| 00:13.10 | *** join/#asterisk Researcher (~user@unaffiliated/unafilliate) |
| 00:17.30 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 00:17.40 | TehRabbitt | any idea what would cause this: |
| 00:17.41 | TehRabbitt | [May 2 20:16:52] WARNING[5156]: channel.c:2589 ast_prod: Prodding channel 'Skinny/500@CISCO-1' failed |
| 00:19.46 | Jumpie | hmm..i changed some settings so it shouldnt look like its sending from another domain |
| 00:20.02 | Jumpie | <PROTECTED> |
| 00:20.08 | Jumpie | but...now i think im gettin what you said...that delay |
| 00:20.52 | p3nguin | Like I said, greylisting isn't new. |
| 00:21.21 | Jumpie | the greylisting is the delay you talking about? |
| 00:21.29 | Jumpie | i know of white/black listing greylisting i dont really hear that term? |
| 00:21.46 | traderz | p3nguin, so i fixed it and seems tons of others have had the same issue and nobody posted that they ever fixed it but i did it.. |
| 00:22.03 | traderz | this is why i wanted to see your files |
| 00:22.35 | traderz | versios 6.x and backwards use the P0S in the OS9* file |
| 00:23.23 | traderz | version 7.x use the P003 in the OS* file but directions also date you need to use the image_version in the SIP files |
| 00:23.33 | traderz | and if you use the same P003 filename it wont ever work.. |
| 00:23.50 | traderz | so you must use P0S in the SIP files and P003 in the OS9* |
| 00:24.16 | p3nguin | image_version:P0S3-08-11-00 ;SIP image |
| 00:24.25 | traderz | ya exactly |
| 00:24.27 | p3nguin | P003-08-11-00 |
| 00:24.37 | traderz | if you would hav showed me that i would have been done days ago lol |
| 00:24.38 | p3nguin | I read the documentation. |
| 00:24.42 | traderz | instead i had to figure it out |
| 00:24.52 | traderz | ehehhe i did too and didnt see that |
| 00:25.02 | p3nguin | No one held my hand when I needed to configure it. |
| 00:25.35 | traderz | anyways - hopefully my time spent will help someone else |
| 00:26.51 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 00:28.15 | TehRabbitt | p3nguin: I still can't figure out how to get the phone to route calls from the SCCP side to the SIP side (outgoing trunk) |
| 00:28.37 | p3nguin | Dial() |
| 00:28.43 | p3nguin | That's all there is to it. |
| 00:28.57 | TehRabbitt | Is that in extensions.conf or sip.conf? |
| 00:29.03 | p3nguin | extensions |
| 00:29.14 | p3nguin | Create an extension, make it Dial(SIP/yourpeer/somenumber). |
| 00:30.10 | traderz | anyone recommend any good wifi sip phones? |
| 00:30.10 | *** join/#asterisk Brookss (~fedora@174.3.119.13) |
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| 00:37.36 | *** join/#asterisk devdvd (~myemail@173-31-160-214.client.mchsi.com) |
| 00:38.56 | *** join/#asterisk davidstrauss (~davidstra@wikimedia/davidstrauss) |
| 00:39.17 | devdvd | hey all, with asterisk 1.4.30 is there anyway to announce a call to an agent when using ackcall? |
| 00:44.11 | *** join/#asterisk jasonjjohnsonjr (~jjohnson@adsl-93-32-130.owb.bellsouth.net) |
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| 00:46.36 | davidstrauss | I have an issue with the voicemail service on * 1.6 on Ubuntu 10.04 |
| 00:46.45 | davidstrauss | It just says "goodbye" when I call it. |
| 00:46.54 | davidstrauss | The * shell doesn't tell me anything useful. |
| 00:47.38 | *** join/#asterisk diegomad (mad@186.81.138.100) |
| 00:53.58 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
| 00:54.34 | ChannelZ | When you call it, as in VoiceMailMain? |
| 00:55.20 | davidstrauss | ChannelZ: yes |
| 00:55.33 | davidstrauss | ChannelZ: In the shell, it says it's playing vm-login and then goodbye |
| 00:55.41 | davidstrauss | ChannelZ: But I only hear it play goodbye |
| 00:56.14 | davidstrauss | ChannelZ: I'm worried the build it broken for Ubuntu 10.04, so I'm building out a CentOS 5.4 machine now. |
| 00:56.15 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:56.21 | ChannelZ | it works fine here |
| 00:56.47 | davidstrauss | ChannelZ: The built-in packages for U 10.04? |
| 00:56.55 | ChannelZ | no not using their packages |
| 00:56.58 | davidstrauss | ChannelZ: Ah |
| 00:57.19 | davidstrauss | ChannelZ: I am using the Ubuntu packages, which worked fine for me for Asterisk 1.4 |
| 00:58.47 | devmod | any good docs about building a queue ? |
| 00:58.51 | davidstrauss | ChannelZ: did you compile it yourself? |
| 00:59.19 | ChannelZ | yeah |
| 01:02.57 | *** join/#asterisk t0rrieri (~Torrieri@nelug/crew/torrieri) |
| 01:03.14 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
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| 01:09.02 | TehRabbitt | Ok, I've looked at dozens of dialplans, tried getting extentions to work internally, but the best i've been able to do was get one phone to be able to call the other, but there was no audio and I couldnt' call back the other way... on top of that I still can't get an outgoing trunk to work and I feel like at the rate i'm going i'll never figure it out... the SIP phone is a desktop softphone (x-lite) and is assigned to extention 100 |
| 01:09.50 | devdvd | TehRabbitt have something that looks like |
| 01:09.58 | devdvd | [default] |
| 01:10.00 | devdvd | errr |
| 01:10.01 | devdvd | hold on |
| 01:10.07 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 01:10.10 | devdvd | actually do a pastebin of your dialplan |
| 01:10.46 | TehRabbitt | there is none now, I deleted the entire .conf file since none of it worked... basically i'm starting off with [default] again lol |
| 01:10.56 | devdvd | ok well do what you had |
| 01:11.02 | devdvd | then we can help you fix it |
| 01:11.04 | TehRabbitt | ok |
| 01:11.05 | joobie | guys need to get the lastest firmware + config for a polycom 320 and polycom 321 |
| 01:11.13 | devdvd | We'll nominate ChannelZ to help you :P |
| 01:11.39 | joobie | do we need a support contract with polycom to get this? |
| 01:11.56 | davidstrauss | joobie: no |
| 01:12.25 | joobie | davidstrauss, know where we can download |
| 01:12.40 | davidstrauss | joobie: http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_2_3_release_sig_combined.zip |
| 01:13.36 | TehRabbitt | Ok here is what I had: |
| 01:13.36 | TehRabbitt | http://pastebin.com/X8MqVz7M |
| 01:13.37 | davidstrauss | joobie: As seen from this page: http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip321_331.html |
| 01:13.42 | TehRabbitt | from about 3 hours ago :( |
| 01:13.43 | TehRabbitt | lol |
| 01:14.29 | TehRabbitt | Basically I just want extention 100 to be able to make a call to extention 500, and vice versa... and also allow it to dial out using [VOIP1] which is the SIP provider |
| 01:14.59 | joobie | thanks davi |
| 01:15.25 | devdvd | and when you try to dial from 100 to 500 |
| 01:15.31 | devdvd | what does your asterisk console say? |
| 01:15.33 | davidstrauss | TehRabbitt: What's the output from the asterisk shell? |
| 01:15.56 | TehRabbitt | [May 2 21:16:32] WARNING[5430]: pbx.c:1832 pbx_extension_helper: No application 'SetCalledParty' for extension (users, 500, 1) |
| 01:16.21 | devdvd | 44.exten => 500,1,SetCalledParty("CISCO Wireless" <500>) |
| 01:17.12 | devdvd | looks to me like an issue with your sccp channel driver |
| 01:17.15 | TehRabbitt | where do I put that? |
| 01:17.16 | p3nguin | tehrabbitt: (1753.31) <p3nguin> I also do not have SetCalledParty() as a valid application, so check your system to see if you do or do not have it. |
| 01:17.29 | devdvd | not supporting that application |
| 01:17.36 | TehRabbitt | huh? |
| 01:17.41 | *** join/#asterisk pabelanger_ (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 01:17.49 | p3nguin | It's not a channel driver that doesn't have support for the application... the application does not exist in Asterisk. |
| 01:17.54 | devdvd | TehRabbitt: that is telling you that the function setcalledparty DOES NOT exist |
| 01:18.03 | TehRabbitt | so what do I do? |
| 01:18.08 | p3nguin | take it out. |
| 01:18.12 | TehRabbitt | from where? |
| 01:18.13 | p3nguin | Read the book, first. |
| 01:18.16 | devdvd | why are you using SetCalledParty? |
| 01:18.17 | p3nguin | ~book |
| 01:18.17 | 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 |
| 01:19.21 | TehRabbitt | where is setcalledparty anyway? |
| 01:19.29 | devdvd | ummm |
| 01:19.30 | devdvd | dude |
| 01:19.33 | devdvd | read your dialplan.... |
| 01:19.39 | p3nguin | I finally got around to getting chan_sccp installed. |
| 01:20.00 | TehRabbitt | ohhhh |
| 01:20.02 | p3nguin | devdvd: I'd recommend the book instead of the dialplan. |
| 01:20.08 | TehRabbitt | so what do I use instead of that then? |
| 01:20.12 | p3nguin | NOTHING |
| 01:20.17 | TehRabbitt | just remove the line? |
| 01:20.22 | devdvd | p3nguin: yea i get that but he asked [18:19] <TehRabbitt> where is setcalledparty anyway? |
| 01:20.31 | devdvd | which is in his dialplan |
| 01:20.50 | devdvd | TehRabbitt: i suggest you read the asterisk book before you start ripping dialplans off the internet :) |
| 01:20.51 | TehRabbitt | so just use: exten => 500,1,Dial(SCCP/Wireless) |
| 01:20.51 | TehRabbitt | ? |
| 01:21.00 | devdvd | ~book |
| 01:21.01 | 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 |
| 01:21.18 | devdvd | i hate telling people to RTFM |
| 01:21.35 | devdvd | but at this point it seems like you haven't even taken the time to understand the basics on your own |
| 01:23.55 | Jumpie | anybody know why my dmsg shows as having 4 cpus..and it mentiosn somethin about they are off synch and that its 'correcting' them |
| 01:24.00 | Jumpie | but its a dualcore atom d510 |
| 01:28.42 | TehRabbitt | devdvd: ok I now kinda understand how the dialplan works, and was able to get one phone talking to the other... my question is though: why no audio? |
| 01:28.55 | devdvd | TehRabbitt |
| 01:28.57 | devdvd | check your codecs |
| 01:29.26 | pabelanger_ | TehRabbitt: or are you behind a firewall / NAT |
| 01:29.44 | p3nguin | No audio? Probably an RTP problem. |
| 01:31.09 | TehRabbitt | devdvd: how would I go about that? |
| 01:31.23 | TehRabbitt | p3nguin: How would I be able to determine if it's an RTP problem? |
| 01:31.51 | devdvd | pastebin the configuration file for your phones |
| 01:32.04 | devdvd | probably like sccp.conf or something like that |
| 01:32.18 | devdvd | (unless your using skinny then it would be something like skinny.conf |
| 01:32.52 | TehRabbitt | alright hold on |
| 01:34.00 | TehRabbitt | http://pastebin.com/QFPtMtuA |
| 01:34.00 | joobie | guys is there a way to find out what bootrom version is on a polycom phone, without physical access (have remote access only atm) |
| 01:34.07 | TehRabbitt | it's the skinny.conf that came with asterisk |
| 01:35.13 | devdvd | ok..actually at this point i need to ask |
| 01:35.17 | devdvd | what kind of phones are you using? |
| 01:35.27 | devdvd | are they sip |
| 01:35.34 | devdvd | or cisco sccp/skinny? |
| 01:35.49 | devdvd | i seen cisco and made the wrong assumption |
| 01:35.51 | TehRabbitt | one SIP softphone (desktop) and one sccp/skinny phone |
| 01:36.06 | TehRabbitt | the SCCP/Skinny phone is the one that is having the issues though |
| 01:36.58 | devdvd | well, you can rule out an rtp issue, bring up another sip softphone |
| 01:37.02 | devdvd | connect it to the system and dial between |
| 01:37.15 | devdvd | if you can get audio between them |
| 01:37.18 | devdvd | its probably not rtp |
| 01:37.27 | devdvd | and at that point, possible codec related |
| 01:38.15 | TehRabbitt | for instance, I can dial out / use the softphone to connect to the digium test server using SIP but the SCCP one connects and has no audio |
| 01:38.25 | TehRabbitt | no audio between the phones either |
| 01:38.42 | devdvd | find the configuration file for your sccp phone |
| 01:38.45 | devdvd | and pastebin it |
| 01:39.17 | TehRabbitt | kk |
| 01:40.04 | TehRabbitt | http://pastebin.com/1eCtUEpF |
| 01:40.11 | TehRabbitt | it's an XML file hosted on a TFTP server |
| 01:40.27 | devdvd | no |
| 01:40.28 | devdvd | not that |
| 01:40.38 | devdvd | im talkin about the file on your asterisk box |
| 01:40.44 | TehRabbitt | Skinny.conf? |
| 01:40.49 | devdvd | if thats what your using |
| 01:40.52 | TehRabbitt | http://pastebin.com/QFPtMtuA |
| 01:40.54 | TehRabbitt | thats the one |
| 01:41.00 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 01:42.09 | devdvd | 12.;allow=all ; see doc/rtp-packetization for framing options |
| 01:42.09 | devdvd | 13.;disallow= |
| 01:42.12 | devdvd | that might be your problem |
| 01:42.27 | devdvd | uncomment allow=all |
| 01:42.32 | devdvd | see if that helps |
| 01:42.33 | devdvd | if not |
| 01:42.36 | devdvd | find the doc its talkin about |
| 01:42.37 | devdvd | and read it |
| 01:43.01 | TehRabbitt | i'm trying that... |
| 01:43.14 | TehRabbitt | there's another option there specifying if the server should handle audio or if the phone should |
| 01:43.21 | TehRabbitt | ;earlyrtp=1 ; whether audio signalling should be provided by asterisk |
| 01:43.27 | TehRabbitt | not sure if that could be it |
| 01:43.34 | devdvd | play with the options |
| 01:45.05 | TehRabbitt | nope nothing... with both settings enabled/disabled |
| 01:45.54 | *** join/#asterisk Slugs_ (~yeah@unaffiliated/slugs-/x-6594848) |
| 01:46.33 | TehRabbitt | the phones can dial eachother but there's like no audio at all on the SCCP phone |
| 01:46.48 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 01:47.21 | TehRabbitt | ok this is weird... |
| 01:47.28 | p3nguin | devdvd: skinny still uses RTP, as far as I know. |
| 01:47.41 | devdvd | p3nguin, you'd know better than I |
| 01:48.00 | TehRabbitt | when I make a call from the SIP phone to the SCCP phone, no audio, but if I hit a button it makes a constant Tone of whatever number I hit and i can't stop the tone unless I hang up |
| 01:48.12 | devdvd | ive not touched a cisco phone that required sccp/skinny in ~5 years |
| 01:48.49 | p3nguin | I'm debating on configuring this sccp.conf now that I got chan-sccp-b installed. |
| 01:49.07 | devdvd | p3nguin, you do much with queues? |
| 01:49.18 | TehRabbitt | p3nguin: what is chan-sccp-b? |
| 01:49.29 | p3nguin | If this channel driver is better than the one that comes with asterisk (chan_skinny), then it has to be worth trying again. |
| 01:49.39 | devdvd | aye |
| 01:49.48 | p3nguin | tehrabbitt: 3rd-party native sccp channel driver |
| 01:49.59 | TehRabbitt | hm... would that fix my problem perhaps with the audio? |
| 01:50.03 | devdvd | wonder if you can still get those cisco 20's for dirt cheap |
| 01:50.06 | p3nguin | chan_sccp rather than chan_skinny |
| 01:50.46 | p3nguin | tehrabbitt: No, it's not going to fix your audio. It has way more to configure than skinny.conf, and you probably haven't even mastered that yet. |
| 01:51.05 | TehRabbitt | Heh... true... i'm just trying to figure out why audio won't work 0_o |
| 01:51.08 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:51.24 | *** join/#asterisk Brookss (~SSJGotenk@174.3.119.13) |
| 01:54.54 | pabelanger_ | TehRabbitt: is your asterisk and phone behind firewalls? |
| 01:55.30 | TehRabbitt | the AP that the phone connects to is directly attached to the server through a switch... internal LAN.. I can't make calls even internally |
| 01:56.14 | TehRabbitt | i've even gone the step of putting the AP directly on one of the network interfaces of the server with a crossover cable and still same issue.. it can connect, it can dial another extention, but no audio incoming or outgoing from the SCCP phone |
| 01:56.26 | TehRabbitt | if I hit a button on the phone I just hear a continious DMTF tone |
| 01:59.20 | *** join/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-gjbkfwfmbqxaqyiq) |
| 02:00.01 | TehRabbitt | and I just checked my /etc/hosts file to confirm that 127.0.0.1 is only set to localhost |
| 02:02.20 | devdvd | TehRabbitt: have you tried restarting asterisk after these changes? |
| 02:02.28 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 02:02.34 | TehRabbitt | Yes |
| 02:02.37 | devdvd | k |
| 02:02.52 | TehRabbitt | I've tried restarting asterisk as well as the entire server at least once and neither helped (thought it'd be worth a try) |
| 02:03.02 | TehRabbitt | not the same phone but I found this on google: https://www.trixbox.org/forums/trixbox-forums/help/sccp-no-audio-cisco-7945g |
| 02:04.32 | TehRabbitt | everthing I am reading is refrencing NAT / Ports not being open but This is pretty much a direct connection from the phone to the server |
| 02:04.39 | TehRabbitt | and the server isnt' blocking anything afaik |
| 02:06.12 | TJNII | Why don't you stop spinning your wheels and try with a SIP softphone. Once you get that working you can work on that Skinny phone or whatever it is you've got. |
| 02:06.22 | p3nguin | Have you pasted all your configs, in their entirety, masking ONLY passwords, yet? |
| 02:06.38 | TehRabbitt | Yes I have |
| 02:07.11 | TehRabbitt | TJNII: I have the SIP softphone able to dial out / hear audio to the asterisk test server... the SCCP phone can dial out / dial in but there is no audio |
| 02:07.46 | TehRabbitt | hold on, i'm going to post each config file as-is (minus passwords) |
| 02:08.08 | [TK]D-Fender | TehRabbitt: In and out from where, to where? |
| 02:09.17 | carrar | Audio is over rated anyhow |
| 02:10.48 | *** join/#asterisk gospch (~gospch@p5088EE9D.dip.t-dialin.net) |
| 02:10.58 | TehRabbitt | Skinny.conf: |
| 02:10.59 | TehRabbitt | http://pastebin.com/k58GzgjW |
| 02:11.43 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 02:11.53 | TehRabbitt | extensions.conf: |
| 02:11.53 | TehRabbitt | http://pastebin.com/yMYJCEjf |
| 02:13.48 | p3nguin | Not causing the problem, but the users context has a failed exten in it. |
| 02:14.11 | p3nguin | 500,2, = invalid |
| 02:14.47 | TehRabbitt | SIP.conf |
| 02:14.54 | TehRabbitt | http://pastebin.com/1ctd6wVe |
| 02:15.10 | TehRabbitt | p3nguin: didn't see that, fixed it |
| 02:16.50 | p3nguin | In that one, line 34... they aren't likely to decide to put their server behind NAT all of a sudden, so you'll never need it. (I'd delete it so there is no confusion.) |
| 02:17.29 | p3nguin | and insecure=very may not be valid anymore. |
| 02:18.18 | p3nguin | also not causing the problem |
| 02:22.33 | TehRabbitt | any ideas? |
| 02:22.46 | p3nguin | Everything looks okay to me. |
| 02:26.37 | p3nguin | tehrabbitt: Can you call successfully from sip 100 to sip 200? |
| 02:26.57 | TehRabbitt | SIP200 isnt' installed yet (will be on my laptop) but I will try that now |
| 02:29.30 | [TK]D-Fender | TehRabbitt: just test with voicemail direct with your single phone. |
| 02:29.47 | [TK]D-Fender | TehRabbitt: prove the phone leg is fine. |
| 02:31.09 | devmod | looking for any docs explaining how to setup queues on asterisk, any links ? |
