| 00:09.37 | echinos | is there a builtin way to paginate asterisk CLI output, other than watching the logfile? |
| 00:10.45 | *** join/#asterisk n3hxs (n=HAMming@pool-70-110-19-76.washdc.fios.verizon.net) |
| 00:13.04 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:26.06 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 00:26.31 | *** join/#asterisk jbebel (n=jbebel@nat/google/x-df350f8d7d12db42) |
| 00:31.18 | *** join/#asterisk xanderp (n=pphillip@c-98-220-163-8.hsd1.in.comcast.net) |
| 00:32.07 | xanderp | anyone with a working asterisk box able to call me via a sip call to verify that my system is receiveing incoming calls? I think my VOIP provider has it's routes screwed up again, I can call out, but not receive calls. |
| 00:34.25 | [TK]D-Fender | xanderp: Call you how? |
| 00:35.06 | xanderp | Hmm |
| 00:35.28 | xanderp | I know there's a way to call a sip extension directly, but am drawing a blank... been about a year since I've done it. |
| 00:35.59 | xanderp | like 6002@www.mypbx.com kind of format... |
| 00:36.08 | xanderp | something like that |
| 00:36.24 | xanderp | crap |
| 00:36.35 | [TK]D-Fender | xanderp: is that the exact URI? |
| 00:36.36 | xanderp | would have to be from a sip software phone, not from asterisk |
| 00:36.49 | xanderp | no... just an example |
| 00:36.56 | [TK]D-Fender | xanderp: xanderp and what do you mean "not from asterisk"? |
| 00:37.21 | xanderp | I forgot that when I was direct dialing for testing it wasn't from asterisk, but from a softphone on the pc. |
| 00:37.43 | [TK]D-Fender | xanderp: Ok, well if you want someone to direct-call you, provider the URI |
| 00:38.00 | xanderp | i'm not sure of the format... |
| 00:38.16 | xanderp | my system is www.xanderp.com and the extension to call would be 6002. |
| 00:38.40 | [TK]D-Fender | xanderp: pastebin your sip.conf & extensions.conf masking only passwords |
| 00:38.57 | xanderp | ? |
| 00:39.00 | [TK]D-Fender | xanderp: and FYI I don't see what this proves about your provider... |
| 00:39.24 | xanderp | i wanted to prove that my router/asterisk box was receiveing incoming calls, just not from them. |
| 00:39.44 | xanderp | guess it wouldn't do that aye? |
| 00:40.32 | [TK]D-Fender | calling |
| 00:40.37 | xanderp | I think i can call directly from googletalk by doing soemthing like 6002:www.xanderp.com or possibly 6002@www.xanderp.com or something like that... i had it working before. |
| 00:41.15 | [TK]D-Fender | xanderp: No SIP response at all |
| 00:41.17 | xanderp | hmm |
| 00:42.33 | xanderp | when i do a sip show peers it shows my trunk as OK, so I would think that inbound calls would work. I haven't changed anything on my configs in forever, and just the past 2 days it stopped incoming calls only. i can still place calls. |
| 00:43.11 | xanderp | i'm going to try to remote into my work and test connecting back into my home from there. |
| 00:43.39 | [TK]D-Fender | xanderp: this is the part where I distrust EVERYTHING. You aren't showing anything. |
| 00:45.15 | xanderp | I think there's a way to telnet to the sip port to see if it's listening isn't there? |
| 00:45.15 | Nugget | telnet is eeeeeeevil! |
| 00:45.58 | [TK]D-Fender | xanderp: Dump your firewalls, check your IP's, check your SIP config, you should have been watching for debug when I called, SHOW YOUR CONFIGS, etc |
| 00:48.25 | [TK]D-Fender | yup, another complete waste of time. |
| 00:49.19 | *** join/#asterisk SiberAIR (n=SibRphre@cpe-67-243-43-136.nyc.res.rr.com) |
| 00:50.56 | *** join/#asterisk jacco (n=root@unaffiliated/jacco) |
| 00:50.59 | jacco | Hey guys. |
| 00:51.18 | jacco | Uh, so I had to move a network and the network topology has changed significantly. |
| 00:51.42 | jacco | Now my sip phone doesn't even register with asterisk, although it responds to ping. |
| 00:51.47 | jacco | Um, where to begin troubleshooting? |
| 00:51.56 | hardwire | how the crap can you configure a 7912 over the web interface? |
| 00:52.00 | hardwire | there is no submit button! |
| 00:52.32 | jacco | freaking internet phones. :( |
| 00:52.40 | *** join/#asterisk giantrobot (n=giant_ro@74-137-137-8.dhcp.insightbb.com) |
| 00:53.04 | [TK]D-Fender | jacco: pastebin your sip.conf masking only passwords and tell us in detail Exactly what networking is sitting between * and your device |
| 00:54.11 | [TK]D-Fender | ~pb |
| 00:54.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 00:54.13 | [TK]D-Fender | ^^^^^^^^^^ |
| 00:54.21 | jacco | Uh, can't really paste sip.conf. There's no x server on here, no ftp either. |
| 00:54.24 | jacco | Not even screen. |
| 00:54.26 | jacco | :| |
| 00:54.41 | hardwire | ssh? |
| 00:54.46 | jacco | although I could paste over ssh I guess. |
| 00:54.54 | hardwire | your brain is neat |
| 00:56.39 | jacco | ohhh wait I think I know the problem. D'oh. |
| 01:01.35 | *** join/#asterisk JoshieP0x (n=JK@unaffiliated/joshuap0x) |
| 01:02.08 | JoshieP0x | I'm new to linux. Is Asterisk easy to setup and use? |
| 01:03.07 | JoshieP0x | Is this a software that will work on the old slow box that has some dust on it in my closet or do i need something new? |
| 01:04.14 | ryoohki | the receptionist phone has stop ringing and is going straight to voice mail |
| 01:04.16 | [TK]D-Fender | JoshieP0x: Depends on the machines spec, and your needs |
| 01:04.19 | jql | it'll run on a wocketwatch. whether is will work depends on what you're asking of it. |
| 01:04.25 | bobnormal | hey i'm having problems setting up sip peers in asterisk realtime. i've got users working fine already. trying to register with voipbuster, but can't get the entry in to the table corresponding to sip_peers (same as sip_users, which works fine) to show up in 'sip show registry' output. anyone done this before? |
| 01:04.25 | jql | s/w/p/ |
| 01:04.58 | ryoohki | http://pastebin.com/m2ea194a5 |
| 01:06.04 | *** join/#asterisk andresmujica (n=andresmu@201.244.105.224) |
| 01:06.19 | ryoohki | also, does anyone know if the polycom sound point ip 601 sip uses the same tftp firmware as the polycom sound point ip 501 sip? |
| 01:07.02 | JoshieP0x | [TK]D-Fender: I have a Athlon processor. Not sure of the speed. It's about 5 years old. The box has 2G of RAM |
| 01:07.16 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:07.18 | bobnormal | polycom = pleaserapemywalletcom |
| 01:07.34 | bobnormal | JoshieP0x, fine |
| 01:07.57 | JoshieP0x | bobnormal: Fine? |
| 01:08.21 | giantrobot | are there any large companies using asterisk in a production environment? |
| 01:09.32 | bobnormal | JoshieP0x, as long as you dont want too many lines and heavy codec conversion |
| 01:09.36 | jql | depends on the definition of large. Lots of customers? Lots of minutes? Lots of dollars? |
| 01:10.00 | bobnormal | JoshieP0x, also depends on the hardware, cheap zaptel cards bleed interrupts which can cause jittery machines |
| 01:10.09 | jql | the answer is probably yes for all of those, anyways |
| 01:10.10 | bobnormal | JoshieP0x, but if you try it and your needs are minimal you'll likely have success |
| 01:10.10 | ryoohki | anyone know the default password on these polycom phones? i can |
| 01:10.14 | giantrobot | jql, well im a native linux person, im looking for a new product... specfically im looking to develop an ivr ap |
| 01:10.22 | jql | ryoohki: admin pw is 456 |
| 01:10.25 | ryoohki | 't get into the advaced settings |
| 01:10.31 | ryoohki | ok |
| 01:10.36 | giantrobot | jql, trying to give my sales person a few names to drop |
| 01:10.40 | ryoohki | thanks! |
| 01:10.41 | JoshieP0x | bobnormal: Nah. just 2 lines, 3 mailboxes |
| 01:10.55 | JoshieP0x | I need a modem for my machine. Any suggestions? |
| 01:11.41 | jql | giantrobot: once you get familiar with asterisk, it's funny how many times you hear the default prompts and music from asterisk on other companies' IVRs |
| 01:11.51 | ryoohki | jql: what is the diff between local setting and device config? |
| 01:12.00 | [TK]D-Fender | JoshieP0x: Modems will not work. You need a compatible FXO interface |
| 01:12.18 | jql | allison smith's voice is also very distinct |
| 01:12.22 | *** join/#asterisk Fester (i=kalin@unaffiliated/fester) |
| 01:12.52 | JoshieP0x | [TK]D-Fender: Any recomendations for one? |
| 01:12.52 | giantrobot | jql, interesting... is the ivr flexible? database connections? can i write straight code, in java or another language? |
| 01:13.29 | jql | giantrobot: it comes with a couple of dialplan scripting languages, and a couple of external interfaces |
| 01:13.33 | [TK]D-Fender | JoshieP0x: Describe your actual needs |
| 01:14.16 | jql | giantrobot: and there's an internal database as well as external plugins for a couple more |
| 01:14.40 | giantrobot | jql, whats the best (cheapest) test config, install the packages and use softphones? |
| 01:15.09 | giantrobot | jql, trying to get a dev environment setup on my laptop ;) |
| 01:15.09 | *** part/#asterisk Fester (i=kalin@unaffiliated/fester) |
| 01:15.15 | jql | giantrobot: yes, exactly. be default, dialing into it triggers a test IVR application |
| 01:15.27 | giantrobot | nice, i got it installed |
| 01:15.41 | giantrobot | jql, i'll go research a softphone for linux |
| 01:16.05 | JoshieP0x | [TK]D-Fender: Well, I'm not sure how this all works. |
| 01:16.17 | JoshieP0x | I just learned about SIPs yesterday |
| 01:16.20 | [TK]D-Fender | JoshieP0x: What do you WANT? |
| 01:16.29 | JoshieP0x | so something to learn with |
| 01:16.33 | giantrobot | jql, the thing is, ive got a customer, straight mac shop... they want softphones and most of the standard pbx companies use proprietary softphone clients |
| 01:16.33 | JoshieP0x | 2 lines |
| 01:16.36 | JoshieP0x | 3 AMs |
| 01:16.43 | JoshieP0x | automated attendant |
| 01:16.44 | [TK]D-Fender | AMs? |
| 01:16.47 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:16.58 | giantrobot | jql, so i think i have an oppurtunity to sell it |
| 01:17.02 | [TK]D-Fender | JoshieP0x: You can learn * with just a PC to run it on and using a soft phone. |
| 01:17.05 | JoshieP0x | answering machines |
| 01:17.05 | jql | meh. free softphones are fine. the proprietary ones are mainly raped with branding |
| 01:17.18 | [TK]D-Fender | JoshieP0x: And when it comes time to consider hardware then that is another matter |
| 01:17.21 | JoshieP0x | thats fine |
| 01:17.57 | giantrobot | jql, what about hunt groups, line appearances |
| 01:18.15 | giantrobot | jql, does that work on softphones and ip hard phones? |
| 01:18.26 | jql | once you desire BLF functionality, softphone selection gets more entertaining |
| 01:18.32 | JoshieP0x | will that Oriely * book teach me about softphones and how to setup * with them |
| 01:18.38 | giantrobot | jql, thats what i was wondering |
| 01:18.39 | JoshieP0x | I have Time Warner |
| 01:18.42 | jql | but, yeah |
| 01:18.47 | [TK]D-Fender | JoshieP0x: very basic stuff.. |
| 01:18.54 | JoshieP0x | I want to know how to get the SIP settings |
| 01:19.00 | [TK]D-Fender | JoshieP0x: Should have some quick samples. We can help out with the rest |
| 01:19.13 | giantrobot | jql, point being, there is an ip hardphone/softphone that do and dont support it |
| 01:19.17 | JoshieP0x | [TK]D-Fender: What should? The book? |
| 01:19.20 | giantrobot | is/are |
| 01:19.22 | [TK]D-Fender | JoshieP0x: only takes about 6 values for Asterisk side, and 3 on your soft-phone |
| 01:19.28 | [TK]D-Fender | ~book |
| 01:19.28 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 01:19.30 | [TK]D-Fender | ^^^^^^^^^^ |
| 01:19.44 | jql | yes |
| 01:19.59 | JoshieP0x | I am already there |
| 01:20.05 | JoshieP0x | thanks. |
| 01:20.17 | giantrobot | jql, you got a preference on the softphone? |
| 01:21.18 | jql | I don't have much linux softphone knowledge, regrettably |
| 01:21.33 | jql | I work with hard phones and windows, to my chagrin. :) |
| 01:21.53 | giantrobot | jql, hard ip phones? |
| 01:22.35 | jql | polycom, linksys, sometimes snom |
| 01:22.43 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 01:22.57 | giantrobot | jql, are you just managing a system or are you reselling? |
| 01:23.37 | jql | company resells |
| 01:24.05 | giantrobot | nice, in the US? |
| 01:24.59 | giantrobot | does asterisk support any video conferencing? |
| 01:25.06 | jql | yeah, supposedly in the US. although some of the phones magically register from india |
| 01:25.08 | jql | go figure. :) |
| 01:25.19 | giantrobot | sure i get that |
| 01:25.49 | giantrobot | well im a systems guy that has always worked for companies that do phone and computers |
| 01:26.04 | bobnormal | jql: i just ordered some cheap hardware sip phones from a chinese company, 38USD each :) |
| 01:26.18 | jql | yeah, cheap |
| 01:26.36 | bobnormal | support IAX too! :) |
| 01:26.36 | giantrobot | ive now got my own business and looking to sell something that i can customize |
| 01:26.52 | jql | that's a couple dollars off the low-end retain grandstream, even |
| 01:26.58 | jql | s/tain/tail/ |
| 01:27.56 | giantrobot | IAX asterisks bastard protocol? for BLF and such? |
| 01:29.48 | Corydon76-dig | giantrobot: video calls, yes. Video conferencing, no. |
| 01:30.10 | giantrobot | Corydon76-dig, nice |
| 01:30.12 | Corydon76-dig | giantrobot: the main issue is the patent-encumbered video codecs |
| 01:30.26 | giantrobot | giantrobot, hmmm |
| 01:31.10 | giantrobot | Corydon76-dig, video voicemail? |
| 01:31.10 | Corydon76-dig | If we could get past that, then we could look into video mixing |
| 01:31.23 | Corydon76-dig | Yes, Asterisk supports video voicemail today |
| 01:31.27 | giantrobot | Corydon76-dig, nice |
| 01:31.47 | giantrobot | can the video voicemail be watched on any of the hardware phones? |
| 01:31.52 | giantrobot | ip of course |
| 01:32.25 | Corydon76-dig | Yes, in fact, I've done it on the Grandstream GXV-3000 |
| 01:32.31 | giantrobot | im trying to justify paying 300 bucks for that high end grandstream |
| 01:32.53 | [TK]D-Fender | high end grandstream = "deluxe" crap? |
| 01:32.56 | [TK]D-Fender | ~gs |
| 01:32.56 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 01:32.57 | Corydon76-dig | Heh, the "high end" Grandstream is the cheapest video phone on the market |
| 01:32.59 | [TK]D-Fender | ~grandstream |
| 01:32.59 | jbot | i heard grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 01:33.07 | giantrobot | ha |
| 01:33.09 | jql | fun thing about video-phones is that you have to order two for "testing purposes". tee hee hee |
| 01:33.52 | Corydon76-dig | If Grandstream is the Yugo, Cisco is the Astin-Martin. Vastly overpriced for the functionality |
| 01:33.56 | giantrobot | are any companies formally offering support? |
| 01:34.26 | giantrobot | i.e. i sell this product, ive got some critical issue, can i pay someone for support |
| 01:34.47 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 01:35.00 | Corydon76-dig | You'd have to ask the individual companies |
| 01:35.28 | kerx | hey tk |
| 01:35.37 | giantrobot | k, im just trying to figure out if the asterisk developers are pulling a redhat model |
| 01:35.45 | giantrobot | give it away, and bill for support kind of thing |
| 01:36.03 | eppigy | hello |
| 01:36.03 | Corydon76-dig | giantrobot: you mean support for Asterisk or for the hard phones? |
| 01:36.05 | eppigy | i am dave |
| 01:36.09 | giantrobot | asterisk |
| 01:36.16 | giantrobot | i'll throw the phone away |
| 01:36.21 | Corydon76-dig | Yes, you can buy contract support for Asterisk |
| 01:36.37 | Corydon76-dig | Or you can go with the free community support |
| 01:36.43 | giantrobot | Corydon76-dig, im going to pitch this to my partner, im trying to get all of my selling points established |
| 01:36.52 | bobnormal | giantrobot: asterisk is the sort of thing companies need someone cluey to analyse needs, explain possibilities, spec a deployment, configure, and maintain |
| 01:36.59 | Corydon76-dig | The only difference is that contracts get priority from Digium |
| 01:37.20 | giantrobot | bobnormal, thats why i think its a good fit for my business |
| 01:37.22 | bobnormal | giantrobot, it can do or integrate with almost ANYTHING ... billing, automation, faxing, presence, single sign on, etc... |
| 01:37.35 | giantrobot | i do custome solutions, software, hardware, dev |
| 01:37.52 | bobnormal | giantrobot, i know what you mean - if i wasn't in china starting something else, i'd be running an asterisk-focused consultancy right now |
| 01:37.59 | JoshieP0x | [TK]D-Fender: What is a live CD? |
| 01:38.08 | Corydon76-dig | giantrobot: for a long time, my employer used the fact that a core Asterisk developer was on his staff |
| 01:38.21 | Corydon76-dig | (as a selling point) |
| 01:38.26 | giantrobot | nice, maybe i should start contributing =) |
| 01:38.37 | Corydon76-dig | giantrobot: or hire somebody like seanbright |
| 01:38.41 | giantrobot | ha |
| 01:38.49 | giantrobot | that you? |
| 01:38.56 | Corydon76-dig | He's one of the remaining independent developers |
| 01:39.01 | jql | programmers are cheap; buy 2 |
| 01:39.02 | giantrobot | oic |
| 01:39.04 | Corydon76-dig | No, not me |
| 01:39.10 | Corydon76-dig | Now, I work for Digium |
| 01:39.21 | JoshieP0x | I'm looking to download CentOS. I see an option of a live CD. I'm not sure what a live cd is. |
| 01:39.24 | JoshieP0x | anyone? |
| 01:39.38 | Corydon76-dig | JoshieP0x: allows you to run off the CD without installing |
| 01:39.39 | giantrobot | ok, ive been on your site |
| 01:39.48 | Corydon76-dig | JoshieP0x: Ubuntu is the same way |
| 01:39.51 | JoshieP0x | oh |
| 01:39.52 | giantrobot | you sell the canned asterisk boxes |
| 01:40.18 | Corydon76-dig | JoshieP0x: except that Ubuntu's install CD is also a live CD. You can choose to install after using the live CD |
| 01:40.22 | JoshieP0x | also, what version of CentOS do you guys recommend? |
| 01:40.32 | Corydon76-dig | JoshieP0x: the latest, 5.2 |
| 01:40.46 | JoshieP0x | I've used LiveCDs before |
| 01:40.47 | giantrobot | Corydon76-dig, do you work with resellers? |
| 01:40.58 | JoshieP0x | thanks Corydon7-dig |
| 01:41.00 | giantrobot | Corydon76-dig, i should say does digium work with resllers? |
| 01:41.06 | Corydon76-dig | giantrobot: the company does, yes. I'm a developer; I don't work directly with resellers. |
| 01:41.33 | giantrobot | Corydon76-dig, interesting... |
| 01:41.52 | Corydon76-dig | And, I see the time, and I need to go get dinner started |
| 01:42.15 | giantrobot | thanks everyone for the convo |
| 01:42.40 | giantrobot | ive been bouncing this around for sometime, then i seen there was a freenode channel |
| 01:43.06 | giantrobot | im convinced :) |
| 01:57.25 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:59.09 | ryoohki | can anyone read this CLI and say why the receptionist's phone has stopped ringing( it's going straight to voice mail) http://pastebin.com/m2ea194a5 |
| 02:00.28 | [TK]D-Fender | ryoohki: Nowhere in there are you trying to dial a phone at all |
| 02:00.32 | [TK]D-Fender | and... |
| 02:00.33 | [TK]D-Fender | ~freepbx |
| 02:00.34 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:00.52 | [TK]D-Fender | ryoohki: You are dumping your caller into an IVR. |
| 02:01.12 | Nugget | http://www.defectiveyeti.com/archives/002657.html <-- ha ha |
| 02:01.59 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 02:04.03 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 02:11.11 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:15.32 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 02:15.47 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:15.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:16.01 | edibrac | i have a test box (apart from production) where I can't seem to get "zap show channels" to even show up as an option in the asterisk CLI |
| 02:21.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:22.03 | [TK]D-Fender | edibrac: Go prove that chan_zap is even loaded |
| 02:22.32 | edibrac | ah good idea |
| 02:23.02 | seanbright | don't hire me, i'm a hack |
| 02:23.03 | ryoohki | [TK]D-Fender: these are inbound calls going straight to vm and not ring the desk phone first - there is no attempt to dial the phone |
| 02:23.42 | [TK]D-Fender | ryoohki: And thats due to your FreePBX configuration. Nothing to do with * itself |
| 02:23.49 | ryoohki | ok |
| 02:24.20 | russellb | seanbright: orly?! |
| 02:24.24 | seanbright | nods |
| 02:24.49 | russellb | oic. |
| 02:24.50 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-203-173-209.dsl.pltn13.sbcglobal.net) |
| 02:24.55 | seanbright | russellb: don't tell russellb |
| 02:25.01 | russellb | k, I won't |
| 02:25.05 | seanbright | russellb: he thinks i'm super awesome |
| 02:25.07 | seanbright | heh |
| 02:25.54 | russellb | not anymore, because I told him you were a hack |
| 02:26.12 | seanbright | you're off my christmas list, judas. |
