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| 04:04.18 | bjhaid | HI |
| 04:04.21 | bjhaid | hi |
| 04:04.35 | bjhaid | how does one get CDRs with asterisk AMI |
| 04:10.58 | [TK]D-Fender | you don't |
| 04:13.32 | bjhaid | D-Fender: so what would you advise if I wanted to retrieve CDR, or call status in about realtime |
| 04:13.50 | MaliutaLap | bjhaid: use a database |
| 04:14.36 | bjhaid | MaliutaLap: I don't understand, I configure asterisk to write CDRs to a database and query the database right? |
| 04:16.26 | [TK]D-Fender | put a TRIGGER on the database if you want to be informed of changed instead of polling blindly |
| 04:17.47 | bjhaid | nice thanks |
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| 07:43.55 | Zogot | what is the format required for voicemails secret.conf in the spool directory |
| 07:44.25 | Zogot | i just keep getting Failed reading voicemail password from /directory/blabla/secret.conf, using secret from config file |
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| 08:24.20 | andycol_500 | hi guys |
| 08:24.34 | andycol_500 | is it possible to put the sip username in the contact field of the sip invite |
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| 08:24.54 | andycol_500 | similar to how freeswitch does it with name="extension-in-contact |
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| 09:20.31 | andycol_500 | any idea ? |
| 09:36.24 | andycol_500 | is it possible to modify the contact field in the sip invite? |
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| 11:36.23 | hypfer | hi, small question: is it possible with asterisk to automatically accept calls and play some message while waiting for a real person to answer a still ringing phone? |
| 11:36.30 | hypfer | and if that doesn't happen send to voicemail? |
| 11:36.43 | WIMPy | sure |
| 11:37.08 | hypfer | thats pretty cool :) |
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| 12:42.42 | t4nk525 | hello all |
| 12:43.00 | t4nk525 | need some help |
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| 12:46.08 | Chainsaw | t4nk525: With an Asterisk installation? |
| 12:46.29 | t4nk525 | have asterisk 11 + fpx 2.11 with 4 port fxo card. worked very well for last two months, then yesterday inbound calls give busy signal and unable to make outbound calls. Anybody experienced this before and how to solve. Thank you for your help |
| 12:46.41 | t4nk525 | yes |
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| 12:47.48 | t4nk525 | chainsaw: yes |
| 12:48.01 | [TK]D-Fender | Show a failed call. |
| 12:48.25 | t4nk525 | the inboud does not show up on log |
| 12:48.53 | [TK]D-Fender | mumbles something about "Got a dead call, then show us a BODY if you want an autopsy".... |
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| 12:53.21 | t4nk525 | <PROTECTED> |
| 12:53.44 | [TK]D-Fender | So far it answered. |
| 12:53.50 | [TK]D-Fender | PASTEBIN the entire call |
| 12:54.03 | [TK]D-Fender | And it would help if you told us what you were using as well.... |
| 12:54.09 | WIMPy | BTW, [TK]D-Fender I think you missed cdr_manager earlier. |
| 12:54.10 | [TK]D-Fender | Which card? |
| 12:55.18 | [TK]D-Fender | WIMPy: News to me, looking up now. |
| 12:57.02 | [TK]D-Fender | WIMPy: Imagine that.. something I've missed for a long time... |
| 12:57.29 | [TK]D-Fender | WIMPy: thanks for passing on the note. Curious how shitty the sample config is compared to a decrepit voip-info WIKI page on it... |
| 12:59.43 | WIMPy | Unfortunatly the guy who asked was also gone. |
