| 00:00.14 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 00:01.02 | *** part/#asterisk gyoo (~gyo@201-42-164-218.dsl.telesp.net.br) |
| 00:22.19 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 00:24.34 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 00:24.54 | k-man | for some reason asterisk runs at 33% cpu even though no calls are taking place |
| 00:25.03 | k-man | is that normal? |
| 00:27.36 | TJNII | No. |
| 00:27.57 | k-man | what should i do to try and diagnose it? |
| 00:29.01 | TJNII | It is normal for asteriak to use some CPU, but it only comes out to under 1% on my P3. You're not running it on a router or something, are you? |
| 00:29.31 | k-man | no, its a full computer |
| 00:29.53 | k-man | intel Pentium D 3.4GHz |
| 00:30.03 | TJNII | Yea, it shouldn't burn CPU like that. |
| 00:30.13 | TJNII | There are som eknown issues, but I can't remember what they are. |
| 00:30.41 | k-man | google didn't give me much to go on |
| 00:35.35 | k-man | this seems interesting |
| 00:35.37 | k-man | https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17217 |
| 00:40.45 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:41.22 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 00:45.07 | TJNII | Is the -c for color? I vaguely remember something about the init script, color, and CPU usage. |
| 00:45.26 | k-man | well, my init script doesn't have -c in it |
| 00:45.33 | k-man | so...that's not my issue |
| 00:45.37 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:45.38 | *** join/#asterisk Baylink (~jra@205.201.151.25) |
| 00:47.44 | k-man | ah... i think i have found part of the problem |
| 00:47.51 | k-man | my music on hold stuff is broken |
| 00:48.29 | *** join/#asterisk jhirley (~jhirley@adsl-3-96-254.mia.bellsouth.net) |
| 00:48.51 | jhirley | o/ |
| 00:49.17 | TJNII | k-man: That could do it. |
| 00:49.20 | TJNII | Straming? |
| 00:49.31 | TJNII | s/Straming/Streaming/ |
| 00:50.16 | k-man | it was unable to find a couple of my moh directories |
| 00:50.28 | k-man | forgot to set them up after reinstalling and copying my old configs accross |
| 00:51.57 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 00:52.39 | k-man | hmm.. well that seems a lot better so far |
| 00:53.04 | k-man | seems as though asterisk, rightly or wrongly, if it fails to find the moh directory, it just keeps trying indefinately |
| 00:56.53 | k-man | well, that was it, asterisk now runs at less than 1%cpu |
| 00:57.19 | k-man | thanks for your support guys :) |
| 01:12.34 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 01:22.31 | *** join/#asterisk `paul (~kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 01:25.29 | `paul | what function should i use to record a custom message from a user? what i want to do is record a message thru phone on create a page where he presses a button and the script automatically calls phone numbers which the user sets and plays the recording. Im already done with the script that calls and plays the message. I just need an idea on how to go about recording a custom message. help please. :) |
| 01:29.24 | carrar | record()? |
| 01:29.38 | carrar | or is this a trick question |
| 01:32.38 | TJNII | Your standard phone keypad has 0 and 1 buttons. Should be able to get any data you want into your PBX through them, as long as you format it properly. |
| 01:33.14 | `paul | carrar: so record would be the best? how about "press 1 to play the message again... etc" ill be the one coding the logic for that? |
| 01:33.29 | TJNII | Simple dialplan logic. |
| 01:35.51 | jblack | tjii: Yeah? How about the precise value for pi? :) |
| 01:36.43 | TJNII | jblack: You'd probably want to use little endian octal for that. To save on keystrokes. |
| 01:37.20 | jblack | heh |
| 01:37.35 | TJNII | I'm half temped to write up some DTMF data transfer monstrosity now/ |
| 01:39.09 | jblack | The problem isn't the code. It's finding the fool that's willing to enter a non-terminating, non-repeating string of numbers. |
| 01:45.10 | `paul | has anybody here used Cepstral text to speech? will it allow to record the audio to a file instead of directly playing it? |
| 01:46.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:46.22 | TJNII | carrar: I guess it was a trick question. |
| 01:55.52 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-219-139.mia.bellsouth.net) |
| 01:56.34 | TJNII | I turned off the ceiling fan and my mp3 player died. |
| 01:56.38 | TJNII | Amazing timing.... |
| 02:08.16 | *** part/#asterisk kazaa_lite (~eddie@212.183.140.52) |
| 02:10.03 | carrar | hahah TJNII yeah |
| 02:10.37 | carrar | everything in here is a trick question or has two parts |
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| 02:26.50 | *** join/#asterisk skrite (~skrite@69.55.26.25) |
| 02:36.40 | skrite | hey all |
| 02:39.12 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 02:42.00 | `paul | has anybody here used Cepstral text to speech? will it allow to record the audio to a file instead of directly |
| 02:47.37 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 03:37.19 | *** join/#asterisk rjk` (rjk@home.melb.org) |
| 03:37.25 | rjk` | is there asterisk for mac with skype integration ? |
| 03:37.47 | ChannelZ | I'd say no. |
| 03:37.58 | carrar | Build it |
| 03:38.14 | carrar | .... and they will come! |
| 03:38.24 | rjk` | shame |
| 03:38.37 | rjk` | i could really use some kind of skype ivr on my mac |
| 03:39.34 | TJNII | You might be able to compile it. What with mac baing based on BSD nowadays and all. |
| 03:39.38 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 03:39.40 | spenguin[work] | test |
| 03:39.52 | ChannelZ | and use Skype for SIP I suppose |
| 03:40.24 | TJNII | Is the Skype for Asterisk module closed source? |
| 03:40.32 | ChannelZ | yes |
| 03:40.35 | TJNII | Actually I suppose it would have to be, wouldn't it..... |
| 03:40.45 | spenguin[work] | try siptosys |
| 03:41.00 | spenguin[work] | www.mhspot.com/sts/siptosis.html |
| 03:41.05 | TJNII | Amazing what conclusions you can come up with when you catually think something through. |
| 03:41.14 | rjk` | looking now |
| 03:41.57 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 03:42.07 | rjk` | urgh its java |
| 03:42.54 | spenguin[work] | rjk`: yeah urg |
| 03:43.00 | spenguin[work] | but its free |
| 03:43.14 | spenguin[work] | and you can use the normal skype client |
| 03:43.20 | *** join/#asterisk zzlane (~zzlane@c-67-175-46-8.hsd1.il.comcast.net) |
| 03:43.23 | spenguin[work] | with no need for the BCP etc |
| 03:43.49 | rjk` | i dont mind paying for a solution but i want it mac os x native |
| 03:45.47 | rjk` | wikipedia says asterisk is also for os x now, im yet to see proof |
| 03:47.54 | spenguin[work] | why would you want to run asterisk on the Mac? |
| 03:48.23 | rjk` | for the ivr features with my skype in numbers |
| 03:48.40 | spenguin[work] | I mean run asterisk ona linux box where itll work best |
| 03:52.39 | Nugget | asterisk compiles fine in os x, but you'd be stupid to use it in production. |
| 03:54.02 | Nugget | I used to run asterisk on my macbook so I could tunnel voip more reliably through the multi-level nat hell you sometimes find in hotels. |
| 03:54.38 | Nugget | I'd connect my softphone to the local asterisk and then trunk iax to my office asterisk server and that worked a lot better than trying to do sip through nat |
| 03:55.18 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 03:56.03 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 03:56.50 | *** join/#asterisk `paul (~kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 04:02.27 | rjk` | i also need voice broadcasting on my mac , to cater for members of our blind community association for meeting notifications, cancelled events etc |
| 04:02.36 | rjk` | theres lots for windows nothing for mac |
| 04:02.44 | rjk` | so asterisk might be the best way to go |
| 04:03.46 | Nugget | I don't know what voice broadcasting is, so I don't have an opinion on that. |
| 04:03.46 | [TK]D-Fender | rjk`: Clarify "voice broadcast" |
| 04:03.55 | [TK]D-Fender | high-5's Nugget |
| 04:03.56 | Nugget | asterisk will compile in os x, though. that much I do know |
| 04:03.57 | [TK]D-Fender | ... |
| 04:04.00 | [TK]D-Fender | telnet |
| 04:04.00 | Nugget | telnet is eeeeeeevil! |
| 04:04.04 | Nugget | pfft |
| 04:04.07 | [TK]D-Fender | pushes some more buttons |
| 04:04.11 | [TK]D-Fender | muahahahah! |
| 04:05.32 | rjk` | [TK]D-Fender: i do the computer support for the victorian blind social network inc. we have 890 or so members, two volunteers in the office, if events are cancelled it would be good to auto call them and play them a message |
| 04:06.37 | rjk` | i might just have to go for a windows solution for that and run parallels |
| 04:07.13 | rjk` | on the linux box idea, i think thats a good idea and the way i will go |
| 04:07.13 | TJNII | Man up and run Linux. |
| 04:07.16 | Nugget | obviously you know your needs better than we do, but I just can't see how that makes sense. |
| 04:07.30 | TJNII | Don't try and pussyfoot by running some craop on Windows. |
| 04:07.31 | Nugget | for the price of windows and parallels you could buy a whole computer and run linux on it. |
| 04:07.50 | Nugget | I hate linux just as much as the next guy, but really it's the path of least resistance when it comes to asterisk |
| 04:08.00 | Nugget | and I mean I hate it. I hate hate hate linux. |
| 04:08.04 | Nugget | but I use it for asterisk |
| 04:08.16 | TJNII | I love linux. Love love love love love. |
| 04:08.22 | Nugget | it's way less awful than running asterisk on os x. |
| 04:08.32 | [TK]D-Fender | rjk`: Sad. FFS just use VirtualBox and install * on Linux :p |
| 04:08.45 | rjk` | i run slackware in vm, i own LINUX personalised number plates, i am not oppoed to linux but i would like a mac solution that is native |
| 04:09.03 | Nugget | asterisk will run native on os x. but you should use linux instead. |
| 04:09.09 | rjk` | btw if you are in vic australia my plates are for sale |
| 04:09.14 | [TK]D-Fender | rjk`: Don't knwo of any Skype connectors that will run on OSX |
| 04:09.24 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 04:09.35 | TJNII | And the conversation goes full circle. |
| 04:09.46 | rjk` | [TK]D-Fender: there is a sip/skype connector in beta on skypes site |
| 04:10.22 | [TK]D-Fender | rjk`: Beta radiation WILL kill you.... |
| 04:10.32 | rjk` | lol |
| 04:10.37 | TJNII | Ohhh. That's good. |
| 04:11.07 | rjk` | the reason for the mac thing is that our volunteers are blind and the mac is best for them |
| 04:11.09 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-xthaflyeiqrlrpmv) |
| 04:11.16 | rjk` | i cant be there all the time |
| 04:11.20 | Nugget | your volunteers will not be adminning the asterisk server. |
| 04:11.28 | Nugget | the platform is irrelevant |
| 04:11.30 | rjk` | no i will remotely |
| 04:11.46 | rjk` | but setting up outbound auto calls they can do |
| 04:12.09 | Nugget | that will be the same no matter what os you're using. |
| 04:12.15 | Nugget | how hard or easy it is will be up to you |
| 04:12.21 | rjk` | true |
| 04:12.40 | rjk` | im thinking out loud and hoping someone can tell me something google hasnt |
| 04:12.49 | rjk` | sorry to be a pain |
| 04:12.50 | Nugget | we have. are you listening |
| 04:12.51 | Nugget | ? |
| 04:13.16 | rjk` | yes i am listening |
| 04:17.27 | rjk` | ok i think i have found a box for linux, does anyone know of any auto dialling outbound apps for asterisk ? |
| 04:17.52 | rjk` | i rang one of our computer volunteers and he has a P4 he can give us |
| 04:21.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:24.58 | [TK]D-Fender | rjk`: Fed by what? What will your call do? |
| 04:27.24 | rjk` | a volunteer will record a message saying that an event has been cancelled or that a social night has changed, or news about the braile library etc, then the system will call the batch of numbers to the people and play them the message |
| 04:27.59 | rjk` | we were using a DECTALK speech synthesis system until it died two weeks ago |
| 04:28.02 | rjk` | it had 8 lines |
| 04:28.15 | rjk` | now im tasked with replacing the tech |
| 04:28.24 | rjk` | and im not good at this stuff anymore |
| 04:28.44 | coppice | the smithsonian might like to have that :-) |
| 04:28.46 | rjk` | im a 43 year old telecoms tech who last worked in telecoms 17 years ago |
| 04:29.09 | rjk` | dectalk was great |
| 04:29.18 | rjk` | unfortunately there isnt even DEC anymore |
| 04:29.27 | rjk` | we cant get it fixed |
| 04:29.44 | rjk` | my idea was skype |
| 04:30.16 | rjk` | somehow automating it all through skype with asterisk which i understand has ivr capability |
| 04:30.56 | TJNII | What you want to do is completely workable with asterisk. I recommend just using a ITSP instead of Skype, though. |
| 04:31.11 | [TK]D-Fender | rjk`: So you planned on using Skype to phone real phone #'s? |
| 04:31.12 | rjk` | ITSP ? |
| 04:31.16 | [TK]D-Fender | ~itsp |
| 04:31.17 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:31.25 | rjk` | yes real numbers |
| 04:31.33 | rjk` | everyone who is being called is blind you see |
| 04:31.39 | [TK]D-Fender | rjk`: Then your idea of using Skype is broken. Just get a real ITSP |
| 04:31.42 | rjk` | most dont even have computers |
| 04:31.50 | [TK]D-Fender | rjk`: And then you can certainly run * on MacOS native |
| 04:32.10 | rjk` | ITSP being ? |
| 04:32.17 | [TK]D-Fender | rjk`: Probelm started when you seemed fixed on an idea without the premise for its selection |
| 04:32.26 | coppice | rjk` were you using the original hardware card in a PDP11 version of DECTALK? |
| 04:32.29 | [TK]D-Fender | [00:31]<infobot>[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:32.35 | [TK]D-Fender | rjk`: ^^ HELLO |
| 04:32.56 | rjk` | coppice: no it was a 8 card dectalk phone rack |
| 04:33.07 | rjk` | 8 lines connected to it |
| 04:33.14 | rjk` | a dos computer connected to it |
| 04:33.34 | coppice | oh, something quite new |
| 04:33.37 | rjk` | you typed the message, then selected the numbers from the database, and it rang and spoke to each number |
| 04:33.48 | rjk` | it wasnt new, it was 20 years old |
| 04:34.07 | coppice | compare to the PDP11 I used to use that is new |
| 04:34.53 | rjk` | coppice: http://old.rjk.melb.org/notpublic/rob2002.jpg |
| 04:35.04 | rjk` | thats me next to my old data general nova :) |
| 04:35.23 | rjk` | i used to collect big iron |
| 04:35.24 | rjk` | lol |
| 04:36.14 | rjk` | [TK]D-Fender: i've been googling a couple, doesn't seem that good for australia |
| 04:36.58 | [TK]D-Fender | rjk`: Well if you've got your phone lines, get an FXO gateway then |
| 04:37.13 | [TK]D-Fender | rjk`: Though I know you've got plenty of options either way |
| 04:37.26 | rjk` | i have an old dialogic card, but i can't find anything to run it |
| 04:37.33 | rjk` | it's ISA |
| 04:37.39 | rjk` | 4 port |
| 04:38.32 | [TK]D-Fender | rjk`: Invest in hardware or get an itsp. Your choice. Given the temporary nature of your call-outs I suspect an ITSP would be far cheaper |
| 04:39.34 | rjk` | yes but convincing this non profit committee to buy milk for the fridge is an effort lol |
| 04:40.05 | [TK]D-Fender | rjk`: Fine tell them. Oh well guess you don't need what that old box was doing for you. |
| 04:40.31 | rjk` | im going to accept this pc donation of the P4 and put ubuntu on it then asterisk then have a play i think |
| 04:40.33 | dzup | what kind of fridge is that? |
| 04:40.45 | rjk` | dzup ? |
| 04:40.57 | dzup | heh |
| 04:40.57 | rjk` | the photo ? |
| 04:41.02 | rjk` | lol |
| 04:41.02 | dzup | yeah |
| 04:41.13 | rjk` | its a data general nova 3 |
| 04:41.20 | rjk` | 16 bit computer |
| 04:41.31 | rjk` | competed with the pdp 11 |
| 04:41.39 | dzup | how many bulbs ? 4? and the rest is metal ha? |
| 04:41.44 | rjk` | lots of nice switches and lights |
| 04:42.27 | dzup | works? |
| 04:42.38 | Beta2K | Did it glow at night? |
| 04:43.45 | dzup | rjk`: if you plugit in, works? |
| 04:44.11 | rjk` | i gave it away to a collector when i couldn't keep the collection anymore |
| 04:44.17 | rjk` | it was fully operational |
| 04:44.34 | dzup | wow |
| 04:44.40 | rjk` | had a 5 megabyte 12 inch disk pack in the bottom |
| 04:45.36 | rjk` | this is the story of the guy who took it off my hands |
| 04:45.38 | Beta2K | One of those drives that you could remove the platters from? |
| 04:45.39 | rjk` | http://www.chookfest.net/nova3/ |
| 04:45.43 | rjk` | yes |
| 04:46.09 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 04:46.38 | rjk` | that link is a funny read |
| 04:46.44 | rjk` | especially the phone call bit |
| 04:49.25 | dzup | can you recall the name of the operating system that use? |
| 04:50.21 | rjk` | DGOS |
| 04:50.56 | rjk` | in my past i used to work on big honeywell machines when computer operators existed |
| 04:51.24 | rjk` | DPS 6 , Level 66, they ran GCOS which was very unix like |
| 04:51.40 | rjk` | thats going back 22 years or more |
| 04:52.55 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 04:53.37 | dzup | damm |
| 04:54.21 | dzup | the oldest one i ever used was imb system 36 heh |
| 04:54.42 | dzup | ibm* |
| 04:55.03 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 04:55.31 | rjk` | oldest i worked on was a Feranti Argus 500 , 12 bit computer |
| 04:55.39 | rjk` | all discreet |
| 04:56.08 | rjk` | had 4 big vertical spinning 1 Megabyte disks in big boxes |
| 04:56.24 | rjk` | they were as high as me |
| 04:56.34 | rjk` | 7 cabinets per computer |
| 04:56.37 | rjk` | there were 3 |
| 04:56.53 | rjk` | paper tape and ASR 33 teletypes for I/O |
| 04:57.07 | rjk` | ran an OS called April |
| 04:57.13 | *** join/#asterisk devoid (~devnull@unaffiliated/devemo) |
| 04:57.16 | rjk` | which was 17k big |
| 04:57.48 | rjk` | it wasnt decommissioned until 1992 |
| 04:57.53 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 04:57.54 | rjk` | as it did the job |
| 04:58.43 | *** join/#asterisk mianos (~mrfish@124-168-146-31.dyn.iinet.net.au) |
| 04:59.18 | rjk` | http://en.wikipedia.org/wiki/Ferranti_Argus |
| 04:59.49 | mianos | haha, I worked on them |
| 04:59.59 | rjk` | seriously ? |
| 05:00.12 | mianos | did the telex switching for the Australian department of aviation |
| 05:00.23 | rjk` | you knew ed parsons |
| 05:00.31 | mianos | real magnetic core |
| 05:00.42 | mianos | 6 layer PCBS |
| 05:00.55 | rjk` | at the DCA under the control tower at tulla ? |
| 05:00.56 | mianos | scary to unsolder |
| 05:01.22 | mianos | Sydney Kinsford Smith |
| 05:01.33 | rjk` | dad worked at melbourne message switching |
| 05:01.36 | mianos | it was in that block underneath the old tower next to the water |
| 05:01.52 | rjk` | you'd have spoken to ed parsons |
| 05:02.01 | mianos | can't recall the name |
| 05:02.03 | rjk` | and alan knight |
| 05:02.11 | mianos | can't really remember anyone's name |
| 05:02.15 | rjk` | and someone bullock |
| 05:02.16 | mianos | hmm, alan knight |
| 05:02.17 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 05:02.20 | mianos | familiaer |
| 05:02.24 | mianos | maybe I can |
| 05:02.30 | rjk` | dad did your job in melbourne |
| 05:02.31 | mianos | shift station |
| 05:02.51 | rjk` | i worked on the ones at mmbw sewerage plant at carrum in the 80s |
| 05:03.36 | mianos | 85 or so I think |
| 05:03.38 | mianos | or 86 |
| 05:04.16 | rjk` | automatic message switching it was called |
| 05:04.20 | rjk` | wow small world |
| 05:04.23 | mianos | hmm ok worked it out |
| 05:04.24 | mianos | 1983 |
| 05:04.46 | mianos | yep that's what it was called |
| 05:04.59 | mianos | then I moved up to "radar bright display" |
| 05:05.02 | rjk` | dad started at essendon airport in radio maintenance and then moved on to tulla radar and then message switching |
| 05:05.05 | mianos | we had a new DG there |
| 05:05.20 | mianos | scan converters and analogie radar |
| 05:05.29 | rjk` | the mock up for that radar system was next to the AMSC building at tulla |
| 05:05.58 | rjk` | looked so modern at the time |
| 05:06.39 | rjk` | you lived in the real computer age, you had 3 systems in sydney ? two working one spare ? |
| 05:06.56 | mianos | not every modern for me, we already had apple IIe and TRS 80 at school |
| 05:07.06 | mianos | that feranti was quite fast though |
| 05:07.18 | rjk` | the console was awesome |
| 05:07.26 | mianos | yer, we used to run chess on the cold spare |
| 05:07.38 | mianos | they had two hots with a 'fast data link' between them |
| 05:07.42 | rjk` | but those vertical spinning disks were out of this world they were so bloody big |
| 05:08.00 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 05:08.05 | mianos | it's all coming back haha |
| 05:08.16 | mianos | I rememebr beign explained the arbitration |
| 05:09.06 | rjk` | wire wrap :) |
| 05:09.10 | rjk` | that was fun |
| 05:09.12 | mianos | the backplane |
| 05:09.26 | mianos | it looked cool, the backplane was quite good |
| 05:09.47 | mianos | the FDL had huge umbilicals |
| 05:09.51 | rjk` | grant lampart |
| 05:09.56 | rjk` | trevor bartlet |
| 05:10.16 | mianos | bartlet, wasn;'t he the boss? |
| 05:12.47 | bmg505 | good morning, have not seen the phrase asr33 in a long time |
| 05:12.47 | rjk` | yeah trevor bartlet became the boss |
| 05:13.40 | bmg505 | wonders when decnet is going to be mentioned :) |
| 05:13.57 | *** join/#asterisk blaines (~blaines@ip68-228-243-100.ph.ph.cox.net) |
