00:00.07 | bkw_ | install openssl |
00:00.11 | bkw_ | thats a simple one |
00:00.16 | devnu112 | heh |
00:00.38 | bkw_ | and if you use packages install openssl-dev too |
00:00.40 | tmarsh | i have it installed but am still getting the error that was the first thing i check |
00:01.03 | docelm0 | I created a simple agi... And when it runs it says it ran but it didnt execute it. Can someone tell me why it didnt do anything? |
00:02.25 | bkw_ | run ldconfig |
00:03.06 | docelm0 | who me? |
00:03.46 | syslod | tmarsh- if ldconfig doesn't work your libs path is wrong. |
00:03.57 | bkw_ | correct |
00:05.26 | sudoer | is there a way to randomly dial out from a different line like either dial through zaptel or voip or different provider |
00:05.47 | Q-At-Home | bbiab, must feed |
00:08.14 | tmarsh | the ldconfig looks like it worked thanks |
00:08.14 | bkw_ | sure |
00:08.50 | Chuji | docelm0: what is your agi written in? |
00:09.04 | bkw_ | ewww agi |
00:09.07 | bkw_ | how like last year |
00:09.23 | tmarsh | i spoke too soon same error |
00:09.38 | bkw_ | path is wrong then |
00:09.41 | bkw_ | look for libssl* |
00:09.48 | sudoer | me? |
00:09.50 | Chuji | bkw_ : When are you guys going to document res_perl better |
00:09.56 | Chuji | intimidating for newbies |
00:10.07 | Chuji | That's why they go to AGI first |
00:10.20 | tmarsh | how do i check the paths i did a default install based on asterisk docs? |
00:10.24 | Chuji | mod_perl has good documentation |
00:10.32 | bkw_ | Chuji when someone documents it |
00:10.35 | bkw_ | haha |
00:10.38 | Chuji | :P |
00:10.39 | bkw_ | Chuji give it time |
00:11.12 | rollotomnasi | tmarsh look in /etc/asterisk/asterisk.conf |
00:11.24 | bkw_ | what the hell would that have to do with his ldpath |
00:12.03 | rollotomnasi | bkw_ absolutely nothing. should have scrolled. going to go crawl away now. :p |
00:12.03 | *** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net) |
00:12.07 | bkw_ | haha |
00:12.12 | *** join/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net) |
00:12.15 | bkw_ | haha |
00:12.16 | justinnnnnn | someone how do i mysqldump all the tables |
00:12.17 | bkw_ | ok |
00:12.23 | rollotomnasi | hehe |
00:12.24 | justinnnnnn | like the hole database ?? |
00:12.27 | bkw_ | justinnnnnn just shut it down and copy it all |
00:12.33 | justinnnnnn | how so ? |
00:12.45 | Chuji | /join #mysql |
00:12.51 | *** join/#asterisk iMediax (~user@00d0a8003aa3.click-network.com) |
00:12.56 | bkw_ | the /var/lib/mysql/DIRNAME |
00:13.02 | justinnnnnn | ooh col |
00:13.02 | bkw_ | or something like that |
00:13.03 | justinnnnnn | thanks :)) |
00:13.04 | bkw_ | just lock em all and back it up |
00:13.13 | bkw_ | then unlock the tables or shut it down and backup.. |
00:15.10 | *** join/#asterisk Darwin35 (~darwin35@pool-68-162-159-172.pitt.east.verizon.net) |
00:15.13 | *** join/#asterisk z_smurf (~z_smurf@jamlikheten-77-34.ip-pluggen.com) |
00:18.50 | tmarsh | ok i am reay missing something cause the /etc/asterisk/ diretory doesnt exist why would this be i followed the new version install doc? |
00:19.13 | bkw_ | ignore that |
00:19.20 | bkw_ | you haven't done a make install |
00:19.24 | bkw_ | go get openssl and install it by hand |
00:19.27 | bkw_ | problem solved |
00:20.55 | pfn | what requires openssl |
00:20.58 | pfn | oh, the rsa stuff |
00:21.04 | blacksmall | is it possible to have a softphone and * on the same pc? |
00:21.28 | pfn | blacksmall it takes a bit of clue, but it's possible |
00:21.41 | blacksmall | pfn> want to start me off somewhere? |
00:21.47 | pfn | no |
00:21.52 | blacksmall | why not? |
00:22.23 | bkw_ | because anyone with a right mind wouldn't do it |
00:22.24 | pfn | because at this point, it's an exercise left up to you |
00:23.36 | blacksmall | wouldn't do it? why? I just want to try to connect to asterisk |
00:23.48 | pfn | because it isn't a recommended configuration |
00:23.58 | pfn | and we aren't going to waste our time trying to encourage it |
00:24.04 | bkw_ | zactly |
00:24.31 | pfn | if you have specific questions/help that you can't answer by reading the wiki, docs, whatever, we can help |
00:24.37 | pfn | but beyond that, we aren't here to handhold |
00:24.53 | bkw_ | *HINT* run each thing on a diffrent port :P |
00:24.56 | syslod | Anyone using * in a large CLEC or large pbx enviroment? |
00:27.10 | bkw_ | brb |
00:29.07 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
00:30.13 | blacksmall | thank you. enough to go on. |
00:31.43 | *** join/#asterisk brc_ (~john@brc.base.supporter.pdpc) |
00:33.42 | tmarsh | ok here is what sequence i ran and i still get the error at the make stage cd /usr/src/asterisk make clean make |
00:34.00 | bkw_ | go install openssl from src |
00:34.03 | bkw_ | www.openssl.org |
00:34.29 | tmarsh | ill try that again but have done that twice |
00:34.36 | bkw_ | installed from src? |
00:34.41 | bkw_ | cat /etc/ld.so.conf |
00:34.50 | pfn | pastebin your make error |
00:35.06 | bkw_ | add the pat to ld.so.conf to your libs |
00:35.10 | bkw_ | maybe /usr/local/lib ? then ld-config |
00:35.14 | bkw_ | er ldconfig |
00:35.31 | tmarsh | include ld.so.conf.d/*.conf |
00:35.51 | bkw_ | what distro do you run? |
00:36.42 | pfn | where'd you install openssl? |
00:37.19 | pfn | -lssl has nothing to do with ld.so |
00:38.00 | tmarsh | i ran cat and i get /usr/X11R6/lib /usr/lib/qt-3.3/lib |
00:38.24 | tmarsh | fedora cat 2 |
00:38.25 | pfn | where'd you install openssl? |
00:39.23 | bkw_ | pfn yes it would if it can't find it |
00:39.37 | pfn | no, ld.so does not determine search path for linking |
00:39.38 | bkw_ | if it can't find the libssl stuff then you be fucked |
00:39.50 | pfn | for linking, you must specify -Lpath |
00:40.00 | pfn | e.g. -L/usr/local/openssl/lib -lssl |
00:40.07 | bkw_ | if you say so |
00:40.14 | pfn | of course, runtime is another matter |
00:40.31 | bkw_ | if it were in ld.so.conf it owuld work |
00:40.33 | pfn | if ld.so cannot find it at runtime, then ld.so.conf must be fixed |
00:40.42 | pfn | unless binutils changed, it shouldn't.... |
00:40.45 | bkw_ | oh ya |
00:40.47 | pfn | binutils should never look at ld.so.conf |
00:40.47 | bkw_ | you might be right |
00:40.53 | bkw_ | its been a long time since I have fucked with that |
00:41.46 | tmarsh | here it is /usr/bin/openssl |
00:42.37 | bkw_ | no |
00:42.38 | bkw_ | the libs |
00:42.52 | tmarsh | fedora 2 |
00:42.55 | bkw_ | locate libssl |
00:43.25 | pfn | you haven't installed openssl-devel |
00:43.27 | pfn | yum openssl-devel |
00:43.38 | bkw_ | er ago |
00:43.44 | pfn | sounds like he didn't do it |
00:46.49 | tmarsh | fedora 2 |
00:47.03 | *** join/#asterisk flewid (~flewid@CPE0050ba8c9a95-CM000f9fac6da2.cpe.net.cable.rogers.com) |
00:48.25 | *** join/#asterisk nassy (~mark@24-193-231-136.nyc.rr.com) |
00:48.49 | *** part/#asterisk z_smurf (~z_smurf@jamlikheten-77-34.ip-pluggen.com) |
00:49.04 | bkw_ | yes yes we get you're using fedora core 2 |
00:50.15 | pfn | then gonna head over to the mall and look at crappy mall diamonds |
00:50.15 | pfn | heh |
00:50.29 | pfn | so I can be thatmuch more impressed with the diamond I ordered that's coming next week |
00:51.35 | bkw_ | haha |
00:51.44 | bkw_ | you mean the glass you paid 20000% for |
00:54.23 | tzanger | pfn getting engaged? |
00:57.52 | *** join/#asterisk Cheng29 (~cheng29@d57-83-90.home.cgocable.net) |
01:00.16 | pfn | tzanger already engaged, just haven't gotten the ring, heh |
01:00.28 | pfn | well, not officially, but same difference |
01:00.29 | tzanger | pfn congrats :-) |
01:00.39 | pfn | thanks |
01:00.47 | pfn | bkw nah, mall shit sucks |
01:03.10 | pfn | getting a half-carat, ags0, hearts & arrows, e, si1 |
01:12.22 | czero | any of you using cisco to terminate pri's then SIP over to asterisk? |
01:14.10 | syslod | czero - we tried that but ended up just getting a PRI card for asterisk. Much simpler. |
01:15.01 | iMediax | been awhile.. meetme needs zap for timing correct? |
01:15.29 | syslod | Yep or at least ztdummy but I haven't seen that work. |
01:15.52 | iMediax | oh ya ztdummy |
01:16.39 | syslod | If you get it to work let me know. I have a few test systems we tried ztdummy but no luck. |
01:17.28 | C4thY | we use ztdummy |
01:17.59 | syslod | :) any pointers on getting it to work/compile? |
01:18.11 | C4thY | we used all 1.0.0 versions |
01:18.40 | C4thY | asterisk zapata and zaptel |
01:20.13 | sudoer | is it normal for fxs cards to have ports that fit ethernet? |
01:20.22 | *** join/#asterisk hades_ (~hades_@200-180-177-229.paemt7006.dsl.brasiltelecom.net.br) |
01:20.30 | sudoer | i got my first fxs tdm card and the ports are ethernet size |
01:21.20 | cypromis | yes |
01:21.32 | cypromis | at least digium fxs modules are all rj45 |
01:21.44 | bkw_ | you can plug rj11's into that |
01:22.05 | sudoer | yes, its digium |
01:22.41 | sudoer | if its a 4 port fxf, do i need to plug power into the power hole on the card |
01:22.42 | file[laptop] | bkw_: should I buy this workstation? |
01:22.48 | sudoer | fxs i mean |
01:22.59 | sudoer | file[laptop] i want to see link too |
01:23.12 | file[laptop] | http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=11218&item=5723438330&rd=1 |
01:23.47 | sudoer | looks nice |
01:24.39 | bkw_ | file haha I can't copy it |
01:25.20 | file[laptop] | it's all like... nifty! |
01:25.25 | bkw_ | ok |
01:25.38 | file[laptop] | of course I'd change it some |
01:25.38 | bkw_ | I wouldn't buy it |
01:25.49 | pfn | xeon 1.0? |
01:25.51 | pfn | yuck |
01:26.18 | file[laptop] | I don't need a new workstation anyway |
01:26.23 | bkw_ | I would rather have a Sun Ultra |
01:26.27 | pfn | it'd be cool if it had a nice case |
01:26.27 | bkw_ | 10 or something |
01:26.29 | file[laptop] | but it's fun to look at everything |
01:26.32 | pfn | ultra10 is ultra slow |
01:26.35 | bkw_ | ya that case is ass |
01:26.40 | bkw_ | pfn slow but solid |
01:26.46 | file[laptop] | ultra160 mmm |
01:26.47 | *** part/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
01:26.47 | pfn | if I wanted an ultra, have to go pizza box, ultra5 |
01:27.03 | bkw_ | ultra5's are great just don't expect to compile alot on them |
01:27.04 | bkw_ | haha |
01:27.22 | pfn | I've always liked sun pizzaboxes |
01:27.26 | file[laptop] | people shouldn't be allowed to visit eBay |
01:27.34 | file[laptop] | I find stuff that I want, but that I don't need |
01:27.58 | pfn | $100, netbooted off of my linux box and worked ok as a 8bpp display |
01:28.16 | pfn | cheaper than getting a multisync 21" monitor at the time, heh |
01:29.26 | bkw_ | not a bad deal |
01:29.32 | bkw_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51107&item=5129693351&rd=1 |
01:30.56 | pfn | that sounds a bit on the expensive side |
01:31.03 | pfn | they were going for under $250 like after the .com crash |
01:31.30 | bkw_ | I would love a sunfire |
01:31.32 | bkw_ | haha |
01:31.39 | tmarsh | thanks all that did the trick i reinstalled openssl-dev i was surprised that locate shoed it installed but it didnt work again thanks.. |
01:31.59 | pfn | locate didn't show it installed, I bet :p |
01:32.02 | pfn | rpm -qa | grep openssl |
01:32.07 | bkw_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=20328&item=5724909140&rd=1 |
01:32.07 | pfn | or whatever the short command to that is... |
01:32.23 | pfn | rpm -q openssl-devel woulda shown it not |