| 02:32.19 | TehRabbitt | [TK]D-Fender: don't have VM set up yet afaik |
| 02:32.35 | TehRabbitt | so far i'm just tryign to get them talking among eachother |
| 02:33.35 | [TK]D-Fender | TehRabbitt: "them"? |
| 02:33.42 | [TK]D-Fender | TehRabbitt: Go set up a VM then. |
| 02:33.52 | TehRabbitt | the desktop softphone and the sccp phone |
| 02:33.53 | [TK]D-Fender | TehRabbitt: its 2 lines |
| 02:34.20 | TehRabbitt | correct, and they can't talk to eachother |
| 02:34.39 | [TK]D-Fender | TehRabbitt: prove each independantly |
| 02:35.53 | *** join/#asterisk gospch (~gospch@p5088EE9D.dip.t-dialin.net) |
| 02:36.30 | devdvd | YAY!!! |
| 02:36.51 | devdvd | figured a work around for the whole pre-announe queue issue :) |
| 02:36.56 | TehRabbitt | Nope they def work |
| 02:37.11 | TehRabbitt | laptop right next to speakers of desktop == Major feedback once call connected |
| 02:37.26 | TehRabbitt | I can take the laptop into the other room, talk into it's mic and hear the desktop and vice versa |
| 02:37.31 | TehRabbitt | crystal clear |
| 02:38.31 | TehRabbitt | so SIP to SIP works |
| 02:38.53 | TehRabbitt | Ext 100 -> 200 and Ext 200 -> 100 works full audio crystal clear |
| 02:40.39 | TehRabbitt | voicemail apparently works as well heh |
| 02:44.29 | TehRabbitt | It's gotta be a codec issue or something how can I be certain? |
| 02:45.41 | ManxPower | set disallow=all and allow=ulaw |
| 02:45.41 | devdvd | TehRabbitt: do this, skinny set debug |
| 02:45.46 | devdvd | then attempt to make a call |
| 02:45.58 | devdvd | then pastebin the output from the console |
| 02:46.25 | TehRabbitt | just type skinny set debug into the CLI? |
| 02:48.07 | *** join/#asterisk coppice (~chatzilla@m121-202-89-14.smartone-vodafone.com) |
| 02:48.08 | TehRabbitt | got it i'm pastbining it now |
| 02:48.22 | TehRabbitt | http://pastebin.com/7EBrhdux |
| 02:50.01 | TehRabbitt | does it help at all? :-\ |
| 02:51.59 | devdvd | TehRabbitt |
| 02:52.09 | devdvd | have you tried putting allow=all |
| 02:52.16 | devdvd | in your phone definition in skinny.conf |
| 02:53.34 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:53.57 | TehRabbitt | Yep tried that |
| 02:54.06 | TehRabbitt | actually *just* tried that... |
| 02:56.02 | TehRabbitt | SIP Host: thoth.tenehawk.com |
| 02:56.08 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 02:56.15 | TehRabbitt | srry wrong post |
| 02:56.24 | TehRabbitt | i was pasting into notepad |
| 02:56.35 | devdvd | try it like that and repaste your debug output |
| 02:56.55 | TehRabbitt | like what? |
| 02:57.04 | TehRabbitt | the allow=all? |
| 02:57.09 | devdvd | try setting allow=all |
| 02:57.12 | devdvd | restarting asterisk |
| 02:57.21 | devdvd | then do skinny set debug |
| 02:57.30 | devdvd | attempt to make a call |
| 02:57.51 | devdvd | and show us the debug output |
| 02:58.02 | Jumpie | any idea why not all of my vm_email.inc isnt being displayed in the email? |
| 02:58.09 | Jumpie | http://jumpie.pastebin.com/4nN1cafn |
| 02:58.11 | Jumpie | is what i have |
| 02:58.18 | Jumpie | it works fine, but anything after the length isnt displayed |
| 02:58.36 | Jumpie | basically everything after the first line..is it some word wrap thing messing it up? |
| 02:58.38 | TehRabbitt | http://pastebin.com/tVNC0W7a |
| 02:58.48 | TehRabbitt | that's the debug output |
| 03:00.30 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 03:00.34 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:02.45 | Jumpie | shit! that was it |
| 03:02.53 | Jumpie | that file doesnt like hard carriage returns..... |
| 03:04.42 | TehRabbitt | anything? |
| 03:16.22 | TehRabbitt | anyone here? |
| 03:16.55 | Jumpie | dunno enougha bout that man |
| 03:17.28 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 03:20.23 | TehRabbitt | oh :( |
| 03:27.07 | *** join/#asterisk gospch (~gospch@p5088D770.dip.t-dialin.net) |
| 03:29.23 | *** join/#asterisk TehRabbitt (~rabbott@c-71-59-82-2.hsd1.nj.comcast.net) |
| 03:30.43 | TehRabbitt | any ideas on how to get this cisco phone able to actually work with audio? |
| 03:34.17 | [TK]D-Fender | TehRabbitt: You have not proven the phone direct to * |
| 03:35.42 | *** join/#asterisk gospch (~gospch@p5088FA92.dip.t-dialin.net) |
| 03:38.34 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:40.33 | TehRabbitt | [TK]D-Fender: How have I not proven it? Extension 100 Can call Extention 200 and hear voice 2 ways |
| 03:40.41 | TehRabbitt | so SIP works, SCCP however not working |
| 03:41.10 | TehRabbitt | thoth*CLI> skinny show devices |
| 03:41.10 | TehRabbitt | Name DeviceId IP Type R NL |
| 03:41.10 | TehRabbitt | -------------------- ---------------- --------------- --------------- - -- |
| 03:41.10 | TehRabbitt | CISCO SEP001AA192AB6D 192.168.1.200 7921 Y 2 |
| 03:41.10 | TehRabbitt | CISCO SEP001AA192AB6D 7921 N 2 |
| 03:41.56 | [TK]D-Fender | TehRabbitt: .... |
| 03:42.11 | [TK]D-Fender | TehRabbitt: Its just not sinking in. |
| 03:44.05 | TehRabbitt | how do you want me to prove it? |
| 03:44.09 | *** join/#asterisk gospch (~gospch@p5088D3AA.dip.t-dialin.net) |
| 03:44.55 | TehRabbitt | when I make a call, I dont even hear ringing on the Cisco handheld... |
| 03:45.14 | ManxPower | TehRabbitt, I think he means protocol debug |
| 03:45.29 | TehRabbitt | i've posted that a few times, i'll post it again |
| 03:45.53 | TehRabbitt | http://pastebin.com/iFpW0mc9 |
| 03:45.57 | [TK]D-Fender | TehRabbitt: I'm going to try this one last time. |
| 03:46.01 | TehRabbitt | that is the debug for SCCP |
| 03:46.10 | [TK]D-Fender | TehRabbitt: Try to follow with me. |
| 03:46.17 | TehRabbitt | ok |
| 03:46.31 | [TK]D-Fender | TehRabbitt: PLACE A FUCKING CALL DIRECTO TO VOICEMAIL AND GO RETREIVE IT AFTERWARDS. |
| 03:46.44 | TehRabbitt | from the Cisco phone? |
| 03:46.47 | [TK]D-Fender | TehRabbitt: YES |
| 03:47.06 | [TK]D-Fender | TehRabbitt: What part of "test these fucking phones independently" was not clear? |
| 03:47.45 | [TK]D-Fender | TehRabbitt: Remove variables to confirm what does, and does not work on the smallest scale. Do NOT involve added devices. |
| 03:47.57 | [TK]D-Fender | TehRabbitt: if it works direct then there is a problem in the conversion. |
| 03:48.05 | [TK]D-Fender | TehRabbitt: But get off your ass and PROVE IT |
| 03:48.59 | TehRabbitt | Ok, Can record a VM on both SIP softphones, When I dial the VM # from the Cisco phone I dont hear *anything* at all, just says "connected" |
| 03:50.08 | devdvd | LOL@TK |
| 03:50.37 | ChannelZ | <PROTECTED> |
| 03:50.43 | ChannelZ | wandered away several hours ago.. |
| 03:50.47 | devdvd | yea likewise |
| 03:50.51 | TehRabbitt | Everything I see online says to use "SCCP" instead of "Skinny" should I even bother or will I have the same issue? |
| 03:50.56 | *** join/#asterisk jtodd (jyc1l5tlkv@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 03:50.56 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 03:51.37 | TehRabbitt | ChannelZ: yes it's still going on, so far I can get 3 softphones and a PAP2 working via SIP but Skinny still won't work |
| 03:51.38 | p3nguin | chan_skinny DOES work. I've used it. |
| 03:51.41 | p3nguin | It doesn't work for all features, but it does make and receive calls. |
| 03:52.25 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 03:52.26 | [TK]D-Fender | TehRabbitt: Go check your firewall on the server |
| 03:52.31 | p3nguin | I'm just about to dive in on configuring sccp.conf to test out chan_sccp. I'll report failures and success in a few hours if all goes well. |
| 03:52.39 | TehRabbitt | where would that be because I've never installed a firewall on that server |
| 03:52.53 | p3nguin | iptables -L -nv |
| 03:53.00 | p3nguin | See if anything is there. |
| 03:53.01 | [TK]D-Fender | TehRabbitt: Really... sure it wasn't part o fhte OS install? |
| 03:53.26 | p3nguin | specifically, iptables -L INPUT -nv |
| 03:53.42 | TehRabbitt | "0 packets 0 bytes" is all it shows |
| 03:54.01 | p3nguin | Also, did you verify that chan_skinny is loaded/working? |
| 03:54.18 | p3nguin | It has to show more than just that. |
| 03:54.30 | p3nguin | It's a verbose app, so there should be quite a bit more. |
| 03:54.41 | TehRabbitt | anything specific I should try? |
| 03:54.52 | TehRabbitt | It wont' let me unload chan_skinny says it is in use |
| 03:54.56 | TehRabbitt | i've tried restarting asterisk |
| 03:54.58 | TehRabbitt | still nothing |
| 03:55.25 | [TK]D-Fender | TehRabbitt: "iptables --list" <- PASTEBIN |
| 03:56.36 | TehRabbitt | http://pastebin.com/bwgFG0VF |
| 03:57.39 | [TK]D-Fender | TehRabbitt: ok, that isn't it then. Good. Teh phone is on the same local subet as *, right? |
| 03:59.15 | TehRabbitt | Yes. it is... The AP is connected to the same switch as teh server, same local subnet |
| 03:59.26 | TehRabbitt | phone == 192.168.1.200 |
| 03:59.33 | TehRabbitt | * = 192.168.1.70 |
| 04:00.53 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 04:01.17 | TJNII | Subnet mask of 255.255.255.0 on both the server and the phone? |
| 04:02.15 | TehRabbitt | Yes |
| 04:03.44 | p3nguin | This sccp.conf is a LOT different from skinny.conf. |
| 04:05.37 | TehRabbitt | p3nguin: how's it different? |
| 04:05.54 | p3nguin | Lots more settings/options available. |
| 04:06.18 | TehRabbitt | ah |
| 04:06.41 | p3nguin | It'll take me a while to wrap my head around the new config before I can put it in use. |
| 04:09.26 | TehRabbitt | here is the latest debug output |
| 04:09.27 | TehRabbitt | http://pastebin.com/tneGNrL2 |
| 04:12.47 | TehRabbitt | Ok... I just rebooted the phone while keeping debug skinny on.... this is the output during boot up: |
| 04:12.47 | TehRabbitt | http://pastebin.com/vaHEkQqZ |
| 04:12.51 | TehRabbitt | it's quite a bit |
| 04:12.53 | TehRabbitt | discusses codecs |
| 04:15.28 | TehRabbitt | anyone? |
| 04:19.14 | [TK]D-Fender | TehRabbitt: Should have "disallow=all", followed by "allow=ulaw" |
| 04:19.22 | [TK]D-Fender | TehRabbitt: Apply, reload, retest |
| 04:20.37 | TehRabbitt | nothing... |
| 04:21.06 | [TK]D-Fender | TehRabbitt: show us |
| 04:22.15 | TehRabbitt | Here is the output of debug again |
| 04:22.15 | TehRabbitt | http://pastebin.com/2Fi1v2ch |
| 04:23.04 | [TK]D-Fender | TehRabbitt: CONFIGS |
| 04:23.13 | TehRabbitt | kk hold on |
| 04:24.05 | TehRabbitt | http://skinny.conf.pastebin.com/FJfBTaJU |
| 04:24.49 | *** join/#asterisk mun27 (~chatzilla@mail.soti.net) |
| 04:24.57 | mun27 | hi |
| 04:25.26 | mun27 | I am unable to register with my asterisk server |
| 04:25.53 | mun27 | when I use xlite I am able to register |
| 04:26.06 | mun27 | but using othe sip soft phone I am not |
| 04:26.16 | [TK]D-Fender | mun27: pastebin the SIP DEBUG of your attempt from * CLI |
| 04:26.18 | [TK]D-Fender | ~pb |
| 04:26.19 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 04:26.47 | mun27 | wait |
| 04:26.55 | TJNII | I thought he was debugging skinny |
| 04:27.06 | TJNII | I thought sip worked.... |
| 04:27.33 | TJNII | Wait, you were talking to someone else. |
| 04:27.34 | ChannelZ | who? different person |
| 04:27.36 | TJNII | Carry on. |
| 04:27.42 | TehRabbitt | lol different person lol |
| 04:28.18 | TJNII | catches his mistakes, but sometimes it doesn't happen fast enough. |
| 04:28.26 | ChannelZ | PLZ HALP my toast does burn toast when I turn it on dark but not light |
| 04:28.42 | TehRabbitt | lol |
| 04:28.44 | TJNII | have you tried running it under the faucet? |
| 04:28.47 | TJNII | To cool the toast? |
| 04:28.49 | devdvd | yeah toast! :) |
| 04:28.53 | TJNII | Whoo toast! |
| 04:29.02 | ChannelZ | yes but then spark come out and house burn, what to do? |
| 04:29.54 | TehRabbitt | [TK]D-Fender: any ideas about the Skinny.conf? |
| 04:29.54 | TJNII | I don't belive that. Pastebin a log of your house burning. |
| 04:30.24 | TehRabbitt | TJNTI: http://pastebin.com/ehWxD116 |
| 04:30.26 | TehRabbitt | lmfao |
| 04:30.57 | TJNII | I don't see a 911 transcription! How am I supposed to debug that! |
| 04:31.08 | TehRabbitt | lmao |
| 04:33.04 | TehRabbitt | Here is my house it's in fire see: http://pastebin.com/P7qkHgis |
| 04:33.05 | TehRabbitt | lmao |
| 04:33.33 | devmod | when using a queue, can I ring the agent phones instead of directly connect the customer to them ? |
| 04:34.02 | mun27 | [TK"D-Fender: http://pastebin.com/c1HU4A8U |
| 04:34.27 | mun27 | [TK]D-Fender: http://pastebin.com/c1HU4A8U |
| 04:36.12 | TehRabbitt | do you think SCCP will work or is it a lost cause at this point? |
| 04:36.43 | devdvd | hey, does asterisk have the ability to dial all agents in a queue n times then drop out to the next priority (ex. Dial 1>2>3>4>1>2>3>4>drop out of queue into next priority) |
| 04:37.15 | devdvd | so in that case it would dial through the queue twice |
| 04:37.16 | [TK]D-Fender | mun27: Ok, that is a local device registering to your *. It is getting challenged, and your device is not responding with auth |
| 04:37.17 | devdvd | then drop out |
| 04:37.38 | TehRabbitt | [TK]D-Fender: any other ideas I should try with skinny? |
| 04:38.16 | [TK]D-Fender | TehRabbitt: I never saw your updated configs |
| 04:38.41 | TehRabbitt | TehRabbitt> http://skinny.conf.pastebin.com/FJfBTaJU |
| 04:38.46 | TehRabbitt | posted them a while ago |
| 04:39.03 | [TK]D-Fender | TehRabbitt: Line 61. <- You Allow=all again |
| 04:39.16 | [TK]D-Fender | TehRabbitt: the peer is more specific than [general]. Now try to do the job RIGHT |
| 04:40.05 | TehRabbitt | Ah. didn't see that one :( |
| 04:40.08 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91) |
| 04:40.19 | mun27 | [TK]-Fender: I have the source code for this softphone client, this registers successfully with the company asterisk server who developed this. But I want it to register with my own *. |
| 04:42.40 | mun27 | [TK]D-Fender: Need some hint so that I can make changes in the source code |
| 04:42.42 | TehRabbitt | [TK]D-Fender: here is the updated skinny.conf still doesn't work with "allow=ulaw" |
| 04:42.43 | TehRabbitt | http://pastebin.com/wNAtZb1G |
| 04:42.54 | p3nguin | Now... do I really want to reflash my phone from SIP to SCCP... that is the question. |
| 04:43.12 | TJNII | Don't be a wuss. Do it! |
| 04:43.14 | devdvd | p3nguin: only if you really hate your phone |
| 04:43.17 | devdvd | and yourself |
| 04:43.30 | devdvd | but honestly, probably be less painful to just go slit your wrists |
| 04:43.42 | p3nguin | I'm doing it for the greater good of mankind. |
| 04:43.49 | p3nguin | I have to test chan_sccp. |
| 04:43.52 | devdvd | ah |
| 04:44.00 | TehRabbitt | p3nguin: if it works well... let me know lol |
| 04:44.09 | p3nguin | chan_skinny sucked, so I went SIP. |
| 04:44.32 | *** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 04:44.42 | TehRabbitt | [TK]D-Fender: here is the updated skinny.conf still doesn't work with "allow=ulaw" http://pastebin.com/wNAtZb1G |
| 04:44.43 | p3nguin | chan_sip is pretty good, but I think I could have more active features on the ole Cisco with SCCP. |
| 04:45.04 | devdvd | what kinda cisco phone you using? |
| 04:45.10 | iluminator101 | i am having trouble provisioning a phone |
| 04:45.53 | iluminator101 | linksys spa941 with elastix and skype |
| 04:46.25 | TehRabbitt | [TK]D-Fender: did you see the pastebin? |
| 04:46.58 | TehRabbitt | these are the codecs my phone supports: Voice CodecsG.729a, G.729ab, G.711u, G.711a |
| 04:48.15 | p3nguin | Here goes... |
| 04:49.44 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 04:49.59 | frk2 | whats happening guys? |
| 04:50.15 | frk2 | is there a way to connect a 4-wire E&M line to asterisk? |
| 04:50.33 | frk2 | got some army guys with REALLY obsolete hardware here |
| 04:50.37 | frk2 | am wondering if its possible |
| 04:52.12 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 04:52.12 | TehRabbitt | [TK]D-Fender: you still here? |
| 04:52.27 | devdvd | might be possible over a T1 interface frk2 but im not sure. |
| 04:52.49 | frk2 | devdvd, but im not getting the E&M over a T1 |
| 04:53.03 | frk2 | its just plain old 4 wire E&M hooked up to a old VHS Radio |
| 04:53.05 | frk2 | VHF |
| 04:53.59 | devdvd | i meant interfacing the E&M with asterisk using a t1 card (like i said, thats just speculation from what i googled) |
| 04:54.36 | devdvd | but i dont know |
| 04:54.39 | devdvd | sorry :( |
| 04:54.43 | *** join/#asterisk Tim_Toady (~moi@77.49.61.52.dsl.dyn.forthnet.gr) |
| 04:54.59 | TehRabbitt | can anyone give me any more suggestions on this skinny.conf issue? |
| 04:55.35 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 04:55.44 | frk2 | any ideas anyone? or should I be using a E&M channel bank? |
| 04:56.53 | *** join/#asterisk Brookss (~SSJGotenk@174.3.119.13) |
| 04:58.10 | TehRabbitt | [TK]D-Fender: you still here??? |
| 04:59.13 | p3nguin | Not having a lot of luck with chan_sccp. |
| 04:59.32 | TehRabbitt | p3nguin: what luck are you having heh |
| 04:59.36 | *** part/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101) |
| 05:00.31 | p3nguin | I've configured sccp.conf to what seems like a reasonable configuration. |
| 05:00.51 | TehRabbitt | and? |
| 05:01.23 | *** join/#asterisk Keal (~chiamuff@unaffiliated/jargon) |
| 05:01.37 | p3nguin | The phone took the SCCP image and registered. |
| 05:01.45 | Keal | Where the Hell is voipmonk? |
| 05:02.01 | p3nguin | But it doesn't make calls. A call to 500 (the echo test) goes to busy. |
| 05:02.22 | Keal | I need some code to blatantly rip off for some major corporation that detects dial tones and identifies them in response. |
| 05:02.27 | Keal | ..in-band. |
| 05:02.51 | TehRabbitt | p3nguin: at least you can hear a busy tone :-p |
| 05:03.15 | p3nguin | chan_skinny never gave me any troubles with dialing phone numbers. |
| 05:03.21 | Keal | I will donate 3500 USD into your paypal after sufficient assistance. |
| 05:03.53 | TehRabbitt | roffle |
| 05:07.09 | Keal | Is anyone going to assist me or do I need to start ordering hits on users. |
| 05:09.34 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-jsxhzvtdeegtfohu) |
| 05:11.21 | p3nguin | Oh, problem solved. default context in sccp.conf was "sccp" and I need my phone in a different context than that. |
| 05:14.35 | *** join/#asterisk gospch (~gospch@p5088B60D.dip.t-dialin.net) |
| 05:14.57 | p3nguin | So far, so good. |
| 05:16.54 | *** join/#asterisk jtodd (hvy7gvrxxe@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 05:16.54 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 05:20.24 | *** join/#asterisk gospch (~gospch@p5088E894.dip.t-dialin.net) |
| 05:24.10 | ChannelZ | god iTunes is a slow piece of crap |
| 05:24.49 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 05:30.45 | TehRabbitt | p3nguin: I joined you over with CHAN-SCCP-B and I have audio but I can't dial out to other extentions... i'm guessing it handles dial() differently? |
