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| 13:07.38 | makubi | hi. I am currently setting up an asterisk server for personal usage. Establishing connections between linphone works, but now I tried using the built-in android sip-client. Sometimes I am unable to contact it, asterisk debug shows "SIP/2.0 404 Not Found": http://paste.coding4coffee.org/?7fe79a36e19939d5#IG45q1OI9j1uvnHGgqQYcNDLHAPL0XQ2V61B8lWHxc0= Can anybody give me a hint what this means and how I can start debugging / try other th |
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| 13:09.06 | oquidave | hey, am having a little trouble setting up asterisk with gtalk. Up untill now, everything is setup, but when i've an incoming call from gtalk, the caller can hear me but i can't hear them. any pointers? thanks |
| 13:10.10 | WIMPy | makubi: Your trying to call a user (extension) that doesn't exist. |
| 13:10.29 | makubi | WIMPy: I am not even trying to call somebody |
| 13:10.43 | makubi | WIMPy: while this log statement occurrs |
| 13:11.07 | makubi | WIMPy: at least not actively. maybe the android client is broken |
| 13:11.39 | WIMPy | Then someone is trying to call (via) you. |
| 13:12.03 | WIMPy | I.e. the usual caracking attempts. |
| 13:12.15 | makubi | WIMPy: Are you sure? I did set a password for that user that is not like 1234 |
| 13:12.31 | makubi | and I did set allowguests=no (if that changes this) |
| 13:13.01 | WIMPy | That pastebin doesn't work for me. But You shoudl just turn up verbose instead of debugging, that should tell you waht's going on. |
| 13:13.02 | makubi | WIMPy: and the IP I am seeing is the one of my android client |
| 13:13.44 | WIMPy | Ok, so it might be a subscription attempt. |
| 13:13.58 | makubi | WIMPy: http://nopaste.info/51ced1487a.html |
| 13:14.51 | WIMPy | Ok, that's fine. OPTIONS are usually only used as keepalives. |
| 13:14.55 | makubi | you mean by starting asterisk with -vvvv ? I did set verbosity to 9 in the config |
| 13:15.08 | makubi | yeah, but what's about that 404? |
| 13:15.51 | WIMPy | That actually the most common answer to options. |
| 13:16.01 | makubi | Ok :) |
| 13:16.10 | makubi | So the problem is somewhere else |
| 13:16.20 | WIMPy | No, it just doesn't matter. |
| 13:16.49 | WIMPy | Unless that client really wants to request options, which I doubt. |
| 13:17.01 | makubi | Even if I am calling somebody and end the call, before the other one answers it, his / her phone rings infinitely |
| 13:17.26 | makubi | Ok |
| 13:17.40 | WIMPy | That might be a networking issue. |
| 13:17.56 | makubi | this does only happen with the native android client |
| 13:18.01 | makubi | not with linphone android |
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| 13:18.49 | makubi | but how can increase the verbosity? I'd really like to see whats going on and try to debug it |
| 13:19.12 | WIMPy | In that case sip debug is the right way. |
| 13:19.56 | makubi | WIMPy: also if I get something like unable to reach the other one? |
| 13:26.11 | makubi | well I'll just try to read that debug. One other question: What should I do to 'secure' the asterisk (like not allowing guests)? |
| 13:27.34 | WIMPy | Read the README-SERIOUSELY-bestpractices.txt |
| 13:28.37 | makubi | WIMPy: thanks. I just entered (in the android sip client) as outgoing-proxy my asterisk-server-address. This may sound dumb, but what for do I need a proxy server? |
| 13:28.59 | makubi | I know why I need STUN, etc. but a proxy for what? isn't the server always the proxy too? |
| 13:31.10 | WIMPy | Not neccessarily. |
| 13:31.18 | makubi | is it just for routing calls? |
| 13:31.38 | makubi | and what if I don't set any proxy server? wouln't I be completely unable to make calls? |
| 13:31.41 | WIMPy | The server itse;f could even be on a private net. |
