| 00:00.41 | *** join/#asterisk Abdu (n=Lgvp@201-25-178-218.mganm702.dsl.brasiltelecom.net.br) |
| 00:00.54 | mafkees | vlt|home: basically, the bristuff patches in support for their cards, that's it |
| 00:01.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:01.08 | mafkees | it will not touch support for PRI and analog stuff |
| 00:01.33 | vlt|home | mafkees: no other non-sip stuff installed. |
| 00:01.50 | Dovid | that u got 3 diffrent places wit diffrent opinions |
| 00:02.27 | Phel | No, I've tried with 3 different SIP service providers |
| 00:02.29 | mafkees | vlt|home: then it's safe to reload everything |
| 00:02.40 | Phel | Follwed their directions |
| 00:02.56 | mafkees | I tried ekiga once |
| 00:03.03 | Phel | DId it work 4 U |
| 00:03.09 | mafkees | I did not like it so I never took time to get it working |
| 00:03.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:03.46 | mafkees | for sip softphone I use x-lite or twinkle |
| 00:03.58 | mafkees | depends wether the client is running kde or not |
| 00:04.03 | Phel | I think it's my router |
| 00:04.53 | vlt|home | JT, mafkees: I loaded chan_zap with the unpatched qozap driver and my card is shown and recognized by zap. Even an incoming call was shown on *CLI ... |
| 00:05.38 | mafkees | what patches for qozap did you use before ? |
| 00:05.39 | vlt|home | mafkees: What traps will I face using the junghanns driver? |
| 00:05.45 | mafkees | <--- never had to patch his qozap |
| 00:05.50 | mafkees | none |
| 00:06.06 | vlt|home | mafkees: I found a patch from someone using his beronet card with bristuff |
| 00:06.15 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:06.17 | JT | vlt|home: ok, can you go further than that? |
| 00:06.22 | JT | vlt|home: can you handle calls? |
| 00:06.26 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:06.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:06.41 | mafkees | hhmm |
| 00:06.42 | DocHolliday | lol |
| 00:06.47 | vlt|home | JT: Never used zap before .. will read a bit and brb ... |
| 00:06.48 | mafkees | I would use stock qozap |
| 00:06.52 | JT | maybe junghanns removed the stuff that stops it working for non junghanns card, or maybe it only blocks beronet, his main rival |
| 00:07.08 | mafkees | no, beronet works fine on stock qozap |
| 00:07.57 | JT | maybe junghanns changed his stance |
| 00:08.06 | JT | otherwise why would people post patches? :) |
| 00:08.12 | mafkees | gheh |
| 00:08.19 | mafkees | look at the digium bugtracker ;) |
| 00:08.24 | mafkees | talking bout patches |
| 00:08.24 | JT | ? |
| 00:08.30 | JT | heh |
| 00:08.42 | mafkees | every software has bugs |
| 00:08.49 | mafkees | bristuff is no exception |
| 00:08.59 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-141-154-130.pppoe.mtu-net.ru) |
| 00:09.07 | vlt|home | JT, mafkees: My card is explicitly mentioned in his original qozap.c as "evaluation board" but I can't find any beronets there, Maybe JT is right ;-) |
| 00:09.27 | JT | mafkees: no but the patch was not for a bug, it was for an intentional vendorid check |
| 00:09.35 | JT | to prevent beronet from working |
| 00:09.52 | mafkees | ah |
| 00:10.00 | mafkees | that's not a block by junghanns |
| 00:10.08 | mafkees | it's simply something they did not know |
| 00:10.20 | vlt|home | ;-) |
| 00:10.23 | mafkees | they wrote their driver with their own pci-id's in mind |
| 00:10.40 | mafkees | actually it was nice |
| 00:10.43 | JT | heh, debatable |
| 00:10.49 | mafkees | it meant they only touched their own hardware |
| 00:10.57 | mafkees | to prevent other hardware drivers to bork |
| 00:11.07 | JT | mafkees: you on the bristuff mailing list? |
| 00:11.19 | mafkees | ehm, yeah |
| 00:11.22 | mafkees | I host it |
| 00:11.27 | JT | that's right |
| 00:11.29 | JT | i forgot |
| 00:11.38 | mafkees | lol, never mind |
| 00:11.45 | mafkees | three-dimensional.net is my company |
| 00:13.18 | mafkees | 36 members right now |
| 00:13.21 | mafkees | still small |
| 00:13.55 | mafkees | 28 in the first day |
| 00:13.58 | mafkees | that was nice to see |
| 00:14.29 | mafkees | let me spam it once more |
| 00:14.47 | mafkees | There's now an in-official bristuff-users mailinglist at http://lists.three-dimensional.net/mailman/listinfo/bristuff-users |
| 00:14.53 | vlt|home | JT: btw: --- Results after 16 passes --- Best: 100.000000 -- Worst: 99.987793 -- Average: 99.994659 |
| 00:15.08 | *** join/#asterisk mog (i=ejabberd@71.207.215.93) |
| 00:15.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:15.14 | mafkees | We tried to contact junghanns.net about it but we got no reply so feel free to subscribe to it |
| 00:15.17 | mafkees | heya mog |
| 00:15.25 | mafkees | you return 1 second too late |
| 00:15.37 | mafkees | vlt|home: that looks a lot better ! |
| 00:15.43 | vlt|home | :) |
| 00:16.18 | mog | ? |
| 00:16.26 | mafkees | There's now an in-official bristuff-users mailinglist at http://lists.three-dimensional.net/mailman/listinfo/bristuff-users |
| 00:16.29 | edgecase | Dovid, re: t.38, a guy in hong kong Steve Copice seems to be the only one in the world working on it. my SIP terminator unlimitel.ca supports t.38 BTW |
| 00:16.58 | *** part/#asterisk lencho (n=lencho@pool-72-78-116-222.phlapa.fios.verizon.net) |
| 00:16.58 | mafkees | ;) |
| 00:17.01 | mafkees | spamming there |
| 00:17.04 | Dovid | what do u mean by working on it |
| 00:17.05 | Dovid | ? |
| 00:17.18 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:17.31 | JT | Dovid: writing code to handle actual t.38 termination |
| 00:18.32 | mafkees | gowd |
| 00:18.43 | mafkees | I need 3 ipods to hold all my mp3z |
| 00:19.01 | mafkees | oh wait, the new ipod is bigger |
| 00:19.16 | mafkees | I thought 20gb was the biggest one |
| 00:19.34 | DocHolliday | edgecase, have you attempted to use it? |
| 00:19.54 | Abdu | CDR Works with canreinvite=yes ? |
| 00:20.12 | mafkees | Abdu: yeah, only the RTP will be peer-to-peer |
| 00:20.19 | edgecase | yeah there's a library used by t38modem (h323) PSTN -> h323 -> softmodem in asterisk, he's adapting for PSTN -> sip terminator's modem -> t.38 over IP -> asterisk t.38 to image file or whatever |
| 00:20.21 | mafkees | the control channel will stay at asterisk |
| 00:20.39 | edgecase | DocHilliday, i'm waiting for the sip -> t.38 part to be finished |
| 00:20.48 | vlt|home | mafkees, JT: I can access the BRI channels!!!!!!11!11!11one! Thank you! |
| 00:20.49 | edgecase | fax over u-law is too flakey for me |
| 00:20.56 | mafkees | vlt|home: congrats :) |
| 00:20.57 | JT | vlt|home: sweet |
| 00:20.59 | mafkees | good work |
| 00:21.10 | JT | edgecase: over u-law over what? |
| 00:21.14 | mafkees | edgecase: fax over u-law will work 5% of the time |
| 00:21.26 | mafkees | I tried with SIP and IAX2 |
| 00:21.34 | edgecase | yeah, ulaw over sip/rtp |
| 00:21.42 | JT | Abdu: yes cdr may work, but is susceptible to toll fraud if you don't control the media path |
| 00:21.44 | mafkees | even on lines with 0.00000% packetloss and latency of 5ms it's not perfect |
| 00:21.48 | JT | edgecase: sip over what? |
| 00:21.49 | JT | :) |
| 00:22.05 | mafkees | JT: asterisk only supports SIP over udp |
| 00:22.17 | mafkees | no tcp support yet |
| 00:22.24 | mafkees | I think oej is working on it |
| 00:22.25 | edgecase | the idea of t.38 is the "modem" is in the PSTN gateway, and fax HDLC frames are encapsulated in udptl or RTP over IP |
| 00:22.27 | vlt|home | edgecase, mafkees: I reached up to 20 % successful fax over u-law ;-) then switched to T.38. Now got 98% |
| 00:22.48 | mafkees | vlt|home: in passthru mode ? |
| 00:22.53 | edgecase | you have a t.38 ATA -> analog fax machine? that is supported in asterisk apparently, yeha passthru |
| 00:22.58 | JT | mafkees: i know, over the internet, a lan.... |
| 00:23.03 | JT | rtp is over udp anyway |
| 00:23.21 | vlt|home | mafkees: To be true: Bypassing asterisk at all. ATA <---> QSC server |
| 00:23.27 | mafkees | lol |
| 00:23.40 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 00:23.47 | DocHolliday | edgecase, i want to use a T.38 ATA --> Asterisk Pass Thru --> T38 Provider (any thoughts?) |
| 00:23.48 | mafkees | I use the fax2mail and mail2fax services my IAX provider gives me |
| 00:23.53 | mafkees | I get 100% succes |
| 00:24.05 | jql | Doc |
| 00:24.10 | mafkees | but they use their sangoma cards to get the fax, forward it with iaxmodem to hylafax |
| 00:24.15 | ManxPower | DocHolliday: As I understand it you need 1.4 to do that. |
| 00:24.26 | mafkees | indeed |
| 00:24.37 | mafkees | 1.4 supports T.38 passthru |
| 00:24.49 | DocHolliday | correct |
| 00:24.58 | JT | some stuff even supports t.38 for real ;) |
| 00:25.05 | DocHolliday | but if i were to do that, would everything be pretty? |
| 00:25.19 | Abdu | JT if i use canreinvite=yes i can have fraud with cdr ? |
| 00:25.29 | edgecase | well t.38 is limited to 19.2k baud yes? not fancy colour faxes etc |
| 00:25.35 | vlt|home | JT: Doesn't OpenPBX support T.38? |
| 00:25.43 | JT | Abdu: yes, i guess it depends on your clients |
| 00:25.46 | JT | vlt|home: yes |
| 00:25.58 | DocHolliday | edgecase, i just want black and white.. would the faxes send/receive with little problems? |
| 00:26.01 | JT | i've never seen a colour fax |
| 00:26.20 | jql | Brother supposedly sells them |
| 00:26.24 | jql | I have yet to buy one |
| 00:26.27 | edgecase | DocHolliday, well that's the dream, to no deal with a fax machine on your end, but it's still vaporware |
| 00:26.36 | jql | damn patents and piss-poor libtiff support... |
| 00:26.43 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:26.52 | edgecase | i hate faxes, i hope people switch to scan/email and colour fax doesn't take off |
| 00:27.00 | *** join/#asterisk osiris (n=osiris@71.205.27.131) |
| 00:27.02 | ManxPower | Reinvites only reinvite the AUDIO, the signalling still stays with the server |
| 00:27.04 | mafkees | same here edgecase |
| 00:27.12 | JT | edgecase: scan email isn't very legally strong |
| 00:27.14 | DocHolliday | JT, yeah.. i own one but the difficulty is finding someone else that owns one :) |
| 00:27.29 | mafkees | JT: a fax is as easy to manipulate as email |
| 00:27.34 | jql | don't need color faxes for transmitting legal signatures, either |
| 00:27.37 | DocHolliday | edgecase, noo i will have a fax plugged in to the ATA that supports T38... |
| 00:27.40 | mafkees | email has the benefit of having gpg/pgp |
| 00:27.48 | edgecase | mafkees, yeah technically weak, but legally strong, what a mess |
| 00:27.51 | JT | mafkees: once you hand off the media path to untrusted parties, you are liable to toll fraud/inaccurate CDRs |
| 00:28.06 | DocHolliday | indeed, i dont mind faxes as long as they work :P |
| 00:28.11 | JT | mafkees: err, faxes go through the pstn |
| 00:28.15 | mafkees | so ? |
| 00:28.19 | edgecase | american bar association is working on legal foundation for digital signatures |
| 00:28.21 | JT | mafkees: all pstn calls have CDRs |
| 00:28.27 | JT | at the telcos |
| 00:28.28 | mafkees | nope |
| 00:28.30 | JT | who are independant |
| 00:28.35 | JT | in 1st world countries :) |
| 00:28.43 | mafkees | lol, you wish |
| 00:28.58 | mafkees | in .nl the digital signatures are legal stuff |
| 00:29.25 | edgecase | ok here's a good question for #asterisk, my SIP termination provider doesn't allow re-INVITE on a hairpin call in my * and out thru them again |
| 00:29.28 | mafkees | I even sign my tax papers with my digital signature and send them with email |
| 00:29.53 | edgecase | smartcards or software keystore or ? |
| 00:29.53 | vlt|home | in .de there's something called "qualified digital signature" |
| 00:29.53 | JT | mafkees: well telcos keep records of calls for quite some time |
| 00:30.15 | mafkees | edgecase: software keystore |
| 00:30.34 | mafkees | JT: that's only because goverment wants to tap your line |
| 00:30.36 | edgecase | i like the german HBCI banking, use your linux + smartcard to do banking |
| 00:30.52 | mafkees | edgecase: we have that in .nl as well |
| 00:30.57 | edgecase | cool |
| 00:31.00 | JT | mafkees: sure, and telcos also like to be able to settle fees with each other |
| 00:31.13 | mafkees | linux/osx/windows/openbsd + bankcard + device to generate key |
| 00:31.19 | JT | and telcos generally just keep records out of habbit |
| 00:31.31 | JT | IT seems to be good at not keeping good records |
| 00:31.43 | jql | the number of phone-calls in the world grows slower than the amount of storage available, it seems |
| 00:32.03 | mafkees | I do hate .nl goverment regulations tho |
| 00:32.05 | edgecase | so serious question, how do i tell * 1.4 not to do reinvite on hairpin PSTN -> SIP bridged to SIP -> PSTN, but still do reinvite for SIP bridged to SIP -> PSTN |
| 00:32.08 | jql | even a billion calls a day can go on a single drive |
| 00:32.16 | mafkees | as an ISP I have to keep logs of all the traffic for 10 years |
| 00:32.21 | mafkees | that's plain sick |
| 00:32.45 | mafkees | edgecase: use a different sip account for it |
| 00:32.48 | edgecase | i've tried the 't' flag in Dial command |
| 00:32.50 | ManxPower | edgecase: you would have two sip peers in sip.conf |
| 00:32.53 | Abdu | JT i need to have nat=no to canreinvite work ? |
| 00:32.59 | mafkees | yeah |
| 00:33.08 | JT | Abdu: nat has nothing to do with it |
| 00:33.17 | edgecase | yes the difference between peer, user, friend, and which register associates with is unclear in docs |
| 00:33.29 | mafkees | JT: with 'nat=always' canreinvite=yes wont work |
| 00:33.44 | Abdu | mafkees with nat=yes dont work to ? |
| 00:33.47 | JT | what is the 1.2 equivalent of always? |
| 00:33.51 | edgecase | ManxPower, yes i was thinking to have a 2nd peer, that calls which shouldn't reinvite would use in Dial() |
| 00:34.02 | JT | mafkees: it depends where your clients are anyway |
| 00:34.06 | mafkees | I think nat=yes will work |
| 00:34.12 | mafkees | JT: true true |
| 00:34.22 | ManxPower | edgecase: Just remember you will still have correct CDRs even if you reinvite |
| 00:34.23 | *** part/#asterisk mog (i=ejabberd@71.207.215.93) |
| 00:34.31 | Abdu | mafkees and with nat=no i need to work ? |
| 00:34.31 | mafkees | canreinvite only works when they are no the same subnet |
| 00:34.50 | mafkees | Abdu: nat=no and canreinvite=yes will work yeah |
| 00:34.57 | Abdu | ok ok |
| 00:34.58 | Abdu | thankz |
| 00:35.00 | Abdu | let me test |
| 00:35.01 | edgecase | ManxPower, ie CDRs follow signaling not media path? |
| 00:35.08 | mafkees | edgecase: indeed |
| 00:35.16 | ManxPower | edgecase: CDRs are part of the signaling. |
| 00:35.24 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 00:35.24 | vlt|home | mafkees: How can I disable span 4 (that's spamming syslog and *CLI)? I commented the lines in zaptel- and zapata.conf, reloaded and even restarted * ... idea? |
| 00:35.34 | jql | they have to follow signalling. early media would wreck things for pstn-style billing otherwise |
| 00:35.47 | edgecase | apparently 1.2/1.4 changed how reinvites work, there should have been another canreinvite= setting IMO |
| 00:36.07 | mafkees | vlt|home: remove it from /etc/zaptel.conf, unload and load module |
| 00:36.24 | mafkees | vlt|home: remove it from /etc/asterisk/zapata.conf after that an restart asterisk |
| 00:36.26 | edgecase | is there any better documentation on how sip.conf work re: friend/peer/user/foe |
| 00:36.36 | Abdu | mafkees if the ATA uses stun the canreinvite dont work ? |
| 00:36.40 | mafkees | edgecase: try the wiki |
| 00:36.51 | edgecase | voip-info.org? that was incomplete |
| 00:36.57 | mafkees | Abdu: I have no idea, I never used stun |
| 00:36.58 | ManxPower | edgecase: type=friend can make and receive calls, a user can only send calls to asterisk, a peer can only receive calls from asterisk |
| 00:37.03 | edgecase | use the source eh? |
| 00:37.15 | mafkees | know the force, read the source |
| 00:37.39 | jql | one of these days, I'll have to learn the magic of stun |
| 00:37.42 | edgecase | hmm well i'll have something worthy of posting for my efforts |
| 00:38.17 | mafkees | I'm off to have sex |
| 00:38.19 | mafkees | latero all |
| 00:39.05 | DocHolliday | holly shit asterisk 1.4 sounds soo clear |
| 00:39.23 | JT | ManxPower: yes CDRs are based of signalling |
| 00:39.33 | DocHolliday | sorry for the profanity |
| 00:39.45 | jql | yeah, they turned it up to 11 |
| 00:39.51 | JT | ManxPower: they are susceptible to toll fraud if you offer a charged service and you allow reinvites. |
| 00:40.03 | ManxPower | JT: How. |
| 00:40.22 | vlt|home | Next problem: "app_queue.c:3244 queue_exec: Unable to join queue 'officeq'" when answering zap channel. Incoming sip calls enter the queue w/o problems. What's different in handling the channels? |
| 00:40.39 | ManxPower | If signalling stays with the server, then how would fraud happen in a way that was diferent if the media stayed with the server. |
| 00:41.47 | JT | ManxPower: because something could happen to the media and the server can't see it, it sees only the view presented by signalling, that's it |
| 00:42.16 | JT | ask any voip provider offering service to members of public what they think of reinvites |
| 00:42.22 | JT | in the end it's a business decision/risk |
| 00:42.57 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:42.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:43.02 | *** join/#asterisk tzanger (n=tzanger@208.68.91.47) |
| 00:43.14 | ManxPower | JT: The only "fraud" I can see is to redirect the audio to some other device. |
| 00:43.21 | ManxPower | It's still not billable. |
| 00:44.29 | edgecase | well if the signaling didn't enforce media disconnect and vice-versa, that would be a weakness, ie reinvite, then signal hangup, but keep media going? |
| 00:44.31 | JT | well not so much on fraud, but billing accuracy, what if you lose the media path? sometimes it's not nicely in the signalling |
| 00:45.04 | JT | edgecase: yep |
| 00:45.13 | ManxPower | JT: If you lose the mediapath so what. User will be billed until the signaling path is torn down. |
| 00:45.29 | edgecase | it seems like a pretty lame PSTN gateway that would allow that |
| 00:45.36 | ManxPower | edgecase: I agree. |
| 00:46.05 | edgecase | signaling and media seem to be split for scalability, perhaps creating this issue |
| 00:46.35 | JT | it's also a LIG problem, reinvites |
| 00:46.43 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 00:46.54 | JT | i think most voip providers offering public voip service handle media all the time |
| 00:46.58 | JT | especially due to NAT too |
| 00:47.05 | jql | yeah |
| 00:47.11 | jql | all media goes through my servers |
| 00:47.16 | edgecase | ok so if str2ba is in libbluetooth.a, it's probably not in libbluetooth.so yes? |
| 00:47.39 | edgecase | oh 2 customers with media re-invited direct between them? seems voip peering should be free to me |
| 00:47.41 | ManxPower | JT: Many ITSPs outsource their PSTN stuff anyway. |
| 00:47.56 | JT | ManxPower: 'pstn'? |
| 00:48.02 | JT | what do you mean pstn stuff |
| 00:48.16 | ManxPower | JT: I mean calling non-voip numbers. |
| 00:48.39 | ManxPower | i.e. Many ITSPs use Level 3 for DIDs, for example. |
| 00:48.47 | JT | ManxPower: sure, you could have a wholesaler, unless you're a virtual itsp, probably still goes through your servers |
| 00:49.31 | ManxPower | JT: reinvites would allow you to not have the media go thru your servers |
| 00:49.46 | ManxPower | the media could go direct to Level 3, for example. |
| 00:50.09 | jql | if my upstream provider actually allowed that, I might consider enabling reinvites |
| 00:50.11 | jql | but, no |
| 00:50.22 | JT | ManxPower: you wouldn't do that anyway, most customers would have NAT |
| 00:50.40 | JT | and you want a high level of call control, which could include cutting off the media |
| 00:50.46 | JT | or recording it |
| 00:50.48 | edgecase | i think there's a difference depending on which end of the media path you want to move |
| 00:50.57 | jql | recording: ding ding ding. CALEA |
| 00:51.11 | edgecase | does that extend beyond PSTN now? |
| 00:51.11 | JT | yeah, CALEA, LIG for the rest of us |
| 00:51.44 | JT | edgecase: if you're calling enum numbers, that's really pstn now even if on voip, but yes, that act includes it |
| 00:51.55 | edgecase | interesting |
| 00:52.21 | edgecase | so voip -> voip with end to end encrytped media stream ? |
| 00:52.30 | JT | seemed like common sense to me that if you offered paid enum service, authorities may come to you wanting to listen in :) |
| 00:52.44 | jql | if you facilitate voip connections, as far as a court is concerned, it's your responsibility to enable tapping |
| 00:52.48 | JT | in the US it's more official now |
| 00:53.08 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 00:53.36 | edgecase | if users were to download a softphone from a 3rd party that enabled encryption, that would get interesting |
| 00:54.09 | jql | you could easily scramble the binary audio data to transmit anything. even setup a vpn over rtp |
| 00:54.11 | JT | sure you can do that already in a p2p scenario |
| 00:54.38 | jql | I implemented something like that in Perl, for laughs |
| 00:55.05 | jql | it sounded like nails on a chalkboard, but whatever |
| 00:55.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:56.36 | *** join/#asterisk orlock (i=jwr@202.44.174.4.static.nexnet.net.au) |
| 00:58.41 | JT | i think the only place i'd consider reinvites is on a LAN where calls aren't charged |
| 00:58.55 | JT | but then reinvites seem a bit academic with that much bandwidth :) |
| 00:59.11 | orlock | hmm.. |
| 00:59.18 | orlock | damn QoS doesnt seem to be working right |
| 00:59.47 | JT | orlock: so much trouble for so many months, are you sure your voip provider isn't just crap? ;) |
| 01:00.19 | edgecase | oh wow $ gcc -o chan_bluetooth.so -lbluetooth -shared -Xlinker -x chan_bluetooth.o |
| 01:00.25 | edgecase | the -l bluetooth is wonderful stuff :) |
| 01:00.38 | vlt|home | "Unable to join queue 'officeq'" on zap was my fault, a typo, its name is just "office". Sorry. |
| 01:01.27 | orlock | JT: nah, just that trying to concentrate on QoS rules when you get inturrupted every 15 minutes aint easy |
| 01:01.43 | JT | QoS is such a bother |
| 01:01.48 | JT | easier to not do it ;) |
| 01:01.55 | orlock | yeah |
| 01:01.56 | orlock | grrr |
| 01:02.17 | orlock | the thing is, i implemented it at home ok |
| 01:02.33 | orlock | but on production ites it doesnt seem to function as well |
| 01:04.24 | orlock | need to hook asterisk into th QoS script so you can change dedicated bandwidth on the fly |
| 01:04.34 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:05.11 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 01:05.14 | JT | how exciting :P |
| 01:08.05 | wunderkin | a note on my problem with polycoms and the problem when you press a button and it does not register... i have now noticed that when this happens, the cursor stops blinking... odd eh? sounds like a software/processor thing |
| 01:10.41 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 01:10.56 | *** join/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 01:10.58 | k-man | hello |
| 01:11.40 | [TK]D-Fender | wunderkin : if it isn't a bungled up config file, then I'd say the board is fried (lightly) |
| 01:11.43 | jql | howdy |
| 01:12.17 | JT | k-man: hi |
| 01:13.43 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:13.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:13.48 | wunderkin | [TK]D-Fender, well i'm back to using a standard config, but i had a new ip501 with a barebones config (only enough to register and a simple dialplan match) i forgot to change imposisble match to 2... but i know they still had key related problems, either keys not registering or the long dtmf thing.. sucks.. the client says he just started having this problem on his phone within the last month.. that phone is from the summer or fall.. |
| 01:14.13 | DocHolliday | for voicemail.conf can i set tz=EST? |
| 01:14.19 | *** join/#asterisk Brian|lfs (n=Brian@208-59-118-159.c3-0.161-ubr1.lnh-161.md.cable.rcn.com) |
| 01:14.30 | Brian|lfs | hello is anyone there |
| 01:14.42 | k-man | jt, do voip providers ever offer voicemail with their service |
| 01:14.43 | k-man | ? |
| 01:14.55 | wunderkin | several people started to report that problem around that same time... so you would think config... but even bare a new phone still sucks... zOmg |
| 01:14.56 | *** join/#asterisk jcool (n=zoro@58.69.226.211) |
| 01:16.00 | JT | k-man: some do |
| 01:16.00 | k-man | ok |
| 01:16.00 | k-man | thanks |
| 01:16.10 | wunderkin | i think the phones need a blessing or something, the sheeps are almost all gone |
| 01:16.32 | Brian|lfs | I installed trixibox on a machien over here and configured a sip client in it but what ports do I need to forward so people from the itnernet can connect to it |
| 01:17.13 | [TK]D-Fender | brian : 5060, 10000-20000 all UDP |
| 01:17.17 | Brian|lfs | I don't see anything about ports int eh documentation on the trixbox home page |
| 01:17.25 | k-man | jt, can you recommend an australian provider that has a voicemail service? |
| 01:17.42 | JT | brb |
| 01:18.02 | jql | australian? scary. the incumbent telco is scary, there |
| 01:18.14 | k-man | jql, yes |
| 01:18.24 | k-man | jql, i'm doing my best not to give them any of my money |
| 01:18.40 | Brian|lfs | cool thanks now I have to go read up on forwarding ports in shorewall |
| 01:18.43 | jql | the best of luck with that. nobody deserves that kind of abuse |
| 01:19.18 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 01:20.06 | jcool | good day guys, today i setup my * server to do b2bua to another * server, i'm encoutering some strange problem i can call to another * pbx, but no on ther way around any idea ? thanks |
| 01:20.43 | jcool | but not on the other way around* |
| 01:20.55 | jql | you can calll them, but they can't call you? |
| 01:21.00 | jql | is there a nat in the way? |
| 01:21.23 | jcool | jql: there is no nat involve since they are all on the same segments |
| 01:21.44 | jql | is your server rejecting the call, or never receiving it in the first place? (sip debug) |
| 01:22.38 | k-man | anyone know if faktortel offer voicemail? |
| 01:22.46 | k-man | i can't see it on their website |
| 01:23.01 | jcool | jql: w8 |
| 01:24.07 | k-man | oh, its ok, i found it |
| 01:24.23 | vlt|home | Does anyone know if and how I can set CFU on a BRI channel? |
| 01:24.56 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 01:25.03 | vlt|home | (CFU is Call Forwarding Unconditional, an ISDN function) |
| 01:25.06 | jcool | jql: pbx1 is not rejecting the call, but using the same type=friend account to go to pbx2 that's the problem started |
| 01:25.52 | jcool | jql: i will just get a fresh svn branch 1.2 then i'll report it back |
| 01:25.57 | jql | ok |
| 01:30.17 | JT | jql: drama much, re. australian telco ;)? |
| 01:30.25 | JT | k-man: i think engin does |
| 01:31.31 | jql | heh, long second-hand history with telstra. :) |
| 01:31.49 | JT | heh |
| 01:34.33 | *** join/#asterisk foxxtrot (n=craig@67.185.241.244) |
| 01:37.25 | DocHolliday | hey guys i want to set voicemail.conf for eastern standard time.. what do i set tz to? |
| 01:37.47 | wunderkin | anyone familiar with hpec? do you have to run zaphpec_enable during ever start up or just once? |
| 01:38.25 | bkruse_home | <3 hpec |
| 01:38.48 | JT | bkruse_home loves a lot of things |
| 01:39.02 | mog | bkruse_home, |
| 01:39.07 | mog | what is up |
| 01:39.19 | bkruse_home | mog: I will <fill in you know what word here> you |
| 01:39.30 | bkruse_home | russellb: tell kpfleming to have another party, that was fun |
| 01:39.31 | mog | exactly |
| 01:39.34 | mog | what i was thinking |
| 01:39.41 | mog | yeah that last one did rock |
| 01:39.51 | bkruse_home | mog: agreed. |
| 01:39.53 | bkruse_home | russellb: im working on the res, its going GREAT |
| 01:40.02 | jcool | jql: now, it's getting worse, the 2nd asterisk server can no longer register on the 1st huhuhu :) |
| 01:40.09 | bkruse_home | mogs been helpin me with my nub errors |
| 01:40.22 | russellb | bkruse_home: i saw your email. i dont' know, i guess i just need to install trunk and try it myself |
| 01:40.33 | bkruse_home | russellb: no! |
| 01:40.34 | bkruse_home | i fixed it |
| 01:40.39 | russellb | oh, ok |
| 01:40.41 | russellb | cool |
| 01:40.49 | bkruse_home | it didnt make much since, it should be if a->argc == e->args not != |
| 01:41.06 | bkruse_home | because then if # of args is equal to the default command, do usage: |
| 01:41.18 | bkruse_home | it made more sense once i thought about it, but thanks anyways, new_cli rocks. |
| 01:41.26 | russellb | right |
| 01:41.26 | jql | jcool: well, that's bad. What does the 1st server say about the attempted registration? |
| 01:41.32 | jql | peer not found? |
| 01:41.36 | bkruse_home | russellb: i wana start that janitor project after this |
| 01:41.43 | bkruse_home | make new_cli be the normal cli_ |
| 01:41.52 | bkruse_home | thats lots and lots of code, but this way is so much more effecient |
| 01:41.56 | orlock | hm. |
| 01:41.57 | bkruse_home | and fun :D |
| 01:41.58 | orlock | janitor |
| 01:42.00 | orlock | that sounds good |
| 01:42.15 | russellb | heh, yeah |
| 01:42.16 | orlock | at least you know the tools to keep the pipes flowing work in that job |