| 13:00.12 | [TK]D-Fender | Well I stand corrected at least and I'll have something new to advise for the next time. |
| 13:00.24 | [TK]D-Fender | I'm good with that. |
| 13:12.18 | t4nk525 | here is the pastebin link for outgoing calls http://pastebin.com/Z7SFkcBd |
| 13:14.14 | t4nk525 | and for inboud the log shows nothing |
| 13:15.27 | t4nk525 | i am running asterisk 11 with freepbx 2.11 on opensuse 13.2 with TDM 410p fxo card |
| 13:15.46 | t4nk525 | this setup worked for two months straight with no issues |
| 13:16.39 | t4nk525 | then yesterday inbound calls were not being logged and the caller only receives a busy signal, whilst outbound calls do not get to the destination even though it is processed |
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| 13:27.13 | t4nk525 | [TK]D-Fender: here is the pastebin link http://pastebin.com/Z7SFkcBd |
| 13:28.34 | t4nk525 | [TK]D-Fender: the inbound call does not show up on the log screen and the caller always gets a busy tone |
| 13:29.27 | t4nk525 | [TK]D-Fender: I unplugged the lines from the TDM 410p card into a telephone box and the line can receive and make calls |
| 13:29.36 | t4nk525 | [TK]D-Fender: thank you |
| 13:31.55 | andycol_500 | has anyone managed to change the contact field in a sip invite in asterisk |
| 13:32.02 | andycol_500 | i need it to show the username not the did |
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| 13:35.21 | [TK]D-Fender | t4nk525: Says it called and answered |
| 13:35.44 | [TK]D-Fender | <PROTECTED> |
| 13:39.39 | t4nk525 | [TK]D-Fender: yes it says so on the log, but in reality it goes silent and does not ring at the destination |
| 13:40.03 | [TK]D-Fender | Again, are we sure that's the correct port? |
| 13:40.11 | [TK]D-Fender | And the correct module on the card... |
| 13:40.32 | WIMPy | Have you tried to connect a phone to check theline? |
| 13:40.56 | t4nk525 | [TK]D-Fender: yes it is, there are 4-fxo modules on the card and all 4 ports are connected and have worked very will till yesterday |
| 13:41.35 | t4nk525 | WIMPy: connecting the lines to a phone works perfectly |
| 13:42.16 | [TK]D-Fender | try another port on the card |
| 13:42.34 | [TK]D-Fender | Isolate taht a module didn't fry, etc |
| 13:43.44 | t4nk525 | [TK]D-Fender: ok will try that |
| 13:44.00 | t4nk525 | [TK]D-Fender: this is dahdi show channels though http://pastebin.com/HEXMr5Pt |
| 13:46.05 | t4nk525 | [TK]D-Fender: and Dahdi Show status http://pastebin.com/W4XVqXtt |
| 13:47.03 | [TK]D-Fender | so far so good |
| 13:47.07 | [TK]D-Fender | go prove the port isn't fried |
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| 13:54.48 | t4nk525 | [TK]D-Fender: i get the same result from all the other 3 ports as well |
| 13:56.09 | *** join/#asterisk todwetsprock (~florian@p5099a594.dip0.t-ipconnect.de) |
| 13:56.14 | todwetsprock | hello |
| 13:56.25 | todwetsprock | howto setup fax->email ? |
| 13:56.36 | todwetsprock | dahdi |
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| 13:56.41 | t4nk525 | [TK]D-Fender: port no. 1 upon outbound just gives static after connecting and does not reach destination, all other ports are give silence |
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| 14:05.08 | [TK]D-Fender | t4nk525: Next test I'd completely shuttdown and restart that machine |
| 14:05.48 | [TK]D-Fender | todwetsprock: Call the fax receive app of your choice based on what you installed. Call an external e-mail script.to send it |
| 14:07.11 | t4nk525 | [TK]D-Fender: ok will do that now |
| 14:07.32 | [TK]D-Fender | t4nk525: I've seen cards seize up and come back fine on reboot many times |