| 13:31.51 | makubi | ok, this makes sense |
| 13:32.03 | WIMPy | No, without proxy the client would connect directly. |
| 13:32.15 | makubi | what do you mean by "directly"? |
| 13:32.28 | makubi | how does the client know how to connect to the other client? |
| 13:32.39 | WIMPy | Without proxy. |
| 13:33.01 | WIMPy | DNS, as usual. |
| 13:33.09 | makubi | alright |
| 13:33.29 | makubi | so each client directly via its IP |
| 13:33.45 | makubi | this may be the error why that android client did not work well |
| 13:35.51 | makubi | WIMPy: thank you very much for explaining things :) |
| 13:39.36 | makubi | WIMPy: wow, now calls do even end |
| 13:43.02 | makubi | WIMPy: If I don't set pedantic=no in my sip.conf, calls do not end (when hanging up before other one answers) on android clients. but why? |
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| 13:59.12 | WIMPy | https://www.youtube.com/watch?v=8pcWlyUu8U4 |
| 14:09.56 | makubi | WIMPy: Ok, I just tried being contacted / contact another client (both android clients) and the call ended instantly after answering the call: http://nopaste.info/5147a2cdb8.html do you have any idea? |
| 14:13.03 | makubi | WIMPy: trying it again a minute later and it works |
| 14:15.50 | makubi | WIMPy: is this video for real? ^^ |
| 14:18.55 | makubi | it is ridiculous how ppl say they dont care that they get spied on but if you ask somebody on the train if you could have his mobile phone they say 'ofc not' |
| 14:20.07 | WIMPy | What yu don;t see isn't real. |
| 14:20.11 | WIMPy | Or something like that. |
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| 14:23.46 | makubi | yup |
| 14:27.33 | makubi | it's interesting how franklin in 1755 said "They who would give up essential Liberty, to purchase a little temporary Safety, deserve neither Liberty nor Safety." |
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| 14:39.51 | WIMPy | True story. |
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| 14:51.37 | makubi | maaaan this android sp client drives me nuts |
| 14:53.20 | makubi | maybe its just the carrier that blocks it |
| 14:53.25 | zamba | we have one of these "wall-hanging" alcatel telephony "servers/centrals".. what word should i be using for this? |
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| 14:54.46 | zamba | this terminates four ISDN land lines from the PSTN.. what we want to do reprogram this so that we can still use this for the analogue fax machine and another analogue modem.. and then we want to hook a asterisk to this to perform some more complex functions.. |
| 14:55.02 | zamba | is this doable? |
| 14:55.20 | zamba | i was thinking about connecting the asterisk to this by using isdn |
| 14:55.32 | zamba | if using "isdn" is the correct term for the in-house connection :) |
| 14:55.34 | zamba | hehe |
| 14:55.48 | zamba | i'm not very savvy in the right terminology here :) |
| 14:56.51 | zamba | and we also want it so that the asterisk server can use all four lines at the same time |
| 14:56.55 | WIMPy | We don't know what ports that thing has (to spare). |
| 14:57.42 | zamba | after i've removed all the phones, it should have more than enough |
| 14:57.57 | zamba | and i know that it already has available ports |
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| 14:58.06 | WIMPy | If the ports for the phones allow you to connect anythign else. |
| 14:58.17 | zamba | but what are these things called? i wanted to search for the model and see if i could dig up any images or manuals for them |
| 14:58.39 | zamba | i know "alcatel".. and i know that it hangs on the wall.. and i know that i can remove a cover to get to the different ports.. and i know that it uses a serial interface to program it.. :) |
