| 00:00.37 | Igramul | Hi, I am new to Asterisk. Is it possible to use an "Intel Corporation 536EP Data Fax Modem" as channel for Asterisk to transfer speech? |
| 00:02.40 | joako | Igramul, have you installed it? What does lspci say? |
| 00:03.30 | joako | The ones I have lspci says: 01:09.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 00:03.59 | Igramul | joako, 00:0a.0 Communication controller: Intel Corporation 536EP Data Fax Modem |
| 00:05.12 | joako | Igramul, probably not |
| 00:06.00 | Igramul | no big deal - I guess I will buy an ISDN card. |
| 00:06.30 | joako | I have one here I am 99% sure is one that works (why else would I keep a 56k modem?) the chip says "Ambient MD3200" |
| 00:06.34 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 00:06.49 | Igramul | However, I have no idea which one. (home office, two D-channels, point-to-multipoint, Europe) |
| 00:07.14 | Igramul | joako, do You use it as analog (Modem) device or as ISDN interface? |
| 00:07.31 | joako | Igramul, What exactly are you tring to do? A card for an analog line will not work for a digital line and vice-versa |
| 00:07.50 | joako | It says ISDN and openSUSE even tries to set it up as an ISDN device... but it's not |
| 00:07.57 | Igramul | joako, the type of line is not determined yet. |
| 00:08.09 | Igramul | But I guess, I will get myself an ISDN line. |
| 00:08.28 | Igramul | s/determined/decided |
| 00:09.36 | joako | Igramul, If you get an ISDN line you will need an ISDN card that is compatible with Asterisk |
| 00:09.47 | joako | You are in Germany? |
| 00:09.52 | Igramul | yes |
| 00:10.10 | joako | It should be well documented what works and how to get it working |
| 00:10.18 | Igramul | ok |
| 00:11.34 | *** join/#asterisk makafre (n=makafre@206-248-162-226.dsl.teksavvy.com) |
| 00:13.21 | makafre | hi guys, how would we tell asterisk to reload its realtime cache for all peers and users whenever the database changed? I tried sip prune realtime all but it seems it only reloads a few, not all of them. |
| 00:15.37 | makafre | the other option I have would be to go through the table and perform a sip prune realtime peer xxx on all of them |
| 00:16.10 | voipmonk | why |
| 00:16.14 | voipmonk | u dont need to do that |
| 00:16.21 | makafre | ok |
| 00:16.21 | voipmonk | all u need to do is update the db |
| 00:16.31 | voipmonk | stop fscking with other options |
| 00:16.34 | voipmonk | and make the update |
| 00:16.35 | voipmonk | and test |
| 00:16.46 | voipmonk | as if you performed a reload - watch what happens |
| 00:16.48 | voipmonk | :) |
| 00:16.59 | makafre | :- ) well it doesn't work whenever you have the cache enabled |
| 00:17.16 | makafre | rtcachefriends=yes |
| 00:21.49 | Gugge | you could enable the timeout option |
| 00:22.09 | makafre | timeout? |
| 00:23.18 | Gugge | rtautoclear |
| 00:24.02 | makafre | ah, let me check |
| 00:24.24 | Gugge | but it wont clear it unless the peer is away for xx time |
| 00:24.46 | Gugge | why do you need rtcachefriends ? |
| 00:25.38 | makafre | good question, the system was setup in 2006, and I can't recall the exact reason (..) |
| 00:27.12 | makafre | maybe for WMI or qualify |
| 00:28.36 | Gugge | if you use those, yes |
| 00:30.38 | makafre | apparently the default is yes (120 secs) |
| 00:37.51 | makafre | Gugge: what do you mean exactly by "away for xx time" |
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| 01:33.18 | chuji | any chance someone here has some mitel firmware? |
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| 01:56.45 | Katty | hi. |
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| 02:04.05 | Katty | hi? |
| 02:04.09 | Katty | where is everyone. |
| 02:04.13 | Katty | pokes about |
| 02:05.51 | jaytee | ow! |
| 02:07.05 | Katty | :> |
| 02:07.13 | Katty | howdy |
| 02:07.33 | GreGrenada | I'm here! If that's any good to you! :-) |
| 02:07.45 | Katty | (= |
| 02:08.30 | jaytee | I'm watchin Doctor Who |
| 02:09.54 | Katty | i'm watching mint julep masque reviews. |
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| 02:12.12 | ChannelZ | ugh. headache. |
| 02:12.19 | Katty | gives ChannelZ midol. |
| 02:13.30 | GreGrenada | I'm watching this screen! |
| 02:13.40 | Katty | you don't have two displays? |
| 02:14.17 | GreGrenada | How do I get it to automagically scrool up? I've spent hours watching it NOT MOVE when people post! |
| 02:15.14 | ChannelZ | just keep sitting there thumbing the wheel on your mouse down |
| 02:15.54 | *** join/#asterisk troy- (n=troy@CPE00907f17e478-CM00186845db94.cpe.net.cable.rogers.com) |
| 02:16.12 | GreGrenada | No, I don't have two displays, unless you count my PowerbookG4, a separate computer on my LAN! |
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| 02:16.56 | GreGrenada | How do I get it to automagically scrool up? Even when I post a msg, I have to manually scroll it to see it! :-( |
| 02:17.06 | ChannelZ | what junk IRC client are you using? |
| 02:18.13 | GreGrenada | qwebirc, got any better ideas??? |
| 02:19.13 | GreGrenada | Seems to be auto-scrolling now, hope I've not spoken too soon...... |
| 02:19.55 | GreGrenada | So, do I have to go drink 7 beers EVERY time I want it to work "properly"? :-) |
| 02:20.44 | GreGrenada | Oh crapp, not scrolling automatically any more :-( What was the name of that k9g0dad5 IRCclient??? :) |
| 02:21.12 | GreGrenada | I ment K9G0NAD5 |
| 02:21.46 | troy- | is it possible using Sangoma U100 usbfxo to ring an exten and answer the call only of the exten is answered? |
| 02:22.06 | troy- | otherwise just let the line continue to ring (do nothing) |
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| 02:41.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:44.47 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
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| 03:14.13 | *** join/#asterisk littleball (n=littleba@cm38.zeta226.maxonline.com.sg) |
| 03:14.18 | littleball | hello |
| 03:14.30 | littleball | what endianness does asterisk use? big endian or little endian? |
| 03:14.54 | GreGrenada | Depends where you compile the source code! :-) |
| 03:15.25 | littleball | when one client call other voip client |
| 03:15.40 | littleball | how is the endiannss decided? |
| 03:16.08 | GreGrenada | auto |
| 03:16.09 | littleball | I cannot find any negotiation of endianness between voip clients |
| 03:16.16 | GreGrenada | none exists! |
| 03:16.16 | littleball | what do you mean auto? |
| 03:17.01 | GreGrenada | endianness = internal! |
| 03:17.48 | littleball | I think there should be some agreement . otherwise, how can remote end unerstand my data? |
| 03:18.11 | GreGrenada | If I set up my 2 machines, 1 = PC, little, 1 = powerPC, Big, no difference at application level, comms is not endian! |
| 03:18.30 | GreGrenada | geddit? |
| 03:19.20 | littleball | For example, PCM data (16 bits signed), |
| 03:20.08 | littleball | when such 16 bits signed data received by remote end, the remote end need to get back the original 16 bits signed data. Endianness agreement must be somewhere |
| 03:20.11 | GreGrenada | Well, if the data itself has endianness, then you just have to convert, surely no biggie!?! |
| 03:20.54 | littleball | "biggie"? |
| 03:21.10 | GreGrenada | "biggie" = big deal |
| 03:21.52 | littleball | for one asterisk installation, if it want to connect to other asterisk servers, then this must be a problem, |
| 03:22.41 | GreGrenada | how so? IAX2 comms transfers data no problem, as far as I know (AFAIK) it's never a problem.... Unless you know different!?! |
| 03:23.35 | littleball | I am coding my own IAX client |
| 03:23.39 | littleball | which support iLBC |
| 03:24.16 | littleball | I call from my PC software to mobile (PSTN termination), I can hear the voice very clear from PC software |
| 03:24.18 | GreGrenada | So native big data on big system is OK, but when transferred to little system becomes a problem? |
| 03:24.25 | littleball | but my mobile is not clear |
| 03:24.51 | littleball | I suspect that it is because of my encoding on PC side |
| 03:25.14 | littleball | pc encoding (iLBC encoding) to asterisk --> PSTN. |
| 03:26.19 | littleball | because iLBC involves some 16 bits short integer to bytes |
| 03:26.21 | littleball | conversion |
| 03:26.30 | littleball | so, endianness is a problem |
| 03:26.32 | GreGrenada | have you attempted changing endianness of sound file (recorded clip) on PC to BIG, send and see if received OK ? |
| 03:26.45 | littleball | I am using macbook |
| 03:26.57 | littleball | let me try big endian |
| 03:27.09 | GreGrenada | yes |
| 03:28.08 | littleball | no need, I have conversion from sampling endianness to whatever I need to be sent to network module |
| 03:28.31 | littleball | anyway, let me try. |
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| 03:28.35 | GreGrenada | 'k |
| 03:31.21 | littleball | the same |
| 03:34.15 | GreGrenada | So, are you telling me that no matter whether you send big or little-endian data, it still comes out "wrong"? |
