| 00:06.36 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 00:11.50 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 01:33.01 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 01:39.52 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:39.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:41.08 | *** join/#asterisk kami` (~user@unaffiliated/kami-) |
| 01:45.11 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:57.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 02:01.43 | *** join/#asterisk bintut (~bintut@cm102.sigma15.maxonline.com.sg) |
| 02:01.49 | bintut | waves |
| 02:02.53 | bintut | is on 2nd day of trying to fix the g729 codec issue |
| 02:04.11 | bintut | information of my setup is at http://www.pastie.org/3970890 |
| 02:23.58 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:37.14 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:49.01 | bintut | what is special in making g729 codec work properly? |
| 02:50.03 | bintut | i don't understand why it doesn't work properly as expected. i bought the g729 codec license from digium and i'm using the binary .deb asterisk v1.8.11 also from digium. |
| 02:50.58 | bintut | digium's readme install is pretty straight forward. |
| 02:52.03 | bintut | even after following instructions, still g729 doesn't work as expected on my home pbx. more information at http://www.pastie.org/3970890 |
| 02:56.30 | bintut | anyone? |
| 03:00.28 | ChannelZ | I just see a warning, does it actually not transcode? |
| 03:02.49 | bintut | ChannelZ: yes, i don't think it transcode. but apart from transcoding since my sip call goes first to a aa/ivr which has a filename of menu.g729, the sound i'm hearing is so fast from the original. |
| 03:04.07 | bintut | ChannelZ: first, i expect that since my call in g729 and it plays a Background() of menu.g729 file, i should be hearing the same thing i hear when using ulaw - the normal sound |
| 03:05.00 | bintut | ChannelZ: second, after timeout from the ivr, it dials to the analog phone via fxs. that part does not transcode. |
| 03:05.01 | ChannelZ | where did menu.g729 come from? |
| 03:05.49 | bintut | ChannelZ: i converted it from my menu.raw using asterisk's "file convert" command |
| 03:06.22 | ChannelZ | well it's probably playing fast due to some problem with the conversion then |
| 03:06.38 | bintut | ChannelZ: unless, the way i converted the menu.raw to menu.g729 is wrong already. |
| 03:07.43 | ChannelZ | Not sure why it's not transcoding though. All I see is that the DAHDI/4 call fails |
| 03:07.43 | bintut | ChannelZ: so i suspect that the first problem maybe because as what you've said is not correctly converted to g729 format but the second part should be a transcoding issue |
| 03:08.04 | ChannelZ | Is the only thing actually working the call to DAHDI/1 ? |
| 03:08.11 | ChannelZ | (is that your FXS, analog phone?) |
| 03:08.45 | bintut | ChannelZ: both 1 and 4 are not working properly. my tdm card is still from the old digium's dev kit |
| 03:09.14 | bintut | i mean, 1 and 4 are not working properly when transcoding from g729 codec is involved |
| 03:09.23 | *** join/#asterisk miltux (~miltux@62.1.166.245.dsl.dyn.forthnet.gr) |
| 03:09.47 | ChannelZ | What I mean is that one of the DAHDI calls is failing period -- see line 241 of your paste |
| 03:10.21 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 03:10.49 | bintut | ChannelZ: yes, you're correct. but i can test again and give you an updated output log if you want with the sip set debug on |
| 03:12.43 | ChannelZ | no.. so what happens when you answer whatever channel DOES work? Do you just get silence? |
| 03:13.53 | ChannelZ | actually forget that - remove the 'r' flag from your Dial |
| 03:14.27 | bintut | ok |
| 03:16.56 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:17.02 | ChannelZ | arggggh f*@#@#ing cable |
| 03:17.15 | ChannelZ | <ChannelZ> actually forget that - remove the 'r' flag from your Dial |
| 03:17.15 | ChannelZ | <ChannelZ> might try removing 'm' too if that doesn't work.. it appears to be a bug |
| 03:18.24 | ChannelZ | And besides all that, it could be your multiple-channel Dial triggering it |
| 03:18.34 | ChannelZ | https://issues.asterisk.org/jira/browse/ASTERISK-17541 |
| 03:20.51 | bintut | ChannelZ: i have "mt" now on my Dial() and transcoding is working great! :) |
