| 00:00.43 | KingDavidNYC | ectopaspam: excuse me, shouldn't it be span=2,1,0,esf,b8zs, since 1=far end T1 provides the clock? |
| 00:01.13 | ectospasm | KingDavidNYC: that's variable. It depends on which is the primary clock source, which you didn't specify |
| 00:01.36 | ectospasm | KingDavidNYC: I assumed you wanted the first span to be primary clock source, so timing for the second is 2 |
| 00:01.52 | ectospasm | ...if the second span is primary clock source, use 1 |
| 00:02.00 | *** join/#asterisk tcliam_ (~liam@203.109.157.87) |
| 00:02.30 | ectospasm | "primary clock source" meaning that span is the primary receiver of timing for the entire card |
| 00:02.49 | KingDavidNYC | ectospasm: I think in the question says they want the 2nd T1 on a 4 port T1 card to do the clocking |
| 00:03.01 | ectospasm | KingDavidNYC: then use 1 |
| 00:03.27 | tcliam_ | Hi I am trying to use fail2ban with asterisk. I have set it up to ban after 5 attempts but it seems that it is too slow to detect hack attempts fast enough. I had an attempt today and fail2ban reported that it banned after "55" attempts rather than 5 attempts. Does anyone know of any info out there on this problem? |
| 00:03.28 | ectospasm | ...although a 2 without a 1 in another span will work. |
| 00:03.36 | KingDavidNYC | ectospasm: great, thanks, and waht about chan_dahdi.conf? |
| 00:03.50 | ectospasm | KingDavidNYC: you need at least signalling=pri_cpe; switchtype=national; and channel => 25-47; << all in chan_dahdi.conf |
| 00:04.24 | ectospasm | KingDavidNYC: that's at minimum, but you will want other stuff (like echocancel, group, etc.) |
| 00:05.11 | KingDavidNYC | Thanks a lot guys, please help me with one more question: Configure the 4th T1 on a 4 port T1 card as e&m with the first 10 channels in group 5 and the rest in group 11 . Show the relevant section in both chan_dahdi.conf and system.conf |
| 00:05.45 | ManxPower | Does anyone happen to know what the maximum length of a sip secret in Asterisk? |
| 00:05.49 | ectospasm | KingDavidNYC: span in system.conf will be similar |
| 00:06.36 | ectospasm | instead of dchan and bchan you'll have e&m=73-96 |
| 00:07.11 | KingDavidNYC | ectospasm: can you please write it for me?, as it will look in chan_dahdi, and system.conf? |
| 00:07.15 | plut0 | ok google voice calls are broken, nothing to do with asterisk |
| 00:07.31 | tyman | ManxPower: Didn't find explicitly, but the username max is 255 char and is "probably" same |
| 00:07.48 | ectospasm | KingDavidNYC: hold on. |
| 00:08.19 | KingDavidNYC | ectopspasm: thanks man, this is the kind of thing you have to do once, to get it right |
| 00:08.50 | KingDavidNYC | ectospasm: mere theory just doesn't cut it |
| 00:08.57 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
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| 00:11.01 | ectospasm | KingDavidNYC: http://pastebin.com/8fz376r0 |
| 00:12.36 | KingDavidNYC | ectospasm: awsome! thanks a lot man!, I owe you lunch!! |
| 00:13.17 | ectospasm | KingDavidNYC: that's not guaranteed to be correct, you may want to compare that with what you'll find on the various Asterisk wikis |
| 00:13.41 | ectospasm | (e.g. voip-info.org, wiki.asterisk.org) |
| 00:14.36 | KingDavidNYC | ectospasm: correct, in any case it is a terrific answer, thanks again |
| 00:15.51 | ectospasm | KingDavidNYC: I will admit my E&M configuration fu is rusty |
| 00:16.01 | KingDavidNYC | mmm |
| 00:16.09 | ectospasm | ...and there are a bunch of E&M signalling types |
| 00:16.25 | KingDavidNYC | I will check it out....it is better than mine :) |
| 00:17.06 | ectospasm | just curious, where are you interviewing? |
| 00:17.45 | ectospasm | KingDavidNYC: ^ |
| 00:19.28 | KingDavidNYC | Washingto DC.. they want a person with knowledge of pri installations, signalings, C, dialplan, etc.... asterisk guy |
| 00:23.15 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:23.30 | ectospasm | Ah, OK |
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| 00:58.15 | mlsmith9999 | DIDForSale... Yay or Nay? |
| 00:58.27 | p3nguin_ | Never heard of it. |
| 00:59.28 | mlsmith9999 | what is a good DID supplier? |
| 01:00.01 | p3nguin_ | Wholesale or for personal/business use? |
| 01:00.10 | mlsmith9999 | business |
| 01:00.17 | p3nguin_ | What country? |
| 01:00.36 | mlsmith9999 | US |
| 01:00.52 | p3nguin_ | Do you only want phone numbers, or do you want VoIP services too? |
| 01:01.31 | mlsmith9999 | have phone number, need them ported so VoIP services. |
| 01:01.48 | p3nguin_ | ~itsplist-us |
| 01:01.49 | 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 |
| 01:02.55 | aidinb | someone's gotta bump flowroute up on that thang |
| 01:03.06 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 01:03.20 | p3nguin_ | In addition to those, there are also plenty of others, but we don't know every ITSP that exists. |
| 01:03.28 | p3nguin_ | I personally use VoIP.ms for my DIDs, and I use both VoIP.ms and Flowroute for termination. |
| 01:05.11 | tyman | has anyone here used the polycom "kirk" phones with asterisk? any comments or alternate recommendations before I shell out some dough? |
| 01:05.12 | atan | Anyone here mess with skinny? This is my config for the phone http://pastie.org/private/0ehj1kfzt4h10gdasah5q |
| 01:05.21 | atan | and the phone just keeps pulling it off the tftp agian & again |
| 01:05.31 | atan | I assume I am missing something |
| 01:07.04 | p3nguin_ | That looks like it should be your XMLDefault.cnf.xml. |
| 01:07.09 | mlsmith9999 | ok, I'll check some of those out.. I wish I could go with Vitelity, but they can't port my number... 501-803 |
| 01:07.41 | atan | p3nguin_, new config http://pastie.org/private/0dbqevsbxd8a4a9fsgbmw |
| 01:07.54 | atan | Just swapped it for something I found while googling |
| 01:08.21 | p3nguin_ | That still looks like it should be your XMLDefault.cnf.xml. |
| 01:08.38 | p3nguin_ | What version of the sccp image do you have? |
| 01:08.50 | atan | P00308000600 is what I just put on the phone |
| 01:09.18 | p3nguin_ | I run P00308010200 here. |
| 01:10.15 | atan | Does the 7940/7960 support P00308010200? |
| 01:10.26 | p3nguin_ | That's what I'm using it on. |
| 01:11.34 | atan | Any chance you would sneak me a look at your SEP<mac>.cnf.xml setup? |
| 01:13.05 | p3nguin_ | XMLDefault.cnf.xml http://pastebin.com/6BdtBJc3 |
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| 01:14.03 | p3nguin_ | SEP<mac>.cnf.xml http://pastebin.com/3f6mGFqX |
| 01:14.19 | p3nguin_ | There really isn't very much info in either file. |
| 01:17.01 | p3nguin_ | If you have those two files and the four files for the firmware, I'd expect it to load up and work. |
| 01:17.24 | atan | I replaced my files with your xml setup there |
| 01:17.27 | atan | just reloading it now |
| 01:17.36 | p3nguin_ | I also have DISTINCTIVERINGLIST.XML and RINGLIST.XML for my ringers. |
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| 01:19.41 | atan | What does your skinny.conf look like for them? |
| 01:19.50 | p3nguin_ | I use chan_sccp-b. |
| 01:19.56 | atan | Hmm. |
| 01:20.06 | p3nguin_ | (sccp.conf) |
| 01:20.26 | p3nguin_ | The samples are very well documented. |
| 01:21.34 | p3nguin_ | http://chan-sccp-b.sourceforge.net/download.shtml#Subversion |
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| 01:25.09 | atan | Well perhaps I should get chan-sccp-b going then to replace the default one installed. |
| 01:26.03 | p3nguin_ | I use it on Asterisk 1.4 because chan_skinny isn't very robust. Maybe it has been further developed in other branches. |
| 01:27.45 | p3nguin_ | I'm using chan-sccp-b v3 svn rev2094 on Asterisk 1.4.36, and I'm pretty happy with it overall. |
| 01:28.48 | tyman | p3nguin: does * operate in full proxy mode if all your phones are using sccp? does that extend the concurrency of * comparable to, say, cisco ccm? |
| 01:29.02 | p3nguin_ | I don't know what "full proxy mode" is. |
| 01:29.23 | tyman | i'm mean, not b2bua like * does by default with sip |
| 01:29.35 | p3nguin_ | Well, Asterisk is a b3bua. |
| 01:29.40 | p3nguin_ | err. |
| 01:29.49 | p3nguin_ | b2bua |
| 01:29.50 | tyman | back to back user agent |
| 01:29.58 | p3nguin_ | typo |
| 01:30.21 | p3nguin_ | It's a b2bua no matter what channel technology you are using. |
| 01:30.34 | p3nguin_ | Asterisk is not a proxy. |
| 01:30.56 | atan | I feel so silly right now. |
| 01:31.08 | atan | How do I best setup chan-sccp-b? |
| 01:31.08 | p3nguin_ | typo in a conf? |
| 01:31.11 | p3nguin_ | oh |
| 01:31.27 | p3nguin_ | Did you check out the latest revision with svn? |
| 01:31.47 | atan | No. I did not. Is there some svn link? |
| 01:31.48 | tyman | really...ok...thought you could use sccp similarly as with directrtpsetup=yes |
| 01:31.55 | p3nguin_ | (1921.33) <p3nguin_> http://chan-sccp-b.sourceforge.net/download.shtml#Subversion |
| 01:31.59 | p3nguin_ | that ^^^ |
| 01:32.26 | atan | Ahh crud! None for 1.8 then? |
| 01:33.05 | p3nguin_ | I'd try it anyway. |
| 01:34.14 | atan | configure: Or run ./configure --with-asterisk=PATH |
| 01:34.24 | atan | would this be /usr/sbin/asterisk ? |
| 01:35.10 | p3nguin_ | Just run ./configure |
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| 01:35.26 | atan | COnfigured error'd out with that message at the end |
| 01:35.26 | p3nguin_ | After you checked out, of course. svn co https://chan-sccp-b.svn.sourceforge.net/svnroot/chan-sccp-b/trunk chan-sccp-b |
| 01:36.32 | atan | uses your svn link |
| 01:36.33 | p3nguin_ | I just use ./configure and it works for me. |
| 01:37.08 | atan | http://pastie.org/private/xi8ffxe2i9q8jmjcqwoa |
| 01:37.57 | atan | grabbed asterisk-dev, trying again |
| 01:38.14 | atan | We're cooking with gas now. |
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| 01:39.44 | p3nguin_ | <PROTECTED> |
| 01:39.57 | p3nguin_ | That's the one that checked ou for me. |
| 01:40.43 | p3nguin_ | To upgrade or not to upgrade... |
| 01:40.52 | carrar | peer pressure |
| 01:40.56 | carrar | everyone is doing it |
| 01:41.03 | p3nguin_ | Might as well upgrade. I can always go back if it's broken. |
| 01:43.45 | p3nguin_ | Seems okay. |
| 01:48.11 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.90) |
| 01:49.09 | atan | Okay got chan_sccp-b all setup & loaded |
| 01:49.18 | atan | It shows up in module show like sccp |
| 01:49.23 | atan | and the other module is now not loading |
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| 01:54.59 | atan | Okay the phone is going wacky =\ still can't get online there with it |
| 01:55.09 | atan | Just keeps pulling files down =( |
| 01:56.54 | atan | I suppose I could stick with SIP for it but I really wanted these little sidecar things to work =) they seem quite interesting |
| 02:02.05 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 02:06.38 | atan | Foolish me. It was the firewall preventing such activities. Hmm. |
| 02:06.46 | atan | Now to figure out how to deal with that one, eeeep. |
| 02:10.25 | atan | hey p3nguin_ , if you're still here, what does the chan_sccp-b have that the default one doesn't? Do you happen to know if it supports the expansion modules whereas the other doesn't? |
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| 02:38.21 | p3nguin_ | atan: It does support the 7914 addon. |
| 02:38.55 | atan | My 7914 lights are all red =\ trying to figure out how to push some firmware to the thing |
| 02:39.22 | atan | Most of the post I have read all refer to using ccm |
| 02:41.44 | p3nguin_ | Did you configure your phone in sccp.conf? |
| 02:42.29 | p3nguin_ | Wow, 855 toll-free numbers are finally active. |
| 02:43.23 | atan | =) |
| 02:43.28 | atan | I have it in sccp.conf |
| 02:43.31 | atan | it is connected up |
| 02:44.04 | atan | I have addon = 7914 in the config =\ |
| 02:44.12 | atan | The lights on the 7914 are all solid red though |
| 02:44.34 | atan | These 7914s are new though. I have never used them. I can't vouch for what firmware is on it... =S |
| 02:45.27 | p3nguin_ | Is the phone operating like you expect, besides the 7914? |
| 02:45.59 | atan | Sure. You could say that. =) |
| 02:46.25 | atan | Server registers all the commands & everything else there is good. |
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| 02:50.06 | atan | You wouldn't have any idea how I might flash the firmware on it using tftp? =\ |
| 02:52.09 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 02:52.12 | p3nguin_ | Not really. |
| 02:53.29 | atan | I see. Hmm. |
| 02:53.41 | atan | I foresee an eBay auction for these coming up soon. |
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| 02:58.54 | p3nguin_ | haha |
| 02:59.05 | atan | And a few Cisco phones. |
| 02:59.07 | p3nguin_ | How many of them do you have? |
| 02:59.16 | p3nguin_ | What all do you have? I might be interested. |
| 03:00.01 | atan | 2x7914, 1x7960, 1x7940, 1xdual stand (for the 7914's), 2xpower cubes |
| 03:00.15 | atan | Oddly though, I really love how the phones are built :| |
| 03:00.26 | atan | They are really commercial feeling. Hard to break feeling, if you will |
| 03:00.43 | atan | I'll just be damned if I can't get them working to their fullest |
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| 03:03.04 | atan | Err, actually, I think there will be another power cube thinger on the way here soon as well. |
| 03:03.13 | atan | Really not all that much gear. |
| 03:03.18 | p3nguin_ | Are you only considering selling, or do you know for sure that you're wanting to get rid of the stuff? |
| 03:04.15 | atan | I'm not sure what you're asking. I'm just fed up with them =) I have been looking at other more SIP compliant phones for awhile now |
| 03:04.45 | p3nguin_ | Are you absolutely ready to get rid of the stuff, or were you only contemplating selling it? |
