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| 01:46.08 | m0sphere | i have a customer who says hes experincing a "1-2 second delay" on all of his calls. i don't really believe him though. is there a way to monitor the latency on a channel or something like that? |
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| 02:22.05 | pabelanger | m0sphere: mixmonitor |
| 02:22.11 | pabelanger | just record the calls |
| 02:24.47 | m0sphere | can it record only from a specific cid, or do i need to record all calls and then listen to just his? |
| 02:27.02 | Penguin | It's your dial plan, so you can make it do whatever you want. |
| 02:27.43 | Penguin | I would consider something as easy as an ExecIf(). |
| 02:28.19 | m0sphere | i'm in canada, i believe recording personal conversations is against privacy laws |
| 02:29.20 | Penguin | Could be. |
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| 02:30.48 | Penguin | I am not a lawyer, but there could be provisions for the telcom engineer to do recording under certain conditions. |
| 02:31.10 | blitzrage | only need to provide notice of the recordings |
| 02:31.22 | m0sphere | hmm |
| 02:31.39 | Penguin | That will be weird on a personal/residential circuit. |
| 02:32.14 | blitzrage | :) |
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| 02:32.52 | m0sphere | with the core show channels verbose command, my channel column seems to have too many characters and cuts off the remainder of the channel, is there a way to increase the number of characters i can see? |
| 02:33.14 | Penguin | Try a different command. Maybe core show channels concise. |
| 02:35.01 | m0sphere | hm that one works, but it's not visually as nice, lol |
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| 04:09.00 | us_0gb | I have a really stupid question, but I have to ask, in case it's not as stupid as I think. If I install Asterisk on my server, is there a way to acquire a phone number to run my home phone through my server? |
| 04:16.32 | Penguin | ~itsp |
| 04:16.32 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 04:16.55 | Penguin | Phone numbers are only for getting calls to you. |
| 04:17.08 | Penguin | If you only want to make outgoing calls, you don't need a phone number. |
| 04:28.31 | us_0gb | Penguin: I've tried making outgoing calls without a number through a regular VoIP account, and the calls are denied. |
| 04:28.58 | Penguin | Your term "regular VoIP account" doesn't mean anything to me. |
| 04:29.57 | us_0gb | Okay, what I mean is that I tried calling out without a number. It did not work. Is there something special I need to do to make outgoing calls without a number? |
| 04:30.28 | Penguin | Both VoIP.ms and Flowroute will terminate your calls without buying a phone number. |
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| 04:31.18 | us_0gb | Okay, so does that mean I am unable to host my own outgoing call solution? |
| 04:32.32 | Penguin | No. You can buy all kinds of hardware and interface directly with the PSTN yourself. |
| 04:32.52 | Penguin | But that doesn't seem cost effective to me. A regular ITSP seems like a better idea. |
| 04:34.04 | us_0gb | Okay, so it would have to be done in hardware, not software. Thank you very much for your time and advice! I will go with a regular ITSP. |
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| 04:34.44 | Penguin | Several thousand dollars per month in circuits is also a deal breaker for me. |
| 04:35.08 | us_0gb | Per month? Oh, wow. Yeah, that is a bit too much. |
| 04:35.16 | Penguin | Both Flowroute and VoIP.ms will terminate your calls without buying a phone number, and they both have good per-minute rates. |
| 04:36.10 | Penguin | But remember, without phone number, you'll have to find some other way to receive phone calls from people. |
| 04:37.14 | Penguin | You can even use two different providers. You can get a phone number from one ITSP for cheap or free and pay for termination from another provider. I did that for years. |
| 04:39.33 | us_0gb | I actually have a temporary incoming call solution, which is gratis. I set it up to work with while I tried to get my own host up, but I could make it permanent now that that isn't happening. |
| 04:40.12 | us_0gb | I might still host my own SIP server for the incoming calls, and get a gratis number. |
| 04:40.59 | us_0gb | Hardly anyone calls me anyway. This is mostly for the people who refuse to use email. |
| 04:42.38 | Penguin | I used to use IPkall for some free phone numbers. As long as Washington state numbers aren't a problem, they get calls from the PSTN and send them to your asterisk. |
| 04:47.41 | us_0gb | I think that might be where I have my number from, actually. |
| 04:48.55 | Penguin | They work. |
