| 00:00.17 | *** join/#asterisk nailbags|work (n=nailbags@149.171.94.134) |
| 00:04.47 | *** join/#asterisk DRoBeR (n=DRoBeR@212.145.81.31) |
| 00:05.04 | DRoBeR | Hello/Hola/Hallo/Ciao all :) |
| 00:05.47 | mog | hi |
| 00:06.04 | DRoBeR | Hello, Developer :) |
| 00:06.55 | *** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) |
| 00:07.20 | aRJAy | Strom_M: that's interesting.. |
| 00:07.47 | bkw_ | HTP |
| 00:12.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-1432.bb.online.no) |
| 00:14.46 | shodan | hmm , if I have this in my sip.cfg <alertInfo voIpProt.SIP.alertInfo.1.value="VISUAL_ONLY" voIpProt.SIP.alertInfo.1.class="2"/> |
| 00:15.06 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:15.10 | *** join/#asterisk ComputerWarm (n=dan@209.29.156.37) |
| 00:15.15 | shodan | and I run this before dial , sipaddheader("alert-info:1") |
| 00:15.22 | shodan | I should get a visual ring only ? |
| 00:15.35 | ComputerWarm | is the creator of A2billing here? |
| 00:15.56 | shodan | I have also this in ringtype <VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/> |
| 00:16.10 | shodan | does my syntax look ok ? |
| 00:18.09 | rickb|server | What ports do I need for forward locally, like for the sip phone and such for asterisk? |
| 00:18.35 | Strom_M | sip is 5060 for signaling and 10000-20000 for media |
| 00:18.46 | Strom_M | iax2 is 4569 for signaling and media |
| 00:18.49 | gambolputty | rtp can be on any port numbers |
| 00:18.51 | ComputerWarm | anyone here using A2bill with the A2BCustomer_UI if so how did you fix it so all the accounts are not blocked, i have also activated it in the admin panel |
| 00:18.57 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:18.59 | TripleFFFF | <PROTECTED> |
| 00:19.02 | TripleFFFF | yo all |
| 00:19.13 | *** join/#asterisk s0lid (n=jlq@210.213.240.129) |
| 00:19.27 | TripleFFFF | wonmdering how to enable WMI in realitime for asteisk to a sip phone behind nat |
| 00:19.31 | TripleFFFF | sinc eits not sending |
| 00:19.35 | TripleFFFF | whats the port it uses ? |
| 00:19.38 | Infamy | any number i dial i get the nice female voice going the number you have dial is not in service |
| 00:19.46 | Infamy | 'evern for *65 |
| 00:20.02 | Infamy | even for *65, what did i miss? |
| 00:20.16 | Strom_M | Infamy, what version of asterisk did you install? |
| 00:20.17 | ComputerWarm | anyone |
| 00:20.37 | Infamy | Strom_M: what came with trixbox let me check |
| 00:20.45 | Strom_M | Infamy, oh hell, not trixbox |
| 00:20.49 | Strom_M | ~trixbox |
| 00:20.52 | jbot | hmm... trixbox is NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 00:21.25 | Infamy | after spending over an hour getting a cisco phone online |
| 00:21.31 | Infamy | sorry... did not mean to insult... |
| 00:21.38 | TripleFFFF | <PROTECTED> |
| 00:21.55 | TripleFFFF | wonmdering how to enable WMI in realitime for asteisk to a sip phone behind nat |
| 00:22.01 | kusznir | Unfortunately, I have yet to find anyone in #trixbox.... |
| 00:22.04 | TripleFFFF | seems my cisco is not getting the WMI\ |
| 00:22.21 | Infamy | lol yeah... |
| 00:22.33 | TripleFFFF | kusnir try freepbx |
| 00:22.50 | Strom_M | TripleFFFF, there are issues with realtime and MWI on sip phones |
| 00:22.53 | TripleFFFF | trisbox is asterisk anti hero in some sort.. since its makeing any pbx cry |
| 00:22.58 | TripleFFFF | oh |
| 00:23.03 | Strom_M | what they are, i dont remember, but i know there are issues |
| 00:23.03 | Rahail | hey guies any one know how to disabel Reg option |
| 00:23.09 | TripleFFFF | Strom_M ..hmmm.. but i do have voicemail=blah |
| 00:23.10 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:23.10 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:23.14 | *** join/#asterisk TrickyPhillips (n=TrickyPh@adsl-68-125-24-158.dsl.irvnca.pacbell.net) |
| 00:23.24 | Strom_M | TripleFFFF, it should be mailbox=blah |
| 00:24.28 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:24.41 | ComputerWarm | any a2biling users ? |
| 00:24.49 | Rahail | yes |
| 00:24.51 | kusznir | Strom_M & others: Per our conversation yesterday with means for managing Asterisk (and "vi" being the recommended one), how does one manage large-scale asterisk deployments (like 1000 extentions, corporation/university, etc)? It seems that trying to do all that by hand won't scale well. |
| 00:25.10 | Strom_M | kusznir, you write macros to help you |
| 00:25.17 | ComputerWarm | Rahail you use a2b? |
| 00:25.21 | Rahail | yes |
| 00:25.21 | Rahail | sir |
| 00:25.28 | Rahail | just start using it from last week |
| 00:25.46 | ComputerWarm | do you have a issue with the A2BCustomer_UI getting blocked account by administrator |
| 00:26.12 | Rahail | nope |
| 00:26.21 | Rahail | everyting working ok exept did |
| 00:26.26 | ComputerWarm | do you use the customer part? |
| 00:26.32 | Rahail | nope not at all |
| 00:26.46 | ComputerWarm | oh ok the did part is due to the extension i bet |
| 00:27.02 | Rahail | I tried everything but couldnt get working |
| 00:27.47 | kusznir | I've also been trying to get a2billing to work with SIP extentions (i.e., use it for postbilling of "regular" (non-callingcard) service), but it still won't pass calls through.... |
| 00:27.50 | ComputerWarm | Rahail not sure i haven`t tried that yet |
| 00:28.11 | Rahail | i think still it got few bugs |
| 00:29.06 | ComputerWarm | Rahail i am thinking its just something i am forgetting because it was working |
| 00:29.21 | ComputerWarm | and then i had to make some small changes to the a2billing.conf and it stopped working |
| 00:29.42 | Rahail | i think cheack your a2billing.conf |
| 00:29.47 | Rahail | you might disabel it |
| 00:29.53 | ComputerWarm | ya i even replaced it |
| 00:29.58 | ComputerWarm | and it doesn`t make the difference |
| 00:30.13 | Rahail | i got wired problem i dont know if is right or wrong |
| 00:30.22 | Rahail | when ever i change something on a2billing it dont work |
| 00:30.26 | Rahail | untill i reboot the box |
| 00:30.27 | Rahail | :( |
| 00:30.30 | Rahail | a2billing.conf |
| 00:30.49 | ComputerWarm | the full server? |
| 00:30.53 | Rahail | yeah |
| 00:30.59 | ComputerWarm | thats screwed up |
| 00:31.13 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 00:32.34 | Rahail | ComputerWarm do you had luck post paid option get working |
| 00:32.45 | Rahail | when you add postpaid client not to ask pin or not to announce minute |
| 00:32.49 | ComputerWarm | Rahail never tried |
| 00:33.00 | ComputerWarm | face it i don`t think i ever will prepay works perfect |
| 00:33.01 | Rahail | ok.. |
| 00:33.48 | Rahail | i got another problem |
| 00:33.53 | rickb|server | Does asterisk by default not allow login from hosts not on the network as the server? |
| 00:34.06 | Rahail | when some have 0 balance it dont ask for diffrent pin |
| 00:34.15 | ComputerWarm | rickb|server do you have host=dynamic set? |
| 00:34.15 | Rahail | notenoughcredit_cardnumber=YES |
| 00:34.25 | rickb|server | Which config file? |
| 00:34.32 | Rahail | a2billing.conf |
| 00:34.34 | ComputerWarm | what are you trying to get to login? |
| 00:35.20 | ComputerWarm | rickb|server is it a sip or iax device? |
| 00:35.27 | rickb|server | sip |
| 00:35.40 | ComputerWarm | rickb|server paste me your config for that phone |
| 00:35.49 | ComputerWarm | just message me with it so you don`t flood the channel |
| 00:35.53 | rickb|server | k |
| 00:36.18 | Strom_M | or... |
| 00:36.20 | Strom_M | PASTEBIN! |
| 00:36.23 | Strom_M | ~pb |
| 00:36.25 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 00:36.42 | ComputerWarm | Strom_C i am on a slow connection he goes to pastebin i can`t help |
| 00:37.08 | Strom_M | since when is pastebin a high-bandwidth application? :) |
| 00:37.20 | ComputerWarm | Strom_C haha try 14.4 lol |
| 00:37.42 | Strom_M | hey, ive done pastebin on worse |
| 00:37.58 | ComputerWarm | Strom_C for some reason it takes along time to load |
| 00:38.04 | ComputerWarm | man i shouldn`t work from home lol |
| 00:38.10 | Strom_M | use pastebin.ca |
| 00:38.17 | Strom_M | pastebin.com is borked |
| 00:38.27 | Strom_M | lemme see something... |
| 00:38.30 | Strom_M | ~pastebin |
| 00:38.31 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 00:39.05 | Strom_M | jbot, no, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 00:39.06 | jbot | okay, Strom_M |
| 00:47.02 | *** join/#asterisk Defraz (n=t0tal@71-213-125-59.slkc.qwest.net) |
| 00:47.07 | Defraz | Hey all! |
| 00:47.18 | map7 | Is there a way to config calls on hold to ring a phone after it's been on hold for a certain amount of time? |
| 00:47.30 | shodan | ok I -think- I have my sip.cfg right , but I'm not sure about my sipaddheader command , which is the right syntax ? sipaddheader("alertinfo:1") sipaddheader("alert-info:1") sipaddheader("alert_info:1") sipaddheader(alert-info;1) etc .... ? |
| 00:47.57 | Defraz | I am having a little trouble with g729. When I use it i loose half my call. Like the other person can her me but I can't hear them. When I switch back to Ulaw it works fine. |
| 00:49.46 | Defraz | Any ideas? |
| 00:50.22 | shodan | omg I GOT IT :))))) |
| 00:50.49 | shodan | it was sipaddheader(alert-info: Visual) |
| 00:51.11 | shodan | in your face stupid polycom phone !! |
| 00:51.40 | Defraz | Are Polycoms good phones? |
| 00:52.03 | shodan | great phone, horrible software |
| 00:52.27 | Defraz | If I were just to configure each phone via the web interface then it sucks |
| 00:52.44 | ComputerWarm | anyone here using A2bill with the A2BCustomer_UI if so how did you fix it so all the accounts are not blocked, i have also activated it in the admin panel |
| 00:52.46 | Defraz | I have always heard good things about them, compared to the Grandstreams |
| 00:52.48 | Strom_M | ciscos are pricier, but are easier to configure and have nicer handsets |
| 00:53.09 | Defraz | Yea Ciscos are $$ |
| 00:53.11 | Strom_M | Defraz, you'll hear good things about a bag of flaming poop compared to Grandstreams |
| 00:53.18 | Defraz | haha |
| 00:53.27 | Ryushin | Well, this is a new one. When you leave a message on a ip430, it shows the icon, and then locks up and auto reboots after a minute. Nice. |
| 00:53.45 | Defraz | hmmm |
| 00:53.51 | ComputerWarm | Strom_C hey my GS works great |
| 00:54.04 | ComputerWarm | everytime i throw it at someone it always hurts |
| 00:54.09 | Strom_M | hahahahahaha |
| 00:54.16 | Strom_M | yes, in that regard it's very reliable |
| 00:54.21 | ComputerWarm | lol |
| 00:54.21 | Defraz | I just recommended the company go with Polycom for teh desk phones, they were switching from a Soft phone solution. |
| 00:54.33 | Strom_M | until about the sixth time at which point the grandstream falls apart |
| 00:54.37 | ComputerWarm | i actually have one i was using the stupid thing would never stay registered to asterisk |
| 00:54.39 | Strom_M | then you're stuck throwing the "6" key at them |
| 00:54.54 | Defraz | haha |
| 00:55.30 | ComputerWarm | Strom_C nope... no joking i was living in a apartment in the 9th floor i have thrown it out the window MANY times |
| 00:55.41 | Ryushin | The hour long bootup time of the ip430 is driving me nuts. The ip601 boots in just a few minutes. |
| 00:56.00 | Strom_M | Ryushin, i have an ip430 that only takes 15 minutes to boot |
| 00:56.32 | Ryushin | I also now have two dead phones that just keep rebooting that have the latest bootroom and flash on it. I was reading that others are having a same problem. |
| 00:57.14 | Ryushin | Strom_M: Then I'm at a loss then why it takes an hour. Which bootroom and flash do you have on it? |
| 00:57.22 | TripleFFFF | i ran into problems using the Message Waiting Indicator with my Cisco 7960g phone using the realtime system under Asterisk 1.2.0. Turns out you need to set the rtcachefriends=yes in the general context of your flat file sip.conf. Once this is done, MWI starts working as usual. |
| 00:57.25 | TripleFFFF | this doesn work |
| 00:57.43 | Ryushin | Does Polycom off support for there phones without a fee? |
| 00:57.46 | Ryushin | offer |
| 00:57.55 | Strom_M | Ryushin, i have whatever the latest is |
| 00:59.22 | Ryushin | Strange. The phones aren't picking up 3.2.1 bootroom or the new 1.6.7 firmware. And the two that did, go through the reboot loop. I was messing with different bootroms (greater than 3.1) and such to try and speed up the boot process. |
| 00:59.28 | Ryushin | All to no avail. |
| 00:59.39 | ido | so who do i need to bribe to get myself an unlimited DID? :( |
| 00:59.49 | ido | on the cheap... |
| 01:00.06 | Strom_M | ido, that's what dreams are for |
| 01:00.31 | ComputerWarm | ido what do you mean unlimited? |
| 01:00.46 | ComputerWarm | ido no per minute incoming? |
| 01:01.01 | ido | i mean unlimited concurrent incoming calls |
| 01:01.04 | ido | and no per minute, preferrably |
| 01:01.31 | ComputerWarm | ido i know a guy that can do it but you have to pay per channel |
| 01:02.07 | ido | :/ |
| 01:04.55 | ComputerWarm | Rahail what verison of a2b are you using |
| 01:05.25 | TripleFFFF | ido lol |
| 01:05.31 | TripleFFFF | lets say ill make this easy |
| 01:06.21 | TripleFFFF | you want an unlimited did with unlimited minutes with unlimited channels.. for free right ? |
| 01:06.28 | TripleFFFF | take the blue pill.. |
| 01:06.29 | ido | well not for free :-/ |
| 01:06.45 | TripleFFFF | if you take the red pill therese no coming back.. |
| 01:06.45 | ido | but unlimited did and unlimited minutes and unlimited channels, sure |
| 01:09.03 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 01:17.42 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 01:23.40 | Strom_M | dear lord, why are none of the polycoms showing the correct time |
| 01:25.13 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 01:25.21 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:25.45 | shap | Wondering about the digium 4 port cards -- is there an echo cancellation option for these? Is one needed? |
| 01:26.05 | JT | 4 ports of what |
| 01:26.16 | JT | T1/E1, POTS, ISDN2 |
| 01:26.53 | shap | Sorry -- I'm look at a 4*FXO Wildcard. The TDM400P |
| 01:27.22 | JT | afaik there's not hardware echo cancellation addon |
| 01:27.37 | shap | I didn't see one. Is one needed in practice? |
| 01:27.48 | Strom_M | if you've got crappy phone lines |
| 01:28.57 | shap | Hard to know, since they aren't provisioned yet. |
| 01:29.34 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:29.42 | shap | Background: I'm setting up a new office. Contemplating IP phones on desks and a few at various homes. Trying to decide between an Asterisk-based solution vs. just buying an Epygi. |
| 01:29.45 | Strom_M | how far away from the CO are you? |
| 01:30.01 | shap | At a guess: 3048 feet |
| 01:30.11 | Strom_M | 3000 feet is relatively short |
| 01:30.17 | Strom_M | you should be fine |
| 01:30.30 | Strom_M | how many stations at the office? |
| 01:31.01 | shap | Stations within the office: probably 3-4. If need be it's no problem to dedicate a separate enet for them or buy a managed switch for QoS handling |
| 01:31.20 | shap | Or for that matter to just configure traffic shaping, since all the clients are linux clients... |
| 01:31.46 | Strom_M | actually, i was going more for analog vs digital |
| 01:31.58 | j0 | anyone here good at sed or awk? when i put stuff into a backreference, eg a regexp like (abcd)(efgh), how can i return just the efgh? |
| 01:32.25 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 01:32.59 | shap | Oh. Planning 4 desks using IP phones. For outbound lines, need two lines, but still trying to decide if I need to do PSTN or can maybe just go with broadvoice. Alleged DSL uplink is 768k, so probably have sufficient uplink for even 3 or 4 lines at a pinch. |
| 01:33.53 | shap | j0: Try making sure that those parens are prefixed by backslashes, then try \2 or &2 in the output pattern. |
| 01:33.53 | Strom_M | pstn will be more reliable |
| 01:34.23 | shap | Strom_M: So I have heard. Is the issue here voice quality or downtime? |
| 01:34.37 | shap | Strom_M: also, I'm not stuck on broadvoice in particular. |
| 01:34.44 | Strom_M | downtime |
| 01:34.50 | Strom_M | itsps can be flaky |
| 01:35.04 | shap | Strom_M: What is a reasonable expectation for that, and what is primary source of downtime? |
| 01:35.26 | Strom_M | well, PSTN line, you're connected directly to the network |
| 01:35.49 | Strom_M | ITSP, you're relying on your internet connection, their internet connection, routing in between, etc etc etc etc etc |
| 01:36.00 | j0 | shap: i guess i'm not sure where to specify the 'output' ex: echo "abcdefgh" | sed 's/(abcd)(efgh)/&1/' returns abcdefgh |
| 01:36.14 | Strom_M | and the PSTN's switches basically never go down unless they're supposed to |
| 01:36.33 | JT | which should be never :) |
| 01:36.37 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 01:37.53 | shap | j0: try &2 where you had &1. You may need to prefix the parens with backslash. If typing at shell, remember that you need to double the backslashes so they won't get treated as escapes by /bin/sh |
| 01:39.52 | shap | JT, Strom: I understand the topology issues, and I did 8 years in the bell system before coming to my senses. The main reason to go with an Epygi or Asterisk solution is so that I can fall back to PSTN. But that doesn't really answer my question. I'm not worried about the routes much -- here in baltimore we're sitting on the major national routing nexus. I'm more worried about the party who provides the PSTN out-link. |
| 01:40.36 | shap | Assuming I have a good route to the provided, is it realistic at this point to put business phones on to a VOIP business plan, or is it plain silly? |
| 01:40.55 | shap | s/provided/provider/ |
| 01:40.56 | JT | sounds fine with pstn backup |
| 01:40.57 | Strom_M | shap, what did you do in the bell system? |
| 01:41.07 | JT | and dialplan failover in event of voip provider unavailability |
| 01:41.10 | Strom_M | shap, yeah, just have a couple of analog lines as backup and you're fine |
| 01:41.52 | shap | Strom: research, initially in NLP and later in software development tools, but in the course of that I did one of the post-mortems on 5ESS. Damn that was a long time ago. |
| 01:42.16 | shap | Strom: failover will work for outbound. What are customers calling in from PSTN who are given lines at the VOIP provider likely to see as a failure rate? |
| 01:42.25 | shap | JT: ack. agree |
| 01:43.35 | Strom_M | shap, sweeet |
| 01:43.51 | Strom_M | ive got like a dozen BSP manuals in the back seat of my car right now :) |
| 01:44.00 | shap | Strom: or am I being an idiot, should get cheapest plan with (e.g.) Verizon for the sake of line termination, have people call in on that, and use the VOIP lines mainly for dialout. |
| 01:44.11 | shap | Damn, man. Give that car a rest. What did it do to *you*. :-) |
| 01:44.16 | Strom_M | HAHAHAHAHA |
| 01:45.06 | shap | Anyway, what I *think* you're telling me is (a) it's plausible, but (b) I should be prepared with a failover plan. Is that about it? |
| 01:45.38 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 01:46.54 | Strom_M | pretty much |
| 01:47.17 | shap | Okay, so two other things... |
| 01:47.45 | shap | Nah. Just one. |
| 01:48.32 | Strom_M | http://www.stromcarlson.com/misc/rear_window.jpg |
| 01:48.36 | Strom_M | that's on my car :) |
| 01:48.55 | shap | Given alternative between Asterisk and Epygi, which way should I lean? I'm pretty confident in Asterisk, and we have some kick-a*s linux people handy, but I'm not in business to support my PBX. It's very important (to me) to have something that is mostly hands-off once configured. Are we likely to get that out of Asterisk, or is that unrealistic? |
| 01:49.14 | shap | Strom: you got ripped off. That bell's whole. :-) |
