IRC log for #asterisk on 20140622

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01:29.30SpaceInvadersAnyone around?
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01:34.00ChannelZish
01:36.00WIMPyBetter than some *ism.
01:37.29SpaceInvadersI was having trouble with the instructions in the 3rd ed of the definitive guide.  They were wrong but I figured it out.
01:38.12WIMPyThere's a 4th one
01:38.29SpaceInvadersIt 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
01:38.37SpaceInvadersYeah I couldn't find the 4th one
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01:39.00WIMPyNether of those are any Asterisk internals.
01:39.07WIMPyIt's all what YOU call it.
01:39.21SpaceInvadersTrue but I've been following the instructions
01:39.36SpaceInvadersso far they have used a consistent context
01:40.03SpaceInvadersThen under voicemail they changed it without warning.
01:40.36WIMPyVoicemail contexts are independent of others.
01:41.54SpaceInvadersIn this case it's the dialplan context for allowing users to dial an extension (8500) to access voicemail
01:42.01SpaceInvadersThe instructions say to create the following entry
01:42.09SpaceInvaders[Services]
01:42.09SpaceInvadersexten => *98,1,VoiceMailMain()
01:42.25SpaceInvaderswrong context, wrong extension :)
01:42.48SpaceInvadersbut if you match with the context used earlier and change the *98 to 8500 it works fine
01:43.15WIMPyWhat's "wrong" with the extension?
01:43.18SpaceInvadersThe good news is I'm starting to get a basic understanding of how Asterisk works
01:43.32SpaceInvadersthe instructions say dial 8500 to access voicemail
01:43.35SpaceInvadersthe entry is
01:43.39SpaceInvadersexten => *98,1,VoiceMailMain()
01:43.51SpaceInvadersperhaps your system works differently than mine
01:43.59SpaceInvadersbut 8500 won't get me voicemail with that entry
01:44.16WIMPyYes, I use "voicemail" or "1155".
01:44.21WIMPyIt's all up to you.
01:44.43WIMPyBut I think *98 is rather common.
01:44.55SpaceInvadersYes.  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
01:45.19WIMPyIt's good if you can help yourself.
01:45.19SpaceInvadersAsterisk so ROCKS
01:45.26SpaceInvadersagreed!
01:45.43SpaceInvadersHey, would you happen to be familiar with Linphone?
01:46.34SpaceInvadersI keep getting this message on CLI
01:46.35SpaceInvaders[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
01:46.36WIMPyno
01:46.42SpaceInvadersbut that phone works and the pw is correct
01:47.06WIMPyPossibly a wrong type=?
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01:47.17SpaceInvadershmmm let me check that....
01:48.03SpaceInvadersnope type=friend
01:48.13SpaceInvadersthat's what I used for all 3
01:48.29SpaceInvadersI thought maybe it was the pw because I initially used 1234 but I changed it
01:48.34WIMPyNumeric usernames are somewhat insecure, BTW.
01:49.00SpaceInvadersYes I need to learn more about security
01:49.04SpaceInvadersthis is just a sandbox system
01:49.11SpaceInvadersand it's firewalled
01:49.16WIMPyWell, if it says wrong password, I guess it must be matching the right peer.
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01:52.41SpaceInvadersis there a command that gives you more detail than sip show peers
01:52.49SpaceInvaderslike sip show 1000
01:52.57SpaceInvadersI thought I read something like that but now can't find it
01:53.26WIMPyuse your tab key :-)
01:53.32SpaceInvadersit looks good when I check it via sip show peers and users
01:53.38WIMPysip show peer ...
01:53.39SpaceInvadersI haven't tried the tab
01:53.53SpaceInvadersSWEET!!!  Thanks!!!!!!
01:54.06SpaceInvadersthat's it sip show user 1000
01:54.50SpaceInvadersok nothing pops up that looks like I should be seeing that error :/
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05:04.21rhineheart_mI am planning to buy licenses (G729).. but I want to hear feedbacks from the community about it..
05:06.36coppicewhat 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.
06:15.24coppice[TK]D-Fender: why did you repeat that?
06:19.25[TK]D-FenderBecause 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
06:22.55coppiceyou're very thorough :-)
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06:25.46[TK]D-FenderThat's just how I roll :)
06:31.07[TK]D-FenderAnd.... time for bed.  Later all
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07:58.56phasipIs 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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08:08.15stiananyone 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.
08:09.33stianbasically 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
08:15.30phasipwhat kind of user and pass?
08:16.14stianbasically just a api key and dummy password
08:16.25stians/a/an
08:16.56stianI'm not very familiar with PHP, so I have been hacking away, but don't know how to debug properly :P
08:17.09stianMind having a look at the file at least?
08:17.43phasiphttp://developer.capsulecrm.com/v1/authentication/ <- is this what you want to do?
