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02:48.04 | pigpen | I have a little problem. |
02:48.23 | pigpen | I accidentally deleted my extensions.conf |
02:48.25 | pigpen | ;-) |
02:48.43 | pigpen | Via the cli, I do not have "save dialplan" or "dialplan save". |
02:48.53 | pigpen | I can do: dialplan show, but the formatting is not right |
02:49.18 | pigpen | any clever way to dump the config to a file or the screen that would be useful? |
02:50.42 | pabelanger | pigpen, what version of asterisk? |
02:52.01 | pigpen | yeah, that would be helpful: 1.6.1.12 |
02:55.59 | pabelanger | well, that is pretty funny and shitty for you |
02:56.00 | pabelanger | https://issues.asterisk.org/jira/browse/ASTERISK-17712 |
02:56.11 | pabelanger | it is a bug that dialplan save is missing |
02:57.12 | pabelanger | you can't even unload and load a patch version cause you'd nuke the local memory |
02:57.14 | pabelanger | so, 2 options |
02:57.32 | pabelanger | 1 dump screen and manually import it all again |
02:57.45 | pabelanger | 2, use source control management for all your configuration files |
02:58.16 | pigpen | yeah. I was figuring that just a dialplan show and re-create. |
02:58.27 | pigpen | I have a copy from about 2 years ago, so better than nothing. |
02:58.43 | pigpen | with show dialplan, I should be able to fill in the blanks. |
02:58.54 | pigpen | thanks though... |
02:59.05 | pigpen | needless to say, I plan to get this on 10 or 11 soon. |
02:59.21 | pabelanger | what is your filesystem? |
02:59.27 | pigpen | reiser. |
03:00.20 | pigpen | It was a bonehead mistake. I was hopping between about 6 live asterisk boxes and a test box. |
03:00.27 | pigpen | then: oops. |
03:00.30 | pigpen | then: shit! |
03:00.33 | pigpen | heh |
03:00.48 | pabelanger | well, you can try to recover it using reiser tools |
03:01.05 | pabelanger | but, if you have 6 boxes, you really need to store your stuff in git or subverison |
03:01.53 | pigpen | heh. yeah, I am a slacker. |
03:02.27 | pigpen | I have setup many with some rsync scripts, but I was planning to move this box to a new one, but, didn't get there yet. |
03:02.35 | pigpen | love svn. |
03:02.51 | pigpen | This is like my first post in this cannel for about 6 months. |
03:04.47 | pigpen | thanks for the info. on to grunt work. |
03:05.15 | Penguin | You'd better get that "dialplan show" before something else goes wrong. |
03:05.41 | pigpen | heh, 3 times actually. |
03:06.05 | Penguin | Why don't you have backups? |
03:07.15 | pigpen | I was quite surprised that I don't. So I am asking the same question. |
03:09.53 | pabelanger | $ git init . |
03:09.57 | pabelanger | $ git add * |
03:10.01 | pabelanger | $ git commit -a -s |
03:10.02 | pabelanger | done |
03:10.05 | pabelanger | now you are using it |
03:11.47 | Penguin | There has to be more to it than that. |
03:11.58 | Penguin | Where does that store the files? |
03:12.18 | Penguin | (I use rsync, so I don't know how git works.) |
03:13.47 | pabelanger | well, that will get you just local source control, so if you delete a file, you can just do: $ git check -- foo.conf |
03:13.50 | pabelanger | and it comes back |
03:14.08 | pabelanger | $ git checkout -- foo.conf |
03:14.29 | pabelanger | git is distributed, so you can push the repo anywhere |
03:14.51 | pabelanger | git remote add upstream http://github.com/foo/bar.git |
03:14.56 | pabelanger | git push upstream master |
03:14.57 | pabelanger | done |
03:21.27 | Penguin | I may try it. |
03:21.38 | Penguin | I'm a little uncomfortable with it. |
03:47.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.199) |
04:16.34 | ChannelZ | (locally it's stored in a dir called .git in the directory you init'd) |
04:18.40 | Penguin | Do I have to init the current dir? Do I have to add files from the current dir? |
04:19.10 | Penguin | E.g., could I git init /some/place and git add /etc/asterisk/* while sitting in my $HOME? |
04:19.11 | ChannelZ | You init at the level you want to control |
04:21.15 | ChannelZ | No, you can't add files "outside" of the directory (that I'm aware of) |
04:23.57 | Penguin | But I could be in some random directory and init /etc/asterisk followed by add /etc/asterisk/* ? |
04:26.16 | ChannelZ | Well the init would work, I'm not actually sure what the syntax for add would be though (or commit, etc) to tell it the repository |