| 01:42.27 | russellb | i think that term came from the kernel community |
| 01:42.31 | russellb | they have kernel janitors |
| 01:43.14 | orlock | <----- digital janitor |
| 01:43.26 | russellb | we don't have official janitors ... but we always have a list of janitor type projects :) |
| 01:43.47 | bkruse_home | russellb: could i be an official janitor :D |
| 01:44.11 | russellb | maybe! |
| 01:44.34 | russellb | ok, i think I should go home before my body can't make it there |
| 01:44.37 | file | clean my office! |
| 01:44.39 | macTijn | blergh |
| 01:44.48 | macTijn | being a janitor isn't all that cool ;) |
| 01:44.52 | bkruse_home | agreed, i am tired sitting here on the couch |
| 01:45.00 | bkruse_home | macTijn: pfft, janitors own you. |
| 01:45.12 | mog | why dont you go to work bkruse_home |
| 01:45.49 | bkruse_home | mog: why dont you!!!!!! |
| 01:45.57 | macTijn | bkruse_home: I used to play janitor on lots of stuff, but all you get back is bitching from coders saying "nono it has to be like this!" |
| 01:45.59 | mog | heh |
| 01:46.06 | orlock | i wonder if janitors ever get complaints cos of the fact the toilet somebody installed upside down on the roof doesnt work right |
| 01:46.15 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:46.24 | orlock | "Hey, my ass gets wet when i use that thing! fix it!" |
| 01:46.28 | macTijn | bkruse_home: so no more janitoring :) |
| 01:46.46 | bkruse_home | :P |
| 01:47.10 | macTijn | you know what's the best way of auto-janitoring stuff ? |
| 01:47.12 | macTijn | rewrites. |
| 01:47.31 | macTijn | (do NOT tell me * doesn't need a rewrite.) |
| 01:50.25 | [TK]D-Fender | . |
| 01:53.52 | *** join/#asterisk n|cotine (i=nicotine@147.202.49.52) |
| 01:53.52 | macTijn | you know you're a bot. |
| 01:53.52 | n|cotine | If a sip device that has canreinvite=yes calls another sip device that has canreinvite=yes - at what point does asterisk remove itself from the media stream? |
| 01:53.52 | JT | bridging |
| 01:53.52 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 01:53.52 | n|cotine | How does one determine if that is happening? |
| 01:54.03 | JT | well can the ends talk to each other? |
| 01:54.07 | n|cotine | Yep. |
| 01:54.22 | JT | i guess they're bridging then |
| 01:54.48 | jcool | jql: now it's getting back to the same problem a whle ago, still i can call from pbx2 to the local extension of pbx1 but not from pbx1 going to local extension of pbx2 :( |
| 01:55.14 | jcool | jql: all i get is circuit busy from pbx1 |
| 01:55.47 | jcool | any idea guys, thank you! |
| 01:55.58 | n|cotine | JT: Not happening. |
| 01:56.05 | n|cotine | tcpdump shows media stream still going through the asterisk server |
| 01:57.42 | DocHolliday | whats the best way to have asterisk restart if it dies? |
| 01:57.51 | jql | I use safe_asterisk |
| 01:58.10 | jql | of course, that doesn't help with the kernel oopses... but, oh well |
| 01:58.13 | jql | can't be perfect |
| 01:59.10 | *** join/#asterisk oQPa (n=uawename@132.Red-81-38-251.dynamicIP.rima-tde.net) |
| 01:59.46 | JT | n|cotine: is both canreinvite and reinvite set to yes? |
| 02:00.02 | JT | jql: need a watchdog for that scenario |
| 02:00.14 | n|cotine | JT: I am confused - see http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+reinvite |
| 02:00.50 | n|cotine | And checking chan_sip.c, there is no mention of the 'reinvite' option |
| 02:01.00 | jql | duplicate server + remote power control = decent uptime |
| 02:01.35 | JT | yep |
| 02:02.03 | DocHolliday | Qwell, are you around? |
| 02:02.05 | jcool | JT: man can i ask a question please? |
| 02:02.14 | JT | sure |
| 02:02.18 | orlock | JT: http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
| 02:02.30 | orlock | JT: reinvite=yes/no is plain wrong, even if you see it mentioned in example .conf files. The correct syntax is canreinvite=yes/no |
| 02:02.44 | orlock | oh, he saw that already |
| 02:02.45 | orlock | :) |
| 02:02.48 | orlock | i was just reading it |
| 02:02.50 | JT | orlock: umm |
| 02:02.57 | JT | not sure about that |
| 02:03.01 | DocHolliday | "Voice-Message: %d/%d (0/0)\r\n", newmsgs, oldmsgs); whats the best way to edit that chan_sip.c line to get rid of (0/0)? |
| 02:03.13 | jcool | JT: i'm performing b2bua scenario on 2 asterisk server, i'm having a small problem, i can call from pbx2 to the local extesion of pbx1 but not on the other way |
| 02:03.35 | jcool | JT: all i get from pbx1 is circuit busy . |
| 02:03.45 | jql | it's safe to just edit it out |
| 02:03.52 | jql | and recompile |
| 02:11.47 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 02:16.13 | DocHolliday | woohoo ported chan_sip.c change to 1.4 :D |
| 02:17.48 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com) |
| 02:17.57 | wunderkin | orlock, how about dtmf=? :D heh |
| 02:18.08 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 02:18.31 | wunderkin | externalip? |
| 02:19.21 | Carp1 | externip |
| 02:19.27 | wunderkin | i know, you missed the joke |
| 02:19.37 | wunderkin | damn you nubbed it all up, thanks ;D |
| 02:19.45 | wunderkin | heh |
| 02:20.15 | Carp1 | lol |
| 02:20.19 | Carp1 | my bad coach |
| 02:21.17 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 02:30.20 | wunderkin | jakdjfsklfjad polycom |
| 02:31.33 | DocHolliday | what is the best way to restart asterisk if it crashes? |
| 02:31.53 | JT | what do you mean? |
| 02:32.43 | wunderkin | found another way to make a polycom reboot, nice |
| 02:33.25 | wunderkin | i think the gerbils in mine are always almost dead by the time i get them |
| 02:34.28 | wunderkin | dialing onhook, i keep dialing numbers until it reaches the limit, it looks like if i press another number after ive reached the limit, that makes it reboot :P sip 2.1.0 |
| 02:35.50 | jql | hah |
| 02:35.57 | jql | remind me not to use that version |
| 02:36.12 | DocHolliday | whats the best voip provider for worldwide calling plans? |
| 02:36.17 | wunderkin | not like anyone normally would notice that anyway |
| 02:36.59 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 02:40.43 | wunderkin | lets just try trusty rusty 1.6.7 |
| 02:40.58 | jql | an oldie, but a goodie |
| 02:45.32 | [TK]D-Fender | wunderkin : On my IP 501 w/ 2.1.0 I cannot replicate that. |
| 02:45.45 | [TK]D-Fender | wunderkin : What model have you encountered this with? |
| 02:46.45 | wunderkin | o rly? on an ip501 also... i formatted it and tried bare config... trying to 'fix' the button problem.. i selected a line key, kept pressing numbers, noticed it stopped.. kept pressing... then pressed menu... to see if it was ok.. and nope it started to reboot after a bit |
| 02:47.53 | wunderkin | i only added registration information, changed impossible match to 2, and changed the dialplan to x.T |
| 02:48.05 | wunderkin | edited the default files |
| 02:48.45 | ManxPower | wunderkin: want my default polycom config files? |
| 02:48.50 | wunderkin | won't do it now.. ugh |
| 02:48.56 | wunderkin | i saved the url :D |
| 02:49.04 | ManxPower | oh, SIP 2.0, nevermind |
| 02:49.07 | wunderkin | i got it to do it twice before |
| 02:49.16 | jql | our combined mental powers have prevented your phone from failing while we focus on it |
| 02:49.28 | wunderkin | keep up the good work |
| 02:50.01 | [TK]D-Fender | wunderkin : eek... mine just hard-locked.... |
| 02:50.11 | [TK]D-Fender | wunderkin : scratch that entirely. |
| 02:50.14 | wunderkin | must be intermittant :) |
| 02:50.17 | [TK]D-Fender | wunderkin : Reboot in progremm. |
| 02:50.20 | wunderkin | jql did you stop? :) |
| 02:50.24 | wunderkin | lol |
| 02:50.27 | [TK]D-Fender | wunderkin : No I sis it ON hook. misread. |
| 02:50.28 | jql | you didn't accidentally hit the reboot combinarion, did ya? :) |
| 02:50.33 | [TK]D-Fender | progress |
| 02:53.03 | wunderkin | at least i finally get you to replicate one of my problems so i'm not alone |
| 02:53.57 | wunderkin | did you keep pressing buttons? maybe it is something with the playing of the sounds for the dtmf... it isnt that immediately when you go over the limit it happens.. and it is intermittant |
| 02:54.24 | wunderkin | perhaps if that is true, that could be related to my other problem? |
| 02:54.41 | wunderkin | maybe turn off the dtmf chord? heh |
| 02:55.08 | *** join/#asterisk timphnode (n=tim@adsl-68-91-95-148.dsl.ksc2mo.swbell.net) |
| 02:55.36 | wunderkin | the problem regarding pressing a button and it stops responding... it acts like a cpu problem... the cursor stops blinking.. nothing responds.. |
| 02:56.13 | [TK]D-Fender | wunderkin : first guess.. input buffer overflow |
| 02:56.21 | wunderkin | information overload! |
| 02:56.29 | wunderkin | johnnie five no more input |
| 02:57.20 | [TK]D-Fender | wunderkin : Johnny Five NOT alive.... |
| 02:57.27 | wunderkin | nooooooo |
| 02:57.49 | wunderkin | the gremlins are after him |
| 02:58.34 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 03:00.13 | [TK]D-Fender | wunderkin : He was far better in Early Edition instead of being jsut another whitey playing and Indian role... |
| 03:00.52 | wunderkin | ... not sure who you're talkin about... i'm not that much into movies.. just the classics |
| 03:01.03 | wunderkin | you know, like office space.. ;) |
| 03:02.00 | wunderkin | since that problem is intermittant i would not think some kind of overflow... i still like my guess with the playing the sound thing |
| 03:02.13 | [TK]D-Fender | wunderkin : the guy who played the scientist following Johhny-5 around in Short Ciruict was in Early Edition, as well as being bad-guy in Hackers. |
| 03:02.44 | wunderkin | oh |
| 03:03.29 | wunderkin | the phone sounds funny when you press multiple buttons at once.. you can make some cool music.. |
| 03:05.04 | *** join/#asterisk hellop (n=hellop@udp112969uds.hawaiiantel.net) |
| 03:05.46 | hellop | hello |
| 03:06.07 | jql | howdy |
| 03:07.41 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:07.42 | *** join/#asterisk stridernzl (n=neville@125-237-114-232.jetstream.xtra.co.nz) |
| 03:07.53 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 03:09.08 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 03:10.57 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 03:11.33 | msw | I have a quick AMI question -- is there a way to originate a call and track its call progress sync |
| 03:11.41 | msw | synchronously? |
| 03:14.55 | wunderkin | there's supposted to be like a eventid or something you can tag to it... there used to be some problems with that... but i dont know.. havent used it |
| 03:14.57 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 03:15.47 | tim0123 | What is the best way to upload .conf files to mysql for Realtime |
| 03:16.05 | msw | *nod* - I can use that eventid to match up with current channels -- not sure if it will let me figure out if the call I originated failed due to busy, etc... |
| 03:24.12 | hellop | tim0123, upload? |
| 03:25.36 | tim0123 | Well read .conf files and update realtime asterisk tables |
| 03:27.42 | hellop | tim, what .conf files? asterisk files? like extensions.conf? |
| 03:27.47 | hellop | do reload command in the CL*I |
| 03:30.25 | tim0123 | Yeah |
| 03:31.10 | tim0123 | Basically Im try to get asterisk realtime working |
| 03:31.39 | tim0123 | But first I have to have my config in the database |
| 03:34.41 | Phel | [TK]D-Fender: Ping |
| 03:38.17 | *** join/#asterisk Foobs (n=chatzill@topaz.vintek.net) |
| 03:38.33 | JacksLivr | why 1.2 over 1.4 |
| 03:39.05 | *** join/#asterisk dacleric (n=dacleric@p54820981.dip0.t-ipconnect.de) |
| 03:39.52 | JacksLivr | do i need to install libnewt, or will the zaptel stuff install that? |
| 03:41.37 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 03:42.24 | *** join/#asterisk jarisu (i=jasmin@uranus.pacifica.ch) |
| 03:45.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:45.43 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:48.00 | msw | JacksLivr: libnewt is only really needed for zttool, which you don't have to have |
| 03:48.46 | JacksLivr | thanks. installed asterisknow yesterday and really wanted a clean, non gui. this is much more involved. trying to get one up and running now |
| 03:49.23 | JacksLivr | installed fedora6 and openssl and bison and made sure gcc is over 4.x |
| 03:50.18 | JT | JacksLivr: 1.2 is more stable, less bleeding edge |
| 03:50.34 | JacksLivr | thanks. about to download one |
| 03:50.42 | JacksLivr | dot 2 |
| 03:51.53 | *** join/#asterisk bmg505 (n=leon@c1-32-13.rndf.isadsl.co.za) |
| 03:53.01 | *** join/#asterisk cnet2 (n=nada@190.10.0.120) |
| 03:53.39 | JacksLivr | asterisk-1.2-current.tar.gz or asterisk-1.2.15.tar.gz ? |
| 03:54.07 | cnet2 | hi guys.. what's the top asterisk used phone? I always been between polycom and snom.. (polycom is really cool, but slow.., was thinking of try-n snom, but first wanted to ask your opinion(... |
| 03:55.44 | cnet2 | jackslivr, isn-t the current version 1.4_ |
| 03:56.06 | *** join/#asterisk Piano_ (n=Piano@unaffiliated/piano/x-000001) |
| 03:56.37 | JacksLivr | yeah, but the fellas here are suggesting 1.2 |
| 03:56.43 | JacksLivr | im learning |
| 03:57.29 | cnet2 | ohh i see. well, i-m using 1.4 since 2 weeks ago.. and it works great |
| 03:57.59 | jql | the snom is awesome for debugging |
| 03:57.59 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 03:58.14 | jql | but, I've gotten shot down on using it in the call center |
| 03:58.23 | jql | damned users didn't like it |
| 03:58.31 | jql | so, I stuck em with Polycoms |
| 03:58.52 | JT | [TK]D-Fender doesn't have too many nice things to say about snom :P |
| 04:00.10 | jql | the snom is a swiss-army knife. It's the Perl of voip phones |
| 04:00.35 | cnet2 | jql, so.._ |
| 04:00.42 | k-man | what do you call the facility to show who is on the phone? |
| 04:00.43 | cnet2 | why didn-t they like them.. too simple? |
| 04:00.58 | k-man | so i could see which users are on the phone? |
| 04:01.12 | jql | bleh. complained about the buttons, complained about the menu, complained about the display, complained about the handset |
| 04:01.20 | jql | whine, whine, whine |
| 04:01.35 | jql | k-man: busy lamp |
| 04:01.46 | cnet2 | so i'm guessing i should stay with polycom |
| 04:01.56 | k-man | does asterisk support busy lamp? |
| 04:02.07 | jql | k-man: to a degree, yes |
| 04:02.31 | k-man | do any voip phones support it? |
| 04:02.35 | cnet2 | in polycom the busy lamp works |
| 04:02.49 | jql | yeah, polycom, grandstream, snom all work. haven't tried the cisco |
| 04:02.51 | cnet2 | with asterisk.. lol |
| 04:02.53 | jpalmer | it works in snom too. |
| 04:03.09 | k-man | linksys? |
| 04:03.15 | jql | I think I tried the sipura, too |
| 04:03.39 | jql | but it didn't work |
| 04:03.44 | JT | jql: cisco must work, people keep talking about a change to the sip code to make it work, in here |
| 04:04.18 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:04.23 | jql | Maybe I'll try out the Cisco's BLF next week |
| 04:04.29 | k-man | blf? |
| 04:04.34 | k-man | what does the f stand for? |
| 04:04.40 | jql | field, I think |
| 04:05.01 | k-man | so linksys doesn't support that? |
| 04:05.35 | jql | not that I recall. Can't take my word as gospel, but I remember trying |
| 04:05.49 | k-man | ok |
| 04:05.50 | k-man | thanks |
| 04:16.11 | ez` | bkw__, again asterlink.com is unreachable ... |
| 04:16.11 | JacksLivr | asterisk-1.2-current.tar.gz or asterisk-1.2.15.tar.gz ? |
| 04:17.03 | *** join/#asterisk anrn (n=maroouch@209.183.16.67) |
| 04:18.55 | *** join/#asterisk Lgvp (n=Lgvp@201-25-178-218.mganm702.dsl.brasiltelecom.net.br) |
| 04:21.08 | Lgvp | ld: crtbegin.o: No such file: No such file or directory |
| 04:21.12 | Lgvp | someone can help me ? |
| 04:23.44 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 04:25.07 | Lgvp | ld: crtbegin.o: No such file: No such file or directory |
| 04:25.08 | Lgvp | someone can help me ? |
| 04:26.13 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 04:26.15 | *** join/#asterisk dseeb_ (n=dcb@58.169.73.237) |
| 04:42.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:42.49 | *** join/#asterisk cryptnix (n=andrew@64.25.197.253) |
| 04:51.08 | k-man | anyone heard of a roland e-35 keyboard? |
| 04:51.28 | k-man | does it have a midi interface? can i use it as a midi keyboard? |
| 04:52.42 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:52.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:54.24 | *** part/#asterisk anrn (n=maroouch@209.183.16.67) |
| 04:55.03 | orlock | k-man: asking some muso geeks i know.... tried google? |
| 04:55.17 | k-man | yeah... couldn't find much |
| 04:55.24 | k-man | i think its about 10-12 years old |
| 04:56.32 | [TK]D-Fender | k-man : Apparently, yes. |
| 04:56.37 | DocHolliday | any voip providers that have world calling plans? |
| 04:56.41 | [TK]D-Fender | k-man : http://reviews.harmony-central.com/reviews/Keyboard+And+MIDI/product/Roland/E-15/10/1 |
| 04:56.48 | [TK]D-Fender | k-man : Do you own it already? |
| 04:56.51 | k-man | [TK]D-Fender, you guys are tops |
| 04:56.59 | k-man | [TK]D-Fender, no, i saw someone offering it for free |
| 04:58.17 | [TK]D-Fender | k-man : Well.... if you don't like it, you can always ask for your money back :) |
| 04:58.24 | k-man | yeah |
| 04:58.40 | k-man | its more whether i can be bothered to pick it up and have it lying around |
| 04:58.46 | k-man | and will i really ever use it? |
| 04:58.51 | [TK]D-Fender | k-man : I own one of these : http://www.m-audio.ca/products/en_ca/KeystationPro88-main.html |
| 04:59.05 | k-man | oh right... |
| 04:59.12 | k-man | are youy happy with it? |
| 04:59.30 | k-man | looks pretty cool |
| 04:59.39 | DocHolliday | nobody has any provier recommendations? :( |
| 04:59.43 | k-man | [TK]D-Fender, have you played around with ardour at all? |
| 05:00.21 | k-man | oh... crap, its already taken... I was too late |
| 05:00.32 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:01.05 | [TK]D-Fender | k-man : nope.... not yet. I have it bookmarked for future reference though |
| 05:01.25 | k-man | [TK]D-Fender, yeah its pretty damn cool - although it doesn't support midi yet |
| 05:01.25 | [TK]D-Fender | k-man : Its a great controller that I'l eventually get around to learning how to really profit from. |
| 05:01.43 | k-man | cool |
| 05:01.50 | k-man | [TK]D-Fender, do you play a lot? |
| 05:03.44 | orlock | k-man: i know some linux geeks who are heavily into techno too |
| 05:04.00 | orlock | k-man: http://www.zog.net.au/ |
| 05:06.28 | [TK]D-Fender | k-man : not so much. I'm a minor hack on piano. I'm primarily a guitarist |
| 05:06.44 | k-man | [TK]D-Fender, ah i see |
| 05:06.52 | k-man | [TK]D-Fender, i'm also a guitarist |
| 05:06.55 | k-man | or at least trying to be |
| 05:07.03 | k-man | i think we might have spoken about this before actually |
| 05:09.07 | [TK]D-Fender | k-man : Shitty clip I made about a month and change ago when jsut starting to learn sweeps : http://aocomputing.net/sweeps1.mp3 |
| 05:09.49 | DocHolliday | i want to create an extension that dials to a menu, how can i accomplish this? |
| 05:10.17 | k-man | [TK]D-Fender, nice |
| 05:10.36 | JT | <PROTECTED> |
| 05:10.41 | jql | if that menu is another context, exten => 666,1,Goto(menu-context,s,1) perhaps? |
| 05:10.45 | JT | i should do some nice prompt editing |
| 05:11.01 | JT | k-man: i was bored the other day, so started recording crompts for "ChopTel" ;) |
| 05:11.05 | JacksLivr | zaptel is not compiling in fc6 || http://pastebin.ca/372599 |
| 05:11.19 | orlock | JT: uncle chop chop? |
| 05:11.22 | JT | yep |
| 05:11.37 | k-man | jt, what is choptel? |
| 05:11.51 | JT | fictional telco with the "voice" of mark "chopper" read |
| 05:11.53 | orlock | k-man: Mark Read |
| 05:12.06 | DocHolliday | Channel 'SIP/4000-b77013f8' sent into invalid extension 's' in context 'mainmenu', but no invalid handler |
| 05:12.25 | k-man | orlock, sorry, who is mark read? |
| 05:12.51 | orlock | k-man: infamouse australian hitman/criminal/standover man |
| 05:12.58 | k-man | oh |
| 05:13.01 | k-man | chopper read |
| 05:13.02 | orlock | whats known as a toecutter.. he only goes after crims |
| 05:13.02 | k-man | right |
| 05:13.05 | k-man | i'm with you now |
| 05:13.21 | [TK]D-Fender | DocHolliday : Maybe you could try pointing that Goto to a place that actually EXISTS... |
| 05:13.45 | k-man | jt, where are you recording it from? |
| 05:14.58 | DocHolliday | [TK]D-Fender, i just want to be able to go from my default context to the mainmenu context |
| 05:15.46 | [TK]D-Fender | DocHolliday : Thats all fine and dandy, but it helps when you've confirmed the target is valid... so go check where you should be going... |
| 05:16.03 | JT | k-man: my voice :) |
| 05:16.10 | JT | i'm not sure how authentic it is |
| 05:16.17 | DocHolliday | fixed |
| 05:16.22 | JT | but i can impersonate a few different people |
| 05:16.25 | k-man | oh, i see.. you are mimicing chopper? |
| 05:16.30 | JT | yes |
| 05:16.37 | k-man | ok... i'm with you now |
| 05:16.42 | k-man | i was a bit slow off the mark |
| 05:16.44 | JT | lame i know :P |
| 05:16.49 | orlock | heh, even william gibson based a character on him |
| 05:17.07 | k-man | jt, i seem to recall hearing that he lives in richmond or somewhere and that he is often seen at the local there |
| 05:17.18 | DocHolliday | [TK]D-Fender, the context where i was supposed to go was commented :P |
| 05:17.18 | k-man | you should just go there and ask him to record it for you |
| 05:18.27 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:18.28 | JT | he already made ringtones and sms alerts for the rohnnie johns half hour |
| 05:18.34 | JT | k-man: richmond in sydney? |
| 05:18.55 | k-man | jt, yeah, i think so |
| 05:19.19 | k-man | i can;t recall who told me that now... but they lived there and said they had seem him there numerous times |
| 05:19.30 | JT | ah ok |
| 05:21.49 | orlock | k-man: which local? |
| 05:21.57 | orlock | k-man: as far as i know he lives in collingwood |
| 05:22.18 | JT | is collingwood near richmond, vic? |
| 05:22.21 | orlock | JT lives in sydney anyway |
| 05:22.36 | DocHolliday | [TK]D-Fender, why is asterisk missing submenuopts and the default mainmenu sound file? |
| 05:22.39 | k-man | orlock, no idea, it is a vague memory - i could well be wrong, or he could have moved since then |
| 05:22.40 | orlock | JT: yeah, pretty close |
| 05:22.49 | JT | orlock: sounds likely then |
| 05:23.03 | [TK]D-Fender | DocHolliday : What do you mean "default mainmenu"? |
| 05:23.20 | JT | i know one of the dudes on the rohnnie johns half hour was from sydney, so either he moved or they film in both states |
| 05:23.31 | [TK]D-Fender | DocHolliday : and "missing submenopts"? |
| 05:24.13 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:24.33 | DocHolliday | [TK]D-Fender, in the default mainmenu context with asterisk 1.4 it references sound files that dont exist |
| 05:26.17 | JT | k-man: i was recording prompts for ChopTel and BoganTel |
| 05:26.17 | [TK]D-Fender | DocHolliday : Ok, apparently no one has told you this, so I'm going to make sure that any illusions you have left are cleanly shattered. The sample files are bloated piles of crap that are at BEST an example on HOW to implement things a few different ways (not necesarily SANE even). |
| 05:26.26 | JT | i can probably also make prompts for PoshTel |
| 05:26.37 | [TK]D-Fender | DocHolliday : They are meant to be studied (if even) and NEVER USED. |
| 05:26.57 | [TK]D-Fender | DocHolliday : Trash absolutely everything you did not do yourself and start from scratch. |
| 05:26.59 | DocHolliday | [TK]D-Fender, much appreciated sir. |
| 05:27.16 | [TK]D-Fender | DocHolliday : Just here to share the holiday cheer :) |
| 05:27.22 | DocHolliday | oh and [TK]D-Fender i ordered the cisco phones ;) |
| 05:27.33 | [TK]D-Fender | DocHolliday : Seriously.... its a psychotic mess.... |
| 05:27.49 | [TK]D-Fender | DocHolliday : At elast you are CONSISTANT in your mosochism :) |
| 05:27.57 | DocHolliday | sure, do you know of a good tutorial for dialpllans and menu systems? |
| 05:28.17 | [TK]D-Fender | DocHolliday : |
| 05:28.18 | [TK]D-Fender | ~book |
| 05:28.28 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:28.44 | [TK]D-Fender | DocHolliday : All there is to say about IVR's can be summed up with an understanding of *'s "Standard Extensions". |
| 05:28.54 | DocHolliday | [TK]D-Fender, thank you oh all powerful one |
| 05:29.13 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:29.56 | k-man | jt, sounds good |
| 05:30.01 | [TK]D-Fender | DocHolliday : set your timeouts. play some sounds. wait for input. pattern match the options that are valid. on timeout, "do somthing". on invalid option "do something else". There really isn't anything more to it. |
| 05:31.01 | DocHolliday | [TK]D-Fender, haha funny considering asterisk came loaded wtih warnings in the CLLI |
| 05:31.18 | JT | k-man: i did some normal prompts at first, but they were a bit boring |
| 05:31.28 | JT | also i probably need a quieter recording environment :P |
| 05:31.31 | [TK]D-Fender | DocHolliday : I think "affirming" of my previous claims would be a more accurate assessment... |
| 05:32.10 | k-man | jt, i wonder if you could get enough graps of say, john howard from TV, and make prompts like that |
| 05:32.16 | k-man | not that i would want john howard tel |
| 05:32.36 | DocHolliday | [TK]D-Fender, sure.. i spent a lot of time today making this machine ready for production asterisk |
| 05:33.07 | JT | k-man: that'd be interesting |
| 05:33.26 | [TK]D-Fender | DocHolliday : Its a lot easier to build something well that to try and fix something from being bad. |
| 05:34.17 | DocHolliday | [TK]D-Fender, sure.. well i take lots of care making sure the box isnt going to trip and die with angry customers |
| 05:35.31 | DocHolliday | can i safely get rid of every dundi context? |
| 05:35.58 | [TK]D-Fender | DocHolliday : Just take the time to evision all of the scenarios yur dialplan could encounter and ensure that you have an option in place to account for it. |
| 05:36.09 | [TK]D-Fender | DocHolliday : Are youplanning on using DUNDI? |
| 05:36.18 | Hmmhesays | well that was a pretty good movie |
| 05:36.18 | DocHolliday | nope |
| 05:36.21 | Hmmhesays | "stranger than fiction" |
| 05:36.47 | [TK]D-Fender | DocHolliday : Then keep these sort of things in mid following my statement that you should "trash everything and strart from scratch". |
| 05:37.42 | [TK]D-Fender | Hmmhesays : I'm thinking of buying a Carvin custom guitar..... going to set me back over 1K$USD..... |
| 05:40.27 | Hmmhesays | ouch |
| 05:40.37 | Hmmhesays | Carvin makes some nice gear though |
| 05:43.57 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:44.47 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 05:45.19 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 05:45.23 | [TK]D-Fender | Hmmhesays : Well it IS a ful custom job. Fretboard, frets, trem, body (its a 1-piece slim through neck), etc.... |
| 05:45.40 | [TK]D-Fender | Hmmhesays : Wilkinson trem & Sperzal locking tuners |
| 05:46.12 | flenders | question about DISA... I added this to my extensions.conf: |
| 05:46.12 | flenders | exten => 98,1,VMAuthenticate(05) |
| 05:46.13 | flenders | exten => 98,2,DISA(no-password|outgoing) |
| 05:46.26 | flenders | I get the dialtone, but when I start dialing, it hangs up |
| 05:46.47 | [TK]D-Fender | flenders : Check your target context. |
| 05:46.49 | JT | maybe it doesn't like what's in outgoing |
| 05:47.14 | flenders | outgoing is my context on extensions.conf, right? |
| 05:47.14 | DocHolliday | [TK]D-Fender, i get errors upon reload that reference extensions.ael |
| 05:47.34 | [TK]D-Fender | DocHolliday : empty that file out completely. |
| 05:47.51 | DocHolliday | what is the purpose of it? |
| 05:47.52 | JT | flenders: yes |
| 05:48.02 | flenders | weird |
| 05:48.05 | flenders | should work then |
| 05:49.12 | [TK]D-Fender | DocHolliday : Alternative extensions programming logic method. Looks like "real code" only nobody but luke-jr_ gives a rip about it ;) |
| 05:49.14 | JT | DocHolliday: different extensions.conf format |
| 05:49.42 | flenders | this is one of my entries on my extensions.conf: |
| 05:49.44 | flenders | exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 05:49.55 | DocHolliday | ahh consider that gone :P |
| 05:50.04 | flenders | so if I try an 8 digit number, it should work right? |
| 05:50.19 | jql | looks like 8 digits to me |
| 05:51.44 | JT | flenders: never used vmauthetnicate for that |
| 05:51.44 | DocHolliday | [TK]D-Fender, is there a way to prevent .ael from loading? (instead of just getting rid of the file)? |
| 05:51.44 | k-man | can anyone recommend a good wireless router with 4 port switch? |
| 05:51.44 | k-man | linksys are all sold out |
| 05:51.44 | k-man | i can't get one for like 2 weeks |
| 05:51.46 | [TK]D-Fender | flenders : I don't trust singular lines pasted like that. Pastebin everything related |
| 05:51.47 | [TK]D-Fender | ~pb |
| 05:51.58 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 05:52.16 | [TK]D-Fender | k-man : First you ask us what we'd advise and then tell us you won't accept our answer. Good start :) |
| 05:52.22 | flenders | JT: that's just to have the same password as voicemail |
| 05:52.32 | k-man | [TK]D-Fender, err... |