01:32.51 | bkw_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51238&item=5724018528&rd=1 <-- THEY ARE SMOKIN CRACK |
01:33.10 | pfn | hah |
01:33.24 | devnu112 | mmmm, crack |
01:33.26 | devnu112 | errr, wait |
01:33.29 | devnu112 | heh |
01:33.54 | bkw_ | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=20328&item=5724856760&rd=1 |
01:34.41 | Chuji | Does asterisk have problems with variables with spaces in them? |
01:34.46 | file[laptop] | why must I have access to Google... |
01:35.00 | czero | syslod when u tired cisco: do you have much cisco experiece? |
01:35.08 | Chuji | I'm having a hard time populating setcidname with a variable that has a space |
01:35.19 | Chuji | it's truncating after the space |
01:35.52 | Chuji | exten => s,2,SetCIDName,${rf_calleridname} |
01:36.15 | Chuji | I've tried it in ()'s too |
01:36.20 | Chuji | any idea? |
01:36.42 | sudoer | can you have cdr do cdr to 2 places at same time? |
01:36.48 | sudoer | like database and to file? |
01:37.00 | pfn | yes |
01:37.14 | Chuji | sudoer : you can pipe the file to a database too |
01:37.18 | Chuji | that is what I do |
01:37.24 | syslod | czero - We have used cisco since 1997. "Back in the day" i guess. We pretty much have eliminated cisco from our archetecture. We have a class 5 switch, DAX, and on most DS1 level connects we now use Adtran. |
01:37.32 | pfn | cdr logs to every cdr module that you have loaded |
01:37.35 | *** join/#asterisk cosmic_ (~yes@ns.serversoft2k.com) |
01:37.39 | pfn | if you have cdr_csv and cdr_pgsql loaded, it'll log to both |
01:38.12 | *** join/#asterisk hcir (~ar@209-193-42-30-cdsl-rb1.sit.acsalaska.net) |
01:38.38 | czero | Syslod: I'll tkae that as a yes then :) I've beenusing IOS hardcore since 96/97 jsut figured the more I coudl do in IOS the better since while I know linux routers are more my thing |
01:38.39 | pfn | ok, I think I'm gonna go cruise the mall now |
01:38.47 | Darwin35 | I can get spark 5 at 20 bucks each at the local computer recycle warehouse here |
01:39.15 | docelm0 | ok Anyone in here know anything about AGI? I have a couple questions.. Namely like why mine will not work.. |
01:39.44 | syslod | czero - What application do you have passing PRI to cisco to SIP to *? |
01:40.00 | *** join/#asterisk Kb1_Kanob (~johnsmith@border.scrd.bc.ca) |
01:40.08 | czero | Syslod I'm not doing it yet |
01:40.31 | czero | I though about useing a t1 win in a 36XX |
01:40.36 | czero | then SIP to * |
01:40.59 | bkw_ | sun blade 100's aren't that expensive |
01:41.33 | syslod | We tested with SIP to * but once we started our R&D we just bought the PRI card. We have used Dialogic and Digium. Both work well and are simple to setup. |
01:41.47 | docelm0 | Can someone help me?!?!? Please.. |
01:42.02 | *** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com) |
01:42.16 | czero | yeah the digum stull is stright forward, I'm thikning about scaling and with IOS I'll get my 729 licences in IOS |
01:42.40 | docelm0 | When I call the extension that runs the AGI it says it runs but it doesnt do anything its designed for.. |
01:43.45 | Chuji | docelm0: I tried to help you earlier |
01:43.48 | Chuji | you didn't answer |
01:43.56 | Chuji | what is the AGI written in? |
01:44.35 | docelm0 | sorry.. doing 1000 things.. |
01:44.38 | docelm0 | PHP |
01:44.50 | docelm0 | Im trying to run the sample.agi on the * AGI PHP page.. |
01:44.51 | Chuji | Are you just wanting it to return a value to the dialplan? |
01:45.09 | docelm0 | Im just wanting to get it to work.. |
01:45.56 | Chuji | give me a link to the AGI you are speaking of |
01:46.44 | syslod | czero - Are you a carrier or just doing private trunking? |
01:47.35 | bkw_ | doh |
01:48.06 | czero | I'm going to be sellign serivce but come form a tier 1 carrier background |
01:48.23 | czero | so I always think long term scalability |
01:48.33 | syslod | PSTN service? |
01:48.53 | czero | I work for iMCI during hte growth years :) |
01:51.09 | syslod | We are trying 8:1 oversubscription on our * projects with 200 user limit per system. This is the testing phase and all looks well. Our exsisting gateways are upwards of 8000 users per slot in a chassis. We really don't know what the limits are yet. |
01:51.56 | Wi_Fi | hey guys |
01:52.03 | Wi_Fi | i might need help |
01:52.04 | syslod | Hello. |
01:52.10 | *** join/#asterisk brc_ (~john@brc.base.supporter.pdpc) |
01:52.38 | czero | I've got data on over sub rates from big telco's but not sure what point you have to get to before they hold true |
01:53.07 | czero | syslod r u US based I assume? |
01:53.32 | syslod | Yes US based. Our data services are something like 1 T1 per 200 subs. |
01:54.28 | czero | you should be able to scale much higher then that past a couple of t1's |
01:55.16 | czero | but thats the number I use myself for planning :) |
01:55.45 | syslod | bbl |
01:55.49 | czero | later |
01:56.27 | *** join/#asterisk wellington (wellington@201008211120.user.veloxzone.com.br) |
01:56.48 | *** join/#asterisk mitcheloc (trilluser@69-169-54-180.anhmca.adelphia.net) |
01:57.06 | file[laptop] | all in favor that I upgrade my wifi network to g say aye |
01:57.18 | czero | aye :) |
01:57.33 | file[laptop] | I could upgrade my laptop too |
01:57.43 | czero | I did my QAP last month |
01:57.50 | Chuji | file: What for? |
01:57.51 | czero | gonna start he devices this month |
01:58.01 | file[laptop] | Chuji: because! I want a router that runs Linux |
01:58.01 | Chuji | file: What's wrong with B? |
01:58.11 | Chuji | ohh, ic |
01:58.26 | czero | yeah u can hack the G routers |
01:58.34 | file[laptop] | what's the cheapest 802.11g router that runs Linux |
01:59.26 | trogs | probably a linksys wrt54g |
01:59.46 | file[laptop] | trogs: tried that already... |
01:59.48 | file[laptop] | D-Link is cheaper |
01:59.51 | mitcheloc | pfn are you around? |
01:59.53 | file[laptop] | same hardware internally though... |
01:59.58 | trogs | which one? |
02:00.13 | file[laptop] | DI-624 |
02:00.17 | trogs | oh ok |
02:00.45 | file[laptop] | hrm it does 108Mbps |
02:01.00 | PatrickDK | I have 3 di624 |
02:01.05 | PatrickDK | and 6 wrt54gs |
02:01.06 | file[laptop] | PatrickDK: how are they? |
02:01.11 | PatrickDK | I like the wrt54gs better |
02:01.22 | PatrickDK | hmm, one di624 is fine |
02:01.29 | PatrickDK | other one crashs under heavy use |
02:01.35 | mitcheloc | anyone know why..after my ip changes, broadvoice bugs out b/c asterisk doesn't re-register fast enough |
02:01.36 | file[laptop] | different revisions? |
02:01.38 | PatrickDK | other one hasn't been used enough by me to know |
02:01.41 | mitcheloc | i'm trying to get it to re register |
02:01.47 | PatrickDK | file, all are B I think |
02:01.50 | trogs | the linksys have rp-tnc connectors though, as opposed to the annoying sma's of the dlinks |
02:01.56 | mitcheloc | and the register => user:pass@server command doesn't do it ? |
02:02.01 | file[laptop] | interesting |
02:02.03 | PatrickDK | I perfer the sma |
02:02.11 | file[laptop] | PatrickDK: does the D-Link run Linux? |
02:02.23 | PatrickDK | hmm, probably |
02:02.27 | trogs | the sma is probably ok, if you don't want to make your own cables :) |
02:02.30 | file[laptop] | nmap one and see :) |
02:02.36 | PatrickDK | make own cables? for what? |
02:02.49 | trogs | antennas |
02:03.00 | PatrickDK | I have alot of antennas with rp-sma |
02:03.14 | PatrickDK | they are actually more common than rp-tnc |
02:03.26 | trogs | mine are usually N type, and i have a bunch of N -> rp-tnc |
02:03.34 | mitcheloc | guys? it's a quick question, how do i use register => user@server in sip.conf to get asterisk to re-register every say 60 seconds |
02:03.57 | bkw_ | go read the sip.conf.sample |
02:04.02 | PatrickDK | N is good, though, not at high freq |
02:04.09 | PatrickDK | sma is high freq version of N |
02:04.12 | trogs | ;defaultexpirey=120 ; Default length of incoming/outoing registration |
02:04.12 | mitcheloc | bkw i did |
02:04.16 | trogs | i assume thats what you want |
02:04.22 | mitcheloc | it says register can use a defined peerid |
02:04.24 | mitcheloc | doesn't work for me |
02:04.50 | bkw_ | defaultexpirey=120 ; Default length of incoming/outoing registration |
02:04.55 | bkw_ | you are blind then |
02:05.05 | *** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net) |
02:05.59 | mitcheloc | no bkw, thats not what i'm talkin about |
02:06.10 | mitcheloc | i think i have the syntax wrong this: register? => <username>@<sip client/peer id in sip.conf>/<extension> :Register with a SIP provider |
02:06.11 | bkw_ | I think it is |
02:06.14 | mitcheloc | thats on the wiki |
02:06.25 | mitcheloc | how do i say register => peercontext |
02:06.37 | mitcheloc | it won't use it to register remotely |
02:06.40 | bkw_ | you don't |
02:06.52 | bkw_ | what is your goal? |
02:06.58 | mitcheloc | i have it all working now |
02:07.03 | mitcheloc | but my house has a dynamic ip |
02:07.05 | mitcheloc | and when it changes |
02:07.08 | mitcheloc | i can't call out BV |
02:07.09 | file[laptop] | bkw_: Ever get that NSLU |
02:07.10 | file[laptop] | ? |
02:07.15 | bkw_ | file SOON |
02:07.20 | file[laptop] | nifty |
02:07.20 | mitcheloc | b/c it loses my registration/ip relationship |
02:07.27 | bkw_ | find a better ISP |
02:07.27 | mitcheloc | so it needs to re-register to them |
02:07.32 | bkw_ | but 60 seconds/ |
02:07.35 | bkw_ | your ip changes that fast? |
02:07.39 | mitcheloc | no every 12 hours |
02:07.42 | bkw_ | DAMN |
02:07.50 | mitcheloc | i go to sleep, wake up, and i have to reload in asterisk to use my phones |
02:07.51 | bkw_ | defaultexpirey would do it then |
02:07.52 | mitcheloc | it sucks |
02:08.12 | mitcheloc | ok, i got that, just don't get what in the context tells asterisk to go out and register to broadvoice? |
02:08.18 | mitcheloc | or defaultexpirey does that automatically |
02:08.29 | mitcheloc | oh |
02:08.31 | mitcheloc | got it |
02:08.32 | bkw_ | yep |
02:08.33 | mitcheloc | it's for general |
02:08.33 | bkw_ | it would |
02:08.40 | mitcheloc | i was thinking it goes under a sip context |
02:08.50 | bkw_ | haha this is why I said look at sip.conf.sample |
02:09.08 | mitcheloc | yep got ya, just didn't make sense to me *sorry* |
02:09.20 | bkw_ | asterisk's peer/user matching in chan_sip is leaves alot to be desired |
02:10.33 | mitcheloc | yep but it works, i kept expecting somehow to make register point to a defined context and use that |
02:10.45 | bkw_ | no |
02:10.53 | bkw_ | you do [peer] and host=ip |
02:11.01 | bkw_ | or [user] |
02:11.09 | bkw_ | but its better to match on ip to an entry |
02:11.23 | Godsey | I'm playing w/ DISA |
02:11.25 | mitcheloc | i got it |
02:11.36 | bkw_ | the best way to learn it is ot look at find_user and find_peer I thin kin chan_sip |
02:11.39 | Godsey | it seems no matter what I do, after I get second dialtone I can't dial :) |
02:11.42 | mitcheloc | disas fun ;) |
02:11.47 | mitcheloc | in the code? |
02:11.48 | Godsey | after 1 or 2 numbers I get error tone |
02:11.57 | bkw_ | yes in the code |
02:12.03 | bkw_ | best place to learn how something works |
02:12.03 | mitcheloc | heh i don't know C |
02:12.10 | bkw_ | I didn't either when I started witha sterisk |
02:12.14 | mitcheloc | i could look though, but i wouldn't understand it |
02:12.29 | bkw_ | know php/ |
02:12.30 | bkw_ | or per? |
02:12.32 | bkw_ | er perl? |
