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| 00:06.15 | raden_work | exten => _*72.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:3}) |
| 00:06.21 | *** join/#asterisk doneir (~cbrunker@appenp.lnk.telstra.net) |
| 00:06.44 | raden_work | instead of setting that t\with {CALLERID(num) is there a way to set to the reg extension ? |
| 00:09.16 | doneir | hey all, having an issue and any advice would be appreciated. I've just receieved my TE220 for a fragmented 10 chan E1 setup and am trying to setup the span, unfortunately i'm receiving Red/Blue alarm (dahdi_tool output), my system.conf file is http://pastebin.ca/1807517 . I've done a hardware loopback test, all fine (so hardware seems ok). I'm using dahdi 2.2.1 |
| 00:09.39 | doneir | From what i've read, it seems Blue alarm indicates issues with the provider (e.g line not provissioned correctly), is that correct? |
| 00:09.50 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:10.57 | voipmonk | unused or 'clear' ? |
| 00:12.01 | doneir | sorry, i take it there's a difference? unused represents unallocated lines, whereas clear is for lines which could be used but aren't? |
| 00:12.05 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:13.10 | doneir | changing to clear provides to difference |
| 00:17.30 | doneir | s/to/no/ |
| 00:17.53 | doneir | well, more on the second match, but thanks mr.infobot |
| 00:17.57 | doneir | ;] |
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| 00:20.45 | p3nguin_ | I wonder if it understands that. |
| 00:21.02 | p3nguin_ | changing to clear provides to difference |
| 00:21.03 | p3nguin_ | s/to/no/2 |
| 00:21.13 | doneir | guess not |
| 00:21.13 | p3nguin_ | IT DOESN'T WORK! |
| 00:21.44 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 00:21.49 | p3nguin_ | Must be some broken implementation of sed. |
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| 00:22.54 | bmoraca_work | it understands /g |
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| 00:40.44 | doneir | http://pastebin.ca/1807551 <--- /proc/dahdi/1 output |
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| 00:40.57 | doneir | doesn't display blue alarm, though dahdi_toll does show "Blue/Red" |
| 00:41.14 | doneir | is it normal to have two alarm colours on the same card at the same time? |
| 00:41.14 | doneir | o_O |
| 00:42.06 | doneir | err, dahdi_tool |
| 00:44.13 | doneir | hrm |
| 00:46.32 | doneir | would a shared IRQ lead to blue error? Or only show a red error? It was my thought blue error was receivership of error signal from the provider |
| 00:47.25 | ManxPower-work | doneir: A BLUE Alarm is a TELCO problem. Report it to your provider. |
| 00:47.39 | doneir | thank you, i just needed clarification :) |
| 00:49.39 | *** join/#asterisk FinboySlick (~shark@74.117.40.9) |
| 00:55.24 | bmoraca_work | woo, it's day over time! |
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| 00:56.43 | Blackgibson | I'd like a clarification regarding call waiting please. If a call comes on line A, and gets forwarded to cell phone b via my Asterisk box, is does line A become available when the forwarded call is connected or is it in use untill the call has been completed? |
| 00:58.02 | Blackgibson | er not call waiting, call forwarding. duh |
| 01:02.40 | cidu | umm, from what i see in daily use, the forward keeps both lines open |
| 01:02.42 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 01:05.04 | Blackgibson | thanks. trying to plan out how many total pstn trunks I need for this new asterisk server and Im very new to telephony |
| 01:05.49 | cidu | have fun, its a deep dark hole with so much wonder and frustration... terrifying and beautiful |
| 01:07.52 | Blackgibson | heh. so far so good. I have one line up and running, have extensions created and working, have an IVR going.. |
| 01:08.11 | coppice | that's an interesting marketing spin on the more pragmatic "its tedious and boring" |
| 01:08.28 | *** join/#asterisk chazzm (~chazz@173-24-238-25.client.mchsi.com) |
| 01:08.31 | Blackgibson | just trying to figure out how im going to set up call forwarding to mobile technician's cell phones |
| 01:08.39 | cidu | coppice, i got no vested intrest, i just love rutting through it |
| 01:09.17 | cidu | you dug through "Asterisk without tears"? |
| 01:09.41 | cidu | free oriley book, might not cover all the little oddities, but its a really good way to start |
| 01:10.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 01:11.07 | Blackgibson | I have poked thru it when needed. It isn't so much actually setting up the forwarding, its making sure i don't tie up all the lines in the office with forwarded calls becasue I told the boss "na, we dont need any more lines" |
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| 02:40.11 | Katty | hi |
| 02:40.16 | carrar | hi!! |
| 02:42.15 | Katty | carrar: would you like to see my latest video feed?\ |
| 02:42.22 | carrar | Well of course I would |
| 02:42.32 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 02:42.34 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 02:42.56 | Katty | carrar: http://ustre.am/bEBU |
| 02:43.51 | carrar | I used to have a ferret |
| 02:43.56 | Katty | just one? :< |
| 02:44.00 | Katty | adjusts camera |
| 02:44.01 | carrar | yeah |
| 02:44.10 | carrar | a sable |
| 02:44.39 | carrar | holy moving camera |
| 02:44.58 | Katty | yeah i'm trying to get a better view |
| 02:44.59 | Katty | hang on |
| 02:45.22 | carrar | Can't see the lower ferrets |
| 02:45.46 | carrar | that'll work |
| 02:45.52 | Katty | anything below that is just the litter |
| 02:46.17 | Katty | WHAT IS THAT MYSTERIOUS GREY THING |
| 02:46.22 | Katty | THAT WILL NOT DO |
| 02:46.30 | carrar | heh |
| 02:46.45 | Katty | k the little brown guy on the right |
| 02:46.47 | Katty | middle of the screen |
| 02:46.49 | Katty | that's BB |
| 02:46.57 | Katty | the moving thing, under the blanket next to him, is sammy |
| 02:47.15 | Katty | pippin is down by the food bowl |
| 02:47.33 | Katty | and merry is probably in the top pink carrier |
| 02:47.49 | Katty | oh and the big dog is obviously riddick |
| 02:48.01 | carrar | Where do you keep the horses? |
| 02:48.08 | vader-- | any of you guys running cisco 7940g phones? |
| 02:48.10 | Katty | out back in the stables |
| 02:48.14 | carrar | heh |
| 02:48.16 | Katty | i'll put a camera on them next |
| 02:48.21 | carrar | HOT |
| 02:48.26 | carrar | vader, a lot of people do |
| 02:48.49 | vader-- | i know that |
| 02:48.50 | carrar | some evern run 7941's! |
| 02:48.54 | vader-- | im looking for someone who has them here |
| 02:48.59 | vader-- | i have a configuration file question |
| 02:49.04 | vader-- | about the backup proxy |
| 02:49.08 | carrar | not much to their config |
| 02:49.48 | vader-- | im wondering is it possible if the phone is inside the network it will try and connect to an internal ip address and if it can't connect to that when plugged in at someone's house it will use an external address |
| 02:49.59 | doneir | contacted the telco, they've acknowledged it's a fault on their network and are looking into fixing it. Thanks for the help |
| 02:50.02 | vader-- | i see a backup proxy setting |
| 02:50.33 | carrar | vader, try it, lets us know |
| 02:50.33 | vader-- | outbound_proxy would be internal ip and proxy_backup would be external ip? |
| 02:51.36 | carrar | don't forget your proxy1_address & proxy2_address |
| 02:52.33 | vader-- | whats the difference between the outbound proxy and proxy1? |
| 02:52.43 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 02:53.06 | hardwire | this is making me grumbly. Using asterisk 1.6.2 dialing a named iax2 connection dials the correct host specified by host= but attempts to use auth that is in [guest] or [iaxtel] if I disable guest |
| 02:57.42 | *** join/#asterisk QAChip (~Christian@201.122.75.250) |
| 02:58.21 | QAChip | hello everyone, a question here, any advice about softphones for linux? |
| 02:58.31 | QAChip | which one is recommended |
| 02:58.32 | QAChip | ? |
| 02:58.59 | Katty | infobot: phones? |
| 02:59.00 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 02:59.15 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 02:59.22 | Katty | adnc_: YOU |
| 02:59.33 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 02:59.33 | adnc_ | Katty, ??? |
| 02:59.37 | adnc_ | yes me |
| 02:59.39 | Katty | adnc_: why i outta just.... |
| 02:59.42 | Katty | adnc_: HUG YOU TO BITS |
| 02:59.44 | Katty | hugs adnc_ |
| 02:59.52 | adnc_ | Katty, ;) |
| 03:00.00 | adnc_ | how are you doing today? |
| 03:00.06 | carrar | QAChip, I've had good luck with the free X-Lite phones |
| 03:00.17 | Katty | adnc_: super. http://ustre.am/bEBU <- my latest feed. |
| 03:00.28 | adnc_ | let me see it |
| 03:00.34 | Katty | adnc_: they just got done playing...so they're mostly sacked out |
| 03:00.52 | adnc_ | Katty, ow |
| 03:00.53 | adnc_ | wow |
| 03:00.59 | adnc_ | where is this? |
| 03:01.03 | Katty | my basement |
| 03:01.30 | adnc_ | what camera is this? |
| 03:01.35 | Katty | logitech c600 |
| 03:01.47 | Katty | broadcasting at 640x480, 10fps, 350kbps upload speed. |
| 03:01.58 | adnc_ | nice images |
| 03:02.06 | adnc_ | logitech makes network cameras? |
| 03:02.13 | Katty | that's not a network camera silly |
| 03:02.15 | Katty | that's a webcam |
| 03:02.29 | adnc_ | ohh |
| 03:02.36 | Katty | it can handle full 720p, but i don't hae the bandwidth for that |
| 03:02.38 | adnc_ | the images are really nice |
| 03:02.51 | Katty | thanks (= |
| 03:02.56 | Katty | i have to keep an eye on my fuzzballs. |
| 03:03.06 | adnc_ | are they called fuzzballs? |
| 03:03.09 | *** part/#asterisk lanning (~lanning@208.87.235.224) |
| 03:03.09 | adnc_ | in english? |
| 03:03.17 | Katty | no, they're called Ferrets in english |
| 03:03.20 | Katty | but i call them fuzzballs |
| 03:03.28 | adnc_ | ferrets |
| 03:03.32 | Katty | mhmm |
| 03:03.44 | adnc_ | one more english word |
| 03:03.51 | Katty | (= |
| 03:04.08 | adnc_ | bad what is that purple/black bag in the back? |
| 03:04.21 | Katty | lower level or upper level |
| 03:04.26 | adnc_ | upper |
| 03:04.32 | Katty | back right? |
| 03:04.36 | adnc_ | yes |
| 03:04.40 | Katty | that's a pet carrier |
| 03:04.50 | adnc_ | they just did drink water, nice ;) |
| 03:05.43 | Katty | yeah it's just a basic soft sided pet carrier that's been retired |
| 03:05.51 | Katty | it gives them a cozy place to hide (= |
| 03:05.56 | Katty | there's another one on the lower level |
| 03:06.40 | adnc_ | ;) |
| 03:07.13 | Katty | i think they've all zonked out now |
| 03:07.28 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 03:07.49 | Katty | Zzzzz naptime |
| 03:07.55 | QAChip | today I just met sipgate, and It seems great, but I'm sure I can dig deeper and find more options, any hint_ |
| 03:07.57 | QAChip | ? |
| 03:08.28 | Katty | google. |
| 03:08.50 | Katty | adnc_: MOVEMENT from lower level ept carrier! |
| 03:08.54 | Katty | s/ept/pet/ |
| 03:09.15 | adnc_ | moevement? |
| 03:09.23 | adnc_ | ohh moment |
| 03:09.28 | Katty | no, movement. |
| 03:09.30 | Katty | it was moving. |
| 03:09.45 | Katty | guess someone wasn't comfortable enough |
| 03:09.55 | adnc_ | ahh |
| 03:10.03 | Katty | what's your primary language? |
| 03:10.12 | adnc_ | Katty, german and turkish |
| 03:10.16 | Katty | k |
| 03:11.09 | *** join/#asterisk fofware (~chatzilla@host147.190-229-137.telecom.net.ar) |
| 03:11.11 | adnc_ | i had that page open but unfortunately i wasn't watching at that moment, i suppose the ferret did go up? |
| 03:11.20 | *** join/#asterisk Dimitripietro (~fasfda@122-110.al.cgocable.ca) |
| 03:11.47 | adnc_ | can you please ask them to go drinking water? that looks funny |
| 03:11.51 | adnc_ | ;) |
| 03:12.41 | Katty | lol |
| 03:12.44 | Katty | i don't want to wake them |
| 03:12.51 | Katty | they've had a full evening of scampering about |
| 03:13.54 | adnc_ | Katty, just joking, we should leave them there peace |
| 03:16.28 | Katty | i'ma bout to go pass out myself |
| 03:17.22 | *** join/#asterisk fofware (~chatzilla@host147.190-229-137.telecom.net.ar) |
| 03:22.57 | *** join/#asterisk trnzmeta (~bleh@secure27.lnk.telstra.net) |
| 03:23.14 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 03:27.41 | adnc_ | Katty, bye bye |
| 03:29.51 | Dimitripietro | Any 2bct user here ? |
| 03:32.29 | joako | I am trying to setup a gateway from PRI to SIP with Asterisk 1.4 and I am having massive issues with calls not going through correctly. How can I resolve this? |
| 03:32.54 | joako | I am getting odd errors such as ] WARNING[3101]: chan_dahdi.c:8652 pri_fixup_principle: Call specified, but not found? |
| 03:38.14 | *** join/#asterisk [netman] (~netman@83.Red-88-8-163.dynamicIP.rima-tde.net) |
| 03:42.40 | etfonhomey | joako, which T1 card are you using? |
| 03:42.52 | joako | etfonhomey, Sangoma A101 |
| 03:43.17 | etfonhomey | Have you contacted their support? Their support is pretty good. |
| 03:44.44 | joako | Yes I am not happy with them |
| 03:44.54 | etfonhomey | joako, what did they tell you? |
| 03:45.28 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 03:45.28 | joako | I had the card already in 2 machines, one with CentOS 5.3 and then 5.4 and another openSUSE 11.2 and all of them have issues one way or another |
| 03:45.44 | joako | they want logs which I can't provide this is at a business and when it goes down they need to reboot it so they can get calls |
| 03:46.12 | joako | The actual PBX where the phones attach to is running Asterisk GUI. |
| 03:46.35 | joako | What is the best way to setup the SIP between the two? Right now I am doing it by IP address without any password or username |
| 03:46.42 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 03:47.52 | etfonhomey | joako, I don't understand your question. Bridging PRI to SIP is just a matter of call routing in your extensions.conf, I would assume. |
| 03:48.35 | joako | etfonhomey, Yes but for example if I put _X.,1,Dial(SIP/${EXTEN}@192.168.1.1) there are issues with call supervision. It has to be _X.,1,Dial(SIP/${EXTEN}@192.168.1.1,120,r) |
| 03:48.59 | joako | and right now I am having issues with call supervision. Call from PBX -> PSTN and the call sometimes when it is answered at the other end drops |
| 03:49.28 | joako | So what is why I want to know is there a configuration that will work without problems for setting up a PRI to SIP gateway? |
| 03:50.31 | etfonhomey | joako, is this a setup from scratch? Or were they using the PRI to with some other phone system? |
| 03:51.56 | joako | PRI was setup directly on a Dell machine with the PBX, it was having issues 2-3 times a day where calls stop going in/out of the PRI. the command "pri restart" sometimes fixed the issue. I replaced the cabling and that did not help. |
| 03:52.05 | joako | PBX = asterisk w/ Asterisk GUI |
| 03:52.35 | joako | So I moved the T1 card to another machine openSUSE minimal install, asterisk build with minimal modules. |
| 03:53.47 | etfonhomey | So, you have never had this PRI working cleanly in any setup Asterisk or not? |
| 03:54.33 | joako | It worked initially for a few months without any issues |
| 03:55.03 | etfonhomey | Dell server, Asterisk GUI, and Sangoma A101? |
| 03:55.10 | joako | Also this is not a "real T1" this is a VOIP service and the carrier installs a Cisco IAD and that is what is connected to the T1 card |
| 03:55.47 | joako | etfonhomey, Correct that was the initial setup. Now I have moved the A101 to another machine that is only acting as a gateway |
| 03:56.24 | etfonhomey | joako, I have the same setup in a non-Asterisk phone system. SIP to a Cisco IAD that hands off a PRI. |
| 03:56.49 | etfonhomey | joako, And nothing changed whatsoever after the few months of working? |
| 03:57.49 | joako | etfonhomey, CentOS updates, but that is it. I tried before I moved it to upgrade CentOS 5.3 -> 5.4 (it shows up with the normal updates), Asterisk, DAHDI & sangoma drivers |
| 03:58.25 | joako | Nooo in this case the Cisco IAD gets it via VOIP (MGCP) and hands it off over PRI. It is retarded. The provider is retraded, they could reconfigure the Cisco to talk with us via SIP but they won't. |
| 03:58.51 | joako | I mean they will but they say you need to order another line and cancel the current contract.... XO Communications. |
| 03:59.09 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 04:00.04 | etfonhomey | My provider is Nuvox. It's probably MGCP as well. I never confirmed. They do sell SIP service now as well, I think. |
| 04:00.17 | etfonhomey | Does AsteriskGUI use DAHDI or Zaptel? |
| 04:01.31 | joako | etfonhomey, DAHDI |
| 04:01.33 | etfonhomey | If DAHDI, pastebin your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf files. |
| 04:01.55 | etfonhomey | I will see if anything sticks out to me. |
| 04:02.06 | joako | etfonhomey, Well I am not running GUI on the gateway machine.... |
| 04:02.25 | joako | But I think right now the issues between the system might be SIP related |
| 04:03.29 | [TK]D-Fender | joako: It is. Your IAD isn't reporting back 180 Ringing which is why you're forcing it on the other side. Not "nice" |
| 04:03.52 | joako | [TK]D-Fender, I always see in the CLI it appears to be passing it correctly |
| 04:04.37 | [TK]D-Fender | joako: Sorry.. the SIP leg |
| 04:05.30 | joako | [TK]D-Fender, So again I go back to the question can someone provide a known working configuration for SIP between two asterisk machines |
| 04:07.21 | [TK]D-Fender | joako: Wait.. this is 2 *s'? |
| 04:07.47 | [TK]D-Fender | joako: * won't pass incomplete progress between just themselves.. its an exterior leg thats responsiblle |
| 04:07.55 | [TK]D-Fender | joako: * would just be a middle-man |
| 04:08.02 | etfonhomey | [TK]D-Fender, your're saying that the IAD is not sending the PRI Ringing message to Asterisk for it to pass as a 180 Ringing to the SIP peer? |
| 04:08.13 | joako | [TK]D-Fender, Yes after having issues with the T1 card in one machine I moved it to a 2nd machine to just act as a gateway from PRI to SIP |
| 04:09.00 | *** join/#asterisk Kumbang (~dsp@rusnas.paume.itb.ac.id) |
| 04:09.19 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 04:10.07 | Katty | yawns |
| 04:10.14 | etfonhomey | joako, Can you recreate the problem right now? |
| 04:11.16 | joako | It is not a consistant thing but yes many times when I dial a PSTN number the call drops when it is answered but no I am not in the office and I do not have a phone I can dial out from this system with |
| 04:13.41 | etfonhomey | So, your setuip is this? XO<-> IAD <---- PRI ----> Asterisk1 <----SIP---> Asterisk2 <----SIP----> Test Phone |
| 04:15.07 | joako | etfonhomey, Correct |
| 04:15.16 | etfonhomey | And the issue is an outgoing call to the PSTN from "Test Phone" that drops when answered? |
| 04:15.47 | joako | etfonhomey, yes but not every time |
| 04:20.03 | joako | Besides the provider what would cause some calls I see "DAHDI/3-1 is ringing" and some calls only " DAHDI/3-1 is proceeding passing it to SIP/hsnp-00000016" |
| 04:21.27 | joako | Ok yes these dropped calls I see DAHDI/2-1 is ringing, then long after it was actually answered I see "DAHDI/2-1 answered SIP/hsnp-00000018" |
| 04:22.54 | etfonhomey | joako, do you Answer() the PRI call before bridging to the SIP channel? |
| 04:23.25 | joako | No. |
| 04:25.16 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:25.29 | etfonhomey | You might try this. 1. Test Phone dials external number. 2. Asterisk2 Answer()'s it, then dials Asterisk1. 3. Asterisk1 Answer()'s the call then dials PSTN through PRI. |
| 04:26.16 | joako | I am sure that will work but the issue is that I am not seeing the correct ringing indication from the PRI when placing calls to the PSTN. Maybe Answer() will work but I do not like that solution because that is not "correct" |
| 04:26.39 | joako | The question now is, can this be my configuration or it it just the carrier and their IAD that is the issue? |
| 04:27.00 | etfonhomey | Have you turned the PRI debuging up on Asterisk2? |
| 04:27.10 | etfonhomey | And watched what you get from the PRI? |
| 04:28.32 | joako | Not yet. |
| 04:30.18 | etfonhomey | I would would want to see the CLI output with SIP debug turned on as well as PRI debug to see the exchange of messages. Especially on Asterisk2. |
| 04:31.09 | etfonhomey | I would want to see that from a working call vs. a failed call. |
| 04:31.45 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 04:33.10 | etfonhomey | I had behavior like that on a Cisco CallManager setup with CallForward All set to an external number. As soon as the user would answer the forwarded call at the external number, the call would go to their voicemail. |
| 04:33.56 | etfonhomey | But, that was SCCP/MGCP/PRI rather than SIP/SIP/PRI. |
| 04:34.38 | joako | Here is just one that failed completely in an odd way: http://pastebin.ca/1807746 |
| 04:36.29 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 04:37.21 | joako | Ok now I get PRI debug messages but calls do not come in. Nothing is shown in CLI w/ verbose 3 |
| 04:39.13 | etfonhomey | joako, I see something weird. |
| 04:39.25 | joako | etfonhomey, Look at this one: http://pastebin.ca/1807753 |
| 04:39.38 | etfonhomey | When you send the INVITE to 192.168.1.1 the first time, it says Reliably Transmitting (no NAT) to 192.168.1.1:5060: |
| 04:40.03 | joako | etfonhomey, Correct they are on the same LAN with no NAT. |
| 04:40.31 | joako | etfonhomey, But can you look at the pri debug I just posted? If this issue with the calls not coming in is still happening, I could care less about SIP. |
| 04:40.35 | etfonhomey | When you send the TRYING to 192.168.1.1, it says <--- Transmitting (NAT) to 192.168.1.1:5060 ---> |
| 04:42.09 | etfonhomey | Gimme a few to look through these. |
| 04:42.28 | joako | etfonhomey, Probably missing nat=no on one end? |
| 04:43.16 | joako | But like I said if this is happening where the calls do not come in and not even and indication of that at verbose 3, there is something wrong with the PRI connection somewhere and that really needs to be sorted out before I even start to look at what is wrong with SIP... or just put the T1 card back where it was originally |
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| 04:46.33 | joako | I might be away from the PC for a bit, but I will still be around... |
| 04:48.37 | etfonhomey | joako, these incoming calls look like they are failing on your end to me. |
| 04:48.59 | joako | etfonhomey, How would I troubleshoot that? |
| 04:49.47 | etfonhomey | Your card may not be setup correctly. |
| 04:50.09 | etfonhomey | What does "dahdi show status" show? |
| 04:50.40 | etfonhomey | and "pri show spans" |
| 04:52.25 | joako | dahdi show status = wanpipe1 card 0 OK 0 0 0; pri show spans = PRI span 1/0: Provisioned, Up, Active |
| 04:52.36 | joako | It was configured using the sangoma autoconfigurator |
| 04:53.42 | etfonhomey | OK. Pastebin your chan_dahdi.conf and your extensions.conf |
| 04:54.12 | etfonhomey | The PRI is up. You might not be handling the incoming call correctly. |
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| 05:03.26 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-mwdgyyrpvjnelthe) |
| 05:04.19 | etfonhomey | joako? |
| 05:09.25 | *** part/#asterisk FinboySlick (~shark@74.117.40.9) |
| 05:15.29 | joako | Yes I am getting those files. |
| 05:16.40 | joako | etfonhomey, http://pastebin.ca/1807784 |
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| 05:20.50 | kazaa_lite | hi all |
| 05:21.38 | kazaa_lite | how to change colour of asterisk CLI? i find it hard to read light grayish text in 1.6.2.x |
| 05:23.19 | ChannelZ | that's up to your terminal |
| 05:25.14 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 05:27.25 | kazaa_lite | it was ok with 1.4.x and 1.6.1.x |
| 05:29.10 | kazaa_lite | like: NOTICE[3335]: cdr.c:1551 do_reload: CDR simple logging enabled. << "1551 do_reload" appears in light grayish text when I boot asterisk |
| 05:30.00 | etfonhomey | joako, I don't see anything glaring. Your call handling looks fine and is pretty basic. I would really try to work with Sangoma's support and get them all of the information they need. |
| 05:32.30 | *** join/#asterisk fofware (~chatzilla@host147.190-229-137.telecom.net.ar) |
| 05:59.40 | joako | TÄ¥ere is something seriously wrong with this |
| 05:59.41 | joako | <PROTECTED> |
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| 06:42.33 | norrec | under the asterisk cli i keep getting a bunch of remote unix connections and dcs, is there any way to determine where thoes connections are coming from? |
| 06:44.27 | p3nguin_ | You're probably using 'asterisk -rx "some command"' in scripts. |
| 06:47.05 | norrec | is there any way to determine the command being run? |
| 06:47.25 | p3nguin_ | Did you not configure the system yourself? |
| 06:49.31 | norrec | not entirely... |
| 06:50.14 | russellb | using freepbx? |
| 06:50.27 | norrec | no |
| 06:50.46 | norrec | but there is a webgui, however its connections look different |
| 06:51.52 | themolester | p3nguin_ do you remember the problem i was having earlier? 20 second dropout issue? |
| 06:53.11 | themolester | i ran a wireshark cap and it seems to go like this trunk > * options, * > phone bye |
| 06:53.11 | p3nguin_ | themolester: Yes. Did you track down the problem? |
| 06:53.21 | norrec | russellb the webgui logs though manager |
| 06:53.27 | norrec | logs in* |
| 06:53.51 | norrec | where the other one is just a remote unix connection |
| 06:53.54 | themolester | inside the bye the x-asterisk-hangupcause is "normal clearing" but the "expert info" shows "unrecognised sip header" |
| 06:56.32 | themolester | so, asterisk is definitely initiating the problem telling the phone to hangup |
| 06:57.06 | p3nguin_ | That's not very nice of it, considering you are trying to make a call. |
| 06:57.19 | themolester | sorry, it's options, ok, THEN bye |
| 06:57.38 | p3nguin_ | It's like the inconsiderate wife/kid/friend who likes to talk really loud when you're on the phone. |
| 06:58.26 | themolester | yea, makes me want to beat up the pound key over and over |
| 06:58.44 | norrec | haha, well i figured out my problem... it was just my own stupidity, forgot a had a watch command going... |
| 06:58.45 | p3nguin_ | I bet if you can get some of the devs to pay attention, they could give you some insight on the matter. |
| 06:59.57 | themolester | you know whats really aweful though? |
| 07:00.22 | *** join/#asterisk grEvenX (~even@apb91b.ip.ssc.net) |
| 07:00.48 | themolester | i kind of expected a ton of problems with setup so when my boss wanted to get a pbx, i built my own, set up accounts at all the same places, etc |
