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| 01:06.24 | luke-jr | upgraded Asterisk and getting "FD 32767 exceeds the maximum size of ast_fdset!", any ideas? |
| 01:09.03 | javier_cintron | Hi there, I am having some strange behavior between 2 ip phones and a normal one: http://pastebin.com/JLMvcMHT. I was trying to get more info about this problem using this command: sip show channel but asterisk cli doesn't show anything and stops to work. Asterisk 1.8.2.3 on Centos 5.5 32 bits |
| 01:10.00 | javier_cintron | I can open another termianl with ctrl+alt+f3, centos is working |
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| 02:25.31 | p3nguin | middleman: Did you solve your problems? |
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| 02:46.49 | Middleman | not yet, cant test it anymore tho |
| 02:47.10 | Middleman | my land line provider suspended my phone |
| 02:47.10 | Middleman | lol |
| 02:47.19 | Middleman | i paid but, takes till monday for the money to go through |
| 02:51.42 | *** join/#asterisk buzubada (~biz@c-24-7-188-35.hsd1.ca.comcast.net) |
| 02:52.11 | buzubada | Greetings all, does anyone have time to answer a few basic questions I have about Asterisk and SIP in general? |
| 02:52.41 | WIMPy | ~ask |
| 02:52.41 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:54.01 | buzubada | When getting SIP trunks, does the number of SIP 'users' I obtain from the provider equal the number of simultaneous in/out calls that can be made (analogous to POTS can make one call per line)? |
| 02:54.39 | WIMPy | No. And |
| 02:54.42 | WIMPy | ~siptrunk |
| 02:54.42 | infobot | from memory, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 02:55.37 | buzubada | ~siptrunk |
| 02:55.37 | infobot | siptrunk is probably something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 02:55.58 | buzubada | Sorry, I did not mean to spam. |
| 02:57.35 | buzubada | So if that is the case, is it safe to assume that the provider sets how many calls can be made and/or recieved at the same time? |
| 02:57.41 | p3nguin | buzubada: Normally, a single "user," which I would call an account, has multiple channels for calls to pass over. |
| 02:57.53 | WIMPy | yes |
| 02:58.23 | p3nguin | buzubada: My accounts all over unlimited channels, but they are metered (I pay for minutes used per channel). |
| 02:58.51 | p3nguin | s/over/offer/ |
| 02:59.07 | buzubada | Cool, thanks for that. If I want to output to a fax machine (yes, I know that fax machine over VoIP is anything but recommended) is there a way to use a standard modem, or is one of the digium FXS/FXO cards required? The EU has an extremely limited budget. |
| 02:59.54 | p3nguin | buzubada: You can use an ATA that supports fax mode. It'll be a lot cheaper than buying a card to go inside the computer. |
| 03:00.08 | WIMPy | Amode to receive fax or a modem to connect your fax to? |
| 03:00.11 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 03:00.26 | WIMPy | A modem to receive fax or a modem to connect your fax to? |
| 03:00.37 | buzubada | To connect the fax to. |
| 03:00.43 | WIMPy | No go |
| 03:00.52 | buzubada | So it can send and receive. |
| 03:01.03 | p3nguin | ATA <-- hint |
| 03:01.19 | buzubada | I am googling that now p3nguin. |
| 03:01.39 | p3nguin | PAP2T might be a good one. Check the specs and capabilities. |
| 03:02.37 | buzubada | That is a very good recommendation p3nguin, thank you! |
| 03:02.38 | p3nguin | If you aren't familiar with what an ATA does, it connects to the LAN by Ethernet for connectivity to Asterisk, and provides two phone jacks to plug in regular phones/fax. |
| 03:03.06 | p3nguin | It's like the box Vonage gives to their customers. |
| 03:03.28 | buzubada | I see that now :D. |
| 03:03.53 | p3nguin | You just program the lines in the ATA to talk to Asterisk. |
| 03:04.18 | buzubada | My final question before I become an annoyance, does anyone have a good, reliable, and preferablly compression-selectable (so I can go light compression for better faxing) VoIP or SIP Trunk provider that works with Asterisk? |
| 03:04.20 | p3nguin | If you buy the PAP2T on ebay or other online entity, make sure they can verify it is unlocked. |
| 03:04.45 | p3nguin | You cannot do compression on faxing over Voice over IP. |
| 03:04.57 | p3nguin | and there is still no such thing as a SIP trunk. |
| 03:05.06 | buzubada | Which is precisely why I want to make sure I can select compression-less VoIP. |
| 03:05.15 | p3nguin | "SIP Trunk provider" = ITSP |
| 03:05.39 | buzubada | I apologize - I saw several sites referring to that. |
| 03:06.11 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 03:06.13 | p3nguin | For fax over Voice over IP, you'll want to use g.711 ulaw or alaw codec. |
| 03:06.51 | buzubada | I assume I have to make sure the VoIP provider supports that. |
| 03:06.57 | p3nguin | For the US and metered SIP channels, I use VoIP.ms and Flowroute as my ITSPs. |
| 03:07.02 | p3nguin | They all do. |
| 03:07.03 | p3nguin | It's the standard. |
| 03:07.13 | buzubada | Okay, that's what I wanted to verify. |
| 03:07.25 | p3nguin | ulaw in the US and alaw in most every other part of the world. |
| 03:08.09 | buzubada | I noticed that on the asterisk site. I was just unsure whether or not I had to verify that the provider supported it. |
| 03:08.30 | p3nguin | You can; they usually offer that info on their web sites. |
| 03:08.41 | p3nguin | http://voip.ms/ |
| 03:08.48 | p3nguin | They show it right on the frontpage. |
| 03:09.03 | p3nguin | - Supported Codecs - |
| 03:09.30 | buzubada | Ahh |
| 03:14.11 | p3nguin | Are there any regular PCI gigabit dual-port NICs available anywhere? |
| 03:14.54 | p3nguin | There are all kinds of PCI-E and PCI-X dual-port gigabit, but I can't find any regular PCI. |
| 03:14.56 | WIMPy | I think there's an Interl one. Otherwise the Sun Quad-GE springs to mind. |
| 03:15.35 | WIMPy | Well, that might be because it's a little slow. |
| 03:16.19 | p3nguin | I have an Intel dual-port 10/100 regular PCI card that I just recently yanked out of my Asterisk box for lack of usage. I was going to recycle it into something else, then I realized it's only 10/100. |
| 03:16.46 | buzubada | http://www.newegg.com/Product/Product.aspx?Item=N82E16833106055 |
| 03:16.54 | buzubada | There's an intel one p3nguin |
| 03:17.01 | buzubada | Way spendish though |
| 03:18.58 | buzubada | Grr didn't pay attention, it's pci-x |
| 03:19.50 | WIMPy | Well, you could put it in a 32-bit slot. But it doesn;t make much sense to have GE then. |
| 03:20.31 | buzubada | Yeah, I doubt PCI 32 has the bandwidth necessary to accomodate dual gig |
| 03:20.40 | WIMPy | More if it's 66MHz. But that's still less than required. |
| 03:20.48 | WIMPy | It doesn't. |
| 03:21.13 | WIMPy | It isn't even enough for single GE. |
| 03:23.41 | WIMPy | Unless it's 66MHz, I should add. |