02:12.42 | file[laptop] | is 802.11g worth it... |
02:12.46 | bkw_ | file yes |
02:12.49 | mitcheloc | php, c#, vb,asp, html, java, those stuff, yea i know it's someone similar |
02:12.56 | bkw_ | then C isn't that far off |
02:12.58 | bkw_ | you'll have it down |
02:13.04 | mitcheloc | bkw, how much would a decent webconfig utility be useful for everyone? |
02:13.04 | file[laptop] | it isn't, I know it isn't |
02:13.05 | bkw_ | my first C project was cdr_odbc |
02:13.07 | mitcheloc | that fully configs asterisk? |
02:13.35 | bkw_ | mitcheloc learn the new realtime stuff |
02:13.46 | bkw_ | brb |
02:13.52 | mitcheloc | new realtime? can you link me somewhere? |
02:15.14 | Godsey | might any of you use DISA? :) |
02:15.33 | mitcheloc | i do |
02:15.37 | mitcheloc | well i've set it up |
02:15.38 | mitcheloc | it's easy though |
02:15.43 | mitcheloc | nothing to it? |
02:15.48 | Godsey | it seems like it should be! |
02:15.49 | Godsey | :) |
02:16.06 | Godsey | I do 1,Answer()... 2,DISA,1234|trunkld |
02:16.20 | Godsey | i dial the #, dial 1234# and get second dialtone |
02:16.29 | mitcheloc | i don't think you pass disa parameters |
02:16.29 | Godsey | I enter any digit and get busy |
02:16.31 | mitcheloc | just go DIsa |
02:16.31 | Godsey | (single) |
02:16.36 | mitcheloc | and it gives the user a dialtone |
02:16.38 | mitcheloc | nothing to it |
02:16.44 | sudoer | is the AMA flag in cdr equivalent to account number , so if i had an account number for each entry in sip.conf or iax.conf, i could assign their own ama #? |
02:16.58 | Godsey | you don't give it a password or context? |
02:18.00 | mitcheloc | whatever context it runs in i presume is the context it uses |
02:18.18 | bkw_ | show application DISA |
02:18.22 | mitcheloc | oh i was using no-password |
02:19.27 | mitcheloc | but thats cause i used it differently |
02:19.28 | sudoer | i cant find more info abouthow exactl ama works,voip-info.org didnt have enough information, can anyone tell me more about ama? |
02:19.40 | mitcheloc | basically i would have someone call in, dial 9 then it would dial them back on a predefined number |
02:19.44 | Godsey | I have a DID setup to answer and give a dial tone |
02:19.46 | mitcheloc | and then send them straight into DISA |
02:19.57 | mitcheloc | that way it's more secure then a password |
02:20.52 | Godsey | I have [disadial] |
02:21.06 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
02:21.06 | Godsey | exten => 1,1,Dial(SIP/95551212@as5400,60) |
02:21.22 | Godsey | expecting that when I dial 1, it'll ring the 555# :) |
02:21.26 | Godsey | I also have _1,1, |
02:22.32 | mitcheloc | post your example fully to pastebin can you? |
02:22.44 | Godsey | trying 1 more thing :) |
02:22.53 | Godsey | having the Anser() bit go to disadial first |
02:24.33 | Godsey | that worked |
02:24.35 | Godsey | yay :) |
02:25.02 | Godsey | [disadial] |
02:25.03 | Godsey | exten => s,1,DISA,1212|disadial |
02:25.03 | Godsey | exten => _.,1,Dial(SIP/9${EXTEN}@as5400,60,t) |
02:26.44 | mitcheloc | bkw, are you back/ |
02:28.23 | czero | has anyone test the ZyXEL WiFiSIP phone yes? |
02:28.25 | czero | yet even |
02:30.41 | bkw_ | czero its shit |
02:30.44 | bkw_ | mitcheloc yes |
02:30.51 | mitcheloc | bkw: what do you mean by the realtime stuff? |
02:31.01 | bkw_ | go read the cvs mailing list |
02:31.18 | czero | bkw have you tired anythere where any good? |
02:31.38 | mitcheloc | bkw: http://www.google.com/search?hl=en&q=asterisk+cvs+mailing+list+real+time&spell=1 |
02:31.39 | mitcheloc | ? |
02:31.40 | sudoer | has anyone played with those xml phone books incorporated inside of phones |
02:31.43 | mitcheloc | <PROTECTED> |
02:31.53 | sudoer | like in cisco's, they have an xml browser? |
02:32.15 | czero | I've used the Cisco one, at Cisco does that count :) |
02:32.20 | czero | but I've never set one up |
02:32.52 | *** part/#asterisk wellington (wellington@201008211120.user.veloxzone.com.br) |
02:33.15 | bkw_ | mitcheloc go check out latest cvs |
02:33.16 | bkw_ | check config.c |
02:33.20 | bkw_ | you'll have to check it out |
02:33.23 | bkw_ | I can't really explain it |
02:33.30 | bkw_ | its real time db stuff for asterisk |
02:34.31 | Godsey | guess I have some work to do on echo cancelation :) |
02:35.36 | Godsey | the newer cisco as5400 modem code is suposed to have good echo cancelation, I just couldn't figure out how to do it in ios |
02:35.47 | mitcheloc | heh bkw, sorry i checked it out and i'm looking at config.c but your talking about code? i don't know enough of how the code is to know whats going on |
02:35.51 | mitcheloc | can you tell me the gist of it? |
02:36.05 | mitcheloc | load realtime? |
02:36.09 | mitcheloc | so no more "reloads?" |
02:38.57 | bkw_ | correct |
02:39.06 | bkw_ | look for the word realtime on all the stuff |
02:39.07 | bkw_ | config.c |
02:39.10 | bkw_ | chan_sip.c |
02:39.12 | bkw_ | app_voicemail.c |
02:39.14 | bkw_ | chan_iax2.c |
02:39.22 | bkw_ | look at app_realtime.c |
02:39.58 | *** join/#asterisk mshades (~mshades@67.176.22.85) |
02:40.22 | *** join/#asterisk Legend` (~Legend@24.244.142.133) |
02:40.26 | mitcheloc | ok very cool |
02:40.36 | mitcheloc | so what does that have to do about the web config tool i was talking about? |
02:40.50 | bkw_ | tiz gonna make it EASY |
02:40.51 | bkw_ | :P |
02:40.58 | bkw_ | README.extconfig |
02:41.48 | *** join/#asterisk mitchel (trilluser@69-169-54-180.anhmca.adelphia.net) |
02:41.56 | mitchel | whoops got disconnected |
02:42.06 | mitchel | bkw: what did that have to do with my question about a web config tool? |
02:42.09 | bkw_ | README.extconfig |
02:43.35 | mitchel | ok |
02:43.40 | mitchel | so say i wrote that web configuration |
02:43.43 | mitchel | use databses? |
02:43.53 | bkw_ | ya |
02:44.10 | bkw_ | you could pull data from anything |
02:44.11 | bkw_ | LDAP |
02:44.14 | bkw_ | DNS |
02:44.15 | bkw_ | what ever you want |
02:44.18 | bkw_ | just write the driver for it |
02:44.57 | *** join/#asterisk mitcheloc (trilluser@69-169-54-180.anhmca.adelphia.net) |
02:45.02 | mitcheloc | damnit, i keep getting d/c |
02:45.04 | mitcheloc | i understand that |
02:45.16 | mitcheloc | so do you think i should write a well indepth config ui? |
02:45.17 | bkw_ | haha |
02:45.25 | bkw_ | mitcheloc sure you could |
02:45.54 | mitcheloc | but your recommendation is? |
02:46.02 | bkw_ | give it a few more weeks before you do anything |
02:46.35 | mitcheloc | ok, i'd like to seriously dedicate some time to making an official user interface though if you think thats possible |
02:46.43 | bkw_ | yes its possible |
02:46.46 | mitcheloc | i promise it would be good |
02:47.21 | mitcheloc | i think i'm going to do that |
02:47.24 | bkw_ | you could build an iax, sip and voicemail stuffs |
02:47.25 | mitcheloc | thing is i think everyone would hate it |
02:47.27 | mitcheloc | since i like c# |
02:47.33 | bkw_ | ewwwwww |
02:47.36 | bkw_ | write it in perl or php |
02:47.39 | mitcheloc | see |
02:47.44 | mitcheloc | theres nothing wrong with c# though |
02:47.47 | mitcheloc | thats whats stoping me |
02:47.54 | mitcheloc | it's quick, efficient |
02:47.59 | bkw_ | how? |
02:48.05 | bkw_ | M$ came up with it.. haha |
02:48.10 | bkw_ | I have never seen it |
02:48.12 | mitcheloc | it's practically 100% compiled code when run on the server |
02:48.15 | mitcheloc | compared to php |
02:48.23 | mitcheloc | oh c# code is easy nice syntax, hold on a sec |
02:48.26 | bkw_ | I can't comment since I haven't written anything in it |
02:49.10 | mitcheloc | here...look at some of this code, it's not my best work but it's something to look at |
02:49.14 | mitcheloc | http://www.titaniumsoft.net/downloads/tsweblib.0.2.src.zip |
02:50.20 | Chuji | where's Corydon when I need him |
02:50.33 | debaser | ok, everyone needs to go watch Team America: World Police |
02:50.40 | debaser | its quite possibly the funniest movie i've ever seen |
02:50.44 | Chuji | gawd that looks stupid debaser |
02:51.06 | debaser | you're a fucking douche |
02:51.08 | Chuji | Anyone good with AGI::Asterisk? |
02:51.11 | debaser | it was hillarious |
02:51.48 | dan2 | C# == java |
02:51.59 | sudoer | is cisco the only phoen that has a 'xml mini browser'? |
02:52.02 | mitcheloc | C# ~ java |
02:52.09 | mitcheloc | it's similar, not the same though |
02:52.25 | dan2 | mitcheloc: ocaml > C# | Java |
02:52.28 | dan2 | :-P |
02:53.22 | mitcheloc | it's perl though? lol |
02:54.40 | mitcheloc | ugh! it's ugly =p |
02:54.44 | dan2 | ocaml? |
02:54.46 | dan2 | ocaml is great |
02:55.32 | mitcheloc | the reason i wouldn't use it is |
02:55.36 | mitcheloc | it doesn't look widely supported |
02:55.42 | dan2 | *COUGH* |
02:55.48 | mitcheloc | is it? |
02:55.50 | dan2 | ocaml is ported to many more platforms than C# |
02:55.54 | dan2 | MANY |
02:56.05 | mitcheloc | aye well i have no say so as i don't know anything about it |
02:56.28 | mitcheloc | bkw: did you see the source yet? |
02:56.30 | dan2 | mitcheloc: ever heard of the file sychronization tool called unison? |
02:56.32 | *** join/#asterisk gregwood (~lynnwood@pcp01073653pcs.andrsn01.tn.comcast.net) |
02:56.48 | dan2 | mitcheloc: its based after rsync protocol, but more powerful, written in ocaml, on every platform |
02:58.56 | mitcheloc | nope haven'theard of it |
02:59.05 | mitcheloc | i'm new to this linux stuff/that area |
02:59.11 | mitcheloc | i was a windows geek before ;) |
03:00.30 | *** join/#asterisk astk_tester (nathan@68.114.199.229) |
03:00.48 | astk_tester | hm, anyone from NuFone here? |
03:01.17 | astk_tester | I'm having some trouble registering I believe |
03:03.48 | astk_tester | anyone at all here? |
03:04.04 | czero | yes but I'm not from nuphone |
03:04.11 | astk_tester | that's alright |
03:04.14 | gregwood | Im here as well |
03:04.45 | astk_tester | anyone know of any good asterisk-compatible VoIP providers with pay-as-you-go and offer toll-free DIDs in the US? |
03:05.02 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
03:10.36 | *** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68) |
03:10.40 | B0ngFrOg | lo |
03:11.22 | mitcheloc | astk i guess nufone |
03:11.29 | mitcheloc | but i don't ever get good quality from them |
03:12.17 | iMediax | they've been great for me |
03:12.42 | czero | how many minutes amonth do you guys push? |
03:12.46 | czero | if I may ask |
03:12.49 | mitcheloc | less then 10 lol |
03:12.55 | mitcheloc | but thats cause i prefer broadvoice |
03:12.58 | mitcheloc | better quality etc |
03:13.00 | czero | :) |
03:13.12 | iMediax | bout 120/mnth |
03:19.15 | *** join/#asterisk telme (~non@c-24-8-57-124.client.comcast.net) |
03:19.38 | telme | so i installed my t100p and nothing happens. neither the read or the green light come on. what gives? |
03:20.02 | mitcheloc | heh |
03:20.04 | mitcheloc | you have to configure it |
03:20.07 | mitcheloc | zapata.conf |
03:20.15 | mitcheloc | and then you do a modprobe |
03:20.18 | mitcheloc | to get lights up |
03:20.50 | telme | but even if zapata.conf and modprobe was wrong wouldn't the green light come on? (by the way i've already modprob'd successfully). |
03:21.02 | telme | sorry not green but any?? |
03:21.43 | Kb1_Kanob | telme: you've got zaptel in, but is wct1xxp also loaded? |
03:22.09 | telme | modprobe wct1xxp doesn't return any errors |
03:22.19 | Kb1_Kanob | does it show in lsmod? |
