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| 07:52.22 | z30 | Hi Guys i use asterisk to passtrought fax but i have this error : dahdi_handle_dtmf: Fax detected, but no fax extension |
| 07:52.47 | z30 | i have a call extension in my extensions.conf and those are prefectly working |
| 07:53.07 | z30 | how can i handle fax extensions? |
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| 08:23.02 | Zogot | z30: you have an extension listed as extenline => fax,blablabla |
| 08:29.42 | z30 | like : exten => fax,1,dial(.........) ? |
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| 08:31.46 | Zogot | z30: exten => fax,1,NoOp(Fax Received) |
| 08:31.47 | Zogot | for example |
| 08:32.00 | Zogot | prefix my name with your messages, else i wont always see them :) |
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| 08:59.51 | z30 | Zogot: i define another extension and name it [fax] and add this : exten => fax,1,NoOp(Fax Received) to it what i do next? i have error yet and when i add fax into my perv extension my fax is droped |
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| 12:36.05 | Reapster | heya, is anyone per chance familiar with ari-py? |
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| 12:46.26 | WIMPy | Don't ask meta questions. Ask you question, maybe someone is able to help. |
| 12:52.56 | Reapster | apologies, I'm wanting to do a dynamic config update (add a SIP endpoint) with ari-py, but haven't found much in the way of documentation on that particular aspect. I can just do with requests module though, but wondering if ari-py supports it |
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| 13:28.01 | mjordan | Reapster: it's a relatively new feature. The good news is, ari-py *should* dynamically figure out the path for you. |
| 13:28.06 | mjordan | it should be: |
| 13:29.46 | mjordan | asterisk.updateObject(configClass='res_pjsip', objectType='endpoint', id='name') |
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| 13:30.04 | mjordan | looks at passing body params |
| 13:31.02 | mjordan | hm. |
| 13:31.04 | mjordan | Should be then: |
| 13:31.54 | mjordan | asterisk.updateObject(configClass='res_pjsip', objectType='endpoint', id='name', fields=[{attribute: 'allow', value: '!all,ulaw,alaw'},...]) |
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| 13:42.11 | Reapster | oh wow, thanks mjordan |
| 13:42.23 | Reapster | thats exactly what I was after, thanks so much |
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| 13:50.35 | mjordan | Reapster: np. Given that it is a relatively new feature, if you run into an issue, please chime in on here. I've fixed a few corner cases that should come out in the next release related to this as well, but they're pretty "weird" edge cases. |
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| 13:52.18 | Reapster | awesome, will do |
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| 13:55.38 | vassilux | I use Dial into my sub and want to keep the initial context into cdr record after hangup, is it possible ? |
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| 14:17.12 | qakhan | is there any logs of dahdi channels. like if PRI disconnect it log in file. |
| 14:17.44 | igcewieling | qakhan: /var/log/messages and (often) /var/log/asterisk/* |
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| 14:19.19 | napnap | Hi all, is it possible to test a time condition before call a ring group ? |
| 14:19.20 | WIMPy | What kind of information are you looking for? |
| 14:19.58 | napnap | (internally) |
| 14:20.06 | WIMPy | 'core show applications like time' |
| 14:20.37 | WIMPy | 'core show functions like TIME' |
| 14:20.47 | WIMPy | Pich one :-) |
| 14:20.51 | WIMPy | Pick |
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| 14:31.54 | qakhan | igcewieling message log file does has pri failure loh |
| 14:32.00 | qakhan | log* |
