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00:09.57 | jameswf | ${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor()) |
00:12.58 | Mango45 | Thanks! |
00:18.26 | Mango45 | Curiously, that doesn't work for me. The variable is empty. |
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00:48.10 | mushroomed | Can someone help me creating a simple ConfBridge? I don't want to type PINCODES, I just want the functionality |
00:48.58 | mushroomed | I've already read voip-info.org ConfBridge cmd but it's too extense, I wan't to keep it simple |
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01:51.11 | WIMPy | mushroomed: There isn't much to do. You just need a default template for users and conferences in confbridge.conf and tehn just call it. |
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02:57.22 | ChannelZ | It's so easy even a caveman could do it |
02:58.03 | MaliutaLap | ChannelZ: That's the case with most * stuff though ... if people would read the damn docs |
02:58.40 | WIMPy | There are plenty if issue without docs explaining what's going on. |
03:01.17 | MaliutaLap | WIMPy: for dumb people ;) |
03:01.46 | MaliutaLap | I really should get that "I see dumb people" shirt |
03:01.46 | WIMPy | must ver very dumb then. |
03:02.42 | MaliutaLap | WIMPy: you seem pretty clued in from what I have seen |
03:03.03 | WIMPy | Definitely not enough. |
03:04.02 | MaliutaLap | WIMPy: one can never be clued in "enough", there is always something more to learn ... it's one of the reasons I like IT work |
03:04.15 | MaliutaLap | not that I like the people I do the work for |
03:04.37 | WIMPy | Only if there is a solution in sight. |
03:04.49 | MaliutaLap | but that's just me and "people". Doing tech support for a couple of years will do that to one |
03:05.12 | MaliutaLap | WIMPy: bend the tech to your will and there is almost always a solution :) |
03:05.49 | WIMPy | The trouble is finding the right spot to bend. |
03:06.13 | WIMPy | And My Asterisk regularly fails after upgrading because I forgot one of the patches. |
03:06.27 | MaliutaLap | what are you patching for? |
03:07.06 | WIMPy | A few little things, but the one I tend to forget is the one that makes outbound calls possible. |
03:07.30 | MaliutaLap | patch for some odd kind of card? |
03:07.44 | WIMPy | No, SIP. |
03:08.15 | WIMPy | My ITSP kills calls if it gets a NAK for AOC messages. |
03:09.58 | MaliutaLap | so you're patching for a fecked up ITSP ... great |
03:10.21 | WIMPy | I consider SIP fucked up by design. |
03:10.26 | MaliutaLap | what have they done, forked SIP and created their own custom version? ... or is just crappy |
03:10.40 | WIMPy | Everyone does so. |
03:11.08 | MaliutaLap | I've been through a few providers and haven't found anything too bad |
03:11.08 | WIMPy | You know what's great about standards? |
03:11.16 | WIMPy | Everyone has his own. |
03:11.28 | WIMPy | That's nowere as true as for SIP. |
03:11.56 | WIMPy | So I consider SIP interoperability pure luck. |
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03:13.06 | WIMPy | I mean, pure SIP is rather useless. So it comes down to what extensions are implemented and in what way. |
03:13.34 | MaliutaLap | pffft SMTP interoperability is a freakin' miracle |
03:13.52 | MaliutaLap | and SMTP has been around a lot longer than SIP |
03:14.02 | WIMPy | Compared to SIP that ALWAYS works. |
03:14.21 | MaliutaLap | laughs |
03:14.25 | MaliutaLap | not always |
03:14.46 | MaliutaLap | there are some pretty screwed up SMTP implementations out there |
03:14.47 | WIMPy | Seriousely, what kind of troble do you have with SMTP? I can't remember anythign I came across. |
03:16.00 | MaliutaLap | Oh, you see issues with people not responding to greylisting properly, and most of the code in stuff like postfix is actually to detect and deal with borked implementations of a [simple] standard |
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03:16.50 | MaliutaLap | people not responding to "temporarily unavailable" codes is becoming more common |
03:17.01 | WIMPy | I have seen implementations that allow obvious spam while filtering legit mail for bogus reasons, but that's not part of SMTP. |