| 02:26.16 | edibrac | chan_zap is from the dahdi-kernel tarball right? |
| 02:26.27 | seanbright | chan_zap is in asterisk |
| 02:27.39 | russellb | i'd still hire you, even though you're a hack. |
| 02:27.49 | seanbright | pay for me to relocate? |
| 02:27.51 | seanbright | :P |
| 02:28.45 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 02:29.12 | Micc | Is there a company where I can get unlimited local SIP calls to the seattle area? |
| 02:29.38 | seanbright | you can get anything unlimited if you are willing to pay for it |
| 02:29.39 | Micc | I want to buy some PRI channels to the seattle area. |
| 02:30.31 | Micc | I'll pay for it if its not just the long distance rate * 24 hours * 31 days |
| 02:32.37 | edibrac | hmm chan_zap doesn't show up in show modules |
| 02:32.59 | edibrac | do you have to explicitly build chan_zap when you make asterisk by source? |
| 02:33.18 | russellb | you just need zaptel installed first, before you compile asterisk |
| 02:33.20 | russellb | and then it's automagic |
| 02:33.21 | seanbright | edibrac: might be called chan_dahdi |
| 02:33.33 | seanbright | oh, probably not in a package though |
| 02:33.39 | russellb | and yeah, you should be using dahdi now. |
| 02:33.41 | [TK]D-Fender | edibrac: Indeed |
| 02:33.45 | ryoohki | has anyone upgraded freepbx 1.2 to the latest? |
| 02:34.48 | Nugget | woudln't you be better off askign that in a freepbx channel? |
| 02:35.03 | seanbright | or kicking yourself in the testicles, at least. |
| 02:35.04 | Nugget | not many people here have ever run freepbx, much less upgraded it. |
| 02:35.20 | ryoohki | ok |
| 02:37.32 | edibrac | asterisk 1.4.22 is high tech for me. we had 1.2 previous |
| 02:37.35 | edibrac | ly. |
| 02:37.41 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 02:44.09 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:44.11 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 02:53.47 | hardwire | sigh |
| 02:54.11 | hardwire | anybody else absent mindedly buy a panasonic globarange? |
| 02:54.34 | *** part/#asterisk aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 02:55.39 | bobnormal | hey i'm having problems setting up sip peers in asterisk realtime. i've got users working fine already. trying to register with voipbuster, but can't get the entry in to the table corresponding to sip_peers (same as sip_users, which works fine) to show up in 'sip show registry' output. anyone done this before? |
| 02:56.20 | edibrac | hmm dahdi show channels - only shows the pseudo channel.. i guess it's misconfigured |
| 02:57.03 | edibrac | i did a dahdi config to create the config -- it's a simple PRI setup |
| 02:57.27 | edibrac | and dahdi finds the right module |
| 02:59.03 | edibrac | oh maybe it's my /etc/dahdi/system.conf that i have to manually ste |
| 02:59.37 | edibrac | hmm but that's created by the Dahdi configurator.. dahdi_cfg |
| 03:02.02 | *** join/#asterisk SiberAIR (n=SibRphre@cpe-67-243-43-136.nyc.res.rr.com) |
| 03:02.40 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 03:10.55 | jaytee | edibrac, you need to setup your chan_dahdi.conf with the proper group and channel info |
| 03:12.31 | edibrac | ah that replaces zapata.conf? |
| 03:13.16 | *** join/#asterisk dec3pti0n (n=dec3pti0@pdpc/supporter/student/dec3pti0n) |
| 03:13.58 | dec3pti0n | anyone here has tried Elastix before ? is it good ? |
| 03:14.35 | dec3pti0n | I'm trying out the asterisknow 1.5 and about to try out the elastix in a bit |
| 03:15.04 | [TK]D-Fender | dec3pti0n: Same shit different smell |
| 03:15.16 | dec3pti0n | hehe ok |
| 03:15.22 | [TK]D-Fender | dec3pti0n: Just another CentOS distro with * slapped on top w/ FreePBX |
| 03:15.50 | dec3pti0n | yeah I haven't tried freePBX yet but I'm not a big fan of php |
| 03:16.48 | jaytee | edibrac, yes. /etc/asterisk/chan_dahdi.conf replaces zapata.conf and /etc/dahdi/system.conf replaces /etc/zaptel.conf |
| 03:21.35 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:29.42 | edibrac | weird i can dial out but i get "Everyone is busy/congested at this time (1:0/0/1)" when I dial out |
| 03:29.51 | edibrac | i mean, i can dial in, but not out |
| 03:33.55 | hesco | is there a way to make an asterisk installation reveal the paths it uses for the Background() and Playback() applications ??? |
| 03:35.19 | hesco | doesn't /var/lib/asterisk/sounds/custom/ sound like a reasonable place to put them ??? |
| 03:41.23 | *** join/#asterisk mog (n=mog@c-68-62-172-152.hsd1.al.comcast.net) |
| 03:41.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:45.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:46.44 | dec3pti0n | why does asterisk prefer freepbx now than freepbx-gui is it that much better ? |
| 03:47.13 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 03:47.39 | drmessano^ | What? |
| 03:48.13 | dec3pti0n | opps sorry I meant asterisk-gui |
| 03:48.20 | jaytee | that's exactly what I thought |
| 03:48.26 | dec3pti0n | and I meant asterisknow |
| 03:48.28 | eppigy | hello |
| 03:48.30 | eppigy | i am dave |
| 03:48.39 | dec3pti0n | sorry bit sleepy here |
| 03:48.39 | jaytee | no you're not! |
| 03:48.45 | drmessano^ | Asterisknow uses FreePBX |
| 03:48.50 | drmessano^ | So what are you talking about? |
| 03:49.02 | justdave | i am dave |
| 03:49.09 | dec3pti0n | why they have changed ? |
| 03:49.14 | bobnormal | edibrac, just a guess: set your debug level higher, probably you can see some kind of error opening the channel ... asterisk -cvvvvvvvvvvvvvvvvvvvvvvvr .. or similar, also core set verbose 999999999999999 :) |
| 03:49.16 | drmessano^ | i, am dave |
| 03:49.30 | drmessano^ | dec3pti0n: Because FreePBX is 10x better than the other GUI |
| 03:49.43 | dec3pti0n | even the lastest one 2.0 ? |
| 03:50.09 | drmessano^ | yep |
| 03:50.40 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:51.24 | dec3pti0n | I trying to see if I should use the asteriskNow 1.5 , I know it's still beta but at least when it's finally release I should be able to just use yum to update to the stable version |
| 03:54.20 | jaytee | dec3pti0n, might want to ask about that in #asterisknow |
| 03:54.31 | jaytee | or ping Quell |
| 03:54.33 | dec3pti0n | oh true |
| 03:54.41 | jaytee | umm, I meant Qwell |
| 04:10.02 | *** part/#asterisk JerJer (n=PhatJ@24-236-207-64.dhcp.aldl.mi.charter.com) |
| 04:11.42 | kerx | hi everyone, anyone know how to make the asterisk CDR's work properly? All my records on outgoing calls are coming back w/ 0 in the billsec, and NO ANSWER in the disposition. |
| 04:14.27 | jeff | kerx: i have that same problem. haven't looked into fixing it yet. :P |
| 04:14.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 04:14.45 | jeff | kerx: are your outgoing calls all made with callfiles? |
| 04:14.47 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:14.55 | kerx | jeff: I've used callfiles and AMI |
| 04:15.56 | jeff | kerx: i'm using callfiles. i think i found a few bugs in the bug tracker related to CDR, but i don't recall if there were any that related to the NO ANSWER / 0 billsecs bit. |
| 04:16.37 | kerx | jeff: this has been around for a while i believe, have you ever got it to work? I have a hard time understanding how people have production environments w/ Asterisk Billing solutions that use Asterisk CDR's? |
| 04:19.42 | jeff | no idea. i'm using 1.4.22. i don't know if it's a recent bug, or if it's the way i'm making my calls, or what. |
| 04:22.33 | *** join/#asterisk outtolunc (n=me@c-67-164-8-168.hsd1.ca.comcast.net) |
| 04:29.53 | kerx | ok |
| 04:32.01 | kerx | jeff: Do you know how to make local calls instead of such as Dial(sip/name/e...) |
| 04:32.02 | kerx | ? |
| 04:32.25 | kerx | local channels i mean |
| 04:36.08 | thehar | anyone use cdr_adaptive_odbc in here? |
| 04:36.45 | *** join/#asterisk keebler (n=keebler@h231.192.20.98.dynamic.ip.windstream.net) |
| 04:37.01 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 04:37.04 | keebler | Anyone here use asterisk with FreeBSD? |
| 04:37.34 | keebler | Just wondering if the port is current. |
| 04:39.47 | keebler | Nevermind, found the info. That being said, anyone use it and like it on FBSD? I'm not a huge fan of CentOS |
| 04:43.18 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 04:48.11 | *** join/#asterisk Leddy (n=Leddy@polar.artica.net) |
| 04:48.17 | *** part/#asterisk keebler (n=keebler@h231.192.20.98.dynamic.ip.windstream.net) |
| 05:02.10 | jeff | kerx: no, i have heard of the concept, but have not tried it. |
| 05:07.25 | *** join/#asterisk luckyaba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 05:13.36 | dec3pti0n | ok has the sip command gone away on 1.4 ? |
| 05:14.11 | drmessano^ | help? |
| 05:14.12 | thehar | uhm no? |
| 05:15.30 | dec3pti0n | hmm u sure cause I attached to the asterisk console and there is no sip commands listed under help |
| 05:15.39 | thehar | sip is there for me :) |
| 05:15.47 | thehar | and i'm on 1.4.22 |
| 05:15.58 | dec3pti0n | strange |
| 05:19.47 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 05:24.06 | Az_au | dec3pti0n: if you are not running asterisk as root make sure the sip.conf is readable by the same user you are running asterisk as. I had this problem |
| 05:24.19 | loather | ok, so i'm trying to configure this set of PRIs for NFAS with zaptel |
| 05:24.31 | loather | sangoma hardware, A102c |
| 05:24.43 | loather | when configure for nfas i can't get any calls in or out |
| 05:25.03 | loather | has anyone done an nfas setup with this hardware before? |
| 05:25.05 | seanbright | loather: pastebin your zapata.conf |
| 05:25.09 | loather | will do |
| 05:25.14 | jql | yep |
| 05:25.20 | jql | works for me |
| 05:25.29 | seanbright | i do it with the a104d |
| 05:26.06 | jql | a108 here, with a 3 T1 nfas group, primary and backup D-channel |
| 05:28.00 | dec3pti0n | Az_au, yep it's readable by everyone |
| 05:29.15 | seanbright | loather: ? |
| 05:29.24 | loather | working on it |
| 05:30.30 | loather | http://pastebin.ca/1288362 |
| 05:30.54 | seanbright | loather: are you sure the logical span numbers are right? |
| 05:31.00 | seanbright | loather: my provider uses 0 and 1, not 1 and 2 |
| 05:31.18 | seanbright | 1,1,0 |
| 05:31.21 | seanbright | 2,1,1 |
| 05:31.44 | loather | ok, let me try that |
| 05:31.50 | seanbright | worth a shot |
| 05:31.50 | seanbright | :) |
| 05:32.17 | seanbright | hmmmm |
| 05:32.26 | loather | any other glaring errors? |
| 05:32.58 | loather | wait, was that in zapata.conf or zaptel.conf? |
| 05:33.06 | seanbright | zapata |
| 05:33.14 | loather | ok. |
| 05:34.09 | seanbright | everything else looks fine to me |
| 05:34.32 | seanbright | jql: thoughts? |
| 05:35.21 | jql | not much short of debugging |
| 05:35.40 | jql | I'd sniff the hell out of the d-channel and look for what's going wrong |
| 05:35.44 | loather | that did it |
| 05:35.47 | seanbright | sweet |
| 05:35.51 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 05:36.04 | jql | well, bingo |
| 05:36.24 | seanbright | that is what we in the business call: a SWAG |
| 05:36.29 | seanbright | ~swag |
| 05:36.29 | jbot | Silly Wild-Assed Guess |
| 05:37.36 | [TK]D-Fender | seanbright: Also a term for Company-branded crap like pens and base-ball caps :) |
| 05:37.41 | seanbright | yes indeed |
| 05:38.10 | loather | ok, someone wanna do me a really big favor? :) |
| 05:38.32 | seanbright | whats up? |
| 05:38.40 | loather | i need to see if calls will actually come through on the second span |
| 05:38.56 | seanbright | so you need to rollover |
| 05:38.58 | jql | I usually test outbound with cli originate, but inbound is good too |
| 05:39.28 | loather | ok, how would I do that? :) |
| 05:39.30 | jql | first and last b-channel on every span, for best results. :) |
| 05:39.39 | seanbright | loather: PM me a DID |
| 05:39.46 | jql | originate <number> application Echo or something |
| 05:40.19 | jql | originate Zap/1/4123345678 application Echo perhaps |
| 05:40.19 | jql | :) |
| 05:40.29 | jql | with a real TN though, of course |
| 05:41.23 | seanbright | loather: i'm read with 23+ channels when you are |
| 05:41.26 | seanbright | ready* |
| 05:41.38 | jql | DOS him, quick |
| 05:41.39 | seanbright | actually, i'm lying |
| 05:41.47 | seanbright | yeah, i'm lying |
| 05:41.48 | seanbright | damnit |
| 05:41.51 | loather | ok :) |
| 05:41.53 | seanbright | toll free PRIs |
| 05:41.54 | jql | is ready for 1000+ channels, if nobody tells on me |
| 05:41.54 | seanbright | :-/ |
| 05:41.58 | [TK]D-Fender | preps a 23 B-Chan Monkey Salute |
| 05:42.12 | loather | i can give you the OCN numbers if you want |
| 05:42.51 | seanbright | and you swear to jesus these are yours? |
| 05:43.01 | loather | let me double-check that 858 number |
| 05:43.11 | jql | california? me too. :) |
| 05:43.25 | jql | whips out a 760 to go against your 858 |
| 05:43.27 | loather | oh, that would have been bad |
| 05:43.34 | loather | heh, i live in 760 area code |
| 05:43.35 | *** join/#asterisk CrashSys (n=kumba@azrael.crashsys.com) |
| 05:43.42 | jql | I live in 619 |
| 05:43.48 | jql | I work in 760 |
| 05:44.03 | loather | and yeah, i swear to jesus these are mine |
| 05:44.05 | CrashSys | 72 repruhsent! |
| 05:44.20 | CrashSys | well, it would have been funnier if I had typed the whole area code |
| 05:44.21 | seanbright | k |
| 05:44.25 | jql | heh |
| 05:44.43 | CrashSys | House of repruhsentin' |
| 05:45.00 | *** join/#asterisk freakazoid0223 (n=mattc@68.238.182.170) |
| 05:45.08 | loather | ok, is aw a bunch of activity |
| 05:45.26 | loather | they all went to voicemail |
| 05:45.28 | seanbright | i show 23 active channels |
| 05:45.37 | seanbright | and now 0 |
| 05:46.00 | seanbright | want to try 40 or so/ |
| 05:46.02 | seanbright | ? |
| 05:46.22 | loather | perfect. and some numb nuts customer called in one of the other DIDs at the same time too |
| 05:46.28 | seanbright | oh good |
| 05:46.33 | loather | looks like they hung up after they said it was closed :) |
| 05:46.37 | seanbright | heh |
| 05:46.41 | loather | but yeah, let's try 40 just in case |
| 05:46.47 | loather | watch them fall through the whole process |
| 05:46.48 | seanbright | headed your way |
| 05:46.48 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 05:47.09 | loather | i see them all |
| 05:47.09 | CrashSys | capacity testing? |
| 05:47.10 | seanbright | 40 active channels on this end |
| 05:47.17 | loather | all hit voicemail |
| 05:47.18 | seanbright | CrashSys: nfas testing |
| 05:47.28 | CrashSys | Ewww |
| 05:47.30 | CrashSys | sorry |
| 05:47.33 | loather | all disconnecting |
| 05:47.39 | seanbright | all gone |
| 05:47.40 | seanbright | yup |
| 05:47.44 | loather | wonderful, thanks so much for helping me test this |
| 05:47.48 | seanbright | my pleasure |
| 05:48.09 | CrashSys | I despise NFAS... specially when all the T's still have the 24th channel as a back-up D... what the hell is the point then? |
| 05:48.25 | jql | not all the T1s should have it. just the first two |
| 05:48.29 | seanbright | i'm off to bed |
| 05:48.31 | seanbright | night folks. |
| 05:48.45 | jql | nfas is somewhat silly for less than 3 T1s |
| 05:48.53 | jql | but still okay |
| 05:49.06 | CrashSys | I'm still not convinced you get your money out of the single extra channel considering the limitations you now have with those T1's... |
| 05:49.19 | loather | yup, i plan on expanding it out to four eventually |
| 05:49.59 | loather | so i had them build it out as nfas now |
| 05:50.12 | CrashSys | Who's the carrier? |
| 05:50.17 | CrashSys | Some are better then others :) |
| 05:50.21 | loather | :( cox communications |
| 05:50.31 | jql | omgcox |
| 05:50.49 | loather | the equipent is colocated in the same facility as my pbx though, so the t1s go maybe 30 feet |
| 05:50.51 | CrashSys | Hmmm... i'm afraid (or fortunate) that I have no experience with them... |
| 05:50.54 | jql | cox and time warner try selling some awful service plans sometimes |
| 05:50.59 | loather | well, their data services are shite |
| 05:51.13 | CrashSys | In a colo and they wont deliver SIP 30-feet away? |
| 05:51.26 | loather | but their pri service has been pretty well spot on, aside from some initial funkery |
| 05:51.54 | loather | billing certain area code customers at 900 rates for 888 numbers -- yeah, that went over well |
| 05:51.56 | CrashSys | I've dealth with TWTC a handful of times and they are horrid |
| 05:52.04 | loather | twtc's data service isn't half bvad |
| 05:52.13 | loather | i have a metro ethernet ds3 through them and it's been rock solid |
| 05:52.28 | jql | I have a loop with TW, but no transit |
| 05:52.42 | loather | but yeah, cox doesn't provide sip service |
| 05:52.46 | jql | they haven't failed at sonet... yet |
| 05:53.07 | CrashSys | TWTC for T1 was horrid |
| 05:53.20 | loather | and i'm afraid of TWTC's sip service |
| 05:53.20 | CrashSys | I know a few people with their data service and it's ok and cheap if the building is already lit |
| 05:53.57 | CrashSys | I'm talking Internet not SIP |
| 05:54.22 | loather | oh, yeah, if i'm doing internet i'll get a tier one carrier |
| 05:54.28 | loather | no sense messing around |
| 05:54.34 | loather | but for P2P links it's a different story |
| 05:57.16 | loather | anyhow, thanks everyone. maintenance was a success! |
| 05:57.21 | loather | bbl. :) |
| 05:58.13 | jql | grats |
| 05:58.47 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 05:59.29 | CrashSys | So, who's used Realtime with Extensions.conf before? I got some questions on how it scaled with load... |
| 06:06.37 | dec3pti0n | anyone here using the latest freepbx ? |
| 06:07.19 | CrashSys | I am using the latest vim? |
| 06:07.47 | kerx | how do you tell AMI to call a local channel? |
| 06:08.27 | kerx | the Originate action on AMI in the Channel: it seems like it doesn't take Local channels |
| 06:09.09 | jql | be firm; don't take any guff |
| 06:09.34 | CrashSys | tell it that it will take locals and like it |
| 06:09.46 | jql | exactlt |
| 06:09.52 | jql | s/lt/ly/ |
| 06:10.05 | kerx | lol |
| 06:10.10 | kerx | ok, let me go have a talk with my server |
| 06:10.12 | CrashSys | someone has too much time |
| 06:10.54 | CrashSys | I hate OTRS... it never forgives, and never forgets... |
| 06:11.31 | jql | marks is "Works for Me" |
| 06:12.16 | kerx | Channel: Local/start@cid88 |
| 06:12.34 | kerx | this doesn't work though, isn't it supposed to be Local/extension_name@context_name ? |
| 06:13.04 | jql | yes, that is what it's supposed to be |
| 06:13.59 | kerx | so it's basically redundant info, when sending the AMI and including Context: , Exten:, Priority: information |
| 06:14.05 | kerx | weird, let me give it a shot though |
| 06:14.19 | jql | no, those are where the dialplan goes once the callee answers |
| 06:14.24 | kerx | Response: Error |
| 06:14.24 | kerx | Message: Originate failed |
| 06:14.34 | kerx | Dec 17 22:14:06] WARNING[21366]: channel.c:3051 ast_request: No channel type registered for 'Local' |
| 06:14.34 | kerx | [Dec 17 22:14:06] NOTICE[21366]: channel.c:2898 __ast_request_and_dial: Unable to request channel Local/start@cid88 |
| 06:14.36 | kerx | weird |
| 06:14.51 | *** join/#asterisk mort_gib (n=mjensen@77.208.101.76) |
| 06:14.52 | kerx | " No channel type registered for 'Local' " |
| 06:14.53 | jql | ahh, lack of chan_local will screw you, and not gently |
| 06:15.00 | kerx | oh shiz-nit |
| 06:15.02 | kerx | slaps himself |
| 06:15.03 | jql | module load chan_local.so |
| 06:15.06 | kerx | punches himself |
| 06:15.14 | CrashSys | Anyone know of an Asterisk Distribution that allows you to control/administer a meetme conference from a web page? I.E. dial a number and join them to the conference, etc? |
| 06:15.18 | kerx | take a 4U server w/ 10 hard-drives in it and drops it on his toes |
| 06:15.34 | jql | ow. I've done that, and it hurts like a mofo |
| 06:15.37 | CrashSys | Customer sent me a contact request but ViciDial is not what he needs... looking to suggest something else... |
| 06:15.43 | jql | although I think it was 3U |
| 06:15.58 | kerx | yeah, i've done it before too but only 2hard-drives w/ raid-controller |
| 06:16.08 | kerx | it was a dual xeon though, with like 4 fan's and a floppy and cdrom |
| 06:16.11 | kerx | the case was a supermicro |
| 06:16.11 | CrashSys | I did that with a supermicro 3U with 15 hard drives in it... that was a fun 5 minutes in the data center :) |
| 06:16.13 | kerx | terrible pain |
| 06:16.17 | kerx | lol |
| 06:16.28 | jql | I've been moving rackmount servers every weekend lately. very exhausting |
| 06:16.37 | CrashSys | the data center was giving a tour to some new clients too :) |
| 06:17.22 | kerx | how stupid |
| 06:17.26 | kerx | now my CDR's have two records |
