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| 02:30.26 | *** join/#asterisk SpaceBass (~SP@pool-98-117-75-200.rcmdva.fios.verizon.net) |
| 02:30.27 | SpaceBass | hey folks |
| 02:31.09 | SpaceBass | Does the new google voice funcationality (and chan jabber, etc) compile automatically with 1.8 or do I need to specify a flag with configure? |
| 02:38.02 | russellb | you will need openssl and libiksemel |
| 02:38.19 | russellb | after running configure, run "make menuselect" and ensure that res_jabber and chan_gtalk are selected to be built |
| 02:38.27 | russellb | they will be built by default if their dependencies have been satisfied |
| 02:38.28 | SpaceBass | russellb, thanks! |
| 02:38.32 | russellb | you're welcome. |
| 02:39.37 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 03:01.54 | SpaceBass | ok, last dumb question of the night - whats the new way to do 'sip show peers'? |
| 03:02.48 | russellb | "sip show peers" |
| 03:03.00 | russellb | it hasn't changed. |
| 03:03.14 | SpaceBass | getting: no such command |
| 03:03.22 | russellb | module show like chan_si |
| 03:03.26 | russellb | is it even loaded? |
| 03:03.40 | russellb | *CLI> sip <tab> |
| 03:03.55 | SpaceBass | yeah, I see chan_sip.so |
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| 03:38.02 | SpaceBass | the darndest thing - chan_sip is loaded, the application was built at the time of compile - but I cannot do a sip show peers |
| 03:39.17 | Kobaz | SpaceBass: module load chan_sip.so |
| 03:39.53 | SpaceBass | interesting, thought it was loaded, but that did the trick |
| 03:41.39 | Kobaz | SpaceBass: do you have autoload yes in your modules.conf ? |
| 03:42.09 | SpaceBass | Kobaz, checking - thanks for the tips |
| 03:42.30 | Kobaz | or, do you have a noload chan_sip |
| 03:42.31 | SpaceBass | migrating away from freepbx (been relying on it for years since early A@H days) trying to re-learn the CLI |
| 03:42.36 | Kobaz | either one will make sip not load up |
| 03:43.01 | Kobaz | well.. not having autoload, will make sip not load, if it's not specifically in the list |
| 03:55.33 | Kobaz | this is really really weird |
| 04:09.25 | Kobaz | oh damnit, my chrome tab crashed in the middle of a reviewboard post |
| 04:11.59 | ketas | are linksys atas good for home use? |
| 04:13.11 | p3nguin | Often |
| 04:14.09 | ketas | but what about ip dects |
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| 04:30.18 | SpaceBass | ketas I had a linksys ata that worked quite well before I bricked it |
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| 05:08.03 | p3nguin | Bricks are made for throwing through windows. |
| 05:16.16 | *** join/#asterisk Janky (~Janky@cpe-173-171-233-220.tampabay.res.rr.com) |
| 05:16.27 | Janky | hello |
| 05:30.24 | *** join/#asterisk kidisgod (~danimal@184.77.195.41) |
| 05:30.30 | kidisgod | hello everyone. |
| 05:32.07 | kidisgod | I am using asterisk 1.8 and I am seeing ${EXTEN} containing 4259967033;phone-context=+1;npdi=yes |
| 05:32.20 | kidisgod | this is different behavior from 1.4 and 1.6 |
| 05:33.17 | kidisgod | does anyone know if there is some new configuration setting |
| 05:34.47 | kidisgod | -- Executing [4259967033;phone-context=+1;npdi=yes@360default:5] NoOp("SIP/360trunk-0000000d", "dialed number => 4259967033;phone-context=+1;npdi=yes") in new stack |
| 05:36.30 | kidisgod | the sip peer sending the call is a Sonus_UAC |
| 05:42.01 | SpaceBass | think I got sip working - what about debug, how does one enable sip debug? |
| 05:49.07 | kidisgod | core set verbose 4 |
| 05:49.19 | kidisgod | core set debug 4 |
| 05:49.29 | kidisgod | sip debug peer <name of peer> |
| 05:50.01 | kidisgod | sip set debug peer <name of peer> |
| 05:50.11 | *** join/#asterisk benklop_ (~ben@c-24-8-170-86.hsd1.co.comcast.net) |
| 05:51.01 | benklop_ | hello, i've got asterisk 1.8 installed on debian, i'm new to this, so this is a very simple config |
| 05:51.20 | benklop_ | i've got 2 phones - an ATA, and a softphone on a linux PC. |
| 05:51.46 | benklop_ | i've also got a gtalk account connected for outbound calls over google voice. |
| 05:52.11 | benklop_ | all is working well, but iv'e got bad echo using the ATA, none using the softphone |
| 05:52.22 | benklop_ | how can I enable echo cancellation on the ATA? |
| 05:52.42 | benklop_ | er, on asterisk for that extension, since the ATA doesn't seem to have any echo settings |
| 05:53.38 | benklop_ | echo is only heard on the other end of the call - the phone attached to the ATA does not get an echo |
| 05:55.11 | benklop_ | i'm not finding a whole lot out there written about 1.8 in specific |
| 05:55.38 | kidisgod | is your ata allowed to reinvite |
| 05:55.47 | kidisgod | and is the gtalk allowed to reinvite? |
| 05:56.37 | kidisgod | what codec is being used by the ata |
| 05:56.57 | kidisgod | what codec is being used by the softphone |
| 05:57.20 | benklop_ | ATA is configures to only allow g711u, |
| 05:57.45 | kidisgod | cisco/linksys ata? |
| 05:58.16 | benklop_ | very crappy cheap ATA, soyo |
| 05:58.32 | benklop_ | i don't know about reinvite |
| 05:58.51 | benklop_ | the extensions are all set to use ulaw only |
| 05:59.41 | benklop_ | so I expect the softphone is using ulaw as well |
| 05:59.48 | kidisgod | all the extensions are on the same network segment |
| 05:59.58 | benklop_ | yes |
| 06:02.32 | benklop_ | ill try calling one extension from the other here shortly, I can no longer remember if the echo occurs in that case |
| 06:03.34 | kidisgod | does your machine have a timing source for asterisk |
| 06:03.44 | kidisgod | the dahdi kernel and tools |
| 06:04.32 | benklop_ | i'm not certain, I don't remember setting that up though |
| 06:06.24 | benklop_ | is there a specific kernel module I can look for |
| 06:06.59 | kidisgod | there s hould be some dahdi kernel modules loaded on the system |
| 06:07.10 | benklop_ | ok, none that I see. |
| 06:07.47 | benklop_ | I was under the impression that dahdi was only for connection to local fxos and such directly in the box |
| 06:08.58 | kidisgod | I know it is used for some of the asterisk applications |
| 06:09.05 | kidisgod | like meetme |
| 06:09.16 | benklop_ | okay |
| 06:09.21 | kidisgod | if there is no pstn hardware it loads a dummy timer |
| 06:09.59 | mun24 | any asterisk guru here? |
| 06:10.20 | kidisgod | I use centos |
| 06:10.21 | kidisgod | yum list installed | grep dahdi |
| 06:10.22 | kidisgod | asterisk18-dahdi.x86_64 1.8.0-1_centos5 installed |
| 06:10.22 | kidisgod | dahdi-firmware.noarch 2.0.2-1_centos5 installed |
| 06:10.22 | kidisgod | dahdi-firmware-hx8.noarch 2.06-1_centos5 installed |
| 06:10.22 | kidisgod | dahdi-firmware-oct6114-064.noarch 1.05.01-1_centos5 installed |
| 06:10.22 | kidisgod | dahdi-firmware-oct6114-128.noarch 1.05.01-1_centos5 installed |
| 06:10.26 | kidisgod | dahdi-firmware-tc400m.noarch MR6.12-1_centos5 installed |
| 06:10.28 | kidisgod | dahdi-linux.x86_64 2.4.0-1_centos5 installed |
| 06:10.30 | kidisgod | dahdi-tools.x86_64 2.4.0-1_centos5 installed |
| 06:10.32 | kidisgod | kmod-dahdi-linux.x86_64 2.4.0-1_centos5.2.6.18_194.26.1.el5 installed |
| 06:10.34 | kidisgod | kmod-dahdi-linux-fwload-vpmadt032.x86_64 |
| 06:10.37 | kidisgod | those are the packages installed related to dahdi |
| 06:10.58 | kidisgod | whats up mun24 |
| 06:10.59 | mun24 | I am getting this error message: No match Their Call ID: 8c5ae271ba446f12113d71b43b96810b@sip:100 Their Tag 369530534 Our tag: as2c82bfbb |
| 06:12.11 | benklop_ | i don't believe I have a timing source then, no.. |
| 06:12.21 | benklop_ | no dahdi modules either |
| 06:13.23 | kidisgod | it seems weird the softphone is ok and the ata is not though |
| 06:13.42 | kidisgod | no firewalls involved? |
| 06:14.33 | mun24 | kidisgod: I am getting this error message: No match Their Call ID: 8c5ae271ba446f12113d71b43b96810b@sip:100 Their Tag 369530534 Our tag: as2c82bfbb |
| 06:14.34 | benklop_ | no firewalls that only one would be subjected to |
| 06:16.41 | benklop_ | there is a firewall/router to the internet, and the asterisk server itself is on a different subnet / physical network than the phones, but that's just a normal, non-nat route and since the phones can make calls I think it's configures ok |
| 06:19.16 | *** join/#asterisk carrar (~tim@2604:5000:11:1::3) |
| 06:21.20 | *** join/#asterisk lpmusic (~dballenge@denetronllc-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) |
| 06:21.29 | benklop_ | ok, s owhen placing a call from the ATA to the softphone, i get echos of anythnig I say on the softphone back to me |
| 06:21.49 | benklop_ | but i do not get echos of anything said on the ATA back to it |
| 06:22.34 | benklop_ | i think that the ATA subjects the other end of whatever is connected to it to echo |
| 06:22.50 | benklop_ | whatever it is callnig or whatever is calling it |
| 06:23.12 | benklop_ | and i'm wondering basically if there is a way to do echo cancellation on that |
| 06:23.51 | benklop_ | i don't believe i am allowing reinvite, to answer antoher question |
| 06:24.31 | benklop_ | I guess you asked because if the call gets reinvited, then asterisk is totally out of the loop as far as audio is concerned and there's no way for echo cancellation to happen |