| 03:05.11 | atan | ponders that question for a moment |
| 03:05.56 | atan | Yeah. They can go. I just need to order in a replacement. I have my eye on one right now though =) |
| 03:06.20 | atan | To be fair though, two items have still not arrived. I am still waiting for the x60 to get here (due any day) and one power cube |
| 03:07.09 | p3nguin_ | Is that a 7960G or non-G model? |
| 03:07.51 | atan | It's marked as G |
| 03:08.17 | atan | I am still waiting for it to arrive to be sure though. The supplier was confused about it, not knowing much about them or something. |
| 03:08.25 | atan | I feel his/her pain now :P |
| 03:08.34 | p3nguin_ | I tried to buy one from another guy on here... I paid him back in August, and I still haven't seen the phone. |
| 03:08.53 | atan | lol. Ouch. |
| 03:09.31 | atan | Well I have like 500+ some odd positive eBay feedback, and however many verified paypal sales if that eases your mind any :P |
| 03:10.17 | atan | I wouldn't make any deal on the 7060g one until it arrives since I can't comment on it just yet |
| 03:10.26 | atan | That would be like selling the milk before getting the cow |
| 03:11.35 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 03:12.41 | p3nguin_ | What phone were you testing with? |
| 03:12.55 | atan | Right now I have one of the two 7914s on a 7940 |
| 03:13.21 | atan | (and now you're going to tell me the 7940 doesn't support it? please?) |
| 03:13.53 | atan | All the bloody lights on the thing are red. Only reference to it on Google I could find was to update the firmware & do something funky with CCM |
| 03:13.55 | p3nguin_ | As far as I know, it does. Sorry. :( |
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| 03:14.29 | atan | Maybe it's just this 7940 is funky |
| 03:14.57 | atan | Perhaps the 7960 will have a different song to sing ^_^ |
| 03:15.35 | p3nguin_ | Don't get up your hopes on that one. The 7940 and 7960 should behave the same. |
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| 04:09.02 | sshock | Can I query the * DB from CLI? |
| 04:09.47 | sshock | hmm, database show ? |
| 04:09.49 | WIMPy | database <tab> |
| 04:10.46 | sshock | something weird is going on; * keeps calling me |
| 04:11.33 | p3nguin_ | haha |
| 04:11.39 | sshock | In the CLI I see messages like: WARNING[9750]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 874372580@10.218.121.72 for seqno |
| 04:12.09 | sshock | it has something to do with Google Voice, which I've been playing with, but I can't figure out why it is doing phantom calls |
| 04:12.38 | sshock | and no, it is not funny; ok, maybe a little bit |
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| 04:25.27 | sshock | so, any ideas? is this coming from GV, or is * doing it? |
| 04:27.09 | p3nguin_ | Are you wanting me to guess or what? |
| 04:28.13 | sshock | I don't know |
| 04:28.47 | sshock | anyone has played with pygooglevoice and noticed this behavior? |
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| 04:43.31 | v1s | is there a way to add sip users to a database and and not have to sip reload or is the only way via a conf file? |
| 04:44.56 | p3nguin_ | ~realtime |
| 04:44.56 | infobot | realtime is probably a feature of Asterisk starting with 1.2 which allows you to map any configuration file (static mappings) to be pulled from the database, or to map special runtime entries which permit the dynamic creation of objects, entities, peers, etc. without the necessity of a reload. |
| 04:47.15 | v1s | p3nguin_: thanks |
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| 05:33.01 | [TK]D-Fender | v1s: [23:42]<infobot>Fullstop wanted to say - [11:58]<fullstop>if he comes back, it's been asked before: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg189267.html |
| 05:33.04 | [TK]D-Fender | v1s: was for you |
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| 05:47.11 | v1s | ok got it thanks. |
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| 06:35.51 | jplank | could someone see whats wrong with this: ExecIf($["${LEN(${CALLERID(num)})}" < "5"],Set,CALLERID(num)=7325551234) |
| 06:35.57 | jplank | I must be missing it |
| 06:36.14 | jplank | its setting the caller ID even when the caller id len is greater then 5 |
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| 06:39.03 | jplank | ahhh got it, comma instead of a pipe |
| 06:40.45 | phix | yeah |
| 06:40.48 | phix | gets me everytime |
| 06:46.18 | jplank | grrr I'm wrong |
| 06:46.25 | phix | yay |
| 06:46.51 | jplank | does that first part look right? |
| 06:47.19 | jplank | $["${LEN(${CALLERID(num)})}" < "5"] that always comes out as true no matter what the caller id num is |
| 06:48.01 | jplank | there has to be something obvious I'm missing |
| 06:48.13 | phix | maybe |
| 06:48.16 | phix | time for python |
| 06:48.18 | phix | brb |
| 06:50.20 | jplank | got it, none of it should of been inside quotes |
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| 06:56.04 | schmidts | good morning |
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| 07:01.10 | AndyRomano | gooood morning ;) |
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| 07:08.23 | Burnz1984 | i can set a char bfore the number with set(calleridnum)=abc${calleridnum}) but how i can set a char behind the number? |
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| 07:34.27 | shamelessn00b | good morning |
| 07:34.29 | shamelessn00b | :) |
| 07:36.26 | kaldemar | Burnz1984: your example has numerous errors in it... but, Set(CALLERID(num)=${CALLERID(num)}abc) |
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| 07:39.59 | Burnz1984 | @<kaldemar> it doen´t work :/ |
| 07:40.54 | Burnz1984 | i use asterisk 1.4 on debian |
| 07:42.29 | kaldemar | show a CLI output where it doesn't work. and which version? 1.4 is a branch, not a version. |
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| 07:48.24 | Burnz1984 | cli output: "CALLERID(num)=sasabc" in new stack ; "sas" is the shorthand symbol for a internal number |
| 07:48.41 | Burnz1984 | cli output: "CALLERID(num)=sasabc" in new stack ; "sas" is the shorthand symbol for a internal number |
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| 07:50.33 | kaldemar | how did that not work? |
| 07:51.42 | Burnz1984 | there is no sasabc at the telephone lcd, there is only the sharthand symbol "sas" at the lcd |
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| 07:53.37 | kaldemar | what is the telephone showing? and, it is bad practice to insert characters in the number part of caller id. maybe you want to change the name part, CALLERID(name). |
| 07:55.46 | Burnz1984 | i make a change because of forwarding...someone call person a, an this person is not available, than there is a forwarding to person b. an that person b see it´s a forwarding the shall be a char after the number like: 0123456789F; "F" shall show that it is a forwarding call |
| 07:57.28 | Burnz1984 | i have set the char before the number...thats no problem |
| 07:57.45 | kaldemar | what is the telephone supposed to show on the screen? the caller id name or the number? |
| 07:58.19 | Burnz1984 | the name, but whenn i make the same with callerid(all) it doesn´t work to |
| 07:58.40 | kaldemar | try CALLERID(name) then. |
| 07:58.48 | Burnz1984 | yes with callerid(name it works ^^ |
| 07:58.58 | Burnz1984 | thanks very much |
| 07:59.06 | kaldemar | CALLERID(all) wants a different syntax. |
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| 08:00.00 | Burnz1984 | ok thanks |
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| 08:03.14 | Burnz1984 | but what is when i want to make this for internal call that works with shorthand symbols and for external calls that works with numbers? |
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| 08:03.50 | Burnz1984 | than i have to amake twiwce a set? one for callerid(num) and one for callerid(name)? |
| 08:03.56 | kaldemar | i don't undestand what you want. |
| 08:04.49 | Burnz1984 | when a call comes from outside, have this call a callerid(name) or a callerid(num)? |
| 08:07.04 | kaldemar | depends on many things. use an extension that shows both in CLI to find out your case. |
| 08:07.23 | Burnz1984 | k |
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| 08:09.42 | kaldemar | anyway, number is usually the part that is always set. nothing prevents you from setting CALLERID(name)=${CALLERID(num)}abc for example, if you want. |
| 08:10.24 | kaldemar | the usual case is that the number is all you get in an inbound call from a service provider. |
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| 08:17.26 | stix | Morning guys. Every time my asterisk has an issue it tries to restart itself. This is fine except after the restart it is suddenly running as user root. This results in a lot of scripts not working etc. How can I resolve this? |
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| 08:21.34 | AlHafoudh | hi all |
| 08:21.52 | AlHafoudh | are there some commands that can check status o ooh323 in asterisk? |
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| 08:27.41 | Burnz1984 | one question more...we have a 8 port isdn panel in our astreisk server...6 ports a used by a provider named "versatel" now we have the chance to get a flat rate for 2 of this ports. is it possible to set a priority for this 2 ports that the calls goes a first out over this 2 ports when they free? |
| 08:28.19 | kaldemar | Burnz1984: yes. you do that in your dialplan. |
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| 08:39.06 | aberrios_ | lo |
| 08:45.42 | ChannelZ | high |
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| 09:14.19 | Intel`` | question guys. i would like to know how can i change the filename of the incoming monitoring |
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| 09:15.52 | AlHafoudh | why i dont have correct CALLERID information in dialplanm script when call is coming form SIP trunk? the info is in SIP header but not in CALLERID variable, i get "" |
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| 09:17.46 | v1s | AlHafoudh: how areu calling it? |
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| 09:24.08 | kaldemar | AlHafoudh: variable or function? |
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| 09:27.53 | kaldemar | AlHafoudh: CALLERID variable was deprecated in 1.4 branch, use the function instead. |
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| 09:36.57 | Burnz1984 | kaldemar, you have said i can handle ports with prority? have you some example howto? |
| 09:37.24 | kaldemar | Burnz1984: for example, make an extension that dials the 2 ports first, then checks DIALSTATUS variable that is set by app Dial, and proceeds by the value. |
| 09:38.21 | kaldemar | if the call goes through, ${DIALSTATUS} should equal to "ANSWER". otherwise, you can dial using the rest of the ports (aka spans). |
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| 09:50.54 | teichsta_ | hi |
| 09:51.02 | teichsta_ | which language is spoken here? |
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| 09:53.58 | metiu | hi all, just a quick question: do I have to manage concurrency in the dialplan? I mean, should I worry about two calls entering the same critical section in parallel? |
| 09:54.15 | kaldemar | teichsta_: english |
| 09:55.10 | kaldemar | metiu: depends on what the dialplan does. |
| 09:56.41 | teichsta_ | :kaldemar thanks |
| 09:56.52 | metiu | I mean, if I have e.g. a group(foo) in more than one extension, and I test group_count(foo) before, should I worry about this critical section and use LOCK() UNLOCK() |
| 09:58.50 | teichsta_ | i have problems to get chan_misdn selectable in make menuselect (asterisk 1.8 sourcebuild) - my OS is Ubuntu Server 10.10 / mISDN, mISDNuser is compiled and installed successfully (as far as i can see - misdn_info writes reasonable output) |
| 09:59.02 | teichsta_ | but no chance to make chan_misdn selectable |
| 09:59.52 | teichsta_ | are some hints you can give me? i issued ./configure after mISDN installation |
| 10:00.19 | kaldemar | metiu: it's possible that the conditions change in between |
| 10:01.56 | metiu | kaldemar: so it's necessary to use a mutex to protect the group_count() check and the group() entry... I suspected it |
| 10:02.16 | metiu | but I couldn't find any mention in the group_count() examples |
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| 10:15.57 | Burnz1984 | who know what the "#" in may zaptel.conf do? like: span=1,1,3...... #bchan=5,6 #dchan=7 bchan=1,2 dchan=3 is the "#" for an alternative span? |
| 10:21.28 | kaldemar | Burnz1984: it is a comment character. |
| 10:23.27 | Burnz1984 | like ";"? |
| 10:24.04 | kaldemar | yes. like ; in asterisk configuration files. |
| 10:24.08 | pif | hi, has someone tried the polycom IP 335 with * ? |
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| 10:28.05 | tzafrir_laptop | Burnz1984, but '#' on asterisk configs means something else (basically: a "preprocessor" directive) |
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| 12:21.01 | stix | Morning guys. Every time my asterisk has an issue it tries to restart itself. This is fine except after the restart it is suddenly running as user root. This results in a lot of scripts not working etc. How can I resolve this? |
| 12:22.04 | fauxalliance | http://www.mdinescu.com/making_asterisk_run_as_its_own_user.php @ stix |
| 12:22.23 | fauxalliance | One of the side effects of this kind of setup is that asterisk won't be able to set ToS bits for VoIP packets anymore. That's because setting the ToS requires elevated privileges. |
| 12:22.35 | fauxalliance | amongst other 'features' |
| 12:24.01 | stix | well I don't know what ToS bits are, so don't know if it's a problem :) |