04:27.57 | Penguin | I also don't understand why my man page says git-init, git-add, etc. rather than git init, git add... |
04:28.37 | ChannelZ | They're just links |
04:29.05 | ChannelZ | git looks at argv0 and if it's called as "git-add", it's the same as 'git add', etc. |
04:32.17 | ChannelZ | There's a pretty good book you can read online, a-la the Asterisk book, at http://git-scm.com/book |
04:32.57 | ChannelZ | I think somewhere on that same site there's a whole interactive 'learn git' thing that explains it piece by piece and has you type in things in a fake virtual shell |
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04:34.04 | Penguin | Are there squirrels in Australia? |
04:34.47 | pabelanger | How random |
04:35.11 | ChannelZ | Yes but they have funny accents |
04:35.41 | Penguin | Good grief, 40m is busy tonight. |
04:36.13 | ChannelZ | I guess it's non-sequitor night in #asterisk |
04:36.48 | Penguin | Why not? |
04:36.55 | ChannelZ | Is that your CB handle? Australian Squirrel? |
04:36.59 | Penguin | haha |
04:38.24 | Penguin | There was something on TV with squirrels and my wife asked if I knew there aren't any squirrels in Australia. I decided I would ask here since we have people from all over the world hanging out here. |
04:40.41 | Penguin | The comment about 40 meters was because I kept hearing some noise coming from my radio, which was tuned to 7200. It sounded off frequency, so I tuned down some and stopped at 7195... but on the way down 5 kHz, there were at least three conversations. And there isn't even enough bandwidth for that. |
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04:40.50 | ChannelZ | I don't actually know, but I'm sure someone has dragged one in as a pet somewhere along the way |
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04:41.21 | Penguin | I tuned around a little more and determined that there is probably more traffic on there than I have heard at any point in the last six months. |
04:41.59 | gravspeed | so does anyone have any tips for me troubleshooting a cisco 7940 that won't register to my asterisk 1.8 box? |
04:42.07 | Penguin | SIP? |
04:42.20 | gravspeed | yes, sip |
04:42.26 | ChannelZ | bigger hammer |
04:42.34 | Penguin | Did you use config files on a tftpd or manually set the phone? |
04:42.46 | gravspeed | lol @channelZ |
04:42.57 | gravspeed | i've tried both, same results |
04:43.31 | gravspeed | the phone can make calls out but does not show as registered therefore cannot recieve calls |
04:43.36 | Penguin | Is the phone in the same subnet as asterisk? |
04:43.57 | gravspeed | i also have s spa504g and a yealinkt28p and t26p that work perfectly. |
04:44.16 | gravspeed | no, the phone is at home with me now, the pbx is in our datacenter |
04:44.40 | Penguin | The other phones you mentioned are also at home with you now? |
04:44.52 | gravspeed | i brought a spa504 with me |
04:45.04 | Penguin | It works from home? |
04:46.16 | Penguin | Working and working in your current location are two different things. |
04:46.32 | gravspeed | i'm setting up my local dhcp now, it works from my office connected through a simulated t1 |
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04:49.49 | gravspeed | i'm kinda new to the voip game, how do i watch the registration process to see where it fails? |
04:50.10 | gravspeed | i'm sure i'm just not googleing it right... |
04:50.20 | ChannelZ | sip set debug on |
04:51.15 | Penguin | I'm going to take a guess that the register never makes it to asterisk. |
04:51.58 | gravspeed | could be true... i can't pass that debug through grep or something can i? |
04:52.30 | ChannelZ | well if you know the phone's IP you can do sip set debug ip x.x.x.x to limit to just that one |
04:53.18 | Penguin | Oh, I guess there's some sort of contesting going on. |
04:53.41 | ChannelZ | But if you have some verbose on and don't see an error on the console about a busted registration when you reboot the phone, it probably isn't trying as Penguin says |
04:54.18 | Penguin | I think the phone is trying, but I bet it never makes it to asterisk. |
04:54.59 | Penguin | I suspect something like a Belkin router. |
04:58.53 | *** join/#asterisk gravspeed2 (~gravspeed@c-50-173-59-122.hsd1.ca.comcast.net) |