| 14:58.48 | WIMPy | You could always have an asterisk qith 8 S0 and put that between the lines and the PBX. |
| 14:58.51 | zamba | unfortunately i'm some miles away, so i can't just read off the model name of it |
| 14:59.07 | zamba | with 8 S0? what does that mean? |
| 14:59.17 | zamba | and the pbx is now the wall-hanging alcatel thing? :) |
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| 14:59.28 | WIMPy | 8 BRI ports. |
| 14:59.46 | WIMPy | 4 to connect the lines and 4 to connect the PBX. |
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| 15:00.34 | WIMPy | Although I wouldn't use Asterisk for that. I'd use LCR and then connect Asterisk to that. |
| 15:00.46 | zamba | meaning to connect asterisk to the PBX? |
| 15:00.52 | WIMPy | But that depends on what you want and what you need. |
| 15:00.53 | zamba | PSTN -> PBX -> asterisk |
| 15:01.08 | WIMPy | We don;t knwo if that's possible, |
| 15:01.19 | zamba | what i really need is to be able to still do fax and have an analoge modem and still utilize asterisk's features |
| 15:01.24 | WIMPy | But you can always do PSTN <> Asterisk <> PBX. |
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| 15:01.39 | zamba | WIMPy: but that would require more hardware in the asterisk box |
| 15:01.52 | zamba | and then we loose fax/modem if the asterisk box ever goes down |
| 15:01.53 | WIMPy | yes |
| 15:02.06 | WIMPy | yes |
| 15:02.27 | zamba | but what is the common term for the boxes that hangs up on a wall and does pbx services? just pbx? |
| 15:02.28 | WIMPy | What do you want from Asterisk? |
| 15:02.49 | WIMPy | yes |
| 15:02.52 | MaliutaLap | zamba: and that is different from losing the fax/modem if the PBX goes down in the other situation |
| 15:02.59 | WIMPy | That must e a PBX. |
| 15:03.33 | WIMPy | be |
| 15:03.50 | zamba | WIMPy: i want better call handling.. we are a radio station and we want to route callers with more flexibility.. we don't want everyone to occupy lines which is not going on air and so on |
| 15:03.52 | zamba | and some just want to leave a message |
| 15:03.54 | Kobaz | anyone notice a performance regression between asterisk 1.8.15 and 1.8.25 |
| 15:03.54 | MaliutaLap | passes WIMPy [s]ed |
| 15:04.58 | WIMPy | Well, how would it help to connect Asterisk to the PBX then, instead of the other way round? |
| 15:05.00 | zamba | alcatel 4200 |
| 15:05.32 | zamba | http://condepot24.de/Telefonanlage-Alcatel-4200-Anlage |
| 15:05.34 | zamba | that's the one |
| 15:06.54 | MaliutaLap | if you can get a SIP card for the PBX you can great */PBX gloop and make things work |
| 15:07.03 | WIMPy | Not too much information, but I assume it's using Up0 as telephone ports. |
| 15:08.03 | MaliutaLap | WIMPy: there are no telephone ports, just * - a PBX and the PSTN :) |
| 15:08.37 | zamba | WIMPy: was that for me? Up0? |
| 15:09.52 | MaliutaLap | FWIW I did a project to use * to increase the number of VM accounts rather than than the rip off price the vendor was going to charge. because of difficulties getting any type of IP comms with the PBX the budget got blown |
| 15:11.08 | MaliutaLap | unless you can get cheap VoIP units for the PBX you're probably better off look at a different solution, say go * all the way, or pay big $$$'s to your PBX provider |
| 15:11.54 | zamba | MaliutaLap: are you talking to me? :) |
| 15:12.01 | WIMPy | What kind of sense would a PBX make without phones? |
| 15:12.26 | MaliutaLap | zamba: I'm just talking |
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| 15:12.47 | MaliutaLap | WIMPy: it would make most peoples lives easier |
| 15:13.06 | MaliutaLap | WIMPy: all automated call attendants :) |
| 15:13.14 | WIMPy | zamba: Might be a good idea if you start by describing what exactely you want to change. |
| 15:13.52 | MaliutaLap | good point, zamba has been a little light on the detail |
