| 00:00.05 | mishehu | has anybody tried today's cvs head and can verify if there is somethign wrong with queue() ? |
| 00:00.08 | blankman | It was listed as having work done on it, but I don't see changes. |
| 00:00.16 | iCEBrkr | I just recompiled asterisk and new kernel |
| 00:00.16 | iCEBrkr | Before this, my shit worked just fine. |
| 00:00.36 | blankman | RoyK, I understand your point. The was why I was asking how your "server" knows that they don't have the bandwidth. |
| 00:00.43 | Brixius | RoyK: Sorry man, I didn't see the coment's in the bug that Mark Threw out the changes, I thought you were bitching because it didn't do what you wanted and you wanted someone to code it to work the way you thought it should. |
| 00:00.58 | Moc | what up ? |
| 00:01.12 | blankman | We have the same issue with softphones vs. iaxy. the stupid iaxys only support adpcm and ulaw. |
| 00:01.15 | Moc | anyone got feature request for MeetMe ? |
| 00:01.27 | *** join/#asterisk UPM-eduardo (UPMeduardo@80.102.40.2) |
| 00:01.30 | RoyK | blankman, the clients are configured individually, centrally, based on customer's bandwidth's or complains |
| 00:01.46 | BigCanOfTuna | OK, so when I do a sip show peers, it indicates that 2205 is unreachable...what might cause this? I turned off iptables in the box that is running that sip phone. |
| 00:01.50 | blankman | boss is back ;-) |
| 00:02.46 | BigCanOfTuna | Anyone? |
| 00:02.47 | Primer | Moc: make meetme list X show the CALLERID...? |
| 00:02.51 | helvetico | RoyK: By all means the CPE should say "I don't like G.711 and I want G.729". The ATA is responsible for this and not * except you remove G.711 in totalis. |
| 00:03.05 | blankman | Look, all I am saying is that on your pstn server send them to a box that only has g.729 and g.726 enabled. On the flip side, have them login to a * server that knows how to forward them based on the pref. you did before ... this way you avoid the problem. It isn't like you are realtime making the change right? |
| 00:03.27 | blankman | The SIP will handle it through the reinvit. |
| 00:03.33 | *** join/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 00:03.45 | blankman | royk, Least I think it should, but as I said I don't do SIP ... only IAX :-) |
| 00:04.18 | iCEBrkr | anyone anyone? |
| 00:04.27 | iCEBrkr | Jan 3 19:02:19 ERROR[5596]: chan_zap.c:5856 mkintf: Unable to get parameters |
| 00:04.37 | iCEBrkr | Wtf is mkintf? no man page, nothing in locate |
| 00:04.39 | *** join/#asterisk bprice20 (~brandon@alb-69-200-30-85.nycap.rr.com) |
| 00:04.46 | bprice20 | yo yo |
| 00:05.33 | blankman | Hey bkw_, do you know if anyone is hitting up the vmail issues/features yet? We heard talk and it was on Marks list at Astricon. |
| 00:06.34 | blitzrage | blankman: not sure that voicemail has gone anywhere yet... I don't think it'll move until this new group that wants to manage the features gets setup |
| 00:07.16 | D1ng0 | uhh crap |
| 00:07.51 | D1ng0 | anyone in here running asterisk on a debian 64bit AMD ? |
| 00:08.15 | *** join/#asterisk Xenesis (Xenesis@ACCEA7F3.ipt.aol.com) |
| 00:08.40 | *** part/#asterisk Xenesis (Xenesis@ACCEA7F3.ipt.aol.com) |
| 00:09.30 | blankman | blitzrage, okay ... that makes me sad ... did we get an eta on the "new" group? |
| 00:10.25 | *** join/#asterisk bkw_ (~brian@65.38.28.11) |
| 00:10.25 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:11.11 | blankman | blitzrage, do you know if the encrypted channels stuff is coming along yet, or is that held up in waiting for the new group too? |
| 00:12.30 | Brixius | Does anyone know of a way/app/addon to asterisk that will call a sip/iax2/h323 channel the first time it registers to play a welcome greeting to the user? |
| 00:12.42 | blitzrage | blankman: the new group discussion has been happening on the mailing list since last night (about 2 AM EST), so its only a few hours into planning... but things seem to be moving along at a pretty decent pace. |
| 00:13.03 | blitzrage | blankman: as for the encrypted channels thing... I think you'll get there faster if you run it over a SSH tunnel :) |
| 00:14.17 | stevekstevek | You can't do that.. (ssh tunnel). But you can use openvpn. |
| 00:14.30 | mquin | ssh is going to stink for voip |
| 00:14.39 | mquin | openvpn works thouhg |
| 00:14.46 | blitzrage | I thought I read someone did it, interesting |
| 00:14.50 | *** join/#asterisk ZX81 (matt@222-152-141-153.jetstream.xtra.co.nz) |
| 00:14.51 | blankman | blitzrage, yeah, but the over head is killing me :-( We only run iax and it is still to chopy to make it happen :-( Also, it won't help for the new ATA's that we want to make. We figured it was worth just putting in a simple signaling encryptor ... but I haven't the time to do the code. I have most of the arch. for it, just not the time to make :-( |
| 00:15.13 | Jameno123 | so you say the gsm codec == shit, and ulaw is the best? |
| 00:15.19 | RoyK | blankman, we only have ATAs that can do SIP so I don't have a choice |
| 00:15.20 | Strom_TM | yes yes yes |
| 00:15.27 | Strom_TM | ulaw == good, gsm == bad :) |
| 00:15.30 | *** join/#asterisk Markie (Mark@208.247.106.165) |
| 00:15.36 | blitzrage | blankman: hrmmm... I haven't heard any mention about encrypted channels right in Asterisk for a while now... |
| 00:15.37 | Jameno123 | why does asterisk default to GSM ? |
| 00:15.43 | Jameno123 | if thats so... |
| 00:15.48 | stevekstevek | Mark wrote something.. |
| 00:15.49 | blitzrage | ulaw is not compressed |
| 00:15.54 | blitzrage | so it makes things a lot easier |
