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| 00:20.32 | russellb | there are 97 files in the directory of sample configuration files |
| 00:20.34 | russellb | that's just insane |
| 00:21.39 | greezmunkey | makes it a bit daunting to say the least. |
| 00:22.14 | russellb | luckily you don't need most of them |
| 00:22.26 | russellb | most people will only use a handful of them |
| 00:22.30 | russellb | there's just a ton of optional features |
| 00:26.52 | thehar | Everyone have a great Turkey day. |
| 00:27.32 | greezmunkey | man this works [xxxx](template), this does not [xxxx] (template) - nice |
| 00:28.26 | greezmunkey | stupid space... |
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| 00:37.04 | yonahw | thehar: have yourself a great Turkey day as well |
| 00:39.00 | yonahw | I am dialing using Dial(${ARG1},10,wW) console shows phone is ringing but doesn't pass the ringing tone back to the caller. Is this to be expected? Wiki indicates that r is only to override normal tones and that ringing is passed as appropriate |
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| 00:53.52 | p3nguin | If you add the 'r' dial option, do you then hear ringing? |
| 00:54.23 | p3nguin | If you add the 'm' dial option, do you hear music? |
| 00:59.31 | p3nguin | yonahw: ^ |
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| 01:03.56 | yonahw | p3nguin: didn't try. I was working from home but going to go back to the office now that the wife came home. Will check it out in a few minutes and post back. |
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| 01:12.02 | beek | If you add the 's' dial option, do you then hear swearing? |
| 01:13.19 | drmessano | I added the 'p' option and my night got way better |
| 01:14.54 | beek | :D |
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| 01:20.08 | greezmunkey | I need to just walk away for a while... |
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| 01:36.37 | yonahw | so "s" produced no swearing and "p" didn't improve my night at all |
| 01:36.49 | yonahw | "r" produced nothing but "m" did play music |
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| 01:42.01 | p3nguin | If you hear music, then you know you have media flow. |
| 01:42.41 | yonahw | indeed media flow exists. if the call is answered there is two way audio it being an incoming call. |
| 01:42.58 | yonahw | I still haven't gotten around to troubleshooting my nat issues yet but I don't think that is the issue here |
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| 02:00.50 | greezmunkey | I have followed the book to set the system up. I have made some progress, but I can't run this problem down: chan_sip.c:21289 handle_request_invite: Call from '' to extension '1002' rejected because extension not found in context 'unauthenticated'. I placed my configs here: |
| 02:00.55 | greezmunkey | [#asterisk] not found in context 'unauthenticated'. I placed my configs here: |
| 02:00.56 | greezmunkey | oops |
| 02:01.04 | greezmunkey | pastebin: |
| 02:01.04 | greezmunkey | [#asterisk] not found in context 'unauthenticated'. I placed my configs here: |
| 02:01.29 | greezmunkey | ...http://pastebin.com/9FzcfCc4 |
| 02:02.07 | yonahw | greezmunkey: where are you dialing from? what device? |
| 02:02.45 | yonahw | greezmunkey: you set in sip.conf to be in context unauthenticated but you didn't create the context in extensions.conf |
| 02:03.24 | greezmunkey | yonahw: I am setting up a test system. I have the asterisk server connected to an access point, and have two spectralink 8020 phones. The phones register, firwall ports are ok |
| 02:04.05 | yonahw | greezmunkey: see my second comment, it definitely won't work without fixing that |
| 02:04.22 | greezmunkey | yonahw: so the line context=unauthenticated needs to go? |
| 02:05.22 | yonahw | well based on your extensions.conf you should probably change it to "context=LocalSets" without the quotes of course |
| 02:06.02 | greezmunkey | yonahw: testing now... |
| 02:11.59 | greezmunkey | yonahw: hmm, I don't get the bad auth message now. I get this instead, which is where I was hours ago: chan_sip.c:21289 handle_request_invite: Call from '' to extension '1002' rejected because extension not found in context 'unauthenticated'. Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) |
| 02:12.38 | p3nguin | You assign a context for every peer. The context you assign needs to exist. That's where calls start. |
| 02:13.16 | p3nguin | You're calling 1002, but extension 1002 does not exist in the unauthenticated context. |
| 02:13.48 | greezmunkey | p3nguin: so the context is not set in the [brackets]? I assumed it was. |
| 02:14.00 | p3nguin | Where? |
| 02:14.22 | greezmunkey | p3nguin: in sip.conf |
| 02:14.25 | yonahw | what does sip show peers say |
| 02:14.39 | greezmunkey | p3nguin: I'll pastebin that. |
| 02:14.48 | p3nguin | The info in the brackets in sip.conf is the name of the device. |
| 02:14.59 | p3nguin | [phil] for example |
| 02:15.06 | yonahw | if you are calling from phone01 and it is setup properly it should already be in context LocalSets which has an extension 1002 |
| 02:15.13 | p3nguin | You would call that device by Dial(SIP/phil) |
| 02:15.23 | greezmunkey | here is is: http://pastebin.com/g0q2K2D0 |
| 02:15.58 | p3nguin | You have no devices associated with those two peer entries. |
| 02:16.26 | p3nguin | Let's start at the beginning. Okay? |
| 02:16.38 | greezmunkey | p3nguin: that would be great |
| 02:16.43 | p3nguin | Choose a name for a phone. Make it something sensible. |
| 02:17.02 | greezmunkey | p3nguin: phoneA |
| 02:17.08 | p3nguin | In sip.conf |
| 02:17.15 | p3nguin | [phoneA] |
| 02:17.21 | p3nguin | type=peer |
| 02:17.28 | p3nguin | wait, let me put it in pastebin. |
| 02:17.37 | greezmunkey | p3 thanks |
| 02:18.02 | greezmunkey | There is a logical jump here I guess I am not getting straight |
| 02:19.13 | p3nguin | What will be the extension number used to reach phoneA? |
| 02:19.24 | greezmunkey | p3nguin: 1001 |
| 02:19.31 | yonahw | what is the cause of a phone which registers with asterisk and can make calls but doesn't show up in sip show peers and can't receive calls? |
| 02:20.13 | p3nguin | http://pastebin.com/DBefZXvB |
| 02:22.04 | greezmunkey | yonahw: I ran across that in the book, appendix B, section Asterisk ad VoIP |
| 02:24.27 | drmessano | yonahw, it's not registered. It can auth to asterisk and make calls, but is not regged |
| 02:24.49 | yonahw | drmessano: got it. phone thinks its registered but clearly it isn't |
| 02:25.02 | *** join/#asterisk okado (~kotoshi@unaffiliated/okado) |
| 02:25.03 | p3nguin | Phones need not be registered to make calls. |
| 02:25.36 | greezmunkey | p3nguin: and that ties back to the extensions file as exten => 1001,1,Dial(SIP/PhoneA), right? |
| 02:26.03 | p3nguin | exactly |
| 02:26.43 | greezmunkey | p3nguin: ok, I'll incorporate those changes and see what it does. |
| 02:27.11 | p3nguin | But before we get there, note that we need to build an hierarchical dial plan so that phones can access various contexts, but not all contexts have access to each other. |
| 02:27.28 | greezmunkey | p3nguin: oh... |
| 02:27.38 | p3nguin | So I chose contexts=phones ... |
| 02:27.59 | p3nguin | [phones] will include other contexts that phones should be allowed to access. |
| 02:28.15 | greezmunkey | p3nguin: making notes... |
| 02:28.58 | p3nguin | perhaps an internal context where phones' extensions will be, and an outbound context where extension patterns for dialing out through a telco will be, and possibly some others. |
| 02:29.09 | okado | i am having prob connecting to asterisk with client |
| 02:29.13 | okado | here's is my config |
| 02:29.14 | okado | http://pastebin.com/qtTNZU4v |
| 02:29.25 | greezmunkey | p3nguin: ok |
| 02:29.55 | okado | http://pastebin.com/H8zqVxC7 |
| 02:30.03 | okado | sorry this is the new one |
| 02:30.18 | p3nguin | Then you will want to have an inbound context where calls will be routed from the telco. |
| 02:30.44 | p3nguin | That inbound context will very likely include your internal context, but never the context that allows calling outbound. |
| 02:32.18 | greezmunkey | p3nguin: are you going to pastebin this? |
| 02:32.38 | p3nguin | Working on an example. |
| 02:32.49 | greezmunkey | p3nguin: ty |
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| 02:35.21 | okado | greezmunkey, can you look at my config please? |
| 02:35.42 | p3nguin | http://pastebin.com/hJVFKVvi |
