IRC log for #asterisk on 20121224

00:00.19greenwolfbeing rude is ignorant
00:00.25greenwolfi was just asking for alittle help
00:00.49greenwolfeveryone isnt perfect you kno
00:01.03jpsharpSounds like your vps provider gave you a non-standard kernel.
00:01.21greenwolfyes jpsharp i am pissed they did this
00:01.24WIMPyMost distro kernels are non-standard.
00:01.40WIMPyProbaly almost all.
00:01.42greenwolfbut other ppl are pertaining its my knowledge of things causing this
00:01.52jpsharpnon-distro-standard, then.
00:02.00greenwolfin which it isnt because obviously they couldn;t figure the problem out either
00:02.03greenwolfhuh
00:02.07WIMPyThe non-standard standard?
00:02.13greenwolflol wimpy good one
00:03.00jpsharpBewarned, 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.
00:03.11jpsharps/there/their
00:03.39greenwolfi have many friends that are voip engineers that recommended these guys to me
00:03.46greenwolfthey are really good and so i used them myself
00:03.54PenguinIf 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.
00:03.57greenwolfthey run multiple actually entire systems on their network with no problem
00:04.13WIMPyAnd 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)
00:05.06greenwolfyea 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
00:05.19PenguinOr, 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.
00:05.21greenwolfcoreyf1513: you said something about timingfd
00:06.09greenwolfyea 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
00:06.27WIMPyI told you that you don;t need dahdi, except for MeetMe. But if you can't upgrade to ConfBridge, you do need dahdi.
00:06.31greenwolfthen 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
00:06.38coreyf1513greenwolf: that only provides timing.. meetme requires dahdi besides timing...
00:06.47greenwolfoh i see..damn
00:07.22greenwolfyea my employer is requiring me to use meetme cuz we have many scripts that were coded for meetme conferences and options within those conferences
00:07.24WIMPyBut in the time you spent trying to install dahdi, you might have been able to change your dialplan from MeetMe to ConfBridge.
00:07.27greenwolfso its a must i use meetme
00:07.51greenwolftrue wimpy...true im starting to see that myself :)
00:07.52greenwolflmao
00:08.15greenwolfwell 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
00:08.28WIMPyAnd ConfBridge will scale acroll more than one CPU unlike MeetMe.
00:08.42WIMPyacross
00:09.56PenguinI would ask them to install the latest kernel and source packages for that CentOS branch.
00:10.16greenwolfyes that is exactly what im requesting them to do now penguin..thanks
00:10.38greenwolfits their fault im running into such a problem becuase they are doing something weird with their kernel sources for some reason
00:10.52jpsharpProbably patched to run properly on the VPS hose.
00:10.52PenguinThen 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.
00:10.54jpsharphost.
00:10.56greenwolfi dont understand why they keep using that non-standard kernel source and why im having such problems with it
00:11.17greenwolfyea i was also thinking of installing it from source myself
00:11.23PenguinIt seems that all VPS providers patch the kernels.
00:12.25greenwolfi see
00:12.35PenguinIf 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.
00:12.53jpsharpCause 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)
00:15.25greenwolf_sorry lost connection..yes jpsharp they are running OpenVZ
00:16.51greenwolf_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?
00:17.06jpsharpThere's your answer as to why they're using a "nonstandard" kernel.
00:17.26greenwolf_yea stupid OpenVZ
00:17.55greenwolf_yup your completely right jpsharp and penguin...bullshit if you ask me lol
00:18.31PenguinYou aren't likely to need an older dahdi version.
00:18.45greenwolf_ok
00:18.50PenguinThe problem is because of packages and package versions, not because of the dahdi version.
00:19.12jpsharpYep, you're going to need matching kernel + source to build dahdi no matter what.
00:39.56*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
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00:58.36*** join/#asterisk FireAndIce (~FireAndIc@123.201.83.42)
01:00.31WIMPywants a pickup for VM.
01:00.48PenguinWhat will it do?
01:01.04WIMPypick up a call that has gone to VM.
01:01.23Penguinchannel redirect  <---
01:01.42WIMPyYes, but it takes too long to type that.
01:01.55gustoi watched now a little of that SIP debug and i see that there is a lot of nonsense in there
01:02.39gustofor 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
01:02.55PenguinAre you sure about that?
01:04.56gustoyes
01:05.51gustowhen 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
01:06.12PenguinSounds right to me.
01:06.45PenguinCommunications between asterisk and the device are two-way, not one-way.
01:06.56gustoyes
01:08.36gustoPenguin: http://pastebin.com/zGxwueME take a look at this sample
01:09.17gusto192.168.27.49 is asterisk here
01:11.46PenguinYeah, 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.
01:11.57gustobut in that sample you only see the other devices (linksys and cisco) mixing up from: and to:
01:11.58Penguins/deive/device/
01:12.13PenguinIt's not mixing up anything.
01:12.37gusto<--- SIP read from UDP:192.168.23.19:5060 ---> To: <sip:dedko@192.168.23.19:5060>;tag=324a73616f09fa1di0 ???
01:12.46gustothat would mean that he is sending that to himself
01:13.00PenguinIt doesn't mean that.
01:13.10gustowhat does it mean then?
01:13.54PenguinIt means that asterisk has a packet that is To: 192.168.23.19, just like it says.
01:14.35gustoand why is it in between of SIP read from UDP:192.168.23.19:5060 then?
01:15.09gustowould it be asterisk's packet then he could take "Reliably Transmitting (no NAT) to 192.168.27.187:5061"
01:15.23gustoor 23.19, i mean, however
01:15.53PenguinWhy is your phone on the wrong port?