| 05:14.11 | dzup | the matrix :p |
| 05:16.01 | bmg505 | our country was under sanctions and no one was suppose to have all these toys, my first "embedded system" I worked on was a PDP8, good to see there is other old geysers around |
| 05:16.25 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 05:17.36 | rjk` | the argus 500 made a pdp 8 look like a supercomputer :P |
| 05:17.43 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:17.48 | mianos | yer but it was quite fast |
| 05:17.58 | rjk` | well it was a real time system |
| 05:18.05 | rjk` | one of the earliest ever |
| 05:18.13 | bmg505 | yea I realized that when I had to copy the papertape ipl's and had to use an argus |
| 05:18.38 | rjk` | the chips were all DTL |
| 05:18.42 | bmg505 | well actually had to make a change and re-write it |
| 05:18.46 | rjk` | not TTL |
| 05:18.52 | rjk` | but DTL |
| 05:19.13 | rjk` | all wirewrap |
| 05:19.20 | rjk` | we used to make our own mods |
| 05:19.23 | bmg505 | argh, this trip down memory lane is making me re-live the nightmare |
| 05:19.29 | rjk` | lol |
| 05:19.38 | rjk` | its fun admit it |
| 05:19.53 | bmg505 | yea I actually still have my 220v wrapping tool |
| 05:19.58 | bmg505 | I kept it when I left |
| 05:20.37 | rjk` | i have an argus memory core that i pinched when they decommissioned our argus at the board of works |
| 05:20.41 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 05:20.49 | rjk` | it ran the sewerage system |
| 05:20.53 | bmg505 | I actually still have 41613 connectors a well, the type they used for the pdp's |
| 05:21.06 | rjk` | the one my dad and mianos worked on ran message switching for the airports |
| 05:21.34 | rjk` | 3 shifts a day |
| 05:22.31 | bmg505 | what was scary was the whole floor dedicated to aircons and ups/generators |
| 05:22.49 | bmg505 | and those thick power cables :) |
| 05:22.50 | rjk` | omg someone is selling an argus http://www.metronome-trading.com/argus/ |
| 05:23.30 | rjk` | thats a little version though |
| 05:23.58 | rjk` | mianos: can you still think in octal ? :P |
| 05:24.25 | bmg505 | Argus was written in a language that could only be understood by one surviving programmer. <--- like rightm I know at least 2 other guys |
| 05:24.50 | rjk` | yeah i read that |
| 05:25.11 | rjk` | APRIL wasn't hard anyway |
| 05:26.06 | rjk` | i had a fortran on the argus at dads work i used to use on the spare system when i spent holidays at work with him |
| 05:26.33 | rjk` | i wrote a tiny basic for it when i worked at MMBW |
| 05:27.04 | rjk` | we even made our own 8" disk drive controller for it |
| 05:27.12 | rjk` | for 8" floppies |
| 05:28.15 | mianos | those shots in that one for same are perfect |
| 05:28.27 | mianos | 6 layer boards, no sockets |
| 05:28.29 | mianos | haha |
| 05:28.47 | mianos | I remember changing chips, all discrete logic |
| 05:29.04 | mianos | looking at it |
| 05:29.07 | mianos | I bet that's it |
| 05:29.27 | mianos | same ones, for sale in sydney |
| 05:30.03 | rjk` | so was trevor PTO of sydney as well as melbourne ? |
| 05:30.22 | rjk` | i know dad refused the job as he was taking a package |
| 05:30.58 | mianos | Principal Technical Officers |
| 05:30.59 | mianos | haha |
| 05:31.14 | rjk` | its funny , in 1983 dick smith was my boss when i had my first job at dick smith electronics and dick smith was dads boss when dad left |
| 05:31.41 | rjk` | im blown away seriously |
| 05:31.56 | rjk` | whats the chances of finding ANYONE who knows an argus 500 |
| 05:32.17 | rjk` | let alone worked the same place , albeit geographically different, as my dad |
| 05:32.35 | mianos | ZN logic |
| 05:32.43 | mianos | ripple adders adn such |
| 05:32.56 | mianos | no board jockying in there |
| 05:33.11 | mianos | they solved a problem to the point of knowing what gate was failing |
| 05:33.16 | rjk` | those machines were kept in service so long because nothing was good enought to replace them |
| 05:34.23 | rjk` | each computer had its own desk |
| 05:34.39 | rjk` | and there was a bench with a row of teletype machines |
| 05:34.43 | mianos | I remember now, 'patches' |
| 05:34.45 | mianos | I did lots of those |
| 05:34.52 | rjk` | heh |
| 05:34.54 | mianos | sodler teflon wire |
| 05:34.59 | mianos | solder |
| 05:35.14 | mianos | maybe run it through a drill through |
| 05:35.18 | bmg505 | rofl I will never forget my first experience with teflon wire |
| 05:35.24 | rjk` | hold old are you mianos ? |
| 05:35.31 | mianos | 47 |
| 05:35.35 | rjk` | im 43 |
| 05:35.41 | rjk` | dads 66 |
| 05:35.43 | rjk` | lol |
| 05:35.56 | bmg505 | 47 this side as well |
| 05:36.07 | bmg505 | 63 model |
| 05:36.20 | mianos | me too, a good year :) |
| 05:36.21 | rjk` | i remember being the youngest at board of works and i became supervisor in 14 months which pissed the old guys off |
| 05:36.58 | rjk` | but i used to play with the AMSC argus from the age of 10 |
| 05:36.59 | mianos | funny thing rjk` is it would be so simple to replace a lot of those really epsnsive ILS systems with DSP now |
| 05:37.03 | rjk` | so i had lots of training |
| 05:37.23 | mianos | direct conversion DSP like in phones |
| 05:37.29 | rjk` | yep but they work and its the aviation department ;P |
| 05:37.42 | mianos | yer long life |
| 05:37.52 | rjk` | they dont like change |
| 05:37.53 | rjk` | lol |
| 05:38.13 | rjk` | do you remember rod irving electronics ? |
| 05:38.19 | *** join/#asterisk sourcode (~code@ppp-61-90-10-179.revip.asianet.co.th) |
| 05:38.31 | bmg505 | my father used to say that moving from analog and valve systems to logic based was the most traumatic move the aviation industry had |
| 05:38.50 | rjk` | rod irving was also at AMSC in the early years before he started Rod Irving Electronics |
| 05:39.54 | mianos | lots of trips on the DOA (remember they chanegd the name, whata choice) cars to Radio Spares |
| 05:41.56 | rjk` | okies i gotta go for a bit, great to catch up |
| 05:42.47 | coppice | bmg505: as traumatic as the early days of jets, when comets were falling out of the sky, and nobody understood why? :-) |
| 05:45.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:45.52 | bmg505 | lol so true |
| 05:47.21 | coppice | the real trauma with electronics in aircraft was the move to fly by wire. being valve and analogue to being digital was relatively easy to manage |
| 05:48.35 | mianos | F16 avionics |
| 05:48.41 | mianos | was a cool change |
| 05:48.48 | mianos | but no choice |
| 05:49.00 | coppice | the big change was before the F16 |
| 05:49.15 | mianos | it was not trimmable to be stable |
| 05:49.42 | mianos | but most airframes were stable |
| 05:50.02 | coppice | the F16 is relatively stable. if you want to see an early truly unstable design look at something like the mirage 2000 |
| 05:50.45 | mianos | fair enough, I don't know anything about the European planes |
| 05:53.05 | coppice | the control arrangements in the F16 were probably the coolest step for the pilot :-) |
| 05:53.25 | mianos | the joystick |
| 05:53.46 | mianos | straign gauges? |
| 05:53.56 | mianos | or was that the airbus |
| 05:54.08 | mianos | it's doesn't actually move on the airbus |
| 05:54.56 | coppice | I don't think any of the modern joysticks move much |
| 05:55.30 | Beta2K | A340 moves a fair bit |
| 05:56.21 | coppice | of course. until the 777-300ER it had the longest range in the world :-) |
| 05:57.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:57.34 | mianos | hey, asterisk question, anyone in AU withe TDM400P? |
| 05:57.56 | Beta2K | asterisk questions are currently off-topic :) |
| 05:58.21 | mianos | FXOs are all good, OPERMODE is Australia in the kenrel |
| 05:58.25 | mianos | kernel |
| 05:58.49 | mianos | <PROTECTED> |
| 05:59.04 | mianos | but my call progress does not work |
| 05:59.09 | dzup | yeah if you want asterisk please /join #BP-MESS-fixup |
| 05:59.35 | mianos | no that's for porn downloading sites |
| 06:00.07 | dzup | fne /join BP-OIL-SPILL |
| 06:00.14 | dzup | :p |
| 06:01.24 | *** join/#asterisk Tim_Toady (~moi@77.49.21.138.dsl.dyn.forthnet.gr) |
| 06:01.26 | coppice | its nice to see Britain still leads the world in some things |
| 06:02.40 | dzup | lol |
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| 06:38.40 | rjk` | can i link asterix to skype as the outbound calling device ? |
| 06:41.09 | kaldemar | ~sfa |
| 06:41.31 | kaldemar | http://www.digium.com/en/products/software/skypeforasterisk.php |
| 06:43.44 | rjk` | thanks |
| 06:43.54 | rjk` | because i found Nerdvittles Teleyapper |
| 06:44.02 | rjk` | which does the job on asterisk |
| 06:44.11 | rjk` | so i could do the lot on the mac i am guessing |
| 06:49.44 | mianos | hey rjk` do you have TDM400P in australia? |
| 06:53.30 | *** join/#asterisk costal (~ivan@203.84.234.194) |
| 06:53.36 | costal | Hi there |
| 06:53.37 | *** join/#asterisk Z_God (~julius@wlan232185.mobiel.utwente.nl) |
| 06:54.06 | costal | I'm wondering if you guys can refer me to some source where the CCR and G549 numbering format is used |
| 06:56.29 | rjk` | mianos: dunno |
| 06:56.45 | rjk` | mianos: however i just found macports has asterisk which is installing now :) |
| 06:58.08 | rjk` | TDM400P http://forums.whirlpool.net.au/forum-replies-archive.cfm/429440.html |
| 06:58.10 | rjk` | yep |
| 07:01.34 | voxter | Kinda weird, authenticating to meetme() if i get the pin wrong, it says invalid, and exits after the first try |
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| 07:05.50 | rjk` | now i wonder if i can find a free asterisk to skype rather than the $66 one from digium |
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| 07:15.26 | ChannelZ | rjk`: there's a shitty one (maybe) still available written by some company that requires a running Skype client and tries to use their public API to communicate with it |
| 07:16.24 | ChannelZ | But really, you spend money on a TDM card but are cranking about $66 for SFA? |
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| 08:08.33 | ijpalmer | good morning, I'm using mixmonitor to record calls, when I make an outbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
| 08:09.12 | ijpalmer | I'm using * 1.6.2.6 |
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| 08:23.21 | mtryfoss | anyone seen this kind of warning? WARNING[25986] chan_dahdi.c: Can't fix up channel from 37 to 41 because 41 is already in use |
| 08:23.30 | mtryfoss | is it harmless? |
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| 08:54.17 | mianos | rjk`no, I have one, just setting it up. Was having issues with the call progress in au |
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| 09:09.55 | Xt0f | hello. Does anybody has got experience with res_odbc + free tds crashes ? Thanks in advance ! |
| 09:10.04 | Xt0f | For realtime |
| 09:10.11 | ijpalmer | good morning, I'm using *1.6.2.6, I use mixmonitor to record calls, when I make an outbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
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| 09:20.55 | yabadabado | I just went through the asterisk book - future of telephony. Its wonderful. |
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| 09:45.13 | ijpalmer | good morning, I'm using *1.6.2.6, I use mixmonitor to record calls, when I make anoutbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
| 09:46.18 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 09:47.19 | *** join/#asterisk Zakar|a (~Me@wana-235-228-12-196.wanamaroc.com) |
| 09:47.22 | Zakar|a | Hello |
| 09:47.29 | Zakar|a | I need help with Asterisk |
| 09:47.41 | Zakar|a | I'm trying to compile Zaptel |
| 09:47.51 | Zakar|a | but it gives me an error while i typed "make" |
| 09:48.21 | Zakar|a | anyone can help? |
| 09:50.40 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 09:51.51 | Zakar|a | anyone can help? |
| 09:52.59 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:56.18 | *** join/#asterisk sidd (~siddharth@59.90.209.162) |
| 09:56.59 | sidd | hi, i am facing aroblem where if a client disconnects after registration, its peer does not come to know for the next 60 seconds and tries to call. can i reduce this timeout? |
| 09:58.02 | WIMPy | Zakar|a: Might help if you at leas tell, what error. |
| 09:58.17 | WIMPy | sidd: Search for qualify* options. |
| 09:58.45 | sidd | WIMPy: ok. am a complete noob but still will search. |
| 09:59.04 | ChannelZ | sidd: in sip.conf |
| 09:59.20 | ChannelZ | assuming these are sip peers |
| 09:59.37 | sidd | they are. it says qualify=1000 for my clients |
| 10:00.35 | sidd | WIMPy: also, i have noticed that for some clients behind nat, i have to keep changing the ip of my client machine or else, the client does not get registered after disconnection due to adsl down. |
| 10:01.14 | sidd | is there a reconnect timeout that i can reduce? |
| 10:01.21 | ChannelZ | sidd: you can set qualifyfreq=30 to have it poke the peer every 30 seconds for instance |
| 10:01.36 | sidd | ChannelZ: thanks...will try that. |
| 10:03.03 | ChannelZ | As for peers changing IPs, it's really their responsibility to re-register |
| 10:04.31 | sidd | but the server keeps the earlier ip as registered (strangely the ip of the client behind nat is seen in the sip show peers) and the client is unable to register |
| 10:05.04 | sidd | for natted client should i put something in sip.conf specifically? |
| 10:06.01 | ChannelZ | nat=yes |
| 10:06.18 | ChannelZ | host=dynamic |
| 10:06.46 | sidd | ChannelZ: its not there for my clients. is that why sip show peers shows the inner ip and not of the natting machine. |
| 10:08.32 | *** part/#asterisk manatails (~admin@reactos/tester/manatails) |
| 10:13.16 | Zakar|a | WIMPy What should I install, Zaptel or DAHDI for Conference in Asterisk ? |
| 10:14.20 | WIMPy | dahdi is the new name of zaptel. |
| 10:14.23 | kaldemar | Zakar|a: if your version of asterisk is 1.4.22 or newer, use DAHDI. zaptel was renamed to dahdi some time ago, 1.4.22 is the first version of asterisk that supports DAHDI. |
| 10:14.40 | WIMPy | And with newer Asterisks neither is required. |
| 10:15.13 | Zakar|a | kaldemar that's what i found in a blog of Digium asking users to use DAHDI instead of Zaptel |
| 10:15.25 | Zakar|a | i really really have a headache because of Zaptel |
| 10:15.32 | Zakar|a | i kept searching and searching |
| 10:15.40 | Zakar|a | when i try to compile "make" |
| 10:15.43 | Zakar|a | it showed an error |
| 10:15.48 | Zakar|a | i did many tries |
| 10:15.57 | Zakar|a | but now i'm going to install Dahdi |
| 10:16.33 | Zakar|a | should i install dahdi package from asterisk's web site? |
| 10:16.50 | WIMPy | If you only need it for meetme, try a current version of Asterisk and forget about zaptel/dahdi. |
| 10:17.03 | Zakar|a | yes it's for meet me |
| 10:17.04 | WIMPy | .. and use confbridge instead. |
| 10:17.16 | Zakar|a | conf bridge? |
| 10:17.23 | Zakar|a | excuse me but how? |
| 10:17.43 | Zakar|a | because when i configure an extension for conference, it says The number of conference is invalid |
| 10:17.46 | Zakar|a | something like this |
| 10:18.09 | WIMPy | confbridge is a similar application to meetme that does not requre dahdi timing. |
| 10:18.10 | Zakar|a | I use the latest version of AsteriskNOW |
| 10:18.20 | Zakar|a | and how can I install it? |
| 10:19.02 | WIMPy | Recente versions of Asterisk have it. No idea about AsteriskNOW. |
| 10:19.19 | Zakar|a | i just made a yum update |
| 10:19.42 | Zakar|a | so |
| 10:19.47 | Zakar|a | how can i install it |
| 10:20.28 | WIMPy | The usual way would be to go to digiums site, download the source tarball and install from there. |
| 10:21.37 | Zakar|a | but in digium i don't find the confbridge |
| 10:22.21 | WIMPy | It's part of Asterisk, just like meetme. |
| 10:22.28 | Zakar|a | how about VisioConference |
| 10:22.33 | Zakar|a | i mean i want to include the Video |
| 10:22.37 | Zakar|a | in the conference |
| 10:22.40 | Zakar|a | possible? |
| 10:22.50 | WIMPy | Asterisk won't do video conferencing. |
| 10:23.11 | Zakar|a | :S |
| 10:23.14 | Zakar|a | are you sure |
| 10:23.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:23.40 | WIMPy | That's a one-to-one. |
| 10:24.21 | WIMPy | There are addons available to do vide switching for conference, AFAIK, but no mixing/tiling. |
| 10:24.47 | Zakar|a | hmm |
| 10:26.03 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 10:26.40 | MAbbas | Hi All, Agent1 has parked incoming customer's call, how can Agent2 pick the parked call? parking slot for call is 701 |
| 10:26.41 | MAbbas | ? |
| 10:27.58 | ChannelZ | You dial 701 |
| 10:29.31 | MAbbas | When logged in Agent2, dials 701 from his softphone nothing happens .. call is still parked |
| 10:29.32 | MAbbas | ? |
| 10:30.13 | MAbbas | pardon me I am new and experimenting with asterisk call transfers .. |
| 10:30.23 | MAbbas | <ChannelZ> ? |
| 10:30.55 | ChannelZ | First just so we're talking about the same thing, you have "include => parkedcalls" in your extensions.conf right? And you transfer a call to 700 to park it? |
| 10:31.17 | MAbbas | yes .. |
| 10:32.06 | ChannelZ | Is the person trying to pick up the call in the same context as that include statement? |
| 10:33.24 | MAbbas | <ChannelZ> yes same context, I name it [test]. My bad for being naieve, from softphone how can agent pick a parked call. |
| 10:34.40 | ChannelZ | You should just be able to dial the extension it told you it parked the call at. |
| 10:36.11 | MAbbas | <ChannelZ> when agent2 dials 701, nothing happens. |
| 10:36.38 | ChannelZ | What does the console say? Something should be happening. |
| 10:38.02 | MAbbas | <ChannelZ> My scenario, 2 agents logged in AgentLogin() to queue. when call comes to Queue(). It is sent to agent1. Agent1 #700 to park call. |
| 10:38.30 | MAbbas | <ChannelZ> now how can agent2 pick, parked call at slot 701? |
| 10:39.36 | MAbbas | <ChannelZ> On agent2 softphone I simply dial 700.. Asterisk console does not shows anything .. |
| 10:40.21 | *** join/#asterisk HenrikJott (~info@h34n2fls33o954.telia.com) |
| 10:40.33 | mtryfoss | anyone got an idea of why asterisk/dahdi tries to move calls from one channel to another ? the channel may be taken by the telco already ? |
| 10:41.02 | HenrikJott | Hi all! I was wondering which is the correct way to make anonymous calls from asterisk? I´m using call-files the set the caller-id... |
| 10:41.42 | ijpalmer | good morning, I'm using *1.6.2.6, I use mixmonitor to record calls, when I make an outbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
| 10:43.16 | ChannelZ | MAbbas: YOU DIAL 701 to pickup the call parked on 701, NOT 700!! |
| 10:43.20 | MAbbas | <ChannelZ> however, if I dial 701 from softphone where no agent is logged in. call is connected |
| 10:43.50 | ChannelZ | Maybe it's an issue with parking queued calls. I have no idea. What version of Asterisk? |
| 10:43.53 | *** join/#asterisk kk7av (~rob@pdpc/supporter/professional/rob-p) |
| 10:43.54 | MAbbas | <ChannelZ> I need to know how can a logged in agent pick a parked call. |
| 10:44.03 | MAbbas | <ChannelZ> asterisk 1.4 |
| 10:44.10 | ChannelZ | 1.4.what |
| 10:44.20 | MAbbas | <ChannelZ> 1.4.29 |
| 10:45.27 | MAbbas | <ChannelZ> I rephrase my question. Can an agent pick a parked call? |
| 10:47.01 | *** join/#asterisk sidd (~siddharth@59.90.209.162) |
| 10:47.16 | *** part/#asterisk sidd (~siddharth@59.90.209.162) |
| 10:47.28 | ChannelZ | I understand your question. The answer is that I think it should, because an agent is really a function of a queue and not otherwise special, but maybe there is a bug, I don't know. |
| 10:48.21 | ChannelZ | The console should show *something*. Make sure you have some verbosity turned on. |
| 10:48.27 | ChannelZ | I have to go to bed so I can be to work in 4 hours. |
| 10:49.33 | MAbbas | <ChannelZ> Just to make sure we are on same page. Agent2(who is free) simply dials 701. * console does not show anything I have verbosity set to 5. |
| 10:49.52 | MAbbas | <ChannelZ> I even tried with #701 |
| 10:50.54 | *** join/#asterisk Beowolf (~Administr@82-69-4-3.dsl.in-addr.zen.co.uk) |
| 10:52.38 | WIMPy | MAbbas: If the console shows nothing, that seems to be an issue on the phone. |
| 10:52.40 | Beowolf | Are the forums at http://forums.digium.com/ down for everyone or just me? |
| 10:53.46 | WIMPy | Beowolf: No route to host |
| 10:54.03 | Beowolf | ah, ok |
| 10:54.05 | Beowolf | damn |
| 10:54.36 | Beowolf | Are there some other widly used forums? |
| 10:54.58 | *** join/#asterisk sourcode (~code@ppp-61-90-10-179.revip.asianet.co.th) |
| 11:00.15 | Beowolf | widely* |
| 11:03.51 | *** join/#asterisk AsteriskNoob (~AsteriskN@host86-128-21-197.range86-128.btcentralplus.com) |
| 11:04.00 | *** join/#asterisk Z_God (~julius@wlan239014.mobiel.utwente.nl) |
| 11:04.37 | AsteriskNoob | Hello all. Can anyone tell me how to prevent a forwarded handset from causing a dial pool failure? |
| 11:07.51 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 11:10.33 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 11:11.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:13.24 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 11:25.57 | *** join/#asterisk glam (~glam@58.252.229.246) |
| 11:26.14 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 11:26.46 | glam | hi, am running freepbx 2.7.0.2. sip show settings shows direct rtp setup = no, how do i enable it ? |