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| 12:25.09 | stix | fauxalliance, but asterisk is already running as user asterisk: "asterisk 6406 1.2 1.2 597584 52436 pts/0 Sl 12:49 0:26 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c" |
| 12:25.26 | stix | it's only when it dies and restarts itself, then it runs as root automatically |
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| 12:36.09 | E-bola | Is there anyway to specify a Set(CHANNEL(musicclass)=Polyteknisk) for a whole context? Its a bit of a pain to have to specify it for every extension... |
| 12:37.01 | russellb | start all extensions at priority 2, and have exten => _X.,1,Set(...) |
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| 12:38.30 | E-bola | ahhh interesting hadnt thought of that |
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| 12:39.26 | E-bola | hmm requires me to re-arrange my context a bit, but i guess it does whaqt i need. Thanks russellb |
| 12:39.35 | russellb | yeah, it's kind of a hack ... |
| 12:39.38 | russellb | you're welcome |
| 12:40.25 | russellb | or you could use 2 contexts, one with the catchall that sets that, and then do a Goto() to the other |
| 12:40.35 | russellb | then you don't have to mess with the extensions at all |
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| 12:46.33 | *** join/#asterisk rajmohan (~rajmohan@122.164.113.66) |
| 12:47.07 | rajmohan | hi |
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| 12:49.05 | Khratos | good morning rajmohan |
| 12:49.34 | rajmohan | good morning khratos |
| 12:50.10 | rajmohan | can you give me nice tutorial / reference to build asterisk as a voip gateway |
| 12:51.32 | Khratos | of course |
| 12:51.39 | Khratos | start reading this: http://asteriskdocs.org/ |
| 12:55.26 | E-bola | Can anybody explain the conclusion of https://issues.asterisk.org/view.php?id=18188 |
| 12:55.53 | E-bola | pabelanger says its been removed recently, but what does that mean? Is it removed in an svn version, or is it removed in 1.8 final |
| 12:56.12 | russellb | the NOTICE was a bug that had already been fixed |
| 12:56.31 | E-bola | So should I be seeing it in 1.8 final? |
| 12:56.47 | E-bola | ahh doh |
| 12:56.54 | E-bola | just ntoiced it says fixed in 1.8.1 at the bottom |
| 13:04.38 | gregd | guys, in freepbx, i'd like to design a custom feature code that simply do SendDTMF(16,250) down the line. Is it so complicated? |
| 13:05.55 | fauxalliance | gregd, perhaps you should try the proper channel (#freepbx) and wait a little longer for a reply... it's still early. |
| 13:06.04 | gregd | ;) |
| 13:06.13 | fauxalliance | however, that does sound trivial. |
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| 13:38.09 | McBoingbo | Are the cool kids still using Sipp to stress test Asterisk? |
| 13:41.30 | fauxalliance | sipvicious? |
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| 13:48.02 | pabelanger | McBoingbo: Sortly yes |
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| 14:55.51 | tzafrir_laptop | McBoingbo, cool kids will give your server a stress test for free |
| 14:56.00 | tzafrir_laptop | Just put it on the internet |
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| 15:12.16 | kjs | yo |
| 15:13.06 | iratik | There is this framework out there called "Vicidial".. We were getting spikes in system load about once every two days⦠Found a slow query log and watched it run a 4 page sql query that took 18 seconds and brought down the box. Just saying â¦. |
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| 15:42.05 | nny | how does asterisk handle a queue in rmmemory if everyone is full? Is it suppose to jump to the next part of the dialplan or put the caller on hold until the queue times out? |
| 15:44.50 | McBoingbo | Asterisk add-ons with 1.8 have become menuselect correct? I want to add cdr mysql support to Asterisk 1.8 |
| 15:45.24 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:45.36 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 15:45.48 | nny | I was/am pretty sure asterisk doesn't do anything with a caller ina queue until the timeout is reached, and then jumps in context. Is this correct? |
| 15:46.13 | nny | (even if the queue's members are all in use or unavailable based on configuration |
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| 16:02.41 | ManxPower | I'm having a problem with MoH. It seems to play the file only once, then you get silence |
| 16:03.20 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 16:05.19 | gregd | is it possible to send a dtmf down a specific channel from a CLI? |
| 16:12.02 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:14.30 | *** join/#asterisk khronos (~tom@38.126.196.5) |
| 16:14.36 | khronos | Hi guys. |
| 16:14.49 | khronos | I have a system running asterisk 1.4.36. |
| 16:15.01 | khronos | What are the requireents for app_voicemailmain? |
| 16:15.13 | Qwell | nothing |
| 16:15.18 | khronos | In teh menuselect I have a selection for app_voicemail but not voicemailmain. |
| 16:15.51 | Qwell | voicemailmain is an app inside of app_voicemail |
| 16:15.52 | khronos | Has this application been removed and replaced by another in this version? |
| 16:16.52 | *** join/#asterisk VonGodric (~albeva@109.231.205.3) |
| 16:18.53 | yonahw | what is module embedding in 1.8 menuselect? anywhere online with more info about the implications? |
| 16:18.53 | _Corey_ | gregd: You could probably use SendDTMF on the channel via Asterisk Manager |
| 16:18.59 | khronos | Ok, if I add the module in to the system do I need to shutdown the asterisk process and start it up again? |
| 16:19.10 | khronos | or can I just do a reload? |
| 16:19.15 | russellb | yonahw: you don't need it. It compiles modules into the main asterisk binary. |
| 16:19.27 | russellb | it was primarily done for platforms that don't support dynamic loading of modules |
| 16:19.34 | yonahw | russellb: thanks |
| 16:25.46 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 16:26.18 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
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| 16:37.39 | ruyo | Is there a feature in Asterisk to return a transfered call to the person who transfered in case it's busy or times out? |
| 16:38.27 | p3nguin_ | In the case of a blind transfer? |
| 16:38.28 | mmlj4 | it's *possible* to do |
| 16:39.28 | ruyo | p3nguin_, either blind or attended. |
| 16:39.53 | ruyo | mmlj4, I mean a "easy" way, without much dialplan programming. :> |
| 16:40.00 | mmlj4 | in that case, no |
| 16:40.31 | ruyo | Ok. I'd better get to work then. |
| 16:43.31 | mmlj4 | a few lines of AGI ought to do it |
| 16:44.56 | mmlj4 | it's a simple matter of capturing the extension of the phone making the transfer, ringing the target phone for 20 seconds, then transferring back to the first phone |
| 16:45.36 | mmlj4 | and maybe not even AGI... |
| 16:46.23 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:50.12 | ruyo | There's a variable, BLINDTRANSFER, maybe that's the easiest way. It's limited to blind transfer, though. |
| 16:50.36 | [TK]D-Fender | You don't need to transfer anything back on an ATTENDED TRANSFER |
| 16:50.46 | [TK]D-Fender | You never let him go |
| 16:52.48 | ruyo | [TK]D-Fender, depends. Sometimes clients want attended transfer capability but still be able to make a semi-blindtransfer. |
| 16:53.09 | ruyo | But you're right, it should be enough. |
| 16:53.54 | ruyo | Even if they need to use diferent keys to make the diferent types of transfer. |
| 16:58.42 | ruyo | Is there a reason for Asterisk not to send a CANCEL to an INVITE when the call was answered on another phone? |
| 16:59.59 | *** join/#asterisk coppice (~chatzilla@p5498ADF6.dip0.t-ipconnect.de) |
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| 17:20.30 | timahvo1 | reloads dahdi modules |
| 17:24.08 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 17:25.13 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 17:28.06 | ShaunR | anybody seen issues with the polycom phones and the newer bootrom/app where it downloads the 000000....cfg file but doesnt listen to the config_files="" params? |
| 17:28.31 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:28.33 | ShaunR | 3.3.1 |
| 17:29.18 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:30.11 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 17:31.10 | yonahw | ShaunR: I had that problem with 3.1.6 the other day |
| 17:31.21 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 17:31.44 | ShaunR | What was the solution? |
| 17:32.06 | ShaunR | i've reset all the configs, formated the fs, and even reset device settings... nothing |
| 17:32.08 | yonahw | I eventually managed to get the phone to upload its app log and saw that it was failing to download the other files. I tried using the credentials for one of the other files in a browser and the phone automagically worked on the next reboot. I have no explanation |
| 17:32.39 | yonahw | ShaunR: is the phone uploading logs to the server? |
| 17:33.02 | ShaunR | i can see it download the 000000000000.cfg file... it also uploads a MAC-boot.log |
| 17:33.41 | ShaunR | 1110173118|copy |3|00|Download of '000000000000.cfg' succeeded on attempt 1 (addr 1 of 1) |
| 17:33.57 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 17:34.07 | yonahw | For me the app log was the key. I was having the problem with a 601 pressing the four arrow keys simultaneously for a few seconds got it to upload the app log. Check the admin guide for your phone if that doesn't help. |
| 17:34.55 | ShaunR | i have no app log |
| 17:37.43 | yonahw | ShaunR: the phone has an app log its just not uploading it. There is a way to force a manual upload of the logs including the app log. On the 601 the mechanism is to press the 4 arrow keys simultaneously for a few seconds. Try doing that with your problematic phone. If that doesn't work check the admin guide for your model to find out how to force a manual upload of the logs. |
| 17:37.55 | *** part/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 17:39.58 | *** join/#asterisk Defraz (~tim@c72co-edge-router.fuzecore.com) |
| 17:41.13 | ShaunR | doing that makes the phone make a noise like it does when issueing a reboot, but server logs show no attempt to upload... |
| 17:41.21 | ShaunR | must be some issue with the connection. |
| 17:41.37 | ShaunR | i seam to recall polycom having a sftp problem... wonder if that still exists. |
| 17:42.01 | yonahw | ShaunR: maybe its the wrong key combination for your phone. |
| 17:42.58 | yonahw | I don't know anything about sftp issues, can't help you there |
| 17:42.58 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:43.25 | ManxPower | I didn't know Polycoms supported SFTP |
| 17:43.37 | ShaunR | there we goo.... |
| 17:43.43 | ShaunR | ManxPower: it's a option.. |
| 17:43.57 | ShaunR | it actually listened and uploaded a app.log now. |
| 17:44.16 | ShaunR | fricken A, that sftp problem existed like 3 years ago... you think they could have fixed it. |
| 17:46.07 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 17:47.43 | ShaunR | yay, it downloaded my config. |
| 17:49.43 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
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| 17:52.00 | yonahw | ShaunR: what was the problem? |
| 17:52.18 | *** join/#asterisk Mango (~iMango@S01060016b6853255.vc.shawcable.net) |
| 17:52.28 | Mango__ | ~itsplist-ca |
| 17:52.28 | infobot | Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
| 17:53.12 | v1s | ~itsplist-ph |
| 17:53.47 | citywok | sorry, no philippines list. just ca & us |
| 17:53.47 | v1s | ~itsplist-us |
| 17:53.47 | 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 |
| 17:54.07 | v1s | is that did providers? |
| 17:54.25 | citywok | i'm pretty sure they all provide DIDs, yes |
| 17:56.37 | v1s | I have for ph if u want to add |
| 17:57.07 | citywok | I'm not sure who manages the bot. one of the opers will know :) |
| 17:57.08 | citywok | ~infobot |
| 17:57.09 | infobot | methinks infobot is in need of training, can someone train me? |
| 17:59.55 | *** part/#asterisk VonGodric (~albeva@109.231.205.3) |
| 18:04.19 | ChannelZ | infobot poop is only on the newspaper or outside |
| 18:04.19 | infobot | ACTION poops on is only on the newspaper or outside |
| 18:04.39 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 18:04.56 | thehar | Has anyone ever had a Polycom IP4000's display stop showing up and all 3 lights on the top flash continously |
| 18:10.21 | hehol | @thehar: yes, I've seen this on a IP4000 with a broken power supply |
| 18:10.33 | thehar | oh really? |
| 18:10.46 | thehar | just the broken power interface module |
| 18:10.47 | thehar | sweet |
| 18:12.11 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 18:16.31 | ShaunR | yonahw: polycom has some type of bug with SFTP config fetching. |
| 18:16.51 | ShaunR | i found it years ago... but figured that with all the versions that have been pushed out over the years that they would have fixed it. |
| 18:17.14 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
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| 18:24.24 | mlsmith9999 | afternoo, still working on my problem, in the mean time, I'm looking for work around. My Idea is simple, everytime one of my employee's encounters the congestion problem, they dial and extension, which sole purpose is to issue the reload command. has anybody done this yet? |
| 18:24.42 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:26.07 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 18:26.23 | marl_scot | any nice folks on here, who could help with an mISDN problem? I have a beronet 4port BRI card, have install mISDN as per (http://www.misdn.org/index.php/Howto_for_Debian) but i cant see the card listed under dmesg , anyone got ay pointers on what im missing? |
