00:00.11 | darkskiez | bowman: i hate the way they change the names in every bloody bit of software |
00:00.52 | bowman | darkskiez: I'm not talking about linphone, I'm talking about Gnophone ;) anyways, I used exactly the options given in the gnophone example at the wiki page |
00:01.13 | darkskiez | BoRiS: what does asterisk say in the logs |
00:01.22 | darkskiez | s/BoRis/bowman |
00:01.25 | *** join/#asterisk matobago (~matobago@65-77-23-11.ptp.ezeronetworks.net) |
00:01.31 | matobago | hi everyone |
00:01.40 | matobago | http://news.zdnet.co.uk/communications/networks/0,39020345,39169076,00.htm |
00:02.19 | *** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
00:02.27 | _Vile | he likes to show me this things |
00:02.30 | _Vile | skdfk |
00:03.09 | darkskiez | is there any good examples of a working enum configuration that doesnt bork up ? |
00:03.23 | *** join/#asterisk m-00kie (3704558@pcp09297172pcs.arlngt01.va.comcast.net) |
00:03.44 | bkw_ | ok why does drama always find me? |
00:03.45 | darkskiez | like checks the enum for what you dialled and falls back properly if it doesnt work |
00:03.46 | bkw_ | WHY OH WHY? |
00:03.46 | bowman | darkskiez: it says pure void , nothing :-) |
00:04.04 | darkskiez | bowman: set debug 15 |
00:04.07 | darkskiez | bowman: set verbose 15 |
00:04.19 | ZX81_AFK | maybe cos u r a drama queen? |
00:04.22 | ZX81_AFK | :-) |
00:04.45 | *** join/#asterisk bonbon-home (~happy@81-86-185-223.dsl.pipex.com) |
00:05.05 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
00:09.09 | bowman | scratch it. asterisk doesn't even react to traffic from gnophone, whereas gnophone sends exactly 2 packets to the server which contain pure garbage. |
00:09.37 | rene- | hey all, does anybody know what happened to zapteldoc.blogspot.com? it hasnt been updated for ages.... |
00:10.36 | ZX81_AFK | dunno |
00:10.38 | ZX81_AFK | sux a |
00:11.45 | darkskiez | WTF does inheritance mean with regards the SetVar applicaiton |
00:12.37 | *** join/#asterisk Corydon76-home (gray@pcp08665860pcs.500ash01.tn.comcast.net) |
00:12.44 | redder86 | if SetVar's parent dies then all of the parent's environment will belong to SetVar |
00:12.55 | redder86 | ;-) |
00:15.03 | blitzrage | is notransfer=yes known to work in Asterisk? I can't seem to stop the calls from being natively bridged... |
00:15.40 | file[laptop] | blitzrage: always works for me |
00:16.06 | blitzrage | file[laptop]: I can't seem to get it to work at all. Very confusing and annoying |
00:16.11 | blitzrage | unless I'm missing something stupid simple |
00:16.29 | file[laptop] | just put it in the two entries... hrm yes |
00:17.19 | blitzrage | I've placed notransfer=yes in the [general] section of iax.conf on all 3 computers. I'm calling from *1 <--> *2 <--> *3 and *2 keeps natively bridging the calls. |
00:17.26 | ChulJin | I thought to prevent [attempts at] native bridging it was 'canreinvite=no' |
00:17.31 | ionix | I get an error when i receive a FWD call on my asterisk gateway :/ |
00:17.32 | ionix | look |
00:17.32 | ionix | <PROTECTED> |
00:17.32 | ionix | <PROTECTED> |
00:17.32 | ionix | <PROTECTED> |
00:17.32 | ionix | Nov 15 19:16:09 NOTICE[-1253721168]: app_dial.c:714 dial_exec: Unable to create channel of type 'SIP' |
00:17.33 | ionix | <PROTECTED> |
00:17.35 | ionix | <PROTECTED> |
00:17.39 | _Vile | pastebin.ca |
00:17.43 | ionix | oks orry |
00:17.44 | ionix | sorry |
00:17.45 | blitzrage | ionix: NO FLOODING please |
00:17.57 | file[laptop] | ChulJin: different things, different things |
00:18.00 | JonR800 | SIP/l100???? |
00:18.01 | Optic2 | sip sip sip sip mooo |
00:18.03 | ionix | yeh |
00:18.04 | ChulJin | ah, I apologise. |
00:18.07 | _Vile | SIP/l100? |
00:18.07 | ionix | l100 is the extension |
00:18.09 | rowter | bbits possible to add ilbc support to cisco 7960, anyone has heard about it? |
00:18.21 | file[laptop] | notransfer=yes will prevent your IAX2 legs from directly connecting, put it in their entries... try not global |
00:18.22 | _Vile | sip debug |
00:18.23 | ionix | I have [l100] in sip.conf configured |
00:18.29 | file[laptop] | canreinvite=no does the same thing, prevents it from directly connecting |
00:18.31 | JonR800 | oh. |
00:18.37 | file[laptop] | they will still native bridge the audio if both sides are using the same codec |
00:18.44 | blitzrage | damnit! |
00:18.47 | blitzrage | not what I want |
00:18.53 | *** join/#asterisk Elshar (~Elshar@just.another.lame.unix-admin.com) |
00:18.58 | blitzrage | I need *2 to stay in the channel path. |
00:19.15 | JonR800 | ionix: do a sip show peers.. and check that it's registered. |
00:19.22 | file | native bridge the audio... as in, not transcode |
00:19.26 | _Vile | is your l100 phone registered, ion? |
00:19.55 | blitzrage | file[laptop]: what do you mean? |
00:20.02 | rowter | ionix, it seems its not registered, try to add l100 to name and shortname as in authname to test it. |
00:20.11 | ionix | status is Unmonitored |
00:20.22 | Elshar | Hey, has anyone had a problem with the zaptel drivers? In particular, I keep getting the error "zaptel: unknown symbol crc_ccitt_table". When trying to modprobe a cleanly compiled version from cvs |
00:20.29 | blitzrage | ionix: qualify=yes to monitor |
00:20.34 | ionix | ok |
00:20.35 | file | blitzrage: native bridging = asterisk just passing the audio from one leg to the other because of the same codec |
00:20.50 | blitzrage | file: right, but I don't want that :) |
00:21.00 | ionix | status is UNKNOWN |
00:21.10 | file | blitzrage: can you go into a conf? |
00:21.19 | blitzrage | file: sure, where? |
00:21.30 | ionix | according to the SPA-2000 Registration State: Registered |
00:21.32 | implicit | man even my regular cdrs are acting weird |
00:21.35 | blitzrage | file: just /msg it to me |
00:21.39 | _Vile | <PROTECTED> |
00:21.49 | implicit | does someone have a bit of time to take a look at them with me? |
00:22.05 | blitzrage | _Vile: I get those every once in a while too - not sure the exact cause |
00:22.14 | blitzrage | _Vile: doesn't seem to affect much... |
00:22.16 | _Vile | me either, but it's simply annoying... |
00:22.19 | _Vile | yeah, no complaints here |
00:22.25 | file | blitzrage: get it? |
00:22.56 | JonR800 | ionix: does it list an ip in sip show peers? |
00:23.12 | ionix | Unspecified |
00:23.13 | ionix | hmm |
00:23.15 | implicit | does anyone know how to get cdr's to be put out for all the legs of a call? |
00:23.21 | JonR800 | it doesn't sound like it's registered. |
00:23.23 | Blackthorn | on a two pri setup can you have both pri's set to primary sync? and would that be a reason that shows the secondary in red alarm |
00:23.24 | implicit | I am dialing a DID that goes into my asterisk box |
00:23.26 | mishehu | what does the zaptel watchdog actually do if a zap device stops taking interrupts? |
00:23.28 | ionix | why is that so ? |
00:23.31 | JonR800 | err not registered |
00:23.34 | JonR800 | sorry typo |
00:23.45 | ionix | sipura says: Registration State: Registered |
00:23.52 | ionix | should I restart the spa ? |
00:24.17 | ionix | ahh |
00:24.19 | ionix | lol my error |
00:24.29 | implicit | that rings my 7960 which forwards back to the local context which uses a voip provider to my cell phone, but the only records that go in are the DID record and the Local dial, there is no dialout to the voip provider |
00:24.42 | implicit | anyone know how to change that ? |
00:25.16 | ionix | Registered SIP 'line1' at 65.94.160.215 port 62263 expires 3600 |
00:25.46 | ionix | however I still cannot receive a call |
00:25.46 | Blackthorn | well got my primary up thats good enophe for the night thanks for everyoens help |
00:25.49 | ionix | <PROTECTED> |
00:25.49 | ionix | <PROTECTED> |
00:26.05 | ionix | so at this point problem is the SPA ? |
00:26.42 | ionix | ok works |
00:26.52 | ionix | I love you guys |
00:27.12 | JonR800 | I'm glad someone does. |
00:27.18 | Elshar | hehe :) |
00:27.26 | implicit | ?????? |
00:27.28 | implicit | lol |
00:27.30 | *** join/#asterisk sleepy_one__ (~chatzilla@dhcp16632045.neo.rr.com) |
00:29.31 | *** join/#asterisk sleepy_one____ (~chatzilla@dhcp16632045.neo.rr.com) |
00:29.35 | blitzrage | file: thanks again for the tip - you're the man! |
00:30.40 | *** join/#asterisk nem (~nemisis@66.220.23.4) |
00:30.42 | Elshar | Anyone have any ideas on that unknown symbol thing? ;) |
00:30.55 | bonbon-home | if you have a quality=yes setting in your iax entity, then wouldn't you expect * to move from one line of extensions.conf to the next in the event that iax2 connection is down? |
00:31.36 | bonbon-home | because it doesn't seem to work like that |
00:31.38 | bonbon-home | :-( |
00:32.08 | *** join/#asterisk nem (~nemisis@66.220.23.4) |
00:32.45 | bonbon-home | it jumps to "busy" context |
00:32.59 | *** join/#asterisk krw (~ken@209.242.52.25) |
00:33.46 | *** join/#asterisk trelane (trelane@lan.trelane.net) |
00:33.53 | *** join/#asterisk sleepy_one_____ (~chatzilla@dhcp16632045.neo.rr.com) |
00:34.23 | *** part/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
00:34.46 | implicit | bonbon-home: sucks :( |
00:35.17 | trelane | anyone have a good FSX interface recommendation besides digium's? Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models? |
00:35.50 | menger | is ftp.digium.com broken? seems to be very slow for me |
00:36.07 | file[laptop] | it's always slow |
00:36.15 | file[laptop] | it's the ftp.digium.com way |
00:36.26 | bonbon-home | implicit: have you got it to work? |
00:36.36 | sleepy_one_____ | try to wget asterisk v1.0.0 and zaptel and libpri instead using http |
00:36.57 | menger | sleepy_one____ url? |
00:37.45 | sleepy_one_____ | wget http://www.asterisk.org/asterisk-1.0.0.tar.gz; wget http://www.asterisk.org/libpri-1.0.0.tar.gz; wget http://www.asterisk.org/zaptel-1.0.0.tar.gz |
00:37.46 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
00:37.49 | Moc | hi all |
00:39.27 | sleepy_one_____ | hey Moc |
00:39.39 | ChulJin | hey Moc |
00:39.49 | ChulJin | ça va? |
00:40.42 | Moc | pas pire toi ? |
00:40.43 | sleepy_one_____ | tres bien merci |
00:40.56 | sleepy_one_____ | pardon my french ;) |
00:41.09 | Moc | ;) |
00:41.17 | sleepy_one_____ | pun intended |
00:41.30 | Moc | you said it right ;) |
00:42.07 | sleepy_one_____ | I know ;) |
00:42.24 | sleepy_one_____ | he was addressing you tho, I was kidding around |
00:43.33 | *** join/#asterisk enzo123 (~jason@cpe-024-211-181-116.nc.rr.com) |
00:46.18 | enzo123 | sup all |
00:46.28 | enzo123 | anyone using a spa3000 |
00:47.53 | *** join/#asterisk sleepy_one______ (~chatzilla@dhcp16632045.neo.rr.com) |
00:48.07 | ChulJin | enzo: yes, but not the pstn side (yet)...so not prepared to answer questions about that part of its functionality |
00:48.27 | sleepy_one______ | anyone know what the ETA is on the S100I v2.0 ? |
00:49.47 | trelane | anyone have a good FSX interface recommendation besides digium's? Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models? |
00:50.00 | trelane | s/FXS FXS |
00:50.29 | *** join/#asterisk myconid (myconid@155.42.19.149) |
00:50.31 | myconid | Hello. |
00:50.32 | enzo123 | ive got a clone |
00:50.42 | myconid | I am trying to connect to my remote * system.. |
00:50.48 | myconid | via SIP (xlite) |
00:50.50 | enzo123 | dont remember the chipset tho |
00:50.58 | myconid | but it just times out.. and im not sure how to see if sip port is open |
00:51.20 | *** join/#asterisk Legend (~legend@24.244.142.134) |
00:51.51 | trelane | enzo123, know what manufacturer made the modem? |
00:52.55 | *** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com) |
00:53.12 | sleepy_one | hello again everyone |
00:55.39 | implicit | trelane fxs with which modems?? |
00:56.27 | myconid | how do I bind multiple ports in sip.conf? |
00:58.35 | enzo123 | dont remember |
00:58.44 | enzo123 | ebay for x100p |
00:58.52 | enzo123 | there are tons of clones |
00:58.57 | implicit | fxo? |
00:59.21 | implicit | x100[p is fxo |
00:59.27 | WilliamK | probably 50-100 on ebay right now |
00:59.29 | enzo123 | FXO |
00:59.32 | WilliamK | just looked about 20 mins ago |
00:59.35 | enzo123 | PSTN |
00:59.39 | implicit | never heard of modems doing fxs |
00:59.46 | enzo123 | they dont |
00:59.52 | implicit | ya |
01:00.05 | implicit | trelane: did u ,mean fxs or jiust typo? |
01:00.19 | enzo123 | got a old quicknet card |
01:00.27 | enzo123 | souunds like crap but it works |
01:00.43 | enzo123 | for fxs |
01:00.54 | folsson | myconid : You don't |
01:01.18 | implicit | i used the tdm400p and iaxy |
01:01.24 | implicit | not too pleased with either's sound quality |
01:01.29 | implicit | so i just bought a 7960 |
01:02.00 | implicit | the sound quality is not too bad either |
01:02.04 | implicit | just not as good as i'd like |
01:02.16 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
01:02.22 | implicit | hi |
01:02.28 | puzzled | hello |
01:02.33 | implicit | whats going on? |
01:02.36 | enzo123 | my 7960 doesn't sound as good as my sipura |
01:02.53 | implicit | oh yeah? |
01:02.59 | Moc | enzo123, I wouldn't say that ;) |
01:02.59 | implicit | which sipura? |
01:03.12 | Moc | but my polycom sound better than my cisco |
01:03.17 | enzo123 | haha |
01:03.20 | enzo123 | prob so |
01:03.29 | puzzled | Moc: polycom kept the good stuff for themselves :) |
01:03.38 | krw | I'm trying to use ser to forward to a voice app on asterisk. I'm rewriting the URI and forwarding the call correctly to asterisk, but I need to get the origonal dialed number in the "To:" sip header. Found RDNIS but that appears to be adopting an old cisco "Diversion" tag, anyone have any thoughts on this? |
01:03.41 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-204-71.client.comcast.net) |
01:03.44 | implicit | you think i should sell cisco and get the polycom? |