| 03:35.07 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 03:36.31 | littleball | no exactly. hard to explain. The voice quality is not good even I can hear. But the decoding side is perfect and voice is very clear |
| 03:36.49 | littleball | that is to say, mobile ---> PC is good |
| 03:37.14 | GreGrenada | so problem = ? |
| 03:38.23 | littleball | <PROTECTED> |
| 03:39.27 | GreGrenada | middle system? first I've heard! I thought you were transferring straight from MAC(big-endian) to PC (little-endian) |
| 03:40.03 | littleball | pc --> asterisk -->pstn gateway -->mobile |
| 03:40.10 | littleball | no idea where is the problem |
| 03:40.33 | littleball | I am testing mac to mobile. Becasue it is easy for me to debug |
| 03:41.02 | littleball | actualyly mac is little endian (intel core 2) |
| 03:41.16 | GreGrenada | 'k |
| 03:41.25 | littleball | never mind. Let me test on windows |
| 03:41.29 | GreGrenada | 'k |
| 03:41.55 | GreGrenada | process of elimination, find out what it is NOT, all that's left is what IS! :) |
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| 04:11.25 | GreGrenada | sleep 28,800 |
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| 06:03.09 | NateHB | hey guys, im trying to set up some routing, could use a hand... |
| 06:03.20 | p3nguin | What's the issue? |
| 06:03.37 | p3nguin | Without a specific question, there will be no answer. |
| 06:04.50 | ChannelZ | wax on, wax off |
| 06:05.03 | NateHB | I've got three nics, eth0 has a static public ip, eth1 is static 192.168.10.20 (managment interface), and eth2.1000 is vlan 1000 192.168.11.1 my phone vlan |
| 06:05.54 | NateHB | I cant seem to figure out what ive done wrong, I can seem at access any of the phones on the 192.168.11.x VLAN |
| 06:06.24 | p3nguin | Access them from where and in what manner? |
| 06:06.35 | NateHB | no route to host when I ping em from the asterisk box, or from my desktop on the 192.168.10.x network |
| 06:06.59 | ChannelZ | what does 'route' say? |
| 06:07.00 | p3nguin | Pastebin the output of "netstat -nr" |
| 06:07.03 | NateHB | right now im trying to get a simple ping back from a phone at 192.168.11.199 |
| 06:09.48 | NateHB | http://pastebin.com/d74977506 |
| 06:11.01 | ChannelZ | hmm |
| 06:11.32 | p3nguin | It doesn't look like the routing table is the problem. Each subnet has a valid device associated with it. |
| 06:11.32 | ChannelZ | nothing seems out of place. Are you running iptables? |
| 06:11.49 | p3nguin | pastebin "iptables -L -nv" |
| 06:14.18 | NateHB | ifcfg-eth2 and 2:1000 scripts - http://pastebin.com/d708224ac |
| 06:14.23 | NateHB | iptables - http://pastebin.com/d2fab7863 |
| 06:15.29 | p3nguin | What an odd rule to have. |
| 06:15.32 | ChannelZ | hmm maybe the phones don't ping?? :) |
| 06:15.41 | NateHB | maybe my network script for the VLAN is setup wrong |
| 06:15.59 | p3nguin | Can you ping the addr on each interface? |
| 06:16.30 | NateHB | yes, i can ping them from my 10.x and from the asterisk box |
| 06:17.13 | ChannelZ | why are you doing this this way by the way? |
| 06:20.32 | ChannelZ | (the eth2:1000 alias I mean) |
| 06:20.32 | p3nguin | What is the value in /proc/sys/net/ipv4/ip_forward ? |
| 06:21.12 | NateHB | we have one drop per workstation, each drop goes to the phone and the workstation plugs into the phone, I dont want the workstations to be able to directly talk to the phones, or see sip traffic, and the asterisk server has a seperate ip than the rest of the network, and the QOS controls give precedence to sip, rtp, and the router interface that eth1 is connected to |
| 06:22.23 | NateHB | ChannelZ, that was for you but I hadnt seen your clarification, to answer, I really don't know alot about settign up linux interfaces to work with VLANs |
| 06:22.36 | NateHB | answer: I dunno |
| 06:22.44 | ChannelZ | well you said all 3 are separate physical interfaces.. eth0 eth1 and eth2 |
| 06:22.57 | ChannelZ | this is the * server I'm assuming |
| 06:23.37 | NateHB | and I've had like 344mg x 2 of caffiene |
| 06:23.48 | ChannelZ | are you saying that eth1 and eth2 really aren't separate, they're on the same physical network (the same wire)? |
| 06:26.24 | NateHB | eth0 and eth2 are connected to the same switch, eth1 bypasses the switch entirely and plugs into the router on the same VLAN as the WAN |
| 06:26.54 | NateHB | by the way, dd-wrt is f*ing great |
| 06:26.58 | ChannelZ | well you have a loop then |
| 06:27.29 | NateHB | where? im not seeing it |
| 06:27.33 | ChannelZ | you have 2 interfaces in one computer hooked up to the same physical network. It's no wonder your routing doesn't work, ARP has no clue what is going on |
| 06:27.51 | NateHB | eth0 and eth2 are hooked up to the same switch, but are on diffrent VLANs |
| 06:28.12 | ChannelZ | VLANs configured in the switch it's self? |
| 06:28.18 | NateHB | eth2 cant see traffic from eth0 and eth0 cant see traffic to eth2 |
| 06:28.20 | NateHB | yes |
| 06:28.34 | ChannelZ | ok then |
| 06:28.57 | ChannelZ | But I'm still unclear why you're defining eth2 as nothing and eth2:1000 with the IP and such |
| 06:30.15 | NateHB | got a crapo cisco switch, but it does a good job, bought it off craigslist, and it would have been useless if one of my buddies wasnt a cisco wiz, he pulled out his baby blue rj45 to rs232, and rebuilt that switch from scratch, and enables the webinterface so I could maintain it |
| 06:30.38 | ChannelZ | unless eth2 is configured DHCP and you're just adding on that additional static IP with the alias? |
| 06:31.11 | NateHB | Ok, i reading more about the VLAN, looks like I might have missed some setting in my eth2:1000 cfg, |
| 06:31.52 | NateHB | eth2:1000 is a virtual interface, i think I need to tell it what the physical interface is and tell it that it is in fact tagged with a vlan |
| 06:32.35 | ChannelZ | interface aliases are aliases of the interface.... |
| 06:32.59 | p3nguin | http://www.cyberciti.biz/tips/howto-configure-linux-virtual-local-area-network-vlan.html |
| 06:33.04 | NateHB | are you saing I dont need the interface, I can just tag eth2? |
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| 06:33.23 | ChannelZ | from what I understand of your setup, yes |
| 06:33.58 | NateHB | ahh, dang, crap that was one of the pages I found but didnt read, arg |
| 06:34.01 | ChannelZ | eth2 isn't already doing anything, it should just be configured as what you have eth2:1000 setup as now |
| 06:38.20 | NateHB | ok, now I'm reading on this page, it is assuming that I want two interfaces, 0 and 0.5 and 0.5 know its a vlan because VLAN=yes |
| 06:38.31 | NateHB | so in my setup, i do VLAN=yes, but how do I specify what VLAN ID to use if I'm just using eth2 and not eth2.1000 ? |
| 06:39.56 | NateHB | wait, I just answered my own question, I have the switch tell it what VLAN |
| 06:40.04 | NateHB | I think, im going to try it |
| 06:42.27 | ChannelZ | well I'm not sure how it works trying to do what you're trying to do by using a separate nic.. as opposed to having the vlan created on eth0 |
| 06:43.42 | NateHB | I think I just enable VLAN=yes, and then use the switch and set that port in cisco speak as a DESKTOP on VLAN 1000 |
| 06:44.45 | NateHB | any way I can identify that switch port with out walking over there and following the wires? |
| 06:45.12 | ChannelZ | shrugs - I dunno what the 'DESKTOP' terminology means |
| 06:45.21 | ChannelZ | unplug it on your end and see which light goes out? |
| 06:46.05 | NateHB | how can I change the speed setting, make the card 10TX in stead of 100TX? |
| 06:47.07 | NateHB | ChannelZ, cisco (or least this one) has this really shitty cisco centric VLAN configuration page, its either a desktop, and router, or switch |
| 06:48.38 | ChannelZ | well my question is, is it really a VLAN from the standpoint of the server, or is all of that being abstracted in your switch for that particular port -- so from your server's point of view it's just another ethernet connection as if it were on a separate physical network |
| 06:50.17 | ChannelZ | (IE is it really getting these 'tagged frames') |
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| 06:53.40 | NateHB | that worked, i just set that switch port to native VLAN1000 |
| 06:54.03 | ChannelZ | and dumped the alias on the linux side? |
| 06:54.30 | NateHB | yup, i dumped the alias, set eth2's script with VLAN=yes |
| 06:54.41 | ChannelZ | I'm not even sure you need that |
| 06:54.59 | NateHB | im going to try it without then |
| 06:55.51 | ChannelZ | I think as far as your server is concerned it has no idea (or doesn't need to care) there's a VLAN on that interface because it's all being sorted out by your switch |
| 06:56.25 | ChannelZ | Now, if you didn't HAVE an eth2 and had to run it over the same physical eth0 interface, that's where you'd need to config the alias and tell it about the right VLAN ID, etc. |
| 06:57.14 | NateHB | hey cool, your right |
| 06:57.26 | NateHB | well, that makes things easier |
