| 00:04.29 | laggo | why does setting my language=en-uk-female under the sip peer config change all sound files being played but not for SayNumber()? |
| 00:04.45 | laggo | must i really specify SayNumber(1,en-uk-female) every time? |
| 00:05.28 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 00:06.01 | dwery | hello. anyone has experience with xorcom usb banks? would you suggest them? |
| 00:06.18 | *** join/#asterisk ambush276 (n=ambush27@ip70-181-112-218.oc.oc.cox.net) |
| 00:06.20 | ambush276 | hey guys |
| 00:06.25 | ambush276 | any good SIP providers? cheap and US calling? |
| 00:06.33 | f0ner00t | Can you guys help me.. I reloaded my sip..and go the following.. See: http://pastebin.ca/1402398 |
| 00:06.39 | ambush276 | im trying to use voxox ? |
| 00:06.42 | ambush276 | anyone know that provider. |
| 00:06.48 | ambush276 | but outgoign calsl are not working? |
| 00:07.52 | [TK]D-Fender | f0ner00t: that isn't a real error, just ignore it |
| 00:08.24 | f0ner00t | [TK]D-Fender: Ok. |
| 00:11.02 | autobus | [TK]D-Fender its possible help me |
| 00:11.19 | autobus | how i make auth per ip but for one range ip/netmask |
| 00:11.26 | autobus | i test configuration: permit=ip/netmask and hostname=dynamic |
| 00:11.32 | autobus | this configuration is correctly for ip auth? |
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| 01:36.18 | VaGoNeTaS | is back from the dead. Gone: 1d 1h 39m 17s |
| 01:36.20 | VaGoNeTaS | hi |
| 01:36.26 | VaGoNeTaS | i need some help |
| 01:36.50 | VaGoNeTaS | i've just installed asterisk with redfone support |
| 01:36.58 | VaGoNeTaS | but when i do dahdi show status |
| 01:37.07 | VaGoNeTaS | doesnt show the correct line |
| 01:37.12 | VaGoNeTaS | i mean the E1 line |
| 01:43.17 | VaGoNeTaS | hello? |
| 01:49.01 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 01:57.34 | Qwell | VaGoNeTaS: turn off your public away message |
| 01:59.16 | VaGoNeTaS | is back from the dead. Gone: 10m 15s |
| 01:59.27 | VaGoNeTaS | dude, why are you here if you are not helping ? |
| 01:59.33 | eppigy | WOW DOG |
| 01:59.36 | VaGoNeTaS | i really need help with this |
| 01:59.37 | eppigy | calm sown |
| 01:59.39 | eppigy | down |
| 02:00.01 | VaGoNeTaS | ive just connected the E1 line to my redfone |
| 02:00.09 | VaGoNeTaS | actually there is 2 E1 lines connected to my E1 line |
| 02:00.20 | VaGoNeTaS | but dahdi is still being loaded as an Dummy |
| 02:05.08 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:05.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:06.00 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
| 02:06.18 | JT | VaGoNeTaS: seriously, disable those messages |
| 02:07.30 | tfrew | it's troy |
| 02:08.51 | drmessano | is a major douche: Spamming IRC with Away messages... <<uHH/WtF>> [ Douching, Page: Woof ] |
| 02:09.10 | *** kick/#asterisk [VaGoNeTaS!i=north@pdpc/sponsor/digium/Qwell] by Qwell (I said to disable the public away messages.) |
| 02:09.10 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 02:10.39 | drmessano | smells: Sour cream and failure... <<uHH/WtF>> [ Omelettes, Yum: Cheese ] |
| 02:11.30 | tfrew | hi Qwell, do you know of any good reference material in starting up a voip provider service, particularly I will be getting acsess to an ss7 connection, and at&t will be lighting up some dark fiber that runs into the building we are leasing for us. |
| 02:13.05 | tfrew | (using asterisk in a clustered/failover setup, with cdr billing for clients) |
| 02:13.30 | JT | this building you are leasing, is it a data centre? |
| 02:14.02 | tfrew | leasing |
| 02:14.08 | tfrew | it was to be a data center back in 2001 |
| 02:14.20 | tfrew | the company who deployed the equipment went bankrupt |
| 02:14.32 | tfrew | and per the lease agreement, the landlord gained control of everything |
| 02:14.59 | tfrew | and that is going to be ours to use |
| 02:15.02 | tfrew | per our lease |
| 02:15.22 | tfrew | including a link to washington dc which lands in an at&t datacenter |
| 02:16.15 | tfrew | i was handling the setup and maintence and sales of our pbx systems to small companies, but my boss would want us to become a full fledged voip provider with these resources |
| 02:16.23 | JT | does it have any facilities that are data centre-like? |
| 02:16.54 | tfrew | raised floors, generator, building wide ups, verizion, cavtel, sprint in building or nextdoor |
| 02:16.58 | tfrew | chillers, etc |
| 02:17.05 | tfrew | it was a 2001 style data center |
| 02:17.20 | tfrew | basically abandoned because the leaser went benkrupt |
| 02:17.53 | tfrew | even has another fiber link to aol (lol) |
| 02:18.01 | tfrew | back then that probably was usefull |
| 02:20.25 | JT | nice |
| 02:20.47 | JT | aol are big in transit these days aren't they? |
| 02:23.31 | *** join/#asterisk joobie (n=joobie@124-168-43-238.dyn.iinet.net.au) |
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| 02:29.20 | tfrew | in there network |
| 02:29.29 | tfrew | with thier peer agreements |
| 02:29.45 | tfrew | time warner and road runner aren't big around here |
| 02:30.31 | drmessano | wonders how good those 2001 UPS batteries are |
| 02:30.46 | tfrew | they are all being replaced heh |
| 02:31.01 | drmessano | The fuel sitting in the generator for 8 years |
| 02:31.04 | tfrew | its all -48volt dc |
| 02:31.05 | drmessano | and.... |
| 02:31.18 | tfrew | the generator was maintained by the landlord |
| 02:31.22 | drmessano | ok |
| 02:31.41 | drmessano | Loaded with awesome 1GHZ Xeons? |
| 02:32.40 | tfrew | no servers |
| 02:32.47 | drmessano | Ah |
| 02:32.48 | tfrew | those apperently where grabbed |
| 02:33.18 | tfrew | but the fiber equipment, the overhead rails, cooling, power, some *really* old cisco, and a rack of modem banks |
| 02:33.51 | drmessano | Sounds like little more than a room with cool old junk and lots of expense to modernize, IMO |
| 02:33.52 | eppigy | P3 XEONS |
| 02:35.21 | tfrew | what about the cost of running a fiber to washington dc |
| 02:35.51 | JT | do you need to run a fibre? |
| 02:36.55 | drmessano | With turning up service with a provider would include fiber termination at the Demarc, unless you had some very specific Point to Point link |
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| 03:00.36 | Aiatek | ~pb |
| 03:00.37 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
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| 03:56.24 | thebbxx2000 | If I need to downgrade from 1.6 to 1.4, should i just install over the existing files? Otherwise, how do I uninstall |
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| 04:12.26 | tfrew | thebbxx2000: are you using rpms? |