01:16.25gustoyou mean that 5061 one?
01:16.29PenguinYes
01:16.54gustobecause these devices have 2 analogue phone connectors and one is on 5060 and the second on 5061 and so on
01:17.11PenguinI see.  I have to do that, sometimes, too.
01:17.37PenguinIf they are on the same LAN as asterisk, that isn't necessary, though.
01:18.06PenguinAt least I think that's true.
01:18.40gustono, it's that device
01:18.46gustoone of them is PAP2T
01:19.05gustoServer: Linksys/PAP2T-3.1.15(LS)
01:19.19gustoyou can not have these lines both on 5060
01:19.32PenguinWhy not?
01:19.51gustobecause then that device would need 2 IP addresses
01:20.28PenguinI'm sure I have tested this theory before, but I can't remember the results.
01:20.39PenguinAnd I have a tendency to not take notes.
01:20.51gustoin theory it would work, but the firmware of this devices will not let you
01:22.09mathihow do I pass a call from one asterisk server to the other ?
01:22.15gustoof 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
01:22.21mathiWIMPy, ?
01:22.38gustomathi: SIP/asterisk2/call
01:22.39PenguinDepending on your definition of pass, you might want to use Dial().
01:22.56mathigusto, what is that?
01:23.04mathia Dial ?
01:23.07PenguinDial()   <--------
01:23.19WIMPymathi: yes
01:23.29mathiwon't I lose the original Caller ID if I dial the other server ?
01:23.42PenguinNo
01:23.59PenguinYou will only if you set a different caller id.
01:24.04PenguinSo don't do that.
01:24.05mathicaller -> server A -> server B.   I was expecting server B to have server A as Caller ID ?
01:24.29PenguinServers don't have callerids.
01:24.31WIMPyOnly if you configure a caller ID for that account on server B.
01:24.53mathinice then
01:25.07WIMPy"that" = the one Server A uses.
01:25.38mathiWIMPy, you mean, i don't have to configure a caller ID on server A
01:25.49PenguinYou shouldn't.
01:25.57WIMPyNo.
01:26.03Penguinhttp://pastebin.com/Ag7tknm2   See here.
01:26.21WIMPyYou shouldn't configure a caller ID for te account that accepts the calls.
01:26.45PenguinI recommend only setting callerid for phones.
01:27.21WIMPyexately
01:27.44mathiPenguin, can you tell me what happens in your code ?
01:27.59PenguinHuh?
01:28.25PenguinThis is sample configuration for two asterisk systems to communicate with SIP.
01:30.07mathiPenguin, well with phones, you need to register them. how does it work here ebtween two servers ?
01:30.32PenguinPhones register to one or the other, or via proxy.
01:30.49WIMPyYou don;t have to register phones.
01:31.03mathihah
01:31.05WIMPyYou have to register if the other end doesn;t know where to find you.
01:31.07PenguinIt's helpful for them to be registered to get phone calls.
01:31.07mathiI always registered them
01:31.37mathiPenguin, so in your example the servers don't register between them
01:31.59PenguinRight, they are configured statically.
01:32.50PenguinIf you need to register each one to the other, that's an easy change to my sample.
01:35.03mathithen all I need to do is pass the call from server A to server B in extensions.conf ? Dial(SIP/miami)
01:36.16WIMPyYou might want to call a specific extension on "miami".
01:36.20PenguinNo, the format for Dial is Tech/peer/extension
01:36.40WIMPyWith "/extension" being optional.
01:36.56WIMPySo mathis exaple is correct.
01:36.59mathiwell tech is SIP, peer is miami ...
01:37.24WIMPyJust probably less than he might want.
01:37.32PenguinI'm quite certain that you won't want to call asterisk itself.  You'll want to call another device that is on that system.
01:37.55WIMPyYou don't call devices. You call extensions.
01:38.03PenguinTrue story there.
01:38.36mathiPenguin, 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"
01:38.48WIMPyBut possibly knowing where the call comes from is enough.
01:38.55PenguinThose are not extensions.
01:39.18mathithey must be, context is pointing to extensiosn
01:39.20PenguinThose are contexts.
01:39.34PenguinWithin those contexts, you will put extensions.
01:39.55mathioops
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01:42.15mathiPenguin, for the servers to register between them, I should so something like this? register => 1234:pwd@...
01:43.04PenguinIf 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.
01:43.18mathiPenguin, it was just out of curiosity
01:43.43PenguinChange host= to dynamic and add a register statement to each one.
01:44.10WIMPyTo each one doesn't make too much sense.
01:44.32mathiWIMPy, it seems it does if IP's are dynmaic ?
01:44.40WIMPyNo
01:44.52PenguinWell, you'll have to know at least one host or the other can't register to it in the first place.
01:45.12mathiah, you have to register only one ?
01:45.18WIMPyIf you don't have at least a dyndns name that you could configure as host, where are you going to send the register?
01:45.38mathiright)
01:45.53mathiWIMPy, but only one server needs to register to the other then ?
01:46.01WIMPycorrect
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01:46.31WIMPyOr none, if both have at least a known name.
01:47.24mathiWIMPy, but even if both have a static IP, is it ok to register ? or is it less performant ?
01:47.43WIMPyYou can do it, but it doesn't make sense.
01:50.38mathiserver B has only one extension matching for the phones, all the other extensions should be forwarded to server A. Is this possible to achieve?
01:51.06WIMPysure
01:51.38WIMPyBut are you sure you need server A at all?