03:22.42 | telme | as unused yes |
03:23.05 | Kb1_Kanob | Hmmm.... and entries in dmesg are all reasonable? |
03:23.13 | *** part/#asterisk ranliv (~root@203.172.11.239) |
03:23.44 | mitcheloc | no it wouldn't come on i doubt it |
03:23.54 | mitcheloc | red might though |
03:24.11 | Kb1_Kanob | unless the span isn't declared in /etc/zaptel.conf? |
03:24.16 | telme | mitcheloc: that's what i'm askin, neither light comes on |
03:24.19 | telme | loadzone=us |
03:24.20 | telme | defaultzone=us |
03:24.20 | telme | span=1,0,0,esf,b8zs |
03:24.20 | telme | fxoks=1-24 |
03:24.20 | telme | fxsks=1 |
03:24.26 | telme | there's zaptel |
03:24.27 | Kb1_Kanob | that will not load. |
03:24.38 | telme | ??? |
03:24.39 | Kb1_Kanob | channel 1 is double-declared. |
03:24.53 | Kb1_Kanob | s/b fxoks=2-24 |
03:24.53 | telme | oks vs. sks |
03:25.05 | telme | no no |
03:25.16 | telme | there is a t100p and an x100 |
03:25.24 | telme | the x100 is the sks |
03:25.58 | telme | here, i'll kill the sks and see what happens |
03:26.18 | Kb1_Kanob | Not tried t100p w/x100p in same box. Likely need something else between the fxs and fxo lines to make wct1xxp understand it's for a different drive. |
03:26.23 | Kb1_Kanob | s/drive/driver/ |
03:28.02 | telme | i killed the fxsks and still no light (red or gree) |
03:28.06 | telme | gree -> green |
03:28.46 | Kb1_Kanob | Sounds like the x100p hardware is confusing something. |
03:29.14 | B0ngFrOg | have you tried to put a loop back in the t1 port ??? |
03:29.31 | telme | what i the opposite of modprobe (to uninstall) |
03:29.37 | telme | i -> is |
03:29.40 | B0ngFrOg | rmmod |
03:29.56 | Kb1_Kanob | in reverse order of usages (rmmod wct1xxp; rmmod zaptel) |
03:30.13 | telme | i'm gonna remove the wcfxo |
03:30.15 | Legend` | or rmmod -r wct1xxp to remove zaptel as well |
03:30.35 | B0ngFrOg | ;p |
03:30.36 | Kb1_Kanob | Legend`: heh - you learn something useful every day. Thanks. |
03:32.57 | mitcheloc | no it shouldnt |
03:33.02 | mitcheloc | just do it in order |
03:33.05 | mitcheloc | you modprobe wct`1xxp |
03:33.12 | mitcheloc | then it's fxsks = 1-24 |
03:33.22 | mitcheloc | then the next one is fxoks=25 |
03:33.25 | mitcheloc | something like that |
03:33.41 | Kb1_Kanob | mitcheloc: will the x100p need a 'span' declaration? |
03:34.32 | mitcheloc | not sure |
03:34.34 | mitcheloc | i don't think so though |
03:35.27 | mitcheloc | hey you want to buy a t100p ? =) i've got an extra one =/ |
03:35.36 | Kb1_Kanob | how much? |
03:36.17 | Kb1_Kanob | Also, country of origin? |
03:36.20 | Kb1_Kanob | (of shipment) |
03:36.27 | telme | still nada you guys, thanks for trying to help me |
03:36.51 | mitcheloc | make me an offer |
03:37.00 | mitcheloc | its new unused |
03:37.05 | Kb1_Kanob | shipping out of the USA? |
03:37.20 | mitcheloc | check fedex.com i don't know it |
03:37.29 | mitcheloc | but if the moeny clears why would i care |
03:37.32 | B0ngFrOg | telme --- are you sure there is no odd refrences in /var/log/messages when you modprobe??? |
03:37.33 | mitcheloc | where it goes to |
03:37.42 | Kb1_Kanob | I would care - taxes, duties etc. |
03:38.00 | telme | B0ngFrOg: i'll check |
03:38.05 | astk_tester | anyone here a VoIP provider? |
03:38.10 | Kb1_Kanob | I meant 'is it shipping from the usa?', sorry. :-0 |
03:38.19 | mitcheloc | yes |
03:38.24 | telme | astk_tester: why, you lookin? |
03:38.31 | astk_tester | yes |
03:38.59 | telme | check out teliax, good prices, I have no complaints |
03:39.17 | Kb1_Kanob | mitcheloc: Let me ponder - don't dissapear. |
03:40.37 | astk_tester | looking for around 3000 min/mo for now, with potential for much much more if we decide to deploy asterisk over many of our commercial and residential properties |
03:40.43 | mitcheloc | ok just msg me if i'm not here on aim: mitcheloc |
03:40.45 | astk_tester | going to try it in our office first |
03:40.59 | mitcheloc | astK: whats your business? |
03:41.11 | astk_tester | I'm a real-estate developer |
03:42.02 | astk_tester | we'll have about 100,000 sq. ft. of class-a medical office space this time next year, and about 300 units of residential rentals |
03:44.52 | andy_b | hey everyone - question if anyone could help me out - just reloaded Asterisk and reloaded the conf files from my prior working config - not I can't log into FWD (times out - failure to register) - any hints from anyone? |
03:45.13 | andy_b | not = now |
03:45.48 | mitcheloc | heh where at? |
03:46.07 | herag | is there an impedance difference between the headset jack on my cell phone and the headset jack on my cordless phone? the earbud that I have for my cell phone works fine, but if I try to use it for in the jack for my cordless phone, the sound is totally |
03:46.22 | herag | low |
03:46.39 | *** join/#asterisk Sickning (~sickning@transkore.com) |
03:46.52 | astk_tester | mitchel: missouri |
03:46.53 | herag | attenuated...it's like there's just an impedance mismatch...if that's so, how would I go about finding an earbud that would work for my phone? |
03:46.54 | Sickning | Anyone around to answer a question ? |
03:48.02 | Kb1_Kanob | Sickning: I take it you haven't been here before? |
03:48.08 | Sickning | correct |
03:48.11 | Sickning | ;-) |
03:48.32 | mitcheloc | hey if they want to hire me on the west coast (orange/la county) let me know =) |
03:48.33 | mitcheloc | i've got to go, be back in 20-30 minutes |
03:48.34 | Kb1_Kanob | Right - best to come right out with the question. |
03:49.21 | Kb1_Kanob | sooooooo? |
03:50.17 | *** join/#asterisk silug (~steve@osiris.silug.org) [NETSPLIT VICTIM] |
03:54.18 | file | 996 |
03:56.47 | *** join/#asterisk PhilM (~a@r42h98.res.gatech.edu) |
03:58.47 | gregwood | Need Help, Have installed asterisk but can not get the demo to work. Can't hear any voice played from the box. iphone to iphone call works fine but echo and demo and voicemail don't play any voice. |
03:59.08 | *** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68) |
04:03.02 | *** join/#asterisk andrew` (~andrew@adsl-67-119-25-173.dsl.snfc21.pacbell.net) |
04:04.03 | gregwood | is this thing working |
04:04.31 | Kb1_Kanob | yes - no one knows the answer to your question at the moment. Or they're all sleeping. |
04:04.41 | gregwood | ok thanks |
04:15.36 | docelm0 | What is the easiest way to create new extensions? Can I use the management API to do this? |
04:15.50 | florz | gregwood: Created the echo extension yourself or default config? |
04:16.04 | gregwood | default config |
04:16.23 | gregwood | actually copied the /etc/asterisk directory from another machine that is working fine |
04:17.01 | gregwood | ip phone call work fine also. it would appear to be a problem with playing back recorded audio. |
04:17.22 | florz | gregwood: How about creating a simple test context with only one extension that answers and then starts the echo app? |
04:17.46 | gregwood | can try |
04:18.44 | docelm0 | Can I create new extensions via the Management API? |
04:19.44 | florz | docelm0: I think someone in here said something like that recently ... |
04:23.09 | docelm0 | I have been out and about working on other things.. But Im guessing from looking at the wiki it cant be done.. Then what is a efficient way to do it? Create the files and have them included in the main conf's? |
04:23.15 | docelm0 | or edit the confs directly? |
04:25.50 | gregwood | florz: ok, the echo test actually works if I don't do the playback(demo- |
04:25.57 | gregwood | echotest) |
04:26.34 | gregwood | if I include that not only do i not hear the playback it would appear it never goes beyond that point. |
04:26.34 | *** join/#asterisk andreg (~andreg@204.249.177.82) |
04:27.54 | file | http://www.jonessoda.com/ |
04:28.27 | file | http://www.jonessoda.com/files/sodas.html |
04:28.59 | florz | gregwood: Tried to start asterisk with -vvvvvvvv |
04:29.01 | florz | ? |
04:29.11 | gregwood | done |
04:30.02 | florz | gregwood: Then what does it say when you try a playback? |
04:31.41 | gregwood | <PROTECTED> |
04:31.42 | gregwood | <PROTECTED> |
04:31.42 | gregwood | <PROTECTED> |
04:32.18 | gregwood | The weird thing is I can hear when it begins to play. Sounds like someone picking up a phone but not talking. |
04:32.28 | florz | hmmm |
04:32.30 | gregwood | iaxphone also show I am receiving audio |
04:32.44 | *** join/#asterisk docelm0 (~me@116-39.202-68.tampabay.rr.com) |
04:33.12 | florz | weird, indeed =:-) |
04:33.41 | florz | already tried a different client (protocol)? |
04:34.05 | iMediax | lol what speed is that? 38081864 bytes received in 3.33 secs (1.1e+04 Kbytes/sec) |
04:34.12 | Kb1_Kanob | gregwood: codec problem? |
04:34.36 | *** join/#asterisk ahyanne (yahnee@dialup-222-126-69-37.infocom.ph) |
04:34.50 | florz | iMediax: about 1.1 MB/sec |
04:34.53 | gregwood | One the working machine when the deom-echtest payback is done it goes into the echo app. on this machine it never goes to the echo app, just stays in the playback. |
04:34.59 | iMediax | ahh duh |
04:35.11 | florz | iMediax: really ;-) |
04:35.17 | iMediax | heh smack |
04:35.20 | florz | iMediax: What software is that? ;-) |
04:35.25 | gregwood | yes I have tried several iax phones and also tried sip |
04:35.35 | iMediax | its off my 1and1 server |
04:35.42 | iMediax | simple ftp |
04:36.26 | florz | iMediax: So it's your FTP client telling you that?! |
04:37.26 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
04:37.29 | iMediax | ya.. reason why i said that.. i was d/l new kernel, and kernels are big... so i thought it was an error since only took 3.3 secs *shrug* long day. |
04:39.17 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
04:40.18 | florz | iMediax: IC - the speed looks much like a 100 Mbps connection ... |
04:40.36 | iMediax | that was from kernel.org |
04:40.39 | gregwood | Kb1_Kanob: the gsm codec works fine when I make a ip phone call. and the echo app works fine. It seems to just be the playback. |
04:41.10 | gregwood | voicemain does not work either but I assume it is using the playback command under the covers. |
04:41.10 | Kb1_Kanob | Hmmm.. that's interesting. |
04:41.10 | Kb1_Kanob | yes, it does. |
04:41.19 | Kb1_Kanob | when you make an IP call via asterisk? |
04:41.31 | gregwood | correct |
04:41.55 | florz | iMediax: Ah, so were logged in via ssh/telnet/whatever to your server and downloaded the kernel there? Then it even might be that this speed might be correct without a local proxy ... =:-) |
04:42.11 | *** part/#asterisk ahyanne (yahnee@dialup-222-126-69-37.infocom.ph) |
04:42.33 | iMediax | ya i'm logged in viz ssh |
04:42.40 | iMediax | via* |
04:42.44 | florz | gregwood: When making a call through *, are you using different codecs with both phones? |
04:43.31 | gregwood | If I understand you question correctly, No. Both phones are using the gsm codec. |
04:43.57 | Kb1_Kanob | is this a recent download from CVS? If so did you pull v1-0 or head? |
04:44.38 | florz | gregwood: Then it still might be a codec problem. As long as a stream goes out in the same format it comes in, no recoding is needed, data is basically passed through transparently |
04:44.49 | Sickning | anyone know of any good softphones besides DIAX |
04:45.26 | gregwood | I have two version currently. One is a recent download from about a week ago. the other is a copy from another machine. complete copy of /etc/asterisk from other machine. Other machine works fine, this one does not. |