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| 05:01.45 | *** join/#asterisk Get_The_Fish (n=IceChat7@173.14.4.113) |
| 05:08.46 | Get_The_Fish | anyone here use opensbc before? |
| 05:31.59 | jeffspeff | how do i access the webvmail cgi page? i compiled it during install, the files are there, but what else do i need to configure to get it to work? |
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| 06:07.24 | joobie | guys one of my polycom 320 phones takes forever to boot |
| 06:07.36 | joobie | just 1.. like each stage it goes thru, it's like it takes tripple the time to go through |
| 06:07.59 | joobie | i dont have a new firmware to flash it with |
| 06:08.12 | joobie | so just thinking if there's something common that can cause this |
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| 06:23.44 | jplank | does anyone know how ARI pairs up a call to a specific call record? |
| 06:23.52 | jplank | I don't see any field in the CDR that shows it |
| 06:32.16 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 06:43.45 | jeffspeff | how do i access the webvmail cgi page? i compiled it during install, the files are there, but what else do i need to configure to get it to work? |
| 06:51.22 | kc8pxy | is there a known saturation ratio of B channels to D channels? (basically, how many B's can use a single D?) |
| 06:57.20 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:02.26 | Get_The_Fish | kc8pxy, it depends :).. some T's use NFAS (non facility associated signalling) and some use FAS... a typical, single T is 23 B to one D, however. |
| 07:02.31 | Get_The_Fish | http://www.voip-info.org/wiki/view/NFAS |
| 07:03.42 | Get_The_Fish | jeffspeff, I get there like this: http://<enter computer name or IP here>/recordings/index.php |
| 07:09.36 | kc8pxy | Get_The_Fish: assuming i have NFAS PRI's. 10PRI /D channel(as in the example at that link) is realistic? |
| 07:11.24 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 07:24.22 | Get_The_Fish | yes |
| 07:24.41 | Get_The_Fish | kc8pxy, if your equipment can support it. |
| 07:25.15 | Get_The_Fish | I have always heard 8 spans to a D, but that doesnt really mean anything |
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| 10:41.47 | proxium | hello, I have this message in CLI:No application 'Conference' for extension (default, 8600051, 1) |
| 10:41.47 | proxium | <PROTECTED> |
| 10:41.57 | proxium | How can I resolv this pb? |
| 10:53.09 | war9407 | had a quick question, I want all callers go to straight to voice mail, except if they come from certain numbers, does anyone have a extensions.conf/dialplan that implementss this that they could show as an example? |
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| 13:25.58 | medjr | does anyone know how to use SipShowPeerAction in asterisk-java ? |
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| 13:42.36 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:42.37 | dwery | hi. I've been consulted to check why a 2 lines ISDB asterisk doesn't tell the telco he's busy when all the channels are in use and another call is signalled. I'll see the system the next week, is there anthing in particular I must check? |
| 13:43.21 | riddlebox | ISDB? |
| 13:43.27 | dwery | oops.. ISDN :) |
| 13:43.54 | riddlebox | ok figured that, just wanted to make sure |
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| 14:08.32 | Get_The_Fish | so, do you think that it's possible that the carrier is trying to send an SIP ACK to a SIP 200 OK to the contact address instead of the from address? |
| 14:12.30 | Get_The_Fish | nevermind, found the answer. |
| 14:15.34 | *** join/#asterisk h-idrisi (n=h-idrisi@www.ppksadns.com) |
| 14:19.37 | tzafrir_laptop | dwery, I'm not sure I understand the problem from your description. Also: ISDN BRI or PRI? |
| 14:20.50 | dwery | tzafrir_laptop: BRI |
| 14:20.51 | h-idrisi | i have dtm caller id but some times it don't appear any help about this ? |
| 14:20.51 | dwery | tzafrir_laptop: two B channels busy, a new call is signalled on D |
| 14:21.19 | h-idrisi | * dtmf |
| 15:01.33 | *** part/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 15:05.05 | *** join/#asterisk _brent_ (n=_brent_@c-67-169-249-208.hsd1.ut.comcast.net) |
| 15:05.47 | kc8pxy | heya guys. I'm looking to fax via my asterisk server. am i correct that all i need is fxs port(in my asterisk box/ATA) and a fax machine?(an obviously, app_fax)?? |
| 15:05.59 | kumarphilly | morning |
| 15:08.27 | riddlebox | kc8pxy, do you have a pots line or a voip line |
| 15:09.36 | kc8pxy | riddlebox: the outbound link is sip. |
| 15:11.07 | kc8pxy | riddlebox: the other end of the sip will have a PRI. does that help? |
| 15:11.48 | riddlebox | you wont get good results |
| 15:12.30 | *** join/#asterisk ipstatic (n=ipstatic@74.197.158.19) |
| 15:12.59 | kc8pxy | riddlebox: what do i need for good results? |
| 15:13.09 | riddlebox | an fxs card in the asterisk server |
| 15:14.14 | riddlebox | but this may help |
| 15:14.16 | riddlebox | http://www.google.com/search?q=faxing+over+voip+asterisk |
| 15:15.39 | kc8pxy | riddlebox: ok, so you didn't read the whole question. so of the server/ata options, it should be in the server, not an aata |
| 15:16.37 | riddlebox | yes no ata, if you want it to be reliable it will need to be in the server,or a dedicated line |
| 15:19.12 | riddlebox | and i did read it all i said an fxs card in the server |
| 15:19.13 | *** join/#asterisk iq (n=chatzill@unaffiliated/iq) |
| 15:19.21 | iq | Hi |
| 15:20.22 | _brent_ | if i remember correctly, safe_asterisk keeps running until you stop it explicitly, right? i.e., it will show in a ps -ef |
| 15:20.26 | kc8pxy | riddlebox: and i won't need to worry that it's a branch-office's service, connecting to the main PRI-connected server? |
| 15:21.41 | _brent_ | i'm running 1.6 rc5 on centos 5.2 and safe_asterisk doesn't appear to keep running. has anyone seen this? |
| 15:22.05 | riddlebox | kc8pxy, so pri is at main site, fax is a remote |
| 15:22.17 | riddlebox | that wont be reliable |
| 15:23.11 | riddlebox | faxing over voip is not reliable, it is best to have a dedicated line imo |
| 15:23.25 | kc8pxy | riddlebox: why not?? noone's got anything like rfc2833 for fax yet? |
| 15:23.42 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 15:24.43 | riddlebox | kc8pxy, read stuff in that link i posted |
| 15:25.16 | riddlebox | voip traffic looses packets, fax transmission cannot loose that many packets |
| 15:25.41 | riddlebox | _brent_, sorry i dont touch 1.6 |