01:53.08mathiWIMPy, 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
01:54.11mathiWIMPy, what do you think?
01:54.31mathiI think I will get rid of all the hassle about these cards and all ...
01:54.49PenguinIf they already have asterisk, why do you need another asterisk?
01:54.54WIMPyDo you think SIP will be easier?
01:56.22mathiPenguin, 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, ...
01:56.33mathiWIMPy, sure, very easy
01:56.40mathiWIMPy, as long as it's local
01:57.03PenguinIs it possible to eliminate their original system completely and use only the bigger/better system?
01:57.57mathiPenguin, 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, ...
01:58.11mathiif I add my server, I can just forward extensions to original server
01:59.05mathiPenguin, the main problem is the card, their phone line, ...
01:59.23WIMPydoesn't see a problem there.
01:59.32WIMPyMore of the opposite.
01:59.44mathiWIMPy, well in a perfect world there is no problem ...
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02:01.03mathibut clearly these are my two options I think. replace their server VS adding my server...
02:01.42WIMPyAnd how is "adding your server" going to work?
02:03.12mathiWIMPy, 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
02:03.25mathiWIMPy, something wrong ?
02:03.42WIMPyBut there's no Server A, yet?
02:03.54PenguinYou basically just need to add an auto-attendant and/or an IVR, so it seems.  Is that correct?
02:03.56WIMPyAnd Server B is local?
02:04.03WIMPyI don't get that setup.
02:04.05PenguinBut the existing box cannot handle it.
02:04.21PenguinI would dump the original.
02:04.24mathiWIMPy, there is, that's the existing server at the doctor's place (in this concrete example, a small modem with asterisk) <= server A
02:04.30WIMPyYes, what do you want to add there?
02:04.33PenguinSet up the new box, configure everything, then cut over.
02:04.40mathiWIMPy, and server B is my server that I will add in the network with Ethernet
02:05.18WIMPyWhy do you want to keep a mini Asterisk if tou have a big one anyway?
02:05.27WIMPyAnd what is that mini Asterisk doing at the moment?
02:05.34mathiPenguin, 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
02:06.08mathiWIMPy, that mini asterisk server is handling incoming calls and outgoing calls, with a very basic auto-attendant
02:06.10PenguinOh, I missed that part of the puzzle.
02:06.15*** part/#asterisk ghost75 (~trechber@dslb-178-002-149-112.pools.arcor-ip.net)
02:06.27WIMPyBut what is that mini Asterisk connected to?
02:06.32WIMPyAnd how?
02:07.40mathiWIMPy, 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 :(
02:07.56mathiand it's not me that has set up that system there
02:08.14WIMPySo that "modem" does have an FXO port?
02:08.22mathiWIMPy, yes
02:08.27WIMPySilly.
02:08.31mathiwhy ?
02:09.05WIMPyToo expensive and too unreliable. If you have an S0 port, use that. Don;t convert it to POTS inbetween.
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02:09.50mathiWIMPy, you mean that it hould be a digital card instead of an analog card ?
02:09.56PenguinYou have a regular phone line going into the building to your device where asterisk lives?
02:10.10WIMPyIs 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?
02:10.14mathiPenguin, yes, they have, in the cellar
02:10.18WIMPySure!
02:10.45PenguinAnd that device where the phone line connects, that has Ethernet also?
02:10.55WIMPyPenguin: No they have a BRI, but it looks like someone connected Asterisk via an analog port.
02:11.20mathiPenguin, of course it's on the internet, how otherwise ?
02:11.33mathiWIMPy, if I udnerstood correctly it has two FXO
02:11.40WIMPyHow otherwise what?
02:11.40PenguinI didn't see it on the internet, so I didn't know.
02:12.03WIMPyWhat's that thing even doing at the moment?
02:12.14mathiWIMPy, Penguin was asking if that modem was connected on the internet, but I thought that was mandatory for things to work out
02:12.23WIMPyThis is very misty.
02:12.40WIMPyWe don;t even know what things.
02:12.44PenguinThere's a phrase that people sometimes use around here:  as clear as mud.
02:12.54mathiok, what is NOT clear?
02:13.09WIMPyThe whole setup.
02:13.13WIMPyWhat's it like now?
02:13.35WIMPyOk, I go the phone line part, but what is it doing?
02:13.49WIMPyWhat's the purpose of that "modem".
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02:14.39mathithe 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
02:15.09mathiso it receives calls from PSTN, and redirects the call to some phone number
02:15.19WIMPyWhy do you forward calls from that line via VOIP elsewhere?
02:15.23mathiit forwards to an external secretary of the doctor
02:15.33mathibecause that's how this doctor's business work
02:15.39mathithe secretary is in another city
02:15.43WIMPyWhy isn't it done in the CO?
02:15.46mathito get the appointments
02:15.51mathiwhat is CO ?
02:15.56WIMPyCentral office
02:16.04mathiWIMPy, doctor office you mean?
02:16.21WIMPyNo the telcos switch.
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02:17.08mathiI don't know
02:17.08WIMPyIs that the service we discusses several months ago?
02:17.48mathiWIMPy, 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
02:17.49WIMPyWhere 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?
02:18.54WIMPyThen get some box with the appropriate interface.
02:18.59mathiWIMPy, well things are evloving only slowly. maybe we were then talking about a ip-centrex but I didn't go on into this idea
02:19.27WIMPyNo it was just about standard PSTN services then.
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02:19.58mathiWIMPy, so you are telling me to replace the doctor's PBX by my bigger server ?
