| 00:00.03 | jblachly | jart: I really don't have time to compile and test a new branch of asterisk. If the correction made it (by accident) into the trunk then i'll hope it makes it into 1.2 at some point. Otherwise, I have medical boards to study for and really don't have time to mess with this any more. Just trying to help. Thanks again. |
| 00:00.09 | *** join/#asterisk apardo (n=apardo@87.217.146.74) |
| 00:00.48 | jart | jblachly: if you don't care about losing karma on the bug tracker, and are /pretty/ sure you're right, just post a bug telling them abou tit |
| 00:00.57 | jart | ugh |
| 00:02.13 | file | all we ask is for people to try the latest version, is that too much to ask for? |
| 00:03.07 | jart | i agree, he acts as though people writing /Free/ software owe him something |
| 00:03.13 | russellb | well, if he is smart enough to look at the code and say there is a bug, why does he need to test? |
| 00:03.24 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 00:03.38 | russellb | if he says he looked and the code hasn't changed, why not give him the benefit of the doubt? |
| 00:04.55 | toerkeium | guys, I am trying to read the status of an extensions with php AMI, how does the * tells me it's the end of the headers it's sending ? any idea? |
| 00:05.17 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 00:05.32 | jart | toerkeium: the statuscomplete event |
| 00:05.56 | toerkeium | excellent, thanks |
| 00:06.03 | jart | toerkeium: assuming you're issuing a status event |
| 00:06.25 | toerkeium | ExtensionState would be? |
| 00:08.34 | jart | that just gives a single response |
| 00:09.02 | toerkeium | what would be the best way to check the status of an extension? I mean, busy, in use, etc |
| 00:09.56 | jart | you do know that extensionstate requires that you use those goofy hint extensions right? |
| 00:10.04 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 00:10.22 | toerkeium | hmm no, in fact no idea what you talking about |
| 00:11.08 | jart | toerkeium: now by extensions state, what exactly do you mean? do you mean whether or not one if you sip phones is currently being used? |
| 00:11.31 | jart | one of* |
| 00:12.03 | toerkeium | yes, that's what I want to check first.. first extensions (SIP/100) and then the same for channels |
| 00:12.40 | jart | toerkeium: I recommend you issue a status action. it will tell you about every single active channel on the system. If any of those channels start with "SIP/100-", then it's active |
| 00:13.18 | jart | it works by you sending Action: Status\r\n\r\n, it sends back a bunch of Event: Status and ends with an Event: StatusComplete |
| 00:13.42 | toerkeium | excellent, going to try that right now |
| 00:14.22 | jart | anyway i gotta bounce padna' |
| 00:14.26 | jart | 1 |
| 00:15.14 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 00:17.24 | toerkeium | Channel, when the call is between SIPs are the called right? |
| 00:19.00 | toerkeium | I mean, if a SIP calls another SIP, the channel shown in Action: status will be the called SIP ? |
| 00:32.07 | *** join/#asterisk jeffgus (n=jeffgus@shaper.zimage.com) |
| 00:32.35 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:34.53 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:37.58 | shmaltz | anybody know how I can make sure that a variable set in one macro is available thruout the duration of the call? |
| 00:39.53 | *** join/#asterisk MacWeenie (n=chatzill@cpe-74-65-228-134.nyc.res.rr.com) |
| 00:40.06 | toerkeium | anyone knows what could be the reason that if I originate a call from a php script with AMI if both SIP softphones hungup, and the call stops, a "sip show channels" will still show both channels active? |
| 00:42.19 | shmaltz | toerkeium, what does your extensions.conf look like? |
| 00:43.15 | toerkeium | it's auto generated by freePBX, but if I start the call from the SIP phone insead of the php script, it doesn't show the channels active |
| 00:43.48 | shmaltz | toerkeium I wish I knew what you were talking about, can you repeat that last one in english? |
| 00:43.59 | toerkeium | heh sure |
| 00:44.25 | toerkeium | ok, I am making a php script which originates a call from SIP/1501 to SIP/1503 |
| 00:44.34 | toerkeium | if I start the very simple script, it works ok |
| 00:44.45 | toerkeium | each SIP get connected |
| 00:44.56 | toerkeium | when I stop the call, from the softphone |
| 00:45.04 | toerkeium | and both SIP are stopped |
| 00:45.18 | toerkeium | if I issue a "sip show channels" I still see the channels active |
| 00:45.29 | toerkeium | this doesn't happen if I originate the call from the SIP device |
| 00:45.39 | toerkeium | understand what I mean? |
| 00:46.27 | shmaltz | toerkeium, how does the php script make the call? |
| 00:46.40 | toerkeium | let me paste it in pastebin.ca |
| 00:47.36 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 00:48.06 | rene1 | i have installed the astmanproxy and the stability on the system has improved i would say |
| 00:48.34 | toerkeium | shmaltz: http://pastebin.ca/119850 |
| 00:49.59 | rene1 | i just cant make use of the http interfase. i have enabled suport for both standard|http interfaces, i can get astmanproxy to respond via http however it just shows continually the result of the agents action, and wont take any actions via GET |
| 00:50.23 | rene1 | havent tried with post or disabling standard i/o and using just http i/o tho |
| 00:52.33 | shmaltz | toerkeium, I don't know PHP nor am I interested in learning it on this beautifull sunday, |
| 00:52.47 | shmaltz | toerkeium, how do you make the call from PHP? |
| 00:53.20 | shmaltz | now I got it you using the manager interface |
| 00:53.25 | toerkeium | heh sorry |
| 00:53.33 | toerkeium | yes, I am using the manager interface |
| 00:53.42 | shmaltz | toerkeium, may I ask again to see your extensions.conf? |
| 00:53.45 | toerkeium | so I send to * the headers to Originate a call |
| 00:53.50 | toerkeium | sure, let me paste it |
| 00:54.00 | *** join/#asterisk tempest1 (n=asf@adsl-153-56-119.chs.bellsouth.net) |
| 00:56.36 | *** join/#asterisk tempest1 (n=asf@adsl-153-56-119.chs.bellsouth.net) |
| 00:59.23 | rene1 | toerkeium, the astmanproxy is a good idea people here had told me.... it is very likely to have asterisk crash when you are using the manager interfase, the astmanproxy does it nicely for you |
| 01:00.05 | rene1 | to connect concurrently to AMI |
| 01:00.30 | rene1 | you dont have to change a line to use astmanproxy, it just sits happily between asterisk and your program |
| 01:01.21 | toerkeium | shmaltz: http://pastebin.ca/119872 |
| 01:01.22 | toerkeium | pretty large |
| 01:01.38 | toerkeium | rene1, what is it? a 3rd party program? |
| 01:03.12 | shmaltz | toerkeium, why are you using exen => h???? |
| 01:04.20 | toerkeium | no idea really, that config is generated by freePBX |
| 01:04.21 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 01:04.23 | shmaltz | toerkeium, comment out all the exten => h and try again |
| 01:04.29 | toerkeium | let me try |
| 01:05.43 | riddlebox | is this a correct way to use gotoiftime? http://pastebin.ca/119734 |
| 01:07.01 | shmaltz | riddlebox, looks ok to me |
| 01:07.10 | riddlebox | cool thanks |
| 01:07.44 | shmaltz | riddlebox, I'm just not sure if sat-sun works, but I'm assuming it does |
| 01:08.25 | riddlebox | yeah thats what I am wondering, but you can do mon-fri so I figured it was ok |
| 01:09.27 | shmaltz | riddlebox, but you cant do 23:00-6:00 therefore you might not be able to do sat-sun or maybe you could, I'm not sure |
| 01:10.17 | riddlebox | but I want it to do 6am to 11am on saturdays and sundays |
| 01:10.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 01:13.02 | shmaltz | riddlebox, then do like this: |
| 01:13.03 | shmaltz | exten => whatever,1,GotoIfTime(6:00-10:59|sat|*|*?whatever) |
| 01:13.05 | shmaltz | exten => whatever,2,GotoIfTime(6:00-10:59|sun|*|*?whatever) |
| 01:13.38 | riddlebox | and then a exten => whatever,3, to do go the rest of the time correct |
| 01:14.02 | shmaltz | riddlebox, correct, in fact my gotofitimes are sometimes 20-30 lines long |
| 01:14.15 | toerkeium | shmaltz: it doesn't happen now, but now I uncommented the => h lines, and it's still working well (I did the reload) |
| 01:14.18 | toerkeium | strange |
| 01:14.40 | riddlebox | shmalts, this is a way for my mom to stop calling me early on the weekends :p |
| 01:14.52 | shmaltz | toerkeium, then the prolem is *not* with the h exten |
| 01:15.18 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 01:15.46 | shmaltz | riddlebox, that is till she doesnt get smart enough to log onto irc.freenode.net/#asterisk and ask shmaltz how to bypass it :P |
| 01:16.04 | riddlebox | if she does that I would have to say wow ;p |
| 01:16.11 | shmaltz | lol |
| 01:16.35 | shmaltz | last time my mom tried using a mouse she was all frustrated that it doens't move in the direction she wants :) |
| 01:17.37 | riddlebox | lol I tell her all the time, that I am young and still go to bars on the weekend and I stay out till 4 in the morning and I do not want her to call me early |
| 01:18.07 | shmaltz | riddlebox, how young are you? |
| 01:18.25 | riddlebox | 26 |
| 01:18.39 | shmaltz | still single I guess |
| 01:18.54 | riddlebox | nope I live with my girlfriend,....kinda single I guess |
| 01:19.14 | shmaltz | any kids? |
| 01:19.17 | toerkeium | put the phone at your girlfriend side.. huh? |
| 01:19.18 | riddlebox | nope |
| 01:19.18 | toerkeium | :P |
| 01:19.44 | shmaltz | lol |
| 01:19.49 | riddlebox | she can sleep through anything, I jump up at any ring or beep |
| 01:20.04 | shmaltz | how long you been living togther? |
| 01:20.08 | riddlebox | 1 year |
| 01:20.17 | shmaltz | you going to marry her? |
| 01:20.38 | riddlebox | maybe someday |
| 01:20.47 | shmaltz | do you want to? |
| 01:20.54 | *** join/#asterisk chiardon (i=Dorphals@pcsp168-254.supercabletv.net.co) |
| 01:21.00 | chiardon | Hello? |
| 01:21.16 | shmaltz | chiardon, helo |
| 01:21.20 | riddlebox | she only wants to use linux now, so there is a good chance :P |
| 01:21.26 | shmaltz | lol |
| 01:21.32 | shmaltz | ubuntu? |
| 01:21.56 | riddlebox | yup I only run ubuntu these days, life is easy when you use it |
| 01:21.58 | chiardon | I was kinda wondering... is there anyway to know which channel answered which call on a queue? |
| 01:22.19 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:22.21 | chiardon | I mean, I want my agents to see the callers data onscreen |
| 01:22.24 | shmaltz | sorry I got disconnected |
| 01:22.26 | chiardon | but I'm using Zap phones |
| 01:22.34 | chiardon | I was kinda wondering... is there anyway to know which channel answered which call on a queue? |
| 01:22.36 | shmaltz | chairdon, what type of phones? |
| 01:22.56 | chiardon | Regular ordinary cheap, non techy phones :( |
| 01:22.56 | shmaltz | riddlebox, why dont you propose? |
| 01:23.08 | shmaltz | chiardon, have you ever heard of CID? |
| 01:23.26 | chiardon | shmaltz ... yeah... but how do I send it to my agent's computer :P |
| 01:23.39 | shmaltz | chiardon, FOP |
| 01:23.42 | shmaltz | ~fop |
| 01:23.43 | jbot | An XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel |
| 01:23.55 | riddlebox | shmaltz, when she is done with school |
| 01:24.15 | shmaltz | riddlebox, why? so she can meet someone else that likes mac? |
| 01:24.27 | shmaltz | or freebsd god forbid? |
| 01:25.52 | riddlebox | lol, no it is easier then |
| 01:25.56 | chiardon | shmaltz --> no other way around? I kinda wanted to make it an automated process |
| 01:26.15 | chiardon | I kinda dont want them messing around with calls that are not for them ;) |
| 01:26.54 | shmaltz | chiardon, what exactly do you want? |
| 01:28.18 | chiardon | Agent's phone rings (with call from queue) -> Agent picks up phone and at the same time on the agent's computer a web page appears with the data relevan |
| 01:28.36 | chiardon | I have the data associated with the cid, so that's the least of my worries |
| 01:28.44 | shmaltz | chiardon, you can configure FOP for that, and they wont be able to do anything |
| 01:29.19 | chiardon | shmaltz... mind giving me the site again? That last one was a funky xml-thingy |
| 01:29.39 | chiardon | :$ |
| 01:29.51 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 01:29.58 | shmaltz | http://www.asternic.org/ |
| 01:30.03 | JunK-Y | some1 having audio problem with jingle here? |
| 01:30.10 | chiardon | thnx! |
| 01:30.22 | toerkeium | anyoe know where to get a list of status codes? like 0 (available) ? |
| 01:31.49 | chiardon | bye |
| 01:32.23 | toerkeium | found |
| 01:43.14 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 01:44.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 01:46.02 | shmaltz | I like this story: |
| 01:46.03 | shmaltz | http://www.stltoday.com/stltoday/business/columnists.nsf/techtalk/story/A0F7FD49EFA6565A862571BF006C005A?OpenDocument |
| 01:47.25 | riddlebox | shmaltz, you from stl? |
| 01:47.28 | Un1x | anyone ever buy a DID from DIDww.com? |
| 01:47.45 | Un1x | !didww |
| 01:47.49 | Un1x | err |
| 01:47.51 | Un1x | ~didww |
| 01:47.58 | Un1x | ~did |
| 01:48.00 | jbot | i heard did is Direct Inward Dialing |
| 01:48.12 | Un1x | lol i alread knew that... |
| 01:48.37 | Un1x | anyone ever Used... DIDWW did's before... because i need to find out how do i mapp it to my server, so i can receive calls :)? |
| 01:51.21 | toerkeium | guys, what does it means this line: |
| 01:51.25 | toerkeium | 201.216.206.221 1503 48b4d32a146 00102/00000 gsm No Init: INVITE |
| 01:51.30 | toerkeium | the Init: INVITE ? |
| 01:57.56 | *** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net) |
| 02:01.11 | Un1x | Damn, Asterisk is dead.. |
| 02:02.19 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 02:08.56 | riddlebox | Un1x, what do you mean? |
| 02:09.33 | *** join/#asterisk topping (n=topping@209-204-141-95.dsl.static.sonic.net) |
| 02:09.57 | *** join/#asterisk tipizo (n=tipizo@c-65-96-95-140.hsd1.ma.comcast.net) |
| 02:11.26 | tipizo | I'm running shorewall as my firewall... I can call.. The person can hear me, but i cannot hear the person. Please help |
| 02:12.10 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 02:12.25 | ManxPower | tipizo, turn off the firewall. does it start working? |
| 02:12.52 | tipizo | you mean stop shorewall? |
| 02:12.54 | riddlebox | do you have all ports open? I think in rtp.conf it has ones that need to be open as well as 5060 and 5061 |
| 02:13.25 | tipizo | I have 5060:5082 open |
| 02:14.12 | tipizo | and 10000:20000 |
| 02:14.21 | ManxPower | tipizo, You did not answer my question. |
| 02:14.31 | ManxPower | Does it work when you turn off the firewall? |
| 02:14.47 | tipizo | nope... I try that already |
| 02:15.29 | ManxPower | so with sorewall off you can still make a call, but there is still no audio? |
| 02:15.41 | *** join/#asterisk michaelo (n=michaelo@adsl-12-115-142.gsp.bellsouth.net) |
| 02:17.08 | tipizo | With shorewall off... I cannot go on the internet, therefore not route to the net... |
| 02:17.44 | ManxPower | so sorewall is not just being a firewall, it is also doing the NAT translations. |
| 02:17.53 | ManxPower | makes it hard to help you. |
| 02:17.59 | tipizo | that's correct |
| 02:18.21 | JT | it uses iptables yeah? |
| 02:18.29 | tipizo | yeap |
| 02:18.58 | JT | iptables -I FORWARD 1 -j ACCEPT |
| 02:19.49 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-179-72.lsanca.fios.verizon.net) |
| 02:21.17 | tipizo | what does it do? |
| 02:21.45 | ManxPower | We basically do not know if it's a firewall issue, NAT issue, routing issue, interface issue, or config issue. |
| 02:23.13 | tipizo | If it was a firewall issue I would I be able to browser sites on the Internet? |
| 02:26.55 | ManxPower | That boy doesn't have a clue. |
| 02:28.19 | Un1x | anyone ever Used... DIDWW did's before... because i need to find out how do i mapp it to my server, so i can receive calls :)? |
| 02:33.26 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 02:34.39 | ManxPower | generally you register with the remote service. |
| 02:39.03 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 02:53.33 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 03:08.08 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 03:09.14 | *** join/#asterisk n838901_2 (n=name@24-117-16-198.cpe.cableone.net) |
| 03:09.17 | n838901_2 | hello |
| 03:10.18 | StewLG | Anyone have a clue how the "voice mail waiting" light gets turned on with a Sipura 941 + Asterisk? |
| 03:10.41 | StewLG | I'm moving from TrixBox to a do-it-from-scratch Asterisk install, and I have no clue how Trixbox was getting it to work. |
| 03:20.12 | ManxPower | StewLG, in the [whatever] section of sip.conf (where whatever is the sip userid of the sipura, there should be a mailbox=vmailbox@vmcontext line |
| 03:20.43 | StewLG | Ah! |
| 03:20.55 | StewLG | Ok, I see it.. |
| 03:20.58 | n838901_2 | can someone with x-lite login to and use my asterisk system from across the internet? |
| 03:21.12 | StewLG | Rebooting, testing.. And yes, that seems to have done it. Thanks! |
| 03:21.32 | StewLG | n83.. is that a question or request? |
| 03:21.40 | n838901_2 | a question :) |
| 03:21.55 | StewLG | I think the answer is yes, you just have to set it up properly. |
| 03:22.21 | JT | i think he wants someone to try |
| 03:22.21 | StewLG | JT: He said it was a question. |
| 03:22.34 | JT | yes and the question was can someone in here do that for him |
| 03:22.37 | JT | still a question |
| 03:22.44 | n838901_2 | lol |
| 03:22.51 | n838901_2 | im just wondering if it can be done and how? |
| 03:22.53 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 03:23.11 | JT | n838901_2: you need to set it up as a sip host |
| 03:23.38 | n838901_2 | also..i am connected to another network via a ssl vpn, the sip client can login, but times out when making a call |
| 03:23.51 | StewLG | I think it would revolve around making the SIP phone a "friend", and giving it a context like your internal phone(s). |
| 03:23.52 | n838901_2 | all i use are sip hosts...voip only..no hardware |
| 03:24.04 | n838901_2 | ive done that too :) |
| 03:24.07 | ManxPower | PHONES are ususally type=friend. |
| 03:24.09 | russellb | no hardware? is asterisk running on toast? |
| 03:24.15 | n838901_2 | these are all x-lite clients |
| 03:24.18 | ManxPower | SERVERS are usually one type=user and one type=peer |
| 03:24.28 | n838901_2 | yeah..toast with jelly |
| 03:24.29 | JT | n838901_2: ssl vpn... not sure how good that is for real time stuff |
| 03:24.33 | StewLG | n83: the call timing out sounds like a NAT/SIP issue. Are you going through NAT? |
| 03:24.41 | ManxPower | russellb, It has better not be running on toast! I have a patent on that! |
| 03:25.05 | n838901_2 | i am through my firewall to the internet, but would that affect the traffic to/from the vpn tunnel? |
| 03:25.25 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 03:25.38 | StewLG | n83: I don't quite understand the question, but I'll ask again: is there a NAT gateway between you and the outide x-lite client? |
| 03:26.11 | n838901_2 | not to the tunnel..ips can be pinged directly |
| 03:26.46 | StewLG | Then I don't know what might be causing that. |
| 03:27.01 | ManxPower | n838901_2, it's pretty easy to mess up the firewall rules when using ipsec tunnels |
| 03:27.15 | n838901_2 | im using ssl, not ipsec |
| 03:27.21 | n838901_2 | amvpn |
| 03:27.23 | ManxPower | or any kind of tunnel, I suppoer. |
| 03:27.25 | ManxPower | suppose. |
| 03:27.34 | JT | n838901_2: does the tunnel run over TCP or UDP? |
| 03:28.08 | ManxPower | The TCP .vs. UDP *should* only affect call quality. |
| 03:28.26 | JT | maybe |
| 03:28.43 | n838901_2 | well, he can communicate with my server over the tunnel in every which way |
| 03:28.52 | n838901_2 | webserver, fileserver, etc... |
| 03:29.07 | JT | none of those things you mentioned are latency crucial |
| 03:29.28 | n838901_2 | ahh |
| 03:29.34 | n838901_2 | i see what you're saying |
| 03:29.47 | JT | but yeah, it may be another problem entirely |
| 03:29.50 | n838901_2 | is there any way to adjust the timeout value of the client? |
| 03:29.52 | JT | it's hard to tell |
| 03:30.07 | JT | a good test would be to try setup an IAX connection over the tunnel |
| 03:30.15 | JT | get the other guy to use an IAX client |
| 03:30.16 | JunK-Y | someone here using chan_h323? |
| 03:30.20 | n838901_2 | he says that x-lite is for some reason trying to send the data out through his router to the net rather than through the tunnel |
| 03:30.31 | JT | as it doesn't have the same difficulties with NAT that SIP does |
| 03:30.35 | n838901_2 | is there a free iax client? |
| 03:30.51 | JT | i don't know of any that cost money |
| 03:30.54 | JT | check voip-info.org |
| 03:30.57 | JT | it has a list |
| 03:31.02 | JT | on one of the iax pages |
| 03:31.40 | JT | the fact that it's trying to send data not over the tunnel is definately something to look into |
| 03:31.58 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:32.01 | quid2478 | hmm... is there any apps for * that will let me test network latency... like an echo test, except it spits back hard numbers? |
| 03:32.04 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 03:35.02 | jart | quid2478: uhm, ping? |
| 03:37.06 | Qwell | Why not just do an echo test? |
| 03:37.48 | quid2478 | Jart: Ping is pure ICMP... I want to quantify latency due to codecs etc |
| 03:37.55 | quid2478 | Qwell: Echo Test is Subjective |
| 03:38.04 | russellb | exten => blah,1,Echo |
| 03:38.10 | Qwell | subjective? |
| 03:38.19 | Qwell | You say a word, it repeats it back.. |
| 03:40.09 | Corydon76-home | repeats it back |
| 03:41.04 | Qwell | How many repeats before it just becomes feedback? |
| 03:41.10 | russellb | lol |
| 03:41.13 | russellb | that's out of hand |
| 03:41.16 | russellb | and what, ChanSpy on it? |
| 03:41.19 | Qwell | heh |
| 03:41.26 | Qwell | with 'w' |
| 03:41.29 | russellb | you'd have to inject some audio in there |
| 03:41.47 | russellb | ooh, idea |
| 03:41.57 | Corydon76-home | As long as each echo diminishes, there's no continual feedback |
| 03:42.06 | Qwell | hmm |
| 03:42.12 | ManxPower | If it doesn't then you get ECFO |
| 03:42.13 | jart | wouldn't you need some advanced telephony equipment for something like that? |
| 03:42.15 | russellb | *CLI> originate Local/MeetMeExten application Echo |
| 03:42.23 | russellb | do that a few times |
| 03:42.24 | ManxPower | ~ecfo |
| 03:42.29 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. |
| 03:42.29 | Qwell | a...few...times? |
| 03:42.35 | Qwell | That would be...interesting |
| 03:42.42 | russellb | Qwell: yeah, then call in and scream |
| 03:42.44 | Qwell | 2 == meh, 3 == ...wtf |
| 03:42.53 | russellb | Qwell: i'm doing it. |
| 03:42.56 | Qwell | haha |
| 03:42.59 | Corydon76-home | Oooo, a 3some? |
| 03:43.28 | Corydon76-home | An orgy of sound! |
| 03:43.37 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 03:43.37 | Qwell | I'll give you like 8 cool points, if you do an inband dtmf |
| 03:43.56 | quid2478 | Yeah... that is what I had in mind... if I could program.. I'd write something that sends out say a 100ms pulse of some kind to another * box running echo test, then have it measure the quality of the return signal |
| 03:43.59 | Qwell | and like...42, if you can let it sit in an echo loop for 5 minutes, and then make the dtmf DO something |
| 03:44.04 | Qwell | like, hit an external ivr |
| 03:44.12 | *** join/#asterisk LoneShadow (n=duh@59.92.205.26) |
| 03:44.29 | russellb | Qwell: oh god ... it was terrible |
| 03:44.32 | Qwell | heh |
| 03:44.38 | Qwell | how quick did it go to hell? |
| 03:44.57 | Qwell | and how many echos? |
| 03:45.00 | Corydon76-home | Faster than he can detect by ear alone |
| 03:45.11 | russellb | one is fine ... |
| 03:45.16 | russellb | two it goes to hell |
| 03:45.24 | russellb | you want to hear? :-p |
| 03:45.34 | Qwell | Did you mixmon it? :D |
| 03:45.42 | Corydon76-home | Conference call! |
| 03:45.58 | Corydon76-home | Can we use BJ's conference bridge? |
| 03:46.03 | Qwell | !!! |
| 03:46.07 | Qwell | You know what app_echo needs?! |
| 03:46.09 | Qwell | a delay |
| 03:46.15 | Qwell | a forced delay |
| 03:46.15 | russellb | heh, just call my digium exten, 6098 |
| 03:46.17 | LoneShadow | can someone tell me why I get calls as Unauthenticated with this config http://pastebin.ca/120027 |
| 03:46.57 | Qwell | russellb: what's the prefix again? |
| 03:46.59 | russellb | guest@misery.digium.com/6098 |
| 03:47.03 | russellb | IAX2/ |
| 03:47.03 | LoneShadow | box1 tries to make an IAX call to box2, and call falls into default context |
| 03:47.15 | Qwell | gonna pstn it.. I'm in the process of moving my desk ;/ |
| 03:47.23 | russellb | heh, alright |
| 03:47.30 | russellb | 877-linuxme |
| 03:47.33 | Qwell | ahh, right |
| 03:48.01 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:48.27 | Qwell | wow |
| 03:48.49 | Qwell | You could hear the third one kick in |
| 03:49.07 | russellb | woah, i put another one in and it stopped |
| 03:49.09 | russellb | it's just quiet now |
| 03:49.13 | russellb | i think it just said "wtf?" |
| 03:49.18 | Qwell | heh |
| 03:49.37 | quid2478 | is there a list of modules I can safely dump from Asterisk? I'm going to set up a box running pure IAX and SIP and want to make it lean and mean |
| 03:49.59 | russellb | quid2478: there is an asterisk slimming page on the wiki somewhere |
| 03:50.03 | *** part/#asterisk n838901_2 (n=name@24-117-16-198.cpe.cableone.net) |
| 03:50.07 | russellb | Qwell: heh, now we know what it would sound like, i guess |
| 03:50.07 | Qwell | I wonder if it would be the same if A) the join noise wasn't there, and/or B) you shouted over the echo'd join noise |
| 03:50.15 | quid2478 | thanks... I should have checked there firs |
| 03:50.17 | quid2478 | thanks... I should have checked there first |
| 03:50.23 | Qwell | russellb: That isn't what I expected at all... I kinda expected just...noise |
| 03:50.35 | russellb | oh, you're right, it's the join noise that makes it blow up |
| 03:50.43 | Qwell | but you could clearly hear the join noise repeat |
| 03:59.15 | *** join/#asterisk jterrero (i=eldude@200.42.248.92) |
| 04:03.33 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 04:11.59 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 04:16.56 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 04:17.06 | LoneShadow | on dynamic ip, if I use IAX2 or SIP register, does asterisk re-register everytime there is an IP change ? |
| 04:24.05 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 04:29.10 | *** join/#asterisk jefft (n=jefft@203.14.171.190) |
| 04:33.40 | *** join/#asterisk cpatry (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 04:35.30 | cpatry | any1 has an h323 termination, id like to make a quick test. |
| 04:36.00 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 04:40.04 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 04:48.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:51.57 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:53.41 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 04:56.29 | Un1x | anyone ever Used... DIDWW did's before... because i need to find out how do i mapp it to my server, so i can receive calls :)? |
| 04:57.23 | L|NUX | Un1x : http://www.didww.com/support/index.php?_m=knowledgebase&_a=view&parentcategoryid=1&pcid=0&nav=0 will help you |
| 05:04.35 | Un1x | hmm well sorta, the Extensions.conf is weird tho :S |
| 05:04.56 | Un1x | [didww-in] |
| 05:04.56 | Un1x | host=212.150.36.116 |
| 05:04.56 | Un1x | dtmfmode=inband |
| 05:04.56 | Un1x | dtmf=inband |
| 05:04.57 | Un1x | type=peer |
| 05:04.57 | Un1x | context=from-didww |
| 05:04.59 | Un1x | insecure=very |
| 05:05.01 | Un1x | nat=never |
| 05:05.03 | Un1x | allow=all |
| 05:05.05 | Un1x | extensions.conf: |
| 05:05.07 | Un1x | [from-didww] |
| 05:05.09 | Un1x | exten=> 972775654199,1,Dial(SIP/100) |
| 05:15.43 | L|NUX | Un1x : 972775654199 is your DID ? |
| 05:23.34 | *** join/#asterisk salaud (n=salaud@h-68-166-181-4.sttnwaho.covad.net) |
| 05:24.13 | salaud | Anyone know what the status of res_perl might be? I've never used it, and it looks like it is no longer in asterisk-addons |
| 05:24.42 | salaud | res_perl also doesn't seem to be in the main asterisk development either.. What's the deal? |
| 05:24.49 | ManxPower | LoneShadow, yes |
| 05:24.59 | ManxPower | LoneShadow, actually, no. |
| 05:25.19 | ManxPower | LoneShadow, if your IP changes you won't get incoming calls until the next registration. |
| 05:33.32 | dlynes_laptop | salaud: I don't think it ever was |
| 05:33.41 | dlynes_laptop | salaud: it's probably included with OpenPBX |
| 05:33.53 | dlynes_laptop | salaud: it's a fork of asterisk |
| 05:34.09 | salaud | dlynes_laptop: didn't know there were forks |
| 05:34.27 | salaud | Maybe res_perl has been ditched in favor of fastAGI? |
| 05:34.59 | dlynes_laptop | salaud: doubtful....res_perl is written by the same guys that forked asterisk to become OpenPBX, I think |
| 05:35.26 | salaud | dlynes_laptop: I see... |
| 05:35.27 | russellb | that's not quite true ... |
| 05:35.32 | dlynes_laptop | salaud: if you look there, you'll see there's also res_js |
| 05:35.45 | dlynes_laptop | russellb: what's not quite true? |
| 05:36.00 | salaud | has anyone else forked asterisk? |
| 05:36.03 | russellb | first, res_perl was pulled out of addons by the author |
| 05:36.16 | russellb | but he still provides it at www.pbxfreeware.org |
| 05:36.38 | russellb | and those guys are not the ones who forked asterisk to openpbx ... they actually started their own project |
| 05:36.58 | salaud | russellb: I got the sources... everything needs to me manually configured to make it... no autoconf... and it requires asterisk be built from source.. and I'm using debian |
| 05:37.30 | salaud | russellb: Because of that lack of configurability and non-inclusion in the add-ons... it feels dead |
| 05:37.38 | salaud | but that's just a feeling |
| 05:37.38 | ManxPower | Uh, no RELEASED version of Asterisk has autoconf |
| 05:38.02 | russellb | heh, yeah ... soon to change |
| 05:38.02 | dlynes_laptop | ManxPower: no trunk version, either |
| 05:38.06 | russellb | dlynes_laptop: liar! |
| 05:38.12 | salaud | I've never had to twiddle the Makefile directly to get asterisk to compile though |
| 05:38.25 | dlynes_laptop | russellb: no...there's a russellb configure script, but no gnu autoconf :) |
| 05:38.36 | russellb | yes, theres is gnu autoconf |
| 05:38.40 | dlynes_laptop | oh |
| 05:38.43 | dlynes_laptop | You changed? |
| 05:38.48 | russellb | no, it was always like that |
| 05:38.55 | ManxPower | I'll stick with the Digium release of Asterisk until such time as one of the ports has a compelling reason to switch. |
| 05:39.17 | ManxPower | nor ports, FORLS |
| 05:39.18 | dlynes_laptop | so ./configure --prefix=/usr works now? |
| 05:39.27 | salaud | If it is packaged and has what I need for 80% of cases that's when I would switch |
| 05:39.30 | russellb | dlynes_laptop: yes |
| 05:39.33 | dlynes_laptop | ah |
| 05:39.34 | dlynes_laptop | ok |
| 05:39.43 | russellb | and that was still autoconf, just not set up the usual way |
| 05:40.09 | dlynes_laptop | ah...ok....figured it was autoconf because of the way it worked |
| 05:40.10 | salaud | res_perl fails to compile without a bunch of monkeying and that says to me, not widely used |
| 05:40.11 | russellb | i didn't write that 30k line script :) |
| 05:40.19 | dlynes_laptop | and running autoconf/automake screwed everything up |
| 05:40.31 | Juggie | russellb, i found out the makefile problem in zaptel i was talking about. |
| 05:40.40 | L|NUX | cparty : there |
| 05:41.01 | Juggie | seems, if you have the smp kernel running, but only the non smp kernel source. |
| 05:41.16 | Juggie | eg, you have kernel & kernel-smp installed but only kernel-devel (and not kernel-smp-devel) |
| 05:41.31 | dlynes_laptop | Juggie: isn't that waht I told you the problem was? |
| 05:41.44 | Juggie | the problem was it needed kernel-smp-devel |
| 05:41.45 | russellb | heh, that's bad, mkay? |
| 05:41.51 | dlynes_laptop | yeah...that's what I told yhou |
| 05:41.53 | Juggie | once i installed that zaptel built fine. |
| 05:42.16 | Juggie | yeah, but zaptel probally shoudnt build when kernel-smp-devel is missing |
| 05:42.20 | dlynes_laptop | you had a symbolic link the non-smp source inside the smp kernel directory |
| 05:42.25 | JT | asterisk is normally fine with smp yeah? |
| 05:42.26 | dlynes_laptop | and that's why it was building |
| 05:42.32 | Juggie | dlynes_laptop, nah i had him remove that. |
| 05:42.34 | dlynes_laptop | that symlink shouldn't have been there |
| 05:42.38 | dlynes_laptop | ah |
| 05:42.39 | JT | or is uniprocessor more reliable? |
| 05:42.39 | Juggie | he gave me root to the box |
| 05:42.55 | Juggie | zaptel uses the non-smp source in /usr/src/kernels for some reason |
| 05:42.57 | Juggie | when it shoudnt. |
| 05:43.01 | dlynes_laptop | JT: doesn't matter...just don't be doing fake smp (aka HT/hyperthreading) |
| 05:43.07 | Juggie | it should just bitch and say source not found. |
| 05:43.19 | JT | dlynes_laptop: why is that? |
| 05:43.27 | dlynes_laptop | Juggie: it's hard-coded to look for /usr/src/linux-2.6 and /usr/src/linux |
| 05:43.34 | dlynes_laptop | Juggie: lord only knows why |
| 05:43.41 | Juggie | yeah i know, i removed those symbolic links he had. |
| 05:43.45 | Juggie | speaking of which |
| 05:43.48 | Juggie | i'm slightly confused |
| 05:43.53 | Juggie | why is there kernel-smp-devel & kernel-devel |
| 05:44.07 | Juggie | isnt that the whole point of /lib/modules/<kernel ver>/build |
| 05:44.08 | dlynes_laptop | Juggie: cause you're on redhat |
| 05:44.32 | Juggie | i see... but thats the point of the build dir right? |
| 05:44.39 | dlynes_laptop | Juggie: correct |
| 05:44.44 | Juggie | thats what i thought. |
| 05:45.00 | dlynes_laptop | JT: I haven't seen one single person on here using HT that's gotten it to work |