| 07:01.10 | themolester | i got all set up with ZERO issues... then I set up at work expecting smooth sailing, and its nothing but trouble |
| 07:01.42 | *** join/#asterisk kamh (~kamh@xdsl-1814.wroclaw.dialog.net.pl) |
| 07:01.45 | themolester | and at home, it's just an extra thing that I really don't need... at work, its not lol |
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| 07:08.56 | themolester | hrmm... everywhere I'm reading seems to imply that "unrecognised sip info" means nothing, and it should work regaurdless |
| 07:09.28 | themolester | though, it certianly doesn't SOUND like its nothing |
| 07:11.58 | norrec | are you sure your sip trunk is configured correctly? |
| 07:12.35 | themolester | it is configured the way flowroute says on their website, but i'll pastebin it for a second opinion |
| 07:12.46 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:12.55 | norrec | k, whos the provider? |
| 07:13.07 | p3nguin_ | lol |
| 07:13.17 | norrec | ? |
| 07:13.27 | themolester | flowroute |
| 07:13.35 | norrec | oh |
| 07:13.44 | norrec | thats why p3nguin_ was laughing |
| 07:13.44 | norrec | haha |
| 07:13.46 | norrec | missed that |
| 07:14.12 | p3nguin_ | I would have laughed more if he would have said some other name. |
| 07:14.43 | norrec | also, you may want to make sure your passing cid/correct cid, some providers require a cid of a number you own though them |
| 07:14.47 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:14.58 | norrec | granted its odd that its not passing you back an error |
| 07:15.10 | themolester | flowroute doesn't, though you missed the first part ... its incoming calls exclusively |
| 07:15.17 | norrec | ah |
| 07:15.25 | p3nguin_ | thinks norrec needs to come up to speed with the rest of this situation. |
| 07:15.56 | norrec | agrees with p3nguin_ |
| 07:15.59 | themolester | http://pastebin.com/xJtC4KhQ |
| 07:16.28 | themolester | norrec issue is, incoming calls will drop 20 seconds (exact) after phone is picked up |
| 07:16.45 | themolester | regaurdless of ring time (i could leave someone on hold/ring indefinitely) |
| 07:16.56 | themolester | outgoing is perfect |
| 07:17.29 | themolester | oh, and there is perfect 2 way audio until the disconnect/hangup |
| 07:17.38 | p3nguin_ | What's with the odd port number? |
| 07:18.19 | p3nguin_ | Do they really operate on the non-standard port? |
| 07:18.29 | themolester | asterisk retried a crapload of times, and 5060 stopped working, they had that port for odd isps |
| 07:18.53 | themolester | they run on standard as well |
| 07:19.13 | themolester | i'll probably be able to switch back to 5060 in a month or so |
| 07:19.20 | p3nguin_ | http://pastebin.com/17RxCSYP |
| 07:19.32 | p3nguin_ | There's what I would try without even looking at their examples. |
| 07:20.07 | norrec | yeah, i was about the suggest the same thing lol |
| 07:20.46 | themolester | any reason to change to nat=no |
| 07:20.56 | p3nguin_ | Yes. They aren't behind nat. |
| 07:21.27 | themolester | but I am... at least thats the way I understood it, possibly from crackpots |
| 07:21.45 | p3nguin_ | Your NAT setting goes in the general section. |
| 07:21.45 | norrec | if you are, you need nat=yes under global |
| 07:21.50 | norrec | er general |
| 07:22.02 | themolester | ok, so nat=yes in general, but not in trunk? |
| 07:22.13 | p3nguin_ | They aren't behind nat, so nat=no. |
| 07:22.26 | themolester | k, gonna try |
| 07:22.50 | p3nguin_ | If you are behind nat, put the correct nat, localnet, and externip or externhost in general. |
| 07:22.58 | norrec | it kinda seems like a firewall problem... |
| 07:22.58 | themolester | also done |
| 07:23.10 | norrec | since the call starts but then gets cut off |
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| 07:23.36 | themolester | norrec i am no sip expert, but wouldn't it just not work? |
| 07:23.36 | p3nguin_ | I would start with the config I pasted, then see what needs changed from there. |
| 07:23.38 | norrec | did you forward 10000-20000 udp to the server? |
| 07:23.42 | themolester | yes |
| 07:23.57 | norrec | hmm, yeah give p3nguin_'s config a try and let us know |
| 07:24.09 | p3nguin_ | Don't forget sip reload after changing and saving. |
| 07:24.57 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
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| 07:27.21 | themolester | hrmm |
| 07:27.36 | themolester | that kind of worked, but now i have no audio |
| 07:27.46 | themolester | got past 20s though |
| 07:28.05 | p3nguin_ | Interesting. |
| 07:28.51 | themolester | hold on... |
| 07:29.12 | themolester | nope... first was a fluke |
| 07:29.16 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:29.21 | themolester | second got audio, and 20s disconnect |
| 07:29.26 | p3nguin_ | erm |
| 07:29.35 | p3nguin_ | You're using my suggested config? |
| 07:29.40 | themolester | yep |
| 07:29.55 | themolester | including the 5060 standard port |
| 07:30.00 | p3nguin_ | Did you ever provide a full sip debug? |
| 07:30.00 | themolester | seems its been unblocked |
| 07:30.22 | themolester | yea, you said you didn't see anything obvious... but i could post again |
| 07:30.50 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:30.56 | norrec | did u post the packet capture? |
| 07:30.58 | p3nguin_ | I don't recall seeing it. |
| 07:31.30 | themolester | I didn't post the capture, not sure how to filter my sensitive info |
| 07:32.17 | norrec | export the packets used to start the call as text and then delete ur info |
| 07:32.37 | p3nguin_ | In a sip debug, no private information should be shown. |
| 07:34.06 | themolester | p3nguin_ user/pass is shown on register if timing is right |
| 07:34.25 | p3nguin_ | The pass would be hashed, which is of no use. |
| 07:34.50 | themolester | sec |
| 07:34.53 | p3nguin_ | Everything else isn't private. The username is transmitted on the public internet in the clear. |
| 07:38.50 | themolester | hmmm.. odd |
| 07:39.36 | themolester | going to remote restart the router (scary) |
| 07:39.50 | themolester | i can't connect remotely anymore, worked earlier today |
| 07:40.11 | themolester | my isp might be fucking with me, so we pay their excessively high voip service |
| 07:41.22 | *** join/#asterisk fofware (~chatzilla@190.229.137.147) |
| 07:41.29 | themolester | nm, filter reload worked |
| 07:41.35 | themolester | lemme see if it fixed anything else |
| 07:43.15 | p3nguin_ | I can't even imagine rebooting my router. |
| 07:43.45 | themolester | i don't think its the router, i think its the isp who claims they don't ever block anything, for any reason |
| 07:44.10 | p3nguin_ | <PROTECTED> |
| 07:44.41 | themolester | yea, my server looked like that once, then I rebooted and it never came back up |
| 07:44.42 | themolester | :-x |
| 07:44.51 | themolester | knocks on wood |
| 07:45.11 | p3nguin_ | I don't reboot most of my stuff. Sometimes the power goes off for longer than the UPSs can run. |
| 07:46.11 | themolester | well, by rebooted, i mean i shut it down to install something, and it never posted again... even after removing the thing i installed |
| 07:48.12 | norrec | whats rebooting? |
| 07:49.15 | norrec | only thing i do that to is my damn windows box... i hate keeping it around but i gotta play my games on something =/ |
| 07:50.10 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
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| 07:51.26 | themolester | hmmm |
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| 07:56.13 | p3nguin_ | Hmm, my Windows server has only been up 53 Days, 10 Hours, 11 Minutes, 42 Seconds. |
| 07:56.23 | p3nguin_ | Wonder what happened there. |
| 07:56.33 | vader-- | is anyone using and/or has seen microsoft OCS/Exchange used as a PBX system? |
| 08:01.09 | norrec | p3nguin_: it was prob just being ms and decided to restart itself lol |
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| 08:05.01 | themolester | hmmm |
| 08:05.32 | themolester | i think I need to either get an extra ip address to accomodate a 1:1, or get a new isp |
| 08:05.58 | themolester | cause some of my forwards aren't working anymore |
| 08:08.19 | p3nguin_ | I ran into that earlier. |
| 08:08.30 | p3nguin_ | Turned out that I already had the ports forwarded to another host. |
| 08:11.05 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:12.30 | themolester | unfortunately thats not it |
| 08:12.44 | p3nguin_ | Too bad -- it would have been an easy fix. |
| 08:13.53 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 08:19.16 | norrec | u should put your server on the dmz first and see if that fixes the problem |
| 08:22.02 | norrec | but it does sound like a firewall/nat issue... make sure you have sip alg disabled |
| 08:22.28 | norrec | and any other sip features, like sip transformations and such |
| 08:22.44 | norrec | also depending on your firewall, you may want to disable spi |
| 08:23.24 | norrec | if ur using something like a linksys or netgear disable it, if your useing an enterprise class make sure your access rules and udp timeouts are set properly |
| 08:25.33 | norrec | and i would also contact the company who provides your sip trunk, they may be able to shed a bit more light on what is going on, on their end |
| 08:26.19 | p3nguin_ | I can't stand that term... "sip trunk." |
| 08:26.37 | norrec | what do you prefer to call it? |
| 08:27.17 | p3nguin_ | I guess it depends on what you really mean. There is no such thing as a sip trunk, so your guess is as good as mine. |
| 08:28.21 | norrec | yeah, but i just go by what the industry calls it =/ or at least the providers i deal with call it |
| 08:29.18 | norrec | but i'm refering to the connection between themolester's inbound provider and his server that is negotiatied with SIP lol |
| 08:30.10 | p3nguin_ | I was pretty sure that was what you were talking about, but it doesn't change the fact that SIP doesn't do trunking. |
| 08:30.35 | norrec | indeed |
| 08:30.43 | themolester | i've always heard that phrase as well |
| 08:30.49 | themolester | if thats not correct, what is? |
| 08:31.08 | p3nguin_ | I'm trying to figure out how to quantify the term. |
| 08:32.13 | norrec | termination and origination via sip? |
| 08:32.13 | norrec | lol |
| 08:32.27 | norrec | of voice via sip* |
| 08:33.02 | themolester | sounds like a mouthful, though i've heard those terms as well, usually separately |
| 08:33.05 | p3nguin_ | I don't know. I can't come up with anything. |
| 08:33.31 | p3nguin_ | I understand WHY people say it, but it doesn't make it any more accurate. |
| 08:33.55 | norrec | but in the end, while it may not be an accurate term it is the one used in the industry and by accepting that we all know what it refers to we can use it to create a meaningful converstaion |
| 08:34.22 | norrec | but i think that occurs a lot p3nguin_ |
| 08:34.38 | p3nguin_ | And until we know what we SHOULD be calling the concept, the term will keep getting used improperly. |
| 08:34.42 | norrec | were ppl use a term that doesnt really make sense, mostly becuse they dont fully understand it |
| 08:34.50 | norrec | where* |
| 08:35.31 | p3nguin_ | ~sip-trunk |
| 08:35.37 | p3nguin_ | ~trunk |
| 08:35.38 | infobot | hmm... trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 08:36.17 | norrec | i think that even if some1 figured out a proper term, most people would continue to use |
| 08:36.22 | norrec | "SIP trunk" |
| 08:36.24 | themolester | p3nguin_ was there a logical and meaningful reason for some guy to call something a byte? or a nibble? |
| 08:36.39 | p3nguin_ | I doubt it. |
| 08:36.45 | themolester | some guy said it, more people repeated it |
| 08:36.50 | themolester | now it IS the correct term |
| 08:36.55 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 08:37.22 | p3nguin_ | But was there something else that correctly described that phenomenon before he coined the terms? |
| 08:38.10 | norrec | is there something else that correctly describes "sip trunks" |
| 08:38.22 | norrec | or i should say, what sip trunks refer to |
| 08:38.33 | p3nguin_ | There is, but I cannot recite it at this moment. |
| 08:38.49 | norrec | is it more than 1-3 words? |
| 08:38.58 | themolester | also, thank you for enlightening me, i had no idea a telcom trunk could be a single line |
| 08:39.02 | p3nguin_ | Ask tk and manxpower in about 8-10 hours. |
| 08:39.06 | themolester | telecom |
| 08:39.48 | themolester | i had always heard that term to refer to larger scale builds |
| 08:40.09 | norrec | i was gonna say that if its a mouthful thats why we have "sip trunk" but i have a feeling it has more to do with the techs trying to explane the concept to sales guys haha |
| 08:40.27 | norrec | so to make things easy they just said, its like trunks over something called sip |
| 08:40.34 | norrec | then the sales guys called them sip trunks |
| 08:40.38 | norrec | and thats what stuck |
| 08:41.25 | themolester | which may actually have a lot to do with the misinformation, since a lot of other people may have also associated the word of 'trunk' to mean multi-line |
| 08:41.33 | p3nguin_ | I think of trunk in the sense of dot1q and iax2. |
| 08:41.55 | norrec | but ur knowledgeable about this |
| 08:41.57 | norrec | most ppl arnt |
| 08:42.32 | themolester | so, what allows a iax2 connection to be a 'trunk' and not sip? |
| 08:42.43 | themolester | just curious |
| 08:42.44 | p3nguin_ | trunking=yes |
| 08:42.47 | p3nguin_ | :) |
| 08:42.50 | norrec | lol |
| 08:42.55 | norrec | sip cant do trunking |
| 08:42.59 | norrec | iax2 can |
| 08:43.09 | p3nguin_ | err, trunk=yes |
| 08:43.51 | themolester | i mean, what specifically... i haven't dug too much into iax since there are so few providers |
| 08:44.04 | themolester | and i haven't needed to link my own yet |
| 08:44.19 | p3nguin_ | In my mind, I picture a trunk like how a T1 operates. |
| 08:44.28 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:44.34 | norrec | hey p3nguin_, do you by chance know if they implemented multi-context call parking in 1.6.0.x? |
| 08:45.53 | p3nguin_ | You join all the information describing the data into an area by itself, letting the data flow without all the headers... |
| 08:46.06 | kaldemar | ~trunk |
| 08:46.07 | infobot | well, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 08:46.21 | p3nguin_ | The more data you push through the trunk, the better the benefits of trunking. |
| 08:47.57 | p3nguin_ | Rather than having headers on each frame, you reduce the overhead by taking out the unnecessary duplicated information (the parts that never change due to the protocol being used). |
| 08:48.22 | p3nguin_ | If you have to keep repeating that info over and over and over, you're taking up space that could be used for more useful data. |
| 08:48.33 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 08:48.37 | themolester | so... let me see if i'm following the logic flow.. |
| 08:48.58 | p3nguin_ | I'm just describing it the way it visualize it. |
| 08:49.06 | p3nguin_ | the way I visualize |
| 08:49.25 | norrec | wikipedia refers to a "sip trunk" as a sip connection, which makes more sence |
| 08:49.44 | norrec | sense* |
| 08:49.47 | *** join/#asterisk J_nx (~skully@217.166.55.142) |
| 08:50.07 | p3nguin_ | People call it a trunk because they have no idea what else to call it. |
| 08:50.28 | norrec | yep |
| 08:50.32 | p3nguin_ | And they are relating it to other technology, where a trunk means the connection. |
| 08:50.38 | norrec | yep |
| 08:50.52 | p3nguin_ | But it still doesn't make it an accurate description. |
| 08:50.57 | norrec | nope |
| 08:51.01 | themolester | if protocol foo ran by creating a single tcp or udp connection which stayed connected for the entirety of the call (or longer), and only transmitted detailed headers once... it would be a trunk? |
| 08:51.21 | p3nguin_ | no |
| 08:52.03 | norrec | a trunk, in reference to voice, is a single connection with the capacity for more than one call at a time |
| 08:52.12 | p3nguin_ | But if you sent two calls, and the description information was combined to reduce the space it needs, that would be a trunk. |
| 08:52.54 | norrec | http://en.wikipedia.org/wiki/Trunking |
| 08:53.08 | themolester | so, a trunk does imply multiple lines |
| 08:53.23 | p3nguin_ | I wish I knew how to correctly describe how it actually works. I can only describe it the way I "see" it. |
| 08:53.59 | norrec | In modern communications, trunking is a concept by which a communications system can provide network access to many clients by sharing a set of lines or frequencies instead of providing them individually. This is analogous to the structure of a tree with one trunk and many branches. |
| 08:54.36 | norrec | A trunk is a single transmission channel between two points, each point being either the switching center or the node. |
| 08:56.14 | norrec | if you think about a dsl connection, all the connections in your area connect to the local network hub |
| 08:56.29 | norrec | and that network hub has a trunk between it and the next routing point |
| 08:56.56 | norrec | and all the peers of the hub are pushed though one connection |
| 08:57.19 | norrec | rather than each peer having their own transmission line between the dsl hub and the next routing point |
| 08:58.37 | norrec | and sip cant be used as a trunk, or at least isnt, because each call requires its own sip initialization |
| 08:58.56 | norrec | which is why you need 10000-20000 for your rtp audio stream |
| 08:59.07 | norrec | each call has uses 2 ports |
| 08:59.11 | norrec | for example |
| 08:59.36 | norrec | 10001 is outbound audio and 10002 is in bound |
| 08:59.47 | norrec | the next call would take 10003-10004 |
| 08:59.52 | norrec | and so on |
| 09:00.02 | norrec | where iax2 uses a single port |
| 09:00.12 | norrec | and all calls are routed inside that port |
| 09:00.16 | norrec | including audio data |
| 09:00.24 | norrec | make sense? |
| 09:00.35 | themolester | mostly |
| 09:01.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:01.16 | norrec | p3nguin_: any problems with that, or did i do alright? lol |
| 09:01.50 | norrec | and i dont mean that in a dick way btw |
| 09:02.04 | norrec | just want to make sure i explaned it correctly |
| 09:05.44 | p3nguin_ | http://imagebin.org/86295 |
| 09:05.57 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:06.13 | p3nguin_ | Heh, can you tell it's 3AM? |
| 09:06.48 | p3nguin_ | I think you did pretty good. |
| 09:07.41 | norrec | oh good, i'm in the same boat as you |
| 09:07.48 | norrec | sadly its only 1am for me |
| 09:07.58 | norrec | but i'm working on ~2hrs of sleep a night |
| 09:08.42 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:09.08 | norrec | anyways, i'm gonna go to sleep before i stay any (more) stupid shit lol |
| 09:09.33 | p3nguin_ | I wanted to be in bed hours ago, but I have a client in GMT+2 that needed some things done, so I stayed up until their business hours. |
| 09:09.33 | norrec | hmm i guess that should be say (any more)* |
| 09:09.44 | norrec | oh fun |
| 09:10.29 | p3nguin_ | I shouldn't try to draw things. |
| 09:10.53 | norrec | lol, i think you did pretty good =D |
| 09:11.03 | *** join/#asterisk themolester (themoleste@cpe-173-171-165-200.tampabay.res.rr.com) |
| 09:11.15 | themolester | sorry about that, lost connection for a sec |
| 09:11.33 | p3nguin_ | That's just how I picture it. The trunk itself contains certain info, while the data has less overhead. |
| 09:11.54 | p3nguin_ | Oh, you probably missed my silly drawing. |
| 09:12.22 | p3nguin_ | http://imagebin.org/86295 |
| 09:13.09 | norrec | hmm |
| 09:13.16 | norrec | i gotta get a vhost methinks |
| 09:13.26 | norrec | just realised it showed my ip =/ |
| 09:13.45 | p3nguin_ | skeered of something? |
| 09:14.06 | norrec | na, i just dont like my ip being displayed for anyone who feels like whoising me |
| 09:14.14 | norrec | or for everyone when i join a channel |
| 09:14.26 | p3nguin_ | probes |
| 09:14.44 | themolester | just one more thing though... if the term trunking was included (incorrectly or not) prominantly inside a rfc... doesn't that pretty much make it a correct term, even if the idea behind it is wrong? |
| 09:14.45 | norrec | i generally dont piss ppl off, but i'd rather not have some bored script kiddie trying to break into my shit |
| 09:14.58 | themolester | http://www.ietf.org/rfc/rfc4904.txt |
| 09:15.38 | p3nguin_ | People say wrong things in technical documentation all the time, and it doesn't make it any more right. |
| 09:17.09 | themolester | you should detail your opinion on some site that pbx/telco newbs like me go to when they start learning asterisk |
| 09:17.16 | themolester | like http://www.voip-info.org/wiki/view/SIP+Trunking perhaps |
| 09:17.45 | p3nguin_ | Someone more qualified than myself probably did it already. |
| 09:18.08 | themolester | if you want the term to change, doing so once every few days in a room totalling 300 is probably nowhere near the exposure that page gets |
| 09:18.37 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 09:18.57 | p3nguin_ | I was just sayin' I don't like the term due to its inaccuracy. |
| 09:19.02 | themolester | :) |
| 09:19.06 | norrec | lol |
| 09:19.23 | themolester | oh yea... wasn't I troubleshooting a hour ago |
| 09:19.24 | themolester | oops |
| 09:19.28 | themolester | :P |
| 09:19.30 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 09:19.45 | norrec | lol |
| 09:20.05 | norrec | try what i told you earlier about the firewall |
| 09:20.24 | norrec | seems like a nat issue to me, happens all the time with my companies customers |
| 09:20.56 | norrec | company |
| 09:20.57 | themolester | i'm not sure if i can do a 1:1 with a single ip on pfsense... theres some cryptic note on the page saying something about needing virtual ips |
| 09:20.59 | norrec | company's |
| 09:21.24 | themolester | and i'm not about to potentially lock up a remote router on a theory that it SHOULD work |
| 09:21.26 | themolester | :) |
| 09:21.48 | norrec | oooooohhhh |
| 09:21.59 | norrec | fuck |
| 09:22.02 | p3nguin_ | It's easy to add addresses to interfaces on FreeBSD. |
| 09:22.14 | norrec | i had this exact problem with a customer who was using monowall |
| 09:22.21 | norrec | and i assume its prob the same problem |
| 09:22.22 | themolester | norrec oh? |
| 09:22.31 | themolester | pfsense is a fork of mono |
| 09:22.34 | norrec | now i just have to remember what the fix was |
| 09:22.40 | themolester | yes, please do |
| 09:22.51 | themolester | jumps on norrec's back |
| 09:22.58 | themolester | tell me tell me tell me |
| 09:22.59 | themolester | :) |
| 09:23.00 | norrec | yeah, but i dont doubt that whatever the pfsense is going monowall was doing as well |
| 09:23.07 | norrec | doing* |
| 09:23.12 | norrec | man i'm tired |
| 09:23.16 | norrec | uhh |
| 09:23.27 | norrec | hold on let me see if i have it in my notes somewhere |
| 09:23.32 | themolester | :) |
| 09:23.38 | norrec | this was like 2 years ago =( |
| 09:23.42 | themolester | :/ |
| 09:25.52 | norrec | let me see if i still have access to the firewall |
| 09:26.15 | themolester | gets norrec an espresso to stay alert |
| 09:26.32 | themolester | i've been pounding my head against the wall with this for two days |
| 09:26.59 | norrec | hmm, looks like they changed the pw |
| 09:27.30 | themolester | those bastards |
| 09:27.46 | norrec | hmm, well check your access rules, and udp timeouts and sip alg and such |
| 09:28.10 | norrec | i'll call the admin over there tomorrow and see if he'll let me access his firewall |
| 09:28.32 | norrec | but like i said, i remember having this exact same problem with their monowall |
| 09:28.39 | norrec | i just cant rememeber what setting we changed |
| 09:28.42 | themolester | didn't they take a moment to consider the fact that sometime in the future someone they had never met might one day need someone who no longer needed access to their system to log in again for a similar issue |
| 09:29.08 | themolester | i mean... really... those inconsiderate bastards |
| 09:29.09 | themolester | :) |
| 09:29.11 | norrec | lol |
| 09:29.12 | norrec | i know |
| 09:29.16 | norrec | what were they thinking |
| 09:29.31 | norrec | lol |
| 09:30.41 | themolester | was it "Clear DF bit instead of dropping" by chance? |
| 09:31.08 | norrec | uhhh i dont think so |
| 09:31.32 | norrec | what u could do, as allow all from the sip providers ip address in your access rules |
| 09:31.48 | norrec | that should allow it to by pass anything being done by the firewall |
| 09:32.24 | themolester | the rules section has optional ports, but nat section needs port/range listed |
| 09:32.28 | *** join/#asterisk incorrect (~fwest@78.33.42.140) |
| 09:32.58 | themolester | was it somthing to do with pf 'scrub' or 'normalization'? |
| 09:33.05 | norrec | could |
| 09:33.07 | norrec | disable thoes |
| 09:33.22 | norrec | you dont want the firewall tampering with the sip or rtp packets |
| 09:33.41 | norrec | but there is a difference between nat and access rules |
| 09:34.10 | norrec | you need to portforward to deal with nat |
| 09:34.17 | norrec | and add in access rules to deal with the firewall |
| 09:34.40 | themolester | right, thats the way its configured now, port forward+access rule |
| 09:35.35 | norrec | hmmm |
| 09:36.16 | norrec | idk, but i cant think straight anymore lol |
| 09:36.27 | norrec | i would go though all your settings for the firewall |
| 09:37.04 | norrec | also make sure youve disabled your linux firewall and selinux |
| 09:37.20 | norrec | on the asterisk server |
| 09:37.53 | norrec | and i think you had mentioned that some of your forwards wernt working |
| 09:38.04 | norrec | you'll prob want to start there and figure out why |
| 09:38.30 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 09:38.32 | Polysics | hello |
| 09:38.44 | norrec | hello |
| 09:38.46 | Polysics | what is the best way to have a webpage with user status on it? |
| 09:38.55 | Polysics | asking * via AMI? |
| 09:39.16 | Polysics | or is that too intensive, and would logging statuses via AMI in a DB then querying that be preferable? |
| 09:39.45 | norrec | i use http://www.asternic.org/ but idk if thats what your looking for |
| 09:39.54 | Polysics | so far i have a working event logger, so i could just hook that up to MySQL |
| 09:40.22 | norrec | are you looking for like reg status or call status? |
| 09:40.30 | Polysics | both, i'd say |
| 09:40.32 | norrec | like aval, rining, inuse? |
| 09:40.38 | Polysics | exactly |
| 09:40.50 | Polysics | problem is, i can't use FOP as this is a "custom" app |
| 09:41.13 | norrec | well asterisk supports watchers |
| 09:41.17 | Polysics | i have Adhearsion running so scripting isn't a problem, was just thinking about repeated AMI queries being a problem |
