| 00:07.43 | *** join/#asterisk Aut0Exec (~Joe@24.244.165.54) |
| 00:10.00 | Aut0Exec | voip provider list? |
| 00:10.03 | Aut0Exec | is there one here? |
| 00:10.10 | Aut0Exec | looking for a cheap prepaid one |
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| 00:17.50 | *** join/#asterisk Amazinzay (~Amazinzay@c-98-211-4-202.hsd1.tn.comcast.net) |
| 00:20.15 | Amazinzay | Hey everyone, I have been working on Asterisk 1.6 for a while and I am trying to set up 1.8 I've got everything installed and working, except when I do a pri show status I get "PRI span 1/0: Up, Active" Should I be concerned that it doesnt say "Provisioned, Up, Active"? |
| 00:20.34 | Aut0Exec | lol |
| 00:20.35 | Aut0Exec | wow |
| 00:20.38 | Aut0Exec | almost the same thing |
| 00:20.42 | Aut0Exec | just a lil different wording |
| 00:21.43 | Amazinzay | yeah, and it seems to be working right, except when I try to make a call via SIP->Asterisk->DAHDI I get a 503 error |
| 00:22.00 | Amazinzay | and when I do a pri debug it is giving me a hangup code 3 saying route not found |
| 00:22.00 | Aut0Exec | oh |
| 00:22.13 | Amazinzay | I wanted to make sure it wasn't because I don't have DAHDI configure correctly |
| 00:22.32 | Aut0Exec | see ya |
| 00:23.43 | Amazinzay | Anyone else know what might be the problem? |
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| 01:07.42 | ectospasm | Amazinzay: cause code 3 is a route problem? That soudns awfully like you're not formatting the DAHDI destination right |
| 01:08.14 | ectospasm | ...or, the telco can't route to that destination for whatever reason |
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| 02:35.30 | nightwalk | Is there NO way to get a tdm410 to NOT back-ring on other phones on the FXS line? |
| 02:35.43 | nightwalk | Ex: the phone rings in, someone answers the phone on a phone not connected to the asterisk system, so the tdm continues sending ring tones out on the fxs line (even though the line has already been 'answered') |
| 02:36.36 | WIMPy | You should be able to disable early media. |
| 02:37.04 | WIMPy | Bu tI doubt any of the cards are able to detect anything else on line. |
| 02:38.34 | nightwalk | I know very little about the POTS, but I don't understand why the fxs line even NEEDS to send a ring on the fxs line, honestly |
| 02:38.48 | tzanger | ? |
| 02:38.49 | WIMPy | It doesn't. |
| 02:38.53 | tzanger | how would you know when to pick up? |
| 02:39.13 | WIMPy | But POTS shouls really be seen as point-to-point. |
| 02:40.28 | nightwalk | tzanger: because the phone rings based on the signal from the pots |
| 02:40.33 | nightwalk | tzanger: as I said, the phone "gets picked up by a phone not connected to the asterisk system" |
| 02:40.36 | nightwalk | WIMPy: that seems more like an excuse for bad behavior from the card than anything :/ |
| 02:41.21 | WIMPy | POTS wasn;t designed for that. Live with it or go digital. |
| 02:41.32 | WIMPy | Or use a PBX. |
| 02:41.43 | nightwalk | But WHY does the fxs port on the card ring back? |
| 02:41.57 | nightwalk | I don't see any useful purpose to it |
| 02:42.11 | WIMPy | Because you have early media enabled? |
| 02:42.39 | nightwalk | Possibly. Not really sure what that is, but the google results seem to tie it to sip...? |
| 02:43.16 | WIMPy | It exists on SIP as well. |
| 02:43.30 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 02:43.41 | WIMPy | I have no idea on its use with analog. |
| 02:44.06 | tzanger | ? multiple phones on a line wasn't ever the intention of the POTS system? |
| 02:45.25 | WIMPy | Not without additional hardware. |
| 02:45.42 | nightwalk | If you REALLY want to get technical, that was EXACTLY the intention, being as back in the day several houses shared a single circuit |
| 02:46.18 | tzanger | I must be missing part of the conversation. when any phone on the line picks up the loop current rises and that signals the CO to stop ringing and connect the circuit |
| 02:46.33 | WIMPy | But they would have some sort of multiplexer between them. |
| 02:46.59 | tzanger | if asterisk is doing the ringing, it doesn't care who picks up. If asterisk is one of the phones and the other picks up, it should still do the right thing because the ringing voltage will go away |
| 02:47.10 | nightwalk | tzanger: the problem here is that the tdm410 sends a ring tone out to the already-connected phone that the tdm's fxs port is sharing with |
| 02:47.29 | nightwalk | IMO, it should NOT be doing that, as I can see no technical reason it SHOULD |
| 02:47.37 | tzanger | "ring tone" -- you're talking audio ringback? |
| 02:48.01 | tzanger | wait, who is doing the ringing? the PSTN or an FXO port on the 410? |
| 02:48.12 | nightwalk | No. It sends out a ring tone of its own as if it thinks the fxs line is still ringing (or something to that effect anyway) |
| 02:48.33 | nightwalk | Forget about the FXOs for a sec...they have nothing to do with this |
| 02:48.46 | WIMPy | understood it as ringback. |
| 02:49.17 | tzanger | ok, so you're saying that an FXS port on a TDM410 is sending audio when it is not doing any dialing or being instructed to pick up? It's merely another extension on a line? |
| 02:50.14 | tzanger | PSTN === FXS == TELEPHONE, PSTN is ringing the line, FXS is reporting ringing and telephoen is ringing... telephone picks up and the TDM410 FXS port is sending some kind of audio signal? |
| 02:50.28 | nightwalk | yes, basically |
| 02:50.33 | nightwalk | the tdm is supposed to be doing what a normal phone would be doing and nothing more |
| 02:50.36 | nightwalk | exactly |
| 02:51.11 | tzanger | nightwalk: that is WEIRD. |
| 02:51.17 | tzanger | kill asterisk (leave zaptel/dahdi running) -- does the same thing happen? |
| 02:51.28 | nightwalk | In fact, if it helps anything, you can think of asterisk in this situation as a basic sip->POTS converter |
| 02:51.37 | p3nguin | It's like you have Ringing() on the inbound extension or something. |
| 02:52.02 | tzanger | p3nguin: yes, which is why I'm asking if he can stop asterisk but leave everything else in place and repeat the experiment |
| 02:52.38 | p3nguin | The card isn't supposed to provide ringing indication to a phone line, is it? |
| 02:52.50 | p3nguin | I can't see why it would. |
