| 00:03.43 | [TK]D-Fender | DropsOfSerenity: Guess that's how it works. It isn't the best documented feature ever.... |
| 00:04.16 | [TK]D-Fender | DropsOfSerenity: Change your logic and do the redial yourself in a single "try" |
| 00:04.33 | DropsOfSerenity | yeah, that may be best, and easiest |
| 00:04.40 | [TK]D-Fender | yup |
| 00:04.50 | MrRobert | thanks tk, found what i needed :) |
| 00:05.39 | [TK]D-Fender | MrRobert: Helps you you don't pin your own keywords over things ;) |
| 00:06.07 | MrRobert | Yes it helps to have the correct terminology for something you're looking for ;) |
| 00:09.23 | [TK]D-Fender | indeed |
| 00:44.26 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 00:59.07 | *** join/#asterisk elico (~Thunderbi@109.67.204.3) |
| 01:28.28 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 01:32.46 | *** join/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
| 01:33.01 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 01:35.09 | *** part/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
| 01:35.32 | *** join/#asterisk deo (~deo@58.71.19.178) |
| 01:37.05 | *** join/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
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| 02:06.28 | *** join/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
| 02:07.42 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 02:20.38 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 02:21.32 | *** join/#asterisk ELSPAZZZ (~Vince@76-242-98-61.uvs.livnmi.sbcglobal.net) |
| 02:21.44 | ELSPAZZZ | Good evening everyone |
| 02:22.30 | WIMPy | Good morning |
| 02:22.44 | ELSPAZZZ | Is anyone in here familiar with setting up a CISCO IP Phone 7960 with SIP Firmware for use with asterisk? I'm having a ton of problems and could use someone knowledgeable to tell me just what I'm fouling up. |
| 02:24.54 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:30.44 | ELSPAZZZ | The specific problem I appear to be having has to do with configuration of the provisioning files. I can't seem to find a straight forward idiots guide for it |
| 02:33.01 | tm1000 | ELSPAZZZ: If only a straight forward idiots guide existed for any cisco/polycom product we would all rejoice |
| 02:33.37 | tm1000 | ELSPAZZZ: http://www.pbxinaflash.com/community/index.php?threads/cisco-7914-using-sip.13234/ |
| 02:34.09 | tm1000 | thats for freepbx,,...,. but the configs are in there and can be deciphered easily |
| 02:41.45 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 02:45.23 | ELSPAZZZ | Thanks TM1000 going to give it a shot |
| 02:45.48 | ELSPAZZZ | Im starting to realize this Cisco gear is not very user friendly unless you have a support contract with those guys |
| 02:47.20 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 02:50.59 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-2-231-146.ph.ph.cox.net) |
| 02:53.22 | ELSPAZZZ | TM1000 that guide seems to be further along than I am. I still can't get the firmware on the phone let alone configured |
| 02:54.27 | ELSPAZZZ | I know I currently have P0S30100 and PC03C300 as the Load ID's in the firmware I have. But I can't tell what that actually MEANS in terms of what version of the firmware I have. Cisco makes reference to identifying what version of the firmware I have but they refer it differently when they talk about the images themselves. |
| 02:54.32 | ELSPAZZZ | I can't make heads or tails of the two |
| 03:06.38 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 03:18.25 | ELSPAZZZ | Jesus, no two people even can call the firmware the same damn thing |
| 03:18.50 | ELSPAZZZ | I have no idea what i'm even messing up to start to fix it |
| 03:29.55 | *** join/#asterisk engrxyz (~uiuihkhjk@host81-150-217-167.in-addr.btopenworld.com) |
| 03:38.35 | KNERD | BULD NATIVE and make CFLAGS="-march=k8 -msse4a" failed :-( |
| 03:47.27 | KNERD | where is the options listed for make? I am looking at asterisk docs and see nothing about compikle |
| 03:52.06 | *** join/#asterisk ideaman55 (~ideaman55@199.30.187.200) |
| 03:53.23 | ideaman55 | I've made some tweaks in my asterisk 1.8svn, and I broke it loading all the codecs. What config file does it parse that would break that part? |
| 04:01.42 | jpsharp | modules.conf |
| 04:02.53 | ELSPAZZZ | Anyone... Please, I've been working on this thing for about 5 hours. Curse the man who designed this stupid hardphone |
| 04:04.02 | ELSPAZZZ | All I can make it do is spit out DEFAULTING TO CM TO TFTP SERVER. The server tells me the phone is requesting a provisionong file using the SCCP protocol even though i'm trying to upgrade it to SIP. I can't find ANY decent guides. I just need someone to help me figure out what i'm doing wrong! |
| 04:04.17 | *** join/#asterisk Smirker (~x@124-171-226-166.dyn.iinet.net.au) |
| 04:04.27 | jpsharp | Which version of firmware are you running on it? |
| 04:04.42 | ELSPAZZZ | I can't even tell that. I know what the phone says |
| 04:04.43 | ELSPAZZZ | hang on |
| 04:05.05 | ELSPAZZZ | Application Load ID P003H310 |
| 04:05.13 | ELSPAZZZ | Boot Load ID PC03C300 |
| 04:06.58 | ELSPAZZZ | I know P00 means its using SCCP. It would say P0S if it was on the SIP firmware |
| 04:07.22 | jpsharp | Do you have your OS79XX.TXT file in place? |
| 04:07.26 | ELSPAZZZ | Yes |
| 04:08.03 | ELSPAZZZ | its set to load a p00 file which to my understanding is supposed to then chainload the P0S firmware |
| 04:09.31 | ELSPAZZZ | I've tried loading the POS3-08-8-00 firmware which has P003-08-8-00.bin as the loader. When that didn't work I tried a couple of older versions |
| 04:10.02 | ELSPAZZZ | The problem is i'm being told that the tftp server should hand it a provisioning file SIP(myphonesmacaddress) |
| 04:10.19 | ELSPAZZZ | but instead the phone is requesting SEP(macaddress) |
| 04:10.38 | ELSPAZZZ | So I know the firmware isn't upgrading but i'm not getting any error messages telling me why |
| 04:11.29 | ELSPAZZZ | and this is about where i'm stuck |
| 04:12.59 | ELSPAZZZ | Any idea where to go from here because I have no clue....... It's frustrating not even knowing what the problem is. |
| 04:13.57 | *** join/#asterisk Preytell (~Preytell@68-186-162-170.dhcp.stls.mo.charter.com) |
| 04:18.01 | Smirker | Hi. Using Asterisk 1.8.4.4, what's the best way to make asterisk start with safe_asterisk on boot (Debian)? I've looked in /etc/init.d/asterisk and it seems safe_asterisk configuration was removed from there (in the comments). |
| 04:18.40 | Smirker | I'm trying to make asterisk automatically restart when it crashes. ;o |
| 04:18.42 | Preytell | Evening all, I have a problem with Polycom and Asterisk with reguard to sip registration. I have looked over everything (Packet Level) and found that Asterisk is ignoring the request, the only thing I see in the registration packet that seems odd is that the phone is sending cseq 1. <-- this may be why * is ignoring the request but shouldn't * give up on the current registration and accept the packet as a new registration at so |
| 04:19.20 | Preytell | nothing short of restarting * fixes this issue. |
| 04:22.44 | ELSPAZZZ | jpsharp: Any idea's? |
| 04:46.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 04:47.19 | ELSPAZZZ | Ok. So I was able to try to get it to load an older SIP firmware. Part of my issue seems to be trying to upgrade too far from the base load the phone already has. I was able to get the phone to ask for the SIP firmware and was able to get a (Upgrading Software) indicator on my phone. However the TFTP server is reporting "Peer Returns ERROR <-> Transfer Aborted |