| 05:52.45 | k-man | [TK]D-Fender, i mean, in the absense of linksys, would you get any thing else? |
| 05:52.47 | JT | DocHolliday: mv extensions.ael extensions.ael.disabled |
| 05:54.13 | DocHolliday | JT and that will prevent the AEL load process from starting? |
| 05:54.15 | DocHolliday | i already moved it to extensions.ael.bak |
| 05:54.34 | JT | yes |
| 05:54.36 | JT | err |
| 05:54.49 | [TK]D-Fender | k-man : Some D-Link's are flaky with SIP, and I couldn't advise any others. |
| 05:54.55 | JT | it will stop the file from loading, you can put a noload directive in for the ael parser too |
| 05:55.07 | k-man | [TK]D-Fender, ok, i'll just have to wait the 2 weeks |
| 05:55.08 | JT | d-links are flaky with staying up for a little shile |
| 05:55.09 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:55.09 | k-man | thanks |
| 05:55.10 | JT | while |
| 05:55.23 | dlynes_laptop | ugs |
| 05:55.31 | DocHolliday | JT, *raises the cisco flag* |
| 05:56.00 | DocHolliday | JT, which file is the AEL parser in? |
| 05:56.13 | [TK]D-Fender | DocHolliday : EW. Cisco PIX = NAT Hell |
| 05:56.21 | JT | you can disable it in modules.conf |
| 05:56.27 | DocHolliday | [TK]D-Fender, i meant in terms of ATAs |
| 05:56.33 | JT | yeah pix are shit at NAT |
| 05:56.37 | DocHolliday | haha |
| 05:56.58 | JT | there's a little known router that's really good with nat |
| 05:57.04 | JT | it's called linux iptables |
| 05:57.17 | jql | JT: I have one of those |
| 05:57.36 | [TK]D-Fender | DocHolliday : Even their ATA's... overpriced..... just too much trouble |
| 05:57.40 | orlock | JT: heh, yeah.. been dealing with a company that uses a PIX.. they cant have different internal destinations for port forwards from the one external IP |
| 05:57.53 | JT | orlock: heh |
| 05:58.18 | jql | bleh. I loathe the PIX |
| 05:58.25 | orlock | so seperate internet IP's needed for every port forward to a different internal IP |
| 05:58.40 | JT | orlock: rofl! |
| 05:58.41 | jql | my company has one, so I can't test voip worth a damn behind it |
| 05:58.45 | orlock | yeah |
| 05:58.56 | orlock | the best soho modems i have found have been Draytek |
| 05:58.58 | JT | jql: good if you're testing nat punching solutions ;) |
| 05:59.00 | jql | the PIX magically "fixes" the sip packets, making debugging useless |
| 05:59.08 | JT | fixes? |
| 05:59.10 | jql | it's got sip-awareness turned on |
| 05:59.23 | orlock | JT: "fixes".. like what you do to dogs, you know? |
| 05:59.32 | jql | rewrites the whole packet, and then opens up the ports for it |
| 05:59.35 | JT | when they bark too much? |
| 05:59.48 | jql | neuters the packets. :) |
| 05:59.54 | JT | hmm |
| 05:59.54 | DocHolliday | jT, any idea where i can disable the parser? |
| 05:59.57 | jql | not spay... neuter |
| 06:00.02 | JT | DocHolliday: YES modules.conf |
| 06:00.12 | JT | jql: only when sip awareness is on? |
| 06:00.27 | jql | yes, only when the sip option is on |
| 06:01.12 | DocHolliday | JT, i dont see it in there.. :0 |
| 06:01.14 | jql | it does the fixup, and my asterisk server sees all the calls coming from the pix, rather than the internal IPs. can't test anything with that... |
| 06:01.24 | flenders | damn it. it works if I dial in through one of my sip providers, but it doesn't if I dial in on my Zap channels |
| 06:01.49 | JT | DocHolliday: you need to add it! |
| 06:02.00 | JT | noload blbhalbhalbhablah.so |
| 06:02.42 | JT | jql: ah yikes, is that actually supposed to be useful in certain scenarios? |
| 06:03.09 | jql | yes. It's useful when you have Cisco phones using SIP, because they're asymmetric. :) |
| 06:03.14 | jql | otherwise, no |
| 06:03.15 | jql | heh |
| 06:03.34 | JT | explain more? :) |
| 06:04.34 | jql | Since the PIX rewrites the packet, it can know both the incoming and outgoing SIP port, and keep them open within itself. Since the Cisco phone likes those ports to be different, a PIX firewall works where other non-sip-aware firewalls wouldn't |
| 06:04.56 | *** join/#asterisk joebob777as7 (n=richard@yoda.peacefulescape.com) |
| 06:06.01 | joebob777as7 | just heard about asterisk and skimmed through the manual... What desk phones work best with asterisk? We currently have a NEC Aspire system |
| 06:06.22 | JT | ah so it pretty much only works with cisco phones a CCM? |
| 06:06.39 | jql | well, the feature works with SIP, which CCM doesn't normally use |
| 06:06.51 | jql | so it's more generally useful for Cisco phones |
| 06:06.57 | JT | hmm ok |
| 06:07.07 | jql | it's the only firewall behind which Cisco phones can reliably be natted, AFAIK |
| 06:07.26 | [TK]D-Fender | joebob777as7 : Polycom. |
| 06:07.26 | jql | and by phones, I mean LOTS of phones |
| 06:07.27 | JT | connecting to what sip server? |
| 06:07.32 | jql | asterisk |
| 06:07.51 | JT | does asterisk support assymetric rtp? |
| 06:07.53 | jql | you can make one or two cisco phones work via port mapping, but a PIX lets you have as many as you want |
| 06:08.08 | jql | the cisco uses symmetric rtp, but not symmetric sip |
| 06:08.14 | jql | that's what's special. :) |
| 06:08.25 | JT | symmetric or assymetric? |
| 06:08.31 | DocHolliday | why would i get this error? NOTICE[4136]: codec_zap.c:364 find_transcoders: No Zaptel transcoder support |
| 06:08.40 | joebob777as7 | [TK]D-Fender only polycom phones? |
| 06:08.58 | jql | the Cisco will send SIP packets from, say, port 12345, but expect responses to port 5060 |
| 06:09.17 | sbingner | that's stupid |
| 06:09.22 | sbingner | lol |
| 06:09.24 | jql | heh |
| 06:09.28 | jql | oh, yeah |
| 06:09.37 | [TK]D-Fender | joebob777as7 : You asked for the best phones. there are a few other acceptable ones... |
| 06:09.44 | [TK]D-Fender | ~phones |
| 06:09.45 | jbot | it has been said that phones is at http://bani.anime.net/phones/, or is In order of quality: Polycom (Any), Aastra 480i, Cisco 7940+, Linksys SPA-94x |
| 06:09.45 | jql | stupid like a FOX |
| 06:09.47 | joebob777as7 | oh ok thanks |
| 06:09.55 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 06:10.10 | jql | The Aastra looks like a east european car from the soviet era, though. :( |
| 06:10.33 | JT | jql: wait... |
| 06:10.34 | JT | so |
| 06:10.41 | orlock | ql: Astra? |
| 06:10.53 | jql | yeah, the Aastra |
| 06:10.57 | orlock | Vauxhall/Opel/Holden Astra? |
| 06:11.04 | DocHolliday | jql, how about a Cisco 7985? only 5,000 |
| 06:11.08 | jql | heh |
| 06:11.24 | orlock | i cant wait till the new Torana comes out |
| 06:11.29 | jql | I considered asking for one |
| 06:11.36 | JT | Cisco phone SPort: 12345 DPrt: 5060 >>> SIP SERVER >>> SPort 5060 DPort 5060 >>> Cisco Phone |
| 06:11.39 | JT | they want that? |
| 06:11.50 | jql | but, really, I'd be stupid to use it without callmanager |
| 06:12.03 | jql | JT: yes |
| 06:12.14 | JT | that's F***ING STUPID |
| 06:12.16 | JT | i mean hi |
| 06:12.25 | jql | Cisco sets the Contact: me@foo:5060, and you're supposed to do the right thing |
| 06:12.36 | jql | unfortunately, the firewall disagrees |
| 06:12.41 | JT | hrm |
| 06:13.12 | jql | unless it's a PIX |
| 06:13.13 | sbingner | sip is F***ING stupid... it shouldn't ALLOW you to do that |
| 06:13.19 | jql | in which case, it agrees heartily |
| 06:13.21 | DocHolliday | anyway, night guys.. thanks for the help |
| 06:13.23 | DocHolliday | ha |
| 06:13.31 | JT | all voip protocols are stupid |
| 06:13.33 | JT | :) |
| 06:13.33 | jql | sbingner: In Cisco's opinion, it's the Right Thing(tm) to do |
| 06:13.41 | jql | they staunchly refuse to budge on it |
| 06:13.45 | sbingner | jql, doesnt linux's SIP masq module handle that too? |
| 06:13.51 | sbingner | I never looked |
| 06:13.54 | jql | sbingner: it might |
| 06:13.58 | JT | almost no other protocol does that |
| 06:14.12 | JT | most expect clients to use random high numbered ports and to respond to it |
| 06:14.16 | jql | no, certainly no protocol abouve port 1024 should be trying that |
| 06:14.29 | jql | below 1024, perhaps there would be a historical security reason for it |
| 06:14.34 | jql | but port 5060? wtf? |
| 06:15.18 | JT | i wasn't sure if you agreed with the cisco stance for a minute there ;) |
| 06:16.08 | jql | naw, it just excludes Cisco from the market for people without a PIX |
| 06:16.15 | Phel | Well if my asterisk server could talk with a SIP service provider, couldn't I point the client to my local asterisk server? |
| 06:16.26 | jql | of course, Cisco would rather lock you in with CallManager, so they probably don't give a damn |
| 06:16.34 | sbingner | jql, I think it's just because cisco doesn't want people to USE SIP |
| 06:16.36 | JT | jql: there seem to be a few people using cisco phones with asterisk? |
| 06:16.44 | sbingner | I bet their skinny works fine over non-pix |
| 06:17.01 | jql | I love my Cisco, and use it all the time with asterisk |
| 06:17.08 | jql | I just dread any customer asking for one |
| 06:17.20 | JT | surely most people use them without a PIX |
| 06:17.51 | jql | they would if they could. I'm pretty sure more than one cisco phone behind a sip-ignorant firewall isn't going to work |
| 06:17.54 | Phel | I know the router can at least do basic port forwarding |
| 06:17.57 | *** join/#asterisk jjhall (n=chatzill@67.60.61.7) |
| 06:18.19 | jql | one cisco phone is easy, more than one makes port forwarding a disaster |
| 06:18.23 | JT | jql: ah but what about on the lan? |
| 06:18.31 | jql | locally, it'd work great |
| 06:18.31 | JT | no firewall |
| 06:18.39 | JT | so it's only a nat issue |
| 06:18.43 | jql | asterisk handles the port business perfectly |
| 06:19.01 | jql | yeah, only nat |
| 06:19.14 | JT | okay, that's what i though |
| 06:19.15 | JT | t |
| 06:19.20 | sbingner | actually, only PAT |
| 06:19.23 | jjhall | I haven't updated in at least 6 months (SVN-branch-1.2-r), should I update to 1.2.15 or go to 1.4? |
| 06:20.02 | sbingner | jjhall, probably :) |
| 06:20.18 | jjhall | LOL Very helpful... ;-) |
| 06:21.07 | jjhall | Its for my home phone, but we primarily use our cells anyway so it isn't horribly critical. |
| 06:22.26 | jjhall | What would be the biggest ups and downs of moving to 1.4? |
| 06:22.57 | JT | possibly less stable |
| 06:24.13 | jjhall | Any advantages? |
| 06:24.35 | JT | some new features |
| 06:24.54 | [TK]D-Fender | ok, checkout time here... back tomorrow. |
| 06:26.34 | joebob777as7 | i'm a complete linux newb started using ubuntu about 1 month ago what are my chances of getting one of these up and running? how long do you think it would take me? |
| 06:27.04 | jjhall | Asterisk? |
| 06:27.18 | Phel | JT: You know, when I try making a call the dubugging output looks like it actually working |
| 06:27.23 | joebob777as7 | yes |
| 06:28.00 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:28.16 | jjhall | joebob777as7: Not long actually. There are some good walkthroughs out there. For a basic system you can be up and running in about a half hour. Do you already have a provider to use? |
| 06:29.06 | Phel | I receive an "INVITE", it waits a minute, then I get a "BYE" |
| 06:29.17 | jql | that's bad |
| 06:29.30 | JT | Phel: cool, what did you change? |
| 06:29.39 | Phel | Nothing |
| 06:29.49 | Phel | It still can't register |
| 06:29.54 | JT | hrn |
| 06:29.56 | JT | lame |
| 06:30.16 | Phel | I'm just saying it *looks* like a call is happening from the dubugging output |
| 06:30.23 | Phel | which is weird |
| 06:30.36 | JT | sure, but clearly the RTP isn't getting through |
| 06:31.02 | jql | RTP doesn't start until asterisk sends back a port to send it to |
| 06:31.04 | Phel | I give up |
| 06:31.08 | joebob777as7 | jjhall: well that's the thing I'm not sure what I'm going to need to get, and what my advantages are going to be... I have standard phone service through my local providor. it's all analog currently and we have 3 phone lines and the wiring looks daunting with our current phone system... it looks like a rat's nest to me lol |
| 06:31.13 | jql | if you only ever receive packets, and never send one, no rtp |
| 06:32.06 | Phel | I'm gonna sledge the router |
| 06:32.50 | Phel | I've tried "Static NAT" meaning forward all unsolicited packets to me, and still nothing |
| 06:33.19 | JT | port forward the rtp ports? |
| 06:34.17 | Phel | everything |
| 06:35.05 | jjhall | joebob777as7: Is this for a business or a home system? Business I'm assuming? |
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| 06:37.14 | *** join/#asterisk antlers (n=antlers@ip68-224-230-141.lv.lv.cox.net) |
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| 06:41.14 | Phel | JT: How about putting myself in the DMZ? |
| 06:42.42 | JT | shrug |
| 06:42.47 | JT | worth a shot i guess |
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| 06:47.56 | joebob777as7 | yeah in home business |
| 06:48.39 | joebob777as7 | we are in the process of buying a building though |
| 06:50.07 | jjhall | joebob777as7: How many phones attached to the 3 lines? |
| 06:50.34 | joebob777as7 | currently 3 standard base phones and 2 cordless |
| 06:51.45 | jjhall | Are you wanting to stick with those phones or upgrade to IP based phones? |
| 06:52.36 | joebob777as7 | i'm not sure... I don't know what the benifit would be... I'm not up to speed on this yet sorry for my extreme phone system newbness... |
| 06:53.17 | jjhall | No problem at all. I guess the first question to ask then is why are you here, as in what are you looking at asterisk to do for you? |
| 06:55.00 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 06:55.02 | joebob777as7 | I'm looking to design my own phone system that isn't some proprietary piece of crap that I actually have some control over... I'm sick of paying between $4000 and $6000 for another pile of junk and I want to do it all myself |
| 06:55.02 | zeeesh | hi |
| 06:55.29 | jql | amen |
| 06:55.39 | joebob777as7 | lol |
| 06:55.41 | jjhall | Ok, what about functionality, do you just want to be able to call between the phones, or are you looking for something more? |
| 06:55.46 | jql | howdy, zeeesh |
| 06:56.24 | JT | argh it's the attack of the j nicks |
| 06:56.25 | *** join/#asterisk Waverly360 (n=mirc@209.149.58.214) |
| 06:56.36 | jjhall | LOL |
| 06:56.40 | jql | jjjjj |
| 06:56.48 | jql | j <tab> |
| 06:56.56 | jql | the result is insanity |
| 06:57.15 | joebob777as7 | I want to be able to monitor calls, record them, call in to hear my email, conference, easy call forwarding, barge in funcionality and whisper |
| 06:57.37 | Waverly360 | Anyone who's played around with fastAGI before, I'm having a problem with asterisk passing single digits (like for menu selections) to my agi script with the proper newline characters on the end. Has anyone else had a problem with this? |
| 06:57.38 | jql | joebob777as7: all noble goals |
| 06:58.02 | joebob777as7 | thx |
| 06:58.03 | Waverly360 | jql: you're everywhere :P |
| 06:58.11 | jql | you cannot escape me! |
| 06:58.16 | Waverly360 | damn :P |
| 06:59.31 | Waverly360 | LMAO |
| 06:59.41 | Waverly360 | crap...he's onto me |
| 07:00.12 | *** join/#asterisk Phel (n=chatzill@adsl-153-185-236.mia.bellsouth.net) |
| 07:00.19 | jjhall | joebob777as7: All things you can do with asterisk. Before taking the plunge, I'd do some serious reading and decide if you want to stick with analog lines or move to digital, and whether you want to replace your phones or use a linecard or adapters. If it were me I would replace the phones so that you can have feature buttons and such, but it also depends on your budget. |
| 07:00.27 | Phel | JT: Answer: Update Firmware |
| 07:00.29 | jql | defeated! |
| 07:00.34 | JT | Phel: works? |
| 07:00.38 | Phel | yep |
| 07:00.44 | Waverly360 | jql: :P |
| 07:00.46 | JT | Phel: everything? |
| 07:00.50 | Phel | yep |
| 07:00.53 | JT | Phel: registrations, calls? |
| 07:00.57 | Phel | yep |
| 07:01.01 | JT | i know your router was a pile of junk :P |
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| 07:01.30 | joebob777as7 | jjhall: I'll definately replace the phones. what is going to be the major benifit of switching to digital? |
| 07:01.41 | jjhall | joebob777as7: For the lines? |
| 07:01.46 | JT | proper call progress signalling, joebob777as7 |
| 07:01.49 | Phel | Plus the router config GUI looks and works much better |
| 07:01.55 | Phel | yay! |
| 07:02.06 | JT | analogue has signalling that make the baby jesus (or computers) cry |
| 07:02.10 | jjhall | joebob777as7: JT couldn't have said it better. |
| 07:02.30 | joebob777as7 | what is that? lol |
| 07:02.44 | JT | detecting when calls are answered, hanged up |
| 07:02.47 | JT | etc etc |
| 07:02.53 | jjhall | The ability for the PBX to know when calls are in progress, when they ended, caller ID, etc. |
| 07:03.24 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:03.33 | joebob777as7 | ok so is that only with voip? what is the cost associated with switching usually? why wouldn't i switch? |
| 07:03.47 | JT | no digital |
| 07:03.59 | JT | voip implements digital over packet switched networks |
| 07:04.06 | JT | digital usually means isdn |
| 07:04.12 | jjhall | Expandability as well. You're fairly limited to 4 analog lines depending on how many free PCI slots you have. With digital or even VoIP your limits are far greater. |
| 07:04.36 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 07:04.39 | xo8ox | hey guys |
| 07:05.01 | joebob777as7 | so i'd have to get an isdn line and drop my phone lines? |
| 07:05.19 | JT | that would be the idea, however it may not be economical depending on the area |
| 07:05.27 | xo8ox | if I have a .wav file and wanna use it for menu.. how do I convert it so it can be placed in the sounds dir in asterisk to be used ? |
| 07:05.35 | JT | usually the minimum number of channels on PRI is 8 lines in the us |
| 07:05.36 | joebob777as7 | yeah i'll probably keep the analog then |
| 07:05.54 | JT | bri is unfavourably priced in most places in the US |
| 07:05.57 | JT | BRI is 2 channels |
| 07:06.05 | jql | very unfavorably |
| 07:06.14 | jql | but only when compared to residential service |
| 07:06.14 | JT | jql: depends on location though |
| 07:06.29 | jql | business lines are generally anti-competitive with BRI. :) |
| 07:06.34 | jjhall | joebob777as7: What kind of Internet connection will you have? T1 or DSL? |
| 07:06.45 | joebob777as7 | DSL with a static ip |
| 07:07.21 | jjhall | Ok. If you had a T1, you can sometimes break part of it off for voice channels. |
| 07:07.31 | joebob777as7 | the DSL is my fourth line that I don't currently have through our phone system. |
| 07:07.52 | joebob777as7 | jjhall: oh ok bummer... well t1 is about $400 a month here lol |
| 07:07.55 | xo8ox | so anybody :) |
| 07:08.04 | jjhall | joebob777as7: That isn't bad actually. |
| 07:08.24 | joebob777as7 | jjhall: is that what you have? |
| 07:08.34 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 07:08.48 | jql | $400 / 23 < $20/line/month |
| 07:08.56 | joelsolanki | Good Evening people |
| 07:09.04 | jql | not a bad deal |
| 07:09.13 | xo8ox | if I have a .wav file and wanna use it for menu.. how do I convert it so it can be placed in the sounds dir in asterisk to be used ? |
| 07:09.26 | jql | xo8ox: wav is supported if format_wav.so is loaded |
| 07:09.31 | joelsolanki | was thinking to implement sip 2 h.323 & h.323 to sip translator in Aserisk. Is this possible ? |
| 07:09.44 | jql | just put file.wav in the sounds dir, and use it like anything else |
| 07:09.47 | xo8ox | if its not |
| 07:09.48 | jjhall | Here at home I use VoIP, but at work we have a T1 for voice, a T1 for data to the Internet, and a fractional T1 that is split among several of our remote offices. |
| 07:09.52 | xo8ox | how do I convert to GSM ? |
| 07:10.15 | joelsolanki | Any hints plz for translation ? |
| 07:10.22 | jql | If audacity can't do it, then I might try sox, and after that I'm lost |
| 07:10.28 | joebob777as7 | jjhall: what would my advantages and disadvantages of VOIP be? |
| 07:10.34 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:10.41 | jjhall | At another location we have a T1 that runs full data speed, then dynamically allocates voice channels as needed. |
| 07:10.45 | Chris-NB | hi |
| 07:11.02 | jjhall | joebob777as7: Reliability. If you have a rock-solid DSL, and a good provider, you'll be fine. |
| 07:12.28 | Phel | JT: Many Thanks |
| 07:12.30 | Chris-NB | is it possible to hav a queue with 3 members. If a call comes in, first ring the 1, then the 2. then the 3. If none of them pick up the phone, call another nr? |
| 07:12.50 | jjhall | If you're using your DSL for both voice and data IP traffic (VoIP, not the analog line the DSL rides on) you'll want to get a router (or use a linux-based solution like Smoothwall) that can provice Quality of Service. |
| 07:12.59 | jql | Chris-NB: the ring pattern is possible, not sure of integration with the queue |
| 07:13.23 | joebob777as7 | ok i still don't understand voip but i think that's ok for now lol i think |
| 07:13.49 | Chris-NB | jql, exten => xxx,1,Dial(SIP/1);...2,Dial(SIP/2);...3,Dial(SIP/3) <- did you mean that? |
| 07:13.50 | jjhall | Hehehe. |
| 07:14.09 | xo8ox | in my modules.conf there is no format_wav.so ..! |
| 07:14.14 | jql | that would indeed call them in sequence |
| 07:14.23 | jjhall | There is a lot to consider when setting up a new system. Do you currently just have 3 lines and several single or multi-line phones or do you have some sort of PBX? |
| 07:14.37 | Chris-NB | jql, but is it possible to pack that into a queue? |
| 07:14.57 | jql | do you want them ringing simultaneously? |
| 07:15.09 | Chris-NB | jql, noop. in sequence |
| 07:15.34 | jql | well, put Local/s@sequence in the queue |
| 07:15.41 | joelsolanki | anybody has done h.323 sip translation in asterisk ? |
| 07:15.43 | jql | and setup the [sequence] context with the above pattern |
| 07:15.45 | joelsolanki | i heard asterisk can do it. |
| 07:15.55 | jql | Dial(),Dial(), ... |
| 07:16.12 | Chris-NB | jql, oh, that would be possible. your right. thanks! |
| 07:16.18 | jql | enjoy. :) |
| 07:16.25 | joebob777as7 | brb |
| 07:16.48 | joelsolanki | nobody ??? |
| 07:17.03 | jql | not I |
| 07:17.08 | joelsolanki | :) |
| 07:17.22 | jjhall | joebob777as7: If I'm not still here when you return, feel free to contact me hall(dot)jeremy(at)gmail(dot)com and I can chat with you more tomorrow. |
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| 07:20.04 | joebob777as7 | back |
| 07:20.29 | jjhall | Ok. I'm headed to bed in about 5 minutes (or whenever my compile gets done) |
| 07:20.49 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:21.18 | joebob777as7 | ok me too I'll email you tonight. but we currently are running 3 phone lines into our NEC apsire system and we have multifuncion phones |
| 07:23.10 | *** part/#asterisk kfhdk (n=kfh@1107ds1-ksa.0.fullrate.dk) |
| 07:23.14 | jjhall | What is your budget for the changeover? |
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| 07:23.54 | joebob777as7 | $1000-$2000 i'd say |
| 07:24.18 | jjhall | Do you already have a computer of sufficient power to run Asterisk or do you need to buy one? |
| 07:24.38 | joebob777as7 | I own a computer business I have plenty laying around |
| 07:25.51 | jjhall | Ok. Are your cordless phones standard phones or are they part of your PBX? |
| 07:26.34 | joebob777as7 | part of the PBX |
| 07:28.54 | jjhall | Ok. Well here is what I would probably do in your situation. Buy one IP phone or an adapter for a regular phone. Set up Asterisk on an extra box, and get setup with a business-class VoIP provider and just give it a try. You can base your decision on whether to change to VoIP based on how that works in your testing. You would only be out $60-$300 depending on what you buy, and won't be... |
| 07:28.55 | jjhall | ...commited to something you won't be happy with. |
| 07:29.43 | jjhall | Anyway, time to assume the horizontal position. I'll check my e-mail tomorrow, and feel free to add me on Google Talk if you use it. Have a good evening! |
| 07:30.36 | joebob777as7 | thanks night |
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| 07:41.36 | xo8ox | guys I placed the wav file in the sounds folder and it doesn't work |
| 07:42.18 | xo8ox | when I dial the voice menu I hear distorted noice |
| 07:45.49 | jql | might be the wrong bitrate or stereo or something |
| 07:46.14 | xo8ox | what format or file conversion does it need to be ? |
| 07:46.24 | xo8ox | I have it in PCM 8000 8bit mono |
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| 07:47.05 | jql | did you ulaw it and name it .ulaw? |
| 07:47.07 | Chris-NB | jql, if you are interested: When you call a queue like that: queue(queuename,n) then every phone in that queue will ring 1 time, then the queue is exited and the next action(s) take place. If you set the ring strategy to roundrobin, so every phone rings in sequence and .... call another nr. after the timeout |
| 07:47.36 | Chris-NB | jql, ring 1 time is wrong, should mean ring |
| 07:47.38 | Chris-NB | : D |
| 07:48.26 | xo8ox | its a .wav file |
| 07:48.39 | xo8ox | the sound folder in asterisk has only .gsm files in it |
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| 08:45.45 | SheriF_SpacE | hmm i have a proble, ... 2 grandstreem phones. one using gsm and one usong PCMU " ulaw as i understan " both can't hear each others " both in the same nat with the server " .. it's codec issue cuz when i change both to pcmu both works .... what is wrong ? aren't asterisk 1.4 should do codec translation ? |
| 08:48.02 | dj-fu | make them both use ulaw? |
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| 09:04.51 | vlt | Hello. How do I activate monitoring on iax2 channels? I tried "qualify=yes" in iax.conf's [general] section (like in sip.conf) but hat seems to be wrong ... |
| 09:06.03 | *** join/#asterisk thekidrio (n=dbzkai@24-205-76-13.dhcp.psdn.ca.charter.com) |
| 09:06.28 | thekidrio | anyone else get this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 09:12.34 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 09:12.38 | tutt9876 | hi |
| 09:12.51 | tutt9876 | get a problem to mach a sip addre in extensions.conf |
| 09:12.57 | tutt9876 | sip address |
| 09:13.35 | tutt9876 | I have tried 's', 'i' extension but didn't work |
| 09:13.41 | tutt9876 | Someone to help? |
| 09:14.05 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:14.48 | tutt9876 | how can I match 'adreddwithonlyletters@domain.com' in extensions.conf? |
| 09:17.14 | sbingner | are you trying to match an extension or a peer? |
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| 09:17.29 | tutt9876 | no a peer |
| 09:17.35 | sbingner | then configure it in sip.conf |
| 09:18.08 | tutt9876 | but i would like to match any sip address like onlyletters@sipdomain.com |
| 09:18.35 | sbingner | I think you may need to explain what you mean better |
| 09:19.09 | tutt9876 | a peer is logged on sterisk proxy (message ready in xlite) and.. |
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| 09:19.53 | tutt9876 | then the peer want to join a sip adress: onlyletters@asipdomain.com |
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| 09:21.05 | tutt9876 | but when dialing onlyletters@saipdomain.com there is no pattern entry for onlyletters@asipdomain.com and the dial failed |
| 09:22.12 | *** part/#asterisk sklim (n=shankuo@60.49.242.225) |
| 09:22.35 | tutt9876 | because s extensions didn't match onlyletters@asipdomain.com in extensions.conf |
| 09:23.27 | tutt9876 | I don't know wich pattern to give in extensions.conf to make the rigth match |
| 09:25.02 | Ahrimanes | _[a-z].@[a-z]. |
| 09:25.57 | tutt9876 | ok thanks i will try this |
| 09:31.13 | tzafrir | is that case-sensitive? |
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| 09:33.46 | jm|work | regex is |
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| 09:35.40 | webman | is it possible to have a guest IAX2 account (no secret,context=noauth) and a IAX2 account (with a secret,context=allfeatures) and have the call routed to the correct context? |
| 09:36.04 | webman | BTW, the wiki seems to say no, and this is the behaviour I seem to see, but I |
| 09:36.30 | webman | BTW, the wiki seems to say no, and this is the behaviour I seem to see, but I must be wrong, ..... well, I think I am :( |
| 09:36.42 | Ahrimanes | tzafrir: yes lower only |
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| 09:45.07 | Ahrimanes | i guess _[a-zA-Z.@]. would be used? |
| 09:45.12 | Ahrimanes | could |
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| 09:58.19 | bobbytux | lo |
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| 10:17.20 | mkl1525 | Hi, I've installed asterisk-bristuff on a debian etch system. * works (I can call other phones etc) problem is I can't hear any messages not my own nor * internal like voicemail etc. sound works on the system I can play mp3 with mpg123 from cli and hear the output - any suggestions what could go wrong? |
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| 10:22.57 | tzafrir | mkl1525, zaptel timing problem? |
| 10:23.18 | tzafrir | mkl1525, what do you get from running 'zttest -v'? |
| 10:24.31 | tzafrir | mkl1525, do you have zaptel built? |
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| 10:40.25 | ozant | hi, will Agent channel deprecated ? i am trying to use it on 1.4.It says AgentCallbackLogin deprecated and dialplan versions seems to use Local channel instead of Agent channel |