02:20.08WIMPyWhat PBX?
02:20.17mathitheir mini modem with asterisk
02:20.21WIMPyBut yes, you could do that as well.
02:20.21mathiit's a PBX after all
02:20.42WIMPyI wouldn't call it a PBX.
02:20.44mathiWIMPy, you don't like my idea to add a server :(
02:21.07WIMPyNo. I don't like the current setup. It's just bad.
02:21.17mathiyou see, I do'nt want to touch their mess, i'm not a telephony guy like you. I'm a programmer
02:21.32mathiand every doctor has a different setup
02:21.53mathiI could just come with my server, add it to the network, and job is done
02:22.45mathiif 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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02:23.18WIMPyThere are always issues when connecting to POTS.
02:24.39mathiWIMPy, 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
02:24.57mathiand I never found a solution to that
02:25.06mathiand that is the reason why it takes months
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02:28.23WIMPyThat might be due to that "modem". Hard to say.
02:28.49WIMPyBut you always have issues like being called by a dialtone.
02:29.25mathiWIMPy, 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
02:29.43mathiWIMPy, being called by a dialtone ?
02:29.56WIMPyThen Id don't understand the whole idea.
02:30.20WIMPyYou prefer to use a device of which you know it makes your mission impossible?
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02:31.02WIMPyWhen someone calls and hangs up short before that "modem" decides to answe and forward the call, you will be called by a dial tone.
02:31.38WIMPyThere's nothing you can do about that. And it's not that unlikely to happen.
02:31.58mathiWIMPy, when you say there is always problem when connecting to POTS, youa re talking about FXO ? with S0 all these problems disappear ,
02:32.07mathi?
02:32.11WIMPyIndeed.
02:32.59WIMPyYou just can't connect to a call that has ended and get a free line instead.
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02:38.58WIMPyThe only easy way is probably the one we talked about way back then.
02:39.20WIMPyDon;t put any hardware locally and just use the telcos features.
02:40.08WIMPyIf you have to configure the customers PBX things can easily become exhausting.
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02:41.37WIMPyAnd that's the only way you won;t have a limit on the number of callers.
02:41.49WIMPyWhile still keeping their phone line free.
02:42.37mathiWIMPy, i don't understand it:( no hardware locally... you mean to host my server?
02:43.09WIMPyFor example.
02:43.28WIMPyOr put it in to your garage.
02:43.35WIMPyOr where the agents are.
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02:43.59mathiWIMPy, but then they will have to pay any incoming call, the redirection from their number to mine
02:44.25WIMPyYes, but they can still use their phone.
02:44.40WIMPyAnd you can receive calls while they're on the phone.
02:44.49mathiWIMPy, how is that possible?
02:44.57WIMPySo it might make sense anyway.
02:45.04WIMPyBy letting the telco do it.
02:45.22mathiWIMPy, and how is this service called ?
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02:45.40WIMPyCall Forwarding.
02:45.54WIMPyOr Call Deflection if you want to do it on a per call basis.
02:51.45mathiWIMPy, 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
02:52.29WIMPyYou think it's easier to keep an eye on may servers?
02:52.51WIMPyYou need to send the calls somewhere else anyway/
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02:53.25WIMPyAre the appointments made fully automatic or are agents involved?
02:55.30mathiWIMPy, 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
02:56.49WIMPyHow is that different then?
02:59.10WIMPyAnyway it's the only easy solution. Otherwise you will always have to adapt to the situation on site.
03:02.53mathiWIMPy, 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
03:03.32WIMPyThey would still pay themselves.
03:03.54WIMPyUnless they first forward to your IVR and then you forward further to a call center.
03:05.06WIMPyneeds some sleep
03:05.07WIMPyCUL
03:06.02mathiWIMPy, 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
03:06.20mathi(they have a web itnerface that activates or not the responder, they can even chose where to redirect the calls)
03:06.26mathi(if responder is off)
03:07.02mathiand 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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03:09.19mathign:-)
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03:11.36SeRiPenguin: you use Bria?
03:11.43PenguinI have, but I don't.
03:13.12SeRiPenguin: what do you use on the ios plataform?
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03:15.26PenguinI can't even remember right now.  It's been so long since I used any iDevice.
03:15.57PenguinI'll grab my iPod and see what I was using on there.
03:16.05SeRiThanks
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03:21.16PenguiniSip
03:22.12PenguinIt 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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03:24.57SeRiI see
03:26.00PenguinI must have liked iSip best, since that is what I used on my iPod touch.
03:27.39SeRiok cool.
03:27.43SeRiI will try that
03:28.05SeRiI was testing earlier to see if I would get my session killed but all was ok.
03:28.09SeRiafter a few changes.
03:28.18SeRiI was connected to your conf for 4hrs... lol
03:28.33PenguinI saw that.
03:31.54SeRiI hope that it was a small issues. I dont want this to blow up more....
03:33.33Penguin2012-12-23 11:25:49 - 2012-12-23 15:27:29; 14500 seconds
03:33.52PenguinI guess that's UTC.
03:34.06PenguinWasn't it last night when you were on?
03:34.10PenguinOr was that today?
03:35.38SeRitoday
03:35.42PenguinI guess that CST.
03:36.01SeRistarting like at 11AM to 3PM
03:36.02PenguinI thought I saw you on there last night when I went to bed.
03:36.18PenguinOh!
03:36.27PenguinI know why I thought that.  I took a nap today!
03:36.39SeRi:)
03:36.56PenguinI confused bed in the day time with bed in the night time.