| 03:24.52 | buzubada | Actually, if the entire bandwidth was available for the network traffic, it could almost suffice... you get 266MB/s on 66mhz, gigabit is only 128MB |
| 03:25.19 | WIMPy | But GE is full duplex. |
| 03:25.32 | WIMPy | So, yes with 66MHz it would be ok. |
| 03:25.34 | buzubada | DOH forgot to double for duplex. |
| 03:25.57 | WIMPy | But you'd have to share with other devices on the same bus. |
| 03:26.21 | buzubada | Yes, so you will never have full gigabit. |
| 03:27.05 | WIMPy | I wouldnt say never, but, chances are it's slower than you want it to be. |
| 03:28.29 | *** join/#asterisk Infested (Infested@24-148-117-190.ip.mhcable.com) |
| 03:28.47 | WIMPy | OTOH it will be faster than 100MBit. |
| 03:29.08 | p3nguin | So I can plug in a long-slot PCI-X card into my regular PCI short slot on the computer's board? |
| 03:29.22 | WIMPy | Then again if you suffer from buffer overruns it might actually get slower than with 100. |
| 03:29.28 | buzubada | Usually. |
| 03:29.38 | WIMPy | As long as it has the right voltage, yes. |
| 03:30.05 | WIMPy | Well, I'm not sure it's a must, but I did work for me. |
| 03:30.32 | buzubada | Some 64-bit cards don't have the ability to work half bus-width. |
| 03:30.48 | buzubada | But it's usually only SCSI controllers and OOB management cards that lack that ability. |
| 03:30.49 | WIMPy | The cards or tehir drivers? |
| 03:30.56 | buzubada | The cards themselves. |
| 03:32.21 | WIMPy | That's easier with PCI-e where there's always only one lane required and the other up to 15 are optionally negotiated after initial setup. |
| 03:32.35 | p3nguin | I only have a single PCI slot, but I don't know what else could operate on the bus in the absense of add-in cards. |
| 03:32.41 | *** join/#asterisk AskewDread (Someone@60-234-236-207.bitstream.orcon.net.nz) |
| 03:32.45 | AskewDread | hey guys |
| 03:32.50 | WIMPy | The on-board stuff. |
| 03:32.55 | AskewDread | is anyone able to help me with this error? [Mar 5 03:29:44] WARNING[22595]: chan_sip.c:5445 sip_call: No audio format found to offer. Cancelling call to 1021815184 |
| 03:33.26 | WIMPy | If you've only got one slot that will certainly sith on the same bus as all the integrated stuff. |
| 03:33.30 | p3nguin | Right, but I'm saying I don't know what all exists that has to operate on the bus. There are no more card slots for more cards, so they aren't an issue. |
| 03:34.03 | buzubada | Your onboard audio, onboard NiC, probably onboard ATA controller |
| 03:34.09 | buzubada | Those will all use the same PCI bus. |
| 03:34.16 | WIMPy | yes |
| 03:34.22 | buzubada | Especially as PCI becomes less and less utilized. |
| 03:34.34 | WIMPy | If you've got more slots, chances are that it's more than one PCI bus. |
| 03:34.36 | p3nguin | askewdread: Sounds like something codec-related. Are you playing sound files, or trying to make calls? |
| 03:34.47 | AskewDread | trying to make a call |
| 03:35.01 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:35.06 | p3nguin | askewdread: Do you know how to get sip debug for us to analyze? |
| 03:35.27 | AskewDread | no, i know how to get into the console? |
| 03:35.41 | p3nguin | Typically you ssh to the asterisk host. |
| 03:35.48 | p3nguin | Then you run asterisk -r. |
| 03:35.58 | AskewDread | yu, in there |
| 03:36.14 | AskewDread | yup* |
| 03:36.38 | p3nguin | Once you are in the Asterisk console, run "sip set debug on" and you'll start seeing all the SIP traffic fly by. |
| 03:38.02 | AskewDread | ok, do you want me to send you the output of it when i make a call? |
| 03:38.17 | p3nguin | I usually like to wait until 1/2 a second before I dial my phone to run sip set debug (on) and then immediately turn it back off after I finish the call attempt. This reduces the crap before and after the call to make debugging easier on me. |
| 03:38.36 | p3nguin | Yes, I'll need to see the entire call, including the failure. Pastebin it for me. |
| 03:39.02 | p3nguin | Don't hide anything in it. Only mask passwords if you see any plain-text passwords. |
| 03:40.05 | AskewDread | http://pastebin.com/fdfuTpwj |
| 03:43.34 | p3nguin | It doesn't look like it's a problem while making a call from your phone into Asterisk, but it looks to me like it happens when Asterisk tries to send the call on to where it's supposed to go. |
| 03:44.05 | AskewDread | hm ok, that would make sense as local calls work fine |
| 03:46.02 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 03:46.22 | AskewDread | its so odd |
| 03:47.08 | buzubada | Provider error? |
| 03:52.31 | AskewDread | hmm |
| 03:52.32 | AskewDread | its odd |
| 03:52.36 | AskewDread | works with other things |
| 03:52.39 | AskewDread | just not this |
| 03:59.49 | *** join/#asterisk strehlow (~strehlow@24-247-41-88.dhcp.mrqt.mi.charter.com) |
| 04:01.08 | buzubada | Would Asterisk for a maximum of 10 users run well on an atom dual-core? |
| 04:02.01 | buzubada | Or, would I be better off running it as another VM appliance on an existing server on site? |
| 04:02.02 | p3nguin | probably |
| 04:02.49 | p3nguin | Most of the professionals are not keen on running Asterisk in a virtual machine. |
| 04:04.10 | buzubada | I figured as well, but the EU's needs are light at best and they have a small budget so I am just trying to figure out best performance vs price. |
| 04:10.16 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 04:10.17 | p3nguin | You say 10 users, but what exactly do you intend to do? |
| 04:11.51 | p3nguin | The cheapest, most power efficient desktop PC you can find could be a good way to go. |
| 04:13.18 | WIMPy | If you only want to do SIP with directmedia, you might get away with a wrist watch :-) |
| 04:25.54 | p3nguin | I'm quite fond of using HP Compaq business class workstation computers for projects. They're great machines, and you can get them used on ebay for good prices. They come with AMD or Intel processors from 2.0 to at least 3.0 GHz and usually support up to 4G RAM. |
| 04:26.51 | p3nguin | I personally own one AMD 2.0 GHz machine and one Intel Pentium 4 2.8 GHz machine. |
| 04:28.32 | p3nguin | I run my office PBX on a 12 year old Gateway desktop with a Pentium 3 in it. |
| 04:30.39 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 04:31.02 | p3nguin | Asterisk doesn't need very much CPU or RAM, so I recycle old computers for Asterisk usage when possible rather than spending unnecessarily for new hardware. |
| 04:32.03 | WIMPy | CPU depends on what you want it to do and what you have connected. |
| 04:33.37 | *** join/#asterisk cerberus_za (~coert@196-215-44-48.dynamic.isadsl.co.za) |
| 04:33.39 | p3nguin | That's true. I don't do anything fancy, so I'm good with a slow (by today's standards) CPU. Some transcoding, MeetMe conferences, recording... |
| 04:34.02 | p3nguin | I almost never see Asterisk's CPU usage over 10%. |
| 04:50.06 | buzubada | Sorry for nonresponsiveness penguin and wimpy, I was taking care of the kid for a few. |