| 03:22.54 | bintut | ChannelZ: i answered my call on my analog phone and i can hear both ends. i check if indeed the transcoding is being used by executing the command "g729 show licenses" and the output is here => http://www.pastie.org/3974639 |
| 03:23.07 | bintut | ChannelZ: thank you for your help! :) |
| 03:23.44 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 03:23.48 | bintut | my only problem now is on how i can properly convert my menu.raw into a menu.g729 in g729 codec format |
| 03:24.53 | bintut | ChannelZ: can you recommend to me the best tool to convert my audio file into a correct g729 format? |
| 03:25.38 | ChannelZ | it's probably your original "raw" file that is wrong. What is it/what did you create it with? |
| 03:28.23 | bintut | ChannelZ: i created it with the use of the GarageBand but i can't remember exactly where i put the original file. |
| 03:30.25 | bintut | ChannelZ: how to check the format of an audio file? in *nix, normally i execute the command: "file filename" from the shell to get an information of the file. but when i do that on my audio files, i am getting the same results as "DOS executable (device driver)" |
| 03:37.57 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 03:42.59 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 03:49.45 | bintut | ChannelZ: brb. thank you! :) |
| 03:49.53 | bintut | waves.. brb. |
| 03:54.50 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 04:08.03 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 04:31.11 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:42.55 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 04:58.47 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.136.154) |
| 05:01.16 | *** join/#asterisk shadebob (~shadebob@41.141.237.252) |
| 05:10.41 | *** join/#asterisk hfb (~hfb@cpe-98-151-249-95.socal.res.rr.com) |
| 05:34.51 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 05:38.31 | atan | Well color me confused! There's something fishing going on here. So I have two boxes, both 1.4, identical configs (like, copies... seriously) one lets me register, one spits out 401 unauthorized. I'm out of ideas on what to change here, the only diff I can think of is the 'working' one happens to be on a VPS which perhaps might use some form of NAT, but that's all I know about it |
| 05:38.37 | atan | ~nat |
| 05:38.37 | infobot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 05:38.52 | atan | !nat |
| 05:39.05 | atan | Hmm I would have sworn there was a link with details on this :) |
| 05:56.18 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 05:57.01 | urvg4 | ~itsp |
| 05:57.01 | 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. |
| 05:57.16 | urvg4 | ~itsplist-us |
| 05:57.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 06:00.58 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 06:09.55 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:11.24 | urvg4 | ~itsplist-ca |
| 06:11.25 | infobot | i guess itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 06:12.31 | kaldemar | atan: sipnat is the keyword you're thinking about |
| 06:14.01 | atan | ~sipnat |
| 06:14.01 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 06:20.10 | atan | Well something isn't correct here. Maybe I goofed up on a config, but from what I can see in the sip debug the peers that connect okay hit the box, get a 401 then send auth details which turn into a 200 response. The Cisco and Polycom phone, as well as softphones hit the box for the 401 but are not following up with sending any auth details |
| 06:21.07 | atan | That or it's sending it but it isn't going to the correct spot and Asterisk isn't seeing it... *bangs head* |
| 06:21.21 | atan | By the end of 2012, I will have this done. |
| 06:39.56 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 06:43.40 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 06:57.05 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 07:03.54 | ChannelZ | are those phones otherwise aware they are supposed to auth? (IE they have a username/password set, etc.) |
| 07:12.34 | atan | ChannelZ, well using the same config (all%, same database, same files, same version of asterisk, same os... but different provider) the phones can connect to the old server if I pull it up. It's just the new server that's causing this... I don't know why. |
| 07:13.04 | atan | s/all%/100%/ |
| 07:17.48 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 07:20.55 | ChannelZ | Have you rebooted the problematic phones? Are they caching bad DNS or something? |