04:59.18 | gravspeed2 | didn't want to give me my username back... dhcp server online, phones starting up... |
04:59.37 | Penguin | gravspeed is still online. |
05:00.08 | gravspeed2 | i know i am... had to restart my router to switch dhcp servers... crap netgear at home... |
05:00.25 | Penguin | I was close. I said Belkin. |
05:00.45 | gravspeed2 | lol... i'm not that cheap... |
05:01.47 | gravspeed2 | cisco switches take forever to start up |
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05:17.15 | gravspeed2 | so it seems my surplus catalyst has some interesting vlan configs in it... |
05:17.28 | gravspeed2 | this might take me a while |
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05:31.18 | *** join/#asterisk Basssie90 (b.kooijan@5ED2711E.cm-7-3b.dynamic.ziggo.nl) |
05:33.29 | Basssie90 | can someone tell me how to install a svn package? I don't want to work on it. but there is no trunk directory availible. |
05:34.05 | Basssie90 | do I need to search for some file? |
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05:52.25 | ChannelZ | no comprende |
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06:14.25 | Basssie90 | I found it, thanks for your reaction ChannelZ |
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07:26.27 | hos7ein | hi |
07:27.32 | hos7ein | for install asterisk of source code, I should be farst install LibPRI and DAHDI and now Asterisk? |
07:28.19 | hos7ein | farst= First |
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07:43.11 | hos7ein | for install asterisk of source code, I should be first install LibPRI and DAHDI and now Asterisk? |
07:43.40 | Basssie90 | just follow the steps: https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source |
07:44.17 | Basssie90 | with svn you can use http://www.asterisk.org/downloads/source-code |
07:45.28 | Basssie90 | do you need LibPri and DAHDI? |
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08:17.20 | hos7ein | Basssie90: i have install asterisk for test |
08:24.23 | ChannelZ | Do you have any TDM hardware? |
08:32.23 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
08:33.14 | hos7ein | ChannelZ: no, i have not the TDM |
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08:39.10 | ChannelZ | then you don't need DAHDI or LibPRI |
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09:39.46 | *** join/#asterisk admin0 (~admin0@5356416B.cm-6-7b.dynamic.ziggo.nl) |
09:40.08 | admin0 | hi all ..is there any function/etension.conf i can use to echo back the callerID/CLI of any incoming call ? |
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11:17.12 | WIMPy | admin0: Look at application SayDigits and function CALLERID. |
11:17.28 | admin0 | WIMPy: thanks |
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11:30.35 | Ice_Strike | Which software can play PCM file? |
11:30.44 | Ice_Strike | VLC dont seem to play it |
11:31.14 | roswell | Ice_Strike, sox |
11:32.29 | Ice_Strike | to convert? |
11:32.51 | roswell | it includes 'play' binary afair |
11:32.56 | Ice_Strike | I see |
11:33.09 | Ice_Strike | Let see if it available on Windows |
11:38.39 | WIMPy | If vlc doesn't play it, I'd check if the file is ok. |
11:39.03 | WIMPy | Usually vlc and mplayer play everything. |
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13:42.00 | roswell | Ice_Strike, did it help you on windows? |
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15:13.27 | Pegasus_RPG | Hello. I'm running * 1.8.13 (via Debian) and just wanted to know how I might get my dialplan to accept letters (and convert them to keypad digits if the SIP provider can't accept letters.) |
15:16.04 | Penguin | There aren't very many DTMF letters, and asterisk should be fine with them already. |
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15:25.45 | Pegasus_RPG | Penguin: I guess my problem is that my dialplan has stuff like exten => _1NXXNXXXXXX, 2, dial(SIP/${EXTEN}@voipprovider) |
15:25.59 | Pegasus_RPG | looks up pattern matching |
15:26.06 | [TK]D-Fender | Pegasus_RPG: So make your pattern accept letters |
15:26.34 | Penguin | And use SIP/voipprovider/${EXTEN} isntead. |
15:26.46 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
15:31.10 | [TK]D-Fender | Well he'd have to do some of that work of "converting", so it's doubtful that he'll be using that variable in the dial itself |
