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01:29.30 | SpaceInvaders | Anyone around? |
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01:34.00 | ChannelZ | ish |
01:36.00 | WIMPy | Better than some *ism. |
01:37.29 | SpaceInvaders | I was having trouble with the instructions in the 3rd ed of the definitive guide. They were wrong but I figured it out. |
01:38.12 | WIMPy | There's a 4th one |
01:38.29 | SpaceInvaders | It said you need to create an entry under [Services] but the context used for the samples is [Local-something]--which I matched and it works |
01:38.37 | SpaceInvaders | Yeah I couldn't find the 4th one |
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01:39.00 | WIMPy | Nether of those are any Asterisk internals. |
01:39.07 | WIMPy | It's all what YOU call it. |
01:39.21 | SpaceInvaders | True but I've been following the instructions |
01:39.36 | SpaceInvaders | so far they have used a consistent context |
01:40.03 | SpaceInvaders | Then under voicemail they changed it without warning. |
01:40.36 | WIMPy | Voicemail contexts are independent of others. |
01:41.54 | SpaceInvaders | In this case it's the dialplan context for allowing users to dial an extension (8500) to access voicemail |
01:42.01 | SpaceInvaders | The instructions say to create the following entry |
01:42.09 | SpaceInvaders | [Services] |
01:42.09 | SpaceInvaders | exten => *98,1,VoiceMailMain() |
01:42.25 | SpaceInvaders | wrong context, wrong extension :) |
01:42.48 | SpaceInvaders | but if you match with the context used earlier and change the *98 to 8500 it works fine |
01:43.15 | WIMPy | What's "wrong" with the extension? |
01:43.18 | SpaceInvaders | The good news is I'm starting to get a basic understanding of how Asterisk works |
01:43.32 | SpaceInvaders | the instructions say dial 8500 to access voicemail |
01:43.35 | SpaceInvaders | the entry is |
01:43.39 | SpaceInvaders | exten => *98,1,VoiceMailMain() |
01:43.51 | SpaceInvaders | perhaps your system works differently than mine |
01:43.59 | SpaceInvaders | but 8500 won't get me voicemail with that entry |
01:44.16 | WIMPy | Yes, I use "voicemail" or "1155". |
01:44.21 | WIMPy | It's all up to you. |
01:44.43 | WIMPy | But I think *98 is rather common. |
01:44.55 | SpaceInvaders | Yes. I was just commenting the instructions were incorrect and I figured it out--and was happy because I'm starting to get a handle on it |
01:45.19 | WIMPy | It's good if you can help yourself. |
01:45.19 | SpaceInvaders | Asterisk so ROCKS |
01:45.26 | SpaceInvaders | agreed! |
01:45.43 | SpaceInvaders | Hey, would you happen to be familiar with Linphone? |
01:46.34 | SpaceInvaders | I keep getting this message on CLI |
01:46.35 | SpaceInvaders | [Jun 21 21:45:25] NOTICE[2813]: chan_sip.c:28144 handle_request_register: Registration from '<sip:1000@192.168.1.58>' failed for '192.168.1.54:60019' - Wrong password |
01:46.36 | WIMPy | no |
01:46.42 | SpaceInvaders | but that phone works and the pw is correct |
01:47.06 | WIMPy | Possibly a wrong type=? |
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01:47.17 | SpaceInvaders | hmmm let me check that.... |
01:48.03 | SpaceInvaders | nope type=friend |
01:48.13 | SpaceInvaders | that's what I used for all 3 |
01:48.29 | SpaceInvaders | I thought maybe it was the pw because I initially used 1234 but I changed it |
01:48.34 | WIMPy | Numeric usernames are somewhat insecure, BTW. |
01:49.00 | SpaceInvaders | Yes I need to learn more about security |
01:49.04 | SpaceInvaders | this is just a sandbox system |
01:49.11 | SpaceInvaders | and it's firewalled |
01:49.16 | WIMPy | Well, if it says wrong password, I guess it must be matching the right peer. |
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01:52.41 | SpaceInvaders | is there a command that gives you more detail than sip show peers |
01:52.49 | SpaceInvaders | like sip show 1000 |
01:52.57 | SpaceInvaders | I thought I read something like that but now can't find it |