04:46.09 | Kb1_Kanob | If you're trying to run cvs head you might have problems. |
04:47.02 | gregwood | I could be wrong but I don't thing it has anything to do with asterisk directly but maybe something to do with the os. os is redhat, same version but different install packages,etc. Also on different type machine. |
04:47.37 | Kb1_Kanob | Shouldn't affect it. It's all just packet shuffling - no hardware unless you specifically add it. |
04:47.44 | Kb1_Kanob | (meaning audio hardware) |
04:48.13 | gregwood | I would agree, but it the same binary and config files on both machines |
04:48.23 | gregwood | it is |
04:49.08 | Kb1_Kanob | in that case - are the config files modified correctly in /etc/asterisk ? |
04:49.18 | Kb1_Kanob | could be a networking problem w/the audio streams. |
04:50.04 | Kb1_Kanob | what is the client? |
04:50.24 | gregwood | the /config files are identical. No sure what I would modify. Don't know about the audio stream but again the echo works and I can make a phone call. |
04:50.30 | gregwood | iaxphone |
04:51.21 | Kb1_Kanob | playback is very simple. If it 'hangs' on that step for longer than the duration of the actual audio file it implies there is a problem pushing the data out - ie. that it's waiting on something to move. |
04:51.32 | *** join/#asterisk Mike (~mike@201.135.48.52) |
04:52.34 | gregwood | it does appear to hang. it never comes back from the payback unless I hang up the phone. I have stayed on as long a 10 mins and it never returns from the playback to next step in the dial plan. |
04:52.48 | Kb1_Kanob | what happens if you use background() instead? |
04:52.58 | gregwood | have not tried |
04:52.59 | gregwood | hold on |
04:54.48 | Kb1_Kanob | gregwood: any chance there is a firewall issue? compare 'iptables -v -L' on both boxes. |
04:54.50 | gregwood | it hangs as well |
04:55.15 | B0ngFrOg | gergwood .. are you sure the file you are playbacking is on the machine??? |
04:55.33 | bkw_ | are you answering the channel? |
04:55.35 | *** join/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net) |
04:56.01 | Kb1_Kanob | bkw_: would not answering stall the playback? not just prevent the audio from making it out? |
04:57.01 | gregwood | iptables are the same on both boxes. iptables not being used on either. |
04:57.37 | gregwood | buth return INPUT FORWARD and OUTPUT chains are empty |
04:57.55 | Kb1_Kanob | You're getting audio on demo, but not a custom playback and custom voicemail? |
04:58.12 | gregwood | no audio on demo either |
04:58.42 | Kb1_Kanob | no audio on voicemail implies something other than a missing Answer() then. |
05:00.37 | gregwood | sorry, in answer to the playback file, yes it is there. I also tried to rename it and I did get a file not found error then. |
05:03.33 | gregwood | If I dial 8500 for voice mall and enter an mailbox number(even though I can't hear it as for it) It does switch to playing the playback for the password (which I also can not hear). |
05:05.08 | Kb1_Kanob | that really sounds like a codec negotiation issue between the ends. but I don't know iaxphone. sorry,. |
05:07.49 | gregwood | could it be a timing issue with the codec and the actual hardware I am running on. The machine is an smp compaq proliant with 2 cpu's. I have the quad t1 care installed. have tried removing the quad t1 and using a x100p but the result is the same. |
05:11.10 | gregwood | also could it have anything to do with the language. Don' t know how languages work as I can only find one version of the sound files. |
05:11.35 | Kb1_Kanob | shouldn't be language. |
05:12.03 | Kb1_Kanob | Playback should work w/o timing source. It's needed for meetme and other coordinated services. |
05:13.18 | Kb1_Kanob | playback is a tiny application - 42 lines of code. |
05:13.22 | *** join/#asterisk Fpl (~Fpl@201.248.181.147) |
05:13.45 | Kb1_Kanob | it starts the stream on the channel and then waits for it to finish before returning. |
05:14.04 | Kb1_Kanob | so the stream isn't ever finishing. |
05:14.28 | Fpl | hi guys |
05:14.30 | Kb1_Kanob | which implies the audio channel from * -> client that carries the stream isn't ok. |
05:14.42 | Fpl | anyone using QUICKNET BOARDS ? ( linejack ) ? |
05:14.50 | Kb1_Kanob | you're not using g729 are you? |
05:14.58 | gregwood | Would the echotest and the playback be carried on the same audio channel? |
05:15.02 | gregwood | no to g729 |
05:15.23 | Kb1_Kanob | they should be. |
05:15.36 | gregwood | and the echotest works fine |
05:15.39 | Fpl | anyone using Linejack from Quicknet ???? |
05:15.40 | Kb1_Kanob | do you have silence supression turned on somewhere? |
05:15.48 | Kb1_Kanob | on the client? |
05:16.04 | gregwood | no |
05:16.07 | Kb1_Kanob | damn. |
05:16.58 | Kb1_Kanob | Can you try a different client, like diax or x-lite? |
05:17.04 | gregwood | sure |
05:17.07 | gregwood | I have both |
05:17.09 | Kb1_Kanob | x-lite is sip, but it would prove out the engine. |
05:18.09 | *** join/#asterisk rene- (~rene-@201.135.255.155) |
05:18.20 | rene- | hello boys and girls |
05:18.41 | Fpl | i have about 7 Linejack boards and would like to know if this is possible. : I had a standard siemens PBX that had capability for 4 PSTN LINES AND 8 extensions. ( all analog ). The PBX burned.....!!! now, i have a box with 7 ISA Linejacks that i wish to implement as my pbx using the standard analog RJ-11 in the linejacks. Use two for incoming lines and on the same token the remaining to transfer the calls to the different extensions..Has anyone done this ? |
05:19.07 | rene- | is anyone experiencing problems with voicepulse where the call gets dropped randomly after 1 or 5 seconds with no-apparent reason |
05:19.09 | Fpl | using quicknet linejacks ?? |
05:19.32 | Kb1_Kanob | Fpl: would the rolm phones work plugged directly into a telco wallplug? |
05:20.09 | Kb1_Kanob | ie. they're not anything fancy for signalling? |
05:20.12 | Kb1_Kanob | like 4 wire? |
05:22.36 | gregwood | Ok, diax has the same results works fine on old server, can do echotest on new server but can't do playback. |
05:22.49 | gregwood | trying x-lite |
05:22.58 | Kb1_Kanob | that's extremely wierd. |
05:27.09 | *** join/#asterisk drmatrix (~sean@cdm-68-228-9-62.laft.cox-internet.com) |
05:28.04 | drmatrix | anybody using a sangoma? |
05:28.48 | gregwood | same with x-lite except that I tend to hear more static/hissing/background noise during the echotest. |
05:29.04 | Kb1_Kanob | now that's got me stumped. |
05:29.10 | bkw_ | dfasdfasdfasdfasdfasdf |
05:29.33 | Kb1_Kanob | IAX clients and SIP clients don't get audio from asterisk, even demo? |
05:29.54 | Kb1_Kanob | noise in echo test is probably just delay/compression artifacts |
05:31.35 | *** join/#asterisk justinnnnnn (~justinm@solid.mpa.net.au) |
05:31.37 | justinnnnnn | hey guys |
05:31.38 | justinnnnnn | guess wat |
05:31.41 | gregwood | also playback and background both hang and never return to next step in dial plan |
05:31.49 | justinnnnnn | my server isnt screwed after the gr8 hdparm incident of yesterday |
05:31.51 | justinnnnnn | reboot fixed her :) |
05:32.05 | justinnnnnn | can someone please help me with my sata hdd's tho.. i cant enable dma.. but i need to do something to speed em up or something |
05:32.10 | justinnnnnn | to stop em interfering with my e100p |
05:32.21 | justinnnnnn | im getting heaps of event 6 warnings and each time it comes up the pstn person can hear a click.. |
05:32.46 | Kb1_Kanob | which sata drivers? |
05:32.52 | justinnnnnn | not sure |
05:32.55 | justinnnnnn | how do i tell ? |
05:33.07 | Kb1_Kanob | erm. lsmod? |
05:34.15 | justinnnnnn | i think its this.. |
05:34.17 | justinnnnnn | ext3 73376 2 |
05:34.51 | Kb1_Kanob | nope. |
05:35.03 | Kb1_Kanob | what kernel are you using? |
05:35.07 | justinnnnnn | 2.4-smp |
05:35.08 | justinnnnnn | redhat 9 |
05:35.20 | justinnnnnn | hid 22308 0 (unused) |
05:35.25 | justinnnnnn | usb-uhci 27404 0 |
05:35.25 | justinnnnnn | ehci-hcd 20456 0 (unused) |
05:35.28 | justinnnnnn | any of thse 3 ?| |
05:35.42 | Kb1_Kanob | perhaps dump output of lsmod into pastebin.ca and bring back the link. |
05:36.13 | *** join/#asterisk BoRiS (boris@wnpgmb01dc2-31-184.dynamic.mts.net) |
05:38.15 | *** join/#asterisk coppice (~Steve_Und@218.198.17.210.dyn.pacific.net.hk) |
05:38.36 | justinnnnnn | http://www.pastebin.com/108890 |
05:39.32 | Kb1_Kanob | odd - looks like its monolithic (ie. not a module) |
05:39.46 | Kb1_Kanob | grep dmesg for ide and paste it in a bin. |
05:40.37 | herag | I have a curious problem with my current setup, cell phone users who have t-mobile and cingular can't call me, all other types of incoming calls seem to work just fine...i'm pretty sure it's a configuration problem on my side, because if I connect my sipura directly to my vsp, then these celll users can call me just fine, any ideas? |
05:41.18 | justinnnnnn | http://www.pastebin.com/108891 |
05:41.27 | justinnnnnn | theres the grep ide |
05:42.33 | *** join/#asterisk habakuk (~chatzilla@adsl-64-168-20-150.dsl.snfc21.pacbell.net) |
05:42.41 | drmatrix | herag, who owns the number they are dialing? |
05:42.42 | Kb1_Kanob | hmmm... which drives are joined w/jbod? |
05:42.52 | justinnnnnn | not sure.. |
05:42.56 | herag | drmatrix: huh? |
05:43.08 | herag | drmatrix: the number is mine, it's a broadvoice line |
05:43.22 | drmatrix | and only tmobile/cingular users can't call? |
05:43.49 | herag | drmatrix: yes, but they can't call only when I use my asterisk server, if my sipura goes directly to broadvoice, they can call |
05:43.54 | drmatrix | what's a call to you sound like to them? |
05:43.58 | drmatrix | with asterisk |
05:44.04 | herag | drmatrix: a bunch of little quick beeps |
05:44.27 | drmatrix | do you have a verbose log of asterisk when the call comes in? |
05:44.30 | herag | drmatrix: I can even see their call come in on the cli, it terminates after the third exten => command, usually |
05:44.49 | drmatrix | which is/ |
05:45.08 | herag | -- Executing Answer("SIP/147.135.0.129-08137430", "") in new stack |
05:45.08 | herag | <PROTECTED> |
05:45.08 | herag | <PROTECTED> |
05:45.28 | herag | after the waitexten, I get a spawn extension...exited non-zero |
05:45.54 | grrant | I think I accidentally deleted a line in sip.conf now Xlite doesnt work with asterisk it says "No compatible codecs" and then plays a busy signal |
05:46.46 | gregwood | you can turn the codecs on and off in x-lite by clicking them in the display window of x-lite. make all the codec black instead of white or grey. |
05:46.54 | herag | it's really peculiar, cause verizon cell users can call, at&t can call, all the land lines I've tried can...it's just really bizarre |
05:47.02 | *** part/#asterisk Fpl (~Fpl@201.248.181.147) |
05:47.10 | drmatrix | what is the next step when other users call |
05:47.39 | herag | drmatrix: normally? assuming they don't push a new extension on the stack? |
05:47.58 | drmatrix | i assume the beeping occurs before 2 secs pass... |
05:48.07 | Kb1_Kanob | herag: how is the call coming in? what interface/hardware? |
05:48.21 | herag | drmatrix: I'm not sure about that, it all happens so fast, I can't be sure |
05:48.27 | grrant | Gregwood: I tried that but it still has the sameproblem |
05:48.41 | herag | Kb1_Kanob: it's all sip, broadvoice number gets dialed, goes to my asterisk server |