08:18.46stianIndeed
08:18.57stianhttps://gist.github.com/stianeklund/e9d7541161d7c205d6b8 This is what I have made so far
08:19.16stiannow the api key and everything is in there, and that's ok as I created the account for testing purposes only :)
08:19.17phasipIf 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"
08:19.31stianyou think so?
08:19.54phasipIt's quite standard, try it atleast
08:20.31stiansounds a hell of a lot easier than having to go around the get_url_contents function
08:20.34stian:P
08:21.36phasipHey,
08:21.43phasipYou are already using CURLOPT_USERPWD
08:21.53stianYes
08:21.58stianbut it's not working :P
08:22.01phasipjust remove the [] around the username and password and you should be fine
08:22.07stianhmm
08:22.35stianthe script that I based it off just calls the get_url_contents function
08:22.47stianbut I'm not that familiar with it, nor do I know if I need it
08:23.02stianprobably not as long as I get a result and eventually return $caller_id :)
08:23.52stianI'm not sure if I'm parsing the data correctly either yet, tbh
08:25.08phasipFirst try to fix authentication, just print whatever result you get to visually see if it contains what you want.
08:26.36stianRight, so I fixe the authentication bit, but that didn't work
08:27.00stianI need to learn how to print out / echo the results somewhere :)
08:27.44phasipHeh, sorry dunno bout the superfecta stuff
08:28.12stianme neither, but I thought it would be easiest to do it that way, rather than screw around with how caller id works
08:28.18phasip$this->DebugPrint("Searching Capsule - {$this->thenumber} ... "); <- seems to be your help
08:28.46stianah, array not found
08:29.10stiantruth be told I haven't touched php until last night, so. :P
08:29.49stianI'm going to give this a go, thanks for the tips, man.
08:29.54stianAppreciate it
08:31.07phasip1: 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);
08:31.35stianright
08:31.43stianI agree that sounds like a good idea
08:32.59phasipgl/hf
08:33.05stianThansk :)
08:33.09stian*thanks
08:33.24stianwould be amazing if I get this working
08:33.33stiancaller id lookups towards our crm at work
08:33.49stiandives in
08:34.49stianany tips on how I can debug this? other than watching the output through the web interface?
08:36.05stiannvm
08:39.10stianhaha :D:D
08:39.21stianphasip: works with user:passwd@url
08:39.31stiannow to figure out how to parse the json
08:39.32stiancheers!
08:40.13stianactually, how do I get json output then, since I no longer use the curlopts?
08:40.58stianit by default returns xml, I guess I can just find out how to parse that instead
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09:44.25kervelhello, 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 ?
09:44.33kervelor will asterisk complain about duplicate dialplan entries
09:46.01WIMPyIt does not complain, but it will never search for includes if it already has a hit.
09:47.10WIMPyYou wil have to ask whoever made that thing. I haven't heard of it before.
09:50.41kervelWIMPy ok, so i just need to switch the order and make sure it includes my file before the generated one
09:51.16WIMPyThat would mean you have to replace the existing configuration and include the original one.
09:51.31kervelwimpy 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.
09:51.40WIMPySo that would be somethig that would have to be repeated on every click at least.
09:52.04kervelwimpy: the main extensions.conf is not generated, only extensions-postel.conf is generated
09:52.05WIMPyWell, that's normal about GUIs.
09:52.13kervelso i can mess around in extensions.conf
09:52.36kervelwimpy: yeah but other GUIs allow you to dial extensions that are not in the "gui generated" dial plan
09:52.37WIMPyOk, if that one is static you could just include your config befor theirs.
09:53.03WIMPyThe only usefull GUI will be the one you wrote yourself.
09:53.38kervel:) ... the positron g-224 is not easy to tweak: its uClinux so very difficult to install other things on it
09:53.46WIMPygoogled and found positrontelecom.com, but they don't seem to offer anything with BRIs.
09:54.26kerveland 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)
09:54.44kervelwimpy: right, its not on their website, but its on voipandgo.be ...
09:55.03kervelhttp://www.voipandgo.be/positron-g-224.html
09:55.19kerveli guess european market product, not relevant for their main market
09:55.23kervelbut software is the same
09:55.53WIMPyIf you're from Europe you should know not to buy anything telephony related from North America.
09:56.03kerveltoo late :)
09:56.14WIMPyCute little thing :-)
09:56.46kervelyeah, and the price is relatively low compared to alternatives ... but there is a reason i guess
09:57.08WIMPyCan't you just install a normal OS with standard Asterisk on it?
09:57.30WIMPyIs it?
09:59.07kervelwimpy: /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
09:59.41WIMPyOuch.
10:00.00WIMPyWell, I gess that's more in the range of a modem/router thing.
