| 00:01.13 | *** join/#asterisk mtaht4 (n=m@12.158.129.130) |
| 00:01.37 | *** join/#asterisk berktr (n=canberk@212.156.213.131) |
| 00:01.59 | berktr | hello, is it possible to learn which provider my voip provider is using to terminate my call? |
| 00:03.14 | *** part/#asterisk ITiliti (n=IceChat7@72.54.46.18) |
| 00:04.24 | generalhan | nentis: im not sure of the rating of this switch ... but i have got 10 of the 9133i phones and they were all plugged into a D-Link DES-1316K and they work perfectly |
| 00:05.10 | nentis | thanks. I have 18 9133i's I need to power. I'll check out the wattage delivery of that dlink. |
| 00:05.13 | generalhan | nentis: its not a GREAT switch but it works. but if you have more than 8 phones you need something else. this is 8 PoE and 8 regular ports |
| 00:05.39 | generalhan | but if you need the information about the power draw you may find more information about this switch than that phone |
| 00:06.18 | berktr | is it possible to learn which provider my voip provider is using to terminate my call? |
| 00:08.21 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:09.00 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 00:09.37 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 00:11.10 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:15.43 | rvhi0 | hi, in sip.conf, i can put multiple lines for "setvar". how do i do it in realtime database? |
| 00:17.39 | *** part/#asterisk nentis (n=nentis@209-162-205-68.dq1mn.easystreet.com) |
| 00:21.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:25.14 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 00:28.21 | Hmmhesays | can you tell vi to put the cursor at the end of a file upon opening it? |
| 00:28.45 | RypPn | <PROTECTED> |
| 00:28.54 | RypPn | iirc |
| 00:29.37 | fujin | anyone have any experiencing ODBCing sip.conf/extensions.conf? |
| 00:29.48 | fujin | a tutorial would be super if anyone has a URI handy. |
| 00:32.20 | berktr | fujin, http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 00:32.58 | fujin | ooh ta. |
| 00:33.15 | berktr | is this what you want fujin |
| 00:39.05 | Hmmhesays | hmm how do you make logger log everything the console does |
| 00:39.29 | Hmmhesays | I don't seem to be getting any verbosity messages in my /var/log/asterisk/messages |
| 00:40.15 | Hmmhesays | nevermind |
| 00:40.16 | Hmmhesays | retarded |
| 00:45.55 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 00:46.29 | kiscokid | I have a question about T1 vs ISDN PRI... |
| 00:47.21 | kiscokid | is the PRI the same if I get a T1 or ISDN? |
| 00:49.09 | wunderkin | pri is a t1 that has a d channel with isdn signalling, a channelized t1 is not the same |
| 00:49.32 | JT | PRI is generally superior |
| 00:49.41 | JT | than channelised T1/CAS/RBS |
| 00:51.59 | anonymouz666 | ciscokid |
| 00:52.50 | kiscokid | so if my provider says they are going to connect me to an ISDN PRI with 23 channels they really mean a T1 with ISDN signalling? |
| 00:54.42 | JT | they mean a T1 ISDN PRI |
| 00:55.04 | kiscokid | and I can use a Sangoma T1/E1 card with this? |
| 00:55.05 | JT | T1 is only the line technology for the link |
| 00:55.08 | JT | yes |
| 00:55.12 | kiscokid | cool |
| 00:55.19 | JT | you should get one with hardware echo cancellation |
| 00:55.34 | kiscokid | the Sangoma not the t1? |
| 00:55.49 | JT | the card you buy |
| 00:55.53 | kiscokid | right |
| 00:55.55 | kiscokid | thanks |
| 00:56.06 | JT | almost doubles the price, but saves hassle later |
| 00:57.14 | kiscokid | have you run a Sangoma quad fxo card and a Sangoma single port t1/e1 in the same box without any problems? |
| 00:57.45 | JT | i haven't had that setup myself |
| 00:57.51 | JT | it's meant to work |
| 00:57.52 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 00:58.00 | kiscokid | ok, thanks |
| 01:01.54 | *** join/#asterisk VJFROMGT (n=vjfromgt@190.80.51.236) |
| 01:03.49 | VJFROMGT | calls drop after 20 seconds,, samesubnet |
| 01:08.37 | *** join/#asterisk ManxPower (n=manxpowe@19.sub-70-218-94.myvzw.com) |
| 01:09.40 | ManxPower | Can anyone think of a shorter way to do this? I originally wrote it for 1.0.x: exten => _XXXX,n,GotoIf($["${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?do-busy:do-no-answer) |
| 01:09.43 | fujin | hey, anyone running * on Ubuntu? |
| 01:09.52 | fujin | trying to work out why the mysql engine isn't part of the asterisk package |
| 01:09.55 | fujin | for res_mysql.conf |
| 01:10.29 | ManxPower | For example, could I do ${DIALSTATUS} = ("BUSY" | "CONGESTION".... |
| 01:11.05 | ManxPower | fujin: The MySQL license is GPL, which is incompatable with Digium's dual licensing |
| 01:11.20 | ManxPower | MySQL even wrote Digium to yell at them for including it. |
| 01:11.42 | [TK]D-Fender | ManxPower, how many values would FAIL that test? |
| 01:11.58 | ManxPower | MySQL used to be licensed under the LGPL license, which is ok. |
| 01:12.26 | ManxPower | [TK]D-Fender: I'm not sure what you mean? |
| 01:12.45 | [TK]D-Fender | ManxPower, which values are you NOT testing that could happen? |
| 01:13.00 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 01:13.21 | JT | mysql is also dual licensed |
| 01:13.22 | ManxPower | [TK]D-Fender: does it matter? |
| 01:13.29 | [TK]D-Fender | ManxPower, yes. |
| 01:13.31 | JT | mysql is crap ;) |
| 01:13.34 | JT | but you can use it |
| 01:13.36 | JT | via odbc |
| 01:14.18 | ManxPower | [TK]D-Fender: Well, if it does not match any of what I'm testing for, it should just go to the no-answer priority. |
| 01:14.56 | ManxPower | I THINK NOANSWER and ANSWER are the only other values documented for Dial. |
| 01:15.20 | ManxPower | I was looking for a more concise way of checking for X OR Y OR Z |
| 01:15.59 | [TK]D-Fender | ManxPower, Well since I know you don't run your dial with "g" then there is in fact only 1 possibility that could happen. So instead of testing ONE, you are testing against THREE :) |
| 01:16.08 | [TK]D-Fender | ManxPower, THATS how you can make it shorter :) |
| 01:16.26 | ManxPower | Um, all my Dial has g on it. |
| 01:16.48 | [TK]D-Fender | ManxPower, ..... either way 2 < 3 |
| 01:16.49 | ManxPower | how else am I going to check the DIALSTATUS |
| 01:17.02 | ManxPower | [TK]D-Fender: so the answer is "no". |
| 01:17.09 | [TK]D-Fender | ManxPower, Do you care to process after the calls HAS terminated? |
| 01:17.28 | [TK]D-Fender | ManxPower, was answered, whats left to do? |
| 01:17.47 | ManxPower | [TK]D-Fender: Yes, I do. I want to see if it was not answered, or if it was busy so I can route the call based on the fall forwarding settings |
| 01:18.07 | [TK]D-Fender | ManxPower, No, what do you need to do if is WAS answered? |
| 01:18.31 | ManxPower | [TK]D-Fender: I need to fall off the dialplan if it was answered. 8-) |
| 01:18.41 | ManxPower | It is EVERY OTHER situation I need to check for. |
| 01:18.42 | [TK]D-Fender | Then with "g"? |
| 01:19.00 | VJFROMGT | does anyone know why calls get dropped after 20 seconds,, (internal , same subnet) |
| 01:19.26 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:19.36 | [TK]D-Fender | ManxPower, you should have no need of continuing processing after that call... |
| 01:19.43 | [TK]D-Fender | ManxPower, I recall that macro.... |
| 01:19.50 | ManxPower | BUSY|CONGESTION|CHANUNAVAIL = handle as call forward busy if set, NOANSWER = handle as call forward no answer, ANSWER = don't do anything special |
| 01:20.12 | [TK]D-Fender | ManxPower, "ANSWER" would get processed jsut the same..... |
| 01:20.15 | ManxPower | [TK]D-Fender: the macro is a mess. I'm trying to update it. |
| 01:20.22 | [TK]D-Fender | ManxPower, pastebin it. |
| 01:20.36 | ManxPower | [TK]D-Fender: ANSWER can just be processed as congestion or fall off the end of the dialplan |
| 01:20.47 | shido6 | ZzZz |
| 01:21.05 | [TK]D-Fender | ManxPower, but you have no NEED to do anything, therefor "g" = useless, and you are testing 3 instead of ONE. |
| 01:21.10 | ManxPower | http://pastebin.ca/613552 |
| 01:21.16 | [TK]D-Fender | ManxPower, you are CREATING work for nothing. |
| 01:21.47 | ManxPower | [TK]D-Fender: This is a tri-state. either fall off the dialplan OR handle as busy OR handle as no answer |
| 01:21.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:22.04 | ManxPower | I'll be happy to remove "g". Then it is a dual state |
| 01:23.04 | [TK]D-Fender | Umm.... its misnumbered or you removed stuff |
| 01:23.08 | berktr | is it possible to learn which provider my voip provider is using to terminate my call? |
| 01:23.45 | ManxPower | [TK]D-Fender: I'm in the middle of rewriting it. If I was done rewriting it, then I would not have questions. |
| 01:24.33 | ManxPower | So the answer to my question is "no, there is no shorter way to test for value OR value OR value |
| 01:25.19 | ManxPower | I should make sure to handle the situation where DIALSTATUS has a value I'm not expecting. |
| 01:28.26 | ManxPower | I suppose a regex would be logical, but all I've seen about regex in Asterisk is questions. |
| 01:29.16 | [TK]D-Fender | ManxPower, ok this is shotening ver well except your "in transition line numbering". |
| 01:29.23 | [TK]D-Fender | ManxPower, like the #5 you don't have... |
| 01:29.33 | x86 | millions of peaches |
| 01:29.34 | ManxPower | [TK]D-Fender: all the numbered priorties are going away |
| 01:29.35 | [TK]D-Fender | and 19+ |
| 01:29.36 | x86 | peaches for me |
| 01:29.43 | x86 | heya manx |
| 01:29.50 | [TK]D-Fender | ManxPower, they aren't in there NOW so I can't tell what they refer to |
| 01:30.03 | [TK]D-Fender | x86, I could eat a peach for hours.... |
| 01:30.17 | ManxPower | [TK]D-Fender: I appreciate your try at helping with my entire macro. I was really only asking about the one thing. |
| 01:31.37 | ManxPower | I did remove "g" |
| 01:32.41 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 01:33.10 | x86 | [TK]D-Fender: me too bro |
| 01:33.11 | x86 | :) |
| 01:33.25 | [TK]D-Fender | ManxPower, http://pastebin.ca/613566 |
| 01:37.40 | Hmmhesays | numbered priorities are going away? |
| 01:38.02 | [TK]D-Fender | ManxPower, GotoIf's restructured to take advanteage of the NON ZERO len checks. no $[] needed |
| 01:38.13 | [TK]D-Fender | Hmmhesays, Nope. |
| 01:38.26 | ManxPower | [TK]D-Fender: I know. I hate that in the dialplan just as much as I hate it in any programming language. |
| 01:38.32 | ManxPower | You know me, I like explicit. |
| 01:38.37 | [TK]D-Fender | Hmmhesays, you CAN'T reall do away wth numbered priorities |
| 01:38.41 | x86 | http://pastebin.ca/613570 |
| 01:38.41 | ManxPower | Hmmhesays: They are going away in MY stuff. |
| 01:39.10 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 01:39.13 | x86 | anyone see why i can not call a Zap extension on my channel bank? (Zap/5 - Zap/28) ? |
| 01:39.33 | ManxPower | x86: That is the values of DIALSTATUS and HANGUPCAUSE after your Dial? |
| 01:40.32 | flewid | exten => 2065551212,3,GotoIf($["${CALLERID(number)}" != "5551211212"]?4:1000) |
| 01:40.34 | lin0xx | if i'm receiving a '484 status incomplete' - which field is this referring to in the packet? |
| 01:40.36 | *** join/#asterisk holiday42 (n=me@71-221-23-43.farg.qwest.net) |
| 01:40.47 | ManxPower | lin0xx: usually the destination number |
| 01:40.57 | lin0xx | destination number, awesome |
| 01:40.57 | flewid | that reads "if the callerid isn't this, go to this context, priority 4, if it does equal this, then go to this context, priority 1000" right? |
| 01:41.34 | lin0xx | ManxPower: thanks :) |
| 01:42.50 | [TK]D-Fender | x86, All that shit should be UNDER channels |
| 01:43.25 | x86 | [TK]D-Fender: ah ok |
| 01:43.27 | ManxPower | most people don't use trunkgroups |
| 01:43.38 | [TK]D-Fender | x86, you = major jumbled |
| 01:43.39 | x86 | [TK]D-Fender: 'wancfg' from the sangoma drivers did that |
| 01:43.42 | lin0xx | ManxPower: would the destination number be in the To: field? |
| 01:43.50 | x86 | [TK]D-Fender: i've never used physical hardware before ;) |
| 01:43.59 | x86 | [TK]D-Fender: just SIP phones and IAX/SIP trunks heh |
| 01:44.32 | x86 | ManxPower: but I want each grouping to be a huntgroup |
| 01:44.41 | x86 | ManxPower: port 2 on the A102D is a T1 |
| 01:44.48 | x86 | ManxPower: aka, T1 to the PSTN |
| 01:44.54 | ManxPower | lin0xx: Assume you Dial(SIP/12345@happypbx). Assume that happypbx has exten => 123456,1,Whatever and nothing else. The number you sent it is "incomplete" because there was not a pattern match, but if you sent a longer number it might match. |
| 01:45.01 | JT | x86: argh, the evil sip trunks!1 :P |
| 01:45.02 | x86 | ManxPower: so i want to be able to dial Zap/g3 and hunt for an available channel |
| 01:45.04 | ManxPower | x86: that has NOTHING to do with trunkgroups |
| 01:45.14 | x86 | no? |
| 01:45.22 | [TK]D-Fender | flewid, exten => 2065551212,3,GotoIf($["${CALLERID(number)}" != "5551211212"]?4:1000) <- nope |
| 01:45.24 | ManxPower | Zap/g3 refers to group= not [trunkgroups] |
| 01:45.30 | x86 | ah ok |
| 01:45.34 | flewid | [tk]: oh? what's wrong with it? |
| 01:45.46 | flewid | i also tried without the ! and also reversing the 1000:4 |
| 01:45.48 | [TK]D-Fender | flewid, exten => 2065551212,3,GotoIf($["${CALLERID(number)}"="5551211212"]?1000) <- better |
| 01:45.52 | JT | x86: wow that is a big mess :P |
| 01:45.52 | flewid | but none seemed to work |
| 01:46.03 | ManxPower | flewid: 4 and 1000 would be PRIORITIES, not extensions |
| 01:46.09 | JT | trunkgroups is for NFAS only |
| 01:46.13 | flewid | manx: yup |
| 01:46.20 | x86 | ManxPower: should i keep what's currently under [channels] also? or can i just nix that? |
| 01:46.20 | JT | forget about trunkgroups if you do not have an NFAS service |
| 01:46.24 | flewid | i have priority 4 which is the normal menu, then priority 1000 which is the exgirlfriend stuff |
| 01:46.29 | x86 | JT: ok |
| 01:46.37 | [TK]D-Fender | JT : Not For A Second do i believe he needs that! ;) |
| 01:46.39 | lin0xx | ManxPower: ah, okay. well, i'm just trying to figure out where in the sip packet i'm building the address would be invalid |
| 01:46.44 | JT | ;) |
| 01:46.47 | lin0xx | however, your explanation does put a 'human' face to it :) |
| 01:46.49 | flewid | let me try without the spaces and without the 4: |
| 01:46.56 | lin0xx | ManxPower: sorry if i'm being difficult |
| 01:47.01 | ManxPower | flewid: I guess you need a Noop(CALLERID(number)=${CALLERID(number)}) as the priority before that to see what is REALLY going on |
| 01:47.36 | flewid | yah, im tossing one of those in too :) |
| 01:48.14 | ManxPower | flewid: I see the problem. CALLERID(number) is NOT a valid variable. |
| 01:48.22 | ManxPower | As you would know if you checked README.variables |
| 01:48.42 | x86 | http://pastebin.ca/613580 |
| 01:48.47 | x86 | JT: this look better? |
| 01:49.20 | [TK]D-Fender | x86, still BAD |
| 01:49.35 | ManxPower | It might be nice if CALLERID(number) was just an alias for CALLERID(num) |
| 01:49.40 | x86 | [TK]D-Fender: i dont understand why :) |
| 01:49.42 | flewid | manx: ah you're right, i forgot this is 1.2 when my home sys is 1.4 :) the same line works at home |
| 01:49.42 | [TK]D-Fender | x86, you are setting all sorts of shit AFTER your channel declarations that will never apply to ANYTHING |
| 01:50.20 | shido6 | :) |
| 01:50.20 | x86 | [TK]D-Fender: yeah i figured i could remove that... i asked and no one answered :P |
| 01:50.20 | ManxPower | x86: When you set an option, that option applies to all channel= lines AFTER that option until you reset the option |
| 01:50.47 | ManxPower | x86: another way to think of it (which is technically not correct) is that /etc/asterisk/zapata.conf is parsed from the bottom to the top.] |
| 01:51.03 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:51.31 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 01:52.30 | magic_hat | hey everyone. I'm experiencing some weird behavior with one of my softphones. When the extension receives a call, it doesn't ring -- it just picks up. Can't figure out if this is an X-Lite issue or an asterisk problem. Here's the console output from one such call... http://pastie.caboo.se/77867 |
| 01:52.36 | x86 | ManxPower: thanks :) |
| 01:53.09 | irule | why is mixmonitor not recording calls from sip to zap channels? |
| 01:54.27 | x86 | now i'm getting this: [Jul 3 14:57:40] WARNING[29272]: Unable to specify channel 1: No such device or address |
| 01:55.21 | flewid | manxpower: as always, thanks for your help :), there were two issues, the number i was given for the incoming calls was reporting a diff cid than i was told to enter in the gotoif, as well as your note on (num) vs (number) :) |
| 01:55.25 | x86 | wtf is that about? |
| 01:55.26 | flewid | works now |
| 01:58.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:58.04 | ManxPower | x86: /etc/zaptel.conf issue |
| 01:58.16 | x86 | ManxPower: can you look at my paste? |
| 01:58.23 | x86 | http://pastebin.ca/613580 |
| 01:58.29 | x86 | my zaptel.conf is there |
| 01:58.35 | JT | x86: did your asterisk install have a default zaptel.conf? |
| 01:59.02 | JT | zapata.conf too |
| 01:59.10 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 01:59.39 | x86 | they did at one point |
| 01:59.47 | x86 | wancfg from sangoma ate them, aparantly |
| 01:59.52 | magic_hat | http://pastie.caboo.se/77867... anyone? |
| 01:59.58 | x86 | wait, i have some defaults in /usr/share/doc/asterisk :) |
| 02:00.27 | ManxPower | x86: next time install from scratch |
| 02:00.32 | ManxPower | ..er... source |
| 02:00.35 | ManxPower | not a package |
| 02:00.39 | JT | x86: basically that looks nothing like what it should look like |
| 02:01.02 | JT | x86: < x86> http://pastebin.ca/613580 |
| 02:01.04 | JT | that url |
| 02:01.10 | JT | is that zapata.conf or zaptel.conf? |
| 02:01.29 | ManxPower | x86: If you are in the USA/Canada you do NOT want busydetect or callprogress set. |
| 02:02.31 | x86 | JT: they are all there, labeled |
| 02:02.42 | ManxPower | (you might if you were connected to PBX FXS analog lines and were desperate for disconnect supervision) |
| 02:02.53 | x86 | ManxPower: ah |
| 02:02.53 | x86 | ok |
| 02:02.54 | JT | x86: http://pastebin.ca/613580 |
| 02:03.03 | x86 | ManxPower: i DID install from source :P |
| 02:03.03 | JT | x86: is that zapata.conf or zaptel.conf? |
| 02:03.30 | JT | x86: it's one or the other, not both |
| 02:03.33 | x86 | JT: ah that was my old zapata.conf |
| 02:03.39 | JT | ah ok |
| 02:03.42 | ManxPower | x86: not if you have /usr/share/doc/asterisk (unless it is one of the new things in 1.4) |
| 02:03.43 | x86 | JT: (i had one post with 4 configs in it first) |
| 02:03.54 | x86 | ManxPower: gentoo compiled it from source |
| 02:04.00 | x86 | ManxPower: and properly installed the docs |
| 02:04.06 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 02:04.12 | x86 | ManxPower: also, there is a makefile target to install the docs as well |
| 02:04.30 | ManxPower | x86: compileing from a package is not "from source" in our world. |
| 02:04.40 | x86 | it's not from a package |
| 02:04.43 | ManxPower | In this world it is downloading, unpackaing, and compiling |
| 02:04.58 | ManxPower | Ah, must be something new then. |
| 02:04.59 | JT | "gentoo compiled it" was the key phrase |
| 02:05.00 | x86 | well, if you call a tarball from ftp.digium.com a package, then yeah i guess it was a package ;) |
| 02:05.08 | JT | so YOU compile it |
| 02:05.12 | JT | compiled |
| 02:05.28 | x86 | well, i had to hand-compile zaptel anyway because sangoma required me to patch it |
| 02:05.38 | JT | yep |
| 02:05.42 | ManxPower | i.e. using "make" and "configure" rather than something like "emerge asterisk gentoo-sucks-cock" |
| 02:05.58 | x86 | ManxPower: emerge is just a front-end to configure, make, etc ;) |
| 02:06.07 | ManxPower | x86: with it's own makefiles |
| 02:06.16 | x86 | ugh, believe what you will |
| 02:06.33 | x86 | anyway, instead of flaming distros, can you please help? |
| 02:06.42 | ManxPower | x86: so emerge downloads the package from Digium (how does it know what actual file to get) then runs configure and make |
| 02:06.43 | JT | x86: yes, please look at the defaults |
| 02:06.50 | x86 | ManxPower: YES |
| 02:06.51 | JT | and info in the wiki on zapata and zaptel.conf |
| 02:07.02 | JT | ~thewiki |
| 02:07.02 | jbot | i heard thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 02:07.15 | JT | x86: also, what is your zaptel.conf like? |
| 02:07.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:07.44 | ManxPower | x86: Then why even use emerge? |
| 02:08.22 | x86 | JT: http://pastebin.ca/613591 |
| 02:08.45 | x86 | ManxPower: because it automatically grabs all the source, including patches like bristuff, and does all the configuration and making for me |
| 02:09.04 | x86 | ManxPower: funny how you flame gentoo but don't even know how it works ;) |
| 02:09.32 | JT | x86: 2 span 1s, see an issue? |
| 02:10.28 | ManxPower | x86: There is the problem. Like with any "package" it applies patches we know nothing about here. |
| 02:10.43 | k-man_ | is there australian sounding ring tones for the linksys spa941 anywhere? |
| 02:11.06 | ManxPower | x86: I hate compiling from source. I use pre-compiled packages whenever I can. But not for Asterisk. For Asterisk I compile by hand. |
| 02:11.40 | ManxPower | x86: I flame anyone here that does not download, compile, and install manually. |
| 02:12.18 | ManxPower | (asterisk, spamassassin, and clamav are pretty much the only things I compile from source) |
| 02:12.46 | x86 | JT: i thought zaptel.conf matched zapata.conf? |
| 02:12.56 | ManxPower | x86: not even close. |
| 02:13.04 | JT | x86: they do different things |
| 02:13.19 | JT | the only thing that matches are channel numbers, that what joins them together |
| 02:13.27 | JT | zaptel.conf configures zaptel |
| 02:13.36 | JT | zapata.conf configues asterisk's chan_zap |
| 02:13.41 | x86 | ah ok |
| 02:13.52 | x86 | so i can't have a matching group and a matching span? |
| 02:14.00 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 02:14.01 | *** join/#asterisk cullenincrease (n=cullen@c-75-66-240-62.hsd1.tn.comcast.net) |
| 02:14.03 | JT | you cannot have two span 1s |
| 02:14.03 | cullenincrease | hey |
| 02:14.13 | cullenincrease | i was wondering if someone could take a look at my sip.conf and point out any possible issues? i've been fighting with it for 3 days now |
| 02:14.20 | x86 | JT: i dont defined "spans" at all in zaptel.conf though |
| 02:14.31 | cullenincrease | its really weird....sometimes calls will work just fine but then all of a sudden there will be no ring and the call doesnt seem to get routed to the phone, unless i restart the phone |
| 02:14.40 | JT | change span=1,1,0,esf,b8zs to span=2,1,0,esf,b8zs |
| 02:14.46 | x86 | ah ok |
| 02:14.47 | cullenincrease | http://pastebin.com/m65c50aee there's my sip.conf and extensions.conf |
| 02:14.55 | x86 | wasn't sure how the numbering went ;) |
| 02:15.09 | JT | x86: check out the wiki entry on it |
| 02:15.17 | x86 | JT: 2,0,0 or 2,1,0? |
| 02:15.24 | *** join/#asterisk LoneShadow (n=vshivash@59.92.145.223) |
| 02:15.35 | ManxPower | x86: A span is a T-1 physical port on the card. |
| 02:15.47 | ManxPower | you can't have two port #1 |
| 02:15.50 | JT | http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax |
| 02:16.02 | JT | x86: 2,1,0 |
| 02:16.18 | *** join/#asterisk bobdrakken (n=root@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 02:16.26 | JT | x86: only the first number is necessarily a unique sequence number |
| 02:16.41 | bobdrakken | hey.. i have a question.. does anyone know of some free software that will do voice distortion or voice changing |
| 02:16.46 | JT | x86: second number is 1 for the connection to telco, 0 to channel bank |
| 02:16.51 | bobdrakken | my friend that uses M$ is looking for something |
| 02:16.54 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 02:16.55 | JT | do you definitely have an e&m link to the telco? |
| 02:17.16 | bobdrakken | any ideas... |
| 02:17.20 | *** join/#asterisk Strom_M (n=strom@12.175.45.219) |
| 02:17.25 | JT | the second number is about timing, but i'm sure you can read the wiki article yourself :) |
| 02:17.26 | bobdrakken | maybe a free softphone that does voice changing or something |
| 02:17.58 | cullenincrease | i was wondering if someone could take a look at my sip.conf and point out any possible issues? i've been fighting with it for 3 days now |
| 02:18.08 | x86 | JT: yes, it's e&m |
| 02:18.12 | bobdrakken | anyone ?? |
| 02:18.20 | JT | bobdrakken: stop being so damned impatient |
| 02:18.26 | bobdrakken | JTsorry |
| 02:18.30 | JT | x86: my condolences :P |
| 02:19.13 | cullenincrease | there are a few cool voip services out there that give you a PIN, let you specify your own caller ID # and optionally change your voice |
| 02:19.27 | cullenincrease | pretty cool little gig there |
| 02:19.39 | bobdrakken | can u give me a name of the provider |
| 02:19.45 | cullenincrease | but unforunately i dont know of any voice changing software |
| 02:19.47 | cullenincrease | sure |
| 02:20.11 | cullenincrease | spooftel.com |
| 02:20.29 | ManxPower | JT On Sangome each PORT can have it's own sync source. Digium cards can have one sync source per CARD. |
| 02:20.36 | bobdrakken | shes saying she just needs it for like one night |
| 02:20.42 | bobdrakken | wonder if they have a pay buy usage plan |
| 02:21.13 | JT | ManxPower: ok |
| 02:21.27 | ManxPower | JT: I don't recall if you use Sangoma or not. |
| 02:21.49 | JT | ManxPower: both |
| 02:21.50 | cullenincrease | i think they charge something like 5 bucks for 60 minutes |
| 02:22.02 | x86 | JT: CAS T1 hehe |
| 02:22.17 | JT | x86: sounds terrible |
| 02:22.27 | x86 | Jul 10 21:21:45 ERROR[13094]: chan_zap.c:12613 setup_zap: Unknown signalling method 'e&m' |
| 02:22.38 | JT | all these idiotic old signalling methods used on T1s in the us |
| 02:22.50 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 02:22.59 | x86 | i like kewlstart, but AT&T wont support that for me ;) |
| 02:23.04 | Corydon76-home | x86: it's e&m in zaptel.conf, but em or em_w in zapata.conf |
| 02:23.15 | x86 | Corydon76-home: cool thanks |
| 02:23.19 | JT | you should LIKE PRI ;) |
| 02:23.39 | techman97_andy | hey all, anyone worked with Polycom 601s connecting through a NAT to an asterisk box? In the syslogs from the Polycom, all I get is: Error Code:480 Temporarily not available. Another grandstream I have onsite here works just fine through the NAT. |
| 02:23.43 | Corydon76-home | I like Q.931 |
| 02:23.50 | Corydon76-home | but I also like SS7 |
| 02:23.55 | JT | heh |
| 02:24.09 | x86 | JT: i'd rather have a PRI, but I can not get "dedicated" LD rates with a PRI... PRI uses "switched" rates for LD, just like a POTS line |
| 02:24.10 | cullenincrease | http://pastebin.com/m65c50aee can anyone take a quick look at that for me and see if anything looks abnormal? |
| 02:24.31 | x86 | JT: and this is for an outbound call center, so we're talking huge savings |
| 02:24.34 | JT | x86: yes... idiotic old signalling methods used in the US for cheap rates, i get it :P |
| 02:26.27 | bobdrakken | i dont see were u can change ur voice |
| 02:26.42 | x86 | JT: *nod* |
| 02:27.00 | x86 | cool, it seems to be working |
| 02:27.00 | ManxPower | x86: e&m is not a valid signalling name. |
| 02:27.14 | x86 | ManxPower: it works fine now :) |
| 02:27.22 | cullenincrease | its there ive used it before |
| 02:27.25 | x86 | ManxPower: (with e&m signalling) |
| 02:27.26 | cullenincrease | says right there on the main page |
| 02:27.44 | ManxPower | it is em or em_w I think |
| 02:27.55 | ManxPower | I would have to look it up |
| 02:28.21 | cullenincrease | "change your voice to male/female, record convo, etc" |
| 02:28.37 | cullenincrease | the male voice will give you the voice changing you're looking for, like on the movies |
| 02:29.00 | ManxPower | cullenincrease: Just tell her to make up with her BF |
| 02:29.01 | cullenincrease | ive played around with the service before, one thing to keep in mind i think it could take up to 24 hours to activate your acct once you register |
| 02:29.08 | cullenincrease | eh? |
| 02:29.20 | ManxPower | Why else would someone need to change their voice for 1 night? |
| 02:29.21 | cullenincrease | you mean bobdrakken: :") |
| 02:29.27 | ManxPower | maybe I do. 8-) |
| 02:29.33 | cullenincrease | dont ask me ask him i was just making a suggestion |
| 02:29.35 | bobdrakken | so |
| 02:29.37 | bobdrakken | after i set up the account |
| 02:29.43 | bobdrakken | it will let me change from male to female |
| 02:29.56 | JT | why don't you try it and find out |
| 02:30.11 | bobdrakken | u want me to pay for a service to see if it does what i want it to |
| 02:30.14 | cullenincrease | absolutely |
| 02:30.20 | bobdrakken | i might as well buy M$ |
| 02:30.24 | bobdrakken | while im at it |
| 02:30.28 | cullenincrease | well it was just a suggestion |
| 02:30.36 | bobdrakken | not a smart one |
| 02:30.37 | cullenincrease | unfortunately i dont know of any free alternatives |
| 02:30.37 | JT | bobdrakken: you might as well continue to whinge and complain like a sook |
| 02:30.43 | cullenincrease | well sorry for trying to help bub |
| 02:30.46 | JT | bobdrakken: go have a cry |
| 02:30.49 | JT | or even better |
| 02:30.50 | JT | piss off |
| 02:31.02 | JT | not even an asterisk question |
| 02:31.05 | JT | wrong channel |
| 02:31.08 | bobdrakken | im sorry i didnt mean to sound mean |
| 02:31.10 | cullenincrease | yeah i couldnt agree more |
| 02:31.34 | bobdrakken | just i mean you told me to buy something and see after words if its what i want |
| 02:31.54 | cullenincrease | im just saying i've used their service before and it will do what you want it to do |
| 02:32.03 | bobdrakken | that makes more sense |
| 02:32.04 | JT | cullenincrease suggested it, and if you can't read the web page, whose fault is that? |
| 02:32.06 | bobdrakken | sorry if i was rude |
| 02:32.10 | JT | he already said that :P |
| 02:32.13 | bobdrakken | and thats what i said jt |
| 02:32.22 | bobdrakken | on the site i dont see info on changing voice |
| 02:32.29 | cullenincrease | plus maybe some of these asterisk guru's in here might find it as an interesting way to make a few bucks off of by setting up a similar service |
| 02:32.55 | JT | cullenincrease: problem is you'd need to do the voice changing some how :) |
| 02:33.02 | ManxPower | Anyone that wants to set up an ITSP is insane. |
| 02:33.25 | *** join/#asterisk jameswf (n=jfinstro@ip70-162-108-73.ph.ph.cox.net) |
| 02:33.28 | JT | a bargain one, anyway |
| 02:33.30 | cullenincrease | yeah but these guys do it and i wouldn't almost doubt they're using asterisk to somehow do it |
| 02:33.54 | cullenincrease | something to think about i guess |
| 02:34.16 | JT | cullenincrease: you'd need to make a customer bit of code that sits in the internals of asterisk |
| 02:34.22 | ManxPower | Here is an Asterisk related question: Is it a bad idea to use a Goto to jump out of a macro? |
| 02:34.24 | JT | custom, even |
| 02:34.34 | JT | bobdrakken: it's on the FRONT PAGE |
| 02:34.44 | JT | You can show any phone number you wish on their call display. You can change your voice to male/female, record the conversation, SMS text message and be able to PROTECT YOUR PRIVACY by not showing your number on their caller ID. |
| 02:35.03 | bobdrakken | crap ur right jt |
| 02:36.46 | cullenincrease | like i said bobdrakken, i've used the service before and it will do exactly what you're looking for, although not free |
| 02:37.16 | cullenincrease | but it will definitely give you the voice changing you're looking for, just like you would see in the movies |
| 02:37.33 | ManxPower | Perfect for extortion schemes |
| 02:37.36 | cullenincrease | lol |
| 02:37.36 | *** part/#asterisk bobdrakken (n=root@75-138-105-182.dhcp.hckr.nc.charter.com) |
| 02:37.40 | cullenincrease | or to mess with your friends :) |
| 02:37.55 | ManxPower | <darth vader>All your base belong to us.</darth vader> |
| 02:38.02 | cullenincrease | manxpower your nick wouldn't be related to VW's would it? |
| 02:38.29 | cullenincrease | ahh im getting a bit off topic here |
| 02:38.32 | cullenincrease | sorry :) |
| 02:38.44 | ManxPower | cullenincrease: it is not related to Volkswagons, Veterans of Wars, Very smart Wabbits |
| 02:39.05 | cullenincrease | ah ok. i've built MANX baja bugs in the past didnt know if that's what it meant or not |
| 02:39.43 | ManxPower | A Manx is a breed of cat |
| 02:39.56 | cullenincrease | that too |
| 02:40.37 | cullenincrease | but there's nothing like slapping a fiberglass body on a VW chassis with a bored out porsche 911 motor and running 12 second quarter miles with 32" tires :) |
| 02:40.59 | x86 | cullenincrease: which service are you talking about? |
| 02:41.09 | cullenincrease | in regards to what x86? |
| 02:41.18 | x86 | voice changing |
| 02:41.26 | cullenincrease | www.spooftel.com |
| 02:41.40 | cullenincrease | there's several of them out there though |
| 02:42.32 | cullenincrease | only thing is most of them are not so savvy marketing wise so with the right resources someone could make a nice buck off something like that if properly set up |
| 02:43.30 | cullenincrease | i might not be an asterisk guru but internet marketing is another story |