| 11:26.53 | *** join/#asterisk BANSAL (~bansal@117.207.83.40) |
| 11:36.47 | Beowolf | glam, uncomment the line ;directrtpsetup=yes in sip.conf |
| 11:36.50 | kaldemar | glam: no idea on enabling it via freepbx, but in asterisk, directrtpsetup=yes in sip.conf. for freepbx support there is #freepbx. |
| 11:37.16 | Beowolf | oh, freepbx... |
| 11:39.43 | Beowolf | I have a strange problem. When asterisk is started from cron, an agi script doesn't run. When asterisk is started manually - with the same command, and as the same user - the script runs fine. I've checked all the permissions I can think of, but can't get it working. Asterisk 1.6.2.2 |
| 11:40.22 | Beowolf | I originally had the script called via a system() call, but that had exactly the same problem |
| 11:44.58 | WIMPy | You start Asterisk via cron? |
| 11:48.48 | *** join/#asterisk rymkus (~pcccp@mx.okhtaform.ru) |
| 11:50.08 | Beowolf | if it dies, a script restarts it |
| 11:55.41 | *** part/#asterisk rymkus (~pcccp@mx.okhtaform.ru) |
| 11:56.53 | Beowolf | ok, I missed being able to add it to inittab. I wonder if that would solve the problem - or at least avoid it. |
| 11:57.27 | *** join/#asterisk stix (~sorena@exchange2003.corporate.billetkontoret.dk) |
| 11:57.57 | stix | Hi guys. Anyone know a sip-soft-phone which can look numbers up in a global directory? |
| 11:58.07 | WIMPy | At least it restarts it faster. |
| 11:59.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:59.51 | mianos | if I have phones of dahdi ports with dahdichan=X defined |
| 11:59.54 | kaldemar | stix: what do you mean by "look numbers up"? |
| 11:59.59 | *** join/#asterisk blinky42 (~quassel@66.54.228.71) |
| 12:00.00 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 12:00.04 | mianos | is there a dial syntax to dial that line? |
| 12:00.17 | mianos | like Dial(DAHDI/robo) ? |
| 12:00.25 | mianos | where I have |
| 12:00.27 | mianos | [robo] |
| 12:00.35 | mianos | dahdichan=2 |
| 12:00.38 | kaldemar | mianos: Dial(DAHDI/<chan>) |
| 12:00.58 | mianos | yer I knwo I can ahrd dial the port |
| 12:01.04 | kaldemar | mianos: in that case Dial(DAHDI/2) |
| 12:01.24 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:01.39 | mianos | yes that works, but I don't want my extensions full of hard coded TDM ports do I? |
| 12:01.52 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:3c13:f086:66f:188) |
| 12:02.12 | mianos | this works: |
| 12:02.14 | mianos | [from-pstn] |
| 12:02.14 | mianos | exten = s,1,Dial(DAHDI/4, 20) |
| 12:02.24 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:02.59 | mianos | but what's the point of the dahdichan line in users.conf? |
| 12:03.03 | kaldemar | mianos: that's what you have. you want the channel from users.conf? |
| 12:03.23 | mianos | yes |
| 12:03.42 | mianos | maro it os something I guess |
| 12:03.45 | mianos | mcaro |
| 12:04.10 | kaldemar | core show function AST_CONFIG |
| 12:04.20 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:04.21 | mianos | Dial(DAHDI/{some macro that converts this to a port:robo}) |
| 12:04.38 | mianos | anyway, you recon that's possible? |
| 12:04.45 | kaldemar | Dial(DAHDI/${AST_CONFIG(users.conf,robo,dahdichan)}) might work. try it. |
| 12:04.53 | mianos | haha |
| 12:04.56 | mianos | ok I get the idea |
| 12:05.57 | mianos | if that's the way to do it thanks |
| 12:05.57 | tzafrir_laptop | mianos, 'dahdichan' is for chan_dahdi.conf (as of asterisk 1.6.1) |
| 12:05.57 | mianos | I just thought with all the trouble to get the dahdichan in users.conf there would be some normal way |
| 12:05.57 | tzafrir_laptop | oh, and also for users.conf in earlier versions |
| 12:06.01 | kaldemar | tzafrir_laptop: it's in users.conf in 1.6.2 also. |
| 12:06.51 | tzafrir_laptop | it's supported in users.conf as of 1.4 . But in chan_dahdi.conf it's only useful as of 1.6.1 (where seperate config sections are supported there as well) |
| 12:07.40 | tzafrir_laptop | it's the same as having 'channel => 2' in the end of that section |
| 12:07.48 | mianos | running 1.6.2 so it's cool |
| 12:07.48 | *** join/#asterisk d4rkstar (~bruno@ip-242-50.sn2.eutelia.it) |
| 12:08.22 | tzafrir_laptop | mianos, users.conf also generates some implicit dialplan for every channel |
| 12:08.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:08.34 | mianos | yes |
| 12:08.35 | tzafrir_laptop | I guess you can use it |
| 12:09.03 | mianos | I always hand written config files but I'm giving asterisk-gui a try and it's filled out users.conf |
| 12:09.29 | mianos | and I was trying to find the best way to map an incoming FXO to a particular user |
| 12:09.32 | [TK]D-Fender | ~users.conf |
| 12:09.33 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 12:10.00 | mianos | yes, I've never used it before |
| 12:11.07 | [TK]D-Fender | mianos: It is unmaintained. I do not recommend it for production. |
| 12:11.27 | ijpalmer | [root@cvt-lpt-01 dialplans]# ls |
| 12:11.31 | ijpalmer | [root@cvt-lpt-01 dialplans]# ls |
| 12:12.06 | ijpalmer | good afternoon, I'm using *1.6.2.6, I use mixmonitor to record calls, when I makeanoutbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
| 12:12.08 | mianos | asterisk-gui, yes I see all the bugs marked as such |
| 12:12.19 | mianos | but it's pretty simple to work with |
| 12:12.29 | mianos | I've fixed 4 bugs so far |
| 12:12.33 | mianos | just to use it |
| 12:12.37 | mianos | it's good for me now |
| 12:13.05 | mianos | otherwise I'll go back to hand configs, I don't buy into these zillion line freePBX thingies |
| 12:13.52 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 12:15.44 | tzafrir_laptop | mianos, look at the dialplan in the context of that channel (dialplan show that-ocntext) |
| 12:16.09 | tzafrir_laptop | Look for lines that do not come from [pbx_config] |
| 12:19.32 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:22.36 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:23.43 | mtryfoss | anyone got an idea of why this happens: chan_dahdi.c: Can't fix up channel from 71 to 72 because 72 is already in use ? |
| 12:29.43 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 12:48.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 12:52.51 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:52.51 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:54.21 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:54.43 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:58.37 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 12:59.31 | t_dot_zilla | anybody ever setup a mp118 with asterisk? |
| 13:00.05 | MAbbas | [TK]D-Fender I am trying to use asterisk consult transfer feature. But when I try to consult other agent, I am unable to communicate with other agent. My dialplan and CLI output is http://dpaste.de/ukwf/ |
| 13:02.06 | [TK]D-Fender | MAbbas: [Jun 1 06:57:11] NOTICE[27121]: res_features.c:1442 ast_feature_request_and_dial: We exceeded our AT-timeout Toot too long |
| 13:02.11 | [TK]D-Fender | Took too long |
| 13:02.21 | [TK]D-Fender | MAbbas: And you didn't show the complete call |
| 13:03.42 | *** join/#asterisk maldous (~maldous@scooby.unicity.com.au) |
| 13:03.44 | MAbbas | [TK]D-Fender well, call was queued and connected to agent 201 |
| 13:04.02 | MAbbas | what do you mean by "Took too long"? |
| 13:04.14 | [TK]D-Fender | MAbbas: I don't do "story time". I do debug. |
| 13:04.20 | *** join/#asterisk DND (~arabia@94.200.7.22) |
| 13:04.33 | [TK]D-Fender | MAbbas: " We exceeded our AT-timeout" <-- TOOK TOO LONG |
| 13:04.42 | DND | hi guys i need help. my asterisknow keeps logging in messages a couple of segfault |
| 13:05.02 | DND | something like this: |
| 13:05.02 | [TK]D-Fender | MAbbas: And what phones are you using again? |
| 13:05.03 | DND | kernel: asterisk[20970]: segfault at 000000000c37f000 rip 00002aaaac109db5 rsp 000000004135b5d0 error 4 |
| 13:05.14 | [TK]D-Fender | DND: ---------> #asterisknow |
| 13:05.33 | maldous | yay. there is a channel. quick question. default centos install. yum install asterisk method and install as root. update manager.conf and http.conf - all web requests are return http 501 not implemented. is this a known? |
| 13:05.33 | DND | its a ghost channel. very rarely reply |
| 13:05.47 | MAbbas | I am using softphone eyeBeam and sjphone |
| 13:06.13 | MAbbas | [TK]D-Fender I am using softphone eyeBeam and sjphone |
| 13:06.41 | [TK]D-Fender | MAbbas: Both of those have normal SIP transfer capabilities. Why are you using DTMF for this? |
| 13:06.53 | maldous | or, can someone point me towards logging for web-gui? |
| 13:07.01 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 13:07.02 | [TK]D-Fender | DND: And you seem to have vague distro issues. |
| 13:07.09 | maldous | or, does web-gui work with 1.6 ? |
| 13:07.34 | [TK]D-Fender | maldous: What are you expecting to see? |
| 13:07.39 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:07.41 | MAbbas | [TK]D-Fender I want to use asterisk feature. Instead of using feature of softphone |
| 13:07.46 | [TK]D-Fender | maldous: What GUI? |
| 13:07.57 | maldous | anything other than "Not Implemented: Attempt to use unimplemented / unsupporte method" |
| 13:08.01 | [TK]D-Fender | MAbbas: Bad idea. DTMF features suck |
| 13:08.21 | [TK]D-Fender | maldous: From what you described you have nothing to show. There is no GUI in your install. |
| 13:08.37 | maldous | i see online documentation about a web gui. perhaps i'm mistaken. |
| 13:08.48 | MAbbas | [TK]D-Fender I just want to verify this feature. |
| 13:08.57 | maldous | the yum install (http://www.asterisk.org/downloads/yum) creates /etc/asterisk/http.conf |
| 13:09.06 | [TK]D-Fender | maldous: All of them are completely separate bolt-ons. Not inciuded. |
| 13:09.22 | maldous | http://www.asterisk.org/asterisknow/developers/gui-guide suggested others. |
| 13:09.28 | maldous | err, suggested otherwise. |
| 13:09.30 | [TK]D-Fender | included* |
| 13:09.37 | maldous | The web server included with both Asterisk 1.4, as well as the AsteriskNOW deployment, is used to display the Asterisk Manager Interface (AMI). |
| 13:09.56 | maldous | oh, this "AMI" isn't part of 1.6 ? |
| 13:10.16 | [TK]D-Fender | maldous: You can view the page source or checkout trunk GUI <----------- |
| 13:10.20 | maldous | i'm unsure why it's "not included", if it's part of the "out of the box configuration" |
| 13:10.20 | Naikrovek | AMI is, but it's not web-browser compatible. |
| 13:10.25 | [TK]D-Fender | maldous: this is nto part of what you described |
| 13:10.52 | Naikrovek | maldous: Asterisk DOES come with a web server, and it does come with AMI, but AMI is not usable via a web browser |
| 13:10.55 | *** join/#asterisk Baylink (~jra@205.201.151.25) |
| 13:10.57 | maldous | <[TK]D-Fender> : eh? I installed via yum. i read the gui-guide. have I not explained something? |
| 13:11.09 | maldous | ah. it's a web api? |
| 13:11.12 | MAbbas | [TK]D-Fender What does AT-timeout mean? That the dialed party did not respond? |
| 13:11.18 | [TK]D-Fender | maldous: I see no confirmation that you downlpaded and installed the GUI. |
| 13:11.24 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-jfxmigegyvqfuzkk) |
| 13:11.39 | [TK]D-Fender | maldous: And keep in mind that for several months now AsteriskGUI has not been maintained, |
| 13:11.50 | [TK]D-Fender | MAbbas: Apparently |
| 13:11.56 | maldous | <[TK]D-Fender> : ah. ok, i wasn't to know that's it's not part of the yum installation. |
| 13:12.07 | maldous | months I can deal with. |
| 13:12.08 | [TK]D-Fender | maldous: Read the page you linked to me. |
| 13:12.15 | [TK]D-Fender | [09:10]<[TK]D-Fender>maldous: You can view the page source or checkout trunk GUI <----------- |
| 13:12.29 | MAbbas | [TK]D-Fender Is my dialplan correct for attended call transfer? |
| 13:13.07 | [TK]D-Fender | MAbbas: Can't you read the error message? You can't have an attended transfer timeout... if you didn't actually start an attended transfer. |
| 13:13.31 | maldous | [tk] - the link to "page source" doesn't work for me. |
| 13:13.51 | maldous | svn it is.. |
| 13:13.58 | [TK]D-Fender | maldous: OLD links.... |
| 13:14.52 | [TK]D-Fender | maldous: the GUI has not been maintained for some time. Many old links are worthless |
| 13:15.13 | maldous | sounding like i shouldn't go down this path. no problem. |
| 13:16.06 | [TK]D-Fender | maldous: Wouldn't advise it. |
| 13:18.03 | maldous | i was just playing with proof of concept idea and wanted to register asterisk up as a simple sip client answering thingy. |
| 13:18.12 | maldous | (hoping the gui would have let me click click click) |
| 13:18.40 | maldous | only because i learnt the sip account details of my corporate desk phone today. |
| 13:20.09 | *** join/#asterisk m_tadeu (~quassel@89.180.0.145) |
| 13:20.10 | [TK]D-Fender | maldous: "answering thingy"? Something tells me what you want is pretty custom anyway which the GUI will not help at all. |
| 13:20.28 | [TK]D-Fender | maldous: Just about all of them are for creating cookie-cutter PBXs |
| 13:21.35 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:21.49 | maldous | well, right now, it's just a quick check to make sure i can register against my sip server. |
| 13:22.25 | [TK]D-Fender | maldous: GUI only slows down that process. getting * to register to your server is ONE LINE of code in sip.conf |
| 13:22.37 | maldous | yeah. figuring. |
| 13:23.19 | m_tadeu | hi...I just installed asterisk from source, setup sip and a dialplan with some "playback"'s...but I hear no sound in my softphone |
| 13:26.22 | [TK]D-Fender | m_tadeu: Describe in proper full detail the networking between * and your soft-phone, and included a failed call with SIP DEBUG enabled in a PASTEBIN |
| 13:26.24 | [TK]D-Fender | ~pb |
| 13:26.25 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:26.26 | [TK]D-Fender | ^^^^^^^6 |
| 13:27.19 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 13:28.33 | kk7av | I'm getting, "404 Not Found" on a regular basis, as I watch the debug output. It's not in response to attempted calls. See an example log snippet here: http://pastebin.com/bCnTykMG Everything *seems* to be working. I just want to know why it's happening, and how to fix it. Any ideas? |
| 13:29.06 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:29.26 | maldous | alrighty. made the sip.conf i think should be working. connected to console. any one know the command to see if i've registered? |
| 13:29.39 | [TK]D-Fender | kk7av: that is in response to a OPTIONS packet. * ALWAYS responds this way and it means nothing |
| 13:29.52 | [TK]D-Fender | maldous: "sip show registry" |
| 13:30.10 | m_tadeu | [TK]D-Fender: it's lan...both in the same 192.168.1.x...no firewalls...the debug that shows is small "<--- SIP read from UDP:192.168.1.10:5060 --->" |
| 13:30.19 | [TK]D-Fender | kk7av: that is only a NAT keep-alive. |
| 13:30.34 | kk7av | [TK]D-Fender: I see. Ok then. I won't fret over it! :-) |
| 13:30.35 | maldous | bah. nada. debug ? |
| 13:30.36 | [TK]D-Fender | m_tadeu: PASTEBIN <-------------- |
| 13:30.44 | kk7av | [TK]D-Fender: tnx |
| 13:30.46 | [TK]D-Fender | maldous: NOTHING? not text at all? |
| 13:31.04 | maldous | Host dnsmgr Username Refresh State Reg.Time |
| 13:31.04 | maldous | 0 SIP registrations. |
| 13:31.17 | [TK]D-Fender | maldous: Means you didn't put it in the right place if at all. |
| 13:31.18 | maldous | where "nada" mean "0 SIP registrations" |
| 13:31.42 | [TK]D-Fender | maldous: permanently remove every commented-out line from your sip.conf and pastebint he whole thing masking only passwords. |
| 13:32.28 | maldous | last line in sip.conf is "register => username@domain:password@proxy" |
| 13:32.37 | [TK]D-Fender | [09:31]<maldous>where "nada" mean "0 SIP registrations" <- Acceptable. I did expect to see the header at least and it tells me that your config doesn't have the right line in the right place |
| 13:33.05 | maldous | np. |
| 13:33.08 | [TK]D-Fender | maldous: All register staments have to come at the END of [general] and before any other section |
| 13:33.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 13:33.53 | ZeXr0 | I have an Asterisk server. It uses SIP line to dial normal phone number. When it dial a normal phone number, I'd like to be able to know if that number is in service. Right now, the "attendant of the phone company" says that the number is not in service. Is it possible to catch that because Asterisk return an error and in $REASON there's the error code 0. |
| 13:34.23 | *** join/#asterisk glam (~glam@58.252.229.246) |
| 13:35.07 | maldous | http://pastebin.com/32zGJq64 |
| 13:35.12 | maldous | [tk] - ah, ok. |
| 13:35.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:35.31 | WIMPy | ZeXr0: Hangupcause = 1 |
| 13:35.43 | m_tadeu | [TK]D-Fender: http://pastebin.com/sifkq7uR |
| 13:36.47 | maldous | ok, have a registry attempt now. sip server complaining, but asterisk is trying. thx. |
| 13:37.02 | ZeXr0 | WIMPy : thanks ! I'll see if I can do something with this :) |
| 13:38.12 | maldous | is register => username@domain:password@proxy a typical format? (similar parameters from x-lite are working) |
| 13:38.17 | ijpalmer | good afternoon, I'm using *1.6.2.6, I use mixmonitor to record calls, when Imakeanoutbound call and transfer the call it stops mixmonitor, I'm using audiohook_inherit and all other transferred calls are working |
| 13:38.20 | *** join/#asterisk Baylink (~jra@205.201.151.25) |
| 13:38.28 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:38.40 | maldous | ah. userid is different to authorisation name. |
| 13:39.00 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 13:39.17 | madduck | what's the usual reason for not hearing the dial tone? the phone otherwise works fine, it's just that there is no ringtone. |
| 13:39.23 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:39.26 | madduck | like when i am waiting for the other side to pick up. |
| 13:39.34 | [TK]D-Fender | madduck: dialtone != ring tone |
| 13:39.41 | [TK]D-Fender | madduck: Please picka story and stick with it |
| 13:39.50 | madduck | ;) |
| 13:39.51 | [TK]D-Fender | madduck: and tone WHERE? on what? |
| 13:39.51 | maldous | yay. i appear to be state registered. |
| 13:39.59 | [TK]D-Fender | maldous: \o/ |
| 13:40.13 | maldous | ;) |
| 13:40.20 | madduck | [TK]D-Fender: when i dial a number on my Siemens handset, I used to hear it go beep-beep-beep until the other side picked up |
| 13:40.22 | madduck | or the busy tone |
| 13:40.24 | maldous | alrighty. that was easy. |
| 13:40.29 | [TK]D-Fender | maldous: Congratulations. Your server should now send calls to *. You'll need a SIP peer to auth the incoming call and dialplan to process it |
| 13:40.36 | madduck | right now, i hear nothing |
| 13:40.57 | [TK]D-Fender | madduck: Right now you've told us NOTHING about your environment |
| 13:41.00 | madduck | and suddenly the other side picks up. so it's all working. there is just no audio feedback of the other side ringing. |
| 13:41.25 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:41.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:41.50 | madduck | [TK]D-Fender: I am using a SIP provider for outgoing calls, so the sequence is S685IP → asterisk → provider → PSTN |
| 13:42.40 | ZeXr0 | I'm also looking for information on SayDateTime and SayNumber. How does it work ? I need it to work both in french and english. Is there something I need to do in order for it to work in both language ? |
| 13:42.53 | [TK]D-Fender | ZeXr0: No. |
| 13:43.14 | [TK]D-Fender | ZeXr0: * has language based syntax for french as well |
| 13:43.36 | [TK]D-Fender | ZeXr0: If the channel language is french so will the output |
| 13:43.50 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 13:43.54 | [TK]D-Fender | ZeXr0: Assuming of course you installed the french sound pack |
| 13:44.03 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 13:44.55 | mtryfoss | anyone got an idea of why this happens: chan_dahdi.c: Can't fix up channel from 71 to 72 because 72 is already in use ? |
| 13:45.11 | ZeXr0 | [TK]D-Fender : Thank you :) |
| 13:45.27 | tzafrir_laptop | mtryfoss, incoming call? |
| 13:45.31 | [TK]D-Fender | ZeXr0: bienvenue |
| 13:45.40 | mtryfoss | seems to be outgoing. |
| 13:45.44 | ZeXr0 | ;) |
| 13:45.55 | mtryfoss | outgoing callcenter generating a lot of calls |
| 13:46.47 | maldous | what's the 'easiest' way to have a sip register asterisk server answer a call and play something? |
| 13:46.52 | maldous | err, sip registered. |
| 13:47.30 | *** join/#asterisk jhirley_ (~jhirley@64.134.176.191) |
| 13:47.33 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:48.12 | [TK]D-Fender | maldous: make that SIP peer to auth the incoming call. Point it to a context in your dialplan. Make 2 lines of dialplan to match the exten theya re dialing in. Answer. Playback. The End |
| 13:48.19 | maldous | answer appears to be "google for asterisk answering machine" |
| 13:49.05 | [TK]D-Fender | maldous: Gte ready for "dead-end" Land |
| 13:49.18 | [TK]D-Fender | maldous: Poor approach |
| 13:49.34 | maldous | plgggh. ;) |
| 13:50.59 | maldous | ; By default we include the demo. In a production system, you |
| 13:51.03 | maldous | hmm |
| 13:53.34 | madduck | [TK]D-Fender: I realise I should provide you with my configuration and logs and all, but I am actually here right now to understand the technical side of the tone you hear when the other side is ringing or busy |