| 18:27.27 | marl_scot | ah, hold opn might have it :) |
| 18:29.16 | marl_scot | ok, got the modules install, but the error im getting from misdn_info says that the address family is not supported, how do i find out the address family to use? |
| 18:30.15 | marl_scot | i did an itial configure of mISDN with --with-AF_ISDN=34 |
| 18:30.24 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 18:30.26 | wcselby | o/ |
| 18:31.20 | ManxPower | mlsmith9999, ExecIf($["${DIALSTATUS} = "CONGESTION"],System,/usr/bin/asterisk -rx "reload") |
| 18:31.49 | mlsmith9999 | ManxPower: nice... |
| 18:32.15 | ManxPower | mlsmith9999, but you are still going to have issues receiving calls |
| 18:32.28 | mlsmith9999 | receiving works like a champ... |
| 18:32.31 | ManxPower | You will of course need the correct number of quotes on DIALSTATUS |
| 18:33.01 | mlsmith9999 | gotcha. |
| 18:33.03 | citywok | are you in need of a reload, or a restart when they hit congestion? |
| 18:33.14 | mlsmith9999 | reload fixes it. |
| 18:34.17 | mlsmith9999 | I'm fighting with my ITSP at the moment. about it. Once I get free sec, I'll research alternatives that can port my number.. in the meantime just trying to find a bandaid. |
| 18:35.04 | ManxPower | citywok, his ISP sucks, that is why |
| 18:35.30 | citywok | really, he needs a reload? or just a sip reload which forces a new register? |
| 18:35.53 | came0 | ok im reading the astbook.asteriskdocs.org book and its quite good but im at the point of setting up my ftp and dhcp daemons and im looking at the example dhcpd.conf but i dont see where it specifies the location of the ftp server. does the phone automatically try to ftp to the same ip as the dhcp server? |
| 18:35.55 | ManxPower | citywok, I suspect a sip reload would do it, but he did not ask for that |
| 18:36.12 | ManxPower | came0, yes |
| 18:36.14 | citywok | Gotcha, I was mostly wondering if I had missed something |
| 18:36.42 | Khratos | came0: There should be a setting in the dhcp server that informs the phone about who's going to be the ftp server |
| 18:36.56 | mlsmith9999 | Indeed.. maybe just a sip reload.. need to try that out still. I told it in the SIP.conf to reregister every 30sec's even... but that didn't fix it either, so it's something else. and yes it was confrimed that I need to register before auth. |
| 18:37.25 | mlsmith9999 | Khratos: there is.. option 66 I think if I remeber right. |
| 18:37.26 | citywok | mlsmith9999: have you looked at a sip debug or a tcpdump to see the conversation? |
| 18:38.01 | mlsmith9999 | citywok: yes when it happens the ITSP's server throws back a 403 Forbidden. |
| 18:38.24 | citywok | how long before was the previous chatter? |
| 18:38.58 | came0 | Khratos: ok but by default the phone will look to the same ip address as the dhcp server? |
| 18:39.22 | mlsmith9999 | ManxPower: So could I put that statement in one of the configfiles so that when it acutally try's the trunk it executes the reload and then retry's? |
| 18:39.34 | Khratos | That would depends on the phone, but my experience with Polycom is: 'no' |
| 18:40.25 | mlsmith9999 | came0: that's where the DHCP option 66 come into play, it tells it the tftp server address. IF I'm right. |
| 18:40.51 | came0 | well im using ftp not tftp if that makes a difference |
| 18:40.56 | Khratos | People, Is there a way to set ODBC_STORAGE available without running make menuselect ? |
| 18:41.22 | mlsmith9999 | citywok: previous chatter? as in phone call? Anywhere from 15min's to an hour. Mostly and Hour as far as I can tell. |
| 18:42.15 | citywok | no, as in previous sip messages between your * and the itsp |
| 18:43.29 | mlsmith9999 | citywok: ah.. good question.. I have to look at that and get back to you. I'm having to leave now and was hoping to put this bandaid in place befor I do. |
| 18:43.51 | marl_scot | ok, anyone know how to sort out 'Cannot open mISDN due to Protocol not supported. (Does your Kernel support socket based mISDN?) |
| 18:43.51 | marl_scot | ' when using kernel 2.6.26 and mISDN V2 ? |
| 18:44.26 | wcselby | came0 - what type of phones are you deploying? |
| 18:45.26 | ManxPower | mlsmith9999, you put it in the dialplan after your Dial line |
| 18:47.44 | atan2 | When a SIP phone makes a call outbound it can hang up, but the phone receiving the call keeps ringing. I assume I missed something somewhere. Hmm. Any ideas? |
| 18:48.06 | ManxPower | atan2, if the far end answers does the call work? |
| 18:48.15 | atan2 | Yes, the call does function |
| 18:48.23 | ManxPower | two-way audio? |
| 18:48.50 | atan2 | Yes, there is two-way audio |
| 18:49.01 | atan2 | I just wanted to try it to be sure, but yup, it works |
| 18:49.36 | atan2 | Err, sorry, let me clarify. |
| 18:50.02 | atan2 | If the receiving phone hangs up the calling SIP phone takes awhile to register that the person hung up on them |
| 18:50.10 | atan2 | Perhaps there is some setting for that somewhere? |
| 18:50.19 | ManxPower | atan2, all SIP, no PSTN? |
| 18:51.35 | atan2 | Current setup <sip phone cisco 7940>---<nat router>---<asterisk server>---<voip.ms>---<pstn?>---<who knows what's in here>---<GSM cell phone> |
| 18:52.13 | citywok | atan we only care about the route from your phone to the itsp, we assume the itsp is working properly. |
| 18:52.14 | ManxPower | atan2, then what you are seeing is expected. It takes a few moments for the far end carrier to send "phone hung up" message |
| 18:52.47 | citywok | if it's a second or two that's fine, if you are talkinga bout 20 seconds then you have an issue between asterisk and your phone most likely |
| 18:53.03 | ManxPower | atan2, pastebin the output of a failed call |
| 18:54.25 | atan2 | Would love to. One moment. |
| 18:56.17 | atan2 | http://pastie.org/private/gn96muvu35piq9ostbqvq is the SIP phone making an outbound call to the cell phone. The SIP phone hung up after hearing just one ring (by me pressing hang up) but phone continued to ring for another 15 or 16 rings, which is about 5 or 16 seconds I believe |
| 18:56.51 | atan2 | Also I noticed something funny about how asterisk is sending my caller ID. |
| 18:57.25 | citywok | you need to set verbosity to like 10, and sip debug should probably be turned on |
| 18:57.41 | atan2 | Randomly the caller is is sent as my true 19055551111 number, but I see it has been sent (according to the cell phone) as 01119025551111 =\ hmm |
| 18:58.07 | citywok | callerid isn't really supported internationally |
| 18:58.31 | citywok | if dialing intl, just send your 11digit number and hope for the best. |
| 19:00.00 | atan2 | citywok, the confusion is that it should only be sending the 19055551111 number. I have no idea who added the 0111 on to it =) |
| 19:00.26 | citywok | oh i read it backwards. my bad |
| 19:03.10 | atan2 | UGH!! There are so many lines flying from sip debug I can't copy any of them. Does it output these to a log file somewhere? |
| 19:04.18 | fenrus | you can add logging to file |
| 19:05.09 | atan2 | I see one line that concerns me in the window though, flying around, and that is Sending to 192.168.1.12 : 5060 (no NAT) |
| 19:05.27 | atan2 | As far as I know Asterisk should not be able to know the phones are behind a router, or their IP ? It should be showing my public IP? |
| 19:05.36 | atan2 | and why it says (no nat) also miffs me |
| 19:06.08 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:07.03 | fenrus | if you have nat between your asterisk server and your phones the nating router needs to be 'sip aware' |
| 19:07.15 | fenrus | if it's just regular routing you're fine |
| 19:07.27 | atan2 | fenrus, it's just your run of the mill linksys router? |
| 19:08.36 | [TK]D-Fender | [14:05]<atan2>I see one line that concerns me in the window though, flying around, and that is Sending to 192.168.1.12 : 5060 (no NAT) <- * believes this call came from a place it can trust the CONTACT from. This is your peer configs fault |
| 19:08.50 | *** join/#asterisk cslamar (~cslamar@ampache/staff/cslamar) |
| 19:09.34 | atan2 | [TK]D-Fender, any idea how I might rectify this? |
| 19:09.42 | atan2 | the 192.* ip thing that is. |
| 19:09.47 | [TK]D-Fender | atanFIX YOUR PEER |
| 19:09.52 | cslamar | has anyone had problems with app_fax giving a 'transmit: Transmission error" while running? The strange part is the fax is written to disk but it doesn't seem to exit cleanly? |
| 19:10.31 | atan2 | My peer refers to the [sip] user, or something configured on the phone itself? |
| 19:11.13 | ManxPower | atan, peer = sip.conf setup for device on 192.168.1.12 |
| 19:12.27 | *** join/#asterisk underdog (~underdog@72.46.208.205) |
| 19:12.28 | atan2 | http://pastie.org/private/ufmpgefhwjxvkcwgod2g is what I have in there |
| 19:12.39 | ManxPower | fenrus, you are wrong. Most routers with SIP/NAT support screw it up and it must be disabled on the router |
| 19:13.08 | ManxPower | atan, Callerid does NOT contain quotes |
| 19:13.16 | ManxPower | atan, pastebin the [general] config as well |
| 19:14.12 | fauxalliance | [TK]D-Fender, Holiday tomorrow? |
| 19:14.17 | *** join/#asterisk doctah- (~doctah@76.77.120.30) |
| 19:14.29 | [TK]D-Fender | fauxalliance: Noe, worse-still all-day meeting |
| 19:14.35 | atan2 | ManxPower, http://pastie.org/private/ds86v4wxh4n11pjrlz4a |
| 19:14.51 | fauxalliance | [TK]D-Fender, eww... I think it should be a national holiday nonetheless. |
| 19:15.01 | fauxalliance | Novemberance Day... |
| 19:15.11 | fauxalliance | ugh, Remembrance Day rather |
| 19:15.38 | ManxPower | atan, and the output of ifconfig |
| 19:15.48 | [TK]D-Fender | atan: You aren't looking at at the call. |
| 19:16.57 | atan2 | ifconfig: http://pastie.org/private/ar32vy8gkqj4wg9pjpkf2q |
| 19:17.49 | ManxPower | either you faked your address or you are behind nat |
| 19:17.51 | ManxPower | 111.222.333.444 |
| 19:18.03 | atan2 | I replaced my public ip with 111.222.333.444, sorry |
| 19:18.05 | p3nguin_ | <atan2> citywok, the confusion is that it should only be sending the 19055551111 number. I have no idea who added the 0111 on to it =) <--- No, it should be sending 9055551111 if you are in the US. |
| 19:18.21 | ManxPower | atan2, you didn't do something stupid like set the phone config on the actual phone for NAT, did you? |
| 19:18.36 | gregd | guys, im trying to senddtmf() down the line, I have an extension that does it, CLI shows the execution, however it does not look like it goes the right channel (it goes back to me instead down to the channel, is it possible at all? |
| 19:18.36 | *** join/#asterisk darksk1ez (~mhb@2001:470:9272:1::1) |
| 19:18.47 | atan2 | p3nguin_, I thought I was to provide the '1'. I'll remove it. |
| 19:19.00 | ManxPower | gregd, that is what it does. what are you trying to accomplish? |
| 19:19.26 | atan2 | p3nguin_, thank you. Removing the 1 resolved the caller ID issue |
| 19:19.50 | atan2 | But not the whole issue of it continuing to ring the phone I am calling =\ |
| 19:19.56 | gregd | ManxPower: senddtmf(flash), senddtmf(1234) down the line when I dial feature code |
| 19:20.22 | atan2 | ManxPower, yes I think I did. |
| 19:20.30 | atan2 | It says NAT: Yes but Nat address: blank |
| 19:21.48 | p3nguin_ | atan2: Is Asterisk behind NAT? |
| 19:22.10 | atan2 | No? Just the SIP phone. |
| 19:22.23 | atan2 | I don't believe it is anyway. It has a public IP address & so on.... |
| 19:22.29 | p3nguin_ | The asterisk system is connected right to the modem? |
| 19:22.53 | atan2 | The Asterisk box is sitting at Rackspace. How it's connected up I'm not 100% sure. |
| 19:23.09 | ManxPower | gregd, features.conf |
| 19:23.09 | atan2 | The SIP phone is sitting on my desk, going through a little linksys router thinger. |
| 19:23.29 | ManxPower | atan2, stop mumbling and tell us your router |
| 19:23.58 | atan2 | ManxPower, WRT300N running dd-wrt |
| 19:24.10 | ManxPower | atan, disable any SIP NAT features. |
| 19:24.20 | ManxPower | aka SIP ALG |
| 19:24.21 | atan2 | I changed the phone setting from NAT=yes to NAT=no and the phone is no longer connecting |
| 19:24.30 | *** join/#asterisk Preytell (~jerry.win@68-188-27-90.static.stls.mo.charter.com) |
| 19:24.31 | ManxPower | atan, the phone must be set to no |
| 19:25.17 | ManxPower | when you If you SIP NAT twice (asterisk AND the phone), nat support is DISABLE when the 2nd device tries to fixup the call for bat |
| 19:25.19 | ManxPower | nat. |
| 19:25.32 | *** join/#asterisk Roconda (~roconda@blackmind.roconda.nl) |
| 19:25.44 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:25.48 | p3nguin_ | You should set the actual phone's NAT setting to disabled and let Asterisk deal with the NAT. This is a typical configuration. |
| 19:26.46 | atan2 | This is going to sound silly, but how do you disable NAT on a router? It has a NAT / QoS tab but inside it are port forwards, range forwards, port triggers, upnp, dmz & qos |
| 19:27.01 | ManxPower | p3nguin_, he is using DDWRT so chances are the router is as well |
| 19:27.05 | p3nguin_ | You don't disable NAT on the router. |
| 19:27.13 | ManxPower | atan2, are you doing any port forwarding |
| 19:27.19 | ManxPower | atan, disable SIP NAT not NORMAL NAT |
| 19:27.28 | p3nguin_ | You need NAT on the router to provide internet for the LAN. |
| 19:27.28 | atan2 | ManxPower, there are no port forwards |
| 19:27.42 | ManxPower | atan, is there an ALG or application page? |
| 19:29.09 | atan2 | There is a Services tab, with Milkfish SIP Router in it |
| 19:29.12 | atan2 | But that's disabled |
| 19:29.49 | atan2 | ManxPower, nope I do not see any |
| 19:29.55 | p3nguin_ | Did you ever try to milk a fish? |
| 19:30.04 | atan2 | ...actually |