05:49.00 | drmatrix | grrant: try allow=gsm in the sip.conf for that user |
05:49.12 | Kb1_Kanob | how long before the hangup (from the time the caller hits send)? and is it consistent? |
05:49.16 | herag | drmatrix: the next step in the extension, assuming no new extension gets pushed on is stopplaytones, and then a Dial(...) |
05:49.38 | drmatrix | herag, msg me with the number; i can try with my cingular |
05:49.41 | herag | Kb1_Kanob: I believe it's happens as soon as the caller connects |
05:50.23 | habakuk | anyone know what "realtime" support is referring to in chan_sip? |
05:50.47 | herag | Kb1_Kanob: and I cannot be sure if the duration happens consistently, because generally a cell phones takes a varying amount of time to connect to a call |
05:50.58 | Kb1_Kanob | I'm having a problem with allstream/rogers, but it's an analog answer supervision problem on my channel bank. Manifests itself as hangup after exactly 60 seconds. Not a problem for other carriers. |
05:50.59 | grrant | allow=gsm is already there |
05:51.14 | grrant | I just typo'd something in sip.conf and this happened |
05:51.34 | herag | drmatrix: you connected? |
05:51.42 | drmatrix | yep... music? |
05:51.46 | herag | yes |
05:51.51 | herag | hmm |
05:51.58 | herag | this is curious |
05:52.11 | drmatrix | once more; another cingular |
05:52.12 | Kb1_Kanob | herag: if it was a telco supervision problem w/broadvoice it would be timered from the pressing of send, not connecting to you. |
05:52.29 | grrant | Weird I just called my DID it rang my softphone Xlite |
05:52.31 | drmatrix | works there 2 |
05:52.36 | herag | yup |
05:52.40 | grrant | BUt when I try to call out with Xlite it says "no compatible codec" |
05:52.49 | herag | hmm |
05:52.55 | herag | now I don't know what to make of this |
05:52.57 | grrant | My account is at only $1.90 but that should be sufficient plus Im calling 800#s |
05:53.32 | grrant | What happens in xlite when you select (or make black) all the codecs, how does it select which one it wants to go to |
05:53.34 | herag | see, I only know what my friends tell me, I'm on verizon, and it works...but I have friends on t-mobile and cingular that say they can't get through |
05:53.40 | herag | and they have similar problems |
05:53.59 | herag | thanks for testing that drmatrix |
05:54.15 | herag | though this makes my problem a little harder |
05:54.30 | drmatrix | they just dont hear the music i'd bet. |
05:55.04 | drmatrix | your ring runs into the original ring... it's hard to tell when asterisk picks up |
05:55.23 | grrant | How could xlite and asterisk work to receive calls through a DID but not to make calls from asterisk? |
05:55.25 | drmatrix | i'd change the ring tone with a background voice prompt, just to be sure they can tell |
05:55.44 | herag | drmatrix: no, that music is what was coming out of my speakers |
05:55.52 | herag | drmatrix: I physically picked up that call |
05:56.08 | drmatrix | instead of the waitexten? |
05:56.45 | grrant | Anyone else having problems making outbound calls with Connect.Voicepulse |
05:57.03 | drmatrix | ah, they don't support all the codecs they say they do |
05:57.05 | herag | drmatrix: the waitexten only lasts for two seconds, if you don't push anything on the stack, it'll go to dial |
05:57.22 | drmatrix | grrant, i'd try ulaw everything until it's working |
05:57.32 | herag | drmatrix: now here's an even more curious situation, I just told one of my friends who can't connect to dial me |
05:57.41 | herag | and I took out the waitexten stuff |
05:57.53 | herag | so all I have is 1,Answer() and 2,Dial() |
05:58.08 | herag | and it rang my phone for a second, and then just hung up |
05:58.13 | grrant | Do i just add disallow = all & allow = ulaw in sip.conf |
05:58.18 | herag | it's almost like something's timing out |
05:58.21 | grrant | Or in iax, and everywhere else I can including xlite |
05:58.23 | drmatrix | grrant, yes |
05:58.41 | grrant | iax? |
05:59.07 | Kb1_Kanob | herag: can you debug where the hangup is occuring in the console? Ie. is broadvoice hanging up the call? |
05:59.08 | drmatrix | same |
05:59.23 | herag | Kb1_Kanob: I don't know how to determine where the hangup is happening |
05:59.28 | herag | Kb1_Kanob: how would I check? |
05:59.34 | drmatrix | what was the log on that last call? |
05:59.53 | grrant | Me? |
05:59.58 | drmatrix | no, herag. |
06:00.01 | herag | drmatrix: you mean the cli output? |
06:00.04 | drmatrix | you: same for iax and sip .conf |
06:00.07 | drmatrix | yes |
06:00.12 | grrant | Yea I didit, still not working |
06:00.12 | drmatrix | set verbose 5 perferred |
06:00.17 | grrant | But my DID still rings to xlite |
06:00.19 | grrant | How can this be |
06:00.24 | drmatrix | still the xlite error? |
06:00.24 | herag | oh, verbose 5, ok I'll try that |
06:00.52 | Kb1_Kanob | herag: or sip debug peer ..... |
06:00.58 | grrant | No xlite error |
06:01.01 | grrant | Well yea |
06:01.04 | grrant | 404 not found |
06:01.22 | drmatrix | ? |
06:01.37 | grrant | The error is in xlite |
06:01.44 | grrant | it says Call failed: 404 not found |
06:01.45 | grrant | Xlite error, before I did these modifications it would give me an CLI error saying "no compatible codecs" and everythin was working fine butI |
06:01.56 | grrant | typo'd something in sip.conf and saved/closed it |
06:02.07 | drmatrix | sip reload from the console |
06:02.08 | grrant | I've looked through sip.conf for syntax errors but nothing |
06:02.19 | grrant | What does sip reload do? |
06:02.24 | drmatrix | rereads the file |
06:02.28 | grrant | yuh |
06:02.34 | grrant | I already did that with the changes you said |
06:02.39 | drmatrix | no error? |
06:02.59 | grrant | Not in CLI |
06:03.10 | grrant | Unless I need to set verbose to 100 first |
06:03.14 | grrant | Now the error is in xlite |
06:03.19 | grrant | 404 |
06:03.28 | riksta | "there is no error" |
06:03.37 | riksta | s/error/spoon/ |
06:03.42 | grrant | Huh? |
06:03.48 | drmatrix | set a verbose, and see if you are dialing an invalid extension |
06:03.57 | riksta | it's the matrix |
06:04.11 | herag | ok, progress has been made...if I have just a single dial command in my extensios 1,Dial(...) they can get in |
06:04.32 | grrant | its set to verbose |
06:04.36 | grrant | I did asterisk -rvvvvc |
06:04.43 | grrant | What level is that considered: 4? |
06:04.46 | drmatrix | yes |
06:04.58 | herag | but if I want to do anything else, it won't work....I'd like to pick up the phone...1,Answer 2,Dial()....why on earth won't it work with something like this? |
06:05.38 | Kb1_Kanob | herag: Paste the exact (unworking) in pastebin |
06:05.44 | herag | ok, one sec |
06:06.15 | drmatrix | herag, you might want to add the obligatory wait,2 right after the answer |
06:06.18 | bkw_ | kram |
06:06.25 | herag | http://pastebin.ca/1358 |
06:06.28 | bkw_ | I swear he doesn't pay attention at all sometimes |
06:08.34 | Kb1_Kanob | herag: they are always hung up on, even without trying to hit a dtmf key? |
06:08.52 | habakuk | hi bkw_ can you tell me what you mean by "REALTIME" extensions to chan_sip.c ? how is that better than MYSQL_FRIENDS? is it just a more modular approach? |
06:08.58 | herag | Kb1_Kanob: yes, they never try to push a dtmf |
06:09.07 | herag | Kb1_Kanob: yet they still are dropped |
06:09.52 | herag | Kb1_Kanob: asterisk would tell me if they pushed a dtmf and are now on a different extension, I mean, the calls are dropped even if all I have is 1,answer 2,dial() |
06:10.10 | Kb1_Kanob | and if you simply change 3 to 'exten => s,3,noop(WaitExten(2))'? |
06:10.15 | bkw_ | habakuk go read |
06:10.18 | bkw_ | I can't really explain it |
06:10.19 | herag | but the calls _work_ if all I have is 1,dial() |
06:10.22 | bkw_ | its alot more modular |
06:10.24 | Kb1_Kanob | Hericom: ah, sorry. |
06:10.33 | Kb1_Kanob | herag, even. |
06:10.35 | drmatrix | herag, just for giggles, does 1,answer; 2 dial work without the ,r ? |
06:11.19 | bkw_ | its ,,r |
06:11.21 | herag | drmatrix: I'm not sure, I will have to try |
06:11.22 | bkw_ | recall the timeout |
06:11.33 | drmatrix | he had a 30, bkw |
06:11.34 | bkw_ | IAX2/blah,,r |
06:11.38 | bkw_ | oh ok |
06:11.40 | bkw_ | just checkin |
06:11.40 | drmatrix | i'm happy leaving that |
06:11.44 | herag | Kb1_Kanob: I nooped it, nothing |
06:11.48 | herag | Kb1_Kanob: still dropped |
06:11.49 | habakuk | bkw_, sure. do you a have a bug# where I can read more about? My main beef with MYSQL_FRIENDS, is there doesn't appear to be a way to create a sip user of type "friend" |
06:12.02 | habakuk | i.e. just user or peer |
06:12.20 | bkw_ | friend is a user and a peer at the same time |
06:12.25 | Kb1_Kanob | herag: try that wait(2) as s,1,... suggestion. |
06:12.48 | drmatrix | i like that one |
06:13.21 | habakuk | bkw_, right, but I can't figure out how to set type=friend from a database. Unless I add a new field to my sipfriends table or something.. |
06:13.22 | herag | Kb1_Kanob: is the wait(2) before or after the answer? |
06:13.24 | Kb1_Kanob | drmatrix: you think far end is getting confused by the fast answer? |
06:13.29 | Kb1_Kanob | herag: before. |
06:13.35 | bkw_ | habakuk it implys friend |
06:13.35 | herag | drmatrix: I tried the removal of the ,r no good |
06:13.38 | bkw_ | no need to do type= |
06:13.40 | herag | Kb1_Kanob: ok, I'll try it |
06:13.45 | bkw_ | thats why its called "sipfriends" |
06:13.48 | drmatrix | yea... i had an extention that played a gsm instead of ringing... |
06:13.58 | drmatrix | worked for everyone but a few callers until i answer()ed |
06:14.29 | drmatrix | and if i remember right, it was particular cellular users... |
06:14.30 | habakuk | bkw_, but if you look closely at the code, it doesn't allow you to things like callerid and mailbox for example |
06:15.16 | herag | drmatrix: sorta like what I'm experiencing? |
06:15.25 | drmatrix | habakuk: and res_odbc doesn't do everything you want? |
06:15.32 | grrant | Thanks for you help I found out it was just a typo in extensions.conf |
06:15.34 | drmatrix | herag: it rhymes.... |
06:15.41 | grrant | instead of NXXNXXXX i had NXXNfNXXXX |
06:15.56 | bkw_ | habakuk the realtime does |
06:16.00 | herag | Kb1_Kanob: I tried the wait(2) like you said |
06:16.03 | bkw_ | MYSQL_FRIENDS no |
06:16.03 | grrant | Is there a way to setcallerID for each call you make without opening sip or reloading asterisk |
06:16.03 | herag | Kb1_Kanob: nothing |
06:16.05 | bkw_ | realtime YES |
06:16.09 | bkw_ | realtime supports all fields |
06:16.13 | bkw_ | you just add them to the database |
06:16.15 | bkw_ | and they are there |
06:16.23 | bkw_ | so if you want callerid |
06:16.26 | bkw_ | you add the field |
06:16.28 | bkw_ | and its there then |
06:16.31 | herag | drmatrix: your description really sounds like my problem...how did you get around it? |
06:16.31 | Kb1_Kanob | herag: and the sipura behaves for the same callers. Wierd. |
06:16.32 | habakuk | bwk_ at least callerid, as this is not part of a peer, but a user. |
06:16.42 | drmatrix | answer, wait, then send audio |
06:16.43 | herag | Kb1_Kanob: yes, the sipura works for everyone |
06:16.46 | bkw_ | I use friends and it works fine |
06:16.52 | bkw_ | the relatime friends that is |
06:17.31 | herag | drmatrix: would it matter if I do wait() answer, playtones... or answer, wait() playtones...? |