10:00.24kervelyes. however it seems to be powerful enough to run asterisk
10:00.56WIMPyI guess, you shouldn't try to run conferences on that thing.
10:01.51WIMPySo for what's in there, I think the proce is rather outrageous.
10:02.45kervelwimpy: a isdn BRI card from digium is already 400 or more ... and i wanted something completely solid state (no fans, no spinning disks)
10:03.45coppicethat's expensive compared to other blackfin based 4 port asterisk boxes
10:03.52WIMPyYou can get quad BRI cards for 100 EUR. And there is certainly more powerfull fanless hardware available.
10:04.23WIMPyOthers use BF as well? Can it do conferences at all?
10:05.04coppicelots of people used to make blackfin based asterisk boxes, even digium. their days have largely passed, though
10:05.07kervelthe web interface supports conferences (i don't know how well it works: i don't need them)
10:05.59coppicethere should be no problems with small conferences, especially if they are only using G.711
10:07.27WIMPyI 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.
10:08.43WIMPyThe 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 :-(
10:10.39WIMPyIs the whole OS on the card?
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10:14.11jmls'tis a fine morning to be conversing with you fine people ;)
10:14.17jmlsis in a good mood for a change
10:15.05jmlsstruggling to get to www.asteriskdocs.org - is the problem me or the site ?
10:15.24coppiceyou
10:15.33WIMPyWorks for me (tm)
10:15.42jmlsdang
10:15.54kervelWIMPy: yay i can override extensions now... i'm  going to try to configure call queueing now
10:15.55jmlsmust be a dns problem this side of the pond
10:16.31WIMPyMaybe it was blocked for security reasons.
10:16.34jmlsheh. I guess I've had too much coffee already ...
10:16.35jmls<PROTECTED>
10:16.43jmlsis not going to work art all, is it ?
10:17.05jmlsguess that could be called an "out by one" error
10:18.49jmlsyay. docs is back up
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11:25.46stianso I've managed to actually get json output now, whoever I was talking to earlier :P
11:26.09stianbut... I can't seem to get any matching results :(
11:29.06stianI had to mod superfecta_base to run curlopt with Accept:application/json in order for it to give me the right output :P
11:29.22stianhowever, I'm very uncertain how to parse the json output :(
11:30.00stianI feel that I'm soo close to actually having a functioning CapsuleCRM caller ID module
11:34.19stianhere is an example of the json output I'm getting: https://gist.github.com/stianeklund/c448c7fde8e3f03fa257#file-gistfile1-json (sorry about the formatting)
11:35.45stiancan 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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12:01.55stiannever mind, fixed it
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13:51.03stianI have fixed the stuff I mentioned earlier btw
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14:30.32Marquelmorning. 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.
14:31.26Marquelbut all of these commands depend on who's the callee.
14:33.29Marquelf.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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17:30.16vlad_starkovQuestion: 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
17:40.30Marquelmorning. 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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17:53.32WIMPyYes
17:54.58WIMPyBut you need to note that n increases the priority no matter if it's the same or not.
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19:29.04zopsiAre there any databases or APIs I can check against for spam phone calls (i.e. telemarketers and scammers)?
19:29.31andy09usaUnable to add Google ICE candidates as ICE support not available or no candidates available
19:29.45andy09usaasterisk  + google voice ((((
19:29.52andy09usa<PROTECTED>
19:29.59andy09usanot work (((((
19:30.04andy09usaAsterisk 11.10.2
19:30.19andy09usahelp me plzz :)
19:35.43TazzNZandy09usa: custom compile ?
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19:36.04andy09usafreepbx
19:37.37WIMPyThat's a frontend and doesn't answer how Asterisk was installed.
19:37.58WIMPyHowever to get help with configuring FreePBX, you need to ask in #freepbx
19:38.17TazzNZandy09usa: have you read https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google ?
19:38.45andy09usathere all dead #freepbx
19:39.29WIMPyBe patient.
19:39.33andy09usaTazzNZ: yes, it does not workw
19:40.01andy09usaWIMPy: thx %)
19:41.14TazzNZandy09usa: well, you need to show us some CLI output
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19:48.10andy09usaTazzNZ: http://pastebin.com/BZb7tcBy
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19:58.42TazzNZandy09usa: "what does module show like motif" show ?
20:00.08TazzNZand "module show like xmpp" ?
20:02.09andy09usalocalhost*CLI> module show like motif
20:02.09andy09usaModule                         Description                              Use Count
20:02.12andy09usachan_motif.so                  Motif Jingle Channel Driver              0
20:02.14andy09usa1 modules loaded
20:02.35andy09usaModule                         Description                              Use Count
20:02.38andy09usares_xmpp.so                    Asterisk XMPP Interface                  0
20:02.40andy09usa1 modules loaded
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20:35.22ChannelZDidn't Google kill external GV support?
