| 00:00.39 | jblack | Shuttleworth made a disturbing statement the other day that he can afford to support canonical for another 3 years or so. |
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| 00:01.06 | jblack | 3-5. |
| 00:01.17 | jblack | wonders what may happen in 2011 |
| 00:01.42 | Katty | ello. |
| 00:02.17 | gcbirzan | My money's on an asteroid hitting Earth |
| 00:02.49 | rob0 | It's life, Jim, but not as we know it. |
| 00:03.17 | jaytee | Katty: mew |
| 00:03.55 | Katty | jaytee: ohai |
| 00:04.44 | jaytee | jblack, I read that. I think what he meant was that it would take another 3 to 5 years for Canonical to become successful enough to not require funding from other sources. |
| 00:07.11 | garywsmith | sadly I abandoned redhat because of version 3 using their hybrid 2.4/2.6 kernel. |
| 00:07.39 | jaytee | Katty I told one of my friends who named her dog Frell what you named your puppy. I think she's still laughing and insane with jealousy at this point and that was hours ago. |
| 00:09.46 | garywsmith | it would be really nice if digium brought their yum repo back online sometime today... |
| 00:10.19 | Qwell | garywsmith: what are you talkingabout? |
| 00:10.22 | Qwell | it's up.. |
| 00:10.43 | garywsmith | http://packages.digium.com/centos/5/current/i386/repodata/repomd.xml: [Errno 4] IOError: <urlopen error (111, 'Connection refused')> |
| 00:10.56 | Qwell | oh, that repo |
| 00:10.59 | Qwell | ...yeah |
| 00:11.36 | jaytee | oooooohhh! thaaaat repo! ooops! |
| 00:12.05 | Qwell | garywsmith: it was being reinstalled last night.. not sure what happened |
| 00:12.33 | jaytee | "Hey, Rocky! Wanna see me pull a repo outta my hat?" "Bullwinkle, that trick never works!" |
| 00:14.58 | garywsmith | Qwell: well, I know it's not up today ;) |
| 00:16.20 | jaytee | maybe they don't "roll" on Shabbos. :-) |
| 00:18.00 | Qwell | good lord.. do not try to type long server names with T9 |
| 00:18.14 | Katty | jaytee: oh? why? |
| 00:18.39 | jaytee | Katty because you're dog's name is soo cool |
| 00:19.54 | garywsmith | jblack: why were you wondering what will happen in 2011? |
| 00:20.14 | garywsmith | oh, never mind. |
| 00:20.19 | Katty | jaytee: ah. |
| 00:20.24 | Katty | jaytee: riddick was rather... |
| 00:20.29 | Katty | jaytee: well, i'm a female. |
| 00:20.32 | Katty | jaytee: you get my drift. |
| 00:21.32 | jaytee | Katty, I guess so but I gave her the FQDN, "Fully Qualified Doggy Name" of Kaiser Riddick der Kleine mit Waggytail. |
| 00:22.11 | Katty | oh. |
| 00:22.14 | Katty | that's not quite right |
| 00:22.22 | Katty | Kaiser Riddick der Kleine Hobbit mit Waggytail |
| 00:22.22 | jaytee | and you have to figure any girl who names her dog Frell (slang for fuck in Farscape) has to have a sense of humor, warped though it might be :-) |
| 00:22.26 | Katty | due to his brothers being 'hobbits' |
| 00:22.34 | Katty | Merry Pippin Sam and Shire, the ferrets |
| 00:22.38 | jaytee | ah, I left out the hobbit part here but not when I told her |
| 00:22.45 | Katty | cheers (= |
| 00:23.40 | Katty | Ryan was amused by Frell as well |
| 00:23.44 | Katty | your friend, gets two kudos. |
| 00:23.55 | Katty | also, i have new pup pictures. |
| 00:24.02 | Katty | we had a fun day at mom's house raking leaves |
| 00:24.58 | jaytee | jblack, in 2011 the LHC will open a rip in the space/time continuum to another dimension allowing Mr Mxyzptlk to cross over. Since Superman isn't here to defeat him we'll be in for an eternity of mischief and mayhem. |
| 00:29.08 | garywsmith | I was thinking about December 21, 2012. That's when we try to schedule all of the project completion dates. Either it ends that day or we find new jobs :) |
| 00:31.31 | jaytee | or Windows 7 will be a true Linux killer and Microsoft OCS will dominate the VOIP industry and I'll be selling oranges on a median strip in downtown L.A. |
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| 00:33.44 | jaytee | [TK]D-Fender, you still here? |
| 00:34.08 | [TK]D-Fender | jaytee: not for long |
| 00:34.19 | jaytee | can I PM you for just a sec? |
| 00:34.55 | [TK]D-Fender | quick |
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| 01:23.36 | garywsmith | I take it there was some type of server crash. |
| 01:25.47 | subdolus | correcto |
| 01:26.24 | subdolus | been a few lately |
| 01:26.29 | subdolus | still not as bad as austnet |
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| 01:33.40 | garywsmith | yay, the repo is finally back online |
| 01:39.25 | mankash | I am not able to connect through asterisk -r, getting an error |
| 01:39.43 | jblack | check to see that asterisk is working. |
| 01:40.11 | mankash | how to check |
| 01:41.06 | jblack | ps aux | grep asterisk |
| 01:42.06 | mankash | I already did that |
| 01:42.17 | mankash | it is showing too many lines for that |
| 01:44.34 | ManxPower | mankash: Are you root? |
| 01:45.48 | mankash | I have root credentials too |
| 01:46.56 | mankash | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 01:47.03 | ManxPower | does it exist? |
| 01:47.18 | ManxPower | ls -l /var/run/asterisk.ctl |
| 01:47.20 | mankash | I guesss asterisk doesn't have rights to write into /var/run |
| 01:47.31 | ManxPower | that file is what Asterisk uses to communicated with the console (asterisk -r) |
| 01:47.33 | mankash | yes but it is 0 bytes |
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| 01:47.39 | ManxPower | that is fine. |
| 01:47.49 | mankash | it is there |
| 01:47.50 | ManxPower | I believe it's a FIFO or other filesystem based IPC |
| 01:47.57 | ManxPower | who owns the file? |
| 01:48.10 | mankash | ok |
| 01:49.11 | mankash | even loggin with root also has same error |
| 01:51.41 | ManxPower | Who owns the file? |
| 01:52.12 | ManxPower | The ls command I gave you above would have shown you the ownership and group of the file. |
| 01:53.00 | mankash | user "asterisk" |
| 01:53.30 | mankash | may be bcoz it is zero bytes' |
| 01:53.54 | ManxPower | I'll go log into a production system and check if it will make you feel better. |
| 01:54.49 | jblack | mankash: It's just a lock file. It's existance is what matters, not it's contents. |
| 01:54.58 | ManxPower | [root@pbx-1 ~]# ls -l /var/run/asterisk.ctl |
| 01:54.59 | ManxPower | srwxr-xr-x 1 root root 0 Oct 29 13:19 /var/run/asterisk.ctl= |
| 01:55.10 | ManxPower | jblack: I believe it's actually a Named Pope |
| 01:56.04 | jblack | Oh, I'm sorry. I was thinking of a different daemon. Yeah. asterisk.ctl is a named pipe. |
| 01:56.26 | ManxPower | so it would only be a problem if it is NOT 0 bytes |
| 01:57.23 | jblack | I've never seen a named pipe have a size other than 0. I could check the standard if we care. |
| 01:57.37 | ManxPower | jblack: *nod* they Just Work |
| 01:57.48 | jblack | I suppose it _might_ be possile that the "size" reflects unread data in the buffer. |
| 01:57.58 | ManxPower | jblack: that's what I was thinking. |
| 01:58.22 | jblack | do we care enough for me to plud up two flights of stairs? |
| 01:58.32 | ManxPower | jblack: gads no! |
| 02:00.41 | jblack | yeah. I'm mildly curious enough to check posix via google after a smoke. But to change my x,y,z coordinates 28' on the z axis.... nah |
| 02:03.55 | jblack | cmere google. |
| 02:07.17 | jblack | Looks like the size reflects the number of messages waiting. |
| 02:07.37 | jblack | http://www.users.pjwstk.edu.pl/~jms/qnx/help/watcom/clibref/mq_overview.html |
| 02:08.06 | jblack | I'm not entirely certain that a "message queue" and a named socket are idempotent, though. |
| 02:08.39 | jblack | pardon, synonymous. |
| 02:09.38 | jblack | No, they're something different. |
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| 02:21.30 | tzafrir_laptop | http://lwn.net/Articles/306364/ |
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| 02:49.34 | plasmid | WOuld anyone please recommend a list of VOIP providers for home use here in the USA? |
| 02:51.03 | seanbright | ~itsp-us |
| 02:51.08 | seanbright | ~itsp |
| 02:51.09 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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 |
| 02:53.05 | plasmid | ~itslplist-us |
| 02:53.37 | seanbright | plasmid: /msg jbot ~itsplist-us |
| 02:53.54 | plasmid | why thank you. |
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| 03:19.46 | MindTheGap | hello all, can somebody please help me with this? http://www.pastebin.ca/1249246 |
| 03:19.56 | MindTheGap | HDLC Bad FCS/ Abort |
| 03:20.13 | MindTheGap | pri intensive debug in the pastebin |
| 03:21.08 | MindTheGap | asterisk1.6.0.1, libpri1.4.7, dahdi2.00. |
| 03:23.16 | MindTheGap | cat /proc/interrupts, hdparm -i, lsmod output here: http://www.pastebin.ca/1249249 |
| 03:26.34 | MindTheGap | incomming and outgoing calls are fine but it will Alarm several times a day, hanging up all calls |
| 03:30.02 | [netman] | find out if your hardware is loosing interrupts |
| 03:32.23 | MindTheGap | [netman], cat /proc/interrupts do not show anything, dahdi_test looks great, 99,994 average... |
| 03:33.39 | MindTheGap | bumping up the te110p latency and lowering the others have no effect... |
| 03:34.29 | MindTheGap | ive tried a a lot of things... its driving me insane. |
| 03:35.35 | MindTheGap | had the telco change the modem, had asterisk machine and modem grounding verified, had the telco test the line, nothing works. |
| 03:36.01 | MindTheGap | funny thin is it started some 2 months ago |
| 03:36.36 | MindTheGap | some days it will alarm 10 times a day, sometimes none. |
| 03:37.22 | MindTheGap | had changed the te110p, changed the hardware, nothing will make it go away. |
| 03:37.24 | jblack | bad switch at the telco? |
| 03:40.13 | MindTheGap | jblack, how will I argue this if on our site their instrument will show a huge OK on its LCD and the guy will say: - i'll let it run for another 30 minutes. and after that it shows another OK at the lcd? i mean, i know computers, not telecom, ISDN is something i just know the name. :) |
| 03:41.59 | jblack | wtf. diamondcard uses 30 second billing increments? |
| 03:42.02 | MindTheGap | jblack, will a bad switch there show up in a loop run trhough the line? (thats how they will test it) |
| 03:42.22 | jblack | MindTheGap: I don't know. I just threw out an idea. |
| 03:42.57 | ManxPower | A TE110P is NOT a modem! |
| 03:43.30 | MindTheGap | hi ManxPower, what do you mean? |
| 03:44.16 | ManxPower | I suggest you ask the telco to "loop the dsu", which should tell the Asterisk card to run a loopback. Then while you have them on the phone and saying there are no problems, unplug the line from Asterisk. Chances are they are looking at the wrong line and will realize it when you tell them you just unplugged the line. |
| 03:44.43 | ManxPower | 'cause they can't see "nothing wrong" if you have the line unplugged. |
| 03:45.44 | ManxPower | HDLC Aborts are usually caused by some device locking interrupts for too long of a time. IDE and SATA contollers do this frequently, So do some RAID and GigabitEthernet devices |
| 03:47.11 | MindTheGap | ManxPower, but it used to work before, same hardware. nothing changed. it just started aout of the blue. |
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| 03:48.07 | MindTheGap | actually ive also changed hardware (also working in another asterisk install) and it shows the exact same problem on this line |
| 03:48.48 | ManxPower | MindTheGap: I know this is not the answer you want to hear, but I had a system that would only get HDLC Aborts when there was more than a tertian amount of disk activity. |
| 03:48.51 | [netman] | I bet you should blame it on your telco |
| 03:49.02 | ManxPower | or the cable from the smartjack to Asterisk |
| 03:49.37 | MindTheGap | ManxPower, for the loop thing i had them hook their instrument on the coax out on the modem and they would get the loop from the telco and vice versa. |
| 03:49.53 | MindTheGap | for the cable, its been changed too... |
| 03:50.02 | MindTheGap | you see, im desperate... :) |
| 03:50.09 | ManxPower | OK. Now we have a problem. You must have an E-1, not a T-1. |
| 03:50.18 | ManxPower | 'cause T-1s don't come in on "coax" |
| 03:50.19 | MindTheGap | yes, E1 |
| 03:50.41 | ManxPower | then that coax goes into come converter which converts it to twisted pair, which is then plugged into Asterisk. |
| 03:50.46 | MindTheGap | its an optical modem, its got both outputs, coax and rj45 |
| 03:50.58 | ManxPower | It is not a modem. |
| 03:51.02 | MindTheGap | its plugged w the rj45 |
| 03:51.36 | ManxPower | I don't care if it sits up and sings Beoyance. It's not a modem. |
| 03:51.51 | carrar | haha |
| 03:51.59 | carrar | That would be impressive however |
| 03:52.00 | ManxPower | In the T-1 world we call "the thing from the telco the customer plugs into" a "smartjack" |
| 03:52.27 | carrar | RJ48C typically |
| 03:52.31 | MindTheGap | whatever, the thing from the telco the customer plugs into" a "smartjack" has been tested and changed |
| 03:52.33 | ManxPower | If the telco central office loops the line then they should see when you unplug the line. |
| 03:52.45 | ManxPower | if they can't see that then chnances are they are testing the wrong line. |
| 03:52.54 | MindTheGap | ManxPower, yes we did it |
| 03:53.01 | MindTheGap | they saw |
| 03:53.05 | jblack | If a smartjack sang beyonce, I bet the title of the song would be "fill my hole" |