| 04:49.00 | us_0gb | Oh, nice. It seems Asterisk is included in my system's repositories. I don't even have to compile it. |
| 04:49.35 | us_0gb | They work incoming, but as we discussed, not so much outgoing. |
| 04:50.10 | Penguin | With most ITSPs, you can set your outgoing caller ID number to your free IPkall phone number so that when you call someone they can see the number to use to call you back. |
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| 04:50.49 | Penguin | I haven't found any good free termination providers. You can get good ones or you can get a free one, but not a good free one. |
| 04:50.55 | us_0gb | Yeah, I better do that then. It would be more convenient for the receiver. |
| 04:51.11 | Penguin | I had a decent free one, but they ended their free services. |
| 04:52.41 | us_0gb | I don't need it to be gratis necessarily. I wanted to host it myself for independence mostly, not cost. Though at a rate of several thousand each month, I'll give up independence. |
| 04:53.08 | Penguin | You can be mostly independent by using the right ITSP. |
| 04:53.29 | Penguin | They'll give you a lot of control and then the only thing you need them for is the gateway between your VoIP system and the PSTN. |
| 04:54.26 | us_0gb | Before signing up, what do I watch for? |
| 04:55.34 | Penguin | I don't know if you have any unusual needs. VoIP.ms used to have a money back policy, where if you put some money into your account and found out you didn't like the service, they'll just refund the unused portion. |
| 04:56.02 | Penguin | I think they require a minimum of $25 per deposit. |
| 04:56.48 | Penguin | If you don't buy a phone number, and if you only use the service for outgoing calls at 1 cent per minute, that will last you a while. |
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| 04:58.29 | us_0gb | That would indeed. Most of my calls are about 15-30 seconds. Assuming voip.ms rounds all calls up to the nearest minute, that would be 2500 calls. |
| 04:59.32 | Penguin | They round to the nearest 6 seconds. |
| 04:59.46 | Penguin | well, round up to the nearest 6 seconds. |
| 05:00.00 | us_0gb | Oh, wow. That's a lot of calls! |
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| 05:00.43 | Penguin | So you don't even have to pay for a full minute if your call only lasts 6 seconds. |
| 05:01.11 | Penguin | The 6-second increment is fairly standard. |
| 05:03.21 | us_0gb | I'm pretty new to this. The head of the household canceled our land line a while back, and replaced it with a cell phone. |
| 05:03.27 | us_0gb | <PROTECTED> |
| 05:04.22 | us_0gb | So I'm choosing VoIP for my own phone. She can do whatever sh wants with the cell, and will not have a valid reason to try to make me have anything to do with it. |
| 05:04.26 | us_0gb | *she |
| 05:04.27 | Penguin | What will you use for phones? |
| 05:04.45 | us_0gb | A ThinkPad X60. |
| 05:04.57 | us_0gb | That, and Linphone. |
| 05:05.14 | Penguin | That'll do the trick. |
| 05:05.39 | Penguin | I haven't used Linphone that I can recall, but a lot of people have mentioned that they use it. |
| 05:06.05 | us_0gb | What do you use, if you don't mind me asking? |
| 05:06.58 | Penguin | With proper configuration, you may not even need asterisk. You can peer the soft phone directly to the ITSP. I'd have to look into how you can get your free IPkall number to the softphone, but it should be pretty easy as well. |
| 05:07.10 | Penguin | I use Cisco 7900 series phones. |
| 05:07.50 | Penguin | Or if I need a soft phone, I usually go for Zoiper for Windows computers or I have Twinkle on my Linux desktop. |
| 05:08.09 | Penguin | I use cSipSimple on my Android phone. |
| 05:08.23 | us_0gb | Oh, a hardware phone. I might get one of those one day after I move out. |
| 05:09.16 | us_0gb | IPKall routes to a SIP account, so that would need to be hosted somewhere, either on my server or through someone else. |
| 05:09.26 | Penguin | No. |
| 05:09.31 | us_0gb | No? |
| 05:09.32 | Penguin | It routes to a SIP URI. |
| 05:09.47 | us_0gb | Oh, interesting. The plot thickens. |
| 05:09.48 | Penguin | If your soft phone can receive the call, you're set. |
| 05:10.22 | us_0gb | I will defiantly look into that. Cutting out the SIP account would make setup simpler. |
| 05:10.59 | us_0gb | *definitely |
| 05:11.56 | Penguin | When I route an IPkall number to asterisk, I always use the URI of phonenumber@my.address, e.g., 4015551212@penguin.asterisk-box.net |
| 05:12.32 | Penguin | And then my asterisk box at penguin.asterisk-box.net has the appropriate extension 4015551212 configured to get calls from the IPkall hosts. |
| 05:15.17 | us_0gb | Oh, I think I misunderstood then. It sounds like now that the soft phone likely will not accept calls directly in most cases. |
| 05:17.13 | Penguin | It should. |