| 15:13.53 | makubi | this android client always sends 486 busy here instead of answering the call. any ideas? |
| 15:14.04 | WIMPy | All auto atendants doesn't make sense with a PBX. That's clearly an Asterisk task. |
| 15:14.14 | zamba | WIMPy: first of all i want to know if it's possible to hook the asterisk to the pbx somehow.. secondly i want to know if it's possible to route 2 of the 20 available numbers to the analoge fax and the analoge modem and the 18 other numbers to asterisk.. |
| 15:14.19 | MaliutaLap | makubi: the phone is busy? |
| 15:14.55 | WIMPy | zamba: Probably, but we don't know what ports you have. |
| 15:15.05 | zamba | in my asterisk i have two of these cards: 01:07.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 15:15.26 | WIMPy | zamba: But you shouldn't care about technical details at this point. What's the goal of whatever might happen? |
| 15:15.27 | MaliutaLap | zamba: you need to be able to get a VoIP/SIP module for your PBX - these can be expensive, if the PBX isn't already SIP/VoIP |
| 15:16.02 | WIMPy | zamba: That's only two ports. |
| 15:16.04 | MaliutaLap | zamba: then you use the PBX to route what you want where you want |
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| 15:16.18 | zamba | MaliutaLap: can't i talk isdn between the pbx and *? |
| 15:16.30 | WIMPy | Also we don't know what kind of lines you have connected. |
| 15:16.37 | zamba | WIMPy: the goal is to have more flexibility for incoming calls.. i've already said that :) |
| 15:16.41 | makubi | MaliutaLap: nah, its not :) |
| 15:16.48 | MaliutaLap | zamba: but what your talking about means that you might aswell get rid of the PBX and terminate your lines onto the * box |
| 15:16.57 | zamba | MaliutaLap: fax and analog modems? |
| 15:17.00 | MaliutaLap | zamba: * handles fax passthrough |
| 15:17.07 | WIMPy | zamba: And I already asekd how that could work if calls go to the PBX first. |
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| 15:17.27 | zamba | WIMPy: i have no idea.. ? i thought the pbx would direct the calls to asterisk? |
| 15:17.37 | makubi | MaliutaLap: WARNING[16635]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 592927d993484794fff18a93f6464cd8@10.31.67.29 for seqno 103 (Critical Request) |
| 15:17.38 | zamba | MaliutaLap: over voip? |
| 15:17.49 | zamba | MaliutaLap: i need a analoge adapter for that, right? |
| 15:17.49 | WIMPy | Yes, but it still has the calls then. So what's the point? |
| 15:18.02 | WIMPy | You've being extremely vague about the goal. |
| 15:18.21 | MaliutaLap | zamba: how much is it going to cost you to keep around the PBX around, and can it be justified over the cost of 1-2 analogue lines dedicated to modems |
| 15:18.28 | zamba | WIMPy: the point is to get the flexibility of the asterisk.. i need to create IVRs with more complexity for the callers.. i need to route them to different queues and i need to handle this programmatically |
| 15:18.50 | zamba | WIMPy: are you saying that the alcatel pbx can do all this for me? |
| 15:19.05 | zamba | WIMPy: i also want to get voicemails that i want to be put into mp3 files and placed into separate folders |
| 15:19.09 | WIMPy | zamba: Ok, that doesn't make sense the way you describe it. You need to connect the PBX to Asterisk, not Asterisk to the PBX. |
| 15:19.34 | zamba | huh? |
| 15:19.35 | WIMPy | The Alcatel can probably do all that. But it might be cheaper to put Asterisk in front. |
| 15:19.41 | zamba | in front of what? |
| 15:19.49 | zamba | the phones or the PSTN? |
| 15:19.52 | WIMPy | The PBX. |
| 15:20.02 | zamba | incoming calls: PSTN -> PBX -> * |
| 15:20.16 | WIMPy | No. |
| 15:20.26 | zamba | incoming calls for fax and modeom: PSTN -> PBX -> fax/modem |
| 15:20.44 | zamba | all other incoming calls: PSTN -> PBX -> * -> <logic> -> whatever |