| 14:41.05 | napnap | WIMPy, thx for your answer, I will check that |
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| 15:32.46 | phormulate | anyone know anything about chan_motif and its lack of srtp? |
| 15:34.10 | phormulate | apparently it was rev eng, so says a mailing list, and yet google's implementation supports srtp based on the dev whitepapers |
| 15:34.31 | phormulate | I can't find a damned thing on it |
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| 17:16.03 | justdave | is there a way to specify a @ as part of a username in a sip register=> line? |
| 17:16.37 | justdave | I have a provider that wants an @ with a domain name as part of the username, and the IP that name resolves to is not the same IP they told us to register to |
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| 17:17.20 | justdave | asterisk seems to ignore everything after the @ because there's another one later with the IP |
| 17:17.57 | justdave | since it seems sorta like a URL I tried %40 but it puts that in literally |
| 17:18.28 | [TK]D-Fender | @ should work. Show us your actual line masking only the secret |
| 17:19.41 | justdave | register => 1234567890@domain.name:{secret}@ipaddress |
| 17:20.21 | [TK]D-Fender | And the register packet going out.... |
| 17:20.50 | justdave | [Sep 21 10:20:12] WARNING[2166]: chan_sip.c:21478 handle_response_register: Got 404 Not found on SIP register to service 1234567890@ipaddress, giving up |
| 17:20.54 | justdave | that's the error I get |
| 17:21.00 | justdave | looking for packet trace, one moment |
| 17:21.04 | [TK]D-Fender | that is not the full packet |
| 17:21.20 | justdave | right I was already on my way to paste the above before you asked, going back for that now |
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| 17:25.20 | justdave | https://it.pastebin.mozilla.org/8847020 |
| 17:25.37 | justdave | I see it in there, maybe the error message is just borked |
| 17:26.14 | justdave | it's a 1.8 asterisk (will upgrade one of these days) |
| 17:27.01 | [TK]D-Fender | And you aren't showing where the packet was sent../.. |
| 17:28.19 | justdave | oh, right... it did go to the IP after the final @ and not the one the domain in the middle resolves to |
| 17:28.52 | [TK]D-Fender | Which is ... exactly what you're telling it to do |
| 17:29.00 | [TK]D-Fender | And you are getting a response back |
| 17:29.14 | justdave | yep. so I need to get with them to find out why they're 404ing it then |
| 17:29.18 | [TK]D-Fender | Go validate that that account is actually valid and not locked, etc |
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| 17:30.27 | justdave | and yeah, the error message is just truncating the username at the @ |
| 17:30.31 | justdave | that's the part that confused me |
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| 18:01.54 | mehale | hi folks. |
| 18:02.31 | mehale | I need to add a calling rule to dial to a number into a pap3000 fxo port, which is configured as a regular user. |
| 18:02.39 | mehale | what should I google for? |
| 18:02.41 | mehale | :) |
| 18:03.34 | [TK]D-Fender | Nothing to google for. |
| 18:03.41 | [TK]D-Fender | Dial(SIP/yourpeer/thenumber) |
| 18:03.58 | [TK]D-Fender | You should ahve a peer setup for the FXO port |
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| 18:29.47 | mehale | [TK]D-Fender, thnaks, trying it now |
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| 18:32.30 | mehale | [TK]D-Fender, on asterisk-gui, should I enter custom destination "Dial(....)"? |
| 18:32.51 | [TK]D-Fender | asterisk-gui has been dead for half a DECADE, is not supported and should never ever be used |
| 18:33.26 | mehale | ops |
| 18:34.02 | mehale | this is what came with my nas... what is the good option for it? |
| 18:34.09 | mehale | good alternative to it? |