| 05:31.32 | ChannelZ | Alright my brain is not working... what is wrong with GotoIf($[${DB(cfwd/work)}=1]?callwork,1) |
| 05:34.01 | p3nguin | chan_sccp doesn't do Dial() at all. |
| 05:34.07 | TehRabbitt | just realized that 0_o |
| 05:34.27 | TehRabbitt | p3nguin: ok so heres what it's doing now... I can receive calls through SCCP but I can't make calls from the phone to other phones |
| 05:34.41 | p3nguin | Where's the debug output? |
| 05:34.46 | TehRabbitt | be right back with it |
| 05:35.53 | TehRabbitt | p3nguin: http://pastebin.com/UWWXvfZs |
| 05:35.55 | TehRabbitt | there ya go |
| 05:36.14 | TehRabbitt | when I call the SIP phones I get a busy signal... when I call the SCCP handheld it goes through |
| 05:36.52 | p3nguin | You enabled realtime. |
| 05:36.59 | p3nguin | I can only assume this was in error. |
| 05:37.08 | TehRabbitt | whoops where? 0_o |
| 05:37.17 | *** join/#asterisk SunnyDP (~scan@bas1-montreal27-1279505166.dsl.bell.ca) |
| 05:37.17 | TehRabbitt | and how do i disable heh |
| 05:37.19 | p3nguin | During the "make" for chan-sccp-b. |
| 05:37.24 | TehRabbitt | oh :( |
| 05:37.28 | TehRabbitt | how do i fix it? |
| 05:37.33 | TehRabbitt | just remake? |
| 05:37.38 | p3nguin | yes |
| 05:38.25 | TehRabbitt | grr how do i remake? i feel stupid |
| 05:38.32 | p3nguin | I'm probably going to go back to SIP before the night is done. This is still too new for me to troubleshoot. |
| 05:38.39 | TehRabbitt | lol |
| 05:38.41 | p3nguin | make clean && make |
| 05:39.00 | TehRabbitt | do I use direct RTP? |
| 05:39.09 | p3nguin | I wouldn't. |
| 05:39.15 | TehRabbitt | oh 0_o |
| 05:40.46 | TehRabbitt | it's showing "200 unknown number" |
| 05:41.55 | TehRabbitt | p3nguin: any ideas? |
| 05:42.37 | p3nguin | I don't know what "it" is. |
| 05:43.00 | TehRabbitt | the phone is showing 200 unknown number then giving busy tones |
| 05:43.16 | TehRabbitt | but it accepts calls with full 2 way audio when it receives a call |
| 05:43.19 | TehRabbitt | just can't place one :( |
| 05:43.37 | TehRabbitt | I have a feeling it's something i've overlooked but I can't find it in the sccp.conf file |
| 05:43.37 | p3nguin | Did you already recompile chan_sccp? |
| 05:43.39 | TehRabbitt | Yes |
| 05:43.45 | p3nguin | and reinstalled it? |
| 05:43.56 | p3nguin | Show me the new debug. |
| 05:44.55 | TehRabbitt | http://pastebin.com/sHfU8gUx |
| 05:45.06 | TehRabbitt | wait reinstalled it how? |
| 05:45.18 | TehRabbitt | (where do i copy the module over to) |
| 05:47.57 | p3nguin | How did you do it the first time? |
| 05:48.55 | TehRabbitt | i made it the right way this time and it's not using realtime but is still giving me the same issue |
| 05:48.55 | TehRabbitt | http://pastebin.com/8wbQCW61 |
| 05:49.26 | p3nguin | Show me your sccp context in extensions.conf. |
| 05:50.25 | TehRabbitt | [wireless] |
| 05:50.26 | TehRabbitt | exten => 500,1,Dial(SCCP/500,120) |
| 05:50.48 | p3nguin | That's your "wireless" context. I clearly asked for your "sccp" context. |
| 05:51.06 | TehRabbitt | ah... dont have one in there 0_o |
| 05:51.13 | p3nguin | Better fix that. |
| 05:51.38 | p3nguin | Either create one with extens in it, or change your phone's context. |
| 05:52.15 | TehRabbitt | changing phone's context would be in sccp.conf right? |
| 05:52.24 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 05:52.54 | p3nguin | yes |
| 05:54.24 | voxter | why is my mac playing the fucking 'alert sound' 4 times in a row every 60 seconds |
| 05:54.28 | TehRabbitt | OWWWW it worked but OUCH loud feedback sound |
| 05:54.41 | TehRabbitt | sounded like a poltergeist inside the phone haha |
| 05:54.43 | voxter | make it stop!! |
| 05:54.56 | TehRabbitt | lol |
| 05:55.23 | TehRabbitt | haha nvm speakerphone was on :-p |
| 05:56.28 | TehRabbitt | sweet transfers work too :-D |
| 05:56.38 | TehRabbitt | just transfered SIP --> SCCP --> SIP |
| 05:57.08 | TehRabbitt | thanks to everyone, and thank you p3nguin :-D |
| 05:57.20 | p3nguin | It's working now? |
| 05:57.27 | TehRabbitt | Yep 2-way audio with SCCP :-D |
| 05:57.33 | TehRabbitt | crystal clear too :-D |
| 05:57.46 | p3nguin | So chan_skinny was where the problem lied. |
| 05:57.51 | TehRabbitt | yep |
| 05:57.54 | p3nguin | interesting |
| 05:58.04 | TehRabbitt | something in chan_skinny is broken |
| 05:58.04 | TehRabbitt | lol |
| 05:58.12 | TehRabbitt | 6 hours later 0_o |
| 05:58.15 | p3nguin | Could have been a configuration error. |
| 05:58.18 | TehRabbitt | true |
| 06:02.20 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:04.02 | TehRabbitt | now just to set up the SIP trunk haha then it'll all work :-D |
| 06:04.07 | TehRabbitt | at least all the internal extentions work heh |
| 06:04.19 | TehRabbitt | ... 10 hours later jk |
| 06:04.54 | p3nguin | I think I found a bug in chan_sccp. |
| 06:05.42 | p3nguin | With the mwilamp set to flash, wink, or blink, the light never comes on, even though sccp show device mydevice says MWI is lit. |
| 06:05.52 | p3nguin | If I set it to on, it actually comes on. |
| 06:06.13 | TehRabbitt | hm |
| 06:06.45 | p3nguin | On solid is better than not working, though. |
| 06:07.11 | TehRabbitt | lol... curious, can I make an exten 411 redirect calls to 1800GOOG411? |
| 06:07.17 | p3nguin | yes |
| 06:07.21 | p3nguin | That's how I do it. |
| 06:07.27 | TehRabbitt | lol |
| 06:08.01 | TehRabbitt | exten => 411,1,Dial(SIP/VOIP1/18004664411}) ? |
| 06:08.24 | p3nguin | lose that } in there, and it should work. |
| 06:08.41 | TehRabbitt | yea that was a typo of the pinky trying to hit backspace lol |
| 06:08.49 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 06:08.59 | p3nguin | presses DND |
| 06:09.06 | DND | hi guys i need help. my phones and softphones are acting lie walkie talkies |
| 06:09.43 | ChannelZ | cool |
| 06:09.51 | TehRabbitt | (whatever DND has enabled that sounds like PTT... figure out how it's working and implement it as a softkey :-D) |
| 06:09.54 | ChannelZ | breaker breaker good buddy |
| 06:10.00 | TehRabbitt | lol |
| 06:10.15 | DND | im calling a mobile phone. then whenever i speak, the other party just silents. then whenever i stop talking, i can hear the other party again |
| 06:10.18 | p3nguin | tehrabbitt: If you don't have 911 service, you might consider exten => 911,1,Playback(no-911-2) |
| 06:10.30 | DND | its like whenever im transmitting voice, the receiving stops |
| 06:10.53 | devmod | if i want to externally execute cmds on asterisk, how would I go about it? (ie an app that will add an agent to a queue) |
| 06:10.58 | TehRabbitt | 411 isnt working :( |
| 06:11.20 | p3nguin | Where's the debug? |
| 06:11.42 | DND | TehRabbitt, you think echo cancel problem? |
| 06:11.52 | DND | no i dont have PTT enabled if there's a module like that |
| 06:13.14 | TehRabbitt | http://pastebin.com/Jh1mETzC |
| 06:13.20 | ChannelZ | aggressive echo cancellation can cause a half duplexy effect |
| 06:13.23 | ChannelZ | Like bad speakerphones |
| 06:13.39 | TehRabbitt | ChannelZ: thats what was causing my banshee noise that was goin on lol |
| 06:14.10 | p3nguin | tehrabbitt: That's the entire thing? |
| 06:14.17 | TehRabbitt | yep :( |
| 06:14.23 | TehRabbitt | that's the entire thing |
| 06:14.23 | DND | ChannelZ, i have a hardware echo cancel from digium |
| 06:14.37 | p3nguin | Try core set verbose 10 and make the call again. |
| 06:15.01 | p3nguin | That sip debug wasn't even a full invite. |
| 06:15.28 | TehRabbitt | http://pastebin.com/F3NXC5EZ |
| 06:15.43 | TehRabbitt | it's immediatally going into "411 not found" on the softphone's display |
| 06:16.00 | TehRabbitt | "address incomplete" |
| 06:17.47 | DND | p3nguin, nothing wrong actually. |
| 06:18.03 | p3nguin | tehrabbitt: You've got no valid dtmfmode. |
| 06:18.03 | DND | to rest, i called an IVR |
| 06:18.10 | DND | *to test |
| 06:18.13 | TehRabbitt | p3nguin: where do i specify that |
| 06:18.21 | DND | then tried talking and blowing off to the mic |
| 06:18.37 | p3nguin | tehrabbitt: in the peer definition |
| 06:18.43 | DND | the IVR just stops talking as if its was blocking something |
| 06:18.54 | p3nguin | tehrabbitt: But there should be a default one that would be inherited. |
| 06:18.59 | TehRabbitt | :-\ |
| 06:19.02 | DND | then after im done talking and blowing on the mic, the IVR continues |
| 06:19.13 | TehRabbitt | the thing is, i can't dial 411 from any of the phones |
| 06:19.22 | TehRabbitt | it's like it's not even seeing that I defined it as an extention |
| 06:19.51 | p3nguin | Show me the output from "dialplan show 411@users" |
| 06:21.57 | TehRabbitt | thoth*CLI> dialplan show 411@users |
| 06:21.58 | TehRabbitt | qThere is no existence of 411@users extension |
| 06:21.58 | TehRabbitt | nvm forgot to reload |
| 06:22.37 | p3nguin | nothing a quick "dialplan reload" can't cure, huh? |
| 06:22.41 | TehRabbitt | p3nguin: this might be more helpful... http://pastebin.com/k9hwN8B1 |
| 06:22.55 | TehRabbitt | no now it's just saying no route :-( |
| 06:23.14 | TehRabbitt | and on the SIP side it's saying "Service Unavailable" on the softphone |
| 06:23.42 | p3nguin | Show me the peer definition for VOIP1, masking ONLY the passwords. |
| 06:24.33 | TehRabbitt | usernames too or just secrets? |
| 06:24.49 | *** join/#asterisk smooth_penguin (~smoove@59.96.95.42) |
| 06:25.19 | p3nguin | For now, you may hide your usernames if you want. |
| 06:25.38 | p3nguin | At this point, I don't think that's the issue, so I don't care to see them. |
| 06:26.27 | TehRabbitt | wait i think i might have fixed it... hold on |
| 06:27.15 | TehRabbitt | this is the new error that stands out in the debug: SCCP: Timeout for callid '2'. Going to dial '411' |
| 06:31.47 | TehRabbitt | p3nguin: http://pastebin.com/NefDbxsg |
| 06:31.50 | TehRabbitt | any ideas? |
| 06:32.17 | p3nguin | For one, proxy1.newyork.talk4free.com is not behind NAT, so get rid of that setting. |
| 06:32.25 | TehRabbitt | k |
| 06:32.38 | TehRabbitt | nat is now set to "no" |
| 06:33.09 | p3nguin | You know you left your usernames and passwords in the file, right? |
| 06:33.16 | TehRabbitt | wait what?!? |
| 06:33.18 | TehRabbitt | where :( |
| 06:33.30 | TehRabbitt | dammit |
| 06:33.31 | TehRabbitt | lmao |
| 06:33.49 | p3nguin | You also left the password in a paste earlier. |
| 06:33.55 | TehRabbitt | sigh :( |
| 06:34.05 | TehRabbitt | i'm tired heh |
| 06:34.19 | p3nguin | Anyway, is proxy1.newyork.talk4free.com's port supposed to be 5070 instead of 5060? |
| 06:34.24 | TehRabbitt | Yes |
| 06:34.28 | TehRabbitt | 5070 |
| 06:34.48 | p3nguin | Did they tell you to use the fromdomain setting? |
| 06:34.59 | TehRabbitt | no |
| 06:35.08 | p3nguin | I'd probably get rid of it, then. |
| 06:35.37 | p3nguin | You want them to think you run a MagicJack? |
| 06:35.39 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 06:35.59 | TehRabbitt | ehh that server yes lol |
| 06:36.18 | TehRabbitt | i'm only using this to test the config for right now until I can mess with the Google voice SIP and Gizmo and such |
| 06:37.15 | p3nguin | Save, run sip reload, try the call again, paste the debug. |
| 06:37.19 | TehRabbitt | still fails even with it :( |
| 06:37.45 | p3nguin | I guess I could play with it since I know your creds. :) |
| 06:38.32 | TehRabbitt | gee thanks lmfao |
| 06:38.37 | p3nguin | heh |
| 06:38.40 | TehRabbitt | :( haha |
| 06:40.25 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 06:40.52 | TehRabbitt | http://pastebin.com/VT5h22hS |
| 06:40.54 | *** join/#asterisk Researcher (~user@unaffiliated/unafilliate) |
| 06:40.56 | TehRabbitt | that's the debug |
| 06:41.28 | TehRabbitt | could this be it: "SIP/2.0 502 Bad Gateway" |
| 06:41.29 | TehRabbitt | heh |
| 06:41.49 | p3nguin | Yes, but why is it a bad gateway? |
| 06:41.54 | TehRabbitt | idk :( |
| 06:43.24 | TehRabbitt | heres the latest with a different hostname |
| 06:43.24 | TehRabbitt | http://pastebin.com/bCaWGRFv |
| 06:44.14 | p3nguin | Didn't you tell me that your * box was connected directly to the internet? |
| 06:44.27 | TehRabbitt | its on the DMZ of my router |
| 06:44.30 | p3nguin | And then we determined that it isn't. |
| 06:44.38 | p3nguin | It's actually behind NAT. |
| 06:44.51 | TehRabbitt | so how do i fix it then? |
| 06:44.53 | p3nguin | But you've still never configured the nat stuff in sip.conf. |
| 06:44.57 | p3nguin | ~sipnat |
| 06:44.58 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:45.11 | p3nguin | This is the third or fourth time I've referenced this for you. |
| 06:46.01 | p3nguin | You've got nat=yes, but you don't have the rest. |
| 06:46.08 | TehRabbitt | ah |
| 06:46.09 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 06:46.13 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 06:46.16 | p3nguin | still. |
| 06:46.19 | p3nguin | after 14 hours. |
| 06:47.03 | *** join/#asterisk noisewaterphd (~noisewate@c-98-202-190-92.hsd1.ut.comcast.net) |
| 06:48.08 | TehRabbitt | Same thing |
| 06:48.14 | TehRabbitt | even with the NAT stuff in there |
| 06:48.22 | p3nguin | Show me the updated sip.conf. |
| 06:48.31 | p3nguin | make sure you run sip reload, too. |
| 06:49.55 | TehRabbitt | http://pastebin.com/kCU6yJDs |
| 06:49.59 | TehRabbitt | i ran sip reload |
| 06:51.13 | p3nguin | I'm not sure if 192.168.1.0/24 is valid for localnet. I always thought it had to be 192.168.1.0/255.255.255.0 |
| 06:52.27 | TehRabbitt | still the same thing :(* |
| 06:52.34 | TehRabbitt | bad gateway |
| 06:55.31 | TehRabbitt | here we go this might help more: http://pastebin.com/LrxJ3Rde |
| 06:57.30 | *** join/#asterisk smooth_penguin (~smoove@59.96.92.51) |
| 06:58.05 | TehRabbitt | p3nguin? |
| 07:00.50 | p3nguin | lost |
| 07:01.06 | TehRabbitt | i cant figure out why it wont work :( |
| 07:01.11 | p3nguin | bad gateway |
| 07:01.46 | TehRabbitt | if its a bad gateway then how come: |
| 07:01.46 | TehRabbitt | [May 3 03:02:17] NOTICE[8381]: chan_sip.c:12718 handle_response_register: Outbound Registration: Expiry for proxy1.newyork.talk4free.com is 120 sec (Schedul |
| 07:01.55 | TehRabbitt | it registers fine |
| 07:02.02 | p3nguin | I guess I'll configure it here. |
| 07:02.20 | TehRabbitt | Heh you saved the cred? lmao |
| 07:05.26 | p3nguin | sip_reg_timeout: -- Registration for 'yourusername@proxy1.newyork.talk4free.com' timed out, trying again |
| 07:05.35 | TehRabbitt | ? |
| 07:05.59 | joobie | guys trying to setup a boot server to pass the configs to my polycom phone.. I went http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html and downloaded the "SoundPoint IP, SoundStation IP and Polycom VVX SIP 3.2.3 [Split]" package.. within it there's a heap of .ld files for each product.. i've loaded that up correctly and it updated the bootrom, but just not sure where sip.ld is - it's not in the zip file.. |
| 07:08.37 | p3nguin | bad gateway |
| 07:08.50 | p3nguin | Need a better gateway. |
| 07:09.01 | TehRabbitt | what defines a better gateway lol |
| 07:09.06 | p3nguin | one that works |
| 07:09.18 | TehRabbitt | that one does work... it's working on a PAP2 right now |
| 07:09.24 | p3nguin | hmm |
| 07:09.31 | TehRabbitt | dialtone and everything |
| 07:09.57 | p3nguin | Can you verify the proxy address in the ATA? |
| 07:10.11 | TehRabbitt | proxy1.newyork.talk4free.com:5070 |
| 07:10.57 | TehRabbitt | If I use the info from one of my other SIP provders it works |
| 07:11.04 | TehRabbitt | however I want this one to be the primary for 411 |
| 07:11.14 | TehRabbitt | and a few other #'s (long distance primarally) |
| 07:12.00 | p3nguin | http://pastebin.com/gDsQaYE6 |
| 07:12.03 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 07:12.24 | TehRabbitt | it works? |
| 07:12.37 | p3nguin | For toll free 411, you don't even need to use magicjack. |
| 07:12.42 | p3nguin | no, it gives me a bad gateway. |
| 07:12.54 | TehRabbitt | oh :( |
| 07:13.07 | p3nguin | But that's the peer definition I'm using. |
| 07:13.11 | TehRabbitt | again i just wanna see if I can get 411 working on that one since it's a good way of testing it |
| 07:13.30 | p3nguin | You could test it to a free toll-free termination service. |
| 07:13.44 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 07:13.46 | TehRabbitt | http://magicjackhacks.blogspot.com/2007/11/changing-proxy-servers-on-magicjack.html |
| 07:13.52 | TehRabbitt | choose your pick 0_o |
| 07:17.06 | p3nguin | same problem |
| 07:17.18 | TehRabbitt | sigh :( |
| 07:17.30 | TehRabbitt | could it be the non-standard port? |
| 07:18.27 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:18.48 | TehRabbitt | 800 termination wont work either btw |
| 07:19.06 | TehRabbitt | exten => _800NXXXXXX,1,Dial(SIP/1{EXTEN}@proxy.ideasip.com,60) |
| 07:19.30 | p3nguin | Now that should be working. |
| 07:19.42 | TehRabbitt | "temp fail" is what the phone says |
| 07:19.46 | TehRabbitt | no route to host |
| 07:19.55 | TehRabbitt | i've reloaded the dialplan several times |
| 07:21.28 | *** join/#asterisk Polysics (~Luca@host207-51-dynamic.24-79-r.retail.telecomitalia.it) |
| 07:21.30 | Polysics | hello |
| 07:22.00 | ChannelZ | o hell |
| 07:22.01 | Polysics | if the Cdr event isn't reliable for me, can i rely on Bridge and Unlink to tell me when a call is picked up and ends? |
| 07:22.15 | p3nguin | What does "dialplan show 8004444444@users" show you? |
| 07:22.23 | *** join/#asterisk coppice (~chatzilla@m121-202-83-86.smartone-vodafone.com) |
| 07:23.03 | TehRabbitt | thoth*CLI> dialplan show 8004444444@users |
| 07:23.03 | TehRabbitt | [ Included context 'external' created by 'pbx_config' ] |
| 07:23.03 | TehRabbitt | <PROTECTED> |
| 07:23.03 | TehRabbitt | <PROTECTED> |
| 07:23.47 | *** join/#asterisk nix8n82 (~nathan@63.162.27.14) |
| 07:24.44 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:26.46 | p3nguin | Your MJ setup doesn't work for me, but ideasip tollfree gateway does. |
| 07:27.39 | TehRabbitt | :( |
| 07:27.44 | TehRabbitt | still not working for me |
| 07:27.52 | ChannelZ | Maybe you should just go back to writing letters |
| 07:28.21 | Polysics | is Unlik ALWAYS fired when an answered call ends? |
| 07:28.25 | Polysics | *unlink |
| 07:28.43 | TehRabbitt | funny ChannelZ |
| 07:28.47 | JAMMAN2110 | Well they would fire the receptionist |
| 07:28.52 | JAMMAN2110 | But shes obviously not there to fire |
| 07:29.29 | ChannelZ | eh? |
| 07:29.34 | JAMMAN2110 | (Please ignore how sexist that comment is) |
| 07:30.21 | ChannelZ | oh. I get it |
| 07:30.35 | JAMMAN2110 | :) |
| 07:30.36 | ChannelZ | I don't know anyone named Unlik |
| 07:30.52 | JAMMAN2110 | There is that guy called "Link" on the Matrix |
| 07:30.59 | JAMMAN2110 | Guess everyone has an opposite |
| 07:31.03 | ChannelZ | Make a good stripper name |
| 07:31.10 | Polysics | the Unlink AMI event, come on :-) |
| 07:31.33 | ChannelZ | Hush, you're interrupting our irreverence |