| 19:31.18 | atan2 | I suppose I should forward the ports to the device? |
| 19:32.40 | ManxPower | atan, only if you want to guarntee it does not work |
| 19:32.52 | ManxPower | ~sipnat |
| 19:32.52 | infobot | rumour has it, sipnat is 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 |
| 19:32.58 | ManxPower | why don't you go read those docs |
| 19:33.23 | atan2 | Thanks =) |
| 19:35.40 | atan2 | Can't get to http://www.aocomputing.net/?p=3 =\ the other link work fine though |
| 19:36.09 | atan2 | I assume I would be #8? "Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk " |
| 19:36.29 | gregd | ManxPower: I've added the script in features.conf but still it everything is send back to me instead of down the line |
| 19:36.43 | atan2 | In the list below it says "#8 is no problem. No NAT in the middle " :P |
| 19:43.27 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 19:44.10 | Micc | I've set maxexpirey=160 in sip.conf general section, but my peers still are able to register with expire of 3600 |
| 19:44.56 | [TK]D-Fender | This is ridiculous... |
| 19:45.21 | [TK]D-Fender | atan2: Pastebin the ENTIRE FUCKING CALL with SIP DEBUG enabled along with your sip.conf masking ONLY paswords. |
| 19:46.07 | [TK]D-Fender | atan2: You last showed 1 line to show where it was responding to, but not WHY because the entire start of the call you didn't feel was relevant to determining WHERE to respond to. This is wrong. |
| 19:46.13 | p3nguin_ | atan2: I think you are #9 on that list. |
| 19:46.36 | [TK]D-Fender | I think.... |
| 19:46.38 | p3nguin_ | atan2: You have a NAT. You have: Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk |
| 19:46.41 | [TK]D-Fender | ~wmmfpb |
| 19:46.41 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 19:47.11 | ManxPower | gregd, then you have the ,<ActivateOn>[/<ActivatedBy>] wrong |
| 19:47.34 | atan2 | [TK]D-Fender, lol. Sorry. I changed nat=no on the phone and now it won't even connect up. Just getting that sorted out then I'll get that debug out =) |
| 19:48.03 | ManxPower | atan, define "nat=no" "on the phone". Do you mean "on the phone" or do you mean "in sip.conf" |
| 19:48.26 | atan2 | ManxPower, in sip.conf nat is set to yes. On the phone, inside the menus, it is set to no |
| 19:48.35 | ManxPower | atan, good |
| 19:51.34 | atan2 | I wish this 7940 would start up quickier each time =\ it's as if it loads Windows ME every time I unplug it. |
| 19:54.37 | gregd | ManxPower: so how do I design a feature to senddtmf down the line and bridge back? |
| 19:55.16 | atan2 | Okay, we're getting somewhere here now. I just asked my provider if my server was behind a nat, and they replied "Michael Q: Yes, they are behind a bridge using NAT" |
| 19:55.23 | *** part/#asterisk nrc2 (~nrc@ip-38-95.sn1.eutelia.it) |
| 19:56.20 | ManxPower | gregd, in features.conf |
| 19:56.32 | ManxPower | gregd, are you using a GUI for Asterisk |
| 19:56.39 | atan2 | They go on to say "The server is still wide open to the internet, especially once your OS firewall has opened its ports" >_< |
| 19:57.30 | gregd | ManxPower: i got his point :) yes I use gui, but it does not matter.. I'm happy to edit the files itself for testing purposes |
| 19:59.17 | ManxPower | gregd, except the GUI will re-edit the files anytime you make a change in the GUI |
| 19:59.27 | ManxPower | gregd, look on the Wiki, it should work. |
| 19:59.55 | ManxPower | gregd, "bridge back"? |
| 20:00.20 | Preytell | anyone seen an issues where outbound calls choose a trunk that is ringing, therefore the outbound call anwsers the incoming line. |
| 20:00.35 | Preytell | this is DAHDI/Pots btw. |
| 20:00.40 | ManxPower | Preytell, that is called "glare" |
| 20:00.48 | gregd | ManxPower: meaning, if I put the call on hold, I got access to features. So I mean, that after executing the feature I'd like to remove the hold |
| 20:00.54 | citywok | Preytell: i had that happen when i used 24 port analog FXS/FXO cards and both sides opened the same trunk at the EXACT same time |
| 20:01.18 | ManxPower | gregd, don't put the call on hold. when you do that your features won't apply to the held line |
| 20:01.33 | Preytell | man, I am using an 8 port digium card, and it happens often. |
| 20:01.36 | ManxPower | Preytell, chage your g1 (or whatever group number you use) to G1 |
| 20:02.04 | ManxPower | if the telco starts at the bottom and moves on up, then you want to start at the top and work your way down. search for "glare" on the wiki |
| 20:02.28 | Preytell | sorry, but which wiki? |
| 20:02.36 | ManxPower | !answers |
| 20:02.39 | ManxPower | ~answers |
| 20:02.39 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 20:02.45 | ManxPower | tge voip-info.org one |
| 20:02.45 | Preytell | thanks |
| 20:03.48 | gregd | ManxPower: but if I dont put it on hold, the asterisk feature is not executed |
| 20:04.37 | Preytell | fun. |
| 20:04.57 | citywok | Who maintains the infobot? |
| 20:05.04 | gregd | *im not able to execute a feature |
| 20:05.10 | ManxPower | gregd, then something ELSE is wrong |
| 20:05.17 | citywok | The asterisk wiki is now asterisk.wiki.org :P |
| 20:05.28 | citywok | whoops, dyslexic much? wiki.asterisk.org |
| 20:05.56 | ManxPower | citywok, but the voip-info.org wiki is the voip-info.org wiki |
| 20:06.21 | ManxPower | the only thing wiki.asterisk.org has on it, as far as I know, is ONLY docs for Asterisk 1.7 |
| 20:06.27 | ManxPower | ..er.. 1.8 |
| 20:07.45 | Qwell | it's a wiki. Add to it. |
| 20:08.08 | Preytell | Ok, one other issue, (And I really hate pots lines), at times when a caller attempts to make an outbound call the system will show that all channels are busy and play the congestion tone. But there are zero lines in use and incoming calls still work on the same pots lines that the system will not allow an outbound call. ???? |
| 20:08.10 | *** join/#asterisk funslug (~muiro@unaffiliated/muiro) |
| 20:08.16 | *** part/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 20:08.56 | *** join/#asterisk elguero (~miguel323@12.187.84.162) |
| 20:09.04 | funslug | I have a question about cmd MYSQL from the asterisk-addons package. How can I specify a port in the Connect command? |
| 20:09.06 | Preytell | where does asterisk keep track, read that "show", you which lines it thinks are in use. Because the core show channels does not show it. |
| 20:09.49 | Preytell | /proc/dahdi/1 shows zero lines in use... I don't understand where the decision that a line is in use comes from. |
| 20:11.13 | ManxPower | Preytell, are you using a GUI for Asterisk? |
| 20:11.22 | tzafrir_laptop | Preytell, a line is in use when a userspace program opens it |
| 20:11.33 | tzafrir_laptop | That program is normally Asterisk |
| 20:11.56 | tzafrir_laptop | If it's asterisk it means that this channel is available for Asterisk to use |
| 20:11.57 | Preytell | yeah, and to make matters worse I am using piaf, asterisk 1.8, and freebpx 2.3 |
| 20:11.58 | ManxPower | Preytell, core show channels shows you the channels that are in use. |
| 20:12.08 | ManxPower | Preytell, why don't you ask on #FreePBX? |
| 20:12.26 | tzafrir_laptop | freepbx 2.3? Does it work with asterisk 1.8 (manager interface, speicifcally) |
| 20:12.27 | tzafrir_laptop | ? |
| 20:12.50 | Preytell | sorry freepbx 2.8 |
| 20:12.57 | ManxPower | Preytell, since you are using FreePBX virtually nothing we tell you will apply. |
| 20:13.31 | Preytell | I guess, but the underlying system, and the dahdi layer should apply to all implementations. |
| 20:13.32 | ManxPower | Preytell, the decision as to what line to use is set up in the Asterisk dialplan |
| 20:13.44 | ShaunR | anything with the queues change from 1.6.0.13 to 1.6.2.13? calls are put into the queue but asterisk doesnt seam to be doing anything with the call after that |
| 20:13.50 | ManxPower | Preytell, you are wrong Mostly |
| 20:14.04 | [TK]D-Fender | [15:08]<Preytell>Ok, one other issue, (And I really hate pots lines), at times when a caller attempts to make an outbound call the system will show that all channels are busy and play the congestion tone. But there are zero lines in use and incoming calls still work on the same pots lines that the system will not allow an outbound call. ???? <-- systenm will show? O RLY? |
| 20:14.10 | ManxPower | ShaunR, if it did it should be listed in the UPGRADE*.txt files that comes with your Asterisk source |
| 20:14.33 | citywok | ShaunR: I went from 1.6.0 to 1.6.1 and then 1.6.2 and I don't think i've done anything to my queues |
| 20:14.53 | ManxPower | Preytell, your problem is not at the DAHDI later, it is at the Asterisk later |
| 20:15.41 | ManxPower | Preytell, Your problem has been experienced by people with PBXs for 20 or 30 years. We know what the issue is, we know what the solution is. (Assuming you correctly described the problem_ |
| 20:16.11 | ManxPower | now, go change your outbound group to hunt from the top down instead of the bottom up. |
| 20:16.35 | Qwell | glare isn't going to be solved the same way on analog... |
| 20:16.42 | ManxPower | On regular Asterisk you do that by changing from a lower case "g" to a uppercase "G" in your Dial line. |
| 20:16.57 | ShaunR | with penalty 10 (Invalid) has taken no calls yet |
| 20:17.03 | ShaunR | hmm, wonder what the invalid part is about |
| 20:17.09 | ManxPower | It will not eliminate the issue, but it will happen much less frequently. |
| 20:17.18 | Preytell | no, I understand the glare thing. And I will change the order. |
| 20:17.19 | [TK]D-Fender | ShaunR: You are using chan_local for members and the modules loaded in the wrong order |
| 20:17.43 | [TK]D-Fender | ShaunR: app_queue tried referring to a channel type that did not exist at the time it loaded |
| 20:18.07 | [TK]D-Fender | ShaunR: you need to preload it in modules.conf |
| 20:18.11 | ShaunR | so chan_local needs to load before queue |
| 20:18.21 | [TK]D-Fender | yes |
| 20:18.24 | Qwell | ManxPower: how do you figure? incoming calls on analog lines aren't going to be predictable at all. |
| 20:18.47 | ShaunR | i had load => chan_local.so before the load => app_queue.so |
| 20:18.50 | Qwell | You'd have just as much luck by using a random selection for outgoing. |
| 20:18.53 | citywok | Lol, set the other end to go the other direction sequentially. |
| 20:18.59 | ShaunR | i have to preload then? |
| 20:19.03 | Preytell | correct, you have no control over which line is dialed if the public has the numbers assoc with the other lines. |
| 20:19.06 | citywok | That's what I did when i had to deal with stupid analog cards |
| 20:19.16 | [TK]D-Fender | ShaunR: Yes. YES. ***YES*** |
| 20:19.29 | ShaunR | [TK]D-Fender: do i need to preload all my chans then? |
| 20:19.29 | [TK]D-Fender | ShaunR: Capiche? |
| 20:19.36 | citywok | [TK]D-Fender: don't blow a gasket |
| 20:19.38 | [TK]D-Fender | ShaunR: not a bad idea |
| 20:19.55 | Preytell | I agree that POTS sucks, I which everyone would let me use sip trunks, pots for failover. |
| 20:20.21 | ManxPower | um, no. They should use PRI |
| 20:20.33 | Preytell | too expensive where I live. |
| 20:20.37 | funslug | how can I specify a port to use with cmd MYSQL Connect? |
| 20:20.49 | Preytell | only a few customers will opt for pri over pots. |
| 20:20.56 | ManxPower | Preytell, and running telephone calls over the internet is unreliable |
| 20:21.03 | Preytell | nod |
| 20:21.24 | Preytell | but I have been using flowroute, and not had a problems for over a year... |
| 20:21.31 | ManxPower | we run tens of thousands of calls each day over VoIP, but they never touch the internet |
| 20:22.44 | citywok | Yes, having SIP termination with your internet carrier directly, riding their network 100%, with their QoS makes things much better. |
| 20:23.27 | Preytell | I wish that AT&T/SBC would offer PRI service for a decent rate to business in IL/MO. It would make things cleaner. |
| 20:23.36 | ShaunR | [TK]D-Fender: thank you, working again... |
| 20:23.44 | ShaunR | for SIP tcp connections what should i set the expires too? |
| 20:23.57 | Preytell | ok, off to move trunks around to avoid glare. :) |
| 20:24.06 | citywok | Preytell: what is a "decent rate"? |
| 20:25.19 | Qwell | Preytell: ask about a partial PRI |
| 20:25.27 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 20:31.02 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 20:33.31 | *** join/#asterisk hectaman (~hectaman@82.218.68.216.DED-DSL.fuse.net) |
| 20:39.37 | *** join/#asterisk fofware (~Fabian@host127.200-82-50.telecom.net.ar) |
| 20:40.51 | hectaman | when "dahdi show status" shows me CRC4 errors on a T1 is there a way to get more detail as to when those errors took place? |
| 20:41.18 | citywok | hectaman: not that i know of, but my experience was all with zaptel not dahdi |
| 20:41.30 | citywok | if there's any way to clear counters i'd do that and see if it's still happening |
| 20:42.51 | hectaman | I'm occassionally dropping calls on one of three circuits; the first two are PRI's, the third is an LD T1 |
| 20:45.33 | hectaman | I'm less than 20 feet from the smartjack; i thought about maybe changing the line build-out to see if I can reduce the occurance |
| 20:45.35 | *** join/#asterisk kannan (1004@115.252.89.44) |
| 20:46.47 | kannan | hello, in asterisk 1.4.35 , i want to get a 'hot key' in IVR , like press 9 at any time during this ivr to do something..how to acheive that? |
| 20:47.14 | WIMPy | hectaman: "Smartjack" meaning modem? |
| 20:48.23 | hectaman | the demarc where the carrier hands us the T1 line |
| 20:49.17 | atan2 | Does the MagicJack use SIP? =S |
| 20:49.35 | atan2 | Err, so it does. |
| 20:49.47 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 20:49.47 | Qwell | atan: effectively no |