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| 10:41.46 | joelsolanki | Hi morning all |
| 10:42.08 | joelsolanki | i have sucessfully configured h323 to sip translation in asterisk. |
| 10:42.34 | joelsolanki | cisco ata --> asterisk (h323 - sip ) --> gateway |
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| 10:43.25 | joelsolanki | cisco ata i have already configured 0x00140014 in audio mode which means VAD ( silence suppression is OFF ) |
| 10:43.48 | joelsolanki | but still i see Feb 26 16:11:35 NOTICE[10513]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 10:43.57 | joelsolanki | what could be the issue ? |
| 10:44.57 | joelsolanki | any hints plz |
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| 10:53.18 | WasntMe | hi all |
| 10:54.08 | WasntMe | can someone help me with bri isdn hcf card installation on asterisknow beta 4 the question is if this beta support bristuff |
| 10:55.08 | JT | nah just don't use asterisknow |
| 10:55.25 | WasntMe | nice |
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| 10:56.53 | mkl1525 | tzafrir, thanks for your help "zttest -v" hangs after "Opened pseudo zap interface, measuring accuracy..." zaptel module was build using m-a a-i and ztdummy, zaptel and rtc is loaded |
| 10:57.08 | tzafrir | right, you're missing a zaptel timing source |
| 10:57.19 | tzafrir | ztdummy is loaded? |
| 10:57.28 | tzafrir | ls -l /dev/zap/pseudo |
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| 11:16.00 | vlt | Hello. Is there a " (${CALLERID} == 00) ? 0 : ${CALLERID} " syntax in dialplans? Do you know what I mean? |
| 11:17.34 | jql | ?: is valie |
| 11:17.36 | jql | valid |
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| 11:20.07 | Daejeo1 | guys plz help. how can I configure php/apache with flite (TTS engine)? |
| 11:20.42 | vlt | jql: hmmm, What's wrong here then? exten => _44440XX.,n,Set(CALLERID(number)=555(${CALLERID} == 00 ? 0 : ${CALLERID})) |
| 11:20.48 | jql | oh, example? |
| 11:20.49 | jql | heh |
| 11:21.08 | jql | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if |
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| 11:22.08 | jql | IF($[ ${CALLERID(num)} = "00" ] ? "0" : ${CALLERID(num)}) |
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| 11:27.38 | puzzled | Daejeo1: iirc search on the nerdvittles website or use google |
| 11:28.14 | puzzled | vlt: use "num" instead of "number" |
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| 11:32.29 | Daejeo1 | puzzled: i tried but could not find |
| 11:33.56 | kippi | what is the best way to set a cid when it is matched to a number? |
| 11:34.03 | puzzled | Daejeo1: wonder what you tried if at all. Second link, not that hard to spot... http://www.google.nl/search?q=nerdvittles+flite&start=0&ie=utf-8&oe=utf-8&client=firefox-a&rls=org.mozilla:en-US:official |
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| 11:37.53 | JT | kippi: |
| 11:37.54 | JT | ? |
| 11:38.53 | kippi | JT: when someone on there mobile rings you, how can you get it to flag there name up on your screen by changing the CID, I would have a big list off them |
| 11:40.49 | JT | something like exten => _X./123456,1,Set(${Callerid(name)}=FRIEND) |
| 11:43.50 | kippi | got a massive list of them, can i load them from a database or somthing? |
| 11:45.01 | JT | hrm maybe, there might be a couple of mechanisms for it |
| 11:45.07 | JT | or you might have to make your own |
| 11:46.51 | Chris-NB | is it possible to say a digium 1 port E1/T1 to get the clock from the pbx? |
| 11:47.03 | Chris-NB | so it recieves the clock |
| 11:48.43 | JT | the pbx is not asterisk? |
| 11:50.40 | mkl1525 | tzafrir, thanks again, after rebooting and doing a ztcfg -vvv it works :) |
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| 11:51.29 | tzafrir | mkl1525, do you have a zaptel hardware? or do you use ztdummy? |
| 11:51.59 | mkl1525 | tzafrir, I've got a hfc-s isdn card |
| 11:52.22 | tzafrir | I just wonder: do you use zaphfc or vzaphfc? |
| 11:54.04 | vlt | puzzled: Why do these variable names like CALLERID(number) change every now and then? |
| 11:54.32 | puzzled | vlt: afaik it was always num |
| 11:55.02 | vlt | puzzled: hmmm ... ok. |
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| 11:55.55 | Chris-NB | JT, correct |
| 11:56.10 | JT | Chris-NB: 1,1,0 |
| 11:56.14 | JT | span definition |
| 11:56.14 | Chris-NB | JT, what I need, is this in /etc/zaptel.conf .... ok. thanks |
| 11:56.17 | JT | assuming span 1 |
| 11:56.28 | Chris-NB | JT, jep. thats what I meant |
| 11:56.44 | Chris-NB | another question. How do I get a BUSY status on sip phones? |
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| 11:58.22 | Chris-NB | normally there is a call limit from 5. when I set this to 1 I get an Error, Call to user '130' rejected due to usage limit of 1 and the channel is CHANUNAVAIL, not BUSY |
| 11:58.29 | Chris-NB | how do I get a busy status? |
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| 12:09.21 | mkl1525 | tzafrir, zaphfc afaik - it's all precompiled in the asterisk-bristuff package of debian |
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| 12:15.13 | soo-hick | Hello |
| 12:15.13 | soo-hick | please any body can help here? |
| 12:15.13 | dj-fu | It's usually better to ask a question |
| 12:15.13 | soo-hick | ok |
| 12:15.13 | dj-fu | than ask if anyone can help |
| 12:15.13 | soo-hick | i have asterisk setup with h232 registered to GNUGK |
| 12:15.13 | dj-fu | mm? |
| 12:15.13 | soo-hick | abd installed zaptel on astresik |
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| 12:15.14 | soo-hick | the call comes in from the GNUGK to asterisk, asterisk push is it to the zaptel, terminating it to the isdn channels |
| 12:15.14 | soo-hick | now the problem i'm having is there is alot of calls been droped |
| 12:16.14 | soo-hick | and that because the carrier i'm using is not terminating these calls for me |
| 12:16.14 | soo-hick | is ther any way i can connect these calls with a small duration to improve my ASR? |
| 12:16.14 | soo-hick | on asterisk i mean? |
| 12:17.41 | dj-fu | sorry, out of my knowledge range |
| 12:17.41 | dj-fu | perhaps someone will read your message and enlighten you |
| 12:17.41 | soo-hick | thank you dj-fu |
| 12:17.41 | soo-hick | i'll be here any way |
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| 12:19.11 | HeinrichSA | Hi guys |
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| 12:19.38 | HeinrichSA | anyone have any XP using PRI and Q.931 |
| 12:19.38 | HeinrichSA | ? |
| 12:20.21 | HeinrichSA | I need to use some of the information elements in the dial plan... |
| 12:20.25 | HeinrichSA | Is there an easy way to do this? |
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| 12:22.29 | kieran491 | would any one happen to know if there is a problem with the freeworld IAX2 server'(s) |
| 12:23.21 | vlt | Hello. Yesterday I installed a 4xBRI card in TE mode (as client) using zaptel drivers. When dialling a number I can hear 0.3-0.5 seconds of another running call on this ISDN bus. What could be set wrong here? |
| 12:23.52 | vlt | Then connection is established correctly. |
| 12:26.07 | JT | did you get ptmp/ptp mode right? |
| 12:28.02 | kieran491 | when reciving an error like `chan_iax2:7344 socket_process: registeration of '838764' rejected: 'Registration Refused' from: '192.246.69.186'` dose that mean its my fault or is it freeworld fault or could be both? |
| 12:29.16 | vlt | JT: Hello again. I'm using it in ptmp mode. Didn't try ptp yet. |
| 12:29.48 | mkl1525 | Is there anything like NoOp($GLOBAL) that shows all variables that are available in the dialplan at this moment (trying to find if I can get the queue agent of a call that finished)? |
| 12:29.48 | vlt | JT: PS: Thanks again for your help getting it working. |
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| 12:31.10 | kieran491 | any one? |
| 12:31.10 | alexandrekeller | hi everyone |
| 12:31.12 | alexandrekeller | <PROTECTED> |
| 12:31.18 | JT | vlt: check with the provider already |
| 12:31.50 | alexandrekeller | sorry ?! |
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| 12:44.53 | kieran491 | no ideas any one relating to my questions? |
| 12:45.26 | Kigh | kieran491: could be both sides fault |
| 12:45.45 | tzafrir | mkl1525, I believe not |
| 12:46.01 | mkl1525 | tzafrir, ok thanks |
| 12:46.11 | Kigh | either you entered the wrong auth-parameters or they didnt set up your account correctly. but in first term: search for a mistake on _your_ side :-) |
| 12:47.00 | kieran491 | been spend a fair amount of time searching.. |
| 12:47.01 | Kigh | kieran491: setting up FWD is a bit tricky, there is a HOWTO you should read on voip-info.org |
| 12:47.04 | kieran491 | the logs dont say much |
| 12:47.18 | kieran491 | i am reading the doc on the asterisk site |
| 12:47.19 | Kigh | check out the howto, if not already |
| 12:47.25 | Kigh | read voip-info.org. |
| 12:47.30 | Kigh | the asterisk docs suck |
| 12:47.37 | kieran491 | oreily? |
| 12:47.51 | Kigh | well you want to use FWD, right? |
| 12:48.03 | kieran491 | yeah |
| 12:48.04 | Kigh | then you are wrong with the asterisk doc |
| 12:48.05 | JT | i didn't find fwd very hard at all |
| 12:48.12 | JT | there's docs on the fwd site |
| 12:48.15 | Kigh | read the howto on voip-info or fwd websitee |
| 12:48.46 | kieran491 | i have tead the fwd site |
| 12:48.53 | Kigh | JT but FWD is not set up like the others ive seen |
| 12:49.18 | JT | kieran491: did you enable IAX on fwd? |
| 12:49.37 | kieran491 | ohh? |
| 12:50.18 | JT | you must log in and enable it |
| 12:50.22 | JT | and wait a little bit |
| 12:50.24 | JT | or it will not work |
| 12:50.34 | JT | it clearly states this in the documentation on their site |
| 12:50.39 | JT | it defaults to sip |
| 12:50.46 | kieran491 | OHHHH |
| 12:50.53 | JT | as they can't accept registrations on the one account for sip and iax2 |
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| 13:12.24 | soo-hick | i have asterisk with zaptel installed on h323 registered to GNUGK |
| 13:12.55 | soo-hick | calls come in from GK to asterisk to the ISDN channels |
| 13:13.14 | soo-hick | some of these call can't be terminated by the carrier |
| 13:13.43 | soo-hick | i want to make these call have a small duration to improve my ASR, any ideas? |
| 13:13.46 | knathraak | Hi, anyone have {howto,sample configs,advice} for making asterisk with 2 TE110Ps talk gr303, emulating 5ess? |
| 13:20.16 | SheriF_SpacE | hmm i have a proble, ... 2 grandstreem phones. one using gsm and one usong PCMU " ulaw as i understan " both can't hear each others " both in the same nat with the server " .. it's codec issue cuz when i change both to pcmu both works .... what is wrong ? aren't asterisk 1.4 should do codec translation ? |
| 13:21.41 | ManxPower | SheriF_SpacE: how are you forcing the codec for each phone? sip.conf? |
| 13:22.08 | ManxPower | I didn't know GS phones even supported GSM |
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| 13:44.30 | SheriF_SpacE | ManxPower: nop in the phone it self. |
| 13:45.10 | ManxPower | SheriF_SpacE: do it in sip.conf. leave all codecs enabled on the phone |
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| 13:47.11 | coppice | I don't want to support codecs. I want codecs to support me. |
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| 14:00.54 | grinsbalu | lo |
| 14:00.54 | *** join/#asterisk omyz (n=sweetsug@mbl-65-158-104.dsl.net.pk) |
| 14:00.54 | omyz | hello people |
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| 14:09.01 | tparcina | omyz: hello omyz! |
| 14:09.09 | soo-hick | how to set ivr on asterisk, any one knows? |
| 14:09.45 | omyz | tparcina..thanks for reply...i thought everyone was away :) |
| 14:09.51 | tparcina | soo-hick: do you need ivr or aa (auto atendant)? |
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| 14:10.17 | tparcina | omyz: you are wellcome |
| 14:11.54 | omyz | thanks. im new to asterisk (and irc). i wanted to try asterisk. i have downloaded the cd image and am ready to install. my question is regarding the wiring the house. |
| 14:11.59 | *** join/#asterisk soo-hick (n=sinan@ip-81-1-98-55.cust.homechoice.net) |
| 14:12.04 | soo-hick | hello |
| 14:12.13 | soo-hick | i have these unconnected calls |
| 14:12.28 | soo-hick | and i want to connect them for a few seconds |
| 14:12.46 | vlt | Hello. How can I prevent callers waiting in a queue hearing the first part of the moh file over and over again? A "workaround" is "random=yes" but continously playing would be much better. How to do this? |
| 14:13.03 | soo-hick | i thought by putting ivr or aa in between the calls that comes form the gk to asterisk will solve that problem, no? |
| 14:13.47 | omyz | whats the best (cheapest) way to wire my house. i mean should i be thinking of analogue phones or sip phone or may be something better? |
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| 14:15.10 | vlt | omyz: What do you want to do with the phones ;-) ? |
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| 14:16.33 | soo-hick | any ideas on ivr or aa |
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| 14:17.16 | omyz | :) i mean extensions. i want to connect all my rooms to asterisk....so we can call each other :) |
| 14:18.52 | pigpen | Could someone tell me what the _% extension is? I am running realtime asterisk and I am getting the following query in postgres debug: |
| 14:18.53 | pigpen | SELECT * FROM extensions WHERE exten LIKE '\_%' AND context = 'office-open' AND priority = '-1' ORDER BY exten |
| 14:19.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:19.39 | sandorp | pigpen: _% looks like a wildcard query for anything that starts with an _ |
| 14:19.45 | *** part/#asterisk HeinrichSA (n=hh@196.25.68.131) |
| 14:20.10 | pigpen | I get this when I finish a call to a successful call. |
| 14:20.14 | soo-hick | anyone in here knows how set up ivr or aa |
| 14:20.15 | soo-hick | ? |
| 14:20.30 | pigpen | soo-hick, check the wiki |
| 14:20.38 | soo-hick | ok |
| 14:20.48 | pigpen | also, google is your friend. |
| 14:21.39 | tzanger | I'm having a major brain fart here -- what do you call the cat5 termination "strip" that is usually located in offices? |
| 14:21.44 | mindframe | is there a way to get a package to install without a dependency (i.e. need mozplugger without iceweasel) |
| 14:21.49 | tzanger | i.e. all the office data jacks go to the back room to one of these "strips" |
| 14:21.54 | pigpen | tzanger, patch panel? |
| 14:21.58 | tzanger | the phone jacks usually go to BIX strips |
| 14:21.59 | tzanger | yes that's it |
| 14:22.01 | tzanger | thank you |
| 14:22.04 | pigpen | np. |
| 14:22.05 | tzanger | I could NOT think of htat word |
| 14:22.16 | pigpen | coffee. |
| 14:22.23 | omyz | :) |
| 14:22.47 | sandorp | fileframe: tried rpm --nodeps file.rpm ? |
| 14:23.30 | omyz | any suggestions about cheapest extension phones that are suitable for asterisk. i need to connect 13 rooms to each other in my home |
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| 14:23.47 | mitcheloc | use a t1 |
| 14:23.57 | pigpen | I am using used polycom 500/501's |
| 14:24.19 | mitcheloc | t1 is cheaper |
| 14:24.58 | pigpen | 13 rooms? |
| 14:25.10 | omyz | t1...is it name of any phone set? |
| 14:25.11 | sandorp | does that include closets? |
| 14:25.14 | sandorp | :) |
| 14:25.19 | mitcheloc | expensive house, cheap phones? |
| 14:25.26 | omyz | he he he...no no closets |
| 14:25.26 | pigpen | I have 12 including 3 bathrooms....and one closet. |
| 14:25.50 | pigpen | dam..since you have that much money, get the Polycom 650's. |
| 14:26.00 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:26.08 | [TK]D-Fender | T1 != Cheaper. 7 x SPA-2002 = cheaper :) |
| 14:26.34 | omyz | well... i wanna start small... i really interested in asterisk and father is gonna pay for those. i dont wanna give him heart attack from begining |
| 14:26.44 | mitcheloc | T1 is cheaper when you can swipe an adtran from a client :) |
| 14:27.16 | [TK]D-Fender | omyz: First you need to consider your current (and/or planned) wiring. |
| 14:27.16 | pigpen | [TK]D-Fender, why would asterisk be polling my database with a query for _% ? |
| 14:27.39 | [TK]D-Fender | mitcheloc: No.... 7 SPA's still undercut the T1 card you'd need ;) |
| 14:27.52 | omyz | okay.... currently we have no wiring |
| 14:27.57 | [TK]D-Fender | pigpen: no clue |
| 14:28.04 | pigpen | thanks...be either. |
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| 14:28.20 | pigpen | well, it is working...but bitching... |
| 14:28.52 | [TK]D-Fender | omyz: Then run Cat5E to each room, terminating on RJ45. Then have them all come back to a central point and have them terminate on a patch panel. From there you can choose what kind of setup you're going to want. |
| 14:29.22 | [TK]D-Fender | omyz: You have a lot of learning to do as to the kinds of equipment you can use with * and to find out what best suits your budget & needs |
| 14:29.33 | sandorp | I suspect _% matches builtin extensions of some kind, since using _ at the beginning of a name is a common way of defining "private" values in a program |
| 14:29.34 | [TK]D-Fender | omyz: Start by downloading and reading THE BOOK |
| 14:29.38 | [TK]D-Fender | ~book |
| 14:29.40 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:29.58 | [TK]D-Fender | ~telephony101 |
| 14:30.07 | [TK]D-Fender | ~telecom101 |
| 14:30.17 | [TK]D-Fender | ~strom_c |
| 14:30.19 | jbot | extra, extra, read all about it, strom_c is just some nub |
| 14:30.22 | omyz | [TK]D-Fender : alrite......im ready to learn whatever i need to learn to get asterisk working |
| 14:30.22 | [TK]D-Fender | lol |
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| 14:31.00 | [TK]D-Fender | omyz: More important at the start is understanding the kinds of hardware you can use, and where each kind is most useful. |
| 14:31.21 | sandorp | I have 4 analog voice lines |
| 14:31.21 | sandorp | It looks like I will need to get a Digium card for those to work |
| 14:31.22 | sandorp | Do I need a VOIP provider to enable remote employees to connect to asterisk to make/receive calls via the analog lines? |
| 14:31.22 | sandorp | I was hoping to use software-based "phones" on people's home PCs + Asterisk to provide a "central office" phone |
| 14:31.24 | pigpen | sandorp, hmm.... yeah..that is what I was thinking. |
| 14:31.26 | omyz | ahan okay...i may have that book already..if not i will download it again |
| 14:31.36 | pigpen | but I only seem to get it upon hangup. |
| 14:31.44 | pigpen | maybe I'll just define it and move along. |
| 14:31.50 | pigpen | to like "hangup" |
| 14:32.28 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:33.07 | sandorp | can asterisk do what I'm trying to accomplish and do I need a VOIP account from my phone company to make it work? |
| 14:33.43 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:35.14 | [TK]D-Fender | sandorp: No. |
| 14:35.31 | [TK]D-Fender | sandorp: To your "do I need a VoIP provider to have remote users) |
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| 14:36.19 | sandorp | D-Fender: thanks |
| 14:36.22 | [TK]D-Fender | sandorp: Get a TDM card of some sort to give * access to your lines, and your remote users will connect directly over your internet connection. Please take your bandwidth into consideration. |
| 14:37.45 | sandorp | is 768k cable fast enough for 4-5 people talking to each via the net connection? |
| 14:38.04 | sandorp | or, how does one calculate the necessary bandwidth ? |
| 14:38.11 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 14:38.45 | pigpen | http://www.google.com/search?client=safari&rls=en&q=asterisk+bandwidth+calculator&ie=UTF-8&oe=UTF-8 |
| 14:38.51 | pigpen | ^^^first hit. |
| 14:39.12 | sandorp | pigpen, thanks |
| 14:39.13 | [TK]D-Fender | sandorp: if thats your upstream, you could do OK, but I'd highly recommend whatever they use run either G.729 or GSM codecs (far lighter than most). |
| 14:39.44 | *** join/#asterisk eald (n=eald@189.157.105.134) |
| 14:40.27 | sandorp | are then any software-only phones that someone could recommend? I'm trying to keep costs *down* |
| 14:40.35 | *** join/#asterisk danp (i=danp@204.118.103.42) |
| 14:40.55 | pigpen | idefisk |
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| 14:41.38 | [TK]D-Fender | sandorp: idefisk ( www.asteriskguru.com ) is a decent choice, followed by X-Lite ( www.counterpath.com ) |
| 14:41.39 | danp | i just had a sangoma card freak out...the wanpipe driver kept logging things like this: http://pastie.caboo.se/43088 |
| 14:42.36 | sandorp | pigpen and [TK]D-Fender: thanks I will look those up |
| 14:43.01 | PupenoR | Can "permit" on manager.conf have a hostname instead o an IP? |
| 14:45.26 | pigpen | PupenoR, to my knowledge it is ip's or ip blocks. |
| 14:45.58 | pigpen | but if using a fqn, then you probably would be opening it up to the internet...which is a bad idea. |
| 14:46.09 | pigpen | if the remote is out there somewhere, use a vpn. |
| 14:46.19 | pigpen | then permit from that ip block. |
| 14:48.25 | PupenoR | pigpen: the remotes are in the LAN as the DNS resolving them. Since they are dynamic IPs I try to use names. I'll have to use static IPs for those connecting to the manager. |
| 14:48.32 | edgecase | heh ok you have to send some obscure dbus command to make bluez "discoverable", before the Motorola HS-850 can connect to chan_bluetooth. now, to actually try making a call to it... |
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| 14:49.21 | pigpen | PupenoR, yeah...then a manual dhcp lease is your best bet. |
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| 14:50.33 | sandorp | what's the difference between an FXS module and and FXO module? the website makes it sound like you plug analog lines into both; I'm trying to figure out which TDM card and modules I will need if I have 4 analog voice lines |
| 14:50.35 | L|NUX | ~docs |
| 14:50.39 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 14:51.04 | [TK]D-Fender | sandorp: FXS module is for connecting PHONES. FXO is for connecting LINES. |
| 14:51.37 | ctaloi | hey all - anyone have any experience using Mediatrix 3000 gateways? |
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| 14:52.19 | sandorp | [TK]D-Fender: thanks (again) :) |
| 14:52.27 | pigpen | sandorp, Remember, FXS (phone Set) , FXO (central Office) |
| 14:52.39 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:52.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:52.45 | [TK]D-Fender | ctaloi: Oh yuck..... Meditrix Digital gateways are PITA Cisco wanna-be's. |
| 14:53.08 | sandorp | pigpen: gotcha |
| 14:53.52 | *** part/#asterisk soo-hick (n=sinan@ip-81-1-98-55.cust.homechoice.net) |
| 14:53.56 | ctaloi | ha - yeah - but i'm trying to get a specific call flow to work reliably (Mediatrix ATA (T38) --> Mediatrix GW --> ISDN and doing it with Cisco gear isn't cost effective |
| 14:54.27 | Schreiber1337 | Anyone here experienced on configuring a TDM400P on Debian based system? |
| 14:54.38 | Schreiber1337 | I keep getting Error: missing /dev/zap! |
| 14:54.58 | pigpen | sounds like you are missing some modules |
| 14:55.11 | pigpen | is /dev/zap even created? |
| 14:55.23 | pigpen | does lsmod show the necessary modules? |
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| 14:55.33 | pigpen | does lspci show any cards? |
| 14:57.12 | sandorp | so, I would use FXS modules if I wanted to plug regular phones into an asterisk PC? the FXO presumably handles the actual management of calls and such |
| 15:01.25 | pigpen | correct. |
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| 15:01.38 | [TK]D-Fender | sandorp: I personally suggest AWAY from using PCS cars for FXS usage. FXO modules are used to plug your analog home LINES into. |
| 15:01.55 | Schreiber1337 | Sorry, my connection got wacked. |
| 15:02.08 | [TK]D-Fender | sandorp: Keep in mind that a call can be from 1 FXS to another, or from an FXO, or in 1 FXO and out another. |
| 15:06.23 | vlt | Does anyone know how to prevent callers waiting in a queue hearing the first part of an moh file over and over again? |
| 15:08.02 | vlt | Do I have to create a separate sound stream on the system running all day? If that's the only way how can I plug that into moh? |
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| 15:11.39 | EmleyMoor | How do I use the first character of a string for comparison? How do I specify it? |
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| 15:13.07 | pigpen | EmleyMoor, you may want to elaborate. |
| 15:13.25 | [TK]D-Fender | EmleyMoor: ${thevariable:0:1} |
| 15:13.33 | EmleyMoor | Ah, thanks |
| 15:13.42 | [TK]D-Fender | EmleyMoor: Go lookup "asterisk variables" on the WIKI while you're at it. |
| 15:13.46 | Mpls-Eric | $[${myvar:0:1} = "a"] |
| 15:13.52 | Mpls-Eric | H's a faster typer |
| 15:14.25 | Mpls-Eric | www.voip-info.org/wiki/view/Asterisk+variables |
| 15:14.31 | *** part/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 15:15.57 | Mpls-Eric | Hey vlt, have more music files, with starts at different points(clip the beginnings) and pick them at random. |
| 15:16.28 | *** part/#asterisk sandorp (n=sandor@dhcp-242.phx3.llnw.com) |
| 15:17.36 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:19.12 | *** part/#asterisk tparcina (n=tparcina@cisco14.fesb.hr) |
| 15:21.33 | *** join/#asterisk Curi (n=Curi@200.24.227.122) |
| 15:22.16 | Curi | hello everyone, does anyone knows if i can rewrite the SIP packet that asterisk is sending in an AGI? |
| 15:23.03 | vlt | Mpls-Eric: hmmm, but then a caller continuing singing the song during the announcement will be confused when the music continues ... I can't take responsibility for this ;-) |
| 15:25.21 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:25.58 | angryuser | to stop dialplan exacution it is softhangup()? |
| 15:26.05 | angryuser | execution* |
| 15:27.01 | [TK]D-Fender | angryuser: "Hangup" or just run out of dialplan. |
| 15:27.16 | [TK]D-Fender | angryuser: Though I suggest "hangup" |
| 15:27.54 | *** join/#asterisk gmcinnes (n=gmcinnes@bas2-toronto63-1088792748.dsl.bell.ca) |
| 15:28.07 | *** join/#asterisk Flauto (n=zhao@adsl-69-212-194-6.dsl.chcgil.ameritech.net) |
| 15:28.26 | *** part/#asterisk omyz (n=sweetsug@mbl-65-158-104.dsl.net.pk) |
| 15:28.31 | Flauto | having a problem, asterisk would stop itself and saying core dumped |
| 15:28.36 | Flauto | what does it mean |
| 15:28.43 | gmcinnes | Hi guys: Any torontonians in here who could recommend a good DID provider with 416 numbers and 1-800 numbers? |
| 15:29.46 | angryuser | [TK]D-Fender: ok, but i i got for example xxxxxxxxx,1,Dial() and then xxxxxxxxxx,2,Hangup and i have "g" option set in Dial, will it hangup immediatly or *After* a call? |
| 15:29.53 | [TK]D-Fender | gmcinnes: www.unlimitel.ca |
| 15:30.13 | [TK]D-Fender | angryuser: Depends who ended the call. |
| 15:30.31 | gmcinnes | [TK]D-Fender: a ha! That's who I was trying to remember. Thanks. |
| 15:30.49 | [TK]D-Fender | angryuser: If the CALLER hung up, the call will end immediately. If the CALLEE hung-up, then "2" would be executed |
| 15:31.20 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:31.22 | [TK]D-Fender | gmcinnes: NP. Several of my clients use them and are very happy with the range and quality of services. |
| 15:31.23 | angryuser | [TK]D-Fender: ok, thank you |
| 15:31.28 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:31.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:32.03 | *** part/#asterisk hellop1 (n=hellop@udp112969uds.hawaiiantel.net) |
| 15:32.04 | Flauto | tkd-fender, i have a problem with voicestick did, it gives error 500 message when caller hangs up or it keeps going in circles within my ivr |
| 15:33.27 | [TK]D-Fender | ~anthm |
| 15:33.28 | jbot | Anthm just f'n rocks..... |
| 15:34.00 | gmcinnes | [TK]D-Fender: d'you know if they do rollover numbers etc.? I'm sure they do. |
| 15:34.18 | angryuser | in priorities if i have 1 2 4 and 3 is not present, 4 will never be executed wright? |
| 15:34.32 | angryuser | right |
| 15:34.34 | [TK]D-Fender | gmcinnes: These guys are really flexible, and even if they don't advertise a service doesn't mean they might not offer it to you. Call them up. |
| 15:34.48 | [TK]D-Fender | angryuser: Correct. |
| 15:35.19 | gmcinnes | [TK]D-Fender: right o. |
| 15:35.51 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 15:35.55 | Flauto | what is anthm |
| 15:36.21 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:36.21 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 15:37.28 | [TK]D-Fender | Flauto: A long-time contributor to * and the VoIP development community at large... |
| 15:37.53 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:38.00 | Flauto | hmm. |
| 15:38.34 | Flauto | have you run into any problem that 1.4 would stop itself and gives a message saying conre dumped? |
| 15:38.49 | [TK]D-Fender | Flauto: I don't use or advise using 1.4 yet. |
| 15:39.41 | gmcinnes | [TK]D-Fender: thanks. Are you in Toronto? |
| 15:39.52 | Flauto | hmm.. |
| 15:39.55 | Flauto | okay |
| 15:40.11 | Flauto | i only have this problem once in a while after i upgraded to 1.4 |
| 15:40.16 | [TK]D-Fender | gmcinnes: Montreal, but I didn't think that pertinent to answering your question :) |