| 06:57.49 | NateHB | now to get the aastra scripts going and auto provision my phones |
| 06:57.59 | p3nguin | It's going to save your configuring trunking, too. |
| 06:58.32 | p3nguin | well, maybe. |
| 07:00.29 | p3nguin | I guess that depends on how you physically wired things. |
| 07:04.03 | NateHB | now if I have nat turned, i can let my phones reinvite, correct, only problem is that the server is thier default gateway |
| 07:04.41 | p3nguin | If * is on the same LAN, you can probably reinvite. |
| 07:05.38 | NateHB | would it be better if I give that interface a ip address on the router, and forward ports to my *, and let the phones deal with the router, is this better, give the * box the ip and let him route the outgoing packets? |
| 07:06.53 | NateHB | router <-----> eth1 [*] eth2 <------> switch <-------> phones |
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| 10:09.15 | benngard | any ${BLINDTRANSFER} expert here? i have noticed that ooh323 dont set that variable |
| 10:10.15 | benngard | have a "dump" of a transfered call but dont have a clue how to "modify" the code |
| 10:14.30 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 10:28.30 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 10:41.58 | *** join/#asterisk DND (n=arabia@94.200.7.26) |
| 10:42.32 | DND | hi guys. i have a line here that disables route password: database put AMPUSER 3137/pinless NOPASSWD |
| 10:42.53 | DND | now i wanted to remove that and re-enable password authentication again |
| 10:49.03 | ChannelZ | so database del AMPUSER 3137/pinless |
| 10:49.24 | DND | ok will try that so i will not include NOPASSWD ? |
| 10:49.33 | ChannelZ | NOPASSWD is the value |
| 10:49.40 | *** join/#asterisk [arctus] (n=arctus@124-171-6-167.dyn.iinet.net.au) |
| 10:50.54 | DND | ok thanks ChannelZ |
| 10:51.06 | ChannelZ | but it sounds like you want to just delete the whole key |
| 10:51.54 | *** join/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 11:05.09 | *** join/#asterisk siera04 (n=chatzill@113.18.8.82) |
| 11:09.53 | ArtemMakhutov | Hello, how should be multiline variables transmitted with the manager API? |
| 11:10.21 | *** join/#asterisk siera04 (n=chatzill@113.18.8.82) |
| 11:10.35 | siera04 | hi, i am newbee and installed 8FXS zaptel card in my asterisk server. |
| 11:10.40 | siera04 | when the analog phone dials '2001', the call is okay, but when the analog phone dials '2001#', the call is failed. |
| 11:10.46 | siera04 | the asterisk said me "Can't match 2001# from '2002' in context DAHDI" |
| 11:10.52 | siera04 | how can i do to dial '2001#' to call other phone? |
| 11:11.05 | siera04 | my DAHDI version is 2.2.0-rc1 |
| 11:17.24 | tzafrir_laptop | recommends 2.2.1-rc2 |
| 11:19.15 | tzafrir_laptop | siera04, you generally don't need the '#' to end a number |
| 11:19.47 | siera04 | my guests are used to use "#" before. |
| 11:20.03 | siera04 | tzafrir_laptop: so i have to fix this problem. |
| 11:21.14 | ChannelZ | so put it in your dialplan....? |
| 11:22.40 | ChannelZ | exten => 2002#,1,Dial(...) |
| 11:23.45 | Meaw | most calling card clients uses # to end the number |
| 11:24.09 | siera04 | ChannelZ: thank u for your help. |
| 11:24.33 | siera04 | i know it's easy to fix the dialplan as you said. |
| 11:25.02 | ChannelZ | was that not the question or..? |
| 11:25.07 | siera04 | but i want to know another way, eg: change the dahdi configuration. |
| 11:25.43 | verywiseman | i need good program for video conf integrated with * pls |
| 11:26.18 | ChannelZ | siera04: that's not really a function of DAHDI as a whole |
| 11:26.19 | siera04 | i have memory which i didn't meet this problem in asterisk1.4.18 and zaptel driver. |
| 11:26.50 | tzafrir_laptop | siera04, the dialplan has changed? |
| 11:27.07 | siera04 | i don't think so.. |
| 11:27.12 | siera04 | let me check it again... |
| 11:27.46 | ArtemMakhutov | wasn't there a option in zap.conf to threat the # as the "end" of the dialed number? |
| 11:28.38 | tzafrir_laptop | verywiseman, no built-in capability |
| 11:28.45 | tzafrir_laptop | See also http://www.voip-info.org/wiki/view/asterisk+video+conference |
| 11:33.39 | ChannelZ | ArtemMakhutov: well if it did DAHDI should have it too.. |
| 11:38.52 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 11:47.32 | siera04 | ChannelZ, i checked it again, SIP phone has same problem now. |
| 11:47.43 | siera04 | asterisk 1.4.18 has no this problem. |
| 11:48.22 | *** part/#asterisk [arctus] (n=arctus@124-171-6-167.dyn.iinet.net.au) |
| 11:50.33 | siera04 | <PROTECTED> |
| 11:50.39 | ChannelZ | well a SIP phone can be a little bit different story.. the phone it's self might allow you to program # as a "send now" digit to hotwire the phone's own dialplan |
| 11:50.42 | siera04 | another box is asteirsk 1.4.18 |
| 11:52.01 | siera04 | ChannelZ, i have tested it with same condition in 2 versions, i have no changed x-lite setting just now. |
| 11:52.13 | siera04 | and, no same result |
| 11:52.26 | m0e | quick question.. what is the Noop() function? |
| 11:52.37 | ChannelZ | m0e: "do nothing" |
| 11:53.11 | m0e | hrm.. i was looking at dialplans (trying to make sense of them, and it seems like they usually start with Noop().. any reason why? |
| 11:53.18 | ChannelZ | m0e: but it has the bonus of printing out in the console, so you can use it as a sort of debug output.. Noop(HEY the phone rang!) |
| 11:53.26 | m0e | aaah.. thats why! |
| 11:53.32 | m0e | ok perfect.. thanks a bunch :) |
| 11:53.41 | ChannelZ | I usually start mine with NoOp(${CALLERID(all)}) |
| 11:54.07 | ChannelZ | not every extension mind you |
| 11:54.11 | ChannelZ | but anyway |
| 11:54.14 | m0e | *nods* |
| 11:54.53 | ChannelZ | siera04: so you're using an identical dialplan on one version vs. another and getting different results? |
| 11:55.17 | siera04 | yes, i make simple dialplan for test just now. |
| 11:55.21 | siera04 | like that. |
| 11:56.07 | siera04 | [from-internal] |
| 11:56.15 | siera04 | exten => _.,1,Set(fromext=${CALLERID(num)}) |
| 11:56.16 | siera04 | exten => _.,n,Set(toext=${EXTEN}) |
| 11:56.52 | siera04 | exten => _.,n,Dial(...) |
| 11:57.36 | siera04 | and ${EXTEN} = '2001#' in 1.6.0.3 version and ${EXTEN} = '2001' in 1.4.18 version |
| 12:09.11 | ChannelZ | I don't know, perhaps this was a bug in the past where # was being ignored |
| 12:24.14 | siera04 | ChannelZ: thanks for your kind. |
| 12:24.20 | siera04 | i will check it again now. |
| 12:24.46 | siera04 | I think SIP phone has no this problem. :) |
| 12:25.10 | siera04 | i think it's only dahdi problem. |
| 12:27.22 | ChannelZ | as I said a SIP phone is different because A. it has a dialplan of it's own and B. depending on that dialplan may or may not include the # in the dial string depending on how it's programmed and what you dialed |
| 12:28.17 | ChannelZ | A softphone doesn't have a dialplan and will simply send everything you type in as soon as you hit the call button, so the # would pretty much always be included in what is sent to asterisk |
| 12:29.41 | ChannelZ | I'm not really seeing anything in the DAHDI or Zaptel configs that would give the # any special behavior |
| 12:34.30 | siera04 | ChannelZ: thansk for your kind. |
| 12:49.14 | *** join/#asterisk glam (n=glam@58.254.68.109) |
| 13:01.36 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.23.118) |
| 13:05.51 | *** join/#asterisk bsdbandit (n=csh11@wsip-24-249-123-207.hr.hr.cox.net) |
| 13:05.59 | bsdbandit | good morning everyone |
| 13:06.50 | bsdbandit | im looking to setup my own home phone system but im a total newbie to voip is asterisk orielly book a good place to start or do i need another book |
| 13:06.53 | bsdbandit | ? |
| 13:09.51 | *** join/#asterisk chuji (n=chuji@c-174-49-16-97.hsd1.tn.comcast.net) |
| 13:11.43 | *** join/#asterisk Keal (n=jargon@unaffiliated/jargon) |
| 13:12.14 | Keal | It says Asterisk is newest version but dkpg -l asterisk* doesn't list it |
| 13:12.50 | glam | bsdbandit: this is a new book and is quite good "http://www.amazon.com/Practical-Asterisk-1-4-1-6-Beginner/dp/0321525663/ref=sr_1_2?ie=UTF8&s=books&qid=1262524354&sr=8-2 |
| 13:23.16 | bsdbandit | oh ok |
| 13:23.17 | bsdbandit | cool |
| 13:23.20 | bsdbandit | thank you |
| 13:23.48 | Keal | hi |
| 13:30.15 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:33.52 | *** join/#asterisk vally (i=vally@ip-95-222-216-123.unitymediagroup.de) |
| 14:01.35 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
| 14:23.16 | Akiraa | If an IP phone has WAN and LAN ports, is it normal for it to stand ahead of a router in order to gain internet connectivity? For some reason, it can't be slaved to an existing router (i.e. get an internet connection from a router's LAN port) |
| 14:25.10 | glam | Akiraa: an IP phone has WAN port ? |
| 14:25.13 | Akiraa | It can get internet connectivity through the WAN port from an ADSL modem, for instance |
| 14:25.39 | Akiraa | and can work as a router between the modem and a separate computer (with NAT functions) |