| 05:45.11 | Juggie | dlynes_laptop, it works fine for me. |
| 05:45.13 | JT | hmm, ok |
| 05:45.21 | dlynes_laptop | ok |
| 05:45.27 | dlynes_laptop | one person's gotten it to work :0 |
| 05:45.41 | Juggie | i have dual xeon boxes EM64T procs |
| 05:45.43 | Juggie | with HT |
| 05:45.51 | Juggie | so the kernel sees 4 processors |
| 05:45.59 | dlynes_laptop | Juggie: do you have a uni xeon with HT, though? |
| 05:46.17 | Juggie | no, all my boxes are dual. i have some pentium4's with HT that work fine though |
| 05:46.18 | dlynes_laptop | Juggie: or uni anything with HT, for that matter? |
| 05:46.21 | Juggie | just not xeon's w/ HT |
| 05:46.32 | Juggie | yes, i have p4's that are HT and work fine. |
| 05:46.33 | JT | i assume you mean it's a problem for Asterisk, dy |
| 05:46.39 | JT | dlynes_laptop |
| 05:46.45 | dlynes_laptop | Juggie: does the piv with HT have HT enabled, though? |
| 05:46.51 | Juggie | yes |
| 05:46.54 | Juggie | linux sees two procs |
| 05:46.58 | dlynes_laptop | ah |
| 05:46.58 | JT | i have uniprocessor JT boxes that works fine for Linux |
| 05:47.09 | dlynes_laptop | heh |
| 05:47.11 | dlynes_laptop | JT boxes :) |
| 05:47.12 | dlynes_laptop | cute |
| 05:47.20 | JT | ?? |
| 05:47.20 | Juggie | dlynes_laptop, i dont see why you would think it would be a problem |
| 05:47.26 | Juggie | its a hardware level thing |
| 05:47.27 | JT | boxes is cute? |
| 05:47.32 | dlynes_laptop | Juggie: i don't |
| 05:47.41 | Juggie | i thought you said it never worked :) |
| 05:47.50 | dlynes_laptop | Juggie: just tons of people have come on here not getting HT processors to work |
| 05:48.00 | dlynes_laptop | Juggie: they disable HT, and magically everything works just fine |
| 05:48.11 | Juggie | well it just fake SMP |
| 05:48.16 | Juggie | so i dont see how it would cause a problem :) |
| 05:48.38 | Juggie | there may be situations where its not perfect for performance |
| 05:48.43 | Juggie | but it shoudnt cause anything to not work |
| 05:52.13 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:53.36 | dlynes_laptop | JT: thought you were doing a play on HT boxes |
| 05:54.23 | JT | ok |
| 05:59.49 | JT | anyone had any experience setting up asterisk as a least cost router? |
| 06:00.12 | ionix | hrm |
| 06:00.13 | ionix | [app_curl.so]Aug 7 01:57:26 WARNING[21413]: loader.c:325 __load_resource: libcurl.so.3: cannot open shared object file: No such file or directory |
| 06:00.13 | ionix | Aug 7 01:57:26 WARNING[21413]: loader.c:554 load_modules: Loading module app_curl.so failed! |
| 06:00.22 | ionix | yet libcurl.so.3 is in /usr/local/lib/ |
| 06:01.05 | JT | it's actually there, or a symlink to somewhere? |
| 06:02.44 | ionix | lrwxrwxrwx 1 root root 16 Feb 13 2005 /usr/local/lib/libcurl.so.3 -> libcurl.so.3.0.0 |
| 06:02.45 | ionix | -rwxr-xr-x 1 root root 618572 Feb 13 2005 /usr/local/lib/libcurl.so.3.0.0 |
| 06:02.54 | ionix | symlinks to an existing file |
| 06:05.52 | dlynes_laptop | ionix: is /usr/local/lib in your ld.so.conf file? |
| 06:06.14 | ionix | hmm I assume so, let me double check |
| 06:06.48 | ionix | hmm it wasn't |
| 06:06.53 | ionix | damn, that's fked up |
| 06:07.01 | dlynes_laptop | ionix: why is that screwed up? |
| 06:07.07 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 06:07.23 | ionix | I had it added like 6 months ago |
| 06:07.33 | dlynes_laptop | ionix: it's not in there by default on a lot of distros |
| 06:07.47 | dlynes_laptop | ionix: did you upgrade your distribution in the last six months? |
| 06:07.53 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-35.range217-44.btcentralplus.com) |
| 06:07.53 | dlynes_laptop | ionix: or your glibc? |
| 06:08.02 | ionix | working now. |
| 06:08.08 | dlynes_laptop | ionix: or your gnu development environment? |
| 06:08.11 | ionix | ahhh glibc |
| 06:08.16 | ionix | that must be it, |
| 06:08.23 | ionix | wow thank you for pointing that out :) |
| 06:08.58 | dlynes_laptop | yeah...that probably replaced your ld.so.conf file then |
| 06:09.05 | dlynes_laptop | and then ran an ldconfig -a |
| 06:09.15 | Dovid | what is the diffrence between qulaify=yes and qualify=2000 ? |
| 06:09.53 | dlynes_laptop | Dovid: qulaify=yes doesn't exist |
| 06:10.03 | dlynes_laptop | Dovid: otoh, qualify=yes and qualify=2000 are the same thing |
| 06:10.12 | Dovid | ok |
| 06:10.18 | Dovid | meant qualify |
| 06:10.37 | Dovid | i am having a problem where I have t phones behind the same NAT connecting to asterisk |
| 06:11.02 | Dovid | as soon as one is in use asterisk says that the second onces qualify is 0 and now unreachable |
| 06:11.17 | Dovid | its killin me |
| 06:11.19 | Dovid | i tried everything |
| 06:11.30 | Juggie | they should just use different sip ports |
| 06:11.47 | Dovid | hmm |
| 06:11.51 | Dovid | that would fix the problem ? |
| 06:12.03 | Dovid | ans there wont be an issues with RTP |
| 06:12.10 | Juggie | rtp wont be an issue |
| 06:12.16 | Juggie | and not the DEST sip port either |
| 06:12.22 | Dovid | cause no matter what they can make outgoing calls. the problem is the incoming calls |
| 06:12.32 | Juggie | i suspect its the nat device. |
| 06:12.39 | Juggie | it should be smart enough to handle this. |
| 06:12.43 | Dovid | that its doing poor job ? |
| 06:12.50 | Juggie | what do you have doing nat? |
| 06:13.00 | Dovid | he has snom's so what port does he switch ? |
| 06:13.03 | Juggie | in * do both phones show up as having the same sip port. |
| 06:13.08 | Dovid | not sure i think he has linksys |
| 06:13.15 | ionix | damn, even after upgrading asterisk, I have the same problem, it cannot find the AGI script file |
| 06:13.20 | NDT | noht in grub.conf looks as if it turns ht off...but if it is on in the bios does it still cause issues? |
| 06:13.24 | ionix | Aug 7 02:12:01 WARNING[22220]: pbx.c:5152 ast_pbx_run_app: No such application 'call.php' |
| 06:13.42 | dlynes_laptop | ionix: do you have bin/php in your path? |
| 06:13.58 | dlynes_laptop | ionix: do you have call.php chmod 755 or something similar? |
| 06:14.03 | ionix | -rwxr-xr-x 1 root root 1208 Aug 7 02:11 /var/lib/asterisk/agi-bin/call.php |
| 06:14.07 | Dovid | Juggie: what ports do i have to change on the phone and in asterisk ? |
| 06:14.18 | ionix | yes and it's first line calls php. I can even run the script from cmd line |
| 06:14.19 | Juggie | Dovid, type 'sip show peers' |
| 06:14.27 | Juggie | do all the phones behind nat have the same sip port. |
| 06:14.35 | dlynes_laptop | ionix: are you calling it using DeadAGI(call.php)? |
| 06:14.37 | Dovid | now nothing comes up |
| 06:14.43 | ionix | no, in a outgoing file |
| 06:14.47 | Dovid | i was messing around. it looks like its working now though |
| 06:14.51 | dlynes_laptop | ionix: oh...no idea about that |
| 06:14.54 | ionix | Application: call.php |
| 06:15.07 | Juggie | Dovid, if all that persons peers are using the same sip por |
| 06:15.11 | Juggie | *sip port |
| 06:15.14 | Juggie | then thats the problem. |
| 06:15.23 | dlynes_laptop | ionix: check asterisk.conf, to make sure it's telling asterisk that your agi-bin directory is actually where you think it is |
| 06:16.13 | dlynes_laptop | ionix: application might need it to be in your path, or have the absolute path specified, too |
| 06:16.24 | dlynes_laptop | ionix: it might assume current directory is /etc/asterisk |
| 06:16.32 | dlynes_laptop | ionix: or something similar |
| 06:16.35 | ionix | hmm ok let me see |
| 06:16.35 | Dovid | so what do i have to switch ? |
| 06:16.45 | Dovid | Juggie: they are both using 5060 for SIP port |
| 06:16.47 | Juggie | Dovid, you still havnt answered my question |
| 06:16.54 | Juggie | Dovid, you cant really switch anything |
| 06:16.55 | Dovid | sip show peers ? |
| 06:17.03 | Dovid | comes up blank |
| 06:17.04 | ionix | astagidir => /var/lib/asterisk/agi-bin |
| 06:17.07 | ionix | correct dir |
| 06:17.21 | Dovid | i have |
| 06:17.21 | Dovid | nat=yes |
| 06:17.26 | Juggie | Dovid, stop talking |
| 06:17.27 | dlynes_laptop | ionix: yeah, so the application keyword doesn't look in /var/lib/asterisk/agi-bin as I suspected then |
| 06:17.31 | Juggie | and listen to my questions |
| 06:17.34 | Dovid | ok |
| 06:17.34 | Dovid | shoot |
| 06:17.44 | dlynes_laptop | ionix: the application keyword checks the path |
| 06:17.49 | Juggie | ok. |
| 06:17.49 | dlynes_laptop | ionix: and absolute paths |
| 06:17.55 | Juggie | you need to get both those peers in sip show peers. |
| 06:18.06 | Juggie | so you can see the source port they are using. |
| 06:18.12 | Dovid | ok. they were before now. they arent. |
| 06:18.21 | Juggie | ok, well, you must get them back ;) |
| 06:18.33 | Juggie | only then can you confirm if they are indeed using the same source port. |
| 06:18.40 | ionix | I tried with absolute path, same error |
| 06:18.46 | Dovid | ok. let me restart asterisk |
| 06:18.49 | Juggie | what you see in your phone setup is not source, its the dest port on the sip server |
| 06:18.53 | Juggie | thats different. |
| 06:19.26 | ionix | ug 7 02:19:13 WARNING[22621]: pbx.c:5152 ast_pbx_run_app: No such application '/var/lib/asterisk/agi-bin/call.php' |
| 06:19.36 | Dovid | Juggie: pb is out of order can i PM u ? |
| 06:19.59 | Juggie | use pastebin.ca |
| 06:20.09 | Dovid | also |
| 06:20.23 | Dovid | not workin |
| 06:20.30 | Juggie | ~pb |
| 06:20.32 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 06:20.40 | ionix | ahh no shit |
| 06:20.48 | ionix | Application should be AGI |
| 06:20.49 | ionix | hehe |
| 06:20.57 | Juggie | pastebin.com is up for me. |
| 06:21.18 | Dovid | yes juggie. i tried. .com loads but once u hit submit it freezes |
| 06:21.53 | Dovid | Argh |
| 06:21.53 | Dovid | 10315/10315 72.144.247.22 D N 50706 UNREACHABLE |
| 06:21.53 | Dovid | 10305/10305 72.144.247.22 D N 50625 UNREACHABLE |
| 06:21.56 | ionix | works now |
| 06:22.07 | Dovid | it must not like me |
| 06:22.14 | Dovid | Juggie thats what I got |
| 06:22.42 | dlynes_laptop | heh |
| 06:23.05 | Juggie | ok, they are using dif source ports. |
| 06:23.09 | Juggie | thats good ;) |
| 06:23.20 | Juggie | Dovid, my friend has a linksys and he has constant problems as well |
| 06:23.22 | Dovid | and juggie I keep getting this as soon as i hang up with the phone |
| 06:23.36 | Juggie | i think its just a shitty nat device. |
| 06:23.44 | Dovid | : chan_sip.c:11564 sip_poke_noanswer: Peer '10305' is now UNREACHABLE! Last qualify: 0 |
| 06:23.44 | Dovid | -- SIP Seeding peer from astdb: '10306' at 10306@69.114.216.204:60060 for 3600 |
| 06:23.51 | Dovid | but the phone is there. |
| 06:23.53 | Juggie | if it keeps closing the nat connection you can do. |
| 06:23.57 | Dovid | so linksys = shit |
| 06:24.01 | Juggie | there, theres not much you can do. |
| 06:24.15 | Juggie | do you have qualify=yes ? |
| 06:24.17 | Juggie | in your sip.conf |
| 06:24.20 | *** join/#asterisk tempest1 (n=asf@adsl-150-206-4.chs.bellsouth.net) |
| 06:24.22 | Dovid | i have qualify=2000 |
| 06:24.28 | Dovid | i amusing real time |
| 06:25.12 | Dovid | should i also have it in general.conf ? |
| 06:25.31 | Un1x | is there a way i can test my did..... |
| 06:25.39 | Juggie | you can also try setting your registration time out to be super low |
| 06:25.43 | Un1x | it's a international number in china was wanting to test it see, if it works or not ;P? |
| 06:25.46 | Juggie | i set mine to 5minutes |
| 06:25.51 | Dovid | where is that set ? |
| 06:25.55 | Juggie | but that perticular nat device still has problem. |
| 06:25.55 | Un1x | i set mines to 2 min :P |
| 06:25.59 | Juggie | Dovid, in sip.conf |
| 06:26.09 | Dovid | what setting in sip.conf ? |
| 06:26.17 | Un1x | lmao |
| 06:26.33 | Juggie | Dovid, its documented |
| 06:26.34 | Juggie | look for it |
| 06:26.34 | Dovid | im tired, sorry Un1x |
| 06:26.55 | dlynes_laptop | Un1x: call it on your landline? |
| 06:27.30 | Dovid | ok this means ? |
| 06:27.30 | Dovid | Last qualify: 0 |
| 06:27.34 | Dovid | ? |
| 06:27.42 | Dovid | that it cant find the phone ? |
| 06:27.46 | Juggie | right |
| 06:27.51 | Juggie | it cant give a ping number |
| 06:27.55 | Juggie | because it got no response |
| 06:28.08 | Dovid | ok. and if i turn qualify off what will happen ? it deff wont find it ? |
| 06:28.10 | Un1x | dlynes_laptop: i dont wanna pay 1$ a minute to try and call :) |
| 06:28.11 | Dovid | ok |
| 06:28.21 | Un1x | i'll just wait till my freind comes online i'll ask him to test ;D |
| 06:28.37 | Juggie | Dovid, you should also see things like |
| 06:28.38 | Juggie | Aug 7 02:22:05 NOTICE[19931]: chan_sip.c:9864 handle_response_peerpoke: Peer '08000f161fb5' is now TOO LAGGED! (936ms / 300ms) |
| 06:28.38 | Juggie | Aug 7 02:24:56 NOTICE[19931]: chan_sip.c:9858 handle_response_peerpoke: Peer '08000f161fb5' is now REACHABLE! (153ms / 300ms) |
| 06:28.47 | Juggie | if theres a high ping for any reason, etc. |
| 06:28.51 | Juggie | thats when it would show a number. |
| 06:29.19 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 06:29.19 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 06:29.34 | Dovid | Juggie: i am not seeing the too laged just that the last qualify is 0 |
| 06:29.50 | dlynes_laptop | Un1x: $1/mi? |
| 06:29.55 | Juggie | the nat is essentially having problems |
| 06:29.58 | dlynes_laptop | Un1x: you've got a crappy LD provider, then |
| 06:29.59 | Juggie | does it work fine w/ just one phone? |
| 06:30.06 | Dovid | i believe so |
| 06:30.15 | dlynes_laptop | Un1x: I get China for less than 1c/mi |
| 06:30.15 | Juggie | try a different nat device if you can |
| 06:30.18 | Dovid | he had the same problem in a diff office |
| 06:30.21 | Juggie | or try turning off one phone. |
| 06:30.27 | Un1x | hmm dlynes it's bell dude... |
| 06:30.29 | Juggie | and see if the problem clears up w/ just one phone. |
| 06:30.30 | Dovid | my friend set the RTP stream to go to one phone and it worke |
| 06:30.34 | Un1x | they will make up some fucking gay fees |
| 06:30.39 | dlynes_laptop | Un1x: never heard of a calling card? |
| 06:30.42 | Juggie | Dovid, like i said, try one phone. |
| 06:30.47 | Un1x | dont have one atm to try :( |
| 06:30.52 | dlynes_laptop | Un1x: try CiCi....they're super cheap |
| 06:30.55 | Un1x | i gotta wait till tomorow for a calling card :) |
| 06:30.59 | dlynes_laptop | Un1x: especially for china |
| 06:31.04 | Un1x | dlynes_laptop you call me wanna try :p? |
| 06:31.05 | Dovid | Juggie: what devices have good NAT ? |
| 06:31.06 | Juggie | unix, theres also the 10-10-xxx dial around numbers |
| 06:31.11 | Juggie | do you have those in the usa? |
| 06:31.15 | Juggie | i assume so |
| 06:31.18 | Un1x | im not in states :p |
| 06:31.21 | dlynes_laptop | Un1x: nah |
| 06:31.21 | Dovid | yes juggie: we do |
| 06:31.22 | Juggie | we have them here in canuck land. |
| 06:31.29 | Un1x | im canuck as well :P |
| 06:31.30 | dlynes_laptop | Un1x: i know...you're in Canada |
| 06:31.38 | Un1x | juggie doesn't.. |
| 06:31.42 | Juggie | un1x, use 10-10-925 |
| 06:31.46 | Juggie | see rates @ www.yak.ca |
| 06:31.51 | Un1x | thanks :) |
| 06:31.58 | *** join/#asterisk sekuru (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 06:31.59 | dlynes_laptop | Un1x: i suggested CiCi because i knew you were in Canada |
| 06:32.06 | Juggie | ah, i find yak good |
| 06:32.09 | Juggie | and it shows up on my bill. |
| 06:32.11 | dlynes_laptop | Un1x: it's a Canadian calling card company |
| 06:32.19 | Dovid | Juggie: what do u suggest for a good NAT device ? |
| 06:32.55 | dlynes_laptop | Juggie: yeah...we're gearing ourselves up for 1010 and 310 numbers |
| 06:33.10 | Juggie | dlynes_laptop, what do you mean? |
| 06:33.12 | dlynes_laptop | Juggie: we got a deal with Telus to start offering them soon |
| 06:33.26 | Juggie | Dovid, i'm not sure, but you need to do testing first |
| 06:33.31 | Juggie | to ensure thats the problem. |
| 06:33.43 | Dovid | yea. he is going live tomorrow so he really cant :( |
| 06:33.48 | Dovid | i am gona get shot |
| 06:33.59 | Juggie | eg, try w/ just one sip phone enabled |
| 06:34.03 | Dovid | oh well. thanks for the help/ i need me a smoke |
| 06:34.06 | Dovid | i will |
| 06:34.09 | Juggie | well, i'm only guessing but on default settings things should just work. |
| 06:34.11 | dlynes_laptop | Dovid: try a linux box with an iptables firewall |
| 06:34.20 | Dovid | that he wont do. cheap fuck |
| 06:34.22 | Juggie | i mean phone wise. |
| 06:34.31 | Dovid | i think i am goan try to mod a wifi router |
| 06:34.33 | dlynes_laptop | Dovid: I've been deploying shorewall firewalls on my asterisk keysystems |
| 06:34.49 | Juggie | if it works w/ one phone and doesnt work w/ two |
| 06:34.53 | Juggie | and they are idential configs |
| 06:34.57 | Juggie | then its almost surely a nat problem. |
| 06:35.02 | Juggie | * is seeing dif source ports |
| 06:35.03 | Juggie | which is good |
| 06:35.05 | dlynes_laptop | Dovid: wait a second |
| 06:35.09 | Dovid | kk |
| 06:35.19 | dlynes_laptop | Dovid: he's a cheap fuck, and he's buying cisco? |
| 06:35.24 | dlynes_laptop | Dovid: that does not compute |
| 06:35.26 | Dovid | not cisco |
| 06:35.27 | Dovid | snom |
| 06:35.32 | dlynes_laptop | even then |
| 06:35.35 | dlynes_laptop | snom is expensive, too |
| 06:35.47 | dlynes_laptop | and ugly |
| 06:35.49 | Juggie | mitel are the best sip phones in my opinion |
| 06:35.55 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 06:36.00 | dlynes_laptop | mitels work great with asterisk? |
| 06:36.07 | Juggie | Un1x, one of the best FG episodes is on teletoon :) |
| 06:36.10 | Juggie | just fyi :P |
| 06:36.18 | Juggie | dlynes_laptop, yep, their sip phones are fantastic |
| 06:36.26 | benjk | hi dlynes |
| 06:36.28 | Juggie | 100% solid business phone. |
| 06:36.34 | dlynes_laptop | Juggie: nortel stm phones are fantastic, too |
| 06:36.53 | Dovid | dylnes_laptop: does shorewall have something for routers. like u can take a linksys router wipe it and replace it. i saw places that had better firmware that run on linux |
| 06:36.56 | dlynes_laptop | Once I get some spare time I've got a nortel softphone that uses stm |
| 06:37.01 | Dovid | i think i am gona play with that tomorrow |
| 06:37.15 | Juggie | Dovid, just buy a new router from best buy or something |
| 06:37.16 | Juggie | and try it |
| 06:37.22 | dlynes_laptop | i'm gonna try playing with the unistm channel driver when i get a chance |
| 06:37.29 | Dovid | Juggie: what kind ? |
| 06:37.35 | benjk | what's stm? |
| 06:37.37 | Dovid | or maybe just go out and get 10 phones |
| 06:37.37 | Juggie | Dovid, something besides linksys. |
| 06:37.41 | Dovid | lol |
| 06:37.50 | dlynes_laptop | Dovid: something besides linksys or dlink |
| 06:37.55 | Dovid | he had a netgear b4hanad and that made a lot more problems |
| 06:37.57 | dlynes_laptop | dlink sucks just as bad |
| 06:38.00 | Dovid | i think i am gona try an SMC |
| 06:38.06 | dlynes_laptop | cheap netgear sucks, too |
| 06:38.14 | Juggie | Dovid, a friend of mine has a linksys and his connection drops all the time |
| 06:38.16 | dlynes_laptop | but the more expensive netgear is apparently fine |
| 06:38.18 | Juggie | eg allways goes unreachable. |
| 06:38.22 | Juggie | meanwhile, mine is perfect |
| 06:38.27 | Juggie | and we have the same phones/same * config. |
| 06:38.28 | Dovid | hmm |
| 06:38.34 | Dovid | so i goto have him try an SMC |
| 06:38.38 | Dovid | or something else |
| 06:38.41 | Dovid | or a custom router |
| 06:38.41 | Juggie | well as i said |
| 06:38.44 | Juggie | try w/ one phone first. |
| 06:38.45 | dlynes_laptop | Juggie: I only have that problem with a cisco firewall that's running a vpn |
| 06:38.50 | Dovid | ok will do that |
| 06:38.55 | Dovid | after all this i need a smoke. |
| 06:38.56 | Dovid | bbs |
| 06:39.05 | dlynes_laptop | almost all of my customers are running linksys wrt54g's |
| 06:39.18 | dlynes_laptop | i've only run into a problem once in a blue moon |
| 06:39.20 | Juggie | dlynes_laptop, it depends on the model of the router i guess. |
| 06:39.25 | Juggie | some of them are good |
| 06:39.27 | Juggie | some of them suck |
| 06:39.37 | Juggie | i'm not sure what my friend has |
| 06:39.39 | Juggie | but it sucks horribly |
| 06:39.44 | Juggie | and drops the nat connection all the time |
| 06:39.57 | Juggie | he can take it out and put a usr router in its place |
| 06:40.01 | Juggie | and it works perfect. |
| 06:40.15 | dlynes_laptop | I've found the best solution is to run a dhcp server on the asterisk box, disable dhcp on the linksys router, and only use the linksys device as a router |
| 06:40.22 | dlynes_laptop | linksys' dhcp server is flaky as hell |
| 06:40.41 | Juggie | i have a USR8054 |
| 06:40.53 | dlynes_laptop | US Robotics? |
| 06:40.53 | Juggie | and i found a version of the firmware that works great. |
| 06:40.56 | Juggie | yep. |
| 06:41.00 | Juggie | its 100% solid |
| 06:41.05 | Juggie | with one fw version ;) |
| 06:41.06 | dlynes_laptop | I thought they got bought out by 3COM? |
| 06:41.11 | Juggie | every other version has some problem |
| 06:42.05 | Juggie | its a USR8054 w/ 1.67fw |
| 06:42.07 | Juggie | if anyone has one |
| 06:42.10 | Juggie | thats the best version. |
| 06:42.13 | benjk | hey guys, I am looking for some feedback on a new option for the Dial command |
| 06:42.19 | Juggie | shoot. |
| 06:42.33 | benjk | http://pastebin.ca/120240 |
| 06:43.08 | Juggie | i can see how its useful |
| 06:43.09 | Juggie | but |
| 06:43.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 06:43.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 06:43.13 | benjk | not sure if "i" is the most straightforward letter to pick |
| 06:43.14 | Juggie | it can be done via a macro. |
| 06:43.33 | dlynes_laptop | benjk: cool |
| 06:43.40 | dlynes_laptop | benjk: but yeah, what juggie said |
| 06:43.45 | benjk | yeah, but there are some issues with doing it with macros |
| 06:43.51 | Juggie | like? |
| 06:43.51 | dlynes_laptop | benjk: but, i think your method is better |
| 06:43.56 | dlynes_laptop | benjk: it's straight forward |
| 06:44.00 | Juggie | maybe we should fix macros then :) |
| 06:44.37 | Juggie | i've done it before to tell someone they had a click to talk call and did they want to accept etc. |
| 06:44.49 | Juggie | what doesnt work doing it in a macro? |
| 06:45.07 | benjk | I have done this with a macro before and it got out of hand, this one is cleaner and more robust |
| 06:45.19 | Juggie | "you bring me the wall street journal, you two fight to the death" |
| 06:45.20 | Juggie | hehe. |
| 06:45.28 | *** join/#asterisk tempest1 (n=asf@adsl-153-35-130.chs.bellsouth.net) |
| 06:45.29 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:45.50 | Juggie | benjk, you still arnt telling me why its insufficient in a macro. |
| 06:45.59 | Juggie | what about it doesnt work. |
| 06:46.37 | Juggie | i'm not insulting your work only asking because it wont get in unless you can justify why this is required and cant already be done. |
| 06:46.55 | benjk | I am not interested in discussing the reasons why I did what I did, I am interested in usability feedback as I am to add more options for the called party |
| 06:47.52 | Juggie | then there is nothing more that i can add. |
| 06:48.11 | benjk | one of the things that got out of hand was language settings |
| 06:48.29 | benjk | because the called party doesn't typically get the same language as the calling party over here |
| 06:48.37 | salaud | Anyone know the best way to dynamically add extensions and logic? |
| 06:48.44 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:48.46 | salaud | Manger Interface? |
| 06:48.49 | salaud | AGI? |
| 06:49.20 | Juggie | realtime extensions. |
| 06:49.35 | benjk | anyway, is there any impediment to using the letter "i" for this? |
| 06:49.41 | salaud | Juggie: Realtime is 1.2? |
| 06:49.51 | Juggie | yes. |
| 06:49.53 | Juggie | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 06:50.06 | salaud | Juggie: Can it be done with the manager interface? |
| 06:50.46 | salaud | Juggie: I want to push new extensions into asterisk, but maybe via FastAGI |
| 06:50.47 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 06:50.47 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:51.29 | Juggie | salaud, no. |
| 06:51.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.40) |
| 06:51.48 | salaud | Juggie: I really don't want to re-load any config files while things are moving... But, I do want to save the current configuration |
| 06:52.16 | salaud | Hmmm... so there was never any way to add more config logic to asterisk before Realtime? |
| 06:52.24 | benjk | there was |
| 06:52.29 | benjk | ands still is |
| 06:52.36 | salaud | benjk: and that is? |
| 06:52.40 | benjk | macros |
| 06:53.04 | Juggie | salaud, perhaps you arnt explaining what you want to do |
| 06:53.08 | salaud | benjk: How would that be used to create a new extension? |
| 06:53.12 | benjk | which retrieve parameters from either astdb or any other database of your choice |
| 06:53.38 | Juggie | if you wanted to add ivr menues on the fly into extensions you would need realtime extensions. |
| 06:53.45 | Juggie | however, if its just phones you are talking about |
| 06:53.49 | benjk | not necessarily |
| 06:54.06 | Juggie | benjk, without reloading, or depending on an external db. |
| 06:54.16 | benjk | you could write a macro that processes voicexml |
| 06:54.26 | benjk | you can use astdb |
| 06:54.32 | Juggie | i suspect what he is asking is to add phones |
| 06:54.42 | Juggie | which doesnt require extensions |
| 06:54.43 | benjk | I am not saying one or another approach is superior |
| 06:54.46 | salaud | Juggie: nope... I don't need to add phones |
| 06:54.52 | benjk | just saying that it was always possible to do |
| 06:54.54 | Juggie | explain your application. |
| 06:54.58 | salaud | I really need to add extensions |
| 06:55.00 | Juggie | what are you trying to do. |
| 06:55.06 | Juggie | what is your extension. |
| 06:55.23 | salaud | so basically I'm getting several RSS feeds which are channels |
| 06:55.34 | salaud | I want to dynamically create an extension for each channel |
| 06:55.57 | Juggie | can you pastebin an example of what you are doing now |
| 06:56.01 | salaud | so that I can do an IVR that drops them into a menu for each channel which ultimately plays some content |
| 06:56.02 | benjk | [macro-DialRSSFeed] |
| 06:56.10 | Juggie | what are you doing reading out the rss? |
| 06:56.30 | salaud | I'm only in the evaluation stage now.. I have no code yet |
| 06:56.42 | benjk | well, that's how I would do it |
| 06:56.44 | salaud | I am going to get URLs out of the RSS feeds |
| 06:56.56 | benjk | write a macro that dials RSS feeds |
| 06:57.04 | salaud | the URLs will either be file://someaudio |
| 06:57.16 | salaud | or http://someaudio |
| 06:57.25 | benjk | and if the macro gets out of hand, turn it into an app |
| 06:57.56 | salaud | I will create an IVR that kind of introduces the channel then let's the users iterate through the various audio files |
| 06:58.08 | Juggie | salaud, if i were you i'd take a bunch of extensions and route them to an agi. |
| 06:58.14 | Juggie | it would probally make it much simpler. |
| 06:58.27 | salaud | Juggie: trouble is the number of extensions will change |
| 06:58.34 | benjk | so? |
| 06:58.46 | Juggie | right, but your agi can still reject an extension that isnt valid. |
| 06:59.02 | benjk | you can add a family to astdb |
| 06:59.07 | salaud | but the agi runs FROM an extension |
| 06:59.11 | benjk | say rss-feeds |
| 06:59.20 | benjk | then add each feed to that |
| 06:59.22 | salaud | I didn't know agi could receive an extension |
| 06:59.29 | Juggie | yes, of course it can |
| 06:59.32 | benjk | and your AGI or macro goes through the list |
| 06:59.46 | benjk | this way you can dynamically add and remove feeds |
| 06:59.51 | benjk | without reload |
| 06:59.52 | salaud | this is sounding like what I want to do |
| 06:59.59 | Juggie | i woudnt use astdb though |
| 07:00.11 | Juggie | hard to manage cant really use a web interface for it, etc. |
| 07:00.27 | salaud | I'm missing the link between extensions and the macro |
| 07:00.27 | benjk | database add rss-feeds foobar "http://foo.bar.baz/blah" |
| 07:00.43 | Juggie | agi passes a ton of info to the agi script |
| 07:00.44 | salaud | Macros are called from extensions |
| 07:00.55 | Juggie | which includes dialed extension, caller id. |
| 07:00.56 | Juggie | etc. |
| 07:01.03 | salaud | but isn't AGI an application called from an extension? |
| 07:01.10 | Juggie | yes. |
| 07:01.19 | benjk | a macro or an AGI can be coded to go through your rss feed list stored in astdb as a family |
| 07:01.23 | salaud | wouldn't AGI get passed the extension where it came from? |
| 07:01.31 | Juggie | yes of course it does. |
| 07:01.38 | salaud | that's the rub |
| 07:01.43 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 07:01.44 | Juggie | but you could do exten=>_4XXX,1,AGI(someagi.php) |
| 07:01.50 | salaud | I want to create extensions that call AGI |
| 07:01.58 | Juggie | so anything from 4000-4999 could call that agi |
| 07:02.05 | benjk | not a problem |
| 07:02.16 | Juggie | but, when the agi was called, it will know the exact extension that called it. |
| 07:02.19 | salaud | that would hard code a certain number of extensions |
| 07:02.46 | Juggie | there has to be some limit :) |
| 07:02.51 | Juggie | that doesnt mean they all have to be active. |
| 07:02.54 | benjk | it wont if you use a macro for all dials |
| 07:02.57 | salaud | not if it is dynamically |
| 07:03.05 | salaud | how does the macro work? |
| 07:03.17 | salaud | it's not really dials... all the calls are coming in |
| 07:03.24 | benjk | exten => _X., 1, Macro(SuperDial,${foobar}) |
| 07:03.25 | salaud | or maybe I'm understanding it wrong |
| 07:03.40 | Juggie | eg, someone calls 4800, and your agi runs, and knows 4800 isnt active, it rejects the request saying "that rss feed does not exist" |
| 07:03.43 | Juggie | or whatever. |
| 07:04.06 | salaud | all calls will be dumped into extension 's' though |
| 07:04.19 | benjk | so what? |
| 07:04.20 | salaud | they just get an initial IVR that directs them what to do next |
| 07:04.53 | Juggie | you are not looking at this in the proper way or something |
| 07:04.56 | salaud | the feeds could be changing all the time... there could be 10 today and then 100 tomorrow |
| 07:04.56 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 07:05.02 | salaud | Juggie: that's possible |
| 07:05.03 | Juggie | you appear to be really overconfusing yourself. |
| 07:05.09 | benjk | you can write a macro that creates feeds and assigns an extension to the feed to be created |
| 07:05.28 | Juggie | salaud, so? how does that matter, i dont understand. |
| 07:05.34 | benjk | then you put a universal extension in your ivr context |
| 07:05.43 | salaud | benjk: I'm reading feeds |
| 07:05.43 | Juggie | eg |
| 07:05.50 | Juggie | [feedivr] |
| 07:05.58 | benjk | and that calls the AGI or macro that checks what the user dialed and whether that maps to a vaild feed |
| 07:06.08 | Juggie | exten=> s,1,background(please-select-your-feed) |
| 07:06.21 | Juggie | exten=> _XXXX,1,AGI(feedrssreader.php) |
| 07:06.29 | benjk | precisely |
| 07:07.08 | salaud | Juggie: That's sorta the thing I want to do... very close |
| 07:07.10 | Juggie | then, the agi can reject feeds that dont exist. |
| 07:07.23 | Juggie | salaud, if you do that be sure to disable autofallthrough |
| 07:07.26 | Juggie | in [general] |
| 07:07.46 | Juggie | or, you can use Read() |
| 07:07.50 | Juggie | and do something like |
| 07:07.54 | Juggie | [feedivr] |
| 07:08.55 | Juggie | exten=>s,1,Read(feedivrexten|plesse-select-your-feed) |
| 07:09.19 | Juggie | exten=>s,2,AGI(feedrssreader.php) |
| 07:09.27 | salaud | The last problem is that I would like to architect this so that if the backend (AGI) is not available that it still works... this is a "would like" not a must have |
| 07:09.34 | Juggie | then, in the agi script you read the variable, feedivrexten and move along. |
| 07:09.53 | Juggie | then you will have to write it as a macro |
| 07:10.30 | Juggie | and use astdb from the dialplan (which would be a pain) or use the MYSQL or ODBC dialplan functions. |
| 07:10.42 | Juggie | so you can access one of those database types from the dialplan. |
| 07:10.51 | Juggie | which in this case would be your macro. |
| 07:10.56 | salaud | I follow that |
| 07:11.30 | benjk | it depends how complex the data structure is that describes a feed |
| 07:11.53 | benjk | if it is just a name-url list, then astdb is good enough |
| 07:12.17 | salaud | Ultimately what would I'm shooting for is a sorta FastAGI pushes changes into multiple "front-end"ish asterisk boxes ... but when the changes come in they are immediately available...but are then saved as the current config in case of a restart |
| 07:12.29 | Juggie | sigh |
| 07:12.32 | Juggie | your still not getting it |
| 07:12.46 | Juggie | * doesnt need EVERY feed to be defined. |
| 07:13.02 | Juggie | you just check if it exists when a user tries to access it |
| 07:13.16 | benjk | somehow you need to know what the URLs are |
| 07:13.27 | salaud | Juggie: Right... but the AGI can't check it... because the AGI might be down or unavailable. |
| 07:13.32 | benjk | and they have to be mapped to something that can be dialed on a keypad |
| 07:13.45 | Juggie | then dont use agi |
| 07:13.47 | Juggie | like i said |
| 07:13.53 | Juggie | use a macro w/ astdb/mysql/odbc |
| 07:13.56 | Juggie | your pick. |
| 07:14.09 | Juggie | and how would the agi be down |
| 07:14.15 | Juggie | if you run a local script on the * box |
| 07:14.23 | Juggie | only if you ran fastagi could it potentially be down. |
| 07:14.32 | salaud | I'm only running FastAGI |
| 07:14.34 | benjk | maybe somebody deleted /bin/perl :) |
| 07:14.50 | Juggie | salaud, no offense, but i think you have alot of reading to do. |
| 07:14.58 | Juggie | you dont seem to have a good understanding of what asterisk can do |