| 03:53.23 | ManxPower | The test I am referring to does NOT require telco people at the customer location. |
| 03:53.26 | carrar | is that even a Beoyance song? |
| 03:53.36 | carrar | I don't listen to them |
| 03:53.36 | jblack | could be, but isn't. |
| 03:53.39 | ManxPower | The test I am referring to can only be done by the equipment in the telco office. |
| 03:54.02 | ManxPower | has a vision from The Onion Movie |
| 03:54.10 | MindTheGap | allright you are referring to a loop test, right? |
| 03:54.17 | carrar | Always carry around a loopback plug |
| 03:55.15 | ManxPower | there are several kinds of loop tests |
| 03:55.34 | ManxPower | but yes, a loopback plug/jack could be used as well |
| 03:55.49 | MindTheGap | haw it that different from a loop analysed onsite? |
| 03:58.37 | MindTheGap | they would send their ppl and the guy would hook ul their equippment on the back of the "thing that came from the telco" and on the cell phone would say, close the loop please... got it... moments after the equippment, (like a huge scientific calculator) would show a big OK on screen. then they will do the same test on the far ent of the link w another person ant the loop would show up as well and also the ok... |
| 03:58.50 | troy- | can asterisk receive SMS over a standard POTS line with FX0 card? |
| 03:59.32 | carrar | troy |
| 03:59.37 | ManxPower | troy-: in most of Europe yes. |
| 03:59.39 | carrar | What did TK tell you |
| 03:59.45 | carrar | heh |
| 03:59.53 | troy- | ^^ which is why i'm asking |
| 04:00.14 | ManxPower | The validity of an answer is not a function of how much you like that answer. |
| 04:00.28 | MindTheGap | ManxPower, if i loop the e110p jack and the jack is deffective will it show the same messages on the log? BAD FCS and Aborts? |
| 04:00.48 | ManxPower | MindTheGap: My POINT is that the telco sometimes tests the WRONG line. |
| 04:00.52 | troy- | carrar, do you know what functionality has been implemented in Europe which allows you to send/receive over PSTN? |
| 04:00.57 | ManxPower | My advice was to try to TRAP them if they do that. |
| 04:01.21 | ManxPower | troy-: the cellular/mobile phone companies have this as part of their service. |
| 04:01.57 | ManxPower | As there are no cellular companies in the USA or Canada that permit random people to connect to their SMS service. |
| 04:02.09 | MindTheGap | ManxPower, man, they are testing the right line, i got ppl several times here this week, the e110 will Alarm as soon as they close the loop.. |
| 04:02.13 | carrar | Sorry, I don't live in Europe, you will need to consult your local providers |
| 04:02.24 | ManxPower | now, if you just want to send SMSs between two Asterisk servers using the PSTN that sould work just nifty. |
| 04:02.47 | ManxPower | But don't expect any company in the USA to permit you do connect to them using the app_SMS |
| 04:03.07 | troy- | ManxPower, i'm just curious what prevents me from being SMS (receive) capable over the PSTN? |
| 04:03.18 | ManxPower | troy-: nothing at all. |
| 04:03.32 | ManxPower | set it up in Asterisk and you can receive all the SMS calls you want. |
| 04:03.42 | ManxPower | I can't think of anything that would actually call you, however. |
| 04:04.07 | troy- | ManxPower, meaning when someone tries to send an sms it wont reach me? |
| 04:04.31 | ManxPower | troy-: When someone tries to send you an SMS *from where/what device*? |
| 04:04.58 | troy- | someone on a cellular network which isn't the same carrier as who I have the POTS / PRI circuit with |
| 04:05.16 | ManxPower | troy-: what country are you in, what country is the caller in? |
| 04:05.26 | troy- | Canada <--> Canada |
| 04:05.50 | ManxPower | not a chance in hell of getting a cell phone company to send that SMS over the PSTN using the SMS protocol. |
| 04:06.07 | Superbartt | in the netherlands that happens... |
| 04:06.12 | ManxPower | They have the technology to do it. They just do not offer that service. |
| 04:07.03 | troy- | ManxPower, why does the originating carrier care what method is used to transmit the SMS (cellular-->cellular, cellular-> PSTN) etc. |
| 04:07.28 | ManxPower | troy-: because they have to run, buy, manage a gateway to translate between technoloogies. |
| 04:07.52 | ManxPower | they do not offer that service. It is as simple as that. You can ask in 500 zillion different ways, but the answer will always be the same. |
| 04:08.14 | Superbartt | sms is like a build in transmit feature of the cell-network. pstn doesn't has that so they need specific hardware to get that, which = money : |
| 04:08.29 | troy- | ah okay, so the originating carrier is able to determine the technology of the destination line and make a decision accordingly |
| 04:08.54 | ManxPower | troy-: obviously the originating carrier COULD if they wanted to. The Euro telcos do it. |
| 04:09.08 | troy- | ManxPower, i just dont understand how it works which is why i'm asking :< |
| 04:09.23 | ManxPower | troy-: PSTN SMS is a 2400 baud modem data burst. |
| 04:09.45 | ManxPower | SMS over cellular network uses network messaging -- don't ask me the protocol, I don't know. |
| 04:10.38 | Superbartt | ManxPower as far as i know it's sent over some network signalling method, and as it got populair telco's started to exploit it :p |
| 04:11.04 | troy- | got it, so when a mobile user attempts to send a message that carrier's message center attempts to negotiatiate a protocol with the receiving party which in POTS case isnt compatible |
| 04:11.53 | drmessano | SMS over POTS not the same as the CallerID data? |
| 04:11.56 | Superbartt | well, it's already sent to a server, as you need to fill in an sms-gateway in your cellphone |
| 04:12.08 | *** join/#asterisk giovani (n=giovani@unaffiliated/giovani) |
| 04:12.48 | ManxPower | drmessano: different protocol. |
| 04:12.52 | troy- | Superbartt, thats normally preset to your carrier you have service with |
| 04:12.57 | drmessano | ok |
| 04:13.05 | drmessano | Interesting |
| 04:13.05 | ManxPower | troy-: in YOUR country it's pre-set. |
| 04:13.12 | drmessano | Seems like there would be some reuse there |
| 04:13.20 | ManxPower | drmessano: I'd have to dig into app_sms.c to know more |
| 04:13.23 | Superbartt | yes it's pre-set most of the times |
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| 04:13.33 | troy- | ManxPower, ah thanks for the correction |
| 04:14.02 | ManxPower | troy-: the USA and Canada are 3rd world countries when it comes to telecom |
| 04:14.13 | ManxPower | as compared to much of the EU |
| 04:14.19 | drmessano | Well, you said it all when you mentioned 2400baud.. U.S. CID is Bell202, 1200baud, 2200hz/1200hz tones |
| 04:14.52 | drmessano | Of course, I don't know what they use in the EU, but anyway |
| 04:15.27 | troy- | ManxPower, haha yeah. My lack of knowledge is really in the 'transactional process' occurring between the send and receive function between carriers and mediums |
| 04:15.51 | ManxPower | http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fwww.rtx.dk%2FAdmin%2FPublic%2FDWSDownload.aspx%3FFile%3D%252FFiles%252FFiler%252Ftekniske%2Bartikler%252FSMStransmissionwithinthePSTN.pdf&ei=uGMWSb37HZSY8wSDip3uCg&usg=AFQjCNHdp0m9FtkJ0lxyrqvVLBkMlyFiwQ&sig2=g7PCXgvxV9_951IYW9I8qw |
| 04:15.54 | drmessano | I think they use MGFY for carrier interchange |
| 04:16.39 | drmessano | "Maybe Go F*** Yourself" |
| 04:16.44 | drmessano | Maybe not :/ |
| 04:16.58 | ManxPower | I wonder if app_SMS could be used as an InterAsterisk Messaging thing. |
| 04:16.59 | drmessano | Which is about how telco's in the US work |
| 04:17.30 | drmessano | ManxPower: Without an additional acronym? |
| 04:17.42 | ManxPower | It's a 1200 baud FSK burst |
| 04:18.51 | drmessano | I wonder what modulation scheme |
| 04:19.03 | ManxPower | I believe FSK is the modulation scheme. |
| 04:19.19 | ManxPower | I'm a bit rusty with the modem protocols these days |
| 04:20.35 | ManxPower | "Data is transmitted on the physical layer using 1200 Baud FSK modulation within a traditional voice-band call. This means that the terminal hardware must be capable of sending and receiving 1200 Baud FSK according to ETSI standard for off hook data transmission. " |
| 04:20.53 | drmessano | V.23 |
| 04:22.13 | drmessano | 1200baud FSK w/ 1300hz/2100hz tones |
| 04:23.41 | troy- | drmessano, once a cell user's SMSC has stored the message how do they initiate the transaction to the receiving party's SMSC? |
| 04:23.53 | drmessano | why me? |
| 04:23.58 | drmessano | I know nothing about this |
| 04:24.03 | troy- | haha |
| 04:24.54 | drmessano | I'm just an old bitbanger who thinks 9600 baud is "damn slick" |
| 04:25.39 | troy- | my accoustic coupler is far from slick ;) |
| 04:25.53 | ManxPower | troy-: Who is "they"? The SMSC? |
| 04:26.31 | troy- | ManxPower, the originating SMSC |
| 04:26.44 | ManxPower | drmessano: I spent a year working at Symantec doing pcANYWHERE (for DOS) tech support. |
| 04:28.44 | jaytee | ManxPower, can I PM you for a sec? |
| 04:28.50 | ManxPower | troy-: They send it on to the destination SMSC (mobile). I imagine they just do an outgoing call for a PSTN destination |
| 04:28.59 | ManxPower | jaytee: does it involve sending me money? |
| 04:29.22 | jaytee | well, not me but it involves the opportunity to make some |
| 04:29.38 | ManxPower | jaytee: you've saved up enough good karma it doesn't matter. go ahead. |
| 04:29.52 | drmessano | ManxPower: I am sorry you had to go through that |
| 04:30.03 | ManxPower | troy-: remember this stuff all applies to the european telcos, not to usa or canada |
| 04:30.42 | ManxPower | drmessano: It was interesting. |
| 04:30.59 | ManxPower | jaytee: pm away |
| 04:31.37 | troy- | ManxPower, gotcha - so basically if the originating SMSC checks it's call route table and realizes the destination isnt mobile - an error is returned to the user? |
| 04:32.10 | ManxPower | troy-: Well Verizon Wireless sends you back a text message offering to call the land line and speak the message. |
| 04:33.19 | troy- | understood, the only question remaining is what flag/identifier differentiates POTS from mobile and how do you set it :P |
| 04:34.15 | ManxPower | troy-: that is all SS7 stuff. |
| 04:34.38 | troy- | gotcha, thanks for putting you through all that ;) |
| 04:40.55 | carrar | had dinner, what did I miss? |
| 04:47.41 | troy- | i learned that SS7 is used by the originating SMSC to determine if the destination is SMS capable |
| 04:49.30 | troy- | FYI - http://www.ncs.gov/library/tech_bulletins/2003/tib_03-2.pdf |
| 04:55.14 | carrar | blah |
| 04:55.17 | carrar | ss7 is old news |
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| 05:28.15 | mankash | how to change the location of asterisk.pid from /var/run to /var/run/asterisk |
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| 05:36.23 | jameswf | http://web.archive.org/web/20020926094908/http://www.digium.com/ |
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| 07:01.30 | drmessano | hmmm |
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| 07:26.06 | ManxPower | mankash: /etc/asterisk/asterisk.conf |
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| 07:41.13 | ManxPower | It is SO nice to be able to download as much as I want without worrying about exceeding my quota |
| 07:42.09 | Mark_Logan | Quota? eeeewwwww. |
| 07:42.32 | ManxPower | Mark_Logan: There are tradeoffs to living in a rural area. |
| 07:43.00 | ManxPower | choices are (in order of preference) Dialup, Satellite, Verizon EVDO |
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| 07:59.04 | tzafrir_laptop | mankash, edit asterisk.conf |
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| 09:51.53 | trentster | hey all, I have a few iax2 trunks connecting to a central asterisk box, if i do an iax2 show peers most of the peers show port 4569, but a couple of them show other ports, why is this, and how do I force those trunks to 4569...any ideas? |
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| 10:06.21 | trentster | bump |
| 10:16.31 | Mark_Logan | bumpity. |
| 10:17.02 | subdolus | boomp |
| 10:24.14 | tzafrir_laptop | trentster, any chance those "others" are behind NAT? |
| 10:24.39 | tzafrir_laptop | anyway, iax can be run on any port. It does not have to use 4569 |
| 10:27.12 | trentster | tzafrir_laptop, yes, they are all behind firewalls and nat... |
| 10:28.05 | tzafrir_laptop | One thing that could be bad is if they register with one port and then call from another |
| 10:28.39 | tzafrir_laptop | sip.conf has insecure.port for that . but chan_iax2 does not have anything equivalent |
| 10:30.02 | trentster | hmmm, its very strange, we can talk and dial each other down the trunk, and I have even locked down the src and destination both incoming and outgoing to 4569 and it still shows a high port number yet we can talk etc... |
| 10:31.11 | trentster | I can even see block in the log coming from remote iax2 peer trying the port number |
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| 10:49.42 | Cybertoy | anyone know why ast_get_srv messages haved moved from verbose level 4 to 3 between versions 1.4.x and 1.6.x? |
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| 13:29.31 | hi365 | is there anyway to see the asterisk-addons version from the cli? |
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| 13:53.26 | getsql08 | does anyone know teliax rates for USA - USA Calling |