| 18:35.58 | [TK]D-Fender | Configure your own system or pick a GUI that isn't completely dead |
| 18:36.02 | [TK]D-Fender | FreePBX is the common choice |
| 18:36.32 | [TK]D-Fender | And then use their channel for support of it and this channel does not currently support any GUI |
| 18:36.52 | mehale | ok |
| 18:37.17 | avb | guys, im setting a variable in the dialplan and executing it via Local/ call. Problem is that Call is 'losing' this variable, while im specifying /n for a number :-/ |
| 18:37.25 | avb | any hints? |
| 18:37.58 | [TK]D-Fender | that doesn't make sense aws you've written it. |
| 18:38.07 | [TK]D-Fender | Rephrase and actuaclly show us what you're doing |
| 18:38.21 | avb | http://pastebin.com/QDhsFbrh |
| 18:38.53 | avb | http://pastebin.com/tyvX7S7R |
| 18:38.58 | avb | thats the test dialplan |
| 18:39.50 | avb | and doesnt matter if i have /n, variable gets set on Local/123@default-0000001d;2 and in bridge context im getting Local/123@default-0000001d;1 |
| 18:40.16 | avb | sounds pretty straightforward thing to do |
| 18:41.34 | [TK]D-Fender | <PROTECTED> |
| 18:41.49 | avb | http://pastebin.com/qfbXQd2h |
| 18:41.54 | avb | thats a call file |
| 18:42.24 | avb | [TK]D-Fender: i were trying without it as well |
| 18:43.00 | [TK]D-Fender | try without any inheritance as well |
| 18:43.06 | [TK]D-Fender | no __ |
| 18:44.01 | [TK]D-Fender | And actually.... there is no actual "dial" involved here.. we're all kinda Local.... |
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| 18:45.18 | [TK]D-Fender | Variables: test2=qqq |
| 18:45.23 | [TK]D-Fender | And where would this be coming from? |
| 18:45.47 | avb | [TK]D-Fender: thats from a call file |
| 18:46.19 | [TK]D-Fender | Ok. Try without the __ and lets see |
| 18:46.24 | avb | removed __ |
| 18:46.28 | avb | same story |
| 18:46.31 | [TK]D-Fender | hrm |
| 18:47.25 | [TK]D-Fender | I think because there is no Dial involved there is no inheritance, and because the call actually resumes on the "Exten" channel side that there is no variable to carry over. |
| 18:47.35 | [TK]D-Fender | Not sure if there is a way to do this on that side of the equation. |
| 18:48.32 | [TK]D-Fender | (reasonable) |
| 18:48.46 | avb | huh, seems need to do a workaround :( |
| 18:49.04 | avb | will play a bit more |
| 18:49.09 | avb | maybe will figure out something |
| 18:50.11 | [TK]D-Fender | You could push them into some more static place so that each side could pick them up in a unique way. |
| 18:50.22 | [TK]D-Fender | like an auto-incremented astDB value or some such. |
| 18:50.34 | [TK]D-Fender | somewhat dirty or course. |
| 18:50.37 | [TK]D-Fender | of* |
| 18:54.25 | avb | :) yeh, thats something i had before |
| 18:54.45 | avb | luckily i got rid of this for most of the functions |
| 18:55.01 | avb | ill workaround by SetVar |
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| 19:04.21 | qakhan | is there any way, we can get exts info(ext number and person name) from sip.conf and publish them on ip phones as contacts |
| 19:05.18 | qakhan | phone directory |
| 19:08.40 | [TK]D-Fender | Depends on the phone. |
| 19:09.07 | [TK]D-Fender | And you can parse that file any time you want. It's your system. Go write a script for it |
| 19:09.10 | qakhan | i have Polycom and grandstream |
| 19:09.16 | [TK]D-Fender | And to getting it on a phone... read your phone's manua;l |
| 19:09.24 | [TK]D-Fender | VERY FINE MANUALS |
| 19:09.34 | [TK]D-Fender | Read them |
| 19:10.26 | qakhan | i have corporate directory on my network and phones are getting info from it. |
| 19:10.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:11.05 | qakhan | but there are some phone which are connected my * over the internet, they cannot access corporate directory |
| 19:11.12 | [TK]D-Fender | So go write a script to parse your configs to fill it in |
| 19:11.42 | qakhan | any example? |
| 19:11.51 | qakhan | script example |