| 07:15.04 | benjk | in any event, it can be done without ARA |
| 07:15.48 | salaud | None taken... and I've done a lot of reading... I think your insight is very helpful... but, no offense, I do think you all are seeing the fault tolerance that I am trying to achieve |
| 07:16.25 | salaud | I'm envisioning something like CFEngine |
| 07:16.27 | benjk | and how would ARA solve your fault tolerance requirement? |
| 07:17.05 | Juggie | salaud, then write an AGI which can use a primary and backup db server. |
| 07:17.16 | Juggie | idealy unless you want to log into every box to edit astdb |
| 07:17.24 | Juggie | you are going to need a real database |
| 07:17.42 | salaud | I would like front-end asterisk boxes to be able to run with no dependencies once they are configured... like pushing static content to front-end html only servers that get generated from a CMS |
| 07:17.43 | Juggie | but theres no point to writeing it two ways |
| 07:18.20 | Juggie | well.... |
| 07:18.22 | Juggie | they will need |
| 07:18.25 | Juggie | a) your macro |
| 07:18.27 | Juggie | b) your agi |
| 07:18.28 | salaud | I know I could create an extensions.conf and push those files |
| 07:18.30 | Juggie | c) fast agi |
| 07:18.50 | Juggie | one of the above will be required. |
| 07:18.58 | Juggie | but they are still going to require some configuration |
| 07:18.58 | benjk | you could use DNS to advertise the feeds as TXT records |
| 07:19.11 | Juggie | all the same |
| 07:19.16 | Juggie | i think you are really overcomplicating this. |
| 07:19.23 | benjk | then have the local boxes read those TXT records in and avoid the database altogether |
| 07:19.25 | Juggie | what you are trying to do is not this complex. |
| 07:19.44 | benjk | advertising TXT records in a DNS zone is the simplest thing to do |
| 07:19.56 | Juggie | i could write what you are trying to do in an hour. |
| 07:19.58 | Juggie | if that. |
| 07:20.02 | benjk | and Asterisk already has a dialplan command to read them |
| 07:20.16 | benjk | that would give you redundancy |
| 07:20.19 | salaud | benjk: certainly don't want to advertise this application security and fault tolerance are the highest concerns ;) |
| 07:20.43 | jart | undead undead undead |
| 07:20.44 | benjk | the rss-feeds are a secret? |
| 07:20.57 | benjk | go away |
| 07:21.00 | Juggie | rss is a bad way to store information if its a secret |
| 07:21.10 | salaud | :) |
| 07:21.21 | Juggie | honestly, you are really adding too much complexity to this. |
| 07:21.24 | salaud | everything will travel across tinc |
| 07:21.28 | Juggie | what do you have to do really... |
| 07:21.32 | benjk | precisely, so you may as well advertise them in your DNS zone |
| 07:21.37 | Juggie | 1) ask for feed number |
| 07:21.42 | Juggie | 2) check to see if feed exists |
| 07:21.45 | Juggie | 3) download feed |
| 07:21.50 | Juggie | 4) read it to user |
| 07:21.54 | Juggie | this isnt a complex app. |
| 07:21.56 | Juggie | at all. |
| 07:21.57 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:21.58 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 07:22.01 | Juggie | and in the time we've been talking about |
| 07:22.19 | Juggie | it, either me or benjk probally could have written a working implementation :) |
| 07:22.27 | salaud | heh heh |
| 07:22.32 | benjk | at least a working prototype ;) |
| 07:22.47 | Juggie | you are really overthinking it. |
| 07:23.03 | salaud | I've got a set of specs and I have to develop to it... |
| 07:23.18 | Juggie | you say you want super redundancy |
| 07:23.24 | Juggie | but then you say you are using fastagi |
| 07:23.29 | Juggie | so i'll tell you outright |
| 07:23.32 | Juggie | w/ fast agi @ 1.2 |
| 07:23.35 | salaud | one of them is the end servers have to be able to run without any ties to a database or external configuration... YET be updated from remote locations |
| 07:23.38 | benjk | in any event, the question was "does one need ARA for this or are there other ways?" and the asnwer is "no, there are other ways: |
| 07:23.45 | Juggie | you cant have redundancy |
| 07:23.54 | Juggie | because agi on 1.2 is bugged |
| 07:24.04 | FlatFoot | ,2,GotoIfTime(17:01-08:59|mon-sun|*|*?bettsmithnight,s,1) |
| 07:24.04 | Juggie | if the fastagi connection is refused agi returns -1 |
| 07:24.27 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 07:24.27 | FlatFoot | opps wrong screen |
| 07:24.27 | Juggie | which then causes asterisk to hang up the call. |
| 07:24.34 | salaud | Juggie: oh... that sounds really bad |
| 07:24.34 | Juggie | so if your fastagi server is down, * will hangup the call regardless. |
| 07:24.38 | Juggie | indeed. |
| 07:24.41 | salaud | Juggie: Doh! |
| 07:24.45 | Juggie | its fixed in 1.4 |
| 07:24.52 | Juggie | and it could easily be patched in 1.2 if you wanted. |
| 07:25.23 | Juggie | i suggest you forget about using FastAGI |
| 07:25.28 | Juggie | and use a macro |
| 07:25.31 | Juggie | or a local agi |
| 07:25.40 | benjk | stick your rss-feeds into TXT records in your DNS zone and write a macro that collect the information about feeds from there, then presents an audible menu to the caller |
| 07:25.54 | benjk | that will give you super redundancy |
| 07:27.03 | salaud | benjk: so the macro would read the digits and jump to an extension? Or would they mere dial a pre-allocated extension? |
| 07:27.41 | benjk | the macro would read the available rss-feeds from your DNS and then present the user with an audio menu |
| 07:27.53 | benjk | the user would then push whatever the menu offers |
| 07:28.16 | salaud | In any case it looks like, unless I just statically generate an extensions.conf I am relying on a database of some for on the local client. Which might be so bad |
| 07:28.18 | benjk | if he pushes in a number the menu didn't offer the macro will send him to an announcement that says "no such feed" |
| 07:28.40 | benjk | if he pushes in a number that the menu did offer, the macro will fetch the feed and read it back to the user |
| 07:28.49 | Juggie | why is relying on a db bad? |
| 07:29.05 | benjk | I am not saying it is |
| 07:29.15 | salaud | benjk: specific question is how the macro will capture that number... by reading it and then jumping or merely sitting wait for an extension be dialed |
| 07:29.18 | benjk | first, DNS is a distributed DB |
| 07:29.32 | salaud | DNS may be too public for this app.. |
| 07:29.57 | salaud | maybe the local db isn't so bad |
| 07:30.05 | benjk | secondly, he seems to be having issues with building the redundant parts himself, so DNS would be a short cut to use existing infrastructure that is redundant and easy to maintain |
| 07:30.21 | benjk | why is it too public if you are using rss feeds |
| 07:30.38 | benjk | if you use rss feeds then the info is already public and it doesn't get any more public |
| 07:31.10 | salaud | an rss feed is just something that exists on a webserver... in this case a webserver that can only be accessed across a VPN |
| 07:31.36 | salaud | perhaps set of DNS servers on a VPN would be a good analogy |
| 07:31.40 | benjk | in that case you have nothing to worry about because anybody who would obtain those URLs from your DNS wouldn't be able to access the feeds anyway |
| 07:32.01 | salaud | benjk: that's a good point |
| 07:32.17 | Juggie | salaud, your going to need a db of some sort |
| 07:32.32 | Juggie | if you dont your going to have to generate .conf files and upload them to every server |
| 07:32.36 | Juggie | and i'm sure you dont want to do that |
| 07:32.57 | salaud | Juggie: You are right... OR create extensions on the fly |
| 07:33.07 | Juggie | right |
| 07:33.09 | Juggie | but |
| 07:33.11 | salaud | THEN save the configuration at intervals |
| 07:33.17 | Juggie | unless you want to use asterisk realtime |
| 07:33.20 | benjk | its either a redundant database server like Postgres etc etc or a distributed database like DNS |
| 07:33.24 | Juggie | there is no way to generate extensions on the fly |
| 07:33.28 | Juggie | unless you use a db of some sort |
| 07:33.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 07:33.30 | salaud | is realtime broken in 1.2? or is it pretty good |
| 07:33.37 | Juggie | eg, dns,asttb,mysql,odbc.... etc |
| 07:34.01 | Juggie | salaud, if i were you |
| 07:34.05 | Juggie | here is what i would do. |
| 07:34.05 | benjk | even if you create extensions on the fly, you still need a database that tells you what feeds are available |
| 07:34.10 | salaud | Juggie: in a macro are you really generating extensions or reading digits? |
| 07:34.10 | benjk | so you win nothing |
| 07:34.25 | Juggie | i read digits |
| 07:34.28 | Juggie | and then check the db |
| 07:34.32 | Juggie | thats what i would do in a macro |
| 07:34.38 | Juggie | or, i could pass the digits into the macro |
| 07:34.40 | Juggie | already read |
| 07:34.45 | Juggie | either way |
| 07:35.00 | Juggie | anyways |
| 07:35.01 | Juggie | if i were you |
| 07:35.03 | Juggie | i would do this |
| 07:35.09 | Juggie | write an AGI |
| 07:35.16 | Juggie | in whatever language you want |
| 07:35.24 | salaud | benjk: not entirely. If I could create extensions (and the logic) on the fly the extensions would already know what's available... because, if the extensions is there, it's available |
| 07:35.33 | Juggie | which is inteligent enough to connect to a primary/backup database server. |
| 07:35.47 | Juggie | and then write your logic in there to do the check for the extension |
| 07:36.01 | Juggie | and play the feed etc |
| 07:36.11 | *** join/#asterisk zedkatuf (n=audela@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 07:36.15 | benjk | and how does whatever creates the extensions on the fly know which rss feeds are avail;able? |
| 07:36.27 | Juggie | exactally, theres still a db :) |
| 07:36.40 | Juggie | there has to be. |
| 07:36.47 | benjk | precisely my point |
| 07:37.04 | Juggie | and mine too |
| 07:37.17 | Juggie | i think we've both exhausted our powers explanation |
| 07:37.18 | salaud | The best way to look at it would be replicating the database to the asterisk server |
| 07:37.22 | Juggie | er, of explanation. |
| 07:37.22 | benjk | so creating extensions on the fly does nothing for you there |
| 07:37.30 | Juggie | salaud, holy shit |
| 07:37.40 | Juggie | how redundant does this need to be |
| 07:37.47 | Juggie | whats wrong with a remote db |
| 07:37.59 | Juggie | or a primary and backup |
| 07:38.02 | salaud | Juggie: Think dialup lines... cell phones ... |
| 07:38.09 | benjk | if you need military strength redundancy to survive a nuclear attack, use DNS |
| 07:38.12 | salaud | Think jungles of south america |
| 07:38.19 | Juggie | salaud, soooooo |
| 07:38.27 | Juggie | their phone isnt running the database |
| 07:38.32 | Juggie | its all contained centrally |
| 07:38.36 | Juggie | same place the webserver is |
| 07:38.59 | Juggie | if you can depend on a webserver, why not an database? |
| 07:39.47 | salaud | That's not exactly true... the connection to the database server (for the purpose of replication) be only accesible via a dial-up or worse... GSM |
| 07:40.29 | benjk | so you need a cache |
| 07:40.53 | benjk | again DNS is your low cost easy maintenance friend |
| 07:40.59 | salaud | That's why having flat text XML/RSS be what is sent from the app server (I was thinking FastAGI... but that's wrong... sounds more like realtime) to the asterisk server and then the asterisk server would be set until the next it could make contact with the app server |
| 07:41.10 | Juggie | i'm so confused. |
| 07:41.30 | benjk | salaud is confused |
| 07:41.43 | salaud | It's a confusing app... |
| 07:41.50 | benjk | that's not what realtime is for |
| 07:41.52 | salaud | or not confusing... |
| 07:41.57 | Juggie | whats the point of deploying multiple boxes around say the jungles of africa |
| 07:42.01 | benjk | its very simple really |
| 07:42.03 | Juggie | why not just dial into one central location |
| 07:42.28 | benjk | long distance charges perhaps |
| 07:42.36 | salaud | that's right |
| 07:42.44 | salaud | everything will actually be done with callbacks |
| 07:43.03 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 07:43.19 | Juggie | is saving long distance really going to be a savings when you consider the cost of installing/maintenance of remote boxes |
| 07:43.26 | benjk | so what kind of redundancy do you need? |
| 07:43.40 | benjk | do you want to cover the loss of a box, or the loss of a site? |
| 07:43.42 | salaud | It is possible that the asterisk servers might be running on WRT54Gs or some sort of embedded platform with a 1GB mem stick |
| 07:43.59 | benjk | even more reason to use DNS as a database then |
| 07:44.00 | *** join/#asterisk engineeer (i=1001@adsl-68-94-12-209.dsl.rcsntx.swbell.net) |
| 07:44.02 | Chunks | Sry.. quick question.. any idea why Xlite phone (x1002) can call Cisco (x1001) ONLY if I dial 1001@192.168.1.11? Am I bypassing *? |
| 07:44.17 | Juggie | yes |
| 07:44.34 | Juggie | assuming the cisco phone ip is 1.11 |
| 07:44.48 | Chunks | so calling either extension directly, going straight to vmail... is that a routing problem? |
| 07:44.49 | Chunks | yes it is |
| 07:45.07 | Juggie | Chunks, you are bypassing * so no vmail will exist |
| 07:45.19 | Juggie | salaud, ok, so if you need to mirror remote boxes |
| 07:45.27 | Chunks | sorry, if I dial them normally, 1001 or 1002, they go straight to vmail, which is * |
| 07:45.46 | Juggie | then it sounds like you need to write a crontab to download all the feeds and the then a macro to read them back. |
| 07:45.54 | Juggie | when someone calls in. |
| 07:46.01 | salaud | I REALLY and I mean really appreciate benjk and Juggie's help... I'm starting to narrow down the options |
| 07:46.09 | zedkatuf | Hi, could someone help me with a local-calling problem I have? I want to be able to dial 9 & then a local number (without the area code) & at the moment, it doesn't work..here's a pastebin of what I've done (please be patient...I'm new to asterisk!): |
| 07:46.11 | Juggie | if you want to go super light weight |
| 07:46.11 | zedkatuf | http://pastebin.ca/120292 |
| 07:46.24 | Juggie | use your crontab to download the rss feeds and name them per extension |
| 07:46.33 | salaud | Juggie: that's sounding more and more like what I need to do... pull rather than push |
| 07:46.41 | Juggie | eg, if its a rss feed for extension 4111 |
| 07:46.46 | Juggie | then name it 4111.rss |
| 07:46.59 | benjk | what if your crond is down though :D |
| 07:47.00 | Juggie | then you can check and see if the feeds exist by just looking for the file when the user inputs what they want |
| 07:47.14 | Juggie | so, user inputs 5111 you check, oh no 5111.rss |
| 07:47.16 | Juggie | so read an error |
| 07:47.21 | Juggie | user puts in 4111 |
| 07:47.22 | Juggie | you check |
| 07:47.26 | Juggie | oh 4111.rss exists |
| 07:47.30 | Juggie | and you read it out |
| 07:47.41 | salaud | Right... and that would be using astdb? |
| 07:47.49 | benjk | it could be |
| 07:47.50 | salaud | or just looking for a file? |
| 07:47.52 | benjk | doesn't have to |
| 07:47.56 | Juggie | that would be using an agi |
| 07:47.59 | Juggie | and just looking for the file. |
| 07:48.04 | benjk | it could, doesnt need to |
| 07:48.04 | salaud | Ok... |
| 07:48.21 | Juggie | benjk, unless theres functions in the dialplan to check for files? |
| 07:48.26 | benjk | I really think you shoudl look into the DNS scenario |
| 07:48.30 | Juggie | what i was suggesting bypasses a db totally. |
| 07:48.36 | benjk | its much simpler than you may think |
| 07:48.37 | Juggie | by just naming the files as the extensions. |
| 07:48.48 | benjk | Astewrisk already has a command to read TXT records from DNS |
| 07:48.56 | benjk | a dial plan command |
| 07:49.03 | Juggie | right, but these boxes are disconnected from the central location |
| 07:49.04 | Juggie | most of the time |
| 07:49.07 | Juggie | only connected to sync |
| 07:49.13 | Juggie | each remote box would need to run a dns server |
| 07:49.19 | benjk | then run a local caching DNS server |
| 07:49.29 | hwt | in my realtime setup, howcome sip friends are listed as Nat=N even though i have Nat=yes in the database? |
| 07:49.41 | Juggie | how can you guarantee that the dns will sync for the say 5minutes a connection is up |
| 07:50.19 | benjk | you could expire the cache and make a query |
| 07:50.23 | engineeer | zedkatuf U need to keep the 9NXXXXXX consistent thru all the dial numbers |
| 07:50.46 | Juggie | i think it would be more productive to connect centrally get a list of extensions and files, and then download all the rss feeds, naming them as their extensions so, if exten 100 is active then download the file and name it 100.rss |
| 07:50.50 | engineeer | zedkatuf U have 2 dial patterns there so its getting confused |
| 07:51.02 | Juggie | then when someone dials in no db/dns etc is required |
| 07:51.16 | benjk | well, the files will be your db |
| 07:51.25 | Juggie | it just goes based on file names if the file exist read the feed, if it doesnt, dont read the feed. and say no feed exists |
| 07:51.32 | Juggie | right. |
| 07:51.43 | benjk | depends on volume and other factors which approach is favourable |
| 07:52.03 | Juggie | well, i would agree with you, if they were being cached realtime |
| 07:52.11 | benjk | on a Flash memory I would want to avoid downloading though |
| 07:52.11 | Juggie | but since the intent is to download them all and disconnect from the remote |
| 07:52.31 | Juggie | i think that may be the best solution. |
| 07:53.00 | salaud | Essentially this application seems to be an Audio Blog reader where it is impossible to know how many blogs there will be, but still allow for the ordering (in terms of which extensions come first second third) |
| 07:53.02 | benjk | at the very least you will need to have some mechanism to avoid downloading stuff that hasn't changed |
| 07:53.15 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 07:53.18 | hwt | also i have problems with "mysql server has gone away". |
| 07:53.21 | hwt | when inactive for a while. |
| 07:53.24 | hwt | how can i increase the mysql socket timeout? |
| 07:53.28 | salaud | benjk: that's one of the motivations behind using RSS |
| 07:53.35 | Shaun2222 | anybody know why i get this error when running asterisk+mysql... `parse error: No category context for line 6 of extconfig.conf` |
| 07:53.47 | Juggie | salaud, i wrote a tool like that not long ago using php. |
| 07:53.53 | salaud | benjk: RSS has some abilities to do "show me what has changed" |
| 07:53.53 | Juggie | it was super simple. |
| 07:54.01 | benjk | but your downloading needs to be able to take advantage of the time stamping |
| 07:54.14 | zedkatuf | engineeer: tnx..I half get what you're saying, but would you be able to post me the solution as I've already spent several hours on it & am going round in circles atm!!! I'm sure it's very easy once you know what you're doing (I've been reading AsteriskTFOT & it's v good but my brain hurts!) |
| 07:54.19 | Juggie | basically, i would do like, http://somehost/?request=filelist |
| 07:54.19 | benjk | its simple as long as you are connected all the time |
| 07:54.33 | Juggie | which would then provide me with a list of files in csv |
| 07:54.35 | salaud | benjk: right |
| 07:54.44 | Juggie | including their md5. |
| 07:54.49 | benjk | but if you are downloading the stuff for local caching, you will need to add some logic to make sure you are only downloading what's necessary |
| 07:55.20 | Juggie | then, i would compare the md5's of the remote file list to my local files |
| 07:55.24 | Juggie | and download what was necessairy. |
| 07:55.45 | benjk | it really depends on how frequently you connect to the backbone and also the conditions under which such a connection is made and for how long etc |
| 07:55.53 | Juggie | salaud, is the connection between remote/central perm or only temporary. |
| 07:55.58 | benjk | then also the volume |
| 07:55.59 | salaud | temporary |
| 07:56.04 | Chunks | ok.. sip debug shows this (when the calls for my two phones go straight to vmail..) X-Asterisk-HangupCause: No route to destination. Any idea how I can check my routes? is there a "route print" command for *? |
| 07:56.06 | Juggie | ok then you need a cacheing model. |
| 07:56.06 | salaud | and slow |
| 07:56.22 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:56.24 | Juggie | ok, i'm gonna tell you how to do this with only the webserver. |
| 07:56.28 | benjk | if it is slow and the number of feeds is large, then downloading could become a bottleneck |
| 07:56.43 | Juggie | benjk, but he has to mirror them all. |
| 07:56.46 | engineeer | zedkatuf http://pastebin.ca/120305 something more like that |
| 07:56.51 | benjk | maybe you want a caching web proxy |
| 07:57.08 | salaud | benjk: I don't think the number of feeds will be large... but, which feeds are available will change fairly often |
| 07:57.12 | benjk | and configure it such that it only does the feeds nothing else |
| 07:57.52 | zedkatuf | engineeer: brilliant, thanks..I'll try it now |
| 07:57.55 | salaud | benjk: The order in which the feeds should be presented will change often too |
| 07:58.10 | benjk | that's up to your AGI or macro |
| 07:58.25 | engineeer | zedkatuf put a :1 at the end of the saydigits line or it will say the 9 as well |
| 07:58.28 | Juggie | salaud, you essentially need a way to download all the feeds and assign them numbers. to be used in * to access them. |
| 07:58.56 | Juggie | so as i suggest, i recomend writing a crontab which connects, downloads them, and names them based on the number you want to be able to access them. |
| 07:58.58 | salaud | Juggie: that's right.... and those numbers are based on some metadata in the RSS feed |
| 07:59.19 | salaud | or I should say.. feedS |
| 07:59.32 | Juggie | so, if the metadata says 4000 then name the file 4000.rss or whatever. |
| 07:59.42 | salaud | Juggie: right.. |
| 07:59.57 | Juggie | that way your local script is very simple, user inputs the number they want |
| 08:00.03 | Juggie | check and see if the file exists. |
| 08:00.10 | Juggie | and then based on that, play feed not found, or read it. |
| 08:00.17 | salaud | I was envisioning the central pushing the way CFEngine does... but, maybe pulling is better |
| 08:00.25 | Juggie | the only other consideration is cacheing of the feeds |
| 08:00.32 | engineeer | zedkatuf move your :1 inside the {} I fat fingered that too sorry |
| 08:00.34 | Juggie | such that you onlydownload what changes. |
| 08:00.45 | zedkatuf | engineeer: ok, tnx :) |
| 08:00.46 | Juggie | since bandwith appears to be a consideration. |
| 08:01.07 | salaud | Juggie: That's why RSS was selected... RSS aggregation modules already exist that remember the last item that was seen |
| 08:01.18 | salaud | and only download what's new |
| 08:01.37 | salaud | no need to write that again |
| 08:01.40 | Juggie | uhhuh |
| 08:01.44 | Juggie | well i dont know anything about that. |
| 08:01.51 | zedkatuf | engineeer: sorry to be a real pain in the butt, but if you could pastebin the new version I would be very grateful |
| 08:01.55 | Juggie | sounds like you would just need to use the rss tool to mirror all the rss |
| 08:01.57 | Juggie | then. |
| 08:02.03 | Juggie | and then write a crontab to read them all |
| 08:02.07 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 08:02.18 | Juggie | and move them into a dir where they will be named and acessible for * to use. |
| 08:02.34 | salaud | Juggie: yep... this is about 98% of the solution.... |
| 08:02.42 | Juggie | ok well good luck. |
| 08:02.45 | salaud | I really appreciate the help and having folks to bounce this off |
| 08:03.00 | salaud | There are so many possible ways to skin the cat |
| 08:03.00 | Juggie | remember K.I.S.S |
| 08:03.14 | Juggie | (Keep it simple stupid) |
| 08:03.47 | salaud | that's the idea... the simplest way I can think to do it would be create an extensions.conf that was totally static (no db needed and no AGI needed) and just push/pull them and reload |
| 08:03.51 | engineeer | zedkatuf the :1 strips the leading nine off of your dial string 120308 |
| 08:04.04 | salaud | I just want to avoid the reload |
| 08:04.40 | engineeer | also looks like its going to repeat the digits 2 times.. do U really want that? |
| 08:04.51 | salaud | let me start hacking it out more a bit... and maybe I'll check back in a day or so and share what I have learned... |
| 08:05.01 | salaud | I appreciate the good asterisk community |
| 08:05.18 | Juggie | do the heavy lifting in an AGI |
| 08:05.31 | Juggie | dont change your extensions.conf |
| 08:05.44 | mitcheloc | ** or the family friendly version -- Keep it super simple ;) |
| 08:06.03 | salaud | what is required to patch the FastAGI bug anyway? Even if this is not the solution? Is there already a patch? |
| 08:06.21 | zedkatuf | engineeer: no, I don't really need it to be saying digits....that was just my attempt to troubleshoot what wasn't working :) |
| 08:06.37 | engineeer | Id pull those lines out and renumber |
| 08:06.46 | zedkatuf | ok |
| 08:08.35 | engineeer | I assume your sip.conf file is correct |
| 08:08.51 | zedkatuf | yes |
| 08:09.05 | zedkatuf | just pastebinning my new version |
| 08:09.28 | zedkatuf | http://pastebin.ca/120311 |
| 08:09.47 | *** join/#asterisk drega (n=drega@mail.amcat.co.uk) |
| 08:10.31 | engineeer | pastbin your sip.conf without the sensitive info... |
| 08:12.31 | zedkatuf | engineeer: I can dial long distance no problem so I think sip.conf is OK |
| 08:12.50 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:12.51 | engineeer | ok just making sure.. |
| 08:12.56 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 08:13.35 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:13.45 | engineeer | did it work? |
| 08:13.56 | zedkatuf | no, still get fast busy signal |
| 08:14.13 | zedkatuf | it looks fine though, that's the frustratinbg thing |
| 08:14.43 | zedkatuf | hehe, but it looked fine before too to me so I guess I shouldn't get too frsutrated so quickly :) |
| 08:14.56 | engineeer | watching the cli does the number look right when * dials out |
| 08:15.07 | engineeer | properly formatted |
| 08:15.21 | zedkatuf | no, it still has the 9 on it at the front & no 0117 prefix |
| 08:15.45 | creativx | so, does anyone have a good reccomendation for a handsfree headset that can be used with any phone jack? |
| 08:15.46 | engineeer | did U reload extensions.conf |
| 08:15.53 | zedkatuf | yep |
| 08:15.56 | L|NUX | then i think you should check your extensions.conf |
| 08:16.01 | fourcheeze | can someone explain SRV records in dns and SIP or point me at a page? I'm trying to find out if they mean anything at registration time or whether it's just for calls. |
| 08:16.18 | L|NUX | if you are using some prefix then you should remove them when its dialing actual number |
| 08:16.39 | L|NUX | e.g _9.,1,Dial(SIP/provider/${EXTEN:1}) |
| 08:16.44 | L|NUX | it will remove 9 |
| 08:16.44 | engineeer | he is trying to put an 0117 before the number he is dialing |
| 08:16.58 | L|NUX | engineeer> : he can use same for the number |
| 08:16.59 | fourcheeze | _9.,1,Dial(SIP/provider/0117${EXTEN:1}) |
| 08:17.03 | L|NUX | right |
| 08:17.12 | L|NUX | fourcheeze : indeed |
| 08:17.20 | engineeer | thats what I put in there.. but he says its not working |
| 08:17.29 | zedkatuf | http://pastebin.ca/120311 |
| 08:17.31 | L|NUX | did he reloaded ? |
| 08:17.35 | L|NUX | let me check |
| 08:17.36 | zedkatuf | is the latest non-working version |
| 08:17.39 | zedkatuf | yep reloaded |
| 08:17.45 | L|NUX | wait |
| 08:17.54 | fourcheeze | although why you want to call Bristol I'll never know |
| 08:17.58 | zedkatuf | lol |
| 08:18.00 | zedkatuf | :) |
| 08:19.15 | L|NUX | zedkatuf : show dialplan outbound-local |
| 08:19.20 | L|NUX | on your * cli |
| 08:19.37 | zedkatuf | hang on....the StripMSD is obviated by the EXTEN:1 perhaps? |
| 08:19.39 | zedkatuf | ok |
| 08:19.43 | zedkatuf | re cli |
| 08:19.50 | zedkatuf | pastebinning....... |
| 08:19.55 | L|NUX | k |
| 08:20.01 | L|NUX | i also need to see your sip.conf |
| 08:20.05 | L|NUX | if you do not mind |
| 08:20.20 | engineeer | he says ld is working.. so he did not want to post it |
| 08:20.36 | L|NUX | well he can remove passwords |
| 08:20.39 | L|NUX | and usernames :) |
| 08:20.50 | engineeer | thats what I suggested... but hey just trying to help |
| 08:20.52 | L|NUX | but i want to see might be its the context issue |
| 08:21.30 | zedkatuf | the context is internal |
| 08:21.33 | L|NUX | engineeer : basically i know the problem |
| 08:21.39 | zedkatuf | for the extension I'm using |
| 08:21.49 | L|NUX | problem is that he is using _9 for both long distance and for local |
| 08:21.53 | zedkatuf | ahh |
| 08:21.58 | L|NUX | :P |
| 08:22.16 | L|NUX | http://pastebin.ca/120319 |
| 08:22.17 | zedkatuf | ok, will try using 8 see if that cracks it :P |
| 08:22.19 | L|NUX | this should work |
| 08:22.19 | L|NUX | :) |
| 08:22.26 | fourcheeze | use +9XXXXXXX for local |
| 08:22.36 | fourcheeze | s/+/_ |
| 08:22.55 | *** join/#asterisk chexum (i=jfarkas@gateway/tor/x-726888265e8cf560) |
| 08:24.06 | engineeer | I would think it would differentiate with diff string lengths but maybee not.. |
| 08:24.31 | engineeer | intersting to see if it works |
| 08:24.55 | fourcheeze | oh sure it would work |
| 08:26.05 | zedkatuf | http://pastebin.ca/120320 |
| 08:26.07 | zedkatuf | no worko |
| 08:26.19 | zedkatuf | reloaded too :) |
| 08:26.29 | engineeer | I like the context idea myself... |
| 08:26.37 | L|NUX | hummm |
| 08:26.40 | L|NUX | show your sip.conf |
| 08:28.43 | engineeer | Hey L|NUX he could take the oubound local out just to test to see if its a conext issue... |
| 08:28.54 | engineeer | context |
| 08:29.14 | engineeer | or comment it out |
| 08:29.44 | L|NUX | humm |
| 08:30.07 | zedkatuf | engineeer: yes, the long distance bit works fine |
| 08:30.19 | L|NUX | zedkatuf : hummm |
| 08:30.47 | engineeer | my idea is to comment out the outbound-local line which will make the ld and the local part of the same context and test |
| 08:31.04 | L|NUX | hummm |
| 08:31.05 | engineeer | will only cost U a ; |
| 08:31.18 | L|NUX | engineeer : lets see |
| 08:32.14 | zedkatuf | yep, have commented out the local bit & long disance works fine on its own....interestingly, the long distnace bit works even with the non-commented-out local dialplan bit there too |
| 08:32.17 | fourcheeze | zedkatuf: |
| 08:32.18 | fourcheeze | exten => _9XXXXXXX,1,StripMSD,1 |
| 08:32.18 | fourcheeze | exten => _9XXXXXXX,2,Prefix,0117 |
| 08:32.21 | fourcheeze | you don't want those two |
| 08:32.33 | zedkatuf | righty...no need for em |
| 08:32.52 | zedkatuf | as they're obviated by the clever stuff later... |
| 08:32.55 | zedkatuf | will try now |
| 08:33.34 | L|NUX | zedkatuf : http://pastebin.ca/120322 |
| 08:33.37 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 08:33.57 | engineeer | thats as basic as it gets |
| 08:34.05 | fourcheeze | zedkatuf: how many users have you? |
| 08:34.25 | L|NUX | i have seen only one extension :) |
| 08:34.26 | L|NUX | 203 |
| 08:34.27 | L|NUX | :P |
| 08:34.35 | zedkatuf | fourcheeze: BINGO |
| 08:34.45 | L|NUX | works |
| 08:34.46 | zedkatuf | thanks for ur patience everyone |
| 08:34.55 | zedkatuf | yep |
| 08:34.55 | zedkatuf | works |
| 08:35.00 | zedkatuf | now I just have to learn how to get incoming working :) |
| 08:35.01 | fourcheeze | zedkatuf: if you have a lot of users you might just want: |
| 08:35.16 | fourcheeze | exten => _XXXXXXX,3,Dial(${OUTBOUNDTRUNK}/0117${EXTEN}) |
| 08:35.22 | fourcheeze | make every 7 digit call local |
| 08:35.40 | L|NUX | incoming is easy :) |
| 08:35.41 | zedkatuf | fourcheeze: that make a lot of sense |
| 08:35.57 | fourcheeze | it saves on training telling people to dial 9 |
| 08:36.02 | benjk | thats very crude though |
| 08:36.07 | zedkatuf | L|NUX: it's all easy when you know what to do |
| 08:36.07 | fourcheeze | why? |
| 08:36.15 | zedkatuf | benjk: |
| 08:36.17 | L|NUX | zedkatuf: you need to register your did or user in your sip.conf or iax.conf |
| 08:36.22 | zedkatuf | It's only for home use |
| 08:36.32 | L|NUX | e.g. user:pass@host:port/exten |
| 08:36.35 | zedkatuf | L|NUX: yep, registered in sip.conf |
| 08:36.37 | fourcheeze | also you could make every 10+ digit number dial out |
| 08:36.41 | benjk | I have done hardcoded dialplans like that for the longest time but was never quite happy with it |
| 08:36.41 | engineeer | he means wife is trainable |
| 08:36.48 | L|NUX | :) |
| 08:36.53 | zedkatuf | yep |
| 08:37.01 | zedkatuf | re trainable wife |
| 08:37.13 | fourcheeze | benjk: it's generally how least cost routing has worked for ages |
| 08:37.17 | zedkatuf | she's quite happy dialling different prefices for calls |
| 08:37.26 | L|NUX | zedkatuf : and you need to define incoming context |
| 08:37.28 | benjk | what I do is ACR |
| 08:37.46 | L|NUX | and in your extensions.conf you need to tell what action to do when call comes in |
| 08:37.46 | L|NUX | :) |
| 08:37.49 | L|NUX | simple |
| 08:37.50 | L|NUX | :) |
| 08:37.51 | zedkatuf | benjk: this is on a low cpu box so I'm minimizing bells & whistles |
| 08:38.12 | benjk | I have a dial macro which first calls another macro that determines the type of call |
| 08:38.12 | fourcheeze | does anyone know if a hint can be passed/received to/from another box? |
| 08:38.16 | zedkatuf | ok, L|NUX - I'll give it a go |
| 08:38.22 | benjk | the type of call then determines the optimal route |
| 08:38.50 | *** join/#asterisk Geliman (n=DrkShdw@unaffiliated/drkshdw) |
| 08:38.59 | benjk | then if the route fails, it will use the second most optimal route until no more routes for this type of call are listed |
| 08:39.03 | benjk | and this is all dynamic |
| 08:39.44 | zedkatuf | benjk: this is sort of how my trixbox does it I guess...I can live with just having a few prefixes for my wife to try if one fails |
| 08:39.46 | benjk | my dialplan has nothing in it other than assigning a few global vars |
