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| 07:16.24 | andycol_500 | Hi All |
| 07:16.31 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 07:16.46 | andycol_500 | how can you pattern match a variable? |
| 07:16.48 | andycol_500 | for example |
| 07:16.57 | andycol_500 | exten => _X.,2,GotoIf($["${CALLERID(num)})"="_27X."]?s-Neo1,1) |
| 07:17.06 | andycol_500 | so i want it to look if the callerid begins with 27 |
| 07:17.30 | andycol_500 | and if does it must go to s-Neo,1 |
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| 07:23.33 | wdoekes | you cannot pattern match variables, but you can regex-match them, using the =~ operator |
| 07:24.05 | *** join/#asterisk ruied (~ruied@192.61.136.95.rev.vodafone.pt) |
| 07:24.13 | wdoekes | but, you can also pattern match the callerid, using exten => _X./_27X.,1,Goto(s-Neo1,1) ; I think |
| 07:25.44 | Rac-on | imho it is more readable to do a GotoIf($["${CALLERID(num):0:2})"="27" |
| 07:29.29 | andycol_500 | hmm |
| 07:29.37 | andycol_500 | let me try both ur guys ideas quick :) |
| 07:30.40 | andycol_500 | wdoekes: i dont think that will work to be honest surely i need a GotoIf? |
| 07:31.16 | wdoekes | surely? no.. the exten will only match if the callerid matches |
| 07:31.26 | wdoekes | it's a Rac-on! |
| 07:31.36 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 07:31.37 | Rac-on | wdoekes: it is indeed \o/ |
| 07:31.38 | andycol_500 | can u give me the full dialplan line you would add? |
| 07:31.53 | wdoekes | did I not just do that? |
| 07:32.05 | andycol_500 | ok let me try it |
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| 07:32.39 | andycol_500 | i have just never seen a / instead of a , after X. |
| 07:33.33 | wdoekes | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching <-- "Matching on Caller ID |
| 07:33.44 | wdoekes | I admit that that did not turn up on my first google search |
| 07:34.15 | andycol_500 | yea but it doesnt seem to work on variables |
| 07:34.41 | wdoekes | obviously, since it will match the callerid only |
| 07:35.16 | andycol_500 | yes but i only need it to match first 3 digits so if it is +27X must go to another context and change the +27 to 0 |
| 07:36.00 | Rac-on | you didnt mention the + before |
| 07:36.10 | andycol_500 | well let me explain what i am trying to do |
| 07:36.12 | Rac-on | so you will have to rewrite both my option and the one from wdoekes |
| 07:36.27 | andycol_500 | so if a customer sends a call with callerid +27 it must rewrite to 0 |
| 07:36.37 | andycol_500 | and if they send with just 27 it must rewrite to 0 |
| 07:37.11 | Rac-on | use a execif? |
| 07:37.34 | andycol_500 | can u give me an example |
| 07:37.40 | Rac-on | ExecIf($["${CALLERID(num):0:3}"="+27"]?Set(CALLERID(num)=0${CALLERID(num):3})) |
| 07:37.54 | Rac-on | something like that. chek if the clid starts with +27, if so, strip 3 digits and add a 0 |
| 07:38.13 | andycol_500 | ok let me see :) |
| 07:38.50 | wdoekes | [fix-cli] |
| 07:38.50 | wdoekes | exten _+27!,1,Set(CALLERID(num)=0${EXTEN:3}) |
| 07:38.50 | wdoekes | exten _27!,1,Set(CALLERID(num)=0${EXTEN:2}) |
| 07:38.50 | wdoekes | exten _[0-9+]!,1,NoOp() |
| 07:38.50 | wdoekes | exten _[0-9+]!,n,Return() |
| 07:38.52 | wdoekes | ... |
| 07:38.55 | wdoekes | exten => _X!,1,Gosub(fix-cli,${CALLERID(num)},1) |
| 07:39.33 | wdoekes | there, cli is "fixed" |
| 07:39.41 | andycol_500 | wdoekes: but i need it to be exten => _X. |
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| 07:40.08 | wdoekes | nono, you're not reading the solution |
| 07:40.13 | wdoekes | you start at _X! |
| 07:40.42 | andycol_500 | umm |
| 07:40.43 | Rac-on | andycol_500: as always, wdoekes makes a solution more elegant than mine |
| 07:41.32 | andycol_500 | like this |
| 07:41.42 | andycol_500 | http://pasteboard.co/h9q8fIrl1.png |
| 07:42.12 | wdoekes | no, shouldn't be in the same [context] |
| 07:42.31 | andycol_500 | ok |
| 07:42.32 | andycol_500 | so |
| 07:42.34 | Rac-on | pls use a text-based pastebin |
| 07:42.40 | andycol_500 | will do |
| 07:42.58 | wdoekes | you have an [incoming] somewhere where you planned on doing this: exten => _X.,2,GotoIf($["${CALLERID(num)})"="_27X."]?s-Neo1,1) |
| 07:43.16 | andycol_500 | http://pastebin.com/a9jc9nCi |
| 07:43.18 | wdoekes | instead you do: exten => _X!,2,Gosub(fix-cli... |
| 07:43.27 | wdoekes | and on the next line your cli is as you wanted it |
| 07:43.28 | andycol_500 | like that? |
| 07:43.53 | Rac-on | yeah |
| 07:44.03 | andycol_500 | then where do i add the dial line so it dials the call out? |
| 07:44.08 | wdoekes | on the next prio in [extensions-test] |
| 07:44.10 | Rac-on | then add the rest of your dialplan after the gosub in [extensions-test] |
| 07:44.11 | andycol_500 | at the bottom of [fix-cli] |
| 07:44.17 | andycol_500 | ok |
| 07:44.48 | andycol_500 | so like this |
| 07:44.54 | andycol_500 | http://pastebin.com/FupgH7bM |
| 07:45.40 | Rac-on | yes, but use ,n, instead of ,2, |
| 07:45.45 | andycol_500 | ok |
| 07:45.59 | andycol_500 | let me test ) |
| 07:46.01 | andycol_500 | let me test :) |
| 07:46.09 | Rac-on | basicly, only number the first line 1 and everything after that 'n'. it makes it much easier to edit your dialplan later |
| 07:46.18 | andycol_500 | true |
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| 07:47.36 | andycol_500 | looks like it is working :) |
| 07:47.38 | andycol_500 | thanks guys |
| 07:47.46 | Rac-on | yw |
| 07:52.46 | jkroon | hi all, i've got a system with around 80 IPv4 addresses configured, and as many asterisk instances bound to each of those (bindaddr). From what I can determine only one bindaddr is supported (ast11 at least). As such - is there any way to bind to a specific IPv4 as well as IPv6 address? Any way to DNAT/SNAT between the two such that asterisk binds to IPv6 (or IPv4) but still enables the alternate stack? |
| 07:53.13 | jkroon | (only looking to solve the problem for chan_sip) |
| 07:53.33 | Kunsi | jkroon: bind asterisk to v6, your OS should take kare to map v4 to v6 addresses |
| 07:54.42 | jkroon | Kunsi, it should? how? I know if I bind to :::X (ie, IPv6 variant of 0.0.0.0:X), but let's say I configure PFX::a.b.c.d - will the kernel automatically map a configured a.b.c.d IPv4 address to the IPv6 socket? |
| 07:56.02 | Kunsi | atlest my router (debian) maps 127.0.0.1 to ::ffff:127.0.0.1 (and so on) |
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| 07:57.26 | jkroon | investigates thanks |
| 07:57.40 | jkroon | unfortunately my laptop isn't connected to my IPv6 network at the moment :( |
| 08:02.12 | jkroon | Kunsi, ok, so I need to bind to public IPv6 in our use-case, eg, 2c0f:XXXX::c0a8:2bc1/64 |
| 08:03.34 | Kunsi | hm, don't know if that works then, my asterisk listens on :: ... but you should be able to do some firewall magic |
| 08:04.17 | wdoekes | yes.. but if you listen on a specific v6 IP, you won't listen to v4 anymore, afaict |
| 08:04.20 | wdoekes | compare: |
| 08:04.22 | wdoekes | >>> s = socket(AF_INET6, SOCK_DGRAM) |
| 08:04.25 | wdoekes | >>> s.bind(('', 1234)) |
| 08:04.27 | wdoekes | >>> print s.recvfrom(4096) |
| 08:04.29 | wdoekes | s.bind(('::1', 1234)) |