| 02:36.20 | p3nguin | oops |
| 02:36.53 | p3nguin | http://pastebin.com/fjeqUzNa bug fix |
| 02:37.57 | okado | p3nguin, is that ur extension.conf? |
| 02:38.02 | p3nguin | no |
| 02:38.22 | p3nguin | mine = -= 293 extensions (984 priorities) in 66 contexts. =- |
| 02:39.02 | okado | mine = blank? |
| 02:40.23 | okado | p3nguin, this is what i have |
| 02:40.28 | okado | http://pastebin.com/H8zqVxC7 |
| 02:40.50 | okado | i am not able to connect client (softphone) to asterisk |
| 02:41.12 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 02:41.17 | boodu | hello |
| 02:41.38 | p3nguin | greezmunkey: Are you with me on my example? |
| 02:42.37 | greezmunkey | p3nguin: yes, looking for the reason for your bug fix, it was subtle, no? |
| 02:42.38 | p3nguin | greezmunkey: Based on my sip.conf entry I gave you, program the phone with UID of "phoneA" and a password of "password" and see if it will register. |
| 02:43.00 | p3nguin | Yes, I fouled up a priority on the extension in the inbound context. |
| 02:43.38 | p3nguin | It went 1, n, 1, but should have been 1, n, n. |
| 02:43.39 | greezmunkey | p3nguin: I see it. 1, n, 1 to 1, n, n |
| 02:44.21 | p3nguin | I sometimes paste before double-checking my work. |
| 02:45.26 | greezmunkey | p3nguin: programming the phones is a pita, my options are username and password. I used 1001:1001 on one phone, and 1002:1002 on the other. Can I use those in your example instead? |
| 02:45.46 | greezmunkey | p3nguin: sometimes I double paste an improper copy! |
| 02:46.06 | p3nguin | If you use my entry for sip.conf, change username to phoneA and password to password. |
| 02:46.26 | greezmunkey | p3nguin: ok, I'll do it... |
| 02:46.42 | p3nguin | Make sure you run sip reload after making the changes to sip.conf. |
| 02:47.43 | greezmunkey | p3nguin: I will also have to add entries for the second phone, ie phoneB, etc etc. |
| 02:47.53 | greezmunkey | working... |
| 02:47.53 | p3nguin | sip reload for sip.conf, dialplan reload for extensions.conf |
| 02:48.05 | greezmunkey | p3nguin: ;) |
| 02:48.43 | p3nguin | You might be surprised how many people restart asterisk entirely for each change. Some even reboot the computer! |
| 02:49.13 | greezmunkey | p3nguin: I did get something out of reading...! |
| 02:49.38 | p3nguin | Do you have an ITSP already? |
| 02:50.37 | greezmunkey | p3nguin: no, I have an audiocodes MP-118 to connect to, but I'm doing one thing at a time. So, make two phones call eachother, then worry about a gate. |
| 02:51.09 | p3nguin | If you can get the phones to register and authenticate, the rest is cake. |
| 02:52.48 | greezmunkey | p3nguin: have a look at this, I have a question about the general section: http://pastebin.com/UZN9nbJE |
| 02:53.12 | greezmunkey | p3nguin: the context= line should read?? |
| 02:53.22 | greezmunkey | phones? |
| 02:53.38 | p3nguin | If you're going to use my example dialplan, yes phones. |
| 02:53.48 | greezmunkey | p3nguin: working |
| 02:54.05 | yonahw | so I am back to tackling my nat issues. I am confused partially because * box is both outside of and behind nat depending on the interface. Incoming calls are coming in on the edge nic and working perfectly. Outbound calls are originating from the nic behind nat seeing as they are coming from the local network but being sent out to the world on the edge nic presumably since that is the default route and also the only ip calls will be accepted from. |
| 02:54.14 | p3nguin | What the context line does is tell Asterisk where the call starts when it comes from that device. |
| 02:55.36 | greezmunkey | p3nguin: ok, here is the dialplan: http://pastebin.com/Bz7EWTHY |
| 02:55.42 | okado | p3nguin, which sip.conf and extensions.conf did you give to greezmunkey ? |
| 02:55.53 | okado | i can not get phones to authenticate |
| 02:56.43 | greezmunkey | p3nguin: programming phones... |
| 02:57.38 | p3nguin | greezmunkey: That should allow dialing 1001 to reach phoneA and dialing 1002 to reach phoneB. Each one will go to voicemail after 30 seconds of not answering the ringing phone. At that point, the call will fall apart because we never configured voicemail.conf. :) |
| 02:58.12 | p3nguin | okado: I gave greezmunkey the sip.conf and extensions.conf that I thought up in my brain and typed out so greezmunkey would have an example to work off of. |
| 02:58.36 | okado | i came here late. i only saw part of the conv |
| 02:58.38 | p3nguin | very basic examples, nothing fancy. |
| 02:59.37 | okado | my soft phone client won't connect to asterisk |
| 03:00.30 | p3nguin | I guess it ran out of magic. |
| 03:00.33 | okado | the commands for asterisk 1.4 and 1.6 is different |
| 03:00.48 | p3nguin | "commands" huh? |
| 03:00.55 | p3nguin | What "commands" are you referring to? |
| 03:01.03 | okado | http://pastebin.com/H8zqVxC7 |
| 03:01.14 | okado | to shut down asterisk |
| 03:01.21 | okado | core stop now |
| 03:01.36 | okado | vs older is something stop now without the 'core' word |
| 03:01.41 | greezmunkey | p3nguin: need to modify the tftp files, brb |
| 03:02.27 | p3nguin | I prefer to not shut down asterisk. It doesn't process calls very well if it isn't running. |
| 03:03.27 | okado | i need to stop it to run -cvvv |
| 03:04.00 | p3nguin | Why do you need to run it that way? Why can't you do core set verbose 3 and get the same result? |
| 03:06.09 | greezmunkey | p3nguin: sip_PhoneA.cfg : http://pastebin.com/CyqMQQC0 |
| 03:07.38 | p3nguin | I know nothing about the phone configuration. |
| 03:08.22 | greezmunkey | p3nguin: ok, I had to fix PhoneA anyway. done. restarting the phones. |
| 03:08.38 | p3nguin | Make sure you know which field does what and use the right info... and it should work. |
| 03:09.29 | greezmunkey | p3nguin: the auth line is simple "user:secret", so since we changed username from 1001 to PhoneA, I reflected those changes. |
| 03:09.46 | greezmunkey | p3 I now have PhoneA, and PhoneB !! |
| 03:10.04 | p3nguin | I hope you made the passwords match between asterisk and phone. |
| 03:10.09 | greezmunkey | p3nguin: I have wireshark running, if necessary |
| 03:10.16 | greezmunkey | p3nguin: ;) |
| 03:10.32 | p3nguin | My example used "password" as the secret, but your phone looks like it uses 1001 as the secret. |
| 03:10.37 | greezmunkey | p3nguin: they look regestered, I'll chec sip show peers |
| 03:11.33 | greezmunkey | p3nguin: sip show peers: http://pastebin.com/15JmeTWQ |
| 03:11.51 | greezmunkey | p3nguin: looks eerily similar to what I had before... |
| 03:12.20 | *** join/#asterisk coppice (~chatzilla@113.160.224.49) |
| 03:12.47 | p3nguin | Are the phones even trying to register? |
| 03:12.56 | greezmunkey | p3nguin: shouldn't the ip address of * show up? |
| 03:13.05 | p3nguin | where? |
| 03:13.13 | greezmunkey | p3nguin: the phones look registered on their displays |
| 03:13.35 | greezmunkey | p3nguin: in sip show peers, where it show (unspecified) |
| 03:13.52 | p3nguin | That should show the IP address of the peer. |
| 03:14.03 | p3nguin | the phone |
| 03:14.40 | greezmunkey | p3nguin: I agree. This has been an issue from the start. I think that maybe my access pint is causing an issue. |
| 03:14.44 | p3nguin | Does wireshark show there was a SIP registration attempt? |
| 03:14.47 | greezmunkey | s/pint/point |
| 03:15.02 | greezmunkey | p3nguin: I'll run that...brb |
| 03:17.14 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 03:17.15 | p3nguin | sip debug should also show registration attempts. |
| 03:19.26 | greezmunkey | p3nguin: looking there next, wireshark shows traffic between the AP and the *, but there is registration traffic... |
| 03:19.40 | p3nguin | The phones are wireless phones? |
| 03:20.41 | greezmunkey | p3nguin: yes, wireless - here is a sample of the cli output: http://pastebin.com/4fyFjmfS |
| 03:21.20 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:22.01 | p3nguin | What is the IP address of your Asterisk system? |
| 03:22.16 | greezmunkey | p3nguin: X.X.11.10 |
| 03:22.49 | p3nguin | So the registr packets look fine. |
| 03:22.58 | p3nguin | s/registr/register/ |
| 03:23.04 | greezmunkey | p3nguin: subnet 192.168.11, *=.10, AP=.200, PhoneA=.21, PhoneB=.22 |
| 03:23.51 | yonahw | happy to report that the source of my nat issues was a simple canreinvite=no. thought I had that in there but on closer inspection I did not. |
| 03:23.52 | greezmunkey | p3nguin: Here is the cli output from a call attempt from PhoneA to PhoneB: http://pastebin.com/sbD7ES5q |
