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| 00:17.20 | MaliutaLap | I have some polycom ip 550's I'd like to upgrade firmware on. Is the UC 4.1.1 AA firware the right stuff? |
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| 00:45.35 | wyoung | MaliutaLap: no idea, this is an asterisk support channel |
| 00:45.41 | wyoung | and I use snom :) |
| 00:46.28 | MaliutaLap | wyoung: I think I know what gets discussed in here, I've only been a regular for about 8 years :P |
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| 00:46.49 | MaliutaLap | wyoung: and my personal preference is Cisco :P |
| 00:46.55 | MaliutaLap | this is for work |
| 00:47.18 | wyoung | MaliutaLap: :P |
| 00:47.30 | wyoung | your personal preference needs some work :) |
| 00:47.36 | wyoung | ah |
| 00:48.01 | wyoung | yeah I use snom for work too, never touched a polycom device in my life |
| 00:48.04 | wyoung | wish I could help |
| 00:48.07 | MaliutaLap | Cisco's with SIP firmware work well for me |
| 00:48.39 | MaliutaLap | Oh, I haven't touched the devices ... I just need to configure/upgrade the ones in other peoples places |
| 00:48.46 | MaliutaLap | we are a work from home outfit |
| 00:49.03 | MaliutaLap | I refused a phone for my place ... given that I don't need to talk to customers |
| 00:49.31 | MaliutaLap | so I have to manage their configs via a webserver |
| 00:59.22 | [TK]D-Fender | MaliutaLap, http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
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| 01:29.34 | snadge | sip default context blocking.. on ancient versions of asterisk |
| 01:29.57 | snadge | i'll google it.. but just wondering if anyone has any useful insight for that |
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| 01:56.52 | snadge | apparently we used to just write default context attempts to a file, which fail2ban scrapes |
| 02:00.19 | [TK]D-Fender | ... what is a "context attempt"? |
| 02:05.00 | snadge | a host attempting to place a call, from the default sip context |
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| 02:21.12 | snadge | its not a big deal, because asterisk is configured to reject them.. but when a host spams you with 50 of those a second, it unnecessarily loads up the server and can cause problems |
| 02:21.49 | snadge | so the idea is.. you log > n blocked attempts.. then block the ip with iptables |
| 02:23.32 | [TK]D-Fender | that is not a "think" |
| 02:23.49 | [TK]D-Fender | thing* |
| 02:23.51 | [TK]D-Fender | SIP knows nothing of "context" |
| 02:24.04 | [TK]D-Fender | the peer you match is he peer you match |
| 02:24.17 | [TK]D-Fender | If you're trying to secure your system then you shouldn't be allowing unauthed calls at all |
| 02:24.22 | [TK]D-Fender | And they should never hit the dialplan |
| 02:26.20 | snadge | you have to have a default context though |
| 02:26.58 | snadge | so under [general] we have.. context=no-unauth-sip |
| 02:28.25 | snadge | in extensions.conf under [no-unuath-sip] we write a line to a text file and hangup |
| 02:29.43 | snadge | allowguest=no is that what you're talking about? |
| 02:34.35 | [TK]D-Fender | YES |
| 02:34.46 | [TK]D-Fender | You should not be allowing them in the FIRST PLACE |
| 02:34.56 | snadge | now i just have to figure out why we're not already using that option.. there must be a reason |
| 02:35.05 | [TK]D-Fender | Not authed as a known peer? GTFO + Security logging |
| 02:36.06 | snadge | yeah its possible we're not using allowguest=no .. specifically so we can log those attempts and ban them from the entire network |
| 02:36.38 | snadge | someone smart turned off the banning though.. so now the logging obviously just creates extra load |
| 02:36.50 | snadge | ie.. we have removed the router/firewall which did that |
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| 04:21.09 | socomm | I'm having problem with auto attendant, whenever I press a digit asterisk detects it as two. |
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| 04:21.17 | socomm | For instance I press 1, asterisk picks up as 11. |
| 04:22.24 | socomm | I tried changing DTMF to rfc2833 to no avail. |
| 04:26.05 | [TK]D-Fender | clairfy exactly what the call is coming in on, show us config, and show us the call |
| 04:26.10 | [TK]D-Fender | ~pb |