| 06:17.32 | jql | dreads the moment when the previously unknown weight of the server is loosed from its screws and put upon his poor flesh |
| 06:17.43 | kerx | the disposition on the Local channel call is set to Answered, but the billsec is 0 |
| 06:17.48 | jql | nurses his bruises |
| 06:17.53 | kerx | the other CDR still has a disposition of NO ANSWER and aa 0 billsec |
| 06:17.54 | kerx | terrible |
| 06:17.59 | kerx | hits Asterisk CDR |
| 06:18.01 | kerx | system |
| 06:18.16 | CrashSys | I'm going away from the big heavy 1U's and going to SuperMicro half-depth 1U mini-cases... the SC-503 with front-mounted IO... going to be double-racking my rack :) |
| 06:18.18 | kerx | it just can't do it |
| 06:18.27 | kerx | whoever is able to get Asterisk's CDR's to work properly |
| 06:18.28 | kerx | I bow to them |
| 06:18.45 | jql | my company managed to drop supermicros. the rails for them suck(ed) |
| 06:18.45 | CrashSys | I am able to get asterisk CDR's to work properly :) |
| 06:18.50 | jql | dell rails are sweet |
| 06:19.06 | kerx | CrashSys, how! |
| 06:19.15 | CrashSys | However, I run everything through a set of asterisk servers that act as a gateway, so everything is in then out :) |
| 06:19.29 | kerx | how did u get that gateway to work? |
| 06:19.32 | CrashSys | I gave up trying to get useful CDR's from vicidial (and it's obsessively high use of locals) |
| 06:19.49 | kerx | i started using Local, because I thought it would work |
| 06:20.00 | kerx | before I was just sending it directly to SIP/provider/xxxxnnn |
| 06:20.00 | CrashSys | I compiled mysql CDR's using asterisk-addon's... |
| 06:20.05 | kerx | I've done that also |
| 06:20.14 | kerx | Still has the Disposition set to "NO ANSWER" and 0 in the billsec |
| 06:20.18 | kerx | on all answered calls |
| 06:20.21 | CrashSys | then my context for outbound basically checks the sending caller-id, if it's allowed, issues a dial command to the PSTN |
| 06:20.23 | CrashSys | and that's it |
| 06:20.38 | kerx | exactly what i do basically |
| 06:20.41 | kerx | Dial() to the PSTN |
| 06:20.44 | CrashSys | one second, i'll pastebin my extensions and iax.conf for you |
| 06:20.54 | kerx | thanks, i appreciate it a lot |
| 06:21.11 | kerx | i was almost ready to trying to setup a OpenSER machine to Proxy all the call's through it |
| 06:21.19 | kerx | and use the acc.so module |
| 06:21.44 | CrashSys | I am stuck with Asterisk because I need to transcode from ulaw (inside) to gsm (outside) on my sip provider :) |
| 06:21.55 | jql | joy |
| 06:22.12 | codefreeze-lap | kerx: known probs in last asterisk release with getting the disposition right; might try the latest svn... are you using 1.4? |
| 06:22.19 | CrashSys | openser is just a packet router essentially |
| 06:22.25 | jql | how kind of your telco enforce downsampling. :) |
| 06:22.27 | CrashSys | and proxy/sbc |
| 06:22.58 | kerx | CrashSys, i know, but i can send all calls through it |
| 06:23.45 | kerx | codefreeze-lap, i've done svn on 1.4 and svn on 1.6, both same stuff |
| 06:24.10 | codefreeze-lap | hmmmm. I'll make another round on the cdr bugs tom. |
| 06:24.29 | kerx | do u want anything from me? |
| 06:24.31 | CrashSys | http://pastebin.ca/1288387 |
| 06:24.49 | CrashSys | the first part is my context that the dialer/server dials into on my gateway |
| 06:24.56 | codefreeze-lap | I did fix some probs; maybe they are still patches. Too tired to look; matter of fact, I'm off to sleep. |
| 06:25.00 | CrashSys | the second part is the iax.conf entry I set up on the dialer/server and the gateway |
| 06:25.29 | CrashSys | I have 3 gateways so I set the userfield equal to the gateway to know where the call came from incase I have issues I can correlate them to providers :) |
| 06:25.37 | kerx | CrashSys, i pretty much do the same |
| 06:26.02 | CrashSys | locals and CDR's get pretty damn ugly, plus with the way ViciDial dialplans are there is no clean way to do CDR's |
| 06:26.20 | kerx | i'm not using vicidial |
| 06:26.22 | CrashSys | So, I measure from the provider end, which is good cause I move all transcoding away from the dialers and just send ulaw to them :) |
| 06:26.25 | kerx | im just originating calls through AMI |
| 06:26.32 | CrashSys | Yeah, but it kind of relates :) |
| 06:26.34 | CrashSys | this is 1.4.21.2 |
| 06:26.46 | CrashSys | and addons 1.4.7 |
| 06:26.51 | CrashSys | MySQL 5.0.67 |
| 06:26.52 | kerx | i see |
| 06:27.19 | kerx | i just finished installing AsteriskNOW on another machine |
| 06:27.22 | kerx | a monster Dell 4u |
| 06:27.29 | kerx | hopefully i don't drop it on my toes :P |
| 06:27.38 | CrashSys | have fun |
| 06:27.40 | kerx | i'm gonna see how my CDR's come out on call originates |
| 06:27.43 | kerx | and see the diff. |
| 06:28.00 | kerx | CrashSys, what does _X. mean in ur dialplan? |
| 06:28.19 | CrashSys | Match any numeric number |
| 06:28.27 | CrashSys | that is at least longer then 1 digit |
| 06:28.29 | kerx | does it actually store it? |
| 06:28.33 | kerx | and use it? |
| 06:28.53 | drmessano^ | A Dell 4U? |
| 06:28.54 | CrashSys | Yes, but the CDR will write the source as recieved not as set later... |
| 06:29.00 | drmessano^ | How many users? |
| 06:29.16 | CrashSys | If the server sends 0000000000 initially that is what will get written in the CDR, even tho I set the CID later |
| 06:29.37 | CrashSys | I haven't had very good luck doing a Set(CDR(source)=7275551212) before |
| 06:30.02 | CrashSys | Plus I have a script that parses it all out from the lastdata column and formats it correctly to give me billable CDR's :) |
| 06:30.11 | CrashSys | it's all done in SQL to so it's pretty quick/easy |
| 06:31.16 | kerx | drmessano^, yeah |
| 06:31.53 | kerx | ok, i'll report if any success w/ asterisknow |
| 06:32.24 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 06:32.45 | CrashSys | Monster servers dont really equal capacity with asterisk :) |
| 06:32.57 | CrashSys | Maybe 1.6 will change that |
| 06:33.04 | jaytee | how so? |
| 06:33.25 | CrashSys | You hit a software limit on channels long before you hit a hardware limit |
| 06:33.37 | CrashSys | 300 or so channels is about where asterisk becomes finicky |
| 06:33.38 | kerx | yes |
| 06:33.58 | kerx | maximum 300 w/ dualcore and 2gb ram i was able to do, before i had call quality problems |
| 06:34.01 | jql | asterisk is sadly O(n) |
| 06:34.08 | jql | at best |
| 06:34.09 | kerx | i was using g729 tho |
| 06:34.28 | CrashSys | you scale Asterisk horizontally, not vertically :) |
| 06:34.38 | jql | although I admit, coming up with a channel hash is a painful idea |
| 06:35.00 | jaytee | 300 channels? so 200 sip phones plus 92 b channels on PRI would be pushing it? |
| 06:35.22 | jql | the number is phones is irrelevant; it's the number of active calls |
| 06:35.35 | CrashSys | Depends on concurrent call volume |
| 06:35.46 | CrashSys | 300 is active channels, not attached devices |
| 06:36.08 | jql | asterisk could probably stomach a couple-thousand phone registrations without choking. openser could go towards a couple hundred-thousand |
| 06:36.55 | jaytee | other than using CDR with SQL and some "nifty" query how would one determine what the peak concurrent call volume is? |
| 06:36.56 | CrashSys | Yeah, but the learning curve from asterisk to openSER is mind-numbing... |
| 06:37.21 | jql | failure_route[1] { omgwtfbbq } |
| 06:37.46 | CrashSys | jaytee do samples of CDR records every 15 minutes for a day, average that over a couple weeks, ?????, profit |
| 06:38.02 | CrashSys | <PROTECTED> |
| 06:38.32 | CrashSys | You could use nagios or something like that to poll the AMI and log how many channels it has active every 5 minutes and generate statistics fromt hat |
| 06:38.33 | jaytee | CrashSys, that's kind of along the lines of what I was thinking |
| 06:38.37 | CrashSys | maybe MRTG can be made to do that |
| 06:39.16 | CrashSys | Infact, I think the guy who wrote MRTG has an Asterisk Plug-In for MRTG to check the AMI, but it does it by channel-type not total channel... |
| 06:39.42 | CrashSys | But for a regular phone system your channel load should be mostly symmetrical so you probably just need to monitor SIP channels :) |
| 06:40.42 | jaytee | I have a Dell PowerEdge 2950 Quad Xeon 4GB RAM with 1 TE212P card sitting on my outbound span of a PRI and connected to an Option 11C's outbound span so calls route between the two pbx's and to the telco. All inbound calls currently go through the Option 11C. I've got a second TE212P card but haven't installed it yet. I've also got a second server for failover/redunancy. |
| 06:41.50 | CrashSys | Your highest feasible concurrent volume is 48 channels (2-T1's) |
| 06:41.51 | jaytee | I'm trying to determine the best strategy for moving forward on this as I migrate more users off the Nortel. The end goal is to decommission the Option 11C. |
| 06:42.02 | CrashSys | unless you have some kind of conference call, but it's still minimal |
| 06:42.11 | jaytee | 46 since 2 channels are D channels |
| 06:42.42 | CrashSys | Why not just a 4-port T1 card? |
| 06:43.49 | jaytee | CrashSys, I work for a non-profit. I have the skillset to make things work but I don't have the authority, budget control etc. to do things the way I think would be best. I'm basically being micromanaged into the ground. |
| 06:44.26 | CrashSys | Join the club :) |
| 06:44.41 | mort_gib | jaytee: It's always business decisions, if you can save them money.... |
| 06:44.57 | mort_gib | jaytee: Yeah, join the club ;-) |
| 06:44.59 | jaytee | I don't want to join the friggin club. I want a .44 with one bullet :-( |
| 06:46.13 | jaytee | although I really don't need to eat a gun. this project will kill me within 6 months anyways |
| 06:46.30 | jql | alcoholism and sleeping pills is the norm |
| 06:46.37 | jql | either/or. both kills |
| 06:46.57 | jaytee | I don't drink and every doctor I've ever had is an asshole that won't prescribe anything useful. |
| 06:47.04 | jql | well, time to start |
| 06:47.17 | CrashSys | self-medicate |
| 06:47.28 | CrashSys | be your own doctor... millions do it everyday :) |
| 06:47.41 | mort_gib | jaytee: Ah, then that's your problem, you need to start drinking -Pronto! |
| 06:47.41 | jql | yes. hell, even nyquil is better than nothing |
| 06:47.51 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 06:47.57 | jaytee | anyways, to get off depressing subjects..... |
| 06:48.14 | jql | can't survive in this industry without killing some braincells *somehow* |
| 06:48.19 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 06:48.24 | jaytee | so if I'm using 2 2 port cards I don't see any caveats. |
| 06:48.50 | CrashSys | I'm surrounded by under-qualified IT staffs all day... I have to dumb it down a few levels so they can sound important and justify their existence to management :) |
| 06:48.56 | jql | if there's caveats, you'd have to warn us about em |
| 06:49.09 | CrashSys | Alcohol helps me accomplish that |
| 06:49.27 | jql | I'm drinking my second bottle of wine right now. :( |
| 06:49.50 | CrashSys | preferably a flash of rum and a coke throughout the day :) takes the edge off nicely... |
| 06:50.00 | CrashSys | flash = flask |
| 06:50.06 | jql | indeed it is |
| 06:50.28 | CrashSys | Never got much into wine... I do enjoy a good beer tho... |
| 06:50.38 | jaytee | I'm just thinking when I add a second card so that I can have both may inbound T1 and my current outbound T1 to the telco both going through Asterisk to my Option 11C whether I should setup 1 card with both spans from the telco or split the spans from the telco over both cards. |
| 06:51.21 | CrashSys | Jaytee: Wont matter, just whatever method you start with stick with it... nothing will mess with your mind like changing methods half-way through an implementation :) |
| 06:52.45 | jql | I don't know the details of card timing, but that's the only consideration I'd make as far as distributing telco T1s |
| 06:52.47 | CrashSys | Is faxing going through these spans or separate copper? |
| 06:52.54 | jql | timing is a bitch |
| 06:53.28 | CrashSys | Yeah, but timing is SUPPOSED to be generated internally on the card when you play CO :) |
| 06:53.45 | jaytee | CrashSys, understood. I'm just looking for input/feedback. In the back of my mind I've got questions as to which would be the best scenario. If I put both spans to the telco on the same card and they take timing from the telco and both spans to the Option 11C providing timing to the Nortel side as PRI_NET it might be better. |
| 06:54.17 | CrashSys | I'd have port 1 on each card be a CO PRI, and port 2 be the option 11C |
| 06:54.32 | CrashSys | just based on past experience with older digium cards/drivers... |
| 06:54.33 | jaytee | currently I've only got the one 2 port card with 1 span taking timing from the telco and the other providing timing to the Nortel and it works fine. |
| 06:54.34 | jql | I'd probably do the same |
| 06:55.04 | jql | in fact, I'm doing something similar |
| 06:55.07 | jaytee | so stick with the config I have and just duplicate it with the other card |
| 06:55.28 | CrashSys | Sure, if it's working stick with it :) |
| 06:55.31 | jaytee | I've even managed to do some tricky dialplan stuff to get 4 digit dial between the 2 systems. |
| 06:55.34 | *** join/#asterisk s0lid (n=s0lid@58.185.110.126) |
| 06:56.46 | jaytee | it's a relief to hear others express the same opinion :-) |
| 06:58.00 | jaytee | in fact I feel way less stressed in just the last 5 minutes running this by you guys |
| 06:58.49 | CrashSys | Wait till you see our consulting bill |
| 06:59.02 | jaytee | lol |
| 07:00.17 | drmessano^ | hmm |
| 07:00.23 | drmessano^ | I think they suck |
| 07:00.31 | drmessano^ | Not sure who "They" is |
| 07:00.35 | drmessano^ | But yesh |
| 07:01.00 | drmessano^ | I've had nothing but problems out of "they" |
| 07:01.07 | drmessano^ | "them" and "those guys" |
| 07:01.43 | jaytee | seriously, I've been hamstrung on so many turns on this project. Initially I'd come up with a plan that my zippy the pinhead director has micromanaged into a nightmare. I'd planned on having both spans equipped before migrating more than just a small group of test users. Now I've been pushed into migrating more than I feel comfortable with just the single card scenario because my boss misplaced his testicles when it came time to request approval |
| 07:01.44 | jaytee | for another card. |
| 07:02.34 | jql | pointy-hairs need to lay off the hardware micromanagement, sometimes |
| 07:02.58 | drmessano^ | s/sometimes/always |
| 07:03.03 | jql | yeah, that |
| 07:03.27 | jql | is in the middle of a terrifying PRI -> VoIP transition. many sleepless nightsx |
| 07:03.33 | CrashSys | jaytee: You know, part of good management (or customer service if you are doing consulting or own your own business) is the ability to say no :) |
| 07:04.06 | CrashSys | Telling them no draws a line in the sane. If they still push you to do it against advisement then what can you do? |
| 07:04.19 | CrashSys | Not worry about it, and just attempt to do whatever their whims may ask of you :) |
| 07:04.21 | jaytee | CrashSys, yeah. I understand what you mean but then I'd have to introduce you to my boss before you'd fully understand the position I'm in. |
| 07:04.37 | jql | I created a phrase to handle such scenarios. "Good luck with that." |
| 07:04.49 | CrashSys | I doubt it... has your boss ever flipped out and waved a 9mm in your face? |
| 07:05.00 | jaytee | CrashSys, no |
| 07:05.05 | jaytee | has yours? |
| 07:05.07 | CrashSys | Then I win :) |
| 07:05.11 | jaytee | damn! |
| 07:05.16 | jql | CrashSys: I wish! I wouldn't need to work anymore with my multi-million dollar judgement |
| 07:05.34 | CrashSys | Yes, I had a boss that I worked for during the whole dot-boom who flipped out when I told him no that I couldn't do something and he started waving a gun around |
| 07:06.00 | jaytee | CrashSys, what did you do? |
| 07:06.14 | CrashSys | It was internet broadcast back before youtube and the like... |
| 07:06.37 | jaytee | no, I mean did you go over his head? or call the cops? |
| 07:06.38 | CrashSys | I was telling him the server was over-loaded, so he was telling me to make it not-overloaded with my magic wand, and I said no :) |
| 07:06.48 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-31-197.dsl.hrlntx.sbcglobal.net) |
| 07:06.54 | CrashSys | I just walked out the door and left |
| 07:07.08 | jaytee | brave move |
| 07:07.36 | jaytee | at my age and with this economy I might just as well commit suicide. it'd be quicker and less painful |
| 07:07.37 | TrentCreek | now who here has done PHP/AGI? |
| 07:07.38 | CrashSys | He was just attempting to initimidate me |
| 07:07.45 | jql | grow without scaling: brilliant |
| 07:08.01 | jql | this economy is teh suck |
| 07:08.13 | jql | and my credit is teh suck |
| 07:08.34 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:08.38 | CrashSys | jaytee: I'm telling you, be upfront about can and cant do's! Being wishy-washy is only going to screw you in the end |
| 07:08.53 | jaytee | CrashSys, I hear ya! |
| 07:09.11 | CrashSys | Cause they will talk you into it, then when you cant do it cause it's beyond your skill-level or whatever they will ask why you didn't say you couldn't do it... |
| 07:10.36 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 07:10.37 | jaytee | it's not a question of skill level. it's a question of my boss constantly changing the plan in the middle of it. |
| 07:11.26 | Micc | jaytee, that sounds familiar. |
| 07:11.35 | CrashSys | Then document the changes, and when he asks why, pull out your neat little log that says on monday it was X, tuesday it changed to Y, and wednesday it changed to C... |
| 07:11.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:11.46 | CrashSys | Be accountable to their mismanagement :D |
| 07:11.46 | *** join/#asterisk zaafouri (n=UniXoiDe@196.203.51.238) |
| 07:12.01 | drmessano^ | jaytee: Patch your asterisk version, ASAP |
| 07:12.02 | CrashSys | Also log your time for all the BS meetings and call conferences and pow-wow's and crap... |
| 07:12.13 | drmessano^ | Make it so custom, if they fire you, they wont be able to fix anything on it |
| 07:12.20 | drmessano^ | Then... work on getting your boss fired |
| 07:12.24 | CrashSys | Cause the next thing they tell you is that you dont put in your 8 a day :) |
| 07:12.28 | drmessano^ | With your new +20 |
| 07:12.32 | *** join/#asterisk qdk (n=qdk@79.138.242.28.bredband.3.dk) |
| 07:12.52 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 07:13.19 | Micc | I'm glad I'm going to be my own boss soon. |
| 07:13.35 | CrashSys | That's even worse man, you have no one to blame but yourself then! |
| 07:13.56 | rue_desk | remember 1 thing, "do what your best, hire the rest" |
| 07:14.08 | jql | it's hard work being incompetent |
| 07:14.13 | rue_desk | its really really really true |
| 07:14.15 | drmessano^ | Micc: You're never your "own boss" |
| 07:14.32 | rue_desk | I worked for 3 yers at 10c an hour for me |
| 07:14.33 | Micc | drmessano: I know. |
| 07:14.45 | rue_desk | I quit |
| 07:14.57 | rue_desk | and I couldn't find anyone to replace me |
| 07:15.04 | jaytee | my boss insisted that this be done so that it wouldn't be "dependent" on me. There's only so much I could do to accomodate that. Neither of the two network people working with me have any motivation to learn linux and less to learn asterisk although I've tried to train them. One had ADD and is a total screwup. |
| 07:15.07 | rue_desk | who would work at that rate |
| 07:15.40 | rue_desk | jaytee, choose the total screwup |
| 07:15.42 | drmessano^ | Micc: Owning your own business makes you the one ultimately accountable to customers, so while being the "Boss" removes some asshole over your head, it brings you to the forefront of accountability.. and now you go from 1 boss to hundreds |
| 07:15.45 | Micc | drmessano, Its funny how every little problem becomes my problem. |
| 07:15.46 | CrashSys | I'd hire another asterisk programmer if I could find one I liked... problem is I need a real asterisk programmer and not a trixbox/switchvox one... |
| 07:16.03 | rue_desk | hmm |
| 07:16.12 | Micc | drmessano, I know. I've been there before. |
| 07:16.30 | rue_desk | CrashSys, the company I work for charges $76 an hour for me to do phone work |
| 07:16.35 | jaytee | I tried to setup Polycom provisioning to be as simple as possible. I created templates and shell scripts so that all you have to do is type ./prephone "macaddress" "4digitextension" |
| 07:16.42 | rue_desk | I can do maintenance mainly now |
| 07:16.43 | Micc | I'd do contract work if I wasn't so busy already. |
| 07:17.04 | CrashSys | polycom FTP provisioning + DHCP = best thing going :) |