15:32.26 | Pegasus_RPG | Can I use a pattern like [a-zA-Z]. as one would in a regex? |
15:34.03 | [TK]D-Fender | that looks like what the docs say you use... |
15:34.33 | Pegasus_RPG | is reading http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
15:35.50 | [TK]D-Fender | Pegasus_RPG: so what does your sample actually mean in the end? |
15:37.07 | *** join/#asterisk kresp0 (~kresp0@gateway/tor-sasl/kresp0) |
15:37.08 | Pegasus_RPG | Well right now, I'm going to try accepting 1NXX[1-9a-zA-Z][0-9a-zA-Z]. and see what my VoIP provider does with that |
15:38.33 | Penguin | If you need fixed length extensions, you just about have to use a pattern like that. But if you don't need to have a fixed length, consider that . will match both numbers and letters. |
15:39.16 | Pegasus_RPG | oh, just . alone? |
15:39.39 | Pegasus_RPG | Okay, so my provider reports congestion when I pass it letters |
15:39.45 | Pegasus_RPG | time to convert |
15:43.10 | Penguin | When you say provider, I think ITSP. ITSP is used for a gateway between the PSTN and a VoIP system. So that brings me to the questions... Why would you want to put DTMF letters onto the PSTN? I've never heard of that before. |
15:43.52 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-216-033.pools.arcor-ip.net) |
15:44.29 | ghost75 | when libspandsp2 segfaults as soon as fax is incoming via receivefax, is this problem of libspandsp2 or receivefax ? |
15:46.19 | Pegasus_RPG | Penguin: well some companies advertise their toll-free numbers with letters, e.g. 800-discover and my VoIP client doesn't show a dialplad so I have to look at a real telephone to figure out the numbers and I'd rather my * do that for me. :D |
15:47.02 | Pegasus_RPG | *dialpad |
15:47.26 | Pegasus_RPG | (of course Q and Z are problematic but are usually avoided) |
15:47.30 | Penguin | That's just weird. |
15:47.47 | Pegasus_RPG | what part? |
15:48.31 | ghost75 | [3442216.866983] asterisk[17684]: segfault at db7f3e3c ip b664319e sp b590c8b0 error 5 in libspandsp.so.2.0.0[b6622000+b4000] |
15:49.05 | ghost75 | ok more asterisk segfaults ... |
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15:49.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
15:59.22 | Penguin | Sending letters, especially ones that are not DTMF letters, is weird. |
16:04.54 | *** join/#asterisk gravspeed (~gravspeed@c-50-173-59-122.hsd1.ca.comcast.net) |
16:05.12 | gravspeed | good morning |
16:05.14 | Penguin | If you insist on doing that, you could use patterns similar to _N. and _1N. and then cut the length to a maximum of 10 and 1+10 respectively, followed by converting each letter to the corresponding number. |
16:05.16 | *** join/#asterisk serafie (~erin@24.96.64.240) |
16:05.25 | gravspeed | penguin, you are still here? |
16:05.31 | Penguin | looks around |
16:05.33 | Penguin | Me? |
16:05.36 | Penguin | Nope. |
16:05.56 | gravspeed | you were on last night when i started my project at home |
16:06.02 | Penguin | I recall that. |
16:06.31 | gravspeed | so i went to bed last night, woke up this moring and got all the vlans out of this switch i have |
16:06.41 | Penguin | That's a good start. |
16:07.02 | gravspeed | so i am getting dhcp to my phones, they both register enough to make outbound calls |
16:07.10 | Penguin | No... |
16:07.17 | Penguin | Registrations has NOTHING to do with outbound calls. |
16:07.19 | gravspeed | but my 7940 will not recieve inbound |
16:07.41 | Penguin | That part is due to registration not working. |
16:08.08 | Penguin | And there is no "register enough." A device is either registered or it isn't. |
16:08.57 | gravspeed | fair enough |
16:09.10 | gravspeed | the spa504 is fully registered and can recieva calls in |
16:09.25 | gravspeed | the 7940 can make calls out but is not registered |
16:09.33 | Penguin | And if you run sip show peers, you see what? |
16:10.46 | gravspeed | status unknown host (unspecified) |
16:11.03 | gravspeed | for the 7940, the spa is ok and my ip address |
16:12.04 | Penguin | sip set debug on; then power cycle the phone. |
16:12.13 | Penguin | See if you can see the REGISTER packets at all. |
16:12.24 | gravspeed | what was the command to only debug one ip |
16:12.26 | gravspeed | ? |
16:12.36 | gravspeed | nevermind |