01:04.08 | puzzled | they are cheaper, that's for sure. and cisco's are basically licensed/OEM'ed polycom technology |
01:04.09 | Moc | implicit, Well you can get 2 polycom for the price of 1 cisco ;) |
01:04.18 | enzo123 | im using my cisco on 729 for pass , but even the vm sounds kinda crapy |
01:04.20 | implicit | what is the best polycom? |
01:04.24 | Moc | implicit, you ever saw polycom ? |
01:04.27 | implicit | seen |
01:04.29 | implicit | but not used |
01:04.29 | puzzled | implicit: IP600 |
01:04.33 | implicit | not seen in person |
01:04.33 | Moc | implicit, IP 500 are good, but IP 600 is the best |
01:04.39 | Moc | I just ordered the ip 600 yesterday |
01:04.44 | enzo123 | my 7960 was only 200 |
01:05.15 | Moc | even if 7960 and polycom was the same price, I would still go to polycom |
01:05.21 | implicit | mine was 250 with headset and power cube |
01:05.42 | Moc | implicit, you ever saw polycom config file ? |
01:05.49 | puzzled | hehe |
01:05.53 | implicit | nope |
01:05.56 | puzzled | xml stuff right? |
01:05.58 | Moc | implicit: check http://www.freedomphones.net/polycom/files/Admin_Guide-SoundPoint_IP_SIP_2004-06-16.pdf |
01:05.59 | Moc | yes |
01:06.17 | Moc | this PDF will show you just how good polycom phone are |
01:06.22 | *** join/#asterisk johaan (~johaan@ip24-56-24-181.ph.ph.cox.net) |
01:06.27 | enzo123 | hey got a question for everyone that has nufone |
01:06.28 | ChulJin | ~seen jerjer |
01:06.31 | jbot | jerjer <~mine@d2-236.rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 3d 17h 56m 37s ago, saying: 'no config necessary'. |
01:06.39 | Moc | enzo123, go |
01:06.45 | ChulJin | ~seen jerjer[mobile] |
01:06.46 | jbot | jerjer[mobile] <~jj@mail.nufone.net> was last seen on IRC in channel #asterisk, 8h 40m 55s ago, saying: 'paging'. |
01:06.54 | enzo123 | whats up with nufone ? |
01:07.19 | ChulJin | haha that's rather broad :P |
01:07.20 | Moc | what wrong with them ? |
01:07.21 | redder86 | enzo123: whaddya mean? |
01:07.23 | ChulJin | (though I know what you mean) |
01:07.50 | enzo123 | all support emails to them go unanswered |
01:07.59 | johaan | Hello everyone, Running Fedora Core2, I did make config to have asterisk start up on boot. And the script started like 14 Asterisk proccess |
01:08.01 | johaan | root 2115 0.0 0.4 115396 5020 tty1 S 03:31 0:00 asterisk -vvvg -c |
01:08.29 | enzo123 | service asterisk start |
01:08.32 | johaan | previously I was just have one running.. is this an error or is it suppose to look like that. |
01:08.44 | Moc | implicit, what do you think now ? ;) |
01:08.56 | Moc | can you do all that with your cisco ? ;) |
01:09.05 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
01:09.21 | enzo123 | anyone else able to get support fomr them ? |
01:09.31 | syslod | hi |
01:09.36 | redder86 | enzo123: yup |
01:09.59 | enzo123 | humm then must just ignore my emails |
01:10.10 | redder86 | enzo123: how long has it been? |
01:10.27 | trelane | implicit, I meant fxs, that was indeed a typo |
01:10.33 | enzo123 | 3 weeks this thr |
01:10.36 | ChulJin | enzo: once I overcame my indignance, I found and talked to JerJer. my DID's not up, but he did have some encouraging words about the progress. |
01:10.43 | trelane | anyone have a good FXS interface recommendation besides digium's? Ambient/Intel chipset modems are noted on voip-info.org but I've not seen anyone being particularly open about what chipsets are on their modems, does anyone have any experience or feedback with specific vendor models? (amended for implicit) |
01:10.57 | ChulJin | enzo: one thing to notice (that I overlooked) was 7-10 business days. |
01:11.01 | Moc | damn you can even use Windows .fon file on the polycom phone .. |
01:11.02 | puzzled | enzo123: me too. wonder if they have "a prepay as you go" thing instead of those plans |
01:11.29 | redder86 | treland: the Ambient/Intels are for FXO. |
01:11.35 | enzo123 | umm oct28th is over 10 buz days |
01:11.51 | trelane | redder86, I was looking for specific model recommendations (I've had signal to noise issues on mine) |
01:11.59 | enzo123 | i dont really care about the time.. just reply saying " hey we are working on it " |
01:12.04 | redder86 | treland: FXO or FXS? |
01:12.13 | redder86 | enzo123: what's the problem? |
01:12.15 | trelane | FXS |
01:12.28 | trelane | to terminate a phoneline |
01:12.35 | redder86 | trelane: FXO, then |
01:12.36 | johaan | I should only see one asterisk process running right? |
01:12.41 | trelane | do I have it backassword? |
01:12.56 | enzo123 | ordered and paid for a 800 DID waiting for it to be turned up 3 weeks thr |
01:12.58 | redder86 | trelane: you plug phone lines into FXO ports and phones into FXS ports |
01:13.16 | redder86 | enzo123: he had mine up in a matter of hours. I didn't order vanity, though. |
01:13.23 | BoRiS | from enzo? |
01:13.30 | enzo123 | vanity |
01:13.58 | redder86 | enzo123: did you pay yet? |
01:14.10 | trelane | redder86, then I need FX0 |
01:14.14 | trelane | sigh |
01:14.18 | trelane | I need a new keyboard |
01:14.21 | redder86 | trelane: how many ports? |
01:14.22 | trelane | I honestly can type |
01:14.22 | enzo123 | redder86 yes |
01:14.23 | trelane | one |
01:14.26 | trelane | single line |
01:14.27 | trelane | voicemail system |
01:15.12 | ChulJin | enzo: the idea is to forego indignance, find JerJer on here, and just PM him. |
01:15.15 | ChulJin | He's actually quite nice. |
01:15.37 | enzo123 | my point is that i should have to chase down someone on IRC to get support |
01:15.43 | enzo123 | shouldn't |
01:15.46 | redder86 | trelane: if you don't want to pay for the dev kit from Digium (X100P) then find an Intel/Ambient that uses the MD3200 chipset. AMI-IA92/IE92 but make darn sure that you check with the vendor on the chipset first. The i537EP chipset is *not* what you want. |
01:15.55 | *** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
01:16.18 | redder86 | enzo123: in my experience he's been good about responding to e-mail |
01:16.35 | trelane | redder86, the intel 837 is also supposed to work, I've got one, it does, signal quality is poor, I think because of inferior modem quality, I got it used for $1 |
01:16.38 | redder86 | enzo123: as long as it got into his "system" |
01:16.54 | *** join/#asterisk AlexZander (~AlexZande@spc2-bolt4-4-0-cust107.bagu.broadband.ntl.com) |
01:16.57 | trelane | redder86, my question wasn't so much one of compatibility but of personal experience with a given part |
01:16.57 | redder86 | trelane: I'll bet you money, though that it's not the i537EP chipset. |
01:17.24 | myconid | anyone use a IAX softphone/ |
01:17.35 | enzo123 | you can make the call via e164.org lookup , but not via pstn |
01:17.46 | redder86 | trelane: I've used two different AMI-IA92/IE92 modems that I purchased from two different vendors, both with no problem. |
01:17.57 | trelane | noted |
01:18.01 | trelane | redder86, that's what I needed to haer |
01:18.01 | trelane | hear |
01:18.20 | redder86 | trelane: one I paid $15 for and the other $65. |
01:18.42 | trelane | any audible difference in quality, or issues with call pickup etc? |
01:19.08 | *** join/#asterisk riksta (~rick@81-178-241-195.dsl.pipex.com) |
01:19.11 | redder86 | trelane: nope. In fact, the cards are identical, except that the vendor of the $65 one rubbed out the chipset ID silkscreen with sandpaper. |
01:19.30 | trelane | jerks |
01:19.31 | redder86 | trelane: I purchased that one directly from Tiger Jet. |
01:19.51 | redder86 | trelane: the other I purchased from eebuy.com |
01:20.17 | redder86 | trelane: I hear that eebuy.com is now listing i537EP cards under the AMI-IA92/IE92 label, though. |
01:20.34 | trelane | noted |
01:20.35 | redder86 | trelane: and those don't work, believe me. If you want some I can sell you a few. |
01:21.07 | enzo123 | mine works like a champ |
01:21.12 | enzo123 | 15 bucks on ebay |
01:21.26 | redder86 | enzo123: it's not an i537EP then |
01:21.29 | trelane | redder86, any intel with speaker+mic is a nogo best I can tell |
01:21.56 | redder86 | trelane: the i537EPs that I have do not have speaker or mic |
01:22.39 | redder86 | trelane: I must correct myself. There was a slight difference between the two cards that I have. |
01:23.03 | redder86 | trelane: the PCI ID codes vary due to some differences in the resistors on the PCI ID portion of the card. |
01:23.21 | redder86 | trelane: the PCI IDs only affect "auto detection" |
01:23.43 | trelane | that's not an issue, I can make just about anything work, I just can't deal iwth a loud whirr throughout the call |
01:23.59 | enzo123 | here |
01:24.01 | enzo123 | http://i12.ebayimg.com/02/i/02/a9/31/db_1.JPG |
01:24.10 | enzo123 | thats the one i have |
01:24.34 | redder86 | enzo123: an "Intel 537", but note that it has the MD3200 chipset. |
01:24.43 | puzzled | enzo123: got mine new in the shop for something like €9.95 |
01:24.44 | redder86 | enzo123: not the i537EP |
01:25.11 | *** join/#asterisk florz (nobody@I9593.i.pppool.de) |
01:25.14 | Syncros | <PROTECTED> |
01:26.01 | redder86 | enzo123: notice also the "AMI-IA92/IE92" silkscreening. I don't think that you will see that on the i597EP cards, although the *vendor* may incorrectly sell you an i537EP under that name. |
01:26.49 | redder86 | Syncros: I went through the effort of looking up "Tiger Jet Network Inc." and it led me to cuphone.com, which is where I bought the $65 version that was adulterated. |
01:26.56 | enzo123 | 00:0e.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 |
01:27.17 | enzo123 | yeah |
01:27.25 | enzo123 | i got my from a buddy that got 30 of them |
01:27.34 | enzo123 | all 6 bucks each |
01:29.15 | redder86 | enzo123: there were numerous vendors of these things. The PCI ID was completely configurable on the vendor-level, and so there's a bunch of different entries in the Linux PCI ID device list... just depending on what particular PCI ID your card uses. |
01:29.35 | enzo123 | guess i got lucky |
01:29.55 | enzo123 | picked up a spa3000 today ill she how that does me |
01:30.00 | *** join/#asterisk vexorg (~vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
01:30.17 | redder86 | enzo123: add/remove resistors to the joints on the top-left (in the picture you gave) to twiddle with the PCI ID value. |
01:30.55 | myconid | lol |
01:40.00 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
01:50.12 | *** join/#asterisk t3t (~t3t@207.67.0.20) |
01:52.38 | redder86 | ssssh! |
01:53.41 | *** join/#asterisk Darwin_35 (~darwin35@c-24-3-204-71.client.comcast.net) |
01:54.03 | ariel_ | wow seems like there is just plain too much going on here tonight. |
01:54.14 | syslod | I'm telling ya. |
01:54.19 | syslod | Lots of activity |
01:54.42 | syslod | Well guess I'll liven it up a bit with a ongoing problem. |
01:54.59 | ariel_ | problem? |
01:55.03 | syslod | Anyone know why MWI would not clear on a poly 300? |
01:55.19 | Optic2 | anybody know when budgetone will support MWI? :) |
01:55.20 | Optic2 | hehe |
01:55.24 | syslod | Reboots don't help nothing. Somehow it thinks I have a message. |
01:55.34 | redder86 | Optic2: mine already does. |
01:55.38 | syslod | grandstream MWI works. |
01:55.40 | Optic2 | redder86: whoa! |
01:55.49 | ariel_ | syslod, sometimes there is some mistry message left in the box. |
01:55.52 | Optic2 | is there a faq somewhere? |
01:55.57 | syslod | poly works too well. It won't turn off. |
01:56.00 | Nugget | http://voip-info.org/ |
01:56.13 | redder86 | Optic2: I have had a Grandstrem BT100 since June and it has always supported MWI |
01:56.27 | syslod | I checked all the boxes. Any debug commands or way to have MWI info in the debug? |
01:56.53 | syslod | Grandstreams seem to work well except for forgetting they are registered. |
01:56.58 | ariel_ | are you sure it's not pointing to the wrong box area? other then default? |
01:57.01 | syslod | Polys work even better. |
01:57.16 | redder86 | Polys are a pain to configure, but work well, yes. |
01:57.18 | syslod | I've checked it a million times. |
01:57.50 | syslod | I even setup a budgettone then configured the poly for the same sip account. Budget tone works poly doens't. |
01:58.08 | redder86 | sounds like a Poly issue |
01:58.28 | syslod | Also is there a way to have the message button or message menu go directly to voicemail password prompt? |
01:58.48 | syslod | Anyone have a working config for the poly they mind sending me? |
01:59.12 | riksta | what clever things can i do with the menu screen of a cisco 7940 ?? :) |
01:59.21 | redder86 | syslod: you can configure all of the buttons, including the message one, to dial whatever you need it to |
02:00.01 | syslod | But what do I program it to? IE voicemail is ext 999 do I just do 999 then mailbox? |
02:00.09 | syslod | Seems to screw up 80% of the time. |
02:00.14 | BoRiS | Optic2: Are you using a Grandstream with realtime config (res_config?) |
02:00.17 | syslod | Is there a pause |
02:00.20 | Optic2 | nope |
02:00.29 | BoRiS | ok |
02:00.52 | *** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net) |
02:02.14 | redder86 | syslod: is your poly configured via ftp? |
02:02.24 | syslod | Yea. |
02:02.55 | syslod | I've played with the menus too. |
02:03.03 | syslod | ftp seems to have more options. |
02:03.46 | redder86 | I can't very well send you my cfg without DCC, now can I? |
02:04.25 | syslod | nope. I'll try and figure out what broke my dcc or return with working client. BRB. |
02:05.00 | redder86 | http://pastebin.ca/2115 |
02:05.37 | *** join/#asterisk syslod (0@65.114.15.106) |
02:05.39 | AlexZander | Sorry to but in but are you talking about Polycom handsets - Last time I looked they weren;t too useable with Asterisk - have things changed now? |
02:05.59 | syslod | K now in working client. |