03:37.14Penguinsiesta
03:38.04SeRihehe
03:41.10rkeeneAnyone used the Bridge channel driver ?
03:41.45rkeeneI'm having a hard time finding documentation for it
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04:10.10SeRiPenguin: I have a project in mind that I need help with. You think you can give me a hand?
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04:27.24rkeeneOr the ConfBridge application, which seems entirely broken..
04:28.20rkeene[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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04:32.53coreyf1513rkeene: what version of asterisk?
04:33.27rkeene11.1.0
04:33.47rkeene[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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04:38.04coreyf1513rkeene: do you have a working timing source?
04:38.24rkeeneMy system clock.
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04:39.21coreyf1513rkeene: from asterisk cli run: timing test
04:39.40rkeene> timing test
04:39.40rkeeneAttempting to test a timer with 50 ticks per second.
04:39.40rkeeneFailed to open timing fd
04:39.40rkeeneCommand 'timing test' failed.
04:41.06coreyf1513rkeene: for confbridge that test needs to succeed.. usually supplied by res_timing_pthread, res_timing_timerfd or res_timing_dahdi
04:41.41rkeeneThanks
04:44.22rkeene:-)
04:48.53Penguinseri: probably
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04:58.07SeRiawesome
04:59.20SeRiPenguin: I want to be able to send sms via the gateway and or asterisk
04:59.29SeRigsm*
05:00.39SeRiPenguin: http://www.portech.com.tw/p3-product1_1.asp?Pid=13
05:01.45SeRiPenguin: http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
05:02.05SeRiThere is a sample php script at the end of the page but does not work very well.
05:16.21PenguinWhat are you trying to do?
05:35.01SeRiPenguin: I am trying to send alerts via sms
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05:41.23PenguinHow are they generated?
05:42.01rkeeneAnyone want to call me via SIP to verify my basic configuration ?  sip:rkeene@oc9.org
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05:55.26SeRithey can be generated via php command
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06:10.37PenguinHow are they initiated?
06:16.11SeRiwell haven got that far. I want to use it with nagios
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12:49.57zambawhat 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
12:55.05WIMPyOn a LAN you could try as audio. New Asterisks can do T.38 gateway as well, or so I read.
12:56.41zambathis is over LAN, yeah
12:56.52zambabut what do you mean try as audio?
12:57.06WIMPyWithout T.38
12:57.20WIMPyMuch easier.
12:57.28zambathe fax machine has to be connected to something, right?
12:57.37WIMPyBut even on a LAN it's not guaranteed to work.
12:57.46WIMPyYou said an ATA.
12:57.50zambayeah
12:58.08WIMPyAny reason you don't connect it directly?
12:58.21zambato asterisk?
12:58.25zambai have no interface cards for that
12:58.44WIMPyNo, to the line with an adapter.
12:59.03zambawhat line?
12:59.03WIMPyOr do you have a ptp line?
12:59.13zambai believe i have a ptp line, yeah
12:59.17WIMPyThe ISDN line.
12:59.29WIMPyBad luck then.
13:00.07zambahow do i check what i have?
13:00.22WIMPyLook at our contract.
13:00.32WIMPyOr at what you configured.
13:00.51zambai have configured ptp
13:01.00zambabut i'm not sure what i *have*
13:01.08zambaisdn isn't really a known technology
13:01.13zambafor me
13:01.40WIMPyptp will work on ptmp lines as well.
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13:01.59zambais there a way i can test if ptmp is what i have?
13:02.07zambaif that's possible?
13:02.10WIMPyBut if you configure ptmp and can't get a TEI, you know it's ptp.
13:02.14zambaTEI?
13:02.32WIMPyTerminal Endpoint Identifier.
13:02.35zambathat wasn't in your glossary :)
13:02.44WIMPyHmm.
13:03.04zambaoh, yeah, there it is
13:03.22WIMPyYes. I also found it.
13:03.29WIMPy:-0
13:03.32WIMPy)
13:03.46WIMPyin the bus? on the bus.
13:03.55zambaERROR Error in /usr/local/lcr/interface.conf (line 216): unknown parameter: 'ptmp'.
13:03.57zambaERROR No interfaces specified or failed to parse interface.conf.
13:03.59zambaNo interfaces specified or failed to parse interface.conf.
13:04.32zambahttp://pastie.org/5572034
13:04.32WIMPyJust remove the ptp. ptmp is default.
13:04.37zambathat's the output from lcr query
13:04.37WIMPyOr comment it.
13:05.10zambanot able to dial now
13:05.37WIMPyOk, so now you know it is ptp only.
13:05.39zambaif i changed back to ptp, the call came through
13:06.08WIMPySo you can't put the fax on to an adapter in parallel to Asterisk.
13:06.23zambahttp://pastie.org/5572037
13:06.35zambathat's the interface.conf, btw
13:06.48zambaa bit uncertain about portnum here?
13:06.52WIMPyDidn't you have two lines?
13:07.02zambayeah, exactly :)
13:07.13zambaportnum 1, then?
13:07.27WIMPyThat would be the 2nd port.
13:07.37WIMPyYou can also use names if you want.
13:07.37zambayeah, just add that?
13:07.54WIMPyBut that's only really interesting for hotpluggable things like USB.
13:08.04zambahttp://pastie.org/5572044
13:08.07zambachange it to that?
13:08.34WIMPyIIRC that't the way, yes.
13:08.55zambaok, cool
13:10.01zambaso, since we can't place the ATA in parallel with asterisk, we have to go through it, right?