01:53.26 | WIMPy | use your tab key :-) |
01:53.32 | SpaceInvaders | it looks good when I check it via sip show peers and users |
01:53.38 | WIMPy | sip show peer ... |
01:53.39 | SpaceInvaders | I haven't tried the tab |
01:53.53 | SpaceInvaders | SWEET!!! Thanks!!!!!! |
01:54.06 | SpaceInvaders | that's it sip show user 1000 |
01:54.50 | SpaceInvaders | ok nothing pops up that looks like I should be seeing that error :/ |
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05:04.21 | rhineheart_m | I am planning to buy licenses (G729).. but I want to hear feedbacks from the community about it.. |
05:06.36 | coppice | what kind of feedback are you looking for? they encode and decode G.729. They lock to a MAC address for licencing purposes. That's about it. |
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06:08.48 | [TK]D-Fender | [01:06]coppicewhat kind of feedback are you looking for? they encode and decode G.729. They lock to a MAC address for licencing purposes. That's about it. |
06:15.24 | coppice | [TK]D-Fender: why did you repeat that? |
06:19.25 | [TK]D-Fender | Because he timed out 4 minutes after you said it without giving any feedback like he heard it and I figured there were odds his buffer might get lost, etc |
06:22.55 | coppice | you're very thorough :-) |
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06:25.46 | [TK]D-Fender | That's just how I roll :) |
06:31.07 | [TK]D-Fender | And.... time for bed. Later all |
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07:58.56 | phasip | Is there any good & simple tool to record ivr messages from a linux computer that also sets the volume at an even level and make the recordings sound the same? |
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08:08.15 | stian | anyone care to help me out a bit? I'm trying to write a new module for superfecta, but my php knowledge is far from good. |
08:09.33 | stian | basically trying to to do lookups towards capsulecrm, however their REST API requires a username and password, and most of the superfecta modules utilize the get_url_contents function from the caller-id.php file |
08:15.30 | phasip | what kind of user and pass? |
08:16.14 | stian | basically just a api key and dummy password |
08:16.25 | stian | s/a/an |
08:16.56 | stian | I'm not very familiar with PHP, so I have been hacking away, but don't know how to debug properly :P |
08:17.09 | stian | Mind having a look at the file at least? |
08:17.43 | phasip | http://developer.capsulecrm.com/v1/authentication/ <- is this what you want to do? |
08:18.46 | stian | Indeed |
08:18.57 | stian | https://gist.github.com/stianeklund/e9d7541161d7c205d6b8 This is what I have made so far |
08:19.16 | stian | now the api key and everything is in there, and that's ok as I created the account for testing purposes only :) |
08:19.17 | phasip | If so, you are probably able to put the password in the url, eg if the site is "https://capsulecrm.com/api" make it "https://username:password@capsulecrm.com/api" |
08:19.31 | stian | you think so? |
08:19.54 | phasip | It's quite standard, try it atleast |
08:20.31 | stian | sounds a hell of a lot easier than having to go around the get_url_contents function |
08:20.34 | stian | :P |
08:21.36 | phasip | Hey, |
08:21.43 | phasip | You are already using CURLOPT_USERPWD |
08:21.53 | stian | Yes |
08:21.58 | stian | but it's not working :P |
08:22.01 | phasip | just remove the [] around the username and password and you should be fine |
08:22.07 | stian | hmm |
08:22.35 | stian | the script that I based it off just calls the get_url_contents function |
08:22.47 | stian | but I'm not that familiar with it, nor do I know if I need it |
08:23.02 | stian | probably not as long as I get a result and eventually return $caller_id :) |
08:23.52 | stian | I'm not sure if I'm parsing the data correctly either yet, tbh |
08:25.08 | phasip | First try to fix authentication, just print whatever result you get to visually see if it contains what you want. |