| 04:59.17 | *** part/#asterisk ELSPAZZZ (~Vince@76-242-98-61.uvs.livnmi.sbcglobal.net) |
| 05:04.02 | Smirker | Any ideas regarding my safe_asterisk question? ;o |
| 05:04.03 | *** join/#asterisk timahvo1 (~rogue@mail.sbakenyaltd.com) |
| 05:04.04 | Smirker | smirks |
| 05:22.57 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 05:27.16 | *** join/#asterisk RingtailedFox (~Ringtaile@d24-57-4-54.home.cgocable.net) |
| 05:27.39 | RingtailedFox | heya, guys... i run windows 7 pro 64-bit... i was wondering if asterisk is what i need to send and recieve phone calls on my computer (using my speakers and mic) |
| 05:32.01 | [TK]D-Fender | RingtailedFox: What you are looking for on your PC is a CLIENT. Asterisk is a telephony & PBX toolkit, most commonly used for building PBXs |
| 05:32.16 | [TK]D-Fender | RingtailedFox: Which would be more akin to saying it's a "server" |
| 05:32.35 | [TK]D-Fender | ~soft-phone |
| 05:32.46 | [TK]D-Fender | ~softphone |
| 05:32.46 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 05:32.50 | [TK]D-Fender | RingtailedFox: What you're referring to ^^^ |
| 05:33.23 | [TK]D-Fender | RingtailedFox: Now this softphone you are loking at may still have value in having a "system" in front of it. That is for you to determine. |
| 05:33.42 | RingtailedFox | i see |
| 05:34.16 | RingtailedFox | so, i need to get ekiga... ok |
| 05:37.48 | [TK]D-Fender | That's a fairly decent one. |
| 05:39.51 | RingtailedFox | it's the only one of those i've heard of :P |
| 05:40.14 | [TK]D-Fender | already good that you have then... |
| 05:41.56 | [TK]D-Fender | RingtailedFox: If your PC isn't on to act as that phone what would want to have happen? Many ITSP's offer voicemail, maybe you want something more. Are you looking at this for travel reasons? (expecting to be mobile all the time with a laptop for this) |
| 05:42.57 | RingtailedFox | no no... i just wanted to see if i was able to use my computer as a second phone, so i don't have to get up and go to the other room to answer it |
| 05:43.10 | RingtailedFox | hmm... everyone seems to charge for dialling-out with software.. |
| 05:44.53 | [TK]D-Fender | Yes, actually getting to the PSTN typically involes costs.... |
| 05:45.34 | RingtailedFox | it's a shame my ISP doesn't offer such software... evne though they use VOIP for telephony (i have their triple-play of cable TV + phone + internet) |
| 05:45.36 | [TK]D-Fender | So you have an existing line and don't want to splurge on a 10$ cheap phone? |
| 05:45.48 | [TK]D-Fender | Or a cordless? |
| 05:45.52 | RingtailedFox | oh, that's it? i thought you'd have to add another 0 on there |
| 05:46.02 | [TK]D-Fender | Those are an easy $25 |
| 05:46.03 | RingtailedFox | no, i don't have a cordless, but i think i could find a phone around here somewhere.... |
| 05:46.20 | RingtailedFox | but, i would like the ability to record calls, especially when i get harassingphone calls... you know, use them as evidence |
| 05:46.35 | [TK]D-Fender | If that's your actual need then a real phone is what is actually called for... |
| 05:46.41 | RingtailedFox | i see |
| 05:46.54 | [TK]D-Fender | Recording is another matter... |
| 05:47.01 | RingtailedFox | yah |
| 05:47.16 | [TK]D-Fender | RingtailedFox: Where are you from? |
| 05:47.19 | RingtailedFox | canada |
| 05:47.26 | [TK]D-Fender | Ok, so no legal issue there. |
| 05:47.29 | RingtailedFox | :) |
| 05:47.54 | RingtailedFox | as long as i tell them "asshole, i'm recording you. i know you live 6 houses down from me, so if you threaten to burn my house down again, i'll email these sound files to the police." |
| 05:47.58 | RingtailedFox | i'll be covered |
| 05:48.04 | RingtailedFox | since i have ot tell them if i'm recording at any point |
| 05:48.22 | [TK]D-Fender | You'd need something to take you home line in first, and at that point, Asterisk would be an intermediary letting you record as well while you're at it |
| 05:49.04 | RingtailedFox | nods |
| 05:50.25 | [TK]D-Fender | RingtailedFox: http://www.canadacomputers.com/product_info.php?cPath=27_1066_1069&item_id=012331your quickest solution : |
| 05:50.32 | [TK]D-Fender | RingtailedFox: http://www.canadacomputers.com/product_info.php?cPath=27_1066_1069&item_id=012331 your quickest solution : |
| 05:51.06 | RingtailedFox | i wonder how that would feed in to my LAN... |
| 05:51.11 | RingtailedFox | i already have a couple ethernet hubs |
| 05:51.55 | [TK]D-Fender | That will take your existing line in and let you send it to Asterisk via SIP on your network. It will also let you plug in your phones so you can use them as SIP devices as well. With Asterisk in the middle you co your call recording, voicemail, and whatever else you might want to do |
| 05:51.55 | RingtailedFox | does that have specialized software bundled with it? do i have to pay to dial out, or cna i have it use my phone line with the cable company? will it work with linux machines and windows machines? |
| 05:52.17 | [TK]D-Fender | This device you would not use the "router" functionality at all on it (unless you really wanted to) |
| 05:52.32 | RingtailedFox | so i'll need much more than just my old trusty-dusty (and missing) USRobotics V.92 56K modem... :P |
| 05:52.50 | [TK]D-Fender | This device just lets SIP solutions using your land line (so you can record it), and lets you plug in phones so they can talk SIP too |
| 05:53.01 | [TK]D-Fender | ~savemoney |
| 05:53.01 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 05:53.03 | [TK]D-Fender | ^^ |
| 05:53.05 | [TK]D-Fender | :p |
| 05:53.11 | RingtailedFox | :D |
| 05:53.23 | [TK]D-Fender | Yeah, modem doesn't cut it... |
| 05:53.38 | RingtailedFox | i remember way back to the days of 3.1 and 95 when it *would* |
| 05:53.40 | RingtailedFox | :P |
| 05:53.43 | [TK]D-Fender | Then again, for < $60 (you can find them on kijiji, etc too) hey, it's already a steal. |
| 05:53.46 | RingtailedFox | modem, sound card, speakers, poof! |
| 05:53.51 | RingtailedFox | yah, i just might ge tone |
| 05:54.24 | RingtailedFox | but with this SIP router, would i even need a modem or phone line access at any point? |
| 05:54.29 | RingtailedFox | or does it have its own phone jack? |
| 05:55.22 | [TK]D-Fender | It has a jack to take your line in, and a jack for your phones |
| 05:55.44 | [TK]D-Fender | All you'd need on top of this is a computer to run Asterisk on. |
| 05:56.00 | RingtailedFox | how resource-intensive is asterisk? |
| 05:56.24 | [TK]D-Fender | RingtailedFox: For what you've described .... my watch can do this... and it's ANALOG :p |
| 05:56.39 | RingtailedFox | oh, so not intensive at all |
| 05:56.44 | [TK]D-Fender | RingtailedFox: basically they cheapest POS you'll find is probably WAY more than enough |
| 05:56.47 | RingtailedFox | i could probably use my desktop machine i'm typing to you on righ tnow, then |
| 05:57.17 | [TK]D-Fender | You'd have to leave it on all the time and as you're running windows You'd pretty much be running a VM for this or something of the sort. |
| 05:57.18 | RingtailedFox | well, this isn't an immediate need... and i did find an old landline to put on my desk, so i can wait a few months to get this |
| 05:57.24 | RingtailedFox | hmm |
| 05:57.27 | RingtailedFox | well, i tend to reboot every 2-3 days |
| 05:57.38 | RingtailedFox | sometimes more, sometimes less... |
| 05:57.42 | RingtailedFox | but that's the average |