| 02:45.19 | JT | it's also legally dodgy |
| 02:47.55 | *** join/#asterisk demiv (n=demiv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 02:49.25 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 02:49.42 | linagee | are there any certifications that come to mind that show that you know how to work with PBXs, T1s, etc? |
| 02:51.06 | JT | telecommunications engineer |
| 02:51.20 | linagee | JT: and that's a certification? |
| 02:51.58 | JT | it's a pretty good one |
| 02:52.05 | JT | usually have to go to some form of college |
| 02:52.48 | linagee | lol |
| 02:53.08 | ManxPower | linagee: I'm sure Cisco has some telephony related certifications. Most of the PBX vendors offer certifications in their products |
| 02:53.17 | linagee | ManxPower: hrm. true. |
| 02:53.27 | JT | dCAP is available in the US |
| 02:53.31 | linagee | ManxPower: thanks. i'll have to add that to my list of cisco certs to get. lol |
| 02:53.46 | JT | cisco certified certificate collector |
| 02:53.55 | linagee | hehe |
| 02:54.03 | ManxPower | C4? |
| 02:54.11 | ManxPower | or C^4 |
| 02:54.12 | linagee | JT: professional certificate accumulator |
| 02:54.23 | linagee | JT: they should have a certificate for proficiency in that |
| 02:55.05 | ManxPower | Certified Asterisk Routing Professional? |
| 02:55.19 | x86 | linagee: CCIE Voice, CCVP |
| 02:55.29 | linagee | x86: hrm... seems to be all about voip. lol |
| 02:55.40 | linagee | x86: nobody cares about a traditional T1 or DS-3 or such? :( |
| 02:56.08 | ManxPower | linagee: Really the best answer is "go work for a CLEC for 2 years, that is better than any certification" |
| 02:56.16 | linagee | ManxPower: hehehe |
| 02:56.24 | x86 | linagee: data T1's are covered on the CCNA |
| 02:56.30 | linagee | x86: ah |
| 02:56.32 | ManxPower | linagee: anyone that is not a moron cares about T-1/E-1s |
| 02:57.09 | ManxPower | linagee: T-1/E-1s are just not all that complicated from an end user standpoint. |
| 02:57.20 | linagee | ManxPower: why sell bw in such small chunks though. it seems silly/stupid. they should be running fibers to businesses. :( |
| 02:57.48 | linagee | ManxPower: such is the way of ma bell i guess |
| 02:57.57 | carrar | cost money to change infrustructure |
| 02:58.09 | ManxPower | linagee: Usually the answer is either "It is too expensive" or "PUC/PSC tariffs" |
| 02:58.31 | linagee | carrar: true. but new businesses are being set up every day and they use T1. (possibly because it's a standardized setup. *shrug*) |
| 02:59.03 | linagee | 1.544Mbps ought to be enough for anyone. (i think i remember bill gates saying something about memory. ;) ) |
| 02:59.17 | ManxPower | About 6 months ago I had a coversation with a CLEC with only RESALE service in my CO and the answer to most of my questions was "Bell isn't tariffed for that service so we can't offer it on a resale line." |
| 02:59.40 | ManxPower | linagee: most of the WAN links I manage are 384k |
| 02:59.53 | linagee | ManxPower: wow. that's tiny |
| 02:59.57 | irule | what is the variable for the number I am calling? sometimes ${EXTEN} is s and I need the number, thanks |
| 03:00.25 | ManxPower | Other than that horrid "3Mbps bonded T-1" that is handed to us as an ethernet connection. I almost shot the IT manager over that one. |
| 03:00.45 | *** join/#asterisk kikoy (n=kikoy@124.106.83.9) |
| 03:00.46 | ManxPower | irule: EXTEN is never "s" unless the called number is not known or your are in a macro |
| 03:01.01 | irule | DIALEDPEERNUMBER may be? |
| 03:01.16 | ManxPower | irule: Maybe you should look in README.variables |
| 03:01.29 | JT | x86: T1s are barely convered in CCNA |
| 03:01.32 | *** join/#asterisk huey23 (n=huey@64.184.8.24) |
| 03:01.35 | linagee | ManxPower: why was a 3Mbps bonded T-1 bad? is it less reliable or something? |
| 03:01.45 | JT | most CCNA grads don't know anything about a T1 apart from how to spell it |
| 03:01.59 | JT | s/convered/covered/ |
| 03:02.01 | ManxPower | linagee: Oh, a 3Mbps bonded T-1 is fine. But when I get handed an Ethernet connection, I do NOT consider that a "T-1" |
| 03:02.05 | tzanger | anyone here have any real-world app_amd settings that seem to work well? |
| 03:02.13 | huey23 | i have some basic linux questions if someone would please help |
| 03:02.16 | linagee | ManxPower: hrm. lol |
| 03:02.25 | x86 | JT: haha |
| 03:02.39 | x86 | JT: it teaches you framing and coding at least :P |
| 03:02.40 | linagee | ManxPower: maybe it's a T-one and a half. :-> |
| 03:02.41 | ManxPower | "The internet is down!" "Well we still have a link light on our ethernet port. It must not be down." |
| 03:02.58 | tzanger | I can teach you framing and coding in 20 minutes |
| 03:03.14 | tzanger | gather round children if you want ot hear the tale of the t1 |
| 03:03.14 | linagee | lol |
| 03:03.16 | ManxPower | linagee: There is no way I can tell the state of the T-1 link when my physical interface is thernet. |
| 03:03.17 | x86 | huey23: ##linux |
| 03:03.26 | x86 | huey23: #linuxhelp |
| 03:03.32 | x86 | huey23: not #asterisk ;) |
| 03:03.35 | linagee | tzafrir: everything you need to know is on wikipedia. :-> http://en.wikipedia.org/wiki/T1 |
| 03:03.38 | tzanger | ManxPower: howso? |
| 03:03.52 | tzanger | tdmoe gives you alarm spans pretty easily |
| 03:03.52 | ManxPower | I can't do "sh interface ser0/1" and know that the physical layer is up, but the HDLC layer is down, and so it must by a crossconnect or provisioning issue with the telco. |
| 03:03.54 | huey23 | i figured this place would be active so i asked ...:) thanks |
| 03:03.56 | tzanger | especially since 1111 is an invalid state IIRC |
| 03:03.58 | linagee | tzafrir: uhm... http://en.wikipedia.org/wiki/Digital_Signal_1 |
| 03:04.01 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.16 | tzanger | bah |
| 03:04.20 | tzanger | my story would have been more fun |
| 03:04.25 | x86 | hehe |
| 03:04.35 | ManxPower | You basically can't know ANYTHING about the state of anything for a WAN when your physical interface is Ethernet |
| 03:04.42 | *** part/#asterisk huey23 (n=huey@64.184.8.24) |
| 03:05.20 | ManxPower | tzanger: I'm referring to a slimy telco that sold a "3 Mbps bonded point to point T-1" and then handed me an ethernet port. |
| 03:05.36 | tzanger | ManxPower: oh, my mistake |
| 03:05.44 | linagee | ManxPower: i bet they're using ADSL going to an ethernet port. hehe. "fake T-1" :-> |
| 03:05.52 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 03:05.57 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 03:06.05 | ManxPower | linagee: you would not win your bet. |
| 03:06.17 | linagee | ManxPower: you have seen the actual smart jack connector then? |
| 03:06.51 | ManxPower | linagee: Since it is 3Mbps in both directions, I doubt they are using ASYMETRICAL lines. |
| 03:07.07 | linagee | ManxPower: s/ADSL/DSL/ |
| 03:07.15 | linagee | SDSL or VSDL or *DSL. |
| 03:07.46 | ManxPower | linagee: There is no service in this building that goes into any kind of telco DSLAM |
| 03:07.58 | linagee | ManxPower: wait... i don't get it. the "frame bit" is counted when they talk about T1 bandwidth? but it's never usable? where's the logic in that... |
| 03:08.40 | linagee | s/1.544Mbps/1.536Mbps/ <--- for every instance when i've ever said 1.544Mbps |
| 03:08.53 | JT | linagee: it's more than one bit. |
| 03:08.58 | ManxPower | linagee: you never get the full 1544Kbps anyway because of protocol overhead |
| 03:09.03 | JT | linagee: both ends must synchronise |
| 03:09.16 | tzanger | heh |
| 03:09.17 | JT | TDM has superframes |
| 03:09.18 | tzanger | nice bit of history |
| 03:09.28 | linagee | ManxPower: true, but if you were running "bob joe's protocol" of sending packets, you'd expect to get 1.544 million bits per second. :( |
| 03:09.32 | tzanger | the 1-bit frame was chosen due to marketing saying "if you give us 8, we'll sell it as another timeslot" |
| 03:09.35 | JT | so it's a bit pattern, that the receiving end will recognise |
| 03:09.38 | linagee | ManxPower: s/packets/bits/ |
| 03:09.54 | tzanger | later on, E1 gave them 8 frame bits and sure enough, they were sold as an addditional timeslot for voice in some inmstances |
| 03:09.57 | tzanger | er instances |
| 03:09.59 | linagee | tzafrir: wtf that's messed up |
| 03:10.06 | ManxPower | linagee: honestly I've never missed those few bits. |
| 03:10.09 | tzanger | linagee: 1.536 is the raw rate (including frame bits) |
| 03:10.17 | tzanger | 1.544 is the acutal datarate |
| 03:10.18 | tzanger | er |
| 03:10.19 | tzanger | vice-versa |
| 03:10.20 | linagee | tzafrir: might as well inflate it a bit more by confusing bits/bytes (like harddrive manufacturers) |
| 03:10.55 | linagee | it's 193kibibytes/sec! really! :) |
| 03:11.09 | ManxPower | If I miss those 8 bits, my network is having more issues that missing 8 bits |
| 03:12.11 | linagee | tzafrir: is there a signaling method that says, "screw throwing away 1-bit every time, let's use that for data"? |
| 03:12.13 | ManxPower | linagee: and are your referring to ESF/B8ZS or AMI/D4? |
| 03:12.17 | ManxPower | I would have to look them up |
| 03:12.22 | linagee | ManxPower: i've never heard of either of those |
| 03:12.33 | tzanger | linagee: nope |
| 03:12.40 | tzanger | now you're pissing about for another 8kbps |
| 03:12.50 | linagee | tzafrir: hrm. deceptive marketing practices... |
| 03:12.56 | tzanger | not worth it, go get your cable or dsl internet |
| 03:13.05 | tzanger | leave me the hell alone with reliable standards |
| 03:13.24 | ManxPower | linagee: ESF/B8ZS would be the standard for most modern T-1s in the USA. It allows 64Kbps per channel, 24 channels per T-1. AMI/D4 (or is it SF/AMI?) supports 56Kbps per channel, 24 channels per T-1./ |
| 03:13.26 | linagee | tzafrir: lol |
| 03:13.42 | tzanger | ManxPower: ?? that's not how I know it |
| 03:13.48 | linagee | ManxPower: hrm |
| 03:14.03 | tzanger | AMI can't be used for data since it does not guarantee you will have the odd '1' ot keep the clock recovery happy |
| 03:14.09 | tzanger | B8ZS forces it |
| 03:14.20 | tzanger | you can do 64k with either, but AMI won't be (as) reliable |
| 03:14.48 | ManxPower | never heard that you could to 64K with AMI |
| 03:14.56 | tzanger | ManxPower: it's not a matter of can or cannot |
| 03:15.01 | tzanger | tis' a matter of reliability |
| 03:15.03 | ManxPower | But honestly, pretty much EVERYTHING I've done required 64K channels |
| 03:15.11 | tzanger | modern clock recovery can probably do very well with AMI and data |
| 03:15.22 | tzanger | and really AMI is almost unheard of these days |
| 03:15.42 | tzanger | much like a true physical T1 instead of the DS1-over-HDSL and DS1-over-HDSL2 we see now |
| 03:15.52 | ManxPower | tzanger: *nod* |
| 03:16.07 | JT | true phyiscal T1 :o |
| 03:16.21 | tzanger | and in your case... looks like dual-ds1-over-two-hdsl2-loops-with-a-funky-bonding-gadget-in-da-middle |
| 03:16.22 | JT | back when everyone was a few hundred feet from the co? |
| 03:16.29 | ManxPower | tzanger: "true" t-1 needed a repeater every 1,800 ft, didn't it? |
| 03:16.35 | tzanger | JT: T1 is both a physical and data specification |
| 03:16.40 | JT | right |
| 03:16.49 | ManxPower | rather than every 18,000 like "modern" T-1s |
| 03:17.59 | JT | i am rewriting the wiki article on span definitions |
| 03:18.02 | JT | becuase it sucked |
| 03:18.09 | ManxPower | If I get handed a DSX-1 interface from the telco and I get 24 64K dedicated point-to-point channels set for ESF/B8ZS then I consider it a T-1 no matter what the wizard is doing behind the curtian. |
| 03:18.23 | JT | and no noobs seem to be able to get span definitions right :) |
| 03:19.46 | ManxPower | I've started specifying "DSX-1 interface to CPE" to telecom vendors to make sure I never get handed another ethernet connection from a telco again |
| 03:19.48 | tzanger | oh for fuck sakes |
| 03:20.06 | tzanger | all that text in "core show application amd" but it doesn't tell you what time units all the settigns are in |
| 03:20.10 | tzanger | is it seconds, miliseconds, what? |
| 03:20.21 | tzanger | ManxPower: :-) |
| 03:20.48 | tzanger | looks like miliseconds |
| 03:21.33 | x86 | ManxPower: telco hands you an ethernet interface? how does that work? |
| 03:21.40 | x86 | ManxPower: MAN? |
| 03:23.06 | JT | there we go, a little better http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax |
| 03:29.01 | ManxPower | x86: They are idiots thats how it works. |
| 03:29.44 | *** join/#asterisk Sci_05 (n=Sci_05@ts.bestserversllc.net) |
| 03:29.49 | x86 | that's no joke |
| 03:30.11 | x86 | AT&T asked me if i wanted 5ESS or NI2 on a CAS T1 |
| 03:30.25 | JT | should've asked for etsi |
| 03:30.32 | x86 | i was like, uh, those are ISDN terms, not related to CAS at all ;) |
| 03:31.43 | Sci_05 | hello all |
| 03:31.45 | ManxPower | x86: I was told that I was going to get a "3Mbps T-1" I assumed that was two DSX-1 ports (standard T-1 from telco), so I ordered two more T-1 interfaces for our router, read up on Multilink, etc. |
| 03:32.26 | ManxPower | Then was handed an ethernet connection. I said a lot of bad words, including things like "fraud" and "contract violation" |
| 03:33.57 | ManxPower | As it is, we settled for just using the line for general data with no QoS. |
| 03:34.01 | Sci_05 | I have learned that when you order from at&t it always open to what you get, hell I got a new t1 from then 10 miles away from my other one. First hope came out in st. lewis, not to bad except I am in INDIANA! |
| 03:34.10 | x86 | JT: oh man that wiki entry made a WORLD of difference to me |
| 03:34.28 | x86 | JT: way easier to grok than the other wiki entry i tried to digest earlier |
| 03:34.34 | x86 | JT: thanks :) |
| 03:35.21 | x86 | ManxPower: what did they bring the ethernet in on? cat5 from the demarc? |
| 03:35.59 | x86 | Sci_05: haha |
| 03:36.23 | x86 | Sci_05: the company i work for has been using SBC for a very long time, with good success |
| 03:36.36 | x86 | Sci_05: but now that AT&T bought SBC back, it's turning into a nightmare |
| 03:38.50 | ManxPower | x86: An "ANDA NETWORKS ETHEREACH". I'd have to spend time in the DMARC closet to see where the two cables going into the box from the telco go. |
| 03:39.46 | x86 | ManxPower: uh, you don't know how they handed it off to you? |
| 03:39.52 | x86 | ManxPower: sounds like LRE to me |
| 03:40.09 | kikoy | hi how do i enforce a g711-only codec on my asterisk pbx? |
| 03:40.21 | tzanger | kikoy: with a big stick |
| 03:41.03 | snuff-away | mm... maybe simple disallow=all allow=ulaw would do |
| 03:41.09 | JT | x86: ah good to see it does help :) |
| 03:41.39 | kikoy | snuff-away: thanks man. alaw or ulaw? i'm confused with the two. |
| 03:42.00 | ManxPower | x86: The told me "You either connect into this box or you don't get a connection. Either way you are going to be billed for the service." |
| 03:42.26 | x86 | haha |
| 03:42.29 | ManxPower | kikoy: what country are you located in? |
| 03:42.35 | x86 | sometimes it helps to read the contracts you sign ;) |
| 03:42.45 | ManxPower | x86: The IT manager read and signed the contract. |
| 03:42.59 | ManxPower | I think he knows better than to do that without consulting me now. |
| 03:43.22 | x86 | i would hope ;) |
| 03:44.08 | ManxPower | x86: At HQ I could prolly pull off the ANDA and plug the two lines from the telco into the T-1 ports of the router (after I confirmed they are really T-1s), but at the remote location they have a NOC and they only hand me an ethernet cable and I do not have access to the physical ports from the telco |
| 03:45.00 | kikoy | ManxPower: Philippines here sir. |
| 03:45.12 | ManxPower | kikoy: then you would generally want "alaw" |
| 03:45.54 | kikoy | ManxPower: i've heard that using g711 would increase the max number of concurrent calls of my server. |
| 03:45.58 | ManxPower | x86: in any case nothing *I* am responsible for goes over that line, so I don't care. |
| 03:46.12 | snuff-away | uhmm kikoy i think u have it backwards |
| 03:46.18 | ManxPower | kikoy: G711 has the lowest CPU usage, but the highest bandwidth usage. |
| 03:46.19 | tzanger | I'm having a real brainfart right now |
| 03:46.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:46.31 | ManxPower | tzanger: the answer is 42 |
| 03:46.45 | kikoy | ManxPower: yeah.. i understand that... i don't have to worry with bandwidth. |
| 03:46.46 | tzanger | GotoIf($[DB(foo/bar) = ooga]) or GotoIf($[${DB(foo/bar)} = ooga]) ?? |
| 03:48.37 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:50.22 | Qwell | tzanger: the latter |
| 03:50.28 | tzanger | Qwell: that's kind of what I thought |
| 03:50.34 | ManxPower | x86: This all happened at a time when IT was under pressure to reduce it's reliance on consultants and so tried to so some projects in-house. When the projects didn't work, they started calling me again. |
| 03:50.53 | ManxPower | That happens every couple of years. |
| 03:52.17 | x86 | of course ;) |
| 03:52.38 | ManxPower | My invoices to them can get pretty hefty at times |
| 03:52.53 | *** join/#asterisk bmg505 (n=leon@196.209.180.30) |
| 03:54.24 | tzanger | can I use modulus in the dialplan? |
| 03:54.43 | tzanger | i.e. GotoIf($[${DB_RESULT} % 2 = 0]?even) ? |
| 03:54.53 | tzanger | it's getting late, I should be sleeping instead of trying to work |
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| 03:59.28 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 04:02.29 | *** join/#asterisk m55 (n=mahyar@S0106004005306d1c.gv.shawcable.net) |
| 04:02.55 | m55 | does anyone know of a voip provider that lists long distance fees by country/regional codes vs. names? |
| 04:03.20 | ManxPower | m55: Most of them |
| 04:04.39 | ManxPower | m55: http://teliax.com/rates.html |
| 04:04.41 | m55 | not that i've found |
| 04:05.05 | *** join/#asterisk jakehow (n=jakehow@cpe-66-108-161-123.nyc.res.rr.com) |
| 04:05.12 | ManxPower | lists the country code as well as any special rate / mobile codes that would have a different rate |
| 04:05.36 | m55 | damn, to call a sat phone costs 3.5 a minute |
| 04:05.43 | JT | cents |
| 04:05.44 | m55 | yet to call from a sat phone is 1.5/minute |
| 04:05.49 | m55 | no not cents |
| 04:05.52 | m55 | dollars |
| 04:06.01 | JT | sounds about right |
| 04:06.26 | m55 | they also have this access number thing where you call a us number and its around 1.4/min |
| 04:09.00 | jakehow | is having an outbound sip proxy enough to deal w/ calls originating from behind a NAT? |
| 04:11.58 | jakehow | is having an outbound sip proxy enough to deal w/ calls originating from behind a NAT? |
| 04:12.33 | JT | i swear i saw that very same question 3 minutes ago |
| 04:13.30 | jakehow | JT: i got disconnected |
| 04:13.35 | jakehow | so i might have sent it twice |
| 04:13.54 | JT | it didn't look like you disconnected, but alright :) |
| 04:14.01 | x86 | m55: there are some sat phone networks that cost upwards of $15 USD/minute to call |
| 04:14.02 | jakehow | nyc is wifi's worst nightmare |
| 04:14.20 | m55 | x86: wow |
| 04:14.27 | x86 | that's wholesale price too |
| 04:14.38 | m55 | then why is iridium so cheap |
| 04:14.39 | x86 | i'd be scared to call one of those numbers retail ;) |
| 04:14.54 | x86 | m55: which iridium network? there's like 3 |
| 04:15.09 | x86 | all with different rates, generally |
| 04:15.10 | JT | there's only one iridium constellation |
| 04:15.12 | jakehow | JT: any insight on the NAT issue? |
| 04:15.26 | x86 | JT: what am i thinking then? |
| 04:15.32 | JT | jakehow: yes, you don't even need a sip proxy, asterisk handles it fine |
| 04:15.33 | m55 | http://www.roadpost.com/satellite/iridium/subscription/iridium.asp |
| 04:15.36 | JT | x86: 3 sat networks? |
| 04:15.39 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.6.251.revip2.asianet.co.th) |
| 04:15.44 | JT | immarsat, iridium, globalstar |
| 04:15.46 | JT | a few others |
| 04:15.47 | jakehow | JT: asterisk is going to be outside of my NAT |
| 04:15.51 | jakehow | with clients inside |
| 04:15.55 | JT | jakehow: ok |
| 04:16.01 | JT | jakehow: will work fine if configured right |
| 04:16.02 | x86 | JT: there are certain sat phone NPA's that are insanely more expensive to call than others |
| 04:16.04 | m55 | cheapest iridium rate is 1.43/min |
| 04:16.16 | jakehow | im currently using a hosted provider and we need to configure an outbound sip proxy |
| 04:16.23 | JT | m55: i think immarsat (sp?) is one of the most expensive |
| 04:16.26 | m55 | where you call a US num and it redirects to the sat phone |
| 04:16.45 | tzanger | what the fuck |
| 04:16.55 | x86 | no, it's not redirecting... the sat phones are actually part of the NANPA |
| 04:17.04 | tzanger | exten => s,n,Set(DB(foo/bar)=0) |
| 04:17.12 | tzanger | database show is showing foo/bar as blank |
| 04:18.11 | m55 | let me find it |
| 04:18.15 | *** join/#asterisk hijacked (i=tvtV@cerebus.clandestineresearch.com) |
| 04:18.24 | m55 | it said something to that effect |
| 04:19.27 | m55 | http://www.roadpost.com/satellite/subscription/popup_iridiumrates.asp (incoming 2 stage dialing) |
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| 04:25.17 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:27.24 | m55 | you got to bloddy kidding me |
| 04:27.48 | m55 | globalstar is offering unlimited north america usage for $75 / month |
| 04:27.53 | m55 | http://www.globalstar.ca/en/content.php?cid=201#UNLIMITED |
| 04:28.26 | JT | m55: they're usually dual mode |
| 04:28.33 | JT | gsm or cdma + sa |
| 04:28.35 | JT | sat |
| 04:28.41 | JT | so that probably isn't for sat use |
| 04:29.50 | m55 | even if it's not for sat use it's still a good deal compared for what's offered in canada |
| 04:30.32 | m55 | a 700 min north america plan is $120+ |
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| 04:35.25 | *** part/#asterisk jaike (n=jaike@125.5.144.90) |
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| 04:40.51 | keys_ | I'm having trouble compiling asterisk on fc6 |
| 04:40.55 | *** part/#asterisk m55 (n=mahyar@S0106004005306d1c.gv.shawcable.net) |
| 04:41.01 | keys_ | chan_phone.c:40:29: error: linux/compiler.h: No such file or directory |
| 04:41.35 | Nugget | googling for that error message yields several promising hits. |
| 04:42.01 | Nugget | top hit is http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg24032.html |
| 04:44.59 | tzanger | hmm |
| 04:45.00 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:45.05 | keys_ | saw that one, got zaptel to compile from it. |
| 04:45.28 | tzanger | is there a way in the CLI to see if there are any pending outgoing calls (i.e. a callfile failed to execute and is waiting for retrytime) |
| 04:45.50 | snuff-away | unsure.. i've never used callfiles |
| 04:46.09 | JT | callfiles have retrytime? |
| 04:46.16 | tzanger | yep |
| 04:46.25 | tzanger | retry limits wait time, time betwene retries |
| 04:47.39 | tzanger | heh, I have a dialplan and callfile which, when combined with a cron job, incessantly call a person asking them to call back |
| 04:47.53 | tzanger | it avoids answering machines and can be enabled and disabled from the dialplan |
| 04:50.29 | snuff-away | heh |
| 04:50.48 | tzanger | if a human answers, it asks them to wait and calls another number (the person whose being avoided's cell, for example) |
| 04:55.46 | *** part/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 05:02.30 | ManxPower | tzanger: I have a few users that I could use that script with. I was planning on just randomly redirecting calls to their extension to other extensions, but I think this sounds more fun. |
| 05:02.43 | tzanger | ManxPower: :-) |
| 05:02.54 | tzanger | it's actually not for me but for a friend whose owed some money by a client |
| 05:03.04 | tzanger | they're avoiding his calls |
| 05:03.24 | ManxPower | I don't let clients get too far behind. |
| 05:03.31 | tzanger | yeah I learned that lesson |
| 05:03.32 | ManxPower | Yay! It's time to go home! |
| 05:03.34 | tzanger | :-) |
| 05:03.41 | *** part/#asterisk ManxPower (n=manxpowe@19.sub-70-218-94.myvzw.com) |
| 05:10.17 | *** join/#asterisk Loneshadow (n=vshivash@59.92.135.199) |
| 05:10.34 | Loneshadow | does anyone have slogsrv.zip from sipura.com ? |
| 05:10.52 | jameswf | hey all I have a file I am sucking into an array with file() each item has something to the effect of " item : value" I need to get all of these aray items into a mysql table any thoughts on easily doing this. |
| 05:12.03 | jameswf | damn wrong channel |
| 05:12.29 | bbryant_ | jameswf, yea, that sounds like ##php and you probably want the explode function ;) |
| 05:12.42 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 05:13.22 | rue_mohr | how can I do a "if time is greater than this go to this state, otherwise go to that one" in the extensions.conf |
| 05:13.34 | jameswf | yeah trying to go easy on the loops though |
| 05:13.44 | [TK]D-Fender | rue_mohr, "show application gotiftime" |
| 05:13.52 | *** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au) |
| 05:14.50 | rue_mohr | gotoiftime |
| 05:15.41 | litage_ | is it possible to configure asterisk to log CDRs for calls that go UA1->*->UA2 to file1/DBtable1, and for calls that go UA2->*->UA1 to file2/DBtable2 ? |
| 05:15.53 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 05:16.22 | [TK]D-Fender | rue_mohr, yeah, that |
| 05:16.32 | rue_mohr | :) |
| 05:17.09 | [TK]D-Fender | litage_, Not initially. Once the record is commited you can split things up of course |
| 05:17.38 | litage_ | [TK]D-Fender: yeah, i was just hoping it'd be possible to for asterisk to do it so that post-processing wouldn't be necessary |
| 05:17.59 | litage_ | s/to for/for/ |
| 05:18.09 | tzanger | dammit manxpower left |
| 05:18.42 | [TK]D-Fender | litage_, Fudging history is not *'s strong suit..... go try www.whitehouse.gov instead.... |
| 05:18.52 | litage_ | hahah |
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| 05:28.39 | [TK]D-Fender | ok, bed time, I'm out.... later all |
| 05:32.38 | De_Mon | 1hour to update to asterisk to 1.4.6 |
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| 05:51.02 | rue_mohr | is there a way to do a gosub/return? |
| 05:51.18 | rue_mohr | I suppose that wouldn't entirely make sense |
| 05:51.50 | yonahw-work | rue_mohr: what are you looking to do exactly? |
| 05:53.20 | rue_mohr | well, I have this duck quack and dog bark in two differerent menus, they play and take you back to the 'main' menu, in both routes to them use different numbers of digits, I think one is 306 and the other is just 8, I wanto make one definition of the each destination |
| 05:53.33 | rue_mohr | would you like to see the extensions.conf? |
| 05:53.37 | JT | De_Mon: what system? |
| 05:53.44 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-00653eb86657df4c) |
| 05:54.12 | yonahw-work | sure |
| 05:54.16 | rue_mohr | ok |
| 05:54.21 | rue_mohr | ~pb |
| 05:54.22 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 05:55.00 | *** join/#asterisk m55 (n=mahyar@S0106004005306d1c.gv.shawcable.net) |
| 05:55.03 | rue_mohr | I'm just working on cleaning it up a little |
| 05:55.07 | De_Mon | JT ? 1.2.14 -> 1.4.6 |
| 05:55.17 | De_Mon | debian packages |
| 05:55.42 | rue_mohr | http://paste.debian.net/32478 |
| 05:55.52 | De_Mon | it was tracking down all the config changes that I missed :) |
| 05:56.06 | rue_mohr | house is the context of local sets, telus is the context ofthe co |
| 05:57.29 | JT | De_Mon: ah ok |
| 05:57.34 | JT | packages are evil btw :P |
| 05:58.19 | yonahw-work | rue_mohr: I dont understand what are you trying to do? are you trying to return from telus to the duck or dog? are you trying to return from duck or dog to whereeever you came from to there? |
| 05:58.33 | De_Mon | JT nonsense! |
| 05:58.44 | JT | De_Mon: for asterisk they are |
| 05:58.51 | De_Mon | why would you say that? |
| 05:59.01 | JT | De_Mon: err, how long have you been in this channel? |
| 05:59.09 | De_Mon | 2 years? |
| 05:59.26 | JT | De_Mon: none of the regulars here recommend installing asterisk and zaptel from packages |
| 05:59.30 | JT | other software, sure |
| 05:59.32 | JT | use packages |
| 05:59.35 | JT | just not asterisk |
| 05:59.38 | De_Mon | because? |
| 05:59.41 | JT | they're out of data |
| 05:59.44 | JT | have weird patch sets |
| 05:59.47 | JT | date |
| 05:59.47 | De_Mon | date? |
| 05:59.56 | JT | and have poor defaults on some of them |
| 06:00.32 | De_Mon | that doesnt make packages evil. sure I could have packaged a more recent release, but this was easier. |
| 06:01.03 | *** join/#asterisk boooHah (n=chatzill@c-76-24-56-44.hsd1.ma.comcast.net) |
| 06:01.05 | De_Mon | bad defaults isn't a problem with packages so much as users who are dumb enough to use them |
| 06:01.10 | rue_mohr | from telus (incomming call) you can select 6 to hear the duck quack, from house, you can pick up and dial.... oops... I forgot that one, should be 402 to hear hte duck quack, anyhow I only want the duck quack programming to be in one place and then to use a goto to it, I have to make sure it returns to the right menu |
| 06:02.17 | JT | De_Mon: ok, just don't expect many to take you seriously here |
| 06:02.26 | JT | pakacged asterisk generally sucks |
| 06:02.37 | De_Mon | thats a retarded generalization |
| 06:02.47 | De_Mon | but I woln't hold it against you ;) |
| 06:02.55 | JT | it's possible you haven't paid any attention here |
| 06:02.59 | JT | but since you're the expert |
| 06:03.00 | JT | sure |
| 06:03.05 | JT | go ahead and do what you like |
| 06:03.12 | rue_mohr | heh, I'm having programming problems with my duck, and your not taking HIM seriously |
| 06:03.36 | yonahw-work | rue_mohr: I don't know if this is the best way to do it but I have set a variable to ${CONTEXT} prior to sending the call to the next context and then you can just have the return set to goto(${Variable},s,1) |
| 06:03.48 | rue_mohr | excelent |
| 06:04.01 | boooHah | general question re setup / system design ... if inappropriate pls do redirect me... |
| 06:04.35 | JT | rue_mohr: who would take HIM seriously? that's an emo band :P |
| 06:04.43 | De_Mon | thanks. Calling packages evil is like saying that car manufacturers are evil and everyone should build their own. maybe you just don't use the right manufacturer |
| 06:04.55 | boooHah | I'll be living in Canada, but making a *lot* of calls to the States -- and want to be easily reached from the States (i.e., via a US #) |
| 06:05.15 | JT | De_Mon: debian packages are some of the lesser evil ones, sure |
| 06:05.23 | JT | but they're still out of date |
| 06:05.28 | JT | due to deb's slow release cycle |
| 06:05.33 | rue_mohr | I'm gonna call it a night |
| 06:05.50 | De_Mon | again, I could have BUILT a deb package of the latest if I wanted to... it would have still been a package |
| 06:06.02 | JT | De_Mon: that would be fine |
| 06:06.03 | boooHah | want to be able to call from cell to States for cheap... assuming I can set up VoIP + asterisk + WiFi phone to do this(?...) |
| 06:06.17 | JT | because it would be the latest, with standard defaults on no unusual patches |
| 06:07.18 | boooHah | am I barkin' up the wrong tree here?... |
| 06:08.58 | JT | boooHah: it's probably possible |
| 06:09.03 | De_Mon | oooh, ok so packages are evil because they arn't bleeding edge enough. nevermind the fact that a packaging mechanism has nothing to do with the software being packaged.. |
| 06:09.39 | De_Mon | just dont go calling pkzip crappy because people only use it for old programs or something... |
| 06:09.40 | JT | De_Mon: most packages are great |
| 06:09.49 | JT | i use packes all the time |
| 06:09.54 | JT | just not for asterisk |
| 06:09.56 | JT | packages |
| 06:10.10 | JT | no need to be so defensive |
| 06:10.20 | JT | why would you need pkzip, generally? :) |
| 06:10.38 | De_Mon | im just trying to get a clear context |
| 06:11.33 | De_Mon | I agree, it'd be really nice if someone setup a debian repository with daily builds of asterisk. who knows... maybe one does exist somewhere |
| 06:12.01 | De_Mon | im just glad unstable has caught up to 1.4! |
| 06:12.05 | JT | it's not hard to compile, either |
| 06:12.20 | boooHah | JT: OK ... that's a start ... I'm trying to put together a strategy for doing this ... if this is not a good place to do so, can you suggest a better place???... |
| 06:12.24 | De_Mon | true, but it does get hard to manage |
| 06:12.29 | JT | also, a lot of people need to choose specific patches for zaptel or asterisk too, hence why packaging isn't good either |
| 06:12.47 | yonahw-work | De_Mon: I just have to say that I have not been around asterisk all that long but you are the first person I have ever seen spend so much time trying to argue that installing from a package is a good idea |
| 06:12.48 | JT | De_Mon: well your pbx shouldn't be automatically upgraded in any case |
| 06:13.09 | JT | boooHah: i suggest lest dots and question marks, and more specifics :) |
| 06:13.12 | De_Mon | or they get a package source and add/remove their own patches. its just as easy and you even get the management bonuses |
| 06:13.15 | yonahw-work | De_Mon: and you have yet to point out a single reason why this would be a good idea |
| 06:13.52 | boooHah | JT: Fair enough |