08:26.36 | stian | Right, so I fixe the authentication bit, but that didn't work |
08:27.00 | stian | I need to learn how to print out / echo the results somewhere :) |
08:27.44 | phasip | Heh, sorry dunno bout the superfecta stuff |
08:28.12 | stian | me neither, but I thought it would be easiest to do it that way, rather than screw around with how caller id works |
08:28.18 | phasip | $this->DebugPrint("Searching Capsule - {$this->thenumber} ... "); <- seems to be your help |
08:28.46 | stian | ah, array not found |
08:29.10 | stian | truth be told I haven't touched php until last night, so. :P |
08:29.49 | stian | I'm going to give this a go, thanks for the tips, man. |
08:29.54 | stian | Appreciate it |
08:31.07 | phasip | 1: try the old code, but with the url authentication. 2: Try to fix what you have played with, I see you try to parse the json before you have any result, should be $ret = curl_exec($url); $result = json_decode($ret,true); |
08:31.35 | stian | right |
08:31.43 | stian | I agree that sounds like a good idea |
08:32.59 | phasip | gl/hf |
08:33.05 | stian | Thansk :) |
08:33.09 | stian | *thanks |
08:33.24 | stian | would be amazing if I get this working |
08:33.33 | stian | caller id lookups towards our crm at work |
08:33.49 | stian | dives in |
08:34.49 | stian | any tips on how I can debug this? other than watching the output through the web interface? |
08:36.05 | stian | nvm |
08:39.10 | stian | haha :D:D |
08:39.21 | stian | phasip: works with user:passwd@url |
08:39.31 | stian | now to figure out how to parse the json |
08:39.32 | stian | cheers! |
08:40.13 | stian | actually, how do I get json output then, since I no longer use the curlopts? |
08:40.58 | stian | it by default returns xml, I guess I can just find out how to parse that instead |
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09:44.25 | kervel | hello, i'm using a positron pbx which is asterisk but the web interface doesn't allow me to put "custom" stuff in the dialplan. I'd like to create a custom extensions.conf, include it after the generated (by web interface) one and have it "override" some extensions . would that be possible ? |
09:44.33 | kervel | or will asterisk complain about duplicate dialplan entries |
09:46.01 | WIMPy | It does not complain, but it will never search for includes if it already has a hit. |
09:47.10 | WIMPy | You wil have to ask whoever made that thing. I haven't heard of it before. |
09:50.41 | kervel | WIMPy ok, so i just need to switch the order and make sure it includes my file before the generated one |
09:51.16 | WIMPy | That would mean you have to replace the existing configuration and include the original one. |
09:51.31 | kervel | wimpy its a canadian company. i bought it because it was reasonably priced and supported ISDN BRI, but i'm disappointed as the web interface only supports a very small subset of the real asterisk features. |
09:51.40 | WIMPy | So that would be somethig that would have to be repeated on every click at least. |
09:52.04 | kervel | wimpy: the main extensions.conf is not generated, only extensions-postel.conf is generated |
09:52.05 | WIMPy | Well, that's normal about GUIs. |
09:52.13 | kervel | so i can mess around in extensions.conf |
09:52.36 | kervel | wimpy: yeah but other GUIs allow you to dial extensions that are not in the "gui generated" dial plan |
09:52.37 | WIMPy | Ok, if that one is static you could just include your config befor theirs. |
09:53.03 | WIMPy | The only usefull GUI will be the one you wrote yourself. |
09:53.38 | kervel | :) ... the positron g-224 is not easy to tweak: its uClinux so very difficult to install other things on it |
09:53.46 | WIMPy | googled and found positrontelecom.com, but they don't seem to offer anything with BRIs. |
09:54.26 | kervel | and it doesn't use asterisk db but some weird custom variable system (a generated file full of SetGlobal, which get re-generated when a variable changes) |