02:06.40 | syslod | Polys seem to work very well. 300's and 500's here |
02:07.57 | *** join/#asterisk nullogic (~nullogic@c-24-98-72-110.atl.client2.attbi.com) |
02:08.59 | syslod | been awhile since I've been on IRC at the console. |
02:09.50 | AlexZander | syslod: Do you use sip or h323 with the polys? |
02:10.02 | syslod | sip |
02:11.50 | AlexZander | nice - I'll have to take a look at them again - I really liked the look of the Poly handsets but was told that I a) couldn;t get reliable sip to work with asterisk and b) you couldn;t configure them fully so they worked with asterisk without compromise |
02:12.26 | syslod | I think most ppl are using poly |
02:13.14 | AlexZander | It's been quite a long time since I last frequented this channel and looked at asterisk |
02:13.27 | *** join/#asterisk smoothjim (~jim@24-159-238-38.jvl.wi.charter.com) |
02:13.51 | AlexZander | Back then, people were going with snom over poly not because they felt they were better but just because they worked - I'm obviously way behind. |
02:13.52 | syslod | It looks like its a replacment for most keys. |
02:14.14 | syslod | We are looking at virtual hosted PBX's |
02:14.41 | redder86 | If Snom works then Polycoms rock. Snoms suck IMO. |
02:15.05 | syslod | redder86: Hopefully things are working now if you don't minding sending again. |
02:15.21 | ariel_ | snom work fine. Polycom are better but are alittle harder to configure. But both are good phones. |
02:16.12 | redder86 | I've had no end of firmware glitches with my Snom 190 |
02:16.19 | redder86 | I have to reboot it frequently. |
02:16.56 | redder86 | Polycom are quite difficult to configure in comparison with, say, a Grandstream. |
02:18.04 | syslod | exit |
02:18.23 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
02:18.35 | *** join/#asterisk hades_ (~hades_@200-203-038-195.paemt7006.dsl.brasiltelecom.net.br) |
02:19.12 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
02:21.41 | *** join/#asterisk mlh407 (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
02:23.10 | *** join/#asterisk QRPartner (~tenikiwon@ns1.accu-com.com) |
02:23.22 | AlexZander | redder86: I guess though that once you;ve configured one successfully, it's quite easy to do it again - Plus you get so much more out of the phone than you do with a Grandstream - A phone that interacts with Asterisk like a traditional PBX is worth the premium. |
02:23.31 | *** join/#asterisk dima202 (~chatzilla@ool-18b9039c.dyn.optonline.net) |
02:23.37 | dima202 | »ey guys |
02:23.56 | dima202 | I am running ppc, would it be possible to run atericks on this processor |
02:23.58 | AlexZander | I'm looking at all the poly stuff on voip-info.org - very happy - none of this was there last time I looked. |
02:24.39 | dima202 | It would be great if someone could point me to the direction |
02:25.29 | funkknob | PPC Mac or PPC IBM? |
02:25.40 | dima202 | mac, sorry |
02:26.00 | funkknob | You want to run under MacOS or Linux? |
02:26.10 | dima202 | I am running Mac right now |
02:26.26 | funkknob | OS X I think you should be able to, or load linux. |
02:26.42 | dima202 | I tried make |
02:26.47 | JamesDotCom | yeah, there's a binary package of asterisk for os x floating around somewhere |
02:26.48 | funkknob | 'Cuz OS X is basically unix now |
02:26.52 | dima202 | but I get an error, I will paste |
02:27.33 | dima202 | for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x depend || exit 1 ; done |
02:27.42 | dima202 | make[1]: Nothing to be done for `depend'. |
02:27.54 | JamesDotCom | that's not an error |
02:27.58 | dima202 | cd editline && unset CFLAGS LIBS && test -f config.h || ./configure |
02:28.01 | JamesDotCom | oh |
02:28.09 | dima202 | make -C editline libedit.a |
02:28.14 | dima202 | and td. |
02:28.24 | dima202 | I mean so on |
02:28.27 | JamesDotCom | no errors in there so far |
02:28.36 | dima202 | gmm |
02:28.38 | JamesDotCom | make install it |
02:28.46 | dima202 | ast_expr.y:110: unrecognized: %locations |
02:28.48 | JamesDotCom | then try run "asterisk -vvvc" |
02:29.14 | dima202 | if [ -d CVS ] && ! [ -f .version ]; then echo 1.0.0 > .version; fi |
02:29.26 | JamesDotCom | well anyway, as i said, there's a compiled version of asterisk for os x floating about a few places |
02:29.45 | dima202 | unknown command |
02:29.54 | JamesDotCom | did you make install? |
02:29.58 | dima202 | yes |
02:30.04 | JamesDotCom | did that error? |
02:30.28 | dima202 | Yes, it occured |
02:30.34 | dima202 | make: *** [ast_expr.c] Error 1 |
02:30.38 | dima202 | here is 1 error |
02:31.00 | *** part/#asterisk smoothjim (~jim@24-159-238-38.jvl.wi.charter.com) |
02:31.01 | JamesDotCom | go to pastebin.ca or something and paste all the output from make |
02:31.06 | *** join/#asterisk Marlow (~marlow@217.67.139.197) |
02:31.48 | dima202 | Argh, it's the damn client, won't let me paste |
02:32.27 | JamesDotCom | which client? |
02:32.51 | dima202 | from mozilla |
02:32.59 | file[laptop] | kram: how are you muffin man? |
02:33.05 | kram | chillin |
02:33.06 | kram | u? |
02:33.23 | dima202 | WOuld you be able to accept txt version? |
02:33.42 | file[laptop] | these reality TV shows are scary |
02:33.47 | JamesDotCom | dima, just get something substantial to look at somewhere :D |
02:34.16 | JamesDotCom | scariest one i've seen here is a reality tv show based on ballroom dancing |
02:36.41 | file[laptop] | creepy |
02:37.15 | paulc | I saw Wife Swap for the first time the other night.. HOLY, that program is CRAZY! |
02:37.16 | JamesDotCom | very |
02:37.23 | file[laptop] | that was on earlier |
02:37.24 | JamesDotCom | it's got sms voting and all |
02:37.37 | *** join/#asterisk Peanut486 (~dustin@net24-164-112-183.neo.rr.com) |
02:38.13 | JamesDotCom | haha, they even showed that show over here in australia |
02:38.30 | JamesDotCom | there was another one just the same, trading spouses? i think |
02:38.35 | file[laptop] | paulc: conf? |
02:38.43 | paulc | Sure, pick a number? |
02:38.47 | JamesDotCom | dima202: any luck? |
02:38.48 | file[laptop] | 666 |
02:38.57 | paulc | cos you're evil personified ;-) |
02:39.03 | file[laptop] | yup |
02:39.05 | dima202 | yeah, I think I found something |
02:39.21 | dima202 | http://www.apple.com/downloads/macosx/unix_open_source/asteriskinstallpackageformacosx.html |
02:39.26 | JamesDotCom | yeah |
02:39.30 | JamesDotCom | very old version though |
02:39.35 | JamesDotCom | i'm sure you'll find newer ones about |
02:39.43 | JamesDotCom | Version: CVS 10-28-03 |
02:41.50 | dima202 | how do u run it? |
02:41.53 | dima202 | lol |
02:42.16 | dima202 | no man pages |
02:42.39 | JamesDotCom | asterisk |
02:42.44 | *** join/#asterisk wolfson (~hehe@208.25.254.124) |
02:42.47 | JamesDotCom | asterisk -vvvc for some verbosity |
02:42.58 | dima202 | hmm |
02:43.34 | dima202 | it tells me it's running |
02:44.04 | JamesDotCom | asterisk -r |
02:44.59 | dima202 | unable to connect to remote asterisk |
02:45.35 | dima202 | hmm |
02:48.06 | QRPartner | I downloaded the *1.0.0.tar.gz file and I seem to unzip them |
02:49.06 | slePP | anyone have a DVG-1120S? |
02:49.25 | *** join/#asterisk Verliba (~Miranda@65-100-56-22.ptld.qwest.net) |
02:50.00 | file[laptop] | slePP: loan me one! |
02:51.11 | slePP | heh |
02:51.14 | slePP | these things suck ass |
02:51.52 | dima202 | what command is to list current processes |
02:52.00 | funkknob | ps -ef |
02:52.03 | slePP | file: keep your PAP2-NA's. don't ever let them go |
02:52.06 | dima202 | thanks |
02:52.13 | slePP | i'd like to toss it out the window, but, alas.... it's the best i can do atm |
02:52.20 | slePP | mind you, it costs more than the SPA-2000 |
02:52.25 | slePP | and the SPA-2000 is much cooler |
02:52.32 | dima202 | there is another command |
02:52.34 | slePP | we're still waiting on our RMA of one of the 5 PAP2's we had |
02:52.48 | paulc | I love the SPA-3000 |
02:52.51 | slePP | and we have 21 more on order.. but, yeh. right.... 'they'll be in in 3 days!' 'bullshit you whoremongeringbitchfacedliar!' |
02:53.05 | paulc | and I'm itching to get my hands on a new Sipura 841 phone |
02:53.15 | slePP | er. |
02:53.15 | file[laptop] | paulc: 666 you ding bat |
02:53.16 | slePP | mouse |
02:53.51 | file[laptop] | mmm.... Swiss Chalet.... |
02:54.06 | funkknob | I haven't used IRC for more than 10 yrs - what's a good windoze client? |
02:54.17 | BoRiS | drive thru? |
02:54.25 | BoRiS | mirc |
02:54.34 | puzzled | gaim |
02:54.36 | mlh407 | chatzilla |
02:54.38 | funkknob | Sorry, I mean free :D |
02:54.40 | file[laptop] | I'm in 666 for all those who know what it is |
02:54.44 | JamesDotCom | irssi4life |
02:54.53 | paulc | I use X-Chat and like it a lot.. www.xchat.org I think |
02:55.06 | JonR800 | I'm with irssi. |
02:55.28 | Peanut486 | I am trying to have the FXO channel on voice1 server ring the FXS on voice2 server. I keep getting "Rejected connect attempt from <HOST>" |
02:55.37 | *** join/#asterisk wolfson` (~hehe@208.25.254.124) |
02:55.52 | funkknob | k I'll check those out thx.... right now using HydraIRC but I have to copy/paste URLs :( |
02:56.32 | dima202 | guys, there is something besides ps |
02:57.01 | funkknob | What do you need to see? |
02:57.08 | dima202 | all processes |
02:57.29 | funkknob | ps -ef did not work? |
02:57.29 | dima202 | on unix I remember there was a command, just can't remember it |
02:57.32 | JamesDotCom | ps aux |
02:57.35 | dima202 | it is same for mac |
02:57.36 | JamesDotCom | on os x |
02:57.52 | dima202 | yes, yyou are correct, there is also a live one |
02:57.59 | funkknob | top |
02:58.03 | JamesDotCom | oh |
02:58.05 | JamesDotCom | yeah top |
02:58.07 | dima202 | hell yeah |
02:58.10 | funkknob | heehee :D |
02:58.10 | dima202 | Thanks guys |
02:58.18 | JamesDotCom | i love os x |
02:58.47 | funkknob | Me too, my powerbook is back in the states :( |
02:58.49 | dima202 | mine is sexier |
02:58.59 | JamesDotCom | nothing is sexier than a 12" powerbook |
02:59.15 | dima202 | yeah, a 17" |
02:59.18 | dima202 | hehe |
02:59.22 | JamesDotCom | nah |
02:59.25 | JonR800 | the 12" screen needs updating bad.. |
02:59.29 | JamesDotCom | i'd take the smaller one any day |
02:59.32 | JamesDotCom | yeah |
02:59.37 | JamesDotCom | i wish it could do more than 1024x768 |
02:59.40 | JamesDotCom | but i'm happy |
02:59.42 | JonR800 | I'm hoping January revisions are WS |
02:59.53 | JonR800 | WS G5 12".. mm ;) |
02:59.56 | JamesDotCom | os x looks a lot better than other os's with a smaller resolution |
03:00.13 | *** join/#asterisk skeeziks (~skeeziks@r80h128.res.gatech.edu) |
03:00.19 | JamesDotCom | heh, i dont need anything more powerful than this, all i run is terminals and a browser |
03:00.23 | JamesDotCom | should last me a good 5 years at least |
03:00.40 | dima202 | I like to game some time |
03:00.51 | dima202 | 17 is great for gaming sometimes |
03:00.54 | JamesDotCom | that's what the xbox is for :) |
03:01.02 | dima202 | Don't have it |
03:01.08 | dima202 | I had a choice one day, a car or p b |
03:01.26 | JamesDotCom | pb! :D |
03:01.29 | dima202 | no xbox |
03:01.34 | dima202 | yeah... |
03:01.38 | JonR800 | the 17" is beautiful. |
03:01.44 | dima202 | Thats how crazy I am |
03:01.47 | JamesDotCom | too big |
03:01.49 | JamesDotCom | haha |
03:01.51 | JamesDotCom | i'd do the same |
03:01.53 | JonR800 | i wish the 12" and 15" were as thin as the 17" |
03:01.56 | JamesDotCom | but i live in the middle of the city |
03:01.59 | JamesDotCom | dont need a car |
03:02.00 | *** join/#asterisk IOscanner (~IOscanner@24.0.186.72) |
03:02.20 | JamesDotCom | the 12" is thin enough |
03:02.24 | JamesDotCom | it's all of an inch? |
03:02.30 | JonR800 | more |
03:02.57 | JamesDotCom | 1.18 |
03:02.57 | *** join/#asterisk jake_ (~jake@gw-kit.locore.ca) |
03:02.58 | JamesDotCom | close enough |
03:03.06 | dima202 | BTW, aluminum is much better then TI |
03:03.12 | JonR800 | that .18 is a lot in person :) |
03:03.12 | JamesDotCom | amen |
03:03.16 | dima202 | It doesn't wear out |
03:03.17 | skeeziks | Does this mean anything to anyone here? |
03:03.21 | skeeziks | Nov 15 21:58:42 WARNING[1092078512]: channel.c:1445 ast_indicate: Unable to handle indication 3 for 'SIP/6782380063-a6da' |
03:03.24 | IOscanner | well it is how you use it... |
03:03.38 | IOscanner | sorry I think I missed a few things.... |
03:03.54 | JamesDotCom | only complaint with mine is that i've scratched the screen and the paint is wearing under where my palms sit |
03:04.17 | JamesDotCom | but i also love it for the fact that when i got the pb and started using the keyboard, my rsi went away |
03:04.28 | dima202 | can someone explain how to use asterisk? |
03:04.37 | dima202 | Asterisk Ready. |
03:04.39 | JonR800 | lol |
03:04.41 | JamesDotCom | dima202: what stage are you at? |
03:04.42 | *** join/#asterisk edguy (~edguy2@host-24-225-213-218.patmedia.net) |
03:04.51 | JonR800 | dima202: carefully. |
03:04.54 | dima202 | I don't even know what this does yet |
03:04.54 | JamesDotCom | well, start reading voip-info.org and start editing the configs in /etc/asterisk/ |
03:04.57 | funkknob | He just got the process running ;) |
03:05.11 | dima202 | But I have an idea how to make free* calling |
03:05.26 | dima202 | and if that works, I will post |
03:05.32 | funkknob | Do you have a phone? |
03:05.40 | funkknob | IP phone? |
03:06.05 | dima202 | I don't I thought this is soft. replacement>? |
03:06.19 | JamesDotCom | hrmm |
03:06.25 | funkknob | Well you can run a softphone |
03:06.27 | dima202 | I was wrong? |
03:06.32 | JonR800 | You have a lot of reading to do. |
03:06.40 | funkknob | Asterisk is a softswitch |
03:06.43 | dima202 | Argh! I knew it! |
03:06.49 | JamesDotCom | there's an x-lite client for os x |
03:06.50 | funkknob | Not a phone |