| 07:17.07 | jaytee | the guy screwed up typing the mac address and so the phone defaulted to downloading the 000000000000.cfg file |
| 07:17.12 | rue_desk | working on selling * systems |
| 07:17.20 | CrashSys | and if I had time to figure out snom provisioning i'd be doing that too |
| 07:17.39 | CrashSys | rue_desk: I charge my customers more to work on their systems |
| 07:18.06 | Micc | rue_desk: yeah I would charge more too. |
| 07:18.12 | jaytee | CrashSys, we're a Windows 2K3 domain. we use Cisco Catalysts setup with QoS. I have the provisioning server setup to pull DHCP from the Windows domain controller and put the phones in their own VLAN. |
| 07:18.28 | rue_desk | I was just saying, if there is remote access and you need help, we can bill ya |
| 07:18.32 | CrashSys | Jaytee: Sounds good |
| 07:19.01 | Micc | rue_desk, good to know. |
| 07:19.22 | CrashSys | rue_desk: You know vanilla Asterisk? Kernel Recompilation? D-CHannel Debug? SIP wackiness? |
| 07:19.25 | rue_desk | right now we just do keyed systems, mainly panasonic and nortel, I'm quite linux friendly |
| 07:19.29 | jaytee | but my network engineer that runs the show can't figure out how to setup the DHCP server to use OUI for which address scope to use. |
| 07:19.42 | CrashSys | OUI? |
| 07:19.44 | jaytee | and he makes 20K more than me :-) |
| 07:19.55 | rue_desk | CrashSys, for the mostpart yes, I haven't needed to do any d channel debugging |
| 07:20.15 | jaytee | Organizationally Unique Identifier, the first 6 digits of a MAC address. |
| 07:20.16 | rue_desk | I'm dealing with sip phone wackiness now |
| 07:20.29 | CrashSys | Ahhh |
| 07:20.31 | jql | I debug the d-channel whenever I need to assign the blame to not-me |
| 07:20.42 | CrashSys | jql: same here :) |
| 07:21.00 | CrashSys | sip debug as well |
| 07:21.25 | rue_desk | yea, I'm all over debugging sip, not too much to it |
| 07:21.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 07:21.49 | jql | it's sad how many "issues" disappear when you remove a firewall. very depressing |
| 07:21.51 | jaytee | Crashys, it should be possible to configure the DHCP server so that if it sees a DHCP request from a MAC address beginning in 0004f21 it knows it's a Polycom phone and what VLAN it belongs in with the correct address scopy to hand out an address from. |
| 07:21.52 | rue_desk | still not sure why my aastra phone wont register, should have that worked out in a bit |
| 07:21.58 | jaytee | s/scopy/scope |
| 07:22.21 | CrashSys | jaytee: depends on the DHCP Server, but some of then do wildcard's |
| 07:22.24 | rue_desk | sounds like a fun network |
| 07:22.32 | CrashSys | or you do a mac range of 00:00:00 to ff:ff:ff |
| 07:22.37 | jql | I have Polycom's mac prefix memorized |
| 07:22.46 | Micc | rue_desk, aastra phones are a little strange in the setup. the phone number field is really the username unless you use phone numbers for your sip contexts. |
| 07:22.49 | CrashSys | You have snom ones too? |
| 07:22.50 | rue_desk | you vlan your phone and workstation networks apart? |
| 07:22.55 | jaytee | so do i, it's 0004f21 |
| 07:22.59 | jql | 0004f21 |
| 07:23.03 | jql | not 2 zeros, 3 zeros |
| 07:23.15 | jql | some of my coworkers get confused |
| 07:23.22 | Micc | rue_desk, the username field and the phone number field need to be the same. |
| 07:23.27 | CrashSys | jql: Ahh, but, what are the MAC's of the older 301's?!?!?!? |
| 07:23.34 | rue_desk | Micc, thats not the trouble, its not even trying to register |
| 07:23.39 | jql | I don't let such filth on my network |
| 07:23.42 | CrashSys | or the IP4K's? |
| 07:23.45 | CrashSys | :) |
| 07:23.55 | rue_desk | I discovered the bad paramiter naming already |
| 07:24.01 | jql | I've been waiting for someone to buy (me) a 4k, though |
| 07:24.16 | CrashSys | YEah, polycom should really come down on the price on those things |
| 07:24.18 | rue_desk | the odd thing is, I had the aastra working through 2 firewalls to my * server at home |
| 07:24.39 | rue_desk | and I cant get it to even try to register with a local server |
| 07:24.49 | rue_desk | got the polycom working |
| 07:24.59 | Micc | rue_desk, I've had strange things happen with those aastras that nothing could fix. Then I reset back to factory defaults and re entered all the data just the same and it worked. |
| 07:25.07 | CrashSys | Just remember, forward UDP 5060:5069 and 10000:20000 to your asterisk server for it to work through a firewall, then define your external IP and internal subnets in sip.conf and have nat=yes :) |
| 07:25.45 | CrashSys | Micc: I ran into that with an old client with Aastra 480i's... they would loose their mind after 3-4 months and then require a factory reset and reprovisioning |
| 07:25.48 | Micc | rue_desk, if you've been changing a lot of settings and just doing the phone reset, its probably time to write down all your settings changes and reset to factory defaults. |
| 07:25.53 | rue_desk | yea no, throught eh two firewalls to home wan't a problem |
| 07:26.09 | rue_desk | it was the machine sitting beside the phone, off the same switch, i couldn't connect to |
| 07:26.13 | Micc | CrashSys, I sure hope that doesn't happen to my clients. They have 3 or 4 480i's. |
| 07:26.23 | rue_desk | I think I tried a reset, I'll do again next try |
| 07:26.29 | rue_desk | bedtime! |
| 07:26.31 | jql | all work and no play make Aastra something somthing |
| 07:26.47 | CrashSys | Micc: Well this client was running an old firmware and the company I was working for had a "If it's not broke, dont fix it" policy... |
| 07:27.36 | CrashSys | But that was an old Microsoft shop I left to start my own company... |
| 07:27.43 | rue_desk | the roads are sheets of ice under 4 inches of snow, tommorow is garbage day, this might not go down good |
| 07:27.51 | CrashSys | the whole microsoft mentality was "pretend there are no boogeymen" |
| 07:28.04 | CrashSys | completely reactive monitoring :) |
| 07:30.17 | jaytee | I love how people are anxious to dump their old copper PBX with it's expensive proprietary closed setup and expensive licensing and support contracts and move to Microsoft OCS with it's closed setup and expensive licensing. |
| 07:30.32 | jql | god, OCS |
| 07:30.44 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 07:31.09 | jql | it's VoIP, so it sucks 28% less! |
| 07:31.23 | Micc | CrashSys, what is your company? |
| 07:31.27 | ruben23 | hi |
| 07:31.45 | CrashSys | Hahahaha... that MicroSoft shop has OCS... for a 10-man office, the MS recommended set-up was 1 MS-SQL server, 1 Exchange Server, and 1 OCS Server :) |
| 07:31.48 | CrashSys | Micc: ViciDial Group |
| 07:32.10 | Micc | CrashSys what do you do? |
| 07:32.13 | jql | hold on while I send one quarter's profit to MS... kthx |
| 07:32.15 | jaytee | "well, you'll need a CAL for the phone, a CAL for voicemail/UM, a CAL for dialtone, a CAL for using URI." "What about MWI on the phone? do I need a CAL for that?" "No, because we didn't build in that feature" |
| 07:32.26 | CrashSys | And the OCS server had to have Raid-10 15K drives and 8-gig drives on 64-bit etc etc |
| 07:32.46 | CrashSys | 8-gig ram not drives |
| 07:33.00 | jaytee | We use Exchange 2007 UM for voicemail on our Asterisk system. |
| 07:33.20 | CrashSys | Micc: Part Owner, Programmer, Accountant, Policies and Procedures Manager, Customer Service/Billing, etc :) |
| 07:33.41 | jaytee | via sipX as a proxy to handle the udp/tcp transform |
| 07:33.51 | CrashSys | Director of Hosting... Director of Sales... Director of Hardware... |
| 07:33.59 | ruben23 | CrashSys:where is your company based... |
| 07:34.03 | CrashSys | Over-worked... under-paid... with an alcoholic tendency :) |
| 07:34.09 | jql | VP of Awesome |
| 07:34.26 | jql | Lead Alcoholic |
| 07:34.33 | CrashSys | Saint Petersburg, FL... in the Tampa Bay area |
| 07:34.53 | CrashSys | Primarily Call Centers based on Asterisk |
| 07:35.13 | CrashSys | Well, ViciDial, which uses Asterisk as the telephony engine |
| 07:35.17 | ruben23 | CrashSys:nice...actually in my company we are using vici... |
| 07:35.37 | CrashSys | Yeah, Matt Florell is one of the Owners as well... |
| 07:36.04 | jaytee | is ViciDial mostly just for call centers that do outbound calling? |
| 07:36.19 | ruben23 | CrashSys:do vici have an IRC support....? |
| 07:36.37 | CrashSys | Outbound, Inbound, Message Broadcasting, Press-1 / Survey Calling... |
| 07:36.56 | CrashSys | We're trying to release version 2.0.5 but no one wants to write the documentation/manuals :D |
| 07:37.22 | Micc | CrashSys, what kind of outbound dialing plan do you give your customers? Anyway to get away with unlimited long distance? We can't seem to work an unlimited LD deal for call centers. |
| 07:38.13 | jql | heh, unlimited calling is a gamble |
| 07:38.14 | *** join/#asterisk reneger (n=reneger@p3EE2E1AC.dip.t-dialin.net) |
| 07:38.19 | CrashSys | "Unlimited Long Distance" is a product offering much like a "Widget" is a product offering... it is not defined by it's literal meaning but as defined in their marketings... it's usually around 30K-minutes/mo :) |
| 07:38.22 | jaytee | I'm a one man operation, not only have I had to design this beast but I've had to write all the documentation for it as well as the technical how-tos and training guides for my coworkers. At least I didn't have to write the end-user training guides, our trainer did that. |
| 07:39.19 | Micc | CrashSys, so what do you charge them for outbound dialing? do you have a rate sheet online? |
| 07:39.19 | CrashSys | Micc: We do a sliding scale depending on per-minute LD usage... 2.0 to 1.3 cents per minute |
| 07:39.57 | CrashSys | if you start peaking 250K/mo we go into negotiated pricing depending on your mix and commitment level... or you can stay at 1.3 without commitment |
| 07:40.13 | Micc | CrashSys, ok, that makes sense. We are thinking of doing the same but we don't know if we need to make money on LD, we may just start the sliding scale at 1.2. |
| 07:40.23 | Micc | But it won't slide from there. |
| 07:40.38 | CrashSys | For Sub-Prime rate centers we charge the difference between or prime-rate center the the rate-center you are dialing to or recieving a call from and just charge you that for the minutes you used... |
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| 07:41.36 | CrashSys | I gave up on asterisk billing programs as they were either too generalized or didn't do exactly what we needed and have started custom-writing our own in PHP/PERL |
| 07:41.48 | CrashSys | it's rudimentary output but it's correct :) I'll eventually make it more user-friendly |
| 07:42.31 | Micc | CrashSys, sounds like you are a lot further along than we are. Would you be interested in selling some of your configs/scripts/programs? |
| 07:42.34 | CrashSys | FreeSide was the closest thing I could find to what we needed but it had too many realtime components for us |
| 07:43.04 | CrashSys | I wasn't sure how much load the Realtime would generate on rapid-dialing in Asterisk |
| 07:43.11 | Micc | Realtime will kill our scalability. |
| 07:43.48 | CrashSys | Yeah... i'm looking at setting up rsync and having a cron job pull down the static files every hour or so :) |
| 07:44.38 | CrashSys | And maybe a perl script to run from cron to query an SQL billing database and set a variable in astdb to lock dialing in/out until bill is paid :) |
| 07:44.55 | CrashSys | Micc: I'll give you the asterisk parts... nothing that special there really |
| 07:45.46 | CrashSys | A good chunk of it is just highly-hacked code from here and voip-info.org anyways :) |
| 07:49.00 | CrashSys | I'll tell you what will help you a lot: A good SIP provider that wont jerk you around and a good independant data center with at least 2 tier 1 bandwidth providers in a LPR bandwidth mix (Least Path Routing) |
| 07:49.27 | CrashSys | Our color has Level3 and Global Crossing providing bandwidth, as well as FPL Fibernet and Time Warner... |
| 07:49.42 | CrashSys | s/color/colo |
| 07:50.34 | jql | good stuff |
| 07:51.35 | jaytee | Time Warner is our telco. I've dealt with so many others and I like TW. Their NOC people are very cooperative. |
| 07:52.20 | jaytee | ATT, SBC and Pac Bell all suck ass. US West used to be ok but when they got gobbled up by Quest it went downhill in a hurry. |
| 07:52.39 | CrashSys | Yeah, we were initially in a qwest colo, it was horrible |
| 07:52.52 | jql | poor qwest |
| 07:53.05 | CrashSys | We colo with E-Solutions in Tampa |
| 07:53.23 | CrashSys | in a carrier hotel so we have access to about 11 providers if we want |
| 07:54.36 | jaytee | bet there's at least 2 or 3 Tarus STA-6's in a locked room there :-) |
| 07:55.21 | CrashSys | could be |
| 07:56.13 | jaytee | "shhh, someone's listening!" |
| 07:56.34 | ruben23 | :-D |
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| 07:58.14 | jaytee | well, I think I'm going to try and get a couple hours shuteye before I have to getup and go back into the suck factory again. |
| 07:58.27 | jaytee | nite CrashSys |
| 07:58.36 | jaytee | nite ruben23 |
| 07:58.40 | CrashSys | Nite, good luck with your stuff |
| 07:58.47 | jaytee | thanks |
| 07:59.10 | ruben23 | hi im compiling zaptel-1.4.9 got this error:http://pastebin.com/m412fba8c |
| 07:59.44 | CrashSys | type "make menuconfig" and remove XPP support |
| 07:59.48 | CrashSys | then recompile... |
| 07:59.59 | CrashSys | well, do a make clean then recompile :) |
| 08:00.06 | CrashSys | Xorcom USB Channel Bank |
| 08:04.56 | CrashSys | Hey, that's kewl, I just figured out that FileZille will authenticate through Pageant for SFTP :D |
| 08:05.50 | CrashSys | err filezilla |
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| 08:06.27 | Micc | CrashSys, we use vitelity for our provider. |
| 08:06.41 | CrashSys | I use them |
| 08:06.55 | Micc | I've used them for years on other projects. always seems to work ok. |
| 08:07.06 | CrashSys | They work with me pretty good... let me hook up directly to their SBC's instead of the normal public ones |
| 08:07.12 | Micc | they were exgn before. |
| 08:07.20 | CrashSys | never used them then |
| 08:07.28 | CrashSys | Also doing inter-op testing with Gafachi |
| 08:07.47 | CrashSys | I prefer to go through other people's mixes because it's easier to hide a call center :) |
| 08:07.53 | Micc | We're doing interop testing with broadvox, but they have problems which they haven't resolved. |
| 08:07.56 | kerx | is it possible to add SIP header's in the Originate ? |
| 08:07.59 | kerx | through AMI? |
| 08:08.08 | CrashSys | a lot of direct providers are really get picky when you hammer their networks |
| 08:08.33 | Micc | CrashSys, what do you mean by they let you hook up directly to their SBC's? |
| 08:08.37 | CrashSys | kerx: I think you'd have to establish a channel first |
| 08:08.45 | CrashSys | Micc: I dont use outbound.vitelity.net |
| 08:08.45 | kerx | so it must be in the dialplan |
| 08:08.50 | kerx | CrashSys, thanks |
| 08:09.12 | CrashSys | I bypass that and plug into their session border controllers directly |
| 08:09.16 | Micc | CrashSys, they've been telling me they will switch me to a different host, but it hasn't happened yet. |
| 08:09.30 | Micc | Did you sign their carrier agreement? |
| 08:09.33 | CrashSys | the outbound.vitelity.net does channel limits and stuff |
| 08:09.36 | Micc | the 299$ thing? |
| 08:10.02 | CrashSys | dont think so... that the take or pay deal? |
| 08:10.17 | Micc | its 299$ setup and you get better pricing. |
| 08:10.48 | Micc | they just want you to have at least 299$ per month in 90 days. |
| 08:10.55 | CrashSys | I didn't pay anything but I have a wholesale acct |
| 08:11.10 | Micc | CrashSys, I got screwed then. |
| 08:11.21 | CrashSys | Micc: Yeah, but how many minutes/mo you doing? |
| 08:11.30 | Micc | CrashSys, none yet. |
| 08:11.38 | CrashSys | They also like saying they have us around and we've sent lots of vicidial business to them |
| 08:13.20 | Micc | We're having a problem getting local calls cheap. |
| 08:13.33 | Micc | It looks like we'll have to do our own PRI. |
| 08:13.49 | Micc | I'd rather pay a third party to do the PRI hosting. |
| 08:13.56 | CrashSys | we get inbound cheaper then outbound unless it's not a prime rate center :) |
| 08:14.30 | Micc | CrashSys, if your doing call centers they do a lot of LD anyways, so local doesn't matter much. |
| 08:14.41 | Micc | I figure about 85% of our calls will be local. |
| 08:14.55 | Micc | So it will make sense for us to have a low cost local option. |
| 08:15.15 | CrashSys | Yeah, i'm lucky if 1-2% of our calls would be considered local |
| 08:15.21 | CrashSys | so we just do all 1+ dialing |
| 08:15.48 | Micc | I'm doing a proposal for an 8 agent call center. |
| 08:15.54 | Micc | Thats doing dialouts. |
| 08:16.19 | Micc | And they want to do multi-line dialouts per agent, where they can hit a button and have it offload that call while it leaves a message. |
| 08:17.02 | CrashSys | Mmmm, not sure I follow you 100% |
| 08:17.16 | CrashSys | but I think it's just semantics getting in the way |
| 08:19.11 | Micc | We'll be making the calls from a web-site which will have a button to leave a message which will allow them to make another dial without waiting. |
| 08:19.23 | Micc | We're itegrating with a local CRM vendor. |
| 08:20.10 | Micc | I need to learn how to use Microsoft InfoPath |
| 08:20.20 | Micc | I need to make a new customer information form. |
| 08:20.26 | CrashSys | Yeah, that's somewhat outside my companies core focus |
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| 08:34.13 | ruben23 | CrashSys:hi...i already make clean still got the error...:-( |
| 08:35.23 | ruben23 | CrashSys:and also with make menuconfig got this: http://pastebin.com/m771faa03 |
| 08:36.13 | CrashSys | Install ncurses to use the menu interface! |
| 08:36.18 | CrashSys | :) |
| 08:37.40 | ruben23 | already installed on the system....... |
| 08:37.58 | CrashSys | well it doesn't think so |
| 08:38.05 | CrashSys | is this redhat/centos/fedora? |
| 08:38.15 | ruben23 | yes its centos 5 |
| 08:38.21 | CrashSys | You are on your own |
| 08:38.35 | CrashSys | I've never had any luck with RedHat distro's |
| 08:38.51 | CrashSys | ask in a #redhat channel, i'm sure there's some reason it's not finding ncurses |
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| 08:39.25 | ruben23 | see this http://pastebin.com/m2fe16e98 |
| 08:40.04 | ruben23 | ok...thanks.. |
| 08:40.18 | CrashSys | type "make clean && ./configure" then try make menuconfig |
| 08:40.22 | CrashSys | see if that helps |
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| 08:44.32 | ruben23 | <PROTECTED> |
| 08:45.51 | CrashSys | http://www.letmegooglethatforyou.com/?q=asterisk+centos+ncurses |
| 08:46.55 | CrashSys | look at third result :) See if that helps |
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| 09:05.49 | ruben23 | CrashSys:is it ok to compile asterisk with usr/local or usr/src.. |
| 09:06.30 | CrashSys | doesn't matter |
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| 09:07.18 | TrentCreek | now who here has done PHP/AGI? |
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| 09:15.55 | Karlitoo | good morning all |
| 09:17.10 | Karlitoo | and to all the people having problems with asterisk I hope you solve them today |
| 09:17.12 | Karlitoo | :) |
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| 09:36.59 | kerx | anyone use STRFTIME() command ? |
| 09:37.21 | mort_gib | Morning Karlitoo |
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| 09:41.47 | yang | hi mort_gib |
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| 09:42.38 | mort_gib | Hi Yang |
| 09:43.54 | mort_gib | Just found out that the FSC (local Financial Services commission) has changed so ALL traders are required to record telephone conversations |
| 09:43.57 | mort_gib | :-) |
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| 09:45.51 | yang | got url ? |
| 09:46.16 | mort_gib | yang: -To their website?? |
| 09:46.28 | yang | Is it an article on the web? |
| 09:46.44 | mort_gib | well, some of my clients are traders... |
| 09:47.04 | yang | Here its illegal to record the conversations without previous notice |
| 09:47.29 | mort_gib | UK is doing the same "shortly" So the local rascals though Gib would be, sort of half a century later :-) |
| 09:47.44 | yang | Usually it goes like this "We record conversations for quality assurance, please hang up if you don't like it...) |
| 09:48.10 | mort_gib | I suppose it's the same here, but as this is a requirement that law is not applicable.... |
| 09:50.13 | kerx | is there a function to know when line has been picked up? |
| 09:50.31 | mort_gib | Dialstatus |
| 09:50.34 | kerx | thx |
| 09:50.45 | mort_gib | Will tell you if call failed |