| 05:19.38 | us_0gb | Okay, I'll try to figure out how to set that up then. Actually ... I wonder if it would be as easy as simply routing it to my machine while the soft phone is running .... |
| 05:20.38 | us_0gb | So using <sip:0gb.us@thinkpadx60-0gb-us.0gb.us> as the SIP address. |
| 05:23.25 | Penguin | You'll have to determine if the extension (username portion of the SIP URI) is configurable on the phone. If it is not, it may be 's' by default. |
| 05:25.36 | us_0gb | Okay, I'll use 's' then. Why 's' though? Do you know why that would be the default? |
| 05:26.48 | Penguin | ATAs use s. I can only guess that soft phones do, too. |
| 05:26.54 | Penguin | ~s |
| 05:26.54 | infobot | When used with FXO ports (analog or T-1), "s" means "device too stupid to send number" |
| 05:30.50 | Penguin | It is probably configurable. |
| 05:31.17 | Penguin | But you will have to forward port 5060 UDP to the soft phone for that to work right. |
| 05:31.23 | Penguin | and the RTP range. |
| 05:32.14 | us_0gb | Yeah, I'm reading about that now. I'm trying to figure out what ports to forward. How do you know the RTP range? It seems to vary. |
| 05:32.43 | Penguin | There should be a setting in the phone for that range. |
| 05:33.31 | us_0gb | Okay, I did see that setting. But how will the incoming connection know what port to use in that case? |
| 05:33.42 | us_0gb | OH! Is that sent on port 5060? |
| 05:33.46 | Penguin | SDP |
| 05:34.31 | Penguin | The call (SIP) is sent on UDP 5060. The SDP will be "discussed" in SIP. |
| 05:35.13 | us_0gb | Okay, awesome.That will be much easier to set up than I feared. |
| 05:35.53 | Penguin | In asterisk, you just set the RTP range in rtp.conf, and then that is the range that asterisk says to use when setting up a call. |
| 05:36.22 | Penguin | The phones do the same type of thing. |
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| 10:51.26 | mjt | Hello. A first-time newbie user is here, trying to set up his first "hello world" PBX. It looks like it works, but so far I found only one soft phone which works with it - it is ekiga. I'm playing with android - csipsimple, linphone, sipdroid, native android 4.3 client - neither of those works, each differently. |
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| 10:52.34 | mjt | for example, linphone - when dialing 999 number (which should say "hello world"), says "disconnected", and the log on asterisk side says "channel.c:4257 __ast_read: Dropping incompatible voice frame on SIP/1000-0000000b of format ulaw since our native format has changed to (gsm)" |
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| 10:54.08 | mjt | with csipsimple i can hear the "hello world" words, but it never detects that the call is terminated, instead, the number of seconds since the call started keeps growing (with silence coming from the speaker), until I hit "hang up" on the app |
| 10:54.50 | mjt | since each client has its own issue, i'm not sure if the server actually works correctly |
| 10:55.28 | mjt | sipdroid can't register, it says "unable to register" even if the server logs says "registered <foo>" |
| 10:56.25 | mjt | native android sip client just hangs up without any additional information |
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| 10:58.55 | mjt | initially i tried to install freeswitch, it worked fine with all mentioned clients, but i had other issues with it, and it has been suggested to me to try asterisk instead, for a start. But with asterisk even the first step - basic client setup - doesn't work for me. |
| 10:59.48 | ChannelZ | are all these phones on the same LAN? |
| 11:00.04 | mjt | on the same android smartphone :) |
| 11:00.19 | ChannelZ | which is where in relation to asterisk |
| 11:00.40 | mjt | wifi network with very good connectivity |
| 11:01.07 | mjt | asterisk is running on the same machine as wifi access point |
| 11:01.18 | mjt | (which is my desktop) |
| 11:01.29 | ChannelZ | so they're on the same subnet? |
| 11:01.37 | mjt | yes. |
| 11:02.17 | ChannelZ | so we need to see some sip debug |
| 11:02.56 | ChannelZ | core set verbose 3 |
| 11:03.00 | ChannelZ | sip set debug on |
| 11:03.16 | ChannelZ | then try a registration |
| 11:03.17 | mjt | i'm already using these |
| 11:03.44 | agoodm | mjt, I suggest not using 999 as your test number as this is emergency services number in UK and possibly some clients may do special things because of this? |
| 11:04.32 | mjt | ok, i'll change it to 1111 |
| 11:04.40 | agoodm | use something like 1000 |
| 11:04.56 | mjt | 1001. 1000 is my test account :) |
| 11:05.18 | agoodm | 111 is the non emergency number for police here, I always make testing extensions in the 5000 range personally, 5001 5002 etc |