| 08:40.19 | zedkatuf | maybe I'll invest time in that idea sometime down the line |
| 08:43.20 | fourcheeze | anyone know how presence can be shared between servers? |
| 08:43.33 | Pj_ | (this is not the kind of whistle and bells that will eat your cpu up too ;) |
| 08:43.34 | JT | benjk: is there anything like that documented on the web? |
| 08:43.43 | benjk | like what? |
| 08:43.43 | JT | benjk: or mind sharing a bit more info about it? |
| 08:43.58 | benjk | http://pastebin.ca/120330 |
| 08:43.58 | JT | your call routing setup |
| 08:45.47 | engineeer | hey benjk are you in the UK if so.. how do U guys keep the dial plans straight up there? |
| 08:45.55 | JT | benjk: am i right in gathering that that is just some of the setup? |
| 08:47.33 | *** join/#asterisk Geliman (n=DrkShdw@unaffiliated/drkshdw) |
| 08:48.41 | benjk | not in the uk |
| 08:48.42 | benjk | http://pastebin.ca/120331 |
| 08:50.10 | engineeer | well that last paste bin asnwers that question |
| 08:50.29 | engineeer | benjk thanks for the info.. |
| 08:50.49 | JT | interesting |
| 08:50.57 | JT | did you include any of the macros? |
| 08:51.30 | benjk | the macros and apps are 3500 lines of code |
| 08:52.03 | JT | ah heh |
| 08:52.10 | *** join/#asterisk svenna (n=svenna@p548D3C58.dip0.t-ipconnect.de) |
| 08:52.21 | JT | i thought i was only seeing an aspect of the solution :) |
| 08:52.27 | benjk | not everybody will require as much flexibility but my point is that the idea should be not to hardcode too much inro your dialplan |
| 08:52.29 | JT | sounds like a nice setup |
| 08:53.04 | benjk | start with two macros, one for outbound and one for inbound calls |
| 08:53.33 | benjk | classify all the outbound calls into things such as local, national, mobile, etc etc |
| 08:54.02 | benjk | classify the inbound calls by who is calling such as internal, external, known/unknown caller etc |
| 08:55.44 | JT | hmm |
| 08:56.29 | benjk | also if you have multiple DIDs, then inbound call processing may want to take into account which DID a call was coming in on |
| 08:57.23 | benjk | do you really want to replicate all the logic for the various contexts you need to hardcode if you hardcode? |
| 08:57.23 | JT | yeah the setup i'm thinking of doing will have multiple DIDs, etc over inbound ISDN2s |
| 08:57.41 | JT | it'll need to then pump them back out over other ISDN2s to the digital key system :) |
| 08:57.48 | JT | no, not really at all, no |
| 08:57.55 | JT | hency why your solutions sounds like a good idea |
| 08:58.08 | benjk | you want somehting like ... |
| 08:58.19 | JT | chucking everything into hardcoded dialplan sounds like a nightmarr |
| 08:58.25 | JT | nightmare |
| 08:58.31 | benjk | ; PSTN DID 1 |
| 08:58.31 | benjk | PSTN_DID1 = 03xxxxxxxx |
| 08:58.32 | benjk | PSTN_DID1_LANGUAGE1 = jp |
| 08:58.32 | benjk | PSTN_DID1_ENUM_MAPPED = YES |
| 08:58.32 | benjk | ; |
| 08:58.32 | benjk | ; PSTN DID 2 |
| 08:58.34 | benjk | PSTN_DID2 = 03xxxxxxxx |
| 08:58.36 | benjk | PSTN_DID2_LANGUAGE1 = uk |
| 08:58.38 | benjk | PSTN_DID1_ENUM_MAPPED = YES |
| 09:00.12 | JT | is that meant to be DID2 on the last line? |
| 09:00.24 | benjk | :) |
| 09:00.52 | alexcf | heh, i hate it when that happens |
| 09:01.00 | benjk | I guess nobody called us via ENUM on the English speaking number and complained ;) |
| 09:01.06 | alexcf | i usually paste something like that, then look at it again and scream d'oh |
| 09:01.42 | benjk | in any event, make sure you have something like this ... |
| 09:01.43 | benjk | PREMIUMRATE_CALLS_BARRED = YES |
| 09:02.36 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 09:02.55 | JT | premium rate call blocking isn't my biggest concern |
| 09:03.20 | benjk | it becomes a concern once you allow call forwarding per extension to be user defined ;) |
| 09:03.39 | JT | heh |
| 09:03.58 | JT | not going to have any end user extensions on asterisk |
| 09:04.31 | benjk | well, this setup is for PBX use |
| 09:04.51 | benjk | and there are many user definable features |
| 09:05.32 | JT | i just basically need to route outbound calls based on prefix and whether or not the SIP provider/Internet is dead |
| 09:05.41 | JT | and inbound calls will almost be passthrough |
| 09:05.55 | JT | with 2 sets of ISDN2 ports |
| 09:05.57 | benjk | well, then its half the work |
| 09:06.17 | benjk | actually, the user definable settings by phone were a big piece of work too |
| 09:06.26 | JT | one connects to conventional telco, other set connects to PABX |
| 09:06.56 | JT | obviously it also needs to replicate MSN number data in signalling so DIDs still work |
| 09:07.07 | benjk | make sure you don't trust Asterisk with your call progress./status reporting |
| 09:07.18 | JT | what do you mean? |
| 09:07.25 | benjk | use the hangupcause codes that the PSTN sends back, not DIALSTATUS |
| 09:07.29 | benjk | DIALSTATUS is a liar |
| 09:07.42 | JT | hmm, i'll keep it in mind |
| 09:09.27 | fourcheeze | anyone know why a client would generate a 489 "Bad Event" ? |
| 09:10.23 | fourcheeze | (it's not in RFC 3261) |
| 09:10.31 | JT | benjk: how do you detect if the SIP provider or network is down? |
| 09:10.35 | benjk | basically you look at HANGUPCAUSE and make your own labels, don't use DIALSTATUS |
| 09:10.37 | benjk | like |
| 09:10.38 | benjk | exten => s-NOANSWER,1,GotoIf($["${HANGUPCAUSE}" = "1"]?s-UNALLOCATED|1) |
| 09:11.08 | benjk | there are plenty of cases you want to cover, not all of them but many more than what DIALSTATUS says |
| 09:11.19 | JT | hmm |
| 09:11.54 | benjk | this way you can actually provide useful feedback info to the callers |
| 09:12.13 | benjk | Asterisk always claims "nobody is available to take your call at this time" |
| 09:12.25 | benjk | which is most of the time incorrect |
| 09:13.50 | creativx | awesome |
| 09:14.05 | creativx | i need to rewrite my outbound macro with the hangupcause var |
| 09:14.27 | benjk | you want to lookup the list of hangup causes, there's 128 of them |
| 09:16.48 | JT | hmm |
| 09:17.11 | JT | benjk: any ideas on detecting a bad voip route? |
| 09:17.52 | benjk | use qualify=yes in the peer entry, then Asterisk will tell you |
| 09:18.44 | benjk | but VoIP routes are a dime a dozen, just move on to the next one |
| 09:19.01 | JT | does that make it check on dial attempt? |
| 09:19.02 | benjk | app_dial will usually report CHANUNAVAIL or CONGESTION back to you |
| 09:19.08 | JT | heh |
| 09:19.10 | JT | hmm |
| 09:19.26 | benjk | use the g flag in the Dial command to make the macro continue |
| 09:19.37 | benjk | then check DIALSTATUS |
| 09:19.51 | JT | in the case of ongoing outage it would be nice to have a global var or something so that each dial attempt doesn't have to waste time checking the first route |
| 09:20.01 | benjk | if it says CHANUNAVAIL or CONGESTION after a VoIP dial out attempt, move on to the next route |
| 09:20.20 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 09:20.30 | benjk | its less time than your user will be able to notice |
| 09:20.50 | JT | won't it have to time out for ip based comms? |
| 09:20.55 | JT | like a couple of seconds |
| 09:21.24 | benjk | and there is the risk that you then don't try anymore when it actually *is* available |
| 09:21.39 | sip_me | Hi, Can I config the Asterisk to not pass through RTP between two sip clients? (I would like the RTP to go directly between the SIP clients) |
| 09:22.14 | benjk | sip_me: canreinvite=yes |
| 09:22.18 | JT | that is true, such a scheme would need a way of periodically connectivity checking and resetting the status bit on service return |
| 09:22.38 | benjk | and how do you know "service return" has occurred |
| 09:22.42 | benjk | not worht the hassle |
| 09:22.55 | JT | heh |
| 09:22.55 | benjk | just dial all your routes one by one until the call succeeds |
| 09:23.11 | benjk | or until you have run out of routes |
| 09:23.43 | benjk | and like I said, ideally you wnat to write a macro that does this all in one place, don't hardcode it |
| 09:23.52 | JT | indeed |
| 09:23.52 | sip_me | benjk, will try. Tnx. |
| 09:24.02 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 09:25.10 | benjk | its a good idea though to have separate macros for determine the route and dialing a route |
| 09:25.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:25.35 | benjk | s/determine/determining |
| 09:25.41 | JT | hmm |
| 09:31.36 | benjk | of course it doesn't need to be macros, you could use AGIs and write the stuff in a language of your choice |
| 09:33.36 | JT | macros are in theory lower overhead though, yes? |
| 09:33.52 | benjk | it depends |
| 09:33.57 | benjk | in principle yes |
| 09:34.28 | *** join/#asterisk Yanie (n=Dave@82-69-4-3.dsl.in-addr.zen.co.uk) |
| 09:34.36 | benjk | but if you do lots of expensive operations, like large loops within loops and things like that, then AGIs may be more efficient |
| 09:35.15 | benjk | the most efficient is to write an app_ |
| 09:35.59 | benjk | looks cleaner in the dialplan too |
| 09:36.20 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:36.37 | benjk | exten _X.,1,SetRouteFor(${EXTEN}, ...) |
| 09:37.02 | JT | does an app need to be written in c? |
| 09:37.04 | benjk | exten _X.,2,DialACR(${ROUTE}, ...) |
| 09:37.10 | benjk | yes |
| 09:37.19 | JT | well that's a problem for me :/ |
| 09:37.25 | benjk | its less troublesome to write an app than it seems though |
| 09:37.40 | benjk | you can use the app_skeleton.c source that comes with asterisk |
| 09:37.48 | benjk | or is it app_skel.c |
| 09:38.02 | JT | hmm |
| 09:38.07 | benjk | and then look at other apps how various things are done |
| 09:38.19 | benjk | most of the time you can do plenty of stuff just by cut and paste |
| 09:38.41 | benjk | and this way you learn how to write apps, not just assembling them |
| 09:39.44 | benjk | but then again, AGI is probably fine |
| 09:39.55 | *** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com) |
| 09:40.11 | phearless | ARG! |
| 09:40.20 | phearless | hello |
| 09:40.26 | benjk | ${ARG!} |
| 09:40.26 | phearless | I am confused |
| 09:40.35 | benjk | ${ARG1} |
| 09:40.52 | phearless | the firmware for my cisco voip phone is a zip file P0S3-08-2-00.zip |
| 09:41.03 | phearless | but it contains P003-08-2-00.bin |
| 09:41.10 | benjk | so? |
| 09:41.19 | phearless | the S (for SIP) is not in the firware file name |
| 09:41.52 | benjk | not all of them have S in them |
| 09:42.22 | phearless | i have read that 0 is for the "skinny cisco protocol" and S if for SIP |
| 09:42.35 | phearless | the third character |
| 09:42.37 | benjk | up to a certain point in time yes |
| 09:42.58 | benjk | but Cisco introduced a universal boot loader firmware |
| 09:43.11 | phearless | ah ? interesting |
| 09:43.27 | benjk | I don't recall exactly at which version that happened though |
| 09:44.22 | JT | benjk: i guess how i go about doing it is just another thing i will have to ponder |
| 09:44.33 | benjk | indeed |
| 09:45.08 | phearless | so if I want to use the last fw, I can use the CISCO universal application loader ? |
| 09:45.34 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 09:47.50 | benjk | I think so, yes |
| 09:48.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 09:49.07 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:00.45 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 10:10.57 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:11.40 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 10:13.10 | delmar | I'm running Asterisk 1.2.10 and Zaptel 1.2.7 with a TDM400. call comes in and rings phones... if you answer the call, or if it goes to voicemail... Segfault and Asterisk is dead. calls via IAX coming in.. no problem. (not sure about it going to voicemail) |
| 10:14.08 | delmar | The system is currently being checked for hardware issues such as memory, then I will test more.. but.. anyone have any thoughts? |
| 10:15.19 | fourcheeze | delmar: was it working fine up to now? |
| 10:16.16 | darviria | anyone had any experience updating the firmware on a Zyxel P2000W? I'm trying it now but it's just hanging at 99%, even after a very long time, then the web config goes to a page saying "(firmware filename) saved to memory" but the phone still says 99%. Does it normally take hours? |
| 10:17.44 | *** join/#asterisk Skymarshal (n=Skymarsc@wlan-157.uni-koblenz.de) |
| 10:17.58 | creativx | them phones.. |
| 10:18.01 | creativx | so tender |
| 10:19.14 | darviria | I tried with konqueror and opera but they wouldnt work, IE looks like its doing something more |
| 10:19.42 | Skymarshal | I try to log into the Asterisk Manager but it allways tells me "Message: Authentication failed". The values in /etc/asterisk/manager.conf seems to be ok. What else could it be? |
| 10:19.55 | E-bola | wrong channel then |
| 10:20.32 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-35.range217-44.btcentralplus.com) |
| 10:25.41 | delmar | fourcheeze, I started with FreePBX... and had no end if issues... I decided to go back to what I know works... a fresh Asterisk... and now perhaps I see why FreePBX wasnt working... im going to roll back to Zaptel 1.2.5 and Asterisk 1.2.7.1 |
| 10:27.01 | fourcheeze | delmar: sounds like a plan |
| 10:30.53 | E-bola | Is there a guide on upgrading from 1.2.9 to 1.2.10? |
| 10:31.57 | delmar | fourcheeze, nah i havent had the tdm400 in this machine long... we have had calls working ok... intermittantly... when i had FreePBX installed... i moved the asterisk configs to one side.. installed samples... then added some real basic vanilla dialplan stuff to make it ring phones when FXO call comes in... and right away.. Segfaults with the call is answered. |
| 10:32.00 | delmar | so thats a bit sus. |
| 10:32.33 | fourcheeze | delmar: anything in dmesg ? |
| 10:32.48 | delmar | ooo |
| 10:32.50 | delmar | ill check |
| 10:37.04 | *** join/#asterisk Tristan_m (n=Tristan@LNeuilly-152-23-33-115.w193-252.abo.wanadoo.fr) |
| 10:39.45 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 10:41.40 | rg1_ | I am starting to maintain some Asterisk scripts for a person who has left our organization - have a quick question regarding WaitForSilence() |
| 10:41.59 | rg1_ | exten => s,n,MixMonitor(${CALLFILENAME}.wav) |
| 10:42.12 | rg1_ | exten => s,n,WaitForSilence(${TQM_SILENCE}) |
| 10:42.55 | rg1_ | I know that MixMonitor() records the channel - does it just kick that off and then continue through the script? Or does it stay in MixMonitor() till the recording stops? |
| 10:43.22 | Tristan_m | MixMonitor records only when a channel is bridge |
| 10:43.24 | Tristan_m | bridged |
| 10:43.31 | Tristan_m | it stops upon bridge completion |
| 10:43.38 | rg1_ | Reason for asking is that the WaitForSilence() is AFTER that MixMonitor() call; |
| 10:43.53 | rg1_ | so when would the WaitForSilence() get executed? |
| 10:43.55 | Tristan_m | but be carefull with that app ... as it may cause asterisk to crash |
| 10:44.17 | *** join/#asterisk djtremors (n=newjacks@ppp121-90.static.internode.on.net) |
| 10:44.23 | djtremors | hey all |
| 10:44.32 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 10:44.37 | rg1_ | Tristan - I would like to use "Record()" but if the user hangs up in the middle of the recording, it doesn't retain the recorded file. At least that is a comment in the script by this guy. |
| 10:45.31 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 10:46.00 | Tristan_m | What do you want to do with this script ? |
| 10:46.03 | Tristan_m | records calls ? |
| 10:46.10 | Tristan_m | or just an user record ? |
| 10:46.32 | Tristan_m | because waitforsilence is usefull only if you want answering machine detection |
| 10:46.49 | Tristan_m | or wants to playback a sound file after the user stops talking |
| 10:47.54 | Tristan_m | rg1_, that's strange as the records should stop if the channel is hung up, but the file should stay in spooldir |
| 10:48.07 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 10:50.46 | jmls | anyone having problems recording calls with local channels and transfers ? (trunk) |
| 10:51.01 | jmls | Ours stops as soon as they are transferred |
| 10:52.41 | jmls | using /n causes some other wierd problems (see #7671) |
| 10:52.46 | Tristan_m | works good under 1.2svn |
| 10:53.23 | jmls | thanks - but we are using trunk because we are making heavy use of res_jabber in the dialplan |
| 10:53.39 | Tristan_m | by issuing a stopmonitor in the transfer context and reusing monitor a few steps later |
| 10:53.47 | Tristan_m | did you tried ? |
| 10:54.29 | jmls | the PHB demands that all calls are recorded as a single file |
| 10:54.44 | jmls | It worked fine in 1.2.xx (but we didn't use local channels then) |
| 10:54.55 | jmls | we now use local channels for dynamic queue members |
| 10:55.19 | jmls | and have to use local in order to post jabber messages at the appropriate time (after call end etc) |
| 10:55.21 | *** join/#asterisk X-Rob (n=rob@dsl-220-235-95-115.vic.westnet.com.au) |
| 10:55.59 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 10:56.06 | Tristan_m | just try to add the next conversation to the file |
| 10:56.14 | Tristan_m | you were using before the transfer |
| 10:56.33 | rg1_ | Tristan_m - I wanting to have someone record and then if they pause, I will ask them if they want to end the recording or continue the recording |
| 10:56.49 | djtremors | can anyone help me in solving my dtmf issue.. 1 server i built with trixbox doesn't respond to dtmf when called between 2 * machines in a iax trunk |
| 10:57.11 | djtremors | dtmfmode=rfc2833 is set.. |
| 10:57.17 | jmls | can you do that wth monitor ? I know you can with mixmonitor, but that causes segfaults with a simple dialplan (see #7411) so we can't use that |
| 10:57.34 | Tristan_m | I know about mixmonitor |
| 10:57.43 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:57.57 | rg1_ | Tristan - does what i just said make sense? |
| 10:59.47 | jmls | Tristan: Sorry, what do you mean by "transfer context" ? We use 7960's and the users use (!) the transfer button. As far as I was aware, this simply goes to the context specified in the sip.conf file. Is that what you mean ? |
| 11:01.51 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 11:03.20 | Tristan_m | jmls, each transfer is done through a specific context you can define ;) |
| 11:03.29 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 11:03.42 | jmls | cool. didn't know that |
| 11:03.54 | jmls | how and where is that setting defined ? |
| 11:04.12 | Tristan_m | see features.conf for example |
| 11:07.38 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 11:10.12 | phearless | do I have to BUY the SIP firmware for the cisco 7960 ? |
| 11:11.37 | jmls | doesn't features.conf mean that I have to map a dtmf sequence (like #1) ? There is no way that I can make the little buggers remember that instead of pressing the transfer button on the 7960 |
| 11:16.53 | DrkShdw | phearless: the firmware updates for the 7960 are available with a support contract from cisco |
| 11:17.00 | delmar | ok this is really anoying. i just tested a pure IAX call to the * box.. the instant it hit voicemail... Segfault |
| 11:17.06 | delmar | im getting so angry with this crap. |
| 11:17.19 | phearless | ok DrkShdw , how much it is ? I heard about 8$ |
| 11:17.59 | DrkShdw | phearless: no clue. I avoid cisco phones, because I don't think I need a support contract for a device that everyone in my office has used since the age of 5 |
| 11:18.13 | phearless | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 |
| 11:18.18 | phearless | here is is *free* |
| 11:18.28 | phearless | "SIP Flash Image for 7940/7960 IP Phone v8.2(0)- Non-CallManager" |
| 11:18.42 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 11:18.46 | phearless | but I am not sure if it is the good thing |
| 11:19.14 | fourcheeze | delmar: nothing in dmesg? |
| 11:19.36 | delmar | fourcheeze, nothing. |
| 11:19.46 | delmar | fourcheeze, this isnt a tdm problem anymore |
| 11:19.54 | fourcheeze | yeah |
| 11:19.59 | fourcheeze | still weird |
| 11:20.05 | delmar | Asterisk is crashing when calls are being answered... voicemail or otherwise |
| 11:20.08 | phearless | it contains for example : |
| 11:20.13 | delmar | useless piece of shit |
| 11:20.26 | phearless | 767K P0S3-08-2-00.sb2 |
| 11:20.26 | fourcheeze | tried straceing * ? |
| 11:20.26 | DrkShdw | phearless: did you actually try to download the file? it asks for your support contract username/pass |
| 11:20.39 | phearless | you can use anonymous:x@x.com |
| 11:20.42 | delmar | fourcheeze, im far from interested in programmer nonsense. im not a programmer... never been interested. |
| 11:21.17 | fourcheeze | delmar: yeah, me too, it's just that it might be some resource unavailable |
| 11:21.44 | delmar | fourcheeze, tell me how to use it and ill see :P |
| 11:22.32 | fourcheeze | delmar: stop * and do something like: |
| 11:22.33 | fourcheeze | strace asterisk -vvvvvvvvvvc |
| 11:22.55 | fourcheeze | better still |
| 11:23.05 | fourcheeze | <PROTECTED> |
| 11:23.37 | fourcheeze | see what the last thing was that it did before segfault |
| 11:26.22 | delmar | yeah i can see stuff but.. nothing standing out that means anything useful |
| 11:26.29 | *** join/#asterisk DoDaT69 (n=DoDaT69@66.240.110.114) |
| 11:27.12 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:27.15 | backblue | * |
| 11:27.28 | delmar | sigreturn() = ? (mask now [RTMIN]) |
| 11:27.29 | delmar | read(0, 0xbffff2db, 1) = ? ERESTARTSYS (To be restarted) |
| 11:27.29 | delmar | --- SIGSEGV (Segmentation fault) @ 0 (0) --- |
| 11:31.02 | tuxick | nice |
| 11:31.31 | tuxick | maybe try strace -f |
| 11:32.00 | tuxick | dunno asterisk architecture though |
| 11:34.47 | delmar | hrm |
| 11:34.53 | delmar | im developing a theory |
| 11:34.56 | delmar | get this guys... |
| 11:35.26 | delmar | when a call comes in via TDM... and its answered by a sip phone or voicemail... we get a Segfault... |
| 11:35.56 | delmar | when a call comes in via IAX (g729) .. the call is answered by a sip ohone... works great... if it hits voicemail... Seg fault |
| 11:36.25 | backblue | what does core dumped says? |
| 11:37.19 | delmar | hrm. |
| 11:37.33 | delmar | used to be that i would find a coredump in the dir where i exited to |
| 11:37.46 | delmar | its a Segfault without a coredump i suspect |
| 11:37.55 | MrChimpy | check ulimit |
| 11:38.02 | delmar | MrChimpy, ? |
| 11:38.37 | MrChimpy | ulimit -c unlimited |
| 11:39.51 | delmar | whats that going to acheive? |
| 11:40.15 | champster | do you have g.729 licenses??? |
| 11:40.40 | delmar | are u kidding? they STOLE it in the first place. NO |
| 11:40.52 | delmar | has nothing to do with that |
| 11:41.00 | MrChimpy | delmar: um, it might get you coredumps |
| 11:41.35 | delmar | i tried switching to gsm only but that was on the later version of * havent tried gsm on this earlier one. brb |
| 11:42.17 | champster | Also, try "show translation" in the cli and see if that crashes it. |
| 11:42.31 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 11:42.36 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 11:44.25 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-35.range217-44.btcentralplus.com) |
| 11:44.37 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 11:46.23 | delmar | no that doesnt crash it... |
| 11:46.26 | delmar | but i tell u waht.. |
| 11:46.33 | delmar | i had been developing a theory... |
| 11:46.40 | delmar | that is was a translation related issue |
| 11:46.51 | delmar | im glad i tried gsm only... that seems to work |
| 11:47.00 | delmar | ill try a few more |
| 11:47.18 | backblue | how are you running asterisk? |
| 11:47.23 | delmar | i wonder if this will work via the FXO tho.. that was the main issue |
| 11:47.33 | backblue | cores are placed in /etc/asterisk by default |
| 11:47.38 | delmar | backblue, as root asterisk -cvvvvvvvvvvvvvvvvvvv(lots and lots) |
| 11:47.50 | backblue | run in deamon |
| 11:47.55 | MrChimpy | root probably won't have ulimits set |
| 11:47.59 | backblue | and check if it cores dumps |
| 11:48.29 | delmar | ok. thanks for all the suggestions guys. i appreciate it. |
| 11:48.31 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 11:49.00 | delmar | backblue, run as daemon u mean.. just run it without -c |
| 11:49.40 | *** join/#asterisk sumasuma (n=chumma@cm233.omega181.maxonline.com.sg) |
| 11:50.37 | delmar | when i run it as background daemon just by typing "asterisk" .. is it normal for it to spawn like.. 17 processes? |
| 11:54.37 | Juggie | no |
| 11:56.20 | *** join/#asterisk EmileC (n=EmileC@beechjet.clarotech.co.za) |
| 11:57.42 | EmileC | hi, i have a gereral * issue and would appreciate some help. I have a fresh load of Asterisk on SUSE with a Sirrix card hooked up to ISDN. I can make/receive calls quite happily. My problem seems to be with the playback of sounds. i.e. voicemail queue prompts etc |
| 11:57.51 | EmileC | i can see them being called but i don't hear the voicemail prompt |
| 11:58.08 | EmileC | it looks similar to when a PRI goes down and Asterisk can no longer get any clocking |
| 11:58.25 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 11:58.33 | CMike | hi all |
| 12:01.15 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 12:02.05 | my007ms | how can i upgrade to asterisk 1.2.10 |
| 12:02.23 | my007ms | i need just upgrade without loss config |
| 12:02.27 | MrChimpy | by downloading and installing it? |
| 12:02.30 | MrChimpy | yes |
| 12:03.01 | MrChimpy | obviously back up config beforehand |
| 12:03.36 | my007ms | and what bit cdr and mysql support that i have in now asterisk |
| 12:11.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 12:11.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 12:12.51 | sumasuma | when i send any messages from the agi i'm not getting on the asterisk console |
| 12:13.01 | sumasuma | anyone please help me |
| 12:13.14 | sumasuma | $err=fopen("php://stderr","w"); |
| 12:13.22 | sumasuma | fprintf($err,"Entering asterisk agi<br>"); |
| 12:13.34 | sumasuma | this is what i'm asking to print |
| 12:15.44 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:15.45 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:18.08 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:19.18 | MrChimpy | susasuma: agi debug on? i think that's the command. |
| 12:19.28 | ionix | <BR> ?? ;) |
| 12:19.29 | MrChimpy | that or agi debug on its own |
| 12:19.39 | sumasuma | it is on |
| 12:19.45 | sumasuma | but still i could not get the messages |
| 12:19.50 | ionix | try \n instead of <BR> |
| 12:19.54 | MrChimpy | turn buffering off on PHP stderr? |
| 12:20.07 | sumasuma | new line is not a problem |
| 12:20.12 | sumasuma | there is no text at all |
| 12:20.32 | MrChimpy | could be... php might not flush until a newline. |
| 12:20.54 | MrChimpy | IMHO you're lucky anything works in PHP anyway ;) |
| 12:21.20 | ionix | php has a framework for asterisk |
| 12:21.25 | ionix | even a module, I suggest you use it. |
| 12:26.02 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 12:26.58 | rg1_ | Is it possible to stop the recording done by Monitor() by having the user hit the "*" (star) key? |
| 12:28.07 | my007ms | what well happen if i download and install asterisk-1.2.10 over my asterisk-1.2.9 |
| 12:28.23 | my007ms | it's asterisk still can read it's config from mysql |
| 12:30.42 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 12:31.49 | DoDaT69 | <PROTECTED> |
| 12:31.49 | DoDaT69 | <PROTECTED> |
| 12:31.58 | DoDaT69 | that keeps displaying over and over |
| 12:32.26 | my007ms | it's like u use FreePBX |
| 12:32.38 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:32.44 | DoDaT69 | yes.. I installed it |
| 12:32.47 | DoDaT69 | I will ask in there |
| 12:36.40 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 12:38.16 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 12:38.41 | *** join/#asterisk drega (n=drega@mail.amcat.co.uk) |
| 12:38.54 | *** join/#asterisk cfassoni (n=cfassoni@200.214.115.29) |
| 12:39.05 | drega | I want to record all calls coming through asterisk. I'm sure im missing something simple. |
| 12:39.06 | Winkie | hey guys, i'm having a weird error after upgrading asterisk, any outbound calls i make via ZAP are immediately hung up and i'm getting an invalid number format error return from our provider |
| 12:39.06 | Winkie | i've redone zapata.conf and i can't see anything immediately wrong with it, have there been any changes recently that are known to cause things like this? |
| 12:39.14 | *** part/#asterisk cfassoni (n=cfassoni@200.214.115.29) |
| 12:39.19 | drega | anyone able to point me in the right direction? |
| 12:40.08 | alexcf | record all calls? |
| 12:40.09 | alexcf | hmm |
| 12:40.26 | alexcf | wexten => _0X.,2,SetVar(CALLFILENAME=${TIMESTAMP}-${CALLERIDNUM}-${DNID}) |
| 12:40.26 | alexcf | exten => _0X.,3,Monitor(wav,${CALLFILENAME},m) |
| 12:40.30 | alexcf | i've got that configured |
| 12:40.40 | alexcf | (drop the w) |
| 12:41.05 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 12:43.24 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 12:45.37 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 12:46.43 | *** join/#asterisk Tili (n=tili@202.133.67.214) |
| 12:49.16 | hwt | alexcf: where will the files end up? |
| 12:49.29 | alexcf | umm |
| 12:50.05 | hwt | apparently /var/spool/asterisk/monitor |
| 12:50.09 | alexcf | yep |
| 12:50.16 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 12:50.33 | Winkie | well I sorted mine with pridialplan=local |
| 12:50.33 | Winkie | good old comedy hardware |
| 12:50.55 | Mike | hi, anyone knows how to force hangup cause using SIP? |
| 12:51.16 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:52.33 | Winkie | force hangup cause? you mean pretend it was hung up because of a particular reason? |
| 12:52.42 | Winkie | or you mean just force a channel to disconnect? |
| 12:54.54 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 12:56.48 | zedkatuf | anyone able to help me set up a simpl incoming context in extensions.conf? Here's what I've got so far: |
| 12:56.51 | zedkatuf | http://pastebin.ca/120604 |
| 12:56.59 | zedkatuf | (my extensions.conf) |
| 12:57.12 | *** join/#asterisk svenna (n=svenna@p548D3C58.dip0.t-ipconnect.de) |
| 12:57.44 | zedkatuf | ...the incoming bit doesn't work..in the sense that *CLI sees nothing |
| 12:58.01 | zedkatuf | I've got all the appropriate ports forwarding to the * box |
| 12:58.04 | jmls | anyone having problems recording calls with local channels and transfers ? (with svn trunk) |
| 12:58.12 | zedkatuf | (from my firewall) |
| 12:58.19 | jmls | Ours stops as soon as they are transferred. using /n causes some other wierd problems (see #7671) |
| 12:59.26 | [TK]D-Fender | zedkatuf: pastebin your sip.conf |
| 12:59.27 | [TK]D-Fender | ~pb |
| 12:59.30 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 13:01.10 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-35.range217-44.btcentralplus.com) |
| 13:06.52 | zedkatuf | [TK]D-Fender: http://pastebin.ca/120614 (I've cut out the register stuff which I know works because I can do outgoing calls OK) |
| 13:06.59 | zedkatuf | going AFK now |
| 13:07.06 | *** join/#asterisk daysmen3 (n=primus@host217-44-108-35.range217-44.btcentralplus.com) |
| 13:07.47 | zedkatuf | actually, still here for a bit! |
| 13:08.15 | *** join/#asterisk ToyMan (n=stuq@74-32-31-237.dsl1.mdl.ny.frontiernet.net) |
| 13:08.52 | ToyMan | morning all |
| 13:08.59 | l-fy | hi ToyMan |
| 13:11.19 | *** join/#asterisk adorah (n=Administ@84.94.208.77.cable.012.net.il) |
| 13:11.41 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 13:12.52 | [TK]D-Fender | zedkatuf: I need your [general] section.... |
| 13:15.08 | *** join/#asterisk clive- (n=pirch@dsl-165-172-230.telkomadsl.co.za) |
| 13:17.15 | [TK]D-Fender | zedkatuf: Aside from the fact that you aren't supposed to use "s" for incoming SIP calls.... the targt # is known... |
| 13:17.16 | *** join/#asterisk B2382F29 (n=daniel@dslb-084-058-229-107.pools.arcor-ip.net) |
| 13:17.50 | zedkatuf | [TK]D-Fender: OK, what should I use for sip? |
| 13:18.03 | B2382F29 | Hello, everyboby. Is this the right channel to ask about a misdn problem in asterisk? (DTMF recognition) |
| 13:18.56 | my007ms | hello what is wrong there http://pastebin.ca/120629 |
| 13:19.35 | my007ms | i config 35 and he stop even in channel 22 |
| 13:20.19 | hmmhesays | wiki wiki wild wild |
| 13:21.21 | queuetue | file: Are you around? Wondering if you recall where you rebuilt * from, so I can rebuilt it with your patch if necessary. |
| 13:24.19 | *** join/#asterisk livesNbox (n=chadkous@user-12l2mv6.cable.mindspring.com) |
| 13:24.40 | livesNbox | hey guys -- is it possible to force an extension to dial another extension as soon as it's off hook. |
| 13:24.55 | livesNbox | like -- a phone with no buttons that is just used to dial customer service or something |
| 13:25.20 | Nivex | livesNbox: you'd have to get the phone itself to do it. |
| 13:25.58 | livesNbox | ok.. so it would be a function of the phone itself.. anyone know a good phone that will do that ? |
| 13:26.03 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:26.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:26.05 | my007ms | http://pastebin.ca/120629 did any one have idea what is wrong |
| 13:26.41 | *** join/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 13:28.18 | [TK]D-Fender | zedkatuf: and exten for your inbound DID based on your register line in sip.conf (whic I still need to see the rest of) |
| 13:30.49 | hmmhesays | anthm, was it you that did the work on hint priorities in the asterisk dialplan? |
| 13:31.31 | anthm | nope one of the few things in asterisk i have no idea about |
| 13:32.21 | hmmhesays | ahh ok, someone mentioned friday you had something to do with that |
| 13:33.09 | *** join/#asterisk eivindtr (n=wingnut-@85.221.124.2) |
| 13:33.14 | hmmhesays | how was cluecon this year? |
| 13:33.42 | anthm | it was tough to run but wen well |