| 13:53.53 | [TK]D-Fender | maldous: Btw, the sample configs are for "flavor" value only. They are not usable as is and only help serve as inspriation in non-consumable little nuggets |
| 13:53.54 | madduck | because it isn't plain audio (inband), it seems, or else it should just work, given that the phone works |
| 13:53.59 | madduck | i mean voice works |
| 13:54.21 | maldous | gotcha. |
| 13:54.35 | maldous | i'm at "type=friend" probably needs to go somewhere. |
| 13:54.41 | [TK]D-Fender | madduck: You didn't even say precisely WHAT you should be hearing this tone ON. Or which tones even. There is no sense of flow in your description |
| 13:55.15 | ZeXr0 | You should try to reexplain that with more detail, and clearly :P |
| 13:55.24 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 13:55.29 | madduck | [TK]D-Fender: when I pick up the phone, dial a number, and press 'dial', I usually hear a tone indicating that the other side is ringing or busy in my earpiece |
| 13:55.35 | madduck | i don't know how else to describe this |
| 13:55.42 | madduck | i want to know what makes this tone happen |
| 13:55.48 | madduck | and what factors are involved. |
| 13:55.56 | [TK]D-Fender | madduck: pastebin a failed call with SIP DEBUG enabled |
| 13:56.27 | madduck | the call does not fail. at some point in time, the other side picks up, and it all works. i just don't hear any indication that it's ringing on the other side. |
| 13:56.31 | madduck | but sec, will do |
| 13:56.40 | maldous | i'm missing something here. i'm not going to have clients registering against me. i just want to answer all calls to a context. |
| 13:57.02 | ZeXr0 | [TK]D-Fender : Isn't the tone you ear is generated by asterisk in the config file ? |
| 13:57.40 | [TK]D-Fender | ZeXr0: such a quaint littel assumption :) |
| 13:57.45 | idespinner | Depends on the tone, if it is 'dial tone' no, if it is the ringback.... no |
| 13:58.01 | [TK]D-Fender | idespinner: latter = MAYBE |
| 13:58.13 | idespinner | although ringback generated by a sip phone needs to get a SIP progress message usually |
| 13:58.18 | idespinner | before the phoen generates it |
| 13:58.26 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 13:58.26 | idespinner | so [TK]D-Fender its kinda a technicality |
| 13:58.42 | [TK]D-Fender | idespinner: Perhaps a required one on his part |
| 13:59.43 | madduck | [TK]D-Fender: http://slexy.org/view/s2EM7rLLrG is the debug log of the s685ip calling the n900, both directly connected to the same asterisk (which is outside of my NAT network where both phones reside). While the n900 is ringing, I hear nothing in the earpiece of the s685ip |
| 13:59.58 | *** join/#asterisk frawd (~francois@83.39.207.254) |
| 14:00.04 | *** part/#asterisk frawd (~francois@83.39.207.254) |
| 14:00.27 | [TK]D-Fender | madduck: COMPLETE call. End to end. pastebin.com please |
| 14:01.05 | maldous | i see "context=default" in sip.conf and "[default] include => demo" in extensions.conf |
| 14:01.19 | maldous | what am i missing here - i'd assume that would automagically answer. |
| 14:01.37 | [TK]D-Fender | maldous: "context=default" in sip.conf <-- WHERE this occurs is important |
| 14:01.54 | madduck | [TK]D-Fender: that is a complete call. |
| 14:01.56 | madduck | http://pastebin.com/8TXi1V0A |
| 14:02.01 | [TK]D-Fender | maldous: Putting gas in your car helps... but not if its in the WINDSHIELD WASHER CONTAINER |
| 14:02.11 | maldous | [tk] - as per the default distribution. |
| 14:02.11 | [TK]D-Fender | :D |
| 14:02.14 | maldous | ;) |
| 14:02.24 | [TK]D-Fender | maldous: Told you... those samples are GARBAGE. Trash them |
| 14:02.34 | [TK]D-Fender | maldous: and WHERE = IMPORTANT |
| 14:02.37 | maldous | blah. |
| 14:03.00 | maldous | well, gosh. this is easy. |
| 14:03.14 | [TK]D-Fender | maldous: keep only the [general] section & your register for sip.conf. Trash extensions.conf in its entirety |
| 14:03.45 | [TK]D-Fender | maldous: It is easy. It gets easier when you aren't distracted by sample crap you hope means something. |
| 14:03.52 | maldous | ah. |
| 14:04.02 | [TK]D-Fender | maldous: You need about 4 lines of extensions.conf to do this. |
| 14:04.24 | madduck | [TK]D-Fender: that SIP/2.0 180 Ringing should be what I am looking for, right? |
| 14:05.06 | [TK]D-Fender | madduck: in each direction |
| 14:06.26 | madduck | [TK]D-Fender: that is not what http://tools.ietf.org/html/rfc3665#section-3.1 suggests |
| 14:06.41 | madduck | i've *never* seen TRYING/RINGING go from the source to the destination. |
| 14:07.35 | [TK]D-Fender | madduck: there are *2* calls going on here. |
| 14:07.43 | [TK]D-Fender | madduck: Go caffienate. |
| 14:08.20 | maldous | chan_sip.c: Call from '' to extension 'XXXX' rejected because extension not found. |
| 14:08.24 | madduck | as a matter of fact, i *am* caffeinating |
| 14:08.29 | [TK]D-Fender | madduck: and pastebin.... THE ENTIRE FUCKING CALL. Details will KILL you in this game :) |
| 14:08.43 | maldous | i blew away extensions.conf AND [TK]D-Fender BROKE MY ASTERISK?!@#? OMG. |
| 14:08.59 | maldous | so, something in there must be a default. |
| 14:09.00 | [TK]D-Fender | maldous: progress. What XXX is it looking for? |
| 14:09.10 | madduck | [TK]D-Fender: sip set debug peer n900, start paste, call n900 from s685ip, pick up, talk a bit, hang up, stop paste? |
| 14:09.13 | [TK]D-Fender | maldous: There is no such thing as "default" in * |
| 14:09.18 | maldous | my userid |
| 14:09.22 | [TK]D-Fender | maldous: Welcome to "absolute control" |
| 14:09.25 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:09.39 | [TK]D-Fender | maldous: pastebin your sip.conf masking only passwords |
| 14:10.09 | maldous | my register and "context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes" |
| 14:10.23 | *** join/#asterisk carl0s- (~carl@host81-149-50-32.in-addr.btopenworld.com) |
| 14:10.38 | ijpalmer | hi does anyone know why MixMonitor would stop recording when I complete a transfer on an outbound call, I'm using AUDIOHOOK_INHERIT=yes. MixMonitor doesn't stop recording when inbound transferred calls are completed |
| 14:11.05 | *** join/#asterisk HerdberH (~Kilian@p4FD63AE9.dip.t-dialin.net) |
| 14:12.07 | carl0s- | I've been thinking.. what's the best way to utilise two separate ADSL lines to increase outbound call capacity? The SIP provider only gives 1 IP address and set of credentials. Should I: Use a NAT router and do 1:1 NAT from a private IP to the SIP Provider, so that my Asterisk box thinks it's onto a provider at 192.168.1.50 when it's actually onto the SIP provider at 222.222.222.222 (e.g.), or should I use OpenSER to proxy it? |
| 14:12.27 | carl0s- | (or none of the above?) |
| 14:12.49 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 14:12.57 | MAbbas | [TK]D-Fender Am I missing something in attended call trasnfer. Dialplan and console output http://pastebin.com/QE33pzic. I am following attended call transfer steps mentioned on http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 14:13.01 | madduck | [TK]D-Fender: yes, two calls, since both phones are connected to the same * |
| 14:13.03 | Chainsaw | carl0s-: Why are you trying to fool your Asterisk installation? |
| 14:13.07 | Chainsaw | carl0s-: Work with it, not against it. |
| 14:13.12 | madduck | anyway, i will try to find an external number to use instead. |
| 14:13.23 | [TK]D-Fender | maldous: then you are accepting the call completely un-authed. Not healthy, but we'll start with this |
| 14:14.15 | carl0s- | Chainsaw, well, the provider has a single IP. I don't see how I can get the linux system to make multiple connections to a single IP address over two different gateways, without using a proxy, or 1:1 NAT or something .. ? |
| 14:14.21 | *** part/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 14:14.23 | [TK]D-Fender | maldous: your XXX : in extensions.conf add [general] [default] and in [default] do : exten => XXXX,1,Answer exten => XXXX,2,Playback(tt-monkeys) |
| 14:14.38 | MAbbas | [TK]D-Fender why my dialplan exten => 0202,1,Dial(Agent/202,,tT). Returning AT-timeout?? |
| 14:14.41 | [TK]D-Fender | maldous: substituting the XXX for the precise thing they dial in |
| 14:14.51 | [TK]D-Fender | MAbbas: do you hear it dial? |
| 14:15.05 | [TK]D-Fender | MAbbas: I see ti start. |
| 14:15.31 | MAbbas | [TK]D-Fender yes I hear it dialing |
| 14:15.47 | pabelanger | does 'dialplan add extension' create the context if it does not exist? |
| 14:15.58 | [TK]D-Fender | pabelanger: * CLI? |
| 14:16.05 | pabelanger | [TK]D-Fender: yar |
| 14:16.06 | [TK]D-Fender | MAbbas: then they aren't answering |
| 14:16.22 | pabelanger | I don't believe so, just confirming |
| 14:16.26 | pabelanger | m17431 |
| 14:16.28 | MuffinMan | [new] [Asterisk] General 0017431: dialplan add failing to create context if context doesn't exist reported by leearcher https://issues.asterisk.org/view.php?id=17431 |
| 14:16.38 | MAbbas | <PROTECTED> |
| 14:17.17 | MAbbas | <PROTECTED> |
| 14:17.24 | [TK]D-Fender | MAbbas: Maybe you should show my something complete to look at and go look. |
| 14:17.53 | madduck | [TK]D-Fender: i identified the culprit: the tone echoes just fine unless i use my Jabra BT500v headset to initiate the call on the s685ip. i'll stop here. |
| 14:19.24 | maldous | yay! i have monkeys. |
| 14:19.31 | carl0s- | So, any ideas? |
| 14:19.52 | MAbbas | <PROTECTED> |
| 14:20.58 | MAbbas | <PROTECTED> |
| 14:23.04 | MAbbas | <PROTECTED> |
| 14:24.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:24.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:24.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:27.13 | [TK]D-Fender | maldous: Ok, no go grab the book and start getting a grasp on how the dialplan works. |
| 14:28.40 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:33.02 | carl0s- | What's the best way to utilise two separate ADSL lines to increase outbound call capacity when the SIP provider only gives one public IP address (thus routing over multiple gateways tough). Should I: Use a NAT router and do 1:1 NAT from a private IP to the SIP Provider, so that my Asterisk box thinks it's onto a provider at 192.168.1.50 when it's actually onto the SIP provider at 222.222.222.222 (e.g.), or should I use OpenSER to |
| 14:33.02 | carl0s- | <PROTECTED> |
| 14:33.46 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 14:36.43 | maldous | thx. working what i needed now. "sorry voicemail is full, followed by, push button and play monkeys" |
| 14:36.51 | [TK]D-Fender | MAbbas: What happens when you use the native SIP attended transfer? |
| 14:36.54 | maldous | that will give me some time to get work done tomorrow. yay. |
| 14:37.10 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 14:37.31 | [TK]D-Fender | CarlYes, proxy it. * will not multi-home well for this |
| 14:39.30 | [TK]D-Fender | carl0s-: Yes, proxy it. * will not multi-home well for this |
| 14:40.27 | *** part/#asterisk HerdberH (~Kilian@p4FD63AE9.dip.t-dialin.net) |
| 14:40.30 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:41.27 | *** part/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:41.54 | carl0s- | [TK]D-Fender, That a job for OpenSER then? (I've never used it before..) |
| 14:42.05 | [TK]D-Fender | carl0s-: Sure |
| 14:42.13 | carl0s- | OK. I'll give it a whirl ;) |
| 14:42.36 | carl0s- | I might have the same problem getting the OpenSER box to make multiple connections to the same IP address over different gateways/routers though ... |
| 14:42.43 | p3nguin | What other proxy choices are there? |
| 14:42.44 | carl0s- | (hence the 1:1 NAT idea) |
| 14:43.27 | carl0s- | p3nguin, dunno.. |
| 14:43.51 | [TK]D-Fender | carl0s-: alternate idea: run 2 *'s and have one bound to a single gatway, and one to the other. then have A pass of to B alternately. |
| 14:44.04 | *** join/#asterisk traxx (~traxx@91-64-51-178-dynip.superkabel.de) |
| 14:44.48 | carl0s- | [TK]D-Fender, hmm. oh yeah. good idea. I could buy a load of Atcom IP01s for the job.. |
| 14:45.04 | maldous | is there a sample to play that sounds like a dialing tone? |
| 14:45.20 | [TK]D-Fender | maldous: Yes. Depends what you want to do in the bigger picture though |
| 14:45.23 | maldous | answer() appears to be like sleep. i want the calling peer to hear ringing. |
| 14:45.30 | maldous | err, dialing. |
| 14:45.46 | p3nguin | By the way, opensips is the current name of the old openser. |
| 14:46.14 | carl0s- | p3nguin, yeah.. or kamailio or something. |
| 14:46.28 | p3nguin | siproxd is also a SIP proxy. |
| 14:46.36 | *** join/#asterisk flapjacks (~flapjacks@wsip-72-214-208-206.ph.ph.cox.net) |
| 14:46.41 | maldous | progress? |
| 14:47.15 | p3nguin | maldous: Answer() brings the channel "Up" |
| 14:47.28 | p3nguin | maldous: It doesn't sleep, pause, wait, or anything else. |
| 14:47.49 | flapjacks | im looking to provide a T1 interface to one of our off site legacy pbx's so they can use are sip services on our asterisk pbx. Any product recomendations? |
| 14:47.54 | maldous | i'm not hearing a ring tone when i dial. |
| 14:48.08 | maldous | err, after i've dialled, waiting for answer. |
| 14:48.24 | *** join/#asterisk gloin (me@unaffiliated/gloin) |
| 14:48.26 | p3nguin | maldous: You hear nothing at all until the far end picks up the phone? |
| 14:48.44 | p3nguin | And where's your dialplan paste? |
| 14:49.37 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
| 14:50.14 | [TK]D-Fender | [10:45]<maldous>err, dialing.<- how long? Then what. work on describing the complete flow |
| 14:50.26 | [TK]D-Fender | maldous: Otherwise we'll give you a command the rings them indefinitely |
| 14:50.28 | carl0s- | [TK]D-Fender, I suppose thinking about it, if I was to use OpenSER, then perhaps I should be looking at 1 OpenSER box per connection as well.. that would simplify things. So it's a choice of 1x OpenSER box per connection/router, or 1xAsterisk box per connection/router. Then I don't have any gateway issues. |
| 14:52.02 | maldous | !v |
| 14:52.16 | krion | hi |
| 14:52.33 | maldous | weird. |
| 14:52.44 | krion | i'm running asterisk 1.6.2.6, and from time to time, the server doesn't accept call anymore |
| 14:52.46 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:52.59 | krion | i see nothing on the console (asterisk -r) or in the log |
| 14:53.10 | krion | even with DEBUG 9 enabled |
| 14:53.15 | maldous | answer(5) gives 5 seconds of silence and then asterisk answer. answer(6) gives 6 seconds, ring tone, and polycom or outlook then kicks in. |
| 14:53.26 | krion | anyone've seen such behaviour ? |
| 14:53.27 | maldous | my sip provider isn't giving a ringtone for the first five seconds. |
| 14:54.08 | p3nguin | Stop providing a number as data to the Answer() application. |
| 14:54.43 | maldous | err, I meant wait(5), wait(6) |
| 14:55.14 | leifmadsen | maldous: that kind of makes sense... |
| 14:55.43 | *** join/#asterisk Seb^ (~sebspiers@188.39.20.226) |
| 14:56.03 | maldous | ok, maybe i got things confused earlier. during the wait(5), I'd like a ringtone to be played. |
| 14:56.14 | p3nguin | Why wait at all? |
| 14:56.24 | leifmadsen | ya... |
| 14:56.40 | leifmadsen | is the incoming channel SIP or... ? |
| 14:56.41 | maldous | it's too fast right now.. |
| 14:56.48 | leifmadsen | too... fast? |
| 14:56.51 | maldous | combination of both pots and sip |
| 14:57.10 | leifmadsen | I think you need to show the dialplan with a description of what you're trying to do |
| 14:57.13 | maldous | yeah. not enough time between pressing dial on the mobile and getting the phone to my ear. |
| 14:57.37 | p3nguin | The person on the other end can't pick up his phone THAT fast. |
| 14:58.23 | maldous | nothing hard here. http://pastebin.com/X67SzXGd |
| 14:58.39 | maldous | nono. i'm calling from a sip client to asterisk. |
| 14:58.58 | maldous | when i call asterisk, asterisk answering too quickly. need a small delay. want sound to played during delay. |
| 14:59.16 | p3nguin | What happens once asterisk answers? |
| 14:59.20 | maldous | it works. |
| 14:59.32 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 14:59.38 | maldous | the wait(10) gives 10 seconds of nothing, unanswered. |
| 14:59.49 | p3nguin | What _happens_ once asterisk answers? |
| 14:59.57 | maldous | exten => XXXXXXXX,n,BackGround(vm-mailboxfull) |
| 15:00.01 | *** join/#asterisk egyDev (~egyDev@fedora/ahmed-araby) |
| 15:00.07 | maldous | (ie, the sample is played) |
| 15:00.29 | maldous | i'm going to presume my sip provider isn't receiving RINGING sensibly. |
| 15:00.38 | maldous | client-sip-sip-asterisk |
| 15:00.44 | leifmadsen | do you actually send ringing from Asterisk to the provider? |
| 15:00.49 | leifmadsen | if you're just waiting, you're doing nothing |
| 15:00.51 | p3nguin | Okay, use Ringing() then Wait(5). |
| 15:00.56 | maldous | ah. |
| 15:00.57 | leifmadsen | you might want something like a Progress() first or Ringing() |
| 15:01.06 | leifmadsen | high fives p3nguin |
| 15:01.07 | egyDev | Hi,I'm trying to use Asterisk Builtin mini-HTTP server, But I always get 'Attempt to use unimplemented / unsupported method' |
| 15:01.25 | leifmadsen | resists the urge to say, "you sure do" |
| 15:02.03 | maldous | perfect. that works. |
| 15:02.46 | maldous | egyDev - it's not actively maintained, apparently doesn't work, and what you've got there is possible the application interface, not the user interface. short answer is "it's not meant to be used as a GUI" |
| 15:03.13 | leifmadsen | the built in HTTP server is actively maintained |
| 15:03.17 | leifmadsen | the GUI itself is not |
| 15:03.23 | carl0s- | I wonder if OpenSER handles the RTP packets as well ? |
| 15:03.30 | leifmadsen | (unless you are a commercial customer, such as those using AA50 etc) |
| 15:03.42 | leifmadsen | carl0s-: it can if you use the rtpproxy |
| 15:04.30 | carl0s- | Sweet. because I think I need that really. I am going to buy another Blackfin thing and have a play. |
| 15:05.02 | leifmadsen | but OpenSER/OpenSIPS etc... are SIP proxies |
| 15:05.12 | leifmadsen | generally |
| 15:05.20 | leifmadsen | (which means just SIP signalling) |
| 15:05.31 | leifmadsen | but like I said, rtpproxy can be used to handle the RTP too if necessary |
| 15:05.59 | egyDev | Well,this is the end of my day and I can hardly see words :),All what I want a nice gui to configure Asterisk ? |
| 15:06.15 | carl0s- | leifmadsen, I am trying to utilise multiple Internet connections on a single Asterisk PBX box, so I think OpenSER/OpenSIPS/whatever is the way forward, with both SIP and RTP proxying. Would you agree ? |
| 15:06.21 | p3nguin | maldous: You can tune the time in the Wait() command for your comfort. A normal US ring is 2 seconds on, 4 seconds off... so Wait(5) is going to let you hear one ring and three seconds of silence before your sound file plays. |
| 15:06.46 | leifmadsen | egyDev: heh, well then you might want to just install AsteriskNOW or something like that |
| 15:07.23 | egyDev | leifmadsen, But AsteriskNOW is complete distribution I think , right ? |
| 15:07.32 | leifmadsen | carl0s-: well depends what you mean by that -- that sounds more like a networking problem that should be done below the application level, but Asterisk can listen to multiple interfaces |
| 15:07.36 | leifmadsen | egyDev: yes |
| 15:07.48 | leifmadsen | egyDev: it installs CentOS for you along with Asterisk and the GUI |
| 15:07.56 | p3nguin | I might go with Wait(7) so I hear one second of the second ring sound. |
| 15:08.07 | carl0s- | leifmadsen, I want to increase outgoing call capacity to a single provider with a single IP address. |
| 15:08.29 | carl0s- | I haven't actually asked the provider if they have another/alias IP address as it happens. They might. |
| 15:08.39 | egyDev | leifmadsen,well you know I can't simply reinstall every thing , It's not that easy :) |
| 15:09.00 | leifmadsen | carl0s-: ya -- asterisk will always respond on the primary interface, but will listen to all interfaces (fyi) |
| 15:09.34 | leifmadsen | you might be able to use OpenSIPS along with a custom SIP header to redirect to different interfaces, but I'm not sure how OpenSIPS, etc... handle interfaces |
| 15:10.00 | egyDev | If there's a link or doc or how about that gui or how to, to use later I'll be grateful |
| 15:12.06 | carl0s- | leifmadsen, I think I want to just tell my main Asterisk box to use each OpenSER as an independent trunk. Or maybe as [TK]D-Fender suggested I just run Asterisk instead of OpenSER on these gateway/proxy boxes |
| 15:12.22 | leifmadsen | egyDev: just google for 'asterisk gui' -- should give you lots of links |
| 15:12.41 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 15:12.52 | egyDev | leifmadsen, Thanks |
| 15:13.07 | leifmadsen | carl0s-: depends how you want the networking to work I guess :) |
| 15:13.29 | leifmadsen | carl0s-: I prefer to use the least number of tools to solve the problem as possible though to keep complexity to a minimum |
| 15:14.18 | carl0s- | leifmadsen, well, I would be much happier if the provider gave a second IP address, then I could just create a static route on the Asterisk boxes via eth2 (additional ADSL line). but they only give one IP address. |
| 15:14.46 | leifmadsen | carl0s-: perhaps you can better explain how the network is setup and what you want to do |
| 15:14.55 | leifmadsen | carl0s-: are you talking networking provider, or ITSP provider? |