| 00:16.00 | mquin | for certain values of good and bad |
| 00:16.01 | Strom_TM | Jameno123, someone doesn't have ears, perhaps :) |
| 00:16.12 | stevekstevek | for encrypted iax, but it's not complete. |
| 00:16.14 | Jameno123 | Strom_TM, heh i hear that :) |
| 00:16.16 | mquin | if you're trying to conserve bandwidth, gsm is better then ulaw |
| 00:16.35 | Jameno123 | mquin, though its minimual gain? |
| 00:16.44 | Jameno123 | like a few kbp/s? right? |
| 00:16.49 | Jameno123 | (per call) |
| 00:16.50 | blankman | stevekstevek, umm lost you on your answer he 'wrote' as in a paper or code? |
| 00:16.51 | Markie | no, not a few. quite a bit. |
| 00:16.52 | Markie | about 5x |
| 00:16.59 | Jameno123 | ouch ;) |
| 00:17.01 | blitzrage | ulaw = 64kbps plus IP (and whatever else) overhead |
| 00:17.01 | Markie | i think it's 14K versus 64K |
| 00:17.01 | stevekstevek | code. In CVS. Not working, AFAIK. |
| 00:17.18 | Jameno123 | now, for 'transcoding' |
| 00:17.23 | Jameno123 | i guess all the *.gsm sounds |
| 00:17.29 | stevekstevek | (re ulaw vs GSM): With overhead, for a single non-trunked call, it's more like 30kbps vs 80kbps |
| 00:17.33 | Jameno123 | converting to ulaw, would be transcoding? |
| 00:17.36 | RoyK | Markie, around 80kbps with 20ms slicing. 126kbps with 10ms slicing |
| 00:18.14 | Markie | RoyK: wow...thats a lot of overhead..why the overhead, wasnt that the point of using UDP?? |
| 00:18.15 | Jameno123 | 10 to 1 half adoze of another... ok i got the point :) |
| 00:18.34 | RoyK | Markie, IP, UDP and RTP? |
| 00:18.48 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:18.51 | Strom_TM | Jameno123, the expression is "six of one, half a dozen of the other" - what you said makes no sense unless one dozen is now 20 |
| 00:18.56 | stevekstevek | UDP overhead is small. IP overhead is big :) |
| 00:19.10 | RoyK | Markie, you can't do it with TCP, too much latency |
| 00:19.10 | JamesDotCom | hrmm... when i use ValetParkCall() it seems to park the call fine, but then when the phone is put back on, gives a "blah got tired of being Valet Parked" and redials the phone that just parked the call... any ideas anyone? |
| 00:19.52 | stevekstevek | that's a lot of latency.. |
| 00:19.55 | D1ng0 | anyone in here running asterisk on a debian 64bit AMD ? compiled in 64bit ? |
| 00:19.58 | blankman | stevekstevek, splain me that ... IP over IAX2 over HTTP? |
| 00:20.00 | Jameno123 | Strom_TM, ya ya.. long day @ work :) |
| 00:20.05 | stevekstevek | I do, really. |
| 00:20.09 | blankman | D1ng0, no i. |
| 00:20.23 | RoyK | D1ng0, a few have done that. works fine afaihh |
| 00:20.24 | Brixius | stevekstevek: udp still uses ip, it should actually be called udpip and tcpip |
| 00:20.26 | iCEBrkr | 10 to 1 have a dozen of the other? I thought it was 6 of 1 |
| 00:20.33 | stevekstevek | It could make very little latency, but then I'd make 50 HTTP requests/sec. |
| 00:20.36 | Markie | it is called tcp/ip |
| 00:20.37 | Markie | silly |
| 00:20.39 | D1ng0 | im hvaing compile issues with both zaptel and * |
| 00:20.51 | iCEBrkr | D1ng0: yea yea yea, me too. :) |
| 00:20.56 | stevekstevek | so, I actually make only 2 HTTP requests/sec, and get about 500ms jitter, total latency under 1 sec.. |
| 00:21.04 | RoyK | Markie, it's called IPv4 |
| 00:21.06 | D1ng0 | i get WARNING: Error inserting zaptel (/lib/modules/2.6.9-10-amd64-k8/misc/zaptel.ko): Invalid module format |
| 00:21.11 | RoyK | Markie, and TCP, and UDP etc |
| 00:21.21 | blankman | You using http1.1? |
| 00:21.21 | RoyK | Markie, different protocols |
| 00:21.23 | iCEBrkr | D1ng0: oh, but you're having different issues, I see. |
| 00:21.29 | blankman | stevekstevek, are you using http1.1 |
| 00:21.32 | Markie | thanks..good to know. |
| 00:21.34 | Markie | heh |
| 00:21.39 | D1ng0 | iCEBrkr, what issue you having ? |
| 00:21.46 | stevekstevek | blankman: do you mean to we use keep-alive? If it's available.. |
| 00:21.47 | RoyK | VoIP over avian carriers :) |
| 00:22.04 | iCEBrkr | D1ng0: Asterisk can't seem to use the zaptel module. |
| 00:22.13 | iCEBrkr | D1ng0: I found my same problem on pastebin |
| 00:22.28 | iCEBrkr | http://pastebin.ca/2967 |
| 00:22.30 | RoyK | n8 |
| 00:22.34 | D1ng0 | iCEBrkr, you using the debian 64bit from pure64 debian ? |
| 00:22.42 | iCEBrkr | P U. |
| 00:22.46 | iCEBrkr | I stay away from Debian |
| 00:22.50 | iCEBrkr | :) |
| 00:22.54 | RoyK | iCEBrkr, wot? |
| 00:23.01 | RoyK | debian is the rule! |
| 00:23.09 | RoyK | all others are exceptions |
| 00:23.13 | blankman | hummm ... well I am off to re-write voicemail I guess ... make sure that the "new" group knows that people have been waiting for stuff or a while now, so they should wait for ever ;-) |
| 00:23.24 | iCEBrkr | hrrm fixed. |
| 00:23.30 | iCEBrkr | odd. |
| 00:23.33 | *** part/#asterisk RoyK (~roy@110.80-203-29.nextgentel.com) |
| 00:23.35 | iCEBrkr | Maybe I had a dirty source directory |
| 00:23.41 | pbd | I have succeeded in installing linux from scratch- new partitions, new kernel, install appropriate stuff, and boot. |
| 00:24.13 | blankman | actually, before I do that, anyone think that pop is better than imap as a store? if so why? |
| 00:24.35 | D1ng0 | i need pure64 debian help |
| 00:24.36 | Jameno123 | pop == for dialup, imap for DSL/broadband. |
| 00:24.39 | stevekstevek | POP is not better, although it's simpler.. |
| 00:24.41 | helvetico | POP used to be good, until IMAP came to existance |
| 00:24.41 | Jameno123 | :) heh... |
| 00:24.51 | iCEBrkr | yay! it all works |