| 06:14.12 | De_Mon | yonahw-work im trying to clarify why its a 'bad' idea to begin with. the benifits for using packagemanagement have been well established in the linux community for some time |
| 06:15.19 | De_Mon | nobody can agree on which implimentation is the best one, maybe its that compileeverything package management system that doesnt even distribute binaries. |
| 06:15.57 | JT | De_Mon: i think apt-get is one of the best package management systems out there |
| 06:16.17 | JT | but we aren't talking about linux in general, we're talking about asterisk |
| 06:17.00 | De_Mon | don't disagree with that |
| 06:17.01 | redax | hi, |
| 06:17.24 | tzafrir | well, package asterisk yourself :-) |
| 06:17.39 | flenders | De_Mon: I've seen people screwing up their systems because they changed their apt/sources file |
| 06:18.02 | flenders | De_Mon: I think asterisk as a package is a bad idea, cause you shouldn't be updating your systems that often anyway. |
| 06:18.17 | De_Mon | lol |
| 06:18.36 | tzafrir | flenders, all the better reason: with a package you should know immediately what has changed (if at all) |
| 06:18.44 | flenders | De_Mon: I installed asterisk 1.4.0 and zaptel 1.4.0 and have been running them for 6 months, and I'm not upgrading them until I see a real bug on MY system |
| 06:19.04 | tzafrir | rpm -Vf /usr/lib/asterisk/modules/* |
| 06:19.25 | De_Mon | flenders what about critical security fixes? you don't apply those either? |
| 06:19.27 | flenders | I just think that a PBX shouldn't be updated that often |
| 06:19.38 | De_Mon | thats a debian repository I check for updates on regularly |
| 06:19.40 | tzafrir | (dpkg -S on the debian side) |
| 06:19.41 | boooHah | What I want to achieve: 1) Make calls from Canadian cell phone to US #s for cheap / free; 2) {vice versa, i.e.} Receive calls from US callers on my Canadian cell phone for cheap / free (i.e., presumably by calling US # that gets patched to my Canadian cell at my [limited!] expense); 3) will spend stretches in US, calling Canadian and US #s, and will not want to pay significant roaming... |
| 06:19.42 | boooHah | ...charges (pay as you go service?...) |
| 06:19.47 | JT | if it doesn't face the Internet, it doesn't matter, De_Mon |
| 06:19.48 | flenders | when we had a NEC phone system, they NEVER came to us to update it. |
| 06:20.21 | flenders | problem is that most sysadmins install asterisk, and they want to manage their servers as they manage a web server |
| 06:20.25 | flenders | it's not the same thing |
| 06:20.35 | tzafrir | flenders, great. apt supports pinning. pin the package apt to a specific version or to a specific repository (yours) |
| 06:20.52 | tzafrir | Nobody forces you to actually run updates, you know |
| 06:20.59 | JT | .win 96 |
| 06:21.18 | De_Mon | management practices for any business critical service are the same wheather you use an automated package management system or a manual administrative method. its just different tools |
| 06:21.29 | De_Mon | *fully* manual |
| 06:21.49 | boooHah | I think the following would work, but I am curious whether I could do it for less: 1) US-based VoIP; 2) Canadian VoIP; 3) Canadian cell; 4) asterisk to route calls around -- or is this [asterisk] unnecessary(?...) |
| 06:22.04 | tzafrir | flenders, but when you do upgrade your four servers, it is better to make the same screwup on all four automatically than to make a different screwup manually on each one. |
| 06:22.07 | flenders | if your apache dies, it doesn't matter, if your phone system dies, you're screwed |
| 06:22.28 | *** join/#asterisk stoffell_w (n=stoffell@fw.catsanddogs.com) |
| 06:22.29 | De_Mon | flenders bullshit. |
| 06:22.44 | JT | De_Mon: phones dieing does not matter? |
| 06:22.45 | tzafrir | flenders, hence using packages makes it easy to reproduce in production what you have in your testing environment |
| 06:22.49 | De_Mon | maybe if YOUR apache dies it doesnt matter. mines business critical |
| 06:23.08 | JT | De_Mon: even for a 1second? |
| 06:23.41 | De_Mon | ? |
| 06:24.14 | JT | http inaccessible for 1 second |
| 06:24.17 | JT | not ok? |
| 06:24.30 | De_Mon | no more ok than my phones being down for 1 second |
| 06:24.48 | JT | err ok |
| 06:25.01 | JT | for the great majority of businesses, phones are far more critical |
| 06:25.08 | De_Mon | so? |
| 06:25.30 | JT | so maybe you should stop being so defensive |
| 06:25.45 | flenders | De_Mon: you wanna maintain packages? do it mate |
| 06:25.47 | De_Mon | all I said is the management practices for maintaining business crtical services is the same regardless of the tools... |
| 06:25.54 | flenders | nothing against you wasting your time |
| 06:26.27 | De_Mon | quit making stuff up, I duno why im waiting my time trying to explain this stuff |
| 06:26.30 | flenders | De_Mon: what you don't understand is that a phone system is not a SERVER for most businesses... it's something you never touch |
| 06:26.48 | De_Mon | its a SERVICE |
| 06:26.49 | JT | De_Mon: "waiting your time" what does that mean? |
| 06:26.57 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:27.06 | JT | De_Mon: most businessed think "appliance" with their PBX |
| 06:27.14 | JT | it's not meant to be down, generally |
| 06:27.23 | JT | it's meant to just work as long as it's receiving power |
| 06:27.25 | flenders | De_Mon: mate, any other phone system would be screwed to a wall on a cupboard somewhere |
| 06:27.31 | flenders | and no one would touch it |
| 06:27.33 | De_Mon | I called it a business critical service |
| 06:27.36 | flenders | it's a phone system |
| 06:27.40 | flenders | like phone lines |
| 06:27.41 | De_Mon | I consider my apache to be a business critical service |
| 06:27.42 | flenders | don't touch it |
| 06:27.45 | De_Mon | they are both treated the same |
| 06:28.25 | flenders | De_Mon: apache is a critical service, no doubt about it... you can get hacked through apache, and unless your asterisk is on the internet, you CAN'T |
| 06:28.25 | De_Mon | the policies for updating and managing it are the same. the tools I use are different from yours. not the policies maybe but the tools man the tools |
| 06:28.32 | redax | if someone using blind transfer the BLINDTRANSFER channel var set. is there something similar to identify if someone used Attended transfer? |
| 06:29.25 | De_Mon | flenders you must not use voip |
| 06:29.46 | JT | De_Mon: voip on lan doesn't go to the Internet |
| 06:29.58 | De_Mon | with an itsp |
| 06:30.14 | De_Mon | finally starting to qualify stuff. wohoo makin progress... |
| 06:30.21 | boooHah | can someone recommend a place where I can ask some questions / try to get some advice on designing a system to meet my needs?... |
| 06:30.46 | flenders | boooHah: sorry mate, we all got carried away on the discussion |
| 06:30.56 | boooHah | flenders: NP |
| 06:31.17 | JT | De_Mon: a lot of us avoid ITSPs for business purposes |
| 06:32.20 | flenders | boooHah: so, what's your question? |
| 06:32.23 | snuff-away | mm some do use it just for international.. but yes generally avoided very well :) |
| 06:32.28 | De_Mon | that doesn't make them evil does it? |
| 06:33.20 | De_Mon | (yer supposed to say no) |
| 06:33.40 | flenders | what version is the current debian package on? |
| 06:33.57 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:34.20 | boooHah | flenders: What I want to achieve: 1) Make calls from Canadian cell phone to US #s for cheap / "free"; 2) {vice versa, i.e.} Receive calls from US callers on my Canadian cell phone for cheap / "free" (i.e., presumably by calling US # that gets patched to my Canadian cell at my [limited!] expense); 3) will spend stretches in US, calling Canadian and US #s, and will not want to pay significant... |
| 06:34.22 | boooHah | ...roaming charges (pay as you go service?...) |
| 06:34.35 | L|NUX | can some one tell me how can i make voicemail messages automatically save into Database |
| 06:34.48 | De_Mon | you have 3 repositories to pick from ranging from 1.2.? to 1.4.6 |
| 06:34.53 | flenders | boooHah: yeah, you can do that with asterisk |
| 06:34.58 | boooHah | flenders: I think the following would work, but I am curious whether I could do it for less: 1) US-based VoIP; 2) Canadian VoIP; 3) Canadian cell; 4) asterisk to route calls around -- or is this [asterisk] unnecessary(?...) You see where I'm coming from?... |
| 06:34.59 | JT | De_Mon: VoIPoI is just less reliable, that's all |
| 06:35.02 | De_Mon | official repositories anyways... |
| 06:35.05 | JT | sometimes it's an acceptable risk |
| 06:35.08 | JT | sometimes not |
| 06:35.40 | flenders | boooHah: you could forward calls on your voip providers, and then you wouldn't need asterisk |
| 06:35.53 | flenders | but it's better to use asterisk for that. |
| 06:36.10 | flenders | how many maintainers? |
| 06:37.37 | boooHah | flenders: sorry - you asking me?... "maintainers?..." umm, *one* - me [obviously a noob] |
| 06:37.37 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:37.51 | flenders | boooHah: no, it was to De_Mon |
| 06:38.02 | boooHah | flenders: aha |
| 06:38.38 | flenders | boooHah: you can dial into your asterisk box, get a dialtone and ring anywhere from there |
| 06:39.18 | flenders | boooHah: search for DISA on the wiki |
| 06:40.48 | De_Mon | flenders duno, theres a voip-maintainers mailist listed in the maintainers section |
| 06:40.52 | *** join/#asterisk grEvenX (n=even@ti500720a080-4273.bb.online.no) |
| 06:42.05 | boooHah | flenders: I see, but this would require *two* lines, would it not?... One to call into asterisk box, and one to call back out -- and could only support one such "pair" of calls at a time?... |
| 06:42.31 | De_Mon | tzafrir might have a better idea of how many maintainers are involved with asterisk |
| 06:43.29 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:46.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:49.36 | flenders | boooHah: most voip providers let you have simultaneous calls on the same 'line' |
| 06:50.15 | boooHah | flenders: any limit to # of simultaneous calls - or ltd only by bandwidth? |
| 06:50.29 | flenders | boooHah: it depends on your provider |
| 06:50.58 | flenders | boooHah: engin, in .au, lets you have as many calls as you want on the same account |
| 06:51.07 | flenders | simultaneous calls |
| 06:51.42 | boooHah | flenders: interesting... any idea re US-based providers?... |
| 06:54.58 | flenders | boooHah: I use via-talk, and I only had 2 simultaneous calls at once |
| 06:55.04 | flenders | so, not sure |
| 06:56.47 | yonahw-work | is there any reason why sip set debug would not show any messages |
| 06:57.15 | yonahw-work | other than sip debugging enabled that is |
| 06:58.02 | litage_ | what is the "calltype" CDR field for? it's null for all of my CDRs.. |
| 06:58.33 | boooHah | flenders: I was thinking, "Then I could eliminate Can. VoIP service," but that would make call in to asterisk (from cell) *international* (when WiFi not avail.) -- and that's not acceptable |
| 06:59.09 | flenders | yeah |
| 06:59.13 | JT | boooHah: have you had a look at the book yet? |
| 07:00.07 | boooHah | flenders: one possibility would be to have a US-based SkypeIn - or SkypeOut (whichever it is) - # that fwd'd to Can. VoIP #. I'd have to pay for that / min., but @ $.02 or whatever it is, it would probably be ok |
| 07:00.20 | boooHah | JT: "the book?..." |
| 07:00.40 | JT | ~thebook |
| 07:00.42 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:00.50 | flenders | wow, jbot is fast today |
| 07:01.04 | JT | boooHah: skype is a proprietary protocol and network, and asterisk cannot use it |
| 07:01.22 | flenders | I think he meant use skype instead of asterisk |
| 07:01.46 | snuff-away | i've seen a commerical product to convert skype to sip i think.. sounded dodgey to me |
| 07:01.57 | flenders | boooHah: problem is when you want to make calls to the US, for example |
| 07:02.21 | boooHah | JT: have *not* read the book... |
| 07:02.58 | flenders | snuff-away: I saw a chan_skype, I windows box running skype, and then it would somehow convert it to sip |
| 07:03.25 | flenders | snuff-away: I saw a chan_skype, I think, but you had to have windows box running skype, and then it would somehow convert it to sip |
| 07:03.41 | flenders | how did I miss the middle part of my sentence |
| 07:03.47 | flenders | some serious brain damage |
| 07:05.18 | boooHah | flenders: I'd need a Can. VoIP plan that incl'd US for free |
| 07:05.23 | JT | snuff-away: they are very didgy |
| 07:05.35 | JT | dodgy |
| 07:05.54 | JT | snuff-away: usually it involves pretending to be a client and bridging with the sound card |
| 07:08.34 | flenders | snuff-away: http://www.chanskype.com/faq |
| 07:08.41 | flenders | Do I need to install Skype in order to use ChanSkype? |
| 07:08.42 | flenders | Yes, you must have an X server and the Skype binary installed, both of which should be configured and running properly. |
| 07:17.28 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 07:25.07 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 07:28.34 | *** join/#asterisk Scrye (i=1127@204.244.252.35) |
| 07:28.56 | Scrye | when i call disa() i get a dialtone, but on any key, i get a fast-busy |
| 07:29.02 | Scrye | what can i use to debug this? it used to work |
| 07:29.27 | JT | Scrye: wrong context, bad entries in the context? |
| 07:29.30 | *** join/#asterisk arcanine (n=saxon_m2@222.127.84.165) |
| 07:29.56 | *** join/#asterisk boooHah (n=chatzill@c-76-24-56-44.hsd1.ma.comcast.net) |
| 07:30.12 | Scrye | exten => 7787863704/16046296899,1,goto(zingotel-in-disa|s|1) |
| 07:30.18 | Scrye | thats my sip channel in |
| 07:30.21 | Scrye | that works |
| 07:30.55 | arcanine | hi |
| 07:31.05 | Scrye | zingotel-in-disa 1=answer(), 2=wait(2), 3= nocdr(), 4= |
| 07:31.17 | Scrye | exten => s,4,disa(no-password|varinet-disa-authorized) |
| 07:31.43 | JT | Scrye: did you upgrade asterisk? |
| 07:31.44 | yonahw-work | ~pb |
| 07:31.45 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 07:31.57 | Scrye | i dont think i did |
| 07:32.23 | JT | Scrye: s is not a suitable context |
| 07:32.26 | JT | extensions even |
| 07:32.34 | Scrye | ok, what should go there? |
| 07:32.37 | JT | you need proper extensions or pattern matches |
| 07:32.41 | JT | for what you want to dial |
| 07:32.56 | Scrye | the guy who set this up is no longer available heh |
| 07:33.38 | JT | Scrye: btw, looks like disa within disa |
| 07:33.41 | JT | very inefficient |
| 07:33.55 | Scrye | disa() is getting called, its just when i hit a button, it fast-busys |
| 07:33.56 | JT | best to pastebin you whole extensions.conf |
| 07:34.09 | JT | ~pb |
| 07:34.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 07:35.09 | Scrye | http://pastebin.ca/613927 |
| 07:35.31 | Scrye | all i want to do is dial into my pbx so i can dial back out and take advantage of my free longdistance |
| 07:36.31 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 07:37.07 | boooHah | So I'm back - and getting a little clearer picture of how to do what I want to do: 1) Can. VoIP - that allows at least 6 simultaneous connections / calls, and cheap / free calling to US; 2) A US-based # that I can fwd to my Can. VoIP #; 3) Asterisk so that calls in to Can. VoIP can be fwd'd to appropriate cell phones - mine, my wife's, our son's; *and* so that each of us can call US via VoIP... |
| 07:37.08 | boooHah | ...line (rather than via cell carrier)... |
| 07:37.43 | JT | Scrye: you have no varinet-disa-authorized context. |
| 07:37.56 | Scrye | ok.. i wonder if i accidently toasted it |
| 07:37.58 | yonahw-work | i have a snom 300 which cannot register with my asterisk box i have posted the relevant sip.conf and sip trace at http://pastebin.ca/613931 can anyone tell me if they see something obviously wrong here please |
| 07:38.25 | yonahw-work | i also see nothing on the cli even with sip debug on |
| 07:38.51 | Scrye | if i change the varinet one to default would that work? |
| 07:39.09 | *** join/#asterisk shinao1 (n=shinao1@dial-pool27.lagos.starcomms.net) |
| 07:40.43 | mosty | yonahw-work, set verbose 10 in the asterisk console, then pastebin what you see in the console when you tell the snom to register |
| 07:41.06 | yonahw-work | mosty: verbose is 10 and I see nothing in the console when it tries to register |
| 07:41.29 | JT | Scrye: perhaps, pretty shakey security having no password on disa, but i guess it should work :) |
| 07:41.37 | mosty | yonahw-work, ok then go one layer deeper, use tshark/tethereal to see what packets arrive from the snom's ip address |
| 07:41.52 | Scrye | JT: callerid security |
| 07:42.00 | JT | Scrye: that's not security in the usa |
| 07:42.10 | JT | Scrye: almost all providers allow outbound callerid spoofing |
| 07:42.16 | JT | in the us |
| 07:42.27 | Scrye | well when it works again i'll throw on a pin |
| 07:43.18 | Scrye | Jul 11 00:45:47 WARNING[11486]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'zingotel-in-disa' |
| 07:43.20 | Scrye | what does that mean |
| 07:43.26 | JT | mosty: sip debug would be more useful than verbose t |
| 07:43.28 | JT | verbose 10 |
| 07:43.40 | yonahw-work | sip debug shows nothing as well |
| 07:43.42 | mosty | jt: yes but the sip debug showed nothing |
| 07:44.31 | JT | verbose 10 certainly won't show more then |
| 07:44.32 | mosty | yonahw-work, "tshark host X.X.X.X" |
| 07:44.37 | JT | and that was not a sip debug |
| 07:44.45 | JT | that was the sip dump from the phone |
| 07:44.53 | JT | the phone must be incorrectly configured |
| 07:45.02 | JT | or the network between the phone and asterisk |
| 07:45.04 | yonahw-work | i posted the sip dump |
| 07:45.15 | JT | yonahw-work: it was from the phone, wasn't it? |
| 07:45.21 | yonahw-work | yes |
| 07:45.57 | yonahw-work | where can i get tshark from |
| 07:46.17 | Scrye | ok i got it working now, thanks |
| 07:46.29 | *** join/#asterisk ozus (n=ozus82@65.110.43.185) |
| 07:47.01 | JT | Scrye: no probs :) |
| 07:47.18 | yonahw-work | is it part of wireshark? |
| 07:47.24 | JT | Scrye: i know you didn't set it up, but keep in mind putting most of your stuff in [general] is a bad idea |
| 07:47.26 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:47.28 | JT | security wise |
| 07:47.36 | Scrye | i think i was trying to clean it up a while ago and ripped apart the extensions.conf |
| 07:48.24 | JT | yonahw-work: i think the problem lies inside the configuration of the snom |
| 07:48.24 | JT | yonahw-work: is it behind nat? |
| 07:48.47 | Scrye | ok now onto a telco problem.. from my cell which is a ilec, i cant call my voip line sometimes, which the number is terminated in floria (4000 miles away) |
| 07:49.01 | Scrye | but from my other lines (seperate providers) i can call it |
| 07:49.03 | mosty | yonahw-work, it's part of ethereral/wireshark |
| 07:49.10 | Scrye | is that some sort of routing issue? |
| 07:49.14 | yonahw-work | JT: that has definitely occured to me especially since it is my first time working with snom phones though it seemed pretty straight forward |
| 07:49.20 | yonahw-work | phone is not behind nat |
| 07:50.26 | mosty | yonahw-work, also set debug to 9 on the snom |
| 07:50.51 | yonahw-work | mosty: where can i set debug on the phone? |
| 07:51.12 | mosty | yonahw-work, on the log page |
| 07:51.25 | mosty | it's called log, not debug, sorry |
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| 07:54.33 | yonahw-work | ok gonna go try all this and see where i get (I am on a different floor of the building on a different network) |
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| 07:57.04 | Swat2 | is SCA the same as SLA? |
| 08:01.14 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 08:03.30 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 08:07.22 | *** part/#asterisk Scrye (i=1127@204.244.252.35) |
| 08:08.00 | JT | Swat2: depends on the context i'd guess |
| 08:13.09 | yonahw-work | ok i posted the phone log and tshark log to http://pastebin.ca/613973 |
| 08:13.20 | *** join/#asterisk wedhorn (n=wedhorn@dsl-58-6-89-11.act.westnet.com.au) |
| 08:13.20 | yonahw-work | it seems to me that the problem must be on asterisks side |
| 08:15.03 | JT | yonahw-work: what's the asterisk server's ip? |
| 08:15.09 | *** part/#asterisk maerlin_ (i=merlin@dk0td.afthd.tu-darmstadt.de) |
| 08:15.11 | yonahw-work | the fact that I do not see any messages from asterisk whatosever is also kind of disconcerting |
| 08:15.23 | yonahw-work | jt the asterisk 192.168.2.254 |
| 08:15.28 | JT | and the phone? |
| 08:15.36 | yonahw-work | 192.168.2.154 |
| 08:15.51 | JT | the fact that the phone is trying to contact itself as a sip server is just a LITTLE disconcerting |
| 08:15.51 | bintut | i have a digium's dev kit here. how do you configure extensions.conf in such a way that whenever a pstn caller calls to my telephone number, the caller will hear a custom ring tone (maybe playing an mp3 file) instead of a default ringtone while the target extension in NoOp() and Dial() rings at the same time? |
| 08:15.56 | JT | see the issue here? |
| 08:16.06 | mosty | yonahw-work, are there any firewall rules on the asterisk box? and can you confirm that asterisk is listening on port 5060? |
| 08:16.23 | JT | mosty: won't help if the phone's only trying to call itself |
| 08:16.34 | JT | bintut: umm, i don't think asterisk can do that |
| 08:16.37 | yonahw-work | JT: where do you see the phone trying to call itself? |
| 08:16.42 | mosty | jt: the register is from .154 to .254 |
| 08:16.43 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 08:16.51 | JT | oh |
| 08:16.55 | yonahw-work | there are no firewall rules on the box |
| 08:16.58 | JT | you and your silly look alike ips |
| 08:17.06 | yonahw-work | JT: sorry about that |
| 08:17.16 | yonahw-work | how can i check that asterisk is listening on 5060? |
| 08:17.26 | mosty | netstat -na | grep 5060 |
| 08:18.00 | yonahw-work | returns udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 08:18.08 | JT | ok that looks fine |
| 08:18.16 | JT | where were you tsharking from? |
| 08:18.27 | yonahw-work | asterisk box |
| 08:19.07 | yonahw-work | i have been working on this for the last 24 hours and am going just a little nuts at this point |
| 08:19.30 | yonahw-work | the thought of taking the box and the phones and beating them with a baseball bat has definitely crossed my mind at this point |
| 08:19.31 | JT | probably doesn't matter, but this is wrong: localnet=192.168.2.254/255.255.255.0 |
| 08:19.39 | JT | 192.168.2.0/255.255.255.0 |
| 08:19.51 | mosty | that could be it |
| 08:19.55 | yonahw-work | really? |
| 08:19.56 | JT | i doubt it |
| 08:19.57 | *** join/#asterisk jm|laptop (n=jm|home@sentry.flags.co.uk) |
| 08:19.58 | yonahw-work | why is that wrong? |
| 08:20.13 | mosty | i've had incorrect localnet settings break sip before |
| 08:20.17 | JT | yonahw-work: because you specify the network address then the subnet mask |
| 08:20.30 | JT | not a random ip on the network and subnet mask |
| 08:20.39 | JT | mosty: i guess it's possible, asterisk can be pretty fickle |
| 08:20.50 | mosty | JT, although can't you determine the network ip given the subnet mask and an ip on the subnet? |
| 08:21.02 | JT | yes, in theory |
| 08:21.07 | JT | it's not technically correct though |
| 08:21.09 | yonahw-work | ok I changed that and still see no difference |
| 08:21.25 | yonahw-work | however I do not understand why it would seem like asterisk is not doing sip at all |
| 08:21.45 | yonahw-work | i can do zap no problem but i see no sip messages and nothing can register to it |
| 08:21.48 | mosty | if you do "sip show peers" does it show your peer definition? |
| 08:21.56 | yonahw-work | mosty: it does |
| 08:21.58 | JT | yonahw-work: on the asterisk box: iptables -L |
| 08:23.35 | yonahw-work | threw the iptables -L output on top at http://pastebin.ca/613982 |
| 08:23.45 | JT | yonahw-work: btw i still think your phone is also configured wrong |
| 08:23.46 | JT | # |
| 08:23.47 | JT | From: "7002" <sip:7002@192.168.2.254>;tag=pjj2j4v11p |
| 08:23.47 | JT | # |
| 08:23.48 | JT | To: "7002" <sip:7002@192.168.2.254> |
| 08:23.55 | *** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 08:23.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:23.56 | JT | from and to addresses are usually different |
| 08:24.33 | yonahw-work | i noticed that and thought it was strange but i reset the phone and just configured the registrar name and password |
| 08:24.40 | yonahw-work | i dont see how the phone could be setup wrong |
| 08:25.08 | *** join/#asterisk LuXten (n=noname@81.74.200.194) |
| 08:25.11 | mosty | what firmware version is it running? |
| 08:25.14 | yonahw-work | i even took the same phone down to a different box change those three settings as appropriate and it worked fine |
| 08:25.27 | yonahw-work | mosty: i think it is 6.5. something |
| 08:25.39 | JT | yonahw-work: iptables -L -v, is there any count against the REJECT rule in RH-Firewall-1-input ? |
| 08:26.20 | yonahw-work | JT: what would that look like? |
| 08:26.33 | JT | a number greater than 0 in the packet count column |
| 08:27.19 | yonahw-work | do you mean something like 14934 11M ACCEPT all -- any any anywhere anywhere state RELATED,ESTABLISHED |
| 08:27.58 | JT | yes, except i'm interested in the REJECT rule |
| 08:28.16 | yonahw-work | only 12497 9829K REJECT all -- any any anywhere anywhere reject-with icmp-host-prohibited |
| 08:30.03 | JT | lol |
| 08:30.12 | JT | ONLY 12497 rejected packets |
| 08:30.27 | JT | please pastebin.ca iptables -L -vv |
| 08:30.35 | JT | iptables -L does not reveal enough |
| 08:30.37 | bintut | JT: our mobile phone operators can provide their subscribers such feature but an additional cost to the subscribers.. hopefully, it will be supported by asterisk.. |
| 08:30.40 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 08:30.55 | JT | bintut: i realise some mobile phone providers do it |
| 08:30.56 | *** join/#asterisk viraptor (n=viraptor@87.194.164.154) |
| 08:31.14 | JT | bintut: their telephone switches and IN setups cost millions of dollars, also |
| 08:31.36 | viraptor | hi - any devs alive here? :) |
| 08:32.29 | LuXten | I have a Digium b410p. I just upgraded Asterisk from 1.2.19 to 1.2.20. After the upgrade asterisk is not recognizing dtmf tones coming in from isdn. The problem is in chan_misdn.so: if I switch it back to the 1.2.19 version, the problem disappears. Can I do something to further investigate the problem? |
| 08:33.10 | JT | LuXten: misdn is completely useless with dtmf recognition, you're very lucky it worked at all |
| 08:33.11 | yonahw-work | JT: I really appreciate your taking the time to help me out with this, I posted to http://pastebin.ca/613988 |
| 08:33.54 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:34.24 | LuXten | JT: Sorry, I'm pretty novice to *. You mean I can not serve an IVR, for example, on a b410p (and other mISDN driver devices)? |
| 08:35.02 | JT | LuXten: it is unfortunate that DTMF is so flakey in misdn |
| 08:35.33 | JT | yonahw-work: yes, you do have a firewall, and it IS blocking SIP. |
| 08:36.33 | yonahw-work | JT: how would I go about fixing that? |
| 08:37.00 | JT | yonahw-work: iptables -D INPUT 1 |
| 08:37.36 | Swat2 | Anyone know of some more detailed SLA Documentation |
| 08:37.43 | wedhorn | JT: that's a tad excessive |
| 08:37.50 | JT | wedhorn: it's perfectly fine |
| 08:37.58 | mosty | Swat2, ask the person you have the SLA with |
| 08:38.07 | viraptor | if someone is familiar with chan_iax - do you see sometimes bogus (I think) INVAL messages after call gets hangup? |
| 08:38.18 | JT | mosty: i think he means shared line appearance not service level agreements |
| 08:38.29 | JT | wedhorn: considering he thought there was no firewall |
| 08:38.34 | mosty | oh |
| 08:38.35 | Swat2 | mosty: nice work smart arse. |
| 08:38.40 | LuXten | JT: OK. Only one thing: I tried upgrading and downgrading mISDN (1.1.2, 1.1.3, 1.1.4). The problem is not there. The problem did arise in the 1.2.19-1.2.20 change in chan_misdn. The changes were not so long, some Ks of diff. Do you think I should signal the problem to the beronet guys, which seem to be the maintainers of chan_misdn? |
| 08:38.40 | wedhorn | JT: ok, I have only seen the last few minutes so don't know much about the box in question |
| 08:38.47 | JT | wedhorn: there is no point having a firewall if it is not maintained |
| 08:38.54 | Swat2 | ;) |
| 08:38.57 | mosty | Swat2, legitemate mistake- people call that BLF i think. |
| 08:39.01 | JT | Swat2: now now, just a misunderstanding |
| 08:39.02 | mosty | Swat2, which phone? |
| 08:39.10 | Swat2 | nah BLF is dif :) |
| 08:39.22 | Swat2 | JT: no malace :) |
| 08:39.25 | yonahw-work | JT: Thank you ever so kindly, I believe that solved the problem |
| 08:39.35 | Swat2 | Snom 340 |
| 08:39.41 | Swat2 | or is it snom 320 |
| 08:39.44 | JT | LuXten: you can also add it to the digium bugtracker, as they promise to support misdn's horrible bugs |
| 08:39.55 | JT | bugs.digium.com |
| 08:40.01 | mosty | don't think there's a snom 340. it's 300/320/360 that i know of |
| 08:40.10 | JT | yonahw-work: no probs :) |
| 08:40.12 | mosty | there's a section in the snom wiki about it |
| 08:40.14 | Swat2 | yeh snom 320 |
| 08:40.22 | JT | yonahw-work: always blame the network before asterisk if asterisk sees nothing ;) |
| 08:40.55 | LuXten | JT: Excuse me, with misdn do you mean mISDN (the kernel driver) or chan_misdn (the channel manager in *?) |
| 08:41.14 | JT | LuXten: well both is needed to make a b41-p function, are they not? |
| 08:41.23 | JT | b410p |
| 08:41.30 | ai-a | also snom 370 |
| 08:41.47 | viraptor | hmm... noone uses IAX around here? :) |
| 08:41.59 | Swat2 | mosty: on the snom website, or the aussie-voip wiki |
| 08:42.24 | Uatec | LuXten, which version of asterisk are you using? |
| 08:42.25 | mosty | Swat2, click "manual" on the snom phones web interface, it will take you to the snom wiki |
| 08:42.43 | LuXten | JT: Yes, sure. I'm just trying to understand if you mean that mISDN is buggy about dtmf, or it is chan_misdn, or you mean both. In my _present_ problem, the problem seem to be only in chan_misdn. |
| 08:43.17 | JT | LuXten: i'm not sure what's necessarily the culprit, i lump them together as one pile of bugs :/ |
| 08:43.28 | LuXten | Uatec: I'm using 1.2.19, now. I need dtmf to work. The problem came when I upgraded to 1.2.20. |
| 08:43.45 | Uatec | may i ask if there is a reason you're not using 1.3 or 1.4? |
| 08:43.45 | JT | LuXten: bristuff is much nicer, unfortunately there's no reports of anyone testing it with a b410p |
| 08:44.11 | Uatec | JT, i am currently in communication with digium business support about bristuff with the b410p |
| 08:44.22 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 08:44.25 | JT | Uatec: asterisk 1.3? :P |
| 08:44.45 | JT | Uatec: what do they have to say about it? |
| 08:45.10 | Uatec | they're being rather silent about it, which leads me to believe that they do not want anything to do with bristuff |
| 08:45.11 | LuXten | Uatec: Simply I'm new to *, I'm just breaking my bones on 1.2.XX, I would like not to introduce a "major change" right now. |
| 08:45.12 | Swat2 | mosty: i think i may of found exactly what i was looking for.. thanks (sometimes takes a fresh set of eyes to see things from a different perspective) |
| 08:45.23 | JT | Uatec: correct, it's all politics |
| 08:45.30 | JT | Uatec: hence why i won't buy a b410p |
| 08:45.37 | JT | since misdn is not what i consider usable |
| 08:45.55 | Uatec | digium make the b410p, but have never tested it in a country where it works, and have no first hand experience of using their own hardware |
| 08:46.11 | JT | :) |
| 08:46.14 | JT | or :( |
| 08:46.15 | Uatec | i pretty much had to figure out how to use the b410p in ABE 1.3 and tell them |
| 08:47.02 | JT | doesn't surpise me |
| 08:49.06 | JT | Uatec: basically, bristuff is disclaimed, digium cannot repackage it into ABE |
| 08:49.16 | Uatec | disclaimed? |
| 08:49.18 | JT | and they cannot remove klaus junghann's copyright messages |
| 08:49.27 | JT | and put mark spencer/digium there instead |
| 08:49.28 | Uatec | oh |
| 08:49.33 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 08:49.45 | Uatec | well i don't blame them |
| 08:49.46 | JT | Uatec: sorry, bristuff is NOT disclaimed |
| 08:49.47 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:49.58 | Uatec | i wouldn't out someone elses copyright information in my product |
| 08:50.26 | LuXten | Anyhow, I'm writing Beronet guys about my problem. chan_misdn stuff did change so little, I hope they are going to help me. |
| 08:50.28 | JT | Uatec: all patches to asterisk/zap/etc must have a faxed/written disclaimer from the author to say they relinquish copyright |
| 08:50.45 | JT | LuXten: but you didn't even buy a beronet card :) |
| 08:51.09 | JT | Uatec: well bristuff, the patches, are klaus's products, not digium |
| 08:51.10 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 08:51.19 | JT | Uatec: there was quite a lot of work involved |
| 08:51.20 | nicox | hello, |
| 08:51.41 | Uatec | if i buy a PC from Dell i expect a Dell logo on it, not an intel logo |
| 08:51.57 | JT | Uatec: yes they have intel logos on the, now why is that? |
| 08:52.05 | Uatec | do they? |