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| 14:21.58 | t4nk525 | [TK]D-Fender: restarted no change |
| 14:25.12 | igcewieling | t4nk525: did you remember to plug power into the card? |
| 14:25.36 | WIMPy | He's FXO only |
| 14:27.31 | igcewieling | I still recommend pluging power into it. |
| 14:28.18 | [TK]D-Fender | yup |
| 14:28.42 | igcewieling | today is a great day: http://www.theguardian.com/technology/2015/oct/07/linkedin-lawsuit-add-connection-feature |
| 14:30.22 | t4nk525 | igcewieling: the card is fxo and has worked for 2 months straight |
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| 14:32.13 | igcewieling | Having given my advice, I'll go back to fixing polycom phones. |
| 14:33.39 | [TK]D-Fender | Did you completely power down the server? |
| 14:33.46 | [TK]D-Fender | Or just "reboot"? |
| 14:39.30 | t4nk525 | [TK]D-Fender: completely powered down, removed cable for 10 secs and powered up again |
| 14:40.01 | t4nk525 | igcewieling: the card is powered |
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| 15:21.26 | millsu2 | I have a setup with 1000+ sip extensions with qualify on. WHen a sip reload is done, the qualify appears to happen for all extensions at the same time. Is there a way to limit the rate that these happen after a reload? |
| 15:21.49 | WIMPy | yes |
| 15:22.22 | WIMPy | qualifygap |
| 15:22.45 | igcewieling | WIMPy: only in Asterisk 13? |
| 15:23.10 | WIMPy | no |
| 15:23.27 | WIMPy | Don't know how old it is. |
| 15:28.13 | millsu2 | Thanks! I'm running asterisk 11, will test it and see if it is available in that version. |
| 15:28.55 | WIMPy | it is |
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| 15:43.25 | millsu2 | looks like that option came in 1.6.2, so prety old. |
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| 15:44.16 | millsu2 | is there a way to show what the cerrent setting is for qualifygap? sip show settings doesn't seem to show it. |
| 15:47.43 | WIMPy | mumbles something about sauce or something |
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| 15:53.45 | millsu2 | on a sip reload, does asterisk mark everything as unreachable until the qualify is done? i set up qualifygap=40 and qualifypeers=100 and when I did a sip reload most extensions went unreachable. |
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| 16:43.42 | cj | moo |
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| 16:48.54 | Kobaz | anyone aware of a digital gateway for phones other than citel? Like hook up an Avaya 1408 to a sip gateway? |
| 16:49.38 | WIMPy | Gateway from what to what? |
| 16:49.55 | Kobaz | gateway from avaya 1408 to sip |
| 16:50.03 | Kobaz | ever use a citel? |
| 16:50.06 | [TK]D-Fender | proprietary PBX digital phones |
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| 16:50.22 | Kobaz | it can take a definity 48xx and it's a sip gateway |
| 16:50.24 | Kobaz | [TK]D-Fender: yeap |
| 16:50.37 | Kobaz | but citel never updated their hardware support to do newer digital phones |
| 16:50.45 | WIMPy | No, but I wonder where they got the specs or if they rely on reverse engeneering. |
| 16:51.46 | [TK]D-Fender | Norstar is only a little off of ISDN specs |
| 16:52.02 | [TK]D-Fender | As I'd been told |
| 16:52.07 | igcewieling | He is talking about one of those horrible devices which tries to take proprietary PBX digital phones from Nortel and Avaya into "SIP" phones. |
| 16:52.26 | [TK]D-Fender | The standards are somwhat documented as many 3rd party interfaces have existed for these makers PBX's |