| 13:34.09 | anthm | it was for sure a notch up from the last one which is all i can hope for |
| 13:34.41 | anthm | we will have all the material on the site soon |
| 13:34.54 | anthm | video clips of all the presentations |
| 13:35.11 | clive- | does anyone know why the cdr-csv seems to be confused by the "congestion" message |
| 13:35.16 | my007ms | can someone help pleas http://pastebin.ca/120629 |
| 13:35.36 | hmmhesays | good deal |
| 13:35.50 | hmmhesays | you just going to have clips, or the whole presentations? |
| 13:36.19 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 13:37.21 | anthm | well by clips i guess i meant cliped at the right spots between them |
| 13:37.35 | anthm | we have 20 hours of video from 3 cameras |
| 13:37.43 | hmmhesays | <chuckle> ok |
| 13:37.51 | anthm | one was jacked into the mic so we have to sync it all etc |
| 13:37.52 | hmmhesays | I look forward to viewing them |
| 13:38.31 | anthm | we will probably have dvd iso and put it on bit torrent and then let you download per speaker off our site or somethign |
| 13:39.12 | hmmhesays | that works |
| 13:39.25 | hmmhesays | you have your own tracker? |
| 13:39.46 | anthm | for? |
| 13:39.57 | hmmhesays | bittorrent |
| 13:40.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:41.13 | *** join/#asterisk Lexapro (n=Lexapro@1Cust648.an3.chi30.da.uu.net) |
| 13:41.20 | Mike | Winkie, some routers expect to get a hangup cause like circuits busy i think its code 32 or no route code 3 |
| 13:41.21 | Lexapro | Top the morning |
| 13:41.54 | *** part/#asterisk Lexapro (n=Lexapro@1Cust648.an3.chi30.da.uu.net) |
| 13:45.06 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 13:45.15 | backblue | anyone knows how this sipbroker it's suposed to work? |
| 13:46.19 | fourcheeze | my007ms: can't seem to get to your pastebin |
| 13:46.35 | backblue | pastebin it's broken |
| 13:46.58 | fourcheeze | certainly starting to smell that way |
| 13:47.10 | backblue | it's broken a couple of days |
| 13:53.35 | hmmhesays | get a different provider? |
| 13:54.05 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:56.11 | *** join/#asterisk eivindtr (n=wingnut-@85.221.124.2) |
| 13:59.23 | my007ms | fourcheeze, thanks much |
| 13:59.26 | my007ms | i solve it |
| 14:01.01 | *** join/#asterisk cbrake (n=cbrake@cpe-65-25-51-117.neo.res.rr.com) |
| 14:01.16 | *** join/#asterisk hohum (n=dcorbe@12.195.58.238) |
| 14:02.06 | cbrake | so what are some favorite IP phones? Looking for something in the $100-$200 range. Thinking about the linksys SPA-922 ... |
| 14:02.40 | hmmhesays | good phone |
| 14:03.55 | l-fy | spa RULEZ |
| 14:03.58 | l-fy | sorry |
| 14:04.04 | l-fy | hey folder |
| 14:04.07 | l-fy | hey fourcheeze |
| 14:04.17 | fourcheeze | hey l-fy |
| 14:04.21 | hmmhesays | I can't get my sla to work though |
| 14:04.24 | hmmhesays | blargh |
| 14:05.25 | sevard | push me |
| 14:06.11 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:06.20 | sevard | your foot tickels my balls of steel |
| 14:06.34 | hmmhesays | I need a decent incoming service provider for a FARGO did |
| 14:06.53 | sevard | one fargo DID? |
| 14:06.55 | hmmhesays | yes |
| 14:06.56 | bkw_ | Fargo... wasn't that a lame movie? |
| 14:06.59 | hmmhesays | yes |
| 14:07.05 | hmmhesays | yes it was |
| 14:07.10 | sevard | bkw_: you know people don't actually talk like that |
| 14:07.22 | hmmhesays | yeah doncha knoow |
| 14:07.30 | MrChimpy | eh? |
| 14:07.43 | sevard | no idea where that shit came from, i've never met a single person in my entire life that talks like that |
| 14:07.59 | MrChimpy | emo phillips comes quite close |
| 14:08.04 | *** join/#asterisk eivindtr (n=wingnut-@85.221.124.2) |
| 14:08.18 | *** join/#asterisk backblue (n=backblue@suporte.3gnt.net) |
| 14:08.25 | hmmhesays | comes from the 1st and 2nd generation scandanavians that live here |
| 14:08.30 | hmmhesays | most of them are dead now though |
| 14:08.41 | sevard | hmmhesays: finish E17 |
| 14:08.56 | *** join/#asterisk lehel (n=t@82.77.41.206) |
| 14:10.47 | GerbilWrk | anyone know how to send variables from one context to another? like in a goto cmd? |
| 14:10.57 | hmmhesays | so service provider ideas anyone? i haven't had to buy an IP DID in long time |
| 14:12.18 | hmmhesays | they should follow your call |
| 14:13.02 | ionix | www.iax.cc |
| 14:13.16 | GerbilWrk | so differen't contexts won't cause them to disappear? |
| 14:13.39 | mocker | cbrake: I like my SPA-942 |
| 14:13.43 | hmmhesays | GerbilWrk: i hope not otherwise I got a lot of dialplan fixing to do |
| 14:13.58 | hmmhesays | yeah sixtel just merged with someone |
| 14:14.06 | GerbilWrk | k, I read something about conext specific and global variables, so i had to make sure :) |
| 14:14.08 | GerbilWrk | thanks |
| 14:14.22 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 14:14.59 | *** join/#asterisk networkjedi (n=networkj@12.36.212.60) |
| 14:15.54 | *** part/#asterisk networkjedi (n=networkj@12.36.212.60) |
| 14:16.02 | *** part/#asterisk backblue (n=backblue@suporte.3gnt.net) |
| 14:16.32 | *** join/#asterisk networkjedi (n=networkj@12.36.212.60) |
| 14:20.27 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 14:20.34 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 14:22.33 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 14:24.04 | *** join/#asterisk eivindtr (n=wingnut-@85.221.124.2) |
| 14:27.21 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 14:27.33 | hmmhesays | sixtel ain't bad |
| 14:28.02 | hmmhesays | $1.49/Month $0.0135/minute |
| 14:30.45 | queuetue | Software fax handling requires patching of asterisk? Or is the wiki out of date? |
| 14:33.08 | brookshire | queuetue: i think trunk will handle faxing without patching |
| 14:33.33 | *** join/#asterisk keith80403 (n=keith804@24-56-189-80.co.warpdriveonline.com) |
| 14:33.53 | queuetue | brookshire: Is trunk what I get when I download the "current" from asterisk.org? |
| 14:34.10 | brookshire | you can only download trunk from svn |
| 14:34.16 | brookshire | trunk is not stable |
| 14:34.18 | cbrake | mocker: how did you handle the POE part w/ the SPA-942? |
| 14:34.46 | cbrake | mocker: I currently don't have any POE infrastructure |
| 14:34.49 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 14:35.10 | queuetue | Gah. I had no idea how much work the A@H guys put into making asterisk usable. :) Switching to do-it-yourself is very tough. |
| 14:35.21 | bkw_ | not really |
| 14:35.34 | queuetue | Yes, really. |
| 14:35.56 | SeicherlBoB | Hi there! i have troubles installing asterisk on my ubuntu-server (dapper). make install throws an error like "gcc cant create executables". any ideas? |
| 14:35.59 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 14:36.02 | clive- | first time is maybe a bit tricky, but its not difficult |
| 14:36.28 | queuetue | Maybe not *difficult*, just labor intensive. |
| 14:37.10 | queuetue | SeicherlBoB: Can you make anything else? Sounds like a build toolchain issue, not a n asterisk one. |
| 14:37.26 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 14:37.36 | SeicherlBoB | queuetue: well, just set up the machine... dont know what i should make else. |
| 14:37.58 | SeicherlBoB | queuetue: going to be a test and training-server for asterisk |
| 14:38.41 | shmaltz | I"m getting errors when compiling asterisk output is here: |
| 14:38.42 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 14:38.42 | shmaltz | http://pastebin.ca/120703 |
| 14:39.38 | c4t3l | has anyone noticed the bug in 1.2.9 that fails to display the output of 'asterisk -rx "blah blah"'? |
| 14:39.50 | *** join/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com) |
| 14:39.55 | Cherebrum | Anyone use Xtunnels? |
| 14:40.33 | SeicherlBoB | shmaltz: mine doesnt work either, but i cant see an error in your output |
| 14:43.49 | queuetue | Is there any way to elave a |
| 14:43.50 | queuetue | real" fax machine inline with an asterisk port and have asterisk make the relationship work? I understand fax over voip, even on an internal LAN is going to be spotty? |
| 14:44.20 | *** part/#asterisk clive- (n=pirch@dsl-165-172-230.telkomadsl.co.za) |
| 14:44.30 | michael-i | anyone have a writeup on res_sqlite? I'm attempting to just use it for my extensions.conf but it want to also replace my sip.conf even though everything's commented out in res_sqlite.conf |
| 14:44.54 | clyrrad | shmaltz - you dont have awk installed |
| 14:45.33 | shmaltz | clyrrad, I do |
| 14:47.01 | shmaltz | clyrrad, interesting, it shows up but when I try running it I get this: |
| 14:47.03 | shmaltz | root@pbx:/usr/src/asterisk-1.2.6# /usr/bin/awk |
| 14:47.05 | shmaltz | -bash: /usr/bin/awk: No such file or directory |
| 14:47.14 | *** join/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 14:48.27 | SeicherlBoB | my install log tells that /usr/bin/ld: crt1.o: No such file: No such file or directory ... how can i fix that? |
| 14:48.56 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:51.15 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:51.30 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:52.53 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 14:54.14 | ManxPower | shmaltz, you need to install "awk" |
| 14:54.35 | shmaltz | ManxPower, I'm installing gawk, is that good enough? |
| 14:54.41 | ManxPower | shmaltz, no idea. |
| 14:55.22 | toerkeium | hello people |
| 14:55.51 | toerkeium | it's advisable to use astmanproxy when working with web based AMI applications? |
| 14:55.55 | mmealling | gah, how hard can it be to find just pc to phone adapter for linux? ghees.... |
| 14:57.29 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 14:57.44 | ph|ber | how do you raise the gain on outbound VOIP calls?? is that in zapata.conf?? |
| 14:57.56 | puppet | can anyone try call my guest iax? |
| 14:59.17 | puppet | just see if it works :) |
| 14:59.22 | hmmhesays | ahh sixtel, one of the few providers with DID's in FARGO |
| 14:59.22 | puppet | guest@home.puppet.nu |
| 14:59.31 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:00.55 | *** join/#asterisk backblue (n=backblue@suporte.3gnt.net) |
| 15:01.00 | ph|ber | anyone?? |
| 15:01.25 | hmmhesays | wiki wiki |
| 15:02.01 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 15:02.08 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 15:02.10 | backblue | ei ppl, there is some function in one asterisk modules, that i can insert and remove or something, in one database, like mysql? |
| 15:02.13 | Dr-Linux|work | hi guys |
| 15:02.16 | backblue | i have seen that |
| 15:02.38 | Dr-Linux|work | gonna have a new asterisk server, just need some suggestions |
| 15:03.09 | Dr-Linux|work | anyone can discuss? |
| 15:03.19 | mmealling | can anyone recommend a USB ATA that works with linux? |
| 15:03.24 | Supaplex | Dr-Linux|work: bankrupt qwest |
| 15:04.11 | Dr-Linux|work | Supaplex, what you just said? :S |
| 15:04.12 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:04.12 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:04.17 | StewLG | Help? http://pastebin.ca/120731 |
| 15:04.36 | Dr-Linux|work | i think mog would be able to help me |
| 15:04.39 | StewLG | Trying to get IAX set up for external calls; I finally have SIP working.. |
| 15:04.43 | Dr-Linux|work | mog, active ? |
| 15:04.47 | hmmhesays | what would be the accounting term for service estimate? |
| 15:05.31 | Dr-Linux|work | mog, we are going to have new asterisk server in USA, |
| 15:05.47 | Dr-Linux|work | i just need some information, everything is ready to go .. but still want to confirm |
| 15:05.47 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 15:06.10 | puppet | dr-linux|work: suggestions for what? |
| 15:06.35 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
| 15:06.54 | SeicherlBoB | can anybody give me a starting-point? I need a pbx that actually checks multible phonenumbers and zip-accounts to redirect them to 1 phone (or softphone) showing both calling parties on the client |
| 15:07.03 | Dr-Linux|work | mog, going to buy TE405P, 4 PRI's/96 fine? |
| 15:07.04 | jtexter3 | I have what I hope is a quick question. I'm writing a custom app for Asterisk. Part of this involves playing and stopping files. Is there a way to know at what position in the file a play stopped at? |
| 15:07.22 | sb_mx | StewLG, im not sure but maybe you dont have chan_iax2 module loaded. type CLI> show modules like iax |
| 15:07.29 | Dr-Linux|work | mog, server dual 3g , 1GB ram |
| 15:07.48 | Dr-Linux|work | mog, Asterisk new version |
| 15:08.01 | mog | yes |
| 15:08.39 | Dr-Linux|work | mog, anything else i need or you recommend? |
| 15:08.50 | mog | nodda |
| 15:08.53 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.63) |
| 15:08.59 | Dr-Linux|work | mog, i guess i need at least 2GB of RAM :S |
| 15:09.00 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:09.48 | Dr-Linux|work | nodda? :S |
| 15:10.00 | Supaplex | nodda = no |
| 15:10.56 | StewLG | sb_mx: Good guess. It might have been that I was following older docs, or maybe I just screwed up: I was saying "IAX" instead of "IAX2". |
| 15:11.04 | mog | nah |
| 15:11.09 | mog | astersik is not ram heavy |
| 15:11.16 | RoyK | IAX1 doesn't exist anymore |
| 15:11.19 | RoyK | IAX2 == IAX |
| 15:11.32 | *** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 15:11.53 | Dr-Linux|work | mog, yes but this server will have hight load |
| 15:12.10 | mog | and? |
| 15:12.13 | RoyK | what is 'high load'? |
| 15:12.16 | kend | Query: 400 analog phones. What should I patch them into? |
| 15:12.18 | Dr-Linux|work | mog, do i need anything esle? |
| 15:12.22 | Dr-Linux|work | like g729 etc? |
| 15:12.25 | mog | not really |
| 15:12.26 | GerbilWrk | if a user modifies their voicemail password from the one listed in voicemail.conf, does that get replaced with the default password after a reload? |
| 15:12.46 | RoyK | Dr-Linux|work: describe the server's load in detail, don't just say 'it'll be high load' |
| 15:13.05 | ph|ber | anyone know how to raise the gain on outgoing sip VOIP calls? |
| 15:13.10 | michael-i | maybe someone new is around :) anyone have any res_sqlite experience? I'm almost there but need a bit of help |
| 15:13.36 | Dr-Linux|work | RoyK, this server will have AGI IVR's, caller will do everything will asterisk IVR |
| 15:14.01 | sb_mx | StewLG, what was the output of the CLI command? |
| 15:15.45 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 15:15.50 | hmmhesays | I wonder if I could get a trixbox setup for 100+ users |
| 15:15.51 | jtexter3 | Okay, got it. I found the ast_tellstream API, which does exactly what I need |
| 15:15.51 | nortex | -dev |
| 15:16.09 | hmmhesays | I feel daring today |
| 15:16.25 | nortex | hmmhesays, yes you are. |
| 15:16.35 | hmmhesays | I bet it can be done |
| 15:16.46 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 15:16.48 | ionix | What provider is good to terminate calls? |
| 15:16.59 | zeedo | hmmhesays: only limited by the specs of the box, but they'd have to be fairly high |
| 15:17.12 | zeedo | depending on how much usage those 100+ threw at it |
| 15:17.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:18.59 | ionix | Anyone from Sixtel here? |
| 15:19.33 | hmmhesays | yeah, this is a dual 3.0ghz xeon with 2gig of ram raid 1 |
| 15:19.45 | nortex | ionix, I use exgn, which merged with sixtel and became someone new. |
| 15:20.05 | ionix | ok exgn, websitE? |
| 15:21.07 | hmmhesays | vitelity |
| 15:21.15 | hmmhesays | is what they became, I've had good service with them |
| 15:21.20 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:21.25 | ionix | yeh, seems like vitelity is not sixtel |
| 15:21.29 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:21.30 | hmmhesays | come to think of it, I forget one of my clients uses it |
| 15:21.36 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 15:21.37 | hmmhesays | *uses them |
| 15:21.57 | hmmhesays | their user portal says so |
| 15:22.11 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 15:22.20 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 15:22.42 | nortex | ionix, When sixtel merged they became vitelity, I have not had any problems with the new setup. |
| 15:22.43 | hmmhesays | and their iax host is *.vitelity.net |
| 15:23.17 | StewLG | http://pastebin.ca/120763. Help appreciated. I'm a novice, so hopefully, my questions are simple.. |
| 15:25.24 | nortex | StewLG, My guess is that the 8 needs to be stripped off |
| 15:25.45 | nortex | StewLG, So try dialing ${EXTEN:1} |
| 15:25.51 | StewLG | Nortex: Ok, will do. |
| 15:25.58 | StewLG | Forgot about that. |
| 15:26.00 | *** part/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com) |
| 15:27.04 | StewLG | Nortex: I think you were totally right about that, but I'm still getting the same error message, even with that fixed. |
| 15:27.36 | *** join/#asterisk tjtoocool (n=none@72-19-25-58.idstelcom.net) |
| 15:27.46 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 15:27.51 | StewLG | I did "iax2 show registry" and its empty, though. I do have a register string in iax.conf that was working on a prior install..? |
| 15:29.00 | *** join/#asterisk anthm (n=anthm@h460894fa.area4.spcsdns.net) |
| 15:29.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:29.50 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-166.usadatanet.com) |
| 15:30.08 | StewLG | Uh, maybe it's a problem with the loading of my IAX driver? http://pastebin.ca/120774 |
| 15:31.03 | hmmhesays | um |
| 15:31.34 | sb_mx | StewLG, Aug 7 01:26:33 ERROR[4594]: chan_iax2.c:8606 set_config: Unable to load config iax.conf |
| 15:31.39 | sb_mx | your config is not loading |
| 15:31.49 | hmmhesays | hence voxee missing |
| 15:32.01 | StewLG | Ok, so.. hmm. |
| 15:32.16 | StewLG | How do I figure out what's wrong with my config? Is there a way to get additional debugging on that? |
| 15:32.42 | sb_mx | does your config exist in /etc/asterisk (assuming your using the default and you didnt change anything) ? |
| 15:33.15 | StewLG | There is an iax.conf file in /etc/asterisk, and yes, I have been modifying it. |
| 15:34.30 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:34.53 | ManxPower | StewLG, Perhaps the userid asterisk is running as does not have permission to read that file. |
| 15:35.08 | StewLG | Manx: I'll check that. |
| 15:35.40 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
| 15:35.49 | hmmhesays | What does digium charge for an asterisk install these days |
| 15:36.04 | tjtoocool | I don't suppose anyone knows how to get the amplitude of an ast_frame? |
| 15:36.10 | file | hmmhesays: I think the prices are online |
| 15:36.32 | hmmhesays | but that means i'd have to alt-tab into firefox |
| 15:36.36 | mmealling | hmmmm..... is quicknet still 'alive'? |
| 15:36.50 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 15:36.56 | file | http://www.digium.com/en/services/custom/configuration.php |
| 15:37.55 | hmmhesays | thanks |
| 15:37.59 | hmmhesays | whoa, spendy buggers |
| 15:38.08 | hmmhesays | that is funny that the price goes up by users |
| 15:38.12 | hmmhesays | lol |
| 15:38.20 | okdo | hi |
| 15:38.26 | StewLG | Do I need to worry about that IAX timing interface warning? |
| 15:38.35 | okdo | anyone have any tips for tuning rxfax application with a PRI line to get cleaner more legible FAXes? |
| 15:38.38 | *** join/#asterisk visba (i=user@70.222.240.255) |
| 15:38.55 | coppice | okdo: fix your timing |
| 15:39.03 | ManxPower | I hate Tuscaloosa |
| 15:39.03 | backblue | StewLG: yes if you will use iax. |
| 15:39.05 | *** part/#asterisk visba (i=user@70.222.240.255) |
| 15:39.08 | RoyK | hmmhesays: is it really that hard? |
| 15:39.08 | *** part/#asterisk B2382F29 (n=daniel@dslb-084-058-229-107.pools.arcor-ip.net) |
| 15:39.27 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 15:39.28 | ManxPower | StewLG, A timing interface is only REQUIRED for IAX2 trunking |
| 15:39.41 | ManxPower | Of you don't do IAX2 trunking, then you should not have to worry about it. |
| 15:39.53 | coppice | ManxPower: and for great comedy :-) |
| 15:39.55 | RoyK | or meetme or moh |
| 15:39.58 | RoyK | ? |
| 15:40.13 | ManxPower | IAX2 timing is not required for MOH. |
| 15:40.24 | ManxPower | Zaptel timing is required for MeetMe. |
| 15:40.43 | hmmhesays | i've used zaprtc a few times |
| 15:40.44 | StewLG | Trunking == PBX <-> PBX interop? |
| 15:40.55 | ManxPower | StewLG, no. |
| 15:41.16 | ManxPower | IAX2 trunking is putting multiple calls between the same two servers into one UDP stream |
| 15:41.32 | RoyK | ...which is rather neat |
| 15:41.49 | okdo | coppice: can you elaborate .... ? I am not entirely sure how to measure, etc. |
| 15:41.55 | RoyK | but iax2 timing == zaptel timing - rightt? |
| 15:42.03 | SeicherlBoB | Is there an easy-to-start-with, webbased config tool to let asterisk connect to and handle several SIP-accounts? |
| 15:42.06 | mog | hey anyone know easy way to get in touch with steveK other than email? |
| 15:42.09 | StewLG | Manx: Oh, right. |
| 15:42.10 | ManxPower | You can usually assume that. |
| 15:42.11 | RoyK | SeicherlBoB: yes, vi |
| 15:42.23 | SeicherlBoB | hehe |
| 15:42.39 | *** join/#asterisk firstknight (n=firstkni@202.155.150.55) |
| 15:42.40 | SeicherlBoB | RoyK: anything webbased? |
| 15:42.45 | RoyK | several |
| 15:42.50 | RoyK | but i've never used them |
| 15:43.01 | SeicherlBoB | RoyK: anything for noobs? |
| 15:43.06 | ManxPower | SeicherlBoB, Try #freepbx |
| 15:43.07 | RoyK | ~book |
| 15:43.08 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:43.09 | RoyK | ~docs |
| 15:43.10 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:43.22 | RoyK | SeicherlBoB: start with the book |
| 15:43.24 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 15:43.36 | tuxick | that was pdf only, afair |
| 15:43.57 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 15:44.14 | tuxick | i'd love to get my hands on an html version of that book |
| 15:44.24 | caloi | morning all, anyone using a Cisco 7905 unit? I updated my Asterisk CVS and now it appears Asterisk isn't parsing the messages sent from the 7905 correctly, any thoughts? |
| 15:44.29 | RoyK | SeicherlBoB: anyway, #freepbx might be worth a try |
| 15:44.38 | eKo1 | caloi: downgrade |
| 15:45.11 | RoyK | caloi: upgrade to 1.2.10 or latest 1.2 from svn, make a dump, report |
| 15:45.12 | caloi | eko1 haven't does that before, and am a newbie when it comes to CVS, how do I proceed with a downgrade? |
| 15:45.31 | eKo1 | caloi: cvs isn't being used anymore by the way |
| 15:45.33 | RoyK | caloi: iirc cvs is no longer supported |
| 15:45.39 | RoyK | use subversion |
| 15:45.51 | *** part/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 15:46.50 | SeicherlBoB | RoyK: ok, i already got it... maybe i should start reading ;) |
| 15:47.25 | StewLG | Well, the reason I didn't have iax.conf was because I made a subversion mistake. It's working now! Thanks all! |
| 15:47.57 | caloi | I misspoke, Subversion is what I am using, not CVS |
| 15:49.26 | StewLG | Has anyone done any work on real-time, on the fly voice modification? Like pitch shifting? Assuming enough CPU horsepower, is it feasible? |
| 15:51.39 | caloi | If I follow the directions on asterisk.org/download and run svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 - that will give me the latest version correct? |
| 15:52.05 | caloi | should I run that command in my /usr/local/src directory where my original asterisk source files already are? or create a new dir? |
| 15:52.54 | eKo1 | caloi: You don't want to download from branches. You want to download from tags |
| 15:53.11 | eKo1 | unless you want unstable versions. |
| 15:54.02 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 15:54.24 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 15:55.01 | caloi | eKo1 thanks for helping, so I should download this: svn checkout http://svn.digium.com/view/asterisk/tags/1.2.9.1/ |
| 15:55.04 | *** join/#asterisk paryl (n=chatzill@209.236.78.59) |
| 15:55.05 | *** join/#asterisk roguebug (n=johnw2@dip-106-210.bras.dsl.breisnet.com) |
| 15:55.07 | roguebug | hi |
| 15:56.13 | *** part/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 15:56.16 | roguebug | a question about non-root: i see it's possible, but not the default, so i guess there must be a reason why most people wouldn't want to run it non-root. what is it? |
| 15:56.51 | paryl | is there any other option for silent monitoring, other than ZapBarge? I need to set up a way for people who are training to listen to live calls |
| 15:57.50 | roguebug | by non-root, i mean asterisk -U askterisk |
| 15:58.10 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:58.40 | queuetue | roguebug: For one, if someone manages to exploit a bug in asterisk, they don't have control of your system. |
| 15:58.54 | sip_me | -- Hi, I would like to get two SIP Clients do RTP to each other without ever passing via the PABX (not event at the begining of the call). Is this supported? |
| 15:58.57 | eKo1 | roguebug: because root is a dangerous user |
| 15:59.10 | roguebug | thx, i understand that |
| 15:59.26 | roguebug | i mean why is non-root NOT set as default? |
| 15:59.54 | eKo1 | sip_me: that is what the canreinvite option is for |
| 15:59.58 | roguebug | i see the advantages of non-root (i.e. the disadvantages of root) but no disadvantage to non-root |
| 16:00.24 | eKo1 | roguebug: that is a dev. decision and I can't speak on their behalf. |
| 16:00.43 | caloi | anyone know what im doing wrong when I try to check the following out of SVN ? svn checkout http://svn.digium.com/asterisk/tags/1.2.10/ asterisk-1.2.10 |
| 16:00.53 | ManxPower | roguebug, It's a hassle to run as non-root. |
| 16:01.10 | eKo1 | caloi: looks fine |
| 16:01.27 | caloi | here is what I get though svn: PROPFIND request failed on '/asterisk/tags/1.2.10' |
| 16:01.28 | caloi | svn: PROPFIND of '/asterisk/tags/1.2.10': 405 Method Not Allowed (http://svn.digium.com) |
| 16:01.57 | ManxPower | caloi, Perhaps you have a broken proxy server |
| 16:02.23 | caloi | ManxPower - i'm not going through a proxy |
| 16:04.23 | roguebug | ManxPower: ok thx. is there any other reason or do you believe it's the only one (though a big one) ? |
| 16:04.46 | sip_me | eK01, canreinvite=no leaves the two clients talking via the PABX |
| 16:05.12 | sip_me | eK01, canreinvite=yes starts by working via the PABX and than reinvite them to talk directly to eahc other |
| 16:05.24 | sip_me | Any other option? |
| 16:06.04 | RoyK | sip_me: iirc that is almost true. the call is still held by the pbx, but the rtp stream goes directly client to client unless nat is in place where reinvites usually fuck up things |
| 16:06.05 | ManxPower | sip_me, If you want the two client to talk directly to each other without ever going thru the PBX, then you must do that via the client. |
| 16:06.40 | ManxPower | However, that will NOT work via NAT |
| 16:06.57 | sip_me | How? |
| 16:07.05 | sip_me | How can I do it via the client? |
| 16:07.15 | ManxPower | sip_me, that would depend on the client. |
| 16:07.19 | *** part/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 16:07.26 | sip_me | You mean to call direct? |
| 16:07.31 | ManxPower | This is not "All SIP client support channel" |
| 16:07.34 | ManxPower | sip_me, correct. |
| 16:07.35 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 16:07.53 | ManxPower | IF you want calls to never touch Asterisk, then this is NOT the place to get help. |
| 16:07.59 | *** join/#asterisk tempest1 (n=asf@adsl-150-207-143.chs.bellsouth.net) |
| 16:08.01 | mmealling | hmm.... looks like I'll just have to get an IAXy |
| 16:08.15 | ManxPower | sip_me, What I'm wondering is WHY you want the calls to never touch Asterisk. |
| 16:08.30 | ManxPower | mmealling, The IAXy is never the solution. |
| 16:08.42 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 16:08.45 | RoyK | ManxPower: removing asterisk would be a good start, then :) |
| 16:08.51 | FlatFoot | anyone help with call quality probs on snom360's ? |
| 16:08.55 | sip_me | I want the calls to be done via the Asterisk. I just do not want Asterisk to handle RTP! |
| 16:09.07 | mmealling | ManxPower, hmm.... so how should I connect the analog phones in the house to my PC running IAX? |
| 16:09.07 | RoyK | sip_me: canreinvite=on, then |
| 16:09.34 | *** join/#asterisk livesNbox (n=chadkous@user-12l2mv6.cable.mindspring.com) |
| 16:09.36 | FlatFoot | got 2 360's that have issues but the 190's on the same network are fine , has anyone found any probs with the 360 ? |
| 16:09.37 | mmealling | without spending $600 on a PCI card with four lines. |
| 16:09.37 | ManxPower | mmealling, Why use IAX? Why not an inexpensive SIP ATA like a SIPura. |
| 16:09.41 | RoyK | sip_me: only with clients behind nat, you're bound to be up against problems |
| 16:10.00 | *** part/#asterisk chexum (i=jfarkas@gateway/tor/x-726888265e8cf560) |
| 16:10.08 | Dr-Linux|work | guys, TE406P << should i buy this echo cancelation module? as i was going to buy TE405P , what's the main difference? |
| 16:10.10 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 16:10.21 | ManxPower | mmealling, The IAXy is for connecting phones to Asterisk. You cannot connect phone LINES to Asterisk with the IAXy |
| 16:10.29 | mmealling | ManxPower, because I want to run it as a PBX. I _hate_ putting application level network protocols into hardware only, non-linux based 'appliances'. |
| 16:10.48 | mmealling | ManxPower, correct.... I want to connect phones.... not a phone line..... |
| 16:10.55 | RoyK | Dr-Linux|work: echocancel will be done in hardware/firmware and helping your box to be able to handle far more traffic |
| 16:10.55 | ManxPower | mmealling, the IAXy does not run Linux. |
| 16:11.02 | RoyK | Dr-Linux|work: i'd say get the one with hwec |
| 16:11.03 | mmealling | ManxPower, I know..... |
| 16:11.05 | ManxPower | <PROTECTED> |
| 16:11.17 | puzzled | or alaw |
| 16:11.18 | ManxPower | The IAXy only support ulaw and alaw protocols. |
| 16:11.29 | ManxPower | puzzled, it doesn't support ALAW? |
| 16:11.36 | puzzled | ManxPower: afaict nope |
| 16:11.40 | coppice | i think it does G.726 |
| 16:11.53 | mmealling | I'm trying to find something that I can plug into my computer (USB preferably) and then plug into the wall RJ-11 so that any phone in the house uses the PC to place/recieve calls. |
| 16:11.55 | Dr-Linux|work | RoyK, Great, what's the price difference and "installation method" difference? |
| 16:11.57 | sip_me | In otherwords RoyK - this will get the RTP going via the Astrisk |
| 16:12.09 | ManxPower | mmealling, The IAXy will not work that way. |
| 16:12.17 | mmealling | ok..... |
| 16:12.26 | mmealling | got any suggestions for what does that doesn't cost $600? |
| 16:12.30 | ManxPower | You have two choices. 1) Get a PCI card, or 2) get an ethernet ATA device. |
| 16:12.41 | mmealling | why an Ethernet ATA device? |
| 16:12.41 | ManxPower | mmealling, Yes, get a SIPura. |
| 16:12.51 | ManxPower | mmealling, Because there are no other ATA devices. |
| 16:13.11 | mmealling | but that puts SIP into a device which is where I don't want it to be. |
| 16:13.27 | mmealling | its yet another device to manage when there doesn't need to be one. |
| 16:13.32 | ManxPower | mmealling, An IAXy puts IAX into a device. |
| 16:13.51 | mmealling | ManxPower, yes, and I didn't _want_ to use an IAXy.... |
| 16:13.52 | ManxPower | mmealling, If you do not want a nother device then you will have to get a PCI card. |
| 16:14.17 | ManxPower | I have listed your choices. Wishing you had more options will not make more options appear out of nothing. |
| 16:14.24 | mmealling | and those start at $600.... how hard is it to have a usb-sound device with a rj-11 out? |
| 16:14.41 | RoyK | sip_me: canreinvite=yes makes asterisk issue a reinvite so the RTP stream goes point-to-point, outside asterisk |
| 16:14.50 | ManxPower | mmealling, Digium used to have one. They never worked right and they stopped selling them. |
| 16:15.07 | ManxPower | nobody else has wanted to make one. |
| 16:15.17 | RoyK | Dr-Linux|work: installation is more or less the same. the price for HWEC is something like +$1k |
| 16:15.29 | RoyK | I'd say it's worth it |
| 16:15.36 | ManxPower | I got HWEC for about $300 |
| 16:15.45 | RoyK | what sort? |
| 16:15.47 | benjk | ManxPower: edit telepathy.conf, add a new entry [2000] with type=mindreader |
| 16:15.52 | ManxPower | RoyK, Tellabs off eBay |
| 16:15.56 | benjk | no device needed |
| 16:15.57 | RoyK | k |
| 16:16.23 | ManxPower | RoyK, it's a fair amount of work to get working, but worth it. |
| 16:16.23 | mmealling | sigh..... ok, I guess I'll have to conform to the bubba joe, "plug that there little box in and it does magic stuff to your phone" method. |
| 16:16.37 | eKo1 | Is there a variable or something that I can use to determine if a line is ringing apart from the usual CLI message? |
| 16:16.49 | ManxPower | eKo1, no. |
| 16:17.03 | *** join/#asterisk frodonbash (n=frodon@AMarseille-153-1-55-197.w86-200.abo.wanadoo.fr) |
| 16:17.07 | eKo1 | Ah rats. |
| 16:17.25 | ManxPower | eKo1, What are you trying to accomplish? |
| 16:17.26 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 16:17.55 | eKo1 | I'm testing out Zap channels because some of them are 'mute', i.e. they don't ring. |
| 16:17.59 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:17.59 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:18.03 | ManxPower | mmealling, in traditional telecom equipment, a 4-port analog card would cost $2,000 |
| 16:18.31 | eKo1 | I want to automate this so I need something that will tell me if the call is ringing. |
| 16:18.41 | ManxPower | I thought a TDM400P loaded costs about $305, maybe I'm wrong. |
| 16:18.59 | ManxPower | eKo1, analog or ISDN? |
| 16:19.02 | Qwell[] | ManxPower: That's about right |
| 16:19.26 | ManxPower | Qwell, Any idea where this guy got the $600 figure from? |
| 16:19.34 | eKo1 | ManxPower: ISDN |
| 16:19.39 | Qwell[] | ManxPower: rediculous resellers? |