| 08:04.31 | wdoekes | ('v4\n', ('::ffff:127.0.0.1', 44896, 0, 0)) |
| 08:04.34 | wdoekes | with |
| 08:04.51 | wdoekes | that last one won't listen on v4: echo v4 | nc localhost 1234 -u4 |
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| 08:12.16 | jkroon | wdoekes, my testing with adding 2c0f:XXXX::192.168.43.173 to lo, and then running nc -6 -l -p 5000 also didn't allow me to telnet to port 5000 on 192.168.43.173 so this is going to be slightly more complex probably. |
| 08:13.19 | jkroon | would prefer to avoid firewall magic if possible ... perhaps some kind of SIT tunnel ... so set up that entire VLAN as IPv6-only and then on the router perform an IPv4=>IPv6 MAP, so the client IP will effectively be ffff::a.b.c.d. |
| 08:13.52 | jkroon | i'm just wondering how that would affect the SDP related stuff, because asterisk will now send SDP containing IPv6 addresses which the poor IPv4 endpoint won't know how to handle. |
| 08:14.17 | jkroon | for IAX/2 it shouldn't be a problem, not for any of the http or any other interfaces that doesn't utilize SDP. |
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| 08:33.27 | wdoekes | jkroon: docker and expose only one v4 IP to asterisk? dunno. I'm not that intimate with docker, but I would assume you can be able to create a limited network with only one of your IPs |
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| 08:37.10 | jkroon | wdoekes, thanks - that's an interesting idea. don't know docker myself either, but you're right, if one can expose only that one IPv4 and one IPv6 to docker then asterisk can bind to "any" and both ipv4 and ipv6 should function. In theory. |
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| 09:35.03 | cervajs2 | is is possible log queue_log to sqlite? |
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| 13:25.43 | Bumbaa | Hello,I'm having a problem with my asterisk not monitoring Peers, I have tried to change the qualify to 1 via commnand line but it is still unmonitored. |
| 13:25.54 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:25.54 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:26.44 | Bumbaa | I went to check on the web interface and advanced setting and SIP and IAX qualifywas yes by default with a qualifyfreq of 60. |
| 13:27.01 | Bumbaa | But when I do sip show peer <name> it still shows unmonitored. |
| 13:33.08 | Bumbaa | This is the output command off sip show peer <name> |
| 13:33.08 | Samot | Show the config for the device peer, mask any passwords. |
| 13:33.08 | Bumbaa | http://pastebin.com/jYzbLHTJ |
| 13:33.32 | Bumbaa | Samot: Is that what you want? |
| 13:34.25 | Samot | Yes, but it was the next request. |
| 13:34.30 | *** join/#asterisk ruied (~ruied@192.61.136.95.rev.vodafone.pt) |
| 13:34.34 | Samot | Still show the config for the device peer. |
| 13:34.36 | Samot | From sip.conf |
| 13:34.39 | Bumbaa | ok |
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| 13:39.58 | Bumbaa | Samot They are two sip.conf one in /etc/asterisk and one in /var/www/html/admin/modules/core/etc/ , I'll give you both. |
| 13:40.19 | Samot | No. |
| 13:40.28 | Samot | You're using a GUI? |
| 13:40.33 | Samot | FreePBX? |
| 13:40.40 | Bumbaa | I have access to FreePBX gui. |
| 13:40.46 | Bumbaa | But I was taking it directly from command line |
| 13:40.46 | Samot | So you should be in #freepbx for this. |
| 13:40.55 | Bumbaa | well |
| 13:41.07 | Samot | Because I would just say, go into the GUI and get the trunk config. |
| 13:41.33 | Samot | FreePBX creates the .conf files differently. |
| 13:41.44 | Bumbaa | I went into the GUI and saw in advanced setting the qualify was already set to yes with a qualifyfreq of 60. |
| 13:41.46 | Bumbaa | Oh,I see. |
| 13:41.57 | Samot | OK, still show the config. |
| 13:42.15 | [TK]D-Fender | <Bumbaa> But I was taking it directly from command line <- there is no such thing as enabling qualify from CLI |
| 13:43.13 | Bumbaa | Isn't it sip qualify peer <name> <input> ? |
| 13:44.25 | [TK]D-Fender | that is a ONE-SHOT. It does not enable monitoring in any on-going capacity, nor does it change any judgement on the peer sttatus |
| 13:45.09 | [TK]D-Fender | "sip qualify peer Send an OPTIONS packet to a peer |
| 13:45.09 | [TK]D-Fender | " <- Singular. Doesn't change the peer's general rules. |
| 13:45.17 | Bumbaa | Samot , This is the sip.conf from /etc/asterisk http://pastebin.com/5YatUpEi and this one is the one from http://pastebin.com/EJ2aYhqz |
| 13:45.52 | Samot | Not what I said after finding out this was FreePBX. |
| 13:45.54 | [TK]D-Fender | #include sip_additional.conf <- that's where your extensions are defined |
| 13:46.02 | Samot | I said go into the GUI and get the config. |
| 13:46.06 | Bumbaa | Ok, thank you for that information Fender, so how would I go at it to make it continuously monitored. From the research I have done, I need to enable qualify for the peer, but I tried different ways and here I am. |
| 13:46.19 | [TK]D-Fender | Fix your peer definition |
| 13:46.20 | Bumbaa | Apologies, Samot. I missunderstood. |
| 13:47.15 | Bumbaa | So, If I understand properly, I must configure my peer in the sip-additional.conf , Fender? |
| 13:48.11 | [TK]D-Fender | no, you have to do it in the GUI |
| 13:48.17 | [TK]D-Fender | you have to do EVERYTHING in the GUI |
| 13:48.22 | [TK]D-Fender | It owns your configs, not you |
| 13:49.19 | [TK]D-Fender | The GUI does not "edit" configs, so there is no hand-editing to make changes you want there. The GUI completely rebuilds them every time you apply changes from the GUI. |
| 13:50.58 | Samot | 10 minutes.. |
| 13:51.11 | *** join/#asterisk ntt (~smuxi@89.21.199.116) |
| 13:54.59 | ntt | Hi, I'm using asterisk 13.10.0 on centos7 with freepbx. Randomly in logs I see a warning: "Unable to write frametype: 2". Rarely this message "floods" the log file and in that case I have some problem with the call (disturbed audio). Can someone help me, please? |
| 13:55.13 | Bumbaa | I'm trying to find where I can find the config file in FreePbx, I'm sorry samot. |
| 13:55.42 | *** join/#asterisk andremar (~andremar@105.211.54.77.rev.vodafone.pt) |
| 13:55.47 | Samot | It's in the same spot you set it up. |
| 13:55.55 | Samot | Connectivity -> Trunks. |
| 13:57.18 | Bumbaa | You want peer details? |
| 13:58.08 | Bumbaa | Oh,I understand.I got it now. |
| 13:58.14 | Bumbaa | You must include in the peer details for the options. |
| 13:58.18 | Bumbaa | D'oh. |
| 13:58.34 | Bumbaa | Thank you very much, samot. |
| 13:58.57 | *** join/#asterisk sekil (~sekil@109-93-199-144.static.isp.telekom.rs) |
| 14:00.41 | Samot | For what? |
| 14:02.51 | Bumbaa | For helping me,you pointed me in the correct direction. |
| 14:02.59 | Bumbaa | Thank you :) |
| 14:03.08 | Samot | I still don't see the config |
| 14:03.26 | Samot | Thank me by providing the information I asked for. |
| 14:04.08 | Bumbaa | http://pastebin.com/kmRTRHvT here you go. |
| 14:04.20 | Bumbaa | Apologies, don't mean to offend or anything. |
| 14:04.52 | Samot | canreinvite is dead. It was replaced with directmedia |
| 14:05.15 | Samot | If this is a provider trunk, I highly doubt they are behind NAT. |