| 20:50.14 | atan2 | Qwell, well I was not making any comment on the quality of the lump of plastic. I was just curious =) |
| 20:51.14 | *** join/#asterisk _zoom_ (~user@41.218.36.192) |
| 20:51.30 | _zoom_ | hey fellas, how to get rid of FAS |
| 20:51.31 | _zoom_ | ? |
| 20:51.41 | _zoom_ | is it possible |
| 20:51.43 | _zoom_ | ? |
| 20:52.01 | pabelanger | kannan: Set it up in your dial plan |
| 20:52.08 | kannan | another question, in an IVR , i have an agi script after WaitExten that sometimes takes upto 10 seconds to execute ; the ivr says please wait , but then there is silence for those 10 seconds, is it possible to set music or something during the time when the agi script is executing .. |
| 20:52.08 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 20:52.26 | kannan | pabelanger , thru features.conf ? and DYNAMIC_FEATURES? |
| 20:52.51 | pabelanger | kannan: no, exten => 9,1,Dosomething() |
| 20:53.38 | pabelanger | Dynamic_features would work too actually |
| 20:54.05 | [TK]D-Fender | No |
| 20:56.52 | phix | hey Mr [TK]D-Fender |
| 20:56.54 | kannan | can comfort noise be set on the DAHDI channel? |
| 20:57.51 | [TK]D-Fender | kannan: No such thing |
| 20:58.27 | p3nguin_ | Asterisk doesn't support comfort noise, anyway. |
| 20:58.58 | kannan | [TK]D-Fender, thanks , i was wondering if something could possibly done to avoid the log silence, on the DAHDI channel it feels like a dead silent |
| 20:59.03 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 20:59.10 | kannan | long silence i meant |
| 21:00.59 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 21:01.00 | kannan | also, in a E-1 crossover to a legacy PBX, any outbound call from the phones connected to the legacy pbx goes to 's' exten in dialplan context only, so i am forced to use a Background to enable dialling out. Setting immediate=no , in chan_dahdi.conf did not change this.. how to get the asterisk dialtone? |
| 21:01.15 | ManxPower | kannan, you mean something like Ringing or MusicOnHold? |
| 21:01.16 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 21:01.30 | kannan | ManxPower , moh is what i thought |
| 21:01.57 | ManxPower | Maybe you just have a badly designed dialplan? Are you sure the AGI is what is taking so long? |
| 21:02.40 | [TK]D-Fender | E1 is not a channel signalling standard |
| 21:02.41 | kannan | ManxPower , the agi uses cURL, the server from which the response is given is sometimes slow, some netwrok issue also |
| 21:02.44 | [TK]D-Fender | get SPECIFIC |
| 21:03.17 | ManxPower | kannan, sounds like MoH is what you need |
| 21:04.06 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:04.39 | kannan | [TK]D-Fender, Euroisdn, CCS , PRI NET on this dahdi trunk |
| 21:04.39 | funslug | okay, lol, cmd MYSQL does not have the ability to take a port in asterisk-addons-1.6.0 or 1.6.2 |
| 21:04.54 | [TK]D-Fender | Katry actually looking at the call |
| 21:04.56 | funslug | that's a pretty nasty oversight |
| 21:05.02 | [TK]D-Fender | kannan: try actually looking at the call |
| 21:05.12 | kannan | ManxPower , ok thanks i will lokk it , so if i set MOH , then it will provide music automatically if there is silence? |
| 21:05.22 | Nugget | anything that's a deterrent to using mysql is a feature. :) |
| 21:05.35 | citywok | lol, realtime odbc?!? |
| 21:05.35 | ManxPower | kannan, it Plays music on hold. "silence" is meaningless |
| 21:06.23 | funslug | luckily, easy to fix |
| 21:06.24 | kannan | ManxPower , oh so i start and stop the music-on-hold when needed |
| 21:07.01 | kannan | [TK]D-Fender, by pri debug on the span? (i cannot understand the output) |
| 21:08.14 | *** part/#asterisk _zoom_ (~user@41.218.36.192) |
| 21:09.37 | *** join/#asterisk JunK-Y (~junky@64.15.77.94) |
| 21:09.47 | pabelanger | anybody have a polycom 301 kicking around? tell me the AC adapter stats? DC 12V ..... mA? |
| 21:10.52 | ManxPower | the 330 and 560 I have are 24VDV 500mA |
| 21:11.31 | kannan | [TK]D-Fender, when the legacy PBX 's phone dials , they have to press 0, to get a dial tone on this span. This works when the PBX is connected to PSTN - dialtone , then after dial the number, it calls out. But with Asterisk as PRI Net , it simply dials the zero alone (in 's' exten), which of course there is no dialplan till i use a Waitexten |
| 21:12.31 | ManxPower | kannan, PRIs don't provide dialtone |
| 21:13.03 | WIMPy | Sure, PRIs provide dial tone if you don't dend any digits. |
| 21:13.18 | ManxPower | No. They. Do. Not. |
| 21:13.27 | WIMPy | But overlap dialling and Asterisk can be quite tricky. |
| 21:13.33 | florz | Yes. They. Do. |
| 21:13.44 | ManxPower | you don't even have a Bchannel up to the telco when you are dialing. |
| 21:14.02 | ManxPower | How exactly will the dialtone audio get from the telco to your endpoint? |
| 21:14.03 | WIMPy | You sure do. |
| 21:14.17 | ManxPower | I wish you the BEST of luck. |
| 21:14.22 | WIMPy | I the B channel you requested. |
| 21:14.29 | WIMPy | In |
| 21:14.40 | ManxPower | WIMPy, you have not requested a dialtone when when you start dialing |
| 21:14.46 | ManxPower | requested a B-channel |
| 21:14.56 | [TK]D-Fender | kannan: Indeed, this is live-overlap... you'll have some tricky dialplan to minimize the waiting |
| 21:15.15 | WIMPy | ManxPower: You usually won't but you could. |
| 21:15.41 | WIMPy | And kannans PBX seems to do. |
| 21:15.47 | ManxPower | How would the PRI detect you "went off hook" when you don't go "off hook" when you are dialing. |
| 21:16.15 | WIMPy | Which I have to gove in seems rather unusual and I haven't seen before. |
| 21:16.21 | ManxPower | he is confused. |
| 21:16.51 | WIMPy | ManxPower: By sending a setup without called party number, as every ISDN phone also does. |
| 21:17.05 | ManxPower | of course by the time he has gotten to the "s" extension you are all done with "PRI" and are in Asterisk./ |
| 21:17.21 | ManxPower | WIMPy, ISDN phones generate local dialtone AFIK |
| 21:17.24 | yonahw | just installed 1.8, made test call, call works but cli doesn't show any call information. verbosity is set to 10. any clues as to where I'm going wrong here? |
| 21:17.26 | ManxPower | Just like SIP phones do. |
| 21:17.36 | ManxPower | yonahw, maybe you did not set up /etc/asterisk/logger.conf? |
| 21:17.39 | kannan | [TK]D-Fender, in fact the same config works fine on another Legacy PBX but not with this one. And the delay in dialling out is what i want to minimize. Otherwise actually, i used a 3 second dialtone.wav with Background |
| 21:17.52 | yonahw | ManxPower: maybe, checking |
| 21:17.57 | kannan | so the users cannot figure out , but it dials out after some 8 seconds |
| 21:18.06 | ManxPower | yonahw, you, read all the UPGRADE*.txt files, right? |
| 21:18.08 | WIMPy | No, they don't. |
| 21:18.30 | yonahw | ManxPower: thought so, but apparently I missed something important |
| 21:18.37 | ManxPower | What ISDN phones out there support PRI anyway? |
| 21:18.51 | WIMPy | ISDN is usually 100% interactive, AKA overlap dialling. |
| 21:18.53 | p3nguin_ | That seems very contradictory. |
| 21:18.55 | WIMPy | None |
| 21:18.59 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 21:19.22 | WIMPy | But the number of B channels on the onterface doesn't make much difference to the signalling. |
| 21:19.35 | ManxPower | BTW, "sending a call with no digits" still send the call to the CO and the CO sends out the dialtone |
| 21:19.46 | florz | ManxPower: see Q.931, section 5.1.3 |
| 21:19.53 | ManxPower | You are still not going "off hook" |
| 21:19.55 | WIMPy | That's the idea. |
| 21:20.21 | florz | ManxPower: whatever you mean by "off hook", PRIs can supply a dialtone |
| 21:21.13 | kannan | with a Samsung PBX , the E1 crossover is fine, but with another PBX (Matrix PBX), this 's' exten thing happens |
| 21:21.26 | ManxPower | Sounds like a simple form of DISA to me. |
| 21:21.30 | yonahw | ManxPower: the upgrade docs don't seem to produce anything in regards to logger.conf other than the removal of LOG_EVENT |
| 21:21.38 | yonahw | runs off to read up on logger.conf |
| 21:21.40 | ManxPower | kannan, "exten s" means "we received no digits |
| 21:21.56 | kannan | ManxPower, yes, thanks, i had understood that |
| 21:22.17 | ManxPower | kannan, start looking at pri debug and make sure the PBX is sending digits |
| 21:22.22 | ManxPower | then you won't have any delay at all |
| 21:22.38 | florz | ManxPower: no, the dialed digits are (probably) not transmitted in-band, or in end-to-end signalling |
| 21:23.53 | WIMPy | kannan: something like s,1,WaitExten(20) should do. |
| 21:24.07 | ManxPower | florz, if they are not transmitted in-band and not are transmitted in the signalling, how are the digits being senr |
| 21:24.11 | p3nguin_ | If the PBX is connected to Asterisk via analog lines, I'd expect exten s to be used on every call. |
| 21:25.21 | kannan | WIMPy, ok i got it , to set the timeout low is working OK on the 's' now |
| 21:25.33 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 21:25.58 | ManxPower | kannan, do you have something like immediate=yes? |
| 21:25.59 | *** join/#asterisk kriebz (~matthewk@office.cawinet.com) |
| 21:26.12 | kannan | i was using background (to provide a false dialtone with a wav file ) and then waitexten , so there was the delay |
| 21:26.33 | kannan | ManxPower, no , i set chan_dahdi conf to have immediate=no |
| 21:26.33 | *** part/#asterisk kriebz (~matthewk@office.cawinet.com) |
| 21:28.23 | ManxPower | kannan, which timeout did you set too low? |
| 21:28.52 | pabelanger | Wish me luck. First time attempting polycom provisioning via Asterisk (res_phoneprov.so) |
| 21:29.30 | kannan | i had not set anything on the timeout like SET(aboule or digit) i came out wrong with the word timeout, I had s,1,Background then, as next priority waitexten(5) |
| 21:30.29 | kannan | now the dialtone.wav plays for 3 seconds, i removed the waitexten, the outbound dialling is better in terms of delay |
| 21:31.12 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 21:32.15 | WIMPy | Well, as I said: creating an overlap friendly dial plan is tricky. |
| 21:32.56 | kannan | what i didnt get is hwo come the same thing works with one PBX and not another |
| 21:33.01 | WIMPy | In fact, I think it's only partially possible. |
| 21:33.26 | WIMPy | Because they're not the same? |
| 21:34.14 | kannan | WIMPy , ha ha yeah well they have to live with it, as they have a buy back from Samsung, and the other is provided free by the telecom company based on some billing minimum commitments |
| 21:34.18 | *** join/#asterisk [canniballllera] (~cannibale@201-2-232-3.fnsce703.dsl.brasiltelecom.net.br) |
| 21:35.14 | florz | ManxPower: I said "end-to-end signalling" |
| 21:38.28 | kannan | thanks for all the help , be back shortly with another list of issues (i cannot get callerid on incoming FXO line), but i am off to get some sleep today, thanks again , bye |
| 21:40.07 | citywok | hmm, so i found the bad line in ne_request.c from libneon. anybody see anything wrong with this line? ne_set_error(sess, "%s: %s", doing, ne_sock_error(sess->socket)); |
| 21:40.28 | citywok | if i change it to sess, "%s: error at line 208" and remove the ne_sock_error part, it works just fine |
| 21:49.20 | *** join/#asterisk ccesario (~ccesario@201-42-150-145.dsl.telesp.net.br) |
| 21:56.33 | *** join/#asterisk E-bola (~bola@x1-6-00-13-46-83-e5-04.k1098.webspeed.dk) |
| 21:59.35 | marl_scot | is it posible to make asterisk use port 4569 TCP as well as/ instead of 4569 UDP ? |
| 21:59.46 | atan2 | Can anyone guess at why if I make three calls, the first two exit correctly when I hang up but #3 continues to ring even if the caller has hung up? |
| 21:59.56 | marl_scot | for IAX port forwarding |
| 22:00.13 | p3nguin_ | IAX2 uses UDP, so what do you care about TCP for? |
| 22:00.22 | WIMPy | marl_scot: no |
| 22:00.57 | marl_scot | :( trying to test an * box behind a firewall that i dont have control over, was wanting to use putty to port forward :( |
| 22:01.32 | marl_scot | in past have have full control of routers, so not hit the problem :( |
| 22:01.56 | WIMPy | marl_scot: You can use openvpn, but the usual warnings about tunneling RTP over TCP apply. |
| 22:02.39 | marl_scot | ok, was wanting to avoid that, but looks like i will have to bite the bullet :(# |
| 22:02.45 | marl_scot | thanks :) |
| 22:03.03 | WIMPy | Yes, you probably want to avoid that. |
| 22:05.31 | p3nguin_ | http://code.google.com/p/udptunnel/ |
| 22:05.50 | p3nguin_ | Wait, that's backward. |
| 22:05.55 | p3nguin_ | disregard. |
| 22:06.06 | marl_scot | thanks p3nguin_ :) |
| 22:06.20 | p3nguin_ | What about using socat or netcat? |
| 22:06.55 | marl_scot | never had a lot of luck with netcat on windows :( |
| 22:07.10 | p3nguin_ | This looks like your solution: http://nardcore.org/ctunnel/ |
| 22:07.27 | p3nguin_ | They don't know English, but maybe they can write a tunnel app. |
| 22:08.17 | atan2 | Is there a cheap 411 provider out there? |
| 22:08.27 | p3nguin_ | oh |
| 22:08.29 | p3nguin_ | 411.com |
| 22:08.44 | p3nguin_ | Free 411 (Bing 411) |
| 22:08.45 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 22:08.46 | atan2 | p3nguin_, to forward my phone calls to? |
| 22:08.52 | atan2 | For Canada, sorry |
| 22:09.02 | atan2 | Bing-411 doesn't support the polar bears |
| 22:09.08 | p3nguin_ | Actually, I'm full of fail today... |
| 22:09.38 | p3nguin_ | Free 411 and Bing 411 are not the same. I was thinking of 800-CALL-411 (it's Bing). |
| 22:09.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:10.12 | p3nguin_ | I'm all messed up. I think I overdosed on Tramadol. |
| 22:10.32 | p3nguin_ | drowsy, puking, generally feel like shit |
| 22:10.45 | atan2 | If only their 411 databses were public records :P one could invent a service that costs $0.10 and provides the same features |