| 15:40.36 | Flauto | i saw that 1.4 is released officially, so, i thought it would be okay |
| 15:40.56 | *** join/#asterisk RaeTheGit (n=sven@dslb-088-073-086-200.pools.arcor-ip.net) |
| 15:40.59 | RaeTheGit | heyho |
| 15:41.03 | gmcinnes | [TK]D-Fender: No, it wasn't :) I was just wondering if there was a * user group or something. I seem to always be getting drawn back to doing telephony programming :) |
| 15:41.24 | gmcinnes | [TK]D-Fender: I'm sure I can look one up and find it tho. |
| 15:42.41 | [TK]D-Fender | gmcinnes: Google up "TAUG" (Toronto Asterisk Users Group). They have regular meet-ups, etc.... |
| 15:42.45 | *** join/#asterisk bertrand^ (n=bertrand@mst45-2-82-242-81-196.fbx.proxad.net) |
| 15:42.53 | bertrand^ | hello |
| 15:43.39 | gmcinnes | [TK]D-Fender: thx again. |
| 15:45.06 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 15:45.14 | RaeTheGit | http://nopaste.biz/?12767 <-- this is what is supposed to deal with incoming calls in my extensions.conf - it is supposed to forward the call (depending on the 7.+8. digit) to different users. now, when i call, say, 30640307, asterisk says, the extension doesnt exist - why? |
| 15:46.17 | edgecase | gmcinnes, yeah i'm using unlimitel.ca they were the first to explicitly support asterisk in canada that i could find |
| 15:46.28 | RaeTheGit | funny enough, before, i made a few tries with the extension "s", which never disappeared in the logs after reload or restart, when i changed the entry to the above |
| 15:46.49 | RaeTheGit | it disappeared from the dialplan, though |
| 15:47.47 | RaeTheGit | any ideas? |
| 15:49.06 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:49.58 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:50.26 | PupenoR | How can I get the IP of a realtime sip peer? |
| 15:50.31 | [TK]D-Fender | RaeTheGit: Pattern matches have to be preceeded by "_". |
| 15:51.24 | JacksLivr | hey guys, where are the asterisk-sounds kept that are talked about in the o'reilly book. I can find many on the web, but I wanna get the right ones that the book refferences. I'm kinda following along in the book. thanks. |
| 15:51.46 | mercestes | PupenoR: "Sip show peers" with "qualify=yes" or "sip show peer <peername" maybe. |
| 15:52.09 | mercestes | JacksLivr: The default sound folder is under /var/lib/asterisk/sounds on most distros |
| 15:52.39 | RaeTheGit | [TK]D-Fender: *bang* thanks.... :o) |
| 15:52.55 | mercestes | JacksLivr: you might need to install asterisk-addons to get some of the extra ones. |
| 15:52.56 | *** join/#asterisk Netranger78 (n=netrange@24.214.236.85) |
| 15:53.02 | PupenoR | mercestes: what do you mean by "with" there? |
| 15:53.04 | JacksLivr | mercestes: this is like an expanded version |
| 15:53.06 | Netranger78 | morning gentlemen |
| 15:53.22 | JacksLivr | oh, they are in the addons |
| 15:53.22 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 15:53.24 | JacksLivr | thanks |
| 15:53.34 | mercestes | JacksLivr: no problem. good luck |
| 15:53.41 | funkmaster | hello ppl :) |
| 15:53.53 | Netranger78 | anyone up for helping me with a 7970 running sccp....ive got some wierd goings on with the speeddials |
| 15:53.57 | mercestes | PupenoR: IP in "sip show peers" requires that qualify=yes in your DB table. |
| 15:54.30 | funkmaster | i have a question, i have a problem with one of my sip providers, the registration time, in sip.conf what the max value i can set for it? or can i make it register only once and then not again or sumthing liek that |
| 15:54.56 | *** join/#asterisk seva (i=seva@66.90.103.12) |
| 15:55.28 | seva | is there a way to restrict the number of peers that can login using set of credentials? |
| 15:55.33 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 15:55.38 | PupenoR | mercestes: oh, thanks. |
| 15:55.41 | seva | like if i have a perr "foo" i want to make sure it's logged in only once |
| 15:55.44 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 15:55.57 | mercestes | PupenoR: NP, good luck |
| 15:56.27 | [TK]D-Fender | seva : if by "logged in" you mean registered, then you can only have 1. Any subsequent register will kill off its predecessors |
| 15:56.37 | mercestes | funkmaster: Maybe you should start with what is going wrong. If you set reg time to like, 9 billion, if it ever fails to register than it won't notice forever. |
| 15:56.54 | seva | [TK]D-Fender: i don't think that's the case.. i have multiple ones logged on right now |
| 15:57.01 | seva | the calls that they've alread placed are working fine |
| 15:57.16 | Netranger78 | here is what i have......got a 7970 running sccp.....i have other sip channels showing up as second lines on here....which i want....and they all ring through....however one of these.....lets call it 3678 ....when i call in on that line the speed dial light that it is assigned to doesnt flash and it looks like it is coming in on the main line of the phone....none of the others do this......any ideas? |
| 15:57.19 | [TK]D-Fender | seva : Using SLA on 1.4? Because thats the only way to multi-register a single user.... |
| 15:57.54 | seva | no, i have iax softphones connecting using a single peer setup |
| 15:57.59 | seva | they all login to a conference call |
| 15:59.12 | funkmaster | mercestes: in sip.conf defualtexpirey and maxexpirey do they have any maximum values? or i can set whatever i want? |
| 15:59.26 | mercestes | funkmaster: Try it. |
| 15:59.57 | bertrand^ | is it possible _not_ to choose a channel when talking to an E1? |
| 16:00.05 | bertrand^ | my provider asks me to do that |
| 16:00.14 | mercestes | funkmaster: I think setting qualify to a numeric value (like 100) is more than likely to fix whatever problem yoru having than setting expirey to an insane value if I am guessing correctly on what you are experiencing. |
| 16:00.45 | mercestes | funkmaster: However, your not telling us what's wrong so I'm guessing the max value for those variables is likely the maximum size of an INT32. |
| 16:01.29 | mercestes | bertrand^: By "choose a channel" you mean "dial any one of 30 or so available channels within an E1 PRI?" then that would be the default behavior. |
| 16:01.45 | *** join/#asterisk ChicagoBud (n=Bud@adsl-70-228-35-78.dsl.chcgil.ameritech.net) |
| 16:02.12 | mercestes | bertrand^: You define your e1 with a 1-30 with dchan=31 or something, assign it a group, then DIAL/ZAP/g1/###### etc. |
| 16:02.17 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 16:02.30 | ChicagoBud | hello, is there a spec for the number of characters in caller id name |
| 16:02.58 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 16:03.08 | Netranger78 | any ideas on the 7970 button issue? |
| 16:03.17 | aydiosmio | remove it |
| 16:03.27 | aydiosmio | then no one can complain it doesn't work |
| 16:03.39 | *** join/#asterisk queuetue (n=scott@70.54.254.134) |
| 16:03.53 | Netranger78 | lol....id love to....but unfortunately this will be sitting on the asssistant to the presidents desk |
| 16:03.54 | Netranger78 | lol |
| 16:04.24 | bertrand^ | mercestes, i've already tried every channel choice methods r1 R1 G1 g1, but i still can't make an outogin call |
| 16:04.37 | mercestes | ChicagoBud: It just cuts it off at like 11 chars or something, I can't remember exactly how many tho |
| 16:04.39 | bertrand^ | my phone provider now asks me "to not choose a channel" |
| 16:04.46 | mercestes | ChicagoBud: Just count them sometime. |
| 16:04.48 | aydiosmio | Netranger78: you say that like you should really care |
| 16:04.51 | bertrand^ | i'm not sure that's possible |
| 16:04.55 | aydiosmio | you shouldn't |
| 16:05.04 | aydiosmio | damn the man and all that. |
| 16:05.09 | [TK]D-Fender | bertrand^: You need to make sure of how you have allocated groups in your zapata.conf |
| 16:05.23 | Netranger78 | im sure in some deep dark part of my soul i dont.....but unfortunately they pay me to care......lol |
| 16:06.02 | funkmaster | mercestes: well the problem is like this, i have a voip sip provider configure in my sip.conf, works fine i can receive make calls etc.. but after a while it does not work anymore, e.g. incoming calls still make my phone ring but it does not show up in the console, though verbose mode and there is no connection established when i pick up, not audio transfer, so then i have to wait a couple hours, stop the registration of that provider during that |
| 16:06.02 | funkmaster | time and then after a couple hours or a day re-registrer, then it works again for while and the same thing starts over... |
| 16:06.07 | mercestes | Netranger78: Sounds like a config issue to me, but I can't tell you exactly *waht* config. |
| 16:06.23 | aydiosmio | Netranger78: sorry I can't provide anything other than comic relief |
| 16:06.27 | *** join/#asterisk darken_darken (n=marco@4.128.76.83.cust.bluewin.ch) |
| 16:07.06 | Netranger78 | im pretty sure i know where it is....i just cannot figure out why this one isnt working......in the /etc/asterisk/sccp.conf i have i line in there for this particular extension that reads |
| 16:07.07 | Netranger78 | autologin= 8955,8978,8985,3678 |
| 16:07.08 | mercestes | funkmaster: Weird, phone should not ring with nothing in the CLI, tho, if you need a faster rereg, qualify=100 is what you need, not expirey= 2 billion. What type of phones are these?? |
| 16:07.29 | JerJer | http://digg.com/software/How_To_Configure_Asterisk_Your_First_Installation_2 |
| 16:07.30 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:07.48 | Netranger78 | <PROTECTED> |
| 16:07.52 | Netranger78 | its wierd |
| 16:07.56 | aydiosmio | JerJer: a.k.a. How to let everyone on the planet make free phone calls through your SIP account |
| 16:07.58 | bertrand^ | [TK]D-Fender, you mena this line: channel => 1-15,17-31 ? |
| 16:08.07 | funkmaster | mercestes: well i have an ata grandstream 486 and use among others skypho, italian provider, which is the one causing the trouble.. |
| 16:08.18 | [TK]D-Fender | bertrand^: No, I mean "group=" for those channels. |
| 16:08.21 | funkmaster | will try with qualify=100, had turned it of so far.. |
| 16:08.38 | [TK]D-Fender | funkmaster: 100 is way to high. 2000 is the norm |
| 16:08.52 | JerJer | aydiosmio: no - a working, basic configuration of Asterisk |
| 16:09.00 | mercestes | lol |
| 16:09.05 | mercestes | ok, maybe 100 is a bit extreme. |
| 16:09.13 | funkmaster | but iqualify just checks if my client is reachable |
| 16:09.14 | *** part/#asterisk seva (i=seva@66.90.103.12) |
| 16:09.14 | mercestes | Maybe 500 |
| 16:09.19 | funkmaster | i don't c how that is gonna help me |
| 16:09.43 | aydiosmio | the last thing we need is everyone and thier brother running asterisk at home |
| 16:10.06 | mercestes | funkmaster: Ditch the grandstream ATA's. That wil help you. |
| 16:10.10 | funkmaster | lol |
| 16:10.16 | [TK]D-Fender | JerJer: Your macro is busted... exten => s,n,Goto(${DIALSTATUS}) <---- |
| 16:10.21 | bertrand^ | [TK]D-Fender: http://nopaste.biz/?12768 |
| 16:10.26 | funkmaster | well the grandstream box ain't the problem.. |
| 16:10.28 | JerJer | [TK]D-Fender: no it aint |
| 16:10.30 | bertrand^ | i use group=1 |
| 16:10.41 | JerJer | i'm going to that extension in the local context (macro) |
| 16:10.53 | [TK]D-Fender | JerJer: Goto with a single parm is to a PRIORITY within the current exten. |
| 16:11.00 | [TK]D-Fender | JerJer: Read again :) |
| 16:11.04 | funkmaster | anyone has worked with sphinx and speech recognition in asterisk? |
| 16:11.22 | [TK]D-Fender | JerJer: That should be exten => s,n,Goto(${DIALSTATUS},1) |
| 16:11.41 | JerJer | oddly enough thats pulled from working config files |
| 16:12.06 | JerJer | but i'll fix it |
| 16:12.09 | JerJer | done |
| 16:12.23 | [TK]D-Fender | JerJer: good :) |
| 16:12.30 | JerJer | any more input ? |
| 16:12.39 | [TK]D-Fender | JerJer: I jsut love erroneous * how-tos :) |
| 16:12.43 | Corydon-w | [TK]D-Fender: you can do a Goto(${DIALSTATUS}) if you have the corresponding label defined |
| 16:12.48 | JerJer | my goal here is to make this a valid document |
| 16:12.59 | [TK]D-Fender | JerJer: exten => ANSWER,1,Hangup |
| 16:13.04 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 16:13.05 | [TK]D-Fender | JerJer: Will never get called. |
| 16:13.09 | aydiosmio | someone should post a * conf validator |
| 16:13.13 | Corydon-w | [TK]D-Fender: in fact, it's generally good, because it preserves the extension |
| 16:13.33 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:13.55 | Netranger78 | well....i kinda figured out the button thing |
| 16:13.58 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 16:14.07 | [TK]D-Fender | Corydon-w: Idea can be nice, and can be worked a number of ways. Th one was simply WRONG. mismatch=bad. |
| 16:14.45 | [TK]D-Fender | aydiosmio: Sorry, its doing exactly what he's telling it to, and * is interpreted, not compiled. You can't know if every way a line will be called is going to be valid every time. |
| 16:14.47 | JerJer | [TK]D-Fender: hmm - i remember seeing Answer get fired for whatever reason |
| 16:14.48 | Netranger78 | looks like i cant make SIP channels ring well on a SCCP extension |
| 16:14.57 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:15.16 | pigpen | anyone know if asterisk "hints" can be stored in the RTA database yet? (last I saw it was moved to a feature request back in '06) |
| 16:15.29 | [TK]D-Fender | JerJer: Would also be nice if you called dial in a consitant format like other apps - Dial(SIP/${MACRO_EXTEN},25) |
| 16:15.30 | JerJer | but its mainly there to ensure a call doesn't make it into voicemail after the other end hangs up |
| 16:15.43 | mercestes | I've got a dumb question. What's wrong with the Wiki and thebook? |
| 16:15.54 | aydiosmio | everything, next question. |
| 16:15.59 | JerJer | oh - kill the pipe |
| 16:16.00 | JerJer | ok |
| 16:16.01 | [TK]D-Fender | JerJer: We both know (or you should) that the only reason for dialplat continue exectution would be if you passed Dial a "g" |
| 16:16.12 | JerJer | thats the old school coder in me |
| 16:16.29 | [TK]D-Fender | JerJer: Mixed school apparently :) |
| 16:16.42 | JerJer | [TK]D-Fender: that could be why I had to add it into my standard macro |
| 16:16.44 | [TK]D-Fender | JerJer: I might not have commented if it were consistant :) |
| 16:16.54 | kanaeda | lol britney gone crazy |
| 16:17.00 | kanaeda | http://www.youtube.com/watch?v=yHLQkWOFFvg |
| 16:17.31 | JerJer | [TK]D-Fender: input is good - Thank you |
| 16:17.54 | bertrand^ | if i used the following line, i get the clock from the line, right? span=1,1,0,ccs,hdb3 |
| 16:18.15 | JerJer | 1 uses that line as the primary clock source |
| 16:18.33 | JerJer | 2 secondary - 3, 4 ... so on |
| 16:18.35 | sweeper | I use your mom as my primary clock source |
| 16:18.43 | aydiosmio | ooooh burrrrn |
| 16:19.08 | Corydon-w | sweeper: It's a trifle bit early for that |
| 16:19.13 | *** join/#asterisk Strom_M (n=strom@65.197.244.4) |
| 16:19.15 | [TK]D-Fender | JerJer: Ok, while you're at it ${MACRO_EXTEN} is horrible IMO, I always pass the device to call as a parameter as the tech may change, etc. All keeping with the "flexible". But this is more optional as you are teacing some more basic stuff first. However the POINT of Macro's is parameters, and having a return point. You have NEITHER. |
| 16:19.25 | Corydon-w | sweeper: I'd suggest keeping that for after 10pm |
| 16:19.35 | sweeper | it IS after 10 |
| 16:19.40 | JerJer | [TK]D-Fender: ok |
| 16:19.41 | Corydon-w | pm |
| 16:19.46 | sweeper | yea, 10 pm |
| 16:19.56 | Corydon-w | Over here, it's 10am |
| 16:20.01 | bertrand^ | zttool says: Sync Source: Internally clocked |
| 16:20.04 | sweeper | I'm sorry |
| 16:20.07 | JerJer | [TK]D-Fender: I will make that point in the next article |
| 16:20.24 | bertrand^ | doesn't that mean it doesn't use the line as a primary clock source? |
| 16:20.46 | [TK]D-Fender | JerJer: Sometimes its an intentional scope lesson. I'll leave you some "artistic/scaling" leaway here :) |
| 16:20.48 | *** join/#asterisk zogulus_ (n=zogulus@58.98.adsl.brightview.com) |
| 16:21.30 | JerJer | yeah - I didn't want to get too complex for this first instance of asterisk article |
| 16:21.54 | zogulus | afternoon |
| 16:21.57 | JerJer | however it is a totally valid point |
| 16:23.08 | [TK]D-Fender | JerJer: sip.conf - mailbox should have the context, and does "peer" now fully take the place of "friend"? Might be nice to include callerID in there as well... |
| 16:23.48 | JerJer | i have always used peer in the SIP world |
| 16:25.09 | sweeper | zogulus: sorry, we run on official Digium time here, it is 10 AM |
| 16:25.39 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 16:25.41 | aydiosmio | at the sound of the fake silence, the time will be 11:24 AM |
| 16:25.56 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:25.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:25.59 | [TK]D-Fender | JerJer: Also in 1.4 IIRC you have to pass the VM message to play as parameter 2 now, not as a character prefix to the box. |
| 16:26.10 | sweeper | you are not allowed to make time of day-inappropriate comments, lest you be reprimaned~ |
| 16:26.16 | bertrand^ | what does your zttool say about your Sync source? |
| 16:26.18 | *** join/#asterisk edit_21 (n=edwho@unaffiliated/edit21/x-00000001) |
| 16:26.25 | edit_21 | evening all |
| 16:26.26 | JerJer | hmm |
| 16:27.51 | edit_21 | noob syupid question time, Can i configure asterisk to recieve phone calls from a pstn line - ie route to the asterisk answerphone |
| 16:27.59 | JerJer | wonder if there is code in there that checks? i serisouly pulled this config from my very functional home pbx |
| 16:28.10 | JerJer | Voicemail(${MACRO_EXTEN}, u) ? |
| 16:28.12 | sweeper | edit_21: yes |
| 16:28.19 | edit_21 | sweeper, thanks |
| 16:28.29 | [TK]D-Fender | JerJer: Correct. do a "core show application voicemail" |
| 16:28.45 | [TK]D-Fender | JerJer: Keeping in mind I read this once, and don't personally use 1.4 :) |
| 16:28.59 | JerJer | in run 1.4 on my home stuff |
| 16:29.04 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 16:29.16 | JerJer | but it does maybe 2 simultaneous calls - on an extreemely busy day :) |
| 16:29.57 | Netranger78 | here is what i have found with me button problem on the 7970......while i CAN make a SIP extension ring on a SCCP phone.....it will only ring in on the main number......it apparently doesnt send either enough or correct enough header information to make it specify WHICH line is ringing.....it just makes any incoming SIP calls ring on the main line. |
| 16:30.06 | pigpen | Sorry to be a noob, but what does the "a" extension provide to the voicemail app, while a greeting is being played? Just another extension like operator? |
| 16:30.16 | JerJer | the admin menu |
| 16:30.28 | JerJer | if you press * in Voicemail asterisk will attempt to call the 'a' exten |
| 16:31.07 | pigpen | while listening to voicemail or while the greeting is being played to a callee. |
| 16:31.09 | pigpen | ? |
| 16:31.15 | JerJer | which brings up another point - there is no way to check voicemail in my article :D |
| 16:31.25 | JerJer | pigpen: greeting |
| 16:31.41 | pigpen | so just assign it to operator is the standard? |
| 16:31.43 | [TK]D-Fender | pigpen: While you're listening to the greeting, you can press * to bomb out and do other stuff, like nag the receptionist to hunt down the person you don't want to leave a VM for. |
| 16:31.47 | zogulus | sweeper, ;) |
| 16:32.02 | pigpen | ah...opererator/receptionist. |
| 16:32.03 | pigpen | k. |
| 16:32.07 | [TK]D-Fender | pigpen: I just jumpts to "a,1" in the current context where you can do "whatever". |
| 16:32.26 | [TK]D-Fender | pigpen: I jstu gave a more common application of what you'd do with it. |
| 16:32.34 | pigpen | Postgress was bitching about it not being a defined extension in RTA. |
| 16:32.39 | pigpen | yes...still on the quest. |
| 16:33.00 | [TK]D-Fender | pigpen: in my home setup, I don't have a real IVR, I jsut ring-all, then hit VM. on * I VMauthenticate, then DISA for full remote access. |
| 16:33.02 | pigpen | thanks. |
| 16:33.22 | *** join/#asterisk seva (i=seva@66.90.103.12) |
| 16:33.27 | pigpen | Still trying to figure out _% ... |
| 16:33.30 | angryuser | is there any option for MISDN dialing to make it select automaticly the upper port, and if it is busy, select another one? or i need to write a script? |
| 16:33.39 | seva | [TK]D-Fender: this is basically what i am looking for http://bugs.digium.com/view.php?id=1164 |
| 16:33.52 | seva | " Currently IAX2 allows a given set of credentials to be used from more than one device without notice or complaint. This leads to situations where the two devices compete for call directed to their channel. This leads to missed calls and also to general confusion." |
| 16:33.52 | [TK]D-Fender | pigpen: In most of my clients setups' I usually use "*" as the way for them to ENTER their box (aside from an extra code). |
| 16:33.55 | seva | i want to limit it to 1 |
| 16:34.46 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 16:35.36 | [TK]D-Fender | seva : Sorry, I accidentally assumed "SIP". I have no idea how IAX2 will react. |
| 16:35.37 | pigpen | [TK]D-Fender, ah..good idea. |
| 16:36.21 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:36.29 | seva | it reacts by allowing multiple iax2 account to register, lat iax peer to register gets incoming call |
| 16:36.34 | seva | but outbound works for all the peers |
| 16:36.45 | [TK]D-Fender | JerJer: In your macro you should also be passing the VM context..... |
| 16:37.01 | [TK]D-Fender | seva : same as SIP then. |
| 16:37.20 | seva | [TK]D-Fender: what i'd like to do is prevent second registration from occuring |
| 16:37.29 | seva | or at least kick the original off |
| 16:37.34 | seva | and then register |
| 16:37.45 | seva | i might be submitting a patch for that soon ;) |
| 16:37.51 | [TK]D-Fender | JerJer: And continuing : exten => _1NXXNXXXXXX,1,Dial,SIP/${EXTEN}@NuFone <- this format has cause DNS issues with others in places where it is assumed to be a hostname, not a peer entry |
| 16:37.56 | seva | if i can't find a solution |
| 16:38.33 | Kritter | s/ 1 |
| 16:38.40 | [TK]D-Fender | JerJer: exten => _1NXXNXXXXXX,1,Dial(SIP/NuFone/${EXTEN}) <- consistant and clearer way to call an entry from sip.conf |
| 16:38.55 | JerJer | so perhaps DIal,SIP/NuFone/${EXTEN} |
| 16:38.55 | JerJer | ok |
| 16:38.57 | JerJer | editing |
| 16:39.07 | JerJer | [TK]D-Fender: do you want editing credit on my article ? |
| 16:39.13 | JerJer | i'll gladly give it to you |
| 16:39.23 | [TK]D-Fender | JerJer: Sure, why not :) |
| 16:41.29 | queuetue | I've tried trixbox and asterisknow, and they both have good points and lots of bad points... Do most of you just write dialplans by hand, or are there other tools out there to simplify life, that aren't quite as buggy or messy? |
| 16:42.30 | [TK]D-Fender | queuetue: This is the "know better and do it ourselves" channel. |
| 16:43.02 | [TK]D-Fender | queuetue: All of those GUI's simply limit you. Some have made custom tools for the most menial bits, but as it implied... CUSTOM. |
| 16:43.43 | mercestes | queuetue: the answer is, we do it by hand. |
| 16:43.55 | *** part/#asterisk seva (i=seva@66.90.103.12) |
| 16:43.58 | mercestes | .... the config files...I mean. |
| 16:44.00 | mercestes | I meant the config files |
| 16:44.09 | [TK]D-Fender | mercestes: TMI <- |
| 16:44.21 | mercestes | what? |
| 16:44.24 | mercestes | Oh you do it too, Fender1 |
| 16:46.08 | file | [TK]D-Fender: the floor is not very comfy |
| 16:46.44 | [TK]D-Fender | file: Careful now ;)\ |
| 16:47.16 | Netranger78 | ok....got another 7970 SCCP question.....how do i get the call forward softkey to work..... i tried adding cfwdall=on to the sccp.conf but it only shows up when you take the phone off hook.....any ideas? |
| 16:47.32 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 16:47.50 | Qwell[] | Netranger78: chan_sccp? yeah, it's there so you can forward to a number, which is why you have to go offhook |
| 16:47.59 | Qwell[] | go offhook, hit the button, dial a number |
| 16:49.22 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 16:51.27 | anonymouz666 | mpg123-0.59r is VERY old, so I am having problems to compile that under x64 arch... |
| 16:51.33 | PupenoR | Des $[] perform arithmetic operations on the dialplan? |
| 16:52.00 | anonymouz666 | format_mp3 is recommended ? |
| 16:52.12 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 16:53.12 | angryuser | anonymouz666: use wav 8 bit mono |
| 16:53.17 | mace | could i trouble anyone for some assistance with linking two asterisk servers? i keep getting "No authority found" errors on the called PBX but for the life of me can't figure out whats setup wrong :( |
| 16:53.50 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 16:54.12 | Netranger78 | ok....then ive got something else wrong then.....after i go off hook and press the cfwdall button it just beeps twice and goes back to dialtone |
| 16:54.19 | elriah | Hi all. In a call queue, what's the setting that tells the queue how long to ring each member before it gives up? |
| 16:54.29 | *** join/#asterisk yonahw (n=yonahw@84.228.169.3) |
| 16:54.39 | Netranger78 | tried to dial a number and hit end call and that didnt work |
| 16:54.49 | *** join/#asterisk apardo (n=apardo@87.217.144.244) |
| 16:55.59 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:58.00 | *** join/#asterisk slayer192 (n=slayer19@pirus.securax.be) |
| 16:58.41 | [TK]D-Fender | elriah: "timeout" |
| 16:59.03 | elriah | [TK]D-Fender: In the queue definition, right? Not from the queue cmd? |
| 16:59.13 | generalhan | Hey all !! i need a bit of direction on setting up a remote Cisco7960 to my local * box ( ver 1.2.10 ). Im not completely sure i have all the right ports forwarded over, or something. but i am NOT using NAT and i thought this would be simple, but the phone just wont connect. can some one point me in the right direction ? |
| 16:59.43 | [TK]D-Fender | elriah: Correct |
| 16:59.47 | generalhan | maybe a link with some good information or troubleshotting advice ? |
| 16:59.52 | elriah | [TK]D-Fender: Thanks. |
| 17:00.41 | elriah | Is there a way to play a specific MOH file instead of having to define a MOH class for a queue? |
| 17:03.47 | mercestes | elriah: You have to use both places. The "timeout=" in the queues.conf tells it how often to check to see if it's timed out, and the timeout in application queue is how long the timeout is. |
| 17:04.01 | *** join/#asterisk adsa (n=adas@S01060016d422485a.ed.shawcable.net) |
| 17:04.06 | elriah | Thanks! |
| 17:04.20 | Netranger78 | ok...back to my call forward issue.....i have just one more question....when i go off hook....press the cfwdall softkey and dial the number.....it actually dials out the number.....is there anywhere i can go to keep it from dialing and instead just forward to that number instead? |
| 17:04.21 | mercestes | elriah: And Set(MusiconHold=) or some nonsense like that should play custom MoH |
| 17:04.45 | elriah | mercestes: hrm.. Ok, thanks! :) |
| 17:04.55 | mercestes | np |
| 17:05.03 | mercestes | send me copies of vista via paypal. :D |
| 17:05.46 | elriah | eh? |
| 17:05.50 | elriah | vista? |
| 17:05.52 | mercestes | yea. |
| 17:06.03 | elriah | As in Windows Vista? |
| 17:06.04 | zogulus | I've got a couple of mods I want to make to the Asterisk server and I wondered if anyone could give me some hints on developing/testing them. e.g. do you always do a "make install" to test changes? |
| 17:06.28 | Qwell[] | zogulus: usually |
| 17:06.37 | Qwell[] | zogulus: I'll generally just copy the module over, if only one module changed |
| 17:06.47 | zogulus | Qwell, ah ok good to know that works |
| 17:07.52 | zogulus | Qwell, how about debugging, ever tried to bring it up in gdb? |
| 17:07.58 | Qwell[] | sure, always |
| 17:09.43 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 17:09.52 | zogulus | Qwell, it's been several years since I've done any C so the curve is looking a bit steep at the moment! ;) |
| 17:11.08 | elriah | mercestes: set(musiconhold=path/to/file) should play moh a specific file? |
| 17:12.59 | Qwell[] | elriah: I don't think so. I think you need to set a class |
| 17:13.07 | elriah | Qwell[]: Thansk. |
| 17:13.10 | Qwell[] | actually, that's only a var...that won't do anything |
| 17:15.54 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:19.01 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
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| 17:24.57 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-49.dhcp.leds.al.charter.com) |
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| 17:34.32 | *** join/#asterisk Abdu (n=Abdu@200.181.220.138) |
| 17:35.16 | Abdu | Someone know if the REINVITE works with AGI ? |
| 17:38.16 | *** part/#asterisk Curi (n=Curi@200.24.227.122) |
| 17:38.58 | *** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:39.03 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 17:39.13 | clyrrad1 | It seems like Canada has absolutly no decent provider for the 905 Exchange area code, does anyone here know of a decent provider? One that you can actually get a hold of? |
| 17:41.04 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 17:41.36 | Kritter | s 2 |