| 04:50.28 | buzubada | I just want to help a client replace their ultra-old POTS-dependant phone system with a cheap PBS> |
| 04:50.35 | buzubada | *PBX |
| 04:51.21 | buzubada | Then they can switch to Comcast's 50/10 service and I can relocate my backup VM host nodes to their site. |
| 04:52.23 | buzubada | A beneficial arrangement, but neither of us have any significant budget for this, so I am trying to keep cost to an absolute minimum. |
| 04:52.39 | buzubada | and by 10 users I mean they have 10 extensions. |
| 04:53.49 | buzubada | (to be concise) |
| 04:56.38 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 05:23.00 | *** join/#asterisk buzubada (~biz@c-24-7-188-35.hsd1.ca.comcast.net) |
| 05:25.51 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 05:25.52 | p3nguin | 10 "extensions" huh? You must be pretty new to Asterisk. |
| 05:39.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 05:44.01 | buzubada | Never tried it yet. |
| 05:44.12 | p3nguin | Seriously? |
| 05:44.38 | AskewDread | hm ok, a new asterisk install has fixed the codec issue... |
| 05:44.50 | buzubada | Yes, I am going to download and play with it tomorrow. |
| 05:44.55 | buzubada | For now just researching it. |
| 05:45.02 | buzubada | But I know how a traditional PBX works. |
| 05:45.08 | p3nguin | Then I'll have to give you a pass on the "extensions" thing for now, but I'm going to tell you my reasoning for the comment. |
| 05:46.22 | p3nguin | Somewhere in time, someone took it to mean that because an extension was used to reach a phone that the phone itself was an extension. |
| 05:46.51 | *** join/#asterisk strehlow (~strehlow@24-247-41-88.dhcp.mrqt.mi.charter.com) |
| 05:47.19 | p3nguin | In Asterisk, extension has a very specific meaning, and an extension does not refer to a phone. Extensions are the dialing rules that make things happen when you pick up a phone and make a call. |
| 05:47.41 | buzubada | I see. |
| 05:48.03 | buzubada | What would the asterisk term for a user/slot/whatever be? |
| 05:48.05 | p3nguin | Phones themselves can be referred to as phones, devices, endpoints, peers, and in certain cases sometimes users. |
| 05:48.18 | buzubada | okay. |
| 05:48.39 | buzubada | So I am looking for 10 endpoints, with up to 10 users and up to 10 vm's |
| 05:49.15 | p3nguin | Extensions in Asterisk are found in the configuration file by the name of extensions.conf. There is absolutely no phone configuration done in extensions.conf. |
| 05:49.45 | p3nguin | Extensions are the dialing rules. Extensions can be used to dial phones using the Dial() applications. |
| 05:50.02 | p3nguin | application, rather. not plural. |
| 05:50.10 | buzubada | Do you think that if I just set that to run on a dual-core atom, that it would sufficiently handle the load as well as the zimbra voicemail tie-in |
| 05:50.28 | p3nguin | Extensions can also do other useful things. Extensions do not have to dial devices or peers. |
| 05:51.21 | p3nguin | 10 phones with 10 people using them, and 10 voicemail mailboxes... that isn't going to create much load in Asterisk no matter what they are doing. |
| 05:51.33 | WIMPy | That should fit easily, even if I've got no clue what zimbra is. |
| 05:51.55 | buzubada | (It's a collaboration software, similar to exchange.) |
| 05:52.24 | buzubada | That is good to know too penguin. |
| 05:52.47 | p3nguin | Asterisk causes load on a CPU when device registrations occur, when call setup happens, and when there is transcoding between codecs going on. |
| 05:53.02 | buzubada | So extensions is where I would need to tinker if (for instance) they made international calls and I wanted it to use different providers based on rate? |
| 05:53.21 | p3nguin | What other significant CPU use load event am I leaving out? |
| 05:53.32 | p3nguin | That's correct. |
| 05:53.52 | buzubada | I am sure there's some cpu loading when VM is retreived etc. |
| 05:53.58 | WIMPy | hardware |
| 05:54.15 | WIMPy | not really |
| 05:54.29 | p3nguin | You would simply make the extension match the number they are dialing from the phone and make it Dial() out a different peer that you've configured. |
| 05:55.41 | p3nguin | wimpy: When you said "hardware," what was that in response to? |
| 05:56.21 | WIMPy | <p3nguin> What other significant CPU use load event am I leaving out? |
| 05:56.26 | WIMPy | That one |
| 05:56.48 | p3nguin | What hardware event causes CPU usage? |
| 05:57.02 | WIMPy | hardware interfaces |
| 05:57.07 | WIMPy | If you have any. |
| 05:57.25 | buzubada | A sidebar question, do you have any suggestions for a good source to get the phones themselves from? I found some decent-seeming SIP phones on Amazon, but the price is pretty high for what they want to spend. |
| 05:57.46 | p3nguin | That depends on what you want and how much you want to spend. |
| 05:58.13 | buzubada | Basic office phones, and ideally $50 each or less. (I know that's asking a lot) |
| 05:58.14 | p3nguin | If you want basic IP phones, many people are plenty happy with Polycom phones and their prices. |
| 05:58.35 | p3nguin | You're probably not going to find any new phones that are worth a crap for $50. |
| 05:58.57 | buzubada | Are Grandstream phones remotely decent? |
| 05:59.02 | p3nguin | You can probably pick up used phones on ebay for that much. |
| 05:59.05 | p3nguin | ~gs |
| 05:59.05 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 05:59.20 | buzubada | LOL |
| 05:59.26 | buzubada | Good to know. |
| 05:59.27 | p3nguin | ~grandstream |
| 05:59.27 | infobot | rumour has it, grandstream is the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 06:00.02 | p3nguin | I don't know the going rate on used Polycoms. |
| 06:00.04 | buzubada | Unfortunately they are hurting bad, so their budget is under $1000 and anything above that I have to eat. |
| 06:00.21 | WIMPy | If you want something really cheap, the AllNet 7950 is usable. The 7960 can only be used as a door stop. |
| 06:00.22 | p3nguin | I know how much used Cisco phones go for. |
| 06:01.17 | p3nguin | Using Cisco phones often has a licensing issue that some people are concerned about. I typically don't worry myself with it, though. |
| 06:02.21 | buzubada | What licensing issue? |
| 06:03.02 | p3nguin | Cisco has rules about using their products and require the user to have a license. |
| 06:03.47 | buzubada | Ahh, I figured you were referring to something more along the lines of required software architecture. |
| 06:04.00 | buzubada | (CCIE here) |
| 06:05.00 | buzubada | Small applications rarely need to worry about second sale issues. |
| 06:07.33 | buzubada | How well do the Linksys/Cisco phones such as this one rate? http://www.amazon.com/Cisco-SPA941-4-line-1-port-Ethernet/dp/B000CETHA8/ref=pd_sim_sbs_e_11 |
| 06:07.54 | p3nguin | Two popular suppliers are voipsupply.com and telephonydepot.com |