| 13:53.35 | getsql08 | its not listed on their rates page of https://www.teliax.com/RatesPage |
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| 14:05.13 | KP7 | i have a paygo account with Teliax |
| 14:05.27 | KP7 | its $0.02 for US calls |
| 14:05.34 | getsql08 | thats expensive |
| 14:05.48 | getsql08 | Their wholesale package is |
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| 14:05.53 | getsql08 | 0.007 cents |
| 14:05.55 | getsql08 | How do you rate calls? Rates for all destinations are included in the rate deck. International calls are “cost +”. Domestic US calls are OCN LATA and broken into Local, Interstate, Intrastate and “Zone 1” catagories. Local calls will be the least expensive averaging $0.005. To qualify as a local call the OCN LATA of the ANI and CPN (called party) must be local according to LERG definition. A good guide to what is considered local can be found at local |
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| 14:06.09 | KP7 | $0.02/min is expensive? |
| 14:06.57 | getsql08 | when your doing 50,000 calls |
| 14:06.58 | getsql08 | it is |
| 14:07.44 | KP7 | tru |
| 14:07.52 | getsql08 | KP7 - are you a developer |
| 14:07.55 | getsql08 | or user of aestrix |
| 14:07.59 | KP7 | btw, you have to go into plan detail for Teliax |
| 14:08.02 | KP7 | https://www.teliax.com/plans/4? |
| 14:08.07 | KP7 | give you more detail |
| 14:08.12 | KP7 | i'm a user |
| 14:08.40 | KP7 | i did help on a patch last night tho :) i feel official |
| 14:09.03 | KP7 | did you have a question |
| 14:09.04 | getsql08 | lol |
| 14:09.08 | getsql08 | ok thanks for the rates question |
| 14:09.11 | KP7 | np |
| 14:09.14 | getsql08 | no i wanted to know if you could actually code |
| 14:09.17 | getsql08 | for aestrix |
| 14:09.39 | getsql08 | did aestrix setup some sort of admin web interface yet |
| 14:09.47 | KP7 | nah, I'm a programmer but i have done anything for * |
| 14:09.49 | KP7 | yea |
| 14:09.55 | getsql08 | you know the link |
| 14:09.56 | KP7 | there's a web admin |
| 14:10.01 | getsql08 | so i can see screen shots |
| 14:10.13 | KP7 | go to www.asterisk.org |
| 14:10.22 | KP7 | and follow the asterisknow links |
| 14:10.51 | KP7 | or hit http://www.voip-info.org/wiki/index.phttp://www.voip-info.org/wiki/index.php?page=Asteriskhp?page=Asterisk |
| 14:11.00 | KP7 | and start your search from there |
| 14:12.53 | KP7 | http://www.voip-info.org/wiki-Asterisk+GUI <-- actually, its there |
| 14:13.11 | getsql08 | thanks |
| 14:13.13 | getsql08 | i'm checking it out |
| 14:13.17 | KP7 | np |
| 14:14.44 | tzafrir_laptop | KP7, what's up? |
| 14:14.50 | Cybertoy | I find voicetrading.com pretty cheap |
| 14:15.20 | KP7 | hey tzafrir_laptop you know what after all that, it was a outage at my ITSP |
| 14:15.36 | KP7 | *laff* |
| 14:16.26 | Cybertoy | I don't qualify to use them though ... but I guess if you do 50000 calls to the USA you should. |
| 14:16.26 | Cybertoy | they're at 0.0069 |
| 14:16.26 | tzafrir_laptop | getsql08, asterix is a comics character . I believe you use asterisk |
| 14:17.08 | KP7 | getsql08: actually another thing on that tho i would be as concerned with reliability (where your ITSP peers, redudancy,etc) as I am with price |
| 14:18.00 | KP7 | at 50,000 mins that is a lot of volume, you want to make sure you have a good provider as well as some redudancy yourself |
| 14:18.10 | getsql08 | I was thinking about going with |
| 14:18.21 | getsql08 | teliax.com |
| 14:18.53 | getsql08 | what do you mean by redudancy myself? |
| 14:19.08 | getsql08 | oh ok |
| 14:19.12 | KP7 | you might want to check out http://www.bandwidth.com/ |
| 14:19.19 | getsql08 | Cybertoy - 0.0069 is fairly cheap |
| 14:19.44 | KP7 | i use teliax for my conference bridges but guess what? 2 of their locations are down |
| 14:19.44 | getsql08 | its more like 100,000 - 200,000 / minutes per week |
| 14:19.51 | getsql08 | those are low estimates |
| 14:20.04 | KP7 | so i had to register to another one |
| 14:20.10 | KP7 | but that was on me to do |
| 14:20.17 | KP7 | and they're not open on the weekends |
| 14:20.26 | KP7 | geez |
| 14:20.57 | getsql08 | hrm, i've just installed the aestrix module on a tesat linux server |
| 14:21.07 | getsql08 | are there any docs in regards to configuration of the GUI |
| 14:21.09 | KP7 | thats a <one of those 7 words deleted> -load ! |
| 14:21.28 | KP7 | configure via the gui? |
| 14:21.40 | KP7 | or of the gui itself |
| 14:22.06 | Cybertoy | getsql08: I use some of their offers aimed at consumers like voipcheap.com or intervoip.com ... and they have excellent quality and I never had an availability problem with them. |
| 14:23.00 | getsql08 | Kp7: I mean configuring the GUI itself |
| 14:23.07 | getsql08 | I've just installed aestrix via cmd prompt on this box |
| 14:23.16 | getsql08 | no idea how to access the GUI |
| 14:23.40 | getsql08 | Cybertoy: who are you refering to? intertix? |
| 14:23.55 | KP7 | its doesn't come with one (as far as i know) you'll have to dl one and install that |
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| 14:29.05 | *** join/#asterisk chigital (n=chigital@tmo-100-77.customers.d1-online.com) |
| 14:31.26 | Cybertoy | getsql08: no... voicetrading.com |
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| 14:34.26 | *** join/#asterisk synchris (n=synchris@athedsl-86032.home.otenet.gr) |
| 14:34.57 | tzafrir_laptop | getsql08, there are several gui-s for Asterisk . But you don't have to use any |
| 14:35.00 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 14:35.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:35.11 | tzafrir_laptop | Some are tricky to install |
| 14:35.26 | tzafrir_laptop | What are you trying to do? |
| 14:36.09 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 14:41.51 | KP7 | <getsql08>what do you mean by redudancy myself? <-- I was talking about how you're going to build your network i.e. breadth/depth, hot-swap abilities, service continuity, etc |
| 14:45.04 | getsql08 | I was just going to use |
| 14:45.05 | getsql08 | aestrix |
| 14:45.10 | getsql08 | link to a SIP provider |
| 14:46.18 | *** join/#asterisk dexpdx (n=jason@75-164-245-74.ptld.qwest.net) |
| 14:46.20 | dexpdx | does anyone have any points on how one would setup outbound queing for calls orignated from spool/outgoing |
| 14:46.27 | dexpdx | s/points/pointers |
| 14:48.14 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 14:50.33 | KP7 | getsql08: right but what if 1) your asterisk box crashes or 2) you ITSP has an interruption |
| 14:51.01 | KP7 | whats the allowable outage window for you (or your client)? |
| 14:51.21 | getsql08 | i see |
| 14:51.28 | KP7 | just things to consider |
| 14:51.29 | getsql08 | so i should get backups |
| 14:51.39 | getsql08 | different provider |
| 14:51.41 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 14:51.47 | getsql08 | duplicate aestrix settings |
| 14:51.51 | getsql08 | in different data centers |
| 14:52.19 | KP7 | not realy related to asterisk but you'll find that asterisk since is runs on LInux will provide for things that you can't do elsewhere |
| 14:52.23 | KP7 | right |
| 14:52.25 | KP7 | there ya go |
| 14:52.28 | KP7 | :) |
| 14:52.35 | KP7 | things like that |
| 14:52.39 | getsql08 | oh |
| 14:52.42 | getsql08 | i got that covered |
| 14:52.48 | getsql08 | just need an aestrix coder |
| 14:53.14 | KP7 | what r you trying to do in * that is not there? |
| 14:53.47 | getsql08 | call a list of numbers in a static file |
| 14:53.52 | getsql08 | play greeting from a sound file |
| 14:53.58 | getsql08 | thats it |
| 14:55.29 | KP7 | sounds like a simple agi |
| 14:55.38 | getsql08 | it will be more complex |
| 14:55.39 | getsql08 | but |
| 14:55.42 | getsql08 | thats the jist of it |
| 14:55.49 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 14:56.00 | getsql08 | where can i find some agi developers |
| 14:56.09 | KP7 | chapter 9 for the asterisk book talks about that |
| 14:56.38 | KP7 | i was going to take a crack at doing something to lookup weather by zip code |
| 14:56.43 | KP7 | no sure |
| 14:56.47 | KP7 | i guess you could ask here |
| 14:57.04 | getsql08 | everyone seems to be dead |
| 14:57.11 | KP7 | LOL |
| 14:57.17 | KP7 | idle me thinks... |
| 14:57.32 | KP7 | someone is always lurking |
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| 14:57.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:58.07 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:58.13 | KP7 | if you can't hang out here, i'd suggest posting on a list or google around |
| 14:58.51 | KP7 | i'd offer but i've got to learn this for myself too |
| 14:59.06 | KP7 | no enough hours in day |
| 14:59.11 | Maliuta | there is normally someone that flicks past and helps on what they can |
| 14:59.27 | Maliuta | but if you're not here when it's active the best place the users mailing list |
| 14:59.48 | KP7 | see getsql08 :) |
| 15:00.04 | KP7 | continues to sip coffee |
| 15:00.10 | Maliuta | yeah, I don't do telemarketer crap anymore |
| 15:00.12 | KP7 | err, no pun intended! |
| 15:00.13 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 15:00.25 | Maliuta | did a thing well before * was a reality, not worth it |
| 15:00.43 | KP7 | i shoot telemarkers, spammers and robocallers on site |
| 15:00.54 | KP7 | and i approve that message >:) |
| 15:00.56 | KP7 | LOL |
| 15:01.06 | Maliuta | KP7: no, you've got to torture them so that the know they have done wrong |
| 15:01.11 | KP7 | Maliuta: what did you use back then? |
| 15:01.40 | Cybertoy | use the script on www.whocalled.us ... you'll never have a telemarketer call you again |
| 15:01.43 | KP7 | Maliuta: i ain't say I'd kill them- use 'em for target practice ;) |
| 15:01.54 | Maliuta | KP7: I ended up just doing a scripting interface for them, there was no way to interface with their PBX ... and I'm glad it didn't come to that |
| 15:02.06 | tzafrir_laptop | what SIP provider? |
| 15:02.15 | KP7 | Cybertoy: yea man as soon as i can get my damn treo to run that ! |
| 15:02.26 | KP7 | Maliuta: I feel ya |
| 15:02.44 | Maliuta | tzafrir_laptop: who you talkin' to tzafrir_laptop? </different_strokes> |
| 15:03.10 | Cybertoy | KP7: hmm... what a great idea for all those open source phones... put asterisk on them.. :) |
| 15:03.31 | KP7 | Maliuta: asterisk is a telemarker's dream, i showed someone that switchvox demo and this dude was drooling 'cause he knew his client would be drooling |
| 15:04.04 | Maliuta | KP7: I can think of heaps of uses better than telemarketing |
| 15:04.18 | KP7 | Cybertoy: dude don't play- you know, i'm contemplating learning java to just think of the android/limo possibilities |
| 15:04.38 | KP7 | Maliuta: oh i know- I'm just saying THOSE folks are thinking big too |
| 15:05.32 | KP7 | I'm putting together several high level tech briefs myself for client- see if I can slowly migrate traditional PBX users over |
| 15:05.36 | tzafrir_laptop | KP7, will any of those actually allow you to use a decent voip stack? |
| 15:05.59 | KP7 | tzafrir_laptop: android or limo? |
| 15:06.06 | tzafrir_laptop | either |
| 15:06.12 | Maliuta | KP7: when I get my health back I have some pitch's to put together |
| 15:06.33 | KP7 | i haven't pulled down the developer stuff but i would assume so |
| 15:06.52 | KP7 | i would assume limo has a leg up |
| 15:07.02 | KP7 | since i think its a pure form on linux |
| 15:07.23 | KP7 | where as android from what i understand is more of a custom build for google's need |
| 15:07.30 | KP7 | i'm make supposition here tho |
| 15:08.51 | KP7 | you gotta figure with the FCC (here in the US) allowing access to white space frequencies, its only a matter of time before wireless services for voice, video and other media blaze onto the scene |
| 15:09.24 | KP7 | even on regular 3g networks- why would i make a cell call if i could link to my pbx with a 3g voip client |
| 15:09.53 | KP7 | (i do that now but its not useable indoors) |
| 15:10.36 | KP7 | Maliuta: get well soon man- you know world domination takes time! |
| 15:10.51 | KP7 | rather "open" domination :) |
| 15:11.21 | Maliuta | KP7: tell the public system here to put me at the top of the waiting list for my hip and 85% of my problem is gone ... I can work once that's done |
| 15:11.33 | Maliuta | the GVH is managable |
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| 15:12.17 | KP7 | Maliuta: where are you? |
| 15:14.00 | Maliuta | KP7: qld.au |
| 15:14.37 | KP7 | qld... that queensland? |
| 15:16.10 | Maliuta | up |
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| 15:16.17 | Maliuta | s/^/Y/ |
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| 15:18.35 | KP7 | hehe |
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| 15:28.03 | KP7 | drops a pin... and listens to it through his Snom m3 <--- i frigg'n lub that, i haven't played with a phone so much since i was a kid looking like that etrade commerical |
| 15:31.00 | KP7 | hey does anyone know if you have to use hints to get the MWI notifies to work? |
| 15:31.23 | KP7 | I thought I could just turn on subscriptions |
| 15:32.09 | _ShrikE | KP7: nope, hints are for presence, not mwi. |
| 15:32.19 | KP7 | ok |
| 15:34.02 | KP7 | do you know how i could test mwi functionality? I reboots my phone and on a sip trace it looked like i was not getting an accurate message count on the message-summary and i definitely have messages |