| 09:41.36 | Polysics | norrec, watchers = AMI events? or is it something else? |
| 09:42.20 | norrec | <PROTECTED> |
| 09:42.57 | norrec | yeah, its events |
| 09:43.06 | norrec | usu used for blf on phones |
| 09:43.10 | norrec | on sip phones |
| 09:43.35 | Polysics | hmm, hinting means every user needs his own extension |
| 09:43.45 | Polysics | let's hope i do not need that |
| 09:43.58 | themolester | don't you generally need that anyway? |
| 09:44.05 | norrec | yeah... thats what i was thinking |
| 09:44.20 | norrec | what do you intend to monitor if each person doesnt have their own exension? |
| 09:44.24 | norrec | extension* |
| 09:44.37 | themolester | don't confuse extensions (virtual, you can create on a whim) with incoming lines |
| 09:44.38 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:45.14 | Polysics | so far i have some fairly complex logic in place for calls |
| 09:45.20 | Polysics | let me explain the whole project: |
| 09:45.37 | Polysics | there are two types of users "translators" and "operators" |
| 09:45.48 | Polysics | translators provide translation services for one or more language to operators |
| 09:46.24 | Polysics | operators need to be able to see the translators' status and eventually call one through a provisioned Zoiper Web ActiveX softphone, which i control via javascript |
| 09:46.35 | Polysics | obviously you cannot call a busy operator |
| 09:46.39 | Polysics | *translator |
| 09:47.00 | norrec | well each one of thoes provisioned softphones needs its own extension... |
| 09:47.16 | Polysics | in addition, an operator needs to be able to enter a queue for a specific language, thing i haven't yet sorted out but will need soon |
| 09:47.28 | Polysics | i can just dial SIP/XYZ |
| 09:47.39 | norrec | but xyz is an extension |
| 09:47.50 | Polysics | nope, it's the client account |
| 09:48.08 | Polysics | you can dial SIP accounts without extensions, been doing that so far :-) |
| 09:48.43 | norrec | ok, well either way, we're talking about the same thing |
| 09:48.50 | norrec | a sip account is an extension |
| 09:48.57 | p3nguin_ | No it isn't. |
| 09:49.04 | norrec | really? |
| 09:49.45 | p3nguin_ | extensions are rules to say what happens when calling numbers (or letters). |
| 09:49.54 | norrec | oh |
| 09:50.24 | norrec | well, what do you know, my terms dont match up with the correct terms yet again ;) |
| 09:50.35 | p3nguin_ | exten => 123,1,Playback(tt-monkeys) |
| 09:50.45 | p3nguin_ | extension 123 does not call any phones. |
| 09:50.57 | norrec | makes sense |
| 09:51.10 | norrec | but his sip users should still give hints correct? |
| 09:51.45 | p3nguin_ | exten => 321,1,Dial(SIP/8675309,30) |
| 09:51.49 | p3nguin_ | SIP account is 8675309 |
| 09:51.55 | p3nguin_ | extension is 321 |
| 09:52.23 | themolester | but 8675309 is an extension too |
| 09:52.27 | p3nguin_ | nope |
| 09:52.31 | Polysics | one sec, phone |
| 09:52.35 | norrec | its the termonology again lol |
| 09:53.04 | p3nguin_ | 8675309 is the device's identifier, and it has absolutely nothing to do with extensions. |
| 09:53.04 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 09:53.31 | themolester | lmao... what the hell is that number from |
| 09:53.33 | p3nguin_ | phones aren't extensions, and extensions don't always take you to phones. |
| 09:53.36 | norrec | an extension is the dialing rules associated with a user correct? |
| 09:53.44 | p3nguin_ | Call Jenny? |
| 09:54.05 | themolester | i have no idea what the song is, but now its stuck in my head |
| 09:54.21 | p3nguin_ | http://en.wikipedia.org/wiki/867-5309/Jenny |
| 09:54.45 | themolester | 8-6-7-5-3-0-9eeeiii-iiinnne |
| 09:55.00 | themolester | i don't remember another damn word from the song though |
| 09:55.03 | themolester | lol |
| 09:55.53 | p3nguin_ | Extensions do whatevertheheck you tell them to do. They can Dial() devices via channel technology, they can Playback() sound files, they can do nothing such as NoOp(), they can disconnect with a Hangup()... |
| 09:56.11 | p3nguin_ | Extensions really aren't related to the users nor their phones at all. |
| 09:57.03 | norrec | ah |
| 09:57.03 | norrec | ok |
| 09:57.23 | norrec | well i called the sip users extensions |
| 09:57.28 | p3nguin_ | Now on the other hand, MOST people use the extension number the same as the account number. So it would be exten => 123,1,Dial(SIP/123). |
| 09:57.33 | norrec | and the extensions i just through into routing |
| 09:57.34 | norrec | haha |
| 09:57.47 | norrec | threw* |
| 09:58.56 | *** join/#asterisk ickmund (~magnus@ada-bcn-fw01.adamoeurope.com) |
| 09:59.34 | p3nguin_ | The correct term would be user, peer, or friend... not extension. |
| 10:00.05 | norrec | yes, i see that now lol |
| 10:00.12 | p3nguin_ | Do you have to add a person in Skype to be able to talk with them? |
| 10:00.33 | norrec | i believe so |
| 10:00.36 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 10:00.38 | Polysics | ok, i am back ,sorry |
| 10:00.45 | themolester | i think they have to approve you as well |
| 10:00.56 | Polysics | an extension can be anything, yes |
| 10:00.57 | p3nguin_ | Ah, I guess not. Just click on "Start a chat with..." rather than clicking OK on the ADD page. |
| 10:01.03 | norrec | ah |
| 10:01.04 | Polysics | i am talking about sip peers/friends |
| 10:01.18 | norrec | well like i said, you should get hints from those |
| 10:01.41 | norrec | and you can subscribe to the hints rather than querying AMI every x seconds |
| 10:01.55 | Polysics | no hints = no events i can use in AMI? |
| 10:01.56 | p3nguin_ | exten => 208,hint,SIP/208 |
| 10:02.03 | Polysics | i already have a subscriber daemon |
| 10:02.31 | Polysics | that currently logs everything it sees to a text file, obviously just for development |
| 10:03.00 | Polysics | p3nguin_, i need the extension and peer number to match for hints to work? i suppose not, but.. |
| 10:03.14 | p3nguin_ | Nah |
| 10:03.17 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 10:03.27 | p3nguin_ | You probably do have to have call-limit set, though. |
| 10:03.37 | norrec | oh yeah, u do need that |
| 10:03.43 | norrec | you wont get hints w.o it |
| 10:03.45 | p3nguin_ | Even if you set it to something extremely high. |
| 10:03.56 | norrec | i usu do 100 |
| 10:03.58 | p3nguin_ | call-limit=99 would be better than not having it at all. |
| 10:04.10 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 10:05.58 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 10:06.35 | *** join/#asterisk Kumbang (~dsp@167.205.24.69) |
| 10:06.42 | Polysics | call limit is? |
| 10:08.12 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 10:08.21 | p3nguin_ | I don't know why it is required for hints to work, but lots of people have said it is. |
| 10:09.51 | p3nguin_ | J\u016bsu bi\u013cete tika sa\u0146emta, un viens no m\u016bsu konsultantiem str\u0101d\u0101, lai atbild\u0113tu uz J\u016bsu jaut\u0101jumu. Zem\u0101k nor\u0101d\u012bti J\u016bsu bi\u013cetes dati. L\u016bdzu neizdz\u0113siet bi\u013cetes numuru rindi\u0146\u0101 \u201cT\u0113ma\u201d. |
| 10:09.58 | p3nguin_ | Chrap. |
| 10:13.14 | Polysics | uhm, but aren't hints a ver yspecific functionality? |
| 10:13.30 | Polysics | let's go back to the original question which is "a webpage with sip statuses on it" |
| 10:13.51 | p3nguin_ | Yes, hints are for showing presence. |
| 10:16.02 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 10:16.23 | *** join/#asterisk themolester (themoleste@cpe-173-171-165-200.tampabay.res.rr.com) |
| 10:17.13 | *** join/#asterisk godsyn (~godsyn@c-67-167-168-141.hsd1.tn.comcast.net) |
| 10:18.42 | Polysics | no hints means i only get to know if a SIP peer is registered or not? |
| 10:19.09 | p3nguin_ | If your app checks for that, I guess so. |
| 10:19.35 | p3nguin_ | Hints would allow the app to check the status of the peer's line: idle, unavail, busy |
| 10:19.46 | p3nguin_ | InUse |
| 10:19.56 | godsyn | Suggestions for googlefu, please: I'm attempting to record DTMF from only one side of a call (caller (initiator)), and mute/ignore all DTMF from/to callee. All I can think of is MeetMe auto-creation/setup when calling. Other ideas? |
| 10:20.05 | p3nguin_ | I don't remember all the statuses. |
| 10:20.28 | godsyn | (1.4) |
| 10:23.46 | Polysics | and how would I check that status? through AMI? |
| 10:24.11 | Polysics | which brings us again to the original point: do i log those in a DB then read them with PHP, or is it better to ask AMI directly? |
| 10:24.25 | Polysics | are those events or need to be read? |
| 10:31.26 | *** join/#asterisk rossh (~ross@host217-40-110-153.in-addr.btopenworld.com) |
| 10:32.19 | norrec | you could create another sip user that subscribes to all the other users |
| 10:32.27 | norrec | so you dont have to query |
| 10:32.36 | norrec | idk how you would do that one though =/ |
| 10:32.43 | norrec | otherwise its core show hints |
| 10:33.09 | benngard | any knows what u gonna set "T38FaxMaxDatagram" to, when u receive a waring like "UDPTL asked to send 75 bytes of IFP when far end only prepared to accept 63 bytes; data loss will occur."? |
| 10:33.24 | norrec | and i dont imagine there is really a diff between logging to a db or just getting them direct with php |
| 10:34.09 | Polysics | can the extension i use as a hint be also used? |
| 10:34.21 | norrec | cause there would be no reason to query the db more often than you query the AMI |
| 10:34.38 | Polysics | eg. exten => 100, hint, SIP/100, but also exten => 100, 1, Dial(SIP/100) |
| 10:34.39 | norrec | an extension that has hints can also subscribe |
| 10:35.02 | p3nguin_ | You can't have those spaces in there. |
| 10:35.31 | p3nguin_ | and yes, that's how you handle the hints. |
| 10:35.45 | p3nguin_ | both a hint and a dial |
| 10:35.53 | norrec | well there you go |
| 10:37.16 | norrec | Polysics: anyways, i'm off for tonight, but i think p3nguin_ is prob a better resouce than me anyways |
| 10:37.38 | norrec | esp when i can barely type lol |
| 10:39.01 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 10:41.01 | godsyn | I'm attempting to mute outbound DTMF (and capture it (only outbound, single channel) to a var). Suggestions? |
| 10:41.57 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 10:47.46 | *** join/#asterisk ickmund (~magnus@ada-bcn-fw01.adamoeurope.com) |
| 10:48.15 | Polysics | p3nguin_, is the dial needed? |
| 10:48.28 | Polysics | i suppose so, since no dial = no activity on the extension |
| 10:48.34 | p3nguin_ | If you want to dial the device, yes. |
| 10:48.35 | Polysics | even if i directly dial the SIP user |
| 10:48.54 | Polysics | as said before, i can directly dial a SIP peer with no extension |
| 10:49.08 | Polysics | but it makes sense if that does not trigger the hint |
| 10:49.51 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 10:49.53 | p3nguin_ | I don't fully understand how the extension relates to the device when you are hinting it. |
| 10:49.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 10:50.00 | themolester | norrec are you still awake? |
| 10:53.41 | Polysics | guess the only sensible test is to hint an extension, then fire up my event logger and see if there is anything there |
| 10:58.31 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 11:01.57 | norrec | Polysics: i dont use hint in my exten dialing but i still get hints |
| 11:02.12 | norrec | themolester: whats up? |
| 11:03.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 11:03.11 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 11:03.21 | themolester | norrec does this look like the issue you fixed |
| 11:03.22 | themolester | http://marc.info/?l=ipfilter&m=121749272404107&w=2 |
| 11:03.37 | themolester | i'm so tired i really don't know what i'm looking at |
| 11:04.49 | norrec | that would be a problem if the call wasnt starting |
| 11:05.10 | themolester | i think i'm going to tell the boss he NEEDS to order the 5 static ip package, and hook the asterisk direct to net with a switch+ iptables |
| 11:05.13 | norrec | what seems to be happening in ur case, is that the sip init goes fine |
| 11:05.22 | norrec | but then the firewall cuts the rtp stream |
| 11:05.56 | norrec | and then one side just sends a bye cause its not getting audio data anymore |
| 11:06.01 | norrec | so it assumes the other side hung up |
| 11:06.43 | norrec | Polysics: also my bad, i do use a hint in all my exten... wasnt thinking straight =/ |
| 11:06.59 | Polysics | :-) |
| 11:07.01 | themolester | if the firewall cut the rtp stream, wouldn't it take a while (more than 1 second) before asterisk hung up? |
| 11:07.27 | themolester | i'm getting audio until nearly (maybe entirely) the last instant |
| 11:07.39 | norrec | uhh, not nessicarily |
| 11:07.45 | norrec | cause once the rtp stream stops |
| 11:07.48 | norrec | the call stops |
| 11:08.03 | norrec | well i guess thats not true either |
| 11:08.13 | norrec | idk, i would need to see a packet capture |
| 11:08.21 | norrec | and some sleep |
| 11:08.44 | norrec | but your other option works as well |
| 11:08.45 | p3nguin_ | You should be able to completely cut off RTP and the call would remain. |
| 11:09.01 | norrec | how are u still up p3nguin_? |
| 11:09.02 | norrec | lol |
| 11:09.12 | p3nguin_ | barely |
| 11:09.19 | norrec | yeah me too |
| 11:09.22 | themolester | yea p3nguin_ you said you were the same timezone as me |
| 11:09.23 | norrec | i cba to think |
| 11:09.31 | themolester | i know why I've been up constant |
| 11:09.31 | p3nguin_ | 05:09 |
| 11:09.36 | themolester | why the hell are you up |
| 11:09.43 | norrec | i think my insomnia is contagious over then internet |
| 11:09.45 | p3nguin_ | working |
| 11:09.46 | themolester | oh, not quite as bad, thought it was 6 there |
| 11:09.56 | norrec | X( |
| 11:09.58 | norrec | anyways |
| 11:10.04 | norrec | i cant think anymore right now |
| 11:10.04 | themolester | its 6 here |
| 11:10.07 | themolester | heh |
| 11:10.13 | norrec | i'm not really sure what i'm even answering atm |
| 11:10.15 | norrec | lol |
| 11:10.24 | themolester | was up till 9 am last night.... err... thismorning |
| 11:10.28 | norrec | in the last 3 days ive had 6 hrs of sleep =( |
| 11:10.28 | themolester | something like that |
| 11:10.40 | norrec | er 4 |
| 11:10.51 | norrec | 4 days |
| 11:10.52 | norrec | 6 hrs |
| 11:10.55 | norrec | wooo |
| 11:11.04 | themolester | i am about a flinch away from telling the boss to contract the pbx out |
| 11:11.23 | norrec | you can do it though my company =p |
| 11:11.31 | themolester | its fun to play with when its at my house, and it really doesn't matter if it goes down for a week or more |
| 11:11.37 | norrec | how many lines do u guys need? |
| 11:11.41 | themolester | but two days without phones in a business is bad |
| 11:11.42 | norrec | just out of curiosity |
| 11:12.07 | themolester | unless you're in the greater tampa area, not really interested |
| 11:12.34 | themolester | i already have a provider in another state who expects me to have everything set up :) |
| 11:12.55 | norrec | *shrug* located in sac, but we have customers nation wide |
| 11:13.18 | themolester | right now, its nothing, the most traffic we've had in 2 months is me calling in with this 20 second bug |
| 11:13.22 | themolester | to test |
| 11:13.23 | themolester | heh |
| 11:13.26 | norrec | lol |
| 11:13.35 | themolester | but in the next few weeks we're starting an ad campaign |
| 11:13.46 | themolester | and the phone system needs to be rock solid, or thats lost money |
| 11:13.52 | norrec | indeed |
| 11:14.13 | norrec | well you can see about getting additional ips |
| 11:14.18 | themolester | yea |
| 11:14.26 | norrec | but i'll give a call over to the customer of mine |
| 11:14.32 | norrec | and see if i can see what we did |
| 11:14.49 | norrec | cause i think once i find it |
| 11:14.54 | norrec | it should be a quick fix |
| 11:15.32 | norrec | and i'm sure it would be easier in the long run to get the firewall working than install more infastructure to support |
| 11:15.54 | themolester | i would really appreciate it |
| 11:16.15 | themolester | honestly, i'm not even sure if it's getting blocked from my router, or the isp |
| 11:16.16 | norrec | *shrug* no problem, i know what its like to be in ur position |
| 11:16.37 | themolester | isps who also provide digital voice have a habit of making things difficult |
| 11:16.59 | norrec | i havent had that issue here |
| 11:17.19 | p3nguin_ | I also don't have that problem here. |
| 11:17.27 | norrec | infact almost all the internet providers i work with do digital voice |
| 11:18.12 | themolester | hmm |
| 11:18.21 | norrec | my hosting facility is a huge voice provider |
| 11:18.26 | norrec | lol |
| 11:18.51 | themolester | well i have no other way to explain why port 5060 outgoing stopped working mysteriously |
| 11:19.00 | norrec | yeah, thats an odd one |
| 11:19.16 | norrec | how big is ur internet providor? |
| 11:19.19 | norrec | provder* |
| 11:19.22 | p3nguin_ | I thought 5060 was working. |
| 11:19.28 | themolester | i was on the phone with the sip provider and they had me try 5060 in a different server location, completely separate |
| 11:19.43 | themolester | and it wouldn't work, then switched to the alt port and it lit up |
| 11:19.55 | themolester | p3nguin_ it worked for a little while, then crapped out |
| 11:19.59 | norrec | still sounds like a firewall issue to me =/ |
| 11:20.14 | norrec | i'd like to see that packet capture |
| 11:20.17 | norrec | tbh |
| 11:20.39 | norrec | but atm i muuusssttt sleep |
| 11:20.43 | norrec | gotta be up in 3 hrs |
| 11:20.54 | norrec | so night guys |
| 11:20.58 | themolester | i agree.. firewall... but whos |
| 11:21.00 | themolester | :) |
| 11:21.05 | norrec | haha |
| 11:21.15 | norrec | blaim ur sip provider =[ |
| 11:21.17 | norrec | =p* |
| 11:21.21 | norrec | tell them to fix it for u |
| 11:21.21 | norrec | haha |
| 11:21.30 | themolester | mine has never had an issue til i started messing with sip, provider says its not them, and sip provider says its not them |
| 11:21.31 | themolester | lol |
| 11:21.56 | p3nguin_ | Whose firewall? Probably yours, since I doubt there are any others present. |
| 11:22.03 | norrec | lol |
| 11:22.07 | norrec | anyways guys |
| 11:22.08 | norrec | night |
| 11:22.17 | norrec | i'll talk to u tomorrow themolester |
| 11:22.19 | p3nguin_ | quickly slides in and removes his firewall from themolester's network |
| 11:22.28 | norrec | does the same |
| 11:22.33 | themolester | p3nguin_ there are always firewalls present at the edges of isps |
| 11:22.46 | themolester | just not nat, and usually are fairly wide open |
| 11:23.07 | themolester | they use em for packet floods and to block hacking attempts |
| 11:23.16 | norrec | yeah, to their network |
| 11:23.22 | norrec | but not between hosts |
| 11:23.38 | themolester | my isp admits they have them, but I'd have to contact the security department to see if anything got blocked |
| 11:23.50 | norrec | hmm |
| 11:24.02 | norrec | yeah, they shouldnt be filtering any of your traffic |
| 11:24.11 | norrec | if they are... well thats kind of a problem |
| 11:24.20 | norrec | i would suggest moving to a larger isp |
| 11:24.40 | themolester | its brighthouse... spin off from timewarner |
| 11:24.58 | norrec | yeah thoes isps troubble me lol |
| 11:25.06 | p3nguin_ | It's all RoadRunner network, though. |
| 11:25.07 | norrec | tier 3 isps do odd shit sometimes |
| 11:25.22 | norrec | i would just see if you can buy direct from roadrunner |
| 11:25.41 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 11:25.55 | norrec | ok sleep |
| 11:26.24 | themolester | yea, me too |
| 11:26.29 | themolester | almost 7am |
| 11:26.32 | p3nguin_ | still waits |
| 11:26.33 | norrec_zzz | lol |
| 11:26.41 | norrec_zzz | what are u waiting for p3nguin_ |
| 11:26.43 | norrec_zzz | ? |
| 11:26.58 | p3nguin_ | Trying to recover several thousand emails. |
| 11:27.04 | norrec_zzz | ooohhh |
| 11:27.08 | norrec_zzz | sounds fantastically fun |
| 11:27.09 | norrec_zzz | =( |
| 11:27.10 | themolester | norrec_zzz if i'm not awake or around when you find the info ... just send it via query to me |
| 11:27.19 | norrec_zzz | i planned to |
| 11:27.21 | themolester | i'll get it when i finally get back |
| 11:27.24 | norrec_zzz | lol kk |
| 11:27.32 | p3nguin_ | I don't know too much about cyrus imap, so it's a little bothersome. |
| 11:27.43 | themolester | gonna be a long day tomorrow |
| 11:28.04 | themolester | thank you both though, even though we never solved it, you've been a ton of help working through theories |
| 11:28.04 | norrec_zzz | p3nguin_: all i know is that its supposed to be bi-directional |
| 11:28.15 | themolester | err... haven't solved it yet |
| 11:28.25 | norrec_zzz | indeed |
| 11:28.32 | norrec_zzz | hopefully we can get this sorted out |
| 11:28.37 | p3nguin_ | For some reason, the old emails just disappeared. |
| 11:28.43 | themolester | i don't want to put a negative out there :) |
| 11:28.53 | p3nguin_ | Cyrus says it did NOT expunge any mails. |
| 11:29.12 | themolester | p3nguin_ is it possibly a client issue? |
| 11:29.38 | p3nguin_ | That was my thought... such as Outlook pop3 downloading the mails and not leaving a copy on the server. |
| 11:29.50 | p3nguin_ | But they said that didn't happen. |
| 11:30.04 | norrec_zzz | isnt it supposed to be a sync protocol |
| 11:30.33 | norrec_zzz | it doesnt delete anything till you delete it at one of the locations youve synced with |
| 11:30.37 | p3nguin_ | So now I'm having someone copying files out of a backup image into the live host. |
| 11:31.01 | norrec_zzz | well good luck with that p3nguin_ |
| 11:31.05 | themolester | yea, imap shouldn't have a 'delete after download' type setting, but there might be a 'delete old emails after x days' |
| 11:31.17 | p3nguin_ | We've gone from 12 files to 1659 files, so the copying must be going well. |
| 11:31.22 | norrec_zzz | lol |
| 11:31.24 | norrec_zzz | good to hear |
| 11:31.24 | norrec_zzz | =D |
| 11:31.38 | themolester | p3nguin_ i'd look for a setting like that somewhere in the clients |
| 11:31.48 | norrec_zzz | well i hope you can find yourself some time to sleep today |
| 11:31.59 | themolester | delete/archive old items |
| 11:32.08 | norrec_zzz | anyways cya for real |
| 11:32.44 | p3nguin_ | I'll try to get them to reveal the true story, but for now I must get the mails back on the box. |
| 11:33.25 | themolester | also, never expect the service provider to do their job... make your own backups this time |
| 11:33.46 | themolester | they had backups for you on this incident, but next time who knows |
| 11:34.08 | p3nguin_ | I think they are taking an image every day. |
| 11:34.45 | themolester | still, you're trusting THEM that THEY do a backup, and THEY don't lose THEIR backup and that THEY will pull the backup out at your first whim |
| 11:34.46 | p3nguin_ | They said they had 24.02.2010 (today)... I asked what about 22.02.2010? They said yes. |
| 11:35.37 | themolester | if they claim they're doing that frequent then you probably won't need nearly that often, but its good to have a copy in your hands to put your mind at ease... |
| 11:35.50 | themolester | but then again, i'm paranoid |
| 11:36.08 | p3nguin_ | This is the first time I have ever talked with the provider, so I don't really know what they do or don't do. |
| 11:36.39 | themolester | just be glad they were doing the job they were contracted to do |
| 11:36.41 | themolester | :) |
| 11:37.07 | themolester | its always expected, but not always carried through |
| 11:37.41 | themolester | anyway, i ahve to get some sleep and go to like 4 jobs tomorrow |
| 11:38.19 | themolester | freelance/self employed work... such a pain in the ass sometimes |
| 11:38.24 | themolester | bbl |
| 11:38.45 | p3nguin_ | 4818 miles between myself and that server, so let's hope they do their job correctly. |
| 11:39.34 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 11:41.24 | *** join/#asterisk ttwhy (~tekkno@p4FECF2AC.dip.t-dialin.net) |
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| 11:55.31 | mnick86 | Hi, I am confused about the SendFax application, do I need a "DIAL" before the SendFax ? |
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| 12:08.27 | *** join/#asterisk carumas (~carumas@a81-84-246-201.static.cpe.netcabo.pt) |
| 12:08.47 | carumas | Hello people! |
| 12:10.54 | carumas | Guys, I have asterisk 1.4 only with gsm codec and i'm using wav and gsm audio files. Is it supposed to work this way? Is asterisk coverting wav files to gsm when the playback is running? |
| 12:11.24 | carumas | the thing is that I want to save bandwith. will i attain that using wav files? |
| 12:11.35 | carumas | @ sip.conf I have only gsm allowed |
| 12:14.46 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:15.26 | kaldemar | carumas: what files you have, has no effect in bandwidth. the chosen codec defines that. you better use gsm sound files to avoid transcoding to gsm. |
| 12:17.13 | carumas | Kaldemar: ok, so what happens is that wen i use asterisk with a wav file and have it set in sip.conf to only allow gsm, it transcodes that wav file to gsm? |
| 12:22.29 | carumas | oh by the way, what codec you would advice to use with music files? Gsm is really crappy for that, but it really saves bandwidth. What other low bandwidth codec could i use to improve sound quality? |
| 12:22.51 | garymc | Yo anyone know how you configure the cfg files for Polycoms so the ringer volume stay high all the time? |
| 12:25.25 | kaldemar | carumas: yes, it transcodes it to gsm for the channel. if you use gsm codec, use gsm sound files. that's pretty much your choice unless you want to pay for G.729 licenses. |
| 12:26.35 | carumas | Thank you for your help |
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| 12:50.56 | garymc | Hi guys, im currently trying to boot my Polycom Ip550 phones from the tftp server. I got the files off Polycom site. Now both my phones are sitting there saying Checking applictaion. Theyve been doing this for about 5- 10 mins now. Are they downloading the new Firmware?? cos thats a pretty big file. Or could my files be wrong |
| 12:52.01 | kaldemar | they don't download or update the firmware for that long. |
| 12:55.35 | *** part/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 12:56.00 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 13:04.05 | garymc | no |
| 13:04.10 | garymc | what could be wrong |
| 13:04.30 | garymc | could i have downloaded the wrong ld files? |