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| 10:06.49 | TrentCreek | now who here has done PHP/AGI? |
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| 10:25.13 | TrentCreek | now who here has done PHP/AGI? |
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| 10:28.01 | creativx | so, incidently, has anyone here ever had "file -s /dev/sdb1" return "/dev/sdb1: data" ? |
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| 10:31.59 | TrentCreek | never heard of it |
| 10:32.11 | creativx | magic partition heh |
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| 10:38.09 | Karlitoo | Morning mort_gib, sry for the late response was working |
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| 11:21.54 | viraptor | could anyone tell me if they can get balanced CPU utilisation with asterisk on more than 2 cores? |
| 11:22.54 | viraptor | I experience pretty good spread on 2 cpus, but a lot lower on no. 3 and 4 in a 4 cores machine, so I was wondering if that's my config-, or asterisk-specific |
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| 11:24.57 | xrmx__ | hi, a bit ot question: does anyone know how auto dial function is called in polycom phones? with auto dial i mean dial a number without pressing the dial button |
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| 11:26.46 | tokozedg | how can i make so that if caller number length is 6 dial exten 10 and if callerid length is another than 6 dial 20? |
| 11:26.55 | tokozedg | using GotoIf |
| 11:27.30 | tokozedg | GotoIf($["${CALLERIDLEN}" = "6"]?yes:no) |
| 11:27.42 | tokozedg | but it doesn`t works |
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| 11:31.41 | Poincare | I'm looking for 'remote asterisk hands' in Mexico, anyone? |
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| 11:34.47 | tokozedg | Poincare, what you want to do? |
| 11:36.26 | Poincare | I need someone to go onsite to a customer in Mexico and connect/check an Asterisk server |
| 11:37.50 | tzafrir_laptop | viraptor, in top, press 'H' (shift-h) to enable display of threads |
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| 11:38.22 | tzafrir_laptop | If there are two specific threads taking most of the CPU time, you can't really spread it better |
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| 11:39.55 | yang | hello tzafrir_laptop |
| 11:43.09 | kerframil | hi. does anyone have any thoughts as to the relative merits of x86 vs x88_64 for asterisk? I need to buy a new box and normally buy Opteron hardware but I'm wondering as to whether it's a comfortable fit for asterisk (in particular, I'm thinking about SIMD instruction support in assembler paths, if any, that kind of thing) |
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| 11:46.25 | viraptor | tzafrir_laptop: one thread does most of the work - it's a system with a very big realtime dialplan, no registrations - do you know if that's our current config? or is it just the way asterisk works? |
| 11:47.11 | viraptor | or can I somehow check what that thread is doing for most of the time? |
| 11:47.15 | tzafrir_laptop | viraptor, thst's surprising. I thought realtime did all the work in the context of the call thread |
| 11:49.26 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 11:50.33 | viraptor | well - I don't know what that thread is doing, so I'm not sure if that's a realtime's, or nic's, or something else's problem... |
| 11:52.44 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
| 11:56.28 | tzafrir_laptop | viraptor, what's the name of that process ? |
| 11:56.54 | tzafrir_laptop | and if you really have no clue, strace is your friend: strace -p PID |
| 12:01.25 | viraptor | hmmm... it's just another 'asterisk' |
| 12:01.57 | viraptor | it's 5%, so acutally 20% of cpu1 |
| 12:03.24 | viraptor | can I somehow get that info from the asterisk itself? I don't want to strace a live process on a production system ;) |
| 12:05.19 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
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| 12:19.22 | *** join/#asterisk andresmujica (n=andresmu@201.244.105.224) |
| 12:29.57 | viraptor | tzafrir_laptop: ok - found out that the process does logging, networking and mysql all in one thread... I'm starting to think that something's messed up in this system - it's not right behaviour, right? |
| 12:30.24 | tzafrir_laptop | viraptor, is it an AGI of some sort? |
| 12:30.38 | *** join/#asterisk phpboy (n=shane@196.36.108.18) |
| 12:31.00 | phpboy | hey guys, how do I make asterisk only allow one call through to an extension on my network |
| 12:32.33 | viraptor | tzafrir_laptop: the porcess/thread itself is the 'asterisk' process, but yes - we run ~20 AGIs at the time |
| 12:33.15 | viraptor | but I see a write("SELECT * FROM rt_extensions WH... as well there |
| 12:34.56 | yang | phpboy: check the GROUP_COUNT syntax - GotoIf($[${GROUP_COUNT(pager)} > 1]?hangup) |
| 12:35.26 | viraptor | and definitely both SIP and rtp packets are visible on an strace of that thread... I tried a couple of the other ones and they're idle (or at least not syscalling) |
| 12:37.44 | phpboy | yang: can I check if an extension is already on a call, this will be easier on my dialplan |
| 12:37.47 | phpboy | ? |
| 12:37.53 | yang | I wonder if this stuff will work as they promise - http://www.provu.co.uk/ipvideo_bvp8882.html |
| 12:38.28 | yang | I am mostly interested about the video... |
| 12:38.30 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 12:38.42 | scruz | hello |
| 12:39.15 | phpboy | hi |
| 12:41.13 | scruz | i'm using Asterisk 1.2 and connected to it from Asterisk.NET using the AMI (successfully). i'm trying to make calls from my application, but Originate basically fails. the extension i'm calling may even ring very briefly and then hangup. can anyone help me with tracing where the problem is? |
| 12:42.54 | scruz | or you could just tell me how i might make multiple automated calls to phone numbers, play a voice message, then quit ;) |
| 12:43.17 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:44.17 | phpboy | scruz: so you say it does in come cases ring briefly? |
| 12:44.25 | scruz | yes it does |
| 12:45.04 | yang | scruz: 1.2 is outdated, you should upgrade |
| 12:45.12 | scruz | i've got it to ring until i picked up, but that was only once |
| 12:45.48 | scruz | it's the office pbx. since we use it for all our outgoing calls, they wouldn't appreciate me messing with it :) |
| 12:47.57 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:51.42 | scruz | "it works. why fix it?" - or something like that |
| 12:57.26 | *** join/#asterisk sasargen (n=chatzill@174-153-210-50.pools.spcsdns.net) |
| 13:17.12 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:17.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:20.31 | viraptor | anyone else? any ideas why everything seems to happen in only one thread? (mysql, sip, rtp, logging, ....) |
| 13:20.57 | scruz | ? |
| 13:21.09 | scruz | any experts with AMI here? |
| 13:22.52 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 13:23.08 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
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| 13:30.05 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 13:32.30 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
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| 13:42.12 | *** join/#asterisk etfonhomey (n=chatzill@74-143-192-77.static.insightbb.com) |
| 13:43.02 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 13:43.14 | verywiseman | what is meaning of "Call Parking" |
| 13:43.53 | scruz | i'm trying to make a call with originate via AMI. my call doesn't display in the console. needless to say, it fails. can anyone give me a reason for this? |
| 13:44.24 | phpboy | scruz: that tells me it's working? |
| 13:44.48 | glaz | scruz: how do you do it? |
| 13:45.05 | phpboy | glaz: you issue commands to the AMI |
| 13:45.08 | scruz | do what? |
| 13:45.36 | glaz | phpboy: duh, I'm asking him how he does it so I might be helpful and see what's wrong. |
| 13:45.44 | phpboy | scruz: when you picked up did it originate the call? |
| 13:45.48 | phpboy | glaz: my bad :P |
| 13:46.07 | scruz | i'm using asterisk.net, a .net port of asterisk-java |
| 13:46.24 | glaz | :\ |
| 13:49.27 | scruz | glaz: here's the trace from the debug window in SD: http://pastie.org/342262 |
| 13:51.16 | phpboy | scruz: why not use the AMI? |
| 13:52.23 | scruz | directly? asterisk.net abstracts it |
| 13:53.21 | phpboy | scruz: please paste the originate code, as that pastebin doesn't help all that much :/ |
| 13:53.57 | phpboy | scruz; from what I can tell, there's DEFINITELY something wrong with your code :( |
| 13:53.59 | scruz | ok |
| 13:55.21 | phpboy | pastebin it |
| 13:58.45 | *** join/#asterisk DarkRift (n=dark@65.92.170.105) |
| 14:01.51 | scruz | http://pastie.org/342262 |
| 14:04.53 | phpboy | scruz: that will NOT work :( |
| 14:05.00 | phpboy | I'll paste you the AMI version |
| 14:05.10 | phpboy | just remove the PHP code around the actual commands |
| 14:05.12 | phpboy | please hold |
| 14:05.14 | scruz | odd that if i capitalize the command, it doesn't work |
| 14:05.27 | phpboy | but it doesn't work period? |
| 14:06.12 | scruz | yeah. but if i capitalize the action, it throws an error about a missing action |
| 14:06.38 | scruz | i'm more-or-less doing copy-and-paste from the book |
| 14:06.51 | phpboy | :( |
| 14:07.10 | phpboy | http://pastie.org/342271 <---- You should be able to work out the rest from that |
| 14:07.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:10.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:10.42 | scruz | that's a problem |
| 14:10.46 | phpboy | ? |
| 14:11.03 | scruz | i want to call an extension on the same asterisk server |
| 14:11.12 | phpboy | which is fine |
| 14:11.47 | scruz | via SIP, so i would normally do something like this (dialplan version): SIP/127.0.0.1/${EXTEN} |
| 14:12.12 | scruz | in other words, the channel would have to be: SIP/127.0.0.1, right? |
| 14:12.44 | phpboy | no |
| 14:13.09 | phpboy | SIP/<ext> |
| 14:13.21 | phpboy | SIP/3000 |
| 14:13.31 | phpboy | that will dial ext 3000 |
| 14:13.35 | scruz | then what data do i supply for the Exten param? |
| 14:14.12 | scruz | since in the dialplan i'd rather just do Dial(SIP/3000), like you said |
| 14:14.39 | phpboy | hang on |
| 14:14.46 | phpboy | I'll hack up a real example for you |
| 14:15.03 | file | "an extension" can mean two things... either an extension in the dialplan in which case you can not use SIP, but you would use Local |
| 14:15.13 | file | or a device configured in sip.conf |
| 14:15.22 | *** join/#asterisk itguru (n=itguru__@host81-134-10-140.in-addr.btopenworld.com) |
| 14:16.04 | scruz | file: a device? i've the extension i want to dial defined in sip.conf |
| 14:16.24 | file | scruz: that is a device, in Asterisk an extension is a set of logical steps that execute applications |
| 14:16.33 | file | scruz: one of the steps may include calling a device |
| 14:16.43 | itguru | is confused, I have a handset that can accept calls, it just can't make them - everytime I do, i get an engaged tone, any ideas? |
| 14:17.07 | scruz | file: now you're completely clashing with Asterisk:TFOT |
| 14:17.10 | scruz | :) |
| 14:17.23 | file | scruz: in which case SIP/3000 will call device 3000, no exten needs to be provided |
| 14:17.24 | phpboy | scruz: http://www.pastie.org/342276 |
| 14:17.31 | phpboy | that laid out as easy as pie |
| 14:17.50 | phpboy | scruz: that example is 3000 calling 3069 |
| 14:18.22 | scruz | phpboy: i was already logged in for the pastie |
| 14:18.31 | phpboy | huh? |
| 14:18.32 | scruz | i wasn't doing a batch |
| 14:18.50 | phpboy | uhm, ok |
| 14:19.06 | phpboy | anyhoo, check my lastest pastie and you'll get the hang of it, very easy |
| 14:20.12 | phpboy | scruz: does it work? |
| 14:20.40 | scruz | i guess this highlights my lack of understanding. i thought Channel was the extension you wanted to call |
| 14:21.52 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 14:22.40 | phpboy | scruz: nope, that's the channel you're calling from |
| 14:22.44 | phpboy | anyhoo, did it work? |
| 14:23.39 | scruz | invalid channel...hold up a few minutes. thanks for the help so far |
| 14:24.02 | scruz | what values can i supply for channel? devices specified in sip.conf? |
| 14:24.03 | *** join/#asterisk kerx (n=prepro@adsl-68-125-34-97.dsl.irvnca.pacbell.net) |
| 14:24.18 | kerx | hi, anyone know why GotoIf() doesn't get executed after a Dial() ? |
| 14:24.29 | phpboy | scruz: which two extensions are you trying to originate between? |
| 14:24.47 | phpboy | also, pastie the CLI output after attempting the code I gave you |
| 14:25.46 | scruz | http://pastie.org/342262 |
| 14:26.28 | scruz | i'm trying to call from 3595009 to 3590003, which in the dialplan will be converted to 013590003 |
| 14:26.36 | TrentCreek | oh..we have a phpboy |
| 14:26.51 | phpboy | TrentCreek: :D |
| 14:27.12 | TrentCreek | give me example of sending PHP varible to Asterisk AGI |
| 14:27.14 | phpboy | scruz: does 3595009 exist in sip.conf? |
| 14:27.26 | TrentCreek | a DIAL command of a phone number entered |
| 14:27.40 | phpboy | TrentCreek: please hold |
| 14:27.50 | phpboy | scruz ? |
| 14:28.41 | TrentCreek | oops..he is gone |
| 14:28.53 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:29.26 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 14:29.39 | scruz | sorry, didn't realize my connection went off |
| 14:30.30 | scruz | is multitasking and making a horrible job of it |
| 14:30.36 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:31.25 | scruz | phpboy: did my pastie make any sense? |
| 14:31.30 | phpboy | scruz: does 3595009 exist in sip.conf? |
| 14:31.35 | scruz | yep |
| 14:31.48 | phpboy | hmmm, is it registered? |
| 14:31.57 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 14:32.13 | scruz | not, it's not atm |
| 14:32.19 | scruz | *no |
| 14:32.41 | scruz | no SIP client using it atm |
| 14:32.58 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:34.43 | phpboy | scruz: that's why it doesn't work |
| 14:34.53 | phpboy | both phones need to be registred |
| 14:35.04 | phpboy | *registered |
| 14:36.37 | scruz | i thought having the softphone registered would be a problem. okay, let me start it first |
| 14:37.57 | scruz | freaking slow-ass computer |
| 14:39.01 | scruz | is running 3 .net apps, chrome, putty, two console windows... on 512MB :) |
| 14:39.05 | TrentCreek | scruz: you can rent a decent VPS for only $15 US a month |
| 14:39.53 | TrentCreek | that is cheaper than running a box at hoem sucking up all our electricity |
| 14:40.47 | scruz | i'm not at home. i'm at work |
| 14:41.30 | scruz | and i can't possibly be sucking up your electricity - we're on different continents, if i'm right |
| 14:41.59 | TrentCreek | typo..YOUR |
| 14:43.13 | *** join/#asterisk [gnubie] (n=[gnubie]@cm248.omega113.maxonline.com.sg) |
| 14:44.00 | scruz | okay, i'm officially pissed off. http://pastie.org/342262 |
| 14:44.35 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:45.44 | tzafrir_laptop | scruz, Action: Originate |
| 14:45.50 | tzafrir_laptop | Is it case sensitive? |
| 14:46.16 | scruz | it already complained when i made it capitalized |
| 14:46.19 | kerx | hi, does Dial() if dialed party pick's always execute? I am trying to understand why if I have a Dial() and a GotoIf(), the GotoIf never get's executed? |
| 14:46.54 | tzafrir_laptop | That error message comes from \n\n or so |
| 14:47.08 | tzafrir_laptop | maybe you forgot \r ? |
| 14:47.22 | *** join/#asterisk freckle (n=chatzill@sokar.enta.net) |
| 14:49.55 | kerx | anyone know if i can do this somehow: |
| 14:49.56 | kerx | http://pastebin.ca/1288626 |
| 14:50.29 | kerx | it seems like there is no way to get a successfull $DIALSTATUS of ANSWER, because on a Dial() if it becomes answered it never executes further in the extension |
| 14:50.29 | scruz | phpboy: it seems channel is the 'line' you're calling |
| 14:50.34 | kerx | any suggestions would be appreciated |
| 14:51.12 | *** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU) |
| 14:51.13 | scruz | because i wanted to make a call as though i were using the softphone, and the call just came in - *to* the softphone :) |
| 14:51.18 | *** join/#asterisk etfonhomey (n=chatzill@32.179.18.86) |
| 14:51.29 | scruz | so the Channel is the line to call |
| 14:52.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:53.11 | kerx | danyone can help me? |
| 14:53.14 | kerx | *anyone |
| 14:53.34 | phpboy | scruz: do you know what originate does? |
| 14:53.58 | phpboy | channel is the phone the call will ultimately be coming FROM |
| 14:54.55 | scruz | i'm faced with a different reality right here, unfortunately. and the docs seem to agree with my reality :) |
| 14:55.54 | phpboy | well, the thing is |
| 14:56.00 | phpboy | the docs aren't doing the trick |
| 14:56.01 | scruz | i used the address of the softphone (3595009) as the channel, and tried to call extension 3590003 (a desk ip phone). the call came in to the soft phone |
| 14:56.16 | phpboy | yes, then answer it |
| 14:56.22 | scruz | and it's consistent with the results i got since i started this yesterday |
| 14:56.24 | phpboy | and it will then call the desk phone |
| 14:57.12 | [gnubie] | which do you prefer for a sip traffic between branches: ipsec vpn or ssl vpn (e.g. openvpn) and why? |
| 14:57.36 | phpboy | [gnubie]: I presonally use L2TP |
| 14:57.42 | phpboy | SIP over LT2P |
| 14:57.44 | kerx | http://pastebin.ca/1288626 |
| 14:57.49 | kerx | anyone know if the above is possible? |
| 14:58.25 | [gnubie] | phpboy: ok.. but if you were to choose between the two choices above, which one would you choose and why? |
| 14:58.46 | phpboy | scruz: you with me? |
| 14:59.38 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 15:00.01 | rue_desk | <PROTECTED> |
| 15:00.29 | kerx | rue_desk, it's weird, when I do the Dial() the GotoIf never get's executed after I pick up the phone |
| 15:00.39 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 15:00.53 | kerx | good morn' TK |
| 15:01.06 | rue_desk | kerx, http://pastebin.ca/1288626 you noticed you duplicated the priority number? |
| 15:01.20 | kerx | oh, please n/m that |
| 15:01.24 | kerx | i was typing it out on pastebin |
| 15:01.32 | scruz | yeah...just looking up the book again. says the channels is the name of the channel to call. then the call gets passed to an application or the Exten/Context/Priority |
| 15:01.32 | kerx | i actually have 1,n on my real diaplan |
| 15:02.03 | kerx | my GotoIf never get's executed |
| 15:02.06 | scruz | *channel |
| 15:02.11 | file | Dial both dials and bridges calls, it will not return unless you set the Dial option to continue in the dialplan after the called party has hung up |
| 15:02.36 | rue_desk | I dont think it ever gets to the goto |
| 15:02.37 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 15:02.38 | kerx | file, Ouch, is it possible to have it continue? |
| 15:03.02 | file | kerx: the way you want? there is no way exposed |
| 15:03.03 | kerx | Or it it possible for me to set up the dialplan somehow to have an exact Timestamp of when the Dial'ed party picked up the line? |
| 15:03.22 | kerx | I am building my own CDR's in the dialplan because Asterisk CDR's are not reliable |
| 15:03.24 | file | you can execute a Macro on the called party |
| 15:03.28 | rue_desk | dial either needs to be given a timeout in which the connecting of the call has failed or it connects the call and the statmachine stops |
| 15:03.34 | file | but that is not the right channel I'm afraid |
| 15:03.47 | kerx | file: what kind of macro? |
| 15:04.08 | file | a macro, a dialplan macro ... |
| 15:04.19 | phpboy | scardinal: and that is 100% correct |
| 15:04.26 | phpboy | scruz: that is 100% correct |
| 15:04.33 | rue_desk | kerx, what are you trying to do? |
| 15:04.34 | phpboy | it calls channel A to call channel B |
| 15:04.45 | kerx | rue_desk, create CDR's |
| 15:04.48 | kerx | in the dialplan |
| 15:04.49 | rue_desk | ?? |
| 15:05.04 | scruz | phpboy: then what's channel B? |
| 15:05.05 | kerx | i want to log the Answer, to the Hangup |
| 15:05.09 | kerx | in a database |
| 15:05.20 | [TK]D-Fender | kerx: Fix your expression |
| 15:05.20 | kerx | in timestamps |
| 15:05.24 | phpboy | scruz: let's rather look at this from another angle, what are you trying to achieve? |
| 15:05.29 | rue_desk | ah |
| 15:05.34 | kerx | [TK]D-Fender, which expression? |
| 15:05.44 | [TK]D-Fender | kerx: in your GotoIf |
| 15:05.52 | scruz | i want to make a bunch of automated calls |
| 15:06.05 | phpboy | 3595009 = channel A | 3590003 = channel B |
