00:00.19 | greenwolf | being rude is ignorant |
00:00.25 | greenwolf | i was just asking for alittle help |
00:00.49 | greenwolf | everyone isnt perfect you kno |
00:01.03 | jpsharp | Sounds like your vps provider gave you a non-standard kernel. |
00:01.21 | greenwolf | yes jpsharp i am pissed they did this |
00:01.24 | WIMPy | Most distro kernels are non-standard. |
00:01.40 | WIMPy | Probaly almost all. |
00:01.42 | greenwolf | but other ppl are pertaining its my knowledge of things causing this |
00:01.52 | jpsharp | non-distro-standard, then. |
00:02.00 | greenwolf | in which it isnt because obviously they couldn;t figure the problem out either |
00:02.03 | greenwolf | huh |
00:02.07 | WIMPy | The non-standard standard? |
00:02.13 | greenwolf | lol wimpy good one |
00:03.00 | jpsharp | Bewarned, though, if you're running on a virtual server, you may get weird timing issues depending on how heavily your provider loads there servers and what virtualization platform they use. |
00:03.11 | jpsharp | s/there/their |
00:03.39 | greenwolf | i have many friends that are voip engineers that recommended these guys to me |
00:03.46 | greenwolf | they are really good and so i used them myself |
00:03.54 | Penguin | If it were MY VPS, I would determine what dahdi version I can get my hands on, determine which kernel version it was built against, and install or have the provider install THAT kernel version and source for me, followed by installing the dahdi version that I just found available. |
00:03.57 | greenwolf | they run multiple actually entire systems on their network with no problem |
00:04.13 | WIMPy | And it seems that it's still not a good idea to use non-standard kernel, BTW. |
00:05.03 | *** join/#asterisk nightrid3r (~kvirc@62.205.65.49) |
00:05.06 | greenwolf | yea i dont kno why they are using a non-standard kernel and why they keep putting this kernel version in all their distros and flavors its really weird if you ask me |
00:05.19 | Penguin | Or, alternatively, I would install whatever version of Linux was available where I could get both the kernel and the source packages, then I would build the dahdi stuff from source so that it does not care which kernel version I am using. |
00:05.21 | greenwolf | coreyf1513: you said something about timingfd |
00:06.09 | greenwolf | yea i have installed many dahdi no problem...seems this system is the only time im getting this error but its ok ill get thru it |
00:06.27 | WIMPy | I told you that you don;t need dahdi, except for MeetMe. But if you can't upgrade to ConfBridge, you do need dahdi. |
00:06.31 | greenwolf | then ill post if on a forum for others who run into this problem and need help down the road so they dont run into this mess |
00:06.38 | coreyf1513 | greenwolf: that only provides timing.. meetme requires dahdi besides timing... |
00:06.47 | greenwolf | oh i see..damn |
00:07.22 | greenwolf | yea my employer is requiring me to use meetme cuz we have many scripts that were coded for meetme conferences and options within those conferences |
00:07.24 | WIMPy | But in the time you spent trying to install dahdi, you might have been able to change your dialplan from MeetMe to ConfBridge. |
00:07.27 | greenwolf | so its a must i use meetme |
00:07.51 | greenwolf | true wimpy...true im starting to see that myself :) |
00:07.52 | greenwolf | lmao |
00:08.15 | greenwolf | well i got one more trick up my sleeve im going to email them to run a different kernel source here..lets see what kind of response i get from them |
00:08.28 | WIMPy | And ConfBridge will scale acroll more than one CPU unlike MeetMe. |
00:08.42 | WIMPy | across |
00:09.56 | Penguin | I would ask them to install the latest kernel and source packages for that CentOS branch. |
00:10.16 | greenwolf | yes that is exactly what im requesting them to do now penguin..thanks |
00:10.38 | greenwolf | its their fault im running into such a problem becuase they are doing something weird with their kernel sources for some reason |
00:10.52 | jpsharp | Probably patched to run properly on the VPS hose. |
00:10.52 | Penguin | Then you at least have a chance that there is a dahdi package that matches. If you don't have one that matches, install it from source. |
00:10.54 | jpsharp | host. |
00:10.56 | greenwolf | i dont understand why they keep using that non-standard kernel source and why im having such problems with it |
00:11.17 | greenwolf | yea i was also thinking of installing it from source myself |
00:11.23 | Penguin | It seems that all VPS providers patch the kernels. |
00:12.25 | greenwolf | i see |
00:12.35 | Penguin | If they can't give you the correct kernel and source packages, maybe they could build a dahdi package that matches the package versions they have provided you. |
00:12.53 | jpsharp | Cause they're all running Xen or OpenVZ, which doesn't always play right with a generic kernel. |
00:15.13 | *** join/#asterisk greenwolf_ (186788ee@gateway/web/freenode/ip.24.103.136.238) |
00:15.25 | greenwolf_ | sorry lost connection..yes jpsharp they are running OpenVZ |
00:16.51 | greenwolf_ | i noticed on the asterisk.org website they only let you download dahdi 2.6.1 is there a way to access the older dahdi files? |
00:17.06 | jpsharp | There's your answer as to why they're using a "nonstandard" kernel. |
00:17.26 | greenwolf_ | yea stupid OpenVZ |
00:17.55 | greenwolf_ | yup your completely right jpsharp and penguin...bullshit if you ask me lol |
00:18.31 | Penguin | You aren't likely to need an older dahdi version. |
00:18.45 | greenwolf_ | ok |
00:18.50 | Penguin | The problem is because of packages and package versions, not because of the dahdi version. |
00:19.12 | jpsharp | Yep, you're going to need matching kernel + source to build dahdi no matter what. |
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01:00.31 | WIMPy | wants a pickup for VM. |
01:00.48 | Penguin | What will it do? |
01:01.04 | WIMPy | pick up a call that has gone to VM. |
01:01.23 | Penguin | channel redirect <--- |
01:01.42 | WIMPy | Yes, but it takes too long to type that. |
01:01.55 | gusto | i watched now a little of that SIP debug and i see that there is a lot of nonsense in there |
01:02.39 | gusto | for example from ATA's it says the right addresses to have something got from, but the header is wrong, for example To: and From: are mixed up |
01:02.55 | Penguin | Are you sure about that? |
01:04.56 | gusto | yes |
01:05.51 | gusto | when he is saying that he just recieved something from addressA and then in the header is TO: addressA and FROM: addressB when addressB is asterisk itself, then its pretty damn sure |
01:06.12 | Penguin | Sounds right to me. |
01:06.45 | Penguin | Communications between asterisk and the device are two-way, not one-way. |
01:06.56 | gusto | yes |
01:08.36 | gusto | Penguin: http://pastebin.com/zGxwueME take a look at this sample |
01:09.17 | gusto | 192.168.27.49 is asterisk here |
01:11.46 | Penguin | Yeah, what's the problem? It's a two-way transaction, so there are data from the deive to asterisk, and from asterisk to the device. |