| 01:49.30 | Strom_M | i can see you're not at all sour about your employment there |
| 01:50.18 | shap | Actually not, but think about the logos I survived while working there: a broken bell, a death star, and a flaming assh*le (the lucent logo) |
| 01:50.34 | TripleFFFF | YO |
| 01:50.35 | kusznir | Is there a way to enable debugging of an AGI script within asterisk? (for example, get it to log what it sends to the script and what it returns)? |
| 01:50.46 | Strom_M | shap, it depends on how you set it up. I've got an all-TDM install with fifty stations and a PRI, and they only call me for moves, adds, and changes |
| 01:51.28 | Strom_M | i have another client that just loves looping the network back onto itself; I'm out there once a month unfucking things |
| 01:52.06 | shap | Clients are way fun that way. |
| 01:54.06 | shap | Strom: Ignoring 911 issues, am I going to get away with IP-only phones within the local office, or should I plan for FXS ports? I'm asking because I'm trying to figure out the right HW config for an Asterisk option. |
| 01:54.25 | Strom_M | all-voip will work just fine |
| 01:54.42 | shap | One other thing. Suspect this is IP/analog neutral. |
| 01:54.50 | Strom_M | assuming you buy good phones like Cisco or Polycom, and not garbage like Grandstream |
| 01:55.52 | shap | Was looking at the SNOM phones. Polycom's are reported to do really badly in the non-local situation (e.g. no STUN support), and I want to use the same equipment at a couple of individual homes. Already got the word on GrandStream, but thanks. |
| 01:56.28 | shap | Also, the polycom's don't appear to handle SIPS/SRTP, unless I missed something. |
| 01:56.35 | Strom_M | Snoms are supposed to be very good |
| 01:56.55 | Strom_M | honestly, if you want the best NAT traversal, send IAXys into the field |
| 01:57.12 | shap | strom: Do I need to be thinking multiline phones, or are single-line IP phones sufficient? From the user perspective, what is perceived difference? |
| 01:57.33 | shap | strom: If I go asterisk, I'll do that. If I go epygi, not an option. I'm *really* nervous about the turnkey issue. |
| 01:57.46 | Strom_M | shap, where are you located |
| 01:57.53 | shap | strom: baltimore area |
| 01:58.01 | Strom_M | shap, for most stations, a two-line IP phone is more than plenty |
| 01:58.13 | Strom_M | for the receptionist or the executives, six-line phones :) |
| 01:59.03 | shap | strom: figured that. Question is: if the user has one-line, can they still get notice of an incoming call and park current call back at PBX (much like three-way on PSTN)? Is the second line a convenience issue or a function issue? |
| 01:59.04 | Strom_M | shap, oh ok...if you were in los angeles, I would say let me be your consultant :) |
| 01:59.18 | Strom_M | shap, i have yet to see a decent quality one-line phone |
| 01:59.28 | Strom_M | what were you thinking of using? |
| 01:59.39 | shap | Strom: thanks for the offer. Actually, one of my PhD students has been hacking Asterisk at home, and I got the okay to pay him for some consulting time. |
| 01:59.48 | Strom_M | hehe ok |
| 01:59.54 | shap | Ack on the single line phones. Like I said, looking at the SNOM 360 series |
| 02:00.40 | shap | Very impressed with SNOM's responsiveness on issues. Those guys are turning new firmware in response to bug reports. Everybody has bugs, but the SNOM guys seem to be seriously motivated about fixing them. |
| 02:01.03 | Strom_M | hm...i have little experience with snom, but the ones ive handled havent struck me as being really nice in build quality...the handsets arent comfortable to use for extended periods |
| 02:01.23 | Strom_M | the keys feel mushy too |
| 02:01.43 | Strom_M | im kind of big on tactile things when it comes to phones :) |
| 02:01.52 | Strom_M | WECo 2500 is the gold standard in that department :) |
| 02:03.00 | gambolputty | snom buttons are ok |
| 02:03.05 | gambolputty | handsets are better now |
| 02:03.12 | gambolputty | sip features rock :) |
| 02:03.12 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 02:03.24 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:03.28 | Strom_M | cisco and polycom have the buttons thing down; cisco's handset is nicer than polycom's IMHO |
| 02:03.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:03.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:04.00 | shap | Problem is that neither cisco nor polycom is doing srtp. Don't care for the local units, but for remote access I do. |
| 02:04.35 | shap | gambolputty: were the handset updates a 300 series change, or they just got better at some point? Also, how is the speakerphone quality? |
| 02:04.58 | gambolputty | 300 series yes |
| 02:05.00 | gambolputty | I have the 300 |
| 02:05.06 | shap | As far as I have found, the only handsets doing SRTP at the moment are the SNOM and the SPA-942s |
| 02:05.07 | gambolputty | speakerphone is good |
| 02:05.22 | gambolputty | I have the 190 also |
| 02:05.30 | Strom_M | the spa-942 is actually a really nice phone for what you pay |
| 02:06.01 | Strom_M | it definitely has that mid-market feel to it, but it's solid |
| 02:06.08 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 02:06.16 | shap | Any others doing SRTP that I ought to look at? We spend a depressing amount of time on phone, so it's worth getting good phones. |
| 02:06.18 | gambolputty | I use mine to dial a SIP URL. |
| 02:06.29 | gambolputty | Haven't done SRTP yet. |
| 02:06.32 | gambolputty | * doesn't do it |
| 02:06.49 | Strom_M | shap, best thing to do, honestly, is set up VPN connections and tunnel your voip over that |
| 02:07.39 | shap | gambolputty: I thought that asterisk dropped out and let the codec-level negotiation happen between the handsets? Even if it doesn't, it's a "prepare for future" kind of thing. |
| 02:07.41 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.50 | gambolputty | don't know |
| 02:08.11 | shap | strom: problem with that is that you need static IP's on both ends to configure IPSec, and most of the OTC routers aren't bright enough to do SSL VPNs |
| 02:08.34 | gambolputty | unless you use openvpn |
| 02:08.40 | gambolputty | which works with domain names |
| 02:08.44 | gambolputty | like dynamic ones :) |
| 02:10.13 | shap | gambolputty: looks good, but the problem in general is that the more stuff you home-brew, the harder it is to rebuild when your primary world-interface box goes south. |
| 02:10.34 | gambolputty | keep notes then |
| 02:11.00 | shap | gambolputty:my time to implement the restore is 10x the cost of buying the hardware that does it out of box |
| 02:11.28 | shap | strom: if I assume VPN, that clearly changes the options on handsets. Any opinions on Aastra? |
| 02:11.29 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 02:11.48 | Strom_M | Aastra stuff is good quality, but i have no experience with config |
| 02:12.07 | Strom_M | but the Aastra stuff is from the old Nortel analog telephone division, so it's quality equipment |
| 02:12.39 | shap | strom: That makes the tunneling approach more tempting. Nortel spun Aastra out? |
| 02:12.49 | Strom_M | nortel sold the analog phone division to aastra |
| 02:14.26 | shap | That explains why their analog and IP phones look alike :-) |
| 02:14.34 | Strom_M | :D |
| 02:14.57 | shap | Some folks are saying that the aastra 9000 series has flimsy buttons. |
| 02:15.20 | Strom_M | ive only touched an aastra once, but i liked it |
| 02:15.26 | Strom_M | that sounded so wrong |
| 02:15.33 | shap | Do you recall the model? |
| 02:15.40 | hads | Yes it did :) |
| 02:15.47 | Strom_M | 480i? |
| 02:15.50 | Strom_M | or something? |
| 02:15.57 | shap | Big rectangular LCD, or two-line? |
| 02:16.01 | shap | hads: what did? |
| 02:16.07 | Strom_M | dont remember |
| 02:16.17 | hads | Sounded wrong. |
| 02:16.27 | hads | The 480i is their top model. |
| 02:16.43 | shap | Definitely wanna be careful SIPing at *that* guy's cup. |
| 02:17.02 | shap | Hmm. Now that *really* sounded wrong. |
| 02:18.11 | shap | All: Would appreciate some info on Asterisk federation. My student, Scott, tells me that if you have two asterisk servers up at two different sites, they can be configured to act collaboratively? |
| 02:19.18 | shap | strom: assuming I go with asterisk, how long should I allow to get it configured for simplish office usage? |
| 02:19.39 | shap | strom: assume dedicated machine, knowledgeable OS people, total * virgins |
| 02:20.17 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 02:21.15 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 02:22.54 | shap | strom: Hmm. I'm going at this the wrong way. Ought to pay for some of your time and get on the phone. You up for it? |
| 02:22.57 | IronMan2000 | I am having problems getting my asterisk to do outbound Toll Free numbers. Can anyone help with getting this setup. |
| 02:23.28 | Strom_M | shap, sure |
| 02:23.38 | Strom_M | let me finish up here and i'll be happy to chat |
| 02:24.09 | shap | strom: email me a phone number to shap at eros-os.com |
| 02:24.18 | Strom_M | ok |
| 02:25.18 | Strom_M | sent |
| 02:26.37 | IronMan2000 | when I try to call any 1-800 number, I get a recording saying "The number you are calling cannot be called from this phone".. I know it's proably due to the setting in the extenstions.conf file, but I cannot seen to be able to get it to where I can make 800 calls. |
| 02:26.52 | IronMan2000 | does anyone here know how to help me with this problem? |
| 02:26.53 | Strom_M | IronMan2000, who are you dialing out over |
| 02:28.31 | IronMan2000 | I have a PRI with a Digium Dual Span card using asterisk. I am connected to my SIP using a GrandStream ATA, and anytime I try to dial a 800 number, it does not work. I am sure something is not setup in the extenstions.conf, but I am so new to this, I am not able to resolve it. |
| 02:30.31 | Strom_M | what are you sending as a dial string |
| 02:30.33 | IronMan2000 | here is my extenstions.conf tollfree section |
| 02:30.35 | IronMan2000 | [trunktollfree] |
| 02:30.35 | IronMan2000 | ; |
| 02:30.36 | IronMan2000 | ; Long distance context accessed through trunk interface |
| 02:30.36 | IronMan2000 | ; |
| 02:30.36 | IronMan2000 | exten => _1800NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 02:30.36 | IronMan2000 | exten => _1866NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 02:30.38 | IronMan2000 | exten => _1877NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 02:30.40 | IronMan2000 | exten => _1888NXXXXXX,1,Macro(outgoingdial25,${EXTEN}) |
| 02:30.42 | Strom_M | man, use pastebin |
| 02:30.46 | Strom_M | please |
| 02:30.49 | Strom_M | ~pb |
| 02:30.52 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 02:31.15 | Strom_M | pastebin your macro |
| 02:32.34 | IronMan2000 | ?? |
| 02:32.42 | JT | ... |
| 02:32.59 | JT | please look at what jbot said |
| 02:33.07 | IronMan2000 | I am |
| 02:33.21 | JT | past there so the channel doesn't get flooded |
| 02:33.57 | IronMan2000 | yes, I understand. did not realize my paste was that huge. |
| 02:34.11 | Strom_M | anything > 2 lines is considered huge |
| 02:34.22 | IronMan2000 | ok |
| 02:34.30 | JT | considering there's 254 people in this channel |
| 02:34.52 | IronMan2000 | just trying to get help. |
| 02:35.14 | JT | just trying to help you getting help :) |
| 02:35.51 | IronMan2000 | well since it's already been pasted and it cannot be un-done.. Can anyone help me? |
| 02:36.02 | Strom_M | like i said, pastebin your macro |
| 02:40.07 | IronMan2000 | I am trying. seems these services don't work very well. |
| 02:40.17 | JT | pastebin.ca |
| 02:40.31 | JT | seems to be one of the best |
| 02:41.11 | *** join/#asterisk aRJAy (n=aRJAy@218-214-130-112.people.net.au) |
| 02:41.37 | IronMan2000 | ok, I sent it to pastebin.ca - listed as "someone" - my extenstions.conf |
| 02:41.40 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 02:41.47 | Strom_M | sigh |
| 02:41.49 | IronMan2000 | that one worked |
| 02:41.59 | shap | All: opinions on the digium boards vs. the sangoma boards for FXO ports? |
| 02:42.00 | Strom_M | give us the URL, please |
| 02:42.21 | Strom_M | shap, on a 3000 foot loop, there is no difference |
| 02:42.23 | IronMan2000 | http://pastebin.ca/160309 |
| 02:42.51 | aRJAy | Hi there.. I'm wondering if someone could help me with a quick question: Is there an OpenSource solution/script that allows a site visitor to 'Click' on a button "Dial-Me" that then initiates a call to a VOIP number? |
| 02:43.09 | shap | strom: Got it. Probably the right model that if I ever move the box I'll be replacing the card anyway. |
| 02:43.20 | aRJAy | N.B. I'm hoping that the client need not have dedicated s/w on their machine :) |
| 02:43.32 | hads | shap: The TDM400 doesn't require the extra wanpipe layer so is a little easier to config but if you have a long loop then the A200 with it's HW echocan option is good for eliminating echo |
| 02:43.33 | Strom_M | IronMan2000, yow |
| 02:43.43 | IronMan2000 | yes strom |
| 02:43.52 | Strom_M | IronMan2000, who wrote that? |
| 02:44.16 | IronMan2000 | www.voipconnection.com |
| 02:44.18 | Strom_M | you know priority numbering is the way-old way of doing things, right? |
| 02:44.19 | shap | aRJAy: problem is that the client browser needs to know how to handle the URL protocol type, and sip just isn't going to be a protocol type known to most browsers. I don't know if IE knows about it. |
| 02:44.19 | Strom_M | ugh |
| 02:44.42 | Strom_M | there is no way i am debugging that mess |
| 02:44.52 | IronMan2000 | ok |
| 02:44.54 | aRJAy | shap: not sure either really... |
| 02:45.02 | aRJAy | shap: you know what I mean though, ey? |
| 02:45.19 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 02:45.45 | shap | aRJAy: I think so. This is more of a thing for a browser expert, though. |
| 02:47.13 | aRJAy | shap: maybe so... |
| 02:47.31 | shap | All: is there an IAXy variant for two-line phones? |
| 02:47.32 | aRJAy | I don't know :\ |
| 02:47.37 | file | Strom_M: where are 'chu?!? |
| 02:47.38 | aRJAy | Anyone else might know? |
| 02:48.27 | Strom_M | i'm near LAX |
| 02:48.32 | Strom_M | across the street from Inglewood |
| 02:49.00 | file | docelmo: you forgot to change the subject |
| 02:49.13 | IronMan2000 | anyone know how I can block certain extenstions from making LD calls? |
| 02:51.09 | Strom_M | IronMan2000, put them in a context that has no access to LD calling |
| 02:51.16 | aRJAy | shap: you know of anyone else that might have experience in this area? |
| 02:52.22 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:52.34 | shap | aRJAy: sorry, no. It's probably not a smart thing to rely on, though, since most people coming to your site won't be able to use it. |
| 02:53.37 | tengulre | benjk: are u japanese? |
| 02:53.43 | IronMan2000 | Strom, do you have any support links that may can get me started in this? |
| 02:54.05 | aRJAy | shap: it's not really a matter of having it as a point of main communication... however that being said, I think that if it were made simple enough, and as a server-side solution, then people would get great benefit from it.. |
| 02:56.04 | Strom_M | IronMan2000, i gave you one earlier: |
| 02:56.09 | Strom_M | ~book |
| 02:56.10 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:57.01 | shap | aRJAy: This can't be a server-side solution. It needs to cause a new app to start on the client side. That requires setting up a handler on the client browser. |
| 02:57.30 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:57.32 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:57.46 | aRJAy | shap: Why do you think that it can't be a server-side/ASP solution? |
| 02:57.53 | aRJAy | I don't see a reason. |
| 03:00.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:00.28 | blitzrage | evening all |
| 03:00.36 | blitzrage | anyone ever create a directory looking for polycom phones? |
| 03:00.45 | blitzrage | s/looking/lookup/g |
| 03:00.51 | JT | aRJAy: err, the whole having to have a client softphone doesn't strike you as a reason? |
| 03:01.07 | shap | aRJAy: because ultimately you need to open a softphone application on the client side. You can't do that from the server. |
| 03:01.31 | blitzrage | unless you use 'dial' from the console |
| 03:01.38 | blitzrage | (although I really have no idea what you're talking about) |
| 03:01.43 | aRJAy | Why can't a softphone application be a server-side app ? |
| 03:01.50 | file | blitzrage: oh no you didn't |
| 03:02.03 | JT | aRJAy: because it has to connect to speakers and microphone??? |
| 03:02.20 | blitzrage | aRJAy: sounds like if you had a soundcard in the server you can plug in a headset into that and perform dialing and answering from the CLI |
| 03:02.33 | JT | i mean you could be able to do it in flash or java |
| 03:02.38 | blitzrage | aRJAy: just need to enable console |
| 03:02.39 | JT | but compatibility would be low |
| 03:02.45 | JT | blitzrage: nah, that's not it |
| 03:02.45 | shap | aRJAy: if it's running on the server, how do you get the audio to/from the user on the web browser? |
| 03:02.48 | aRJAy | JT: Yes... I do understand that :) But I see no reason why that can't be server-side driven.. even if it's coded in java! |
| 03:03.05 | blitzrage | file: I did! |
| 03:03.06 | JT | he wants someone to be able to click on something on a web page to initiate a phone call |
| 03:03.20 | JT | aRJAy: java solutions really are client side |
| 03:03.27 | blitzrage | agreed |
| 03:03.34 | blitzrage | by nature of the application |
| 03:03.42 | blitzrage | you could make one server side using PHP if it creates a callfile |
| 03:03.56 | blitzrage | which is kinda a neat idea I hadn't thought of before |
| 03:04.03 | blitzrage | would only take like 20 lines of code :) |
| 03:04.10 | JT | he wants the person clicking the link to be able to talk to them |
| 03:04.14 | JT | so softphone |
| 03:04.15 | aRJAy | JT: technically, yes... but from a users perspective they don't need to d/w a specific application just for that session.. |
| 03:04.17 | shap | aRJAy: if you do this with java, you are going to need to download something that requires special authorization by the user, and then you will *still* have the problem that the user doesn't necessarily have any softphone app installed. |
| 03:04.19 | blitzrage | lol, yah.... |
| 03:04.34 | JT | or speaker and micrphone |
| 03:04.35 | file | java softphone |
| 03:04.40 | blitzrage | have it connect the call to a cell phone by calling out in two directions simultaneously |
| 03:04.40 | aRJAy | I think you get my meaning though.. |
| 03:05.06 | blitzrage | NEXT!!! |
| 03:05.12 | aRJAy | blitzrage: that's one option and that can be done via a simple PHP or HTML page.. |
| 03:05.12 | JT | that would be open to abuse, blitzrage, if it didn't require registering on website first |
| 03:05.14 | blitzrage | bkw_: oh yes I did :) |
| 03:05.29 | blitzrage | JT: so use authentication... thats simple |
| 03:05.29 | Strom_M | hahahahaha |
| 03:05.46 | JT | sure |
| 03:05.47 | JT | simple |
| 03:05.52 | JT | depends what his target is |
| 03:06.05 | blitzrage | use google, takes like 5 seconds of searching to find authentication code |
| 03:06.21 | blitzrage | JT: I've solved the calling parties from a webpage, authentication, and not having a softphone... |
| 03:06.25 | blitzrage | what else is the issue? :O |
| 03:06.28 | aRJAy | Personally.. I see a great future in this kind of web-person interaction from a site.. |
| 03:06.55 | JT | blitzrage: you mean obfuscated images? |
| 03:08.09 | blitzrage | you could make it significantly more complex if you wanted to build a full webapp softphone. If thats what you REALLY want use the ActiveX FreeWorldDialup application and connect your Asterisk to FWD. Call the PBXs FWD number, enter a hidden option and pin, then dial back out from there |