| 02:53.11 | nightwalk | p3nguin: I asked about this before, and someone here told me it was a hardware feature and it wasn't possible to disable it. HOWEVER, after doing some reading, I'm now thinking they weren't understanding what I was asking for exactly |
| 02:53.14 | p3nguin | A phone certainly wouldn't, and I'd guess that a card with an FXO port would be made to emulate a phone. |
| 02:53.24 | tzanger | nightwalk: it is not a hardware feature |
| 02:53.41 | tzanger | I have personally designed equipment with the SLICs that the TDM400 uses. |
| 02:53.52 | p3nguin | It sounds like a setting in your dahdi config to enable ringing. |
| 02:54.06 | p3nguin | But I don't use analog, so I can't say with any certainty. |
| 02:55.28 | nightwalk | tzanger: I can't stop the asterisk server atm, but I'll check that as soon as the line is freed |
| 02:55.54 | tzanger | nightwalk: ok. if the problem goes away then you have a problem in the incoming context that your fxs line is set to use. |
| 02:56.11 | tzanger | nightwalk: if the problem does NOT go away, it sounds like you might have a funky port, although I need more information about the sound you're hearing |
| 02:56.43 | tzanger | just to make sure we're not runnign omething weird... you are using an FXO port with the PSTN line you're connected to, right? |
| 02:57.05 | nightwalk | nope...port 1 is an fxs, and that's the port I connected to the pots |
| 02:57.15 | tzanger | you do not connect FXS ports to POTS |
| 02:57.20 | p3nguin | Well that's a problem. |
| 02:57.20 | tzanger | you connect FXS ports to phones |
| 02:57.35 | p3nguin | FXO ports connect to the POTS line. |
| 02:57.37 | tzanger | FXS ports GENERATE ringing voltage. FXO ports ACCEPT ringing voltage |
| 02:57.41 | p3nguin | FXS ports connect to phones. |
| 02:58.10 | tzanger | I remember it this way: FXS = S = telephone Sets. FXO = O = central Offices |
| 02:58.52 | p3nguin | That's confusing. |
| 02:59.23 | tzanger | how do you remember it? |
| 03:00.00 | p3nguin | I don't use any mnemonic devices. I just remember them literally for what they do. |
| 03:00.15 | IsUp | nightwalk: do you have any other card on your pbx? |
| 03:00.27 | p3nguin | Phones have FXO ports. Phones must connect to FXS ports. |
| 03:00.32 | p3nguin | That's how I remember it. |
| 03:01.43 | IsUp | http://www.digium.com/images/graphics/pitch_catch.gif |
| 03:03.54 | nightwalk | Ok...the problem is that I'm an idiot. I just looked at my order and it was supposed to be 3x FXS, 1x FXO, but I got sent 3x FXO and 1x FXS. I'm guessing that got me thinking about it bass-ackward, and I didn't see a problem because it 'worked' :/ |
| 03:04.21 | p3nguin | Except that it doesn't work. |
| 03:05.28 | nightwalk | Which I, guess, leads me to another question: HOW can that work? FXOs don't generate voltage, so the phones should have been dead, should they not? |
| 03:05.34 | nightwalk | Also seems like FXS<->FXS should've burned out the port on the card |
| 03:05.56 | p3nguin | Are you plugging phones into FXO ports on the card? |
| 03:06.24 | IsUp | /etc/init.d/dahdi restart; dahdi_cfg -vvv; |
| 03:06.27 | nightwalk | p3nguin: yup. 3x FXO ports, each with a phone attached (basic, non-wireless phones) |
| 03:07.47 | nightwalk | http://pastebin.com/x78jYZaD <--- cat /proc/dahdi/1 |
| 03:08.56 | WIMPy | That's the signalling, which is the opposite of the port type. |
| 03:10.01 | IsUp | nightwalk: run dahdi_scan |
| 03:10.31 | nightwalk | Yep...sure enough, they're correct |
| 03:10.43 | nightwalk | Now that I think of it, I think that confused me when I was originally setting this up too :( |
| 03:10.57 | nightwalk | It's been a while though |
| 03:11.35 | nightwalk | So, for the record, I *do* have 1x FXO, and 3x FXS. The 3x FXS are connected to phones, and the 1x FXO is hooked to the pots line |
| 03:11.57 | p3nguin | I think I'd go make a change to the wires right now. |
| 03:14.57 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:17.24 | nightwalk | Ok...I just tried it again, and it isn't happening anymore. Think I'm going to have to go back through the commits and see what all I changed between then and now |
| 03:18.08 | nightwalk | Also need to do a bit more reading about the pots, evidently. Any ideas what I might have mangled (so I can narrow my search)? |
| 03:18.31 | ruied | How can I resolve this: B party calls to PBX; PBX transfers B to A; A doesn't answer and PBX push A call again. I've been thinking alot about this but do not know how to solve it .... |
| 03:19.05 | p3nguin | Look at your dial plan. |
| 03:20.31 | nightwalk | p3nguin: as I understand it, the dialplan shouldn't have come into play since the pbx wasn't the one who answered though...? |
| 03:20.43 | p3nguin | I was responding to ruied. |
| 03:21.16 | nightwalk | p3nguin: ah...sorry. I was still trying to make sense of ruied's message when I replied :) |
| 03:24.06 | nightwalk | p3nguin, tzanger, WIMPy, IsUp: thanks for your help, and sorry my confusion ended up making you waste a bit of your time |
| 03:24.44 | p3nguin | It was an honest mistake, no worries. |
| 03:25.28 | nightwalk | Will dig into this a little more tomorrow when I (hopefully) have a clearer head |
| 03:26.23 | ruied | sorry my english... let me try again, someone from the a mobile phone dial to PBX, PBX person answers and than transfer the call to phone 'A'. Since phone 'A' didn't answear, how PBX can have the call aggain with the mobile phone's person ? |
| 03:27.30 | ruied | don't know how to recover a transferred call... |
| 03:27.53 | p3nguin | Blind transfers cannot be recovered. Attended transfers are not lost. |
| 03:28.45 | p3nguin | In the case of a blind transfer, I would expect voicemail to pick up when phone-A didn't answer. |
| 03:29.07 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 03:29.54 | ruied | p3nguin, attended transfer has anything to do with call park? |
| 03:30.47 | ruied | if I pick up the call ant than transfer it with the phone's transfer button, is this a 'blind transfer' ?? |
| 03:34.11 | tzanger | depends |
| 03:34.24 | tzanger | blind transfer means the call is sent over to the other phone without you talking to the person you're transferring to |
| 03:34.37 | tzanger | so if you hit transfer, enter the destination extension and hang up, that's a blind transfer |