| 04:26.10 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:26.11 | [TK]D-Fender | ^^^^ |
| 04:27.04 | socomm | SIP trunk. |
| 04:27.12 | socomm | [TK]D-Fender: ^ |
| 04:27.44 | socomm | [TK]D-Fender: You need config for autoatendant? |
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| 04:36.09 | [TK]D-Fender | Autoattended does do DTMF decode. |
| 04:36.17 | [TK]D-Fender | the CHANNEL driver does |
| 04:36.22 | [TK]D-Fender | So that's SIP config for the peer |
| 04:36.29 | [TK]D-Fender | and the call from CLI with SIP DEBUG enabled. |
| 04:36.40 | [TK]D-Fender | does NOT do* |
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| 04:36.42 | [TK]D-Fender | rather |
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| 06:21.56 | ram_ | hi i want to know from which file the dialpan currently in use by asterisk is getting loaded, i noticed the default context in extensions.conf is not the dialplan currently in use, when i make any new changes to it & then do a dialplan reload is not taking effect, i checked in cli with dialplan show command, the context name which it is showing pbx_config could not find anywhere within /etc/extensions folder |
| 06:22.51 | juned | is it plain asterisk or you're using freepbx or something ? |
| 06:23.14 | ram_ | i m using goautodial |
| 06:24.11 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 06:24.29 | ram_ | infact the dialplan which i had set in goautodial admin panel for the carrier settings is also not the one which is currently in use |
| 06:25.17 | ram_ | the context pbx_config which is pointing to i could not find anwhere under /etc/extensions folder |
| 06:25.37 | juned | ram_: sorry I've never used goautodial so not sure how they're handling stuff |
| 06:26.07 | juned | don't search for pbx_config search for the exact dialplan |
| 06:26.09 | [TK]D-Fender | No such thing in * as "/etc/extensions" |
| 06:26.23 | [TK]D-Fender | pbx_config is the asterisk MODULE responsible for loading the dialplan. |
| 06:26.45 | juned | Correct pbx_config is not a dialplan |
| 06:26.46 | [TK]D-Fender | And that is extensions.conf |
| 06:26.59 | [TK]D-Fender | pbx_ael loads extensions.ael |
| 06:27.02 | [TK]D-Fender | which is another things |
| 06:27.49 | [TK]D-Fender | <PROTECTED> |
| 06:28.09 | [TK]D-Fender | Or maybe your changes are bad so they aren't getting acknowledged when it tries loading |
| 06:28.51 | ram_ | D-Fender: when i typed dialplan show command in cli i have noticed pbx_config next to all exten => which is currently in use |
| 06:29.26 | [TK]D-Fender | Ther is no "currently in use" that isn't a thing |
| 06:29.31 | ram_ | then i searched for pbx_config under all files in /etc/extensions folder |
| 06:29.38 | [TK]D-Fender | those are lines it parsed and loaded |
| 06:30.01 | [TK]D-Fender | and I did NOT say you should look for pbx_config |
| 06:30.01 | ram_ | D-Fender let me copy paste the dialplan show command now |
| 06:30.05 | [TK]D-Fender | that is an asterisk MODULE |
| 06:30.09 | [TK]D-Fender | it is in the MODULES folder |
| 06:30.17 | [TK]D-Fender | and it is not something you have any reason to even look for |
| 06:30.43 | [TK]D-Fender | pbx_config is not a CONFIG FILE |
| 06:30.57 | [TK]D-Fender | it is EXECUTABLE CODE that loads extensions.conf |
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| 06:35.41 | ram_ | D-Fender: please check http://pastebin.com/9sq2df7h |
| 06:36.13 | [TK]D-Fender | I don't see a context there. |
| 06:36.15 | [TK]D-Fender | or anything else |
| 06:36.19 | ram_ | which is the dialplan i need to modify |
| 06:36.20 | [TK]D-Fender | including the command itself |
| 06:36.33 | [TK]D-Fender | I'm not going to comment on some hacked up tiny piece like that |
| 06:36.45 | juned | ram_: there is no context in that pastebin link |
| 06:38.20 | [TK]D-Fender | go find the extensions.conf it's loading and go do whatever you want with it |
| 06:39.17 | ram_ | here is the whole http://pastebin.com/xN4FXqgA |
| 06:39.31 | ram_ | it is default context |
| 06:40.13 | ram_ | however the changes to default context in extensions.conf is not taking effect even after dialplan reload |
| 06:40.40 | [TK]D-Fender | And we don't see these "changes" |
| 06:40.45 | [TK]D-Fender | maybe you did them wrong. |
| 06:40.53 | [TK]D-Fender | You aren't showing us anything useful |
| 06:41.02 | juned | Do you see the same dialplan in your extensions.conf file ? |