| 03:08.34 | blitzrage | both of those solutions could solve the problem in relatively little time |
| 03:08.35 | aRJAy | ah ha |
| 03:08.39 | JT | blitzrage: is that what you mean by "authentication"? |
| 03:08.48 | file | blitzrage: you should get paid for this stuff! |
| 03:08.50 | blitzrage | JT: no, from the PHP you could have htaccess auth |
| 03:08.54 | blitzrage | file: no shit eh? :) |
| 03:09.08 | JT | yeah, well we still have no idea what aRJAy' target is |
| 03:09.08 | aRJAy | So, who's up for the challenge? :) |
| 03:09.16 | JT | prospective customers, pre-existing, etc etc |
| 03:09.19 | blitzrage | aRJAy: I am a consultant for hire funny enough :) |
| 03:09.31 | blitzrage | JT: thats for him to decide |
| 03:09.35 | blitzrage | I'm just offering suggestions |
| 03:09.47 | blitzrage | depends how high-scale it needs to be |
| 03:09.51 | file | blitzrage is also a body guard for hire |
| 03:09.59 | blitzrage | if its high-scale, then the solution is to hire a developer to create a web-based softphone |
| 03:10.03 | JT | it helps decide the solution too |
| 03:10.10 | aRJAy | JT: The target is for all web users to be able to call web owners via a web page without having to install a dedicated softphone. |
| 03:10.35 | JT | "web owners"? |
| 03:10.54 | aRJAy | Site owners... |
| 03:11.04 | JT | right |
| 03:11.08 | aRJAy | www.thisismysite.com/callmeonvoip.php |
| 03:11.08 | aRJAy | :) |
| 03:11.15 | file | see I've already waited too long... and all my hope is gone |
| 03:11.21 | blitzrage | aRJAy: right -- go hire a developer to build you a web app |
| 03:11.44 | blitzrage | if you need a kludge, then anything I mentioned would work |
| 03:11.51 | aRJAy | blitzrage: right -- gimme the dosh to hire a developer to build me a web app |
| 03:12.01 | aRJAy | :) |
| 03:12.07 | file | dosh? |
| 03:12.12 | blitzrage | aRJAy: if you don't have the cash, then its not really that important -- not EVERYTHING is free |
| 03:12.15 | aRJAy | dosh = moolah |
| 03:12.16 | blitzrage | s/dosh/cash |
| 03:12.27 | aRJAy | casholla |
| 03:12.31 | file | interesting |
| 03:12.36 | JT | dosh, cash, moolah, dough |
| 03:12.38 | JT | etc etc |
| 03:12.41 | blitzrage | aRJAy: like, is this just for you to use at an office for a couple peeps, or what? |
| 03:12.55 | aRJAy | blitzrage: I'm really not after it just for myself... I think it'd be a great project to take on... |
| 03:13.04 | blitzrage | aRJAy: yep, it would. |
| 03:13.08 | blitzrage | EOF |
| 03:13.10 | file | can't have it all for nothing though, we all have to eat |
| 03:13.34 | blitzrage | aRJAy: if your intension is to come in here and convince someone to build it and give it away free, you're wasting your time |
| 03:14.30 | blitzrage | move along... nothing to see here people.... move along... nothing to see here OH WHAT A HORRIBLE PLANE CRASH! Everybody gather round, don't be shy, gather round |
| 03:15.03 | aRJAy | file: I'm not making a commercial offer... I wasn't making any requests that someone take of their time specifically... My suggestions was one of pure enquiry. |
| 03:15.48 | file | aRJAy: I know what you're talking about does exist, and there are companies doing it |
| 03:15.50 | aRJAy | file: personally, I'd like to see such an app available for most people to have the benefit out of... there need not be commercial gain from it.. |
| 03:16.09 | aRJAy | file: aye.. as I've seen.. |
| 03:16.25 | aRJAy | they are all commercially available... |
| 03:16.40 | aRJAy | non are opensource from what I can fathom.. |
| 03:18.12 | aRJAy | personally, I'd prefer to see an O/S solution as being the industry standard with this kind of thing, rather than having Skype or Google snap up the market... |
| 03:18.21 | aRJAy | so.. with all that said... |
| 03:18.27 | aRJAy | gtg |
| 03:18.50 | file | wow, I already wrote something the way I needed to without realizing it |
| 03:19.28 | blitzrage | handy |
| 03:21.52 | blitzrage | hrmmmm... TV doesn't like the rez Linux is booting up in |
| 03:22.53 | shodan | hmm , who would phone2 put a call on hold or forward to if my setup is like this phone1 => asterisk => (spa-2102 fxs) => analog phone2 ? do I have to setup the spa-2102 to do this or is asterisk listening for dtmf commands ? |
| 03:23.26 | blitzrage | shodan: well, depends if you are doing an asterisk transfer, or SIP transfer |
| 03:23.52 | blitzrage | if SIP, then the spa-2102 would be the device transferred |
| 03:25.09 | shodan | no I think that would be an asterisk transfert |
| 03:25.23 | IronMan2000 | when I call my asterisk server to get voice mail, it tells me login incorrect. I have double check my voice mail box number, and password and sure there both correct. anyone got any ideas??? |
| 03:25.24 | shodan | how can an analog phone send dtmf commands to asterisk during a call ? |
| 03:25.59 | docelmo | setup DTMF correctly |
| 03:26.03 | blitzrage | shodan: well, the SPA can listen for DTMF, then sends those to Asterisk via RFC2833 probably |
| 03:26.24 | blitzrage | could also do inband if you were using ulaw on a local network |
| 03:27.08 | blitzrage | but yah, you want to enable transfers using the 't' and/or 'T' option in Dial() and control the keys via features.conf |
| 03:28.00 | *** join/#asterisk Strom_M (n=strom@dsl001-174-194.lax1.dsl.speakeasy.net) |
| 03:28.01 | shodan | k, I'll read up on features.conf , there are probably examples on how to do just that |
| 03:28.09 | blitzrage | shodan: I just gave you the example :) |
| 03:28.19 | *** part/#asterisk DRoBeR (n=DRoBeR@212.145.81.31) |
| 03:28.35 | blitzrage | shodan: show application dial -- then just see the dtmf sequence in features.conf (will be in there by default if you did make samples) |
| 03:28.43 | file | blitzrage: what do I feel like listening to? |
| 03:28.50 | blitzrage | file: hrmmmmmm.... David Bowie |
| 03:28.52 | Strom_M | file, you feel like listening to a-ha |
| 03:29.01 | blitzrage | a-ha sux0rz |
| 03:29.07 | Strom_M | !!!!!!!!!!!! |
| 03:29.12 | blitzrage | yah that's right, I said it |
| 03:29.21 | blitzrage | apple sucks too |
| 03:29.22 | file | David Bowie? good idea |
| 03:29.30 | blitzrage | file: of course! that's why you ask me |
| 03:29.37 | file | you mean myself |
| 03:29.45 | blitzrage | I did mean that |
| 03:30.06 | file | so I'm going to buy myself a drink @ Astricon? |
| 03:30.15 | blitzrage | we're perfect dualities of each other -- you're op'd, I am not :) |
| 03:30.29 | blitzrage | file: of course! |
| 03:30.42 | blitzrage | file: with my own money! :) |
| 03:30.58 | file | I'm confused now |
| 03:31.03 | blitzrage | aren't we both |
| 03:31.19 | file | we can be heroes! |
| 03:31.33 | blitzrage | damn why doesn't wanpipe compiled on 0.4.3 astlinux -- boooo |
| 03:31.45 | blitzrage | not sure I have the skill set to fix it either.... |
| 03:32.05 | blitzrage | and my desktop won't grab an IP from linux, and I can't see anything since the TV won't accept the screen res |
| 03:32.25 | file | lame |
| 03:33.36 | blitzrage | agreed |
| 03:34.42 | blitzrage | file: yay file cleanup! :) |
| 03:36.12 | file | :) |
| 03:39.40 | *** join/#asterisk hb2hb (i=hb2hb@wishmaster.vttc.donpac.ru) |
| 03:42.15 | file | blitzrage: you better shape up! |
| 03:45.05 | *** join/#asterisk natrixgli (n=natrixgl@CPE-24-211-0-207.wi.res.rr.com) |
| 03:49.40 | *** part/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 03:51.07 | blitzrage | file: yah... I need to go for a bike ride tomorrow :) |
| 03:51.10 | blitzrage | or a rollerblade |
| 03:51.50 | file | that was not the reply I was looking for |
| 03:52.35 | blitzrage | :D |
| 03:53.00 | blitzrage | although I started at like 2pm... so there has been a lot of time between beers |
| 03:53.28 | file | gradual absorption into your system |
| 03:53.37 | blitzrage | aye |
| 03:53.43 | *** join/#asterisk Katty (n=Administ@dialup-4.244.123.250.Dial1.StLouis1.Level3.net) |
| 03:53.44 | blitzrage | thus no drunk effect :) |
| 03:53.49 | blitzrage | Katty: welcome! |
| 03:54.41 | blitzrage | group hug! |
| 03:54.43 | blitzrage | lol |
| 03:54.44 | Katty | kay. |
| 03:55.14 | *** join/#asterisk thx2000 (n=the@adsl-66-51-192-221.dslextreme.com) |
| 03:55.14 | docelmo | ME ME ME! |
| 03:55.17 | Katty | no. |
| 03:55.19 | Katty | you go home. |
| 03:55.21 | docelmo | :P |
| 03:55.39 | thx2000 | Does anyone have any recomendations for echo cancelling w/ a tdm400p? |
| 03:55.51 | docelmo | hold your breath |
| 03:56.06 | file | Digium can not be held responsible for deaths caused as a result of your TDM400P. |
| 03:56.14 | thx2000 | hehe |
| 03:56.15 | Katty | is a tdm400p for analog lines? |
| 03:56.21 | file | ie: holding your breath because of echo, using it as a weapon, etc |
| 03:56.30 | blitzrage | Katty: aye |
| 03:56.47 | Katty | thx2000: yeah, you're screwed. |
| 03:57.05 | Katty | thx2000: we ended up porting stuff down a t1, into a channel back, into our tdm cards. |
| 03:57.19 | docelmo | whois we? |
| 03:57.34 | thx2000 | I love how, before I bought this thing everyone said "oh yea, echo cancelling via software works fine w/ a tdm400p" now, im screwed |
| 03:57.50 | file | thx2000: it does work fine for most circumstances |
| 03:58.10 | russellb | if you have problems, please contact support@digium.com |
| 03:58.10 | file | if tweaked properly |
| 03:58.16 | russellb | it is their job to help you. |
| 03:58.22 | russellb | complaining here gets you nowhere :) |
| 03:58.24 | file | yeah, they have experience with echo issues - they know how to solve them |
| 03:58.30 | Katty | who i do i contact for hugging help? |
| 03:58.35 | file | Katty: meeeeee |
| 03:58.40 | Katty | i have all sorts of support issues. |
| 03:58.56 | file | Katty: do you have your warranty card handy? |
| 03:58.57 | thx2000 | Not complaining, just askin for a help. Sarcasm != help |
| 03:59.07 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:59.08 | Katty | file: i have so many issues, you can't afford to subscribe! |
| 03:59.54 | docelmo | Katty, no you just have issues.. |
| 03:59.59 | blitzrage | thx2000: yah, russellb is correct in that hardware issues are best dealt at Digium support since its included with your hardware purchase |
| 04:00.17 | Katty | docelmo: i don't think you know a thing about my issues. |
| 04:00.33 | file | Katty: you are good as a comfy pillow |
| 04:00.36 | docelmo | nope.. I just know your female and thats enough for me |
| 04:00.47 | Katty | file: when i'm not trying to steal your OJ, anyway. |
| 04:00.51 | Katty | docelmo: k |
| 04:01.35 | InHisName | I try to use VMAuthenticate(1021@context) and I am NOT prompted for pwd but account no anyway. |
| 04:10.45 | InHisName | If ${EXTEN} identifies the extn dialed, what identifies the exten used to make the dialing FROM ? |
| 04:11.47 | benjk | CALLERID |
| 04:12.37 | InHisName | CALLERID is active in extensions.conf file ? |
| 04:13.23 | blitzrage | InHisName: look in the doc/README.variables file (or doc/channelvariables.txt in trunk) |
| 04:13.35 | blitzrage | for a list of variables created by Asterisk |
| 04:15.17 | InHisName | Just tried it by changing exten => _10XX,nsaydigits(callerid) to CALLERID and it started to work! Things are case sensitive |
| 04:15.39 | blitzrage | InHisName: ${CALLERID} |
| 04:16.20 | InHisName | I use lower case in my sip.conf file and yet it needs all caps in extensions.conf, interesting. |
| 04:16.23 | blitzrage | InHisName: because you're calling a dialplan function which has to be uppercase |
| 04:16.35 | InHisName | OK, I see |
| 04:16.36 | blitzrage | *CLI> show functions |
| 04:17.05 | InHisName | Thanks. Now for something harder. |
| 04:17.27 | InHisName | I try to use VMAuthenticate(1021@context) and I am NOT prompted for pwd but account no anyway. More syntax issues ? |
| 04:17.28 | blitzrage | :) |
| 04:17.46 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 04:17.49 | blitzrage | not sure, don't have the syntax for that app memorized |
| 04:18.25 | InHisName | The helps all seem to point to what I typed, but still response is not right. |
| 04:19.06 | InHisName | The prompts are the same with and without stuff inside the (). |
| 04:22.00 | blitzrage | are you pointing at a real VM context? |
| 04:23.36 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 04:29.02 | InHisName | blitzrage the VM account exists, I can give it the 1021 and pwd and it approves me. Just wont prompt for pwd ONLY for 1021 like helps say. |
| 04:33.49 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 04:34.30 | Qwell | InHisName: You have to pass it a valid vm box |
| 04:34.39 | Qwell | oh, vmauthenticate.. |
| 04:43.27 | JT | heh, steve irwin died |
| 04:43.35 | JT | incase anyone hadn't hear already :P |
| 04:43.44 | Strom_C | yes, i just found that out on the other IRC channel I hang out on |
| 04:43.53 | docelmo | JT who's steve irwin? |
| 04:44.18 | Qwell | docelmo: The Croc Hunter. :P |
| 04:44.32 | JT | aussie guy who dressed only in khaki and did crocidile hunting, likes to say "crikey!" |
| 04:44.34 | Qwell | word travels far too fast on IRC |
| 04:44.35 | docelmo | the blonde hair'd guy? |
| 04:44.41 | JT | sandy blonde |
| 04:44.42 | docelmo | ohh ya that guy |
| 04:44.45 | Qwell | http://en.wikipedia.org/wiki/Steve_Irwin |
| 04:44.59 | docelmo | What happened? Did he finally have a fatal run in with a croc? |
| 04:45.04 | Strom_C | would it be insensitive of me to say "Crikey! He's dead!"? |
| 04:45.06 | Qwell | docelmo: sting ray...go figure |
| 04:45.24 | docelmo | http://www.news.com.au/heraldsun/story/0,21985,20349890-5005961,00.html |
| 04:45.45 | JT | Strom_C: nah |
| 04:45.55 | JT | 14:35 < JT> crikey, a sting ray! |
| 04:45.55 | JT | 14:35 < `2L> lol |
| 04:45.55 | JT | 14:35 < JT> these little critters are deadly |
| 04:46.03 | JT | australians make fun of irwin the most |
| 04:46.27 | docelmo | ya.. I have a friend that lives there that hates him |
| 04:49.40 | Qwell | new word: manifying |
| 04:49.54 | Qwell | just saw it used in the subversion build, heh |
| 04:51.34 | InHisName | Qwell vm account 1021 exists, do I need to do more to make it valid also ? |
| 04:51.49 | Qwell | InHisName: I would have to look at the code.. What version of asterisk? |
| 04:52.05 | Qwell | if the docs say it should work like that, then it should probably actually work like that |
| 04:52.13 | InHisName | Can I get version from CLI ? |
| 04:52.17 | Qwell | show version |
| 04:53.06 | InHisName | version 1.2-r7386M |
| 04:53.12 | Qwell | umm |
| 04:53.16 | Qwell | That's like...old |
| 04:53.24 | InHisName | yea sorta |
| 04:53.29 | Qwell | You're only about 30,000 revisions behind |
| 04:53.48 | Qwell | no, make that nearly 35,000 |
| 04:54.27 | Strom_C | updating?! what's that??? |
| 04:54.36 | InHisName | ha ha arent you exaggerating by just a tad ? 35K??? Several hundered a day ? |
| 04:54.41 | Qwell | No, I'm not |
| 04:54.54 | Qwell | The last revision was 41974 |
| 04:55.28 | Qwell | The last 1.2 commit was 41882 |
| 04:55.31 | InHisName | I am running astlinux off a CF memory. It updated once but needs bigger CF. I am stuck for the moment. |
| 04:57.14 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.11.124.revip2.asianet.co.th) |
| 05:11.40 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 05:17.41 | *** join/#asterisk new2voip (i=new2voip@user-24-214-152-5.knology.net) |
| 05:27.48 | new2voip | how do I make a test call from my PC to * using soft client? nothing I have tried has worked. |
| 05:28.00 | *** join/#asterisk sthfrnth (n=sthfrnth@211.97.195.195) |
| 05:28.45 | *** part/#asterisk sthfrnth (n=sthfrnth@211.97.195.195) |
| 05:30.34 | pyrom | dial the extension from asterisk itself? |
| 05:30.39 | pyrom | in the CLI |
| 05:30.46 | pyrom | use: dial extensionnumber |
| 05:31.22 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 05:37.29 | *** join/#asterisk axarob (n=ebash@spc1-barn1-0-0-cust964.asfd.broadband.ntl.com) |
| 05:38.47 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:39.36 | new2voip | dial 1234 gives "unable to re-open DSP device" but now I have a mail message telling me I have a wav file msg. How do I get that message now? |
| 05:40.32 | new2voip | now I need to call from client machine since server has no audio card I cannot play from server? |
| 05:41.12 | X-Rob | new2voip, dial *43 from your softphone |
| 05:41.17 | X-Rob | that'll do an echo test |
| 05:41.45 | X-Rob | voicemail is *97 I think |
| 05:42.35 | pyrom | new2voip, do you have a soundcard on your asterisk machine? |
| 05:42.45 | pyrom | if not that would explain the "unable to open dsp device"! |
| 05:42.54 | *** join/#asterisk DEADer (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 05:43.06 | DEADer | hello |
| 05:43.14 | DEADer | i have a question |
| 05:43.30 | DEADer | i have a asterix server i setup on fedora core 5 |
| 05:43.50 | DEADer | after the installations is complete and i start the server |
| 05:43.59 | DEADer | is there any thing else i should do |
| 05:44.06 | DEADer | because i can't connect to the server |
| 05:44.49 | X-Rob | can't connect to it with what? |
| 05:45.07 | new2voip | pyrom, no I don't have a soundcard in * machine. I was told that I don't need one. |
| 05:45.14 | DEADer | with a soft phone i installed |
| 05:45.35 | DEADer | x lite softphone |
| 05:45.57 | new2voip | pyrom, let me rephrase the audio isn't working under Fedora c5 |
| 05:47.57 | pyrom | hehe |
| 05:47.58 | pyrom | ok |
| 05:48.02 | pyrom | well, guess that's a kernel issue |
| 05:48.09 | pyrom | and you dont need one |
| 05:48.20 | pyrom | but if you have one you can dial directly from the asterisk machine via the "console" |
| 05:48.24 | pyrom | for testing it's fine |
| 05:48.26 | pyrom | but NOT needed |
| 05:48.30 | pyrom | ...gone |
| 05:49.33 | new2voip | thx, pyrom |
| 05:51.06 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:58.16 | *** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 05:58.31 | tengulre11 | how to connect GnuGK with asterisk? anybody can give me some ideas? |
| 06:00.51 | docelmo | dont do it! |
| 06:01.00 | docelmo | H323 and asterisk sucks money balls |
| 06:02.28 | tengulre11 | docelmo: really?? but my friend was successfull connect GnuGK with h323.conf? |
| 06:03.18 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 06:03.18 | docelmo | Not to mention H323 eats resources and such |
| 06:03.22 | new2voip | does the room have a preferred or recommend a soft client? |
| 06:03.24 | docelmo | And H323 is being phased out over the next year in most companies |
| 06:03.33 | docelmo | XTEN or SJPhone |
| 06:03.42 | docelmo | Xten == Eye Beam |
| 06:04.42 | docelmo | tengulre11 if your gonna do h323 and you must.. Pick a channel version and go with it. There are 3 or 4 that I know of |
| 06:05.14 | tengulre11 | docelmo, version 3 and 4? |
| 06:05.33 | tengulre11 | what 's full name CLI? |
| 06:05.47 | *** join/#asterisk lehel (n=t@82.77.41.206) |
| 06:06.07 | docelmo | There are 3 or 4 types of H323 versions for asterisk |
| 06:06.18 | docelmo | full name cli? CLI is caller id essentially |
| 06:06.42 | tengulre11 | docelmo: thank you ! :) |
| 06:11.18 | new2voip | when I try to dial *43 from sjphone I get and error "number does not exist call rejected 404 blah" I can call 6000 exten which dials myself so sjphone is talking to *. What am I doing wrong? |
| 06:11.40 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:11.42 | *** join/#asterisk daysmen3 (n=primus@host86-139-212-12.range86-139.btcentralplus.com) |
| 06:12.51 | *** join/#asterisk adorah (n=Administ@84.94.115.133.cable.012.net.il) |
| 06:19.04 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:27.22 | Dico_ | hello guys |
| 06:27.42 | Dico_ | are the files extensions.ael and extensions.conf compatible ? |