| 15:43.11 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 15:43.15 | tzafrir_laptop | KP7, try a different phone? |
| 15:43.30 | tzafrir_laptop | e.g. some soft phone |
| 15:44.18 | KP7 | the problem is on the * side- i have 3 message total and 1 new and the message-summary says 0/0 on the voicemail line |
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| 15:51.43 | sulan | is it possible in asterisk 1.4 to have all contexts switched using RealTime? |
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| 16:22.56 | ManxPower | Soundslike you don't have a context in your mailbox= line. |
| 16:23.41 | ManxPower | mailbox=mailboxnumber@voicemailcontext (not extensions.conf context) |
| 16:24.05 | x86 | voicemail.conf context |
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| 16:37.08 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 16:42.06 | hi365 | I have a server that is both nat-ed to the local network (i.e. its on 192.168.0.0, and there is nat between 192.168.1.0 and 192.168.2.0) and its open to the outside world. how do i setup the externip/localnet settings? |
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| 16:52.38 | *** join/#asterisk BrokenNoze (n=BrokenNo@212.183.134.129) |
| 16:53.09 | BrokenNoze | hi, is anyone aware of an issue with music on hold suddenly cutting in on a call for no apparent reason? |
| 16:54.10 | *** join/#asterisk Sonarcade (n=chatzill@76.91.206.48) |
| 16:54.44 | Sonarcade | hi. does anyone know where I'd be able to get information on where I can buy a 3 rj11 to 1 rj45 adapter? |
| 16:56.08 | tzafrir_laptop | The RJ45 is for BRI or E1? |
| 16:57.16 | Sonarcade | tzafrir_laptop: sorry. I'm quite new to this. My pbx box has one rj45 input and I'm trying to circumvent having to crimp telephone cables into one rj45 plug |
| 16:57.36 | ManxPower | hi365: Set localnet to whatever your localnet is, set externip to your actuall routable exterenal IP |
| 16:57.58 | tzafrir_laptop | Sonarcade, in most cases RJ45 would be for ethernet |
| 16:58.07 | Sonarcade | oh |
| 16:58.10 | hi365 | ManxPower: thanks, I have three localnet's should i just list them? or do i only need the local net asterisk is on? |
| 16:58.23 | Sonarcade | then I mean just a modular plug that uses 6 pins |
| 16:58.25 | tzafrir_laptop | but rj11: would that be for phones, or to connect to a PSTN line? |
| 16:58.42 | ManxPower | hi365: Localnet lets Asterisk determine if it needs to enable the special NAT stuff or not for a packet. |
| 16:58.48 | Sonarcade | like an rj11 |
| 16:59.01 | ManxPower | so, if you have 3 networks that don't have NAT between them, then try listing 3 localnets. I am not sure. |
| 16:59.23 | hi365 | ManxPower: I do have nat in between them... (err, does that mean there not local?) |
| 16:59.27 | ManxPower | Sonarcade: You can't just go around plugging things in. What specific port type is the RJ-45 on the PBX |
| 16:59.51 | ManxPower | hi365: The DEFINITION of localnet in Asterisk is "no NAT between me and the specified network" |
| 17:00.02 | hi365 | gotchya! |
| 17:00.12 | hi365 | ManxPower: ^ thanks |
| 17:01.06 | ManxPower | Sonarcade: For example if you plug a phone line into an FXS port on your PBX (or Asterisk) you will BLOW THE PORT and have to have it replaced |
| 17:01.30 | Sonarcade | ManxPower: http://www.ablecomm.info/d308.htm . I'm not too familiar with the code #s for plugs. I do know that it takes a phone sized plug that has 6 pins |
| 17:01.37 | ManxPower | If you plug a T-1 into an ethernet port then you are likely to blow the port as well. |
| 17:01.50 | ManxPower | Sonarcade: you CANNOT know what the port is based on the plug/jack. |
| 17:02.18 | Sonarcade | how do I find out? |
| 17:02.21 | ManxPower | Sonarcade: RJ-11 has 6 pins. RJ-45 has 8 pins. |
| 17:02.25 | [TK]D-Fender | Sonarcade: Read the MANUAL |
| 17:02.32 | ManxPower | Sonarcade: I dunno. Go ask the person that installed your PBX? |
| 17:02.37 | Sonarcade | i don't have the manual handy with me |
| 17:02.41 | Sonarcade | then it's an rj-11 |
| 17:02.44 | [TK]D-Fender | Sonarcade: How about the model? |
| 17:02.54 | Sonarcade | it's a td308 |
| 17:03.39 | [TK]D-Fender | I'd be betting its a T1/E1 port |
| 17:03.45 | ManxPower | Sonarcade: It is starting to sound like you are trying to guarantee failure. It's like trying to install a Ford engine into a mazda car. You are guaranteed to fail. |
| 17:03.53 | [TK]D-Fender | Sonarcade: At which point it has nothing to do with FXO/FXS at all |
| 17:04.07 | Sonarcade | what's fxo/fxs |
| 17:04.13 | [TK]D-Fender | Sonarcade: ANLOG lines |
| 17:04.17 | [TK]D-Fender | ANALOG |
| 17:04.18 | ManxPower | Sonarcade: You should go read the Asterisk Book. |
| 17:04.39 | [TK]D-Fender | Sonarcade: Go call someone to look at what you've got since you can't seem to identify it yourself |
| 17:04.45 | ManxPower | Sonarcade: At this point you don't even know enough to know how to ask the right questions. You need to go read the book., |
| 17:04.58 | Sonarcade | instead of that, can you help me with directing me to a place where my methods won't be framed in the context of setting up an asterisk box or whatever? |
| 17:05.16 | Sonarcade | b/c I was told to come here for general pbx related questions |
| 17:05.22 | [TK]D-Fender | Sonarcade: So far your methods could fry your equipment. |
| 17:05.25 | ManxPower | Sonarcade: Then you were told wrong./ |
| 17:05.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:05.48 | ManxPower | Actually we can usually give you general PBX info. What you are looking for is SPECIFIC PBX info. |
| 17:05.52 | Sonarcade | [TK]D-Fender: how so? I've just said that I'm looking to merge 3 rj11 cables into one rj11 plug |
| 17:05.57 | [TK]D-Fender | Sonarcade: You are looking to crimp connectors for devices you don't know the electrical charateristics of. |
| 17:06.11 | feeds | hi |
| 17:06.17 | ManxPower | [TK]D-Fender: just let him do it. It will serve him right if he blows up his PB X |
| 17:06.20 | Sonarcade | after having mistakenly said that I wanted an rj45 plug involved in there somehow |
| 17:06.34 | [TK]D-Fender | Sonarcade: for proper RJ11 thats quite Googleable under RJ11 spec |
| 17:06.35 | Sonarcade | how would I overload the input? |
| 17:06.41 | ManxPower | Sonarcade: you will have to wire the cable yourself. |
| 17:06.54 | [TK]D-Fender | Sonarcade: But you've brought a piece of equipment into this picture that does seem to mnatch your goal |
| 17:06.54 | Sonarcade | so there's no "merging" adapter out there |
| 17:06.59 | feeds | Is there some command in * to reload the configs wihtout restarting the whole server?? |
| 17:07.00 | ManxPower | Sonarcade: A ringing phone line places 90 volts on the line. |
| 17:07.12 | ManxPower | feeds: The command is "reload" |
| 17:07.19 | feeds | ManxPower, thanks |
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| 17:07.27 | ManxPower | feeds: you need to start reading some docs |
| 17:08.19 | Sonarcade | I spoke to a dealer since the manufacturer directed me to one and the guy was mum on it unless I'd hire him to look at it. the thing is, it's already installed and up and running with a plug that has all 6 pins filled up |
| 17:08.30 | Sonarcade | presumably an rj11 |
| 17:09.01 | KP7 | <ManxPower>Soundslike you don't have a context in your mailbox= line. <--- spot on man, soon as i fixed that and reloaded sip i got the notification |
| 17:09.33 | [TK]D-Fender | Sonarcade: There are... RadioShack carries that stuff. |
| 17:11.41 | [TK]D-Fender | Sonarcade: Well who knows what that device's wiring spec is? Who says it its proprietary to it? |
| 17:11.50 | rob0 | In point of fact, I believe that Ford and Mazda have collaborated, and many of them might have interchangeable parts. |
| 17:12.02 | [TK]D-Fender | Sonarcade: Guessing puts voltage where there shouldn't be. |
| 17:12.28 | [TK]D-Fender | rob0: Or your Frankenvehicle could blow up and kill you. |
| 17:13.07 | rob0 | IIRC a Mazda Protege is a Ford Escort. |
| 17:13.30 | rob0 | but it was years ago when I was told that |
| 17:14.30 | rob0 | I'm jes funnin' wicha |
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| 17:16.54 | *** join/#asterisk Tuxguy (i=Jimi@cpe-024-025-040-206.ec.res.rr.com) |
| 17:17.10 | Tuxguy | Anyone know a VoIP trunk service in the USA that can offer multiple phone #s, etc? |
| 17:17.29 | jjshoe | Tuxguy any |
| 17:17.52 | Tuxguy | Like who? I dont even know of any companies yet, except residential ones. And that is all I keep finding on google. |
| 17:18.02 | Sonarcade | vonage? |
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| 17:18.21 | jjshoe | Tuxguy teliax, voicepulse, cbeyond, etc. etc. etc. etc. |
| 17:18.26 | [TK]D-Fender | ~itsplist-us |
| 17:18.27 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:18.37 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 17:18.49 | Tuxguy | Ah, bandwidth.com based out of raeligh, nc |
| 17:18.50 | Tuxguy | cool |
| 17:19.08 | [TK]D-Fender | Vonage offers reisdential grade service. Avoid at all costs |
| 17:19.22 | Tuxguy | ok |
| 17:19.36 | Dougy | vonage blows |
| 17:20.30 | Bad_Robot- | a client had 3 vonage boxes behind nat and it was hit or miss if they would ring |
| 17:20.35 | Tuxguy | I have used them for home service before. I thought they only did residential where you get a voice modem, and connect to their service. I didnt know you could use it through your pbx etc |
| 17:20.40 | RB2 | I've been fairly happy with ipcomms so far. Has anyone else had good or bad experiences with them? |
| 17:21.17 | Tuxguy | I have been testing and configuring asterisk with SipDiscount using their free call system, just to get everything configured. |
| 17:22.26 | *** join/#asterisk ManxPower (n=manxpowe@24.196.31.252) |
| 17:22.51 | *** join/#asterisk datacompboy (n=datacomp@213.187.250.81) |
| 17:23.34 | drmessano | It's not a voice modem, it's an ATA |
| 17:23.40 | drmessano | We don't use the word "modem" here |
| 17:23.46 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 17:24.00 | Bad_Robot- | drmessano why not? |
| 17:24.13 | Tuxguy | I am sorry. I am just getting into the voip and telefony setup |
| 17:24.15 | Bad_Robot- | is learning |
| 17:24.28 | maxxim | hi, i'm getting 'SIP/2.0 401 Unauthorized' error for incomming connection. could you help me to fix it? http://rafb.net/p/2BgnOH98.html |
| 17:24.39 | drmessano | Because modem usually comes up with "WHY CANT MY US ROBOTICS FAX MODEM WORK IN AKERISK".. and it doesnt |
| 17:24.48 | drmessano | So we avoid that term period.. |
| 17:24.53 | Bad_Robot- | ok |
| 17:25.18 | drmessano | Vonage provides an Analog Telephony Adapter (ATA) which is completely unlike a modem |
| 17:25.24 | drmessano | Modem's are evil |
| 17:25.36 | drmessano | Errr didn't need the apostrophe |
| 17:25.43 | drmessano | Modems are evil |
| 17:25.49 | drmessano | wakes up a little |
| 17:26.09 | Bad_Robot- | can a modem be used for remote access to an asterisk box? |
| 17:26.19 | Tuxguy | So I would need an ATA to make a regular wireless phone work with voip right? |
| 17:26.20 | drmessano | That's kinda scary |
| 17:26.25 | *** join/#asterisk miguel3239 (n=elguero@ns1.nashuacs.com) |
| 17:26.29 | drmessano | Tuxguy: yes |
| 17:26.43 | Tuxguy | Ok, I was wondering what the adapter was called. Wasnt sure where to look or even how to google it. |
| 17:26.56 | drmessano | Bad_Robot-: Assuming you have a real internet connection to Asterisk, why would you want a modem for terminal access? |
| 17:27.00 | jjshoe | vgetty ftw :P |
| 17:27.17 | Bad_Robot- | i was thinking sshd died or soemthing |
| 17:27.28 | rob0 | "Whut happened to de grass by the hyway?" "Looks like dey modem." |
| 17:27.41 | drmessano | Tuxguy: ATA + Consumer cordless makes a good cordless solution, as most Wifi SIP Phones suck |
| 17:28.19 | [TK]D-Fender | ~ata |
| 17:28.20 | jbot | ata is, like, Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 17:28.27 | Tuxguy | Are they expensive? I am setting up a 1 telephone number voip service for my family. I am going to set up an auto attendant, ie to talk to my wife, press 0, me, 1.. my son 2.. etc although my son is 3.. lol |
| 17:28.44 | drmessano | Naw |
| 17:28.53 | drmessano | ~$50 or so for a 2 line ATA |
| 17:28.55 | maxxim | hi, i'm getting 'SIP/2.0 401 Unauthorized' error for incomming connection. could you help me to fix it? http://rafb.net/p/2BgnOH98.html |
| 17:28.57 | ManxPower | Bad_Robot-: Asterisk does not support remote access via modem. Your OS, however, probably does support remote access via modem. Your question is an OS question, not an Asterisk question. |
| 17:29.09 | ManxPower | maxxim: that means you have a user or password problem |
| 17:29.19 | drmessano | Damn.. ManxPower provided +4 on the pwnage |
| 17:29.32 | maxxim | ManxPower> this is an incomming connection to my * |
| 17:29.47 | maxxim | ManxPower> who is rejecting, my * or remote end? |
| 17:29.47 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 17:29.58 | Bad_Robot- | ManxPower ahh that's right that would be OS related but would that be a bad idea to have as a backup in the box or could it possible interfere with asterisk |
| 17:30.21 | drmessano | Bad_Robot-: Gonna justify the cost of a phone line to have it sit there waiting? |
| 17:30.32 | ManxPower | maxxim: your asterisk is rejecting if if it's an incoming connection to the Asterisk server. |
| 17:30.49 | ManxPower | Bad_Robot-: I have no opinion on that. |
