00:06.48 | [TK]D-Fender | phix, "can barely hear" means one (or both) ends of your call doesn't have sane gain levels |
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01:21.01 | LemensTS | How does a multi-line polycom work when you put a call on hold? Does it use a line key for each call on held? |
01:22.07 | [TK]D-Fender | No. A line key does not gfet taken up at THAT point in the process. |
01:22.18 | [TK]D-Fender | If a line key is used, it's from the moment a call is presented to the phone at all. |
01:22.25 | [TK]D-Fender | Long before you answer, or put them on hold |
01:22.40 | [TK]D-Fender | And as to how your line keys work... it depends how you configured them |
01:23.17 | [TK]D-Fender | I had an old IP301 that had 2 line-keys... but I only used ONE of them ... and that single line-key supported up to 5 simultaneous calls. |
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07:13.01 | Guest8893 | Looking in extensions.conf I see [public]\n include => from-voip |
07:13.17 | Guest8893 | what exactly does this mean? Is the context from-voip included in the public context? |
07:13.23 | Guest8893 | Cause I did not see it defined in sip.conf |
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07:28.44 | r00f | Guest8893, yes it is included |
07:29.31 | r00f | if it is not present, on dialplan reload you will see a warning that public tries to include non-existing context |
07:30.16 | Guest8893 | r00f: you mean present in sip.conf? |
07:30.29 | Guest8893 | Cause I receive no warning with dialplan reload |
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07:32.06 | Guest8893 | Or can context's be defined elsewhere? |
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07:41.26 | Guest8893 | Also, can the cdr_mysql function write to two databases? Did not find this in the docs |
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07:44.49 | kleszcz | Guest8893: i dont thins so, you can use replication |
07:45.11 | Guest8893 | An option I wished to avoid, I'll solve it another way. |
07:45.52 | kleszcz | hmm or maybe user cdr_odbc ? |
07:45.55 | kleszcz | *use |
07:46.34 | kleszcz | i never try, but maybe its possible |
07:48.07 | kleszcz | you know what i mean? |
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08:01.42 | Guest8893 | kleszcz: Nah am good, got my way around it. |
08:05.19 | kleszcz | so maybe you can tell me waht way you prefer to aroudn it ;) |
08:05.22 | kleszcz | hmm? |
08:16.54 | elitas | hello, how to register a number i sip.conf ? |
08:17.29 | elitas | register => 'phone number'@server ? |
08:18.13 | kleszcz | register => user:password@server |
08:18.29 | kleszcz | user/number |
08:19.06 | elitas | register => user:password@server:5060 port required ? |
08:19.30 | kleszcz | if the port number is dfferent |
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09:41.27 | elitas | i got some errors with network when trying to register my acc http://pastebin.com/2cX9wz3n |
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10:24.00 | haribotec | Hello all, Linux 6.6 & Asterisk 11 newbie here looking for a little help / guidance. Can anybody give me some hints on how to solve the dependencies for an asterisk 11 install on centos 6.6 (libsqlite3x-devel and gmime22-devel). This shows when I run /usr/src/./install_prereq test |
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10:35.24 | haribotec | can anybody tell me if this room is working ok, I have only seen my own question along side people entering & leaving messages |
10:36.16 | zamba | channel is ok |
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10:36.37 | haribotec | thanks zamba |
10:36.57 | Chainsaw | haribotec: Yes, room is working fine. I can compile Asterisk from source, but I've not done so on RedHat/CentOS. |
10:37.14 | Chainsaw | haribotec: Is there a #centos that you could ask on, now that you know what software packages you need? |
10:37.43 | haribotec | great idea thanks Chainsaw |
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11:14.37 | Freeaqingme | Hi folks. I was wondering how you guys monitor your asterisk installations |
11:14.47 | Freeaqingme | Any ideas on how to verify all is sstill in working order? |
11:15.13 | vjosh | can anyone help with ARI integrations? |
11:15.40 | Freeaqingme | vjosh, just ask hwatever questions you may have. If someone knows, you might just get the answers you're looking ofr |
11:15.41 | Freeaqingme | *for |
11:15.54 | vjosh | ok thanks |
11:16.45 | vjosh | So I have build a stasis app in ruby which interacts with asterisk server. But my problem is I bridge both the channel when the call is picked up |