| 15:15.21 | carl0s- | leifmadsen, ITSP/SIP provider |
| 15:15.29 | leifmadsen | carl0s-: oh, then just create multiple SIP accounts |
| 15:15.33 | leifmadsen | SIP/itsp-1 |
| 15:15.36 | leifmadsen | SIP/itsp-2 |
| 15:15.59 | leifmadsen | in sip.conf just setup [itsp-1] and [itsp-2] |
| 15:16.03 | carl0s- | leifmadsen, but there is just 1 IP address for the provider.. so it's hard to get the Asterisk box to use more than one Internet connection/router. |
| 15:16.17 | [TK]D-Fender | leifmadsen: No, he needs to use a different source IP. separate peers won't change *'s routing |
| 15:16.26 | carl0s- | which is why I'm looking at OpenSER or Asterisk proxies |
| 15:16.39 | [TK]D-Fender | leifmadsen: mult-homes, not "spread targets" |
| 15:16.42 | leifmadsen | [TK]D-Fender: ah gotcha... then ya -- he can't do that with Asterisk |
| 15:16.50 | [TK]D-Fender | leifmadsen: Yes... with multiple :) |
| 15:16.50 | leifmadsen | his description was confusing |
| 15:16.55 | leifmadsen | well ya |
| 15:17.04 | [TK]D-Fender | :) |
| 15:17.05 | leifmadsen | multiple instances binded to a single IP address |
| 15:17.29 | [TK]D-Fender | leifmadsen: Yup, I suggested that VS running multiple SER's. He know's * already, and only need enough to pass through |
| 15:17.31 | carl0s- | I will first ask if they have an additional IP address, else I will play with OpenSER on Blackfin boxes. I think 1x Blackfin per Internet connection will be most straightforward. |
| 15:17.35 | leifmadsen | I don't know if OpenSIPS helps with this or not -- I don't know enough about OpenSIPS to know if you can force certain connections out physical ports |
| 15:17.46 | leifmadsen | ya, I'd rather run multiple asterisk instances |
| 15:17.56 | leifmadsen | ok, back to making releases |
| 15:19.00 | carl0s- | For some reason I hadn't thought of 1x proxy (Asterisk instance, or OpenSER instance) per connection. I was trying to cram multiple connections onto a single instance. Thanks for the ideas and help. |
| 15:24.37 | carl0s- | Hmm. If I could get a cheapo ADSL2+ VoIP router that can be used as a trunk from Asterisk, well that'd sure be neat! |
| 15:24.44 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 15:25.45 | [TK]D-Fender | carl0s-: there are many |
| 15:26.39 | carl0s- | [TK]D-Fender, yeah, but wifi not needed/wanted, and Draytek are slightly pricey. It just would be a bit cleaner than having [ADSL Router + Blackfin Asterisk Proxy] x 2 --> Asterisk PBX |
| 15:27.23 | *** join/#asterisk BANSAL (~bansal@117.207.83.40) |
| 15:27.54 | carl0s- | I think there will be too many firmware/configuration limitations. Much better to have full Asterisk on an Atcom IP01, bound to its own ADSL router. Just lots of plugs and wires that's all. |
| 15:28.15 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 15:31.54 | [TK]D-Fender | carl0s-: I might jsut run VM as your other instances would only be routing packets |
| 15:32.13 | [TK]D-Fender | carl0s-: Run your primary as the host of course to give it the real power |
| 15:32.38 | carl0s- | [TK]D-Fender, Hmm. Yeah that's an idea. Thanks :) |
| 15:33.09 | [TK]D-Fender | carl0s-: I think basic KVM or whatever that other low-end one is should do. |
| 15:34.01 | carl0s- | KVM's great but it can't keep the guest clocks in sync with the host! VMWare ESXi (free) works well for my server whatsits here |
| 15:34.24 | carl0s- | or rather, on my desktop it doesn't keep the time synced anyway ... |
| 15:34.49 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 15:36.11 | [TK]D-Fender | carl0s-: Keep in mind I know EXTREMELY little in this dept. I just see pieces sitting around and think "Hey yeah... I'm pretty sure these bits can go together" |
| 15:37.49 | *** join/#asterisk thansen (~thansen@c-67-177-0-194.hsd1.ut.comcast.net) |
| 15:38.54 | thansen | do nortel system support SIP? or is it possible to make asterisk a client to a nortel system over IP? |
| 15:40.40 | [TK]D-Fender | thansen: Depends which obviously |
| 15:41.17 | thansen | [TK]D-Fender: ok, I haven't been told the model number yet so I'll get back with more info..thanks! |
| 15:47.39 | *** join/#asterisk grungies1138 (~grungies1@cpe-66-74-197-34.san.res.rr.com) |
| 15:48.01 | *** join/#asterisk hfb (~hfb@pool-98-112-109-195.lsanca.dsl-w.verizon.net) |
| 15:49.25 | grungies1138 | I'm very familiar with an Avaya phone system and on it, I am able to build an auto-attendant that allows me to have options (press 1 for XXX press 2 for XXX) and then send that call to a 'hunt group'. I am also able to send a 'caller ID' of the group name so I can identify the call as say a Sales or Support call. Can I do something similar in Asterisk? (using straight Asterisk. no GUI) |
| 15:51.05 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:54.30 | [TK]D-Fender | grungies1138: Yes. And far more |
| 15:55.01 | rjk` | does anyone know a free asterisk to skype connector ? |
| 15:55.02 | [TK]D-Fender | grungies1138: Call from Indiana, its rainging in NYC and the Mets lost their last home game? Send call to YYY |
| 15:55.13 | rjk` | other than the digium one which is $66 |
| 15:55.32 | [TK]D-Fender | rjk`: Why are you so centered on using Skype for PSTN in your solution? |
| 15:55.51 | rjk` | because i have every component in place now to do it on the mac |
| 15:56.00 | grungies1138 | Let me be more specific... I want to send some indication to the phone of what kind of call it is. |
| 15:56.08 | rjk` | found macports had the latest asterisk |
| 15:56.13 | *** join/#asterisk Z_God (~julius@wlan232185.mobiel.utwente.nl) |
| 15:56.27 | ijpalmer | hi does anyone know why MixMonitor would stop recording when I complete a transfer onan outbound call, I'm using AUDIOHOOK_INHERIT=yes. MixMonitor doesn't stop recording when inbound transferred calls are completed |
| 15:56.31 | rjk` | also found a good blind tty text to speech generator |
| 15:56.44 | [TK]D-Fender | grungies1138: Sure, distintive ringing, selective call appearance, manipluated caller ID, extra headers (phone must support), etc |
| 15:57.14 | [TK]D-Fender | rjk`: I fail to see where Skype for PSTN figures into this. |
| 15:57.29 | [TK]D-Fender | rjk`: Your other solution isn't * so that's already out of the picture |
| 15:57.32 | rjk` | voip providers for au are crap |
| 15:57.56 | [TK]D-Fender | rjk`: http://www.google.ca/#hl=en&source=hp&q=free+skype+to+sip+gateway&aq=0&aqi=g2&aql=&oq=free+skype+to+SIP&gs_rfai=&fp=290a43085aba2861 |
| 15:57.58 | grungies1138 | So if I use, say, a SNOM or Polycom phone I can send a unique caller ID from my incoming Dial plan for a particular selector code? |
| 15:58.02 | [TK]D-Fender | rjk`: JFIG <- |
| 15:58.34 | [TK]D-Fender | grungies1138: Sure, you might for instance prefix the name. So "John Smith" will be set as "Sales:John Smith" |
| 15:58.41 | [TK]D-Fender | JFGI* |
| 15:58.44 | [TK]D-Fender | dangit |
| 15:59.06 | grungies1138 | Are you aware if Softphones will accept it? Just curious |
| 15:59.13 | rjk` | lol |
| 15:59.20 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:59.34 | grungies1138 | I'm using X-lite as my test phone while I learn |
| 16:00.07 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 16:00.44 | bent_screwdriver | anyone know good source for NPA/NXX listing? |
| 16:01.17 | [TK]D-Fender | grungies1138: You can change the CID any tiem. Many also accet distintive rings |
| 16:01.29 | [TK]D-Fender | grungies1138: CID si the "freebit" though. That "just works" |
| 16:02.14 | [TK]D-Fender | frebie* |
| 16:02.17 | [TK]D-Fender | GAH |
| 16:04.19 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 16:05.10 | grungies1138 | gotcha. And here's the newbie question: which application in the dialplan should I use for that? |
| 16:06.12 | [TK]D-Fender | grungies1138: "core show application set" , "core show function CALLERID" |
| 16:06.12 | grungies1138 | thanks. I may ask stupid questions, but the good thing is I usually only have to ask them once. :P |
| 16:10.26 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 16:14.46 | iscario | Hi, |
| 16:14.46 | iscario | I set up an asterisk server that i use with iax accounts. |
| 16:14.46 | iscario | Everything is working fine, but for personal reason i need to insert a proxy on my server which role is to encrypt data. |
| 16:14.46 | iscario | Here, I encountered what i suppose to be a timeout prb. I can register with my iax account, but i can not handle a call... |
| 16:14.46 | iscario | In the log, asterisk gave me (among other...) an error cause 16... |
| 16:14.47 | iscario | Does it mean anything to someone ? |
| 16:14.49 | iscario | Thx! |
| 16:16.46 | leifmadsen | not enough information provided |
| 16:18.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:18.56 | iscario | what do you need ? |
| 16:19.10 | iscario | i do not have the log report here... |
| 16:20.21 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:20.51 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:22.10 | iscario | where am I supposed to look at leifmadsen ? |
| 16:22.18 | Knightfal | What would cause hold music to start playing on an active call, it seems very random and to only play for about 5 seconds. Any Ideas? |
| 16:22.35 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:22.48 | leifmadsen | Knightfal: look at the SIP trace (presuming you're using SIP channels) |
| 16:22.54 | Chainsaw | Knightfal: Other party playing with hold button out of boredom? |
| 16:23.11 | Chainsaw | Knightfal: Hard to say without a trace of it happening. core set debug 10, core set verbose 10, wait for the issue to reoccur. |
| 16:23.13 | leifmadsen | iscario: without logging or console output, it will be impossible to know what is the cause of the hangup |
| 16:23.28 | Knightfal | ok thx guys |
| 16:23.40 | leifmadsen | ~debug |
| 16:23.41 | infobot | ACTION DeBuggers $1 |
| 16:23.41 | Chainsaw | Knightfal: You can leave it in this logging mode and observe /var/log/asterisk/full later (or whatever you configured it to). |
| 16:23.48 | Knightfal | Its PSTN-> sip exten |
| 16:23.55 | leifmadsen | ~debug-info |
| 16:23.59 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:24.02 | Knightfal | sorry t1 to sip exten |
| 16:24.42 | leifmadsen | infobot: debug-info is In order to debug your problem please read the How-To document on collecting debug information here: http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:24.43 | infobot | leifmadsen: okay |
| 16:25.01 | leifmadsen | [TK]D-Fender: see above for future reference |
| 16:26.04 | iscario | leifmadsen: i noticed two informations : code cause 16 and in the end called timeout. The prb just comes from delay of the encryption because, if i comment the encryption/decryption on my home made proxy, it is working. I can't do anything with these information ? |
| 16:26.29 | leifmadsen | I can't help you with that information other than it appears to be a networking or configuration problem |
| 16:27.03 | Knightfal | It seems to be when a page goes out |
| 16:27.17 | Knightfal | all calls goto hold that are in the page grp |
| 16:28.02 | leifmadsen | in that case when channels place or receive calls I'd set a custom GROUP() on the channel and build the Page() dialstring only for channels with less than 1 GROUP_COUNT() |
| 16:28.20 | leifmadsen | or something else equally clever :) |
| 16:30.57 | iscario | ok leifmadsen. to save the log, how can i act (usally i just set iax2 set debug on, and paste it to my text file, but i do not have mouse on my server ^^) ? |
| 16:31.08 | leifmadsen | ~debug-info |
| 16:31.09 | infobot | debug-info is probably In order to debug your problem please read the How-To document on collecting debug information here: http://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 16:32.08 | iscario | thx! |
| 16:41.19 | Trixboxer | hi |
| 16:41.50 | Trixboxer | asterisk disconnecting call after every 11 mins , with broadvoice.. internal calling works fine |
| 16:42.29 | Trixboxer | BYE send from broadvoice message in logs when call gets disconnected |
| 16:46.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:46.12 | [TK]D-Fender | Trixboxer: And we see no debug to comment on |
| 16:48.24 | Trixboxer | [TK]D-Fender: Im catching new logs.. hold on |
| 16:48.50 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:49.34 | grungies1138 | Trixboxer: you using Time Warner Internet? |
| 16:49.38 | Knightfal | @leifmadsen: the s option in Page seems to have done the trick |
| 16:51.35 | Trixboxer | grungies1138: multiple ISP |
| 16:52.01 | grungies1138 | I've seen that exact issue and the problem turned out to be TWC |
| 16:52.14 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:52.24 | Trixboxer | I hope they dont have any such limitation on any specific plan |
| 16:52.41 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.137.237.dsl.dyn.forthnet.gr) |
| 16:53.27 | grungies1138 | I talked with Tier 3 and they said no, but we switched them to a different provider and it worked perfectly. I read some Forums saying the same thing about SIP on TW |
| 16:54.17 | Trixboxer | grungies1138: the problem is that the same trunk is working fine on other asterisk |
| 16:54.34 | grungies1138 | hmm. My issues were not with Asterisk... |
| 16:54.46 | Trixboxer | with same ISP as well |
| 16:58.23 | grungies1138 | So I want to add CALLERID to my DIAL(). What would that look like? |
| 16:58.36 | leifmadsen | you would execute that prior to Dial() |
| 16:59.20 | grungies1138 | oh so Line 1(example:) is CALLERID(Blah,blah) then line 2 is DIAL()? |
| 16:59.39 | leifmadsen | no, CALLERID() is a function |
| 16:59.44 | leifmadsen | Dial() is an application |
| 16:59.56 | leifmadsen | CALLERID() needs to be executed within the Set() application |
| 17:01.02 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 17:01.48 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 17:01.49 | mtryfoss | chan_dahdi.c: < Message Type: RESTART (70) means that the other side has requested a channel reset (based on the direction of the '<') ? |
| 17:02.07 | leifmadsen | mtryfoss: I believe so -- asterisk received |
| 17:02.37 | mtryfoss | strange.. they send that the same second they send CALL PROCEEDING. |
| 17:03.03 | DND | guys anyone know what's this error means? |
| 17:03.06 | DND | Received MDL/TEI managemement message, but configured for mode other than PTMP! |
| 17:03.26 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
| 17:03.43 | Trixboxer | [TK]D-Fender: is there any way to turn off all other logs.. because im not able to extract the exact logs |
| 17:03.54 | Trixboxer | so many calls are going |
| 17:05.17 | *** join/#asterisk d4rkstar (~bruno@93-42-1-91.ip84.fastwebnet.it) |
| 17:08.20 | grungies1138 | @leifmadsen: So if I wanted the callerID but prepended would I set the callerid(num) in a variable then set the variable with the prepended text and then DIAL()? |
| 17:09.00 | [TK]D-Fender | grungies1138: No, you would manipulate the NAME, not the NUMBER |
| 17:09.30 | grungies1138 | but what if I just wanted to send the number plus prepended text? |
| 17:09.57 | [TK]D-Fender | grungies1138: Do the smart thing and do NOT mangle a numebr |
| 17:10.08 | [TK]D-Fender | grungies1138: You won't be able to dial it back, etc |
| 17:10.17 | grungies1138 | oh I see. |
| 17:10.20 | [TK]D-Fender | grungies1138: Stop shoving text into a numeric field. |
| 17:10.47 | [TK]D-Fender | grungies1138: Its people with a midset like this taht made MySQL the mostrosity that it is :) |
| 17:11.02 | [TK]D-Fender | And I'm losing 10% of my keystrokes today.. |
| 17:11.20 | grungies1138 | but if I wanted say SALES: XXX-XXX-XXXX then I could just manipulate the name and change it to just SALES: |
| 17:11.37 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
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| 17:13.48 | grungies1138 | so like SET(CALLERID(name,SALES:))? |
| 17:14.15 | Trixboxer | strange, now call is not getting cut |
| 17:14.18 | [TK]D-Fender | grungies1138: Nope, read up on your "how to set a function" syntax |
| 17:14.24 | Trixboxer | almost more than 15 mins |
| 17:15.23 | grungies1138 | oh, SET(CALLERID(name)=SALES:)? |
| 17:15.40 | *** join/#asterisk vstemen (~vince@ip72-198-66-128.ok.ok.cox.net) |
| 17:16.30 | *** join/#asterisk devoid (~devnull@unaffiliated/devemo) |
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| 17:19.56 | [TK]D-Fender | grungies1138: Getting warner. That replaces the entire callerid, not prefixes. |
| 17:20.01 | [TK]D-Fender | warmer* |
| 17:20.23 | *** join/#asterisk darkskiez_ (~dz@62-50-207-75.client.stsn.net) |
| 17:20.23 | [TK]D-Fender | grungies1138: Now make sure to add the EXISTING value at the end. Time to brush on up REFERENCING functions |
| 17:20.38 | grungies1138 | but if I don't care about the name? |
| 17:20.48 | grungies1138 | will that replace the number as well? |
| 17:22.32 | grungies1138 | and D-Fender, is it really brushing up if you didn't know in the first place? :P |
| 17:25.14 | *** join/#asterisk darkskiez (~dz@62-50-207-121.client.stsn.net) |
| 17:25.55 | grungies1138 | I tested it and it does replace the number. |
| 17:26.22 | [TK]D-Fender | grungies1138: Nope. |
| 17:26.27 | ChannelZ | you mean the name |
| 17:26.42 | grungies1138 | oh I guess it didn't |
| 17:26.45 | *** join/#asterisk sahafeez_ (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 17:27.50 | grungies1138 | say So I'm guessing that I just add the standard caller ID to a variable then in my set I do like Sales: ${Varname}? |
| 17:31.34 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 17:31.44 | AsteriskNoob | does anyone know a way to get asterisk to load configuration from a webpage or XML or PHP or something? |
| 17:32.01 | AsteriskNoob | I'd like to have dynamic conf files if possible |
| 17:32.05 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 17:32.21 | ManxPower | Was the "o" option to Dial replaced by the f(x) - Force the outgoing callerid to 'x' option? |
| 17:32.46 | ChannelZ | AsteriskNoob: google 'realtime asterisk' |
| 17:33.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:33.55 | AsteriskNoob | ChannelZ: Thanks... |
| 17:37.17 | grungies1138 | I got it. Thanks D-F |
| 17:38.40 | grungies1138 | A prince among men |
| 17:38.52 | grungies1138 | or Princess among women as the case may be |
| 17:39.56 | ManxPower | I see my karma is still no good here. |
| 17:40.06 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 17:41.09 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 17:41.11 | wcselby | o/ |
| 17:41.35 | wcselby | so putting " " (quotes) around a CALLERID(name) variable breaks SIP calling.... :( |
| 17:43.27 | leifmadsen | why would you use quotes? |
| 17:45.19 | wcselby | because the callerid variable in SIP.conf sets it "NAME" <NUMBER>, and I just made a bad assumption. I've done it before, is the annoying part |
| 17:45.35 | wcselby | i should know better |
| 17:46.25 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:49.07 | ChannelZ | wcselby: actually you shouldn't do that either |
| 17:49.25 | ChannelZ | wcselby: just use callerid=Foo Manchu <1234567> in sip.conf |
| 17:50.40 | wcselby | ChannelZ - huh, you're probably right |
| 17:51.19 | wcselby | ChannelZ - yeah, that's the way it shows in the sample sip.conf file |
| 17:52.22 | ChannelZ | A long time ago I had quotes which seemingly caused no problems, but occasionally I would get these weird warnings on the console when a certain extension would hang up |
| 17:53.18 | ChannelZ | once I actually investigated it, turned out to be the quotes |
| 17:53.36 | wcselby | yeah |
| 17:53.51 | wcselby | were quotes needed in 1.2? I'm trying to think why I've always added them... |
| 17:54.08 | wcselby | probably just a mistake on my part |
| 17:54.55 | ChannelZ | not sure |
| 17:55.24 | [TK]D-Fender | They are appropriate |
| 17:57.22 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 17:59.28 | ChannelZ | well with them I'd get "Bad request protocol Line 4@192.168.2.2 SIP/2.0", 4 or 5 in a row over the course of a few seconds, when someone hung up |
| 18:02.08 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 18:02.41 | mtryfoss | anyone got an idea of why this happens: chan_dahdi.c: Can't fix up channel from 71 to 72 because 72 is already in use ? |
| 18:03.18 | wcselby | lunch |
| 18:13.05 | tzafrir_laptop | mtryfoss, can you provide a PRI-level trace? |
| 18:13.10 | tzafrir_laptop | pri debug or whatever |
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| 18:17.59 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:24.12 | mtryfoss | yeah. seems to be triggered by an ALERTING message. |
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| 18:26.00 | mtryfoss | tzafrir_laptop: http://pastebin.com/Sf2iY5sY |
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| 18:40.26 | grungies1138 | So I want to create a shared mailbox. I want to have my Autoattentant give options (1-3 for example) Ring a group and if there is no answer then drop into a 'group' mailbox |
| 18:40.35 | p3nguin | maldous: I know this is several hours old, but I just looked at your http://pastebin.com/X67SzXGd paste... Are you literally using XXXXXXXX as the extension? If so, you need to put _ on the front of it to make it a pattern. |
| 18:40.53 | [TK]D-Fender | p3nguin: No, and he has long since solved that part./ |
| 18:41.27 | [TK]D-Fender | grungies1138: So go make a box. Dial the phones. Then fall to VM in that box. |