| 17:30.57 | drmessano | Bad_Robot-: You could always create an asterisk extension using System() to restart SSHd |
| 17:31.04 | maxxim | can you please see the output, and tell me how to allow this incomming connection: http://rafb.net/p/2BgnOH98.html |
| 17:31.06 | Bad_Robot- | good point it's probably a waste of 20 bucks a month but i was curious if it's done on asterisk systems or just a bad idea |
| 17:31.21 | drmessano | Bad_Robot-: You could always create an asterisk extension using System() to restart SSHd |
| 17:31.26 | drmessano | System() is your friend |
| 17:31.31 | ManxPower | maxxim: what SIP device are you using? |
| 17:31.32 | Bad_Robot- | drmessano wow that's an insane idea.. wow i LOVE IT |
| 17:31.56 | maxxim | ManxPower, this is from remote provider, i don't know |
| 17:31.59 | Bad_Robot- | that is a much better solution :) |
| 17:32.19 | ManxPower | I strongly doubt the provider is connecting as user 101 |
| 17:32.52 | Bad_Robot- | thanks for the input |
| 17:32.57 | maxxim | ManxPower> i've setup 101 peer for outoing connecton via that provider |
| 17:33.06 | maxxim | ManxPower> now i have problem with incomming connections |
| 17:33.17 | ManxPower | maxxim: What did you set up for incoming connections from that provider? |
| 17:33.18 | drmessano | Bad_Robot-: Create a handful of extensions using System() to restarts SSHd, reload asterisk, etc.. use VMAuthenticate() in front to give some basic auth if someone should stumble across your obscure extensions numbers (HINT: MAKE THEM OBSCURE) |
| 17:34.01 | maxxim | ManxPower> i don't know how to setup properly incomming connection for it. coul you tel me the hint? |
| 17:34.16 | Bad_Robot- | can you give me an example of what you'd make one look like? meaning the obscure part |
| 17:34.27 | ManxPower | maxxim: no. How you set up incoming depends on your provider. |
| 17:34.27 | maxxim | ManxPower> i want * to asnwer to the incomming call and to play something |
| 17:34.53 | ManxPower | maxxim: How far thru The Book have you gotten? |
| 17:35.01 | drmessano | Like *999101, *999102, or anything just WAY off your internal dialplan |
| 17:35.15 | drmessano | If you have 101, 102, 103, dont make it 109 and 110 (duh) |
| 17:35.24 | drmessano | Use your imagination |
| 17:35.29 | Bad_Robot- | ok i see what you mean so no accidents hitting it |
| 17:35.40 | maxxim | ManxPower> i see what do you mean. i have to read it futher... |
| 17:35.53 | drmessano | Unused NANPA assignments work too |
| 17:35.55 | drmessano | lol |
| 17:39.36 | Tuxguy | drmessano, you in the USA? |
| 17:39.41 | Sonarcade | are there any measures I can take to prevent overloading of the pbx input? |
| 17:39.49 | drmessano | yes |
| 17:39.53 | Sonarcade | like buying a multimeter |
| 17:39.55 | Sonarcade | or something |
| 17:40.03 | jjshoe | lololol |
| 17:40.13 | drmessano | overloading of the PBX input? |
| 17:40.15 | Tuxguy | drmessano, do they sell ATAs at bestbuy/circuit city? |
| 17:40.24 | drmessano | Tuxguy: No |
| 17:40.43 | drmessano | http://www.telephonydepot.com is a good source |
| 17:40.58 | Tuxguy | I wonder if I can use my old vonage ata then? if i add an entry in /etc/hosts for whatever IP address it uses? |
| 17:41.01 | ManxPower | Sonarcade: Yes, like knowing what the PORT IS. |
| 17:41.12 | Sonarcade | drmessano: yeah. that's definitely something I hadn't considered coming in here before ManxPower and [TK]D-Fender set me straight |
| 17:41.42 | ManxPower | Sonarcade: You can ask the same thing 500 zillion different ways. It doesn't matter if you like the answer or not, the answer is correct. |
| 17:41.44 | Sonarcade | ManxPower: barring that, unless there's a way to know from just looking at the unit itself |
| 17:41.58 | [TK]D-Fender | Sonarcade: Want to know how not to overload your PBX input? Go find out what you HAVE. |
| 17:41.59 | ManxPower | Sonarcade: if there was a way, we would have told you. |
| 17:42.18 | *** join/#asterisk ddfire (n=sdf@94-240-16-190.fibertel.com.ar) |
| 17:42.29 | [TK]D-Fender | Sonarcade: Barring that, do you possess any substantial psychic capabilities? Because everything else is a guess. |
| 17:42.49 | ddfire | hi |
| 17:42.50 | Tuxguy | drmessano, How do you configure them once you plug them into the lan? Do you have like a webserver running on them that you go to and edit the settings? |
| 17:42.55 | drmessano | Tuxguy: What kind is it? |
| 17:43.04 | ddfire | i am looking some info on how to identify a transfer from the manager? |
| 17:43.36 | ManxPower | Tuxguy: I know it's hard to believe but these devices come with installation instructions. Read them. |
| 17:43.39 | [TK]D-Fender | Tuxguy: Those ATA's are locked and typically worthless. |
| 17:43.48 | Tuxguy | drmessano, Just in general, wasn't sure how they worked. I assume its an interface like a router's config page |
| 17:44.02 | [TK]D-Fender | Tuxguy: With a fair amount of work and a lucky model you might be able to unlock it. |
| 17:44.07 | Tuxguy | ManxPower, I dont own a device yet. |
| 17:44.09 | Tuxguy | Thanks TK |
| 17:44.10 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 17:44.32 | drmessano | Tuxguy: Some can be unlocked, and can be a quick start if you need one |
| 17:44.54 | Tuxguy | So they are similar to cell phones, where they are locked to one provider? |
| 17:44.56 | ManxPower | Tuxguy: real routers don't have a web interface. Usually you configure the ATA using DHCP or TFTP or FTP, or HTTP or HTTPS, or a built in web server. |
| 17:45.10 | drmessano | Tuxguy: Yes |
| 17:45.17 | Sonarcade | oh awesome |
| 17:45.20 | Sonarcade | I found the manual online |
| 17:45.21 | drmessano | Tuxguy: Is it a Linksys? |
| 17:45.26 | ManxPower | Tuxguy: they are only locked if you buy them with some service provider. Get a non-locked device and you don't have to worry about it. |
| 17:45.54 | Tuxguy | drmessano, I am just looking at that website you pointed to earlier, telephony depot. But, leaning towards the linksys device. |
| 17:46.05 | ManxPower | As an example, the Linksys PAP2 is normally locked. The PAP2-NA is not normally locked. |
| 17:46.51 | Tuxguy | Ah ok, thank you. |
| 17:46.57 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 17:47.15 | drmessano | Tuxguy: Personally, my first SIP device, even before using a softphone, was a PAP2 that I unlocked. For me, I wasn't gonna buy crap 3 years ago for some "PBX server thing" I had little use for, and getting it unlocked ==> asterisk working got me hooked |
| 17:47.29 | drmessano | But it may or may not be worth it to you to buy a new one unlocked and not worry about it |
| 17:48.46 | Tuxguy | I would rather just buy one that is unlocked and not have to worry with it. I am just doing this for home use. I might look for a multi-line version, like 2-3 lines, since that is how many devices there will be. |
| 17:49.27 | drmessano | So get one unlocked, get it working, and make the old one a weekend project later :) Ask me later at some point, and we can work on unlocking, if the device is capable |
| 17:49.39 | rob0 | A year or so ago, I got a couple of used Sipuras for ~US$35 on ebay. |
| 17:49.40 | drmessano | But yeah, that will get you up and going |
| 17:50.00 | Bad_Robot- | i ended up buying a sipura 3000 and it's an ata with too many settings for me |
| 17:50.01 | rob0 | (each, not both :) ) |
| 17:50.18 | drmessano | Too many settings? |
| 17:51.01 | [TK]D-Fender | drmessano: AKA the 90% of things you'll never need to touch and works anyway |
| 17:51.20 | Bad_Robot- | well i say that because i used telasip and they told me it was a bad choice for their service and it took a few diff configs to actually get it working. too many meaning I nor the sip provider knew how to config it |
| 17:51.31 | ddfire | please some help here... i need to know when an agent transfer a call |
| 17:51.33 | drmessano | ROFL |
| 17:51.48 | ManxPower | ~ask |
| 17:51.49 | jbot | i guess ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:51.52 | drmessano | Bad_Robot-: I call bullshit, at least from their end |
| 17:52.00 | drmessano | ATA setup for MOST providers: |
| 17:52.02 | Bad_Robot- | telasip techsupport == FAIL |
| 17:52.07 | drmessano | Facotry reset (if not new) |
| 17:52.12 | drmessano | Add SIP user |
| 17:52.17 | drmessano | Add SIP Proxy |
| 17:52.20 | drmessano | Add SIP Password |
| 17:52.22 | drmessano | Apply |
| 17:52.30 | drmessano | O.o |
| 17:52.46 | Bad_Robot- | well that worked but lots of one way audio until i setup dyndns and used that in teh setup |
| 17:52.53 | drmessano | .... |
| 17:53.05 | [TK]D-Fender | Bad_Robot-: Your provider was full of shit |
| 17:53.11 | drmessano | That has nothing to do with an ATA to a provider |
| 17:53.30 | drmessano | No dyndns, no one way audio with a NAT'ed ATA to a provider |
| 17:53.30 | ManxPower | Bad_Robot-: One way audio is usually caused by the person setting up the device not understanding SIP, RTP, or NAT. |
| 17:53.34 | Bad_Robot- | if that's the case i'd recommend staying away from telasip |
| 17:53.38 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 17:53.59 | drmessano | User, password, proxy is all you need |
| 17:54.10 | Bad_Robot- | well ata was in the dmz and still needed outside ip for audio that worked consistently |
| 17:54.13 | ddfire | how i know when someone trasnfer a call, using #<exten> from the manager? |
| 17:54.19 | drmessano | DOesnt need DMZ either |
| 17:54.23 | drmessano | Cant be fully NAT'ed |
| 17:54.26 | drmessano | Can* |
| 17:54.27 | ManxPower | Bad_Robot-: DMZ means NOTHING to Asterisk |
| 17:54.43 | ManxPower | ddfire: No, nobody knows this. |
| 17:54.49 | Bad_Robot- | shouldn't the dmz forward all ports to that ip |
| 17:54.51 | drmessano | Bad_Robot-: Was this ATA connected to the provider |
| 17:54.56 | drmessano | or TO asterisk |
| 17:54.56 | Bad_Robot- | yes |
| 17:54.56 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 17:55.01 | Bad_Robot- | straight to them |
| 17:55.07 | ManxPower | Bad_Robot-: yes, but you don't need to forward ports to the ATA. |
| 17:55.11 | drmessano | Then you did something wrong |
| 17:55.26 | Bad_Robot- | probably me :) |
| 17:55.26 | ManxPower | Your question indicates that you don't understand NAT or SIP |
| 17:55.28 | drmessano | That ATA needs 3 params and can be fully NAT'ed |
| 17:55.46 | Bad_Robot- | i understand sip is on 5060 and audio on 10000-20000 |
| 17:56.02 | ManxPower | drmessano: I had my SIPura box roaming between NAT and no-NAT seamlessly |
| 17:56.09 | ManxPower | Bad_Robot-: Correct. |
| 17:56.23 | ManxPower | well, audio is on whatever ports the two devices can agree on. |
| 17:57.04 | ManxPower | Bad_Robot-: But SIP does not look at the source IP/port and destination IP/port. SIP looks into the DATA part of the packet and that part of the packet does not have NAT stufff on it. |
| 17:57.07 | drmessano | ManxPower: Easily done.. tweaking is the #1 NAT problem.. not NAT and not the NAT params in Asterisk.. it's the tweaks made outside those that break the REAL params from working |
| 17:58.28 | ManxPower | Bad_Robot-: HTTP uses port 80, HTTPS uses port 443. You don't need to put your web browsing machine in the DMZ do you? |
| 17:58.39 | ManxPower | drmessano: I was using SRV records to do the roaming |
| 17:59.26 | drmessano | ManxPower: I've thought of that.. right now I use the external address for the devices and let iptables handle the inside stuff.. But it's FAIL if the inet goes down |
| 18:00.48 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:00.52 | ManxPower | drmessano: not really an issue as asterisk usually fails of the internet goes down. 8-| |
| 18:02.33 | Tuxguy | Can someone point me at documentation for setting up an auto attendant |
| 18:02.55 | [TK]D-Fender | ~book |
| 18:02.56 | jbot | book is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 18:02.57 | [TK]D-Fender | ^^^ |
| 18:03.06 | [TK]D-Fender | Tuxguy: IVR is the work you're looking for |
| 18:03.32 | ddfire | Tuxguy http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:03.39 | Tuxguy | Oh ok, so that is "press 2, press 3" ? |
| 18:03.45 | [TK]D-Fender | Tuxguy: Yes |
| 18:03.49 | ddfire | Tuxguy read that pdf |
| 18:03.54 | Tuxguy | ty |
| 18:04.30 | ddfire | Tuxguy there you will find all the answers and it have a lot of examples, you can use it for 1.4 and 1.6 |
| 18:04.37 | Tuxguy | woot |
| 18:04.56 | drmessano | HAW |
| 18:05.07 | drmessano | The digium site now has a redirect for the book |
| 18:05.10 | drmessano | YAY for hit counting |
| 18:05.19 | ManxPower | You should still read the UPGRADE*.txt files. |
| 18:07.25 | *** join/#asterisk CapriCorN^80 (i=50d8dcb6@gateway/web/ajax/mibbit.com/x-fa1abe15091308c7) |
| 18:07.31 | CapriCorN^80 | hi |
| 18:07.46 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:08.17 | CapriCorN^80 | i am looking for SIP working with service instant messing and presence . |
| 18:08.26 | CapriCorN^80 | please refer me some doc or link |
| 18:08.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:09.23 | [TK]D-Fender | CapriCorN^80: *'s presene is the "hint" priority in the dialplan. * does not support SIP IM |
| 18:09.42 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:10.39 | CapriCorN^80 | [TK]D-Fender: can you explain it more . secondly if you can refer me some doc with good example of it |
| 18:10.40 | CapriCorN^80 | thx |
| 18:11.05 | [TK]D-Fender | CapriCorN^80: Go lookup "presence" on the WIKI |