| 03:34.55 | tzanger | if you hit transfer, enter the destination, wait for them to pick up and say "you got a call" then hangup, that is an attended transfer |
| 03:36.10 | p3nguin | You could park the call, then call over to phone-A and say, "You have a call parked on 701," and then when they say okay, you hang up. If they don't call 701 to take the call, it will return to you after the timeout period. |
| 03:37.53 | ruied | tzanger, thanks for the explanation |
| 03:38.06 | p3nguin | Another choice, which may or may not be suited to your application, is using call pickup. If you blind transfer to phone-A and you hear it ringing and ringing, you could then "steal" that call by dialing *8 from your phone (if that's the code you use for pickup). |
| 03:39.21 | ruied | p3nguin, thanks, I think that's what I was looking for ;) |
| 03:39.37 | ruied | the first solution, the second one is ok too :) |
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| 03:46.11 | p3nguin | Call parking is great for several reasons. |
| 03:46.48 | p3nguin | One, you can park a call and go run around the office looking for someone to tell him he has a call, and he can take that call from any phone in the building instead of just his own. |
| 03:46.58 | WIMPy | It reminds you of the days, before PBXs? |
| 03:47.15 | p3nguin | Pretty much. It's useful for simulating a key system. |
| 03:51.10 | ruied | I'll try it |
| 03:53.49 | pabelanger | we |
| 03:54.04 | pabelanger | wrong window |
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| 04:11.29 | jamko | Hello all. I am trying to use a host name rather than IP address for one of my origination providers, but asterisk fails to authenticate on inbound calls when I set the peer's host parameter to a host name. Does Asterisk have a setting to do a DNS lookup on peers? It can obviously resolve the hostname to IP address |
| 04:11.34 | jamko | based on sip show peers. |
| 04:12.05 | p3nguin | What host name are you using? |
| 04:13.02 | jamko | the host name of the origination provider's switch. tampa.voip.ms |
| 04:13.27 | p3nguin | If you use the IP address, does it work? |
| 04:13.32 | jamko | yes. |
| 04:13.54 | p3nguin | Which IP address are you using that is working? |
| 04:14.14 | jamko | The same one that asterisk resolves to when I set it to the host name. |
| 04:14.41 | p3nguin | So it is resolving, then. |
| 04:15.17 | jamko | yes in sip show peers, it shows the correct ip, when I set the host parameter to tampa.voip.ms |
| 04:15.41 | jamko | And it shows that IP in the failed to authenticate message as well. |
| 04:16.14 | jamko | Obviously I can just use the ip address, but just curious why it would cause a problem, if DNS is apparently working properly |
| 04:16.42 | p3nguin | Maybe it's a bad configuration. Here's mine, but using chicago rather than tampa: http://pastebin.com/rC6eGsuH |
| 04:17.12 | *** join/#asterisk greenwolf (greenwolf@cpe-74-77-221-5.buffalo.res.rr.com) |
| 04:17.56 | jamko | Ok so you have the IP address in the permit parameter though. |
| 04:18.05 | jamko | I put the host name in the permit parameter. |
| 04:18.14 | jamko | Kind of defeats the purpose of using a hostname, no? |
| 04:18.31 | p3nguin | The permit line creates an ACL. It's not required. |
| 04:19.09 | p3nguin | I don't think you can use a hostname in the ACL entries, but I could be mistaken about that. |
| 04:19.56 | jamko | Hmm.. Does voip.ms send out notification e-mails when their IP address is changing for a POP? |
| 04:20.17 | p3nguin | They have in the past. |
| 04:20.36 | p3nguin | They don't change often, though. |
| 04:21.02 | jamko | Well I'll roll the dice then, and wrestle for automated perfection another day. |
| 04:21.06 | jamko | Thanks p3nguin |
| 04:22.42 | p3nguin | If you don't need anonymous SIP calling, you could create your ACL at the firewall and forget about the ACL in asterisk. |
| 04:26.16 | jamko | I think it might have something to do with an incorrect global setting, because I always get this message whenever I reload sip when I have a host name rather than IP address, IE- flowroute. acl.c:3 |
| 04:26.20 | jamko | woops |
| 04:27.03 | jamko | acl.c:328 ast_append_ha: Invald ip address in sip.flowroute.com .. Bad ACL entry in configuration line... doing dnsmgr_lookup |
| 04:27.29 | jamko | so maybe remove the acl like you were saying, but I use IP address authentication, not registration. |
| 04:27.35 | p3nguin | So in its own way, it is telling you that it does not accept host names in the ACL entries. |
| 04:28.18 | p3nguin | If it were me, I'd simply use the IP address in the permit line and go on with my day. |
| 04:28.55 | jamko | yea that makes sense. |
| 04:30.39 | jamko | Is it a security risk to remove the permit/deny entry for origination providers, if I am doing IP auth? It works with the host name if I remove the permit/deny. |
| 04:31.36 | p3nguin | I guess it could be, technically. |
| 04:32.33 | jamko | I am not sure in what way one would exploit it though. |
| 04:32.36 | p3nguin | If someone knows your IP address and knows how to appear as tampa.voip.ms, and knows what extensions you accept calls on and can figure out how to do bad things with those extensions, it could be a problem. |
| 04:33.12 | p3nguin | You shouldn't be allowing inbound calls to go back out, so that's out of the picture. |
| 04:33.46 | p3nguin | The most they could really do, in my opinion, is create a DoS, which isn't exactly a security problem in the same sense that most people think of security problems. |
| 04:35.17 | jamko | So they would pretend to be my provider and send me a shit load of fake calls I suppose. From my experience that is not the purpose of sip attacks. Either way, I will just use the ip address to be safe. Don't need anything weird. |
| 04:36.27 | p3nguin | Most of the SIP attacks I have had against me do not try to appear as my provider, probably because they'd have to be guessing which provider my host is expecting. |
| 04:36.44 | p3nguin | The attacks most often are trying to find a route back out of the system to make free calls. |
| 04:37.17 | p3nguin | I got to the point where I accept all those calls and send them to ludacris extensions. |
| 04:38.05 | tzanger | p3nguin: I have the same thing |
| 04:38.08 | p3nguin | Err, uh, ludicrous. |