| 14:26.58 | Akiraa | this is convenient for small offices that need only one IP phone line |
| 14:31.45 | *** join/#asterisk af_ (n=getsmart@88-149-240-72.dynamic.ngi.it) |
| 14:46.19 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:88d:2fee:a077:acde) |
| 14:46.21 | cusco | hi |
| 14:46.58 | cusco | core show channel "channel" tells me lots of info, including the time that the call was answered. But it doesn't tell me fo how long it has been answered |
| 14:47.29 | Akiraa | glam: aha, apparently I have to enable bridging mode, which disables downstream lan connectivity from the phone itself |
| 14:47.32 | cusco | is there any thing else that tells me for how long is the call going |
| 14:50.53 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 14:53.17 | *** join/#asterisk hohum (i=dcorbe@apollo.corbe.net) |
| 14:54.39 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 14:55.34 | shamelessn00b | hi guys, I'm currently stuck configuring my sangoma card for ss7 config and I get this error when I do a module load chan_dahdi.so |
| 14:55.45 | shamelessn00b | ERROR[3962]: chan_dahdi.c:15424 linkset_addsigchan: Unable to open SS7 sigchan 16 (Device or resource busy) |
| 14:56.00 | shamelessn00b | pastebinning my chan_dahdi file |
| 14:58.14 | shamelessn00b | http://pastebin.com/m2c58f246 |
| 14:58.18 | shamelessn00b | please have a look |
| 15:01.59 | *** join/#asterisk e4voip (n=chatzill@96-42-136-87.dhcp.trcy.mi.charter.com) |
| 15:04.37 | Keal | Why would Asterisk say newest version already in place during attempt to reinstall, but dpkg -l asterisk* not list the asterisk? |
| 15:04.42 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-37-228.acanac.net) |
| 15:05.11 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:10.03 | Keal | What Asterisk add-on module bridges a mp3 or shoutcast stream into a conference? |
| 15:11.37 | Keal | I need it to be able to have the Asterisk box downlink the feed or file directly into a conference as an audio stream from a remote source http or shoutcast server directly into the conference using an on-the-fly method of user controls within the conference. |
| 15:11.50 | voipmonk | Nice |
| 15:12.02 | Keal | This is for http://thekealshow.com/ for the 857 conference on both Telephreak and The Crib |
| 15:14.28 | Keal | If there exists no known addin for Asterisk, where can I find opensource samples along with tuts on how to make my own for Asterisk that accomplishes such? |
| 15:15.02 | Keal | I prefer samples with Gnu Public License 2.0 |
| 15:15.21 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 15:16.07 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 15:16.21 | Keal | I don't really agree with the terms for GPL 3.0 |
| 15:16.52 | Keal | ...a lot of open source fanatics don't. |
| 15:18.44 | Pan3D | oh god, not in #asterisk. please no. |
| 15:20.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:21.36 | shamelessn00b | http://pastebin.com/m519c3d0d |
| 15:21.47 | shamelessn00b | when I originate a call using dahdi my asterisk crashes |
| 15:21.57 | shamelessn00b | thats my chan_dahdi conf file |
| 15:22.48 | shamelessn00b | keal use the MeetMe |
| 15:22.50 | shamelessn00b | application |
| 15:22.59 | shamelessn00b | I'm using that |
| 15:23.01 | shamelessn00b | aswell |
| 15:23.06 | shamelessn00b | for the same purpose |
| 15:26.42 | Keal | The English phrasing of the Gnu Public License seems to imply that dervative works of independent origin such as the original Compaq BIOS would not properly be protected under such a license. |
| 15:27.26 | Keal | The English phrasing of the Gnu Public License 3.0 * seems to imply that dervative works of independent origin such as the original Compaq BIOS would not properly be protected under such a license. |
| 15:27.43 | Keal | In 2.0 there seems to be no such issue in GPL. |
| 15:36.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:40.09 | zaoul | I have a clipcom cp100 voip phone; made an exten for it along with an exten for an ekiga soft phone. Imade an exten for MP3Player, sounds pretty good! my question is when does it not sound good or what makes it sound like crap? many users? using poor codec? |
| 15:49.56 | cusco | is there a cli command that lists for how long the call is ongoing? |
| 15:51.05 | *** join/#asterisk a1 (i=a1@c-24-19-90-75.hsd1.wa.comcast.net) |
| 15:51.45 | a1 | Hey guys, it is possible to setup SS7 on asterisk and use my own LIDB? Is there any way to force LIDB dips through my LIDB server? |
| 15:52.19 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:00.40 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:00.59 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 16:01.01 | *** join/#asterisk smooth_penguin (n=smoove@59.95.30.48) |
| 16:03.02 | shamelessn00b | Failed to open /dev/dahdi/transcode: No such file or directory |
| 16:03.07 | shamelessn00b | any idea what this means?? |
| 16:03.44 | Chainsaw | shamelessn00b: That means that whatever telephony adapter you're using does not transcode between different codecs. |
| 16:04.10 | Chainsaw | shamelessn00b: Unless it was supremely expensive or does not have any actual telephony ports on the board, that is expected. |
| 16:04.10 | shamelessn00b | Im trying to configure sangoma cards on ss7 |
| 16:04.17 | shamelessn00b | using libss7 library |
| 16:04.33 | shamelessn00b | it does transcode |
| 16:04.53 | shamelessn00b | A1xx series cards |
| 16:04.57 | shamelessn00b | sangoma |
| 16:05.07 | shamelessn00b | it even has hardware echo cancellation |
| 16:05.21 | shamelessn00b | also |
| 16:05.38 | Chainsaw | Hardware echo cancellation is less special than transcoding really. |
| 16:05.43 | shamelessn00b | if I try and originate a call to a context in my dialplan using dahdi, my asterisk is crashing |
| 16:05.53 | Chainsaw | But other then that it works great? |
| 16:06.14 | shamelessn00b | how can I call another dahdi channel |
| 16:06.26 | shamelessn00b | originate dahdi/1/??? |
| 16:06.29 | Chainsaw | DAHDI/1, DAHDI/2, etc |
| 16:06.47 | Chainsaw | I use capitals, but I'm not convinced it's case-sensitive. |
| 16:06.58 | shamelessn00b | its not case sensitive |
| 16:07.14 | shamelessn00b | can you please tell me how its done using the originate command |
| 16:07.16 | shamelessn00b | ?? |
| 16:07.16 | m0e | with regards to HW echo cancelation.. any docs on that? |
| 16:07.43 | m0e | i'm uding the digium tdm800p card |
| 16:08.07 | m0e | using* |
| 16:08.25 | shamelessn00b | Chainsaw: how to originate dahdi->dahdi call using originate |
| 16:08.38 | shamelessn00b | originate dahdi/1/dahdi,1?? |
| 16:08.56 | shamelessn00b | originate dahdi/1/dahdi,2?? |
| 16:08.58 | shamelessn00b | rather |
| 16:09.13 | Chainsaw | shamelessn00b: Never had the need, sorry. |
| 16:11.30 | shamelessn00b | http://pastebin.com/m6d2e1689 |
| 16:11.39 | shamelessn00b | does this indicate my configuration is wrong?? |
| 16:12.17 | Chainsaw | No, it suggests that your DAHDI driver is horrendously broken. |
| 16:12.53 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 16:13.17 | shamelessn00b | its working if I ry to run on ccs |
| 16:13.26 | shamelessn00b | but if I configure it for ss7 |
| 16:13.30 | shamelessn00b | I get this |
| 16:13.46 | shamelessn00b | im using libss7 btw |
| 16:13.49 | Chainsaw | shamelessn00b: Well, seeing how it's not a regular DAHDI driver, but some emulation layer supplied by Sangoma... |
| 16:13.52 | Chainsaw | shamelessn00b: Have you talked to them? |
| 16:13.58 | shamelessn00b | no |
| 16:14.08 | Chainsaw | Maybe you should. |
| 16:14.17 | Chainsaw | Emulation tends to have limits. Maybe you just found one. |
| 16:14.42 | shamelessn00b | I just want to make sure my configuration is correct |
| 16:14.56 | shamelessn00b | can you have a look at my chan_dahdi file?? |
| 16:15.23 | shamelessn00b | http://pastebin.com/m519c3d0d |
| 16:15.27 | Chainsaw | I don't have Sangoma hardware, so I can't reproduce the bug locally. |
| 16:15.57 | shamelessn00b | have you configured ss7 on asterisk before? |
| 16:16.37 | Chainsaw | Nope. |
| 16:16.48 | Chainsaw | This is why I can't help you. |
| 16:17.32 | shamelessn00b | theres a huge probability I'm not configuring the point code etc etc properly |
| 16:17.38 | shamelessn00b | so it doesnt know where to route the call |
| 16:18.11 | m0e | Chainsaw, do you know a place where I can read about hardware echo cancellation? |
| 16:18.17 | m0e | I cant seem to find much online |
| 16:18.28 | Chainsaw | m0e: I'd call/mail Digium. |
| 16:18.35 | Chainsaw | m0e: You get support with hardware you buy from them. |
| 16:18.47 | m0e | Chainsaw, yeah I bought it from them.. I guess I'll do that.. thanks |
| 16:22.03 | m0e | One more question if I may.. what are all these signlaing types.. LIke what would be the diff between Loop, Ground, or Kewl start.. |
| 16:22.22 | m0e | I'm just using the default specified by the dahdi generation scripts.. but I dont know exactly what they are |
| 16:22.23 | Chainsaw | Ah, I believe that has a reasonable explanation on wikipedia. |