| 06:45.45 | ram_ | juned: no it is different let me copy paste that part in extensions.conf |
| 06:46.24 | juned | is there any other file do you see included in extensions.conf ? |
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| 06:52.14 | juned | I'm originating a call using call files but even if I'm rejecting the call as soon as it rings to my cell phone I'm getting status as NO ANSWER instead of BUSY |
| 06:52.59 | [TK]D-Fender | Because that is the status reported back |
| 06:53.06 | [TK]D-Fender | reject != busy |
| 06:54.24 | juned | [TK]D-Fender: so in which case I will get busy status, cause if I'm making extension to extension call and if i'm rejecting the call it's giving me BUSY status, so the same status I'm expecting when I'm making call using trunk |
| 06:54.58 | [TK]D-Fender | Your expectation is clearly wrong |
| 06:55.05 | [TK]D-Fender | And you should actually look at the call |
| 06:55.35 | juned | [TK]D-Fender: http://pastebin.com/a5NYunVj |
| 06:55.58 | [TK]D-Fender | SIP/2.0 180 Ringing |
| 06:56.15 | ram_ | please refer to http://pastebin.com/Eh58ZBXF |
| 06:56.17 | [TK]D-Fender | <PROTECTED> |
| 06:56.27 | ram_ | will paste what is happening in cli as well |
| 06:56.27 | [TK]D-Fender | it rang and there was no answer in the timeout period |
| 06:56.35 | [TK]D-Fender | Which means exactly what it says |
| 06:57.36 | juned | It was ringing that's true but I just rejected the call at the same time |
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| 06:59.14 | [TK]D-Fender | Telco doesn't just take that and pass back busy |
| 06:59.24 | [TK]D-Fender | If it were busy it wouldn't have been RINGING |
| 06:59.40 | juned | Another case someone was calling me so I didn't receive the call even in that case its giving status as NO ANSWER |
| 06:59.48 | juned | Okay I'm testing one more scenario |
| 07:02.47 | [TK]D-Fender | ram_, Indeed you have made a screwed up dialplan and that's why it failed to load |
| 07:02.54 | juned | [TK]D-Fender: so we can say that Trunk is not sending busy |
| 07:03.08 | [TK]D-Fender | ram_, Pay attention to what you actually add. |
| 07:03.23 | [TK]D-Fender | juned, We don't have to say. We can see. It's right there |
| 07:03.31 | [TK]D-Fender | and it shouldn't send a busy |
| 07:03.34 | [TK]D-Fender | because it isn't |
| 07:03.47 | [TK]D-Fender | if it can ring it isn't busty. |
| 07:04.17 | [TK]D-Fender | just because you told them "stop ringing me" doesn't mean it gets to say "Oh forget the fact I told you he was ringing, he's actually *BUSY*". |
| 07:04.21 | [TK]D-Fender | Doesn't work like that |
| 07:04.27 | [TK]D-Fender | No take-backs |
| 07:04.54 | juned | I got it |
| 07:05.09 | juned | May be i should test it with another provider and see |
| 07:05.18 | [TK]D-Fender | No provider should do otherwise |
| 07:05.34 | [TK]D-Fender | when you reject on the caller side, if there is progress then the telco will just ring through |
| 07:05.41 | [TK]D-Fender | it also isn't the PROVIDER doing it... |
| 07:05.47 | [TK]D-Fender | you are using them to hit the PSTN. |
| 07:05.58 | [TK]D-Fender | the "reject is at the FAR SIDE telco |
| 07:06.18 | [TK]D-Fender | that decision and awareness isn't at yYOUR provider, its at the telco of the CALLEE |
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| 07:07.00 | juned | Then how one can get the actual status , is there any other way ? |
| 07:07.38 | [TK]D-Fender | no. |
| 07:07.42 | [TK]D-Fender | and it isn't a "status" |
| 07:07.50 | [TK]D-Fender | it's a reaction. |
| 07:08.23 | [TK]D-Fender | ther is no "I decided I don't want to talk to you" response normally after accpeting ringing |
| 07:10.24 | ram_ | D-Fender: if there are issues with new dialplan changes is that system is using .old version? |
| 07:10.59 | [TK]D-Fender | no. |
| 07:11.07 | [TK]D-Fender | * doesn't jsut pick alternative files to load |
| 07:11.09 | ram_ | why is it not listing errors when dialplan reload is invoked |
| 07:11.15 | [TK]D-Fender | It's bad and the bad bits get ignored |
| 07:11.31 | ram_ | oh ok! |
| 07:11.52 | [TK]D-Fender | And it ashould tell you if you have your verbose up properly |
| 07:11.57 | ram_ | while dialplan reload it is not pointing lines of bad bits |
| 07:12.16 | ram_ | yes i m loading asterisk -vvvvr |
| 07:12.36 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvv |
| 07:12.41 | [TK]D-Fender | Then show me |
| 07:12.48 | ram_ | ok sure thanks |
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| 07:16.49 | ram_ | please find http://pastebin.com/vK3VBY7V |
| 07:17.10 | ram_ | i just pasted only warnings |
| 07:18.13 | [TK]D-Fender | and there they are |
| 07:18.40 | [TK]D-Fender | "t is not pointing lines of bad bits" <- Oh hell yes it is. Right there |
| 07:22.14 | ram_ | D-Fender: Thanks a lot, i fixed the issue, i just learnt about priority bit |
| 07:22.23 | ram_ | which is duplicated |
| 07:22.41 | [TK]D-Fender | And killed 3 lines |
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| 07:22.57 | ram_ | yes |
| 07:23.01 | [TK]D-Fender | because the first dupicate start the failure and the next 2 n's become duplicates as well. |
| 07:23.29 | [TK]D-Fender | Lesson : LOOK at the stuff you make. |
| 07:23.58 | ram_ | infact the new one which i added is Set(CALLERID which is having 1 as priority instead should be n |
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| 07:24.42 | ram_ | D-Fender technology is ocean & hard to swim whole :) |
| 07:24.58 | [TK]D-Fender | These are dialplan basics |
| 07:25.07 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 07:25.10 | [TK]D-Fender | You need to understand contest, exten & priority |
| 07:25.24 | [TK]D-Fender | Dialplan is 95% of learning *. |
| 07:26.27 | [TK]D-Fender | And on that note ..... bed time |
| 07:27.01 | ram_ | i agree, i dived into asterisk recently as no one there for support out small callcenter |
| 07:27.49 | ram_ | thanks again |
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| 07:44.02 | wyoung | Asterisk <3 |
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| 08:38.13 | napnap | Hi all. I have a vlan voip(192.168.50.0) with asterisk as ipbx. All works fine, but now I have soft phone under other subnetwork (192.168.100.0). And When I call phone with this softphone the call hangup after few seconds. However when I call the softphone the call don't hangup. |
| 08:39.04 | napnap | As I read on net it is a nat/route problem . But I don't know how to fix this. |
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| 08:42.24 | TandyUK | sounds like you need to add rules in your firewall to let the networks talk |
| 08:42.53 | TandyUK | NOTE: the firewall between the networks i suspect, not the firewall o nyour asterisk box |
| 08:45.33 | napnap | TandyUK, but the call begin. So I don't think it is the fault of this firewall |
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| 09:03.15 | napnap | X-Asterisk-HangupCause: No user responding |
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| 10:45.04 | babak | Hi , I installed 2 instance of Asterisk 13.7 on vmware ing timing better ? and wants to connect them with TDMoE, Dahdi channels ar up but dahdi_test results are around 2.5% , is there any configuration for mak |
| 10:45.15 | babak | making timing better |
| 10:45.16 | babak | ? |
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| 11:01.32 | guest9019 | Hi, I've built a dialplan to format callerids to E.164. It works 99% of the time but it looks like crap. Are there any modules for this or is it all about dialplan scripting? |
| 11:01.41 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 11:02.00 | guest9019 | The formats coming in can range from 00COUNTRYCODE, COUNTRYCODE, +COUNTRYCODE, LOCAL, anonymous etc |
| 11:02.39 | guest9019 | Now it's a butload of gotoifs |
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| 11:06.50 | kryl | hi |
| 11:07.48 | kryl | is it possible to use an android phone to connect to asterisk to use as an sms gateway ? |
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| 12:50.51 | SirLagz | anyone had any experience getting a Cisco 7945G registered on Asterisk ? I can't seem to get mine registered. Asterisk seems to think it's registered, but the phone just keeps displaying 'Registering' |
| 12:51.22 | SirLagz | I've flashed all the firmware and all that, and the tcpdump seems to show some activity between my asterisk box and my cisco phone |
| 12:57.46 | file | what channel driver, and do you have any NAT settings enabled on the Asterisk side? |
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| 12:59.20 | SirLagz | file: no NAT settings enabled, not sure what you mean by channel driver, sorry |
| 12:59.28 | file | chan_sip or chan_pjsip? |