| 15:25.56 | ipstatic | hello all, after much googling, I just want to confirm, there should be no huge harm in compiling the newest version of Asterisk over an older one, as long as I make backups of the configs and the modules in /usr/lib/asterisk/modules correct? |
| 15:26.05 | _brent_ | riddlebox: do you know how safe_asterisk works, more or less? |
| 15:27.31 | riddlebox | _brent_, i know what its meant to do |
| 15:27.50 | _brent_ | riddlebox: if you run ps -ef on your machine, does safe_asterisk show up in the list of processes? |
| 15:28.03 | knarfly | I was using zaptel with *-1.2.23 on a FreeBSD server. I then switched to Fedora 10 with *-1.6.0.9 with Dahdi-2.1.0.4 Echo Cancellor |
| 15:28.03 | knarfly | when I make a call via Dadhi channel I see chan_dahdi ... unable to enable echo cancellation |
| 15:28.03 | knarfly | and of course there is an annoying echo when I use my sip phones.... |
| 15:28.06 | _brent_ | i think it's supposed to stay running, but it's exiting on me |
| 15:34.09 | kc8pxy | riddlebox: even t.38 (which upon reading, seems to be similar to rfc2833, in method) id "droppable" even with QoS to protect the link? |
| 15:34.50 | riddlebox | you can do as much as you can its just not reliable |
| 15:36.25 | riddlebox | i admin a 50k endpoint hospital with an avaya switch and we tell people to dial out and back in if they want to fax someone inside our network |
| 15:37.22 | _brent_ | ah, it appears that asterisk 1.6 always returns when you start it, but previous versions would present you with the asterisk CLI if you ran something like asterisk -vvvc |
| 15:37.53 | _brent_ | if this is the case, will safe_asterisk work at all? |
| 15:38.00 | _brent_ | it appears that it won't |
| 15:38.25 | _brent_ | even with the no fork (-f) flag, asterisk returns |
| 15:46.37 | riddlebox | _brent_, asterisk -vvvvvvvvvr |
| 15:46.54 | _brent_ | riddlebox: that works if asterisk is already running |
| 15:47.00 | _brent_ | (reattach) |
| 15:47.10 | riddlebox | ohh you want to start it |
| 15:47.19 | _brent_ | asterisk -vvvvvvvvvc should start it without forking and attach me to a console |
| 15:48.05 | riddlebox | yeah didnt realize what you were doing |
| 15:49.09 | _brent_ | yeah, safe_asterisk is exiting because the command `asterisk -f -vvvg` returns instead of not forking |
| 15:49.42 | riddlebox | hows the stability of 1.6 |
| 15:49.51 | _brent_ | so far, so good |
| 15:50.11 | _brent_ | i've been running the release candidates for about two months |
| 15:50.48 | riddlebox | i run two servers at my house and the last time i tried to use it, i couldnt get it to start |
| 15:50.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.23.211.242) |
| 15:51.24 | anonymouz666 | how to disable chan_sip of using ASTDB? |
| 15:51.33 | anonymouz666 | is there any way? |
| 15:51.46 | anonymouz666 | ASTDB is crashing my system, I don't need it. |
| 15:53.05 | *** join/#asterisk Urthwhyte (n=urthwhyt@0x5da320aa.cpe.ge-1-1-0-1101.oebrnqu2.customer.tele.dk) |
| 15:53.51 | *** join/#asterisk af_ (n=getsmart@88-149-240-185.dynamic.ngi.it) |
| 15:56.07 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:57.53 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 15:58.27 | anonymouz666 | [TK]D-Fender: do you know if it's possible to disable chan_sip of using ASTDB? |
| 15:58.51 | [TK]D-Fender | anonymouz666: "vi chan_sip.c" |
| 15:59.16 | anonymouz666 | that sucks |
| 16:01.08 | bbsf1 | I need help debugging why a SIP connection to Hong Kong (HKBN - 2b) ain't working (used to work before I tried to update everything over from Zap to DAHDI) |
| 16:01.45 | mmlj4 | that sounds strange |
| 16:02.16 | *** join/#asterisk ta^3 (n=tacvbo@200.95.162.52) |
| 16:02.21 | bbsf1 | been stumbling around for 2 days, now tearing out hair (which is becoming sparse) |
| 16:07.21 | mmlj4 | ok... let's pretend that DAHDI isn't your problem... what else changed? |
| 16:08.14 | mmlj4 | what version * were you using before, and now? |
| 16:08.33 | bbsf1 | I would also agree DAHDI isn't the problem. long story, short version - f10 system, I de-installed everything then re-installed pkgs (version 1.4 from atrpms). |
| 16:09.06 | mmlj4 | SIP works otherwise? |
| 16:09.20 | bbsf1 | SIP works ok with another supplier |
| 16:09.23 | *** join/#asterisk eric2 (n=ejc@i209-195-64-196.cia.com) |
| 16:09.40 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:09.52 | rob0 | sip debug peer |
| 16:09.58 | bbsf1 | HKBN register is ok, trouble comes on INVITE, which gets a 487 (Request Terminated) reply |
| 16:10.05 | eric2 | anyone get SLA working with snom phones? the process seems a bit convoluted to me |
| 16:10.12 | rob0 | hmm |
| 16:10.33 | mmlj4 | I need to read RFCs at some point, I suppose |
| 16:11.08 | *** join/#asterisk dieguito84 (n=diego@host247-195-dynamic.12-79-r.retail.telecomitalia.it) |
| 16:11.11 | bbsf1 | windoze machine using HKBN's windoze client connects ok, I've been trying to analyse what the differences are, but they are numerous. |
| 16:12.25 | bbsf1 | using nat with own-version router (all unchanged) |
| 16:13.33 | bbsf1 | tcpdump on router seems to show no other traffic between INVITE and 485 response, which leads me to concentrate on the INVITE packet. |
| 16:13.44 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 16:13.53 | seb- | [TK]D-Fender: hello!? you there? |
| 16:14.22 | mmlj4 | the HK end... anything changed there? or where is your * box? |
| 16:14.53 | seb- | [TK]D-Fender: sending pm now |
| 16:15.10 | bbsf1 | no change to HK that I know of, but they're a kinda funny organization (Nortel system) - * box is in San Francisco |
| 16:15.40 | mmlj4 | nortel-- |
| 16:16.17 | mmlj4 | <bbsf1> SIP works ok with another supplier |
| 16:16.36 | mmlj4 | where is that supplier? HK? maybe I'm not understanding what your setup is |
| 16:16.48 | mmlj4 | you're a provider of sorts? |
| 16:17.34 | bbsf1 | SIP works OK with Broadvoice. HKBN is a HK supplier, my * system is just a home pbx system. |
| 16:18.46 | mmlj4 | so... SIP with broadvoice works, HKBN doesn't... |
| 16:18.51 | bbsf1 | correct |
| 16:19.00 | mmlj4 | can you connect out-of-band to HKBN? |
| 16:19.59 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 16:21.38 | bbsf1 | that's a little complicated. They use a proxy system, domain is s2hkbntel.net (a fishy name), must go through proxy which is s22.hkbntel.net. |
| 16:22.03 | mmlj4 | yeah... |
| 16:22.25 | bbsf1 | but since the registration is all working, I'm assuming that part is OK. |
| 16:22.25 | mmlj4 | great firewall of china? |
| 16:22.36 | bbsf1 | no, HK is pretty open (so far) |
| 16:22.45 | mmlj4 | so far |
