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| 04:36.00 | KNERD | Something has to be done about Asterisk and virtualization |
| 04:36.47 | KNERD | " sip set debug on No such command 'sip set debug on' (type 'core show help sip' for other possible commands) " COME ON NOW....F*****G REALLY!! |
| 04:47.43 | jpsharp | That's not a virtualization issue. That sounds like you just don't have chan_sip loaded or built. |
| 04:48.14 | KNERD | wanna bet? |
| 04:48.36 | KNERD | https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 04:49.00 | KNERD | i have been trying to make another successfull build for 3 months now!!!! |
| 04:49.19 | KNERD | this crap is beyone old |
| 04:49.31 | KNERD | i am about to flee to FreeSwitch |
| 04:49.48 | KNERD | but before I do I am going to an old version |
| 04:49.56 | KNERD | and see what happens |
| 04:50.59 | jpsharp | I see an "illegal instruction" bug. I don't see a "chan sip didn't load or build" bug. |
| 04:52.02 | KNERD | it is related |
| 04:52.12 | KNERD | they stuck in that NATIVE BUILD crap |
| 04:52.18 | KNERD | and now it si causing havoc |
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| 04:53.24 | jpsharp | Dunno. Every time I see someone say something about chan_sip not showing up or not loading its cause they didn't install libssl, so res_crypto didn't build, and chan_sip didn't build because of that. |
| 04:53.54 | KNERD | I used their suggestion of CFLAGS="-march=k8 -msse4a" and now i am getting unpredictable behavior |
| 04:54.14 | KNERD | dude..it was working seconds before then all of a suggen no such command...derp |
| 04:55.09 | jpsharp | Okay. Just offering a second opinion :) |
| 04:55.30 | KNERD | yeah, but after 3 months of this crap.. |
| 04:57.44 | KNERD | i had a nice working system until I moved to a new server and got the newer version...now I will be stuck at 1.8.5 or so until either I learn FreeSwitch, or Digium fixed this crap |
| 04:59.12 | KNERD | i thought it was fixed until I saw all my phones getting a SIP 401 Unathorized error |
| 04:59.50 | jpsharp | I personally blame the virtualisation providers. Requiring hacked up kernels and doing a crap job of providing reasonable virtual CPUs. |
| 05:02.13 | KNERD | I was using on OpenVZ, but after this issue I though it may have been OpenVZ because I switched to a new server with an updated version of OpenVZ |
| 05:02.21 | KNERD | so I switched to Xen |
| 05:02.36 | ChannelZ | Maybe I'm thick but this just seems like a compiler options change |
| 05:02.36 | KNERD | wellI think it was partially OpenVZ problem |
| 05:02.52 | KNERD | as not even DAHDI would run on OpenVZ |
| 05:03.25 | KNERD | well the problem with that is their make script is not seeing the correct CPU |
| 05:04.06 | jpsharp | If you've got full access to the server, might I suggest VMWare ESXi? I'm running Asterisk on it without a problem. |
| 05:04.21 | KNERD | no I don't |
| 05:04.42 | KNERD | I would run my own box if I had the bandwidth somewhere |
| 05:04.49 | jpsharp | Ah. Gotcha. |
| 05:05.33 | KNERD | looking at 1.8.5 the BUILD NATIVE is missing |
| 05:06.21 | KNERD | yes I know and I aprecciate it |
| 05:07.04 | KNERD | some of us just dont have time to play with these things and the hundreds to CPU flags to see which one may work |
| 05:08.30 | jpsharp | You can go bottom of the barrel and just use "-march=i386" |
| 05:08.42 | ChannelZ | Can you look at the results of an old version configure? |
| 05:09.21 | KNERD | not sure...I guess I can look. |
| 05:09.50 | KNERD | i396?? uggg |
| 05:09.57 | KNERD | err i386 |
| 05:10.07 | KNERD | well the no flag did not work either |
| 05:12.07 | KNERD | the config log? |
| 05:17.00 | KNERD | i got it here... |
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| 07:08.21 | bulkorok | hi |
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| 07:26.11 | _zoom_ | server internal error, cause & reasons do they have standards values? |
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| 08:31.34 | schmidts | good morning |
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| 08:34.24 | ChannelZ | blah! |
| 08:35.15 | creativx | np |
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| 08:37.50 | schmidts | maybe thats a stupid question but how can i set the verbose level in asterisk 11 so it might stay when i reconnect to it? |
| 08:38.05 | schmidts | i have the problem when i enter asterisk with rasterisk i allways have to reset the verbose level |
| 08:38.41 | ghost75 | asterisk.conf options |
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| 08:42.02 | schmidts | ghost75 thats where i set the default level but for example when i set verbose to level 10 i didnt want to reset it the next time |
| 08:49.49 | bulkorok | schmidts: asterisk -rvvv gives you verbose 3 |
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| 08:53.24 | schmidts | bulkorok thanks but why is this behavior changed since 10? |
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| 08:56.20 | ChannelZ | has it? |
| 08:56.38 | bulkorok | I don't think so too... |
| 08:56.41 | ChannelZ | You can also set some defaults in cli.conf |
| 09:01.00 | ChannelZ | (which is possibly something you had before and don't now..) |
| 09:03.40 | Gugge | if "verbose = something" is set in asterisk.conf, asterisk -r picks that up in 11, in 1.8 it it only picks it up if the setting in asterisk.conf is higher |
| 09:03.43 | Gugge | (on my machines) |
| 09:03.46 | Gugge | so something changed :) |
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| 09:05.37 | Gugge | schmidts: but why do you care about the verbose level (when you dont use the cli)? |
| 09:07.07 | schmidts | gugge why do you think i dont use it? |
| 09:07.23 | Gugge | use what? |
| 09:07.31 | schmidts | cli :) |
| 09:07.41 | Gugge | i think you use it :) |
| 09:07.45 | schmidts | ok |
| 09:07.51 | schmidts | monday morning, i need more coffee |
| 09:08.01 | Gugge | jusat remove verbose=x from asterisk.conf if you want the custom set verbose level to stick :) |
| 09:08.04 | Gugge | just |
| 09:08.19 | Gugge | Or if you want a specifik verbose level set each time you start the cli, set the option |
| 09:08.29 | Gugge | And if you want to log a specifik level, set that in logger.conf :) |
| 09:09.50 | schmidts | ok once again, i should read CHANGES file :) |
| 09:10.08 | Gugge | it was the logger stuff you wanted? :) |
| 09:11.04 | schmidts | its exactly described what have changed in the CHANGES file |
| 09:11.14 | Gugge | yep :) |
| 09:11.32 | ghost75 | somebody use monitoring on asterisk? |
| 09:11.45 | schmidts | i only wondered cause normally if i set the verbose level with core set verbose it stays that way, but as it says in the CHANGES file it only is changed for the current terminal session |
| 09:12.35 | schmidts | s/terminal session/remote consol/ |
| 09:15.34 | _zoom_ | on exten.conf can I associate certain exten with specifi IP address? |
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| 09:15.57 | ghost75 | what? |
| 09:16.28 | schmidts | _zoom_ not directly in extensions.conf you have to made this in sip.conf for example, there you can create peers for every IP you want, and then set a different context for each peer |
| 09:17.13 | _zoom_ | am working with carriers they drop calls directly @ipaddress |
| 09:19.44 | ghost75 | schmidts: do you remember how that software was called for seeing voip quality? |
| 09:20.29 | schmidts | ghost75 i have to look in my mails, you mean this open source project around 2 years ago or something? |