01:11.57 | gusto | but in that sample you only see the other devices (linksys and cisco) mixing up from: and to: |
01:11.58 | Penguin | s/deive/device/ |
01:12.13 | Penguin | It's not mixing up anything. |
01:12.37 | gusto | <--- SIP read from UDP:192.168.23.19:5060 ---> To: <sip:dedko@192.168.23.19:5060>;tag=324a73616f09fa1di0 ??? |
01:12.46 | gusto | that would mean that he is sending that to himself |
01:13.00 | Penguin | It doesn't mean that. |
01:13.10 | gusto | what does it mean then? |
01:13.54 | Penguin | It means that asterisk has a packet that is To: 192.168.23.19, just like it says. |
01:14.35 | gusto | and why is it in between of SIP read from UDP:192.168.23.19:5060 then? |
01:15.09 | gusto | would it be asterisk's packet then he could take "Reliably Transmitting (no NAT) to 192.168.27.187:5061" |
01:15.23 | gusto | or 23.19, i mean, however |
01:15.53 | Penguin | Why is your phone on the wrong port? |
01:16.25 | gusto | you mean that 5061 one? |
01:16.29 | Penguin | Yes |
01:16.54 | gusto | because these devices have 2 analogue phone connectors and one is on 5060 and the second on 5061 and so on |
01:17.11 | Penguin | I see. I have to do that, sometimes, too. |
01:17.37 | Penguin | If they are on the same LAN as asterisk, that isn't necessary, though. |
01:18.06 | Penguin | At least I think that's true. |
01:18.40 | gusto | no, it's that device |
01:18.46 | gusto | one of them is PAP2T |
01:19.05 | gusto | Server: Linksys/PAP2T-3.1.15(LS) |
01:19.19 | gusto | you can not have these lines both on 5060 |
01:19.32 | Penguin | Why not? |
01:19.51 | gusto | because then that device would need 2 IP addresses |
01:20.28 | Penguin | I'm sure I have tested this theory before, but I can't remember the results. |
01:20.39 | Penguin | And I have a tendency to not take notes. |
01:20.51 | gusto | in theory it would work, but the firmware of this devices will not let you |
01:22.09 | mathi | how do I pass a call from one asterisk server to the other ? |
01:22.15 | gusto | of course, asterisk is also using only one port and you have a lot of connections going into it, and of course it could hold the lines apart over SIP registrations/usernames or whatever, but these devices are not capable to do this |
01:22.21 | mathi | WIMPy, ? |
01:22.38 | gusto | mathi: SIP/asterisk2/call |
01:22.39 | Penguin | Depending on your definition of pass, you might want to use Dial(). |
01:22.56 | mathi | gusto, what is that? |
01:23.04 | mathi | a Dial ? |
01:23.07 | Penguin | Dial() <-------- |
01:23.19 | WIMPy | mathi: yes |
01:23.29 | mathi | won't I lose the original Caller ID if I dial the other server ? |
01:23.42 | Penguin | No |
01:23.59 | Penguin | You will only if you set a different caller id. |
01:24.04 | Penguin | So don't do that. |
01:24.05 | mathi | caller -> server A -> server B. I was expecting server B to have server A as Caller ID ? |
01:24.29 | Penguin | Servers don't have callerids. |
01:24.31 | WIMPy | Only if you configure a caller ID for that account on server B. |
01:24.53 | mathi | nice then |
01:25.07 | WIMPy | "that" = the one Server A uses. |
01:25.38 | mathi | WIMPy, you mean, i don't have to configure a caller ID on server A |
01:25.49 | Penguin | You shouldn't. |
01:25.57 | WIMPy | No. |
01:26.03 | Penguin | http://pastebin.com/Ag7tknm2 See here. |
01:26.21 | WIMPy | You shouldn't configure a caller ID for te account that accepts the calls. |
01:26.45 | Penguin | I recommend only setting callerid for phones. |
01:27.21 | WIMPy | exately |
01:27.44 | mathi | Penguin, can you tell me what happens in your code ? |
01:27.59 | Penguin | Huh? |
01:28.25 | Penguin | This is sample configuration for two asterisk systems to communicate with SIP. |
01:30.07 | mathi | Penguin, well with phones, you need to register them. how does it work here ebtween two servers ? |
01:30.32 | Penguin | Phones register to one or the other, or via proxy. |
01:30.49 | WIMPy | You don;t have to register phones. |
01:31.03 | mathi | hah |
01:31.05 | WIMPy | You have to register if the other end doesn;t know where to find you. |
01:31.07 | Penguin | It's helpful for them to be registered to get phone calls. |
01:31.07 | mathi | I always registered them |
01:31.37 | mathi | Penguin, so in your example the servers don't register between them |
01:31.59 | Penguin | Right, they are configured statically. |
01:32.50 | Penguin | If you need to register each one to the other, that's an easy change to my sample. |
01:35.03 | mathi | then all I need to do is pass the call from server A to server B in extensions.conf ? Dial(SIP/miami) |
01:36.16 | WIMPy | You might want to call a specific extension on "miami". |
01:36.20 | Penguin | No, the format for Dial is Tech/peer/extension |
01:36.40 | WIMPy | With "/extension" being optional. |
01:36.56 | WIMPy | So mathis exaple is correct. |
01:36.59 | mathi | well tech is SIP, peer is miami ... |
01:37.24 | WIMPy | Just probably less than he might want. |
01:37.32 | Penguin | I'm quite certain that you won't want to call asterisk itself. You'll want to call another device that is on that system. |
01:37.55 | WIMPy | You don't call devices. You call extensions. |
01:38.03 | Penguin | True story there. |
01:38.36 | mathi | Penguin, I need to enter in an IVR in server B, so I need to get into a specific extension. I thought that was the reason of your extensions "from-miami" and "from-tampa" |
01:38.48 | WIMPy | But possibly knowing where the call comes from is enough. |
01:38.55 | Penguin | Those are not extensions. |
01:39.18 | mathi | they must be, context is pointing to extensiosn |
01:39.20 | Penguin | Those are contexts. |
01:39.34 | Penguin | Within those contexts, you will put extensions. |
01:39.55 | mathi | oops |
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01:42.15 | mathi | Penguin, for the servers to register between them, I should so something like this? register => 1234:pwd@... |
01:43.04 | Penguin | If you don't have static IP addresses and you don't have dynaic DNS host names, you can register each to the other like that. |
01:43.18 | mathi | Penguin, it was just out of curiosity |
01:43.43 | Penguin | Change host= to dynamic and add a register statement to each one. |
01:44.10 | WIMPy | To each one doesn't make too much sense. |
01:44.32 | mathi | WIMPy, it seems it does if IP's are dynmaic ? |
01:44.40 | WIMPy | No |
01:44.52 | Penguin | Well, you'll have to know at least one host or the other can't register to it in the first place. |
01:45.12 | mathi | ah, you have to register only one ? |
01:45.18 | WIMPy | If you don't have at least a dyndns name that you could configure as host, where are you going to send the register? |
01:45.38 | mathi | right) |
01:45.53 | mathi | WIMPy, but only one server needs to register to the other then ? |
01:46.01 | WIMPy | correct |
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01:46.31 | WIMPy | Or none, if both have at least a known name. |
01:47.24 | mathi | WIMPy, but even if both have a static IP, is it ok to register ? or is it less performant ? |