| 15:06.09 | kerx | [TK]D-Fender, unfortunately it seems that the GotoIf never get's executed if someone pick's up the line |
| 15:06.11 | scruz | call subscribers from a list, play a voice message, hang up |
| 15:06.21 | kerx | [TK]D-Fender, so what it seems like I am trying to do will never work |
| 15:06.26 | phpboy | so originate will call Channel A and then connect it to Channel B |
| 15:06.30 | [TK]D-Fender | kerx: Pick up what line? |
| 15:06.44 | kerx | I want the following use-case w/ my dialplan |
| 15:06.45 | [TK]D-Fender | kerx: (The GotoIf is still bad of course) |
| 15:06.48 | kerx | Dial() |
| 15:06.51 | rue_desk | there is a way, cause I know of doing a quality survey after the call is over |
| 15:06.52 | scruz | and Channel B would just happen to be Exten, right? |
| 15:07.00 | kerx | If answered, log the timestamp in a database |
| 15:07.11 | kerx | If not answered, send to the Hangup() extension, which log's the timestamp of the hangup |
| 15:07.35 | [TK]D-Fender | kerx: "core show application dial" <- M() |
| 15:07.38 | kerx | It seems like w/ a Dial() the dialplan doesn't continue |
| 15:07.56 | [TK]D-Fender | kerx: it can, depending on the OPTIONS you give it. |
| 15:08.07 | phpboy | scruz: correct |
| 15:08.14 | [TK]D-Fender | kerx: And a proper knowledge of Asterisk Standard Extensions. |
| 15:08.24 | phpboy | the exten you're trying to dial, be it internal or external |
| 15:08.33 | file | [TK]D-Fender: if he uses M it'll execute on the called channel, not the calling... and you can't exchange vars and info... |
| 15:08.33 | phpboy | ultimately trying to dial |
| 15:08.59 | file | although depending on how you architect it you might not keep state, might just write it directly out to the db in which case you could use an inheritable variable to the dialed channel to associate it |
| 15:09.09 | [TK]D-Fender | file: He just said he wants to make a log entry. That's more than fine for a System() call, etc |
| 15:09.11 | kerx | [TK]D-Fender, hrmm. interesting :) I see. Let me investigate on this, and try some stuff out. Will report |
| 15:09.17 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:09.24 | [TK]D-Fender | file: Who said anything about PROMPTING for anything? ;) |
| 15:09.24 | kerx | [TK]D-Fender, thanks again . btw, good morning to u again :) |
| 15:09.31 | [TK]D-Fender | file: That box is too small for you! |
| 15:09.46 | rue_desk | g - Proceed with dialplan execution at the current extension if the |
| 15:09.46 | rue_desk | <PROTECTED> |
| 15:09.47 | kerx | [TK]D-Fender, Execute the Macro for the *called* channel before connecting to the calling channel. |
| 15:09.56 | phpboy | scruz: We on the same page? |
| 15:10.01 | scruz | hmm |
| 15:10.06 | file | [TK]D-Fender: true, and I didn't mention prompting... I have no idea what you are talking about now :P |
| 15:10.15 | kerx | [TK]D-Fender, looks like it's executing something before the channel is "connected" |
| 15:10.21 | [TK]D-Fender | kerx: SAME SHIT. Just put your logging line in that macro |
| 15:10.22 | kerx | I wonder what it means by "connected" ? ? ? |
| 15:10.32 | kerx | SHIT SAME? |
| 15:10.33 | kerx | ok |
| 15:10.36 | scruz | but the phone doesn't ring on the called party (B) |
| 15:10.37 | kerx | SESAME SHIT |
| 15:10.59 | [TK]D-Fender | kerx: with SPRINKLES |
| 15:11.02 | kerx | heh |
| 15:11.36 | [TK]D-Fender | file: Oh.... and Mythbusters have crushed the idiom of "turd polish". You CAN polish shit.... |
| 15:11.44 | kerx | exten => 1,n,Dial(SIP/provider/number,M(phoneisanswered)) |
| 15:11.49 | [TK]D-Fender | file: To "high gloss" no less |
| 15:11.51 | kerx | exten => 1,n(phoneisanswered),.... |
| 15:11.54 | kerx | does that look right? |
| 15:11.57 | kerx | along those lines |
| 15:12.02 | [TK]D-Fender | kerx: No. |
| 15:12.17 | [TK]D-Fender | kerx: who said that had anything to do with a Dial PRIORITY? |
| 15:12.24 | [TK]D-Fender | kerx: Read the instructions <- |
| 15:12.36 | kerx | isn't that a macro |
| 15:12.42 | kerx | exten => 1,n(macroname) |
| 15:12.44 | kerx | isn't that it? |
| 15:12.48 | [TK]D-Fender | swears he can't even hand out answers outright anymore... |
| 15:12.55 | [TK]D-Fender | kerx: NO |
| 15:13.03 | scruz | phpboy: so, basically, originate spoofs a call, then connects the 'calling party' with the called party - if i get you properly? |
| 15:13.09 | [TK]D-Fender | kerx: that is a label that represents a priority |
| 15:13.19 | kerx | [macro_name] |
| 15:13.21 | kerx | is that a macro? |
| 15:13.28 | [TK]D-Fender | kerx: Go read the book, you are missing WAY too many dialplan basics |
| 15:13.45 | [TK]D-Fender | kerx: No, the syntax is off.... try AGAIN |
| 15:13.46 | kerx | http://www.jeremy-mcnamara.com/2007/04/14/how-to-configure-asterisk-using-macros/ |
| 15:13.51 | scruz | kerx: chapter 5 |
| 15:13.55 | kerx | k |
| 15:14.22 | kerx | pg. 119 |
| 15:14.26 | kerx | let's see if Macro's are in here |
| 15:14.35 | *** join/#asterisk mog (n=mog@nat/digium/x-da6f31bb13d8dd85) |
| 15:14.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:15.00 | kerx | pages 157-160 is about macros |
| 15:15.03 | kerx | let me just skip to that |
| 15:15.16 | [TK]D-Fender | kerx: Yes.... you're good at "skipping" :) |
| 15:15.19 | kerx | nice, theres a big crease on that page |
| 15:15.25 | kerx | used books are awesome |
| 15:15.41 | kerx | [TK]D-Fender, dude, that's all about me.... if i had to actually read every book, every webpage |
| 15:15.50 | kerx | i'd be toast |
| 15:16.06 | kerx | although asterisk dialplan's should not be skipped |
| 15:16.09 | [TK]D-Fender | hands kerx some PB&J |
| 15:16.12 | kerx | slaps himself... whatya thinking buddy |
| 15:16.25 | kerx | i love PB&J |
| 15:16.33 | scruz | no crease on my copy - unless i somehow invent a way to crease pdfs |
| 15:16.50 | scruz | oh. that's what it meant |
| 15:17.04 | scruz | phpboy: we still on the same page? |
| 15:17.10 | rue_desk | so you said asterisk CDR is flakey, whats klakey about it? |
| 15:17.39 | kerx | asterisk CDR's are as useless as my cigarettes |
| 15:17.51 | rue_desk | but I asked why |
| 15:17.57 | kerx | disposition is off |
| 15:18.00 | kerx | billsec is always 0 |
| 15:18.10 | kerx | start and end time's are off |
| 15:18.19 | rue_desk | off? by a lot? |
| 15:18.31 | kerx | 7-12 seconds each call ive made |
| 15:18.35 | TrentCreek | [TK]D-Fender: Those PHP/AGI examples suck |
| 15:18.42 | rue_desk | an a zaptel? |
| 15:18.50 | kerx | rue_desk, nope, Dial(SIP/provider/number) call's |
| 15:19.06 | rue_desk | hmm I'd expect sip timing to be really good |
| 15:19.31 | kerx | rue_desk, i donno. i've been told to use a few diff versions, including latest svn's and ive tried all |
| 15:19.37 | rue_desk | I could see zap channels being out of the supervision on the lines isn't set right |
| 15:19.39 | kerx | i've used 1.2.x, 1.4.x, 1.6.x |
| 15:19.49 | phpboy | scruz: I'm thinking that originate is prolly not what you're looking for :( |
| 15:19.51 | rue_desk | kerx, they have bugs filed for it? |
| 15:20.08 | kerx | it might even be something i've done, but i've shown my dialplan + sip.conf to a few guru's here, and they say it looks ok |
| 15:20.30 | kerx | yeah, ive seen some bugs filed for .call's, but i've done AMI Originates |
| 15:20.45 | kerx | i've also tried sending the call's to Local channels which dialed out, but did me no luck besides adding duplicate cdr's |
| 15:22.04 | kerx | [TK]D-Fender, you still here? |
| 15:22.35 | rue_desk | off to work! |
| 15:22.40 | [TK]D-Fender | kerx: Yes |
| 15:22.45 | kerx | i understand Macro's now, but having a hard time seeing the light using M() w/ Dial() because it executes the macro before connecting it to the channel |
| 15:23.12 | kerx | what i understand from the core show application dial is that the macro will be executed until the call is picked up? |
| 15:23.40 | [TK]D-Fender | kerx: You said you want to make a log entry if its answered. well... that macro gets called when its ANSWERED |
| 15:24.08 | kerx | oh, roger that... thanks for clarifying that. it doesn't exactly state that in the 'core show app dial' |
| 15:24.19 | kerx | let me give it a shot then. muchos gracias! |
| 15:24.20 | [TK]D-Fender | kerx: Yes it does. |
| 15:24.32 | scruz | thanks, phpboy, file. have a meeting to attend past some major traffic |
| 15:24.35 | scruz | later |
| 15:24.43 | kerx | it does? |
| 15:24.57 | kerx | "Execute the Macro for the *called* channel before connecting to the calling channel." |
| 15:24.57 | file | it is after they have answered but before being connected to the caller |
| 15:25.10 | kerx | that doesn't really sound like "Executes the Macro when the *called* channel is picked up." |
| 15:25.15 | [TK]D-Fender | kerx: prior to bridging audio |
| 15:25.34 | kerx | ok... i'm probably not fully good w/ the "terminology" yet |
| 15:25.36 | [TK]D-Fender | kerx: How can you run a macro on a channel that hasn't been established? |
| 15:25.47 | *** join/#asterisk stimpie (n=michiel@84-104-5-227.cable.quicknet.nl) |
| 15:25.52 | [TK]D-Fender | kerx: It IS established, it simply isn't BRIDGED yet. |
| 15:26.31 | kerx | ok, so i need a macro w/ some logic |
| 15:26.57 | kerx | it's executing the macro even while it's in a ringing state |
| 15:27.24 | [TK]D-Fender | kerx: Nope. |
| 15:27.25 | kerx | it would have been awesome for my needs if it executed the macro, immediately when it hit a ANSWERED state :) |
| 15:27.33 | kerx | really? |
| 15:27.35 | [TK]D-Fender | kerx: You're clearly not paying attention to the "big picture" |
| 15:27.47 | kerx | if you have time please can u clarify the big for me? |
| 15:28.05 | kerx | i'm writing the macro in the d.p. right now |
| 15:28.25 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:28.28 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 15:31.20 | TrentCreek | phpboy: still there? |
| 15:31.20 | kerx | testing it now |
| 15:32.09 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 15:33.23 | phpboy | TrentCreek; I am |
| 15:33.40 | Carlos_PHX | Anyone aware of an IP phone that has hotel-specific buttons like "room service" and such? |
| 15:34.07 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.130.105) |
| 15:34.13 | CGMChris | I am having a weird problem with my Asterisk. Everything works great, but after the asterisk server is left on for several days we start having problems. For example, right now, I can make outgoing calls and they get connected... but no sound is being passed thru the line and DTMF does not work either. Almost like my provider is switching codes on me or something. Thoughts? Anyone? |
| 15:34.16 | TrentCreek | phpboy: got time for those examples? |
| 15:34.26 | TrentCreek | at this time |
| 15:34.38 | kerx | nice worked |
| 15:34.41 | Carlos_PHX | CGMChris: This is very likely a NAT issue. What router do you have? |
| 15:34.41 | kerx | thanks [TK]D-Fender |
| 15:34.51 | CGMChris | to clarify, if I restart my asterisk service or do a SIP reload, the problem goes away. |
| 15:35.11 | CGMChris | Carlos_PHX: I am using a Peplink Balance 380 enterprise multi-wan router with SIP passthru and QoS |
| 15:35.51 | CGMChris | just like an expensive linksys that load balances 3 lines |
| 15:35.55 | CGMChris | :) |
| 15:36.39 | Carlos_PHX | Never heard of it. What fixes the audio issue when it happens? |
| 15:37.02 | Carlos_PHX | Sorry, just saw you answered that. |
| 15:37.06 | phpboy | TrentCreek: never done AGI stuff in PHP before |
| 15:37.13 | phpboy | I will be doing it soonish though |
| 15:37.16 | phpboy | but only next week |
| 15:37.16 | Carlos_PHX | Typically an audio issue is related to NAT. |
| 15:37.37 | CGMChris | Its just odd that it works for days, and then on the 3rd or 4th day of running, it will stop working. |
| 15:37.41 | TrentCreek | phpboy: Well then in any script? |
| 15:37.58 | Carlos_PHX | My guess is the router is forgetting the connection. Have you changed the re-register time from the default? |
| 15:38.00 | phpboy | echo $var; ? |
| 15:38.09 | phpboy | if in a function |
| 15:38.14 | phpboy | return $var; |
| 15:38.15 | TrentCreek | I am just trying to figure out how to get a script to pass a number and DIAL |
| 15:38.15 | CGMChris | Carlos_PHX: No, i have not altered re-register time |
| 15:38.36 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 15:38.37 | TrentCreek | phpboy: I see the AGO commands only do BS stuff and notihing like transfer...etv |
| 15:38.45 | TrentCreek | oops.. AGI |
| 15:39.11 | CGMChris | Carlos_PHX: I am using Asterisk in G729 passthru mode, so the codec may have something to do with it. Not sure yet. |
| 15:39.18 | Carlos_PHX | CGMChris: There's probably no really clear answer to this. Losing audio while calls still go through has always been a NAT problem in my experience. I think you're facing some troubleshooting like replacing the router or trying a different server and/or different ITSP. |
| 15:39.34 | fexy | Any of you folks attempted to listen for dhcp events and then created extensions on the fly in mysql? |
| 15:39.45 | CGMChris | Carlos_PHX: Looking thru sip debug logs now... we will see. |
| 15:39.51 | fexy | I wish to do this for sccp phones |
| 15:40.07 | itguru | is confused, I have a handset that can accept calls, it just can't make them - everytime I do, i get an engaged tone, any ideas? |
| 15:40.37 | CGMChris | Carlos_PHX: "SIP/2.0 403 Forbidden" |
| 15:41.24 | TrentCreek | phpboy: well pasing the number is no big deal...but the book, and web, I see no way to make the system dial that number |
| 15:41.35 | Carlos_PHX | CGMChris: You can't place a call at all? I thought you said it was just audio that didn't go through? |
| 15:41.47 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 15:42.13 | CGMChris | I can place the call, just no sound goes thru at all. |
| 15:42.28 | [T]ank | I am getting a bunch of remote unix connect/disconnect message on one of my asterisk machines. http://pastebin.ca/1288660 any ideas of what they could be or how to stop them? |
| 15:42.29 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:42.31 | phpboy | TrentCreek: hmmmm, that sounds like an AGI problem and not PHP? |
| 15:42.51 | Carlos_PHX | The forbidden message above suggests a total call failure. |
| 15:42.55 | *** join/#asterisk BlackRayne (n=nsx@70.94.0.201) |
| 15:43.21 | [TK]D-Fender | [T]ank: AMI or CLI connection taking place. You should already know what would be polling your server |
| 15:43.24 | kerx | [Dec 18 07:41:11] WARNING[25579]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 15:43.24 | kerx | =ANSWER |
| 15:43.24 | kerx | ^ |
| 15:43.34 | kerx | Anyone seen that warning/error before? |
| 15:43.42 | TrentCreek | phpboy: yes..I see all the AGO commanded listed , but nothing like in the CONF files that say EXEN --> |
| 15:43.43 | [TK]D-Fender | kerx: What part of "fix your expression" didn't you get earlier? |
| 15:44.03 | kerx | that word Fix |
| 15:44.14 | [T]ank | [TK]D-Fender: there shouldnt be anything polling my server. all I have is 6 phones pointed at it |
| 15:44.15 | kerx | i don't know how to do it by myself |
| 15:44.18 | kerx | [TK]D-Fender, i need u |
| 15:44.18 | [T]ank | no apps or anything else |
| 15:44.27 | CGMChris | Carlos_PHX: Got unsupported a:fmtp in SDP offer ? |
| 15:45.17 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a9981a3d563db033) |
| 15:45.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:45.39 | kerx | my expression looks exactly like another statement i have |
| 15:45.39 | kerx | exten => start,5,GotoIf($[${AMDSTATUS}=MACHINE]?machine:human) |
| 15:45.42 | kerx | and this works fine |
| 15:46.22 | [T]ank | [TK]D-Fender: what could be polling my server? |
| 15:48.12 | Carlos_PHX | CGMChris: I've never seen that specific error, but it does suggest that the CODEC is wrong. Could be the far end (or your end even) is out of g.729 licenses. |
| 15:48.17 | Carlos_PHX | How many licenses do you have? |
| 15:49.11 | file | that messages comes up if you have sip debug turned on, there is something in the SDP that Asterisk does not support, usually nothing to be worried about |
| 15:49.38 | Poincare | I'm looking for 'remote asterisk hands' in Mexico, someone who can go onsite to connect everything etc. Anyone here from Mexico? |
| 15:50.27 | [TK]D-Fender | kerx>my expression looks exactly like another statement i have <- no, it doesn't |
| 15:50.41 | CGMChris | Carlos_PHX: I have 5 licenses. The licenses are only used when transcoding takes place, such as when the phone is ringing or when silence is being detected when callers leave a voicemail message. Licensing errors are different, so we can safely rule that out. |
| 15:50.59 | kerx | [TK]D-Fender, i'm reading about it in channelvariables.txt about my syntax error....still no lluck |
| 15:51.07 | kerx | but i'm trying to investigate what i've done wrong |
| 15:51.10 | CGMChris | [TK]D-Fender: <--- SIP read from 209.249.3.59:5060 ---> SIP/2.0 403 Forbidden. this was "Read from" my provider's IP. Which side is saying 403 forbidden? Their side? |
| 15:51.15 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:51.15 | [TK]D-Fender | kerx: Your 2 lines there are NOT the same |
| 15:51.24 | kerx | i didn't know that |
| 15:51.31 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 15:51.38 | kerx | it looks exactly the same to me, besides the variable and the labels |
| 15:52.16 | [TK]D-Fender | kerx: LOOK AGAIN |
| 15:52.16 | [TK]D-Fender | CGMChris: READ <- that should tip you off |
| 15:52.17 | CGMChris | Ok, thanks. |
| 15:52.20 | kerx | GotoIf($[${AMDSTATUS}=MACHINE]?machine:human) |
| 15:52.20 | kerx | GotoIf($[${DIALSTATUS}=ANSWER]?press1bad:answer) |
| 15:52.23 | kerx | hrmm.... |
| 15:52.27 | *** join/#asterisk dec3pti0n (n=dec3pti0@pdpc/supporter/student/dec3pti0n) |
| 15:52.39 | *** join/#asterisk cjk (n=cjk@vodsl-9252.vo.lu) |
| 15:52.46 | awk_r | could it be that ${DIALSTATUS} isn't set? I think I've had similar issues, try putting "" around each side of hte '=' |
| 15:53.06 | [TK]D-Fender | kerx: repastebin your dialplan and CLI output |
| 15:53.09 | kerx | awk_r, I've tried that also :) I've placed ${DIALSTATUS} as "${DIALSTATUS}" but it gave me the same error |
| 15:53.09 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-37ed367a79ad093f) |
| 15:53.09 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:53.10 | awk_r | *similar issues where variables = null or are empty |
| 15:53.18 | cjk | hi, maybe this is not the right channel but is there a way to tell sipsak to not wait for an answer so that messages like "timeout after 2000 ms" do not appear |
| 15:53.21 | rue_desk | kerx, you set that option, right? |
| 15:53.26 | file | kerx: if you are doing that in your macro then DIALSTATUS will not be set ... |
| 15:53.44 | Skyp | Hm... is it possible to set the Callerid(name) for a channel created with Dial()? For realtime-statistics mainly |
| 15:53.50 | kerx | http://pastebin.ca/index.php |
| 15:54.04 | kerx | rue_desk, which option? |
| 15:54.13 | kerx | file, i must be doing that then :-( |
| 15:54.15 | [TK]D-Fender | Skyp: "core show function CALLERID" |
| 15:54.20 | rue_desk | the one tk told you to look up and the one I pasted |
| 15:54.26 | [TK]D-Fender | kerx: http://pastebin.ca/1288626 <-- this PB from earlier = bad |
| 15:54.32 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 15:54.40 | kerx | i know |
| 15:54.43 | kerx | i'm not doing that |
| 15:55.00 | file | kerx: if your macro gets called then the channel has been answered, it is a fact |
| 15:55.00 | Skyp | TK: but how can I call that in the dialed channel? |
| 15:55.01 | [TK]D-Fender | kerx: Then go show me what I requested |
| 15:55.02 | glaz | missing the $ ? |
| 15:55.05 | [T]ank | [TK]D-Fender: so were you suggesting to me that there might be an application that is polling my server? |
| 15:55.20 | rue_desk | kerx, you have your * console available there? |
| 15:55.25 | [TK]D-Fender | [T]ank: No. That was a statement. |
| 15:55.27 | Skyp | TK: I already do that for the calling channel, but I would like to set it on the other side |
| 15:55.51 | [TK]D-Fender | Skyp: Same answer. |
| 15:55.55 | teknoprep | hey all... i have some polycom phones.. does anyone now what the option is to have the phone remember the volume when i pick up the handset? i always have to crank it up when i pick the handset up.. it remembers the speakerphone volume tho |
| 15:55.56 | [T]ank | [TK]D-Fender: If I did not set anything up, how could I figure out what it is that is polling it? |
| 15:56.11 | kerx | http://pastebin.ca/1288670 |
| 15:56.11 | [TK]D-Fender | [T]ank: then someone else did |
| 15:56.16 | kerx | [TK]D-Fender, ^ please look above |
| 15:56.28 | [T]ank | teknoprep: when you change the volume there is a save button on the screen |
| 15:56.42 | teknoprep | no |
| 15:56.46 | teknoprep | [T]ank, no |