| 14:05.24 | Samot | So you want that to be nat=no |
| 14:06.10 | ntt | Samot: have you idea of what the message "unable to write frametype: 2" means? |
| 14:06.39 | Bumbaa | Ok, what did careinvite do? And does directmedia do now? I'm sure you know that I am pretty new at this. |
| 14:06.55 | Bumbaa | And do you know where I could list of the available parameters list I can put in the peer details box? |
| 14:08.30 | Samot | wiki.asterisk.org |
| 14:08.54 | Samot | ntt: It's voice related. You need to provide more that a single error in the log. |
| 14:09.01 | Samot | Context to the error would be helpful. |
| 14:10.14 | Bumbaa | Thanks samot, have a nice day. |
| 14:11.24 | ntt | Samot: you can check here more details -> http://pastebin.com/rkZCUdie |
| 14:11.52 | *** join/#asterisk kharwell (kharwell@nat/digium/x-yuevtwjpzvvupplr) |
| 14:12.24 | Samot | rfc2833compensate=yes <-- I don't even think that's valid anymore |
| 14:12.46 | Samot | Wasn't that dropped in 1.4? |
| 14:13.48 | ntt | Samot: are you talking with me? |
| 14:13.55 | Samot | No. |
| 14:15.12 | Samot | Here we go. Another FreePBX user. |
| 14:15.21 | Samot | ntt: We need to see the *whole* call. |
| 14:16.35 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:16.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.06 | cervajs2 | is it possible play announcement after queue join like "all agents busy"? |
| 14:19.43 | Samot | Immediately when the caller joins the queue? |
| 14:19.55 | Samot | I'm not sure there's something that does that per user join. |
| 14:21.11 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 14:21.37 | cervajs2 | Samot: yes |
| 14:22.08 | [TK]D-Fender | cervajs2, No, but you can play one BEFORE they go in |
| 14:23.15 | cervajs2 | [TK]D-Fender: yea i know. but i need info in queue_log :( |
| 14:23.55 | [TK]D-Fender | What "info"? I fail to see a question anywhere... |
| 14:26.29 | ntt | Samot: http://pastebin.com/fBhmbUL5 please let me know if this is a complete log |
| 14:27.04 | cervajs2 | [TK]D-Fender: info about missed calls. we are using data from queue_log about missed calls |
| 14:28.24 | [TK]D-Fender | Ok.... |
| 14:28.32 | [TK]D-Fender | I still fail to see a problem. |
| 14:32.09 | cervajs2 | <PROTECTED> |
| 14:33.18 | cervajs2 | i need statistics if he immediatelly hangup the call when he hear this message |
| 14:34.00 | Samot | ntt: So this is all over SIP trunks? |
| 14:34.28 | ntt | yes.... on the same lan there is also a ATA adapter with another trunk (not used on the pbx) |
| 14:36.22 | [TK]D-Fender | <cervajs2> i need statistics if he immediatelly hangup the call when he hear this message <- if it's a message before they enter the queue then it's your dialplan... do whatever you want |
| 14:37.11 | [TK]D-Fender | <cervajs2> [TK]D-Fender: is it possible play announcement after queue join like "all agents busy"? <- what part of "no" are you having trouble with? |
| 14:39.18 | Samot | ntt: Show a call with a SIP debug: sip set debug on |
| 14:39.48 | ntt | Samot: ok, but I'm connected via ssh and there are a lot of calls on the pbx.... |
| 14:39.57 | Samot | cervasjs2: So catch the hangup in the announcement context. |
| 14:40.15 | Samot | Send call to announcement, catch hangup there before even going to the queue. |
| 14:40.39 | WIMPy | Well, you can craft a special MOH context for your queue listing your announcement and a list of (possibly the same) music file(s). |
| 14:40.40 | Samot | Then do: sip debug peer <peer> |
| 14:40.48 | *** join/#asterisk v1r0 (~viro@unaffiliated/viro) |
| 14:40.49 | Samot | <peer> being the extension you are calling from. |
| 14:41.11 | [TK]D-Fender | WIMPy, I don't recall MoH necessarily starting from start of some source |
| 14:41.25 | [TK]D-Fender | WIMPy, And there is still no specifical logging. |
| 14:41.27 | Penguin | I think you mean <peer> being the peer you are calling from. |
| 14:41.34 | Penguin | Since you don't call FROM extensions. |
| 14:41.35 | *** join/#asterisk Hatsjoe (~Hatsjoe@unaffiliated/hatsjoe) |
| 14:41.37 | WIMPy | You can have MOH without random, can't you? |
| 14:41.42 | Samot | Penguin: He's using FreePBX, I'm speaking in those terms. |
| 14:42.03 | Penguin | Why are you not doing it in #freepbx? |
| 14:42.32 | Samot | Because the app_dial issue isn't a GUI issue. |
| 14:43.11 | ntt | Actually I have: app_dial.c:1526 wait_for_answer: Unable to write frametype: 2 |
| 14:43.15 | ntt | wait_for_answer ?? |
| 14:43.21 | Hatsjoe | In the debian jessie package of asterisk (version 11.13.1), the module 'res_config_ldap.so' is missing.. Not sure how to get it, besides building asterisk from source, am I missing something? |
| 14:43.39 | Samot | You would need to talk to the people who maintain that package. |
| 14:43.45 | [TK]D-Fender | Have you checked if it was provided in a separate package? |
| 14:43.46 | Samot | Other than that, souce. |
| 14:43.56 | Samot | ^ That too. |
| 14:44.26 | Hatsjoe | Couldn't find any package providing it |
| 14:44.34 | Hatsjoe | But will check with the maintainer indeed |
| 14:46.21 | ntt | Samot: http://pastebin.com/k44PceWD can you check this? |
| 14:46.36 | ntt | I'm not sure that this is a complete call .... |
| 14:46.41 | ntt | but the error happens |
| 14:50.28 | Samot | <--- Transmitting (NAT) to 192.168.1.131:3645 ---> |
| 14:50.47 | Samot | Your endpoints are local to the PBX, why do you have Asterisk thinking they are behind NAT? |
| 14:51.30 | ntt | yes, phones are on the same lan, but the pbx doesn't have a public ip. Should I remove the nat in the extension settings? |
| 14:55.13 | Samot | nat=no |
| 14:55.19 | Samot | Because they aren't behind NAT. |
| 14:56.01 | WIMPy | If your Asterisk isn't on the internet there should not be nat anywhere. |
| 14:57.35 | ntt | ok, but only for extension, right? In "asterisk sip settings" -> "chan sip settings" nat should be on yes? |
| 14:58.08 | WIMPy | Why? Where is any NAT going to happen? |
| 14:58.11 | Penguin | Are we still talking about configuring FreePBX? |
| 14:58.47 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 14:59.03 | igcewieling | It is rarely harmful to enable asterisk's nat handling even when there is no nat. |
| 14:59.17 | ntt | WIMPy: I'm confused |
| 14:59.45 | [TK]D-Fender | Peer audio RTP is at port 0.0.0.0:20812 |
| 14:59.46 | igcewieling | ntt: Most people who ask FreePBX questions on #Asterisk are confused. |
| 14:59.52 | [TK]D-Fender | Yeah, THAT seems trustworthy... |
| 14:59.59 | WIMPy | igcewieling: True. |
| 15:00.31 | Penguin | That's why FreePBX chatter should be on #freepbx instead of here. |
| 15:01.43 | Penguin | My general nat setting is nat=force_rport,comedia |
| 15:01.59 | Penguin | I don't disable it for peers not behind nat. |
| 15:02.16 | Penguin | So that means that every peer on my asterisk has nat enabled. |
| 15:02.24 | ntt | this is my setting for all extension |
| 15:02.31 | clopez | Why the security advisories don't have CVE numbers? Distributions won't notice if you dont ask for CVE numbers |
| 15:02.37 | Penguin | It's my settings for all peers, since extensions don't have that type of setting. |