| 06:26.52 | kidisgod | exactly |
| 06:27.05 | kidisgod | asterisk is acting as man in the middle for the call. |
| 06:27.17 | kidisgod | do you have 2 softphones |
| 06:27.23 | kidisgod | to try locally |
| 06:27.44 | kidisgod | or set up an echo extension |
| 06:28.31 | benklop_ | i can do that, but I have no microphone on one of them |
| 06:28.41 | benklop_ | that shouldn't matter I guess |
| 06:28.45 | kidisgod | exten => echotestextension,1,NOOP(====================================>) |
| 06:28.45 | kidisgod | exten => echotestextension,n,NOOP(== Echo Test Extension =>) |
| 06:28.45 | kidisgod | exten => echotestextension,n,NOOP(====================================>) |
| 06:28.45 | kidisgod | exten => echotestextension,n,ExecIf($[${CHANNEL(state)} = Ring]?Answer()) |
| 06:28.45 | kidisgod | exten => echotestextension,n,ExecIf($[${CHANNEL(state)} = Up]?Echo()) |
| 06:28.46 | kidisgod | exten => echotestextension,n,ExecIf($[${CHANNEL(state)} = Up]?Hangup()) |
| 06:28.55 | pabelanger | ~pb |
| 06:28.55 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 06:29.00 | pabelanger | kidisgod: ^^ |
| 06:32.24 | benklop_ | kidisgod: so then ,as I said i'm rather new to this, how do I add an extension to dial to call that? |
| 06:32.52 | kidisgod | what is the context your phones are in. |
| 06:33.06 | carrar | ~book |
| 06:33.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 06:33.13 | carrar | read that |
| 06:33.44 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 06:33.51 | benklop_ | carrar: thanks, I'll definitely pick up a copy. I assume it's not about 1.8, so the changes are pretty easy to sort out? |
| 06:34.41 | carrar | it applies to your question |
| 06:34.58 | benklop_ | ok |
| 06:35.36 | benklop_ | kidisgod: so, i'm honestly not completely sure i have my contexts set up in a straightforward way.. |
| 06:35.38 | benklop_ | http://pastebin.ca/1997697 |
| 06:35.44 | benklop_ | that's my extensions.conf |
| 06:36.06 | benklop_ | pretty simple, copied from a few places with some changes |
| 06:36.19 | carrar | http://downloads.oreilly.com/books/9780596510480.pdf |
| 06:36.23 | carrar | download it down |
| 06:36.26 | carrar | skim it over |
| 06:36.31 | carrar | err downlad it now |
| 06:36.40 | kidisgod | add the echo test extension to the context stations |
| 06:37.01 | benklop_ | carrar: i've got it open now |
| 06:37.03 | kidisgod | change the name echotestextension to something like 300 |
| 06:37.12 | carrar | change "echotestextension" to a number you can dial |
| 06:37.13 | benklop_ | ok, that's kinda what I thought |
| 06:37.17 | kidisgod | then call 300 from your phones |
| 06:37.32 | kidisgod | or you can goto extensiontest,1 |
| 06:39.50 | benklop_ | so I can send a hup to asterisk to make it reload this right? |
| 06:40.05 | carrar | dialpla reload |
| 06:40.09 | carrar | dialplan reload |
| 06:40.19 | carrar | from the CLI |
| 06:40.35 | benklop_ | cool ok |
| 06:41.09 | benklop_ | well that works fine |
| 06:41.26 | carrar | see page 72 |
| 06:41.27 | benklop_ | i'm not totally sure what to expect if it is not worknig though |
| 06:42.17 | benklop_ | i mean, i'm getting the sort of echo I would expect since it is supposed to be echoing it to me |
| 06:42.20 | carrar | or 73 |
| 06:42.25 | benklop_ | from the softphone |
| 06:42.35 | benklop_ | i'll try it from the ATA |
| 06:43.14 | benklop_ | ok, so from the ATA it's like shouting into a tunnel |
| 06:43.17 | carrar | dump the ATA |
| 06:43.19 | benklop_ | multiple echos |
| 06:43.54 | kidisgod | on the ata can you adjust the txgain and rxgain |
| 06:44.24 | kidisgod | or are you using a speaker phone connected to the ata? |
| 06:45.01 | benklop_ | no, just a normal phone that when plugged into a pots line works as you would expect with no noticable echo |
| 06:45.19 | benklop_ | but it does sound a bit like it might from a poor speakerphone |
| 06:46.26 | carrar | read page 200 |
| 06:47.49 | benklop_ | thanks |
| 06:47.54 | benklop_ | reading now |
| 06:49.04 | *** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au) |
| 06:49.16 | benklop_ | so essentially, just because this phone does not cause echo on a pots circuit does not mean it may not cause echo in this setup |
| 06:51.20 | carrar | like kidisgod suggested, you might try lower some of the volumn settings |
| 06:51.49 | carrar | make sure you don't have much for latency |
| 06:51.50 | benklop_ | I don't see any mention of how to deal with echo on a sip phone, only on zaptel interfaces |
| 06:52.12 | kidisgod | is there a web page to set up the ata? |
| 06:52.17 | benklop_ | yes |
| 06:52.22 | kidisgod | look in there |
| 06:52.31 | kidisgod | for something about txgain or rxgain |
| 06:52.31 | benklop_ | it is just s simple page with a billion text fields for settings |
| 06:52.38 | benklop_ | as I said, a cheap ATA |
| 06:52.52 | carrar | might have to read it |
| 06:52.56 | kidisgod | I have not used the soyo ata's |
| 06:53.01 | kidisgod | but I have used many others. |
| 06:53.14 | kidisgod | there was some way to always get them to work |
| 06:53.18 | kidisgod | or work around |
| 06:53.31 | benklop_ | ok |
| 06:53.55 | kidisgod | I have had the most issues with the grandstreams |
| 06:54.22 | kidisgod | the least issues with the linksys/cisco 2102 |
| 06:54.44 | benklop_ | ok, so I have a "handset out" and a "handset in", a "jitter size" |
| 06:55.03 | kidisgod | is there a setting for sip rtp packet size? |
| 06:55.13 | kidisgod | .020 for asterisk |
| 06:55.58 | benklop_ | and a few checkboxes like vad, agc, and aec |
| 06:56.08 | benklop_ | and an audio frames setting |
| 06:56.26 | benklop_ | no rtp packet size, but maybe audio frames is the same setting? |
| 06:56.38 | benklop_ | currently set to 2 |
| 06:56.45 | kidisgod | try to change the agc setting and see if it makes a difference |
| 06:57.09 | benklop_ | aec (which I am guessing is auto echo cancellation) is checkedm as are the others |
| 06:57.13 | benklop_ | ok, i'll try that |
| 06:58.04 | benklop_ | the darn thing reboots every time you change settings |
| 06:58.10 | benklop_ | in order to apply them |
| 06:58.11 | kidisgod | I am thinking it is auto gain control |
| 06:58.19 | kidisgod | that is how most sip phones / ata's are |
| 06:58.24 | benklop_ | yeah, that would make sense |
| 06:58.29 | benklop_ | why would that cause echo though? |
| 06:58.35 | kidisgod | polycom's take a long time to boot. |
| 06:58.44 | kidisgod | and some cisco phones |
| 06:59.50 | kidisgod | echo can come from a lot of places from an ata |
| 07:00.03 | benklop_ | hmm. Well, if no tweaking can fix this, I may eventually buy a linksys ATA |
| 07:00.08 | benklop_ | agc did not fix it |
| 07:00.13 | kidisgod | is it a full or half duplex network adapter |
| 07:00.21 | benklop_ | i'll try playing arouns a bit here |
| 07:00.40 | benklop_ | i think it is connecting at 100 full |
| 07:01.17 | benklop_ | not sure though, why? |
| 07:01.45 | kidisgod | one of may things that may introduce echo |
| 07:02.15 | kidisgod | usually the first thing I look at is the gain on the ata |
| 07:02.27 | benklop_ | in or out? |
| 07:02.43 | benklop_ | if i turn off "aec" it gets worse.. so that's something :) |
| 07:03.20 | kidisgod | maybe try to change the jitterbuffer settings for the sip peer in asterisk |
| 07:05.04 | benklop_ | well, handset out volume is set rather high (30 out of 31) and hanset in set lowish (8) |
| 07:05.22 | benklop_ | so i turned down handset out and the echo got better but is not totally gone |
| 07:05.32 | benklop_ | but of course is now a bit quieter |
| 07:05.59 | kidisgod | so maybe the audio echo is being generated not by the sip call? |
| 07:06.38 | benklop_ | i have 2 handsets plugged into this ATA, i'll try another one |
| 07:09.22 | *** join/#asterisk ElvisR (~elvis@horace.dionysian-mind.net) |
| 07:11.24 | benklop_ | well the echo is definitely still there |
| 07:11.56 | benklop_ | it's much easier to intenify when calling something with more latency though |
| 07:13.26 | benklop_ | kidisgod: would you say that getting another ATA, for instance a cisco, would probably have less of a chance of having echo? |
| 07:13.40 | benklop_ | or is this something I'll probably encounter no matter the ATA? |
| 07:14.17 | kidisgod | the echo is not there on the same call from the softphone? |
| 07:14.26 | benklop_ | correct |
| 07:15.03 | kidisgod | have you tried the ata to some other sip setup? |
| 07:16.05 | carrar | skip asterisk phones and use a megaphone! |
| 07:16.13 | benklop_ | heh |
| 07:16.34 | benklop_ | you mean like a sip provider such as sipgate? |
| 07:17.09 | benklop_ | I have not, but that would certainly prove its a problem with the ATA |
| 07:17.10 | kidisgod | yeah anywhere |
| 07:17.23 | kidisgod | or some local sip pbx like 3cx |
| 07:17.46 | benklop_ | i'm pretty sure it is anyway, I just thought asterisk had the ability to do echo cancellation on it |
| 07:18.26 | benklop_ | I have a sipgate account I could set it up with |
| 07:19.20 | kidisgod | just to test the ata |
| 07:20.15 | benklop_ | yeah, I can do that |