| 06:08.41 | p3nguin | I've heard some complaints abou the 941 and others in that series. Take a look at the new 500 series. |
| 06:08.46 | WIMPy | doesn't like the sipura user interface. |
| 06:09.00 | WIMPy | But tehn I'm non north american. |
| 06:09.19 | p3nguin | http://www.telephonydepot.com/Catalog/Cisco-Phones They have some 500 series phones here. |
| 06:09.54 | WIMPy | But then I haven't seen a VOIP phone yet, I really liked. |
| 06:12.30 | p3nguin | Ebay has the Polycom 321s for around $100. |
| 06:12.41 | buzubada | Well, the reality is they are currently used to a phone system that wasn't even new when they acquired it in '95. |
| 06:13.08 | buzubada | So I am sure even the GrandSuck phones will be decent compared to their current system. |
| 06:13.18 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 06:13.49 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:15.22 | p3nguin | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=330534926615 |
| 06:15.31 | p3nguin | Buy a dozen of these. |
| 06:16.03 | p3nguin | For $55, you'll have a better phone than any GrandStream. |
| 06:16.40 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:16.40 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 06:17.35 | p3nguin | Heck, I might even get one just so I can play with a Polycom phone. I'm using all Cisco desk phones, and I have a generic cordless SIP phone for my home phone. |
| 06:17.44 | buzubada | That does sound like a good deal. |
| 06:18.30 | p3nguin | Maybe Leif will give us his opinion on the SoundPoint IP 320. |
| 06:19.06 | buzubada | I wonder how this person has 37 sold and more than 10 still available though |
| 06:19.26 | p3nguin | He must have had more than 47 when he started. |
| 06:20.11 | p3nguin | Ebay doesn't say 187 available... they just say more than 10 if you have more than 10. |
| 06:20.29 | WIMPy | Some call cernter that used them? |
| 06:20.41 | p3nguin | They are all new in box. |
| 06:20.55 | WIMPy | didn't read that far. |
| 06:21.10 | buzubada | I know, what I am getting at is that's suspicious that a relatively new person has more than 47 of these phone NIB for about half what the normal going price seems to be. |
| 06:21.54 | p3nguin | Member since: May-29-07 |
| 06:22.00 | buzubada | Nevermind, I am seeing some in google shopping in the $70 range, so that's not so far-fetched a discount. |
| 06:22.15 | WIMPy | would also like to get his hands on a polycom, but they are quite rare over here. |
| 06:23.04 | WIMPy | And I'd also like to try one of the Siemens OpenStage things. |
| 06:23.24 | p3nguin | If you need one, you can buy it and have it sent to me and then I can handle the international shipping. I have no clue how much it would cost you to ship from the US to where you live, though. |
| 06:23.39 | psilikon | I am trying to figure out why I get no ring-back on my asterisk box. The sip provider is telling me that according to their logs the switch sends a SIP 183 but my asterisk box is failing to forward the early media. Any pointers would be extremely helpful. |
| 06:23.48 | WIMPy | Probably too much. |
| 06:23.56 | p3nguin | Quite possibly. |
| 06:25.02 | p3nguin | psilikon: While I don't have a reason for the lack of ringing, I experience it on my SCCP phones. Rather than worry with why it's like that, I just add the r dialing option to the Dial() command to provide false ringing. |
| 06:26.12 | WIMPy | I remember when I did that. Getting announces with ringback overlaid is funny. |
| 06:26.13 | p3nguin | Oh, earlyrtp... that gives me an idea. Rather than providing false ringing with the r option, I could switch on earlyrtp for these phones to see if that allows ringing to pass. |
| 06:26.43 | psilikon | p3nguin: i tried adding an 'r' and a 'R'. Still no jpy |
| 06:26.48 | psilikon | jpy=joy |
| 06:26.54 | p3nguin | The small r should have taken care of it. |
| 06:27.09 | p3nguin | Maybe you fouled up the syntax. |
| 06:27.27 | WIMPy | Or it's the phone. What kind of phone do you use? How is it connected to Asterisk? |
| 06:27.37 | p3nguin | If you did Dial(SIP/something,r) that's incorrect. |
| 06:27.47 | *** join/#asterisk kriegerod (~krieger@user-204.45.infomir.com.ua) |
| 06:27.49 | WIMPy | classic |
| 06:29.57 | psilikon | Dial(SIP/itsp/${EXTEN:1},60,tTor) |
| 06:30.50 | psilikon | The issue has to be related to the fact that the phone is a pots phone connected to a channel bank that connects to a Sangoma a104d |
| 06:30.54 | p3nguin | Two way DTMF transfers. Nice. |
| 06:31.58 | WIMPy | Oh, yes, I had quite some trouble to get early media going with dahdi. |
| 06:32.31 | WIMPy | But it does work now. |
| 06:33.07 | WIMPy | But still no go on late media. |
| 06:33.23 | p3nguin | What is late media used for? |
| 06:34.12 | WIMPy | The same as early media: Tones and announcements |
| 06:35.02 | p3nguin | I guess I don't understand why that stuff isn't sent during the normal media portion of the call. |
| 06:35.34 | WIMPy | Because the call end defines the end of billing. |
| 06:36.13 | WIMPy | And if it no connection is possible it would also be lat in order to indicate that the call failed before playing the announcement. |
| 06:36.14 | p3nguin | I need a phone number that will provide ringing so I can test my new theory. It seems like every number I think of answers damn near immediately. |
| 06:36.30 | WIMPy | Call your cell phone? |
| 06:37.41 | psilikon | I forgot to place a test call from a pure sip phone to see if I had ringing before I left. |
| 06:38.01 | psilikon | WIMPy: so what did you end up doing to get early media going? |
| 06:38.33 | WIMPy | Apply some patches. |
| 06:38.57 | WIMPy | I guess they are included in the latest release. |
| 06:39.20 | psilikon | what about dropping this in sip.conf progressinband=never prematuremedia=no? |
| 06:44.55 | WIMPy | Thinking about that, it might be a good ide, not to send a SIP BYE upon a DISCONNECT, but rather ignore it and send te BYE on REL COMP. |
| 06:46.20 | buzubada | So, if there is no such thing as SIP trunking, what are the providers (Such as Nextiva) who say SIP trunking referring to? |
| 06:46.59 | WIMPy | A SIP account that allows more than one concurrent connection? |
| 06:47.48 | p3nguin | My theory was correct. When I made sure earlyrtp was set to none and did not have the r dial option, I have no dialout sound. |
| 06:48.29 | p3nguin | The new default earlyrtp setting is progress, and I have the outbound ringing without the r option. |
| 06:49.02 | p3nguin | So the problem was never the ITSP, it was the chan_sccp-b default earlyrtp setting. |
| 06:49.15 | p3nguin | I'm glad this was brought up tonight. |
| 06:49.20 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 06:58.21 | psilikon | Something else I don't understand is why you don't need ports 5060 and 10000-20000 open on a firewall. Is it because most firewalls already accept ESTABLISHED? |
| 07:04.01 | *** join/#asterisk dimm (~kvirc@unaffiliated/dimm) |