| 05:57.57 | [TK]D-Fender | just rebooting isn't the most crucial amount of "down time" for you I suspect. |
| 05:58.00 | RingtailedFox | i have more pressing needs, like getting more ram and a better video card, first |
| 05:58.11 | [TK]D-Fender | I would probably get a cheap-ass netbook for this. |
| 05:58.26 | [TK]D-Fender | All in one, built in batter backup..... |
| 05:58.30 | [TK]D-Fender | +y |
| 05:58.40 | RingtailedFox | ehh, the most-critical downtime was when Windows 7 Pro 64-bit's SP1 installed sideways and screwed everything up... took 14 hours in safe mode and a ton of rebootings, but i was able to fix it :p |
| 05:58.52 | RingtailedFox | spread out over 2 and a half days... as i don't work when i'm asleep |
| 05:59.07 | RingtailedFox | i'm a fox, not a mindless robot! |
| 05:59.14 | [TK]D-Fender | RingtailedFox: That's not very practical. You should work on that :p |
| 05:59.19 | RingtailedFox | i kno |
| 05:59.21 | RingtailedFox | *know |
| 05:59.24 | RingtailedFox | but no other updates have done it before or since |
| 05:59.33 | RingtailedFox | everything's been smooth.... even re-installed SP1... no issues |
| 06:00.06 | RingtailedFox | so, a dedicated machine is strongly recommended... ok |
| 06:00.07 | RingtailedFox | i can get that |
| 06:00.51 | RingtailedFox | i have an old P3-533 MHz machine upstairs i could probably put to use, but it bogged down when i put Win2000 on it... might have to use something like Puppy Linux instead |
| 06:01.47 | RingtailedFox | hm. i'd also need a monitor... i can just use my HDTV, it has VGA input for a few minutes, or use something like VNC or Remote Desktop to dial-in |
| 06:03.49 | RingtailedFox | it's kinda dizzying... how much computers have changed since the late 1980s |
| 06:03.51 | RingtailedFox | and early 1990s |
| 06:04.48 | RingtailedFox | back then, we had those old mickey-mouse phone modems for the Commodorre 64... used a "blazing-fast" 9600-bps modem (later a 14.4... lol) on Win 3.0... i remember when i thought using Nestcape Communicator on Win95 with my 33.6 modem was like "warp speed" |
| 06:04.51 | RingtailedFox | man, i feel old |
| 06:07.32 | RingtailedFox | hope i didn't kill the chat x.x |
| 06:12.08 | [TK]D-Fender | Sorry, e-mail I had to write |
| 06:12.14 | [TK]D-Fender | That's not old |
| 06:12.43 | [TK]D-Fender | I've used acoustic coupled 300-baud modems, and ones you had to hold a stiwch on to even initiate carrier |
| 06:13.20 | [TK]D-Fender | And your 64 beat my IC-20, Commodore PET, and WORSE I've had in my lifetime... |
| 06:13.24 | [TK]D-Fender | VIC* |
| 06:13.29 | Preytell | that's where I started...and my phone had square ends, I had to go buy another that would fit the coupler. |
| 06:14.26 | Preytell | I used to repair PET's, 64's, 128's and dare I say it Amiga, go Fat Agnes. |
| 06:23.02 | Smirker | Hi guys. Can anyone suggest the best way to ensure Asterisk 1.8 is restarted when it crashes? So far I'm looking into safe_asterisk (although I'm worried about it being started in /tmp) and inittab 'respawn'. |
| 06:26.29 | jpsharp | heck, I was still using an A2000 for broadcast work 3 years ago. |
| 06:26.46 | [TK]D-Fender | safe_asterisk already does this and is the first thing to do. |
| 06:27.53 | Smirker | I'm not sure how to configure the server to use safe_asterisk on boot though. It appears the option was removed from /etc/init.d/asterisk in 2007 - and I've been searching Google for some time ;p |
| 06:28.03 | Smirker | Could you make any recommendations? |
| 06:29.37 | Preytell | I haven't used it in a few years, but http://wiki.koshatul.com/Installing_Asterisk_-_Daemon_Tools |
| 06:30.33 | Preytell | DJB makes a great tool in daemontools, which can be installed on most linux based systems... can be a slight learning curve. |
| 06:31.58 | Smirker | Thanks - I'll have a read into that. |
| 06:32.49 | [TK]D-Fender | Smirker: typically a distro-based init script will call this for you |
| 06:33.19 | [TK]D-Fender | ones for RH, debian, etc come included with * using "make config" |
| 06:33.54 | RingtailedFox | i've used my grandfather's VIC-20 and PET :) |
| 06:34.00 | RingtailedFox | of course, i was just RingtailedPup back then :p |
| 06:34.30 | Smirker | [TK]D-Fender: I can run safe_asterisk manually, however I'm not sure how to configure it to run on boot. i.e. what startup options to modify/etc. |
| 06:34.54 | Smirker | [TK]D-Fender: It's definitely not using safe_asterisk at the moment. When Asterisk crashes, it stays down until I issue a start command. |
| 06:35.00 | [TK]D-Fender | Generally nothing much to do aside from the user to run * as |
| 06:52.56 | ChannelZ | see the contrib/init.d folder |
| 06:55.52 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:56.49 | [TK]D-Fender | checkout time, later all |
| 07:04.39 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:13.16 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:16.38 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:18.47 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 07:47.34 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 07:49.06 | *** join/#asterisk Vince-0 (~vincent@105-236-228-152.access.mtnbusiness.co.za) |
| 07:49.11 | verywiseman | i have grandstream HT503, i register fxo to asterisk , if i want to dial land line , is it just Dial(SIP/${landlinenumber}) ? |
| 07:55.36 | kaldemar | verywiseman: no, you must tell asterisk which device to use. SIP/<peernameofht503>/<number> |
| 07:56.52 | verywiseman | kaldemar, is <peernameofht503> which is defined in sip.conf ? |
| 07:59.20 | kaldemar | yes |
| 08:01.38 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 08:11.04 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:16.53 | *** join/#asterisk Krolo (~krolo@gw.ptr-80-238-141-253.customer.ch.netstream.com) |
| 08:20.54 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.168) |
| 08:22.34 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 08:25.56 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 08:33.02 | *** join/#asterisk evilman_home (kvirc@2.94.90.125) |
| 08:43.03 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 08:47.10 | verywiseman | i have grandstream HT503, and i register fxo and assign context for it , and all things seem good , but , when i call the landline asterisk detect it , and i saw in log file that , the applications in pstn context is executing , but there was not actions was taken , i mean , one of applications is Answer() , it appeared in log file that asterisk execute it , but the call is not answered by asterisk , why? |
| 08:48.11 | kaldemar | what makes you think that asterisk does not answer the call? |
| 08:48.25 | kaldemar | and what are you expecting to happen when you make a call? |
| 08:48.44 | kaldemar | answer those and show the CLI output you get for a call. use pastebin. |
| 08:48.46 | kaldemar | ~pb |
| 08:48.46 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 08:53.18 | verywiseman | kaldemar, because the log didn't stop in Answer() application , and i expect the call is been answered by * , but i still hear ring tone . http://fpaste.org/NMmu/ |
| 08:58.10 | verywiseman | kaldemar, there is something you must know , the line which come from pstn , firstly it enter into panasonic pbx , then enter into ATA. |
| 08:58.56 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 08:59.52 | verywiseman | kaldemar, i am sorry my internet is disconnect |
| 09:00.11 | verywiseman | kaldemar, i write my answer for your questions again |