| 04:38.10 | tzanger | exten => _.,1,Hangup |
| 04:38.13 | p3nguin | Damn rappers. |
| 04:38.47 | tzanger | actually I think the first step is a reporting NoOp() but you get the idea |
| 04:39.10 | p3nguin | I have a Verbose() so I know it is going on. |
| 04:41.21 | jamko | interesting. I see mainly brute force on my simplest extensions, but maybe I am not seeing the whole picture. What is an example of how they would get "back out" of the system? |
| 04:42.22 | tzanger | jamko: if you allow that peer to get "out" (dial out of your PBX) then you have a potential problem |
| 04:42.36 | p3nguin | People don't have any understanding of what contexts are used for, so they often incorporate inbound extensions and outbound extensions in a single context, or they'll do silly things with includes, allowing someone to reach an extension on incoming and dial back out of another extension. |
| 04:43.16 | tzanger | the only way out of my system from an inbound connection is through DISA, and that is only available on one DID. it's about as safe as it gets without locking it down to specific caller ids (whcih can be faked anyway) |
| 04:44.49 | p3nguin | That's another way: people don't know when to use DISA and when not to use it. |
| 04:45.14 | p3nguin | DISA isn't a shortcut, but is often used as one. |
| 04:46.22 | tzanger | I think the single biggest problem I have with my pbx is that I keep forgetting how I set up the parking lot since I (try to) use it maybe 3 times a year |
| 04:46.32 | jamko | The hosts in question are providers, so they don't really dial out of the system per se. All of the peers capable of dialing out, are using registration, not static ip authentication. |
| 04:47.13 | jamko | lol tzanger ... I feel you on that. |
| 04:48.01 | p3nguin | But if I call your phone number via that provider, I reach some extension on your box, which runs BackGround() and/or WaitExten(), and I enter in a full phone number that matches an outbound dialing rule in the same or in an including context, now I'm making calls on your dime. |
| 04:50.18 | jamko | interesting.. But if I have strictly an incoming_calls context, which does not have any includes, and just routes to another context with the S variable, is that still a risk? |
| 04:50.19 | p3nguin | Maybe I craft a dial string to exploit one of your bad macros or something. Let's say I'm being given time to enter in an extension, so I somehow manage to send "1234&SIP/voipms/13145551212" in. |
| 04:50.55 | p3nguin | If your dialplan is built well, the most they could do is make calls into your system and call people's phones internally. |
| 04:51.32 | tzanger | p3nguin: that's why I strongly advocate many small contexts and gradually build up more complex dialplans |
| 04:51.47 | tzanger | the modules are all easy to parse and debug by mere humans |
| 04:52.14 | p3nguin | Designing good contextual hierarchy is near critical. |
| 04:53.10 | p3nguin | inbound can call extens in internal, but internal cannot dial outbound. phones are assigned to a context called 'phones' which includes outbound and internal contexts. |
| 04:53.15 | p3nguin | and so on. |
| 04:54.00 | p3nguin | Never allow an inbound call into a context where outbound calling is possible. That's asking to either run up your bill or run out your prepaid balance. |
| 04:55.01 | jamko | wow thanks for the info. I'm going to try exploiting that on my box. very interesting. |
| 04:55.10 | p3nguin | Luckily, most ITSPs in the residential market are prepaid, and most people who prepay don't keep extremely high balances. |
| 04:56.18 | p3nguin | I think if you could exploit my outbound extensions right now, you could steal about $18 from me. Whoop-de-doo. |
| 05:02.14 | jamko | So for that exploit to work though, my incoming calls context must include a context that allows outgoing calls. But if my incoming calls context is included in a context that includes both incoming and outgoing calls, but incoming calls does not exclusively include outgoing, I should be okay, right? |
| 05:02.39 | jamko | Yea $25 is the highest I go. |
| 05:04.28 | p3nguin | Your incoming calls context cannot include itself, and there's no reason it should ever include a context that allows outgoing calls. |
| 05:05.27 | p3nguin | I don't see any reason for an outgoing context to include incoming, in most cases, either. |
| 05:05.39 | p3nguin | In rare cases, I know it happens. |
| 05:06.26 | jamko | Yea i think im good but ill try to exploit it anyway. |
| 05:06.58 | jamko | Meanwhile I'll put the ip address back in for the host. #wayofftracknow |
| 05:07.02 | p3nguin | Also look for the ability to craft an extension like I mentioned earlier. |
| 05:07.35 | jamko | ok |
| 05:08.01 | p3nguin | It is recommended to filter for only numbers to prevent such things, but most people don't filter. |
| 05:10.02 | jamko | I don't think there is anyway into my outgoing context unless you are registered. Incoming rings phones and stops. No automated attendants at the moment or wait extens. |
| 05:11.11 | p3nguin | I don't have to be registered to you to be able to make calls on your system. I just have to know one of your peer names and the password for it. |
| 05:12.50 | p3nguin | I'd have to hope you didn't create an ACL for said peer, though. |
| 05:13.03 | tzanger | I actually allow anonymous calls to dial my business line... but that's all they can do |
| 05:13.43 | p3nguin | I allow all sorts of anonymous calls to come in. I'm not scared, and it gives me something to do at night, as well. |
| 05:14.52 | p3nguin | It also helps that my IP address for asterisk it not publicly advertised. |
| 05:19.23 | p3nguin | I do find it strange that I get so many forms of attacks on an unadvertised host. I guess people scan netblocks all day and night looking for UDP 5060 to respond with something other than closed or filtered. |
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| 05:58.56 | tiuman | http://pastebin.ru/320382 what not call? |
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| 06:16.37 | dschuett | has anyone figured out a good way to bypass the annoying call screening issue? - even if this option is turned off it still asks you to press 1 to accept or 2 to send to voicemail. I have tried the answer, wait, send dtmf ..... all with no luck |