| 22:10.50 | atan2 | Everyone would use it. I swear. |
| 22:10.52 | marl_scot | p3nguin_, not to much of an overdose, that ctunnel thing looks like it could be REALLY handy :) |
| 22:10.54 | atan2 | Everyone without a computer anyway. |
| 22:19.55 | atan2 | Hmm... I went wrong somewhere here. exten => _*661NXXNXXXXXX,1,Set(callerid=0000000000) |
| 22:20.10 | atan2 | I just want to set it for the current call, not forever |
| 22:20.18 | ManxPower | atan2, not using the correct syntax |
| 22:20.35 | *** join/#asterisk hipitihop (~denis@202.153.71.87) |
| 22:20.35 | p3nguin_ | Set(CALLERID(num)=1234567890) |
| 22:22.09 | [TK]D-Fender | atan2: And that isn't "forever". These are CHANNEL variables. Guess when they die... |
| 22:23.08 | atan2 | :P =) |
| 22:23.57 | atan2 | So _*661NXXNXXXXXX would match *6619055551212 without any troubles but of course it would pass the entire *66+ portion of it over to ${EXTEN}. Any advice on how to snip the *66 off it? |
| 22:24.29 | atan2 | I wouldn't just remove the *66 from line #2 would I? |
| 22:25.05 | [TK]D-Fender | atan2: CHANNELVARIABLES.TEX <- read it |
| 22:25.22 | a1fa | awww |
| 22:25.25 | a1fa | asterisknow sucks :) |
| 22:25.33 | a1fa | or is it freepbx thats awful |
| 22:25.34 | a1fa | ;) |
| 22:26.08 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 22:26.09 | ManxPower | ~tier2 |
| 22:26.09 | infobot | somebody said tier2 was #asterisk is not Tier 2 FreePBX/Trixbox Support! |
| 22:26.25 | a1fa | lol |
| 22:26.26 | a1fa | :( |
| 22:27.04 | [TK]D-Fender | ~3rd |
| 22:27.04 | infobot | Don't even THINK about it! Not third, forth OR fifth tier support! |
| 22:27.08 | [TK]D-Fender | .. |
| 22:27.12 | [TK]D-Fender | ~5th |
| 22:27.13 | infobot | Good grief..... ENOUGH already! |
| 22:27.21 | [TK]D-Fender | ~6th ? |
| 22:27.21 | infobot | You though you just needed to pass 5 didn't you? |
| 22:27.28 | [TK]D-Fender | ~32nd ? |
| 22:27.29 | infobot | And no, I'm not falling for this either! |
| 22:27.34 | [TK]D-Fender | :D |
| 22:28.55 | p3nguin_ | a1fa: If you think AsteriskNOW sucks, you're obviously doing it wrong. |
| 22:29.08 | a1fa | which part |
| 22:29.20 | [TK]D-Fender | probably all of it :) |
| 22:29.33 | p3nguin_ | That was my thought. |
| 22:29.33 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 22:29.52 | a1fa | hehe :) hello d-fender |
| 22:30.03 | p3nguin_ | While I am not a fan of a GUI, AsteriskNOW was quite nice when I tested it. |
| 22:30.12 | a1fa | the freepbx part? |
| 22:32.02 | a1fa | not a very well thought of layout |
| 22:32.09 | a1fa | i need not complain |
| 22:32.20 | a1fa | its free :) |
| 22:35.22 | ManxPower | So is an STD |
| 22:35.51 | thehar | lol |
| 22:36.16 | a1fa | lol |
| 22:36.34 | [TK]D-Fender | No, that prostitute was most certainly NOT free.... |
| 22:36.40 | p3nguin_ | You can apparently install AsteriskNOW without a GUI. |
| 22:36.57 | citywok | haha, [TK]D-Fender some of us don't have to pay for it :P |
| 22:38.02 | a1fa | :X |
| 22:38.20 | a1fa | anyway, I am withdrawing my pbx from a dmz to the inside network |
| 22:39.35 | a1fa | and also virtualizing the hardware too |
| 22:42.10 | pabelanger | p3nguin_: I'm surprised to hear that wasn't an option before |
| 22:42.37 | E-bola | is wondierinbg if using realtime will get him a distributed asterisk |
| 22:43.16 | citywok | wondierinbg? sounds german to me |
| 22:43.29 | E-bola | wondering |
| 22:44.19 | citywok | is wondering if e-bola was asking a question, or just thinking out loud |
| 22:45.01 | E-bola | bit of both :) |
| 22:45.17 | citywok | ~question |
| 22:45.17 | infobot | somebody said question was If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 22:45.27 | E-bola | I'm still searching for the best way to end up with an active-active asterisk cluster |
| 22:45.43 | citywok | so is everybody else. I'd recommend looking up asterisk SCF |
| 22:45.57 | E-bola | thats not released out of beta yet? |
| 22:46.13 | citywok | it's not really... anything yet |
| 22:46.39 | E-bola | hence, i dont understand why you would recomend look at that? |
| 22:47.30 | citywok | b/c there isn't really an active-active solution out of box unless you feel like doing a lot of work. http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions |
| 22:48.30 | E-bola | reading http://comunidad.asterisk-es.org/documentos/astricon2008/why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf atm seems like ma nice doc |
| 22:48.38 | E-bola | -m |
| 22:48.51 | citywok | s/ma/a would probably work |
| 22:48.58 | citywok | :P |
| 22:49.46 | E-bola | Hmm i bet ile end up just usiong active/passive since there's a ton of options for that and its pretty easy |
| 22:49.55 | E-bola | i just detest active/passive from a general standpoint :) |
| 22:50.49 | citywok | that doesn't even make sense. you can't detest it. |
| 22:50.57 | *** join/#asterisk iratik (~Adium@74-84-99-12.client.mchsi.com) |
| 22:51.17 | citywok | you can definitely prefer active/active to active/passive, which would make sense. but you can't detest something that provides greatly increased reliability for free :P |
| 22:51.46 | E-bola | well i detest to have to chose it over active/active :) |
| 22:52.00 | E-bola | But my reasons are ofcourse lame and stupid, but they remain valid for me :) |
| 22:52.00 | *** join/#asterisk guilhermebr (~Guilherme@189.63.46.106) |
| 22:52.42 | citywok | feel free to develop a good, functional active/active solution, i'm sure it will be accepted in to the code base with much thanks! |
| 22:52.53 | citywok | I know i'd appreciate it greatly |
| 22:54.41 | E-bola | i dont think an awefull lot is missing codebase wise |
| 22:54.50 | E-bola | im just looking for an example |
| 22:55.05 | citywok | how many concurrent phone calls are you supporting? |
| 22:55.23 | E-bola | not much at all ~50 |
| 22:55.28 | E-bola | but its gonna grow |
| 22:56.17 | citywok | yea... what do you do that can't handle 50 dropped calls & 3 seconds of downtime? |
| 22:56.43 | E-bola | nothing |
| 22:57.58 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 23:00.44 | E-bola | citywok: http://danielaliaman.com/blog///index.php/2009/03/15/load_balance_asterisk_with_ultramonkey?blog=2 pretty much outlines what i want |
| 23:00.50 | E-bola | but he leaves out the interesting part :/ |
| 23:02.47 | citywok | what's the interesting part? |
| 23:03.13 | E-bola | the realtime part |
| 23:03.31 | E-bola | of basically how he handles integrating 3 asterisk servers in regards to having distributed queues, extensions etc |
| 23:03.42 | citywok | that's by no means interesting. |
| 23:03.46 | E-bola | the whole ip failover stuff etc, is basic linux ha setup |
| 23:03.49 | citywok | and he isn't providing support for distributed queues |
| 23:04.33 | E-bola | how do you conclude that? |
| 23:04.57 | citywok | it doesn't look like he has anything provided to even check where a SIP client has registered to be able to route calls from one host to the other |
| 23:05.00 | citywok | because realtime * has no support for distributed queues, it has queue support, but it doesn't know which host the client has registered on. |
| 23:05.13 | E-bola | citywok: dundi handles that? |
| 23:07.24 | citywok | that isn't active/active either, lol |
| 23:07.52 | citywok | so really, it isn't what you are looking for unless you need to load balance thousands of calls (but if * failed, all calls on that node would be lost) |
| 23:08.10 | E-bola | i dont care so much for calls |
| 23:08.27 | citywok | if it has a heartbeat & a shared cluster ip it's active/passive |
| 23:08.42 | E-bola | my main concern is, that if something fails. I want everything to automatically failover to another server. Either manually or automatically. And preferably as quick as possible |
| 23:08.44 | atan2 | Inside an exten can you use if? like exten =>*1,n,if(${SERVICELEVEL} == 1) { Playback(enabled); } else { Playback(disabled); } ? |
| 23:08.56 | citywok | if you don't care about calls then all you need is active passive w/ heartbeat. not active/active. |
| 23:08.58 | E-bola | citywok: it has both, did u read the pdf? |
| 23:09.23 | citywok | i'm looking at their diagram, the pdf just talks about setting up their load balancers and makes no mention of anything * |
| 23:09.38 | [TK]D-Fender | atan2: "core show application gotoif" |
| 23:09.43 | E-bola | citywok: my reason for not wanting to use active/passive, is that i dont have time to test it. With an active/active setup your sure it works. With an active/passive you dont know before something dies, or you test it |
| 23:10.02 | atan2 | ty =) |
| 23:10.09 | E-bola | citywok: I know thats what anoys me :) |
| 23:10.10 | p3nguin_ | also ExecIf |
| 23:10.18 | citywok | so spend 200 hours trying to find an active/active solution, or 2 hours setting up active/passive and spend 5 minute testing it? seems logical to me :) |
| 23:10.50 | E-bola | citywok: point is you have to continueally test it. It can potential break after every single software update etc etc. |
| 23:11.02 | E-bola | active/active has "built-in" testing. If its setup ideally anyway |
| 23:11.11 | citywok | if you have time to update the software, you should have time to spend a couple seconds testing failover :P |
| 23:11.31 | E-bola | citywok: true, but remember then potential i make the cluster go down |
| 23:11.37 | E-bola | which isnt nice.... |
| 23:12.07 | *** join/#asterisk deltalytic (~lork@154.5.144.132) |
| 23:12.07 | citywok | if you have the ability to update your code i'm assuming you have an outage window to do that... i hope you don't just throw code on production servers and cross your fingers mid-day |
| 23:12.40 | deltalytic | Did 1.6 rid of the DNS bug that brings down all the phone if the asterisk server looses connection with the isp? |
| 23:12.59 | citywok | when i lose connection to my itsp i don't lose connectivity between phones |
| 23:13.05 | E-bola | citywok: i do just throw code on production servers |
| 23:13.17 | E-bola | afterhours, but i do just run apt-get update |
| 23:13.23 | deltalytic | city, and you do not have your asterisk box pointed to a internal dns server? |
| 23:13.25 | E-bola | i dont test anything in a test enviroment beforehand |
| 23:13.46 | citywok | deltalytic: yes, i do have it pointed internally. i use a lot of internal resources. |
| 23:14.03 | deltalytic | okay |
| 23:14.13 | citywok | E-bola: lol, i am assuming you either very rarely change anything, your changes are minimal, or your track record is terrible. lol |
| 23:14.30 | E-bola | citywok: change anything? |
| 23:14.36 | deltalytic | I am uncirtain how to create a simple internal dns. Not been able to get mine working since we had changed isps. |
| 23:15.00 | fenrus | bind is neat. |
| 23:15.05 | citywok | you are saying if i lose connection to DNS, * stops letting phones connect? |
| 23:15.25 | deltalytic | yes, in 1.4 all phone loose registration with LOCAL asterisk box. |
| 23:15.35 | deltalytic | if loose dns from isp |
| 23:15.42 | citywok | E-bola: if i were you i'd change my process and get a test server. 50 concurrent calls (or did you mean users?) is a lot to be running around like a retard breaking things. |
| 23:15.43 | p3nguin_ | Sounds like you need to update. |
| 23:15.44 | deltalytic | or just isp or down |
| 23:15.51 | deltalytic | Im running 1.6 now |
| 23:16.01 | deltalytic | But still, cannot get local phone to register. |
| 23:16.03 | citywok | can you link to the bug report on the issue tracker? |
| 23:16.10 | p3nguin_ | I run 1.4, and I'm not familiar with that problem. |
| 23:16.10 | E-bola | citywok: i dont think upgrading debian ever broke my asterisk |
| 23:16.12 | citywok | what does sip debug say? |
| 23:16.24 | E-bola | and no i meant calls |
| 23:16.36 | deltalytic | p3nguin_, mmm running a internal dns? |
| 23:16.37 | citywok | E-bola: yet. you mean yet. |
| 23:16.54 | citywok | and are you running testing branch? why do you need to apt-get update? |
| 23:16.56 | E-bola | yes, after 2 years of doing it this way, it havent broken yet |
| 23:17.07 | E-bola | citywok: no i run stable/lenny |
| 23:17.10 | p3nguin_ | deltalytic: I have an internal dns forwarder, which uses external DNS. |
| 23:17.20 | p3nguin_ | Typical LAN stuff. |
| 23:17.24 | E-bola | why i need to apt-get update? umm are you serious? |
| 23:17.35 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 23:17.54 | deltalytic | p3nguin_, if there was no internal dns, and you were to unplug the dsl, then all phones would drop off line. |
| 23:17.58 | citywok | stable tree doesn't change a whole lot, i don't update my production boxes very often in order to avoid breaking anything. |
| 23:18.05 | *** join/#asterisk simplydrew_ (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 23:18.12 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:18.14 | E-bola | citywok: there's constantly updates, so i dont know where you had that idea.... |
| 23:18.18 | citywok | good firewalling + updated * is my strategy |
| 23:18.40 | deltalytic | I MIGHT turn my asterisk box into a firewall |
| 23:18.41 | citywok | i haven't updated in 3 months, and have a total of 27 packages that need to be updated. the only one i'd worry about is the openssl update. |
| 23:18.46 | E-bola | just because packages dont change version, doesnt mean there arent security updates and bugfixes |
| 23:19.08 | E-bola | do you include volatile? |
| 23:19.19 | citywok | definitely not |
| 23:20.05 | p3nguin_ | deltalytic: What if I configure no nameserver or a bad nameserver on the machine? That would be the same as an external nameserver and no internet connection, right? |
| 23:20.06 | citywok | volatile is worse than testing in debian land |