| 03:24.10 | greezmunkey | yonahw: ftw |
| 03:24.16 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 03:24.51 | greezmunkey | p3nguin: I just don't get it... |
| 03:25.03 | p3nguin | Until sip show peers shows an IP address for both phones, you can't call to them. |
| 03:25.18 | greezmunkey | p3nguin: fair enough... |
| 03:25.26 | p3nguin | I don't know why it isn't registering properly since the REGISTER is arriving at Asterisk. |
| 03:26.19 | p3nguin | I guess you could set the host statically in sip.conf and disable the phone from registering. |
| 03:26.26 | greezmunkey | p3nguin: Perhaps because i statically defined everything. maybe it would behave better if I set up dns, and dhcp... |
| 03:26.39 | greezmunkey | p3nguin: really... |
| 03:27.09 | p3nguin | Disable the phone from sending registration, set host=192.168.11.21 and .22 respectively. |
| 03:27.16 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 03:27.31 | greezmunkey | p3nguin: I have this line in the general section: udpbindaddr=192.168.11.10:5060 |
| 03:27.48 | p3nguin | That's good. |
| 03:27.59 | yonahw | so now that I have solved my nat issue I am revisiting my lack of ringing tone issue. I have pasted the relevant bits from the dailplan here http://pastebin.com/YE3ua7ZF |
| 03:28.05 | p3nguin | Without it, asterisk may not know what IP address and port to listen on. |
| 03:28.22 | p3nguin | Well, you could use 0.0.0.0 for the IP address. |
| 03:28.29 | p3nguin | I probably would have. |
| 03:28.41 | greezmunkey | p3nguin: the host= line should go under the [PhoneA], and [PhoneB] sections?? |
| 03:29.09 | greezmunkey | p3nguin: I have three NICs here, so was defining it incase i used the others in the future. |
| 03:29.18 | p3nguin | You should already have host= in both of them. Change each from dynamic to the appropriate IP address for each phone. |
| 03:29.39 | greezmunkey | p3nguin: ah, ok... |
| 03:29.50 | p3nguin | dynamic is for peers which register. |
| 03:30.41 | greezmunkey | p3nguin: I'm not quite sure what to do to keep the phones from trying to register, will it matter if they do? |
| 03:31.25 | p3nguin | It might. Asterisk doesn't like it when static peers send registrations. |
| 03:31.55 | *** join/#asterisk root52 (~root52@ip68-228-177-7.cl.ri.cox.net) |
| 03:33.26 | greezmunkey | p3nguin: I went ahead and reloaded sip...there was a flurry of activity that seems to have subsided now! |
| 03:33.57 | greezmunkey | resetting wireshark, and making a test call, brb |
| 03:35.24 | root52 | Question. I want to set a channel custom variable on a SIP channel. (shared?) That part is no problem. I want to then do something in another channel based on weather or not that channel var is set and the same? So I guess I am having trouble understanding how to implement the "shared" channel vars. Does that question make sense? |
| 03:37.39 | yonahw | so apparently my other problem was that I was answering the call instead of just forwarding to the correct context |
| 03:39.01 | p3nguin | "forwarding" to a context? |
| 03:39.04 | p3nguin | Goto() ? |
| 03:39.56 | yonahw | yes goto rather than forwarding |
| 03:41.23 | okado | where is a list of asterisk functions? |
| 03:41.43 | yonahw | well folks as much fun as its been sitting here at work until 10:30 pm, I think I'm going to take off |
| 03:41.59 | greezmunkey | p3nguin: Interesting, I dialed 1002, from PhoneA, here is the result: http://pastebin.com/0Fk2qYYP |
| 03:42.05 | greezmunkey | yonahw: g'night |
| 03:42.13 | yonahw | good night |
| 03:42.19 | yonahw | good luck with your setup |
| 03:42.23 | WIMPy | okado: core show functions |
| 03:43.08 | p3nguin | No matching peer for 'PhoneA' from '192.168.11.200:1066' |
| 03:43.17 | p3nguin | So your AP is breaking SIP. |
| 03:44.02 | okado | WIMPy, thx |
| 03:44.37 | greezmunkey | p3nguin: I've been wondering that for hours now...I've checked the configs and they look right. It's running dd-wrt. |
| 03:44.59 | greezmunkey | p3nguin: I'm wondering if it's natting for some reason. |
| 03:46.03 | p3nguin | I doubt it's THAT intelligent that it can nat within the same subnet. |
| 03:46.34 | greezmunkey | p3nguin: I know, but I've been questioning everything since I started this! |
| 03:46.50 | p3nguin | The phones don't have Ethernet on them? |
| 03:47.45 | greezmunkey | p3nguin: so, I'm not dense, or stupid. What I had configured before probably would have worked...The phones, unfortunately. I can probably rig up a couple of softphones pretty quick though. |
| 03:48.09 | greezmunkey | unfortunately, no I should have typed... |
| 03:48.50 | p3nguin | This is just one more case of wifi sip gone bad. |
| 03:49.21 | greezmunkey | p3nguin: yeah, well the project that this was to lead to in the near future *is* wifi sip! |
| 03:49.44 | p3nguin | I'd say you may as well scrap it now. |
| 03:50.50 | greezmunkey | I've been having issues with this AP from the start apparantly. The project will be using *qualified* gear, or so they say. Unfortuneatly I don't have access to the big boy right now. |
| 03:51.56 | *** join/#asterisk coppice (~chatzilla@113.160.224.49) |
| 03:52.18 | WIMPy | What happened? The IP in the SIP message got rewritten? |
| 03:52.45 | greezmunkey | p3nguin: I really apprieciate you working with me through this. It has been quite a learning experience. I will stick with it by setting up a couple of lappy's with softphones just to proove it works. I'll deal with the wireless later. |
| 03:53.12 | greezmunkey | p3nguin: you're the bomb |
| 03:53.14 | p3nguin | At least you now know how to set up a basic phone switch. |
| 03:53.26 | greezmunkey | p3nguin: it has been awesome, ty |
| 03:53.54 | WIMPy | greezmunkey: What happened? The IP in the SIP message got rewritten? |
| 03:53.55 | greezmunkey | p3nguin: I've been wanting to do this for a while now, and here it is! |
| 03:54.57 | greezmunkey | WIMPy: yeah, it seems that the ap is natting on the same subnet as * and the phones. I dont see how, but I will find out. |
| 03:55.26 | p3nguin | It's doing something because it looks like it is routing rather than just passing traffic. |
| 03:55.30 | greezmunkey | I've got three lappy's here that I can use to test with |
| 03:55.56 | WIMPy | greezmunkey: rmmod nf_nat_sip |
| 03:56.27 | greezmunkey | WIMPy: eh? in the dd-wrt ap? |
| 03:56.35 | WIMPy | yes |
| 03:56.40 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 03:57.10 | greezmunkey | WIMPy: hmm, what can it hurt...I'll check... |
| 04:01.39 | p3nguin | Wouldn't it be something if it were that easy to fix? |
| 04:02.35 | greezmunkey | WIMPy: ok, I was able to run that in the command window. I also set it to be saved for use at startup, I'll try a call... |
| 04:02.54 | WIMPy | Ha |
| 04:02.57 | greezmunkey | WIMPy: or should I reregister the phones first? |
| 04:03.26 | p3nguin | If you didn't change host back to dynamic, registration is moot. |
| 04:03.37 | WIMPy | Erm, im not fully into your setup. Just got in here. |
| 04:03.54 | WIMPy | But you should be able to see that the IP doesn;t get changed any more. |
| 04:04.12 | greezmunkey | p3nguin: you are right, I'll reverse that since it was troubleshooting to begin with, brb |
| 04:05.32 | WIMPy | There still might be some natting going on. So registering might be the safe way. |
| 04:05.53 | *** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 04:06.11 | TehRabbitt | p3nguin: ya around? |
| 04:07.54 | WIMPy | BTW: The other option would have been to switch of nat support in Asterisk. That should work as well. But I advise against using nf_nat_sip. |
| 04:10.56 | greezmunkey | WIMPy: I have nat=no in the [PhoneA/B] sections...No love on the change btw, even after reregistering. I'm going to restart the AP, and try again. |
| 04:11.33 | greezmunkey | WIMPy: are you kidding! I just loaded the image, I didn't look to see what was there! ;) |
| 04:11.52 | greezmunkey | call me scriptmunkey |
| 04:12.03 | WIMPy | Err, what? |
| 04:12.54 | greezmunkey | WIMPy: I was kidding about the AP, like I just installed dd-wrt without knowing what it really does behind the scenes. |
| 04:12.59 | p3nguin | yawns |
| 04:13.11 | WIMPy | I suspect it will do nat. But if you can;t fugure how to disable that you could switch nat support on now. Not the right way, but it should work. |
| 04:13.27 | WIMPy | aye |
| 04:13.36 | TehRabbitt | not sure why this is happening, but I can't dial out from my SCCP phone that I just set up. |
| 04:13.47 | p3nguin | wrong context |