| 11:05.57 | agoodm | it shouldnt make any difference, but if it did and it was something that simple youd be kicking yourself! |
| 11:06.03 | mjt | here's linphone registration (after starting linphone): https://paste.debian.net/79624/ -- linphone client says "Unregistered" |
| 11:07.06 | mjt | damn. and i can't dial any number anymore using linphone. |
| 11:07.14 | agoodm | can you show your sip.conf entry please? |
| 11:07.32 | ChannelZ | well you either cut the debug off early, or didn't paste it all, or linphone is broken because it's not trying to auth properly |
| 11:07.35 | mjt | [1000] |
| 11:07.35 | mjt | type=friend |
| 11:07.35 | mjt | host=dynamic |
| 11:07.35 | mjt | secret=1000 |
| 11:07.47 | mjt | that's all :) |
| 11:08.06 | mjt | just started from the tutorial |
| 11:08.18 | mjt | ok, let's try something else. |
| 11:08.43 | mjt | csipsimple registered ok |
| 11:08.44 | agoodm | ill give you an example sip.conf entry I use in a live system, works with most phones ive tried |
| 11:09.17 | agoodm | http://pastebin.com/2jpCzZVB |
| 11:09.55 | mjt | the only notable difference from mine appears to be the codecs config |
| 11:10.25 | agoodm | directmedia=no will ensure that the sip server always try to remain in media path, avoids lots of nat/reinvite issues |
| 11:10.59 | agoodm | do you have anything else in your sip.conf ? a general section? |
| 11:11.03 | mjt | i see. Actually I'd try keep it the other way around, we don't have any NAT in the path |
| 11:11.45 | agoodm | i know, but there is nat in your network and when you try to progress to connecting this to the internet itll spoil your day |
| 11:12.36 | agoodm | but anyway; as ChannelZ suggests, your sip client didnt try to authenticate or you didnt get all the debug |
| 11:12.53 | mjt | what i want to use this thing for is to connect several branches of our firm together, all using internal network, for now there's no plans to connect to outside |
| 11:13.26 | mjt | heck. now linphone worked just fine |
| 11:13.28 | agoodm | i see |
| 11:14.16 | mjt | er. i'm mixing them all already. |
| 11:14.28 | mjt | csipsimple worked fine. linphone does not. |
| 11:15.00 | mjt | ..and finally linphone crashed :) |
| 11:15.02 | agoodm | to me it looks like linphone is attempting to make a call without registering |
| 11:15.10 | agoodm | at least from your debug log |
| 11:16.23 | mjt | ekiga crashed too |
| 11:16.25 | mjt | ;) |
| 11:16.34 | mjt | that's.. ecoraging :) |
| 11:16.53 | agoodm | i just noticed something thats glaringly missing from your sip.conf entry |
| 11:17.04 | agoodm | im fairly confident you NEED a context= line |
| 11:17.07 | mjt | context/ |
| 11:17.09 | mjt | eh |
| 11:17.22 | mjt | it is "public" by default (i learned it by trial and error :) |
| 11:17.27 | agoodm | it gives the phone a home location in extensions.conf for want of a better word |
| 11:17.33 | mjt | yeah |
| 11:18.08 | agoodm | dont use the public / default context for stuff authenticated phones need to do... if you ever connected that to the internet youd get toll fraud or random phone calls within minutes |
| 11:18.22 | agoodm | in your extensions.conf create [phones] |
| 11:18.34 | agoodm | then add all your extension logic below that |
| 11:18.36 | mjt | context=users, or =phones, etc |
| 11:18.37 | mjt | yeah |
| 11:18.54 | agoodm | then context=phones in your sip.conf entry and dialplan reload and sip reload |
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| 11:19.13 | agoodm | then try again with your testing starting with linphone |
| 11:20.22 | agoodm | I would also suggest in sip.conf making sure you have a [genera] section at the top, which as a minimum has a context=public line |
| 11:21.36 | mjt | sigh |
| 11:21.51 | mjt | retries getting the debug logs |
| 11:22.13 | agoodm | have you done what I suggested? |
| 11:22.20 | mjt | yes |
| 11:22.39 | agoodm | sip.conf first section should be [general] - I typoed |
| 11:22.57 | mjt | yes |
| 11:23.11 | agoodm | just nipping to grab a cuppa coffee, be back in less than 5 |
| 11:23.16 | mjt | so, linphone crashes now, i'll reinstall it. appears to be buggy |
| 11:23.51 | mjt | http://pastebin.com/ynVwaQJV -- that's debugging from ekiga on the same host as asterisk. There, the problem is that it does not hangup, even the server already dyd |
| 11:23.54 | mjt | did |
| 11:24.12 | mjt | only manual hangup |
| 11:24.14 | mjt | works |
| 11:24.29 | agoodm | by the way, just so you know, I once tried to roll out an asterisk installation with wifi sip phones and it was a complete disaster, tread very carefully with wifi connected phones - call quality issues seem rife |
| 11:24.51 | mjt | interesting |
| 11:25.35 | agoodm | remember wifi is a hubbed technology, only one traffic stream can be broadcasted at a time |