| 15:20.50 | WIMPy | If you want to do queues with Asterisk, it would mean that te call would go PSTN > PBX > Asterisk > PBX. |
| 15:21.01 | zamba | huh? |
| 15:21.07 | zamba | the queues should never leave the asterisk..? |
| 15:21.27 | WIMPy | And as far as the fax/modem thing goes it would help to know what kind of lines you have. |
| 15:21.34 | zamba | analoge lines |
| 15:21.38 | zamba | i believe i've already said that |
| 15:21.55 | WIMPy | So far you only mentioned 4 ISDN lines. |
| 15:22.23 | zamba | we have ISDN to the building.. and the alcatel pbx is interfaced to the PSTN (is this the correct way of saying it?).. the PBX has some analoge ports (quite a few, i believe) and the fax machine and some phones are connected to these |
| 15:22.55 | WIMPy | That's waht I got. |
| 15:23.06 | WIMPy | But what are those ISDN lines? |
| 15:23.11 | zamba | huh? |
| 15:23.16 | zamba | the connection to the outside world? |
| 15:23.17 | WIMPy | Are they one trunk or seperate or both? |
| 15:23.31 | zamba | there's two "boxes" on the wall for them :p |
| 15:23.36 | zamba | i have no idea what they're called |
| 15:23.45 | WIMPy | Two? |
| 15:23.48 | zamba | yup |
| 15:23.54 | WIMPy | So it's only two lines? |
| 15:24.05 | WIMPy | (with 2 channels each) |
| 15:24.09 | zamba | oh.. lines/channels.. stupid me.. |
| 15:24.11 | zamba | yeah |
| 15:24.16 | zamba | i meant four channels, two lines |
| 15:24.35 | WIMPy | Ok. |
| 15:24.46 | WIMPy | What kind of lines? |
| 15:24.59 | zamba | and i've previously been able to connect my asterisk directly to these lines by using the card i pasted the lspci output of earlier |
| 15:25.05 | WIMPy | And what exactely do you want to do with your callers? |
| 15:25.07 | zamba | Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 15:25.19 | MaliutaLap | zamba: I don't see an easy way of adding VoIP to the PBX. If you don't want to use T.38 pass through for the fax/modem then I suggest pitching the PBX, bringing the 2/3 PSTN pairs you need for the fax/modem. Then just run the rest through * |
| 15:25.29 | WIMPy | The cards don;t care about the type as long it's a BRI. |
| 15:26.08 | zamba | WIMPy: i don't know exactly what i want to do with the callers once they hit the asterisk.. but once i've got them there i can start being creative with the dialplan and add new features.. |
| 15:26.12 | zamba | WIMPy: does that matter? :) |
| 15:26.37 | zamba | WIMPy: this is for incoming calls.. once they hit the asterisk, they won't ever be routed back to the pbx, unless it's a new outgoing call |
| 15:26.51 | WIMPy | It does matter if you want to be able to send them to the PBX. |
| 15:26.51 | MaliutaLap | zamba: make it easy, just play tt_monkeys and hang up ;) |
| 15:27.05 | zamba | WIMPy: why should i send them back to the PBX? that makes no sense :) |
| 15:27.18 | MaliutaLap | WIMPy: I think the pbx is moot |
| 15:27.23 | WIMPy | What does "new outgoing call" mean in that context? |
| 15:27.28 | zamba | WIMPy: Dial() |
| 15:27.54 | WIMPy | If you don't want to send them to the PBX, what's the PBX good for now? Only a adaptor for the fax? |
| 15:28.32 | MaliutaLap | WIMPy: I think it should be removed from the proposition. A) it complicates things. B) it's redundant |
| 15:28.42 | WIMPy | Are you going to use any phones on the PBX? |
| 15:29.38 | zamba | WIMPy: yup.. the fax is a "phone", right? |
| 15:29.43 | zamba | WIMPy: and the modem could be a phone as well |
| 15:29.54 | WIMPy | No. Phone, not fax. |
| 15:29.55 | zamba | WIMPy: i basically just need two analoge lines (meaning no adapter) |
| 15:30.07 | WIMPy | So you only want to connect those two devices? |
| 15:30.15 | zamba | WIMPy: directly to the PBX, yeah |
| 15:30.23 | zamba | WIMPy: the rest is with sip adapters |
| 15:30.50 | zamba | "digitalen UA- Ports" |