| 09:20.53 | ghost75 | was it voip monitor? |
| 09:21.15 | schmidts | yes i think so |
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| 09:39.26 | zamba | will 'reload' break any channels already up? |
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| 09:51.09 | wdoekes | zamba: no |
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| 09:58.00 | zamba | good |
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| 10:49.45 | bombev | Hi guys |
| 10:49.53 | bombev | I bought two channel of G729 |
| 10:50.08 | bombev | How can I install it on my system |
| 10:52.21 | Vince-0 | let me google that for you |
| 10:52.35 | Vince-0 | there are lots of instructions for that - even on digiums site |
| 10:55.16 | bombev | I found it :) already |
| 10:56.40 | bombev | Vince-0 how to check |
| 10:56.49 | bombev | whether the machine is 32bit or 64? |
| 10:59.55 | kaldemar | bombev: uname -m |
| 11:01.17 | bombev | well I got this |
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| 11:01.19 | bombev | No such command 'uname -m' (type 'core show help uname -m' for other possible commands) |
| 11:01.30 | kaldemar | bombev: not in asterisk, in shell. |
| 11:01.37 | bombev | oh my vad |
| 11:01.38 | bombev | bad |
| 11:02.29 | bombev | hm I got this x86_64 |
| 11:02.35 | Maliuta | how can you not know if your using a 32 or 64 bit system? |
| 11:02.47 | Maliuta | 64 bit |
| 11:03.31 | bombev | so 32 bit or 64? |
| 11:08.29 | bombev | Maliuta thanks |
| 11:08.56 | bombev | I dont know because somebody else have installed the system |
| 11:09.15 | bombev | I just wanna make sure :) |
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| 11:16.44 | Vince-0 | uname -a |
| 11:17.57 | Maliuta | bombev: what is the OS/distribution ... most of them make it perfectly clear what the architecture is in one way or another. |
| 11:20.42 | bombev | Well I think /sbin/init: ELF 64-bit LSB executable, AMD x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked (uses shared libs), for GNU/Linux 2.6.9, stripped |
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| 11:28.35 | hugo | ERROR[-1] tcptls.c: Unable to connect SIP socket to 213.125.83.26:61581: Connection refused <--- how can I convince * to stop trying ? (other than removing the db) |
| 11:28.53 | hugo | it just seems to retry.. forever |
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| 11:53.38 | ghost75 | is capturing voip packets cpu intensive? i have only atom cpu |
| 11:59.20 | schmidts | ghost75 only capturing in a pcap file shouldnt be a big problem, cause it runs mostly without cpu, but many IO and allways the cpu of your nic |
| 11:59.56 | ghost75 | nic has no cpu |
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| 12:25.31 | gavimobile | sorry for going off topic over here, I have a linksys spa2102 which has the ability to use Block ANC Act Code: *77. I want to use *77 as a dialplan rule in my pbx for something else. I do not wish to disable only this service in my pap device, I would rather disable ALL service feature codes. is this possible? or should I just change my dialplan rule from *77 to somethine which my spa2102 isn't using? |
| 12:31.10 | ghost75 | is not possible to run 2 asterisk server behind nat conncting to outside sip accounts like sipgate or? |
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| 12:35.44 | jmls | morning all |
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| 12:36.27 | jmls | I thought that there was a way to execute ami / cli commands from the dialplan using asterisk-11. Am i mistaken ? |
| 12:37.06 | jmls | i know that I can system asterisk -rx "somecommand", but thought that there was a diaplan option as well |
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| 12:39.34 | ghost75 | system cmd in dialplan? |
| 12:40.05 | ghost75 | why execute ami cmd over dialplan? |
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| 12:42.38 | jmls | because I want to do (for example) a "sip reload" and provide a mechanism to allow you to do this from a "command" phone without having to logon to a web service etc |
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| 12:43.48 | ghost75 | start script from dialplan |
| 12:44.44 | GeoGeek | Does anyone here have experience with Hylafax integration? I have Hylafax installed and working with Asterisk, but am having iax2 registration issues. |
| 12:46.11 | GeoGeek | Even just basic iax extension experience might be able to help me... |
| 12:49.19 | *** join/#asterisk AviMarcus (~avi@bzq-79-183-240-31.red.bezeqint.net) |
| 12:49.20 | AviMarcus | Hey. I've got an SPA-2102 that returns "busy" when a call comes in while one is still on the line. But I have "call waiting" turned on. How do I debug this? I got the debugging logs but they just show sending a 100 trying then 486 0.2 seconds later. |
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| 13:08.10 | schmidts | AviMarcus are you sure the analog phone connected to the 2102 is capable of call waiting? |
| 13:08.41 | AviMarcus | schmidts, isn't that a function in the spa? |
| 13:08.59 | AviMarcus | it just needs a flash button or something to actually send the signal to pick up |
| 13:09.33 | schmidts | AviMarcus i am not sure about this to be honest. normally you should hear a ringer signal for this |
| 13:09.53 | schmidts | Maybe you can set a syslog server in the system tab of the spa and check if you will see somthing there |
| 13:11.31 | AviMarcus | schmidts, I did get this, but it doesn't seem very helpful.. maybe I don't have enough debugging on: http://pastebin.com/57mxj6Bj |
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| 13:16.25 | schmidts | yes you need debug level 3 (i think its the highest) and also on line 1 you can enable sip debuging that also helps |
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| 13:16.55 | AviMarcus | I'm pretty sure I did that but I thought it was supposed to send the SIP trace with it |
| 13:17.11 | AviMarcus | but sip trace won't help, I need to know why the box is doing what it's doing.... mmmm. |
| 13:17.25 | AviMarcus | maybe I should toggle call waiting on it, that might fix it. |
| 13:17.30 | AviMarcus | thx for looking |
| 13:22.16 | schmidts | AviMarcus in the system tabs you have a setting about the debug level and system debug level, you have set both to the highest level |
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| 13:33.42 | bombev | pf please need help about g729 codec |
| 13:34.08 | bombev | I did everything |
| 13:34.47 | bombev | When I try to load the codec i got this |
| 13:34.53 | bombev | [root@call /]# asterisk -rx "module load codec_g729a.so" |
| 13:34.53 | bombev | Unable to load module codec_g729a.so |
| 13:34.53 | bombev | Command 'module load codec_g729a.so' failed. |
| 13:35.23 | kaldemar | does the license not include support? |
| 13:35.43 | bombev | well I included the license |
| 13:36.18 | bombev | I think |
| 13:37.01 | bombev | I did everthing http://downloads.digium.com/pub/telephony/codec_g729/README |
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| 13:43.15 | bombev | kaldemar I have the license var/lib/asterisk/licenses |
| 13:43.35 | bombev | file with name g729-*********.lic |
| 13:44.46 | kaldemar | i really meant that digium surely supports the products they sell, you can ask them. |
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| 13:51.26 | [TK]D-Fender | bombev, Don't just -rx it. Do it from full CLI in case it's sending messages you won't see from there |