| 00:24.58 | Jameno123 | imap sorta sucks under low banwidth situations |
| 00:25.01 | blankman | I meant to act as the transport for the vmail application. |
| 00:25.17 | mquin | different strokes - pop is good for transferring mail, imap is good if you want to access a remote mailbox without downloading everything |
| 00:25.20 | blankman | I am going to re-write vmail to use a mail server as it's backend. |
| 00:25.46 | Jameno123 | blankman, ack, no :) |
| 00:25.49 | stevekstevek | blankman: sounds ugly to me. |
| 00:26.01 | helvetico | I think you are not using any of them. You are outblasting mails only with an SMTP engine |
| 00:26.02 | D1ng0 | no help here |
| 00:26.06 | Jameno123 | hehe email servers have enough load on them :) |
| 00:26.07 | stevekstevek | blankman: why?? |
| 00:26.13 | blankman | heheeheh ... it actually is quite elegant I think :)) |
| 00:26.34 | pbd | blankman- why not really hit it.. use the 'googlefs' driver. :) |
| 00:26.50 | pbd | ~googlefs |
| 00:26.59 | stevekstevek | What's wrong with any networked file system? |
| 00:27.04 | stevekstevek | Or a database? |
| 00:27.23 | pbd | Googlefs would be best of all worlds.. it uses your gmail account as a filesystem under linux. :) |
| 00:27.25 | stevekstevek | If those aren't suitable, I'd use WebDAV before trying to make it talk to a mail server.. |
| 00:27.26 | blankman | Cause it offerse the ability to have a very, very scalable system. The way I look at it, mailservers run store and process more transactions than anyother system created so far. |
| 00:27.42 | dant | why not just make use of unixodbc and store everything in an access database |
| 00:27.43 | BigCanOfTuna | What are some of the reasons my my SIP phone may be unreachable? It successfully registers with asterisk, however, I can't call it. |
| 00:27.44 | stevekstevek | Still WebDAV wins. |
| 00:28.02 | blankman | stevekstevek, file locks to start ;-) NFS big issue... smb ... mapping issues. iSCSI to expensive ... NAS same thing ... |
| 00:28.27 | stevekstevek | Okay, then why not WebDAV. |
| 00:28.29 | blankman | Thought about db ... still thinking about it, but it seemed easier and KIS to use pop or imap. |
| 00:28.49 | Jameno123 | well pop cant upload to the server can it? |
| 00:29.00 | mquin | Jameno123: that's correct |
| 00:29.05 | stevekstevek | It won't be simple to use POP or IMAP. I'm not sure you can write to POP, and Imap is very complicated. |
| 00:29.26 | Silik0n | top - 19:32:23 up 6:48, 4 users, load average: 9.97, 4.67, 2.34 |
| 00:29.30 | stevekstevek | But you still didn't say why you don't like WebDAV. |
| 00:29.32 | blankman | stevekstevek, ummmm webdav ... haven't looked into it enough to answer no, but it doesn't seem to be as tested as mail server... |
| 00:29.35 | bkw_ | bull shit |
| 00:29.37 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 00:29.37 | bkw_ | imap is easy |
| 00:29.40 | bkw_ | so is pop |
| 00:29.46 | blankman | Jameno123, nope, smtp only for mailservers. |
| 00:29.50 | Silik0n | imap + ssl is easy as hell |
| 00:29.52 | bkw_ | god jump on it with NFS and Maildir |
| 00:29.53 | BigCanOfTuna | Come on, some one can help me out....oh woe is me. |
| 00:30.01 | dant | bkw_, for a VM backend? |
| 00:30.10 | bkw_ | dant yes |
| 00:30.13 | Brixius | BigCanOfTuna: do you have a qualify= line in sip.conf |
| 00:30.16 | pbd | blankman.. although it pains me to say it.. hotmail uses webdav.. OutlookWebAccess.. |
| 00:30.27 | D1ng0 | bkw_, yo got any clue on * on debian pure64 ? |
| 00:30.39 | stevekstevek | bkw_: I don't think IMAP is easy, but using a maildir-derived store for VM solves "locking" problems so you can use NFS. |
| 00:30.41 | bkw_ | seeeeeeeeeeeeeeee |
| 00:30.50 | blankman | bkw_, you saying that to me? the bullshit stuff? ... or are you saying I am not nuts ;-) |
| 00:31.03 | BigCanOfTuna | Brixius: yes I do. 300. |
| 00:31.03 | stevekstevek | no, I think he was calling bullshit on me. |
| 00:31.25 | bkw_ | your not nuts... its a wise idea to mount the vm spool and use Maildir and just read and write base 64 encoded messages |
| 00:31.27 | bkw_ | we did it alrady |
| 00:31.29 | bkw_ | er already |
| 00:31.33 | Brixius | BigCanOfTuna: try remarking it out and then doing a sip reload |
| 00:31.33 | bkw_ | but people said it was fucking stupid |
| 00:31.53 | stevekstevek | and, I'm defending myself: Try to talk IMAP against Cyrus, and see how often it complains about some single-characted violations of RFC-2822 |
| 00:31.53 | blankman | who said? |
| 00:32.01 | dant | bkw_, I don't think people were saying using Maildir was stupid, but using POP3..... |
| 00:32.01 | stevekstevek | (or 2821, I guess) |
| 00:32.09 | *** join/#asterisk knight_ ([HrQ8MXnPR@voip.phunc.com) |
| 00:33.03 | bkw_ | just bypass all that bullshit |
| 00:33.03 | bkw_ | and use NFS |
| 00:33.03 | bkw_ | :P |
| 00:33.03 | knight_ | grrr, 500 internal server error for broadvoice |
| 00:33.03 | bkw_ | but I can and have done imap to Cyrus |
| 00:33.03 | blankman | I would, but it has locked every ten hours or so in testing... |
| 00:33.03 | bkw_ | via telnet |
| 00:33.03 | stevekstevek | that is what I said in the first place. because IMAP is _not_ simple :) |
| 00:33.03 | bkw_ | its a bitch |
| 00:33.03 | bkw_ | but you can do it |
| 00:33.03 | blankman | Not good for prod boxes. |
| 00:33.04 | Silik0n | i dont think pop would be that great for VM storage... its ass at best rught now |
| 00:33.06 | bkw_ | see we have this format_base64.c we wrote for asterisk |