| 18:11.07 | [TK]D-Fender | ~wikis |
| 18:11.07 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:11.16 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 18:11.28 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:11.40 | *** part/#asterisk baliktad (i=baliktad@c-24-17-254-250.hsd1.wa.comcast.net) |
| 18:11.47 | ManxPower | CapriCorN^80: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 18:12.13 | drmessano | I need to add a wiki page: SIMPLE messaging: Why it's someone else's fault |
| 18:12.43 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 18:13.32 | CapriCorN^80 | ManxPower: this doc is more with asterisk |
| 18:13.38 | CapriCorN^80 | i need some general concept |
| 18:13.41 | drmessano | .... |
| 18:14.03 | [TK]D-Fender | CapriCorN^80: http://www.ietf.org/rfc/rfc3261.txt |
| 18:14.04 | drmessano | http://www.ietf.org/rfc/rfc3261.txt |
| 18:14.07 | drmessano | Damn you |
| 18:14.10 | [TK]D-Fender | :p |
| 18:14.15 | drmessano | My broken get stuck when I copied |
| 18:14.21 | drmessano | GRRRR |
| 18:14.26 | drmessano | Broken = browser |
| 18:14.30 | drmessano | Yeah |
| 18:14.37 | drmessano | Firefox screwed me on the paste |
| 18:14.53 | [TK]D-Fender | drmessano: I was slowed down because I already primed his nic then autocompleted it again... I should have been about 1-2s faster :) |
| 18:15.12 | [TK]D-Fender | drmessano: And I'm using FF as well :) |
| 18:15.26 | drmessano | I'm using Windows <-- 5 sec handicap |
| 18:16.43 | [TK]D-Fender | drmessano: Same here.... |
| 18:16.43 | ManxPower | CapriCorN^80: This is an Asterisk channel. We generally only deal with Asterisk related issues. |
| 18:16.43 | drmessano | I'm on Comcast <--- My connection was probably throttled |
| 18:16.43 | ManxPower | drmessano: well turn off all the eye candy |
| 18:16.43 | [TK]D-Fender | drmessano: Equally armed, it is a battle of wits between us! |
| 18:16.43 | drmessano | ManxPower: Doesnt matter.. FF3 sucks |
| 18:16.44 | [TK]D-Fender | drmessano: But alas I shant fight an unarmed opponent! |
| 18:16.45 | ManxPower | drmessano: I was referring to your OS eye candy. |
| 18:16.51 | drmessano | I have it all off |
| 18:16.58 | drmessano | I dont do eye candy |
| 18:17.15 | CapriCorN^80 | Thanks. but i have searched this RFC but didnt get any example or working of SIP with presence and instant messaging |
| 18:17.17 | ManxPower | My XP interface looks exactly like the Win2k interface with all it's eye candy turned off, which looks just like Win98 |
| 18:17.20 | drmessano | OS isn't a problem anyway.. machine is plenty fast.. I was just trying one-up fender |
| 18:17.29 | CapriCorN^80 | Manxpower: yea i can understand |
| 18:17.38 | ManxPower | CapriCorN^80: Well that is all the info there is. |
| 18:17.38 | CapriCorN^80 | sorry for asking other information |
| 18:17.43 | [TK]D-Fender | CapriCorN^80: then go look at SER, etc |
| 18:17.54 | CapriCorN^80 | but i am not getting it thats why i refer to you people |
| 18:18.27 | [TK]D-Fender | CapriCorN^80: Sorry, you'd be looking for #neurosurgery |
| 18:18.45 | drmessano | Sorry, we use Asterisk, we know nothing about SIP as per the RFC, just what Digium tells us it's supposed to work like |
| 18:18.48 | drmessano | <-- Sheeple |
| 18:18.51 | [TK]D-Fender | CapriCorN^80: We can't make you "get it", and we're not about to read up on all of this just to try to explain it to you |
| 18:19.29 | drmessano | I didnt know SIP used TCP until I downloaded 1.6 |
| 18:19.32 | drmessano | Baaaah |
| 18:19.37 | ManxPower | CapriCorN^80: What you want to do is poorly supported and poorly documented. Asterisk doesn't even support one of the things you want to do. |
| 18:19.55 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 18:20.00 | CapriCorN^80 | Manxpower: ok |
| 18:20.14 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:20.31 | ManxPower | idly ponders trying to use SIP over EVDO |
| 18:20.53 | drmessano | I thought it was cool how SIP TCP, T.38 faxing, and G722 came out the same day as Asterisk 1.6 |
| 18:20.55 | drmessano | Baaaah |
| 18:21.27 | CapriCorN^80 | Manxpower: any # on freenode for this type of chat ? |
| 18:21.37 | [TK]D-Fender | CapriCorN^80: #ser |
| 18:22.20 | ManxPower | drmessano: Yeah. We can now die happy. |
| 18:22.27 | drmessano | Maybe a nice book on SIP from the pirate bay |
| 18:22.37 | CapriCorN^80 | thx |
| 18:22.40 | drmessano | (thats where I got my asterisk source code... ssshhh_ |
| 18:22.43 | drmessano | (thats where I got my asterisk source code... ssshhh) |
| 18:22.52 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 18:23.38 | Bad_Robot- | sry cable took a dump after you said i didn't understand sip/nat so i didn't see the msg |
| 18:24.24 | drmessano | Bad_Robot-: Probably a NAT issue |
| 18:24.45 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:24.46 | Bad_Robot- | ok |
| 18:25.00 | Bad_Robot- | i had a cheesy airlink at the time |
| 18:30.34 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:35.47 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 18:37.42 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:44.09 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:46.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:48.52 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:50.16 | *** join/#asterisk timburke (i=timburke@unaffiliated/timburke) |
| 18:50.53 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:52.46 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:54.50 | *** join/#asterisk lxroot (n=ALAMI@ce3.net.hrz.tu-darmstadt.de) |
| 18:54.52 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 18:54.55 | *** part/#asterisk lxroot (n=ALAMI@ce3.net.hrz.tu-darmstadt.de) |
| 19:04.21 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 19:22.45 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 19:28.13 | *** join/#asterisk hi365_m (n=hi365@213.151.47.18) |
| 19:28.21 | ManxPower | . |
| 19:29.58 | rob0 | I need to mess with tc(8) on Linux to give priority to SIP and phone calls. |
| 19:32.22 | rob0 | damn phone just rang, and the caller couldn't hear me because of /dev/wife's downloads :( |
| 19:35.06 | ManxPower | rob0: you understand that QoS only works on TRANSMITTED data, right? |
| 19:35.31 | rob0 | I think that's the problem, actually. I can hear callers just fine. |
| 19:35.36 | ManxPower | and if they could not hear you, I imagine that would be an UPLOAD issue. |
| 19:35.37 | rob0 | ADSL |
| 19:35.47 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 19:35.51 | rob0 | um |
| 19:35.56 | rob0 | right |
| 19:36.19 | ManxPower | rob0: Welcome to the world of VoiceOverIPOverInternet |
| 19:36.47 | ManxPower | My recommendation is to get rid of the wife actually. |
| 19:36.52 | rob0 | haha |
| 19:36.53 | [TK]D-Fender | rob0: http://www.voip-info.org/wiki/view/QoS+with+Linux+using+PRIO+and+HTB |
| 19:42.14 | Bad_Robot- | agrees to getting rid of the wife ;) |
| 19:42.47 | rob0 | Are there any bids? |
| 19:43.22 | ManxPower | rob0: It's a buyers market |
| 19:43.26 | SteelSide | Just a Q: does * have some builtin module for AT commands for a data/voice modem? |
| 19:43.32 | ManxPower | "free to good home" might be the best way. |
| 19:43.45 | ManxPower | SteelSide: No, it does not. Asterisk does not support modems. |
| 19:44.01 | SteelSide | that sounds sad |
| 19:44.22 | *** join/#asterisk EI5GTB (n=Paul@78.16.183.196) |
| 19:44.31 | SteelSide | i was pretty sure i read somewhere that it did :/ |
| 19:45.05 | EI5GTB | evening guys, why is it that when i restarted asterisk, and started it again, tty9 dont have asterisk in it? |
| 19:45.07 | ManxPower | Not really. "voice modems" do not have the low latency required for two way voice. They are high latency -- works for IVRs and voicemail. They would work so poorly Digium does not want to spend the money to write the drivers for each and every voicemodem. |
| 19:46.22 | ManxPower | SteelSide: you read wrong. The chan_modem, which never worked right (I think it was written by a couple of drunken college kids during spring break and never looked at again) was removed from Asterisk 1.2, I believe. |
| 19:47.05 | ManxPower | But you are welcome to write kernel drivers and zaptel compat drivers for your modem. But we cannot help with that. |
| 19:47.10 | SteelSide | Arrggh |
| 19:47.54 | EI5GTB | lol, windows forgot i have a soundcard.. must be due a scheduled restart |
| 19:47.56 | EI5GTB | brb |
| 19:48.04 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:49.33 | SteelSide | ManxPower, but it *would* be possible to compile chan_module against 1.4 or the like and have a quick peek at it? |
| 19:50.25 | ManxPower | SteelSide:Best of luck with that. The fact you don't like the answer does not mean the answer is incorrect. |
| 19:55.42 | tzafrir_laptop | chan_modem was not intended to get data modems to work as modems |
| 19:56.20 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:56.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:56.32 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 19:56.48 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:58.05 | *** join/#asterisk timburke (i=timburke@unaffiliated/timburke) |
| 19:58.36 | ManxPower | tzafrir_laptop: I suspect the same people wrote chan_modem as wrote wu_ftpd. 8-( |
| 19:59.47 | *** join/#asterisk loproc (i=d40a3735@gateway/web/ajax/mibbit.com/x-cebcfc853f605df0) |
| 20:01.06 | tzafrir_laptop | ManxPower, http://qa.debian.org/popcon.php?package=wu-ftpd :-( |
| 20:03.15 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:03.33 | *** join/#asterisk diegows (n=diegows@motherbox.xtech.com.ar) |
| 20:03.42 | loproc | Hi again. [TK]D-Fender and ManxPower you helped me yesterday with some NAT issues. Now I've just stumbled over another problem I've had for some time. I've got a handytone 286 hooked up from home, but there's "0"-way audio for the first 40 seconds of a call... (tested when calling the echo application) |
| 20:04.08 | [TK]D-Fender | loproc: By now your pastebin should already be up... |
| 20:04.38 | loproc | [TK]D-Fender: Yup. I'll just enable sip debug :o) |
| 20:06.53 | diegows | anybody knows why an Linksys SPA 3102 puts 127.0.0.1 on SIP messages? |
| 20:07.11 | drmessano | Misconfig |
| 20:07.27 | loproc | [TK]D-Fender: http://pastebin.com/da607f6 |
| 20:07.29 | diegows | drmessano: do you have a spa 3102? |
| 20:08.15 | drmessano | I do |
| 20:08.32 | drmessano | Is that important? |
| 20:08.55 | diegows | yes, because I think it isn't a misconfiguration problem |
| 20:09.08 | drmessano | It is |
| 20:09.29 | *** join/#asterisk Segnale007 (n=Pietro@host36-253-dynamic.36-79-r.retail.telecomitalia.it) |
| 20:09.34 | diegows | ok, so you know how to fix it? |
| 20:09.50 | drmessano | Factory reset and start over and be careful what you're editing |
| 20:10.12 | diegows | I tried a lot of configuration examples that I found in the web, read the manuals and I don't undertand why this stupid device puts 127.0.0.1 |
| 20:10.48 | *** join/#asterisk timburke|laptop (i=timburke@unaffiliated/timburke) |
| 20:10.51 | diegows | I tried with minimal configuration and nothings, 127.0.0.1 is always there |
| 20:10.55 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
| 20:11.02 | drmessano | Where do you see this 127.0.0.1 |
| 20:11.10 | drmessano | And what are you configuring exactly? |
| 20:11.11 | hardwire | heh |
| 20:11.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:11.40 | diegows | drmessano: I see it when I have sip debug enabled |
| 20:12.41 | diegows | I want to use the fxs port as a simple asterisk channel and the pstn to use the pstn line |
| 20:12.42 | ManxPower | are you sure it's not your Asterisk sending that? |
| 20:14.19 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 20:14.44 | diegows | look this: http://pastebin.com/d72822d4a |
| 20:14.54 | diegows | 192.168.0.1 is the spa |
| 20:15.19 | ManxPower | loproc: I suspect your problem is because you are allowing every codec under the sun. |
| 20:15.30 | drmessano | lol |
| 20:16.00 | ManxPower | so what is 192.168.0.2? |
| 20:16.03 | diegows | asterisk |
| 20:16.23 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:16.29 | ManxPower | do you have SRV lookups enabled in Asterisk or the device? |
| 20:16.48 | diegows | no |
| 20:17.15 | loproc | ManxPower: Yup, I've enabled all codecs... The HT286 dosn't have SRV lookups enabled |
| 20:17.25 | drmessano | diegows |
| 20:17.46 | diegows | drmessano: |
| 20:17.52 | ManxPower | loproc: allowing all codecs almost guarantees problems. Enable one codec for each peer/friend/user |
| 20:17.54 | drmessano | Factory reset the box.. set up the user/password/proxy on both the PSTN and Line1, and touch NOTHING else |
| 20:18.33 | diegows | drmessano: ok, i'll do again :) |
| 20:18.47 | drmessano | again is fine, do it correctly this time |
| 20:18.55 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:19.54 | loproc | ManxPower: OK - I'll limit the codecs. Should i enable SRV lookups in the HT286? |
| 20:20.27 | loproc | is watching Muppets... Great show! |
| 20:20.49 | ManxPower | no, don't enable it on asterisk or the device |
| 20:21.25 | loproc | ManxPower: OK - It was enabled on asterisk |
| 20:21.37 | Sonarcade | can using one of these: http://www.allelectronics.com/make-a-store/item/MT-105/3-JACK-2-LINE-MODULAR-T-ADAPTER/-/1.html lead to a short or overload? |
| 20:22.01 | ManxPower | Sonarcade: yes. |
