| 00:14.46 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 00:33.14 | *** join/#asterisk jetlag (~jetlag@pool-71-168-201-186.cmdnnj.east.verizon.net) |
| 00:34.07 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 00:52.49 | volga629 | Is any body know about this error app_voicemail_imapstorage.c: Couldn't find mailbox 2000 in context device |
| 00:53.48 | volga629 | It just filling the log and for this user voice mail disabled |
| 01:12.37 | autofsckk | hello, im trying to make a call, but my phone sounds like if i have already called somebody and the line is busy, that is without making any call |
| 01:17.13 | autofsckk | well there isnt a busy tone, its more like 6 short beeps and then i can call |
| 01:17.56 | autofsckk | i know its because a mailbox thing |
| 01:24.58 | volga629 | What format for tlsclientmethod |
| 01:32.16 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:07.44 | autofsckk | i have a friend connected with x-lite from a mac, im not getting audio, but she can hear me? what can that be fixed? she is behing a router |
| 02:15.04 | ChannelZ | her return audio is either not making it out of her router or is not making it to you, for any number of reasons |
| 02:16.40 | ChannelZ | Is your Asterisk behind a firewall? |
| 02:19.33 | autofsckk | not |
| 02:19.36 | autofsckk | no sorry |
| 02:19.44 | autofsckk | she is behind a router |
| 02:21.51 | ChannelZ | Is her peer set to nat=yes in sip.conf? |
| 02:22.16 | ChannelZ | or actually nevermind the audio is working in that direction |
| 02:23.45 | autofsckk | yes it is |
| 02:23.58 | autofsckk | she is using x-lite 4 with a mac |
| 02:24.25 | ChannelZ | Her packets either aren't making it out of her router or are going off into space |
| 02:24.29 | autofsckk | the thing is i have tested with fing for example with a very similar kind of network and it was working excellent |
| 02:24.54 | autofsckk | i tried to debug from the cli while in the call, and i didnt see any kind of error |
| 02:26.10 | ChannelZ | You wouldn't necessarily see any kind of err |
| 02:26.22 | ChannelZ | The device you're talking on, is it behind NAT? |
| 02:27.07 | autofsckk | yes |
| 02:27.14 | autofsckk | through a pap2 |
| 02:27.47 | autofsckk | but it works great with all kinds of calls, with land lines, cellular, other extensions |
| 02:28.39 | ChannelZ | It can be any number of problems. Hard to guess without seeing a sip debug of the call |
| 02:28.56 | autofsckk | i see, wel thanks |
| 02:29.00 | autofsckk | i have to o out now |
| 02:29.11 | ChannelZ | both ends have to be aware of their REAL IPs, or know that the other side is lying in order to send media to the right place. |
| 02:29.16 | autofsckk | ill try to debug it later and look for some help if i cant do it :D thanks a lot ChannelZ |
| 02:29.28 | ChannelZ | It also depends on if your Asterisk is in the media stream or not. |
| 02:39.56 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 02:42.21 | p3nguin | WTF is wrong with this silly thing... |
| 02:42.37 | p3nguin | WARNING[25523]: ast_expr2.fl:415 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end; Input: + 1 ^ |
| 02:42.51 | p3nguin | The ^ points to the + |
| 02:43.01 | p3nguin | exten => s,n,Set(DB(callerid/${CALLERID(num)}/lookups)=${IF($[${DB_EXISTS(callerid/${CALLERID(num)}/lookups)}]?$[${DB(callerid/${CALLERID(num)}/lookups)} + 1]:1)}); |
| 02:43.37 | p3nguin | It is supposed to record the number of times a CID number is looked up locally. |
| 02:43.58 | p3nguin | Set the db entry to the value of... |
| 02:44.13 | p3nguin | if the key already exists, set it to the existing value plus 1, or... |
| 02:44.31 | p3nguin | if the key does not exist, set it to 1. |
| 02:45.30 | p3nguin | The syntax looks perfectly good to me, but asterisk 1.4.42 doesn't seem to agree. |
| 02:45.36 | p3nguin | Thoughts? Anyone? |
| 02:50.10 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-170-021.sc.res.rr.com) |
| 02:52.06 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:56.50 | p3nguin | And 1.8 thinks it's wrong, too, but I just don't see where the mistake is. |
| 02:57.10 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001744.dsl.bell.ca) |
| 02:57.17 | dijib | come one come all |
| 02:57.31 | dijib | its the greatest show on earth |
| 02:57.55 | dijib | welcome to the magnificent |
| 02:58.04 | dijib | the wonderfull |
| 02:58.14 | dijib | the phono |
| 02:58.17 | dijib | i dont even know |
| 02:58.21 | dijib | cuz im not home. |
| 02:58.23 | dijib | #asterisk |
| 02:58.39 | ChannelZ | Right then. |
| 02:58.40 | dijib | this place is a graveyard |
| 02:58.59 | dijib | why am i such a noob, i ponder always |
| 02:59.27 | dijib | howcome i cant set proper routes on cent |
| 02:59.31 | dijib | why am i so useless |
| 02:59.44 | dijib | why #asterisk gods why |
| 03:00.12 | ChannelZ | Well you could start by making some sense? :) |
| 03:00.20 | dijib | never shall i good sir |
| 03:00.21 | dijib | never |
| 03:00.38 | ChannelZ | p3nguin: remove the spaces around the + |
| 03:00.41 | dijib | i still have a sangoma card sitting here im never going to use |
| 03:01.15 | dijib | im building a new server |
| 03:01.23 | dijib | but its going to be my main serving box. |
| 03:01.32 | dijib | leaded with daemons |
| 03:01.36 | dijib | loaded |
| 03:01.40 | dijib | what do i need? |
| 03:01.46 | dijib | how do i configure bind? |
| 03:01.48 | dijib | named |
| 03:01.57 | ChannelZ | how long is a piece of string? |
| 03:02.06 | dijib | as long as you wrap it? |
| 03:02.23 | ChannelZ | exactly. 'how do I configure bind' is like that. |
| 03:02.26 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 03:02.33 | ChannelZ | It's not a quick answer |
| 03:02.41 | dijib | { |
| 03:02.42 | dijib | } |
| 03:03.15 | ChannelZ | You make a zone file and load it through the named.conf. There. |
| 03:03.35 | p3nguin | I tried removing the spaces earlier and it didn't seem to make any change, but I will do it once more just to make sure. |
| 03:03.43 | dijib | i think im missing my zone |
| 03:04.07 | dijib | where does zone reside? |
| 03:04.18 | ChannelZ | Wherever they reside. |
| 03:04.23 | ChannelZ | Start by reading named.conf |
| 03:05.09 | ChannelZ | Your distro might pre-configure some additional configs you can add to, and/or you can look for a 'directory' statement which specifies where the zones typically live |
| 03:05.32 | ChannelZ | But it's not automatic, you also have to add the zone to the config |
| 03:05.47 | dijib | p3nguin: i still have this sangoma fxo card ... what do>? |
| 03:06.10 | ChannelZ | but you can load them from anywhere |
| 03:06.41 | *** join/#asterisk ZenKen (~ZenKen@99-62-61-118.lightspeed.toldoh.sbcglobal.net) |