| 00:33.11 | Silik0n | just for email |
| 00:33.13 | bkw_ | that writes RFC822 files |
| 00:33.19 | stevekstevek | :) see "it's a bitch" != simple in my book. I had a bitch of a time trying to convert mailboxes to mail cyrus happy :) |
| 00:33.20 | Jameno123 | blankman, soft mount your disks :) so they throw I/O Error, instead of bringing the box down |
| 00:33.21 | Jameno123 | heh |
| 00:33.21 | knight_ | <PROTECTED> |
| 00:33.23 | bkw_ | and you can read and write.. play and record them |
| 00:33.32 | bkw_ | and leave them in base 64 format |
| 00:33.36 | echinos | I am thinkin of stting up my * system with 1 fxo card, with 3 analog phones split from a sipura or IAXy, so 1 extension is used for all the analog phones. Any comments? Good/bad/prone to problems? |
| 00:33.40 | bkw_ | but nobody thought that was a good idea :P |
| 00:34.31 | BigCanOfTuna | Brixius: That did it, thanks! |
| 00:34.31 | Jameno123 | bkw_, id start to think only the 'base64' idea to be bad |
| 00:34.41 | Silik0n | knight_ stfu |
| 00:34.43 | tzanger | I think having VM as RFC822 in Maildir/ would be *awesome* |
| 00:34.58 | bkw_ | yes tzanger me too |
| 00:35.03 | tzanger | Silik0n: say I bet you win all the diplomacy awards |
| 00:35.03 | bkw_ | but nobody thinks that |
| 00:35.19 | blankman | bkw_, what did you do with your old code? |
| 00:35.20 | Silik0n | yeah pretty much |
| 00:35.30 | Silik0n | tzanger and it helps that i know knight_3 |
| 00:35.35 | tzanger | :-) |
| 00:35.51 | bkw_ | blankman its around here somewhere |
| 00:35.56 | blankman | in cvs? |
| 00:36.05 | Silik0n | besides i've never been accused of actually using tact |
| 00:36.06 | blankman | you remember the date ranges approx? |
| 00:36.17 | *** join/#asterisk gustavoz (~gustavoz@gustavoz.developer.gentoo) |
| 00:36.20 | bonbon-home | latest cvs asterisk-addons doesn't compile by the way |
| 00:36.29 | PTG123 | actually maildir format would be smart, no mutex locking, easy maintence, etc |
| 00:37.04 | PTG123 | was looking into doing that in something else of mine recently |
| 00:37.11 | Jameno123 | Maildir is excellent storage format, heh, though base64 could bring a heavy loaded box to a halt if the file sizes were big enough. |
| 00:37.43 | mquin | I can see how storing voicemail on an imap server would be attractive - voicemail messages show up as emails with any metadata (time, caller id etc) in text format and the audio as an mp3/wave attachment |
| 00:37.50 | blankman | I actaully was thinking about that issue, so I was thinking about having the system work like a web service :-) |
| 00:37.50 | BigCanOfTuna | Brixius: I just read the documentation on qualify....so commenting it out works, but doesn't explain why the SIP phone is unreachable when qualified is set to timeout number. |
| 00:37.50 | tzanger | ok... 8400 *incoming* calls in 3 months |
| 00:38.05 | Jameno123 | voicemail doing 3 messages a minute @ 2MB/peice :) == death. |
| 00:38.08 | mquin | and by taking that approach your VM system doesn't have to worry about the storage issues - it just lets the ail server deal with it |
| 00:38.26 | mquin | (this may, however, piss off your email admin) |
| 00:38.32 | Brixius | BigCanOfTuna: what's your ping time from the * box to the phone? |
| 00:38.43 | syslod | Finally chan_sip seems to be working again :) |
| 00:39.07 | BigCanOfTuna | Brixius: .288ms. |
| 00:39.17 | blankman | mquin, that is the point. There is only one place to "manage" the entire store. That at least is what we decide made it "production" friendly ;-) |
| 00:39.18 | mishehu | bah. |
| 00:39.47 | mishehu | anybody try out g711ulaw-over-SIP/RTP->speex-over-IAX ? |
| 00:40.03 | BigCanOfTuna | Brixius: I misread the documentation...I suppose that could be why. I interpreted it to be seconds...not milliseconds. |
| 00:40.09 | mishehu | audio cuts out for me after about a second and a half |
| 00:40.59 | *** join/#asterisk Curren (~curren@ip242.174.235.64.susc.suscom.net) |
| 00:41.02 | Jameno123 | if your 'going' to use a mailserver, use IMAP ;) that way you can upload/download w/o the neccessary overhead of an SMTP session. |
| 00:41.28 | Curren | anyone have experience with one way voice on an IAXy |
| 00:41.39 | tzanger | hmm |
| 00:41.58 | Jameno123 | unncessary* |
| 00:42.06 | Nugget | one way voice seems to always be "nat problems" when it comes up in here. |
| 00:42.33 | *** join/#asterisk Madiman (~patdelan@66-188-106-58.mad.wi.charter.com) |
| 00:42.42 | *** join/#asterisk greg_work (~greg@ik-dynamic-66-102-67-53.kingston.net) |
| 00:42.51 | Curren | is there anything you suggest I look for... have opened ports still doesn't seem to help |
| 00:43.08 | BigCanOfTuna | Brixius: Actually, that doesn't make sense, I set up the qualify=5000, and it is still unreachable. |
| 00:43.12 | Curren | beginning to think its bad, and needs replaced |
| 00:43.29 | tzanger | 8400 incoming calls in 3 months averaging 217 seconds per call. 30621 minutes over that 3 month period, averaging to 10207 minutes per month INCOMING only |
| 00:43.32 | blankman | Well ... thanks for the feed back. |
| 00:43.47 | Brixius | BigCanOfTuna: when it say's unreachable, what's the time it's saying. |
| 00:44.43 | Curren | also the phone connected to the iaxy doesn't get a dial tone... just static |
| 00:44.43 | BigCanOfTuna | Brixius: It doesn't...it just says UNREACHABLE (according to sip show peers)....the one that is reachable is 15ms. |
| 00:44.47 | Madiman | I'm an IRC newbie.... what's the correct protocol for asking a question? |