| 08:52.06 | JT | s/yes/yet |
| 08:52.07 | JT | yes. |
| 08:52.10 | JT | Intel Inside |
| 08:52.13 | Uatec | oh |
| 08:52.18 | Uatec | i haven't had a dell for... 100 years |
| 08:52.20 | JT | and if you open them |
| 08:52.22 | JT | omg intel |
| 08:52.23 | JT | ! |
| 08:52.36 | Uatec | if you open anything you'll find other stuff |
| 08:52.47 | Uatec | the branding on my keyboard here is microsoft |
| 08:52.53 | LuXten | JT: I did'nt even know they produce them, I'm very fresh in thelephony world. I just saw they seem the only people making changes to chan_misdn. |
| 08:52.55 | Uatec | but if i open it i bet i'll find other non-microsoft components |
| 08:52.56 | JT | Uatec: so then what the hell does it matter what the comments in source code says? |
| 08:53.04 | wedhorn | the intel logo is for marketing though (or maybe it's part of a deal between intel and dell) |
| 08:53.10 | JT | LuXten: that's the only reason they maintain the sofware |
| 08:53.14 | Uatec | JT, i have no idea. |
| 08:53.32 | JT | Uatec: the problem is digium want to dual license everything |
| 08:53.38 | JT | for ABE and GPL |
| 08:53.44 | Uatec | i don't see that as a problem |
| 08:53.51 | JT | you can't do that if a component is not disclaimed |
| 08:53.58 | JT | it is a problem |
| 08:54.09 | JT | if the code you want to put in ABE is not disclaimed |
| 08:54.28 | Uatec | i use ABE, and i will continue to use ABE becuase then if something goes wrong i can ask digium for help, and ultimately sue them if it fucks up beyond all recognition |
| 08:54.37 | JT | they also want to rebrand it all |
| 08:54.58 | JT | Uatec: nice in theory, hardly ever hear of anyone successfully suing a software company for software fucking up |
| 08:55.00 | Uatec | why not rebrand it |
| 08:55.03 | JT | and software almost always fucks up |
| 08:55.05 | mightnare | is using the plunger or flash button on an analog phone to do attended call transfers not recommended? |
| 08:55.16 | JT | Uatec: because they would breach copyright laws |
| 08:55.29 | Uatec | i mean why not let digium rebrand it? |
| 08:55.37 | Uatec | it's not like any end users will actually go out and get bristuff... |
| 08:55.40 | LuXten | JY: I'm just writing them as the maintainers. |
| 08:55.49 | yonahw-work | JT: well that did the trick, time for me to go learn about iptables and firewalls |
| 08:55.59 | Uatec | JT, yes i have. |
| 08:56.02 | Uatec | not just minor fuckups |
| 08:56.05 | Uatec | but major ones |
| 08:56.06 | JT | Uatec: because some people actually want to keep the copyright on stuff they put big time and effort in to |
| 08:56.09 | Uatec | breach of contract ones |
| 08:56.12 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:56.27 | yonahw-work | JT: , mosty: I really appreciate both of you helping me out like that saved my ass and I got a flight overseas tonight |
| 08:56.29 | Uatec | you can keep copyright and still allow rebranding |
| 08:56.36 | JT | yonahw-work: no probs |
| 08:56.44 | JT | Uatec: no, you cannot, not with code in asterisk |
| 08:57.00 | JT | Uatec: to keep copyright, your copyright message must remain |
| 08:57.09 | JT | and you must allow them to do what they want so they can make ABE |
| 08:57.18 | Uatec | not if money changes hands |
| 08:57.31 | JT | what? |
| 08:57.34 | Uatec | i have published forums with no developer branding on at all |
| 08:57.36 | wedhorn | JT: disclaiming and allowing it into the asterisk base only impacts on two rights, adding your name, and ensuring that it can't be included in close source |
| 08:57.38 | Uatec | and the copyright still remains |
| 08:57.46 | wedhorn | however, it will still be open source as well |
| 08:57.57 | JT | wedhorn: exactly, so pretty much all your rights. |
| 08:58.14 | wedhorn | how, it is basically dual source, open and closed |
| 08:58.26 | JT | Uatec: it's not about branding, it's about the copyright to the code. |
| 08:58.59 | JT | wedhorn: if you reqlinquish copyright, you relinquish your rights to the intellectual property, i though that was sort of obvious |
| 08:59.08 | Uatec | you can still maintain copyright without copyright markings |
| 08:59.11 | Uatec | well you can in my country |
| 08:59.22 | JT | s/though/thought/ |
| 08:59.37 | wedhorn | but what is the value of intellectual property of open source |
| 08:59.45 | JT | quite high |
| 08:59.57 | JT | for useful pieces of software |
| 09:00.12 | wedhorn | how, if it is open source anyway? |
| 09:00.16 | mosty | intellectual property is a vauge term that has no real meaning |
| 09:00.22 | mosty | vague, even |
| 09:00.55 | Uatec | mosty, apart from when it has been legally defined, which it has in most countries |
| 09:00.56 | JT | wedhorn: you must be pretty ignorant if you think intellectual property has no value simply because it has been released under a FOSS licence |
| 09:01.12 | Uatec | JT, we're not talking about relinquishing copyright, just removing the copyright markings |
| 09:01.23 | Uatec | my car does not have my name on it but it's still mine |
| 09:01.31 | JT | Uatec: read the damn disclaimer, it IS RELINQUISHING COPYRIGHT |
| 09:01.33 | JT | okay |
| 09:01.38 | JT | i've signed one |
| 09:01.52 | JT | Uatec: i am trying to explain why bristuff is not included |
| 09:01.55 | mosty | Uatec, i've never seen it defined in legal documents. they tend to use the actual terms copyright, patent, etc |
| 09:01.56 | JT | if you can't accept that |
| 09:01.58 | JT | that's too bad |
| 09:02.04 | JT | i'm only trying to make it clear |
| 09:02.27 | Uatec | the point of the conversation is, digium to not support or advocate Bristuff, only misdn |
| 09:02.41 | JT | or provide bristuff |
| 09:02.43 | JT | correct |
| 09:02.55 | Uatec | i say they support misdn, it's damned hard to get any help out of their support at all |
| 09:03.03 | JT | what's new |
| 09:03.14 | JT | they only provide installation support, not configuration |
| 09:03.21 | Uatec | i know |
| 09:03.22 | JT | ever try to get support out of microsoft? :) |
| 09:03.27 | Uatec | but i haven't got it installed properly |
| 09:03.38 | Uatec | yes, i have, but we're microsoft gold partners, so it's not that hard |
| 09:03.54 | JT | most people can't get any support out of microsoft |
| 09:03.57 | nicox | hello, JT, are this data of zttest okay ? Best: 100.000000 -- Worst: 99.816895 -- Average: 99.987385 |
| 09:04.17 | Uatec | i will consider it "installed properly" when Dial(misdn/g:isdn/123456) dials out on round robin over all lines and channels |
| 09:04.24 | JT | nicox: no, it fell bellow 99.97% at some stage |
| 09:04.27 | Uatec | at the moment it only dials out on channel 1 on any line |
| 09:04.30 | JT | i don't like zttest totals too much |
| 09:04.55 | JT | Uatec: TE or NT mode? |
| 09:04.56 | mosty | Uatec, how many lines are defined in misdn.conf? |
| 09:05.09 | Uatec | 4 |
| 09:05.11 | Uatec | TE mode |
| 09:05.23 | JT | hmm okay |
| 09:05.27 | JT | thought that did work |
| 09:05.33 | JT | but not surprised :P |
| 09:05.35 | Uatec | not in ABE |
| 09:05.44 | JT | 4 TEs, why not go PRI |
| 09:05.53 | nicox | i see 99.902344% at the output where some of the got reject-prbomlem occurs |
| 09:06.08 | Uatec | cos it's a research project |
| 09:06.44 | Uatec | also PRI only becomes economic when we need more than 8 lines.. |
| 09:07.13 | JT | nicox: ouch, that's not cool |
| 09:07.21 | JT | Uatec: ah that sucks |
| 09:07.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:07.48 | nicox | whats the problem? |
| 09:08.13 | JT | nicox: 99.90% is a major problem |
| 09:08.22 | JT | poor zap timing |
| 09:09.08 | Uatec | meh, research project is a research project |
| 09:09.09 | nicox | zap timing? is it possible to have the "client" of the clock on the NT-side of an E1? |
| 09:09.15 | Uatec | but i think the conclusion of it is that we will go PRI |
| 09:09.23 | Uatec | and we will tell our customers to go pri |
| 09:09.34 | JT | nicox: i don't understand the question |
| 09:09.55 | JT | Uatec: yes i'd avoid bri if possible, especially if bristuff was not an option |
| 09:10.14 | JT | digital is nice |
| 09:10.28 | JT | but misdn can be more annoying to work with than analogue phone lines |
| 09:10.34 | nicox | on each E1 there is an net and cpe mode, also an clock "master" and clock "client" |
| 09:10.44 | JT | nicox: correct |
| 09:10.54 | nicox | is it possible to be a clock client if the E1 is in net mode? |
| 09:11.29 | JT | it's possible, it's not normal |
| 09:11.53 | JT | nicox: anyway, i can't remember your setup |
| 09:11.54 | Uatec | i'm just annoyed that digium sell a product about which they know nothing |
| 09:11.56 | Uatec | not just sell, develop |
| 09:12.03 | nicox | is it possible to have 3 clock sources (clock "clients") on one digium card? |
| 09:12.04 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 09:12.06 | wedhorn | JT: I understand the value of intellectual property, the question was what is the value to the author if they put that property under the GPL. |
| 09:12.07 | JT | Uatec: it is frustrating |
| 09:12.45 | JT | wedhorn: they are free to do with it as they wish, including relicense it, and they ovbiously receive the credit for it |
| 09:13.03 | *** join/#asterisk oej (n=olle@174.82-134-80.bkkb.no) |
| 09:13.06 | JT | nicox: you mean, receiving timing, or sending timing? |
| 09:13.17 | nicox | receiving |
| 09:13.22 | JT | nicox: yes. |
| 09:13.36 | JT | use different priorities for them of course |
| 09:14.07 | wedhorn | JT: ok, I'll have to go reread my disclaimer, but I didn't actually think you were removing your own rights, but rather giving digium the same rights |
| 09:14.37 | JT | wedhorn: you're removing all legal recourse with respect to the code you submit, and ownership |
| 09:14.48 | nicox | so each E1 i can use their own timer? priorities, i think thats the span=1,1,x,ccs,hdb3,crc4 |
| 09:14.56 | nicox | can i set it from 1 to 4 ? |
| 09:14.58 | JT | no |
| 09:15.11 | JT | span=1,1,x x is line build out |
| 09:15.14 | JT | almost always is 0 |
| 09:15.25 | *** join/#asterisk Gled|work (n=gled@LPuteaux-151-42-17-115.w193-252.abo.wanadoo.fr) |
| 09:15.27 | JT | nicox: read http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax |
| 09:15.32 | JT | i updated that article today |
| 09:16.02 | nicox | sorry, |
| 09:16.05 | nicox | i understand |
| 09:16.18 | nicox | and also the priority 3 and 4 is possible? |
| 09:16.47 | JT | yes |
| 09:17.09 | nicox | so, now i have to restart with 60 calls on it ..... |
| 09:17.13 | nicox | shit! |
| 09:17.24 | nicox | sorry |
| 09:17.32 | JT | what was your config like before? |
| 09:18.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:18.38 | nicox | span 1 and 2 has no clock source (both in net mode) and the priority was 0) and the third span had the priority 1 and was in cpe mode |
| 09:18.50 | JT | err |
| 09:18.54 | JT | why change it? |
| 09:19.03 | JT | if they are indeed acting as network |
| 09:19.20 | JT | network to what btw, a channel bank? |
| 09:19.38 | wedhorn | JT: may not be the latest, but http://asterisk.org/files/disclaimer.txt indicates that you give digium a "non-exclusive" licence, so you could still use your own code in any licence you want, including selling it to microsoft (non-exclusive licence as well of course) |
| 09:20.02 | nicox | to a soft switch from digitalk, but they have another clock source and are not so configurable |
| 09:20.03 | Habbie | of course you give them a non-exclusive license, how else could they gpl-license it :) |
| 09:20.43 | JT | <PROTECTED> |
| 09:21.07 | JT | you can still maintain copyright and release under open source licences |
| 09:21.31 | Habbie | true |
| 09:21.39 | JT | any code you do release under those licenses will be subject to the conditions of those licenses |
| 09:21.58 | JT | which may include the source being available |
| 09:22.14 | Habbie | yeah i mixed it up |
| 09:22.15 | JT | nicox: anyway, you're barking up the wrong tree |
| 09:22.18 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 09:22.28 | nicox | what else is the problem? |
| 09:22.32 | JT | nicox: timing issues are almost always a hardware problem |
| 09:22.43 | JT | or closely related hardware configuration issue |
| 09:22.44 | nicox | hardware-problem |
| 09:22.47 | nicox | hm.... |
| 09:22.54 | JT | nicox: digium cards are quite finicky |
| 09:23.01 | JT | what is the specifications of the machine |
| 09:23.06 | JT | and is it doing irq sharing? |
| 09:23.56 | nicox | pentium d 3.2 GHz |
| 09:23.58 | nicox | 58: 746806349 503265659 IO-APIC-level wct4xxp |
| 09:24.02 | nicox | no irq-sharing |
| 09:24.44 | JT | chipset? |
| 09:24.48 | JT | back in a bit.. |
| 09:24.56 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:27.13 | nicox | and a machine of Supermicro 5015 Intel E7230 (Mukilteo) Smithfield chipset |
| 09:34.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:35.48 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-00653eb86657df4c) |
| 09:38.11 | nicox | but not the pdsme board as described on voip-info.org, in this server there is the pdsmi-board |
| 09:40.21 | bintut | JT: sorry, i was out lately.. |
| 09:41.31 | nicox | no problem |
| 09:42.27 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:45.25 | nicox | no i got this error on a second server, the first time that is error occurs there, and nothing was changed.... very dubious! |
| 09:46.24 | *** join/#asterisk ozus (n=ozus82@65.110.43.185) |
| 09:48.12 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.94) |
| 09:50.13 | nicox | do you have any idea? |
| 10:00.49 | *** join/#asterisk aiksa[LV] (n=root@195.2.112.177) |
| 10:00.57 | *** join/#asterisk duncanmhor (n=chatzill@outmail.fandisolutions.co.uk) |
| 10:01.31 | JT | nicox: try changing pci slot |
| 10:02.55 | nicox | its an 1 U server |
| 10:02.59 | *** join/#asterisk mightnare (n=mike@p5142-ipad12motosinmat.mie.ocn.ne.jp) |
| 10:03.59 | duncanmhor | Hi. I've been asked by our Telco (virgin media, UK) to provide them with our Type of Number and Screening Indication type. We're running asterisk 1.2.9.1. Can anyone point me to where I would find this information? |
| 10:04.10 | aiksa[LV] | exit |
| 10:05.13 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
| 10:09.11 | nicox | any other hint? |
| 10:09.18 | colde | Hi, im trying to do the classic "press 1 for sales, press 2 for other department" thing, however, when asterisk is finished listing the choices using Background(), it hangs up, any ideas why? |
| 10:11.55 | Sci_05 | colde: after you have the background(), is the following line something like this |
| 10:11.59 | Sci_05 | exten => did,n,WaitExten |
| 10:12.54 | colde | nope |
| 10:13.32 | Sci_05 | toss that line in there, it tells * to wait for an extension |
| 10:13.42 | colde | nice |
| 10:13.44 | colde | thanks |
| 10:14.29 | colde | can i do it without specifiing number of seconds to wait and it will just wait indefinately? |
| 10:16.18 | Sci_05 | I believe it will wait while the message is being played from background(), after that it will go to the following line |
| 10:16.28 | Sci_05 | my following line after that is this |
| 10:16.48 | Sci_05 | exten => did,n,Dial(SIP/102&SIP/101&SIP/111,25,tr) |
| 10:17.17 | Sci_05 | so the exten => did,n,WaitExten should just wait till the background sound is done |
| 10:17.46 | colde | oh ok |
| 10:17.58 | colde | arh, i just slap in the t extension to repeat it all after timeout |
| 10:19.40 | Sci_05 | well my default is, if they don't enter an extension then i have it ring all the phones |
| 10:20.28 | Sci_05 | crap g2g catch a plane, god luck colde |
| 10:21.12 | cy- | holy crap 1.4.7.1 already out?! |
| 10:21.19 | cy- | seems like I JUST upgraded to 1.4.5 |
| 10:21.22 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 10:22.05 | Sci_05 | cy ya they are fixing a lot of bugs, 1.4.7 was just out the other day then the 1.4.7.1 cam out a few days later |
| 10:22.12 | Sci_05 | later all |
| 10:23.18 | cy- | heh |
| 10:23.23 | cy- | ridiculously fast development |
| 10:23.33 | cy- | which isn't a bad thing at all, it's just like uhhhh, damn, I have to upgrade again now! :P |
| 10:27.09 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 10:32.51 | *** join/#asterisk xezz (n=xxx-4011@83.235.189.2) |
| 10:32.55 | xezz | #trixbox |
| 10:33.01 | xezz | soz |
| 10:35.01 | xezz | hello, i would like to make some specific calls via shell script, like: dial <phone> <extention>, any idea ? |
| 10:37.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:47.36 | cy- | after upgrading to 1.4.7.1 I get error .. my modules directory contains modules that were not installed by this version of asterisk.. chan_features, format_mp3.so, app_saycountpl.so and chan_ooh232.so .. is this okay? |
| 10:48.49 | JT | xezz: i've already point you which way to go |
| 10:48.53 | JT | pointed |
| 10:49.02 | JT | xezz: please take a look into the block |
| 10:49.05 | JT | ~thebool |
| 10:49.08 | nicox | hi JT, any idea? |
| 10:49.09 | JT | ~thebook |
| 10:49.10 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:49.33 | JT | nicox: no |
| 10:49.56 | JT | nicox: get a new card, new server, try a few tricks |
| 10:50.25 | Uatec | hey, i'm using the standard asterisk voicemail. When leaving a message, it plays my recording, and then after that it says "Please leave you message after the tone. When done hangup or press the # key." How can i turn this off? We have no need, nor desire for this message after we have left our own. |
| 10:52.40 | nicox | nice idea, thanks. |
| 10:52.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:52.58 | *** join/#asterisk JunK-Y (n=junky@modemcable094.9-83-70.mc.videotron.ca) |
| 10:54.07 | snuff-away | Uatec, ... add a 's' to the voicemail options when u call it |
| 10:54.14 | snuff-away | that will skip that message |
| 10:54.26 | snuff-away | the 'instructions' per say |
| 10:55.52 | Uatec | ahh |
| 10:56.11 | Uatec | as in VoiceMail(us123) where 123 is my mailbox number... |
| 10:57.18 | snuff-away | yes |
| 10:57.37 | Uatec | YAY |
| 10:57.38 | Uatec | thanks |
| 10:57.44 | snuff-away | i think.. new format is Voicemail(123|su) but anyway :) |
| 10:59.29 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 11:00.16 | Uatec | oh |
| 11:00.18 | Uatec | well my way works |
| 11:02.39 | *** part/#asterisk wedhorn (n=wedhorn@dsl-58-6-89-11.act.westnet.com.au) |
| 11:12.26 | yonahw-work | does anyone know where I can obtain the latest unixODBC? on unixODBC.org the lastest is 2.2.12 but I see references on voip-info and elsewhere to 2.2.7 and even 2.2.9 |
| 11:12.42 | *** join/#asterisk Strom_M (n=strom@12.175.45.209) |
| 11:13.21 | mosty | 2.2.12 is newer thab both of those |
| 11:13.28 | mosty | than |
| 11:13.46 | mosty | 12 > 9 > 7 |
| 11:14.03 | tzafrir | yonahw-work, hmmm... why not use the package from your distro? |
| 11:14.13 | yonahw-work | mosty: thank you for clarifying for some stupid reason i was thinking that it was 2.2.1.2 |
| 11:14.16 | *** join/#asterisk Stormfr (n=StorM@stardust.noc.frontier.fr) |
| 11:14.27 | yonahw-work | tzafrir: I saw mention that people have had problems using the packages |
| 11:14.32 | yonahw-work | is this not really so? |
| 11:14.40 | tzafrir | what distro is that? |
| 11:14.47 | yonahw-work | centos 4.4 |
| 11:15.16 | tzafrir | which specific problems? Are those not Asterisk problems? |
| 11:15.41 | yonahw-work | http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc |
| 11:16.10 | Stormfr | any use TC400B (DSP G729/3 card) and get success to get command show transcoder, i don't find also where it could in the source code ? |
| 11:16.25 | mosty | yonahw-work, use cdr_pgsql and postgresql |
| 11:16.29 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 11:16.33 | MrWup | hey guys im having echo problems |
| 11:16.34 | duncanmhor | 've been asked by our Telco (virgin media, UK) to provide them with our Type of Number and Screening Indication type. We're running asterisk 1.2.9.1. Can anyone point me to where I would find this information? Are there alternative search terms I should be using ? |
| 11:16.43 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:16.44 | mosty | stormfr: you need to load the correct zaptel module first |
| 11:16.48 | MrWup | im using BRI ISDN lines on a digium b410p card |
| 11:16.56 | MrWup | ive set echocancel=yes in misdn.conf |
| 11:16.56 | Stormfr | mosty : driver is up and card is detect |
| 11:17.05 | Stormfr | mosty, also asterisk view the card detected |
| 11:17.06 | MrWup | and my rcv amp is 8 and transmit amp is 4 |
| 11:17.10 | Dr-Linux | which one is best and friendly asterisk billing opensource program? |
| 11:17.11 | yonahw-work | the database already exists on MSSQL and is there for other reasons I don;t really have a choice here |
| 11:17.12 | MrWup | but still im getting pretty bad echo |
| 11:17.15 | MrWup | i have no idea why |
| 11:17.16 | Dr-Linux | need suggestion |
| 11:17.27 | mosty | yonahw-work, also, can't you use cdr_tds for mssql? |
| 11:17.40 | yonahw-work | mosty: yes that is also possible |
| 11:17.45 | yonahw-work | what are the advantages |
| 11:17.56 | mosty | 1. it isn't odbc |
| 11:18.38 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 11:18.39 | yonahw-work | mosty: what is so bad about odbc |
| 11:18.44 | MrWup | also, echotraining=no as i understand this interferes with the hardware echo cancelling on the card |
| 11:19.08 | mosty | odbc is one more level of indirection that can break. plus it's less efficient than native database drivers |
| 11:19.45 | *** join/#asterisk jeffik (n=Valued@206-248-128-113.dsl.teksavvy.com) |
| 11:20.48 | yonahw-work | hmm, I hear you but I actually have used odbc in the past and have only heard about FreeTDS today which leaves me feeling more comfortable using ODBC |
| 11:21.29 | *** join/#asterisk s0lid (n=jlq@210.213.240.236) |
| 11:22.12 | *** part/#asterisk s0lid (n=jlq@210.213.240.236) |
| 11:23.33 | *** join/#asterisk figu78 (n=figu78@26.Red-80-26-83.staticIP.rima-tde.net) |
| 11:23.41 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 11:23.57 | figu78 | hi |
| 11:24.13 | figu78 | anyone has 5 minutes to help me?? |
| 11:24.18 | MrWup | anyone had echo problems with b410p? |
| 11:24.44 | mosty | MrWup, what kind of phones? |
| 11:24.58 | mosty | ie are the calls SIP <-> BRI? |
| 11:26.06 | Err | figu78: it would be a lot easier for someone to help you if you asked a question :-) |
| 11:26.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:27.00 | figu78 | I need to transfer calls from reception to some extensions (hardphones) |
| 11:27.10 | figu78 | basically |
| 11:27.23 | JT | MrWup: i thought the b410p was supposed to have hardware EC |
| 11:27.28 | JT | maybe it's not enabled |
| 11:28.11 | *** join/#asterisk Corydon76-home (i=gray@pdpc/supporter/sustaining/Corydon76-home) |
| 11:28.11 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:29.40 | figu78 | someone call from outside, the recepcionist pick up the phone, send this call to a wait state, the recepcionist, call the extension, and if there is someone at the end, the recepcionist send the waiting call to the final extension |
| 11:30.22 | mosty | figu78, do an attended transfer |
| 11:31.00 | figu78 | mosty, ummm as you can see im a newbie |
| 11:31.21 | mosty | do a search for asterisk attended transfer on voip-info.org |
| 11:31.45 | figu78 | mosty, doing (thanks) |
| 11:32.05 | *** join/#asterisk bintut (n=bintut@cm177.gamma177.maxonline.com.sg) |
| 11:33.34 | *** join/#asterisk oej (n=olle@174.82-134-80.bkkb.no) |
| 11:37.45 | figu78 | mosty, This is excatly what i was doing |
| 11:38.59 | figu78 | mosty, but everytime i did it in the last days, the server sent a panic and i had to restart the box |
| 11:39.04 | *** join/#asterisk xand (i=alex@zero.xand.co.uk) |
| 11:39.36 | MrWup | JT - i know, thats what it says. but according to all the help files and google etc, the hardware echo canc is turned on by setting echocancel=yes in misdn.conf |
| 11:39.41 | MrWup | and ive done that so dont know whats wrong |
| 11:40.02 | MrWup | some calls i get no echo, some calls i get terrible echo |
| 11:44.12 | Err | figu78: if you mean a *kernel* panic, that's not an asterisk problem - that's a hardware or hardware driver problem |
| 11:45.00 | figu78 | err, i tried with several drivers with the same result |
| 11:45.18 | mosty | figu78, drivers for what? |
| 11:45.18 | *** join/#asterisk basty (n=basty@212.218.65.230) |
| 11:45.20 | basty | Hi |
| 11:46.07 | basty | I have Problems using Asterisk 1.2.19 with SpanDSP. SpanDSP is answering the call and hangsup the fax after 10 Secs. Anyone used to have the same issue ? |
| 11:46.30 | figu78 | I saw there is an open bug about it, but obviously not everybody is getting the same thing |
| 11:47.02 | mosty | an open bug for what? |
| 11:47.23 | figu78 | http://bugs.digium.com/view.php?id=6876 |
| 11:47.58 | figu78 | it seems the threewaycalling is not working properly with analog phones |
| 11:48.43 | mosty | but you're not doing three way calling |
| 11:49.25 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 11:50.18 | *** join/#asterisk MrPloddy29` (n=ploddy@87-194-62-72.bethere.co.uk) |
| 11:51.03 | *** join/#asterisk s0lid (n=jlq@210.213.240.236) |
| 11:51.33 | figu78 | no |
| 11:51.42 | figu78 | but i use the same procedure |
| 11:52.07 | figu78 | as Voip-info recommends |
| 11:52.14 | *** join/#asterisk version5 (i=version5@nat/ibm/x-1c06a60c6b5776a2) |
| 11:52.20 | mosty | but what does that have to do with the three way calling bug? |
| 11:53.03 | figu78 | to transfer the incoming call in this way, you need to setup the threewaycalling=yes in zapata.cof |
| 11:53.55 | mosty | i haven't had to do that for attended transfers |
| 11:54.43 | figu78 | <PROTECTED> |
| 11:54.44 | figu78 | <PROTECTED> |
| 11:54.44 | figu78 | <PROTECTED> |
| 11:54.44 | figu78 | <PROTECTED> |
| 11:54.44 | figu78 | <PROTECTED> |
| 11:55.18 | *** join/#asterisk MrPloddy29` (n=ploddy@87-194-62-72.bethere.co.uk) |
| 11:56.16 | *** join/#asterisk GaryH (n=chatzill@host217-37-44-41.in-addr.btopenworld.com) |
| 11:57.29 | figu78 | mosty, is what you do??? |
| 11:57.42 | mosty | i use the features.conf method |
| 11:59.59 | figu78 | using the callparking, isn't it? |
| 12:00.35 | mosty | no |
| 12:00.38 | mosty | atxfer |
| 12:01.13 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:01.51 | figu78 | ok |
| 12:03.05 | basty | after updating spandsp 0.0.2 to 0.0.4 - do I need to restart asterisk, or does it work suddenly after the compile ? |
| 12:03.24 | mosty | restart |
| 12:03.43 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:04.32 | basty | mosty: okay..what about the rxfax.so ? On the softswitch page there is no rxfax.so/txfax.so incl. the makefile to download |
| 12:05.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:05.59 | mosty | no idea, i don't do fax with asterisk |
| 12:06.37 | basty | okay..thanks anyway. |
| 12:07.03 | figu78 | mosty, it seems nothing happend, when I press *2 i didn't get the dial tone |
| 12:07.20 | figu78 | and the same happends if i push the R button |
| 12:08.08 | mosty | do "show features" in the console, does it say *2 is atxfer? |
| 12:08.31 | MrWup | anyone had echo problems with b410p? |
| 12:08.45 | figu78 | yes |
| 12:08.55 | figu78 | in the current colum |
| 12:11.53 | figu78 | (yesterday the R button works, but for some reason now is not working)maybe is related |
| 12:12.06 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.40 | Faithful | Hey whats a good software phone? |
| 12:14.02 | [TK]D-Fender | Faithful: eyeBeam |
| 12:16.58 | *** join/#asterisk VJFROMGT (n=vjfromgt@190.80.51.236) |
| 12:25.24 | *** join/#asterisk BouYYY (n=BouYYY@81-86-77-70.dsl.pipex.com) |
| 12:25.42 | BouYYY | Hi I am running 1.4.6, on a call between to iax peers trying to put on end on hold, seems to have an issue between IAX -> IAX. The call does not go on hold, and I get a unlink and link event in the AMI. Using SIP -> IAX causes no problems when inintiating the hold from the IAX channel. Any idea's anyone? |
| 12:26.06 | Stormfr | anyone know a bit of the new dsp card TC400B ? seems asterisk current or 1.4 don't include all code for support it (asterisk cli command don't find in source code) |
| 12:26.12 | version5 | hey guys, if i have two peripheral asterisk servers( A and C) and one central (B) one. Is there a way to set them up so that i can write a call file on B to initiate a call between a sip phone registered on A and another on C |
| 12:26.48 | mosty | Stormfr, do you have a recent zaptel? |
| 12:27.56 | Stormfr | lastest |
| 12:28.27 | Stormfr | anyway the problem is in asterisk, show trancoder doesn't exist |
| 12:28.52 | Stormfr | card is correctly detected too |
| 12:29.16 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:29.17 | mosty | asterisk 1.2.19 plus the current 1.2 zaptel worked fine for me once i loaded the module |
| 12:29.36 | Stormfr | you have show transcoder command ? |
| 12:30.10 | Stormfr | maybe they plan to put this but not yet commit the source (it's in the manual btw) |
| 12:30.55 | mosty | yes, but only if the module is loaded |
| 12:31.02 | Stormfr | i haven't |
| 12:33.13 | Stormfr | Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) so i think it's load correctly and get zapg729 something detected in asterisk |
| 12:34.38 | mosty | does show translation show g729 conversion times? |
| 12:38.51 | nicox | hello, which distribution you are using for 65.000 minutes a day with asterisk and a Digium 4 Port E1 Card? |
| 12:41.19 | *** join/#asterisk ToyMan (n=Stuart@74-32-24-175.dsl1.mdl.ny.frontiernet.net) |
| 12:42.33 | mosty | whatever you're most familiar with |
| 12:43.31 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-230.net.novis.pt) |
| 12:43.46 | MrMister2 | Hi everyone |
| 12:43.58 | MrMister2 | I'm having issues with SIP on Asterisk 1.4.6 |
| 12:44.29 | *** join/#asterisk VJFROMGT (n=vjfromgt@190.80.51.236) |
| 12:44.32 | MrMister2 | How can I recompile that part of Asterisk? Or do I have to the full package? |
| 12:45.24 | anonymouz666 | www.digium.com does not open |
| 12:46.17 | Err | anonymouz666: that would be a problem on your end, since it works for me :-) |
| 12:46.20 | MrMister2 | anonymouz666: It's opening here |
| 12:46.41 | Stormfr | mosty, show transcoder is not show translation |
| 12:46.49 | MrMister2 | anonymouz666: Try to open a command line prompt and do "tracert www.digium.com" and see there it gets stuck |
| 12:46.51 | Uatec | what was the name of that massive nazi battleship in WWII, that was sunk before it even headed out of port for the first time, and there's that horrible footage of the thing rolling over with hundres of people trying to run over the hull to stay on top? |
| 12:46.56 | Stormfr | show transcoder should display number or call process on the dsp |
| 12:47.53 | mosty | Stormfr, but show translation will show you if you can transcode to g729 |
| 12:48.20 | Stormfr | yes but it's not the same usage at all |
| 12:48.40 | Stormfr | show transcoder is for manage the card see if you reach limit, see if call correctly handle by the dsp |
| 12:49.16 | mosty | what about show g729? |
| 12:51.01 | Stormfr | i haven't this command |
| 12:51.44 | *** join/#asterisk shinao1 (n=shinao1@41.205.185.69) |
| 12:52.35 | mosty | show transcoder works on my box, but as i said it's asterisk 1.2.19 |
| 12:53.33 | *** join/#asterisk jm|laptop (n=jm|home@sentry.flags.co.uk) |
| 12:56.37 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 12:57.12 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:57.57 | Stormfr | mosty, ok 1.4 don't have the code for this command, 1.2.20 have it |
| 12:58.14 | Stormfr | seems 1.2 is more update for driver ;) will try to add it to 1.4 thanks for info |
| 12:58.51 | JT | quite an expensive card considering how much support it has |
| 12:59.22 | VJFROMGT | is there a way to set verbose on a single peer only? |
| 13:01.14 | *** join/#asterisk Op3r (n=op3r@125.212.11.166) |
| 13:02.04 | [TK]D-Fender | VJFROMGT: No. |
| 13:05.11 | *** join/#asterisk E-bola (i=snowman@cpe-76-179-4-233.maine.res.rr.com) |
| 13:05.26 | E-bola | Can seombody help me and try to udnerstand the sip show channels command better? |
| 13:05.40 | E-bola | it seems to be cluttered with mmm unknown channels which i cant quite figure out what is |
| 13:05.44 | Holos | If I want a user to press one to accept, two to re-enter a number, without breaking out of the "s" extension, I should use a Read and a Goto right? any other options? |
| 13:05.46 | E-bola | its definately not conversations going on |
| 13:06.01 | E-bola | last message for all fo them is: ACK |
| 13:06.10 | E-bola | form is unknown |
| 13:06.44 | VJFROMGT | does anyone know the most common issue for trunks to drop calls after 16 to 20 secs? |
| 13:06.49 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:07.14 | JT | trunks being what? |
| 13:07.21 | JT | and no, that does not sound like a common issue |
| 13:07.48 | VJFROMGT | jt |
| 13:07.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.58 | VJFROMGT | calls just drops after 20 secs |
| 13:08.14 | mocker | VJFROMGT: Do you get in errors in the * console? |
| 13:08.18 | JT | i asked you what the trunks were, not to repeat what has already been said :) |