| 16:52.28 | igcewieling | The problem is, of course, the device is almost as expensive per port as it would be to buy all new phones. |
| 16:52.43 | [TK]D-Fender | igcewieling: Absolutely |
| 16:52.48 | Kobaz | igcewieling: i know, it's terrible |
| 16:52.55 | WIMPy | It's putting the gateway in the wrong place. |
| 16:52.59 | Kobaz | igcewieling: customer has 950 stations and wants to put 0 money into new wiring |
| 16:53.21 | igcewieling | Kobaz: then make them a former customer. |
| 16:53.22 | Kobaz | but yet we're bidding on a $250,000 phone system |
| 16:53.24 | [TK]D-Fender | The only reason to get something like a Citel is if the customer is either retarded and insists on reusing their old crap, or they are in a terrible building restricted by single-pair runs. |
| 16:53.34 | Kobaz | throw in another 100k for wiring, what's the big deal? |
| 16:53.34 | Kobaz | hehe |
| 16:53.41 | igcewieling | they are obviously rerranged, |
| 16:53.43 | Kobaz | they want new handsets |
| 16:53.53 | [TK]D-Fender | So NEW old crap ;) |
| 16:53.54 | Kobaz | they dont want to use the old nec's |
| 16:53.58 | Kobaz | exactly |
| 16:54.01 | Kobaz | avaya makes the 1408 |
| 16:54.05 | Kobaz | which is a nice digital gateway |
| 16:56.21 | igcewieling | what wiring do they have now? |
| 16:58.21 | WIMPy | I'd also like to know what kind of hardware is in those citel things. |
| 16:58.25 | Kobaz | single pair |
| 16:58.35 | Kobaz | 780 digital phones on single pair |
| 16:58.39 | Kobaz | 950 phones total |
| 16:59.06 | Kobaz | <Kobaz> avaya makes the 1408 <Kobaz> which is a nice digital gateway.. whoops, typo... nice digital phone |
| 16:59.07 | WIMPy | And why not keep them on the PBX? |
| 16:59.15 | igcewieling | ah, yes, I'd run away from them. |
| 16:59.17 | Kobaz | because the pbx is 30 years old |
| 16:59.36 | Kobaz | if only polycom made a nice digital phone |
| 16:59.43 | igcewieling | I've never regretted firing a customer. |
| 17:00.39 | igcewieling | the last one who I fired tried sueing me for not installing the latest OS updates -- 3 months after I fired them. |
| 17:00.51 | Kobaz | somehow cisco is competing on the bid as well |
| 17:01.03 | Kobaz | but they can't pull it off, they have no digital phones |
| 17:01.19 | Kobaz | and since the customer is no-rewire |
| 17:01.24 | Kobaz | i don't know why they are bothering |
| 17:02.00 | WIMPy | They can put a DSL modem under each phone :-) |
| 17:02.48 | Kobaz | yeah |
| 17:02.52 | Kobaz | at $300 bucks a port |
| 17:03.02 | Kobaz | that's + $200k in hardware |
| 17:03.06 | Kobaz | that we don't make a penny on |
| 17:03.16 | Kobaz | it would be cheaper to re-wire |
| 17:03.26 | WIMPy | If you also want POE so you really only need that single pair. |
| 17:17.32 | *** join/#asterisk airjump (~Thunderbi@p20030070CE363059612F13CE40B11771.dip0.t-ipconnect.de) |
| 17:24.09 | *** join/#asterisk airjump (~Thunderbi@p20030070CE363059612F13CE40B11771.dip0.t-ipconnect.de) |
| 17:36.59 | *** join/#asterisk Stefan26 (~Stefan27@195-67-255-105-no207.tbcn.telia.com) |
| 17:38.27 | Stefan26 | if a queue has a single agent who whenever he receives a sdp invite always answers busy here immediately, is the asterisk queue supposed to try calling him instantly, not waiting for next ring cycle? |
| 17:38.41 | Stefan26 | calling him again* |
| 17:42.45 | *** join/#asterisk dimitry7 (~dimitry7@gate.aaamerica.com.mx) |
| 17:43.14 | dimitry7 | Hey guys, I have my asterisk running behind an IPtables firewall PBX IP: 192.168.14.253, the Firewall is 192.168.14.254 |
| 17:43.43 | dimitry7 | and I get this error on another iax peer: Host 192.168.14.254 failed to authenticate as iaxaereo |