| 16:19.58 | Qwell[] | or...a 4 port tdm2400 maybe? |
| 16:20.00 | *** join/#asterisk knarfly_wk (n=bwatson@12.42.132.26) |
| 16:20.21 | Qwell[] | that's a bit more than that though, I thought |
| 16:20.22 | ManxPower | Qwell, Heck, if he needs 4 ports, then 4 IAXys would cost more than a fully loaded TDM400P |
| 16:20.29 | Qwell[] | ManxPower: indeed |
| 16:20.43 | file | SIP it to me sir! |
| 16:21.46 | ManxPower | RoyK, The nice thing about the Tellabs is that the chassis I got came with like 12 EC cards for 12 T-1s |
| 16:23.58 | RoyK | erm. 12xT1 EC for $300? |
| 16:24.15 | RoyK | not too expensive |
| 16:24.32 | RoyK | still, I'd guess Dr-Linux|work will be using E1s |
| 16:24.44 | coppice | about a dollar a channel. cheap at half the price |
| 16:25.13 | RoyK | ...unless .pk has started downgrading to T1s |
| 16:25.30 | coppice | there is far more E1 that T1 kit in this world, yet 90% of the second hand stuff seems to be T1 |
| 16:25.58 | ManxPower | RoyK, depends on where he needs the EC. |
| 16:26.00 | mmealling | that's the thing, I don't need 4 ports..... just one |
| 16:26.16 | ManxPower | mmealling, then you would pay about $250 |
| 16:26.20 | ManxPower | for a card. |
| 16:26.41 | Qwell[] | ManxPower: for a tdm400? |
| 16:26.46 | RoyK | methinks digium tdm hardware is overpriced.... |
| 16:26.49 | Qwell[] | more like...$130ish |
| 16:27.04 | Qwell[] | RoyK: price Avaya hardware. |
| 16:27.08 | RoyK | mmealling: use isdn instead if you can. lots of cheap stuff available there |
| 16:27.11 | mog | price cisco hardware |
| 16:27.14 | *** join/#asterisk [hC] (n=hardcore@201.198.6.194) |
| 16:27.15 | mmealling | isdn? why? |
| 16:27.16 | mog | price dialogic hardware |
| 16:27.17 | mog | etc |
| 16:27.22 | Qwell[] | price any non-Digium hardware :P |
| 16:27.53 | RoyK | mmealling: get a $15 HFC card that can do BRI just fine |
| 16:28.39 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 16:29.57 | *** join/#asterisk jart (n=user@ix66.ctlinc.com) |
| 16:30.03 | mmealling | hmm..... according to voip-info.org: Support for BRI in Asterisk is bewildering and highly dependant upon the driver |
| 16:31.30 | ManxPower | mmealling, That would be correct. |
| 16:31.43 | ManxPower | Also not supported for USA/Canada BRI. |
| 16:33.59 | ionix | hmm, what UDP ports must be open for asterisk to work correctly? |
| 16:34.10 | ManxPower | ionix, no. |
| 16:34.13 | ionix | I opened only 5060 but I can't get any sound from it so I assume there is a range I need. |
| 16:34.30 | ManxPower | You only need to open UDP ports if you want to use a VoIP protocol. |
| 16:34.45 | ionix | SIP... |
| 16:34.50 | ionix | Asterisk is calling a SIP phone |
| 16:34.55 | ManxPower | ionix, You could have said so in the first place. |
| 16:35.21 | ManxPower | ionix, The UDP ports for audio are dynamically negotiated for each call. |
| 16:35.30 | StewLG | Is there an alternative to pastebin? It isn't responding, and I don't want to flood the channel with my question. |
| 16:35.38 | ManxPower | However, you can force the range on one side of the call with /etc/asterisk/rtp.conf |
| 16:35.40 | *** join/#asterisk InfraRed (n=infra@bb-87-81-42-111.ukonline.co.uk) |
| 16:35.44 | ionix | ok let me check |
| 16:36.20 | sb_mx | ionix, i think you also need to open the ports both ways. at least thats how it worked for us with iax. not sure if its the same with sip |
| 16:36.46 | ionix | works |
| 16:37.51 | StewLG | Asterisk doesn't use mp321 anymore, I gather. Is it still true that a timing source is needed? |
| 16:38.09 | InfraRed | hi all |
| 16:38.14 | Qwell[] | StewLG: asterisk never used mpg321...only mpg123 |
| 16:38.34 | ManxPower | In fact if you used mpg321 it would not even work with Asterisk |
| 16:38.39 | Qwell[] | and it still can, if you'd like |
| 16:38.51 | StewLG | I'm just trying to get it to work at all. |
| 16:38.53 | InfraRed | can anyone point me for a possible solution for a voice stream loss after transferring a call ? |
| 16:39.04 | ManxPower | StewLG, perhaps you missed my earlier message. MOH DOES NOT REQUIRE TIMING |
| 16:39.09 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 16:39.18 | StewLG | Manx: Yes, I did miss that. |
| 16:39.25 | ManxPower | InfraRed, NAT and reinvites |
| 16:39.26 | StewLG | Ok then.. so .. that's not my problem. |
| 16:39.29 | InfraRed | the asterisk server is on public ip, phones are on nat behind the server. incoming call is ok, whne i transfer the call the incoming voice is lost but the outgoing is ok |
| 16:39.55 | StewLG | Infra: Is installing a SIP proxy on your NAT gateway an option? |
| 16:39.58 | StewLG | It worked for me. |
| 16:40.02 | ManxPower | InfraRed, So you have canreinvite=no |
| 16:40.18 | InfraRed | StewLG: the gateway is the * server |
| 16:40.34 | InfraRed | caninvite=no |
| 16:40.36 | Dr-Linux|work | RoyK still, I'd guess Dr-Linux|work will be using E1s >> No, we will use T1s , as this server is going to be setup in USA |
| 16:40.40 | StewLG | Oh, you're right. I wasn't paying attention. Sorry! |
| 16:40.53 | InfraRed | hmm |
| 16:41.00 | InfraRed | ManxPower: i dont |
| 16:41.04 | InfraRed | should i put it in ? |
| 16:42.13 | StewLG | http://tinyurl.com/bn57o suggests I need format_mp3 installed for MOH. Is this true? |
| 16:42.19 | StewLG | Or rather, is this *still* true? |
| 16:42.35 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 16:43.23 | InfraRed | ManxPower: www.subhi.com/sip.txt |
| 16:44.59 | _alex_mx_ | StewLG, it is if you want to have your moh in mp3 format |
| 16:45.00 | *** part/#asterisk michael-i (n=michael@141.41.38.58) |
| 16:45.27 | *** join/#asterisk Ne0traX (n=mmm@213.255.65.47) |
| 16:45.40 | StewLG | Alex: I do.. so.. I'm running Debian.. how do I get this? |
| 16:46.20 | StewLG | I can't find an "asterisk-addons" package for debian. |
| 16:46.55 | mmealling | ManxPower, it looks like I should get an SPA-3000? |
| 16:47.11 | _alex_mx_ | StewLG, are you compiling from source? |
| 16:47.14 | ManxPower | mmealling, an SPA-1000 would be find if you just need a single port |
| 16:47.31 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 16:47.33 | StewLG | Alex: No, I'm not compiling from source, and would prefer not to. I will if necessary, though. |
| 16:47.35 | ManxPower | find == fine |
| 16:47.39 | *** join/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org) |
| 16:47.43 | ManxPower | ~fxofxs |
| 16:47.46 | jbot | i heard fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 16:47.57 | mmealling | but the 1000's single port is an FXS which means I can only connect a single phone, not the entire house. right? |
| 16:48.28 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
| 16:48.32 | _alex_mx_ | StewLG, as of 1.2 you don't need mpg123 support for mp3 is "built in" not sure if there's a package for debian |
| 16:48.36 | ManxPower | mmealling, it has a single phone port. You can plug multiple phones into it using a splitter. |
| 16:48.39 | *** part/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org) |
| 16:48.40 | hmmhesays | caught here in a fiery place |
| 16:48.48 | RoyK | ManxPower: no BRI support in northern america?? |
| 16:49.05 | ManxPower | RoyK, Correct. Nobody in NA uses BRI for voice. |
| 16:49.18 | StewLG | alex: I'm running 1.2. : http://tinyurl.com/bn57o suggests I need format_mp3 installed for MOH. I do need format_mp3, right? |
| 16:49.19 | Qwell[] | or data, or much of anything :p |
| 16:49.25 | RoyK | that's stupid :P |
| 16:49.28 | ManxPower | in most states BRI is 2 x more expensive than 2 analog lines. |
| 16:49.35 | RoyK | oh |
| 16:49.36 | RoyK | ok |
| 16:49.42 | ManxPower | TN is the exception |
| 16:49.52 | RoyK | in .no a BRI is like 20% more expensive than an analog line |
| 16:49.58 | _alex_mx_ | StewLG, format_mp3 is a module...check /usr/lib/astersik/modules and see if you have it |
| 16:50.10 | Qwell[] | StewLG: You need asterisk-addons |
| 16:50.11 | _alex_mx_ | or asterisk/modules |
| 16:50.12 | RoyK | show modules like mp3 |
| 16:50.24 | ManxPower | mmealling, you do not have a local phone line? |
| 16:51.12 | hmmhesays | ok avenged sevenfold just rocks |
| 16:51.22 | ManxPower | Qwell[], Before *DSL many people used BRI for data in the USA |
| 16:52.59 | RoyK | Deutsche telecom did it right. the ditched all pots stuff in the early ninetees, going for isdn all round |
| 16:52.59 | ManxPower | RoyK, also in the USA you must buy your own ISDN box |
| 16:52.59 | ManxPower | the telco does NOT provide it. |
| 16:53.03 | Qwell[] | and, if that wasn't bad enough... |
| 16:53.06 | RoyK | hm. doesn't make it easy to migrate to isdn, then |
| 16:53.11 | Qwell[] | every time you get ISDN in the US, god kills a kitten |
| 16:53.18 | RoyK | even though isdn is easier to support and maintain |
| 16:53.28 | RoyK | and quite possibly cheaper in the long run |
| 16:53.40 | ManxPower | RoyK, Same goes for PRIs |
| 16:53.46 | Qwell[] | RoyK: which is why they charge more for it |
| 16:53.53 | Qwell[] | because...they can |
| 16:54.24 | RoyK | ManxPower: so with a PRI you need to get your own modem? |
| 16:55.26 | ManxPower | RoyK, Doesn't really apply to PRI. MY point is that a 23 B-channel PRI in the USA is almost always MUCH more expenisve than 23 POTS lines. |
| 16:55.35 | RoyK | Knowledge = Power = Energy = Mass |
| 16:55.37 | ManxPower | Even though PRI uses less copper, is easier to maintain, etc. |
| 16:56.07 | mmealling | ManxPower, I do but that's what I want to get rid of... |
| 16:56.31 | ManxPower | mmealling, What will you use for 911? |
| 16:56.32 | *** part/#asterisk Ne0traX (n=mmm@213.255.65.47) |
| 16:56.37 | mmealling | cell phone |
| 16:56.54 | mmealling | an e911 when that gets figured out |
| 16:57.13 | mmealling | I'm probably going to go with sunrocket and they're attempting an e911 rollout |
| 16:57.32 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 16:57.33 | mmealling | out of curiosity, is anyone doing anything with ENUM? |
| 16:58.25 | InfraRed | depends on your country really |
| 16:58.40 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 16:58.57 | *** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 17:00.11 | hmmhesays | sunrocket? |
| 17:00.35 | hmmhesays | lets take two random nouns and make a company name! |
| 17:00.48 | c4t3l | crap |
| 17:01.00 | mmealling | so if I take the FXS port on the SPA-1000 and plug directly into the wall (the home wiring for that wall RJ-11 outlet is _not_ connected to the PSTN. Just the other house phones) then I should be able to pick up any phone in the house it should get dialtone from the SPA-1000, right? |
| 17:01.15 | hmmhesays | shoetire! |
| 17:01.22 | c4t3l | oh i see |
| 17:01.24 | mmealling | hmmhesays, yep.... when you're company name is determined by what domain-names are available you get silly stuff like that. |
| 17:01.53 | svenna | has someone linphone workin with * ? |
| 17:02.04 | hmmhesays | shelfchair! |
| 17:02.44 | tasat- | hello -- I'm struggling to find a termination/origination provider that fully clamps DTMF tones -- with most that I've tried, I'm hearing short blips. Anyone know if there's another solution? |
| 17:02.59 | c4t3l | adolfsaddam telecom |
| 17:03.09 | hmmhesays | tasat: for what purpose? |
| 17:03.48 | tasat- | hmmhesays: it's an issue with conferencing when users press DTMF for config -- all the others hear it. Any they don't like it |
| 17:04.11 | hmmhesays | conferencing via what? |
| 17:04.50 | tasat- | hmmhesays: SIP w/ Meetme and app_conference |
| 17:05.36 | tasat- | hmmhesays: probably will go with app_conference |
| 17:06.32 | InfraRed | canreinvite=no solved it \o/ |
| 17:06.38 | InfraRed | thanks ManxPower |
| 17:07.47 | hmmhesays | app_conference likes to segfault asterisk |
| 17:08.04 | hmmhesays | and if you are using meetme, you have something setup wrong if you are hearing dtmf tones in there |
| 17:08.57 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 17:09.13 | okdo | hi |
| 17:09.35 | okdo | trying to do an autoconfig of the thomson st2030 ... I see this line. NtpZoneNum=23 .... does anyone know what these zone numbers are? |
| 17:11.48 | tasat- | hmmhesays: I'm hearing short, sub-second blips resulting from a partially clamped tone from the provider -- what the destination or source user hear is the same whehther in a conference (with app_conf or meetme) or not. I've double checked this with a port capture. The providers that send DTMF via RFC2833 are all trying to clamp, but are not quite getting all the way... |
| 17:15.38 | hmmhesays | you sure they are sending rfc2833? |
| 17:15.53 | hmmhesays | and your inbound settings are for rfc2833 and not inband? |
| 17:17.02 | tasat- | hmmhesays: positive to both -- I've seen the RFC2833 packets in ethereal and gone over my config several times... you can't hear so much of a tone, but rather a noise blip, almost more like a click impulse |
| 17:17.45 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 17:17.58 | *** join/#asterisk livesNbox (n=chadkous@user-12l2mv6.cable.mindspring.com) |
| 17:18.18 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:18.41 | livesNbox | hey guys -- I can't seem to get queuemetrics to work correctly with dynamic agents and queues.. is there a better app for realtime monitoring and reporting on queues ? |
| 17:22.17 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
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| 17:24.27 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:24.51 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
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| 17:29.48 | *** part/#asterisk backblue (n=backblue@suporte.3gnt.net) |
| 17:31.36 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 17:32.29 | *** join/#asterisk agallo (n=agx@ip-37-53.sn1.eutelia.it) |
| 17:33.09 | agallo | Hi there, anyone here can give me an hand for linking 2 PBX with IAX trunking? i can call only in 1 direction atm |
| 17:34.32 | *** join/#asterisk ToyMan (n=stuq@74-32-7-1.dsl1.mdl.ny.frontiernet.net) |
| 17:35.10 | eKo1 | agallo: sure |
| 17:37.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:38.01 | Juggie | ~seen theplot |
| 17:38.14 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 23h 39m 23s ago, saying: 'I did set in the address field to match the username too'. |
| 17:40.35 | *** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
| 17:41.09 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 17:41.21 | kfudge | Can anyone tell me the general reasons why a call would have audio totally drop out in one direction at random times? |
| 17:42.07 | hmmhesays | unstable upload |
| 17:42.41 | hmmhesays | and I can't remember the joke but the punchline is "your mothers a whore" |
| 17:43.01 | kfudge | hmmm. Possible regular network traffic causing issues? |
| 17:43.18 | hmmhesays | yes |
| 17:43.50 | agallo | Anyone has a dual asterisk server? i'm trying "example 3" from the wiki but i can make calls only in 1 direction... |
| 17:44.03 | hmmhesays | heh, example 3? |
| 17:44.05 | hmmhesays | what is that |
| 17:45.52 | agallo | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 17:46.10 | agallo | check "example 3": master server has static IP, slave server has dynamic IP |
| 17:46.26 | agallo | i can originate call from slave to master but not viceversa |
| 17:47.19 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 17:48.29 | hmmhesays | pastebin your sip.conf and the corresponding extensions.conf |
| 17:49.02 | agallo | a second :) |
| 17:50.49 | viperdude | anyone had load average issues with MOH in Asterisk 1.2.x? |
| 17:51.48 | eKo1 | define 'load average issues' |
| 17:52.38 | viperdude | well if i have say 20 calls in a queue using mpg123 then load average is around 0.08 but if i use native in asterisk 1.2.x then load avergae junps to over 1.00 |
| 17:53.30 | ManxPower | viperdude, Well, with mpg123 MoH the audio is only being decoded once. With native MoH, the audio is being decoded once for each channel on hold. |
| 17:54.10 | viperdude | ManxPower: yes i realise that ... so how is that an advantage over mpg123? |
| 17:54.54 | ManxPower | viperdude, I never thought it was unless you needed the MoH to play a MESSAGE that you wanted to start over for each call on hold or if you are using non-MP3 audio files. |
| 17:55.44 | viperdude | ManxPower: well i wanted to try native as I was having asterisk crashing when using MixMonitor and i read some where it may be a mpg123 issue |
| 17:56.11 | ManxPower | viperdude, How in the world would mpg123 screw up mixmonitor? |
| 17:56.41 | viperdude | i dont know , but i read somewhere it maybe a issue... i readily accept it could be wrong |
| 17:56.56 | agallo | here you are: http://pastebin.ca/120964 |
| 17:59.37 | viperdude | if anyone could give a hint on why mixmonitor crashes when one of the parties hangup up then u would be grateful |
| 17:59.42 | hmmhesays | um you are trying to dial iax2 to a sip user |
| 17:59.58 | viperdude | i would be grateful even :-) |
| 18:00.05 | agallo | hmmhesays, yes it works from slave to master |
| 18:00.08 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 18:00.54 | *** join/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 18:00.55 | agallo | hmmhesays, i also tried sip<->sip and iax2<->iax2, same result |
| 18:00.59 | negativecreep | hi folks.. |
| 18:01.25 | agallo | hmmhesays, also tryied with and without port-forwarding on slave server (this is a total of 8 attempts) |
| 18:01.34 | ManxPower | Must. Resist. Buying. Stroopwafels. Mailorder. |
| 18:01.48 | hmmhesays | randomly trying shiat isn't going to get you anywhere |
| 18:01.53 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 18:02.08 | negativecreep | i am unable to login in an astpp fresh install..the accounts table in the astpp db doesnt contain an entry for admin user.. |
| 18:02.17 | hmmhesays | the 'master' has the iax2 user configured? |
| 18:02.26 | ManxPower | negativecreep, Perhaps you have the wrong channel. |
| 18:02.28 | *** join/#asterisk tewfik (n=SpazzinS@81.52.161.75) |
| 18:02.31 | tewfik | hi all |
| 18:02.36 | negativecreep | ManxPower: which channel for astpp? |
| 18:02.42 | agallo | hmmhesays, yes as showed in pastebin |
| 18:02.43 | ManxPower | negativecreep, Not this one. |
| 18:02.49 | hmmhesays | if that is the case, then your 'slave' to 'master' sip->iax2 call is not working |
| 18:03.01 | hmmhesays | -- Called sedecentrale/200 -- IAX2/sedecentrale-9 is circuit-busy |
| 18:03.05 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 18:03.10 | negativecreep | ManxPower: thnx |
| 18:03.18 | *** join/#asterisk pigpen2 (n=mark@fwipsec.tmi-sa.org) |
| 18:03.32 | agallo | hmmhesays, se, then your 'slave' to 'master' sip->iax2 call is not working |
| 18:03.33 | agallo | <hmmhesays> -- Called sedecentrale/200 -- IAX2/sedecentrale-9 is circ |
| 18:03.40 | agallo | ops sorry |
| 18:04.08 | agallo | i'm also trying iax2<->iax2 and its the same result |
| 18:04.12 | tewfik | i need some technical help about the way asterisk handles modems on linux/bsd distros, is the "only" thing to do is to get a winmodem works on linux so asterisk can recognize/ use it ? thanks in advance |
| 18:04.25 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:04.26 | hmmhesays | stick with the same protocols going asterisk to asterisk |
| 18:04.35 | hmmhesays | start from scratch and set them up |
| 18:04.37 | hmmhesays | then debug |
| 18:04.40 | hmmhesays | this is a logical progression |
| 18:04.44 | tewfik | i want to do a simple voice/fax switch using asterisk / |
| 18:04.50 | tewfik | + winmodem |
| 18:04.55 | agallo | hmmhesays, what to debug when all i get is "hangup cause is 50" ? |
| 18:05.08 | tewfik | googled around and got confused, thank you |
| 18:05.10 | hmmhesays | instead of jumping to a completely different configuration, debug your iax2 to iax2 calls between the servers |
| 18:05.16 | sb_mx | agallo, turn your debug on cause its saying "received reject". maybe something's wrong there |
| 18:05.26 | hmmhesays | iax2 debug |
| 18:05.30 | Qwell[] | tewfik: You need something with drivers for zaptel |
| 18:05.41 | Qwell[] | tewfik: of which, there is only one - and it's junk |
| 18:05.54 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:05.58 | Qwell[] | tewfik: You would be better off getting a proper telephony card |
| 18:06.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:06.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:06.23 | ManxPower | tewfik, the reason you are confused is that you can't use Winmodems with Asterisk |
| 18:06.27 | tewfik | zaptel cards are pstn compatible ? |
| 18:06.58 | tewfik | ManxPower : nor can i use modems that have builtin chipsets ? |
| 18:07.13 | *** join/#asterisk ppdm (i=pdm@pdm.sh) |
| 18:07.16 | ManxPower | tewfik, correct. |
| 18:07.26 | ppdm | anyone use a 7961 on asterisk? |
| 18:07.32 | ManxPower | You can use Fax modems with OTHER software like Hylafax, of course. |
| 18:07.32 | ppdm | i read conflicting reports ;( |
| 18:07.39 | hmmhesays | i bet some ubergeek in some dark corner of his moms basement has written a winmodem channel for asterisk |
| 18:07.40 | *** part/#asterisk tjtoocool (n=none@72-19-25-58.idstelcom.net) |
| 18:07.42 | ManxPower | ppdm, does it use SIP? |
| 18:07.48 | Qwell[] | ppdm: yes, there is SIP firmware for it |
| 18:07.56 | ppdm | nice |
| 18:08.02 | Qwell[] | and, even if not, the answer is still yes |
| 18:08.07 | ppdm | now i gotta wait for my friend to get to work to get a damn CCO login |
| 18:08.42 | tewfik | i planned to do something which can answers to fax and voice (mp3 message is played in fact, a little interactio with some menus); can you please advice some cards that works well with asterisk / pstn line ? |
| 18:09.34 | agallo | hmmhesays, ok debug says that code 50 is "No authority found", and google it seems to say that maybe its a typo in my config... |
| 18:09.46 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 18:10.02 | agallo | hmmhesays is Dial(IAX2/sectioname/extension,30,r) correct? |
| 18:10.11 | hmmhesays | you have found your problem |
| 18:10.38 | hmmhesays | you are having an auth error |
| 18:10.53 | hmmhesays | your origination and/or your termination server are not set up right |
| 18:11.12 | tewfik | ManxPower : one last question; since asterisk can't support winmodem, why not use a windows box and route calls on ethernet , i'm a dreaming ? |
| 18:11.24 | Qwell[] | oh god |
| 18:11.29 | hmmhesays | rockin |
| 18:11.36 | sb_mx | agallo, does the context=local exists? |
| 18:11.48 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:11.52 | agallo | sb_mx, yes is where i've all the phones (sip and iax2) |
| 18:11.57 | hmmhesays | if you're quite giving us bits and pieces and pastebin what I told you |
| 18:12.20 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 18:12.45 | tewfik | i think i just said something silly, anyway thanks guys for help |
| 18:13.01 | tewfik | have a good day / night whichever is your timezone (-: |
| 18:13.17 | hmmhesays | 1. pastebin your iax.conf and extensions.conf for the server originating the call, then your iax.conf and extensions.conf for the termination server with iax2 debug enabled on the termination server |
| 18:13.35 | hmmhesays | there was going to be a second point there, but I smashed it all into one |
| 18:14.50 | agallo | hmmhesays, http://pastebin.ca/120964 |
| 18:16.14 | hmmhesays | i don't see any extensions.conf or iax2 debug information |
| 18:16.39 | agallo | hmmhesays, http://pastebin.ca/121000 |
| 18:17.41 | quid2478 | here's an interesting question... I want to evalluate call quality and latency from two different terminatino providers... I am going to dial the Bureau of Standards atomic clock time line as a timecode... is there a way to initiate two calls at *precisely* the same moment? I don't have to connect personally to either of the calls, as they will be recorded with monitor. |
| 18:18.00 | hmmhesays | your termination server credentials don't match the origination |
| 18:18.42 | *** join/#asterisk [hC] (n=hardcore@201.198.6.194) |
| 18:19.02 | agallo | hmmhesays, i'm lost: does it means i've a typo in username/password? |
| 18:19.10 | hmmhesays | that could be one reason |
| 18:20.39 | agallo | hmmhesays, FIXED! ty |
| 18:20.39 | sb_mx | agallo, sry to ask again. but does the context [local] exist in the slave? |
| 18:21.06 | agallo | honestly the Wiki is really ambiguos about [sectioname] , register and stuffs |
| 18:21.37 | agallo | is was not using [sectioname] == username, changed to it and started to work |
| 18:22.23 | hmmhesays | so read the configuration file notes |
| 18:22.53 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 18:24.37 | ManxPower | agallo, phones should be type=friend and the [sectionname] should contain username=sectionname |
| 18:24.56 | ManxPower | for remote servers you should have two entries, one type=user and one type=friend |
| 18:25.20 | file | erm? |
| 18:26.09 | ManxPower | [sectionname] is what is matched for INCOMING connections, username= is what is matched for OUTGOING connections. |
| 18:26.33 | ManxPower | This is a bit less divided in 1.2, but it's still a good habit to have. |
| 18:26.50 | agallo | ManxPower, strange i've section names in sip.conf like this [trunk-sip-1], [trunk-sip-2] and never had a problem with incoming sip connections O.o |
| 18:27.03 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-0b1fdeb1f71d0873) |
| 18:27.13 | ManxPower | agallo, what is the type= for those? |
| 18:27.44 | agallo | ManxPower, i always use friend for trunks and phones |
| 18:28.00 | ManxPower | agallo, you will have problems with that when the remote side expects a different username than it sends. |
| 18:28.11 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 18:28.33 | agallo | ManxPower, for SIP i use UltraCom and Eutelia; they seem not affected from it |
| 18:28.51 | *** join/#asterisk podzap (n=podzap@83-102-12-142.dnayritysnetti.net) |
| 18:28.51 | ManxPower | agallo, so the far end sends a username of sip-trunk-1? |
| 18:28.53 | podzap | hi there! |
| 18:28.54 | agallo | ManxPower, will change my script too, its a 1 min work :-) |
| 18:29.12 | podzap | is * supposed to work on fc5, because it doesn't work for me? |
| 18:29.26 | agallo | ManxPower, dunno i just fill in username= and fromuser= fields |
| 18:30.00 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 18:30.27 | ManxPower | do you have insecure=very |
| 18:30.34 | brodiem | when making changes to zaptel.conf and running ztcfg to load the changes, does * needs to be restarted to take effect? It seems a CLI reload isn't doing anything (can't shut it down though due to active calls) |
| 18:30.52 | ManxPower | brodiem, ztcfg will kill all active zap calls. |
| 18:31.22 | ManxPower | brodiem, for the most part a "reload" or "reload chan_zap.so" will load the changes except for a few options. |
| 18:31.49 | agallo | ManxPower, yes: insecure?very on all the SIP trunks |
| 18:31.57 | ManxPower | ah, that would explain it. |
| 18:32.01 | brodiem | ManxPower, hmm after running ztcfg, it displays all of the channel state changes, but a CLI zap show channels doesn't display my new channels |
| 18:32.10 | ManxPower | insecure=very basically means "turn off all authentication" |
| 18:32.25 | ManxPower | brodiem, what version of Asterisk? |
| 18:32.34 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-6.getnet.net) |
| 18:32.36 | brodiem | ManxPower, specifically from "unused" to PRI 23B+1D |
| 18:32.41 | ManxPower | agallo, I hope you don't allow any incoming SIP connections to dial out. |
| 18:32.44 | brodiem | ManxPower I'm on 1.2.6 |
| 18:33.01 | agallo | ManxPower, incomign SIP connection goes to a specific context, there is a context for every SIP trunk |
| 18:33.06 | ManxPower | brodiem, what does "reload chan_zap.so" do. |
| 18:33.10 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 18:33.32 | brodiem | ManxPower before I do that, will it disconnect active zap channels? |
| 18:33.38 | *** join/#asterisk [tasty]freeze (n=simba@pm3.5-23.wctc.net) |
| 18:33.59 | agallo | ManxPower, and that context contains: black list check, date and time check, fax detection, etc., etc. at the end if not already redirect goes directly to the default extension |
| 18:33.59 | ManxPower | agallo, Um, since there is no authentication these incoming calls may not be going to the context you expect them to. put context=INVALID in [general] if your calls stop working then the context you specified in the [sections] are not being used. |
| 18:34.08 | ManxPower | brodiem, "realod" will NEVER KILL ACTIVE CALLS |
| 18:34.14 | *** join/#asterisk apardo (n=apardo@87.217.147.151) |
| 18:34.39 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 18:34.39 | ManxPower | restart will kill active calls, a unload/load will kill active calls. reload will not kill active calls. |
| 18:34.40 | agallo | ManxPower, yes i've context=NULL in [general]... should i put insecure=no and re-check if my trunks are working? |
| 18:35.01 | brodiem | ManxPower hmm, doesn't list any of my new channels |
| 18:35.11 | ManxPower | agallo, you never want an insecure= setting unless you need it. |
| 18:35.29 | ManxPower | brodiem, what "ignoring" messages do you get on a reload chan_zap.so |
| 18:35.31 | agallo | ManxPower, but when i started on VOIP italian forum everyone said to use "insecure=yes" ... |
| 18:35.40 | cytrak | what is a good echo canceller for zap ? |
| 18:36.11 | cytrak | yesterday I compiled zaptel with MG2 and Agressive suppressor and that seems to make it worse |
| 18:36.15 | *** part/#asterisk podzap (n=podzap@83-102-12-142.dnayritysnetti.net) |
| 18:36.25 | brodiem | ManxPower, there were no ignore messages. All just "Reconfigured channel X...", but none of them were the newly configured channels |
| 18:36.50 | _alex_mx_ | brodiem, asterisk will parse /etc/asterisk/zapata.conf when it starts/reloads....zaptel.conf is parsed when you insert the zaptel module into the kernel |
| 18:36.55 | ManxPower | brodiem, there should be ignoring switchtype, signaling |
| 18:37.34 | brodiem | _alex_mx_, using ztcfg will parse zaptel.conf again though won't it? |
| 18:37.36 | brodiem | hmm |
| 18:37.52 | brodiem | maybe I'll just wait till the zap calls are done and run ztcfg again |
| 18:37.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:37.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:38.24 | ManxPower | brodiem, IN THE ASTERISK CLI |
| 18:38.31 | ManxPower | "reload chan_zap.so" |
| 18:38.38 | brodiem | yes I did that. |
| 18:38.46 | ManxPower | there should be messages about ignoring stuff. |
| 18:39.04 | quid2478 | Is there a way in the Dialplan to branch into two paralell instructions (instead of stepping sequentially)? |
| 18:39.40 | *** join/#asterisk adorah (n=Administ@84.94.208.77.cable.012.net.il) |
| 18:39.44 | brodiem | ManxPower at verbose 5, every line looks like "Reconfigured channel 1, FXS Kewlstart signalling" with each of my configured channels. There's no "ignore" messages at all |
| 18:41.20 | _alex_mx_ | brodiem, in my experience...you need to stop asterisk, unload/reload card driver, run ztcfg then your new config will show up |
| 18:41.29 | hmmhesays | quid2478: what do you want to do? |
| 18:41.34 | [TK]D-Fender | cytrak: Otasic or Tellabs |
| 18:41.54 | cytrak | what ? |
| 18:42.13 | brodiem | I'll give it a shot after phone hours are over, thanks |
| 18:42.18 | tdonahue-laptop | has anyone had any issues compiling zaptel on CentOS 4.3? |
| 18:42.29 | [TK]D-Fender | cytrak: : For your EC question. |
| 18:42.40 | cytrak | tdonahue-laptop: you getting a bunch of errors on vpm50m ? |
| 18:42.45 | [TK]D-Fender | tdonahue-laptop: : there is 1 commonly know and well known bug |
| 18:42.49 | [TK]D-Fender | ~centosbug |
| 18:42.51 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 18:43.00 | tdonahue-laptop | torisa.c |
| 18:43.01 | [TK]D-Fender | easy fix |
| 18:43.06 | quid2478 | hmmhesays: start a thread that branches off into two paralell commands... both start recording, dial out via a diffrent SIP provider and record the answered call |
| 18:44.04 | cytrak | [TK]D-Fender: sorry but you got me totally lost now |
| 18:44.04 | [tasty]freeze | I know I am not doing something right, so please forgive me, but when I run ztcfg (X100P Card) it tells me its configured. I have used freepbx to setup trunk 1, and an extension (1111) that dials it. it is telling me it is unable to create channal of type zap (cause 0, unknown). |
| 18:44.06 | brodiem | I assume ztcfg in test mode won't disconnect zap calls right? |
| 18:44.07 | quid2478 | then I can A/B the calls later for quality and latency (calling a time signal line); |
| 18:44.16 | quid2478 | a poor man's VOIP quality check |
| 18:44.20 | Qwell[] | [tasty]freeze: please re-read the channel topic |
| 18:44.28 | hmmhesays | exten => _X.,1,Dial(Local/1@command&Local/2@command); 1 and 2 are extensions under context command |
| 18:44.45 | cytrak | [TK]D-Fender: I set it to ECHO_CAN_MG2 and AGGRESSIVE_SUPPRESSOR before compiling zaptel |
| 18:44.56 | [tasty]freeze | join #freepbx |
| 18:44.58 | quid2478 | hmmhesays: Okay, looks line a winner... will give that a shot. |
| 18:45.11 | hmmhesays | what do I win? |
| 18:45.12 | agallo | ManxPower, i removed insecure=very and changed [sectionname] into username. How now become my Dial string? Dial(SIP/${EXTEN}@voipprovider.com,30) ? |
| 18:45.30 | cytrak | [TK]D-Fender: before I used ECHO_CAN_KB1 and it seemed to work well enough and yesterday I decided to try out the MG2 |
| 18:46.18 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 18:47.05 | quid2478 | hmmhesays: Hmm... I'll send you some free TCP packets. |
| 18:47.10 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 18:47.18 | quid2478 | thanks |
| 18:47.19 | hmmhesays | i'd rather have a $20 so I can buy some new guitar strings |
| 18:47.32 | *** part/#asterisk [tasty]freeze (n=simba@pm3.5-23.wctc.net) |
| 18:47.36 | [TK]D-Fender | cytrak: I was suggesting that if Zaptel isn't working for you its time to invest in HARDWARE and get the job done right. |