| 13:06.14 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:10.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:13.08 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:13.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:15.26 | garymc | [TK]D-Fender : Got my tftp boot server working. Now both my IP phones are sitting there saying "Checking Application" They have been sitting at that for atleast 30 mins..... is this normal on a firmware upgrade? |
| 13:15.54 | [TK]D-Fender | garymc: No |
| 13:16.14 | garymc | Hmm what could be causing this. Could I be trying to put the wrong firmware on? |
| 13:16.28 | [TK]D-Fender | garymc: possibly |
| 13:16.40 | garymc | what should I check? |
| 13:16.57 | [TK]D-Fender | garymc: ... if you're putting the wrong firmware on. |
| 13:17.15 | garymc | Well i think im putting the right opne on. Just not 100% on it |
| 13:18.09 | garymc | i download the IP550 3.2.2 combined from polycom site. I know Version 2.2.2 was on the phone |
| 13:18.15 | [TK]D-Fender | garymc: When you're able to read SIP Application release Notes and version numbers you might stand a chance |
| 13:18.23 | garymc | well i did |
| 13:19.48 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:20.10 | [TK]D-Fender | garymc: Use your PC TFTP client and verify that you can grab the same files your phone would. |
| 13:20.13 | garymc | Hmm have I got to upload the bootrom first? |
| 13:20.29 | [TK]D-Fender | garymc: No, and you should not change that unless you absolutely have to |
| 13:20.53 | garymc | it says to use the latest version i need bootrom 4.0 or higher |
| 13:22.03 | ariel_ | Morning |
| 13:23.17 | *** join/#asterisk TommyBotten (tommy@91.205.185.145) |
| 13:23.23 | ariel_ | I have an issue with asterisk 1.6.015 and 1.6.0.21, I am not able to use iax2, the first server sends the call to the 2nd server no error is reported on the first server, but nothing at all happens on the 2nd no error, no debug mess no rejection. |
| 13:24.50 | [TK]D-Fender | garymc: What do you have on them currently? |
| 13:25.30 | garymc | I dont know what bootrom version but i think its less than 4.0 im just checking now |
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| 13:29.06 | *** join/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 13:29.52 | garymc | [TK]D-Fender : my tftp client downloads the files off the server |
| 13:30.21 | [TK]D-Fender | garymc: What have you got for config files so far? |
| 13:31.17 | garymc | Ive got sip.cfg macaddressofphones.cfg and phone204.cfg and phone205.cfg |
| 13:31.40 | garymc | plus sip550.ld |
| 13:31.52 | garymc | which ive just removed for now so the phones will boot |
| 13:31.55 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:32.10 | garymc | well atleast one of them will |
| 13:32.59 | *** join/#asterisk tdjacobs (~tiagoj@host00100.ht2p.net) |
| 13:33.04 | *** join/#asterisk Chodorenko (~chodorenk@84.201.225.23) |
| 13:33.22 | garymc | so i can find out the bootrom version |
| 13:34.03 | garymc | I think one of my IP550's is actually broken |
| 13:34.09 | Chodorenko | hello All |
| 13:34.14 | garymc | That just keeps rebooting |
| 13:34.59 | tdjacobs | hi, does speex support 16khz audio? |
| 13:35.07 | tdjacobs | or i need to resample to 8khz? |
| 13:35.22 | Chodorenko | howe to i can say asterisk send CANCEL to all agents in queue if one of agent answer on incoming call ? |
| 13:35.58 | *** part/#asterisk Boardy (~chatzilla@kirakira.xs4all.nl) |
| 13:37.04 | Chodorenko | in phone all operators have message XXX non answered cals |
| 13:37.59 | garymc | [TK]D-Fender : ok Bootblock = 2.8.1 ------Bootrom = 4.0.0 |
| 13:38.30 | *** join/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 13:38.31 | tdjacobs | I have a speex transcoder, i receive speex (16khz), asterisk only accepts speex 8khz... So, i will decode to PCM 16khz and send to asterisk... and read 16khz back from asterisk.. its possible? |
| 13:38.38 | tdjacobs | or asterisk is limited to 8khz audio? |
| 13:38.44 | [TK]D-Fender | Chodorenko: You can't |
| 13:38.49 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 13:39.03 | garymc | i need to be using split not combined |
| 13:39.05 | garymc | :S |
| 13:39.33 | Chodorenko | if i use command Dial(sip/XXXX&sip/YYYY) asterisk send command CANCEL to XXXX if YYYY Answer call |
| 13:40.22 | Chodorenko | [TK]D-Fender: in Road map planning add this Features ? |
| 13:40.29 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 13:41.45 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99) |
| 13:41.51 | [TK]D-Fender | Chodorenko: There is no common signal to tell a phone not to count a call that it was send as "missed" |
| 13:42.04 | [TK]D-Fender | Chodorenko: there isn't a roadmap because there is no standard <- |
| 13:42.04 | *** join/#asterisk voipmonk (~voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 13:42.14 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-oxmjyqkbdqctbiyf) |
| 13:42.21 | ManxPower-work | Many phones can be configured to not log missed calls. |
| 13:43.09 | Chodorenko | [TK]D-Fender: ??? in Dial "c" key create this functional |
| 13:43.54 | ManxPower-work | Chodorenko: you are welcome to code that feature and submit it to issues.digium.com |
| 13:44.39 | incorrect | is there a webui to make configuring asterisk easier? |
| 13:44.55 | incorrect | sure there are many, but which one should i sue |
| 13:44.56 | incorrect | use |
| 13:45.20 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:45.24 | ManxPower-work | incorrect: Yes, there are several 3rd party addon GUIs for Asterisk. None of them, in my opinion, make configuring Asterisk any easier. Remember, we don't support GUIs on this channel. |
| 13:45.25 | Chodorenko | http://www.asterisk.org/docs/asterisk/trunk/applications/dial c |
| 13:45.26 | Chodorenko | If the Dial() application cancels this call, always set the flag to tell the channel driver that the call is answered elsewhere. |
| 13:45.50 | ManxPower-work | Chodorenko: Are you running TRUNK? |
| 13:45.52 | incorrect | ManxPower-work, fair enough |
| 13:46.14 | incorrect | is there anything i can do to make my life easier? |
| 13:46.21 | [TK]D-Fender | Chodorenko: And if you do this direct on a phone with 5 consecutive DIAL's.... does it work as expected? |
| 13:46.45 | *** part/#asterisk Chodorenko (~chodorenk@84.201.225.23) |
| 13:46.49 | *** join/#asterisk Chodorenko (~chodorenk@84.201.225.23) |
| 13:46.58 | [TK]D-Fender | Chodorenko: And you certainly can't do this in a queue because the call CANNOT be "answered". It wasn't. |
| 13:47.07 | ManxPower-work | incorrect: Asterisk is not a PBX. Asterisk is a TOOLKIT that helps you build an PBX. Much like a compiler and library help you create an application. |
| 13:47.17 | incorrect | i see |
| 13:47.29 | ManxPower-work | You should understand Asterisk, Linux, NAT, routing, IP ports, telecom, and SIP |
| 13:47.59 | [TK]D-Fender | [08:47]<ManxPower-work>You should understand Asterisk, Linux, NAT, routing, IP ports, telecom, and SIP <- at least the first 2. the rest... as needed |
| 13:49.22 | incorrect | you must get a lot of people who just installed a distribution for the first time |
| 13:49.41 | [TK]D-Fender | incorrect: It happens |
| 13:50.07 | ManxPower-work | incorrect: not really. Most people realize they will fail at Asterisk if they don't know Linux. |
| 13:50.15 | Gugge | as far as i know app_queue supports the cancel_elsewhere stuff on 1.6.2 |
| 13:50.21 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 13:50.26 | Gugge | and 1.6.0+1.6.1 supports it in app_dial |
| 13:50.28 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 13:50.56 | ManxPower-work | You don't use Dial for queues |
| 13:51.12 | [TK]D-Fender | ManxPower-work: You do sometimes. |
| 13:51.13 | Gugge | no, thats why i said app_queue supported it in 1.6.2 :) |
| 13:51.54 | Chodorenko | [TK]D-Fender: Why ? if one of agents answer , other phone cancel ringing, but they were not sent a message that the call is accepted elsewhereб Why dont send message i don`t understand , may be its better of now situation |
| 13:52.14 | Gugge | Chodorenko: app_queue supports that in 1.6.2 |
| 13:52.20 | ManxPower-work | Chodorenko: if it does not work as documented, then report it as a bug. |
| 13:52.47 | [TK]D-Fender | Chodorenko: Only in a ring-all scenario |
| 13:53.30 | Chodorenko | ManxPower-work: in queue not documented and not make |
| 13:53.48 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 13:53.50 | Chodorenko | [TK]D-Fender: in 1.6.2 ? |
| 13:55.15 | Chodorenko | Gugge: by default ? without of different setting ? only RingAll policy setting . i true understand you |
| 13:55.44 | ManxPower-work | Fortunately, I found a work around for the missed calls issue. |
| 13:56.10 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 13:56.25 | Gugge | Chodorenko: the C flag should 'Mark all calls as "answered elsewhere" when cancelled' |
| 13:56.36 | Gugge | but i admit, i havent tried it |
| 13:57.23 | ManxPower-work | Gugge: get your caps correct. The C option resets the CDR for the call. |
| 13:57.28 | *** join/#asterisk faint2000 (~faint2000@78.110.96.32) |
| 13:58.05 | Gugge | ehm ... sure ? |
| 13:58.12 | faint2000 | Hi , how i use queue , how to calculate agent ring time for each agnet? |
| 13:58.18 | Gugge | C: Mark all calls as "answered elsewhere" when cancelled. |
| 13:58.18 | faint2000 | agent |
| 13:58.25 | Gugge | c: Continue in the dialplan if the callee hangs up. |
| 13:58.31 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.168.47.dsl.dyn.forthnet.gr) |
| 13:58.37 | Gugge | http://www.asterisk.org/docs/asterisk/trunk/applications/queue |
| 13:58.39 | ManxPower-work | Gugge: then you are using a different version of Asterisk then I am. |
| 13:59.02 | ManxPower-work | Ah, you are referring to the queue C option, not the dial c option. |
| 13:59.09 | Gugge | yes :) |
| 13:59.18 | ManxPower-work | WHY do people inisit in quoting docs for UNRELEASED Asterisk versions? |
| 13:59.27 | ManxPower-work | TRUNK == development version not released version |
| 13:59.29 | Gugge | couldnt find a link to 1.6.2 docs :) |
| 13:59.36 | Ad-Hoc | hi |
| 13:59.48 | ManxPower-work | Gugge: The docs are on your own system. |
| 14:00.10 | Gugge | yes, but i cant link my system here :P |
| 14:00.21 | ManxPower-work | Gugge: dpm |
| 14:00.39 | ManxPower-work | Gugge: I'm sure you'll eventually realize how much of a problem linking to the wrong documentation is. |
| 14:01.28 | Gugge | i do actually |
| 14:02.23 | Gugge | and im sorry for the wrong link |
| 14:02.45 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:03.03 | ManxPower-work | Ah, I see Verizon is out screwing up our lines again. |
| 14:03.37 | Gugge | But, aside from the wrong link, the C and c option does what i said, on 1.6.2.4 |
| 14:03.55 | Gugge | (if "core show application queue" is right) |
| 14:04.23 | garymc | [TK]D-Fender : I just downloaded the Split version of my upgrade as im running boot version 4.0 and there are 10 versions of sip.ld I read the readme and it says put the sip.ld file in correct folder on server. But there are many of them. All written like this "2345-11402-001.sip.ld" and so on |
| 14:04.45 | ManxPower-work | garymc: STOP! |
| 14:04.48 | drfreeze | I'm getting a channel.c:3033 __ast_request_and_dial: Unable to request channel DAHDI error due to congestion |
| 14:05.04 | garymc | ManxPower-work : ??? |
| 14:05.05 | ManxPower-work | garymc: read the admin guide or the polycom whitepaper on provisioning |
| 14:05.13 | drfreeze | on my PRI. This happened yesterday and went away on it's own |
| 14:06.16 | faint2000 | how to calculate the ring time at queue for agnets? |
| 14:06.25 | ManxPower-work | Last month my truck made a weird noise. Does anyone know the solution? |
| 14:06.47 | garymc | yes : scrap it |
| 14:07.02 | ManxPower-work | Maybe we need a #asterisk-queue? With a /topic of It sucks to be you |
| 14:07.04 | coppice | ManxPower-work: yeah. see your Toyota dealer |
| 14:07.16 | ManxPower-work | coppice: they work on Ford trucks? |
| 14:07.39 | ManxPower-work | drfreeze: might see the sillyness of his report. |
| 14:07.48 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 14:09.03 | coppice | ManPower-work they can offer you 60 months interest free finance right now to help you get rid of that noisy Ford. Remember Ford had some scary stuff with their accelerators recently |
| 14:09.29 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:09.35 | ManxPower-work | coppice: They might offer YOU 60 months of interest free financing. They would only offer me a good laugh. |
| 14:09.49 | Chodorenko | Gugge: where can read minimal version of asterisk support C in app_queue ? |
| 14:10.04 | drfreeze | ManxPower-work: there is no one on the line. Calls can come in, but not go out |
| 14:10.06 | Gugge | Chodorenko: huh? |
| 14:10.15 | ManxPower-work | Gugge: thank you for offering to work with Chodorenko |
| 14:10.15 | drfreeze | Please show me my sillyness. :) |
| 14:10.24 | Gugge | ManxPower-work: im bored :) |
| 14:10.41 | ManxPower-work | drfreeze: thinking we might be able to diagnose something that happens once and happened yesterday |
| 14:10.55 | drfreeze | ahh |
| 14:11.01 | Katty | WHO AUTHORIZED THIS SNOW |
| 14:11.21 | drfreeze | I assume PRI and asterisk is stable |
| 14:11.22 | Gugge | Katty: if you find him send him this way, i want to kill him! |
| 14:11.28 | drfreeze | I'm wondering if this is a telco problem |
| 14:11.29 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:11.40 | Katty | Gugge: k, you can have em when i'm through with em |
| 14:11.59 | ManxPower-work | drfreeze: It is. I suspect the telco bounced the line or something |
| 14:12.40 | tzafrir | drfreeze, "stable" as in 1.6.2.4? |
| 14:13.08 | tzafrir | btw: apache can't even do math properly: http://lwn.net/Articles/375806/ |
| 14:13.21 | Chodorenko | Gugge: Thx by help me |
| 14:13.23 | Katty | ahhh the fuzzies are sleeping |
| 14:14.03 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 14:15.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:15.25 | Katty | hugs jaytee and leifmadsen |
| 14:15.32 | leifmadsen | w00t |
| 14:16.08 | Katty | leifmadsen: http://ustre.am/bEBU |
| 14:16.14 | Katty | leifmadsen: teh bebes are napping :> |
| 14:17.52 | *** join/#asterisk nickaugust (~anonymous@71-33-207-229.hlrn.qwest.net) |
| 14:19.19 | Katty | oooh |
| 14:19.22 | Katty | BB's awake! |
| 14:19.51 | garymc | Ok I need to upgrade my bootrom. Do i just add bootrom.ld to the tftpboot folder and the phones will upgrade? |
| 14:20.06 | Katty | polycom has tons of documentation for that |
| 14:20.10 | *** join/#asterisk Czindy (~Czindy@91.120.30.42) |
| 14:20.20 | ManxPower-work | Katty: he doesn't want to read the docs. |
| 14:20.32 | Katty | ManxPower-work: i can see that. |
| 14:20.42 | ManxPower-work | garymc: if you do not read the docs you WILL FAIL. |
| 14:20.57 | [TK]D-Fender | garymc: No, you DON'T |
| 14:21.03 | garymc | im readin the docs. It doesnt say exactly how to upgrade tohe bootrom |
| 14:21.22 | ManxPower-work | garymc: then you are reading docs that are at least 4 years old. |
| 14:21.32 | garymc | they are on the polycom website |
| 14:21.39 | ManxPower-work | garymc: where EXACTLY does it say that? |
| 14:21.45 | garymc | http://downloads.polycom.com/voice/voip/relnotes/BootROM_4_2_0_release_notes.pdf |
| 14:21.58 | garymc | it doesnt say nothing about installing it |
| 14:22.07 | garymc | it just tells me what bootrom file i need |
| 14:22.16 | garymc | to upgrade .4.2 |
| 14:22.17 | ManxPower-work | garymc: Maybe you should read the docs I referred you do. If that was a good doc I would have suggested you read it. |
| 14:22.23 | *** join/#asterisk aandrade (~aandrade@201.22.27.252.dynamic.adsl.gvt.net.br) |
| 14:22.37 | Katty | Qwell: YOU |
| 14:22.43 | Katty | Qwell: IT IS WEDNESDAY |
| 14:22.46 | Katty | Qwell: :> |
| 14:22.48 | garymc | Err i dont think you refered me to any docs? |
| 14:23.26 | ManxPower-work | Oh look! It took me 5 seconds to find http://www.polycom.com/global/documents/support/technical/products/voice/white_paper_configuration_file_management_on_soundpoint_ip_phones.pdf |
| 14:24.13 | ManxPower-work | (9:05:05 AM) ManxPower-work: garymc: read the admin guide or the polycom whitepaper on provisioning |
| 14:24.19 | ManxPower-work | This doesn't refer you to the docs? |
| 14:24.39 | garymc | ok thanks |
| 14:24.40 | Katty | bmoraca_work: sooooooooooooooo |
| 14:24.42 | garymc | :S |
| 14:24.44 | ManxPower-work | I assumed you were smart enough to find them on the polycom web site. |
| 14:24.47 | Katty | bmoraca_work: you brought me peach shrimp recipe??? |
| 14:24.50 | Katty | bmoraca_work: RIGHT?! |
| 14:24.52 | Katty | bmoraca_work: :>>>> |
| 14:25.14 | Czindy | Hello! Please help me on realtime iax2 and queue problem: when I'm watching the queue (queue show Q6) the state of the members (between: available / not in use) are always changeing, but the iax peers not logging out, only waiting calls. Why is that happening? |
| 14:28.20 | Katty | holy crap |
| 14:28.42 | Katty | some guy was tearing out his bathroom to redo it and found a wells fargo envelope packed FULL of 100 dollar bills |
| 14:28.58 | Katty | http://i.imgur.com/Z8dY7.jpg |
| 14:29.07 | garymc | ManxPower-work : that white paper was useless |
| 14:29.27 | [TK]D-Fender | Czindy: Maybe you should SHOW US |
| 14:29.28 | ariel_ | what free money, where? |
| 14:29.32 | garymc | I know all that info that was there. It still doesnt tell me how to upgrade bootrom version |
| 14:29.43 | Katty | ariel_: i know, right? it makes me wnat to tear out my bathroom too! |
| 14:29.44 | ManxPower-work | garymc: I wish you the BEST of luck. |
| 14:30.15 | garymc | thanks |
| 14:31.17 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99) |
| 14:32.33 | Czindy | Here is a sample http://pastebin.com/gbJBnm7x you can see this on member 357 |
| 14:34.29 | Czindy | And here is the iax2 show peer 357 inculded: http://pastebin.com/uJHQcKxE |
| 14:34.35 | tuxx- | When i set a global variable, how long will this be kept in memory? Like im afraid the memory will be full after the systems runs for a couple of weeks, is there any way to clear a global variable? I saw a thread on the interwebz that said: 'setting the global variable to null would help' but its not really clear to me yet. Anyone know this? :) |
| 14:34.44 | Katty | just a remidner folks: today is the day that microsoft is releasing the optional patch KB971033 windows 7 piracy patch. |
| 14:35.21 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-egspkiasegdqpmps) |
| 14:35.23 | tuxx- | who uses windows anyway? |
| 14:35.40 | coppice | they keep the cold out |
| 14:36.01 | [TK]D-Fender | Czindy:I don't see 257 as registered |
| 14:36.51 | [TK]D-Fender | 357* |
| 14:38.45 | Czindy | how is it possible that status OK 14ms without registration? |
| 14:40.07 | [TK]D-Fender | Czindy: Go look at your peer |
| 14:41.30 | Czindy | The peers in this case are registering by a home-deployed software. Do you think that this program is not register corrctly? |
| 14:41.57 | [TK]D-Fender | Czindy: I'd go look at it if i were you |
| 14:43.06 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:45.02 | Czindy | is it correct that the command iax2 show registry shows: 0 IAX2 registrations |
| 14:45.06 | *** join/#asterisk V4mpire (~Gary@82.118.111.253) |
| 14:45.32 | V4mpire | hi does anyone here know an easy way to setup a network card in windows as a DHCP server for a phone ? |
| 14:46.14 | garymc | Cool phones now working. Both Bootroms upgraded and configs done. But I could do with understanding these config files better |
| 14:46.22 | ManxPower-work | Czindy: iax2 show registry shows remote servers ASTERISK is registered to. |
| 14:46.40 | Czindy | ok thank you |
| 14:47.41 | [TK]D-Fender | V4mpire: perhaps you should ask in #windows |
| 14:47.46 | garymc | anyone know of a decent XML editor for windows. Preferbly a free one. As im editing these files in notepad and they are really jumbled up |
| 14:51.16 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 14:51.25 | Czindy | thank you [TK]D-Fender we are trying to correct this. |
| 14:51.47 | garymc | [TK]D-Fender : This ip550 ive been having issues with. Upgraded the bootrom and booted to the extension i set 205. But it says network link down??? Is the phone broken The cable is fine and the connection and freepbx shows that extension is connected |
| 14:52.28 | [TK]D-Fender | garymc: Do I care what a GUI thinks? |
| 14:52.44 | garymc | No you dont :P |
| 14:52.47 | [TK]D-Fender | garyif it thinks the network link is down, then that is physical media |
| 14:53.07 | garymc | yes but how can the link be down if it just downloaded files from the boot server? |
| 14:53.11 | [TK]D-Fender | garymc: So either your phone, cable, or switch has issues |
| 14:54.01 | coppice | [TK]D-Fender: I care *if* a GUI thinks |
| 14:54.01 | [TK]D-Fender | garymc: maybe the connector itself is busted |
| 14:54.01 | garymc | connector on the phone? |
| 14:54.01 | [TK]D-Fender | coppice: a local ISP here -> http://skynetcanada.com/ |
| 14:54.01 | ManxPower-work | Maybe he upgraded the sip firmware without upgrading sip.cfg and phone1.cfg? |
| 14:54.01 | [TK]D-Fender | coppice: I don't think I trust these guys... |
| 14:54.30 | garymc | ManxPower-work. I upgraded all of it. I got 2 of these phones and done both of them |
| 14:55.07 | garymc | one is working and one is saying network link down. Thsi particular phone wouldnt even connect to the dhcp until this bootrom upgrade etc |
| 14:55.22 | ManxPower-work | that didn't answer my statement |
| 14:55.25 | [TK]D-Fender | garymc: So maybe its screwed up |
| 14:55.45 | coppice | [TK]D-Fender: Wireless Fiber is an interesting concept |
| 14:55.45 | tdjacobs | guys, is speex wideband supported in asterisk? |
| 14:55.46 | garymc | I upgraded firmware and sip.cfg and phone1.cfg |
| 14:55.57 | [TK]D-Fender | garymc: Flush the phone, provision from FTP from a new folder with the latest bootrom & sip app extracted from scratch |
| 14:56.06 | [TK]D-Fender | coppice: AMAZING isn't it? |
| 14:56.41 | garymc | [TK]D-Fender thats what ive just done. It formated and everything |
| 14:56.50 | [TK]D-Fender | garymc: NEW provisioning folder. |
| 14:56.58 | *** join/#asterisk Iskorptix_ (~iskorptix@213.197.141.162) |
| 14:57.02 | Iskorptix_ | hello |
| 14:57.03 | *** join/#asterisk moy (~chatzilla@74.12.129.100) |
| 14:57.09 | ManxPower-work | garymc: which phone model? |
| 14:57.17 | garymc | POlycom IP550 |
| 14:57.26 | Iskorptix_ | I got a simple question, could anyone of you recommend good book for absolute beginner of asterisk ? |
| 14:57.31 | Iskorptix_ | or manual* or anything else |
| 14:57.41 | ManxPower-work | ~answers |
| 14:57.42 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 14:58.27 | Iskorptix_ | thanks |
| 14:58.48 | ManxPower-work | garymc: what file contains your phone specific registration info? |
| 14:59.10 | ManxPower-work | garymc: are you running LLDP on your network? |
| 14:59.14 | garymc | 2345-12500-001.sip.ld |
| 14:59.25 | ManxPower-work | garymc: FAIL! Try again. |
| 14:59.35 | ManxPower-work | What file contains the phone specific REGISTRATION info? |
| 15:00.05 | garymc | oh my mac address.cfg? |
| 15:00.10 | garymc | or phone205.cfg? |
| 15:00.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:00.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:00.41 | ManxPower-work | garymc: it would be the file that contains the reg.* entries telling your phone what it's userid, password, etc is. |
| 15:01.02 | *** join/#asterisk ralonso (~ralonso@140.Red-88-2-26.staticIP.rima-tde.net) |
| 15:01.06 | garymc | oh ohone205.cfg |
| 15:01.13 | garymc | *phone205.cfg |
| 15:01.16 | ralonso | hi |
| 15:01.16 | ManxPower-work | Is that file listed in your 00000000000.cfg? |
| 15:01.49 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:02.37 | ManxPower-work | If you have MAC.cfg files (which were deprecated in SIP 2.1)? |
| 15:02.39 | garymc | yes |
| 15:04.04 | Katty | grooves |
| 15:04.34 | garymc | it was 000000000000.cfg I edit it for each phone and rename it with the MAc address of the phone . right? |
| 15:04.52 | garymc | Well Ip550 ext 204 works. 205 is setup the same and it doesnt |
| 15:05.34 | ManxPower-work | garymc: That is correct for the old pre 2.1 firmware. |
| 15:06.01 | ManxPower-work | It should work for more modern firmwares, but it's not recommended per polycom. |
| 15:06.26 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:06.53 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:08.21 | ralonso | anyone can help me with one cuestion about record in dialplan? |
| 15:08.27 | leifmadsen | ~ask |
| 15:08.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:10.48 | ralonso | ok, i have a digium aex2400 digium card with a FXO modules connect to a alcatel analog extensions, im trying to call from alcatel extension to asterisk the call progress w/o any problem this is my dialplan: |
| 15:11.21 | ManxPower-work | ~pb |
| 15:11.22 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 15:11.23 | ralonso | exten => s,1,Wait(2) |
| 15:11.23 | ralonso | exten => s,2,Answer |
| 15:11.23 | ralonso | exten => s,3,Record(/tmp/proba1:gsm|100|10|) |
| 15:12.05 | ManxPower-work | garymc: Page A-2 of the admin guide. |
| 15:12.39 | garymc | you got a link please. I got about 20 admin guides going |
| 15:13.41 | ChannelZ | ralonso: Record(/tmp/proba1.gsm,100,10) |
| 15:14.08 | ralonso | until here everything works fine, but when I hang up the alcatel side |
| 15:14.33 | ralonso | on the side of the asterisk is not short me and recorded busy tone |
| 15:14.37 | garymc | Well im guessing there is definatley somethign wrong with one of my phones. Why it says network link down is beyond me. |
| 15:14.41 | ManxPower-work | ChannelZ: That would depend on the Asterisk version |
| 15:14.54 | ManxPower-work | garymc: network link down is a network issue |
| 15:14.55 | ChannelZ | ralonso: press # to stop recording |
| 15:15.12 | Czindy | Hello again. We tested the iax peers and it seems that they lost their registered state after a minute automatically: http://pastebin.com/iF1Pd2An ([TK]D-Fender could you take a look at this please) |
| 15:15.18 | ChannelZ | ralonso: or lower the first 100 arg (at least in * 1.6 that is seconds of silence) |
| 15:15.41 | ManxPower-work | garymc: there is only ONE admin guide |
| 15:15.47 | ChannelZ | ralonso: for whatever reason your system isn't detecting the hangup or it probably would terminate the recording long before |
| 15:16.06 | garymc | ManxPower-work : But I plug all phones into same network they all work. I swap a phone for this one it works? |
| 15:16.34 | ManxPower-work | ChannelZ: that's pretty typical when trying to interface to PBX analog station lines. They don't provide disconnect supervision |
| 15:16.48 | [TK]D-Fender | Czindy: Sorry, can't help from this point |
| 15:16.52 | ChannelZ | Hmm.. mine do |