| 04:13.53 | TehRabbitt | ah |
| 04:14.03 | TehRabbitt | so change the context in sccp.conf? |
| 04:14.13 | p3nguin | yep |
| 04:14.45 | *** join/#asterisk costal (~ivan@corpnat.comindico.com.au) |
| 04:14.49 | costal | Hi |
| 04:15.42 | costal | I have a context on asterisk in that context I have 3 extensions with only a dial command |
| 04:15.46 | costal | exten => 0288100001,1,Dial(SIP/0288100001@10.61.52.55) |
| 04:16.19 | costal | but when I try to call another extension I got this error |
| 04:16.20 | costal | -- Got SIP response 482 "Loop Detected" back from 10.61.52.55 |
| 04:16.37 | costal | I've doing a bit of google |
| 04:17.10 | p3nguin | I guess extension 0288100001 in the context on 10.61.52.55 where your call landed probably sent the call back to you. |
| 04:17.13 | costal | I still haven't figured out how to stop the loop |
| 04:18.56 | *** join/#asterisk syntxerr (ae702326@gateway/web/freenode/ip.174.112.35.38) |
| 04:19.01 | syntxerr | hey all |
| 04:20.29 | *** join/#asterisk superm1 (~superm1@ubuntu/member/superm1) |
| 04:21.18 | syntxerr | I have a weird problem... I had my mailcmd set to ssmtp and it was working for a long time... I recently noticed I wasnt getting any email but have no idea why |
| 04:21.22 | superm1 | hey guys, in looking through res_jabber.c, it appears to potentially support 'invisible' mode if IKS_SHOW_INVISIBLE is defined. i checked through iksemel source and didn't even see any references to this. is there a special way to get invisible support working? |
| 04:21.40 | syntxerr | Ive tried outputting my mailcmd to a file but that isnt working - or so it appears |
| 04:25.47 | kaldemar | costal: 10.61.52.55 is the address of your asterisk box, right? |
| 04:27.14 | costal | hi |
| 04:27.23 | costal | yes 10.61.52.55 is my asterisk box |
| 04:27.26 | kaldemar | costal: if you want to dial a sip device that you have defined in sip.conf, don't use an ip address, just SIP/devicename, where devicename is something inside brackets in sip.conf. |
| 04:28.04 | costal | the problem that I have is I'm using realtime |
| 04:28.08 | costal | mysql sip_buddies |
| 04:28.17 | costal | all the sip users are there |
| 04:28.21 | costal | I can register in the asterisk box |
| 04:28.27 | costal | but I can't call |
| 04:28.46 | pabelanger | superm1: To my understanding no |
| 04:29.49 | p3nguin | It's no wonder that calling yourself results in a loop. |
| 04:31.14 | costal | haha you guys are awesome |
| 04:31.16 | superm1 | pabelanger, that's weird to have that dead code there then :( |
| 04:31.18 | costal | I found the problem |
| 04:32.20 | kaldemar | costal: no problem there. then you need a realtime switch in the context. switch => Realtime/... |
| 04:32.36 | pabelanger | superm1: ya, res_jabber.c needs some love. |
| 04:32.39 | pabelanger | patches welcome |
| 04:33.05 | kaldemar | costal: then again, if your extensions are not in a CB, disregard my last statement. |
| 04:33.14 | kaldemar | s/CB/DB/ |
| 04:34.37 | costal | kdaldemar thanks for your help I was wrong in using SIP/0290080687@ipaddress |
| 04:34.52 | costal | I removed the ipaddres as you advice and its working fine |
| 04:34.56 | costal | thanks dude appreciate it |
| 04:35.10 | superm1 | pabelanger, well if i get down to the bottom of why there was ever thought to be invisible support, i'll be glad to submit a patch to make it work. i'll keep digging then |
| 04:35.57 | superm1 | some random urls from google seem to show it in an iksemel source tree (http://www.koders.com/c/fidB35A848B8C5925C24BA0B299C4DAFB6E165B7279.aspx?s=search#L109) but it's not in the real upstream trunk |
| 04:38.32 | *** part/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 04:39.25 | *** join/#asterisk coppice (~chatzilla@113.160.224.49) |
| 04:41.25 | *** join/#asterisk Corydon76-dig (silver@c-69-137-80-31.hsd1.tn.comcast.net) |
| 04:41.25 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 04:43.03 | *** join/#asterisk wimt_ (wimt@freenode/staff/wikipedia.wimt) |
| 04:55.45 | greezmunkey | well, I'm pretty sure it's the AP at this point. p3nguin I wouldn't have come this far without you, thanks. I'll go about prooving the AP business tomorrow. I'll report back with what I find then. Happy Thanksgiving everyone. |
| 05:01.24 | *** part/#asterisk root52 (~root52@ip68-228-177-7.cl.ri.cox.net) |
| 05:01.43 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 05:14.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 05:16.13 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
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| 05:24.02 | *** part/#asterisk skyion (~bradc@siza.thusa.net) |
| 05:25.03 | *** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 05:25.32 | TehRabbitt | any way to get my asterisk install to let me reload the dialplan? it just says "dialplan reload" is not a valid option |
| 05:25.47 | TehRabbitt | No such command 'dialplan reload' (type 'help dialplan reload' for other possible commands) |
| 05:28.10 | TehRabbitt | any ideas? :-\ |
| 05:28.22 | *** join/#asterisk thansen (~thansen@74-47-186-12.dr01.hmdl.id.frontiernet.net) |
| 05:29.37 | drmessano | 1.4? |
| 05:31.04 | TehRabbitt | yes |
| 05:31.53 | TehRabbitt | i think i messed something up when I installed the chan_sccp_b driver because it did something with "autoadd dundi" or something |
| 05:33.44 | TehRabbitt | drmessano: any ideas how I can get it back to normal? |
| 05:33.57 | TehRabbitt | 1.4.30 |
| 05:34.16 | *** join/#asterisk Tachoh (459e59dc@gateway/web/freenode/ip.69.158.89.220) |
| 05:35.26 | p3nguin | module reload pbx_config.so |
| 05:36.30 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 05:36.36 | TehRabbitt | p3nguin: still nothing :-\ |
| 05:36.59 | TehRabbitt | [Nov 25 00:35:40] NOTICE[25700]: loader.c:685 ast_module_reload: The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized. |
| 05:37.14 | p3nguin | If dialplan isn't a valid command, then pbx_config.so is jacked up. |
| 05:37.21 | Tachoh | i am trying to program a tsp for windows in c#, can anyone direct me anywhere i can get more info on how to do this? TAPI enabled stuff is very abundant, but making a regular old COM tsp seems like it's a massive task. |
| 05:37.26 | TehRabbitt | so basically just recompile? |
| 05:37.31 | p3nguin | no |
| 05:37.37 | TehRabbitt | aight.. |
| 05:37.38 | p3nguin | You could restart asterisk. |
| 05:37.46 | TehRabbitt | i've tried it, multiple times |
| 05:38.58 | TehRabbitt | so i'm basically stuck :-\ |
| 05:39.38 | kaldemar | module unload pbx_config.so and then module load... |
| 05:39.46 | p3nguin | Installing chan_sccp.so involves compiling it and copying it to the modules directory. I don't see how that could break pbx_config.so. |
| 05:50.36 | *** join/#asterisk dee-kay-tee (~D@120.28.174.172) |
| 06:01.10 | TehRabbitt | p3nguin: well i ran some command afterwards following a guide on the site which i think might have set it to automaticlaly think it was realitime |
| 06:01.29 | TehRabbitt | http://chan-sccp-b.sourceforge.net/doc_setup.shtml |
| 06:02.28 | TehRabbitt | i don't have mysql installed |
| 06:02.30 | TehRabbitt | so it can't be setup |
| 06:02.49 | TehRabbitt | but i ran the original sccp.conf file with the following: |
| 06:02.59 | TehRabbitt | devicetable=sccpdevice ; datebasetable for devices linetable=sccpline ; datebasetable for lines |
| 06:03.12 | TehRabbitt | so idk if that did something automagically |
| 06:03.24 | p3nguin | The only file installed from chan_sccp is usr/lib/asterisk/modules/chan_sccp.so |
| 06:03.37 | TehRabbitt | this sadly works: server1*CLI> dialplan show sccpregistration |
| 06:03.54 | TehRabbitt | correction it doesn |
| 06:03.54 | TehRabbitt | 't |
| 06:04.18 | TehRabbitt | thoth*CLI> dialplan show sccp [ Context 'sccp' created by 'SCCP' ] |
| 06:04.37 | TehRabbitt | the phones should be registering to the "users" context |
| 06:04.41 | TehRabbitt | which is in the dialplan |
| 06:04.42 | p3nguin | Do you have an sccp context? I bet you don't. |
| 06:05.04 | p3nguin | So go change it in sccp.conf like I suggested multiple hours ago. |
| 06:05.14 | TehRabbitt | i did |
| 06:05.16 | TehRabbitt | to "users" |
| 06:05.37 | TehRabbitt | but it wont read the extensions.conf file and it seems like it wont change context either |
| 06:06.05 | TehRabbitt | http://pastebin.com/eLjnJxbu |
| 06:06.06 | p3nguin | I've tested chan_sccp on at least asterisk 1.4.36 and .37 ... |
| 06:06.14 | p3nguin | Did you consider an upgrade? |