| 17:42.21 | clyrrad1 | anybody? |
| 17:42.29 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 17:42.33 | elriah | les.net |
| 17:42.43 | *** join/#asterisk sivana (n=sivana@208.68.91.47) |
| 17:43.01 | clyrrad1 | Do you use them? |
| 17:43.14 | elriah | Yep, great service. |
| 17:43.30 | clyrrad1 | are they able to Port / LNP DID's? |
| 17:43.33 | elriah | Yep. |
| 17:43.43 | clyrrad1 | how is the quality of Voice? |
| 17:43.49 | elriah | Great. |
| 17:44.03 | clyrrad1 | and what your opinions on the Tech support? |
| 17:44.08 | clyrrad1 | are they fast slow ? |
| 17:44.35 | elriah | I've never really had problems so dunno. |
| 17:44.55 | clyrrad1 | eheh |
| 17:44.59 | clyrrad1 | guess thats a good sign |
| 17:45.13 | Druken | clyrrad1: yes we do... |
| 17:45.24 | clyrrad1 | im checking them out - I was using telantek before - but these guys never answer the phone |
| 17:45.28 | clyrrad1 | Druken: who is that? |
| 17:45.33 | Abdu | Someone know if the REINVITE works with AGI ? |
| 17:46.13 | Druken | clyrrad1: you looking for a selfserve option or a large wholescale option ? |
| 17:46.55 | clyrrad1 | wholesale |
| 17:47.28 | ManxPower | Abdu: AGI itself should not prevent reinvites. |
| 17:47.33 | Druken | clyrrad1: www.thinktel.ca or 866-92think |
| 17:47.52 | ManxPower | Abdu: However, many other things can prevent reinvites from happening. |
| 17:48.09 | clyrrad1 | Druken: you work for this company? |
| 17:48.16 | Druken | nope... |
| 17:48.47 | clyrrad1 | ah you just know they do the 905 exchange? |
| 17:49.34 | Druken | clyrrad1: yeah... i used to deal with them on a regular basis, and have an extensive knowledge of their network :) |
| 17:51.13 | clyrrad1 | I have been having such a hard time to find a good reliable provider for these DID's |
| 17:51.14 | clyrrad1 | I use Unlimitel for the rest - those guys are great - I need a provider like that on the 905 DID's |
| 17:51.14 | Druken | well, keep in mind, the 905 is a HUGE regional area code.... |
| 17:51.14 | Druken | what ratecenters? |
| 17:51.14 | ManxPower | All Voip Phone Companies suck. |
| 17:51.15 | ManxPower | Some suck less that others, however. |
| 17:51.15 | Druken | ManxPower: how come i can never get them to suck??? :) |
| 17:51.15 | clyrrad1 | Drunken: yea this is true 905 is huge but I want some of the ones close to Toronto |
| 17:51.41 | clyrrad1 | I was using Telantek but seems like a one man show - been waiting a week to get a call back - horrible service |
| 17:52.07 | Druken | so you have numbers you want to port then... |
| 17:52.14 | clyrrad1 | yep |
| 17:52.26 | clyrrad1 | some for Alberta too |
| 17:52.45 | Druken | upper or lower? |
| 17:53.12 | clyrrad1 | im not sure the DID is 403-255-XXXX |
| 17:53.18 | [TK]D-Fender | Druken: Have you considered MAKEUP? ;) |
| 17:53.23 | Druken | lower, 403 is lower alberta |
| 17:53.41 | Druken | [TK]D-Fender: nah... i don't like the rainbow down there :) |
| 17:53.56 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 17:55.47 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:56.41 | xo8ox | guys whats the exact format settings for the asterisk voices so we can convert our own voice greetings to match it |
| 17:56.49 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 17:56.52 | Qwell[] | xo8ox: there are many formats |
| 17:57.00 | xo8ox | like .gsm 8000khz.. 8bit.. stereo/mono etc... |
| 17:57.18 | Qwell[] | yes, gsm is 8khz, mono |
| 17:57.20 | ManxPower | xo8ox: 8khz, mono, don't know if it is 16-bit or 8-bit |
| 17:57.25 | Qwell[] | 8 I think |
| 17:57.29 | [TK]D-Fender | xo8ox: If you're using any diecrt to PSTN hardware, then ULAW/ALAW depends on your region |
| 17:58.56 | xo8ox | well I did convert the wav that we have to the right GSM 8000khz and placed it in the sounds dir and used it but when we call that voice menu we hear distorted noise |
| 18:00.17 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:00.28 | generalhan | ok guys, i have an * box - ver 1.2.10 behind an IPCop box forwarding ports 5060 for Sip, 69 for TFTP, and my rtp port range from *. My Cisco 7960 is behind a little DI-624 router with the same ports forwarding, but the phone wont register. i tailed the logs on the * server and i cant see that the phone even hits that machine ... am i missing some ports that need to be forwared ?? |
| 18:00.39 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:03.27 | *** part/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 18:03.41 | ManxPower | 5060 is all you need for registration. |
| 18:04.45 | [TK]D-Fender | generalhan: 2 things can easily be screwd up. Your sip peer entry, or your general settings for NAT overall on your * box. |
| 18:05.06 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 18:05.49 | generalhan | well im not using NAT so that shouldnt be an issue ... |
| 18:06.04 | generalhan | does my sip entry look differently for a remote user than a local one ? |
| 18:06.08 | ManxPower | If you are forwarding ports then you are doing nat |
| 18:06.24 | generalhan | uh oh ... |
| 18:06.29 | [TK]D-Fender | generalhan: " ok guys, i have an * box - ver 1.2.10 behind an IPCop box " |
| 18:06.36 | ManxPower | Are you sure you cisco phone is configured to register? |
| 18:06.39 | [TK]D-Fender | generalhan: I dunno... SOUNDS like itt to me.. |
| 18:06.46 | generalhan | lol my fault. |
| 18:07.25 | generalhan | this is a phone that i ripped from the local net and was working. i was hoping to make it work at home that i would just have to forward the correct ports. |
| 18:07.31 | [TK]D-Fender | ManxPower: Sounds like a duck, walks like a duck, and its duck season, does that mean I can shoot him? ;) |
| 18:07.52 | [TK]D-Fender | generalhan: You sure the phone is pointed to your public IP now , and not the private one? |
| 18:07.58 | generalhan | yes |
| 18:08.29 | [TK]D-Fender | generalhan: Well it'd still be nice to verify that yoursip.conf is properly set up... |
| 18:08.48 | generalhan | ive tried 2 ways. connecting to the TFTP server on my local * machine. and making a TFTP server on a remote computer on the same net as the remote phone. niehter work. though when i check the sip settings on the phone it shows the public ip |
| 18:08.53 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 18:10.25 | generalhan | [TK]D-Fender: http://generalhan.pastebin.ca/373277 |
| 18:11.14 | generalhan | this is how all my Cisco 7960s are setup on the local network. i didnt change anything when i took the phone to a remote location. |
| 18:12.08 | *** join/#asterisk saftsack (n=oliver@pD9E0628F.dip.t-dialin.net) |
| 18:13.51 | generalhan | does that look like it should ? |
| 18:14.18 | *** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net) |
| 18:14.48 | tutt9876 | hi, got a problem to match an extension in my dialplan |
| 18:15.25 | tutt9876 | I am trying to make Dial(SIP/siapdres@sipdomain.com) |
| 18:16.03 | tutt9876 | with a exten => _[a-z0-9].@[a-z0-9]. patern but I can't get the domain |
| 18:16.08 | tutt9876 | any idea? |
| 18:16.10 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 18:16.37 | KuJaX | What is the command to hangup a line via the asterisk console? I am showing two of my extensions are using minutes and I need to manually hang them up via asterisk console. |
| 18:17.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:17.29 | tutt9876 | ${EXTEN] only get sipaddress |
| 18:18.07 | generalhan | i think you can do a 'hang up <channel>' id check on that though before you just trust me ! lol. wouldnt want it to hang up ALL channels ! lol |
| 18:18.20 | Qwell[] | soft hangup <channel>, actually |
| 18:18.46 | ManxPower | tutt9876: see README.variables in the asterisk source |
| 18:18.58 | *** join/#asterisk jesster_ (i=jesster@jesster.org) |
| 18:19.02 | tutt9876 | yes but didn't find the good one |
| 18:19.20 | generalhan | see this is why i say not to trust me ! lol |
| 18:19.43 | KuJaX | it says unknown channel when I put "soft hangup <extension>" |
| 18:19.53 | Qwell[] | not extension, channel |
| 18:20.01 | KuJaX | how do i see the list of chnnals? |
| 18:20.02 | generalhan | do a sip show channels |
| 18:20.10 | Qwell[] | hit tab |
| 18:20.15 | Qwell[] | soft hangup <tab> |
| 18:20.45 | jesster_ | Hey guys - I have some 7961 and 7941 phones running SIP 8.0.4 SR2 and am having problems getting the time to show. It keeps showing the wrong date and time. I see it's querying my NTP server - any suggestions where to look? my SEPmac.cnf.xml file is pretty much copy/paste from the voip-info wiki |
| 18:20.52 | tutt9876 | sip show channels? |
| 18:21.05 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 18:21.05 | KuJaX | Okay cool. Thanks! :) |
| 18:21.26 | ManxPower | jesster_: make sure the timezone offset is correct in the phone |
| 18:21.42 | generalhan | ManxPower: did you see my sip config for my cisco phone? i just need to make sure that they dont need to be set up differently for remote phones |
| 18:21.46 | KuJaX | Cool it worked! thanks, |
| 18:22.23 | jesster_ | ManxPower: would a time offset help the date? It's showint Jan 10th, 2007 UTC |
| 18:22.36 | PupenoR | Where can I read how to use labels in extensions.conf? |
| 18:22.37 | [TK]D-Fender | generalhan: With * being behind NAT as well as your remote phone, NEITHER is at all ready to function as-is |
| 18:22.54 | BrianR___ | I think I may have found a rather interesting bug in Asterisk 1.2.10. I have an asterisk box with two t1 interfaces sitting between another asterisk box and the pstn. It has two simple contexts which are causing it to forward calls in both directions without doing much else (like an exten => _X.,1,Dial(othert1,${EXTEN}) type of dialplan). Calls are working in both directions, but active channels don't show up in "show channe |
| 18:23.02 | generalhan | [TK]D-Fender: wonderful ... |
| 18:23.13 | [TK]D-Fender | generalhan: Actually... you never showed the [general] section... so maybe * is ready as a core... but I can't SEE that |
| 18:23.26 | BrianR___ | I'd say about 90% of calls don't show up in "show channels" or Master.csv. I don't even see the Dial() get executed when I turn up verbose. |
| 18:23.32 | BrianR___ | But the calls go through. |
| 18:24.09 | generalhan | [TK]D-Fender: sorry ... i has been a long time since i have had to think about this system. ill put together a post with as much information that is relevant ! |
| 18:25.17 | BrianR___ | Show channels says "0 active channels / 0 active calls" |
| 18:26.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:28.28 | [TK]D-Fender | BrianR___: Perhaps a user has unplugged * from the middle... |
| 18:28.49 | generalhan | [TK]D-Fender: [incoming] |
| 18:28.49 | generalhan | exten => _XXXXXXXXXX,1,NoOP(Call From-- ${CALLERID(number)} ** Calling to-- ${EXTEN}) |
| 18:28.50 | generalhan | exten => _XXXXXXXXXX,2,GotoIf($["${CALLERID(number)}"=""]?1000) |
| 18:28.50 | generalhan | exten => _XXXXXXXXXX,3,Goto(phone-numbers,${EXTEN},1) |
| 18:28.50 | generalhan | exten => _XXXXXXXXXX,1000,Set(CALLERID(number)="No-Number") |
| 18:28.50 | generalhan | exten => _XXXXXXXXXX,1001,Goto(phone-numbers,${EXTEN},1) |
| 18:28.56 | generalhan | OMG im sorry |
| 18:29.02 | BrianR___ | [TK]D-Fender: I thought that was a possibility too - I actually went and checkted the wiring closet. Of course that doesn't explain why some calls show up but not others. |
| 18:29.12 | [TK]D-Fender | generalhan: And I don't care about DIALPLAN. that isn't the issue here |
| 18:29.14 | generalhan | http://generalhan.pastebin.ca/373299 <------ what i meant to paste |
| 18:29.45 | generalhan | i just tossed it in cause it was referenced in the sip.conf ... didnt know if you wanted to see it ! |
| 18:29.50 | ManxPower | I doubt most phones will accept "no-number" as the callerid digits |
| 18:30.00 | [TK]D-Fender | generalhan: Ok, my first statement sticks... you have NONE of the NAT settings on either end for this to work. |
| 18:30.05 | generalhan | ManxPower: its for mine ! lol |
| 18:30.14 | generalhan | for labeling purposes |
| 18:30.50 | generalhan | [TK]D-Fender: what NAT settings ? im trying to learn HOW to do this ... ive never attempted before. |
| 18:31.13 | Qwell[] | ~nat |
| 18:31.15 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:32.49 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-7ff37d58558e8bd6) |
| 18:38.07 | *** join/#asterisk egypcio (n=egypcio@200.194.102.234) |
| 18:45.06 | *** join/#asterisk lukketto (n=lukketto@87.7.26.7) |
| 18:45.31 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 18:49.20 | *** part/#asterisk edgecase (n=jjackson@CPE00a0c9841796-CM000f9fa6b7d6.cpe.net.cable.rogers.com) |
| 18:49.29 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
| 18:52.53 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 18:58.49 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 19:04.17 | BrianR___ | [TK]D-Fender: Ok. I figured it out. Someone futzed with my wiring, and only calls which were overflowing onto a second span were going through the box. |
| 19:04.58 | BrianR___ | [TK]D-Fender: In fact, it may have been possible that I wired it wrong myself, since it appears the labeling is wrong :( |
| 19:07.41 | jesster_ | I'm trying to setup a 7941 / 7961 and am having problems with the date. It always shows 7:45 10/01/07 on it. I have the SEPmac.cnf.xml file set to Pacific Standard/Daylight Time and the dateTemplate is D/M/Y -- I verified with wireshark the phone is polling our NTP server, however, phone does not have correct time/date. Any suggestions would be great |
| 19:10.39 | *** join/#asterisk CrazyTux (n=CrazyTux@70.142.26.244) |
| 19:15.42 | *** part/#asterisk lukketto (n=lukketto@87.7.26.7) |
| 19:16.04 | *** join/#asterisk ozant (n=ozan@reverse-89-106-0-124.grid.com.tr) |
| 19:21.59 | cpatry | what could makes this happen, [Feb 26 14:20:10] WARNING[483]: channel.c:2380 ast_indicate_data: Unable to handle indication 3 for 'SIP/204-101588e8' , just just sometimes. |
| 19:24.10 | ManxPower | cpatry: not having a /etc/asterisk/indications.conf |
| 19:24.51 | cpatry | i do, if im calling directly, thats perfect, but if im using a background(foobar), it happens. |
| 19:25.28 | ManxPower | Exactly. |
| 19:25.47 | ManxPower | Asterisk uses /etc/asterisk/indications.conf to provide inband indications, like ringing after answer, etc. |
| 19:26.15 | cpatry | ive that file in my config dirs. |
| 19:26.55 | ManxPower | If the line has not been answered then Asterisk can use out of band indications and not need /etc/asterisk/indications.conf. cpatry, try copying the indications.conf.sample to /etc/asterisk/indications.conf in case it was corrupted. |
| 19:27.13 | cpatry | is res_indications.c is a must for that? |
| 19:27.16 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 19:27.26 | cpatry | ive no /etc/asterisk/ my config dir is at other place. |
| 19:28.00 | ManxPower | cpatry: perhaps asterisk is still looking for it in the default location. There has been issues with getting asterisk to look in non-default locations |
| 19:28.24 | ManxPower | cpatry: Yes, I imagine that res_inidcations.so would be required |
| 19:28.27 | cpatry | i will make few tests, but even my defaults.h doesnt point there. |
| 19:28.38 | cpatry | let me reload my res_indications |
| 19:30.20 | aydiosmio | can one of you fine lads tell me how to diagnose why asterisk is offering gsm and g711 but not g729 to my gateway for outgoing calls? |
| 19:30.41 | ManxPower | aydiosmio: I That is configured in sip.conf. |
| 19:30.48 | ManxPower | aydiosmio: do you have a G729 license? |
| 19:30.51 | aydiosmio | 0/0 encoders/decoders of 1 licensed channels are currently in use |
| 19:31.05 | Juggie | because you havnt set allow=g729 for your sip peer i would imagine |
| 19:31.46 | aydiosmio | ah yeah, there was no global allow-g729 |
| 19:31.59 | aydiosmio | thanks |
| 19:32.47 | *** join/#asterisk elpropagandista (n=user@63.110.13.126) |
| 19:34.36 | *** join/#asterisk elpropagandista (n=user@63.110.13.126) |
| 19:34.40 | *** join/#asterisk Daouid (n=daouid@zux221-071-059.adsl.green.ch) |
| 19:45.44 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 19:48.26 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:49.07 | *** part/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 19:50.55 | *** join/#asterisk sloth_ (n=chatzill@209.10.153.194) |
| 19:50.58 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 19:51.20 | iceyp | hey guys, I had a asterisk (on debian) server die yesterday, after the reboot my meetme has stopped working |
| 19:51.29 | iceyp | Feb 27 08:48:32 WARNING[2480]: chan_zap.c:915 zt_open: Unable to open '/dev/zap/pseudo': No such device or address |
| 19:51.37 | iceyp | However that file still exists |
| 19:51.43 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:51.44 | iceyp | crw-r--r-- 1 root root 196, 255 May 25 2006 /dev/zap/pseudo |
| 19:51.52 | iceyp | and i've also tried to run modprobe zaptel |
| 19:53.41 | cpatry | lsmod|grep zap |
| 19:54.25 | iceyp | zaptel 234340 0 |
| 19:54.26 | iceyp | crc_ccitt 2368 1 zaptel |
| 19:54.36 | Juggie | modprobe ztdummy |
| 19:54.44 | Juggie | unless you have a real card in there |
| 19:55.53 | iceyp | WARNING: Error inserting rtc (/lib/modules/2.6.16.18/kernel/drivers/char/rtc.ko): No such device |
| 19:55.56 | EmleyMoor | I'd like to add some additional country free calls to my Asterisk system - is there a listing of what can easily be done anywhere? |
| 19:56.07 | iceyp | and FATAL: Error inserting ztdummy (/lib/modules/2.6.16.18/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see |
| 19:56.18 | *** join/#asterisk darken_darken (n=marco@233.191.76.83.cust.bluewin.ch) |
| 19:56.33 | Juggie | iceyp, looks like you should reinstall zaptel, you probally updated your kernel |
| 19:57.19 | *** join/#asterisk darken_darken (n=marco@233.191.76.83.cust.bluewin.ch) |
| 19:57.44 | dlynes_laptop | iceyp: you need to rebuild your kernel, and enable the rtc module (real time clock) |
| 19:58.24 | EmleyMoor | I have US, UK, NL, NO and DE already |
| 19:58.43 | iceyp | ahh damn |
| 19:58.43 | iceyp | ok |
| 19:58.59 | iceyp | i usually use apt for that :/ |
| 19:59.02 | dlynes_laptop | Juggie: ztdummy requires the rtc and crc_ccitt modules; but somehow he's able to compile, link and install the drivers without support for it in his kernel |
| 19:59.24 | Juggie | dlynes_laptop, he said the problem occured without a reboot so i would imagine he had compiled on a old kernel |
| 19:59.31 | Juggie | er, after a reboot |
| 19:59.47 | dlynes_laptop | Juggie: yeah, but look closely at the error...it's trying to load a non-existent rtc module |
| 20:00.19 | dlynes_laptop | Juggie: and it's ztdummy that's trying to load it |
| 20:00.23 | iceyp | those files exist -rw-r--r-- 1 root root 19429 May 25 2006 /lib/modules/2.6.16.18/kernel/drivers/char/rtc.ko |
| 20:00.43 | iceyp | probably my apt-get update , updated my kernel with a non rtc module'd kernel |
| 20:00.54 | cpatry | uname -a? |
| 20:01.01 | iceyp | same with -rw-r--r-- 1 root root 152923 May 25 2006 /lib/modules/2.6.16.18/misc/ztdummy.ko |
| 20:01.06 | iceyp | Linux voip.unix.co.nz 2.6.16.18 #1 SMP PREEMPT Thu May 25 22:03:47 NZST 2006 i686 GNU/Linux |
| 20:01.25 | dlynes_laptop | iceyp: try the following: insmod /lib/modules/2.6.16.18/kernel/drivers/char/rtc.ko |
| 20:01.33 | dlynes_laptop | iceyp: what do you get when you try that? |
| 20:01.36 | *** join/#asterisk Nugget (i=nugget@68.93.27.60) |
| 20:01.40 | iceyp | insmod: error inserting '/lib/modules/2.6.16.18/kernel/drivers/char/rtc.ko': -1 No such device |
| 20:01.50 | dlynes_laptop | you don't have a real time clock? |
| 20:02.05 | iceyp | *shrug* this all worked before the reboot |
| 20:02.07 | dlynes_laptop | that's kinda like impossible, afaik |
| 20:02.14 | *** join/#asterisk foobar778 (i=johhny@ip68-100-41-120.dc.dc.cox.net) |
| 20:02.21 | dlynes_laptop | try the following, then: |
| 20:02.23 | dlynes_laptop | depmod -a |
| 20:02.27 | aydiosmio | when I Dial from an AGI, the calls don't seem to get logged to the CDR, do I have to do something so they get logged as a separate call? |
| 20:02.28 | Juggie | what does rtc depend on? |
| 20:02.36 | ozant | hi, anyone used app_icd with asterisk ? |
| 20:02.37 | Juggie | what hardware i mean |
| 20:02.55 | cpatry | icd? never heard about it. |
| 20:02.56 | dlynes_laptop | Juggie: I think it's the 8259 pic, but I'm not sure |
| 20:03.07 | iceyp | depmod -a run |
| 20:03.17 | foobar778 | Fender solved the problem about fxo found a better solution and free |
| 20:03.19 | ozant | cpatry, it is like app_queueu |
| 20:03.25 | RoyK | icd - intelligent call distribution iirc |
| 20:03.28 | cpatry | ozant: isnt in trunk. |
| 20:03.30 | dlynes_laptop | iceyp: what happened after you did 'depmod -a'? |
| 20:03.33 | RoyK | dunno if it's that intelligent, though |
| 20:03.36 | dlynes_laptop | iceyp: any errors? |
| 20:03.40 | iceyp | dlynes_laptop just went to the next line |
| 20:03.41 | cpatry | RoyK: hehehe |
| 20:03.48 | iceyp | dlynes_laptop no errors |
| 20:03.53 | dlynes_laptop | iceyp: let's try this |
| 20:04.05 | cpatry | isnt just include in openbpx? |
| 20:04.07 | ozant | RoyK, it seems it is just on opbx |
| 20:04.23 | cpatry | ozant: so better ask the opbx guys, no? |
| 20:04.23 | Juggie | cpatry, go leafs tonight :) |
| 20:04.25 | dlynes_laptop | iceyp: cp -R /lib/modules/2.6.16.18 /lib/modules/2.6.16.18-backup && rm -rf /lib/modules/2.6.16.18 |
| 20:04.32 | dlynes_laptop | iceyp: and then reinstall your kernel modules |
| 20:04.35 | cpatry | Juggie: bah, habs rocks. |
| 20:04.36 | RoyK | ozant: I beleive you can compile it for asstrix as well |
| 20:04.54 | iceyp | dlynes_laptop I feel a little embarresed but i wouldnt know how to do that, I'm a freebsd man |
| 20:04.56 | ozant | RoyK, hmm it worth a try |
| 20:05.00 | dlynes_laptop | cpatry: dood...everyone knows the canucks kick the habs' ass |
| 20:05.13 | Juggie | dlynes_laptop, thats kind of extreme... why not just apt remove and then apt install them. |
| 20:05.14 | dlynes_laptop | iceyp: didnm't you just finish saying you did that using apt-get? |
| 20:05.17 | foobar778 | [TK]D-Fender: are u there |
| 20:05.28 | [TK]D-Fender | foobar778: Yeah, whats up? |
| 20:05.33 | iceyp | dlynes_laptop yeh, so can i just do an apt-get update? |
| 20:05.40 | cpatry | dlynes_laptop: ya, you're too hot for yus |
| 20:05.41 | cpatry | us |
| 20:05.42 | Juggie | cpatry, the leafs are going to win tonight! :) |
| 20:05.43 | dlynes_laptop | Juggie: yes, it is extreme, but it leaves it in a known state...i.e. no leftover modules |
| 20:05.44 | cpatry | but not toronto. |
| 20:05.50 | foobar778 | I found a bettr soution to fx0 and free |
| 20:05.54 | RoyK | ozant: it's probably quicker to just try it with opbx first |
| 20:06.10 | foobar778 | most usa cities can go staright in and its not peering |
| 20:06.22 | [TK]D-Fender | foobar778: Oh, better,and free? Explain |
| 20:06.27 | foobar778 | ok |
| 20:06.37 | foobar778 | I have one did number ok |
| 20:06.51 | dlynes_laptop | iceyp: i really donm't know...i'm not a debian guy...juggie seems like he would know, though |
| 20:06.53 | iceyp | dlynes_laptop i ment apt-get upgrade |
| 20:06.54 | cpatry | damn, im still getting Driver for channel 'SIP/202-10155480' does not support indication 3, emulating it |
| 20:07.04 | iceyp | dlynes_laptop ok |
| 20:07.14 | foobar778 | i set it upon call in to dial an extesion wait send a DTMF and then disa |
| 20:07.18 | dlynes_laptop | cpatry: that's not an error...that's a warning |
| 20:07.39 | cpatry | dlynes_laptop: the ringing tone is so odd from that channel |
| 20:07.42 | cpatry | but okay from pstn. |
| 20:07.56 | Juggie | isnt ringing just audio on SIP? |
| 20:08.04 | *** join/#asterisk quetwo (n=quetwo@pplant-336.user.msu.edu) |
| 20:08.16 | Juggie | its just part of the RTP its not like the sip phone generates the ringing |
| 20:08.22 | foobar778 | I then route my number from free craigsnumber or grandcentral to that number in almost all udsa cities thats why the dtmf |
| 20:08.32 | dlynes_laptop | Juggie: yes, it's not an indication |
| 20:08.33 | foobar778 | usa |
| 20:08.55 | cpatry | not sure how to fix that thought. |
| 20:09.05 | dlynes_laptop | cpatry: nothing to fix |
| 20:09.06 | Juggie | the only way it could be an indication would be if chan_sip generated the ringing rather then * |
| 20:09.09 | dlynes_laptop | cpatry: nothing's broken |
| 20:09.10 | Juggie | but it would still be audio. |
| 20:09.10 | foobar778 | So anyone in the usa dials a local number and with DISA is in the pbx straight away |
| 20:09.27 | cpatry | dlynes_laptop: the tone is horrible and doesnt sound like normal. |
| 20:09.38 | dlynes_laptop | cpatry: then you need to fix your region |
| 20:09.38 | cpatry | when im background(foo) |
| 20:09.42 | foobar778 | Thus can do more than having a pstn fx0 |
| 20:09.45 | cpatry | but if im dialing directly, thats fine. |
| 20:09.50 | dlynes_laptop | cpatry: You're probably set for some country that's not yours |
| 20:10.03 | [TK]D-Fender | foobar778: Sounds ugly, but possibly effective. I wouldnt do taht to a company, but for your own free use you can use whatever makeshift method you want I guess. |
| 20:10.10 | cpatry | and how come, directly and from pstn is all fine? |
| 20:10.20 | foobar778 | works splendidly |
| 20:10.32 | Juggie | cpatry, je ne sais pas :) |
| 20:10.32 | dlynes_laptop | cpatry: pstn isn't generated |
| 20:10.46 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 20:10.52 | foobar778 | voicestick free did but credit card signup |
| 20:11.15 | dlynes_laptop | ~clue cpatry |
| 20:11.17 | jbot | ACTION fans out Vegas-style a stack of clues for cpatry "Pick one, any one! ...and don't show it to me." |
| 20:11.18 | cpatry | and why that behavior isnt the same for direct call? |
| 20:11.36 | Juggie | cpatry, perhaps you need to explain your call path thats working and the one thast not |
| 20:11.40 | Juggie | and which direction its going |
| 20:11.47 | dlynes_laptop | because for a direct call, asterisk isn't intervening, because you have canrevinvite=yes |
| 20:11.55 | cpatry | phones a is calling phones b directly, all okay. |
| 20:12.13 | dlynes_laptop | or because you're not going through asterisk at all, if you're using a sip url on the one phone to dial the other phone |
| 20:12.30 | cpatry | but if a is dialing my ivr, which is just background(foobar), then im dialing b'extensions, tone is odd to A. |
| 20:12.41 | Juggie | what is A? |
| 20:12.45 | Juggie | a cell phone, sip phone, etc. |
| 20:12.48 | dlynes_laptop | cpatry: CHECK YOUR REGION |
| 20:12.57 | cpatry | a,b are boths sip phones. |
| 20:13.07 | *** join/#asterisk netsurfer (n=netsurfe@user-5444b2bb.lns6-c10.dsl.pol.co.uk) |
| 20:13.08 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 20:13.09 | dlynes_laptop | cpatry: if the ringing soudns like aliens or somehting, you're probably set for Japanese region |
| 20:13.21 | dlynes_laptop | cpatry: not for Canada |
| 20:13.24 | cpatry | country=us |
| 20:13.34 | [TK]D-Fender | foobar778: Well more power to you then... |
| 20:14.08 | dlynes_laptop | See? There's your problem....your ringing is set up like George Bush's |
| 20:14.17 | dlynes_laptop | You've got an Iraqi ringtone |
| 20:14.57 | dlynes_laptop | cpatry: have you used a call to Set() in your dialplan anywhere to change the default region? |
| 20:15.10 | Juggie | cpatry, pastebin the working/broken dialplan |
| 20:15.12 | dlynes_laptop | cpatry: or modified indications.conf? |
| 20:15.20 | cpatry | dlynes_laptop: nope. |
| 20:15.35 | dlynes_laptop | cpatry: try juggie's suggestion then |
| 20:15.37 | dlynes_laptop | ~pb |
| 20:15.38 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 20:16.55 | cpatry | Juggie: http://www.pastebin.ca/373456 |
| 20:18.06 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 20:19.34 | [TK]D-Fender | cpatry: What kind of devices are those 2 SIP channels? |
| 20:20.04 | cpatry | 2 analog phones thru an mediatrix 1104 |
| 20:20.28 | *** join/#asterisk backblue (n=moo@87-196-99-21.net.novis.pt) |
| 20:22.38 | [TK]D-Fender | :/ |
| 20:22.44 | cpatry | tkfender: in what that matters? |
| 20:23.30 | [TK]D-Fender | cpatry: Just wondering if perhaps one end didn't have a complete SIP indications set on the remote side. |
| 20:25.44 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 20:28.17 | Carp1 | Alright, my internet is working again and I'm not tired anymore! |
| 20:28.28 | generalhan | [TK]D-Fender: ok i think 3rd times the charm here !! http://generalhan.pastebin.ca/373471 hows that look now !? |
| 20:28.30 | Carp1 | Anyone willing to help me with my one-way audio issuse...once again. |
| 20:28.48 | Mpls-Eric | Wow, I just added a single extension in extensions.conf and on reload, this is what I got: "pbx.c:929 pbx_find_extension: Maximum PBX stack exceeded" Any ideas? Running 1.4 SVN updated yesterday... |
| 20:29.09 | Mpls-Eric | New bug or stupid human trick? |
| 20:29.12 | [TK]D-Fender | generalhan: Getting warmer.... missing canreinvite=n (everywhere), and "nat=yes" for [general] |
| 20:29.44 | Carp1 | 1.4 is really buggy, |
| 20:29.48 | Carp1 | I had to downgrade to 1.2. |
| 20:29.55 | tzanger | Carp1: compared to trunk? :-) |
| 20:29.59 | Mpls-Eric | I'm learning that quickly... |