| 07:31.46 | Polysics | oh, i am sorry |
| 07:32.15 | JAMMAN2110 | Polysics - do you see it being "fired" everytime a call isnt answered? |
| 07:32.57 | TehRabbitt | hm is it true https://www.future-nine.com will pay you for toll-free calls? |
| 07:33.23 | *** join/#asterisk drcode (~c7cbb864@gateway/web/freenode/x-tunoadprlifnkkhy) |
| 07:33.23 | TehRabbitt | because if so, i'll set them up for the Tech support #'s that put me on hold for 2 hours + |
| 07:33.24 | TehRabbitt | lol |
| 07:33.25 | drcode | hi all |
| 07:34.13 | ChannelZ | HAI! |
| 07:35.14 | p3nguin | tehrabbitt: Where did you hear that? |
| 07:35.51 | drcode | is there support in h323? |
| 07:35.56 | drcode | in astriks? |
| 07:36.11 | p3nguin | perhaps you meant asterisk |
| 07:36.21 | Polysics | JAMMAN2110, it apparently does, yes |
| 07:36.23 | ChannelZ | yaes thar is |
| 07:36.27 | drcode | yes |
| 07:36.34 | TehRabbitt | http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers |
| 07:36.36 | TehRabbitt | that's where |
| 07:36.37 | JAMMAN2110 | Well then Polysics - You've answered your own question :) Good work! |
| 07:36.38 | drcode | I mean somthing like mcu |
| 07:36.43 | TehRabbitt | Future Nine Requires an account to be created - but you can terminate toll-free traffic for free. No payment required. If your toll-free volume is high Future Nine may even compensate you (pay you) for those minutes. (SIP registration not required) |
| 07:36.59 | TehRabbitt | Simwood eSMS Require an account to be created but offers a small credit to your account for all US toll-free traffic. If volume is higher, commercial outpayments are available on normal settlement terms. |
| 07:38.04 | ChannelZ | ok WTF, pulldown menus in Illustrator are suddently opening to the left of the menu instead of the right. |
| 07:38.59 | JAMMAN2110 | Bad karma |
| 07:39.00 | Polysics | wiki says multiple unlinks can be seen for a single call |
| 07:39.14 | TehRabbitt | p3nguin: see what i mean? |
| 07:39.25 | p3nguin | yep |
| 07:39.37 | TehRabbitt | do you think it's true? |
| 07:39.37 | Polysics | so i am not sure i can reliably use that |
| 07:39.44 | p3nguin | probably |
| 07:39.49 | TehRabbitt | they support 729 codec too which is nice |
| 07:40.45 | p3nguin | You bought some g.729 licenses? |
| 07:40.46 | Polysics | i have never seen more than one Unlink for a call, then again, i have never seen asterisk not being able to set up a native bridge |
| 07:40.55 | Polysics | tbh i have no idea how to produce tath to test |
| 07:41.56 | TehRabbitt | no lmao but it's good for future use i suppose |
| 07:42.15 | *** join/#asterisk CleanerX (~nix@HSI-KBW-109-192-057-206.hsi6.kabel-badenwuerttemberg.de) |
| 07:43.08 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 07:44.21 | TehRabbitt | booo ok something is wrong here :( |
| 07:48.04 | TehRabbitt | any reason my dialplan son't show existsance of 8004444444@users? |
| 07:48.29 | TehRabbitt | exten => _1800NXXXXXX,1,Dial(SIP/futurenine/${EXTEN}) |
| 07:48.47 | ChannelZ | There's no 1 |
| 07:50.27 | TehRabbitt | works now :) |
| 07:52.36 | TehRabbitt | what is an incoming DID? |
| 07:53.06 | p3nguin | ~did |
| 07:53.07 | infobot | extra, extra, read all about it, did is Direct Inward Dialing, or just a phone number |
| 07:53.13 | TehRabbitt | ah lol |
| 07:54.42 | TehRabbitt | hey p3nguin, what was that 911 announce you told me to set up? |
| 07:55.01 | TehRabbitt | exten => 911,1,Playback(no-911-2) right? |
| 07:55.06 | p3nguin | yeah |
| 07:55.17 | TehRabbitt | what happens if i dial 911 afterwards? just an error? |
| 07:55.39 | p3nguin | It plays the sound file by the name of no-911-2, and then exits. |
| 07:56.29 | TehRabbitt | hm, question... how do I setup music on hold? |
| 07:56.30 | TehRabbitt | is it easy? |
| 07:56.34 | TehRabbitt | i have an MP3 I want to use |
| 07:56.34 | TehRabbitt | lol |
| 07:59.48 | *** join/#asterisk Polis_ttt (~lasse@irc.mussla.se) |
| 07:59.51 | TehRabbitt | ?? |
| 08:00.27 | nix8n82 | yes it's easy if you know what you are doing |
| 08:00.33 | ChannelZ | it's much easier to just convert it to an 8khz wav and use it that way |
| 08:00.55 | TehRabbitt | well where do i configure the location to the wav file? |
| 08:01.04 | ChannelZ | musiconhold.conf |
| 08:06.13 | TehRabbitt | ok i put an audio file in the /var/lib/asterisk/moh directory but MOH doens't work |
| 08:06.42 | TehRabbitt | nvm |
| 08:06.43 | TehRabbitt | lmao |
| 08:07.17 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:08.29 | *** join/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
| 08:09.47 | Boardy | I have 3 phone numbers and so I register 3 times (with the same provider) in my sip.conf. But it seems my provider is a little bit confused by this at startup of asterisk. |
| 08:09.51 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 08:09.55 | DelphiWorld | good morning |
| 08:10.07 | Boardy | Is it possible to have these 3 registrations happen with a certain interval? |
| 08:10.07 | davidstrauss | Boardy: You shouldn't have to register three times just for three DIDs. |
| 08:10.19 | davidstrauss | Boardy: Who is your provider? |
| 08:10.28 | DelphiWorld | anyone know how do i build a asterisk based call generatore? |
| 08:10.29 | Boardy | xs4all, dutch provider. |
| 08:11.05 | davidstrauss | Boardy: Generally, providers let you route any number of DIDs to peers you configure on your acount. |
| 08:11.07 | davidstrauss | account* |
| 08:11.08 | Boardy | They have their VoIP handled by b3g (French) |
| 08:11.31 | davidstrauss | Boardy: Does your provider create a new peer per DID? |
| 08:11.37 | Boardy | yes. |
| 08:11.57 | davidstrauss | Boardy: And it's a problem if you register them all at once? |
| 08:12.02 | Boardy | Exactly. |
| 08:12.12 | davidstrauss | Boardy: Why is that a problem? |
| 08:12.30 | ChannelZ | call your provider and tell them to fix their shit |
| 08:12.35 | Boardy | The problem is that Asterisk "thinks" they are all registered, but I can't be called. |
| 08:12.48 | davidstrauss | Boardy: I'm worried you're creating a workaround to a problem you don't fully understand. |
| 08:13.02 | davidstrauss | Boardy: I'm not even convinced that staggered registration would fix it. |
| 08:13.03 | Boardy | That might well be the case... |
| 08:13.45 | Boardy | Well... I turned off my server last weekend, because of a thunderstorm and had to get things going afterwards. |
| 08:13.53 | davidstrauss | Boardy: Your provider almost certainly has thousands of DIDs and peers it manages registrations for. It would take *extra effort* for them to throttle registrations between your DIDs/peers. |
| 08:15.00 | Boardy | Ok... I don't know exactly what's the problem but: I can call myself from one DID to another. But I can't call myself from my cell phone |
| 08:15.23 | Boardy | I finally managed to solve it by uncommenting all but 1 registration |
| 08:15.47 | Boardy | Sorry... should be "commenting" |
| 08:16.01 | Boardy | Then that one DID worked. |
| 08:16.01 | TehRabbitt | w00t MoH works... SIP works... SCCP works... Toll Free dialing works... 411 and 911 work... now all I have to do is get the damn incoming / outgoing SIP working lmao |
| 08:16.09 | TehRabbitt | anyway i'm off to bed... night everyone |
| 08:16.14 | TehRabbitt | thanks again to p3nguin |
| 08:16.15 | TehRabbitt | :-D |
| 08:16.35 | davidstrauss | TehRabbitt: I'm sending a hitman to your location just to verify that 911 *really* works. |
| 08:16.48 | TehRabbitt | LOL |
| 08:16.51 | Boardy | After that I uncommented one more registration |
| 08:16.57 | TehRabbitt | night all |
| 08:16.57 | Boardy | Then the 3rd. |
| 08:17.50 | Boardy | So it's a little bit strange to get it working this way, but now everything is Ok. (Until the next thunderstorm) |
| 08:21.21 | davidstrauss | Boardy: If delaying the registrations does, indeed, fix things, you may be able to set the peers to *not* automatically register and then have a script that uses the CLI or management interface to do them in a staggered way. |
| 08:22.20 | Boardy | Ok... |
| 08:22.48 | Boardy | Sounds sensible. |
| 08:23.03 | Boardy | But indeed twisted to have to do it this way. |
| 08:24.02 | davidstrauss | Boardy: It's your provider's fault. |
| 08:24.30 | Boardy | Yes. It definitely is. |
| 08:24.44 | Boardy | Thanks for your help. |
| 08:25.00 | *** join/#asterisk MiserySoft (~LND@109.180.149.188) |
| 08:30.57 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
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| 08:43.56 | *** part/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
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| 09:02.16 | *** join/#asterisk daMull (~beutenmue@80.64.183.252) |
| 09:03.16 | daMull | Hey I got a problem using SNOM 3x0 phones and an hosted asterisk pbx. The phones won't respond to the "Unauthorized" answer of the asterisk with Digest auth |
| 09:04.51 | daMull | it's basically the same sip trace as in: http://wiki.snom.com/SIP_Traces, but the last step of answering the "Unauthorized" correctly doesn't happen. any ideas? (Snoms are 320, 360 and 370 using firmware 7.3.30 and factory defaults) |
| 09:05.11 | daMull | (I know it's slightly offtopic ..) |
| 09:05.29 | Gido-E | i like to be offtopic |
| 09:05.40 | Gido-E | ontopic is boring. |
| 09:08.02 | daMull | ok ;-) any interesting ideas? |
| 09:08.39 | Gido-E | daMull dont use SNOW phones, good enough? |
| 09:09.53 | daMull | Gido-E: someone here bought bout 20 pcs .. so not an option |
| 09:10.02 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 09:10.52 | Gido-E | daMull yea, default problem. |
| 09:11.35 | *** join/#asterisk noisewaterphd (~noisewate@c-98-202-190-92.hsd1.ut.comcast.net) |
| 09:11.35 | daMull | Gido-E: so basically I'm stuck and need to get it working ;-) |
| 09:11.37 | *** part/#asterisk noisewaterphd (~noisewate@c-98-202-190-92.hsd1.ut.comcast.net) |
| 09:14.22 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 09:15.07 | Gido-E | daMull yep, something like that. |
| 09:15.50 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 09:22.10 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.91) |
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| 09:38.28 | *** join/#asterisk ttwhy (~tekkno@p4FECF52E.dip.t-dialin.net) |
| 09:40.35 | ttwhy | Hi, can you tell me are possible reasons if sometimes (50% of the calls) asterisk is ringing and the client is receiving the call, but if the client picks up the phone asterisk receive a hungup from the client and the call will be rejected (tried 2 differend softphones) |
| 09:41.07 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 09:45.18 | *** join/#asterisk kartik (~koolkarti@117.207.83.247) |
| 09:56.45 | ttwhy | and maybe a nother problem -> ReceiveFAX breaks after a fax is received. The Fax will be stored in the spool, but the commands after the ReceiveFAX command will be skipped (most of the times). But they tif files are 100% okay. (i use a cron now to pull them out of the spool directory which is quite lame) |
| 09:57.20 | Gido-E | yea. |
| 09:57.30 | Gido-E | i use agx_fax for my fax |
| 09:57.39 | Gido-E | txfax and rxfax and work great. |
| 09:57.53 | frk2 | hmm. Is it a good idea to use asterisk as a large scale VOIP gateway? |
| 09:58.00 | frk2 | or should i look at freeswitch for that |
| 09:59.56 | Gido-E | frk2 gateway? |
| 10:00.26 | Polysics | i am a newbie, but i have never heard about a voip gateway |
| 10:00.34 | Polysics | server, proxy, not gateway :-) |
| 10:02.10 | drcode | any one did use astk with video ? |
| 10:02.19 | drcode | can it support h323? |
| 10:03.29 | coppice | ttwhy: sounds like you are not picking up the hangup condition, so your scripts behaviour depends on which end hangs up first |
| 10:05.04 | ttwhy | HMM |
| 10:05.30 | ttwhy | so, i need to insert the Hangup condition? |
| 10:05.57 | ttwhy | i will try that |
| 10:05.59 | ttwhy | thanks |
| 10:09.27 | frk2 | Yes :) |
| 10:09.29 | frk2 | a Proxy |
| 10:09.45 | frk2 | a VOIP service provider, to sell wholesale minutes,etc |
| 10:09.57 | Gido-E | asterisk is not a proxy |
| 10:10.03 | frk2 | Gido-E, I know |
| 10:10.10 | frk2 | but it can be used as such |
| 10:10.26 | Gido-E | I wouldn't use asterisk for it. |
| 10:11.44 | frk2 | Gido-E, I know im not comfortable eitehr |
| 10:11.50 | frk2 | but the customer insists |
| 10:12.09 | frk2 | is it just a bad idea or simply wont work with that scale (and accuracy of billing) |
| 10:12.55 | Gido-E | frk2 the customer pays, why even worry? |
| 10:13.22 | frk2 | Gido-E, the customer also comes to you when things break and makes your vacations impossible :) |
| 10:14.03 | Gido-E | frk2, don be that stupid to give him that support level. |
| 10:16.13 | *** join/#asterisk davidstrauss_ (~davidstra@wikimedia/davidstrauss) |
| 10:16.17 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
| 10:16.22 | frk2 | but i cant just sell them stuff that sucks |
| 10:16.32 | frk2 | while i've used asterisk for 150-200 concurrent calls |
| 10:16.42 | frk2 | i dont know how it fares with 1000 node concurrent |
| 10:16.46 | frk2 | 1000 concurrent calls |
| 10:16.56 | frk2 | and then there is the problem with SIP interoperability |
| 10:28.28 | ManxPower | How do I confirm hardware EC is enabled on a Digium card? |
| 10:31.09 | frk2 | Has anybody here used a2billing? |
| 10:35.13 | *** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl) |
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| 11:12.06 | daMull | how can I manually test a sip register from a pc? |
| 11:12.23 | daMull | are there tools for that? |
| 11:12.35 | ManxPower | daMull, your question makes no sense |
| 11:13.26 | kaldemar | daMull: use a software phone |
| 11:14.17 | daMull | kaldemar: what would you recommend on a Linux host? (probably including the capability to log) |
| 11:17.58 | ttwhy | daMull, ekiga or zoiper |
| 11:20.25 | Gido-E | twinkle doesn't compile annymore for kde4, is my experience. SO for me it is now, Ekiga. |
| 11:23.07 | ManxPower | drat! Digium support is much more clever at avoiding customers that I thought. |
| 11:26.10 | gelo | ?¿?¿ |
| 11:26.28 | ManxPower | It is pretty cool. You must have a serial number in order to get support on a card. There is no way that I know of to get the serial number without opening up the server. |
| 11:26.30 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 11:26.40 | gelo | hehe |
| 11:26.57 | gelo | you should have thought of it prior to installing the card, shouldn't you? :) |
| 11:27.21 | ManxPower | gelo, considering our history with Digium cards, yes, we should have. |
| 11:28.01 | Gido-E | Or just email them all, with the text, One of these. |
| 11:28.08 | ManxPower | it is also pretty cool that a card can only be registered to one account. So we need a "company" account rather than an account for each tech. |
| 11:28.23 | ManxPower | Gido-E, I did. the message got rejected. |
| 11:28.43 | Gido-E | ManxPower :-) Yea, they dont need you. |
| 11:29.05 | Gido-E | ManxPower registering cards, i have never done that. |
| 11:29.17 | ManxPower | If I had the money I'd buy the customer a Sangoma out of my own pocket and just avoid these problems |
| 11:29.23 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 11:29.28 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 11:29.38 | Gido-E | ManxPower why register the hardware at all? |
| 11:29.51 | ManxPower | Gido-E, because they won't provide support if you don't register your card? |
| 11:30.12 | Gido-E | support on the hardware, which hardware? |
| 11:30.23 | ManxPower | You can't even open a case without selecting your problem card from a list of cards registered on that account. |
| 11:31.02 | ManxPower | Gido-E, TDM24xxP |
| 11:31.06 | *** part/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 11:31.50 | Gido-E | pci expres, ISDN PRI? No problems with those cards. |
| 11:32.01 | Gido-E | I didn't even know you could file a bug. |
| 11:32.28 | ManxPower | wctdm24xxp 0000:06:08.0: Found a Wildcard TDM: Wildcard AEX2400 (0 digital modules, 24 analog modules) |
| 11:32.52 | ManxPower | wctdm24xxp 0000:06:08.0: Missed interrupt. Increasing latency to 23 ms in order to compensate. |
| 11:32.55 | ManxPower | bunches of those. |
| 11:33.29 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 11:35.32 | carrar | nice |
| 11:35.47 | carrar | I use a T1 or sip for that many analog ports |
| 11:36.03 | carrar | to some other analog outputting device |
| 11:36.03 | ManxPower | carrar, I would to if I had a choice. |
| 11:36.15 | ManxPower | But these are PLAR to a trading floor |
| 11:36.20 | carrar | inform the customer it's the better option |
| 11:36.44 | ManxPower | Already have, already tried. |
| 11:37.11 | carrar | they rather take the chance of a blackout? |
| 11:37.28 | carrar | aka card crapping out |
| 11:37.36 | ManxPower | turns out the "carrier" (aka some company that does PLAR lines for brokerage houses) can't even tell us the signaling of the CAS channels of the T-1 going into their locked channelbank. |
| 11:37.37 | carrar | or something else |
| 11:38.05 | carrar | heh |
| 11:38.24 | carrar | no remote access? |
| 11:38.30 | ManxPower | I just want Digium to tell us 1) card is broken 2) card is out of warrenty. Then I can buy a Sangoma |
| 11:38.58 | ManxPower | carrar, no access into the carrier's locked channelbank, no. |
| 11:40.29 | carrar | using fxo_ks? |
| 11:40.57 | carrar | we use that with ADCI600 is you can see that in their cage |
| 11:41.00 | carrar | ADIC |
| 11:41.05 | carrar | is=if |
| 11:41.11 | carrar | man my typign sucks |
| 11:43.05 | DND | guys what is trunkrealloc=yes ? |
| 11:43.08 | DND | what does it do? |
| 11:43.23 | DND | and iaxcompat=yes |
| 11:44.45 | ManxPower | DND, the .sample config files doesn't explain the option? |
| 11:45.59 | Gido-E | DND where is the manual of users.conf? |
| 11:46.13 | Gido-E | ManxPower documentation is poor. |
| 11:47.26 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:48.56 | *** join/#asterisk coppice (~chatzilla@m121-202-10-104.smartone-vodafone.com) |
| 11:52.56 | ManxPower | ~users.conf |
| 11:52.56 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 11:54.14 | frk2 | hahahahah |
| 11:54.20 | frk2 | toaster grade |
| 11:54.32 | coppice | I like toast |
| 11:54.37 | frk2 | yeah |
| 11:54.39 | *** join/#asterisk bminish (~bminish@pdpc/supporter/professional/bminish) |
| 11:54.45 | frk2 | doesnt that mean robust as hell? :) |
| 11:54.52 | frk2 | my toast from 1973 is still toasting away |
| 11:55.07 | frk2 | toaster |
| 11:55.35 | bminish | just tried to go to 1.6.2.6 on centos 5.4 64bit and now asterisk segfaults on startup, ideas? |
| 11:55.50 | coppice | and what other config file can offer you crumpets |
| 11:56.48 | bminish | http://pastebin.com/vtLicdED |
| 11:57.08 | carrar | DND: http://tinyurl.com/27f9weh |