| 09:00.18 | verywiseman | kaldemar, because the log didn't stop in Answer() application , and i expect the call is been answered by * , but i still hear ring tone . http://fpaste.org/NMmu/ |
| 09:00.26 | verywiseman | kaldemar, there is something you must know , the line which come from pstn , firstly it enter into panasonic pbx , then enter into ATA. |
| 09:03.26 | *** join/#asterisk engrxyz (~uiuihkhjk@host81-150-217-167.in-addr.btopenworld.com) |
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| 09:09.18 | kaldemar | verywiseman: you must do something with the call. the answer application just answers the line and moves on in the dialplan. it doesn't mean that asterisk would start a conversation with you by itself. you're seeing an immediate hangup after the answer. what are you trying to to with the call? |
| 09:11.01 | verywiseman | kaldemar, ok i will put some apps , but why the call didn't hangup though i put hangup() app? |
| 09:12.48 | kaldemar | verywiseman: that's because of the ATA and you panasonic PBX. asterisk will hang up the call on the SIP side to the ATA. |
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| 09:14.40 | verywiseman | kaldemar, are you meaning , if i remove panasonic pbx , and connect the pstn line to ata directly , the call will hangup? |
| 09:15.48 | kaldemar | no. you might experience an issue there too it the ATA is to blame. |
| 09:16.15 | kaldemar | s/it/if/ |
| 09:17.03 | verywiseman | infobot, i know :) |
| 09:17.03 | infobot | You know :)? |
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| 10:42.02 | salz212 | guys, do we have support of codec_amr in asterisk 11?. A patch.. or something to test.. |
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| 11:12.58 | Rahail | hi there new in asterisk trying figer out this We have 2 Asterisk pbx |
| 11:13.10 | Rahail | each have around 5-10 sip Extensions |
| 11:13.30 | Rahail | how can we call each other on different Asterisk server |
| 11:13.51 | Rahail | IE; Server A exten X want call Server B exten Y |
| 11:14.01 | Rahail | so what do i need to configure on extensions.conf |
| 11:16.45 | kaldemar | ~book |
| 11:16.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:17.25 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291281 |
| 11:20.09 | Rahail | i follow those doc |
| 11:20.12 | Rahail | <PROTECTED> |
| 11:20.45 | Rahail | 7906320375 is the extensions we trying to reach from server B |
| 11:21.40 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 11:24.54 | kaldemar | you have not made a matching peer in sip.conf. that's why the calls land to context 'default'. |
| 11:25.30 | kaldemar | or you have but you're using default as a context, which is bad practice. |
| 11:31.22 | Rahail | sip extensions is on context office1 |
| 11:31.24 | Rahail | office2 |
| 11:31.26 | Rahail | like that |
| 11:31.33 | Rahail | from Server A how can we call those |
| 11:31.41 | Rahail | sorry i been trying so many way to figerout still no luck |
| 11:32.54 | kaldemar | what have you done so far? use pastebin to show sip.conf (mask secrets) and extensions.conf. |
| 11:32.57 | kaldemar | ~pb |
| 11:32.57 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 11:35.58 | Rahail | http://pastebin.com/MyvTsJL5 |
| 11:37.00 | Rahail | on Server A we have dial plan we put 9Serversip user info so this bring call to server B |
| 11:37.19 | Rahail | but when it comes to server B I can not figerout how to add the dailplan to go to correct sip user |
| 11:41.03 | kaldemar | exten => _x.,3,SIP(${EXTEN}) is not valid |
| 11:41.23 | kaldemar | context=office1 <-- don't see such a context |
| 11:41.44 | kaldemar | [7906320375] <-- don't see anything that would use this peer |
| 11:42.08 | Rahail | sorry i am bit lost |
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| 12:26.02 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 12:28.37 | verywiseman | It is may seem stupid question :) , any way , i have ATA and ADSL modem , which is better : 1- to insert pstn line into splitter and then to ATA and adsl modem K , or 2- insert pstn line into FXO and adsl modem into FXS ? |
| 12:29.28 | Rahail | kaldemar |
| 12:29.31 | Rahail | figure it out |
| 12:29.35 | Rahail | exten => _X.,1,dial(sip/${EXTEN}) |
| 12:29.38 | Rahail | this did the trick |
| 12:32.15 | kaldemar | is it like in the example from the book? |
| 12:32.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:33.22 | kaldemar | verywiseman: only one of those will work. |
| 12:35.33 | Rahail | i found it one of the mailing list |
| 12:35.46 | Rahail | simular to what i need another person was trying |
| 12:35.52 | Rahail | so i just uset it |
| 12:37.56 | kaldemar | your current config will not enable dialing between all phones. |
| 12:46.15 | verywiseman | kaldemar, thank you for your support , i am really appreciate that :) |
| 12:56.36 | afink | kind off topic but can you guys get to msn.com? |
| 12:57.53 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 12:58.44 | gavimobile | I need help with a dialplan 072XXXXXXX I want the number 2 to be changed to 2 || 4 || 5 || 6 || 7 || 8 || 9 |
| 12:58.50 | gavimobile | note that I don't want 073 |
| 12:59.16 | gavimobile | how do I do that? 07[2|4-9]XXXXXXX ? |
| 13:00.06 | gavimobile | or 07[2,4-9]XXXXXXX |
| 13:01.23 | gavimobile | 07[24-9]XXXXXXX |
| 13:01.26 | gavimobile | like that, right? |
| 13:01.43 | kaldemar | gavimobile: [24-9]. you should just try those yourself. |
| 13:02.10 | gavimobile | kaldemar: I couldn't find it in the definitive guide |
| 13:02.12 | gavimobile | sorry and thanks |
| 13:03.14 | salz212 | Do we have any patch available for amr codec in Asterisk 11? |
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| 13:38.20 | acidfoo | ~book |
| 13:38.21 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 14:23.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:25.00 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:49.46 | verywiseman | kaldemar, Thank you , All things good now , Thank you again Guru :) |
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| 15:07.10 | verywiseman | kaldemar, would you please look at this ,http://fpaste.org/1T3e/ ? |
| 15:07.39 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:09.19 | kaldemar | allow = gsm,alaw,g726aal2,adpcm,slin,lpc10,g729,speex,ilbc,g726,g722,jpeg,png,h261,h263,h263p,h264 <-- use one codec per allow line. |
| 15:09.49 | kaldemar | and try again. this time enable sip debug to see what really happens. |
| 15:10.33 | [TK]D-Fender | verywiseman, Also other could help with this. Walking in an calling him out by name limits those who might answer you. Also make sure not to have extra chars tacked onto the URLs you provide. |
| 15:20.31 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:35.48 | verywiseman | kaldemar, this is sip.conf after modification http://fpaste.org/PgrS/ , but problem is still existing. |
| 15:36.09 | *** join/#asterisk vfabi (~fabi@93.116.255.140) |
| 15:37.55 | verywiseman | [TK]D-Fender, Would you please explain more ? |
| 15:41.04 | kaldemar | verywiseman: i told you to enable sip debug. pastebin the output. |
| 15:41.22 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:42.19 | kaldemar | verywiseman: he meant that if you have a question, don't direct the question to a single person. |
| 15:45.07 | ChannelZ | SIP/100/37626934 probably makes very little sense |