| 11:25.35 | mjt | as i mentioned before, my first attempt was with freeswitch, -- tried in 2 places (at home and in office), used several android phones as clients (all over wifi) -- worked very reliable. |
| 11:26.11 | agoodm | problem is wifi networks are inherently extremely jittery unless totally idle |
| 11:26.27 | mjt | our networks _are_ idle :) |
| 11:26.45 | *** join/#asterisk Chotaire (chotaire@host-089-207-249-134.vipri.net) |
| 11:26.55 | agoodm | even when idle interference from other networks can ruin your day - wireless lan devices are supposed to listen to see if anyone else is transmitting before they do... |
| 11:26.56 | mjt | except of those android devices sometimes using them for email and - guess it - for voip (to be set up) |
| 11:27.25 | mjt | that's why i consider switching to 5GHz @home |
| 11:27.29 | agoodm | dont forget broadcast packets on your lan will be broadcast on your wifi ;) only likely to become an issue in larger networks though |
| 11:27.45 | mjt | no, we don't have broadcasts like that |
| 11:28.16 | agoodm | err yes you do... if you didnt have any broadcast on your LAN it wouldnt function... ARP packets are a good example of broadcast |
| 11:28.17 | *** join/#asterisk linjan (~linjan@176.195.43.46) |
| 11:28.18 | agoodm | DHCP is another |
| 11:28.25 | mjt | well |
| 11:28.35 | agoodm | coffee! brb |
| 11:28.59 | mjt | this has little to do with linphone as the only client being unable to auth, or with built-in android client hanging up |
| 11:29.36 | mjt | i need somethig to start with, I don't need it to be very reliable and stuff |
| 11:30.00 | mjt | we don't plan to use wifi really, exept for the tesing |
| 11:31.03 | mjt | (actually i did plan to use wifi - for this my android phone to be a client for my office phone when @home, and so far I don't see big issues here, -- wifi conditions are very good here) |
| 11:31.44 | mjt | ekiga is sending more packets "I'm online" than all broadcasts together |
| 11:32.46 | agoodm | im back, could you paste your extensions.conf stuff into pastebin please? so I can check it |
| 11:39.49 | agoodm | mjt, btw; you are correct. wireless will work for your testing etc :-) so its not a bad way to test |
| 11:43.13 | agoodm | also the output of iptables -L -v -n would be useful, just to double check your firewall rules look OK :) |
| 11:55.33 | mjt | i don't have iptables installed |
| 11:55.45 | agoodm | you get command not found when you run iptables ? |
| 11:55.53 | mjt | no, i didn't install it |
| 11:56.08 | agoodm | its part of the linux kernel, what operating system did you start with? |
| 11:56.24 | mjt | i'm old scool, so to say, I run only stuff which I explicitly installed and configured |
| 11:56.38 | mjt | iptables is not part of linux kernel |
| 11:56.49 | mjt | i mean the control utility |
| 11:56.58 | agoodm | of course yeah |
| 11:57.14 | mjt | i was writing tutorials for hat stuff back when it was called ipchains |
| 11:57.26 | agoodm | well lets hope you've got policy accept and move on then, if you're getting audio in test calls you're all good tbh |
| 11:57.34 | mjt | heh. |
| 11:57.49 | agoodm | could I see extensions.conf please? |
| 11:58.33 | mjt | $ cat /proc/net/ip_tables_names |
| 11:58.33 | mjt | cat: /proc/net/ip_tables_names: No such file or directory |
| 11:58.37 | mjt | that should answer it |
| 11:58.45 | mjt | (so there's no policies, either) |
| 11:59.12 | agoodm | good to hear :-) |
| 12:00.03 | mjt | # cat extensions.conf | grep -v '^;' | pastebinit |
| 12:00.03 | mjt | http://paste.debian.net/79629/ |
| 12:00.16 | mjt | useless use of cat(1) award |
| 12:00.47 | mjt | this is a default debian jessie (actually wheezy backport) config |
| 12:01.01 | mjt | asterisk 11.7.0 |
| 12:01.12 | agoodm | the code you wrote is in [users] ? |
| 12:01.21 | mjt | yes |
| 12:01.29 | mjt | i moved it from [public] as you suggested |
| 12:01.35 | mjt | s/moved/renamed/ |
| 12:01.44 | agoodm | exten => 10,1,blah new line exten => 10,n,blah |
| 12:02.04 | mjt | ..and chanhed the extension from 999 to 10 |
| 12:02.06 | agoodm | ive no idea how important the > is :D |
| 12:02.55 | *** join/#asterisk tzafrir (tzafrir@conference/fosdem/x-fdqnuopsoouznaxi) |
| 12:03.08 | mjt | the number to n change shoudl not be important, according to the docs, as long as the numbers are consecutive(sp) |
| 12:03.16 | agoodm | thats correct |
| 12:03.32 | agoodm | but it should be exten => 10,1, in my opinion |
| 12:03.44 | agoodm | notice the chevron |
| 12:03.49 | mjt | yeah |
| 12:03.54 | agoodm | you just have = |
| 12:03.56 | mjt | object vs declaration |
| 12:04.05 | mjt | reload |
| 12:04.07 | mjt | er miswin |
| 12:04.34 | mjt | nothing changed on ekiga side |
| 12:05.03 | mjt | i'm reinstalling linphone |
| 12:05.27 | agoodm | ok, in your sip.conf you have context=users ? |