| 06:06.24 | TehRabbitt | thats what i'm considering right now since i'm on 1.4.3 |
| 06:06.30 | TehRabbitt | .30 |
| 06:06.54 | p3nguin | Now you've created the sccp context anyway? |
| 06:07.15 | p3nguin | Nice workaround for a misaligned context setting. |
| 06:07.17 | TehRabbitt | I created it in extensions.conf but i can't reload dialplan so it's not seeing that sccp context exists |
| 06:07.52 | TehRabbitt | [users] should and i mean *should* be the default context for the phones |
| 06:07.57 | TehRabbitt | but it won't take the change |
| 06:07.58 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 06:08.08 | p3nguin | Normally, a change to the context setting in sccp.conf to the correct context, followed by reloading chan_sccp, would solve it. |
| 06:09.04 | TehRabbitt | http://pastebin.com/nLM010D3 |
| 06:10.07 | TehRabbitt | http://pastebin.com/VZ2FRkf2 |
| 06:14.07 | TehRabbitt | p3nguin: http://pastebin.com/uMSzWt31 |
| 06:14.18 | TehRabbitt | you might remeber that setup from like last winter heh |
| 06:19.05 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 06:21.20 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 06:27.31 | TehRabbitt | p3nguin: how can i remove the old copy of asterisk before installing the new copy? or just a new make install will overwrite it? |
| 06:29.16 | TehRabbitt | anyone here know how to remove a source-built install of asterisk? |
| 06:30.15 | costal | quit |
| 06:30.16 | costal | exit |
| 06:33.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ewueqbzkqrrlzbbc) |
| 06:35.45 | *** join/#asterisk dee-kay-tee-2 (~D@112.198.147.194) |
| 06:36.29 | *** join/#asterisk thansen (~thansen@74-47-186-12.dr01.hmdl.id.frontiernet.net) |
| 06:36.40 | kaldemar | TehRabbitt: make uninstall |
| 06:39.09 | TehRabbitt | ok so i'm upgraded to the latest version, i can now reload dialplan (thank god), but I still can't dial out / seem to be having issues with my context :-\ |
| 06:39.23 | TehRabbitt | context is set in extensions.conf right? |
| 06:41.29 | *** join/#asterisk dee-kay-tee (~D@112.198.146.56) |
| 06:47.10 | ChannelZ | well no, extensions are IN contexts |
| 06:47.23 | ChannelZ | channels/devices are assigned to contexts |
| 06:47.28 | TehRabbitt | ChannelZ: so where would I deifine the context? :-\ |
| 06:47.34 | TehRabbitt | which .conf file? |
| 06:47.36 | ChannelZ | (in sip.conf, iax.conf, or whatever the device might be) |
| 06:47.42 | kaldemar | TehRabbitt: depends what you mean by setting the contetxt. |
| 06:48.03 | TehRabbitt | I have X number of phones, I want them to all be able to dial eachother and also dial outside |
| 06:48.09 | *** join/#asterisk tish (~dna6@unaffiliated/tish) |
| 06:48.11 | TehRabbitt | I put them all as "context=sccp" |
| 06:48.21 | TehRabbitt | what do I need to put in extensions.conf for it to see that :-\ |
| 06:48.28 | TehRabbitt | just [sccp]? |
| 06:48.38 | ChannelZ | things under [sccp] will be in that context |
| 06:48.44 | kaldemar | then you need a context by the name of sccp in extensions.conf that has extensions for your phones. |
| 06:49.13 | TehRabbitt | [sccp] exten => 5000,1,Dial(SCCP/5000@11,45,m) exten => 5000,n,VoiceMail(4000@default) |
| 06:49.16 | TehRabbitt | so that should work in theory |
| 06:49.40 | ChannelZ | yes |
| 06:50.09 | *** join/#asterisk AndyRomano (~Adium@195.34.154.20) |
| 06:50.10 | ChannelZ | assuming whatever device tries to dial 5000 is in the sccp context |
| 06:50.21 | TehRabbitt | and if i assign the phones to the "[users]" context in extensions.conf, and "include" each other context, it should let me dial anywhere within the system, right? |
| 06:50.23 | AndyRomano | echo 'hello world' :) |
| 06:50.42 | ChannelZ | You assign phones to contexts IN THE DEVICES' CONFIG |
| 06:51.04 | ChannelZ | Groups of extensions belong to contexts as set in extensions.conf |
| 06:51.26 | TehRabbitt | Ohhh |
| 06:52.11 | TehRabbitt | so if i set the context in sccp.conf to "myContext", and then i put "[myContext" and include 3 other contexts, any devices within "myContext" should be able to dial the other 3 context's devices? |
| 06:52.36 | ChannelZ | well not exactly |
| 06:53.54 | ChannelZ | If you include one context from another in extensions.conf, the 'included' extensions sort of act as if they are in the 'included-in' context |
| 06:55.08 | ChannelZ | A device's context only tells asterisk where to start when that device dials something |
| 06:55.32 | TehRabbitt | so what would I be doing wrong that I can't dial ANYTHING including other SCCP extensions? |
| 06:55.44 | ChannelZ | An extension can be programmed to do anything to pretty much any device no matter what context it happens to be in |
| 06:56.36 | ChannelZ | ?? Your dialplan is mis- or non-configured I guess |
| 06:57.14 | ChannelZ | pastebin it |
| 06:59.07 | ChannelZ | God, MacOS's remote desktop/VNC/whatever they call it implementation is a piece of shit |
| 07:01.42 | *** join/#asterisk sourcode (~code@ppp-61-90-16-205.revip.asianet.co.th) |
| 07:01.48 | TehRabbitt | ChannelZ: http://pastebin.com/0xmeD9v1 |
| 07:01.53 | TehRabbitt | sorry it took a lil while |
| 07:02.46 | ChannelZ | ok and what exactly are you trying to do |
| 07:02.56 | ChannelZ | or what isn't working when you do what |
| 07:02.57 | TehRabbitt | I want to be able to place calls on my sccp phones |
| 07:03.07 | TehRabbitt | i pick up the handset, i get a dialtone |
| 07:03.14 | TehRabbitt | i type a number in, i get busy signal |
| 07:03.27 | TehRabbitt | get *a busy signal |
| 07:03.35 | ChannelZ | too basic.. what are you dialing |
| 07:03.58 | TehRabbitt | 5002, 5566, 5905, 411, 18005551212 |
| 07:04.04 | TehRabbitt | no matter what I dial, it does the same thing |
| 07:04.25 | ChannelZ | ok. Well you've said the device you're dialing from is set to the 'sccp' context in your config yes? |
| 07:04.34 | TehRabbitt | afaik Yes. |
| 07:04.44 | TehRabbitt | it doesn't say "context" it says "regcontext=" |
| 07:04.48 | ChannelZ | well you should know |
| 07:04.58 | TehRabbitt | and it does some sort of automagic "autoregistration" |
| 07:05.05 | ChannelZ | that's different |
| 07:06.09 | TehRabbitt | ChannelZ: http://pastebin.com/K6bgm8xt |
| 07:06.12 | TehRabbitt | theres my sccp.conf |
| 07:06.18 | TehRabbitt | perhaps i missed something in there |
| 07:06.38 | TehRabbitt | regcontext = sccp ; SCCP Lines will we added to this context in asterisk for Dundi lookup purposes |
| 07:07.20 | ChannelZ | well I see context=sccp in the [defaultline] macro. So which device are you dialing FROM? one of the 5000 ones? |
| 07:07.34 | TehRabbitt | Yes |
| 07:07.54 | TehRabbitt | Ah, maybe i missed that :-\ |
| 07:08.03 | ChannelZ | ok. Then based on your extensions.conf that device can only dial 5000, 5001, 5002, 5003, and 5004 |
| 07:08.42 | ChannelZ | If it needs to dial to the outside world, it's got to be able to get to the extensions in your 'outgoing' context for instance |
| 07:09.13 | TehRabbitt | or the "users" group would work? |
| 07:09.15 | ChannelZ | same for 'tollfree' or whatever. |
| 07:09.44 | ChannelZ | Yes, your users context includes everything else |
| 07:10.12 | TehRabbitt | i set that to "context=users" and still no change after a reload |
| 07:10.13 | TehRabbitt | :-\ |
| 07:10.47 | ChannelZ | show the console output of a failed call |
| 07:11.53 | TehRabbitt | ChannelZ: http://pastebin.com/mpgzQFyv |
| 07:12.17 | ChannelZ | you need to reload your sccp config, not the dialplan |
| 07:12.31 | TehRabbitt | reloaded that one too buffer just didnt' show it in my console |
| 07:13.31 | ChannelZ | sheesh this sccp outputs wierd crap |
| 07:13.40 | TehRabbitt | ChannelZ: http://pastebin.com/YRHRRKtZ |
| 07:14.07 | ChannelZ | is this with verbose turned on? To at least 3? |
| 07:14.48 | TehRabbitt | higher the number the more verbose, right? |
| 07:15.00 | ChannelZ | yes |
| 07:15.06 | TehRabbitt | 11 highest? |
| 07:15.14 | TehRabbitt | i'll set to 5 and try again hold on |
| 07:15.15 | ChannelZ | I have no idea but that's not necessary |
| 07:16.17 | TehRabbitt | http://pastebin.com/atkGsmyL |
| 07:16.47 | ChannelZ | not debug... verbose |
| 07:16.51 | ChannelZ | core set verbose 3 |
| 07:17.10 | TehRabbitt | what should i set debug to then |
| 07:17.11 | TehRabbitt | off? |
| 07:17.14 | ChannelZ | core set debug off |
| 07:18.32 | TehRabbitt | ChannelZ: http://pastebin.com/YbczxV3R |
| 07:19.33 | ChannelZ | hmm this doesn't look any different. I don't see anything even hitting the dialplan so I have no idea what is going on |