| 18:48.25 | cytrak | [TK]D-Fender: I thought digium cards were good |
| 18:48.34 | quid2478 | hehe, nothing like the sound of a good 4 string guitar |
| 18:48.52 | hmmhesays | we call that a banjo |
| 18:49.01 | quid2478 | haha |
| 18:49.04 | ManxPower | brodiem, http://pastebin.ca/121058 |
| 18:49.29 | ManxPower | agallo, if you dial @provider.com then sip.conf section will not be used. |
| 18:49.35 | ManxPower | dial @sipconfsection |
| 18:49.55 | *** join/#asterisk bmg505 (n=leon@c1-44-3.rndf.isadsl.co.za) |
| 18:50.38 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 18:52.17 | brodiem | ManxPower its like mine isn't reading from zapata.conf correctly or something |
| 18:52.25 | tdonahue-laptop | [TK]D-Fender, does that fix apply to all the close relatives to RHEL or just CentOS? |
| 18:52.45 | ManxPower | brodiem, or your /etc/asterisk/logger.conf is not sending WARNING to the console |
| 18:54.10 | _alex_mx_ | brodiem, pastebin dmesg output |
| 18:54.43 | ManxPower | brodiem, what specific option are you changing? |
| 18:55.01 | _alex_mx_ | he's adding a T1 |
| 18:55.50 | brodiem | ManxPower my bad, it wasn't printing warning msgs. I found it in my log along with a " ERROR[29635] chan_zap.c: Unable to reconfigure channel '25-47'" |
| 18:55.53 | brodiem | =/ |
| 18:58.13 | hmmhesays | haha timidity sucks with guitar pro 5 |
| 18:58.39 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 18:58.52 | brodiem | yeah I'm on a channelized T1 and switching to a PRI, so the PRI is in the 2nd TE210 port |
| 18:59.38 | [TK]D-Fender | tdonahue-laptop: Both |
| 19:00.01 | tdonahue-laptop | thanks, i'll have to make a note about that |
| 19:00.11 | hmmhesays | bang bang, shoot em up |
| 19:00.47 | brodiem | ManxPower but yeah I also saw the "Ignoring switchtype, signalling" msgs |
| 19:01.26 | *** join/#asterisk backblue (n=moo@87-196-102-213.net.novis.pt) |
| 19:05.58 | *** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net) |
| 19:06.48 | salaud | anyone know if the DB function can be used in a way to return more than one record... or in some way to loop through multiple records with the same key? I know this can be achieve with app_mysql |
| 19:06.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:06.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:07.12 | *** join/#asterisk froguz (n=xxxxx@pc-120-155-104-200.cm.vtr.net) |
| 19:07.18 | froguz | hi! |
| 19:07.43 | hmmhesays | fly me, courageous |
| 19:07.57 | froguz | russellb, are you the same russell who wrote a book about SS7? |
| 19:08.08 | russellb | heh, no |
| 19:08.18 | froguz | aps |
| 19:09.00 | froguz | well.... somebody knows the way to look in the CLI for the codecs being used in current calls? |
| 19:09.33 | ManxPower | brodiem, chances are you can't change the signalling for channels without a unload/load |
| 19:09.52 | ManxPower | frodonbash, <protocol> show channels |
| 19:10.00 | ManxPower | like sip show channels |
| 19:10.08 | ManxPower | or sip show channel thechanneid |
| 19:10.51 | [TK]D-Fender | hmmhesays: "The party never ends" |
| 19:11.12 | quid2478 | hmm, I'm looking at the Wiki for the Dial Command and the hangup after x time option... is the correct format to ahng up after 80 seconds? ie. Dial(SIP/exten@sipprovider,S80) or is Dial (SIP/exten@sipprovider,S(80))? |
| 19:12.04 | froguz | ManxPower, thanks |
| 19:12.27 | russellb | quid2478: you're missing a comma, add one more |
| 19:13.07 | salaud | So... there is no way to fetch through multiple rows using the DB function? |
| 19:13.15 | hmmhesays | [TK]D-Fender: heh |
| 19:14.11 | quid2478 | russellB: Hmm I'm waiting the call to hangup 80 seconds after it's picked up... not a timeout |
| 19:14.51 | hmmhesays | quid2478: I would follow russelb's instructions |
| 19:15.11 | brodiem | ManxPower thanks I'll try that after phones shut down this evening |
| 19:15.41 | quid2478 | So ...provider,S,80? |
| 19:15.51 | hmmhesays | provider,,s80 |
| 19:15.52 | quid2478 | So ...provider,S,80)? |
| 19:15.57 | [TK]D-Fender | hmmhesays: Skid Row was one of the groups I cut my teeth on learning guitar... |
| 19:16.00 | quid2478 | oh okay, thanks. :) |
| 19:16.07 | hmmhesays | [TK]D-Fender: oh yeah? |
| 19:16.30 | [TK]D-Fender | hmmhesays: Yup, Warrant, Whitesnake, Poison, Motley Crue, Skid Row, etc |
| 19:16.31 | hmmhesays | a lot of skid row is beyond my skill level |
| 19:16.32 | russellb | provider||S(80) to be exact |
| 19:16.41 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 19:16.41 | hmmhesays | comma's work too |
| 19:16.47 | russellb | i know ... |
| 19:16.58 | russellb | but they are translated into '|' anyway |
| 19:17.06 | [TK]D-Fender | hmmhesays: Nah! I'm sure you could pick it up easy... 18 & life, I Remember you, Youth Gone Wild. All pretty easy. |
| 19:17.07 | russellb | '|' is a good habit, since that is what we have standardized on |
| 19:17.37 | Qwell[] | and | is the only thing that works in realtime |
| 19:17.45 | hmmhesays | [TK]D-Fender: I suppose i'd have to try first |
| 19:17.46 | Qwell[] | or in doing some major hacks...like... |
| 19:17.49 | [TK]D-Fender | russellb: | kills visibility, I'm suprised you'd forego standard delimiters in lieu of pipes... |
| 19:17.57 | russellb | you'll get over it |
| 19:17.59 | Qwell[] | Set(blah=abc\|123) |
| 19:18.06 | hmmhesays | i need some good tutorials on increasing picking speed |
| 19:18.09 | Qwell[] | Macro(blah|${blah}) |
| 19:18.13 | [TK]D-Fender | hmmhesays: That would help :) Go to Harmony Central and grab some tabs. |
| 19:18.25 | hmmhesays | I got like 28k tabs in guitar pro format |
| 19:18.42 | Qwell[] | (yes, the above does work) |
| 19:18.59 | [TK]D-Fender | hmmhesays: You need to tighten up your grip and reduce your picking depth. Also only used the very tip. |
| 19:19.13 | hmmhesays | picking depth yes |
| 19:19.14 | [TK]D-Fender | Wow... and that couldn't POSSIBLY be taken completely out of context :D |
| 19:19.18 | hmmhesays | i have trouble with that |
| 19:19.20 | russellb | Qwell[]: considering Set doesn't try to separate on | anyway ... |
| 19:19.36 | Qwell[] | russellb: try it without the \ :D |
| 19:19.36 | file | [TK]D-Fender: I had to read... previous messages to clue in |
| 19:19.39 | salaud | Can someone pretty please verify that there is no way to fetch through multiple records using DBGet / DB function |
| 19:19.45 | Qwell[] | it b0rks real nice |
| 19:19.53 | russellb | salaud: correct |
| 19:19.53 | trelane_ | [TK]D-Fender, good song "Youth Gone Wild" |
| 19:19.55 | salaud | Does DBM even support that? |
| 19:20.03 | salaud | russellb: thanks |
| 19:20.06 | russellb | no, it does not support it, anyway |
| 19:20.22 | russellb | for that kind of stuff, look up func_odbc |
| 19:20.32 | file | oh! |
| 19:20.33 | file | russellb: !!! |
| 19:20.39 | salaud | russellb: excellent... or MYSQL |
| 19:20.40 | russellb | what? |
| 19:20.45 | Qwell[] | and actually...set does parse | |
| 19:21.00 | Qwell[] | Set(var1=value1|var2=value2) |
| 19:21.01 | russellb | salaud: still, use the ODBC function. there is no MYSQL function |
| 19:21.06 | russellb | Qwell[]: oh, d'oh |
| 19:21.15 | salaud | russellb: app_mysql |
| 19:21.20 | russellb | that's not a function |
| 19:21.26 | *** join/#asterisk jart (n=user@ix66.ctlinc.com) |
| 19:21.26 | russellb | that's an application |
| 19:21.31 | russellb | :) |
| 19:21.35 | salaud | russellb: right... but you can fetch through multiple records |
| 19:21.42 | russellb | cool |
| 19:21.47 | russellb | i've never used it |
| 19:21.53 | salaud | I don't care how the job is done.. other than not wanting to use a heavy weight database |
| 19:22.13 | *** join/#asterisk docelm0 (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 19:22.14 | russellb | can you use odbc with sqlite? |
| 19:22.44 | salaud | russellb: perhaps... I'll probably just abstract it and use an AGI so I can change the backend later |
| 19:22.49 | russellb | cool. |
| 19:22.59 | salaud | you ever used res_perl? |
| 19:23.02 | russellb | no |
| 19:23.11 | salaud | I don't think many people have |
| 19:23.20 | salaud | I guess it's FastAGI all the way |
| 19:23.29 | [TK]D-Fender | file : Like I said... CONTEXT ;) |
| 19:23.47 | file | [TK]D-Fender: sorry, my contexts aren't setup correctly |
| 19:24.14 | froguz | i have a licence for G.729, so i decided to convert asterisk sounds into g729 format, my question is: if a left a copy in g729 and other in g711 for each sound. will asterisk use the g711 automatically after the licence have been used in another proccess? |
| 19:24.29 | salaud | russellb: can't find func_odbc in the standard distribution.. is it in add-ons? |
| 19:24.40 | Qwell[] | russellb: fyi; set also has options...heh |
| 19:24.46 | russellb | salaud: it's only officially in trunk, but there is a backport |
| 19:24.57 | russellb | salaud: asterisk-backports.org has a link to it, i think |
| 19:25.25 | salaud | russellb: is it stable (ish) |
| 19:25.45 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 19:26.01 | Qwell[] | salaud: it works very well |
| 19:26.36 | russellb | svn co http://svncommunity.digium.com/svn/func_odbc/1.2 func_odbc-1.2 |
| 19:26.38 | salaud | that's good... that could provide the abstraction that I need without running another FastAGI daemon |
| 19:27.13 | salaud | russellb: will it compile with just the asterisk-dev headers? I'm using a debian package |
| 19:27.34 | russellb | um ... i'm not quite sure |
| 19:27.44 | russellb | in theory, but the makefile may not be set up that way |
| 19:29.06 | salaud | yeah... the makefile for res_perl is only setup (such that it is "setup" at all) to compile against an asterisk source tree not just headers |
| 19:30.06 | salaud | I think considering an backport or other module in the light of folks who primarily use a package would be good |
| 19:30.06 | froguz | how smart is asterisk selecting file formats from /var/lib/asterisk/sounds/?? will use the correct format if i have more than one? |
| 19:30.22 | russellb | froguz: yes |
| 19:30.52 | froguz | thanks russell |
| 19:33.49 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
| 19:34.33 | NDT | Probably sounds friggin dumb...but anyone know if HT could possibly cause RDNIS not to show up? heh |
| 19:34.47 | NDT | Tried every thing I can and the other box is fine |
| 19:34.52 | Qwell[] | HT could cause a lot of issues :p |
| 19:35.24 | NDT | Hey Qwell: You know if adding noht to grub.conf would be enough or does it actually need to be disabled in the bios? |
| 19:35.33 | Qwell[] | no clue |
| 19:35.59 | NDT | yeah noht showed 2 processors which is correct unlike when it shows 4 heh...but then asterisk wouldn't start LOL |
| 19:36.43 | NDT | Wish there was a utility to change the bios settings without a drac on these poweredge servers |
| 19:36.58 | NDT | damn server is 3.5 hours away |
| 19:37.03 | adorah | I don't mind using older cpu without HT..much cheaper but it is time to get over that problem.. |
| 19:45.57 | *** join/#asterisk roguebug (n=johnw2@dip-106-210.bras.dsl.breisnet.com) |
| 19:45.59 | roguebug | hi. |
| 19:49.06 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:51.43 | roguebug | i'm using wav files (mono, 16 bit, 8000 Hz sample rate) for my soundfiles. however, i find that the sound in this sampling freq clearly isn't as good as the normal speech quality over direct isdn. can i use any higher sample rate ? |
| 19:54.02 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 19:54.05 | dos000 | howdy |
| 19:55.14 | dos000 | i am trying to build zaptel and zaptel is barking because /lib/modules/`uname -a`/build does not exist .. any ideas ? |
| 19:55.35 | dos000 | uname -r actually |
| 19:56.40 | ManxPower | dos000, it should like your currently installed kernel is not the same as your currently running kernel or your currently running/installed kernal does not match the kernel-source installed or the kernel-headers installed. |
| 19:58.31 | dos000 | ManxPower, the kernel headers package does not install a build directory tho ! |
| 19:59.22 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 19:59.36 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 19:59.40 | Assid | heya |
| 19:59.40 | ManxPower | dos000, There's a reason the zaptel build is looking for the wrong kernal module directory. |
| 20:00.17 | Assid | okay if im getting an incoming call.. shouldnt the name pop if the person is in my directory? |
| 20:00.22 | Assid | on a poly301 ? |
| 20:00.57 | Shaun2222 | dos000: what os you running? |
| 20:01.06 | dos000 | ManxPower, it says ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) |
| 20:01.06 | dos000 | <PROTECTED> |
| 20:01.30 | h3x | this is so god damn funny |
| 20:01.31 | ManxPower | dos000, so where is it getting $(KVERS) from? |
| 20:01.41 | dos000 | Shaun2222, debian sarge with a custom kernel |
| 20:01.54 | dos000 | ManxPower, is the uname -r |
| 20:02.04 | h3x | i called this place they has a ivr and it sounds like they re-recorded the voices for call queues |
| 20:02.07 | h3x | but they left the default asterisk moh |
| 20:02.32 | ManxPower | dos000, so uname -r is not correct for your kernel? |
| 20:03.22 | Shaun2222 | dos000: build should be a symlink to the kernel source you used to build that custom kernel |
| 20:03.37 | dos000 | ManxPower, uname -r points to the right place .. its just that the build dir its looking for does not exist and it is not installed by the kerne-headers pacakge |
| 20:03.53 | ManxPower | dos000, sounds like you need to put in some symlinks |
| 20:04.25 | dos000 | ManxPower, so build points to the src ? or the include ? |
| 20:04.53 | Shaun2222 | points to what is usually /usr/src/linux in the old days |
| 20:05.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:05.37 | plexdk | Hi there.. where can i find some examples on schuedeled opening hours, for my extensions? |
| 20:06.00 | ManxPower | plexdk, /path/to/src/asterisk/configs/extensions.conf.sample |
| 20:06.24 | dos000 | ManxPower, thnx ! |
| 20:07.58 | plexdk | ah drat, doesnt exist in the freebsd port version of asterisk |
| 20:09.47 | plexdk | another thing.. When my asterisk is going to the answering machine, its playing the typical "you have called..1..0..1" etc.. How do I make it play only one file i define? |
| 20:10.01 | plexdk | instead of the typical answering machine, plus the number gsm files |
| 20:10.38 | caloi | anyone know if it's possible to have the output of the Asterisk CLI write to a file? Example: I want to capture SIP off one peer and write to a file, is that possible? |
| 20:10.44 | Corydon-w | You mean voicemail? |
| 20:10.49 | plexdk | Yeah voicemail =) |
| 20:11.13 | Corydon-w | Go into VoicemailMain and in the Mailbox options (0), record your name and greeting |
| 20:11.49 | Corydon-w | caloi: man script |
| 20:12.15 | ManxPower | plexdk, download the asterisk source |
| 20:12.17 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 20:12.17 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:12.24 | plexdk | ManxPower: yeah im on it =) |
| 20:12.26 | plexdk | thx |
| 20:12.55 | ManxPower | You never know what those *BSD packagers are going to our poor innocent Asterisk source code. |
| 20:13.02 | caloi | @Corydon - that's the easiest way huh? I've used script before, I just wish I could get the console to right directly to a file |
| 20:13.27 | Corydon-w | What do you think script(1) does? |
| 20:13.47 | ManxPower | caloi, you mean like you can define in /etc/asterisk/logger.conf |
| 20:13.48 | caloi | script outputs everything on my screen to a file |
| 20:14.15 | Corydon-w | and you said you wanted everything on your screen output to a file |
| 20:14.45 | caloi | i sais "example, if i want to capture SIP off one peer and write that to a file, is that possible" |
| 20:14.50 | plexdk | Corydon-w: i dont quite follow you.. do you mean i have to set it in this extension: exten => 101,4,VoiceMail,u101 |
| 20:14.53 | *** join/#asterisk ComputerWarm (n=donc@209.29.159.11) |
| 20:14.58 | caloi | if I do this in the aster console: |
| 20:14.58 | caloi | SIP Debugging Enabled for IP: 66.218.16.66:5060 |
| 20:15.09 | caloi | enabled on that IP |
| 20:15.12 | ComputerWarm | hello all quick question |
| 20:15.19 | caloi | would like that output to a file |
| 20:15.26 | caloi | not everything on the screen |
| 20:15.35 | caloi | sip debug peer |
| 20:15.36 | ComputerWarm | with asterisk/manager interface is there anyway to pull the active calls... and see which is connected and whats not? |
| 20:15.39 | Corydon-w | caloi: that is the only way to do it |
| 20:15.49 | caloi | thanks |
| 20:15.58 | ManxPower | caloi, there is no way to do that, about as close as you can get is by setting options in /etc/asterisk/logger.conf |
| 20:16.21 | Corydon-w | ManxPower: yeah, but he wants non-logger output |
| 20:16.45 | ManxPower | Corydon-w, he should run a terminal with scrollback then! |
| 20:16.53 | plexdk | Corydon-w: i dont quite follow you.. do you mean i have to set it in this extension: exten => 101,4,VoiceMail,u101 |
| 20:16.57 | Corydon-w | ManxPower: or script(1) |
| 20:17.01 | Corydon-w | plexdk: No |
| 20:17.08 | Corydon-w | plexdk: read what I said. |
| 20:17.12 | ManxPower | plexdk, READ THE BOOK! |
| 20:17.15 | ManxPower | ~book |
| 20:17.16 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 20:17.29 | plexdk | k |
| 20:19.40 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 20:23.38 | knarfly_wk | anyone offer some help on my dialplan? |
| 20:23.56 | hmmhesays | depends on how fast you want it done |
| 20:25.10 | knarfly_wk | well, I just added a DID and the PSTN provider has been a big help but they left me hanging |
| 20:25.22 | Assid | ManxPower: you using polycom? |
| 20:25.59 | ManxPower | The drain for the condenser for the AC unit in the computer room at one of my client's offices stopped up, the float shut down the A/C this weekend. Dead server this morning. |
| 20:26.03 | knarfly_wk | when I receive a call into my * server coming in on my DID (VOIP) not FXO/FXS, I want to play a message that says please enter the extensions you want to call |
| 20:26.19 | *** part/#asterisk ComputerWarm (n=donc@209.29.159.11) |
| 20:26.25 | Assid | your computer dies with no AC ? |
| 20:26.30 | knarfly_wk | when the user dials that extension I want it to be transferred to that extension |
| 20:26.34 | ManxPower | Assid, Hard driver burned up |
| 20:26.50 | knarfly_wk | I followed the docs but it's not working and the PSTN guy says he can't help |
| 20:26.54 | Assid | seriously??? wow.. which hdd was it |
| 20:27.01 | ManxPower | knarfly_wk, your questions are SO basic, it's pretty obvious you have not read much. |
| 20:27.06 | Assid | knarfly: check out the docs on IVR |
| 20:27.13 | ManxPower | Assid, The main one. |
| 20:27.22 | Assid | er.. as in what make? |
| 20:27.26 | Shaun2222 | lol |
| 20:27.35 | Shaun2222 | what was the server in a box? |
| 20:27.41 | ManxPower | Assid, Hell if I know, I'm a 7 hr drive away from it. |
| 20:27.56 | knarfly_wk | ManxPower: That's your opinion....and please stfu if you don't have any answers |
| 20:28.04 | Assid | hehe |
| 20:28.05 | ManxPower | Shaun2222, the server was in a closet. It's the largest NOC the office manager would allow. |
| 20:28.24 | Qwell[] | knarfly_wk: Good luck getting any help... |
| 20:28.31 | knarfly_wk | I've been reading docs and trying the examples and I read seom more but it still doesn't work like it says it supposed to |
| 20:28.33 | Qwell[] | attitudes should be checked at the door |
| 20:28.34 | Shaun2222 | ManxPower: ah, ya guess it could get pretty hot in a small closet |
| 20:28.39 | ManxPower | knarfly_wk, Oh, I can help you, but I would spend most of my time teaching you the basics before we could actually accomplish your task |
| 20:28.44 | obiwanmikenolte | Haha. knarfly's got claws. Next, he'll tell us that he's using ASterisk@Home |
| 20:28.57 | ManxPower | Shaun2222, I'm suprized the Cisco Cayalyst switch didn't burn up too. |
| 20:28.58 | Assid | dammit.. is it me or shouldnt the name be shown if i have the user in my directory? |
| 20:29.03 | Assid | like a cell phone |
| 20:29.12 | knarfly_wk | ManxPower: stop proving my point |
| 20:29.21 | Qwell[] | Assid: heh, my cell phone only checks the last 7 digits |
| 20:29.36 | Assid | i think most cell phones do that.. |
| 20:29.46 | Qwell[] | so if you get a call from somebody in a diff area code... |
| 20:29.52 | Qwell[] | it flags it as that user |
| 20:30.08 | knarfly_wk | if you could help you would have asked something besides I have read the docs....did you not see that I have everything working but when I added the DID I cannot get that part working...I'm sick of people saying read the docs.... |
| 20:30.18 | Assid | wow.. never had that happen to me before |
| 20:30.35 | Qwell[] | Assid: it's quite humourous at times |
| 20:30.47 | knarfly_wk | kma |
| 20:31.19 | Assid | Qwell[]: what about the ip phones.. |
| 20:31.26 | Qwell[] | Assid: dunno |
| 20:31.44 | Assid | shouldnt they show the caller id name if the person is in my directory? |
| 20:31.52 | obiwanmikenolte | knarfly: in the context that the DID is using to come into your dialplan, put: exten => DID,1,Background(whatever-the-hell) |
| 20:31.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.172) |
| 20:32.17 | ManxPower | Would be nice if pastebin.ca worked |
| 20:32.32 | ManxPower | Assid, I guess that would depend on the PHONE you are using |
| 20:32.33 | Assid | ManxPower: you could try pastebin.com too |
| 20:32.38 | obiwanmikenolte | PS Don't really put DID. Put the numbers that are passed |
| 20:32.42 | Skinzy | or paste.skinzy.com :p |
| 20:32.47 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 20:32.53 | ManxPower | My IVR: http://pastebin.ca/121200 |
| 20:33.27 | Assid | any way of getting the polycoms (301 in particular) to do that |
| 20:33.53 | ManxPower | obiwanmikenolte, assuming of course that the DID provider sends the dialed number, and assuming that the context the provider is in is the same as where the exten line is. |
| 20:34.07 | Qwell[] | and that the peer is setup right in the first place |
| 20:34.26 | ManxPower | Qwell, *nod* That too. |
| 20:34.35 | obiwanmikenolte | ManxPower: You're going to confuse the guy, and he seems unhinged already. |
| 20:34.36 | Qwell[] | all of which could very easily be false |
| 20:34.55 | Assid | false false false false |
| 20:35.01 | obiwanmikenolte | Haha. We have to give him doubt benefits here |
| 20:35.21 | obiwanmikenolte | PRETEND like we aren't vainglorious wankers |
| 20:35.46 | Qwell[] | obiwanmikenolte: You must be new here |
| 20:35.59 | ManxPower | obiwanmikenolte, I'll bet you a copy of my [macro-std-exten] that he doesn't have any of that right. |
| 20:36.36 | obiwanmikenolte | Haha. Thanks, but we probably downloaded it from the same place anyway |
| 20:36.38 | Assid | i think maybe the polycoms dont support it |
| 20:36.51 | ManxPower | obiwanmikenolte, Mine is not downloaded from anyehere. |
| 20:37.03 | obiwanmikenolte | And now, I'm chastised for trying to actually help someone... |
| 20:37.04 | ManxPower | lovingly hand crafted byte by byte |
| 20:37.14 | obiwanmikenolte | Whoa! And apparently, I'm British |
| 20:37.36 | ManxPower | obiwanmikenolte, Hint: start out by finding out what his Asterisk CLI Terminal says when a call comes on on his DID. |
| 20:37.38 | Qwell[] | british? |
| 20:38.26 | hmmhesays | hmm for some reason this freepbx box is not showing iax2 registries |
| 20:38.28 | obiwanmikenolte | Chastise. I know that it's correct, but it sure looks British, like "colonise", "theatre", |
| 20:38.37 | Juggie | ~seen theplot |
| 20:38.51 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 1d 2h 40m ago, saying: 'I did set in the address field to match the username too'. |
| 20:38.51 | obiwanmikenolte | "mayonnaise", "colour", etc. |
| 20:38.51 | hmmhesays | it should show something if you have register lines in the iax_additiona.conf |
| 20:38.51 | hmmhesays | iax_additional.conf |
| 20:38.53 | Qwell[] | meh, you were't chastised |
| 20:38.56 | Assid | okay definitely think it doesnt suport now |
| 20:39.03 | ManxPower | hmmhesays, You are using FreePBX? Pervert. |
| 20:39.05 | obiwanmikenolte | Qwell. Yes-huh |
| 20:39.10 | Assid | i cant findit |
| 20:39.11 | Qwell[] | when? |
| 20:39.19 | obiwanmikenolte | Hahaha |
| 20:39.23 | ManxPower | Assid, look in the admin guide. |
| 20:40.29 | Assid | okay i think i see it |
| 20:40.30 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 20:41.07 | ManxPower | I thought the plot was total world domination? Did I miss ANOTHER memo? |
| 20:41.16 | caloi | anyone doing route advances in sip on asterisk? |
| 20:41.30 | obiwanmikenolte | caloi: RTFM, n00b!!!!111 |
| 20:41.42 | obiwanmikenolte | : ) |
| 20:41.43 | caloi | ?? |
| 20:41.46 | hmmhesays | ManxPower: it has its place |
| 20:42.12 | caloi | i've RTFM - just asking a question |
| 20:42.20 | ManxPower | caloi, define "route advances" |
| 20:43.15 | caloi | if i get 3 routes back from ITSP in my 300 multiple choices for routing, if the first route to destination fails, is it possible to failback to the second choice? |
| 20:43.35 | obiwanmikenolte | Haha. I was referring to a conversation moments ago. I'm sure it scrolled past by now. PS Is there any way to make jokes more obvious? If so, please note them. |
| 20:44.57 | Mercestes | obiwanmikenolte: j/k sometimes helps. |
| 20:45.13 | obiwanmikenolte | I did the colon-close paren.! |
| 20:45.17 | caloi | Mercestes: good point :) |
| 20:45.33 | Qwell[] | obiwanmikenolte: the space caused a parse error... |
| 20:45.42 | hmmhesays | anyone know if iax2 registry should show something even if you're failing to register? |
| 20:45.56 | ManxPower | caloi, See "show application dial". Pay special attention to DIALSTATUS. Asterisk will not do LCR / Failover routing by itself, but you can do it with clever macros in the dialplan |
| 20:45.59 | obiwanmikenolte | Qwell: I see now. Thanks. Err... I mean: thx:) |
| 20:46.21 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 20:46.25 | caloi | ManxPower - thanks for the info |
| 20:46.55 | ManxPower | caloi, there's also something called "superdial" macro, but I've never used it. |
| 20:48.23 | caloi | ill take a look |
| 20:48.26 | caloi | thanks |
| 20:48.36 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 20:48.55 | ManxPower | caloi, The major problem with doing failover is that a call can fail in ways that, as far as Asterisk is concerned, are not failures. |
| 20:49.05 | froguz | are loopstart and groudstart signalling methods for analog telephony? is there another signalling concept in analog phones? |
| 20:49.17 | salaud | I'm getting "connect refused" when trying do a FastAGI ... the FastAGI daemon is running and I can telnet to it |
| 20:49.17 | Nugget | telnet is eeeeeeevil! |
| 20:49.19 | *** join/#asterisk linlin (n=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 20:49.23 | ManxPower | frodonbash, yes. and yes. |
| 20:49.35 | frodonbash | lol |
| 20:49.46 | caloi | ManxPower true, but shouldn't asterisk know a 503 Service Unavailable is a failed call? |
| 20:49.53 | salaud | Anyone know what might be preventing asterisk agi from connecting? |
| 20:50.11 | ManxPower | Loopstart, Groundstart, E&M, are all signaling protocols for analog. Voice T-1s also support these protocols. |
| 20:50.11 | salaud | or how to test things? |
| 20:50.13 | ManxPower | caloi, yes, assuming your provider sends that back for a failed call. |
| 20:50.35 | ManxPower | caloi, what if the far server is offline, the call will fail, but asterisk will wait for the timeout before considering the call failed. |
| 20:50.39 | caloi | in this scenario I am the ITSP and the UA - Working on some interop |
| 20:50.44 | sb_mx | hey guys. in sip.conf for example, you're able to define a set of codecs with "allow = alaw", "allow = ulaw" etc. my question is if alaw has priority 1 (being the first line in sip.conf) when will it use ulaw? will it use it when the other side doesnt have alaw available? |
| 20:51.23 | ManxPower | sb_mx, you almost never want both alaw and ulaw |
| 20:51.28 | froguz | ManxPower, T-1's too ??? doesn't T1 use CAS signalling?? |
| 20:51.35 | hmmhesays | haha, wow freebpx shredded that iax2 config up |
| 20:51.50 | salaud | anyone know anything about FastAGI? |
| 20:51.50 | ManxPower | froguz, CAS to emulate loopstart, e&M, etc. |
| 20:51.50 | froguz | sb_mx, yes |
| 20:52.21 | froguz | thanks ManxPower |
| 20:52.48 | sb_mx | ManxPower, yeah I thought so since alaw is the "carriers' choice". however we set alaw first since ulaw has core-dumped our asterisk several times now (yes, we havent been able to compile asterisk with the proper flags to post the bug since its a production environment). |
| 20:53.23 | *** part/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 20:53.37 | froguz | ManxPower, wich other signalling concepts do exist in analog telephony? |
| 20:53.58 | sb_mx | anyway, I just did a "sip show channel" and i can see alaw, ulaw and unkn. if that is the case, then we're gonna have to restrict the codec to alaw only :S |
| 20:55.48 | ManxPower | froguz, Just remember that Digium analog cards only support loopstart, IIRC |
| 20:55.49 | ManxPower | froguz, winkstart |
| 20:56.22 | ManxPower | froguz, POTS lines in the USA are all loopstart w/disconnect supervision (called kewlstart by Digium) |
| 20:56.30 | salaud | FastAGI anyone? |
| 20:56.52 | froguz | ManxPower, never heard about winkstart. i understand the difference between loop and ground start, but i can't find any information related to kwell start. |
| 20:57.19 | ManxPower | salaud, No thanks, but I'll take "Questions Without Enough Information for $500, Alex." |
| 20:57.34 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:57.36 | ManxPower | froguz, kewlstart is not a telco term, it is a Digium term |
| 20:57.44 | mog | nope |
| 20:57.47 | mog | it is a telco term |
| 20:57.49 | mog | google it |
| 20:57.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:58.03 | mog | you will find other references then ones relating to asterisk / digium |
| 20:58.06 | ManxPower | mog, try ordering it from BellSouth |
| 20:58.29 | ManxPower | kewlstart = loopstart with far end disconnect supervision" |
| 20:58.31 | mog | i have been on support calls for customers will bellsouth tech new what it was |
| 20:58.43 | mog | when i said kewlstart he knew what i was talking about |
| 20:58.46 | mog | maybe sales doesnt |
| 20:59.00 | mog | i mean the longer term may be more common |
| 20:59.10 | mog | but kewlstart is still real term not created by us to confuse |
| 20:59.44 | froguz | if it's just related to digium, how can i use it when i connected to the chilean PSTN? in fact is the siganlling method that works better for me |
| 21:00.03 | *** join/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 21:00.04 | mog | no he means the term |
| 21:00.11 | mog | not what it discribes |
| 21:00.18 | mog | most people do what it describes or offers it |
| 21:00.27 | mog | not every one knows the term kewlstart |
| 21:00.32 | mog | although it is a true term |
| 21:00.58 | froguz | aps |
| 21:01.00 | Jon335 | How do you make a dial pattern that matches 5551212 but sends 5555551212? |
| 21:01.00 | mog | its like how not everyone knows what dihydrogen monoxide is but every one knows what water is |
| 21:01.34 | ManxPower | Jon335, exten => 5551515,1,Dial(Zap/1/555${EXTEN}) |
| 21:02.19 | Jon335 | ManxPower: Thanks |
| 21:02.38 | froguz | mog is kewelstart AKA other term? |
| 21:02.44 | mog | no |
| 21:02.47 | ManxPower | Jon335, extensions.conf.sample did not show that? |
| 21:02.54 | mog | everyone just says loops start with disconnect supervision |
| 21:02.55 | mog | instead |
| 21:03.04 | mog | kewlstart is just easier to say |
| 21:03.07 | *** part/#asterisk Jon335 (i=Jon335@unaffiliated/jon335) |
| 21:03.15 | froguz | aaaaaaaaaaahhhhh, that's what i wanted to read |
| 21:03.36 | froguz | loops start with disconnect supervision, jejeje... so, that's the difference |
| 21:03.49 | mog | yes |
| 21:03.54 | mog | they are the same though |
| 21:04.33 | froguz | and does irt prevent from the hence issue known in loop start? |
| 21:04.54 | froguz | i mean |
| 21:04.56 | froguz | glare |
| 21:05.12 | froguz | the glare issue |
| 21:05.37 | ManxPower | froguz, no, it does not prevent glare. |
| 21:06.07 | ManxPower | far end disconnect supervision prevents asterisk from keeping your phone line open for days because it did not detect when the far end disconnected. |
| 21:06.23 | *** join/#asterisk hohum (n=dcorbe@12.195.58.238) |
| 21:07.35 | *** join/#asterisk wagtag (n=wagtag@adsl-70-132-18-173.dsl.snfc21.sbcglobal.net) |
| 21:08.11 | froguz | good to learn with you guys |
| 21:08.17 | hmmhesays | what is the accounting term for a service estimate? |
| 21:08.35 | hohum | "service estimate?" |
| 21:08.53 | hwt | in my realtime setup, howcome sip friends are listed as Nat=N even though i have Nat=yes in the database? |
| 21:08.56 | mmealling | quote |
| 21:09.08 | hwt | also i have problems with "mysql server has gone away". |
| 21:09.14 | hwt | how can i increase the mysql socket timeout? |
| 21:09.41 | hohum | stop using MySQL for realtime |
| 21:09.43 | hmmhesays | an estimate of the cost of future services to be rendered |
| 21:09.47 | hohum | ODBC is your friend |
| 21:10.04 | hohum | if you insist on having MySQL as a back end then use a MySQL ODBC driver |
| 21:10.25 | hwt | hohum: i suspect it's the CDRs that gets the timeout problems. |
| 21:10.37 | hwt | which is possibly worse. |
| 21:10.42 | hohum | yes, that is bad |
| 21:10.49 | hwt | hohum: is that more robust in any way? |
| 21:10.53 | hwt | odbc. |
| 21:10.55 | hohum | and ODBC does a much better job of connection management than the MySQL drivers do |
| 21:11.19 | hohum | you can still use MySQL, through ODBC |
| 21:11.41 | hwt | even for cdrs? |
| 21:11.48 | hohum | yes, even for CDRs |
| 21:12.06 | hwt | k. |
| 21:12.08 | ManxPower | one would hope you are not generating more than 10 cdrs per second |