| 19:11.55 | [TK]D-Fender | Write it yourself |
| 19:12.48 | [TK]D-Fender | If you can't tell how to write one then you aren't a programmer and should hire a consultant to write it for you. |
| 19:13.03 | [TK]D-Fender | sip.conf is as dumb and flat a config as any. |
| 19:13.26 | [TK]D-Fender | [section] |
| 19:13.30 | [TK]D-Fender | paramter=value |
| 19:14.26 | qakhan | i am thinking about contact directory and publish it as webservice |
| 19:14.54 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-pznlazbhozuehvif) |
| 19:14.55 | qakhan | so phones over the internet can get it |
| 19:16.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:28.29 | igcewieling | ugh. I've trapped and relocated 19 squirrles from my yard. Another 2 just showed up. |
| 19:29.00 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:29.40 | [TK]D-Fender | Or they just came back and it's the same squirrels all over ... did you tag them? |
| 19:30.38 | igcewieling | [TK]D-Fender: I took them 15 miles away on the far side of a bay. I'm reasonably confident none of them have returned. |
| 19:31.02 | [TK]D-Fender | natural predators <- |
| 19:31.05 | [TK]D-Fender | #darwin |
| 19:34.46 | igcewieling | [TK]D-Fender: I prefer not to kill them if I can avoid it. |
| 19:35.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:35.38 | robmal | The real question is how did you catch 19 squirrels? |
| 19:35.48 | TazzNZ | carefully ? |
| 19:35.50 | TazzNZ | :) |
| 19:36.07 | igcewieling | they also have to cross 2 rivers to get back to me. |
| 19:36.21 | [TK]D-Fender | igcewieling: You wouldn't be killing them. You'd be creating a supportive habitat for another creature! |
| 19:37.01 | igcewieling | robmal: I have some woods along the back of the property and I've put out bird feeders for birds. If the squirrels didn't destroy the bird feeders I'd not care. |
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| 19:37.22 | avb | seems you live in a nice place :) |
| 19:38.44 | igcewieling | If there is one thing a squirrel cannot resist is looking for food inside of a trap. It's just a matter of time. |
| 19:38.49 | [TK]D-Fender | igcewieling: https://www.youtube.com/watch?v=WGcneF0wx2s |
| 19:39.20 | igcewieling | heh, I've seen that one. |
| 19:40.28 | igcewieling | you should see them whe they try to jump on the pole supporting the bird feeders which I covered with vasaline. 8-| |
| 19:42.44 | igcewieling | HAHA! one down, one to go. |
| 19:46.15 | robmal | So, where's the camera stream for us to enjoy? |
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| 20:04.19 | *** join/#asterisk Shaan (xop@unaffiliated/shaan) |
| 20:04.43 | Shaan | Hi guys, im looking for a VoIP phone that has atleast 5 line buttons does anyone know of such? |
| 20:05.00 | Shaan | not 5 different sip accounts |
| 20:05.08 | [TK]D-Fender | can be the same thing |
| 20:05.12 | [TK]D-Fender | take your pick out of apile |
| 20:05.21 | [TK]D-Fender | polcyom 6XX, |
| 20:05.40 | [TK]D-Fender | several obvious cisco/linsys SPA's |
| 20:05.51 | [TK]D-Fender | Yealink has several |
| 20:05.56 | Shaan | well im loking to have an ip phone connect to my pbx, and then use different line buttons for predefined callerid(s) for various business of mine. |
| 20:05.59 | [TK]D-Fender | all pretty clearly stated |
| 20:06.21 | [TK]D-Fender | that requires separate accounts <- |
| 20:06.41 | [TK]D-Fender | the buttons don't get a magical identifier without being "separate" from the rest |
| 20:06.54 | Shaan | well can you not define them as extensions |
| 20:07.03 | [TK]D-Fender | There is no "but this is button #2" while still being the same "account" |
| 20:07.12 | [TK]D-Fender | What you do in your dialplan is up to you |
| 20:07.20 | Shaan | why not it would just set a different callerid |
| 20:07.22 | [TK]D-Fender | but they will be separate SIP peers for each |