13:10.06zambaand then we have to use T.38?
13:10.10WIMPyyes
13:10.35WIMPyNo, you don't have to, but on the IP side it's supposed to be more reliable.
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13:10.52zambai want the simplest and most reliable solution, so let's go for that, then
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13:11.07WIMPyOr you could have gone for a 4-port card and connect it that way.
13:11.20WIMPyI have never tried T.38.
13:11.44zambatoo late.. already bought two of those single port ISDN cards :)
13:11.46WIMPyIn fact I haven't tried fax at all in many years. More than I've been using Asterisk.
13:11.48zambawhich seems to be working just fine
13:12.03WIMPyYou could use a 3rd :-)
13:12.11zambanot enough room in server, actually :)
13:12.24WIMPyOr you go for USB.
13:12.28zambayeah
13:12.38zambaor just get rid of fax
13:12.44WIMPyNot to my liking, but I read from people using it on their servers.
13:12.53WIMPyThat's a good idea :-)
13:13.03WIMPyOr let Asterisk do the faxing.
13:15.03zambayeah, but i need some howtos for that :)
13:15.11zambabtw.. late/early media.. how can i check that it's turned off?
13:17.49WIMPyYou can set "earlyb no" but defaults should be just fine.
13:18.28zambawhere do i set that?
13:18.38WIMPyunder the interface.
13:18.58WIMPyBut you shouldn't need to configure anything more than what you have.
13:19.04zambaok, good
13:19.33WIMPyYou might want to use screen-in if you want formatted numbers in CALLERID(num).
13:19.51WIMPyOtherwise you get them the way they are sent from your telco.
13:24.15zambascreen-in?
13:24.29zambai should try setting callerid here
13:24.32zambasee if i can change that
13:25.45zambai see that i get the numbers prefixed with '0' first
13:26.18WIMPyThat's unusual.
13:26.32WIMPyNormally the 0s aren't transmitted.
13:26.58WIMPyThat's where screnn-in comes handy to add them depending on the number type.
13:27.43zambadoesn't look like portnum 1 worked
13:27.52zambawasn't able to dial out anymore when i had that defined
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13:28.35WIMPyThen the 2nd port doesn't seem to work.
13:29.08zambaoh.. seems like i had to do portnum 0, ptp and then portnum 1, ptp
13:29.37zambaguess portnum 0 tried to use ptmp then
13:29.38WIMPyah
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13:30.06WIMPyInterestimg. So you can even mix different types of line on one logical interface.
13:30.21zambalooks like it
13:30.57zambawhat's the "correct" way of setting callerid in asterisk these days?
13:31.05zambai'm just going to try changing it
13:31.19WIMPySet(CALLERID(num)=
13:32.17zambawhich worked :)
13:32.44zambalovely
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13:58.05mathiWIMPy, hey! I was thinking about our conversation yesterday. If I do call forwarding... won't I lose the caller ID ???
13:58.31jmetroYou can set your own callerid in dialplan
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14:02.58mathijmetro, I mean to get the number of the person that called
14:03.50WIMPymathi: no
14:05.46mathiWIMPy, 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
14:06.45mathiWIMPy, 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
14:06.45WIMPyIf they use one.
14:06.55mathiWIMPy, I will give them one
14:07.13mathior they cna buy one themselve
14:07.55WIMPyBut then they would need to forward all calls to their lindline they want to get.
14:07.59mathiWIMPy, look 1 euro per month!!  http://www.ovh.co.uk/VoIP/packages/online_sip_individual.xml
14:08.36mathiWIMPy, I didn't understand your last comment
14:09.18WIMPyHmm. free calls to 40 countries for £0.99/month? Sounds too good to be true.
14:10.19mathiWIMPy, 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
14:10.43mathiWIMPy, to have two simultaenous calls I need this offer: http://www.ovh.co.uk/VoIP/packages/online_sip_enterprise.xml
14:12.31WIMPyWould seem like two of the former ones would be a much better deal.
14:12.40WIMPyI still wonder where the catch is.
14:13.22mathiWIMPy, but can I use one account to do ingoing and another account to do outgoing ?
14:13.36WIMPyWhy not?
14:14.17mathiWIMPy, but then if I do outgoing with another account, the people will see a different number
14:14.28jmetroyou can set caller id
14:14.50WIMPyThat depends on the provider.
14:14.53mathijmetro, really?? but then I can spoof any number? sounds strange
14:15.16WIMPyNothing too special, really.
14:15.29jmetromy coworker frequently prank calls members of our office with the caller ID of our boss
14:15.37mathiLOL
14:15.43WIMPyThat service has been available in the PSTN for a ling time as well.
14:16.12mathiWIMPy, hy does it depends on the provider? it depends only on the caller ID set in the dialplan, no?
14:16.31WIMPyNot all providers will accept what you send.
14:17.34leifmadsenmany 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
14:21.10mathiso 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
14:21.52WIMPyBut 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
14:31.22leifmadsenI 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
14:32.32leifmadsenya just reading now
14:32.48leifmadsenwe 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
14:33.16mathiWIMPy, indeed it doesn't matter for my specific case :)
14:35.29leifmadsen_Corey_: sounds right :)
14:37.48[sr]_Corey_: nice, buy cisco buy... cisco is good.. (cof)
14:38.19WIMPyDo you really think that's a bug?
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14:45.17mathiif a phone is registered on two servers, and an extension is dialed, both servers will try to match the extension ? how does that work ?
14:53.07WIMPynot
14:59.06mathiWIMPy, a phone can only be registered to one server ??