| 06:27.55 | blitzrage | ael is a parsing language that converts everything to extensions.conf which is then run by asterisk |
| 06:28.30 | blitzrage | you can use both ael and extensions.conf at the same time since both ael and extensions.conf are just brough down to dialplan languge |
| 06:28.42 | Dico_ | i see |
| 06:28.58 | Dico_ | but what is the translation into .conf |
| 06:29.05 | Dico_ | of this ael code : |
| 06:29.05 | Dico_ | context queues-manip { |
| 06:29.06 | Dico_ | + |
| 06:29.06 | Dico_ | + // Raquel Squelch |
| 06:29.06 | Dico_ | + _[IO]6121 => { |
| 06:29.06 | Dico_ | + &queue-addremove(dispatch,10); |
| 06:29.08 | Dico_ | + &queue-success(); |
| 06:29.09 | blitzrage | so asterisk will try to merge them together -- you just can't overlap -- so you can't have two contexts of the same name in both your ael and extensions.conf file since its all one dialplan |
| 06:29.10 | Dico_ | + } |
| 06:29.13 | Dico_ | + |
| 06:29.15 | Dico_ | + // Brittanica Spears |
| 06:29.17 | Dico_ | + _[IO]6165 => { |
| 06:29.20 | Dico_ | + &queue-addremove(dispatch,20); |
| 06:29.21 | blitzrage | ~pb |
| 06:29.23 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 06:29.23 | Dico_ | + &queue-success(); |
| 06:29.23 | Dico_ | + } |
| 06:29.25 | Dico_ | + |
| 06:29.28 | Dico_ | + // Rock Hudson |
| 06:29.30 | Dico_ | + _[IO]6170 => { |
| 06:29.31 | Dico_ | + &queue-addremove(sales-general,10); |
| 06:29.34 | Dico_ | + &queue-addremove(customerservice,20); |
| 06:29.36 | Dico_ | + &queue-addremove(dispatch,30); |
| 06:29.38 | Dico_ | + &queue-success(); |
| 06:29.39 | Dico_ | + } |
| 06:29.43 | hads | My god man |
| 06:29.43 | Dico_ | + |
| 06:29.45 | Dico_ | + // Saline Dye-on |
| 06:29.47 | Dico_ | + _[IO]6070 => { |
| 06:29.48 | blitzrage | lol |
| 06:29.50 | Dico_ | + &queue-addremove(sales-general,20); |
| 06:29.52 | Dico_ | + &queue-addremove(customerservice,30); |
| 06:29.53 | Dico_ | + &queue-addremove(dispatch,30); |
| 06:29.54 | *** kick/#asterisk [Dico_!n=north@unaffiliated/qwell] by Qwell (Qwell) |
| 06:30.04 | hads | Ah... |
| 06:30.24 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 06:30.32 | Dico_ | meuh ^^ |
| 06:31.22 | blitzrage | I guess I'm way behind AEL |
| 06:31.22 | tecnico | anybody from Spain ? would you understand "/" if I say "diagonal" instead of "barra" ? |
| 06:31.34 | blitzrage | tecnico: front-slash |
| 06:31.44 | blitzrage | "\" back-slash |
| 06:31.50 | tecnico | blitzrage: from Spain... (in spanish) |
| 06:32.03 | blitzrage | my names are universal :) |
| 06:32.15 | tecnico | blitzrage: :) |
| 06:32.19 | Dico_ | tecnico, lol i think i may understand |
| 06:32.46 | tecnico | Dico_: r u from Spain ? |
| 06:33.13 | Dico_ | tecnico, no but i think i may understand spanish ;0 |
| 06:33.18 | Dico_ | ;) |
| 06:33.52 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 06:34.11 | tecnico | see the things is that there are localized versions of certain words from region to region. |
| 06:35.14 | tecnico | in Spain, the "/" is said "barra", everywhere else pretty much is said "diagonal" . . And likewise, in Spain they say "almoadilla" for # , but other places say "gato" or "numero" or "tecla de numero" |
| 06:35.27 | *** join/#asterisk RoyK (n=roy@ti211210a080-1432.bb.online.no) |
| 06:37.01 | map7 | are incoming PSTN lines into Asterisk given an extension number? |
| 06:37.25 | lowlevel | map: more like port/interface names/numbers. |
| 06:38.24 | map7 | I have a snom320 and want to link an incoming line to a function key, is it possible? |
| 06:43.20 | benjk | Paypal are such a bunch of wonkers |
| 06:43.20 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:47.11 | Defraz | I am having a little trouble with g729. When I use it i loose half my call. Like the other person can her me but I can't hear them. When I switch back to Ulaw it works fine. |
| 06:48.05 | Defraz | Any ideas? |
| 06:48.19 | JT | do you have adequate G.729 licences? |
| 06:48.31 | Defraz | yes I bought 10 from digium |
| 06:48.33 | new2voip | can the room make recomendations for PSTN-to-voip # providers |
| 06:48.38 | Defraz | and regged them to the box |
| 06:48.42 | Assid | could be your provider doesnt support 729 properly |
| 06:48.58 | Defraz | Yes they do it is Teliax. |
| 06:49.02 | Assid | most providers just use passthrough |
| 06:49.08 | Defraz | I have a box that is running g729 installed the same way and it works fine. |
| 06:49.12 | Defraz | both ulaw and g729 |
| 06:49.21 | Assid | and sometimes.. their providers (from certan locations) dont support 729 |
| 06:49.32 | Defraz | this machine doesn't have as much ram is the only thing I can think of. |
| 06:50.35 | map7 | can I link a zaptel channel to an extension number for the sake of monitoring? |
| 06:50.48 | blitzrage | RAM isn't really a big factor... its almost always CPU horsepower you need to worry about |
| 06:51.00 | blitzrage | map7: yes |
| 06:51.16 | blitzrage | map7: chanspy and mixmonitor |
| 06:51.33 | map7 | thanks |
| 06:51.41 | blitzrage | np |
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| 07:58.13 | kuto | hi all, i'll be getting E1 line anytime this week, any recommendation for a digium card i can used for a 250 workstation i can used, i'll be getting the xten sip software |
| 07:58.29 | kuto | also |
| 08:01.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:04.55 | Assid | anyone know any good routers? |
| 08:13.11 | EyeCue | know any good peices of string? |
| 08:13.32 | Qwell | EyeCue: StringCo |
| 08:13.37 | EyeCue | awesome |
| 08:13.39 | mitcheloc | String Cheese? |
| 08:13.40 | *** join/#asterisk hotroot (n=michael@pD9E96D37.dip.t-dialin.net) |
| 08:13.42 | EyeCue | is the string cheap? |
| 08:13.47 | EyeCue | how long is it? |
| 08:13.48 | EyeCue | :~) |
| 08:13.53 | mitcheloc | tastes good |
| 08:13.58 | EyeCue | amen |
| 08:14.10 | *** join/#asterisk cian (n=cian@cian.ws) |
| 08:18.05 | mitcheloc | man...i want some cheese now |
| 08:18.14 | mitcheloc | there is NEVER anywhere good to eat around this time of the night =/ |
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| 08:24.39 | Winkie | hey guys |
| 08:24.52 | Winkie | I am trying to figure out the best method of checking if an extension is currently in use |
| 08:24.54 | Winkie | and by that i don't mean a phone |
| 08:25.14 | Winkie | we have several numbers (8500+) which are delivered via PRI and for some we need to check whether they already have a call active |
| 08:25.26 | Winkie | I thought of doing it by setting a global var when their call first comes in |
| 08:25.36 | Winkie | but this seems a little wasteful if there is a function built in that does something like this |
| 08:25.39 | Winkie | any suggestions? |
| 08:25.48 | creativx | extension hints? |
| 08:26.06 | Winkie | any more info? :) |
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| 08:26.53 | creativx | yeah |
| 08:26.56 | creativx | ~extension hints |
| 08:27.01 | creativx | i had to try |
| 08:27.06 | creativx | check voip-info.org :-) |
| 08:27.13 | avidal_ | hello everybody!!!!! |
| 08:27.27 | Winkie | creativx: from what i've read it's for monitoring phones, not particular extensions? |
| 08:27.29 | Winkie | i could be wrong :) |
| 08:27.31 | Winkie | also hi avidal_ |
| 08:28.16 | avidal_ | yes, hint is to monitor the state of the device |
| 08:28.41 | avidal_ | exten = 100,hint,sip/235 |
| 08:29.17 | Winkie | avidal_: i'm trying to monitor the state of a particular extension, it's hard to explain i suppose |
| 08:29.37 | Winkie | I need to ensure certain numbers which call in over our PRI are only allowed one phone call at a time |
| 08:29.43 | Winkie | if a second one calls in they get a busy tone |
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| 08:29.51 | avidal_ | and that extension is all the time the same device? |
| 08:30.13 | Winkie | no, it actually goes into a series of queues |
| 08:30.24 | Winkie | there's a lot of agents the calls are distributed to |
| 08:30.39 | creativx | ah |
| 08:30.41 | creativx | my bad |
| 08:30.44 | avidal_ | you could try to do it not with hints, but with some kind of LIMIT |
| 08:30.50 | Winkie | i thought of using setglobalvar but i don't know how wasteful that would be and how easy it would be to pick up when calls are hung up |
| 08:30.57 | Winkie | i checked call-limit but that's device only :( |
| 08:31.28 | avidal_ | you can use groups for sure |
| 08:31.44 | avidal_ | let me chek |
| 08:31.45 | avidal_ | 1 min |
| 08:35.07 | avidal_ | hi winkie |
| 08:35.11 | avidal_ | that's what you need |
| 08:35.13 | avidal_ | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup |
| 08:35.30 | Winkie | avidal_: thanks i'll check it out |
| 08:37.53 | *** part/#asterisk sergee (n=opera@195.94.224.197) |
| 08:38.17 | Winkie | avidal_: yeah that looks ideal, a little annoying i'm going to have to create a group for each but not a big deal, i can do it in a specific context |
| 08:38.56 | avidal_ | is a bit more complex for the extensions context |
| 08:39.02 | avidal_ | but makes what you need |
| 08:39.11 | Winkie | indeed it looks just fine to be honest |
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| 08:39.57 | avidal_ | i've one question, maybe someone can answer |
| 08:40.09 | avidal_ | i connect to some of my Asterisk boxes via SSH |
| 08:40.20 | avidal_ | with asterisk -vvr |
| 08:41.35 | avidal_ | and quite often the console goes "inverted", black background, and white text, making it harder to read, any suggestion to let the console as is begins (whit background, black text) ? |
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| 08:42.26 | Winkie | avidal_: that's very odd, i've never experienced that |
| 08:42.33 | Winkie | what terminal emulator do you use and what is TERM set to? |
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| 08:44.25 | avidal_ | the test is set to xtem-color |
| 08:44.31 | scage_ | hello everyone, |
| 08:44.31 | scage_ | I have a small problem regarding priority of codecs (for anyone who can answer it) |
| 08:44.32 | scage_ | <PROTECTED> |
| 08:44.32 | scage_ | I have two clients X [g.729,ulaw,alaw are allowed] and Y [ulaw allowed] |
| 08:44.32 | scage_ | when i tried to call from Y to X, it works (ulaw channel) but when i try to call from X to Y , call is dropped with error message |
| 08:44.34 | scage_ | WARNING[1958]: channel.c:2727 ast_channel_make_compatible: No path to translate from SIP/phonny-081866f0(256) to SIP/wrong1-08196da0(4) |
| 08:44.37 | scage_ | WARNING[1958]: channel.c:3605 ast_channel_bridge: Can't make SIP/phonny-081866f0 and SIP/wrong1-08196da0 compatible |
| 08:44.40 | scage_ | <PROTECTED> |
| 08:44.42 | scage_ | PLZ dont tell me to install a codec for translating g.729 to ulaw |
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| 08:49.45 | avidal_ | ;anyone having these "inversed" effect on CLI ? |
| 08:51.25 | benjk | inversed effect on CLI? whazzat? |
| 08:53.23 | phearless | desrevni ? inversed ? |
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| 08:56.35 | avidal_ | yes, i get into cli, via asterisk -vvr |
| 08:57.02 | avidal_ | and after a few seconds the screen goes "reversed", black background |
| 08:58.08 | benjk | avidal Adria is that you? |
| 08:58.24 | thx2000 | I'm trying to get asterisk to generate all my text2wave prompts when it starts up (and idealy even when i do a reload)...is there a way to go about this? |
| 08:59.26 | benjk | thx2000, write a module |
| 09:01.13 | avidal_ | yes, benjk |
| 09:03.55 | kuto | can anyone help me find a card i can use with E1 lines? |
| 09:05.24 | phearless | ok now I am really confused |
| 09:05.30 | phearless | I still got an echo :( |
| 09:05.43 | avidal_ | E1 cards:http://www.asterisk.org/hardware |
| 09:05.53 | *** join/#asterisk tlg (n=tlg@248.200-78-194.adsl-fix.skynet.be) |
| 09:12.51 | kuto | does the xten eyebeam compatible with Digium Wildcard TE412P? |
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| 09:14.03 | laichzeit | anyone have an idea what would cause this: http://pastebin.ca/16972 ? |
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| 09:15.17 | phearless | I do not know how to describe the problem |
| 09:15.34 | phearless | I got a TDM400P + FXO module + a big echo |
| 09:16.48 | *** join/#asterisk Muck- (n=Muck@145.253.170.162) |
| 09:17.11 | Muck- | hi |
| 09:22.23 | hads | phearless: Lower your txgain |
| 09:23.05 | phearless | i will try this |
| 09:24.13 | phearless | when I change zapata.conf, I ONLY have to restart asterisk ? |
| 09:25.19 | hads | Correct |
| 09:27.42 | tzafrir | if you just changed gain settings, a reload will do |
| 09:29.00 | phearless | same echo |
| 09:29.11 | phearless | I tried the fxotuner trick too |
| 09:29.12 | phearless | same echo |
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| 09:37.18 | phearless | any ideas ? |
| 09:37.34 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 09:38.13 | hads | So... what exactly did you try? |
| 09:43.20 | vlt | Hello. Any idea how to force my VoIP provider to suppress my CallerID? It must be possible 'cause when I connect to that provider from my AVM SIP hardware and switch on CLIR on the client phone it works ... |
| 09:47.05 | *** join/#asterisk RoyK (n=roy@ti211210a080-1432.bb.online.no) |
| 09:47.22 | phearless | hads: I tried fxotune |
| 09:49.20 | avidal_ | exit |
| 09:49.22 | tzafrir | BTW: regarding a reload and chan_zap: http://bugs.digium.com/view.php?id=7877 |
| 09:50.13 | tzafrir | phearless, where is the echo coming from? Do you have local analog extensions? What's the other side of the call? |
| 09:51.02 | phearless | when I call or receive a call, I got an echo on the side of the voip phones linked to asterisk |
| 09:51.23 | phearless | they are cisco 7960 voip phones |
| 09:51.48 | phearless | I use a not-ISDN line |
| 09:52.30 | phearless | for ex : mobile-POTS-fxomodule-tdm400p-asterisk-cisco7960 |
| 09:52.33 | hads | You need to learn to explain more carefully. When I said what exactly did you try I meant it. Details are the key to answers. |
| 09:52.40 | phearless | and... the cisco 7960 got an echo |
| 09:53.11 | phearless | If the cisco phone call or receive a call, it got the echo |
| 09:53.23 | phearless | but the mobile phone for ex, got no echo |
| 09:53.49 | DarKnesS_WolF | i have a small question is there is any application i can record directory to gsm format ? i mean without using astersik. |
| 09:53.57 | tzafrir | phearless, try playing with rxgain/txgain? |
| 09:54.12 | phearless | http://paste-bin.com/277 |
| 09:54.18 | phearless | here is my zapata.conf file |
| 09:55.51 | hads | So... when I said "Lower your txgain" what did you do? All you said was "same echo" which means nothing. |
| 09:56.25 | phearless | I just tried with rxgain=8.2, same |
| 09:56.32 | phearless | sorry hads |
| 09:56.55 | tzafrir | phearless, that bug report centainly applies to you. is there any chance that your PBX behaves differently after a reload? |
| 09:57.04 | tzafrir | http://bugs.digium.com/view.php?id=7877 |
| 09:57.23 | tzafrir | not relevant to the echo issue, though. Try a negative gain there |
| 09:57.42 | phearless | <hads> So... when I said "Lower your txgain" what did you do? All you said was "same echo" which means nothing. <--- like txgain=0 ? |
| 09:57.49 | tzafrir | A positive gain increases the call volume, and thus a stronger echo is generated |
| 09:57.59 | tzafrir | like txgain=-3 |
| 09:58.04 | phearless | pl |
| 09:58.05 | phearless | ok |
| 09:59.11 | hads | Start at 0 for both and take txgain down in increments of one until you get a satisfactory result. Rinse and repeat with different rxgain values. |
| 10:01.07 | phearless | txgain=-30 <-- same ! |
| 10:01.10 | phearless | is it normal ? |
| 10:02.25 | hads | With txgain at -30 you shouldn't be able to dial. Doesn't sound like the settings were picked up. Did you reload? |
| 10:02.42 | phearless | i restarted asterisk |
| 10:02.57 | phearless | /etc/init.d/asterisk restart |
| 10:06.28 | tzafrir | look at zttool. It should show you current gain settings |
| 10:11.24 | kuto | anyone here using asterisk on E1? |
| 10:11.40 | kuto | i need some few information |
| 10:13.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:14.28 | phearless | <PROTECTED> |
| 10:14.35 | phearless | In zttool |
| 10:14.38 | phearless | 0 ?! |
| 10:17.22 | creativx | hm.. i could need some input on the best way to differentiate inbound or outbound calls in the AMI |
| 10:17.34 | creativx | im trying to log call progress, both inbound and outbound |
| 10:21.30 | *** join/#asterisk JohnRobert (n=john@ip-222-200.net-220.dhcp.soton.ac.uk) |
| 10:21.49 | JohnRobert | anyone know of a softphone for gnome which supports IAX2? |
| 10:22.08 | JohnRobert | or rather, anything other than kiax? |
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| 10:23.58 | puzzled | morning |
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| 10:25.44 | MadRio | hello |
| 10:25.50 | JohnRobert | word |
| 10:26.20 | MadRio | Can somebody help me with sip under asterisk? |
| 10:26.57 | MadRio | I need write my owen module for sip |
| 10:27.24 | MadRio | where can i get info for this theme? |
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| 10:35.47 | phearless | <tzafrir> look at zttool. It should show you current gain settings |
| 10:35.50 | phearless | are you sure? |
| 10:40.25 | MrChimpy | they look MUCH better made than the TE4xxPs |
| 10:41.14 | tzafrir | phearless, hmmm... no |
| 10:42.56 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:51.56 | RoyK | MrChimpy: they work better too |
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| 10:57.34 | Nhu | Hi everyone, another * newbie is here ! |
| 10:57.50 | Ahrimanes | yay! |
| 10:58.34 | Nhu | I have been reading on this stuff non stop for the last several days. Really cool. |
| 11:01.34 | Nhu | Do you know if I have a 1 line SIP phone, can I have call waiting indicator/switch between SIP callers with the flash button? |
| 11:11.08 | tzafrir | Nhu, I don't know, but it differs between phones. Which phone do you have? |
| 11:14.18 | Nhu | I am just using softphones Xlite now for testing. What I wanted is that if both lines are in use (xlite has 2 lines) and someone calls, instead of indicating missed call, can it give a chance to switch to the new caller? |
| 11:14.23 | *** join/#asterisk Di[LV] (i=grikis@in.delfi.lv) |
| 11:15.32 | Nhu | It's almost like having another line but the logic is on the server, not the phone. |
| 11:18.15 | MrChimpy | royk: yep, that's why we've ordered them. given up on digium cards chewing up CPU |
| 11:18.41 | RoyK | MrChimpy: they eat just as much cpu, though |
| 11:18.45 | RoyK | that is, wanrouter does |
| 11:19.01 | *** join/#asterisk s0lid (n=jlq@58.69.210.140) |
| 11:19.02 | RoyK | it's asterisk's design there that's bad |
| 11:19.04 | RoyK | zaptel |
| 11:19.04 | MrChimpy | that's not what i've read around |
| 11:19.13 | RoyK | well, that's what i'm seeing on my boxes |
| 11:19.17 | MrChimpy | claims they're 50% lighter |
| 11:19.22 | MrChimpy | anyway, we're evaluating |
| 11:19.28 | RoyK | i know people disagree, but i just talk about what we're seeing here |
| 11:19.46 | MrChimpy | but even if they're the same we'd go with sangoma. much better built and firmware upgrades |
| 11:19.50 | RoyK | most time is spent in echocancel anyway, so better get hardware EC boards |
| 11:19.53 | MrChimpy | that's assuming nowt crap |
| 11:20.07 | MrChimpy | royk: yeah, we got echo cancelling digium and sangomas |
| 11:22.00 | MrChimpy | yep, just got two of them |
| 11:22.11 | MrChimpy | i'm building 240 line boxes |
| 11:22.30 | MrChimpy | they collapse at about 600 calls/min with digium and perl fastagi |
| 11:22.47 | *** join/#asterisk Di[LV] (n=I@3priedez.net) |