| 07:20.31 | CaneToad | I'm new with asterisk, but have got it to the point where two ip phones (extensions 1 and 2) are successfully registering with asterisk and can call each other, and asterisk is also registering with an external sip server. Invite requests from the external sip server are arriving, but my asterisk is telling them 401 Unauthorized. Where should I be looking to fix this? |
| 07:21.57 | benklop_ | kidisgod: thanks for the help |
| 07:22.09 | kidisgod | anytime |
| 07:22.28 | benklop_ | carrar: thanks for pointing out the book |
| 07:22.50 | benklop_ | any idea if there will be a new edition since 1.8 is the new LTS? |
| 07:23.13 | carrar | probably |
| 07:23.15 | kidisgod | CaneToad is the external sip server a pstn gateway |
| 07:23.26 | carrar | books take time |
| 07:23.56 | CaneToad | kidisgod: yes...my ATA can successfully directly register with it and make PSTN calls |
| 07:24.16 | kidisgod | in your sip.conf |
| 07:24.28 | kidisgod | for the sip peer for the pstn gateway |
| 07:24.35 | kidisgod | how is the canreinvite set |
| 07:24.58 | CaneToad | set to no |
| 07:25.55 | kidisgod | can you sip show peers and see the peer online? |
| 07:27.38 | CaneToad | The peer is online, I have another ATA using a different number and udp port that works on it right now. |
| 07:28.06 | kidisgod | so you are not getting inbound calls |
| 07:28.14 | kidisgod | or outbound calls from the asterisk? |
| 07:28.53 | CaneToad | on an inbound invite from the peer, asterisk is telling it 401 Unauthorized |
| 07:29.22 | kidisgod | it wants to authenticate |
| 07:29.42 | kidisgod | what is the insecure setting for the peer |
| 07:29.49 | kidisgod | in sip.conf |
| 07:30.16 | CaneToad | hmm it has none |
| 07:30.31 | kidisgod | set it to no |
| 07:30.46 | kidisgod | reload chan_sip.so |
| 07:30.50 | kidisgod | to apply the changes |
| 07:32.28 | CaneToad | if I set insecure=no, does that mean it will try to use secure comms? |
| 07:33.05 | CaneToad | at this point I want everything un-encrypted so I can see what is going on when snooping |
| 07:33.29 | kidisgod | it will not try to authenticate the sip sessions for the peer |
| 07:34.04 | kidisgod | usually used with ip based auth |
| 07:35.52 | kaldemar | CaneToad: the insecure parameter has nothing to do with secure communications, it defines if the peer must authenticate to make a call. |
| 07:36.37 | CaneToad | ok I completely restarted asterisk after setting insecure=no and my asterisk is still sending unauthorized |
| 07:37.53 | kidisgod | the sip peer is originating and the asterisk is terminating the call |
| 07:37.59 | CaneToad | yes |
| 07:38.32 | kidisgod | and when the sip peer originates the call with the invite asterisk is responding with a 401 unauthorized |
| 07:38.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 07:38.51 | CaneToad | yes |
| 07:39.56 | CaneToad | see capture here: http://www.aaa.net.au/campbell/fails.png |
| 07:41.25 | kidisgod | 135.10 is the asterisk |
| 07:41.34 | CaneToad | yes |
| 07:43.18 | kidisgod | if you sip show peer <peername> |
| 07:43.27 | kidisgod | does it show the insecure setting |
| 07:43.47 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 07:44.34 | CaneToad | How do I get to the prompt to enter that ship show peer ? |
| 07:44.39 | CaneToad | sip |
| 07:44.49 | CaneToad | on the cmdline |
| 07:45.38 | carrar | read the readme file |
| 07:46.00 | CaneToad | I'm in an embeeded device (dd-wrt) |
| 07:46.11 | CaneToad | embedded |
| 07:46.20 | CaneToad | I'll go looking for the readme |
| 07:46.24 | kidisgod | not sure |
| 07:50.11 | CaneToad | ah ok got the commandline now, needed to run asterisk -vvvvr |
| 07:52.39 | kidisgod | who can I chat with about a possible regession in asterisk |
| 07:52.52 | kidisgod | bug #7761 |
| 07:55.58 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 07:58.19 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-180.mia.bellsouth.net) |
| 07:58.43 | CaneToad | kidisgod: you can see the full output of sip show peer here: http://fpaste.org/Qpze/ |
| 08:00.35 | CaneToad | and yes you can see insecure is no |
| 08:00.56 | CaneToad | shouldn't we set insecure=yes to disable security? |
| 08:04.19 | kidisgod | insecure=invite |
| 08:04.54 | kidisgod | when the call comes in do you see anything on the console |
| 08:06.50 | CaneToad | just this: |
| 08:06.51 | CaneToad | <PROTECTED> |
| 08:07.34 | kidisgod | core set verbose 4 |
| 08:07.42 | kidisgod | core set debug 4 |
| 08:07.46 | kidisgod | and try again |
| 08:08.41 | CaneToad | router*CLI> core set verbose 4 |
| 08:08.41 | CaneToad | Verbosity is at least 4 |
| 08:08.41 | CaneToad | router*CLI> core set debug 4 |
| 08:08.41 | CaneToad | Core debug was 0 and is now 4 |
| 08:08.52 | CaneToad | made the call |
| 08:08.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:08.57 | CaneToad | got absolutely identical output |
| 08:09.04 | CaneToad | <PROTECTED> |
| 08:13.45 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 08:20.33 | *** join/#asterisk ElvisR (~elvis@horace.dionysian-mind.net) |
| 08:23.56 | joako | What causes the error FXO PCI Bus Master Abort? |
| 08:28.07 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 08:49.55 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 08:55.47 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 08:56.51 | v1s | ok I was told by digium to check my installed version asterisk against the version of asterk-dev can some one tell me how to do that on deb? |
| 09:17.53 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
| 09:18.00 | ik_5 | hello |
| 09:19.34 | ik_5 | I'm using chan_local for Originate, but when it arrive to Dial command it acts like it's not blocking and continue using the dialplan. I'm using the dial with /n and no answer, and can't figure out what's wrong |
| 09:19.48 | ik_5 | Asterisk 1.4.37 from digium repositories |
| 09:30.41 | *** part/#asterisk v1s (~v1s@202.84.107.67) |
| 09:46.37 | *** join/#asterisk ElvisR (~elvis@horace.dionysian-mind.net) |
| 10:17.17 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 10:24.18 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 10:45.40 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 10:53.40 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 11:06.47 | *** join/#asterisk TimeRider (~steve@5ac31891.bb.sky.com) |
| 11:12.35 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 11:16.47 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 11:28.40 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 11:39.25 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 11:39.29 | [sr] | howdy people |
| 11:41.10 | JerJer | yawns |
| 11:43.31 | [sr] | night is for sleep |
| 11:43.32 | [sr] | :p |
| 11:51.49 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 12:48.33 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 12:56.18 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:59.30 | *** join/#asterisk oneman (~oneman@pool-173-79-250-133.washdc.fios.verizon.net) |
| 13:17.19 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 13:27.12 | *** join/#asterisk johnnygreek (~john@79.107.159.25) |
| 13:27.16 | johnnygreek | hello everyone |
| 13:27.20 | johnnygreek | greetings from greece |
| 13:27.30 | *** join/#asterisk cusco (~trilili@57.218.108.93.rev.vodafone.pt) |
| 13:28.23 | johnnygreek | is anybody here? |
| 13:30.14 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 13:30.34 | johnnygreek | anybody here? |
| 13:30.56 | z4nD4R | hi all, i need interconnect asterisk with sipxec ... somebody any tutorials? |
| 13:31.54 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 13:32.47 | johnnygreek | here you go http://sipx-wiki.calivia.com/index.php/HowTo_interconnect_sipXecs_IP_PBX_with_the_Asterisk_PBX |
| 13:32.53 | johnnygreek | a little googling wont hurt |
| 13:34.27 | z4nD4R | i tryit... but..this is out of date... |
| 13:36.06 | johnnygreek | sorry then, i can't help |
| 13:36.29 | johnnygreek | does anybody have 1.8 here? |
| 13:37.27 | z4nD4R | i have 1.8 |
| 13:38.00 | johnnygreek | i am trying to figure out wether the attended transfer caller id issue was solved |
| 13:38.05 | johnnygreek | do you know what i am talking about ? |
| 13:38.48 | z4nD4R | sry but no :D |
| 13:39.15 | johnnygreek | when someone calls you |
| 13:39.19 | johnnygreek | and you see the caller id |
| 13:39.24 | johnnygreek | then you transfer the call |
| 13:39.26 | johnnygreek | to another extension |
| 13:39.37 | johnnygreek | but the other extensions see's your number not the caller's number |
| 13:39.42 | johnnygreek | as caller id |
| 13:40.20 | johnnygreek | for example if you pickup a call fro number 123456 , your extension is 11 and you transfer to extension 12, then extension 12 will see 11 as caller id |
| 13:40.27 | johnnygreek | legacy pbx transfer the caller id |
| 13:40.40 | johnnygreek | and it was told that it would be resolved on asterisk 1.8 |
| 13:40.42 | z4nD4R | oh.. i am undterstand but... i dont how fix it |
| 13:43.27 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 13:45.57 | Gugge | johnnygreek: for attended transfer i would expeect to see 11, as 11 is the person in the other end when 12 picks up |
| 13:46.11 | Gugge | for unattended i would expect 123456 |