| 21:12.12 | hwt | maybe i should check that out. |
| 21:12.36 | hohum | Manx: the problem is he isn't generating enough CDRs to keep the MySQL connection alive |
| 21:12.48 | hohum | not enough data going back and fourth |
| 21:13.05 | hohum | I had the same problem until I switched to ODBC |
| 21:13.30 | hwt | but the mysql driver handles more cdrs than the odbc driver? |
| 21:13.41 | hohum | hwt: who said that? |
| 21:13.50 | RoyK | hohum: we've been using mysql support from -addons for more than two years without problems |
| 21:14.01 | hohum | RoyK: k |
| 21:14.15 | RoyK | both on 1.0.x and 1.2.x |
| 21:14.19 | hwt | hohum: ManxPower. |
| 21:14.20 | RoyK | and quite high traffic |
| 21:14.25 | hohum | RoyK: good for you, but I like to have the ability to migrate from back-end to back-end |
| 21:14.39 | ManxPower | ODBC, of course, would be expected to be "slower", but unless you have a MASSIVE number of calls it should not matter. |
| 21:14.43 | RoyK | hohum: how? |
| 21:14.43 | hwt | RoyK: the problem here is, actually, that i have too little traffic. :) |
| 21:14.52 | hohum | RoyK: ODBC drivers |
| 21:14.56 | hohum | like for example |
| 21:15.04 | ManxPower | so the real issue is which do you prefer, native, or ODBC |
| 21:15.11 | RoyK | hwt: ok, what sort of traffic do you expect will limit mysql? |
| 21:15.21 | hohum | if I feel like I don't want to have Oracle's cock firmly implanted in my ass, I'll migrate away from MySQL to something more scalable like Postgres |
| 21:15.29 | hmmhesays | LOL |
| 21:15.30 | ManxPower | one might assume you can set the connection timeout |
| 21:15.53 | hmmhesays | nice mental picture |
| 21:15.57 | hwt | RoyK: huh? the problem is the the mysql connection times out. |
| 21:16.08 | RoyK | doesn't happen |
| 21:16.24 | *** join/#asterisk adorah (n=Administ@84.94.208.77.cable.012.net.il) |
| 21:16.30 | hwt | RoyK: i get a "mysql server has gone away" message. |
| 21:16.41 | hwt | RoyK: in the * console. |
| 21:16.44 | RoyK | in the logs as well? |
| 21:16.52 | RoyK | if so, I can grep through |
| 21:17.05 | salaud | I'm trying desperately to get my AGI command to connect to my FastAGI server... I keep getting "connection refused" anyone got this? or know how to fix it? |
| 21:17.11 | RoyK | I have at least a year's worth logs with full verbosity |
| 21:17.41 | RoyK | salaud: fix the fastagi server? |
| 21:17.46 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 21:17.51 | ManxPower | salaud, sounds like your server isn't working. |
| 21:18.00 | salaud | I connect to my server via telnet... no problem |
| 21:18.05 | hwt | RoyK: Jul 25 07:42:53 ERROR[6926] cdr_addon_mysql.c: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away |
| 21:18.19 | hwt | RoyK: in messages |
| 21:18.39 | hwt | RoyK: (/var/log/asterisk/messages) |
| 21:18.51 | RoyK | i kinda understood that :P |
| 21:19.03 | salaud | Will "connect refused" happen even if it can connect but has some protocol error? I'm using the example from the FastAGIDemo website |
| 21:19.04 | hwt | RoyK: i hope that it will actually try to reconnect, and not only pass the cdr to /dev/null. |
| 21:19.19 | hwt | RoyK: i know, sorry. :) |
| 21:19.49 | RoyK | hwt: I only have those logs since mid march, but no matches since then |
| 21:20.09 | hwt | RoyK: it could be nasty if it does not reconnect. |
| 21:20.17 | hwt | RoyK: hey, btw, do you use realtime sip friends? |
| 21:20.25 | RoyK | yes |
| 21:20.38 | hwt | RoyK: my friends are listed as nat=N, even though it says nat=y in the database. |
| 21:21.07 | hwt | RoyK: is that only a listing bug, or does it actually look at the IP adress of the SIP singalling? |
| 21:21.45 | hwt | RoyK: what does it say in your sip show peers for nat=y-friends? |
| 21:22.11 | hwt | i suspects it's only a listing bug, but on the other hand i do realtime caching. |
| 21:22.17 | RoyK | we have nat=yes for all clients, and they all show up as with nat=always |
| 21:22.27 | *** part/#asterisk javar (n=javar@200.118.174.253) |
| 21:23.16 | hwt | RoyK: strange. |
| 21:23.39 | salaud | ManxPower: RoyK: any additional thoughts on the FastAGI "connection refused" given the information? Or perhaps just a debug route? |
| 21:23.58 | ManxPower | salaud, how about a copy of the FastAGI extension line? |
| 21:24.14 | *** join/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 21:24.24 | salaud | exten => _X.,1,AGI(agi://127.0.0.1) |
| 21:24.39 | salaud | The server is running on the standard port... |
| 21:24.44 | hwt | RoyK: and i also have global nat=yes. |
| 21:24.56 | salaud | This is the code that I am running: http://www.miselconsulting.com/?page=fastagi |
| 21:25.03 | hwt | RoyK: but i haven't had any problems with one-way audio, even on full-cone nats. |
| 21:25.05 | salaud | just trying to get past step one |
| 21:25.13 | RoyK | hwt: doesn't matter. realtime will override that |
| 21:25.23 | RoyK | hwt: what version? |
| 21:25.24 | ManxPower | salaud, define "standard port" |
| 21:25.31 | salaud | 4577 |
| 21:25.39 | hwt | RoyK: |
| 21:25.46 | hwt | RoyK: 1.2.9.1 |
| 21:25.56 | RoyK | ok. should work then |
| 21:26.03 | RoyK | try 'sip show peer xxx load' |
| 21:26.21 | salaud | I imagine you can put a different port in the AGI extension if you had to |
| 21:27.36 | hwt | RoyK: Nat : Always |
| 21:27.44 | Mike | guys why can i register with iax if i have host=ip? i really need this for trunking |
| 21:28.07 | hwt | RoyK: but nat=N in 'sip show peers'. |
| 21:28.17 | Qwell[] | because you don't need to register if it knows the host... |
| 21:28.26 | RoyK | hwt: then perhaps there's some old shite in sip.conf..... |
| 21:28.43 | Qwell[] | if the other end knows the host, that is |
| 21:28.47 | *** join/#asterisk bhima (n=bhima@server1.mdsamerica.com) |
| 21:29.01 | Mike | Qwell[], its not working the trunk i have zaptel cards on all my servers, i have trunk=yes and jitter=no |
| 21:29.05 | bhima | Is there a way to get DTMF during calls, after an extension has been selected? |
| 21:29.09 | Mike | Qwell[], its really giving me a hard time |
| 21:29.13 | hwt | RoyK: grep -i nat sip.conf does not reveal anything interesting. |
| 21:29.39 | salaud | is there anyway to get more debug info from app_AGI? |
| 21:29.56 | RoyK | anyway...... pastebin your sip.conf, hwt |
| 21:33.20 | bhima | I'd like to connect two people together, and pay attention to their DTMF activity. |
| 21:34.44 | *** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com) |
| 21:36.53 | bhima | Like, say, if you press "44" it speaks "Your next appointment is in an hour" |
| 21:41.14 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:41.27 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 21:44.23 | salaud | Ok... I have google 100 different ways and can find no information about FastAGI and connection refused... I seem to be getting the message even when the FastAGI server daemon is NOT running |
| 21:45.04 | salaud | When I do AGI() to some other server, I get unreachable |
| 21:46.18 | salaud | or rather... it says it times out.. |
| 21:46.34 | salaud | I'm dead in the water |
| 21:47.23 | bhima | I think most people here are too busy drooling over His Steveness's latest toys to pay attention to the channel. :) |
| 21:47.53 | *** part/#asterisk roguebug (n=johnw2@dip-106-210.bras.dsl.breisnet.com) |
| 21:48.12 | salaud | in other words... be nice |
| 21:49.02 | ManxPower | sounds to me like asterisk is trying to connect to a different port than you think it's connecting to. specify the port. |
| 21:49.31 | salaud | ManxPower: I have tried that also... any port I specify on the localhost gives the same message whether the FastAGI daemon is running or not |
| 21:49.42 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 21:49.51 | ManxPower | salaud, any iptables/ipchains installed on the system? |
| 21:50.00 | ManxPower | use tcpdump to see what's going on |
| 21:50.18 | salaud | ManxPower: possibly ... it could be that kind of thing... |
| 21:51.18 | salaud | ManxPower: perhaps messages log would have dropped packets... I'll go down that road and see where that leads. I really want to know if there is someway to increase the level of debugging in app_agi (I did 'agi debug' already) |
| 21:53.56 | salaud | ManxPower: one other piece of info... I can telnet from the localhost to the FastAGI server daemon with no problem... |
| 21:54.16 | salaud | I checked iptables and everything is ACCEPT |
| 21:54.34 | ManxPower | salaud, what is your telnet command? |
| 21:54.34 | salaud | starting to look like a bug |
| 21:54.34 | Juggie | sigh |
| 21:54.39 | Juggie | can you telnet to it. |
| 21:54.53 | Juggie | eg, telnet fastagiserver:port |
| 21:54.56 | salaud | Juggie: yes... see above |
| 21:55.05 | Juggie | fastagi works fine. |
| 21:55.27 | salaud | telnet 127.0.0.1 4577 |
| 21:55.28 | ManxPower | salaud, what is your telnet command? |
| 21:55.48 | salaud | I connect fine and the daemon sees me and I when I disconnect it cleans up |
| 21:55.55 | ManxPower | good, now force fastagi to connect to that port. |
| 21:56.07 | Juggie | 4577 isnt the standard port. |
| 21:56.34 | Juggie | so, agi://127.0.0.1:4577 |
| 21:56.45 | ManxPower | salaud, what makes you think 4577 is the standard port. |
| 21:56.58 | ManxPower | if you had forced it in the first place... |
| 21:57.25 | Juggie | #define AGI_PORT 4573 |
| 21:57.27 | Juggie | hmmmm |
| 21:57.40 | salaud | ManxPower: The damn FastAGIDemo online has 4577 |
| 21:57.56 | salaud | http://www.miselconsulting.com/?page=fastagi |
| 21:58.07 | ManxPower | did you try official digium docs? |
| 21:58.22 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 21:58.27 | rene1 | hello |
| 21:58.32 | salaud | give me two secs... I don't think this is going to solve the problem though |
| 21:58.40 | Juggie | oh i'm pretty sure it will |
| 21:58.47 | rene1 | what is the difference of a channel definition like SIP/109 versus SIP/109-41205d38 |
| 21:58.59 | salaud | because I get connection refused on all ports... and timeout on any other host.. |
| 21:59.00 | rene1 | the first one is available as ${CHANNEL} |
| 21:59.03 | salaud | BRB |
| 21:59.04 | RoyK | rene1: the latter is the name with the uniqueid |
| 21:59.04 | rene1 | how do i know the second one |
| 21:59.11 | ManxPower | salaud, a basic tcpdump would have shown you what port it's connecyting to |
| 21:59.25 | RoyK | Juggie: let him think it's a bug :P |
| 21:59.42 | salaud | ManxPower: That's true... there is a bug in the http://www.miselconsulting.com/?page=fastagi |
| 21:59.44 | Juggie | dont get me wrong, in 1.2 res_agi is somewhat broken |
| 21:59.45 | salaud | ;) |
| 21:59.53 | Juggie | but it does work :) |
| 22:00.00 | ManxPower | salaud, then report it to the author, not us. |
| 22:00.28 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.199) |
| 22:00.31 | ManxPower | I still don't understand what the problems with "show application agi" is |
| 22:00.32 | rene1 | RoyK: |
| 22:00.35 | rene1 | how is it calulcated |
| 22:00.41 | rene1 | calculated |
| 22:00.42 | Juggie | i'm off to finish cleaning the washroom which weirdly enough is more exciting then salaud's bugs ;) |
| 22:00.47 | RoyK | rene1: iirc by random |
| 22:00.53 | twisted[asteria] | RANDOM! |
| 22:01.03 | salaud | Juggie: be nice... this application is going to help people |
| 22:01.04 | Qwell[] | skinny isn't random :D |
| 22:01.04 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 22:01.17 | Qwell[] | well, okay, the protocol is |
| 22:01.23 | Qwell[] | but the callid isn't :P |
| 22:01.49 | AJaymn | I have a user using a Linksys PAP2 over the internet, after about 5 mins hes unable to receive calls.. It still rings on the callers end but he doenst hear anything.. HE can still make outbound calls... Any IDEAS? |
| 22:02.04 | ManxPower | salaud, will it help me place more cash in my bank account? |
| 22:02.29 | ManxPower | AJaymn, classic nat timeout issue. |
| 22:02.37 | salaud | ManxPower: Perhaps it may save you tax dollars |
| 22:02.39 | ManxPower | AJaymn, enable the net keep alive in the sipura |
| 22:02.47 | [TK]D-Fender | AJaymn : set "qualify=yes" for the peer |
| 22:02.48 | twisted[asteria] | AJaymn, no, my IDEAS have all gone out the window, but i have an idea ;) |
| 22:02.50 | ManxPower | salaud, HAHAHAHAHA!!!!! |
| 22:02.51 | salaud | anyway... community... that's what this channel is about anyway |
| 22:03.13 | ManxPower | ending the war in iraq or the war on (some) drugs will save my tax dollars |
| 22:03.19 | twisted[asteria] | nonono |
| 22:03.32 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 22:03.33 | ManxPower | qualify=yes is asking for trouble |
| 22:03.34 | twisted[asteria] | what will save your tax dollars are people on welfare getting real fucking jobs |
| 22:03.43 | ManxPower | better to set the NAT keep alive on the sipura |
| 22:04.00 | ManxPower | twisted[asteria], a tiny amount compared to the two I referenced. |
| 22:04.07 | salaud | btw, that port change seemed to work. But... I'll try to send the app_agi developers some notice that error messages are a little wacked |
| 22:04.34 | Qwell[] | uhh... |
| 22:04.36 | salaud | I would expect to have gotten a timeout if I couldn't reach the port, just as it does for any other host it can't reach |
| 22:04.40 | docelm0 | AJaymn, yes NAT Keep alive |
| 22:04.40 | Qwell[] | "COnnection refused" is not "wacked" |
| 22:05.02 | salaud | Qwell[]: It is when it gives you a different message when it can't reach a host/port |
| 22:05.11 | twisted[asteria] | ManxPower, i dunno... i think on average welfare spending on people who wind up with cadillac escalades far outweighs the amount of spending in the war on drugs... |
| 22:05.15 | Qwell[] | salaud: telnet localhost 3824 |
| 22:05.20 | Qwell[] | You'll get a connection refused |
| 22:05.30 | ManxPower | salaud, timeout would mean the host did not respond. connection refused means the host responded |
| 22:06.10 | salaud | ManxPower: If there is nothing running on a port, you still get a response? |
| 22:06.17 | Qwell[] | uhh..yeah |
| 22:06.21 | Qwell[] | it says "no, go away" |
| 22:06.38 | Qwell[] | salaud: like I said, telnet to a random port on your machine, and watch what it does |
| 22:07.08 | salaud | Qwell[]: Hmm... even responding at all when someone is trying to compromise your system by scanning ports would be bad |
| 22:07.26 | Qwell[] | salaud: Tell it to the inventor of tcp/ip |
| 22:07.26 | salaud | But that answers the question... and that's good info |
| 22:08.03 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 22:10.18 | ManxPower | twisted[asteria], Um, the cost of the war on drugs in 1999 was over $30 billion |
| 22:10.45 | ManxPower | and the cost of welfare 2005 was under $25 billion |
| 22:10.45 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:11.56 | ManxPower | the $30 billion was just DIRECT costs, it does not factor in the other costs like prosecution, prisons, etc |
| 22:12.07 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 22:12.11 | *** join/#asterisk queuetwo (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 22:12.16 | salaud | ManxPower: that's the truth... |
| 22:12.21 | bhima | twisted: Do you have any actual stats that lead you to this belief, or is it just based on the stereotype that black people are on welfare and black people like cadillac escalades? |
| 22:13.08 | ManxPower | at least welfare does SOME good. |
| 22:13.12 | salaud | In the 1930's there two choices for the established elite in the US... 1) Welfare 2) Revolution |
| 22:13.36 | salaud | They choose option 1 and thus we de did not have a workers revolution here. |
| 22:13.39 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 22:14.05 | ManxPower | AND current welfare lifetime benefits cap out at 3 years. |
| 22:14.09 | salaud | Paying off people to stay in there place is a great strategy from a Neo Con perspective |
| 22:14.17 | salaud | s/there/their |
| 22:14.20 | ManxPower | for what most people would consider "welfare" |
| 22:15.15 | ManxPower | that cap doesn't apply to retarded people on SSI, or assistance to children, etc. |
| 22:15.21 | ManxPower | etc. |
| 22:15.27 | salaud | So if you are a right winger ... look at the bright side of welfare... at least you are able to still be able to have your lifestyle without be affected by some pesky revoultion |
| 22:15.49 | ManxPower | one of my boyfriends is in the Social work Phd program at a university. |
| 22:18.32 | Qwell[] | ManxPower: one of? |
| 22:19.10 | salaud | ManxPower: must be nice... finding time for relations between fits of asterisk hacking |
| 22:19.23 | froguz | /quit Spawn extension (normal, 98544419, 1) exited non-zero on 'SIP/140-6b79' |
| 22:19.30 | ManxPower | Qwell, correct, one of. |
| 22:20.10 | Qwell[] | tsk, tsk |
| 22:20.24 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 22:20.25 | ManxPower | Qwell, they are friends. |
| 22:22.12 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:23.38 | *** join/#asterisk angom_w (n=angom@red-corp-201.130.139.249.telnor.net) |
| 22:24.59 | salaud | crap.... has anyone tried to pass encode variables in an agi:// URL? Of course, http://www.miselconsulting.com/?page=fastagi is wrong about this too... I tried AGI(agi://localhost?somevariable=test) it don't work... "can't locate host" |
| 22:25.16 | *** join/#asterisk Nitrus^ (n=ntisog@72-34-76-86.skyriver.net) |
| 22:25.37 | salaud | I wrote the guy who made that example document and told him about the errors.. RTFM screw that WTFM |
| 22:25.39 | Nitrus^ | how do i get music on transfer working? is there something in features.conf? |
| 22:26.02 | ManxPower | salaud, tried AGI(agi://localhost/?somevariable=test) |
| 22:26.08 | ManxPower | or AGI(agi://localhost/somevariable=test) |
| 22:26.17 | ManxPower | salaud, treat it JUST LIKE HTTP |
| 22:26.36 | salaud | I did... ?somevariable=test |
| 22:26.44 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 22:26.47 | ManxPower | In theory you shouldf be ablle to do a GET VARIABLE |
| 22:27.08 | ManxPower | set the variable in the dialplan, then GET VARIABLE in your AGI |
| 22:27.10 | salaud | ManxPower: I was trying to avoid that extra operation if at all possible.. but, if I have to |
| 22:28.43 | *** join/#asterisk pbd (n=PDavidso@12.144.118.36) |
| 22:31.02 | *** join/#asterisk ppdm (i=pdm@pdm.sh) |
| 22:31.21 | salaud | It worked with agi://localhost/somevariable=test Remind me to be even MORE skeptical of documentation and examples I find online |
| 22:34.32 | *** part/#asterisk Mike (n=mike@201.112.50.158) |
| 22:34.49 | *** part/#asterisk angom_w (n=angom@red-corp-201.130.139.249.telnor.net) |
| 22:35.23 | puppet | What do people think about this? http://h10010.www1.hp.com/wwpc/us/en/ss/WF05a/15351-241434-241475-241475-f78-446006.html for freepbx and some other? |
| 22:35.36 | ppdm | anyone ever have a 7961 with a busted display? |
| 22:36.07 | pbd | Not I- and I've got about 250 of them in production. What did you do, step on it? |
| 22:37.00 | Nitrus^ | im trying to dial out on an fxs group with four lines and only one is taken but i keep showing that there is congestion, any thoughts? |
| 22:38.25 | ppdm | no? |
| 22:38.28 | ppdm | i was configuring it |
| 22:38.30 | ppdm | and now |
| 22:38.31 | ppdm | no display |
| 22:38.34 | ppdm | its brand new |
| 22:38.35 | ppdm | its fucked up |
| 22:39.48 | AJaymn | how much bandwidth does a IAX connection use for 1 call? |
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| 22:40.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:40.16 | ManxPower | AJaymn, what codec? |
| 22:40.39 | ManxPower | You can sort of roughly assume 16k of overhead |
| 22:40.40 | AJaymn | gsm? |
| 22:40.45 | delmar | Hey everyone... I'm having audio volume issues via TDM FXO. VoIP calls between * systems (polycoms on each system) sound great. FXO call volumes are always low... for both the caller and the party on the SIP phone.. I have ramped up the RXGain and TXGains and am starting to get some improvement but I fear if I ramp it up any more.. I will start to get echo issues... what else can I do to increase volume apart from the phone itself? |
| 22:41.09 | ManxPower | AJaymn, gsm = about 13k |
| 22:41.21 | ManxPower | so about 29kbps |
| 22:41.47 | *** join/#asterisk froguz (n=xxxxx@pc-120-155-104-200.cm.vtr.net) |
| 22:41.52 | delmar | The problem with increasing volume on the phones is.. when an IAX call comes in.. the volume is too loud.. has to be decreased on the phone back to normal... when the FXO call comes in.. its too quiet.. so the volume needs to go up again. |
| 22:41.54 | ManxPower | ~bandwidth |
| 22:42.00 | jbot | extra, extra, read all about it, bandwidth is This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 22:42.19 | ManxPower | Well, THAT is a useless jbot entry |
| 22:42.33 | ManxPower | delmar, you will need to play with the fxo gains |
| 22:42.34 | delmar | ManxPower, im using gsm here for testing. I know g729 is supposed to be better, but im finding gsm pretty damn cool |
| 22:42.38 | froguz | russellb, why isn't the autoconf and menuselect branch svn available?? |
| 22:42.40 | delmar | ManxPower, yeah i have been... |
| 22:43.25 | russellb | froguz: because it is in the trunk now |
| 22:43.27 | delmar | ManxPower, im up to ... RX 8.5 and TX 3.5 now and ive never had to have them that high... im getting a bit worried about the increasing volume of the side-tone |
| 22:43.42 | froguz | jihaa |
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| 22:43.48 | delmar | ManxPower, the sidetone is getting quite noticable the more I increase it. soon it will be good volume but unusable |
| 22:44.10 | ManxPower | delmar, does a regular analog phone have any issues when you plug it into the line? |
| 22:44.18 | delmar | ManxPower, none what so ever |
| 22:44.26 | froguz | russellb, does that autoconf detect the cpu architecture??? |
| 22:44.35 | ManxPower | I had a similar issue. turned out to be a partial short sonehwere on the telco side. |
| 22:44.41 | russellb | it should yes |
| 22:45.27 | delmar | ManxPower, ok. im going to get my old man to jack a phone into the wall again and retest some calls.. perhaps you are right.. if there is a noticable lower volume on the line in general.. perhaps we can log a fault with the Telco |
| 22:45.29 | froguz | that would be great, i don't like to have to uncomment the i586 line in Makefile everytime i build a new embedded system |
| 22:46.27 | froguz | so i'd build an script for that "choose arch".. but autoconf is better |
| 22:47.32 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 22:49.58 | firejon | is anyone using a sangoma card with echo cancellation for fxo? |
| 22:50.17 | froguz | russellb, tell me something... is menuselect for doing manually what autoconf would try to do by automagically? |
| 22:51.42 | russellb | not really, no |
| 22:52.05 | russellb | We have close to 200 modules, and handling each one of them in autoconf is completely unmanagable |
| 22:52.24 | russellb | so we let autoconf do its thing, which is finding libraries and checking if they are usable |
| 22:52.35 | russellb | and then menuselect handles applying that to the asterisk build |
| 22:52.48 | russellb | which also includes an optional menu interface for tweaking to the user's needs |
| 22:53.10 | froguz | great |
| 22:54.35 | wvtrixbox | yes firejon |
| 22:54.41 | wvtrixbox | i'm using a pri config sangoma card |
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| 22:55.04 | froguz | i think 1.4 would be a bigger jump than 1.2 was |
| 22:55.06 | froguz | hope |
| 22:55.38 | firejon | wvtrixbox: i am thinking about getting one... how well does the echo cancellation work? |
| 22:56.02 | wvtrixbox | i haven't had any problems..but getting it setup and working was a bitch |
| 22:56.13 | wvtrixbox | you have to use beta software and it doesn't play well with others |
| 22:56.46 | firejon | define "others" :) |
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| 22:57.25 | froguz | is the ajam http manager in the trunk too? |
| 22:59.13 | wvtrixbox | i've got one tdm400p with 4 fxo, 1 tdm400p with 4 fxs and 2 sangoma pri cards.. sagnoma uses it's own software - wanrouter/wanpipe to load and do the echo can and call progression stuff. Zaptel isn't neede...however when you thrown in the digium devices you have to make sure to load the cards/modules in a precise order or it will fall on it's face |
| 22:59.29 | wvtrixbox | when you do the install a couple times its a breeze just context and setup sensitive |
| 23:00.00 | tasat- | hello all -- I posed this question several hours ago, but I'll try again... I'm looking for a SIP origination/termination provider that fully masks DTMF calls, alternatively, a way for me to cover their mistakes. Any ideas? I'm hearing short noise blips when callers on the other end send DTMF... |
| 23:00.03 | firejon | i see |
| 23:01.10 | firejon | i have just been having non stop echo issues with this Digium TDM card and i am getting sick of messing with it |
| 23:01.45 | froguz | tasat- i haven't had any DTMF problems with BroadVoice, maybe you should try an account |
| 23:01.49 | wvtrixbox | yeah same here..i just use the FXO TDM card as emergency lines in case the PRI fails |
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| 23:02.40 | firejon | but setup aside the echo cancellation works well? |
| 23:02.41 | tasat- | froguz: do they do wholesale? |
| 23:05.24 | froguz | tasat- yep... just look in their website www.broadvoice.com |
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| 23:21.32 | froguz | will IAXy support G.729 in the future? |
| 23:21.49 | Qwell[] | froguz: I highly doubt it |
| 23:22.14 | froguz | why? |
| 23:22.30 | Qwell[] | why would they? |
| 23:22.42 | Qwell[] | it doesn't make a whole lot of sense |
| 23:22.42 | froguz | to save bandwidth?? |
| 23:23.23 | froguz | saving bandwidth is allways a good reason |
| 23:23.52 | froguz | i don't understand why it doesn't sense to you |
| 23:23.54 | Qwell[] | but then you need to make it more powerful, which means the cost goes up... |
| 23:23.57 | froguz | have* |
| 23:24.20 | [TK]D-Fender | Qwell : Yeah... 90$ a port doesn't look so good NOW :) |
| 23:26.04 | froguz | well, that's true. maybe they should build a second edition (for about $130) with more codecs support. |
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| 23:26.31 | froguz | or maybe the should support g.729 just in pass thru mode |
| 23:26.43 | hads | Huh? |
| 23:26.43 | Qwell[] | it's...an endpoint... |
| 23:27.00 | [TK]D-Fender | froguz : Not possible.. the ATA has to do the transform.. its not "in the middle", its THE ENDPOINT |
| 23:27.01 | Qwell[] | What's it going to pass it through...to? |
| 23:27.41 | hads | An analog phone that does g729 of course. |
| 23:27.44 | [TK]D-Fender | froguz : Sipura ATA's already do G729 at far under half the price / port as it is.... |
| 23:27.55 | froguz | well i went confused, you're right |
| 23:28.02 | [TK]D-Fender | And IAX2 has viirtually no savings over SIP for a single channel anyways... |
| 23:28.22 | froguz | [TK]D-Fender, yep. but sipura is not IAX |
| 23:29.30 | [TK]D-Fender | froguz : Correct, it has "usable on every other solution out there", "dual-port", "possibly gateway", "web admin", "g.729", and a whole lot more instead :) |
| 23:29.45 | Qwell[] | [TK]D-Fender: those are all drawbacks |
| 23:30.02 | [TK]D-Fender | IAX2 was not meant for clients... inst Inter-ASTERISK eXchange..... |
| 23:30.22 | [TK]D-Fender | Qwell : yeah, they keep drawing me back to buy more :) |
| 23:30.28 | Qwell[] | :p |
| 23:32.02 | froguz | well i should admit, i love PAP2. but is allways good to count with an IAX alternative, doesn't it? |
| 23:32.47 | froguz | i don't understand why you still refering to sipura, as i know sipura is linksys since a long time ago |
| 23:33.04 | Qwell[] | no, the devices are sipuras still |
| 23:33.27 | Qwell[] | by that logic, why do people call the WRT54G'd Linksys? We all know cisco owns them |
| 23:34.17 | Qwell[] | Why do people still call it skype? It's eBay now |
| 23:35.15 | Qwell[] | same with SuSE...it's Novell now |
| 23:35.19 | Qwell[] | I could go on for weeks |
| 23:35.57 | froguz | mmmm... i have just noticed, sipura does still exist. here in chile nobody sells sipura, we all know their product throw linksys |
| 23:36.56 | *** join/#asterisk unlord (n=nathan@va-65-41-104-14.dyn.embarqhsd.net) |
| 23:37.10 | unlord | I'm trying to figure out if one of my old winmodem's will work as an FXO |
| 23:38.06 | [TK]D-Fender | unlord : Try "load chan_driveryouwriteyourselfbecauseyouclearlyhavewaytoomuchfreetime.so" |
| 23:38.18 | [TK]D-Fender | unlord : And when you give up on that get a supported card :) |
| 23:38.44 | Qwell[] | [TK]D-Fender: one of these days...one of these nubs will teach us a lesson |
| 23:38.45 | [TK]D-Fender | </sarcasm> |
| 23:38.51 | unlord | ahh, so I should just spend the $20 and buy a real Digium one |
| 23:38.58 | [TK]D-Fender | Qwell : yeah.. they'll show US..... |
| 23:39.17 | [TK]D-Fender | unlord : If you jsut wannt putz around sure. |
| 23:39.26 | Qwell[] | "real Digium one"? |
| 23:40.08 | unlord | [TK]D-Fender: I've got home run catV all over my [very small] house |
| 23:40.08 | froguz | unlord, don't waste youre time, x100p clones, ambient chips and all that stuff are horrible working as FXO interfaces |
| 23:40.39 | unlord | all I'm trying to accomplish is integrating my home phones w/ VOIP so that when I recieve a call, my home phone rings |
| 23:40.58 | unlord | I'm not even sure how I'd place a call via the home phones |
| 23:40.59 | Qwell[] | unlord: get a Digium tdm400p, or a sipura spa-3000 |
| 23:41.23 | Qwell[] | of course, if your phones are analog, you'll need some fxs' too |
| 23:41.31 | Qwell[] | is that right? "fxs'"? |
| 23:41.36 | froguz | maybe just if you have a lot of time and just wanted to get exited of just hearing x-lite ringing from your cellphone |
| 23:41.46 | froguz | after that, go and buy a real one |
| 23:41.56 | [TK]D-Fender | unlord : Plug your whole home on an SPA-3000 and the line into it as well. End of story. |
| 23:42.09 | [TK]D-Fender | ok, well off to volleyball, bbiab |
| 23:42.48 | unlord | so the 4 ports are for the 4 lines I have? |
| 23:42.53 | unlord | in the catV? |
| 23:42.58 | Qwell[] | catV? |
| 23:43.11 | unlord | right, I have a star network of catV run throughout my house |
| 23:43.20 | unlord | it has 4 twisted pairs, which are lines 1,2,3,4 |
| 23:43.22 | Qwell[] | cat...5? |
| 23:43.27 | unlord | right |
| 23:43.37 | unlord | whatever you want to call it |
| 23:43.59 | Qwell[] | how many different extensions do you have? 4? |
| 23:44.05 | Qwell[] | How many actual lines from the telco? |
| 23:44.31 | unlord | I have only one actual line from the telco |
| 23:44.39 | unlord | but if I wanted to pay them more, they'd surely give me another |
| 23:45.09 | Qwell[] | well, you need one fxo per line from the telco, and one fxs per phone extension |
| 23:45.25 | Qwell[] | You *can* have multiple phones on a single fxs, but they won't have different extensions...they'll all ring at once |
| 23:45.38 | unlord | my basic thought was to take the demarcation in from the telco, and run it up to the PC where the VOIP hooks in (POTS) and then send it back down to the cable box where it goes out to all the analog lines |
| 23:46.00 | unlord | I don't really need more than one extension I don't think |
| 23:46.02 | Qwell[] | So, where does VoIP come into this? |
| 23:46.13 | unlord | I want to have incoming calls ring the home line |
| 23:46.13 | Qwell[] | sounds like 100% analog to me |
| 23:46.25 | unlord | yah, all my phones are analog |
| 23:46.40 | Qwell[] | so then you need either a tdm400p with an fxo and an fxs, or a sipura spa-3000 |
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| 23:47.22 | unlord | does the FXS let me 'dial' voip numbers and intercept it to call out over the internet |
| 23:47.41 | Qwell[] | an fxs lets you connect your phone to asterisk |
| 23:47.52 | unlord | I thought that was what the FXO did |
| 23:47.53 | Qwell[] | whatever you do with asterisk beyond that, is up to you |
| 23:48.02 | Qwell[] | no, an fxo lets you take a phone line, and connect it to asterisk |
| 23:48.06 | Qwell[] | ~fxofxs |
| 23:48.08 | jbot | i guess fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 23:48.17 | Qwell[] | neither of those have anything to do with voip |
| 23:48.49 | unlord | so the FXO is what comes in from the telco, and the FXS is the other port |
| 23:48.50 | marl | hi folks, can someone point me in the rite direction? im tryingf to setup * to dial a zap channel, and prompt for a # to accept the call, the docs i can find on the site indicate that i should use Dial(Zap4c/mobile_no) the 'c' being the switch to prompt for the '#' but its not working, can someoune point me to a working example, or tell me what ive doen wrong with this line? |
| 23:48.57 | unlord | like on a modem where it says PHONE and LINE |
| 23:49.01 | Qwell[] | no |
| 23:49.09 | Qwell[] | the "phone" port on a modem is passthrough |
| 23:49.19 | Qwell[] | it's literally hardwired to the "line" port |
| 23:49.24 | unlord | ahh |
| 23:49.33 | unlord | even if I get a V92 winmodem |
| 23:49.42 | Qwell[] | You could physically remove a modem from a PC, and the ports would still "work" |
| 23:49.53 | unlord | yah, true |
| 23:50.02 | Qwell[] | in fact, a friend of mine and I used a bunch of modems to string rj11 to his house from mine |
| 23:50.12 | Qwell[] | plastic baggies, buried underground, with modems in them... |
| 23:50.22 | unlord | wow, thats great |
| 23:50.27 | Qwell[] | yes, it was ghetto, yes, it worked |
| 23:50.38 | unlord | did you then use it as a null modem and play Doom? |
| 23:50.45 | Qwell[] | ... |
| 23:51.00 | unlord | hehe |
| 23:51.01 | unlord | n/m |
| 23:51.12 | Qwell[] | anyhow...bbl |
| 23:57.51 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 23:58.10 | unlord | hmm, too many choices |
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