| 00:44.57 | JamesDotCom | you ask it |
| 00:44.58 | mishehu | ooh I wish I could remember who it was I was talking about speex with about a week or so ago... |
| 00:45.01 | JamesDotCom | and dont ask how to ask it ;( |
| 00:45.06 | Madiman | LOL |
| 00:45.14 | mishehu | Madiman: just asking, giving as much useful detail as possible. |
| 00:45.25 | Brixius | Madiman, you ask it, and wait for someone to answer, don't ask lots of times in a row if you don't get an immediate response. |
| 00:45.29 | tzanger | Madiman: just ask, tell us what you've tried and what you're getting and if it's more than 4 lines, use www.patebin.ca to dump the info to and give us the URL it gives back |
| 00:45.44 | mishehu | not "hi, my asterisk is borked", but "I did x, y, and z, and now it such-and-such a module won't work..." |
| 00:45.57 | mishehu | yes, pastebin is a friend. |
| 00:46.03 | tzanger | that goddamned such-and-such module, I always have trouble with it |
| 00:46.09 | Madiman | I just built up a small asterisk server based on the online docs, JUst SIP for now, I'm waiting for the rest of the hardware... It all seems to work except for Music on hold |
| 00:46.15 | tzanger | It's almost always broken in -HEAD |
| 00:46.23 | Strom_TM | madiman: did you install mpg123? |
| 00:46.29 | Brixius | BigCanOfTuna: can you call it when you have qualify commented out |
| 00:46.36 | Strom_TM | and are you completely sure it's mpg123 and not mpg321? |
| 00:46.36 | BigCanOfTuna | Brixius: Yes, I can. |
| 00:46.40 | tzanger | Madiman: that is good -- music on hold requires a zap timing source, or a SIP source that doesn't do VAD IIRC |
| 00:46.41 | Madiman | Music in hold doesn't work it's kind of choppy |
| 00:46.57 | Madiman | yes mpg123 is installed |
| 00:47.10 | BigCanOfTuna | Brixius: But the status becomes UNMONITORED.....not that it is a bad thing, but I just want to understand why it is unreachable when I have qualify set. |
| 00:47.29 | Markie | there alreayd is a working group for IMAP Voice |
| 00:47.37 | Markie | for years now. standards too |
| 00:48.05 | Markie | http://quimby.gnus.org/internet-drafts/draft-ema-vpim-imap-01.txt |
| 00:48.08 | Brixius | BigCanOfTuna: then I'm not sure, I just saw your question and have seen the error before on my * box, the solution was to increase the quialify= line or coment it out. |
| 00:48.18 | tzanger | BigCanOfTuna: qualify doesn't ensure it's gonna be reachable, it just keeps an eye on it and won't try to dial it if it's unreachable |
| 00:48.27 | *** join/#asterisk ZX81 (matt@222-152-141-153.jetstream.xtra.co.nz) |
| 00:48.51 | Madiman | so when I get the card installed it should work? |
| 00:49.23 | johngalt | could someone point me at the docs I need to get all my config files written? * is compiled running the sample config w/ a tdm400/1 fxo/1 fxs. I have a iaxtel nbr, a fwd nbr, another sip provider, a pots phone, and pots telco line to setup. |
| 00:49.25 | BigCanOfTuna | tzanger: Yea...I figured that. Any idea why it would be ureachable though? Firewall or something (although I killed iptables)...I'm just trying to understand.....thanks for you help though. |
| 00:49.36 | BigCanOfTuna | Brixius: thanks for all your help. |
| 00:50.02 | greg_work | johngalt: voip-info.org is very useful |
| 00:50.10 | Brixius | ~docs |
| 00:50.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 00:50.20 | tzanger | BigCanOfTuna: yeah basically unreachable is usually due to bad registration information (bad user/pass) or connectivity issues |
| 00:50.38 | Brixius | johngalt: see the above links for documentation on config files. |
| 00:52.05 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
| 00:53.04 | blankman | hey bkw_, can you go in and fix the rss feeds for the iax channel on mantis? You and a bunch of others have a <B> bing closed by a </b> |
| 00:53.46 | blankman | That isn't valid xml so the readers are being mad. |
| 00:56.34 | iCEBrkr | woah |
| 00:56.47 | iCEBrkr | I just tested my MoH and it was like insanely over modulated/loud |
| 00:57.08 | Atacomm | looking to find out how I can get a commercial use license on some music for promotional videos |
| 00:57.22 | HoopyCat | icebrkr: if it's too loud, you're too old ;-) |
| 00:57.27 | iCEBrkr | HoopyCat: hahaha |
| 00:57.43 | Brixius | Atacomm: do you know who wrote it/produced it? |
| 00:58.03 | HoopyCat | icebrkr: in musiconhold.conf, try using quietmp3: instead of mp3: |
| 00:58.03 | mishehu | Atacomm: bend over for the music monarchs |
| 00:58.05 | *** part/#asterisk BigCanOfTuna (~chatzilla@S0106000c41b313f3.cg.shawcable.net) |
| 00:58.13 | iCEBrkr | yea, just about to do that |
| 00:58.28 | HoopyCat | atacomm: try ASCAP or BMI |
| 00:58.41 | Moc | Go people, so on bugs.digium.com and test some of those NICE patch up there... and comments about them please. It will help the process of getting those new feature in HEAD. Thanks |
| 00:58.57 | Atacomm | brixius: havent officially decided on anything yet ... one that we are looking at is an avril lavigne song... i'm sure no matter what even though we only expect a few thousand views max we'll still get reamed up the ass by big RIAA |
| 00:59.06 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 00:59.26 | iCEBrkr | damn, it didn't make a difference. |
| 00:59.31 | syslod | Moc: How does the multi-file boot patch work? BTW the count and intro meet me patches are sweet thanks. |
| 00:59.37 | HoopyCat | icebrkr: you will need to restart the asterisk process |
| 00:59.40 | iCEBrkr | I did |