03:18.36 | MaliutaLap | that has more to do with peoples configurations |
03:18.49 | WIMPy | Indeed |
03:18.51 | MaliutaLap | and the spam filtering software they use ontop of that |
03:19.35 | MaliutaLap | but things like exchange ignore parts of the protocol, at least in terms of what comes before the '@' symbol |
03:20.19 | WIMPy | You don't seriousely expect anythign form Microsoft to be compatible to known standards, do you? |
03:20.42 | MaliutaLap | no, that's why I don't touch any of their shit |
03:22.58 | MaliutaLap | I like to say: "Windows are for opening up and jumping out of, 1929 stock broker style, as a last resort" |
03:25.58 | WIMPy | wonders what Asterisk would be... |
03:26.31 | MaliutaLap | a pixie in sky? ;) |
03:27.12 | WIMPy | Probably the sledge in a time when the wheel not available. |
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07:15.54 | bhavikpatel6842 | Hello everyone |
07:16.16 | bhavikpatel6842 | I am facing one issue regarding webrtc using ASterisk |
07:16.34 | bhavikpatel6842 | I am trying to integrate SIPml5 demo. |
07:16.42 | bhavikpatel6842 | my Asterisk version is 11.9 |
07:17.20 | bhavikpatel6842 | I am not able to sip to sip calls using two chrome browser which is latest version. |
07:17.39 | bhavikpatel6842 | I use this link for reference p, li { white-space: pre-wrap; } https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
07:17.58 | bhavikpatel6842 | configuration and settings |
07:17.58 | bhavikpatel6842 | http://pastebin.com/7KCvtcNf |
07:18.25 | bhavikpatel6842 | Sip dialogs for |
07:18.25 | bhavikpatel6842 | incoming calls : http://pastebin.com/ykeS0JCi |
07:18.25 | bhavikpatel6842 | and outbound call log : |
07:18.25 | bhavikpatel6842 | <PROTECTED> |
07:18.35 | bhavikpatel6842 | Let me know if I am doing wrong. |
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08:00.50 | songoku1610 | hi |
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10:18.38 | cusco | hi |
10:21.33 | cusco | I must be missing something to add a member to a queue with hint'ed state interface |
10:22.02 | cusco | queue show <queue> shows: 150 (Local/150@agents) (dynamic) (Invalid) has taken no calls yet |
10:23.26 | cusco | is there a queue option needed? |
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11:11.09 | margeas | hello ppl, I need to set-up the SIP caller-Id in asterisk11 in e164 format, and the service provider is expecting a SIP header like this "From: "447937947990" <sip:APIKEY@sip.nexmo.com>". I'm using the SipAddHeader() dialplan function but it seems I end up with *two* From: header lines in the sip dialogs, one added automatically by asterisk, the second it's the SipAddHeader one.How can I replace the one's added by asterisk, |
11:11.10 | margeas | without adding another one? |
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12:12.23 | margeas | Does anybody know how I can "replace" the "From:" sip header before calling a peer? thank you |
12:21.01 | file | you can't replace it but you can configure in sip.conf aspects of it |
12:21.17 | file | fromuser to control the user portion, fromdomain to control the domain part |
12:27.26 | margeas | file: could you please look at https://docs.nexmo.com/index.php/voice-api/sip, the callerid portion... |
12:28.28 | file | they've given you the portion for domain and APIKEY, looks like they are using the name portion to convey number |
12:28.41 | file | so you can set that using Set(CALLERID(name)=447937947990 |
12:28.46 | file | in the dialplan before dialing |
12:29.05 | margeas | file: ok let me try.. |
12:34.12 | cusco | taking the oportunity that there is som activity here... |
12:34.29 | cusco | queue show <queue> does not show member with hint |
12:34.38 | cusco | I don't quite understand why |
12:34.43 | cusco | and shows as Invalid.. |
12:34.53 | cusco | if I set state_interface to SIP/bla i'ts ok |
12:34.58 | cusco | but to hint:ext@context |
12:35.04 | cusco | doesn't show in the queue |
12:35.06 | cusco | .... |
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12:37.30 | margeas | file: after adding this to the dialplan: same => n,Set(CALLERID(name)=447937947990) the From header is From: "Anonymous"<sip:74f2ecf3@anonymous.invalid>;tag=as383edf3d back from nexmo and I get a SIP/2.0 401 Unauthorized |