14:59.26WIMPyDepends on the phone.
14:59.53WIMPyBut you will only dial out to one of the configured servers.
15:00.45mathiWIMPy, 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
15:01.07WIMPyBy the user selection an account.
15:01.27jmetrothose are line separate lines
15:01.35jmetrolike separate*
15:01.46WIMPyThe same way you'd select the outgoing number on an ordinary phone.
15:02.03WIMPyOr the same concept rather.
15:02.59mathiWIMPy, are there phones which allow easily to select a specific server (friendly labeled) ?
15:03.12WIMPySure.
15:03.22mathiWIMPy, cool:) then it is not a problem
15:03.26WIMPyI'd say most of them.
15:03.43mathiWIMPy, SIP phones ?
15:03.53WIMPyyes
15:04.45mathiWIMPy, can you recommend one?
15:05.02WIMPynot really.
15:05.19WIMPyI find the Snom 300 series least bad.
15:05.26mathiwhere can I buy the WIMPy phones?
15:05.36Sicelolol
15:06.37WIMPyI seriousely thought about bulding a phone some time back. But it just doesn't make sense.
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15:13.46mathiWIMPy, omg it's > 100 € ? I thought a phone was sold 15 € :p
15:14.37WIMPySIP phones are in the range of proprietary PBX phones.
15:15.21WIMPyPlus the running costs for electricity.
15:17.14jmetrosip phones are nicer
15:17.34WIMPyIn what way?
15:18.19WIMPyThey are surely nicer than rotary dial antique phones. But that's about it.
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15:21.19gnubienwant to use voip ATA when pc is poweroff; should i buy a cheap ethernet switch or router?
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15:35.05mathiWIMPy, doesnt the Snom 300 do too many for my needs ?
15:36.26WIMPyDo you wan to use it to replace whatever is at the customers office?
15:38.01SupaYoshiI think all IP phones are ugly as hell
15:38.09SupaYoshiThey need a good designer :P
15:38.14SupaYoshiToo bad apple doesnt make IP phones x
15:38.26jmetroif they did wed be paying 800$ for a desk phone
15:38.29WIMPy*cough*
15:38.39jmetroand couldnt provision them
15:39.00WIMPyAnd they would probably only support ilbc or something.
15:39.08SupaYoshi:P
15:39.13SupaYoshimwah atleast they would look pretty
15:39.25jmetrothe front of the phone would be glass..
15:39.25SupaYoshii mean I like these phones from philips :P but theyre not VOIP
15:40.57SupaYoshihttp://www.philips.nl/c/telefoons/design-collection-slim-line-zwart-id5551b_22/prd/nl/
15:41.39jmetrohttp://www.e-netsource.com/images/SNOM-821-IP-VoIP-Phone-with-Expansion-Module-Key-Panel-Black-e-NETSource.jpg?321
15:41.54SupaYoshio.o nice
15:42.02WIMPyprefers desktop phones.
15:42.26jmetroand the snom 800 is IP
15:42.36WIMPyAnd I really dislike the upstanding ones. Who cam up with that idea?
15:42.44jmetrothey are quite tall
15:42.56jmetroyou can reformat the base to make it lower but you cant see the pretty graphics then
15:43.38WIMPyA tiltable display is a must.
15:45.26jmetrolike the snom 340?
15:45.48WIMPy340?
15:47.49jmetro60*
15:48.12WIMPy320, 360 and 370 are all tiltable, yes.
15:48.58WIMPyNot having that is the most obvious drwback of my beloved Eurit 40.
15:49.22jmetrothats an odd looking phone. i like the keyboard though
15:49.53WIMPyIt's an extremely handy phone.
15:50.38WIMPyEven though it looks important it doesn't have that may features. But it does exactely what a phone should do.
15:50.50WIMPyVery easy handling of multiple calls.
15:52.14gnubienare Grandstream voip phones mostly junk?
15:52.36WIMPySome people say so.
15:52.41WIMPy~gs
15:52.41infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
15:52.46WIMPy~grandstream
15:52.46infobotfrom memory, grandstream is the Yugo of VoIP hardware.  Run...  Run away now.  Though, therealcircut says that they're not that bad.
15:53.30gnubienwhat brand of ATA will has the best quality for the lowest price?
15:54.29WIMPyFirst you have to pick one: best quality or lowest price?
15:55.33gnubienWIMPy: cisco spa2102 good quality?
15:56.06WIMPyI have no idea. Never used an ATA.
15:56.08gnubienthink lynksys sold to cisco, or visa versa
15:56.24WIMPyBut I'd look out for stuff like a symmetrical line.
15:56.38WIMPyCisco bought Linksys.
15:56.53gnubienok, will google symmetrical line
16:00.28jmetroi think all lines are symmetrical
16:00.51MaliutaWIMPy: current rumor is that cisco hasd engaged Lyods to find a buyer for linksys
16:01.09WIMPyjmetro: Unlikely as that's more expensive.
16:01.48jmetro</geometry humor>
16:02.07WIMPyoh
16:02.23gnubienare ATA's with built in routers better or worse than using a ATA with a ethernet switch or router?
16:02.47WIMPyThat obviousely depends on where you want to use them.
16:03.22gnubienwant to use the ATA when pc is power off so just ISP >> cable modem >> eth on pc now
16:04.02WIMPyDoes the modem include a router?
16:04.18gnubienno, just a cable modem, no router in the modem
16:04.37WIMPyThen you probably need a router somewhere.
16:04.57gnubienwould a ethernet switch be adequate?