| 12:05.55 | mjt | yes |
| 12:06.12 | mjt | for this account anyway |
| 12:06.55 | agoodm | I think you should be good then really... bar buggy sip clients and stuff like that |
| 12:07.08 | agoodm | let me try linphone on my android and see if I have the same issues |
| 12:07.30 | mjt | linphone appears to work after reinstall |
| 12:07.48 | agoodm | thats good to hear; ekiga doesnt hang up ? |
| 12:08.03 | mjt | yes |
| 12:08.13 | snadge | lol.. ekiga |
| 12:08.36 | agoodm | linphone doesnt appear to be available for android kit kat i believe |
| 12:08.48 | agoodm | i only see linphone video, not the linphone i remember using before? |
| 12:09.03 | mjt | damn. i can't terminate linphone using regular ways |
| 12:09.09 | mjt | it is the linphone video |
| 12:09.15 | agoodm | ah cool ok |
| 12:09.24 | mjt | dunno what they mean by "video", it is a regular sip client |
| 12:09.25 | snadge | on android i use zoiper |
| 12:09.36 | snadge | bria is another one |
| 12:09.56 | agoodm | zoiper I have vague memories of |
| 12:10.37 | mjt | i just love these applications. No way to quit, so it constantly draws battery. |
| 12:11.02 | mjt | task manager => kill. Oh well. |
| 12:12.04 | agoodm | i spose for stuff like a sip phone client having a close option would be nice |
| 12:12.42 | agoodm | but generally for stuff that doesnt continue int he background there is no battery hit despite the fact android keeps the processes running in the background till it needs the resources for something else |
| 12:13.24 | mjt | this one is using network hence draws battery |
| 12:13.36 | mjt | i dont mind sleeping proceses ofcourse |
| 12:13.44 | snadge | you can exit out of zoiper |
| 12:13.48 | snadge | and tell it not to register automatically |
| 12:13.51 | agoodm | eurgh 188.138.83.56 has been aggresively scanning for open sip accounts now for a while :( ive logged onto several client systems recently and found it hammering away :( |
| 12:14.01 | snadge | set up fail2ban bro |
| 12:14.12 | snadge | not difficult |
| 12:14.25 | agoodm | I could do, but the config is secure and it doesnt cause issues for me beyond annoyance when I want to use asterisk terminal |
| 12:14.46 | snadge | convenient.. after x failed attemps, ip is blackholed |
| 12:14.50 | mjt | agoodm: so after a few changes in the config as you suggested it appears to be working fine now (except of the ekiga issue which does not detect hangup). |
| 12:15.11 | mjt | agoodm: i'll try to find which change it was, to understand what's going on |
| 12:15.32 | mjt | (and aftere reinstalling of linphone which was crashing and behaving strangely) |
| 12:15.35 | mjt | after* |
| 12:16.21 | agoodm | mjt, I think its probably the context= line |
| 12:16.37 | mjt | it was context=public by default |
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| 12:17.29 | agoodm | linphone is functioning normally for me |
| 12:17.47 | agoodm | since you're using WiFi youll find the following options for the top of sip.conf useful: |
| 12:17.52 | agoodm | jbenable = yes |
| 12:18.10 | agoodm | jbimpl = fixed (or adaptive) |
| 12:18.15 | agoodm | jbmaxsize = 100 |
| 12:18.21 | mjt | removed context= line from sip.conf, renamed [users] to [public] in extensions.conf, restarted (just in case), -- it works so far :) |
| 12:18.46 | agoodm | ok wonder if it was as simple as missing the > out in extensions.conf ? |
| 12:18.50 | mjt | but it looks like my issue with linphone was that it was having some bad stuff in its config on the phone |
| 12:19.05 | mjt | nope, removing ">"s does not change anything :) |
| 12:19.49 | agoodm | so sounds like issue was buggy softphone all along |
| 12:20.56 | mjt | even built-in android client works now |
| 12:20.58 | agoodm | missing chevron doesnt break on my asterisk 1.8 install either |
| 12:21.43 | mjt | sipdroid doesn't work, and ekiga doesn't hangup. android client, csipsimple and linphone works. Yay. ;) |
| 12:22.02 | agoodm | i think those other issues you're seeing are buggy clients vs broken asterisk |
| 12:22.57 | mjt | at least this lets me to move |
| 12:23.06 | mjt | thank you very much for the help and especially for patience! :) |
| 12:23.15 | agoodm | you're welcome :) |
| 12:24.14 | agoodm | my knowlege of what you can get away with in newer asterisk is quite limited... i started with asterisk 1.2 if I recall and it was quite finicky about what itd work with and without |
| 12:24.38 | agoodm | all my 'templates' have been built from there, probably include stuff thats not technically needed |
| 12:25.02 | mjt | btw, are there good softphones for windows? For one, I'd like to use an "usb phone" - a device which looks like a cell phone - with buttons, speaker and microphone, connected to an USB port, so that it is possible to answer a call by pressing a button on that handset. |