| 03:06.48 | dijib | ive set y zone in named.conf ut im missing zonefile |
| 03:06.50 | dijib | i think |
| 03:06.55 | ChannelZ | p3nguin: your line works here, under 10 anyway, by removing the spaces |
| 03:06.56 | p3nguin | It looks like it cleared up the error on 1.8, but it still complains on 1.4. |
| 03:07.31 | dijib | also y routes are mangled as im replacing a box that pppoes so the ip was temp-tested changed to that box's |
| 03:07.38 | p3nguin | Now it complains about "+1" rather than "+ 1" |
| 03:10.20 | ZenKen | Hi everyone, quick question for you guys about SIP: I'm running an asterisk box behind a NAT, and I'm aware that I have to forward ports for it to be able to accept incoming calls. The one thing that bothers me though is: I also have a hardware SIP phone sitting here. If I register my SIPGate account on my SIP phone, why can my SIP phone accept incoming calls without any port forwards? Thanks |
| 03:11.48 | ChannelZ | There are various tricks you can do with UDP to allow an incoming connection through a firewall.. |
| 03:12.44 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:12.50 | ChannelZ | For instance you send a dummy packet out on the IP on the port number you really want to receive traffic back from them from. This makes the NAT router expect the traffic that comes from them and maps it to the correct LAN IP |
| 03:13.06 | ChannelZ | s/out on the IP/out to the IP/ |
| 03:13.26 | ChannelZ | Your phone could be doing this |
| 03:13.51 | ZenKen | ChannelZ: Can I implement those tricks in Asterisk too? I'm not really fond of opening up 5060 in addition to a range of ~1000 ports to the WAN. |
| 03:13.53 | *** join/#asterisk koffel (~whome@173-167-212-105-ip-static.hfc.comcastbusiness.net) |
| 03:13.56 | ChannelZ | This is the same method by which Skype for instance is able to communicate through firewalls correctly without having to setup forwards |
| 03:14.12 | koffel | has anyone used sipstation before |
| 03:14.25 | ChannelZ | You don't need 1000 ports |
| 03:14.45 | dijib | how amany realistically do u need there? |
| 03:14.52 | ChannelZ | If you only plan to have 10 simultaneous calls for instance, do 10 plus a few spares |
| 03:15.06 | ChannelZ | specify the range in rtp.conf |
| 03:15.19 | p3nguin | For 10 calls, you'll want like 40 ports. |
| 03:15.25 | ZenKen | ChannelZ: I was just referring to the range that asterisk has set for RTP |
| 03:15.33 | ChannelZ | ZenKen: so was I |
| 03:15.43 | ZenKen | ChannelZ: OK, I'll try that then |
| 03:16.00 | ChannelZ | the port range can be configured in rtp.conf - which is the range that Asterisk will request the remote end to send their media to. |
| 03:16.34 | dijib | where did i configure centos routes? |
| 03:16.36 | ChannelZ | You have no control over what the remote end will ask YOU to send your media to them on, so your firewall pretty much needs to allow all outgoing traffic |
| 03:16.39 | dijib | ive lost my way |
| 03:17.03 | ZenKen | ChannelZ: Alright, I've got a better understanding of that now. Thanks for explaining how my SIP Phone handles it! |
| 03:17.36 | dijib | me2 im going to lessen my open ports |
| 03:17.47 | ChannelZ | I thought there used to be some STUN settings in Asterisk a long time ago which can assist in things of that nature, but I've never used them, not sure if they even exist |
| 03:18.05 | ChannelZ | (I seem to recall somewhere along the way reading that they were never properly implemented and got removed maybe..hmm) |
| 03:18.46 | ZenKen | Interesting... Do you remember what version you were running? (I'm currently running 1.4) |
| 03:19.03 | ChannelZ | I started with 1.4 so it would have been that |
| 03:19.33 | ZenKen | I guess I'll have to do some digging in the docs |
| 03:20.25 | ChannelZ | It's honestly just easier to port forward |
| 03:21.03 | ZenKen | It is much easier, but I'm a pedantic perfectionist :D |
| 03:22.36 | ChannelZ | I see a note in the v10 config, "...Asterisk *always* uses 'symmetric RTP' mode as defined by RFC 4961; Asterisk will always send RTP packets from the same port number it expect to receive them on." |
| 03:23.11 | ChannelZ | Which would pretty much do what I described. Dunno if that applies to 1.4 or not... where did I put the sample configs? |
| 03:23.51 | ChannelZ | Is this something you've tried and isn't working or are you pre-barking up the tree? |
| 03:24.10 | p3nguin | Okay, another piece of evidence has arisen. After the first call where it actually sets the key to 1, the error goes away. I think I'll add another command to check if it exists and set it to 1 on the first pass if it does not exist... FIRST. Then I bet this command won't complain about it. |
| 03:24.38 | p3nguin | I think it is showing the error because it is trying to evaluate what the value would be if it set an empty value + 1. |
| 03:24.54 | p3nguin | null + 1 = error |
| 03:24.57 | ChannelZ | Yeah it's probably the 'if DB exists' returning a NULL or something and getting misinterpreted |
| 03:25.04 | p3nguin | Exactly. |
| 03:25.25 | p3nguin | I couldn't see that before, but now that I have thought about it, it makes sense. |
| 03:25.27 | ZenKen | ChannelZ: I've got everything working with the normal port fowards, but the fact that my SIP phone was able to do it without forwards was drilling at me. |
| 03:25.53 | ChannelZ | ZenKen: well Asterisk might very well work without the port forwards for RTP. You'd still want 5060 though. |
| 03:26.50 | ZenKen | ChannelZ: I think I'll disable my RTP forwards and do some tests, since you mention that it always practices symmetric RTP. |
| 03:28.10 | ChannelZ | Well, as I said that is mentioned in the Asterisk 10 and 1.8 configs, not sure if it's been in there all along or got added somewhere else along the way. But it's worth a test. Or look at your 1.4 dist sip.conf and see if it mentions RFC 4961 |
| 03:29.43 | ChannelZ | I used to keep copies of the dist configs for each major version around but apparently I got bored with that cuz they aren't where I remember putting them |
| 03:30.16 | ZenKen | ChannelZ: Ah, I thought you included 1.4 as well, I don't see any mention in my sip.conf for RFC 4961, but I'll give it a try anyway. |
| 03:31.55 | ChannelZ | is this a new setup? |
| 03:33.22 | ZenKen | ChannelZ: Pretty much, it's a clean install on a MIPS linux system |
| 03:33.53 | ChannelZ | just wondering why you went with 1.4, quite old |
| 03:33.55 | *** join/#asterisk mefistofeles (~Tathar@unaffiliated/mefistofeles) |
| 03:33.59 | *** part/#asterisk mefistofeles (~Tathar@unaffiliated/mefistofeles) |
| 03:34.25 | *** join/#asterisk mefistofeles (~Tathar@unaffiliated/mefistofeles) |
| 03:34.43 | *** join/#asterisk Resilience (fran@212.97.186.171.dyn.user.ono.com) |