03:06.59 | JamesDotCom | get that, and you'll be able to get it to call to an asterisk server |
03:07.21 | JonR800 | dima202: http://www.voip-info.org/wiki-Asterisk |
03:08.10 | *** join/#asterisk michael12345 (~mick@staff2.tsn.cc) |
03:08.24 | michael12345 | I am told asterisk can auth off radius |
03:08.47 | michael12345 | is this true |
03:09.48 | mishehu | I heard rumors |
03:09.52 | JonR800 | http://lists.digium.com/pipermail/asterisk-users/2004-March/040837.html |
03:10.17 | mishehu | dammit, can't seem to find a heatsink that fits this mobo+proc |
03:10.31 | dima202 | guys, lets say my isp offers voip through cable modem |
03:10.34 | dima202 | and I have server |
03:11.03 | freestyle_networ | anyone here experienced with SIP DTMF? |
03:11.56 | mikegrb | :D |
03:12.38 | *** join/#asterisk daveGS (~none@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
03:13.02 | funkknob | dima202: no need to subscribe to your provider's voip service |
03:13.21 | funkknob | There are plenty of providers around |
03:13.27 | funkknob | And free solutions too |
03:13.27 | dima202 | I know that |
03:13.29 | scrubb | any broadvoicers around? |
03:13.38 | dima202 | free solutions? |
03:13.40 | scrubb | tryign to patch against todays tree. |
03:13.42 | dima202 | already available? |
03:13.54 | funkknob | http://fwd.pulver.com |
03:13.57 | funkknob | I think |
03:14.20 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
03:14.36 | funkknob | You can get free inbound numbers in US, UK and several others |
03:14.53 | funkknob | Calling to PSTN you can get flat-rate packages |
03:15.30 | dima202 | WHat is astericks for then guys? |
03:15.52 | funkknob | http://www.ipkall.com/ |
03:16.01 | funkknob | It's a phone switch |
03:16.16 | funkknob | and pbx |
03:16.20 | dima202 | WHat is the main funtion? briefly explained? |
03:16.37 | funkknob | It routes calls between phones |
03:16.57 | funkknob | Just like the switch at your PSTN or GSM provider |
03:17.17 | funkknob | So a network would look like this: |
03:18.00 | funkknob | IP Phone ----- Asterisk box --------provider(s) ------- PSTN or other IP subscriber |
03:18.26 | funkknob | Also it can provide voicemail, conference calling, etc. |
03:18.27 | dima202 | So do I really need it? |
03:18.33 | funkknob | Probably not |
03:18.36 | JonR800 | dima202: how can we tell you that? |
03:18.37 | dima202 | or is it like an addon |
03:18.37 | funkknob | But it can be fun |
03:19.14 | funkknob | You can use an IP phone at home, send calls to the PSTN using a Digium card or via IP to many different networks |
03:19.50 | funkknob | You can have multiple IP phones at home and ring diff. extensions within your house/business |
03:19.54 | JonR800 | It makes the thing with the handset go ring ring! |
03:20.00 | skeeziks | How stable is chan_sip2 vs. chan_sip? |
03:20.02 | mlh407 | if I have 3 phone janks all on one line, and someone is on jack #1, can I disconnect the call from jack #2. I have tried crossing the pairs but that does not do it, it only "flashes" the line. Is there anyway to disconnect the call? |
03:20.03 | funkknob | And have voicemail, IVR, etc. |
03:20.03 | *** join/#asterisk jcrock7 (~jcrock7@209.130.128.34) |
03:20.08 | freestyle_networ | anyone use the nagios plugin for asterisk? |
03:20.36 | *** join/#asterisk gregk (gregk@CPE0080c832cd7b-CM014340017512.cpe.net.cable.rogers.com) |
03:20.46 | funkknob | mlh407: yes, but you need a special little inline adaptor |
03:21.06 | funkknob | mlh407: on each phone |
03:21.19 | mlh407 | funkknob: how do I make/buy one? |
03:21.30 | funkknob | Hmmm it's been awhile, lemme look |
03:21.58 | *** join/#asterisk int19h (Miranda@219.95.154.87) |
03:23.40 | daveGS | hi there, anyone know how to use a softphone with vonage without paying for there extra softphone line |
03:24.05 | daveGS | They gave me a Linksys RT31P2 Device |
03:24.08 | mikegrb | daveGS: haw haw |
03:24.34 | daveGS | is that a yes i know how haw haw or a good luck youll need it one |
03:25.49 | mikegrb | number two |
03:26.03 | *** join/#asterisk PilotPTK-Home (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net) |
03:27.25 | funkknob | mlh407: Still looking, it's been years since I saw those things.... |
03:27.32 | mlh407 | funkknob: thanks |
03:27.44 | tzanger | funkknob: what the things that disconnect other phones when a call is being made? |
03:28.03 | ChulJin | it seems perhaps the opposite... |
03:28.15 | ChulJin | disconnected a call in progress on another jack on the same line |
03:28.36 | tzanger | sounds like some kind of device priority module |
03:29.28 | mlh407 | I need it for my kids |
03:29.45 | tzanger | mlh407: why... TDM440P and you have 4 lines :-) |
03:30.13 | mlh407 | tzanger: I use asterisk at work, I really don't want to use it at home. But you are right, that would also work ... "soft hangup ..." |
03:30.24 | tzanger | no no |
03:30.36 | tzanger | I mean you could carry 4 simultaneous conversations |
03:30.48 | tzanger | provided you only needed one POTS line |
03:31.11 | AlexZander | All these companies that provide local telephone numbers - are they just getting a bank of ISDN lines and hooking them up to an Asterisk type box to act as the bridge between voip and POTS or is it more complicated than that? |
03:31.13 | dima202 | I am having a problem registering everything right to the prog |
03:31.16 | mlh407 | I need to disconnect then when they are on late |
03:31.26 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
03:31.34 | dima202 | All they give me is SIP Phone Number, SIP Proxy, e-maail and pw |
03:31.41 | dima202 | Is that enough info to recieve calls? |
03:31.51 | tzanger | mlh407: meh, I just instill the fear of god in my kids |
03:32.00 | tzanger | "I catch you on the phone late and you lose it." |
03:32.22 | mlh407 | yes but I sleep too well for that.... |
03:32.42 | tzanger | heh |
03:32.57 | tzanger | well you could always put a relay in line with the tip or ring |
03:32.59 | tzanger | hell X10 it |
03:33.05 | funkknob | mlh407: http://www.sandman.com/lineshar.html - Maybe something there you can use, scroll down to exclusion privacy devices |
03:33.30 | tzanger | the rat shack modules would work too |
03:33.36 | dima202 | guys, is that sufficient info>? |
03:33.36 | tzanger | just pick up your phone to disconnect the others |
03:33.39 | PilotPTK-Home | use a script. absolutetimeout(23:00 - now) |
03:33.41 | PilotPTK-Home | basically... |
03:33.49 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
03:33.55 | PilotPTK-Home | and then have the context they dial out from only include during the hours you want them to be able to call. |
03:33.56 | funkknob | mlh407: You may just need to rewire your house, and put a switch in your bedroom |
03:33.57 | dima202 | I put username and Authorization username the phone number I was assigned |
03:34.01 | PilotPTK-Home | then you can still allow them 911 and other things after hours. |
03:34.42 | PilotPTK-Home | that way, at 11 (assuming thats the time you want to use), the phone would disconnect, and they could only dial a few selected numbers that you allow. |
03:34.48 | tzanger | yeah speaking of 911, I didn't realize that 911 was not working since we moved in June... oopsie |
03:35.16 | doughecka_ | tzanger: oh well, that person who had the heart attack wasnt liked anyway |
03:35.22 | tzanger | doughecka_: heh |
03:35.31 | tzanger | it's not like there aren't a dozen cell phones in the place |
03:35.35 | doughecka_ | yea |
03:35.42 | doughecka_ | till the cell towers are taken out |
03:35.44 | tzanger | and it's working now :-) |
03:35.45 | doughecka_ | :P |
03:35.54 | tzanger | well when the towers are taken out there will be bigger issues |
03:36.06 | doughecka_ | yea :P |
03:36.19 | tzanger | if your cell towers disappear it's likely your internet connectivity will die too |
03:36.56 | tzanger | I should play with DUNDi more... how stable is it in terms of call quality? I mean if you're just using anyone's lines I imagine there'd be a lot of change in quality from call to call |
03:36.58 | doughecka_ | this week I am in vmware training, and theres this guy that works in nashville who is in my same class... he was talking about hot sites and stuff like that... |
03:37.06 | doughecka_ | actually I have never used dundi :P |
03:37.23 | doughecka_ | well, basicly, if the hospital gets taken out... what use is the client data gonna be? |
03:37.26 | tzanger | vmware training? |
03:37.30 | tzanger | what exactly do they train you on? |
03:37.34 | doughecka_ | eh |
03:37.37 | doughecka_ | vmware ESX server |
03:37.43 | tzanger | ok :-) |
03:38.05 | tzanger | is there that much of a speedup from eliminating the host OS with a slim vmware scheduler and I/O abstractor? |
03:38.13 | doughecka_ | speedup? |
03:38.24 | tzanger | difference between GSX and ESX |
03:38.29 | doughecka_ | oh |
03:38.38 | doughecka_ | well, ESX is redhat 7.2 |
03:38.49 | funkknob | dima202: You can receive calls from other users on the system, if you want to receive PSTN calls, try http://www.voip-info.org/wiki-Telesthetic |
03:38.50 | doughecka_ | with ANOTHER kernel sittin beside it doing direct hardware stuff |
03:38.53 | tzanger | IIRC GSX runs on a host OS still, whereas ESX uses a thin kernel that pretty much abstracts I/O only |
03:38.58 | tzanger | ahh maybe I have GSX and ESX mixed up |
03:39.05 | doughecka_ | no, you are correct |
03:39.17 | doughecka_ | but ESX is a contained package |
03:39.21 | doughecka_ | gsx is vmware only |
03:39.23 | dima202 | You mean it is not a real number? |
03:39.26 | tzanger | ahh |
03:39.28 | dima202 | just like a nick |
03:39.38 | doughecka_ | esx is 80%-98% cpu speed |
03:39.53 | doughecka_ | gsx is more like 60%-80% |
03:39.56 | dima202 | wonder why they speak of washington numbers currently free |
03:40.13 | tzanger | doughecka_: GSX is slower than ESX? WTF's the advantage then? |
03:40.28 | doughecka_ | esx is more expensive |
03:40.41 | tzanger | but esx runs on a host OS? |
03:40.43 | doughecka_ | gsx is far cheaper |
03:40.47 | tzanger | <-- confused |
03:41.01 | tzanger | perhaps I need some vmware training :-) |
03:41.02 | doughecka_ | ok, you have 3 products |
03:41.05 | doughecka_ | workstation |
03:41.07 | doughecka_ | gsx |
03:41.09 | doughecka_ | and esx |
03:41.36 | tzanger | yes I use workstation at least twice a week |
03:41.38 | funkknob | dima202: |
03:41.41 | doughecka_ | gsx == a better workstation |
03:41.42 | funkknob | dima202: http://phone.ipkall.com/ipphone/ |
03:41.54 | tzanger | played briefly with either GSX or ESX on slackware |
03:41.56 | doughecka_ | with a web gui |
03:42.01 | dima202 | I am getting time outs when i TRY TO log in |
03:42.06 | doughecka_ | then you played with gsx |
03:42.08 | dima202 | though the phone prog |
03:42.24 | tzanger | and ESX's scheduler/abstraction is just RH7.2?? |
03:42.50 | doughecka_ | well, yes and no |
03:43.03 | doughecka_ | it runs redhat 7.2 as the base OS, for basic booting... |
03:43.06 | funkknob | Xlite? |
03:43.11 | dima202 | yes |
03:43.14 | dima202 | can I pm you? |
03:43.21 | doughecka_ | but theres a vmkernel that runs beside the redhat kernel |
03:43.22 | funkknob | sure |
03:43.37 | doughecka_ | and it accesses most hardware directly |
03:44.01 | doughecka_ | NICs, drives, SANs, long range photon cannons, etc |
03:45.41 | tzanger | heh |
03:46.03 | doughecka_ | so the vmkernel adds a whole level of speed increase |
03:46.07 | daveGS | what is a good voip provider that doesnt have vonages restrictions |
03:46.13 | tzanger | daveGS: nufone |
03:46.26 | tzanger | I use them pretty much exclusively with vpc as a backup |
03:46.30 | daveGS | tzanger: thanks |
03:46.30 | doughecka_ | while GSX (running on a windows/linux host OS) offers a cheaper, but less performance product |
03:46.34 | tzanger | and then my PRI as a backup backup :-) |
03:46.41 | tzanger | doughecka_: right i Understand now |
03:47.17 | doughecka_ | GSX DOES offer more hardware support, anything the host OS can support, but ESX supports fiberchannels and junk like that. |
03:48.21 | tzanger | right right |
03:48.37 | *** join/#asterisk brettnem (~brettnem@user-0ccsr4b.cable.mindspring.com) |
03:48.38 | h3x0r | esx is for those who want to spend a shitload of money to save money doing virtual server colos |
03:48.38 | h3x0r | heh |
03:48.58 | doughecka_ | well... |
03:49.10 | doughecka_ | and other people |
03:51.42 | *** join/#asterisk Kb1_Kanob (johnsmith@sec2d7.dial.uniserve.ca) |
03:52.09 | int19h | word up all |
03:52.09 | int19h | , is anyone anywhere in the world having trouble getting calls through IAXtel, in either direction? |
03:53.24 | Kb1_Kanob | evening all. |
03:53.39 | Kb1_Kanob | Has mantis started eating it's own children? I seem to have lost a bugreport... |
03:54.48 | gregk | i've given up on IAXtel a week ago. then again I never had it running before, so maybe I was doing it wrong |
03:55.47 | mishehu | I'm going to kill intel. |
03:56.19 | gregk | Any bug marshalls out here? How do I go about having a bug reopened? |
03:56.36 | mlh407 | What bug # |
03:56.38 | redder86 | gregk: just repost to the bug report, it will reopen |
03:56.39 | JunK-Y | grefk: which #? |
03:56.45 | int19h | hmmm, I had it working great... till my ISP (pretty much the only broadband ISP in malaysia) decided to grciously 'upgrade' all it's customers... 384->512... 512->1M.... and since IAXtel been broken |
03:56.47 | int19h | bah |
03:57.07 | gregk | bug_id=0001822 |
03:57.07 | kram | what's wrong with iaxtel btw/ |
03:57.12 | kram | what is the symptom? |
03:57.23 | kram | it's on my list to work on today |
03:58.16 | gregk | mlh407, Junk-Y: bug_id=0001822 " Quick Net Internet Phonejack Goes Bezerk" |
03:58.26 | twisted | kram: it's up and down and up and down and up and down |
03:58.33 | twisted | and generally nobody can make calls through it |
03:58.53 | JunK-Y | gregk: maybe twisted could re-open it. |
03:58.57 | JamesDotCom | is there a way to get more debug output from the h.323 driver? |
03:59.10 | twisted | gregk, one sec |
03:59.13 | BoRiS | hey kram, Can you check the callerid column from a database in realtime config? It doesn't seem to be taking the name to show as callerid (SIP dialing). Also, MWI doesn't work with realtime config. :( |
03:59.32 | redder86 | Use iaxtel for real calls? Hah! |
04:00.00 | twisted | gregk, i closed that out over 2 months ago, because it was inactive for almost a month before that. |
04:00.08 | int19h | kram: this is as far as I can get... Called iaxtel-outbound/18009797900@iaxtel... and then WARNING[1096813488]: chan_iax2.c:1473 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel-outbound/16384 (type = 6, subclass = 9, ts=12610, seqno=2) |
04:00.43 | twisted | gregk, i'll re-open it, but it will get closed again if nothing happens for awhile. |
04:01.04 | twisted | gregk, it's open. |
04:01.13 | *** join/#asterisk adavee (adavee@S01060090f8009e11.va.shawcable.net) |
04:01.18 | redder86 | yup... iaxtel is down |
04:01.43 | gregk | twisted:I can reproduce it very easily. We'll see |
04:01.51 | gregk | twisted: thanks |
04:01.51 | twisted | gregk, okay. post away :) |
04:02.09 | adavee | is it ok for a newbie to ask a couple questions? |
04:02.15 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-40-71.dsl.pltn13.pacbell.net) |
04:03.44 | Kb1_Kanob | a good evening question - Is incoming audio data from a zaptel interface passed through a high-pass filter anywhere to remove DC offset effects? I have noticed the "base" signal level on my calls seems to vary wildly - yet my end is PRI. So, the serving switch must be passing me a line artifact from the other parties analog line, no? |
04:03.50 | h3x0r | i should set up a iaxtel clone, plus DIDs |
04:03.57 | h3x0r | its probably not a bad idea |
04:04.13 | h3x0r | i've got a couple local PRIs im not doing anything with |
04:04.21 | Kb1_Kanob | adavee: usually best to just ask, albeit politely... |
04:05.22 | adavee | I just signed up for Voip and am severing my two telco lines (one for us and one for the kids). I'd like to get some kind of simple setup where calls to the one voip # could be redirected to the kids phone. I thought a simple PBX setup might work. Would Asterisk work well for this kind of setup? |
04:05.30 | Kb1_Kanob | following on from the "dc offset", I get complaints of bad audio on some calls that are pumped over a gsm codec. Not all though - could a reduced dynamic range be a source of problems? |
04:05.32 | *** join/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net) |
04:06.05 | redder86 | adavee: yes |
04:06.12 | h3x0r | adavee: many voip providers lock you to their hardware and won't let you use your own software/hardware |
04:06.12 | Kb1_Kanob | adavee: how would you differentiate between the calls? |
04:06.19 | h3x0r | such as vonage |
04:06.21 | adavee | I just wasted some $$ on a VoicePro 206 that doesn't seem to work....so searching around led me to Asterisk |
04:07.12 | adavee | someone would call my voip # and then the voicemail attendant would ask .."for the kids, hit 5" or something like that?? |
04:07.35 | redder86 | adavee: yes, if you wanted, you could do that |
04:08.48 | Kb1_Kanob | adavee: troll through the voip-info.org wiki a bit. Look at implementing an "IVR" |
04:09.06 | adavee | thanks, I will |
04:09.14 | redder86 | adavee: your VP206 doesn't work? |
04:09.23 | adavee | no, it doesn't seem to |
04:09.36 | adavee | it just sits there with the power light on |
04:10.12 | redder86 | adavee: what's it supposed to do? |
04:10.55 | adavee | its supposed to give me a dial tone when I pick up a phone...and then it can direct calls to other extensions in my house - it has 5 phoune outlets |
04:11.07 | adavee | but it does nothing |
04:11.19 | redder86 | got it on eBay? |
04:11.44 | adavee | no, not directly |
04:12.25 | redder86 | well, I see them on eBay for $170. You'll spend at least double that for your setup with Asterisk. |
04:12.59 | adavee | I paid $100 for it but a lesson learned - I'd rather do it the "right" way :) |
04:13.31 | redder86 | adavee: you'll need one or two FXO ports (~$20 each), and then an ATA device or two ($100 each). |
04:13.51 | redder86 | adavee: you have broadband? |
04:14.00 | adavee | high speed cable |
04:14.18 | adavee | I have an ATA device from Broadvox - you mean that? |
04:14.26 | redder86 | possibly |
04:14.40 | redder86 | is it a SIP device? |
04:15.05 | adavee | hmm, not sure what that means.. |
04:15.22 | tzanger | any recommended places to purchase Polycom IP300 phones? Looking for about a dozen of 'em |
04:15.24 | redder86 | What is the model number information? |
04:15.47 | adavee | Mediatrix 2102 |
04:16.18 | Mavvie | seems that my question with regarding to modem emulation for SIP/IAX was too difficult :-/ |
04:16.32 | *** join/#asterisk rje (~rje@c-67-160-57-179.client.comcast.net) |
04:16.52 | Kb1_Kanob | anyone familiar with the concept of "padding" a PSTN line? either digital or analog forms? |
04:17.13 | redder86 | adavee: "Deployable in SIP VoIP networks" that's a yes |
04:17.32 | adavee | you just searched for that? :) |
04:17.51 | redder86 | adavee: www.voip-info.org, then I went to the manufacturer's website, yet |
04:17.52 | redder86 | yes |
04:18.19 | redder86 | Mavvie: what are you doing? |
04:18.37 | redder86 | adavee: do you want to keep your existing PSTN number? |
04:18.50 | adavee | no, that doesn't matter |
04:18.52 | redder86 | adavee: do your kids have their own number that they solicit? |
04:19.25 | redder86 | adavee: well you wouldn't need Asterisk, then, unless you wanted it. You should be able to set up your Mediatrix with a VoIP provider. |
04:19.37 | adavee | I was thinking that the kids would just get an ext. # off the main voip # |
04:19.40 | redder86 | adavee: they would provide you a number, called DID |
04:20.04 | Mavvie | redder86: http://lists.digium.com/pipermail/asterisk-users/2004-November/072668.html |
04:20.38 | Mavvie | redder86: the idea is to have some kind of modem emulator device which does do the calls via a SIP link instead of a PSTN link |
04:20.39 | redder86 | Mavvie: okay, gotcha, but what are you trying to do with the emulated modem? |
04:21.00 | redder86 | slePP: around? |
04:21.13 | Mavvie | redder86: I'm trying to get rid of a BRI card ;-) |
04:21.19 | Mavvie | well, that's the effect of it. |
04:21.20 | rje | ADSI question... After posting to the users list and thy helped me part of the way I still cannot get a greeting/idle screen to show up on my 390. I've got the slots set right and the keys show up when i go off hook. Just no idle gretting. Anyone have some ideas? |
04:21.21 | adavee | ok, I'll read up a bunch at the listed sites! |
04:21.23 | redder86 | Mavvie: for faxing? |
04:21.35 | Mavvie | dialin / dialout and faxing. |
04:22.31 | Mavvie | redder86: it's just that it's easier for fall-over that I don't have to worry about too many ISDN connections. One PRI is cheaper than four+ BRIs. |
04:22.39 | redder86 | Mavvie: most of the code for faxing exists already in spandsp. slePP was working on getting an IAX client to connect with the spandsp libraries, which already has a T.31 modem available, and then interface that with a device node. |
04:23.17 | Mavvie | oh! that's interesting. |
04:23.17 | int19h | hey, has anyone seen an * app that can apply audio filters to a call? like reverb or distortion or whatever? |
04:23.17 | redder86 | Mavvie: as for data calls, I don't know anything about that. |
04:23.42 | Mavvie | but only fax. data calls is then still an issue. |
04:24.08 | redder86 | Mavvie: coppice, the spandsp author, was going to make a pseudo-modem channel driver instead. So it would provide pseudo-modems on the same system as Asterisk. |
04:24.27 | redder86 | Mavvie: spandsp may work for data, but I don't know. I only concern myself with fax. |
04:24.41 | Mavvie | redder86: aha, that sounds like a way to go for legacy apps which want to talk to /dev/something. |
04:25.09 | *** join/#asterisk rustyb (~rustyb@68.235.250.116) |
04:25.21 | redder86 | Mavvie: the only difference, in the end-result, I think between slePP's and coppice's approach is that slePP's would provide device nodes on remote systems. |
04:25.36 | Mavvie | redder86: I'm going to figure out if I can get it up and running, will at least get rid of three of the four cards if I get faxing running. |
04:25.50 | redder86 | Mavvie: coppice worries about frame slips if the machine is remote |
04:26.10 | redder86 | Mavvie: I don't think that there is anything in a finished state, yet. |
04:26.38 | dima202 | Discovered Full Cone NAT Firewall |
04:26.41 | dima202 | is that bad? |
04:27.11 | redder86 | Mavvie: right now to integrate Asterisk and faxing you've got to either connect a separate fax device to an FXS port on Asterisk, or you have to use spandsp's txfax/rxfax. |
04:28.07 | daveGS | nufone sux they have horrible service, cant even sign up for a phone number |
04:28.21 | redder86 | daveGS: NuFone's been good to me. |
04:28.31 | twisted | daveGS, if you say so |
04:28.38 | twisted | i've not had a problem |
04:28.42 | adavee | one last question, can I use my existing cordless phones with Asterisk, or do I need specific phones? |
04:28.59 | h3x0r | http://www.knopsterisk.com/ |
04:29.01 | tzanger | daveGS: you just asked me not even an hour ago who to use ... and now you're an expert on nufone? |
04:29.04 | h3x0r | thats violating so many gpl's |
04:29.07 | h3x0r | i dont know where to begin |
04:29.08 | redder86 | adavee: the purpose of an ATA device is to allow you to use regular old phones with VoIP |
04:29.23 | adavee | hehe, ok |
04:29.47 | dima202 | so guys what does this mean? Discovered Full Cone NAT Firewall |
04:29.56 | dima202 | and Discovered Blocked Firewall |
04:29.58 | redder86 | h3x0r: how does it violate the GPL? |
04:30.07 | herag | my vsp gave me a patch to chan_sip.c the other day and is shoving it down my throat, if I just patch and recompile that one file, can I just replace it's respective .so in the /var/lib/asterisk/modules dir and just restart *? or do I have to go through the entire recompile process? |
04:30.35 | tzanger | ahh herag's another happy broadvoice customer |
04:30.43 | herag | ya, no frick |
04:30.47 | h3x0r | you can't download it |
04:31.03 | herag | I'm getting a little pissed |
04:31.18 | redder86 | h3x0r: does the GPL require it to be downloadable? |
04:31.45 | herag | I finally had a little time to work on configuration of my extensions, and now they go about screwing with my registration functions....my server won't even register with them properly now |
04:32.01 | tzanger | herag: in theory you can just do the one file, unload chan_sip.so or whatever it is and reload it without taking all of * down |
04:32.04 | redder86 | herag: try the former and if that doesn't work then the latter |
04:32.21 | tzanger | anyway it is my beddy bye time, later all |
04:32.49 | redder86 | herag: is Broadvoice going to submit a bug report on that thing or what? |
04:33.07 | Kb1_Kanob | tzanger: gnight. |
04:33.08 | herag | yes, actually, they claim their patch will be incorporated into the cvs |
04:33.15 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
04:33.32 | redder86 | herag: that's what their e-mail claims, but they have to jump through Digium's hoops to get it there. I wonder if they will. |
04:33.40 | tzanger | Kb1_Kanob: :-) Didn't see you there, 'night |
04:33.59 | Kb1_Kanob | quietly slipped in the back door... |
04:34.09 | herag | redder86: well, supposedly, the patch was made by astricon devs, so I dunno |
04:34.18 | tzanger | herag: well it was a * dev who made the patch for them and the patch is good, I don't doubt it is already in CVS HEAD |
04:34.36 | tzanger | in fact I am certain I saw email about it in the asterisk-cvs list |
04:34.50 | skeeziks | Anybody have this problem with SIP? WARNING[1092078512]: channel.c:1445 ast_indicate: Unable to handle indication 3 for 'SIP/6785550063-dee1' |
04:34.54 | redder86 | herag: the noise on the -users was confusing |
04:35.07 | tzanger | something about "fixing olle's sneakiness" or sommat :-) |
04:35.09 | herag | olle johansson and steve sokol wrote the patch |
04:35.11 | tzanger | anyway 'night for real now :-) |
04:35.38 | herag | this is the error I'm getting now when trying to dial out: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP' |
04:35.44 | herag | what the frick does that mean? |
04:35.47 | redder86 | I would have thought that if it were in CVS HEAD that Broadvoice would have told their customer base to upgrade to CVS HEAD rather than distributing a patch to all of them. |
04:36.07 | redder86 | herag: it means that SIP isn't working |
04:36.28 | herag | redder86: is that a source problem, or registration problem? |
04:37.01 | redder86 | herag: not sure |
04:37.04 | herag | grr |
04:37.16 | redder86 | herag: I would guess a SIP configuration problem |
04:37.28 | redder86 | kram: was that Broadvoice patch integrated into CVS HEAD? |
04:37.29 | herag | hmm |
04:38.00 | *** join/#asterisk adavee (adavee@S01060090f8009e11.va.shawcable.net) |
04:39.32 | Mavvie | h3x0r: I'm not really sure how it violates the GPL... |
04:40.04 | Mavvie | they say it's plain asterisk, and their FAQ shows how to make one yourself. |
04:40.20 | herag | sigh... |
04:40.31 | herag | it's the patch's fault |
04:40.35 | herag | I gotta recompile everything |
04:40.40 | herag | fricken bv |
04:40.49 | redder86 | hehe... there's like 100 votes' difference in the gubenatorial race in WA |
04:42.17 | adavee | ok, I'm looking at pic of a FXO module @ Digium - I've read that I'll need one - it doesn't look like it goes in a PCI or ISA slot, - how are the connected? |
04:42.54 | redder86 | adavee: those go on the TDM cards. You want the X100P |
04:43.10 | adavee | ok |
04:43.44 | adavee | ahh, there it is |
04:45.06 | redder86 | adavee: the X100P is an Ambient/Intel winmodem (AMI-IA92/IE92) for which Asterisk has its own driver and comes with installation/setup support from Digium |
04:46.12 | *** join/#asterisk tetraz (~asko@S010600067b03be4c.vs.shawcable.net) |
04:46.16 | adavee | does anyone have any user websites that show some detailed steps in setting up a small home steup - even just one line? I'm just so new to all this. But I'm eager to learn. I should only need one X100P with one line correct? |
04:46.29 | *** join/#asterisk charlesIII (~charlesII@c-67-161-110-59.client.comcast.net) |
04:46.36 | redder86 | adavee: yes, one x100P for one line |
04:46.46 | tetraz | Is there an asterisk newbie channel? |
04:46.53 | adavee | haha |
04:46.54 | charlesIII | anyone have spandsp working with freebsd? |
04:46.56 | PilotPTK-Home | tetraz: www.voip-info.org |
04:47.08 | adavee | I'm reading lots there right now |
04:47.28 | adavee | still quite confusin |) |
04:47.31 | JunK-Y | tetraz: ask ur questions. |
04:47.48 | tetraz | :) How do I know if asterisk recognized my soundcard? |
04:47.58 | adavee | in a nutshell this would be my simple setup: |
04:48.03 | redder86 | adavee: unfortunately the entry level learning curve requirement for Asterisk is quite high |
04:48.20 | tetraz | redder86: No kidding :) |
04:48.35 | redder86 | adavee: voip-info.org has some good links. Don't trust everything you read there (or anywhere else, either), though. |
04:49.01 | *** join/#asterisk florz (nobody@I811c.i.pppool.de) |
04:49.04 | JonR800 | does anyone else have trouble with voipjet taking about a full minute to dial out??? |
04:49.24 | redder86 | adavee: even if you're in-the-know with Linux and telephony you'll probably spend many, many hours getting a basic Asterisk system going |
04:49.42 | adavee | internet connection --> cable modem --> hub/switch --> one connection to Asterisk box in --> Asterisk box out to my cordless phones |
04:50.17 | adavee | other connection --> router --> my main computer |
04:50.20 | charlesIII | freebsd --- spandsp... anyone? |
04:50.58 | redder86 | adavee: this looks promising: http://www.surfcity.com/Asterisk/default.htm |
04:51.43 | redder86 | charlesIII: s/freebsd/linux/ ? |
04:52.07 | charlesIII | one can hope, you know? |
04:52.18 | charlesIII | nah, specifically looking for freebsd |
04:52.36 | BoRiS | Anyone know why when I try to dial an H323 phone (exten => 123,1,Dial(H323/1800xxxxxxx@192.168.x.x). When I dial 123, My console keeps showing "Nov 15 22:53:34 NOTICE[16306]: pbx.c:1341 pbx_extension_helper: Cannot find extension context 'default'" even though I just want it to dial? |
04:53.12 | redder86 | charlesIII: the way that I see it, the box is there for the purpose of running the applications, and if the application that you need to run is Asterisk+spandsp, then well, Linux is the OS you should use. Likewise, if you want to run MS Access... well, then you're going to need to run Windows. |
04:53.39 | charlesIII | I don't think inquiring about ports of software is too weird, considering asterisk runs well on freebsd |
04:53.45 | redder86 | BoRiS: you need to have a context where it can dial from |
04:54.09 | redder86 | charlesIII: but spandsp doesn't |
04:54.23 | charlesIII | I am asking if someone has ported it or is in the process... |
04:54.28 | redder86 | oh |
04:54.56 | redder86 | your message "freebsd --- spandsp... anyone?" was unclear about that |
04:55.01 | charlesIII | sorry |
04:57.51 | *** join/#asterisk hmodes (hmodes@dsl092-231-153.phl1.dsl.speakeasy.net) |
04:58.41 | hmodes | oy |
05:02.20 | abombss | anyone in here use kdevelop with * |
05:02.49 | abombss | or anyone else use any other ide? |
05:04.30 | kram | well you can restart yah |
05:05.57 | dima202 | hey guys, is this site for real? http://phone.ipkall.com/ipphone/ |
05:06.19 | dima202 | DO they really give you a phone number and can be reached from land line phone for free? |
05:06.26 | *** part/#asterisk PilotPTK-Home (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net) |
05:06.43 | kFuQ | dima202: yes |
05:06.45 | brettnem | dima202: I used them.. it works |
05:06.50 | kFuQ | i have 2 #'s |
05:06.53 | dima202 | I dont understand it |
05:06.54 | kFuQ | thru them |
05:07.07 | dima202 | I configured everything properly to make calls |
05:07.16 | dima202 | but not to recieve using their service |
05:07.31 | brettnem | I think they are termination only.. |
05:07.36 | brettnem | er.. origination |
05:07.43 | dima202 | 360-519-5157 |
05:07.50 | brettnem | meaning they send calls to your did |
05:07.50 | dima202 | here is my number they issued |
05:08.18 | dima202 | I dont even know where to include the info config of SIP Proxy an d SIP Phone, PW |
05:11.35 | dima202 | How do I tell them to use my farward number so I can reach the call? |
05:15.28 | dima202 | can anyone help me set this up |
05:19.30 | *** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
05:19.45 | Moc | hail |
05:19.54 | Dr_Ray | Moc yesterday you talked about a dial around 411 provider, any name of a company? |
05:20.10 | Moc | Dr_Ray, I just play a prompto use 411.ca |
05:20.25 | Moc | someone talked about a provider that sold in bulk or something |
05:20.40 | Dr_Ray | oops, thought that was you |
05:20.41 | Dr_Ray | my bad |
05:21.30 | Moc | let me check my logs |
05:22.37 | Moc | ok found it |
05:22.48 | daveGS | tzanger: ya i actually went to sign up with them, their web app doesn't register any phone numbers either michigan or toll free, theres no number for customer support |
05:22.57 | daveGS | thats why i say they suck |
05:23.09 | Moc | <AMAG> You can redirect them to a toll-free 411 service for like $0.30/call with no real volume commitment |
05:23.40 | Moc | that all I see |
05:23.58 | daveGS | and they didnt even register me under the correct plan |
05:24.30 | Dr_Ray | thanks Moc |
05:25.29 | bkw_ | YAY asterlink is live |
05:25.31 | bkw_ | :P |
05:27.18 | Moc | so what new today ? |
05:27.37 | daveGS | i just signed up for it with ipkall, its working fine |
05:28.20 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
05:28.43 | Dr_Ray | Dave in Seattle? |
05:28.56 | Moc | I hope to get my polycom at the end of the month.. REally hope |
05:29.10 | JunK-Y | moc: ip500? |
05:29.11 | Moc | but I guess I'll have it by new year .. |
05:29.18 | Moc | JunK-Y: IP 600 |
05:29.20 | JunK-Y | nice |
05:29.27 | JunK-Y | going to bed, tty tomorrow moc. |
05:29.29 | Moc | yea I decided to try it out |
05:29.34 | Moc | allright cya |
05:29.36 | JunK-Y | hehe nice guess |
05:29.39 | JunK-Y | :) |
05:31.08 | *** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
05:32.41 | _Vile | mmmmm |
05:32.46 | _Vile | HI |
05:33.48 | _Vile | hi |
05:33.50 | _Vile | sdjld |
05:33.53 | outtolunc | howdy |
05:35.27 | freestyle_networ | i have a 600 ..love it |
05:36.57 | *** join/#asterisk ender (~me@ender.fedora) |
05:37.40 | fOSSiL | anyone know how to upgrade IAXy firmware on an early-release box? |
05:38.05 | ManxPower | fOSSiL: You have to send it to Digium. |
05:38.32 | ManxPower | I think digium can tell you based on the srial number of they have to do the first firmware upgrade |
05:39.10 | ManxPower | I think Digium can tell you, based on the serial number, if they have to do the first firmware upgrade. |
05:39.49 | _Vile | man I keep taking these billing tickets and they don't seem to be stopping.. damn the top 10 newest unowned tickets.. |
05:39.54 | fOSSiL | yeah, i got that, thanks :) |
05:40.48 | fOSSiL | i think this one definetely needs a digium upgrade -- the MAC addy is d00d00f00dXX |
05:41.02 | _Vile | d00d f00d |
05:41.13 | fOSSiL | ya :P |
05:41.25 | ManxPower | fOSSiL: Yes. That MAC address made me spend 2 hours tracking it down. |
05:41.25 | *** join/#asterisk Guest^DJ (~mmmm@219.94.64.226) |
05:41.32 | fOSSiL | lol |
05:41.58 | ManxPower | It was on an unmanaged switch connected to an unmanaged switch connected to our backbone managed switches. |
05:42.41 | fOSSiL | heh, how the heck did you find it then? |
05:42.43 | _Vile | fun |
05:43.07 | fOSSiL | the cable it was hooked up with did not have any dust on it? :) |
05:43.09 | ManxPower | fOSSiL: Pinged the IP, then started unplugging stuff |
05:43.14 | _Vile | hahahaha |
05:43.21 | _Vile | i was just thinking that |
05:43.40 | ManxPower | We localzed it to MIS/accounting/marketing |
05:44.27 | outtolunc | sounds localized to me <G> |
05:45.10 | Guest^DJ | anyone knows where to get FXS clones for * ? |
05:45.15 | fOSSiL | all network hubs and switches need a "send 10,000 volts to <this> port" function |
05:46.13 | robl^ | at 10,000 watts |
05:46.13 | Nugget | I am in full support of that and also SITFoIP technology. (Stab In The Face over IP) |
05:46.13 | ManxPower | Guest^DJ: They don't exist. |
05:46.13 | _Vile | Guest, you'll be happier with a t100p and a channel bank full of fxs's |
05:46.13 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:46.13 | Dr_Ray | digium derves the money |
05:46.13 | Dr_Ray | er, deserves |
05:46.24 | ManxPower | outtolunc: 8-) |
05:46.38 | ManxPower | Can I call Guest^DJ "cheap"? Can I? Can I? |
05:46.48 | Guest^DJ | sure u can, ManxPower lol |
05:47.01 | Guest^DJ | well, place i am in T1 dont exist |
05:47.02 | _Vile | guest, they don't exist, and if they do, you'll have a bitch of a time with echo.. |
05:47.09 | _Vile | Guest, doesn't matter |
05:47.26 | *** join/#asterisk |Blaze| (dirc@d142-59-247-192.abhsia.telus.net) |
05:47.31 | _Vile | Analog Lines <-> FXS Ports on Channel Bank <-> T1 <-> Asterisk |
05:47.49 | _Vile | depending on # of lines you're looking for |
05:47.49 | ManxPower | Guest^DJ: A Channel bank splits a T-1 to Asterisk into 24 analog ports. |
05:48.02 | _Vile | yeah that |
05:48.03 | ManxPower | Theefore T-1 no connection to the telco needed |
05:48.03 | _Vile | too |
05:48.05 | *** part/#asterisk int19h (Miranda@219.95.154.87) |
05:48.21 | Guest^DJ | hmmmm, i am in asia, so only E1 is available here |
05:48.43 | Dr_Ray | e1 gives you 32 ports |
05:48.51 | Mavvie | 30 |
05:48.53 | Dr_Ray | 30 |
05:48.57 | Dr_Ray | :) |
05:49.07 | Dr_Ray | which I was tempted to use here in the states |
05:49.15 | Guest^DJ | yup, i have a T410P |
05:50.16 | Guest^DJ | Dr_Ray, i am going nuts over * |
05:50.24 | Dr_Ray | good nuts? |
05:50.50 | Guest^DJ | after whole night of reading, dont know what the FXXXX is going on |
05:51.00 | Dr_Ray | more reading |
05:51.07 | Dr_Ray | hitchikers guide to asterisk |
05:51.10 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
05:51.13 | ManxPower | To search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms. |
05:51.27 | Guest^DJ | then i am gonna change from good nut to bad nut |
05:52.55 | _Vile | what are you trying to do, guest? |
05:54.17 | Guest^DJ | just trying to built 2 *, 1 with 3 FXO and 4 FXS on each end, and connect both * using IP |
05:55.31 | _Vile | and you have 1 T410P? |
05:56.16 | Guest^DJ | oh, that 1 T410P is already running in my firm, it was built by a developer from here |
05:56.42 | Himeko | not all of asia is t1 |
05:56.45 | Himeko | er e1 |
05:56.52 | Guest^DJ | now trying to built analog based |
05:56.57 | Himeko | but most is |
05:57.04 | Guest^DJ | Himeko, we have E1/t1 here |
05:57.48 | Himeko | i wouldn't think they woudl have both |
05:58.02 | Himeko | t1 is NA and japan |
05:58.05 | Himeko | e1 everywhere else |
05:58.22 | Corydon76-home | I thought Japan was J1 |
05:58.35 | ManxPower | Japan is J1 |
05:59.00 | Guest^DJ | E1 is common here, but T1 is also available here, but telco guys are trained with E1 knowledge |
05:59.03 | Himeko | they might call it that too |
05:59.30 | Himeko | i seen it called t1-Jsomthing |
05:59.40 | Corydon76-home | Isn't J1 basically the same as T1, but with enough incompatibilities introduced to ensure that T1 equipment will not work there? |
05:59.53 | Himeko | probably |
06:00.02 | outtolunc | the question is not what you can do for your country, the question is what you can do with your channel |
06:00.38 | Guest^DJ | why cant they just sit down and agree on the same standard |
06:01.04 | Himeko | hehe |
06:01.08 | outtolunc | i'll take ego's for $100 |
06:03.18 | Himeko | There are just a few minor |
06:03.19 | Himeko | differences, designed purely to make it incompatible. They just changed |
06:03.19 | Himeko | some alarm flags and CRC patterns, if I recall correctly. |
06:03.48 | outtolunc | just look at the ITU specs |
06:04.05 | Himeko | i guess taiwan used to use it too |
06:04.13 | Himeko | callignit a T1M |
06:04.37 | outtolunc | not saying that US providers follow any better |
06:04.38 | Himeko | but when the telecom was deregulated there they use standard t1 and e1 |
06:05.14 | outtolunc | i faught with SBC abount SIT tones ..'hard' about 7 months ago |
06:05.59 | outtolunc | seems they have forgetten that 'some people' still require inband compliance |
06:07.16 | outtolunc | wouldn't they just love my publishing their in-compliance |
06:07.59 | Guest^DJ | hi |
06:08.06 | Himeko | i guess 2 bits have to be twidlled in the framer for a t1 |
06:08.15 | Himeko | but the isdn spec is slightly off too |
06:10.22 | *** join/#asterisk brc_ (~bob@brc.base.supporter.pdpc) |
06:10.23 | ZX81 | can you have host=dynamic with a peer? i.e. and have him register for peer and user account with one line? or do I have to create two register lines (i.e. 1 for peer and one for user?) |
06:12.23 | ZX81 | ah well type=friend then |