| 15:17.15 | garymc | [TK]D-Fender : It says link down on the phone but the phone can make and recieve calls ..........??????? |
| 15:17.20 | ManxPower-work | ChannelZ: are you sure you're not interfacing with PBX analog CO ports? |
| 15:17.27 | [TK]D-Fender | garymc: Maybe your phone simply has issues |
| 15:17.29 | ManxPower-work | garymc: now you are just plain not making sense. |
| 15:17.38 | ralonso | but ChannelZ, its posible configure in chan_dahdi.conf? |
| 15:17.41 | garymc | Sorry it rings when i call it |
| 15:17.48 | ManxPower-work | If the phone says "link down" there is no way they are going to receive calls. |
| 15:17.59 | garymc | cant make any calls |
| 15:18.00 | ManxPower-work | maybe it actually says something else? |
| 15:18.11 | ManxPower-work | (10:17:14 AM) garymc: [TK]D-Fender : It says link down on the phone but the phone can make and recieve calls ..........??????? |
| 15:18.16 | ManxPower-work | You just SAID IT CAN MAKE CALLS! |
| 15:18.23 | garymc | NO: it says "LINK DOWN" |
| 15:18.27 | garymc | Yeah i made mistake |
| 15:18.44 | garymc | it also says 1 missed call. As I just called it |
| 15:19.02 | ManxPower-work | garymc: I wish you the BEST of luck. |
| 15:19.33 | garymc | OK i just called it from ext201. It ran I picked up 205(link down phone) and it worked |
| 15:19.49 | garymc | But if I pick up 205 and try to dial. No dial tone and cant dial |
| 15:20.12 | garymc | all this going on while NETWORK LINK DOWN is displayed |
| 15:20.49 | garymc | Im at the end of my tehter. It must be a faulty phone. I did buy it second hand |
| 15:21.07 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:21.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:21.16 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 15:21.22 | garymc | Do you know where I set the SNTP server in the CFG files? |
| 15:21.32 | ruben23 | hi, any problem with the DEBUG message appearing on my asterisk CLI -----> http://pastebin.com/CunbpwBx |
| 15:21.55 | [TK]D-Fender | garymc: Ctrl-F in the admin guide, and in sip.cfg <- |
| 15:22.04 | Katty | hugs Deeewayne |
| 15:22.13 | garymc | ok |
| 15:22.15 | garymc | thanks |
| 15:22.18 | ManxPower-work | ruben23: I don't see a problem |
| 15:22.31 | ralonso | ManxPower-work u think the problem is in alcatel side? |
| 15:22.34 | ManxPower-work | If you don't like that message then turn off debug. |
| 15:22.38 | Deeewayne | offers Katty waffles |
| 15:22.41 | Katty | Deeewayne: http://ustre.am/bEBU <- showing off my ferrets, again :> |
| 15:22.42 | [TK]D-Fender | ruben23: No |
| 15:22.51 | ManxPower-work | ralonso: I think it is a problem with the DESIGN of the Alcatel analog station ports |
| 15:22.55 | Katty | Deeewayne: ooooh waffles :> |
| 15:22.57 | Katty | Deeewayne: <3 |
| 15:23.06 | *** part/#asterisk iq (~iq@unaffiliated/iq) |
| 15:23.14 | Czindy | [TK]D-Fender: Is it possible to make a debug why the iax clients lost their connections? Could you help to do a little debug please? (I suppose that it could connect to the maxregexpire or qualify...freq...ok...notok) |
| 15:23.17 | ralonso | okok, i know |
| 15:23.46 | [TK]D-Fender | Czindy: No, I cannot |
| 15:23.50 | ralonso | thx for help ;) |
| 15:23.55 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 15:24.42 | Deeewayne | Katty, aaawww..... |
| 15:24.47 | ruben23 | ManxPower-work: ok, how about experiencing like on a random time client cant barely hear the caller, but sometimes also there is good quality on voice..worst case is sometimes client cant really hear the caller even the caller shout out.. |
| 15:25.16 | ruben23 | im uisng voip trunk... |
| 15:25.34 | [TK]D-Fender | ruben23: `then they are at fault |
| 15:26.36 | ruben23 | <PROTECTED> |
| 15:26.46 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 15:28.13 | ralonso | ManxPower-work one more cuestion, is posible to capture the cadence of busytone and implement if this rate is detected cut the call? |
| 15:28.32 | [TK]D-Fender | [10:25]<[TK]D-Fender>ruben23: `then they are at fault |
| 15:29.12 | [TK]D-Fender | ralonso: "callprogress=yes", "busydetect=yes" <--- for your channel defs |
| 15:29.43 | ManxPower-work | [TK]D-Fender: what did ralonso do to piss you off? |
| 15:29.59 | ralonso | if i configure the callprogess to yes and busdetect yes call not progress |
| 15:30.01 | ralonso | jeje |
| 15:30.05 | ManxPower-work | callprogress=yes should be renamed to be randomlydisconnectmycalls=yes, because that's what it will do. |
| 15:30.14 | ruben23 | TK]D-Fender: you mean the voip carrier...? |
| 15:30.32 | ManxPower-work | I was also not aware callprogress even worked on SIP |
| 15:30.46 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 15:31.13 | garymc | Has anyone got a less jumbled up phone1.cfg file than the one polycom provide? |
| 15:31.24 | [TK]D-Fender | ruben23: YES |
| 15:31.35 | garymc | Im having a hard time trying to find all the stuff i need to configure |
| 15:31.37 | [TK]D-Fender | garymc: Do as I have advised for that phone. |
| 15:31.40 | ManxPower-work | <PROTECTED> |
| 15:31.45 | vader-- | is anyone using and/or has seen microsoft OCS/Exchange used as a PBX system? |
| 15:32.41 | garymc | You mean reboot from fresh folder? |
| 15:32.59 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 15:33.09 | [TK]D-Fender | garymc: I mean what i said earlier |
| 15:33.19 | garymc | ManxPower-work : I have to edit and save as phon205.cfg otherwise how does the phone know what extension it is etc |
| 15:34.54 | ralonso | uhm then it is possible to do something with the parameters of "cadence" |
| 15:38.01 | garymc | Right im going to have to re read all this stuff |
| 15:38.12 | garymc | f-sake another wasted day |
| 15:42.21 | garymc | nah im sure its this dodgey phone that doesnt work and its making me question the setup, but the other phone worls fine |
| 15:42.32 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 15:42.53 | garymc | Where do you set the number of line keys in phone1.cfg I cant find the option |
| 15:43.30 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 15:44.15 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 15:47.50 | *** join/#asterisk korcan (~kshamoun@ip65-44-169-89.z169-44-65.customer.algx.net) |
| 15:51.03 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 15:51.13 | garymc | OK anyone know where I can find detials on all the options in the phone.cfg files? |
| 15:51.24 | garymc | that way Ill just trial and error it all |
| 15:52.22 | garymc | No? |
| 15:52.27 | garymc | I guess not |
| 15:54.16 | TheDavidFactor | I'm trying to install Asterisk 1.6.2 w/res_odbc on CentOS 5.4; I've installed unixODBC and unixODBC-devel re-ran configure, but res_odbc is still X'd out, can anyone tell me what I've missed? |
| 15:55.48 | *** join/#asterisk Rich_T9980 (~rich@ool-4572cc0e.dyn.optonline.net) |
| 15:56.06 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:56.28 | *** part/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:56.48 | Rich_T9980 | hello is there anyone that can help me with the asterisk web server? |
| 15:57.04 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 15:58.13 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:59.18 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
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| 16:00.26 | leifmadsen | TheDavidFactor: did you install the unixODBC-dev package? |
| 16:00.30 | seanbright | ~ask |
| 16:00.31 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:00.33 | seanbright | Rich_T9980: ^^^^ |
| 16:00.58 | TheDavidFactor | leifmadsen, it was called unixODBC-devel in yum, but yes I did |
| 16:01.12 | leifmadsen | TheDavidFactor: oh ya, using too much ubuntu lately :) |
| 16:01.17 | *** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net) |
| 16:01.31 | leifmadsen | TheDavidFactor: umm, at this point I would check configure.log (I think that's the file...) and look through for ODBC to see what it is trying to find that it isn't |
| 16:01.55 | leifmadsen | TheDavidFactor: also, sometimes a 'make distclean' will verify that something isn't messing with you |
| 16:02.22 | vader-- | Adtran just wrote me back, they said my demo unit is going to be delayed because of a production issue or something with the TA9xx series |
| 16:02.24 | vader-- | wtf |
| 16:03.25 | thehar | When not using zaptel is there a way to increase gain on voicemail recordings? Besides the g(#) on VoiceMail? |
| 16:03.48 | [TK]D-Fender | thehar: read the sample config <- |
| 16:03.57 | thehar | hrm is there an rxgain in there? |
| 16:03.59 | thehar | loox |
| 16:04.11 | thehar | ooooh volgain=. |
| 16:04.14 | thehar | dur de dur.. ty [TK]D-Fender |
| 16:04.21 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:04.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:04.49 | thehar | that is on recording, [TK]D-Fender? |
| 16:05.00 | TheDavidFactor | leifmadsen, thanks I'll try that |
| 16:05.25 | *** join/#asterisk Rich_T9980 (~rich@ool-4572cc0e.dyn.optonline.net) |
| 16:05.34 | thehar | tests it out |
| 16:06.00 | leifmadsen | thehar: I think the VOLUME() function MAY help if you're using 1.6.x (I think 1.6.0.x and greater) |
| 16:06.19 | thehar | leifmadsen: ty! |
| 16:06.20 | leifmadsen | thehar: although I'm not sure if that can increase gain in both directions |
| 16:06.28 | leifmadsen | I haven't used it yet :) |
| 16:06.38 | thehar | elderly customers complain of "quiet voicemails". |
| 16:06.47 | thehar | and refuse to turn up their phone volume or are unable to figure it out. |
| 16:06.50 | leifmadsen | ah yes, they will do that |
| 16:06.51 | thehar | hehe |
| 16:06.59 | leifmadsen | the + sign is certainly confusing |
| 16:07.04 | thehar | laff |
| 16:07.12 | thehar | leifmadsen: how are you doing? been a while! |
| 16:07.24 | leifmadsen | oh not too bad! Just busy busy busy! |
| 16:07.28 | thehar | ditto |
| 16:07.38 | leifmadsen | been working on TFoT 3 each morning for an hour for the last 2 weeks |
| 16:07.45 | leifmadsen | gonna be a long route to getting stuff done, but we'll get there |
| 16:07.54 | thehar | nods |
| 16:08.00 | leifmadsen | hopefully ready in time shortly after 1.8.0 release |
| 16:08.06 | leifmadsen | which is a few months away |
| 16:08.11 | thehar | i've been having to help interview a new director of netops .. == not fun |
| 16:08.19 | thehar | 1.8 yay |
| 16:08.19 | leifmadsen | eep |
| 16:08.41 | thehar | oh wow.. my boss is on slashdot |
| 16:08.45 | leifmadsen | heh |
| 16:08.53 | thehar | http://yro.slashdot.org/story/10/02/24/025225/Utah-Considers-Warrantless-Internet-Subpoenas |
| 16:09.18 | thehar | also dealing with that bill as well |
| 16:09.20 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:10.08 | leifmadsen | oy |
| 16:10.35 | thehar | << excited for 1.8 |
| 16:12.36 | TheDavidFactor | ok, I've looked at the config.log file I and I see the following: ac_cv_lib_iodbc_SQLConnect=no then a few more lines down: ac_cv_lib_odbc_SQLConnect=yes what's the difference between iodbc and odbc? |
| 16:15.37 | Corydon76-dig | Two different libraries |
| 16:15.44 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 16:15.56 | Corydon76-dig | odbc=unixODBC |
| 16:15.59 | TheDavidFactor | are both required for res_odbc? |
| 16:16.07 | Corydon76-dig | At least one |
| 16:16.35 | TheDavidFactor | I'm trying to figure out why menuselect is showing res_odbc x'd out |
| 16:16.39 | Corydon76-dig | And no, they are not both required |
| 16:16.51 | Czindy | hi again, we tested the iax clients with zoiper also, and the peers lost the connection after a little time. |
| 16:16.51 | Corydon76-dig | Probably because you don't have ltdl |
| 16:17.22 | Czindy | Is it possible to debog somehow this... here is a log from this: http://pastebin.com/tbZgynr5 |
| 16:17.28 | Corydon76-dig | Blame RedHat for not making ltdl a dependency of odbc |
| 16:18.02 | Corydon76-dig | Because even though they didn't, odbc won't work without it |
| 16:18.03 | *** join/#asterisk Peste_Bubonica (~eduardo@189.63.241.226) |
| 16:18.11 | TheDavidFactor | do I need ltdl-devel? because the ltdl is installed |
| 16:18.16 | Corydon76-dig | Yes |
| 16:18.20 | TheDavidFactor | ok, thx |
| 16:18.36 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:18.51 | Corydon76-dig | paulc! |
| 16:19.53 | paulc | Oi oi :) |
| 16:19.56 | paulc | how goes? |
| 16:20.16 | Corydon76-dig | Not bad, LTNS |
| 16:20.58 | paulc | Yeah, s'been a while.. day job's been keeping me busy.. a bit too busy.. and this stuff's more fun, so I've been lurking a bit, on and off.. what's new with you? |
| 16:21.01 | Katty | okay parents. |
| 16:21.04 | Katty | i need your help. |
| 16:21.07 | TheDavidFactor | Corydon76-dig: that was it, thanks! leifmadsen, thanks for your help too! |
| 16:21.12 | Katty | they make these foamy type blocks, that fit together |
| 16:21.13 | Rich_T9980 | can anyone help me with the asterisk web server config files? |
| 16:21.16 | Katty | and you can build things from them |
| 16:21.27 | Katty | they're intended for preschoolers i believe. and rather large blocks like 12" or so |
| 16:21.32 | Katty | anyone have any idea what i'm talking about? |
| 16:21.46 | leifmadsen | TheDavidFactor: under the list of deps in menuselect, does res_odbc show that it needs ltdl? might be a useful addition if not |
| 16:22.28 | paulc | Katty: Like this? http://www.amazon.com/gp/product/B000FK9AVA?ie=UTF8&tag=wwwbestchildt-20&linkCode=as2&camp=1789&creative=390957&creativeASIN=B000FK9AVA |
| 16:22.51 | TheDavidFactor | yes it does, but I thought ltdl was installed because I saw it get installed as a dependency, until Corydon76-dig enlightened me, I didn't actually check to see if it was the devel version |
| 16:23.08 | Katty | paulc: no, they're flat |
| 16:23.12 | Katty | paulc: about an inch thick maybe |
| 16:23.49 | Katty | paulc: they're rectangular and interconnect |
| 16:23.53 | paulc | Katty: Hmm.. I dunno, sorry :$ |
| 16:23.57 | Katty | paulc: k |
| 16:24.57 | Katty | http://www.viewpoints.com/images/review/2009/102/17/1239574740-73929_full.jpg <- like these |
| 16:25.06 | Katty | kinda. |
| 16:26.20 | leifmadsen | Katty: oh! just go to a place like Home Depot |
| 16:26.34 | leifmadsen | Katty: I've got a set that look like diamond plate, but they had coloured ones too |
| 16:26.57 | Katty | leifmadsen: do you think you could build a 3 dimensional object with them? |
| 16:27.01 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 16:27.02 | Katty | leifmadsen: or are they not very sturdy |
| 16:27.03 | leifmadsen | Katty: yes you could |
| 16:27.13 | leifmadsen | Katty: you could build a block with them yes |
| 16:27.19 | leifmadsen | they foam is reasonably ridgic |
| 16:27.20 | Rich_T9980 | hello? |
| 16:27.21 | leifmadsen | ridgid |
| 16:27.22 | Katty | leifmadsen: do you think lowes would carry it? we don't have a home depot around here |
| 16:27.24 | leifmadsen | rigid* |
| 16:27.36 | *** join/#asterisk khronos (tom@sip02.kineticvoip.com) |
| 16:27.42 | Katty | :> |
| 16:27.47 | leifmadsen | Katty: ya, any of those places should carry them. I've seen them up at the checkout lane, or at the back near the wood or tools |
| 16:28.02 | Katty | so it's sold as a mat? |
| 16:28.06 | leifmadsen | Katty: they gets used for things like cold floors or cement floors when you're standing on them (that's how they are marketed there) |
| 16:28.12 | Katty | k |
| 16:28.23 | leifmadsen | they are sold as a package of about 8 or so I think |
| 16:28.26 | Katty | nods |
| 16:28.33 | Katty | that's enough to build a little ferret hidaway |
| 16:28.33 | leifmadsen | they are all individual peices |
| 16:28.42 | leifmadsen | yep for sure |
| 16:28.47 | leifmadsen | they aren't too expensive either |
| 16:28.53 | paulc | LOL is that what you're building, Katty? |
| 16:28.57 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 16:29.01 | Katty | mhmm |
| 16:29.07 | leifmadsen | you could build a 1x1 block for sure |
| 16:29.10 | Katty | i was looking through the preschool stuff at target the other day |
| 16:29.18 | Katty | and they had these plastic 'houses' |
| 16:29.22 | Katty | with a little door and such |
| 16:29.32 | seanbright | Rich_T9980: hello |
| 16:29.40 | leifmadsen | I'm thinking like a 1 high, 2 or 3 long would be easily built if you covered one end to make it steady |
| 16:29.42 | Katty | but all the extra stuff in there would just get in the way |
| 16:30.05 | geneticx_wrk | morning |
| 16:30.16 | seanbright | Rich_T9980: just ask your question. if someone knows the answer, they will give you the answer. |
| 16:30.33 | Katty | ah HA |
| 16:30.37 | Katty | they're called Edu tiles |
| 16:30.51 | Katty | and really expensive |
| 16:30.56 | Katty | i'll stick with the interlocking mats |
| 16:31.09 | garymc | Im just trying to find out how I set the ring volume to highest in the sip.cfg . Is there a global setting. cos phones are always set to medium ring volume |
| 16:31.15 | garymc | for polycom |
| 16:31.26 | khronos | Anybody know how I can lock down a Linksys spa8000 unit so that a user can not reset the unit to defaults? |
| 16:31.50 | Rich_T9980 | thanks sean, I am having a problem with the web server for the asterisk-gui. I have replaced my 'http.conf and manager.conf' files and now have the web server running. How ever I still get a 501 error =( |
| 16:32.08 | Katty | i should get the ferrets a little fisher price phone for their cage :> |
| 16:32.43 | paulc | Katty: Not a Polycom? ;) |
| 16:33.22 | Katty | well. |
| 16:33.28 | Katty | i guess i could mount one on the wall next to the cage |
| 16:33.37 | Katty | so i can call them on add sip header auto answer. |
| 16:34.02 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:34.07 | Katty | might be kinda weird tho |
| 16:34.15 | garymc | [TK]D-Fender : Do you know what I set in sip.cfg to get the phones to have the loudest volume setting set all the time? |
| 16:34.27 | Katty | what do you call doll houses for boys? |
| 16:34.31 | Katty | the big like pirate houses |
| 16:34.37 | Katty | and lego castle thing |
| 16:35.02 | [TK]D-Fender | garymc: Just set the "persist" options |
| 16:35.31 | [TK]D-Fender | garymc: And if you feel you have to do that then either your users are deaf, or you have not normalized to PSTN connectivity |
| 16:36.15 | garymc | iam slightly deaf so is my boos |
| 16:36.25 | garymc | The persist are set to 0 |
| 16:36.34 | garymc | What do I set them to for full? is it 1? |
| 16:37.00 | geneticx_wrk | if I want to use a digium TDM410 PCI card with asterisk, do I need to install the ZAP channel Module |
| 16:37.02 | geneticx_wrk | ? |
| 16:37.27 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:38.27 | paulc | geneticx_wrk: I think DAHDI is the new ZAP these days |
| 16:38.31 | garymc | [TK]D-Fender : Normalized to PSTN connectivity. Why would I not have and how would I check? |
| 16:38.42 | *** join/#asterisk rare1980_ (~rare1980@115.186.26.218) |
| 16:39.06 | [TK]D-Fender | garymc: Call from phone to phone with mid-way volume. If the PSTn isn't roughly equal then you should adjust your gains |
| 16:39.12 | geneticx_wrk | paulc: Aha..been out of the loop for a while. Thank you sir |
| 16:39.24 | garymc | ok |
| 16:39.34 | garymc | That would be handset volume |
| 16:39.50 | garymc | ITs just when the phone rings they are very quiet. The actual calls are fine |
| 16:40.08 | paulc | garymc: the ringer volume should be sticky though |
| 16:40.23 | paulc | garymc: crank it to max while the phone is idle, and all subsequent calls should ring at max volume too, no? |
| 16:40.25 | garymc | no it goes back to mid way after a little while |
| 16:40.41 | paulc | garymc: really? hmm.. is that after a config change/reboot? or "just on its own"? |
| 16:40.51 | garymc | just on its own |
| 16:41.59 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:42.21 | [TK]D-Fender | garymc: [11:34]<garymc>[TK]D-Fender : Do you know what I set in sip.cfg to get the phones to have the loudest volume setting set all the time? <--- this didn't say RINGING VOLUME |
| 16:42.41 | [TK]D-Fender | garymc: and your ringer IS persistent automatically |
| 16:42.43 | garymc | No I know, but Ive since cleared that one up |
| 16:42.49 | [TK]D-Fender | garymc: Except through reboots |
| 16:42.56 | Katty | hmmmmm. |
| 16:42.58 | garymc | ahh ok reboots |
| 16:43.03 | Katty | i bet the make teeny tiny dog houses. |
| 16:43.07 | Katty | for toy dogs. |
| 16:43.12 | garymc | maybe thats the problem . Cos I ve been rebooting the bugers that often |
| 16:43.13 | Katty | plots |
| 16:43.27 | garymc | *buggers |
| 16:43.29 | garymc | :P |
| 16:45.05 | coppice | Katty: they make plenty of teeny tiny people houses, too |
| 16:45.19 | garymc | Why is there 2 files for the IP330 sip.ld ? |
| 16:45.55 | Katty | coppice: something that would fit a ferret? |
| 16:46.15 | etfonhomey | garymc, there is one that containes all images in a giant sip.ld and one that contains only the image for the 330, I believe. |
| 16:46.36 | Katty | coppice: http://www.callingalldogs.com/images/products/detail/SweetDreamHouse.jpg <- that would be pefect |
| 16:46.42 | coppice | Katty: I've seen bedrooms less than 6' in each direction. maybe those are for ferrets |
| 16:47.04 | garymc | etfonhomey : no they are both the same size files |
| 16:47.28 | Katty | coppice: i think that would qualify as a closet. |
| 16:47.56 | coppice | they seem to be bedrooms your children can really outgrow |
| 16:48.06 | Katty | no need for kids rooms. |
| 16:48.23 | etfonhomey | garymc, what zip file did you get them out of? |
| 16:48.24 | Katty | but if i moved to a house with a 6ftx6ft room, i think i'd probably turn it inot a closet |
| 16:48.37 | Katty | or the ferret playroom |
| 16:48.42 | etfonhomey | garymc, or where did you download them? |
| 16:48.48 | Katty | and put up a big 3ft by 2ft plexiglass room divider |
| 16:48.49 | carrar | or a Monkey den |
| 16:48.52 | Katty | so they can't get out |
| 16:48.57 | coppice | Katty: you wouldn't fit in very well here :-) |
| 16:49.12 | Katty | probably not |
| 16:49.14 | paulc | wishes the marketing team would shut up about being in the media for their olympic protesting yesterday - who cares about Muk Muk?! |
| 16:49.27 | garymc | etfonhomey I got them from polycom support site. 3.2.2split.zip |
| 16:49.48 | Katty | coppice: i'm one of those people who would carpeted shelves up the wall for a cat. |
| 16:50.03 | etfonhomey | Which files are for the 330? |
| 16:50.12 | etfonhomey | (Just being lazy about looking it up) :) |
| 16:50.51 | garymc | 2345-12200-001.sip.ld |
| 16:50.54 | garymc | 2345-12200-004.sip.ld |
| 16:51.07 | Naikrovek | just download the split 3.2.2 SIP firmware and you'll be fine |
| 16:51.17 | Naikrovek | grab the latest bootloader while you're there |
| 16:51.49 | garymc | Naikrovek this is the latest. But for IP330 there are 2 sip.ld files. Doesnt specify which one to use |
| 16:51.58 | Naikrovek | you don't need to specify |
| 16:52.02 | Naikrovek | the phone knows which one it needs |
| 16:52.12 | Naikrovek | you just dump the entire zip file into a directory |
| 16:52.17 | Naikrovek | the phone grabs the one IT needw |
| 16:52.19 | Naikrovek | needs* |
| 16:52.25 | etfonhomey | garymc, that's a good question, even the Release notes, lists both but doesn't explain the diff. |
| 16:52.32 | Naikrovek | FORGET THE DIFF |
| 16:52.35 | Naikrovek | YOU DON'T NEED TO KNOW |
| 16:52.38 | Naikrovek | THE |
| 16:52.40 | Naikrovek | PHONE |
| 16:52.41 | Naikrovek | KNOWS |
| 16:52.48 | Katty | coppice: http://www.petstrends.com/wp-content/uploads/pet-living-tips-cat-catwalk.jpg <- that's the kind of person i am. |
| 16:52.50 | carrar | YOU CAN'T HANDLE THE TRUTH |
| 16:52.51 | seanbright | yes. what he said. |
| 16:53.09 | garymc | Yeah the phone knows? so which one do i put in there? |
| 16:53.11 | *** join/#asterisk Mhaddog (~Mhaddog@74-221-243-58.ds1-static.mia1.net.ststelecom.com) |
| 16:53.14 | garymc | :) |
| 16:53.17 | Naikrovek | download the firmware |
| 16:53.18 | seanbright | both |
| 16:53.22 | seanbright | stop being a jackass |
| 16:53.23 | *** join/#asterisk roni (~roni@190.196.71.206) |
| 16:53.35 | roni | hi everybody ! what s up ? |
| 16:53.42 | Naikrovek | place the .zip into wherever directory the phone pulls its config from |
| 16:53.44 | Naikrovek | unzip it |
| 16:53.45 | Naikrovek | DONE |
| 16:53.49 | Naikrovek | stop making it hard |
| 16:54.02 | garymc | but my 000000000cfg tells it to download a specific file. So do i put them both in under same name. I know I cant do that |
| 16:54.15 | Naikrovek | that should be "sip.ld" |
| 16:54.17 | Naikrovek | don't fuck with it |
| 16:54.18 | etfonhomey | garymc, the phone will look for the specific one |
| 16:54.25 | etfonhomey | regardless of the config file. |
| 16:54.38 | Naikrovek | it will look for phone-part-number.[whatever you put in the config file] |
| 16:54.51 | Naikrovek | so if your phone is 1234-12312312.12312312312312fkoasjf;sadfjfuck!!!!!!! |
| 16:55.03 | Naikrovek | it will look for "1234-12312312.12312312312312fkoasjf;sadfjfuck!!!!!!!.sip.ld" |
| 16:55.10 | etfonhomey | garymc, turn verbose up on your tftp server (or what ever you use) and look at the logs, you'll see it request the phone specific one. |
| 16:55.14 | Naikrovek | sorry for the attitude |
| 16:55.18 | Naikrovek | i'm really sick |
| 16:55.25 | Naikrovek | leave the config file to say "sip.cfg" |
| 16:55.30 | Naikrovek | that's only the suffix |
| 16:55.33 | paulc | Sicky McSickerson? Or sick of your job? no, wait, that's me... |
| 16:55.33 | roni | just a liittle help , i got a digium with 8 lines conected , i have my extension.conf configured like this |
| 16:55.36 | roni | exten=>_8X.,1,DIAL(DAHDI/8/${EXTEN:1}) |
| 16:55.36 | roni | exten=>_8X.,2,Hangup() |
| 16:55.37 | garymc | fuk me im sick too, im bleeding out my arse eyes and mouth |
| 16:55.42 | Naikrovek | heh |
| 16:55.43 | Naikrovek | ow |
| 16:55.44 | garymc | cos of this shit |
| 16:55.45 | carrar | or just read the admin guide |
| 16:55.46 | carrar | <PROTECTED> |
| 16:55.59 | paulc | garymc: NIIIICE mental image there mate :-s |
| 16:56.05 | garymc | your welcome |
| 16:56.16 | roni | how can i do , if i want to use another line to call .. like if i call 8-xxxxxx and the line dahdi-8 is busy .. to use another one |
| 16:56.20 | Naikrovek | garymc: it's simple |
| 16:56.24 | garymc | coughs his guts up and farts them out at the same time |
| 16:56.33 | Naikrovek | garymc: make one config file for each phone |
| 16:56.44 | Naikrovek | garymc: 0004[rest of mac address].cfg |
| 16:56.49 | garymc | yes i do that. |
| 16:57.03 | garymc | But in that file I have to specify the particular sip.ld |
| 16:57.03 | Naikrovek | okay |
| 16:57.06 | Naikrovek | no you don't |
| 16:57.09 | drfreeze | Anyone had a D-channel go bad on a TE121 card (digium) |