| 03:34.46 | *** part/#asterisk mefistofeles (~Tathar@unaffiliated/mefistofeles) |
| 03:35.34 | ZenKen | ChannelZ: To be exact it's running on a DD-WRT router, I went with the first version I could pull from optware |
| 03:35.54 | ChannelZ | oh. |
| 03:37.17 | ZenKen | ChannelZ: I thought, the earlier the version, the less resources it should use (blatant assumption although) |
| 03:37.50 | ZenKen | Afterall, 680 MHz and 128 MB RAM isn't exactly top of the line. |
| 03:40.36 | Resilience | I have some questions about the "why2 of the cost of pci cards and networks appliances for ATA's, is this the right channel to ask it? |
| 03:41.35 | ChannelZ | Well, we're bored. But I don't understand the question. |
| 03:43.01 | p3nguin | Okay, once I wrapped my head around that error, the fix was pretty easy. Two commands instead of one: check if it exists, initialize to 0 if it does not; then set to existing value + 1. |
| 03:43.20 | *** join/#asterisk jetlag (~jetlag@pool-71-168-201-186.cmdnnj.east.verizon.net) |
| 03:43.28 | *** join/#asterisk radic (~radic@dslb-094-216-234-100.pools.arcor-ip.net) |
| 03:43.42 | *** join/#asterisk DMBoyCloud (~cloud@201.191.198.28) |
| 03:43.47 | Resilience | ChannelZ, a Cisco SMB SPA8000 8-Port IP Telephony Gateway costs about 2hundred euros for 8 ports and a wahtever you like from http://www.voiplink.com/OpenVox_Cards_Asterisk_FXS_FXO_T1_E1_Linux_VoIP_s/112.htm) costs much more for one port, I don't understand why, is there anything I do't see here? |
| 03:46.08 | *** part/#asterisk DMBoyCloud (~cloud@201.191.198.28) |
| 03:47.40 | ChannelZ | I don't know why ATAs are so expensive but they all seem to be |
| 03:48.19 | ChannelZ | I don't know anything about the SPA8000 |
| 03:50.07 | Resilience | ChannelZ, ATA's don't seem so expensive to me, but a pci card + one fxs and one FXO seemsm to expensive for me, is it like an ATA, right? |
| 03:51.16 | ChannelZ | Yeah. |
| 03:51.45 | p3nguin | Cards are expensive. An SPA-3102 (one FXS and one FXO) isn't so bad. |
| 03:51.48 | Resilience | ChannelZ, and it has an enclusore lees, and a power system less.. but it costs sommething like a 3foldl than an external AtA.. seems ilogical to me xD |
| 03:52.24 | Resilience | p3nguin, but why a card with just one FXS and ONE FXO costs a lot more than an 8 port FXO appliance? I don't understand... |
| 03:52.37 | p3nguin | It's different technology. |
| 03:55.34 | Resilience | p3nguin, iwhat'as the difference? I don't see, I only see less components in the pci card |
| 03:56.08 | Resilience | (i'm trying to uderstand) |
| 03:58.52 | ChannelZ | That SPA8000 I think is the cheapest per-port ATA I've ever seen. Can't speak for the quality of it |
| 04:00.16 | Resilience | ChannelZ, well, it is a Cisco product, I don't think it be bad ;D |
| 04:00.31 | Resilience | (by the way, FXS ports are prots to the phone terminals, right?) |
| 04:00.36 | ChannelZ | Honestly they are hit-and-miss |
| 04:01.01 | ChannelZ | FXS are what you'd plug an analog phone into. FXO is what you'd plug a phone line from your telco into |
| 04:01.28 | Resilience | ok |
| 04:02.52 | Resilience | ChannelZ, right, so this appliance would be somethin like an 8 terminal card, right? I still don't understand why an ATA pci card is so damn expensive (compared with an external ethernet ATA) |
| 04:03.40 | Resilience | isn't really there an inexpensive ATA in the form of a pci card? |
| 04:04.04 | ChannelZ | Not comparitively, no. Maybe chineese clones are a bit cheaper |
| 04:04.16 | Resilience | ChannelZ, I know that CXisco is a hit-and-miss for all ofo its line of products ;D |
| 04:04.35 | ChannelZ | The PCI card is more direct, in the sense that they appear as first-class channels in Asterisk. There will probably be less latency but maybe not enough for you to care. |
| 04:05.08 | Resilience | ChannelZ, I try to get out of ochinese goods the most I can.. when the y have workers instead of slaves AND when they manufacture half as good as german products, then I will check chinese products ;D |
| 04:05.59 | p3nguin | A card is a completely different interface. It becomes part of the system. An ATA is just another phone on the IP network. |
| 04:06.59 | Resilience | ChannelZ, that's a good point for marketing, not for physical good, for my, one FXS + ONE FXO port card is just for not having another box in my desk (another box, another wallmart plug, more cabling, more dust, etc) |
| 04:07.50 | Resilience | p3nguin, I understand that, but from th epoint of view of real costs ti does not add o rests anything to the cost of manufacturing (if something, it lessens its costs) |
| 04:09.58 | ChannelZ | I choose not to try and justify it. There. |
| 04:11.00 | Resilience | ChannelZ, ha ha haha, really what I was asking is if there was anything more than marketing reasons, if I was missing something in the technology |
| 04:12.37 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 04:15.44 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 04:15.51 | ChannelZ | *shrug* |
| 04:16.34 | Resilience | ChannelZ, what does it mean? |
| 04:16.44 | ChannelZ | "I don't know" |
| 04:16.51 | Resilience | ah ok |
| 04:16.55 | Resilience | me neither xD |
| 04:18.08 | Resilience | p3nguin, when you say that a pci crd is 2part of the system", what do youo mean? isn't it like another phone? I'm thinking in phones (lines) as ports.. I thought that the interface to asterisk was that, lines/phones, is not right? |
| 04:19.13 | ChannelZ | as I'd said, a card manifests its self to Asterisk as channels. That SPA actually just talks SIP over the network to Asterisk, just like a SIP phone. |
| 04:19.56 | ChannelZ | Channels are more tightly integrated into Asterisk for lack of a better description, but it's not necessarily more advantageous either |
| 04:20.34 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 04:21.05 | Resilience | ChannelZ, mm I thought there would someway to make it show like something like and FXS/FXO port (that's what I thought, not based on anything real) |
| 04:21.30 | ChannelZ | With the SPA8000? |
| 04:22.01 | ChannelZ | FXO/FXS has no meaning to SIP |
| 04:22.02 | Resilience | yes (i'm new at voip, I was searching for somehardware just to try) |
| 04:22.30 | Resilience | I understand that FXO/FXS has no meaning to SIP (it was.. some way to talk, sorry) |
| 04:22.36 | ChannelZ | Actually it doesn't have any meaning to Asterisk either even with a PCI-based ATA, except for the actual hardware configuration |
| 04:23.23 | Resilience | I understand that |
| 04:24.15 | Resilience | but I thought that an external AtA could be managed as FXS/FXO channels, like its internal pci counterpart, I was wrong |