12:39.20 | margeas | it's like if the callerid name get not set...while if I add the number in the name portion of my phone set, I get authenticated and the call is successful |
12:40.29 | file | add Set(CALLERID(name-pres)=allowed) |
12:42.39 | margeas | file: THANK YOU! It's finally working...what is that name-pres? I haven't found nothing related in the docs... |
12:42.46 | file | it's presentation |
12:43.29 | file | callerid can be marked with different presentation values, so for example if someone blocks their callerid it may traverse the PSTN with their callerid until it reaches the end... where it is then supposed to not send it to the end device |
12:45.17 | margeas | so I guess the default is to *not* present the callerid which is why my number was not passing through? |
12:45.45 | file | it's not, without knowing more about the setup I couldn't say why |
12:46.22 | margeas | file: well It's all there in that link from nexmo |
12:46.40 | file | that's only outgoing |
12:46.44 | file | that's not inbound |
12:46.58 | file | cusco, what is your queues.conf and the output of queue show <name>? |
12:47.37 | margeas | file: I don't have an inbound number with them so my configuration is set-up to use nexmo only as peer.... |
12:47.54 | margeas | type=peer |
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12:47.59 | file | margeas, what is initiating the call? |
13:02.13 | margeas | file: just a moment... |
13:03.26 | cusco | file: queue is configured in realtime database |
13:03.41 | cusco | 999 (Local/999@agents) with penalty 2 (dynamic) (Invalid) has taken no calls yet |
13:04.07 | cusco | would queues.conf still be important? |
13:04.31 | file | no, but it just means I can't reproduce it without a lot of effort |
13:04.39 | file | from config file I can say that looks different and does work |
13:04.50 | cusco | I see but I can select its configuration from sql |
13:04.56 | file | hint:1000@users (Local/1000@users from Local/1000@users) (ringinuse disabled) (Not in use) has taken no calls yet |
13:05.24 | cusco | yes I don't know why it doesn't show hint in queue show <name> |
13:06.07 | cusco | also queue members are configured in realtime sql |
13:06.14 | cusco | I tried adding it dynamically |
13:06.20 | cusco | but same |
13:06.30 | file | what version of Asterisk? |
13:06.38 | cusco | well yea.. 1.6.2 |
13:06.39 | cusco | :( |
13:06.45 | file | uhhhh |
13:06.51 | cusco | hides |
13:07.16 | file | 1.6.2 app_queue does not have support for using hints like that |
13:07.40 | cusco | but I found on the interwebs people using hints in 1.6.2 |
13:07.49 | cusco | well would there be anotehr way? |
13:07.51 | file | hints exist |
13:07.59 | file | but app_queue does not support using them |
13:08.03 | file | (in that version) |
13:08.04 | cusco | at the moment we use group_count in dialplan ... |
13:08.48 | cusco | we tried migrating to 11 |
13:08.55 | cusco | but had to rollback |
13:09.02 | cusco | need to try that again :/ |
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13:35.03 | Ricmacas | Hello everybody, I'm trying to send a sip MESSAGE through a trunk proxy |
13:35.21 | Ricmacas | and since I'm a noobie, I have no idea, can anyone give me a hand? |
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14:05.07 | Ricmacas | Uhm, should I repeat my question? |
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16:48.48 | Kobaz | hmm |
16:49.00 | Kobaz | is there an easy way to play generic comfort noise just all by itself |
16:49.18 | Kobaz | of course you can find some and stick it in music on hold |
16:49.28 | Kobaz | but is there a way to generate it |
16:50.11 | [TK]D-Fender | * doesn't generate it |
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16:54.07 | Kobaz | do de do |
16:54.13 | Kobaz | building an outbound dialer |
16:54.33 | Kobaz | i have until july |
16:54.52 | Kobaz | should be able to make a kick-butt dialer by then |
16:57.16 | Kobaz | http://soundbible.com/suggest.php?q=background+noise&x=0&y=0 |
16:57.18 | Kobaz | kind of close |
17:01.33 | [TK]D-Fender | totally different. |
17:01.44 | Kobaz | yeah but i can play with it |