| 12:59.42 | SirLagz | file: oh, chan_sip I believe...let me double check though |
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| 12:59.57 | file | and ensure nat=no really is set for it... if using chan_sip |
| 13:00.28 | SirLagz | file: ok, will do that now |
| 13:02.22 | SirLagz | file: yep, nat=no is definitely set |
| 13:02.40 | file | then I'd suggest putting a full SIP trace on a pastebin so people can see if the traffic is flowing as expected |
| 13:03.00 | SirLagz | file: ok sounds good. I'll do another dump now and chuck it up. |
| 13:03.39 | SirLagz | file: when you say SIP trace, you mean tcpdump right? any special parameters i should use? |
| 13:03.51 | file | sip set debug on |
| 13:03.58 | file | (in Asterisk) |
| 13:03.59 | SirLagz | file: oh right, whoops |
| 13:07.27 | SirLagz | file: would 'sip set debug ip x.x.x.x' also present the needed information? I have asterisk registering with my VSP as well as a couple of other extensions which clutter the log up |
| 13:07.55 | file | yes |
| 13:08.07 | SirLagz | file: thanks, doing a dump now |
| 13:13.19 | SirLagz | file: http://paste.debian.net/411020/ |
| 13:13.32 | SirLagz | While looking through that dump, I noticed some 401 errors =/ |
| 13:15.09 | file | from the Asterisk side it looks fine |
| 13:15.23 | file | we challenge, they retry with auth, we accept |
| 13:15.32 | SirLagz | file: ah right. |
| 13:15.35 | SirLagz | file: makes sense |
| 13:15.50 | SirLagz | file: I guess I've mucked up the phone config somehow then =/ |
| 13:16.30 | file | has not dealt with Cisco phone configs in years, he simply ensures that a few fundamentals aren't broken in Asterisk to make sure they can still talk to them |
| 13:16.40 | SirLagz | file: probably a good thing lol |
| 13:19.06 | SirLagz | file: thanks for the help anyway |
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| 13:20.30 | file | suggests a hammer |
| 13:21.04 | SirLagz | file: lol. maybe once I get a replacement phone, I can hammer it into oblivion |
| 13:21.32 | file | Digium sells phones! but I'm biased. |
| 13:21.47 | SirLagz | file: the phone + shipping would probably bankrupt me :P |
| 13:23.30 | file | gives options |
| 13:23.52 | SirLagz | file: they do look like nice phones though! |
| 13:24.26 | file | they even make phone calls |
| 13:24.36 | SirLagz | unlike mine! |
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| 13:26.29 | SirLagz | tries another config |
| 13:30.55 | SirLagz | WOOOT. I buggered the config, got another one AND NOW IT WORKS !@#!@@!@!!!! |
| 13:31.28 | file | can't act surprised at that |
| 13:32.44 | file | Cisco phone configs are like a black art finding the write mixture |
| 13:32.46 | file | er right |
| 13:33.39 | SirLagz | file: yep. I swear I did everything the same |
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| 14:33.01 | *** mode/#asterisk [+o cresl1n] by ChanServ |
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| 14:39.30 | julius | hi |
| 14:40.30 | julius | in this picture: http://service.avm.de/support/media/filter/l/transfer/img/4c52ce91-41dc-4124-9307-37aeac100096/anschluss_ntba_tae_y_kabel.png do i have to connect the fritzbox with the dsl connection from my wall plug? because im already using a router thats working fine. i just want asterisk to manage some calls |
| 14:40.36 | julius | the left part of the picture |
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| 14:45.02 | WIMPy | I would. What's the point of havong two routers? |
| 14:46.15 | julius | current router is more current, got openwrt on it for traffic shaping. better wlan |
| 14:46.39 | julius | i ordered a very old fritzbox |
| 14:46.58 | WIMPy | The old boxes didn't have the option to use an existing connection. |
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| 14:47.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:47.58 | WIMPy | And even worse they might not even try to register a SIP account when they don't have an internet connection set up themselves. |
| 14:48.31 | WIMPy | So generelly no, you don't have to technically, but it depends on the firmware. |
| 14:49.23 | julius | i thought sip was just for making the phone know to the box, why would it want to use the internet connection for that? |
| 14:49.58 | WIMPy | Because it expects the other end to be in the internet. |