| 04:24.23 | ChannelZ | Well speaking strictly of VoIP, you don't need any hardware. You could do SIP over the internet right now. |
| 04:24.49 | ChannelZ | Well the FXO/FXS is abstracted by the hardware. From Asterisk's point of view, it doesn't actually care. It's just "a channel" |
| 04:25.35 | Resilience | I understand that, but obviously, I woul dlike to tinker a bit with POTS phone terminals ;D |
| 04:26.35 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 04:28.26 | ChannelZ | Yeah just get an SPA3102 |
| 04:29.38 | Resilience | ChannelZ, but wiil it work the same as a pci card from th epoint of view of configuring asterisk or another pbx? or won't? |
| 04:29.57 | ChannelZ | No it's totally different but the end result is more or less the same |
| 04:30.45 | Resilience | the end result I understand is the same |
| 04:31.20 | Resilience | I think that, fro the poit of view of configuring asterisk, it is another SIP terminal more, which is different from an FXO+FXS ports, right? |
| 04:31.37 | ChannelZ | Yes |
| 04:34.15 | p3nguin | If you only want to use your POTS line and phone, the 3102 is good enough to do that. If you want to configure specifically the channels of the dahdi interface, you'll need the card. |
| 04:34.26 | Resilience | ChannelZ, that happens becuase the firmare in the SPA "exports" the FXS/FXO as SIP "channels", and not another "anything", so asterisk can only manage what is provided by the SPA, that is, the SIP "channels", right? |
| 04:34.40 | ChannelZ | essentially yes |
| 04:35.27 | Resilience | p3nguin, I would like to tinker a bit with FXS/FXO channels, that was the point (my point). I would like to have both things, use my POTS lines/phones + tinker a bit |
| 04:35.32 | p3nguin | If you can configure asterisk with a SIP phone, you can configure asterisk with an ATA. |
| 04:35.48 | Resilience | ChannelZ, ooook, thank you, you have been very supportive and patient with me |
| 04:36.07 | p3nguin | For FXS/FXO configuration in asterisk, an ATA will be of no use. |
| 04:36.26 | Resilience | p3nguin, and configuring it with a SIP/ATA is the same as configuring it with FXS/FXO ports? or isn't? |
| 04:36.30 | ChannelZ | but the point is that it doesn't even matter. |
| 04:37.05 | p3nguin | Configuring SIP in Asterisk is totally different from configuring Dahdi hardware in Asterisk. |
| 04:37.06 | Resilience | ChannelZ, I understand that from the point of view of use/user it does not matter, becuase it works externally the same |
| 04:37.23 | Resilience | p3nguin, sorry but, what's dahdi? |
| 04:37.46 | ChannelZ | DAHDI talks directly to hardware devices (PCI cards..) |
| 04:37.47 | p3nguin | Dahdi is the interface driver that you will probably use with your PCI card. |
| 04:38.08 | p3nguin | An ATA's interface is Ethernet. |
| 04:38.17 | Resilience | ah, so it is the "drivers"? the interface to the filesystem? |
| 04:39.01 | ChannelZ | sort of |
| 04:39.22 | ChannelZ | it's hardware drivers AND an SDK through which Asterisk communicates to the hardware |
| 04:39.46 | p3nguin | The driver is the connection between the hardware and the kernel. |
| 04:39.47 | Resilience | kind of protocol to comuunicate over the device of the filesystem, right? |
| 04:40.15 | ChannelZ | filesystem isn't the right word per se but yes |
| 04:40.16 | p3nguin | kernel, not file system |
| 04:40.42 | p3nguin | You don't format your PCI card with ext3. :/ |
| 04:40.53 | ChannelZ | but in Linux, you can try |
| 04:40.54 | Resilience | I was meaning the character/stream dive showed in the filesystem |
| 04:40.57 | *** join/#asterisk moy_ (~moy@bas5-toronto47-2925109168.dsl.bell.ca) |
| 04:41.21 | Resilience | dive*device |
| 04:41.23 | Resilience | xD |
| 04:41.53 | p3nguin | The kernel will provide an interface to the hardware through /proc in Linux. |
| 04:43.32 | p3nguin | Without the dahdi driver, that device node will not work. It may not even exist without dahdi. |
| 04:43.36 | Resilience | I tought the drivers would export the device to some entry in the /dev part of the filesystem, soemthing like... /dev/telephony/cardXXX/fxsXXX and /fxoXXX (something like serial lines or sound channels) |
| 04:44.15 | ChannelZ | yes /proc is more of a status |
| 04:44.50 | p3nguin | Oh. Perhaps I was thinking of checking some bit of info on the card rather than controlling it. |
| 04:45.33 | ChannelZ | DAHDI is both a set of hardware (kernel) drivers which talk to specific pieces of hardware (Digium/clone TDM cards of various kinds) and an interface by which Asterisk can talk to any piece of hardware with DAHDI drivers. |
| 04:45.52 | p3nguin | I don't use hardware like that, anyway. Around here, it's all VoIP all the time. |
| 04:45.53 | ChannelZ | It's a HAL |
| 04:46.13 | Resilience | p3nguin, yes, I understood you we rethinking about the inards od driving the card, i was thinking about using that card, as I unxderstand dahdi is the "protocol2 which makes the middleware between the high level orders os asterisk (ring, call, cut) and how it is done really by the physical card, right? |
| 04:46.29 | Resilience | chaneright (I called it middleware, sorry) |
| 04:47.02 | p3nguin | If you know what a sound card driver does, you should know what dahdi does. |
| 04:47.05 | Resilience | ChannelZ, and that dahdi interfaces don't show FXS/FXO channels the same as a "SIP channel"? or am I saying soemthing complete misleaded? |
| 04:47.24 | ChannelZ | No that's correct. |
| 04:47.33 | p3nguin | SIP is completely unrelated to FXS/FXO. |
| 04:47.38 | ChannelZ | But channels in Asterisk are (mostly) treated the same, they just have different names. |
| 04:47.54 | Resilience | p3nguin, yes, in fact, I was thinking of FXS/FXO cards as soundcars, or better, as serial interfaces (like rs323, rs485) and asterisk like a giant multiplexer os rs2322 channels over tcp/ethernet.. right? |
| 04:47.58 | ChannelZ | There are DAHDI channels, SIP channels, IAX channels, and any number of other types. |
| 04:48.45 | Resilience | ChannelZ, so in fact, the only difference between using an xternal ATA and an internal pcicard.. is configurint the card, but iti will somethin not ver difeerent? or very verru different? |
| 04:49.07 | p3nguin | With an ATA, you configure only SIP for the device. |
| 04:49.31 | ChannelZ | well.. there's some that can do TDMoE but let's not go there |
| 04:49.33 | p3nguin | The ATA is like another IP phone on Ethernet. |
| 04:49.45 | Resilience | I understand that, lets call it.. mmmm ATAFXS and ATAFXO (though they are not fxs and fxo) |
| 04:50.05 | p3nguin | How about no, let's don't call it that. |
| 04:50.09 | Resilience | the asterisk configuration file would the same like the oe with an internal crd an channels PICFXS and PCIFXO, right? |