| 12:25.25 | agoodm | example; you can omit HangUp() and it should clean up all by itself, I think it did that back in 1.2 but the person that taught me had been using it since year . when auto fall through didnt exist |
| 12:25.39 | agoodm | mjt xlite and eyebeam are my favorite |
| 12:25.44 | agoodm | zoiper close 2nd |
| 12:26.09 | mjt | i tried xlite. it was crashing left and right :) |
| 12:26.36 | mjt | but i'll try again (it was some obscure testing PC) |
| 12:26.40 | agoodm | I used to use zoiper because it was the only soft phone I could find that supported iax2, and my softphone was on wifi, jitter made calls not usable and iax2 had jitter buffer way way way before sip did :D |
| 12:28.44 | mjt | oh. there's whole [demo] section in my (debian) extensions.conf |
| 12:29.28 | agoodm | mjt I would suggest starting from empty extensions.conf, it makes things less covoluted... all you need is a [general] and a [users] or whatever context you put your phones in |
| 12:30.46 | agoodm | get out of the habit of putting your phones in a public context though, otherwise unauthenticated users and malicious third parties (if you connect it to interwebz) will make calls to or through your installation... best case scenario it spams your logs, middle scenario you get random silent calls to your extensions, worse case toll fraud |
| 12:31.10 | mjt | it isn't a habit yet :) |
| 12:32.08 | agoodm | public context is set with context=something in general section of sip.conf, set it to something and create the context in sip.conf then dont put anything that doesnt need to be there in it. My configs have lines which grab the inbound telephone calls from the sip provider which delivers them to me anonymously and thats it |
| 12:34.31 | agoodm | deny/permit rules on your sip.conf entries are also a very wise move... if your sip phone only registers from local ip ranges deny everything then permit just your local ranges :) |
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| 12:39.56 | mjt | this host does not have access from outside anyway ;) |
| 12:40.26 | mjt | i've been with networks and linux in heavy ways for over 20 years |
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| 13:00.00 | mjt | hm. ekiga is trying to use the same port when run on the same host as asterisk |
| 13:31.18 | mjt | speaking of network. Is it considered unsafe to expose asterisk to the outside world? I mean, like, to be able to use our in-office sip server from these same android phones elsewhere? |
| 13:32.03 | mjt | i asked the same question about freeswitch, they said it is okay, but within about 5 minutes of looking at the code I found remote exploitable buffer overflow. |
| 13:32.52 | file | if you don't properly secure it then it's a bad idea |
| 13:34.05 | file | for example leaving anonymous calling on allows someone to enumerate valid devices (as the response differs) |
| 13:34.46 | file | choosing poor passwords also means that someone could bruteforce an account, unless you use something like fail2ban that will automatically firewall a bruteforce attacker |
| 13:35.01 | mjt | yeah, i don't plan to allow anonymous calling, plan to allow only a few accounts from "outside" and will use stronger passwords for these |
| 13:35.59 | file | then not as bad |
| 13:36.54 | file | oh, and outdated Asterisk... that'll also get you since we fix reported security issues |
| 13:36.55 | mjt | plus a usual higiene on the host - will most likely run it in a separate (virtual) machine in a dmz |
| 13:37.32 | file | all fine in my book tthen |
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| 14:02.00 | minotaur01 | o/ morning everyone, I have a quick question about sound packages, I just did a fresh install of asterisk from the repos and the default install only installed the asterisk-sounds-core-en-gsm sounds package shoud I also install the ulaw and alaw versions or is it not nessary? |
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| 14:07.06 | minotaur01 | anyone? |
| 14:08.05 | mjt | hmm |
| 14:08.07 | mjt | [Feb 1 18:07:00] WARNING[16518][C-0000002f]: chan_oss.c:488 setformat: Unable to re-open DSP device /dev/dsp: Device or resource busy |
| 14:08.23 | mjt | wonder why it is tryin to use OSS instead of ALSA |
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| 15:07.19 | agoodm | mjt, im my opinion it is safe to expose a properly secured asterisk install to the internet |
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| 15:09.54 | agoodm | mjt, best practices for good security include not using numeric user names, use mac address, something else, secure passwords, being careful what you put in default context, ensuring outbound contexts are not included in default context and being careful about using exten variable in dial command on untrusted contexts - you can inject additional destinations if you're not careful |