| 15:02.41 | clopez | http://www.asterisk.org/downloads/security-advisories |
| 15:02.49 | WIMPy | Neither do I. But if you know you definitely can't have NAT, you can disable ot. |
| 15:02.51 | WIMPy | it |
| 15:03.18 | Penguin | If the nat value causes a problem, setting nat=no is an easy override. |
| 15:04.25 | WIMPy | ANY kind of NAT support is known to sometimes backfire. So if the situation is that clear, just play it safely. |
| 15:04.55 | ntt | but in asterisk sip settgins -> chan sip settings nat should be = no if the pbx is behind a router? |
| 15:05.15 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 15:05.27 | WIMPy | Depends if there's NAT involved. |
| 15:05.37 | ntt | all extensions are on the same lan |
| 15:05.48 | Samot | As far as the endpoints go... |
| 15:05.56 | WIMPy | Didn't you say it wasn't on the internet? |
| 15:06.16 | Samot | They should be nat=no, there's no NAT between them and Asterisk. |
| 15:06.41 | ntt | endpoints = only extensions? |
| 15:06.54 | WIMPy | extensions are in extensions.conf. |
| 15:07.04 | [TK]D-Fender | <ntt> all extensions are on the same lan <- and those clientts are passing garbage media IP's so you can't trust them and leaving as NAT=YES is GOOD |
| 15:07.46 | Penguin | endpoints are phones. Extensions are the commands that make connections to endpoints. |
| 15:08.03 | WIMPy | Configure those phones correctly. |
| 15:08.07 | Penguin | endpoints are configured in sip.conf; extensions are configured in extensions.conf |
| 15:11.11 | ntt | if I understand, I can set NAT=no for each extension... I tried with "202" and now I see "Reliably Transmitting (no NAT) to 192.168.1.131:3645" |
| 15:11.22 | Samot | ntt: Just go to #freepbx. You need to configure things properly which means you need to be in the GUI. Which is not supported here. |
| 15:11.42 | Penguin | extensions don't have nat values, phones do. |
| 15:11.51 | Samot | OK. |
| 15:11.56 | Samot | Just got to #freepbx. |
| 15:12.22 | WIMPy | You can call them extensions in #freepbx. For Asterisk it means somethin different. |
| 15:23.39 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
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| 15:56.11 | Mango45 | I love looking at the audio IPs in my CDR. Particularly when I see one I don't recognize, and discover my audio is traveling directly from me to the carrier of the company I called. |
| 15:57.53 | Mango45 | Well, maybe 'love' is a strong word for that. But it is mildly amusing if you are interested in VoIP. |
| 16:00.39 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-zzycvefmplfhxlic) |
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| 16:53.09 | igcewieling | I don't know if this will be helpful to anyone, but here is a copy of my CentOS Kickstart file I use to do the OS install for FreePBX boxes we install. http://pastebin.com/g8LHMirj |
| 16:54.16 | *** join/#asterisk dandann00dle (uid120491@gateway/web/irccloud.com/x-hpkovkgmatdnasgm) |
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| 17:23.41 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 17:30.42 | Samot | Mango45: There are quite a large number for VoIP providers that have no hand in the RTP stream of the calls. |
| 17:31.41 | WIMPy | So lawful interception is now done on the customers line? |
| 17:32.05 | Samot | Or at the carrier level. |
| 17:32.34 | WIMPy | What carrier? |
| 17:32.53 | Samot | Are we just talking theory? |
| 17:33.12 | WIMPy | Where is the difference? |
| 17:33.17 | Samot | Was that question in response to my RTP stream statement? |
| 17:33.29 | WIMPy | yes |
| 17:33.35 | Samot | OK. |
| 17:33.38 | *** join/#asterisk pvoigt (~Linux@unaffiliated/pvoigt) |
| 17:33.39 | Samot | VoIP Innovations. |
| 17:33.51 | Samot | They are a VoIP Provider.. |
| 17:34.00 | Samot | But when you buy a DID, the actual carrier is Level 3 or Onvoy... |
| 17:34.10 | WIMPy | There are rumors that some ITSPs got paid to make sure all RTP goes through their infrastructure. |
| 17:34.19 | Samot | I'm not saying they don't. |
| 17:34.37 | Samot | I'm saying a large number of VoIP Providers do not have a hand in the RTP stream. |
| 17:34.43 | Samot | Flowroute uses Directmedia. |
| 17:34.53 | Samot | Which means, they don't touch the RTP stream. |
| 17:34.56 | WIMPy | But they wouldn't be involved. |
| 17:35.05 | Samot | I think that's what he is saying.. |
| 17:35.15 | Samot | He expects the media IP to belong to X Voip Provider. |
| 17:35.21 | Samot | Instead it belongs to Y carrier. |
| 17:36.17 | WIMPy | So only one of two ITSPs optimizing themselves away. |
| 17:36.47 | Samot | I'm giving examples. |
| 17:37.25 | Samot | I'm not saying that ALL voip providers don't have a hand in the RTP stream. |
| 17:37.58 | WIMPy | That's not what I understood, either. |
| 17:38.45 | Samot | That's because of the typo |
| 17:38.48 | Samot | s/for/of |
| 17:38.58 | Samot | "large number of.." |
| 17:42.57 | igcewieling | (usa only) my employer would be fined up to $10,000/day for failure to process a lawful wiretap order. It is one of the reasons I've not pushed harder for SRTP. |
| 17:44.53 | igcewieling | Samot: Do they do direct media when the asterisk side is behind NAT? |
| 17:45.20 | Samot | Flowroute? They do direct media period. |
| 17:46.02 | Samot | Now, they *may* proxy the media. |
| 17:46.16 | Samot | Which in turn allows them to record the media. |
| 17:46.33 | *** join/#asterisk v1r0 (~viro@unaffiliated/viro) |
| 17:47.34 | igcewieling | I wonder how they do that with NAT. |
| 17:47.53 | Samot | That's why you would proxy media. |
| 17:47.55 | Samot | For NAT. |
| 17:48.36 | igcewieling | Correct, those are not direct media. |
| 17:49.53 | igcewieling | seems like nearly 100% of endpoints are behind NAT. |
| 17:50.17 | WIMPy | It could work with NAT as well. |
| 17:50.20 | Samot | Yup and they're letting the carrier deal with that. |
| 17:51.28 | Samot | Even when you proxy media.. |
| 17:51.39 | Samot | In order to record or inject media into the call.. |
| 17:51.45 | Samot | You need something that supports media. |
| 17:51.53 | igcewieling | When media is proxied, I don't ever call that 'direct media' |
| 17:52.18 | Samot | At the core the proxy just helps with NAT. |
| 17:52.22 | WIMPy | Nothing direct about that. |
| 17:52.22 | Samot | It does nothing with the call. |
| 17:52.31 | Samot | The media is still controlled by the two endpoints. |
| 17:53.17 | WIMPy | It always is. |
| 17:53.40 | Samot | Right.. |
| 17:53.50 | Samot | But with Asterisk, endpoint A can be g711 |
| 17:54.02 | Samot | and endpoint B can be g722 |
| 17:54.10 | Samot | Asterisk will transcode the media. |
| 17:54.29 | WIMPy | With Asterisk you can't make calls between Endpoints. |
| 17:54.38 | Samot | I know. |
| 17:54.48 | Samot | With a media proxy.. |
| 17:54.58 | Samot | The same endpoints.. |
| 17:55.05 | Samot | It would fail. |
| 17:55.14 | Samot | Because the endpoints can't talk on the same codec. |
| 17:55.22 | Samot | And the media proxy *does not* transcode. |
| 17:55.49 | WIMPy | Indeed |
| 17:56.15 | Samot | Therefore it's not touching the RTP stream, really. It stays out of it. |
| 17:56.24 | Samot | It just flows through it. |