| 07:04.10 | dimm | hello! where i can setup that inbound calling at some days will route to some destination ? |
| 07:08.53 | WIMPy | Ha! |
| 07:09.23 | WIMPy | After I changed from TLS back to UDP, I get that PUBLISH issue here as well. |
| 07:09.35 | WIMPy | But I still don't know when they happen. |
| 07:09.49 | p3nguin | dimm: extensions.conf |
| 07:10.12 | p3nguin | dimm: You will be using the GotoIf() application in your dial plan. |
| 07:12.06 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 07:12.06 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 07:13.06 | p3nguin | Why can't variables be used in hints? |
| 07:13.44 | WIMPy | Can't you use EXTEN? |
| 07:14.15 | WIMPy | Anything else seems ... intertesting. What are you trying to do? |
| 07:15.45 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 07:18.08 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 07:23.51 | p3nguin | Rather than writing SIP/asdf in the hint, I was trying to store the device info in the AstDB and use a DB lookup as the device in the hint. When that didn't work, I tried a variable (unfortunately I don't remember what variable I tried), and it just has blank information when I check the configured hints. |
| 07:25.12 | p3nguin | exten => 211,hint,SIP/${EXTEN}; |
| 07:25.18 | p3nguin | <PROTECTED> |
| 07:25.46 | p3nguin | I don't understand what would prevent it from working. |
| 07:28.04 | p3nguin | Maybe there's no code to make it work? I have no idea. |
| 07:28.07 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:37.02 | p3nguin | wimpy: Any thoughts about that? |
| 07:37.09 | *** part/#asterisk dimm (~kvirc@unaffiliated/dimm) |
| 07:38.02 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 08:03.03 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 08:03.31 | benngard | oops, if u type wrong in the dial plan exten => 5,n.Hangup(), a . instead of a , and do a reload u crash astersik |
| 08:04.09 | benngard | just did it :( |
| 08:04.21 | benngard | and ofc on a production server |
| 08:10.26 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 08:14.46 | p3nguin | It shouldn't be that sensitive. I've made plenty of typing mistakes while writing dial plan, and it never crashes Asterisk. |
| 08:15.08 | benngard | i am 100% sure, did it 2 times just to verify |
| 08:15.30 | p3nguin | When you run dialplan reload, it should just screw up pbx_config, not crash Asterisk. |
| 08:15.31 | benngard | saturday morning in sweden so its no big isue if i crasch it |
| 08:16.26 | p3nguin | I'll do it now. |
| 08:17.08 | florz | benngard: just read the parser a bit and you won't be surprised anymore ;-) |
| 08:17.17 | p3nguin | [Mar 5 02:17:11] WARNING[20342]: pbx_config.c:2266 pbx_load_config: Invalid priority/label 'n.Hangup' at line 45 |
| 08:17.22 | p3nguin | Nope, no crash. |
| 08:17.31 | p3nguin | Exactly as predicted. |
| 08:17.33 | benngard | ok let me try, |
| 08:17.54 | benngard | sip*CLI> reload |
| 08:17.54 | benngard | sip*CLI> |
| 08:17.54 | benngard | Disconnected from Asterisk server |
| 08:18.07 | p3nguin | Why would you run "reload" in the first place? |
| 08:18.29 | p3nguin | No one with any sense would do that. |
| 08:18.46 | p3nguin | If you modify dialplan, the correct command is diaplan reload. |
| 08:19.38 | benngard | i was "lazy" and didnt write "dialplan reload" just reload |
| 08:19.58 | benngard | but i dont think u should crash asterisk anyway |
| 08:20.07 | p3nguin | I doubt it did. |
| 08:20.15 | p3nguin | ps -C asterisk |
| 08:21.28 | benngard | sip*CLI> dialplan reload |
| 08:21.28 | benngard | sip*CLI> |
| 08:21.28 | benngard | Disconnected from Asterisk server |
| 08:21.46 | p3nguin | "ps -C asterisk" says what? |
| 08:22.08 | benngard | its dead, there is no asterisk running anymore, do belive |
| 08:22.12 | p3nguin | I'm concerned that you have a bug. What version are you using? |
| 08:22.30 | benngard | Asterisk 1.8.2.3 |
| 08:22.47 | p3nguin | Can you upgrade? |
| 08:22.52 | benngard | noop |
| 08:23.04 | benngard | have some t38 patches from may thats not offical |
| 08:23.24 | p3nguin | If you could upgrade, you could see if it still happens in the newest version and then submit a bug report. |
| 08:24.04 | benngard | i know i have to speak to may so we can inlude his t38 patches in the official asterisk |
| 08:30.57 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:41.17 | *** join/#asterisk mimage (~mimage@fedora/mimage) |
| 08:57.04 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 08:57.32 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 09:15.47 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 09:21.14 | *** join/#asterisk rethus (~suther@p549A738D.dip.t-dialin.net) |
| 09:21.26 | rethus | how can i check, if ami-connection works? |
| 09:45.06 | kaldemar | telnet to port 5038 (or some other you have set), you should see a string sent by asterisk. |
| 10:00.49 | *** join/#asterisk war9407 (war@71.6.165.232) |
| 10:01.24 | war9407 | question - using an spa3102 and asterisk, can the asterisk/spa3102 send incoming calls (outside POTS line) to a phone connected to the spa3102? |
| 10:01.36 | war9407 | I can place calls outbound but not inbound |
| 10:01.57 | benngard | no dial entry to your phone? |
| 10:02.10 | war9407 | also, voice mail/etc works but it doesnot ring my phone, hmm will go read more |
| 10:02.11 | war9407 | thx. |
| 10:02.30 | benngard | whats the name of your phone? |
| 10:03.20 | *** join/#asterisk phretor (~phretor@dynamic-adsl-78-12-226-76.clienti.tiscali.it) |
| 10:03.56 | phretor | hi I've been reading about Sphinx having troubles with Asterisk because of the 8khz acoustic model limitation |
| 10:04.09 | phretor | is it really an issue? |
| 10:06.06 | *** join/#asterisk pp07 (~fedup@182.71.111.126) |
| 10:06.16 | pp07 | Hello ! |
| 10:11.26 | pp07 | I am testing for outbound IVR with vicidial my outbound campaign is running sucessfully but , i want to setup a IVR when coustmore pick up the phone & after listning the prerecoded message the call auto disconnect... help me . Thanks in advance.. |
| 10:12.09 | *** join/#asterisk TimeRider (~steve@92.41.107.54.sub.mbb.three.co.uk) |
| 10:13.49 | *** join/#asterisk datarecal (~data@loxely.illusivecreations.com) |
| 10:13.52 | datarecal | morn |
| 10:14.27 | datarecal | having trouble, changed out my router and now im getting registration time outs, i opened all the ports again and forwarded them to the asterisk box and changed externip |
| 10:14.31 | datarecal | something I am overlooking ? |
| 10:14.50 | pp07 | I am from INDIA any Indian here?? |
| 10:16.45 | rethus | have someone asterisk run on vpn ? |
| 10:16.57 | rethus | how can i telnet to tun0 5038 to check AMI-connection |
| 10:17.03 | rethus | i got alway Connection refused |
| 10:25.04 | benngard | u have probably not enable the manager |
| 10:25.37 | phretor | anyone using sphinx with asterisk? |