| 07:21.25 | atan | The phones have been rebooted. They're hitting the right server. |
| 07:21.52 | atan | What is killing me is one device registers fine. It's a Grandstream ATA. It hits, gets a 401, sends auth, and away it goes... |
| 07:22.29 | atan | But the other 4 or 5 devices no such love. One is a Cisco IP 7940, 7960, Polycom something or other, CSipSimple soft phone... none of these work. |
| 07:23.19 | ChannelZ | are any of them local to you such that you could wireshark and see if they are sending a new register off into space? |
| 07:24.09 | atan | Actually, I wonder........ I think you're on to something there ChannelZ. The box does have more than 1 IP let me try connecting the phone to another. |
| 07:24.27 | ChannelZ | Without being able to see anything, it sounds like your new machine is maybe identifying its self as something else such that they are sending their responses elsewhere |
| 07:24.48 | atan | Snap. You've got it buddy! |
| 07:24.58 | ChannelZ | or that, if it's multi-homed and seeing the traffic as coming from a different IP than you're actually listening on |
| 07:25.07 | atan | Okay so here's the deal: asterisk is replying from a different IP. Phone/network is ignoring it. |
| 07:25.16 | atan | Dude I love you so much right now.... |
| 07:25.28 | atan | sends ChannelZ pizza and chocolate |
| 07:25.39 | ChannelZ | did you bind it to the desired IP in sip.conf? |
| 07:26.06 | atan | Nope -- I figured 0.0.0.0 would hit all interfaces, but, not so much it seems |
| 07:26.46 | atan | The phone was connecting to sip.domain.com, right, which let's say resolves to x.x.x.123. However, Asterisk was sending out on x.x.x.124 |
| 07:27.01 | atan | I can't explain why. Not sure that's a bug, or feature. I'll go with feature. |
| 07:27.51 | ChannelZ | well yes it can listen to 'everywhere' but the outgoing traffic is going to route however it routes... is it an aliased interface? |
| 07:29.58 | atan | Yeah |
| 07:30.05 | atan | Which I can' |
| 07:30.47 | atan | s/can'/can't quite explain. But, Asterisk was dealing with things on eth0:2 instead of eth0, but setting the IP in sip.conf resolves this/ |
| 07:33.50 | ChannelZ | Well without specifying an IP I think the behavior is determined by the routing table |
| 07:35.14 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:43.45 | shadebob | R850F here ? |
| 07:43.55 | shadebob | s/ / user |
| 07:51.46 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.12.183) |
| 07:54.32 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 07:58.42 | *** join/#asterisk shadebob (~shadebob@196.206.235.45) |
| 08:11.08 | *** join/#asterisk kami` (~user@unaffiliated/kami-) |
| 08:15.49 | *** join/#asterisk shadebob (~shadebob@196.206.235.45) |
| 08:20.09 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 08:33.43 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 09:03.37 | *** join/#asterisk shadebob (~shadebob@adsl196-45-235-206-196.adsl196-8.iam.net.ma) |
| 09:05.54 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 09:10.01 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.136.154) |
| 09:10.12 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 09:26.58 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.136.154) |
| 09:32.47 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 09:43.53 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 10:00.21 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 10:39.46 | *** join/#asterisk reber (~reber@tsm83-4-78-232-65-13.fbx.proxad.net) |
| 10:41.40 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 10:46.27 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 10:47.31 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 10:47.32 | Ice_Strike | Hey |
| 10:49.52 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 10:55.58 | *** join/#asterisk din3sh (din3sh@41.136.241.218) |
| 10:58.06 | *** join/#asterisk twanny796 (~Twanny@46.11.8.201) |
| 10:58.27 | twanny796 | build breaking on debian |
| 11:07.36 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-tjorouokmegwpuwf) |
| 11:21.10 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 11:27.32 | *** join/#asterisk gusto (~gusto@88.128.55.35) |
| 11:30.21 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 11:45.38 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 11:56.03 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 12:01.29 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.136.154) |