| 15:50.45 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:56.58 | [TK]D-Fender | veyYou have no explained what those SIP devices are talking to, and you have not enabled SIP DEBUG and provided a new call as requested |
| 15:57.26 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:58.21 | ChannelZ | was that a truncated "Oy vey!" at the front there? :) |
| 15:58.46 | *** join/#asterisk cheasee (~cheasee@vie-188-118-242-061.dsl.sil.at) |
| 15:58.49 | cheasee | hi |
| 15:59.17 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:59.18 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:00.09 | cheasee | is there maybe some sip client known/available for linux which is able to run without an x-window-system installed? just running on shell |
| 16:01.57 | kaldemar | asterisk is one. :) |
| 16:02.13 | verywiseman | kaldemar, http://fpaste.org/CIbi/ |
| 16:02.19 | cheasee | ive googled but found just pjsip which is a library, not a client |
| 16:03.21 | cheasee | kaldemar: can asterisk be configured to use bluetooth? |
| 16:03.44 | [TK]D-Fender | Asterisk is a lousy SIP client |
| 16:03.46 | [TK]D-Fender | Get something else |
| 16:04.07 | kaldemar | cheasee: there is pjsua under the project. |
| 16:04.14 | WIMPy | cheasee: sflphone, IIRC |
| 16:04.30 | [TK]D-Fender | cheasee, http://www.pjsip.org/pjsua.htm |
| 16:04.34 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.133) |
| 16:04.41 | [TK]D-Fender | cheasee, it is not just a "library" |
| 16:04.52 | kaldemar | cheasee: linphonec |
| 16:05.40 | *** join/#asterisk greenwolf (cf1d2a02@gateway/web/freenode/ip.207.29.42.2) |
| 16:05.47 | cheasee | not? i just downloaded pjproject, configured and compiled it but i see no binary or similar to use? after reading "getting started" there writes something about library ready to use in applications.. |
| 16:07.52 | cheasee | WIMPy: sflphone-daemon looks good thanks :) |
| 16:09.05 | cheasee | kaldemar: linphone is a graphical client |
| 16:09.08 | ChannelZ | verywiseman: your ATA is rejecting the call. Does it have a dialplan of its own? |
| 16:09.21 | kaldemar | ChannelZ: i said linphonec, not linphone. |
| 16:09.24 | cheasee | ah there's a -nogtk package too |
| 16:09.28 | kaldemar | ChannelZ: oops, sorry. |
| 16:09.35 | kaldemar | cheasee: ^^ |
| 16:09.35 | ChannelZ | no worries |
| 16:09.53 | cheasee | heh :) |
| 16:10.28 | cheasee | kaldemar: i cant find a package in debian for linphonec, so i guessed a typo and looked for linphone |
| 16:10.59 | cheasee | there is linphone-no-gtk :) |
| 16:11.03 | [TK]D-Fender | verywiseman, under [100] add "fromuser=100", "directmedia=no", remove "canreinvite = no", and remove all but ONE of those codecs you "allow" there |
| 16:11.09 | cheasee | linphone-nogtk even |
| 16:11.24 | [TK]D-Fender | verywiseman, and change type to "peer" |
| 16:11.40 | [TK]D-Fender | verywiseman, You are using a lot of 1.4 and lower habits and parameters. |
| 16:15.13 | Katty | hai. |
| 16:15.49 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:18.35 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 16:25.26 | *** join/#asterisk P-NuT (2e41fea1@gateway/web/freenode/ip.46.65.254.161) |
| 16:25.58 | P-NuT | Hi all. |
| 16:26.46 | P-NuT | I have a question about codec bandwidth |
| 16:27.11 | P-NuT | if, with ethernet overhead you calculate a g.729 call at roughly 30kbps |
| 16:27.35 | P-NuT | is that roughly 15kbps one way and 15k in the other for a total of 30kbps? |
| 16:27.50 | WIMPy | There are lots of voip bandwidth calculators out there. Try to ask google for them. |
| 16:28.26 | WIMPy | Probably each way, but that depends on thw version of G.729 and the packetization. |
| 16:28.50 | WIMPy | 15kbps per direction is probably not possible. |
| 16:29.04 | [TK]D-Fender | that's UDP overhead, not ethernet. |
| 16:29.36 | [TK]D-Fender | for the RTP packet header based on packetization rate, etc |
| 16:29.48 | WIMPy | A little UDP, a little IP, possibly a little ethernet, but certainly lots of RTP. |
| 16:30.11 | [TK]D-Fender | ~20kbps overhead @ 20ms (standard) packet size. |
| 16:30.17 | [TK]D-Fender | PER direction. |
| 16:30.33 | [TK]D-Fender | so that's 30 UP *and* 30 DOWN |
| 16:30.43 | P-NuT | @Wimpy - Thanks, I worked it out here.. http://www.asteriskguru.com/tools/bandwidth_calculator.php |
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| 16:42.24 | jmetro | hm.. anyone know the MOS of Speex? |
| 16:46.04 | verywiseman | i did what [TK]D-Fender said , and allow ilbc codec in sip.conf for my sip client conf and fxo conf , as well as in my ATA i let the first choice is ilbc , but this output appear , http://fpaste.org/zv92/ |
| 16:46.21 | *** join/#asterisk moos3 (~moos3@pool-72-73-84-223.ptldme.east.myfairpoint.net) |
| 16:46.35 | moos3 | anyone know why 58 channels of sip would eat 8G of ram |
| 16:47.36 | [TK]D-Fender | verywiseman, Stop wasting time providing CLI without SIP DEBUG. you have SIP issues and you are picking oddball codecs. Start with the BASICS. ULAW/ALAW. |
| 16:48.24 | danfromuk | Hi, can someone help me with this error message? Unable to request echo training on channel 4: Invalid argument |
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| 16:54.43 | danfromuk | Are the different echo canceller documented anywhere? |
| 17:00.16 | [TK]D-Fender | http://forums.digium.com/viewtopic.php?f=1&t=80000 |
| 17:00.29 | [TK]D-Fender | I'd simply disable it |
| 17:05.08 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 17:09.33 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.1.2 (2013/01/03), 10.11.1 (2013/01/02), 1.8.19.1 (2013/01/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 17:23.33 | Katty | aww |
| 17:23.35 | Katty | infobot: crittercam |
| 17:23.35 | infobot | methinks crittercam is Katty's Critter Cam http://tinyurl.com/b5k3lt4 |
| 17:23.42 | Katty | adorable squirrely is addorable ^_^ |
| 17:26.29 | moos3 | anyone ever seen this app_meetme.c:3352 conf_run: Unable to write frame to channel |
| 17:31.06 | *** join/#asterisk Micc_ (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 17:31.29 | leifmadsen | moos3: seen those, usually just means someone hung up right as asterisk was about to send audio |
| 17:31.40 | leifmadsen | not seen it with meetme, but I've seen that with sip |
| 17:31.44 | leifmadsen | usually just ignore it as it causes no real issues |
| 17:31.57 | moos3 | so how do i correct so my log doesn't have millions of lines in that |
| 17:32.13 | Micc_ | anyone know a good mitel reseller? |
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| 17:38.05 | zerohalo | hungry squirrel |
| 17:40.01 | jmetro | that squrrels been there for like at least 15min |
| 17:40.39 | zerohalo | snowman is not impressed |
| 17:51.49 | infinity1 | is teliax down? |
| 17:52.13 | Qwell | call and ask? |
| 17:54.22 | infinity1 | you can't. just hangs up. i tried on my cell. |
| 17:55.05 | infinity1 | i tried their 888 # and 303 #. nothing works. |
| 17:57.27 | Qwell | there you go then |
| 17:57.53 | Russ | an hour ago I did a 25 minute call to finland |
| 17:58.14 | Russ | but now... |
| 17:58.33 | Russ | lax.teliax.net not happy |
| 17:58.38 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 17:58.47 | infinity1 | Russ: on teliax? looks like i had a successfull call at 8:53 PST |
| 17:59.02 | infinity1 | Russ: I'm on den.teliax. no love there either. |
| 17:59.19 | infinity1 | Russ: i also tried forwarding my DIDs. nothing. |
| 17:59.22 | infinity1 | all dark. |
| 17:59.58 | Russ | https://teliax.zendesk.com/entries/22859386-trouble-open-network-event-causing-services-down |