| 13:46.29 | johnnygreek | yes but most legacy pbx do transfer caller id on attended transfer |
| 13:46.34 | johnnygreek | and somehow it's more convenient |
| 13:46.55 | johnnygreek | everyone said that asterisk 1.8 would support it |
| 13:46.56 | Gugge | sure, 12 picks up and says "hey honey" but its a coworker in the other end ... sounds convenient :P |
| 13:47.14 | johnnygreek | my honey calls my mobile :p |
| 13:47.21 | Gugge | you get the idea :) |
| 13:47.24 | johnnygreek | i get the idea |
| 13:47.42 | johnnygreek | but somehow that's how my workers were used to |
| 13:47.47 | johnnygreek | with the legacy panasonic |
| 13:48.10 | johnnygreek | and when you change things the least wanted thing is to hear them bitch about it |
| 13:48.40 | johnnygreek | and since asterisk 1.4 |
| 13:48.42 | johnnygreek | i see on forums |
| 13:48.53 | johnnygreek | that this thing would be covered on asterisk 1.8 |
| 13:49.38 | Gugge | i guess you could set some variables before you dial, and then set the transfer context to a context where you use that variable to set the callerid |
| 13:52.53 | johnnygreek | hmmmm my knowledge is very limited to do that :p |
| 13:53.11 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 14:00.38 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 14:01.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 14:01.39 | z4nD4R | if i want interconnect asterisk and Sipxec, is the same conf. on asterisk side, when i interconnect 2 asterisk servers? |
| 14:04.26 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 14:15.10 | *** join/#asterisk wesphillips (~wphillips@adsl-75-53-137-6.dsl.hstntx.sbcglobal.net) |
| 14:15.46 | johnnygreek | dude |
| 14:19.01 | *** join/#asterisk jlewisat1 (~jlewis@solo.atlantic.net) |
| 14:20.19 | jlewisat1 | quiet this morning...everyone sleeping? |
| 14:20.51 | z4nD4R | morning? with timezone? :D |
| 14:21.54 | jlewisat1 | morning here |
| 14:22.32 | jlewisat1 | I've got a really weird issue...technically, not an asterisk problem, but a telco one...looking for anyone who might be able to offer a clue |
| 14:22.32 | z4nD4R | ;) |
| 14:23.05 | jlewisat1 | we've had this problem on and off (nights and early mornings) since last Thursday, and the telco has been clueless |
| 14:23.28 | cusco | hi.. |
| 14:23.39 | cusco | with any variable I can use the :offset right? |
| 14:24.07 | WIMPy | yes |
| 14:24.42 | cusco | and it is ${var:start:end}, right? |
| 14:24.53 | cusco | (can't open http://www.voip-info.org/wiki/view/Asterisk+variables for some reason) |
| 14:24.54 | leifmadsen | ${var:offset:length} |
| 14:25.00 | leifmadsen | cusco: ^^^ |
| 14:25.06 | cusco | ah! |
| 14:25.12 | cusco | kk thanks |
| 14:25.23 | jlewisat1 | if you call our 800 number, it sounds like hell. if you call the DID the 800 number points to, it sounds fine |
| 14:25.48 | leifmadsen | sounds like a possible problem with the provider then assuming it goes the same route to your system |
| 14:26.13 | jlewisat1 | that's what we decided last Thurs...but they've been unable to find the problem |
| 14:26.27 | jlewisat1 | at some point during the day, it clears up, but then reverts in the evening |
| 14:26.29 | leifmadsen | doesn't sound like an asterisk issue to me |
| 14:27.03 | jlewisat1 | I verified it wasn't asterisk before realizing it didn't happen on the DID...replaced the T1 card, moved the PRI to another server, problem persisted |
| 14:27.17 | jlewisat1 | anyone care to call it and listen to the noise? |
| 14:27.22 | cusco | leifmadsen: and should I use ${var:0:5} or ${var::5} ? |
| 14:27.23 | leifmadsen | still doesn't sound like an asterisk issue ;) |
| 14:27.31 | leifmadsen | cusco: I'd use the 0 |
| 14:27.37 | leifmadsen | just to make it clear it's not a typo if anything |
| 14:28.43 | cusco | and finally, comparing strings in AEL, should be if ( ${var} == "string" ) ? or "${var}" == "string" |
| 14:28.48 | cusco | or only one = |
| 14:29.19 | leifmadsen | no idea, I don't use AEL |
| 14:30.03 | cusco | sorry |
| 14:30.06 | cusco | what do you use? |
| 14:30.16 | cusco | lua? |
| 14:31.17 | jlewisat1 | it starts out with a tone that actually sounds vaguely familiar...and then its all staticy |
| 14:31.32 | leifmadsen | cusco: I use extensions.conf ... |
| 14:31.43 | leifmadsen | shocking! |
| 14:32.28 | cusco | a not related question... |
| 14:33.01 | cusco | problems with echo on one gateway presists. I changed echocanceler to oslec but the echo presists |
| 14:33.05 | cusco | only from landlines |
| 14:33.24 | cusco | txgain and rxgain in dadhdi conf is 0 |
| 14:33.51 | cusco | it has been a problem for a while and I don't know what do do about it |
| 14:34.06 | cusco | buying a card with echocanceler is too expensive.. |
| 14:34.22 | leifmadsen | it sure is |
| 14:34.25 | cusco | why doesn't it work like the other asterisk gateway? |
| 14:34.28 | leifmadsen | but it's the only sure fire way to solve that problem |
| 14:34.54 | cusco | see, we have two asterisk gateways.. phisically distant |
| 14:34.56 | leifmadsen | you can try enabling software echo cancelling (as you've done) but beyond that, I'm not sure what else to suggest |
| 14:35.09 | leifmadsen | yes, different lines and different line lenghts |
| 14:35.14 | leifmadsen | anyways, I'm outta here. I should be cleaning. |
| 14:35.17 | cusco | the default echocanceler (software) works fine on the first one |
| 14:35.23 | cusco | ok |
| 14:35.26 | cusco | thanks lanning |
| 14:40.23 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 14:45.46 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 14:48.02 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.24.142.37) |
| 14:49.08 | coppice | if OSLEC doesn't work for you, why would you expect a hardware echo canceller to perform any better? |
| 14:51.39 | Scorcerer | in the book "Asterisk: the future of telephony" i read that Asterisk doesn't have to use DSP. my question is: can Asterisk make use of internal DSP (like those on sound card) or it there is just no code that would make use of it ? |
| 14:52.16 | WIMPy | There is no code. |
| 14:52.29 | WIMPy | But what sound card has a DSP? |
| 14:52.29 | Scorcerer | thanks |
| 14:53.16 | Scorcerer | dunno, i theoretized that it's sound card, so it should have sound processor, but guess i was wrong on that too :) |
| 14:54.16 | WIMPy | That should be a very high end one then. Usually it isn't much more than a FIFO. |
| 14:54.24 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 14:54.31 | WIMPy | Like almost any interface. |
| 14:54.31 | *** part/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 14:55.11 | Scorcerer | and how about any cards compatible with asterisk ? i know that Zaptel doesn't have any DSPs. Is it true with any other cards ? |
| 14:55.40 | WIMPy | There are cards with DSP available from different vendors. |
| 14:56.00 | WIMPy | And there are software ones as well. |
| 14:56.28 | WIMPy | Oh, and zaptel has been replaced by dahdi some time ago. |
| 14:58.15 | Scorcerer | ok, thanks :) |
| 15:08.02 | GhOnDiE | im having a problem with my dahdi setup |
| 15:08.09 | GhOnDiE | or the hardware im not sure |
| 15:08.27 | GhOnDiE | no audio if i dial out on the trunk |
| 15:08.32 | GhOnDiE | but inbound works fine |
| 15:08.41 | GhOnDiE | using an openvox a400p |
| 15:08.59 | GhOnDiE | has been working fine but just stopped after the server was reinstalled |
| 15:11.25 | cusco | pri debug span? |
| 15:12.01 | GhOnDiE | its a standard analogue line |
| 15:12.11 | GhOnDiE | card has fxo and fxs port |
| 15:12.30 | cusco | sorry never used one of those |
| 15:12.34 | GhOnDiE | if i spy on the channel though i have inbound audio |
| 15:12.39 | cusco | o.O |
| 15:12.48 | cusco | the problem is probably in sip |
| 15:12.51 | GhOnDiE | it seems like a codec problem maybe |
| 15:13.06 | cusco | check nat, carnreinvite/directmedia |
| 15:13.32 | cusco | what does asterisk cli say during the call? |
| 15:13.54 | GhOnDiE | all the normal that you would expect to see |
| 15:14.00 | cusco | no warnigs? |
| 15:14.03 | GhOnDiE | nope |
| 15:14.05 | cusco | verbose 15 |
| 15:14.10 | GhOnDiE | im on 17 |
| 15:14.13 | GhOnDiE | and debug 17 |
| 15:14.13 | cusco | lol |
| 15:14.39 | cusco | is the sip phone on the same network as asterisk? |
| 15:14.48 | GhOnDiE | yeah |
| 15:14.51 | GhOnDiE | all local |
| 15:14.57 | cusco | you have nat = no then, right? |
| 15:15.00 | GhOnDiE | tried sip, iax, and sccp |
| 15:15.09 | GhOnDiE | yeah set to no |
| 15:15.19 | cusco | try setting it to yes |
| 15:15.21 | GhOnDiE | ok |
| 15:15.24 | cusco | also try canreinvite=no |
| 15:15.27 | cusco | and directmedia=no |
| 15:15.37 | cusco | (I really don't know..) |
| 15:15.56 | GhOnDiE | im not using sip for my phones anyway |
| 15:16.02 | GhOnDiE | im suspecting hardware fault |
| 15:20.53 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 15:22.03 | GhOnDiE | think i may try a recompile |
| 15:22.13 | GhOnDiE | just incase its a codec problem or something |
| 15:25.39 | SpaceBass | hey folks |
| 15:26.14 | SpaceBass | I migrated to a new server box lastnight, same configs, same version (1.8) - my sip trunks are all showing "request sent" |