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| 14:08.31 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:15.17 | Katty | morning |
| 14:16.15 | schmidts | welcome katty |
| 14:16.32 | schmidts | katty did you asterisk survive the new years eve? |
| 14:16.48 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 14:18.29 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 14:25.25 | Katty | nope. |
| 14:31.10 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:35.12 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 14:36.11 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.206) |
| 14:37.37 | qakhan | hi all, here i have a question |
| 14:38.45 | qakhan | i want to setup exts, which can communicate one to one and one to many exts |
| 14:43.18 | Penguin | No, you want to set up phones, which can dial many extens. |
| 14:45.38 | qakhan | i want to setup exts which can dial many exts same time and one to one dial |
| 14:46.11 | leifmadsen | no, you want a device that can dial an extension that can dial many devices |
| 14:46.35 | leifmadsen | extensions are just what triggers the dialplan logic -- extensions are no devices |
| 14:47.04 | leifmadsen | qakhan: all of this is easily possible -- if you asteriskdocs.org and work through some of the dialplan it should become pretty obvious how to do that |
| 14:47.22 | qakhan | hmmmm |
| 14:48.09 | qakhan | @leifmadsen can you tell me about Push to Talk in asterisk |
| 14:48.17 | leifmadsen | I can not |
| 14:49.00 | leifmadsen | I'm not even sure what that question really means... PTT is really a device level thing |
| 14:49.05 | leifmadsen | gotta reboot |
| 14:49.48 | qakhan | ok is there any setting in dialplan if i dial any ext it answer call on phone automatically |
| 14:58.32 | wdoekes | qakhan: depends on the phone. you'll need to add a SIP header |
| 14:58.54 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:58.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:59.06 | wdoekes | SIPAddHeader(Alert-Info: Auto Answer) ; Polycom |
| 14:59.09 | Rac-on | qakhan: SIPAddHeader(Call-Info: answer-after=0) seems to work for most phones |
| 14:59.15 | wdoekes | SIPAddHeader(Call-Info: <http://>\;answer-after=0) ; Grandstream/Linksys |
| 15:00.02 | *** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net) |
| 15:23.31 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
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| 15:34.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:36.39 | jacekowski | my asterisk seems to have problem reregistering after reload |
| 15:36.43 | jacekowski | as in, it's not doing it |
| 15:36.49 | *** join/#asterisk moos3 (~moos3@72.95.108.32) |
| 15:37.14 | jacekowski | [Jan 7 15:36:04] NOTICE[21638] chan_sip.c: Sending fake auth rejection for device "+447XXXXX" <sip:+447XXXX@sbc.freeconet.pl>;tag=as466088a4 |
| 15:37.22 | jacekowski | after waiting for a while |
| 15:37.25 | jacekowski | it will work fine |
| 15:37.47 | moos3 | anyone know why Asterisk SVN-branch-1.8-r378217 ignores rtp.conf rtpstart and rtpend ? |
| 15:38.01 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-poggxnrlyrcfqxby) |
| 15:38.23 | moos3 | I have it set to rtpend=20000 but its trying to use 63501 on client |
| 15:39.29 | kaldemar | moos3: the ports are what adterisk uses, peers may still use what ever they want. |
| 15:40.26 | leifmadsen | +1 |
| 15:40.38 | moos3 | Registered SIP '9043' at 10.121.0.17:63592 |
| 15:40.45 | leifmadsen | nothign wrong with that |
| 15:40.46 | moos3 | its causing me to have one way audio |
| 15:40.59 | leifmadsen | the port range is what asterisk advertises as being valid for it to listen on |
| 15:41.09 | leifmadsen | the client can listen on whatever port they want |
| 15:41.28 | leifmadsen | one way in terms of other end can't hear you? then their firewall is the problem |
| 15:41.37 | moos3 | they can hear me |
| 15:41.40 | moos3 | but I can't hear them |
| 15:41.50 | moos3 | doesn't matter if i'm vpn'd in to the network or not |
| 15:42.29 | leifmadsen | if you're remote to asterisk as well (not same lan) check your firewall etc. Also check to make sure SIP connection is setting up correctly and that you're getting RTP from them |
| 15:42.34 | kaldemar | and that is a port for SIP, not RTP |
| 15:42.47 | leifmadsen | RTP is negotiated at call setup time |
| 15:43.03 | moos3 | k, because we are comming into 5060 |
| 15:43.03 | leifmadsen | rtp set debug ip xx.xx.xx.xx |
| 15:43.13 | moos3 | k will set it up |
| 15:43.14 | leifmadsen | that won't have anything to do wtih the RTP stuff |
| 15:43.25 | leifmadsen | you're listening on 5060, other end probably isn't if they are behind NAT |
| 15:44.18 | *** join/#asterisk k3asd` (~k3asd`@static-94-32-127-180.clienti.tiscali.it) |
| 15:44.40 | k3asd` | hi |
| 15:47.15 | moos3 | leifmadsen https://gist.github.com/8b3a59c0c3b63bd82c50 |
| 15:47.47 | moos3 | that was me making a call, they could hear me but I can't hear them |
| 15:48.02 | moos3 | 1.6.2 this worked, 1.8 only effecting 2 users |
| 15:51.54 | moos3 | leifmadsen ideas ? |
| 16:01.30 | leifmadsen | moos3: ya, looks like asterisk was only getting RTP from one directly |
| 16:01.33 | leifmadsen | direction* |
| 16:01.43 | moos3 | yeah |
| 16:01.51 | moos3 | I can't figure out why its just effecting me |
| 16:06.41 | *** join/#asterisk bchia (~Adium@nat/digium/x-rvomxedfcxcsvvaj) |
| 16:11.14 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 16:13.12 | *** join/#asterisk navaismo (~navaismo@189.144.207.195) |
| 16:22.11 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
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| 16:33.59 | qakhan | Rac-on and wdoekes SIPAddHeader(Call-Info: answer-after=0) is not working |
| 16:34.35 | wdoekes | qakhan: there were three distinct headers posted |
| 16:34.44 | wdoekes | with a short summary of phone types |
| 16:35.02 | wdoekes | surely you can come up with a better report than "option 1 doesn't work" |
| 16:37.58 | qakhan | do i need to enable auto answer somewhere in sip.conf? |
| 16:40.04 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 16:42.30 | Greenlight | Hi folks, and happy new year! I'm having troubles with CallerID on an ISDN30 (in UK - Virgin Media/Telewest). Asterisk 1.8.0 and Dahdi 2.6.1. Outgoing callerid used to work and now does not, we're not sure when it stopped working and in that time both asterisk and dadhi have been updated to what were latest versions. Are there any known issues with CallerID or is it likely something has been |
| 16:42.30 | Greenlight | chagned externally? |
| 16:43.10 | *** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 16:46.49 | *** join/#asterisk rue_work (~rue_mohr@24-207-100-190.eastlink.ca) |
| 16:47.10 | rue_work | anyone awake I have a problem with mgcp.conf |
| 16:47.55 | rue_work | if I specify more than one mgcp device, only the last one defined works |
| 16:48.39 | rue_work | taps on the glass...'hello?' |
| 16:51.06 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:52.50 | Rac-on | qakhan: certain phones require you to enable auto-answer in the phone-configuration |
| 16:53.04 | Rac-on | qakhan: and like wdoekes said, try 1 of the other mentioned headers |
| 16:53.41 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:55.52 | qakhan | Rac-on i tried all |
| 16:55.57 | qakhan | but not working |
| 16:56.19 | rue_work | anyone ever used mgcp or am I the only one? |
| 17:00.49 | SuperNull | i laughed when one of our providers wanted to use MGCP over sip ;) heh. |