| 13:08.41 | VJFROMGT | i have so many cals at the same time i am trying to find a way to isolate the message, any hints? |
| 13:08.52 | VJFROMGT | regular verbose shows for all calls |
| 13:08.58 | JT | what technology? |
| 13:09.04 | VJFROMGT | sip to gsm |
| 13:09.15 | *** part/#asterisk figu78 (n=figu78@26.Red-80-26-83.staticIP.rima-tde.net) |
| 13:09.40 | JT | gsm is not a channel driver in asterisk as far as i'm aware |
| 13:09.54 | VJFROMGT | i am using it as a trunk |
| 13:10.17 | JT | please be less vague |
| 13:10.20 | JT | and more specific |
| 13:10.31 | VJFROMGT | just a sec,, let me paste configs |
| 13:10.56 | VJFROMGT | http://pastebin.ca/614234 |
| 13:11.13 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:11.19 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 13:12.21 | [TK]D-Fender | Holos: Why wouldn't you run it as a standard IVR? |
| 13:12.55 | mocker | i need a web app to start tracking DIDs. ;) |
| 13:13.30 | Holos | [TK]D-Fender: It's inside a macro for agents to log in, just wanted to keep it simple, I know I can use an IVR, but the thought of having a 1 and 2 extension in the macro seems wrong to me... Not sure why... |
| 13:13.51 | JT | VJFROMGT: i'm still failing to see what that snippet of sip.conf HAS TO DO WITH GSM, could you tell me? |
| 13:14.15 | mocker | Holos: What about having them dial something like *60, then starting your IVR? |
| 13:14.21 | [TK]D-Fender | Holos: If its a simple test you could jsut use READ. |
| 13:14.22 | mocker | Instead of just them picking up the phone and dialing 1 |
| 13:14.49 | [TK]D-Fender | Holos: it is the best way if you do need to do it from a macro, because running off "s" ends your macro. |
| 13:14.59 | Holos | [TK]D-Fender: Thats what I'll be doing, a Read and GotoIf |
| 13:15.20 | [TK]D-Fender | Holos: Then thats the way to do it. |
| 13:15.31 | Holos | [TK]D-Fender: Thanks.. |
| 13:15.47 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 13:18.23 | naitram | looking at asterisk channel.h, struct ast_channel_tech has send_digit_begin(chan*...) and send_digit_end(chan*...,duration), anyone know which to use to send a dtmf tone on a channel, or both? |
| 13:18.34 | VJFROMGT | the call is failing between asterisk and the sip trunk (which is really a sip to gsm module) |
| 13:21.02 | VJFROMGT | JT,, here you go http://pastebin.ca/614249 |
| 13:21.12 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 13:21.55 | JT | VJFROMGT: so what you mean to say is "i am using a SIP to GSM gateway"? |
| 13:21.55 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:23.02 | [TK]D-Fender | VJFROMGT: You provided only PART of ONE side's sip.conf config, and eventually USELESS cli output with no SIP debug enabled. Are you sure you're actually looking for HELP!? |
| 13:23.23 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 13:23.32 | [TK]D-Fender | VJFROMGT: And making getting a proper FULL description of everything involved remarkably painful. |
| 13:23.39 | docelmo | Anyone know anything about zaptel having issues with dropped calls? |
| 13:24.06 | *** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 13:24.12 | [TK]D-Fender | docelmo: No, Zaptel has no trouble dropping calls at all... |
| 13:24.19 | [TK]D-Fender | *THUD* |
| 13:24.24 | [TK]D-Fender | :D |
| 13:24.40 | VJFROMGT | sorry TK,, i will get more details |
| 13:24.52 | JT | yesm try typing them |
| 13:24.55 | docelmo | No seriously.. :) I have a customer that is complaining of dropped calls and they are using zaptel. Someone said awhile back that Zaptel had some issues where it would drop the calls |
| 13:25.08 | JT | "i am doing this, and that, using this method" |
| 13:25.15 | JT | not random obscure references to gsm |
| 13:25.40 | *** join/#asterisk Ameritech (n=Ameritec@mail.accinc.info) |
| 13:26.17 | Ameritech | Anyone know how I publish extension presence to a remote IAX server |
| 13:26.35 | [TK]D-Fender | docelmo: What card, what modes, pastebin a config....... |
| 13:27.04 | [TK]D-Fender | Ameritech: You can't |
| 13:27.09 | VJFROMGT | TK,, is this what is needed http://pastebin.ca/614257 |
| 13:27.14 | [TK]D-Fender | Ameritech: * does not "span" for presence |
| 13:27.21 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:27.39 | JT | wow, pretty patient |
| 13:27.43 | mocker | ugh. |
| 13:27.50 | mocker | queue call just went to voicemail. |
| 13:27.52 | mocker | That's not good. |
| 13:27.56 | mocker | ;) |
| 13:27.58 | JT | i'm thinking +m with 5minutes for +v would be optimal here |
| 13:28.02 | *** join/#asterisk Ameritech (n=Ameritec@mail.accinc.info) |
| 13:28.11 | JT | if you can't wait the 5min, you can't be helped |
| 13:28.49 | Ameritech | (GOT BOOTED)... Anybody know how to publish extension presence to a remote IAX server? |
| 13:29.09 | JT | <PROTECTED> |
| 13:29.44 | Ameritech | sorry....to be honest, i clicked the close button...i know its not the same...but it was still an accidental, unexpected loss of connection |
| 13:29.54 | JT | hehe |
| 13:30.00 | [TK]D-Fender | set_destination: set destination to 192.168.20.33, port 12 <- whats with port 12? |
| 13:30.19 | [TK]D-Fender | VJFROMGT: looks whacked. Also the mix of public VS private IP's |
| 13:30.42 | [TK]D-Fender | Ameritech: You can't |
| 13:30.44 | [TK]D-Fender | Ameritech: * does not "span" for presence |
| 13:30.45 | Uatec | in my extensions, can i access the information of another call? |
| 13:30.55 | [TK]D-Fender | Uatec: like? |
| 13:31.06 | Uatec | like the oad and the dad of another call |
| 13:31.14 | [TK]D-Fender | Uatec: Typically only through AGI |
| 13:31.25 | Uatec | what i want to do is frig the caller ID of a user when they are transfering a call |
| 13:31.39 | Uatec | for example, if i transfered a call to you it would come up with "From: Uatec" |
| 13:31.40 | [TK]D-Fender | VJFROMGT: And at least now we know its your GSM gateway that is killing the call. |
| 13:31.48 | Uatec | but it's not really from me, it's from "Bob" |
| 13:32.01 | tzafrir | hmmm... if I do any calcualtion in MATH, must I use SPRINTF to get rid of the excess digits? |
| 13:32.06 | Uatec | i want to frig it so that it says "Uatec-Xfer-Bob" or something... |
| 13:32.16 | [TK]D-Fender | Uatec: Wrong approach. Stop doing Attended transfers when what you want to do is a BLIND TRANSFER |
| 13:32.33 | tzafrir | ${MATH(1+2)} ? |
| 13:32.47 | Uatec | [TK]D-Fender, i know that a blind transfer does it |
| 13:32.53 | Uatec | but i want to do it with an attended transfer too |
| 13:33.06 | tzafrir | any simpler alternative? |
| 13:33.25 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 13:33.27 | [TK]D-Fender | Uatec: And you can't miraculously make the CID seen on screenr ight after a transfer. Once the call is PLACED, its TFB |
| 13:33.49 | [TK]D-Fender | Uatec: You can't. Thats not the way its designed. |
| 13:34.05 | [TK]D-Fender | tzafrir : $[1+2] |
| 13:34.28 | [TK]D-Fender | tzafrir : Need some "Bat"? ;) |
| 13:34.34 | mocker | Answer() is a tricky beast. |
| 13:36.45 | Uatec | i know [TK]D-Fender, but i want it to say who the call that is being transfered it from... |
| 13:37.13 | [TK]D-Fender | Uatec: TFB, these are LAWS of physics, not SUGGESTIONS. |
| 13:37.28 | [TK]D-Fender | Uatec: they are designed to be a certain way extremely explicitly. |
| 13:37.32 | Uatec | i don't want to change the caller ID mid call |
| 13:37.38 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-62d93b75f7ccbf1b) |
| 13:37.57 | [TK]D-Fender | Uatec: An attended transfer shows yours on purpose for extremely good reasons. |
| 13:38.09 | Uatec | i know, but i want to show both... |
| 13:38.14 | [TK]D-Fender | Uatec: You cannot change this without massive recoding to *. |
| 13:38.17 | [TK]D-Fender | ~wglwat |
| 13:38.17 | jbot | hmm... wglwat is well, good luck with all that |
| 13:38.59 | Uatec | so there is no other way of accessing the data of another call which is currently in progress? |
| 13:39.27 | mosty | agi/ami |
| 13:39.50 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:41.20 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:41.59 | *** join/#asterisk Capps- (n=andrew@67-67-242-2.ded.swbell.net) |
| 13:43.17 | mocker | Uatec: Maybe you should just have people park calls. |
| 13:43.21 | mocker | And not use transfer at all. |
| 13:44.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:44.49 | Uatec | oh, now that soundsa bit complicated |
| 13:44.56 | Uatec | my users aren't the most phone enlightened bunch |
| 13:45.29 | [TK]D-Fender | Uatec: If they can't handle a blind transfer, then SHOOT THEM. There is noting remotely as easy. |
| 13:45.32 | Strom_M | that's why you teach them |
| 13:45.48 | mocker | So I think I'm going to have to setup a seperate group of extensions that are 'Agent' phones for call queues. |
| 13:46.08 | mocker | Because my std. extensions have an explicit Answer() which does not play nicely w/ queues. |
| 13:46.13 | Uatec | they can handle a blind transfer, but don't always want to do so. |
| 13:46.39 | Uatec | awww, poor mocker, you have an STD |
| 13:46.55 | mocker | Uatec: At least my users can blind transfer. |
| 13:46.56 | [TK]D-Fender | Uatec: oh now you want these "laws" of physics to not only CHANGE, but only apply SOMETIMES? |
| 13:46.58 | mocker | :) |
| 13:47.22 | [TK]D-Fender | Uatec: "Put down the crack pipe!" (c) JerJer |
| 13:51.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:51.18 | *** join/#asterisk Peri (n=redanti@xtreme-14-56.dyn.aci.on.ca) |
| 13:52.41 | Uatec | my users can blind transfer |
| 13:52.44 | Uatec | they just choose not to |
| 13:52.54 | Uatec | retards |
| 13:53.22 | *** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 13:53.29 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 13:53.55 | [TK]D-Fender | Uatec: Indeed they are. The first step is admitting you have a problem..... |
| 13:54.28 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:54.32 | [TK]D-Fender | Uatec: And they will continue to follow whatever new scheme you try to devise with equal commitment no doubt. |
| 13:54.57 | Fl1p | it seems that since version 1.4.7 Musiconhold cant be played saying "Unable to set channel SIP/... to format Unknown" using the default fpm*.wavs. Any idea anyone ? |
| 13:55.15 | mosty | Fl1p, submit a bug report |
| 13:55.25 | file | upgrade to 1.4.7.1 |
| 13:55.26 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 13:55.29 | Fl1p | maybe it's not a bug |
| 13:55.36 | Fl1p | file : is it a known bug ? |
| 13:55.44 | [TK]D-Fender | f|1p : UPGRADE |
| 13:55.52 | Fl1p | allowed Codecs are : ulaw,alaw,gsm,g726,g729 |
| 13:57.08 | Fl1p | ok thx for the help |
| 13:59.17 | NOT_guru | hello |
| 13:59.50 | *** join/#asterisk jpmcallister (n=ec06113@kapla.escelsa.com.br) |
| 13:59.54 | NOT_guru | so when you have 2 extensions, and you want a line to appear on both, Line 1 , Line 2 on both handsets |
| 13:59.58 | NOT_guru | is that possible |
| 14:00.34 | mocker | I wish there were reserved XXXX numbers that were never allocated by LECs to be used for internal extensions. |
| 14:00.38 | NOT_guru | so guy at extension 1 can see line one is in use and/or if need be pick up line one while guy at extension 2 is on line 1 as well |
| 14:00.50 | mocker | Sorta like how there is IP space that can never be used... |
| 14:01.02 | NOT_guru | you mean like 555 |
| 14:01.11 | mocker | NOT_guru: Exactly.. |
| 14:01.18 | mocker | Except the last 4 digits. ;) |
| 14:01.23 | NOT_guru | ahso |
| 14:01.25 | NOT_guru | LOL |
| 14:01.31 | mocker | Too bad it'd burn so many numbers out of the pool. |
| 14:02.40 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 14:02.52 | [TK]D-Fender | NOT_guru: that made NO sense. |
| 14:02.59 | NOT_guru | sorry I am a bit distracted patching a windows box on hold with cingular |
| 14:03.50 | [TK]D-Fender | NOT_guru: Actually... you basically want to let users share a line in a hop-on, hop-off way? |
| 14:03.59 | NOT_guru | yes |
| 14:04.09 | NOT_guru | sorry I was trying to think of another way to phrase it |
| 14:04.12 | NOT_guru | and thats the way |
| 14:04.14 | NOT_guru | LOL |
| 14:04.15 | Peri | anyone encountered an extensionstate of 1 when it should be 0? |
| 14:04.18 | [TK]D-Fender | NOT_guru: You appear to be the ONLY person who needs 1.4's new "SLA" <- |
| 14:04.20 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:04.21 | [TK]D-Fender | :D |
| 14:04.22 | NOT_guru | thank you fender insightfull as always |
| 14:04.30 | tzafrir | Note: a default 'make install' of zaptel 1.4 will now install zaptel perl stuff |
| 14:04.37 | tzafrir | enjoy... |
| 14:04.45 | *** join/#asterisk salvatore2 (n=cn@teknopet.com) |
| 14:04.49 | JT | zaptel...perl... ? |
| 14:04.54 | [TK]D-Fender | NOT_guru: There's your answer, and it does pretty much exactly what you asked. |
| 14:04.55 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:04.58 | NOT_guru | surely you jest... I can't be only one |
| 14:05.31 | salvatore2 | hello everybody |
| 14:05.43 | salvatore2 | i am new to asterisk and i want to ask you one thing |
| 14:05.44 | [TK]D-Fender | NOT_guru: Pretty much, YES. Its not "real " SLA, and is nearly useless in its current implementation (IMO). Your limited needs are the only thing its good for. |
| 14:05.49 | salvatore2 | what is jitter buffer? |
| 14:05.56 | tzafrir | JT: ./xpp/utils/zaptel_hardware |
| 14:05.58 | NOT_guru | another questionLOL |
| 14:06.02 | NOT_guru | at least I am easy |
| 14:06.07 | tzafrir | and more |
| 14:06.08 | JT | tzafrir: what does perl do for zaptel? |
| 14:06.23 | [TK]D-Fender | salvatore2: I do nt's k n ow, y ou' r e b reak ing u p ! ! |
| 14:06.29 | NOT_guru | thanks again fender I will have to take a peak at 1.4 ( its for my mechanic ) |
| 14:07.00 | tzafrir | JT, just a bunch of small perl modules. Because writing in perl is much nicer than writing in C (userpsace code) |
| 14:07.03 | *** join/#asterisk Corydon76-home (i=yellow@pdpc/supporter/sustaining/Corydon76-home) |
| 14:07.04 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:07.39 | salvatore2 | [TK]D-Fender, sorry? |
| 14:07.40 | JT | i don't imagine perl would be too nice for very time critical stuff |
| 14:07.57 | tzafrir | JT, all those modules do is expose information already available. They don't use ioctls |
| 14:07.58 | Mercestes | [TK]D-Fender, What?? I can't understand you! You need a jitterbuffer to cache your data and error correct it so it comes out correctly! |
| 14:08.22 | salvatore2 | :), what kind of education is this here ? :) |
| 14:08.23 | [TK]D-Fender | Mercestes: z0mg, everything sounds so much cleaner now! |
| 14:08.23 | jm|laptop | # A-jitterbug |
| 14:08.25 | tzafrir | They are used for control (setup, startup and such) |
| 14:08.36 | [TK]D-Fender | salvatore2: load chan_sarcasm.so <- |
| 14:08.37 | Mercestes | [TK]D-Fender, Ah, much better. |
| 14:08.40 | JT | tzafrir: hmm ok |
| 14:08.41 | salvatore2 | :) |
| 14:08.42 | mocker | lol |
| 14:08.44 | tzafrir | And also work well with miniperl and such |
| 14:08.59 | JT | i want a COBOL interface to zaptel |
| 14:09.15 | [TK]D-Fender | salvatore2: This was a dramatic reenactment or a call before, and AFTER applying a jitter buffer |
| 14:09.16 | mocker | I should have asked Mark if official fax support will ever make it into mainline asterisk last Friday. |
| 14:09.24 | salvatore2 | i actually knew this answer from my google search but what I want to know is how it works, I mean is it my server jittering the rtp packages or the sender server that does that |
| 14:09.46 | [TK]D-Fender | salvatore2: odds are, NEITHER. Its all the crap in BETWEEN |
| 14:09.56 | tzafrir | If you like perl, have a look at the stuff under xpp/utils/zconf |
| 14:10.27 | tzafrir | hmmm, people who like both perl and Zaptel..... ... |
| 14:10.40 | salvatore2 | hmm confused, so let's say when i ping my provider, ping results vary between 20 and 24ms, so if a package comes before the other one, the sound becomes crappy, and jitter buffer solves this right |
| 14:10.56 | [TK]D-Fender | salvatore2: It may. |
| 14:11.08 | [TK]D-Fender | salvatore2: Depending on how bad things get. |
| 14:11.09 | JT | s/package/packet/ |
| 14:11.48 | salvatore2 | how can i enable it on sip conversations? |
| 14:12.02 | salvatore2 | is it me enabling it for receiving side? |
| 14:12.05 | [TK]D-Fender | salvatore2: its documented in the sample sip.conf. Go read. |
| 14:12.19 | salvatore2 | yes but what i don't understand is how can i activate for receiving channel |
| 14:12.29 | [TK]D-Fender | salvatore2: same thing. |
| 14:12.33 | NOT_guru | I thought " load chan_sarcasm.so" is alpha code my system is a production box |
| 14:12.36 | [TK]D-Fender | salvatore2: Go. Read. |
| 14:12.40 | NOT_guru | sorry late timing on phone |
| 14:12.44 | salvatore2 | jbenable=yes |
| 14:13.25 | JT | NOT_guru: if you patch it against mISDN it will be very stable |
| 14:13.50 | [TK]D-Fender | JT : There is nothing more stable than a "dead stop" :D |
| 14:14.33 | *** join/#asterisk ksng (n=chatzill@62.147.219.15) |
| 14:14.42 | ksng | hello all :) |
| 14:15.00 | NOT_guru | LOL |
| 14:15.08 | *** join/#asterisk Strom_C (i=strom@nat/digium/x-b419d22443cc7125) |
| 14:15.27 | ksng | i find documentation of asterisk svn version for DEVSTATE, someone have something? ;) |
| 14:15.56 | [TK]D-Fender | ksng: huh? |
| 14:17.24 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:17.36 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 14:17.43 | mocker | Ahh! Using Progress() will still get me callerid name information in place of Answer() |
| 14:17.46 | mocker | ! |
| 14:17.48 | ksng | i'm looking for documentation of core function of Asterisk (SVN version SVN-trunk-r71577) |
| 14:18.24 | [TK]D-Fender | ksng: You need to be more specific. |
| 14:18.45 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 14:20.10 | nicox | anybody an idea why this happened?: Got reject for frame 82, retransmitting frame 82 now, updating n_r! |
| 14:21.17 | ksng | How in the Asterisk CLI or in The manager interface, i can get the value of variables like DEVSTATE for a SIP channel. |
| 14:21.32 | *** join/#asterisk riddlebox (n=JamesMid@75-128-170-26.static.stls.mo.charter.com) |
| 14:22.14 | [TK]D-Fender | ksng: probably listed under HINTS like it is meant to emulate. |
| 14:22.27 | ksng | Ok |
| 14:23.22 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:23.33 | ksng | thanks |
| 14:24.28 | MrWup | anyone had echo problems with b410p? |
| 14:24.54 | MrWup | ive enabled hardware echo cancelling with dsp_hwechocan=1 and still i get terrible echo when calling some people |
| 14:27.05 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-180-39.bflony.east.verizon.net) |
| 14:31.07 | Polis_ttt | why do i get: Jul 11 16:23:14 NOTICE[23341]: chan_sip.c:11173 handle_request_subscribe: Got SUBSCRIBE for extension Unknown@default from 192.168.0.105, but there is no hint for that extension |
| 14:31.15 | Polis_ttt | all other lines in asterisk is working, even this one, but this computer wont work, it never gets properly connected to asterisk, and astersisk cli prints this message. The workstation is using windows xp and xlite, worked before, but now it's ot working :( |
| 14:31.23 | Polis_ttt | the computer connect to server via internet, 192.168.0.105 si the computers lan-ip. |
| 14:32.21 | [TK]D-Fender | Polis_ttt: The error is telling you to your face that a phone is trying to get Presence for "Unknkown" and that is not an exten with a hint in [default] |
| 14:32.29 | [TK]D-Fender | Polis_ttt: How much clearer can you be?! |
| 14:33.02 | Mercestes | <PROTECTED> |
| 14:33.13 | Mercestes | And I meant every word of it. |
| 14:33.18 | [TK]D-Fender | Mercestes: Yes... you are completely transparent. |
| 14:33.48 | *** part/#asterisk mtaht4 (n=m@12.158.129.130) |
| 14:34.05 | MrWup | in misdn-init.conf it says unknown variable: dsp_hwechocan |
| 14:34.06 | MrWup | <PROTECTED> |
| 14:34.22 | MrWup | but in all the helpfiles it says dsp_hwechocan=1 should be set to enabkle echo cancelling |
| 14:34.25 | MrWup | i dont know whats up |
| 14:35.09 | Strom_C | MrWup: perhaps you should just call digium support |
| 14:35.21 | Strom_C | you /do/ have install support for that card included in the price |
| 14:35.33 | MrWup | erm |
| 14:35.35 | MrWup | i have no idea |
| 14:35.52 | MrWup | oh u mean |
| 14:35.54 | MrWup | i do |
| 14:35.59 | MrWup | thought that was a question |
| 14:36.01 | Polis_ttt | [TK]D-Fender: but sometimes is says 66052@default that is the extension/number for this phone |
| 14:36.04 | MrWup | how do they even know i bought the card though? |
| 14:36.09 | Mercestes | MrWup: He even escaped it for you to be implicit. |
| 14:36.10 | MrWup | do i have to give em my card's serial number? |
| 14:36.29 | Mercestes | MrWup: feeling guilty? |
| 14:36.33 | MrWup | no im not |
| 14:36.38 | Mercestes | uh huh |
| 14:36.41 | MrWup | im just curious how they know i paid for it |
| 14:36.46 | [TK]D-Fender | Polis_ttt: You are not getting it. someone got a call from "unknown" caller saved it to his stupid addressbook and tried putting a WATCH ON IT. |
| 14:36.46 | MrWup | cause ill have to take apart my server to find out |
| 14:37.13 | Mercestes | They have a GPS chip on it that activates upon power being applied to it that updates via sat-link to the Digium home office giving them the exact location of every card they have ever manufactured with serial number information and customer information. |
| 14:37.30 | Strom_C | yeah, you should see the GPS room - it's like "wargames" |
| 14:37.42 | JT | well |
| 14:37.52 | JT | let's think about this another way |
| 14:38.19 | JT | MrWup: why would you want tech support on a card you DO NOT HAVE? :) |
| 14:38.55 | MrWup | maybe im a tech support pirate |
| 14:38.57 | MrWup | =] |
| 14:39.11 | Dr-Linux | which one is best and friendly asterisk billing opensource program? |
| 14:39.16 | Dr-Linux | need suggestion |
| 14:39.24 | [TK]D-Fender | Dr-Linux: www.drphil.com |
| 14:39.33 | MrWup | ah hell |
| 14:39.39 | MrWup | this is going to be an international call |
| 14:39.49 | Dr-Linux | ? |
| 14:40.04 | mosty | MrWup, call the distributor in your country |
| 14:42.15 | mosty | Dr-Linux, look up asterisk billing on voip-info.org |
| 14:43.15 | Dr-Linux | mosty: yeah i went through some of ... but not sure which one is optimal |
| 14:43.22 | Dr-Linux | maybe AstBill ? |
| 14:43.39 | mosty | Dr-Linux, it's subjective. try a few, pick the one you like |
| 14:44.25 | Dr-Linux | mosty: for that maybe i'd get a testing server first to check them out.. |
| 14:44.40 | mosty | you should be doing that anyway |
| 14:45.12 | *** join/#asterisk _fbs_ (n=felipe@190.8.81.97) |
| 14:45.54 | Dr-Linux | mosty: thanks |
| 14:48.22 | MrWup | well have tried emailing support@digium.com |
| 14:48.24 | MrWup | hope it helps |
| 14:48.42 | MrWup | hope i dont get an answer like "is the card plugged into the pci slot?" |
| 14:48.57 | [TK]D-Fender | MrWup: go phone them. |
| 14:49.16 | tzanger | well |
| 14:49.24 | tzanger | I'm gonna try running kubuntu with a xen kernel |
| 14:49.29 | tzanger | so I can run winxp in a domu |
| 14:49.38 | tzanger | and then throw up mini systems for testing and whatnot |
| 14:49.42 | rob0 | Digium isn't an international call! It's x500 in the demo! |
| 14:49.47 | mosty | digium's emails support can take a long time to get back to you |
| 14:50.20 | [TK]D-Fender | rob0: I was going to save that for when he would start whining about it ;) |
| 14:50.26 | rob0 | oops :) |
| 14:50.27 | Dr-Linux | i got new cards from digium. Very nice |
| 14:52.37 | MrWup | [TK]D-Fender, cant find a UK person i phone |
| 14:52.40 | MrWup | still looking though |
| 14:53.29 | MrWup | im hoping this echo cancel problem is well known |
| 14:53.36 | MrWup | i saw someone post the very same problem on a forum |
| 14:53.42 | MrWup | but they went into private message mode |
| 14:57.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:59.55 | *** join/#asterisk cullenincrease (n=cp@c-75-64-44-200.hsd1.tn.comcast.net) |
| 14:59.56 | cullenincrease | hey |
| 15:00.50 | cullenincrease | could anyone take a look at my sip.conf and extensions.conf and see if there are any obvious errors? my asterisk system is acting really strange for some reason. http://pastebin.com/m65c50aee |
| 15:01.16 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 15:01.20 | cullenincrease | specifically i keep having to restart my phones for some reason |
| 15:01.24 | cullenincrease | for them to work |
| 15:02.16 | [TK]D-Fender | cullenincrease: I take it they are behind NAT on the remote side? |
| 15:03.02 | cullenincrease | yeah just the phones |
| 15:03.11 | cullenincrease | the asterisk server isn't though |
| 15:03.27 | [TK]D-Fender | cullenincrease: they need "nat=yes", "canreinvite=no", and "qualify=yes" for NAT keep-alives |
| 15:03.35 | [TK]D-Fender | cullenincrease: (the phones that is) |
| 15:03.36 | cullenincrease | hmm ok |
| 15:03.44 | cullenincrease | i had everything except for the qualify=yes |
| 15:04.21 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 15:04.25 | ZaVoid | hey all |
| 15:04.43 | ZaVoid | anyone got any decent exp with agi and php work? |
| 15:04.49 | ZaVoid | i'm having a problem and couldn't find a good answer on voipwiki |
| 15:05.19 | ZaVoid | trying to set agi_accountcode with a value from my DB.. and it say it inserts it.. then when i query on it, it fails |
| 15:07.58 | [TK]D-Fender | cullenincrease: Qualify sends the packets that keep the NAT RTP path open otherwise the phone will cease to be contactable for incoming calls. |
| 15:08.15 | [TK]D-Fender | ZaVoid: PASTEBIN |
| 15:08.20 | De_Mon | Aarr nobodys fixed up 8446 :( |
| 15:08.26 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 15:08.26 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:08.38 | [TK]D-Fender | cullenincrease: "qualify = yes" does not belong in extensions.conf |
| 15:08.44 | ZaVoid | ${ACCOUNTCODE} or somthing i guess |
| 15:08.46 | ZaVoid | ok i'll do a pastebin |
| 15:08.52 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 15:08.58 | De_Mon | im surpprised more people havent monitored the whisper for meetme issue |
| 15:09.44 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:10.04 | cullenincrease | awesome! |
| 15:10.04 | FreezeS | hey guys |
| 15:10.04 | FreezeS | I was just curious about the new features that will be in 1.6 |
| 15:10.08 | FreezeS | as I know, there are 3 important (for me) changes: asynchronous call control, redundancy and different CDR system |
| 15:10.09 | *** join/#asterisk Lewis999 (n=chatzill@static-70-108-241-182.res.east.verizon.net) |
| 15:10.10 | cullenincrease | added qualify=yes and all seems well now |
| 15:10.13 | cullenincrease | you rock man |
| 15:10.15 | FreezeS | can anyone give me more details ? |
| 15:10.42 | Lewis999 | Hello, Can anyone help me with AGI programming? |
| 15:10.53 | De_Mon | FreezeS do you have any specific questions? |
| 15:10.54 | *** join/#asterisk windex (n=amp@host-69-95-137-4.mil.choiceone.net) |
| 15:11.13 | [TK]D-Fender | cullenincrease: Quite welcome |
| 15:11.32 | Fl1p | whats the new Meetme called in 1.4.7.1 ? |
| 15:11.39 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 15:11.44 | Fl1p | says cant find application Meetme |
| 15:11.57 | [TK]D-Fender | f|ip : hasn't changed. |
| 15:11.59 | windex | hey guys, i have done everything i can think of to fix: "Zap/3-1: ring timeout reached with no data or events?!?", anyone have any handy hints where i'm missing something? it can dial out fine, it just wont answer. |
| 15:12.00 | Fl1p | ? |
| 15:12.09 | [TK]D-Fender | f|ip : if you don't have it then you did not install Zaptel properly |
| 15:12.34 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 15:12.45 | [TK]D-Fender | Lewis999: .... |
| 15:12.46 | [TK]D-Fender | ~ask |
| 15:12.47 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:12.57 | Fl1p | D-Fender : yes possible installed a clean asterisk on a second server, can i compile zaptel now or do i have to use order Zaptel -> asterisk ? |
| 15:13.01 | JT | ~question |
| 15:13.02 | jbot | well, question is 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 |
| 15:13.20 | FreezeS | De_Mon: I'm especially curious about the call control |
| 15:13.33 | FreezeS | how easy will it be to create on the fly conferences |
| 15:13.35 | Lewis999 | When I use AGI->exec('Dial','SIP/buster/0014109888888') I did get any information in the asterisk console. |
| 15:13.45 | [TK]D-Fender | FreezeS: this is clearly a question for #asterisk-dev <- |
| 15:13.49 | Lewis999 | I installed asterisk-perl-0.008 |
| 15:13.54 | FreezeS | aha |
| 15:13.59 | FreezeS | thanks |
| 15:14.24 | [TK]D-Fender | Lewis999: pastebin the whole mess, the CLI output of your call at verbose 10 and debug enabled. |
| 15:14.35 | rob0 | FWIW once I came --->][<--- this close to giving ESR his own "Smart Questions" link. He was asking not-so-smart questions in #postfix. |
| 15:15.12 | De_Mon | FreezeS according to the stuff I've read it'll be as easy as bridging channels togeather "instantly" being a quotable qord in there |
| 15:15.24 | Lewis999 | I set debug leve 10 and I don't get any message in CLI window. |
| 15:15.34 | FreezeS | de_mon, that is what I read aswell |
| 15:15.42 | FreezeS | but I'm curious how exactly is it planned |
| 15:15.47 | FreezeS | from an agi script ? |
| 15:15.52 | [TK]D-Fender | Lewis999: and "verbose 10". pastebin yrou dialplan, your scripts, EVERYTHING. |
| 15:15.53 | FreezeS | or from the management interface |
| 15:15.55 | [TK]D-Fender | ~pb |
| 15:15.56 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 15:15.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 15:16.09 | De_Mon | ahhh, now your asking good questions |
| 15:16.14 | De_Mon | take em to -dev ;) |
| 15:16.32 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 15:18.50 | Lewis999 | Ok. I changed verbose from 21 to 10 now. I only have sip account set up for outbound call. There is one section in sip.conf [buster] for my voip account information. |
| 15:19.16 | Holos | Are there any known issues with using a Goto(xxxx) in a context where extension xxxx is only available through an include? |
| 15:19.45 | Corydon76-work | Holos: No |
| 15:20.03 | [TK]D-Fender | Lewis999: Stop providing useless commentary about what yout hink is going on and show us the CODE & OUTPUT. |
| 15:20.41 | [TK]D-Fender | Holos: So pastebin your dialplan and we'll show you where you went wrong. |
| 15:20.54 | Holos | [TK]D-Fender: ok will do.. |
| 15:20.58 | Holos | quit |
| 15:22.49 | Lewis999 | In extensions.conf I have one ; outbound call |
| 15:22.51 | Lewis999 | exten => _9NXXNXXXXXX,1,Dial,SIP/buster/001${EXTEN:1} |
| 15:22.52 | Lewis999 | exten => _9NXXNXXXXXX,2,Congestion |
| 15:22.54 | Lewis999 | test.pl |
| 15:22.55 | Lewis999 | use Asterisk::AGI; |
| 15:22.56 | Lewis999 | $AGI=new Asterisk::AGI; |
| 15:22.58 | Lewis999 | $AGI->exec('Dial','SIP/buster/0014109888888'); |
| 15:23.12 | [TK]D-Fender | Lewis999: PASTEBIN!! |
| 15:23.14 | [TK]D-Fender | ~pb |
| 15:23.15 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 15:23.18 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:23.34 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:23.55 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-aa4e0113b792df95) |
| 15:24.00 | Lewis999 | sorry guys |
| 15:24.11 | *** join/#asterisk ber111 (i=brad@neu.cow.org) |
| 15:24.25 | ber111 | how do i configure a backup SIP route for an extension |
| 15:24.34 | Holos | [TK]D-Fender: Here is the information: http://pastebin.ca/614429 |
| 15:24.41 | ber111 | like try dialing XXX@SIP_PROVIDER_A and if failure XXX@SIP_PROVIDER_B |
| 15:24.46 | neverblue | ~thebook |
| 15:24.46 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:25.20 | [TK]D-Fender | Holos: you have forgotten Goto's PARANTER ORDER |
| 15:25.28 | [TK]D-Fender | Holos: you have forgotten Goto's PARAMETER ORDER |
| 15:25.39 | *** join/#asterisk ecoleman (n=eric@24.75.47.130) |
| 15:25.42 | [TK]D-Fender | Holos: that is looing for that PRIORITY in the CURRENT EXTEN |
| 15:25.42 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-191-128.inter.net.il) |
| 15:25.51 | *** join/#asterisk KaczX (n=KaczX@gate.ieurop.net) |
| 15:25.58 | [TK]D-Fender | Holos: "show application goto" |
| 15:26.17 | [TK]D-Fender | ber111: Just dial them back-to-back |
| 15:26.26 | Holos | [TK]D-Fender: ah, I need Goto(from_iax,${EXTEN:3}) |
| 15:26.37 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:26.42 | [TK]D-Fender | Holos: getting WARMER.... |
| 15:26.56 | ecoleman | is it possible to use AMD and dial together and get a proper response? I'm tring to detect if the party im connecting to is actually answering, rather than it's voicemail |
| 15:27.08 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 15:27.28 | Holos | [TK]D-Fender: so.. maybe Goto(from_iax,${EXTEN:3},1) |
| 15:27.45 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 15:28.15 | *** join/#asterisk Dj_FlyBy (n=abc@mail.imonkeyit.com) |
| 15:28.43 | [TK]D-Fender | Holos: MUCH closer....... |
| 15:29.12 | Holos | [TK]D-Fender: We're not jumping contexts in this case, so I only need (${EXTEN:3},1) |
| 15:30.05 | Lewis999 | [TK]D-Fender:I pate it to channels.debian.net/paste |
| 15:30.07 | [TK]D-Fender | Holos: Actually.... that WOULD work. |