| 15:57.24 | kerx | exten => 1,n,Dial(SIP/provider/18885551212,3650,M(press1pickup)) |
| 15:57.30 | [TK]D-Fender | kerx: $"{DIALSTATUS}"=ANSWER <-- I don't see "'s in your dialplan. RELOAD YOUR CHANGES |
| 15:57.31 | kerx | [TK]D-Fender, I forgot to post that above in pastebin |
| 15:57.53 | kerx | i've tried that one also, and it failed |
| 15:57.55 | kerx | let me try |
| 15:57.55 | [T]ank | [TK]D-Fender: in this instance, I am the only one with access to this machine. Is there not a way to see where the requests are coming from? I am not the only person who has set things up on this network. I wonder if it is something somewhere else accidentally pointed here. |
| 15:57.56 | [TK]D-Fender | kerx: and there is NOTHING to check in that macro. |
| 15:57.57 | eppigy | hello |
| 15:57.59 | eppigy | i am dave |
| 15:58.04 | [TK]D-Fender | kerx: just make your log entry |
| 15:58.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:58.17 | [TK]D-Fender | teknoprep: "persist" <- 3 values |
| 15:59.03 | *** join/#asterisk dhill (n=dhill@dhcp-222.iserv.net) |
| 15:59.05 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 15:59.18 | kerx | oh |
| 15:59.19 | kerx | weird |
| 15:59.23 | kerx | it must have never taken affect |
| 15:59.27 | kerx | it did now, but i get a 0 in there? |
| 15:59.32 | kerx | DIALSTATUS is set to 0 |
| 15:59.33 | dhill | when i reload, there is no comment about it loading udptl.conf. i do not think it is loading... |
| 15:59.34 | kerx | GotoIf("SIP/flowroute-166d6860", "0?press1bad:answer") |
| 15:59.39 | teknoprep | hey [TK]D-Fender , thanks alot man... that about drove me nuts |
| 15:59.49 | [TK]D-Fender | kerx: you don't need to check ANYTHING |
| 16:00.18 | kerx | oh duh... |
| 16:00.21 | kerx | slaps himself to wake up |
| 16:00.57 | dhill | I am using 1.4.22. there is the udptl CLI command. |
| 16:01.57 | dhill | i have verbose > 1, and am not seeing the UDPTL allocating from port range message |
| 16:02.04 | dhill | argh |
| 16:02.28 | dhill | does udptl.conf need to be specified in asterisk.conf or extconfig.conf? |
| 16:04.25 | *** join/#asterisk a1fa (n=a1f@unaffiliated/a1fa) |
| 16:04.33 | a1fa | hello |
| 16:04.47 | a1fa | [TK]D-Fender : sorry to bug you, what was that telephony depot site? |
| 16:05.00 | [TK]D-Fender | a1fa: telephonydepot.com |
| 16:05.11 | a1fa | they changed their website |
| 16:05.21 | a1fa | so i was kind of confused |
| 16:05.54 | [TK]D-Fender | a1fa: http://www.telephonydepot.com/ |
| 16:06.00 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 16:06.03 | a1fa | yeah.. i am in.. they changed their webdesign |
| 16:06.17 | [TK]D-Fender | a1fa: much nicer |
| 16:06.46 | dhill | anyone have ideas for me? |
| 16:07.14 | a1fa | i liked the old one better |
| 16:07.18 | a1fa | :P |
| 16:07.26 | TrentCreek | phpboy: I found the answer |
| 16:08.17 | itguru | My extension can recieve calls - but everytime I go to make one, I get an instant engaged tone - if there was a problem with the configuration, shouldn't calls in both directions be affected? |
| 16:09.06 | [TK]D-Fender | itguru: No, your phone can refuse calls based on its dialplan. It can also accept calls regardless of being registered, etc. |
| 16:09.39 | itguru | [TK]D-Fender, Aww man! - you mean the phone can refuse to dial out because of its dial plan? |
| 16:09.48 | [TK]D-Fender | itguru: |
| 16:09.51 | [TK]D-Fender | yes |
| 16:10.08 | itguru | how the hell am I to track that down!? |
| 16:10.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:11.34 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:11.44 | [TK]D-Fender | itguru: enable SIP debug and make sure the phone isn't actually trying to talk to *. t hen go look at your phone's configs |
| 16:11.56 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:12.33 | a1fa | number porting is pita. |
| 16:12.50 | a1fa | i am going to keep broadvoice at 9/month :( and use teliax for outgoing calls |
| 16:13.12 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 16:13.29 | dhill | argh |
| 16:13.40 | itguru | [TK]D-Fender, nothing comes up in the log at all - the handset says it's registered, every other extention is set up the same way, and they work |
| 16:16.04 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-c3bad70afa88ef10) |
| 16:16.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:16.13 | itguru | the word forbidden came up on the LCD display of the phone. Every other extension calls out just fine |
| 16:19.41 | dhill | ls -lu tells me udptl.conf is not accessed |
| 16:19.46 | dhill | lovely |
| 16:20.51 | [TK]D-Fender | itguru: if you enabled SIP debug on * you'd BETTER see something if the phone 403's |
| 16:21.10 | dhill | oh, its only loaded on startup? |
| 16:23.23 | *** join/#asterisk n3glv (n=n3glv@c-71-60-107-110.hsd1.pa.comcast.net) |
| 16:23.27 | dhill | nope |
| 16:23.27 | kerx | is Record() or Monitor() better for recording a bridged Dial() ? |
| 16:23.30 | dhill | not on startup either |
| 16:24.38 | [TK]D-Fender | kerx: Record is not for recording Dial |
| 16:24.50 | kerx | k, thanks |
| 16:25.51 | eppigy | hello |
| 16:25.54 | eppigy | i am dave |
| 16:25.55 | kerx | [TK]D-Fender, do you know if it's possible to pass arguments w/ the M() ? I can't really find the doc's in voip-info |
| 16:26.26 | kerx | I have 3 variables set in my context, but the macro context never receives those. |
| 16:26.40 | [TK]D-Fender | kerx: "core show application dial" <-------- |
| 16:27.28 | itguru | [TK]D-Fender, |
| 16:27.37 | itguru | [TK]D-Fender, nothing at all |
| 16:28.07 | [TK]D-Fender | itguru: Fix your phone, or fix your networking. Either way, packets aren't getting to * |
| 16:28.28 | eppigy | or iptables -L |
| 16:28.31 | eppigy | and flush |
| 16:30.12 | *** join/#asterisk lucasb (n=lucasb@office.telifon.com) |
| 16:33.39 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:33.47 | jjshoe | is it possible to get the serial # of a sangoma card from the console? |
| 16:35.25 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:35.32 | mikealeonetti | hahaha. |
| 16:35.47 | mikealeonetti | I installed the zaptel module on asterisk and now the timing is all screwed up |
| 16:36.07 | mikealeonetti | the voicemails and auto attendant will repeat itself |
| 16:36.16 | itguru | [TK]D-Fender, but incoming calls work |
| 16:36.31 | mikealeonetti | I wonder if the RTC is screwed up |
| 16:39.15 | *** part/#asterisk n3glv (n=n3glv@c-71-60-107-110.hsd1.pa.comcast.net) |
| 16:40.52 | fexy | Have any of you configured asterisk with mysql under debian etch without install packages from sid? |
| 16:41.45 | xuser | fexy: install from source |
| 16:42.44 | fexy | I was trying to avoid that :p |
| 16:43.03 | fexy | are there deb source packages I can use or do I just have to grab tarballs from the asterisk repository? |
| 16:43.20 | xuser | tarballs |
| 16:43.26 | xuser | ~book |
| 16:43.27 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 16:45.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:45.26 | thehar | hrm. leif's domain is failed right now |
| 16:45.35 | *** join/#asterisk DaVe_LeMe (n=Michel@189.123.65.93) |
| 16:50.54 | *** join/#asterisk reneger (n=reneger@p3EE2E1AC.dip.t-dialin.net) |
| 16:50.58 | mikealeonetti | damn |
| 16:51.04 | mikealeonetti | what could I be doing wrong |
| 16:51.14 | mikealeonetti | hr |
| 16:51.15 | mikealeonetti | m |
| 16:51.57 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 16:52.26 | *** join/#asterisk hugh_cox (n=chatzill@216.65.195.52) |
| 16:52.31 | hugh_cox | I need to dial about 200 #s to be sure our IVR is answering all of them, is there a way to have asterisk dial them and detect if the call was answered? |
| 16:52.39 | *** join/#asterisk itguru (n=itguru__@host81-134-10-140.in-addr.btopenworld.com) |
| 16:55.18 | xuser | sipp |
| 16:55.27 | fexy | does aterisk-addons-1.4.2 match up with asterisk-1.4.22 ? |
| 16:56.16 | fexy | will just get asterisk 1.4.2 to be safe |
| 16:56.41 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:58.33 | iratik | In your guys' opinion... what is the best softphone out there for vista/xp ... ? (I've tried just about all of them .. but that was a year ago now) |
| 16:58.46 | Qwell | ~best |
| 16:58.47 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 16:59.04 | iratik | ~best in your opinion.... |
| 16:59.11 | fexy | oingo boingo! |
| 16:59.20 | mikealeonetti | ~best game |
| 16:59.24 | Qwell | fexy: I said this yesterday, and I'll say it again today. |
| 16:59.25 | itguru | iratik, x-lite |
| 16:59.28 | Qwell | Boingo > Oingo Boingo |
| 17:00.32 | iratik | you guys are very familiar with asterisk ... and are extremely familiar with the types of challenges softphones have to face in larger deployments .. . i can't pretend i know all the factors that make a softphone the best |
| 17:00.32 | Qwell | iratik: same rules apply - best for what? |
| 17:00.46 | mikealeonetti | timing was working perfectly before until I installed the zaptel module |
| 17:00.49 | mikealeonetti | maybe I'll upgrade i |
| 17:00.50 | mikealeonetti | t |
| 17:00.51 | Qwell | call quality? ease of use? ease of setup? |
| 17:00.56 | Qwell | expandability? |
| 17:01.00 | iratik | reliability |
| 17:01.15 | *** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 17:01.24 | iratik | its got to just work... all the time.. and never have problems ... |
| 17:01.56 | iratik | command line options would be nice |
| 17:02.02 | Qwell | so then you don't want a softphone |
| 17:02.13 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:02.15 | iratik | SJPhone meets those criteria... but doesn't work in vista |
| 17:02.38 | Qwell | so, SJPhone, which doesn't work, works all the time? |
| 17:02.52 | russellb | skype? |
| 17:02.53 | russellb | ducks |
| 17:03.46 | russellb | or eyebeam. |
| 17:03.54 | TrentCreek | didnt your name used to be xxxx_skype? |
| 17:04.22 | TrentCreek | i skipped out on skype |
| 17:04.52 | russellb | do what? |
| 17:05.42 | TrentCreek | nope |
| 17:05.51 | TrentCreek | dont DO skype |
| 17:05.58 | a1fa | bye bye broadvoice |
| 17:06.03 | russellb | my name was never xxxx_skype. |
| 17:06.10 | a1fa | 35/month x 12 months x 4 years = |
| 17:06.17 | iratik | Qwell: It doesn't work on vista |
| 17:06.20 | a1fa | $1680 |
| 17:06.21 | iratik | Its been a dream on XP |
| 17:06.24 | a1fa | no more business for you |
| 17:06.48 | TrentCreek | X = varible |
| 17:07.00 | russellb | my name has never been anything skype. |
| 17:07.21 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 17:07.21 | TrentCreek | okay..someone had a skype something |
| 17:07.32 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 17:07.48 | iratik | need something that just works on vista... x-lite had a ton of problems last time i tried it |
| 17:08.05 | TrentCreek | it works fine on mine |
| 17:08.31 | iratik | If i remember right... x-lite makes sounds upon pressing keys etc.. through the systems audio hardware.. on at least 2 vista machines... it seemed like x-lite's processor usage would spike every time it made a sound |
| 17:08.33 | iratik | odd... |
| 17:08.56 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:09.28 | russellb | another windows option ... http://www.softphone.com/asterisk/ |
| 17:09.41 | fexy | is chan-sccp-b the most uptodate sccp protocol for asterisk? |
| 17:10.27 | iratik | oh yeah... you can't transfer with x-lite |
| 17:11.33 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:11.49 | Corydon76-dig | fexy: actually, I think chan_skinny is |
| 17:12.29 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 17:12.37 | fexy | Corydon76-dig, thanks |
| 17:12.54 | ariel_ | afternoon everyone |
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| 17:30.32 | GoRK | Question: I have never been able to get the jitterbuffer working right with sip channels using jbimpl=adaptive. jbimpl=fixed works and disabling the jb works, but both are non-optimal for a situation where I have both local and remote sip users bridging calls onto a PRI. It doesnt work for me in any version of 1.4 or 1.6 (tested to 1.6.0.3-rc1). The main symptoms are problems with no audio or garbled audio when transferring, parking, and holding calls. Am I m |
| 17:32.25 | GoRK | FWIW I can't get transfer, parking, and holding working 100% using anything but canreinvite=no and directrtpsetup=no either. I am hoping maybe they are related |
| 17:32.39 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
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| 17:39.09 | a1fa | wow |
| 17:39.16 | a1fa | i just changed my primary number @ broadvoice |
| 17:39.20 | a1fa | and it screwed up everything |
| 17:39.49 | a1fa | "Probably a DNS error for registration to" |
| 17:40.52 | a1fa | [Dec 18 11:40:44] WARNING[16373]: chan_sip.c:16773 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 25 |
| 17:41.01 | a1fa | it doesnt like broadvoices register => line |
| 17:42.14 | a1fa | register => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension> |
| 17:45.03 | a1fa | strange |
| 17:45.35 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 17:48.32 | *** join/#asterisk galeras (n=galeras@190.159.51.107) |
| 17:48.46 | galeras | please take a look of http://pastebin.com/m6583e6bc, and tell me which extension will haveoutrt-005-Extbogo-custom context to be executed before of outrt-005-Extbogo context. |
| 17:53.32 | fexy | what would be the best way to assign extensions on the fly? |
| 17:53.57 | fexy | I was thinking of listening to my dhcp server in some fashion and then assigning the extension to it's MAC |
| 17:54.08 | fexy | All the phones are cisco |
| 17:54.13 | fexy | using SCCP |
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| 17:57.57 | hugh_cox | I need to dial about 200 #s to be sure our IVR is answering all of them, is there a way to have asterisk dial them and detect if the call was answered? |
| 18:01.59 | a1fa | i have a problem with register line |
| 18:02.03 | a1fa | it tells me a dns problem |
| 18:02.07 | a1fa | i dont see how |
| 18:02.44 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 18:03.11 | a1fa | Probably a DNS error for registration to number@sip.broadvoice.com |
| 18:03.12 | a1fa | ??! |
| 18:03.24 | lowtek | Hi all. Is there some secret to setting a context for a type=friend in sip.conf? My calls are coming in just fine but will only be accepted by my default context regardless of what I have set in my peer definition in sip.conf, any ideas? |
| 18:10.11 | lowtek | Or are all calls accepted in my context=whatever in sip.conf/general no matter what? |
| 18:11.36 | GoRK | lowtek: It sounds like the sip peers are not registering properly. Are they authenticated by ip/host or password? |
| 18:12.23 | lowtek | They are authenticating via password: register => 01248-1:ZMF7Kc@in.flexpulse.com <= Do I need to somehow specify my context here? |
| 18:12.44 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
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| 18:14.21 | *** part/#asterisk n3glv (n=n3glv@c-71-60-107-110.hsd1.pa.comcast.net) |
| 18:14.31 | GoRK | lowtek: uh.. well first, did you just paste your password into a public IRC channel? second, if this is for a provider you will do better to split the configuration into a seperate type=peer and type=user entry |
| 18:14.43 | lowtek | Yea, I've already changed it. Oops. |
| 18:16.27 | a1fa | lol |
| 18:16.36 | lowtek | Ok, thanks for the help. I have type=friend right now on my inbound definition. Is the context= in my peer definition handled differently depending on the type= ? |
| 18:19.12 | GoRK | well the context in terms of a sip 'user' doesnt really apply .. you should probably just have a peer defined with context=whatever and host=in.flexpulse.com |
| 18:19.34 | lowtek | GoRK: Yep, that's what I have. Tha tpeer is type=friend |
| 18:19.40 | lowtek | Tha t = That |
| 18:19.48 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 18:19.50 | GoRK | but yeah break it apart into a seperate user and peer sections instead of type=friend |
| 18:19.56 | lowtek | Got it! That makes sense. |
| 18:20.03 | lowtek | It just clicked. Ok, will do, thanks, Gork. |
| 18:20.29 | GoRK | that may not solve your problem per se but it will help you better determine whats going on |
| 18:21.05 | GoRK | i suspect that you are receiving the call from another host and its not falling into that peer definition |
| 18:21.24 | [TK]D-Fender | GoRK>but yeah break it apart into a seperate user and peer sections instead of type=friend <--- no need and generally not recommended |
| 18:21.24 | GoRK | so its going to the context you have in [general] |
| 18:21.25 | lowtek | Like I mentioned, calls are coming in just fine, just not to the context I'm specifying in my inbound peer definition. They are all coming in to whatever I have set in the general section of sip.conf under context= |
| 18:21.32 | lowtek | Yep. |
| 18:21.38 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 18:21.40 | lowtek | Ok, I see. |
| 18:21.45 | lowtek | Another epiphany (sp?) |
| 18:22.04 | [TK]D-Fender | lowtek: Go enable SIP DEBUG and actually watch what comes in |
| 18:22.28 | lowtek | Thanks guys for your help (GoRK/TK) |
| 18:22.34 | a1fa | fixxxd |
| 18:22.39 | a1fa | damn broadvoice |
| 18:29.41 | hugh_cox | Can i make a .call file that will be placed in /var/spool/asterisk/outgoing call to more than 1 number? i cant find an example of how anywhere |
| 18:30.08 | *** join/#asterisk freckle_home (n=jon@84.45.168.57) |
| 18:30.14 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 18:31.12 | beek | hugh_cox: I'm not 100% certain but I believe the answer to be "no". Why would you want to do that over one file per call? |
| 18:32.07 | GoRK | do you just want the call to ring two destinations and then whichever can answer same as like Dial(SIP/100&SIP/101) might do? |
| 18:32.22 | hugh_cox | i need to call about 200 of my 800 #s to be sure they are all working. should i just make a file for each call? |
| 18:32.37 | GoRK | yes |
| 18:32.42 | beek | hugh_cox: That would be the quickest way -- easily done with a script. |
| 18:33.46 | hugh_cox | I can do that, but the next question is then what do i specify for the channel for each file? i did a test, and it calls a local phone, once i answer it then it calls the 800 #, i cant very well answer 200 pones. |
| 18:34.30 | hardwire | codec negotiation is a pain |
| 18:35.34 | hugh_cox | Does my question make sence? |
| 18:35.49 | beek | hugh_cox: Are you trying to tie up 200 channels at once? |
| 18:35.54 | hugh_cox | yes |
| 18:36.14 | hugh_cox | i want it to just dial them all at once, i just need to log if it was answered by the IVR or not. |
| 18:36.37 | hugh_cox | looks like adding Achive: yes will do that for me |
| 18:36.38 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-48-166.w86-215.abo.wanadoo.fr) |
| 18:37.21 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-48-166.w86-215.abo.wanadoo.fr) |
| 18:37.37 | hugh_cox | i guess it can do a few chunks at a time. but I definitely don't want to sit by my phone and answer it 200 times if you know what i mean. |
| 18:38.32 | beek | hugh_cox: I'm not sure exactly what you're expecting but you do not need to answer the phone at all. |
| 18:38.40 | hugh_cox | really? |
| 18:38.50 | hugh_cox | how so? |
| 18:39.44 | hugh_cox | Channel: <channel>: Channel to use for the outbound call <--- I put my desk phone for this field, when i mv the file it calls my desk phone, oExtension: <ext> Extension definition in extensions.conf nce i answer it then dials the number spicified in |
| 18:40.01 | beek | http://pastebin.com/m7c5b1914 |
| 18:40.14 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:40.15 | GoRK | have it originate to the remote number instead of your local phone .. Zap/g1/8005551212 for instance |
| 18:40.17 | hugh_cox | cool, thx ill take a look |
| 18:40.17 | beek | Here's one I wrote as a joke to inform a coworker that I was ready for lunch. |
| 18:40.27 | hugh_cox | ha |
| 18:40.36 | beek | Asterisk can answer and then hang up and that will get your log filled. |
| 18:44.25 | hugh_cox | i think im confused |
| 18:44.41 | *** join/#asterisk freckle_home (n=jon@84.45.168.57) |
| 18:44.41 | beek | hugh_cox: How so? |
| 18:45.05 | beek | hugh_cox: Is this system in production yet? |
| 18:45.40 | hugh_cox | so in your example it calls Zap/3/140, once its answered it calls ext 8000 plays a message then hangs up? |
| 18:46.14 | beek | Extension 8000 in the 'hungry' context acts as the caller. |
| 18:46.34 | hugh_cox | We dont use asterisk for our production system, but i use linux @ my desk, so i set it up with asterisk to be able to call out to the pstn and other exts in the building |
| 18:46.45 | hugh_cox | beek: ahhhhh, i see |