| 17:01.07 | SuperNull | Anyone use Linksys ATAs ? (spa-21XX) |
| 17:02.03 | rue_work | ok I have two mgcp gateways that I'm trying to link calls between, but I cant get them both to work at the same time with asterisk |
| 17:02.21 | rue_work | I can get one to work |
| 17:02.29 | rue_work | and it can be either one |
| 17:02.35 | rue_work | but not both |
| 17:02.48 | rue_work | I dont know if mgcp.conf is limited to one gateway |
| 17:03.14 | rue_work | *afk* |
| 17:05.01 | Katty | hello my asterisk does not work at all it seems to have a dinosaur on it. how to fix plz??? |
| 17:06.00 | WIMPy | Wait a few thousand years until the dinosaurs become etinct. |
| 17:08.41 | Faustov | walk the dinosaur |
| 17:13.17 | leifmadsen | turn the dinosaur into gasoline and power the generator that runs your asterisk box |
| 17:13.36 | *** join/#asterisk fullstop (~fullstop@64-121-16-14.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 17:13.55 | fullstop | What is a reliable ATA these days? I'm looking for one with 1xFXO and 1xFXS |
| 17:14.20 | WIMPy | doesn't think that technology existed when the dinosaurs did. |
| 17:15.07 | Katty | leifmadsen: clever boy. |
| 17:15.19 | Katty | WIMPy: ancient aliens disagrees with you! lol |
| 17:15.38 | Katty | ALIENS, i tell you. ALIENS. |
| 17:15.41 | Katty | flashes the crazy hair |
| 17:15.51 | WIMPy | You think they killes the dinosaurs? |
| 17:18.01 | Katty | something sure did. |
| 17:18.11 | Katty | they're lyin all about the place. skeletons everywhere. |
| 17:18.22 | Katty | i'm not so sure about Aliens doing it. |
| 17:18.31 | Katty | but they're definately extinct. |
| 17:18.37 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:18.52 | fullstop | IF the aliens were around and had access to reliable hardware, which ATA would they use? |
| 17:19.04 | Katty | *hee* |
| 17:19.05 | fullstop | and don't say alienware |
| 17:19.07 | Katty | i like this guy |
| 17:19.23 | qakhan | Rac-on any update? |
| 17:20.15 | Katty | fullstop: if you don't get an answer, call voipsupply.com and ask them what their best seller is. |
| 17:20.29 | Katty | fullstop: i don't use them, so can't really give any advice. |
| 17:21.01 | fullstop | Often times the best seller is something which has been discontinued, but I'll give it a shot. |
| 17:22.01 | Katty | that happens. |
| 17:22.08 | Katty | you can ask them about their second best seller too ;) |
| 17:22.32 | fullstop | Are ATA's that uncommon now? |
| 17:22.52 | Katty | well since i've never used one, i can't say i have a good answer |
| 17:23.14 | Katty | but..i don't see a whole lot of plain ole phones sittin around when i go visit clients |
| 17:23.42 | Katty | there's another vendor that's pretty good. telephony depot |
| 17:23.56 | Katty | i think fender bender introduced me to them |
| 17:24.00 | fullstop | Right. I normally don't deal with analog, but this would be for home. |
| 17:24.25 | fullstop | I used to be in this channel quite often, but I don't do as much with voip these days. |
| 17:24.33 | Katty | i have a number for telephony depot if you want it. |
| 17:24.35 | fullstop | You might remember my daughter vs her soup: http://i.imgur.com/9NijB.jpg |
| 17:26.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.206) |
| 17:26.50 | Katty | i have a retired polycom at my house. |
| 17:27.05 | fullstop | do you feed it crackers? |
| 17:27.11 | Katty | it mostly collects dust. |
| 17:27.22 | Katty | hehe |
| 17:27.29 | fullstop | that is not a very ambitious thing to collect. |
| 17:27.49 | WIMPy | has lots of phones collecting dust. |
| 17:28.05 | fullstop | my phone collects dust... because nobody calls me. |
| 17:28.41 | WIMPy | Redirect all invalid extensions to your phone and you will get lots of calls. |
| 17:28.45 | Katty | ALSO |
| 17:28.53 | Katty | leifmadsen: happy birthday two ewe! |
| 17:28.55 | fullstop | interesting idea. |
| 17:28.57 | Katty | leifmadsen: happy birthday two ewe! |
| 17:29.06 | Katty | leifmadsen: happy birthday deer ewe!!!! happy birthday two ewe!!!!! |
| 17:30.41 | *** part/#asterisk GeoGeek (~steve-o_@69.26.219.66) |
| 17:32.56 | tzanger | oh yeah it's his birthday |
| 17:33.01 | tzanger | he's gotta be what now, at least 15 |
| 17:33.26 | fullstop | Driving into work today, I realized that I didn't know my age. |
| 17:33.50 | fullstop | I did the math and was a year older than I thought I was. |
| 17:33.52 | tzanger | fullstop: I always have to ask my wife |
| 17:33.53 | Katty | yeah i have to subtract years all the time |
| 17:34.40 | Katty | guess it's not important enough to remember ;) |
| 17:37.44 | [TK]D-Fender | <qakhan> but not working <- you haven't told us what you're using. |
| 17:49.24 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:50.52 | leifmadsen | tzanger: at least! |
| 17:54.08 | tzanger | :-) How old are you now? |
| 17:58.19 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 18:00.22 | rokjan | o/ |
| 18:03.49 | rokjan | where to get table structure ARA Asterisk 1.8? which exists in: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure it's only for SIP accounts. |
| 18:03.51 | rokjan | help me |
| 18:04.10 | fullstop | 34 |
| 18:07.39 | gusto | so |
| 18:18.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:22.02 | leifmadsen | tzanger: 32 :) |
| 18:23.25 | WIMPy | leifmadsen: Happy 6th bit then. |
| 18:23.38 | leifmadsen | thanks :D |
| 18:23.43 | leifmadsen | clever, I like that |
| 18:24.00 | leifmadsen | hopefully I can make it to the 7th bit |
| 18:24.06 | leifmadsen | 7 is my lucky number... |
| 18:24.27 | leifmadsen | that sounds somewhat ominous |
| 18:24.39 | WIMPy | Well, you know it takes twice the time for each additional one. |
| 18:26.16 | fullstop | Even turtles have difficulty making it to the 8th bit. |
| 18:32.28 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 18:32.45 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 18:34.25 | SuperNull | so if i looped through an SQL result is there a way to put the contents into a search able .. variable some how ? |
| 18:34.52 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:34.52 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:45.48 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 18:48.38 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:51.31 | *** join/#asterisk ryan42 (unix@doc-72-47-18-129.brenham.tx.cebridge.net) |
| 18:52.38 | ryan42 | Greetings. I have a quick question as I can't seem to figure this out. I have FreePBX + Digium phones. I'm trying to set a custom ring for an incoming route with Alert Info in FreePBX |
| 18:52.39 | reisi | should ubuntu 1.8 include cli originate command? (as in http://www.voip-info.org/wiki/view/Asterisk+cli+originate ) |
| 18:52.54 | ryan42 | Something like "alert alert_info="normal" ringtone_id="Digium" ring_type="normal" |
| 18:53.28 | ryan42 | however that isn't working, and I can't figure out if alert info should carry the actual alert info, or if it's just a reference that's defined elsewehre like sip_additional.conf |
| 18:54.15 | navaismo | reisi, you mean Asterisk? |
| 18:54.46 | navaismo | maybe your alias conf file is not setup and you should try with: Channel Originate |
| 19:03.42 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:06.57 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.206) |
| 19:13.28 | *** join/#asterisk PipBoy (~PipBoy@66.212.187.33.tor.pathcom.com) |
| 19:20.40 | jacekowski | ryan42: you need DPMA for that to work |