| 16:57.10 | Naikrovek | that's where you're confusing things |
| 16:57.13 | *** join/#asterisk Omorika (~omorika@193.198.31.85) |
| 16:57.15 | garymc | eg for my ip550 They use sip550.ld |
| 16:57.15 | Omorika | hi |
| 16:57.19 | Omorika | chan_iax2.c: Received trunked frame before first full voice frame |
| 16:57.25 | Naikrovek | garymc: it should be "sip.ld" ONLY |
| 16:57.26 | paulc | roni: why not use a group? Get 9 to dial out using a group, then 8X to dial out on line X |
| 16:57.27 | Omorika | what does this mean??? |
| 16:57.33 | Omorika | my log is full of this message |
| 16:57.33 | Rich_T9980 | is there any other config files for the asterisk web server other than manager.conf and http.conf? |
| 16:57.42 | [TK]D-Fender | roni: typically you will assign your dahdi channels into GROUPS so you can dial from the first available channel in the group. |
| 16:58.09 | garymc | Naikrovec even though im using 3 differnt types of polycom phones |
| 16:58.21 | Naikrovek | garymc: yes, even if you're using 20 different polycom phones |
| 16:58.22 | roni | nice , this is what i need .. lets go to google .... THANKS ! |
| 16:58.27 | [TK]D-Fender | roni: like "group=1" , "channel => 1", "group=1" , "channel => 2", "group=1" , "channel => 3", etc. then you can Dial(DAHDI/g1/1234567890) |
| 16:58.28 | Naikrovek | garymc: leave it as sip.cfg |
| 16:58.37 | garymc | yes i left that one |
| 16:58.40 | Naikrovek | garymc: the phone will know to look for it's part number + "sip.ld" |
| 16:58.53 | [TK]D-Fender | roni: g1 = group 1, ascending order. G1 = group1 descending order |
| 16:59.02 | Naikrovek | garymc: i have 5 different polycom phones here, all my config files say "sip.ld" |
| 16:59.07 | [TK]D-Fender | roni: Number as appropriate, group as appropriate |
| 16:59.08 | Katty | so what's for lunch today |
| 16:59.15 | Naikrovek | Katty: jimmy john's |
| 16:59.17 | Naikrovek | yum |
| 16:59.25 | roni | <[TK]D-Fender> : Thank you .. i will do that |
| 16:59.28 | Katty | we have one of those, but i've never been there |
| 16:59.39 | Naikrovek | Katty: delish. #1 pepe if you like ham & cheese |
| 16:59.49 | etfonhomey | DAHDI, DAHDI, we likes to party. We don't cause no trouble. We don't bother nobody. |
| 16:59.59 | Katty | checks their website for nutritional information |
| 17:00.08 | Katty | Naikrovek: which sandwich would you recommend? |
| 17:00.17 | Naikrovek | Katty: i get a #1 |
| 17:00.33 | Naikrovek | which is ham, cheese, mayo, lettuce, tomato |
| 17:00.41 | Naikrovek | and is yum |
| 17:00.52 | Katty | wowser |
| 17:00.55 | Katty | 620 calories |
| 17:00.57 | Naikrovek | no kidding |
| 17:00.59 | Naikrovek | wtf |
| 17:01.10 | Katty | yeah |
| 17:01.10 | Naikrovek | how can ham and mayo be that many calories |
| 17:01.20 | Katty | without maybe it's 425 |
| 17:01.25 | Katty | s/maybe/mayo/ |
| 17:01.35 | florz | .62 kcal? that's not much ... |
| 17:01.38 | Katty | 361 without the ham |
| 17:01.38 | *** join/#asterisk rare1980_ (~rare1980@115.186.26.218) |
| 17:01.46 | carrar | without bread? |
| 17:01.47 | Naikrovek | ah the french bread is like 250 |
| 17:01.54 | carrar | Just the meat pls |
| 17:01.55 | rare1980_ | hey all, |
| 17:02.08 | Naikrovek | unreal |
| 17:02.15 | Naikrovek | not eating there again any time soon |
| 17:02.18 | Katty | lol, no bread, no mayo, 175 calories |
| 17:02.24 | carrar | heh |
| 17:02.34 | Naikrovek | trying to lose weight here, jimmy john's |
| 17:02.45 | carrar | walk to jimmie johns? |
| 17:02.45 | Katty | the vegetarian is 575 |
| 17:03.02 | Katty | you can get 6 inch subs from subway for around 350 calories |
| 17:03.04 | Naikrovek | carrar: it's only a few hundred feet away. |
| 17:03.09 | carrar | oh |
| 17:03.11 | *** join/#asterisk AndyGraybeal (~AndyGray@128-177-27-78.ip.openhosting.com) |
| 17:03.16 | Katty | and bullets from quiznos for about 350 too |
| 17:03.34 | carrar | large glass of water |
| 17:03.40 | rare1980_ | any one is it possible.. that if i hev 2 sip accounts for each call each sip account will be in use? |
| 17:03.42 | carrar | and some popcorn |
| 17:03.44 | carrar | heh |
| 17:03.53 | Katty | that doesn't sound very lunch-worthy |
| 17:03.56 | Katty | :/ |
| 17:04.11 | Omorika | anyone? |
| 17:04.14 | Omorika | need help, pls |
| 17:04.17 | rare1980_ | means all :) |
| 17:04.46 | etfonhomey | Omorika, ask and you shall receive... ...maybe |
| 17:04.55 | Omorika | i asked |
| 17:05.01 | Naikrovek | Omorika: (s)he did ask |
| 17:05.03 | Omorika | "chan_iax2.c: Received trunked frame before first full voice frame" <------ what this means? |
| 17:05.22 | Omorika | asterisk is spamming me with this message |
| 17:05.27 | Naikrovek | Omorika: are you able to make calls across that trunk |
| 17:05.39 | Katty | Jimmy John's is really somethin. |
| 17:05.50 | Katty | an arby's beef and cheddar melt is like 350 calories |
| 17:05.57 | Katty | there's no reason for a sub to be 600 something |
| 17:06.15 | Naikrovek | no kidding |
| 17:06.21 | Naikrovek | beef & cheddar = awesome |
| 17:07.27 | Rich_T9980 | is this an asterisk channel? |
| 17:07.41 | Naikrovek | Rich_T9980: doesn't seem like it sometimes, but yes |
| 17:07.47 | carrar | food channel |
| 17:08.11 | Rich_T9980 | i wasnt sure i was asking my question in the correct place as i am a newb to irc. |
| 17:08.26 | Naikrovek | go ahead and ask your question |
| 17:08.37 | Rich_T9980 | i did nobody answered |
| 17:08.40 | Rich_T9980 | lol |
| 17:08.46 | Naikrovek | oh i missed it |
| 17:08.49 | Naikrovek | ask again? |
| 17:09.15 | Rich_T9980 | im having a problem with the asterisk web server working properly |
| 17:09.17 | Naikrovek | ah i see it |
| 17:09.22 | Naikrovek | asterisk doesn't have a web server |
| 17:09.34 | Naikrovek | are you using FreePBX? Trixbox? Elastix? AsteriskNOW? |
| 17:09.39 | Rich_T9980 | probly why i had received no response then |
| 17:09.40 | Rich_T9980 | lolo |
| 17:09.54 | Rich_T9980 | im trying to run asterisk-gui |
| 17:10.09 | Naikrovek | Rich_T9980: you need to /j #asterisk-gui |
| 17:10.12 | Rich_T9980 | i was told that the webserver was a component of asterisk not asterisk-gui |
| 17:10.35 | Rich_T9980 | hence my presence in this channel |
| 17:10.37 | Naikrovek | well the web server is apache probably |
| 17:10.53 | Naikrovek | but #asterisk-gui can probably help you |
| 17:11.00 | Rich_T9980 | thats what I thought originally until someone from the asterisk-gui channel said otherwise |
| 17:11.05 | Naikrovek | really |
| 17:11.21 | Rich_T9980 | yes because i am also running mediawiki |
| 17:11.25 | Naikrovek | well i'm sure asterisk has no web server of its own |
| 17:11.42 | Rich_T9980 | thought that may have some config issue I had to figure out to make apache work for both |
| 17:11.55 | Naikrovek | why don't you just run asterisk on its own hardware |
| 17:12.06 | Naikrovek | or its own virtual machine or something - better yet - put media wiki in a virtual machine |
| 17:12.24 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 17:13.02 | Rich_T9980 | i would believe the QOS would be affected running in a vm |
| 17:13.37 | Naikrovek | i run asterisk in a VM and it's fine |
| 17:13.40 | *** join/#asterisk Victor_Yure_ (~victor@unaffiliated/victoryure/x-837844) |
| 17:13.50 | Naikrovek | but maybe you're right, my environment is not your environment |
| 17:13.54 | *** part/#asterisk Omorika (~omorika@193.198.31.85) |
| 17:14.33 | Rich_T9980 | it really is a simple process from what i gather on the net getting this working....<as i bang my head into the desk> |
| 17:14.55 | seanbright | asterisk has a web server |
| 17:14.59 | seanbright | built-in |
| 17:15.05 | Naikrovek | really |
| 17:15.06 | Naikrovek | wtf |
| 17:15.10 | seanbright | really |
| 17:15.11 | Naikrovek | since ... oh never mind |
| 17:15.18 | Naikrovek | shoots himself in the face |
| 17:15.28 | seanbright | AJAM |
| 17:15.33 | Rich_T9980 | yes AJAM |
| 17:15.39 | Rich_T9980 | theres the acronym |
| 17:15.55 | Rich_T9980 | my apology for not using it |
| 17:16.09 | seanbright | no one asked you to apologize |
| 17:16.33 | seanbright | is there anything in the logs relevant to the error you are seeing? |
| 17:16.40 | *** join/#asterisk brian (~brian@unaffiliated/brian) |
| 17:17.06 | Rich_T9980 | not that I can see, if I run 'make checkconfig' it actually says everything is ok |
| 17:17.09 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 17:17.45 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 17:17.47 | seanbright | so where are you seeing the 501? |
| 17:18.14 | Rich_T9980 | there really isnt anything other than file permissions that would cause this outside from the 2 configuration files 'manager.conf & http.conf' ? |
| 17:18.28 | Rich_T9980 | when i goto http://192.168.10.20:8088/static/config/index.html |
| 17:21.16 | seanbright | then there should be something in the logs |
| 17:21.32 | seanbright | /var/log/asterisk/full maybe? |
| 17:22.34 | Rich_T9980 | which log file should i look at specifically? |
| 17:22.42 | seanbright | /var/log/asterisk/full maybe? |
| 17:22.52 | Rich_T9980 | i doo not see /var/log/asterisk/full |
| 17:23.02 | seanbright | then don't look at that file |
| 17:23.11 | seanbright | do you have any files in /var/log/asterisk |
| 17:23.26 | Rich_T9980 | yes but empty |
| 17:24.51 | Rich_T9980 | event_log event_log.1 messages etc etc |
| 17:25.01 | Rich_T9980 | no contents though |
| 17:25.09 | seanbright | hmm |
| 17:25.53 | tzafrir | Rich_T9980, do you happen to be familiar with apache? |
| 17:26.11 | Rich_T9980 | no =( |
| 17:26.44 | tzafrir | IIRC requests are not logged by default |
| 17:26.46 | Rich_T9980 | that is why i did not know at first that AJAM was seperate |
| 17:27.04 | tzafrir | (In my setup I proxy requests through apache, and this also provides me logs) |
| 17:27.13 | Rich_T9980 | seanbright, here are some errors that come up when i reload asterisk.. http://paste.ubuntu.com/383133/ |
| 17:27.51 | ManxPower-work | Rich_T9980: IF you're not using ldap then those are not errors |
| 17:27.52 | tzafrir | Rich_T9980, those are unrelated |
| 17:27.53 | Rich_T9980 | tzafrir, I see the advantages as I read running it that way. I am going to try it now. |
| 17:28.21 | Rich_T9980 | ok ty |
| 17:28.37 | TheDavidFactor | do you have to do anything special to get dahdi_dummy installed? I downloaded dahdi-linux-complete 2.2.1 and did a basic make, make install and dahdi_dummy isn't there |
| 17:28.49 | seanbright | Rich_T9980: is there an error that goes along with your 501? |
| 17:28.53 | seanbright | in the browser |
| 17:29.10 | Rich_T9980 | "Attempt to use unimplemented / unsupported method" |
| 17:29.13 | [TK]D-Fender | TheDavidFactor: isn't where? |
| 17:29.25 | Rich_T9980 | " NOT IMPLEMENTED" |
| 17:30.19 | seanbright | i think i see a bug |
| 17:30.21 | tzafrir | so it's not a static file? |
| 17:30.33 | tzafrir | What do you see in 'http show status' ? |
| 17:30.59 | seanbright | no. no bug. |
| 17:31.20 | TheDavidFactor | [TK]D-Fender: isn't in asterisk, when I try dahdi show status I get: No DAHDI found. Unable to open /dev/dahdi/ctl: No such file or directory |
| 17:31.32 | Rich_T9980 | here is the 'http show status' http://paste.ubuntu.com/383140/ |
| 17:31.32 | seanbright | probably a path issue |
| 17:31.52 | seanbright | ...? |
| 17:31.55 | [TK]D-Fender | TheDavidFactor: "modprobe dahdi_dummy" "dahdi_cfg -vvvv" THEn start * manually |
| 17:32.13 | seanbright | oh, nm. |
| 17:32.26 | tzafrir | Rich_T9980, try: http://192.168.10.20:8088/asterisk/static/config/index.html |
| 17:32.26 | Rich_T9980 | i dont know where else other than those 2 config files that I would need to address a path change |
| 17:32.27 | seanbright | oh |
| 17:32.28 | seanbright | yeah |
| 17:32.30 | seanbright | heh |
| 17:32.37 | seanbright | what tzafrir said |
| 17:32.47 | TheDavidFactor | [TK]D-Fender: thx |
| 17:32.59 | Rich_T9980 | tzafrir, "404 - not found" |
| 17:33.27 | tzafrir | now it's the issue of location of the static files (static-http) |
| 17:33.27 | Rich_T9980 | "The requested URL was not found on this server." |
| 17:33.45 | Rich_T9980 | tzafrir, how does one define that path? |
| 17:34.14 | tzafrir | Where are the files right now? |
| 17:34.16 | *** join/#asterisk V4mpire (~gary@82.118.111.254) |
| 17:34.21 | seanbright | it's /var/lib/asterisk/static-http |
| 17:34.24 | seanbright | or it should be |
| 17:34.41 | Rich_T9980 | yes |
| 17:34.46 | Rich_T9980 | that is correct seanbright |
| 17:34.54 | tzafrir | The package has /usr/share/asterisk/static-http |
| 17:34.55 | V4mpire | hi all does anyone know of a free voip provider that offers free calls even if its a small amount just to test that my phone works correctly so i can test settings |
| 17:35.21 | seanbright | well it uses ast_config_AST_DATA_DIR |
| 17:35.28 | seanbright | Rich_T9980: pastebin your /etc/asterisk/asterisk.conf |
| 17:35.48 | tzafrir | (it's not set in asterisk.conf, it's set in the code) |
| 17:36.04 | Rich_T9980 | tzafrir, I have that path with no files located in it '/usr/share/asterisk/static-http' |
| 17:36.04 | seanbright | ast_config_AST_DATA_DIR is overridable in asterisk.conf |
| 17:36.32 | tzafrir | basically two things to do: |
| 17:36.40 | tzafrir | 1. symlink around the issue |
| 17:36.41 | Rich_T9980 | copy the files to that path? |
| 17:36.54 | tzafrir | 2. install the files to /usr/share/asterisk/static-http |
| 17:37.04 | tzafrir | (2) is what I have in my package) |
| 17:37.05 | [TK]D-Fender | V4mpire: test what settings? |
| 17:37.13 | seanbright | or change asterisk.conf |
| 17:37.25 | *** part/#asterisk Mhaddog (~Mhaddog@74-221-243-58.ds1-static.mia1.net.ststelecom.com) |
| 17:37.26 | seanbright | astdatadir=/var/lib/asterisk/ |
| 17:37.48 | V4mpire | test that everything connects correctly and it can make/recieve calls so i know if i will need to change any router settings for it to work |
| 17:38.41 | [TK]D-Fender | V4mpire: free outbound is rare. free inbound has a few |
| 17:38.49 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 17:38.55 | [TK]D-Fender | V4mpire: or just set up a service like ekiga.net and test with any external client |
| 17:38.59 | Rich_T9980 | seanbright, here is the asterisk.conf http://paste.ubuntu.com/383143/ |
| 17:39.14 | seanbright | astdatadir => /usr/share/asterisk |
| 17:39.18 | Rich_T9980 | yup |
| 17:39.21 | seanbright | best to just install static-http there |
| 17:39.35 | seanbright | but you could also just update that path and restart asterisk |
| 17:39.39 | seanbright | but... meh |
| 17:40.01 | Rich_T9980 | i would set that path to astdatadir=/var/lib/asterisk/ ? |
| 17:40.06 | seanbright | yeah |
| 17:40.12 | seanbright | but just copy the files |
| 17:40.18 | seanbright | to /usr/share |
| 17:40.27 | Rich_T9980 | ok |
| 17:40.31 | seanbright | makes life less complicated |
| 17:40.36 | Rich_T9980 | ill try that |
| 17:41.14 | Rich_T9980 | just the files in static-http correct? |
| 17:42.39 | seanbright | i believe so |
| 17:42.48 | seanbright | i'm just making up this as i go |
| 17:43.04 | Rich_T9980 | your good at this on the fly |
| 17:43.21 | seanbright | i learned everything i know from tzafrir |
| 17:43.27 | Rich_T9980 | sorry i would have had it done but im a newb with shell so I dont understand the cp command well |
| 17:43.28 | Rich_T9980 | lol |
| 17:44.28 | Rich_T9980 | what the hell syntax would I use 'cp *.* <from inside /var/lib/asterisk/stattic-http>'? |
| 17:45.02 | seanbright | cp -R /var/lib/asterisk/static-http /usr/share/asterisk |
| 17:45.07 | seanbright | i think |
| 17:46.01 | Rich_T9980 | guess what |
| 17:46.11 | Rich_T9980 | YOU THOUGHT RIGHT! |
| 17:46.14 | seanbright | heh |
| 17:46.17 | Rich_T9980 | hehe |
| 17:48.53 | Rich_T9980 | ummmm |
| 17:49.04 | Rich_T9980 | ok this is interesting lol |
| 17:49.44 | Rich_T9980 | I logged in and it did somethings and said I had to restart asterisk....so I reloaded and logged back in and Im stuck in a loop of pages refreshing? |
| 17:50.39 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 17:50.45 | seanbright | no clue |
| 17:51.12 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 17:51.18 | Rich_T9980 | hey I got into the gui I dont know how to ty enough! |
| 17:52.03 | *** join/#asterisk Zuchmir2 (~Zuchmir-i@ool-18bd3bfc.dyn.optonline.net) |
| 17:54.06 | Zuchmir2 | is there a way to force a SIP channel to change RTP audio to an alternate IP, in middle of a call? |
| 17:54.29 | Zuchmir2 | i.e. using the UPDATE or re-invite? |
| 17:55.42 | *** join/#asterisk QubeZ (~qube@64.128.254.34) |
| 17:55.45 | idespinner | Zuchmir2, dont the monitor/bargein features do that sorta? |
| 17:55.47 | QubeZ | hello all |
| 17:55.51 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 17:55.52 | idespinner | that may be a place to look... |
| 17:55.55 | sun28 | moin \o/ |
| 17:56.12 | *** join/#asterisk Elvis__ (~elvis@s69-163-39-5.in-addr.arpa.static.dsn1.net) |
| 17:56.14 | QubeZ | I am running Centos 5.4 Final and installed kernel-devel (/usr/src/kernels/2.6...) however, when I run 'make all' for dahdi it errors: You do not appear to have the sources for the 2.6.18-164.el5 kernel installed. |
| 17:56.27 | Zuchmir2 | purpose: i have 2 internet feeds, if 1 feed goes down i want to flip all calls to secondary feed |
| 17:56.31 | QubeZ | Am I suppose to point to the source or should it be found automagically? |
| 17:56.31 | idespinner | QubeZ, you need the kernel headers... |
| 17:56.32 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 17:56.55 | QubeZ | idespinner headers and sources should be installed via 'yum install kernel-devel' no? |
| 17:56.59 | Elvis__ | hello... I have a question I was wondering if anybody could help me with... I have installed asterisk Asterisk 1.6.2.0~rc2 on ubuntu server 9.10 and I am having an issue with it not loading the extensions.conf at all. The only thing it loads in the extensions.ael. Anyone know what I could be doing wrong? |
| 17:57.44 | idespinner | QubeZ, maybe yum install kernel-headers |
| 17:57.46 | QubeZ | idespinner: Package kernel-headers-2.6.18-164.11.1.el5.i386 already installed and latest version |
| 17:58.05 | idespinner | uname -a |
| 17:58.18 | QubeZ | Package kernel-headers-2.6.18-164.11.1.el5.i386 already installed and latest version |
| 17:58.20 | QubeZ | err |
| 17:58.31 | QubeZ | Linux lab-asterisk01 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux |
| 17:58.44 | [TK]D-Fender | QubeZ: .... 11 <- |
| 17:58.44 | idespinner | hmmm |
| 17:59.04 | [TK]D-Fender | QubeZ: 2.6.18-164.11.1.el5 != 2.6.18-164.el5 |
| 17:59.21 | idespinner | QubeZ, maybe a reboot. You may have updated your kernel recently |
| 17:59.30 | *** join/#asterisk djdefi (~rtrauntve@209-234-189-154.static.twtelecom.net) |
| 17:59.36 | QubeZ | possibly, didn't think about that because I just installed 5.4 final from iso |
| 17:59.38 | Zuchmir2 | idespinner: i wonder if blindxfer is what i'm looking for |
| 17:59.41 | idespinner | and installed the latest headers, which do not match your current kernel version |
| 17:59.42 | QubeZ | rebooting now |
| 17:59.55 | idespinner | Zuchmir2, what do you mean by a feed... |
| 17:59.59 | QubeZ | makes sense, damn, feel stupid :\ |
| 18:00.02 | Elvis__ | anyone? |
| 18:00.05 | idespinner | and how do you propose switching? |
| 18:00.35 | idespinner | Elvis__, how did you install asterisk? |
| 18:00.43 | sun28 | Elvis__: split asterisk on a voip server and router |
| 18:00.50 | Elvis__ | idespinne: just with an apt-get install asterisk |
| 18:01.11 | Elvis__ | sun28: huh? |
| 18:01.26 | Rich_T9980 | seanbright, how would I change the write permissions for /var/lib/asterisk/stattic-http ? apparently I cant write there unless Im root |
| 18:01.41 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 18:01.43 | Zuchmir2 | idespinner: i have verizon internet, and cable, if one goes down, i can flip (either manually or automatically - based on outside program) between the 2 |
| 18:01.45 | idespinner | Elvis__, any errors with a dialplan reload? |
| 18:02.06 | sun28 | Elvis__: oops |
| 18:02.07 | Elvis__ | idespinne: no, it just shows extensions.ael loading |
| 18:02.20 | Elvis__ | totally ignores extensions.conf |
| 18:02.22 | sun28 | Zuchmir2: split asterisk on a voip server and router |
| 18:02.24 | Zuchmir2 | i'd like to send an "UPDATE" or re-invite to the remote server |
| 18:02.24 | sun28 | ^__^ |
| 18:02.52 | Zuchmir2 | sun28: please explain |
| 18:03.56 | sun28 | got PA+AS .. bgp peer to ISP ... bgp peer to yet another ISP ... |
| 18:06.34 | Elvis__ | the extensions.conf file doesn't need to be in an #include in any other files, does it? |
| 18:07.08 | [TK]D-Fender | Elvis__: No |
| 18:08.29 | sun28 | split extensions.ael on a different files like extensions_{dial,incoming,outgoing,local}.ael .. |
| 18:08.48 | Zuchmir2 | sun28: that's not an option at the moment |
| 18:08.55 | Elvis__ | is there a way I can get it to ignore the extensions.ael file and just use extensions.conf instead? |
| 18:09.04 | [TK]D-Fender | Elvis__: It will use both |
| 18:09.34 | Elvis__ | right, but is can I tell it to ignore the extensions.ael? |
| 18:09.40 | Zuchmir2 | was hoping * can send out an UPDATE to callers with no IP address |
| 18:09.47 | tzafrir | module unload pbx_ael.so |
| 18:09.48 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:09.49 | tzafrir | :-( |
| 18:09.55 | sun28 | Zuchmir2: I do it by hand in some configurations, edit {sip,h323}.conf files and change bindaddr= .. :( |
| 18:10.16 | Zuchmir2 | ... for RTP |
| 18:10.52 | Zuchmir2 | sun28: but that only changes NEW calls, the existing calls stay on existing IP |
| 18:10.59 | sun28 | Elvis__: delete unused modules from modules.conf |
| 18:11.14 | tzafrir | Elvis__, but seriously: edit /etc/asterisk/modules.conf and include: unload => pbx_ael.so |
| 18:11.29 | tzafrir | You might as well have unload => pbx_lua.so |
| 18:11.34 | tzafrir | if you have 1.6.2 |
| 18:11.44 | Elvis__ | tzafrir: ok. Still doesn't seem to load the extensions.conf file, when I do a dialplan show it displays only what's defined in extensions.ael and nothing that's defined in extensions.conf |
| 18:12.14 | tzafrir | Elvis__, what happens if you run: module reload pbx_config.so |
| 18:12.57 | sun28 | Elvis__: e.g. http://dpaste.com/164262/ |
| 18:13.47 | Elvis__ | ah... that was it... found an error in my extensions.conf file |
| 18:13.57 | tzafrir | sun28, I'm not sure I'd recommend explicit modules loading to someone who's not familiar with all the dialplan applications and such |
| 18:14.57 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 18:22.10 | *** join/#asterisk aruntomar (~aruntomar@61.17.193.163) |
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| 18:23.09 | *** part/#asterisk djdefi (~rtrauntve@209-234-189-154.static.twtelecom.net) |
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| 18:31.17 | Katty | hmm |
| 18:33.55 | aruntomar | hi! we are getting a beep sound while talking when recording the calls, even when we've not set it up to beep..... any ideas where i should start looking |
| 18:34.36 | [TK]D-Fender | aruntomar: * cli. |
| 18:35.58 | aruntomar | [TK]D-Fender: we are using zoiper, and using sip as the protocol |
| 18:36.30 | *** part/#asterisk LemensTS (~customgt@70.238.161.207) |
| 18:36.57 | aruntomar | [TK]D-Fender: i didn't understand the meaning of * cli |
| 18:37.12 | Naikrovek | * cli = "asterisk CLI" |
| 18:37.29 | Naikrovek | cli = "command line interface" |
| 18:37.56 | aruntomar | Naikrovek: i'm already there |
| 18:38.09 | *** join/#asterisk yang (~yang@freenode/sponsor/cacert.assurer.yang) |
| 18:38.43 | aruntomar | we get this beeping sound on our zoiper softphone |
| 18:39.06 | Naikrovek | is the beeping in the recording? |
| 18:39.17 | Naikrovek | also |
| 18:39.20 | Naikrovek | have you tried it with a real phone |
| 18:39.29 | aruntomar | we are recording every call by default |
| 18:39.42 | Naikrovek | why are you doing that |
| 18:39.46 | [TK]D-Fender | aruntomar: And you are simply enlarging the story and not showing us anything |
| 18:39.50 | Naikrovek | why are you recording every call |
| 18:40.01 | aruntomar | customer req |
| 18:40.16 | Naikrovek | well make sure your employees know otherwise it's illegal in most states |
| 18:40.25 | Naikrovek | anyway |
| 18:40.32 | Naikrovek | does the beeping exist in the recordings? |
| 18:40.39 | Naikrovek | and does the beeping happen when you use a real, hard, phone |
| 18:40.49 | sun28 | record every call by res_monitor |
| 18:41.42 | aruntomar | we are using only softphones, & don't have the hard phone, i guess, will need to buy one for all these purposes and testing |
| 18:42.04 | Naikrovek | well at the moment I, personally, suspect zoiper. you could try alternative soft phones |
| 18:42.11 | aruntomar | we've informed the service providers and customers of the recordings |
| 18:42.48 | Naikrovek | alright. just fyi, in all states except nevada, phone calls can't be recorded unless all participants on the call know about the recording |
| 18:43.02 | Naikrovek | fallout from monica lewinsky |
| 18:43.03 | aruntomar | we are on linux (ubuntu) and ironically, none of them work well except zoiper |
| 18:43.11 | Naikrovek | really |
| 18:43.17 | Naikrovek | have you tried ekiga? |
| 18:43.31 | Naikrovek | that's the only one i've used, but i've used it on windows |
| 18:43.56 | aruntomar | we tried ekiga, xlite, iax, and many more and then only settled down on zoiper |
| 18:44.01 | Naikrovek | oh okay |
| 18:44.26 | Naikrovek | get a physical voip phone and see if it continues |
| 18:44.27 | Naikrovek | otherwise |
| 18:44.37 | Naikrovek | find out if there are any error messages in the asterisk cli |
| 18:44.47 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 18:44.57 | Naikrovek | also (third time) do the beeps exist in the recordings? |
| 18:45.20 | aruntomar | it was working fine before but since the time we restarted the server once for hardware maintenance, we find that the telecallers can't hear the ring for 27 sec, after 27 sec they can hear the ring, i tried to google a lot, but i guess there is not explanation to it |
| 18:45.44 | Naikrovek | i guarantee there's a reason for it somewhere |
| 18:45.47 | aruntomar | i'll check right away, if the recordings have the beep sound |
| 18:45.47 | Naikrovek | you just need to find it |
| 18:46.07 | ManxPower-work | ringback issues are frequently caused by /etc/asterisk/indications.conf missing |
| 18:46.30 | aruntomar | i understand that, and i tried very hard to find it, when i couldn't then i turned up here to some help and guidance |
| 18:48.11 | aruntomar | indications.conf is there, i just checked that |
| 18:49.47 | ManxPower-work | aruntomar: Are you using the "m" option to Dial? |
| 18:50.19 | [TK]D-Fender | 10 minutes later.... where is the CALL to look at? |