| 11:26.56 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:29.02 | RoyK | ok |
| 11:33.30 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:34.14 | *** join/#asterisk foxmjay (n=root@ll81-144-114-192-81.ll81.iam.net.ma) |
| 11:35.51 | *** join/#asterisk [shodan] (n=shodan@ip061.99-113-216.pppoe4.joliette.intermonde.net) |
| 11:42.01 | [shodan] | is there a way to do the following , caller call * box , press "2" I playback a message then I voicemail(s101) , next the caller either hangups or returns to the main menu , now whatever happens I want to caller 101's phone and playback a message "you received a message , press 1 to play" , and if user press 1 then play the message |
| 11:45.46 | *** join/#asterisk ramtha (n=tk@195.14.234.162) |
| 11:45.48 | ramtha | hi |
| 11:46.02 | ramtha | a carrier trys to connect over our asterisk |
| 11:46.09 | ramtha | signaling works, rtp not |
| 11:46.52 | ramtha | he says that our asterisk is sowing a list of supported codecs after invite |
| 11:47.01 | ramtha | instaed of picking one of the invite |
| 11:47.08 | ramtha | how can i change that? |
| 11:55.05 | MrChimpy | fuck. gotta send the sangoma cards back |
| 11:55.17 | MrChimpy | they don't do knight rider impressions like the digium ones! |
| 11:56.24 | *** join/#asterisk ScaryGary (i=ayxe@ns20322.ovh.net) |
| 11:56.33 | Muck- | can anyone give me a hint, why my asterisk is segfaulting? log is here: http://paste.debian.net/12231 |
| 11:57.28 | ScaryGary | Can i connect using TCP to the asterisk server from Windows messenger or can i somehow enable the "search for contacts" through udp? |
| 12:15.04 | *** join/#asterisk voipmagic (n=voipmagi@myw-stp-196-34-112-59.sentechsa.net) |
| 12:20.32 | RoyK | shouldn't 1.2.12 be out soon? |
| 12:28.13 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 12:29.26 | gaspiz | hi I'm trying to set a variable from agi and doesn't work.... it worked fine before... anyone has any idea why? maybe I did some configuration changes somewhere? |
| 12:29.49 | RoyK | gaspiz: prolly just a typo :) |
| 12:29.54 | RoyK | works for me (tm) |
| 12:31.14 | gaspiz | it gives me : 501 invalid or unknown command |
| 12:32.13 | *** join/#asterisk Makenshi (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 12:33.48 | gaspiz | I use phpagi |
| 12:34.20 | gaspiz | this is why it's so wierd, it's a fine class and I shouldn't have these kind of troubles |
| 12:34.59 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 12:40.17 | *** join/#asterisk _deg_ (n=deg@201-24-224-220.ctame704.dsl.brasiltelecom.net.br) |
| 12:43.12 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.16) |
| 12:45.49 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:53.47 | gaspiz | exit |
| 12:55.01 | phearless | [root@ouistiti:~]# /usr/src/zaptel/ztmonitor 1 -vv |
| 12:55.13 | phearless | # * ############ * |
| 12:55.25 | JohnRobert | for a one to one relationship, would I use belongsTo or hasOne (or both?) |
| 12:55.26 | phearless | I got this kind of thing with ztmonitor |
| 12:55.45 | phearless | the "<----------------(RX)---------------->" is too small |
| 12:56.02 | phearless | but I use rxgain=30 |
| 12:56.09 | phearless | it should be enough, no ? |
| 12:56.45 | *** part/#asterisk tarmot (n=tarmo@turing.uiah.fi) |
| 12:58.22 | phearless | tzafrir, hads any idea ? |
| 12:59.44 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 12:59.59 | *** join/#asterisk Leob (n=chatzill@w2kvpn-14.media.mit.edu) |
| 13:00.20 | JohnRobert | lol, sorry, totally wrong chan :p |
| 13:01.16 | Leob | Novice question: hello there, I've made a mistake during installation and now I'm getting the following msg: "asterisk dead but subsys locked" what shall I do to get rid of this error? |
| 13:02.15 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 13:07.10 | phearless | asterisk gurus ? |
| 13:07.13 | phearless | where are you ? |
| 13:07.54 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 13:10.26 | RoyK | phearless: sleeping |
| 13:10.47 | phearless | my question seems obvious :-( |
| 13:10.57 | *** join/#asterisk Niklas- (n=Niklas@213.237.44.34) |
| 13:11.33 | Niklas- | Hey.. I wonder if there is any command i can run, that dials *every* sip unit connected - is that possible without adding every sip unit to an extension? |
| 13:11.41 | phearless | @#! |
| 13:12.01 | RoyK | Niklas-: queues |
| 13:12.13 | Niklas- | But i then need to add every member to that queue, right? |
| 13:12.18 | RoyK | but you'll need to specify all sip unites, yes |
| 13:12.31 | RoyK | there's no broadcast stuff in asterisk, no |
| 13:12.32 | pablus | hi |
| 13:12.45 | Niklas- | ok thanks |
| 13:12.53 | RoyK | echo 'Buy viagra!' | mail *@*.* |
| 13:12.54 | RoyK | :P |
| 13:13.13 | Nugget | heh |
| 13:15.06 | Niklas- | Hmm, would it be possible to make asterisk play a shoutcast stream instead of plain mp3 files? |
| 13:15.34 | RoyK | Niklas-: everything is possible :) |
| 13:16.36 | Niklas- | Just wondering if theres a native support for it |
| 13:20.13 | *** join/#asterisk Lorbas (n=b@ns2.4commerce.de) |
| 13:21.10 | xheliox | Niklas-: I've done it before using mpg123. |
| 13:21.27 | xheliox | It's not "native", but it doesn't require anything extra other than some configuration.. |
| 13:21.27 | Niklas- | oi ok |
| 13:21.35 | xheliox | There's a howto someone if you Google for it. :) |
| 13:21.43 | xheliox | somewhere* |
| 13:22.15 | Niklas- | great thanks |
| 13:28.35 | *** join/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk) |
| 13:29.00 | E-bola | If i wanna look at some hardware cards for my asterisk server (considering switchin from a sip to a pstn setup) what sites should i be looking at? |
| 13:29.09 | E-bola | Whats the best cards to get for use with asterisk? |
| 13:29.28 | E-bola | I heard alot about some asian "fakes" or something like that |
| 13:31.55 | tzafrir | phearless, 30 is a very large value for rxgain/txgain |
| 13:32.59 | DarKnesS_WolF | i wnat to do something on my dialplan after the user leaves a voice message the line got hanged up |
| 13:33.02 | coppice | changing rx and tx gain is the biggest cause of problems in * installations. |
| 13:33.09 | DarKnesS_WolF | exten => s,5,Dial(${ARG2},20) ; 20sec timeout |
| 13:33.10 | DarKnesS_WolF | exten => s,6,VoiceMail(su${ARG1}@itsyn) |
| 13:33.10 | DarKnesS_WolF | exten => s,7,hangup |
| 13:33.16 | DarKnesS_WolF | this is not working. |
| 13:33.21 | phearless | tzafrir: but I always got : |
| 13:33.25 | phearless | # * ############ * |
| 13:33.37 | phearless | so the RX is far too small ! |
| 13:34.10 | DarKnesS_WolF | after i leave the message it's want me hangup |
| 13:38.11 | *** join/#asterisk EzWayz (n=ez@c207.134.174-180.clta.globetrotter.net) |
| 13:40.12 | EzWayz | hi , i hear a ticking noise every 65 seconds exactly while speaking with someone, how to solve this problem , thanks a lot for you help |
| 13:41.04 | *** join/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 13:41.15 | *** join/#asterisk CyberPony (n=CyberPon@cpe-071-075-174-216.carolina.res.rr.com) |
| 13:42.10 | phearless | tzafrir? |
| 13:43.21 | *** join/#asterisk aixa (n=Miranda@www.crediweb.lv) |
| 13:43.37 | tzafrir | maybe you always get a saturated audio |
| 13:43.39 | aixa | hi |
| 13:43.42 | tzafrir | anyway, I got to go |
| 13:44.07 | aixa | any news on changes to available spandsp source? |
| 13:48.34 | phearless | <tzafrir> maybe you always get a saturated audio <---- yes it is my problem |
| 13:49.08 | phearless | rxgain=-30 |
| 13:49.08 | phearless | txgain=-10 |
| 13:49.13 | phearless | and still the SAME echo |
| 13:49.19 | phearless | and I rebooted the machine |
| 13:51.11 | *** join/#asterisk _deg_ (n=deg@201-24-224-220.ctame704.dsl.brasiltelecom.net.br) |
| 13:52.06 | *** join/#asterisk Katty (n=Administ@dialup-4.244.123.165.Dial1.StLouis1.Level3.net) |
| 13:52.24 | Katty | ... |
| 13:52.44 | EzWayz | satured audio can produce tick sound repeated exacly every 65 second ? |
| 13:55.19 | JohnRobert | http://www.ipchitchat.com/products/ipc-168.htm << can anyone tell me if this ATA looks ok? I'm unsure of whether it supports the best/most common codecs |
| 13:55.33 | phearless | I am fed up by this echo. |
| 13:56.12 | Muck- | hehe |
| 13:56.12 | dlynes_laptop | totally |
| 13:56.14 | dlynes_laptop | totally |
| 13:56.43 | Muck- | can anyone give me a hint, why my asterisk is segfaulting? log is here: http://paste.debian.net/12231 |
| 13:58.33 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 14:02.39 | coppice | people love echo. their words say they don't, but their actions says they do |
| 14:05.59 | JohnRobert | you know ATA's that claim IAX2 support...does that mean they can connect directly to an asterisk server via the IAX2 protocol? |
| 14:06.04 | JohnRobert | (and make calls through it etc) |
| 14:06.25 | JohnRobert | the voip supplier I want to use is IAX2 only and not SIP |
| 14:08.14 | phearless | coppice: don't have fun of my echo please ;-( |
| 14:08.19 | phearless | echo? |
| 14:08.20 | phearless | echo. |
| 14:09.38 | phearless | rxgain=30 |
| 14:09.38 | phearless | txgain=30 |
| 14:09.54 | *** part/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 14:09.57 | phearless | even with dumb settings like this, NOTHING CHANGE |
| 14:10.06 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.200) |
| 14:10.09 | Kernel_Core | hi all |
| 14:10.28 | coppice | phearless: with the gains set like that everything is in serious overload, and nothing will work right |
| 14:10.29 | phearless | Kernel_Core: are you an echo guru ? |
| 14:10.52 | phearless | coppice: with 30-30, it is not worse than 0-0 |
| 14:10.56 | phearless | just the same crap |
| 14:11.18 | Kernel_Core | asterisk by default plays 45 as " -- Playing 'digits/40' (language 'en')" and -- Playing 'digits/5' (language 'en') , how do I force asterisk to play 45.gsm file instead of playing 40.gsm and 2.gsm ? |
| 14:11.50 | Kernel_Core | phearless: what is your equipment ? |
| 14:11.55 | Lorbas | phearless, try to set the tx gain lower than rx gain. |
| 14:12.23 | phearless | Kernel_Core: TDM400P + FXO module + cisco 7960 |
| 14:12.23 | Lorbas | phalacee, to minimize the difference between outgoing signal and the returning echo |
| 14:12.30 | phearless | ok Lorbas |
| 14:12.50 | Lorbas | phalacee, this way the software has a better chance to detect, that the returning signal actually is echo |
| 14:13.15 | phearless | but do I have to reboot the computer to apply the settings? |
| 14:13.25 | phearless | because I never see any difference |
| 14:14.01 | Lorbas | no, but maybe to unload the zap modules |
| 14:14.12 | Kernel_Core | phalacee: use echocancelwhenbridged=no echotraining=800 rxgain=9 txgain=1.7 in your zapata.conf |
| 14:14.25 | Niklas- | Hmm, isn't it possible to use DB functions in an extension? Like Dial(SIP/moo,DB(..)) ? |
| 14:14.47 | Lorbas | phearless, eg. i do txgain=-4.0 and rxgain=0 |
| 14:15.11 | Lorbas | phearless, but it depends on your phones as well |
| 14:15.26 | Kernel_Core | Lorbas: it makes echo , because your txgain is higher than your rxgain |
| 14:16.10 | Lorbas | Kernel_Core, it doesnt make echo with me -- but as you (maybe) can see, my tx is lower than my rx |
| 14:16.20 | *** join/#asterisk xyklopz (n=xyklopzi@86.122.8.28) |
| 14:16.30 | Lorbas | Kernel_Core, and thats what i suggested to phearless |
| 14:16.44 | Lorbas | Kernel_Core, but maybe i am wrong here |
| 14:16.54 | xyklopz | I need help with setting up DTMF call hold/resume + transfer in 1.2.11 Asterisk - I'm using some cordless phones plugged in a SIP ATA |
| 14:17.21 | xyklopz | I was just recently told something to the effect of # # (exten) # could transfer a call to exten |
| 14:18.06 | Lorbas | phearless, Kernel_Core: anyway these settings work for me -- and i dont think we differ here |
| 14:18.21 | Kernel_Core | Lorbas: Okey |
| 14:19.49 | Niklas- | Any guide on howto dynamic enable/disable voicemail via the phone? |
| 14:20.31 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 14:21.11 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 14:21.43 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:22.37 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:22.41 | MRH2 | Hi can anybody suggest why recently I am getting an NMI error on zaptel load and a "PCI Parity error"(on a dell 2650) |
| 14:23.10 | MRH2 | could the card (digium 411) be faulty? |
| 14:23.26 | *** join/#asterisk shortguy_ (i=shortguy@panoramix.divadeli.nl) |
| 14:23.49 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 14:24.13 | coppice | it could. a dirty PCI socket is often at fault, though. try plugging the card in a few times to clean up the contacts |
| 14:24.52 | RoyK | ping |
| 14:25.17 | MRH2 | ok i will give it a shot, anything else i can check while i'm in there? |
| 14:25.51 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:26.28 | Lorbas | phearless, as i understand it, the trick is to intensify the echo signal (rx) to such an extent, that the echo canceller can effectively detect the similarities between the tx and the rx signal |
| 14:26.46 | Lorbas | phalacee, this implies txgain < rxgain |
| 14:27.13 | *** join/#asterisk markQing (n=mark@82-197-202-242.dsl.cambrium.nl) |
| 14:27.26 | MRH2 | if the slot looks ok how do i decide whether the card is at fault? |
| 14:27.56 | coppice | trying different slots is a good idea too. if that fails, suspect the card |
| 14:29.36 | BjornRobertsson | There is some sort of weird delay in SIP, so when dialling a SIP extension, it takes up to 10 secs before the phone rings |
| 14:31.30 | *** join/#asterisk willy_1234 (n=icechat5@62.231.36.101) |
| 14:32.49 | MRH2 | ok while i am messing with it - are there any other diagnostic tools that may indicate a card error - ztest result is ok, zttool looks ok |
| 14:33.24 | willy_1234 | ok how do u get asterisk to hang up |
| 14:33.40 | willy_1234 | im using a TDM04B |
| 14:34.23 | phearless | <----------------(RX)----------------> <----------------(TX)----------------> |
| 14:34.36 | phearless | <PROTECTED> |
| 14:34.38 | Di[LV] | MRH2 try put # in the end of number or check extensions.conf |
| 14:34.46 | phearless | it is like this during a conversation |
| 14:34.50 | phearless | Lorbas |
| 14:34.55 | *** join/#asterisk S^P (n=Masood@203.148.73.236) |
| 14:35.04 | MRH2 | huh? |
| 14:35.13 | phearless | with : |
| 14:35.17 | phearless | rxgain=0 |
| 14:35.17 | phearless | txgain=-4 |
| 14:35.26 | Di[LV] | sory it was for BjornRobertsson |
| 14:35.55 | BjornRobertsson | never mind, it was #1 as dialling plan |
| 14:36.30 | willy_1234 | hrlp |
| 14:36.36 | willy_1234 | help |
| 14:36.38 | *** join/#asterisk s0lid (n=jlq@202.124.153.200) |
| 14:37.19 | willy_1234 | ok how do u get asterisk to hang up an incoming call as when the caller hangs up the line does not stop ringing |
| 14:37.58 | blitzrage | willy_1234: Hangup() |
| 14:39.15 | willy_1234 | yeah but when some one dials the call goes to a queue and the extensions do not stop ringing if someone dials |
| 14:39.22 | S^P | how we can check the status of an ongoing call, i mean cdr only written when call done but i wana knw while call going on. any CLI command? |
| 14:41.02 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:41.27 | DrukenHME | morning everyone |
| 14:41.35 | *** join/#asterisk ESCulapio_ (n=ESCulapi@200.88.44.66) |
| 14:41.49 | xyklopz | i need help setting up the blindxfer feature |
| 14:41.50 | Chaz6 | hey, it's afternoon here! |
| 14:41.58 | xyklopz | I am using the 't' option in the Dial |
| 14:42.10 | xyklopz | but whenever I press #1 nothing happens in turn of transfering the call |
| 14:42.38 | xyklopz | and I'm getting ast_bridge_call: Bridge failed... warnings |
| 14:42.45 | DrukenHME | anyone know the iax2 debug cause codes? |
| 14:43.58 | DrukenHME | http://pastebin.ca/160604 |
| 14:44.47 | ESCulapio_ | Hi as it is I cosay of Italy area? |
| 14:45.05 | ESCulapio_ | are code of italy ? |
| 14:45.50 | *** join/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 14:46.21 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 14:46.32 | TripleFFFF | wahts the damn comman dto read digits ? like 4 digits ? |
| 14:47.09 | *** part/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 14:47.46 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 14:48.25 | xyklopz | can anyone help me with the blind transfer feature? |
| 14:49.44 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 14:49.49 | Seba_soy | hello guys |
| 14:49.53 | DrukenHME | xyklopz: on what hardware? |
| 14:50.07 | Seba_soy | someone installed DIALOGIC cards on asterisk? |
| 14:50.31 | coppice | dialogic? ewwww |
| 14:50.45 | Seba_soy | dialogic cards,yes... |
| 14:50.53 | coppice | aren't they eicon cards now? |
| 14:50.59 | Seba_soy | <PROTECTED> |
| 14:51.06 | Seba_soy | those models |
| 14:51.09 | coppice | nasty |
| 14:51.26 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 14:51.27 | Seba_soy | I want to know if it can be installed or I am wasting my time |
| 14:52.02 | coppice | they can, but i think you need to pay for the drivers. they also work badly, as they have rather high latency |
| 14:52.27 | Seba_soy | Do you know steps to install it cards? |
| 14:52.33 | xyklopz | ~[ DrukenHME ]~ between two SIP ATAs (Linksys PAP2) |
| 14:52.34 | Seba_soy | this cards. |
| 14:53.07 | coppice | you need to contact digium for the channel driver. |
| 14:53.10 | xyklopz | DrukenHME, I have the blindxfer => ## in features.conf |
| 14:53.15 | *** join/#asterisk lunaphyte (n=lunaphyt@71.120.128.10) |
| 14:53.18 | Seba_soy | aaaa ok, ... |
| 14:53.20 | Seba_soy | I will try |
| 14:53.52 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 14:53.56 | Seba_soy | so, zaptel standard driver will not work... :( |
| 14:53.59 | xyklopz | DrukenHME, but I keep getting the bridge failed warning's whenever I press # and hangup |
| 14:54.03 | S^P | how we can check the status of an ongoing call, i mean cdr only written when call done but i wana knw while call going on. any CLI command? |
| 14:54.10 | coppice | nope |
| 14:54.15 | xyklopz | DrukenHME, nor do I hear the "Transfer" as stated on voip-info.org |
| 14:57.03 | *** part/#asterisk axarob (n=ebash@spc1-barn1-0-0-cust964.asfd.broadband.ntl.com) |
| 14:57.29 | Lorbas | phearless, sorry i had a phone call |
| 14:57.40 | phearless | ok |
| 14:57.50 | Lorbas | phearless, so what about your echo |
| 14:57.55 | *** join/#asterisk zerthimon (n=Zerthimo@80.74.110.153) |
| 14:57.59 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 14:58.26 | phearless | same echo, I do not really see how to modify the ###### rate, in ztmonitor |
| 14:58.45 | phearless | I can change the txgain txgain, nothing seems to change |
| 14:58.51 | DrukenHME | xyklopz: do you dial with the t dial option |
| 14:59.02 | Lorbas | phearless, you can scale up the rxgain |
| 14:59.30 | DrukenHME | coppice: do you know what cause code 29 is for an iax2 registration ? |
| 14:59.35 | phearless | to 4, for ex? |
| 14:59.48 | Lorbas | can you pastebin your zapata.conf? |
| 15:00.02 | zerthimon | question: how do I pick up someone's else call from my softphone ? |
| 15:00.10 | creativx | pickup() |
| 15:00.17 | phearless | yes Lorbas |
| 15:00.31 | *** join/#asterisk bhickey (n=chatzill@213-202-166-115.bas503.dsl.esat.net) |
| 15:00.42 | phearless | http://paste-bin.com/282 |
| 15:00.44 | phearless | zapata.conf |
| 15:00.47 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:01.26 | zerthimon | creativx: how do I configure a dialplan so I can pickup any call ? |
| 15:02.10 | zerthimon | creativx: can you point me to an example for this ? |
| 15:02.28 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 15:03.07 | coppice | DrunkenHME: yes thanks. I know. I just spent 30 seconds looking it up |
| 15:03.29 | *** part/#asterisk bhickey (n=chatzill@213-202-166-115.bas503.dsl.esat.net) |