| 00:59.49 | HoopyCat | icebrkr: install foam over your earpiece |
| 00:59.50 | Moc | syslod, multi file boot ? |
| 00:59.58 | iCEBrkr | haha |
| 01:00.07 | Atacomm | although figuring out how to spin a PCI card without covering any of it to hold it vertical is going to be more tough, that and green screening / blue screening against a greenish/blue PCB |
| 01:00.09 | syslod | Yea there are a couple of files to patch. Just use the last one or all? |
| 01:00.13 | iCEBrkr | Before I upgraded my Asterisk, I remember MoH being too damn quiet. |
| 01:00.22 | iCEBrkr | now it's just the opposite |
| 01:00.23 | Moc | syslod, for the reboot thng ? |
| 01:00.27 | syslod | Yea |
| 01:01.01 | HoopyCat | atacomm: the process is somewhat expensive but easy; you give the money to ASCAP and/or BMI, and they give you permission to use whatever is in their library, and then they divy up the money accordingly |
| 01:01.02 | Moc | check it again, Ive done some cleanup, and also renamed it to sip notify, so I modified the patch information : check bug 3231 |
| 01:01.14 | Moc | syslod, I hope they get into CVS ASAP ;) |
| 01:01.35 | Markie | Moc: how do we see the new patches? that only lists bugs? |
| 01:01.37 | Atacomm | HoopyCat: ASCAP, who is that? |
| 01:01.52 | HoopyCat | atacomm: The American Society of Composers, Authors and Publishers |
| 01:01.59 | Moc | Markie, no, patchs and bugs are all on the same site rightnow |
| 01:02.08 | flewid | sup |
| 01:02.09 | syslod | Moc: Yea I know I've got like 20 patches right now fixing sip problems. |
| 01:02.13 | *** join/#asterisk bjohnson (~bjohnson@ip186-172.dsl.istop.com) |
| 01:02.16 | Moc | [patch] mean it a patch, a new feature, or something to fix a problem |
| 01:02.17 | Markie | Moc: so how can i figure out the difference :) |
| 01:02.24 | Markie | theres thousands :) |
| 01:02.40 | Moc | Markie, well hundred open.. |
| 01:02.50 | HoopyCat | atacomm: there's ASCAP and BMI which cover most things around these parts, then there's SESAC for other parts of the world |
| 01:02.54 | Moc | 92 open, be sure resolved and closed are hidden |
| 01:03.04 | syslod | Moc: Ahh I was looking at it eariler. I'll give it a try. |
| 01:03.06 | Markie | ok..i'll just broswe them all |
| 01:03.23 | Beave | I just did a upgrade CVS 05-05-2004 (which was working fine) to CVS-v1-0-01/03/05-19:30:33. Now, my HDLC is not working. ztcfg -vvvv gives a "ZT_CHANCONFIG failed on channel 13: Function not implemented (38) |
| 01:03.23 | *** join/#asterisk jstorm (~none@ip68-11-89-95.no.no.cox.net) |
| 01:03.25 | Beave | " |
| 01:03.31 | Moc | also if you look at Severity column, if it show feature with [patch] then it probably a new feature |
| 01:03.43 | Beave | does zaptel need to be compiled with a special HDLC flag now or something? |
| 01:03.45 | Curren | ok... iaxy is oneway voice... firefly works fine through the same setup, any thoughts |
| 01:04.04 | WilliamK | Beave, not to my knowledge |
| 01:04.09 | TripleFFF | now been 21 days ups holding my darn fxo card |
| 01:04.18 | Moc | TripleFFF, hey |
| 01:04.24 | Moc | you tryed my dialconfirm app ? |
| 01:04.33 | TripleFFF | yes never worked |
| 01:04.34 | Beave | WilliamK: same kernel, HDLC is compiled directly in. Has been working fine. hrmph. |
| 01:04.57 | Beave | been googling, but just not finding much. |
| 01:05.12 | WilliamK | try running ztcfg -vvvvvvv ? |
| 01:05.12 | Markie | hey..can i just umm..make a general comment.. |
| 01:05.16 | TripleFFF | wen can try to make it work if u want |
| 01:05.16 | syslod | dialconfirm? |
| 01:05.29 | Moc | TripleFFF, maybe you can't specify the full path for the recording |
| 01:05.42 | TripleFFF | hmm |
| 01:05.52 | Markie | based on a number of opensource projects i've been on...the ones that have worked really well, we had a rule, that you can't submit a patch unless you also include patched documentation |
| 01:05.58 | Beave | WilliamK: yep, thats when i get the "Function not implemented (38)" on channels 13 (which is the first HDLC channel) |
| 01:06.10 | Moc | syslod, allow to dial multiple # at the same time, and only the first one that Answer AND press # (or * dont recall) will get the call, and only then the others called parties get hangup |
| 01:06.28 | Markie | it's amazing how just adidng the docs when people do a patch build up the docs tremendously..and vice versa..it's amazing how quickly the docs become obsolete if people dont update them. |
| 01:06.29 | TripleFFF | exten => 3036298337,1005,Dialconfirm(SIP/triplef|60|A(asterisk-recording-temp)) |
| 01:06.58 | syslod | Is that SIP only or ZAP channel too? |
| 01:07.03 | Moc | syslod, so you can call, home,cell,work, and even if voicemail get on on any of them, it will continue |
| 01:07.08 | TripleFFF | now if u call me with no caller id it will send to dialconfirm |
| 01:07.08 | Moc | syslod, EVERYTHING ;) |
| 01:07.17 | WilliamK | Beave, no idea... I've got issues with it not sending DNIS to the switch |
| 01:07.36 | syslod | Moc: I paid like $15,000 for that one-number feature on a class 5 switch. |
| 01:08.28 | Moc | ouch |
| 01:08.40 | syslod | Moc: That last one failed to compile. |
| 01:08.49 | Moc | dialconfirm ? |
| 01:09.00 | Moc | your running stable ? |
| 01:09.20 | syslod | sip_notify |
| 01:09.28 | syslod | CVS HEAD |
| 01:09.35 | Beave | WilliamK: I could probably drop back to the older zaptel version, but i really dont want to do that. |
| 01:09.39 | Beave | oh well. |
| 01:09.45 | Moc | weird... let me check |
| 01:09.45 | knight_ | tgrrrrr why is broadvoice 500ing |