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21:06.11dan_jHi. Has anyone used sipsak to do a test registration to an asterisk server?
21:06.20dan_jI'm struggling to get the right command line.
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21:28.05syadnomhi 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)
21:30.35zopsiHas anyone used pfsense with asterisk? I'm having trouble receiving audio on outbound calls.
21:32.41ChannelZsyadnom: 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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21:34.45syadnomChannelZ, 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.
21:34.51syadnomAdditionally, I want to use G.729
21:35.33syadnomNot sure if that little NUC box is appropriate for those queues.
21:36.51ChannelZWell 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
21:38.38ChannelZIf you keep everything g729 it wont matter much (IE if your handsets can do g729, if you encode your MOH as g729..)
21:39.41ChannelZAlthough 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
21:39.49syadnomChannelZ, 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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21:41.45syadnomI 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.
21:43.19syadnomChannelZ, sounds like you've looked at asterisk 12.  Have you looked at multiple registrations yet?
21:44.14ChannelZIm running it at home.  And you mean multiple incoming to the same peer?  Yeah
21:45.18syadnomChannelZ, no, multiple registrations from same device.
21:46.04syadnomsay 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
21:47.51ChannelZWell yes I've registered several devices to the same peer.  But it's not any sort of automatic SLA that I'm aware of.
21:49.07ChannelZI 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
21:54.33syadnomChannelZ, 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
21:55.49ChannelZhmm
21:59.18syadnomits 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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22:41.35zopsiIs 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.
22:42.26zopsiThose vars are for a trunk in pjsip.conf
22:42.44zopsitrunk nat route
22:43.15TazzNZanybody know how I can override my "Contact:" header ?
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23:12.14TazzNZmjordan: do you have any idea how I can override/append the "Contact:" header in Asterisk - I suspect it's "protected" or something
23:12.54fileyou can't
23:13.30TazzNZcool - thanks file - just wanted to confirm that - Is there an "official" reason behind it ?
23:13.51TazzNZsomething like "because you can break X with it"
23:14.20TazzNZ(my ITSP is asking us to do this for "A-Party pass thru")
23:14.26fileit'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
23:15.30TazzNZcool - I need to set it in the dialplan so I will go back and ask for another way of doing this
23:15.32TazzNZthx
23:18.45fileyou're everywhere... you're everywhere! tell me how I got this far just tell me why and who you are... *sings*
23:19.05TazzNZhehe......what song is that file ?
23:19.19fileapparently Everywhere by Michelle Branch
23:19.33fileI'm on the '80s, '90s, and Today station
23:19.35TazzNZnever heard it before.....googles
23:20.39TazzNZnot bad
23:21.10TazzNZaaahhhhh - chorus - now I know :)
23:21.24fileit was popular back in the day, apparently enough for me to be able to sing to it
23:21.55TazzNZyeah - the chorus was what I needed
23:22.43TazzNZwow - 2001 - I thought it was way older than that
23:23.26filethat's still 13 years
23:23.48TazzNZdamn - true
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23:25.52fileooh baby do you know what that's worth? ooh heaven is a place on earth!
23:28.06TazzNZnow that one I know :) lol
23:34.41fileTazzNZ, what is your ITSP asking you to change in the Contact?
23:35.18TazzNZfile: 1 sec - otp
23:37.17TazzNZfile: they want something like Contact: <sip:044718608;tgrp=44718607;trunk-context=itsp.domain.com@192.168.1.12:5060;transport=UDP>
23:39.09fileinterestin'
23:39.12TazzNZthose varibles seems to be part of RFC4904
23:40.09TazzNZyeah - look at this from the RFC
23:40.11TazzNZContact: <sip:0100;phone-context=example.com;tgrp=TG1-1;
23:40.13TazzNZ<PROTECTED>
23:40.33TazzNZthat might be a feature request for Asterisk
23:43.31fileyeah, no real way to manipulate Contact header generation to include that
23:46.39TazzNZfile: unless it's done to comply with the RFC
23:46.44TazzNZso I am thinking...
23:47.02TazzNZsomething like, either create tgrp vars in sip.conf
23:47.18TazzNZor specific vars that create a RFC type header
23:47.53TazzNZso if TGRP_something is set and TRUNK_CONTEXT is set, the create the contact header in that way
23:48.35TazzNZin a nutshell, have varibles that allow you to create that header. This will still keep the header "protected"
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23:51.37TazzNZsweet - got the another method to work :D
23:51.49TazzNZI'll log a feature request on Asterisk shortly
23:52.02filewe don't accept feature requests without patches on the issue tracker
23:52.16TazzNZeeekkkk
23:52.38TazzNZI'll have to see if work will Auth my time for that :|
23:52.43filehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines?src=search#AsteriskIssueGuidelines-Howtorequestafeature

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