| 15:26.24 | SpaceBass | the IP is the same as the old box, so its not a NAT issue... |
| 15:26.25 | SpaceBass | any ideas? |
| 15:33.10 | *** join/#asterisk MiserySoft (~LND@host81-139-123-63.in-addr.btopenworld.com) |
| 15:33.45 | chuckf | I'm assuming the old box is offline at this point |
| 15:34.02 | chuckf | My first guess would be a firewall issue |
| 15:35.06 | SpaceBass | chuckf, I can bring the old box back on line (booting it now) |
| 15:35.30 | SpaceBass | chuckf, I would have thought firewall too - but the new box has the old box's IP and is not running a firewall |
| 15:35.36 | SpaceBass | I did migrate from Centos to Ubuntu |
| 15:36.34 | chuckf | well you can only have one or the other up with the same ip address. |
| 15:36.49 | SpaceBass | changed the old box to a new IP |
| 15:37.10 | SpaceBass | newbox = .45 oldbox =.46 (oldbox was .45 while in production) |
| 15:39.10 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207.dsl.dyn.forthnet.gr) |
| 15:39.20 | SpaceBass | would something like a host name change matter? |
| 15:39.34 | chuckf | it shouldn't under most conditions |
| 15:39.57 | SpaceBass | I've got externip and locanet set on the new box |
| 15:40.03 | chuckf | when you do a netstat -an|grep 5060 on the new box is it showing as open? |
| 15:40.18 | SpaceBass | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 15:41.27 | SpaceBass | good troubleshooting trick - thanks, hadn't thought to check that before - but doesnt appear to be the problem |
| 15:42.19 | chuckf | it should have just kinda come back up. |
| 15:43.20 | chuckf | I'm just trying to think of what else might be in the network causing an issue |
| 15:43.28 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 15:46.07 | SpaceBass | phones are working and I can call out - so chan_sip must be working properly |
| 15:49.00 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 15:49.16 | *** part/#asterisk MiserySoft (~LND@host81-139-123-63.in-addr.btopenworld.com) |
| 15:51.34 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:53.29 | SpaceBass | think I see the problem |
| 15:53.55 | SpaceBass | on the old box, registrations are sent with my LAN ip, on the new box they are 127.0.0.1 |
| 15:54.21 | chuckf | that's a problem |
| 15:54.27 | SpaceBass | what determines that? |
| 15:54.50 | chuckf | not sure, I've not seen that before |
| 15:55.14 | WIMPy | Usually routing. Does your resolver work? |
| 15:55.48 | SpaceBass | yes |
| 15:56.21 | WIMPy | So it is trying to send to the correct IP, but from 127.0.0.1? |
| 15:56.35 | SpaceBass | yes |
| 15:56.59 | WIMPy | Sounds interesting. |
| 15:57.06 | SpaceBass | just changed /etc/hosts to reflect the correct IP and hostname |
| 15:57.29 | SpaceBass | the registration messages in debugmode look right, but stil not registering |
| 15:58.04 | WIMPy | time for tcpdump, I guess. |
| 15:58.29 | SpaceBass | just confirmed that both hosts (old box and new) are resolving the sip proxies the same |
| 15:58.37 | SpaceBass | exit |
| 15:58.43 | SpaceBass | oops :D wrong window |
| 16:00.44 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:14.40 | *** join/#asterisk SuD (~alex@89.7.113.215) |
| 16:15.11 | [sr] | howdy WIMPy |
| 16:15.24 | SpaceBass | hummm its like there's no traffic coming back from the proxies - nothing confirming or rejecting the registrations |
| 16:20.22 | *** join/#asterisk MiserySoft (~LND@host81-139-123-63.in-addr.btopenworld.com) |
| 16:29.37 | *** join/#asterisk [netman] (~netman@83.55.243.112) |
| 16:31.10 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
| 16:35.39 | WIMPy | Salve [sr]! |
| 16:43.58 | *** join/#asterisk mpe_ (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:45.39 | [sr] | :) |
| 16:47.41 | *** part/#asterisk wesphillips (~wphillips@adsl-75-53-137-6.dsl.hstntx.sbcglobal.net) |
| 16:50.27 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 16:50.47 | SpaceBass | hell if I can figure this out - the tcpdumps are identical - one box registers, the other doesnt |
| 17:02.17 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 17:02.50 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 17:04.08 | *** join/#asterisk RoyK (~roy@cE6BABF51.dhcp.bluecom.no) |
| 17:05.58 | *** join/#asterisk RichardLynch (~Richard_L@c-67-162-103-133.hsd1.il.comcast.net) |
| 17:07.47 | RichardLynch | I'm told that the internal "database get" is MUCH faster in an AGI script. True or False? And, if so, any pointer to Docs about bulk transfer/sync of MySQL MyISAM table into (and out of) said database? Cuz right now, the table definitions don't even match up... :-? |
| 17:12.15 | *** part/#asterisk RoyK (~roy@cE6BABF51.dhcp.bluecom.no) |
| 17:15.22 | SpaceBass | think im closer, my firewall is permitting 5061 - secure sip - but my host doesnt seem to be listening on that port |
| 17:16.15 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 17:22.54 | p3nguin | Any particular reason you're not using the standard port? |
| 17:23.57 | SpaceBass | p3nguin, Im using 5060 too - thought both might be required, turns out not to be the case |
| 17:24.11 | SpaceBass | as my box has randomly decided to start working |
| 17:25.14 | p3nguin | I'm not so sure that using the correct port is random. |
| 17:25.35 | SpaceBass | I had been using 5060 all along |
| 17:25.40 | p3nguin | oh, okay |
| 17:27.38 | SpaceBass | exit |
| 17:29.34 | *** join/#asterisk andyoutside (6161c46a@gateway/web/freenode/ip.97.97.196.106) |
| 17:30.13 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 17:30.31 | SpaceBass | oops wrong window :D |
| 17:32.02 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:22.10 | *** join/#asterisk timholum1 (~timholum3@75-121-142-165.dyn.centurytel.net) |
| 18:26.32 | timholum1 | is voip-info.org down or is it just with my isp |
| 18:27.19 | coppice | its refusing connexions |
| 18:28.21 | p3nguin | http://www.downforeveryoneorjustme.com/voip-info.org |
| 18:29.15 | timholum1 | ok, I thought so, by my isp has had some issue's where certain sites get blocked for a day or two and then come back up |
| 18:29.43 | timholum1 | p3nguin: ill have to remember that site :) |
| 18:32.55 | thehar | yawns |
| 18:37.20 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:57.13 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 18:58.30 | andyoutside | I am not seeing dahdi show up in core show help |
| 18:58.39 | andyoutside | What am I missing? |
| 18:58.52 | p3nguin | You're wanting to see core help dahdi? |
| 18:59.17 | p3nguin | I don't have core help, even, so I don't know how that would work. |
| 19:00.31 | *** join/#asterisk CaneToad (~CaneToad@CPE-58-174-135-10.mjcz1.woo.bigpond.net.au) |
| 19:00.51 | *** join/#asterisk maimouditsos (~johnspiro@athedsl-424560.home.otenet.gr) |
| 19:00.53 | maimouditsos | hey all |
| 19:01.01 | andyoutside | just tired restarting dahdi |
| 19:01.17 | andyoutside | and got http://pastebin.com/03Nn8fdF |
| 19:01.22 | maimouditsos | is there any software i right click a number and click CALL and then asterisk calls and connects to my extension ? |
| 19:02.01 | p3nguin | It can call an extension, but it doesn't connect to extensions. |
| 19:02.34 | maimouditsos | :( |
| 19:02.56 | maimouditsos | i want to right click call number and then my ip phone rings |
| 19:03.01 | maimouditsos | and the number is dialed |
| 19:03.08 | maimouditsos | like outlook integration |
| 19:03.22 | p3nguin | If you can program, you can probably accomplish it. |
| 19:03.23 | andyoutside | there is a way to do that maimouditsos but you may have to do all the programing |
| 19:03.48 | maimouditsos | can you give me some more info ? |
| 19:04.57 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.135.18) |
| 19:05.38 | timholum1 | maimouditsos: I have a phpagi script that will basicly allow you to write a web page to do that |
| 19:05.50 | maimouditsos | yes i could live with that |
| 19:05.53 | maimouditsos | copy paste the number |
| 19:06.01 | timholum1 | one sec |
| 19:06.46 | maimouditsos | thanks |
| 19:07.07 | andyoutside | http://webcache.googleusercontent.com/search?q=cache:OUlMqAlAEKUJ:www.voip-info.org/index.php%3Fcontent_id%3D3956+link+to+call+asterisk&cd=1&hl=en&ct=clnk&gl=us |
| 19:07.25 | p3nguin | You could use a soft phone or a web phone to input the number and press DIAL. Write a dial plan to take that number and connect your phone and that number together. |
| 19:07.38 | andyoutside | p2nguin: do you have anyidea for the dahdi? |
| 19:07.49 | maimouditsos | hm lets see |
| 19:08.04 | maimouditsos | it would be too much trouble to make another 10 extensions for my employees |
| 19:08.33 | timholum1 | sorry I miss typed not phpagi, a ami script written in php |
| 19:08.36 | p3nguin | andyoutside: You've left out a lot of information needed to provide an answer. |
| 19:09.37 | andyoutside | ok what information do you want? |
| 19:11.46 | timholum1 | http://pastebin.com/Qu3PK5q1 |
| 19:12.12 | maimouditsos | andyoutside: thanks for the information |
| 19:13.30 | timholum1 | and you will have to set enable=yes in manager.conf along with creating a user with write=call |
| 19:19.08 | *** join/#asterisk kidisgod (~danimal@184.77.195.41) |
| 19:19.17 | maimouditsos | thanks timholum1 |