| 17:56.38 | Samot | Like the SIP signaling would. |
| 17:56.53 | WIMPy | Not touching and staying out are different things to me. |
| 17:56.58 | Samot | Fair. |
| 17:57.53 | Samot | But when Joe Schmoo the VoIP Guy sets up your trunk and just peers you directly with his VoIP Innovations account... |
| 17:58.00 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 17:58.10 | Samot | Joe Schmoo can't comply with the wiretap. |
| 17:58.19 | Samot | He has to send it up to VoIP Innovations.. |
| 17:58.26 | Samot | Who in turn don't handle the media either. |
| 17:58.27 | WIMPy | Whicht was my point. |
| 17:58.41 | Samot | The inbound will be from one carrier... |
| 17:58.54 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.82) |
| 17:59.00 | Samot | The outbound will be based on which ever carrier Joe picks in his LCR setup. |
| 17:59.15 | Samot | So again, it has to go to the carrier. |
| 17:59.21 | aiksa[LV] | Hi everyone! It's been ages since I have been here! |
| 17:59.24 | Samot | The actual entity responsible for the call. |
| 18:00.07 | Samot | I've been in meetings with Feds. |
| 18:00.39 | aiksa[LV] | I just ran into a PJSIP issue today, I cant seem to be able to explain - as it happens on 25% of all test attempts. In short - CANCEL fails, becuase of "No call / transaction found" |
| 18:00.54 | Samot | Not just for phone taps, tapping servers and Internet traffic. |
| 18:01.35 | igcewieling | CALEA doesn't apply to internet traffic, does it? |
| 18:02.09 | aiksa[LV] | anyone, knowledagble enough to discuss it with? |
| 18:02.32 | aiksa[LV] | RFC specifies that CANCEL should mimck the values of the original INVITE, right |
| 18:02.39 | aiksa[LV] | with all the tags, and everything |
| 18:02.49 | aiksa[LV] | To:, From:, Callseq, and Callid |
| 18:03.19 | igcewieling | I've seen similar issue with chan_sip when the endpoint was set for a proxy. |
| 18:03.22 | aiksa[LV] | how should it treat ;tag=... added by other party in Ringing status |
| 18:04.00 | aiksa[LV] | igcewieling: with chan_sip we had strict=yes/no or something like that which got rid of issue |
| 18:05.36 | Samot | CALEA applies to Internet traffic for digital voice... |
| 18:06.09 | Samot | But we also had a datacenter and direct backbone access, servers were for phishing scams that ended up on our servers. |
| 18:06.12 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 18:07.33 | Samot | In one case they just showed up with a warrant and took a server. |
| 18:08.33 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.82) |
| 18:08.41 | Samot | By passed the customer who was a hosting provider. Came straight to us because we were the datacenter. |
| 18:08.43 | aiksa[LV] | sorry - crappy wifi at home |
| 18:08.54 | aiksa[LV] | did I miss anything? |
| 18:10.56 | *** join/#asterisk klow (~textual@96-81-150-137-static.hfc.comcastbusiness.net) |
| 18:13.58 | aiksa[LV] | just wonder if this is something I should ask devs' about |
| 18:14.24 | aiksa[LV] | what strikes me -is that the behaviour is different in various scenarios |
| 18:14.35 | aiksa[LV] | the same scenarios |
| 18:15.08 | aiksa[LV] | sometimes it works and sometimes it does not; SIP package wise it looks exactly the same. |
| 18:16.10 | aiksa[LV] | checked out bug req database - the last issues regarding this were closed somewhere in May this year |
| 18:27.26 | OldHoster | <PROTECTED> |
| 18:27.43 | Samot | Not my server. Some end user of the customer. |
| 18:28.08 | OldHoster | Samot: i mean, it was a shared server and they took all of the server or just a snap shot? |
| 18:28.26 | Samot | I don't know. |
| 18:28.30 | Samot | It wasn't my server. |
| 18:28.33 | Samot | The customer had a rack. |
| 18:28.36 | Samot | Bandwidth.. |
| 18:28.39 | Samot | And an IP block. |
| 18:28.46 | Samot | We were the datacenter. |
| 18:29.19 | *** join/#asterisk seiggy (~seiggy@74.203.105.194) |
| 18:29.28 | aiksa[LV] | It's nice when customer have a rack - much more easy to seize. I have seen whole cluster of VMs being taken away - because of one offending instance |
| 18:29.30 | OldHoster | Samot: Right, I used to be a partner in a colo-facility and we had a good relationship with LEO's and usually they wouldn't take the entire server. And if it was a rack customer they would come to us. |
| 18:30.06 | Samot | When the FBI comes in with a warrant that says they want the server, intact, you comply. |
| 18:30.59 | aiksa[LV] | Samot - the problem starts when that server is hosting numerous customers. |
| 18:31.04 | OldHoster | Samot: I understand that, but we always had a great relationship with them. But they would ususally ask if it was a VM or shared host at any of the DC's that I have ran. |
| 18:32.15 | aiksa[LV] | our (Latvian) security services would never ask that |
| 18:32.25 | aiksa[LV] | they would just show up |
| 18:32.26 | OldHoster | Samot: We have even went as far as providing them a way during a maint window to snapshot a customers server. And helped with FISA warrants in the past. |
| 18:32.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 18:33.28 | aiksa[LV] | sounds so much more civilized than over here |
| 18:34.11 | Samot | When a customer buys a rack.. |
| 18:34.19 | Samot | We don't ask what's one each individual server. |
| 18:34.24 | Samot | Why would we? |
| 18:34.57 | Penguin | I think you've missed the entire point. |
| 18:35.01 | OldHoster | aiksa[LV]: We had a forum once(~2006/07) that they paid the bill for them to be online and we captured traffic for them which they also paid all of the bills for our hardware and services etc |
| 18:35.42 | Samot | We've let servers run.. |
| 18:35.47 | Samot | Let them collect data.. |
| 18:36.10 | Samot | One instance of them taking a server directly in 22 years. |
| 18:36.11 | OldHoster | We just either do a port mirror or provide netflows or both long with what we had in our archives. |
| 18:36.17 | Samot | That's my experience. |
| 18:36.25 | Samot | The rest of the time, it was monitoring or data dumps. |
| 18:36.36 | OldHoster | Samot: I have had them take entire servers, but never take an entire VM node. |
| 18:36.52 | Samot | I didn't say it was a VM node. |
| 18:37.02 | Samot | I don't know if it was a VM node. |
| 18:37.15 | Samot | It was a server that an end user of the customer who owned the rack put in. |
| 18:37.39 | OldHoster | I had a 20x20 cage that was sized once and before they took all of the equipment they paid all of the bills for the rest of the contract before removing everything. |
| 18:37.56 | Samot | Like I said, it was once. |
| 18:38.15 | Samot | The rest of the time we had to deal with it, it was logs/data dumps.. |
| 18:38.21 | Samot | Letting the traffic run to be monitored.. |
| 18:38.30 | aiksa[LV] | OldHoster: I meant taking the whole VM cluster because they suspected that on one of the nodes something bad might be going on. |
| 18:38.33 | OldHoster | Samot: Then from my experience they either had flows, a compromise on the machine(have seen many times), or something to show that was basically a single customer box. |
| 18:38.42 | aiksa[LV] | there were like 50 other customers there |