| 15:05.55 | creativx | zerthimon: you could set up an extension |
| 15:06.03 | creativx | that grabs dtmf input |
| 15:06.06 | creativx | and pickups that |
| 15:06.40 | creativx | http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 15:06.41 | zerthimon | aha, ok, thanks, out to try that |
| 15:06.48 | Lorbas | phearless, seems ok to me. does the amplitude of ztmonitor change while you speek? |
| 15:07.05 | Lorbas | phalacee, i dont know how the wink parameter influences stuff |
| 15:07.07 | DrukenHME | coppice: mind informing me where you looked it up? since i googled it, and got bumkiss |
| 15:07.24 | phearless | Lorbas: yes |
| 15:07.28 | phearless | it changes |
| 15:07.30 | coppice | its very very obscure. its called causes.h |
| 15:08.00 | DrukenHME | ahh,... yes.. i guess that would work... |
| 15:08.13 | DrukenHME | however, NOT being a programmer, i tend not to think of that... |
| 15:08.16 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:08.23 | creativx | gotta love them header files |
| 15:08.34 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 15:08.56 | blebleble | I have a digium card in my asterisk box, (don't need it anymore) but as soon as i remove it asterisk will no longer start, any idea where i need to pull a config out of or why it would occur? |
| 15:09.17 | xyklopz | DrukenHME, the t option is in the Dial cmd |
| 15:09.38 | [TK]D-Fender | blebleble : empty out your zapata.conf & zaptel.conf |
| 15:09.43 | *** join/#asterisk theblue (n=theblue@pool-72-83-124-196.washdc.fios.verizon.net) |
| 15:10.15 | theblue | Hi all. |
| 15:10.38 | blebleble | TK-D-Fender: than you |
| 15:10.59 | theblue | I've got a Nazi ISP, and it blocks port 5060 incoming, is there anyway to allow SIP traffic in on a different port? |
| 15:11.14 | theblue | That the remote server (fwd.pulver.net) would be able to operate on? |
| 15:11.15 | DrukenHME | try 5061? |
| 15:11.38 | theblue | But is FWD able to use 5061? |
| 15:11.39 | xyklopz | DrukenHME, I'm getting a Bridge failed |
| 15:11.42 | [TK]D-Fender | theblue : Use IAX for FWD |
| 15:11.52 | theblue | [TK]D-Fender, They block the IAX port, too. |
| 15:11.57 | xyklopz | Druken, and it only happens when I try to transfer |
| 15:12.06 | [TK]D-Fender | theblue : Ok, they're assholes alright.... |
| 15:12.15 | phearless | Lorbas : |
| 15:12.15 | Lorbas | phearless, maybe the low return amplitude is due to a similar ec on the other side? |
| 15:12.18 | theblue | [TK]D-Fender, What do you expect, it's a telco. |
| 15:12.21 | file | [TK]D-Fender: eep you're back |
| 15:12.24 | Lorbas | phearless, did you try with different destinations? |
| 15:12.25 | *** join/#asterisk lorinc (n=ang@caracas-4252.adsl.interware.hu) |
| 15:12.36 | [TK]D-Fender | file : Indeed, and the ocean is missing many fish :) |
| 15:12.59 | phearless | the "* = Max Audio Hit" is still low in the RX |
| 15:13.24 | theblue | [TK]D-Fender, The end result is that no incoming calls are accepted, and even outgoing ones cut off after about 30 minutes. |
| 15:13.44 | phearless | <Lorbas> phearless, maybe the low return amplitude is due to a similar ec on the other side? <-- what is an ec ? |
| 15:14.08 | phearless | <Lorbas> phearless, did you try with different destinations? <-- yes, there is always an echo on the side of the cisco 7960 phones |
| 15:14.11 | Lorbas | depends on context - here echo canceller |
| 15:14.32 | phearless | there is no echo on the other side |
| 15:14.42 | Lorbas | phalacee, i meant did you try to ztmonitor with different destinations |
| 15:14.57 | phearless | my map is : mobile-POTS-fxomodule-tdm400p-asterisk-cisco7960 |
| 15:15.13 | phearless | I use : [root@ouistiti:~]# /usr/src/zaptel/ztmonitor 1 -vv |
| 15:15.34 | phearless | and when I call from a landline or a mobiel to asterisk, asterisk got an echo |
| 15:15.35 | theblue | [TK]D-Fender, Any advice? |
| 15:15.38 | phearless | mobile* |
| 15:16.27 | xyklopz | anyone? I keep getting Bridge failed on two SIP channels and it is occurring when I am trying to use the blindxfer feature for transferring between two extensions |
| 15:16.29 | Lorbas | phalacee, ok i meant: is the low rx amplitude independent of the (outbound) call destination |
| 15:16.38 | af_ | wow a fourbri from digium! |
| 15:17.26 | phearless | <Lorbas> phalacee, ok i meant: is the low rx amplitude independent of the (outbound) call destination <-- i will retry, one minute... |
| 15:17.26 | Ciber311 | anyone here using voicepulse? |
| 15:17.37 | Seba_soy | so, anybody have installed DIALOGIC on asterisk? |
| 15:18.01 | phearless | and I am not phalacee by the way |
| 15:18.17 | Lorbas | phalacee, try to rise rxgain to a very hig value, maybe 60, just to check any difference in the output |
| 15:18.49 | Lorbas | phalacee, if it doesn't change, you maybe didn not unload the driver fully or sth. |
| 15:19.28 | Lorbas | phearless, (sorry) to my experience the amplitude changes at some point |
| 15:20.04 | Lorbas | phearless, even if it is distorted then. But there should be a visible change |
| 15:20.18 | phearless | I reboot the box between each change |
| 15:20.19 | phearless | ok |
| 15:20.21 | inv_arp[work] | is there a cmdline fast way to create a blank .wav file |
| 15:21.03 | Lorbas | phearless, strange, that. |
| 15:21.06 | hotroot | does anyone know, what I've to do to make asterisk playing a stuttered dialtone on my ISDN-Phone if there are new voicemails? |
| 15:21.33 | S^P | any way we can check active calls time? |
| 15:23.46 | phearless | with another phone, same values, in ztmonitor |
| 15:24.56 | Lorbas | phearless, i meant a different destination in landline or mobile |
| 15:24.58 | [TK]D-Fender | theblue : I think they may be interfereing with RTP as well.... |
| 15:25.10 | phearless | I mean I called another mobile and landline |
| 15:25.17 | theblue | [TK]D-Fender, What leads you to this? |
| 15:25.27 | Lorbas | phearless, ah ok |
| 15:25.33 | phearless | I have set rxgain=100, and rebooted |
| 15:25.35 | phearless | I will see |
| 15:26.27 | xyklopz | okay ... the blindxfer option in features.conf being set to only one # solved the problem |
| 15:26.47 | xyklopz | but now I'm getting a weird NOTICE and WARNING |
| 15:27.24 | [TK]D-Fender | theblue : The fact that calls in progress cut off as well. |
| 15:27.34 | theblue | True. |
| 15:27.50 | theblue | But is there anything I can do, short of changing my provider, or upgrading to the "Business" package? |
| 15:28.59 | phearless | SAME SAME SAME SAME CRAP with : |
| 15:29.01 | phearless | rxgain=100 |
| 15:29.01 | phearless | txgain=-4 |
| 15:29.10 | phearless | I can call |
| 15:29.15 | phearless | receive call |
| 15:29.17 | phearless | ... |
| 15:29.21 | phearless | but I got a damned echo |
| 15:29.26 | phearless | SAME echo |
| 15:29.34 | phearless | since the beginning |
| 15:29.43 | phearless | and I reboot the pc between each try |
| 15:29.51 | phearless | these settings does nto change anything |
| 15:30.21 | phearless | I modify /etc/asterisk/zapata.conf |
| 15:30.26 | phearless | it is the good file ?!!!!!1??? |
| 15:32.02 | [TK]D-Fender | phearless : pastebin your zapata.conf |
| 15:32.06 | Lorbas | phearless, yes, if your hardware is driven by zap. hmm. version of asterisk? |
| 15:32.24 | Lorbas | [TK]D-Fender, http://paste-bin.com/282 |
| 15:32.27 | phearless | [TK]D-Fender: http://paste-bin.com/282 |
| 15:32.30 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 15:33.33 | Lorbas | phearless, i have a different ordering of statements in zapata. conf, i will reread yours |
| 15:33.36 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 15:34.02 | phearless | Asterisk 1.2.10 |
| 15:34.04 | phearless | from svn |
| 15:34.19 | phearless | zaptel from svn, from last week |
| 15:35.09 | MRH2 | getting no audio on sip transfer when originating a call from Agent (Local) and reinvite=yes - will sip debug be enough output for a bug report? |
| 15:38.40 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:39.56 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 15:40.01 | phearless | so... |
| 15:40.11 | Lorbas | phearless, i think that is it |
| 15:40.22 | phearless | ? |
| 15:40.39 | Lorbas | phearless, i ithink you need to put the rx/txgain stuff before the channel=>1 |
| 15:41.04 | phearless | it seems completly retarded but I will try |
| 15:41.24 | Lorbas | phearless, please read the section "Creating Channels" from http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 15:41.43 | Lorbas | phearless, no it does not once you read the quoted section |
| 15:41.56 | MRH2 | woah - is that the mixmonitor issue fixed thanks to file?? |
| 15:42.00 | S^P | any way we can check active calls time? |
| 15:43.05 | Lorbas | phearless, i dont know if that is still valid for 1.2.x but it makes a plausible explanation |
| 15:43.26 | Lorbas | phearless, for the fact that changein values dont change anything |
| 15:43.52 | Lorbas | phearless, and it implies that a lot of other stuff you have in there does not have any effect either |
| 15:44.22 | phearless | okay so there is an "order" for the config file lines |
| 15:44.39 | phearless | is there any why to guess/find this order ? |
| 15:45.02 | Lorbas | yes, there must be, since you can change values from channel to channel (provided you have some more) |
| 15:45.13 | phearless | I know that : |
| 15:45.15 | phearless | [channels] |
| 15:45.16 | phearless | signalling=fxs_ks |
| 15:45.20 | phearless | must be at the top |
| 15:45.28 | phearless | but I did not know that there is a setting order |
| 15:46.17 | phearless | is it writte somewhere on the voipwiki ? |
| 15:46.17 | Lorbas | dunno - its on voip-info.org, the one i pasted |
| 15:46.26 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 15:46.53 | Lorbas | phearless, the section is one simple sentence, that explains it all. |
| 15:47.01 | shap | I have a user who wants wireless handset. Anybody have experience with the Uniden 1686, or other suggestions to consider? Any decent wifi handsets out there? |
| 15:47.16 | Lorbas | phearless, just set up all the values *before* the channel=> stetement |
| 15:47.29 | phearless | yes I will do this :) |
| 15:49.30 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:50.33 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:52.33 | Lorbas | phearless, so does it have any effect? |
| 15:54.30 | phearless | it seems better |
| 15:54.41 | phearless | I will let you know in a few minutes |
| 16:07.27 | Lorbas | phearless, need to go now |
| 16:07.57 | Lorbas | hope you succeed |
| 16:07.59 | *** part/#asterisk Lorbas (n=b@ns2.4commerce.de) |
| 16:10.57 | phearless | IT WORKS !!!!!!!! |
| 16:11.00 | phearless | YEPEEEEEE |
| 16:11.09 | phearless | and lorbas is gone :( |
| 16:11.39 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:11.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.245) |
| 16:16.02 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:16.14 | flujan | guys I am in a hurry for help |
| 16:16.25 | flujan | i had a 406P which have problems to dial. .. |
| 16:16.39 | flujan | no I bought a 407P and I am having the same issues... |
| 16:16.44 | phearless | FREE BEER FOR EVERYBODY !!!!!!!!!!!!! |
| 16:16.45 | phearless | FREE BEER FOR EVERYBODY !!!!!!!!!!!!! |
| 16:17.19 | pyrom | one for me, one for me & one for me! |
| 16:19.57 | flujan | http://pastebin.ca/160682, please guys... take a look |
| 16:20.01 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 16:20.58 | phearless | don't put a , just near your URL |
| 16:21.19 | flujan | sorry phearless |
| 16:21.21 | flujan | http://pastebin.ca/160682 |
| 16:21.43 | flujan | I dunno what is happening... I just receive hangup requests... |
| 16:23.32 | wunderkin | probably need a pri debug, but you arent specifying a pridialplan or prilocaldialplan.. but i dont know what the problem is without a debug |
| 16:24.02 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 16:24.06 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 16:24.17 | *** part/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 16:24.53 | Ciber311 | [TK]D-Fender: you alive???? |
| 16:24.54 | flujan | wunderkin, I tried the same call using a sip softphone and have success... |
| 16:25.05 | flujan | It could be a IAX2 related issue? |
| 16:25.26 | [TK]D-Fender | Ciber311 : Yes, what is it? |
| 16:26.01 | Ciber311 | what provider do you recommend for voip inbound/outbound? |
| 16:26.51 | [TK]D-Fender | Ciber311 : What do you need? |
| 16:27.09 | Ciber311 | just a couple of channels |
| 16:27.12 | file | if I hear the word "cheap" in the response I'm going to run in circles |
| 16:27.19 | file | semi-circular circles a that |
| 16:27.26 | Ciber311 | looking at junction networks right now |
| 16:28.02 | Ciber311 | file: are you talking to me? :P |
| 16:31.02 | flujan | guys, any idea about this error? Just My IAX2 softphone is not working... http://pastebin.ca/160693 |
| 16:31.04 | wunderkin | flujan, i dont think so from what you posted |
| 16:31.14 | file | Ciber311: yes |
| 16:31.15 | Qwell | file: not oblong circles? |
| 16:31.24 | file | Qwell: no! |
| 16:31.24 | flujan | wunderkin, I showed a example |
| 16:31.36 | Qwell | semi-oblong? |
| 16:31.59 | wunderkin | flujan, you still arent showing the pri debug span 1 |
| 16:32.18 | Ciber311 | hehe file, i just need it to be reliable, yet not cost more than a regular pots line from verizon :P |
| 16:32.30 | Ciber311 | otherwise, what's the point? ;) |
| 16:32.39 | file | well, it won't be as reliable as pots |
| 16:32.52 | Ciber311 | well i know that |
| 16:34.06 | flujan | wunderkin, http://pastebin.ca/160696 |
| 16:34.08 | flujan | :) |
| 16:34.30 | Ciber311 | who are you using right now, file? |
| 16:34.54 | file | nobody, I have a cellphone and I'm a Digium employee so I use our PBX when I need to |
| 16:35.30 | *** join/#asterisk ellisdee (i=ellisdee@cpe-70-116-118-236.houston.res.rr.com) |
| 16:35.34 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 16:35.43 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:36.05 | Ciber311 | i see |
| 16:36.13 | manopulus | hello, i want to set accontcode, for further authentication. can i just use setvar(account=xxx)? |
| 16:37.03 | wunderkin | flujan, i think it is getting garbage from the caller id on the iax phone and setting your caller id to 9s which it does not like.. try removing your caller id from the iax phone |
| 16:38.21 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 16:39.27 | wunderkin | manopulus, i think it is Set(CDR(account)=blah) |
| 16:39.31 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:40.18 | wunderkin | no it is accountcode |
| 16:40.25 | wunderkin | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr |
| 16:44.00 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
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| 16:49.44 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 16:51.48 | flujan | wunderkin, thanks and congratulations..... |
| 16:51.59 | flujan | wunderkin, You just killed the problem... thanks so much!!! :D |
| 16:52.26 | wunderkin | yey |
| 16:59.45 | Dr-Linux|work | is is there anythat that we can provide an option to a user that he/she can check his/her mailbox space? |
| 16:59.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:59.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:01.09 | eKo1 | You mean voicemail ? |
| 17:02.05 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.139) |
| 17:03.56 | [TK]D-Fender | Dr-Linux|work : ....huh? |
| 17:05.50 | Dr-Linux|work | eKo1, yes |
| 17:06.16 | *** join/#asterisk beuster (i=beu@freenode/developer/gentoo.developer.beu) |
| 17:06.22 | Dr-Linux|work | [TK]D-Fender, i mean voicemail box size |
| 17:07.08 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:07.11 | eKo1 | By space, do you mean amount a limit on the amount of messages or the size in MB? |
| 17:07.37 | eKo1 | Well, whatever the answer is, you're going to have to hack app_voicemail* |
| 17:07.59 | Dr-Linux|work | eKo1, no other way? |
| 17:08.03 | [TK]D-Fender | Dr-Linux|work : Describe what you want * to do exactly. What is your measurement of "size"? |
| 17:08.11 | Dr-Linux|work | eKo1, like agi? |
| 17:08.31 | [TK]D-Fender | Dr-Linux|work : leave eKo1's comment alone for a minute, you weren't clear enough about what you want. |
| 17:08.53 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp186-119.adsl.forthnet.gr) |
| 17:08.54 | Dr-Linux|work | [TK]D-Fender, actually we are going to setup a separate server which will have ONLY voicemail, nothing else |
| 17:09.15 | Dr-Linux|work | this server will have more than 5k VM users |
| 17:09.40 | Dr-Linux|work | so we will provide them different packages, also different size |
| 17:09.58 | [TK]D-Fender | Dr-Linux|work : again what MEASUREMENT of size? Mewssages? bytes? time? |
| 17:10.32 | Dr-Linux|work | so we want to give them an option that they can check their Voicemailbox size in MB |
| 17:11.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:11.36 | [TK]D-Fender | Dr-Linux|work : If you want to tell the the size you'll have to do some system calls at the very least and then read off the size back. Probably best done in AGI |
| 17:11.40 | Dr-Linux|work | time, mmm i don't know about time, but MB and bytes are concerend |
| 17:12.55 | [TK]D-Fender | Dr-Linux|work : I'd probably expect this done in a perl AGI that would recurse the VM box's root folder for the recording failes and add up the size |
| 17:14.40 | Dr-Linux|work | [TK]D-Fender, actually our setup will be kinda different, the IVR options are on ServerA and after hitting voicemain checking option will forward the user to ServerB (Which is ONLY voicemail server) |
| 17:15.37 | Dr-Linux|work | [TK]D-Fender, there are other bunch of feature already running in AGI (C) for our cardholders |
| 17:15.47 | [TK]D-Fender | Dr-Linux|work : Doesn't matter where it is, the request can be sent along betweent he servers. |
| 17:16.32 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 17:16.39 | blitzrage | anyone played around with voicemail in a clustered environment? |
| 17:16.46 | blitzrage | (asterisk clusters) |
| 17:16.49 | Dr-Linux|work | [TK]D-Fender, yes these things are cleared, but we thinking that how to provide them voicemailbox size option |
| 17:17.33 | Dr-Linux|work | [TK]D-Fender, actually backend DB (API) also involved, and those stuff are already in production |
| 17:17.34 | blitzrage | thinking of using an NFS partition so all asterisk servers access the same virtual partition for VM |
| 17:17.48 | blitzrage | either that, or I move it to a DB, but not sure if MWI is working on that yet |
| 17:17.51 | Dr-Linux|work | all we are going to ad a new feature VoiceMail System |
| 17:18.49 | Ryushin | I guess pastebin is just really overloaded. Could someone take a look at my zapta.conf file and see if there is a reason asterisk isn't pulling up my PRI? http://pastebin.com/783532 |
| 17:18.52 | *** join/#asterisk Avalone (n=Avalone_@dial-094.vl-cen-as1.avtlg.ru) |
| 17:19.00 | Dr-Linux|work | blitzrage, walking around i saw on the web the same question you asked, but i didn't see the answer |
| 17:19.05 | Strom_C | Ryushin: use pastebin.ca |
| 17:19.38 | Dr-Linux|work | Ryushin, not opening for me |
| 17:20.52 | Ryushin | http://pastebin.ca/160721 |
| 17:20.55 | Strom_C | Ryushin: pastebin your zaptel.conf as well |
| 17:21.01 | blitzrage | Dr-Linux|work: probably just need to setup both and find the weaknesses |
| 17:21.39 | Strom_C | Ryushin: really, you should be putting your T1 card higher in the sort order so that the T1 is channels 1-24 and the four-port card is channels 25-28 |
| 17:22.14 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
| 17:22.15 | Ryushin | I've updated it. |
| 17:22.21 | Dr-Linux|work | Ryushin, where is context? |