09:54.44 | kervel | wimpy: right, its not on their website, but its on voipandgo.be ... |
09:55.03 | kervel | http://www.voipandgo.be/positron-g-224.html |
09:55.19 | kervel | i guess european market product, not relevant for their main market |
09:55.23 | kervel | but software is the same |
09:55.53 | WIMPy | If you're from Europe you should know not to buy anything telephony related from North America. |
09:56.03 | kervel | too late :) |
09:56.14 | WIMPy | Cute little thing :-) |
09:56.46 | kervel | yeah, and the price is relatively low compared to alternatives ... but there is a reason i guess |
09:57.08 | WIMPy | Can't you just install a normal OS with standard Asterisk on it? |
09:57.30 | WIMPy | Is it? |
09:59.07 | kervel | wimpy: /proc/cpuinfo says ADSP-BF537 600(MHz CCLK) 120(MHz SCLK) (mpu off) ... never heard of it, it will be hard to find a normal linux that runs on it |
09:59.41 | WIMPy | Ouch. |
10:00.00 | WIMPy | Well, I gess that's more in the range of a modem/router thing. |
10:00.24 | kervel | yes. however it seems to be powerful enough to run asterisk |
10:00.56 | WIMPy | I guess, you shouldn't try to run conferences on that thing. |
10:01.51 | WIMPy | So for what's in there, I think the proce is rather outrageous. |
10:02.45 | kervel | wimpy: a isdn BRI card from digium is already 400 or more ... and i wanted something completely solid state (no fans, no spinning disks) |
10:03.45 | coppice | that's expensive compared to other blackfin based 4 port asterisk boxes |
10:03.52 | WIMPy | You can get quad BRI cards for 100 EUR. And there is certainly more powerfull fanless hardware available. |
10:04.23 | WIMPy | Others use BF as well? Can it do conferences at all? |
10:05.04 | coppice | lots of people used to make blackfin based asterisk boxes, even digium. their days have largely passed, though |
10:05.07 | kervel | the web interface supports conferences (i don't know how well it works: i don't need them) |
10:05.59 | coppice | there should be no problems with small conferences, especially if they are only using G.711 |
10:07.27 | WIMPy | I have to admit that there's usually at least one channel being transcoded involved for me, but with a 3 party conference going on, Asterisk uses a lot of CPU on the 2.9 GHz one. |
10:08.43 | WIMPy | The old MeetMe thing is surely a lot less hungry, but a lot less good as well. And using CPU the DAHDI way can end up really ugly :-( |
10:10.39 | WIMPy | Is the whole OS on the card? |
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10:14.11 | jmls | 'tis a fine morning to be conversing with you fine people ;) |
10:14.17 | jmls | is in a good mood for a change |
10:15.05 | jmls | struggling to get to www.asteriskdocs.org - is the problem me or the site ? |
10:15.24 | coppice | you |
10:15.33 | WIMPy | Works for me (tm) |
10:15.42 | jmls | dang |
10:15.54 | kervel | WIMPy: yay i can override extensions now... i'm going to try to configure call queueing now |
10:15.55 | jmls | must be a dns problem this side of the pond |
10:16.31 | WIMPy | Maybe it was blocked for security reasons. |
10:16.34 | jmls | heh. I guess I've had too much coffee already ... |
10:16.35 | jmls | <PROTECTED> |
10:16.43 | jmls | is not going to work art all, is it ? |
10:17.05 | jmls | guess that could be called an "out by one" error |
10:18.49 | jmls | yay. docs is back up |
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11:25.46 | stian | so I've managed to actually get json output now, whoever I was talking to earlier :P |
11:26.09 | stian | but... I can't seem to get any matching results :( |
11:29.06 | stian | I had to mod superfecta_base to run curlopt with Accept:application/json in order for it to give me the right output :P |
11:29.22 | stian | however, I'm very uncertain how to parse the json output :( |
11:30.00 | stian | I feel that I'm soo close to actually having a functioning CapsuleCRM caller ID module |