| 19:19.20 | kidisgod | greetings all |
| 19:19.39 | kidisgod | who should I talk to about a possible regression in asterisk 1.8 on bug #7761 |
| 19:20.25 | p3nguin | You could comment on the issue page or you might find someone in #asterisk-dev who is interested. |
| 19:21.02 | kidisgod | this was originally resolved in circa 2006 |
| 19:21.27 | p3nguin | So it wasn't on 1.8, then, so create a new issue report. |
| 19:21.36 | timholum1 | maimouditsos: no problem, if you have any question on it and if Im not in the room feel free to email them to me at pythonholum at gmail dot com |
| 19:22.08 | maimouditsos | thanks mate :) you are really help |
| 19:22.44 | kidisgod | on an inbound call to asterisk 1.8 ${EXTEN} is set to the following |
| 19:22.46 | kidisgod | 4259967033;phone-context=+1;npdi=yes |
| 19:23.28 | p3nguin | You're saying that the EXTEN is more than just the number? |
| 19:23.33 | kidisgod | correct |
| 19:23.49 | kidisgod | on 1.4 and 1.6.2 asterisk would trim anything after the first ; |
| 19:23.49 | p3nguin | Sounds like a problem someone else would have ran into by now. |
| 19:24.03 | kidisgod | my sip peer is a sonus gw |
| 19:24.25 | p3nguin | Submit an issue with all the necessary information on how to reproduce the problem. |
| 19:24.35 | WIMPy | p3nguin: I thought so many times before. |
| 19:24.40 | kidisgod | ok I can do that |
| 19:24.50 | kidisgod | thanks for the input p3nguin |
| 19:26.36 | SuD | random question: as AEL dialplans look more clean and all, do you think AEL will eventually become the mainstream syntax? |
| 19:28.20 | kidisgod | Sud I think that would depend on the dialplan coder |
| 19:30.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 19:31.53 | p3nguin | I can't imagine switching to AEL primarily -- overall, I don't think that many people use AEL dial plans. |
| 19:33.57 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 19:45.57 | *** join/#asterisk benklop_ (~ben@c-24-8-170-86.hsd1.co.comcast.net) |
| 19:46.07 | *** join/#asterisk splnet (~splnet@128.239.28.2) |
| 19:46.35 | *** part/#asterisk splnet (~splnet@128.239.28.2) |
| 19:55.05 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:6233:4bff:fe0e:bc1) |
| 20:01.05 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.139.117) |
| 20:01.15 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 20:14.10 | andyoutside | ok why am I getting DAHDI_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:14.53 | andyoutside | when I now run dahdi_cfg |
| 20:15.43 | WIMPy | Looks like it can't find the configured hardware. |
| 20:16.48 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 20:18.07 | andyoutside | why would it not find the hardware? |
| 20:21.12 | andyoutside | it seems to find it then when I restart dahdi it does not |
| 20:22.42 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 20:28.21 | *** join/#asterisk wesphillips (~wphillips@75.53.137.6) |
| 20:31.21 | benklop_ | hey, new to asterisk, is there a way to do echo cancellation on a specific SIP extension? |
| 20:31.31 | benklop_ | my ATA produces some echo |
| 20:33.03 | benklop_ | everything I am finding seems to be for an fxo or something.. a zaptel / dahdi interface, but nothing for sip |
| 20:33.29 | WIMPy | hasn't heard of EC for SIP. |
| 20:42.12 | p3nguin | benklop_: SIP isn't an extension, extensions aren't SIP, and SIP doesn't echo. |
| 20:42.23 | p3nguin | benklop_: Welcome to the world of Asterisk. |
| 20:45.31 | benklop_ | p3nguin: thanks :-P so, I have a sip softphone and an ATA that are registered with asterisk, is that a correct way of putting it |
| 20:45.32 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 20:45.44 | p3nguin | benklop_: yes |
| 20:46.38 | benklop_ | ok, so then I believe the ATA is causing echo on whatever calls it |
| 20:46.40 | p3nguin | benklop_: If you hear echo on the ATA, it's in the analog portion of the device, and other than enabling the ATA's echo cancellation, there's probably nothing else you can do. |
| 20:47.01 | benklop_ | the echo is on the other device, but only happens when calling the ATA |
| 20:47.22 | p3nguin | You hear it on the other device, but it's occurring in the ATA. |
| 20:47.23 | benklop_ | ok. ATA's echo cancellation is on, but apparently crappy |
| 20:47.35 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:47.36 | p3nguin | I bet if you turn it off, it's really bad. |
| 20:48.01 | benklop_ | p3nguin: yeah, it gets somewhat worse when disables, but I wouldn't call either one bearable |
| 20:48.17 | p3nguin | What kind of ATA is it? |
| 20:48.20 | benklop_ | honestly it's probably my fault for buying a 15 dollar ATA |
| 20:48.23 | benklop_ | soyo |
| 20:48.52 | benklop_ | i might have been 20, but in any case.. cheap. |
| 20:49.46 | p3nguin | I don't know of anything else to do to reduce echo caused by an ATA. Maybe someone else has some tricks. |
| 20:50.18 | benklop_ | p3nguin: well, I'm actually thinking the best way of dealing with it is to buy an ATA with better echo cancellation |
| 20:50.34 | benklop_ | any suggestions? |
| 20:50.41 | WIMPy | Stay digital |
| 20:50.43 | kidisgod | I would try the ata with some other type of sip server and or service |
| 20:50.48 | kidisgod | and see if the results are the same. |
| 20:50.52 | kidisgod | then you know it is the ata |
| 20:51.04 | benklop_ | kidisgod: I did, like you suggested and it is the ATA |
| 20:51.16 | p3nguin | If you can go full IP without the ATA that's the best thing to do. |
| 20:51.30 | kidisgod | maybe try a different network cable and or a different port on the hub too |
| 20:51.39 | kidisgod | just to make sure it is not somethign on the local network |
| 20:52.44 | benklop_ | p3nguin: i'd like to, but.. do they really make sip cordless phones of a style that would look normal in my kitchen and living room? |
| 20:53.15 | WIMPy | YEs, but you don't want them. |
| 20:53.46 | WIMPy | But you can go dect via SIP or ISDN. |
| 20:55.31 | kidisgod | I have seen a lot of analog cordless phones connected via ata's |
| 20:55.37 | benklop_ | WIMPy: dect? |
| 20:55.48 | p3nguin | I guess it depends on what type of look you're comfortable with when it comes to phones. |
| 20:55.51 | p3nguin | ~dect |
| 20:55.51 | infobot | Digital European Cordless Telecommunications (telecommunication) [source: V.E.R.A.] |
| 20:55.51 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 20:56.22 | WIMPy | European? |
| 20:56.42 | WIMPy | That should be "Enhanced" |
| 20:57.06 | p3nguin | Only because it changed. |
| 20:57.48 | WIMPy | Ages ago. |
| 20:58.18 | benklop_ | anyway, looking on google, dect seems to find me some decent looking phones, but how do they communicate with asterisk if they are not SIP? |
| 20:58.48 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 20:58.51 | WIMPy | You gan get SIP base stations. |
| 20:58.53 | p3nguin | You must use DECT which is SIP (or some other IP-based transport). |
| 20:59.19 | WIMPy | But I had the impression that the affordable one are still a litte experimental. |
| 20:59.30 | benklop_ | hm |
| 20:59.46 | WIMPy | benklop_: What's your location? |
| 21:00.00 | benklop_ | U |
| 21:00.03 | benklop_ | er US |
| 21:00.34 | WIMPy | Bad starting point for telephony. |
| 21:00.50 | benklop_ | hm? |
| 21:00.58 | benklop_ | i'm not sure what you mean |
| 21:01.12 | p3nguin | I think he means we suck. |
| 21:01.19 | benklop_ | lol |
| 21:01.22 | WIMPy | But if your budget isnt too tight you should find decent SIP/dect bases from the well known PBX vendors. |
| 21:01.31 | benklop_ | probably not totally wrong |
| 21:02.05 | benklop_ | I was noticing some sip dect phones on google , made by panasonic.. look decent |
| 21:02.26 | benklop_ | so you're saying ill never get an ATA to be totally echo free |
| 21:02.27 | p3nguin | Also look at Siemens. |
| 21:03.13 | WIMPy | had lots of trouble with Siemens. |
| 21:03.19 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:03.39 | WIMPy | But it seems to work since I set up a 2nd base so that each of them only has one mobile peered. |
| 21:04.07 | *** part/#asterisk wesphillips (~wphillips@75.53.137.6) |
| 21:05.20 | *** join/#asterisk telex (telex@gateway/shell/anapnea.net/x-vkggadkamahekybd) |
| 21:07.06 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 21:09.43 | *** join/#asterisk mintee (1000@static-64-115-220-213.isp.broadviewnet.net) |
| 21:10.10 | mintee | anyone using hylafax with iaxmodem? |
| 21:10.22 | mintee | i know it's unrelated, but i'm stuck |
| 21:10.48 | benklop_ | i'll do some research I guess |
| 21:11.04 | benklop_ | i was hoping to just use my existing phones |
| 21:12.39 | p3nguin | A better ATA could help. |
| 21:13.13 | p3nguin | |
| 21:14.08 | benklop_ | any recommendations? |
| 21:14.18 | benklop_ | i was thinking about gonig with a linksys |
| 21:14.44 | p3nguin | A lot of people use the Linksys PAP2T-NA. |
| 21:15.00 | p3nguin | Others like the Cisco ATA 186. |
| 21:19.27 | benklop_ | anybody mind helping me with my dialplan a bit? I'm trynig to make it so I don't have to dial 1 before a number, but optionally can. Currently, I get an error from google voice when I dial a number without the 1 first. |