| 17:22.52 | Strom_C | Ryushin: what hardware do you have |
| 17:22.58 | *** part/#asterisk Avalone (n=Avalone_@dial-094.vl-cen-as1.avtlg.ru) |
| 17:23.06 | Dr-Linux|work | blitzrage, what are you trying to get using voicemail? |
| 17:23.17 | blitzrage | Dr-Linux|work: ? |
| 17:23.34 | *** join/#asterisk budairc (n=chatzill@urano.mhnet.com.br) |
| 17:23.35 | Ryushin | Dr-Linux|work: It's not there. |
| 17:23.39 | Ryushin | Probably the problem. |
| 17:23.56 | Ryushin | Sangoma A104D and the A200 4 port FXS card. |
| 17:24.52 | Dr-Linux|work | i see |
| 17:24.59 | Dr-Linux|work | Ryushin, what problem you are facing? |
| 17:25.17 | Ryushin | The 4 analog channels come up fine, the PRI channels aren't coming up. |
| 17:25.34 | Dr-Linux|work | ooo |
| 17:25.48 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 17:26.03 | Ryushin | The context is there for both cards. |
| 17:27.05 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 17:27.28 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 17:27.57 | sip_me | Hi, Can I have the /var/log/asterisk/messages or other log file log the SIP Debug output? |
| 17:28.56 | Dr-Linux|work | yes |
| 17:28.58 | Dr-Linux|work | hhm.. |
| 17:29.15 | Dr-Linux|work | Ryushin, do you want to see my setting for my T1 card? |
| 17:29.20 | Strom_C | Ryushin: do you have the spans set correctly? |
| 17:29.33 | Strom_C | Ryushin: what framing, line coding, and switch type is your PRI using |
| 17:29.34 | Dr-Linux|work | Ryushin, what about your zaptel.conf? |
| 17:30.14 | Ryushin | I've posted both there. |
| 17:30.20 | Ryushin | I've added the messages as well. |
| 17:30.36 | Dr-Linux|work | Ryushin, i can't see your zaptel.conf :S |
| 17:30.58 | Ryushin | Dr-Linux|work: I would very much like to see it. It's a US PRI, with NI2. I have channels 1-9 with 24 being the d-channel. |
| 17:31.11 | Dr-Linux|work | Ryushin, do >> cat /etc/zaptel.conf and show |
| 17:31.25 | Strom_C | Ryushin: what framing / line coding did your telco tell you to use? |
| 17:31.39 | Ryushin | b8zs esf. |
| 17:31.42 | Strom_C | ok |
| 17:31.53 | *** part/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 17:31.56 | Strom_C | except you got those backwards ;) |
| 17:32.06 | Dr-Linux|work | Ryushin, span=1,1,0,esf,b8zs ? |
| 17:32.08 | Ryushin | The zaptel.conf starts on line 32. |
| 17:32.19 | Dr-Linux|work | Ryushin, i'm also using US PRI |
| 17:32.54 | Dr-Linux|work | huh? |
| 17:33.07 | Dr-Linux|work | where is line 32 and zaptel? >> http://pastebin.ca/160721 |
| 17:33.14 | Ryushin | http://pastebin.ca/160726 |
| 17:33.43 | Ryushin | Does it give me a new number each time I update it? |
| 17:33.50 | Strom_C | yes |
| 17:33.56 | Ryushin | Oh, then that is the problem. |
| 17:34.23 | *** part/#asterisk TrickyPhillips (n=TrickyPh@adsl-68-125-24-158.dsl.irvnca.pacbell.net) |
| 17:34.27 | Ryushin | Dr-Linux|work: Can I see your config? |
| 17:34.29 | Strom_C | Ryushin: so wait, if you have a quad-span card, why are the analog ports on channels 1-4? |
| 17:34.51 | Strom_C | the quad-span card should be channels 1-96, and analog should be 97-100 |
| 17:34.53 | Dr-Linux|work | # |
| 17:34.54 | Dr-Linux|work | bchan = 25-34 |
| 17:34.54 | Dr-Linux|work | # |
| 17:34.54 | Dr-Linux|work | dchan = 48 |
| 17:34.57 | Dr-Linux|work | what's this? |
| 17:34.59 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 17:35.06 | sip_me | Hi, ca anyone help in turning on logging of sip debug? |
| 17:35.17 | blitzrage | sip_me: sip debug |
| 17:35.18 | Dr-Linux|work | is it only 9 channel PRI? |
| 17:35.31 | Ryushin | The A200 uses 24 channels. So I start the PRI at 25. Yes, only a 9 channel PRI. |
| 17:35.54 | Strom_C | ...the A200 uses 24 channels on a four-port card? |
| 17:35.57 | Strom_C | that's really dumb |
| 17:36.18 | Ryushin | Well, The cards can be daisy chained together up to a maximum of 24. |
| 17:36.28 | Ryushin | Without using any more PCI slots. |
| 17:36.45 | Dr-Linux|work | odd setup |
| 17:36.46 | Strom_C | yeah, but taking up all the available space anyway |
| 17:36.59 | Ryushin | True. |
| 17:37.03 | [TK]D-Fender | Strom_C : no, the A200 is built on the same frame as their T1 cards and are treated effectively like a CAS T1. As you expand the card past 4 towards 24 ports you'll see the logic in not having all your numbering pushed around. |
| 17:37.03 | Dr-Linux|work | Ryushin, what you get when you do >> /sbin/ztcfg -vvv |
| 17:38.02 | Strom_C | [TK]D-Fender: im shuddering at the wiring disaster that would turn into |
| 17:38.28 | Strom_C | they invented 25-pair connectors for a reason :) |
| 17:38.32 | Ryushin | Line 54: http://pastebin.ca/160734 |
| 17:38.49 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 17:39.28 | rowter | anyone knows how could I change on unicall the way it sends the call to dtmf instead of mfc? |
| 17:41.19 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 17:42.05 | Ryushin | [TK]D-Fender: Did you say a while back that you've had good success with the ip430's? I've been having nothing but headaches with them. When I try to have them update their firmware, they go into a reboot loop with dumping the boot log with this error: Error application is not present |
| 17:42.41 | Ryushin | Any thoughts on my config? |
| 17:44.36 | manopulus | a question. asterisk-cmd-dia, G(context^exten^pri): If the call is answered, transfer both parties to the specified context.... - any example for it? |
| 17:44.45 | [TK]D-Fender | Ryushin : What release ar your trying to install on them? |
| 17:45.21 | Ryushin | 2.3.1, 1.6.7 |
| 17:46.28 | [TK]D-Fender | 2.3.1? |
| 17:46.38 | Ryushin | If I erase the filesystem and have it reload, it doesn't. So far I've killed three phones with doing that. Not going to make that mistake again. If I have the other ip430 phones boot, they use the original firmware and bootroom. |
| 17:46.41 | Ryushin | Sorry, 3.2.1 |
| 17:47.27 | [TK]D-Fender | Ryushin : Not sure what you did, perhaps bad authority on the firmware.... double check your SIP app and the version file are there and proper |
| 17:48.07 | Ryushin | I downloaded the firmware and bootrom from: http://www.freedomphones.net/polycom/files/ |
| 17:49.07 | Ryushin | The single ip601 I had updated perfectly from the same files. |
| 17:49.14 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 17:49.56 | Ryushin | Do you know the key config to reset the phone to factory defaults? The 468* doesn't work on the ip430's. |
| 17:54.44 | [TK]D-Fender | <PROTECTED> |
| 17:57.08 | [TK]D-Fender | Ryushin : 1,3,5& 7 for factory reset |
| 17:58.48 | *** join/#asterisk Tim__P (n=nospam@80.168.59.43) |
| 18:00.48 | Tim__P | Hi all. Trying to get N-way calling working with 1.2.9.1 under Gentoo - I keep seeing references to SVN-trunk, but can't find anywhere which actually explains what that is. Is it a module? How can I make Gentoo include this during an emerge? Anyone? thanks.. |
| 18:01.37 | Strom_C | Tim__P: SVN-trunk is the development branch of asterisk |
| 18:01.43 | Strom_C | Tim__P: www.asterisk.org |
| 18:02.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:03.25 | Tim__P | Strom_C: i see. So the modules required for n-way calls (ChannelRedirect etc) are still in development stage, and therefor I need to go a bit more bleeding edge to get there... that about right? |
| 18:03.38 | Strom_C | exactly |
| 18:04.34 | Tim__P | Strom_C: gotcha. is there any other method for performing n-way calls? |
| 18:04.34 | Strom_C | how many ways do you want to call, exactly? |
| 18:04.40 | wunderkin | 42! |
| 18:04.47 | Tim__P | Strom_C: 3 or 4..? |
| 18:04.52 | Strom_C | well three-way is easy |
| 18:05.17 | Strom_C | sip phones do it natively with the "conference" button, and analog phones do it by flashing |
| 18:05.41 | Strom_C | greater than three-way you could always stick everyone in a meetme conference |
| 18:06.06 | Tim__P | yeah, meetme is my current method, just a bit of a fiddle. anyway, thanks for the info! |
| 18:07.58 | RoyK | there's a guy working on a 3g video phone gateway for asterisk :) |
| 18:11.53 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 18:12.20 | *** join/#asterisk d4rkstar (n=d4rkstar@ip-46-85.sn2.eutelia.it) |
| 18:12.23 | d4rkstar | hi all |
| 18:12.32 | jart | has anyone ever had a really nasty bug where an application segfaults (e.g. asterisk) and then a file system becomes read only? |
| 18:12.41 | jart | and a ton of files end up in lost+found |
| 18:12.44 | d4rkstar | a question about asterisk |
| 18:13.01 | EzWayz | hi , why there is ticking noise (1/10 second length) every 65 seconds exactly, while speaking with someone, how to solve this problem , thanks a lot for you help, tell me where to read in case of it ... |
| 18:13.02 | Strom_C | d4rkstar: asterisk!? never! :) |
| 18:13.18 | Strom_C | EzWayz: what kind of connection? |
| 18:13.26 | d4rkstar | Strom_C ? |
| 18:13.31 | *** join/#asterisk volp (n=volp@200.44.157.60) |
| 18:13.36 | Strom_C | d4rkstar: it was a joke |
| 18:13.40 | d4rkstar | :) |
| 18:13.41 | Strom_C | d4rkstar: ask your question |
| 18:13.58 | d4rkstar | is there a way to simply build some ivr on asterisk (like on bayonne, i mean)? |
| 18:14.03 | *** part/#asterisk asterisk_bounty (n=pcm@72.146.14.209) |
| 18:14.14 | Strom_C | d4rkstar: yes, IVRs are easy |
| 18:14.36 | d4rkstar | Strom_C, can you give me a link from where i can start? |
| 18:14.42 | Strom_C | ~book |
| 18:14.44 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:14.49 | d4rkstar | i premit I've dialogic hardware on linux box |
| 18:14.54 | EzWayz | zap/chan1 1 fxo clone card pstn --> asterisk @ extension 600 local IAX phone ATA X100P , |
| 18:14.58 | *** part/#asterisk volp (n=volp@200.44.157.60) |
| 18:15.24 | EzWayz | this is not a random tic, always every 60-65 seconde |
| 18:15.31 | [TK]D-Fender | d4rkstar : * does not support Dialogic yet, though some amout of support is coming in 1.4 apprently (release date is uncertain) |
| 18:15.44 | Strom_C | EzWayz: hmmmm |
| 18:16.23 | Ryushin | [TK]D-Fender: I'll have to add the reset to the wiki. |
| 18:16.23 | Strom_C | EzWayz: what kind of PC? |
| 18:16.51 | EzWayz | P3 Dell OptiPlex 256 RAM |
| 18:17.05 | Ryushin | Is the bootrom where you can set the ip address and other such information. If that is the case, then yes. |
| 18:17.08 | Strom_C | is the iax phone local to the machine? |
| 18:17.14 | EzWayz | yep |
| 18:17.17 | Strom_C | hmmm |
| 18:17.44 | d4rkstar | so [TK]D-Fender i can't use asterisk with dialogic hardware |
| 18:17.56 | InHisName | I try to use VMAuthenticate(1021@context) and I am NOT prompted for pwd but account no anyway. More syntax issues ? |
| 18:18.09 | [TK]D-Fender | Ryushin : The bootrom IS where you set the IP parameters... |
| 18:18.17 | [TK]D-Fender | d4rkstar : Not yet |
| 18:18.28 | Ryushin | Okay, then I can get into it. |
| 18:18.46 | [TK]D-Fender | d4rkstar : I suggest you go buy a known compatible card |
| 18:18.58 | Ryushin | I've just reset it using the 1,3,5,7. Well see what happens in an hour. The ip430's take an hour to boot. |
| 18:19.26 | [TK]D-Fender | Ryushin : That makes no sense at all... you trying to provision them over the internet? |
| 18:19.35 | [TK]D-Fender | Ryushin : on some sort of dialup? |
| 18:19.36 | d4rkstar | [TK]D-Fender, what about a 12 or 16 analog channel? |
| 18:20.12 | [TK]D-Fender | d4rkstar : Use a T1/channel bank. There are analog cars for that density thouugh. The Sangoma A200 scales to 24 ports, as does the Digium TDM2400 |
| 18:20.16 | Ryushin | 100 mb internet. I know it makes not since. That's why I'm not so happy with them. Lets see what happens after the reset. |
| 18:20.36 | [TK]D-Fender | Ryushin : Make sure of your provisioning point... try to move it local |
| 18:21.14 | Ryushin | It is. It's on the LAN. |
| 18:21.26 | Ryushin | I can see it pulling the files via ftp from the server. |
| 18:22.05 | wunderkin | and it takes a long time to download the file? |
| 18:23.21 | *** join/#asterisk adorah (n=Administ@84.94.115.133.cable.012.net.il) |
| 18:25.36 | InHisName | any experts on VMAuthenticate ? |
| 18:28.10 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.10) |
| 18:28.13 | [TK]D-Fender | InHisName : pastebin your diaplan segment useing it and voicemail.conf |
| 18:32.07 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:32.11 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 18:32.45 | *** join/#asterisk sulan (n=ksjoberg@1-1-4-23a.lio.sth.bostream.se) |
| 18:33.56 | sulan | Hi, I'm using RealTime and can't get GotoIfTime to work properly. I have GotoIfTime in my dialplan with the arguments "12:00-20:32|*|4|sep?transfertovirtel,${EXTEN},1" but I get "Cannot find extension '660' in context 'screen-external'" and "Priority 'transfertovirtel,660,1' must be a number > 0, or valid label" in Asterisk |
| 18:35.00 | brimstone | or Strom_C |
| 18:35.01 | file | oh, okay! |
| 18:35.08 | Strom_C | well then |
| 18:35.27 | brimstone | just making sure you guys are clear on that :D |
| 18:35.46 | Strom_C | I have discovered a potential candidate for Official Asterisk Song |
| 18:36.36 | Assid | Asterisk you make me hungry.. All time making a call.. |
| 18:37.08 | Assid | Call me a pizza with chee eeze .. and a drunken ball |
| 18:37.23 | Strom_C | no, that's not it |
| 18:37.36 | Assid | then? |
| 18:37.50 | Strom_C | Ric Ocasek - Connect Up to Me |
| 18:38.19 | *** join/#asterisk IronMan2000 (n=kent@65.124.236.24) |
| 18:39.40 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:39.52 | IronMan2000 | can someone help me to setup my extensions.conf to be able to do outbound 800 toll free calling? |
| 18:40.58 | sulan | It seems that the documentation is outdated; you type a priority only, not a context+extension+priority like Goto |
| 18:41.19 | sip_me | blitzrage, sory for the delay. When I do sip debug I see all the debug info - but it is not logged into the /var/log/messages |
| 18:43.01 | Strom_C | sulan: you can do a number of things |
| 18:43.04 | Strom_C | priority |
| 18:43.09 | Strom_C | extension, priority |
| 18:43.14 | Strom_C | or context, extension, priority |
| 18:43.48 | IronMan2000 | can anyone help me debug my extenstions.conf to be able to make outbound 800 calls? I can make outbound calls, but whenever I call an 800 number, it will not work. I am very new to asterisk, and cannot figure out how to do this. Maybe if someone had an example .conf I could take a look at would really help. |
| 18:44.22 | Strom_C | IronMan2000: i tried to help you yesterday, and your config is a mess |
| 18:44.52 | IronMan2000 | that's why I am asking again to see if anyone can help. |
| 18:44.55 | Strom_C | you should really read the book and write your own extensions.conf instead of using someone else's config |
| 18:45.26 | IronMan2000 | I understand. I just need some examples to start with. If I knew how to do this, I would not be here asking.. |
| 18:45.36 | sulan | Strom_C: for some reason, it doesn't work (see the error messages posted earlier) |
| 18:46.17 | Strom_C | sulan: pastebin your extensions.conf; use pastebin.ca |
| 18:47.48 | Strom_C | IronMan2000: read this: |
| 18:47.50 | Strom_C | ~book |
| 18:47.58 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:54.55 | bkw_ | "Asterisk: The Future of Telphony".. I use mine for a monitor stand. |
| 18:55.25 | bkw_ | so glad I didnt' pay for it |
| 18:56.36 | IronMan2000 | looked at the .pdf on Future of Telephony, but has very little regarding setting up toll free |
| 18:56.51 | Strom_C | IronMan2000: you set up toll free just like any other eleven digit call |
| 18:57.10 | IronMan2000 | still not finding a solution. Seems this would be simple enough, but it seems to not be. |
| 18:57.18 | Strom_C | just because the call is rated differently doesn't change the fact that it's still in the same numbering space |
| 18:57.33 | bkw_ | Strom_C, I gather he doesn't understand that either |
| 18:57.44 | Strom_C | IronMan2000: what channels do you have your PRI set up on? |
| 18:58.02 | IronMan2000 | 1-23 |
| 18:58.15 | Strom_C | ok, so you have that defined as a group in zapata.conf? |
| 18:58.22 | IronMan2000 | yes. |
| 18:58.25 | Strom_C | try this |
| 18:58.37 | Strom_C | just set up an extension where you have the following: |
| 18:59.07 | Strom_C | exten => 18002255288,1,Dial(ZAP/Gx/18002255288) |
| 18:59.13 | Strom_C | where x is your group number that you defined |
| 18:59.34 | IronMan2000 | ok, I will try that.. Thanks. |
| 18:59.46 | Strom_C | and tell me what happens |
| 19:01.03 | IronMan2000 | ok |
| 19:01.39 | *** join/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 19:02.29 | bkw_ | So when do you think Mark is going to admit that Asterisk is flawed and needs to be rewritten? |
| 19:03.37 | IronMan2000 | added to my .conf and get the old "cannot be complete as dialed from the telephone you are using" |
| 19:03.43 | *** part/#asterisk cgabriel (n=cgabriel@217-133-96-194.b2b.tiscali.it) |
| 19:03.55 | *** join/#asterisk dlynes_ (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 19:03.57 | Strom_C | IronMan2000: pastebin your console output from that call |
| 19:04.04 | Strom_C | use pastebin.ca |
| 19:05.59 | *** join/#asterisk saftsack (n=saftsack@p54A7E0D2.dip.t-dialin.net) |
| 19:06.40 | IronMan2000 | where can I find my console outpu |
| 19:07.04 | Strom_C | connect to the console by typing "asterisk -r" at the bash prompt |
| 19:07.21 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 19:07.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:07.41 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 19:08.24 | *** join/#asterisk knobo (n=Knut@c85-196-83-87.static.sdsl.no) |
| 19:08.52 | knobo | how does operator=yes work (in voicemail.conf) |
| 19:09.19 | Strom_C | user can press 0 to be sent to the "o" extension |
| 19:09.35 | knobo | ah, the o, not the 0 |
| 19:10.12 | IronMan2000 | hmm. When I SSH into my asterisk and do the asterisk -r at the bash I get "bash: asterisk: command not found" |
| 19:10.46 | Strom_C | IronMan2000: what user is asterisk running as? |
| 19:10.46 | *** join/#asterisk Boggi (n=boggi@luleavnpc047-dot1x.studentby.uit.no) |
| 19:11.26 | Boggi | I get "Rejected connect attempt from xx.xxx.xx.xxx, request 'xxxxxxxx@inbound' does not exist", where do solve that problem, and how? |
| 19:11.41 | IronMan2000 | root |
| 19:11.56 | Strom_C | IronMan2000: and are you root when trying to issue that command? |
| 19:12.12 | knobo | ah! and fooled by my own foolishness. |
| 19:12.52 | IronMan2000 | yes, I used the SU and entered my root password. |
| 19:12.56 | knobo | in my test i used Playback(invalid), and it told me that "o" was not a valid extension |
| 19:13.05 | knobo | haha! |
| 19:13.13 | Strom_C | hahaha |
| 19:13.35 | Strom_C | IronMan2000: which version of asterisk are you running |
| 19:14.23 | IronMan2000 | 1.2.10 |
| 19:14.37 | Strom_C | on what linux distribution |
| 19:15.23 | Qwell | You didn't "su -", did you? |
| 19:15.27 | Qwell | ~su |
| 19:15.29 | jbot | Password: |
| 19:15.36 | Qwell | nice |
| 19:15.45 | IronMan2000 | Red Hat Fedora |
| 19:16.21 | Strom_C | maybe try /usr/sbin/asterisk -r |
| 19:17.26 | joat | try "whereis asterisk" |
| 19:17.31 | IronMan2000 | Qwell, what are you asking? |
| 19:18.33 | IronMan2000 | Strom, I pasted it but I dont think it's what you need. |
| 19:18.33 | S^P | IronMan2000: use "su - " instead of just "su" while changing from normal user to "root". |
| 19:18.37 | IronMan2000 | http://pastebin.ca/160830 |
| 19:19.06 | Strom_C | IronMan2000: um ok |
| 19:19.21 | Strom_C | IronMan2000: now place the call that fails and then paste the console output from that |
| 19:19.49 | *** join/#asterisk Mike800 (n=mike800@68-171-44-18.vnnyca.adelphia.net) |
| 19:19.50 | IronMan2000 | ok |
| 19:20.27 | Strom_C | hey mike |
| 19:20.34 | Mike800 | hey |