16:12.38 | gravspeed | i remember |
16:12.46 | Penguin | Do you have other phones on the system? |
16:13.03 | gravspeed | several |
16:13.07 | Penguin | okay |
16:13.09 | gravspeed | only 2 here with me |
16:13.27 | gravspeed | phone is rebooting now |
16:13.42 | Penguin | In many cases, debugging by IP is discouraged. |
16:13.52 | Penguin | In this case, I think it will be okay. |
16:14.09 | gravspeed | maybe not... phone is rebooted, not a single line from debug |
16:15.03 | gravspeed | ok, when i make a call out it gives info |
16:15.43 | gravspeed | when i call into it it also gives info... |
16:15.45 | Penguin | If there are no REGISTER packets, maybe you forgot to tell the phone to register. |
16:16.06 | gravspeed | the phones are autoconfigured |
16:16.18 | Penguin | With magic? |
16:16.19 | gravspeed | i can check it's config... let me grab a copy of it |
16:16.21 | Penguin | its |
16:18.31 | gravspeed | <PROTECTED> |
16:19.37 | gravspeed | that is from the SEP%MAC%.cnf.xml |
16:20.16 | Penguin | That file isn't really relevant. |
16:20.33 | Penguin | The important file is SIP<MAC>.cnf. |
16:20.45 | gravspeed | my bad |
16:21.20 | gravspeed | so then proxy_register: "1" |
16:21.41 | gravspeed | proxy address is correct |
16:22.18 | gravspeed | so are the names and password |
16:22.25 | Penguin | I keep all those settings in the SIPDefault.cnf, and only the phone-specific things in SIP<MAC>.cnf. |
16:22.41 | gravspeed | nat_enable: "1" |
16:22.45 | Penguin | Change that. |
16:22.51 | Penguin | nat_enable 0 |
16:22.55 | gravspeed | NO NAT ENABLE? |
16:23.15 | Penguin | Correct. That breaks asterisk's ability to deal with things behind NAT. |
16:23.26 | gravspeed | oic |
16:27.02 | gravspeed | ok, so this is where it gets fun |
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16:27.43 | Penguin | Yeah? |
16:27.53 | gravspeed | this is a bicom pbxware multi tenant |
16:28.11 | gravspeed | our support contrct isn't active yet or i would be talking to them (no offense) |
16:28.30 | gravspeed | the cnf files do not exist until they are requested |
16:29.08 | Penguin | That name means nothing to me. If it's asterisk and Cisco config files, that's enough for me to work on it. |
16:29.33 | gravspeed | i have changed the settings for the 7940 to nat = no, however the file still comes to me with nat enabled |
16:30.27 | Penguin | The cnf needs to have nat_enable: 0, and asterisk should have nat=yes for that peer. |
16:30.51 | gravspeed | oh, in the ui i did show nat as yes |
16:30.59 | gravspeed | i turned it to no |
16:31.20 | gravspeed | reloaded configs and re-requested the sipMAC.cnf |
16:31.32 | Penguin | If the device corresponding to the peer is behind a NAT, the setting should be nat=yes. |
16:32.08 | gravspeed | and by peer you mean phone correct? i am defianlty behind nat here |
16:32.24 | gravspeed | and i appearently can't spell this morning |
16:32.31 | Penguin | The phone is a peer, but I mean the peer definition in sip.conf. |
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16:33.35 | gravspeed | ok, i do have a sip.conf and all extensions have nat=yes |
16:33.45 | Penguin | Extensions go in extensions.conf. |
16:34.16 | Penguin | There are only peers/devices in sip.conf. |
16:34.25 | gravspeed | i guess these would actually be users then... |
16:34.28 | Penguin | Nope. |
16:34.34 | Penguin | Users are people. |
16:34.53 | gravspeed | so then i need to get my terms straight... |
16:34.58 | Penguin | And we don't configure people in asterisk. |
16:35.43 | Penguin | Phones are phones or devices. Users are people. Extensions are the numbers that phones send to do useful things like calling other phones. |
16:36.06 | gravspeed | here in the sip.conf i have each configured extensions information broken down by what is refered to in the ui as username |
16:36.07 | Penguin | Or, if you are [tk]d-fender, extensions are used to make coffee. |
16:36.20 | gravspeed | mmm, coffee |
16:36.37 | Penguin | There are no extensions found in sip.conf. Extensions are configured in extensions.conf. |
16:36.50 | Penguin | sip.conf contains the device configurations. |
16:37.51 | gravspeed | so if i have nat=yes in all of my devices... |
16:38.28 | Penguin | That tells asterisk that those devices are to be treated as if they are behind a NAT. |