| 16:22.25 | Chainsaw | Let me link you. |
| 16:22.32 | m0e | that would be great, thanks! |
| 16:23.20 | Chainsaw | m0e: http://en.wikipedia.org/wiki/Ground_start_trunk |
| 16:23.27 | Chainsaw | m0e: http://en.wikipedia.org/wiki/Loop_start |
| 16:23.29 | m0e | thanks a bunch :) |
| 16:23.51 | Chainsaw | m0e: Kewlstart is basically an extension of loop start, the article's a bit sparse: http://en.wikipedia.org/wiki/Kewlstart |
| 16:23.58 | m0e | thanks |
| 16:24.26 | Chainsaw | But as a short answer, you very likely want kewlstart. |
| 16:24.59 | m0e | I'll try that then.. I believe the gen script added groundstart |
| 16:25.01 | m0e | *checking* |
| 16:25.04 | Chainsaw | (Without you can't know that the other end has hung up, or you'd have to rely on the control tones) |
| 16:25.25 | m0e | oh nope.. its kewlstart! |
| 16:25.35 | Chainsaw | *nod* That looks valid to me. |
| 16:25.51 | m0e | great, thanks! |
| 16:32.07 | shamelessn00b | FINALLY |
| 16:32.10 | shamelessn00b | got it working |
| 16:32.32 | shamelessn00b | ok, heading home |
| 16:32.37 | shamelessn00b | take care all bye |
| 16:32.39 | Chainsaw | shamelessn00b: Glad to hear :) |
| 16:32.40 | Chainsaw | Bye. |
| 16:33.13 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 16:33.20 | shamelessn00b | sorry, what were you saying |
| 16:33.22 | shamelessn00b | ?? |
| 16:33.37 | shamelessn00b | Chainsaw: |
| 16:33.55 | Chainsaw | shamelessn00b: Glad to hear :) |
| 16:34.00 | Chainsaw | And then I said bye. |
| 16:34.03 | Chainsaw | But you were gone already. |
| 16:34.09 | shamelessn00b | lol |
| 16:34.13 | shamelessn00b | been working for 14 hrs |
| 16:34.15 | shamelessn00b | non stop |
| 16:34.21 | Chainsaw | It was worth it. |
| 16:34.25 | shamelessn00b | yeah |
| 16:34.35 | shamelessn00b | still have one more tricky issue to resolve |
| 16:34.48 | Chainsaw | But some sleep or other recreation is a good thing. |
| 16:34.53 | Chainsaw | You'll think clearer afterwards :) |
| 16:34.58 | shamelessn00b | im using ogg123 to stream from my icecast server into asterisk using musiconhold |
| 16:35.10 | shamelessn00b | and its going defunct at random |
| 16:35.15 | shamelessn00b | the ogg123 process |
| 16:35.45 | shamelessn00b | like out of 8 ogg123 processes (8 different streams) one goes defunct at random |
| 16:35.51 | shamelessn00b | and comes back up after some time |
| 16:36.02 | shamelessn00b | ok, see ya all |
| 16:36.06 | shamelessn00b | bye |
| 16:51.02 | *** join/#asterisk Meaw (i=dino@somnium.fakereality.net) |
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| 17:12.11 | m0e | I have a very weird problem if anyone wants to have a go at.. I setup Authenticate followed by DISA in the dialplan.. sometimes this works, other times it hangs up |
| 17:12.19 | m0e | this is what is outputted if it hangs up |
| 17:12.25 | *** join/#asterisk rare1980_ (n=rare@116.71.180.118) |
| 17:12.38 | m0e | << [ HANGUP (NULL) ] [DAHDI/1-1] |
| 17:12.41 | m0e | WARNING[11363]: file.c:744 ast_readaudio_callback: Failed to write frame |
| 17:12.43 | m0e | thats it |
| 17:16.50 | tzafrir_laptop | the caller hung up? |
| 17:17.02 | m0e | the caller is me.. the system hangs up automatically |
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| 17:48.38 | *** join/#asterisk tgrman (n=jcmoore@unaffiliated/tgrman) |
| 17:48.53 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:49.31 | *** join/#asterisk smooth_penguin (n=smoove@59.95.3.169) |
| 17:50.56 | *** join/#asterisk e4voip (n=chatzill@96-42-136-87.dhcp.trcy.mi.charter.com) |
| 18:17.03 | gr0mit | ok what is wrong with exten => s,1,noop,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) ? |
| 18:17.30 | gr0mit | itgives me NoOp("SIP/home303-b6671af0", "Sun Jan 3 18:14:37 2010)}") |
| 18:17.39 | gr0mit | which is not what i want :-( |
| 18:19.57 | m0e | NoOp does nothing |
| 18:20.08 | m0e | check the other functions |
| 18:20.13 | gr0mit | i know i am just trying to get the time in the right format |
| 18:28.29 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.230.99.dsl.dyn.forthnet.gr) |
| 18:38.17 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 18:40.59 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:42.39 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
| 18:42.39 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:43.52 | p3nguin | gr0mit: What do you want it to look like? |
| 18:45.58 | p3nguin | 20100103-124555 ? |
| 18:46.12 | gr0mit | yyyy-mm-dd hh:mm:ss |
| 18:47.36 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
| 18:48.21 | p3nguin | exten => time,n,NoOp(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) |
| 18:48.26 | p3nguin | <PROTECTED> |
| 18:48.45 | FlaPer87 | hey guys, I was trying to install g79 codec but I'm getting this message "g729tolin did not update samples 0" when I start asterisk and after that I get a segfault |
| 18:48.48 | FlaPer87 | any idea? |
| 18:49.11 | p3nguin | gr0mit: Do you want the hyphens and colons in the outputted date/time? |
| 18:49.57 | p3nguin | Personally, I use Verbose() for outputting the date/time in the CLI, but NoOp() still works fine. |
| 18:50.04 | gr0mit | yup |
| 18:50.22 | gr0mit | this is a precursor to assigning it to a variable! |
| 18:50.48 | p3nguin | exten => time,n,NoOp(${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) |
| 18:50.52 | p3nguin | <PROTECTED> |
| 18:51.09 | gr0mit | copies and pastes ! |
| 18:51.13 | gr0mit | thank you sir! |
| 18:53.18 | *** join/#asterisk naif (n=naif@93-32-160-185.ip34.fastwebnet.it) |
| 18:53.20 | naif | ok |
| 18:53.29 | naif | hi to all |
| 18:53.31 | p3nguin | Hopefully that will get you what you're wanting. |
| 18:54.36 | naif | I have a dialer script that use SendDTMF. It call another asterisk. It went out in ulaw, reach my carrier and then comeback from another termination carrier always via uLaw. INBAND DTMF detection does not work. I tried to tune the timing of DTMF, in a random fashon i get some random number. |
| 18:54.47 | naif | I think that's because the quality of the channel is poor. |
| 18:55.11 | naif | What to do in that case? I already spent 1 day on this |
| 18:55.17 | naif | increased ms of playing tone |
| 18:55.24 | naif | from 30ms up to 300ms |
| 18:55.27 | naif | up to 3000ms |
| 18:55.35 | naif | nothing to do |
| 18:55.43 | naif | DTMF inband it's completely unreliable |
| 18:56.11 | naif | are there something else, that's not DTMF but use similar technique |
| 18:56.14 | naif | to signal data |
| 18:56.19 | naif | and receive data |
| 18:56.24 | naif | across asterisk voice stream? |
| 19:03.19 | *** join/#asterisk e4voip (n=chatzill@96-42-136-87.dhcp.trcy.mi.charter.com) |
| 19:04.35 | *** join/#asterisk _cgc (n=_cgc@94-194-207-211.zone8.bethere.co.uk) |
| 19:13.36 | *** join/#asterisk rare1980_ (n=rare@116.71.189.250) |
| 19:18.13 | *** join/#asterisk NateHB (n=noone@72.34.90.74) |
| 19:19.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:22.02 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 19:28.26 | cusco | hi |
| 19:28.32 | fenrus | evening |
| 19:28.37 | cusco | :) |
| 19:28.53 | cusco | what cli command would show me for how long a call as been answered |
| 19:30.53 | [TK]D-Fender | cusco: "core show channel [channel]" |
| 19:31.57 | cusco | [TK]D-Fender: it shows that it was answered at $time |
| 19:32.12 | cusco | but not for how long it is online |
| 19:32.35 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 19:32.40 | [TK]D-Fender | cusco: Show me... |
| 19:33.42 | cusco | http://paste.debian.net/55526/ |
| 19:35.04 | cusco | the "duration" is since it came on asterisk, listening to the ivr, queuwing.. before being answered |
| 19:35.33 | [TK]D-Fender | cusco: Wrong. * answered it IMMEDIATELY. |
| 19:35.54 | [TK]D-Fender | cusco: the fact * called SOMETHING ELSE afterward Is irrelevent |
| 19:36.06 | cusco | ahhh |
| 19:36.07 | [TK]D-Fender | cusco: THAT call has its own CDRs and is separate from this call |
| 19:36.07 | cusco | right |
| 19:36.35 | cusco | so I mist look at the channel SIP/blah |
| 19:36.40 | cusco | instead of DAHDI/blah |
| 19:36.42 | cusco | ok |
| 19:36.58 | cusco | thanks |
| 19:45.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:06.01 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 20:11.32 | ArtemMakhutov | How can I get better voice quality over a 256kbit WAN link? I am using an IAX trunk with ilbc as codec and have jitterbuffer enabled on both sides: |
| 20:11.34 | ArtemMakhutov | jitterbuffer=yes |
| 20:11.35 | ArtemMakhutov | forcejitterbuffer=yes |
| 20:11.37 | ArtemMakhutov | maxjitterbuffer=1000 |
| 20:11.39 | ArtemMakhutov | maxjitterinterps=10 |
| 20:11.40 | ArtemMakhutov | resyncthreshold=1000 |
| 20:11.42 | ArtemMakhutov | jittertargetextra=40 |
| 20:11.44 | ArtemMakhutov | trunkfreq=20 |
| 20:12.33 | _cgc | bandwidth = low? |
| 20:13.31 | ArtemMakhutov | What exactly does bandwith = low do? It just choose the audio codec, right? |
| 20:14.53 | ArtemMakhutov | I have allready set the audio codec manually, so bandwith=low should not have any influence? |