| 10:25.43 | benngard | in manager.conf make sure u have [general] |
| 10:25.43 | benngard | enabled = yes |
| 10:25.55 | plundra | And if you did, did you bind the right address? Can this be reached via your vpn? |
| 10:26.15 | plundra | Routes set up correct etc. |
| 10:27.37 | datarecal | anyone help me with my problem i know its something stupid |
| 10:31.37 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 10:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:42.01 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:45.45 | phretor | where can I get free DID numbers that work with Asterisk (i.e., PBX, not SIP) |
| 10:46.35 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 10:46.35 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 10:46.36 | rethus | how can i exit the telnet-connection to 5038 ? |
| 10:46.41 | rethus | is there a command? |
| 10:51.02 | *** part/#asterisk phretor (~phretor@dynamic-adsl-78-12-226-76.clienti.tiscali.it) |
| 10:51.55 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 10:51.55 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 11:01.40 | *** join/#asterisk gerts (~gerts@ti0013a380-dhcp0255.bb.online.no) |
| 11:01.56 | gerts | Are there any FOSS project which have had great success by attending at FOSS evens to promote and build thefurther a community around? |
| 11:08.10 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 12:00.52 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
| 12:13.11 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 12:30.39 | SunTsu | I have a SIP account, sipgate, and 4 SIP phones, which I need to ring all at the same time and access the same mailbox. With sipgate itself I can just use the same account on multiple phones, which does not to be possible with asterisk1.6 - what would be the way to go? |
| 12:31.33 | SunTsu | er. there is a "seem" missing in the former line. Add it where you see fit ;) |
| 12:35.38 | *** part/#asterisk rethus (~suther@p549A738D.dip.t-dialin.net) |
| 12:44.34 | WIMPy | Create four accounts. |
| 12:55.51 | SunTsu | WIMPy: what's the most elegant way to let them ring all? Putting them all into Dial will make that a unmaintainable beast when more phone will be added. In fact I have 5 SIP lines and all four phones are able to use multiple accounts, and I don't want it to turn into a confusing setup |
| 12:56.56 | WIMPy | You could put a variable in between. |
| 12:57.16 | WIMPy | Or use other dialplan logic to concentrate repeating stuff. |
| 12:57.24 | WIMPy | But that's the way it's done. |
| 12:59.56 | *** join/#asterisk qjb (~qjb@2001:888:17b4:0:21f:5bff:fef1:f005) |
| 13:03.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:04.36 | SunTsu | WIMPy: thanks. Then I'll make my mind up about minimizing the dialplan |
| 13:05.39 | WIMPy | I thik many of us use their own config files and scripts to generate the Asterisk configuration. |
| 13:05.45 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 13:16.29 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-242-220.pools.arcor-ip.net) |
| 13:19.41 | *** join/#asterisk philfine (~philfine@56.4.166.178.rev.vodafone.pt) |
| 13:19.46 | philfine | Hello everyone |
| 13:21.15 | philfine | What should I see in tcpdump on the asterisk server during call that uses SayNumber ... I mean a call to the server itself |
| 13:21.30 | philfine | I am asking since tcpdump doesn't show anything :D |
| 13:34.09 | *** join/#asterisk dr__ (~dr@79.113.187.158) |
| 13:52.07 | *** join/#asterisk Dr-Linux (~Dr-Linux@182.177.233.197) |
| 13:56.51 | *** join/#asterisk philfine (~philfine@a213-22-178-36.cpe.netcabo.pt) |
| 13:58.33 | philfine | I want to be able to connect from NAT from a different public ip address |
| 13:58.42 | philfine | What exactly should be done for this to work ? |
| 13:59.33 | *** join/#asterisk neurosys (~neurosys@c-98-254-246-250.hsd1.fl.comcast.net) |
| 14:05.26 | *** join/#asterisk BeeBuu (b71b2db6@gateway/web/freenode/ip.183.27.45.182) |
| 14:07.30 | BeeBuu | hello,all |
| 14:10.39 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:37.57 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 14:40.45 | benngard | shouldnt NoOp(${CALLERID(DNID)}) return the called number? |
| 14:41.08 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:41.25 | benngard | its empty in my case |
| 14:43.39 | *** join/#asterisk hairyraven (~nobody@95.73.205.70) |
| 14:45.59 | Gugge | the called number should be the extension |
| 14:50.00 | benngard | i must be doing something very wrong here |
| 14:52.49 | benngard | exten => 0317998985,n,NoOp(${CALLERID(name)}) woks, exten => 0317998985,n,NoOp(${CALLERID(DNID)}) is empty, strange |
| 14:54.20 | WIMPy | I'm not sure how it is defined in Asterisk, but my guess would be that it is only set when you receive a redirected call. |
| 14:56.13 | benngard | u must be able to find out what number have been dialed |
| 14:56.28 | WIMPy | That's the extension. |
| 14:56.51 | benngard | yes i know that, but for example, u can reach my extension in many ways |
| 14:56.59 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:57.18 | benngard | i can join a queue for example, could be nice to know how the call came to me |
| 14:57.29 | WIMPy | You get it in ${EXTEN} |
| 14:58.00 | WIMPy | Save it in another variable if it gets lost in the dilplan. |
| 14:59.45 | benngard | yes, i know i can do that, but still cant figure out why DNID is not working |
| 15:00.54 | *** part/#asterisk mimage (~mimage@fedora/mimage) |
| 15:01.44 | WIMPy | My guess is still that it's only for redirected calls. |
| 15:01.46 | *** join/#asterisk aanev (~aanev@79.126.229.50) |
| 15:03.17 | aanev | Hi, Can someone help me to store DID number in userfield of database, where to put exten => Set(CDR(userfield)=${EXTEN}), or maybe there is some other way to see it in my custom destination.. |
| 15:03.57 | *** join/#asterisk fejj (~fejj@c-24-91-170-86.hsd1.ma.comcast.net) |
| 15:04.51 | fejj | hey I'm working on a patch to port asterisk from gmime2.2 to 2.4 (which changed API/ABI) |
| 15:05.02 | fejj | where should I send my patch? |
| 15:06.17 | *** join/#asterisk coppermine (~inside@unaffiliated/coppermine) |
| 15:06.46 | coppermine | i have freepbx/asterisk/centos in virtualbox, when i run nmap on it i dont see port 5060 udp, but netstat -plantu | grep 5060 shows it there |
| 15:07.04 | *** join/#asterisk fofware (~Fabian@host63.190-31-62.telecom.net.ar) |
| 15:07.10 | coppermine | but i cant connect to it, not even on the local machien with telnet localhost 5060 |
| 15:08.08 | WIMPy | benngard: Looking at the source it actually looks completely unused. |
| 15:08.17 | blitzrage | p3nguin: may be a bit late, but I really like my IP335 (Polycom). I've used (and still have) a 430 and a 501, and have liked every polycom phone I've ever used. |
| 15:08.33 | blitzrage | While I haven't used the 320 specifically, I presume it'll be just as good as other Polycom phones |