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| 07:18.24 | ChannelZ | What, from Google Voice? |
| 07:27.57 | p3nguin | If yes... the answer and senddtmf method works just fine for me, but other people send the digit from the dial() command with success. |
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| 08:13.17 | Arch1mede | anyone know where i need to go to resolve wrong password, getting chan_sip.c24884 handle_request_register: Registration from (display name) <sip:1101@(asterisk server 192.168.1.4) failed for 192.168.1.6 (SPA):5060 - Wrong Password |
| 08:34.58 | *** join/#asterisk AlecTaylor (~AlecTaylo@unaffiliated/alectaylor) |
| 08:35.00 | AlecTaylor | hi |
| 08:35.50 | AlecTaylor | I'm looking for a Flash or Java project featuring "Click to call" functionality from the browser. Do you know of such a project? |
| 08:44.09 | *** join/#asterisk _trine (~trine@trine1-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 08:44.56 | _trine | is there an asterisk command from cli which will show me the asterisk passwd |
| 08:45.29 | _trine | my SPA will not register and I wanted to check if asterisk has the same passwd as my SPA |
| 08:51.47 | tzafrir | _trine, SIP user's password? |
| 08:53.49 | _trine | tzafrir, well in sip.conf I have [1101](sets) |
| 08:53.49 | _trine | secret=xxxx |
| 08:54.17 | tzafrir | sip show user 1101 ? |
| 08:54.25 | _trine | I think the passwd and ID I have set in my SPA 3102 should allow a registration |
| 08:54.34 | _trine | if it matches |
| 08:55.54 | _trine | when the SPA 3102 tries to register with asterisk, asterisk is saying 'wrong password' |
| 09:00.36 | tzafrir | What error do you see in the asterisk CLI when it tries to register? |
| 09:06.46 | _trine | chan_sip.c:24884 handle_request_register: Registration from 'trine <sip:1101@192.168.1.4>' failed for '192.168.1.6:5060' - Wrong password |
| 09:07.35 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:10.10 | _trine | .4 is the dockstar running asterisk and .6 is the SPA |
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| 09:10.46 | _trine | a dockstar is a small router type device which runs debian |
| 09:11.04 | _trine | and I have asterisk running inside it |
| 09:13.35 | tzafrir | do you have the type for [1101] (or for [sets]) set to 'friend'? |
| 09:13.36 | _trine | am I right in thinking the ONLY place the passwd needs to be is in asterisk '[1101](sets) secret=xxxx' in sip.conf and in the SPA 3102 |
| 09:13.58 | tzafrir | Also: host=dynamic ? |
| 09:14.11 | _trine | as far as I understand it this should allow registration? |
| 09:14.23 | _trine | let me check |
| 09:17.00 | _trine | where would 'host=dynamic' be ? |
| 09:17.26 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:24.21 | tzafrir | in that section |
| 09:24.28 | _trine | yes it is set to host=dynamic in sip.conf for internal sets |
| 09:24.42 | _trine | and sets |
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| 09:30.38 | iscario | hi! i set up an asterisk server for my personnal voip use and it works as expected. I would like now to 'upgrade' my services and connect it to the telephony network. So, i would like to know : which hardware devices should i bought to -1- allow my softphone to call a hardphone and -2- allow a hardphone to call my softphone. my sofphones are using SIP and IAX protocol/ Thank you! |
| 09:31.29 | _trine | iscario, a cheap way would be to get either an SPA3000 or an SPA3102 |
| 09:32.04 | iscario | _trine, i'll google this ; ) |
| 09:32.28 | _trine | they are on sale on Ebay |
| 09:33.55 | AlecTaylor | yay |
| 09:35.38 | iscario | _trine, so in fact i would have no changes to do on my server, it is a kind of external router , right ? |
| 09:35.59 | _trine | kind of |
| 09:36.07 | _trine | yes |
| 09:36.31 | _trine | it allows you to connect a telephone |
| 09:36.47 | _trine | and it talks with asterisk on your computer |
| 09:37.50 | _trine | then you need to have a subscription with a voip provider such as voipstunt |
| 09:38.05 | _trine | or you can just use sip direct or IAX |
| 09:38.38 | iscario | yes i don't want to rely on an external voip provider |
| 09:38.52 | iscario | interesting! |
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| 09:44.02 | iscario | another question then, because at a time, i thought to buy a gsm modem to be directly connected to gsm network... What should i configure in asterisk to be able to redirect voip calls to gsm ? (i would need to configure chan_mobile and dahdi ?) |
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| 12:33.08 | AlecTaylor | I'm looking for a FOSS Flash or Java project featuring "Click to call" functionality from the browser. Do you know of such a project? |
| 12:34.22 | tzafrir | AlecTaylor, search the archives of the asterisk-users mailing list or whatever |
| 12:34.31 | tzafrir | I recall this question asked there |
| 12:34.45 | AlecTaylor | asked it |
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| 13:04.17 | Dovid | when doing set(codec=foo) and then I do a dial it still sends 2 codecs. any way to force the outgoing invite to only have one? |
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| 13:14.54 | spnl_ | Hi |
| 13:15.55 | spnl_ | i was wondering if i could get some help with getting a ISDN 1 port card to work..? |
| 13:16.14 | spnl_ | "Network controller: Device 1ae7:0520 (rev 02)" |
| 13:17.31 | spnl_ | tried asterisk 1.4 and 1,6 various versions of mISDN last one was mISDN-1_1_9.1 which i patched for my card |
| 13:17.54 | spnl_ | running debian squeeze, and at the moment lenny with kernel 2.6.26 |
| 13:23.37 | spnl_ | or any one could advice a OS/kernel anything to get my ISDN card working with asterisk? |
| 13:28.19 | WIMPy | If you have a version for that card, waht's the issue? |
| 13:29.28 | spnl_ | that i cant get it to work... |
| 13:29.59 | spnl_ | and if im lucky to find a patch on the net, it doesnt seem to match up wih the version(s) i can find to patch |
| 13:31.04 | spnl_ | for example "for line in $(${LSPCI} -n | ${GREP} "1397....." |
| 13:31.34 | spnl_ | in 1.9 isnt that supposed to just be grep and not ${GREP} ? |
| 13:32.05 | spnl_ | then is also no done added to the end of "for line in" |