| 20:07.32 | Shaan | the traffic is still going to go out same sip |
| 20:07.40 | [TK]D-Fender | because no phone sets different parameters for the same account |
| 20:07.44 | [TK]D-Fender | that is not how these work |
| 20:08.11 | [TK]D-Fender | No phone does this. There is no concept of "2nd button" as a distinct entity |
| 20:08.31 | Shaan | [TK]D-Fender: okay so let me explain i just want to be able to hit button 1 and it dail an extension which per my dialplan has a certain caller id |
| 20:08.39 | [TK]D-Fender | If you want a physical line-key on one of those to go out differently, then it is going to have be it's own distinct SIP peers |
| 20:08.45 | [TK]D-Fender | I understand what you want |
| 20:08.50 | [TK]D-Fender | And it does not work that way |
| 20:09.15 | Shaan | say if i use dialplan to prefix dial 8 before number couldn't i set callerid that way |
| 20:09.40 | Shaan | becayse my termination provider allows me to set any callerid i wanna pass to them. |
| 20:09.41 | [TK]D-Fender | You could. that would be using your dialplan to determine what to do. |
| 20:09.50 | [TK]D-Fender | And would ahve nothng to do with phone config |
| 20:10.10 | [TK]D-Fender | Rememebr you phone has nothing to do with your provider. Asterisk is what is calling them, not your phone. |
| 20:10.15 | Shaan | ya but then couldn't i program those ' buttons ' to be press 7 or 8 or 5 for me |
| 20:10.21 | [TK]D-Fender | So it's a question of what * sees from the phone's request |
| 20:10.25 | Shaan | or what other way would you suggest |
| 20:10.32 | Shaan | without using multiple sip peers |
| 20:11.05 | [TK]D-Fender | Havintg it dial things like that would be speed-dials, not separate line appearances |
| 20:11.45 | [TK]D-Fender | at which point you COULD set up something for them to dial, but then you'd have to pick up the actual number to dial out via the dialplan.(actually Read, Waitexten, etc) |
| 20:11.56 | [TK]D-Fender | And this would screw up your CDR aws well |
| 20:12.07 | [TK]D-Fender | Would make for a nice mess to have to clean up after |
| 20:12.20 | [TK]D-Fender | but gets a little closer to the end result |
| 20:12.21 | Shaan | so what would be the clean way to accomplish what i am trying to do |
| 20:12.58 | [TK]D-Fender | Either set up multiple regs per phone, or just dial the prefix as part of the numebr yourself and forget having multiple line-keys for this |
| 20:14.03 | Shaan | ya but i dont want to remember each prefix, thats what i was hoping i could use phone labels on the buttons |
| 20:14.16 | Shaan | like for instance on cisco 8860 its digital |
| 20:14.36 | [TK]D-Fender | means nothing |
| 20:14.53 | [TK]D-Fender | SIP doesn't have signalling to use the same peer and have the "button" indicate anything extra |
| 20:15.33 | Shaan | so what is the best approach to this then? |
| 20:15.38 | Shaan | have multiple regs? |
| 20:15.41 | [TK]D-Fender | I just gave you your 2 real options |
| 20:15.55 | [TK]D-Fender | if you don't want to remember the prefix then make them separate regs |
| 20:16.03 | Shaan | speed dial is the only way i can do the prefix right? |
| 20:16.09 | Shaan | if i want buttons? |
| 20:16.14 | [TK]D-Fender | Or as speed-dials on a single reg and sacrifice sane CDRs |
| 20:16.58 | [TK]D-Fender | And setup more dialplan to take in the actual number since you'll be in the dialplan |
| 20:18.13 | Shaan | i can do multiple reg's to my PBX and then to the termination provider it would still be one reg correct?, sorry havent messed around with asterisk in several years. |
| 20:18.44 | [TK]D-Fender | What you call out to the provider with is it's own peer and has nothing to do with your phones' |
| 20:18.55 | [TK]D-Fender | So that's just 1 peer normally |
| 20:19.09 | [TK]D-Fender | And in your dialplan you'd set the CID accordingly before placing the call out |