| 19:21.00 | jacekowski | ryan42: and phone has to be provisioned from DPMA |
| 19:22.28 | fullstop | So who here is sick / getting over being sick? |
| 19:22.32 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 19:22.45 | fullstop | Apparently a good portion of the world is sick at the moment, based on my limited sampling. |
| 19:26.17 | leifmadsen | yep, for sure |
| 19:26.19 | leifmadsen | wife is sick |
| 19:26.23 | leifmadsen | I haven't been sick this year yet |
| 19:26.32 | fullstop | I'm just getting over something my daughter gave me. |
| 19:26.43 | leifmadsen | but I work from home and avoid sick people -- I'm almost a dick when it comes to avoiding people who are sick around me |
| 19:26.52 | fullstop | I know it's from her because she stuck her finger in her mouth and then poked me in the eye. On purpose. |
| 19:28.32 | fullstop | I assume that you have no children. If you plan on having them, know that they are tiny germ factories. |
| 19:28.56 | fullstop | And then you send them to school, which turns into a gigantic germ factory. |
| 19:29.13 | file | I banned my mother from visiting when she was sick because of a trip, but then a snow storm had my delay it anyway |
| 19:29.19 | file | erm had me |
| 19:29.33 | file | hates his travel luck |
| 19:30.26 | mjordan | file: part of this may have something to do with your location |
| 19:30.51 | file | mjordan, it's rarely on the legs for here! |
| 19:30.52 | fullstop | twist: file lives in Arizona |
| 19:34.24 | leifmadsen | fullstop: kid due in Feb, and I'm well aware of how much more sick I'm going to start getting in the coming years |
| 19:34.58 | fullstop | congratulations, btw. On the kid, not the sickness. |
| 19:38.18 | leifmadsen | :) |
| 19:40.43 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:42.19 | tzanger | whoa wait |
| 19:42.23 | tzanger | wow congrats leif! |
| 19:42.28 | tzanger | do you know what it is? |
| 19:42.48 | fullstop | a baby, you fool! |
| 19:42.49 | tzanger | file: why, were you on this flight? http://www.nydailynews.com/news/national/drunk-passenger-duct-taped-gagged-aboard-flight-article-1.1233554 |
| 19:42.53 | tzanger | (who do you think they tied up?) |
| 19:43.03 | file | no comment |
| 19:43.37 | tzanger | fullstop: amen to that |
| 19:44.01 | *** join/#asterisk citywok (~kvirc@208.186.203.146) |
| 19:44.16 | citywok | ~itsp |
| 19:44.16 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:44.20 | tzanger | late last year I was reading to my 3yo and he barfed on me, the chair, the carpet, the dresser, his stuffed puppy... there wasn't anything I could do either since that would involve me tracking it through the house |
| 19:44.44 | citywok | I need an ITSP for a number in France, and a number in Germany. Does anybody have any suggestions? (U.S. based here, so flying blind) -- Thanks! |
| 19:50.28 | *** part/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 19:50.59 | fullstop | My girls have strong stomaches, but both had explosive diapers. As in, shot poo up their backs and into their hair. |
| 19:51.24 | fullstop | We keep a chair in a particular spot in the room because of that. |
| 19:51.54 | cusco | hi |
| 19:52.05 | cusco | I'm using AMI action originate |
| 19:52.49 | cusco | I'm setting a variable, and not having it in dialplan.. the correct syntax to set a var is "Varialbe: webId=$callid\r\n" |
| 19:52.52 | cusco | right? |
| 19:55.05 | cusco | ahh typo |
| 19:55.09 | cusco | Varialbe vs Variable |
| 19:55.11 | cusco | argh |
| 19:57.38 | *** join/#asterisk epaphus (~user1@108.174.50.29) |
| 19:58.06 | epaphus | Hello. Is this the correct way to set a custom callerid when dialing out... exten => _8X.,2,Set(CALLERID(num)="954678987") |
| 19:58.15 | epaphus | obviously this would match when i dial 8 as prefix.. right |
| 19:59.04 | epaphus | for some reason when i add that line, calls are not getting out. Dead air. If i remove it then calls out work |
| 20:00.44 | [TK]D-Fender | epaphus, pastebin the entire relevant section of your dialplan and the call attempt |
| 20:01.00 | [TK]D-Fender | epaphus, Also, NO QUOTES after the = |
| 20:01.06 | [TK]D-Fender | epaphus, First thing... |
| 20:01.20 | ryan42 | jacekowski: I believe it is |
| 20:01.30 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 20:01.36 | epaphus | it was the quotes! thanks |
| 20:01.36 | epaphus | :) |
| 20:02.03 | ryan42 | jacekowski: I have the DMPA addon installed |
| 20:02.19 | ryan42 | The phones are listed in the Digium Phones section of FreePBX |
| 20:20.45 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:25.01 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:28.28 | ghost75 | is there a command to get the callerid from a phone? |
| 20:28.56 | rrittgarn | you mean from a channel? |
| 20:28.59 | ghost75 | from all phones |
| 20:29.24 | rrittgarn | well the ${CALLERID} can be called to get the CID info or set it |
| 20:29.52 | ghost75 | there is no call, i have a list with all internal extensions and want to include callerid from sip.conf |
| 20:31.01 | rrittgarn | what's your goal here then? just a list of extensions with caller id? |
| 20:31.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:32.00 | ghost75 | i have webpage that displays all extensions with their state and you can do click2dial |
| 20:32.02 | jacekowski | ryan42: digium_phones show sessions |
| 20:32.30 | ryan42 | it shows my three phones |
| 20:32.36 | ryan42 | total active sessions:3 |
| 20:34.41 | rrittgarn | ghost75: Are you running real time? We run realtime here and its just a query against my device table (sip.conf effectively) |
| 20:34.55 | ghost75 | https://www.sugarsync.com/piv/D234635_65941903_648551 <- how it looks |
| 20:35.57 | ghost75 | i dont want to give rights on sip.conf to apache |
| 20:35.58 | rrittgarn | ah yeah I don't use that webGUI... you could use the AMI to grab a sippeer and parse that out if you're customizing that |
| 20:36.29 | ghost75 | of course you dont use it because i did it |
| 20:37.00 | rrittgarn | looked a bit like the web GUI project |
| 20:37.27 | ghost75 | as soon i am gonna parse it over ami, the webserver needs read rights on sip.conf |
| 20:37.37 | ryan42 | jacekowski: so I have "ring-group1" as alert info from my ring group but I don't know where to define that it should be a separate ringtone |
| 20:38.21 | rrittgarn | ghost75: with AMI you would just need read rights in the manager.conf for it, Thus not giving apache direct access to sip.conf. I assume you're using AMI to do the click action? |
| 20:39.24 | ghost75 | no ami is then just able to execute the commands |
| 20:40.57 | rrittgarn | ghost87: i use the http://serverip:8088/asterisk/mxml? page for loading dynamic content on my front end, but the asterisk server's web UI is only accessible via my apache server (different box/different network) separated by Firewall rules |
| 20:42.05 | ghost75 | whats on that port ? |
| 20:42.26 | rrittgarn | 8088 is the default port |
| 20:42.34 | rrittgarn | but you can do a curl request to grab the xml formatted info |
| 20:42.37 | rrittgarn | and then parse that out |
| 20:42.39 | ghost75 | port for what |
| 20:42.48 | rrittgarn | sorry asterisk listens on that |
| 20:42.50 | rrittgarn | misread |
| 20:43.11 | rrittgarn | if you have webenabled=yes |
| 20:44.07 | ghost75 | ah the diginum gui |
| 20:44.14 | ghost75 | dont have that installed |
| 20:44.24 | rrittgarn | its in stock asterisk |
| 20:44.48 | ghost75 | didnt they remove it in later versions? |
| 20:44.56 | rrittgarn | nope, still running it on 11 |