| 18:38.49 | Samot | Well considering this was 15 years ago. |
| 18:38.53 | aiksa[LV] | with live systems, accounting tc. |
| 18:38.55 | Samot | I'm going to guess it wasn't a node. |
| 18:39.55 | *** join/#asterisk areski (~areski@80.174.128.12.dyn.user.ono.com) |
| 18:40.25 | OldHoster | Samot: We have also denied(only four times in my carrer) flows/logs for certan warrants asking for a judges signature and appealed it. |
| 18:40.44 | Samot | I've denied requests too. |
| 18:41.09 | Penguin | OldHoster: What country are you in? |
| 18:41.18 | OldHoster | Penguin: United States |
| 18:41.39 | Samot | I've also done the minimum requirements |
| 18:41.51 | Penguin | You've just indicated that a warrant was presented which had not been signed by a judge. |
| 18:41.52 | Samot | Dragged out things.. |
| 18:42.09 | OldHoster | Samot: Alwas do only what is required on the warrant. |
| 18:42.14 | Penguin | If it's not signed by a judge, it's just a piece of paper with meaningless words on it. |
| 18:42.20 | Samot | ^ Yup. |
| 18:42.21 | OldHoster | Penguin: There are FISA warrants that don't have to be signed. |
| 18:43.11 | Samot | OldHoster: I do. If the request has 5 requirements, I do those. |
| 18:44.13 | Penguin | Even a warrant from a FISC should have a federal judge's signature on it. |
| 18:44.20 | OldHoster | Samot: But off the record depending on the customer more than once we have said hey why don't you look here or here. |
| 18:44.41 | Penguin | Otherwise it's just a piece of paper. |
| 18:44.57 | Samot | Translated: There are times I've only done it once forced and would not comply behind the extent of that. Forcing them to make every request a requirement. |
| 18:45.08 | Samot | Other times, I'll give them extra. |
| 18:45.12 | Samot | It all depends. |
| 18:45.23 | OldHoster | Penguin: You are 100% right but you would be surprised what the local FBI office in small town america will try to get away with. |
| 18:46.37 | OldHoster | Or what they ask for on fiber in a larger office such as a complete trap. |
| 18:47.19 | Penguin | I find that very hard to believe. If they would serve a "warrant" on a person or entity (and I say "warrant" because it's not a valid warrant if not executed properly), there's a good chance that any evidence seized using a bogus warrant would not be admissible in court. |
| 18:47.27 | Penguin | I doubt the FBI would want to risk that. |
| 18:47.45 | Penguin | Unless the people trying to pull this off are just pure retards. |
| 18:48.09 | Samot | 99% of the time.. |
| 18:48.35 | Samot | The agents we dealt with weren't even local. |
| 18:48.41 | OldHoster | They will try to call it FISA but it is something that they try to force on you. This is long ago, but we a customer that had a wave and they had tried to get the dark fiber provider to give them a splice of the cable(bad term) and they had the gall to ask us for a port mirror of it directly off of the switch. |
| 18:49.56 | OldHoster | Samot: We actually after Enron(Had a 150sq/ft facility in downtown houston) had a ton of warrants none FISA, and I estimate thant 80% of them were local |
| 18:50.13 | Samot | Oh I mean there were local agents involved back in the day.. |
| 18:50.32 | Samot | But the division dealing with it didn't have a presence in that office. |
| 18:50.55 | OldHoster | When in Houston we didn't have a bunch of local stuff, but while in Atlanta over 70% were local to the Atlanta Computer Crimes office there, and when in VA a ton of stuff was local too. |
| 18:51.27 | OldHoster | FBI warrants anyway. |
| 18:51.52 | Samot | Luckily, in 22 years the amount of dealings with the FBI has been relatively low. |
| 18:52.55 | OldHoster | We had one customer that got 10+ warrants a month for their customers it seemed like |
| 18:53.21 | Penguin | They should choose better customers. |
| 18:53.31 | Samot | And that customer wouldn't be my customer. |
| 18:53.37 | Samot | Because of all their customers. |
| 18:54.02 | OldHoster | Samot: When they are renting over 15k servers from you they will be. |
| 18:54.20 | Samot | I guess it really depends on how much that 15K matters. |
| 18:54.34 | Penguin | The revenue would need to seriously outweigh the crap you have to put up with. |
| 18:54.45 | Samot | There does come a point where the revenue doesn't mean anything. |
| 18:54.47 | OldHoster | Many millions a month.... |
| 18:55.14 | Samot | And if that millions keeps getting you FBI visits. |
| 18:55.18 | OldHoster | Samot: When you get to customers that are in the millions a month it gets hard to turn it away just hiresomeone to deal with it. |
| 18:55.25 | Samot | Is it really worth being associated with it and dealing with it? |
| 18:55.37 | Penguin | I guess it would become routine paperwork after a while. The people show up, show the warrant, you look it over, open the door, they go in, take the stuff, come back out, say thanks have a nice day... |
| 18:56.01 | Samot | I guess. |
| 18:56.05 | OldHoster | Samot: the FBI never did bother us as long as we complied and helped them and yes we had more than one someone that delt with it. |
| 18:56.17 | OldHoster | it actually made some of our other relationships better |
| 18:57.34 | *** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za) |
| 18:58.03 | OldHoster | eg we were hosting a customer, and one of their customers was hosting a forum website related to terrorisim and because we had a good relationship they came to us and asked us for some stuff and paid us for it |
| 19:01.07 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 19:01.22 | Samot | Again, it just goes back to if the revenue is worth the headaches that come with it. |
| 19:01.43 | Samot | In larger companies, I'm sure that headache has to be really, really big. |
| 19:02.05 | Samot | In smaller ones, I'm sure they want as little headaches as possible. |
| 19:02.25 | OldHoster | Hey back on asterisk subject... I haven't messed with asterisk in quite a while personally, what is the current just off the cuff best distribution setup, we are looking to have 4-5 locations with a centralized office pbx. |
| 19:02.55 | Samot | That's going to be based on a few factors. |
| 19:03.15 | OldHoster | Samot: I have a friend that works for L3 that manages a infosec team and I know he has more than 20ppl that deal with just warrants |
| 19:03.35 | Samot | Well L3 is a carrier, that's part of business. |
| 19:04.02 | Samot | Global companies are a little different. |
| 19:04.19 | OldHoster | True |
| 19:04.38 | Penguin | I prefer packaged software so I can manage everything properly (that is to say using the package manager). And I prefer stability. So I would say probably CentOS or Debian. |
| 19:05.07 | OldHoster | Penguin: I would prefer CentOS. |
| 19:05.52 | Penguin | You can either use the packages someone else makes and throws into the repos or you can roll your own. I prefer to roll my own only when the packages in the repos don't have something I need. |
| 19:06.22 | *** join/#asterisk Chotaire (chotaire@oahu.chotaire.net) |
| 19:07.11 | OldHoster | Penguin: I was looking at incredible pbx or something freepbx just to make it simple |