01:47.43 | WIMPy | You can do it, but it doesn't make sense. |
01:50.38 | mathi | server B has only one extension matching for the phones, all the other extensions should be forwarded to server A. Is this possible to achieve? |
01:51.06 | WIMPy | sure |
01:51.38 | WIMPy | But are you sure you need server A at all? |
01:53.08 | mathi | WIMPy, well, first I thought I would replace the PBX systems at the doctors, but then I was thinking I can just add my server to their existing server, with the technique of peer between server that penguin pasted. I think it is better solution |
01:54.11 | mathi | WIMPy, what do you think? |
01:54.31 | mathi | I think I will get rid of all the hassle about these cards and all ... |
01:54.49 | Penguin | If they already have asterisk, why do you need another asterisk? |
01:54.54 | WIMPy | Do you think SIP will be easier? |
01:56.22 | mathi | Penguin, yes I could integrate my dialplan into their servers, but unfortunately they have asterisk running on small modems, and my IVR requires a server running applications among php, curl, ... |
01:56.33 | mathi | WIMPy, sure, very easy |
01:56.40 | mathi | WIMPy, as long as it's local |
01:57.03 | Penguin | Is it possible to eliminate their original system completely and use only the bigger/better system? |
01:57.57 | mathi | Penguin, yes, but adding a server is easier. If i need to replace their system, it will cost more too, need to buy a card, need to recode all their extensions, register their phones, ... |
01:58.11 | mathi | if I add my server, I can just forward extensions to original server |
01:59.05 | mathi | Penguin, the main problem is the card, their phone line, ... |
01:59.23 | WIMPy | doesn't see a problem there. |
01:59.32 | WIMPy | More of the opposite. |
01:59.44 | mathi | WIMPy, well in a perfect world there is no problem ... |
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02:01.03 | mathi | but clearly these are my two options I think. replace their server VS adding my server... |
02:01.42 | WIMPy | And how is "adding your server" going to work? |
02:03.12 | mathi | WIMPy, well with Penguin technique to make a peer between the two servers. and the original server (A) will forward the calls to my server (B) through SIP locally |
02:03.25 | mathi | WIMPy, something wrong ? |
02:03.42 | WIMPy | But there's no Server A, yet? |
02:03.54 | Penguin | You basically just need to add an auto-attendant and/or an IVR, so it seems. Is that correct? |
02:03.56 | WIMPy | And Server B is local? |
02:04.03 | WIMPy | I don't get that setup. |
02:04.05 | Penguin | But the existing box cannot handle it. |
02:04.21 | Penguin | I would dump the original. |
02:04.24 | mathi | WIMPy, there is, that's the existing server at the doctor's place (in this concrete example, a small modem with asterisk) <= server A |
02:04.30 | WIMPy | Yes, what do you want to add there? |
02:04.33 | Penguin | Set up the new box, configure everything, then cut over. |
02:04.40 | mathi | WIMPy, and server B is my server that I will add in the network with Ethernet |
02:05.18 | WIMPy | Why do you want to keep a mini Asterisk if tou have a big one anyway? |
02:05.27 | WIMPy | And what is that mini Asterisk doing at the moment? |
02:05.34 | mathi | Penguin, the problem is the card and the phone line. I need to check if the service that they use allow me to buy a cheap digital card as WIMPy suggested, but if it's not possible then it sucks |
02:06.08 | mathi | WIMPy, that mini asterisk server is handling incoming calls and outgoing calls, with a very basic auto-attendant |
02:06.10 | Penguin | Oh, I missed that part of the puzzle. |
02:06.15 | *** part/#asterisk ghost75 (~trechber@dslb-178-002-149-112.pools.arcor-ip.net) |
02:06.27 | WIMPy | But what is that mini Asterisk connected to? |
02:06.32 | WIMPy | And how? |
02:07.40 | mathi | WIMPy, PSTN through that Belgacom TWIN product that I showed you. But I don't know more details, as you can see with all my questions I'm still a beginner in telephony :( |
02:07.56 | mathi | and it's not me that has set up that system there |
02:08.14 | WIMPy | So that "modem" does have an FXO port? |
02:08.22 | mathi | WIMPy, yes |
02:08.27 | WIMPy | Silly. |
02:08.31 | mathi | why ? |
02:09.05 | WIMPy | Too expensive and too unreliable. If you have an S0 port, use that. Don;t convert it to POTS inbetween. |
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02:09.50 | mathi | WIMPy, you mean that it hould be a digital card instead of an analog card ? |
02:09.56 | Penguin | You have a regular phone line going into the building to your device where asterisk lives? |
02:10.10 | WIMPy | Is that "modem" connected to the other POTS port? Or will it answer after a long time oder is it switched on and of or how does that work? |
02:10.14 | mathi | Penguin, yes, they have, in the cellar |
02:10.18 | WIMPy | Sure! |
02:10.45 | Penguin | And that device where the phone line connects, that has Ethernet also? |
02:10.55 | WIMPy | Penguin: No they have a BRI, but it looks like someone connected Asterisk via an analog port. |
02:11.20 | mathi | Penguin, of course it's on the internet, how otherwise ? |
02:11.33 | mathi | WIMPy, if I udnerstood correctly it has two FXO |
02:11.40 | WIMPy | How otherwise what? |
02:11.40 | Penguin | I didn't see it on the internet, so I didn't know. |
02:12.03 | WIMPy | What's that thing even doing at the moment? |
02:12.14 | mathi | WIMPy, Penguin was asking if that modem was connected on the internet, but I thought that was mandatory for things to work out |
02:12.23 | WIMPy | This is very misty. |
02:12.40 | WIMPy | We don;t even know what things. |
02:12.44 | Penguin | There's a phrase that people sometimes use around here: as clear as mud. |
02:12.54 | mathi | ok, what is NOT clear? |
02:13.09 | WIMPy | The whole setup. |
02:13.13 | WIMPy | What's it like now? |
02:13.35 | WIMPy | Ok, I go the phone line part, but what is it doing? |
02:13.49 | WIMPy | What's the purpose of that "modem". |
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02:14.39 | mathi | the Belgacom TWIN is an ISDN to get two calls (if i'm not wrong), and it is connected to that mini modem with asterisk through FXO. That modem thus receives calls from PSTN, and forward to a classic phone number through a VOIP provider |
02:15.09 | mathi | so it receives calls from PSTN, and redirects the call to some phone number |
02:15.19 | WIMPy | Why do you forward calls from that line via VOIP elsewhere? |
02:15.23 | mathi | it forwards to an external secretary of the doctor |
02:15.33 | mathi | because that's how this doctor's business work |
02:15.39 | mathi | the secretary is in another city |
02:15.43 | WIMPy | Why isn't it done in the CO? |
02:15.46 | mathi | to get the appointments |
02:15.51 | mathi | what is CO ? |
02:15.56 | WIMPy | Central office |
02:16.04 | mathi | WIMPy, doctor office you mean? |
02:16.21 | WIMPy | No the telcos switch. |
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02:17.08 | mathi | I don't know |
02:17.08 | WIMPy | Is that the service we discusses several months ago? |