| 17:44.44 | dimitry7 | I thought there was a problem with username / password but it isn't because if I place the asterisk service in the 192.168.14.254 server it can authenticate with no problem. |
| 17:44.48 | dimitry7 | what can be wrong? |
| 17:45.13 | *** join/#asterisk mub (~jub@50-205-171-170-static.hfc.comcastbusiness.net) |
| 17:51.17 | *** join/#asterisk italorossi (~Adium@187.60.66.11) |
| 17:57.39 | igcewieling | You are using a protocol almost nobody used anymore. |
| 18:00.10 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:1413:8219:34ff:fecf:17f0) |
| 18:07.25 | [TK]D-Fender | dimitry7: Only thing that is likely is that the packets are getting rewritten and because they are coming from 14.254 they fail |
| 18:07.43 | [TK]D-Fender | That's the first suspicion based on what you've said and the fac you haven't shown us the call... |
| 18:08.28 | *** join/#asterisk gusto (~gusto@2001:4c50:62e:1413:8219:34ff:fecf:17f0) |
| 18:08.43 | robmal | dimitry7: Follow this guide: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 18:10.58 | *** join/#asterisk millsu2 (~Brad_Gard@69.167.49.201) |
| 18:12.10 | dimitry7 | [TK]D-Fender, exactly, they are being rewritten :S. Although I have this same config on serveral IAX boxes and it fails only in some of them |
| 18:12.18 | dimitry7 | 70% works good! |
| 18:18.23 | *** join/#asterisk u0m3 (~u0m3@92.80.81.240) |
| 18:20.15 | *** join/#asterisk u0m3 (~u0m3@92.80.81.240) |
| 18:23.35 | pankid | What could cause a phone to not send its secret after the first unauthorized from the server? |
| 18:23.37 | pankid | https://paste.debian.net/314851/ |
| 18:26.53 | igcewieling | pankid: generally masking anything other than passwords just annoys people |
| 18:27.05 | pankid | yeah sorry |
| 18:27.55 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 18:28.15 | pankid | https://paste.debian.net/314853/ unmasked |
| 18:28.16 | [TK]D-Fender | dimitry7: rewriting the network packet off your firewall is a networking violation. It should not be treated as a MITM |
| 18:30.24 | igcewieling | pankid: are you experienced in using Polycom phones? |
| 18:30.45 | dimitry7 | [TK]D-Fender, I think that would solve the problem!, I found this to rewritte packets: https://sandilands.info/sgordon/address-spoofing-with-iptables-in-linux |
| 18:30.49 | dimitry7 | SNAT |
| 18:31.00 | [TK]D-Fender | Don't |
| 18:31.03 | pankid | igcewieling: Somewhat. |
| 18:31.09 | [TK]D-Fender | never fuck with the SOURCE. |
| 18:31.10 | robmal | dimitry7: Yes, very don't. |
| 18:31.28 | dimitry7 | [TK]D-Fender, no? okay. Why? |
| 18:31.32 | igcewieling | paste all the reg lines from the polycom config, be sure to mask the password |
| 18:31.42 | igcewieling | dimitry7: because it breaks stuff. |
| 18:31.47 | robmal | pankid: You're looking for retry maximum count. |
| 18:31.49 | [TK]D-Fender | type=peer ID by tyhe IP ADDRESS |
| 18:31.55 | [TK]D-Fender | IAX is not a routed protocol. |
| 18:32.00 | [TK]D-Fender | It has to COME from the right place |
| 18:32.02 | dimitry7 | ohh... god. |
| 18:32.04 | [TK]D-Fender | You can't go fucking with that |
| 18:32.11 | dimitry7 | okay ok |
| 18:32.20 | dimitry7 | I'll pastebin my config, hold on |
| 18:32.26 | [TK]D-Fender | This is not like TCP. There is no "connection" to track back |
| 18:32.39 | [TK]D-Fender | The smarter you think you are the more wrong you'll discover you are |
| 18:32.59 | [TK]D-Fender | Forget the config. Look at the CALL. |
| 18:33.09 | [TK]D-Fender | Verify that it is sourcing properly first |
| 18:33.18 | [TK]D-Fender | And then as the right user, etc |