| 16:23.15 | bbsf1 | normally I live there, SF is temporary location for me. |
| 16:23.20 | mmlj4 | ah, ok |
| 16:23.49 | mmlj4 | and you have no * gear at the HK end? |
| 16:23.57 | bbsf1 | no |
| 16:24.23 | mmlj4 | or can you revert back to your previous zaptel setup? |
| 16:24.36 | shyam_k | i just got a pci card with two phone jacks, one labeled as "phone" and other labelled as "line" i don't know which model or what kind of card it is;-) but i have plugged it into the system and then lspci should say about that right? i didn't see anything on that.. |
| 16:24.38 | mmlj4 | did you change * versions? |
| 16:24.57 | mmlj4 | shyam_k: theoretically |
| 16:25.17 | mmlj4 | shyam_k: but unless it's rated to do asterisk stuff... it won't. evar. |
| 16:25.46 | mmlj4 | shyam_k: chances are, what you have is known as a "modem" |
| 16:26.02 | shyam_k | its a digium card bought two years ago by my prof but he hasn't touched it yet though |
| 16:26.41 | mmlj4 | oh? then pop it in and boot... /var/log/messages ought to see it, as will dmesg and lspci |
| 16:26.52 | mmlj4 | xp100? |
| 16:27.09 | mmlj4 | but it does say "digium"? |
| 16:27.36 | shyam_k | mmlj4: oh and that may not qualify it as an fxo then? |
| 16:27.55 | bbsf1 | mmlj4: now you're getting into some nebulous area for me - before I de-installed the previous installation I copied the whole /etc/* directory; I didn't note the precise version number which was de-installed. Reverting to that previous directory now (and changing all 'Zap' to 'DAHDI') gets me to the same point (everything works except HK) |
| 16:27.58 | mmlj4 | shyam_k: does it actually say "digium" |
| 16:28.10 | Talkradio | buy real equipment and you'll be much happier in the end |
| 16:28.45 | mmlj4 | bbsf1: ah, and ugh |
| 16:29.11 | bbsf1 | yeah - if you heard some of the rest of the "long story" you'd be even more depressed.... |
| 16:29.18 | mmlj4 | heh |
| 16:29.23 | shyam_k | mmlj4: hmm..actually i just have that board which doesn't have any kind of name in it.. say one on board chip is of motorola.. and cat /var/log/messages | grep "digium" says nothing |
| 16:30.01 | mmlj4 | so... what do you really have to accomplish? you want to have your HK line ring in SF while you're on furlough? and place calls to HK? |
| 16:30.21 | bbsf1 | and receive calls from numerous HK callers. |
| 16:30.46 | bbsf1 | yeah, precisely. |
| 16:31.13 | bbsf1 | just like what I get with the SIP from Broadvoice, or the DAHDI connection to Ma Bell. |
| 16:31.18 | mmlj4 | shyam_k: you have a cheap knock-off.... it /might/ work, but you're better off getting a linksys or other ATA (but NOT A GRANDSTREAM), and using that intstead |
| 16:31.53 | mmlj4 | bbsf1: ok... get you a working HK number... trivial, many many providers exist... you admitted broadvoice does |
| 16:32.23 | bbsf1 | no good for HK people calling me. |
| 16:32.31 | mmlj4 | transfer your HK line to ring your new HK number... bing |
| 16:32.38 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096591569.dsl.bell.ca) |
| 16:32.38 | mmlj4 | solved |
| 16:32.47 | mmlj4 | want to call to HK? broadvoice |
| 16:33.21 | coppice | If I want to call to HK I just open the window and yell |
| 16:33.59 | mmlj4 | again, the key it getting your HK phone to transfer to another HK number (the broadvoice DID) |
| 16:34.21 | mmlj4 | coppice: rent yourself out as a proxy |
| 16:35.25 | bbsf1 | calling from SF to HK is certainly not a problem. It's the other direction (HK people who are not computer-literate) able to contact me by calling a (HK) local number. The HKBN 2b service has worked fantastically well for the last couple of years, and should still work (like I said, it works fine from my Windoze system, just not from my * system) |
| 16:36.06 | bbsf1 | except I (somehow) screwed it up and now can't seem to figure out how to fix it. |
| 16:36.20 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 16:36.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:36.40 | shyam_k | mmlj4: yeah it matches with that photo from digium site explaining those cheap hardware |
| 16:36.49 | mmlj4 | hope you get it straight :-) |
| 16:36.57 | bbsf1 | And I take the point that HKBN's Nortel system does some nasty things, but * used to be able to circumvent it. |
| 16:36.58 | shyam_k | with lots of electrolitic capacitors and all.. |
| 16:37.08 | mmlj4 | now, I have to take a bath using 1 gallon of water from a jug... backhoes ate our water line this morning (and took out POTS to the neighborhood, too) |
| 16:37.40 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
| 16:38.22 | my007ms | i have skype mobile phone any idea allow me to use it with my asterisk PBX |
| 16:44.19 | drmessano | If the phone doesnt do SIP, then no |
| 16:44.41 | my007ms | :( it's does not |
| 16:44.54 | my007ms | i was think i can create something like skype trunk or something |
| 16:45.51 | drmessano | If the Skype channel driver was out you could, but it is not |
| 16:46.06 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 16:47.54 | my007ms | drmessano Skype not used normal sip thy have thy own protocol ?? |
| 16:48.01 | drmessano | Correct |
| 16:48.17 | drmessano | Closed proprietary protocol |
| 16:55.56 | eppigy | they use encruption and stuff |
| 16:56.01 | eppigy | encryption |
| 17:01.14 | *** join/#asterisk eliel (n=eliels@120-17-235-201.fibertel.com.ar) |
| 17:14.09 | *** join/#asterisk DavidR2008 (n=chatzill@nc-76-4-4-127.dhcp.embarqhsd.net) |
| 17:18.57 | Urthwhyte | Skype is like IAX on firewall worming steroids |
| 17:27.04 | *** part/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 17:29.35 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 17:29.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:36.35 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:46.27 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 17:51.02 | *** join/#asterisk Diblo (n=chatzill@0x573fc6ce.cpe.ge-1-1-0-1101.arcnqu1.customer.tele.dk) |
| 17:54.26 | Diblo | hey I'm looking for a web ext for asterisk that works as a call me button |
| 17:54.57 | Diblo | some like a webphone |
| 18:00.35 | Urthwhyte | Diblo: You mean so clients can call you directly from their web browser? |
| 18:01.13 | Diblo | Yes wit a click and no software |
| 18:01.44 | Urthwhyte | http://blog.tringme.com/tringme-introduces-flash-enabled-web-based-sip-phone-tringphone/ |
| 18:02.08 | Urthwhyte | Looks like it'd do what you want. |
| 18:02.32 | *** join/#asterisk crevetor (n=crevetor@bas1-montreal07-1176434423.dsl.bell.ca) |
| 18:02.42 | crevetor | hi there |
| 18:02.47 | Diblo | What I am really looking for is a Support System (call and chat or chat) |