| 20:45.14 | rrittgarn | its built into the AMI |
| 20:46.06 | rrittgarn | only issue with it that i've found is that you have to use curl to create the cookie to pull it from the pbx properly |
| 20:46.16 | ghost75 | http://192.168.2.2:8088/asterisk/ <- doesnt start |
| 20:46.18 | rrittgarn | so parsing out AMI might be easier depending on your preference via socket |
| 20:46.29 | rrittgarn | you have webenabled=yes in your manager.conf? |
| 20:46.39 | ghost75 | ah manager, wait |
| 20:47.08 | rrittgarn | then restart asterisk and a netstat should show it listening on that port as well as the usual 5038 (unless you specify otherwise of course) |
| 20:48.11 | ghost75 | in general section? |
| 20:48.48 | rrittgarn | yep |
| 20:49.16 | rrittgarn | used to come in the sample docs |
| 20:50.03 | ghost75 | doesnt show |
| 20:50.16 | rrittgarn | timeout or page not found? |
| 20:50.31 | ghost75 | nmap doesnt even show port |
| 20:51.11 | rrittgarn | from the asterisk cli: manager show settings |
| 20:51.16 | rrittgarn | what's enabled? |
| 20:51.38 | ghost75 | Web Manager (AMI/HTTP): Yes |
| 20:51.54 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 20:52.13 | rrittgarn | did you sent the bindaddr in the manager.conf? |
| 20:52.44 | ghost75 | ok, it was localhost thats reason |
| 20:53.49 | ghost75 | but still not works lol |
| 20:53.55 | rrittgarn | so http://192.168.2.2:8088/asterisk/mxml?action=login&username=myuser&secret=mysecret doesn't yeild anything? |
| 20:54.06 | ghost75 | no |
| 20:54.17 | ghost75 | maybe something is missing in install |
| 20:54.25 | rrittgarn | server still isn't listening on 8088 if you do nestat -a |
| 20:54.31 | kaldemar | http.conf |
| 20:54.37 | serafie | ghost75: This is documentation for Asterisk GUI (which does not come with Asterisk, however Asterisk does come with an HTTP *server* not *gui*) This documentation includes information on troubleshooting the HTTP server which may help you: https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI#AsteriskGUI-Troubleshooting |
| 20:55.29 | rrittgarn | ghost75: Kaldemar is right... i forgot about that one... |
| 20:55.48 | *** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 20:56.13 | serafie | ghost75: "http show settings" and "manager show settings" CLI commands may give you additional information as well. |
| 20:59.54 | jacekowski | ryan42: you have to define it all in res_digium_phones.conf |
| 21:00.05 | jacekowski | ryan42: as in ringtone and stuff |
| 21:00.34 | ryan42 | okay. Right now my res_digium_phone.conf file just has includes for 4 other files (due to freepbx) |
| 21:00.43 | ryan42 | I have tried putting it in _devices.conf but it gets overwritten |
| 21:00.44 | ghost75 | http://192.168.2.2:8088/asterisk -> The requested URL was not found on this server. |
| 21:00.54 | ghost75 | at least port is now open |
| 21:01.24 | ryan42 | so can I just put the overrides in say _additional? for instance [100] \n type=phone \n alert=alert=alert-from-pstn |
| 21:01.57 | serafie | ghost75: you may need to add prefix=asterisk to http.conf |
| 21:01.59 | jacekowski | ryan42: http://pastebin.com/zNpMC1e6 |
| 21:02.01 | serafie | or remove it |
| 21:02.09 | serafie | experiment. |
| 21:02.13 | jacekowski | ryan42: that's how it's supposed to loo |
| 21:02.38 | ryan42 | ah okay. that's pretty much waht I have come up with but now I just need to put it somewhere where freepbx doesn't overwrite it |
| 21:02.47 | jacekowski | ryan42: where did you get digium phones module for freepbx |
| 21:03.15 | ryan42 | I just installed it through PBX and put my key and such for asterisk in |
| 21:03.26 | ryan42 | err key from digium rather |
| 21:03.54 | rrittgarn | ghost75: add /mxml or another one of the subfolders to there |
| 21:04.03 | ghost75 | http://192.168.2.2:8088/asterisk/httpstatus <- that worked |
| 21:04.06 | rrittgarn | the /asterisk directory will yield a 404 by default |
| 21:04.19 | rrittgarn | yep so http://192.168.2.2:8088/asterisk/mxml |
| 21:04.39 | rrittgarn | there's also one for plain text, and one with a hoakey click do dial via AMI for testing things |
| 21:04.59 | ghost75 | This XML file does not appear to have any style information associated with it. The document tree is shown below. |
| 21:05.06 | ghost75 | <generic response="Error" message="Missing action in request"/> |
| 21:07.11 | rrittgarn | have you used the AMI before? it accepts ami commands via get request |
| 21:07.38 | rrittgarn | so http://192.168.2.2:8088/asterisk/mxml?action=login&username=MyUserName&secret=MySecretPassword |
| 21:07.40 | ghost75 | i have used ami over normal manager access 5038port |
| 21:08.02 | rrittgarn | same theory here, just all at once. So after you log in you can do stuff like action=sippeers |
| 21:08.47 | rrittgarn | ghost75: http://www.voip-info.org/wiki/view/Asynchronous+Javascript+Asterisk+Manager+(AJAM) old but most of it is accurate |
| 21:08.49 | ghost75 | http://192.168.2.2:8088/asterisk/amanager <- get a login here but fails to login in |
| 21:09.22 | rrittgarn | credentials line up with your manager.conf or manager.d/* |
| 21:09.31 | ghost75 | yes |
| 21:10.26 | ghost75 | maybe rights problem |
| 21:10.52 | rrittgarn | possibly. I don't use that very often (the amanager), just the mxml from my protected networks. |
| 21:13.13 | ghost75 | i dont know path from this webserver files |
| 21:15.08 | rrittgarn | its just the GET requests... so like i have it above. just replace the action= stuff |
| 21:16.28 | ryan42 | jacekowski: so in your example... [paging-alert] defines the alert and alert=paging-alert activates it for teh phone, right? |
| 21:16.46 | ryan42 | paging-ring-answer is the text that must be specified as alert-info for the incoming trunk to activate that alert? |
| 21:21.00 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 21:24.36 | ghost75 | rrittgarn: <generic response="Success" message="Already authenticated"/> |
| 21:24.38 | ghost75 | :) |
| 21:24.45 | jacekowski | ryan42: yes |
| 21:24.56 | ryan42 | jacekowski: doesn't seem to be working here :-/ |
| 21:26.04 | rrittgarn | ghost75: after you get authorized, you issue commands like : http://192.168.2.2:8088/asterisk/action=listcommands |
| 21:26.13 | rrittgarn | making sense? |
| 21:27.02 | ghost75 | http://192.168.2.2:8088/asterisk/mxml?action=listcommands |
| 21:27.40 | ryan42 | i have to go afk for a bity |
| 21:30.14 | ghost75 | k i have now sip.conf, but looks a bit ugly |
| 21:30.59 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:32.46 | rrittgarn | yeah you will have to parse it in whatever language your looking for... I assume you're not hard coding all this, so you will need to parse the XML into something useable, and probably loop through to grab the sippeer info for each host to get the CID, but that's whatever you're programming in, and not asterisk |
| 21:32.46 | ghost75 | hard to extract callerid from there |
| 21:33.50 | epaphus | Could anybody help me with this? Call from '13010' (63.233.226.97:5060) to extension 's' rejected because extension not found in context 'from-external'. i dont find any extensions "s" in that context... :S |
| 21:34.20 | WIMPy | Yes, that's what it says. |
| 21:35.02 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 21:35.05 | ghost75 | good old dial patterns |
| 21:36.42 | *** join/#asterisk gusto (~gusto@bband-dyn133.95-103-217.t-com.sk) |
| 21:37.31 | jmetro | epaphus: you answered your own question - but to clarify, somewhere you have a peice of code that is going to from-external with the target "s" and since S isnt there... |