| 18:33.22 | robmal | No, the internet quote for today is: Clever people solve problems, smart people avoid them. |
| 18:34.48 | pankid | igcewieling: This is from /tftpboot. Not sure how to get the config currently on the phone. voIpProt.server.1.retryTimeOut="0" voIpProt.server.1.retryMaxCount="0" |
| 18:34.54 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 18:35.16 | igcewieling | pankid: that was for robmal |
| 18:35.35 | robmal | No forwarding tickets! |
| 18:35.37 | robmal | ;-) |
| 18:35.37 | pankid | oh yeah, suorry |
| 18:35.56 | dimitry7 | [TK]D-Fender, okay this is my config: https://paste.debian.net/314857/ |
| 18:35.57 | robmal | pankid: What firmware are you using and why not 4.x.x? |
| 18:36.13 | igcewieling | say my name when you have the reg info and I should notice it. |
| 18:36.17 | robmal | dimitry7: Good job. |
| 18:36.43 | dimitry7 | robmal, :-) |
| 18:37.00 | c|oneman | Digium has been pushing 3.2.x polycom firmware and stil does :/ |
| 18:37.08 | [TK]D-Fender | dimitry7: I don't see a basic TYPE in there or any other settings yet. From what I see those peers will not even get defined |
| 18:37.11 | robmal | dimitry7: Now go read this http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 18:37.42 | igcewieling | [TK]D-Fender: have you considered quitting and letting the n00bs fend for themselves? |
| 18:37.55 | robmal | c|oneman: Because they sell their own phones. |
| 18:38.04 | c|oneman | heh. |
| 18:38.19 | dimitry7 | robmal, [TK]D-Fender thanks, I'll adjust my config |
| 18:38.22 | robmal | igcewieling: I'm sure he's addicted to 'trunks' |
| 18:38.30 | robmal | Instead of coffee he waits. |
| 18:38.31 | c|oneman | well, maybe their support team should try to upsell me, not be all like, oh, uh er, reconfigure the phone from scratch bro |
| 18:38.45 | hexanol | when using an "#include my_dir/*.conf" construct inside an asterisk configuration file, is there a guarantee that the file will be include in the alphabetical order ? |
| 18:39.08 | robmal | c|oneman: But setting up tftp takes less time than explaining the problem. |
| 18:41.52 | [TK]D-Fender | igcewieling: https://africacheck.org/wp-content/uploads/2012/08/Someone-is-wrong-on-internet.png |
| 18:42.15 | robmal | :-)))))) |
| 18:42.18 | igcewieling | [TK]D-Fender: once you let go of that, life is better. |
| 18:42.30 | igcewieling | I know I'm right, everyone else can go to hell. |
| 18:42.40 | igcewieling | see? 8-) |
| 18:42.51 | [TK]D-Fender | igcewieling: I still log in at work because I can only surf up distraction so much... |
| 18:43.42 | *** join/#asterisk deweydb (~deweydb@d172-218-160-124.bchsia.telus.net) |
| 18:43.54 | *** join/#asterisk hecatae (~philip@host-89-240-2-225.static.as13285.net) |
| 18:43.59 | igcewieling | our sales people keep things too interesting. |
| 18:44.11 | deweydb | Hi! i have a strange question. I'm looking for a VOIP provider that is a century link reseller, and not a Level 3 communications reseller. anyone know of one? |
| 18:45.09 | *** join/#asterisk bkruse (~Adium@64.89.97.100) |
| 18:45.25 | robmal | deweydb: Check out 373k.net, they spammed here lately. |
| 18:46.28 | mub | My proxmox box says I have no quorum? We had a power-outage last night which may have affected some of the nodes |
| 18:46.43 | mub | But ... Oh wait this is #asterisk |
| 18:52.23 | hexanol | ok I found the answer by myself, it is sorted since it's using glob(3) without the GLOB_NOSORT flag |
| 18:53.00 | pankid | how can I grab the config form a polycom phone? I know where the configs are stored on /tftpboot, but not sure how to grab the config actually on the phone |