| 20:15.51 | ArtemMakhutov | Even with bandwith=low I get a lot of packet loss, while skype has absolutly no problems |
| 20:16.19 | ArtemMakhutov | What parameters should I try to change? |
| 20:24.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 20:31.21 | *** join/#asterisk TheDavidFactor-H (n=TheDavid@76.73.16.26) |
| 20:35.12 | *** join/#asterisk timholum (n=chatzill@d30-13.rb2.lax.centurytel.net) |
| 20:35.40 | timholum | Finaly after 4 days of working on it I got skype for sip working :) |
| 20:36.07 | TheDavidFactor-H | which version did you implement and on which version of asterisk? |
| 20:37.09 | timholum | asterisk 1.4 ( if you where talking to me :) ) |
| 20:37.39 | TheDavidFactor-H | timholum, I was, sorry for not sticking your name in there |
| 20:38.09 | timholum | no problem, I just got in here so i was not sure if there was an exsisting conversation going |
| 20:38.25 | TheDavidFactor-H | nah, it's dead 'round these parts ;-) |
| 20:39.12 | timholum | 4 days of troubleshooting my skype for sip and it turns out I forgot to put creadits on my skype account :) I felt like an idiot |
| 20:39.16 | ChannelZ | has to go into work and figure out what happened to his PoE switch |
| 20:39.44 | TheDavidFactor-H | ChannelZ, that sucks |
| 20:39.51 | ChannelZ | yah way to end a 4-day weekend |
| 20:40.13 | TheDavidFactor-H | the only good thing about it, is that the weekend is almost over anyway |
| 20:40.59 | TheDavidFactor-H | or just swear ;-) |
| 20:41.26 | TheDavidFactor-H | ^that was the last part of a statement that got lost |
| 20:42.17 | TheDavidFactor-H | timholum, those things are when you just smack youself in the forehead and |
| 20:42.33 | TheDavidFactor-H | swear you won't that again, or just swear ;-) |
| 20:44.18 | timholum | :) ya |
| 20:45.22 | TheDavidFactor-H | which skype for sip product are you using? |
| 20:50.56 | FlaPer87 | how can I wait for an extension (or digit) during a call? |
| 20:51.24 | TheDavidFactor-H | Wait |
| 20:52.52 | TheDavidFactor-H | sorry, meant WaitExten |
| 20:53.06 | TheDavidFactor-H | shot from the hip, without checking first |
| 20:53.40 | FlaPer87 | TheDavidFactor-H: so I should put it righ after the Dial command |
| 20:54.00 | TheDavidFactor-H | depending on what you want to do you may want to look at Read |
| 20:54.11 | FlaPer87 | TheDavidFactor-H: the thing is that I would like to add an exten to start recording the call and another to stop recording |
| 20:54.16 | FlaPer87 | TheDavidFactor-H: with Monitor |
| 20:54.29 | TheDavidFactor-H | oh, that's different |
| 20:54.56 | FlaPer87 | TheDavidFactor-H: any idea about how can I do that? |
| 20:54.58 | TheDavidFactor-H | what you want to do there is allow codes to be entered during a call |
| 20:55.09 | TheDavidFactor-H | what version of *? |
| 20:55.13 | FlaPer87 | 1.6 |
| 20:55.20 | timholum | isint that in features.conf? |
| 20:55.49 | TheDavidFactor-H | yes |
| 20:56.35 | TheDavidFactor-H | FlaPer87, you want to look at features.conf and the Dial w/W options |
| 20:59.53 | TheDavidFactor-H | FlaPer87, parts of a blog I wrote might be helpful for you: http://tdf-asterisk.blogspot.com/ |
| 21:00.12 | FlaPer87 | TheDavidFactor-H: thanks a lot |
| 21:00.29 | TheDavidFactor-H | I don't use the w/W options in the blog |
| 21:00.41 | TheDavidFactor-H | sure thing, best of luck |
| 21:04.43 | *** join/#asterisk af_ (n=getsmart@88-149-240-72.dynamic.ngi.it) |
| 21:07.43 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 21:08.49 | timholum | what soft phones would everyone recommend? |
| 21:09.07 | Chainsaw | timholum: Ekiga. |
| 21:10.09 | TheDavidFactor-H | timholum, what OS? |
| 21:10.16 | timholum | sorry, for windows |
| 21:10.36 | timholum | I am currently using xlite |
| 21:10.51 | TheDavidFactor-H | do you have any problems with it? |
| 21:11.57 | timholum | for some reason I can not use the dial pad in the middle of a call |
| 21:12.07 | timholum | So if I call a IVR I cant use it |
| 21:12.30 | *** join/#asterisk moy (n=moy@189.162.161.141) |
| 21:13.03 | TheDavidFactor-H | ok, well I use a simple phone called SJPhone, can't say it's the best or anything but it does the dialing stuff and we develop IVRs so I've used it to test them |
| 21:13.17 | TheDavidFactor-H | it is free, but I think it' |
| 21:13.19 | TheDavidFactor-H | s |
| 21:13.23 | TheDavidFactor-H | closed source |
| 21:13.50 | timholum | im fine with closed source as long as it works :) ( and is free ) |
| 21:19.17 | [TK]D-Fender | Timthen you set the wrong DTMF mode. |
| 21:19.31 | [TK]D-Fender | timholum: then you set the wrong DTMF mode. |
| 21:19.42 | [TK]D-Fender | Timthere is nothing wrong with X-Lite |
| 21:19.50 | Keal | IS voipmonk a real person or just a spambot? |
| 21:20.29 | Keal | ..or just a mentally imbalanced dumbass/ |
| 21:20.30 | timholum | I am pritty sure he is real, I think he helped me out a few days ago |
| 21:20.31 | Keal | ? |
| 21:20.49 | etfonhomey | Keal, both |
| 21:21.56 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 21:22.02 | Keal | I am pretty convinced the guy has some mental issues regarding proper netiquette regarding pms |
| 21:22.22 | etfonhomey | [TK]D-Fender, you have any Polycom phones running their SIP 3.2.2 image? |
| 21:22.30 | [TK]D-Fender | etfonhomey: Not yet |
| 21:22.31 | Keal | He did the equivalent of leave a telemarker call in a answermachine with my pm |
| 21:22.55 | [TK]D-Fender | etfonhomey: I have 2 x IP 320's at work that can at least support it... mine are all older now (600 / 501 /301 |
| 21:23.05 | etfonhomey | Keal, he'll help you and then PM you to let you know he accepts "donations". |
| 21:23.22 | Keal | I never pmed him, he just sent me a pm that sounded like a business card and poofed. |
| 21:23.54 | Keal | I never even addressed him. |
| 21:24.03 | etfonhomey | [TK]D-Fender, when I do a check-sync to force the phones to reboot, they send back 2 x SIP 403 - Forbidden messages, but they still reboot. |
| 21:24.23 | Keal | I mean I understand such things are acceptable on freenode, but he is really digging. |
| 21:24.33 | etfonhomey | I don't recall seeing that under earlier versions of their SIP image. |
| 21:25.11 | [TK]D-Fender | etfonhomey: Doesn't sound like something worth worrying about |
| 21:25.17 | etfonhomey | Doesn't affect functionality, ju |
| 21:25.21 | etfonhomey | Exactly. |
| 21:25.57 | Keal | Me on the otherhand, I am a bard. |
| 21:26.18 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 21:27.06 | fenrus | i am bored. |
| 21:27.09 | etfonhomey | [TK]D-Fender, thought I would check and see if you'd seen it before and knew of an easy fix, but it's really no big deal. |
| 21:27.16 | Keal | I do stuff for people, then a year down the road of them being helped without them putting into effort into it and not allowing me to give them the reins I hold it ransom, then when they pay me I give it to them in full for them to take care of, then put the money toward an emergceny fund for homeless hackers. |
| 21:27.46 | Keal | That is how I roll, because I am a bard, just an artist that programs etc. |
| 21:28.33 | [TK]D-Fender | Keal: "Note to subway musicians : just because nobody understands you ... doesn't mean you're an 'artist'" |
| 21:28.58 | Keal | I don't like being stuck with dns, hosting, pbx's, etc indefinately that I only have on my hands to train the owner how to administrate. |
| 21:29.55 | etfonhomey | [TK]D-Fender,You may already know this, but presence works on the 301's with the newer SIP images. Where it used to just have the dialpad icon when you had a Buddy on your line appearance. |
| 21:30.16 | etfonhomey | I think 3.1.2 is last to support 301. |
| 21:30.18 | [TK]D-Fender | etfonhomey: really... news to me... |
| 21:30.34 | [TK]D-Fender | etfonhomey: I only have 1 left anywhere regardless |
| 21:34.05 | zaoul | I have setup ekiga as an extensions and a clipcom voip phone, they both can call each other but no voice is transmitted. What could be wrong? |
| 21:35.24 | Keal | People that take advantage of me to put money into stuff in-order to teach them, but never take upon themselves to actually start paying for it thru their own independent ownership and adminstration I attempted to teach them tick me off. |
| 21:35.52 | Keal | I usually pay for people to teach them. |
| 21:36.21 | Keal | That is why when they take advantage of me I hold the server ransom until they pay the difference to aquire it. |
| 21:48.35 | ChannelZ | well that was a wasted trip. |
| 21:49.05 | TheDavidFactor-H | back already? that wasn't to long a trip, but why was it wasted? |
| 21:49.49 | ChannelZ | well because nothing was dead, I'm not even sure what went wrong. I just reset the switch and it's all fine. But I could have done that in the morning had I known |
| 21:50.23 | TheDavidFactor-H | that sucks |
| 21:50.33 | ChannelZ | I thought one of the power supplies or the switches it's self had died, cuz like 6 of my phones went offline all at the same time the day before yesterday |