| 04:50.12 | p3nguin | You'll just confuse yourself. |
| 04:50.18 | Resilience | no, listen |
| 04:50.21 | p3nguin | No, stop. |
| 04:50.25 | Resilience | listen*read (he) |
| 04:50.32 | Resilience | ok, I'll hear, tell me |
| 04:50.33 | p3nguin | You're confusing yourself. |
| 04:50.47 | p3nguin | The ATA is only SIP. Forget about the FXS/FXO. |
| 04:50.55 | Resilience | I'm trying to udnerstand what hardware do I want to buy |
| 04:51.07 | Resilience | I know the ATA is only SIP, I understand that |
| 04:51.08 | ChannelZ | It's the difference between doing Dial(SIP/line1/phonenumber) and Dial(DAHDI/1/phonenumber) |
| 04:51.20 | p3nguin | If you are using the ATA, you do not configure the FXS/FXO at all in asterisk. |
| 04:51.37 | p3nguin | You only configure another SIP device. |
| 04:51.46 | ChannelZ | It's a different configuration for the channel, but the rest of the way they are treated more or less the same. |
| 04:51.53 | Resilience | but, I configure the SIP channel in and the SIP channel out, or they are the same? maybe taht's my error? |
| 04:52.00 | Resilience | aha |
| 04:52.18 | Resilience | so tehre si the SIP channel, which isi bidirectional, not "two channels,2 like the FXS/FXO? |
| 04:52.24 | p3nguin | There is no sip channel in and sip channel out. There is only a sip DEVICE, which is configured in sip.conf. |
| 04:52.36 | p3nguin | You don't explicitly configure sip CHANNELs. |
| 04:52.41 | p3nguin | Only devices. |
| 04:52.45 | Resilience | and the FXS/FXO devices would be confiured in... where? |
| 04:52.50 | ChannelZ | The SPA |
| 04:53.15 | Resilience | chaneyes, for the ATA devic, but fot he internal card? they wuld configured in waht file? |
| 04:53.19 | Resilience | what* |
| 04:53.25 | ChannelZ | the DAHDI config |
| 04:53.40 | p3nguin | chan_dahdi.conf? |
| 04:53.41 | ChannelZ | But none of this is particularly important |
| 04:53.48 | Resilience | ok, different file from sip.conf, right? |
| 04:53.57 | hurdman | system.conf |
| 04:53.59 | ChannelZ | yes |
| 04:54.19 | p3nguin | chan_dahdi.conf/system.conf you configure dahdi CHANNELs. sip.conf, you configure sip DEVICES. |
| 04:54.34 | Resilience | s, if I want to tinker and configure a system with FXS and FXO channels, I need to buy the oci cards, like oro not, right? |
| 04:54.40 | ChannelZ | NO |
| 04:54.52 | ChannelZ | Just buy an SPA3102 |
| 04:54.59 | Resilience | for myself yes |
| 04:55.25 | ChannelZ | It has an FXO and an FXS port. You configure various attributes related to them on the SPA its self, through its built-in web server. |
| 04:55.36 | ChannelZ | The SPA talks to Asterisk over Ethernet, using SIP. |
| 04:55.38 | Resilience | but, if I have.. 2 FXO lines and (oet's say) ( FXS internal lines.. I can't do it with ATA's.. I would need to make it with FXS/FXO devices, right? |
| 04:55.50 | p3nguin | You could use two SPA-3102s. |
| 04:56.09 | p3nguin | And configure another SIP entry in sip.conf. |
| 04:56.21 | Resilience | p3nguin, and the configuration would be similar with two ATA's than to pci cards? |
| 04:56.25 | p3nguin | Nope. |
| 04:56.35 | Resilience | ah, that was the point |
| 04:56.42 | p3nguin | There is no similarity between a sip device configuration and a dahdi channel configuration. |
| 04:56.53 | Resilience | would it be more simillar to having some kind of.. pbx appliances, right? |
| 04:57.07 | ChannelZ | Who cares! |
| 04:57.14 | Resilience | me cares xD |
| 04:57.33 | p3nguin | If you need a bunch of FXO ports, get the appropriate ATA. Or buy the appropriate card. |
| 04:57.53 | p3nguin | Asterisk does not care HOW they interface. It speaks DAHDI and SIP. |
| 04:58.24 | Resilience | p3nguin, is form, personal, I only wanted to tinker a bit, I don't want to expend too much money, I'm erh, searching to better job right now ;D |
| 04:58.58 | ChannelZ | Which is why we've answered several times over, JUST GET AN SPA3102 |
| 04:59.05 | p3nguin | You'll spend more money on a card than you'll spend on a 3102. |
| 04:59.20 | Resilience | I understand that, that was clear to me since the beggining |
| 04:59.35 | Resilience | begining* |
| 04:59.36 | p3nguin | I think good cards are $250 or more. A 3102 is $65-$70. |
| 04:59.47 | Resilience | begining? or beggining? |
| 04:59.58 | p3nguin | beginning |
| 04:59.59 | Resilience | p3nguin, yes, that's the range of prices |
| 05:00.07 | Resilience | oh, double nn, xD |
| 05:00.13 | Resilience | thaNNk you |
| 05:04.14 | Resilience | I tink I will start with one or two SPA's and time will tell |
| 05:28.27 | *** part/#asterisk TheCops (~mdb@72.55.132.180) |
| 06:04.34 | ChannelZ | .....aaaannnnndd scene. |
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| 08:48.02 | ChannelZ | wow |
| 08:48.34 | dym | wow what? |
| 08:49.21 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:54.10 | ChannelZ | Place went dead |
| 09:02.00 | dym | ohwell |
| 09:02.37 | ChannelZ | I worked out, took a shower, watched an entire movie AND TV show between words |
| 09:06.56 | dym | Why congratulations! |
| 09:06.58 | dym | (: |
| 09:15.57 | ChannelZ | Ohhhh and a timewarp! |
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| 09:53.12 | *** join/#asterisk PbxMan (~user1@213.37.35.26.dyn.user.ono.com) |
| 09:54.31 | PbxMan | hello Is anybody familiar with any call center management apps for asterisk, with the usual stuff, campaigns, skills, treatments, etc...??? |
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| 10:08.11 | brian98 | pbxman - queuemetrics I guess is good |
| 10:08.23 | brian98 | orderlq orderly stats are an other option |
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| 10:14.08 | PbxMan | do you have any experience working with them brian98 ? |
| 10:14.21 | brian98 | a little with the orderly and stats |
| 10:14.29 | PbxMan | what do you think of them? |
| 10:14.32 | brian98 | good price, good support, easy to install and maintain once setup |
| 10:14.34 | brian98 | hoever |
| 10:14.37 | brian98 | however even |
| 10:14.57 | brian98 | the market leader in uk/ireland and parts of europe is queuemetrics |
| 10:15.07 | brian98 | it is VERY powerful |
| 10:15.17 | brian98 | but has a quite steep learning curve |
| 10:15.31 | brian98 | I've installed it for 2 customers who had experience of it themselves |
| 10:15.31 | PbxMan | is there an Opensource solution? |
| 10:16.05 | brian98 | as in free? |
| 10:16.30 | PbxMan | well at least as in "free" to make my own code touch ups |
| 10:17.07 | brian98 | not being smart but |
| 10:17.07 | brian98 | http://www.google.com/search?client=safari&rls=en&q=opensource+call+center+management&ie=UTF-8&oe=UTF-8#hl=en&client=safari&rls=en&sa=X&ei=dHtcT--uGsHMhAeV4smoBA&ved=0CEMQBSgA&q=open+source+call+center+management&spell=1&bav=on.2,or.r_gc.r_pw.r_qf.,cf.osb&fp=a8d18c00ad240132&biw=1342&bih=1004 |