| 20:30.00 | generalhan | [TK]D-Fender: ohh i didnt notice that in the docs ... so im looking for a nat=yes line under the general context? and i need canreinvite=no for ALL sip entries ? or just for the remote phones ? |
| 20:30.02 | [TK]D-Fender | Mpls-Eric: 10 PRINT "I AM GOING INSANE" 20 GOTO 10 |
| 20:30.17 | Carp1 | tzanger? |
| 20:30.28 | [TK]D-Fender | generalhan: Highly advisable for ALL. |
| 20:30.56 | mercestes | [TK]D-Fender: ROFLMAO |
| 20:30.57 | generalhan | hmm ok ... will that restrict functionality that i have gotten used to on the local phones ? lol |
| 20:31.03 | cpatry | Mpls-Eric: hey dude, whats up? |
| 20:31.14 | mercestes | [TK]D-Fender: 10 bucks says he has no clue what your talking about. |
| 20:31.14 | cpatry | Mpls-Eric: and if you remove that extension, its all fine? |
| 20:31.24 | [TK]D-Fender | generalhan: No, nothing to lose, all to gain |
| 20:31.31 | Mpls-Eric | I though I should run 1.4 in the hopes of improving it. Never thought it was going to be released in such bad shape. |
| 20:31.36 | Mpls-Eric | About to try removing now. |
| 20:31.59 | Mpls-Eric | Google was no help, tried that first, now its IRC, next its used my head....! |
| 20:32.07 | generalhan | [TK]D-Fender: perfect ! thanks for the help ... again. it was you and Qwell about a year and a half ago that helped me get this thing up and running in the first place !! |
| 20:32.18 | cpatry | Mpls-Eric: change #define AST_PBX_MAX_STACK 128 for 256 to see. |
| 20:32.24 | cpatry | in ur pbx.c |
| 20:32.30 | Mpls-Eric | Problem solved... Removed line. |
| 20:32.34 | [TK]D-Fender | generalhan: Bill is in the mail ;) |
| 20:32.50 | generalhan | lol! |
| 20:33.14 | cpatry | whats ur line looks like btw? |
| 20:34.13 | Mpls-Eric | Here's the real problem, I'm calling a destination inside a macro, ie Dial(blah/blah|o)... After the remote party hangs up, my polycom used to go back "onhook", now it sits with dead air... CLI shows hangup, but polycom still sits. |
| 20:34.35 | Mpls-Eric | autofallthrough=yes |
| 20:35.17 | Mpls-Eric | It was stupid after thinking about it. I added ;exten => h,Hangup() |
| 20:36.06 | cpatry | and if ur increase ur value like i said, no more warnings? |
| 20:36.57 | Mpls-Eric | I'm guessing that calling Hangup from h, creates the loop. I haven't tired changing AST_PBX_MAX_STACK |
| 20:37.28 | Carp1 | http://pastebin.ca/373491 this is my sip.conf...... |
| 20:37.33 | Carp1 | I'm getting one way audio. |
| 20:37.50 | Mpls-Eric | Carp1, do you have NAT anywhere? |
| 20:38.23 | vlt | Hello. Does anyone know how to play moh but prevent callers waiting in a queue hearing the first part of an moh file over and over again? |
| 20:38.25 | Carp1 | only inbound calls. |
| 20:38.32 | Carp1 | I believe I took care of the NAT |
| 20:38.34 | cpatry | Mpls-Eric: so all fixed now? |
| 20:38.37 | Carp1 | Yes, I have a router.... |
| 20:39.16 | Mpls-Eric | Well, the real problem is the polycom phone not going onhook when the remote party hangs up the line. CLI shows hangup, but phone sits. |
| 20:39.58 | Mpls-Eric | Carp1, have you made any adjustments in your sip.conf file so that asterisk understand the nat you have? |
| 20:40.27 | Carp1 | http://pastebin.ca/373491 |
| 20:40.30 | Carp1 | Thats my sip.conf |
| 20:40.37 | Carp1 | I did localnet |
| 20:40.54 | Carp1 | I had externip but I commented it out becuase it wouldnt allow incoming calls when I had it. |
| 20:41.56 | Mpls-Eric | One-way audio as in works from PBX to phone, but not phone to pbx? |
| 20:42.36 | Carp1 | yes. |
| 20:42.56 | Carp1 | When I call out, it works both ways, when someone calls in, only one way works. |
| 20:43.00 | Carp1 | I cant hear anything but they can hear me. |
| 20:43.46 | Mpls-Eric | Are you forwarding the UDP ports to the pbx in your nat box? That's your problem I think. |
| 20:43.54 | Carp1 | Yes, I am. |
| 20:44.02 | Carp1 | the port range I defined in rtp.conf |
| 20:44.08 | Mpls-Eric | What ports? Are you sure? What NAT are you using? |
| 20:44.28 | Carp1 | It's a Linksys router, and I fowarded the defaults.....10000-20000 I believe. |
| 20:44.56 | *** part/#asterisk backblue (n=moo@87-196-99-21.net.novis.pt) |
| 20:45.19 | Mpls-Eric | I'd use tcpdump and filter for the remote IP space and see what you get incomming on thiose ports. I'm guessing bug or misconfiguration. |
| 20:45.41 | Carp1 | I dont know how to do that :( |
| 20:47.00 | J4k3 | wow.. now my f1000g spontanious reboots. |
| 20:48.23 | *** join/#asterisk ToyMan (n=Stuart@dpc6714368169.direcpc.com) |
| 20:51.39 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 20:51.52 | *** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader) |
| 20:52.14 | Carp1 | hmm. |
| 20:52.30 | *** join/#asterisk gmcinnes (n=gmcinnes@bas2-toronto63-1088792748.dsl.bell.ca) |
| 20:53.42 | generalhan | Anyone know if you can edit config settings on a 7960 via telnet .. as opposed to just looking at what the settings are ? |
| 20:53.42 | Nugget | telnet is eeeeeeevil! |
| 20:53.56 | generalhan | ... |
| 20:54.36 | mercestes | ftp is eeeeeeeeevil. |
| 20:56.28 | jesster_ | I'm trying to setup a 7941 / 7961 and am having problems with the date. It always shows 7:45 10/01/07 on it. I have the SEPmac.cnf.xml file set to Pacific Standard/Daylight Time and the dateTemplate is D/M/Y -- I verified with wireshark the phone is polling our NTP server, however, I don't see any packets back from the NTP server to the phone (NAT problem?) Any suggestions would be great |
| 20:56.46 | rad07 | Hi, I have MultiModem ZPX (MT5634ZPX-PCI Series) Can I use it with Asterisk |
| 20:58.06 | rad07 | If somebody can have a look at ftp://ftp.multitech.com/3rd-party-patches/unix/mt5634zpx-pci_linux.txt to see if it is usable |
| 20:59.38 | spaceinvader | hmm I have Modem: Intel Corporation 82801AB |
| 21:00.27 | rad07 | spaceinvander: Is it working? |
| 21:01.04 | spaceinvader | havent trie |
| 21:01.10 | dlynes_laptop | cpatry: Juggie had asked you paste the working/broken dialplan, not your log file |
| 21:01.14 | spaceinvader | i want to make a poor-mans personal PBX |
| 21:01.26 | dlynes_laptop | spaceinvader: get a long string and two tin cans |
| 21:01.27 | spaceinvader | using that modem for the line and the sound card for the phone |
| 21:01.43 | spaceinvader | dlynes_laptop: where do I stick the ethernet for SIP? ;P |
| 21:01.50 | cpatry | dlynes_laptop: let me write ya a quick one to simplify things. |
| 21:02.20 | dlynes_laptop | cpatry: before giving me the quick one to simplify things, test it, and make sure it still shows the same problem |
| 21:02.37 | cpatry | dlynes_laptop: of course. |
| 21:02.50 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 21:02.51 | spaceinvader | dlynes_laptop: would it work? |
| 21:03.01 | generalhan | anyone know if configuration options can be changed via telnet for a Cisco 7960 ? |
| 21:03.04 | rad07 | Is your voice modem used as FXS, FXO or both FXO/FXS. Can I connect my regular phone and dial out and receive calls as well as receiving inbound calls and route it first to Asterisk PBX if I don't answer (this should be FXO function) and can somebody call me via SIP software and will my regular phone ring (FXS function)? |
| 21:03.05 | spaceinvader | dlynes_laptop: you see the box I want to use has no spare PCI slots |
| 21:03.08 | dlynes_laptop | spaceinvader: stick it in your ear :) |
| 21:03.20 | spaceinvader | dlynes_laptop: :) |
| 21:03.29 | spaceinvader | rad07: voicemodems can only be used for the line afaik |
| 21:03.35 | dlynes_laptop | spaceinvader: obviously you've never heard of firewire or usb ethernet adapters? |
| 21:03.48 | spaceinvader | dlynes_laptop: noooo |
| 21:03.56 | spaceinvader | dlynes_laptop: i meant the FXO/FXS |
| 21:04.06 | dlynes_laptop | spaceinvader: sipura 3000 |
| 21:04.08 | spaceinvader | dlynes_laptop: it has 2 PCI NIC's, a modem and a tv card |
| 21:04.12 | spaceinvader | dlynes_laptop: no more space |
| 21:04.21 | dlynes_laptop | spaceinvader: a tv card in a phone server? that makes sense |
| 21:04.28 | spaceinvader | dlynes_laptop: 2 nics for routing, modem is on its own silly slot, and tv for streaming tv |
| 21:04.35 | spaceinvader | dlynes_laptop: its my home server |
| 21:05.00 | elriah | What does ALI actually stand for? |
| 21:05.14 | elriah | Automatic Location Information? |
| 21:05.23 | dlynes_laptop | elriah: don't you mean ANI? |
| 21:06.06 | rad07 | spaceinvade: voicemodems can only be used for the line afaik. What do you mean? Aren't we talking about FXO/FXS functionality |
| 21:06.57 | spaceinvader | same thing |
| 21:07.02 | dlynes_laptop | rad07: besides...voicemodems with the shorted jumper so that they function as an x100p or x101p give you horrible quality audio with a shitload of echo |
| 21:07.18 | dlynes_laptop | rad07: and you also get dropped calls on those from time to time |
| 21:07.29 | spaceinvader | FXO goes to the line, FXS goes to the telephone |
| 21:07.38 | dlynes_laptop | spaceinvader: sipura 3000 |
| 21:07.44 | spaceinvader | dlynes_laptop: no space... |
| 21:08.09 | dlynes_laptop | spaceinvader: no space? do you live in a microwave oven? |
| 21:08.25 | spaceinvader | dlynes_laptop: oh, thats an ATA |
| 21:08.31 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:08.40 | dlynes_laptop | spaceinvader: no, it's not an ATA |
| 21:08.44 | dlynes_laptop | spaceinvader: it's a SIP gateway |
| 21:08.51 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 21:08.52 | dlynes_laptop | spaceinvader: one fxo port, one fxs port |
| 21:09.05 | dlynes_laptop | spaceinvader: an ATA has one or more FXS ports, and no FXO ports |
| 21:09.09 | rad07 | dlynes_laptop: I just want to see if it works and allow my parents to ring me on a regular phone. MultiModem ZP MT5634ZPX-PCI should be a good voice card configurable via script? |
| 21:09.23 | dlynes_laptop | rad07: no |
| 21:09.44 | dlynes_laptop | rad07: unless you are feeling overly ambitious and you feel like writing your own driver for it to work with asterisk |
| 21:10.07 | justdave | hmm, what's this mean? :) |
| 21:10.08 | justdave | Feb 26 13:07:12 NOTICE[29117]: chan_iax2.c:3161 iax2_read: I should never be called! |
| 21:10.12 | rad07 | spaceinvader: Am I right to say as you said FXO goes to the line, FXS goes to the telephone that proper voice modem card can be used instead of FXO/FXS. |
| 21:10.27 | spaceinvader | rad07: no |
| 21:10.48 | spaceinvader | rad07: FXO goes to the line, FXS to the telephone, and you can only use a voice modem as a FXO |
| 21:10.51 | dlynes_laptop | rad07: a voicemodem card has an analog in port, and an analog pass-through port |
| 21:11.01 | dlynes_laptop | rad07: it does not have an fxs port |
| 21:11.16 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:11.23 | spaceinvader | dlynes_laptop: seems a waste of money since its only going to be used for the FXO/FXS and nothing else |
| 21:11.52 | spaceinvader | dlynes_laptop: this is going to be a poor-mans^Wstudents version ;p |
| 21:12.26 | dlynes_laptop | spaceinvader: so get your trusty student's pcb designer, and acid bath, and copper clad boards and parts, and make your own :) |
| 21:12.51 | spaceinvader | dlynes_laptop: a possibiliy but it would be nicer to use existing junk |
| 21:12.57 | dlynes_laptop | spaceinvader: do you really need an fxs port? |
| 21:13.08 | rad07 | dlynes_laptop: "it does not have an fxs port"!! spaceinvader is saying that FXS goes to the telephone |
| 21:13.11 | [TK]D-Fender | spaceinvader: You had us at "junk". |
| 21:13.39 | spaceinvader | FXS is any device that, from the point of view of a telephone, seems to be a telephone exchange. |
| 21:13.40 | [TK]D-Fender | rad07: it is a PASSTHROUGH, not a CONTROLLED port. |
| 21:13.55 | spaceinvader | dlynes_laptop: probably not |
| 21:14.14 | spaceinvader | dlynes_laptop: it would be ok if i could just rig up a "normal" phone and use it for VoIP activities |
| 21:14.15 | Carp1 | Has anyone here been through the asterisk bootcamp? |
| 21:14.29 | [TK]D-Fender | spaceinvader: And to end this : The FXO passing through is what makes the phone work. Your computer can't DO anything with it however. |
| 21:14.29 | Mpls-Eric | Hey, cpatry... I tied again to add an extension, this one wasn't like the h,Hangup() and it did the same as the other... I tried the 128 to 256 setting and it didn't help. |
| 21:15.01 | [TK]D-Fender | spaceinvader: Buy and ATA or get a card with proper FXS on it. I'd personally suggest the former. |
| 21:15.41 | rad07 | [TK]D-Fender: Can you explain plz. I just want my folks oversees to ring me on my normal phonen via Asterisk SIP call. Is the regular phone plugged in the phone jack acting as passthrough in that case (and not a controlled port that can receive ring/calls via voip) |
| 21:15.47 | justdave | Carp1: I'm going to the one in April in San Jose, but I haven't been to one before, so I'm probably no help to you |
| 21:15.52 | Mpls-Eric | OK, never mind again, stupid human trick again. I forgot to add a priority when coding it up. |
| 21:16.19 | [TK]D-Fender | rad07: Buy an ATA or get a real FXS card. Some cheap junk modem will NOT work. |
| 21:16.41 | dlynes_laptop | rad07: if you want to be able to ring the analog phone from asterisk, whether it's coming from the analog line or from a sip call, you either need a true FXS port, or a SIP gateway |
| 21:16.42 | spaceinvader | [TK]D-Fender: Ive seen several things saying you could use a sound card as a FXO |
| 21:16.57 | dlynes_laptop | rad07: a pass-through port will not work for those purposes. PERIOD. |
| 21:16.59 | [TK]D-Fender | spaceinvader: And what else do your Rice Crispies say to you? |
| 21:17.06 | mercestes | lol |
| 21:17.09 | spaceinvader | [TK]D-Fender: they say use gentoo |
| 21:17.13 | mercestes | Space, didn't we just have this conversation in #gentoo-chat? |
| 21:17.13 | spaceinvader | [TK]D-Fender: its ricylicous |
| 21:17.14 | rad07 | How were x100p or x101p cards used then? |
| 21:17.17 | spaceinvader | mercestes: yep |
| 21:17.29 | mercestes | Didn't *I* tell you it wouldn't work that way? |
| 21:17.36 | [TK]D-Fender | rad07: X100/X101 were FXO. for LINES. get it? not FXS. |
| 21:17.38 | dlynes_laptop | mercestes: he's an invader from space...that should explain it right there... |
| 21:17.39 | Carp1 | What is the best book to read reguarding asterisk.....The asterisk handbook? |
| 21:17.40 | spaceinvader | well, the internet suggests its possible |
| 21:17.59 | mercestes | of course it's possible. Anything is possible. I fyou wish to build/write it yourself. |
| 21:18.00 | *** join/#asterisk Schreiber1337 (i=cee4b403@gateway/web/cgi-irc/ircatwork.com/x-2029cbf872bddcb6) |
| 21:18.01 | spaceinvader | dlynes_laptop: OK, so i am after a budget FXS card |
| 21:18.05 | mercestes | I could use my flashlight as a cattleprod too. |
| 21:18.10 | dlynes_laptop | spaceinvader: you're dreaming |
| 21:18.14 | dlynes_laptop | spaceinvader: there's no such thing |
| 21:18.19 | spaceinvader | dlynes_laptop: they are $$$? |
| 21:18.21 | [TK]D-Fender | spaceinvader: "The internet" is not a quoteable or reliable source. Perhaps you should provide a specific reference. |
| 21:18.32 | spaceinvader | [TK]D-Fender: guess so |
| 21:18.40 | [TK]D-Fender | spaceinvader: Suggestion : Linksys SPA-2002 |
| 21:18.41 | J4k3 | spaceinvader: theres a bluetooth module floating around that claims it'll make your bt earpiece work as an 'extension' |
| 21:18.48 | spaceinvader | hmm |
| 21:18.51 | J4k3 | not an fxs, but your sound card wouldn't be either. |
| 21:18.53 | dlynes_laptop | spaceinvader: tdm400p with one fxs module is about the cheapest you're going to get there |
| 21:18.56 | [TK]D-Fender | spaceinvader: $70 for 2 ports. Decent quality. |
| 21:18.57 | dlynes_laptop | spaceinvader: about $150 |
| 21:18.58 | J4k3 | (a true fxs, simply an extension) |
| 21:19.11 | spaceinvader | hmm |
| 21:19.19 | [TK]D-Fender | dlynes_laptop: One of many reasons I never suggest PCI FXS :) |
| 21:19.21 | J4k3 | I think what spaceinvader wants is a cheap extension device, not a true rj-11-on-the-back fsx |
| 21:19.22 | dlynes_laptop | spaceinvader: you probably want more like a sipura 3000, (one fxo, one fxs) for about $80 or so |
| 21:19.22 | J4k3 | er fxs |
| 21:19.22 | spaceinvader | how much would a native IP phone cost? |
| 21:19.26 | *** part/#asterisk ozant (n=ozan@reverse-89-106-0-124.grid.com.tr) |
| 21:19.38 | J4k3 | spaceinvader: you can buy one that "works" for about $40 USD. |
| 21:19.41 | dlynes_laptop | spaceinvader: about $35-40 for a grandstream budgetone |
| 21:19.45 | J4k3 | a good one.. expect to pay 2-3x that |
| 21:19.49 | Mpls-Eric | softphone = free |
| 21:19.55 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 21:19.58 | spaceinvader | hardware... ;p |
| 21:20.04 | dlynes_laptop | spaceinvader: but it has super low gains, and looks like a toy |
| 21:20.10 | spaceinvader | 0 items found for |
| 21:20.14 | spaceinvader | sipura 3000 |
| 21:20.16 | spaceinvader | ebay fails |
| 21:20.18 | J4k3 | it looks like a $12 walmart POTS phone |
| 21:20.21 | dlynes_laptop | spaceinvader: spa-3000 |
| 21:20.21 | J4k3 | with an ethernet jack on the back |
| 21:20.25 | J4k3 | and a webserver |
| 21:20.27 | [TK]D-Fender | spaceinvader: If you'd like to use a SINGLE phone for both your VoIP calls to your folks, as well as having access to an analog line where you are, then the Linksys SPA-3102 is a more appropriate choice for you. |
| 21:20.37 | spaceinvader | I dont really need analog |
| 21:20.42 | spaceinvader | Thats just a nice addition |
| 21:20.49 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 21:20.55 | Bobthehunter | anyone have comments on david levine |
| 21:20.57 | [TK]D-Fender | spaceinvader: SPA-3102 has 1 FXS and 1 FXO in a single ATA frame. |
| 21:21.34 | spaceinvader | £85.00 GBP :( |
| 21:21.43 | tzanger | with the asterisk sounds strutcure with en/ es/ fr/ etc is it necessary to have the sounds in /var/lib/asterisk/sounds anymore or will asterisk automatically try to locate the correct sound in the en/ directory (for example) |
| 21:21.53 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-148-242.ny325.east.verizon.net) |
| 21:22.13 | J4k3 | spaceinvader: if you're not after super-high-quality, just get a grandstream budgetone 101 and be happy with it |
| 21:22.23 | [TK]D-Fender | spaceinvader: Nobody said all this hardware was FREE. If you're that cheap, then get a computer headset and use a soft-phone. |
| 21:22.35 | [TK]D-Fender | spaceinvader: Cost = $5 |
| 21:22.48 | J4k3 | treat yourself to a USB headset, $8 |
| 21:22.50 | J4k3 | hehe |
| 21:23.46 | jesster_ | anyone have any problems using NTP with Cisco's ? |
| 21:24.01 | *** join/#asterisk Snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 21:24.33 | spaceinvader | J4k3: headsets ftl! |
| 21:24.48 | mercestes | ftl? |
| 21:24.56 | J4k3 | ft...linux? |
| 21:25.01 | dlynes_laptop | mercestes: fark the loogan |
| 21:25.19 | generalhan | [TK]D-Fender: so still wont connect !! i think i know why, when i telnet into the remote phone and check the config it shows "nat_enable 0" ive been checking with cisco's docs to see if i can change that setting from telnet, rather that just see that its wrong ! lol. any idea if thats possible ? |
| 21:25.44 | spaceinvader | for the loss |
| 21:25.54 | mercestes | that makes no sense. |
| 21:25.54 | rad07 | [TK]D-Fender: What exactly is "a true rj-11-on-the-back fsx"? is it =ATA =combination of FXO/FXS? |
| 21:26.16 | [TK]D-Fender | rad07: TDM400 + FXS modules. |
| 21:26.21 | J4k3 | "ATA... hey my motherboard has 6 of those connectors!!! Super ATAs!!!" |
| 21:26.25 | J4k3 | hehe |
| 21:26.32 | spaceinvader | ;p |
| 21:26.34 | [TK]D-Fender | rad07: Thats a PCI card based solution. |
| 21:26.46 | Schreiber1337 | Anyone ever run across a situation where the 1st callers voice is lost after a transfer... Happens on all internal transfers. |
| 21:27.25 | [TK]D-Fender | rad07: You can also get an ATA. These are Ethernet FXS devices that let you plug in a regular phone and allow it to talk SIP to other devices (like Asterisk) |
| 21:27.49 | [TK]D-Fender | Schreiber1337: "canreinvite=no" <- your friend |
| 21:27.50 | spaceinvader | hah my school has the worst PBX ever |
| 21:27.52 | spaceinvader | its ancient |
| 21:28.23 | Bobthehunter | anyone tried serveriron... |
| 21:28.36 | Bobthehunter | and ser + server iron |
| 21:29.01 | rad07 | [TK]D-Fender: I would like to have a device that will work in both Windows/Unix environment (what are the best choices of USB and PCI based devices. I would like to play with only one FXO/FXS for now). I need a device that can be connected to Microsoft Speech server (to have TIM drivers for SIP) |
| 21:29.08 | spaceinvader | GDK-162 |
| 21:29.13 | *** join/#asterisk techie (n=techie@67.181.184.170) |
| 21:29.56 | [TK]D-Fender | rad07: Depends what your software supports. |
| 21:29.59 | [TK]D-Fender | BBIAB |
| 21:30.01 | [TK]D-Fender | heading home |
| 21:31.18 | rad07 | [TK]D-Fender: Let's say I wanted to program in .NET (web based multipoint conferencing) and interconnect users via Asterisk |
| 21:32.10 | spaceinvader | Price: $241.00 |
| 21:32.11 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 21:32.11 | spaceinvader | hmm |
| 21:32.35 | spaceinvader | The Wildcard TDM11B Retail Package is the most budget overall setup I could find |
| 21:32.47 | nextime | hi. Is chan_misdn in * 1.4.0 stable usable, or it is better to upgrade it from the latest chan_misdn snapshot? |
| 21:32.52 | rad07 | Is there any difference between ATA device and PCI based FXO/FXS card regarding receiving/sending calls from a regular phone (from PSTN and voip calls) |
| 21:33.04 | *** join/#asterisk frenzy_ (n=frenzy@unaffiliated/frenzy) |
| 21:34.20 | spaceinvader | rad07: an ATA dosent need a pc |
| 21:34.26 | spaceinvader | rad07: its just a consumer device like a router |
| 21:34.46 | spaceinvader | rad07: whereas a PCI based FXO/FXS card can exploit the full capability of asterisk |
| 21:35.05 | *** join/#asterisk Aces1Up (n=rich@wsip-24-234-88-23.lv.lv.cox.net) |
| 21:35.06 | jesster_ | Any idea why my TFTP logs request United_States/g3-tones.xml for Cisco phones? |
| 21:35.09 | rad07 | spaceinvader: Can it work with asterisk then? If yes, how? |
| 21:35.25 | Aces1Up | does anyone here have a good suggestion for a softphone that works great with asterisk?> |
| 21:36.10 | E-bola | Aces1Up: xlite |
| 21:36.34 | spaceinvader | rad07: I dunno with an ATA |
| 21:36.44 | spaceinvader | rad07: a FXO/FXS is basically designed for it |
| 21:36.48 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 21:38.45 | Aces1Up | xlite keeps crashing' |
| 21:38.51 | fetcher | rad07: I've used ATAs as Asterisk ports quite a bit. As FXS's they usually do OK |
| 21:38.51 | *** join/#asterisk lymers (n=Lyme@manufacturerstransportation.com) |
| 21:39.10 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 21:39.26 | fetcher | rad07: ATAs with FXO (telco line) ports are less common, and the only one I messed with, a Sipura 3000, had some problems with low audio volume, or echo if the gain was turned up to compensate |
| 21:39.44 | DocHolliday | what do you guys use to power PoE phones? |
| 21:40.12 | Nugget | I hook them up to treadmills and make the call center staff burn off the calories from lunch. |
| 21:40.24 | spaceinvader | heh |
| 21:40.27 | lymers | have a quick question, im trying to setup a ivr that actually dials outside lines... could i use something like Dial("SIP/334", "SIP/MTI-PBX/18005556666|300|") ? or would that not work? |
| 21:40.30 | fetcher | rad07: one drawback is that you'll have a bit more latency (speech delay) with the ATA, because of internal buffers inside the device |
| 21:41.38 | fetcher | lymers: I don't think you can dial two endpoints at once like that |
| 21:42.02 | rad07 | fetcher: Is this any better? Sipura SPA-3102 http://www.telephonyware.com/telephonyware/tw00307.html |
| 21:42.48 | lymers | fetcher: i figured the first of it was caller-id info? |
| 21:42.57 | spaceinvader | Under misc HW support it says ALSA, what kind of setup could you use that for? |
| 21:43.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:43.18 | rad07 | fetcher: If I install one of Sipura ATA's will I be able to dial out from asterisk to my cell when my parents oversea call me via SIP call |
| 21:43.43 | fetcher | lymers: Caller ID should contain the SIP phone's extension by default. To override it and show something else, you have to SetCallerId() on a separate line, prior to the Dial() |
| 21:43.56 | Schreiber1337 | [TK]D-Fender: That did it.. Thanks. |
| 21:44.22 | Mpls-Eric | <PROTECTED> |
| 21:44.56 | fetcher | rad07: That looks like an updated version of the SPA-3000. OK as an answering machine or IVR |
| 21:45.22 | *** join/#asterisk topping (n=topping@adsl-68-122-119-108.dsl.pltn13.pacbell.net) |
| 21:46.22 | fetcher | rad07: What would you use as the second outside line, to dial the cellphone after receiving a call? SPA-3xxx has only one line port-- the other jack is for a local phone |
| 21:46.46 | jesster_ | Does anyone have a copy of g3-tones.xml and mk-sip.jar |
| 21:46.58 | jesster_ | for Cisco |
| 21:47.53 | DocHolliday | jesster_, why do you need those? mine works fine without em' |
| 21:48.24 | dlynes_laptop | cpatry: i guess the stripped down version wasn't eliciting the same problem? |
| 21:48.24 | Bobthehunter | yo.. can serveriron do LOAD balance on RTP ? |
| 21:48.26 | *** join/#asterisk tecolote99 (n=user@63.110.13.126) |
| 21:48.27 | jesster_ | DocHolliday: NTP for setting date and time is ignored unless it is able to download the locale configuration files |
| 21:49.20 | DocHolliday | jesster_, if you find copies let me know :P |
| 21:49.25 | rad07 | fetcher:1.VoIP to PSTN and PSTN to VoIP Gateway (both origination and termination) 2.Forward calls to and from the PSTN / VoIP service 3.Advanced inbound and outbound call routing and dialplan support |
| 21:50.00 | jesster_ | DocHolliday: there's some on the CCO site but none are in en |
| 21:50.09 | lymers | Executing Goto("SIP/105-0899dd58", "Dial("SIP/MTI-PBX/600|300|")") anyone have a idea how to make that actually dial ext 600? |
| 21:50.24 | lymers | /lost |
| 21:50.38 | DocHolliday | jesster_, sucks, just realized the time on my phone is off :P |
| 21:50.49 | jesster_ | DocHolliday: :) |
| 21:51.09 | lymers | maybe use from-internal ? |
| 21:51.13 | rad07 | fetcher: These are some of the features of this device? It seems it can replace Asterisk. But I wanted to use some inteligence from my program to route the calls. Let's say I wanted to use an application to text-to-speech call out a client and allow some interaction with a program with DB connectivity |
| 21:51.51 | jesster_ | The phone has internal en locale files but always requests them from TFTP |
| 21:52.10 | DocHolliday | indeed |
| 21:52.11 | *** join/#asterisk dj-fu (n=ajc@203.211.96.8) |
| 21:55.26 | fetcher | rad07: you can use it with * for basic FXO functions. The worst problem may be the low audio level, if they haven't fixed that in the newer model, but for personal home use it's mostly ok... just an annoyance |
| 21:55.38 | vlt | Hello. Can I change the codec during a running call? |
| 21:55.49 | fetcher | rad07: the SPA will be harder to setup compared to a PCI card, and less flexible in some ways. e.g. I never figured out a way to make mine pick up the line while it was already in use by another POTS phone |
| 21:55.51 | vlt | ^^SIP or IAX |
| 21:55.55 | rad07 | fetcher: Thanks, I hope this version is supported by Asterisk. This is an USB device |
| 21:56.49 | fetcher | rad07: USB? SPA-3xxx? That doesn't sound right. It should connect through Ethernet |
| 21:57.02 | fetcher | vlt: No way to do that, as far as I know :( It would be a useful feature |
| 22:00.35 | lymers | anyone have any ideas? |
| 22:01.37 | lymers | could anyone help me with using goto to dial a number? |
| 22:02.21 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:03.06 | dlynes_laptop | lymers: exten => _X.,1,Goto(my_new_dialplan_context,${EXTEN},1) |
| 22:04.10 | dlynes_laptop | lymers: then in my_new_dialplan_context, exten => _X.,1,Dial(SIP/mysippeername/${EXTEN}) |
| 22:04.51 | dlynes_laptop | vlt: you can, if your phone supports it |
| 22:05.06 | dlynes_laptop | vlt: but it's a very phone-dependent method |
| 22:05.11 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:05.11 | lymers | dlynes_laptop: thanks, ill see what i can do with that... |
| 22:05.22 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqvl.cable.mindspring.com) |
| 22:06.42 | *** join/#asterisk gatuno (n=gatuno@230.red-82-158-212.user.auna.net) |
| 22:08.40 | fetcher | dlynes_laptop: that would affect only transmitted audio from the phone, not the receive path, right? |
| 22:09.04 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 22:10.03 | lymers | dlynes: thanks for your help, but i actually got from-internal,600,1 to work =) |
| 22:10.13 | fetcher | Sipura/Linksys ATAs can switch to T.38 on the fly on hearing a FAX tone, which is sort of the same thing |
| 22:10.18 | rad07 | fetcher: USB? It is a mistake |
| 22:10.19 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 22:10.33 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 22:10.52 | *** join/#asterisk zaf (n=zaf@rader-laf-gw.radersolutions.com) |
| 22:11.35 | dlynes_laptop | fetcher: yeah...not sure...never used it |
| 22:11.57 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 22:15.07 | *** join/#asterisk netsurfer (n=netsurfe@user-514f09ee.l1.c3.dsl.pol.co.uk) |