| 18:02.57 | Diblo | Ok |
| 18:03.10 | crevetor | I was wondering if anybody had heard about issues when using asterisk with ethernet bonding (using 802.3ad) |
| 18:03.44 | Urthwhyte | Diblo: You should be able to tie something together in a web page fairly easily |
| 18:04.23 | Diblo | Jep nice thanks for the help :) |
| 18:08.06 | [TK]D-Fender | crevetor: as long as its counted as a single interface by the time its done, * shouldn't care |
| 18:09.16 | crevetor | [TK]D-Fender: It doesn't have a problem using it (We have 4 * boxes that use this setup) but I was wondering if anybody had had weird issues such as dropped calls, etc due to this... |
| 18:10.00 | [TK]D-Fender | crevetor: That would boil down to "is my link busted all by itself?". |
| 18:17.32 | Urthwhyte | [TK]D-Fender: What's your opinion on using SIP trunks for International calls? Bad idea to stay away from, or something that would be fine for <5 users? |
| 18:17.54 | DavidR2008 | [TK]D-Fender: I got my redirect thing working. You suggested using AMI Redirect and that worked. I had to apply a patch to chan_sip to add the SIPRemoveHeader application and I had to background the execution of the AMI script. thanks again |
| 18:17.56 | [TK]D-Fender | Urthwhyte: its as reliable as your internet connection and your provider |
| 18:18.20 | [TK]D-Fender | Urthwhyte: Its fine |
| 18:18.33 | Urthwhyte | [TK]D-Fender: Any suggestions for a non-cheap provider? |
| 18:18.48 | [TK]D-Fender | Urthwhyte: Depends on several things. |
| 18:19.15 | Urthwhyte | [TK]D-Fender: Need information on usage first to make a reccomendation? |
| 18:19.42 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 18:20.17 | seb- | [TK]D-Fender: can i ask you another question? i tried MeetMe and it says 'that is not a valid conf #" |
| 18:20.32 | Urthwhyte | seb-: Do you have ztdummy or zaptel hardware installed? |
| 18:20.41 | [TK]D-Fender | seb-: ^ |
| 18:21.30 | seb- | Urthwhyte: i read that in docs but i naively hoped ubuntu package would be smart nuff to include that by default? |
| 18:21.30 | [TK]D-Fender | seb-: that can be a misleading error for not having installed zaptel/dahdi which is required for mixsing |
| 18:21.30 | seb- | hides |
| 18:21.30 | Urthwhyte | seb-: What version of * are you running? |
| 18:21.30 | [TK]D-Fender | seb-: dahdi/zaptel are separate packages |
| 18:21.39 | seb- | <PROTECTED> |
| 18:21.43 | seb- | Urthwhyte: ^ |
| 18:22.33 | seb- | I installed zaptel..i believe it needs a conf file it doeesn't provide for you |
| 18:22.39 | seb- | /etc/zaptel.conf perhaps |
| 18:22.40 | Urthwhyte | run lsmod |
| 18:23.00 | seb- | Urthwhyte: nothing w/ z in name when do lsmod |
| 18:23.06 | Urthwhyte | what about dahdi? |
| 18:23.14 | seb- | no |
| 18:24.14 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 18:24.20 | seb- | Urthwhyte: Zaptel telephony kernel driver: FATAL: Module ztdummy not found. |
| 18:24.52 | eppigy | why would you not compile zaptel from source? |
| 18:24.54 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 18:25.07 | seb- | eppigy: why would you not install the ubuntu package? |
| 18:25.17 | eppigy | seb-: scroll up |
| 18:25.20 | eppigy | and look |
| 18:25.46 | seb- | Urthwhyte: there doesn't seem to be ANY ubuntu package that mentions ztdummy |
| 18:25.49 | eppigy | YEAH WHAT DO YOU HAVE TO SAY NOW? |
| 18:26.48 | seb- | eppigy: heh |
| 18:26.53 | drmessano | eppigy: HE TOLD YOU |
| 18:27.12 | Urthwhyte | seb-: What about DAHDI? |
| 18:27.14 | eppigy | I WON AN ARGUMENT ON THE INTERNET |
| 18:27.15 | Urthwhyte | that's the new name of Zaptel |
| 18:27.25 | seb- | Urthwhyte: oh you're joking |
| 18:27.40 | Urthwhyte | wut? |
| 18:27.45 | drmessano | seb-: Packages for an application with less components are one thing, but people seem to only know how to screw up Asterisk packages |
| 18:27.56 | seb- | Urthwhyte: name changes confuse the money spending customers |
| 18:28.08 | drmessano | Umm |
| 18:28.22 | drmessano | Dahdi is much better at this point than Zaptel |
| 18:28.26 | seb- | Urthwhyte: ztdummy and dahdi are not in ubuntu..guess i need to d/l the source like eppigy says |
| 18:28.39 | eppigy | BOOYA |
| 18:28.41 | drmessano | Install Dahdi |
| 18:28.49 | drmessano | Dont do a new install with Zaptel |
| 18:29.00 | mmlj4 | define "better" |
| 18:29.13 | drmessano | Progressed.. more stable |
| 18:29.16 | drmessano | Better tools |
| 18:29.25 | seb- | How in the fruitcake did Ubuntu's Zaptel *NOT* have ztdummy innit? |
| 18:29.32 | seb- | who's running the show over there? |
| 18:29.38 | drmessano | seb-: because not everyone needs it |
| 18:29.56 | drmessano | seb-: Same argument as EVERY fuxored up Asterisk package attempt |
| 18:30.04 | eppigy | they probably assumed if you neet zaptel you need it for zap type interfaces |
| 18:30.07 | drmessano | seb-: Its whatever the packager decided |
| 18:30.10 | eppigy | *need |
| 18:31.53 | Urthwhyte | before you start though, this isn't running in a VM, is it? |
| 18:31.58 | seb- | no |
| 18:32.31 | *** join/#asterisk bbryant (n=bbryant@m4d5336d0.tmodns.net) |
| 18:32.36 | Urthwhyte | Just grab the source and compile and you should be good to go then |
| 18:32.49 | seb- | Urthwhyte: thanks..why did they do a name change? |
| 18:32.57 | mmlj4 | drmessano: do you don't know of any non-whacked package? |
| 18:33.00 | Urthwhyte | seb-: Lawsuit I think |
| 18:33.08 | mmlj4 | drmessano: slackware's sbo package seems to work for mr |
| 18:33.10 | mmlj4 | me |
| 18:33.13 | seb- | ah ok..well that's a good reason |
| 18:33.33 | mmlj4 | but lenny's only dumps core for me |
| 18:33.47 | drmessano | mmlj4: yes, works for YOU.. does it include every module and leave ANYTHING out that I would get from source? Probably not |
| 18:34.16 | mmlj4 | it's rather lightweight, actually... few files in /etc/asterisk |
| 18:34.17 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:34.31 | drmessano | mmlj4: yes, so that just backs up what I said |
| 18:34.36 | mmlj4 | yeah |
| 18:35.02 | drmessano | Personally, I like the kitchen sink |
| 18:35.06 | mmlj4 | i need to look at suse's again |
| 18:35.25 | eppigy | man |
| 18:35.31 | drmessano | I use Chan Mobile, I use jabber integration, I drop all the sound files in so I can limit some of the transcoding |
| 18:35.33 | eppigy | compilign form source is not hard |
| 18:35.35 | drmessano | I do a lot of things |
| 18:35.41 | eppigy | i mean whats the deal |
| 18:35.49 | drmessano | Its not hard |
| 18:35.52 | Urthwhyte | eppigy: wget and make install are hard! |
| 18:35.56 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 18:35.59 | eppigy | :[ |