06:13.47 | ZX81 | compliance with |
06:13.57 | outtolunc | read up |
06:14.01 | ZX81 | i.e. certifications / permits? |
06:14.03 | ZX81 | ok |
06:14.20 | ZX81 | ah yeah |
06:15.01 | _Vile | i don't like sbc |
06:15.03 | _Vile | nor qwest |
06:15.26 | outtolunc | like or dislike have nothing to do with this |
06:16.53 | _Vile | incompetancy and care-free ...? getting warmer? |
06:16.59 | _Vile | bad companies |
06:17.34 | outtolunc | is your phone provider SBC under the hood, is it working? |
06:17.57 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-85-82.sentechsa.net) |
06:19.36 | outtolunc | maybe you aren't aware of the issues with 'detecting SIT' |
06:19.52 | outtolunc | inband |
06:21.43 | outtolunc | well |
06:26.59 | outtolunc | ITU q.35, e.182, e.181 should help |
06:27.26 | WilliamK | so who is everyone's favorite 1800 number provider? |
06:27.57 | outtolunc | _vile: any comments? |
06:29.13 | ZX81 | does anyone know what the snom 190 is expecting for the music on hold server? |
06:32.05 | outtolunc | anyone else want to go head to head on issues regarding SIT detecting and thier dislike of thier local provider? |
06:32.07 | *** join/#asterisk alphaque (~Alphaque@218.208.238.245) |
06:32.25 | *** join/#asterisk davegsx (fuzzy@CPE000d6d60d-CM014280007905.cpe.net.cable.rogers.com) |
06:33.53 | *** join/#asterisk serdiehard (~serdiehar@202.65.128.18) |
06:35.24 | pfn | SIT detection? |
06:36.44 | outtolunc | google is your friend http://66.102.7.104/search?q=cache:FoltOTpWaR0J:www.ahk.com/Special%2520Information%2520Tones.pdf+itu+SIT+&hl=en&start=3 |
06:37.12 | outtolunc | ever heard of 'TRI-Tone' |
06:37.32 | outtolunc | da da dee |
06:38.29 | outtolunc | we're sorry, the person you attempted to call has fled the country or is no longer available <G> |
06:39.57 | outtolunc | you can't tell me all you were born after inband detection |
06:44.08 | outtolunc | then again, i guess you can be <G> |
06:45.18 | outtolunc | i seem to forget i'm a bit of a old fart here sometimes |
06:48.21 | outtolunc | and given that i've been the only one stating anything for what 20 lines, ... nevermind i'm just talking to my damn self |
06:48.33 | outtolunc | thought so |
06:49.09 | outtolunc | going back to other channels |
06:57.20 | *** join/#asterisk scratchrf (~ryan@67-40-182-169.tukw.qwest.net) |
06:57.33 | *** join/#asterisk ard (ard@goatse.kwaak.net) |
06:58.21 | *** join/#asterisk wolfson (hehe@cpe-68-187-190-045.man.nc.charter.com) |
07:00.04 | scratchrf | if i start * from a "regular" login, then connect from my win box with putty and run asterisk -r, how do I disconnect from the -r session w/o killing * completely? |
07:00.32 | ard | ^D maybe? |
07:00.40 | ard | quit? |
07:00.57 | scratchrf | thx, i'll try and let you know |
07:01.09 | ard | to get asterisk down you'll have to type down or something like that |
07:01.12 | Slimey | quit or exit |
07:01.27 | herag | can I kill an active call/channel? |
07:02.04 | scratchrf | quit works.... |
07:05.09 | ZX81 | herag: soft hangup <tab> |
07:06.27 | herag | it's cool the zombie channel just died on its own eventually |
07:06.34 | scratchrf | if i can do everything in webvmail but save preferences, probably a cookie issue? i can see new vm's, move, etc. but w/o saving preferences I can't actually listen to them... |
07:07.13 | scratchrf | ..boots me back to "login incorrect" login screen |
07:07.37 | ZX81 | ctrl c |
07:12.10 | brc_ | ~seen jerjer |
07:12.12 | jbot | jerjer <~mine@d2-236.rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 4d 2m 18s ago, saying: 'no config necessary'. |
07:12.34 | brc_ | ~seen jerjer[mobile[ |
07:12.35 | jbot | i haven't seen 'jerjer[mobile[', brc_ |
07:12.37 | brc_ | ~seen jerjer[mobile] |
07:12.38 | jbot | jerjer[mobile] <~jj@mail.nufone.net> was last seen on IRC in channel #asterisk, 14h 46m 47s ago, saying: 'paging'. |
07:13.46 | herag | what does the ACL A flag mean when I do sip show peers? |
07:15.03 | *** join/#asterisk SuperMMan (~SuperMMan@clgrtnt5-port-123.dial.telus.net) |
07:15.38 | SuperMMan | Evening all, I was wondering if anyone knows of any companies that are offering in/out dids in Edmontion Alberta? |
07:19.14 | *** part/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net) |
07:20.27 | Himeko | contact netmonks.ca |
07:22.26 | ZX81 | anyone know why music on hold would not be able to start? with default mp3's? |
07:22.32 | SuperMMan | Himeko, ok thanx willd o |
07:25.36 | *** join/#asterisk znoG (~speedy@134-134-126-200.fibertel.com.ar) |
07:31.10 | skeeziks | Can anybody give me a hand with gdb and Asterisk? I'm having trouble getting symbol lookups to work for the various modules. |
07:32.03 | *** join/#asterisk Stephie3 (Stephie@203.81.192.172) |
07:32.14 | Stephie3 | hello |
07:32.54 | Stephie3 | anybody know about quintum A800 configuration? plz msg me |
07:38.06 | clive- | stephie do they do sip yet those quintum boxeS? |
07:43.04 | *** join/#asterisk RoyK (~roy@110.80-203-29.nextgentel.com) |
07:47.41 | *** join/#asterisk juice (~juice@mo-205-240-40-86.dyn.sprint-hsd.net) |
07:48.46 | _Vile | clive, no sip |
07:50.24 | *** join/#asterisk EisNerd (eisnerd@outpost.cyberadaptor.de) |
07:51.26 | EisNerd | is there a german channel? |
07:53.14 | florz | Dunno about any, but how about opening one if there is none? |
08:02.28 | *** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
08:04.35 | Guest^DJ | ~seen Essobi |
08:04.37 | jbot | essobi <kstone@75.137.26.216.host.teledvance.com> was last seen on IRC in channel #asterisk, 129d 1h 48m 23s ago, saying: 'Freak. ;)'. |
08:07.34 | *** join/#asterisk los415 (~los415@adsl-69-104-178-68.dsl.pltn13.pacbell.net) |
08:10.00 | SuperMMan | ~seen TestMasTer |
08:10.01 | jbot | testmaster <~testmaste@S0106000ea67573e2.cg.shawcable.net> was last seen on IRC in channel #asterisk, 102d 11h 15m 19s ago, saying: 'sivana: there is a company from here in alberta that says they can do it, but won`t give me a price lol'. |
08:10.24 | *** join/#asterisk kivi (~martin@djz.net) |
08:16.10 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
08:22.43 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
08:27.54 | *** join/#asterisk scannachiappolo (~scannachi@81-174-16-211.f5.ngi.it) |
08:32.14 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
08:32.20 | *** join/#asterisk soulz2 (~Soulz-@host-137-132-45-241.imcb.nus.edu.sg) |
08:32.57 | libpcp | hi everyone |
08:33.12 | libpcp | anyone has an asterisk write-ups? where can i get it? |
08:33.59 | *** join/#asterisk ed0wn (ed0wn@dontvisit.us) |
08:34.18 | *** join/#asterisk Michiel1983 (~Michiel19@81-17-62-133.dsl.uwadslprovider.nl) |
08:34.48 | ed0wn | I was wondering with asterisk...how does camophone.com use no special hardware? |
08:35.24 | Dr_Ray | click to call |
08:35.52 | Dr_Ray | it calls you when it makes the bridge |
08:35.53 | tafazzi | Hi all. I would like my asterisk to find a free extension before answering the incoming call, to avoid having the caller paying the call to hear ringing the default extensions. Anybody has a dialplan suggestion for me? |
08:36.44 | libpcp | anyone has an asterisk write-ups? where can i get it? |
08:40.09 | RoyK | anyone that have used the queue_log here? |
08:40.30 | *** join/#asterisk skeeziks (~skeeziks@r80h128.res.gatech.edu) |
08:41.06 | skeeziks | Does anybody have oej's version of the chan_sip2.c patch that integrates the BroadVoice patch? |
08:42.16 | skeeziks | The latest bugnotes says that he integrated the BV patch, but there's not a new attachment yet |
08:42.47 | skeeziks | I'm having trouble with call progress on a SIP channel once it's been answered, so I thought I'd give his patch a atry |
08:42.51 | Michiel1983 | Hello all. I get a segmentation fault when I make a dail from my SIP softphone. I've a Winbond ISDN card with i4l driver the error messages are after a few seconds when I make a call "Executing Dial("SIP/msn2-fed4", "Modem/ttyI1:...") in new stack" "Segmentation fault (core dumped)" Did anyone know what the problem is or where I can find information? |
08:46.52 | *** join/#asterisk monst3r (~monst3r@adsl-59-29.swiftdsl.com.au) |
08:47.40 | *** join/#asterisk CleanerX (AC142907@nat-ph3-wh.rz.uni-karlsruhe.de) |
08:47.55 | skeeziks | RoyK: Do you know of any sources of more info about chan_sip2? |
08:48.57 | *** join/#asterisk Pryk (~tmalkut@host-ip226-209.crowley.pl) |
08:50.01 | pfn | cvs |
08:51.20 | RoyK | pfn, I don't think sip2 is in cvs |
08:51.32 | RoyK | pfn, stuff from sip2 is being backported to cvs, that's all |
08:51.40 | skeeziks | Yeah, it's not there |
08:52.02 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
08:52.26 | skeeziks | Have you guys heard of any problems with call progress once a SIP channel has been answered? |
08:52.33 | pfn | chan_sip2 is in cvs |
08:52.36 | skeeziks | I'm looking for a bug report but can't find anything |
08:52.36 | pfn | search for it |
08:52.41 | skeeziks | OK, looking |
08:52.48 | pfn | there is no call progress once a channel has been answered |
08:52.58 | *** join/#asterisk zoa (~zoa@213.16.46.130) |
08:53.12 | pfn | 'cept hanging it up, maybe |
08:53.21 | pfn | or inband |
08:54.05 | skeeziks | Well, I answer a SIP call and prompt the user for a number, then dial out again on SIP, and I get no ringing or busy or anything, and this error: |
08:54.32 | skeeziks | Nov 16 03:28:44 WARNING[14215]: channel.c:1451 ast_indicate: Unable to handle indication 3 for 'SIP/6785550063-7fe9' |
08:55.28 | skeeziks | And I don't see sip2 anywhere in CVS - where did you find it? |
08:56.07 | pfn | chan_sip2 |
08:56.16 | skeeziks | Oh, it's a module... |
08:56.16 | skeeziks | OK |
08:56.30 | pfn | oops, doid I say cvs |
08:56.38 | zoa | its on mantis |
08:56.40 | pfn | I meant mantis |
08:56.44 | zoa | bugs.digium.com |
08:57.41 | skeeziks | Yeah, I found it there, but oej mentions that he's made some changes that I want since the last version was posted - I guess I'll have to wait |
08:57.50 | ed0wn | softphone isnt working to test my damn asterisk setup |
08:57.53 | ed0wn | i wonder what the hell i did wrong |
08:57.55 | monst3r | how are we all this evening |
08:58.19 | skeeziks | pfn: Any idea on that warning? If I dial out a zap channel I hear ringing and stuff properly... |
09:01.09 | pfn | the ringing should happen on the other channel, then |
09:01.13 | pfn | and you should get ringing just fine |
09:01.20 | pfn | figure out what indication 3 is and why you can't play it |
09:01.41 | skeeziks | I've been trying to trace things through GDB but I'm having trouble getting all the symbols available |
09:02.42 | skeeziks | indication 3 is AST_STATE_DIALING |
09:03.06 | pfn | ast_control_ringing |
09:03.53 | pfn | asterisk can't find the ringing tone for the zone that the channel belongs to |
09:04.36 | skeeziks | OK |
09:04.42 | skeeziks | So what should I do? |
09:05.39 | pfn | I dunno, figure out your indication situation |
09:07.35 | skeeziks | But you think the zone thing is the underlying issue? |
09:10.15 | *** join/#asterisk darkskiez (~mbryars@usergc137.dsl.pipex.com) |
09:11.06 | darkskiez | <PROTECTED> |
09:12.45 | postel | darkskiez: heh, funky idea |
09:13.07 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
09:14.01 | darkskiez | Yay, there is even Hal9000: (From 2010)There is a message for you |
09:14.07 | monst3r | is there away to make asterisk if you dial any number thats not part of the dial plan goto a default context, i thought using the bogon-calls context did this.. have i missed something? |
09:14.21 | funkknob | Calling voicemail, is this a codec issue? Nov 16 16:06:29 WARNING[1168520128]: file.c:550 ast_readaudio_callback: Failed to write frame |
09:14.40 | darkskiez | monst3r, with the 'i' extension? |
09:15.19 | monst3r | hmmm ok im not using that so using the i, and tell it where to go will do that? |
09:18.14 | *** join/#asterisk monkey- (user3@port-219-88-128-149.orcon.net.nz) |
09:31.52 | darkskiez | DISA times out far too quickly, the dialtone lasts for a couple of seconds and then dies. |
09:32.39 | *** part/#asterisk monkey- (user3@port-219-88-128-149.orcon.net.nz) |
09:33.19 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
09:34.26 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
09:44.21 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
09:45.41 | Michiel1983 | Hello all. I get a segmentation fault when I make a dail from my SIP softphone. I've a Winbond ISDN card with i4l driver the error messages are after a few seconds when I make a call "Executing Dial("SIP/msn2-fed4", "Modem/ttyI1:...") in new stack" "Segmentation fault (core dumped)" Did anyone know what the problem is or where I can find information? |
09:48.28 | funkknob | Hi Michiel1983 - sorry I don't know |
09:48.37 | funkknob | Seems everyone's gone to sleep |
09:51.54 | funkknob | Does anyone know the syntax for all the codecs in sip.conf and h323.conf? "show codecs" in the CLI does not - i.e. g.711a syntax is "allow=alaw" |
09:53.22 | *** join/#asterisk jaXon` (~jaxon@ip116.66.1311D-CUD12K-03.ish.de) |
09:54.41 | funkknob | Hi jaXon |
09:54.50 | jaXon` | hi |
09:55.12 | funkknob | Do you know the allow= for g.729a in h323.conf? |
09:55.52 | funkknob | I keep getting: |
09:55.57 | funkknob | Nov 16 16:46:37 WARNING[1076298368]: Cannot allow unknown format 'g.729a' |
09:56.47 | jaXon` | what about allow=g729 ? |
09:56.57 | funkknob | thx I'll try |
09:57.55 | funkknob | Hey that works! thx |
09:58.32 | jaXon` | np |
09:59.21 | funkknob | The * process started with no errors, but the h323 segment is still using g.711u :( |
10:00.03 | RoyK | prolly cause it's h323 :P |
10:01.24 | funkknob | I'm using * as a protocol contverter, SIP phone ------ Proxy --------- * -------- AS5350 (H.323) |