| 15:08.45 | blitzrage | when I do installations for customers I don't recommend anything else, if that helps. |
| 15:10.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:12.30 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 15:13.30 | aanev | anyone can tell me a way to read DID in my custom destination dailplan ? |
| 15:16.33 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 15:25.53 | wdoekes2 | ${EXTEN} ? |
| 15:26.08 | *** join/#asterisk wonderworld (~ww@port-92-201-68-31.dynamic.qsc.de) |
| 15:27.01 | wdoekes2 | or if you mean you want to read the DID interatively, try the Read() application |
| 15:27.12 | *** join/#asterisk coppermine (~inside@unaffiliated/coppermine) |
| 15:27.18 | wdoekes2 | s/interatively/interactively |
| 15:27.26 | coppermine | i have udp 0 0 0.0.0.0:5060 0.0.0.0:* 3320/asterisk |
| 15:27.32 | coppermine | but it wont connect to a sip softphone |
| 15:27.39 | coppermine | nmap doesnt show it listenning either |
| 15:27.48 | coppermine | what can I check? |
| 15:27.52 | coppermine | firewall is disabled on centos |
| 15:30.09 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 15:30.53 | *** join/#asterisk mimage (~mimage@fedora/mimage) |
| 15:31.05 | wdoekes2 | nmap won't show whether applications are "listening" |
| 15:31.24 | aanev | wdoekes2: but {exten} in custom destionation is equal to s, so I need to put exten=>s,a,Set(CDR(userfield)=${EXTEN}) somewhere before my custom destination |
| 15:31.26 | coppermine | well i cant even do "telnet localhost 5060" |
| 15:31.26 | wdoekes2 | try sipsak instead |
| 15:31.28 | coppermine | on the localhost |
| 15:31.32 | coppermine | so its not listenning |
| 15:31.39 | wdoekes2 | telnet doesn't do udp |
| 15:31.51 | coppermine | so what should I check |
| 15:32.04 | coppermine | i set allow 0.0.0.0/0.0.0.0 |
| 15:32.07 | coppermine | and nothing in deny |
| 15:32.41 | wdoekes2 | aanev: if ${EXTEN} is 's', how do you plan on getting anywhere? |
| 15:32.51 | wdoekes2 | if the DID has been mangled earlier, fix that |
| 15:33.03 | *** part/#asterisk fejj (~fejj@c-24-91-170-86.hsd1.ma.comcast.net) |
| 15:33.11 | wdoekes2 | (I'm suspecting you're using some kind of prefab dialplan here.. blame it on the prefab) |
| 15:33.30 | wdoekes2 | coppermine: allow/deny is for codecs, don't worry about those for now |
| 15:33.52 | wdoekes2 | or was it allow/reject |
| 15:34.11 | aanev | wdoekes2: I like to store DID number in userfield of CDR database so I'm asking where to put this because in extensions_additional it gets owerwriten |
| 15:34.28 | wdoekes2 | I don't remember.. permit 0.0.0.0/0.0.0.0 is the "allow" for the ip acl |
| 15:34.43 | wdoekes2 | aha.. extensions_additional.. smells like #freepbx |
| 15:36.04 | aanev | wdoekes2: or maybe there is some variable that stores this did number |
| 15:36.07 | wdoekes2 | you're in #asterisk |
| 15:36.17 | wdoekes2 | you should move to somewhere where they know that stuff |
| 15:36.34 | aanev | sorry |
| 15:36.45 | aanev | my mistake |
| 15:39.28 | aanev | wdoekes2: so there is no variable that stores DID number? |
| 15:40.53 | wdoekes2 | there is a CALLERID(dnid), but I'm not sure if that is ever set |
| 15:41.36 | WIMPy | (freom earlier re dnid) Looking at the source it actually looks completely unused. |
| 15:41.37 | wdoekes2 | but for all we know, freepbx has some kind of ${ORIGINAL_EXTEN} |
| 15:43.27 | wdoekes2 | heh, tnx WIMPy :) |
| 15:48.48 | coppermine | where can ifind a howto to setup a local sip server |
| 15:48.55 | coppermine | so users on the lan can call eachother |
| 15:49.12 | WIMPy | You don't need one to do so. |
| 15:49.17 | WIMPy | ~newbook |
| 15:49.17 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 15:49.35 | WIMPy | That's good reading if you want to use Asterisk. |
| 15:55.55 | *** join/#asterisk Infested (Infested@24-148-117-190.ip.mhcable.com) |
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| 16:23.36 | *** part/#asterisk aanev (~aanev@79.126.229.50) |
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| 16:32.19 | *** part/#asterisk gerts (~gerts@ti0013a380-dhcp0255.bb.online.no) |
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| 16:52.25 | buzubada | For FreePBX, is disk encryption bad, or can I go ahead and ancrypt? |
| 16:55.39 | buzubada | *go ahead and encrypt? |
| 16:59.27 | wdoekes2 | => #freepbx |
| 16:59.51 | buzubada | grazie |
| 17:09.25 | saxa | hi, anybody can tell me how to park a call ? Is it just transferring it to 700 ? |
| 17:09.56 | saxa | what should I have in the extensions.conf under exten => 700,1,????? |
| 17:10.53 | saxa | also when the call is parked, is possible to call from the answering phone (sip) to another extension and ssay that the call is waiting for them at exten nr.700 ? |
| 17:11.27 | saxa | how exactly this thing work, I read it in the asterisk book, but could not understand it very well. |
| 17:11.48 | saxa | http://ofps.oreilly.com/titles/9780596517342/ch11.html |
| 17:17.20 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 17:19.54 | *** join/#asterisk jmd (~john@p54B196C4.dip0.t-ipconnect.de) |
| 17:20.18 | jmd | What is the licence of Asterisk? |
| 17:22.10 | saxa | i think its gpl |
| 17:22.19 | saxa | not sure tough |
| 17:28.15 | *** join/#asterisk MiserySoft (~LND@host81-139-80-30.in-addr.btopenworld.com) |
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| 18:00.07 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 18:13.35 | sawgood | In /etc/asterisk/rtp.conf (why have such a large range for UDP like 10000-20000) ... |
| 18:14.14 | *** join/#asterisk hairyraven (~nobody@95.72.117.20) |
| 18:14.15 | ChannelZ | why not? Change it if you want |
| 18:14.18 | cobra2 | sawgood: make a call and pop open wireshark and watch those ports |
| 18:15.15 | sawgood | Right ... I've done that ... and it is sort of 'random' from call to call; Asterisk box to Asterisk box, etc. |
| 18:17.13 | cobra2 | sawgood: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-D-SECT-37.html |
| 18:17.15 | sawgood | So, if I put a range of like 120001-12099, this is what Asterisk would use everytime? |
| 18:17.40 | cobra2 | sawgood: yup, it would |
| 18:18.11 | sawgood | perfect ... thank you! |
| 18:18.53 | cobra2 | i personally don't have many devices hooked up to my asterisk box so i restrict the ports down to a 2k range |
| 18:25.51 | *** join/#asterisk strehlow (~strehlow@24-247-41-88.dhcp.mrqt.mi.charter.com) |
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| 19:35.16 | Dovid | hello all |
| 19:41.00 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 20:01.51 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 20:10.24 | ChannelZ | ahoy |
| 20:23.01 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
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| 20:39.00 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 20:39.17 | Kobaz | hmm, i hate when fail2ban stops working |