| 18:36.08 | drmessano | OMG MAKE MENUSELECT.. TOO MANY CHOICES |
| 18:36.19 | eppigy | like |
| 18:36.28 | eppigy | many daemons |
| 18:36.29 | drmessano | "SIP? Hell no, i just want asterisk" |
| 18:36.31 | eppigy | and server things |
| 18:36.39 | eppigy | itis a great idea to use a package |
| 18:36.40 | mmlj4 | eppigy: no, it's not hard... but on mutliple production servers, I don't want to hassle when updating... I want to run my little doupdate script and reboot afterward if needed, and have everything Just Work(tm) |
| 18:36.45 | eppigy | for ease of administarttion |
| 18:36.52 | eppigy | you can use puppet |
| 18:36.52 | drmessano | svn update |
| 18:36.57 | drmessano | make install |
| 18:36.57 | eppigy | and all sorts of good stuff |
| 18:37.02 | mmlj4 | puppet? |
| 18:37.15 | eppigy | haha |
| 18:37.20 | eppigy | yeah dude |
| 18:37.22 | eppigy | google it |
| 18:37.29 | mmlj4 | you google it |
| 18:37.35 | eppigy | im not asking abotuit |
| 18:37.41 | drmessano | YOu asked HIM about it |
| 18:37.47 | drmessano | Why would HE google it? |
| 18:37.53 | eppigy | mmlj4 is way too defensive |
| 18:37.58 | drmessano | Vista user |
| 18:38.00 | drmessano | My guess |
| 18:38.30 | eppigy | and if you really have that many * instances |
| 18:38.38 | eppigy | then you make your own packages and repository |
| 18:38.41 | mmlj4 | I've got that many clients |
| 18:38.46 | eppigy | so dont give me that excuse either |
| 18:39.07 | mmlj4 | that's what SBO is for |
| 18:39.35 | seb- | eppigy: will dahdi work w/ asterisk 1.1.4 w/o much fuss? |
| 18:39.43 | eppigy | what |
| 18:39.49 | seb- | eppigy: old * doesn't expect old zaptel? |
| 18:39.57 | eppigy | im pretty sure it does |
| 18:40.02 | drmessano | 1.4.22 is minimum for Dahdi |
| 18:40.05 | drmessano | I believe |
| 18:40.21 | drmessano | Newer 1.4 will work with Dahdi |
| 18:40.40 | eppigy | seb-: are you doing a new install? |
| 18:41.01 | drmessano | If youre gonna install the latest 1.4, there is no issue |
| 18:41.21 | eppigy | yeah i was hoping that was a transposed number |
| 18:41.39 | drmessano | If he had 1.1, I want a copy |
| 18:42.15 | drmessano | Just like when someone told me they had 1.5 |
| 18:42.28 | seb- | eppigy, drmessano: i have 1.1.x...that is * version in latest Ubuntu Jaunty |
| 18:42.45 | seb- | eppigy: i'm hoping zaptel source install works w/ * ubuntu package |
| 18:42.49 | [TK]D-Fender | hordes his copy of res_fluxcapacitor.so |
| 18:42.53 | eppigy | i dont know what to say bro |
| 18:42.59 | seb- | eppigy: what do you mean? |
| 18:43.05 | drmessano | There is no 1.1 |
| 18:43.49 | seb- | (server /home/seb) % dpkg -l | grep aster |
| 18:43.50 | seb- | ii asterisk 1:1.4.17~dfsg-2ubuntu1 Open Source Private Branch Exchange (PBX) |
| 18:44.00 | drmessano | God |
| 18:44.07 | drmessano | Thats not... |
| 18:44.09 | seb- | so ubuntu b0rked the name of their package too |
| 18:44.11 | drmessano | Thats 1.4.17\ |
| 18:44.21 | drmessano | Notice the COLON |
| 18:44.28 | eppigy | THATS WHAT SHE SAID |
| 18:44.29 | seb- | drmessano: :) |
| 18:44.30 | keith4_ | is there a simple way to test if a string is numeric? |
| 18:44.41 | seb- | shoot me now |
| 18:44.48 | drmessano | 1.4.17 is old and broken |
| 18:44.50 | eppigy | seb-: are you using ext4? |
| 18:44.56 | drmessano | You need more recent from source |
| 18:44.59 | seb- | eppigy: ext3 |
| 18:45.02 | eppigy | why dog |
| 18:45.10 | eppigy | i mean you uprade to juanty |
| 18:45.13 | eppigy | and dont use ext4 |
| 18:45.16 | eppigy | whats the deal |
| 18:45.23 | drmessano | 1.4.17 through 19 had problems |
| 18:45.26 | seb- | eppigy: i didn't know how stable it was so i passed on it |
| 18:45.28 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 18:45.33 | eppigy | :[ |
| 18:45.45 | eppigy | it was declared stable for that kernel release |
| 18:45.47 | eppigy | come on dog |
| 18:46.05 | seb- | eppigy: what does ext4 have to do w/ *? |
| 18:46.11 | eppigy | nothing |
| 18:46.15 | eppigy | i am just nerdin out |
| 18:46.39 | eppigy | well I mean of course |
| 18:46.50 | eppigy | unless you mean filesystem performance |
| 18:47.01 | eppigy | like if you are using a huge realtime database |
| 18:47.18 | eppigy | or recording a shit load of channels |
| 18:47.19 | drmessano | Im more concerned about this package business |
| 18:47.37 | seb- | drmessano: ? |
| 18:47.40 | eppigy | lets get down to brass tax |
| 18:47.50 | drmessano | You dont need to be using 1.4.17 |
| 18:47.53 | drmessano | Grab source |
| 18:47.56 | drmessano | Install |
| 18:47.57 | drmessano | Be happy |
| 18:48.06 | eppigy | insert ubuntu disk |
| 18:48.07 | eppigy | reboot |
| 18:48.09 | [TK]D-Fender | eppigy: No New Taxes! |
| 18:48.17 | [TK]D-Fender | dumps all of his brass into the river |
| 18:48.21 | eppigy | D: |
| 18:48.31 | seb- | drmessano: is it as easy as emacs install? i.e. configure ; make ; make install? |
| 18:48.45 | drmessano | Pretty much |
| 18:49.02 | drmessano | You need to install Dahdi first |
| 18:49.15 | seb- | eppigy: looks like dahdi is only available from subversion...there is no feel good "stable" tarball |
| 18:49.22 | eppigy | huh? |
| 18:49.27 | drmessano | uh huh? |
| 18:50.04 | eppigy | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/dahdi-linux/releases/dahdi-linux-2.1.0.4.tar.gz |
| 18:50.05 | drmessano | http://downloads.digium.com/pub/telephony/dahdi-linux-complete/ |
| 18:50.09 | drmessano | Really now? |
| 18:50.10 | eppigy | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/dahdi-tools/releases/dahdi-tools-2.1.0.2.tar.gz |
| 18:50.17 | eppigy | http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/libpri/releases/libpri-1.4.9.tar.gz |
| 18:50.22 | eppigy | you are trippin me out dog |
| 18:50.27 | eppigy | i mean whats the deal |
| 18:50.29 | drmessano | This isnt apache |
| 18:50.35 | drmessano | Its not a toaster |
| 18:50.42 | seb- | eppigy: the prophets at voip-info.org saw fit to only mention the svn deal... |
| 18:50.48 | eppigy | man |
| 18:50.48 | drmessano | Do a little work here |
| 18:50.57 | eppigy | put asside your foolish wiki |
| 18:50.58 | drmessano | voip-info is old and sucky documentation |
| 18:51.13 | drmessano | That site should have a server crash |
| 18:51.32 | seb- | heh...here i am all this time thinking that's where all the cool kids and gurus go |
| 18:51.46 | drmessano | Um no |
| 18:52.07 | seb- | in my defense..can i mention voip-info.org is the first hit on a dahdi google search? |
| 18:52.42 | drmessano | Yeah, why go to DIGIUMS site for Dahdi info.. they're only the DEVELOPER |