11:17.12 | vjosh | but if its a wrong number or busy then it doesnt work |
11:18.20 | vjosh | so how do I pass the information to my incoming channel if other one is busy/wrong no etc |
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11:39.46 | Moggie | Hi, just to check that I understand this correctly. If Asterisk fetches SIP users from a database, it cannot support subscriptions, voicemail waiting notifications, or NAT keep alive to maintain registrations? |
11:40.38 | Moggie | I'm been reading various docs and may have confused myself slightly. If someone could please clarify, that would be great, thanks. |
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12:28.20 | skyroveRR | Has anyone used chan_celliax module before? |
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12:45.39 | skyroveRR | Is there a way to interface cell phones with asterisk using only GSM? No SIP or mobile internet... |
12:50.18 | [TK]D-Fender | Clarify "interface cellphones" |
12:50.36 | skyroveRR | Sure. |
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12:53.23 | skyroveRR | I'm trying to do the following: <SIP> -- <HOME_CELL_PHONE> -- <CELL_PHONE>; I have a spare cell phone that I don't use much, and it is AT compatible, I'd like to make and receive phone calls from the internet through that spare cell phone onto my actual phone, but without using VoIP. I came across chan_celliax, but I was unable to succeed at doing what I just said, is there some other way of doing it? |
12:54.09 | skyroveRR | I basically want to proxy all the calls through the home cell phone. |
12:54.40 | [TK]D-Fender | I don't see any service provider in there anywhere.. or Asterisk |
12:55.15 | [TK]D-Fender | Draw a proper and complete chain for your next example. |
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12:56.19 | skyroveRR | <internet> <----> <home_phone_with_internet enabled> <----> <current_phone_without_internet>.. |
12:56.53 | [TK]D-Fender | Still vague about what is used for what, and where * sits in there... |
12:57.09 | [TK]D-Fender | And how they actually talk between each leg |
12:57.24 | skyroveRR | http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Network |
12:58.29 | skyroveRR | The home phone is interfaced using a USB cable to the computer, so it's a basic modem, and the computer runs asterisk... I want asterisk to handle all the calls signalled by that phone. |
12:58.33 | [TK]D-Fender | I asked for a basic description from YOU. |
12:58.40 | [TK]D-Fender | Not some guide that may describe something else |
12:59.34 | [TK]D-Fender | I know of no method for using that over USB. |
12:59.53 | [TK]D-Fender | Three is chan_mobile for talking to a cellphone over BLUETOOTH |
12:59.57 | [TK]D-Fender | There |
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13:00.58 | skyroveRR | [TK]D-Fender: can it handle phone calls? |
13:02.21 | skyroveRR | OR only sms? |
13:02.52 | [TK]D-Fender | both |
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13:06.36 | skyroveRR | [TK]D-Fender: is chan_celliax supported anymore? |
13:06.45 | [TK]D-Fender | No idea. |
13:07.30 | [TK]D-Fender | What does their site say? |
13:07.43 | [TK]D-Fender | Because that isn't part of the actual * package |
13:07.55 | skyroveRR | Their site is inactive since 2011.. |
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13:08.15 | skyroveRR | But their github page looks current. |
13:08.54 | [TK]D-Fender | Their webpage times out |
13:09.00 | [TK]D-Fender | That is old crap.... |
13:09.07 | [TK]D-Fender | chan_mobile <- |
13:09.35 | skyroveRR | Alright.. |
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14:14.58 | elitas | after sip reload i got message "chan_sip.c:32005 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use" |
14:15.06 | elitas | how could i fix it ? |
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14:17.01 | Chainsaw | elitas: That normally means you've got multiple Asterisk instances running. |
14:17.08 | Chainsaw | elitas: Or you are competing with some other SIP daemon on the host, of course. |
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14:25.55 | rexwin_ | [2015-04-20 10:24:53] NOTICE[25876]: res_pjsip/pjsip_distributor.c:250 log_unidentified_request: Request from '"rex" <sip:512@54.149.125.66>' failed for '163.47.13.179:42985' (callid: ZTA2MmE0NjAyMmUwZGI0ZTk3MDRkMjI4OTdlNDAwYTE) - No matching endpoint found |
14:26.16 | rexwin_ | I am getting when tring to register my softphone to freepbx server. |
14:26.45 | eppigy | Hello |
14:26.48 | eppigy | I am Dave |
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14:37.49 | [TK]D-Fender | eppigy: You... are dave |