16:38.37 | gravspeed | which they are, so that's good... |
16:38.48 | Penguin | It tells asterisk to look at the source of the packet rather than the address that the packet says to use. |
16:39.30 | Penguin | Because the PBX usually can't do anything with an RFC1918 address found inside the packet coming from a device behind a NAT. |
16:40.07 | Penguin | But then setting a nat traversal setting on the phone itself ruins all that. |
16:40.23 | Penguin | It tries to change things before sending the packets. |
16:40.35 | Penguin | Mix the two and it usually ends in failure. |
16:41.33 | gravspeed | hmm... i just grapped the sipMAC.cnf for my spa, it also has nat enable: "1" |
16:41.43 | gravspeed | nat_enable |
16:41.53 | gravspeed | but it works... |
16:42.20 | Penguin | Consider that phone fortunate. |
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16:42.53 | gravspeed | lol... i have several phones at the office also working (all behind nat) |
16:43.26 | gravspeed | i might wager they are all set that way... let me check |
16:43.35 | Penguin | But then you run into something that behaves differently and proper configuration is required. Crazy stuff. |
16:44.56 | gravspeed | all the other phones seem to be set the same... |
16:45.09 | gravspeed | crazy stuff |
16:45.58 | gravspeed | so i can't just go edit these files, they don't exist on the server... unless they are somewhere locate can't find them... |
16:46.57 | gravspeed | it seems that these files are created when they are requested |
16:47.20 | Penguin | locate might not be the best tool for that job. I don't know what you know about systems, but locate requires a pre-built database to too you where your files are. |
16:47.30 | gravspeed | updatedb |
16:47.36 | Penguin | s/too /tell / |
16:48.11 | Penguin | I have to assume people don't know that. |
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16:53.23 | gravspeed | it seems that there are a series of php files that create those cnf files when they are requested |
16:54.28 | gravspeed | but they are encoded somehow, not plain text except for the first line (copyright info) |
16:54.39 | Penguin | How annoying! |
16:54.47 | gravspeed | it's kinda stagne |
16:54.49 | gravspeed | strange |
16:55.25 | gravspeed | but it is a fully supported multi-tenant asterisk with all the bells and whisles |
16:55.32 | gravspeed | i can't spell this morning |
16:56.10 | gravspeed | all i have to do is wait for our support contract to activate, hopefully wednesday, just trying to figure out as much on my own as i can |
16:59.21 | gravspeed | even if the nat travesal were screwed, shouldn't we see something in the sip debug? |
17:02.18 | ChannelZ | ideally |
17:02.37 | gravspeed | ahh, in a perfect world |
17:04.05 | ChannelZ | and your PHP is probably 'precompiled' with eAccellerator or XCache or one of those |
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17:09.01 | gravspeed | so i copied all the files the phone requested that actually exist on the "real"tftp server and copied them to my own, changed the nat_enable and testing now |
17:10.34 | ghost75 | great, * is still crashing with old libspandsp2 |
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17:13.19 | gravspeed | well that didn't work out either... |
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17:33.11 | gravspeed | is there a way to turn nat traversal off on the cisco 7940? |
17:33.26 | gravspeed | without being able to edit the sipMAC.cnf file? |
17:37.13 | ghost75 | isnt this more a feature of the router |
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18:03.16 | ghost75 | can i use gdb to debug asterisk? |
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19:17.11 | gravspeed | so i got my 7940 to register |
19:18.00 | gravspeed | firmware update to the phone fixed it... hopefully didnt break anything else |
19:18.50 | gravspeed | thank you penguin and channelz for your tips |
19:19.48 | gravspeed | i have an issue with dhcp options i was wondering if you guys have any tips for. |
19:21.09 | gravspeed | the spa phones work when option 66 is set to "tftp://IP/", the 7940 works with option 150 set to IP |
19:21.43 | gravspeed | the spa phone will not even get its config when option 150 is enabled. |
19:22.21 | gravspeed | the 7940 works properly with both options enabled. |