| 12:01.34 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 12:01.52 | kami | Hello. |
| 12:02.38 | WIMPy | Hi kami |
| 12:05.14 | kami | Hi WIMPy, I had to take care of my kids yesterday, so sorry for leaving without notice :) |
| 12:05.43 | kami | You wrote that kernel 3.1.7 solved the problem. What distro/version are you using? |
| 12:06.24 | WIMPy | Both 3.1.7 and 3.4 are working. |
| 12:06.35 | WIMPy | I'm using original kernels only. |
| 12:07.03 | kami | WIMPy: yes, we already talked about the distro kernel patching mania. |
| 12:07.36 | kami | However, I don't know whether I'd run into issues if I upgrade to 3.1 on my lucid box. |
| 12:08.07 | WIMPy | Anything older should be ok as well. |
| 12:09.17 | kami | WIMPy: and did you install the current mISDN master or do you use the kernel misdn modules? |
| 12:10.38 | WIMPy | On the production boxes I use the kernel drivers. |
| 12:10.47 | WIMPy | On the test box I used git. |
| 12:12.42 | kami | So (just to be sure that I understand): the test box has a) HFC USB dongle b) kernel 3.1.7 (or 3.4) and c) git misdn2 + git mISDNuser + git lcr) ? |
| 12:13.24 | WIMPy | Yes. And the same issues as you have. |
| 12:14.00 | kami | Wait: you wrote that going back to kernel 3.1.7 SOLVED the issues, or did I misunderstand? |
| 12:14.19 | WIMPy | Yes, meaning the kernel drives. |
| 12:14.28 | WIMPy | I.e. remove the standalone modules. |
| 12:16.11 | *** join/#asterisk din3sh (din3sh@41.136.241.218) |
| 12:16.11 | *** join/#asterisk SeRi (~wtf@c-98-200-177-50.hsd1.tx.comcast.net) |
| 12:16.11 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:16.11 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:16.11 | *** join/#asterisk CommaCrazy (~CCK@nikola.office.bvcom.net) |
| 12:16.11 | *** join/#asterisk Igneous (~igneous@therealigneo.us) |
| 12:16.12 | *** join/#asterisk vervelak (~vervelak@147.52.2.12) |
| 12:16.12 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 12:16.12 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 12:17.56 | kami | Sorry for being dense. What I now understand is, that this is the working solution: b) kernel 3.1.7 c) misdn2 which comes with kernel 3.1.7 (and NOT the git version) |
| 12:18.17 | WIMPy | Exactely. |
| 12:19.22 | WIMPy | Just leave out the mISDN and use the kernel version. |
| 12:19.22 | WIMPy | The version shouldn't matter. |
| 12:19.22 | kami | WIMPy: thank you. Will do! |
| 12:35.35 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:45.50 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:3d50:bff0:85c:1c01) |
| 12:46.07 | Z_God | hey, anyone here running asterisk on a fritzbox? |
| 12:46.32 | Z_God | I just installed it here, but I'm a bit confused with the capi configuration |
| 12:47.09 | WIMPy | I tried it once. |
| 12:47.18 | WIMPy | And to my astonishment it actually worked. |
| 12:47.35 | Z_God | WIMPy: cool, you have some ideas on the configuration? :) |
| 12:47.54 | Z_God | do I have to configure the internal and external ports separately? |
| 12:48.07 | WIMPy | First of all you need an old kernel. |
| 12:48.12 | WIMPy | yes |
| 12:48.19 | Z_God | also I wonder if this also deals with the DECT & analog parts of the box |
| 12:48.29 | Z_God | why do I need an old kernel? |
| 12:48.50 | Z_God | right now, I attached a 80GB disk to the fritzbox with a Debian Squeeze chroot |
| 12:49.18 | Z_God | in there I have installed asterisk from the repository and at least the capi channel loads |
| 12:49.51 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 12:50.01 | WIMPy | Because the remote capi is too lod. |
| 12:50.12 | WIMPy | Ah, ON the FB? |
| 12:50.19 | Z_God | yes |
| 12:51.24 | WIMPy | ok, so no remote capi. Should help. |
| 12:51.52 | Z_God | I have the kernel from the firmware, which is 2.6.28 I think |
| 12:51.56 | Z_God | it seems to be ok |
| 12:59.44 | Z_God | I just ran capiinfo, it says: Number of Controllers : 1280 |
| 12:59.54 | WIMPy | Err |
| 13:00.01 | WIMPy | Not convincing. |
| 13:00.07 | WIMPy | It should be 5. |
| 13:00.16 | Z_God | they all have Manufacturer: AVM Berlin |
| 13:00.17 | Z_God | ok |
| 13:00.21 | Z_God | but why 5? |
| 13:00.52 | Z_God | the first 5 have Serial Number: 0000000 |
| 13:00.57 | WIMPy | S0 int, S0 ext, FXO, FXS and DECT. |
| 13:01.00 | Z_God | the others have Serial Number: Illegal |