| 15:31.08 | WIMPy | UA? |
| 15:31.14 | zamba | http://condepot24.de/Telefonanlage-Alcatel-4200-Anlage |
| 15:31.17 | zamba | that's from that page |
| 15:31.25 | WIMPy | Is that Alcatel speak for Up or for a/b? |
| 15:31.30 | zamba | WIMPy: you're german, right? |
| 15:31.49 | WIMPy | There's no sip on that page. |
| 15:31.53 | WIMPy | yes |
| 15:31.59 | zamba | i've never said i wanted to interface to the PBX using SIP |
| 15:32.17 | WIMPy | Yes, I guess UA is Alcatelese for Up |
| 15:32.23 | zamba | i was thinking about using the Cologne Chip cards i've pasted the description of two times now :) |
| 15:32.34 | WIMPy | <zamba> WIMPy: the rest is with sip adapters |
| 15:32.46 | zamba | for the phones behind *, yeah |
| 15:32.54 | zamba | since that will be handled in the *-domain, so to speak |
| 15:33.02 | WIMPy | You can use them to connect your Asterisk to your lines. |
| 15:33.18 | zamba | aha |
| 15:33.32 | zamba | meaning connect * to the PBX? |
| 15:33.39 | WIMPy | Ok, so you wan't to connect all phones to Asterisk and need two analogou ports for fax an modem. And that's all? |
| 15:33.45 | zamba | WIMPy: that's it |
| 15:33.50 | WIMPy | No, to the lines. |
| 15:34.01 | zamba | what are "my lines"? |
| 15:34.16 | zamba | that's ISDN, right? i can't use SIP adapters for that.. i need to interface this over ISDN? |
| 15:34.27 | WIMPy | I don;t know if the PBX has any spare S0 ports you could connect the cards to. |
| 15:34.39 | zamba | i believe it has |
| 15:34.59 | WIMPy | If it has, you can do it like that. |
| 15:35.01 | zamba | but i was with you with using the analoge ports on the PBX and then hook some SIP adapters to those and then interface |
| 15:35.10 | zamba | exactly :) |
| 15:35.18 | zamba | that was basically what i was asking 45 minutes ago ;) |
| 15:35.30 | zamba | but i don't know the right terms |
| 15:35.49 | WIMPy | It wasn't clear what the goal was back then. |
| 15:36.28 | WIMPy | If you fax number is on a ptmp line, you could just connect the PBX in parallel to Asterisk to the line. |
| 15:36.37 | zamba | i don't know what a ptmp line is |
| 15:36.51 | zamba | the fax machine is connected directly to the pbx |
| 15:37.04 | zamba | over an analoge line |
| 15:37.20 | WIMPy | Are they one trunk or seperate or both? ptp or ptmp? |
| 15:37.32 | zamba | ... i have no idea what you're saying now :) |
| 15:37.37 | zamba | are what what? :) |
| 15:37.40 | WIMPy | Don't call a port on the PBX a line. |
| 15:37.56 | zamba | ah, ok |
| 15:37.58 | WIMPy | See the contract with your telco. |
| 15:38.12 | WIMPy | Or the configuration of the PBX. |
| 15:38.26 | zamba | yeah, i have to dig up the programming manual for the pbx somehow |
| 15:38.52 | zamba | but i have set up lcr towards the lines earlier |
| 15:39.01 | zamba | if that can help you figure that out? |
| 15:39.33 | zamba | i have lcr installed here |
| 15:40.07 | zamba | interfaces.conf says [ISDN] extern, portnum 0, ptp, portnum 1, ptp |
| 15:40.08 | WIMPy | The call debug would tell what it is, yes. |
| 15:40.35 | zamba | but they're not connected atm. |
| 15:40.36 | WIMPy | Ah, so it's ptp. In that case you can only connect one device per line. |
| 15:41.05 | zamba | but fax on ptmp line? this is still ISDN.. the fax machine is analoge, of course |
| 15:41.13 | WIMPy | So you need to connect te PBX and Asterisk. |
| 15:41.32 | zamba | ok.. so.. lines -> PBX -> asterisk/fax/modem, right? |
| 15:41.54 | WIMPy | ptmp only means the line hase management capabilities to manage multiple devices. |
| 15:42.04 | WIMPy | yes |
| 15:42.15 | WIMPy | If the PBX has the neccessary ports. |
| 15:42.17 | zamba | well.. i don't KNOW for sure that the line is ptmp or ptp.. i just know what the configuration of lcr says |
| 15:42.20 | zamba | S0, yeah? |