| 11:57.56 | carrar | Let me know if you need help with iaxcompat=yes |
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| 11:59.20 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 11:59.26 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:00.08 | ManxPower | bminish, "asterisk -cvvv" to test |
| 12:01.45 | bminish | ManxPower: segfaults after func_rand.so see http://pastebin.com/9XgPwVAP |
| 12:02.28 | plundra | Hmm, do I need to restart something for set{interface,queueentry,queue}var to affect new calls? |
| 12:02.57 | Gido-E | ManxPower nice discription. But where is the howto/documentation of users.conf? |
| 12:02.57 | plundra | I've done a queue reload all, as well as reloading the dialplan, but doens't seem to set any variable anyway. |
| 12:03.54 | ManxPower | bminish, then remove it |
| 12:04.17 | ManxPower | Gido-E, I don't know since I don't use it. |
| 12:05.21 | bminish | ManxPower: did that, it's not the func_rand module it's whatever comes after it .. |
| 12:05.28 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 12:06.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:06.18 | ManxPower | bminish, then remove everything from /usr/lib/asterisk/modules and reinstall Asterisk |
| 12:06.29 | bminish | got as far as func_channel.so this time around, I kinda need that module.. |
| 12:07.42 | bminish | ManxPower: good call |
| 12:08.28 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 12:09.11 | bminish | ahh, it's G279 from 1.6.1.x that's upsetting things |
| 12:11.07 | daMull | damned snom phones .. zoiper registers perfect .. so it cannot be the router |
| 12:11.18 | ManxPower | of course it can be the router1 |
| 12:11.40 | ManxPower | did you do something stupid like enable NAT support on the SNOM? |
| 12:12.01 | ManxPower | Remember phone nat + asterisk nat = not working NAT |
| 12:12.05 | daMull | ManxPower: I have them on factory defaults |
| 12:12.25 | ManxPower | daMull, is the factory default to enable NAT or disable NAT support? |
| 12:12.40 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 12:16.29 | daMull | ManxPower:the only settings I find in the nat category are a possible stun server |
| 12:17.43 | daMull | ManxPower: Besies I see incoming udp packets from the asterisk server in the phone sip trace, so connection probably works |
| 12:17.57 | ManxPower | I assume you actually mean "<daMull> ManxPower:the only settings I find in the nat category are a possible stun server and those settings are disabled" |
| 12:18.09 | carrar | haha |
| 12:18.28 | daMull | ManxPower: Yes ;-) sorry for typos and crappy english |
| 12:18.33 | ManxPower | carrar, I'm sometimes amazed at how difficult it is to get people to answer a simple question. |
| 12:19.25 | daMull | I am astonished how difficult it is facing hundreds of weird names options, to answer a simple question. |
| 12:19.50 | ManxPower | daMull, And yet you even told me the option you were looking at and STILL didn't say if it was disabled or enabled. |
| 12:20.08 | Gido-E | :-) |
| 12:20.29 | ManxPower | daMull, I wish you the BEST of luck. |
| 12:21.41 | bminish | ManxPower: thanks for the help, sorted now |
| 12:22.01 | carrar | Internet Licenses, great idea! |
| 12:23.44 | daMull | ManxPower: I honestly don't know if the stun server settings, are the only nat related settings in those damned phones |
| 12:24.47 | bminish | one last question, upon a restart my hints for the parked calls slots always come up as 'in use' when the are really idle, parking and then unparking a call in eahc slot cures this but is this a bug ? |
| 12:26.48 | devmod | if i want to externally execute cmds on asterisk, how would I go about it? (ie an app that will add an agent to a queue) |
| 12:27.18 | ManxPower | a command is not an app |
| 12:27.48 | ManxPower | asterisk -rx "any CLI command" |
| 12:32.37 | *** join/#asterisk sourcode (~code@ppp-61-90-16-18.revip.asianet.co.th) |
| 12:34.07 | devmod | ManxPower, I see. So, how could a remote server execute cmds on my asterisk server? |
| 12:34.41 | ManxPower | devdvd, you would have to ssh into the server and execute the CLI commands |
| 12:35.07 | ManxPower | It's good that you are not trying to execute applications, because those can only be done as part of a manager connection or dialplan call. |
| 12:35.37 | *** join/#asterisk davidstrauss (~davidstra@wikimedia/davidstrauss) |
| 12:35.51 | devmod | ManxPower, I meant programmatically |
| 12:38.20 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 12:38.39 | Dovid | hi. is it possible to bindport to two ports ? |
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| 12:49.52 | [TK]D-Fender | Dovid: no |
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| 12:53.21 | Dovid | TK: can i set bindport for a peer ? (i doubt it) |
| 12:53.28 | Dovid | if not I will just set up OpenSIps |
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| 12:57.59 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 13:12.19 | guyvdb_ | What is the equiv of Hook-On/Hook-Off time in asterisk? |
| 13:13.05 | Katty | morning |
| 13:13.36 | Baylink-stillafk | Morning, Katty. You find a place to do your shades? |
| 13:13.41 | Baylink-stillafk | guyvdb_: Expand? |
| 13:15.12 | Katty | Baylink-stillafk: no i just sat around on my tail all weekend. |
| 13:15.36 | Baylink-work | Heh. |
| 13:15.47 | guyvdb_ | Baylinl-stillafk : In an analog system hook-on is the minimum time a system recognize as an SLT hang up |
| 13:15.56 | Baylink-work | I changed out battery strings in 3 3kVA UPSs. Damn, lead's heavy. |
| 13:16.20 | Katty | i was supposed to learn how to play magictg this weekend. |
| 13:16.29 | Katty | we were gonna go to one of the quieter bars and play there, but.. |
| 13:16.41 | Baylink-work | I hadn't heard the phrase before, guyvdb_. Since SIP is an authoritative control channel, I don't understand that it matters. You send the SIP equivalent of "DISCONNECT" and the call's gone. |
| 13:16.42 | Katty | we got to drinking and giggling and it never happened |
| 13:16.56 | Baylink-work | Some of the best times you'll never remember, yes... |
| 13:17.05 | Katty | and then came the ill |
| 13:17.16 | guyvdb_ | Ok thx |
| 13:17.32 | devmod | What would I used if i wanted to have a web app make use of AMI? is there a lib out there for any web tech i could use? |
| 13:18.32 | Baylink-work | You're looking for "language bindings" to AMI for web CGIish languages, like PHP, devmod? |
| 13:18.53 | *** join/#asterisk MiserySoft (~LND@dyn-62-56-86-50.dslaccess.co.uk) |
| 13:19.28 | devmod | Baylink-work, right that is what i meant |
| 13:19.44 | Baylink-work | How hard are you planning to hammer your Asterisk instance, and which version is it? |
| 13:20.24 | ManxPower | Katty, I guess Ferrets are not so bad after all: http://fukung.net/v/21342/6ffa66b6e179d6a3cdd5f5cd9e31fb9e.jpg |
| 13:20.43 | devmod | 1.6.2.6 and I am right now developing just a proof of concept. But eventually might not be so hammered but definitely several concurrent request might occur |
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| 13:21.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:21.28 | devmod | Baylink-work, 1.6.2.6 and I am right now developing just a proof of concept. But eventually might not be so hammered but definitely several concurrent request might occur |
| 13:21.32 | Baylink-work | Ok. I ask this, because the guy who wrote Vicidial used to have my job, and he tells me that older Asterisks got unhappy if you did too many connection-setups to AMI per minute. Lock-up unhappy. Sounds like you'll be ok. |
| 13:21.35 | |amadeus| | guten morgen |
| 13:21.39 | Baylink-work | Morn. |
| 13:21.49 | MiserySoft | hi all , anyone have an experience (good or bad) with TDM400 clones from ebay ? My authentic digium board has died. Looking for a cheap replacement. |
| 13:21.59 | devmod | Baylink-work, right now all i was trying to accomplish really was a web interface for the agents to log into |
| 13:22.39 | Baylink-work | "agents" |
| 13:23.49 | devmod | agents as in queues |
| 13:24.32 | Baylink-work | Have you *looked* at Vicidial? |
| 13:25.17 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:25.49 | Baylink-work | If you're doing nearly anything that involves Agents and Queues, it might be a better solution to your problem (#include <stddisclaimer.h>) |
| 13:27.01 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 13:29.46 | Andrew_M_ | Hi Baylink-work: What can Vicidial do that Asterisk cannot? |
| 13:30.21 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
| 13:30.27 | hurdman | is there a nice way to convert an existing diaplan to a schemas ? |
| 13:30.51 | Baylink-work | Well, vicidial can't do *anything* without Asterisk, of course, but other than that, it does substantially everything you can get out of any commercial callcenter management package, including full web management and reporting, and a web interface for agents. |
| 13:32.32 | [TK]D-Fender | [09:21]<Baylink-work>Ok. I ask this, because the guy who wrote Vicidial used to have my job, and he tells me that older Asterisks got unhappy if you did too many connection-setups to AMI per minute. Lock-up unhappy. Sounds like you'll be ok. <-astmanproxy |
| 13:33.22 | Andrew_M_ | Baylink-work: Oh, OK, Is it like Qmetrics? |
| 13:33.41 | Baylink-work | I'm not familiar with that, but probably... |
| 13:34.19 | Baylink-work | In fact, I think that merely instruments the built in queue facilities on Asterisk. |
| 13:34.24 | Baylink-work | VD replaces them completely. |
| 13:36.15 | Baylink-work | VD is also GPL. |
| 13:36.53 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:37.04 | Andrew_M_ | Baylink-work: Thanks! |
| 13:37.10 | Baylink-work | NP. |
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| 13:45.14 | maruz | originate use dtmf to call out? |
| 13:46.47 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:47.10 | devdvd | hey, does asterisk have the ability to dial all agents in a queue n times then drop out to the next priority (ex. Dial 1>2>3>4>1>2>3>4>drop out of queue into next priority) or is it all based on timeouts |
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| 13:57.52 | pabelanger | Anybody have any documentation on the distance limitations for T1/E1? |
| 13:59.34 | pabelanger | I believe it is 655ft |
| 13:59.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
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| 14:00.10 | devdvd | pabelanger: you talkin from dmarc to device or outside of dmarc? |
| 14:00.37 | devdvd | putnopvut: master of queues eh? |
| 14:00.54 | putnopvut | heh, yeah leifmadsen came up with that one :) |
| 14:01.00 | devdvd | hehe |
| 14:01.02 | leifmadsen | :D |
| 14:01.04 | leifmadsen | because he is |
| 14:01.10 | pabelanger | devdvd: Total cable length between any T1 device. |
| 14:01.12 | leifmadsen | let it be said; let it be known! |
| 14:01.18 | devdvd | well maybe you can answer me a question or 2 :) |
| 14:01.26 | devdvd | does asterisk have the ability to dial all agents in a queue n times then drop out to the next priority (ex. Dial 1>2>3>4>1>2>3>4>drop out of queue into next priority) or is it all based on timeouts |
| 14:01.34 | leifmadsen | I can only please one person per day. Today is not your day. Tomorrow doesn't look good either. |
| 14:01.52 | devdvd | who do i have to kill for it to be my day? |
| 14:02.04 | putnopvut | devdvd: no there's no configurable number of cycles. The best you can do is base it on timing. |
| 14:02.08 | leifmadsen | devdvd: ummm... yes you can do that. It has to do with number of retries |
| 14:02.12 | putnopvut | leifmadsen ? |
| 14:02.18 | leifmadsen | putnopvut: I could have sworn you could do that... |
| 14:02.29 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 14:02.29 | leifmadsen | but I'm no master of queues! |
| 14:02.33 | putnopvut | The only thing I'm familiar with is the n option, which will go through one cycle of members. |
| 14:02.36 | putnopvut | And then drop |
| 14:02.46 | leifmadsen | I'm thinking in queues.conf there is a retries option... hmmm |
| 14:02.58 | putnopvut | there's a retry option, which will tell how long to wait before retrying. |
| 14:02.59 | devdvd | there is |
| 14:03.00 | devdvd | but |
| 14:03.09 | devdvd | the retry options tells it howlong to wait before retrying again |
| 14:03.19 | devdvd | yea what put said :) |
| 14:03.30 | leifmadsen | putnopvut: ah yes, you are correct -- I just looked at the docs |
| 14:03.37 | leifmadsen | appears as if queues is all based on timing -- not cycles |
| 14:03.41 | devdvd | ok |
| 14:03.43 | devdvd | thats fine :) |
| 14:03.47 | devdvd | i can suffer with that |
| 14:04.33 | [TK]D-Fender | devdvd: just Dial() the bunch of them. |
| 14:05.37 | devdvd | yea but that would negate the queue wouldnt it? |
| 14:09.58 | [TK]D-Fender | devdvd: yeah i would kill ordering... depend if you have the volume to deal with |
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| 14:16.33 | roe | what is the preferred method of ensuring that multiple digium cards come up in the same order at boot? |
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| 14:19.58 | Polysics | how do i ask * for its exact version on the command line? |
| 14:20.28 | Mark22 | core show version |
| 14:20.33 | Mark22 | if I remember correctly |
| 14:20.36 | Polysics | thx |
| 14:20.37 | ManxPower | asterisk -rx "core show version" |
| 14:21.26 | Polysics | anyone ever had problems with 1.6.11, AGI and hangups? |
| 14:21.59 | Polysics | apparently, when the calleR hangs up a call, then you request DIALSTATUS via AGI, you get HANGUP200 result=1 (CANCEL), which is one HANGUP too much :-) |
| 14:22.22 | Polysics | i'd blame some buffer not being flushed, if i had the slightest idea :-) |
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| 14:33.09 | devyll | is there any knows issues with Fax machines via Asterisk ? |
| 14:33.24 | ManxPower | devyll, yes. |
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| 14:35.35 | devyll | Is it possible to create an error free env with asterisk and fax machines ? (tunning asterisk in a specific way / buying a specific type of fax machine with some .. special types of modems or features) |
| 14:37.14 | ManxPower | devdvd, stick to PSTN, no VoIP and chances are it will work fine. |
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| 14:46.34 | devyll | ManxPower, same for app_rx and app_tx ? |
| 14:46.49 | devyll | fax2email I mean |
| 14:46.59 | devyll | incoming and outgoing |
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| 14:51.36 | mnick86 | hi, how can I log all the manger-events sent by asterisk ? |
| 14:51.56 | russellb | mnick86: hi :-) |
| 14:52.18 | russellb | log on the asterisk console? |
| 14:52.21 | mnick86 | ahoi russell :) |
| 14:52.29 | mnick86 | I want it in a file |
| 14:52.33 | russellb | gotcha. |
| 14:52.43 | mnick86 | I am currently on telnet, but that's not nice :) |
| 14:52.43 | Nugget | telnet is eeeeeeevil! |
| 14:52.46 | russellb | there is nothing built in for asterisk for it |
| 14:53.10 | mnick86 | okay I see ... I will use telnet and some pipes |
| 14:53.41 | russellb | k |
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| 14:58.49 | ManxPower | devdvd, you changed your question |
| 14:59.27 | Bartockbatz | okay - dumb-ass newb question |
| 15:00.23 | Bartockbatz | I have a SIP trunk - when I call the number, I would want it to ring extension xxxx for 30 seconds and if no answer go to voicemail |
| 15:00.31 | *** join/#asterisk smooth_penguin (~smoove@59.95.23.52) |
| 15:01.12 | [TK]D-Fender | Bartockbatz: then Dial() that device. Then call Voicemail. |
| 15:01.51 | *** part/#asterisk maour (~gnu@unaffiliated/maour) |
| 15:02.45 | Bartockbatz | register => john:johnspassword@sipprovider.com/6000 |
| 15:02.52 | Bartockbatz | goes to vm |
| 15:03.08 | [TK]D-Fender | Bartockbatz: That has nothing to do with your DIALPLAN |
| 15:03.40 | [TK]D-Fender | Bartockbatz: That tells them to send calls to your server targeting exten "6000" in whatever context is used by whatever peer gets matched and authed for the call |
| 15:03.58 | [TK]D-Fender | Bartockbatz: it does not imply any action * will take when a call is received |
| 15:04.17 | ManxPower | What would cause this: wctdm24xxp 0000:02:08.0: VPM: Support Disabled |
| 15:04.23 | Bartockbatz | I am a little clueless - so excuse my dumb questions |
| 15:04.50 | Bartockbatz | so - to register I should use in the sip.conf the following: |
| 15:05.06 | [TK]D-Fender | Bartockbatz: You did just show us you're registr line. |
| 15:05.15 | [TK]D-Fender | Bartockbatz: But again that jsut tells them where to send calls to <- |
| 15:05.24 | [TK]D-Fender | Bartockbatz: What you DO with calls is based on yoru dialplan. |
| 15:05.57 | Bartockbatz | okay - so register line is fine - but should not send directly to extension 6000 |
| 15:06.17 | [TK]D-Fender | Bartockbatz: You made the classic mistake of thinking a SIP DEVICE is an EXTENSION |
| 15:06.21 | [TK]D-Fender | Bartockbatz: it is not. |
| 15:06.27 | Bartockbatz | oh - okay |
| 15:06.35 | ManxPower | Bartockbatz, you really should read the Asterisk book |
| 15:06.37 | Baylink-work | ManxPower: VPM is apparently related to echocan, according to google. |
| 15:06.47 | ManxPower | Baylink-work, correct. now why is it disabled? |
| 15:06.49 | [TK]D-Fender | Bartockbatz: And "extension" is a number that can be dialed in yoru dialplan. What action is taken when it is dialed need not have anything to do with any kind of phone at all |
| 15:07.05 | Baylink-work | Hellifino. :-) |
| 15:07.10 | [TK]D-Fender | ManxPower: is it present ont he card? |
| 15:07.14 | Bartockbatz | okay - I will crack the books - I guess I am just a little impatient |
| 15:07.27 | ManxPower | [TK]D-Fender, it is supposed to be, but I can't fly to NYC right this moment to check for sure. |
| 15:07.32 | Baylink-work | There is a traditional litany of "why isn't my HW echocan working" things to check, is there not? |
| 15:07.40 | [TK]D-Fender | Bartockbatz: Do you also have a peer set up to auth the calls they should send you? |
| 15:07.45 | ManxPower | Baylink-work, if there is I've not been able to find it |
| 15:07.50 | [TK]D-Fender | ManxPower: Call someone local :) |
| 15:08.07 | Baylink-work | I infer this is a fresh deployment, ManxPower |
| 15:08.08 | Baylink-work | ? |
| 15:08.39 | [TK]D-Fender | I infer SFA |
| 15:09.14 | Bartockbatz | [TK]D-Fender : like you said - I need to read the Asterisk book - very clueless |
| 15:09.29 | ManxPower | Baylink-work, no, it is fresh look at the problem. |
| 15:09.33 | [TK]D-Fender | Bartockbatz: Slow and steady. Little steps will get you functional in short order. |
| 15:09.37 | ManxPower | We've had EC problems for a LONG time. |