| 18:51.16 | aruntomar | yes, we are using the m option |
| 18:51.31 | ManxPower-work | aruntomar: remove it and see if that fixes the issue. |
| 18:52.04 | aruntomar | ManxPower-work: with the same settings, it was working fine earlier |
| 18:52.55 | ManxPower-work | aruntomar: You can take my advice or not. I did not say that removing the "m" option is the solution. It is only one step. |
| 18:53.30 | ManxPower-work | The "m" option tells the Dial command to play music on hold instead of ringing. |
| 18:53.42 | ManxPower-work | This has not changed since 1998 |
| 18:54.28 | [TK]D-Fender | ManxPower-work: This one doesn't seem to listen much.... |
| 18:54.29 | ManxPower-work | So if you got ringing before when using the "m" option then your Music on Hold was BROKEN before. |
| 18:54.29 | aruntomar | ManxPower-work: i really appreciate ur help, i'll try that also |
| 18:54.29 | ManxPower-work | [TK]D-Fender: I noticed. |
| 18:54.33 | ManxPower-work | I'm just killing time while a firmware downloads. |
| 18:54.45 | aruntomar | i'll just check the recordings and will be back shortly |
| 18:54.51 | ManxPower-work | He has about 8 mins of my time left. |
| 18:55.18 | ManxPower-work | I take it he had multiple issues since he's talking about recordings? |
| 18:55.30 | *** join/#asterisk clouddeveloper (~clouddeve@zeus.clouddeveloper.co.uk) |
| 18:55.37 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 18:55.40 | [TK]D-Fender | ManxPower-work: His actual issue was an unwanted "beep' during his recordings |
| 18:55.54 | ManxPower-work | [TK]D-Fender: Why was he babbling about ringing then? |
| 18:56.48 | [TK]D-Fender | ManxPower-work: res_randomtopicjump.so |
| 18:57.01 | ManxPower-work | [TK]D-Fender: Ah. He's just an idiot then. |
| 18:57.27 | [TK]D-Fender | ManxPower-work: Ask random question. Ignore requests for proper information. Change topic. Wash. Rinse. Repeat. |
| 18:57.59 | ManxPower-work | we both need to just start ignoring those people. |
| 18:58.24 | [TK]D-Fender | ManxPower-work: http://www.benz0.com/stuff/pancake%2520bunny.jpg |
| 18:59.31 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:00.35 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 19:04.20 | Elvis__ | alright... so, I seem to be having another problem... I have phones connected to the PBX, I can see them authenticated if I do a sip show peers... but if I try and dial any extension from the phone, even it's down, I get a fast busy and the console says the call is rejected because the extension can't be found. |
| 19:04.42 | Elvis__ | *dial any extension, even it's own |
| 19:04.54 | p3nguin_ | its own |
| 19:04.58 | [TK]D-Fender | Elvis__: Becaus there is no match in the context 8 is loking in <- |
| 19:05.21 | [TK]D-Fender | Elvis__: Contexts matter. Go verify that is even LOOKINg in the one you think it should be, and that there is something in there that should match |
| 19:05.31 | [TK]D-Fender | Elvis__: "sip set debug on" <---- |
| 19:07.07 | Elvis__ | does the context need to be defined for each extension? |
| 19:08.49 | ManxPower-work | Elvis__: the context= line in the sip.conf entry tells Asterisk which context to send the call to! |
| 19:08.59 | ManxPower-work | What is the context= line in sip.conf for that device? |
| 19:09.21 | ManxPower-work | Elvis__: you do not define extensions in sip.conf. You define DEVICES. |
| 19:09.47 | [TK]D-Fender | Elvis__: extension = an "exten =>" line in EXTENSIONS.CONF |
| 19:11.10 | Elvis__ | hm... yea, that's what I thought... but those contexts are definitly defined... and work on all my other asterisk boxes. |
| 19:11.16 | *** join/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 19:11.20 | ManxPower-work | Elvis__: I'm waiting for the answer to my question |
| 19:11.49 | [TK]D-Fender | Elvis__: And the reason we don't see your failed call attempt at * CLI with SIP DEBUG and your dialplan is....? |
| 19:11.51 | [TK]D-Fender | ~pb |
| 19:11.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 19:11.53 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 19:11.59 | ManxPower-work | (2:08:58 PM) ManxPower-work: What is the context= line in sip.conf for that device? |
| 19:12.14 | *** join/#asterisk nny (~scott@64.203.239.83) |
| 19:12.25 | nny | any way to compile dahdi 2.2.1 with glibc 2.5 ? |
| 19:12.42 | nny | centos 5.4 doesn't even have 2.8 in it's repos, and hitting dependency hell trying to add it |
| 19:13.29 | ManxPower-work | Elvis__ is going to be one of those users where we die of old age waiting for his answers. |
| 19:13.36 | *** join/#asterisk ecrane (~ecrane@o1-69-19-166-10.static.o1.com) |
| 19:14.12 | Elvis__ | yea... lemme piece of together... sorry guys... |
| 19:14.28 | ManxPower-work | nny: um, it workse just fine with glibc 2.5.42 CentOS 5.3, I think |
| 19:14.59 | ManxPower-work | nny: What SPECIFIC issue are you having with GLIBC 2.5? |
| 19:15.11 | nny | scripts/mod/modpost: /lib64/libc.so.6: version `GLIBC_2.8' not found (required by scripts/mod/modpost) |
| 19:15.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:15.32 | nny | that's the error when I try to run make in dahdi-linux |
| 19:15.55 | ManxPower-work | nny: delete config.cache |
| 19:15.58 | nny | all this for ztdummy :\ |
| 19:16.17 | nny | in dahdi-linux-2.2.1 ? |
| 19:16.28 | ManxPower-work | Oh! You are on a 64-bit platform. I must have missed that critical piece of information when you told us. |
| 19:16.37 | nny | har har smart ass |
| 19:17.01 | ManxPower-work | No. Over and over again I see users having issues with 64-bit platforms (especially with CentOS). |
| 19:17.05 | nny | ffs |
| 19:17.12 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.16 | nny | hold on let me call rackspace and tell them to go fuck themselves brb |
| 19:17.20 | nny | oh wait i can't |
| 19:17.50 | *** join/#asterisk iq_ (~iq@ext-B14-257.omhq.uprr.com) |
| 19:17.55 | nny | eh, ok thanks for the "input" |
| 19:17.57 | p3nguin_ | nny: "its repos" |
| 19:18.10 | nny | p3nguin_: come again? |
| 19:18.24 | nny | i have tried pbone x86_64 repo, no love |
| 19:18.45 | p3nguin_ | (1312.42) <nny> centos 5.4 doesn't even have 2.8 in it's repos, and hitting dependency hell trying to add it <-- its repos |
| 19:19.15 | ManxPower-work | nny: he's a grammer queen |
| 19:19.18 | paulc | heard our IT ops guys moaning about Centos, after the new IT manager said it was the way to go (throwing out Ubuntu) - thoughts? |
| 19:19.30 | ManxPower-work | paulc: we use only centos where I work |
| 19:19.45 | p3nguin_ | manxpower-work: What makes me a queen? |
| 19:19.52 | nny | has worked fine before, this is a vm though, and I am stuck with this kernel and this architecture |
| 19:19.59 | ManxPower-work | p3nguin_: I guess I could have said you are a grammer nazi |
| 19:20.01 | p3nguin_ | manxpower-work: I don't see any crown or tiara on my skull. |
| 19:20.10 | nny | i also have another box (running an older rackspace kernel) that still uses ztdummy etc just fine |
| 19:20.14 | nny | so the question remains |
| 19:20.17 | p3nguin_ | manxpower-work: The word is spelled grammar, too. |
| 19:20.30 | nny | i would just use that image but rackspace can't copy images between accounts |
| 19:20.58 | nny | and I need ztdummy for DISA on this box |
| 19:21.06 | paulc | ManxPower-work: And all is well? (It may be that our guys aren't up on using alternate repos etc) |
| 19:21.24 | ManxPower-work | paulc: works just fine in the default repos |
| 19:21.35 | ManxPower-work | nny: you do not need ztdummy for DISA. |
| 19:21.44 | nny | hmm |
| 19:21.50 | nny | well it isn't working without it |
| 19:22.01 | nny | http://www.freepbx.org/forum/freepbx/users/xen-rtc-ztdummy-im-ticked |
| 19:22.21 | nny | just an example, excuse the source |
| 19:22.28 | ManxPower-work | nny: conferencing needs ztdummy. DISA does not. |
| 19:22.32 | ariel_ | humm I just finished up yesterday a new asterisk on rackspace CentOS 5.3 x64 no issues |
| 19:22.42 | paulc | ManxPower-work: Their issue was a newer version of SVN I think? *shrugs* they got it fixed and we're all happy now, 1.6.6 :) |
| 19:22.53 | ariel_ | I used dahdi_combine |
| 19:23.08 | nny | ariel_: yeah I dunno, this is 5.4 centos |
| 19:23.09 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 19:23.13 | p3nguin_ | I don't really see how my insisting the use of a possessive pronoun as opposed to a completely different word, which is a contraction, and doesn't make sense at all in the context, makes me a nazi. |
| 19:23.13 | carrar | nny, dahdi-linux-complete-2.2.1+2.2.1 compiles just fine on 32bit centos 5.4 |
| 19:23.28 | nny | all: I do NOT haver a choice in kernels here |
| 19:23.34 | ManxPower-work | carrar: he's not using 32-bit CentOS, he's using 64-bit centos |
| 19:23.35 | nny | it's a VM hosted by rackspace |
| 19:23.45 | p3nguin_ | ariel_: And there is no such thing as CentOS x64. Maybe you mean x86_64? |
| 19:23.46 | *** part/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 19:23.55 | ariel_ | yes p3nguin_ |
| 19:24.11 | nny | ManxPower-work: wonder why DAHDI ignores all key presses and just hangs up after dialing then |
| 19:24.15 | nny | er |
| 19:24.18 | nny | DISA |
| 19:24.19 | nny | not DAHDI |
| 19:24.26 | nny | acronym soup |
| 19:24.32 | p3nguin_ | mmmmmmm soup |
| 19:24.33 | ManxPower-work | nny: without pastebins it's all speculation |
| 19:24.47 | nny | i'll pastebin but console says nada when it happens |
| 19:25.06 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 19:26.29 | nny | hmmph |
| 19:26.32 | nny | i feel aloof |
| 19:26.44 | *** join/#asterisk adnc_ (~numer@unaffiliated/adnc) |
| 19:26.47 | nny | disa code had a pipe delimiter, all aboard the failboat |
| 19:27.06 | nny | was a config copy over from 1.4, didn't catch that error the first time |
| 19:27.11 | nny | I owe you a chicken dinner |
| 19:27.19 | ManxPower-work | I guess you didn't read the UPGRADE*.txt files that talk about the | and comma changes?????? |
| 19:27.31 | nny | nope, I wiped my ass with them and set them on fire though |
| 19:27.38 | ManxPower-work | Sounds to me that your problem is fatigue. |
| 19:27.40 | nny | and then I ate glue and cried myself to sleep |
| 19:27.55 | ManxPower-work | well go re-read them again. |
| 19:28.06 | nny | ok i'll clean them up first |
| 19:28.26 | ManxPower-work | read them as many times as needed to stop having problems with changes that are clearly documented in the UPGRADE*.txt files. |
| 19:28.41 | nny | so is conference the only thing that requires dahdi_dummy |
| 19:28.43 | nny | ? |
| 19:29.13 | ManxPower-work | app_conference is rumored to not need zaptel mixing. app_meetme has always needed zaptel mixing. |
| 19:29.15 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:29.19 | nny | gotcha |
| 19:29.20 | nny | ty |
| 19:29.26 | ManxPower-work | I don't think app_conference has been updated in years. |
| 19:29.31 | [TK]D-Fender | nny: IAX2 trunk mode, MoH "appreciates' it as well |
| 19:30.00 | nny | k none of those things exist on this box, it's just a glorified Auto Attendant and DISA to save some guy some bucks on international calling |
| 19:30.20 | ManxPower-work | make sure you have allowguest=no in sip.conf [general] |
| 19:30.30 | nny | heh |
| 19:30.33 | nny | that won't work for DIDWW |
| 19:30.37 | nny | for some shitty reason |
| 19:30.43 | nny | disa is pw protected |
| 19:30.58 | ManxPower-work | What did you say the IP of the PBX is? |
| 19:31.10 | nny | why? |
| 19:31.16 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 19:31.19 | V4mpire | hi anyone know of a voip service that offers multiple landline numbers from different countries for free which u can just top up to make outgoing calls |
| 19:31.41 | ManxPower-work | nevermind. If you get a big phone bill because someone is routing guest calls via your PBX I'm sure you'll add the option. |
| 19:31.49 | nny | how would they dial out? |
| 19:31.52 | ManxPower-work | Until then, why bother? |
| 19:31.59 | nny | they have no access to any context that allows dialing out |
| 19:32.10 | ManxPower-work | nny: without knowing your entire dialplan I can't say. |
| 19:32.24 | ManxPower-work | Do you wear a seatbelt? Afterall you' |
| 19:32.30 | ManxPower-work | re not going to be in an accident? |
| 19:32.51 | nny | i dunno, last time I tried allowguest=no and set the default context on *this box* to invalid DIDWW had a shit fit |
| 19:33.17 | nny | all my other boxes have allowguest=no by default |
| 19:34.01 | ManxPower-work | nny: If that happens then your incoming calls from that device were not matching the entry you configured for them. |
| 19:34.12 | ManxPower-work | Another reason to have allowguest=no. |
| 19:34.28 | ManxPower-work | nny: so basically any device that tried calling thru your server would work. |
| 19:34.35 | nny | well |
| 19:34.43 | nny | any device calling my server would hit the "default": context |
| 19:34.54 | ManxPower-work | nny: same place the DIDWWW calls hit |
| 19:35.00 | nny | correct |
| 19:35.01 | ManxPower-work | since both calls would have been "guest" |
| 19:35.10 | nny | which is where they should go |
| 19:35.18 | nny | the same efefct could be achieved by dialing the DIDWW number |
| 19:38.02 | nny | hmmph |
| 19:38.07 | nny | their server IP changed |
| 19:38.12 | nny | fixed that at least |
| 19:38.24 | nny | i do enjoy it when they do that |
| 19:39.12 | nny | dammit ha |
| 19:39.19 | nny | I have to add all their IP addresses in |
| 19:39.24 | nny | since they switch |
| 19:40.13 | nny | can you define "host" by a subnet vs a single IP? |
| 19:40.34 | [TK]D-Fender | nny: No. allow/deny |
| 19:41.29 | ManxPower-work | allow/deny is for INCOMING calls. host= is for OUTGOING calls. |
| 19:42.45 | nny | gotcha |
| 19:43.09 | nny | they use "insecure=very" which is depricated, whats a safe alternative? port, invite or both? |
| 19:43.10 | [TK]D-Fender | ManxPower-work: Host is for incoming as well... |
| 19:43.56 | nny | host works for incoming here, just made 5 entries for the servers as they suggest |
| 19:48.45 | carrar | nny, I just installed CentOS 5.4 64bit in vmware and dahdi-linux-complete-2.2.1+2.2.1 compiled just fine |
| 19:49.22 | p3nguin_ | nny: port,invite should be the new equivalent of the old very |
| 19:50.15 | voipmonk | what version of vmware, carrar ? |
| 19:50.44 | carrar | vmware fusion |
| 19:50.47 | carrar | latest |
| 19:50.57 | carrar | 3.0.2 |
| 19:51.27 | carrar | file linux/drivers/dahdi/dahdi_dummy.ko |
| 19:51.27 | carrar | linux/drivers/dahdi/dahdi_dummy.ko: ELF 64-bit LSB relocatable, AMD x86-64, version 1 (SYSV), not stripped |
| 19:53.06 | carrar | Linux centos5464 2.6.18-164.el5 #1 SMP Thu Sep 3 03:28:30 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux |
| 19:54.06 | *** part/#asterisk tdjacobs (~tiagoj@host00100.ht2p.net) |
| 19:54.27 | *** join/#asterisk d00gster (~d00gster@94.99.5.69) |
| 19:55.04 | Elvis__ | alright guys... sorry to be such a bother, but I've got one last problem... trying to make outbound calls, using a t1 card, all the t1s are up and configured if I do a dahdi show status, but when I try and make a call I get: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) |
| 19:56.54 | [TK]D-Fender | Elvis__: Where do we see your failed call and configs? |
| 19:57.07 | [TK]D-Fender | Elvis__: ......PASTEBIN! |
| 19:57.09 | [TK]D-Fender | ~pb |
| 19:57.09 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 19:57.10 | [TK]D-Fender | ^^^^^^^^6 |
| 19:57.40 | Elvis__ | http://pastebin.com/Byg5pdkX |
| 19:58.55 | Elvis__ | call log: http://pastebin.com/eirF9J57 |
| 20:03.27 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 20:03.31 | ManxPower-work | Elvis__: do you have a CAS T-1 or a PRI T-1? |
| 20:03.40 | Elvis__ | PRI |
| 20:04.47 | carrar | He's having a blue Christmas |
| 20:05.05 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:05.12 | ManxPower-work | Elvis__: pastebin your chan_dahdi.conf |
| 20:05.33 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 20:06.03 | ManxPower-work | My bet is that you set pridialplan, contrary to what is recommended in the config file, but I'd have to see the config file first. |
| 20:06.34 | ManxPower-work | Elvis__: It is disapointing that your two pastebins don't match. |
| 20:07.09 | ManxPower-work | in your extensions.conf pastebin you use g0 where in the CLI pastebin you are using G0. Something somewhere is very, very, very wrong. |
| 20:07.14 | Elvis__ | http://pastebin.com/BkEtmkH3 |
| 20:07.26 | Elvis__ | mmm... good catch |
| 20:08.12 | ManxPower-work | BTW, the term for that here is "the start of not believing anything the user stays" |
| 20:08.38 | ManxPower-work | Elvis__: try pasting the file without the comment lines |
| 20:08.57 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 20:09.20 | *** join/#asterisk duckz (~duckz@86.107.84.186) |
| 20:09.37 | *** join/#asterisk Jhirley (~Jhirley@h69-21-54-248.ldlwvt.dsl.dynamic.tds.net) |
| 20:11.21 | ManxPower-work | I suspect you'll see your problem then. |
| 20:12.04 | Elvis__ | yea... I'll play around with that... |
| 20:12.46 | ManxPower-work | no, just remove all the lines that start with a ; |
| 20:12.53 | ManxPower-work | then you should immediatly see the problem |
| 20:13.55 | Elvis__ | k |
| 20:14.09 | *** join/#asterisk ParanoyaM (~ParanoyaM@34-213-132-95.pool.ukrtel.net) |
| 20:14.13 | ParanoyaM | Hi All |
| 20:14.13 | *** join/#asterisk elkato (~kaka@34-202-17-190.fibertel.com.ar) |
| 20:14.43 | elkato | Hi ther.. |
| 20:14.48 | ParanoyaM | can anybody help me? i wrote a macros that handle errors, but for some reason it isn't work, can somebody check it? |
| 20:15.43 | [TK]D-Fender | ParanoyaM: PASTEBIN it and the CLI output showing the failure and perhaps |
| 20:15.45 | [TK]D-Fender | ~pb |
| 20:15.46 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 20:15.47 | [TK]D-Fender | ^^^^^^^^6 |
| 20:16.26 | *** part/#asterisk elkato (~kaka@34-202-17-190.fibertel.com.ar) |
| 20:17.11 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:18.14 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 20:19.41 | ParanoyaM | here it is: http://pastebin.ru/310719 |
| 20:19.58 | ParanoyaM | but i still recieve an error 28549] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 20:22.48 | ParanoyaM | [TK]D-Fender: no ideas? |
| 20:22.50 | [TK]D-Fender | ParanoyaM: where is the CALL OUTPUT? |
| 20:23.13 | ManxPower-work | ParanoyaM: you do not know enough to decide if something is important for us to know. |
| 20:23.14 | ParanoyaM | [TK]D-Fender: i am sorry, i am not sure i know what is call output |
| 20:23.28 | [TK]D-Fender | ParanoyaM: Asterisk CLI <------- |
| 20:23.29 | ManxPower-work | ParanoyaM: the CLI output of a failure |
| 20:23.33 | ParanoyaM | oh |
| 20:23.38 | ParanoyaM | one moment |
| 20:23.56 | ParanoyaM | i know why it is cause 20 because one of gw is down |
| 20:24.13 | [TK]D-Fender | ParanoyaM: that would do it.. |
| 20:24.18 | ParanoyaM | but in my macros i made a handler for this cause |
| 20:26.05 | Elvis__ | ok... I'm not too familiar with the chan_dahdi.conf file... here's what I get when I removed the commented line: http://pastebin.com/RULpi4MF |
| 20:26.12 | Elvis__ | what vital parts am I missing? |
| 20:26.32 | ManxPower-work | Elvis__: Pretty much everything |
| 20:26.38 | *** join/#asterisk brezular (~brezular@adsl-dyn169.91-127-8.t-com.sk) |
| 20:26.46 | ManxPower-work | you have no switch type, group, channels |
| 20:26.47 | Elvis__ | heh... |
| 20:26.56 | Elvis__ | ok... one sec |
| 20:27.12 | ManxPower-work | remember, channels must come AFTER the settings |
| 20:27.15 | *** part/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 20:27.54 | *** join/#asterisk qdot (~qdot@89.16.172.7) |
| 20:27.58 | qdot | :) |
| 20:28.00 | qdot | hello! |
| 20:29.46 | qdot | I'm new to asterisk.. I have an issue setting it up with SIP - I'm using voipuser.org as my DID, I have configured the termination to s@ourscale.eu (my domain name, which runs the asterisk) |
| 20:30.26 | ManxPower-work | qdot: your service is sending the calls to extension "s"? |
| 20:30.33 | ManxPower-work | "s" is NOT a wildcard. |
| 20:30.56 | qdot | hm.. let's try with a named extension.. |
| 20:31.15 | ManxPower-work | don't just randomly pick extensions. Pick the extension your provider is sending the call to |
| 20:31.52 | ManxPower-work | sip debug should show you where the provider is sending the call. |
| 20:32.48 | qdot | my provider is letting me pick where it would send the calls to |
| 20:33.14 | *** part/#asterisk nny (~scott@64.203.239.83) |
| 20:33.24 | p3nguin_ | Surely you didn't choose to send it to 's' extension. |
| 20:33.29 | p3nguin_ | 'Cause that would be silly. |
| 20:33.39 | ManxPower-work | qdot: You should use sip debug to CONFIRM htat. |
| 20:33.42 | carrar | Use DNIS Randomizor |
| 20:33.42 | qdot | I tried couple options.. |
| 20:33.48 | Elvis__ | does this look better? http://pastebin.com/zMhfELCJ |
| 20:36.46 | qdot | I set it to ourscale@ourscale.eu (if I get this working, I would randomize it) |
| 20:36.50 | [TK]D-Fender | Elvis__: No |
| 20:36.54 | qdot | http://pastebin.com/gF6qVe1v |
| 20:37.03 | qdot | and that's the incoming packet headers on my asterisk |
| 20:37.50 | [TK]D-Fender | Elvis__: you have not defined a single "channel =>" in there |
| 20:37.53 | qdot | so it's clearly getting there |
| 20:38.41 | Elvis__ | is there some sort of documentation that would explain this to me? I'm looking at all my other servers with the same conf and none of that is defined and they are working fine. |
| 20:38.43 | qdot | but I'm getting this error when handling this: [Feb 24 20:37:54] NOTICE[28287]: chan_sip.c:20012 handle_request_invite: Call from '' to extension 'ourscale' rejected because extension not found. |
| 20:39.14 | [TK]D-Fender | Elvis__: perhaps that is in an INCLUDED file whose INCLUD directive you just killed when wiping out comments <------- |
| 20:39.15 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 20:39.58 | Elvis__ | ah... yea... I'm an idiot... |
| 20:40.40 | [TK]D-Fender | Elvis__: watch it with the spray&pray there Rrambo... :p |
| 20:43.49 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 20:46.07 | afink | Hello everyone I am having users complaining about bad audio after being on the phone for 3-5 minutes.. They say the audio is cutting in and out and people complain of a ringing noise in the backgroud. This * system is using SIP to via an ITSP to connect to the outside world. Is this something I should play with the jitter buffer to attempt to correct? |
| 20:46.45 | p3nguin_ | qdot: So I guess you know that your call isn't going to a context containing that extension. |
| 20:47.28 | p3nguin_ | Or actually, it _is_ going to a context that does _not_ contain that extension. |
| 20:47.48 | *** join/#asterisk doubletoker (~warlock@adsl-17-233-129.jax.bellsouth.net) |
| 20:48.22 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 20:48.36 | *** join/#asterisk rizwank (~rizwank@76.89.131.47) |
| 20:48.59 | QbY | Anyone have any suggestions for dipping TNS/Verisign for CNAM? I don't know if our Asterisk can do this. We can dip via OPTIONS or SUBSCRIBE. |
| 20:49.00 | rizwank | Is there a favorite Conference setup GUI for Asterisk? |
| 20:49.00 | p3nguin_ | afink: How many calls are you doing simultaneously, what codecs are you using, and how much bandwidth do you have? |
| 20:49.52 | [TK]D-Fender | rizwank: FreePBX |
| 20:50.00 | afink | p3nguin_: on average 3 max calls, checking on the codecs and 3mb up and down |
| 20:50.02 | Katty | allllllllllloooooooooooooooooooooooooooooooooooooooooooooooooooooo nurse. |
| 20:51.13 | afink | p3nguin_: all I have in sip.conf is disallow gsm |
| 20:51.35 | p3nguin_ | You should disallow=all and then allow whichever codec you want. |
| 20:51.56 | doubletoker | I'm having problems with asterisk dropping my incoming calls, I'm getting this error no reply to our critical packet (see doc/sip-retransmit.txt). |
| 20:52.40 | p3nguin_ | If you have g.729 licenses, or if you want to run all your devices with g.729, you could try g.729 for less bandwidth consumption to see if that helps any. |
| 20:52.40 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:52.47 | doubletoker | I tried setting in sip.conf, session-timers=refuse and stuff like that going by other peoples problems and it's not working |
| 20:52.51 | p3nguin_ | afink: That was for you. ^^^ |
| 20:53.01 | *** join/#asterisk aandrade (~aandrade@201.22.27.252.dynamic.adsl.gvt.net.br) |
| 20:53.09 | doubletoker | if anyone has any ideas how I can fix this, I would appreciate it |
| 20:53.31 | afink | p3nguin_: I do not have any g.729 licenses |
| 20:53.47 | p3nguin_ | afink: Notice I gave you an alternative to licensing. |
| 20:54.16 | p3nguin_ | Most providers offer g.729, so use it everywhere and you don't have to transcode. |
| 20:58.30 | afink | p3nguin_: so I need to download the opensource implementation to try it? |
| 20:58.37 | p3nguin_ | no |
| 20:59.42 | afink | How do I try it out? Do you have a link or something you can toss me? In sip.conf I tried disallow=all then allow=g729 but that didn't work |
| 21:00.56 | Elvis__ | ok⦠here is my dahdi channel definitions: http://pastebin.com/kbx40Cvr and here is my call log: http://pastebin.com/Le5HPum3 but I just get dead-air on the phone⦠never rings. |
| 21:01.34 | Elvis__ | is that maybe a telco issue? or is there still something wrong with my config? |
| 21:02.11 | [TK]D-Fender | Elvis__: Where is your phone relative to *? |
| 21:03.18 | Elvis__ | not sure what you mean by that... dialing from an internal phone on the pbx to an external cell #. |
| 21:04.02 | p3nguin_ | afink: You have to do that on each device (peer) as well as your "trunk" peer. If every device/peer uses g729, then no transcoding is needed. It is possible that you will have to enable g729 on your provider through your customer portal. |
| 21:04.19 | afink | ok thank you! |
| 21:05.12 | [TK]D-Fender | Elvis__: Prove that that phone works 10% direct to * first. |
| 21:05.25 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:38e4:24ad:2ac0:762b) |
| 21:05.31 | cusco | hi |
| 21:05.33 | Elvis__ | I can dial internally, to other extensions fine |
| 21:05.57 | cusco | pri -> local asterisk -> iax -> remote asterisk -> sip extensions in remote network |
| 21:06.16 | cusco | now that remote asterisk is givving me messages like: [Feb 24 21:04:51] NOTICE[10774]: chan_sip.c:18223 handle_response_peerpoke: Peer '609' is now Lagged. (2043ms / 2000ms) |
| 21:06.29 | cusco | I can't figure out why peers are lagged |
| 21:06.38 | cusco | what would be most common reason? |
| 21:06.58 | cusco | http://paste.debian.net/61337/ |
| 21:08.37 | *** join/#asterisk dobry (~d@95.111.7.95) |