| 15:30.16 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:30.34 | [TK]D-Fender | Holos: but do you really need to INCLUDE that other context? looks like it SHOULD be 100% seperate and your goto EXPLICIT |
| 15:31.02 | *** part/#asterisk GaryH (n=chatzill@host217-37-44-41.in-addr.btopenworld.com) |
| 15:31.07 | [TK]D-Fender | Lewis999: LINK PLEASE. I have no intention of going on a search for it.... |
| 15:31.27 | Holos | [TK]D-Fender: So you're saying leave the from_iax in or not? |
| 15:32.07 | ber111 | TK thanks |
| 15:32.14 | [TK]D-Fender | Holos: http://pastebin.ca/614437 |
| 15:32.28 | [TK]D-Fender | ber111: np |
| 15:32.44 | ber111 | if the first one connects correctly will that cause an issue if i dial after it |
| 15:33.46 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 15:33.57 | ifnotwhynot | hi TK |
| 15:34.47 | ifnotwhynot | did a instaltion of suse and asterisk 1.4 whith dare i say terrible results |
| 15:35.51 | Strom_C | this iphone artivle is a load of sensationalist garbage; half of these policies have been true for mobile phone networks since the beginning of time |
| 15:36.06 | Strom_C | this "telecom analyst" is incompetent :) |
| 15:36.07 | Qwell[] | Strom_C: link? |
| 15:36.27 | ecoleman | Strom_C: you talking about the 'Fine Print' article? |
| 15:36.36 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 15:37.12 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 15:37.35 | Strom_C | yes |
| 15:37.45 | Strom_C | but i meant to type that into a different IRC channel |
| 15:38.01 | coppice | Strom_C: sounds like its pretty well written. it gots its page click revenue from you :-) |
| 15:38.52 | Strom_C | no, it was linked by someone to me |
| 15:39.58 | [TK]D-Fender | ber111: No, typically after an answered call, * hangs up the channel unless you used the "g" option. |
| 15:40.58 | [TK]D-Fender | ecoleman: AMD? huh? |
| 15:41.45 | ifnotwhynot | anyone know where can get the lastest release of the asterisk book with all new dialplan applications for asterisk v1.4? |
| 15:41.47 | [TK]D-Fender | ifnotwhynot: SUSE is well known to make any attempts at compiling kernel modules friggen hell. Take heed. |
| 15:42.05 | ecoleman | [TK]D-Fender: im using dial to connect someone to another party, but I want to determine if that party is actually answering, or if it's an answering machine/voice mail picking up |
| 15:42.08 | [TK]D-Fender | ifnotwhynot: Its about to be released. Ask in a month |
| 15:42.35 | [TK]D-Fender | ecoleman: AH.... I've heard of some detection code out there, but I don't recall it being part of * itself. |
| 15:42.39 | alrs | Did you guys see that Digium bought Astricon? |
| 15:42.45 | ifnotwhynot | do you recommend i use fedora 7 [TK]D-Fender? |
| 15:42.56 | ecoleman | [TK]D-Fender: there's AMD() which does it, but i can't seem to get it to work properly calling it from a macro |
| 15:42.58 | [TK]D-Fender | ifnotwhynot: I recommend either CentOS or Slackware. |
| 15:43.27 | coppice | no AMD scheme ever works properly |
| 15:43.35 | [TK]D-Fender | ecoleman: pastebin what you're doing so we can see if there's a way to rearrange it to make it work. |
| 15:43.51 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:43.56 | Lewis999 | [TK]D-Fender: Here is my paste http://pastebin.ca/614453 |
| 15:44.43 | file | alrs: Sokol & Associates, which produces Astricon |
| 15:44.47 | [TK]D-Fender | Lewis999: Ok, so whats the problem? |
| 15:45.13 | Lewis999 | When I run the test program there is no output in the CLI console. |
| 15:45.51 | Lewis999 | I can make phone call using soft phone. It means asterisk is running fine. |
| 15:45.56 | [TK]D-Fender | Lewis999: Where are you trying to run it from? It isn't being called ANYWHERE in your dialplan. |
| 15:46.29 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:47.35 | ecoleman | [TK]D-Fender: actually, it's working |
| 15:47.54 | ecoleman | the message on my voicemail was too short |
| 15:47.56 | ecoleman | :'( |
| 15:48.54 | Lewis999 | I run the program on the same Linux box. |
| 15:49.01 | yonahw | is there something wrong with Dial("Zap/22-1", "SIP/7001 & SIP/7002") in new stack? |
| 15:49.34 | [TK]D-Fender | ecoleman: I'm resolved to make a message that says "Hello? ........... Hi.............. yeah hold on a sec (chatter) ................. What was that? .........................Oh yeah sorry forgot to tell you I'm not really here right now so... :D" |
| 15:49.43 | tzanger | hahaha |
| 15:49.48 | yonahw | sorry never mind that, ill be back in a bit |
| 15:49.48 | ecoleman | heh |
| 15:49.49 | tzafrir | yonahw, extra spaces? |
| 15:50.06 | yonahw | tzafrir: I dont see any |
| 15:50.06 | [TK]D-Fender | Lewis999: Calling WHERE? AGI doesn't just HAPPEN you know. |
| 15:50.45 | yonahw | tzafrir: do you see any? |
| 15:50.45 | [TK]D-Fender | Lewis999: You can't call that from anywhere EXCEPT the dialplan. |
| 15:50.45 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:50.46 | tzanger | yeah the dialplan's goofy combination of spacing critical and spacing-agnostic is... frustrating |
| 15:50.46 | tzafrir | SIP/7001* *&* *SIP/7002 |
| 15:50.46 | irule | what do you recommend to convert Mixmonitor() recordings to text transcript? |
| 15:50.56 | Qwell[] | irule: Humans |
| 15:50.57 | yonahw | tzafrir: thanks i did not realize those did not belong there |
| 15:50.58 | Lewis999 | So I can't run a perl file and call AGI function? |
| 15:51.06 | yonahw | appreciate that one |
| 15:51.12 | irule | Qwell[] how about software? lol |
| 15:51.49 | [TK]D-Fender | Lewis999: NO. You completely misunderstand AGI. |
| 15:52.38 | Lewis999 | My mistake. I am new to asterisk. So what interface I can use to make call throgh asterisk? I couldn't find a good auto dialer so I want to write one myself. |
| 15:52.39 | [TK]D-Fender | Lewis999: AGI is means to give you programming control over the * call that EXECUTES it to control dialplan flow. |
| 15:53.03 | tzafrir | Lewis999, make a call from where exactly? There are a number of methods |
| 15:53.07 | [TK]D-Fender | Lewis999: Go read THE BOOK. You are taking blind random stabs in the name of development. |
| 15:53.10 | [TK]D-Fender | ~book |
| 15:53.10 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:53.26 | [TK]D-Fender | tzafrir : He's looking for call files or AMI originate. |
| 15:53.38 | [TK]D-Fender | tzafrir : All cluelessly. |
| 15:53.38 | Lewis999 | I read that book already. |
| 15:53.45 | [TK]D-Fender | Lewis999: read it again. |
| 15:54.14 | [TK]D-Fender | Lewis999: AGI isn't used to TRIGGER calls, its use in the PROCESSING of them. |
| 15:54.26 | [TK]D-Fender | ~osmosis |
| 15:54.27 | jbot | osmosis is probably the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:54.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:55.00 | tzanger | [TK]D-Fender: I wrote up a dialplan+cron script that calls a set of numbers every 3 minutes and if a human answers, passes them to my cell phone |
| 15:55.11 | tzanger | a friend was having some trouble getting paid for osme work :-) |
| 15:55.43 | Lewis999 | I see. Can you give me a clue which interface I can use to make phone call from a standalone program written with C or perl? |
| 15:55.58 | Mercestes | <PROTECTED> |
| 15:56.17 | tzanger | it calls 7am-7pm mon-fri and you can toggle it by dialing **BUG :-) |
| 15:56.30 | Mercestes | hehe |
| 15:56.59 | Lewis999 | tzanger, can you share your code with us? It sounds like what I need. |
| 15:58.07 | [TK]D-Fender | Lewis999: You haven't been listening to everything I've been saying.... |
| 15:59.00 | Lewis999 | :) I am not talking about AGI now. |
| 15:59.09 | tzanger | [TK]D-Fender: I don't listen to anything you're saying, either. :-) |
| 15:59.26 | tzanger | when I get my laptop back I can pastebin it |
| 15:59.30 | [TK]D-Fender | Lewis999: And you STILL aren't PAYING ATTENTION. |
| 15:59.34 | *** join/#asterisk perf3kt (i=perf3kt@149.166.33.238) |
| 15:59.36 | Uatec | Hey, does anybody know where I can get some free hold music? |
| 15:59.49 | Uatec | just in the interim till we find some proper stuff... |
| 15:59.58 | [TK]D-Fender | Uatec: Comes with * already |
| 16:00.21 | Mercestes | Uatec, Limewire. |
| 16:00.25 | Strom_C | Uatec: use the Record() application with a comb and a piece of toilet paper |
| 16:00.30 | Uatec | umm |
| 16:00.36 | Uatec | Mercestes, i was meaning legal |
| 16:00.50 | perf3kt | tk: if I deleted mine, can I reload it somehow |
| 16:00.51 | Uatec | Strom_C, music? |
| 16:00.57 | Strom_C | it was a joke |
| 16:00.59 | Lewis999 | tzanger, Can you send a copy to standerby@yahoo.com? Thanks. |
| 16:01.10 | Uatec | [TK]D-Fender, how do i turn it on then? cos at the moment all I get it silence. |
| 16:01.17 | Qwell[] | Strom_C: no M? |
| 16:01.29 | Strom_C | Qwell: oh |
| 16:01.30 | Uatec | oh, there's a musiconhold.conf |
| 16:01.31 | Uatec | DOI |
| 16:01.33 | Qwell[] | heh |
| 16:01.33 | Strom_M | therw! |
| 16:01.42 | [TK]D-Fender | Uatec: Verify your MoH configs and that you have the decoder for the format your MoH is in. |
| 16:02.19 | *** join/#asterisk CVirus (n=GoD@196.218.186.175) |
| 16:02.25 | mocker | sigh, why must all softphones be ugly? |
| 16:02.49 | Mercestes | mocker: genetics |
| 16:03.08 | Mercestes | They are all made in West Virginia |
| 16:04.01 | *** part/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru) |
| 16:04.22 | NOT_guru | cisco softphone is perty |
| 16:04.35 | NOT_guru | but its sccp only I believe |
| 16:06.46 | yonahw | tzafrir: the problem was indeed those extra spaces which leads me to point out the wiki is misleading |
| 16:07.05 | yonahw | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial shows they syntax with spaces |
| 16:07.08 | denon | hmm .. you know, it could be fun to write a network driver, that's essentially a shim, that acts as an sccp to iax2 proxy |
| 16:07.10 | mocker | yonahw: If it's a wiki, fix it! |
| 16:07.11 | mocker | :) |
| 16:07.14 | tzafrir | yonahw, which leads me to ask you to fix the wiki |
| 16:07.17 | Qwell[] | mocker: design me a softphone |
| 16:07.24 | irule | how much is the stand alone lumenvox? I cont find any price on the website :s |
| 16:07.45 | Strom_M | Qwell: you have to fedex him some crayons |
| 16:07.48 | Qwell[] | irule: The lumenvox stuff is not going to be able to give you a transcript of text |
| 16:07.54 | denon | it could terminate sccp and sip, and push iax2 out the other end |
| 16:08.04 | irule | Qwell[] why? |
| 16:08.05 | mocker | Qwell[]: Actaully, I might have found one that isn't horrible. |
| 16:08.07 | Qwell[] | Strom_M: I've been thinking about writing a softphone, and everybody always says "don't make it ugly" |
| 16:08.10 | denon | wildly complex, but could be pretty cool |
| 16:08.10 | mocker | Actually even. |
| 16:08.12 | Qwell[] | irule: Because it can't |
| 16:08.13 | yonahw | tzafrir: I intend to I am reading the posting guidlines now |
| 16:08.14 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 16:08.24 | irule | Qwell[] what can it do? |
| 16:08.28 | Qwell[] | speech recognition |
| 16:08.41 | tzafrir | yonahw, login, fix, and be done with it |
| 16:08.42 | Qwell[] | simple detection of grammars |
| 16:08.55 | Qwell[] | "grammars"? |
| 16:08.59 | Qwell[] | oh, Strom_M |
| 16:09.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:09.03 | Qwell[] | I need you to school me |
| 16:09.07 | tzafrir | yonahw, however, just don't forget to write a small log message describing what you fixed |
| 16:09.09 | coppice | granmas |
| 16:09.16 | mocker | Qwell[]: My main thing is I don't like it to not look like a windows app. |
| 16:09.20 | irule | Qwell[] you mean sppech recognition is way our of their league? |
| 16:09.29 | Qwell[] | irule: no, speech recognition is what they do |
| 16:09.31 | mocker | I know it's a softphone, please don't try to make it skinned like a real phone. :) |
| 16:09.37 | Qwell[] | speech recognition IS NOT the same as text-to-speech |
| 16:09.49 | Habbie | it's speech to text ;) |
| 16:09.52 | Qwell[] | mocker: yeah... |
| 16:09.57 | Qwell[] | erm, right |
| 16:10.00 | irule | I am looking for speech-to-text |
| 16:10.08 | Qwell[] | ~wglwat |
| 16:10.09 | jbot | from memory, wglwat is well, good luck with all that |
| 16:10.35 | Strom_M | why do i keep wanting to pronounce that "wiggle-what"? |
| 16:10.43 | Qwell[] | Strom_M: {a,e}ffect |
| 16:10.44 | Qwell[] | discuss |
| 16:10.49 | *** part/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 16:10.50 | Strom_M | discus? |
| 16:10.58 | Qwell[] | school me :p |
| 16:11.01 | Strom_M | one is a noun, one is a verb |
| 16:11.02 | yonahw | tzafrir: in the comments? |
| 16:11.05 | Strom_M | affect == verb |
| 16:11.08 | Strom_M | effect == noun |
| 16:11.21 | Uatec | [TK]D-Fender, well i have done SetMusicOnHold(), but that looks like it only sets it for hold done by asterisk, what about the simple hold button on my phone, i fear that the hold button on my phone might not actually tell asterisk it's onh old |
| 16:11.22 | tzafrir | no. Edit the text itself. You need to be logged in for that |
| 16:11.28 | Qwell[] | can effect be verbified |
| 16:11.30 | Qwell[] | ? |
| 16:11.35 | Strom_M | generally no |
| 16:11.40 | yonahw | tzafrir: I did edit the text, I am reffering to the log |
| 16:11.42 | Uatec | oh no, it does |
| 16:11.46 | Qwell[] | what else? |
| 16:11.55 | Uatec | it says "Stopped music on hold", that means that it was trying to play music on hold, just never did |
| 16:12.08 | Strom_M | ? |
| 16:12.15 | Qwell[] | that's all you've got? :p |
| 16:12.19 | tzafrir | there's an empty line to fill when you post a change |
| 16:12.26 | Qwell[] | I guess that's all there is, heh |
| 16:12.27 | Strom_M | Qwell: well that's the cliff notes version |
| 16:12.47 | Strom_M | but that's generally all you need to remember |
| 16:13.13 | lilalinux | how can I make "sip debug" ignore one peer? |
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| 16:13.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:13.35 | yonahw | tzafrir: for some reason I am not understanding you, would you mind pointing out an example of what you mean to me? |
| 16:14.33 | tzafrir | yonahw, I'm talking about voip-info.org |
| 16:14.45 | tzafrir | when you edit a page |
| 16:16.02 | yonahw | tzafrir: i understand that you are talking about an extra line commenting my edit of a page on voip-info.org, I just dont understand where one would put this comment since I don't really see any on the site |
| 16:17.49 | mightnare | guys... if i set transfer=no in zapata.conf, will i still be able to atxfer and blindxfer from features.conf ? |
| 16:17.51 | tzafrir | This is the "Comment" field in the edit screen. Now press the "History" link of a page |
| 16:18.03 | tzafrir | mightnare, yes |
| 16:18.18 | mightnare | tzafrir: via dtmf right? |
| 16:18.29 | tzafrir | right |
| 16:18.40 | mightnare | i see... thanks |
| 16:18.49 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 16:19.21 | yonahw | tzafrir: Thanks I finally got it, sorry for my idiocy |
| 16:19.32 | mightnare | also, does the flash button on some analog phones generate dtmf? or is it just the same as pressing the plunger? |
| 16:19.40 | tzafrir | Sadly most users don't use comments |
| 16:20.27 | Uatec | Class: default |
| 16:20.27 | Uatec | aste*CLIMode: quietmp3 |
| 16:20.27 | Uatec | aste*CLIDirectory: /var/lib/asterisk/mohmp3 |
| 16:20.27 | Uatec | aste*CLIFormat: slin |
| 16:20.31 | Uatec | oops, sorry |
| 16:20.35 | Uatec | well, now it's there. |
| 16:20.43 | Uatec | I am getting that, when i do "moh classes show" |
| 16:20.52 | coppice | mightmare: flash normally acquires a DTMF decoder, so it wouldn't be much good generating DTMF |
| 16:21.23 | Uatec | but when I put someone on hold it says "-- Started music on hold, class 'default', on channel 'mISDN/4-1'" |
| 16:21.25 | Uatec | but i hear nothing |
| 16:21.30 | Uatec | Why could this be? |
| 16:21.33 | Uatec | what am i doing wrong? |
| 16:21.37 | mightnare | coppice: so i'm better off with using for example *2 which is the default on features.conf? |
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| 16:21.42 | yonahw | tzafrir: for some reason it was not obvious to me that the comment box i was looking at was a comment box for the eid |
| 16:22.13 | yonahw | s/eid/edit/ |
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| 16:24.40 | tzanger | http://www.pastebin.ca/614507 <-- that's the bug the guy script |
| 16:25.15 | mightnare | tzafrir: the reason why i have to disable three-way calling (which in turn affects transfers) is that i think the astribank doesn't handle it well |
| 16:26.10 | mightnare | sometimes i see this with "show channels": |
| 16:26.33 | mightnare | Zap/1-1 (None) Up Bridged Call(Zap/3-1) |
| 16:26.33 | mightnare | Zap/3-1 s@macro-trunkdial:1 Up Dial(Zap/g1/414xxxx) |
| 16:27.36 | [TK]D-Fender | *b00m* |
| 16:27.36 | mightnare | awww |
| 16:27.36 | tzanger | heh my bugging script caused a netsplit :-) |
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| 16:27.51 | [TK]D-Fender | Uatec: bad or missing MPG123 |
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| 16:28.45 | Uatec | [TK]D-Fender, i just installed mpg123 with conary, i'll see if i can check if it's working |
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| 16:30.35 | enjay5150 | anyone know if Polycom phones can work with a Mitel 3300 |
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| 16:31.04 | Mercestes | enjay5150, given that Polycom phones work, and work well, and Mitel does not, I'm going to say .....no. |
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| 16:32.25 | enjay5150 | Mitel does not? dont get me wrong Im no fan, but Its implemented at several 1000+ user customers I have, Im pretty sure it works. Im just trying to determine if the Mitel will accept other SIP phones.. |
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| 16:32.56 | Juggie | enjay5150, they might w/ v6 of the software |
| 16:32.57 | Mercestes | enjay5150, I'd guess no, but, try it and see. |
| 16:33.02 | mightnare | tzafrir: the reason why i have to disable three-way calling (which in turn affects transfers) is that i think the astribank doesn't handle it well |
| 16:33.06 | mightnare | Zap/1-1 (None) Up Bridged Call(Zap/3-1) |
| 16:33.07 | Juggie | as i recall that was supposed to enable sip |
| 16:33.08 | mightnare | Zap/3-1 s@macro-trunkdial:1 Up Dial(Zap/g1/414xxxx) |
| 16:33.11 | Juggie | but i am not 100% sure |
| 16:33.23 | Juggie | it might of just been sip trunks and not sip phones |
| 16:33.31 | enjay5150 | right on.. I'll check into that thanks! |
| 16:33.55 | Juggie | their phones use minet |
| 16:34.02 | irule | can sometone in plain english explain to ma what exactly is the dictate() application? http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dictate makes me think that it is a single caller leaving a sound file in asterisk, and thats it? :s |
| 16:34.05 | tzafrir | mightnare, what do you mean by "have to"? |
| 16:34.18 | Juggie | enjay5150, btw, their newer dualmode phones have great sip firmwares. |
| 16:34.39 | tzafrir | mightnare, what happens if you don't disable it? |
| 16:35.12 | tzafrir | mightnare, this is an FXS module of an Astribank, right? |
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| 16:35.50 | mightnare | tzafrir: FXO and FXS astribank modules |
| 16:35.57 | RedComet- | Has anyone had issues with voip providersand 403 forbidden? |
| 16:36.24 | Uatec | nope, mpg123 is definately there |
| 16:36.39 | enjay5150 | Juggie; Cool thanks, I want to see if the IP4000 conference phones work with the mitel.. |
| 16:36.45 | rob0 | Probably most folks who don't register with their provider get a 403. |
| 16:36.53 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 16:36.55 | tzafrir | mightnare, again, what bad things happen if you don't disable threewaycalling? |
| 16:37.10 | RedComet- | According to asterisk I am registered, but when I make an outbound call I get 403 |
| 16:37.14 | RedComet- | In bound works great |
| 16:37.26 | tzafrir | or any other zapataspeakverylongoptionname |
| 16:39.06 | mightnare | sometimes i see these in "show channels" (both are fxo ports): Zap/1-1 (None) Up Bridged Call(Zap/3-1) |
| 16:39.12 | mightnare | Zap/3-1 s@macro-trunkdial:1 Up Dial(Zap/g1/414xxxx) |
| 16:39.16 | *** join/#asterisk sysmonk (n=nobody@unaffiliated/sysmonk) |
| 16:39.24 | *** join/#asterisk shinao1 (n=shinao1@41.205.185.41) |
| 16:39.32 | *** join/#asterisk Zaphod80013 (n=chatzill@74.93.236.61) |
| 16:39.38 | [TK]D-Fender | Uatec: Go verify that your MoH files are there, in the right format, and the version of MPG123 is correct. |
| 16:41.19 | Zaphod80013 | Can anybody answer a Day 0 newbie question, what is the difference between the 1.2.x & 1.4.x versions? |
| 16:41.40 | Mercestes | Zaphod80013, -2. |
| 16:42.09 | Habbie | Mercestes, -.2. ;) |
| 16:42.16 | *** join/#asterisk hypa7ia (n=hypatia@judecca.aculei.net) |
| 16:42.18 | Mercestes | Habbie, Precisely! |
| 16:42.40 | Zaphod80013 | ok i was looking for something a little more incitefull than -2 but its a start! |
| 16:42.50 | Mercestes | Zaphod80013, Read the changelog. |
| 16:43.20 | Habbie | Zaphod80013, i don't know actually :) |
| 16:44.53 | Zaphod80013 | Fair enough! the change log looked pretty big & i'm also new to VoIP so I'm not sure I'll appreciate the subtlety's but thanks |
| 16:45.04 | tzafrir | mightnare, the problem is that chan_zap's hook time is way over permissive |
| 16:45.06 | Uatec | <PROTECTED> |
| 16:45.11 | Uatec | mpg123 can decode them |
| 16:45.18 | Uatec | and i don't get any errors when i put people on hold |
| 16:45.21 | Uatec | i just hear nothing |
| 16:45.23 | [TK]D-Fender | Uatec: pastebin all the proof of the point I have mentioned. |
| 16:46.14 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 16:46.27 | [TK]D-Fender | Zaphod80013: There are a lot of articles out there to show the key differences. Go check the WIKI for old news, and get Googleing. |
| 16:46.53 | Zaphod80013 | ok will do thanks |
| 16:46.54 | tzafrir | mightnare, by default chan_zap has rxflash set to 1250ms |
| 16:47.02 | tzafrir | Which is rather insane |
| 16:47.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:47.29 | tzafrir | IIRC the spacs say that a flash should be 100ms - 700ms |
| 16:47.49 | tzafrir | in practice we use locally 300ms |
| 16:47.55 | mightnare | tzafrir: what's the recommended value for astribank interfaces? |
| 16:48.16 | tzafrir | mightnare, it depends on the phones used |
| 16:48.30 | tzafrir | (or on the people flashing...) |
| 16:48.55 | tzafrir | Not really something different in the Astribank |
| 16:49.23 | tzafrir | (that I'm aware of) |
| 16:51.33 | mightnare | tzafrir: i'd be better off using dtmf-mode transfers then i guess... |
| 16:51.55 | mightnare | just to be safe |
| 16:51.59 | tzafrir | mightnare, just set the flash times to a sane value |
| 16:54.22 | mightnare | tzafrir: ok, i'll give it a try for one day and see what happens... thanks |
| 16:55.49 | lilalinux | there should be a note in the topic: make sure you have a working reverse lookup for SIP |
| 16:57.58 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:58.56 | *** join/#asterisk ifnotwhynot (n=davidh@c1-52-11.rrba.isadsl.co.za) |
| 16:58.59 | rbd | I have an AGI script that runs off of an extension. Calls come in from an H323 channel....sometimes, the asterisk doesn't return the call and gives a fast busy instead. I am seeing messages like: Jul 10 20:37:51 VERBOSE[6835] logger.c: -- H.323 call 'ip$10.6.200.21:62238/498-7cb8e66f' cleared, reason 8 (Transport failure), established (2093 sec) |
| 17:00.22 | rbd | and Jul 10 20:45:01 VERBOSE[6835] logger.c: -- H.323 call 'ip$10.6.200.21:63624/553-7cb8e6c3' cleared, reason 4 (Cleared by remote user [47 - Resource unavailable, unspecified]), established (55 sec) |
| 17:00.28 | rbd | looks like a bunch of h323-related problems |
| 17:01.50 | ifnotwhynot | hi there i have a softphone that connects to a free tel server i have a username and password how can i set this up to use asterisk to dial out on this account? |
| 17:01.54 | mocker | I'm suprised by how well queuemetrics seems to work.l |
| 17:02.21 | [TK]D-Fender | ifnotwhynot: probably just about the same way as you setup any other ITSP |
| 17:03.01 | ifnotwhynot | tried all the readme's on the net doesn't seem to work |
| 17:04.56 | tzafrir | ifnotwhynot, telling us the name of that service might be a good start |
| 17:05.08 | ifnotwhynot | telfree |
| 17:05.13 | ifnotwhynot | in south africa |
| 17:05.18 | irule | does the directory application wtill work in 1.2.20? I noticed in sounds.txt that there are dir-*.gsm files related to some application that will dial by name by pressing the first 3 digits of a name, whats up with that? |
| 17:05.26 | tzafrir | next: search for telfree asterisk |
| 17:05.35 | tzafrir | anything interesting comes up? |
| 17:05.49 | ifnotwhynot | the guy from tell free send me a how to that does not make any sence |
| 17:06.11 | ifnotwhynot | where do i paste again if someone may wanna have a look |
| 17:07.15 | ifnotwhynot | http://pastebin.com/m155e3289 |
| 17:07.58 | [TK]D-Fender | irule: "show application directory". Yes it works. It wasn't created for nothing. Get off your ass are READ. |
| 17:08.08 | ifnotwhynot | if you set up connection like this do you need to use iax or sip.conf? |
| 17:08.48 | [TK]D-Fender | ifnotwhynot: might be nice to see the failed call attempts with SIP debug you know. We're not PSYCHIC |
| 17:09.46 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 17:10.20 | irule | [TK]D-Fender I dont think its necesary to stand up from my desk to read lol |
| 17:11.00 | [TK]D-Fender | irule: Considering you've got your head up your ass, it WILL. let the blood CIRCULATE! |
| 17:11.25 | coolfreecode | i want to do 50 calls simultanious what hardware i need? |
| 17:12.00 | irule | [TK]D-Fender just because your head is up your ass, does not mean you should asume that everyone elses heads are in the same positiojn :) |
| 17:12.12 | [TK]D-Fender | coolfreecode: a computer and some medium that can support 50 calls. ethernet would do nicely. |
| 17:14.57 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 17:15.49 | ifnotwhynot | would i need to set up iax or sip any help welcome? |
| 17:18.03 | [TK]D-Fender | ifnotwhynot: Depends what yourprovider is using.... |
| 17:19.15 | coolfreecode | i have a pri |
| 17:19.20 | ifnotwhynot | they are using sip |
| 17:19.35 | [TK]D-Fender | ifnotwhynot: well I guess you don't need IAX now do you? |
| 17:19.44 | coolfreecode | exactly 2 pri |
| 17:19.44 | ifnotwhynot | you think |
| 17:20.02 | [TK]D-Fender | coolfreecode: other than a PRI, what else is going to connect to * and how? |
| 17:20.16 | [TK]D-Fender | ifnotwhynot: I do, quite well in fact. |
| 17:20.35 | coolfreecode | well i have a call center |
| 17:20.47 | coolfreecode | but with panasonic |
| 17:20.54 | coolfreecode | so i want to use asterisk |
| 17:21.25 | coolfreecode | and i want to buy a server HP Proliant ML150 G3 |
| 17:21.28 | [TK]D-Fender | coolfreecode: background is nice, but please focus on the question asked. |
| 17:21.32 | *** join/#asterisk neuwald (n=neuwald@200.199.198.61) |
| 17:21.43 | coolfreecode | sorry for my english |
| 17:21.51 | neuwald | Hi folks |
| 17:21.57 | coolfreecode | i speak spanish =) |
| 17:22.21 | neuwald | I'm having some problems with queues. I have one queue called q-suporte, that have this 'entities' that aren't specified in queues.conf: |
| 17:22.22 | neuwald | q-suporte has 1 calls (max unlimited) in 'rrmemory' strategy (41s holdtime), W:0, C:33, A:14, SL:6.1% within 0s |
| 17:22.22 | neuwald | <PROTECTED> |
| 17:22.22 | neuwald | <PROTECTED> |
| 17:22.23 | neuwald | <PROTECTED> |
| 17:22.24 | ifnotwhynot | how would i frase a question in mr google to get the rights results in linking asterisk to server using sip? |
| 17:22.36 | neuwald | a queue is a member of other queue (!?) |
| 17:22.58 | *** join/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 17:23.13 | rob0 | ~book |
| 17:23.13 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:23.17 | trevarthan | Hello. How do I get asterisk to ignore a busy tone and continue ringing until the line picks up? |
| 17:23.23 | rob0 | ifnotwhynot: ^^ |
| 17:23.56 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.133) |
| 17:24.28 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.133) |
| 17:25.40 | [TK]D-Fender | neuwald: You dynamically added that, most likely in the dialplan due to coding error. |
| 17:26.52 | neuwald | [TK]D-Fender ok, but now I already restarted asterisk and it still there. |
| 17:26.59 | neuwald | how to remove it? |
| 17:27.15 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 17:28.31 | [TK]D-Fender | neuwald: go look at removequeuemember |
| 17:29.32 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-209-191.dhcp.insightbb.com) |
| 17:29.52 | etfonhomey | Any of the Aussie people around? |
| 17:30.00 | neuwald | [TK]D-Fender thanks |
| 17:30.04 | neuwald | :-) |
| 17:30.21 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 17:34.12 | trevarthan | Is there any way to make Dial() ignore busy tones and continue dialing? |
| 17:34.16 | coolfreecode | i want to buy a server for call center with 50 calls simultaneously and 2 PRI TE205P(E1), what hardware e do you use? |
| 17:35.25 | [TK]D-Fender | trevarthan: Why would you do that? |
| 17:35.50 | [TK]D-Fender | coolfreecode: I've asked you THREE TIMES NOW. What ELSE are you planning on haveing connect to you * server, and HOW? |
| 17:36.23 | coolfreecode | Only two lines PRI(E1) |
| 17:36.32 | trevarthan | [TK]D-Fender: Because my stupid Intertel phone system's call queue is brain dead and generates a busy tone before dropping into the queue. |
| 17:36.51 | MrWup | whats the diff between TE and NT mode isdn? could it be the cause of my echo? |
| 17:36.56 | [TK]D-Fender | trevarthan: ditch your call progress in zapata |
| 17:37.03 | [TK]D-Fender | MrWup: No. |
| 17:37.24 | [TK]D-Fender | MrWup: one means act like the TELCO, the other says act like a PHONE |
| 17:37.36 | MrWup | oh |
| 17:37.42 | trevarthan | [TK]D-Fender: ok. what sort of nasty side effects can I expect when doing that? |
| 17:37.54 | MrWup | well my asterisk PBX is using the digium b410p in TE mode |
| 17:37.59 | MrWup | so i assume thats all how it should be |
| 17:38.18 | ai-a | Hi, anyone here get Sangoma ISDN working in the UK ? or have any tips ? CLI> pri show span 1 shows Status: Provisioned, Down, Active |
| 17:38.39 | [TK]D-Fender | trevarthan: well... you won't know a "real" busy when you get one..... |
| 17:38.55 | [TK]D-Fender | trevarthan: try, suffer, adapt, therapy. |
| 17:39.23 | trevarthan | :) will it just bridge the audio that way? Will the user hear the busy tone at least? |
| 17:40.00 | trevarthan | or will there basically just be two states: ringing, answered? |
| 17:41.39 | [TK]D-Fender | trevarthan: Not even. |
| 17:41.46 | [TK]D-Fender | trevarthan: it wil be "answered" |
| 17:42.04 | tzafrir | ai-a, are there any drivers for it published? |
| 17:42.31 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 17:42.32 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:42.39 | ai-a | tzafrir: yep,, followed their install guide... installed new zap / pri / wanpipe.. all running,, getting alarms on the system.. |
| 17:42.54 | ai-a | Jul 9 21:28:47 asterisk kernel: wanpipe1: RAI alarm is OFF Jul 9 21:28:48 asterisk kernel: wanpipe1: RAI alarm is ON Jul 9 21:28:48 asterisk kernel: wanpipe1: RAI alarm is OFF Jul 9 21:28:49 asterisk kernel: wanpipe1: RAI alarm is ON |
| 17:43.04 | *** join/#asterisk gardo (n=gardo@121.97.197.207) |
| 17:43.15 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 17:43.15 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 17:43.23 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 17:44.06 | etfonhomey | [TK]D-Fender, how's it going? Haven't been on here in a while. Messed with the new Polycom 330's yet? |
| 17:44.44 | [TK]D-Fender | etfonhomey: not personally, but I've provisioning them for a cliient |
| 17:45.15 | etfonhomey | [TK]D-Fender, any complaints about them from your client? |
| 17:45.58 | [TK]D-Fender | etfonhomey: not a one. |
| 17:46.46 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 17:47.02 | etfonhomey | [TK]D-Fender, Is that what you're gonna recommend to your clients for a regular cubicle/office phone? |
| 17:47.46 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 17:49.10 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 17:50.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:52.14 | *** join/#asterisk MrChicken (n=Dorphals@200.71.58.39) |
| 17:52.21 | MrChicken | Hello |
| 17:52.35 | MrChicken | I have a funny problem with my dialplan |
| 17:52.48 | MrChicken | I'm supposed to recieve a fax and email it to a specified email address |
| 17:52.55 | MrChicken | the tiff file comes without problem |
| 17:53.07 | MrChicken | BUT the emailing script does no get executed |
| 17:55.07 | *** part/#asterisk Zaphod80013 (n=chatzill@74.93.236.61) |
| 17:55.49 | MrChicken | maybe I'm doing something wrong with my priorities. Perhaps somebody could take a look? Its about 5 lines only. Pastebin is: http://www.pastebin.ca/614625 |