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| 18:00.19 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 18:01.04 | *** join/#asterisk MrRobert (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 18:02.10 | infinity1 | that wasn't there last time i was on that page lol |
| 18:02.24 | MrRobert | Asterisk 11, Agents logged in to multiple queues, using SkipBusyAgents = Yes (ringinuse=no), but when agents are on outbound calls, the queue still tries to send calls to them, am I missing a setting? |
| 18:02.38 | Russ | '11:00 MST - Access to most services has been restored at this time.' |
| 18:03.11 | Russ | not for me… |
| 18:03.17 | Russ | go ben go |
| 18:03.39 | MrRobert | to what are you referring Russ? |
| 18:03.48 | Russ | a teliax outage |
| 18:03.56 | infinity1 | Russ: not for me either. |
| 18:04.21 | infinity1 | thank god for ipkall. its the only thing working |
| 18:05.03 | infinity1 | Russ: i just received an inbound call. no outbound calls yet. |
| 18:05.54 | Russ | same here, inbound works, outbound not so much |
| 18:06.21 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 18:07.02 | feeshon | Looking to auto provision hookswitch mode for polycom phone, can't seem to find proper documentation. |
| 18:07.13 | feeshon | can anyone help? |
| 18:07.25 | *** join/#asterisk gusto (~gusto@bband-dyn188.178-41-67.t-com.sk) |
| 18:11.38 | Qwell | ~google polycom admin guide |
| 18:13.30 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:21.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:21.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:23.55 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 18:27.04 | greenwolf | question... |
| 18:27.48 | greenwolf | if i wanted to achieve this effect: when a sip device/ or user picks up their telephone or handset and wanted an IVR menu to be played instantly with options rather then just an actual dialtone |
| 18:28.13 | greenwolf | could i achieve this efftect by simply putting the extension or sip device is the context lets say [localsets] |
| 18:28.33 | [TK]D-Fender | greenwolf, the PHONE would have to call * immediately on pickup. |
| 18:28.34 | greenwolf | and start with exten => s,1,background(IVR-to-be-played-upon-pickup) |
| 18:28.36 | greenwolf | would that work? |
| 18:28.43 | [TK]D-Fender | greenwolf, Check your phone's manual to see if it supports it |
| 18:28.50 | greenwolf | i see |
| 18:28.57 | [TK]D-Fender | greenwolf, Nothing in * can't force this to happen. |
| 18:28.59 | greenwolf | so your saying some devices wont support that? |
| 18:29.01 | [TK]D-Fender | can* |
| 18:29.20 | greenwolf | even if i put the device in a context and started it off with exten =>s? |
| 18:29.27 | greenwolf | wouldn't it just start in the S exten? |
| 18:29.46 | [TK]D-Fender | no |
| 18:29.58 | greenwolf | the problem is this company that hired me to do their phone system wanted one of their main lobby phones to do this |
| 18:30.08 | greenwolf | once you pick up the device it starts playing the IVR menu |
| 18:30.12 | [TK]D-Fender | <[TK]D-Fender> greenwolf, Check your phone's manual to see if it supports it |
| 18:30.19 | greenwolf | rather than allowing anyone to dial anything...i guess for security purposes |
| 18:30.29 | greenwolf | ok what would that be called in the manual? |
| 18:30.33 | WIMPy | A sip phone will never call nothing. |
| 18:30.34 | [TK]D-Fender | As for what context they are in, that is another matter. |
| 18:30.48 | [TK]D-Fender | But there is no "automatic menu" unless the phone dials something on pickup |
| 18:31.02 | greenwolf | what if i use a TDM device with FXO and FXS ports? |
| 18:31.11 | greenwolf | oh isee |
| 18:31.14 | greenwolf | ok that gives me a really good idea |
| 18:31.26 | WIMPy | If it's an ATA, it's the same situation. |
| 18:31.32 | greenwolf | so within my sip phones programming menu i can have it randomly dial a built in exten for say *7 |
| 18:31.40 | WIMPy | If it's a card, it would work. |
| 18:31.43 | greenwolf | which would automatically bring them to the IVR |
| 18:31.43 | [TK]D-Fender | RANDOM? |
| 18:31.47 | [TK]D-Fender | That doesn't sound smart |
| 18:31.55 | greenwolf | not random per say |
| 18:32.01 | greenwolf | i mean i can set any extension |
| 18:32.05 | [TK]D-Fender | greenwolf, Stop guessing. |
| 18:32.07 | greenwolf | thats what i meant by random |
| 18:32.43 | SuPrSluG | search for batphone type applications |
| 18:32.47 | greenwolf | when you say if its a card it could work |
| 18:33.07 | greenwolf | so if i use a digital or anolgue equipment or cards i can set this? |
| 18:33.07 | [TK]D-Fender | greenwolf, Why are you asking about other things and not looking at what you have? |
| 18:33.30 | greenwolf | i was just wondering just in case i occur this and they have tdm cards |
| 18:33.37 | greenwolf | but they are using just SIP devices |
| 18:33.59 | [TK]D-Fender | greenwolf, DAHDI FXS channels can do this. |
| 18:34.23 | [TK]D-Fender | For anything else... check its manual |
| 18:34.31 | WIMPy | Or BRIs. |
| 18:34.39 | greenwolf | ok so if i wanted to achieve this same effect on sip device i would have to program within the ATA devices menu to automatically dial an extension upon pickup |
| 18:34.51 | greenwolf | are you familiar if most sip ATA devices and sip phone allow this? |
| 18:34.53 | [TK]D-Fender | greenwolf, yes |
| 18:34.54 | WIMPy | indeed |
| 18:35.08 | greenwolf | ok thanks |
| 18:35.46 | greenwolf | yea i was in the elevator and was thinking about how I would achieve this when i went on break, so i decided to jump on IRC to see what others thought |
| 18:36.56 | greenwolf | ok great thanks [TK]D-Fender, thanks WIMPy for your input on this matter |
| 18:37.11 | greenwolf | its great to have smarter ppl than me around to help me in things like this lol :-) |
| 18:37.41 | greenwolf | you guys are great have a good one i gotta get back and finish this up for them..i will check the ATA devices manual on setting this in their phone web menu |
| 18:37.44 | greenwolf | later guys!! |
| 18:37.57 | gusto | what ATA? |
| 18:38.40 | gusto | maybe i ll try out more ATA's ... nowadays i have only two PAP2T's and one SPA112 |
| 18:39.33 | WIMPy | The S&M scene is alive. |
| 18:39.59 | gusto | aha |
| 18:40.03 | gusto | sh*** |
| 18:42.24 | gusto | but i like my ATA's |
| 18:42.48 | gusto | seems like there would be only SPA112 out there, like there would be no other manufacturers |
| 18:46.13 | gusto | like funkwerk ... maybe |
| 18:49.47 | gusto | i am going back to care about my little jails ;-) |
| 18:56.21 | MrRobert | Asterisk 11, Agents logged in to multiple queues, using SkipBusyAgents = Yes (ringinuse=no), but when agents are on outbound calls, the queue still tries to send calls to them, am I missing a setting? |
| 18:57.34 | [TK]D-Fender | MrRobert, You're missing having properly defined which device to use for state information for your "members" |
| 18:57.51 | [TK]D-Fender | MrRobert, And I would refrain from using the word "agents", probably indefinitely |
| 18:57.55 | WIMPy | Do yu have callcounters and coutersonpeers enabled? |
| 19:01.09 | [TK]D-Fender | Lets see how they were even added first... |
| 19:02.49 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:03.58 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:03.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:04.57 | *** join/#asterisk televoip (~kbushong@2620:0:280:12:2e41:38ff:feb1:35b9) |