11:34.19 | stian | here is an example of the json output I'm getting: https://gist.github.com/stianeklund/c448c7fde8e3f03fa257#file-gistfile1-json (sorry about the formatting) |
11:35.45 | stian | can anyone push me in the right direction as to how to parse that my module: https://github.com/stianeklund/Caller-ID-Superfecta/blob/release/2.11/sources/source-CapsuleCRM.module |
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12:01.55 | stian | never mind, fixed it |
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13:51.03 | stian | I have fixed the stuff I mentioned earlier btw |
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14:30.32 | Marquel | morning. suppose, i have an external line which supports showing the callerid. now i do want to run some commands on every incoming call, some depending on time of day and then some on every call. currently i do have a parameterized macro run from different contexts which are included based on time of the day. |
14:31.26 | Marquel | but all of these commands depend on who's the callee. |
14:33.29 | Marquel | f.ex. if a call comes in, i want to run a Set(), process the call using the time of day, return to the basic context and run another Set(). would something like {{ exten => _X.,1,Set(...); exten => 1234,2,Dial(...); exten => _X.,n,Set(...) }} work here? |
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17:30.16 | vlad_starkov | Question: Does anyone know how to use "audiofilter" utility to optimise your own voice prompts? See https://wiki.asterisk.org/wiki/display/AST/About+the+Sounds+Tools |
17:40.30 | Marquel | morning. reasking my question: can i use something like {{ exten => _X.,1,Set(...); exten => 1234,2,Dial(...); exten => 5678,2,VoiceMail(...); exten => _X.,n,Set(...) }} to set and reset variables independently of the called extension while doing specific things depending on the called extension in between both Set() actions? |
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17:53.32 | WIMPy | Yes |
17:54.58 | WIMPy | But you need to note that n increases the priority no matter if it's the same or not. |
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19:29.04 | zopsi | Are there any databases or APIs I can check against for spam phone calls (i.e. telemarketers and scammers)? |
19:29.31 | andy09usa | Unable to add Google ICE candidates as ICE support not available or no candidates available |
19:29.45 | andy09usa | asterisk + google voice (((( |
19:29.52 | andy09usa | <PROTECTED> |
19:29.59 | andy09usa | not work ((((( |
19:30.04 | andy09usa | Asterisk 11.10.2 |
19:30.19 | andy09usa | help me plzz :) |
19:35.43 | TazzNZ | andy09usa: custom compile ? |
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19:36.04 | andy09usa | freepbx |
19:37.37 | WIMPy | That's a frontend and doesn't answer how Asterisk was installed. |
19:37.58 | WIMPy | However to get help with configuring FreePBX, you need to ask in #freepbx |
19:38.17 | TazzNZ | andy09usa: have you read https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ? |
19:38.45 | andy09usa | there all dead #freepbx |
19:39.29 | WIMPy | Be patient. |
19:39.33 | andy09usa | TazzNZ: yes, it does not workw |
19:40.01 | andy09usa | WIMPy: thx %) |
19:41.14 | TazzNZ | andy09usa: well, you need to show us some CLI output |
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19:48.10 | andy09usa | TazzNZ: http://pastebin.com/BZb7tcBy |
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19:58.42 | TazzNZ | andy09usa: "what does module show like motif" show ? |
20:00.08 | TazzNZ | and "module show like xmpp" ? |
20:02.09 | andy09usa | localhost*CLI> module show like motif |
20:02.09 | andy09usa | Module Description Use Count |
20:02.12 | andy09usa | chan_motif.so Motif Jingle Channel Driver 0 |
20:02.14 | andy09usa | 1 modules loaded |
20:02.35 | andy09usa | Module Description Use Count |
20:02.38 | andy09usa | res_xmpp.so Asterisk XMPP Interface 0 |
20:02.40 | andy09usa | 1 modules loaded |