17:01.46 | [TK]D-Fender | CNG is generated by the client to cover missing audio |
17:01.54 | [TK]D-Fender | that isn't really "background" |
17:01.57 | Kobaz | i'm not using it for that |
17:02.20 | [TK]D-Fender | You really haven't been too clear on what you actually intend |
17:02.33 | Kobaz | like, an agent is going to dial into a bridge basically |
17:02.48 | Kobaz | and then be connected with calls that the dialer finds |
17:02.59 | Kobaz | so i want some sort of subtle indication that the agent is connected to the system |
17:03.28 | Kobaz | and it's not going to be MusicOnHold(default) |
17:03.35 | Kobaz | that will drive people insane |
17:05.10 | [TK]D-Fender | Sounds like you just want non-music..... music |
17:05.16 | Kobaz | yeah |
17:05.18 | Kobaz | exactly |
17:05.22 | [TK]D-Fender | that it's a thing... it's your set-list |
17:05.22 | Kobaz | like comfort noise |
17:05.39 | [TK]D-Fender | Go by some cricets, record them and feed a local iguana later |
17:05.46 | Kobaz | hehe |
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17:09.31 | Kobaz | where the heck is the option to convert from stereo to mono in audacity |
17:10.03 | Kobaz | oo |
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17:10.04 | Kobaz | there it is |
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17:40.06 | hayden | anyone using pjsip in production already? |
17:42.47 | file | I have heard of people using it. |
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18:07.29 | Ricmacas | [14:35] <Ricmacas> Hello everybody, I'm trying to send a sip MESSAGE through a trunk proxy [14:35] <Ricmacas> and since I'm a noobie, I have no idea, can anyone give me a hand? |
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18:10.49 | [TK]D-Fender | Ricmacas: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend |
18:11.12 | Ricmacas | [TK]D-Fender: That's what I'm using |
18:11.28 | Ricmacas | though when I attempt to send the message, it sends it "outside" the proxy |
18:12.20 | [TK]D-Fender | PASTBIN is your friend.... show us all of the related bits & debug |
18:12.22 | [TK]D-Fender | ~pb |
18:12.22 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
18:12.24 | [TK]D-Fender | ^^^ |
18:12.42 | Ricmacas | Oh, I'm actually debugging with Wireshark |
18:12.54 | Ricmacas | Which I'm not sure if its fine |
18:13.15 | Ricmacas | but what happens is quite simple, the host sms.sip.provider.com doesnt exist |
18:13.32 | Ricmacas | but if i launch the provider's sip client, it communicates with it through the proxy |
18:13.50 | [TK]D-Fender | * SIP debug is all you should be using |
18:14.09 | Ricmacas | so, i cant use MessageSend(number@sms.sip.provider.com) because that will attempt to resolve sms.sip.provider.com |
18:14.26 | Ricmacas | which isnt even a public domain, its only accessible through the trunk's proxy |
18:15.33 | Ricmacas | which requires authentication, of course. |
18:15.54 | [TK]D-Fender | Don't specify the host then, use a peer |
18:16.05 | Ricmacas | Ok, how so? |
18:16.17 | [TK]D-Fender | change what you put after the @ |
18:16.34 | [TK]D-Fender | and make a peer for it |
18:17.53 | Ricmacas | Uhm... |
18:18.32 | Ricmacas | You mean that I create a trunk for that domain? |
18:18.46 | [TK]D-Fender | [14:16][TK]D-Fenderand make a peer for it |
18:19.12 | [TK]D-Fender | sip.conf. [thisisapeer] type=peer ... |
18:19.17 | Ricmacas | I have a sip.provider.com peer |
18:19.24 | [TK]D-Fender | that is a HSOTNAME |
18:19.36 | [TK]D-Fender | not an asterisk SIP peer entry in sip.conf |
18:19.43 | Ricmacas | let me be more specific |
18:19.49 | [TK]D-Fender | [14:19][TK]D-Fendersip.conf. [thisisapeer] type=peer ... |
18:19.55 | [TK]D-Fender | MAKE THIS |
18:19.57 | Ricmacas | I have a [provider] type = peer |
18:20.00 | Ricmacas | ok no need for caps |
18:20.22 | [TK]D-Fender | and then use it instead of a hostname on your MEssage() |
18:20.33 | Ricmacas | so i'd use |
18:20.42 | Ricmacas | number@peer? |
18:20.51 | [TK]D-Fender | yes. |
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18:31.08 | mic_ | hello |
18:31.17 | mic_ | the pain of dtmf length when a call is bridged |
18:31.30 | mic_ | when asterisk intercepts dtmfs and generates new ones |
18:31.46 | mic_ | should I jump off the bridge before attempting to fix that/ |