| 01:09.51 | TripleFFF | shit i think it kindof works |
| 01:09.57 | TripleFFF | i pressed start lol |
| 01:09.58 | Markie | knight_: problems with broadvoice? |
| 01:10.04 | WilliamK | Beave, I tried that to no success |
| 01:10.09 | Markie | knight_: i suprisingly havent had problems in a few days |
| 01:10.17 | *** join/#asterisk WarchildX (user@gso26-154-166.triad.rr.com) |
| 01:10.20 | *** join/#asterisk greg_work (~greg@ik-dynamic-66-102-67-22.kingston.net) |
| 01:10.24 | syslod | Moc: http://pastebin.ca/3713 |
| 01:10.59 | TripleFFF | OHOHW |
| 01:11.05 | TripleFFF | moc u made that dialconfirm ? |
| 01:11.11 | Moc | yes |
| 01:11.16 | TripleFFF | SIP/triplef-529b (default 1 ) Ringing (None) (None) |
| 01:11.16 | TripleFFF | SIP/triplef-e101 (default 1 ) Ringing (None) (None) |
| 01:11.28 | TripleFFF | it has a ammajor bug once out of 2 it redials my sip |
| 01:11.37 | *** part/#asterisk Madiman (~patdelan@66-188-106-58.mad.wi.charter.com) |
| 01:11.44 | *** join/#asterisk Madiman (~patdelan@66-188-106-58.mad.wi.charter.com) |
| 01:11.48 | TripleFFF | and i get a 482 loop detected on second channel |
| 01:11.54 | TripleFFF | it doenst playback my mess |
| 01:12.07 | *** part/#asterisk Madiman (~patdelan@66-188-106-58.mad.wi.charter.com) |
| 01:12.15 | TripleFFF | tried again even if no msg playbacked i pressed # ... it continued to ring on incoming line.. then loop |
| 01:12.30 | TripleFFF | now landline and sip hungup |
| 01:12.34 | TripleFFF | and still show channels |
| 01:12.34 | TripleFFF | SIP/triplef-529b (default 1 ) Ringing (None) (None) |
| 01:12.34 | TripleFFF | SIP/triplef-e101 (default 1 ) Ringing (None) (None) |
| 01:12.42 | TripleFFF | still rigning in asterisk some way |
| 01:12.44 | Moc | ok let me check syslod compiling problem, I'll test dialconfirm right after |
| 01:12.57 | *** join/#asterisk PMantis (~PrayingMa@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 01:13.46 | TripleFFF | is there a way to drop channels in * ? like to reset or i need to stop now and restart |
| 01:13.57 | *** part/#asterisk WarchildX (user@gso26-154-166.triad.rr.com) |
| 01:14.20 | *** join/#asterisk WarchildX (user@gso26-154-166.triad.rr.com) |
| 01:14.36 | JunK-Y | TripleFFF: ya mean terminates a channel? |
| 01:15.03 | TripleFFF | yeah |
| 01:15.03 | JunK-Y | soft hangup 'chan' |
| 01:15.11 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 01:15.21 | TripleFFF | they still there |
| 01:15.28 | TripleFFF | soft hangup SIP/triplef-529b |
| 01:15.28 | TripleFFF | Requested Hangup on channel 'SIP/triplef-529b' |
| 01:15.33 | TripleFFF | nut still shows |
| 01:15.45 | PMantis | I need some direction... My phone server is down, and wife is not happy. |
| 01:15.46 | Moc | TripleFFF, when a channel deadlock, you can't shut it down |
| 01:15.53 | TripleFFF | yeah it looped |
| 01:16.08 | TripleFFF | seemd when i accepted call ( even if i didnt get the greeting) it looped on calling me back |
| 01:16.11 | PMantis | I rebooted to add a new TDM daughterboard... I can't get the drivers to load now |
| 01:16.12 | JunK-Y | restart when convenient works? |
| 01:16.31 | TripleFFF | yep |
| 01:16.32 | johngalt | ls |
| 01:16.34 | TripleFFF | that did it |
| 01:16.53 | TripleFFF | nope didnt |
| 01:16.53 | johngalt | sorry wrong window |
| 01:16.53 | TripleFFF | lol |
| 01:17.03 | Moc | syslod, your running today head ? |
| 01:17.13 | TripleFFF | hopefully you wont type rm -rf /* here and end up in unix |
| 01:17.18 | syslod | Yea. |
| 01:17.26 | Moc | make clean |
| 01:17.28 | Moc | make install |
| 01:17.29 | syslod | Plus the stuff that broke your patch. |
| 01:17.54 | Silik0n | 4you could do like I did today and thats dd if=/dev/zero of=/dev/hda count..... |
| 01:17.57 | Moc | i mean make clean ; make |
| 01:18.10 | Moc | Silik0n, BAD.. |
| 01:18.23 | Silik0n | Moc it was you script that did it |
| 01:18.34 | Silik0n | i was testing some changes to it and forgot to comment out the dd |
| 01:18.37 | ds2 | depending on the value of count, that is quite harmless |
| 01:18.41 | Moc | lol... |
| 01:18.51 | Silik0n | ds2 the count value was high enuff to wipe the partition table |
| 01:19.03 | ds2 | partition table can be safely wiped out |
| 01:19.11 | ds2 | done it many times and lost 0 data |
| 01:19.12 | Silik0n | until you reboot |
| 01:19.17 | Moc | syslod, I just tryed, and it fine with head, you probably didnt make clean or something |
| 01:19.17 | ds2 | reboot is harmless |
| 01:19.22 | syslod | Moc: It won't work. Take a look at 3227. |
| 01:19.31 | ds2 | just come back with a boot disk, and restore partitiontable w/fdisk from memory |
| 01:19.32 | Beave | Okay. just for future reference.... |
| 01:19.39 | syslod | I had to apply those to get sip to work as well as other people. |
| 01:19.47 | tzanger | -- Message check requested for mailbox 8880110/folder (null) but voicemail not loaded. |
| 01:19.59 | tzanger | ^^ what's that mean? voicemail is loaded and is clearly working for mailbox 8880110 |
| 01:20.03 | syslod | They haven't commited to CVS but I expect they will soon. |
| 01:20.05 | PMantis | I ran modprobe zaptel and modprobe wctdm... loaded, but no lights :( |
| 01:20.24 | *** join/#asterisk NormAst (NormAst@Ottawa-HSE-ppp4119047.sympatico.ca) |
| 01:20.25 | Beave | if your doing HDLC, modify the zconfig.h (under zaptel) and enable "CONFIG_ZAPATA_NET". |
| 01:20.32 | Beave | ztcfg seems to run fine now. |
| 01:22.25 | syslod | Moc: They changed ASTOBJ_CONTAINER_TRAVERSE |
| 01:22.30 | Moc | syslod, I donno, it look fine over here, Im running head, and SIP work fine, and my patch applyed fine |