| 13:01.05 | Z_God | ah great :) |
| 13:01.10 | Z_God | is that the correct order as well? |
| 13:01.13 | Z_God | then everything makes sense |
| 13:01.23 | WIMPy | No |
| 13:01.40 | WIMPy | You can identify some by the number of channels. |
| 13:02.19 | Z_God | bchannels? |
| 13:02.27 | Z_God | they are like 512, 256, 1280 |
| 13:03.10 | WIMPy | Something is wrong there. |
| 13:03.54 | Z_God | a bit of googling reveals these are endian bugs |
| 13:04.11 | Z_God | I guess capiinfo wasn't tested much on mips |
| 13:06.51 | Z_God | WIMPy: did you still have to configure anything through the fritzbox webinterface itself or is it better to just empty all the configuration there? |
| 13:11.35 | WIMPy | IIRC it didn't use its own config while CAPI was connected. |
| 13:12.30 | Z_God | ok |
| 13:13.20 | Z_God | it seems there is much more info on this when googling about remote capi, I will see what I can learn from that |
| 13:13.34 | Z_God | that was a good pointer :) |
| 13:24.15 | Z_God | yep, asterisk doesn't have the endian issues, I get 5 controllers with normal channel counts with capi info in asterisk |
| 13:25.43 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 13:49.41 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-219.bois.qwest.net) |
| 13:50.53 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:05.36 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 14:06.04 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 14:07.59 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 14:11.57 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 14:20.13 | *** join/#asterisk puzzled (~patrick@53533D66.cm-6-4a.dynamic.ziggo.nl) |
| 14:33.10 | *** join/#asterisk mazpe (~mazpe@c-174-61-76-85.hsd1.fl.comcast.net) |
| 14:37.17 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:42.04 | mazpe | Anything in particular that will make "cdr show status" display Adaptive ODBC but won't show ODBC |
| 14:43.04 | Z_God | WIMPy: it seems asterisk & the fritzbox own services are still in conflict, at least when using DECT |
| 14:44.00 | WIMPy | I only managed to use the two S0 ports. |
| 14:44.10 | WIMPy | But I didn't try for long. |
| 14:44.30 | Z_God | maybe I need to enable the capi server |
| 14:44.39 | Z_God | I can imagine it might disable the internal stuff |
| 14:45.23 | WIMPy | I think that just makes it listen for remote CAPI. |
| 14:47.22 | Z_God | the problem is that when I dial for instance 1000, I get smth like **500 in my display |
| 14:48.03 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:48.18 | Z_God | and I just get some messages from capi debug in asterisk |
| 14:53.59 | mazpe | or is Adaptive ODBC a better alternative to plain ODBC |
| 14:55.17 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 14:57.16 | *** join/#asterisk gusto (~gusto@88.128.33.3) |
| 14:59.36 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.136.154) |
| 15:01.05 | *** join/#asterisk argief (~johan@d11128.upc-d.chello.nl) |
| 15:01.51 | argief | hi guys, i am struggeling to connect a SIP client from outside my LAN. Any ideas? |
| 15:03.24 | argief | I would like to be able to access my asterisk server from outside. But, it doesnt work? I have a SIP client that I can connect on the lan, but if I put it outside and try to connect I get a Forbidden (Bad Auth) error? |
| 15:04.02 | WIMPy | smells an evil router/IAD. |
| 15:04.07 | argief | I used the same credentials, the only change I made was the server IP |
| 15:04.17 | WIMPy | Unless you specified an IP range in sip.conf. |
| 15:05.34 | argief | i am using freepbx for config. In there, i have specified my local IP as 192.168.0.0/255.255.255.0. but even if i take it out, it gives thes same error |
| 15:06.25 | WIMPy | Are you sure your phone reaches Asterisk? |
| 15:06.43 | WIMPy | For freepbx configuration, better ask in #freepbx. |
| 15:07.22 | argief | ok, thanx guys... i'll go give it s try! |
| 15:10.17 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 15:10.25 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 15:11.33 | argief | Im new to asterisk, but, can I use the CLI to see some live debugging? For my particular problem I should see the auth request comming in and then some explantion for why the auth is rejected. |
| 15:12.36 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 15:13.03 | shadebob | argief : sip set debug on |
| 15:13.06 | WIMPy | 'core set debug 9' and 'core set verbose 9' |