| 20:50.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:58.47 | Kobaz | this is odd |
| 20:59.04 | Kobaz | make install for dahdi-2.4.1 doesn't put the kernel modules into /lib/modules |
| 21:04.44 | Kobaz | oh, never mind |
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| 21:28.01 | *** join/#asterisk coppermine (~inside@unaffiliated/coppermine) |
| 21:28.03 | coppermine | why would freepbx report there are 2 online ip phones when there is none? |
| 21:28.08 | coppermine | need to truncate the db table? |
| 21:30.05 | Kobaz | #freepbx |
| 21:30.25 | coppermine | thank you sir! |
| 21:30.27 | *** part/#asterisk coppermine (~inside@unaffiliated/coppermine) |
| 21:59.29 | *** join/#asterisk nate_h (ae3dbc42@gateway/web/freenode/ip.174.61.188.66) |
| 22:03.29 | *** join/#asterisk freetx (~freetx@99-190-181-112.lightspeed.sprntx.sbcglobal.net) |
| 22:03.38 | nate_h | are there any forums that have a user section for buying and selling voip hardware? I have a few items I wanted to try and sell and figured maybe someone here might know. |
| 22:04.24 | freetx | Have a strange issue, hoping someone can point me in the right direction. Playback or Background won't produce any sound on IAX2 trunk. |
| 22:04.40 | freetx | But does work if I call auto-attendant via SIP phone. Any ideas? |
| 22:07.12 | nate_h | NEW Cisco 4000 IP Soundstation w/ mics, USED polycom 7636 Conference Phone, NEW Polycom 330 IP soundpoint. |
| 22:07.19 | nate_h | guess i'll just google |
| 22:07.23 | Kobaz | freetx: make sure you do an Answer() before any playbacks or anything that produces audio |
| 22:08.47 | freetx | Yes, I do have an Answer, very simple dialplan....Answer(), Background(), WaitExten(5) --- doing a verbose connection I see it firing off in the stack, but its silent from IAX dialin....however, I can dial an extension and connect through.... |
| 22:09.59 | Kobaz | what asterisk version |
| 22:10.20 | cobra2 | freetx: stupid question: ports open? |
| 22:10.33 | freetx | version 1.6.2-5-0ubuntu1.3 |
| 22:10.47 | Kobaz | cobra2: iax2 uses one port, if he can make the call he should get audio |
| 22:11.06 | Kobaz | freetx: download the source directly from asterisk.org |
| 22:11.53 | cobra2 | Kobaz: hence the stupid question. I had a similar issue once. |
| 22:12.13 | Kobaz | cobra2: sip you can have those problems all the time, but if you can make the call, audio can pass as well |
| 22:12.19 | Kobaz | with iax anyway |
| 22:12.31 | freetx | Yes, I can actually talk fine once I connect through....only no Playback() or Background() function. Won't even play the Beep. But once connects through to SIP extension can talk fine. |
| 22:13.08 | Kobaz | don't know, try with a build off the official asterisk source |
| 22:13.40 | freetx | I will do compile from source. |
| 22:13.58 | Kobaz | because if you go and start digging for a problem, you don't know if you're trying to track down a configuration issue or a bug introduced from the debian maintainers |
| 22:15.05 | Kobaz | and you should be trying with the newest stable |
| 22:15.41 | freetx | Agreed. Will do. Should my config basically drop in between 1.6 and whatever the latest is? |
| 22:15.59 | freetx | Or is there big changes? |
| 22:16.33 | Kobaz | it'll be mostly the same, 1.8.x is the newest |
| 22:16.48 | Kobaz | but if you're starting out with really simple stuff it should be a drop in replcement |
| 22:17.38 | freetx | Yes, I just have a very basic dialplan with 2 SIP clients + IAX trunk working. So probably no worries. |
| 22:17.45 | *** join/#asterisk qjb (~qjb@a83-163-158-168.adsl.xs4all.nl) |
| 22:21.08 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:22.13 | nate_h | Anyone interested in buying some voip hardware, or know a forum I can post this stuff on? NEW Cisco 4000 IP Soundstation w/ mics, USED polycom 7636 Conference Phone, NEW Polycom 330 IP soundpoint. |
| 22:22.38 | Kobaz | you said that a few minutes ago |
| 22:24.32 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 22:36.31 | ChannelZ | what, you haven't changed your mind since then? |
| 22:39.27 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 22:39.28 | gray_ | hi |
| 22:40.13 | gray_ | I need logic help in creating an IVR that can store the ${EXTEN} and assign it to an actually EXTENSION |
| 22:41.16 | gray_ | so if I call SIP/100 and it asks me for my extension, I dial a number and I want it to take that extension and assign it to a SIP/200 address. So that if I hangup and call SIP/200 it dials that number that I stored through SIP/100 |
| 22:45.30 | *** join/#asterisk fauxalliance (~fauxallia@142.162.117.116) |
| 23:06.15 | Kobaz | you mean like Set(EXTENSION=${EXTEN}) |
| 23:11.07 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 23:14.25 | p3nguin | Sounds rather not well thought out. |
| 23:15.04 | Kobaz | gray_: are you trying to do some sort of call forwarding? |
| 23:15.38 | WIMPy | p3nguin: re hints, I don't think there's ant normal dialplan logic involved. |
| 23:16.46 | p3nguin | Maybe not, but I still don't see any reason variables shouldn't be allowed to work. |
| 23:17.59 | WIMPy | But if I got that right, you wanted to use a function? |
| 23:22.48 | *** join/#asterisk BarthezZ (~bart@ip51cd391a.speed.planet.nl) |
| 23:28.39 | p3nguin | My original intention was to use a function, but when it didn't work I then tested a standard variable. |
| 23:30.03 | WIMPy | Ok, and that didn't work, either? |
| 23:31.20 | p3nguin | Correct. I supplied the results last night. |
| 23:31.43 | *** join/#asterisk carrar (~tim@osburn.com) |
| 23:31.47 | carrar | WINNING! |
| 23:34.59 | *** join/#asterisk coppermine (~inside@unaffiliated/coppermine) |
| 23:35.05 | coppermine | which is the best trunk provider? |
| 23:35.09 | coppermine | cost/value |
| 23:35.22 | gray_ | Kobaz, if i set the extension to be global and set it to not clear global exten that works |
| 23:35.28 | gray_ | thank you |
| 23:35.43 | gray_ | I totally forgot that I could just use global variables |
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| 23:43.04 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 23:45.33 | *** join/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 23:46.00 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 23:46.18 | *** join/#asterisk CRCinAU_ (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 23:46.24 | CRCinAU_ | Does anyone here use a Cisco 7970 with asterisk? |
| 23:46.43 | CRCinAU_ | and also, why would I get this: |
| 23:46.43 | CRCinAU_ | #asterisk Cannot change nickname while banned on channel |
| 23:47.04 | CRCinAU_ | I also can't send anything under the CRCinAU nick - am I hitting something I don't know about? |
| 23:48.19 | CRCinAU_ | fwiw, it seems the 7970 is sending bad packets to asterisk - and I'm not sure which end is failing... |
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