| 21:38.24 | WIMPy | Or someone calling s or nothing. |
| 21:39.52 | ghost75 | or the given exten doesnt exist and jumps to s ? |
| 21:40.15 | *** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com) |
| 21:40.22 | jmetro | ghost75: i believe thats "i" for invalid. |
| 21:40.39 | WIMPy | indeed |
| 21:42.31 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 21:47.39 | epaphus | iam going to paste my from-external context |
| 21:48.12 | WIMPy | Why? |
| 21:48.20 | WIMPy | We all know it has no s. |
| 21:48.53 | WIMPy | Maybe you should actually ask a question. If you're looking for some sort of answer, that is. |
| 21:49.28 | jmetro | 13010 looks like an internal number, why is it hitting from-external |
| 21:49.33 | jmetro | start there? |
| 21:50.12 | WIMPy | Possibly. We don't know. |
| 21:50.39 | jmetro | ^ was directed at epaphus |
| 21:57.12 | Docfxit | I would like to record a voice prompt. I added extension 130 to extensions.conf. The lines I added are at pastebin.com/3rMxnTLs I have restarted Asterisk. It isn't working. Could someone please suggest what I need to do to get it working? |
| 21:57.54 | jmetro | what is the problem you are having..what goes wrong |
| 21:58.12 | WIMPy | First of all it would help if you post full URLs. |
| 21:58.58 | Docfxit | jmetro: When I dial the extension I don't hear a beep. No file gets created. |
| 21:59.05 | WIMPy | And second I'd try to Answer() before recording. |
| 21:59.14 | jmetro | get rid of the first wait - not necessary. also [referencing mine] it is asterisk-recording.gsm not :gsm |
| 22:00.03 | WIMPy | that too |
| 22:00.03 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:00.16 | jmetro | and why are you using numbers instead of n, and extens instead of same. |
| 22:01.41 | Docfxit | Jmetro: I just copied off the internet. |
| 22:03.11 | Docfxit | I just changed the numbers to n. I don't know how to code same instead of extens. |
| 22:03.28 | jmetro | wimpy's suggestion is probably the most effective for fixing it though. mine are just coding OCD. |
| 22:04.19 | jmetro | exten => 130,1,answer followd by same=> n,Playback(please record blah blah..). |
| 22:04.39 | jmetro | same => (step#),(action) |
| 22:04.53 | jmetro | read up on dialplan basics from the book or wiki |
| 22:05.49 | jacekowski | ryan42: have you reconfigured the phone |
| 22:06.03 | jacekowski | ryan42: digium_phones reconfigure all |
| 22:09.23 | *** join/#asterisk NightMonkey_ (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 22:10.59 | *** join/#asterisk navaismo (~navaismo@189.191.2.44) |
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| 22:15.25 | *** join/#asterisk schultza (~schultza@rc1.rcherbals.com) |
| 22:15.47 | schultza | does running two asterisk servers on the same network conflict with each other? no sip trunk of any kind... |
| 22:16.42 | serafie | schultza: done many times. |
| 22:16.57 | schultza | how does it conflict?> |
| 22:17.08 | serafie | shouldn't. I meant it has been done many times successfully. |
| 22:17.33 | schultza | ok.. so i can run a test environment from another machine (asterisk instance) on the same network? |
| 22:21.26 | *** join/#asterisk jkroon (~jkroon@41.22.181.231) |
| 22:25.04 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-heryiddnuriyyofu) |
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| 22:42.42 | ryan42 | jacekowski: I did do that. |
| 22:43.01 | ryan42 | let me pastebin my config |
| 22:44.54 | ryan42 | jacekowski: http://pastebin.com/FmfhyLjZ see anything out of whack? |
| 22:50.36 | ryan42 | ah wait I think I got it |
| 22:50.38 | ryan42 | syntax error in another file |
| 22:52.43 | ryan42 | Thanks very much for pointing me in the right direction! |
| 23:01.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:04.54 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:17.43 | Docfxit | jmetro: wimpy: Thank you both for the help. I applied the suggestions. It still isn't working. I have to go now. I'll be back in touch to try to get it working. |
| 23:19.32 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
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| 23:25.54 | *** join/#asterisk Get_The_Fish (~get_the_f@173-164-50-49-colorado.hfc.comcastbusiness.net) |
| 23:26.38 | Get_The_Fish | Does anyone know how to force a polycom phone to reset to factory defaults? I just cannot get this phone to accept the fact that there is no blessed boot server to connect to any longer |
| 23:27.32 | epaphus | Hello, can anybody please help me interpret this error.. iam unable to make calls... [Jan 7 17:26:21] WARNING[6199]: chan_sip.c:20552 handle_response_invite: Received response: "Forbidden" from '"585005" <sip:585005@16.228.204.43>;tag=as29bcfad6' |
| 23:27.36 | epaphus | Forbidden by who? |
| 23:27.51 | [TK]D-Fender | Get_The_Fish: Where's the problem? |
| 23:28.07 | Get_The_Fish | "I just cannot get this phone to accept the fact that there is no blessed boot server to connect to any longer" |
| 23:28.16 | [TK]D-Fender | Get_The_Fish: Doesn't matter if a boot server is there. If it has configs on it then it will boot |
| 23:28.30 | [TK]D-Fender | There is no "don't loks for a bot server" option that I've seen. |
| 23:28.39 | [TK]D-Fender | But it doesn't matter if there is or not |
| 23:28.59 | Get_The_Fish | And thats exactly what I DONT want it to. I need to reconfigure the thing, and I cant because it keeps reverting to its old configuration |
| 23:29.06 | [TK]D-Fender | epaphus: pastebin the complete call with SIP DEBUG enabled |
| 23:29.07 | [TK]D-Fender | ~pb |
| 23:29.08 | 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. |
| 23:29.21 | zerohalo | Get_The_Fish: what model? |
| 23:29.32 | Get_The_Fish | Soundpoint 301 |
| 23:29.36 | [TK]D-Fender | Get_The_Fish: the only way it reverts is if there IS a boot server it's connecting to. |
| 23:29.46 | Get_The_Fish | Well that isnt the case. |
| 23:29.50 | zerohalo | hold all of 4,6,8 and * |
| 23:29.52 | Get_The_Fish | I did |
| 23:31.18 | zerohalo | can you get into the admin interface? |
| 23:31.21 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 23:32.33 | Get_The_Fish | zerohalo, the admin interface as in the web interface? No. And since the phone is a reboot loop I only have access to the limited setup menu. But that doesnt matter because it cant find a boot server and keeps reverting to it's previous configuration |
| 23:33.16 | zerohalo | no - the interface on the phone |
| 23:33.19 | zerohalo | admin settings... |
| 23:33.33 | Get_The_Fish | "since the phone is a reboot loop I only have access to the limited setup menu" |
| 23:33.39 | zerohalo | ah |
| 23:33.51 | Get_The_Fish | "But that doesnt matter because it cant find a boot server and keeps reverting to it's previous configuration" |
| 23:34.01 | [TK]D-Fender | Reboot loop is the kind of thing that happens when it's got corrupted configs and can't even minimally boot |
| 23:34.08 | zerohalo | right |
| 23:34.10 | Get_The_Fish | No, it isnt |
| 23:34.22 | Get_The_Fish | Because I'm watching it |
| 23:34.52 | [TK]D-Fender | Get_The_Fish: Then how far does it get? How long does it last? |
| 23:35.06 | [TK]D-Fender | Get_The_Fish: What version of BR & SIP are you running on it? |
| 23:35.22 | Get_The_Fish | It cant pull an IP address, because it still thinks that it's on VLAN 2. It's not. So, it reboots. |
| 23:35.44 | zerohalo | you can remove the vlan in the ethernet menu |