16:05.11WIMPyUnless you ISP is as braindead as KDG and allows multiple public IPs.
16:05.16WIMPyNo
16:05.30WIMPyWell, probably no.
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17:34.19SupaYoshihi
17:34.29SupaYoshithe command cdr status doesn't work from me in ASTERISK CLI
17:34.46SupaYoshiis there anything im doing wrong? I read some guides, and you should be able to type that, according to these.
17:36.59SupaYoshinvm
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17:43.09SupaYoshimy asterisk keeps doing this
17:43.10SupaYoshidoing dnsmgr_lookup for 'sip.voipbuster.com'
17:44.33[TK]D-FenderSupaYoshi: It's a notice of general activity.  It isn't saying anything is wrong.
17:46.24SupaYoshitrue its annoying though
17:46.32SupaYoshishould i just type the ip in instead of sip.voipbuster.com :P
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17:47.28SupaYoshiMY 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-FenderDo you see the backend being loaded?
17:52.43[TK]D-FenderSo you see records being added?
17:52.57[TK]D-FenderHave you confirmed the table structure?
17:53.08[TK]D-FenderHave you tested the auth you provided for * to do so?
17:55.01SupaYoshiMhm i see records being added in the CSV Master file
17:55.09SupaYoshiI also see records being logged in the database
17:55.25SupaYoshiI checkec the passwords for CRD_
17:55.38SupaYoshifor cdr_mysql.conf and freepbx.conf
17:55.41SupaYoshithey match.
17:56.00SupaYoshiAlso 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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18:04.12SupaYoshiI can see the table cdr in mysql
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18:06.23[TK]D-FenderSupaYoshi: Asterisk has nothing to do with "your browser"
18:07.30carrarwait, Asterisk doesn't work in EI?
18:07.33carrar@#($@~!#$
18:07.38carrarerr IE
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18:10.58Penguin<SupaYoshi> Oh i thought maybe in verbose 10 the answer would be different
18:11.00Penguin<Penguin> It won't get more verbose above 3.
18:11.01Penguin<Penguin> Verbose 4 adds the dnsmngr stuff, I think it's verbose 7 where CDR is added...
18:11.06Penguinsupayoshi: Remember this?
18:11.33PenguinIf you are annoyed by the dnsmgr lookups, reduce your verbose level BELOW 4.
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18:30.16pabelangerPretty sure I patched that a while ago, cannot remember the verbose level I changed it too
18:37.16moos3I just installed asterisk 11 my sip clients have no audio ideas ?
18:46.04PenguinYou think you changed the dnsmgr lookups to a higher verbose level?
18:47.17pabelangerDon't think, I did. Change it to 6
18:47.18pabelangerhttp://svnview.digium.com/svn/asterisk/branches/1.8/main/dnsmgr.c?r1=353371&r2=360471
18:47.36pabelangerhttp://svnview.digium.com/svn/asterisk?view=revision&revision=360471
18:48.44pabelanger1.8.10.0+ has it
18:56.44PenguinAnd that carries into 10 and 11 as well?
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19:03.49pabelanger10 yes
19:03.58pabelanger11+ I changed it to a debug message I think
19:05.26PenguinI 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.
19:16.44jmetrosip set debug off
19:16.47jmetrocore set verbose 0
19:16.50jmetrocore set debug 0
19:21.51Penguincore set debug off
19:21.56Penguincore set verbose off
19:25.31carrarcore reboot now
19:27.20jmetrocore stop destructively
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19:42.51ChannelZcore stop from melting
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19:46.21Mololeifmadsen:  you there?
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20:24.10Penguinseri: Siphon on the iPhone 3G
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22:03.39SeRiPenguin: Thanks and Merry Christmass to all here @ #asterisk
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22:04.55gustoheh
22:05.18gustoi did set up my first TLS connection today
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22:20.36rkeeneHmm... Grr... nf_conntrack_sip seems to be broken with SIP over TCP
22:37.15*** join/#asterisk zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189)
22:37.34zopsiDoes anyone from Digium monitor this channel?
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22:38.25dymmerry christmas, everyone!
22:39.06PenguinMary Crispness
22:39.59filezopsi, there are Digium people here at times yes
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22:49.59zopsi@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.
22:50.37filenot at this moment, and tech support may still be there
22:50.47WIMPyBefore chrismas? Here it's already after christmas.
22:51.36zopsi@file, Doubtful. My support ticket hasn't been answered since its initial comment on Saturday.
22:51.45zopsi@WIMPy where are you from?
22:51.53fileI was talking to people in technical support earlier, there just isn't many
22:51.59WIMPyde
22:52.37zopsi@file, yeah completely understandable.
22:52.50zopsi@WIMPy, in that case Merry Christmas!
22:52.59dymWIMPy doesnt celebrate christmas.
22:53.10zopsigreat...
22:54.07zopsiDifferent religion or just scrooge?
22:54.51WIMPyI don't suffer from any of those deseases.
22:55.05zopsiIf anyone happens to know anything about DPMA and using the can_monitor and can_intercom functions in contacts let me know :)
22:55.47zopsichristmas in my household has very little religion involved (other than the name).
22:56.24dymit shouldnt have...
22:56.41dymchristmas is a time for gathering, not religious beliefs, or anything else randomly connected.
22:56.57WIMPyIt used to be a nice quiet time when you could work without disturbances.
22:57.20zopsi@dym : exactly
22:57.41zopsi@WIMPy: maybe in de
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23:01.50zopsiseriously though if anyone could help me out... I'd be very thankful.
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