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| 15:42.33 | WIMPy | yes |
| 15:42.39 | zamba | if not i could still use a sip adapter and then use some analoge ports (four in total) and then SIP towards asterisk, right? |
| 15:42.58 | WIMPy | Ok, ptp configuration usually works on ptmp lines. But never the other way round. |
| 15:43.18 | WIMPy | SIP adaptor where? |
| 15:43.21 | zamba | http://www.altel.ro/en/alcatel-4200E-presentation.html |
| 15:43.39 | zamba | WIMPy: so i could try changing this to ptmp and see if it works? |
| 15:43.44 | zamba | WIMPy: between the PBX and * |
| 15:43.52 | WIMPy | yes |
| 15:44.34 | WIMPy | Unless te PBX speaks SIP itself. You don't want to use any adaptor between te PBX and Asterisk. |
| 15:44.57 | zamba | it definitely doesn't |
| 15:45.09 | zamba | i don't think SIP was even invented when this PBX was installed |
| 15:45.33 | zamba | 4. S0T0 ISDN card - is a basic ISDN access and allows to connect 4 S0 or T0 busses or busses 4 S0 T0 + 4 busses or busses S0 8, or 8 busses T0. This card may have a daughter board to power S0PS4 4 sites S0 bus and board a bus transfer T0CT daughter at a bus S0 T0. |
| 15:45.35 | zamba | 5. T2 ISDN card ISDN primary access (30 B channels and one D channel). |
| 15:46.03 | zamba | the T2 is for the "uplink" to the PSTN and the S0T0 isdn card provides me with the S0 ports, right? |
| 15:46.44 | WIMPy | If you have that card, you should have to spare ports, yes. |
| 15:47.19 | WIMPy | s/to/two/ |
| 15:47.30 | zamba | hehe |
| 15:47.45 | zamba | let's just see if i'm able to dig up the manual for that pbx |
| 15:49.51 | WIMPy | And for the cars make sure you use a kernel that was released after yesterday or a longer time ago. The driver for those cards had a bug. |
| 15:50.54 | zamba | Linux pbx 3.2.0-56-generic-pae #86-Ubuntu SMP Wed Oct 23 17:51:27 UTC 2013 i686 i686 i386 GNU/Linux |
| 15:51.34 | WIMPy | 3.2.30 was ok. |
| 15:52.01 | zamba | oh |
| 15:52.30 | WIMPy | Your should be fine. |
| 15:52.31 | zamba | running ubuntu 12.10 here |
| 15:52.52 | WIMPy | Unless they messed it up. |
| 15:53.43 | zamba | so to summarize.. what i need to figure out is how many S0 ports i have (i need 2, right) and i also need to get the installation manual for the alcatel 4200 |
| 15:54.23 | WIMPy | You need one port per 2 call you want to send to Asterisk. So probably 2, yes. |
| 15:54.34 | WIMPy | s/ll/lls/ |
| 15:56.51 | zamba | god damn.. it wasn't easy to find that manual |
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| 16:31.26 | gusto | hi WIMPy |
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| 17:07.31 | ghost75 | callerid(name) has a limit of 18 chars, can this be changed? |
| 17:08.03 | WIMPy | It doesn't. |
| 17:08.16 | WIMPy | Maybe your phone has a limit. |
| 17:08.20 | ghost75 | oO |
| 17:11.31 | Kobaz | ghost75: you can set it to as long as you want... (but carriers will truncate it) |
| 17:11.40 | ghost75 | ok outcall showing full in log :> |
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| 18:49.20 | eirirs | nickserv finally worked |
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| 19:04.37 | zamba | WIMPy: looks like i have S0 here |
| 19:04.50 | zamba | T0 ISDN TRUNK, four connectors |
| 19:06.29 | zamba | and it's an alcatel 4200 C |
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| 20:24.07 | admin0 | hi all .. can I register at 2 providers and receive all incoming calls form one provider to send to another provider ? |
| 20:24.29 | Penguin | Sure. |
| 20:25.17 | Penguin | Create a peer entry in sip.conf for both providers. Create an extension that runs Dial(). Done. |
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| 22:35.30 | WIMPy | zamba: Tell us when it's talking to you. |
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