| 13:32.06 | WIMPy | You said you that 1_1_9.1 included support for the card. So does it or doesn't it? |
| 13:32.24 | spnl_ | which leads me to believe i could be climbing the wrong tree |
| 13:32.42 | WIMPy | That grep thing doesn;t make any sense to me. What's that about? |
| 13:32.52 | spnl_ | heres the full line |
| 13:33.10 | spnl_ | for line in $(${LSPCI} -n | ${GREP} "1397:\(2bd\(0\|6\|7\|8\|9\|a\|b\|c > \)\|b100\)\|1ae7:0520\|1043:0675\|0871:ffa\(1\|2\)\|1051:0100\| 15b0:2bd0 > \|114f:007\(0\|1\|2\|3\)\|13d1:2bd1\|182d:3069"); do > addcard "hfcpci" "" 1 'mode="te" link="ptmp"' |
| 13:33.38 | spnl_ | of course the ">" are from new lines and the spaces must also be removed |
| 13:33.55 | WIMPy | What (or from what) is that? |
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| 13:34.17 | spnl_ | from the instructions that are mean to patch mISDN for my card |
| 13:34.21 | WIMPy | Do you have a module for that card or not? |
| 13:35.03 | spnl_ | that should be hfcpci? |
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| 13:35.15 | spnl_ | or mISDN |
| 13:35.30 | WIMPy | Probably. |
| 13:36.36 | spnl_ | im just not really sure how to get the card installed |
| 13:37.04 | spnl_ | and after it is installed should it still be ""Network controller: Device 1ae7:0520 (rev 02)"" or detected as something else? |
| 13:37.16 | spnl_ | how am i meant to know if its correctly working |
| 13:37.39 | spnl_ | asterisk:~# mISDN scan 0 mISDN compatible device(s) found: |
| 13:37.54 | spnl_ | asterisk:~# misdn-init scan [OK] found the following devices: card=1 |
| 13:38.07 | spnl_ | which confuses me..... |
| 13:38.25 | spnl_ | and it also complains about it can not find module 1 on start |
| 13:38.54 | spnl_ | <PROTECTED> |
| 13:39.24 | spnl_ | though is lsmod mISDN_dsp seems to loaded fine |
| 13:40.34 | spnl_ | the card is a "x-tensions xc-520" a copy of the B100P |
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| 13:45.52 | spnl | this is the only info i can find --- http://forum.ipfire.org/index.php?topic=3311.0 |
| 13:48.46 | spnl | so no-one any ideas? IRC was kind of my last hope... |
| 13:50.52 | WIMPy | Do yu have hfcpci loaded? |
| 13:51.18 | WIMPy | That is referring to LCR, which uses misdn2. You seem to be on to misdn1. |
| 13:51.22 | WIMPy | Which one do you intend to use?> |
| 13:52.00 | spnl | i was under the impression i must use mISDN v1.1.9 |
| 13:52.22 | spnl | though i'd use any OS/version/ of anything as long as it works |
| 13:52.25 | WIMPy | According to that forum thread it seem to run with misdn2. |
| 13:52.38 | spnl | though id be more comfortable running debian |
| 13:52.48 | spnl | correct |
| 13:52.49 | WIMPy | That means don;t install any drivers, just use a current kernel. |
| 13:53.05 | spnl | though when downloading the git, the syntax seems to be different than it is in the patch |
| 13:54.00 | WIMPy | If I understand that right, you don;t need any patch. It should work out of the box. |
| 13:54.42 | spnl | so this card that is called the "noname" should be running with the current git version of mISDN? |
| 13:55.04 | WIMPy | Hmm. later ithey say it doesn't. |
| 13:55.36 | WIMPy | So I'm not sure you need a patch. |
| 13:55.45 | WIMPy | Wat kernel are you using? |
| 13:55.56 | spnl | ive been at it for 4 days with just as many hours sleep :S |
| 13:56.01 | spnl | at the moment 2.6.26 |
| 13:56.13 | spnl | asterisk:/usr/src/asterisk# uname -r 2.6.26-2-amd64 |
| 13:56.35 | WIMPy | IIRC that was the very first version. |
| 13:56.52 | WIMPy | Try a current one and see if it just works. |
| 13:57.03 | spnl | i was running 2.6.30 |
| 13:57.15 | spnl | latest version in debian netboot install |
| 13:57.21 | WIMPy | That's not really current. |
| 13:57.46 | spnl | which version do you advise? |
| 13:58.15 | WIMPy | I'm currently on 3.1-rc1 |
| 13:58.58 | spnl | and will going over to CentOS have any benifit? |
| 13:59.37 | WIMPy | Ok, looks like that patch wasn't included. So it doesn't matter much. |
| 13:59.58 | WIMPy | "If the Distro makes a difference, it's no good." |
| 14:00.19 | spnl | or trying bristuff? which i believe is outdated right? |
| 14:00.22 | WIMPy | If 2.6.30 is the latest, debina offers, go with that. |
| 14:00.24 | spnl | ok good then im going to stick with debian |
| 14:00.38 | WIMPy | Yes, that's become jnet-dahdi. |
| 14:00.52 | ruied | for serves and asterisk I've been using debian and I like it alot... |
| 14:01.19 | spnl | i also had a go with dahdi, when i was running the newer kernel |
| 14:01.22 | WIMPy | I'd go with either misdn2 or one of the zaphfc tings. Taht depends on what features are imoprtant to you. |
| 14:01.51 | spnl | at the moment id be happy that it worked |
| 14:02.18 | WIMPy | Use the .30 kernel and apply the patch from that forum thread. |
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| 14:04.46 | spnl_ | so, getting most recent kernel version and having a go with mISDN2 seems the way this is going to work right? |
| 14:05.11 | spnl_ | though it will also be needed to be patched right? :S |
| 14:05.27 | WIMPy | Yes. |
| 14:05.49 | spnl_ | though what would the correct syntax be then? |
| 14:06.12 | WIMPy | That seems to fit my kernel. |
| 14:07.03 | WIMPy | Yes, it applies to 3.1. |
| 14:07.44 | spnl_ | how do you mean it applies to 3.1? the patch |
| 14:07.44 | WIMPy | And do not install any additional mISDN stuff. Only mISDNuser. |
| 14:08.17 | WIMPy | II tried to apply that patch to my kernel source and that worked. |
| 14:08.52 | spnl_ | this http://git.ipfire.org/?p=ipfire-2.x.git;a=blob;f=src/patches/mISDN_hfc-s_add_id.patch;h=d544b1fcf7014a10453e9f93fac30085ec50474a;hb=e7d2f07094a3cd74293a10123184489b1c916c57 |
| 14:09.06 | spnl_ | is supposed to applied to the kernel source? and not the mISDN source?! |
| 14:09.15 | WIMPy | Oops. |
| 14:09.20 | WIMPy | wait |
| 14:11.45 | spnl_ | this is from hfcpci.c from the git mISDN |
| 14:11.45 | spnl_ | http://git.ipfire.org/?p=ipfire-2.x.git;a=blob;f=src/patches/mISDN_hfc-s_add_id.patch;h=d544b1fcf7014a10453e9f93fac30085ec50474a;hb=e7d2f07094a3cd74293a10123184489b1c916c57 |
| 14:11.48 | spnl_ | whoops |
| 14:11.57 | spnl_ | { PCI_VDEVICE(SITECOM, PCI_DEVICE_ID_SITECOM_DC105V2), (unsigned long) &hfc_map[22] }, |
| 14:12.24 | spnl_ | where this should be added 24 + {0x1ae7, 0x0520, 25 + PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long) &hfc_map[23]}, |