| 15:13.18 | WIMPy | If tha't not enough ... what shadebob said. |
| 15:13.41 | shadebob | hi WIMPy |
| 15:13.57 | argief | ok, I have seen these commands on the net, but how to I get there? telenet 127.0.0.1 5038 ? |
| 15:14.40 | argief | or do I set it in config and restart asterisk? |
| 15:16.42 | JamKo | argief: I would suggest checking out one of the books on Asterisk. It's going to really help you in getting started to understand some basics. Connecting to the CLI is very fundamental in working with Asterisk. |
| 15:16.49 | WIMPy | you start the remote console with rasterisk. |
| 15:18.15 | argief | thx |
| 15:19.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:21.04 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 15:21.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:22.52 | argief | sory.. how can i capture the ourput? It seems to be going quite fast |
| 15:24.31 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 15:28.03 | JamKo | argief: tcpdump with a network dissector might work better for you. You can also limit the debug to a peer with sip setdebug peer <peer name>, which will cut down on the output. |
| 15:28.35 | JamKo | sorry "sip set debug peer" |
| 15:42.17 | WIMPy | No. he can't. |
| 15:42.28 | WIMPy | The peer hasn't registered yet. |
| 15:44.34 | [TK]D-Fender | Get a bgger buffer |
| 15:44.47 | [TK]D-Fender | And he can set the debug to the IP it should come from. |
| 15:47.03 | WIMPy | Yes. But I still suspect the phone doesn;t even reach the Asterisk box. |
| 15:54.26 | hypknight | [TK]D-Fender: if I throw a sip debug up from the server and remote phone perspective, are you willing to take a peek and give me an idea (or rather confirm my suspiscion) as to why the remote endpoints are shown as UNREACHABLE by asterisk, but can make outbound calls? |
| 15:54.42 | hypknight | wow I can't speel |
| 15:54.46 | hypknight | <sigh> |
| 15:55.10 | [TK]D-Fender | hypknight: they go unreachable when they fail to respond to qualify packets (OPTIONS) |
| 15:55.19 | [TK]D-Fender | can be for a number of reasons. |
| 15:55.40 | hypknight | right, i think a firewall in the middle is doing something to the packet, but I was just looking for confirmation |
| 15:55.55 | [TK]D-Fender | You won't see it on the * end |
| 15:56.13 | hypknight | i have sip captures from the * and endpoint |
| 15:56.25 | hypknight | simultaneously |
| 15:57.42 | hypknight | if you're busy it's cool... you just seem pretty well-versed in sip debug |
| 15:58.20 | [TK]D-Fender | Show what you've got and someone will tell you if they see something |
| 16:15.22 | hypknight | Question in general about the option message... if * transmits an OPTION with a public address of * in the from, and then receives it back from the endpoint with a private address of *, it looks like it discards it... is that true? |
| 16:16.02 | WIMPy | How could that reach Asterisk? |
| 16:16.23 | hypknight | if the inbound from the endpoint is being translated by a firewall when it shouldn't be |
| 16:16.27 | [TK]D-Fender | And how would the endpoint know that internal address? |
| 16:16.39 | hypknight | the endpoint doesn't, the firewall on the * side does |
| 16:16.52 | [TK]D-Fender | * should have it set to WAN already, and be viewable as coming fro that same IP |
| 16:17.24 | hypknight | it's not the IP of the ENDPOINT that's being translated by the inbound firewall, that bit is fine |
| 16:17.47 | WIMPy | That sounds seriousely screwed up. |
| 16:17.52 | hypknight | i'm talking about the "FROM" field on the "00 OK" |
| 16:17.57 | hypknight | err "200 OK" |
| 16:18.11 | hypknight | WIMPy: it's really not... the firewall is manipulating the address... |
| 16:18.18 | hypknight | that's what ALGs do all the time. |
| 16:18.39 | WIMPy | Yes, but you don;t want that. |
| 16:19.03 | hypknight | my question is, on the "200 OK" reply from the endpoint, if it has *'s private IP in the FROM (as opposed to the public IP address it submitted) will it discard it (even though the private IP is right)? |
| 16:20.27 | argief | ok, in my debugging I see a 'notice' messages that is interesting: 1. Registration from "name" <sip:110@192.168.0.5> failed for 'device_ip_on_internet' - Wrong password. Yet it is the same secret I use locally and works fine locally? |
| 16:21.28 | JamKo | argief: Do you have your UA set as static or dynamic? |
| 16:21.54 | argief | static |