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14:51.36 | ChkDigit | I'm sorry Dave, I'm afraid I can't do that. |
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15:05.59 | maddawg2 | lol |
15:10.18 | trurl | ChkDigit: open the pod bay doors!!1 |
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15:37.19 | nickaugust | does anyone know how I disable the damn call waiting beep on a polycom 550? the docs show *71 for disabling on a per-call basis but I can't see how to make it permanent |
15:37.58 | nickaugust | I think most people use rollover lines I'm not sure why this is the default. and there's no setting I see in the web ui |
15:38.21 | PHunter | Polycom is weird compared to others. :/ |
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15:39.11 | PHunter | At least from what I have seen. |
15:43.25 | PHunter | cdr_mysql -> cdr_adaptive_odbc.. where do you specify the remote addr? |
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15:51.26 | [TK]D-Fender | nickaugust: Where do you see this? |
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15:55.21 | qakhan | i have exts on polycome lines like line2 = 1002, line3 = 1003, line4 = 1004 etc.... when i need to dial an ext i just press line number. is there any way when any ext already on call that line number on my phone start blinking. |
15:55.54 | qakhan | like if ext 1003 is on call then line 3 start blinking on my phone. |
15:56.24 | [TK]D-Fender | You said you "have them" on your phone. have them HOW? |
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15:58.07 | qakhan | i want to see on my phone if an ext alreay on call before i call them |
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15:59.15 | [TK]D-Fender | You said you already "have them" on your phone. |
15:59.18 | [TK]D-Fender | Have them HOW? |
15:59.29 | anonymouz666 | watches [TK]D-Fender |
15:59.47 | [TK]D-Fender | hands anonymouz666 a spoon |
15:59.53 | qakhan | i added then as speed dial on polycom |
16:00.05 | qakhan | them* |
16:00.39 | [TK]D-Fender | enable Buddy Watch on them, and have hints set up in the dialplan |
16:01.11 | qakhan | is buddy watch in asterisk or polycom phone? |
16:01.18 | [TK]D-Fender | on the phone |
16:04.05 | qakhan | thanks [TK]D-Fender. i will check it |
16:07.51 | anonymouz666 | netflix is studying to use https to secure the media streams |
16:08.02 | anonymouz666 | what about admin's that caches the content? |
16:08.03 | anonymouz666 | he he |
16:09.07 | anonymouz666 | nice to know that freebsd is still rock solid and being heavily used by many sites |
16:09.17 | anonymouz666 | my last contact with fbsd was in 2002/2003. |
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16:45.32 | skyroveRR | I'm trying to setup a minimal asterisk configuration, I can't get to the asterisk console for some reason, am I forgetting a console module? Here's my modules.conf file http://tracefl.in/pastes/modules.conf.txt |
16:45.46 | skyroveRR | I'm using asterisk -c command. |
16:47.06 | [TK]D-Fender | skyroveRR: You have disabled autoload and you aren't loading enough to get anywhere |
16:47.48 | skyroveRR | Ok. I uncommented that line. Even then, I can't get a console. |
16:48.16 | [TK]D-Fender | You also haven't shown us an ATTEMPT |
16:48.30 | skyroveRR | Let me paste the output. |
16:50.41 | skyroveRR | [TK]D-Fender: http://tracefl.in/pastes/asterisk.error.1.txt |
16:51.07 | [TK]D-Fender | Error loading module 'chan_iax.so': /usr/lib/asterisk/modules/chan_iax.so: |
16:51.11 | [TK]D-Fender | You are missing modules |
16:51.16 | [TK]D-Fender | You have a messed up install |
16:51.27 | [TK]D-Fender | Unable to load config skinny.conf, Skinny disabled. |
16:51.32 | [TK]D-Fender | And missing configs |
16:51.35 | [TK]D-Fender | This is just a mess |
16:51.42 | skyroveRR | :| |
16:51.48 | skyroveRR | I have iax2.so... |
16:52.40 | [TK]D-Fender | Not in the right state in the place it is looking |
16:53.44 | tzafrir | skyroveRR, for starters, you don't need all of those "noload" lines |
16:53.50 | skyroveRR | I'd just like to load the celliax module by getting the prompt. |
16:54.02 | tzafrir | (if you don't use autoload = yes) |
16:54.52 | tzafrir | What you miss are enough -v-s |
16:55.08 | tzafrir | asterisk -vc , maybe? |
16:55.55 | [TK]D-Fender | "I just" doesn't help. * is reporting an inability to load modules. Common regular ones. Your install is broken |
16:57.06 | skyroveRR | [TK]D-Fender: which are the common ones? |
16:57.25 | [TK]D-Fender | Those channel driver modules it's failing to load |
16:57.55 | [TK]D-Fender | And you're missing a LOT of configs |
16:58.15 | [TK]D-Fender | This is in "major screwup" territory |