| 21:09.28 | Elvis__ | also, if it makes any difference... when I dial *in* (eg from a cell phone) to any of the DIDs active on the PBX, it immediately hangs up, with nothing posted to the asterisk console... |
| 21:09.34 | Elvis__ | which makes me think maybe telco? |
| 21:10.23 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.29.19) |
| 21:11.10 | idespinner | cusco, network is slow... |
| 21:11.16 | Katty | jitters |
| 21:11.23 | Katty | wtb blood transfusion |
| 21:11.29 | Katty | blood without caffeine |
| 21:12.37 | *** join/#asterisk brian (~brian@unaffiliated/brian) |
| 21:12.55 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 21:14.22 | afink | p3nguin_: when I did that I got an error in the cli on incoming call "no audio format to offer" something that I need to set on the phones possibly? |
| 21:15.05 | afink | http://pastebin.ca/1808572 |
| 21:15.58 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 21:17.05 | idespinner | afink, you need to allow compatible codecs in the peer config |
| 21:17.13 | idespinner | like: allow=all |
| 21:17.25 | idespinner | or allow=ulaw,alaw, |
| 21:17.42 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 21:18.19 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:19.23 | p3nguin_ | disallow=all allow=g729 |
| 21:19.26 | p3nguin_ | on all peers |
| 21:19.39 | p3nguin_ | Then make sure every device is using g729. |
| 21:20.05 | p3nguin_ | If some devices cannot use it, then we're going to hit a dead end. |
| 21:20.15 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:20.28 | afink | I have snom320s it is listed in the codecs so I think I should be ok |
| 21:22.24 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 21:22.34 | *** join/#asterisk ArtemMakhutov (~ArtemMakh@ip-95-223-6-41.unitymediagroup.de) |
| 21:23.29 | ArtemMakhutov | Hello |
| 21:23.55 | carrar | HARRO |
| 21:23.56 | ArtemMakhutov | Can it be possible that jitterbuffer is broken in asterisk 1.6.2.x ? |
| 21:23.58 | Katty | HARRO DER |
| 21:24.17 | carrar | Hows areyou doinz Katty! |
| 21:24.23 | Katty | jitteries |
| 21:24.27 | Katty | i had to much caffeine |
| 21:24.32 | carrar | woah |
| 21:24.33 | carrar | I miss coffee |
| 21:24.56 | Katty | no you don't |
| 21:24.58 | Katty | i feel ill |
| 21:25.01 | carrar | heh |
| 21:25.10 | Katty | we could do blood transfusion |
| 21:25.18 | Katty | we'd probably both feel better |
| 21:25.21 | carrar | Instead of coffee in the morning I have a large glass of freshly juiced carrot juice |
| 21:25.31 | carrar | It's really not the same |
| 21:25.51 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 21:26.01 | carrar | But it does wake me up crazyly enough |
| 21:26.14 | Katty | really? |
| 21:26.15 | Katty | hmmmmmm |
| 21:26.20 | Katty | how many carrots do you have to juice |
| 21:26.23 | carrar | all the natural sugars in the carrot |
| 21:26.24 | *** join/#asterisk Whitor (~Whitor@rrcs-24-97-4-148.nys.biz.rr.com) |
| 21:26.29 | carrar | 5-6 |
| 21:26.52 | carrar | People at costco think I own horses |
| 21:26.55 | carrar | heh |
| 21:27.00 | Katty | carrots are cheap tho |
| 21:27.11 | carrar | I buy 20 lbs of carrots every 2 weeks |
| 21:27.23 | Katty | how much does 20lbs of carrots cost? |
| 21:27.29 | [TK]D-Fender | checkout time, BBIAB |
| 21:27.33 | Whitor | 5$ ? |
| 21:27.42 | [TK]D-Fender | Katty: About twice as much as 10 lbs |
| 21:27.44 | carrar | about that |
| 21:27.45 | paulc | Income Tax: The reason that my $2500 bonus ends up being closer to $1300. FML. |
| 21:27.48 | carrar | per 10lb bag |
| 21:28.03 | carrar | $10 or so |
| 21:28.07 | carrar | for 20lbs |
| 21:28.18 | Katty | well you'd pay wayyyy more for that on carrot juice |
| 21:28.23 | Katty | plus carrot juice would have tons of additives |
| 21:28.27 | Katty | and obviously not be fresh |
| 21:28.27 | carrar | yeah |
| 21:28.35 | idespinner | paulc, i'd be happy to make that much per month! |
| 21:28.36 | carrar | I juice them at home |
| 21:28.44 | Katty | how does carrot juice.... taste |
| 21:28.49 | Katty | if you say carroty, i'm going to laugh |
| 21:28.53 | carrar | to me it taste fine |
| 21:28.54 | Katty | is it sweet? |
| 21:28.58 | carrar | yeah |
| 21:28.59 | Whitor | carrot juice is great |
| 21:29.03 | Whitor | it is sweet |
| 21:29.03 | Katty | hmmmmm |
| 21:29.08 | Katty | hmmmmmmm |
| 21:29.12 | carrar | it's sweeeeeeeeeeet |
| 21:29.19 | Katty | HMMmMMmmMmmMMmmm |
| 21:29.30 | carrar | wife only likes it with some apples in it also but I think it's fine plain |
| 21:29.32 | Katty | carrar: think you could find me a quality juicer from target's website or somethin? |
| 21:29.43 | Whitor | What's orange and sounds like a parrot? (this joke work best when said out loud) |
| 21:29.50 | Whitor | ... |
| 21:29.56 | carrar | katty, thats my juicer |
| 21:29.56 | Whitor | a carrot ! |
| 21:29.56 | carrar | http://pics.osburn.com/album/1119 |
| 21:30.12 | carrar | it's a good one |
| 21:30.12 | Katty | looks |
| 21:30.26 | paulc | idespinner: what do you do? |
| 21:30.28 | carrar | was on sale at Macy's 20% off plus $80 off that |
| 21:30.28 | Katty | that looks like a pain to clean |
| 21:30.36 | carrar | it's easy |
| 21:30.43 | carrar | 3 parts to clean |
| 21:30.49 | Katty | dish washer safe? |
| 21:30.50 | carrar | takes 5 mins to clean |
| 21:30.59 | carrar | i don't put it in the dish washer |
| 21:31.03 | Katty | k |
| 21:31.10 | Katty | makes note to stop by macy's this weekend |
| 21:31.11 | carrar | You could, cept the screen/cutter |
| 21:31.35 | Katty | on a side note |
| 21:31.39 | Katty | do you think it could juice chicken? |
| 21:31.43 | Katty | raw chicken |
| 21:31.44 | carrar | yes |
| 21:31.46 | carrar | yes |
| 21:31.49 | Katty | excellent |
| 21:31.53 | carrar | and beef |
| 21:31.59 | Katty | well the ferrets don't like raw beef |
| 21:32.03 | Katty | but they do like raw chicken |
| 21:32.07 | Katty | mushy |
| 21:32.13 | carrar | though that would be weird |
| 21:32.20 | Katty | why? |
| 21:32.21 | carrar | I'd like to see pics of that being juiced |
| 21:32.25 | Katty | k |
| 21:32.36 | carrar | raw chicken juice |
| 21:32.49 | carrar | sounds salmanily |
| 21:32.52 | Katty | ferrets will be all OMNOMNOMNOMLIQUIFIEDCHICKENYGOODNESSOMNOMNOMNOM |
| 21:33.00 | Katty | and have it all over their whiskers |
| 21:33.16 | carrar | I bet it could even juice a iPhone |
| 21:33.18 | idespinner | paulc, just an engineer doing voip work.... |
| 21:33.22 | Katty | carrar: hot. |
| 21:33.59 | paulc | idespinner: I'd rather be doing that than what I'm doing right now.. but it begs the question (which British people have no problems asking, but a lot of the world don't like answering) "What's the going rate for that these days?" |
| 21:34.24 | Katty | paulc: two dollah |
| 21:34.43 | paulc | no five dollah? love you loooong time? |
| 21:35.14 | Katty | we don't do that high class stuff here |
| 21:35.21 | *** join/#asterisk briancline (~brian@unaffiliated/briancline) |
| 21:35.58 | idespinner | paulc, probably alot more... |
| 21:37.35 | Katty | http://www.gfxferret.com/images/stuff/hangers2.jpg <- waaaaaaaaaannnnnnnnnnttttt |
| 21:38.07 | adnc_ | hi Katty |
| 21:38.15 | Katty | i need to buy about 3 of those playpens for the boys when they play outside this summer |
| 21:38.22 | Katty | adnc_: ohaider |
| 21:38.34 | adnc_ | hello, does someone know how this error needs to be interpreted? asterisk[3623]: rc_avpair_new: unknown attribute 1490026597 |
| 21:38.40 | beek | hugs katty |
| 21:38.46 | Katty | hugs beek |
| 21:38.56 | adnc_ | ;) |
| 21:39.07 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:39.10 | andres833 | hi all |
| 21:39.18 | andres833 | i need help! |
| 21:40.11 | andres833 | I load the iLBC codec in asterisk 1.6.0.13 |
| 21:40.12 | paulc | andres833: ask away! |
| 21:40.13 | andres833 | I load the iLBC codec in asterisk 1.6.0.13 |
| 21:40.36 | paulc | runs away from low bandwidth codecs |
| 21:41.00 | andres833 | paulc, i try load but in voipinfo the solution don't load :( |
| 21:41.27 | andres833 | paulc, why? |
| 21:41.53 | andres833 | paulc, i need use this codec for a siptrunk |
| 21:42.17 | paulc | andres833: nothing personal - I just prefer my G.711 ulaw for quality :) |
| 21:42.20 | idespinner | i havent heard of anybody who uses iLBC.... |
| 21:42.21 | andres833 | paulc, if i use other coder the comunication sucks :( |
| 21:42.40 | paulc | what sort of bitrate/compression is iLBC - I'm a bit rusty on that stuff |
| 21:43.28 | andres833 | paulc, wait a moment |
| 21:43.45 | Katty | thinks they need to make ferret Runs as well as playpens |
| 21:44.35 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 21:45.27 | andres833 | paulc, look this page www.ilbcfreeware.org |
| 21:45.31 | idespinner | dont let Lief see this link but: http://www.voip-info.org/wiki/view/iLBC |
| 21:46.28 | roe | surprisingly, I seem to be having trouble locating a feature comparison between the 1.4 branch and the 1.6 branch. I am trying to determine which version to install |
| 21:46.53 | andres833 | idespinner, thanks but i see and try follow this steps but no run :( |
| 21:48.30 | paulc | andres833: ok, I'm a bit more in the know now... but still wonder why? not enough bandwidth for G.711? |
| 21:48.34 | Katty | http://www.farmdogs.net/picts7/show06b-5.JPG <- they're doin it right |
| 21:49.01 | Katty | http://contribute.chron.com/ver1.0/Content/images/store/8/15/d8ce06bc-65c6-411d-af79-0bc6ca4e7c19.Full.jpg <- they're also doin it right |
| 21:49.18 | Whitor | Is anyone familiar with PRI to ethernet to PRI tunnels? I'm looking to tie our legacy PBX's over an ethernet link ... |
| 21:49.26 | QubeZ | hello all |
| 21:49.41 | Katty | http://www.drsfostersmith.com/images/Categoryimages/normal/p_25461_34337P_009.jpg <- NOT doin it right |
| 21:49.46 | Katty | ^- NOT a playpen |
| 21:49.48 | idespinner | Whitor, you mean tdmoe? |
| 21:50.02 | Whitor | idespinner, I don't know what tdmoe is |
| 21:50.09 | *** join/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 21:50.25 | idespinner | Whitor, how are you doing pri over ethernet? |
| 21:50.36 | QubeZ | Asterisk runs fine in VMWare ESXi environment right? I have a box with ESXi and just put latest * on it. Dual 3.0 Ghz w/ 4G of ram of which 1G is dedicated to *. I ran dahdi_test and it producing horrible numbers: < 99.800% |
| 21:50.41 | Whitor | right now our pbx's talk together over a T1 going through a csu/DSU at both ends :) |
| 21:50.55 | Whitor | idespinner, I'm not yet ... that is what I'm exploring |
| 21:51.00 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:51.37 | *** part/#asterisk theBruno (~ChrisBrun@casanueva.wifi.frognet.net) |
| 21:52.30 | idespinner | Whitor, have you looked at any manufacturers yet? |
| 21:55.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:57.40 | Whitor | idespinner, not really... didn't even know if it was possible... just feeling things out at this point |
| 21:57.54 | Whitor | RAD IT mux looks like it might do what I want though |
| 21:58.36 | Whitor | so yeah... I guess I have sorta looked around |
| 21:59.17 | idespinner | Whitor, redfone is a generally well respected manufacturer of tdmoe products. Ive met the guys at astricon a while back. |
| 21:59.38 | idespinner | i dont have any personal experience with their products however. |
| 22:00.09 | Whitor | idespinner, thanks a lot for the company names! I'll check them out. |
| 22:00.51 | Whitor | ty for the disclaimer :) |
| 22:03.15 | [TK]D-Fender | Redfone.. BLEH |
| 22:03.15 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:03.25 | [TK]D-Fender | Noone uses TDMoE... |
| 22:05.34 | jaytee | Redfone does! |
| 22:06.35 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 22:06.44 | jaytee | bbiab |
| 22:08.11 | adnc_ | noone any ideas what this error (in syslog) could have caused? rc_avpair_new: unknown attribute 1490026597 |
| 22:08.18 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 22:08.52 | carrar | If only there was a place you could search for stuff |
| 22:10.31 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:12.57 | *** part/#asterisk lanning (~lanning@208.87.235.224) |
| 22:13.28 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 22:14.18 | ariel_ | does anyone knpw how to get IAX to send packets in 60ms instead of 20ms? |
| 22:14.19 | *** join/#asterisk Neo31 (~Neo31@unaffiliated/neo31) |
| 22:14.44 | roe | is there a configuration 'tester' that will evaluate a config based on support in a given release? |
| 22:17.13 | carrar | ariel, you mean your codec's RTP packet size? |
| 22:17.19 | [TK]D-Fender | roe: no |
| 22:17.19 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 22:17.52 | idespinner | ariel_, the closest I have seen is the last 3 lines of "general NAT issues" : http://www.voip-info.org/wiki/view/IAX |
| 22:18.02 | roe | so the best way to see if an upgrade from 1.4 to 1.6 will work smoothly with the current configuration is to just try it? |
| 22:18.19 | idespinner | changing the iax2 timing may be a similar ordeal |
| 22:18.39 | idespinner | roe, yep |
| 22:18.49 | roe | 10-4 |
| 22:18.49 | *** join/#asterisk Whitor (~Whitor@rrcs-24-97-4-148.nys.biz.rr.com) |
| 22:18.55 | idespinner | just keep in mind the differences between 1.4 and 1.6 dialplans |
| 22:19.09 | idespinner | where you use , nstead of | |
| 22:19.39 | roe | will do |
| 22:21.38 | doubletoker | anyone here use ipcomms with asterisk 1.6 |
| 22:21.51 | doubletoker | having the hardest time setting this up |
| 22:22.28 | doubletoker | keeps dropping the rtp, no voice after like 10 seconds or so, but my cell is still on the line |
| 22:22.41 | *** join/#asterisk Reemo (~Reemo@p5B3EFEB0.dip.t-dialin.net) |
| 22:23.16 | Reemo | Hey guys, is there a way to implement a h-Extension on a specific exten? |
| 22:24.00 | doubletoker | put that extension in a different context |
| 22:24.01 | p3nguin_ | h is the specific extension. |
| 22:24.39 | Reemo | again i have troubles describing my problem :) hey p3nguin_ btw |
| 22:24.40 | doubletoker | like you mean when they transfer to a different ivr? |
| 22:24.43 | ariel_ | idespinner: what timers? |
| 22:25.18 | Reemo | Well, say i have the exten $foo. $foo is answered by a Voicemail, and i want to send a Jabbermessage depending on wether the caller left a message on this specific exten & voicemail |
| 22:25.44 | Reemo | Currently i have to implement the h-extension for the full context |
| 22:26.05 | p3nguin_ | Okay, doubletoker gave you a good suggestion, then. |
| 22:26.15 | doubletoker | anyone know how to fix the error [Feb 24 17:20:57] WARNING[13133]: chan_sip.c:3550 retrans_pkt: Maximum retries exceeded on transmission 1705885-3476038837-592939@msx71.mydomain.com for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 22:26.15 | doubletoker | [Feb 24 17:20:57] WARNING[13133]: chan_sip.c:3577 retrans_pkt: Hanging up call 1705885-3476038837-592939@msx71.mydomain.com - no reply to our critical packet (see doc/sip-retransmit.txt). |
| 22:26.24 | p3nguin_ | Just break the exten off into its own context and give it its own h exten. |
| 22:27.39 | Reemo | no other way? i have the feeling that many things in the dialplan have to be solved rather complicated (I'm a programmer, so I miss a more structured language) |
| 22:28.35 | ManxPower-work | doubletoker: 1) that is commonly the user hanging up the phone 2) it could be a NAT or firewall problem if it happens all the time and calls drop |
| 22:29.30 | *** join/#asterisk t0rrieri (~Torrieri@nelug/crew/torrieri) |
| 22:30.11 | ManxPower-work | Reemo: Dialplans ARE complicated. The more you work to make them uncomplicated the less they will work. |
| 22:31.07 | Reemo | ManxPower-work, it shouldnt be like this |
| 22:31.30 | beek | Reemo: What's wrong with the dialplan BASIC-like syntax? |
| 22:31.47 | leifmadsen | Reemo: for a specific extension? you'd likely have to put the extension inside its own context then include it within another context |
| 22:32.20 | ManxPower-work | Reemo: I don't know your dialplan, but some dialplans can be a thousand lines or more. A typical FreePBX dialplan (which in my opinion is overly complicated): -= 718 extensions (2465 priorities) in 103 contexts. =- |
| 22:32.21 | p3nguin_ | Does the dialplan actually have a named language? |
| 22:32.52 | ManxPower-work | p3nguin_: "extensions.conf" |
| 22:33.01 | p3nguin_ | right |
| 22:33.19 | carrar | the language of love |
| 22:33.22 | leifmadsen | http://pastebin.ca/1808645 |
| 22:33.26 | leifmadsen | p3nguin_: "dialplan" is the name |
| 22:33.30 | p3nguin_ | is it considered to be any LANGUAGE other than plain? |
| 22:33.32 | Reemo | beek, basically everything. Something that has to be as dynamic as a dialplan shouldn't be limited by the language it is written in |
| 22:33.40 | leifmadsen | Reemo: http://pastebin.ca/1808645 <-- something like that might work (with tweaks, totally untested) |
| 22:33.53 | leifmadsen | p3nguin_: it's considered a scripting language I suppose |
| 22:34.01 | p3nguin_ | Okay, great. Which one? |
| 22:34.08 | ManxPower-work | This is a production FreePBX: -= 1239 extensions (5032 priorities) in 156 contexts. =- |
| 22:34.09 | leifmadsen | p3nguin_: it's unique to asterisk |
| 22:34.14 | carrar | extension conf script language |
| 22:34.23 | carrar | ECSL |
| 22:34.27 | beek | Reemo: I was being sarcastic. |
| 22:34.29 | p3nguin_ | ;) |
| 22:34.30 | ManxPower-work | I usually use AEL these days |
| 22:34.36 | Reemo | beek, sry, didnt get it |
| 22:35.02 | doubletoker | yea, my firewall, is setup for accept all |
| 22:35.15 | ManxPower-work | doubletoker: firewall or nat? |
| 22:35.20 | doubletoker | firewall |
| 22:35.35 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.especially.bridget.silverelitez.org) |
| 22:35.37 | ManxPower-work | doubletoker: do a sip debug and put it on pastebin.ca |
| 22:35.44 | doubletoker | the nat only forwards ports 5060, 11000 - 11100 |
| 22:35.58 | doubletoker | alright |
| 22:36.09 | p3nguin_ | is that range the range that you configured in rtp.conf? |
| 22:36.16 | ManxPower-work | doubletoker: I take it you have localnet/externip, rtp.conf setup? |
| 22:36.16 | doubletoker | yes |
| 22:36.30 | doubletoker | yes I do |
| 22:36.40 | Reemo | thanks leifmadsen . It seems like your solution would be the cleanest |
| 22:36.52 | ManxPower-work | we will know if you didn't when we see your sip debug info. |
| 22:36.53 | doubletoker | let me pastebin it |
| 22:37.06 | doubletoker | ok |
| 22:37.09 | doubletoker | lets find out |
| 22:38.14 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 22:39.21 | Reemo | I would love to see some language features like return values and the ability of controlling when a exten should really return. |
| 22:39.41 | ManxPower-work | Reemo: Let us know when you code it. |
| 22:40.01 | ManxPower-work | First you'll have to redesign the Asterisk core to support that sort of thing. |
| 22:40.11 | [TK]D-Fender | And every app |
| 22:40.11 | Reemo | The Voicemail issue is very disturbing. If the caller presses the # key $VMSTATUS is filled and the next Line is executed. But if he just hangs up it doesn't. |
| 22:40.35 | ManxPower-work | Reemo: What is the problem with that? |
| 22:40.46 | Reemo | see above. |
| 22:41.07 | ManxPower-work | Reemo: Oh, that's simple.; |
| 22:41.19 | ManxPower-work | see externotify (or something very similar) in voicemail.conf |
| 22:42.04 | ManxPower-work | Reemo: You can either fight Asterisk's oddities and life a miserable life or you can embrace Asterisk's oddities and be happy. |
| 22:42.16 | ManxPower-work | live, of course. |
| 22:42.26 | ManxPower-work | Most of us stopped fighting Asterisk *years* ago. |
| 22:42.35 | doubletoker | http://pastebin.ca/1808661 |
| 22:43.31 | ManxPower-work | that looks like you are calling yourself. |
| 22:43.44 | ManxPower-work | doubletoker: pastebin your sip.conf [general] section, masking ONLY passwords |
| 22:43.46 | doubletoker | that's a phone call coming in |
| 22:44.01 | *** join/#asterisk Alagar (~Administr@122.164.40.142) |
| 22:44.23 | ManxPower-work | the from and the to are the same. |
| 22:44.38 | ManxPower-work | doubletoker: do you have canreinvite=no? |
| 22:45.35 | doubletoker | http://pastebin.ca/1808664 |
| 22:45.40 | doubletoker | no I don't |
| 22:46.23 | doubletoker | been looking this up for hours finally figured I needed some help |
| 22:46.25 | ManxPower-work | canreinvite is NOT an optional setting. |
| 22:46.29 | [TK]D-Fender | doubletoker: Contact: <sip:4015424728@192.168.1.4> <-- clearly you have NOT set your system up properly to work behind NAT |
| 22:46.32 | [TK]D-Fender | ~sipnat |
| 22:46.33 | 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 |
| 22:46.36 | ManxPower-work | ~answers |
| 22:46.37 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 22:46.38 | [TK]D-Fender | doubletoker: READ ^^^ |
| 22:46.49 | ManxPower-work | [TK]D-Fender: it looks right to me |
| 22:47.11 | doubletoker | that is twinkle |
| 22:47.20 | [TK]D-Fender | ManxPower-work: Not when the IP is being presented to an outside address. |
| 22:47.23 | ManxPower-work | doubletoker: also you need to set allowguest=no |
| 22:47.47 | [TK]D-Fender | Lines 285 & 297 |
| 22:47.48 | [TK]D-Fender | BAD |
| 22:47.51 | doubletoker | is that in general |
| 22:48.10 | ManxPower-work | doubletoker: yes. |
| 22:48.17 | ManxPower-work | [TK]D-Fender can help you now. |
| 22:48.29 | ManxPower-work | you should also upgrade your Asterisk |
| 22:50.41 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
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| 23:04.08 | doubletoker | alright |
| 23:04.23 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 23:06.34 | *** join/#asterisk cguerrero (~cuauhtemo@200.79.231.94) |
| 23:09.00 | idespinner | net splits.... wierd |
| 23:09.42 | *** join/#asterisk errr (~errr@fedora/errr) |
| 23:10.12 | errr | where do you find the life span of asterisk releases? Like how much longer each will be supported for |
| 23:10.28 | leifmadsen | errr: www.asterisk.org/asterisk-versions |
| 23:10.32 | errr | thanks |
| 23:10.54 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 23:11.02 | idespinner | any of you familiar with CDR issues and asterisk AMI originate? |
| 23:11.50 | errr | what app are most people going with for Text to Speech on asterisk 1.2.x? |
| 23:12.02 | idespinner | festival if you need free... |
| 23:12.11 | idespinner | cepestral if you have $20 |
| 23:12.21 | errr | no, Im happy paying for cepstral, but I am having trouble finding app-swift |
| 23:12.25 | idespinner | (i like cepestral, it sounds pretty descent) |
| 23:12.43 | errr | it seems who ever used to maintain it has vanished.. |
| 23:12.48 | idespinner | as far as the source? or in your version? |
| 23:13.16 | errr | well either.. I found a link for the 1.6 version but I didnt think that would work at all for 1.2 due to the massive changes |
| 23:13.33 | paulc | I like Cepstral with Allison's voice - we used the high quality version recently to record city names for our IVR app (downsampled to 8KHz for our IVRs) |
| 23:13.34 | idespinner | its hit or miss, with probably more misses... |
| 23:13.49 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 23:13.57 | idespinner | paulc, yea, thats exactly were doing for an application |
| 23:14.09 | idespinner | its suprisingly good for short sentences |
| 23:14.26 | errr | can I use system to have swift output the file and then get asterisk to play that file with playback? |
| 23:14.49 | errr | or is app-swift the way to go? |
| 23:15.52 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:16.13 | paulc | idespinner: yeah - we did "That caller dialed the local number in.." then recorded 15,000 city names (most of which sound decidedly wonky) |
| 23:19.16 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 23:19.50 | Micc | what does exception flag set on '<channel>' but no exception handler mean? |
| 23:20.09 | Micc | and why do I get channels with <ZOMBIE> on the end? |
| 23:21.03 | Corydon76-dig | It means you're running an outdated version of Asterisk |
| 23:21.33 | Corydon76-dig | Those zombies should only continue to exist for a fraction of a second |
| 23:23.27 | ManxPower-work | Corydon76-dig: could they be caused by AGIs that don't catch SIGHUP? (or whichever signal Asterisk sends when the channel hangs up on an AGI) |
| 23:23.53 | Corydon76-dig | ManxPower-work: channels, not processes |
| 23:24.17 | ManxPower-work | Corydon76-dig: I was thinking the channel would hang around until AGI existed. |
| 23:24.20 | Corydon76-dig | Zombies are the leftovers when a masquerade occurs |
| 23:24.27 | Corydon76-dig | Nope |
| 23:24.41 | Corydon76-dig | So, transfers... |
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| 23:31.52 | Micc | I see zombies when parking calls. Is that normal? |
| 23:33.19 | Corydon76-dig | Yep |
| 23:34.14 | Deeewayne | Katty, ferrets are waking |
| 23:40.20 | doubletoker | what is canreinvite in? |
| 23:40.40 | doubletoker | sip.conf?? cause I don't see in any sample file |
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| 23:47.30 | ManxPower-work | look in /path/to/src/asterisk/configs/sip.conf.sample |
| 23:48.13 | ManxPower-work | doubletoker: have you read the sip nat stuff |
| 23:48.17 | ManxPower-work | ~answers |
| 23:48.18 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 23:50.20 | doubletoker | yea, but my configuration is different |
| 23:50.46 | doubletoker | internet -> nat -> asterisk - phone |
| 23:51.24 | doubletoker | I'm really using that same machine, my phone connects to 127.0.0.1 |
| 23:52.08 | p3nguin_ | If nat is between your asterisk system and the internet, you need to configure nat settings. |
| 23:52.23 | p3nguin_ | nat, localnet, and externip or externhost |
| 23:52.33 | p3nguin_ | all in the general section of sip.conf. |
| 23:52.34 | [TK]D-Fender | Basically EVERYTHING IN THE GUIDE |
| 23:52.41 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 23:53.11 | ManxPower-work | [TK]D-Fender: I'd love to order a ClueBat (tm) made from that Nerf foam material. |
| 23:53.41 | [TK]D-Fender | ManxPower-work: @#%^ that! Rock maple or Ionwood. Accept no substitutes! |
| 23:53.45 | [TK]D-Fender | Ironwood* |
| 23:54.15 | ManxPower-work | Bah, just belittle 'em until they cry. |
| 23:54.22 | doubletoker | ok |
| 23:54.30 | doubletoker | made the changes let me check it out |
| 23:54.31 | jaytee | The ClueBat(tm) must have 3" spikes at the top |
| 23:54.44 | [TK]D-Fender | <ivandrago> I will break you... |
| 23:54.56 | [TK]D-Fender | jaytee: Just a rusty nail.. |
| 23:55.05 | jaytee | I was thinking rusty spikes |
| 23:55.11 | jaytee | tetanus is fun! |
| 23:55.27 | p3nguin_ | Yeah it is. I used to play that on GameBoy all the time. |
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| 23:55.38 | jaytee | hehe |
| 23:59.36 | hardwire | tetanus eh? |
| 23:59.44 | hardwire | what a weird word to read. |