| 16:22.21 | JamKo | Try dymanic and see if it registers. |
| 16:22.46 | WIMPy | Yes, you can't register unless host=dynamic. |
| 16:23.27 | argief | sory, i thought you guys were talking about external ip. No, my host for the user is set to dynamic |
| 16:25.33 | argief | any other ideas? why would the same password fail from external ip, yet work on a local ip? |
| 16:27.54 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 16:33.49 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 16:36.31 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 16:37.06 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 16:39.14 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:41.55 | hypknight | another question... if I have qualify turned on for an extension, and the OPTIONS thing is failing as I mentioned before, does * not even TRY to send the call to the endpoint? Does it dump it directly to VM? If I turn qualify off, will it try to send it? |
| 16:42.00 | *** join/#asterisk kayfox (~kayfox@xheotris.zerda.net) |
| 16:42.01 | hypknight | (as a temporary fix) |
| 16:43.08 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 16:43.41 | [TK]D-Fender | the point of qualify is as a NAT keep-alive and on failure anlos a way to tell * to not bother for dead hosts. if you want to avoid that simply DISABLE IT and have the device employ its own keep-alive if needed |
| 16:44.56 | [TK]D-Fender | and also* |
| 17:26.43 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 17:34.10 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 17:40.29 | hypknight | sweet.... thanks bud |
| 17:47.43 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 17:48.08 | volga629 | I have some problem with TLS registration tcptls.c: == Problem setting up ssl connection: error:140760FC:SSL routines:SSL23_GET_CLIENT_HELLO:unknown protocol |
| 17:48.23 | volga629 | Is this cert problem or client ? |
| 17:51.39 | volga629 | WARNING[23479] tcptls.c: FILE * open failed! |
| 17:52.55 | SeRi | how canI change this to take double digits? |
| 17:53.00 | SeRi | exten => _*00100[0-9],1,NoOp() |
| 17:53.16 | SeRi | I now have a phone on 1010 and I cant use under that context |
| 17:54.18 | kaldemar | SeRi: elaborate |
| 17:55.33 | SeRi | kaldemar: If I want to activate the inetercome on my polycomes I dial *00 + extension number. which use to range from 0 to 9 so I use exten => _*00100[0-9],1,NoOp() to start my plan. |
| 17:56.17 | SeRi | now I have another phone which goes to the next number 1010 and that plan wont work because it goes from 1000 to 10009 |
| 17:56.40 | SeRi | if I dial *001010 it dials 10010 |
| 17:57.28 | SeRi | 1009* |
| 17:57.58 | kaldemar | show something real to look at. your explanations don't match with the exten. |
| 17:59.21 | SeRi | kaldemar: http://pastebin.com/DBrvciJb |
| 18:00.25 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 18:02.07 | WIMPy | Is it possible to allow SIP users to suppress their Caller ID? |
| 18:02.28 | WIMPy | Without allowing them to send whatever they want, that is. |
| 18:02.29 | kaldemar | *0010[01][0-9] will match *001010 but others too |
| 18:02.56 | [TK]D-Fender | SeR2 patterns |
| 18:03.02 | [TK]D-Fender | SeRi: 2 patterns required |
| 18:03.29 | SeRi | kaldemar: Thanks. |
| 18:03.44 | SeRi | [TK]D-Fender: ah!!! I see now. |
| 18:05.56 | SeRi | [TK]D-Fender and kaldemar Thanks. |
| 18:41.53 | *** join/#asterisk adeel (~adeel@72.53.65.233) |
| 18:45.44 | *** join/#asterisk angryuser_laptop (~angryuser@187-244-135-95.pool.ukrtel.net) |
| 18:48.24 | argief | hi, so this afternoon I posted a question regarding connecting to * from outside local network. The client kept responding with 'Forbidden'. After MANY man hours, I finally got it to authenticate if I leave the password for the extention as nothing. Any ideas? Why would it not be able to authenticate with password from outside local network? |
| 18:52.33 | [TK]D-Fender | argief: And over those hours you haven't shown us configs & debug. |
| 18:54.53 | argief | Still new at this... Only learned from WIMPy and the others today how to do debugging! |
| 18:55.26 | argief | what do you want to see? |
| 18:57.06 | argief | id I set sip set debug on: I get a NOTICE: Registration from "name" <sip:110@192.168.0.5> failed for 'device_ip_on_internet' - Wrong password. |
| 18:58.02 | argief | And the device says Unauthorised / Forbidden / (Bad Auth) |
| 19:17.55 | [TK]D-Fender | ~pb |