| 18:53.04 | drmessano | What the hell do they know, right? |
| 18:54.42 | seb- | lol |
| 18:56.48 | seb- | drmessano: what were these problems 1.4.17 had? |
| 18:56.56 | keith4_ | does REGEX support \d and \D ? |
| 18:57.58 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 18:58.53 | drmessano | seb-: There were lots of issues with early and mid 1.4 releases.. Im not here to convince you to use the latest Asterisk or Dahdi.. if youre hell bent on using the packages from your distro, jsut do it.. If you want more stable, secure, and current code, and those things concern you at all, you need source |
| 19:01.53 | seb- | drmessano: ok...i'll do the source for * and dahdi....can i ask 1 more thing...did you say you must install the dahdi source first before the * source for some reason? |
| 19:02.10 | drmessano | Yes you need it there for Asterisk to install with Dahdi support |
| 19:02.14 | drmessano | Otherwise it does NOT work |
| 19:02.26 | seb- | 11:49 < drmessano> You need to install Dahdi first |
| 19:02.35 | seb- | ah ok so i understood you right |
| 19:02.35 | seb- | thanks |
| 19:02.48 | seb- | drmessano: you helped a lot |
| 19:02.51 | seb- | i appreciate it |
| 19:03.38 | seb- | crap they have asterisk 1.6 now |
| 19:04.04 | drmessano | Yeah |
| 19:04.09 | drmessano | Im using 1.6 |
| 19:05.04 | drmessano | I've been happier with it, and keep hearing reports of things working better on 1.6 |
| 19:05.15 | drmessano | But arguing asterisk versions is like arguing religion |
| 19:06.02 | seb- | drmessano: hey if it is newer it must be better and more stable |
| 19:06.08 | drmessano | IMO, 1.4 was never as stable as 1.2.. and 1.6 has proven to be more stable than both from reports and personal experience with it |
| 19:06.18 | seb- | drmessano: i'll use that one instead |
| 19:06.24 | seb- | drmessano: unless you disapprove |
| 19:06.28 | drmessano | Things seem to work as they're supposed to in 1.6 |
| 19:06.40 | drmessano | I think 1.6 is fine |
| 19:07.35 | seb- | drmessano: wow..this will be awesome..a solid 1.6 * and dahdi from source...now that's cool |
| 19:07.39 | drmessano | A lot of people are jaded from the issues 1.4 had early on.. some idea that 1.6 must be out for ages before its worth using.. Ive not seen any such issues |
| 19:08.07 | drmessano | Except one little buggy with SIP TCP... but thats workign great now |
| 19:08.11 | drmessano | heh |
| 19:13.11 | DavidR2008 | drmessano: are you using PRIs with 1.6, if so what cards? |
| 19:13.34 | drmessano | Nope |
| 19:14.07 | drmessano | I'm sure anything working with 1.4 would work with 1.6 |
| 19:14.16 | drmessano | Thats Dahdi and Libpri |
| 19:19.44 | *** join/#asterisk Deeewayne (n=dwayne@95.104.105.2) |
| 19:19.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 19:23.39 | Qwell | drmessano: which is funny, because it was a huge part of why we moved to the new release process |
| 19:24.05 | Qwell | it sucked because nobody used it before it was released. well...of course it's going to have bugs if nobody actually uses it |
| 19:24.12 | Qwell | (1.4 that is) |
| 19:26.19 | drmessano | Yeah.. there was a lot of "Let me try 1.4... Hm ok, this is broke.. going back to 1.2" |
| 19:26.44 | Qwell | when 1.2 was released, we had to encourage people to use it instead of trunk |
| 19:27.14 | Qwell | back then, everybody ran trunk.. then we finally convinced people to not use it...then nobody did any testing before 1.4 was released... |
| 19:27.46 | Qwell | now we get the best of both worlds with 1.6 :D |
| 19:27.52 | drmessano | Yep |
| 19:28.37 | drmessano | Its good that 1.6 started off better and theres been very specific reports of things working better in 1.6 |
| 19:29.04 | drmessano | Those things go a long way |
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| 20:43.01 | eric2 | anyone know hot to fix this message? wanpipe1:w1g1 Pipemon command failed, Driver busy: try again. |
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| 22:56.58 | horvath | If I have two outbound SIP providers and I want to send all outbound calls to company a ... but if company a had a problem (ie not registered or asterisk failed to dial it) I want to send calls to company b should I be using the Dial jump feature or is there a better way? |
| 23:02.02 | jblack | Two dial lines in a row. |
| 23:02.31 | jblack | NXXXXXX,1,Dial(A) .... NXXXXXX,2,Dial(B) |
| 23:09.22 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:12.20 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:12.53 | ambush276 | anyone have voxox as sip provider? |
| 23:14.20 | *** join/#asterisk joobie (n=joobie@203-217-64-151.dyn.iinet.net.au) |
| 23:16.59 | zamba | anyone used asterisk (or some other sip enabled server) to keep track of/connect hamradios? |
| 23:17.28 | drmessano | What do you mean "keep track of" |
| 23:18.00 | zamba | drmessano: something like http://www.irlp.net/ or http://www.echolink.org/ |
| 23:18.49 | drmessano | I dont understand the "keep track of" part |
| 23:19.05 | zamba | well.. you have to be able to "see" who's online |
| 23:19.22 | drmessano | Ok, you mean presence |
| 23:19.23 | zamba | without having a contact list of some sort |
| 23:19.26 | zamba | yeah |
| 23:19.35 | zamba | i was actually about to write that :) |
| 23:20.05 | zamba | i'm pretty sure asterisk can't do stuff like that without modification, but maybe you know of something that can with a more simple modification |
| 23:20.15 | zamba | because i guess SIP can be used for signaling here |
| 23:23.52 | joobie | guys i'm playing music on hold for callers ringing through the ISDN lines.. i know the SIP phones are all using g729 encoding, though not sure what encoding is used on the ISDN (or how to find out). I'm thinking if I convert the music on hold to the same encoding the ISDN lines are using, there will be less load on the * box |
| 23:25.07 | joobie | I read that RAW format is good because asterisk won't need to decode.. but if it's playing it on the ISDN line, and the ISDN line is alaw (for example), isn't it better to feed the music on hold music as alaw rather than raw? |
| 23:25.08 | zamba | joobie: i think isdn uses alaw |
| 23:25.21 | zamba | or ulaw |
| 23:25.38 | joobie | yerr it's defeinitely one or the other |
| 23:25.41 | joobie | just duno which |
| 23:25.54 | zamba | i think alaw is used in the us and ulaw in europe? |
| 23:26.05 | zamba | or the other way around |
| 23:26.07 | joobie | but given it's alaw/ulaw.. is it better to encode the music on hold files in alaw / ulaw to reduce load on asterisk? or use raw format |
| 23:26.14 | joobie | im in austrlia |
| 23:26.17 | joobie | australia |