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20:35.22 | ChannelZ | Didn't Google kill external GV support? |
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21:06.11 | dan_j | Hi. Has anyone used sipsak to do a test registration to an asterisk server? |
21:06.20 | dan_j | I'm struggling to get the right command line. |
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21:28.05 | syadnom | hi all, I'm looking for some help sizing a unique pbx setup. Only about 6 handsets, but a queue of up to 25 callers (its a ticketing call desk) |
21:30.35 | zopsi | Has anyone used pfsense with asterisk? I'm having trouble receiving audio on outbound calls. |
21:32.41 | ChannelZ | syadnom: what is the actual question? None of that is very resource-hungry. If you're doing voip for your calls, it's more of a bandwidth issue than anything |
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21:34.45 | syadnom | ChannelZ, I've been using the little intel NUC w/ dual core celeron and 2GB RAM for small PBXs. I predict I can run 50 sip calls on that device, but much of the system is just passing data. I'm don't typically do the queue and I don't know what it would take CPU wise to handle 25 queued callers, playing them MOH and annoucing position. |
21:34.51 | syadnom | Additionally, I want to use G.729 |
21:35.33 | syadnom | Not sure if that little NUC box is appropriate for those queues. |
21:36.51 | ChannelZ | Well someone in a queue is not much different than someone in a call. The G.729 for that machine I don't know. 'core show translation' should give you an idea on a machine that has g729 on it |
21:38.38 | ChannelZ | If you keep everything g729 it wont matter much (IE if your handsets can do g729, if you encode your MOH as g729..) |
21:39.41 | ChannelZ | Although I wonder how that's affected by Asterisk 12, where everything is a bridge. I assume for a single point-to-point bridge it can still do pass-thru... hmm |
21:39.49 | syadnom | ChannelZ, I didn't think of encoding the MOH to g.729, that seems obvious now. I'll have a couple IVR on there too. I typically stick to G.711 but with up to 31 potential calls, I'll need to use g.729 because of the DSL speeds available |
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21:41.45 | syadnom | I can get Centurylink's 12x1Mb service there which according to digium should handle 128 calls over IAX. I'm using a single SIP trunk but that's not much more overhead and my max is 31. |
21:43.19 | syadnom | ChannelZ, sounds like you've looked at asterisk 12. Have you looked at multiple registrations yet? |
21:44.14 | ChannelZ | Im running it at home. And you mean multiple incoming to the same peer? Yeah |
21:45.18 | syadnom | ChannelZ, no, multiple registrations from same device. |
21:46.04 | syadnom | say 3 phones registering/subscribing to the same trunk via pjsip, so they all ring on same line and share call appearances. SLA/BLA type thing but in the sip channel driver |
21:47.51 | ChannelZ | Well yes I've registered several devices to the same peer. But it's not any sort of automatic SLA that I'm aware of. |
21:49.07 | ChannelZ | I haven't played with doing SLA in 12. In theory it should be much easier because of the bridging aspect of calls, whereas in older versions it was total hackery with conferences |
21:54.33 | syadnom | ChannelZ, yeah, I played with it in a freepbx beta w/ asterisk 12. clearly it was still beta (to distro) so SIP trunks werent working well w/ pjsip, but the multiple registrations worked nicely and I could see 'line' status on all registered devices (polycom vvx) and pick up held calls. This is my primary goal, emulating key systems for 4-5 phone offices which are my primary customers |
21:55.49 | ChannelZ | hmm |
21:59.18 | syadnom | its a hassle trying to both convince and then teach people to park calls. I typically use some soft buttons to BLF each parking lot so they can visually see a blinky 'hold' light, but it's less than ideal. I'm glad to see * devs are finally willing to make asterisk perfect for the small office. |