| 14:50.36 | WIMPy | And as we told you yesterday, old firmwares don't support local SIP devices. |
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| 14:51.50 | *** join/#asterisk alsaiduq (~alsaiduq@200.239.65.110) |
| 14:51.56 | alsaiduq | hello everyone |
| 14:51.58 | *** join/#asterisk theron (~theron@2620:10d:c091:200::4:83d4) |
| 14:52.49 | alsaiduq | does anyone has ever had this error? |
| 14:52.52 | alsaiduq | -> [2016-03-02 11:48:10.790] ERROR[2574][C-00000006]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("brasil", "(null)", ...): Name or service not known |
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| 14:54.58 | WIMPy | Looks like you tried to dial "brasil", which is neither a valid peer name nor a valid host name. |
| 15:01.02 | alsaiduq | yes |
| 15:01.41 | alsaiduq | but i hae already connected asterisk to other asterisk know as 'brasil' and they can see each other |
| 15:01.55 | alsaiduq | i connected them via IAX |
| 15:02.22 | WIMPy | Computer says no. |
| 15:03.04 | alsaiduq | yes, and i dunno why... |
| 15:04.00 | WIMPy | Check your config again. |
| 15:04.00 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 15:04.15 | WIMPy | 'iax2 show peer brasil' |
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| 15:07.09 | alsaiduq | let me see |
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| 15:18.36 | alsaiduq | im stupid |
| 15:18.48 | alsaiduq | i was writing the server name wrong |
| 15:18.56 | alsaiduq | now everyting is fine |
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| 15:26.00 | alsaiduq | can i connect 3 asterisks? |
| 15:26.13 | alsaiduq | is it possible? |
| 15:27.18 | [TK]D-Fender | "Connect" is vague |
| 15:27.35 | [TK]D-Fender | If you're referring to call from one tto the otther then yes |
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| 16:00.07 | julius | WIMPy, true...but; SIP-konform nach RFC 3261 |
| 16:00.22 | julius | it says: sip compliant with rfc 3261 |
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| 16:09.29 | WIMPy | What does that mean? |
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| 16:24.01 | julius | thats a good question |
| 16:24.10 | julius | hopefully that it will support sip |
| 16:25.29 | WIMPy | It will. |
| 16:25.46 | WIMPy | Atlthough that is a very vague statement. |
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| 16:29.02 | julius | yes |
| 16:29.09 | julius | doesnt matter, if its not working. i can send it back |
| 16:29.35 | julius | that thing originally was like 169â¬, now its 15⬠including postage |
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| 17:21.33 | file | Wednesday is. |
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| 17:26.04 | voip | hi |
| 17:26.21 | WIMPy | lo |
| 17:27.25 | newtonr | wut |
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| 17:33.43 | Kevin` | is there a headless client for asterisk that has better sound support than asterisk's chan_alsa/oss? |
| 17:35.13 | [TK]D-Fender | pick a CLI-based softtphone |
| 17:35.22 | [TK]D-Fender | There are a few when I checked ages ago |
| 17:35.26 | [TK]D-Fender | Probably still around |
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| 18:20.28 | shudon | hi all :) we use isymphony at work. asterisk isn't really my responsibility, but i'm learning more about it for fun, mostly. how does isymphony "barge" a call? |
| 18:25.26 | *** join/#asterisk antler (uid39524@gateway/web/irccloud.com/x-ndnhzxammrjlmobl) |
| 18:26.36 | [TK]D-Fender | TThey should have a support forum, etc to explain that... |
| 18:27.53 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 18:27.59 | [TK]D-Fender | It's not a product we support here directly, but should basically be using AMI to do some combination of Bridge or ChanSpy most likely with Originate to bring the barger in |
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| 18:58.09 | shudon | hm |
| 18:58.31 | shudon | [TK]D-Fender: oh i understand, i don't need isymphony support. i just wanted to know how it might facilitate this within asterisk. |
| 18:58.49 | shudon | ideally i can replace isymphony altogether since we really only use it for a couple of features |
| 18:58.54 | shudon | but mostly this is just for my curiosity |
| 18:58.55 | *** part/#asterisk jfindley (~jfindley@104-181-196-33.lightspeed.hstntx.sbcglobal.net) |