| 21:50.43 | p3nguin | "itself" |
| 21:51.11 | TheDavidFactor-H | oh no, a grammer nazi ;-) |
| 21:51.17 | p3nguin | grammar |
| 21:51.30 | TheDavidFactor-H | see :-P |
| 21:51.33 | ChannelZ | says the guy who can't type an 'e' in 'penguin' |
| 21:51.34 | p3nguin | spelling |
| 21:54.46 | TheDavidFactor-H | but it's not a normal pebguin, it's a l33t penguin :-D |
| 21:55.18 | p3nguin | Someone that doesn't know the difference between "it's" and "its" certainly doesn't have any room to criticize my choice of IRC nickname, which, coincidentally, doesn't even HAVE to be a word. I could have chosen random characters and still had a valid nickname. |
| 21:55.27 | TheDavidFactor-H | *penguin, i'm one handed here |
| 21:55.41 | ChannelZ | Glass houses |
| 21:56.06 | TheDavidFactor-H | for the record, I'm just picking |
| 21:56.10 | jaytee | somehow, somewhere out there, someone's cornflakes have been pissed in. |
| 21:56.24 | p3nguin | I know the difference between "it's" (a contraction) and "its" (a possessive pronoun). |
| 21:56.30 | p3nguin | No glass house here. |
| 21:56.43 | ChannelZ | Good for you. |
| 21:56.57 | p3nguin | After this, maybe you'll understand it as well. |
| 21:57.35 | ChannelZ | No I'll type words wrong just to give you a sense of purpose |
| 21:58.07 | p3nguin | I was trying to help you learn, where you, on the other hand, are just being a defensive douchebag over it. |
| 21:58.15 | *** join/#asterisk SirWhit (n=sirjames@wsip-24-234-247-44.lv.lv.cox.net) |
| 21:58.43 | SirWhit | Anyone online today? |
| 21:58.54 | ChannelZ | Nope |
| 21:58.57 | SirWhit | ;) |
| 21:59.00 | Carlos_PHX | Nobody. |
| 21:59.01 | TheDavidFactor-H | not me |
| 21:59.38 | SirWhit | let me rephrase then... anyone online today who knows about the TC400 series cards? |
| 21:59.48 | FlaPer87 | awesome it works |
| 22:00.00 | FlaPer87 | now I want to get transfer working |
| 22:00.06 | FlaPer87 | oh and conference calls too |
| 22:00.08 | FlaPer87 | =D |
| 22:00.36 | SirWhit | I have the weirdest problem, I'm trying to trunk 2 * servers together |
| 22:00.36 | Katty | infobot: crittercam |
| 22:00.37 | infobot | rumour has it, crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 22:00.46 | Katty | ^- the end of the world is coming, the birds say so |
| 22:01.19 | Katty | there's an insane ammount of birds in the front yard. i think they're trying to tell me something. |
| 22:01.19 | SirWhit | I've tried iax and sip and it works great with ulaw and gsm.. but when I try to use g729 using 2 of those TC400 cards.. dead air |
| 22:01.22 | Katty | maybe... snowstorm? |
| 22:01.50 | Carlos_PHX | SirWhit: How many g.729 licenses do you have? |
| 22:01.58 | [TK]D-Fender | Katty: We're getting pummeled here ... hard |
| 22:02.02 | SirWhit | those cards include 120 G.729 licenses |
| 22:02.08 | Katty | [TK]D-Fender: oh? |
| 22:02.14 | TheDavidFactor-H | FlaPer87, Dial options t/T and MeetMe |
| 22:02.23 | [TK]D-Fender | Katty: White shit... mountains of it |
| 22:02.28 | Katty | oh i see. |
| 22:02.35 | Katty | maybe i should run to the store |
| 22:02.43 | ChannelZ | It just started 'flurrying' here |
| 22:02.44 | Katty | and buy a case of toilet paper, and makeup remover. |
| 22:03.02 | *** join/#asterisk GreGrenada (i=4549f3c2@gateway/web/freenode/x-ogrvaokfdjneqtsz) |
| 22:03.13 | FlaPer87 | TheDavidFactor-H: and for conference calls? |
| 22:03.29 | TheDavidFactor-H | MeetMe, it was two parts |
| 22:03.40 | Katty | no snow in the forecast. this is very weird. |
| 22:05.03 | e4voip | anyone here familiar with res_phoneprov ? |
| 22:07.52 | ChannelZ | neeooooope. |
| 22:08.06 | Katty | hi ChannelZ |
| 22:08.18 | ChannelZ | Howdy Katty |
| 22:10.42 | ChannelZ | If I sick logrotate on asterisk's logs, do I need to kick it afterwards? (does it leave the filehandle open as it writes?) |
| 22:11.29 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 22:11.42 | ChannelZ | I guess I can use it's built-in rotating |
| 22:12.14 | Kobaz | ChannelZ: the built in rotating is backwards to logrotate, it's kinda annoying |
| 22:12.23 | Kobaz | you can use logrotate, you just need to send asterisk a logger reload |
| 22:12.50 | p3nguin | "its" |
| 22:14.42 | zaoul | I just bought a loffs |
| 22:14.45 | zaoul | arg |
| 22:14.47 | zaoul | offs |
| 22:15.05 | ChannelZ | A what now? |
| 22:15.05 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 22:15.19 | zaoul | Eye leik uewsing asterisk ets teH gude |
| 22:15.28 | zaoul | ryte p3nguinz? |
| 22:15.36 | p3nguin | what? |
| 22:15.46 | ChannelZ | "What" |
| 22:15.51 | zaoul | asterisk ez teH gude |
| 22:16.01 | p3nguin | oh |
| 22:17.01 | zaoul | Eye hov rly gude speluing |
| 22:17.34 | p3nguin | Maybe that's true, but your typing SUCKS! |
| 22:19.52 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 22:21.57 | nny | working with fop2 beta, and still getting used to the code, but working on an issue with it's ability to call park. See http://pastebin.com/m4fcb803c I am not zesty enough to fully understand how the park context is working, and wondering if anyone with eyes better than mine can see why it would fail to announce the parking lot number, (note: it works great otherwise) |
| 22:21.57 | FlaPer87 | I'm having problems whit the transferwaittime, When I do #1 it ask me for an extensions but when I try to write the extension the time is too short, so I'm able to click just the first digit of the extension |
| 22:23.03 | p3nguin | WTF... does NO ONE know the difference between "it's" and "its" these days? |
| 22:23.07 | p3nguin | I find it hard to believe THAT wasn't taught in schools. |
| 22:23.36 | nny | p3nguin: lol |
| 22:23.58 | ChannelZ | Maybe you're in the wrong channel. |
| 22:24.04 | nny | p3nguin: i assumed the "ability to park" belonged to "it", my bad. was a typo, and ffs, I went to school in the 80s and 90s |
| 22:24.57 | p3nguin | nny: Your assumption wasn't wrong, but "it's" means "it is." "Its" is the possessive pronoun you were looking for. |
| 22:25.25 | nny | p3nguin: yes I know that know that you pointed it out, it should be able to be replaced by it has or it is, otherwise it's "its" |
| 22:25.32 | nny | know that now* |
| 22:26.01 | nny | so yeah /leave #grammarnazisonsundayswithhangovers /join #asterisk |
| 22:26.23 | nny | (i get it enough from my GF) :P |
| 22:26.25 | ChannelZ | Don't mind him, he just needs more sex |
| 22:26.38 | FlaPer87 | ChannelZ: looooooooooool |
| 22:26.48 | nny | what's that say about my girlfriend? |
| 22:26.50 | p3nguin | nny: If this wasn't an active topic just a few minutes before you got here, I may not have even noticed the mistake in your sentence. |
| 22:26.52 | nny | oohhhhhhhhhh |
| 22:27.00 | nny | hmm brb need to take care of something |
| 22:27.02 | nny | :D |
| 22:27.07 | ChannelZ | nny, not you |
| 22:27.27 | p3nguin | And because you weren't here, I didn't specifically nail it on you, but rather decided to comment generally instead. |
| 22:27.40 | nny | was making a joke about how lack of sex leads to grammarpolice.. and how my GF is a super grammar.. oh hell joke explanation is a lost art |
| 22:27.58 | nny | idc, i am used to it, I mangle the english language far too much to not be used to it |
| 22:28.25 | nny | and I am natively born hear, whiter than alfonso from fresh prince |
| 22:28.28 | nny | here* |
| 22:28.30 | nny | :\ |
| 22:29.00 | p3nguin | lol |
| 22:30.45 | p3nguin | nny: I also notice when someone makes a typo on occasion and when the mistake is a persistent one. The former means they don't pay attention to what they are typing, where the latter means they didn't pay attention in school. |
| 22:31.52 | p3nguin | Neither of which I am applying to your mistake; again, just a generalized comment. |
| 22:32.03 | nny | :) |
| 22:32.22 | nny | so according to the app it's ParkAndAnnounce(announce:template|timeout|dial|return_context)Â |
| 22:33.01 | TheDavidFactor-H | FlaPer87, how long does it wait if you don't enter any ext? |
| 22:33.02 | nny | I am going to NoOp the first line, trying to figure out what it is trying to do |
| 22:33.42 | ChannelZ | Oh now I'm confused, there are two people with parking problems |
| 22:34.01 | nny | eh this one is more of I am playing with beta code |
| 22:34.19 | nny | the developer tbh said he didn't want to have to add things to the dial plan to make it work, but this was ane xception |
| 22:34.22 | nny | exception* |
| 22:35.06 | FlaPer87 | TheDavidFactor-H: 3 |
| 22:35.13 | p3nguin | nny: I've never had to explicitly define parking contexts in extensions.conf for parking and the announcement of the slot number to work. Is there any particular reason you decided to do it that way? |
| 22:35.13 | FlaPer87 | TheDavidFactor-H: 3 seconds |
| 22:35.18 | [TK]D-Fender | Beta code can be blocked by 4" thick barrier of water |
| 22:35.51 | nny | p3nguin: I am not entirely clear, I will need to find the bit of code that handles the parking in this (FOP2). |
| 22:36.57 | p3nguin | Without a GUI mucking up things, I think you only need to include => parkedcalls. |