| 23:35.56 | epaphus | [TK]D-Fender, http://pastebin.ca/2300253 |
| 23:35.59 | [TK]D-Fender | So lack of network interface then? |
| 23:36.02 | Get_The_Fish | I remove the setting VLAN 2, tell it to start, it gets to the "Welcome! Processing cfg" and fails. |
| 23:36.16 | [TK]D-Fender | Sthen that part is still corrupted |
| 23:36.19 | Get_The_Fish | ?? Lack of network interface?? No. |
| 23:36.25 | Get_The_Fish | That part is not corrupted. |
| 23:36.27 | [TK]D-Fender | And the VLAN is its own layer of fail |
| 23:36.39 | Get_The_Fish | It used to be on VLAN 2. It is not anymore. |
| 23:36.41 | [TK]D-Fender | "Welcome! Processing cfg" and fails. <- evidence appears otherwise |
| 23:37.15 | [TK]D-Fender | So make a new provisioning folder and dump in stock sample configs & firwmware for it to pick up |
| 23:37.16 | Get_The_Fish | What in the world are you talking about TK? |
| 23:37.23 | Get_The_Fish | I cant do that. |
| 23:37.30 | [TK]D-Fender | because? |
| 23:38.11 | Get_The_Fish | Because I dont have unfettered access to this entire subnet, and since it's looking for a boot server on a 10.x.x.x network, and is now on a 192.168 subnet, it fails. |
| 23:38.53 | [TK]D-Fender | There is NOTHING you can point the phone to ANYWHERE? |
| 23:39.29 | [TK]D-Fender | epaphus: From: "100" <sip:100@46.228.204.43>;tag=as29817a6b <- set fromuser=(yourusername) |
| 23:39.42 | [TK]D-Fender | epaphus: as well as "sendrpid=yes" |
| 23:40.01 | Get_The_Fish | And how am I supposed to tell the phone to do that, do you suggest? Because it will not accept my settings. |
| 23:40.30 | Get_The_Fish | Hence the original question, how do you reset a polycom phone to factory default? |
| 23:41.05 | [TK]D-Fender | Get_The_Fish: So you power the phone, drop immediately into the boot menu. Manually change the server type, address info, etc and it doesn't stick at all? |
| 23:41.15 | Get_The_Fish | NO |
| 23:41.43 | Get_The_Fish | Not to mention the fact that it is so old I wouldnt even know which firmware to use on it. |
| 23:42.00 | [TK]D-Fender | "no", what i said isn't true? or "no, it fails like you just described?" |
| 23:42.10 | [TK]D-Fender | And the latter is NOT the problem |
| 23:42.28 | Get_The_Fish | No as in the setting do not stick |
| 23:42.30 | [TK]D-Fender | Get_The_FishNot to mention the fact that it is so old I wouldnt even know which firmware to use on it. <- not knowing what to give it has no impact with being able to TRY it. |
| 23:42.48 | [TK]D-Fender | Get_The_Fish: So you go in, make changes. hit save. Immediately go back in and they're gone? |
| 23:42.59 | Get_The_Fish | Here TK, I have a few more hairs over here you can split. |
| 23:43.12 | epaphus | TK thanks! |
| 23:43.36 | Get_The_Fish | No TK, when I enter the settings manually and it attempts to boot and fails, it reverts to the previous config. |
| 23:44.10 | epaphus | TK it worked just by setting the from-user.. but what is sendrpid = yes for? |
| 23:44.36 | [TK]D-Fender | epaphus: So that they still accept CID sent in the Remote-party ID |
| 23:44.55 | [TK]D-Fender | Get_The_Fish: Give it a place to grab configs from then so it doesn't fail |
| 23:45.10 | Get_The_Fish | With which firmware? |
| 23:45.23 | [TK]D-Fender | Any that it supports |
| 23:45.30 | Get_The_Fish | because this phone isnt supported anymore |
| 23:45.33 | [TK]D-Fender | Hopefulle >= what it has now |
| 23:46.15 | Get_The_Fish | SO, apparently the answer to the question, at the end of the day |
| 23:46.21 | Get_The_Fish | (the part I care about) |
| 23:46.23 | Get_The_Fish | is that you cant |
| 23:46.37 | [TK]D-Fender | ... |
| 23:46.58 | [TK]D-Fender | If what it's got is baked and can't boot to a USER INTERFACE on the phone then you've already fucked it up |
| 23:47.10 | [TK]D-Fender | You have to restore a certain minimum for it NOT to have a heart attack |
| 23:47.42 | Get_The_Fish | Sure pal. I appreciate you trying to help. |
| 23:48.09 | [TK]D-Fender | I see great resistance at your accepting it |
| 23:48.23 | [TK]D-Fender | The tone is most than evident |
| 23:48.27 | [TK]D-Fender | more* |
| 23:48.47 | Get_The_Fish | Because it isnt valid assistance. |
| 23:48.51 | [TK]D-Fender | It is. |
| 23:49.19 | Get_The_Fish | And frankly TK, this isnt my first rodeo with Polycom. Nor with you. |
| 23:49.55 | Get_The_Fish | The IRC "you already fucked it up" really isnt appreciated, and after several years of it from you it's rather old. |
| 23:50.13 | [TK]D-Fender | You are far too sensitive and think this is a game |
| 23:50.15 | [TK]D-Fender | I've been working with Polycom phones for the past EIGHT years and this past week have reprovisioned over a dozen IP 600/601 and more to firmware from LAST YEAR. |
| 23:50.26 | [TK]D-Fender | I know what I'm talking about and have real experience with these |
| 23:50.42 | [TK]D-Fender | You want to waste you time , well apparently YOU knwo better because YOUR setup it working, riight? |
| 23:50.47 | [TK]D-Fender | Oh wait. IT ISN'T |
| 23:50.57 | Get_The_Fish | So have I, and so do I. And screw you for assuming I don't. My last polycom shop was a 50 seat all polycom call center. |
| 23:50.59 | [TK]D-Fender | So Don't dream of trying to shove my help back in my face. |
| 23:51.23 | Get_The_Fish | then drop the attitude and you wont get it shoved back in your face. |
| 23:51.35 | Get_The_Fish | because frankly I'm sick and tired of it. |
| 23:52.10 | tzanger | Get_The_Fish: I have a polycom ip501 here |
| 23:52.30 | tzanger | Get_The_Fish: menu, 2, 456, enter, 2, I have "4. Reset to default" -- does that not work? |
| 23:52.36 | Get_The_Fish | I dont have that menu |
| 23:52.40 | tzanger | mind you if you can get to the admin menu you should be able to do whatever you want |
| 23:52.47 | tzanger | which menu are you missing specifically? |
| 23:52.59 | Get_The_Fish | the menu button does nothing. |
| 23:53.16 | tzanger | Get_The_Fish: it sounds like you have to call polycom; this is not something I've got experience with |
| 23:54.41 | [TK]D-Fender | tzanger: Config failure prevents the SIP app from loading. |
| 23:54.56 | [TK]D-Fender | tzaSame sort of thing a corrupted app would o as well depending on how bad. |
| 23:55.14 | tzanger | odd, that seems like a really broken thing to do on polycom's part |
| 23:55.36 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 23:55.40 | [TK]D-Fender | tzanger: Nothing pointing it to a fresh provisioning source can't fix as long as you didn't kill the BootROM |
| 23:55.58 | [TK]D-Fender | Which is a big plus for it coming in 2 parts |
| 23:56.02 | tzanger | [TK]D-Fender: if the phone's network config isn't hosed as well |
| 23:56.24 | [TK]D-Fender | tzanger: that's the BR's side. He apparently can get in there so its all app-side now |
| 23:56.25 | tzanger | as a designer of embedded systems having a device refuse to work due to bad config is a bad design |
| 23:56.30 | tzanger | and I love polycomps |
| 23:56.32 | tzanger | er polycoms |
| 23:56.42 | tzanger | ahh |
| 23:57.10 | [TK]D-Fender | tzanger: Well the app itself could be corrupted too....but yes, bad configs will prevent full boot. |
| 23:58.15 | [TK]D-Fender | And this twit let his history and attitude get in the way of getting help from people who do know better. |
| 23:58.26 | [TK]D-Fender | [18:55][TK]D-Fendertzanger: Nothing pointing it to a fresh provisioning source can't fix as long as you didn't kill the BootROM <- can* |
| 23:59.12 | tzanger | [TK]D-Fender: yeah I saw the more recent scrollback, it's unfortunate but that too happens sometimes |