02:17.48 | mathi | WIMPy, well, that's what they have right now, now I want to add a feature to their system, which allow them to redirect wherever they want, and to witch on/off a powerful responder that allow patients to make appointments |
02:17.49 | WIMPy | Where you tried to find out how to get calls from different doctors and needed to know both the caller and the doctor that was called? |
02:18.54 | WIMPy | Then get some box with the appropriate interface. |
02:18.59 | mathi | WIMPy, well things are evloving only slowly. maybe we were then talking about a ip-centrex but I didn't go on into this idea |
02:19.27 | WIMPy | No it was just about standard PSTN services then. |
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02:19.58 | mathi | WIMPy, so you are telling me to replace the doctor's PBX by my bigger server ? |
02:20.08 | WIMPy | What PBX? |
02:20.17 | mathi | their mini modem with asterisk |
02:20.21 | WIMPy | But yes, you could do that as well. |
02:20.21 | mathi | it's a PBX after all |
02:20.42 | WIMPy | I wouldn't call it a PBX. |
02:20.44 | mathi | WIMPy, you don't like my idea to add a server :( |
02:21.07 | WIMPy | No. I don't like the current setup. It's just bad. |
02:21.17 | mathi | you see, I do'nt want to touch their mess, i'm not a telephony guy like you. I'm a programmer |
02:21.32 | mathi | and every doctor has a different setup |
02:21.53 | mathi | I could just come with my server, add it to the network, and job is done |
02:22.45 | mathi | if the existing server can communicate with my server with that peer thing, I don't see the problem with that setup |
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02:23.18 | WIMPy | There are always issues when connecting to POTS. |
02:24.39 | mathi | WIMPy, my server won't connect to the POTS, it will connect through SIP with their existing server. But you know what ? I tried this before, and I had a problem when my IVR was asking a patient to enter their phone number. The line would just cut after the patient entered 5 or 6 numbers, 1 time out of 10 ... maybe that's what you're talking about |
02:24.57 | mathi | and I never found a solution to that |
02:25.06 | mathi | and that is the reason why it takes months |
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02:28.23 | WIMPy | That might be due to that "modem". Hard to say. |
02:28.49 | WIMPy | But you always have issues like being called by a dialtone. |
02:29.25 | mathi | WIMPy, it is due to that modem. I was watching network traffic and it sends me a BYE packet for some obscure reasons when a user types some numbers |
02:29.43 | mathi | WIMPy, being called by a dialtone ? |
02:29.56 | WIMPy | Then Id don't understand the whole idea. |
02:30.20 | WIMPy | You prefer to use a device of which you know it makes your mission impossible? |
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02:31.02 | WIMPy | When someone calls and hangs up short before that "modem" decides to answe and forward the call, you will be called by a dial tone. |
02:31.38 | WIMPy | There's nothing you can do about that. And it's not that unlikely to happen. |
02:31.58 | mathi | WIMPy, when you say there is always problem when connecting to POTS, youa re talking about FXO ? with S0 all these problems disappear , |
02:32.07 | mathi | ? |
02:32.11 | WIMPy | Indeed. |
02:32.59 | WIMPy | You just can't connect to a call that has ended and get a free line instead. |
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02:38.58 | WIMPy | The only easy way is probably the one we talked about way back then. |
02:39.20 | WIMPy | Don;t put any hardware locally and just use the telcos features. |
02:40.08 | WIMPy | If you have to configure the customers PBX things can easily become exhausting. |
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02:41.37 | WIMPy | And that's the only way you won;t have a limit on the number of callers. |
02:41.49 | WIMPy | While still keeping their phone line free. |
02:42.37 | mathi | WIMPy, i don't understand it:( no hardware locally... you mean to host my server? |
02:43.09 | WIMPy | For example. |
02:43.28 | WIMPy | Or put it in to your garage. |
02:43.35 | WIMPy | Or where the agents are. |
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02:43.59 | mathi | WIMPy, but then they will have to pay any incoming call, the redirection from their number to mine |
02:44.25 | WIMPy | Yes, but they can still use their phone. |
02:44.40 | WIMPy | And you can receive calls while they're on the phone. |
02:44.49 | mathi | WIMPy, how is that possible? |
02:44.57 | WIMPy | So it might make sense anyway. |
02:45.04 | WIMPy | By letting the telco do it. |
02:45.22 | mathi | WIMPy, and how is this service called ? |
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02:45.40 | WIMPy | Call Forwarding. |
02:45.54 | WIMPy | Or Call Deflection if you want to do it on a per call basis. |
02:51.45 | mathi | WIMPy, a centralized asterisk server has many disadvantages: if it fails (hardware, internet connection failure, ...) all the doctors will lose patient calls, and it's not acceptable for their business. per consequent I will have to keep a very close eye on this server with maintenance and control. I will sleep better at night if the hardware is at each doctor's place |
02:52.29 | WIMPy | You think it's easier to keep an eye on may servers? |
02:52.51 | WIMPy | You need to send the calls somewhere else anyway/ |
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02:53.25 | WIMPy | Are the appointments made fully automatic or are agents involved? |
02:55.30 | mathi | WIMPy, well, if the responder is on, t's automatic. if not, the call is forwarded with an SIP provider. And ehre again a problem if centralized: I will have to pay all the communications and know how much each doctor has called, and manage the invoices and blah blah blah |
02:56.49 | WIMPy | How is that different then? |
02:59.10 | WIMPy | Anyway it's the only easy solution. Otherwise you will always have to adapt to the situation on site. |
03:02.53 | mathi | WIMPy, because i they have the hardware lcoally I cna just use their SIP account in the dialplan, and they would pay the invoice themselves. If the system is centralized, I get into the problem: which doctor needs to pay how much |
03:03.32 | WIMPy | They would still pay themselves. |
03:03.54 | WIMPy | Unless they first forward to your IVR and then you forward further to a call center. |
03:05.06 | WIMPy | needs some sleep |
03:05.07 | WIMPy | CUL |
03:06.02 | mathi | WIMPy, my IVR needs to check if responder is on or off, if off, it needs to forward; so you see, it is unfortunately my server that forwards |
03:06.20 | mathi | (they have a web itnerface that activates or not the responder, they can even chose where to redirect the calls) |
03:06.26 | mathi | (if responder is off) |
03:07.02 | mathi | and my server is checking what to do, and forwarding further to e.g. the call center, if that's where the doctor decided to forward the calls |
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03:09.19 | mathi | gn:-) |