| 18:41.38 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:41.39 | grungies1138 | oh right. I remember now |
| 18:42.11 | p3nguin | I somehow forgot to read his paste earlier. |
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| 18:47.58 | grungies1138 | So I created a Variable (ExistingCLI) to grab caller ID so I can alter it. Can I make that a global? |
| 18:48.45 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 18:48.47 | grungies1138 | Or does it have to be in the flow? |
| 18:49.00 | [TK]D-Fender | grungies1138: No point. Also you don't need to use a temp var to alter it |
| 18:49.15 | grungies1138 | k |
| 18:50.10 | grungies1138 | wait... I don't need a temp var? |
| 18:50.18 | [TK]D-Fender | grungies1138: Nope |
| 18:50.41 | p3nguin | Not likely. |
| 18:51.25 | grungies1138 | what I currently have is 1,1,SET(ExistingCLI=CALLERID(name)) and 1,n,SET(CALLERID(name)=Sales: ${ExistingCLI}) |
| 18:51.48 | grungies1138 | and it works. |
| 18:52.20 | ChannelZ | ExistingCLI is the same as ${CALLERID(name)} |
| 18:52.43 | p3nguin | Set(CALLERID(name)=Sales:${CALLERID(name)}) |
| 18:52.59 | grungies1138 | ahh! |
| 18:53.01 | [TK]D-Fender | ;) |
| 18:53.02 | grungies1138 | thanks guys |
| 18:53.08 | [TK]D-Fender | grungies1138: You're getting there... |
| 18:54.32 | p3nguin | It works because it doesn't actually change until after that command it processed. |
| 18:54.46 | p3nguin | is processed, rather |
| 18:55.41 | p3nguin | Any subsequent references to ${CALLERID(name)} would then be the new value. |
| 18:55.56 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 18:55.56 | grungies1138 | gotcha |
| 18:56.46 | p3nguin | And CLI means Command Line Interface, so that was a terrible choice for a variable name in the first place. :) |
| 18:57.38 | grungies1138 | Sorry I'm used to the Avaya IP Office and some of it's stuff refers to Caller ID as CLI |
| 18:57.48 | p3nguin | Not CLID? |
| 18:58.02 | grungies1138 | sometimes. they are retarded |
| 18:58.28 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 18:58.42 | grungies1138 | I'm finding that pbxes are like programming languages, the second one is the hardest to learn. :) |
| 19:00.27 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 19:00.34 | lordvadr | Is there any way to get the sip (or IAX2 for that matter) "register => " lines into an odbc database... i.e. with sippeers/iaxpeers in extconfig.conf instead of storing the entire sip.conf in odbc? |
| 19:01.26 | lordvadr | Oh, I just saw the "sipregs" in extconfig.conf. Is there an equivalent iaxregs? |
| 19:02.37 | grungies1138 | in the 'group mailbox' that I'm building, should I do the DIALSTATUS stuff as well? |
| 19:02.41 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 19:03.43 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
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| 19:04.26 | Deeewayne | I just received a text with a picture of a Coleman cooler with the message, "At least Gary Coleman's coffin will have his name on it." |
| 19:05.13 | beek | Deeewayne: Now that is tasteless... |
| 19:07.20 | beek | copies and pastes it into an email. |
| 19:07.59 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:08.53 | [TK]D-Fender | grungies1138: Little poitn. If someone had answered the channel would die when the call does. |
| 19:09.50 | [TK]D-Fender | grungies1138: So the only reason to fall is that it didn't get answerwed. I doubt there is a reason to care why. Also harder to judge ca "collective" reason. |
| 19:11.06 | grungies1138 | I think maybe I should clarify my question. Would it fail to the 'user' mailbox? |
| 19:11.09 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 19:12.09 | [TK]D-Fender | grungies1138: No, it falls to the next line in your dialplan and does precisely what you tell it |
| 19:12.35 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 19:13.00 | grungies1138 | gotcha |
| 19:15.14 | grungies1138 | and to add 'group mailbox' MWI to my extensions I just add a second mailbox entry in sip.conf? |
| 19:16.15 | p3nguin | MWI shouldn't have anything to do with extensions. |
| 19:16.45 | grungies1138 | Message Waiting Indication? |
| 19:17.37 | p3nguin | I'll rephrase. The Message Waiting Indicator shouldn't have anything to do with extensions. |
| 19:17.38 | *** join/#asterisk darksk1ez (~mhb@2001:0:53aa:64c:1447:4b2e:a9e1:1ef7) |
| 19:18.12 | grungies1138 | ok... then how to I tell my users that a 'group message' is waiting? |
| 19:18.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:18.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:18.22 | wcselby | grungies1138 - i use email to do that |
| 19:18.35 | [TK]D-Fender | grungies1138: Do not confuse a SIP DEVICE with an EXTENSION |
| 19:18.46 | p3nguin | If app_voicemail supports it, configure it in voicemail.conf and sip.conf. |
| 19:18.49 | [TK]D-Fender | grungies1138: An extension is jsut a line in your extensions.conf |
| 19:19.10 | grungies1138 | ok... then how do I send MWI to my sip peers? |
| 19:19.33 | [TK]D-Fender | grungies1138: I dial extension 1111 here... and it has NOTHIGN to do with making a "phone" ring, or anything conceptually related to a "user" |
| 19:19.39 | *** join/#asterisk MiserySoft (~LND@109.180.135.125) |
| 19:19.44 | [TK]D-Fender | grungies1138: Yes, jsut add another mailbox line to their sip.conf |
| 19:19.51 | grungies1138 | thank you. :P |
| 19:19.52 | rrb3942 | so you have a line like 'mailbox = 1234@default', right and then your adding another such as 'mailbox = 1234@default,1235@default' |
| 19:19.56 | jblack | grungies1138: I set it up i my polycom config |
| 19:20.40 | wcselby | rrb3942 - i think you add the new one with an &, i.e 'mailbox = 1234@default&1235@default', but I could be wrong |
| 19:21.12 | rrb3942 | oh yeah your right its & |
| 19:22.09 | [TK]D-Fender | or multiple lines. |
| 19:24.30 | rrb3942 | man, that should be you're and it's, sigh...but yes, multiple lines work to |
| 19:24.53 | p3nguin | rrb3942: Thank you. |
| 19:27.31 | beefpastry | Trying to salvage IVR hangups by calling my secretary ring group (via external script->callfile) and then dial(<hungupnumber>)...works fine except that the caller id sent to the hungup number is the ringgroup extension...tried several things to no avail. 1. Is there a more appropriate way to do this, 2. Is there something I can read to figure out the callerid issue? |
| 19:30.43 | grungies1138 | I'm building my first Asterisk system. It's super simple, but I'm looking for a good, compatible wireless SIP phone. Any recommendations? |
| 19:31.40 | rrb3942 | beefpastry: what does your call file look like? |
| 19:32.20 | beefpastry | just a sec... |
| 19:32.30 | [TK]D-Fender | ~wifisip |
| 19:32.31 | infobot | [~wifisip] Please refer to ~wifivoip for info on this. |
| 19:32.40 | [TK]D-Fender | ~wifivoip |
| 19:32.41 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 19:32.44 | [TK]D-Fender | grungies1138: ^^ |
| 19:32.51 | grungies1138 | :nods |
| 19:33.08 | p3nguin | ~a850 |
| 19:33.15 | p3nguin | ~a580 |
| 19:33.16 | infobot | [~A580] The Siemens Gigaset A580 IP with ECO DECT technology is multi-line so you are free to register up to 6 handsets for 6 SIP accounts from different providers and make up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call. See http://gigaset.com/hq/en/product/GIGASETA580IP.html for details. Cost is about $70USD for one handset with base. Extra A58H handsets are around $40USD each. |
| 19:34.24 | [TK]D-Fender | Eww.. |
| 19:34.35 | [TK]D-Fender | p3nguin: Party-lined up almost as bad as Aastra :) |
| 19:35.27 | beefpastry | rrb3942: http://pastebin.com/bsRcivZv |
| 19:35.32 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 19:37.07 | rrb3942 | beefpastry: is 300 your ring group? |
| 19:37.13 | beefpastry | yeah |
| 19:37.50 | beefpastry | rrb3942: http://pastebin.com/jiQUtwKz is context FYI |
| 19:38.14 | rrb3942 | So the callerid is wrong when it rings the ringgroup? |
| 19:38.43 | beefpastry | No...it's right there, the dial to the hungup number shows "300" |
| 19:39.01 | rrb3942 | ah gotcha |
| 19:39.07 | beefpastry | I've tried to Set(CALLERID...) before the dial to no avail |
| 19:39.57 | [TK]D-Fender | beefpastry: Show us. |
| 19:40.10 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 19:40.18 | beefpastry | I've also tried to directly dial the trunk to go around the freePBX shenanigans, but same result. |
| 19:42.05 | beefpastry | [TK]D-Fender: http://pastebin.com/01jZqNgW |
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| 19:42.39 | [TK]D-Fender | beefpastry: Show us the CALL |
| 19:43.00 | [TK]D-Fender | beefpastry: exten => s,n,Set(${CALLERID(number)}=8888675309) <-- INVALD |
| 19:43.09 | [TK]D-Fender | beeYou are referencing a function you need to set |
| 19:43.32 | [TK]D-Fender | beefpastry: exten => s,n,Set(${CALLERID(name)}=My Company) <--- ditto |
| 19:44.01 | p3nguin | Set(CALLERID(all)=My Company <8888675309>) |
| 19:44.57 | beefpastry | p3nguin: that's my current dialplan, actually. |
| 19:45.59 | p3nguin | or Set(CALLERID(num)=8888675309) and Set(CALLERID(name)=My Company) |
| 19:46.28 | p3nguin | If that's your current dialplan, why would [tk]d-fender throw out these invalid lines? |
| 19:46.44 | beefpastry | oh...crap...because he's not overlooking what I am ;) |
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| 19:51.34 | grungies1138 | lol. I just went to linksys.com and tried to search for the word 'phone' their search engine auto-corrects it to the word piano... |
| 19:51.49 | rrb3942 | hah |
| 19:51.50 | p3nguin | lol |
| 19:53.05 | beek | It's because they make better pianos than they do phones. |
| 19:56.09 | grungies1138 | apparently |
| 19:56.23 | grungies1138 | looking at the SNOM m3 DECT phone. anyone used it? |
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| 19:58.10 | [TK]D-Fender | grungies1138: The M3 has gotten very specifically shit reviews |
| 19:58.26 | rrb3942 | beefpastry: if fixing your Set doesn't help, i have a way to help bypass freepbx wanting to set the callerid. |
| 19:58.40 | grungies1138 | hmm. |
| 19:58.52 | p3nguin | I have a way to bypass FreePBX, too. |
| 19:59.11 | beefpastry | p3nguin: working on that one |
| 19:59.23 | grungies1138 | apt-get remove? |
| 19:59.39 | p3nguin | or other related removal commands, yes. |
| 19:59.51 | grungies1138 | heh |
| 19:59.53 | [TK]D-Fender | It does help... to actually get your code right |
| 20:00.08 | beefpastry | rrb3942: that solved it...simple mental block |
| 20:00.08 | *** join/#asterisk Baylink-work (~jra@205.201.151.25) |
| 20:00.26 | beefpastry | Yep |
| 20:00.41 | grungies1138 | same with the m9 D-Fender? |
| 20:01.18 | beefpastry | Thanks for the extra eyes. |
| 20:01.22 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 20:01.24 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 20:01.46 | rubberneck | I want to stop an asterisk server from accepting any more calls, how can this be done? core stop when convienient> |
| 20:01.48 | rubberneck | ? |
| 20:02.19 | rrb3942 | gracefully? |
| 20:02.36 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:02.42 | rubberneck | I want it to let the calls that exist fall off before it shuts down. |
| 20:03.02 | wcselby | core stop graceful |
| 20:03.21 | rubberneck | wcselby: Are you certain? |
| 20:03.43 | *** join/#asterisk Chotaire (chotaire@to.to.to) |
| 20:03.44 | rrb3942 | it should let all the current calls finish but not accept anymore, and then stop |
| 20:03.47 | wcselby | rubberneck - on 1.6.x it's core stop gracefully, on 1.4.x it's just stop gracefully. I'm pretty sure. |
| 20:03.52 | Chotaire | morning buddies... |
| 20:04.28 | rubberneck | Thanks guys |
| 20:04.29 | Chotaire | someone might wanna check if the xmission.org mirror is down for you aswell. since that mirror is hardcoded on the asterisk side, this is quite a nice downage. |
| 20:06.25 | wcselby | Chotaire - got a link I can test? |
| 20:06.57 | Chotaire | http://mirrors.xmission.org/asterisk/AsteriskNOW-1.5.0-i386-1of1.iso |
| 20:07.17 | Chotaire | it's the larger files that get forwarded to this mirror by default. |
| 20:07.38 | Qwell | Chotaire: hold off on that. |
| 20:07.47 | Chotaire | ? |
| 20:07.48 | wcselby | Chotaire - if I click your link, it doesn't work |
| 20:07.50 | Qwell | but yes, you're probably right. |
| 20:08.05 | Qwell | Chotaire: 1.7.0 is going to be syncing to mirrors in the immediate future |
| 20:08.07 | wcselby | Chotaire - if I click th elink for asteriskNOW on the asterisk.org page, it works, it's coming from a different mirror apparently |
| 20:08.18 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.137.237.dsl.dyn.forthnet.gr) |
| 20:08.23 | Qwell | and yeah, there's a load balancer in front |
| 20:08.24 | wcselby | Qwell - oooooooh |
| 20:08.25 | Chotaire | Qwell: Don't worry, I ain't using any precooked distributions. |
| 20:08.45 | Qwell | wcselby: I just submitted the IT ticket about 90 seconds ago to rsync it :p |
| 20:08.54 | Chotaire | Qwell: Use this one instead... http://ftp.arklinux.org/pub/asterisk/ |
| 20:10.35 | Chotaire | QWell: Asterisk 1.7.0 or AsteriskNow 1.7.0? |
| 20:10.40 | Qwell | AsteriskNOW |
| 20:10.46 | Chotaire | aah k ;) |
| 20:11.05 | wcselby | so Qwell, tell us about AN 1.70.... :) |
| 20:11.17 | Qwell | wcselby: blog post forthcoming |
| 20:11.30 | Qwell | CentOS 5.5, updated freepbx, new dahdiconfig module for Digium hardware |
| 20:11.43 | Qwell | Asterisk 1.6.2 by default |
| 20:11.45 | *** join/#asterisk CrimsonX (~ccc1490a@gateway/web/freenode/x-eaaxcijkvmvknbvt) |
| 20:11.54 | grungies1138 | Does the free version of Asterisk do Fax to email? |
| 20:12.09 | wcselby | Qwell - OSLEC built in? |
| 20:12.13 | Qwell | wcselby: no |
| 20:12.26 | wcselby | Qwell - :( ah well, can't have everything I guess... :) |
| 20:12.41 | Chotaire | I'm still on 1.4.17... does meetme still require high precision timers in later versions for ztdummy? |
| 20:13.04 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:13.11 | Qwell | Chotaire: it isn't the timers it needs. it's the mixer in DAHDI. and yes. |
| 20:13.50 | CrimsonX | grungies1138: i have had success with writing an AGI script to do fax to email but it requires each DNIS to have an assigned user |
| 20:14.04 | Chotaire | I run my pbx under xen... works perfectly with ztdummy. then I probably don't even have to try on virtualbox even with 1.6.*, right? |
| 20:14.41 | grungies1138 | AGI script huh? oh well. :) |
| 20:15.03 | wcselby | Chotaire - I don't understand what you just said? if you go to 1.6.x, you'll need to go to dahdi and dahdi_dummy |
| 20:15.39 | wcselby | Chotaire - you'll also have access to ConfBridge, a MeetMe replacement, also the new internal timers. |
| 20:15.47 | Chotaire | wcselby: in that case, will dahdy_dummy work on virtualbox (even without proper high precision timing)? |
| 20:15.51 | *** join/#asterisk scardinal (~supreme@90.184.100.119) |
| 20:15.56 | CrimsonX | yup |
| 20:16.12 | wcselby | Chotaire - pretty much. |
| 20:16.16 | Chotaire | a meetme replacement? is it downwards compatible? hell, I put virtually years of code into meetme... |
| 20:17.16 | russellb | MeetMe() isn't going away, but ConfBridge() is built on a new, better architecture |
| 20:17.17 | wcselby | meetme is still in 1.6.x, confbridge is just a new app available to 1.6.x. It may have a port back to 1.4.x, I'm not sure as I've never tried it. the biggest selling point is that it doesn't rely on the mixers in zaptel / dahdi codebase |
| 20:17.23 | russellb | and the goal is to eventually make it feature compatible |
| 20:19.03 | Chotaire | I will check it out but I am kinda scared already. I've been using all imaginable tricks around meetme and built a completely custom, unique product which I am afraid to rewrite should meetme ever leave us. |
| 20:19.44 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 20:20.18 | russellb | MeetMe won't be going away unless we managed to make it perfectly backwards compatible (very unlikely) |
| 20:20.20 | Chotaire | just don't throw out meetme() please... |
| 20:20.44 | Chotaire | Ok, word. |
| 20:22.23 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:22.45 | [TK]D-Fender | Chotaire: You remind me of a guy who live-linked a copyrighted pic from some other server onto his site and when they removed the picture he tried to SUE them to put it back. "Hey, you removed that picture I was stealing from you! PUT IT BACK!!!" |
| 20:23.09 | [TK]D-Fender | Chotaire: Remember that you're building on quicksand is all I'm saying ;) |
| 20:24.14 | rubberneck | exit |
| 20:24.18 | *** part/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 20:24.27 | Chotaire | Fender: I just don't like losing years of development because I have to redo from scratch. The comparison with copyrighted pics I don't understand however. I ain't selling this product. |
| 20:24.31 | ChannelZ | that's like The Clapper of IRC |
| 20:24.34 | [TK]D-Fender | Chotaire: Plenty of people stuck on 1.2 because of similar lack of foresight and migration work. They are doomed to obsolescence and disintigrating security |
| 20:25.17 | [TK]D-Fender | Chotaire: Try coding for Apple :) They change their API's and are all "deal with it". Just the way the world turns... |
| 20:25.39 | ChannelZ | The Steve knows what is best for you. |
| 20:25.42 | Chotaire | Fender: I should code flash applets for the iphone then. |
| 20:25.52 | [TK]D-Fender | Chotaire: You're looking at 1.4. There have been 3 major branches released since then, and another one not too far off... |
| 20:25.59 | [TK]D-Fender | Chotaire: :p |
| 20:27.16 | *** join/#asterisk SaiSoma (~IceChat7@client105.jdcc.edu) |
| 20:27.22 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:27.43 | Chotaire | Fender: When it all started, I was recompiling on each new version, been with you guys for many many years. It's just I don't have the time anymore to beta-test and follow the development. |
| 20:28.26 | [TK]D-Fender | Chotaire: Outsource :) |
| 20:28.28 | Chotaire | Right now I am glad I have a stable setup on Xen on an otherwise LTS supported platform, ztdummy can be quite a pain sometimes. |
| 20:28.38 | *** join/#asterisk bbkt-trix (~bbkt-trix@unaffiliated/bbkt-trix) |
| 20:28.43 | SaiSoma | Anyone here familiar with grandstream ata's? I'm having an issue with them not allowing dialing after the initial number (long distance code for example) |
| 20:28.53 | wcselby | ChannelZ - http://shirt.woot.com/ is appropriate for your last comment |
| 20:30.48 | [TK]D-Fender | checkout time, bbiab |
| 20:31.31 | Chotaire | Fender: An ISP outsorcing a private unique asterisk project... sounds strange to me ;) The custom hacking makes updating a bit of a hassle. but I'll definitely hop onto the next major release. |
| 20:32.07 | Chotaire | (if it works for me) |
| 20:33.39 | Chotaire | can someone pastebin a show application confbridge for me? |
| 20:33.52 | Chotaire | that would be really nice. |
| 20:34.05 | ChannelZ | wcselby: hehe nice |
| 20:34.45 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:37.20 | ChannelZ | I'm not on 1.6.2 but if you look at the head of http://svn.digium.com/svn/asterisk/trunk/apps/app_confbridge.c you can see the args and such |
| 20:38.53 | Chotaire | Thank you. It seems like it will take quite a while until I can even think of conversion. I make good use of meetme options. |
| 20:39.18 | Chotaire | lot of stuff missing. |
| 20:39.41 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 20:40.29 | Chotaire | anyway, it relaxes me you wanna keep meetme. Thanks. |
| 20:43.11 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:43.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:44.58 | *** join/#asterisk Lyma (~lyma@unaffiliated/lyma) |
| 20:50.12 | *** join/#asterisk bbkt-trix (~bbkt-trix@unaffiliated/bbkt-trix) |
| 20:50.45 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 20:51.58 | wcselby | YAY! * Add ability to generate ASCII documentation from the TeX files by running |
| 20:51.58 | wcselby | <PROTECTED> |
| 20:52.10 | idespinner | i'm trying to find a dialplan application which looks to see if a file exists or not |
| 20:52.30 | idespinner | what happens when i use ReadFile and the file does not exist? |
| 20:52.48 | Chotaire | While talking about Apple.. Especially Steve pisses me off lately. I ain't buying an iphone simply because I don't want anyone to tell me that something is "the past" eventhough it is world standard and even if apple released the next generation super devkit for css/html5, Flash will still stay rich media standard for probably another decade. I do like choice. |
| 20:53.33 | Chotaire | Mr. Jobbs that is... Mr. Wozniak is a cool dude. |
| 20:53.37 | wcselby | Chotaire - so you don't run windows or IE? |
| 20:53.43 | wcselby | Chotaire - ;) |
| 20:54.21 | Chotaire | I run windows too.. but I don't run IE unless I am on an xp machine where I need windows updates or when I need to debug "code". |
| 20:55.14 | Chotaire | IE6, IE7, IE8.. all different... always new surprises. I hate it. |
| 20:55.56 | leifmadsen | idespinner: try STAT() |
| 20:56.03 | leifmadsen | idespinner: it's a dialplan function |
| 20:56.16 | leifmadsen | wcselby: that was done at the request of Manxpower |
| 20:56.48 | leifmadsen | wcselby: I was tired of hearing about him bitch about lack of text files, so I just figured out how to convert it from TeX to ASCII and put it in as a Makefile option |