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| 19:02.50 | jhester | shudon: A healthy curiosity is a great ally when getting started with Asterisk. |
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| 19:04.34 | jhester | Without actually cracking it open and looking we can't really know how isymphony does the deed. But [TK]D-Fender makes an excellent estimation in my ⦠estimation. |
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| 19:33.52 | shudon | thanks guys |
| 19:34.03 | shudon | i was able to do it from asterisk console with chanspy :) |
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| 19:39.12 | *** join/#asterisk klow (~klong@c-73-53-31-109.hsd1.wa.comcast.net) |
| 19:39.44 | klow | Does anyone know of a SIP trunk provider in Poland , I just need a single DID there to tie into my FreePBX system .. |
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| 19:41.11 | shudon | so the only thing i couldn't do |
| 19:41.18 | shudon | is pass the 'q' option to ChanSpy from asterisk console |
| 19:41.57 | shudon | i'm extreme asterisk newb. i did it like this: channel originate SIP/myext application ChanSpy SIP/otherext q |
| 19:42.07 | shudon | is that the correct way to pass the option 'q'? |
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| 19:42.46 | shudon | the command worked, but the 'q' option seemed to be ignored. i still got a beep and a voice telling me the "otherext" |
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| 19:52.21 | rmudgett | shudon: What does "core show application chanspy" say it wants for the parameter string? |
| 19:52.38 | rmudgett | What didn't you provide with the parameter string? |
| 19:53.03 | rmudgett | You were missing the comma between SIP/otherext and q |
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| 20:02.18 | shudon | rmudgett: oh my gosh i dont' have a 'q' |
| 20:02.21 | shudon | thanks rmudgett :) |
| 20:03.58 | shudon | oh wait |
| 20:04.05 | shudon | rmudgett: it does mention 'q' |
| 20:04.08 | shudon | <PROTECTED> |
| 20:04.08 | shudon | <PROTECTED> |
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| 22:16.42 | lunaphyte | hi. i'm looking for some guidance on equipment for voip. is that too off topic for here, since it's not an asterisk question? |
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| 22:26.35 | WIMPy | lunaphyte: Only a little. Give it a try. |
| 22:26.44 | lunaphyte | great, thanks. |
| 22:29.44 | lunaphyte | i have a device which is in essence an ip phone. it's a module in a piece of teleconference equipment, and it uses sip. so it wants to connect to a sip registrar/gateway/whatever [apologies if my terms are horrible] |
| 22:30.13 | lunaphyte | i also have a crappy proprietary phone system, which doesn't support sip. |
| 22:30.46 | lunaphyte | so what i'd like to do is provision a traditional analog line on the phone system, and put a device in between the two |
| 22:31.54 | WIMPy | There's a Linksys ATA that can do the reverse thing. IIRC SPA1001. I'm sure someone will correct that name. |
| 22:32.35 | lunaphyte | i found the linksys/cisco pap2t - is that it? |
| 22:32.49 | lunaphyte | googles SPA1001 |
| 22:33.01 | WIMPy | No, that's just a plain ATA I think. |
| 22:33.04 | lunaphyte | oh, ok |
| 22:33.06 | WIMPy | I'm totally not in to the analog stuff, sorry. |
| 22:33.40 | lunaphyte | i wouldn't be either - only out of necessity :) |
| 22:35.27 | lunaphyte | if i've got the terminology right, i believe i need a device with an fxo port? |
| 22:35.59 | WIMPy | Sounds correct. |
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| 22:51.21 | lunaphyte | the cisco spa3102 seems to indicate it has an fxo port |
| 22:52.01 | hagbard | There appear to be numerous customized linux distros + web front ends available for asterisk. Any suggestions which to try to replace an aging fonality pbxtra machine? |
| 22:52.06 | [TK]D-Fender | 1 fxs + 1 fxo |
| 22:52.15 | lunaphyte | [TK]D-Fender: yeah |
| 22:52.30 | lunaphyte | the fxs port would be unused, which i guess would be ok, right? |
| 22:52.38 | [TK]D-Fender | hagbard, the FreePBX official distro |
| 22:52.48 | [TK]D-Fender | lunaphyte, Depends on your need. |
| 22:53.02 | [TK]D-Fender | lunaphyte, It can also be bridged in the case of a power failure |
| 22:57.46 | [TK]D-Fender | BRB |
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