| 01:22.35 | HoopyCat | box saves self from being chucked off of roof of building with sudden, last-minute cooperation. film at 11. |
| 01:22.39 | syslod | I can fix it. |
| 01:22.54 | Moc | you must have applyed something else.. |
| 01:24.36 | syslod | I applied ASTOBT fixes. |
| 01:24.50 | TripleFFF | let me know wenhn ready moc |
| 01:25.06 | NormAst | Hi All.. |
| 01:25.07 | bkw_ | syslod but see I don't see these locks like you described |
| 01:25.20 | bkw_ | and I don't have the patch applied |
| 01:25.49 | NormAst | Anyone know if there is a way to set the Max. number of calls between two * boxes |
| 01:28.05 | mishehu | bah. |
| 01:28.24 | syslod | All I know is after applying all the patchs it mysteriously started working. ??? |
| 01:29.25 | PMantis | Can someone please think through this with me, to help see what I'm doing wrong? zaptel kernel modules not loading right |
| 01:30.52 | Moc | TripleFFF, dl clean CVS, and installed dialconfirm, I'll try it rightnow |
| 01:31.18 | TripleFFF | just did |
| 01:31.19 | TripleFFF | lol |
| 01:33.25 | *** join/#asterisk robert_vcch (~robert@pcp517130pcs.nash01.tn.comcast.net) |
| 01:33.43 | Moc | ok got the problem I think |
| 01:33.58 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 01:34.08 | TripleFFF | what is it lol |
| 01:34.22 | Moc | I'll find it hold |
| 01:35.29 | iCEBrkr | wow this is crazy.. |
| 01:35.36 | iCEBrkr | Music on hold is loud as hell |
| 01:35.43 | iCEBrkr | I tried different mp3s too |
| 01:36.02 | *** join/#asterisk ProAtWork (~procyan@69-170-27-50.chvlva.adelphia.net) |
| 01:36.17 | iCEBrkr | PMantis: what are you doing wrong? :) |
| 01:36.20 | ProAtWork | hi all.. I'm having trouble getting asterisk to pass dtmf to a 3rd party vmail system |
| 01:36.28 | syslod | Moc: Reboot works with poly. ALso I have the patch to work with ASTOBJ patchs if you want them. |
| 01:36.40 | ProAtWork | the vmail system only supports rfc2833 |
| 01:36.46 | redder86 | ProAtWork: did you install disAsterisk? |
| 01:37.32 | TripleFFF | guys |
| 01:37.34 | TripleFFF | 65.110.45.10 - - [03/Jan/2005:17:42:10 -0500] "CONNECT 208.179.62.126:6667 HTTP/1.0" 200 1299 "-" "-" |
| 01:37.40 | ProAtWork | disasterisk? |
| 01:37.42 | TripleFFF | what this |
| 01:37.46 | ProAtWork | aparently not.. what is that? |
| 01:37.55 | redder86 | ProAtWork: hehe ;-) |
| 01:38.36 | redder86 | Disasterisk is a dysfunctional Asterisk. |
| 01:38.45 | ProAtWork | I see |
| 01:38.59 | ProAtWork | well.. my asterisk mostly functions as long as you don't expect to use dtmf |
| 01:39.00 | redder86 | looks like TripleFFF has one too |
| 01:39.05 | Faithful | I'm trying to test * with linphone... * tells me that the client is registered, but when I try to connect to sip:1000@pbx I get a linphone error saying "User need to authenticate to INVITE!" |
| 01:39.15 | TripleFFF | ?? |
| 01:39.26 | redder86 | ProAtWork: how do you have DTMFSignalling set up in your client config? |
| 01:39.48 | ProAtWork | dtmfmode=rfc2833 |
| 01:40.09 | redder86 | ProAtWork: and the phone is configured to use rfc2833 |
| 01:40.09 | redder86 | ? |
| 01:40.13 | ProAtWork | oh.. _on_ the client? |
| 01:40.30 | ProAtWork | yes.. I'm using xten which I think defaults to rfc2833 |
| 01:40.32 | *** join/#asterisk cnw (~cnw@holsten.conradwood.net) |
| 01:40.46 | redder86 | why do you think it defaults to rfc2833? |
| 01:41.03 | ProAtWork | because you have to specifically tell it to go inband |
| 01:41.15 | ProAtWork | also.. my incoming calls to a digium have the same problem |
| 01:41.16 | redder86 | what about SIP info ? |
| 01:41.23 | *** join/#asterisk r0d3nt|m (~RatMan@adsl-69-104-33-81.dsl.scrm01.pacbell.net) |
| 01:41.25 | ProAtWork | and actually thats the problem I'd like to solve |
| 01:41.48 | redder86 | your incoming calls to Digium? You work at Digium? |
| 01:41.53 | ProAtWork | redder86 actually.. pstn->digium->3rd party vmail does not work |
| 01:42.05 | *** part/#asterisk JunK-Y (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 01:42.05 | ProAtWork | redder86 no, incoming calls to my digium board in the asterisk |
| 01:42.07 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 01:42.17 | redder86 | TE405P ? |
| 01:42.23 | redder86 | TDM card? |
| 01:42.32 | redder86 | X100P ? |
| 01:42.48 | ProAtWork | 4-port analog.. getting part number |
| 01:43.02 | redder86 | no problem. |
| 01:43.08 | ProAtWork | TDM400P |
| 01:43.21 | iCEBrkr | errrrgh |
| 01:43.31 | ProAtWork | loaded with 1 FXS and 3 FXO |
| 01:43.57 | TripleFFF | moc? |
| 01:44.01 | *** join/#asterisk bjohnson (~bjohnson@ip186-172.dsl.istop.com) |
| 01:44.05 | redder86 | well, I don't know anything about connecting third-party voicemail systems to Asterisk, but I'd say that the problem lies with Asterisk not sending DTMF to it in a way that it recognises. |
| 01:44.08 | ProAtWork | TDM13B is the part number |
| 01:44.21 | ProAtWork | redder86 yes, I agree |
| 01:44.27 | redder86 | Third party voicemail systems are just another extension? |
| 01:44.29 | *** join/#asterisk bzwi (~w_w_zhang@ip67-153-142-73.z142-153-67.customer.algx.net) |
| 01:44.37 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 01:44.37 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 01:44.40 | ProAtWork | but I'm sure how to verify that asterisk is actually sending using rfc2whatever |
| 01:44.53 | ProAtWork | redder86 yes, you dial a number and it does an invite |
| 01:45.10 | ProAtWork | grrr but I'm _not_ sure how to verify |