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03:11.36 | SeRi | Penguin: you use Bria? |
03:11.43 | Penguin | I have, but I don't. |
03:13.12 | SeRi | Penguin: what do you use on the ios plataform? |
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03:15.26 | Penguin | I can't even remember right now. It's been so long since I used any iDevice. |
03:15.57 | Penguin | I'll grab my iPod and see what I was using on there. |
03:16.05 | SeRi | Thanks |
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03:21.16 | Penguin | iSip |
03:22.12 | Penguin | It seems like I was using something different on my iPhone 3G, but I'd have to charge it up and see what I was using there. |
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03:24.57 | SeRi | I see |
03:26.00 | Penguin | I must have liked iSip best, since that is what I used on my iPod touch. |
03:27.39 | SeRi | ok cool. |
03:27.43 | SeRi | I will try that |
03:28.05 | SeRi | I was testing earlier to see if I would get my session killed but all was ok. |
03:28.09 | SeRi | after a few changes. |
03:28.18 | SeRi | I was connected to your conf for 4hrs... lol |
03:28.33 | Penguin | I saw that. |
03:31.54 | SeRi | I hope that it was a small issues. I dont want this to blow up more.... |
03:33.33 | Penguin | 2012-12-23 11:25:49 - 2012-12-23 15:27:29; 14500 seconds |
03:33.52 | Penguin | I guess that's UTC. |
03:34.06 | Penguin | Wasn't it last night when you were on? |
03:34.10 | Penguin | Or was that today? |
03:35.38 | SeRi | today |
03:35.42 | Penguin | I guess that CST. |
03:36.01 | SeRi | starting like at 11AM to 3PM |
03:36.02 | Penguin | I thought I saw you on there last night when I went to bed. |
03:36.18 | Penguin | Oh! |
03:36.27 | Penguin | I know why I thought that. I took a nap today! |
03:36.39 | SeRi | :) |
03:36.56 | Penguin | I confused bed in the day time with bed in the night time. |
03:37.14 | Penguin | siesta |
03:38.04 | SeRi | hehe |
03:41.10 | rkeene | Anyone used the Bridge channel driver ? |
03:41.45 | rkeene | I'm having a hard time finding documentation for it |
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04:10.10 | SeRi | Penguin: I have a project in mind that I need help with. You think you can give me a hand? |
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04:27.24 | rkeene | Or the ConfBridge application, which seems entirely broken.. |
04:28.20 | rkeene | [Dec 23 22:27:59] ERROR[23151][C-00000077]: app_confbridge.c:1028 join_conference_bridge: Conference bridge '1000' could not be created. Is there any way to see WHY it could not be created ? |
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04:32.53 | coreyf1513 | rkeene: what version of asterisk? |
04:33.27 | rkeene | 11.1.0 |
04:33.47 | rkeene | [Dec 23 22:32:01] DEBUG[23219][C-00000004]: bridging.c:512 ast_bridge_new: Bridge technology softmix failed to setup bridge structure 0x7f9ce80010b8 |
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04:38.04 | coreyf1513 | rkeene: do you have a working timing source? |
04:38.24 | rkeene | My system clock. |
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04:39.21 | coreyf1513 | rkeene: from asterisk cli run: timing test |
04:39.40 | rkeene | > timing test |
04:39.40 | rkeene | Attempting to test a timer with 50 ticks per second. |
04:39.40 | rkeene | Failed to open timing fd |
04:39.40 | rkeene | Command 'timing test' failed. |
04:41.06 | coreyf1513 | rkeene: for confbridge that test needs to succeed.. usually supplied by res_timing_pthread, res_timing_timerfd or res_timing_dahdi |
04:41.41 | rkeene | Thanks |
04:44.22 | rkeene | :-) |
04:48.53 | Penguin | seri: probably |
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04:58.07 | SeRi | awesome |
04:59.20 | SeRi | Penguin: I want to be able to send sms via the gateway and or asterisk |
04:59.29 | SeRi | gsm* |
05:00.39 | SeRi | Penguin: http://www.portech.com.tw/p3-product1_1.asp?Pid=13 |
05:01.45 | SeRi | Penguin: http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk |
05:02.05 | SeRi | There is a sample php script at the end of the page but does not work very well. |
05:16.21 | Penguin | What are you trying to do? |
05:35.01 | SeRi | Penguin: I am trying to send alerts via sms |
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05:41.23 | Penguin | How are they generated? |
05:42.01 | rkeene | Anyone want to call me via SIP to verify my basic configuration ? sip:rkeene@oc9.org |
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05:55.26 | SeRi | they can be generated via php command |
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06:10.37 | Penguin | How are they initiated? |
06:16.11 | SeRi | well haven got that far. I want to use it with nagios |
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09:54.51 | mathi | hi |
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12:49.57 | zamba | what are my options when trying to get analogue faxes working with asterisk? i will be running an ATA and my connection to my telco is over isdn |
12:55.05 | WIMPy | On a LAN you could try as audio. New Asterisks can do T.38 gateway as well, or so I read. |
12:56.41 | zamba | this is over LAN, yeah |
12:56.52 | zamba | but what do you mean try as audio? |
12:57.06 | WIMPy | Without T.38 |
12:57.20 | WIMPy | Much easier. |
12:57.28 | zamba | the fax machine has to be connected to something, right? |
12:57.37 | WIMPy | But even on a LAN it's not guaranteed to work. |
12:57.46 | WIMPy | You said an ATA. |
12:57.50 | zamba | yeah |
12:58.08 | WIMPy | Any reason you don't connect it directly? |
12:58.21 | zamba | to asterisk? |
12:58.25 | zamba | i have no interface cards for that |
12:58.44 | WIMPy | No, to the line with an adapter. |
12:59.03 | zamba | what line? |
12:59.03 | WIMPy | Or do you have a ptp line? |
12:59.13 | zamba | i believe i have a ptp line, yeah |
12:59.17 | WIMPy | The ISDN line. |
12:59.29 | WIMPy | Bad luck then. |
13:00.07 | zamba | how do i check what i have? |
13:00.22 | WIMPy | Look at our contract. |
13:00.32 | WIMPy | Or at what you configured. |
13:00.51 | zamba | i have configured ptp |
13:01.00 | zamba | but i'm not sure what i *have* |
13:01.08 | zamba | isdn isn't really a known technology |
13:01.13 | zamba | for me |
13:01.40 | WIMPy | ptp will work on ptmp lines as well. |
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13:01.59 | zamba | is there a way i can test if ptmp is what i have? |
13:02.07 | zamba | if that's possible? |
13:02.10 | WIMPy | But if you configure ptmp and can't get a TEI, you know it's ptp. |
13:02.14 | zamba | TEI? |
13:02.32 | WIMPy | Terminal Endpoint Identifier. |
13:02.35 | zamba | that wasn't in your glossary :) |
13:02.44 | WIMPy | Hmm. |
13:03.04 | zamba | oh, yeah, there it is |
13:03.22 | WIMPy | Yes. I also found it. |
13:03.29 | WIMPy | :-0 |
13:03.32 | WIMPy | ) |
13:03.46 | WIMPy | in the bus? on the bus. |
13:03.55 | zamba | ERROR Error in /usr/local/lcr/interface.conf (line 216): unknown parameter: 'ptmp'. |
13:03.57 | zamba | ERROR No interfaces specified or failed to parse interface.conf. |