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22:41.35 | zopsi | Is there any way to get external_signaling_address and external_media_address to use a hostname instead of an IP? My IP changes almost weekly. |
22:42.26 | zopsi | Those vars are for a trunk in pjsip.conf |
22:42.44 | zopsi | trunk nat route |
22:43.15 | TazzNZ | anybody know how I can override my "Contact:" header ? |
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23:12.14 | TazzNZ | mjordan: do you have any idea how I can override/append the "Contact:" header in Asterisk - I suspect it's "protected" or something |
23:12.54 | file | you can't |
23:13.30 | TazzNZ | cool - thanks file - just wanted to confirm that - Is there an "official" reason behind it ? |
23:13.51 | TazzNZ | something like "because you can break X with it" |
23:14.20 | TazzNZ | (my ITSP is asking us to do this for "A-Party pass thru") |
23:14.26 | file | it's a core part of SIP and generally both chan_sip and chan_pjsip forbid touching such things, although some settings can influence the creation of the header |
23:15.30 | TazzNZ | cool - I need to set it in the dialplan so I will go back and ask for another way of doing this |
23:15.32 | TazzNZ | thx |
23:18.45 | file | you're everywhere... you're everywhere! tell me how I got this far just tell me why and who you are... *sings* |
23:19.05 | TazzNZ | hehe......what song is that file ? |
23:19.19 | file | apparently Everywhere by Michelle Branch |
23:19.33 | file | I'm on the '80s, '90s, and Today station |
23:19.35 | TazzNZ | never heard it before.....googles |
23:20.39 | TazzNZ | not bad |
23:21.10 | TazzNZ | aaahhhhh - chorus - now I know :) |
23:21.24 | file | it was popular back in the day, apparently enough for me to be able to sing to it |
23:21.55 | TazzNZ | yeah - the chorus was what I needed |
23:22.43 | TazzNZ | wow - 2001 - I thought it was way older than that |
23:23.26 | file | that's still 13 years |
23:23.48 | TazzNZ | damn - true |
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23:25.52 | file | ooh baby do you know what that's worth? ooh heaven is a place on earth! |
23:28.06 | TazzNZ | now that one I know :) lol |
23:34.41 | file | TazzNZ, what is your ITSP asking you to change in the Contact? |
23:35.18 | TazzNZ | file: 1 sec - otp |
23:37.17 | TazzNZ | file: they want something like Contact: <sip:044718608;tgrp=44718607;trunk-context=itsp.domain.com@192.168.1.12:5060;transport=UDP> |
23:39.09 | file | interestin' |
23:39.12 | TazzNZ | those varibles seems to be part of RFC4904 |
23:40.09 | TazzNZ | yeah - look at this from the RFC |
23:40.11 | TazzNZ | Contact: <sip:0100;phone-context=example.com;tgrp=TG1-1; |
23:40.13 | TazzNZ | <PROTECTED> |
23:40.33 | TazzNZ | that might be a feature request for Asterisk |
23:43.31 | file | yeah, no real way to manipulate Contact header generation to include that |
23:46.39 | TazzNZ | file: unless it's done to comply with the RFC |
23:46.44 | TazzNZ | so I am thinking... |
23:47.02 | TazzNZ | something like, either create tgrp vars in sip.conf |
23:47.18 | TazzNZ | or specific vars that create a RFC type header |
23:47.53 | TazzNZ | so if TGRP_something is set and TRUNK_CONTEXT is set, the create the contact header in that way |
23:48.35 | TazzNZ | in a nutshell, have varibles that allow you to create that header. This will still keep the header "protected" |
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23:51.37 | TazzNZ | sweet - got the another method to work :D |
23:51.49 | TazzNZ | I'll log a feature request on Asterisk shortly |
23:52.02 | file | we don't accept feature requests without patches on the issue tracker |
23:52.16 | TazzNZ | eeekkkk |
23:52.38 | TazzNZ | I'll have to see if work will Auth my time for that :| |
23:52.43 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines?src=search#AsteriskIssueGuidelines-Howtorequestafeature |