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17:02.54 | skyroveRR | I took a complete backup of all the config files, let me restore them and let you guys know :) |
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17:06.45 | babak | Hi, how to update asterisk 13.0 to13.1 ? |
17:09.50 | [TK]D-Fender | compile. Install over. |
17:10.27 | skyroveRR | [TK]D-Fender: ok. Restored the files. Got the command line :) |
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17:42.08 | Kobaz | what's a good way to get the device name from a channel |
17:42.18 | Kobaz | you have say SIP/foo-bar-baz-000203ab |
17:43.01 | Kobaz | in my own platform i have generated set vars on all the devices for convenience, so my sip realtime db generates a SetVar=ExtenDevice=SIP/foo-bar-baz |
17:43.12 | Kobaz | but i'm working on a pure-asterisk system that doesn't have my facnyness |
17:43.17 | Kobaz | *fancyness |
17:45.44 | [TK]D-Fender | You can do that with the regular configs just the same... |
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17:49.53 | Kobaz | yeah |
17:49.57 | Kobaz | i know you can do that in regular configs |
17:50.08 | Kobaz | but is there a way to not have to do a setvar on 5000 peers |
17:50.21 | Kobaz | is there a channel function i don't know about, that'll get the base device name? |
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18:13.57 | someone7 | SEND onyx47 I am around when you are, i'll look out for you in the #asterisk channel :-) |
18:17.04 | Kobaz | [TK]D-Fender: i guess i can add a function for it |
18:17.29 | [TK]D-Fender | "core show function SIPPEER" |
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18:54.46 | Kobaz | [TK]D-Fender: oh |
18:54.58 | Kobaz | [TK]D-Fender: you need to know the peername, but that put me on the right track |
18:55.01 | Kobaz | CHANNEL(peername) |
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18:56.08 | [TK]D-Fender | yeah, that's where I was heading... |
18:58.49 | Kobaz | thanks |
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19:19.35 | ChkDigit | Are there any tricks to having presence hints move through an IAX2 trunk? |
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20:14.26 | Kobaz | http://www.watchersweb.com/submissions/BRIYD1428283830/febPU1428283830.jpg |
20:15.21 | [TK]D-Fender | funny they put that bar in the way that makes it especially hard for fat people to take the escalator as it suggests ;) |
20:17.11 | MaliutaLap | Maybe it's to prevent ram-raiding |
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20:25.38 | Kobaz | hah |
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22:18.14 | maddawg2 | i thought you meant a bar that served booze :-( |
22:18.20 | maddawg2 | i was like "This I have to see" |
22:18.49 | maddawg2 | but those are there to prevent people from trying to take baby strollers up them |
22:18.52 | maddawg2 | or wheelchairs |
22:19.00 | maddawg2 | people actually try that shit and fall down all the time |
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22:40.19 | LemensTS | TKD-Fender: Ok, sorry just now replying on the line keys. Would I be able to use the polycom line keys as held lines with programming then? I never use hold, I am used to parking lots |
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23:00.24 | [TK]D-Fender | LemensTS, No. They are not RESERVED to just indicate held calls |
23:00.34 | [TK]D-Fender | LemensTS, They arepresent CALL APPEARANCES |
23:00.59 | [TK]D-Fender | that means as status indicators of being forwarded, NOT inuse, on-call, etc |
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23:05.59 | [TK]D-Fender | heads out for a bit |
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23:14.30 | PHunter | NOTICE[29063][C-000014dc]: res_rtp_asterisk.c:4367 ast_rtp_read: Unknown RTP codec 72 received from 'x.x.x.x:x' |
23:14.50 | PHunter | I know its low severity, but how would i go about solving that? |
23:15.08 | PHunter | only 1 or 2 endpoints experience it. |
23:15.14 | PHunter | Behind NAT |
23:16.03 | file | what is there to solve? |
23:16.27 | PHunter | Apparently the call is not making it to the endpoint due to lack of RTP connectivity after spamming that for 30 seconds. |
23:17.27 | PHunter | Ill have to poke at their devices a bit tomorrow and see what their setup is like. |
23:17.51 | PHunter | I have 85 endpoints and only 1 or 2 at their location is experiencing that. |
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23:18.33 | PHunter | Maybe someone changed the media types..? |
23:18.37 | PHunter | codec.. |
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23:27.29 | PHunter | o/ |
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