13:03.59 | zamba | No interfaces specified or failed to parse interface.conf. |
13:04.32 | zamba | http://pastie.org/5572034 |
13:04.32 | WIMPy | Just remove the ptp. ptmp is default. |
13:04.37 | zamba | that's the output from lcr query |
13:04.37 | WIMPy | Or comment it. |
13:05.10 | zamba | not able to dial now |
13:05.37 | WIMPy | Ok, so now you know it is ptp only. |
13:05.39 | zamba | if i changed back to ptp, the call came through |
13:06.08 | WIMPy | So you can't put the fax on to an adapter in parallel to Asterisk. |
13:06.23 | zamba | http://pastie.org/5572037 |
13:06.35 | zamba | that's the interface.conf, btw |
13:06.48 | zamba | a bit uncertain about portnum here? |
13:06.52 | WIMPy | Didn't you have two lines? |
13:07.02 | zamba | yeah, exactly :) |
13:07.13 | zamba | portnum 1, then? |
13:07.27 | WIMPy | That would be the 2nd port. |
13:07.37 | WIMPy | You can also use names if you want. |
13:07.37 | zamba | yeah, just add that? |
13:07.54 | WIMPy | But that's only really interesting for hotpluggable things like USB. |
13:08.04 | zamba | http://pastie.org/5572044 |
13:08.07 | zamba | change it to that? |
13:08.34 | WIMPy | IIRC that't the way, yes. |
13:08.55 | zamba | ok, cool |
13:10.01 | zamba | so, since we can't place the ATA in parallel with asterisk, we have to go through it, right? |
13:10.06 | zamba | and then we have to use T.38? |
13:10.10 | WIMPy | yes |
13:10.35 | WIMPy | No, you don't have to, but on the IP side it's supposed to be more reliable. |
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13:10.52 | zamba | i want the simplest and most reliable solution, so let's go for that, then |
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13:11.07 | WIMPy | Or you could have gone for a 4-port card and connect it that way. |
13:11.20 | WIMPy | I have never tried T.38. |
13:11.44 | zamba | too late.. already bought two of those single port ISDN cards :) |
13:11.46 | WIMPy | In fact I haven't tried fax at all in many years. More than I've been using Asterisk. |
13:11.48 | zamba | which seems to be working just fine |
13:12.03 | WIMPy | You could use a 3rd :-) |
13:12.11 | zamba | not enough room in server, actually :) |
13:12.24 | WIMPy | Or you go for USB. |
13:12.28 | zamba | yeah |
13:12.38 | zamba | or just get rid of fax |
13:12.44 | WIMPy | Not to my liking, but I read from people using it on their servers. |
13:12.53 | WIMPy | That's a good idea :-) |
13:13.03 | WIMPy | Or let Asterisk do the faxing. |
13:15.03 | zamba | yeah, but i need some howtos for that :) |
13:15.11 | zamba | btw.. late/early media.. how can i check that it's turned off? |
13:17.49 | WIMPy | You can set "earlyb no" but defaults should be just fine. |
13:18.28 | zamba | where do i set that? |
13:18.38 | WIMPy | under the interface. |
13:18.58 | WIMPy | But you shouldn't need to configure anything more than what you have. |
13:19.04 | zamba | ok, good |
13:19.33 | WIMPy | You might want to use screen-in if you want formatted numbers in CALLERID(num). |
13:19.51 | WIMPy | Otherwise you get them the way they are sent from your telco. |
13:24.15 | zamba | screen-in? |
13:24.29 | zamba | i should try setting callerid here |
13:24.32 | zamba | see if i can change that |
13:25.45 | zamba | i see that i get the numbers prefixed with '0' first |
13:26.18 | WIMPy | That's unusual. |
13:26.32 | WIMPy | Normally the 0s aren't transmitted. |
13:26.58 | WIMPy | That's where screnn-in comes handy to add them depending on the number type. |
13:27.43 | zamba | doesn't look like portnum 1 worked |
13:27.52 | zamba | wasn't able to dial out anymore when i had that defined |
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13:28.35 | WIMPy | Then the 2nd port doesn't seem to work. |
13:29.08 | zamba | oh.. seems like i had to do portnum 0, ptp and then portnum 1, ptp |
13:29.37 | zamba | guess portnum 0 tried to use ptmp then |
13:29.38 | WIMPy | ah |
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13:30.06 | WIMPy | Interestimg. So you can even mix different types of line on one logical interface. |
13:30.21 | zamba | looks like it |
13:30.57 | zamba | what's the "correct" way of setting callerid in asterisk these days? |
13:31.05 | zamba | i'm just going to try changing it |
13:31.19 | WIMPy | Set(CALLERID(num)= |
13:32.17 | zamba | which worked :) |
13:32.44 | zamba | lovely |
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13:58.05 | mathi | WIMPy, hey! I was thinking about our conversation yesterday. If I do call forwarding... won't I lose the caller ID ??? |
13:58.31 | jmetro | You can set your own callerid in dialplan |
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14:02.58 | mathi | jmetro, I mean to get the number of the person that called |
14:03.50 | WIMPy | mathi: no |
14:05.46 | mathi | WIMPy, I found another option for my problem. I can do call forwarding on a number from an SIP provider, then get the call by SIP. then I don't even need to connect my server to theirs, or to change their system |
14:06.45 | mathi | WIMPy, for example a SIP provider will give me a belgian landline number, and I jsut need the number of the doctor to be forwarded to the number of that SIP provider |
14:06.45 | WIMPy | If they use one. |
14:06.55 | mathi | WIMPy, I will give them one |
14:07.13 | mathi | or they cna buy one themselve |
14:07.55 | WIMPy | But then they would need to forward all calls to their lindline they want to get. |
14:07.59 | mathi | WIMPy, look 1 euro per month!! http://www.ovh.co.uk/VoIP/packages/online_sip_individual.xml |
14:08.36 | mathi | WIMPy, I didn't understand your last comment |
14:09.18 | WIMPy | Hmm. free calls to 40 countries for £0.99/month? Sounds too good to be true. |
14:10.19 | mathi | WIMPy, well, i won't be able to do outgoing, because it's only 1 simultaenous call. right ? So I can't get an incoming call, then dial a mobile number or landline number |
14:10.43 | mathi | WIMPy, to have two simultaenous calls I need this offer: http://www.ovh.co.uk/VoIP/packages/online_sip_enterprise.xml |
14:12.31 | WIMPy | Would seem like two of the former ones would be a much better deal. |
14:12.40 | WIMPy | I still wonder where the catch is. |
14:13.22 | mathi | WIMPy, but can I use one account to do ingoing and another account to do outgoing ? |
14:13.36 | WIMPy | Why not? |
14:14.17 | mathi | WIMPy, but then if I do outgoing with another account, the people will see a different number |
14:14.28 | jmetro | you can set caller id |
14:14.50 | WIMPy | That depends on the provider. |
14:14.53 | mathi | jmetro, really?? but then I can spoof any number? sounds strange |
14:15.16 | WIMPy | Nothing too special, really. |
14:15.29 | jmetro | my coworker frequently prank calls members of our office with the caller ID of our boss |
14:15.37 | mathi | LOL |
14:15.43 | WIMPy | That service has been available in the PSTN for a ling time as well. |
14:16.12 | mathi | WIMPy, hy does it depends on the provider? it depends only on the caller ID set in the dialplan, no? |
14:16.31 | WIMPy | Not all providers will accept what you send. |