06:17.33 | habakuk | bkw_, hmm so is there a new struct defined? How do you combine attributes of user and peer? is what I'm getting at... |
06:18.11 | drmatrix | i dunno. i had to answer, then wait(2), then playtones |
06:18.24 | herag | hmm |
06:18.35 | herag | drmatrix: well, I'll try it |
06:18.40 | drmatrix | i've looked through it, and it's not in my current setup anymore, i was playing with stuff then |
06:18.42 | bkw_ | habakuk you must be misunderstanding the code or something |
06:18.48 | bkw_ | hold please |
06:18.52 | Kb1_Kanob | herag: sorry, you're beyond me with this one. You might need to talk to broadvoice for a trace but have a 100% non working cellphone handy. |
06:19.03 | herag | Kb1_Kanob: : / |
06:19.23 | herag | Kb1_Kanob: that's hard to manage...considering that it's not my cell that's experiencing the problem |
06:19.28 | drmatrix | the closest i have now is http://pastebin.ca/1359 |
06:19.44 | bkw_ | yep |
06:19.50 | bkw_ | habakuk you can do all that stuff with realtime |
06:19.59 | herag | drmatrix: and that works for those people that were having problems? |
06:20.01 | bkw_ | callerid and all settings |
06:20.17 | Kb1_Kanob | herag: yes, that can be a pain. But without repeatable fault telco debug isn't going to happen. :-/ |
06:20.29 | drmatrix | it works for everyone, with some hearing nothing, some hearing a full ring tone before connecting |
06:20.43 | drmatrix | 0.9 was the lowest i could put it |
06:21.05 | drmatrix | timing. |
06:21.31 | herag | ? |
06:21.40 | drmatrix | the answer to your wonders |
06:22.00 | habakuk | bkw_, you could be right. My point is now with the code there is mysql_user() and mysql_peer(). at registration time, it's always calling mysql_peer() only and doesn't grab the callerid from my database |
06:22.11 | herag | what does timing have to do witht his? |
06:22.13 | herag | *this |
06:22.36 | drmatrix | with the complications? or with your problem? |
06:22.38 | habakuk | bkw_, hmm maybe I should check out the realtime stuff. Is there a bug # I can reference? or is it already in CVS? (I'm using 1.0.1) |
06:22.43 | herag | my problems |
06:23.02 | drmatrix | i think your asterisk is too fast for some providers |
06:23.11 | herag | hmm |
06:23.14 | Kb1_Kanob | herag: wierd things start to happen with inter-carrier calls. Many things that would never be an issue otherwise crop up. you're probably spanning three carriers. |
06:23.17 | drmatrix | i had some callers that could hear music BEFORE i answered the phone |
06:23.28 | drmatrix | others would get disconnected |
06:23.35 | herag | haha ; P |
06:23.38 | drmatrix | and still others heard nothing until the answer |
06:24.08 | docelm0 | can someone tell me where I can get more info on how to setup the conf's in the database? I think it was res_config? |
06:24.22 | herag | I think the very fact that we can do this is causing so many problems on the administration on the telco side, I don' think they know what to do with all of it |
06:24.43 | drmatrix | res_config_odbc or res_odbc; not sure which |
06:24.46 | Kb1_Kanob | heh... that's an understatement. |
06:25.00 | docelm0 | thanks matrix.. |
06:25.14 | habakuk | drmatrix, I thought there were more options besides odbc ? |
06:25.16 | docelm0 | Now if I can only get ODBC to work on linux.. |
06:25.18 | docelm0 | ack.. |
06:25.27 | herag | those old monster bells need to release some docs that explain how their networks are configured so we can acutally interface with them...but they'll never do that, so we're just gonna have these fun problems |
06:25.33 | drmatrix | sure, if you want to read the source yourself |
06:25.53 | Kb1_Kanob | herag: oh, they do. |
06:25.53 | drmatrix | herag: id be scared if I were them |
06:26.12 | habakuk | drmatrix, is this checked into cvs yet, or is this still a patch? |
06:26.19 | drmatrix | uh, good question |
06:26.33 | drmatrix | i'm on CVS-HEAD-10/03/04-17:32:36, |
06:26.35 | Kb1_Kanob | herag: it's getting them without paying > $1k for a copy... |
06:26.52 | Kb1_Kanob | herag: http://www.voip-info.org/wiki-Telco+Engineering+Information |
06:26.52 | bkw_ | habakuk realtime is cvs-head only |
06:26.53 | herag | screw them...that should be public info |
06:27.04 | Kb1_Kanob | google is your friend. ;-) |
06:27.16 | drmatrix | but 1.0.0 has them both too |
06:27.29 | docelm0 | Anyone in here good with ODBC and Linux? |
06:27.32 | bkw_ | yes |
06:27.35 | habakuk | drmatrix: the one I was talking about was here: http://svn.asteriskdocs.org/res_data/ast_data/ not sure if this is the same thing you are referring to |
06:27.45 | bkw_ | docelm0 whats the problem? |
06:27.46 | docelm0 | You wrote the damn hack.. I hope so.. :) |
06:27.52 | bkw_ | hack? |
06:27.53 | bkw_ | what hack? |
06:27.55 | docelm0 | I cant get ODBC to work |
06:28.01 | bkw_ | haha |
06:28.02 | docelm0 | well.. res_odbc |
06:28.09 | bkw_ | I didn't write that |
06:28.13 | bkw_ | tony did |
06:28.15 | docelm0 | I thought you did.. |
06:28.15 | bkw_ | I wrote cdr_odbc |
06:28.15 | docelm0 | ohh |
06:28.23 | bkw_ | so I do know odbc |
06:28.28 | bkw_ | but what are you trying ot get working? |
06:28.32 | bkw_ | cvs-head or 1.0.1? |
06:28.35 | drmatrix | docelm0: whacha need, whatsit worth? |
06:28.46 | BoRiS | show me the money! |
06:29.00 | bkw_ | and in the recent changes in res_odbc and res_config_odbc |
06:29.12 | herag | s,1,dial() all by itself works... |
06:29.12 | drmatrix | bkw_: it's looking nice |
06:29.14 | drmatrix | (er) |
06:29.14 | herag | how is it working? |
06:29.21 | docelm0 | well either way.. I am figuring in stead of using the flat files to add/delete users I thought I would use the DB aspect.. Faster and my code to interface w/ it will be cleaner.. |
06:29.23 | herag | it must be sending something that the other sutff isn't |
06:29.24 | drmatrix | with the ,r? |
06:29.28 | herag | drmatrix: yes |
06:29.32 | docelm0 | but I want to dump my sip.conf and extensions.conf into the DB |
06:29.40 | docelm0 | so I can work on the fly.. |
06:29.51 | docelm0 | add users and dialplans etc.. |
06:30.03 | bkw_ | www.bkw.org/load.txt |
06:30.06 | drmatrix | try 1,answer,2,ringing;3,dial |
06:30.11 | bkw_ | that should load into ast_config for you |
06:30.25 | herag | ringing...what does ringing do? |
06:30.32 | bkw_ | change the indication |
06:30.32 | drmatrix | same as ,r |
06:30.33 | bkw_ | to ringing |
06:30.40 | bkw_ | no accually its not relly the same |
06:30.49 | bkw_ | Ringing can just provide ringing.. ie fake it |
06:30.52 | drmatrix | er, r just does ringtones? |
06:31.00 | herag | oh...hmm |
06:31.03 | bkw_ | you can smack a wait on that and play like you called someone |
06:31.05 | docelm0 | ok bk.. what am I supposed to do with this perl script? Is there anything I need to setup? install etc? |
06:31.07 | bkw_ | totally fool people |
06:31.12 | herag | what the difference between ringing and playtones(ring) ? |
06:31.18 | bkw_ | www.bkw.org/~brian/res_config.doc |
06:31.21 | bkw_ | have fun |
06:31.29 | bkw_ | er |
06:31.33 | bkw_ | www.bkw.org/~brian/doc/res_config.doc |
06:31.40 | bkw_ | BEEFCAKE!!! |
06:31.40 | *** part/#asterisk gregwood (~lynnwood@pcp01073653pcs.andrsn01.tn.comcast.net) |
06:31.41 | drmatrix | playtones is just noise; ringing is status. |
06:31.54 | bkw_ | well Ringing DOES NOTHING but fake it |
06:32.00 | bkw_ | ,r on a dial is accually dialing something |
06:32.05 | bkw_ | yes it does ringing |
06:32.25 | bkw_ | but its diffrent in the way you can just do Ringing with a wait and totally never dial someone |
06:32.36 | herag | well, it's worth a shot |
06:32.36 | bkw_ | but to the caller it sure would sound like it |
06:32.46 | drmatrix | what i do is immaterial, i'm talking about the channel's picture here |
06:33.13 | bkw_ | Thank you Mr. Mayor |
06:33.36 | drmatrix | it's a nitpicky thing, with a nitpicky difference |
06:33.41 | drmatrix | i've been bit by it |
06:33.58 | bkw_ | so why are you needing the r anyway? |
06:33.59 | bkw_ | whats up? |
06:34.00 | herag | drmatrix: I hope that it's nitpicky enough to make a difference ; P |
06:34.25 | drmatrix | i do too, because the headache isn't worth it |
06:34.28 | herag | why do I need the ,r? cause otherwise my callers think nothing's working and just hang up |
06:34.38 | bkw_ | and you're going out sip? |
06:34.40 | bkw_ | or iax? |
06:34.55 | drmatrix | sneaky |
06:34.59 | herag | bkw_: I don't know who you're talking to, but I'm going out sip |
06:35.05 | bkw_ | ok |
06:35.07 | docelm0 | so bk.. The perl script.. It will dump sip.conf or whatever conf I need to dump into the db? |
06:35.09 | bkw_ | going to cisco gear? |
06:35.17 | bkw_ | docelm0 yes |
06:35.27 | herag | bkw_: me? cisco? |
06:35.37 | bkw_ | herag yes going to a cisco gateway? |
06:35.40 | docelm0 | sweet.. this is gonna be MUCH FREAKING easier than what I have been racking my brains about.. |
06:35.46 | herag | bkw_: broadvoice -> asterisk linux -> sipura device |
06:35.54 | bkw_ | OH broadvoice |
06:35.59 | drmatrix | heh |
06:36.09 | bkw_ | have you looked at sip.conf.sample? |
06:36.10 | bkw_ | progressinband=no ; If we should generate in-band ringing always |
06:36.29 | herag | I don't have that setting in there |
06:36.31 | bkw_ | you should try that |
06:36.32 | herag | should I? |
06:36.34 | herag | hmm ok |
06:36.39 | bkw_ | in /usr/src/asterisk/configs/sip.conf.sample |
06:36.41 | bkw_ | it sure is |
06:36.42 | bkw_ | :) |
06:36.48 | bkw_ | ya I would try it |
06:37.01 | bkw_ | but if you don't get ringing then broadvoice needs to get that fixed |
06:37.06 | bkw_ | they might not be responding correctly |
06:37.11 | bkw_ | a sip debug would be most helful |
06:37.12 | bkw_ | er helpful |
06:37.24 | herag | bkw_: ? |
06:37.33 | herag | bkw_: my problem doesn't have to do with ringing |
06:37.50 | bkw_ | you said they don't hear anything right? |
06:37.52 | drmatrix | that may be why i needed my ringing. |
06:38.06 | bkw_ | no ringing? |
06:38.06 | herag | bkw_: no, no...their calls get dropped |
06:38.11 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-236-246.red.bezeqint.net) |
06:38.14 | bkw_ | god be clear about it then |
06:38.23 | BoRiS | lol |
06:38.26 | drmatrix | heh |
06:38.37 | bkw_ | its like pulling teeth in here sometimes.. |
06:38.42 | herag | bkw_: no offense, but I was...go back to the beginning, drmatrix and Kb1_Kanob are quite clear about my issue |
06:38.46 | drmatrix | lmao |
06:38.53 | riksta | with the default configuration of installing asterisk, can one test if it can receive a connection on iax2? i just want to see if i have set up my routers port forwards correctly, before i start to continue |
06:38.55 | bkw_ | you think I have time to read everything |
06:39.01 | bkw_ | this channels S/N is too high |
06:39.02 | BoRiS | then ask them instead. |
06:39.06 | drmatrix | herag: i think bkw_ does know where you are |
06:39.18 | bkw_ | I thought I knew exaclty what you were talking about .. but thats ok |
06:39.20 | drmatrix | just some details are missing |
06:39.27 | bkw_ | inbound not outbound right? |
06:39.31 | drmatrix | bkw_ his calls drop from some users inbound |
06:39.34 | herag | bkw_: yes, inbound only |
06:39.46 | bkw_ | ok if you get dropped calls we need a sip debug |
06:39.50 | bkw_ | of a dropped call |
06:39.56 | bkw_ | I suspect maybe its coming from a diffrent IP |