| 20:57.00 | *** join/#asterisk devdvd (~myemail@173-31-160-214.client.mchsi.com) |
| 20:58.17 | wcselby | leifmadsen - lol, yeah, I believe I commiserated with him about that at some time as well |
| 20:58.18 | Chotaire | wcselby: Why referring to Windows? Any software that requires Appstore's holy batch to get installed? |
| 20:58.51 | idespinner | leifmadsen, no luck, what branch/trunk is the dialplan app STAT() in? |
| 20:58.53 | wcselby | Chotaire - lol, just the statement " I don't want anyone to tell me that something is "the past" eventhough it is world standard" would seem to apply to MS |
| 20:59.03 | leifmadsen | idespinner: not sure -- probably post-1.4 |
| 20:59.10 | wcselby | idespinner - he said function, not app |
| 20:59.11 | idespinner | 1.6 here.... no luck |
| 20:59.18 | leifmadsen | idespinner: trunk is trunk -- branches are 1.6.0, 1.6.1, or 1.6.2 |
| 20:59.20 | leifmadsen | 1.6 means nothing |
| 20:59.36 | leifmadsen | there is no 1.6 branch |
| 20:59.51 | idespinner | well maybe I should say 1.6 from the 1.6 debian repos which is likely a standard build |
| 21:00.01 | leifmadsen | which again means nothing |
| 21:00.08 | leifmadsen | 1.6 is not a valid version / branch |
| 21:00.13 | wcselby | idespinner - when you run asterisk -r what version does it say it is? |
| 21:00.16 | wcselby | should be the first line |
| 21:00.21 | idespinner | ah sorry |
| 21:00.34 | idespinner | didnt realizze 1.6.0/1/2 were drastically different |
| 21:00.49 | leifmadsen | yes, they are the equivelent of 1.2 vs 1.4 |
| 21:00.49 | idespinner | hrmm well actually it is 1.4.21.2 |
| 21:00.52 | leifmadsen | ha |
| 21:00.55 | wcselby | idespinner - the difference between 1.6.0 - 1.6.1 is like from 1.2 to 1.4, etc |
| 21:00.56 | Chotaire | wcselby: whatever strange prophecies coming out of bill's mouth.. atleast with windows I don't need to use AT&T only, jailbreak to get rid of nazistore just to find out that I can't install a software product because the developer is not allowed to release it. |
| 21:00.57 | leifmadsen | try 1.6.2.x |
| 21:01.01 | idespinner | ah good to know |
| 21:01.13 | leifmadsen | 1.6.0.x and 1.6.1.x are security support only now |
| 21:01.20 | leifmadsen | ~asterisk-versioning |
| 21:01.21 | infobot | hmm... asterisk-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 21:01.23 | Chotaire | wcselby: I can also replace the battery in a windows computer ;) |
| 21:01.29 | leifmadsen | ~asterisk-versions |
| 21:01.29 | infobot | it has been said that asterisk-versions is Information about the maintenance support and when branches will move into security fix only mode, and eventually end-of-life is available at http://www.asterisk.org/asterisk-versions |
| 21:01.42 | idespinner | ok thank you |
| 21:01.58 | wcselby | Chotaire - lol, i'm not arguing, was just trying to be funny and make a point |
| 21:02.49 | idespinner | ok, lets say i'm stuck with 1.4.... :( could I get the return value of ReadFile and do a goto if? How does one get the return value of a DP application? |
| 21:03.01 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:03.10 | wcselby | idespinner - you always have System() available to you |
| 21:03.25 | wcselby | idespinner - or AGI |
| 21:03.43 | wcselby | idespinner - and there may be a way to do what you want in the dialplan, but I don't know about it off the top of my head |
| 21:03.54 | idespinner | yea but using system seems a little messy |
| 21:03.56 | Chotaire | wcselby: I might have forgotten a ":)". Still, as much as I disagree with microsoft's management strategies... Windows 7 is a good, solid product. |
| 21:04.09 | idespinner | i see in the source for 1.4, Readfile will return -1 if the file does not exist |
| 21:04.10 | leifmadsen | wcselby: ya, the easy way is with STAT(), but that doesn't exist in his version, so AGI is likely the only recourse |
| 21:04.30 | wcselby | Chotaire - yes, it is. so was XP. I also liked Vista, but I seem to be in the minority with that opinion... |
| 21:04.35 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:04.42 | idespinner | is: exten s,1,var=ReadFile(some_file) valid?? |
| 21:04.51 | leifmadsen | wcselby: SHELL() in later versions (1.6.0.x and greater) will return data |
| 21:04.56 | leifmadsen | no |
| 21:05.00 | leifmadsen | idespinner: no |
| 21:05.05 | leifmadsen | you're missing several parts |
| 21:05.28 | leifmadsen | exten => start,1,ReadFile(...) |
| 21:05.40 | leifmadsen | your syntax is wrong |
| 21:05.49 | Chotaire | Windows 7 is Vista R2 to me. Vista was rock solid too, but it was (and still is) considerably slower on weaker machines. I still run Vista SP2 on my workstation which can't get any faster by upgrading anyway. XP was ok. |
| 21:05.51 | idespinner | yea sorry, i just typed it up real quick |
| 21:05.54 | leifmadsen | 's' is also not a wildcard / catchall (if that's what you're expecting 's' to do) |
| 21:06.19 | idespinner | is: exten => s,1,var=ReadFile(some_file) valid?? |
| 21:06.25 | idespinner | s would just be any extension |
| 21:06.28 | leifmadsen | at this point I'd setup a test extension and just just test it out. Run 'core show application Readfile' |
| 21:06.37 | wcselby | idespinner - exten => s,1,ReadFile(VARIABLE=filename,length) would be the correct syntax, where VARIABLE is what will hold the result, and filename is the name of the file to check |
| 21:06.39 | leifmadsen | idespinner: again, neither of those statements are right |
| 21:07.02 | leifmadsen | <idespinner> s would just be any extension <-- incorrect |
| 21:07.14 | leifmadsen | assuming your statement means "match anything" |
| 21:07.37 | wcselby | leifmadsen - i took it to mean he's using s as a placehold for his real extensions....i.e exten => 123,1,... |
| 21:07.51 | idespinner | lol |
| 21:07.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:07.54 | wcselby | a placeholder for here in IRC, not for his dialplan |
| 21:08.00 | idespinner | wcselby, yes thats what i meant |
| 21:08.10 | idespinner | s is only a place holder in my example |
| 21:08.24 | wcselby | idespinner - but still, the rest of you syntax is wrong |
| 21:08.36 | Chotaire | wcselby: nice to meet people who understand there is not only one religious choice of OS. |
| 21:08.53 | idespinner | my ultimate question I believe is how does one store the return value of a DP application into a variable? |
| 21:09.05 | idespinner | (which I am testing out different things now) |
| 21:09.26 | leifmadsen | idespinner: it's up to the application to support that, otherwise that's what functions are for |
| 21:09.32 | wcselby | Chotaire - haha, i use win 7 at home simply because I like a few of the features over Vista's, but I used vista for a long time. i also have an iPhone, which is my only Apple device. I've been on AT&T for years though, so that wasn't a problem for me |
| 21:09.36 | leifmadsen | like in the case of the Readfile() application |
| 21:10.00 | wcselby | idespinner - the best way I can think of to set variables in DP is with AGI |
| 21:10.23 | idespinner | gotcha, so what you are saying is that unless the application specificaly supports it, the return value is not apparently visible in the dialplan during runtime |
| 21:10.23 | wcselby | idespinner - well, that's not exactly true, but for the instances we're talking about, I mean |
| 21:10.42 | [TK]D-Fender | idespinner: Dialplan apps don't return values. They may set variables hoever. FUNCTIONS return values |
| 21:10.49 | leifmadsen | see above |
| 21:11.11 | [TK]D-Fender | idespinner: This has been a rule in programming for what... 40 years? |
| 21:11.44 | Chotaire | wcselby: I run 7 different operating systems 24/7, and I love all of them. Instead of Apple, I went into the HTC world. Iphone is an incredibly grownup and optimized product, but I am glad I have a choice. ;) |
| 21:12.15 | leifmadsen | OS religious wars are so 1999 |
| 21:12.28 | Chotaire | leifmadsen: I wish you were right. |
| 21:12.29 | leifmadsen | I use at least 3 different OS's on a daily basis |
| 21:12.53 | leifmadsen | Chotaire: people who wage wars over OS choice have nothing better to do and are typically a waste of my time |
| 21:12.58 | idespinner | [TK]D-Fender, I wouldnt, know that old.... :P |
| 21:13.06 | wcselby | Chotaire - haha, the more I use my iPhone and the more I hear about Androids, the more I want one. I really like the open source philosphy behind their OS (at least, as I've heard it told from Androd fans). |
| 21:13.15 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:13.20 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 21:13.23 | [TK]D-Fender | idespinner: You don't need to be ancient history to know it... |
| 21:13.34 | wcselby | haha, 40 years old is ancient history? |
| 21:13.38 | wcselby | :P |
| 21:13.45 | [TK]D-Fender | wcselby: In computer time.. Yes :p |
| 21:14.14 | *** join/#asterisk sawgood (~sawgood@205.167.47.158) |
| 21:14.20 | Chotaire | wcselby: HTC also made windows mobile a good mobile OS. without htc sense/touch, windows mobile 6.* is early 90s. |
| 21:15.44 | wcselby | Chotaire - I've never liked the windows mobile os'es, but I haven't tried since 5 or 6. when I see them though, I still haven't been impressed. I really like my blackberry, but I hated it for calling. I never thought I would want internet on a phone until my buddy got an iPhone. iPhone does phone calls and Internet very well. It does email okay (I wish it was more BB-like). Although I hear one fo the new improvements to the iphone OS is going to be |
| 21:16.07 | Chotaire | wcselby: both android and windows mobile is great with HTC Sense. Iphone is still superiour in terms of quality and speed. Nothing beats a 3GS, it's like a vic-20 compared to a c64. |
| 21:16.25 | wcselby | Chotaire - haha yeah. |
| 21:16.52 | [TK]D-Fender | Chotaire: tried a snap-dragon powered android device yet? |
| 21:16.53 | wcselby | Chotaire - I've read about people loading android onto their iPhones now, I don't know if they're really loading it or just modding jailbroken iphones with it, but I may look into doing something like that soon. |
| 21:16.59 | Chotaire | give it another year or two and if apple ain't reinventing their own wheel, there is no reason to get an iphone anymore. |
| 21:17.51 | *** join/#asterisk jroysdon (~jroysdon@2001:470:1f05:36a::3) |
| 21:17.55 | Chotaire | wcselby: I understand people like hacking around... but it is the combination of iphone OS and iphone hardware that make the product outstanding, I wouldn't do that. if I want android, I get a high-end HTC mobile (e.g. HD2 and up) |
| 21:18.31 | Chotaire | Fender: nope, not yet. |
| 21:19.39 | [TK]D-Fender | Chotaire: Then you may want to revists your "speed test". The closest runner up below that would be the Motorola Droid/Milestone. Tried either of those? |
| 21:19.54 | Chotaire | yup the milestone. |
| 21:20.23 | [TK]D-Fender | Chotaire: Well mileston is only about a 600mhz CPU. Snapdragon starts at 1ghz and is going up. |
| 21:20.36 | Chotaire | fender: well, the usability starts with the touchscreen. capacitve multitouch.. haven't seen any finer than on an iphone 3gs yet. |
| 21:20.44 | [TK]D-Fender | (new 800mhz are in the works.. but lets not think backwards here) |
| 21:21.01 | mianos | re android on the iphone, it's booting yes |
| 21:21.10 | mianos | a friend of mine was showingme it |
| 21:21.13 | mianos | a few days ago |
| 21:21.16 | [TK]D-Fender | Chotaire: reviews show the iphone touchscreen is indeed the best tracking. but now > 1/2 the res. |
| 21:21.25 | mianos | he's got a boot loader that asks OSX or adndoid |
| 21:21.35 | mianos | very complete, quite surprising |
| 21:21.42 | wcselby | mianos - ahhhh, nice. have a link to a guide on how to do it? |
| 21:21.54 | Chotaire | mianos: dualboot is an option, yes. otherwise I wouldn't do it. |
| 21:22.23 | mianos | i don't have an iphone, sorry |
| 21:22.29 | mianos | it makes calls and such |
| 21:22.35 | wcselby | [TK]D-Fender, who is releasing these snapdragon phones? |
| 21:22.40 | mianos | I was amazed, sure boot it, display driver |
| 21:22.44 | wcselby | or should I say phones with snapdragron procs? |
| 21:22.48 | mianos | but the touchscreen and calling is way cool |
| 21:23.24 | Chotaire | Fender: This one you want ;) http://www.htc.com/de/product/desire/overview.html |
| 21:23.42 | Chotaire | so I correct myself, I have seen a snapdragon 1ghz. |
| 21:24.13 | Chotaire | it's coming close, but iphone 3gs still feels a step more complete to me. |
| 21:24.53 | Chotaire | Fender: sorry for gibberish link.. http://www.htc.com/www/product/desire/overview.html |
| 21:26.10 | *** join/#asterisk x303 (~X303@wifi222-144.meruwifi.fit.edu) |
| 21:26.12 | Chotaire | Fender: And this is the windows beast variant... love it.. http://www.htc.com/www/product/hd2/overview.html |
| 21:26.38 | [TK]D-Fender | Chotaire: Yes, the iPhone as a global platform IS tighter, but the swam forming around Android has that something extra that Apple's closed nature may soon find themselves completely blown over by |
| 21:26.49 | [TK]D-Fender | Chotaire: And we see that gap closing FAST |
| 21:26.59 | Chotaire | Fender: Hopefully... Flash is going native on Android ;) |
| 21:26.59 | wcselby | [TK]D-Fender - that's what I've been hearing and sensing as well. |
| 21:27.20 | [TK]D-Fender | Chotaire: Already is with Froyo |
| 21:27.29 | Chotaire | Fender: check the video for the windows phone... it looks different than you probably remember ;) |
| 21:29.21 | [TK]D-Fender | Chotaire: Yeah I follow the mobile market very closely.. I'm a little trapped by the awesmoe cost-effectiveness of my ancient HTC Touch |
| 21:29.53 | Chotaire | Fender: I still have a HTC Touch Diamond... no need to get a new one yet... it's all good. ;) |
| 21:29.55 | [TK]D-Fender | Chotaire: $10/mo unlimited data saves me $240/y worth of what any other device would cost me. |
| 21:30.05 | [TK]D-Fender | Chotaire: And it IS a good phone... |
| 21:30.18 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 21:30.42 | [TK]D-Fender | Chotaire: that plan is the only thing keeping me tied to it |
| 21:31.02 | [TK]D-Fender | Chotaire: My next phone will likely be something that isn't even out yet |
| 21:31.50 | Chotaire | Fender: I guess my next phone will be something like the HD2, but with Android. Just a guess, we'll see what's happening. |
| 21:32.32 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 21:32.34 | niekvlessert | hello! |
| 21:32.40 | Chotaire | hi there. |
| 21:33.07 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 21:33.20 | niekvlessert | what would i need to setup to forward a call on a gsm (which originated from asterisk) to someone else on the pbx with the # key? |
| 21:33.53 | niekvlessert | and, hello Chotaire |
| 21:33.54 | niekvlessert | :) |
| 21:35.19 | niekvlessert | well, no one? |
| 21:35.40 | *** join/#asterisk Lantizia (~lantizia@93-97-23-110.zone5.bethere.co.uk) |
| 21:35.41 | Chotaire | niek: "t" parameter on dial command? |
| 21:36.05 | Lantizia | Evening :) In zaptel/system.conf what would the number 3 control in this line? span=1,1,3,ccs,ami |
| 21:36.45 | Lantizia | dahdi_genconf always has a 0 for that third value, but my card manufacturers example config uses a 3 |
| 21:37.33 | Chotaire | niek: the dial command that dials your gsm number should have the t parameter, in that case you can press a defined sequence to transfer the call somewhere else. |
| 21:37.43 | Chotaire | (features.conf) |
| 21:38.36 | *** join/#asterisk CaZaOrCaS (~chatzilla@200.124.10.66) |
| 21:38.43 | CaZaOrCaS | hi |
| 21:40.14 | niekvlessert | damn, chotaire, I just found that out myself, thanks: |
| 21:40.16 | niekvlessert | <PROTECTED> |
| 21:40.16 | niekvlessert | <PROTECTED> |
| 21:40.16 | niekvlessert | <PROTECTED> |
| 21:40.17 | niekvlessert | <PROTECTED> |
| 21:40.28 | Chotaire | niek: exactly. Don't mix with "T" ;) |
| 21:40.38 | niekvlessert | called/calling |
| 21:40.43 | Lantizia | Anyone got one of those irritating bot triggers to show me the DAHDI manual? |
| 21:41.24 | leifmadsen | I don't believe there is a "DAHDI manual" |
| 21:41.50 | *** join/#asterisk CaZaOrCaS (~manuel@200.75.220.99) |
| 21:41.52 | Chotaire | http://docs.tzafrir.org.il/dahdi-tools/README.html |
| 21:42.21 | Lantizia | 2.4.1. Span Configuration |
| 21:42.24 | Lantizia | cool thanks Chotaire |
| 21:42.28 | Chotaire | :) |
| 21:43.03 | CaZaOrCaS | Hello, anybody can tell me if can use the asterisk logo on a banner outside of my business |
| 21:43.04 | *** join/#asterisk jhirley (~jhirley@adsl-6-32-97.mia.bellsouth.net) |
| 21:43.07 | Lantizia | so that third value is the LBO |
| 21:43.24 | Lantizia | should that be the distance to the ISDN2e port on the wall from the PBX ? |
| 21:43.58 | Lantizia | ah hmm if it is BRI then it's not used lol |
| 21:43.59 | wcselby | haha, twitter trend for the day - #gorepickuplines |
| 21:44.51 | Chotaire | lantizia: then you set 0 |
| 21:45.23 | Lantizia | Chotaire, Junghanns.NET sample system.conf has it set to 3... the only other difference from the DAHDI generated file is the echocanceller is different |
| 21:45.44 | Lantizia | kb1 instead of mg2 |
| 21:46.07 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca) |
| 21:46.29 | Lantizia | Chotaire, even if it's Junghanns (not the Digium BRI that DAHDI natively supports) that value isn't used? |
| 21:46.57 | Chotaire | lantizia: you might wanna ask someone else, DAHDI is not my friend yet. I remember the junghanns admin (or was it CEO?) is a user here too. |
| 21:47.03 | Chotaire | he still around? |
| 21:47.24 | wcselby | night all |
| 21:48.26 | Chotaire | I don't remember his nick, it's been many many years ago. |
| 21:48.48 | Lantizia | lol ok |
| 21:50.48 | Chotaire | lantizia: I'd say you try with 0.. if you hear clicks, noises, you will know that was no good idea. |
| 21:51.02 | Lantizia | :) ok |
| 21:51.22 | Lantizia | at the moment I'm trying to figure out why the spans keep going down the up every 30 seconds |
| 21:51.27 | Lantizia | however they don't go down if in use |
| 21:53.58 | Chotaire | have you set resetinterval? (just a wild guess) |
| 21:55.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:57.11 | Chotaire | another wild guess is that your BRI provider does not use continuous sync (have DSL?) |
| 21:59.50 | Chotaire | lantizia: If you can't find a solution yourself and nobody of the asterisk gurus here can help you, call your telephony provider and request they switch layer 1 synchronization to continuous (not quite sure how that is called in the UK). that might solve your problem. |
| 22:01.54 | Chotaire | lantizia: neitherless, if you have problems with sync, lbo set to 3 is probably a better idea. sorry I can't help you any further.. I haven't done bri in many years. |
| 22:02.11 | Lantizia | it's ok :) |
| 22:03.03 | Lantizia | I'm pretty sure my settings are fine, I've got the chan_dahdi.conf and system.conf from a working zapata.conf and zaptel.conf from another system that uses the same card on the same ISDN2e providers line |
| 22:03.11 | Lantizia | I think it must be them |
| 22:03.38 | Chotaire | yup, many providers offer continuous sync only when requested. |
| 22:03.59 | Chotaire | just call them. they will probably know what you want, once you find someone who knows more than telemarketing. |
| 22:04.05 | Lantizia | would not having continuous sync mean I see... |
| 22:04.05 | Lantizia | <PROTECTED> |
| 22:04.05 | Lantizia | <PROTECTED> |
| 22:04.05 | Lantizia | <PROTECTED> |
| 22:04.06 | Lantizia | <PROTECTED> |
| 22:04.06 | Lantizia | <PROTECTED> |
| 22:04.07 | Lantizia | <PROTECTED> |
| 22:04.09 | Lantizia | <PROTECTED> |
| 22:04.11 | Lantizia | <PROTECTED> |
| 22:06.30 | Chotaire | lantizia: stupid question.. have you h ad problems with incoming or outgoing calls or is it just that you are bothered by the messages? |
| 22:08.04 | Chotaire | lantizia: as you say, once you place a call, the d-channel stays up... have you tried hammering some incoming calls to find out if you would miss any call? have you had problems placing outgoing calls when the channel was down? |
| 22:08.42 | jroysdon | I'm trying to add an extension to my system, and I keep getting Config File Error 0x20 on this Polycom 450. Any idea where to start? I copied the MAC.cfg file from another 450 phone and made edits to the extensions, and I configured the new phone to download, and it gets some files, but then reports that error. |
| 22:09.18 | Lantizia | Chotaire, bothered by the messages |
| 22:09.19 | *** join/#asterisk AI_Rob (~rob@c-76-127-249-66.hsd1.ct.comcast.net) |
| 22:09.39 | *** join/#asterisk Ta^3 (~tacvbo@189.146.180.11) |
| 22:09.45 | Lantizia | Chotaire, the calls are fine... if a span is in use (either incoming or outgoung) then it stays up |
| 22:09.53 | jroysdon | I see this error message in MAC-boot.log: 0101001213|app1 |4|00|Loaded application sip.ld successfully, errors 0x20. |
| 22:10.56 | Chotaire | lantizia: you should definitely check outgoing calls once the d-channel is down. if you ever run into problems (congested) and you would still use chan_misdn: pmp_l1_check=no |
| 22:11.41 | Chotaire | lantizia: I hope that helped a bit. no more I can do. otherwise, just ignore it and call your provider. make sure you try high quality cables. |
| 22:12.04 | Lantizia | have already called them, they should be getting back to me tomorrow |