| 19:08.11 | Penguin | The official FreePBX distro was built on CentOS. I haven't looked at it in a couple of years... it might still be. That would make things easy, especially if you require FreePBX. |
| 19:08.52 | Penguin | I don't need FreePBX and won't install FreePBX. |
| 19:09.13 | *** join/#asterisk Akuma (~Akuma@AYLMPQ0104W-LP130-01-1279778401.dsl.bell.ca) |
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| 19:25.44 | OldHoster | Penguin: In an larger enviroment I agree |
| 19:26.12 | OldHoster | Penguin: for 10-20 extensions hand configuring and not having someone dedicated is what I am looking for. |
| 19:28.25 | Penguin | I'm not following what you're saying. You want the users to admin their own PBX or you want an actual admin to admin the PBX? |
| 19:29.07 | OldHoster | Penguin: I am wanting someone that doesn't have a full technical knowledge of asterisk to admin it if needed and not have to hand configure everything for the simple stuff. |
| 19:46.38 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 19:47.35 | *** join/#asterisk foobar2605 (5de9c7cc@gateway/web/freenode/ip.93.233.199.204) |
| 19:47.40 | foobar2605 | hi all |
| 19:49.25 | foobar2605 | I want to setup an extension where all calls are routed to. I was successful in doing so. Now I want to signal the caller a "busy" if the extension rejects the call. How can this be achieved? |
| 19:51.13 | foobar2605 | in addition I want to signal the caller a ringing when the extension isn't registered in asterisk |
| 19:51.27 | foobar2605 | I achieved the last one so far by doing this: |
| 19:51.53 | foobar2605 | exten => XXXMYNUMBERXXX,1,Dial(SIP/10&LOCAL/00@ringing,30,Ttr) |
| 19:52.22 | foobar2605 | [ringing] exten => _00,1,Ringing exten => _00,n,Wait(30) exten => _00,n,Hangup |
| 19:52.58 | foobar2605 | now if a call comes in, the extension 10 is called and in addition the "virtual" extension "ringing" |
| 19:53.20 | foobar2605 | if 10 isn't registered the caller still hears the ringing |
| 19:53.37 | foobar2605 | but if the extension is registered and rejects the call, the caller still hears the ringing... |
| 19:58.28 | WIMPy | Do it sequentially instead of parallel. |
| 19:58.32 | [TK]D-Fender | Dial both as local channels and set a timeout on the actual dial and either set channel variable or answer in that leg to resume call processing |
| 19:58.44 | WIMPy | The 2nd (dummy) call should probably be conditional then. |
| 19:59.31 | *** join/#asterisk areski (~areski@80.174.128.2.dyn.user.ono.com) |
| 19:59.46 | foobar2605 | sequentially means by using a queue? |
| 19:59.55 | WIMPy | The usual DIALSTATUS/HANGUPCAUSE thing. |
| 20:00.16 | WIMPy | No, by first dialling the device and then continue in the dialplan. |
| 20:01.03 | aiksa[LV] | and GotoIF with ${DIALSTATUS} as switch |
| 20:01.09 | WIMPy | If it wasn't available, do the fake ringing. |
| 20:01.28 | aiksa[LV] | or Goto a label named as a DIALSTUS or HANGUPCAUSE |
| 20:01.38 | foobar2605 | you mean exten => XXXMYNUMBERXXX,1,Dial(SIP/10,30,Ttr) exten => XXXMYNUMBERXXX,2,Dial(LOCAL/00@ringing,30,Ttr) |
| 20:01.56 | aiksa[LV] | Dont do that local channel |
| 20:02.06 | aiksa[LV] | it will just mess with your head |
| 20:02.16 | WIMPy | Yes, just that you don't want to do it e.g. when the device was busy. |
| 20:02.20 | *** join/#asterisk miralin (~Thunderbi@85.115.248.166) |
| 20:02.29 | foobar2605 | hm |
| 20:02.40 | WIMPy | An, yes, there's no need for the 2nd dial any more. Just do it in place. |
| 20:03.47 | aiksa[LV] | exten=>_XXX.,1,Ringing() exten=>_XXX.,n,Dial(SIP/10,30,Ttr) exten=>_XXX.,n,Goto(${HANGUPCAUSE}) exten=>_XXX.,n(BUSY),Goto(1) |
| 20:04.03 | aiksa[LV] | hope you get the idea |
| 20:04.26 | aiksa[LV] | add Wait(1) somewhere as a step as well so you dont flood the preocessing |
| 20:04.40 | WIMPy | I'd go fo a simple if HANGUPCAUSE=BUSY Hangup. |
| 20:04.54 | WIMPy | And then do the ringing and wait. |
| 20:05.48 | aiksa[LV] | I think he wants call to countinue if BUSY, and only signal BUSY if rejected |
| 20:05.54 | WIMPy | The story reminds me that I've said that Dial needs a BOB feature for ages. |
| 20:06.16 | aiksa[LV] | would have to check how exactly DECLINE is manifested in HANGUPCAUSE |
| 20:06.42 | foobar2605 | exactly |
| 20:08.02 | foobar2605 | AST_CAUSE_CALL_REJECTED 21 |
| 20:08.04 | foobar2605 | ? |
| 20:08.12 | aiksa[LV] | look like one |
| 20:08.26 | WIMPy | It depends on the called device. |
| 20:08.38 | aiksa[LV] | just test it |
| 20:08.58 | aiksa[LV] | make a call, reject and Noop HANGUPCAUSE |
| 20:08.59 | foobar2605 | hm |
| 20:09.03 | foobar2605 | tricky |
| 20:09.17 | WIMPy | should put the BOB feature on his to-do list. It can't be that hard to add. |
| 20:09.33 | aiksa[LV] | BOB? |
| 20:09.47 | WIMPy | Busy On Busy |
| 20:09.56 | aiksa[LV] | Ok. |
| 20:09.57 | foobar2605 | will try with the noop... |
| 20:10.20 | aiksa[LV] | pjsip is killing me softly today :) |
| 20:10.46 | aiksa[LV] | SIP/2.0 481 Call/Transaction Does Not Exist |
| 20:11.21 | aiksa[LV] | on exetrnal CANCEL |
| 20:11.36 | aiksa[LV] | and happens sometimes, under exactly the same conditions |
| 20:13.15 | WIMPy | How exactely is exactely? |
| 20:14.49 | aiksa[LV] | almost identical; |
| 20:14.52 | aiksa[LV] | the setup: |
| 20:15.18 | aiksa[LV] | one trunk connection |
| 20:15.40 | aiksa[LV] | call gets recieved, extension modified (synbol stripped) and sent back to the same trunk |
| 20:15.49 | WIMPy | I'd go and write a script to compare the situations. It's far too easy to miss tiny differences when there have to be some differences anyway. |
| 20:15.50 | foobar2605 | hangupcause was 17... |
| 20:15.57 | aiksa[LV] | one liner basically |
| 20:16.14 | aiksa[LV] | then under no other calls or any other external influence |
| 20:16.41 | aiksa[LV] | the call gets sent in (from Nokia's MSS) and then canceled |
| 20:16.52 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 20:17.01 | aiksa[LV] | same source number |
| 20:17.04 | aiksa[LV] | same destination number |
| 20:17.10 | aiksa[LV] | always |
| 20:17.21 | aiksa[LV] | in some cases the cancel is processed as expected |
| 20:17.24 | *** join/#asterisk tzafrir (~tzafrir@37.26.147.191) |
| 20:17.28 | aiksa[LV] | and in some cases I get that |
| 20:17.43 | aiksa[LV] | checked against original INVITE; |
| 20:18.17 | aiksa[LV] | CANCEL uses exactly same From:, Call-Id; Seq number, with all tags and branches and shit |
| 20:18.32 | aiksa[LV] | also same To: as original INVITE as per RFC |
| 20:19.03 | aiksa[LV] | but for some mystical reason pjsip fails to find the channel it is addressed for |
| 20:19.17 | *** join/#asterisk klow (~textual@96-81-150-137-static.hfc.comcastbusiness.net) |
| 20:19.36 | aiksa[LV] | And I know - there is no such thing as voodoo with code |
| 20:19.51 | WIMPy | If you're sure there are no differences, open a ticket on issues.asterisk.org. |
| 20:20.18 | igcewieling | hugs chan_sip |
| 20:20.31 | aiksa[LV] | I am gathering debug output right now |
| 20:23.07 | *** join/#asterisk foobar2605 (5de9c632@gateway/web/freenode/ip.93.233.198.50) |
| 20:27.47 | foobar2605 | thanks a lot for your help |