14:17.34 | leifmadsen | many providers will not pass on whatever you set -- the smartest ones will only let you get the callerID of numbers that are known to exist and be owned by your own account |
14:21.10 | mathi | so using one account for ingoing and another for outgoing might not be the best idea then, if I want to show only one number for ingoing and outgoing |
14:21.52 | WIMPy | But you use it for different purposes, don't you? So why does it matter? |
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14:31.10 | _Corey_ | Anyone seen this? http://spectrum.ieee.org/computing/embedded-systems/cisco-ip-phones-vulnerable |
14:31.22 | leifmadsen | I have now :) |
14:32.20 | _Corey_ | Apparently the hack can be performed on the wire or using the phone's AUX port... The guy basically turns it into a microphone without the phone showing any evidence thereof |
14:32.32 | leifmadsen | ya just reading now |
14:32.48 | leifmadsen | we might start to roll out some cisco phones over the next couple of months -- will have to verify that they have the fixes in them |
14:33.04 | _Corey_ | seems like all the 79xx-type stuff is vulnerable |
14:33.16 | mathi | WIMPy, indeed it doesn't matter for my specific case :) |
14:35.29 | leifmadsen | _Corey_: sounds right :) |
14:37.48 | [sr] | _Corey_: nice, buy cisco buy... cisco is good.. (cof) |
14:38.19 | WIMPy | Do you really think that's a bug? |
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14:45.17 | mathi | if a phone is registered on two servers, and an extension is dialed, both servers will try to match the extension ? how does that work ? |
14:53.07 | WIMPy | not |
14:59.06 | mathi | WIMPy, a phone can only be registered to one server ?? |
14:59.26 | WIMPy | Depends on the phone. |
14:59.53 | WIMPy | But you will only dial out to one of the configured servers. |
15:00.45 | mathi | WIMPy, if the phone is registered on server A and server B, and an extension is dialed, how wil the phone chose where to send the request ? server A/server B |
15:01.07 | WIMPy | By the user selection an account. |
15:01.27 | jmetro | those are line separate lines |
15:01.35 | jmetro | like separate* |
15:01.46 | WIMPy | The same way you'd select the outgoing number on an ordinary phone. |
15:02.03 | WIMPy | Or the same concept rather. |
15:02.59 | mathi | WIMPy, are there phones which allow easily to select a specific server (friendly labeled) ? |
15:03.12 | WIMPy | Sure. |
15:03.22 | mathi | WIMPy, cool:) then it is not a problem |
15:03.26 | WIMPy | I'd say most of them. |
15:03.43 | mathi | WIMPy, SIP phones ? |
15:03.53 | WIMPy | yes |
15:04.45 | mathi | WIMPy, can you recommend one? |
15:05.02 | WIMPy | not really. |
15:05.19 | WIMPy | I find the Snom 300 series least bad. |
15:05.26 | mathi | where can I buy the WIMPy phones? |
15:05.36 | Sicelo | lol |
15:06.37 | WIMPy | I seriousely thought about bulding a phone some time back. But it just doesn't make sense. |
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15:13.46 | mathi | WIMPy, omg it's > 100 ? I thought a phone was sold 15 :p |
15:14.37 | WIMPy | SIP phones are in the range of proprietary PBX phones. |
15:15.21 | WIMPy | Plus the running costs for electricity. |
15:17.14 | jmetro | sip phones are nicer |
15:17.34 | WIMPy | In what way? |
15:18.19 | WIMPy | They are surely nicer than rotary dial antique phones. But that's about it. |
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15:21.19 | gnubien | want to use voip ATA when pc is poweroff; should i buy a cheap ethernet switch or router? |
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15:35.05 | mathi | WIMPy, doesnt the Snom 300 do too many for my needs ? |
15:36.26 | WIMPy | Do you wan to use it to replace whatever is at the customers office? |
15:38.01 | SupaYoshi | I think all IP phones are ugly as hell |
15:38.09 | SupaYoshi | They need a good designer :P |
15:38.14 | SupaYoshi | Too bad apple doesnt make IP phones x |
15:38.26 | jmetro | if they did wed be paying 800$ for a desk phone |
15:38.29 | WIMPy | *cough* |
15:38.39 | jmetro | and couldnt provision them |
15:39.00 | WIMPy | And they would probably only support ilbc or something. |
15:39.08 | SupaYoshi | :P |
15:39.13 | SupaYoshi | mwah atleast they would look pretty |
15:39.25 | jmetro | the front of the phone would be glass.. |
15:39.25 | SupaYoshi | i mean I like these phones from philips :P but theyre not VOIP |
15:40.57 | SupaYoshi | http://www.philips.nl/c/telefoons/design-collection-slim-line-zwart-id5551b_22/prd/nl/ |
15:41.39 | jmetro | http://www.e-netsource.com/images/SNOM-821-IP-VoIP-Phone-with-Expansion-Module-Key-Panel-Black-e-NETSource.jpg?321 |
15:41.54 | SupaYoshi | o.o nice |
15:42.02 | WIMPy | prefers desktop phones. |
15:42.26 | jmetro | and the snom 800 is IP |
15:42.36 | WIMPy | And I really dislike the upstanding ones. Who cam up with that idea? |
15:42.44 | jmetro | they are quite tall |
15:42.56 | jmetro | you can reformat the base to make it lower but you cant see the pretty graphics then |
15:43.38 | WIMPy | A tiltable display is a must. |
15:45.26 | jmetro | like the snom 340? |
15:45.48 | WIMPy | 340? |
15:47.49 | jmetro | 60* |
15:48.12 | WIMPy | 320, 360 and 370 are all tiltable, yes. |
15:48.58 | WIMPy | Not having that is the most obvious drwback of my beloved Eurit 40. |
15:49.22 | jmetro | thats an odd looking phone. i like the keyboard though |
15:49.53 | WIMPy | It's an extremely handy phone. |
15:50.38 | WIMPy | Even though it looks important it doesn't have that may features. But it does exactely what a phone should do. |
15:50.50 | WIMPy | Very easy handling of multiple calls. |
15:52.14 | gnubien | are Grandstream voip phones mostly junk? |
15:52.36 | WIMPy | Some people say so. |
15:52.41 | WIMPy | ~gs |
15:52.41 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
15:52.46 | WIMPy | ~grandstream |
15:52.46 | infobot | from memory, grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
15:53.30 | gnubien | what brand of ATA will has the best quality for the lowest price? |
15:54.29 | WIMPy | First you have to pick one: best quality or lowest price? |
15:55.33 | gnubien | WIMPy: cisco spa2102 good quality? |
15:56.06 | WIMPy | I have no idea. Never used an ATA. |
15:56.08 | gnubien | think lynksys sold to cisco, or visa versa |
15:56.24 | WIMPy | But I'd look out for stuff like a symmetrical line. |
15:56.38 | WIMPy | Cisco bought Linksys. |
15:56.53 | gnubien | ok, will google symmetrical line |
16:00.28 | jmetro | i think all lines are symmetrical |
16:00.51 | Maliuta | WIMPy: current rumor is that cisco hasd engaged Lyods to find a buyer for linksys |
16:01.09 | WIMPy | jmetro: Unlikely as that's more expensive. |
16:01.48 | jmetro | </geometry humor> |
16:02.07 | WIMPy | oh |
16:02.23 | gnubien | are ATA's with built in routers better or worse than using a ATA with a ethernet switch or router? |
16:02.47 | WIMPy | That obviousely depends on where you want to use them. |
16:03.22 | gnubien | want to use the ATA when pc is power off so just ISP >> cable modem >> eth on pc now |