| 18:46.54 | beek | So, * dials extension 140 (although you could dial an outside number as easily) and once the call is answered extension 8000 gets executed. |
| 18:47.19 | beek | So, the callee hears the "barry is famished" message, "thank you", and then is hung up on. |
| 18:47.26 | hugh_cox | ok i get it, that will work perfectly |
| 18:48.27 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:48.44 | hugh_cox | sometimes i need someone to spell it out for me i guess, but cool man, thanks for your help. this will save me alot of time |
| 18:48.54 | beek | hugh_cox: you're welcome. |
| 18:48.56 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-4154dc4840e10220) |
| 18:49.42 | beek | hugh_cox: the kicker to the joke was that I used 'at' to schedule the call so that I was standing in her office when her phone rang. |
| 18:51.08 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
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| 18:52.33 | fexy | have any of you installed unixodbc-dev on etch? |
| 18:53.00 | fexy | I have a perfectly fine headless system here and for some reason it wants to start installing all these x11 libraries |
| 18:53.48 | beek | fexy: Doesn't the unixodbc package include some GUI application for configuring data sources? |
| 18:54.11 | fexy | you are probably right |
| 18:54.28 | fexy | hmmm |
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| 19:10.21 | mikealeonetti | what would cause asterisk to skip a lot after installing the zaptel module? |
| 19:11.20 | mikealeonetti | it's all SIP, no Zap devices needed. But ever since I installed it so I can attempt meetme the voicemail is skippy. |
| 19:13.06 | mikealeonetti | is it a timing issue? |
| 19:13.46 | viraptor | does anyone have an idea why asterisk may look like it's handling everything in one thread? (network, realtime, logs) I see all other threads on my system idling... |
| 19:24.58 | *** join/#asterisk codaine (n=Onur@198.64.168.130) |
| 19:33.02 | mikealeonetti | what can I search for on google in this instance |
| 19:33.03 | mikealeonetti | ? |
| 19:33.12 | mikealeonetti | even if I take zaptel off |
| 19:33.27 | mikealeonetti | the voicemail voices are skippy |
| 19:33.41 | jjshoe | probably timing, if you have no hardware timing source |
| 19:33.48 | jjshoe | computers them sevles are notoriously bad at timing |
| 19:34.07 | mikealeonetti | the strange thing is, it was working just fine |
| 19:34.43 | mikealeonetti | it's running on a virtual machine |
| 19:34.47 | mikealeonetti | I wonder if that is causing timing issues |
| 19:35.32 | *** join/#asterisk ElCheapo (n=elcheapo@d137-186-181-17.abhsia.telus.net) |
| 19:36.36 | mikealeonetti | apparently somebody had a similar issue http://forums.virtualbox.org/viewtopic.php?p=29430&sid=e240b75f4a8fd7ec34cd04ef160f99c3 |
| 19:36.58 | mikealeonetti | I need to disable all modules that will use hardware timing then |
| 19:37.29 | [TK]D-Fender | software timing on a VM? INSANE |
| 19:37.43 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 19:37.48 | *** join/#asterisk deeperror (n=deeperro@75.46.62.85) |
| 19:37.50 | mikealeonetti | well it was working fine until I installed the zaptel module |
| 19:38.10 | [TK]D-Fender | mikealeonetti: Yes... the thing that PROVIDES timing |
| 19:38.34 | mikealeonetti | well, it's either I fix that or get rid of it |
| 19:38.38 | mikealeonetti | and I'm trying to do either |
| 19:38.42 | deeperror | is there anything that would allow asterisk to function similar to a modem answering an inbound call? |
| 19:39.23 | deeperror | say i wanted a dial up connection with a DID? or does that not fly? |
| 19:39.48 | mikealeonetti | [TK]D-Fender: not unless you want to point me in a direction? |
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| 19:45.24 | hugh_cox | if i use a .call file to make a outbound call, how can I tell if the call was answered? I use the Dial app to place the call, but what if i get "im sorry this # is...." or fast busy or something? how can I log what happened? |
| 19:46.06 | *** join/#asterisk reneger (n=reneger@p3EE2E1AC.dip.t-dialin.net) |
| 19:47.42 | fexy | hugh_cox, there are return codes |
| 19:47.50 | fexy | let me see if I have a list somewhere |
| 19:48.00 | mikealeonetti | damn |
| 19:48.26 | fexy | hugh_cox, http://www.telos-systems.com/?/techtalk/cause.htm |
| 19:48.42 | fexy | now how to grab those with asterisk well I have no clue :p |
| 19:50.02 | deeperror | hugh_cox, seems like you could handle the h or t priority in your dial plan to look at call status and then handle it from there however you see fit |
| 19:51.41 | deeperror | anyone know if there is anything that would allow asterisk to handle a v90 connection on an inbound call? |
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| 19:54.03 | hugh_cox | let me see....... |
| 19:56.32 | hugh_cox | deeperror: i cant find a page that details h and t priorities.... |
| 19:57.04 | deeperror | http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension |
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| 19:59.02 | deeperror | hugh_cox, http://www.voip-info.org/wiki/view/DIALSTATUS |
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| 20:10.34 | deeperror | anyone know a way to get asterisk to handle a v90 connection on an inbound call? |
| 20:14.31 | *** join/#asterisk sasargen (n=chatzill@174-153-210-50.pools.spcsdns.net) |
| 20:23.18 | fexy | how do you go about using mysql to store extension info. Or more directly how come there isn't a device field in the extensions table definition? |
| 20:23.36 | thehar | odbc |
| 20:23.54 | fexy | erm |
| 20:23.59 | fexy | how about the second question :D |
| 20:26.13 | fexy | I am attempting to dynamically assign extensions and register cisco 7961's using sccp |
| 20:26.24 | fexy | So I can just plug them in and let them go |
| 20:26.52 | fexy | I have everything compiled and table definitions created |
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| 20:34.25 | rue_desk | anyone know of a set of template configs for sip phones? |
| 20:36.09 | [TK]D-Fender | rue_desk: Every make is different |
| 20:36.39 | rue_desk | exactly |
| 20:36.53 | rue_desk | there should be a repository of config files for them |
| 20:38.19 | *** join/#asterisk freckle_home (n=viperdud@84.45.168.57) |
| 20:39.32 | [TK]D-Fender | rue_desk: O RLY? And who is payed to do this? And how mmuch can anyone provide in configs considering that auth is private? What are you expecting here? |
| 20:39.54 | thehar | hides behind a rock. |
| 20:40.07 | *** join/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 20:40.27 | kb3ien | anyone good advice for FXO atas? |
| 20:40.47 | jjshoe | rue_desk several people provide tools for producing the diff. configs |
| 20:41.52 | rue_desk | http://lirc.sourceforge.net/remotes/ nobody was paid to make those config files, they were submitted cause the developers asked people to |
| 20:42.59 | rue_desk | jjshoe, where can I get it/them? |
| 20:43.01 | thehar | rue_desk: you should do it then :) |
| 20:43.21 | jjshoe | rue_desk i think trixbox has a tool, freepbx might, you need to do some looking around |
| 20:44.02 | rue_desk | think they could give me sipconfigs.asterisk.org ? |
| 20:44.19 | jjshoe | rue_desk they could, but would they? |
| 20:44.40 | jjshoe | rue_desk if a page doesn't already exist, you could easily make one over at voip-info.org |
| 20:44.42 | [TK]D-Fender | rue_desk: Well Cisco, Polycom, and all these other compaines have their OWN samples, and don't feel the need to repo them in the way you want. |
| 20:44.45 | fexy | how do I define sccp devices dynamically in asterisk? |
| 20:44.53 | rue_desk | have a system for submitting them, something like pastebin might help |
| 20:45.19 | [TK]D-Fender | rue_desk: Repo's are ytpically for real software as opposed to phone configs that are jsut off 1 base folder whose location doesn't matter at all |
| 20:45.43 | rue_desk | [TK]D-Fender, well no thats the thing, I couldn't find any exampesl from aastra or polycom that actually had all the config options for each phone in them |
| 20:46.11 | [TK]D-Fender | rue_desk: Really didn't look too hard then. Polycom's are all in their firmware pack. |
| 20:46.21 | jjshoe | rue_desk mostly because you don't need all the options |
| 20:46.30 | jjshoe | rue_desk the admin guide is a good place to look for aastra |
| 20:46.32 | rue_desk | [TK]D-Fender, I didn't expect that |
| 20:46.59 | [TK]D-Fender | rue_desk: Expect? You said you couldn't find. |
| 20:47.06 | [TK]D-Fender | rue_desk: So which is it? |
| 20:47.19 | rue_desk | the i33 has a 1200 page manual, I'm writing a config file slowly as I find things on the phone menu that I need to set then i look up their conifg file name and write it in on the file |
| 20:47.36 | rue_desk | [TK]D-Fender, would you look in the breadbox for cookies? |
| 20:47.49 | rue_desk | how about the cookie jar for bread? |
| 20:48.05 | rue_desk | I dont consider config files firmware |
| 20:48.53 | GoRK | I am having trouble with echo on a PRI using a TE122B with hardware echo canceller. What gives? I have tuned it using milliwatt test to the CO & a loopback to Milliwatt on asterisk, but I get better results just changing txgain to -10.0 .. still have faint echoes though and when I speak over incoming audio the incoming audio cuts out. |
| 20:49.33 | rue_desk | hmm |
| 20:49.35 | GoRK | I would expect this to basically not be happening. Local endpoints are polycom ulaw all default gain settings |
| 20:49.59 | jjshoe | rue_desk ignoring everything [TK]D-Fender just said, voip-info.org is a great place to do what you just said |
| 20:50.15 | rue_desk | GoRK, I know a follow who had to use an external echocan on his T1 specifically |
| 20:50.24 | rue_desk | jjshoe, ok |
| 20:50.31 | rue_desk | jjshoe, who would I talk to? |
| 20:51.05 | jjshoe | rue_desk you wouldnt, its a wiki |
| 20:51.23 | rue_desk | hey you know, thats even better |
| 20:52.33 | thehar | Weird. |
| 20:52.44 | GoRK | it would seem to be insane yet to add an additional EC into the mix; I mean there is already 3 at minimum in the path: Polycom phone, digium hardware EC, CO's EC |
| 20:53.27 | kb3ien | anynone care to recomend any ATAs? |
| 20:53.53 | rue_desk | GoRK, http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers there is the article the fellow I was talking about wrote |
| 20:55.31 | rue_desk | heh, it might be me, but it looks like peopel added to that |
| 20:55.49 | rue_desk | yea they did, cool |
| 20:57.09 | rue_desk | I have a set of 2531's from kb1kanobe |
| 20:59.38 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 21:00.24 | rue_desk | haha I have an account! |
| 21:00.39 | rue_desk | Fri 13 of May, 2005 [15:49] rue_mohr |
| 21:01.17 | rue_desk | wonder what my password is :/ |
| 21:01.20 | eppigy | hello |
| 21:01.22 | eppigy | i am dave |
| 21:01.23 | deeperror | is there a way to get asterisk to handle a v90 connection on an inbound call? |
| 21:04.11 | jjshoe | rue_desk voip-info.org is much easier then arguing with someone about who/where/when/why/how information should be shared :) |
| 21:04.15 | *** join/#asterisk huisnah (n=nhuisman@aeko.ifa.hawaii.edu) |
| 21:04.28 | kb3ien | a random googling suggest the sipura ata 3000 is worth having. objections ? |
| 21:04.31 | huisnah | does anyone know how you might setup an outside calling for voicemail on asterisk using the gui? |
| 21:04.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:04.57 | [TK]D-Fender | kb3ien: SPA-3102 replaced it several years ago |
| 21:06.01 | [TK]D-Fender | rue_desk: configs are specific to the firmware revision, so YES, you WOULD lookin the firmware pack for your samples. |
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| 21:44.27 | keith4_ | is it possible to make a Linksys SPA941 auto-answer an incoming call? |
| 21:47.22 | *** join/#asterisk seanmh (n=seanmh@216.31.101.11) |
| 21:48.31 | keith4_ | i've found lots of people asking, on forums, and pasting dialplan snippets that *don't* work... but no conclusive example |
| 21:49.44 | *** join/#asterisk flexpulse (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:49.48 | flexpulse | Greets, all. |
| 21:50.05 | *** join/#asterisk LND (n=LND@89.193.244.183) |
| 21:52.08 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 21:53.35 | keith4_ | oh, nevermind. my problem was with the phone, not with asterisk |
| 21:53.38 | bkw_ | keith4_: it easy |
| 21:53.53 | keith4_ | bkw_: yah, seems to be working now. thanks anyway |
| 21:54.27 | bkw_ | k |
| 21:56.16 | voxter | hm, i wonder how hard it is to set up a hot line phone with a PAP2 or something |
| 21:56.50 | voxter | oh. easy. |
| 21:59.15 | eppigy | hello |
| 21:59.16 | eppigy | i am dave |
| 21:59.45 | *** join/#asterisk jacco (n=root@unaffiliated/jacco) |
| 21:59.47 | jacco | Hey guys. |
| 21:59.56 | jacco | So, I went from using static IP to DHCP. |
| 22:00.22 | jacco | However, I'm still not using dynamic=yes for the asterisk server, but I was thinking of using an agents.conf and everything. |
| 22:00.47 | jacco | Anyway, my question is... should I move it to dynamic routing or try to get things working on static? |
| 22:01.50 | freckle_home | sighs |
| 22:03.12 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 22:11.25 | jaytee | voxter, it's pretty easy to setup a hotline with a PAP2 |
| 22:11.49 | voxter | yeah i found it pretty quickly |
| 22:12.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:13.15 | jaytee | quittin time, bbiab |
| 22:15.39 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:15.43 | rue_desk | I dont have to register with polycom to download the firmware do I? cant find it.... |
| 22:17.40 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:20.03 | rue_desk | aha |
| 22:28.13 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
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| 22:34.37 | *** join/#asterisk Jacco (n=root@unaffiliated/jacco) |
| 22:34.49 | Jacco | Hey, how do I test dialing out with the asterisk shell? |
| 22:37.06 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:38.27 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 22:42.10 | flexpulse | Jacco: originate |
| 22:42.38 | flexpulse | rue_desk: yes, to get the latest contact whoever your bought your phones from |
| 22:43.51 | Jacco | No such command 'originate' (type 'help' for help) |
| 22:44.21 | flexpulse | Which version of asterisk? |
| 22:44.36 | Jacco | flexpulse: uh, how do I check? |
| 22:44.53 | flexpulse | You don't know what version of asterisk you're running? Did you compile from source? |
| 22:45.22 | Jacco | Nope. It was on a really old server. : |
| 22:45.24 | Jacco | *:| |
| 22:45.27 | Jacco | with linux 2.4. |
| 22:45.30 | Jacco | So I'm guessing pretty old. |
| 22:46.19 | flexpulse | Right when you login to the remote console with "asterisk -r", it will show the version. You can "exit" out and then "asterisk -r" to see the version. |
| 22:46.52 | flexpulse | Or you can type "show version" in the console but I'm not sure how far that goes back. |
| 22:47.34 | Jacco | uh, 1.0.7 |
| 22:47.50 | *** join/#asterisk spo1 (n=raul@dumont.nacorp.com) |
| 22:48.01 | flexpulse | Is that a debian install where asterisk is possible installed via packages? |
| 22:48.04 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:49.40 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:50.21 | flexpulse | Jacco: At any rate, you'll have to configure a SIP peer to test calling if originate isn't implmented in that version. |
| 22:50.21 | Jacco | flexpulse: I think this is slackware. |
| 22:50.32 | Jacco | flexpulse: I have a couple hardware peers. |
| 22:50.42 | Jacco | But they keep saying reorder when I try to dial out. |
| 22:51.24 | flexpulse | Jacco: That's a really old version at this point, it's going to be hard to get help in channel without upgrading to at least 1.2. I'm not familiar with the 1.0 branch (sorry) |
| 22:51.39 | *** join/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 22:51.58 | Peaceful | Can you set SIP headers in a call file? |
| 22:55.55 | Peaceful | Wow, I don't think I've ever seen it this quiet in here before. |
| 22:56.40 | Jacco | flexpulse: it's okay, the phones randomly started working again. |
| 22:56.41 | Jacco | Yaaaay. |
| 22:56.45 | Jacco | :p Anyway thanks, cya. |
| 22:59.02 | *** join/#asterisk Strom_C (n=strom@netblock-208-127-61-171.dslextreme.com) |
| 22:59.15 | flexpulse | Yuck, I don't like random. |
| 23:02.44 | [TK]D-Fender | Peaceful: You can if you seriously look at the list of channel types |
| 23:03.18 | Madkiss | what is the correct check in AEL to test whether ${CALLERID(number)} is exactly two chars long? |
| 23:04.26 | Peaceful | [TK]D-Fender: PERFECT. Thanks! |
| 23:04.38 | [TK]D-Fender | Madkiss: $[${LEN(${CALLERID(number)}) =2}] |
| 23:04.57 | Madkiss | [TK]D-Fender: thanks :) |
| 23:07.09 | eppigy | hello |
| 23:07.11 | eppigy | i am dave |
| 23:07.52 | eppigy | 16:56 < Jacco> flexpulse: it's okay, the phones randomly started working again. |
| 23:07.55 | eppigy | 16:56 < Jacco> Yaaaay. |
| 23:07.57 | eppigy | 16:56 < Jacco> :p Anyway thanks, cya. |
| 23:07.59 | eppigy | that is quality IT |
| 23:08.01 | eppigy | 16:56 -!- Jacco [n=root@unaffiliated/jacco] has quit ["leaving"] |
| 23:08.20 | eppigy | OH HEY FOR SOME STRANGE REASON THEY ARE FUNCTIONING CYA |
| 23:11.43 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:12.05 | Peaceful | [TK]D-Fender: well, almost perfect. I'm trying to set the "Alert-Info:" header, and your suggestion led me to a place that said you could set ALERT_INFO and the channel would automatically set the corresponding header. But it didn't work and I found: |
| 23:12.13 | Peaceful | "* The ALERT_INFO dialplan variable is deprecated and will be removed in coming versions of Asterisk. Please use the dialplan application sipaddheader()" in UPGRADE.txt |
| 23:12.50 | Peaceful | but you can't call an application from a call file and connect to an extension too, as far as I understand :-( |
| 23:13.26 | TrentCreek | Is this [TK]D-Fender: ?? http://tinyurl.com/4rrsnz |
| 23:13.33 | flexpulse | eppigy: lol |
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| 23:19.06 | *** join/#asterisk SparFux (n=raoul@e182027231.adsl.alicedsl.de) |
| 23:19.41 | SparFux | Hello all. Short question. Is there a way to use CAPI with mISDN and kernel 2.6.26 as of now? |
| 23:20.28 | Peaceful | [TK]D-Fender: and yet peeking into channels/chan_sip.c shows that ALERT_INFO is still supposed to be being handled....hmmm... |
| 23:26.02 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 23:26.06 | tessier_ | Hello all |
| 23:26.14 | tessier_ | Does anyone from Teliax hang out here? Or know anyone at Teliax? |
| 23:26.30 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 23:26.38 | cvnet | hello |
| 23:26.50 | tessier_ | I am this >< close to pulling all of my stuff off of them. |
| 23:27.07 | tessier_ | Not that I have a whole lot of minutes going through them so they probably won't care. |
| 23:30.18 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 23:30.31 | madduck | trying to replace a leading + with 00, I am running my head against the wall... |
| 23:30.37 | madduck | I have this dialplan: |
| 23:30.38 | madduck | [outgoing] |
| 23:30.38 | madduck | exten => _+.,1,Goto(outgoing,00${EXTEN:1},1) |
| 23:30.48 | madduck | why wouldn't that work? |
| 23:31.08 | madduck | it falls through end it appears as if it triggers _X. instead |
| 23:31.48 | Peaceful | [TK]D-Fender: Found it! By reading chan_sip.c lines ~7170-7200 revealed that you can set SIPADDHEADER=(whatever header you want to set). And you can do that in the call file. Yay! |
| 23:32.34 | madduck | and I cannot figure out how to actually debug this... |
| 23:32.56 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
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| 23:35.59 | *** part/#asterisk Peaceful (n=Peaceful@70.102.57.178) |
| 23:36.59 | tessier_ | So far teliax doesn't have a working fax machine and they have dropped my call 4 times during transfer. |
| 23:37.15 | madduck | hm, weird, now it suddenly works... |
| 23:37.47 | tessier_ | And when they finally figure out how to transfer me it goes to someones voicemail. Swell. |
| 23:37.57 | tessier_ | If this isn't resolved tomorrow I'm moving my stuff elsewhere. Ugh. |
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| 23:42.10 | tessier_ | So I just emailed Teliax and told them where the nearest Staples is so they can buy a working fax machine. |
| 23:42.22 | tessier_ | If the problem is the phone line they have larger troubles than I had feared... |
| 23:42.37 | drmessano^ | ROFL |
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| 23:59.27 | SparFux | anybody using mISDN v2? |
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