| 14:12.52 | spnl_ | in the patch there is no reference to PCI_VDEVICE |
| 14:13.12 | WIMPy | http://git.ipfire.org/?p=ipfire-2.x.git;a=blob_plain;f=src/patches/mISDN_hfc-s_add_id.patch;hb=e7d2f07094a3cd74293a10123184489b1c916c57 |
| 14:13.30 | WIMPy | This one looks good, but does not apply out of the box for me. |
| 14:13.55 | spnl_ | how should this patch be applied? |
| 14:14.09 | WIMPy | with patch :-) |
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| 14:14.40 | spnl_ | so $ patch thepatch-from-link.patch |
| 14:15.13 | spnl_ | before make && make install, or after? |
| 14:15.34 | WIMPy | you go to /usr/src/linux and do patch -p1 < the-saved-file.patch |
| 14:16.15 | spnl_ | ok awesome |
| 14:16.41 | spnl_ | after make install? |
| 14:16.49 | WIMPy | before make |
| 14:16.59 | spnl_ | ok :) |
| 14:17.06 | spnl_ | thanks |
| 14:17.18 | spnl_ | sure i was starting to lose my hair P |
| 14:17.34 | spnl_ | so first upgrade kernel to >= 3.1 |
| 14:18.03 | WIMPy | Just use to most recent, debian has to offer. |
| 14:18.39 | WIMPy | will try to make it work for 3.1 |
| 14:19.17 | spnl_ | :) and just a simple kernel upgrade will do the trick? |
| 14:21.11 | WIMPy | First get the most current kernel source package. then apply the patch, make, make install, reboot and hopefully it will work then. |
| 14:21.37 | WIMPy | Ten you can install lcr and asterisk, if you haven't already. |
| 14:23.25 | spnl_ | where can i find the most recent version? |
| 14:23.59 | WIMPy | In your package manager. |
| 14:24.20 | spnl_ | oh not just wget it ? |
| 14:24.23 | spnl_ | http://www.kernel.org/pub/linux/kernel/v3.0/testing/linux-3.1-rc3.tar.bz2 |
| 14:25.23 | WIMPy | Sur, you can do that. I just thought you'd like to stick to the debian packages. |
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| 14:27.40 | spnl_ | i cant seem to find it through apt-cache search |
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| 14:40.58 | spnl_ | im going for a clean netboot install of debian 6.0 squeeze to get rid of any mess i may have made... |
| 14:41.59 | WIMPy | Can't hurt, except on the clock. |
| 14:42.13 | spnl_ | yeh.... |
| 14:42.28 | spnl_ | cant really figure out how to just update the kernel though :( |
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| 14:46.57 | ruied | where can I look at this error message? : [call=5 ast=lcr/5] Message 15 from LCR unhandled. |
| 14:47.42 | ruied | Message 15 (chan_lcr) |
| 14:47.45 | WIMPy | ruied: Where/when do you see that? |
| 14:48.18 | ruied | in asterisk prompt when a place an outgoing call to LCR |
| 14:48.38 | ruied | it makes the call, but reports that error... |
| 14:49.39 | WIMPy | Don;t know what 15 is. Maybe an AOC message? |
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| 14:53.07 | ruied | maybe... |
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| 15:01.08 | spnl_ | WIMPy: "And do not install any additional mISDN stuff. Only mISDNuser." mISDN of course aswell right? |
| 15:01.21 | spnl_ | and just the versions of the git? |
| 15:01.38 | WIMPy | No, no mISDN. |
| 15:01.47 | WIMPy | Only mISDNuser. |
| 15:02.04 | WIMPy | Well, you can, but it's already in the kernel. |
| 15:02.09 | WIMPy | [12:38] Gokee2 has joined #asterisk (~gokee2@204-195-14-78.wavecable.com) |
| 15:02.14 | WIMPy | [12:38] echinos has joined #asterisk (~echinos@67.196.136.211) |
| 15:02.21 | WIMPy | Ooops |
| 15:04.01 | dschuett | has anyone figured out a good way to bypass the annoying call screening issue? - even if this option is turned off it still asks you to press 1 to accept or 2 to send to voicemail. I have tried the answer, wait, send dtmf ..... all with no luck |
| 15:04.51 | dschuett | with gtalk ^^ |
| 15:05.20 | spnl_ | thanks WIMPy thats probably where i was going wrong with the new kernel |
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| 15:11.52 | WIMPy | Ah, it worked. |
| 15:11.53 | [sr] | anmyone from NY? |
| 15:12.48 | spnl_ | what did? |
| 15:13.03 | WIMPy | I did a new patch. |
| 15:13.26 | spnl_ | nice, what is it patching? |
| 15:14.49 | WIMPy | Your issue. |
| 15:15.30 | spnl_ | WOW, super! |
| 15:15.42 | WIMPy | http://voice.yeti.dk/patches/ |
| 15:15.48 | spnl_ | you are a superstar :P |
| 15:16.24 | WIMPy | It compiled with 3.1.0-rc3. Now it just has to work as well :-) |
| 15:17.26 | WIMPy | As we found out that the LCR git seems a little messed up, you might have to take chan_lcr from there, as well. |
| 15:17.31 | spnl_ | im looking forward to trying it |
| 15:17.49 | spnl_ | "X-Tensions XC-525 USB ISDN" is not my issue right? |
| 15:17.51 | WIMPy | +yesterday |
| 15:18.07 | WIMPy | No, the X-Tensions PCI. |
| 15:18.18 | WIMPy | You see the PCI ID. |
| 15:18.42 | spnl_ | just checking we didnt get mixed up :) |
| 15:19.02 | WIMPy | ooops. I think I need to check permissions. |
| 15:20.00 | WIMPy | That's better. |
| 15:20.35 | spnl_ | lol yes |
| 15:21.14 | spnl_ | "hfcpci-1AE7:0520.patch" :D :D |
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| 15:29.46 | spnl_ | ok debian 6.0 is installed though kernel = 2.6.32 :S |
| 15:31.45 | [sr] | is using 3.1.0-rc3 |
| 15:31.45 | WIMPy | Hmm. Are colons evil? |
| 15:31.57 | spnl_ | WIMPy: do i just download the link to the kernel above and extract to /usr/src/linux && run patch && make && make install mISDNuser |
| 15:32.04 | spnl_ | my colons pretty evil :P |
| 15:32.38 | spnl_ | 2.6.32-5-686 |
| 15:32.46 | WIMPy | There's nothing to extract. Just do the patch as decribed earlier. |
| 15:33.02 | WIMPy | I think I better rename that stuff. |
| 15:33.19 | spnl_ | sorry.... |
| 15:34.16 | WIMPy | html is evil. |
| 15:34.22 | spnl_ | "Unfortunately connecting Asterisk to the ISDN can be rather confusing for beginners as there are several different routes available to do so. Be prepared for some mess. " lol i have mess |
| 15:34.52 | spnl_ | BTW your link for the 520 patch is not correct |
| 15:35.10 | spnl_ | well working correctly |
| 15:35.18 | WIMPy | No it just wasn't html compatible. Try to reload. |
| 15:35.24 | spnl_ | think your missing a slash |
| 15:35.26 | spnl_ | ooo ok |
| 15:35.48 | spnl_ | yeh its good now |
| 15:35.58 | spnl_ | i just copied and pasted it above |
| 15:36.04 | spnl_ | (before) |