| 19:17.55 | 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. |
| 19:17.56 | [TK]D-Fender | ^^ |
| 19:18.13 | [TK]D-Fender | now actually show us the complete communication and the matching configs. |
| 19:18.17 | [TK]D-Fender | mask ONLY the pass |
| 19:39.17 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:55.50 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 20:11.41 | *** join/#asterisk jsjc (~Adium@109.Red-83-45-143.dynamicIP.rima-tde.net) |
| 20:30.45 | *** join/#asterisk imox1234 (~imox1234@g231226007.adsl.alicedsl.de) |
| 20:33.31 | *** join/#asterisk imox1234 (~imox1234@g231226007.adsl.alicedsl.de) |
| 20:36.52 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-084-057-202-003.pools.arcor-ip.net) |
| 20:38.35 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 20:41.36 | fornax | Hi, i try to establish a connection from my asterisk to a cucm using a sip trunk. Calling my asterisk from cucm works but the other way around rises a 404. I think this might be a NAT problem, because the CUCM is behind a router that forwards the port. is it in some way possible to change the host so that it tries to call number@internalIP instead of number@externalIP that is not found by the cucm? |
| 20:57.00 | *** join/#asterisk jackmcbarn (47c77dd2@gateway/web/freenode/ip.71.199.125.210) |
| 20:57.51 | jackmcbarn | WaitExten isn't working if i use it immediately when a call begins. ive ran several tests on this and detailed my results at http://pastebin.com/YXZtwY8i |
| 21:01.33 | *** join/#asterisk wonderworld (~ww@dsdf-4db55dde.pool.mediaWays.net) |
| 21:14.53 | [TK]D-Fender | jackmcbarn: playback(silence/2) and then show us your configs and the actual call attempt. |
| 21:19.09 | jackmcbarn | Playback(silence/2) makes it work |
| 21:19.38 | jackmcbarn | but so does a Wait(0.03) before or after answering |
| 21:19.39 | *** join/#asterisk jbolo (~jonathan@190.43.69.227) |
| 21:19.43 | *** part/#asterisk jbolo (~jonathan@190.43.69.227) |
| 21:22.00 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 21:22.42 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 21:23.19 | volga629 | res_config_ldap.c:1328 update_ldap: Couldn't modify why asterisk will try request modification in ldap server |
| 21:23.43 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-084-057-202-003.pools.arcor-ip.net) |
| 21:24.13 | [TK]D-Fender | jackmcbarn: RTp setup may take a second or two dependsing... |
| 21:27.02 | jackmcbarn | if i call WaitExten immediately, no DTMF ever reads correctly, no matter how long i wait to send it |
| 21:28.05 | *** join/#asterisk shadebob (~shadebob@41.143.16.62) |
| 21:32.37 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 21:36.48 | jackmcbarn | when i set my phone to SIP INFO for dtmf, it works fine with all of them |
| 21:37.45 | [TK]D-Fender | because that is SIP, not RTP |
| 21:38.01 | [TK]D-Fender | SIP is used to negotiate RTP, but the SIP part is already accepted |
| 21:39.41 | jackmcbarn | so the bug happens when i call WaitExten before RTP is set up and the DTMF comes over RTP |
| 21:39.54 | [TK]D-Fender | that'd ddo it |
| 21:57.38 | *** join/#asterisk jsjc (~Adium@109.Red-83-45-143.dynamicIP.rima-tde.net) |
| 22:01.54 | jackmcbarn | found another oddity. with the same config, dialing 3003[send]1000 really fast sends the server crazy for a few seconds. it spams Unable to write to alert pipe on SIP/redacted-00000001 (qlen = 100): Resource temporarily unavailable! |
| 22:01.55 | jackmcbarn | and Failed to write to alertpipe: Resource temporarily unavailable like crazy to the console, and the server seems to hang until its done |
| 22:02.16 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 22:16.48 | *** join/#asterisk jsjc (~Adium@109.Red-83-45-143.dynamicIP.rima-tde.net) |
| 22:19.34 | *** join/#asterisk wonderworld (~ww@dsdf-4db55dde.pool.mediaWays.net) |
| 22:48.01 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 23:01.11 | WIMPy | jackmcbarn: Put a Progress() and/or Proceeding() before WaitExten. |
| 23:04.14 | WIMPy | Or as that's not dialling anyway, Answer() might be more appropriate. |
| 23:08.06 | *** join/#asterisk wonderworld (~ww@dsdf-4db55b7f.pool.mediaWays.net) |
| 23:37.16 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 23:46.43 | jackmcbarn | i already have an Answer() before WaitExten |
| 23:46.47 | jackmcbarn | oh, he left |
| 23:56.12 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |