| 00:00.19 | ManxPower | Sgeo: depends on what you call "call processing". |
| 00:00.29 | Carlos_PHX | What is your definition of call processing, and why are you asking? |
| 00:00.50 | lesouvage | Sgeo: I suppose that the sip protocol is classified in the session layer |
| 00:00.52 | [TK]D-Fender | ManxPower: its just like food processing.. except this bull is much harder to swallow ;) |
| 00:02.14 | Sgeo | I'm asking because I'm trying to figure out what layers cdmaOne uses. I know it's on 1 and 2, and a third layer related to call processing (at least according to Wikipedia) |
| 00:02.30 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 00:03.18 | lesouvage | Sgeo: see http://devcentral.f5.com/weblogs/images/devcentral_f5_com/weblogs/macvittie/125/o_osi-model-7-layers.png for a nice picture |
| 00:03.26 | Sgeo | According to Wikipedia, "IS-95 is widely described as a three-layer stack, where L1 corresponds to the physical (PHY) layer, L2 refers to the Media Access Control (MAC) and Link-Access Control (LAC) sublayers, and L3 to the call-processing state machine." |
| 00:03.27 | ManxPower | Sgeo: real world stuff tends not to fit very well into the OSI model 8=) |
| 00:03.47 | Sgeo | knows what the OSI model is |
| 00:06.45 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 00:06.58 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ed9cab03995b843d) |
| 00:08.56 | lesouvage | Sgeo: Maybe this picture is helpful http://www.protocols.com/pbook/images/cellular-protocols.gif |
| 00:10.45 | Sgeo | ty, but I don't see IS-95 or CDMA on there |
| 00:14.44 | Sgeo | Bye all, and thanks for the help |
| 00:16.59 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 00:18.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:32.51 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 00:33.50 | FruitBasket | How can I make it so that a phone can only receive one call? Our support techs don't like getting beeped in the ear. I've set call limit 1, which works, but they can't transfer calls... because they aren't allowed to call out since they've reached the limit.. |
| 00:34.18 | FruitBasket | might the aastras have such an option? |
| 00:35.32 | [TK]D-Fender | FruitBasket: No, for the same reasons. |
| 00:35.44 | FruitBasket | ... no what? |
| 00:35.51 | [TK]D-Fender | FruitBasket: If you don't want them gettign beeped then check if they are on the phone beofre passing them the call. |
| 00:36.01 | FruitBasket | how can I do that? |
| 00:36.10 | drmessano | turn off call waiting? |
| 00:36.16 | FruitBasket | hmm. |
| 00:36.19 | [TK]D-Fender | FruitBasket: "no" you probably won't find a phone based feature for this |
| 00:36.31 | [TK]D-Fender | FruitBasket: jsut check in the dialplan before calling. |
| 00:36.33 | FruitBasket | fender: then, what are the reasons you're referring to? |
| 00:36.40 | FruitBasket | doesn't know how to check in the dialplan |
| 00:36.51 | drmessano | disable call waiting on the phone |
| 00:37.13 | FruitBasket | drmessano: ... :-D |
| 00:38.11 | [TK]D-Fender | FruitBasket: "core show application chanisavail" |
| 00:39.05 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:39.28 | FruitBasket | fender: thanks :-) |
| 00:40.36 | *** join/#asterisk synchris (n=synchris@athedsl-154703.home.otenet.gr) |
| 00:40.37 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 00:44.53 | grndslm | anybody here used ooma? |
| 00:45.25 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 00:45.44 | grndslm | from what i understand... they use linux & asterisk on their ooma boxes |
| 00:46.10 | grndslm | i want to pay $230 for their hub & scout boxes... but not if they're not gonna be around in another year or so |
| 00:46.49 | *** part/#asterisk `paul (n=admin@125.252.70.126) |
| 00:48.03 | hardwire | anybody measuring sip qualifications off to rrd's? |
| 00:49.40 | FruitBasket | does anyone ever get odd, seemingly random disconnected calls? Maybe for one in 50, someone will come up to me and say, "I just got disconnected in the middle of a sales call." It's not always the same people, it's with different phones, we have a T1 with great service -- low/no packet loss. I've tried two servers, different hardware, processor, ram, hard drives, motherboard, I've tried a datacenter colo, our in house T1, differen |
| 00:49.40 | FruitBasket | t service providers.. |
| 00:49.50 | hardwire | fruity |
| 00:49.59 | FruitBasket | really. What's up? I can't see _anything_ except Asterisk that would cause it. Or is voip just inherently... unreliable? |
| 00:50.36 | Carlos_PHX | Yes, VoIP is entirely unusable. |
| 00:50.38 | Carlos_PHX | Doesn't work. |
| 00:50.57 | Carlos_PHX | FruitBasket: Who is your ITSP? |
| 00:51.08 | FruitBasket | most of the disconnections are remote, not the local phones. But sometimes it is the phones here.. and those people are _sure_ they didn't hit the wrong buttons. I ask the provider and they either tell me that the PSTN network reported a hangup (on reconnect no one knows what happens).. |
| 00:51.13 | FruitBasket | we use Vitelity and NexVortex. |
| 00:51.42 | stencil | Hello, if i wanted the best sounding audio codecs and bandwidth wasn't an issue, what is my best option? |
| 00:51.47 | FruitBasket | and we've had similar issues with both, as far as I can tell... perhaps more with one than the other, I suppose I should start tracking that. |
| 00:51.54 | FruitBasket | stencil: ulaw |
| 00:52.05 | FruitBasket | stencil: or g729.. but that's patented. |
| 00:52.19 | [TK]D-Fender | stencil: ulaw/alaw |
| 00:52.28 | [TK]D-Fender | G.729 is way down the list |
| 00:52.39 | stencil | thanks guys |
| 00:52.53 | FruitBasket | but.. even our phones are on a physically separate network. Nowhere do they mingle with computers. The routers have changed. Everything is different, and the same problems persist. |
| 00:52.59 | FruitBasket | I just don't get it... |
| 00:53.09 | Carlos_PHX | stencil: ulaw |
| 00:53.20 | Carlos_PHX | FruitBasket: Your experience is not the norm. |
| 00:53.32 | Carlos_PHX | Something is breaking in an unusual way. |
| 00:53.35 | FruitBasket | carlos: would you think it's the provider? I can't change anything else. :-| |
| 00:53.43 | Carlos_PHX | Who is your ISP? |
| 00:53.47 | Carlos_PHX | You said it is T1? |
| 00:53.56 | Carlos_PHX | How is voice quality? Is that good, but you just drop calls? |
| 00:54.02 | FruitBasket | carlos: we've gone with a colo datacenter and we have a T1. I've used both. |
| 00:54.20 | FruitBasket | Voice quality is pretty good, but today one person was telling me of a few seconds voice loss every 10 minutes or so |
| 00:54.22 | Carlos_PHX | Colo...your server is in colo? |
| 00:54.32 | FruitBasket | it was, yeah. It's not now. |
| 00:54.47 | Carlos_PHX | Interesting. Who is the ISP? What router do you use on the T1? Or is the router provided by the ISP? |
| 00:54.51 | FruitBasket | Asterisk sometimes "forgets" to hang up calls -- the phone is obviously put down, Vitelity recorded the call as a 5.7 minute call... I had to do a softhangup in Asterisk, and the call was recorded as 2 hours. |
| 00:55.16 | Carlos_PHX | That part may or may not be relevant, it can indicate that the "hang up" packet was lost. |
| 00:55.21 | FruitBasket | I honestly can't figure it out and it's really starting to bother me. |
| 00:55.22 | Carlos_PHX | Have you done packet loss testing? |
| 00:55.45 | FruitBasket | carlos: they would have to both be lost; the one from the phone, the one from the upstream provider, but not the audio of the last seconds before the hangup. It's weird. |
| 00:55.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 00:55.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:56.00 | Carlos_PHX | Hang up is just one packet. |
| 00:56.06 | FruitBasket | I haven't done much testing on the T1.. |
| 00:56.10 | FruitBasket | carlos: one for each side. |
| 00:56.12 | Carlos_PHX | Can you answer my other questions on the ISP? |
| 00:56.15 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 00:56.30 | FruitBasket | ISP is Qwest, or Bresnan, or Lariat... we've tried all the local ones. |
| 00:56.33 | Carlos_PHX | Also, what is your network/telephony background like? |
| 00:56.49 | Carlos_PHX | So you have had the same issue on multiple ISPs, all T1? |
| 00:57.07 | FruitBasket | mine personally? network is pretty good; I served as the network guy for the building of a router recently. Telephony not so much, but I've learned quite a bit inthe last year of working with Asterisk. |
| 00:57.46 | Carlos_PHX | I would look to the networking on this first. Packet loss in particular. |
| 00:57.54 | FruitBasket | Carlos: only one T1. The other was business class cable internet, which they told me prioritized VoIP. The T1 is quest, with very low packet loss. The Lariat is wireless which was supposed to provide prioritized voip and dedicated bandwidth.. but sucked. |
| 00:58.09 | Carlos_PHX | All of these had the same call drops? |
| 00:58.17 | FruitBasket | yes. |
| 00:58.22 | Carlos_PHX | Wow |
| 00:58.27 | Carlos_PHX | What phones? |
| 00:58.29 | FruitBasket | carlos: I was indeed questioning that. I suppose I'll set up some more pings from inner network to router/server/voip provider. |
| 00:58.39 | *** part/#asterisk km2 (n=x@mobile-166-217-255-036.mycingular.net) |
| 00:58.42 | *** join/#asterisk km2 (n=x@mobile-166-217-255-036.mycingular.net) |
| 00:58.45 | FruitBasket | aastra 57i's, aastra something else, grandstream 1200's, 2020's, polycom's... |
| 00:58.50 | Carlos_PHX | In my experience, call drops are very rare. |
| 00:58.57 | Carlos_PHX | All the phones have the same problem? All brands? |
| 00:58.59 | FruitBasket | hmm. |
| 00:59.06 | FruitBasket | everything. Always. |
| 00:59.13 | Carlos_PHX | We are much more likely to have people complain of quality, not drops. |
| 00:59.20 | FruitBasket | maybe 1 in 50 just drop somewhere in the middle. |
| 00:59.24 | Carlos_PHX | WOW |
| 00:59.27 | FruitBasket | and that can't be one small packet loss. |
| 00:59.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-18daf2617f927bb8) |
| 00:59.35 | FruitBasket | and I _know_ there isn't major packet loss. |
| 00:59.39 | Carlos_PHX | One packet lost wouldn't drop the call. |
| 00:59.46 | Carlos_PHX | Actually a lot of packet loss wouldn't. |
| 00:59.49 | FruitBasket | so.. I have to think it's Asterisk.. |
| 00:59.58 | Carlos_PHX | A call dropped while voice is passing has to be signalled. |
| 00:59.58 | FruitBasket | but I .. just don't have anything to go on. |
| 01:00.05 | FruitBasket | yeah. |
| 01:00.18 | Carlos_PHX | Asterisk or the ITSP, but you said you used two, both did the same thing? |
| 01:00.26 | FruitBasket | Usually it's the provider end that is marked: exited non-zero in SIP/vitelity-blah blah |
| 01:00.27 | Carlos_PHX | I know Vitelity, good network. Don't know the other. |
| 01:00.46 | FruitBasket | Nexvortex says they're business grade. Most of our calls come from Vitelity. |
| 01:00.58 | Carlos_PHX | Yes, everyone says that. |
| 01:01.09 | FruitBasket | Most of the dropped calls I've looked into are Vitelity, too. Some are nexvortex.. fewer, but that may be just volume. |
| 01:01.29 | Carlos_PHX | Do you have the ability to fire up another Asterisk box? |
| 01:01.41 | FruitBasket | carlos: I've done that. At both physical locations. |
| 01:01.45 | FruitBasket | same issues. |
| 01:01.46 | Carlos_PHX | Wow |
| 01:01.58 | Carlos_PHX | So the only thing left is your LAN. |
| 01:02.03 | FruitBasket | at this point.. the only thing I can really say is voip is inherently unreliable.. |
| 01:02.19 | CrazyTux | FruitBasket: I have no reliability issues |
| 01:02.24 | FruitBasket | carlos: wrong. We've tried the phones on the computer side. We've run a physically separate network strictly for the phones. Same issues. |
| 01:02.25 | Carlos_PHX | Yeah, but we run tens of thousands of calls/day without these issues. |
| 01:02.29 | CrazyTux | You know what they say... it's not the car... it's the driver. |
| 01:03.08 | FruitBasket | we've tried a couple different routers.. but not recently. Currently it's a FreeBSD box routing the calls using PacketFilter for NAT. |
| 01:03.22 | FruitBasket | UDP connections are set to disconnect after 5 minutes.. but that's without traffic. |
| 01:03.44 | Carlos_PHX | FruitBasket: I'm going to say this in the most helpful way possible. I do not mean to be an asshole about it, so hope it doesn't sound that way. This is not an inherent issue so you should continue to look for local sources of the problem. |
| 01:03.52 | FruitBasket | I can't see any reason an active call would be dropped. Some calls last hours, others are dropped within minutes. |
| 01:04.06 | Carlos_PHX | That's pretty wild. |
| 01:04.15 | Carlos_PHX | How many calls/day do you make, and drop? |
| 01:04.20 | FruitBasket | carlos: I am. I'm checking cabling, suspecting maybe they're not plugged into the back of the server tight, changing switches, it's mostly Netgear switches.. but still. |
| 01:04.33 | Carlos_PHX | Most consumer switches are just fine. |
| 01:04.43 | FruitBasket | uhm.. drop, maybe 2-10, depending on the day. Calls, 1-200 I would imagine. |
| 01:04.56 | Carlos_PHX | What version of Asterisk? |
| 01:05.29 | FruitBasket | it's been 1.4.18 until last night. I upgraded to 1.4.22 last night, but there were a couple dropped calls today. I think there was a fix for the non-hung-up-calls, though, somewhere around .19 |
| 01:05.29 | Carlos_PHX | Any pattern between oubound/inbound calls? |
| 01:05.33 | Carlos_PHX | Which are dropped? |
| 01:06.15 | Carlos_PHX | I assume there's nothing odd/old about the machine? Something reasonable? |
| 01:06.15 | FruitBasket | uhm.. at the colo, it would sometimes drop _all_ calls at the same time, but that was infrequent. Usually it says it's disconnected from the provider, but sometimes it says the local phone hung up -- where the local user says he's sure it wasn't him. |
| 01:06.28 | FruitBasket | newish intel, I think one was a xeon quad-core. |
| 01:06.38 | FruitBasket | Tested ram, no errors in dmesg, no filesystem errors.. |
| 01:06.44 | FruitBasket | is stumped. |
| 01:06.49 | FruitBasket | It's really beyond me. |
| 01:06.50 | Carlos_PHX | So server is in colo, phones use a T1 to connect to internet? |
| 01:07.07 | Carlos_PHX | Have you tried the server in another location? |
| 01:07.37 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 01:07.42 | FruitBasket | carlos: that was the last configuration, on 1.4.18. Now, phones connect to the server over LAN (but still through the router -- only one NIC on the server :-/) and the server interacts with the provider through the T1 |
| 01:08.23 | FruitBasket | it's connected directly to the internet, static IP, basic firewall; phones connect to the server through the external IP, but from the same subnet, so it's not actually going over the internet. |
| 01:08.32 | Carlos_PHX | So the router has three ports or more, with the server, LAN, T1 on their own ports. |
| 01:08.38 | Carlos_PHX | Ah |
| 01:09.09 | FruitBasket | we haven't been fully utilizing bandwidth.. |
| 01:09.30 | Carlos_PHX | Any pattern to inbound/outbound being dropped? |
| 01:09.34 | FruitBasket | I should run more pings, I guess. I really don't know of any other way to check for packet loss. I wish I could listen for BYE's, but I'd be overwhelmed getting them on every call. |
| 01:10.00 | FruitBasket | carlos: I'm unable to identify one, but I admit I really haven't tried for numbers vs locations. I should start recording all numbers, directions, etc I guess. |
| 01:10.20 | FruitBasket | wishes there was an Asterisk log parser so he could rip out the full dial plan of a single call |
| 01:10.36 | Carlos_PHX | That's crazy stuff. |
| 01:10.45 | FruitBasket | sip debug -- impractical; I can't apply it to every call. But is there a chance I can get the last message that ended each call? |
| 01:11.05 | FruitBasket | mock sip logging.. I don't like the sole message being "blah blah exited non-zero in macro dialextension" |
| 01:11.20 | FruitBasket | thanks for listening/suggesting, btw.. |
| 01:11.30 | FruitBasket | I need.. more information. But I'm not sure how to get it. |
| 01:11.33 | Carlos_PHX | NP, intriguing issue. |
| 01:11.48 | Carlos_PHX | What I do is give each user a log sheet to make notes of issues. |
| 01:11.55 | Carlos_PHX | Sometimes you find patterns that way. |
| 01:11.56 | FruitBasket | maybe I should modify chan_sip so it produces extra logging info, logging the sip messages maybe. |
| 01:11.57 | FruitBasket | hmm. |
| 01:12.16 | Carlos_PHX | Modifying a channel driver has its own risk, so you'd muddy the waters. |
| 01:12.29 | FruitBasket | arright, I'll keep that in mind. Even if they write down _just_ a time, I can do a lot more.. usually they just say, "I had a dropped call.. bout 45 minutes ago." == 4500 lines ago. |
| 01:12.51 | Carlos_PHX | 1-2 days of GOOD logging can tell you a lot. |
| 01:12.56 | Carlos_PHX | If the users comply. |
| 01:12.56 | FruitBasket | carlos: well, it'd be mostly file writing/logging calls. It might, but it would give me information relating to what I need. Any extra issues and I'd change it back pretty quick. |
| 01:12.59 | FruitBasket | yeah :-) |
| 01:13.02 | FruitBasket | ohhh, that logging :-) |
| 01:13.12 | FruitBasket | is gonna make that up tonight. |
| 01:13.47 | Carlos_PHX | In your case I'd look for time, direction, length of call when dropped, and the number dialed or at least NPANXX |
| 01:14.05 | FruitBasket | I can get all that from the logs :-P |
| 01:14.05 | Carlos_PHX | Oh, also, did the phone hang up, or did the sound just disappear. |
| 01:14.25 | Carlos_PHX | True, so you could just go with time. |
| 01:14.38 | FruitBasket | it's _usually_ but not always reported as a hangup, I believe. _sometimes_ they say the sound just disappeared... maybe 50/50 when I hear of it. |
| 01:15.01 | FruitBasket | I do core verbose 9 logging; would the extra memory/file access cause delays that could do something? any idea? |
| 01:15.05 | *** join/#asterisk Archide (n=Justieve@cpe-69-204-5-149.buffalo.res.rr.com) |
| 01:15.09 | Archide | Evenin |
| 01:15.17 | FruitBasket | or does every asterisk write full.blah.asterisk file which includes every dialplan step? |
| 01:15.18 | Carlos_PHX | Sounds like you know the difference there. Lost sound could be a number of things, but if the phone hangs up it was told to. |
| 01:15.36 | Carlos_PHX | I haven't done that level of logging, don't know. |
| 01:15.41 | FruitBasket | right. That's interesting. It would suggest that Vitelity is the odd party out, though. |
| 01:15.46 | Carlos_PHX | Try it and check load. |
| 01:15.51 | Carlos_PHX | What does load normally wrong? |
| 01:15.54 | Carlos_PHX | run? |
| 01:16.00 | FruitBasket | huh. |
| 01:16.04 | FruitBasket | load? run? |
| 01:16.12 | FruitBasket | ohhh right. |
| 01:16.14 | Carlos_PHX | Sorry, what does load normally run? |
| 01:16.31 | Carlos_PHX | I'd be open to giving you an account on our network just to try out for a day, rule out Vitelity. |
| 01:16.35 | FruitBasket | uhm.. gawd, I was looking with one or two calls, just saw the CPU at 99% idle. Disk/load.. don't recall :-) |
| 01:16.39 | FruitBasket | huh. |
| 01:16.39 | Carlos_PHX | Assuming it's just US traffic. |
| 01:17.01 | Carlos_PHX | Yeah, figured load would be low, but thought I'd ask. |
| 01:17.10 | FruitBasket | carlos: yep :-) I might take you up on that at some point.. but not sure. It'd be a little awkward. For outbound calls, you mean, or routing inbound through it too? |
| 01:17.20 | Carlos_PHX | Just outbound as a test. |
| 01:17.26 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:17.31 | Carlos_PHX | Could do inbound, Vitelity lets you forward your numbers. |
| 01:17.39 | FruitBasket | phones are off for the last hour, so can't check load. Usually 4 concurrent calls. |
| 01:18.01 | FruitBasket | huh.. we're mostly inbound, though. I'll need to do some of the logging first. |
| 01:18.03 | Carlos_PHX | From my perspective, I'm both curious about the problem all around, plus we're probably going to sign a contract with Vitelity for some wholesale carriage, so I'm interested all around. |
| 01:18.12 | FruitBasket | mm. |
| 01:18.24 | Archide | win win |
| 01:18.54 | FruitBasket | They've been pretty open with me. On about three of the ones I've asked about, each time I was told that they received a hangup from their upstream provider. On the last one, they said they'd need to be logging the sip messages to tell exactly what happened. |
| 01:19.03 | Carlos_PHX | So if you want to give it a shot, shouldn't be terribly painful. Even just try with a few DIDs or something. I think Vitelity charges you 1.5cpm on forwards. |
| 01:19.14 | FruitBasket | Obviously they have more information than me by default; I'd like to know if it was a hangup or what caused the disconnection. |
| 01:19.31 | Carlos_PHX | Right. I'd be very curious whether we'd see the same. |
| 01:19.36 | FruitBasket | ohhh, that. I could set up a subaccount and register such things ;-) Cost isn't so much a concern, especially for a couple days. |
| 01:19.42 | Carlos_PHX | If we put you on our PRIs, then that's one less variable. |
| 01:19.49 | FruitBasket | ahh, true. |
| 01:19.53 | Carlos_PHX | No charge for a few days testing. |
| 01:20.04 | FruitBasket | hmmm. |
| 01:20.11 | FruitBasket | e-mails Carlos_PHX to himself ;-) |
| 01:20.16 | Carlos_PHX | Heh |
| 01:20.17 | Archide | lol |
| 01:20.46 | Archide | Can I interject iwth a few asbsolute beginner questions and leave quickly and peacefully? |
| 01:20.59 | FruitBasket | sure. |
| 01:21.10 | Carlos_PHX | Jump in, we don't own the channel. |
| 01:21.24 | Carlos_PHX | Of course, we might flame you if the question is REALLY bad, but hey, it's the internet. |
| 01:21.25 | FruitBasket | Carlos: I may take you up on that. It really sounds like a great idea :-) but, not until I have some better logging, so not for a few days, at least. |
| 01:21.42 | FruitBasket | thanks, really. |
| 01:21.47 | Carlos_PHX | Yeah, that's my recommendation. Get more details, then see if an outbound test will tell us something. |
| 01:21.55 | Carlos_PHX | If I'm not around here, carlos@televolve.com |
| 01:21.59 | Archide | I'm very intrested in beginning work with Asterisk, small background in telephony. I taught myself some perl/bash a long time ago and small amount of linux knowledge. All command line mostly with CHAP clients, DNS, APACHE stuff. |
| 01:22.13 | Carlos_PHX | ~book |
| 01:22.13 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 01:22.45 | Carlos_PHX | Archide: That's your best start. If you have specific questions shoot them, but for general knowledge you need the book. |
| 01:22.47 | Archide | I'd like to know where to start. I'm reading Asterisk - The future of telephony. But not sure if I need to work more on linux, or scripting before I begin this venture. |
| 01:23.03 | FruitBasket | great, thanks :-) |
| 01:23.07 | Carlos_PHX | I would say basic Linux is important. |
| 01:23.26 | Archide | Ahh ok perfect yeah reading that now. So read more up and play around with linux administration first? |
| 01:23.31 | Carlos_PHX | Sure, do let me know what you find out, like I say, I am curious about the problem and the potential Vitelity involvement. |
| 01:23.59 | FruitBasket | archide: for the linux task, it's a matter of compiling asterisk, probably. It's really straight forward though: ./configure, make menuselect, make, make install. After that, start asterisk -- /etc/init.d/asterisk start. Maybe add it to start automatically if it doesn't do it.. then it's all dialplan. |
| 01:23.59 | Carlos_PHX | Archide: Do it all, and use your best learning method. For me it was building servers and doing it. |
| 01:24.18 | Carlos_PHX | If nothing else, you can build VMware machines on your own computer. |
| 01:24.21 | [TK]D-Fender | Archide: if you understand the basics of Linux, given that you've done bash & perl, I'd lay bets you can just about fly into * |
| 01:24.27 | Carlos_PHX | Read voip-info.org of course. |
| 01:24.38 | Carlos_PHX | Yeah, what [TK]D-Fender said. |
| 01:24.56 | Archide | Ok, beautiful then I'll continue reading this book. I didn't want to begin learning about asterisk and be too far behind in other aspects is all. |
| 01:24.58 | FruitBasket | the only actual linux stuff I've done on the voip server is add a firewall rule for ssh and vim to read the log files ;-) |
| 01:25.11 | Archide | ok good to know. |
| 01:25.13 | Archide | perfect |
| 01:25.39 | FruitBasket | ok.. off to make log sheets. |
| 01:25.44 | [TK]D-Fender | Archide: need a basic understanding of iptables, routing tables, file permissions (barely), etc |
| 01:25.45 | Carlos_PHX | Good luck |
| 01:25.54 | Archide | Is one asterisk box able to run as a PBX for multiple sights which host IP phones? |
| 01:26.00 | Carlos_PHX | Yeah, really basic Linux admin will get you by. |
| 01:26.09 | Archide | I've worked with shorewall in the past as wel |
| 01:26.14 | Carlos_PHX | Archide: That's what we do. |
| 01:26.16 | Carlos_PHX | So yes. |
| 01:26.19 | Archide | ahh nice |
| 01:26.31 | Carlos_PHX | It gets complex and you have to build your own, but can be done. |
| 01:26.37 | Archide | I was trying to figure out how you could offer more of as a service then a one time sale without residual income |
| 01:26.49 | Carlos_PHX | Just remember that Asterisk doesn't really "do" anything, it just follows your commands. |
| 01:26.54 | Archide | Ok, I know not to ask too many new questions. |
| 01:27.01 | Carlos_PHX | Like, "what does a web server do?" |
| 01:27.03 | Archide | Yeah I love that concept. |
| 01:27.20 | Archide | Are the "commands" perl/bash/proprietary scripting? |
| 01:27.20 | Carlos_PHX | Archide: I run a hosted service provider. |
| 01:27.31 | Carlos_PHX | Proprietary, weird, and often dumb. |
| 01:27.36 | Archide | lol |
| 01:27.38 | Archide | nice |
| 01:28.12 | Carlos_PHX | A friend of mine said: Asterisk is like crack. It's horrible, and you know you want to stop doing it, but every time you try something else it sucks you back in. It's crap, but it's the best thing out there. |
| 01:28.21 | Archide | I'm just trying to move foward with my career, I have the talents to do more but job limiting situation. |
| 01:28.33 | Carlos_PHX | I moved from general IT consulting. |
| 01:28.49 | Carlos_PHX | I did have quite a telephony background in data, so that helped. |
| 01:28.51 | Archide | self employed? |
| 01:28.53 | Carlos_PHX | Yes |
| 01:29.05 | Carlos_PHX | And currently have a partner running a hosted PBX service. |
| 01:29.13 | Archide | nice |
| 01:29.33 | Archide | Open source offers so much more in community help though too if you don't abuse it |
| 01:29.41 | Carlos_PHX | What you're describing is "hosted PBX" where the customer has phones and nothing more. |
| 01:30.14 | Carlos_PHX | I'd tell you not to get into it lightly though, we have a lot of infrastructure and years of pain getting it stable. |
| 01:30.50 | Archide | No I have a partner willing to jump in this, if I can get the tech stuff down |
| 01:31.07 | Archide | I'd play with my house and his house first actually. |
| 01:31.22 | Archide | That should be pain enough, listening to his wife and mine when there's no dial tone |
| 01:31.24 | Archide | :) |
| 01:31.57 | Archide | All data lines or using analog as well? |
| 01:32.14 | *** join/#asterisk KP7 (n=chatzill@dsl001-145-071.phl1.dsl.speakeasy.net) |
| 01:32.28 | Archide | data - digital I meant |
| 01:35.40 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 01:35.55 | Carlos_PHX | No analog. |
| 01:35.59 | Carlos_PHX | That would suck. |
| 01:36.14 | Carlos_PHX | We have local PRIs in three locations in town for primary local service. |
| 01:36.14 | FruitBasket | Can I turn sip debugging on globally and write to a dedicated file? |
| 01:36.32 | Carlos_PHX | Then we use wholesale SIP carriers to bring in out-of-town DIDs. |
| 01:37.04 | FruitBasket | also, does sip debugging impact performance meaningfully? |
| 01:37.19 | Carlos_PHX | FruitBasket: I've never noticed an impact, but use it rarely. |
| 01:37.33 | Carlos_PHX | On our system with 1k+ handsets, it's not real useful... :-) |
| 01:38.11 | FruitBasket | yeah.. ours with 20 and 5 concurrent calls.. I can't see it being useful. It'll take me a week to write a parser and get anything meaningful from them.. but.. I need information :-| |
| 01:38.34 | FruitBasket | but, it writes to full.<hostname>.log :-D |
| 01:39.14 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 01:39.21 | Carlos_PHX | I would start with user feedback in a written log. |
| 01:39.26 | Carlos_PHX | It's remarkably effective. |
| 01:39.54 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 01:39.55 | FruitBasket | Warning: 392 192.168.1.50:5060 "Noisy feedback tells: -- is this line meaningful?... |
| 01:40.10 | FruitBasket | noisy feedback? |
| 01:40.57 | Carlos_PHX | Huh, don't know what that is. |
| 01:41.03 | *** join/#asterisk ecrist (n=ecrist@chunk.ip6.secure-computing.net) |
| 01:42.05 | *** join/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 01:42.07 | Carlos_PHX | Google seems to say this is meaningless. |
| 01:42.47 | FruitBasket | agreed :-) |
| 01:42.47 | *** part/#asterisk kisu (n=kisu@daniel1117.broker.freenet6.net) |
| 01:43.40 | Carlos_PHX | When you mentioned that Vitelity said they got a hangup, was that from the telco side or your side? |
| 01:43.45 | *** part/#asterisk ecrist (n=ecrist@chunk.ip6.secure-computing.net) |
| 01:46.07 | *** join/#asterisk diter (n=martin@83.140.233.103.dyn.rp80.se) |
| 01:47.50 | diter | what pci card shud I by if I only whant test asterisk with a my normal telephone ? |
| 01:48.55 | *** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com) |
| 01:50.13 | telnettech | can someone explain how to make the *8 feature work. I was under the impression that the different users were to be in the same call group but I have a system that is not working that way |
| 01:51.26 | *** join/#asterisk JonCat (n=JonCup@cpe-67-49-245-26.dc.res.rr.com) |
| 01:52.54 | diter | telnettech I belive every one is sleeping. Do you know what pci card I shud by ? |
| 01:53.17 | telnettech | what are you wanting to do |
| 01:53.42 | telnettech | analog stations, PSTN, T-1/E-1 |
| 01:54.07 | harry_v | hoe many |
| 01:54.13 | harry_v | how many extentions |
| 01:54.39 | diter | I am new to asteisk and have set it up to use sip software. Know I whant to thest with a analog pstn phone |
| 01:55.14 | diter | one extentions maby two ? |
| 01:55.29 | harry_v | two is always better |
| 01:56.00 | harry_v | depends if you have existing cat3 or cat5 where you want these phones located. |
| 01:56.20 | telnettech | TDM410 is the pci card you want. you can get upto 4 analog stations or PSTN lines or a combination card.. |
| 01:57.02 | diter | thanks telnettech I will check that card out |
| 01:57.18 | telnettech | http://www.digium.com/en/products/analog/tdm410.php |
| 01:57.27 | telnettech | this is the card on the digium site |
| 01:57.35 | harry_v | depends what and where his lines terminate at unless he is going to have everything at one spot like a lab. |
| 01:58.15 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 01:58.59 | telnettech | harry_v: any idea how to setup an extension to be able to answer a ringing phone? |
| 01:59.24 | harry_v | tons of docs online. asteriskguru asterisk.org ect. |
| 01:59.53 | harry_v | if you downloaded the samples then you can work from those. |
| 02:00.13 | [TK]D-Fender | telnettech: "core show applications like pickup" |
| 02:00.28 | telnettech | im using 1.2 version |
| 02:01.13 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 02:01.21 | telnettech | core doesnt work D-Fender |
| 02:05.32 | harry_v | tel type help |
| 02:06.10 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-e922b07d5b9e5908) |
| 02:23.50 | [TK]D-Fender | telnettech: Consider upgrading. |
| 02:23.58 | [TK]D-Fender | telnettech: 1.2 is no longer supported |
| 02:24.13 | *** join/#asterisk imcdona (n=imcdona@c-24-22-222-90.hsd1.wa.comcast.net) |
| 02:24.19 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:24.20 | telnettech | Development team is testing 1.4.20 with the other application that is running on same server |
| 02:24.57 | telnettech | should be ready late january or early february. But I need to make this work for 2 customers we have |
| 02:26.05 | [TK]D-Fender | telnettech: asterik -rx "show applications"|grep pick |
| 02:26.12 | [TK]D-Fender | telnettech: asterisk -rx "show applications"|grep pick |
| 02:26.29 | [TK]D-Fender | telnettech: asterisk -rx "show applications"|grep PICK <- probably uppercase |
| 02:26.37 | telnettech | yeah i can find that but it really doesnt say how to set it up |
| 02:26.50 | telnettech | i have check a couple websites but im still looking |
| 02:27.10 | imcdona | Is there a way to have a sip phone act like an old analog handset where a particular line is lit on a multiline phone, and when somone puts the call on hold, have somone else at a different extension pick up the call? |
| 02:27.22 | imcdona | without parking the call |
| 02:27.24 | [TK]D-Fender | telnettech: Go read the WIKI then... its largely dated. |
| 02:27.53 | [TK]D-Fender | imcdona: Not really. * does not support true SLA |
| 02:28.05 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 02:28.11 | imcdona | SLA is single line appearance? |
| 02:28.20 | [TK]D-Fender | ~sla |
| 02:28.21 | jbot | extra, extra, read all about it, sla is service level agreement, or shared line appearances |
| 02:28.31 | imcdona | ahh ok...thanks |
| 02:28.45 | *** part/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 02:29.20 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 02:39.16 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 02:41.08 | *** join/#asterisk apeiron (n=apeiron@c-76-124-253-149.hsd1.pa.comcast.net) |
| 02:44.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 02:47.33 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:51.30 | *** join/#asterisk errr (n=errr@fedora/errr) |
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| 03:00.23 | *** join/#asterisk talntid (n=eric@c-67-185-179-75.hsd1.wa.comcast.net) |
| 03:01.03 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-161-34.dsl.stlsmo.sbcglobal.net) |
| 03:02.53 | LemensTS | Im having people upload a wav file to asterisk via a webpage, Im trying to figure out what specs the wav file needs to be for asterisk Playback cmd |
| 03:03.03 | LemensTS | having trouble getting it to play. |
| 03:06.21 | *** join/#asterisk onepointfive (n=dwillemb@118.142.4.226) |
| 03:06.28 | onepointfive | hi there |
| 03:07.20 | LemensTS | hi |
| 03:09.31 | [TK]D-Fender | <PROTECTED> |
| 03:10.00 | onepointfive | have some questions about using asterisk primarily as a call authentications/recording device - want to make sure it can do what I would like it to do before going out and getting hardware, anybody around willing to answer a few questions on this? |
| 03:12.24 | [TK]D-Fender | onepointfive: Shoot |
| 03:12.31 | onepointfive | thanks :) |
| 03:12.37 | onepointfive | here is what I am looking to do |
| 03:17.17 | onepointfive | a call comes in, and the caller gets prompted for a username and password via recorded message (similar to logging into any conference call). Once the user has passed authentication, the call goes directly to something that can record it( voicemail box?) which is decided by the auth credentials - giving each authenticated user the ability to record a message in their own area) - once the message has been recorded and encoded, the voicemail should b |
| 03:17.26 | onepointfive | am I asking the impossible? |
| 03:18.58 | LemensTS | I just did something like this. You would want to use a mysql database with this. |
| 03:19.21 | LemensTS | It can all be coded in the dialplan. |
| 03:19.44 | onepointfive | am completely new to asterisk, will read up on what a dialplan is :) |
| 03:19.58 | stencil | ~book |
| 03:19.59 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 03:20.06 | LemensTS | Your message did cut off, are you trying to have the record it as there voicemail message? |
| 03:20.20 | [TK]D-Fender | onepointfive: "YE" |
| 03:20.28 | [TK]D-Fender | YES* |
| 03:20.39 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-3-82.phlapa.east.verizon.net) |
| 03:20.48 | onepointfive | bascially I want to record authenticaed messages and have them forwared as email attachments |
| 03:21.30 | onepointfive | and to ensure boss happiness and my Christmas bonus, getting it to actually recored entire conference calls and have them available as audio-files would be great too |
| 03:22.07 | [TK]D-Fender | onepointfive: All doable, hardly that complicated |
| 03:22.18 | [TK]D-Fender | onepointfive: No real need for DB's either |
| 03:22.22 | LemensTS | your probabaly gonna want to use deadAGI and perl |
| 03:22.29 | LemensTS | or php whatever language u like |
| 03:22.31 | [TK]D-Fender | LemensTS: none of the above |
| 03:22.47 | [TK]D-Fender | LemensTS: Nothing he can't do with pure * |
| 03:24.15 | diter | n |
| 03:24.30 | onepointfive | reading through the docs all the pieces are built in for sure, but I want to be sure I can string them along in the correct order - call, authentication, recording , email (or just store and I have a cron job to email them automatically) |
| 03:24.42 | diter | Any one used the cheep AX100P card ? |
| 03:24.46 | [TK]D-Fender | onepointfive: no need for cron either. |
| 03:24.56 | [TK]D-Fender | diter: X100P is a junk card. |
| 03:25.09 | [TK]D-Fender | dittPoor PCI design, CID & hangup issues. |
| 03:25.30 | [TK]D-Fender | onepointfive: Virtually everything you need centeres on the voicemail app. |
| 03:25.30 | diter | yes junk but dos it work can I recive a call ? |
| 03:25.43 | [TK]D-Fender | diter: probably |
| 03:26.45 | diter | D-Fender I only need one fxs and dont whant to spend a lot, |
| 03:27.08 | [TK]D-Fender | diter: That card isn't FXS. |
| 03:27.09 | drmessano | You're not gonna get something that really works with an X100P |
| 03:27.58 | onepointfive | So, any word on the ability to conference and record conference calls? |
| 03:28.00 | [TK]D-Fender | onepointfive: MeetMe + Monitor |
| 03:28.30 | onepointfive | I assume these are plugins/modules or some sort? |
| 03:28.39 | [TK]D-Fender | onepointfive: All dialplan applications |
| 03:28.52 | onepointfive | ah, give me 5mins to read about dailplan pls |
| 03:28.56 | [TK]D-Fender | onepointfive: Everything you need is part of * |
| 03:29.07 | carrar | telco works fine with a x100p |
| 03:29.11 | [TK]D-Fender | onepointfive: 5 mins? You'll spend a few DAYS learning it. |
| 03:29.12 | carrar | :) |
| 03:29.21 | saftsack | diter, maybe try a cheap ata with fxo port |
| 03:29.22 | carrar | as a FXO interface |
| 03:29.32 | [TK]D-Fender | onepointfive: it is the most complex and important part of *. |
| 03:30.44 | diter | Thx all I go read about ata |
| 03:31.16 | saftsack | but what's up with the x100p. is this card really crap (maybe some links to threads, serious reviews etc.) or just because it is from digium? |
| 03:31.30 | onepointfive | D-Fender, I just meant to read up and get an idea of what it is :) |
| 03:31.59 | onepointfive | never used * before, so finding out what all the parts are and how they fit together is important |
| 03:32.39 | onepointfive | been considering getting one of the digium asterisk appliances, purely because it is neat and tidy - are they worth it, or should I just build a box myself ? |
| 03:33.04 | carrar | saftsack, I've had a x100p working for years a basic fxo card |
| 03:33.18 | carrar | originaly purchased from Digium |
| 03:33.29 | carrar | not some copy card |
| 03:33.51 | saftsack | your fxo card ---connected---> carrier ? |
| 03:34.08 | carrar | your phone line from your phone provider plus into it |
| 03:34.13 | carrar | plugs |
| 03:34.19 | saftsack | ah ok :) |
| 03:34.32 | carrar | carrier is providing tone and ring |
| 03:34.35 | saftsack | sometimes a little bit hard to understand what is meant with fxo and fxs |
| 03:34.36 | carrar | aka FXO |
| 03:35.02 | saftsack | no echo issues? |
| 03:35.04 | carrar | O to office |
| 03:35.06 | carrar | S to stations |
| 03:35.39 | saftsack | yes i know that but this is the same as talking with a person who is opposite to you where is right and left |
| 03:35.40 | carrar | no |
| 03:36.09 | carrar | but echo can varry depending on your setup and carrier |
| 03:36.18 | saftsack | yes thats true |
| 03:36.19 | [TK]D-Fender | onepointfive: Costs a fortune and is rather limited on CPU, etc... it was not meant for anything other than light PBX work.. and costs a LOT more that what you need to actually get the job done |
| 03:36.25 | [TK]D-Fender | onepointfive: It is a TOASTER |
| 03:36.39 | onepointfive | hahaha |
| 03:36.47 | saftsack | but at all the problem is, that it isnt possible to get informations how good a card really is but the most persons get paid for informations :( |
| 03:37.28 | carrar | Well the newer cards are better |
| 03:37.33 | carrar | obviosuly |
| 03:37.46 | [TK]D-Fender | saftsack: plenty easy to get info on how good a card is. The X100P is crap. There see? |
| 03:38.09 | diter | lol |
| 03:38.53 | onepointfive | D-Fender - this will live in our datacenter so I guess a beefy 1U box with some PCI should be fine? Is Asterisk particularly heavy on anything (drive space? memory? cpu?) |
| 03:39.07 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 03:39.20 | saftsack | that it isnt a telco grade card is known. but if it is crap if your telco has a nice echocancelled fxo line? |
| 03:39.32 | saftsack | but at all i have the same opinion |
| 03:39.33 | [TK]D-Fender | onepointfive: You haven't said how you intend on connecting it to the PSTN, or the kind of load it'll be under |
| 03:39.56 | [TK]D-Fender | saftsack: Telcos don't EC lines. Otherwise faxes wouldn't work |
| 03:40.02 | carrar | x100p is not high grade |
| 03:40.05 | carrar | but it works |
| 03:40.20 | saftsack | [TK]D-Fender, but telcos can detect if there is a fax so they are able to ec a line |
| 03:40.25 | saftsack | carrar, true |
| 03:40.26 | carrar | depends what your requirements are |
| 03:40.37 | [TK]D-Fender | carrthe problems are more than echo. |
| 03:40.46 | [TK]D-Fender | saftsack: the problems are more than echo. |
| 03:40.59 | carrthe | w00t! |
| 03:41.00 | onepointfive | D-Fender : this is where my knowledge breaks down - I come from a purely IP background. That being said I would no predict any more than two to three calls happening at a time, max. |
| 03:41.07 | carrthe | h4X0rs!! |
| 03:41.39 | saftsack | also the hfc chipset cards are crap ... but just because of bad drivers. within patton gateways they work |
| 03:42.02 | saftsack | someone here who tested eicon diva BRI cards? |
| 03:42.45 | [TK]D-Fender | onepointfive: Good answer. Now HOW do you intend to access the PSTN? |
| 03:43.10 | [TK]D-Fender | kicks carrar in the nads |
| 03:43.23 | carrar | WTF@#(*$&^!@(*#$!@# |
| 03:43.39 | carrar | good thing I have my IRON cup on |
| 03:43.41 | saftsack | onepointfive, what do you try? |
| 03:43.46 | onepointfive | D-Fender, I really haven't done any research in that direction yet, but am more than willing to take some advice |
| 03:44.17 | [TK]D-Fender | onepointfive: Given your volume, analog lines or an ITSP |
| 03:44.21 | [TK]D-Fender | ~ITSP |
| 03:44.21 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:44.32 | saftsack | or BRI? |
| 03:44.32 | onepointfive | saftsack - nothing yet, just working out if a box with asterisk on it is the right tool for the job I am trying to accomplish :) |
| 03:44.35 | [TK]D-Fender | ~ISPLIST-us |
| 03:44.49 | carrar | saftsack, just by a T1 card and do everything via T1's! |
| 03:45.02 | carrar | or sip |
| 03:45.15 | saftsack | do you know what E1 costs in germany per month? |
| 03:45.25 | carrar | $1 |
| 03:45.51 | saftsack | SIP isnt often an alternative in germany. dunno why but many people want to have a real telephone line |
| 03:45.52 | [TK]D-Fender | onepointfive: Actually... you in HK as you appear? |
| 03:46.07 | onepointfive | yep |
| 03:46.18 | [TK]D-Fender | onepointfive: Is that to be your calling area as well? |
| 03:46.24 | saftsack | carrar, 280€ at t-home deutsche telekom |
| 03:46.27 | onepointfive | yes |
| 03:46.32 | saftsack | per month. just for having the e1 line |
| 03:46.49 | carrar | Get a Digum TDM410 |
| 03:47.07 | [TK]D-Fender | onepointfive: Ok, well analog lines may be economical. Check around for ITSP's serving your area. I don't know how competitive your options will be. |
| 03:47.21 | [TK]D-Fender | onepointfive: But for analog lines I'd recommend the Sangoma A200d |
| 03:47.47 | onepointfive | will do - just been briefed that if we do sell this service to other customers of ours, we may go a bit higher |
| 03:48.04 | onepointfive | so assume max cap of about 20 calls |
| 03:48.12 | [TK]D-Fender | onepointfive: Simultaneous? |
| 03:48.17 | saftsack | yes. the a200DDDD <- EC!!! or a patton gateway with fxo lines |
| 03:48.31 | onepointfive | worst case simultaneous |
| 03:48.35 | saftsack | then you can build it up on an embedded solution if there are maximum of 5 calls |
| 03:48.42 | onepointfive | (or best case, if you ask the sales guys hahahA) |
| 03:48.59 | [TK]D-Fender | onepointfive: based on your variable scalablity, an ITSP has a high likelyhood of being your best bet economically speaking. |
| 03:49.05 | saftsack | amd geode 800lx for about 90€ as a complete pc. then an fxo gateway and some ip phones and then you are complete |
| 03:50.13 | onepointfive | D-Fender - and if I said I had a big budget, what should I be looking at :) |
| 03:50.31 | saftsack | sangoma a200d |
| 03:50.36 | saftsack | or patton fxo gateway |
| 03:51.19 | [TK]D-Fender | onepointfive: well its a question of wastage as well... PRI is for sure the strongest option, but it really costs... |
| 03:51.47 | [TK]D-Fender | onepointfive: in terms of hardware and service fees |
| 03:51.52 | saftsack | maybe try an itsp and some analog lines as backup? |
| 03:52.46 | saftsack | do you have an async call amount? if you are a company which just makes calls and receives not many calls then you can call out over sip with fallback and get calls over your analog lines |
| 03:53.17 | onepointfive | this will be nothing more than receiving and recording calls |
| 03:53.21 | [TK]D-Fender | saftsack: He's designing a call-in system |
| 03:53.34 | saftsack | sorry didnt read it from the beginning |
| 03:54.10 | saftsack | do you have monthly costs for PRI in the usa? |
| 03:54.59 | onepointfive | quality and reliability are paramount here - our customers area happy to spend over the odds on things like that (and bill them for it we certainly will!) - Telecoms in Hong Kong is also generally quite cheap, I would guess this area should be too |
| 03:55.15 | [TK]D-Fender | saftsack: Depends where and with whom, and he is not IN the USA |
| 03:55.36 | [TK]D-Fender | onepointfive: on the quality & reliability front, PRI is king ( |
| 03:55.38 | [TK]D-Fender | ~pri |
| 03:55.39 | jbot | [pri] [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 03:55.41 | [TK]D-Fender | ~j1 |
| 03:55.59 | [TK]D-Fender | onepointfive: J1 is the carrier circuit in HK IIRC |
| 03:56.11 | [TK]D-Fender | No wait... thats just Japan... or is it.. |
| 03:56.33 | [TK]D-Fender | onepointfive: one of T1, E1, or J1.... other can confirm, but PRI signalling over it |
| 03:57.13 | onepointfive | I would imagine HK to be E1 - it was part of the UK until 1997 |
| 03:57.42 | onepointfive | most things here follow the UK model |
| 03:57.49 | saftsack | onepointfive, in which country do you live? |
| 03:57.55 | onepointfive | Hong Kong |
| 03:58.16 | onepointfive | not really a country I guess, but it likes to think it is :) |
| 03:58.55 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:59.03 | [TK]D-Fender | saftsack: PLEASE get with the program! |
| 03:59.03 | saftsack | my recommendation goes to pri |
| 03:59.17 | saftsack | [TK]D-Fender, later entries arent ok? ;) |
| 03:59.44 | [TK]D-Fender | [22:54]<onepointfive>quality and reliability are paramount here - our customers area happy to spend over the odds on things like that (and bill them for it we certainly will!) - Telecoms in Hong Kong is also generally quite cheap, I would guess this area should be too |
| 03:59.53 | [TK]D-Fender | saftsack: this was 5 mins ago. |
| 04:00.03 | [TK]D-Fender | looks inside saftsack's skull.... |
| 04:00.20 | saftsack | i was smoking :) |
| 04:00.21 | [TK]D-Fender | yup... the light are on... the wheel is spinning... but the hamster is F-ING DEAD :p |
| 04:00.43 | [TK]D-Fender | saftsack: [22:53]<saftsack>sorry didnt read it from the beginning |
| 04:00.45 | jaytee | he was smoking........ and probably drinking the bong water |
| 04:00.57 | [TK]D-Fender | saftsack: What are you smoking exactly? And do you have extra? |
| 04:01.28 | saftsack | let's drop those things ... |
| 04:01.57 | saftsack | onepointfive, get a pri line. itsp isnt as reliable as pri. |
| 04:03.10 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 04:06.33 | onepointfive | thanks for all your help guys, particularly D-Fender :) going to go do some reading and pricing now |
| 04:06.47 | onepointfive | sure I will be back to bother this # soon enough hahaha |
| 04:14.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:14.40 | *** join/#asterisk SanityIO (n=SanityIO@77.242.102.18) |
| 04:16.27 | *** part/#asterisk diter (n=martin@83.140.233.103.dyn.rp80.se) |
| 04:19.16 | [TK]D-Fender | ~bri |
| 04:19.17 | jbot | well, bri is [~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
| 04:19.18 | [TK]D-Fender | ~pri |
| 04:19.19 | jbot | i guess pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 04:28.07 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:38.49 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
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| 05:11.15 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 05:19.12 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 05:27.51 | *** part/#asterisk onepointfive (n=dwillemb@118.142.4.226) |
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| 05:33.09 | *** part/#asterisk advorak (n=advorak@c-76-102-231-132.hsd1.ca.comcast.net) |
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| 05:42.26 | imcdona | This is an easy question but I am braindead at the moment. I have a destiation for a certain extension, but, I dont want the extension to be dialed unless a user presses 1. if they press 1 it then rings the extension |
| 05:44.16 | jblack | just do a getdigit with no timeout. |
| 05:44.52 | imcdona | i'll look that up, thanks |
| 05:46.34 | imcdona | is getdigit an application? |
| 05:46.49 | imcdona | I dont see it listed on: http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands |
| 05:47.09 | jblack | I don't see it either. Hold |
| 05:47.41 | jblack | the actual name is read, and it's a cmd. |
| 05:48.00 | imcdona | cool ty |
| 05:52.39 | [TK]D-Fender | "Application" |
| 05:53.02 | [TK]D-Fender | imcdona: your terminology is all ovre the map |
| 05:53.20 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 05:53.30 | [TK]D-Fender | Who is dialing what, and then who presses what to do what? |
| 05:54.30 | imcdona | manager api makes a call outbound to a PSTN phone, if the user presses 1, they accept the call and then it moves along and connects the desired extension, if they do not press one, it hangs up the call |
| 05:56.01 | [TK]D-Fender | imcdona: you need to pass it to an extension that will ask for input. afterward you can do whatever you want |
| 05:57.16 | [TK]D-Fender | imcdona: So dump them in an IVR or another exten with a looping Read or similar |
| 05:57.40 | imcdona | http://pastebin.com/d7a05b3a0 |
| 05:57.44 | *** join/#asterisk moy (n=moy@187.133.5.154) |
| 05:58.04 | *** join/#asterisk workdraft (n=acxide@203.215.94.239) |
| 05:58.12 | workdraft | got a question. |
| 05:58.30 | drmessano | ~ask |
| 05:58.31 | jbot | i guess ask is 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. |
| 05:58.41 | imcdona | so it dumps to that extension, but, if they press 1 for example and I redirect it to another extension for processing, then I lose the extensin its supposed to call |
| 05:59.01 | workdraft | forget being biased about asterisk. which would you recommend, freepbx or asterisk? |
| 05:59.44 | workdraft | in a 100 seat setting. ive been reading about asterisk lately but i havent read a lot about freepbx. |
| 05:59.58 | workdraft | i need to ask some good opinions or facts perhaps |
| 06:00.43 | drmessano | Thats not a direct comparison |
| 06:00.53 | drmessano | FreePBX USES asterisk |
| 06:00.56 | workdraft | oh |
| 06:00.57 | [TK]D-Fender | imcdona: that is not a proper IVR. |
| 06:01.08 | [TK]D-Fender | imcdona: go read the book on how to make one. |
| 06:01.56 | drmessano | FreePBX is a dialplan configuration interface for Asterisk that creates a PHP web based GUI experience with Asterisk as the underlying engine |
| 06:02.11 | [TK]D-Fender | better phrased : FreePBX builds configs via GUI following its limited logic to make an appliance-grade setup out of your system. None to bright, but does basic IVR stuff. |
| 06:02.17 | drmessano | Asterisk is still the call processor, the worker, the B2BUA |
| 06:04.08 | workdraft | any suggestion besides FreePBX that runs on the same platform (LAMPA)? |
| 06:04.30 | jameswf | dont make me sick the god warier on you http://video.stumbleupon.com/#p=4zkouqqy43 |
| 06:04.41 | drmessano | No, nothing else is usable.. which is why everyone, including Digium are using it where a GUI is desired |
| 06:04.48 | [TK]D-Fender | workdraft: Druid, ScopServ, Switchvox, ThirdLane |
| 06:05.05 | workdraft | @<[TK]D-Fender>: thnx... |
| 06:06.08 | drmessano | Druid being the only free one in that bunch |
| 06:07.16 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 06:11.48 | ManxPower | I was at a 2 day Druid conference a few months ago. |
| 06:12.03 | ManxPower | Pretty amazing stuff, but you really have to be a web 2.0 weenie. |
| 06:12.28 | ManxPower | And quite honestly I don't think the current web is anything past 0.01alpha |
| 06:17.30 | jblack | I've always thought of druids as weenies.... |
| 06:18.17 | [TK]D-Fender | Druids never use their weenies... |
| 06:18.18 | workdraft | have any one checked out vicidialnow? |
| 06:18.27 | [TK]D-Fender | makes for poor expansion ;) |
| 06:18.45 | [TK]D-Fender | I smell a tele-spammer... get the torches... |
| 06:19.12 | [TK]D-Fender | Going from "chump GUI user" to "hated by the masses" |
| 06:19.16 | ManxPower | Release the lions! |
| 06:20.01 | jblack | Doesn't irreperable harm sound expensive to you? |
| 06:20.17 | jblack | I don't think I like irssi with split screens |
| 06:22.37 | [TK]D-Fender | Alrighty... checkout time.. later all |
| 06:23.15 | workdraft | gtg. thn ya'll. |
| 06:23.18 | workdraft | quit |
| 06:34.55 | *** join/#asterisk BeeBuu (n=beebuu@218.13.68.239) |
| 06:36.24 | BeeBuu | i connected 2 asterisks,can i set a queue member in another server? |
| 06:46.01 | *** part/#asterisk LemensTS (n=matthew@adsl-70-238-161-34.dsl.stlsmo.sbcglobal.net) |
| 06:47.13 | grndslm | you guys know of any good voip forums where different providers/codecs/etc. are discussed? |
| 06:48.54 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 06:49.31 | rue_mohr | any comments on the aastra 9143 for a small office? |
| 06:51.38 | *** join/#asterisk TrentCreek (n=kvirc@ppp-70-244-17-140.dsl.hrlntx.swbell.net) |
| 06:54.42 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
| 06:54.48 | speekac | hi all |
| 06:55.06 | speekac | does anyone has implemented SRTP and TLS support in asterisk before? |
| 06:55.40 | demonist | nope |
| 06:55.59 | demonist | how about using ipsec instead |
| 06:56.17 | demonist | is that any good or too much overhead |
| 06:56.41 | speekac | nope, there is a specific requirement that SRTP and TLS support must be in place |
| 06:57.24 | demonist | hmm, i would not be able to help you |
| 06:57.30 | demonist | i have yet to build a pbx yet |
| 06:57.37 | demonist | or read much into this asterisk book |
| 06:57.41 | demonist | hopefully soon though |
| 06:57.55 | speekac | thanks anyway |
| 06:58.14 | demonist | i have yet to build a pbx |
| 06:58.16 | demonist | soon though |
| 06:58.17 | demonist | =) |
| 06:58.33 | Corydon76-dig | speekac: bug oej about it |
| 06:58.49 | Corydon76-dig | speekac: there is experimental support for TLS in 1.6.0 |
| 07:00.09 | Corydon76-dig | SRTP is still a ways away, though |
| 07:02.06 | TrentCreek | How can I get Asterisk to cut my lawn? |
| 07:03.04 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:08.30 | TrentCreek | then what about SMS? |
| 07:10.00 | speekac | oej: hi there |
| 07:10.04 | speekac | Corydon76-dig: thx |
| 07:12.06 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 07:16.09 | *** join/#asterisk dlynes (n=daniel@d206-116-179-20.bchsia.telus.net) |
| 07:16.41 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:19.56 | *** part/#asterisk speekac (n=alwin@60.51.217.61) |
| 07:21.35 | cvnet | hi |
| 07:21.49 | cvnet | what ports do you need to have open to have asterisk work properly? |
| 07:21.56 | cvnet | connecting sips .... ? |
| 07:22.54 | TrentCreek | http://www.google.com/search?q=asterisk+sip+port&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a |
| 07:26.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:31.40 | drmessano | ~ports |
| 07:31.42 | jbot | i guess ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 07:31.42 | drmessano | ~port |
| 07:31.43 | jbot | somebody said port was To port something, you translate the code for a program from one platform to another. You could port a program you wrote on a PC over to a Macintosh, for example. Port |
| 07:31.49 | drmessano | ~sipnat |
| 07:31.50 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 07:31.58 | drmessano | That has the port info |
| 07:34.56 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 07:37.58 | *** join/#asterisk eliyahud (n=eliyahu@bzq-219-217-130.pop.bezeqint.net) |
| 07:41.21 | cvnet | I configured the firewall, but sip gets registered but no voice |
| 07:41.34 | cvnet | i open tcp/udp 5060-7000 |
| 07:41.40 | cvnet | any other suggestions? |
| 07:43.36 | TrentCreek | turn on SIP debug |
| 07:45.14 | eliyahud | what ab0ut the rtp ports |
| 07:45.20 | eliyahud | udp 10000:20000 |
| 07:45.34 | cvnet | sip debug doesnt show anything |
| 07:48.02 | cvnet | <eliyahud thanks taht did it |
| 07:48.05 | cvnet | thanks a bunch |
| 07:48.11 | eliyahud | woo! |
| 07:48.27 | cvnet | for sip if i have 5060:7000 <-- that should be good right? |
| 07:48.55 | eliyahud | why do you need up to 7000, afaik the only port you need open is 5060 |
| 07:49.12 | cvnet | what if i got more than one client connected? |
| 07:49.42 | eliyahud | so they also connect to 5060 |
| 07:49.48 | cvnet | hum |
| 07:49.57 | cvnet | so just leave 5060 udp/tcp open? |
| 07:50.58 | eliyahud | yeah |
| 07:51.03 | cvnet | im really paranoid i got hacked 2 days ago |
| 07:51.57 | eliyahud | how |
| 07:52.14 | *** join/#asterisk hackbanger (n=hackbang@213.209.114.6) |
| 07:52.44 | cvnet | duno |
| 07:52.49 | cvnet | i had a really simple password |
| 07:52.52 | cvnet | and firewall was off |
| 07:53.10 | cvnet | plus im not really good with unix/linux |
| 07:53.23 | cvnet | he is still on teh server |
| 07:53.37 | cvnet | i just build this one and im moving it over as soon as the firewall is configured |
| 07:53.44 | cvnet | he is doing some sort of attack i think |
| 07:54.27 | eliyahud | you should run a root kit detector and change the passwords and keys |
| 07:54.45 | cvnet | i changed hte root pass like 5 times, and he would change it back |
| 07:55.16 | eliyahud | how is he connecting, ssh? |
| 07:55.20 | cvnet | yes |
| 07:55.32 | cvnet | thanks that worked, i got only 5060 open now |
| 07:55.44 | cvnet | from Romania |
| 07:56.00 | eliyahud | you can change ssh to deny access to IP blocks, or allow access to only certain IP blocks |
| 07:56.03 | cvnet | root pts/1 2008-11-19 20:58 (79.118.156.194) |
| 07:56.28 | cvnet | i was in linux channel, and they told me your best bet is to install it freshly |
| 07:56.34 | cvnet | so that is what i did the whole day today |
| 07:56.42 | eliyahud | yeah if that's an option, that's the best |
| 07:56.47 | cvnet | with firewall and hard usernames passwords ... |
| 07:56.58 | cvnet | i hope i wont get rooted again |
| 07:57.09 | cvnet | anything else you would suggest? |
| 07:57.29 | cvnet | where do you change the ssh to deny allow ips? |
| 07:57.32 | cvnet | what file is it? |
| 07:57.44 | eliyahud | /etc/ssh/sshd_config i think |
| 07:58.27 | *** join/#asterisk af_ (n=getsmart@88-149-230-152.dynamic.ngi.it) |
| 07:58.34 | cvnet | ill do that |
| 07:58.35 | cvnet | thanks a bunch |
| 08:00.28 | *** join/#asterisk invalidrecord (n=fares@92.40.24.253.sub.mbb.three.co.uk) |
| 08:00.37 | invalidrecord | hi guys/girls |
| 08:02.01 | invalidrecord | i have just installed asterisk and the gui on my local box to dev a small rails app against, what do i need to set up to have a working voip iax setup locally with say 3 softphones |
| 08:02.54 | eliyahud | cvnet: actually sorry, you do it in /etc/hosts.deny |
| 08:03.28 | *** join/#asterisk X-Rob (n=rob-x@dsl-210-15-202-248-static.QLD.netspace.net.au) |
| 08:03.55 | X-Rob | Can someone re-open bug 13927 please |
| 08:04.17 | eliyahud | you need to configure iax.conf and extensions.conf |
| 08:05.17 | TrentCreek | Watch this vid |
| 08:05.18 | TrentCreek | http://www.youtube.com/watch?v=UP9b_FEZuUE |
| 08:05.28 | TrentCreek | invalidrecord: |
| 08:07.08 | *** join/#asterisk ManxPower (n=manxpowe@63.sub-70-214-21.myvzw.com) |
| 08:07.24 | invalidrecord | TrentCreek: thanks |
| 08:07.52 | TrentCreek | sure |
| 08:08.24 | TrentCreek | invalidrecord: you can download a higher res at the link they show, and example files |
| 08:09.14 | invalidrecord | cool thanks im ok with this just not used the voip on asterisk b4 used it a long time ago with telephony cards |
| 08:20.53 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:24.41 | *** join/#asterisk jpastore (n=jpastore@173.9.147.165) |
| 08:25.09 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 08:26.14 | casix | hello |
| 08:29.07 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0d5985f0bcd19f07) |
| 08:30.36 | *** join/#asterisk donnib (n=aaaa@0x555281d0.adsl.cybercity.dk) |
| 08:30.40 | donnib | hi all |
| 08:30.58 | donnib | i have a big problem with my newly ip asterisk setup |
| 08:31.28 | donnib | i have 2 phones. one in india and one in denmark. both connected on the same network so without nat. |
| 08:31.45 | donnib | i have turned off qualify since i have 300 ms latency |
| 08:32.23 | donnib | i can see both phones are connected but when i try to call from DK -> IN i see by looking on the web interface of the other phone that it's not claling |
| 08:32.45 | donnib | i looked in the asterisk log and i can see it sends out (Resending packets) |
| 08:32.59 | donnib | some times everything work ok and sometimes it's not working. |
| 08:33.19 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:33.21 | donnib | anyone have an idea what my problem could be ? is it because of the network ? |
| 08:41.35 | SwK | it sounds network related |
| 08:49.00 | donnib | is there any way i can see what is the problem ? |
| 08:49.45 | donnib | any debug or someting ? maybe it's clear that the network is the problem since i see the SIP messages resending but the other end does not receive them. |
| 08:50.12 | donnib | or is it because of the latency ? i mean 300 ms is alot but maybe it should work anyway. ?? |
| 08:52.54 | cvnet | what does dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) mean? |
| 08:56.15 | invalidrecord | ii have a local install of asterisk and i have defined two extensions but i dont seem to be able to call between them can anyone walk me through it quickly i dont need anything clever just two ext to test a rails app on? |
| 08:58.46 | cvnet | invalid one sec |
| 08:59.07 | invalidrecord | cvnet: thanks |
| 08:59.07 | *** join/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 08:59.10 | cvnet | r u able to register it? |
| 08:59.53 | invalidrecord | im not sure i have two defined in the user section of the asterisk gui |
| 09:00.01 | cvnet | on sip, you type in context=from-internal |
| 09:00.02 | invalidrecord | but i dont seem able to connect the softphone |
| 09:00.17 | cvnet | did you create a sip user? |
| 09:00.24 | invalidrecord | i was going for iax if possible |
| 09:00.31 | cvnet | something |
| 09:00.40 | cvnet | can you register your softphone with your system? |
| 09:00.58 | invalidrecord | no it seems not to connect |
| 09:01.08 | cvnet | is it on the same network? |
| 09:01.10 | invalidrecord | but i havent used it b4 so it may |
| 09:01.15 | invalidrecord | yes on local machine |
| 09:01.20 | kaldemar | invalidrecord: you'll get better help in #asterisk-gui. most (if not all) people don't use GUI's here. |
| 09:01.27 | *** join/#asterisk trogs (i=dwarf@nz1.jedi.net.nz) |
| 09:02.21 | trogs | hi, just wondering if anyone has firmware for a mitel 5055 sip phone? |
| 09:02.29 | trogs | their firmware download page appears to have gone away. |
| 09:02.35 | invalidrecord | is the gui a bit sucky then i do most other things in vi so dont see why i should do asterisk diffrent |
| 09:03.27 | cvnet | invalid if you have a nat=yes or check mark try to enable that |
| 09:04.18 | invalidrecord | im running the server on 127.0.0.1 and phone is on same box |
| 09:04.28 | invalidrecord | im not using nat |
| 09:04.56 | cvnet | login to box and type--> asterisk -r |
| 09:05.04 | cvnet | then -> sip show peers |
| 09:05.14 | cvnet | see if you can see the users there |
| 09:05.34 | cvnet | im not using gui so i wont be much help to you |
| 09:06.02 | invalidrecord | cvnet: i am happy with a console ill ditch the gui :-) |
| 09:06.21 | invalidrecord | nothing in sip peers |
| 09:06.23 | cvnet | lol i installed the gui first too, but then moved to no gui |
| 09:06.26 | kaldemar | ~book |
| 09:06.27 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 09:06.32 | cvnet | so you do not have a record there |
| 09:06.36 | invalidrecord | nope |
| 09:06.38 | cvnet | let me give you something |
| 09:06.39 | cvnet | one sec |
| 09:06.49 | kaldemar | invalidrecord: ^ that is a good reference for your needs if you stop messing around with the gui. |
| 09:06.59 | invalidrecord | i just did :-) |
| 09:07.09 | invalidrecord | im not a gui fan |
| 09:08.56 | cvnet | open /etc/asterisk/sip.conf |
| 09:09.03 | cvnet | and add http://pastebin.com/m12ff1705 |
| 09:09.27 | cvnet | secret is the pass, so you can change it to anything you wish |
| 09:11.10 | cvnet | then go to /etc/asterisk/extensions.conf |
| 09:11.12 | cvnet | and add http://pastebin.com/m29702c80 |
| 09:11.36 | cvnet | then try calling from user 100 sip (dail anything) it should ring to 101 |
| 09:11.49 | cvnet | and when u type sip show peers it should show you both users |
| 09:12.19 | cvnet | oo after editing each file you have to type reload or for sip --> sip reload for extensions -> dialplan reload |
| 09:12.24 | cvnet | good luck im going for smoke |
| 09:17.25 | invalidrecord | ok done and thanks i have sip |
| 09:18.17 | cvnet | did it work? |
| 09:18.22 | invalidrecord | yes |
| 09:18.26 | cvnet | cool |
| 09:18.27 | invalidrecord | thanks |
| 09:18.29 | cvnet | np |
| 09:18.48 | invalidrecord | that config looks a lot frendlier than sendmail i think ill dithc the gui |
| 09:18.56 | invalidrecord | ditch |
| 09:19.09 | cvnet | you dont have much control over the gui |
| 09:19.28 | cvnet | i mean with gui over asterisk |
| 09:19.58 | invalidrecord | now all i need is sound output lol |
| 09:20.14 | cvnet | what u mean? |
| 09:20.35 | invalidrecord | from zoiper |
| 09:20.42 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 09:20.48 | cvnet | oo |
| 09:21.15 | Karlitoo | can any 1 help me out, I'm trying to figure out how to complite h323plus for asterisk |
| 09:21.26 | Karlitoo | compile |
| 09:22.24 | *** join/#asterisk Dragoon_nz (n=Dragon@ip-58-28-152-209.static-xdsl.xnet.co.nz) |
| 09:24.54 | *** join/#asterisk talntid (n=eric@c-67-185-179-75.hsd1.wa.comcast.net) |
| 09:31.34 | cvnet | is there any command in cli to see how many actives calls are in process ? |
| 09:33.25 | *** join/#asterisk propellerhead (n=yogurt2u@host204.201-252-190.telecom.net.ar) |
| 09:36.34 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:37.30 | Maliuta | core show channels |
| 09:40.12 | invalidrecord | ok brb |
| 09:40.50 | cvnet | thanks |
| 09:45.14 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 10:01.32 | *** join/#asterisk tapic (n=tap@88.255.77.215) |
| 10:02.12 | tapic | hi all, does anyone have knowledge about hunting policies for ss7 channels? |
| 10:02.46 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:03.56 | tapic | I experience lots of chanunavail & congestion problems when I try to outdial through an ss7 link and suspect that it is an hunting policy issue but have a very limited knowledge about it. |
| 10:05.51 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:09.32 | *** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com) |
| 10:14.12 | *** join/#asterisk oej (n=olle@213.136.49.145) |
| 10:19.17 | stix_ | When using the queue show command the output isn't sorted. Can I do that somehow? |
| 10:19.21 | stix_ | by extension |
| 10:21.54 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:22.31 | *** join/#asterisk OhSlap (n=ohslap@202.55.148.56) |
| 10:22.42 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 10:23.59 | disposable | does *1.6 have a larger/smaller set of files in /var/lib/asterisk/sounds/xx/* than 1.2? |
| 10:24.29 | mark_csi | stix: I'm not sure how to do this but I know you can add "like <regular_expression>" to the end of the command |
| 10:26.21 | *** join/#asterisk pcrane (n=pcrane@125-238-255-249.broadband-telecom.global-gateway.net.nz) |
| 10:29.09 | pcrane | Hi guys |
| 10:29.13 | pcrane | how's is everyone? |
| 10:29.57 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 10:33.22 | pcrane | can anyone tell me why callerid(num) is empty for *all* calls in on a PRI? |
| 10:33.31 | pcrane | or where to start looking for why? |
| 10:37.01 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:39.02 | pcrane | I think I answered by own question: |
| 10:39.18 | pcrane | callerid=asreceived should be set to pass the caller id forward |
| 10:40.40 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 10:43.49 | mark_csi | stix: I found a command I used a while back to order sip extensions 'asterisk -rx "sip show peers" | grep 5060 | sort -n" - don't know if this'll help |
| 10:44.24 | stix_ | mark_csi, i'll give it a try |
| 10:45.02 | stix_ | works great, thanks :) |
| 10:45.17 | stix_ | if I set a global var, shut down asterisk and start it again. Will it remember the var's? |
| 10:46.14 | florz | stix_: the var's what? |
| 10:46.59 | stix_ | variable |
| 10:48.25 | *** join/#asterisk ElDios (n=ElDios@213-140-6-112.ip.fastwebnet.it) |
| 10:48.33 | ElDios | hey guys |
| 10:48.45 | ElDios | where do I find the auth IMAP data in kolab? |
| 10:49.03 | ElDios | I need to connect directly to the Cyrus IMAP server but I don't know what AUTH data do I have to use |
| 10:49.07 | ElDios | any idea? |
| 10:49.11 | ElDios | uops |
| 10:49.13 | ElDios | :D |
| 10:49.15 | ElDios | wrong window |
| 10:49.17 | florz | the var's variable? what's that? |
| 10:49.39 | ElDios | sorry florz |
| 10:49.41 | stix_ | florz, you don't know what a variable is? |
| 10:51.28 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 10:51.41 | florz | stix_: I just don't know what a variable belonging to a var is. |
| 10:52.30 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:52.40 | stix_ | florz, okay but do you know that "var" is short for the word variable? |
| 10:52.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:52.52 | florz | stix_: I supposed that. |
| 10:53.22 | stix_ | okay, have you heard of the asterisk command SetGlobalVar ? |
| 10:53.57 | florz | stix_: I think so. |
| 10:54.14 | stix_ | but still you don't understand my question? |
| 10:54.14 | florz | stix_: supposing that you mean the dialplan application? |
| 10:54.50 | florz | stix_: Well, how should I understand - I'm not familiar with the concept of variables having variables. |
| 10:56.10 | stix_ | florz, then I think you shouldn't ask me further questions |
| 10:57.36 | florz | stix_: Actually, I have asked you just one question so far. |
| 10:58.05 | florz | stix_: Didn't intend to ask any further ones ATM. |
| 11:08.30 | puppet | Has anyone ever had a problem with that when ou Answer() in *, * goes on but the call dont really get answered? |
| 11:12.38 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:18.36 | *** join/#asterisk donnib (n=aaaa@0x555281d0.adsl.cybercity.dk) |
| 11:18.40 | donnib | hi people |
| 11:19.11 | donnib | i have two phone which i have disabled qualify but after doing that when i call from one to the other the other phone does not ring |
| 11:19.21 | donnib | both clients sits on the smae network without nat |
| 11:19.37 | donnib | and they both register to the server. i can see that by looking at sip show peers |
| 11:19.59 | donnib | anyone have an idea ? |
| 11:25.01 | mark_csi | donnib: could it be a dialplan problem? |
| 11:25.49 | donnib | no because it works some time. |
| 11:25.54 | donnib | i keep seeing Resending in the log |
| 11:26.30 | donnib | there is a 300 ms latency between these two clients |
| 11:29.03 | puppet | my problem seem to be a dialplan issue, strange, when i removed allow anonymous i go voice |
| 11:33.03 | *** join/#asterisk fprior (n=chatzill@host105.200-43-137.telecom.net.ar) |
| 11:38.12 | fprior | Hi all.I'm new in Asterisk's hardware.I would install a small call center with 8 PSTN incoming line. How choose between PCI card or External gateway ? |
| 11:40.06 | donnib | so i guess no one has other ideas ? |
| 11:49.08 | UnixDawg | morning |
| 11:50.41 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 12:04.57 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 12:09.10 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 12:10.51 | *** join/#asterisk invalidrecord (n=fares@92.40.15.60.sub.mbb.three.co.uk) |
| 12:12.10 | invalidrecord | any of you guys used adhearsion im not sure if its asterisk bit im getting wrong or adhearsion bit, i think this code means any call to an internal extension should run the handler but it dosent seem t0 http://pastie.org/319559 |
| 12:13.28 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:24.05 | *** join/#asterisk homeins6 (n=homeins6@ip-208-109-154-197.ip.secureserver.net) |
| 12:24.26 | homeins6 | If a phone says that it is a single line phone, does that mean it is incapable of 3 way calling or call waiting? |
| 12:26.59 | *** join/#asterisk DarkRift (n=dark@65.92.166.59) |
| 12:27.59 | UnixDawg_ | no |
| 12:28.02 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 12:30.21 | homeins6 | What is the benefit of getting one that has 2 lines? Does that give you the ability to transfer calls, or conference calls? |
| 12:31.42 | puppet | homeins6: transfer, conferense is handled by * |
| 12:31.47 | UnixDawg_ | it gives you a way to put a person on hold and make another call with out the chance of hanging up on them |
| 12:32.01 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 12:32.02 | UnixDawg_ | as you use line 2 to dial out |
| 12:40.04 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 12:41.34 | puppet | hmm, anyone knows why that wehn i have several peers, to same host, it uses just ONE of them at SIP/08XXXXX2375-08d1d9e0" always same doesnt atter who calls in |
| 12:58.33 | puppet | http://pastebin.ca/1262584 my incoming peer looks like this, only diffrence is the username and fromuser, and then the name of the trunk. |
| 12:58.36 | puppet | feels like there is some problem with several trunks to same host |
| 12:59.26 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 13:04.27 | *** part/#asterisk invalidrecord (n=fares@92.40.15.60.sub.mbb.three.co.uk) |
| 13:08.37 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:09.08 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 13:10.38 | fprior | anyone use GrandStream GXW-4108 or GXW-4104 ? |
| 13:13.28 | [TK]D-Fender | ~gs |
| 13:13.29 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:13.32 | [TK]D-Fender | ~grandstream |
| 13:13.33 | jbot | methinks grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 13:13.54 | *** join/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 13:14.03 | *** part/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 13:22.48 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:24.00 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:24.37 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:24.42 | fcois93 | hello all |
| 13:25.05 | fcois93 | I try to insert a header in a ancel sip! have you an idea? |
| 13:25.28 | fcois93 | I know how to insert header in an invite but dont know how to do for a cancel... |
| 13:26.45 | [TK]D-Fender | fcois93: vi chan_sip.c <--- |
| 13:27.17 | fcois93 | [TK]D-Fender: you have a solution? is it possible? you understand what I need? |
| 13:30.47 | fcois93 | [TK]D-Fender: in fact, with openser, I insert a header in the CANCEL, but asterisk don't forward the header to the next user... |
| 13:32.43 | *** join/#asterisk zchaos (n=none@CPE0018f85982ba-CM001ac35b1a7e.cpe.net.cable.rogers.com) |
| 13:32.45 | [TK]D-Fender | fcois93: And what I jsut told you says that you cannot do this in * without recoding the channel driver |
| 13:32.49 | zchaos | anyone here understand how patch panels work for cat5/phone lines? and know how to punch down other ends etc? |
| 13:32.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:32.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:33.25 | [TK]D-Fender | fcois93: * is a B2BUA, not a PROXY. You can't just ADD things.. |
| 13:33.32 | fcois93 | [TK]D-Fender: waouh! I think I cant do it... |
| 13:33.49 | [TK]D-Fender | zchaos: Depends on what kind of path panel |
| 13:33.54 | [TK]D-Fender | patch |
| 13:33.57 | fcois93 | [TK]D-Fender: always the same problem, * is a B2BUA... :( |
| 13:34.43 | [TK]D-Fender | fcois93: Keep that in mind for the next similar request. |
| 13:35.35 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 13:38.38 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:41.05 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 13:41.32 | farah | hello everyone |
| 13:42.10 | farah | i am doing a project on asterisk |
| 13:42.26 | fcois93 | farah: as everyone... |
| 13:42.53 | farah | my project is to perform the call quality |
| 13:43.45 | farah | i configured the .conf files, and i thought to use the command "iax2 show netstats" |
| 13:43.45 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 13:43.52 | fcois93 | farah: use the right codec, a good SDSL access... |
| 13:44.20 | farah | thank you fcois93 |
| 13:44.36 | fcois93 | farah: and need to have 1Adsl for data and 1Sdsl for voice |
| 13:44.46 | farah | my question is about "iax2 show netstats" . |
| 13:45.00 | fcois93 | farah: I don't use IAX |
| 13:45.04 | farah | ah ok |
| 13:45.20 | Carlos_PHX | What's the question about netstats? |
| 13:45.21 | fcois93 | farah: but, know you a proxy IAX as OPENSER... ? |
| 13:45.54 | farah | fcois93: i am a beginner in asterisk so i didnt get your question |
| 13:46.10 | [TK]D-Fender | fcois93: GRAMMAR FAIL :) |
| 13:46.16 | Carlos_PHX | I'm not a beginner and I didn't get his question. |
| 13:47.08 | angryuser | openser can not be used as iax proxy |
| 13:47.10 | fcois93 | in fact, need to have an IAX-proxy same as OPENSER is a SIP-proxy |
| 13:47.22 | fcois93 | angryuser: yes I know :( |
| 13:48.12 | farah | Carlos_PHX: the question is: when i did some tests, i used the command "iax2 show netstats" and i tried it with many configurations of the file iax.conf, and the result changes. So is it better or put on the configuration of iax.conf "jitterbuffer"=yes and "forcejitterbuffer"=yes, or is it better to disable the jitter? |
| 13:48.15 | angryuser | i dont know any product besides asterisk who can do anything in server side for iax |
| 13:48.37 | angryuser | maybe some proprietary |
| 13:49.49 | Carlos_PHX | I wouldn't use that as a pre-config tool, but a troubleshooting tool if you have problems. Generally I would say jitterbuffer=no unless you have a need. |
| 13:50.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:50.16 | Carlos_PHX | If you have good connections that should work. |
| 13:50.27 | [TK]D-Fender | fcois93: What do you need a IAX proxy for? |
| 13:50.35 | farah | Carlos_PHX: When i enable the jitter (jitterbuffer"=yes and "forcejitterbuffer"=yes), i get the following result: voip*CLI> iax2 show netstats |
| 13:50.35 | farah | <PROTECTED> |
| 13:50.35 | farah | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 13:50.35 | farah | IAX2/i55333-2455 1500 3 62 1 0 0 0 0 0 0 0 0 0 0 0 |
| 13:50.35 | farah | 1 active IAX channel |
| 13:50.37 | farah | voip*CLI> iax2 show netstats |
| 13:50.38 | Carlos_PHX | We try to avoid IAX for anything other than special purposes. |
| 13:50.39 | farah | <PROTECTED> |
| 13:50.41 | farah | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 13:50.43 | farah | IAX2/i55333-2455 1500 2 62 1 0 0 0 0 0 0 0 0 0 0 0 |
| 13:50.45 | farah | 1 active IAX channel |
| 13:50.47 | farah | voip*CLI> iax2 show netstats |
| 13:50.49 | farah | <PROTECTED> |
| 13:50.51 | farah | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts |
| 13:50.52 | [TK]D-Fender | farah: PASTEBIN! |
| 13:50.53 | farah | IAX2/i55333-2455 18 2 63 1 0 0 0 1 0 40 0 0 0 0 0 |
| 13:50.55 | farah | 1 active IAX channel |
| 13:50.55 | [TK]D-Fender | ~pb |
| 13:50.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:51.03 | [TK]D-Fender | farah: Do not spam in here like that |
| 13:51.09 | farah | oops sorry |
| 13:51.18 | farah | i am really sorry |
| 13:51.56 | Carlos_PHX | farah: Are you have a sound quality problem? |
| 13:52.05 | fcois93 | [TK]D-Fender: I need to do loadbalancing to others asterisk servers, control some parameters from my customers... |
| 13:52.15 | farah | no i don't |
| 13:52.46 | fcois93 | [TK]D-Fender: somethings that a B2BUA cant do :) |
| 13:52.50 | Carlos_PHX | fcois93: How are you keeping the servers in sync? Calls between servers? |
| 13:53.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:53.08 | *** join/#asterisk botox93 (n=Miranda@213.221.82.242) |
| 13:53.23 | fcois93 | Carlos_PHX: what do you mean with sync ? |
| 13:53.24 | [TK]D-Fender | fcois93: IAX is not designed to be proxied |
| 13:53.34 | botox93 | hello |
| 13:53.49 | Carlos_PHX | You said you have multiple servers. You need to keep the configs across both and allow calls between them. Asterisk realtime? |
| 13:54.27 | farah | when i disable the jitter, i have in the "lost" field a value equal to -1 |
| 13:54.28 | farah | and that seems strange no? |
| 13:54.49 | fcois93 | the config in asterisk is very simple and never change. the changing config is in openser and check in mysql |
| 13:54.59 | farah | i didn't say that i have multiple servers |
| 13:55.16 | fcois93 | farah: it was for me |
| 13:55.23 | farah | ah ok:) |
| 13:55.56 | Carlos_PHX | fcois93: Have you considered using DNS SRV for load balancing? |
| 13:56.41 | fcois93 | Carlos_PHX: to do loadbalancing, with asterisk, I use 'gotoif, sippeer...' |
| 13:57.19 | fcois93 | Carlos_PHX: in fact, I have 2openser and multiple asterisk after, so I need to do loadbalancing openser->asterisk and loadbalancing asterisk->openser |
| 13:57.53 | fcois93 | Carlos_PHX: and that was done without real pbs |
| 13:58.14 | [TK]D-Fender | fcois93: SER is a load-balancer and prozy. Trying to use * as that role is retarded. |
| 13:58.30 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 13:58.42 | dominic1 | hello botox93 |
| 13:59.09 | farah | can someone help me please |
| 13:59.24 | fcois93 | [TK]D-Fender: yes I know but add an other openser to do loadbalancing to others openser isn't a good idea I think |
| 13:59.31 | Carlos_PHX | farah: What's broken? |
| 13:59.48 | [TK]D-Fender | fcois93: You'd be wrong. |
| 14:00.15 | fcois93 | [TK]D-Fender: I don't know |
| 14:00.32 | farah | Carlos_PHX: when i disable the jitter buffer, i get a value equal to -1 in the "lost" field...is it because there is no statistics for this? |
| 14:01.06 | Carlos_PHX | I don't know. Probably not a lot of IAX users in here. But if it's not broken, what are you trying to fix? |
| 14:01.41 | Carlos_PHX | I would guess that without a jitter buffer you can't measure jitter. |
| 14:02.14 | farah | i am doing a trainee and they want me to find a method to improve the call quality even if it's good...they want to be able to detect if there is a failure...and i thought to do this using iax show stats |
| 14:02.25 | *** join/#asterisk Ast001 (n=uros@81.18.55.102) |
| 14:03.22 | Katty | mew. |
| 14:03.27 | Carlos_PHX | Improve call quality even if it's good? Huh? Each CODEC has a baseline call quality level. It can only go down from there if there are network problems or other issues. If there are not, you can't make it better. |
| 14:03.30 | [TK]D-Fender | Katty: Mew |
| 14:03.51 | Carlos_PHX | http://www.speedextreme.com/temp/nov/lolcat_level68.jpg |
| 14:03.55 | Katty | [TK]D-Fender: hmmyeahhi |
| 14:04.06 | Katty | [TK]D-Fender: i think i'm getting ill |
| 14:04.21 | farah | do you think there is a treshold for the % of lost packets with the iax show stats, after which we can say that the quality is not good? |
| 14:04.53 | Katty | Carlos_PHX: heh |
| 14:04.58 | Carlos_PHX | I don't know on show stats specifically. I do know that when our customer see 3-5% loss, it is audible. |
| 14:04.58 | [TK]D-Fender | farah: Place a call, start messing with it, monitor, and judge for yourself |
| 14:04.59 | Katty | Carlos_PHX: onward, to northrend! |
| 14:05.10 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 14:05.24 | Katty | Qwell: 73 :> |
| 14:05.24 | coppice | does improving a good call mean in goes a rough angry voice and out comes the soothing tones of Morgan Freeman? |
| 14:05.40 | farah | I want to find a way to detect automatically if there is a failure |
| 14:05.55 | farah | coppice: lol |
| 14:05.58 | Katty | and i want a millino dollars and a ranch house |
| 14:06.12 | Carlos_PHX | just wants coffee. |
| 14:06.22 | Carlos_PHX | is a man of simple desires. |
| 14:06.30 | Katty | you're a man. |
| 14:06.33 | Katty | that's to be expected. |
| 14:06.42 | Katty | ;> |
| 14:06.55 | Carlos_PHX | However the coffee machine is at least 20 feet away and I have to press the button on it. |
| 14:06.56 | Carlos_PHX | Twice. |
| 14:07.03 | Katty | oh no. |
| 14:07.08 | Carlos_PHX | You can see my dilemma. |
| 14:07.10 | Katty | DOOM |
| 14:08.11 | farah | Carlos_PHX: last question please:) value of lost= -1 what does it mean? |
| 14:08.38 | Carlos_PHX | I don't know, I can only guess that it means there is no data. Without a jitter buffer, I think you can't measure jitter. But I do not know that for sure. |
| 14:09.20 | farah | ok thank you |
| 14:09.32 | Carlos_PHX | farah: Understand that there are products out there for analyzing call quality, most pretty expensive. Because there aren't simple stats that will tell you how call quality is. |
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| 14:10.06 | farah | Carlos_PHX: yes i know...but my diploma project is about this so i don't have the choice |
| 14:10.07 | Carlos_PHX | Katty: I made the trek and pressed the button. Mmmmm |
| 14:10.14 | Katty | you brave soul. |
| 14:10.27 | Carlos_PHX | It was like...dark out there. And cold. |
| 14:10.37 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:10.44 | Katty | :< |
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| 14:14.03 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 14:15.12 | [TK]D-Fender | Carlos_PHX: 20 feet away? Thats some nasty localized weather you have there... |
| 14:15.29 | [TK]D-Fender | Carlos_PHX: wait.... I feel a Micro-Burst coming on... |
| 14:15.32 | [TK]D-Fender | waffles |
| 14:16.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:16.11 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:16.32 | iCEBrkr | looks around and yawns |
| 14:16.38 | Carlos_PHX | Well, it is in another room. |
| 14:16.45 | Carlos_PHX | Computer room/office = warm |
| 14:16.50 | Carlos_PHX | Kitchen = cold |
| 14:17.28 | Carlos_PHX | which flavor of Slim Jim goes best with coffee? They're not just for breakfast any more. |
| 14:17.56 | iCEBrkr | So in voicemail.conf there's a emailcmd option.. What's the default path it'll look for that script? or do I need to full path it? |
| 14:18.24 | *** join/#asterisk dhill (i=dhill@dhcp-222.iserv.net) |
| 14:18.43 | dhill | Dial(Local/XXX@context) |
| 14:18.45 | dhill | chan_local.c:617 local_alloc: No such extension/context |
| 14:18.49 | dhill | which is correct |
| 14:19.17 | dhill | anyway I can create a GotoIf when there is no extension? |
| 14:19.34 | dhill | Dial returns CHANUNAVAIL |
| 14:19.36 | iCEBrkr | how about an 'i' extension? |
| 14:20.20 | dhill | ok, let me give that a shot |
| 14:20.38 | iCEBrkr | exten => i,1,Playback(invalid) or something |
| 14:22.51 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 14:23.41 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:24.07 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:24.57 | farah | iax is developped by default with udp...but is it compatible with tcp too? |
| 14:25.32 | coppice | farah: no streaming media protocol will behave well over TCP |
| 14:26.01 | farah | ok thank you |
| 14:27.21 | *** join/#asterisk SuPrSluG (n=SuPrSluG@72.237.213.90) |
| 14:27.28 | Carlos_PHX | IAX and TCP, that would be...interesting |
| 14:27.45 | dhill | iCEBrkr: not working.. hmm |
| 14:28.26 | Carlos_PHX | coppice: You happen to know anything about T.38 with Vitelity? Going to give it a try later in the week, but wonder if you have any experience with them. |
| 14:28.48 | RMod | vitelity doesnt offer t38 |
| 14:29.16 | *** join/#asterisk Great_Anta_Baka (i=c4249986@gateway/web/ajax/mibbit.com/x-e0387f26687c8856) |
| 14:29.18 | coppice | RMod seems to know far more than me :-) |
| 14:29.48 | RMod | not at all, just know vitelity doenst offer t38 =) |
| 14:29.59 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 14:29.59 | Carlos_PHX | Vitelity doesn't offer it, but does do it. |
| 14:30.12 | Carlos_PHX | "Not supported, best effort..." |
| 14:30.19 | coppice | ah, so Carlos is the true expert |
| 14:30.30 | Carlos_PHX | Since we are going to do some voice with them, would like to try T.38. |
| 14:30.46 | Carlos_PHX | We have yet to settle on a T.38 carrier, though testing is going really well with ipcomms.net. |
| 14:31.14 | RMod | broadvox offers t38 term. to all destinations |
| 14:31.17 | Carlos_PHX | Vitelity is on the same network as Broadvox. |
| 14:31.34 | coppice | Carlos_PHX using what at your end? |
| 14:31.40 | Carlos_PHX | Broadvox's services and plans don't really fit our needs, though we looked at them carefully. |
| 14:31.46 | mort_gib | does anyone have any advice on ATA's ?? |
| 14:31.49 | Carlos_PHX | coppice: Asterisk 1.6 |
| 14:31.58 | mort_gib | I have used Zyxel but.... |
| 14:31.58 | Carlos_PHX | mort_gib: SPA 2102 |
| 14:32.57 | coppice | mort_gib: don't trust anything they say on a ATA's carton |
| 14:33.28 | mort_gib | coppice: I know, the Zyxels worked fine though |
| 14:33.34 | Carlos_PHX | Heh, yeah, don't trust anything any box says about any SIP device... |
| 14:33.43 | mort_gib | :-) |
| 14:34.09 | mort_gib | I need to be able to connect it to a pstn line too |
| 14:34.36 | coppice | spa3102 |
| 14:35.35 | Great_Anta_Baka | anyone have any luck with getting snom presence to work with asterisk? |
| 14:36.39 | mort_gib | coppice: Thanks |
| 14:36.48 | mort_gib | Carlos_PHX: Thanks |
| 14:37.04 | mort_gib | Great_Anta_Baka: Yes |
| 14:37.06 | Carlos_PHX | I'm new with ATAs altogether, but have been really happy with the ease/reliablity of the Linksys. |
| 14:37.35 | mort_gib | Yeah, well this one is slightly odd. This is not really for use with asterisk |
| 14:37.35 | Great_Anta_Baka | mort_gib: any advice you can give me |
| 14:37.40 | coppice | mort_gib: worked fine within the limits of what you asked them to do would be more accurate. I have seem multiple ATAs which say T.38 on the box and have no T.38 support at all. if you don't try to use that feature, you'll probably never realise its missing |
| 14:37.52 | mort_gib | Eh. what do you want to know?? |
| 14:37.53 | Great_Anta_Baka | i am on snom firmware version 7.130 |
| 14:38.04 | Carlos_PHX | coppice: So, which ones do you like for T.38, since that's my next project. |
| 14:38.18 | Great_Anta_Baka | but cant get my lights to light up when someone is on a call |
| 14:38.19 | Carlos_PHX | Once we get on a carrier for the back end, deploy ATAs to client sites. |
| 14:39.13 | mort_gib | Great_Anta_Baka: Do you use hint in extensions.conf |
| 14:39.25 | Great_Anta_Baka | i dnt think so |
| 14:39.30 | Great_Anta_Baka | where do i put it? |
| 14:40.19 | [TK]D-Fender | Great_Anta_Baka: go read up on presence on the WIKI |
| 14:40.23 | mort_gib | 1. Use hint in extensions.conf 2. use friend rather than peer in sip.conf 3. Map Extensions to buttons on the phon |
| 14:40.34 | mort_gib | HI TK |
| 14:40.34 | [TK]D-Fender | Great_Anta_Baka: you need to set up your dialplan hints |
| 14:40.46 | Great_Anta_Baka | i see |
| 14:40.53 | *** join/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
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| 14:49.30 | rue_mohr | any comments on the aastra 9143 for a small office? |
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| 14:52.07 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 14:52.53 | rue_mohr | can I have a second call ring in on a line thats already being used? |
| 14:53.05 | iCEBrkr | Like call waiting? |
| 14:53.21 | rue_mohr | no, like actaully ring |
| 14:53.37 | iCEBrkr | umm, if you have a second line appearance. |
| 14:54.08 | rue_mohr | k, so I need atleast 2 line appearances to have calls ring in while someone is using them |
| 14:54.36 | iCEBrkr | If line 1 is in use and you send a call to it, it'll work like call waiting.. |
| 14:54.52 | iCEBrkr | But if it's a multi-line phone, you'll have to setup individual sip accounts for each line. |
| 14:55.02 | Great_Anta_Baka | [TK]D-Fender: i have set it up like this exten = 201,hint,SIP/201 |
| 14:55.03 | rue_mohr | I'm trying to understand how a receptionist of say 4 incomming numbers works for a phone with less than 4 appearances ( the aastra 9143 is 3) |
| 14:55.23 | Great_Anta_Baka | but still no luck :/ |
| 14:55.34 | iCEBrkr | rue_mohr: you'll have to have dial-plan logic to route the calls to the correct lines |
| 14:55.53 | Akiyuki | Anyone know if fwrite() is acceptable method for creating call files in /var/spool/asterisk/ ? |
| 14:56.04 | rue_mohr | iCEBrkr, the page for that phone says 9 calls simotanious, so I take it will accept 3 calls on each appearance |
| 14:56.47 | iCEBrkr | ummmm |
| 14:56.52 | brad_mssw | I'm trying to switch from zaptel to dahdi ... but dahdi won't start, error message is: line 0: Unable to open master device '/dev/dahdi/ctl' ... I don't have a /dev/dahdi/*, I instead have /dev/dahdictl, /dev/dahdichannel, /dev/dahdipseudo, /dev/dahditimer |
| 14:56.52 | iCEBrkr | sure |
| 14:57.07 | brad_mssw | how do I let dahdi know this? |
| 14:57.18 | iCEBrkr | rue_mohr: We create sip accounts like 100, 100a, 100b, 100c for a 4 line phone. |
| 14:57.38 | iCEBrkr | rue_mohr: and then we have logic in the dial-plan to route to those lines if one is busy. |
| 14:57.48 | rue_mohr | k, do you use 4 line sip phones for the situation I'm talking about |
| 14:58.00 | iCEBrkr | Yeah, Polycoms |
| 14:58.47 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 14:59.10 | rue_mohr | 4+ lines seem hard to find, are they acually 'lines' or are you using any tricks |
| 14:59.31 | rue_mohr | polycom > aastra? |
| 14:59.42 | iCEBrkr | For sure |
| 14:59.45 | rue_mohr | hmm |
| 14:59.53 | rue_mohr | $polycom > $aastra? |
| 14:59.56 | iCEBrkr | THough, I heard Aastra's have improved on quality |
| 15:00.04 | rue_mohr | hmm |
| 15:00.15 | stintel | our Aastra's rock! |
| 15:00.17 | mort_gib | Snoms will do the job too |
| 15:00.28 | iCEBrkr | rue_mohr: Polycoms are best known for their sound quality -- and general overall hardware quality |
| 15:00.35 | rue_mohr | stintel, tell me, how many incomming numbers do you have? |
| 15:00.44 | [TK]D-Fender | rue_mohr: in north America Polycom is pretty much par on cost with Aastra |
| 15:00.53 | iCEBrkr | We did a 500+ Aastra install and we sent back at least 110 of them after a week of use. |
| 15:00.53 | espent | anybody running zaptel/digium card with freebsd on amd64? |
| 15:00.56 | Akiyuki | Or, does anyone know of a way to issue a call through the CLI? Or asterisk management interface? |
| 15:01.08 | [TK]D-Fender | Aastra's are unstable and I have a large # of "hate" points for the 5i series |
| 15:01.36 | [TK]D-Fender | Akiyuki: cli "originate". AMI "originate" |
| 15:01.37 | stintel | rue_mohr: don't know by heart, but at least 8 configurable, iirc |
| 15:01.51 | [TK]D-Fender | Polycom > all |
| 15:01.59 | iCEBrkr | haha |
| 15:02.00 | rue_mohr | hmm, the aatra 9143 is an i33 I think... |
| 15:02.19 | iCEBrkr | [TK]D-Fender: I dunno.. Snom's are coming along :P |
| 15:02.28 | [TK]D-Fender | iCEBrkr>rue_mohr: We create sip accounts like 100, 100a, 100b, 100c for a 4 line phone. <--- YUCK |
| 15:02.33 | iCEBrkr | hehe |
| 15:03.12 | rue_mohr | [TK]D-Fender, k, your reccommended way of doing reception for 4 different phone numbers? |
| 15:03.13 | [TK]D-Fender | iCEBrkr: In the race to "most unstable"? ;) |
| 15:03.21 | iCEBrkr | [TK]D-Fender: Our system 'auto-magically' creates the stuff based on the model of the phone :) |
| 15:04.37 | rue_mohr | I'm working with people used to nortel isdn stuff, and who currently have 6 slt's on 3 lines |
| 15:04.55 | Carlos_PHX | Akiyuki: You can certainly make a call through AMI. |
| 15:05.00 | [TK]D-Fender | rue_mohr: What does the user need to know? |
| 15:05.02 | Carlos_PHX | I don't know the commands off the top of my head. |
| 15:05.31 | [TK]D-Fender | Carlos_PHX: I just answered both questions... |
| 15:06.06 | [TK]D-Fender | 9143i seems identical to the 9133i except for an XML browser (on a shit screen) and a $20+ price hike |
| 15:06.14 | rue_mohr | [TK]D-Fender, there is a push to get equpiment, I need to know if aatra 9143's are gonna work for their 4 line ( 4 seperate bussnisses answered by up to 2 receptionists ) setup |
| 15:06.29 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:06.39 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 15:06.44 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 15:06.48 | [TK]D-Fender | rue_mohr: So they want the line-key to really represent the division of origin? |
| 15:06.50 | rue_mohr | or if it shoudl be switched to something else |
| 15:07.01 | lmadsen | I like how people assume there is good recommendations about hardware for their application on IRC... I think people should actually test hardware to determine if it suits their needs |
| 15:07.02 | rue_mohr | [TK]D-Fender, they will udnerstand that best |
| 15:07.07 | [TK]D-Fender | rue_mohr: In that case I highly recommend a Polycom IP 6XX |
| 15:07.18 | Akiyuki | Carlos_PHX: Can you put me at the documentation? I cant find it. |
| 15:07.28 | lmadsen | but that's just me :) |
| 15:07.33 | [TK]D-Fender | ~book |
| 15:07.33 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 15:07.37 | rue_mohr | we have to keep them simple looking phones too, thats part of the reason for the selection of the aastra 91x3 |
| 15:07.40 | [TK]D-Fender | Akiyuki: Go read the chapter on AMI. And on the WIKI |
| 15:07.55 | Carlos_PHX | Akiyuki: Hve you looked here? http://www.voip-info.org/wiki-Asterisk+manager+API |
| 15:08.15 | Akiyuki | No, I didnt. I was just googling randomly. |
| 15:08.22 | Carlos_PHX | I'm guessing you're looking at it for the predictive dialing? |
| 15:08.30 | [TK]D-Fender | rue_mohr: I'd be VERY sure about the call-handling capabilities on the Aastra... they are very simple phones and their division of lines and lock of interface could mean you can't handle multiple lines per "key". |
| 15:08.32 | Akiyuki | Is AMI a different package/installation? |
| 15:08.38 | [TK]D-Fender | rue_mohr: Which is Polycom's strong suit |
| 15:08.44 | [TK]D-Fender | Akiyuki: No. |
| 15:08.49 | [TK]D-Fender | Akiyuki: Go read the book. |
| 15:08.51 | Akiyuki | Carlos_PHX: Yes, instead of generating the .call files |
| 15:08.52 | rue_mohr | ok |
| 15:08.53 | Carlos_PHX | Asterisk manager is just there, waiting. |
| 15:09.15 | Akiyuki | PHP's fwrite seems to write 1 line at a time like copy |
| 15:09.18 | Carlos_PHX | Akiyuki: Yes, I thought I would discuss that with you when we have our call. I am very interested in what you've done so far, but believe you can do more with manager interface. |
| 15:09.23 | [TK]D-Fender | rue_mohr: A base IP 600 can juggle 8 calls off 6 line-keys. |
| 15:09.33 | [TK]D-Fender | rue_mohr: 8 calls EACH |
| 15:09.50 | Akiyuki | Carlos_PHX: Sounds good. Just waiting for Ryan to get here. |
| 15:09.57 | rue_mohr | ok, their a little more desk realestate |
| 15:09.59 | Akiyuki | You know how those IT guys are :P |
| 15:10.09 | Carlos_PHX | ROFL |
| 15:10.19 | Carlos_PHX | Yeah, it's a true anomaly that I'm up this early myself. |
| 15:10.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:10.33 | Carlos_PHX | Manager interface is like connecting to an SMTP server and issuing commands in a way. |
| 15:10.55 | Carlos_PHX | We use it one one call center app to dial from an analog dialer and create two SIP channels. |
| 15:11.00 | Akiyuki | ah ok, that would probably be better, as we were thinking of building a webservice request from the machine that hosts our web server to our asterisk box, etc |
| 15:11.20 | Carlos_PHX | There's a web server in Asterisk also that can do that. |
| 15:11.30 | Carlos_PHX | I've only barely played with it. |
| 15:11.36 | Akiyuki | Ah thats neat. |
| 15:11.36 | mikealeonetti | damn |
| 15:11.42 | mikealeonetti | people are jerks |
| 15:11.45 | Carlos_PHX | Akiyuki: http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) |
| 15:11.55 | [TK]D-Fender | EW |
| 15:12.03 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:12.17 | rue_mohr | mikealeonetti, with the accpetion of all of us? |
| 15:12.36 | Carlos_PHX | Akiyuki: Security is a huge concern with this. It's not horribly INsecure, but not great either. |
| 15:12.41 | mikealeonetti | rue_mohr: for the most part :P |
| 15:12.54 | [TK]D-Fender | rue_mohr: yes.. we accept jerks here :) |
| 15:12.54 | rue_mohr | goood stuff |
| 15:12.59 | rue_mohr | arg |
| 15:14.54 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 15:16.34 | *** join/#asterisk mog (n=mog@nat/digium/x-5acd40cb56dec29a) |
| 15:16.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:16.52 | Akiyuki | Carlos_PHX: Well, its all internal at this point. |
| 15:17.26 | [TK]D-Fender | Akiyuki: AMI from your web server to *. Leave them separate |
| 15:17.34 | [TK]D-Fender | (if they already are) |
| 15:17.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:18.25 | Akiyuki | They are |
| 15:18.50 | Akiyuki | I was trying to reinvent the wheel by running apache + php files to listen to $_POST and build .call files, but this is so much simpler |
| 15:18.51 | [TK]D-Fender | Akiyuki: Keep them that way then. |
| 15:19.52 | [TK]D-Fender | Akiyuki: AMI calls = much simpler. No drive mapping, file system issues, etc |
| 15:22.44 | rue_mohr | how do you mean distict ring detect not reliable? |
| 15:23.24 | rue_mohr | what makes it worse is they have call waiting on the analog lines that have the second number with the distinctive ring |
| 15:23.50 | rue_mohr | I can quite easily say that the standard nortel isdn system CAN NOT handle their setup |
| 15:23.59 | [TK]D-Fender | HOLY MCFUCK |
| 15:24.15 | [TK]D-Fender | Nothing can |
| 15:24.35 | [TK]D-Fender | they guy thinks a Ma Bell line can do magic and they are WRONG. |
| 15:24.41 | [TK]D-Fender | this entire setup is retarded |
| 15:24.42 | rue_mohr | well, the call waiting isn't a problem, they can deal with it as part of the main call |
| 15:24.57 | Akiyuki | When I try to visit the asterisk demo page, it loads, but gives an alert with 404 not found on any actions. |
| 15:25.03 | [TK]D-Fender | rue_mohr: except they have no idea who to answer as <- |
| 15:25.05 | rue_mohr | and it "works" between them begging me to fix it |
| 15:25.13 | rue_mohr | good point |
| 15:25.28 | rue_mohr | thats prolly how that call got mixed up yesterday... |
| 15:25.29 | *** join/#asterisk stmaher (n=stephen@mateus.province5.tv) |
| 15:25.31 | stmaher | Hello everyone.. |
| 15:25.39 | [TK]D-Fender | rue_mohr: and I missed that we just took this public :) |
| 15:25.46 | stmaher | Im trying to get paging working with a poly com 430 hard IP phone.. |
| 15:25.57 | rue_mohr | I find it annoying when people detract me from a channel |
| 15:26.17 | [TK]D-Fender | stmaher: you need to set up your Alert info in provisioning, and set the header before dialing the phone. |
| 15:26.29 | [TK]D-Fender | stIts documented on the WIKI. read up. |
| 15:26.29 | [TK]D-Fender | ~WIKIS |
| 15:26.29 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:26.39 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-9bd21b276a6a76f1) |
| 15:26.39 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:26.50 | stmaher | [TK]D-Fender yep.. tried the wiki and everything.. not working. |
| 15:27.10 | [TK]D-Fender | stmaher: PASTEBIN is your friend... show us what you've done. |
| 15:27.12 | stmaher | [TK]D-Fender i realise you have to send a sip alert info header.. which is send to the phone via an tcpdump |
| 15:27.12 | [TK]D-Fender | ~pb |
| 15:27.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:27.19 | [TK]D-Fender | ^^^^^^^^^^^ |
| 15:28.00 | [TK]D-Fender | stmaher: And you adjusted the "alert info" stanza in your provisioning? |
| 15:28.18 | donnib | i have this question before..i have problems that my call reaches a client that is on the same network. is there anyway to find out why ? |
| 15:29.00 | rue_mohr | [TK]D-Fender, ok, off to work |
| 15:29.00 | donnib | might sound weird but i can explain |
| 15:29.00 | Akiyuki | http://localhost:8088/asterisk/manager?action=Login&username=admin&secret=secret5 gives me a 404, so do most of the urls on the AMI |
| 15:29.16 | [TK]D-Fender | donnib: Hope so, because we have no details yet |
| 15:29.19 | stmaher | [TK]D-Fender thanks http://www.pastebin.ca/1262679 |
| 15:29.34 | [TK]D-Fender | Akiyuki: FORGET AJAM. Just do an AMI socket connect |
| 15:30.05 | [TK]D-Fender | Akiyuki: you are asking for pain. |
| 15:30.29 | Akiyuki | ok |
| 15:30.37 | Akiyuki | Nothing is running on port 5083 |
| 15:30.39 | Akiyuki | er |
| 15:30.42 | Akiyuki | 5038 |
| 15:31.07 | [TK]D-Fender | Akiyuki: Well go look at your config file then. |
| 15:31.25 | donnib | ok. have 3 phones on same network (different subnets). two phones are in denmark, one is in india. we are running a vpn line between two locations. |
| 15:31.32 | Akiyuki | port = 5038 |
| 15:31.53 | donnib | the server is in denmark. all registers fine. all have qualify disabled and nat as well. |
| 15:31.56 | [TK]D-Fender | Akiyuki: PASTEBIN... don't jsut spit out 2-word bits and pieces |
| 15:32.28 | [TK]D-Fender | donnib: Please describe the netwoking on BOTH ends |
| 15:32.28 | donnib | i have a 300 ms roundtrip between india and DK. if i am running without qualify the phone in india does not ring when i call from DK |
| 15:33.21 | donnib | same network but different subnets. network are connected with VPN thru a 2mbps line |
| 15:33.45 | [TK]D-Fender | donnib: pastebin the SIP debug of a failed call. |
| 15:33.47 | [TK]D-Fender | ~pb |
| 15:33.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:34.02 | donnib | my problem is that i keep seeing Resending in the asterisk as the packets does not get to the other end but i can get to the webserver of the client. |
| 15:34.07 | donnib | hang on |
| 15:34.22 | stmaher | [TK]D-Fender did you forget about me :-( |
| 15:34.23 | Akiyuki | [TK]D-Fender: http://pastebin.ca/1262682 |
| 15:35.38 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:36.41 | [TK]D-Fender | stmaher: <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class="" voIpProt.SIP.alertInfo.2.value="Ring Answer" voIpProt.SIP.alertInfo.2.class="4" voIpProt.SIP.alertInfo.3.value="Auto Answer" voIpProt.SIP.alertInfo.3.class="3"/> |
| 15:36.42 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 15:36.47 | raasdnil | evening all |
| 15:37.02 | [TK]D-Fender | stmaher: And put your ring answer class back to normal |
| 15:38.31 | stmaher | [TK]D-Fender ???? |
| 15:38.56 | stmaher | [TK]D-Fender thats what the webpages are sayin to do.. |
| 15:39.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:41.29 | raasdnil | hey all, question. How to provide a filler in the silence that follows Asterisk routing the call to a channel and preceeds the called party ring tone that gets generated. Some phone companies I have called have short tones about half a second apart to let the user know something is happening, and then it changes to ring or busy as appropriate... |
| 15:41.47 | raasdnil | there should be a question mark somewhere in there. :) |
| 15:41.53 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 15:42.09 | raasdnil | Proceed, Backgroup, and Dial(m) don't seem to foot the bill |
| 15:43.59 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:44.25 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 15:44.32 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 15:44.39 | donnib | http://pastebin.com/d332d2183 |
| 15:45.39 | *** join/#asterisk ManxPower (n=manxpowe@221.sub-75-251-173.myvzw.com) |
| 15:45.50 | raasdnil | hmmm.... found playtones... looks like it |
| 15:46.00 | *** part/#asterisk ManxPower (n=manxpowe@221.sub-75-251-173.myvzw.com) |
| 15:46.08 | *** join/#asterisk ManxPower (n=manxpowe@221.sub-75-251-173.myvzw.com) |
| 15:46.46 | Akiyuki | [TK]D-Fender: Is ther any thing you can think of as to why this wont bind to 5038? netstat doesnt show anything listening there |
| 15:47.24 | Carlos_PHX | Assuming you enabled it in manager.conf? |
| 15:47.42 | Carlos_PHX | Then restart Asterisk |
| 15:47.54 | Akiyuki | It has been restarted and enabled |
| 15:47.56 | neurosys | <PROTECTED> |
| 15:48.17 | ManxPower | neurosys: I autoload and then noload the modules I don't want. |
| 15:48.22 | Akiyuki | stupid centos packages |
| 15:48.25 | *** part/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 15:48.25 | ManxPower | Asterisk modules are VERY interdependent |
| 15:49.09 | neurosys | ManxPower: The dependencies are why i was afraid to turn off autoload. Thats a good idea. :) thanks |
| 15:49.27 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:49.51 | [TK]D-Fender | akiyuka enabled = no <-- maybe THIS has something to do with it? :p |
| 15:50.02 | [TK]D-Fender | Twit had to leave... SHEESH |
| 15:52.07 | donnib | [TK]D-Fender : did u see my pastebin ? |
| 15:52.50 | neurosys | This may be a newb question, But in Linux, How do you activate scrollback? |
| 15:52.53 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 15:53.01 | [TK]D-Fender | neurosys: WHERE? |
| 15:53.12 | [TK]D-Fender | neurosys: this is a client question... |
| 15:53.14 | neurosys | In Freebsd, i just hit scrollock and can scroll back at the shell. |
| 15:53.28 | ManxPower | I just use an xterm or putty, they come with scrollback |
| 15:53.57 | neurosys | ManxPower: Yeah, im at the machine itself. I could ssh into it.. but im at the screen itself. |
| 15:54.46 | Carlos_PHX | Anyone aware of a wi-fi VoIP phone with video? |
| 15:55.35 | neurosys | [TK]D-Fender: Ok ok.. I'll just ssh into the box and use scrollback. :-P |
| 15:55.42 | [TK]D-Fender | neurosys: Ctrl PgUp/Ctrl PgDn |
| 15:56.14 | [TK]D-Fender | neurosys: http://www.linuxforums.org/forum/linux-tutorials-howtos-reference-material/2531-keyboard-shortcuts-x-windows-command-line.html |
| 15:56.24 | [TK]D-Fender | neurosys: When in doubt, JFGI |
| 15:56.26 | neurosys | [TK]D-Fender: I'm not in X. |
| 15:56.35 | [TK]D-Fender | neurosys: Neither am I <- |
| 15:56.42 | ManxPower | An XWindows command line, is that anything like a green orange? |
| 15:57.17 | neurosys | ManxPower: Wel. that page is working with xterm. |
| 15:57.54 | neurosys | [TK]D-Fender: Shift-PgUp/Down |
| 15:58.18 | neurosys | [TK]D-Fender: Promise to JFGI nexttime ;) |
| 15:59.49 | [TK]D-Fender | donnib: check your firewalls, and youting. |
| 15:59.54 | [TK]D-Fender | routing* |
| 16:00.08 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 16:02.02 | stmaher | [TK]D-Fender heya.. any thoughts on the polycom issue? |
| 16:02.24 | [TK]D-Fender | stmaher: I told you want to do... |
| 16:02.53 | stmaher | [TK]D-Fender could you please explain it abit better? |
| 16:03.31 | [TK]D-Fender | stmaher: better? I gave you a complete replacement line to cut & paste and told you another line to put back into stock condition. What more is there to say? |
| 16:03.34 | [TK]D-Fender | stmaher: 2 lines. |
| 16:03.51 | [TK]D-Fender | stmaher: My instructions aren't Raw-Cat Science. |
| 16:04.06 | mikealeonetti | [TK]D-Fender: dude |
| 16:04.29 | neurosys | Always so testy. :( |
| 16:04.38 | [TK]D-Fender | aims his Raw-Cat Lawn Chair @ mikealeonetti |
| 16:04.53 | stmaher | [TK]D-Fender Thanks Ill try that |
| 16:05.06 | raasdnil | is trying to get some playtones working... any ideas on what I am stuffing up? http://www.pastebin.ca/1262711 |
| 16:06.15 | *** join/#asterisk pwasek (n=pwasek@pool-96-246-173-146.nycmny.fios.verizon.net) |
| 16:06.17 | [TK]D-Fender | raasdnil: pastebint he ACTUAL CLI output, and "R" should override your playtones... |
| 16:06.23 | [TK]D-Fender | raasdnil: that enforces ringing... |
| 16:06.46 | mikealeonetti | lol |
| 16:07.26 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:08.31 | raasdnil | [TK]D-Fender: that is the CLI output. Just not all of it. one sec, I'll dump it all there |
| 16:08.44 | stmaher | [TK]D-Fender Thanks! that worked perfectly! |
| 16:09.12 | [TK]D-Fender | raasdnil: Yeah... we're only missing proof about what was actually called and the dialplan to back up that the local channel is valid too ;) |
| 16:09.19 | [TK]D-Fender | stmaher: You're welcome. |
| 16:09.39 | [TK]D-Fender | stmaher: Next step for you is probably "Page" |
| 16:09.45 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-44-197.w86-215.abo.wanadoo.fr) |
| 16:09.46 | *** join/#asterisk jeffspeff (i=Administ@c-98-240-113-191.hsd1.ky.comcast.net) |
| 16:09.49 | raasdnil | [TK]D-Fender: bah, mere technicalities :) |
| 16:09.54 | [TK]D-Fender | stmaher: Chain up a bunch of them and scream bloody moneys to everyone |
| 16:10.00 | [TK]D-Fender | monkeys* |
| 16:10.15 | mikealeonetti | is there a way to monitor calls on other extensions to see if you want to receive them? |
| 16:10.25 | raasdnil | [TK]D-Fender: oh.. hang on a sec... 'dialplan to back up that the local chanel is valid too'.... I think one of those intelligent light bulbs just glimmered to life above my head |
| 16:10.32 | raasdnil | has no dialplan for Local |
| 16:10.33 | [TK]D-Fender | raasdnil: Concorde & Challenger had "technicalities" too... wanna pick up your boarding pass now? ;) |
| 16:10.46 | [TK]D-Fender | raasdnil: :p |
| 16:10.46 | raasdnil | hah! |
| 16:10.57 | raasdnil | no thanks. Titanic has my seat already :/ |
| 16:11.10 | [TK]D-Fender | mikealeonetti: Depends how you want to "monitor" them |
| 16:11.20 | [TK]D-Fender | raasdnil: Your ship has come in! |
| 16:11.23 | *** join/#asterisk farah (n=fauf@11-71-72-193.adsl.switzerland.net) |
| 16:12.04 | mikealeonetti | [TK]D-Fender: well, just to see what calls are coming in to the company in real time |
| 16:13.28 | ManxPower | mikealeonetti: "asterisk -rvvv" |
| 16:13.40 | ManxPower | You would know that if you had read the Asterisk book. |
| 16:13.40 | [TK]D-Fender | mikealeonetti: before ringing phones maybe call a script that will push out an IM or something. maybe a script that will just page the number read out over speakers. Maybe just ring MULTIPLE phones at once. |
| 16:14.06 | raasdnil | ok... where do i define the Local channel then? chan-local.conf? |
| 16:14.08 | [TK]D-Fender | ManxPower: Nobody wants to monitor * CLI jsut to see calls come in... |
| 16:14.13 | [TK]D-Fender | ManxPower: in a user environment. |
| 16:14.20 | [TK]D-Fender | ManxPower: this is for selective pickup.. |
| 16:14.33 | [TK]D-Fender | raasdnil: No, it jsut points to something in your dialplAN. |
| 16:14.43 | [TK]D-Fender | raasdnil: Local/exten@context |
| 16:14.50 | raasdnil | oh, I get it. |
| 16:14.52 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 16:14.53 | [TK]D-Fender | raasdnil: and I advise "/n" at the end |
| 16:15.04 | raasdnil | yeah, I read that bit. |
| 16:15.18 | [TK]D-Fender | raasdnil: What are you doing that you don't know how to use it, what its for, and yet is stillin your dialplan in the first place? |
| 16:15.34 | [TK]D-Fender | raasdnil: Inherited config? |
| 16:15.57 | raasdnil | no... experimenting from the voip-info wiki on a test box |
| 16:16.02 | mikealeonetti | [TK]D-Fender: is it possible to ring a group of people, but have that call clearly come in saying it's for somebody else and then if you want to take that call having to press a confirmation key? |
| 16:16.07 | stmaher | [TK]D-Fender How do I make it play monkeys right after i make an auto answer to the phone? |
| 16:16.33 | mikealeonetti | that was more an in general question actual |
| 16:16.38 | mikealeonetti | didn't mean to direct it at [TK]D-Fender |
| 16:16.50 | mikealeonetti | but I"m sure he likes the attention |
| 16:17.16 | [TK]D-Fender | raaWIKI is often outdated and something flat out wrong... |
| 16:17.44 | [TK]D-Fender | stmaher: Go read up on "call files" and "AMI originate" and you'll see how to direct them to it |
| 16:17.53 | stmaher | thanks! |
| 16:18.14 | [TK]D-Fender | stmaher: I fired up a 20-person simultaneous tt-monkeys page in my office. Freaked the living shit outta people. |
| 16:18.47 | [TK]D-Fender | stmaher: Next time, I'm going to do tt-driveby with the sounds of automatic weapons fire and breaking glass |
| 16:18.50 | stmaher | [TK]D-Fender You are psyhic!.. you read my mind :-) |
| 16:19.06 | stmaher | [TK]D-Fender LOL.. |
| 16:19.32 | mikealeonetti | what's up pussy cat |
| 16:19.36 | mikealeonetti | woah woah woah woah |
| 16:19.51 | stmaher | Ill have one for each day of the week |
| 16:19.53 | stmaher | Blue monday |
| 16:19.57 | stmaher | Happy tuesday |
| 16:20.03 | stmaher | (dont know one for wed or thur) |
| 16:20.07 | stmaher | Im so excited for friday! |
| 16:20.17 | [TK]D-Fender | mikealeonetti: dial multiple local channels with M() called for the non-primary contact |
| 16:20.22 | raasdnil | [TK]D-Fender: thanks... |
| 16:20.27 | raasdnil | seeya all (off to sleep) |
| 16:20.37 | [TK]D-Fender | raanp. All working? |
| 16:20.43 | mikealeonetti | [TK]D-Fender: M(), hrm. lemme look it up |
| 16:24.05 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
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| 16:24.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 16:29.48 | mikealeonetti | OK. So M() executes a macro when it connects with the phone (presumably when picked up). Then I guess I just have to create a macro that has an extension that when it is pressed it Answer()s. Then I just modify the caller ID to show where it is going. |
| 16:29.57 | mikealeonetti | that's pretty brilliant |
| 16:31.03 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 16:34.05 | *** join/#asterisk mathcubes (n=chatzill@host81-149-211-7.in-addr.btopenworld.com) |
| 16:36.57 | *** join/#asterisk dushantch (n=chatzill@adsl-ppp-2169.yubc.net) |
| 16:38.08 | dushantch | Hi, why this works : exten => _#,1,Dial(SIP/4), and this: exten => #,1,Dial(SIP/4) |
| 16:38.33 | dushantch | and exten => #4,1,Dial(SIP/4) but not this exten => _#4,1,Dial(SIP/4) ? |
| 16:38.47 | ManxPower | dushantch: both should work, you must be doing something else wrong. The CLI will show you |
| 16:38.54 | [TK]D-Fender | dushantch: because the "_" in front alny says that reserved patter chars are not to be take literally. |
| 16:39.23 | [TK]D-Fender | dushantch: And both patterns are OK. |
| 16:39.31 | [TK]D-Fender | dushantch: What are you calling that # using? |
| 16:39.36 | ManxPower | Well if you have both # and #4 how does the phone or Asterisk know what you are dialing. But without a pastebin of the CLI output I really can't comment further. |
| 16:39.37 | dushantch | well I'm trying to make something like : exten => _#x,1,Dial(SIP/${EXTEN:1}) |
| 16:40.10 | [TK]D-Fender | dushantch: thats fine |
| 16:40.21 | ManxPower | You ARE doing a reload or dialplan reload after you make changes, right? |
| 16:40.21 | [TK]D-Fender | dushantch: So again, what are you calling with? |
| 16:40.33 | dushantch | I am doing reload |
| 16:40.44 | ManxPower | good. |
| 16:40.48 | ManxPower | What are you dialing from? |
| 16:41.57 | dushantch | I'm dialing from linksys spa3102 configured as SIP to forward anything it gets to asterisk server. It worked before :) |
| 16:42.13 | dushantch | also trying with nokia e65 as sip phone |
| 16:42.22 | [TK]D-Fender | dushantch: you'll have to make sure your SPA's dialplan allows that pattern |
| 16:42.26 | ManxPower | dushantch: the SPA has a dialplan as well that you have to update |
| 16:42.45 | [TK]D-Fender | dushantch: And ou should be looking at CLI with SIP debug enabled to see what's actually coming in |
| 16:44.43 | dushantch | [TK]D-Fender: , ManxPower : thanks I'll give it a look |
| 16:45.08 | ManxPower | dushantch: In SIP the phone collects all the digits then sends them all at once to the PBX. |
| 16:47.27 | dushantch | this is spa's dialplan: (9[2-5]<:@gw0>|xx.S0|<#:>xS0) |
| 16:47.59 | [TK]D-Fender | dushantch: that is not an FXS dialplan... |
| 16:48.19 | [TK]D-Fender | dushantch: and the FXO (line) port should not be "dialing" into * with patterns |
| 16:49.13 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:49.27 | *** join/#asterisk mohawk (n=mohawk@host217-40-110-154.in-addr.btopenworld.com) |
| 16:49.30 | dushantch | [TK]D-Fender: this is the dialplan for hansets, and for pstn is: (S0<:110>) |
| 16:51.15 | [TK]D-Fender | dushantch: (*x.T|x.T|#x.T) |
| 16:51.36 | dushantch | hmm it worked for year and a half with * 1.2 |
| 16:52.30 | dushantch | I'm a little rusty haven't messed with it for so long :), thanks for the patience |
| 16:53.18 | dushantch | [TK]D-Fender: can you enlighten me what T does in that dialplan you gave? |
| 16:54.18 | [TK]D-Fender | (T)imeout |
| 16:54.40 | [TK]D-Fender | dushantch: AKA "accept jsut about anything as long as the guy stops typing digits for 3 sec |
| 16:55.01 | neurosys | [TK]D-Fender: What would you recommend for speech recognition? LumenVox? |
| 16:55.26 | [TK]D-Fender | neurosys: ASR = bleh... but its better than Sphinx |
| 16:56.58 | neurosys | [TK]D-Fender: Oh! and its already built in to asterisk? Free is always great :) |
| 16:58.55 | neurosys | heh just a recompile. cool. |
| 16:58.56 | dushantch | you guys were right, I did this removal of # in this line in spa3102 <#:>xS0 few years ago, as AFAIK <#:> removes # . Thanks a lot |
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| 17:02.00 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 17:05.23 | ManxPower | "SIP accounts are about users (not extensions or devices). " <-- well someone has a lot to learn. |
| 17:06.47 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:07.03 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 17:09.11 | Carlos_PHX | Ah yes, I remember the day that I set up a SIP account as a user and tried to register four devices to it. |
| 17:10.13 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 17:10.15 | ManxPower | *grumble* I guess I should go do something productive. |
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| 17:12.12 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:12.14 | Carlos_PHX | Productivity is over-rated |
| 17:20.49 | *** join/#asterisk hi365_m (n=hi365@213.151.43.64) |
| 17:28.59 | *** join/#asterisk highzeth (n=highzeth@hoiseth.no) |
| 17:31.50 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 17:32.24 | beniwtv | hi all... is there a way for my dialplan to know the IP address of the IAX/SIP phone connected? |
| 17:35.17 | [TK]D-Fender | beniwtv: "core show function SIPPEER" |
| 17:36.28 | beniwtv | [TK]D-Fender: thanks |
| 17:44.20 | maqr | have any of you played with the linksys spa9000? anyone have opinions on it? |
| 17:47.24 | [TK]D-Fender | maqr: Its a toaster... one that toasts only one side, unevenly, doesn't accept rye, and lacks an appeling stainless steel finish |
| 17:50.13 | maqr | [TK]D-Fender: so... good? |
| 17:51.42 | [TK]D-Fender | is not even going to validate that... |
| 17:55.40 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 17:55.45 | Tuxguy | Carlos_PHX: What's your telephone #? |
| 17:57.16 | maqr | [TK]D-Fender: well it was a silly answer to begin with :P |
| 17:57.40 | [TK]D-Fender | maqr: I'd like to think "strangely apt" |
| 17:57.46 | maqr | heh |
| 17:58.12 | root52 | Since we are on linksys ;-) what are the thoughts on Linksys SPA3102. I really just need it to ring an exten on * after someone calls a DID hosted on a ROLM system. |
| 17:58.36 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:59.06 | root52 | it would be DID -> ROLM -> ROLM Exten -> Linksys SPA3102 -> Asterisk |
| 17:59.38 | hi365_m | any links on how to debug why asterisk forks? |
| 18:00.42 | [TK]D-Fender | hi"Because there is no spoon" |
| 18:00.44 | Qwell | Asterisk is supposed to fork. |
| 18:01.02 | [TK]D-Fender | root52: Thoughts on it? Sure its an idea. There you go. |
| 18:01.49 | root52 | [TK]D-Fender: Thank You |
| 18:04.07 | rwaite | spoon(1) |
| 18:05.49 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:05.52 | hardwire | moo.. |
| 18:06.33 | hardwire | I have rtcachefriends=yes in sip.conf and realtime load sippers name 24308 returns the DB entry |
| 18:06.45 | hardwire | but I can't see anything in the sip peers |
| 18:06.54 | hardwire | and I need to make calls through this account from local. |
| 18:07.43 | hardwire | 1.4.17 |
| 18:09.18 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 18:17.56 | hi365_m | when is asterisk supposed to fork? |
| 18:18.29 | hardwire | :P |
| 18:20.55 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 18:22.38 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-44-197.w86-215.abo.wanadoo.fr) |
| 18:24.14 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:24.42 | rhousand | will asterisk work with vodavi 6800 mgcp phones? |
| 18:28.01 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:28.07 | *** join/#asterisk synchris (n=synchris@athedsl-243679.home.otenet.gr) |
| 18:29.11 | rhousand | has anyone had luck connecting vodavi 6800 mgcp phones to an * server? |
| 18:30.21 | [TK]D-Fender | rhousand: extremely few people use MGCP with *. Your odds of getting details on one specific phon are extremely slim |
| 18:31.14 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 18:31.22 | ManxPower | hi365_m: Asterisk forks any time System or other external command is called, it's called a fork/exec |
| 18:31.36 | rhousand | has anyone had luck connecting vodavi 6800 mgcp phones to an * server? |
| 18:31.54 | ManxPower | rhousand: repeating your self all the time will do nothing but piss people off. |
| 18:32.25 | ManxPower | rhousand: Chances are you are the only person on this channel (out of 289 people) that is trying to use MGCP. |
| 18:32.45 | rhousand | sorry, network droped for a sec. I did not think my message made it to you |
| 18:32.51 | ManxPower | rhousand: Your best option is to ask on the Asterisk-Users mailing lists and/or search the mailing list archives. |
| 18:32.52 | ManxPower | ~mailinglist |
| 18:32.53 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 18:33.10 | hi365_m | ManxPower: hello. Let me restate my question then. Many a time, I notice that asterisk seems to be running twice. Some of the systome include the cli not responding to commands. *Usualy* if I do service asterisk stop it seems to stop one instance and then continues to wrok |
| 18:34.01 | ManxPower | hi365_m: You have something else going on. |
| 18:34.17 | hi365_m | ive had this alot. what could it be? |
| 18:34.23 | ZaVoid | so is iax2 just completely busted in 1.4.22? |
| 18:34.32 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:34.37 | hi365_m | it happens on many differnt systems (mostly 1.4.18) |
| 18:34.42 | ManxPower | hi365_m: No idea. I've never in my 5 years of using Asterisk ever heard of someone with the problem you are having. |
| 18:34.55 | ManxPower | ZaVoid: Cite your souce. |
| 18:34.58 | ManxPower | source, even |
| 18:35.29 | ZaVoid | well all my 1.4.15 boxes have no problems with my iax2 clients... i installed 1.4.22 for the tc400b card and i can get 1 call to connect to the box and follow my dialplan out of 100 attempts |
| 18:35.38 | ManxPower | hi365_m: it is common to see multiple asterisk "processes" (if you see more than 2 then you are seeing THREADS not processes). I have never heard of killing one of those processes making Asterisk work. |
| 18:35.42 | ZaVoid | sip clients are fine |
| 18:36.12 | hi365_m | is it also normal to have safe_asterisk running twice? |
| 18:36.47 | [TK]D-Fender | hi365_m: No |
| 18:39.42 | ZaVoid | is there any major changes to iax registration since 1.4.15 that i have to account for in iax.conf or via realtime? |
| 18:39.55 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:40.59 | cvnet | hello |
| 18:41.05 | cvnet | does asterisk supports IPSEC tunnel ? |
| 18:41.21 | hardwire | linux does |
| 18:41.21 | hi365_m | [TK]D-Fender: well, ive had it. any idea what would cause it? |
| 18:41.32 | hardwire | cvnet: use racoon and set up ipsec between some peers |
| 18:41.37 | hardwire | then test the VoIP over it |
| 18:41.41 | hardwire | and tune it to what you need |
| 18:42.32 | ZaVoid | any thoughts manx? |
| 18:42.54 | cvnet | thanks |
| 18:43.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 18:46.02 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-3e302fefc436dcfc) |
| 18:48.46 | *** part/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 18:57.33 | *** join/#asterisk seaq (n=seaq@200.62.48.246) |
| 19:01.27 | [TK]D-Fender | hi365_m: * doesn't care how the packets get there. |
| 19:01.43 | hi365_m | packets? |
| 19:02.42 | [TK]D-Fender | cvnet: * doesn't care how the packets get there. |
| 19:03.00 | hi365_m | :) |
| 19:03.12 | [TK]D-Fender | hi365_m: wrong targt. For yours, watch out for someone else running safe_asterisk by hand, or perm issues wher it doesn't see the PID and double-runs |
| 19:03.30 | hi365_m | weird, but ok |
| 19:04.42 | jameswf | sweet: http://www.dealextreme.com/details.dx/sku.4355 |
| 19:04.54 | jameswf | kill all cell phones.. |
| 19:05.00 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-87-202.w86-215.abo.wanadoo.fr) |
| 19:07.01 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 19:07.26 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 19:09.50 | *** join/#asterisk Tuxofred (n=Fred@ip-80-236-227-103.dsl.scarlet.be) |
| 19:10.00 | Tuxofred | hello |
| 19:10.18 | Tuxofred | i'm setting up a asterisk server.. |
| 19:10.46 | Tuxofred | if i call to a friend, i can hear him but he can't hear me |
| 19:10.53 | Tuxofred | what's the matter |
| 19:11.00 | Deeewayne | ~nat |
| 19:11.01 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:11.32 | Deeewayne | ~sipnat |
| 19:11.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:12.27 | awk_r | Tuxofred, fyi, Deeewayne's jbot triggers are directed toward you. (hint: Investigate your NAT issues) |
| 19:13.08 | Tuxofred | awk_r> i create one account by friend.. i added nat=yes |
| 19:13.37 | Deeewayne | thanks awk_r :-) |
| 19:14.44 | awk_r | Deeewayne, heh np. |
| 19:14.53 | Maliuta | Tuxguy: read the docs you've been pointed at |
| 19:15.18 | awk_r | Tuxofred, ^^ |
| 19:15.26 | Maliuta | s/Tuxguy/Tuxofred/ |
| 19:15.37 | awk_r | pets jbot. |
| 19:15.38 | Maliuta | damn fingers |
| 19:15.56 | Maliuta | when do I get my Andromeda style neural interface? |
| 19:16.24 | [TK]D-Fender | Maliuta: Right this way! |
| 19:16.28 | Tuxguy | ?? |
| 19:16.31 | Tuxguy | oh |
| 19:16.40 | [TK]D-Fender | grabs his B&D power drill |
| 19:16.44 | awk_r | Maliuta, when was the last time you upgraded? That feature is so 2050s! |
| 19:16.50 | awk_r | get with the times. |
| 19:16.57 | Maliuta | questions [TK]D-Fenders surgical abilities |
| 19:17.09 | awk_r | Maliuta, no time for questions! |
| 19:17.28 | Maliuta | awk_r: actually something like CY6000's ;P |
| 19:17.32 | hi365_m | is there a problem viewing the svn? |
| 19:17.34 | [TK]D-Fender | HEEEEEEEERRRRRRRE'S JOHNNY! |
| 19:17.43 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 19:17.46 | *** join/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 19:17.52 | hi365_m | http://svn.digium.com/view/asterisk?view=rev&revision=108530 returns, The requested URL /view/asterisk was not found on this server |
| 19:17.54 | *** part/#asterisk feeds (n=feeds@85-135-232-31.adsl.slovanet.sk) |
| 19:18.01 | Maliuta | SBS is having Kuberick week ATM :) |
| 19:18.23 | awk_r | hi365_m, the svn server is down for maintenance and should be up roughly around 5pm Central |
| 19:18.34 | hi365_m | svn.digium.com gets redirected to the bug tracker?! |
| 19:18.40 | hi365_m | oh - cool |
| 19:18.54 | awk_r | Greetings, |
| 19:18.54 | awk_r | We recently moved our public subversion mirror to a new server. It is |
| 19:18.54 | awk_r | currently down for maintenance while we resolve some unforeseen |
| 19:18.54 | awk_r | problems. It should be back up by the end of the day. |
| 19:18.54 | awk_r | I apologize for the inconvenience, |
| 19:18.55 | hi365_m | wonders why people do such things without consulting with him first |
| 19:18.56 | awk_r | -- |
| 19:18.58 | awk_r | Russell Bryant |
| 19:19.02 | awk_r | (copied from asterisk-users mailing list) |
| 19:19.04 | Maliuta | ~pastebin |
| 19:19.04 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:19.19 | Maliuta | don't make me slap you! |
| 19:19.41 | hi365_m | err, what time is it now in central? |
| 19:19.46 | awk_r | 1:19 PM |
| 19:19.48 | [TK]D-Fender | ~cluebat awk_r |
| 19:19.49 | jbot | ACTION pulls out a ClueBat (tm) and thwaps awk_r. |
| 19:19.58 | [TK]D-Fender | awk_r: Don't spam like that |
| 19:20.19 | awk_r | sighs. |
| 19:20.20 | awk_r | kk |
| 19:20.35 | hi365_m | thats a while... are there any mirrors? |
| 19:20.56 | esaym | how do I get my local time in /var/log/asterisk/cdr-csv/Master.csv instead of the default UTC? |
| 19:21.12 | hi365_m | [TK]D-Fender: under what license is cluebat distributed? |
| 19:21.12 | awk_r | hi365_m, I think the tarballs are still available |
| 19:21.33 | hi365_m | neh, I wanted to see some revisions. i guess ill just have to wait |
| 19:21.51 | [TK]D-Fender | hi365_m: I'm licensed... the thwap! |
| 19:22.21 | hi365_m | checks to see if [TK]D-Fender license has been faxed in to the legal department |
| 19:22.55 | Maliuta | legal department here. [TK]D-Fender is cleared for thwapage |
| 19:24.01 | Maliuta | all queries about my validity as legal dept and issuer of thwapage licences are to be directed to rather large man in the corner with the torture equipment |
| 19:24.37 | hi365_m | runs for his life, clutching his paintbrush and a bucket of paint from mspaint |
| 19:25.27 | hi365_m | btwm who is TK and why does he need to be d-fended? |
| 19:25.40 | hi365_m | s/btwm/btw, |
| 19:25.51 | [TK]D-Fender | Maliuta: Legal Notice : "Torture" is henceforth to be termed as "aggressive interrogation methodologies" |
| 19:26.36 | Maliuta | [TK]D-Fender: thanks for the reminder, I have that memo from PR around here somewhere |
| 19:27.06 | hi365_m | and i though you were going to redefine it "a gentel method teaching duma$$'s to RTFM" |
| 19:27.09 | Maliuta | hi365_m: if you have to ask you don't need to know |
| 19:27.30 | hi365_m | wonders if he should get his won defener as well |
| 19:29.17 | jaytee | [TK]D-Fender: Legal Notice : "Torture" is henceforth to be termed as "just us Republicans show ya da love!" |
| 19:29.17 | hi365_m | hec, with [TK]D-Fender walking around swinging his cluebat around like Griff Tannen, I think we all need a defender |
| 19:30.22 | Maliuta | I think the only people in danger are the clueless who think they are too good to RTFM |
| 19:30.31 | Maliuta | and they're in danger from more than just [TK]D-Fender |
| 19:30.32 | lmadsen | anyone know how to set the privacy=full in the RPID? |
| 19:32.22 | lmadsen | *crickets* |
| 19:32.34 | jaytee | what's a RPID? |
| 19:32.42 | lmadsen | remote-party ID |
| 19:32.53 | lmadsen | is the callerID portion of a SIP message (basically) |
| 19:33.12 | hi365_m | was just reading up on that |
| 19:34.09 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 19:34.10 | jaytee | ah, thanks. I just ordered SIP Demystified as Jared recommended it. I haven't spent alot of time dealing with the headers except with Exchange and haven't had to do alot of it so I'd never seen the term. |
| 19:34.28 | Maliuta | jaytee: you _still_ name dropping? ;P |
| 19:35.06 | *** join/#asterisk km2 (n=x@mobile-166-217-143-182.mycingular.net) |
| 19:35.33 | [TK]D-Fender | Maliuta: Only because LEIF MADSEN said it was OK ;) |
| 19:36.08 | [TK]D-Fender | na na na na na! |
| 19:36.16 | Qwell | jaytee: Just say you know me, and it'll all be okay |
| 19:36.30 | Qwell | You should see the attention you get when you drop my name at a bar or a club. |
| 19:36.36 | Qwell | blank. stares. |
| 19:36.44 | jaytee | I know Qwell!!!! |
| 19:37.36 | jaytee | now I need to mail my book with return postage to Leif so he can sign it too. :-) |
| 19:39.12 | lmadsen | to answer my own question |
| 19:39.20 | lmadsen | exten => s,n,SetCallerPres(prohib_not_screened) |
| 19:39.28 | lmadsen | that's how you change it |
| 19:39.39 | jaytee | I've spent the morning rewriting portions of my IVR to make it more streamlined and I implemented some changes that file suggested the other day and it's running smooth and the voice rec is usually dead on unless you really mumble a command. |
| 19:40.36 | Maliuta | jaytee: gonna cost you more to send it to me to get it signed ;P |
| 19:41.10 | jaytee | why would I care if you signed it? you're not Jim are you? |
| 19:41.56 | Qwell | jaytee: my name is in there too - I'll sign it :p |
| 19:42.05 | Qwell | my signature will probably devalue it though |
| 19:43.11 | codefreeze-lap | Qwell: I've got your sig in my book! Almost everybody's but lmadsen... sigh |
| 19:43.13 | jaytee | Qwell, I should have had you and russell sign it before I left. I didn't think of having Jared sign it till just at the end of the dCAP |
| 19:43.55 | *** join/#asterisk porter (i=terdon@unaffiliated/porter/x-000001) |
| 19:44.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:44.55 | jaytee | Qwell, and it wouldn't devalue the book having your signature in it. |
| 19:45.08 | Maliuta | jaytee: you'd care because it's me :) |
| 19:45.54 | jaytee | Maliuta, with a first name like Nikolai I'm already suspicious of you and I've hidden the silverware, the cash and the chickens. |
| 19:47.29 | Maliuta | jaytee: are you a Nazi? because I think you have something against Slavic people |
| 19:47.31 | Maliuta | :P |
| 19:48.59 | jaytee | no Slavic people per se. I actually find cute women with slavic features and accents to be really hot. I'll just never trust them damn commie russkies. Blame it on a having spent too much of my life living through the cold war. |
| 19:49.58 | Maliuta | Ich bin nicht untermensch |
| 19:50.26 | Maliuta | Well I am a) Ukrainian. b) a socialist. |
| 19:50.29 | jaytee | you are not subhuman? |
| 19:50.46 | Maliuta | and I take offense to the term "commie" |
| 19:51.00 | Maliuta | the bolsheviks did not practice socialism |
| 19:51.04 | lmadsen | steals the silverware, hash, and chickens, then points at Maliuta |
| 19:51.15 | dushantch | Maliuta: you're not alone |
| 19:51.42 | Maliuta | jaytee: the ss refered to Slavs as untermensch |
| 19:51.53 | jaytee | Maliutu, I'm sorry you find the term "commie" offensive. From now on I shall refer to such followers of Marx as "broke dick people who stand in line just to get toilet paper". Does that help? |
| 19:51.55 | *** join/#asterisk Howie69 (n=Owner@host-12-173-142-207.nctv.com) |
| 19:52.12 | Maliuta | jaytee: there is nothing wrong with Marx |
| 19:52.42 | Maliuta | jaytee: the problem is with Bolshevism, things were fine from Feb 5 1917 -> Oct |
| 19:53.25 | jaytee | all economic systems have their failures. communism is flawed. it robs the individual of ownership and kills the motor of the world which is the human ego. read Ayn Rand and you might get a clue. Marx was pathetic whining loser. |
| 19:53.29 | Maliuta | Lennin was a power mad asshole. He had to convince his own people to get violent |
| 19:54.03 | dushantch | socialism? |
| 19:54.16 | lmadsen | looks around and could have sworn he was in #asterisk |
| 19:54.21 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 19:54.22 | mikealeonetti | he convinced eople to get a violin? |
| 19:54.22 | jaytee | hahahaha |
| 19:54.31 | mikealeonetti | Lenin was a musician? |
| 19:54.36 | *** join/#asterisk alexmontoanelli (n=alexmont@200.193.10.187) |
| 19:54.38 | mikealeonetti | didn't he marry Ouno? |
| 19:54.43 | jaytee | rofl |
| 19:54.53 | jaytee | we need drmessano in here NOW! |
| 19:54.53 | alexmontoanelli | Hello all, |
| 19:55.16 | alexmontoanelli | the svn. on http://svn.digium.com/svn/ is down, |
| 19:55.19 | Maliuta | dushantch: I get the impression that most um-ericans are still living with the MacCartist view on anything left of the far right |
| 19:55.25 | alexmontoanelli | anybody could confirm to me? |
| 19:55.58 | jaytee | Maliuta, in my case you couldn't be more wrong. |
| 19:56.00 | [TK]D-Fender | alexmontoanelli: yes |
| 19:56.00 | awk_r | alexmontoanelli, it is currently down for maintenance |
| 19:56.12 | awk_r | alexmontoanelli, http://lists.digium.com/pipermail/asterisk-dev/2008-November/035393.html |
| 19:56.34 | awk_r | [TK]D-Fender, who needs pastebin when you have mailing lists :-) |
| 19:56.44 | alexmontoanelli | awk_r:, thanks. |
| 19:56.49 | awk_r | alexmontoanelli, np |
| 19:57.12 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- SVN IS DOWN FOR MAINTENANCE -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 19:57.58 | jaytee | but I am a left leaning liberal democrat who believes in a moderated capitalist economy, not a completely unregulated capitalist economy and I would still prefer either to a socialist or communist economy because the latter are typcially more stagnant |
| 19:58.31 | lmadsen | 302 Redirect #politics |
| 19:59.48 | Maliuta | the US is so far to the right of the rest of the world what you think is "left leaning" is actually leaning toward the centre right |
| 19:59.53 | dushantch | jaytee: true |
| 20:00.01 | *** join/#asterisk ice_croft (n=nolan@85.172.54.214) |
| 20:00.19 | Howie69 | so, I think I have my SIP registration problem narrowed down to Via-Rhine III module |
| 20:00.21 | *** join/#asterisk wfaulk (i=wfaulk@beaglebros.com) |
| 20:00.24 | Howie69 | via-rhine II chipsets work fine |
| 20:00.57 | Maliuta | wonders how a hardware module is causing a software issue |
| 20:05.15 | wfaulk | I purchased an X100P from x100p.com and I can't get it to work. After loading the zaptel module, it just spews out "FXO PCI Master Abort" over and over and otherwise seems to lock the machine |
| 20:05.57 | wfaulk | I've assigned distinct IRQs to all of the PCI devices in the BIOS and disabled everything I didn't need |
| 20:06.12 | lmadsen | the X100P was EOL'd like... 2 years ago I think |
| 20:06.18 | lmadsen | there was a reason for that |
| 20:06.28 | wfaulk | heh |
| 20:06.33 | wfaulk | um, okay |
| 20:06.35 | [TK]D-Fender | wfaulk: pastebin "cat /proc/interrupts" and tell us what SW versions you're running |
| 20:06.41 | [TK]D-Fender | ~pb |
| 20:06.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 20:06.44 | [TK]D-Fender | ^^^^^^ |
| 20:07.04 | wfaulk | do you have a suggestion as to a decent reasonably low-cost alternative? |
| 20:07.34 | wfaulk | right now, I'm trying to run a livecd and the module gets loaded on boot (init script, maybe?) and I can't even get to a login |
| 20:08.02 | wfaulk | I can probably boot off of a ubuntu livecd. would /proc/interrupts from there be relevant? |
| 20:08.26 | mikealeonetti | I am the conversation stopper |
| 20:08.32 | mikealeonetti | and the beat goes |
| 20:08.44 | mikealeonetti | (what?) |
| 20:10.42 | *** join/#asterisk ar_howard (n=allan@ppp118-208-90-91.lns2.bne4.internode.on.net) |
| 20:11.48 | [TK]D-Fender | wfaulk: Please provide what I hav requested |
| 20:12.50 | wfaulk | I can get it from a different OS install. is that okay? |
| 20:13.26 | [TK]D-Fender | wfaulk: When you actually have an installed system to debug, let us know |
| 20:13.38 | [TK]D-Fender | wfaulk: Debuggin a fixed CD image is a waste of time |
| 20:14.06 | rwaite | a waste of time... or a useful time? |
| 20:15.06 | jaytee | Maliuta, sorry I had to run to another building for a minute. Yeah, I'd agree that my "left leaning liberalism" would actually put me in the real middle in most countries :-) |
| 20:15.44 | Maliuta | jaytee: I'm on the real left :) |
| 20:15.48 | jaytee | but that's only because most Americans have a room temperature I.Q. and either never studied history or forgot everything they learned about it. |
| 20:16.35 | Maliuta | that and a US centric focus |
| 20:17.06 | [TK]D-Fender | jaytee: We believe that even more in Canada ;) |
| 20:17.07 | [TK]D-Fender | ZING!!!! |
| 20:17.14 | jaytee | Maliuta, that's going to change next year. Obama will save us all! |
| 20:17.45 | [TK]D-Fender | executes another UOM drive-by RAT-TAT-TAT-TAT-TAT-TAT!!!! |
| 20:18.11 | Howie69 | ouch |
| 20:18.12 | Howie69 | not the driver |
| 20:18.17 | Howie69 | the mtu is coming up as 576 |
| 20:18.18 | jaytee | "And the moon is in the seventh house. peace will shine on the planets and the world will be as one. This is the dawning of the age of Obamaness, age of Obamaness.........." damn, I shouldn't have had seconds on the Kool-Aid. |
| 20:18.22 | Howie69 | whenever the link comes up |
| 20:18.26 | Maliuta | jaytee: I had fun with my Eastern European history lecturer at uni. He was a yank, educated at Stanford in the '50s and then served in the US Army as a sovietologist |
| 20:18.31 | Howie69 | changing the mtu back to 1500 lets sip registrations work fine |
| 20:18.44 | Howie69 | now why is my card always coming up at 576? |
| 20:18.51 | jaytee | screwing with the MTU usually mucks things up big time |
| 20:18.55 | Maliuta | jaytee: he had a distinctly Russian view on things, so we had some bumps |
| 20:19.18 | *** join/#asterisk Segnale007 (n=Pietro@host219-248-dynamic.18-79-r.retail.telecomitalia.it) |
| 20:19.28 | Maliuta | 576 is awfully small for an MTU |
| 20:19.45 | jaytee | ok, gotta get back to testing and debuggin this IVR and some changes to the grammar files. back in a few |
| 20:19.56 | *** join/#asterisk ManxPower (n=manxpowe@73.sub-70-222-191.myvzw.com) |
| 20:19.59 | Maliuta | that'd result in alot of packets for even small data transmission like SIP initiation |
| 20:20.07 | Howie69 | so, is MTU assigned by the DHCP server? |
| 20:20.20 | Maliuta | no |
| 20:20.35 | mikealeonetti | it's very strange when women have an odd number of breasts |
| 20:20.39 | Maliuta | MTU has to be set before you can even think about getting DHCP info |
| 20:21.00 | Howie69 | Is the module setting the mtu? |
| 20:21.08 | Maliuta | MTU is set when the interface is brought up, then DHCP happens |
| 20:21.19 | [TK]D-Fender | mikealeonetti: Go watch "The Fifth Element" or "Kung-Pao: Enter The Fist" |
| 20:21.23 | Maliuta | module is 2 steps before the interface coming up |
| 20:21.30 | Howie69 | so |
| 20:21.36 | Howie69 | My interface is dhcp |
| 20:21.39 | Maliuta | Howie69: I think you need to go to school on networking |
| 20:21.46 | Howie69 | Been to school on networking |
| 20:21.51 | Howie69 | just never had a mtu issue before |
| 20:21.52 | Maliuta | obviously not |
| 20:22.04 | Howie69 | when I bring my link up , it defaults to a mtu of 576 |
| 20:22.06 | Maliuta | if you don't know where an MTU is being set you don't know enough |
| 20:22.25 | Howie69 | even when I add mtu 1492 |
| 20:22.44 | ManxPower | Howie69: That is very, very, very weird. 576 is the default for some ppp interfaces, but it's pretty uncommon |
| 20:22.52 | ManxPower | Howie69: Linux, I assume? |
| 20:22.54 | Howie69 | ManxPower: yes |
| 20:23.03 | mikealeonetti | [TK]D-Fender: not that it's a bad thing... |
| 20:23.07 | Howie69 | I can change it just fine with ifconfig |
| 20:23.39 | ManxPower | Howie69: what distro? |
| 20:23.45 | Howie69 | debian |
| 20:23.59 | Howie69 | added mtu=1496 in /etc/network/interfaces |
| 20:24.04 | Howie69 | but doesn't seem to work with dhcp |
| 20:24.30 | Howie69 | iface eth2 inet dhcp |
| 20:24.30 | Howie69 | <PROTECTED> |
| 20:24.35 | Howie69 | well, 1500 |
| 20:24.46 | Howie69 | when the link comes up, it is still 576 |
| 20:25.18 | ManxPower | Howie69: you realize that mtu is not the same as MTU, right? |
| 20:25.53 | Howie69 | huh? |
| 20:25.54 | hardwire | can't seem to qualify cached realtime sip peers |
| 20:25.55 | hardwire | :( |
| 20:26.17 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 20:26.53 | puppet | hardwire: "2008-08-27 - In v1.4 (SVN only) "qualify=yes" is ignored if the peer is realtime and caching is not turned on. See http://bugs.digium.com/view.php?id=13383. " |
| 20:27.04 | ManxPower | Howie69: mtu=1500 may not be the same variable as MTU=1500. |
| 20:27.05 | hardwire | caching is turned on |
| 20:27.13 | ManxPower | What is the caps of the other stuff in /etc/sysconfig/network? |
| 20:27.20 | Howie69 | ManxPower: it is |
| 20:27.30 | Howie69 | ManxPower: http://www.ubuntugeek.com/how-to-change-mtu-maximum-transmission-unit-of-network-interface-in-ubuntu-linux.html |
| 20:27.53 | Howie69 | nothing is capped in /etc/sysconfig/network |
| 20:28.09 | ManxPower | Howie69: then I cannot help you as your distro works differently than mine. |
| 20:28.23 | Howie69 | what is your distro? |
| 20:28.27 | ManxPower | that page also does not show = in the settings |
| 20:28.39 | ManxPower | Howie69: CentOS |
| 20:29.43 | ManxPower | Howie69: Do you have a /etc/sysconfig/network-scripts/ifup? |
| 20:29.53 | Howie69 | yes |
| 20:30.02 | ManxPower | in mine I have: |
| 20:30.03 | ManxPower | if [ -n "${MTU}" ]; then |
| 20:30.03 | ManxPower | <PROTECTED> |
| 20:30.03 | ManxPower | fi |
| 20:30.11 | ManxPower | seems pretty simple to me |
| 20:30.35 | Howie69 | I was just headed in that direction to look at the script |
| 20:31.46 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 20:31.48 | [TK]D-Fender | BBAIB |
| 20:32.08 | Peaceful | I just upgraded from asterisk 1.2.x to 1.4.22 and am trying to fix this error: |
| 20:32.10 | Peaceful | [Nov 20 13:30:53] WARNING[9934]: chan_dahdi.c:1095 dahdi_digit_begin: Couldn't dial digit 9: No data available |
| 20:32.23 | ManxPower | Peaceful: did you read all the UPGRADE.txt files? |
| 20:32.28 | Peaceful | ^-- which happens when people call out to the POTS and try to press digits |
| 20:32.33 | Peaceful | Manx, yes, lots of them |
| 20:32.39 | Peaceful | which is why I didn't die |
| 20:32.45 | Peaceful | but I must have missed this issue |
| 20:33.08 | ManxPower | Peaceful: seems like almost nobody that comes in here with 1.2 -> 1.4 or 1.4 -> 1.6 issues ever read the upgrade.txt fles. |
| 20:33.25 | Peaceful | Well I spent several hours reading them, but I'm unusual |
| 20:33.48 | Peaceful | I was able to catch quite a few issues by reading those, maybe I should grep it for dtmf and see if I missed this one... |
| 20:35.10 | rwaite | hay guys how do i build 1.6??? |
| 20:35.18 | rwaite | laughs |
| 20:35.19 | Peaceful | Neither UPGRADE-1.2.txt nor UPGRADE.txt deal with anything related to my problem that contains "dtmf" |
| 20:37.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:38.04 | Peaceful | so anyway, with a dtmfmode of "rfc2833" or "inband" my agents can login, but outbound calls can't interact with IVRs. With dtmfmode "info" outbound calls can interact with IVRs, but my agents can't log in. What am I missing? |
| 20:38.49 | trogs | i recall that 1.2 had weird dtmf handling. try setting mode to auto |
| 20:39.32 | Peaceful | trogs: trying that... |
| 20:40.21 | Peaceful | trogs: auto works with vmail/agent login, not with IVR across the PSTN |
| 20:40.34 | Peaceful | weeird |
| 20:40.50 | Peaceful | maybe I should be rebooting my phone when I change the sip config too? |
| 20:41.14 | ManxPower | Peaceful: there is a toneduration= option (chan_dahdi.conf?) that specifies the length of outgoing generated tones, I set mine to between 300 and 400 |
| 20:41.14 | trogs | you could give it a go |
| 20:41.39 | ManxPower | set it to rfc2833 and set the toneduration |
| 20:42.20 | *** join/#asterisk fabbari (n=mrwho@80.79.152.149) |
| 20:43.23 | fabbari | Hi all! Can I simply ask a question in the channel or there is a more polite way of handling this? |
| 20:44.10 | ManxPower | ~ask |
| 20:44.11 | jbot | hmm... ask is 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. |
| 20:44.27 | Peaceful | ManxPower: in zapata.conf I put toneduration=350 in [channels], set dtmfmode=rfc2833 in sip.conf -- no effect. vmail works, IVR on PSTN doesn't still |
| 20:44.48 | Peaceful | (I haven't upgraded to the dahdi-named stuff yet--I planned this migration before that came out) |
| 20:45.28 | Peaceful | (oh, I did a "reload" before trying it, obviously, though I didn't restart my phone--a polycom 550) |
| 20:46.42 | ManxPower | Peaceful: did you restart asterisk? |
| 20:47.05 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 20:47.22 | ManxPower | Peaceful: if you are using dahdi I don't think it will read zapata.conf will it? |
| 20:47.32 | Un1x | hey was wondering can you still transfer calls even if your using TDM technologie? because yestorday i found out that you cannot signal hold with TDM technologie |
| 20:48.04 | ManxPower | Un1x: do read the zapata.conf.sample and you will become enlighetened |
| 20:48.08 | *** join/#asterisk oomph (n=oomph@wsip-70-164-41-74.dc.dc.cox.net) |
| 20:48.27 | oomph | anyone how to make the # button not forward to parked calls? |
| 20:48.27 | ManxPower | I bet the Asterisk book also talks about it and the voip-info wiki too! |
| 20:48.34 | Un1x | well ManxPower i was told yestorday with analogue phones you cannot signal hold so, im wondering can you still transfer calls? |
| 20:48.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 20:48.56 | ManxPower | Un1x: EXACTLY like 3-way calling with the telephone company. |
| 20:49.06 | fabbari | I'm trying to configure sip.poivy.com as my outbout sip peer. I'm using SIP to connect to the asterisk box. I can connect and talk between two extensions that are local to the asterisk box - two different SIP clients I mean. After I configured the SIP outbound peer I can call, the phone rings - so the outgoing part of SIP is working - when I answer the SIP client keeps getting the ringing tone, and there is no sound on the called phone. When I hang up the |
| 20:49.10 | Un1x | ManxPower, yes indeed it does im not asking you to spoonfeed me simply yes or no so i can try and read about it |
| 20:49.16 | Un1x | ManxPower thanks |
| 20:49.32 | [TK]D-Fender | \o/ for local profile copy! |
| 20:52.59 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.186) |
| 20:53.19 | Daejeo | Meow Meow :) |
| 20:53.39 | Daejeo | Katty |
| 20:54.27 | *** join/#asterisk magic_hat (n=geoffdou@h-64-105-84-216.chcgilgm.dynamic.covad.net) |
| 20:55.01 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 20:55.33 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:56.14 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 21:02.51 | magic_hat | hey all. i installed asterisk via ubuntu package. I know that's not the preferred way to go, but my system's working great and I don't want to muck with it too much. I'm trying to install appconference, and need to point the Makefile to my asterisk headers. any idea where one might find them? |
| 21:03.00 | *** join/#asterisk szallol (n=szallol@86.105.195.113) |
| 21:03.02 | *** join/#asterisk Trionnis (i=Trionnis@s233-51-251.nap.wideopenwest.com) |
| 21:03.35 | Maliuta | magic_hat: ubuntu should have a headers or src package |
| 21:03.38 | ManxPower | magic_hat: you get them from the asterisk source |
| 21:03.50 | Maliuta | or what ManxPower said |
| 21:04.07 | jpeeler | you probably want the asterisk-dev package |
| 21:04.18 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 21:04.26 | Howie69 | ManxPower: I'm not sure what DHCP option the ISP is sending, but I set my mtu to 1500. Then do dhclient3 -r eth2 to release the ip, then dhclient3 eth2 to renew it, mtu goes back to 576 |
| 21:04.28 | Trionnis | anyone here that can tell me what would happen to existing calls between 2 asterisk servers if I remove trunk=yes from iax.conf and reload iax ? is it like the other changes that would only take effect from that point on, or would it break the existing trunk? |
| 21:05.27 | *** join/#asterisk jplank (n=GBove@ool-18bb018e.dyn.optonline.net) |
| 21:05.59 | magic_hat | okay, i'll mess around with that. |
| 21:06.02 | magic_hat | thanx |
| 21:11.37 | tzafrir_laptop | magic_hat, what version of asterisk do you have? |
| 21:12.28 | tzafrir_laptop | some earlier versions of the asterisk deb (before 1.4.18, IIRC) incorrectly placed /usr/include/asterisk.h in /usr/include/asterisk/ |
| 21:12.44 | tzafrir_laptop | you can work around that with a symlink |
| 21:13.33 | jaytee | Katty, what happened to sleekgeek.org? |
| 21:15.40 | Trionnis | anyone? :) |
| 21:16.02 | ManxPower | Trionnis: I've never seen a reload terminate calls. |
| 21:17.00 | ManxPower | Just issue the reload. If calls drop and users complain look puzzled and concerned and say "I'll look into it right away" |
| 21:17.12 | Trionnis | that's why I'm a little worried about it... I've never seen it either, but since I'm changing the underpinnings of how it works, it might cause issues |
| 21:17.14 | Trionnis | lol |
| 21:17.24 | Trionnis | I'll tell our VP of sales to come talk to you then! ;) |
| 21:17.41 | jplank | I have a customer with a bunch of ip650's, they say when they go on speakerphone the have to either get real close to the phone or yell to have the other side hear them, does anyone know of a way to increase the volume of the mic or something like that? |
| 21:17.54 | ManxPower | Trionnis: If it's SO important to do it during business hours.... |
| 21:18.03 | Trionnis | kinda the point I made |
| 21:18.13 | Trionnis | unfortunately she outranks me :) |
| 21:18.14 | ManxPower | jplank: sip.cfg and phone1.cfg |
| 21:18.23 | jplank | I can do it from there? |
| 21:18.31 | *** part/#asterisk root52 (n=f745082a@ip70-191-120-39.cl.ri.cox.net) |
| 21:18.34 | jplank | I have to reread the config files, I must of missed it |
| 21:18.45 | jplank | so you know which section it is? |
| 21:18.50 | jplank | do you* |
| 21:18.54 | ManxPower | jplank: less than 10 days ago someone posted a message on the mailing lists with all the gain settings |
| 21:19.13 | jplank | is their gain settings just for speakerphone? |
| 21:19.14 | *** join/#asterisk lord_nikon (n=lord@host-216-153-131-74.roc.choiceone.net) |
| 21:19.25 | ManxPower | Trionnis: You have several choices. Wait until after hours, do it now and hope for the best, or set up 2 servers, test it then decide it |
| 21:19.41 | ManxPower | jplank: you can set the gains for almost EVERYTHING in the polycoms |
| 21:19.46 | ManxPower | grep gain sip.cfg |
| 21:19.52 | ManxPower | it's not rocket science |
| 21:20.02 | jplank | thats awesome, I don't know how I missed gain settings for the speakerphone |
| 21:20.09 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:20.53 | ManxPower | jplank: they probably don't call it "speakerphone". What I suggest is set the volume.persist option that way and manual volume adjustments should be remembered |
| 21:21.03 | ManxPower | handsfree might be a term they use |
| 21:21.29 | jplank | ahhh I see it now |
| 21:21.44 | jplank | wait no |
| 21:21.45 | jplank | err |
| 21:22.06 | jplank | handset, headset, chassis, ringer |
| 21:22.16 | jplank | chassis? |
| 21:22.23 | ManxPower | that would be my guess. |
| 21:22.31 | ManxPower | The admin guide should tell you |
| 21:22.39 | jplank | but is that going to increase the gain for the speaker or the mic? |
| 21:22.40 | ManxPower | You have the Admin guide? |
| 21:22.49 | jplank | I'm looking in it, really not much mention about it |
| 21:22.50 | jplank | yea |
| 21:22.51 | magic_hat | tzafir: i'm on 1.4.17. i just installed asterisk-dev, which put asterisk.h in /usr/include/asterisk, as you mentioned. |
| 21:23.03 | magic_hat | where does it need to be symlinked to? |
| 21:23.04 | jplank | I'm reading 3.1.1's guide |
| 21:23.10 | *** part/#asterisk Howie69 (n=Owner@host-12-173-142-207.nctv.com) |
| 21:24.06 | ManxPower | jplank: set all the persist options and see if users can just adjust their own gains |
| 21:24.38 | jplank | thats the thing, can they adjust the gain for the mic? |
| 21:24.46 | jplank | its not that they can't hear fine, its the other side |
| 21:24.51 | ManxPower | Ah! |
| 21:25.02 | ManxPower | no they can't do that for the mic |
| 21:25.09 | jplank | I would think its a QOS issue, but they said it works perfectly when on the handset |
| 21:25.18 | Katty | jaytee: dydns is having issues |
| 21:26.07 | [TK]D-Fender | checkout time, BBIAB |
| 21:26.49 | ManxPower | voice.gain.tx |
| 21:27.15 | jplank | hmmm, it can't hurt to try it |
| 21:27.30 | jplank | I never heard anyone complain about sound being too loud |
| 21:27.41 | jplank | well, a lot less then to low at least |
| 21:27.45 | ManxPower | jplank: I have some sample files on http://www.fnords.org/~eric/polycom-config-examples |
| 21:28.14 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:28.14 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:28.53 | jplank | thanks, I'm going to try it |
| 21:29.09 | jplank | I guess I'll do it to chassis first, if that doesn't work, I'll do everything |
| 21:29.32 | jplank | looks like the default is 12 |
| 21:29.45 | jplank | do you know how high that field can do? |
| 21:29.48 | jplank | 0-99? |
| 21:29.52 | jplank | that sounds too high |
| 21:29.56 | *** join/#asterisk hummb (i=anon@theos.org) |
| 21:30.17 | magic_hat | hey, folks. I've got appconference installed with no errors. have stopped and restarted *. but getting this when I try to dial the conference: No application 'Conference' for extension (outbound, 99, 3) |
| 21:30.46 | hummb | are they are desktop apps yet that can perform presence/call transfer my connecting to asterisk manager ? |
| 21:31.11 | jplank | gastman |
| 21:31.12 | jplank | old |
| 21:31.14 | jplank | but works |
| 21:31.18 | jplank | ugly though |
| 21:31.39 | hummb | :( |
| 21:31.45 | hummb | it does look old |
| 21:31.53 | jaytee | FOP |
| 21:31.57 | hummb | screw fop |
| 21:32.04 | hummb | i mean like a HUDlite :) |
| 21:32.09 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:32.13 | hummb | without needing the hud server junk |
| 21:32.17 | jaytee | hummb, that's what I'd say because I'm a Dapper Dan man myself |
| 21:32.34 | jaytee | hummb, can't use Hudlite without the backend stuff |
| 21:32.38 | hummb | i know |
| 21:32.49 | hummb | and id like to use something with the digium aa50 appliances |
| 21:33.21 | *** part/#asterisk fabbari (n=mrwho@80.79.152.149) |
| 21:34.36 | lord_nikon | ive got a question about IAX2, i am trying to forward a call comming in to server1 to a different server, is that possible ? |
| 21:35.04 | jaytee | lord_nikon, no because Crash Override and Acid Burn will stop you |
| 21:35.17 | lord_nikon | :) |
| 21:35.28 | jaytee | lord_nikon, actually yes you can |
| 21:35.45 | lord_nikon | alrighty, in which case |
| 21:35.55 | lord_nikon | im trying it like so: switch => iax2/bills/devicein |
| 21:35.59 | jaytee | but it depends how you set it up in your dialplan |
| 21:36.14 | lord_nikon | however, when i try that, i get -- Executing Dial('IAX2/bills/h@devicein') |
| 21:36.19 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 21:36.20 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:36.25 | lord_nikon | it keeps sending it to extension h |
| 21:37.41 | lord_nikon | if i use Dial instead of trying to switch it works fine, but i want the call to be redirected to the other server, not going through the first server to the second |
| 21:40.56 | magic_hat | so i'm installing appconference, and I'm a bit confused on what's supposed to be happening here. should I be seeing new .so files in /usr/lib/asterisk/modules? |
| 21:41.16 | lmadsen | magic_hat: probably not. you might have to copy them there manually |
| 21:41.24 | lmadsen | you should see them in /usr/src/asterisk/apps/ though |
| 21:41.27 | lmadsen | if they built |
| 21:41.34 | *** join/#asterisk WHYS (i=lpfm@137.28.94.209) |
| 21:41.48 | magic_hat | never mind, too much coffee, too little sleep. it's fine now. |
| 21:42.36 | WHYS | any strong votes for best free softphone. I am using x-lite, but loooking to browse others. |
| 21:42.46 | jaytee | back later |
| 21:43.17 | kaldemar | you won't see app_conference in /usr/src/asterisk/apps/. make install installs app_conference.so to /usr/lib/asterisk/modules/ |
| 21:48.42 | WHYS | Seriously? No one likes any other softphones? |
| 21:50.13 | lmadsen | WHYS: I use X-Lite and Zoiper. Those are the only ones I like. |
| 21:53.01 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 21:53.46 | WHYS | Thanks Leif |
| 21:54.09 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:55.38 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) |
| 22:02.17 | *** join/#asterisk rhousand (n=rhousand@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:02.47 | rhousand | is there a max number of extension in a blastgroup? |
| 22:04.26 | ManxPower | rhousand: perhaps you are looking for #trixbox |
| 22:04.28 | [TK]D-Fender | rhousand: "blastgroup" is not an appropriate term, and its limited by string length in the dialpln (like all apps), and then system load if you chain local channels |
| 22:06.56 | rhousand | [TK]D-Fender: thanks, sorry about the incorect term. |
| 22:10.10 | *** join/#asterisk telnettech (n=telnette@12.236.122.2) |
| 22:10.27 | ManxPower | [TK]D-Fender: wasn't the line length increased n 1.4? |
| 22:11.16 | [TK]D-Fender | ManxPower: Not AFAIK. I can't remember who it was in here like half a year ago who hacked the source for this... |
| 22:11.20 | *** join/#asterisk CrazyTux (n=brandon@rrcs-67-52-124-226.west.biz.rr.com) |
| 22:11.39 | [TK]D-Fender | ManxPower: but then since it isn't a "bug" it woul not have been included in 1.4 now would it? |
| 22:11.57 | [TK]D-Fender | ManxPower: It was for a mass-page in his instance |
| 22:12.05 | *** join/#asterisk test34 (n=alex@unaffiliated/test34) |
| 22:12.38 | test34 | Which VOIP protocol is the most reliable ? |
| 22:12.52 | [TK]D-Fender | test34: Poor question. |
| 22:13.02 | [TK]D-Fender | test34: What makes a protocol "reliable"? |
| 22:13.27 | harry_v | Interesting New domain name extention .tel avaiable to the public on december 3rd. |
| 22:13.29 | test34 | tkd, can recover from errors for example ? |
| 22:13.37 | harry_v | wonder what took so long |
| 22:13.38 | harry_v | ;) |
| 22:13.49 | [TK]D-Fender | test34: what kind of errors? |
| 22:14.13 | test34 | tkd-fender, any and all |
| 22:14.20 | [TK]D-Fender | test34: How... generic. |
| 22:14.21 | test34 | or most |
| 22:14.52 | [TK]D-Fender | test34: That is an almost completely empty question |
| 22:14.53 | test34 | well I don't know exactly how they work in the background |
| 22:15.01 | test34 | so I can't be specific |
| 22:15.19 | [TK]D-Fender | test34: Consider them largely equal then. |
| 22:15.54 | test34 | tkd, which one do you like the most then ? |
| 22:16.08 | harry_v | test34, the protocols are most likly reliabile but how thay are delivered is the question. Or at all. Depends on the reliability of the transmission method bandwith ect between end points. |
| 22:16.21 | Carlos_PHX | Any way to over-ride the DND on a phone? Polycom in this case. |
| 22:16.38 | [TK]D-Fender | test34: SIP, because its common and interoperable |
| 22:16.58 | [TK]D-Fender | Carlos_PHX: Not unless there's a hidden header they didn't document. |
| 22:17.10 | test34 | harry_v: I just don't think that's the only difference... what would be the need to have more then one ? |
| 22:17.13 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-b2d885a06ce3c55a) |
| 22:17.15 | harry_v | Carlos_PHX why would you want to disable it? |
| 22:17.19 | ManxPower | Carlos_PHX: no official way. |
| 22:17.20 | [TK]D-Fender | Carlos_PHX: Because each endpoint does what it feels like. |
| 22:17.28 | Carlos_PHX | Yeah, that's what I figured. |
| 22:17.33 | Carlos_PHX | Wonder if there's a way to lock out DND. |
| 22:17.43 | harry_v | test34, evolution in packet delivery methods :) |
| 22:17.47 | ManxPower | Carlos_PHX: I'm sure there is. |
| 22:18.12 | Carlos_PHX | My answer was, "Fire the employee who is using DND when you said not to." Apparently was not the right answer. |
| 22:18.18 | [TK]D-Fender | test34: becasue there are differences in encryption, routing capabilities, signalling VS media, etc |
| 22:18.24 | talirk81 | In a AGI scrit i am setting SET VARIABLE HID=xxxx and i get a 200 result=1 in the console. However NoOp(${HID}) doesnt return anything , i also tried setting __HID to make it more global. Any ideas? |
| 22:18.59 | [TK]D-Fender | talirk81: pastebin is your friend <- |
| 22:19.01 | [TK]D-Fender | ~pb |
| 22:19.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:21.19 | test34 | ok thanks to both of you, that will help me in finding some more information |
| 22:22.09 | *** join/#asterisk ldsjohn (n=jbunn@c-24-22-52-147.hsd1.wa.comcast.net) |
| 22:22.45 | talirk81 | http://pastebin.com/m29a0898c |
| 22:23.23 | talirk81 | also just tried HID=xxxxx without the quotes and got the same result |
| 22:23.57 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 22:24.12 | ldsjohn | I have an asterisk phone system that worked fine behind a linksys router, but the router broke and my company shipped me a cisco 871w router configured with a firewall and vpn tunnel, now I can make calls out but when I get calls in my sip response packet disapears, does anyone happen to know what might cause that? |
| 22:25.17 | test34 | SIP has not defined procedures for handling device failure. If a proxy fails, the user agent detects this through timer expiration. It is the responsibility of the user-agent to send a re-INVITE to another proxy, leading to long delays in call establishment. |
| 22:25.38 | ManxPower | ldsjohn: There may be a "sip fixup" option enabled by default. Did you forward the ports? |
| 22:25.53 | [TK]D-Fender | talirk81: Please give the AGI Appendix chapter a good read on that particular command. |
| 22:26.08 | ldsjohn | I forwarded 5060 and the rtp ports, and I have turned off inspection on sip, that didn't make a difference |
| 22:26.35 | ldsjohn | I also tried removing all the access lists for the firewall and turning off all inspection and forwarding all traffic to the asterisk box, and still I get the same thing |
| 22:27.32 | harry_v | has anyone here configure a bcm50 before ? |
| 22:27.37 | ldsjohn | if someone calls me, my asterisk box starts the call procedure and sends a packet and hten it disapears in the router and never comes out, since I use callcentric as a voip provider, people hear about 30 seconds of silence while it waits for the packet, then the call times out and they hear The callcentric user you are calling can not be reached |
| 22:27.44 | StephenF | Anyone ever heard of ASterisk integration with ConnectWise CRM? |
| 22:28.04 | ManxPower | ldsjohn: make sure you have canreinvite=no in each sip.conf entry |
| 22:28.28 | ldsjohn | yup its in all of them including my trunk |
| 22:30.00 | ldsjohn | I have tried nat=yes externip=x.x.x.x and localnet in my sip.conf and I get the same thing |
| 22:30.18 | [TK]D-Fender | ldsjohn: APSTEBIN your sip.conf masking only passwords |
| 22:30.21 | [TK]D-Fender | \~pb |
| 22:33.43 | ManxPower | harry_v: No, but I know someone that will consult on BCM50 installs if you need someone |
| 22:34.29 | file | that was disturbing... just heard Allison's voice on a TV commercial |
| 22:34.37 | Qwell | O.o |
| 22:34.57 | lmadsen | neat |
| 22:35.03 | *** join/#asterisk OhSlap (n=ohslap@202.55.146.218) |
| 22:35.47 | ldsjohn | http://pastebin.ca/1263112 |
| 22:35.55 | ldsjohn | I think I got all the passwords out, they are replaced with ****** |
| 22:35.56 | [TK]D-Fender | %#@&^#@%#@##@ CRTC sided with Bell on throttling their wholesale customers! |
| 22:36.48 | [TK]D-Fender | ldsjohn: And why is 17-24 commented out? |
| 22:37.00 | lmadsen | [TK]D-Fender: welcome to 5 hrs ago |
| 22:37.17 | stencil | [TK]D-Fender: exactly those neo-cons are in Bell's pocket |
| 22:37.37 | ldsjohn | its commented out because I have uncommented them and tried it and then recommented them and tried it |
| 22:37.58 | [TK]D-Fender | ldsjohn: well right now its BAD, and I see no peer for your ITSP at all which is another guranteed bad thing. |
| 22:38.09 | ldsjohn | no difference if they are or arn't commented, with my old router just forwarding ports 5060 tcp / udp and ports 10000-20000 udp it worked |
| 22:38.14 | [TK]D-Fender | ldsjohn: Fix your configs and include a failed call attempt. |
| 22:38.25 | ldsjohn | uncomment those? |
| 22:38.30 | [TK]D-Fender | ldsjohn: with SIP debug enabled |
| 22:38.46 | [TK]D-Fender | ldsjohn: Fix your NAT settings. |
| 22:39.46 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 22:40.08 | ldsjohn | um I forgot the last bit of the sip.conf somehow |
| 22:40.14 | ldsjohn | http://pastebin.ca/1263117 shows the part I missed |
| 22:40.23 | ldsjohn | at the bottom the Peer Entry for callcentric |
| 22:41.01 | [TK]D-Fender | ldsjohn: Also bad because they should be "nat=no" |
| 22:41.03 | SkramX | Hi |
| 22:41.22 | ldsjohn | oh? |
| 22:41.33 | [TK]D-Fender | ldsjohn: And you you have not globally disabled reinvites. |
| 22:41.41 | SkramX | how can I do an agentcallbacklogin but have the extension satisfy a pattern? (example: agent login is actually a macro that dials their home phone.. not a SIP extension) |
| 22:41.49 | [TK]D-Fender | ldsjohn: 102 will definitely fail |
| 22:42.13 | ldsjohn | 102 can is what I am using to test everything, its the phone that inbound calls go to |
| 22:42.14 | [TK]D-Fender | SkramX: Pastebin is your friend <- |
| 22:42.26 | ldsjohn | so you might have discovered the problem |
| 22:42.42 | [TK]D-Fender | ldsjohn: it should not be allowed to reinvite. in fact NONE of your devices should |
| 22:43.02 | ldsjohn | I need to uncomment the externip and localnet stuff first off |
| 22:43.07 | SkramX | [TK]D-Fender: okay- just thought of something though.. be right back |
| 22:43.27 | ldsjohn | should I also uncommnet canreinvite=no and nat=yes in the general section? |
| 22:44.00 | ldsjohn | then add Nat=no and canreinvite=no to my peer? |
| 22:44.04 | [TK]D-Fender | ldsjohn: yes |
| 22:47.12 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 22:51.24 | jplank | any of you guys digium distributors ? |
| 22:52.39 | ldsjohn | http://pastebin.ca/1263139 |
| 22:53.01 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 22:53.05 | ldsjohn | thats my full log from just before calling to just after I get callcentrics message saying the person is not available |
| 22:54.53 | [TK]D-Fender | ldsjohn: [Nov 20 14:51:08] VERBOSE[30098] logger.c: -- Executing [17772484176@from-pstn:2] Gosub("SIP/66.193.176.35-091504e8", "cidlookup|cidlookup_1|1") in new stack |
| 22:55.03 | [TK]D-Fender | ldsjohn: this is the last thing executing before THEY cancelled the call |
| 22:55.26 | CrazyTux | [TK]D-Fender: do you know where I can get a list of the Reason: clause values |
| 22:55.29 | CrazyTux | [TK]D-Fender: for the Ast AMI |
| 22:55.32 | CrazyTux | Events |
| 22:55.44 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:55.51 | [TK]D-Fender | CrazyTux: go check the usual places |
| 22:55.59 | *** join/#asterisk Akiyuki (n=root@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 22:56.10 | [TK]D-Fender | ldsjohn: I'm wondering if that line is hanging too long. |
| 22:56.15 | Akiyuki | Is it possible to use commands strung together? like foo(bar(baz) ) ? |
| 22:56.25 | CrazyTux | [TK]D-Fender: voip info dosent have to much info |
| 22:56.33 | [TK]D-Fender | Akiyuki: You can use FUNCTIONS insider of applications... |
| 22:57.07 | lmadsen | and functions inside of functions |
| 22:57.27 | Akiyuki | so SayPhonetic(Festival(foo) ) would work? |
| 22:57.36 | [TK]D-Fender | Akiyuki: No |
| 22:57.43 | [TK]D-Fender | Akiyuki: Festival is not a FUNCTION |
| 22:57.49 | SkramX | [TK]D-Fender: http://pastie.org/320102 - agent login issue |
| 22:58.05 | Akiyuki | Looks like a function in programming languages. What is it called then? |
| 22:58.06 | ManxPower | For a list of applications do "core show applications" for a list of functions do "core show functions" |
| 22:58.23 | ManxPower | functions are UPPER CASE, BTW. |
| 22:58.41 | ManxPower | Akiyuki: In Asterisk functions act much more like variables |
| 23:01.04 | SkramX | :\ |
| 23:01.08 | [TK]D-Fender | SkramX: dump your dialplan |
| 23:01.23 | SkramX | my dialplan is partyly in an AGI |
| 23:01.29 | SkramX | what part are you wanting to see? |
| 23:02.34 | SkramX | wait |
| 23:02.39 | [TK]D-Fender | SkramX: dump the whoe thing |
| 23:02.42 | SkramX | i think it's working now - did i not reload? |
| 23:03.35 | SkramX | yay |
| 23:04.41 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 23:05.06 | [TK]D-Fender | Ok, time to head off for martial arts. Back later |
| 23:05.09 | [TK]D-Fender | is off |
| 23:05.10 | Simon-- | anybody else experiencing VNAK/INVAL IAX frame storming on 1.4.21.2? |
| 23:05.56 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 23:15.05 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 23:20.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 23:20.32 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 23:21.05 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:22.23 | trogs | anyone point me in the right direction of some good sip trunk providers, probably don't need DDIs just bulk outbound minutes. |
| 23:31.01 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 23:37.42 | baliktad | http://www.dslreports.com/gbu/ |
| 23:39.27 | baliktad | I have used and like VoicePulse. DID rates not the best but they have great domestic rates, most of the time less than $.01/min |
| 23:40.17 | *** join/#asterisk eit (n=eit@64.122.178.15) |
| 23:41.23 | *** join/#asterisk jeffgus (n=jeffgus@alpha.zimage.com) |
| 23:45.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:46.10 | Simon-- | svn: PROPFIND of '/svn/asterisk/branches/1.4': 405 Method Not Allowed (http://svn.digium.com) |
| 23:46.15 | Simon-- | pft..where did svn.digium.com go? |
| 23:46.43 | jameswf | I use some fancy scripting with gizmo which makes a chunk of my calls free |
| 23:48.16 | eit | Hello everyone, I have a weird one that I'm hoping someone has encountered or has some better insight than I am coming up with. They system is a freepbx version 2.3.1. Asterisk is version 1.2.26. they are using PRI's with digium te210p. The issue is that certain calls to specific numbers will have issues once the far end VM system answers. On issue in particular appears to be that asterisk mistakes the leave-a-message-tone to be a a fax tone and dumps the c |
| 23:48.41 | jameswf | ~freepbx |
| 23:48.41 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:49.01 | eit | Okay thanks. |
| 23:56.15 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