| 23:20.08 | E-bola | citywok: u run ur * distributed/clustered in anyway? |
| 23:20.15 | E-bola | citywok: nonesence |
| 23:20.29 | deltalytic | p3, very possibly |
| 23:20.40 | deltalytic | I need a way to make my 1.6 work without a isp |
| 23:20.44 | citywok | active/passive, every 5 minutes i rsync everything from the production server to the standby server. |
| 23:21.03 | p3nguin_ | deltalytic: You don't need an internet connection to make Asterisk work on a local network. |
| 23:21.07 | E-bola | citywok: so if it fails over you loose all "states" and registrations? |
| 23:21.17 | E-bola | queue members, hints etc? |
| 23:21.19 | deltalytic | in 1.4 you do if not running dns. google it. |
| 23:21.21 | citywok | i've never had a production failure, so i don't use an automatic failover mechanism, anybody on our IT team goes and clicks a button, which fires up the slave. |
| 23:21.26 | deltalytic | in 1.6, i am not sure. |
| 23:21.59 | p3nguin_ | deltalytic: I've just removed my nameserver entry from resolv.conf. How long before I can't call another phone in-house? |
| 23:22.04 | citywok | queue members are dynamic so stored in astdb which is rsync'd. once the IP moves from primary to standby the phones will have to re-register before making a call. |
| 23:22.21 | deltalytic | p3, it may take a few min. |
| 23:22.33 | citywok | my 25 concurrent calls and 150 users can wait 60 seconds, and the dropped calls aren't E911, we can just call back. |
| 23:22.48 | E-bola | citywok: did u adjust the registry attemp timer on the phones down then? |
| 23:22.53 | deltalytic | city, got a standard 911 land line? |
| 23:23.18 | citywok | deltalytic: i mean we aren't an e911 center. lol. no, we don't have 911 land line. we use e911 from our provider. |
| 23:23.30 | deltalytic | okay |
| 23:23.56 | citywok | if anybody ever dials 911 we get an automatic notifcation of it sent to HR/reception as well as the persons manager, so everybody knows something is happening |
| 23:23.59 | E-bola | citywok: when did u last test that the failover actualy works? :) |
| 23:24.19 | citywok | 71 days ago, the last time i rebooted my servers and made any big changes |
| 23:24.44 | citywok | i actually upgraded my slave server, used the failover, then upgraded my primary server to * 1.6.2 from 1.6.1 |
| 23:24.48 | deltalytic | citywork, that is cool! |
| 23:25.09 | citywok | yea, write a simple perl scirpt and put it in front of the dial(911@itsp) |
| 23:25.11 | deltalytic | That would be a good selling point to sell a asterisk system |
| 23:25.18 | E-bola | that sort of active/passive is built into any newer virtualization platform if ur running it virtual |
| 23:25.46 | citywok | i have a 6 node hyper-v cluster, but i don't run * in it, i don't want to have to debug call quality issues through my cluster. |
| 23:25.55 | citywok | and thato nly provides hardware fault tolerance, not software |
| 23:26.06 | citywok | that's kind of like thinking raid5 is a backup solution |
| 23:26.09 | p3nguin_ | deltalytic: What else do I need to do to make my phone stop working? |
| 23:26.32 | deltalytic | p3, are you running 1.4? |
| 23:26.33 | citywok | deltalytic: ~pb a sip debug of the failure |
| 23:26.43 | p3nguin_ | (1716.10) <p3nguin_> I run 1.4, and I'm not familiar with that problem. |
| 23:26.50 | deltalytic | k |
| 23:26.59 | E-bola | citywok: it provides as much software failover as running an rsync script every xx mins |
| 23:27.05 | atan2 | Can one SIP account have more than one context? So, err, you could hide features within each context? context=local,international,callreturn ? |
| 23:27.10 | citywok | deltalytic: i also have it send the call chanspy() to the receptionist so they know what is going on. |
| 23:27.16 | p3nguin_ | atan2: No. |
| 23:27.32 | deltalytic | citywok, nice. are you the admin that put in asterisk? |
| 23:27.38 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 23:27.42 | p3nguin_ | atan2: You must build an hierarchical structure with your contexts. |
| 23:27.48 | citywok | deltalytic: yes, i wrote everything for it |
| 23:27.56 | deltalytic | nice |
| 23:28.02 | atan2 | p3nguin_, okay. So each context itself can include another context? |
| 23:28.10 | p3nguin_ | atan2: Do you need to see an example? |
| 23:28.14 | citywok | E-bola: i'm not worried about that stuff getting corrupted (which is tarred and backed up every hour to another server), i'm worried about debian shitting itself. |
| 23:28.16 | atan2 | Just not within SIP, but rather the context itself? |
| 23:28.45 | atan2 | What's the deal for inclusion? include => contextname ? |
| 23:28.48 | E-bola | citywok: restoring a snapshot from ur virtualisation would protect against that |
| 23:29.00 | E-bola | wonders how much his voip traffic is going to rise |
| 23:29.13 | E-bola | already doing 17gb out and 17gb in everyday |
| 23:29.29 | deltalytic | E-bola, what service do you sell |
| 23:29.48 | E-bola | deltalytic: what do you mean? |
| 23:30.14 | deltalytic | <E-bola> already doing 17gb out and 17gb in everyday |
| 23:30.26 | citywok | he wants to know what... you do |
| 23:30.31 | E-bola | Its from our hosted pbx * |
| 23:30.36 | deltalytic | okay |
| 23:30.45 | E-bola | still pretty small though |
| 23:30.55 | citywok | deltalytic: if you are interested in how i do e911 i'd be happy to share it outside of this channel |
| 23:30.57 | E-bola | i just hadnt imagine it would use this much traffic to be honest |
| 23:31.07 | citywok | stop using ulaw :) |
| 23:31.41 | E-bola | lol ya i know. But since its in our datacenter i dont really care yet. Im just supprised at how much it accumulates to hehe |
| 23:31.48 | deltalytic | well, I want to get my system up and running. Again! But, get past the blasted nat issue in client pcs. Did a demo once, failed big time because it could not register with my isp. |
| 23:31.52 | deltalytic | err tisp |
| 23:32.13 | E-bola | sip + nat= continous nightmare |
| 23:32.34 | deltalytic | yea tell me about it. we have a new telus router, sucks. wont pass sip or rtp |
| 23:32.56 | deltalytic | So, my want to put my asterisk box as the router to the isp. |
| 23:32.58 | citywok | 50 concurrent calls of ulaw is 3,200kbps according to bandcalc.com |
| 23:33.21 | citywok | that seems lower than it should be, i'd guess probably double that. |
| 23:33.52 | deltalytic | Is there a way a client can click into a web site to see port 5060 and rtp are open? |
| 23:33.59 | E-bola | citywok: would be about 250gb a day |
| 23:34.13 | p3nguin_ | atan2: http://pastebin.com/KcXQhU1E |
| 23:34.15 | E-bola | but we are very very far from having 50 con. calls 24/7 so far |
| 23:34.16 | citywok | ah, the asteriskguru calculator says 50 concurrent calls is 7.79mbps |
| 23:34.47 | deltalytic | So who here knowns how to set up a basic dns? |
| 23:34.49 | citywok | 7.79mbit is not 250gb, 7.79mbit is probably around 70gb |
| 23:34.49 | E-bola | peaking is real bad for voip setups :) |
| 23:34.51 | deltalytic | server? |
| 23:35.03 | atan2 | p3nguin_, ty =) |
| 23:35.18 | p3nguin_ | deltalytic: Still waiting for something exciting to happen. I can still call other phones. |
| 23:35.27 | deltalytic | try |
| 23:35.32 | citywok | E-bola: 7.79mbit is roughly 84,132mb/day |
| 23:35.59 | citywok | 7.79*60/8 = 58MB/minute * 60 * 24 = 84,132/day |
| 23:36.21 | E-bola | yes but having to handle 50 con. call during peak is extremely far from handling on avg 50 con. call always |
| 23:36.30 | E-bola | luckily people tend not to use the work phones much at night :) |
| 23:37.08 | *** join/#asterisk roninarg (~asteriskr@190.113.140.28) |
| 23:37.21 | citywok | If you ran 50 concurrent calls for 24 hours you would rack up 84gb of ulaw, or 25gb of g729 |
| 23:37.32 | p3nguin_ | deltalytic: If you can't tell me how to replicate the problem, I have to assume the problem does not exist. |
| 23:37.44 | deltalytic | p3, google is. |
| 23:37.46 | deltalytic | it |
| 23:37.50 | deltalytic | its online. |
| 23:37.56 | p3nguin_ | You google it. My phones still work. |
| 23:37.58 | citywok | if you show us a sip debug we might be able to help |
| 23:38.00 | E-bola | lol |
| 23:38.08 | deltalytic | :) |
| 23:38.19 | deltalytic | I am not running 1.4 |
| 23:38.28 | p3nguin_ | But I am, and the problem does not exist. |
| 23:38.36 | E-bola | I'm off to bed, nighty |
| 23:38.50 | deltalytic | but I upgraded to 1.6 since I did read the dns but was rectified in this version. |
| 23:39.01 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 23:39.06 | citywok | so deltalytic what are you talking about? |
| 23:39.12 | p3nguin_ | Oh, Karl. |
| 23:39.15 | citywok | you installed it, and it works. what's your problem? |
| 23:39.27 | kfife | p3nguin_: Hey |
| 23:39.50 | deltalytic | https://issues.asterisk.org/view.php?id=12941 |
| 23:40.15 | deltalytic | http://fonality.com/trixbox/forums/trixbox-forums/trunks/sip-trunk-registration-problem-brings-local-calls-troubles |
| 23:40.31 | deltalytic | known issue with 1.4 |
| 23:40.52 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:40.55 | deltalytic | http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=433779 |
| 23:41.04 | deltalytic | anyway back to my work |
| 23:41.09 | *** join/#asterisk pampa (~pampa_2@190.113.140.28) |
| 23:41.28 | *** join/#asterisk Chilling_Silence (~Master@222-153-107-169.jetstream.xtra.co.nz) |
| 23:42.05 | citywok | the issue was never reproduced, and was just closed... that doesn't prove anything |
| 23:42.12 | kfife | oops |
| 23:42.32 | p3nguin_ | It looks like the problem existed on 1.4.21 and disappeared in 1.4.22. |
| 23:42.40 | p3nguin_ | TWO YEARS AGO. |
| 23:42.54 | citywok | good thng he spent 20 minutes in here talking about it |
| 23:42.55 | deltalytic | yea long ago |
| 23:42.55 | pabelanger | Heh, I did know res_phoneprov only worked with users.conf |
| 23:43.02 | citywok | kinda like e-bola being a dumbass. lol |
| 23:43.24 | deltalytic | Ive been using asterisk for a long time. But dumped 1.4 for this reason. |
| 23:44.13 | p3nguin_ | YOu could have just upgraded a version and been done with it. |
| 23:44.14 | citywok | i "dumped" 1.4 b/c 1.6 had more features and worked better w/ dahdi |
| 23:44.56 | p3nguin_ | I'll dump 1.4 when 1.8.3 comes out. |
| 23:45.08 | citywok | haha |
| 23:45.40 | *** join/#asterisk panicou (~panicou@cust-188-2.on3.ontelecoms.gr) |
| 23:45.44 | atan2 | If you wanted to give a SIP user the option to press '7' while on a call to record the call, can you think of a way to do it without causing hell for IVRs? Like if they call their bank and start entering card number 6667... |
| 23:45.45 | Chilling_Silence | Hi all, I'm having issues with connecting Cisco Call Manager and Asterisk (Asterisk 1.4.21.2~dfsg-3+lenny1). If we have a call come in through CCM which handles our PRI, it gets passed off to an asterisk IVR just fine. However when I make a call from a Softphone (ZoIPer) or SPA942 attached to the Asterisk, to any form of extension on the CCM, the quality is terrible, even though they're ulaw (Native bridging). Same poor qua |
| 23:45.56 | deltalytic | btw, I cannot get my phones to register with 1.6. Not using internet at this time. Anyone care to look at the conf configurations? |
| 23:46.04 | p3nguin_ | atan2: Don't use just '7' |
| 23:46.12 | deltalytic | this is perhaps a different issue. |
| 23:46.15 | citywok | lolololol |
| 23:46.18 | p3nguin_ | atan2: Use the normal feature code for automon. See features.conf. |
| 23:46.23 | [TK]D-Fender | deltalytic: show us the FAILURE |
| 23:46.33 | citywok | [TK]D-Fender: the only failure is in his head |
| 23:46.58 | p3nguin_ | There was failure... in June 2008. |
| 23:47.00 | pabelanger | p3nguin_: So your upgrading you boxes in February? |
| 23:47.44 | [TK]D-Fender | atan2: "core show application dial" + features.conf |
| 23:47.48 | p3nguin_ | I refuse to use 1.8 while it is still so new. By 1.8.3, I expect many of the problems will be worked out. |
| 23:48.00 | *** join/#asterisk pkecastillo (~pirruar@190.113.140.28) |
| 23:49.17 | p3nguin_ | For production systems, that is. I'd probably test 1.8.0 without too much fuss. |
| 23:50.55 | p3nguin_ | atan2: Oh, I see a typo I made in that example. That'll give you something to debug if you copy my example and try to use it. |
| 23:51.17 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 23:51.46 | atan2 | p3nguin_, en => _NX? |
| 23:52.08 | p3nguin_ | atan2: What? |
| 23:52.53 | p3nguin_ | exten => _NXXNXXXXXX,1,Dial(SIP/itsp/1${EXTEN}) |
| 23:53.00 | p3nguin_ | This provides 10-digit dialing. |
| 23:53.06 | p3nguin_ | Nothing wrong with it that I can see. |
| 23:54.00 | pkecastillo | hola roninarg |
| 23:54.30 | pkecastillo | hello roninarg , sorry guys! |
| 23:54.48 | roninarg | hi teacher pkecastillo |
| 23:54.49 | pampa | hi pkecastillo |
| 23:55.02 | deltalytic | denada |
| 23:55.47 | atan2 | p3nguin_, using automon (I uncommented it in features.conf) it doesn't seem to record? is there a second file that relates to it? Do I need to build it into the dialplan? |
| 23:56.05 | atan2 | I'd like to save recordings based on their accountcode where possible =) |
| 23:56.27 | p3nguin_ | atan2: (1747.45) <[TK]D-Fender> atan2: "core show application dial" + features.conf |
| 23:56.48 | atan2 | ^_^ will do |
| 23:56.58 | p3nguin_ | Hint: Take a look at w and W. |
| 23:58.01 | atan2 | That just enables it for the user? =) |
| 23:58.16 | p3nguin_ | What is the "user" you keep talking about? |
| 23:58.31 | atan2 | Whoever has control of the SIP phone connected |
| 23:58.37 | atan2 | Not the caller who is calling it |
| 23:58.58 | p3nguin_ | The Dial() options have nothing to do with the people. |
| 23:59.06 | *** join/#asterisk Dovid (Dovid@69.167.68.146) |
| 23:59.29 | Dovid | anyone here ever use sangoma transcoder card ? |