06:40.05 | drmatrix | i think the callers' providers are unhappy by call progress from broadvoice+herag's * setup |
06:40.07 | bkw_ | and not matching a context/peer and gets rejected maybe? |
06:40.12 | Kb1_Kanob | herag: ringing means two things - sound in channel (to indicate to user) and signal on channel (to indicate to far end pbx). The audio will be ignored by far end, but the signalling may be confusing things. |
06:40.26 | herag | o...hmm |
06:40.44 | Kb1_Kanob | bkw_: some cingular wireless inbound calls are disconnected but not all when answer() is in the dialplan. |
06:40.44 | herag | bkw_: it's only some cell phone users, specifically some t-mobile and some cingular users |
06:40.45 | bkw_ | accually I think its ip issues/peer matching |
06:40.58 | bkw_ | herag haha good luck I know whats wrong but you'll have hell fixing it |
06:41.16 | herag | bkw_: if I do strictly a s,1,Dial() it will work, but if I s,1,answer() first, it'll drop |
06:41.23 | Kb1_Kanob | bkw_: cross-carrier answer supervision? |
06:41.33 | bkw_ | herag sip inbound to sip? |
06:41.43 | herag | bkw_: huh? |
06:41.44 | bkw_ | s,1,Answer then playback a file doesn't work? |
06:41.50 | herag | no, that will drop |
06:41.56 | bkw_ | get a sip debug |
06:41.59 | bkw_ | something is'nt right then |
06:42.24 | BoRiS | what is your * version or cvs date? |
06:42.32 | herag | hmm, ok...I turned on sip debug at the command line, how do I get the debug? |
06:42.43 | herag | BoRiS: it's 1.0RC1 |
06:42.43 | bkw_ | make a call that gets dropped |
06:42.46 | bkw_ | OH GOD |
06:42.47 | bkw_ | get 1.0.1 |
06:42.56 | bkw_ | 1.0.1 has ALOT of fixes |
06:42.58 | BoRiS | always a GOOD idea to upgrade to the latest |
06:43.13 | BoRiS | (well, MOST of the time...hehe) |
06:43.25 | Kb1_Kanob | bkw_: has the v1-0 vs. v1-0-1 debate been resolved? |
06:43.32 | bkw_ | use v1-0 |
06:43.34 | bkw_ | as I said |
06:43.38 | bkw_ | nothing needed to be resolved |
06:43.40 | Kb1_Kanob | noted. |
06:43.52 | bkw_ | just people seem to not wanna pay any attention to me |
06:43.58 | bkw_ | but thats ok |
06:44.00 | bkw_ | :p |
06:44.04 | BoRiS | we luv you! |
06:44.07 | bkw_ | I know honey |
06:44.17 | bkw_ | luv you too |
06:44.20 | BoRiS | :) |
06:44.32 | bkw_ | guess people don't know how close I am to all the dev work on asterisk |
06:44.42 | bkw_ | and how involved I am in everything |
06:44.46 | bkw_ | haha |
06:44.53 | bkw_ | ya when I get going I might not make much sense |
06:44.58 | bkw_ | but I got alot of info |
06:45.07 | bkw_ | my fingers can't keep up with my mind sometimes :P |
06:45.11 | Kb1_Kanob | info is good. |
06:45.20 | cypromis | yes and no |
06:45.45 | cypromis | too much info about latest cvs changes |
06:45.53 | *** join/#asterisk af_ (~af@62.94.148.227) |
06:45.55 | cypromis | can make jo average user go nuts |
06:45.55 | cypromis | lol |
06:46.00 | bkw_ | ya really |
06:46.20 | Kb1_Kanob | but it's good to understand why they're happening. |
06:46.27 | Kb1_Kanob | and tidbits are better than nothing. |
06:46.39 | bkw_ | it does clue you in on what you might or might not like |
06:46.42 | habakuk | bkw_, yeah you're a great resource |
06:46.44 | bkw_ | you can get pointed in the right direction |
06:46.55 | bkw_ | which is what alot of people need |
06:46.55 | riksta | can someone test my IAX2 connection on 81.178.236.177 please? |
06:47.01 | bkw_ | need more info |
06:47.12 | bkw_ | IAX2/user@ip/exten |
06:47.16 | bkw_ | gotta have info |
06:47.26 | bkw_ | ie IAX2/guest@pbx.moosepenis.com/996 |
06:47.29 | bkw_ | :P |
06:47.31 | BoRiS | Hey! |
06:47.33 | bkw_ | aka pbx.bkw.org |
06:47.36 | BoRiS | Sssssssssssssh |
06:47.45 | riksta | ahh sorry, i dont know what to do, i've just installed asterisk, followed the FWD IAX setup guide |
06:47.52 | Kb1_Kanob | just the philosophy behind decisions is very helpful - like why log levels are split the way they are... |
06:47.55 | riksta | bkw_ what do i need to make an ext? |
06:48.09 | bkw_ | riksta you have alot of reading and playing to do |
06:48.19 | bkw_ | ok simple |
06:48.19 | bkw_ | iax.conf |
06:48.19 | bkw_ | you see [guest] |
06:48.22 | bkw_ | and it has context= |
06:48.32 | bkw_ | that context= matches [something] in extensions.conf |
06:48.33 | riksta | yes |
06:48.35 | bkw_ | which has an extension |
06:48.35 | bkw_ | ie |
06:48.42 | riksta | ok, i follow |
06:48.46 | riksta | context=default is mine |
06:48.51 | bkw_ | exten => s,1,Answer and exten => s,2,Playback(demo-congrats) |
06:48.53 | drmatrix | riksta: what's your fwd number then? |
06:48.59 | riksta | 479332 |
06:49.08 | bkw_ | riksta ok you're good then :) |
06:49.14 | BoRiS | exten => s,3,Playback(moosepenis) |
06:49.26 | bkw_ | show-me.gsm is better |
06:49.32 | bkw_ | www.bkw.org/show-me.gsm |
06:49.36 | riksta | bkw_ i don't understand that exten => command |
06:49.51 | bkw_ | riksta look at the examples included in the extensions.conf.sample |
06:50.00 | bkw_ | its alot to take in |
06:50.01 | herag | that's a lotta output, I'll have it pasted in a sec |
06:50.13 | riksta | should i set context= demo |
06:50.19 | bkw_ | herag update your asterisk first |
06:50.40 | bkw_ | show me on the dolly where the bad IVR developer touched you! |
06:51.03 | bkw_ | ok i'm sleepy |
06:51.05 | bkw_ | bed time |
06:51.06 | bkw_ | ttyl |
06:51.09 | riksta | Oct 10 07:50:46 WARNING[98311]: requested inkey 'freeworlddialup' for RSA authentication does not exist |
06:51.12 | riksta | i got this error |
06:51.23 | riksta | ahh wait i missed making the RSA key |
06:51.23 | drmatrix | nite, bkw |
06:51.37 | drmatrix | i guess from my call? |
06:51.47 | riksta | please can whoever called, try again |
06:51.52 | riksta | 65.39.205.121 |
06:52.05 | drmatrix | it's by your fwd number... still bad |
06:52.16 | riksta | huh? |
06:52.23 | riksta | i restarted asterisk just then |
06:52.26 | riksta | pls try |
06:52.31 | drmatrix | you are setting iax up with fwd, right? |
06:52.37 | riksta | yeah, trying to :S |
06:52.46 | herag | http://pastebin.ca/1361 this is the sip debug, anyone have a clue what went wrong? |
06:52.48 | riksta | Oct 10 07:52:41 NOTICE[98311]: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xff0 |
06:52.50 | drmatrix | so anyone that can call your fwd can check... |
06:52.59 | drmatrix | er, gsm not there? |
06:53.15 | riksta | do i have to enable gsm codec? |
06:53.23 | riksta | i think i allowed ulaw (if that is the same kinda thing) |
06:53.25 | BoRiS | allow=all in case |
06:53.27 | riksta | ok |
06:53.31 | drmatrix | i'm gsm preferred. |
06:53.38 | riksta | one moment pls |
06:53.50 | BoRiS | just for testing, when it works then you should be specific |
06:54.02 | riksta | ok restarted with allow=all |
06:54.17 | BoRiS | try that |
06:54.20 | drmatrix | busy |
06:54.28 | riksta | same error message |
06:54.45 | BoRiS | is their anything that says "disallow=all?" |
06:54.50 | riksta | sec |
06:54.51 | drmatrix | <PROTECTED> |
06:54.51 | drmatrix | <PROTECTED> |
06:54.51 | drmatrix | <PROTECTED> |
06:54.51 | drmatrix | <PROTECTED> |
06:55.00 | Kb1_Kanob | herag: what is s,3, currently defined as? |
06:55.16 | riksta | just dissallow=lpc10 |
06:55.36 | riksta | disallow* |
06:55.36 | herag | Kb1_Kanob: s,3,WaitExten(2) |
06:55.37 | drmatrix | what do you have for context= |
06:55.40 | riksta | demo |
06:55.51 | riksta | shall i change back to default? |
06:56.07 | drmatrix | can you set verbose? |
06:56.18 | riksta | oh hold on no thats wrong im sorry |
06:56.21 | riksta | i have for context |
06:56.26 | Kb1_Kanob | herag: note 'Spawn extension (incoming, s, 3) exited non-zero ....' followed by 'Destroying call 'SD30qg701-d6bc9152f0059d0aa2c8967461798787-js11002' |
06:56.28 | riksta | [fromiaxfwd] |
06:56.39 | riksta | and can i paste the 3 lines that are in extensions.conf? |
06:56.40 | BoRiS | make sure that is set in your context= |
06:56.52 | riksta | yes it is set in my context= |
06:56.53 | BoRiS | context=fromiaxfwd |
06:56.55 | docelm0 | you know if this actually works I am gonna be happy and amazed.. I have NEVER had any luck with unixODBC.. |
06:57.01 | herag | Kb1_Kanob: ya, I've been seeing that, what am I supposed to understand from it? |
06:57.07 | riksta | can i sho you the three lines in extentions.conf for [fromiaxfwd] ? |
06:57.09 | Kb1_Kanob | herag: that is the call being hung up by ast. note line 047 in pastebin. that is the matching ID. |
06:57.14 | BoRiS | yeah |
06:57.18 | riksta | exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r) |
06:57.18 | riksta | exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX} |
06:57.18 | riksta | exten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX} |
06:57.26 | herag | Kb1_Kanob: the cli keeps telling me that it blows up at the wiatexten(2) line...the question is why? |
06:57.30 | BoRiS | replace ${FWDNUMBER} with s |
06:57.41 | BoRiS | exten => s,1,Dial(${FWDRINGS},20,r) |
06:57.46 | herag | Kb1_Kanob: and why only for these certain callers? |
06:57.50 | riksta | i had FWDNUMBER defined above |
06:58.07 | Kb1_Kanob | herag: agreed. |
06:58.07 | riksta | can i just ask, the FWDRINGS, do i have to have some client set up connected into asterisk? |
06:58.20 | drmatrix | herag: try insecure=yes in your sip.conf. |
06:58.25 | riksta | i have this set FWDRINGS=sip/office ; the phone to ring what is sip/office |
06:58.35 | herag | drmatrix: it already is |
06:58.41 | drmatrix | hmph. |
06:58.49 | BoRiS | make sure office is a useraccount defined in sip.conf |
06:59.00 | drmatrix | FWDRINGS and FWDNUMBER need to be set in globals for that exten to work |
06:59.07 | riksta | ok do i make it like [office] or [sip/office] |
06:59.21 | drmatrix | what phone are you using with *? |
06:59.24 | BoRiS | [office] |
06:59.26 | herag | maybe it's just a bug, and I have to update...I think I've banged my head against the wall quite enough for tonight, I'll make the upgrade tomorrow |
06:59.27 | Kb1_Kanob | herag: before ast gives up gathering digits broadvoice is termianting the call (lines 047 Call-ID: SD30qg701-d6bc9152f0059d0aa2c8967461798787-js11002 |
06:59.29 | drmatrix | sip/office it is |
06:59.36 | riksta | im going to use a softphone, what do you recommend |
06:59.37 | Kb1_Kanob | oops. lines 135 onwards. |
07:01.12 | *** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net) |
07:01.51 | drmatrix | sjphone |
07:01.56 | drmatrix | but x-lite is pretty |
07:02.25 | drmatrix | Kb1_Kanob I was starting to wonder how bad my sip was |
07:02.29 | Kb1_Kanob | herag: bkw_ may have been refering to a silence detection issue w/broadvoice (http://lists.digium.com/pipermail/asterisk-users/2004-August/060340.html) but that's not your issue. |
07:02.43 | BoRiS | riksta: Maybe try setting this a test in your extensions.conf exten => s,1,Answer exten => s,2,Playback(demo-congrats), then dial your fwd # and you should hear something |
07:02.48 | Kb1_Kanob | drmatrix: oh? |
07:02.57 | drmatrix | wjat |
07:03.01 | drmatrix | what's the point of line 39? |
07:03.35 | BoRiS | (of course as long as you have your fwd setup in your extensions.conf for dialing other fwd users) |
07:03.36 | drmatrix | i don't recall completing a call when that happens |
07:03.36 | BoRiS | oy |
07:03.52 | Kb1_Kanob | drmatrix: indeed. broadvoice appears from multiple origins? |
07:04.10 | drmatrix | that's what bkw_ alluded to |
07:04.24 | Kb1_Kanob | Ahhh... yes. I see now. |