| 18:53.20 | igcewieling | pankid: go into the web interface of the phone. |
| 18:53.23 | robmal | pankid: So i repeat the question, what firmware are you using? |
| 18:53.34 | robmal | 3.X didn't have 'download config' |
| 18:53.40 | igcewieling | though if you are not running v4.0+ the web interface is almost useless. |
| 18:53.47 | pankid | robmal: Not sure hwo to tell. This webgui doesnt really tell me anything |
| 18:53.48 | robmal | It had pain and sorrow. |
| 18:53.58 | robmal | pankid: Is it red/gray ? |
| 18:54.01 | pankid | yes |
| 18:54.05 | pankid | pain and sorrow? |
| 18:54.07 | robmal | So 3.X it is. |
| 18:54.20 | robmal | What models are you using? |
| 18:54.28 | pankid | soundpoint 501 |
| 18:54.40 | robmal | Only that? |
| 18:54.48 | pankid | for the most part |
| 18:54.52 | pankid | this one is a 501 |
| 18:55.02 | robmal | :-/ Thats bad. |
| 18:55.05 | wdoekes | hexanol: that's right |
| 18:55.09 | pankid | bad? |
| 18:55.10 | robmal | 501 don't support 4.X |
| 18:55.15 | robmal | Observe: |
| 18:55.19 | robmal | The matrix: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 18:55.38 | robmal | Anything below 4.X is pain and sorrow. |
| 18:56.04 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
| 18:57.13 | [TK]D-Fender | I've used 1.5 up... it's ALL fine |
| 18:57.16 | [TK]D-Fender | It all works. |
| 18:57.25 | [TK]D-Fender | Documentation is FINE. |
| 18:57.28 | [TK]D-Fender | VERY FINE MANUALS |
| 18:57.35 | pankid | lol, is that sarcasm I sense? |
| 18:57.50 | [TK]D-Fender | Nope |
| 18:57.56 | robmal | No, no, everything was documented very fine as 'not supported'. |
| 18:58.30 | robmal | pankid: Check your logs if the phones are trying to upload their local config via tftp. |
| 18:58.35 | [TK]D-Fender | robmal3.X didn't have 'download config' <- real admins config their phones not ask their phones to hand stuff over. |
| 18:58.35 | robmal | They do in 4.X |
| 18:58.58 | [TK]D-Fender | You're mixing up who should be boss |
| 18:59.05 | robmal | You? |
| 18:59.06 | robmal | ;-) |
| 19:01.24 | robmal | [TK]D-Fender: BTW, i don't remeber 3.X too well, was there screencapture? |
| 19:02.07 | pankid | Is this a relevant error |
| 19:02.09 | pankid | 1007162835|copy |4|00|Download of 'bootrom.ld' FAILED on attempt 1 (addr 1 of 1) |
| 19:02.09 | pankid | 1007162835|copy |4|00|Server 'phone.latitudesubro.com' said 'bootrom.ld' is not present |
| 19:02.09 | pankid | 1007162835|cfg |4|00|Could not get all 512 bytes of the header |
| 19:02.39 | robmal | As long as you don't format file system it shouldn't matter. |
| 19:03.02 | robmal | But yeah. |
| 19:03.16 | robmal | Use The Matrix to download your firmware and put it in /tftpboot just in case. |
| 19:03.21 | robmal | It won't hurt. |
| 19:03.58 | robmal | I'm sure guys at Polycom will have a drink this friday since it will be the first download of this version in 3 years. |
| 19:06.13 | *** join/#asterisk mub (~jub@50-205-171-170-static.hfc.comcastbusiness.net) |
| 19:06.32 | pankid | lol |
| 19:06.41 | pankid | so will this effect all the other phones on my network? |
| 19:09.48 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-uqkrvznddesifuog) |
| 19:09.48 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:15.22 | *** join/#asterisk glNito (6ca1d582@gateway/web/cgi-irc/kiwiirc.com/ip.108.161.213.130) |
| 19:15.58 | dimitry7 | Can I route a call through another E1 on asterisk? I have a sangoma card with 2 ports, each with a different E1 |