| 22:12.05 | Chotaire | kapejod.. now I remember his nick.. how is he doing? |
| 22:12.36 | Lantizia | kapejod is Junghanns guy? |
| 22:13.33 | Chotaire | we have a lot of celebrities here ;) |
| 22:14.52 | Chotaire | jroysdon: which firmware? |
| 22:15.04 | jroysdon | Chotaire, I think I found the next step, I needed to create MAC_phones.cfg |
| 22:15.20 | jroysdon | (which I have done, and now I got it to boot with the extension showing) |
| 22:15.42 | Chotaire | ok |
| 22:15.56 | jroysdon | Anyone else use FreePBX? Does one still have to create all these cfg files manually even when using a GUI front-end like FreePBX? |
| 22:16.08 | *** join/#asterisk grungies1138 (~grungies1@cpe-66-74-197-34.san.res.rr.com) |
| 22:16.59 | grungies1138 | If I wanted to set up a submenu off my main attendant I just start a new context and use a Goto(<context>) yes? |
| 22:17.03 | Chotaire | jroysdon: AsteriskNow (with FreePBX) v1.7.0 coming anytime soon (today?) |
| 22:17.10 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 22:17.25 | jroysdon | Chotaire, yeah, I'm just trying to get this working with what I have now |
| 22:17.37 | jroysdon | FreePBX 2.6.0.1 :/ |
| 22:18.17 | Chotaire | grungies: that's how you can do it. |
| 22:18.38 | grungies1138 | Thanks. Just a sanity check. :) |
| 22:18.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:19.27 | grungies1138 | If I do a Goto() do I still need a Hangup()? |
| 22:19.49 | grungies1138 | in the Main menu that is... |
| 22:19.52 | Chotaire | no, you want to go somewhere, not hangup the call ;) |
| 22:19.58 | grungies1138 | k. |
| 22:20.42 | Chotaire | you can still use hangup() for timeout and the like... |
| 22:20.48 | jroysdon | Can you configure a system such that you could take a Polycom 450 out of the box, and not have to touch the phone to configure it? (Using Option 66, for instance?). I'm used to the Cisco world and that's how we do it, just need to know the phone MAC address ahead of time to program it, and no need to touch the physical phone to config it since DHCP pushes the tftp server option which pushes the config files... |
| 22:21.26 | jroysdon | (I'm just assisting here with this phone system, the guy that set it up is gone, and seems to have been clueless) |
| 22:21.40 | [TK]D-Fender | jroysdon: Yes |
| 22:21.45 | gregd | hello guys, can someone help me finding a problem with my configuration? my asterisk box acts as a sip server for only one client and at the same time is a client for my sip provider. My client can easily connect to asterisk box and the asterisk box connects to my sip provider but i cannot place any calls... i get waiting signal but the phone does not ring... i've got output of 'asterisk -vvvgc' : http://pastebin.com/9R7De |
| 22:22.15 | [TK]D-Fender | gregd: broken link |
| 22:22.15 | gregd | if anyone could point me out my mistake i would really appreciate it :) |
| 22:22.18 | jroysdon | [TK]D-Fender, so that is the "normal" way to set it up, right? I mean, unless you had just like 1-2 phones, but if you have many phones, you'd almost always do the method I spoke of, right? |
| 22:22.29 | *** join/#asterisk voxter (~voxter@173.183.118.56) |
| 22:22.29 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 22:22.46 | Chotaire | gregd: - Unknown paste ID, it may have expired or been deleted! |
| 22:22.50 | [TK]D-Fender | jroysdon: its called "provisioning". And yes you can set it up to work from cardboard to service immediately from being plugged in |
| 22:23.06 | jroysdon | right |
| 22:23.24 | gregd | Chotaire: can u check it again, it works for me: http://pastebin.com/9R7DeAyi |
| 22:23.32 | gregd | or what other 'paste-place' do u use? |
| 22:23.44 | Chotaire | that link looks better. |
| 22:23.57 | jroysdon | I mean, put it this way from my point of view, if someone said you had to manually touch a Cisco phone to set it up with Cisco's CallManager, I'd say they were nuts. You never should need to touch a physical phone except to debug (even then you do it remotely) or do one-off situations. |
| 22:24.19 | jroysdon | ok, some progress. When I try to make a call, I see this in the asterisk log: [Jun 1 15:22:19] NOTICE[2698] chan_sip.c: Registration from '<sip:208@phone>' failed for '192.168.1.28' - Wrong password |
| 22:27.30 | Chotaire | gregd: who is 192.168.1.150? |
| 22:27.49 | gregd | Chotaire: that's me sip phone on internal network |
| 22:28.35 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 22:28.42 | Chotaire | gregd: and 192.168.1.2? |
| 22:28.52 | gregd | Chotaire: asterisk box |
| 22:29.10 | Chotaire | k |
| 22:29.34 | Chotaire | so you have tried to place an outgoing call using your .pl sip provider? |
| 22:29.51 | gregd | Chotaire: correct |
| 22:30.49 | jroysdon | hmm, where would I set the password that Asterisk is expecting or that this Polycom 450 is sending? FreePBX lists a "secret" which is just the MAC address of the device. |
| 22:30.52 | Chotaire | gregd: can you call internal extensions? |
| 22:31.23 | gregd | Chotaire: yes, i can call myself and got a callwaiting |
| 22:32.00 | Chotaire | gregd: you have a router behind the asterisk box? |
| 22:32.48 | gregd | Chotaire: SIP_PROVIDER ------ MY_GW ---- ASTERISK-BOX |
| 22:32.51 | jroysdon | ah, found it, it is in the MAC_phones.cfg file |
| 22:33.08 | Chotaire | what kinda gateway is that? |
| 22:33.20 | gregd | Chotaire: so, yes, i do, but the ROUTER and ASTERISK BOX are bridged together |
| 22:33.55 | gregd | Chotaire: it's regular internet gateway with external ip |
| 22:33.58 | Chotaire | bridged as in ethernet bridging or just connected to the same hub/switch? |
| 22:34.12 | gregd | Chotaire: ethernet bridging |
| 22:37.39 | gregd | Chotaire: however, can we make it so it will work behind a regular NAT as well? |
| 22:38.14 | gregd | Chotaire: as I am thinking of changing the schema |
| 22:38.34 | Chotaire | gregd: if connections are forwarded to the sip box, sure. |
| 22:38.50 | Chotaire | gregd: I see it is using NAT for the transmissions... |
| 22:38.54 | Chotaire | sip box = asterisk box.. duh. |
| 22:39.02 | gregd | Chotaire: yeah, i defined nat=yes |
| 22:39.09 | Chotaire | can you paste the relevant parts of the messages logfile? |
| 22:39.44 | Chotaire | gregd: what kinda router is that? |
| 22:39.45 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 22:40.10 | gregd | Chotaire: my NAT is just a regular wifi router on DDWRT |
| 22:40.25 | Chotaire | k |
| 22:44.14 | *** join/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com) |
| 22:44.19 | kam187 | hey guys |
| 22:44.34 | kam187 | does anyone know if any of the h323 implementations support NAT? |
| 22:45.00 | gregd | Chotaire: what logging lvl do u want for the messages? |
| 22:45.50 | Chotaire | gregd: why is it bridged anyway? is there a bridging firewall on that box? I'd just try with a regular NAT setup, as a starter make sure all ports are forwarded to the sip box and then see if that works better. that siplogfile gives me headache, I'd like to see the message log paste instead. -vvvvvvvvv whatever... |
| 22:46.29 | gregd | Chotaire: here are messages: http://pastebin.com/5M4rX2tv |
| 22:49.38 | Chotaire | does that phone number ever ring? (probably your own, right?) |
| 22:49.49 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 22:50.57 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 22:52.14 | Chotaire | gregd: it doesn't help me much more, but from what I understad you have a problem with sip retransmits... |
| 22:52.34 | gregd | Chotaire: so nat/gateway problem |
| 22:52.37 | gregd | ? |
| 22:53.21 | Chotaire | I'd say so... NAT / sip-proxy / header rewrite / firewall / packet-loss |
| 22:55.58 | Chotaire | it's ringing, it's retransmitting, I don't see any errors with dialplan, it dies. |
| 22:57.13 | Chotaire | I don't see a few acks... try working on your network first. |
| 22:59.23 | Chotaire | standard nat setup, forward all ports to asterisk box, verify you have no sip server running on router that could interfere... try again.. still not working? define that stun proxy of your .pl provider, try again... still not working? use a direct internet connection to your asterisk box (e.g. using an adsl/cable modem), try again.. still not working? that's how I'd start from here. |
| 23:00.42 | Chotaire | if anyone else would like to take a look at http://pastebin.com/5M4rX2tv and tell me I am terribly wrong, please do so. but that's how I would start find the cause. |
| 23:02.56 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 23:03.40 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 23:04.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-mptabfppqzfaclmu) |
| 23:09.32 | grungies1138 | <PROTECTED> |
| 23:09.41 | grungies1138 | I don't have an 'incoming' context |
| 23:09.48 | grungies1138 | it's labeled incoming_calls |
| 23:11.04 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 23:11.30 | Chotaire | grungies1138: cd /etc/asterisk ; grep -R incoming * | grep context |
| 23:11.34 | Chotaire | maybe you have elsewhere ;) |
| 23:12.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:12.59 | gregd | Chotaire: i got the point, thanks.. will try it out this way |
| 23:13.17 | grungies1138 | nope |
| 23:14.50 | grungies1138 | multiple entries from sip.conf, but the word incoming is in the comments. |
| 23:15.11 | Chotaire | grungies: how does your directory command look like? |
| 23:15.21 | jroysdon | Anyone know what config file tells these Polycom 450 vs. 560 phones what .LD file to use? |
| 23:15.31 | Naikrovek | yes |
| 23:15.36 | Naikrovek | leave it set to sip.ld |
| 23:15.46 | Naikrovek | don't attempt to outsmart the phone and put the full filename in |
| 23:15.47 | *** join/#asterisk mindCrime (~chatzilla@64.134.184.64) |
| 23:15.48 | jroysdon | well it gets a wrong version complaint |
| 23:16.03 | Naikrovek | did you rename any files |
| 23:16.05 | jroysdon | I am not, I just have a 560 that appears to not have the same software as other 560s I have already |
| 23:16.11 | jroysdon | I did not, but I'm taking over for someone else |
| 23:16.17 | grungies1138 | well, first it's option 3 off my Main Menu. exten => 3,1,Goto(directory,*70,1) |
| 23:16.20 | Naikrovek | do you use ftp |
| 23:16.24 | Chotaire | gregd: unless you run a bridging firewall on the asterisk box or virtual machines, I don't get why running it bridged. hm. |
| 23:16.29 | jroysdon | Yes, using ftp |
| 23:16.42 | jroysdon | This 560 is requesting 450 .LD files it appears |
| 23:16.44 | grungies1138 | exten => *70,1,Answer() |
| 23:16.44 | grungies1138 | exten => *70,n,Background(/var/lib/asterisk/sounds/custom/Submenu) |
| 23:16.44 | grungies1138 | exten => *70,n,WaitExten() |
| 23:16.44 | grungies1138 | exten => *70,n,Hangup() |
| 23:16.44 | grungies1138 | exten => 1,1,Directory(default,incoming,f) |
| 23:16.44 | grungies1138 | exten => 2,1,Directory(default,incoming) |
| 23:16.46 | jroysdon | (which then it errors on) |
| 23:16.51 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:16.56 | Chotaire | so there is your "incoming" |
| 23:17.02 | Naikrovek | and there are lots of sip.ld files in the root directory when the ftp user logs in? |
| 23:17.19 | Naikrovek | ~pb |
| 23:17.20 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 23:17.20 | grungies1138 | *scratching head* |
| 23:17.23 | Kobaz | i'm having problems doing a 'Command' action in the manager |
| 23:17.24 | Naikrovek | grungies1138: ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 23:17.34 | Kobaz | it doesn't like calling the dialplan series of commands |
| 23:17.41 | jroysdon | Naikrovek, yup |
| 23:17.44 | Chotaire | default is your vm-context, incoming is your dial-context |
| 23:17.54 | Naikrovek | jroysdon: did you put those files there? or did you inherit them |
| 23:17.56 | jroysdon | Naikrovek, ones ending in 450.sip.ld and 560.sip.ld |
| 23:18.02 | jroysdon | Naikrovek, I inherited them :/ |
| 23:18.03 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:18.19 | jroysdon | I can power cycle another 560 and see what it does if that matters |
| 23:18.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:18.23 | Naikrovek | jroysdon: delete them, download new SPLIT firmware (important) and unzip the firmware file there |
| 23:18.36 | Kobaz | http://pastebin.ca/1875716 |
| 23:18.40 | jroysdon | Naikrovek, from where? |
| 23:18.40 | Naikrovek | you'll have many more .sip.ld files but the phone will cooperate at that point |
| 23:18.48 | Kobaz | is there some commands you can't use via the AMI 'Command' |
| 23:18.54 | Naikrovek | jroysdon: polycom.com. click voice, then support |
| 23:19.01 | jroysdon | ok, thanks |
| 23:19.18 | grungies1138 | I'm an idiot. sorry guys |
| 23:19.25 | Kobaz | it looks like it's failing to process the arguments or something |
| 23:19.26 | Chotaire | you're welcome ;) |
| 23:19.28 | jroysdon | so I'll just remove all .sip.ld files and unzip these new ones, right, Naikrovek ? |
| 23:19.44 | Naikrovek | well back them up for now. |
| 23:19.49 | Naikrovek | just mv them to a directory |
| 23:20.01 | jroysdon | Naikrovek, right, I will move them to ./old |
| 23:20.03 | Naikrovek | but yes remove them from that directory |
| 23:20.22 | jroysdon | Ok, at polycom.com -> support -> voice |
| 23:20.26 | Naikrovek | then wget the split firmware file into the directory the old ones were in |
| 23:20.30 | Naikrovek | yes, support then voice |
| 23:20.32 | Naikrovek | sorry |
| 23:20.40 | Naikrovek | once you get the file, unzip it in place |
| 23:20.45 | Naikrovek | then reboot the phone |
| 23:21.00 | jroysdon | yup, I am there, but where do they list the split file? In the phone model? Do I need to then download it for the 560, 450, and each model I have? |
| 23:21.05 | Naikrovek | one sec |
| 23:21.16 | Naikrovek | the 560 and 450 will actually fetch the same .zip file i think |
| 23:21.23 | jroysdon | ah, ok |
| 23:21.47 | jroysdon | oh, I see: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 23:21.55 | Naikrovek | http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_2_3_release_sig_split.zip |
| 23:21.59 | jroysdon | So get 3.2.3 split? |
| 23:22.03 | Naikrovek | yes |
| 23:22.04 | jroysdon | right |
| 23:23.15 | Naikrovek | all of the soundpoint IP phones will use that same set of firmware |
| 23:23.19 | Naikrovek | that's why it's called split |
| 23:23.30 | Naikrovek | there are many phones supported, each phone gets its own .sip.ld file |
| 23:23.47 | jroysdon | gotcha. What is the diff between the split and combined files? |
| 23:23.57 | Naikrovek | combined is one sip.ld file that is huge |
| 23:24.00 | Naikrovek | and that causes issues |
| 23:24.08 | Naikrovek | takes forever to download to each phone |
| 23:24.14 | Naikrovek | split just plain works better |
| 23:24.46 | jroysdon | ah, gotcha, makes sense |
| 23:24.56 | Naikrovek | the combined is there only for phones with really early bootrom software, but modern bootroms can know which file to download, and get just that one file |
| 23:25.11 | jroysdon | right |
| 23:25.11 | Naikrovek | gotta run, daughter's tee-ball game |
| 23:25.17 | jroysdon | thanks |
| 23:25.29 | jroysdon | baseball practice for my son in an hour |
| 23:25.33 | jroysdon | (trying to get out of here) |
| 23:27.00 | *** join/#asterisk phunyguy (~phunyguy@h69-128-205-67.mtjltn.dsl.dynamic.tds.net) |
| 23:27.09 | *** join/#asterisk gregd (~gregd@cpc9-sgyl29-2-0-cust103.sgyl.cable.virginmedia.com) |
| 23:27.56 | phunyguy | Hello, quick question here... Where might I find some decent AsteriskNOW documentation? Like a "getting started" thing? |
| 23:28.12 | phunyguy | All I can seem to find is just for the Asterisk software |
| 23:28.49 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:31.16 | gregd | Chotaire: can u please again look at this log: http://pastebin.com/WDJ9jPeh In your opinion, why the retransmissions happen? |
| 23:31.40 | *** join/#asterisk x303 (~X303@wifi222-144.meruwifi.fit.edu) |
| 23:34.54 | Lantizia | Hey... If I want to use qozap (not in DAHDI)... do I _have_ to compile a new version of DAHDI with it included? or can I some how compile qozap.c / qozap.h files and include the module in an existing distribution copy of DAHDI? |
| 23:35.41 | grungies1138 | phunyguy: http://www.madikonda.com/downloads/asterisknow.pdf |
| 23:35.54 | phunyguy | gracias |
| 23:36.19 | Chotaire | gregd: who is 172.16.128.1? |
| 23:36.52 | grungies1138 | I did a 4 second google search... FYI |
| 23:37.28 | gregd | Chotaire: that's actually kind of strange... my sip phone has to network... one is main (connected to asterisk box, internet etc..) and the other is it's own private.... not used for any sip related things |
| 23:37.42 | *** join/#asterisk Baylink (~jra@cerberus.vicimarketing.com) |
| 23:38.08 | Chotaire | gregd: you really have som fun with your network there... show me a route -n |
| 23:38.22 | gregd | Chotaire: it's kind of complicated... ok, sec |
| 23:38.55 | Lantizia | any DAHDI experts about? |
| 23:38.57 | Chotaire | gregd: ip addr ls ; route -n ; route -A inet6 -n ; uname -a |
| 23:39.18 | Chotaire | gregd: also: iptables -L -n -v ; iptables -t nat -L -n -v |
| 23:39.22 | gregd | Chotaire: that's my phone http://pastebin.com/xegR7yZS |
| 23:39.36 | Lantizia | grungies1138, that documentation is ancient and isn't even FreePBX based |
| 23:41.39 | ChannelZ | See why FreePBX is such a waste? |
| 23:41.49 | gregd | Chotaire: im wondering myself why the sip phone takes the 172.16. domain |
| 23:42.20 | Chotaire | gregd: run those commands on your asterisk box and then we'll see... |
| 23:42.30 | grungies1138 | *shrugs* |
| 23:42.31 | gregd | Chotaire: but now I just try to connect my asteriskbox straight to my provider without nat... and there are the retransmit and problems.. ok, moment |
| 23:42.48 | Lantizia | Is it possible to give DAHDI new firmware (a new module... like qozap) without recompiling DAHDI? |
| 23:42.55 | Lantizia | it is modular after all! |
| 23:44.10 | gregd | Chotaire: http://pastebin.com/3YVQnbMU and http://pastebin.com/TxDs7qnP |
| 23:47.50 | Chotaire | ok back.. gregd.. reading... |
| 23:49.57 | Chotaire | gregd: ok, so that was your asterisk box... I see you are forwarding all icmp packets, no matter where they are coming from and where they are going to 192.168.1.2.. that rule can be deleted. |
| 23:50.46 | Chotaire | can you show me your router ip addr ls and route -n ? |
| 23:51.52 | Chotaire | oh, you already did.. sorry. well.. you have a dhcp server running on that router? |
| 23:52.15 | phunyguy | grungies1138: Apparently I have a newer version of AsteriskNOW, the one in your doc uses rPath Linux, mine uses CentOS 5. I also didnt get any kind of setup prompts at the web gui. |
| 23:52.38 | gregd | Chotaire: no, so far i showed u sip client and asteisk box.. |
| 23:52.39 | Chotaire | gregd: I believe the dhcp server on that router is leasing addresses from the 172.16.* range. |
| 23:53.17 | phunyguy | grungies1138: It just brought me to a page with 3 options, one of them being FreePBX setup page. |
| 23:53.20 | Chotaire | gregd: ok. what sip client/OS is that? |
| 23:53.27 | gregd | Chotaire: osx |
| 23:53.53 | gregd | Chotaire: but now, can we concentrate on asterisk<->provider? |
| 23:54.12 | gregd | Chotaire: when i disable nat for my asterisk, it does not want to register with my provider |
| 23:54.34 | gregd | Chotaire: moreover now my asterisk is on DMZ so has direct internet access |
| 23:54.49 | Chotaire | gregd: if you disable NAT, how should it connect to your provider? cable->modem->ethernet card of asterisk box? |
| 23:55.06 | phunyguy | yeah disabling NAT makes that address public. |
| 23:55.08 | gregd | ethernet |
| 23:55.20 | phunyguy | ..even if it is a private range |
| 23:55.24 | phunyguy | most ISPs block that |
| 23:55.41 | Chotaire | gregd: you are not directly connected to the internet because your ISP only allocates one dynamic ip address for you. |
| 23:56.17 | gregd | ok, but every request to this address is passed over to asterisk box |
| 23:56.21 | Chotaire | gregd: a direct connection means you terminate the dsl connection in your ethernet card of the asterisk box and use whatever pptp method to authenticate to your ISP. |
| 23:56.48 | Chotaire | gregd: it still needs to be NAT'ed because your provider does not know your private address range and won't route it. |
| 23:57.20 | gregd | ok, will change it back now |
| 23:57.20 | phunyguy | might as well plug your DSL right to the asterisk box. |
| 23:57.32 | phunyguy | it would work then, but nothing else will |
| 23:57.45 | Chotaire | phunyguy: that only works if he has a dsl modem. |
| 23:57.57 | phunyguy | what is he using? |
| 23:58.03 | phunyguy | internal modem? |
| 23:58.11 | Chotaire | or a dsl router that can be set to forward dsl authentication to another port on the lan. |
| 23:58.22 | Chotaire | he's using a dsl router. |
| 23:58.38 | phunyguy | ahhh... most can be configured to passthrough PPPoE |
| 23:58.46 | phunyguy | bridged mode |
| 23:58.54 | Chotaire | it's not too many at all ;( |
| 23:59.07 | phunyguy | all of mine did, even the all in one routers |
| 23:59.25 | Chotaire | fritzbox can do pppoe passthrough and/or even act as a client.. that's how it should be done. |
| 23:59.30 | phunyguy | could plug a PC into any port on it, and have that PC do the PPP auth |
| 23:59.47 | phunyguy | or in my ISPs case, set it up as a layer 2 bridge, and have the modem as DHCP |