19:23.45 | gravspeed | can i make the spa ignore option 150? or just not get it without making reservations for all the phones? |
19:28.21 | dym | Evening. |
19:29.42 | gravspeed | good evening dym |
19:32.03 | dym | Hello there, gravspeed |
19:32.48 | gravspeed | you have any experience with cisco spa phones by any chance dym? |
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19:41.40 | dym | Indeed I have. But not conecerning your issue |
19:43.15 | dym | I've used a lot of Linksys SPA 941 phones in one of my past installations. |
19:47.59 | gravspeed | dym: are you using dhcp option 66 or 150? |
19:48.58 | dym | I dont use provisioning. |
19:49.11 | dym | s/dont/didn't/ |
19:49.33 | ChannelZ | I do but don't use dhcp options |
19:49.52 | gravspeed | i have about 700 phones to replace over the next month... auto-provisioning is important to me |
19:50.21 | dym | hah, i love infobot! |
19:50.29 | gravspeed | channelz: how are you provisioning without dhcp options? |
19:50.43 | dym | Peace, ChannelZ! |
19:51.51 | ChannelZ | If I have to install a new phone, I just have a little script that hits a URL on the phone and tells it to download a bootstrap config |
19:51.53 | Pegasus_RPG | If an outbound call is proceeding through the dialplan, how can I replace certain digits and have it continue through the dialplan with the new stringreplaced digits? |
19:51.57 | ChannelZ | Then they provision themselves from there |
19:52.25 | gravspeed | hmm... can i see that script by any chance? |
19:53.08 | dym | Pegasus_RPG: Like first digits? |
19:53.25 | navaismo | Pegasus_RPG, you can use the ${EXTEN} with the strip methods like ${EXTEN:1} to remove first digit: 33${EXTEN:1} to add 33 and remove first digit and so on |
19:53.29 | Pegasus_RPG | dym: sure, that'll point me in the right direction |
19:53.37 | ChannelZ | http://pastebin.com/WU4mjM30 |
19:53.41 | dym | Pegasus_RPG: there you go :) |
19:54.05 | Pegasus_RPG | ah ok. So EXTEN is just a variable like anything else huh. |
19:54.11 | dym | indeed. |
19:54.19 | dym | It's the dialled "Number". |
19:54.20 | ChannelZ | so once I know the IP of the phone (which I usually pre-assign from the MAC address) I just do thatscript.php 1.2.3.4 |
19:55.14 | ChannelZ | then spa922.cfg gives it the firmware I want and tells it the 'permenant' provisioning URL for its self |
19:55.42 | gravspeed | hmm... i'm trying to avoid having to set reservations for all the phones, like i said i have sereal hundread to do in short order... |
19:55.42 | ChannelZ | It updates, reboots, and then fetches its own config. |
19:56.01 | gravspeed | and i still can't spell... |
19:57.44 | gravspeed | according to what i'm reading from cisco it should process option 66 before option 150, but if option 150 is present it will not fetch config... |
20:01.46 | gravspeed | they make mention of a DHCP_Option_To_Use which sounds like it should be part of a config file... |
20:01.56 | gravspeed | but they do not say which one |
20:04.36 | ghost75 | my 7941 is also using 150 |
20:07.32 | ChannelZ | Well it seems like 66 is more flexible in that you can specify the whole url.. with 150, what does it try to grab? |
20:07.46 | ghost75 | DHCP Option 150 is Cisco proprietary. The IEEE standard that matches with this requirement is Option 66. Like option 150, option 66 is used to specify the Name of the TFTP server. |
20:08.08 | ghost75 | Difference between option 150 and option 66 |
20:08.09 | ghost75 | •DHCP option 150 supports a list of TFTP servers (Multiple Server IPs) |
20:08.09 | ghost75 | •DHCP option 66 only supports the IP address or the hostname of a single TFTP server. |
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20:29.56 | Pegasus_RPG | dym: Hmm, I am setting EXTEN to something else but it doesn't appear to get replaced. exten => _NX!,2,Set(EXTEN=${KEYPADHASH(${EXTEN}):0:10}) |
20:30.59 | Pegasus_RPG | That is, the log show that it is replaced but the rest of the dialplan that should match the replacement, isn't. I can confirm by changing it to match the original and it does |
20:33.00 | Pegasus_RPG | sigh, nvm. FOund it in the docs "cannot be modified with the set command- just use the GoTo to change the EXTEN variable!" |
20:34.32 | dym | :) |
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