| 15:09.47 | [TK]D-Fender | Bartockbatz: If your peer is already set up, its a matter of 3 lines of dialplan <- |
| 15:09.54 | Bartockbatz | [TK]D-Fender : Good advice - thank you |
| 15:10.01 | Baylink-work | Bartockbatz: I have a little time for remedial Asterisk, if you'd like to go off-channel |
| 15:12.53 | *** join/#asterisk TimeRider (steve@5ac3182f.bb.sky.com) |
| 15:15.09 | pabelanger | heh, all theses Asterisk webinars and can't view them because they don't offer a Linux client. |
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| 15:20.14 | Naikrovek | pabelanger: run windows in a virtual machine or switch to a more supported desktop OS |
| 15:20.25 | Naikrovek | i would go the virtual machine route, personally |
| 15:20.38 | Naikrovek | there are a lot of problems that virtual machines solve |
| 15:20.59 | pabelanger | Naikrovek: lol. not an option |
| 15:21.17 | Baylink-work | ManxPower: Isn't there some driver debugging you can turn on for boot time that will tell you *why* it doesn't like the EC? |
| 15:21.26 | Naikrovek | how is a virtual machine not an option? download vmware player (free) and download windows 7 enterprise trial (free) |
| 15:21.26 | Baylink-work | Can you go to SWEC temp? Or too much load? |
| 15:21.43 | gelo | pabelanger: ask for your girlfriend's laptop with vista |
| 15:21.49 | gelo | that's what i do :P |
| 15:22.06 | Mark22 | lol |
| 15:22.52 | pabelanger | Naikrovek: I understand the concept of running is a virtual machine. However, it is not an options for me. |
| 15:23.07 | pabelanger | gelo: gf running Ubuntu on her eeepc |
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| 15:24.27 | pabelanger | I just find it a little humorous the concert of the webinar is to promote Asterisk (open source) and the client of the webinar is Windows only. |
| 15:24.49 | [TK]D-Fender | pabelanger: Who's webinar is it? |
| 15:27.19 | pabelanger | [TK]D-Fender: this one is from Xorcom (http://bit.ly/ckNCSy) |
| 15:27.59 | leifmadsen | GotoMeeting right? |
| 15:28.12 | leifmadsen | they should be using vyew.com (which is not windows only) |
| 15:28.13 | pabelanger | leifmadsen: yar! |
| 15:28.20 | leifmadsen | hates gotomeeting for that very reason |
| 15:28.39 | russellb | digium has started doing asterisk intro webinars |
| 15:28.40 | leifmadsen | oh, and the fact that everytime I leave a meeting they spam me by opening a new browser window with a link to buy gotomeeting |
| 15:28.42 | russellb | i wonder what they use |
| 15:28.47 | leifmadsen | probably gotomeeting ;) |
| 15:28.50 | russellb | probably |
| 15:28.54 | pabelanger | russellb: gotomeeting :) |
| 15:28.56 | leifmadsen | would NOT be shocked |
| 15:29.06 | russellb | looks |
| 15:29.06 | leifmadsen | :| |
| 15:29.13 | Qwell | we do, and I've heard complaints about it. which I've passed on |
| 15:29.30 | russellb | indeed |
| 15:29.51 | russellb | oh well, at least there is a webinar at all |
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| 15:31.42 | moos3 | can any one help me with hylafax? |
| 15:31.50 | russellb | nope |
| 15:31.53 | ManxPower | Baylink-work, I don't know. Do you have a link? |
| 15:32.00 | ManxPower | What would cause this: wctdm24xxp 0000:02:08.0: VPM: Support Disabled |
| 15:32.25 | Baylink-work | ManxPower: Digium or Sangoma? |
| 15:32.36 | ManxPower | Baylink-work, Digium |
| 15:32.53 | Baylink-work | It's in the zaptel.conf, I think; lemme look |
| 15:33.11 | ManxPower | Baylink-work, echocancel=yes is set |
| 15:33.36 | ManxPower | Baylink-work, and I'm using DAHDI, not zaptel |
| 15:34.00 | Baylink-work | You may have to generalize then; I have no Dahdi here. Wait one. |
| 15:34.13 | ManxPower | I wish dahdi_cfg didn't lie about the EC being used. |
| 15:34.44 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 15:34.49 | ManxPower | Baylink-work, I am familiar with Asterisk and DAHDI. I am not familiar with the details of debugging VPM issues. That is why I am here. |
| 15:35.04 | Baylink-work | Rog. |
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| 15:39.46 | Baylink-work | ManxPower: mattf tells me that VPM is a compile time option for DAHDI/Digi, he thinks in wct4xxp.c |
| 15:39.56 | Baylink-work | Have you been down that road? |
| 15:40.03 | Baylink-work | Or did it used to work ok? |
| 15:40.09 | Baylink-work | (ie: not disables) |
| 15:40.12 | ManxPower | I cannot say if it ever worked. |
| 15:40.25 | Baylink-work | Ok. Hopefully that's a pointer then. |
| 15:40.40 | ManxPower | No, it is not disabled in wctdm24xxp.h as far as I can tel. |
| 15:41.03 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:41.35 | Baylink-work | That's the equivalent file for your model of card, then? |
| 15:41.39 | ManxPower | We ended up using HPEC just to keep the users from screaming |
| 15:41.45 | ManxPower | Baylink-work, correct. |
| 15:42.07 | Baylink-work | Hmmm. Google tells me there's a debug=1 option for the module, but what it'll log, I couldn't tell you. |
| 15:42.14 | ManxPower | I'll rebuild DAHDI from the official tarball this evening to confirm. |
| 15:42.53 | ManxPower | Baylink-work, it is OK to say you have no idea how to fix the issue. |
| 15:43.05 | Baylink-work | I know that. |
| 15:43.56 | Baylink-work | Don't know you well enough to predict what you've already tried. |
| 15:47.39 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
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| 15:57.29 | leifmadsen | has anyone tried loading SFA with asterisk trunk? |
| 15:57.46 | leifmadsen | I tried a while ago and it failed, but I think that was a problem in 1.6.2 that got fixed as well and I haven't had a chance to try again recently |
| 15:58.05 | leifmadsen | really wants the calendar integration stuff, but not lose the the SFA and G.729 modules ;) |
| 16:00.48 | *** join/#asterisk scarecrowed (~fabiano@201.11.153.136) |
| 16:09.01 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 16:09.43 | raden_work | why does exten => s,n,Dial(SIP/101,20) not follow what i have forwarded in the database |
| 16:09.52 | raden_work | just rings that extensions |
| 16:12.14 | paulc | raden_work: What do you expect it to do? That's a standard "Dial" command right there.. |
| 16:12.19 | *** join/#asterisk davidstrauss (~davidstra@wikimedia/davidstrauss) |
| 16:15.26 | leifmadsen | Dial() is doing it exactly what you're telling it to do... |
| 16:15.40 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.142.15.dsl.dyn.forthnet.gr) |
| 16:16.01 | ManxPower | raden_work, AGAIN, that is a FreePBX thing |
| 16:17.03 | raden_work | ok yes im a mornon it has to check the database |
| 16:17.08 | raden_work | thanks :) |
| 16:17.35 | leifmadsen | asterisk does what you tell it to do |
| 16:23.05 | Defraz | Just trying to create a bash script to dial out and play message when a server is down. I have my bash script doing the check and even emailing the message but I wanted my asterisk server to call out. I have the sip trunks setup and I have a registered phone calling out and it works. |
| 16:23.18 | Defraz | No just need to write an agi script I think to do the calling out. |
| 16:23.21 | Defraz | and say the message |
| 16:23.45 | Defraz | I can send commands at the CLI to do the call out and such just don't know how to tell it to do it from the bash script. |
| 16:23.50 | Defraz | is there a tutorial for that? |
| 16:26.55 | leifmadsen | Defraz: trigger a call using a callfile |
| 16:27.15 | Defraz | Just like have it dial then wait for an answer then play the text file that would be created. |
| 16:27.21 | leifmadsen | Defraz: search google for callfile, then use bash to write the file and move it to /var/spool/asterisk/outgoing/ |
| 16:27.36 | Defraz | okay thanks |
| 16:28.57 | Defraz | nice that might be what I want. |
| 16:29.19 | [TK]D-Fender | Defraz: asterisk -rx "originate ..........." |
| 16:30.38 | leifmadsen | assuming the originate CLI command is present in his version of Asterisk |
| 16:30.52 | leifmadsen | I also don't like the idea of using the CLI to process commands from a script |
| 16:30.58 | leifmadsen | but that's just me |
| 16:32.01 | [TK]D-Fender | leifmadsen: In many cases no.. but for a single call like this with no need of a reult code, etc... more than fine, and saves a lot of other code |
| 16:32.12 | *** join/#asterisk MiserySoft (~lnd@host81-139-167-173.in-addr.btopenworld.com) |
| 16:32.18 | [TK]D-Fender | defYou could also netcat an AMI call for this :) |
| 16:32.23 | leifmadsen | hmmm.... I guess I would use the [applicationmap] in features.conf to add a call recording DTMF trigger for when I'm in a conference room? |
| 16:32.44 | leifmadsen | the 'r' option to MeetMe() implies I'm recording all conferences |
| 16:33.00 | [TK]D-Fender | leifmadsen: well.. that conference anyway |
| 16:33.08 | leifmadsen | obviously |
| 16:33.13 | leifmadsen | I mean all conference calls |
| 16:43.08 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 16:58.50 | Defraz | WEll that worked quite well thanks. |
| 17:00.13 | Brian_H | I have polycom 330 phones with the latest firmware/boot rom, when I reload my asterisk config the phones reboot, I'm not sure why. Does anyone know of a way to prevent that from happening? |
| 17:00.34 | Naikrovek | they all reboot at once? |
| 17:00.39 | Brian_H | yea |
| 17:00.49 | Naikrovek | did you use some BS helper app to configure your phones |
| 17:00.52 | Brian_H | as soon as I type "reload" in the asterisk console |
| 17:01.09 | Brian_H | no just the example configs I'm serving them up via ftp |
| 17:01.24 | Naikrovek | hm. |
| 17:01.39 | Naikrovek | the phones actually reboot? |
| 17:01.42 | Brian_H | I've got logging enabled, but I can't make heads or tails of it |
| 17:01.46 | ManxPower | Brian_H, are you using a GUI like FreePBX or Trixbox? |
| 17:01.52 | Brian_H | yea they say "rebooting now" |
| 17:02.05 | ManxPower | Brian_H, no they do not. |
| 17:02.05 | Brian_H | no gui, just asterisk configs |
| 17:02.08 | Naikrovek | freepbx and trixbox don't do this |
| 17:02.40 | Naikrovek | there's something set up to tell the phones to check for new configs, and if they're new, to reload |
| 17:02.48 | Naikrovek | i experimented with this for a few hours |
| 17:02.53 | Naikrovek | but it was annoying |
| 17:02.53 | Brian_H | I was saying, no I was not using a gui |
| 17:02.54 | ManxPower | they might say something else, but they don't say "rebooting now". |
| 17:03.13 | ManxPower | your incorrect report makes us suspect everything you tell us. |
| 17:03.41 | ManxPower | In 10 years of using Asterisk and Polycom phones I've never heard of what you are reporting |
| 17:03.58 | Naikrovek | the phones can check for new config whenever they get a SIP OPTIONS packet i think, and reload if they see the new config files |
| 17:04.01 | [TK]D-Fender | Brian_H: How did you configure them? |
| 17:04.29 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 17:04.31 | ManxPower | BTW, the message is "Restarting Phone" |
| 17:04.35 | Brian_H | via config files, on the ftp server, I can post the configs if you wish |
| 17:04.55 | [TK]D-Fender | Brian_H: PB the logs |
| 17:05.58 | lordvadr | I'm experiencing trouble with Playtones in 1.6.2.6. Only plays about 500ms of the tone and then goes silent. I've tried Answer and Progress (and both) prior to with no change. Is this a known issue or am I doing something wrong? |
| 17:06.45 | [TK]D-Fender | lordvadr: do a Playbac of 2s of silence before playtones |
| 17:07.31 | *** join/#asterisk c0dyhi11 (~chill2@edaisgroup.com) |
| 17:18.45 | iscario | hi, i would need some help to use the encryption with IAX2. here is my probleme ; http://pastebin.org/199641 . I can register as usual with my client, but i cannot make a call. is there is something special i have to configure with my client ? |
| 17:18.55 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 17:20.10 | ManxPower | On this date in 1977 the first telephone was installed in the White House |
| 17:20.23 | ManxPower | better 1877 |
| 17:21.57 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 17:22.18 | *** join/#asterisk smooth_penguin (~smoove@59.95.23.52) |
| 17:22.19 | [TK]D-Fender | iscario: chan_iax.conf:7540 authenticate_verify ; call terminated, incomming call is unencrypted while forceencrypted is enabled. <- your CLIENT isn't encrypting the call |
| 17:23.36 | iscario | so that's mean that is doesn't support the encryption feature, right ? [TK]D-Fender |
| 17:23.58 | iscario | or is it a bad client configuration ? |
| 17:24.03 | ManxPower | iscario, either |
| 17:24.05 | ManxPower | check your client |
| 17:24.09 | [TK]D-Fender | iscario: or |
| 17:26.00 | iscario | mmmh ok |
| 17:28.26 | *** part/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 17:28.58 | *** join/#asterisk imox1234 (~imox1234@p4FC5C50B.dip0.t-ipconnect.de) |
| 17:29.24 | imox1234 | hello, can somebody give me a good instruction to install CDR mysql ? |
| 17:29.55 | Qwell | imox1234: yum install asterisk-addons |
| 17:29.59 | [TK]D-Fender | imox1234: its in the addons docs |
| 17:30.11 | iscario | my client is idefisk (http://www.asteriskguru.com/tutorials/idefisk_softphone.html) how could i know if it provides this IAX2 encryption ? |
| 17:30.17 | imox1234 | i have installed the asterisk addons |
| 17:30.29 | imox1234 | and edit the cdr_mysql.conf |
| 17:30.32 | idespinner | warning: chan_dahdi.c pri_dchannel: PRI error on span 0: we think we're the CPE, but they think they're the CPE too. |
| 17:30.38 | idespinner | ^ does span 0 really mean span 1?? |
| 17:30.42 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:31.27 | imox1234 | [TK]D-Fender: there are the docs ? |
| 17:31.29 | [TK]D-Fender | imox1234: Sample table layouts and DB create scripts are included. |
| 17:31.46 | [TK]D-Fender | imox1234: Go actually look at the contents of your tarball |
| 17:31.51 | ManxPower | idespinner, you have a loopback on the line |
| 17:32.10 | [TK]D-Fender | iscario: Go read its manual |
| 17:32.13 | idespinner | ManxPower, yes |
| 17:32.33 | [TK]D-Fender | iscario: Idefisk is OLD. Good odds that encryption didn't even exist back then. |
| 17:32.37 | [TK]D-Fender | ~zoiper |
| 17:32.37 | infobot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 17:32.38 | imox1234 | [TK]D-Fender: sorry i see only readme ..... |
| 17:32.41 | idespinner | although im curious about the SPAN numbering |
| 17:32.49 | ManxPower | idespinner, it was not a question, it was a statement. You have a loop on the line so you will get that error |
| 17:32.53 | [TK]D-Fender | iscario: Also no guarantee that its newer version does. You'll have to actually go check for yourself |
| 17:33.01 | [TK]D-Fender | imox1234: Look harder |
| 17:33.15 | imox1234 | [TK]D-Fender: how called the files ? |
| 17:33.30 | ManxPower | imox1234, We are not the IDEFisk support channel |
| 17:33.40 | imox1234 | bl abla |
| 17:33.47 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 17:34.13 | [TK]D-Fender | imox1234: How about that blatantly obvious "doc" folder? |
| 17:34.17 | idespinner | ManxPower, TY, but I know SPAN 0 does not have a loopback. I'm just trying to see if someone knows if SPAN 0 is the same as span 1 in chan_dahdi.conf |
| 17:34.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:34.41 | idespinner | 8 ports of PRI in this box... |
| 17:34.43 | ManxPower | idespinner, I guess I could go experiment for you. |
| 17:34.54 | ManxPower | I imagine it means span 1 |
| 17:35.00 | idespinner | I aswell.... |
| 17:35.14 | imox1234 | [TK]D-Fender: ok i found this thanks :D |
| 17:35.26 | idespinner | ManxPower, just unplugged loopbacks on ports 7 and 8... messages seemed to have went away... |
| 17:36.02 | iscario | thx [TK]D-Fender |
| 17:36.11 | imox1234 | [TK]D-Fender:which module i have to include ? |
| 17:36.26 | Brian_H | [TK]D-Fender, app log http://pastebin.com/eHvkqRbv boot log http://pastebin.com/gaMCZP7k this is right after typing reload in the asterisk console |
| 17:39.01 | *** join/#asterisk Arsenick (~y@69.70.231.230) |
| 17:43.42 | *** join/#asterisk hfb (~hfb@pool-98-112-146-142.lsanca.dsl-w.verizon.net) |
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| 17:46.07 | imox1234 | i need the cdr_addon_mysql.so or ? |
| 17:46.12 | imox1234 | i dont have this odule |
| 17:46.12 | imox1234 | <PROTECTED> |
| 17:46.21 | imox1234 | but i have installed the asterisk addons |
| 17:46.24 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 17:48.52 | saisoma | imox1234, did you do a make menuselect when you were installing the asterisk addons to ensure that the cdr mysql piece was going to install? |
| 17:49.50 | imox1234 | yes i make menuselect. but by the cdr_addon are XXX and i cant change this |
| 17:50.02 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 17:50.13 | [TK]D-Fender | imox1234: Because you are missing PRE-REQUISITES |
| 17:50.15 | saisoma | imox1234, what OS are you on? you will need the mysql and mysql-devel packages |
| 17:50.20 | [TK]D-Fender | imox1234: which are clearly listed below |
| 17:50.33 | imox1234 | i have centos 5.4 |
| 17:50.42 | imox1234 | and i have mysql and mysql-devel |
| 17:50.42 | leifmadsen | imox1234: you also need to re-run ./configure after you install deps |
| 17:51.01 | imox1234 | yes I did |
| 17:51.06 | saisoma | leifmadsen, is def correct imox1234 |
| 17:51.08 | leifmadsen | then you're missing a dep still |
| 17:51.57 | imox1234 | and what is PRE-REQUISITES ? |
| 17:52.13 | saisoma | http://www.merriam-webster.com/dictionary/prerequisite |
| 17:52.22 | [TK]D-Fender | imox1234: "stuff you need to compile it" |
| 17:52.29 | imox1234 | ahh ok i have this |
| 17:52.30 | imox1234 | ;) |
| 17:52.37 | imox1234 | i have complied asterisk |
| 17:52.38 | [TK]D-Fender | imox1234: Can't bake a cake without the ingredients |
| 17:52.47 | *** part/#asterisk mboeru (~zen@thpallady.net.hostway.ro) |
| 17:52.52 | [TK]D-Fender | imox1234: Well you need OTHER shit to compile addons with MySQL support |
| 17:53.30 | imox1234 | [TK]D-Fender: WHICH ???? |
| 17:53.45 | leifmadsen | look in config.log |
| 17:53.55 | imox1234 | ok |
| 17:53.55 | imox1234 | thx |
| 17:57.14 | imox1234 | hmm now i have installed this all, mysql-devel was not installed. but the module will not load too |
| 17:58.47 | saisoma | imox1234, haveyou re-ran configure since mysql-devel installed? |
| 17:58.59 | imox1234 | yes |
| 17:59.15 | imox1234 | an restart asterisk :D |
| 17:59.24 | saisoma | so you then ran make menuselect for asterisk-addons and cdr mysql was there? |
| 17:59.35 | imox1234 | now a * |
| 17:59.39 | imox1234 | and i have installed all |
| 17:59.42 | saisoma | k |
| 18:00.23 | saisoma | imox1234, is this: load => cdr_addon_mysql.so |
| 18:00.26 | saisoma | in your modules.conf? |
| 18:00.32 | imox1234 | yes |
| 18:01.32 | saisoma | imox1234, if you run this from the command line |