| 17:56.02 | MrChicken | I'm supposed to recieve a fax and email it to a specified email address the tiff file comes without problem BUT the emailing script does no get executed. maybe I'm doing something wrong with my priorities. Perhaps somebody could take a look? Its about 5 lines only. Pastebin is: http://www.pastebin.ca/614625 |
| 17:56.08 | *** part/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 17:56.41 | Mercestes | MrChicken, Can you manually sendmail ? |
| 17:56.56 | enjay5150 | with fax2mail |
| 17:57.41 | MrChicken | yeap |
| 17:57.49 | MrChicken | i think it may be in the pipes... |
| 17:58.16 | Mercestes | try removing the 's |
| 17:59.11 | *** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:00.27 | *** join/#asterisk dijungal (i=user@nat/digium/x-3a46a3e4433dd640) |
| 18:00.47 | dijungal | heh |
| 18:00.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:01.02 | dijungal | no asterisk box giving trouble today i c |
| 18:02.08 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:04.46 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 18:08.56 | [TK]D-Fender | etfonhomey: IP 320 more like. |
| 18:10.20 | *** join/#asterisk my007ms (n=my007ms@213.158.171.165) |
| 18:10.39 | dijungal | hey guys... |
| 18:10.57 | dijungal | anyone familliar with chan_zap? |
| 18:11.09 | Strom_M | no |
| 18:11.10 | dijungal | sorry... chan_spy |
| 18:11.15 | MrChicken | Mercestes ... nope |
| 18:11.17 | Strom_M | no one is at all familair with chan_zap |
| 18:11.20 | dijungal | i want to monitor a channel |
| 18:11.22 | Strom_M | it's a giant mystery |
| 18:11.24 | MrChicken | removed the pipes and the ' and nothing |
| 18:11.27 | my007ms | what driver i can use with TDM800P |
| 18:11.32 | dijungal | lol.. sorry about that... not chan_zap... chan_spy |
| 18:11.52 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:12.46 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:12.58 | flujan | hi all... |
| 18:13.08 | *** join/#asterisk g1powermac (n=g1powerm@adsl-18-105-164.sdf.bellsouth.net) |
| 18:13.27 | flujan | I am using 1.4.5 with static configuration files... is it possible to have queues configured in the queues.conf and queue_members in a realtiime database? |
| 18:14.03 | flujan | have you tried this configuration? |
| 18:15.15 | MrChicken | I'm supposed to recieve a fax and email it to a specified email address the tiff file comes without problem BUT the emailing script does no get executed. maybe I'm doing something wrong with my priorities. Perhaps somebody could take a look? Its about 5 lines only. Pastebin is: http://www.pastebin.ca/614625. I have already tested and fax2mail works fine by itself (when I run it from the console it sends the fax). |
| 18:16.27 | *** join/#asterisk stompyj28 (n=stompyj2@c-65-96-94-132.hsd1.ma.comcast.net) |
| 18:16.41 | stompyj28 | hey guys |
| 18:16.43 | stompyj28 | quick question |
| 18:16.50 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 18:17.03 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 18:17.08 | stompyj28 | is it possible to install a codec to allow you to stream mp3s out thru the SIP channel? |
| 18:17.17 | stompyj28 | I know MOH uses MP3s |
| 18:17.26 | stompyj28 | but can you extend that outside of MOH? |
| 18:17.46 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 18:19.19 | Mercestes | stompyj28, I believe playback/background will do that for you. |
| 18:19.25 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM] |
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| 18:19.25 | *** join/#asterisk joe (n=nnnnnnnn@ip66-107-33-195.z33-107-66.customer.algx.net) [NETSPLIT VICTIM] |
| 18:19.25 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
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| 18:19.27 | *** mode/#asterisk [+oo angler file] by irc.freenode.net |
| 18:19.43 | MrChicken | I'm supposed to recieve a fax and email it to a specified email address the tiff file comes without problem BUT the emailing script does no get executed. maybe I'm doing something wrong with my priorities. Perhaps somebody could take a look? Its about 5 lines only. Pastebin is: http://www.pastebin.ca/614625. I have already tested and fax2mail works fine by itself (when I run it from the console it sends the fax). |
| 18:19.43 | stompyj28 | Mercestes: are those AGI commands? |
| 18:20.29 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 18:20.36 | Mercestes | stompyj28, no. Grep your agi file for the appropriate AGI commands and syntax to call those two applications |
| 18:20.43 | stompyj28 | AHHHHHH |
| 18:20.47 | stompyj28 | perfect |
| 18:20.51 | stompyj28 | thanks so much Mercestes :) |
| 18:20.58 | Mercestes | NP. Fight well, AGI warrior |
| 18:21.58 | stompyj28 | LOL classic |
| 18:22.15 | Hmmhesays | AGI fun |
| 18:23.07 | Kwakwa | MrChicken ur lucky, I can't even receive faxes.. get segfaults at the point of fax :) |
| 18:23.45 | tzanger | MrChicken: change that system to the 'extren |
| 18:23.52 | tzanger | I am betting rxfax hangs up before system gets to execute |
| 18:24.18 | [TK]D-Fender | MrChicken: You've already been told to remove the quotes from your system call. Why haven't you? |
| 18:25.01 | Kwakwa | MrChicken, why are you piping to `| --cid-name` |
| 18:25.10 | Kwakwa | thats not gonna work |
| 18:25.15 | Kwakwa | remove that pipe |
| 18:25.29 | Kwakwa | and the latter one for that matter |
| 18:26.05 | etfonhomey | [TK]D-Fender, I think we've discussed this before, but you don't use the switches on IP phones, you recommend 2 data ports in an office/cube, correct? 1 data/ 1 VoIP? |
| 18:26.47 | [TK]D-Fender | etfonhomey: Correct |
| 18:26.49 | Kwakwa | etfonhomey: If u can afford it yeah |
| 18:27.06 | [TK]D-Fender | etfonhomey: Of course you can afford it! |
| 18:27.17 | [TK]D-Fender | etfonhomey: Don't be ridiculous! |
| 18:27.30 | Kwakwa | :) |
| 18:27.33 | *** join/#asterisk huey23 (n=huey23@64.192.209.132) |
| 18:28.32 | MrChicken | Kwakwa... there is a small note in the wiki that says you must change something before ./configure so it wont segfault |
| 18:28.34 | Kwakwa | MrChicken: What version of * are u running to get ur rxfax working? |
| 18:28.49 | MrChicken | [TK]D-Fender ... I removed the single quotes.... |
| 18:28.55 | MrChicken | Kwakwa 1.4.5 |
| 18:29.01 | etfonhomey | [TK]D-Fender: Are your clients using completely different switches for data and VoIP? PoE for the VoIP? Gigabit for the data? |
| 18:29.11 | Kwakwa | MrChicken: oo intersting, which wiki? voip-info? |
| 18:29.17 | MrChicken | Yeap |
| 18:29.22 | [TK]D-Fender | etfonhomey: Yes, almost always PoE. |
| 18:29.26 | Kwakwa | thx, will check that out |
| 18:29.45 | etfonhomey | [TK]D-Fender, still 10/100 for the VoIP PoE switches? |
| 18:29.49 | *** join/#asterisk UKCoder (n=UKCoder@h-72-244-53-172.snfccasy.covad.net) |
| 18:30.09 | Kwakwa | etfonhomey: If u can afford... nah, thats not even funny |
| 18:31.01 | [TK]D-Fender | etfonhomey: Add up the cost of 24 power briks, and that pays for the 24 port PoE Switch. Add-up to cost different of the phones with a switch in them and that pasy for SEPERATE WIRING. You've expanded your topology and opened new possibilities at no net cost. |
| 18:31.21 | MrChicken | [TK]D-Fender ... Should I remove ALL the quotes? |
| 18:31.31 | [TK]D-Fender | etfonhomey: Yeah, PHONES *reall* need that extra bandwitch gigabit offer.... |
| 18:31.40 | [TK]D-Fender | MrChicken: the OUTER |
| 18:31.42 | UKCoder | Hola peeps... I've been using Grandstream GXP-2000's for a while as they've been cheap and relatively good, but their speakerphone quality sucks a lil. I've got a few Polycom SoundStations for real speakerphone usage but can anyone suggest reliable regular desk phones that have decent speakerphone capabilities? |
| 18:31.45 | *** part/#asterisk sysmonk (n=nobody@unaffiliated/sysmonk) |
| 18:31.50 | MrChicken | [TK]D-Fender Did that a long time ago.... |
| 18:31.58 | tzanger | [TK]D-Fender: now now, I could definitely use a phone with a gige switch for our drafting guys |
| 18:32.02 | MrChicken | [TK]D-Fender And no lock |
| 18:32.05 | tzanger | saves having to pull another line for each of them |
| 18:32.08 | [TK]D-Fender | UKCoder: Polcyom > All |
| 18:32.11 | MrChicken | Kwakwa ... did you find it? |
| 18:32.15 | MrChicken | *luck |
| 18:32.21 | [TK]D-Fender | UKCoder: Ditch them all and get normal polycom IP phones. |
| 18:32.41 | etfonhomey | UKCoder: ANd get them from me. :) |
| 18:32.42 | Kwakwa | MrChicken: No, I can't find it |
| 18:32.50 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:32.53 | MrChicken | Kwakwa ... lemme search find it for you |
| 18:32.56 | Kwakwa | MrChicken: I found bits about the patch but not about what flag |
| 18:32.58 | Kwakwa | Thank u |
| 18:32.59 | etfonhomey | Unless of course you're in the UK... |
| 18:33.03 | UKCoder | [TK]D-Fender: do you have a particular model you'd recommend |
| 18:33.08 | [TK]D-Fender | tzanger: and the PRICE of such a phone would MORE than pay for the seperate line and remove a diasychain pile of stupidity :) |
| 18:33.12 | UKCoder | etfonhomey: whereabouts are you? :p |
| 18:33.17 | tzanger | [TK]D-Fender: true |
| 18:33.28 | etfonhomey | In the US |
| 18:33.42 | [TK]D-Fender | UKCoder: depends on your needs. basic users = Polycom IP 320. |
| 18:34.12 | [TK]D-Fender | UKCoder: Receptionists = IP 650. the freeakish few inbetween, est w/o PoE = IP 501 |
| 18:34.28 | UKCoder | etfonhomey: :) I'd guessed that... whereabouts? I'm in norcal here |
| 18:34.41 | UKCoder | [TK]D-Fender: I'd like to run them all PoE :) |
| 18:34.42 | etfonhomey | UKCoder, Kentucky. |
| 18:34.43 | Kwakwa | I've got the Polycom IP 330 for PoE, they're reliable phones. |
| 18:34.54 | Kwakwa | Nice speaker phone too, really nice actually. |
| 18:35.23 | tsurko | I have to implement some features in Asterisk, but I have a problem - I can't identify the caller and there is no way to search the data base and understand which services he has activated/deavctivated. Could you tell me a way to set the caller's username to a variable? How to get it? |
| 18:35.33 | UKCoder | thanks guys, I'll take a look |
| 18:35.46 | MrChicken | Yes |
| 18:35.47 | UKCoder | etfonhomey: do you have a website with prices? |
| 18:35.52 | MrChicken | And I already did that |
| 18:35.56 | MrChicken | and no luck :( |
| 18:36.04 | MrChicken | It just wont call the damned system() |
| 18:37.09 | UKCoder | [TK]D-Fender: I take it that getting those polycoms configured with * is just as easy as any other phone? |
| 18:38.32 | *** part/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net) |
| 18:38.35 | Kwakwa | MrChicken: System(/usr/bin/fax2mail --cid-number ${ARG1} --cid-name ${ARG1} --dest-email dorphalsig@gmail.com -f ${FAXFILE}) |
| 18:38.38 | Kwakwa | U tried it like that? |
| 18:39.21 | MrWup | the higher the number of taps the more powerful the echocancel right? |
| 18:39.39 | MrChicken | Kwakwa http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax |
| 18:40.05 | [TK]D-Fender | UKCoder: Not at all :) |
| 18:40.10 | MrWup | im guessing that if theres a higher number of taps then there is a longer period of time which is being processed to filter out echo right? |
| 18:40.17 | [TK]D-Fender | UKCoder: There is a bit of a learning curve, but its entirely worth it |
| 18:40.52 | UKCoder | [TK]D-Fender: crap *lol* I'm too busy to learn, I just wanna edit my sip.conf and get on with my regular job ;) |
| 18:41.14 | etfonhomey | UKCoder, if I can learn it then anyone can. |
| 18:41.30 | MrWup | ive been doing asterisk for 6 months now |
| 18:41.35 | MrWup | one hell of a learning curve heh |
| 18:41.41 | etfonhomey | [TK]D-Fender, for an installation from scratch which PoE switches do you recommend to your clients? |
| 18:41.55 | MrWup | oh ive got a d-link POE switch |
| 18:41.59 | MrWup | damn good and realiable |
| 18:42.00 | UKCoder | oh, nah, I get *, I've been with it for a few years, just some phones are easier than others |
| 18:42.02 | mocker | Anyone taken the 1.4.x dCAP? |
| 18:42.03 | MrWup | and only £150 |
| 18:42.06 | MrWup | for 24 port poe |
| 18:42.15 | [TK]D-Fender | etfonhomey: http://www.antonline.com/antonline.php?op=inventory&st=DES-1228P |
| 18:42.18 | MrWup | seems very reliable here... its working fine on shitty old cat5 cables |
| 18:42.22 | MrWup | yeah that des one |
| 18:42.23 | [TK]D-Fender | etfonhomey: used to be #76 on their site |
| 18:42.24 | MrWup | is the one ive got |
| 18:42.27 | [TK]D-Fender | $376 |
| 18:43.09 | MrWup | yeap. ee rocks |
| 18:43.13 | [TK]D-Fender | UKCoder: Hire a consulant to get you setup then. |
| 18:43.31 | MrWup | consultant conshmoltant |
| 18:43.39 | MrWup | ive become a consultant in 6 months |
| 18:43.41 | MrWup | =] |
| 18:43.50 | mocker | I'm a consultant on the side. |
| 18:43.55 | mocker | I just don't get any money for it! |
| 18:43.56 | UKCoder | [TK]D-Fender: I'm actually in the process of looking for someone who can be here in person in the SF bay area to do exactly that. I'm a coder, I don't want to spend my time doing this IT stuff ;) |
| 18:44.01 | [TK]D-Fender | MrWup: And still you come in here flooding us with your incessant newb questions! ;) |
| 18:44.05 | MrWup | heeh |
| 18:44.13 | Hmmhesays | in person is over rated |
| 18:44.20 | neverblue | hehe |
| 18:44.21 | Hmmhesays | specially with this voip thing |
| 18:44.23 | UKCoder | try telling my boss that :) |
| 18:44.27 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 18:44.40 | Hmmhesays | yeah bosses stuck with the old business models are going to be left in the dust |
| 18:44.43 | MrWup | probably a really bad habit |
| 18:44.53 | MrWup | the amount of polymer i must be ingestic as a result of chewing |
| 18:44.57 | Hmmhesays | you get the right people and give them freedom and they will be more productive for you |
| 18:46.09 | etfonhomey | Hmmhesays, that's for sure! |
| 18:46.36 | DarylVOIP | Anyone have any luck setting the accountcode from one AGI, and then picking it back up with an $agi->request[agi_accountcode] from another AGI in the same extension (later in the dial plan)? I think I'm lissing something pretty basic, but I just can't figure out how to get it set. |
| 18:47.31 | DarylVOIP | I've even tried a Set(CDR(accountcode)=(variable picked up from agi with account code in it) in the dial plan and it doesn't seem to "stick". |
| 18:47.32 | shido6 | ZZzz |
| 18:47.41 | shido6 | um |
| 18:47.51 | shido6 | you want to set an accountcode and do what with it |
| 18:47.58 | etfonhomey | [TK]D-Fender, any experience with Dell's PoE switches? |
| 18:48.19 | DarylVOIP | When my next AGI fires off, it needs to know the accountcode fo billing/rating purposes. |
| 18:48.31 | [TK]D-Fender | etfonhomey: Nope.... I've never heard anything GOOD about their networking gear |
| 18:48.33 | shido6 | does it do any writing to the db during or just after the call? |
| 18:49.01 | flujan | guys, it is possible to run the queues configuration from the queues.conf file and the members from a table in postgresql? |
| 18:49.05 | DarylVOIP | Yes, I could do it other ways, but I'm re-using scripts. Normally, the users are already registered so I have their account number. In this case, they aren't (as they're calling in on a DID and getting forwarded to another number after I figure out the account number based on the dialed number) |
| 18:49.15 | cpm | heard lots of *bad* about their switches, especially if you actually load them up. |
| 18:49.21 | shido6 | any other variables changing around? |
| 18:49.25 | cpm | As long as they don't do much, apparently they are okay |
| 18:49.57 | DarylVOIP | I'm setting other variables initially, but they don't matter much between AGIs. |
| 18:50.15 | etfonhomey | I've heard good things from people, but they're not doing VoIP. The two VoIP people I've asked this question to said the same thing. |
| 18:51.20 | cpm | what good things have you heard? |
| 18:51.42 | DarylVOIP | THe good part is that they're cheap. |
| 18:52.01 | cpm | what I've heard is they are cheap and easy to administer, which turns out to be of little value when you end up having to replace them because of the packet loss |
| 18:52.13 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:52.15 | DarylVOIP | Not bad for PoE access points. But not so good for VoIP, as they don't seem to have a very good processor and choke on high PPS rates. |
| 18:52.44 | lirakis | is there a way to play a different sound file with the Authenticate() function? |
| 18:52.49 | cpm | fragmented packets locking up NT boxes, printers, all kinds of wierd problems that automagically go away when replaced with cisco gear |
| 18:53.42 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 18:53.45 | cpm | I think there is some lower-end Linksys 'workgroup' layer 3 PoE switchgear out there, that is rationally priced |
| 18:55.25 | prashant_jois | I'm having some problems with echo and crackle. Regular phone conversations are perfectly fine, but when a call goes to voicemail the quality just degrades and I get severe echo and cracke. Does anybody know why this would happen? |
| 18:55.32 | [TK]D-Fender | cpm: Same class as the D-Link |
| 18:55.58 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:55.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:56.23 | cpm | [TK]D-Fender, sorry, hearing D-Link and networking gear in the same context makes me mildly nauseous. |
| 18:56.35 | cpm | But yes, same class |
| 18:56.41 | [TK]D-Fender | cpm: I've been using them for years quite happily. |
| 18:56.46 | cpm | Good. |
| 18:57.03 | MrChicken | Its funnt |
| 18:57.08 | MrChicken | I've nailed down the problem |
| 18:57.08 | cpm | it is |
| 18:57.19 | MrChicken | it just never gets to the System part... |
| 18:57.41 | MrChicken | it finishes everything after the rxfax |
| 18:58.42 | Strom_M | is there any way to stop res_features from adding 700 as a call parking extension? commenting that line out and reloading res_features.so and/or restarting asterisk fails to work |
| 18:58.56 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 19:00.02 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-47-120.dsl.tul2ok.sbcglobal.net) |
| 19:00.13 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 19:00.18 | tzanger | Strom_M: you can change the parking extension |
| 19:00.38 | Strom_M | yeah, but I dont want to change it; I just want to eliminate it completely |
| 19:01.03 | bkw__ | you don't want call parking? |
| 19:01.28 | tzanger | hello bkw__ |
| 19:01.31 | bkw__ | hey |
| 19:02.32 | [TK]D-Fender | Strom_M: Just don't include the context. |
| 19:02.50 | bkw__ | well you can't really get rid of it |
| 19:02.52 | bkw__ | its kinda hard coded |
| 19:04.18 | bkw__ | you can't get rid of it because ast_bridge_call lives in res_features |
| 19:04.32 | cpm | Hey, urrrp, thanks for the spork! |
| 19:05.16 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 19:05.50 | festr__ | hello, i've problem with E1 sangoma -> asterisk -> SIP -> asterisk . asterisks are 1.4 |
| 19:05.51 | UKCoder | Question for all... I'm looking at replacing my SIP origination/termination provider that I use for my own personal use (currently BroadVoice) for someone else just due to inconsistent levels of service. What SIP origination/termination providers would you suggest for home use? |
| 19:05.54 | festr__ | any suggestions? |
| 19:07.07 | *** join/#asterisk ifnotwhynot (n=davidh@c1-52-11.rrba.isadsl.co.za) |
| 19:07.08 | bkw__ | festr__, well describe the problem |
| 19:07.20 | ifnotwhynot | TK TK TK |
| 19:07.46 | ifnotwhynot | got it to work sip to link to remote asterisk server |
| 19:08.03 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:08.21 | festr__ | bkw__: SIP are g729 codec. passing dtmf to E1 -> ast -> sip -> ast B. on ast B i can hear first dtmf distorted and all next dtmf are muted |
| 19:08.33 | bkw__ | what are you on? |
| 19:08.36 | bkw__ | sip phone? |
| 19:08.51 | ifnotwhynot | i can dial out no problem but get this error whiles trying to come into my asterisk serverr |
| 19:09.39 | festr__ | bkw__: bkw__ i've tryed rfc and info no success |
| 19:10.52 | UKCoder | no suggestions from anyone? :( |
| 19:11.19 | [TK]D-Fender | ifnotwhynot: Yippi-kay-yay-mo............ NOW what? ;) |
| 19:12.15 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:13.25 | Hmmhesays | its live free or die hard damnit |
| 19:13.38 | ifnotwhynot | what would this error meenUnable to find a codec translation path from g729 to gsm |
| 19:14.34 | [TK]D-Fender | Hmmhesays: Several working names..... |
| 19:14.57 | [TK]D-Fender | ifnotwhynot: Because you didn't pay for the codec from digium perhaps? |
| 19:15.32 | ifnotwhynot | the asterisk server i link to uses g729 |
| 19:16.25 | etfonhomey | UKCoder, I use Vitelity their decent. |
| 19:16.28 | etfonhomey | or they're decent |
| 19:16.32 | Hmmhesays | I also use vitelity |
| 19:16.41 | Hmmhesays | price is right, quality is decent |
| 19:16.49 | [TK]D-Fender | ifnotwhynot: "thats nice". Your server is trying to answer a call it can't translate to. |
| 19:18.05 | *** join/#asterisk ToyMan (n=Stuart@74-32-41-143.dsl1.mdl.ny.frontiernet.net) |
| 19:18.51 | [TK]D-Fender | Yay.. I've become an accidental web-admin :) Setup a CMS, photo-gallery system, Pastebin, in a sub-domained server :D |
| 19:19.15 | ifnotwhynot | kk i see so you need g729 on my server as well? |
| 19:19.36 | [TK]D-Fender | ifnotwhynot: If * is going to answer the call, yes. |
| 19:19.53 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 19:23.41 | *** part/#asterisk ecoleman (n=eric@24.75.47.130) |
| 19:29.03 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 19:29.43 | *** join/#asterisk shinao1 (n=shinao1@41.205.183.123) |
| 19:32.14 | *** join/#asterisk heartones (n=root@196.205.141.114) |
| 19:33.36 | prashant_jois | I'm having some problems with echo and crackle. Regular phone conversations are perfectly fine, but when a call goes to voicemail the quality just degrades and I get severe echo and cracke. Does anybody know why this would happen? |
| 19:34.06 | enjay5150 | are you using Queues? |
| 19:34.12 | prashant_jois | yes |
| 19:34.20 | enjay5150 | SMP? |
| 19:34.53 | prashant_jois | I don't think so, lemme check though |
| 19:35.01 | *** part/#asterisk huey23 (n=huey23@64.192.209.132) |
| 19:35.04 | NOT_guru | question... any idea how to correct this... in the asterisk console I see the full caller I string of an incomming call, yet all that gets sent to my phones is the last name and first initial |
| 19:35.14 | prashant_jois | enjay5150: not smp |
| 19:35.26 | Mercestes | MrChicken, BTW, why are you quoting your variables? Don't do that. |
| 19:35.34 | enjay5150 | prashant_jois: do you have a zaptel card for timing or you using ztdummy? |
| 19:35.47 | NOT_guru | I would like the whole callerid string sent to the phones |
| 19:35.48 | Mercestes | MrChicken, remove all quotes. |
| 19:35.52 | prashant_jois | enjay5150: zaptel card |
| 19:36.31 | waKKu | afternoon folks ;) |
| 19:36.33 | enjay5150 | lsmod shows only that specific zaptel card module not other cards/ztdummy? |
| 19:36.43 | waKKu | folks.. how can I do to send fax using spandsp ??? |
| 19:36.51 | *** join/#asterisk stoffell (n=stoffell@d51A4D441.access.telenet.be) |
| 19:36.55 | prashant_jois | enjay5150: just tor2 (and zttranscode) |
| 19:37.06 | enjay5150 | what kind of card do you have? |
| 19:38.06 | prashant_jois | enjay5150: TE410P, i.e. TDM card |
| 19:38.26 | enjay5150 | wrote card.. |
| 19:38.32 | enjay5150 | wrong mod rather.. |
| 19:38.36 | enjay5150 | TE410P does not use tor2 |
| 19:38.40 | enjay5150 | it uses wct4xxp |
| 19:39.25 | prashant_jois | enjay5150: oh okay, well the 410P card must be in the other box, how can I find out what model this one is? |
| 19:40.00 | enjay5150 | open the case? |
| 19:40.09 | enjay5150 | is it a quad card? single card? |
| 19:40.12 | prashant_jois | enjay5150: quad |
| 19:40.19 | enjay5150 | analog card? |
| 19:40.24 | enjay5150 | or PRI? |
| 19:40.31 | enjay5150 | erm digital.. whatever. |
| 19:40.36 | prashant_jois | enjay5150: digital yeah |
| 19:40.45 | prashant_jois | enjay5150: PRI |
| 19:40.48 | enjay5150 | okay well pop the case and see what it says on it.. |
| 19:41.09 | prashant_jois | enjay5150: ok, brb |
| 19:43.50 | NOT_guru | any suggestions on my c allerID issue? |
| 19:44.34 | enjay5150 | what kind of phone? |
| 19:46.21 | [TK]D-Fender | NOT_guru: pastebin the full CLI output dumping the CID, and write down what you SEE. Also your phone model. |
| 19:46.59 | NOT_guru | thanks D-Fender I will start gathering |
| 19:50.11 | prashant_jois | enjay5150: The machine is in a server rack and I can't get access to it, is there any other way to figure out the model? Would the card even run if it didn't have the right module? |
| 19:52.41 | enjay5150 | cat /proc/interrupts and look for wct or something alike. |
| 19:53.20 | g1powermac | Hi All |
| 19:53.45 | g1powermac | has anyone played with the High Performance Echo Cancellation program sold by Digium? And is it really $10? |
| 19:53.47 | waKKu | folks... i'm getting a annoying problem.. when i terminate a call (on zap channel) and put the phone on hook, the call dont Hangup on asterisk.. its start a music on hold insted of .... can someone help me ? |
| 19:54.26 | [TK]D-Fender | g1powermac: its free if your card is still under warranty IIRC |
| 19:54.27 | prashant_jois | enjay5150: I don't see any wct, but I see this: 11: 1632752692 XT-PIC libata, ehci_hcd, uhci_hcd, tor2 |
| 19:54.38 | [TK]D-Fender | prashant_jois: bad module. |
| 19:54.42 | g1powermac | [TK]D-Fender, ahh, neat |
| 19:54.53 | g1powermac | [TK]D-Fender, but do you know if it works well? |
| 19:54.54 | [TK]D-Fender | prashant_jois: modprob -r tor2 |
| 19:55.09 | [TK]D-Fender | prashant_jois: modprob wct4xxp |
| 19:55.18 | [TK]D-Fender | g1powermac: For most, yes |
| 19:55.23 | g1powermac | cool |
| 19:55.33 | [TK]D-Fender | g1powermac: in so much as I've heard LITTLE... |
| 19:55.43 | g1powermac | ok |
| 19:55.46 | [TK]D-Fender | g1powermac: I'd check some other sources like the mailing list. |
| 19:56.17 | NOT_guru | D-Fender think I need to mask out the phone numbers from the cli output? |
| 19:56.34 | [TK]D-Fender | NOT_guru: Believe me. Nobody gives a @&#^% |
| 19:56.36 | g1powermac | and have heard that will help reduce the echoing |
| 19:56.43 | *** part/#asterisk heartones (n=root@196.205.141.114) |
| 19:56.46 | NOT_guru | sure just checking |
| 19:57.01 | [TK]D-Fender | g1powermac: EW, why? |
| 19:57.07 | NOT_guru | I did mask out my public IP |
| 19:57.21 | [TK]D-Fender | NOT_guru: sure fine, whatever. |
| 19:57.34 | prashant_jois | [TK]D-Fender: how do you know it's a bad module? |
| 19:57.49 | g1powermac | [TK]D-Fender, I need to use existing analog equipment |
| 19:58.02 | NOT_guru | D-Fender : http://www.pastebin.ca/614803 |
| 19:58.05 | [TK]D-Fender | prashant_jois: thats the wrong one... tor is for the most ancient revision of the tormenta cards. |
| 19:58.12 | g1powermac | don't have a big enough budget to replace the analog stuff at this point |
| 19:58.13 | enjay5150 | prashant_jois what happens when you modprobe wct4xxp |
| 19:58.58 | [TK]D-Fender | NOT_guru: What do you see on the phone? |
| 19:59.03 | [TK]D-Fender | NOT_guru: And what model is it? |
| 19:59.08 | enjay5150 | prashant_jois; point being I think your stutter/moh issues are timing related.. |
| 19:59.15 | NOT_guru | Billy G (phone numner) |
| 19:59.16 | [TK]D-Fender | NOT_guru: nvm I see it |
| 19:59.31 | [TK]D-Fender | NOT_guru: so it chopped the number off the name? |
| 19:59.44 | prashant_jois | enjay5150: Well the card is currently live with calls ... would modprobe wct4xxp affect that? |
| 19:59.46 | NOT_guru | chopped of the oat in Goat |
| 20:00.07 | NOT_guru | my callerid is literally " Billy Goat" |
| 20:00.09 | enjay5150 | wait till you can take it down temporarily.. |
| 20:00.10 | g1powermac | [TK]D-Fender, so I don't have much of a choice at this point |
| 20:00.33 | enjay5150 | D-Fender will a TE410/11/12P card load with tor2? or can he truly have an old T400P |
| 20:00.36 | NOT_guru | shows on all phones Billy G |
| 20:01.04 | NOT_guru | my inbound route goes directly to a ringgroup |
| 20:01.13 | prashant_jois | enjay5150: well, how come it doesn't affect normal calls, only voicemail? |
| 20:01.25 | [TK]D-Fender | NOT_guru: Ok, just damned wierd |
| 20:01.25 | neverblue | java and asterisk, having an info about it? |
| 20:01.43 | enjay5150 | voicemail and music on hold right? |
| 20:02.00 | NOT_guru | agreed |
| 20:02.07 | NOT_guru | is that enough info for ya though |
| 20:02.16 | [TK]D-Fender | NOT_guru: Keep that PB around, maybe someone else will have a better idea. |
| 20:02.22 | [TK]D-Fender | NOT_guru: Yeah, thats good |
| 20:02.23 | prashant_jois | enjay5150: just voicemail, moh seems fine |
| 20:02.24 | NOT_guru | all my inbounds get stripped like that |
| 20:02.33 | [TK]D-Fender | NOT_guru: maybe its your telco |
| 20:02.34 | NOT_guru | ok thanks man |
| 20:02.50 | NOT_guru | but it makes it to the box in full |
| 20:02.51 | prashant_jois | enjay5150: actually I take that back, that channel doesn't have moh so I don't know one way or another |
| 20:02.54 | NOT_guru | dropped at the box |
| 20:02.54 | ifnotwhynot | if i set up my connection to the remote server using sip in my sip.conf Remote context i have host=free.net works fine for outgoing but does not work for incoming if i change it to host=dynamic i can get incomming calls t my local address but the outgoing does not help |
| 20:03.04 | NOT_guru | thats the wierd part |
| 20:03.32 | DarylVOIP | Ahhhhhh! SIP/2.0 603 Declined |
| 20:03.38 | bkruse | owned. |
| 20:03.49 | [TK]D-Fender | -- Executing NoOp("Zap/2-1", "user-callerid: Billy G 3147278896") in new stack |
| 20:03.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:03.55 | DarylVOIP | This DID works on two other boxes that appear to be configured the same. |
| 20:03.57 | [TK]D-Fender | NOT_guru: doesn't look like it comes in full |
| 20:04.01 | ZaVoid | so is there anything that would make an asterisk send a proxy auth required in response to a sip invite when insecure=very in the sip context and the general context also has allowguest=yes |
| 20:04.12 | enjay5150 | prashant_jois: might have something to do witht he module sharing interrupts with all that other shit. |
| 20:04.51 | NOT_guru | Executing Set("SIP/1278-083a07b0", "TRUNKOUTCID="Billy Goat"<3147278896>") in new stack |
| 20:04.59 | prashant_jois | enjay5150: I'm gonna try loading the wct driver and see what happens |
| 20:05.01 | NOT_guru | err I see |
| 20:05.04 | NOT_guru | bad example |
| 20:05.11 | NOT_guru | lemme look closer |
| 20:05.12 | NOT_guru | sorry |
| 20:05.16 | [TK]D-Fender | NOT_guru: thats being set before calling OUT, problem is thats not how it comes IN. |
| 20:05.23 | NOT_guru | I see that now |
| 20:05.24 | NOT_guru | sorry |
| 20:05.27 | [TK]D-Fender | NOT_guru: np |
| 20:05.39 | [TK]D-Fender | NOT_guru: So check your telco..... they may be mangling it. |
| 20:05.55 | NOT_guru | those punks |
| 20:05.55 | *** join/#asterisk ManxPower (n=manxpowe@015-811-440.area5.spcsdns.net) |
| 20:05.57 | NOT_guru | LOL |
| 20:06.33 | [TK]D-Fender | NOT_guru: what are you using for zap 1 & 2? |
| 20:06.46 | [TK]D-Fender | (card / protocol) |
| 20:06.47 | NOT_guru | tdm4XX |
| 20:07.15 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 20:07.23 | [TK]D-Fender | NOT_guru: Uhhh....... ANANLOG?! |
| 20:07.25 | NOT_guru | with 2 FXO modules |
| 20:07.28 | NOT_guru | yes |
| 20:07.33 | yannj_fr | by the way [TK]D-Fender, my problem with IMAP storage and mailbox default at context default, was a bug, I reported it and it was corrected... |
| 20:07.34 | [TK]D-Fender | NOT_guru: You can't SET callerid on analog! |
| 20:07.36 | NOT_guru | zap1 telco |
| 20:07.42 | Mercestes | hehehe |
| 20:07.46 | NOT_guru | zap2 sunrocket gizmo into it |
| 20:07.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 20:08.00 | NOT_guru | well I am not trying to set it |
| 20:08.17 | NOT_guru | when I use that line ( say I just hook up a phone ) and call a buddy |
| 20:08.17 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 20:08.21 | NOT_guru | they see Billy Goat |
| 20:08.33 | NOT_guru | I am literally listed as last name billy first name goat |
| 20:08.41 | [TK]D-Fender | NOT_guru: your incoming CID is what that line is designed to do. |
| 20:08.43 | NOT_guru | in phone book |
| 20:08.52 | ifnotwhynot | what can this mean chan_sip.c:13185 reload_config: Section '21' lacks type ? |
| 20:10.31 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 20:10.47 | NOT_guru | I will do more testing when I get home |
| 20:10.58 | russellb | ifnotwhynot: you have a section [21] without a type=something |
| 20:11.02 | NOT_guru | I have like 4 numbers on my home box |
| 20:11.14 | NOT_guru | 1 zap 1 |
| 20:11.17 | NOT_guru | 2 zap 2 |
| 20:11.27 | NOT_guru | 1 sip to viatalk |
| 20:11.51 | ZaVoid | so is there anything that would make an asterisk send a proxy auth required in response to a sip invite when insecure=very in the sip context and the general context also has allowguest=yes |
| 20:12.30 | *** join/#asterisk figu78 (n=root@147.46.220.87.dynamic.jazztel.es) |
| 20:12.59 | *** part/#asterisk figu78 (n=root@147.46.220.87.dynamic.jazztel.es) |
| 20:13.03 | *** join/#asterisk figu78 (n=root@147.46.220.87.dynamic.jazztel.es) |