| 23:26.18 | joobie | even |
| 23:26.29 | zamba | then it's alaw |
| 23:26.38 | zamba | E1 in australia? |
| 23:26.44 | joobie | ya |
| 23:26.56 | zamba | alaw then |
| 23:27.03 | joobie | thanks zamba |
| 23:27.08 | zamba | but that as much help i'm able to provide for this :) |
| 23:27.14 | joobie | hehe no worries |
| 23:27.21 | *** join/#asterisk xbmodder_ (n=Sargun@atarack/Staff/Sargun) |
| 23:27.27 | zamba | <- passes the ball to someone awake and able |
| 23:27.31 | joobie | see im reading http://www.orderlyq.com/asteriskqueues.html which says to use raw |
| 23:27.58 | joobie | but i duno.. my thought is asteirsk will have to encode it to whatever format the channel it's sending it through to on |
| 23:28.09 | zamba | that makes sense, yeah |
| 23:28.09 | joobie | so might be better to just save it in the native format rather than raw |
| 23:28.16 | joobie | sorta zamba |
| 23:28.29 | joobie | the other thought I have is that maybe asterisk is reading it as an input and encoding ti anywhere |
| 23:28.33 | joobie | -anywhere +anyway |
| 23:28.43 | joobie | like it "has to encode it" as a stream for the channel |
| 23:29.09 | joobie | in which case the performance would be in is it quicker for asterisk to read RAW or ALAW .. and RAW would win that |
| 23:32.10 | drmessano | zamba: Im not sure how you would do the live presence |
| 23:32.53 | drmessano | Im thinking AGI to get the users in the conference |
| 23:33.02 | drmessano | But you would need to build this |
| 23:33.45 | zamba | drmessano: well.. this has to scale for potentially hundred thousand users.. |
| 23:34.24 | zamba | drmessano: it's basically a replacement solution than the proprietary solution presented above |
| 23:34.29 | drmessano | 1/5th of the US ham populationg? lol |
| 23:34.34 | drmessano | -g |
| 23:34.56 | joobie | never heard of Ham Radios.. |
| 23:35.03 | zamba | i'm not a ham radio user myself, so i wouldn't know :) |
| 23:35.12 | zamba | " There are more than 200,000 validated users worldwide â in 162 of the world's 193 nations â with about 4,000 online at any given time." |
| 23:35.24 | zamba | as it was presented on echolink's page |
| 23:36.01 | drmessano | Those are very interesting numbers.. |
| 23:36.11 | joobie | it's just streaming a radio station via voip ya? |
| 23:36.15 | drmessano | No |
| 23:36.18 | joobie | so when it refers to users, it means how many voip connections? |
| 23:36.24 | joobie | ok.. way off the target then :P |
| 23:36.24 | drmessano | Its PEOPLE |
| 23:36.27 | drmessano | Yes |
| 23:36.51 | drmessano | Its like high-class CB radio |
| 23:37.59 | drmessano | Honestly, echolink is not a good yardstick here |
| 23:38.02 | joobie | so what is *'s role in this wholething |
| 23:38.16 | drmessano | There is a network using Asterisk already called All-Star link |
| 23:38.25 | drmessano | and its distributed nodes |
| 23:38.46 | drmessano | Thats a better model than something hosting all the conversations on a few server clusters |
| 23:39.14 | drmessano | So now you're looking at getting x number of users on a single node perhaps |
| 23:39.25 | drmessano | and maybe a page listing the number of users on each node |
| 23:42.29 | drmessano | For the user interface, I see maybe a softphone with a PTT button.. Add a tab, page, slideout, whatever that displays a page showing all the nodes and the usercount.. along with a link to call the node, which basically dumps everyone into a meetme |
| 23:43.02 | drmessano | You would need some script running on each node that reports it being online and the number of active users |
| 23:43.33 | drmessano | You would also need authentication at the nodes |
| 23:44.00 | drmessano | Since we're talking about potentially connecting the connected users to a licensed radio |
| 23:44.25 | drmessano | I wouldnt let someone connect to a node that I didnt know was a valid license holder |
| 23:45.20 | drmessano | Also need a way for the admin of each node to kick/ban users from their node |
| 23:48.05 | drmessano | So theres the base of your echolink killer |
| 23:48.48 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:48.58 | drmessano | Although killing or replacing echolink is pretty lame.. Something new and different that gives some of the same functionality but does it all in a much better way is what is really needed here |
| 23:49.30 | ambush276 | hey |
| 23:49.34 | ambush276 | i need some help |
| 23:49.38 | ambush276 | with the uplink program |
| 23:49.41 | ambush276 | for the skype to asterisk |
| 23:49.44 | ambush276 | im getting this error. |
| 23:49.52 | ambush276 | No such command '*CLI> [Apr 25 16:49:09] NOTICE[6576]: chan_sip.c:15236 handle_request_register: Registration from '<sip:12345@192.168.1.102>' failed for '192.168.1.103' - No matching peer found' (type 'help *CLI> [Apr' for other possible commands) |
| 23:50.04 | ambush276 | anybody? |
| 23:50.11 | drmessano | I would say |
| 23:50.14 | drmessano | No matching peer |
| 23:50.19 | ambush276 | ok.. |
| 23:50.23 | ambush276 | but in SIP i have |
| 23:50.37 | ambush276 | [skype] |
| 23:50.38 | ambush276 | host=dynamic |
| 23:50.40 | ambush276 | user=12345 |
| 23:50.42 | ambush276 | secret=XXXXX |
| 23:50.44 | ambush276 | type=peer |
| 23:50.45 | ambush276 | context=from-sip |
| 23:50.46 | drmessano | BAH |
| 23:50.50 | drmessano | ~pastebin |
| 23:50.50 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:51.08 | drmessano | Your config is wrong in the uplink app |
| 23:51.12 | drmessano | Check it again |
| 23:51.32 | drmessano | IP is in the wrong place |
| 23:52.25 | rob0 | Heart is in the right place, though. |
| 23:52.57 | drmessano | Youre saying "he means well"? |
| 23:53.09 | ambush276 | do |
| 23:53.19 | ambush276 | i want the IP to be the IP of the server, or the IP of the computer where Skype is running? |
| 23:54.44 | drmessano | Right now you have the IP in the username fireld |
| 23:54.45 | drmessano | field |
| 23:54.49 | drmessano | and that just WONT work |
| 23:54.54 | *** join/#asterisk ingenius (n=alektro@host57.190-138-60.telecom.net.ar) |
| 23:54.58 | ambush276 | its not thou |
| 23:55.01 | ambush276 | let me screen shot |
| 23:56.09 | rob0 | Yes, clearly, he means well. |
| 23:56.26 | rob0 | screen shot? Why not pastebin? |
| 23:57.09 | ambush276 | http://img27.imageshack.us/img27/8950/pizp.jpg |
| 23:57.16 | ambush276 | sceren shot of the uplink |
| 23:57.34 | ambush276 | b/c ok teh Skype is running on a windows computer w/ uplink |
| 23:57.45 | ambush276 | and the asterisk is on a linux server |
| 23:59.22 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