18:31.47 | mic_ | ? |
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18:38.27 | puzzled | hi |
18:40.00 | puzzled | Anyone have an idea why Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})}) would give this error: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: s != |
18:40.31 | puzzled | I'm using asterisk 11.9.0 |
18:40.49 | [TK]D-Fender | Show us the actual call. |
18:41.31 | puzzled | [TK]D-Fender: hi, log or dialplan? |
18:41.41 | [TK]D-Fender | actual call |
18:42.31 | puzzled | ok, just a sec |
18:44.48 | [TK]D-Fender | While you're at it.. why not the full dialplan around it as well... |
18:45.03 | puzzled | [TK]D-Fender: http://pastebin.com/eZ6KQWbK |
18:45.53 | [TK]D-Fender | Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})}) <- this is certainly pointless given where you are |
18:47.31 | [TK]D-Fender | And that is an unexpected end... because it returns nothing |
18:48.09 | puzzled | [TK]D-Fender: dialplan snippet http://pastebin.com/xTyLXeCJ |
18:48.37 | [TK]D-Fender | didn't end up needing for the above assessment... |
18:50.12 | puzzled | [TK]D-Fender: I'm afraid I don't get your assessment. Why pointless? EXTEN is 8099 and I want to deduct SAFE_EXTEN. Just like in the README.seriously. |
18:50.35 | [TK]D-Fender | "EXTEN is 8099" ... no ... no it isn't |
18:51.23 | puzzled | you mean because of the hand-off via DSTEXTEN to the database context? |
18:51.47 | [TK]D-Fender | Because of the dialplan extension |
18:52.45 | puzzled | sorry I don't get what you mean |
18:53.33 | [TK]D-Fender | <PROTECTED> |
18:54.43 | [TK]D-Fender | hint : it's immediately after the first [ |
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18:57.48 | puzzled | [TK]D-Fender: got it, thanks |
18:58.43 | [TK]D-Fender | Never forget what ${EXTEN} is..... |
18:59.49 | [TK]D-Fender | same => n,Set(EXTEN=${DSTEXTEN}) <- and that it's READ-ONLY ... which makes this pointless. |
19:00.48 | puzzled | got it, the read-only part escaped me |
19:00.54 | [TK]D-Fender | And you have a better variable you could be using... |
19:01.17 | [TK]D-Fender | except that if your coding is expected to be consistant, you never needed to set it like that at all |
19:01.27 | [TK]D-Fender | You should learn to pass args to your Gosubs.... |
19:03.41 | [TK]D-Fender | http://pastebin.com/y8w3KUR6 |
19:04.19 | [TK]D-Fender | But then another idea arises... look where you're using it in the first place... from a pattern that can never fail the filter. Do you have it from one that should? |
19:04.25 | [TK]D-Fender | Somewhere else.... |
19:05.06 | [TK]D-Fender | AKA "where am I calling this from that can actaully fail and validates the filter at all?" |
19:05.27 | puzzled | nope I was just playing with new stuff in 11. Thanks for the Gosub snippet. That's much easier |
19:05.44 | puzzled | if I allow guests than that's going to be at the top :) |
19:05.45 | [TK]D-Fender | Gotta read your apps instructions :) |
19:06.00 | puzzled | very much so. mea culpa |
19:07.13 | [TK]D-Fender | It's a question of validating whether any of your patterns necessitate a filter at all. "guest" doesn't imply much by itself |
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19:10.19 | puzzled | hmm the core show do suggests Gosub(database,s,1(${EXTEN})) unless I'm reading that wrong |
19:10.43 | puzzled | Gosub([[context,]exten,]priority[(arg1[,...][,argN])]) |
19:11.03 | [TK]D-Fender | Yes, bracket was called for there, not commas |
19:11.07 | [TK]D-Fender | my bad on that one |
19:11.29 | puzzled | no problem. it made me read stuff and question it so that's a good thing :) |
19:11.37 | [TK]D-Fender | I mirrored Macro syntax reflexively |
19:11.59 | [TK]D-Fender | Crying shame they couldn't have actually done it tht way... |
19:12.47 | puzzled | I haven't seen that notation before. not very consistent especially coming from 1.4, 1.2 and ancient versions |
19:15.00 | [TK]D-Fender | one of the 1.6 branches changed Gosub's syntax to includ args and that's hot they chose to do it. |
19:15.03 | [TK]D-Fender | bad call. |
19:17.41 | puzzled | yup, guess I'll be doing a lot of core show application ... |
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