| 19:05.47 | MrRobert | that makes sense, they are using soft - sip phones, how can i configure this [TK]D-Fender |
| 19:06.10 | [TK]D-Fender | MrRobert, Nothing makes sense yet. We didn't see exactly how you added your memebers, etc |
| 19:06.23 | MrRobert | callcounters are enabled yes |
| 19:07.17 | [TK]D-Fender | ... |
| 19:07.18 | [TK]D-Fender | pastebin |
| 19:09.28 | MrRobert | http://pastebin.com/xh2K7jEy |
| 19:10.30 | Russ | btw infinity1, my issues are leared |
| 19:10.32 | Russ | er, cleared |
| 19:12.30 | [TK]D-Fender | MrRobert, We do not see precisely how they were added as a memeber of your queue. Please be complete about this. |
| 19:12.54 | [TK]D-Fender | type=friend <- should be "peer" |
| 19:16.39 | *** join/#asterisk navaismo (~navaismo@189.144.207.195) |
| 19:20.01 | jmetro | type=phone |
| 19:20.07 | WIMPy | I'm a little irritated. Is it still not possible to log changing information to do billing? |
| 19:20.36 | [TK]D-Fender | <jmetro> type=phone <- PARDON? |
| 19:20.55 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 19:25.41 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 19:25.55 | *** join/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
| 19:26.51 | saint_ | Hi all - So I have an Asterisk server with a public IP address. Right now, the firewall only allows incoming connections to 5060 tcp/udp and 10,000:20,000 udp coming from my IP address. When I call the server, most of the time I do not hear voice. But every other call, it will work. What could cause this issue ? |
| 19:28.02 | MrRobert | http://pastebin.com/s9spJK3J |
| 19:33.58 | saint_ | mmhh....? |
| 19:34.55 | Primer | I still haven't been able to figure out why the hell a polycom 501 seems to have a hard-on for 1004. I've created a small home phone network with 4 Polycom 501s as well as 1 Polycom soundstation, assigned to extensions 1002, 1003, 1004, 1005, and 1006. The digitplan has this in it: 100x |
| 19:35.04 | [TK]D-Fender | MrRobert, We'd have to see the dialplan FreePBX is using to do that dirtywork, and that is the channel you should be in for all of these matters |
| 19:35.13 | [TK]D-Fender | MrRobert, FreePBX controls your configs, not you. |
| 19:35.23 | Primer | When any of these dial 1002, 1003, 1005, 1006, it works. When any dial 1004, the dial tone returns, and it reads 'Enter more digits' |
| 19:35.42 | Primer | I don't suppose anyone has ever run into this? |
| 19:35.48 | MrRobert | i'm trying to get a better understanding |
| 19:35.50 | [TK]D-Fender | Primer, Show us the call attempt |
| 19:36.00 | Primer | [TK]D-Fender: it never reaches asterisk |
| 19:36.03 | MrRobert | btw changiing them to peers totally screwed it up haha |
| 19:36.16 | [TK]D-Fender | MrRobert, Join #freepbx and pastebin the CLI of the login |
| 19:36.21 | Primer | it's some sort of internal Polycom thing |
| 19:36.38 | [TK]D-Fender | Primer, Go prove it first. SIP DEBUG @ * CLI |
| 19:37.23 | Primer | nothing |
| 19:38.41 | saint_ | i changed the firewall rule to accept all 1:65535 tcp/udp from my IP to my asterisk server, and it's still giving me a had time. Anyone has an idea by any chance ? |
| 19:39.01 | Primer | [TK]D-Fender: actually, let me make sure |
| 19:39.31 | Primer | [TK]D-Fender: ok, nm, seems you are correct |
| 19:39.50 | [TK]D-Fender | Primer, And now what do you see? |
| 19:42.22 | Primer | I'm trying to capture it now |
| 19:44.20 | Primer | [TK]D-Fender: http://ceregatti.org/p/25485 |
| 19:44.24 | Primer | yes, I have my own pastebin |
| 19:44.59 | saint_ | [TK]D-Fender: do you have any hint on the kind of issue I have ? asterisk -> public_ip <-- internet --> public_ip_on_my_home_router <- private network <- Computer |
| 19:45.12 | Primer | Sending to 10.2.1.176 : 5060 (no NAT) <-- is suspect, as there is no route to that network |
| 19:45.17 | saint_ | I tured NAT=ON in the user's template in sip.conf |
| 19:45.49 | Primer | I have nat=yes set for every extension in sip.conf |
| 19:46.10 | Primer | saint_: nothing to do with what you're talking about, btw, in case you're wondering |
| 19:46.35 | rrittgarn | saint_ what version of asterisk? in 10 its nat=yes, in 11 its nat=force_rport,comedia |
| 19:46.44 | saint_ | rrittgarn: 1.8 |
| 19:47.21 | Primer | [TK]D-Fender: http://ceregatti.org/p/25486 (sip show peers) |
| 19:48.23 | rrittgarn | saint_: pretty sure its nat=yes not nat=on |
| 19:48.31 | [TK]D-Fender | Primer, Looking for 1004 in teliax-unrestricted (domain voip.ceregatti.org) <- no DIALPLAN match |
| 19:48.41 | saint_ | rrittgarn: i meant nat=yes, sorry |
| 19:49.12 | [TK]D-Fender | saint_, improper handling of NAT |
| 19:49.26 | Primer | [TK]D-Fender: indeed. Good catch. Thanks! |
| 19:50.08 | saint_ | [TK]D-Fender: on asterisk side or x-lite ..? |
| 19:50.38 | Primer | and it works |
| 19:50.40 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 19:50.44 | Primer | kicks self |
| 19:51.01 | Primer | [TK]D-Fender: thanks again, this had been driving me nuts |
| 19:52.27 | [TK]D-Fender | Primer, Tip #1:Look. Tip #2:Look AGAIN. Tip #3:Your eyes suck, show US :p |
| 19:53.25 | Primer | [TK]D-Fender: hardest part of all this was sucking it up and running sip debug again |
| 19:57.06 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 19:57.53 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 20:00.05 | rogers- | Any way to troubleshoot a call that gets disconnected other than looking in "full" ? Every once in a while, a random call will get disconnected mid coversation. full is only showing hangupcall going in the new stack |
| 20:01.19 | navaismo | rogers-, activate sip debug |
| 20:02.19 | rogers- | we're talking 5 calls in the month of december |
| 20:02.41 | rogers- | sip debug logs would be pretty hefty right? |
| 20:04.18 | navaismo | yes your log will increase a lot |
| 20:06.09 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:06.10 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:07.39 | rogers- | and that will log everything to full? |
| 20:08.26 | navaismo | yes |
| 20:10.15 | *** join/#asterisk bmg505 (~leon@196-209-99-65.dynamic.isadsl.co.za) |
| 20:10.42 | saint_ | Anyone using Asterisk at home ? |
| 20:10.54 | saint_ | Specifically in the USA ... What carrier do you have, if you do ? |
| 20:11.11 | saint_ | I have Vonage, they don't do sip. I heard about VoicePulse.. |
| 20:11.23 | file | I use Flowroute |
| 20:11.25 | _Corey_ | Vonage does SIP... just not with you :) |
| 20:11.51 | saint_ | _Corey_: yeah, well.. I do not want a business account, and I do not want to pay extra for the soft phone so I could hack the sip account. |
| 20:12.15 | saint_ | _Corey_: i'm looking for a residential carrier, that provide sip channel, at no crazy cost |
| 20:13.34 | _Corey_ | Plenty to choose from |
| 20:13.40 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 20:13.44 | a1fa | +1 for teliax |
| 20:14.11 | a1fa | they ported successfully both of my numbers from crapvoice |
| 20:14.41 | saint_ | a1fa: how much do you pay /month about ? |
| 20:14.53 | a1fa | $10/month for both numbers |
| 20:14.57 | a1fa | pay as you go + usage |
| 20:15.20 | a1fa | i am keeping my 800 number disconnected; some douche usually dials and hangs around for 30 minutes at the time |
| 20:17.02 | a1fa | i need to fix my voice menus anyway.. i dont think timeout worked well |
| 20:17.17 | a1fa | i was jumping many contexts.. should've had a global timeout |
| 20:17.41 | saint_ | a1fa: mmhh.. voicepulse has actually business sip account but way cheaper than vonage.. 4 chans, 1 did, caller id included, 1.3c/min for outgoing .. not too bad |