| 19:16.37 | dimitry7 | I want an ext: 1602 to make calls through the second E1, although I see both as one interface: w1g1 |
| 19:16.45 | robmal | pankid: Basically yes. Depends. |
| 19:16.58 | pankid | yeah, that is what I thought |
| 19:17.39 | pankid | what is a good way to revert the phone to defaults and just try manually authenticating without a provisioning server? |
| 19:18.42 | robmal | Each has a web panel. |
| 19:18.47 | glNito | you don't want to even start doing that if you don't have to |
| 19:19.27 | robmal | You can, i was just told it's very well documented. |
| 19:20.24 | [TK]D-Fender | dimitry7: It's up to you to configure each port separately |
| 19:21.40 | dimitry7 | [TK]D-Fender, how can I list the available ports? |
| 19:21.55 | [TK]D-Fender | man wanrouter |
| 19:22.01 | [TK]D-Fender | read the docs on your card |
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| 19:24.16 | dimitry7 | [TK]D-Fender, okay I can configure the ports in dahdi/system.conf. now, in extensions.con, I have a TTRUNK=DAHDI/g1. How do I set the second trunk and how do I route the call through it? |
| 19:24.30 | dimitry7 | * extensions.conf |
| 19:24.35 | [TK]D-Fender | Set up your channels. make your groups. Use them |
| 19:24.58 | [TK]D-Fender | By asking that question it's looking like you don't understand what "TTRUNK=DAHDI/g1" even means... |
| 19:28.42 | dimitry7 | [TK]D-Fender, I thought it was the trunk interface |
| 19:32.30 | dimitry7 | I don't know much about telecommunications. I guess that's used by asterisk to understand singnalling, no? |
| 19:32.37 | dimitry7 | * signalling |
| 19:34.10 | dimitry7 | I thought I needed to separate the TRUNK because that TRUNK is for a E1, and I have 2 E1s |
| 19:34.39 | [TK]D-Fender | ~book |
| 19:34.39 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:34.48 | dimitry7 | Yes Im reading it |
| 19:34.49 | [TK]D-Fender | You need to understand your basics on DADHI |
| 19:34.59 | dimitry7 | it still isn't clear for me |
| 19:35.03 | [TK]D-Fender | that is used in a DIAL command to tell * what device to use |
| 19:35.05 | dimitry7 | yes I read the basics |
| 19:35.14 | [TK]D-Fender | that is a GROUP of channels as defined by chan_dahdi.conf |
| 19:35.18 | [TK]D-Fender | Read them again. |
| 19:35.30 | dimitry7 | Yes, now, how do I tell an extension to use that group? |
| 19:35.50 | [TK]D-Fender | You shouls already understnd what a group is and then how to assign the channels on the SECOND port to a different group and use that to dial out |
| 19:36.00 | [TK]D-Fender | dimitry7Yes, now, how do I tell an extension to use that group? <- DIAL the other group |
| 19:36.19 | dimitry7 | Dial(${TRUNK}/${EXTEN}) |
| 19:36.21 | dimitry7 | okay let me try |
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| 21:00.31 | mub | I'm going to leave work at 5 today! :'D |
| 21:00.33 | mub | I'm so happy |
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| 22:58.30 | deweydb | i am setting up asterisk and freepbx and i got to this step in the install where it's asking me: "Enter the IP ADDRESS or hostname used to access the AMP web-admin:" |
| 22:58.36 | deweydb | should that be my server's public ip? |
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| 23:05.16 | [TK]D-Fender | that actual IP on it |
| 23:05.26 | [TK]D-Fender | and this should be in #freepbx |
| 23:06.22 | deweydb | you mean the internal ip? |
| 23:06.39 | deweydb | i don't know what you mean by "actual ip on it" |
| 23:07.08 | [TK]D-Fender | that server has an IP |
| 23:07.11 | [TK]D-Fender | the one ON the server |
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