| 07:20.10 | ChannelZ | == SEP0015632CDAC0: Ending call 4 on line 5000 (InvalidNumber) |
| 07:20.13 | TehRabbitt | everything in the conf looks fine though right? |
| 07:20.44 | ChannelZ | That's the only clue I see but I don't know why it thinks it's invalid. This sccp channel doesn't output like other channels |
| 07:21.24 | ChannelZ | The only thing I can suggest is to make sure 5000 is in the 'users' context by specifically putting context=users under it's config |
| 07:21.34 | WIMPy | No, it checks the dialplan itself instead of just entering it. |
| 07:21.36 | TehRabbitt | won't work |
| 07:21.45 | TehRabbitt | http://chan-sccp-b.sourceforge.net/doc_setup.shtml |
| 07:21.48 | TehRabbitt | thats what I followed |
| 07:22.03 | ChannelZ | well I'm not going to read all that shit and I don't have any skinny devices to test with |
| 07:22.10 | WIMPy | TehRabbitt: Have you tried restarting Asterisk? |
| 07:22.20 | TehRabbitt | WIMPy: yes, AND reinstalling |
| 07:22.36 | TehRabbitt | and upgrading lol |
| 07:22.50 | WIMPy | I wouldn't trust reload to work. |
| 07:23.01 | ChannelZ | are there 'sccp' console commands or something that the channel driver provides? |
| 07:23.18 | TehRabbitt | some |
| 07:23.31 | ChannelZ | (for a SIP device I'd do "sip show peer Foo" for instance to look at Foo's config) |
| 07:23.48 | TehRabbitt | add, debug, dnd, message, no, reload, remove, reset, restart, show, system, unregister |
| 07:24.22 | TehRabbitt | sccp show: channels, device, devices, globals, line, lines, mwi, sessions, softkeyssets, version |
| 07:24.37 | ChannelZ | sccp show device 5000 maybe |
| 07:25.23 | TehRabbitt | http://pastebin.com/zfkBgQpU |
| 07:25.35 | TehRabbitt | it's a nice layout but doesn't say much to why it wont work |
| 07:27.38 | ChannelZ | well I don't see it mention the context anywhere, but maybe sccp doesn't. I think I'm at the end of my help since I don't even run SCCP |
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| 07:33.31 | schmidts | good morning |
| 07:35.12 | ChannelZ | Hidely-ho |
| 07:35.16 | TehRabbitt | anyway i'm off to bed, night guys |
| 07:35.31 | ChannelZ | night |
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| 08:42.26 | Hatrix76 | Hello, is there some documentation about the event system? I do not seem to be able to find it. Is it possible for an external application to registier to recieve events? I want to receive events of agents in a queue when they receive calls and when they finished the call so we can display Click2Call in our website more accurate without having to bombard the AMI |
| 08:44.22 | schmidts | does anybody use hylafax? i want to know if i can create a wildcard info file to set some parameters for a range of numbers |
| 08:58.19 | *** join/#asterisk John_Mendlov (~John_Mend@ip174-71-87-246.om.om.cox.net) |
| 08:59.04 | John_Mendlov | With asterisk can you have a branch system? |
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| 09:39.01 | _structz | Hi all! |
| 09:39.13 | _structz | Does anyone knows some patch that allow asterisk 1.6 to pickup more than 63 groups(callgroup/pickupgroup)? |
| 09:39.21 | WIMPy | Lo you! |
| 09:39.38 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 09:39.52 | WIMPy | has seen the question before, but not an answer. |
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| 10:18.27 | X-Raimo | hello. Does asterisk supports PBX delivers audio or something else |
| 10:20.10 | X-Raimo | the idea is our asterisk supports g729 codec. Most of our softphone doesn't have g729. We want to speak between softphones using g729. How does this possible? |
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| 10:31.45 | shamelessn00b | X-Raimo: asterisk can do transcoding |
| 10:39.41 | _structz | Does Asterisk 1.8 supports more then 63 callgroup/pickupgroup ? |
| 10:41.28 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 10:41.31 | sam_affable | hi |
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| 10:47.02 | X-Raimo | shamelessn00b: what require to make asterisk do transcoding? |
| 10:48.35 | sam_affable | hey im new to asterisk , is there a decent guide to compile and install asterisk 1.8 on debian lenny ? |
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| 10:52.04 | Gugge | sam_affable: i dont think so ... its the standard ./configure; make; make install way |
| 10:53.54 | sam_affable | Gugge: okie , i thought so ....just wanted to be sure ...if there'll be any dependency failures |
| 10:54.41 | sam_affable | Gugge: ty i'll go ahead and install , will post if there's any error |
| 10:54.51 | *** join/#asterisk telenieko (581adc52@gateway/web/freenode/ip.88.26.220.82) |
| 10:55.45 | telenieko | Hi ppl. In 1.4 what would be the SIP equivalent to a Dial string like that: IAX2/user:pass@host/extension ? (ie: SIP/user:pass@host/extension does not work) |
| 10:57.52 | X-Raimo | telenieko: Dial(SIP/extension,timeout) |
| 10:58.24 | telenieko | X-Raimo: I need to specify the host where extension lives. I look up a DB and get host + extension from there, where host is another asterisk box |
| 10:58.39 | telenieko | I am using IAX2 now (with the string above) but want a SIP equivalent for that |
| 10:59.01 | telenieko | user & pass are from a peer defined, but the host needs to be set there ;\ |
| 10:59.54 | X-Raimo | telenieko: i don't know, but we use Realtime for host definition |
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| 11:00.30 | telenieko | X-Raimo: I'll keep looking, thx! |
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| 11:18.23 | WIMPy | Can someone perhaps help me out with a snippet of pri debug using NFAS? |
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| 11:52.59 | zamba | how can i view asterisk variables through the console? |
| 11:53.16 | zamba | variables for the different channels |
| 11:53.43 | X-Raimo | zamba: connect to asterisk -r |
| 11:55.13 | Hatrix76 | <PROTECTED> |
| 11:55.29 | zamba | X-Raimo: and? |
| 11:56.06 | WIMPy | zamba: You can't. Except for what shows up in core show channel. |
| 11:56.16 | zamba | WIMPy: ok |
| 11:56.59 | zamba | how do i define the channel? |
| 11:57.06 | zamba | tried sip show channels and specify one of them, but no |
| 11:57.16 | zamba | (Call ID) |
| 11:57.29 | zamba | asty*CLI> core show channel 7fad4c251f7 |
| 11:57.29 | zamba | 7fad4c251f7 is not a known channel |
| 11:57.38 | WIMPy | core show channels |
| 11:58.06 | WIMPy | You can do sip show channel <sipid> |
| 11:59.09 | zamba | yeah, but i don't get the variables then |
| 11:59.30 | zamba | will i get the HANGUPCAUSE variable when issuing a agi-script from DeadAGI()? |
| 11:59.33 | WIMPy | No, only a selected few. |
| 11:59.59 | WIMPy | I would think so. |
| 12:07.46 | zamba | GET VARIABLE <var>, right? |
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| 12:26.33 | a_bug | Hi guys, a small question please - if I have a card with no h/w echo cancellation in it, will setting the tx gain and rx gain have an effect at all ? |
| 12:26.47 | a_bug | or does it have an effect only on cards with H/W DSP capabilities |
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| 12:29.11 | WIMPy | That's not related to hwec. |
| 12:30.19 | WIMPy | Oh, and I think you misunderstood the meaning of inband information. |
| 12:36.32 | *** part/#asterisk cHarNe2 (~cHarNe2@217-211-55-108-no79.tbcn.telia.com) |
| 12:45.22 | a_bug | oh |
| 12:45.50 | a_bug | but I read here that it is : http://lists.digium.com/pipermail/asterisk-users/2005-December/131620.html |
| 12:46.08 | a_bug | it says that most people need a rx gain of 8.0 db to have decent echo cancellation |
| 12:46.38 | a_bug | ya, I think I misundesratood the meaning on inband info :P Richard clarified it in his reply :( my bad |
| 12:46.41 | a_bug | thanks |
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| 13:02.17 | WIMPy | You shouldn't need/want to play with the gain unless using FXO/FXS. |
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| 13:20.26 | a_bug | oh |
| 13:20.36 | a_bug | ok thanks |
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| 14:01.58 | leifmadsen | December 2005 :) |
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| 14:11.31 | nzw | Hi all, i have problem with sending event: reboot to my cisco phone. It works if I Auth Resync-Reboot: No in my phone's configuration |
| 14:12.00 | nzw | do somebody have any advices ? |
| 14:13.16 | nzw | Anyone have any experiece with that feature? |