| 10:19.05 | PbxMan | OK thank you brian98 you've saved me some time doing research |
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| 11:06.23 | omashhour | HI question, how can I solve the nat issue when using directrtpsetup=yes |
| 11:06.27 | omashhour | Thanks in advance |
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| 11:22.44 | Resilience | ChannelZ, are you stil there? I have one more question O :o) |
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| 12:52.53 | vlt | Hello. Between two asterisk 1.6 machines I set the codec to gsm. It works but the bandwidth used is 28.5 kbps (in each direction). Any idea why it is that high? |
| 12:55.12 | coppice | why would you expect it to be any lower? |
| 12:59.55 | vlt | coppice: Because astbook says that gsm operates at 13 kbps: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-3.html |
| 13:00.37 | coppice | add RTP overheads to that, and you have your answer |
| 13:02.39 | vlt | coppice: Ok, thank you. |
| 13:05.14 | omashhour | HI question, how can I solve the nat issue when using directrtpsetup=yes |
| 13:05.45 | omashhour | when user A calls user B on the same wifi network they can hear each other |
| 13:06.02 | omashhour | when caller A is on 3g and caller B is on wifi I can't hear any audio |
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| 13:36.52 | davlefou | i, |
| 13:38.09 | davlefou | when i use register on my iax trunk, i was rejected! |
| 13:38.48 | vlt | coppice: Can I reduce overhead between the two asterisks a little? Maybe by using IAX2 instead of SIP? |
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| 13:44.47 | davlefou | it seems have twice register, one with the = et an old =>? |
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| 14:11.29 | coppice | vlt: using a larger packetisation time will reduce the overhead |
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| 14:15.30 | mutex7c | Hello everybody, is it ok to ask some digium hardware related questions in this channel - it is asterisk related. |
| 14:15.46 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 14:20.14 | mutex7c | No objections at all - so here is my question: Is it possible to simulate a T1 or E1 connection to a legacy phone system which is currently connected to a real T1 or E1 PRI? I don't know if there is anything like a TN Mode for PRI ... |
| 14:21.24 | mutex7c | In order to detach the existing T1/E1 PRI from the system, hook it up to an asterisk and connect the existing phone system to a VoIP trunk - is that possible? |
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| 15:26.53 | [sr] | i'm sleepy! |
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| 16:21.01 | *** join/#asterisk benlangfeld (~Adium@unaffiliated/benlangfeld) |
| 16:22.40 | benlangfeld | Hello gents, I'm having a problem with the AMI Bridge command on Asterisk 10 (not sure if the same issue exists on 1.8). It seems that once two channels are bridged, if one hangs up, the other channel is also hung up (also firing a SoftHangupRequest event). Is there a way I can prevent this? |
| 16:28.05 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
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| 16:36.18 | benlangfeld | I'm seeing this happening: https://gist.github.com/85d1137c39730809f05e |
| 16:36.46 | oli_004 | Hi. I'm trying to get the Fax for Asterisk addon with T.38 over sip to work. However I could not receive any fax until now. T.38 is negotiated successfully every time and asterisk even says that it can send packages over the sip channel. But after 12-13 sec, the sending fax machine hangs up, without having sent anything. Does anyone know this? |
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| 16:39.13 | oli_004 | btw. I'm using asterisk 10.2.0 and the latest fax for asterisk |
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| 16:41.16 | dijib | hello? |
| 16:41.21 | dijib | im connected? |
| 16:41.31 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 16:41.48 | dijib | ok. ip's are up. |
| 16:42.00 | dijib | try to ping 10.0.0.1 nothing |
| 16:42.20 | dijib | im even reciving a dhcp address on one of the NICs |
| 16:42.25 | dijib | what am i doing wrong? |
| 16:43.03 | dijib | even have the 0.0.0.0 10.0.0.1 0.0.0.0 UG 0 0 0 ethX entry in route |
| 16:43.06 | dijib | whats my issue? |
| 16:43.24 | dijib | have named installed ut i believe malconfigured |
| 16:43.32 | dijib | but |
| 16:43.43 | [TK]D-Fender | oli_004: I don't see your AGI dialing any devices or calling a FAX app. What are you expecting? |
| 16:48.54 | oli_004 | [TK]D-Fender: hmm. I'm sorry, I do not use an AGI script. Did you maybe mean benlangenfeld's github output? |
| 16:49.30 | [TK]D-Fender | oops |
| 16:49.33 | [TK]D-Fender | bad aim... |
| 16:49.38 | oli_004 | ;-) |
| 16:49.50 | [TK]D-Fender | oli_004: You're right ... for your issue... we see NOTHING :p |
| 16:51.28 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 16:53.26 | oli_004 | [TK]D-Fender: you're right :-). I'll paste the debug output to gist.github. one moment |
| 17:00.26 | oli_004 | [TK]D-Fender: https://gist.github.com/2017121 |
| 17:08.41 | volga629 | Is this correct syntax for AGI "AGI(agi://127.0.0.1/findConference?conference=${EXTEN})" ? |
| 17:08.47 | volga629 | asterisk 10.2 |
| 17:23.38 | [TK]D-Fender | For a FastAGI, it could be |
| 17:24.13 | [TK]D-Fender | Question is... do you ahve a script listening on the right port... |
| 17:24.19 | volga629 | Yes |
| 17:24.41 | [TK]D-Fender | Well, it could be fine... |
| 17:25.30 | volga629 | I am trying set custom konference app syntax look ok, but still missing some thing that will make work properly |
| 17:25.58 | volga629 | it is app_konference.so |
| 17:27.05 | volga629 | May I post dial plan section ? |
| 17:31.22 | volga629 | http://pastebin.com/AjEhGib1 |
| 17:32.19 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:41.36 | [TK]D-Fender | exten => s,n,GotoIf($[${EXTEN} = ${CONFERENCE_FOUND}]?valid:invalid) <------ BROKEN |
| 17:45.30 | p3nguin | What's wrong with it? |
| 17:46.11 | p3nguin | If the conference found is 's', it would go to valid, if the conference found is not 's', it would go to invalid. |
| 17:46.48 | volga629 | that I can't get where is mistake |
| 17:47.05 | p3nguin | EXTEN is always the current exten. In that case, it is always 's'. |
| 17:47.06 | [TK]D-Fender | Why on earth would a conference be named "S"? There is no point in doing a variable to variable comparison when one side will never be different |
| 17:50.13 | volga629 | so it should i and on exit t ? |
| 17:50.57 | p3nguin | That doesn't really make much sense, either. |
| 17:51.30 | [TK]D-Fender | volga629: You don't see to understad what you are even comparing |
| 17:52.13 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:56.33 | volga629 | I am still learning about dial plans |
| 17:57.55 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:02.25 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 18:03.23 | volga629 | that why I asking GURUs that I can understand the right way |
| 18:04.16 | [TK]D-Fender | volga629: Well you now see what it is really comparing and you need to think about what you should actually be comparing.... |
| 18:09.27 | volga629 | Ues, that true I just want to know the konference app |
| 18:09.33 | volga629 | Yes |
| 18:10.16 | p3nguin | I doubt more than two people here even know what the "konference app" is. We all use MeetMe and/or ConfBridge. |
| 18:12.36 | volga629 | all difference that konference not depend on PRI |
| 18:17.45 | p3nguin | Your statement does not compute. |
| 18:17.57 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 18:20.51 | [sr] | who wants to buy me lunch? |
| 18:23.18 | volga629 | So what best way create dynamic conference ? |
| 18:23.43 | ChannelZ | Disco music! |
| 18:29.41 | [TK]D-Fender | volga629: NONE of *'s conferencing options requires a PRI |
| 18:33.37 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.31.66) |
| 18:42.53 | *** join/#asterisk buffyinca (~zack@31-192-198-241.customer.universal.se) |
| 18:44.03 | buffyinca | How doesn the consumer pstn end of the line signal the telco that one has hangup? My SPA3000 doesn't release the line until the remote party hangsup. |
| 18:46.54 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 18:49.24 | drfreeze | I'm looking how to provision the GigasetS675 IP as a SIP client. Anyone have instructions on how to do that? |
| 18:51.46 | ChannelZ | buffyinca: Usually a polarity reversal but it depends on the country |
| 18:54.01 | drfreeze | anyone? Beuller. Beuller |
| 18:54.08 | ChannelZ | Are you using the SPA3000 to do an analog phone to VoIP? Or are you using a VoIP phone through it to your analog phone line? |
| 18:55.51 | drfreeze | want to use the phone as a SIP ext |
| 18:56.02 | buffyinca | ChannelZ: Does the phone or in this case the spa3000 also do the pol reversal? I'm using a softphone to call out through the spa3000 (pstn) |
| 18:56.56 | ChannelZ | Well the SPA should in this case. The softphone terminates the call when you hang up by a completely different means, and the SPA should hang up the POTS line. |
| 18:57.07 | buffyinca | ChannalZ: Right now I'm thinking that the spa3000 must have a way (a tone perhaps) to tell the telco that it needs to revers the polarity. |
| 18:57.40 | dijib | can someone help me with a networking issue? |
| 18:58.18 | ChannelZ | ~ask |
| 18:58.18 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:58.19 | drfreeze | dijib: shoot |
| 18:58.50 | dijib | centos |
| 18:58.53 | buffyinca | ChannalZ: on hook voltage is ~ -50V, on hook is ~ 7V, but if I hangup the softphone the line is 50V postive until the remote end hangsup. |
| 18:58.55 | dijib | have 2nics |
| 18:59.14 | dijib | one statis 10.0.0.5 subnet correct the other dhcp client thats recives address |
| 18:59.41 | dijib | resolve.conf = search lan ; nameserver 10.0.0.1 |
| 18:59.57 | dijib | bind installed. zones setup correctly everything points to 10.0.0.1 |
| 19:00.17 | dijib | routes have 0.0.0.0 10.0.0.1 0.0.0.0 UG entry |
| 19:00.21 | dijib | cant ping |
| 19:00.30 | dijib | ???? |
| 19:00.47 | drfreeze | what? |
| 19:01.03 | dijib | statis = static |
| 19:01.09 | dijib | i cant ping anywhere |
| 19:01.16 | dijib | but im getting a dhcp'ed address |
| 19:01.21 | ChannelZ | Your other nic isn't listed |
| 19:01.44 | ChannelZ | You should have a default route to your ISP |
| 19:01.44 | drfreeze | what does ping 10.0.0.1 give |
| 19:01.49 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 19:01.51 | dijib | oth are listed and configured approprient in ifconfig |
| 19:02.07 | dijib | just takes 103XXXms and 100% lost |
| 19:02.10 | dijib | loss |
| 19:02.52 | dijib | shit its my firewall. i just stoppped iptables im pinging |
| 19:02.55 | ChannelZ | buffyinca: well there are various settings under 'PSTN Line' in the SPA for disconnect detection but I'm not sure what any of them should be for your setup |
| 19:04.03 | ChannelZ | but I'm not sure if they affect how the SPA hangs up its self. |
| 19:05.03 | buffyinca | ChannalZ: Aren't they just for detection? I don't see where I would setup what the spa3000 is to send out trouh the line |
| 19:05.15 | [sr] | people, |
| 19:05.24 | [sr] | i'm accessing my server from the outside, everything fine |
| 19:05.32 | [sr] | rtp ports defined, and OK |
| 19:05.45 | [sr] | but when i use an FXO card i have, no sound is transmited |
| 19:06.06 | [sr] | if i use the BRI or SIP functions, everything works as expected |
| 19:06.21 | buffyinca | ChannalZ: The spa3000's infopage says it has hangup but the line is still not terminated |
| 19:10.54 | *** join/#asterisk timahvo1 (~rogue@197.176.191.214) |
| 19:26.18 | vlt | Hello. A few hours ago I asked whether I can reduce overhead between the two asterisks a little. There were 28.5 kbps for each gsm stream (13 kbps) on the line. Maybe by using IAX2 instead of SIP? Someone answered using a larger packetisation time will reduce the overhead. |
| 19:26.36 | vlt | Where can I control this packetisation time value? |
| 19:37.44 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 19:56.59 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 20:32.37 | ChannelZ | gsm IS 13kbps |
| 20:33.46 | ChannelZ | If you have a lot of calls going between the servers, using an IAX2 link with trunking can be more effecient than SIP but increasing the packet size increases latency |
| 20:38.03 | vlt | ChannelZ: Thank you. Yes, gsm is 13 kbps but I didn’t expect more than twice as much being used on the line. I tried g729 now (8 kbps) which results in 22 kbps. |
| 20:39.00 | vlt | Will trunking be anebled by default when switching to IAX? |
| 20:39.06 | vlt | *enabled |
| 20:39.28 | p3nguin | Use IAX2 with trunking mode enabled. |
| 20:39.29 | vlt | And how to increase the packet size? |
| 20:39.51 | p3nguin | Set the codec to something like g729:30. |
| 20:40.22 | p3nguin | To enable trunking, you trunk=yes in the peer entry. |
| 20:41.46 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 20:42.38 | vlt | p3nguin: Thank you! |
| 20:51.36 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 20:54.23 | ChannelZ | vlt well remember that audio goes in both directions... |
| 21:03.49 | [TK]D-Fender | vlt: How many simultaneous calls do you calculate you'll be wanting between these 2 servers? |
| 21:08.40 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 21:27.09 | vlt | [TK]D-Fender: max. six simulaneous calls on a 256 kbps wire |
| 21:29.30 | vlt | With IAX and trunking it seems to work at last. |
| 21:31.49 | [TK]D-Fender | vlt: Packetization will save you about 7kbps per channel outside of IAX trunk mode. In IAX2 trunk mode it will save you about 7kbps period. IAX2 trunk mode itself will save you 105kbps |
| 21:32.27 | [TK]D-Fender | frankly that 7kbps of changing the frame size at the risk of added latency seems pointless |
| 21:32.40 | [TK]D-Fender | VS just using IAX2 trunk mode as-is |
| 21:33.31 | vlt | [TK]D-Fender: Yes. |
| 21:34.04 | vlt | [TK]D-Fender: What I saved by trunking should be enough for that connection. Thanks again. |
| 21:34.42 | [TK]D-Fender | vlt: GSM = 6 * 13 + 21 |
| 21:34.59 | [TK]D-Fender | 99kbps |
| 21:35.13 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-170-021.sc.res.rr.com) |
| 21:36.16 | vlt | [TK]D-Fender: g729 = 6 * 8 + 21, right? |
| 21:36.34 | [TK]D-Fender | 9.6 |
| 21:50.46 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 21:55.14 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 22:08.52 | *** join/#asterisk jnxa (~xvd@seven.medozas.de) |
| 22:10.36 | jnxa | According to http://tinyurl.com/7g5yldj , /usr/sbin/asterisk shall be owned by the asterisk user. That sort of makes the book untrustworthy, and I wonder what _really_ needs to be writable by asterisk. |
| 22:11.02 | Kobaz | doesn't have to be owned by asterisk at all |
| 22:11.10 | jnxa | What about the other directories? |
| 22:11.28 | Kobaz | most stuff can be owned by root, and non-writable to asterisk |
| 22:11.41 | Kobaz | /var/spool/asterisk should be asterisk:asterisk |
| 22:11.58 | Kobaz | and /etc/asterisk/voicemail.conf should be asterisk:asterisk |
| 22:12.24 | Kobaz | and /var/lib/asterisk/astdb should be asterisk:asterisk |
| 22:12.25 | Kobaz | that's about it |
| 22:19.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:21.52 | *** join/#asterisk keepguessing (7aacad8c@gateway/web/freenode/ip.122.172.173.140) |
| 22:22.12 | keepguessing | I am just looking for #idespinner |
| 22:22.49 | keepguessing | idespinner* new him long time ago .. Do not know if he is here. Found on irc that he could be here ... |
| 22:23.07 | keepguessing | If you guys know him please let me know |
| 22:25.04 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 22:25.19 | ChrisInSydney | g'day all |
| 22:25.54 | keepguessing | p3nguin: Hi the logs here show that you know him. http://ibot.rikers.org/%23asterisk/20111102.html.gz Any idea on where he can be found? |
| 22:26.40 | keepguessing | p3nguin: Sorry I meant it shows that u had some conversation with him. Just enquiring to see if u know him :) |
| 22:28.44 | *** join/#asterisk wonderworld (~ww@dsdf-4db53703.pool.mediaWays.net) |
| 22:32.14 | dym | ChrisInSydney: ellow there |
| 22:32.40 | ChrisInSydney | hi, Good talking, albeit briefly on Saturday / Friday night |
| 22:32.51 | dym | yeah, my thoughts exactly. |
| 22:33.14 | ChrisInSydney | I stayed connected until around 2:30pm my time. Some random south african dropped in via a Blink client |
| 22:33.22 | dym | oh, okay |
| 22:33.24 | ChrisInSydney | wanted to know how hge could make free calls using the service |
| 22:33.31 | dym | (: |
| 22:33.31 | ChrisInSydney | funny |
| 22:33.43 | ChrisInSydney | sounded a bit dodgy if you ask me |
| 22:33.51 | dym | free calls - no way! (: |
| 22:34.18 | ChrisInSydney | yep. He was saying how he "helps" old aged people and people in mental hospitals with their conputer problems |
| 22:34.26 | ChrisInSydney | hmmmm. |
| 22:34.34 | ChrisInSydney | I added one and one together |
| 22:34.38 | ChrisInSydney | came up with 2 |
| 22:34.42 | dym | by getting free calls via a sip bridge |
| 22:34.46 | dym | i see what he was doing there... |
| 22:35.08 | ChrisInSydney | yeah. I said he could get other clients to connect to sip2sip and then he could call / help them for free |
| 22:35.39 | ChrisInSydney | but his, get this, his username, im serious, on sip2sip.info, I am not kidding, was..... |
| 22:35.42 | ChrisInSydney | suspense |
| 22:35.46 | dym | :D |
| 22:35.49 | dym | lovely |
| 22:35.53 | ChrisInSydney | microsoft@sip2sip.info |
| 22:36.01 | dym | lol |
| 22:36.02 | ChrisInSydney | wow |
| 22:36.03 | dym | :D |
| 22:36.04 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 22:36.13 | dym | decent enough |
| 22:36.16 | ChrisInSydney | thats when alarms went off |
| 22:36.21 | ChrisInSydney | really went off |
| 22:36.28 | dym | yeah, i can imagine |
| 22:37.31 | ChrisInSydney | still, he didn't say that he had the sum of TWELVE MILLION FIVE HUNDRED THOUSAND DOLLARS in US currancy he needed to transfer to investment in my couyntry |
| 22:37.45 | dym | did he not |
| 22:37.48 | dym | how surprising |
| 22:37.49 | ChrisInSydney | nahhh |
| 22:38.06 | ChrisInSydney | next time maybe I might me bor elucky ;D |
| 22:38.15 | ChrisInSydney | might be more lucky |
| 22:38.18 | ChrisInSydney | cant type |
| 22:38.21 | ChrisInSydney | today |
| 22:39.05 | dym | :D |
| 22:39.12 | dym | cut the odd substances :D |
| 22:39.23 | ChrisInSydney | too much caffene |
| 22:39.27 | ChrisInSydney | monday mornin meetings |
| 22:39.40 | dym | ah, whats the time over there? |
| 22:39.46 | ChrisInSydney | 9:39am |
| 22:39.54 | dym | 11:40pm here |
| 22:40.05 | ChrisInSydney | been going since 6:05 when I got in the car |
| 22:40.13 | dym | going what? |
| 22:40.17 | dym | caffeine? |
| 22:40.18 | ChrisInSydney | thats about the time I called it quits last nighht |
| 22:40.40 | ChrisInSydney | yup |
| 22:40.44 | dym | right |
| 22:42.47 | ChannelZ | eh? |
| 22:44.03 | dym | hum? |
| 22:50.38 | ChrisInSydney | arghhh. Customers. Want to know prices in advance |
| 22:50.49 | jnxa | Give them one. |
| 22:50.57 | dym | randomize one |
| 22:51.01 | ChrisInSydney | cant they just say. "Chris, do this for me and send me a bill. We trust you" |
| 22:51.14 | dym | ... and then you wake up? |
| 22:51.16 | dym | (: |
| 22:51.24 | ChrisInSydney | true, |
| 22:52.16 | ChrisInSydney | this is 15 handsets in Sydney and 15 in Melbourne. Nothing fancy. BRI + a 10Mb EFM for SIP |
| 22:52.21 | ChrisInSydney | 10Mb is shared with data |
| 22:52.38 | dym | sounds okay |
| 22:53.09 | ChrisInSydney | yeh. about 2 days each site + 2 days setup and management |
| 22:53.19 | ChrisInSydney | bit of hardware and we are sweet |
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| 23:25.04 | ChrisInSydney | out to see customers |
| 23:25.07 | ChrisInSydney | c yaz |
| 23:25.20 | dym | tc |
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