16:04.02 | WIMPy | Does the modem include a router? |
16:04.18 | gnubien | no, just a cable modem, no router in the modem |
16:04.37 | WIMPy | Then you probably need a router somewhere. |
16:04.57 | gnubien | would a ethernet switch be adequate? |
16:05.11 | WIMPy | Unless you ISP is as braindead as KDG and allows multiple public IPs. |
16:05.16 | WIMPy | No |
16:05.30 | WIMPy | Well, probably no. |
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17:34.19 | SupaYoshi | hi |
17:34.29 | SupaYoshi | the command cdr status doesn't work from me in ASTERISK CLI |
17:34.46 | SupaYoshi | is there anything im doing wrong? I read some guides, and you should be able to type that, according to these. |
17:36.59 | SupaYoshi | nvm |
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17:43.09 | SupaYoshi | my asterisk keeps doing this |
17:43.10 | SupaYoshi | doing dnsmgr_lookup for 'sip.voipbuster.com' |
17:44.33 | [TK]D-Fender | SupaYoshi: It's a notice of general activity. It isn't saying anything is wrong. |
17:46.24 | SupaYoshi | true its annoying though |
17:46.32 | SupaYoshi | should i just type the ip in instead of sip.voipbuster.com :P |
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17:47.28 | SupaYoshi | MY CDR is enabled btw, but the logging is simple, would that mean that it logs to mysql too? |
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17:52.36 | [TK]D-Fender | Do you see the backend being loaded? |
17:52.43 | [TK]D-Fender | So you see records being added? |
17:52.57 | [TK]D-Fender | Have you confirmed the table structure? |
17:53.08 | [TK]D-Fender | Have you tested the auth you provided for * to do so? |
17:55.01 | SupaYoshi | Mhm i see records being added in the CSV Master file |
17:55.09 | SupaYoshi | I also see records being logged in the database |
17:55.25 | SupaYoshi | I checkec the passwords for CRD_ |
17:55.38 | SupaYoshi | for cdr_mysql.conf and freepbx.conf |
17:55.41 | SupaYoshi | they match. |
17:56.00 | SupaYoshi | Also the calls are logged, but in my browser i cant find any calls, when i export to CSV, the file is empty |
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18:04.12 | SupaYoshi | I can see the table cdr in mysql |
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18:06.23 | [TK]D-Fender | SupaYoshi: Asterisk has nothing to do with "your browser" |
18:07.30 | carrar | wait, Asterisk doesn't work in EI? |
18:07.33 | carrar | @#($@~!#$ |
18:07.38 | carrar | err IE |
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18:10.58 | Penguin | <SupaYoshi> Oh i thought maybe in verbose 10 the answer would be different |
18:11.00 | Penguin | <Penguin> It won't get more verbose above 3. |
18:11.01 | Penguin | <Penguin> Verbose 4 adds the dnsmngr stuff, I think it's verbose 7 where CDR is added... |
18:11.06 | Penguin | supayoshi: Remember this? |
18:11.33 | Penguin | If you are annoyed by the dnsmgr lookups, reduce your verbose level BELOW 4. |
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18:30.16 | pabelanger | Pretty sure I patched that a while ago, cannot remember the verbose level I changed it too |
18:37.16 | moos3 | I just installed asterisk 11 my sip clients have no audio ideas ? |
18:46.04 | Penguin | You think you changed the dnsmgr lookups to a higher verbose level? |
18:47.17 | pabelanger | Don't think, I did. Change it to 6 |
18:47.18 | pabelanger | http://svnview.digium.com/svn/asterisk/branches/1.8/main/dnsmgr.c?r1=353371&r2=360471 |
18:47.36 | pabelanger | http://svnview.digium.com/svn/asterisk?view=revision&revision=360471 |
18:48.44 | pabelanger | 1.8.10.0+ has it |
18:56.44 | Penguin | And that carries into 10 and 11 as well? |
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19:03.49 | pabelanger | 10 yes |
19:03.58 | pabelanger | 11+ I changed it to a debug message I think |
19:05.26 | Penguin | I don't know what version he's running, but if it's 11, then he shouldn't complain about too much crap if he has debug turned on. That's kind of what debug does, after all. |
19:16.44 | jmetro | sip set debug off |
19:16.47 | jmetro | core set verbose 0 |
19:16.50 | jmetro | core set debug 0 |
19:21.51 | Penguin | core set debug off |
19:21.56 | Penguin | core set verbose off |
19:25.31 | carrar | core reboot now |
19:27.20 | jmetro | core stop destructively |
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19:42.51 | ChannelZ | core stop from melting |
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19:46.21 | Molo | leifmadsen: you there? |
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20:24.10 | Penguin | seri: Siphon on the iPhone 3G |
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22:03.39 | SeRi | Penguin: Thanks and Merry Christmass to all here @ #asterisk |
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22:04.55 | gusto | heh |
22:05.18 | gusto | i did set up my first TLS connection today |
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22:20.36 | rkeene | Hmm... Grr... nf_conntrack_sip seems to be broken with SIP over TCP |
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22:37.34 | zopsi | Does anyone from Digium monitor this channel? |
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22:38.25 | dym | merry christmas, everyone! |
22:39.06 | Penguin | Mary Crispness |
22:39.59 | file | zopsi, there are Digium people here at times yes |
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22:49.59 | zopsi | @file, anyone that works with DPMA? Having my final issue to fix before christmas and I'm assuming Digium tech support is off for the holidays. |
22:50.37 | file | not at this moment, and tech support may still be there |
22:50.47 | WIMPy | Before chrismas? Here it's already after christmas. |
22:51.36 | zopsi | @file, Doubtful. My support ticket hasn't been answered since its initial comment on Saturday. |
22:51.45 | zopsi | @WIMPy where are you from? |
22:51.53 | file | I was talking to people in technical support earlier, there just isn't many |
22:51.59 | WIMPy | de |
22:52.37 | zopsi | @file, yeah completely understandable. |
22:52.50 | zopsi | @WIMPy, in that case Merry Christmas! |
22:52.59 | dym | WIMPy doesnt celebrate christmas. |
22:53.10 | zopsi | great... |
22:54.07 | zopsi | Different religion or just scrooge? |
22:54.51 | WIMPy | I don't suffer from any of those deseases. |
22:55.05 | zopsi | If anyone happens to know anything about DPMA and using the can_monitor and can_intercom functions in contacts let me know :) |
22:55.47 | zopsi | christmas in my household has very little religion involved (other than the name). |
22:56.24 | dym | it shouldnt have... |
22:56.41 | dym | christmas is a time for gathering, not religious beliefs, or anything else randomly connected. |
22:56.57 | WIMPy | It used to be a nice quiet time when you could work without disturbances. |
22:57.20 | zopsi | @dym : exactly |
22:57.41 | zopsi | @WIMPy: maybe in de |
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23:01.50 | zopsi | seriously though if anyone could help me out... I'd be very thankful. |
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