| 20:20.05 | Shaan | Okay sounds good i'll do it that way, now can you please suggest a phone? do you know if the cisco 8860 works with asterisk? |
| 20:20.18 | Shaan | or could you please suggest a polycom ? |
| 20:20.30 | [TK]D-Fender | Depends which way you want to do this |
| 20:20.39 | robmal | Where did the squirrels plot go! |
| 20:20.41 | [TK]D-Fender | And if you want anything fancier |
| 20:20.57 | Shaan | i would like a digital display |
| 20:21.03 | [TK]D-Fender | Polycom 6XX can to 6 visible regs or 1 reg + 5 SD'd |
| 20:21.14 | [TK]D-Fender | display isn't the question. |
| 20:21.20 | robmal | Oh, polycoms <3 |
| 20:21.21 | [TK]D-Fender | the METHOD you are deciding upon is |
| 20:21.39 | Shaan | yes multiple reg's |
| 20:21.40 | [TK]D-Fender | Aastra's 6739i (or whever the newer ver is) |
| 20:22.14 | [TK]D-Fender | Polycom 6XX, Yealink has some obvious labelled options but are lower quality. |
| 20:22.20 | [TK]D-Fender | Several Aastra also would work |
| 20:22.33 | robmal | Polycom 650 |
| 20:22.35 | [TK]D-Fender | Check their lineups , but Poly wins on quality hands down |
| 20:22.54 | robmal | Yealink t41p |
| 20:23.11 | robmal | Or the vvx line on polycoms, they're fun. |
| 20:24.11 | Shaan | how are grandstream? |
| 20:24.16 | Shaan | and aastra is now mitel |
| 20:24.29 | [TK]D-Fender | ~gs |
| 20:24.29 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:24.31 | [TK]D-Fender | ~grandstream |
| 20:24.31 | infobot | [grandstream] the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 20:24.34 | robmal | If you want problems go to grandstream. |
| 20:25.16 | Shaan | http://www.voipsupply.com/mitel-6869i |
| 20:25.27 | Shaan | aastream/mitel how are those? |
| 20:25.54 | [TK]D-Fender | Decent. Not a bad choice. Few things I like more, others I like less |
| 20:26.01 | Shaan | whats the best one/ |
| 20:26.11 | Shaan | let me ask that lame question |
| 20:26.16 | [TK]D-Fender | Depends on your needs & wants |
| 20:26.33 | [TK]D-Fender | Mitel might be an easier choice for you to manage |
| 20:27.46 | Shaan | would the cisco 8860 work |
| 20:27.50 | Shaan | or the 88xx series |
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| 20:29.41 | [TK]D-Fender | Cisco = PAIN |
| 20:29.49 | [TK]D-Fender | pain you would be best to avoide |
| 20:30.16 | [TK]D-Fender | And on that note: checkout time |
| 20:30.28 | [TK]D-Fender | head for the water... |
| 20:30.36 | Shaan | http://www.voipsupply.com/cisco-cp-9971-c |
| 20:30.39 | Shaan | ok cool |
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| 20:37.27 | cmendes0101 | In extensions.conf, is there a variable for checking the SIP user that is in the context? |
| 20:37.50 | file | not unless you make one. |
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| 21:13.59 | jfred | This might be the wrong place, because I know this is the general Asterisk channel, but... does anyone with a Digium Switchvox system know if there's a way to find out which virtual users are logged in at a specific SIP extension via the API? |
| 21:19.21 | jfred | There's an API call to show which queues a user is a member of, but there doesn't seem to be one for logged-in agents |
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| 22:40.08 | xochilpili | hi all, how can i activate "console" in asterisk? |
| 22:41.07 | WIMPy | rasterisk |
| 22:43.19 | xochilpili | WIMPy, no, i mean the module console |
| 22:43.31 | WIMPy | What? |
| 22:43.32 | xochilpili | but i found it; module load chan_oss |
| 22:43.37 | xochilpili | console dial << |
| 22:43.53 | WIMPy | Configure one of the console channels. |
| 22:44.15 | WIMPy | chan_alsa, chan_oss or err chan_console? |
| 22:44.52 | xochilpili | WIMPy, yes, but i did it |
| 22:44.54 | xochilpili | thanks ! |
| 22:44.57 | xochilpili | :D |
| 22:45.46 | WIMPy | You should only have one of them loaded. |
| 22:47.38 | wyoung | WIMPY!!!!!!!!1 |
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