| 22:16.11 | dlynes_laptop | fetcher: actually...you use the dialplan variable: Set(${SIP_CODEC}=g729) |
| 22:19.57 | [TK]D-Fender | dlynes_laptop : Alsmot got it... jsut a little closer :) |
| 22:20.44 | dlynes_laptop | [TK]D-Fender: ? |
| 22:20.55 | DocHolliday | anyone here used the Cisco ATA 186 I1? |
| 22:21.06 | dlynes_laptop | [TK]D-Fender: almost got what? |
| 22:21.49 | [TK]D-Fender | dlynes_laptop : that statement right :) |
| 22:23.28 | fetcher | dlynes_laptop: interesting! Is there a parallel to that for IAX? |
| 22:23.53 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-73-237.vic.bigpond.net.au) |
| 22:24.32 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:24.39 | fetcher | dlynes_laptop: right now I'm defining multiple copies of peers, like [customer1-g729], [customer1-ulaw] differing only in the allow= lines, which seems like an ugly kludge |
| 22:25.28 | dlynes_laptop | fetcher: that's not channel specific...it's a general dialplan variable |
| 22:25.40 | dlynes_laptop | [TK]D-Fender: i still don't follow you |
| 22:25.55 | [TK]D-Fender | dlynes_laptop : "Set(${SIP_CODEC}=g729)" <- formatting error |
| 22:26.17 | dlynes_laptop | [TK]D-Fender: how so? |
| 22:26.38 | [TK]D-Fender | dlynes_laptop : Thats not how you set a variable..... |
| 22:26.57 | [TK]D-Fender | dlynes_laptop : Think how that parses... |
| 22:27.48 | dlynes_laptop | [TK]D-Fender: oh yeah...sorry...1.0 method |
| 22:27.53 | mercestes | lol |
| 22:27.55 | mercestes | he still doesn't get it |
| 22:27.56 | dlynes_laptop | Set(SIP_CODEC=g729) |
| 22:28.03 | mercestes | oh, yea he does. |
| 22:28.03 | mercestes | :) |
| 22:28.04 | mercestes | yay |
| 22:28.05 | [TK]D-Fender | EGADS! |
| 22:28.06 | Snapple42 | hey all.. out of curiosity... what do people out there use to monitor calls... jitter/loss/etc...? |
| 22:28.18 | dlynes_laptop | mercestes: in the old 1.0 days you had to put the braces in there |
| 22:28.24 | mercestes | really? |
| 22:28.25 | mercestes | nice. |
| 22:28.32 | mercestes | I think I prefer it that way, actually. |
| 22:28.35 | dlynes_laptop | mercestes: I just don't use variables regularly, so my brain was still in 1.0 mode |
| 22:28.42 | [TK]D-Fender | dlynes_laptop : No you didn't.... and back then it'd be SetVar ;) |
| 22:28.56 | dlynes_laptop | [TK]D-Fender: yeah...but you'd specify the braces, too :) |
| 22:29.16 | [TK]D-Fender | dlynes_laptop : You never set a variable on the left within braces... |
| 22:29.31 | [TK]D-Fender | dlynes_laptop : thats what you use to GET the value of a variable, never to set |
| 22:29.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:30.00 | [TK]D-Fender | <- Syntax Nazi.... NO SYNTAX FOR YOOUUUUUUUU!U!!!!!@!@!@!@ |
| 22:30.15 | dlynes_laptop | yeah...ok |
| 22:30.20 | dlynes_laptop | so i haven't had my timmy's yet today |
| 22:31.11 | [TK]D-Fender | dlynes_laptop : I know the feeling... after my first 3 cups in the morning I'm the best so-and-so around ;) |
| 22:31.28 | dlynes_laptop | unca timmah's is da bomb :) |
| 22:31.46 | dlynes_laptop | too bad they don't have any in washington state |
| 22:32.03 | dlynes_laptop | so when I drive south, I end up having to go to dunkin donuts or whatever |
| 22:32.07 | dlynes_laptop | those places suck |
| 22:32.22 | dlynes_laptop | and starpukes |
| 22:32.24 | dlynes_laptop | ewwww |
| 22:33.00 | Mavvie | is there here somebody who wants to share a polycom configuration file with me? |
| 22:33.00 | [TK]D-Fender | dlynes_laptop : You forgot to rant on American beer! |
| 22:33.23 | dlynes_laptop | [TK]D-Fender: it's good if you need to go to the washroom bad |
| 22:33.26 | [TK]D-Fender | Mavvie : next thing you know you'll be looking to pass used needles around.... stop the INSANITY!!!! |
| 22:33.28 | dlynes_laptop | [TK]D-Fender: it's pretty close to water |
| 22:33.40 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 22:33.42 | [TK]D-Fender | dlynes_laptop : Starts the same as it ends.... piss water :D |
| 22:33.55 | ChicagoBud | Anyone know what's up with the SpanDSP website? soft-switch.org |
| 22:34.00 | Mavvie | [TK]D-Fender: we're moving from Ciscos to Polycoms. I don't consider it really getting worse to be honest :-) |
| 22:34.38 | [TK]D-Fender | Mavvie : Worse? You just jumped on the best bad-wagon in town! |
| 22:34.40 | lymers | isnt there a way to dail directly to someone's voicemailbox? i forget how though =( |
| 22:35.20 | [TK]D-Fender | lymers : Yeah, make an exteen that just calls voicemail. End of story. |
| 22:35.27 | Carp1 | Voicemail({$EXTEN}) |
| 22:35.51 | [TK]D-Fender | Carp1 : Hazardous for all that implies.... |
| 22:35.57 | lymers | err no i thought there was a shortcut allready installed? |
| 22:36.12 | Carp1 | hmm |
| 22:36.26 | [TK]D-Fender | lymers : Nothing exists that you didn't create yourself. |
| 22:36.38 | mercestes | lol |
| 22:36.53 | *** join/#asterisk p0w3r3d (n=p0w3r3d@201.255.171.162) |
| 22:37.01 | ChicagoBud | lymers, or Voicemailmain(${EXTEN}) if that is what you really want |
| 22:37.39 | [TK]D-Fender | ChicagoBud : No, I seriously doubt that it is... UNPREFIXED... thats going to seriously clash with any sane dialplan.... |
| 22:37.44 | [TK]D-Fender | WAKE UP TIME PEOPLE! |
| 22:38.37 | Carp1 | Well, whats the answer then? |
| 22:38.47 | mercestes | and for those of you with screwed up dialplans and strange error message, I do offer consulting to fix what these other guys did! :D |
| 22:39.05 | ChicagoBud | [TK]D-Fender, true. VoicemailMain(${CALLERID(num)}) |
| 22:39.18 | ChicagoBud | for internal use |
| 22:41.22 | dlynes_laptop | ChicagoBud: what's wrong with it? |
| 22:41.23 | Carp1 | ChicagoBud: THat doesnt work for me, |
| 22:41.57 | fetcher | after some testing, SIP_CODEC appears to only affect SIP, not IAX |
| 22:41.57 | mercestes | Why not just "VoicemailMain()"? |
| 22:41.58 | ChicagoBud | dlynes_laptop, rxfax seems to crash * for me |
| 22:42.09 | dlynes_laptop | ChicagoBud: because it doesn't work with asterisk 1.4 |
| 22:42.24 | dlynes_laptop | ChicagoBud: it's not supported on anything higher than 1.2.9.1 |
| 22:42.29 | [TK]D-Fender | ChicagoBud : No... NOT Voicemailmain. that is NOT what he asked for. |
| 22:42.34 | fetcher | only channels/chan_sip.c references that variable, as of 1.2.13 anyway |
| 22:42.35 | [TK]D-Fender | YOU ARE ALL GETTING COLDER |
| 22:42.39 | JT | fetcher: doesn't tjat sound obvious? :) |
| 22:42.44 | JT | s/tjat/that/ |
| 22:42.47 | file | [TK]D-Fender: you are amusing me |
| 22:43.03 | dlynes_laptop | http://www.zarzamora.com.mx/mirror/www.soft-switch.org <-- here's a mirror for anyone that's having difficulty accessing soft-switch.org |
| 22:43.08 | fetcher | JT: yeah, thought so, but someone early said it might work for IAX... worth a try :) |
| 22:43.13 | dlynes_laptop | Steve's sleeping, so no chance of getting it online atm |
| 22:43.15 | [TK]D-Fender | file : Moreso because you similarly know the answer and are laughing at the chaos of it all :) |
| 22:43.17 | fetcher | earlier |
| 22:43.27 | JT | fetcher: ah ok, probably not then |
| 22:43.31 | [TK]D-Fender | file : Almost pathetic, isn't it? |
| 22:43.51 | ChicagoBud | dlynes_laptop, really? It seems like there are reports of it working: http://forums.digium.com/viewtopic.php?t=13448&highlight=fax |
| 22:44.14 | dlynes_laptop | ChicagoBud: perhaps, but the author doesn't officially support anything newer than 1.2.9.1 |
| 22:44.28 | ChicagoBud | dlynes_laptop, I think I have a bad libtif or a bad version of spandsp |
| 22:44.58 | ChicagoBud | dlynes_laptop, Interesting. Bad for me. |
| 22:45.11 | dlynes_laptop | ChicagoBud: i've had good experience using the version of spandsp distributed with iaxmodem, though |
| 22:45.17 | *** join/#asterisk remmo (n=chatzill@smack.isp.net.au) |
| 22:45.18 | dlynes_laptop | ChicagoBud: iaxmodem.sf.net |
| 22:45.27 | ChicagoBud | dlynes_laptop, under 1.4? |
| 22:45.43 | dlynes_laptop | ChicagoBud: no...I don't have a spare system to test 1.4 on |
| 22:45.49 | dlynes_laptop | ChicagoBud: so I'm still using 1.2 |
| 22:46.32 | ChicagoBud | dlynes_laptop, yeah, sometimes I thing 1.4 was a bad choice but I figurred the next time I would upgrade would be 2.0 |
| 22:47.04 | dlynes_laptop | ChicagoBud: You mean in another 5 years? |
| 22:47.27 | ChicagoBud | dlynes_laptop, LOL yeah... I hope |
| 22:47.41 | dlynes_laptop | ChicagoBud: that wasn't meant to be funny...I was serious |
| 22:48.12 | dlynes_laptop | ChicagoBud: It took more than 2 years to go from 1.2 to 1.4 |
| 22:48.26 | dlynes_laptop | ChicagoBud: and 1.0 had been out for probably 3 or 4 years before 1.2 came out |
| 22:48.48 | ChicagoBud | dlynes_laptop, I mean I hope I don't have to upgrade for 5 years |
| 22:49.04 | dlynes_laptop | ChicagoBud: you're running 1.4.0.0.0 |
| 22:49.09 | dlynes_laptop | ChicagoBud: of course you're going to have to upgrade |
| 22:49.15 | ChicagoBud | dlynes_laptop, I know... |
| 22:49.20 | Corydon-w | dlynes_laptop: uh, no. It was only about 2 years between 1.0 and 1.2 |
| 22:49.34 | Corydon-w | dlynes_laptop: and only about 1.5 years between 1.2 and 1.4 |
| 22:49.38 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 22:49.40 | dlynes_laptop | Corydon-w: oh...so there was a lot of people running on a pre 1.0 asterisk? |
| 22:49.40 | Qwell[] | eh? I thought it was more like a year |
| 22:49.48 | Corydon-w | dlynes_laptop: yes |
| 22:50.01 | Corydon-w | Qwell[]: it was at least 14 months |
| 22:50.10 | Qwell[] | a far cry from 2 years :p |
| 22:50.16 | dlynes_laptop | Corydon-w: ummmm....1.4 was supposed to come out last June, and it didn't come out until December(?) |
| 22:50.23 | Corydon-w | Qwell[]: I meant between 1.2 and 1.4 |
| 22:50.32 | Qwell[] | I meant 1.0-1.2 |
| 22:51.03 | Corydon-w | dlynes_laptop: there are still people running on pre-1.0 Asterisk |
| 22:51.19 | *** join/#asterisk type0 (i=type0@216-67-30-183-cdsl-rb2.cwc.acsalaska.net) |
| 22:51.26 | type0 | wassup? |
| 22:51.33 | *** join/#asterisk axisys (i=vadud3@anapnea.net) |
| 22:51.45 | dlynes_laptop | Corydon-w: what's it good for, besides a conversation piece? |
| 22:51.58 | Corydon-w | dlynes_laptop: if it ain't broke... |
| 22:52.14 | type0 | what's the average sip session use (data wise) with a semi decent codec? |
| 22:52.25 | JT | 85kbit/s each way |
| 22:52.27 | JT | g.711 |
| 22:52.34 | [TK]D-Fender | dlynes_laptop : It did come out around June... if you consider the "Beta" the "real deal" |
| 22:52.46 | type0 | damn.. that's quite a bit |
| 22:52.57 | type0 | considering a PRI would use the 64k timeslot |
| 22:53.09 | Qwell[] | type0: plus part of the D channel |
| 22:53.19 | type0 | fair enough.. but still no where near 85 ;0 |
| 22:53.20 | Qwell[] | type0: consider network overhead, SIP packets, rtp overhead |
| 22:53.35 | type0 | yeah.. I'm just REALLY limited to the bandwidth I have |
| 22:53.42 | Qwell[] | then don't use g711 |
| 22:53.44 | JT | type0: tdm is obviously more efficient if you want high quality |
| 22:54.00 | type0 | ie, t-1 over a 500 mile microwave link with 7 timeslots open (64k) |
| 22:54.12 | Qwell[] | gsm or g729 would be far better |
| 22:54.22 | JT | ilbc :) |
| 22:54.28 | Qwell[] | lpc10 FTW |
| 22:54.45 | file | lpc10 is *the* best codec |
| 22:54.53 | type0 | doesnt 729 need a license? |
| 22:54.57 | Qwell[] | type0: yes |
| 22:54.57 | JT | yes |
| 22:55.08 | Qwell[] | but, if you're limited...what other choice do you really have? |
| 22:55.09 | type0 | what's better about lpc? |
| 22:55.11 | Qwell[] | That's kinda the point of it |
| 22:55.12 | [TK]D-Fender | file : Domo friggen' Arigato! |
| 22:55.27 | type0 | this is for a project traveling over a governemtn microwave link |
| 22:55.32 | type0 | I really dont care about cost.. within reason |
| 22:55.42 | Qwell[] | then use g729 |
| 22:55.47 | [TK]D-Fender | type0 : the LPC10 is for you. Its ALSO half-baked..... |
| 22:55.53 | JT | g.729 isn't expensive |
| 22:56.27 | Defend | cheaper then a week worth of smokes thats for sure |
| 22:56.28 | Defend | lol |
| 22:57.01 | dlynes_laptop | [TK]D-Fender: even then, hte beta didn't come out until September or so, didn't it? |
| 22:57.19 | type0 | so if I have 320k available |
| 22:57.25 | type0 | what's the max calls I could have concurrent? |
| 22:57.40 | fetcher | counting the packet headers, iLBC uses slightly less bandwidth than G.729 for about the same quality. |
| 22:58.15 | dlynes_laptop | beautiful |
| 22:58.24 | fetcher | About 22kb/s vs. 25kb. No need for licenses either, but there usually has to be an Asterisk box on either end to transcode |
| 22:58.34 | dlynes_laptop | ftp.digium.com is down...you'll need to use ftp2.digium.com if you want to access the files |
| 22:58.42 | [TK]D-Fender | dlynes_laptop : "when its ready". All other claims are vapourware :) |
| 22:58.48 | *** join/#asterisk soo-hick (n=sinan@ip-81-1-98-55.cust.homechoice.net) |
| 22:58.54 | soo-hick | hello |
| 22:59.05 | generalhan | Anyone know if you can make changes to the configuration of a 7960 via a telnet connection? ive been through almost all of cisco's docs and all i can find are logging/debugging commands. i just want to adjust a setting ... with out driving 2 hours out to the location ! |
| 22:59.25 | type0 | my mode of thinking was to get some wireless sip phones and use a 22dbi 5.8ghz antenna to beam the signal off a mountain 15 miles into a city.. just have them connect to the asterisk box ontop of the mountain with a 'wifi' sip phone.. and ba-dow.. dialtone |
| 23:00.12 | type0 | i was redundant with the sipphone part.. sorry |
| 23:00.47 | *** part/#asterisk dahunter3 (n=dahunter@pool-71-110-4-30.lsanca.dsl-w.verizon.net) |
| 23:01.56 | type0 | anyone use asterisk in a scenario like that? |
| 23:04.18 | JT | wifi sip phones suck |
| 23:04.34 | type0 | really? |
| 23:04.37 | JT | yes |
| 23:04.41 | JT | they're all rubbish |
| 23:04.46 | type0 | well.. give me an alternative then? |
| 23:04.47 | type0 | heh |
| 23:04.52 | type0 | cat5 isnt an option |
| 23:04.57 | JT | also it's a fundamental problem with the tehnology |
| 23:05.01 | *** join/#asterisk Dimik_ (n=Dimik@unaffiliated/dimik) |
| 23:05.13 | JT | walking around with wireless network stuff gives you variable lag, packet loss and jitter |
| 23:05.19 | JT | normal cordless phone |
| 23:05.23 | type0 | tin cans dont stretch 15 miles, that ive seen |
| 23:05.31 | type0 | a cordless phone with an ATA? |
| 23:05.39 | JT | type0: stable point to point wireless links are ok |
| 23:05.48 | JT | not moving around |
| 23:05.53 | JT | yes |
| 23:06.37 | type0 | so if its the technology.. a sip or iax (if they even make one) wireless phone wouldnt matter |
| 23:06.49 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:07.41 | JT | correct |
| 23:07.51 | JT | iax hard phones don't really exist anyway |
| 23:08.20 | *** join/#asterisk De_Mon (n=de_mon@fl-76-4-98-162.dhcp.embarqhsd.net) |
| 23:09.55 | type0 | hmm. what about if it has a REALLY high gain antenna? |
| 23:10.25 | JT | sorry |
| 23:10.31 | JT | are you talking about point to point |
| 23:10.44 | JT | or walking around with a high gain antenna on your wireless phone? |
| 23:11.00 | type0 | the idea is.. point to point wireless to an access point on the ground at the other side |
| 23:11.06 | type0 | so its like |
| 23:11.32 | type0 | microwave -----wireless---- access point --- access point (high gain antenna) --- sip phone |
| 23:12.05 | spaceinvader | you can get budget FXO's but not FXS's :( |
| 23:12.13 | Qwell[] | s/budget/cheap/ |
| 23:12.30 | type0 | I have a shitload of FXS and FXO cards in my mux |
| 23:12.40 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 23:12.41 | JT | type0: that should work if the link is stable, and not too high latency |
| 23:12.53 | JT | type0: lo/no packet loss, no/low jitter |
| 23:13.00 | type0 | over the microwave link, im getting 100ms pings |
| 23:13.00 | Qwell[] | over microwave? ha |
| 23:13.01 | spaceinvader | type0: do you have a shitload of money too then? |
| 23:13.06 | Qwell[] | ~wglwat |
| 23:13.16 | jbot | i guess wglwat is well, good luck with all that |
| 23:13.21 | type0 | this is a project for a 3rd generation radar site for the US airforce |
| 23:13.43 | JT | if you have tdm timeslots, use them |
| 23:13.53 | JT | don't fart around with voip if you can avoid it |
| 23:13.57 | [[blah]asfd | can anyone tell me the keystrokes on a polycom 301 to reset the phone to factory if it gets locked up. |
| 23:14.54 | type0 | the only problem with that is.. I want to avoid using the goverment pbx on the end of the microwave link |
| 23:15.04 | xo8ox | guys when we call in to our * box the sounds is choppy ?! |
| 23:15.17 | xo8ox | when we dial localy the sound is just fine |
| 23:15.18 | type0 | I was just going to add another RLB (radio lan bridge) to the coastcom and assign some timeslots to that |
| 23:15.39 | xo8ox | and we r using voicepulse |
| 23:15.40 | JT | type0: err so why can't you send the timeslots to asterisk? |
| 23:16.07 | type0 | I can send the timeslots to asterisk.. off the mux |
| 23:16.13 | type0 | I cant straight off the microwave equipment |
| 23:16.15 | JT | ok |
| 23:16.20 | JT | so what's the problem |
| 23:16.35 | type0 | 1. can i use wireless sip phones effectively |
| 23:17.08 | JT | yes but i've already told you they're rubbish |
| 23:17.16 | JT | use a conventional cordless phone |
| 23:17.35 | type0 | alright |
| 23:17.47 | type0 | i guess we'll have to get another generator for the site then |
| 23:17.52 | type0 | since there's no power in the city either |
| 23:18.04 | type0 | do they make wireless ATA's? |
| 23:18.15 | JT | don't think so |
| 23:18.29 | mercestes | no. Why don't you just use cell phones if you want a wireless solution? |
| 23:18.34 | mercestes | Nextel or some nonsense like that? |
| 23:18.41 | type0 | there's no cellular service either |
| 23:18.50 | mercestes | Oh. |
| 23:18.52 | type0 | this city is connected to the world via US postal service and satellite phone |
| 23:19.01 | type0 | there is NOTHING else. |
| 23:19.01 | Qwell[] | how are you going to use a wifi phone over microwave, exactly? |
| 23:19.11 | mercestes | ok, if you set up enough wireless repeaters in bridge mode and you *flood* that area in wireless signals, you can get a wifi phone to work. |
| 23:19.14 | orlock | urgh.. slept in |
| 23:19.16 | type0 | microwave brings in the IP based traffic over an RLB |
| 23:19.27 | type0 | then connect the RLB with the assigned timeslots to a wireless access point |
| 23:19.30 | mercestes | but you take however far you can get with a laptop at full signal, and half that distance...that's about as far as a wifi phone will get you. |
| 23:19.35 | Qwell[] | and how are you going to power the AP? |
| 23:19.39 | mercestes | about 10 feet from a standard consumer Linksys router. |
| 23:19.43 | type0 | generator |
| 23:20.16 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 23:20.26 | *** join/#asterisk echosyp (n=echosyp@wsip-70-183-82-175.dl.dl.cox.net) |
| 23:20.37 | JT | type0: so my recommendation would be to run tdm voice over the microwave link if you can |
| 23:21.36 | echosyp | assuming i don't have an analog to digital converter how could i go about setting up asterisk |
| 23:21.47 | type0 | so basically come out of the FXS card on the city end and just connect the asterisk box to the that? |
| 23:21.53 | echosyp | with a VoIP service subscription |
| 23:22.20 | JT | type0: sorry, is the city end this site or somewhere else? |
| 23:22.38 | vlt | Hmmm, can I get something like ${CALLERS_IN_QUEUE} in a dialplan? |
| 23:22.54 | echosyp | i guess what im asking is, how would the computer connect to the VoIP box my ISP gives me |
| 23:23.29 | mercestes | echosyp: Via broadband internet presumably. Or some other broadband data connection. |
| 23:23.46 | mercestes | echosyp: Wait. a .."voip box?" |
| 23:23.51 | mercestes | echosyp: Is that an ATA device? |
| 23:23.52 | Qwell[] | mercestes: an ATA |
| 23:23.54 | type0 | hang tight |
| 23:24.02 | mercestes | oh |
| 23:24.12 | mercestes | why would you hook your computer to a voip box your ITSP gave you? |
| 23:24.21 | echosyp | well, my experience is the modem my isp gives me has rj-11 ports on it |
| 23:24.33 | echosyp | im n00b, obviously |
| 23:24.39 | mercestes | obviously. |
| 23:24.41 | echosyp | heh |
| 23:24.43 | JT | it's an ATA, not a modem :) |
| 23:24.44 | mercestes | :) |
| 23:24.49 | mercestes | it's an itsp not an isp |
| 23:24.54 | Qwell[] | JT: no, it's probably the modem |
| 23:25.15 | echosyp | its a broadband modem too |
| 23:25.21 | echosyp | avis or something like that |
| 23:25.24 | mercestes | Well, the answer to your question is you provide the "voip box" an FXO port and you plug that into your rj11 where you normally plug a phone. |
| 23:25.25 | Qwell[] | echosyp: Do you mean you get your "voip subscription" from Cox? |
| 23:25.30 | mercestes | and then you run an fxs port to your telephone device. |
| 23:25.40 | echosyp | yes |
| 23:25.43 | echosyp | cox |
| 23:25.43 | Qwell[] | ugh |
| 23:25.45 | Qwell[] | cancel it :p |
| 23:25.49 | Qwell[] | get a real provider |
| 23:26.05 | mercestes | or do what Qwell said and ignore my smart-@$$'d answer because...it will only cause pain. :) |
| 23:26.06 | Qwell[] | You're probably paying about 5x more than you should be |
| 23:26.07 | echosyp | seriously? well i was thinking id just get an analog line and get an converter |
| 23:26.22 | Qwell[] | What's it costing, $40+/month for just voice? |
| 23:26.25 | mercestes | oh well, g'nite. |
| 23:26.27 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 23:26.45 | echosyp | i don't have the service anymore, im just brainstorming here |
| 23:27.02 | Qwell[] | well, find a real ITSP, and just connect directly to them with asterisk |
| 23:27.28 | echosyp | figure i could get a converter for $100-200 and pay $10/month for an analog line |
| 23:28.23 | Qwell[] | That would be ideal |
| 23:28.23 | echosyp | im calling it a converter, is there a real term for it |
| 23:28.23 | echosyp | FXO? |
| 23:28.23 | Qwell[] | FXO for that, yes |
| 23:28.23 | JT | if it's a hardware box it's an ATA |
| 23:28.27 | echosyp | oh |
| 23:28.31 | echosyp | then im talking ATA |
| 23:28.34 | echosyp | i guess |
| 23:30.26 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 23:31.13 | echosyp | and this ATA will let my computer talk to an analog house line? |
| 23:31.24 | echosyp | give me something to read so i can stop asking stupid questions |
| 23:31.30 | JT | yes an FXO will talk to analogue house line |
| 23:31.31 | JT | ~thebook |
| 23:31.33 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:31.33 | Carp1 | as long as I have port 5060 fowarded to my * box, a remote IP phone should be able to connect, correct? |
| 23:31.50 | JT | Carp1: no |
| 23:32.00 | JT | Carp1: you must forward all rtp ports too |
| 23:33.36 | vlt | How can I achieve the following: exten => s,1,IF( $[callers_in_queue > 0] ? Queue() : Goto(dialZap,1) ) // exten => dialZAP,1,Dial(Zap/xy) // exten => dialZAP,102,Queue() |
| 23:33.38 | vlt | to achieve |
| 23:33.46 | vlt | (If anyone can guess what I mean) |
| 23:34.03 | vlt | s/to achieve// |
| 23:34.20 | *** join/#asterisk _paulos_ (n=paulos@201-27-153-150.dsl.telesp.net.br) |
| 23:34.35 | *** join/#asterisk CrazyTux (n=CrazyTux@70.142.27.21) |
| 23:34.41 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 23:35.06 | Bobthehunter | anyone have a viable solution for SER accounting ? |
| 23:36.08 | *** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 23:36.23 | *** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca) |
| 23:36.44 | JerJer | Bobthehunter: no but give me a week or so |
| 23:36.52 | Bobthehunter | lo |
| 23:36.56 | Bobthehunter | lwhat are you gonna do ? |
| 23:37.09 | Bobthehunter | hold CHARGEBACKING a ebay fraud again |
| 23:37.23 | *** join/#asterisk ManxPower (n=manxpowe@71-8-61-102.dhcp.leds.al.charter.com) |
| 23:37.23 | JerJer | either extend rtpproxy or write my own media proxy application |
| 23:37.45 | Bobthehunter | hmm |
| 23:37.49 | Bobthehunter | why rtpproxy |
| 23:38.16 | JerJer | I use SER+Asterisk in production now - its nice but seems overkill for straight call term/orig |
| 23:38.38 | JerJer | cuz Andreas won't support mediaproxy |
| 23:38.55 | JerJer | and osas suggested I take a look at rtpproxy |
| 23:39.18 | Bobthehunter | welll thing is accoutning sucks |
| 23:39.52 | Bobthehunter | one could pull the plug or simply block all BYE packets going out and bang |
| 23:40.01 | JerJer | yes - exactly |
| 23:40.08 | [TK]D-Fender | vlt : Break that crazy shit up.... and stop trying to do everything on 1 line! |
| 23:40.29 | JerJer | one could use SST but i really dont' want to trust a timer |
| 23:40.47 | JerJer | i would rather know that there is no more media flowing and the call actually came down |
| 23:40.50 | Bobthehunter | watswouldsst do ? |
| 23:40.57 | JerJer | session timers |
| 23:41.15 | JerJer | its a newish part of the SIP RFC |
| 23:41.32 | Bobthehunter | also how u use modparam("acc", "dst_leg_avp_id", 111) |
| 23:41.44 | Bobthehunter | "fr_inv_timer", 90) |
| 23:41.46 | JerJer | i don't do ser acc at all |
| 23:41.47 | Bobthehunter | got does also |
| 23:41.50 | JerJer | its a joke |
| 23:41.54 | Bobthehunter | yes its a joke |
| 23:41.58 | Bobthehunter | lets say i GW to L3 |
| 23:42.05 | Bobthehunter | htf am i supposed to know length |
| 23:42.31 | JerJer | the mediaproxy app could provide it |
| 23:42.32 | Bobthehunter | so i need asterisk as last hop always.. then whats the use of SER lol |
| 23:42.38 | JerJer | as long as you are proxying that media |
| 23:42.38 | Bobthehunter | and SER+asterisk aint compat CDR's |
| 23:43.02 | JerJer | asterisk can be made to produce decent CDRs - but that take custom code |
| 23:43.03 | Bobthehunter | again why would i want that.. why not offload the media/bwidth to term box |
| 23:43.05 | JerJer | +s |
| 23:43.28 | Bobthehunter | then OSP form transnexyus could manage the CDR's |
| 23:43.38 | Bobthehunter | BUT it still not really managable as i see it |
| 23:48.32 | [TK]D-Fender | JerJer : SST? Are you really in that much of a hurry? |
| 23:49.33 | vlt | [TK]D-Fender: Do you mean one line in the dialplan or here in the channel? (Didn't want to flood here) |
| 23:49.46 | vlt | [TK]D-Fender: I'll try to use words: |
| 23:50.15 | vlt | [TK]D-Fender: When there are callers in the queue -> join it. |
| 23:50.34 | vlt | [TK]D-Fender: If not -> Dial |
| 23:50.37 | JerJer | no no no - I prolly won't ever run SST on any of my systems |
| 23:50.55 | JerJer | plus we have private routes to our carriers, so we have to proxy anyways |
| 23:51.26 | JerJer | thus i need to either hook accounting into rtpproxy or write a better mediaproxy app and properly document and support it |
| 23:51.27 | vlt | [TK]D-Fender: If Dial gets BUSY (102?) -> join queue |
| 23:52.02 | vlt | [TK]D-Fender: If Dial was successfull or UNAVAILABLE -> hangup |
| 23:52.08 | vlt | [TK]D-Fender: EOF |
| 23:53.43 | ManxPower | vlt: priority jumping is no longer supported in 1.4 Use the value of DIALSTATUS |
| 23:54.07 | Bobthehunter | jerjer how you account for L3 atm ? |
| 23:54.16 | [TK]D-Fender | vlt : break that up into a step-by-step dialplan section. |
| 23:54.17 | JerJer | SER+Asterisk |
| 23:54.25 | [TK]D-Fender | vlt : gotoif = your friend |
| 23:54.42 | DocHolliday | [TK]D-Fender, my phones are arriving tomorrow :P |
| 23:54.47 | Bobthehunter | so sEr talks to ASTERISK (wich v?) that goes to L3 ? |
| 23:54.52 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:54.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:54.53 | Bobthehunter | so SER LB's the AST ? |
| 23:54.58 | [TK]D-Fender | DocHolliday : Congrats, whats coming in? |
| 23:55.04 | JerJer | 0.99 |
| 23:55.11 | JerJer | correct |
| 23:55.16 | Bobthehunter | lol is 0.99 a version ? lol |
| 23:55.18 | DocHolliday | 3 Cisco 7941Gs + Cisco 186 ATA I1 |
| 23:55.38 | JerJer | i run 'SER' in production |
| 23:56.17 | JerJer | been playing with OpenSER for quite a while here in the batcave and have deployed a handful of smallish systems using v.1.1.1 |
| 23:56.30 | elriah | DocHolliday: How's the firmware battle going? |
| 23:56.54 | DocHolliday | elriah, appreciated your help earlier.. everything is going smoothly |
| 23:57.05 | DocHolliday | so much so i decided to order more =P |
| 23:57.15 | [TK]D-Fender | DocHolliday : elriah here has just been our altest guinea pig, he'll be able to help you out with them if you're lucky (or NOT) |
| 23:57.25 | [TK]D-Fender | latest* |
| 23:57.37 | DocHolliday | [TK]D-Fender, haha.. well mine is really stable.. |
| 23:57.38 | elriah | DocHolliday: Anything worthwhile to note? i.e., did you find any magic pixie dust to make it work with NAT? |
| 23:57.59 | elriah | [TK]D-Fender: lol, you know we solved that shit by just building IPSEC tunnels with the lowest settings and no encryption. |
| 23:58.07 | elriah | [TK]D-Fender: Works great, now. |
| 23:58.11 | DocHolliday | elriah, bah it worked out of the box.. phone and asterisk are on the same subnet |
| 23:58.13 | [TK]D-Fender | Darn.... Lymers left before he could fully incriminate himself!@!@!@ |
| 23:58.21 | JerJer | openvpn SSL tunnels work nicely too |
| 23:58.50 | [TK]D-Fender | elriah : And have you worked a way out of that clause making claims on your first-born? ;) |