| 20:27.55 | foobar2605 | the thing with the hangupcause solved my problem |
| 20:28.52 | foobar2605 | I'm doing a Dial(extension) first, then a GotoIf(Hangupcause=17)?busy, then a Dial(virtual) |
| 20:29.13 | foobar2605 | and of course a (busy),Busy |
| 20:31.06 | foobar2605 | if the device is now connected and rejects the call, a busy will be returned to the caller. if the extension is not connected, the virtual phone rings |
| 20:31.23 | WIMPy | Well, as we pointed out, there's no need for that dial and the extra extension. Just do a Ringing() and a Wait(). |
| 20:31.53 | foobar2605 | ah ok |
| 20:31.57 | foobar2605 | right |
| 20:32.39 | aiksa[LV] | WIMPy, just did a test again |
| 20:32.57 | aiksa[LV] | now - CANCEL fails on 5 calls , and works on one random one |
| 20:33.24 | aiksa[LV] | SIP packages exactly the same, with a diffrence that IDs are obviously different in each acse |
| 20:33.28 | aiksa[LV] | each case |
| 20:33.39 | WIMPy | So you have a random distribution of randomness? :-) |
| 20:33.53 | aiksa[LV] | yes kinda; |
| 20:34.06 | aiksa[LV] | looking at geomagnetic storm data |
| 20:34.17 | aiksa[LV] | damn you sun! |
| 20:34.29 | aiksa[LV] | sounds like bugtracker |
| 20:35.00 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 20:35.15 | aiksa[LV] | will only have to do a replace of sensitive names in log file though |
| 20:35.22 | foobar2605 | another question now comes to my mind: what if the extension reconnects to asterisk while a call is coming in. then it won't be dailed. is there a chance to change this? |
| 20:35.51 | aiksa[LV] | if you do Goto(1) |
| 20:36.09 | aiksa[LV] | the call will always loop untill Busied or cut by the timer |
| 20:36.11 | *** join/#asterisk tzafrir (~tzafrir@37.26.148.181) |
| 20:36.32 | aiksa[LV] | and every time it gets to the specific Dial it will attempt to coonect again |
| 20:36.40 | foobar2605 | aha |
| 20:37.18 | WIMPy | Yes, just retr as long as the call exists. But do add a Wait() between attepmts or you might get seriousely spammed when you ever try to debug anything. |
| 20:37.22 | foobar2605 | so do a Dial(SIP/XY),30,Ttr) then a GotoIf, then ringing, then wait(1), then goto 1, right? |
| 20:37.44 | *** join/#asterisk areski (~areski@80.174.128.5.dyn.user.ono.com) |
| 20:37.51 | WIMPy | Jepp |
| 20:39.02 | foobar2605 | but if i goto 1 will it ever stop after 30 seconds? |
| 20:39.10 | foobar2605 | i mean if the extension won't register at all |
| 20:39.23 | WIMPy | No. It would stop when the caller hangs up. |
| 20:39.50 | foobar2605 | yea, but i want to let it ring just for 30 seconds |
| 20:40.35 | WIMPy | Then you have to either add a counter or check the time at the beginning and before every attempt. |
| 20:41.15 | foobar2605 | hm |
| 20:41.18 | foobar2605 | just tried it and it doesn't work |
| 20:41.27 | foobar2605 | if i reconnect the extension it won't be called |
| 20:41.46 | foobar2605 | and I get lot's of app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) in my log |
| 20:41.54 | WIMPy | Does it have qualify enabled? |
| 20:41.59 | foobar2605 | yes |
| 20:42.08 | WIMPy | 20 is subscriber absent. |
| 20:42.42 | WIMPy | With qualify the divice will become available again after the next successfull qualify attempt. |
| 20:42.49 | WIMPy | Whenever that may happen. |
| 20:42.56 | foobar2605 | hm |
| 20:43.09 | foobar2605 | <PROTECTED> |
| 20:43.13 | WIMPy | If you want Asterisk to always try (and time out), you need to disable qualify. |
| 20:43.17 | foobar2605 | Peer 'XYZ' is now Reachable. (20ms / 2000ms) |
| 20:43.29 | foobar2605 | but when I call the number I still get unavailable |
| 20:43.44 | WIMPy | If you have qualify enabled, it will check the status before attempting to call. |
| 20:44.08 | WIMPy | Ther must be another issue then. |
| 20:44.24 | foobar2605 | if i restart asterisk it works again |
| 20:45.22 | foobar2605 | woops...I now get a Forbidden - wrong password on authentication for REGISTER |
| 20:46.13 | foobar2605 | seems my service provider doesn't like to get a register that often |
| 20:46.13 | WIMPy | Definitely another issue. But I'm AFK now. |
| 20:46.25 | foobar2605 | ok, thanks a lot so far! |
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| 23:02.33 | dan_j | wiki seems to be running a little slow tonight |
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| 23:05.36 | *** join/#asterisk SupaYoshi (~SupaYoshi@104.223.1.186) |
| 23:06.40 | dan_j | In PJSIP, in the peer definition that has type=identify, can the match field contain wildcards? such as 10.10.0.% |
| 23:07.20 | newtonr | Nope, just CIDR I believe |
| 23:08.00 | newtonr | If you look at the command line config help for the key it should tell you what is acceptable |
| 23:08.58 | newtonr | Yeah the wiki isn't loading for me.. |
| 23:09.21 | scv | been down for a bit for me as well |
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| 23:11.00 | dan_j | newtonr: Has type=identify been removed? |
| 23:11.07 | file | no |
| 23:11.31 | dan_j | I can see it in the same config files, but not in the 'config show help'. Nor in the wiki. |
| 23:11.45 | dan_j | same should say sample |
| 23:12.12 | file | it's in the res_pjsip_endpoint_identifier_ip module |
| 23:12.19 | dan_j | Thanks |
| 23:12.27 | file | "config show help res_pjsip_endpoint_identifier_ip identify" outputs the deets |
| 23:12.29 | dan_j | Just found it in the wiki too |
| 23:12.38 | dan_j | Thank you |
| 23:13.21 | newtonr | Wiki is loading normally for me now. scv and dan_j is it the same for you? |
| 23:13.41 | dan_j | Seems to be ok now. |
| 23:13.51 | scv | looks good now |
| 23:14.03 | newtonr | weird |
| 23:14.13 | scv | it was connecting, just stalled out mid-session |
| 23:14.25 | file | it's sporadic |
| 23:14.27 | scv | maybe a rogue cronjob or summat ;) |
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| 23:14.55 | dan_j | I take it back. Still slow |
| 23:16.54 | dan_j | If I've got a group of internal servers that need to dial through asterisk, and i set match=192.168.0.0/24, do i still need to specify an aor and contract=something? Follow on question, do i need to specify peer info for every server, rather than making one for all servers to use? |
| 23:19.21 | scv | i'd like to hear the answer to this too actually, i wasn't able to figure out a similar issue and kinda hacked around it |
| 23:19.36 | file | an aor and contact is for outgoing |
| 23:19.49 | file | you can have multiple going to a single endpoint |
| 23:19.54 | file | for inbound traffic |
| 23:20.06 | file | and it's fine if the endpoint has no aor associated with it |
| 23:20.45 | scv | what about dialing to arbitrary peers |
| 23:20.56 | file | SIP URI can be specified in the dial string |
| 23:21.02 | file | PJSIP/endpoint/<SIP URI> |
| 23:21.07 | file | so you still don't need an AOR |
| 23:21.12 | scv | yeah i'm doing it like that now |
| 23:21.21 | scv | so that's correct then ok |
| 23:21.27 | file | yes |
| 23:24.00 | dan_j | Perfect. Thanks. |
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| 23:38.21 | Penguin | Anyone here using pfSense 2.3.2 as a NAT router with Asterisk behind it? |
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