| 15:36.06 | WIMPy | The colon was bad. |
| 15:36.36 | spnl_ | so i dont need to download the 3.1 , just begin directly with patching |
| 15:37.23 | WIMPy | Not sure if there's a relevant difference between 2.6.32 and 3.1.0. Just try it. |
| 15:38.03 | spnl_ | ok, just downloading build-essential |
| 15:42.24 | spnl_ | should i have a /usr/src/linux folder? |
| 15:42.32 | spnl_ | or must this be a symlink |
| 15:45.44 | WIMPy | link |
| 15:46.15 | spnl_ | to which linux-headers-2.6.32-5-686 linux-headers-2.6.32-5-common linux-kbuild-2.6.32 |
| 15:48.17 | WIMPy | Usually it's just linux-version. |
| 15:49.59 | spnl_ | did a bit of googling, linux-headers-xxxx-(guessing 686 as that is my arch) |
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| 15:50.30 | spnl_ | but when running patch -p1 ../hfcpci-1AE7-0520.patch its just hanging |
| 15:50.56 | WIMPy | you're missing a < or a -i. |
| 15:51.10 | spnl_ | lol |
| 15:51.12 | spnl_ | i am |
| 15:51.23 | WIMPy | But headers doesn't sound like a kernel source. |
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| 15:53.30 | spnl_ | hmmm |
| 15:55.22 | spnl_ | some googling leeds me to believe most debian users are point /usr/src/linux to the headers |
| 15:58.01 | [sr] | spnl_: apt-get install linux-source-2.6 |
| 15:58.27 | spnl_ | thanks :) |
| 16:00.22 | spnl_ | then extract and symlink to /usr/src/linux right? |
| 16:01.06 | [sr] | think so |
| 16:01.13 | [sr] | i always use vanila source |
| 16:01.27 | WIMPy | Extract? |
| 16:01.37 | WIMPy | What kind of package is that? |
| 16:01.52 | spnl_ | downloaded linux-source-2.6.32.tar.bz2 into /usr/src |
| 16:02.33 | [sr] | be sure u know what your doing |
| 16:02.56 | [sr] | the debian kernel u're running has alot of stuff in modules that run via initrd etc etc, |
| 16:03.06 | [sr] | wrong compile of the kernel you won't boot again |
| 16:03.12 | spnl_ | yeh kinda hanging from WIMPy on this one.. :S |
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| 16:04.13 | spnl_ | not booting is not so bad, login to iLo and netboot a new install (just a pain in regards to time) |
| 16:05.00 | WIMPy | If you're unsure about the configuration, the packaged source probably includes the default config you're using now. |
| 16:05.50 | spnl_ | so just run the patch |
| 16:06.43 | WIMPy | You need to compile the stuff. |
| 16:07.01 | spnl_ | mISDNuser? |
| 16:07.21 | WIMPy | no, Linux |
| 16:07.43 | WIMPy | and mISDNuser |
| 16:08.01 | spnl_ | compile linux? a clean install is not enough? |
| 16:08.10 | spnl_ | i though mISDN was in the by default now |
| 16:08.38 | WIMPy | yes, but not for your card |
| 16:08.51 | spnl_ | aaaah ha |
| 16:09.29 | spnl_ | so run patch, compile linux then mISDN |
| 16:09.59 | WIMPy | misdnuser |
| 16:10.21 | spnl_ | yeah thats what i meant |
| 16:12.36 | spnl_ | Hunk #3 FAILED at 2235. |
| 16:14.12 | WIMPy | looks like there has been some change since then. |
| 16:14.31 | WIMPy | maybe you can patch it manually |
| 16:15.00 | spnl_ | yeh i think so |
| 16:15.09 | spnl_ | <PROTECTED> |
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| 16:19.05 | spnl_ | should be like this right |
| 16:19.05 | spnl_ | {PCI_VENDOR_ID_XTENSIONS,PCI_DEVICE_ID_XTENSIONS_XC520, PCI_ANY_ID_PCI_ANY_ID,0,0,(unsigned long)&hfc_map[23]}, |
| 16:20.04 | WIMPy | that line should be added. |
| 16:20.30 | spnl_ | i missed a comma and some spaces, bit fixed now and added |
| 16:21.15 | spnl_ | so next compile the new kernel right? |
| 16:21.37 | WIMPy | yes |
| 16:22.59 | spnl_ | like.. |
| 16:23.07 | spnl_ | fakeroot make-kpkg --initrd --revision=custom.1.0 kernel_mISDN |
| 16:23.46 | spnl_ | or must make-kpkg clean be run first? |
| 16:23.51 | WIMPy | no idea about debian specific stuff. |
| 16:24.43 | spnl_ | ok |
| 16:28.35 | spnl_ | BTW thanks again WIMPy i would of never got this far without you :) |
| 16:29.54 | spnl_ | i didnt have to build mISDNuser before the kernel? |
| 16:30.43 | [sr] | better after, my opinion |
| 16:30.50 | spnl_ | phew :P |
| 16:31.13 | WIMPy | yes |
| 16:32.17 | WIMPy | when that kernel is running |
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| 16:58.29 | Fritz09 | hi |
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| 17:28.26 | treborsux | I have a 560 polycom that runs just fine but after sitting idol i will find it froozew with running sip.app on the screen? |
| 17:28.31 | treborsux | anyone seen this before? |
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| 17:35.20 | x86 | did you upgrade to the latest firmware? |
| 17:39.17 | p3nguin | Is there a core module, such as pbx_config, that is responsible for checking variables in dialplan and whether they have underscores prepended or not? |
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| 18:20.05 | spnl__ | well building kernel died at the end lguest.c:37:18: error: zlib.h: No such file or directory make[1]: *** [lguest] Error 1 |
| 18:20.25 | spnl__ | im just installing CentOS 5.5 and see where that goes |
| 18:20.52 | WIMPy | Wouldn't it be easier to just install zlib? |
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| 18:30.17 | dym | :D |
| 18:30.18 | dym | agreed |
| 18:30.30 | spnl__ | maybe be after 4 days of things not working, im going with CentOS, seems to be the OS of choice, plus i know my card will work |
| 18:30.43 | dym | spnl__: why is that? you were clearly missing zlib. |
| 18:30.55 | dym | aka libzlib |
| 18:31.19 | WIMPy | And why do you think, CentOS is better? |
| 18:31.19 | spnl__ | and after installing that i recompile would be needed right? |
| 18:31.28 | spnl__ | took it 2 hours to get this far |
| 18:31.31 | dym | spnl__: obviously |
| 18:31.47 | WIMPy | No, just compressing the result. |
| 18:32.23 | dym | ?! |
| 18:32.31 | WIMPy | Make is there to find out what has already been done and what needs to be done. |
| 18:32.45 | dym | well |
| 18:32.53 | dym | make is basically the "compile" process :D |
| 18:32.58 | dym | :P |
| 18:33.21 | WIMPy | Yes, but that was finished. |
| 18:33.34 | spnl__ | going to grab some food, and have a go later |
| 18:33.37 | spnl__ | thanks for the help guys |
| 18:33.50 | dym | np |
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| 18:57.44 | doolittlework | hi there people |
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