| 20:13.06 | *** part/#asterisk figu78 (n=root@147.46.220.87.dynamic.jazztel.es) |
| 20:16.28 | *** join/#asterisk vn (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 20:17.22 | vn | hi, just got my spa2102 and reading the book...it mentions configuration of zapata and zaptel...do I need to do that since my device is nothing related to zap*? |
| 20:18.05 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491790F.dip.t-dialin.net) |
| 20:18.13 | *** part/#asterisk s1gny|wrk (n=s1gny@p5491790F.dip.t-dialin.net) |
| 20:18.49 | vn | and uh...since I'm behind a router, do I plug a wire from it to my ethernet plug on the spa2102 or the internet even if its not directly internet? |
| 20:18.55 | [TK]D-Fender | vn: for that device nope. Exception : IAX2 trunking and MeetMe require Zaptel (ZTDMMY at least) for timing |
| 20:19.10 | *** join/#asterisk snowy_owl (n=sauron@189.21.62.26) |
| 20:19.34 | vn | well I plan on using IAX if SIP sucks as much as ppl are saying |
| 20:19.39 | [TK]D-Fender | vn: you plug the "internet" (blue) jack to your main router |
| 20:19.51 | vn | dunno if IAX trunking is part of the game |
| 20:19.52 | vn | ok |
| 20:20.05 | vn | cool, that simple! |
| 20:20.18 | [TK]D-Fender | vn: SIP doesn't suck, and your SPA will speak SIP to * which is fine, but you may want to use IAX AFTER to connect to an IAX-speaking ITSP |
| 20:20.40 | vn | well to begin I'll use it without * |
| 20:20.47 | mocker | I read on the list people complaining about quality problems w/ ITSPs using IAX. |
| 20:20.55 | mocker | And that SIP yielded better sounding calls.. |
| 20:20.55 | vn | then I plan on using the *'s VM |
| 20:21.07 | [TK]D-Fender | vn: Use you ATA without Asterisk?! Connecting to who then? |
| 20:21.57 | ZaVoid | fender got any ideas about the proxy auth? |
| 20:21.58 | vn | phone---ata---router---simplitel |
| 20:22.53 | vn | I HEArd it can work.. |
| 20:23.12 | [TK]D-Fender | vn: Certainly. |
| 20:23.21 | [TK]D-Fender | vn: but this IS #asterisk you know.... |
| 20:23.39 | vn | yeah :p |
| 20:23.50 | [TK]D-Fender | vn: normally you'd be learning how to set it up with * and then get * set up with your ITSP |
| 20:24.34 | vn | yeah well first I wanna learn my spa2102 |
| 20:24.58 | vn | I'll have to use * if I want a voicemail and such |
| 20:25.30 | vn | btw I'm reading the book and I think I noticed some errors...who should I tell this to? |
| 20:25.45 | [TK]D-Fender | VN : well the next release is about to be published :) |
| 20:25.55 | [TK]D-Fender | vn: And there were all sorts of little things :) |
| 20:25.59 | vn | k |
| 20:26.00 | [TK]D-Fender | vn: such is life |
| 20:26.05 | vn | yeap! |
| 20:26.13 | [TK]D-Fender | vn: Ask in here and you're get where you need to go. |
| 20:26.43 | *** part/#asterisk UKCoder (n=UKCoder@h-72-244-53-172.snfccasy.covad.net) |
| 20:27.35 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 20:27.38 | *** join/#asterisk VJFROMGT (n=vjfromgt@190.80.51.236) |
| 20:28.16 | VJFROMGT | is there a way to put a delay after using a trunk, eg, when i hangup the trunk i just hung up will not be available for the next 3 seconds |
| 20:28.57 | snowy_owl | Hi all. I have a stupid question. I'm sending a INVITE to asterisk (A) and it sends it out to another point (B). But, this B doesnt answer. So, I send the same INVITE to asterisk WITH A DIFFERENT RURI. But * doesnt answer. There are a variable in any confs that I can set to avoid this behavior? |
| 20:30.46 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 20:31.34 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:41.45 | *** join/#asterisk vitor_ctba (n=vitor@200.150.64.60) |
| 20:42.40 | vitor_ctba | hi there...! does anybody have a regular faxmachine working under VOIP ????? |
| 20:43.19 | Strom_M | the extra question marks really help |
| 20:43.37 | Mercestes | Yes... |
| 20:43.52 | Mercestes | the first four times I was like, "man, is this *really* a question?" the fifth time tho, I was certain. |
| 20:44.23 | Mercestes | and one normally does not pause briefly in an excited manner, so the ...! really confuses me. |
| 20:45.51 | Strom_M | well, he could be calmly saying "hi there" followed by a brief pause where he grins ecstatically and shakes a bit |
| 20:46.08 | Mercestes | Strom_M, that must be it |
| 20:46.33 | Strom_M | vitor_ctba: are you attempting to use a fax machine, or are you considering doing so and merely want to know whether it's doable? |
| 20:47.20 | vitor_ctba | iam trying to have a faxmachine working under voip.... |
| 20:47.44 | vitor_ctba | but it does not work :( |
| 20:48.18 | Strom_M | vitor_ctba: break it down |
| 20:48.21 | Strom_M | fax over voice over ip |
| 20:48.22 | vitor_ctba | is there any trick ?! |
| 20:48.26 | Strom_M | doesn't work :) |
| 20:48.32 | Strom_M | look into using t.38 |
| 20:48.59 | ChkDigit | vitor_ctba: Does your ATA gateway support T.38? |
| 20:49.07 | tzanger | so the current voip fax solution for asterisk is t38modem and h323? |
| 20:49.39 | Strom_M | what does h323 have to do with it? |
| 20:50.31 | vitor_ctba | ChkDigit, yes... |
| 20:51.44 | ChkDigit | Does the other end? |
| 20:53.01 | JT | tzanger: no the current asterisk voip fax solution is "no" |
| 20:53.22 | Strom_M | no no no / no no no / tesla girls, tesla girls / etc |
| 20:53.23 | tzanger | JT: yeah you were mentioning callweaver, which is something Iĺl have to investigate |
| 20:53.29 | tzanger | Strom_M: hahaha |
| 20:53.38 | tzanger | I haven heard that in forever |
| 20:53.40 | vitor_ctba | yes, the both sides... but, the fax don't start. |
| 20:53.47 | Strom_M | tzanger: i love that album :) |
| 20:53.55 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable159.131-56-74.mc.videotron.ca) |
| 20:54.08 | vitor_ctba | i dont know what happens.... :( |
| 20:54.16 | JT | vitor_ctba: fax over VoIP DOES NOT WORK |
| 20:55.14 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:55.23 | DrukenLPY | afternoon everyone..... |
| 20:55.30 | JT | morning |
| 20:55.54 | DrukenLPY | i have a stupid question... how do i make the directory search by first names...?? shouldn't it be an option in voicemail.conf? |
| 20:55.57 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 20:56.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 20:57.01 | *** join/#asterisk kannan (n=kannan@61.17.179.124) |
| 20:57.07 | kannan | hi all |
| 21:00.22 | ChkDigit | JT/vitor_ctba: Fax over VoIP works in some situations. I use it here without trouble, but that is over a LAN... |
| 21:01.33 | tzanger | JT: not entirely true |
| 21:01.47 | tzanger | fax over voip wherein youŕe trying to send the audio around rarely works well |
| 21:01.52 | fujin | anyone seen aything like this? Jul 12 09:01:17 WARNING[26721]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8135228', 10 retries! |
| 21:01.53 | tzanger | fax over voip using t37 or t38 works just fine |
| 21:01.59 | fujin | happens whenever I make an outgoing call |
| 21:04.38 | vitor_ctba | ChkDigit, thanks..... do you know a text/howto/bibliography about fax over VOIP ?? |
| 21:05.30 | ChkDigit | Nope, but have you checked the traffic between the ATA/Asterisk box and the other side? |
| 21:05.52 | ChkDigit | Do they even start a conversation? |
| 21:06.01 | tzanger | fujin: have you tried restarting *? I am not sure what else to suggest at this point |
| 21:06.07 | tzanger | any hung channels or anything? |
| 21:07.19 | vitor_ctba | ChkDigit, yes, even start a transmission... and... for calls, they work fine... |
| 21:07.46 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:08.27 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:11.03 | *** join/#asterisk Cardoe (n=cardoe@gentoo/developer/Cardoe) |
| 21:11.30 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 21:12.34 | *** join/#asterisk errr (n=mike@fedora/errr) |
| 21:13.53 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:14.01 | errr | if I need to update my asterisk from 1.2.16 to the current 1.2.21.1 do I have to update my zaptel too because if I dont need to I dont really want to |
| 21:14.26 | [TK]D-Fender | errr, almost certainly. |
| 21:14.57 | errr | :( |
| 21:15.04 | errr | ok then, thanks |
| 21:15.11 | Strom_M | what's the big deal? |
| 21:15.21 | Strom_M | compile, install, unload, reload, restart |
| 21:15.22 | Cardoe | For some reason I'm not getting $CALLERIDNUM, $CALLERID, and $DATETIME working for me on Asterisk 1.4.4 |
| 21:15.27 | Cardoe | exten => s,1,System(/usr/bin/mythtvosd --template=cid --caller_name="${CALLERID}" --caller_number="${CALLERIDNUM}" --caller_date="${DATETIME}") |
| 21:15.32 | Strom_M | Cardoe: read UPGRADE.txt |
| 21:15.38 | Cardoe | am I calling that improperly |
| 21:15.42 | errr | its just a pain in the ass because this means I have to redo the wanpipe stuff too because it has to patch zaptel or what ever |
| 21:15.43 | Strom_M | those variables were deprecated in 1.2 and removed in 1.4 |
| 21:16.09 | Cardoe | Strom_M: well.. this was a new install.. so I never read the UPGRADE.txt... I assume that would be documented else where.... |
| 21:16.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 21:16.31 | Cardoe | Strom_M: what would be the new variables? |
| 21:16.34 | Strom_M | UPGRADE.txt details all changes from previous version |
| 21:16.42 | Strom_M | Cardoe: UPGRADE.txt will tell you |
| 21:16.47 | Cardoe | since voip-info.org seems to only reference those. |
| 21:16.52 | Strom_M | no... |
| 21:16.58 | *** join/#asterisk fauxalliance (n=fa@stjhnf0120w-142162214053.pppoe-dynamic.nl.aliant.net) |
| 21:17.17 | Cardoe | Strom_M: my point still stands. A user installing for the first time does not read UPGRADE.txt since they're not performing an upgrade. |
| 21:17.20 | fauxalliance | Hello everyone. |
| 21:17.45 | fauxalliance | Is there any way to tell asterisk to ignore a ringing zap channel if there is no appropriatly defined inbound route?? |
| 21:18.02 | Strom_M | Cardoe: and a user installing the latest version should still be aware of changes from previous versions in case he or she decides to use code samples from old versions of asterisk which may contain deprecated items :) |
| 21:18.03 | fauxalliance | as opposed to declaring that the number is not in service? |
| 21:18.16 | fordfrog | hi, is there a way in * to force sending "183 Session Progress" using command, something like Ringing causes "180 Ringing" to be sent? |
| 21:18.24 | file | fordfrog: Progress |
| 21:19.03 | fordfrog | file: uf, that was fast, thx :-) |
| 21:19.18 | Cardoe | Strom_M: I didn't use a code sample. I wrote that myself. |
| 21:20.07 | Strom_M | well, you're still referencing old code :) |
| 21:20.13 | Cardoe | That's fine. |
| 21:20.14 | Cardoe | I'll update it. |
| 21:20.19 | Cardoe | I came in here to ask what I was doing wrong. |
| 21:20.27 | Strom_M | you used deprecated variables. |
| 21:21.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:23.25 | Cardoe | Strom_M: some of the documentation is a bit off as well.. such as the replacement for $DATETIME |
| 21:25.59 | Cardoe | it's also terrible fashion that $STRFTIME doesn't match the standard C function. But oh well. |
| 21:26.22 | Cardoe | Strom_M: http://www.voip-info.org/wiki-Asterisk+variables look at the $DATETIME reference |
| 21:33.19 | bkruse | voip-info.org == old |
| 21:33.24 | bkruse | ${EPOCH}! |
| 21:34.49 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:34.57 | shmaltz | whats wrong with the list? |
| 21:35.50 | bkruse | shmaltz: its down, will be back up shortly |
| 21:36.05 | bkruse | or not down, i dont believe, just overloaded |
| 21:36.22 | shmaltz | i c |
| 21:37.54 | bkruse | either way its being worked on :] |
| 21:46.44 | *** join/#asterisk sharp (n=sharp@c-68-81-156-176.hsd1.pa.comcast.net) |
| 21:49.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:50.50 | *** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 21:51.35 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 21:51.59 | *** join/#asterisk enjay5150 (n=enjay@74.202.4.2) |
| 21:52.19 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:55.31 | Strom_M | reload res_brain.so |
| 21:55.58 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:56.47 | *** join/#asterisk [[blah]asdf (n=ckwall@206.71.78.172) |
| 21:57.12 | [[blah]asdf | i am trying to run ./zttest but it does not reply. what would cause this? |
| 21:57.20 | [[blah]asdf | i have chan_zap running |
| 21:57.24 | [[blah]asdf | ztcfg was run |
| 21:57.27 | [[blah]asdf | with no errors |
| 21:57.37 | [[blah]asdf | what else could I have missed? |
| 21:58.32 | tzafrir_laptop | [[blah]asdf, you don't have a working zaptel timing source |
| 21:58.42 | enjay5150 | what does ztcfg -vvvv return? |
| 21:58.46 | tzafrir_laptop | do you have a zaptel card? |
| 21:59.18 | [[blah]asdf | yes on the zap card... after all channels are configured then the result is 96 channels configured. no errors |
| 22:00.05 | enjay5150 | zttest returns what? |
| 22:00.15 | *** join/#asterisk idefix321 (n=correspo@p5B0757FE.dip.t-dialin.net) |
| 22:00.17 | idefix321 | hi |
| 22:00.33 | [[blah]asdf | nothing... just sits there. |
| 22:00.41 | idefix321 | is there a way to let the asterisk gui (webinterface) read the already configured files? |
| 22:00.55 | [[blah]asdf | "Opened pseudo zap interface, measuring accuracy..." |
| 22:00.59 | [[blah]asdf | then nothing |
| 22:02.05 | enjay5150 | hmmm |
| 22:02.48 | tzafrir_laptop | [[blah]asdf, are those spans TE? |
| 22:02.54 | [[blah]asdf | t1 |
| 22:03.01 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:03.34 | tzafrir_laptop | are they configured to get timing from the remote party? what do you have on the span lines in zaptel.conf? |
| 22:03.39 | *** join/#asterisk MrPloddy29 (n=ploddy@87-194-62-72.bethere.co.uk) |
| 22:05.11 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:08.23 | [[blah]asdf | tzafrir_laptop: http://pastebin.ca/615017 |
| 22:10.14 | tzafrir_laptop | [[blah]asdf, do you get interrupts from the cards? watch -d -n1 cat /proc/interrupts |
| 22:10.20 | tzafrir_laptop | ctrl-c to exit |
| 22:12.33 | [[blah]asdf | http://pastebin.ca/615035 |
| 22:13.50 | [[blah]asdf | nothing changes on irq 11 at all while i do a watch |
| 22:14.45 | *** join/#asterisk vn- (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 22:14.47 | idefix321 | hmmm |
| 22:18.26 | putnopvut | What's the 1.2 equivalent of 1.4 and trunk's "module reload" cli command? |
| 22:18.34 | neverblue | callerid="Jane Smith"<5678> <-- will this return an error, because of the lack of the space between the "< ? |
| 22:19.58 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
| 22:21.54 | [TK]D-Fender | putnopvut, "reload [modulename]" |
| 22:21.56 | file | putnopvut: reload <filename> |
| 22:22.32 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 22:22.45 | *** part/#asterisk [[blah]asdf (n=ckwall@206.71.78.172) |
| 22:22.51 | putnopvut | Thanks [TK]D-Fender and file. |
| 22:24.03 | JT | tzanger: T.37 and T.38 are NOT fax over VoIP. |
| 22:24.39 | JT | fax over VoIP isn't very reliable. fax over VoIP over Internet is basically utterly useless |
| 22:26.50 | idefix321 | no one a idea? |
| 22:28.49 | neverblue | or is the syntax not that specific? |
| 22:28.53 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:29.55 | JT | neverblue: it will probably return an error because you used quotation marks, iirc |
| 22:30.24 | neverblue | huh? |
| 22:30.40 | neverblue | what is the syntax for defining a callerid then? |
| 22:30.52 | neverblue | ~thebook |
| 22:31.01 | jbot | i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:32.05 | idefix321 | how to import the configuration to the asterisk web gui? |
| 22:32.47 | JT | idefix321: i don't think you can |
| 22:32.50 | JT | why use a gui? |
| 22:33.01 | idefix321 | good question |
| 22:33.05 | idefix321 | thanks by the way |
| 22:33.21 | idefix321 | to make win user my slaves... |
| 22:33.26 | idefix321 | ;-) |
| 22:34.01 | *** join/#asterisk NotHere (n=steve@athedsl-247008.home.otenet.gr) |
| 22:35.19 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 22:35.35 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 22:35.47 | NotHere | I use asterisk 1.4.7.1 with a TDM411B card and when I call an outside line, or answering a line from inside there is only noise heard to the one calling |
| 22:35.58 | NotHere | and I hear just silence |
| 22:37.36 | sevard | how do you hear silence |
| 22:37.57 | NotHere | if someone calls from the outside, the call is correctly routed to my telephone and I hear the ring and see a correct caller id. So there is a connection between the points. But the one calling just hears a "vrrrrr" |
| 22:38.23 | NotHere | sorry. He hears the "vrrrrrr" when I pickup the phone |
| 22:38.47 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 22:39.16 | NotHere | sevard, when I answer the phone I hear nothing. And don't ask me how I hear the "nothing" :) |
| 22:39.58 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:41.32 | NotHere | my outside line is an ISDN line but I have connected asterisk to the analog port of the device doing the termination of the isdn line. That's the same port I can connect a regular telphone and use it just fine |
| 22:41.40 | NotHere | is this somehow relevant? |
| 22:41.59 | JT | it's relevant in that it's a retarded way of connecting asterisk |
| 22:42.01 | JT | ;) |
| 22:42.08 | JT | i assume you mean BRI ISDN |
| 22:42.29 | NotHere | JT, Yes. And I don't have the money for an isdn asterisk card... |
| 22:42.44 | JT | i guess 30 euros is pretty expensive... |
| 22:43.19 | NotHere | JT, the cards I saw where in the range of 500 Euros and more. Can you provide a link? |
| 22:43.56 | idefix321 | conceptronic cars will do it, i guess |
| 22:44.01 | idefix321 | conceptronic cards will do it, i guess |
| 22:44.16 | JT | NotHere: any single port HFC-s based card |
| 22:44.22 | JT | they're pretty cheap in europe |
| 22:44.35 | NotHere | you say that asterisk can't deal with my compination and the source of my problems is the isdn? |
| 22:44.55 | JT | no |
| 22:45.15 | JT | i'm just saying it makes no sense to remove all the advantages of isdn by connecting with analogue |
| 22:45.24 | JT | i have no idea what the cause of your problem is |
| 22:46.07 | *** join/#asterisk vn (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 22:46.29 | NotHere | from digium there is just this card for BRI: http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=B410P which is $950... |
| 22:46.52 | JT | so don't buy that card. |
| 22:46.57 | JT | you don't need 4 ports |
| 22:47.03 | idefix321 | i can only show you a german page for that card... |
| 22:47.06 | JT | and digium is not the only maker of cards |
| 22:47.38 | *** join/#asterisk dijungal (i=user@nat/digium/x-80d985c8cadc94de) |
| 22:48.00 | sweeper | YEA |
| 22:48.07 | sweeper | sangooooooma |
| 22:48.20 | sweeper | actually, there's probably quite a few BRI card makers |
| 22:48.39 | *** join/#asterisk MrMister2 (n=mrmister@89-180-10-204.net.novis.pt) |
| 22:49.02 | JT | beronet and junghanns were around way before on the bri scene |
| 22:49.05 | JT | but they're not cheap |
| 22:49.08 | bkruse | yep |
| 22:49.09 | bkruse | b410p -> digium |
| 22:49.17 | JT | the cheap ones are only single ports |
| 22:49.18 | JT | there's heaps of clones |
| 22:49.20 | bkruse | right |
| 22:49.25 | JT | using the hfc-s chipset |
| 22:49.32 | idefix321 | http://www.bluehand.biz/eshop.php?action=article_detail&s_supplier_aid=156326&rid=shopinfo |
| 22:49.35 | sweeper | crazy eu and their crazy isdn usage |
| 22:49.36 | idefix321 | 8 euro |
| 22:49.46 | JT | sweeper: don't be jealous of them |
| 22:49.56 | sweeper | I'm not :v |
| 22:49.59 | idefix321 | isdn rocks |
| 22:50.06 | enjay5150 | jelous of isdn? haha |
| 22:50.15 | sweeper | I'd much rather pay the same prive for a 20/5mb internet connection :D |
| 22:50.22 | sweeper | and just get sip termination~ |
| 22:50.34 | JT | enjay5150: what's so amusing? |
| 22:50.51 | JT | sweeper: err, i'm talking about for VOICE use |
| 22:50.54 | JT | not data |
| 22:50.54 | JT | ... |
| 22:50.55 | bkruse | Whats a good internet provider in the states? |
| 22:50.56 | sweeper | me too! |
| 22:51.22 | sweeper | yay FIOS <3 |
| 22:51.22 | enjay5150 | ISDN is a shit technology.. |
| 22:51.22 | JT | enjay5150: in what way? |
| 22:51.22 | sweeper | bkruse: uh, depends where you live :P |
| 22:51.22 | bkruse | huntsville |
| 22:51.26 | bkruse | hsv, alabama |
| 22:51.30 | sweeper | yea, so google who provides service there |
| 22:51.31 | JT | sweeper: well sip terminations aren't reliable, but if that's what you like, okay |
| 22:51.39 | sweeper | odds are you're stuck with cox or bell |
| 22:51.48 | bkruse | I know who, im wondering if anyone has had good experience with anyone |
| 22:51.52 | sweeper | cox is pretty decent, prices in LA were great for 10/2 |
| 22:52.07 | sweeper | JT: dude, you can get voice circuits from verizon over fios |
| 22:52.25 | sweeper | if you care about it that much |
| 22:52.31 | sweeper | it's like an extra $10 |
| 22:52.39 | JT | sweeper: so what does that have to do with isdn and sip? |
| 22:52.54 | enjay5150 | reliable? I push out over 700k calls per day between centers all via SIP, I'd say its reliable. |
| 22:52.55 | sweeper | I'm not jealous of eu's and their ISDN~ |
| 22:53.04 | sweeper | because I have fios :3 |
| 22:53.13 | JT | enjay5150: if you have private networks, sure |
| 22:53.25 | JT | but over the Internet if there's elements you don't control |
| 22:53.29 | enjay5150 | its as reliable as your carrier and provider.. |
| 22:53.38 | sweeper | JT: so get some peering :P |
| 22:53.41 | JT | which is where most people get their sip terminations through |
| 22:53.46 | JT | sweeper: for home use? |
| 22:53.48 | enjay5150 | hence why typically reliability relies on clearing houses.. not single providers.. |
| 22:53.53 | sweeper | JT: yes! XD |
| 22:54.02 | JT | enjay5150: thanks for the obvious |
| 22:54.12 | JT | enjay5150: now, back up your claim "isdn is a shit system" |
| 22:54.17 | enjay5150 | you're clearly a prick.. |
| 22:54.25 | JT | clearly you're new here |
| 22:54.40 | JT | and you're the one refusing to back up your claim |
| 22:54.44 | JT | i am curious |
| 22:54.48 | JT | what the reasoning is |
| 23:00.05 | rvhi0 | hi, in agi, if cmd dial is used, the script is over. How do I take back the control of call, e.g. redirect it to vm and etc. |
| 23:05.19 | JT | enjay5150: i would still appreciate if you could tell me why isdn is a shit system |
| 23:07.30 | Juggie | rvhi0, using dial directally in agi works but also sucks. |
| 23:07.40 | Juggie | because of what you just said |
| 23:08.07 | Juggie | may i suggest you set the extension/context/priority using agi, and whatever other variables you need |
| 23:08.15 | Juggie | then kill your agi returning control to the asterisk dialplan |
| 23:08.46 | Juggie | let the dialplan handle the dial, and then you can either re-run your agi again (passing in a paremeter of course) or just run vm from the dialplan |
| 23:09.48 | Juggie | dial with the g paremether might work from agi but i am not 100% sure |
| 23:12.58 | De_Mon | see anything wrong with this retrydial command for asterisk 1.4? -- dialplan add extension 31491,1,RetryDial(|1|3|SIP/bandwidth/+15551231234||m) into local |
| 23:13.31 | De_Mon | show application retrydial kinda lists a different set of arguments :] |
| 23:14.30 | Juggie | frankly i'm not sure why it even exists |
| 23:14.34 | Juggie | to be quite honest |
| 23:14.45 | De_Mon | no, i guess thats right |
| 23:15.19 | Juggie | i dont see the point of doing something in code you can do in like <5 lines of dialplan code |
| 23:15.25 | Juggie | or a macro |
| 23:16.05 | De_Mon | macros rock |
| 23:16.45 | De_Mon | Juggie how do you feel about ael? |
| 23:17.06 | Juggie | i dont use it because i already know the dialplan |
| 23:17.31 | Juggie | ael doesnt give any extra functionality, and asterisk just converts it to dialplan before it loads it anyways |
| 23:17.37 | Juggie | so anything you can do in ael you can do in dialplan |
| 23:17.46 | Juggie | it might look simpler but its funationally the same |
| 23:17.55 | De_Mon | i totally freaked out when I acidentally edited extensions.ael instead of extensions.conf |
| 23:18.04 | Juggie | so, to answer your question i havnt bothered :) |
| 23:18.12 | Juggie | thus i have no opinion :) |
| 23:18.40 | De_Mon | it convert it to dialplan eh? one less layer of abstraction! |
| 23:19.40 | *** join/#asterisk vn (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 23:28.35 | Juggie | De_Mon, yes.. that doesnt mean you shoudnt write in ael, and that someday it wont be directally read in, but it isnt as it stands right now |
| 23:31.42 | snuff-away | mm i've not yet been tempted by ael.. just because it seems even more hacked up than a straight extensions.conf |
| 23:31.46 | snuff-away | ;) |
| 23:35.42 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 23:36.10 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 23:36.49 | *** join/#asterisk wunderkin (n=wunderki@dslstat-bvi4-344.fastq.com) |
| 23:37.46 | ManxPower | I take it the mailing lists are down? |
| 23:38.34 | _DAW | I understand they are being fixed. |
| 23:38.36 | De_Mon | http://pastebin.ca/615166 ; broken retrydial attempt |
| 23:38.37 | De_Mon | http://pastebin.ca/615165 ; compared with a working retrydial attempt |
| 23:39.05 | De_Mon | The only thing ive changed between these retrydials is the number dialed.. why oh why do I get different responses? |
| 23:39.56 | ManxPower | De_Mon: Have you made sure there is a blank line at the end of the .call file? |
| 23:40.10 | De_Mon | its not a call file |
| 23:40.15 | ManxPower | There was a bug fixed recently in 1.2 that caused retry to not work under some situations |
| 23:40.20 | ManxPower | ah, nevermind then |
| 23:40.21 | De_Mon | this is 1.4.6 |
| 23:40.38 | De_Mon | http://pastebin.ca/615166 ; broken retrydial attempt |
| 23:40.49 | De_Mon | damn copy and paste |
| 23:40.50 | De_Mon | RetryDial("SIP/office2-08259b10", "|1|3|SIP/bandwidth/+15551231234||m") |
| 23:41.26 | ManxPower | I would be most interested in the value of HANGUPCAUSE and DIALSTATUS |
| 23:41.35 | ManxPower | Are you SURE your carrier wants a + at the beginning of the number? |
| 23:41.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:41.45 | De_Mon | do I have to noop those variables? |
| 23:41.49 | ManxPower | normally "+" means "use your toll prefix. |
| 23:41.52 | De_Mon | ManxPower yeah.. |
| 23:41.54 | ManxPower | De_Mon: yes, noop |
| 23:43.46 | ManxPower | _DAW: Just making sure. |
| 23:43.50 | *** join/#asterisk vn- (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 23:44.00 | vn- | is it me or the AC of the spa2102 sucks? |
| 23:44.07 | _DAW | I saw bkruse talking about it earlier. |
| 23:44.08 | vn- | looks like its inverted... |
| 23:45.44 | *** join/#asterisk wunderkin- (n=wunderki@dslstat-bvi4-344.fastq.com) |
| 23:45.50 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-104-93-53.pskn.east.verizon.net) |
| 23:45.52 | ManxPower | NotHere: try a coherent description of the problem |
| 23:46.02 | De_Mon | ManxPower err... when retrydial fails it throws some warnings about a nonexistant announce file and doesn't process the 2nd priority |
| 23:46.15 | _DAW | sounds somewhat schizophrenic |
| 23:46.17 | NotHere | ManxPower, I think I did before, but let me try again |
| 23:47.00 | ManxPower | De_Mon: try adding "g" as a dial option |
| 23:47.50 | *** join/#asterisk lmoreira (n=lmoreira@201008244102.user.veloxzone.com.br) |
| 23:48.19 | NotHere | I call my asterisk server and the phone that is connected to it, rings and correct caller id is shown. I answer the phone, asterisk console acknowledges it, but the caller hears just noise and the called hears nothing |
| 23:48.52 | ManxPower | NotHere: you are calling from your cell phone, PSTN land line, voip carrier, or telepathy/PSTN gateway |
| 23:49.12 | ManxPower | the call comes into Asterisk on SIP, IAX, T-1, or POTS? |
| 23:49.13 | NotHere | ManxPower, I have tried both my cell phone and a pstn land line |
| 23:49.18 | De_Mon | ManxPower already did, no change |
| 23:49.21 | De_Mon | dialplan add extension 31491,1,RetryDial(|1|3|SIP/bandwidth/+18666197109||mg) into local replace |
| 23:49.35 | JT | ManxPower: no, even better, analogue port from an ISDN NT1 |
| 23:49.38 | De_Mon | dialplan add extension 31491,2,NoOp("${HANGUPCAUSE} and ${DIALSTATUS}") into local |
| 23:49.47 | NotHere | ManxPower, the call comes in from POTS. A TDM400 cards FXO module |
| 23:49.48 | ManxPower | JT: That is too kinky even for me. |
| 23:49.56 | JT | hehe |
| 23:49.57 | lmoreira | I running Asterisk 1.2.20 and facing deamon crash. Please, take a look on log http://pastebin.ca/615176 |
| 23:50.01 | De_Mon | its an ITSP if you hadn't noticed |
| 23:50.13 | NotHere | ManxPower, and the internal phone is connected to a FXS module on the same card |
| 23:50.17 | ManxPower | NotHere: then you need to start looking at your TDM card. Check for interrupt conflicts "cat /proc/interrupts" |
| 23:50.27 | ManxPower | NotHere: confirm you have POWER plugged into the card. |
| 23:51.05 | NotHere | ManxPower, I guess the phone wouldn't ring if I hadn't connected the power correctly, would it? |
| 23:51.26 | lmoreira | Asterisk crash about 4 times a day. |
| 23:51.34 | ManxPower | NotHere: The power connector is there for a reason (well for FXS, at least) |
| 23:51.44 | ManxPower | lmoreira: then use a different version of Asterisk |
| 23:52.00 | De_Mon | okay... that I think is a bug |
| 23:52.02 | De_Mon | check this out |
| 23:52.08 | lmoreira | The crash is happing since 1.2.18 |
| 23:52.11 | russellb | lmoreira: that log says nothing about crashes |
| 23:52.20 | lmoreira | see at the and |
| 23:52.29 | NotHere | ManxPower, the same thing happens if I configure asterisk to auto answer the phone and say some number. So it's not the FXS or its power the problem |
| 23:52.42 | russellb | lmoreira: there is nothing at the end, either |
| 23:52.48 | ManxPower | NotHere: and interrupts? |
| 23:52.55 | lmoreira | Something with cdr.c and what is this acl.c debug?? |
| 23:53.19 | russellb | that is debug output, and does not indicate any type of error |
| 23:53.22 | ManxPower | lmoreira: try the latest 1.2.x. I had crashes with 1.2.17/1.2.18 |
| 23:53.22 | russellb | especially a crash |
| 23:53.42 | ManxPower | or try 1.2.15 (but that has other bugs, but they did not impact my usage) |
| 23:53.57 | NotHere | ManxPower, all seem ok. zttool doesn't report misses and /proc/interrupts indicates that the card is sharing the irq with just one usb bus (and I don't have any usb connected) |
| 23:53.58 | De_Mon | dialplan add extension 31491,2,NoOp("${HANGUPCAUSE} and ${DIALSTATUS}") into local |
| 23:54.00 | JT | NotHere: is the power connector connected? |
| 23:54.02 | De_Mon | grr |
| 23:54.07 | De_Mon | -- Executing [31491@local:2] NoOp("SIP/office2-0826b310", "34 and |
| 23:54.07 | De_Mon | CONGESTION") in new stack |
| 23:54.11 | lmoreira | russellb: the deamon crash and safe_asterisk reload the service. |
| 23:54.23 | russellb | lmoreira: then you need to get a backtrace |
| 23:54.34 | russellb | see doc/backtrace.txt |
| 23:54.42 | russellb | start asterisk with -g |
| 23:54.50 | russellb | then, gdb /usr/sbin/asterisk core.12345 |
| 23:54.52 | lmoreira | russellb: I'll. |
| 23:54.53 | russellb | (gdb) bt |
| 23:54.56 | russellb | (gdb) bt full |
| 23:55.43 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:55.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:55.45 | ManxPower | De_Mon: I suspect your carrier is just running out of ports. I get that even on PSTN telcos w/PRI. |
| 23:55.58 | ManxPower | I have my dialing macros just wait 1 and retry the dial |
| 23:56.15 | De_Mon | ManxPower the funny thing is I can dial the same number using Dial and it never has this problem |
| 23:56.16 | lmoreira | russellb: the command gdb is not working |
| 23:56.24 | *** join/#asterisk vn (n=vn@modemcable083.222-80-70.mc.videotron.ca) |
| 23:56.28 | De_Mon | however, it ALWAYS has the problem when I use retrydial |
| 23:56.35 | NotHere | ManxPower, I should also say that my external analog line comes from an isdn nt. This same port is acknowledged to be working perfectly when a normal analog phone is connected to it |
| 23:57.00 | ManxPower | De_Mon: have you tried a different carrier |
| 23:57.37 | russellb | lmoreira: then you need to install it :) |
| 23:57.50 | De_Mon | not really you think they are out of ports for that particular calling area? |
| 23:58.05 | De_Mon | but i would expect the same behavior from DIAL if that were the issue... |
| 23:58.09 | De_Mon | let me try another carrier |
| 23:58.54 | *** join/#asterisk [[blah]asdf (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 23:59.00 | JT | De_Mon: you could always do a sip debug to see what is ACTUALLY happening |
| 23:59.03 | ManxPower | De_Mon: All ITSPs suck. |
| 23:59.20 | ManxPower | JT: He did do a SIP debug. I hate reading those damn things |
| 23:59.23 | NotHere | could it be that I have a faulty card? |
| 23:59.55 | [[blah]asdf | hey all... I have a digium te405p that i am selling. I have dropped the price to $500. its only been in use for about 1 month. works fine. changed to sip provider and no longer need it. any takers? |