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| 14:35.19 | schmidts | nzw if i understood you right you dont want to set this phone option? |
| 14:36.44 | nzw | schmidts: yes |
| 14:37.27 | nzw | schmidts: i think i would be security problem when I set if to No |
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| 14:39.44 | schmidts | nzw if your phone is reachable from outside with an external IP, maybe. but if its behind nat i wouldnt be a problem |
| 14:41.19 | nzw | schmidts: you're right, but we use ~300 linksys ATA, and in some cases, they use external IP |
| 14:42.15 | schmidts | nzw maybe you should try it if a phone will reboot if this notify comes from another ip then this phone is registered to (a second asterisk for example) |
| 14:44.53 | nzw | schmidts: hmm. Good advice. |
| 14:45.58 | nzw | schmidts: i wrote a script, that sends notify to ATA's. It works, even with "Auth Resync-Reboot: Yes" |
| 14:46.14 | schmidts | nzw :D |
| 14:46.38 | nzw | schmidts: i think, that asterisk has some kind of bug |
| 14:47.13 | nzw | schmidts: I set more verbose logging on ata |
| 14:47.32 | nzw | schmidts: when i look in logs, i see, that ATA send Z jÄzyka:polskiâ¼Na jÄzyk:angielskiâ¼ |
| 14:47.54 | nzw | schmidts: challenge-notify* |
| 14:48.12 | nzw | schmidts: but no response from asterisk |
| 14:48.28 | nzw | schmidts: event where i set sip debug on on that peer |
| 14:48.39 | nzw | schmidts: when* |
| 14:48.51 | nzw | schmidts: sorry for my terrible english :) |
| 14:49.05 | schmidts | nzw the problem IMO is that asterisk doesnt have a "call" for this communication so its just a out of dialog message and will be ignored |
| 14:49.32 | schmidts | nzw i am from austria and my english isnt much better ;) |
| 14:51.11 | nzw | schmidts: you're telling me, that isn't some piece of code, that will generate response to "challenge-notify" ? |
| 14:51.46 | nzw | schmidts: did i understood you right? |
| 14:52.21 | schmidts | nzw thats one possible thing, but i think sending the notify to reboot is just a sip message which doesnt create a dialog in asterisk. this means asterisk will not resend it and even doesnt recognize any reply for this notify |
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| 14:54.26 | AdvoWork | any reason why id be getting this? Connect attempt from '127.0.0.1' unable to authenticate (im at the CLI (asterisk -rvvvv) |
| 14:58.16 | nzw | schmidts: thank you. i think that i should read some source code. maybe i'll find something interesting in it |
| 15:00.05 | nzw | thanks 4 help, bye |
| 15:00.09 | *** part/#asterisk nzw (~rychu@85.11.67.125) |
| 15:00.24 | schmidts | nzw: channels/chan_sip.c but i prefer doxygen: http://www.asterisk.org/developers/documentation |
| 15:00.32 | schmidts | np |
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| 16:04.45 | anonymouz666 | don't know what happens but wct4xxp from dahdi 2.3.0.1 and 5th generation cards doesn't look so stable as before :( |
| 16:05.03 | anonymouz666 | I'll contact the digium support |
| 16:06.39 | anonymouz666 | another strange case... os spans are green... just the last one is yellow. i did some loop tests and everything telling that the port isn't working fine. then I removed the card from on machine to another... and the loop just works. exactly the same config. |
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| 16:25.35 | root52 | Hi Can someone help me figure out what I am doing wrong here? I am trying to increment a var like this. exten => 2,n(retry),Set(RETRYCOUNT=$[${RETRYCOUNT} + 1]) but all I get is a syntax error complaining about the '+'. I have done this in the past just fine. I am doing this for the first time now on 1.8. |
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| 16:29.42 | root52 | [TK]D-Fender: Thanks!! as you pointed out I failed to initialize the var first. ::egg on my face:: |
| 16:42.30 | Hatrix76 | Can I bridge a call ringing in a queue to an agent via AMI? |
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| 17:22.00 | John_Mendlov | With asterisk can you have a branch system? |
| 17:23.44 | p3nguin | ~pbx |
| 17:23.44 | infobot | i guess pbx is a Private Branch eXchange |
| 17:23.58 | p3nguin | Asterisk allows you to build your own PBX. |
| 17:24.20 | p3nguin | So I guess the answer to your question is yes, you can have a branch system. |
| 17:24.24 | John_Mendlov | I mean Have the main asterisk system in a vps/ds and have one at each site |
| 17:25.00 | p3nguin | I meant that, too. |
| 17:25.18 | John_Mendlov | How do I do that, google had no help there |
| 17:26.10 | p3nguin | It's not something we can discuss within just a few sentences. Building a phone system with a main and several branch offices will take time and effort. |
| 17:26.38 | John_Mendlov | I was asking does anyone know of any sites that would have infomation |
| 17:26.44 | p3nguin | I don't even know your needs. Did you already design the system? |
| 17:28.11 | John_Mendlov | It will be all be over SIP I do not have any need for a pots line. I already have the hardware, I just wanted to know if it was possible to link tow outer asterisk boxes and the main uplink togeather |
| 17:28.46 | p3nguin | I would probably use IAX2 between the asterisk systems. That's what it was made for. |
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| 17:35.31 | DelphiWorld | hi guys |
| 17:35.42 | DelphiWorld | i have no clue using aculab with dahdi |
| 17:35.51 | DelphiWorld | is unable to by detected |
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| 17:37.04 | DelphiWorld | tzafrir: :P |
| 17:37.24 | tzafrir | Not well familiar with aculab |
| 17:37.51 | DelphiWorld | tzafrir: :P don't mean help me;) |
| 17:38.08 | DelphiWorld | tzafrir: we spock about it allready befaure, so i know allready hehehe;) |
| 17:38.46 | DelphiWorld | tzafrir: your dahdi help is allready apreciated |
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| 17:41.32 | p3nguin | Playback() won't play mp3 files. Is this normal? |
| 17:42.48 | heffer | if there's no mp3 codec yes |
| 17:43.22 | heffer | for performance reasons you might want to convert sound files to their respective native formats |
| 17:43.31 | heffer | depending on which codecs you use |
| 17:43.58 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:44.22 | pabelanger | p3nguin: format_mp3 installed? |
| 17:44.44 | p3nguin | /usr/lib/asterisk/modules/format_mp3.so |
| 17:44.46 | p3nguin | yes |
| 17:45.25 | p3nguin | If that's all that is required to Playback mp3 files, then maybe it's a permission problem. |
| 17:47.42 | p3nguin | file convert also can't convert the mp3 for me. That was my first idea. |
| 17:47.49 | p3nguin | heffer: ^ |
| 17:48.57 | heffer | p3nguin: i use sox to convert my files into slin16 wav first |
| 17:49.16 | heffer | i have a post on my blog for that here: http://fetzig.org/2010/07/24/converting-audio-files-to-g-722-for-use-with-asterisk/ |
| 17:49.28 | heffer | mostly as a note to myself :D |
| 17:52.19 | heffer | a big pro being that you have the files available in the best quality and natively for VoIP |
| 17:53.14 | p3nguin | The file does play fine through moh, but Playback chokes on it. |
| 17:58.56 | DelphiWorld | company? |
| 18:02.05 | neurosys | slaps DelphiWorld |
| 18:02.24 | p3nguin | slaps delphiworld with neurosys |
| 18:02.28 | DelphiWorld | slap neurosys |
| 18:02.30 | neurosys | hehe |
| 18:02.40 | DelphiWorld | slap p3nguin with digium |
| 18:04.27 | neurosys | malcolmd: You coming back to the ITEXPO East in 11'? |
| 18:06.37 | ruben23 | hi guys any familiar on this notice----->http://pastebin.com/d6QpbzTV |
| 18:08.48 | p3nguin | Are you trying to use MGCP? |
| 18:21.00 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 18:21.46 | *** join/#asterisk king313 (~king313@unaffiliated/king313) |
| 18:24.20 | king313 | Hi. I have several doubts about Asterisk. Can I take multiple classic analogic phone calls simultaneously using a traditional one line system of the public telephony system? |
| 18:24.50 | p3nguin | no |
| 18:25.07 | p3nguin | You'll take as many calls as the line has channels. |
| 18:25.13 | king313 | thanks p3nguin |
| 18:25.43 | p3nguin | If your line can ordinarily support only one call, hooking it to Asterisk will not increase the number of calls. |
| 18:26.24 | king313 | and the last one, to make a call from a traditional analogic phone (called A) to a VoIP phone (B), having B asterisk installed |