| 22:36.59 | nny | ok that makes sense, the code isn't using the context anyways hahaha |
| 22:37.07 | p3nguin | (after adjusting the things in features.conf, that is) |
| 22:37.12 | nny | p3nguin: yeah that's how I used to do it |
| 22:37.21 | nny | p3nguin: i am just testing out implementing FOP2 on systems |
| 22:37.23 | p3nguin | So FOP changed that behavior? |
| 22:38.02 | ChannelZ | fap fap fap |
| 22:38.11 | nny | p3nguin: no, I am unclear as to why he needs the extension addition, and don't want to speak for him. I am sure it's buried in the forums for the software in a discussion. The new version works wicked awesome, parking is a beta feature |
| 22:38.24 | FlaPer87 | btw, Is it possible to dial a number just if the previous dial didn't got answered? |
| 22:38.36 | p3nguin | flaper87: sure |
| 22:38.41 | [TK]D-Fender | FlaPer87: Dial them back to back |
| 22:39.26 | nny | p3nguin: and it actually still works, just no announce |
| 22:39.26 | nny | oh |
| 22:39.27 | nny | i think I know |
| 22:39.27 | nny | one sec |
| 22:39.27 | FlaPer87 | TheDavidFactor-H: any idea? =P |
| 22:39.56 | FlaPer87 | another thing, is it possible to set kKtTwW as default options instead of writting them on each Dial? |
| 22:39.58 | p3nguin | I had recently considered FOP, but now I am rethinking it. |
| 22:40.14 | nny | actually |
| 22:40.33 | [TK]D-Fender | [17:39]<FlaPer87>another thing, is it possible to set kKtTwW as default options instead of writting them on each Dial? <- no |
| 22:40.57 | FlaPer87 | [TK]D-Fender: thanks =D |
| 22:40.57 | nny | p3nguin: try it out, (FOP2, not FOP) |
| 22:41.10 | nny | p3nguin: I have a working deal here you can peep |
| 22:42.29 | nny | p3nguin: my experience so far is that the developer is available and courteous, I started this on thursday and still had help over the holidays. The parking lot thing is all beta code, I am just testing it out |
| 22:44.33 | ChannelZ | It'd be hard to say without seeing exactly what args he's sending to ParkAndAnnounce |
| 22:44.49 | nny | ChannelZ: well, i don't think it uses that context right now at all |
| 22:45.19 | nny | ChannelZ: it looks like it is using the regular parking method |
| 22:45.51 | ChannelZ | as in sending to extension 700 or whatever? |
| 22:46.25 | nny | ChannelZ: I think it uses the Asterisk Manager to park calls actually |
| 22:46.26 | nny | yes |
| 22:46.33 | nny | http://www.fop2.com/forum/viewtopic.php?f=3&t=42 |
| 22:48.02 | ChannelZ | ok this guy should be killed for this floating 'digium award' graphic on his web page |
| 22:48.10 | nny | awww ha |
| 22:48.31 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 22:49.15 | nny | at least it's documented well |
| 22:49.40 | nny | i tried to use astercrm (astercc) the other day, and have been guinea pig testing a bunch of stuff over the last month. Some of it is downright scary |
| 22:50.07 | nny | astercc (to be fair) is written by someone in another language, it just didn't make the translation. |
| 22:50.39 | nny | That and all the passwords are available via the web interface in plaintext |
| 22:51.35 | zaoul | I believe in a coalition for ... good hold music, anyone with me? |
| 22:51.42 | nny | heh |
| 22:51.49 | nny | you don't like robot ditty?!!! |
| 22:51.50 | zaoul | I mean ffs your already pissed off being on hold |
| 22:52.39 | zaoul | I just kinda caught myself getting a little frustrated here with the parking, but put on a good song... felt better, even listened to the whole thing on he phone LOL |
| 22:53.14 | nny | I am still curious about finding some nice open license classical, I have looked a bit previously |
| 22:53.23 | zaoul | heh your customers are like "put me back on hold damn it" ... certainly |
| 22:53.35 | nny | speaking of which anyone notice the IRS plays "The nutcracker" when you are on hold? |
| 22:53.44 | nny | funny one IRS! |
| 22:53.45 | zaoul | really? har |
| 22:53.52 | zaoul | yea they got jokes huh |
| 22:54.03 | nny | yeah but it distorts and sounds like the nutcracker played over loudspeakers in hell |
| 22:54.09 | nny | which is fitting |
| 22:54.16 | zaoul | i was asking about that earlier, what causes that? |
| 22:54.26 | zaoul | hold music breaking up, but the call itself is clear |
| 22:54.42 | zaoul | QoS? |
| 22:55.13 | nny | could be an issue with the range of frequencies in the MOH vs the codec compression.. or something |
| 22:55.35 | nny | that's an uneducated Holiday Inn Express answer |
| 22:56.02 | TheDavidFactor-H | <[TK]D-Fender> FlaPer87: Dial them back to back <- does this work? |
| 22:56.21 | [TK]D-Fender | TheDavidFactor-H: What.. do you think I jsut lied to him? |
| 22:56.30 | FlaPer87 | lol |
| 22:56.51 | TheDavidFactor-H | no, but I never saw a follow up and I had to be gone for a few minutes |
| 22:57.07 | TheDavidFactor-H | some people can't hit buttons that fast :-P |
| 22:57.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:57.48 | *** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) |
| 22:58.09 | nny | p3nguin: an SS of FOP2 if interested http://www.atlantiatech.com/Untitled.png |
| 22:58.49 | nny | has a shiny popup with a picture if the person calling if they are in the phonebook too |
| 22:59.18 | p3nguin | I'm thinking about seeing if I can integrate it without breaking anything. |
| 22:59.51 | nny | p3nguin: it is standalone for the most part, the extensions addition I mentioned is the only one, and even that is 1.) from beta and 2.) doesn't seem to be used yet |
| 23:01.03 | nny | sorry if I sound zealous, I get excited when I get a malformed cheeto. Easily amused they say... :) |
| 23:01.41 | TheDavidFactor-H | l8r all |
| 23:03.01 | p3nguin | nny: I'm concerned about one thing. The downloads specifically list glibc version that should be used. I have glibc 2.11.1, which is not listed at all. |
| 23:03.50 | nny | p3nguin: to be honest I didn't check that |
| 23:03.58 | nny | p3nguin: how do you show which version you have installed? |
| 23:04.26 | p3nguin | I simply ran "pacman -Qs glibc" |
| 23:04.33 | p3nguin | I'm sure your system is different, though. |
| 23:06.14 | nny | according to the forums (this is a rackspace cloud host, CentOS5) it's 2.5-12 |
| 23:06.23 | p3nguin | "ls /lib/ld-2.*.so" might also be a good way to see what is installed. |
| 23:06.48 | nny | yeah 2.5 it appears |
| 23:07.23 | p3nguin | I guess I'm too updated for their taste. Maybe it'll work anyway. |
| 23:07.52 | nny | it creates /usr/loca/fop2, /var/www/html/fop2 and /etc/init.d/fop2 (on my RH based system) |
| 23:07.57 | p3nguin | They also expect that everyone runs either Debian or CentOS. |
| 23:08.20 | nny | oddly enough the source is non distro specific |
| 23:08.25 | p3nguin | I'll have to break open one of the RPMs and make my own package. |
| 23:08.38 | p3nguin | I didn't see any source, but I didn't look very hard. |
| 23:08.41 | nny | i ended up with the source, let me grab you the link |
| 23:08.58 | nny | i have the beta 32 bit source |
| 23:09.10 | FlaPer87 | what programming language was used to write fop2? |
| 23:09.17 | nny | wget http://www.fop2.com/downloads/fop2.tgz |
| 23:09.33 | nny | I dunno, it's php, js and flash according to what's in the webdir |
| 23:09.46 | FlaPer87 | nny: nice, thanks |
| 23:09.49 | nny | there is also "fop2_server" |
| 23:10.05 | FlaPer87 | I see |
| 23:10.13 | FlaPer87 | I'll take a look to it tomorrow |
| 23:10.15 | p3nguin | What's the difference? |
| 23:10.16 | FlaPer87 | gtg guys |
| 23:10.30 | nny | oh sorry that's whats running in the background p3nguin |
| 23:10.34 | nny | same thing afaik |
| 23:10.57 | p3nguin | Okay, so the fop2.tgz is the correct software to run on the * server system? |
| 23:11.18 | nny | p3nguin: yes |
| 23:11.27 | nny | p3nguin: afaik, untar and "make install" |
| 23:11.50 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 23:11.57 | p3nguin | I just looked, and that's not "source" at all. |
| 23:12.29 | nny | p3nguin: sorry I haven't poured through it yet, I guess it's just the files that are distributed |
| 23:12.32 | p3nguin | It's just a tar archive version instead of deb or rpm. |
| 23:12.47 | p3nguin | (which is what I wanted, anyway) |
| 23:12.49 | zaoul | you see a directory named 'passwords'.. to ls or not to ls? |
| 23:13.04 | p3nguin | I'll see what I can do with it. |
| 23:13.26 | nny | yeah let me nkow, sorry I am not that familiar with it yet |
| 23:13.30 | nny | know* |
| 23:13.46 | p3nguin | I don't run apache on the * system, so looks like it's going to require some additional work first. |
| 23:15.33 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 23:26.05 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 23:26.44 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 23:32.47 | p3nguin | nny: I had to email them about getting the correct tarball for my glibc, so I'll have to put the project on hold until I get a reply. |
| 23:39.10 | nny | p3nguin: cool let me know how it goes |
| 23:45.35 | p3nguin | nny: You might have to remind me. |