| 20:18.11 | a1fa | hard to beat 10/month for two dids |
| 20:18.32 | a1fa | unlimited channels :) |
| 20:19.23 | saint_ | yeah, now i just need to buy a bad ass VoIP PoE phone to try it out. |
| 20:19.52 | Qwell | saint_: Got you covered - http://www1.digium.com/en/products/phones |
| 20:19.56 | saint_ | i guess i could keep the analogs for now. |
| 20:20.16 | saint_ | Qwell: i know that.. right now I have all analog in the house, but My IC Phone in my office |
| 20:20.39 | a1fa | digium still the best? |
| 20:21.00 | a1fa | saint_ : i use and old linksys pap2-na |
| 20:21.57 | a1fa | interesting.. i'm getting a failed to authenticate on inbound calls |
| 20:22.18 | saint_ | a1fa: i m thinking about doing some work in the house, so i minus well hide ethernet cables behind the walls, and run them to the basement where my IT stuff it.. |
| 20:24.25 | a1fa | what is rammification of using insecure=very in sip context? |
| 20:29.16 | *** part/#asterisk dpilon (~dpilon@c-50-138-200-155.hsd1.ct.comcast.net) |
| 20:35.53 | *** join/#asterisk niluje (~niluje@bdv75-4-82-227-67-242.fbx.proxad.net) |
| 20:38.11 | a1fa | i need a hand with my inbound calls |
| 20:38.33 | a1fa | should inbound from any to context be started with exten => s,1,.... |
| 20:39.43 | *** join/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 20:40.45 | Penguin | a1fa: No. |
| 20:40.57 | Penguin | a1fa: It should use whatever extension needs to be used. |
| 20:40.57 | a1fa | it has been a while |
| 20:41.18 | a1fa | its an inbound call |
| 20:41.26 | a1fa | i want pbx to answer it r |
| 20:41.42 | *** part/#asterisk Dovid (~Dovid@ool-43523983.dyn.optonline.net) |
| 20:41.43 | [TK]D-Fender | a1fa, Depends what the call sends IN |
| 20:41.49 | [TK]D-Fender | a1fa, Look at the call. |
| 20:42.14 | Penguin | Right. You have to have an extension for whatever number you are trying to receive calls on. Could be 1, could be 44234432525, could be s. |
| 20:43.11 | a1fa | To sip:s@my-ip |
| 20:44.21 | Penguin | Looks like s. |
| 20:44.31 | a1fa | he |
| 20:44.43 | a1fa | exten => s,1,Goto(local-inbound,s,1) |
| 20:44.52 | a1fa | i used that to move it to a different context |
| 20:45.04 | a1fa | [local-inbound] |
| 20:45.04 | a1fa | exten => s,1,Playtones(dial) |
| 20:45.04 | a1fa | exten => s,2,WaitExten() |
| 20:47.00 | a1fa | Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" |
| 20:47.05 | a1fa | SIP/2.0 487 Request Terminated |
| 20:48.58 | a1fa | hm |
| 20:49.01 | a1fa | i wonder why its terminating |
| 20:51.38 | Penguin | You're doing it wrong. |
| 20:51.47 | a1fa | i think i figured it out |
| 20:52.06 | Penguin | If you're trying to make it so you can dial back out, use Authenticate() followed by DISA(). |
| 20:52.19 | a1fa | its only dialing local extensions |
| 20:52.52 | a1fa | no outbound calls |
| 20:53.46 | a1fa | how do i catch anything? |
| 20:53.59 | a1fa | i want to hang-up any other extension |
| 20:54.24 | a1fa | _ |
| 20:56.40 | a1fa | perfect.. got it working the way i want it |
| 20:57.18 | a1fa | what is the absolute timeout |
| 20:58.05 | *** join/#asterisk navaismo (~navaismo@189.144.207.195) |
| 21:01.18 | a1fa | [TK]D-Fender : autofallthrough=yes is recommended, right? |
| 21:01.39 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 21:03.07 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:08.07 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:18.19 | a1fa | what's a good iax softphone for android? |
| 21:24.17 | a1fa | where can i find skype for *? |
| 21:25.02 | jmetro | http://lmgtfy.com/?q=iax+android+softphone |
| 21:28.17 | a1fa | there are about 1000 results |
| 21:28.26 | a1fa | im looking for some experienced users |
| 21:32.00 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/session) |
| 21:33.25 | *** part/#asterisk Primer (~daniel@www.ceregatti.org) |
| 21:33.51 | saint_ | are the soft keys on the digium ip phone personalizable ? I'm thinking about displaying dynamic content, and eventually program action behind keys ? |
| 21:35.56 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:35.56 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:36.30 | Qwell | summons teh malcolmd |
| 21:42.57 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 21:44.11 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 21:44.54 | *** join/#asterisk IIVQ (~Melusine@a82-161-178-122.adsl.xs4all.nl) |
| 21:45.41 | IIVQ | Hello |
| 21:46.42 | IIVQ | I'm having DTMF trouble. When I use a SIP phone <->Asterisk <-> SIP (to phone provider) then an outside number will recognize my keypresses |
| 21:47.11 | IIVQ | but when I use DTMF phone <--> ATA <--> Asterisk <-> Sip (to phone provider) it won't |
| 21:47.55 | IIVQ | the extensions have dtmfmode = inband, for the external line I've tried inband, auto, frc2833 |
| 21:48.04 | IIVQ | Any idea what this might be? |
| 21:48.47 | IIVQ | Oh, and on the external line: disallow=all, allow=ulaw, allow=alaw. So it *shouldn't* be the codec |
| 21:49.33 | IIVQ | Clearly, Asterisk sees the DTMF from the phone (otherwise I couldn't call out at all), but somehow the DTMF doesn't arrive at the external party. No idea what's the problem though |
| 21:51.05 | leifmadsen | well, note that when you place the call, there is probably not DTMF at all |
| 21:51.35 | leifmadsen | with SIP, the request is typically accepted locally then the entire request is sent as a single INVITE |
| 21:51.46 | IIVQ | leifmadsen, With sip everything works |
| 21:51.47 | leifmadsen | so when you're dialing, it's all local to the device |
| 21:52.08 | IIVQ | It's the analog phone via an ATA that I can dial out with, but cannot access external phone menus with |
| 21:52.17 | IIVQ | haven't tried internal menus though |
| 21:52.38 | leifmadsen | right, the ATA is accepting the DTMF and sending a request |
| 21:52.45 | leifmadsen | asterisk is not in the path at that point of dialing |
| 21:53.07 | leifmadsen | it still sounds like a DTMF issue, but the fact you can place a call isn't proving anything |
| 21:55.08 | IIVQ | ok |
| 21:55.59 | IIVQ | Whenever I look at the Asterisk console during the call, I see messages like Signal=3 whenever I press the 3 button |
| 21:56.09 | IIVQ | another reason I think that Asterisk does receive the "3" |
| 21:57.33 | IIVQ | e.g. http://pastebin.com/FRYs3vz3 |
| 21:58.04 | *** join/#asterisk wltjr (~wltjr@unaffiliated/wltjr) |
| 21:58.06 | IIVQ | note that 192.168.15.1 is both my internet router (gateway) and my ATA (two entirely different roles) |
| 21:58.10 | Qwell | INFO? Use rfc2833. |
| 21:59.03 | IIVQ | Owell, I've tried, same effect |
| 22:04.41 | IIVQ | Strange |
| 22:05.03 | IIVQ | I've re-loaded with rfc2833 and now I get this (sip.conf for the external line is on top) http://pastebin.com/Qa5pnmhv |
| 22:05.16 | IIVQ | does that mean that It's still sending out SIP INFO? |
| 22:24.13 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 22:56.23 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:01.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:08.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:13.58 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 23:20.47 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-jflkcjsiiclgyyvw) |
| 23:28.14 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 23:38.23 | *** join/#asterisk deo_ (~deo@112.198.79.121) |
| 23:57.48 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.133) |