10:05.30 | *** join/#asterisk shadebob (~shadebob@ll81-134-144-192-81.ll81.iam.net.ma) |
10:21.50 | Michiel1983 | Hello all, I want to call with a SIP softphone to an extern number with an ISDN card my dial command in extensions.conf is: exten => 21,1,Dial(Modem/ttyI0/69:00123456789). When I make a call I get: "Executing Dial("SIP/msn2-e371", "Modem/ttyI0/69:00123456789") in new stack" - "WARNING[1144486832]: chan_modem.c:828 modem_request: Requested device 'ttyI0/69' does not exist" - " NOTICE[1144486832]: app_dial.c:743 dial_exec: Unable |
10:23.00 | skeeziks | pfn: ARGH! I was missing indications.conf and res_indications.so. |
10:23.06 | skeeziks | pfn: Thanks for your help. |
10:33.30 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
10:41.36 | RoyK | hi |
10:47.29 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
10:48.30 | ZX81 | how were you missing those files?! |
10:48.49 | ZX81 | :-) |
10:49.21 | RoyK | lol |
10:51.27 | RoyK | does anyone know if asterisk supports T.38? |
10:55.02 | *** join/#asterisk Mike_tk (~Mike_@213.180.245.62) |
11:00.43 | [Sim] | royk: it does not support t38 yet |
11:01.10 | *** part/#asterisk PCadach (~paul@www.east.telecom.kz) |
11:01.43 | RoyK | [Sim], any idea how much it'll take to make it? |
11:02.15 | [Sim] | sorry, no idea |
11:06.14 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
11:06.39 | *** join/#asterisk Jayjay (~Jasonrive@host213-120-116-134.in-addr.btopenworld.com) |
11:06.45 | Jayjay | hi peeps |
11:07.08 | Jayjay | anyone awake? |
11:07.36 | RoyK | nah |
11:07.37 | RoyK | sleeping |
11:07.40 | Jayjay | l0l |
11:08.05 | ZX81 | playing counterstrike |
11:08.10 | Jayjay | i'm on Red Hat 9 (don't ask why yet) and i'm having some problems installing Asterisk.... |
11:08.20 | RoyK | SIP:0xc0ffee |
11:08.30 | Jayjay | ...while i make Zaptel, i get "zaptel.c:6213: variable `zt_fops' has initializer but incomplete type" |
11:08.39 | RoyK | Jayjay, redhat kernel or stock kernel? |
11:08.58 | RoyK | Thou Shalt Not Use Distribution Specific Kernels With Zaptel |
11:09.19 | darkskiez | And Jesus cried. |
11:09.39 | RoyK | isn't he dead? |
11:09.47 | darkskiez | s/Jesus/Jayjay/ |
11:09.47 | *** join/#asterisk oller (oller@nemec.strul.net) |
11:09.52 | RoyK | :) |
11:09.58 | Jayjay | RoyK, i'm not entirely sure, its been on this machine for err... a couple of years, stuffed on a shelf |
11:09.59 | ZX81 | jayjay is dead? |
11:10.04 | ZX81 | nope |
11:10.06 | ZX81 | :-) |
11:10.21 | Jayjay | i can only assume RedHat Kernel |
11:10.28 | RoyK | Jayjay, uname -r |
11:10.42 | ZX81 | 0.0.0.1a |
11:10.50 | RoyK | rotflmao |
11:10.55 | Jayjay | just says 2.4.20-8 |
11:11.03 | ZX81 | cant be that old |
11:11.06 | RoyK | that's the redhat standard |
11:11.10 | RoyK | redhat 9 standard |
11:11.14 | Jayjay | yup |
11:11.16 | RoyK | ...and it REALLY REALLY SUCKS |
11:11.23 | Jayjay | i know, its not my machine |
11:11.34 | Jayjay | if i had my way, it would be debian or gentoo |
11:11.38 | RoyK | upgrade to 2.4.27 or 2.6.9 from kernel.org |
11:11.51 | Jayjay | but i'm under orders of *they take too long to install so we're sticking with this for now* |
11:11.59 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
11:12.19 | Jayjay | ok |
11:12.25 | RoyK | Jayjay, then tell Them to Stick The RedHat CD Up Their Behind |
11:12.31 | Jayjay | l0l |
11:12.35 | Jayjay | its my old mans ;) |
11:12.42 | Jayjay | might not go down too well |
11:13.09 | Jayjay | what am i going for 2.4 or 2.6 |
11:13.10 | Jayjay | ? |
11:13.10 | RoyK | well. then gently tell them that Asterisk Can Not Be Installed In A Hurry On The RedHat Standard Kernel |
11:13.15 | RoyK | whatever |
11:13.23 | RoyK | 2.4 will be smoothest |
11:13.29 | RoyK | 2.6 scales better, is better etc |
11:13.38 | RoyK | but will require some other updates |
11:16.22 | Jayjay | do we know what updates? |
11:18.02 | RoyK | read Documentation/Changes in the kernel source tree |
11:18.14 | RoyK | that gives a brief list |
11:18.28 | RoyK | and a complete list... |
11:19.40 | Dr_Ray | is there a problem with fedora core 3 and asterisk? kernel 2.6? I just installed it yesterday |
11:19.54 | *** join/#asterisk speedwagon (~Ariel@fl-nked-ubr2-c6a-125.miamfl.adelphia.net) |
11:25.27 | Michiel1983 | Hello all, I want to call with a SIP softphone to an extern number with an ISDN card my dial command in extensions.conf is: exten => 21,1,Dial(Modem/ttyI0/69:00123456789). When I make a call I get: "Executing Dial("SIP/msn2-e371", "Modem/ttyI0/69:00123456789") in new stack" - "WARNING[1144486832]: chan_modem.c:828 modem_request: Requested device 'ttyI0/69' does not exist" - " NOTICE[1144486832]: app_dial.c:743 dial_exec: Unable |
11:33.42 | PoWeRKiLL | kram are you here ? |
11:33.57 | *** join/#asterisk zyke (~xirak@host-212-158-223-3.bulldogdsl.com) |
11:35.06 | ionix | anyone knows how to add a "*" before an extension ? Trying to parse the 1-800 numbers |
11:35.30 | brc_ | . |
11:35.40 | ionix | i.e if asterisk receives _1800xxxxxxx, I want it to send *18001234567 to SIP/fwd |
11:35.43 | brc_ | . |
11:35.53 | ionix | I don't understand brc |
11:35.57 | brc_ | no you don't |
11:36.11 | ionix | I have |
11:36.12 | ionix | exten => _1800.,1,Dial(SIP/#*${EXTEN}@fwdmax) |
11:36.12 | ionix | exten => _1800.,2,Macro(fastbusy) |
11:36.12 | brc_ | why would you want to send a asterisk? |
11:36.21 | ionix | because fwd needs an asterisk |
11:36.33 | brc_ | k |
11:40.05 | Jas_Williams | ionix: change your dial string to exten => _1800.,1,Dial(SIP/*${EXTEN}@fwdmax |
11:40.45 | brc_ | buzz |
11:41.10 | ionix | ok thx |
11:41.31 | ionix | I tought # was required. Thx |
11:42.00 | ionix | btw, any SIP provider that offers long distance without subscription ? Like I can fuel a card 10$ and make outgoing call to them without having a phone number ? |
11:42.26 | ionix | I would like my gateway to use my phone number for local call and that sip provider for long distance. + use 1800 with fwd |
11:42.43 | ionix | i.e plenty of rules to save money (the real reason is to have fun but shhht) |
11:42.45 | brc_ | ~nufone |
11:42.45 | jbot | hmm... nufone is Visit http://www.nufone.net for an excellent, native IAX termination service. |
11:42.50 | brc_ | 2.0 |
11:42.59 | ionix | I'll look into that |
11:47.37 | Makenshi | ~sipgate |
11:47.49 | Makenshi | oh well :> |
11:48.53 | ZX81 | ~fish |
11:48.54 | jbot | hmm... fish is FISHFISHFISH! DO THE FISH DANCE! "Give a man a fish and you'll feed him a day. Teach him how to fish and he'll feed himself for the rest of his life." This is so appropriate, instead of asking us to tell you exactly what to do, why not read some docs, then come back and ask specific questions which aren't covered?, or ... |
11:48.56 | ionix | I signed up for nufone.net seems nice the pay as you go |
11:49.06 | ZX81 | lol |
11:49.23 | Jas_Williams | ~docs |
11:49.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
11:49.38 | Jas_Williams | ~sipgate |
11:50.09 | ZX81 | ~cow |
11:50.11 | jbot | I am a cow, hear me moo. I eat grass and weigh twice as much as you. |
11:50.15 | ZX81 | lol |
11:50.34 | Jas_Williams | ~sipgate is http://www.sipgate.de |
11:50.35 | jbot | Jas_Williams: okay |
11:50.40 | Jas_Williams | ~sipgate |
11:50.41 | jbot | [sipgate] http://www.sipgate.de |
11:50.43 | ZX81 | ~pig |
11:50.44 | jbot | *oink* *oink* *oink* |
11:50.48 | Jas_Williams | better |
11:50.55 | Jas_Williams | ~lart ZX81 |
11:51.00 | ZX81 | lol |
11:51.10 | ZX81 | ~thwap Jas_Willia |
11:51.12 | jbot | ACTION bends over Jas_Willia and grins happily |
11:51.20 | ZX81 | y:-) |
11:51.31 | Jas_Williams | tooche' |
11:52.56 | ZX81 | ~daily asterisk news |
11:53.25 | ZX81 | ~daily asterisk news is http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss |
11:53.26 | jbot | okay, ZX81 |
11:53.28 | ZX81 | ~daily asterisk news |
11:53.29 | jbot | daily asterisk news is probably http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss |
11:53.44 | ZX81 | ~news |
11:53.54 | darkskiez | does anyone know where the asterisk news site is? |
11:54.02 | ZX81 | ~daily asterisk news |
11:54.03 | jbot | somebody said daily asterisk news was http://www.sineapps.com/news.php for html and http://www.sineapps.com/rssfeed.php for rss |
11:54.04 | ZX81 | :-) |
11:54.06 | brc_ | are you dense? or do you just pretend to be |
11:54.23 | zoa | hello there |
11:54.26 | brc_ | damn |
11:54.32 | ZX81 | where's there? |
11:54.41 | darkskiez | brc_, sorry, easily amused. |
11:54.45 | ZX81 | :-) |
11:55.06 | brc_ | ~asterisk daily news is <reply> see daily asterisk news |
11:55.07 | jbot | brc_: okay |
11:55.14 | brc_ | ~adn is <reply> see daily asterisk news |
11:55.15 | jbot | ...but adn is already something else... |
11:55.19 | brc_ | ~adn |
11:55.20 | jbot | methinks adn is AcidoDexosirriboNucleico |
11:55.25 | brc_ | ~no adn is <reply> see daily asterisk news |
11:55.26 | jbot | okay, brc_ |
11:55.35 | brc_ | ~news is <reply> see asterisk daily news |
11:55.45 | brc_ | ~seen Makenshi |
11:55.47 | jbot | makenshi is currently on #asterisk (2h 42m 40s). Has said a total of 2 messages. Is idling for 7m 58s |
11:56.39 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
11:57.00 | ZX81 | :-) |
11:57.08 | ZX81 | i died |
11:57.15 | ZX81 | headshot while jumping |
11:57.17 | ZX81 | sucky |
11:58.00 | brc_ | halo2? |
11:58.11 | ZX81 | :-) |
11:58.13 | ZX81 | nah |
11:58.17 | ZX81 | still stuck in CS |
11:58.20 | ZX81 | :-) |
11:58.24 | brc_ | CS |
11:58.26 | brc_ | HA |
11:58.32 | ZX81 | lol sad a |
11:58.57 | ionix | Silly question but can I use my nufone account to place 2 simultanous call ? |
11:59.06 | ionix | i.e I have 2 lines on my spa-2000... |
11:59.14 | ionix | Or do I need to register 2 accounts |
12:00.32 | PatrickDK | twoaccounts |
12:01.17 | ZX81 | dunno, voicepulse i was doing 4 |
12:01.18 | brc_ | PatrickDK, su |
12:01.35 | brc_ | ionix, of course you don't need two accounts with nufone |
12:01.43 | PatrickDK | oh heh |
12:01.46 | RoyK | sudo rm -rf / |
12:01.49 | PatrickDK | no, one nufone account |
12:02.01 | brc_ | touch rm\ \-rf |
12:02.02 | PatrickDK | I didn't see the first part of the question |
12:02.35 | brc_ | riiiiiiiiiight |
12:02.58 | ionix | ok thx :) |
12:06.21 | darkskiez | <PROTECTED> |
12:09.15 | ionix | Also, how will I know if there are insuffisent funds in the nufone account ? I mean if balance is insuffisent, how can I notify the caller on my pbx that long distance calls cannot be dialed at the moment ? |
12:09.47 | RoyK | mm |
12:10.12 | RoyK | is it possible to turn off echo cancellation for certain calls? |
12:10.16 | RoyK | on zap? |
12:10.54 | *** join/#asterisk SuB_X (Sahm_X@adsl-135-24-192-81.adsl.iam.net.ma) |
12:10.55 | ZX81 | you can have it on or off per line i think, but not dynamicly |
12:11.09 | PatrickDK | na, it turns it off for fax's I know |
12:11.36 | ZX81 | k |
12:11.42 | ZX81 | so then must be possible |
12:11.53 | PatrickDK | ya, just don't know if there is a command to do it |
12:12.00 | SuB_X | hi, plz i'm searching a frenchspeaking webdevelopper, |
12:12.03 | PatrickDK | or if it's hardcoded into the fax detection stuff |
12:12.03 | ZX81 | grep source |
12:12.11 | ZX81 | my wife is french |
12:12.19 | SuB_X | hi, plz i'm searching a frenchspeaking webdevelopper,, could you help me |
12:12.19 | ZX81 | from lyon |
12:12.20 | SuB_X | ? |
12:12.23 | FuzzyCat | SuB_X: rentacoder |
12:12.39 | ZX81 | i am a developer and my wife is french, i don't speak a word of it tho |
12:12.41 | ZX81 | :-) |
12:12.41 | SuB_X | rentacoder ? |
12:12.48 | SuB_X | lol |
12:13.02 | FuzzyCat | www.rentacoder.com |
12:13.03 | SuB_X | could you help |
12:13.04 | SuB_X | .? |
12:13.39 | RoyK | PatrickDK, how can I signal that THIS IS FAX? |
12:13.46 | RoyK | ...if it doesn't detect it, I mean |
12:13.57 | PatrickDK | you don't |
12:14.01 | RoyK | well... |
12:14.13 | RoyK | but is fax detection supposed to really work? |
12:14.15 | ZX81 | put faxdetect=both in zapata.conf |
12:14.22 | ZX81 | or incoming |
12:14.24 | ZX81 | or whatever |
12:14.35 | RoyK | problem is outgoing faxes... |
12:14.48 | RoyK | are there any echo cancellation in chan_sip? |
12:14.56 | PatrickDK | no |
12:14.58 | ZX81 | nope |
12:15.09 | PatrickDK | what are you using to connect the fax to asterisk? |
12:15.21 | ZX81 | ygsm codec |
12:15.23 | ZX81 | :-) |
12:15.45 | RoyK | fax -> SIP ATA -> SIP/IAX2 gw (*) -> IAX2/PSTN gw (*) -> ISDN PRI |
12:15.57 | RoyK | using ALAW |
12:16.37 | PatrickDK | hmm, the problem is here is no fax detect logic on the sip side |
12:17.06 | PatrickDK | inside of asterisk |
12:17.55 | ZX81 | what defines the time a box re-registers at? like some of my peer servers are around every 10 mins, some every hour or so |
12:18.12 | ZX81 | all the same in iax.conf |
12:18.20 | PatrickDK | it depends on the peer |
12:18.30 | ZX81 | what do they set? |
12:18.38 | PatrickDK | anything they want |
12:18.46 | ZX81 | i.e. what variable |
12:18.50 | ZX81 | xxx=xxx |
12:19.04 | PatrickDK | dunno, it's different |
12:19.08 | ZX81 | k |
12:19.23 | PatrickDK | I mean, it would be different on a supira200, a snom phone, and a cisco 7960 |
12:19.32 | ZX81 | no, all iax |
12:19.32 | PatrickDK | you can't control it in iax.conf |
12:19.44 | ZX81 | intracompany boxes |
12:19.48 | PatrickDK | you can only control how often you register to someone else |
12:19.50 | Makenshi | brc_, had you confused, my bad |
12:19.52 | ZX81 | hmmm...strange |
12:20.30 | PatrickDK | you can control registeration timeouts |
12:20.39 | ZX81 | ah maybe it is not registering the other box becuase of qualify |
12:20.46 | ZX81 | so it already knows |
12:20.48 | tzanger | any recommended places to purchase Polycom IP300 phones? Looking for about a dozen of 'em |
12:25.51 | ionix | SuB_X: I am french heh |
12:26.40 | ionix | tzanger: check ebay or www.froogle.com |