| 18:01.32 | saisoma | rasterisk -x "module show"|grep mysql |
| 18:01.41 | saisoma | does it show cdr_addon_mysql.so? |
| 18:02.09 | imox1234 | dr_addon_mysql.so MySQL CDR Backend 0 |
| 18:02.09 | imox1234 | app_addon_sql_mysql.so Simple Mysql Interface 0 |
| 18:02.09 | imox1234 | res_config_mysql.so MySQL RealTime Configuration Driver 0 |
| 18:02.18 | saisoma | ok, so the module is loaded |
| 18:02.18 | saisoma | :) |
| 18:02.22 | saisoma | why do you think it isn't? |
| 18:02.46 | imox1234 | cdr show status |
| 18:02.57 | imox1234 | are only csv and cdr-costum |
| 18:03.01 | imox1234 | or its right ? |
| 18:03.37 | saisoma | no, i show mysql under registered backends |
| 18:03.41 | saisoma | on my system |
| 18:04.01 | saisoma | have you setup cdr_mysql.conf, created the database and such? |
| 18:04.09 | saisoma | setup permissions for the mysql user? |
| 18:04.11 | imox1234 | yes |
| 18:04.44 | imox1234 | how can i check if asterisk can connect to my database ? |
| 18:05.05 | saisoma | is it on the same server or another server? |
| 18:05.05 | leifmadsen | look at your database connection log? |
| 18:05.13 | saisoma | or test it from the cli. |
| 18:05.23 | imox1234 | how ? |
| 18:05.50 | saisoma | mysql -u<username> -h<hostname> -p <databasename> |
| 18:05.57 | saisoma | it will prompt for the password |
| 18:06.01 | saisoma | if you can get in |
| 18:06.03 | saisoma | and do a |
| 18:06.05 | saisoma | show tables; |
| 18:06.16 | saisoma | then you should be able to query via * |
| 18:06.30 | saisoma | but you'll need to double check your perms for write access, etc |
| 18:06.59 | imox1234 | and now when i call asterisk will wirte it in the mysql table ? |
| 18:07.28 | saisoma | imox1234, i can't say that will fix everything, but it's a step in the right direction |
| 18:07.39 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 18:07.54 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:08.20 | imox1234 | ?? |
| 18:08.45 | imox1234 | how can i test from the asterisk CLI if the asterisk connect to mysql ? |
| 18:08.49 | saisoma | imox1234, did you test your mysql connection? |
| 18:08.56 | saisoma | mysql -u<username> -h<hostname> -p <databasename> |
| 18:08.59 | imox1234 | my connecten work |
| 18:09.10 | saisoma | that's not from the cli, but it's from the server's cli |
| 18:09.17 | imox1234 | yes |
| 18:09.20 | imox1234 | its work |
| 18:09.37 | imox1234 | and how can test if asterisk work ? |
| 18:09.48 | saisoma | imox1234, ok. make a call through the system and see if it's recorded |
| 18:09.55 | imox1234 | ok # |
| 18:09.57 | imox1234 | ;) |
| 18:10.54 | *** join/#asterisk theshadow (~xguzman@173-14-11-29-Colorado.hfc.comcastbusiness.net) |
| 18:18.49 | *** join/#asterisk mkad (~mkad@169-202.surfsnel.dsl.internl.net) |
| 18:18.51 | mkad | Hi |
| 18:19.15 | theshadow | I'm trying to create a sip trunk between my asterisk box and Junction Networks I can't seem to get through the following is what my config and logs look like yes is shows registered. http://pastebin.com/2a9Aw2GS Any help would be greatly appreciated |
| 18:19.18 | mkad | When I want to shape outgoing traffic is it better to do it as egress on ISP interface or as ingress or local interfaces ? |
| 18:19.26 | mkad | I mean VoIP traffic |
| 18:19.41 | mkad | and which qdisc is best for SIP |
| 18:23.11 | [TK]D-Fender | theshadow: Go PB a call with SIP DEBUG enabled |
| 18:24.08 | *** join/#asterisk stix_ (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 18:29.27 | *** join/#asterisk imox1234 (~imox1234@p4FC5C27B.dip0.t-ipconnect.de) |
| 18:30.01 | imox1234 | hello, i have set the cdr_mysql settings and restart my asterisk but now i can run asterisk ? |
| 18:30.21 | devmod | I think I saw someone saying there was a flickering issue with etherpad on firefox. has it been fixed? |
| 18:30.31 | devmod | wrong chan :P |
| 18:30.41 | imox1234 | what can i do :-) |
| 18:32.47 | [TK]D-Fender | imox1234: Go run asterisk |
| 18:33.25 | imox1234 | dont work |
| 18:35.25 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 18:36.30 | [TK]D-Fender | imox1234: Neither does your description |
| 18:37.09 | imox1234 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:37.09 | imox1234 | , |
| 18:38.17 | ariel_ | it appears to not be running. Try asterisk -vvvgc |
| 18:39.36 | imox1234 | [May 3 22:38:31] WARNING[29837]: config.c:1107 process_text_line: parse error: No category context for line 14 of /etc/asterisk/cdr_mysql.conf |
| 18:40.57 | [TK]D-Fender | imox1234: Go fix your config |
| 18:41.06 | imox1234 | but what is wrong :D |
| 18:41.42 | [TK]D-Fender | imox1234: LINE 14 |
| 18:42.02 | [TK]D-Fender | imox1234: "No category context" <- go read the SAMPLE config and see what heading you're missing |
| 18:43.54 | imox1234 | ok now work but i dont get CDR's in my mysql database |
| 18:45.21 | imox1234 | Not currently connected to a MySQL server. |
| 18:45.46 | imox1234 | but the cdname username password all right |
| 18:46.01 | [TK]D-Fender | imox1234: Did you jsut place a call? |
| 18:46.08 | imox1234 | yes |
| 18:47.04 | imox1234 | mom i will check my settings :d |
| 18:48.34 | imox1234 | hmm all right |
| 18:48.37 | imox1234 | but dont connect |
| 18:49.57 | [TK]D-Fender | imox1234: Go prove that you can conenct with the supplied user & pass |
| 18:50.10 | imox1234 | i can connect |
| 18:50.14 | imox1234 | with this user |
| 18:50.14 | [TK]D-Fender | imox1234: and that the datase & tables are set up right |
| 18:50.18 | [TK]D-Fender | imox1234: Show us |
| 18:50.31 | imox1234 | how show you ;) ? |
| 18:51.07 | [TK]D-Fender | imox1234: PASTEBIN |
| 18:51.15 | [TK]D-Fender | ~pb |
| 18:51.16 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:51.24 | imox1234 | yes i now |
| 18:51.24 | imox1234 | know |
| 18:51.40 | imox1234 | but what ? |
| 18:51.56 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 18:52.50 | imox1234 | [TK]D-Fender: what should i past ? |
| 18:52.51 | [TK]D-Fender | imox1234: Show us your configs and connections attempt, and what * CLI shows on load, etc |
| 18:53.04 | imox1234 | ok |
| 18:56.29 | imox1234 | http://pastebin.com/f6WTeWWk, |
| 18:56.29 | imox1234 | |
| 18:56.30 | imox1234 | http://pastebin.com/f6WTeWWk |
| 18:57.56 | *** join/#asterisk megalomano (~klonstein@38.124.169.126) |
| 18:58.16 | megalomano | hya people |
| 18:59.34 | vader-- | are you any of you guys consulants who do asterisk installs? |
| 18:59.57 | imox1234 | member:%5BTK%5DD-Fender: whats wrong :-) ? |
| 19:01.20 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:01.46 | [TK]D-Fender | imox1234: Where is the proof of you connecting with that user? showing us the tables? Where is the ERROR you said you were getting? |
| 19:01.54 | [TK]D-Fender | vader--: Plenty of us |
| 19:03.06 | megalomano | i have some doubts about the caller id , i wish to customize this variable , i.e , if the caller # is 65464 ,the softphone shows "lolo" |
| 19:03.15 | imox1234 | member:%5BTK%5DD-Fender: can you say me what i have to write in the terminal. i only connect with mysqladministrator and this work. but there dont have any output log |
| 19:03.50 | [TK]D-Fender | imox1234: mysql <- at CLI |
| 19:04.00 | imox1234 | what ? |
| 19:04.06 | [TK]D-Fender | imox1234: mysql <- at CLI |
| 19:04.08 | imox1234 | mysql and what ? |
| 19:04.22 | [TK]D-Fender | and connect with your user & pass |
| 19:07.03 | imox1234 | and how ? |
| 19:07.11 | imox1234 | mysql -u USERNAME -p PASSWORD ? |
| 19:07.30 | imox1234 | sorry i always use phpmyadmin |
| 19:08.01 | [TK]D-Fender | imox1234: We aren't here to teach you MySQL |
| 19:08.07 | [TK]D-Fender | imox1234: Try #mysql |
| 19:08.22 | imox1234 | ok i login ? and what should you past ? |
| 19:08.25 | imox1234 | sorry |
| 19:09.52 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
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| 19:10.23 | imox1234 | [TK]D-Fender: hello ?? |
| 19:11.24 | [TK]D-Fender | imox1234: You never showed us the actual errors. You don't seem to be capable of even Google-ing how to conenct to your own MySQL DB with the proper user to prove that its even in good shap to be used. We can't help you until you do |
| 19:12.59 | megalomano | some help |
| 19:14.31 | [TK]D-Fender | megalomano: "core show function CALLERID" |
| 19:14.37 | [TK]D-Fender | megalomano: "core show application set" |
| 19:14.42 | [TK]D-Fender | megalomano: "core show application gotoif" |
| 19:15.59 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:151f:c3f8:13cd:37eb) |
| 19:17.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.17 | megalomano | ÂAdler says: |
| 19:18.17 | megalomano | <PROTECTED> |
| 19:18.18 | megalomano | <PROTECTED> |
| 19:18.18 | megalomano | <PROTECTED> |
| 19:18.18 | megalomano | <PROTECTED> |
| 19:18.18 | megalomano | <PROTECTED> |
| 19:18.37 | megalomano | [TK]D-Fender: thanks |
| 19:19.08 | *** join/#asterisk kartik (~koolkarti@117.199.121.144) |
| 19:19.48 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:21.43 | [TK]D-Fender | megalomano: And please don't flood the channel |
| 19:22.01 | imox1234 | [TK]D-Fender: whats you problem ? my user can connect to mysql and my database are right |
| 19:22.05 | imox1234 | this is not the problem |
| 19:22.30 | *** join/#asterisk ccesario_ (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 19:22.31 | [TK]D-Fender | imox1234: You haven't even shown us the error, or proven that your database is there, that the tables are right or anything |
| 19:23.03 | imox1234 | [TK]D-Fender: which error ? asterisk said only not connected to mysql database |
| 19:23.12 | imox1234 | i dont know WHICH ERROR ? |
| 19:23.16 | [TK]D-Fender | imox1234: Not in your pastebin it didn't |
| 19:23.24 | imox1234 | what ? |
| 19:24.03 | imox1234 | http://pastebin.com/40Seni7e |
| 19:24.51 | [TK]D-Fender | imox1234: module reload cdr_mysql.so |
| 19:25.34 | imox1234 | same too |
| 19:26.27 | [TK]D-Fender | imox1234: PASTEBIN |
| 19:27.00 | imox1234 | http://pastebin.com/Dpmh31B4 |
| 19:27.30 | [TK]D-Fender | imox1234: do and uload then a reload of it |
| 19:28.43 | imox1234 | http://pastebin.com/769T6rvr |
| 19:31.15 | [TK]D-Fender | imox1234: now go prove that the user can connect via the CLI app and that the DB is there and it has rights to it |
| 19:33.37 | imox1234 | i use phpmyadmin |
| 19:33.44 | imox1234 | and it works |
| 19:33.49 | [TK]D-Fender | imox1234: SHOW US |
| 19:33.55 | imox1234 | HOW |
| 19:34.00 | imox1234 | i can give you my login |
| 19:34.04 | imox1234 | for my phpmyadmin |
| 19:34.19 | [TK]D-Fender | imox1234: imagebin.ca |
| 19:36.35 | imox1234 | http://imagebin.ca/view/Uuhrg4WR.html |
| 19:36.54 | imox1234 | http://imagebin.ca/view/CqFtYSY4.html |
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| 19:38.41 | imox1234 | [TK]D-Fender: its ok ? or you need more ? |
| 19:38.56 | vader-- | i was just wondering what the average cost would be for a small 4 ip phone/ 2 FXO setup |
| 19:39.42 | [TK]D-Fender | vader--: price of PC + price of card + price of 4 phones. |
| 19:39.58 | vader-- | they have the PC And the line card and the phones |
| 19:40.03 | vader-- | im just wondering configuration cost |
| 19:40.08 | imox1234 | [TK]D-Fender: ?? |
| 19:40.19 | vader-- | for someone to do it |
| 19:40.20 | [TK]D-Fender | imox1234: That user has no priveleges <- |
| 19:40.31 | [TK]D-Fender | imox1234: assigned priveleges = BLANK |
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| 19:41.01 | imox1234 | ohhh sorry wrong screenshot |
| 19:41.42 | vader-- | wasn't sure how people priced it |
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| 19:53.16 | devmod | any recommendations on best AMI bindings for php or ruby? |
| 19:55.38 | paulc | Make me an offer and I'll tell you if it's too low ;-) |
| 19:57.06 | *** join/#asterisk acxty (~acxty@201.220.136.118) |
| 19:57.47 | acxty | Hi guys, I am register to the provider. But when I receive a call it says that my extension was not found |
| 19:58.51 | paulc | acxty: what context does the inbound call land in, and what extension is being presented? |
| 20:00.20 | acxty | context [110] and the extension configure is 110 |
| 20:00.53 | acxty | I made some test with xlite and it can receive the phone calls |
| 20:03.40 | [TK]D-Fender | acxty: that is a SIP PEER, not an EXTENSION |
| 20:05.21 | paulc | acxty: your context should be something like [in-provider-A] and have an extension 110 in there |
| 20:05.46 | paulc | or by "extension" do you mean "DID" - the number that the provider is sending you (public phone number, long, not a short extension number?) |
| 20:06.02 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 20:09.16 | [TK]D-Fender | paulc: Its the error he gets when he forgets to actually creat his dialplan properly |
| 20:13.13 | paulc | Yes.. I hear you.. |
| 20:13.22 | paulc | a low quality day at the day job, and out there in the real world too it seems |
| 20:13.49 | paulc | dreams of alternative options and fidgets a bit |
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| 20:29.25 | drift- | i've got issue with 2 phone lines disconnecting when they want to... 1 call comes in it connects then disconnects 2 nd call comes in say hello then disconnects 3rd time it stays connected |
| 20:29.39 | drift- | all on iax2 with voips.ms |
| 20:34.15 | drift- | anyone heh |
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| 21:04.53 | voxter | I nee to be able to send "Supported: 100rel" when i dial a nortel SBC from asterisk. Anyone know if theres an option to do this? |
| 21:05.50 | *** join/#asterisk smooth_penguin (~smoothp@122.182.1.135) |
| 21:05.52 | wdoekes2 | SIPAddHeader? |
| 21:06.03 | voxter | the header is already there, i need to amend it. |
| 21:06.11 | *** part/#asterisk mnick86 (~Matthias@whhem00016.cip.uni-regensburg.de) |
| 21:06.32 | wdoekes2 | sipaddheader inserts the header. editing a header is not easily done, afaik |
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| 21:08.03 | wdoekes2 | amend #define SUPPORTED_EXTENSIONS "replaces, timer" in the source? :) |
| 21:08.46 | voxter | Wow. I hope thats not the actual fix! lol. |
| 21:09.02 | voxter | Considering sip interop is such a bitch most of the time, you think that'd be configurable. |
| 21:09.48 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 21:09.59 | wdoekes2 | you could sipaddheader the complete thing and hope that the nortel reads the first |
| 21:10.06 | *** part/#asterisk RockyMountains (~RockyMoun@b538D.static.pacific.net.au) |
| 21:11.51 | voxter | hmm. it seems that the Supported: is something established in peer dialog. |
| 21:14.05 | wdoekes2 | the sip_options[] indicate that chan_sip does not support the 100rel PRACK stuff.. that may be a reason that it's not listed in the supported_extensions ;) |
| 21:14.38 | voxter | Are you seeing this in chan_sip.c or somewhere else? |
| 21:14.44 | wdoekes2 | chan_sip.c indeed |
| 21:15.29 | voxter | ok, so asterisk simply doesnt support PRACK (100rel) ok. |
| 21:16.26 | wdoekes2 | correct.. so your initial "question" was bad |
| 21:17.03 | wdoekes2 | you don't want to set a header, you want a feature |
| 21:17.13 | voxter | Right. The bitch of it is my ITSP (Nortel SBC) says i cant call certain numbers because we are not sending 100rel. The question now will be, if i arbitrarily send 100rel, will that simply make their end work, or will it break things.. |
| 21:17.20 | voxter | I'll have to investigate that |
| 21:17.24 | russellb | no. |
| 21:17.32 | russellb | we plain don't support it |
| 21:17.53 | russellb | put asterisk behind kamailio or something that does support it |
| 21:17.57 | russellb | that's the best solution I think |
| 21:18.10 | voxter | russellb: im curious, if i did send 100rel to them, if their switch will then respond in a way that asterisk wont be able to adhere to, or are they rejecting my call simply on the basis that 100rel is not in the header? |
| 21:19.06 | russellb | if they expect that to work, the call will fail anyway |
| 21:19.13 | russellb | it's not just a little diddy in a header |
| 21:19.19 | voxter | gotcha. |
| 21:19.20 | russellb | it means additional messaging in the call |
| 21:19.30 | russellb | 100rel - reliable transmission of provisional responses |
| 21:19.36 | voxter | Whats messed up is that its only required for me to call "some" toll free numbers. |
| 21:19.39 | voxter | go figure. |
| 21:20.45 | wdoekes2 | mm.. could be they want ACK's on early media (like to tell you how expensive/free the call is, before starting the billing) |
| 21:20.55 | dohd | does anyone know phones that have 't9 like dictionary lookups', like e.g. avaya has? |
| 21:22.52 | dohd | eeh, directory I mean of course |
| 21:25.44 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 21:28.02 | [TK]D-Fender | dohd: The phone isn't the part that has the brain. |
| 21:28.11 | [TK]D-Fender | dodthis is done PBX-side |
| 21:29.40 | *** join/#asterisk aceio (~c2cbd7fe@gateway/web/freenode/x-arilrntcanxvfqbm) |
| 21:29.50 | dohd | yeah |
| 21:30.00 | dohd | but most phones I see have horrible interfaces for them |
| 21:30.12 | dohd | like the polycom I played with, you have to spell the name you are looking for |
| 21:30.25 | dohd | so c is (3x pressing the 2), etc |
| 21:30.57 | dohd | I can live with having to distribute directory information |
| 21:31.08 | dohd | or create a seperate app for it perhaps |
| 21:32.34 | dohd | I'm trying to pick telephones for an asterisk setup to replace the stuff they currently use |
| 21:32.46 | dohd | and they have phones with a seperate keyboardish thing |
| 21:33.08 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 21:33.10 | dohd | very old phones, but I'm afraid they'll complain if they don't get something good enough in return |
| 21:33.25 | *** part/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 21:37.35 | [TK]D-Fender | dohd: t9 dial by name is boring dialplan stuf |
| 21:38.09 | [TK]D-Fender | dohd: with polycom you aren't actually turning text into numbers, you are LITERALLY dialing alpha chars |
| 21:38.33 | dohd | well, I wasn't thinking of vanity dialing |
| 21:39.00 | dohd | and I was talking of the dictionary lookup stuff |
| 21:39.36 | dohd | dial by name wouldn't work if you had 10 jansen's (or smith or whatever) |
| 21:41.12 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 21:41.36 | dohd | on the avaya you'd go to the directory, press 526 and it would let you scroll through all the directory entries that matched [jkl][abc][mno] |
| 21:41.42 | [TK]D-Fender | dohd: It would if it prompts you to clarify the ambiguity |
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| 21:46.11 | dohd | hmm... I'll give it some more thought... |
| 21:46.37 | dohd | I've considered dial by name, but it didn't make me feel confident yet about being accepted as a solution |