| 15:11.37 | agoodm | I have dozens of asterisk installations exposed to the internet. the last security issue was caused after a web application running on the same host was compromised and used to read my sip.conf details, then my trunk details were abused :-( lack of security at trunk provider however they did have good anti fraud and shut my account off rapidly |
| 15:11.46 | agoodm | this was about 4 years ago |
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| 15:54.56 | xaristax | Hi i need some guide, im trying to config my phones in the file sip.conf but not using this syntax [100] instead i see in some places that for security i have to use something like this [92392914] but when im trying to call my extension 100 it doesnt match so i use rexexten but still dont work can someone give me a help on this |
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| 15:58.42 | xaristax | ??? |
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| 16:13.20 | Penguin | ~devicenames |
| 16:13.20 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 16:13.26 | Penguin | ~book |
| 16:13.27 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:19.23 | xaristax | thanks penguin i get the idea but whats drive me crazy is how can i reach for example extension 100 im a newby on this so sorry if the question is a little bit dumb |
| 16:19.58 | Penguin | Extensions execute commands. |
| 16:20.23 | Penguin | You create the dialplan, so you can make extension 100 do anything you want. |
| 16:21.45 | xaristax | right but if a call from extension 102 to 100 i know i can write this on the dialplan and it may sound lazy but i have something like this _XXX is there a way to make this work |
| 16:22.00 | Penguin | You can't call FROM extensions. Extensions execute commands. |
| 16:22.43 | Penguin | You send calls into asterisk from phones. Asterisk will try to match the extension that you're calling, and if there's a match, the extension will execute. |
| 16:24.06 | xaristax | thanks for the time penguin and sorry if im a little bit slow :P i know asterisk place the calls but do i have to put the ID tar or asset in the dialplan |
| 16:24.20 | xaristax | or is a way that i can explicit say in sip.conf |
| 16:24.26 | xaristax | that that id is the extension 100 |
| 16:24.50 | Penguin | The extension pattern _XXX isn't the most ideal patten to use if you mean 100. |
| 16:25.06 | Penguin | If you want extension 100 to make a call to a device, you use the Dial() application. |
| 16:25.50 | Penguin | exten => 100,1,Dial(SIP/0000aaaaffff,26) |
| 16:26.08 | xaristax | right now i have something like this _XXX,1,Dial(SIP/${EXTEN},30,) |
| 16:26.23 | Penguin | This says extension 100 dials a SIP phone which has the name of 0000aaaaffff, and it will try for 26 seconds before giving up. |
| 16:26.55 | xaristax | oooooooo right i thing im the lazy XD that wants everything easy |
| 16:27.04 | xaristax | but you are right im gonna do it that way |
| 16:27.11 | Penguin | ${EXTEN} is always the extension you are executing. In your case, it would be 100. And you don't have a phone named 100. At least you shouldn't because it's a terrible name for a phone. |
| 16:28.06 | xaristax | you save me on this one penguin i own you a big one :D |
| 16:28.27 | xaristax | really thanks for the time and clarify things for me and sorry for the bad english |
| 16:28.31 | Penguin | You can use patterns like that if you prefer, but you won't be using ${EXTEN} as the device to call since that's the extension not the device name. |
| 16:31.46 | xaristax | man really thanks again :D |
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| 20:15.34 | skrusty | evening |
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| 20:41.28 | sectech | PJSIP... does it give us any kind of performance increase over chan_sip? |
| 20:41.49 | sectech | or is it mostly just cleaner easier to use code. |
| 20:44.16 | pabelanger | sectech, no, just easier to maintain right now |
| 20:44.47 | pabelanger | I'd suspect chan_pjsip preforms less then chan_sip right now |
| 20:44.59 | pabelanger | until it has been baked in for a few releases |
| 20:45.07 | sectech | agh ok |
| 20:45.54 | pabelanger | test and let us know how it goes |
| 20:48.24 | sectech | It's compiling now... |
| 20:49.54 | sectech | Asterisk 12 is compiling rather... |
| 20:51.13 | sectech | Really looking forward to playing with the RestAPI... I can already think of a few cool things to build. |
| 20:56.25 | ibercom | What about performance of asterisk 1.8 or 11 ? Which one uses more CPU ? |
| 21:01.46 | ibercom | Anybody know ? |
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