| 20:22.06 | Sonarcade | oh ok |
| 20:22.20 | Sonarcade | I'm still not understanding just how phone cables can be powerful enough to do that |
| 20:22.26 | ManxPower | There is NOTHING YOU CAN PURCHASE that will allow you to plug unknown lines into unknown ports. |
| 20:22.29 | [TK]D-Fender | Sonarcade: What part of "unless you KNOW exactly how that jack is wired you can FRY your system" don't you get? |
| 20:22.40 | Sonarcade | ok |
| 20:22.42 | ManxPower | [TK]D-Fender: maybe he's retarded? |
| 20:22.43 | Sonarcade | I'm learning |
| 20:22.55 | ManxPower | Sonarcade: No, you are not learning. |
| 20:22.59 | [TK]D-Fender | Apparently not. |
| 20:23.04 | drmessano | ManxPower: Maybe he needs an etherkiller? |
| 20:23.09 | ManxPower | you are just trying everything you can to not accept our answers. |
| 20:23.18 | Sonarcade | no, I accept it now |
| 20:23.23 | Sonarcade | that was my last question about that issue |
| 20:23.29 | ManxPower | drmessano: he has no idea what the port on his PBX IS. |
| 20:23.30 | Sonarcade | just wanted to make sure |
| 20:23.45 | Sonarcade | that we were talking about the same thing |
| 20:23.57 | drmessano | So its an unknown RJ-11? |
| 20:24.00 | ManxPower | drmessano: so basically he wants to plug random lines into that port and expect it to work. |
| 20:24.03 | [TK]D-Fender | Sonarcade: Sure of what? That'd we give you the same consistent answer to the same question over & over again? |
| 20:24.05 | drmessano | HAW |
| 20:24.08 | Sonarcade | esp. since I don't have much familiarity with the electronic side of this matter |
| 20:24.46 | Sonarcade | [TK]D-Fender: I'm not equipped with the proper terminology so there's going to be a bit of orbiting around the crux |
| 20:24.48 | ManxPower | Sonarcade: What you are trying to do is the equiv of trying to do open heart surgery when you don't even know anatomy. |
| 20:24.49 | Sonarcade | and now I understand |
| 20:24.58 | *** join/#asterisk timburke (n=timburke@unaffiliated/timburke) |
| 20:25.02 | Sonarcade | at least that part |
| 20:25.13 | Sonarcade | that you guys have been trying to drive home |
| 20:25.16 | Sonarcade | sorry |
| 20:25.17 | ManxPower | Sonarcade: until you know what the port is on your pbx you cannot proceed. |
| 20:25.24 | drmessano | ManxPower: No, hes in the OR to perform open heart surgery and he's taking the wall clock apart |
| 20:25.26 | diegows | drmessano: I did it --> http://pastebin.com/m1c1e6b11 |
| 20:25.32 | diegows | 127.0.0.1 is still there |
| 20:25.43 | drmessano | What did you set for proxy? |
| 20:25.50 | drmessano | and you did the 3 settings, correct? |
| 20:25.56 | diegows | yes |
| 20:25.58 | drmessano | proxy, user, password? |
| 20:26.01 | drmessano | What did you set for proxy? |
| 20:26.01 | diegows | asterisk is my proxy server |
| 20:26.09 | drmessano | On Line 1 and PSTN? |
| 20:26.23 | diegows | proxy = 192.168.0.2 (asterisk) |
| 20:26.28 | diegows | yes, on line1 and pstn |
| 20:27.02 | drmessano | Pastebin the asterisk setup for both |
| 20:27.02 | ManxPower | diegows: put a copy of your sip.conf on pastebin.ca masking ONLY passwords. |
| 20:27.30 | diegows | ManxPower: ok |
| 20:28.13 | diegows | http://pastebin.com/m4d12d27d |
| 20:28.34 | ManxPower | diegows: your sip.conf does not have a [general] section. There is your problem |
| 20:28.45 | diegows | ManxPower: i didn't paste it |
| 20:29.11 | ManxPower | diegows: What part of "put a copy of your sip.conf on pastebin.ca masking ONLY passwords." did you not understand? |
| 20:29.39 | diegows | ManxPower: updated now |
| 20:29.40 | ManxPower | Now we have to waste time because you did not do what I requested. |
| 20:30.03 | ManxPower | not updated on my screen, even after a reload |
| 20:30.50 | drmessano | http://pastebin.com/m6be04bad |
| 20:30.53 | diegows | wget --no-cache http://pastebin.com/m6be04bad |
| 20:30.55 | diegows | :-P |
| 20:30.56 | drmessano | Lame, not updated.. new paste |
| 20:31.18 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 20:31.20 | drmessano | <PROTECTED> |
| 20:31.27 | ManxPower | drmessano: this is starting to get annoying. |
| 20:31.32 | diegows | ups, ok |
| 20:31.32 | drmessano | starting? |
| 20:31.59 | diegows | ManxPower: take it easy ok, i'm not a stupid newbie. |
| 20:32.09 | diegows | was only a mistake perfect people |
| 20:32.47 | ManxPower | diegows: at a bash prompt type "iptables -L -t nat" and put the output on pastebin. |
| 20:33.03 | loproc | ManxPower: Enabling only alaw, ulaw and g723 didn't do the trick... keep in mind that other connected phones work fine... |
| 20:33.05 | ManxPower | diegows: It is OK to be a newbie. It is not OK to not carefully instrucitons. |
| 20:33.10 | drmessano | diegows: Nobody is a newbie |
| 20:33.17 | ManxPower | loproc: STOP ENABLING G.273 |
| 20:33.26 | [TK]D-Fender | ManxPower: Grammar FAIL :) |
| 20:33.37 | ManxPower | loproc: stop enabling both ulaw and alaw. Enable one or the other. |
| 20:34.02 | loproc | ManxPower: Okay - I'll stick to one of them... |
| 20:34.08 | diegows | iptables -L -t nat -vn http://pastebin.com/m773b1657 |
| 20:35.07 | rob0 | fwiw, "iptables-save" is far better than iptables -L |
| 20:35.21 | ManxPower | diegows: now the output of "netstat -rn" and pastebin the output. |
| 20:35.37 | ManxPower | rob0: I was just trying to see if there was some iptables entry that was nating |
| 20:36.11 | diegows | my new sip.conf: http://pastebin.com/d67722e2 |
| 20:36.23 | ManxPower | diegows: and the output of " iptables -L" |
| 20:36.24 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 20:36.31 | diegows | rob0: I agree |
| 20:37.05 | ManxPower | diegows: again, you are getting ahead of yourself. Your old sip.conf was perfectly fine. |
| 20:37.13 | diegows | iptables -L: http://pastebin.com/m7074fcb7 |
| 20:37.56 | diegows | ManxPower: I want to start again with the most simple configuration |
| 20:38.07 | ManxPower | diegows: Best of luck with that. |
| 20:38.30 | diegows | iptables shouldn't be a problem, because the 127.0.0.1 is in sip headers |
| 20:39.41 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 20:39.45 | ManxPower | diegows: and there never SOULD be 127.0.0.1 in a SIP header. We are not dealing with "shoulds" here. |
| 20:40.09 | diegows | i agree |
| 20:41.12 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-b144a677ad6e4be4) |
| 20:41.34 | diegows | netstat -rn http://pastebin.com/d4f3548bc |
| 20:41.39 | wnspark | are there any free ways to use Asterisk to forward to a cell phone? |
| 20:42.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:44.21 | beek | wnspark: Free ways? I don't know. Cheap ways? Yep. I use PAYG @ Teliax. Other ITSPs offer similar plans. |
| 20:45.40 | rob0 | I think there are some free termination services, but there are limits and strings attached, you probably wouldn't want to rely on it. |
| 20:48.01 | wnspark | Thanks beek, I will look into Teliax. What other services do people in here use for termination services? |
| 20:49.01 | beek | ManxPower: Do you usually leave your PRIs reset to the default 3600 seconds or do you disable the "feature?" |
| 20:49.02 | loproc | ManxPower: Now I've tried alaw and ulaw one-at-a-time but neither of them fixes the problem... Anyhow, I'll make sure not to enable them both |
| 20:51.05 | *** join/#asterisk jeffspeff2 (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:53.12 | ManxPower | beek: I've never had to change that option. |
| 20:54.28 | diegows | any ideas? |
| 21:00.24 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 21:01.14 | [TK]D-Fender | wnspark: ... |
| 21:01.18 | [TK]D-Fender | ~itsplist-us |
| 21:01.19 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 21:01.20 | [TK]D-Fender | ~itsplist-ca |
| 21:01.21 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 21:01.45 | wnspark | jbot: thanks that will help out a lot. |
| 21:01.45 | jbot | my pleasure, wnspark |
| 21:03.15 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 21:04.59 | ManxPower | "900MHz MiniPCI 802.11G 200mW" <-- Well I'm not buying from THAT company. |
| 21:10.02 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
| 21:10.35 | *** join/#asterisk zecrazytux (n=zecrazyt@boulz.org) |
| 21:10.40 | Cybertoy | wnspark: I've been using voipcheap.com for more than 2 years. |
| 21:11.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:11.35 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-4dcdfb15a9eed113) |
| 21:12.45 | beek | ManxPower: Thanks |
| 21:20.28 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 21:20.44 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:20.44 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:21.07 | orkid | Cybertoy: u dont get disconnects, no connects, static/etc ? |
| 21:26.09 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:26.37 | *** part/#asterisk zecrazytux (n=zecrazyt@boulz.org) |
| 21:33.58 | Tuxguy | Probably a dumb question, but does asterisk generate the dial tone, or is that the client that does it? |
| 21:34.44 | ManxPower | Tuxguy: Yes. No. State the tech you want to know about. |
| 21:36.05 | Tuxguy | For instance, when I dial a number in my sip client... it makes the dial tone sound. I was just wondering if that was something in the client, or the asterisk server... I was thinking maybe the client does it, but gets told when to make it by the asterisk server? |
| 21:36.24 | ManxPower | For SIP the dialtone and digit collection is done by the SIP device. |
| 21:37.06 | ManxPower | This may or may not apply to other technologies -- depending on what those technologies are. |
| 21:37.06 | loproc | Noone with any clue to why a handytone adapter has no audio for the first 40 seconds? |
| 21:37.33 | ManxPower | loproc: I sort of assume if someone had an answer to your question they would have provided it. |
| 21:37.54 | loproc | Right... |
| 21:38.43 | ManxPower | Tuxguy: In fact Asterisk doesn't even know you went off hook or that you are dialing until the SIP device sends the dialed digits to Asterisk as one block of numbers. |
| 21:38.53 | Tuxguy | oh ok |
| 21:39.06 | Tuxguy | I wasnt sure how that all worked. I am extremely new to telephony. |
| 21:39.27 | ManxPower | NEVER assume that Zaptel, DAHDI, SIP. MGCP, etc work the same. |
| 21:40.12 | lmadsen | because they don't :) |
| 21:40.49 | Tuxguy | ? |
| 21:40.51 | Tuxguy | me? |
| 21:42.38 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 21:43.31 | zchaos | can anyone here help me figure out why my PBX system cannot make calls? no dial tone or anything... however i can receive calls and the ata is registering.... |
| 21:43.54 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 21:43.54 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 21:52.26 | [TK]D-Fender | zchaos: show us the failed call with debug in a pastebin |
| 21:52.28 | [TK]D-Fender | ~pb |
| 21:52.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:57.17 | Tuxguy | Is SIP tcp or udp? |
| 21:57.28 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 21:57.29 | [TK]D-Fender | Tuxguy: Can be either |
| 21:57.47 | [TK]D-Fender | Tuxguy: Until 1.6 * only supported SIP via UDP |
| 21:57.52 | Tuxguy | I tried to telnet locally to port 5060 and it says connection refused. |
| 21:57.59 | Tuxguy | But, my SIP client can connect fine |
| 21:58.13 | [TK]D-Fender | Tuxguy: See above. What ver are you running? |
| 21:58.24 | Tuxguy | 1.6 |
| 21:58.38 | [TK]D-Fender | Tuxguy: there may be certain config options to enable this support |
| 21:58.52 | Tuxguy | Ah ok, if not, I can upgrade. |
| 21:58.52 | [TK]D-Fender | Tuxguy: basically you should just use the deafult until you have a reason to care. |
| 21:59.08 | Tuxguy | Trying to build a perl client that can prank my wife :) |
| 21:59.13 | [TK]D-Fender | Tuxguy: upgrade to what? You're already on 1.6 |
| 21:59.14 | Tuxguy | Since Net::SIP sucks |
| 21:59.28 | [TK]D-Fender | Tuxguy: and what would this client do? |
| 21:59.45 | Tuxguy | Just connect, and do an invite, then play a prank, and loop every 30 seconds. |
| 22:00.07 | [TK]D-Fender | Tuxguy: No need for that. Go read up on "call files" and "AMI Originate" on the WIKI |
| 22:00.09 | [TK]D-Fender | ~wikis |
| 22:00.09 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:00.39 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:01.30 | Tuxguy | Cool, I may end up doing that. I was just trying to roll my own. |
| 22:01.33 | loproc | Bollocks! The handytone-thing seemed to be caused by my router... Darned thing! |
| 22:02.08 | [TK]D-Fender | Tuxguy: Don;'t waste time trying to reinvent the qwheel. Go learn * THEN look at the ways to do what you want. |
| 22:02.36 | Tuxguy | Not sure if you would know this, but are there any PHP sip modules or modules for working with * ? |
| 22:02.52 | drmessano | ... |
| 22:03.17 | drmessano | TK put it best |
| 22:03.17 | [TK]D-Fender | Tuxguy: PHZP is not typically an "interactive" language, and no, and I doubt anyone would try if they even could |
| 22:07.04 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 22:07.29 | [TK]D-Fender | Tuxguy: "To make an apple pie from scratch one must first create the Universe" <- this might work as an approach to * you are still wasting incredible amounts of effort on something that would be otherwise trivial. |
| 22:08.42 | gambler1 | hi, does anyone know is it possible to manipulate inside the dialplan (agi script) with cdr data before * write it to a file od db? |