| 21:19.29 | benklop_ | http://pastebin.ca/1997697 |
| 21:19.37 | benklop_ | that's my dialplan |
| 21:20.05 | p3nguin | You want both 10- and 11-digit dial? |
| 21:20.06 | benklop_ | so if i'm understanding right, "exten => _1NXXNXXXXXX,1,Dial(Gtalk/gmail/${EXTEN:1}@voice.google.com)" |
| 21:20.22 | benklop_ | yes |
| 21:20.30 | benklop_ | that line is the one that is working correctly |
| 21:20.36 | p3nguin | Do you need to send 10 or 11 digits? |
| 21:20.39 | p3nguin | okay, so that's 10 digits. |
| 21:20.45 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 21:20.49 | p3nguin | dial 11, send 10. |
| 21:20.53 | pabelanger | Your missing the '+': exten => _1NXXNXXXXXX,1,Dial(Gtalk/gmail/+${EXTEN:1}@voice.google.com) |
| 21:21.06 | p3nguin | So take off the 1 in the extension and take out the :1 in the variable. |
| 21:21.45 | p3nguin | +${EXTEN:1} would dial +3149691077 which would be failure. |
| 21:22.02 | p3nguin | +13149691077 might work, though. |
| 21:22.21 | pabelanger | Google requires the +{10digit number} |
| 21:22.26 | p3nguin | WHAT?! |
| 21:22.36 | p3nguin | Why would they require that? It's not valid. |
| 21:22.44 | pabelanger | Valid for what? |
| 21:22.48 | pabelanger | It is their network |
| 21:22.58 | p3nguin | But he said it's working as he wrote it, so why argue it? |
| 21:23.35 | p3nguin | _1NXXNXXXXXX,1,Dial(Gtalk/gmail/${EXTEN:1}@voice.google.com) and _NXXNXXXXXX,1,Dial(Gtalk/gmail/${EXTEN}@voice.google.com) |
| 21:23.44 | benklop_ | i'm checking now to see specifically what is currently working.. |
| 21:23.45 | p3nguin | Those both dial the same thing. |
| 21:23.45 | pabelanger | Who says I am arguing? I am telling him how I did my setup |
| 21:23.45 | benklop_ | hold on |
| 21:24.36 | benklop_ | ok, so the line I pasted in here is in fact the currently working line |
| 21:25.04 | benklop_ | so i think i'm still a bit confused about what I need to do |
| 21:25.09 | p3nguin | see above |
| 21:25.17 | p3nguin | Create another extension using the info I gave you. |
| 21:25.36 | p3nguin | One for 10-digit dialing, the other for 11-digit dialing. |
| 21:25.38 | *** join/#asterisk imox1234 (~imox1234@p4FC5C3A3.dip0.t-ipconnect.de) |
| 21:26.10 | benklop_ | p3nguin: ok, i'm trying that now |
| 21:26.18 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.147.178) |
| 21:26.20 | pabelanger | benklop_: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 21:26.36 | p3nguin | And if you find out that you have to dial using a + in it, then you can add the + accordingly. |
| 21:27.16 | benklop_ | well that works much better :) |
| 21:28.39 | p3nguin | ${EXTEN:1} means to offset the extension number called. If you dial 11 digits on the phone, ${EXTEN:1} will only be the last 10 digits. |
| 21:28.40 | benklop_ | in my area, there are 2 area codes that are considered local, and you don't normally have to dial a 1 before dialling those, so people are used to that |
| 21:29.02 | p3nguin | Likewise, if you dial 10 digits on the phone, ${EXTEN:1} will only be the last 9 digits. |
| 21:29.07 | benklop_ | ok, so it offsets by the number after the colon? |
| 21:29.14 | *** join/#asterisk tyman_ (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 21:29.52 | benklop_ | so if say i wanted to ignore the first 4 digits, i'd do ${EXTEN:4} ? |
| 21:30.02 | p3nguin | In that particular instance, yes. You can also use things like ${EXTEN:1:1} for something entirely different. |
| 21:30.17 | p3nguin | yes, ${EXTEN:4} is an offset of 4. |
| 21:30.29 | *** join/#asterisk scrooge-mcduck (~scroogey@80-219-24-66.dclient.hispeed.ch) |
| 21:30.32 | scrooge-mcduck | hello |
| 21:30.35 | p3nguin | ${EXTEN} has no offset |
| 21:30.55 | *** join/#asterisk trumee (~nobody@cpc5-cmbg14-0-0-cust982.5-4.cable.virginmedia.com) |
| 21:31.11 | scrooge-mcduck | i am trying to get asterisk working and i am getting |
| 21:31.14 | benklop_ | so :1:1 would, i'm guessing here obviously, grab 1 digit, 1 digit into the string? |
| 21:31.25 | scrooge-mcduck | handle_request_register: Registration from '"101" <sip:101@192.168.1.6>' failed for '192.168.1.20' - No matching peer found |
| 21:31.26 | trumee | guys, i have installed asterisk in Ubuntu LTS 10.10. The bloody thing doesnt start. |
| 21:31.35 | benklop_ | or is it more convoluted than that? |
| 21:31.36 | scrooge-mcduck | while i am having a friend user for that :| |
| 21:31.56 | trumee | asterisk -vvvvv just spits out a config and retuns back to console without any error or warning message. |
| 21:32.10 | p3nguin | benklop_: I think that's right, yes. |
| 21:32.17 | p3nguin | ${VARIABLENAME[:start[:length]]} |
| 21:32.19 | trumee | last message is " == Manager registered action VoicemailUsersList" |
| 21:32.50 | benklop_ | p3nguin: cool |
| 21:32.57 | scrooge-mcduck | how can i find out the available extensions and sip users? |
| 21:33.07 | p3nguin | ${EXTEN:-7} gives you only the last 7 numbers regardless of the length. |
| 21:33.20 | trumee | strace asterisk is also not indicating anything. is there a trick in ubuntu i am missing? |
| 21:33.44 | p3nguin | benklop_: http://www.the-asterisk-book.com/unstable/variablen.html |
| 21:35.11 | p3nguin | Anyway, that's why your 10-digit extension wasn't working. |
| 21:35.20 | *** join/#asterisk telex (telex@gateway/shell/anapnea.net/x-hkckgkegyxodeoff) |
| 21:36.06 | benklop_ | pabelanger: thanks a lot for that wiki page, it's helping with some other things too |
| 21:36.12 | benklop_ | namely call presentation stupidity |
| 21:36.26 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:36.38 | benklop_ | p3nguin: thanks |
| 21:36.48 | benklop_ | i have that book open right now |
| 21:37.29 | scrooge-mcduck | anyone know what could be up? |
| 21:38.16 | benklop_ | one other thing, how can I get this to ring all extensions on an incoming call? |
| 21:38.29 | p3nguin | You probably mean ring all devices. |
| 21:38.36 | p3nguin | You'll only have one extension. |
| 21:38.51 | benklop_ | yes, that's what I mean |
| 21:38.52 | kidisgod | you can ring them all via a queue |
| 21:38.54 | p3nguin | Dial(SIP/bill&SIP/john&SIP/frank) |
| 21:38.58 | kidisgod | or chain them together in the dial |
| 21:39.38 | benklop_ | kidisgod, p3nguin: thanks |
| 21:40.07 | p3nguin | You could also use more than one Dial() if you want offset ring times. |
| 21:40.35 | p3nguin | Dial(SIP/bill,10) |
| 21:40.38 | p3nguin | Dial(SIP/bill&SIP/john,20) |
| 21:40.51 | p3nguin | Bill's phone would ring for 10 seconds before John's starts. |
| 21:40.59 | benklop_ | ok, so bill would ring for 10, then bill and john would ring for 20 |
| 21:41.37 | WIMPy | And bill gets a missed call after 10s. |
| 21:41.51 | p3nguin | Unfortunately, that's sometimes a problem. |
| 21:41.54 | benklop_ | not what i'm looking for exactly, ringing all at once is fine |
| 21:42.20 | p3nguin | You can get around that problem by using Local channels and crafting more well-thought-out dial plans. :) |
| 21:44.50 | scrooge-mcduck | hmm, no errors in confnig but doesnt work |
| 21:44.52 | scrooge-mcduck | keep getting 404 |
| 21:45.13 | trumee | Guys, no error and warning message and asterisk doesnt start. http://paste.ubuntu.com/535012/ does it mean anything? |
| 21:45.43 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 21:47.13 | trumee | anybody? |
| 21:47.25 | pabelanger | trumee: $ asterisk -vvvvvvvcg |
| 21:47.27 | pabelanger | what happens? |
| 21:48.02 | trumee | pabelanger: same thing it just exits. |
| 21:48.31 | pabelanger | trumee: PB the result, asterisk maybe segfaulting |
| 21:48.48 | trumee | pabelanger: it doesnt say that it is segfaulting |
| 21:49.02 | trumee | pabelanger: should there by any core dumps? |
| 21:49.10 | pabelanger | if you pass -g |
| 21:49.23 | pabelanger | also, your previous examples are missing -c (console) |
| 21:50.16 | trumee | pabelanger: http://paste.ubuntu.com/535016/ |
| 21:50.29 | scrooge-mcduck | Anyone know why I am getting this error: http://paste.ubuntu.com/535017/ ? |
| 21:50.43 | trumee | pabelanger: it simply comes back to the shell. |
| 21:51.01 | pabelanger | trumee: this is voicemail bug, at noload => app_voicemail.so to modules.conf |
| 21:51.14 | kidisgod | scrooge: no sip peer defined |
| 21:51.25 | kidisgod | scrooge: sip show peers |
| 21:51.52 | pabelanger | trumee: https://issues.asterisk.org/view.php?id=18240 |
| 21:52.04 | pabelanger | should be fixed for the next release |
| 21:52.46 | trumee | pabelanger: oh!. |
| 21:52.55 | trumee | pabelanger: is there any way i can get past it? |
| 21:53.12 | pabelanger | <pabelanger> trumee: this is voicemail bug, add noload => app_voicemail.so to modules.conf |
| 21:53.20 | pabelanger | typo |
| 21:53.32 | trumee | pabelanger: thanks. let me try this |
| 21:54.22 | pabelanger | scrooge-mcduck: dynamic=yes is not a valid setting, remove it |
| 21:55.06 | trumee | pabelanger: didnt help |
| 21:55.24 | pabelanger | You also set nat=yes, but your SIP trace shows an local IP address |
| 21:55.29 | benklop_ | pabelanger: does caller ID work for you with google? for me, incoming calls are shown on my sip phone as having a blank caller ID, but if I watch in the console, I can see the caller ID number is identified on the incoming call |