| 18:26.54 | king313 | its needed that the B system have both a telephone and internet access? |
| 18:27.21 | p3nguin | You don't have to have an internet connection to have both IP and analog phones connected to Asterisk. |
| 18:28.00 | king313 | I want to receive calls from a the traditional phone line |
| 18:28.18 | king313 | to a VoIP phone |
| 18:28.18 | p3nguin | Then you must have said phone line connected to your Asterisk system. |
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| 18:28.47 | king313 | yes, that is what I wanted to say |
| 18:29.09 | p3nguin | But you don't need an internet connection for that to happen. |
| 18:29.58 | p3nguin | If you do have an internet connection, though, you could send the call over a greater distance. |
| 18:30.04 | king313 | True, I can make asterisk handle the analogic line as well |
| 18:30.14 | king313 | thanks p3nguin, all doubt solved |
| 18:30.51 | p3nguin | What are your goals? |
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| 18:33.02 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.221.20.10) |
| 18:33.44 | merlin8282 | Hi all ! I'm back after almost 2 years without doing things with asterisk... |
| 18:33.54 | merlin8282 | Now I've a question (maybe more will come) |
| 18:34.19 | merlin8282 | I have to set up an asterisk server for a client, who has an ISDN line in germany. |
| 18:35.43 | merlin8282 | And I wonder what would be the best solution for him. I had set up a server internally that now works since 03/2009, it's bristuff-0.4.0-RC4-xr5. |
| 18:36.14 | merlin8282 | With a Junghanns QuadBRI ISDN PCI card. |
| 18:37.12 | merlin8282 | I see that there have been a couple of new versions of bristuff made by junghanns. Would it be a good choice to buy a junghanns card, or would you recommend something else ? |
| 18:38.49 | *** join/#asterisk tish (~dna6@unaffiliated/tish) |
| 18:43.16 | heffer | merlin8282: depends. if you are happy with the junghanns support then they're a good choice |
| 18:43.31 | heffer | but i'd go for hardware that is natively supported by dahdi |
| 18:43.41 | heffer | or has patches for it |
| 18:44.26 | merlin8282 | so hardware such as ? Is there a list or something ? |
| 18:44.30 | heffer | i think the junghanns devices are supported by dahdi |
| 18:45.22 | heffer | the quadrobri should be supported by qozap |
| 18:48.20 | *** join/#asterisk Mango (~iMango@209.121.85.243) |
| 18:49.16 | Mango | A phone connected to a PAP2T, behind NAT, registered to Asterisk, will always ring. |
| 18:49.29 | Mango | However, sometimes when the call is answered, there is no audio in both directions. |
| 18:49.38 | p3nguin | Yep. You have a NAT problem. |
| 18:49.52 | Mango | Really? I would have thought that since the phone rings, the NAT hole is open. |
| 18:50.09 | p3nguin | SIP isn't really affected by NAT... |
| 18:50.29 | p3nguin | SDP/RTP, on the other hand, break severely. |
| 18:50.43 | Mango | Ahhh. |
| 18:51.03 | merlin8282 | heffer: ok, thanks. |
| 18:51.27 | p3nguin | If you watch your sip debug, you'll see that the invites are good and the call starts just fine. You'll even see the right IP addresses. |
| 18:51.41 | p3nguin | If you watch an rtp debug, you'll see the wrong IP address. |
| 18:52.10 | Mango | Hrm, ok. |
| 18:52.36 | Mango | RTP debug on the Asterisk server? So you think it's sending to a 192.168.x.x address? |
| 18:52.53 | p3nguin | yes |
| 18:53.27 | Mango | would setting a STUN server on the PAP2T confirm that suspicion as well as watching the RTP debug? |
| 18:53.57 | p3nguin | not sure |
| 18:54.05 | Mango | ok, I'll give it a shot |
| 18:54.06 | Mango | Thanks :) |
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| 19:34.21 | flujan | hi guys, i am trying to build asteirsk 1.6 on a machine with postgresql installed. |
| 19:34.30 | flujan | asterisk is not compiling the engine pgsql |
| 19:35.39 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 19:36.59 | flujan | I am not finding the res_pgsql file too. |
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| 20:20.51 | boodu | hello |
| 20:22.34 | ironm | good evening - is there a source rpm for asterisk 1.8.0 or just the .spec file? - thank you in advance for any hints |
| 20:22.34 | boodu | I have a problem with a phone, a draytek vigorphone 350. The phone works but there problem with moh. It is silent when doing transfer or press hold button. |
| 20:22.44 | boodu | moh works with other phone |
| 20:23.38 | boodu | here sip log when hold button is pressed : http://pastebin.com/UfSrrZAq |
| 20:28.41 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-235-199.pools.arcor-ip.net) |
| 20:31.50 | boodu | I don't see anything could be source of the problem |
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| 20:40.53 | *** join/#asterisk ali533 (~ali533@2001:18e8:2:393:225:64ff:fe5c:ab4) |
| 20:41.17 | ali533 | i have a question about srtp |
| 20:42.30 | WIMPy | Can someone perhaps help me out with a snippet of pri debug using NFAS? |
| 20:42.54 | ali533 | anybody know to what extent srtp is supported and tested ? |
| 20:44.13 | WIMPy | ali533: If you have a Snom Phone, Asterisk will crash. |
| 20:45.36 | ali533 | k....what client can i use? SLEphone? |
| 20:45.46 | ali533 | s/SLE/SLF |
| 20:46.14 | ali533 | sorry mean SLFphone |
| 20:46.26 | ali533 | *SFL* jeez |
| 20:47.03 | ali533 | WIMPy: have you tried it your self |
| 20:47.19 | WIMPy | Yes |
| 20:48.35 | ali533 | WIMPy: anyother client you'd recommend ? |
| 20:48.59 | WIMPy | No idea. |
| 20:49.10 | WIMPy | That's the only one I have that supports srtp. |
| 20:50.41 | ali533 | did you use any proxy in between? opensips maybe ? |
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| 20:50.51 | WIMPy | no |
| 20:51.24 | ali533 | won't the key exchange be in the open? |
| 20:52.32 | WIMPy | It is, as usual. |
| 20:54.16 | ali533 | not good |
| 20:54.57 | ali533 | i guess i just need a softphone |
| 20:55.02 | WIMPy | There's no other way unless you transfer a static key before making a connection. |
| 20:55.10 | ali533 | yep....good idea |
| 20:56.13 | ali533 | so if i install asterisk from the source, is there anything more i need to configure to make a simple phone call? |
| 20:57.42 | WIMPy | You need at least a dialplan definig an extension and an account that lets you conect to it, or allow anonymous connections. |
| 20:57.47 | WIMPy | ~book |
| 20:57.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
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| 20:58.01 | WIMPy | That should be good reading. |
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| 21:22.11 | *** join/#asterisk Kate6 (~six@184-100-222-210.ptld.qwest.net) |
| 21:22.33 | Kate6 | Is there anyone around that I could ask for help with my dial plan from? |
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| 21:30.54 | *** join/#asterisk srini (~srini@219.91.201.74) |
| 21:31.01 | srini | Hi room |
| 21:31.18 | srini | Is there a backup script available for asterisk? |
| 21:39.32 | pabelanger | srini: bacula? |
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| 21:47.36 | Kate6 | Does anyone know if the new PITCH_SHIFT function only works on certain codecs? And if so, which ones? |
| 21:48.23 | WIMPy | I'm pretty sure it will always transcode to/from slin. |
| 21:49.15 | ali533 | WIMPy: thanks |
| 21:50.53 | Kate6 | slin? |
| 21:51.33 | WIMPy | signed linear |
| 21:52.23 | Kate6 | So I need this one? |
| 21:52.23 | Kate6 | ;load=format_sln.so ; Raw Signed Linear Audio support (SLN) |
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| 21:52.37 | Kate6 | *tries it* |
| 21:54.23 | Kate6 | Nope... Still not getting any audio on any call that has pitch shift enabled. |
| 21:54.48 | p3nguin | I have a backup script for asterisk, but you'd have to alter it to work on your media. I run it via cron.daily and it backs up to a connected backup media. |
| 21:55.01 | p3nguin | But he left, anyway, so nevermind. |
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| 22:22.08 | p3nguin | Huh. Our rain just switched to snow. |
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| 22:26.51 | jsjc | any freelancer for a very small project? |
| 22:48.15 | p3nguin | jsjc: What do you have going on? |
| 22:54.34 | pabelanger | jsjc: #asterisk-consultants |
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| 23:13.19 | robl^laptop | can anyone explain how you tell parkandannounce where to go after timeout? |
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