| 22:09.56 | drmessano | You're overthinking.. Most people with coding knowledge or with the desire to hack install asterisk as a protocol handler and start looking at how they can hook into with the skills they have.. Which is fail. This isn't an HTTP or a FTPd or something equally as stupid of a unitasker. Learn asterisk a little and you'll find what you can and can't do, and where you need to pick up. |
| 22:11.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:20.05 | *** part/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 22:20.57 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 22:30.04 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-74-234.vif.net) |
| 22:38.08 | flush | hi |
| 22:38.37 | flush | if i buy an 1940 old inter phone on ebay, can i make it work with my asterisk box.. like plug a dial pad on it and make it work properly without much complications ? |
| 22:40.06 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 22:40.46 | [TK]D-Fender | flush: Link to model please.. |
| 22:40.55 | flush | ok gimme sec |
| 22:42.00 | flush | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=130266569981&ssPageName=ADME:X:RTQ:US:1123 |
| 22:42.55 | orkid | wow cool :) |
| 22:43.02 | flush | bid fight |
| 22:43.05 | orkid | lol |
| 22:43.07 | orkid | naw |
| 22:43.11 | flush | haha, would it work you think ? |
| 22:43.25 | orkid | ud have to guy it probably, and rerun wiring, |
| 22:43.28 | flush | i have found another one, guy says in item description "there are no connector it would need to be modified to work with landline"| |
| 22:43.31 | orkid | gut |
| 22:43.44 | orkid | yeh.. basically, u get a cool shell, and then u do all the internals/etc |
| 22:43.47 | flush | is this pretty simple |
| 22:43.52 | flush | yea |
| 22:44.13 | flush | what do you think of this one http://cgi.ebay.ca/VINTAGE-ERICSSON-BAKELITE-1950s-PHONE-DIAL-BUTTON_W0QQitemZ170277088055QQcmdZViewItem?hash=item170277088055&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318 |
| 22:44.15 | orkid | depends how good u are with electronics and stuff like that |
| 22:44.57 | orkid | i like the first one more :) |
| 22:44.59 | [TK]D-Fender | flush: Well if its got a hook flash, it should be compatible with an ATA if you include a TT generator |
| 22:45.00 | jaytee | why would anyone want a phone like that? |
| 22:45.07 | [TK]D-Fender | jaytee: Nostalgia... |
| 22:45.18 | [TK]D-Fender | jaytee: It ain't what it used to be you know! |
| 22:45.41 | jaytee | Ok, maybe. I could probably go for a red rotary dial phone |
| 22:45.59 | flush | [TK]D-Fender i would need an ata box only if i use voip or to plug it on the landline ? |
| 22:46.24 | [TK]D-Fender | flush: "something" that will let * use it |
| 22:46.26 | orkid | take the internals out of a sip phone :) |
| 22:46.33 | jaytee | I'm at the Holiday Inn in Huntsville and Jared Smith just checked in. |
| 22:46.33 | orkid | and make vintage styled IP phones :) |
| 22:46.50 | orkid | who the fk is 'jared smith' |
| 22:46.56 | orkid | and why should we care |
| 22:46.58 | flush | no idea |
| 22:47.23 | flush | [TK]D-Fender sorry but im not sure i get it.. this is an old phone and to use it with my bell's landline i would need an ata? |
| 22:47.44 | jaytee | orkid, have you read the book? he's one of the 3 authors and is going to be the instructor in my class tomorrow. |
| 22:47.47 | [TK]D-Fender | flush: so this question has nothing to do with *? |
| 22:47.47 | orkid | if u knew what an ata was u wudnt b asking :) |
| 22:47.52 | orkid | no i havnt jaytee |
| 22:48.05 | orkid | well depending on what book |
| 22:48.14 | jaytee | ATFOT |
| 22:48.30 | [TK]D-Fender | ~cluebat orkid |
| 22:48.31 | jbot | ACTION pulls out a ClueBat (tm) and thwaps orkid. |
| 22:48.37 | orkid | asterixdicks.org ? |
| 22:48.45 | flush | <PROTECTED> |
| 22:50.31 | flush | [TK]D-Fender what is a "TT generator" ? |
| 22:50.35 | [TK]D-Fender | flush: then you need something to plug it into that will let * use it |
| 22:50.40 | [TK]D-Fender | flush: Touch Tone. |
| 22:50.43 | [TK]D-Fender | flush: DTMF |
| 22:51.17 | jaytee | converts the pulses from rotary dial phones to DTMF tones |
| 22:51.27 | flush | kk |
| 22:51.29 | orkid | get a sip phone and take out the innards, and putem in ur cool fone |
| 22:51.52 | orkid | but then u have a new pad wif n old fone |
| 22:51.53 | jaytee | wouldn't be the same thing. |
| 22:51.56 | flush | i dont have voip provider i guess i should take a plan |
| 22:52.56 | orkid | so i was driving thru arkansa, and the coolest thing they had there on this one stop was a mcds |
| 22:53.00 | orkid | :O |
| 22:53.36 | [TK]D-Fender | flush: * is software... unless you by some kind of interface your phone isn't going to magically talk with * |
| 22:53.49 | orkid | put the phone in the computer :P |
| 22:53.57 | flush | [TK]D-Fender affirmative |
| 22:54.37 | *** join/#asterisk RobertLaptop (n=rmiddle@m910736d0.tmodns.net) |
| 22:54.40 | phix | [TK]D-Fender: :D |
| 22:55.46 | flush | but one more thing, the first ebay link i posted is about an "inter phone" it does not seem to have a dial function at all.. could it be possible to add a dial pad ? |
| 22:56.02 | [TK]D-Fender | flush: I already answered this... |
| 22:56.22 | [TK]D-Fender | flush: and then answered your failure to follow terms related to it |
| 22:57.03 | flush | hrmm |
| 22:58.18 | Spirits-Sight | Does anyone know of a service that allows a number of calls at once using VOIP, right now I have Vocalocity and I want to cut cost more, right now we only have a 800 number and one ext, I would like to be able to make and recive calls using the 800 number I pay 60 dollars a month for this, this is for a non-profit, I don't have the money to keep paying so I figured setup my own |
| 23:01.09 | [TK]D-Fender | Spirits-Sight: What do you pay for now, and what does that give you exactly? |
| 23:04.55 | *** join/#asterisk timburke (n=timburke@unaffiliated/timburke) |
| 23:05.02 | Spirits-Sight | 19 for unlimitied incoming calling but 39.95 for the base extion which is unlimited incoming calls and out going |
| 23:06.21 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:06.39 | [TK]D-Fender | Spirits-Sight: what about the 800, no LD? |
| 23:07.17 | Spirits-Sight | I don't pay any more then the 20 for it, no min, no based on ld nothing |
| 23:07.43 | [TK]D-Fender | Spirits-Sight: And they only allow you 1 channel? |
| 23:08.36 | Spirits-Sight | what do you mean one channel? if you mean to call only one no, I can call like 4 people at once as my phone has for line buttons and I have tested that |
| 23:12.46 | Spirits-Sight | how much space should asterisk have for voicemail stuff |
| 23:13.41 | *** join/#asterisk af_ (n=getsmart@88-149-230-138.dynamic.ngi.it) |
| 23:16.26 | Spirits-Sight | how much space should asterisk have for voicemail stuff and what type of part should it use |
| 23:16.43 | *** join/#asterisk [netman] (n=netman@200.Red-88-25-139.staticIP.rima-tde.net) |
| 23:17.27 | [TK]D-Fender | Spirits-Sight: depends on how much VM you need to store |
| 23:18.34 | Spirits-Sight | Not a lot for now, I never have more then 10 :( I not like much LOL |
| 23:19.23 | Spirits-Sight | what type of part should it be |
| 23:19.44 | Spirits-Sight | I am installing Ubuntu-Server right now and want to setup for this setup of Asterisk |
| 23:20.28 | [TK]D-Fender | Spirits-Sight: Ubuntu has a few special concerns for installing *, but nothing too large |
| 23:20.40 | [TK]D-Fender | Spirits-Sight: You system should not be an issue |
| 23:22.36 | Spirits-Sight | why you say this? |
| 23:22.52 | Spirits-Sight | I am new to all of this so please please help and stay on line and help |
| 23:24.02 | [TK]D-Fender | Spirits-Sight: Ubuntu treats root funny. There are a few special steps todo and there are plenty of guides you can google up for this |
| 23:24.41 | ManxPower | Spirits-Sight: WAV format is something around 64k per second. WAV49 would be about 13K per second. |
| 23:25.01 | Spirits-Sight | thats good |
| 23:25.09 | ManxPower | I like WAV49 because the files are reaonably small and they will play on most any player and OS by default. |
| 23:25.41 | ManxPower | so, figure out how many seconds of voicemail you want to be able to keep at one time and you can figure out how much disk space you need. |
| 23:26.02 | Spirits-Sight | So what would be a good amount for VM if had say three extions and had music on hold |
| 23:26.24 | Spirits-Sight | ok so if I have say 10gb of space that would be plenty |
| 23:26.30 | [TK]D-Fender | Spirits-Sight: You just gave a TINY # for your requirement. this is an afterthought |
| 23:26.47 | ManxPower | Or you could just assume 4GB and use WAV49 and take the default of "no more than 100 msgs in a mailbox" setting and not ever have to worry about it again. |
| 23:27.33 | Spirits-Sight | I like that LOL, so 4gb would be good :-) what type of partion should it be |
| 23:27.51 | [TK]D-Fender | Spirits-Sight: If you are clueless about Linux as well as * you are serious trouble |
| 23:28.19 | ManxPower | 4 GB with WAV you get something around 4660 mins of voicemail storage. |
| 23:28.52 | ManxPower | I do actually recommend putting voicemail on a separate partition and not the same partition as /var. |
| 23:29.08 | ManxPower | If you run out of disk space Askerisk will freak out. |
| 23:30.12 | ManxPower | Oh, sorry! That is 4660 HOURS of voicemail. |
| 23:31.11 | Spirits-Sight | I know there is many type of parts and I know for normal use ext3 seems to be the choice, and yes I would be clueless thats why I am asking for assistance with this, this is for a non-profit setup I am trying to do this because I can not aford to keep paying all the money I am paying right now, thats also why I ask about which VOIP company would be good to use so I can use that with the system |
| 23:31.53 | Spirits-Sight | wow, HOURS 4660 thats plonty of time LOL, I could get away with a 1gb of space |
| 23:32.04 | ManxPower | Spirits-Sight: Analog .vs. ISDN PRI .vs. Internet ITSP (provider) .vs. QoS's ITSP all depends on how reliable it has to be. |
| 23:33.27 | ManxPower | A friend has Vitelity and using a cablemodem connection just said to me about 5% of the time when he wants to use the phone it does not work. |
| 23:33.49 | ManxPower | So that would be how reliable it is for him using the "Internet ITSP" type of service. |
| 23:34.25 | ManxPower | On the other hand a client using Asterisk and a PRI has something near 99.5% uptime including nights, weekends, etc. |
| 23:34.51 | ManxPower | so maybe an hour per month of downtime and most of that is at night and scheduled. |
| 23:35.21 | Spirits-Sight | wow thats sounds good |
| 23:36.01 | Spirits-Sight | ManxPower: I PM you |
| 23:36.02 | ManxPower | They do get telephone line outages every once in a while. but that is mainly because they are near New Orleans. |
| 23:36.07 | ManxPower | Spirits-Sight: I don't use PM |
| 23:36.25 | Spirits-Sight | oo ok :-( |
| 23:36.46 | Spirits-Sight | ok, so what type of part should I use for the partion |
| 23:37.34 | ManxPower | For low budget I recommend (in the USA/Canada) for a small office get a couple of analog lines and the required analog cards, use VoIP for the overflow. have the analog lines hunt to the VoIP numbers when the analog lines are busy. put fax on it's own dedicated line. |
| 23:38.00 | ManxPower | For other parts of the world (especially europe) I suggest ISDN BRI instead of analog lines. |
| 23:38.14 | ManxPower | same VoIP for the occasional usage. |
| 23:38.29 | ManxPower | Spirits-Sight: You should go read the Asterisk Book |
| 23:38.31 | ManxPower | ~book |
| 23:38.31 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:40.16 | ManxPower | For larger offices replace the analog lines and/or ISDN BRI with an ISDN PRI with T-1/E-1 card |
| 23:40.32 | jameswf | j1 in japan |
| 23:42.16 | Spirits-Sight | wow, to much over my head, I need a cheap way to do it using VOIP as I don't have analog in my house and I don't have money to buy cards as I am using a laptop for the server and only have eithernet and wifi for connection and a router, I have eithernet type phone that is plug into my router that is VOIP |
| 23:43.53 | ManxPower | Spirits-Sight: you have a lot to learn. Linux, Telecom, Networking (including routing, ports and NAT), and Asterisk. This is time consuming and hard work. |
| 23:44.16 | *** join/#asterisk RobertLaptop (n=rmiddle@m910736d0.tmodns.net) |
| 23:45.12 | Spirits-Sight | I agree, I wish I did not have to do this rout at this time but I need to save money because I don't have much my self |
| 23:45.52 | Spirits-Sight | are there some good readings to do only that are easy to understand |
| 23:47.02 | ManxPower | Spirits-Sight: No. |
| 23:47.41 | [TK]D-Fender | Spirits-Sight: you will have much to learn for this. |
| 23:47.57 | [TK]D-Fender | Spirits-Sight: there is the BOOK, and here for some inspiration : |
| 23:47.59 | [TK]D-Fender | ~jerjerguide |
| 23:47.59 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 23:50.49 | Spirits-Sight | I would look at the book but I am blind and can not see it |
| 23:51.52 | ManxPower | The book is available in PDF format |
| 23:52.46 | Katty | mmm, ice creams. |
| 23:53.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 23:58.48 | Spirits-Sight | could you please tell me where and how much does it cost? |
| 23:59.13 | Katty | yard sale, .50 |