| 21:55.36 | trumee | pabelanger: i added noload => app_voicemail.so at the bottom of /etc/asterisk/modules.conf |
| 21:56.26 | pabelanger | benklop_: Yes, sounds like to need to add some logic to your dialplan |
| 21:56.36 | pabelanger | trumee: okay |
| 21:56.48 | trumee | pabelanger: it still does the same thing. |
| 21:57.16 | pabelanger | trumee: is voicemail still being loaded? |
| 21:57.34 | trumee | pabelanger: asterisk -vvvvvvvcg |grep voicemail ? |
| 21:57.50 | pabelanger | scrooge-mcduck: Also, set type=peer for [101] |
| 21:58.55 | pabelanger | trumee: You'll need to get a backtrace and see what is going on. https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:59.02 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 21:59.03 | pabelanger | start Asterisk under GDB |
| 21:59.12 | trumee | pabelanger: http://pastebin.com/6X4Qsgnm |
| 21:59.40 | *** join/#asterisk tehrabbitt (~tehrabbit@pool-71-172-89-155.nwrknj.fios.verizon.net) |
| 21:59.42 | pabelanger | trumee: You are using the Debian / Ubuntu build right? |
| 21:59.54 | trumee | pabelanger: yes Ubuntu LTS 10.10 |
| 22:00.16 | pabelanger | you need to no load _all_ app_voicemail_odbc.so, app_voicemail_imap.so, app_voicemail.so |
| 22:00.53 | pabelanger | It should be done by default within modules.conf |
| 22:01.42 | trumee | pabelanger: ah! now it works |
| 22:01.53 | trumee | pabelanger: is that an ubuntu issue? |
| 22:02.21 | pabelanger | trumee: No, Asterisk but file an issue with Ubuntu and I can backport the patch to it. |
| 22:02.36 | pabelanger | I'm on the mailing list |
| 22:02.51 | trumee | pabelanger: thanks |
| 22:03.07 | pabelanger | trumee: what version of Asterisk is it? |
| 22:03.29 | trumee | pabelanger: Asterisk 1.6.2.7-1ubuntu1 |
| 22:03.38 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 22:03.41 | pabelanger | ok |
| 22:04.07 | benklop_ | pabelanger: can you be a little more specific about what I need to add to my dialplan? or what I need to read? |
| 22:04.11 | trumee | pabelanger: thank you very much. |
| 22:04.26 | pabelanger | trumee: np |
| 22:05.40 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 22:06.34 | pabelanger | benklop_: same => n,Set(CALLERID(all)=${GoogleVoiceVariabeYouHaveParsed}) |
| 22:06.43 | pabelanger | same => n,Dial(SIP/1234) |
| 22:07.14 | benklop_ | ah, okay |
| 22:07.28 | benklop_ | i'll see if i can figure that out |
| 22:08.00 | pabelanger | benklop_: Might also be able to just pass CALLERID(num) |
| 22:08.01 | benklop_ | what is same => ? so far all my lines have been exten |
| 22:08.23 | pabelanger | in 1.8.0 you can do: |
| 22:08.36 | pabelanger | exten => s,1,NoOp(top) |
| 22:08.47 | pabelanger | same => n,Dial(SIP/123) |
| 22:09.15 | pabelanger | exten => s can now be same => |
| 22:10.16 | trumee | pabelanger: it is freepbx fault. |
| 22:10.29 | trumee | pabelanger: freepbx overwrote modules.conf |
| 22:11.10 | trumee | pabelanger: before i installed freepbx, asterisk did start correctly i think. |
| 22:12.46 | p3nguin | weeeeeeeeee |
| 22:13.05 | p3nguin | We loves us some FreePBX up in here. |
| 22:18.38 | scrooge-mcduck | hmm |
| 22:19.15 | *** join/#asterisk moy_ (~moy@64.231.53.158) |
| 22:20.09 | benklop_ | so, when i have done a sip set debug on, I can see "From: "+15155515115@voice.google.com/srvres-MTAuMjI5LjY4LjEwMjo5ODQw" <sip:asterisk@192.168.2.7>;tag=as00022739" |
| 22:20.23 | benklop_ | does that make "From" the variable I need to parse? |
| 22:22.36 | benklop_ | ok, apparently not |
| 22:23.44 | kidisgod | benklop look at ${EXTEN} |
| 22:25.07 | kidisgod | or ${CALLERID(all)} |
| 22:25.14 | scrooge-mcduck | i tried to change the settings pabelanger but no lck |
| 22:25.16 | kidisgod | in the context the call is coming into |
| 22:26.04 | *** part/#asterisk scrooge-mcduck (~scroogey@80-219-24-66.dclient.hispeed.ch) |
| 22:26.43 | p3nguin | benklop_: What's wrong now? |
| 22:29.11 | *** join/#asterisk scrooge-mcduck (~scroogey@80-219-24-66.dclient.hispeed.ch) |
| 22:29.15 | scrooge-mcduck | i got ti working ! |
| 22:29.24 | scrooge-mcduck | needed to add hassip=yes to the users |
| 22:29.42 | scrooge-mcduck | because i used users.conf |
| 22:29.46 | CaneToad | I have asterisk (1.6.x) registering successfully with a couple of external PSTN providers and have a couple of corresponding contexts in sip.conf for those...but in the appropriate context in extensions.conf, I have the following, but it complains about audio compatibility - I presume it somehow isn't matching with the context in sip.conf: exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@whatever.com) ..... how can I properly say "c |
| 22:31.00 | benklop_ | p3nguin: just trying to figure out what variable I need to parse to get caller ID working on incoming google voice calls. I can see the info in debugging output |
| 22:31.39 | p3nguin | benklop_: CallerID is extracted using the CALLERID() function. |
| 22:31.46 | pabelanger | scrooge-mcduck: You have better control and more options if you use sip.conf |
| 22:34.10 | scrooge-mcduck | yeah, i learned that now |
| 22:34.42 | scrooge-mcduck | back to learning about dial rules :) |
| 22:34.43 | scrooge-mcduck | thanks all! |
| 22:35.57 | p3nguin | benklop_: And if CallerID is sent with a call, you shouldn't have to do anything for it to work. |
| 22:39.45 | benklop_ | p3nguin: i'm thinking gtalk is setting name but not number.. just a guess |
| 22:41.20 | p3nguin | benklop_: When you get a call, what shows up on the phone's display? |
| 22:41.28 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
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| 22:45.36 | benklop_ | p3nguin: nothing, a few dashes with blanks in between |
| 22:46.05 | benklop_ | p3nguin: but in the softphone I get a rather long string beginning with the number |
| 22:47.13 | benklop_ | p3nguin: I added verbose lines in the incoming context, same => n,Verbose(${CALLERID(num)}) outputs nothing, but same => n,Verbose(${CALLERID(all)}) does show that string my softphone gives me |
| 22:48.31 | benklop_ | +15151551555@voice.google.com/srvres-MTAuMjI5LjEzMS4zMDo5ODUw <> in speciic |
| 22:49.05 | p3nguin | So the CID name is +15151551555@voice.google.com/srvres-MTAuMjI5LjEzMS4zMDo5ODUw and the number is empty. |
| 22:49.13 | benklop_ | yep |
| 22:49.45 | benklop_ | so i'm thinking I can use Set(CALLERID(num) = ${CALLERID(name):1:10} |
| 22:49.48 | benklop_ | right? |
| 22:49.59 | benklop_ | or do I want to include the +? |
| 22:50.07 | p3nguin | Better go :2:10 instead. |
| 22:50.17 | benklop_ | ok |
| 22:50.25 | p3nguin | The number really doesn't have a + or a 1 on it. |
| 22:50.39 | benklop_ | ok |
| 22:54.25 | p3nguin | (not in the US, anyway) |
| 22:55.12 | benklop_ | cool, that worked :D |
| 22:55.37 | benklop_ | OK, so the only thing left bothering me is the echo on my ATA, and that looks to be a case of getting a better ATA. |
| 22:56.03 | benklop_ | thanks for the help kidisgod, p3nguin, and pabelanger! |
| 22:56.15 | p3nguin | Was there a guide you used to get your GV calling working? |
| 22:57.21 | benklop_ | p3nguin: yeah, it was really a combination of a couple guides, i did it a few weeks ago, might have been more than a month |
| 22:57.48 | benklop_ | i know there was a typo to contend with downloading the 1.8 tarball still |
| 22:58.21 | benklop_ | why? |
| 22:58.38 | p3nguin | I've noticed a lack of GOOD documentation on the topic. |
| 22:58.52 | benklop_ | that's definitely true |
| 22:59.10 | p3nguin | If there is any good documentation, there are very few pieces of it out there. |
| 22:59.22 | benklop_ | that wiki page someone pointed me to looked to be pretty helpful though |
| 23:00.59 | benklop_ | later this evening i'll see about adding the caller ID stuff to it |
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| 23:19.58 | benklop_ | is there any reliable way to look up names to associate wth the numbers I |
| 23:20.03 | benklop_ | now get |
| 23:25.00 | benklop_ | i'm seeing the callerID superfecta stuff on nerd vittles, but that requires freepbx |
| 23:25.12 | benklop_ | and i've had bad experiences with freepbx :-P |
| 23:25.25 | benklop_ | i'd rather just tweak all my config files myself thank you |
| 23:30.48 | scrooge-mcduck | whats an easy way to forward all non-existing extensions to a SIP call? |
| 23:32.17 | scrooge-mcduck | i tried |
| 23:32.17 | scrooge-mcduck | exten => _9.,1,Dial(SIP/${EXTEN:1}@voipbuster) |
| 23:32.20 | scrooge-mcduck | but this doesnt work |
| 23:33.06 | scrooge-mcduck | hmm maybe X |
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| 23:46.03 | KavanS | for some reason in my cdr, my "src" column is getting populated with the OUTBOUND callerid not the internal extension where it is originating from - any suggestions? |
| 23:46.30 | wwalker | anyone know of a product (other than iptty) that supports TTY/TDD over SIP? |
| 23:49.48 | mmlj4 | a tty over sip? |
| 23:50.12 | KavanS | doh :( - ERROR[21976]: cdr.c:309 ast_cdr_setvar: Attempt to set the 'src' read-only variable!. |
| 23:54.15 | wwalker | mmlj4: yes, tty as in 45.5 baudot |
| 23:54.32 | wwalker | not as in a hayes modem connection |
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