| 21:47.03 | *** join/#asterisk Ta^3 (~tacvbo@189.146.183.88) |
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| 22:00.41 | devdvd | hey all, im setting up asterisk 1.4 using the agentcallbacklogin function. What i want to do is have the agent be dynamic, so if they dial in from their non-sip phone and enter user,pass,ext (ex. 100,1234,100) then it will log that user in as agent 100 on extension 100 then extension 100 will point to wherever the agent is (sip phone or non-sip phone) |
| 22:01.40 | [TK]D-Fender | devok, it already does this |
| 22:01.50 | devdvd | is there a way to get the number the agent called in from (i know it is set in a variable) but from the looks of it the variable is created at runtime |
| 22:02.09 | devdvd | TK, i know it does part of it..i guess im missing something |
| 22:02.32 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net) |
| 22:02.34 | [TK]D-Fender | devdvd: I dunno... maybe the CALLERID?! |
| 22:02.37 | devdvd | the part that is confusing to me is the "new extension" |
| 22:02.49 | devdvd | no , thats not what i mean TK |
| 22:03.18 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:04.21 | devdvd | what im saying is when an agent calls in, i want them to enter their user, pass and extension, and i want them to be able to enter the same extension every time, and it will basically just find them (ie Dial(SIP/100) will change di Dial(SIP/trunk/1234567890) |
| 22:05.26 | devdvd | i dont even know if such a thing would be possible, just one of these "it would be nice to do" type of things |
| 22:05.37 | [TK]D-Fender | devWell you tell Agentcallbacklogin where to call.... |
| 22:05.55 | devdvd | == Setting global variable 'AGENTBYCALLERID_1234567890' to '869' |
| 22:06.03 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:06.05 | devdvd | is the runtime variable that gets set, but it looks like thats dynamic |
| 22:06.18 | devdvd | ok |
| 22:06.21 | devdvd | i got ya tk |
| 22:06.25 | ruben23 | hi anyone have tried dialing autralian number..? |
| 22:06.26 | devdvd | and that being the case |
| 22:06.29 | devdvd | i cant do what i want |
| 22:06.53 | devdvd | because i could enter the callback number as 1234567890 but it still wouldnt know the trunk |
| 22:06.54 | [TK]D-Fender | devdvd: which is? |
| 22:07.12 | devdvd | when an agent calls in, i want them to enter their user, pass and extension, and i want them to be able to enter the same extension every time, and it will basically just find them (ie Dial(SIP/100) will change di Dial(SIP/trunk/1234567890) |
| 22:07.17 | [TK]D-Fender | devdvd: I don't think you get it.... |
| 22:07.27 | devdvd | TK your probably right |
| 22:07.49 | [TK]D-Fender | devdvd: * dials a LOCAL CHANNEL in the context targeted and ANY dialplan you want can be there WAITING for it. |
| 22:08.53 | devdvd | hmm...i think youve given me an idea :) thanks :) |
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| 22:40.24 | Brian_H | I just installed a polycom 330 phone, I can get the phone to register with asterisk, however if I leave a voicemail for the extension it sends the phone into a reboot |
| 22:40.44 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
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| 22:43.19 | davidstrauss | Brian_H: Check your sip.cfg. It's likely there's an invalid configuration for message waiting. |
| 22:44.01 | davidstrauss | Brian_H: That will crash the phone when the SIP server instructs it to show the message waiting indicator LED/sound. |
| 22:44.49 | Brian_H | ok that makes me feel a little better :) |
| 22:44.56 | Brian_H | I'll try playing with that |
| 22:45.01 | Brian_H | goes to look at the admin guide |
| 22:45.10 | davidstrauss | Brian_H: try a stock sip.cfg |
| 22:45.27 | davidstrauss | Brian_H: or pastebin you sip.cfg (and other xml config) for me |
| 22:45.33 | Brian_H | this is the one that came with the latest firmware/boot rom :( |
| 22:45.39 | Brian_H | I'll paste them though :D |
| 22:46.13 | davidstrauss | Brian_H: I have a whole office filled with 320 and 321 models. This exact thing happened to me when the MWI config was bad. |
| 22:46.30 | Brian_H | ok I really appreciate this |
| 22:46.31 | Brian_H | uploading now |
| 22:47.02 | davidstrauss | Brian_H: Also, a diff versus the contents of the zip from Polycom would be helpful. |
| 22:47.23 | davidstrauss | Brian_H: And the MAC addr of one of the phones haven a problem alongside its config |
| 22:48.19 | Brian_H | pastebin is fighting me |
| 22:48.26 | Brian_H | can I email you the files maybe? |
| 22:48.28 | davidstrauss | Brian_H: i use pastie |
| 22:48.34 | *** part/#asterisk andreas-- (~andy@unaffiliated/slacky) |
| 22:49.49 | Brian_H | http://pastie.org/944375 |
| 22:49.58 | Brian_H | thats mac.cfg |
| 22:50.18 | davidstrauss | Brian_H: I'm suspicious of the avanphone201.cfg, |
| 22:50.26 | davidstrauss | kavanphone201.cfg |
| 22:50.37 | Brian_H | http://pastie.org/944376 |
| 22:51.31 | Brian_H | won't let me post the sip.cfg too big |
| 22:52.01 | davidstrauss | Brian_H: post the diff versus the stock sip.cfg, if any |
| 22:52.09 | davidstrauss | Brian_H: I really need to see kavanphone201.cfg |
| 22:52.24 | Brian_H | http://pastie.org/944376 |
| 22:52.42 | Brian_H | ^ thats it |
| 22:53.02 | davidstrauss | if that's kavanphone201.cfg, then what's the mac-specific phone config? |
| 22:53.30 | Brian_H | a pointer to that file, I followed a howto on voip.net or something |
| 22:53.43 | Brian_H | http://pastie.org/944375 |
| 22:53.49 | Brian_H | thats the mac.cfg |
| 22:54.36 | Brian_H | getting diff now |
| 22:55.09 | davidstrauss | Brian_H: OK, then I also need to see server.cfg and phone1.cfg |
| 22:55.41 | Brian_H | those are the defaults as well, but I will post them |
| 22:57.10 | davidstrauss | Brian_H: Is the diff vs. the stock sip.cfg empty? |
| 22:59.22 | Brian_H | davidstrauss, http://pastie.org/944390 thats the diff |
| 22:59.29 | Brian_H | I changed the timezone offset |
| 22:59.50 | davidstrauss | < <MESSAGE_WAITING se.pat.misc.1.name="message waiting" se.pat.misc.1.inst.1.type="chord" se.pat.misc.1.inst.1.value="1" se.pat.misc.1.inst.2.type="chord" se.pat.misc.1.inst.2.value="2" se.pat.misc.1.inst.3.type="chord" se.pat.misc.1.inst.3.value="1" /> |
| 22:59.50 | davidstrauss | --- |
| 22:59.50 | davidstrauss | > <MESSAGE_WAITING se.pat.misc.1.name="message waiting" /> |
| 22:59.53 | davidstrauss | you can't do that |
| 23:00.14 | davidstrauss | just change "chord" to "silent" in each instance instead of ripping out the types and values |
| 23:00.36 | Brian_H | oh, man that howto I followed apparently is not good :( |
| 23:02.21 | davidstrauss | <MESSAGE_WAITING se.pat.misc.1.name="message waiting" se.pat.misc.1.inst.1.type="silent" se.pat.misc.1.inst.1.value="1" se.pat.misc.1.inst.2.type="silent" se.pat.misc.1.inst.2.value="2" se.pat.misc.1.inst.3.type="silent" se.pat.misc.1.inst.3.value="1"/> |
| 23:02.27 | davidstrauss | that is what we use to silence it |
| 23:02.49 | Brian_H | ok going to put the default back and change only the tzsetting and your setting |
| 23:03.10 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:03.32 | davidstrauss | Brian_H: you probably need that digitmap |
| 23:03.56 | Brian_H | thanks for your help with this, I really appreciate it |
| 23:05.02 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:06.14 | *** join/#asterisk gospch (~gospch@p5088F4FB.dip.t-dialin.net) |
| 23:08.28 | Brian_H | davidstrauss, is there a way to clean up these .cfg files, short of manually, so they look a little nicer? it drives me nuts that they are like that |
| 23:08.44 | Brian_H | rebooting the phone now with the new sip.cfg in place |
| 23:09.06 | *** join/#asterisk adurotec (~davidc@64.245.227.214) |
| 23:09.06 | davidstrauss | Brian_H: Typically, you should be putting the overrides into a separate file. |
| 23:09.17 | *** part/#asterisk adurotec (~davidc@64.245.227.214) |
| 23:09.20 | davidstrauss | Brian_H: I forget whether you need to load the overrides before or after sip.cfg. |
| 23:09.40 | Brian_H | from the howto, before, but that howto apparently was crap :p |
| 23:09.55 | davidstrauss | Brian_H: also, you're using a funny file scheme. that's why i was confused about the mac.cfg. |
| 23:10.06 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:10.24 | Brian_H | if you don't mind, what config are you using |
| 23:10.32 | Brian_H | again the howto suggested that |
| 23:10.34 | Brian_H | :/ |
| 23:11.36 | davidstrauss | Brian_H: You should generally have one 0000...snip...00.cfg file that specifies the file paths and basic config. Phones will load a file with their MACADDR-phone.cfg on their own. |
| 23:12.10 | davidstrauss | Brian_H: Phones read both 000000000000.cfg and MACADDR-phone.cfg to determine their config. |
| 23:12.21 | sawgood | Is it 'do-albe' with Asterisk 1.6.2.x to have Asterisk make bulk outbound calls (and when the phone is answered by the end user) to have a generic greeting played .... |
| 23:12.41 | sawgood | What is the term for this activity called? |
| 23:12.48 | davidstrauss | sawgood: yes |
| 23:12.54 | davidstrauss | sawgood: and the term for that is "annoying" |
| 23:13.00 | Brian_H | lol |
| 23:13.02 | sawgood | I agree ... |
| 23:13.12 | sawgood | I have a client wanting me to do this for them over a SIP trunk |
| 23:13.17 | Brian_H | davidstrauss, :D phone DID NOT reboot |
| 23:13.18 | sawgood | a call campaign is what I call it |
| 23:13.19 | Brian_H | :) |
| 23:13.19 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 23:14.02 | sawgood | So, I could do some reading about it online, what keywords should I use? |
| 23:14.57 | Brian_H | davidstrauss, so is it recommended to just put all these customizations in the 00mac00.cfg file? |
| 23:15.08 | sawgood | I guess basically it is a pre-quailifed leads out to phone numbers which have shown an interest in the product (to remind them they can have help if they call back) |
| 23:15.16 | davidstrauss | Brian_H: If the customizations are for your complete system, no |
| 23:16.02 | davidstrauss | Brian_H: You should have something like myoffice.cfg listed in 000000000000.cfg |
| 23:16.06 | *** join/#asterisk nitram (foo@superblob.com) |
| 23:16.25 | davidstrauss | Brian_H: only per-phone customization should go in MAC-phone.cfg |
| 23:16.37 | davidstrauss | (btw, MAC is always the mac addr of the specific phone) |
| 23:17.41 | davidstrauss | sawgood: You're looking for the Originate command from the manager interface, btw |
| 23:18.04 | davidstrauss | sawgood: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 23:18.06 | devmod | Lets say I have a video call between two endpoints going on, can I originate a voice call and make the audio part of the existing call? kinda like a conference on the fly? |
| 23:18.48 | davidstrauss | devmod: merge separate audio and video calls? |
| 23:20.29 | devmod | david, the existing videocall stays as it is, but the audio get merged into a "conf call" |
| 23:21.09 | davidstrauss | devmod: so there's an existing conf call? |
| 23:21.50 | *** join/#asterisk KNERD (~KNERD@129.113.44.146) |
| 23:21.57 | Brian_H | davidstrauss, before with my crazy sip.cfg file, (don't know if that was the cause) if I typed "reload" in my asterisk console it would reboot the phone, would those settings potentially be the cause? |
| 23:22.06 | devmod | davidstrauss, hopefully not. basically an agent receives a video call, and then the agent calls a third party through a voice call only. I want to bridge the audio streams like in a conference |
| 23:22.10 | Brian_H | doesn't appear to be doing it now |
| 23:22.25 | davidstrauss | Brian_H: reload causes asterisk to send fresh message waiting notifications |
| 23:22.39 | Brian_H | ahh that would do er then |
| 23:22.41 | devmod | davidstrauss, I see how i could create a conference call for every one of these instances but that doesnt sound right for some reason... |
| 23:22.41 | Brian_H | man you're pro |
| 23:22.43 | Brian_H | thanks! |
| 23:22.49 | davidstrauss | ;-) |
| 23:23.16 | davidstrauss | devmod: your problem is still a little vague to me |
| 23:23.36 | Brian_H | davidstrauss, are you located near portland? |
| 23:24.08 | davidstrauss | Brian_H: I live in Austin, TX. I will probably head to portland for CLS and maybe OSBridge |
| 23:24.21 | Brian_H | you should look me up :) I'll buy you a beer |
| 23:24.24 | devmod | davidstrauss, let me give you a little more context. A customer calls into a call center and connects to an agent using audio and video. Then I need to somehow bring a third party into this existing call , this third party would participate using audio only |
| 23:24.24 | davidstrauss | ;-) |
| 23:24.50 | davidstrauss | devmod: just have the end user initiate and bridge the call using his handset |
| 23:25.06 | davidstrauss | devmod: is the audio coming into a decent voip handset? |
| 23:25.14 | devmod | its a softphone |
| 23:27.02 | davidstrauss | devmod: http://www.voip-info.org/wiki/view/Asterisk+cmd+Bridge |
| 23:27.23 | davidstrauss | devmod: use the manager interface to originate the new call and then bridge the channels |
| 23:27.50 | davidstrauss | devmod: that may do the trick |
| 23:28.02 | devmod | davidstrauss, looks like it, gonna look into it. Thanks |
| 23:28.49 | davidstrauss | Brian_H: btw, sntp is best set for your polycoms via dhcp |
| 23:28.59 | *** join/#asterisk c0dyhi11 (~c0dyhi11@ip70-190-105-213.ph.ph.cox.net) |
| 23:29.50 | *** join/#asterisk cy3o3 (~cy@it.was.otherkids.net) |
| 23:29.51 | Brian_H | I have my ntp server pushed out via dhcp but the phones were still off |
| 23:30.32 | carrar | 1 hour off? |
| 23:30.42 | davidstrauss | Brian_H: off in what way? |
| 23:30.51 | Brian_H | off as if it were not being pushed out |
| 23:30.56 | davidstrauss | option ntp-servers north-america.pool.ntp.org; |
| 23:31.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:31.13 | davidstrauss | option time-offset -21600; # Fix for your non-DST time zone |
| 23:31.19 | Jumpie | anybody know if its possible to set a static ip address in an aastra phone's MAC.cfg file, instead of doing it on the phone ui? |
| 23:31.25 | Jumpie | im sure there is, but dunno syntax |
| 23:31.39 | Brian_H | davidstrauss, add that in the dhcp conf or in the sip.cfg? |
| 23:31.44 | Brian_H | I have that in the sip.cfg |
| 23:31.47 | davidstrauss | Brian_H: dhcp conf |
| 23:31.56 | davidstrauss | Jumpie: Is that file hosted on a server or on the phone? |
| 23:32.29 | davidstrauss | Jumpie: if that file is hosted on a provisioning server, you have a chicken/egg problem with your approach |
| 23:32.30 | Jumpie | server in /tftpboot |
| 23:32.45 | Jumpie | the phone already knows the tftp server ip |
| 23:32.48 | c0dyhi11 | ch040887 |
| 23:32.56 | Jumpie | and sends out a broadcast even if it has no config, can always find the server |
| 23:33.00 | Jumpie | thats why i love aastra phonse |
| 23:33.21 | davidstrauss | Jumpie: but it needs to have an ip to talk to the tftp server |
| 23:33.27 | Jumpie | basically the issue is...i am not locally at the phones, and not tryin tos tep the client into configuring it as such |
| 23:33.35 | Jumpie | davidstrauss no it doesnt |
| 23:33.50 | Jumpie | if i preemptively place that phones mac.cfg in the server..w.hich i have...the phone can find it |
| 23:34.01 | Brian_H | ok changed and rebooting phones to see if it works |
| 23:34.02 | Jumpie | but..its up now anyway, its on dhcp though |
| 23:34.12 | davidstrauss | Jumpie: how is the phone talking to the tftp server without an ip? |
| 23:34.22 | Jumpie | i want to statically assign ips to these phones now...so i can give customer a sheet of addresses/passwords |
| 23:34.29 | Jumpie | probably liek bootp or something |
| 23:34.31 | Jumpie | l2 broadcast |
| 23:34.38 | Jumpie | after all a mac address isnt a layer 3 thing : |
| 23:34.40 | Jumpie | :P |
| 23:34.57 | Jumpie | but..it only works that way IF i already have the MAC.cfg on the server...which i do |
| 23:35.00 | davidstrauss | Jumpie: just configure the dhcp server to hand out fixed addresses by mac |
| 23:35.03 | Jumpie | the phone is up now but its not static |
| 23:35.07 | sawgood | So, If I wanted to have my Asterik box 'blast' out a pre-recorded message to tell all the end points it dials what the latest status is with the project ... where I can read more about how to do this? |
| 23:35.16 | Jumpie | davidstrauss..the pbx isnt handing out the dhcp..the router is |
| 23:35.29 | davidstrauss | Jumpie: and you can't configure the router? |
| 23:35.33 | sawgood | A pre-recorded message telling people where to call to get more information |
| 23:35.36 | Jumpie | this is just some netgear |
| 23:35.40 | Jumpie | i dont think you can get that granular |
| 23:35.47 | Jumpie | if i was using a linux box or cisco sure |
| 23:35.50 | Jumpie | but ill check hold on |
| 23:35.53 | davidstrauss | Jumpie: almost every crappy router lets you reserve IPs by mac |
| 23:36.25 | carrar | Jumpie, why would they need static IP's? |
| 23:36.27 | Jumpie | i didnt know that..thought you could pretty much set scope and thast it |
| 23:36.38 | Jumpie | carrar they dont 'need' it really but, the customer may want to edit speeddial settings |
| 23:36.46 | davidstrauss | Jumpie: nah, plenty of $20 routers let you fix the ip by mac |
| 23:36.48 | Jumpie | i dont think he's going to want to trty to figure out what the ip is if its changed |
| 23:36.50 | carrar | You don't push speed dials to the phone? |
| 23:37.05 | Jumpie | carrar i tried...but it didnt take |
| 23:37.08 | Jumpie | i had to do it with webgui |
| 23:37.10 | Jumpie | plus...realize |
| 23:37.14 | Jumpie | this is a non very tech sav vy customer |
| 23:37.18 | Jumpie | its gotta be GUI GUI GUI for him |
| 23:37.23 | Jumpie | and yes davidstrauss i found it...good call |
| 23:37.27 | Jumpie | i didnt think it'd be on such a cheap router |
| 23:37.46 | p3nguin | I would I would have thought of that two-three days ago. *sigh* |
| 23:38.07 | p3nguin | s/would/wish/ |
| 23:42.13 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.26 | davidstrauss | Brian_H: I just requested a connection with you on linkedin |
| 23:43.00 | Brian_H | :) |
| 23:45.22 | Kobaz | wow |
| 23:45.23 | Kobaz | so |
| 23:45.40 | Kobaz | apparently avaya definity system 6 has broken callerid name support for pri |
| 23:46.30 | Kobaz | it's sending a malformed DISPLAY IE |
| 23:52.03 | davidstrauss | Kobaz: You're lucky. I'm having trouble getting any CID records out of this Definity unit: http://www.olay.com/boutique/definity/products/de1015 |
| 23:52.44 | Kobaz | so, adtran tech support has given me the info that the definity is botching up... i wonder if it's fixable |
| 23:53.05 | Kobaz | sangoma + libpri can suck out the name from the malformed message.. why can't adtran |