| 19:20.35 | Mike800 | :-) |
| 19:20.53 | Mike800 | whats going on? |
| 19:21.04 | Strom_C | oh not much...lazing about |
| 19:21.23 | Mike800 | cool |
| 19:21.29 | Strom_C | i was expecting this weekend's project to last three days, but i got it done last night |
| 19:21.32 | Mike800 | so i part of that install yesterday |
| 19:21.41 | Mike800 | just waiting for their t1 connection to come in |
| 19:21.45 | Strom_C | cool |
| 19:21.55 | IronMan2000 | http://pastebin.ca/160834 |
| 19:21.59 | Mike800 | but now asterisk crashed :-) |
| 19:22.27 | Strom_C | IronMan2000: well, something's totally buggered up with your dialplan |
| 19:22.35 | Strom_C | you're not telling asterisk where to send the outbound call |
| 19:23.29 | RoyK | evening |
| 19:23.33 | IronMan2000 | Do you know what needs to be done to fix this |
| 19:23.35 | Strom_C | Mike800: heh, oops |
| 19:23.49 | Strom_C | IronMan2000: you need to build a sane dialplan |
| 19:24.16 | Mike800 | IronMan2000: whats the dial statement that matches to the number you dialed? |
| 19:24.42 | RoyK | IronMan2000: perhaps r the eff em |
| 19:24.44 | RoyK | :P |
| 19:24.50 | RoyK | ~book |
| 19:24.52 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:25.04 | RoyK | ~rtfb |
| 19:25.07 | jbot | rtfb is, like, Read the Fscking Binary, or the only way drivers get written for Apple hardware. or "No source for the buggy port driver? Aaargh! I *hate* proprietary operating systems. Time to RTFB." |
| 19:25.15 | IronMan2000 | exten => 18002255288,1,Dial(ZAP/g1/18002255288) |
| 19:25.16 | RoyK | rotfl |
| 19:25.28 | Strom_C | IronMan2000: well that one isnt matching it |
| 19:25.34 | RoyK | IronMan2000: you're calling SIP, not Zap |
| 19:26.10 | Strom_C | IronMan2000: like i said...you dont understand what you're doing. you really need to start from step 1, build a SMALL asterisk box, understand the concepts, and become comfortable with them before you embark on this project |
| 19:26.11 | Mike800 | exten => _1800NXXXXXX,1,Dial(Zap/g1/${EXTEN}) would match it |
| 19:26.27 | Mike800 | ya....what strom said! |
| 19:27.19 | Strom_C | IronMan2000: or, if you need it fixed like RIGHT NOW, hire a consultant |
| 19:27.27 | *** join/#asterisk luke-jr_ (n=luke-jr@CPE-24-31-245-218.kc.res.rr.com) |
| 19:27.39 | Strom_C | disclaimer: whatever follows is qwell's fault: |
| 19:27.41 | Strom_C | ~hafc |
| 19:27.43 | jbot | extra, extra, read all about it, hafc is hire a freaking consultant |
| 19:27.44 | luke-jr_ | is there a particular reason g723 can't be translated to ulaw? |
| 19:27.54 | Mike800 | lol |
| 19:28.00 | Strom_C | luke-jr_: because asterisk only supports g723 passthrough |
| 19:28.09 | Strom_C | g723 is a patented codec |
| 19:28.30 | Strom_C | the TC400P will support that codec when it comes out, though |
| 19:28.52 | Mike800 | anyone wanna help me fix my asstricks box? (spandsp problem :D) |
| 19:28.54 | IronMan2000 | thanks for trying to help guys. |
| 19:29.28 | Strom_C | i havent played with spandsp yet |
| 19:29.42 | RoyK | Strom_C: spandsp just works imho |
| 19:29.50 | Mike800 | http://pastebin.ca/160812 |
| 19:29.52 | RoyK | Mike800: oh. the app_txfax bug? |
| 19:30.02 | Mike800 | is there a bug i'm unaware of? |
| 19:30.08 | Ryushin | How do I get support through Polycom? |
| 19:30.53 | RoyK | Mike800: hehe. perhaps you're having spandsp 0.0.3 around? |
| 19:31.41 | Mike800 | i installed 0.0.2pre26 |
| 19:31.43 | mitcheloc | oof |
| 19:32.12 | Mike800 | RoyK: did you see the pastebin? |
| 19:32.29 | RoyK | Mike800: try removing all old spandsp libs - find /usr/lib /usr/local/lib -name "*spandsp*" -exec rm -f {} \; |
| 19:32.40 | RoyK | Mike800: try that first, please, and then reinstall spandsp |
| 19:32.49 | Mike800 | RoyK: yessir |
| 19:32.50 | RoyK | i don't know if it helps, but after that it's a lot easier to debug it |
| 19:33.20 | Mike800 | so should i just rm -rf all the files that come up with that find? |
| 19:33.52 | Mike800 | oops..nm |
| 19:33.52 | Mike800 | :-) |
| 19:34.13 | luke-jr_ | Strom_C: I thought that was g729? |
| 19:34.19 | RoyK | just -f. find lists all of those |
| 19:35.19 | Strom_C | luke-jr_: g729 is also a patented codec |
| 19:37.49 | Strom_C | what a bargain |
| 19:38.11 | *** join/#asterisk kwagga (n=corne@wbs-196-2-112-144.wbs.co.za) |
| 19:40.13 | kwagga | whois kwagga |
| 19:41.01 | RoyK | Strom_C: http://dictionary.reference.com/search?q=bargain&x=0&y=0 |
| 19:42.23 | S^P | is it possible we can ass custom field in cdr logs with every user? |
| 19:42.55 | Strom_C | well, full-on assing maybe not, but half-assing is a possibility |
| 19:43.02 | Mike800 | ahahahaha |
| 19:43.11 | RoyK | S^P: how do you 'ass' custom fields? |
| 19:43.15 | RoyK | are we talking engrish here? |
| 19:43.33 | Strom_C | maybe he wants to slip information into the crack |
| 19:43.49 | RoyK | asstricks is nice |
| 19:43.56 | S^P | lol s/ass/assign/ |
| 19:44.09 | Strom_C | S^P: that's what ACCOUNTCODE is for |
| 19:44.26 | RoyK | S^P: show function CDR |
| 19:44.31 | RoyK | accountcode or userfield |
| 19:44.31 | S^P | ok |
| 19:44.49 | RoyK | just stuff whatever you want into userfield as comma separated or something |
| 19:47.21 | *** join/#asterisk Piston (n=what@206M28.oasis.mediatti.net) |
| 19:47.59 | S^P | cool, thanks all |
| 19:49.11 | Mike800 | RoyK: im recompiling asterisk |
| 19:49.26 | RoyK | Mike800: wow! tell me more!!!! |
| 19:49.36 | Mike800 | i followed the steps on asteriskguru to install spandsp :-) |
| 19:52.56 | *** join/#asterisk new2voip (i=new2voip@user-24-214-152-5.knology.net) |
| 19:53.20 | *** join/#asterisk vlt (n=daniel@dslb-088-073-228-004.pools.arcor-ip.net) |
| 19:53.49 | *** join/#asterisk j0 (i=dan@CABLE-72-53-45-212.cia.com) |
| 19:54.45 | new2voip | Hi again. I used the call me feature from FWD website and * answered and took a msg. The msg was blank thou. Is this normal or should I have heard something? |
| 19:55.36 | RoyK | Mike800: any luck? |
| 19:55.39 | S^P | check microphone first :). |
| 19:56.10 | kwagga | Has anyone created a test server with an ISDN BRI running in NT mode |
| 19:57.32 | Mike800 | RoyK: enjoy your wine...removing all the libs worked...spandsp installed :-) |
| 19:58.25 | luke-jr_ | any shortcut to do disallow=passthruonly |
| 19:58.25 | luke-jr_ | ? |
| 19:58.25 | Strom_C | luke-jr_: what are you trying to do |
| 19:58.29 | luke-jr_ | or better yet, renegotiate codec when translation is impossible? |
| 19:58.39 | luke-jr_ | Strom_C: avoid can't-translate-these errors |
| 19:58.48 | Strom_C | luke-jr_: get the other side to not use g.723.1 |
| 19:58.59 | RoyK | luke-jr_: iirc you can't renigotiate codec |
| 19:59.13 | luke-jr_ | Strom_C: that's why I want to just disallow any of these problem codecs |
| 19:59.17 | luke-jr_ | RoyK: that sucks :/ |
| 19:59.19 | Strom_C | how abour |
| 19:59.20 | Strom_C | er, |
| 19:59.21 | Strom_C | about |
| 19:59.24 | Strom_C | disallow=all |
| 19:59.28 | Strom_C | allow=whatever |
| 19:59.30 | Strom_C | etc |
| 19:59.44 | luke-jr_ | then I need to manually select translatable codecs |
| 19:59.57 | RoyK | luke-jr_: exactly |
| 20:00.40 | luke-jr_ | maybe I'll make a patch to let me disable non-translatable codecs... |
| 20:00.42 | RoyK | luke-jr_: it shouldn't be too hard |
| 20:00.54 | luke-jr_ | RoyK: it's not the proper way to do it |
| 20:01.06 | luke-jr_ | it means I'd need to update the list when Asterisk upgrades |
| 20:01.09 | RoyK | then just rewrite the code :) |
| 20:01.16 | luke-jr_ | maybe I will =p |
| 20:01.32 | RoyK | SMOP |
| 20:01.35 | RoyK | ~smop |
| 20:01.36 | jbot | i heard smop is just a simple matter of programming or encrypted floor cleaning tool |
| 20:01.44 | Strom_C | hahahahahahahaha |
| 20:01.49 | RoyK | :) |
| 20:02.32 | eKo1 | unfortunately, programming is not a simple manner |
| 20:03.19 | RoyK | eKo1: depends what you're going to do. rewriting asterisk is a PITA, but apart from that, it can be quite like a stroll in the park |
| 20:06.03 | shap | All: anybody have experience with uniden 1686 handsets? |
| 20:06.26 | shap | Or opinions about other 5Ghz wireless options? |
| 20:06.27 | Strom_C | uniden makes voip phones now?? |
| 20:06.30 | Strom_C | sweet |
| 20:06.36 | Mike800 | they have for a while |
| 20:06.47 | Mike800 | but they're not that good |
| 20:07.30 | shap | Mike, strom: I have a user who wants to be able to walk and talk (no guarantees about gum). Either needs a radio handset or a wifi option (if those are any good yet) |
| 20:07.52 | Strom_C | shap: the wifi phones available pretty much all suck right now |
| 20:08.02 | Mike800 | my understanding about the uniden is that only the base unit connects to the voip provider. so if you wanted an extension assigned to each handset, it couldnt happen. |
| 20:08.11 | Strom_C | my solution is a good regular cordless plus an ATA |
| 20:08.16 | shap | strom: That's more or less what I figured about the wifi. |
| 20:08.32 | shap | Mike: correct on the base unit. He just wants to be able to pace. |
| 20:08.58 | Mike800 | oh |
| 20:08.59 | shap | strom: advice on the ATA? IAXy if asterisk, but if other? |
| 20:09.14 | Strom_C | shap: panasonic makes excellent cordless phones |
| 20:09.24 | Strom_C | for SIP ATAs, the SPA-2002 has never let me down |
| 20:10.05 | shap | strom: never again on panasonic. I'm just annoyed that Siemens exited US market (have a bunch of that at home), but my Uniden analogs have done pretty well. |
| 20:10.09 | shap | Thanks on SPA-2002 |
| 20:11.36 | shap | strom: PAPT-NA? |
| 20:11.58 | Strom_C | that'll work too |
| 20:12.44 | shap | s/PAPT/PAP2/ |
| 20:16.58 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 20:17.41 | *** part/#asterisk new2voip (i=new2voip@user-24-214-152-5.knology.net) |
| 20:22.22 | Ryushin | For those of you that have pri's working, do you see the channels listed in the CLI, when you run zap show channels? |
| 20:22.57 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:24.57 | _DAW | Ryushin: when I so zap show channels I see see channels |
| 20:25.02 | _DAW | er.. do |
| 20:26.07 | Ryushin | crud. Then my zapta.conf still doesn't work. |
| 20:26.20 | Ryushin | I just say my four analog lines. |
| 20:27.04 | _DAW | pastebin zaptel and zapata |
| 20:28.18 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:29.03 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:30.59 | *** join/#asterisk crshman (n=crshman@c-24-126-70-86.hsd1.ca.comcast.net) |
| 20:31.07 | Ryushin | _DAW: http://pastebin.ca/160912 |
| 20:31.29 | lesouvage | I'm exploring the callcenter options of Asterisk. If I logged in using the AgentLogin(<agent>) command what command do I use to log out. AgentLogout(<agent>) doesn't seem to exist. |
| 20:31.45 | Ryushin | I've tried changing the PRI to group = 2, but that doesn't seem to have an affect. |
| 20:32.33 | crshman | hello everyone, i'm having a problem with my callerid settings, i set the callerid field in the user context, and it doesn't get sent, when i try to set the callerid (SetCallerID(${CALLERID})) it says the variable is empty any ideas? |
| 20:34.29 | lesouvage | I seem to have find it, you simply end the connection :) |
| 20:34.43 | lesouvage | crshman: what card are you using? |
| 20:34.57 | crshman | no cards, i am purely voip |
| 20:34.59 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:35.42 | eKo1 | I wish I could go purely VoIP... |
| 20:36.49 | crshman | i can't seem to figure out why the variable is empty... =/ |
| 20:37.27 | InHisName | I try to use VMAuthenticate(1021@context) and I am NOT prompted for pwd but account no anyway. Is there something wrong with my syntax ? |
| 20:39.28 | knobo | asterisk is not trying to registrate to the sip provider, what did I forget? |
| 20:39.51 | lesouvage | chrhman: if you add exten => x,x,NoOp(${CALLERID}) somewhere in your context, what does the cli give as output? |
| 20:40.07 | crshman | nothing it's just blank "" |
| 20:40.14 | knobo | I have register=>400:400@192.168.1.4:5062 |
| 20:40.58 | knobo | sip show peers says |
| 20:40.59 | knobo | yate/400 127.0.0.1 5062 Unmonitored |
| 20:41.11 | eKo1 | ah, a yate user |
| 20:41.25 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:41.30 | delmar | wtf is a "yate" when its at home? |
| 20:42.14 | knobo | Yate (pop. 23,000), near Bristol, England |
| 20:42.14 | crshman | lesouvage, btw...CLI output: Executing NoOp("IAX2/3002-1", "") |
| 20:42.58 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:43.42 | Ryushin | Any idea on what is wrong with my zapata.conf file? |
| 20:44.33 | knobo | Ryushin: I know. looks like it's missing. :) |
| 20:44.43 | delmar | Ryushin, at a glance.... I have a question or two |
| 20:45.02 | delmar | Ryushin, 4 FXO's ? |
| 20:45.33 | knobo | ah pastebin.. |
| 20:45.42 | delmar | Ryushin, or are they FXS? |
| 20:45.49 | RoyK | ~lart knobo for old time's sake |
| 20:46.04 | Ryushin | delmar: yea, the 4 FXO's were to provide analog to our office faxes, which would then go out the pri. |
| 20:46.18 | knobo | Linux Advanced Radio Terminal (LART) |
| 20:46.33 | delmar | Ryushin, not sure about Pri's.. never played with them... but... FXO ports use FXS signaling |
| 20:46.39 | lesouvage | crshman: this is working for me SetCallerID(0031${CALLERIDNUM:1}) |
| 20:47.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 20:47.11 | knobo | Røyk: Help me with this register thing. |
| 20:47.19 | Ryushin | I was going to plug in the faxes directly into my analog card. Do I want to provide fXO to the faxes? |
| 20:47.28 | delmar | Ryushin, sure |
| 20:47.30 | Ryushin | knobo: Did you see my pastibin? |
| 20:47.39 | delmar | Ryushin, fxoks = 1-4 |
| 20:47.40 | crshman | lesouvage: i don't fully understand...but let me try it....you are adding "0031" before the callerid number and removing the first digit? :1 |
| 20:47.43 | delmar | Ryushin, that looks wrong |
| 20:47.59 | delmar | Ryushin, your ztcfg output shows 01-04 as FXO |
| 20:48.16 | knobo | Ryushin: I'm actualy not an expert... |
| 20:48.17 | delmar | Ryushin, so fxoks = 1-4 is incorrect i think |
| 20:48.19 | lesouvage | crshman: it adds the Dutch international country code and cut of the first number. Forget about the 0031 and the :1 and the caller id will show up. |
| 20:48.30 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 20:48.51 | Ryushin | Okay, so it's fxo in zaptel, and fxs in zapata? |
| 20:49.15 | knobo | I'm not even able to make asterisk register to a provider... |
| 20:49.24 | crshman | lesouvage: "1,noop(${CALLERIDNUM})" returns: "Executing NoOp("IAX2/3002-6", "")" |
| 20:49.27 | knobo | but i guess that is asterisks fault |
| 20:49.44 | Ryushin | Well, asterisk isn't even seeing my pri right now |
| 20:50.09 | RoyK | knobo: hva skjer_ |
| 20:50.10 | RoyK | ? |
| 20:50.11 | lesouvage | crshman: is it an internal number or an inbound ip-telphony number that doesn't show up. |
| 20:50.50 | delmar | Ryushin, if its an FXO channel ... zapata/zaptel should refer to it as FXS |
| 20:51.00 | delmar | Ryushin, its confusing |
| 20:52.19 | crshman | lesouvage: it is an internal extention that is going to dial out to the world |
| 20:52.37 | delmar | Ryushin, I suggest you check the Wiki .. there is alot of information on how it all works and how to set it up |
| 20:53.14 | *** join/#asterisk h4mm3r` (n=h4mm3r@213-140-16-186.fastres.net) |
| 20:57.02 | *** join/#asterisk _007 (n=brian@bhlkwihed01-pool0-a31.bhlkwi.tds.net) |
| 20:57.34 | Ryushin | Okay, that's odd. If I switch the channel statements in the zapata.conf, the PRI now comes up, but the znalog channels don't. |
| 21:00.58 | lesouvage | chrman: the variabe you are lookingfor is ${EXTEN} |
| 21:01.35 | lesouvage | crsman=crshman |
| 21:03.37 | crshman | really? if i noop that variable it just gives me the number i dialed |
| 21:04.38 | lesouvage | crshman: sorry, I was to fast, this is the variable that holds the numbr your dialing. Let me walk through my own asterisk configs. There must be an example somewhere. |
| 21:12.30 | InHisName | any experts on VMAuthenticate ? |
| 21:13.30 | lesouvage | crshman: it's just a guess but if you try ${MACRO_EXTEN} does that give a result? It seem to be the variable that is given to any macro holing the extension number wheere the macro is called from. It's the same as ${ARG1} in any macro. |
| 21:13.41 | wunderkin | crshman, what are you trying to do |
| 21:15.01 | bkw_ | ah |
| 21:15.03 | bkw_ | fun fun |
| 21:22.26 | crshman | wunderkin: my outbound callerid is not being set for an extension |
| 21:23.02 | wunderkin | crshman, show me where you are setting it, it is pointless to do a SetCallerID(${CALLERIDNUM}) and that way is also deprecated now |
| 21:24.15 | knobo | ethereal says: no data catured |
| 21:24.23 | crshman | wunderkin: as i have read, i set callerid in the extension context, but if i call the callerid variable and noop it it returns nothing |
| 21:24.24 | knobo | eh.. captured |
| 21:25.42 | wunderkin | crshman, show the line in iax.conf where you set the caller id |
| 21:26.55 | crshman | wunderkin: "callerid="16264xxxxxx" <16264xxxxxx> |
| 21:27.05 | knobo | funny that a so simple thing as "register" is so difficult with asterisk :( |
| 21:27.06 | crshman | x's replaced with numbers of course |
| 21:28.51 | *** join/#asterisk harryv (n=harry@0x3e428c6b.adsl.cybercity.dk) |
| 21:30.16 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 21:34.14 | wunderkin | crshman, that looks ok... maybe that peer is not being matched for some reason and using the guest account? you should be able to do a Set(CALLERID(num)=6261231234) and work |
| 21:35.05 | *** join/#asterisk amer (n=ame@202.125.143.65) |
| 21:35.18 | amer | how do i create the realtime db? |
| 21:35.45 | crshman | i can do that, however caller id is not being set for any extension.....i see it as very cumbersome to set the caller id for every extension like that |
| 21:35.45 | amer | i have already compiled res_mysql |
| 21:36.28 | wunderkin | crshman, no, it is not necessary, i was just saying that i think the only reason that setting it in iax.conf was not working is because it was not matching that peer |
| 21:36.30 | brimstone | amer, look on voip-info.org |
| 21:36.58 | amer | i did, not very helpful |
| 21:37.00 | crshman | wunderkin: o ok sorry for the misunderstanding, how do i check to see if it is matching the peer or not? |
| 21:38.03 | *** join/#asterisk Sarum4n (n=some@saruman.demon.nl) |
| 21:38.24 | wunderkin | crshman, well it does say IAX2/3002 so i would think that it is matching 3002, that is where you have that callerid line set? not sure what else to say |
| 21:38.56 | crshman | yup, that's where it is |
| 21:45.54 | *** join/#asterisk andresmujica (n=andresmu@201.244.243.150) |
| 21:47.27 | knobo | strace -ff -e trace=network asterisk -f does not mention anything that can indicate that asterisk is trying to contact the sip server |
| 21:48.23 | *** join/#asterisk daysmen3 (n=primus@host86-140-208-167.range86-140.btcentralplus.com) |
| 21:52.40 | tzafrir_laptop | what is strace -ff ? |
| 21:53.36 | tzafrir_laptop | knobo, is there any conflicting report from tcpdump? |
| 21:54.06 | *** join/#asterisk tr2x (n=alvar@84-73-197-108.dclient.hispeed.ch) |
| 21:55.45 | knobo | tzafrir_laptop: tcpdump shows no traffic that has anything with sip to do |
| 21:57.36 | *** join/#asterisk daysmen3 (n=primus@host86-140-208-167.range86-140.btcentralplus.com) |
| 21:57.37 | tzafrir_laptop | knobo, have you put the proper register => line in the [general] section of sip.conf? |
| 21:57.38 | Ryushin | Is there some setting on the polycom phones ftp files to specify dial 9 to get out? |
| 21:57.45 | *** part/#asterisk harryv (n=harry@0x3e428c6b.adsl.cybercity.dk) |
| 21:58.46 | Strom_C | Ryushin: you specify pattern matches in the phone in sip.cfg but specify the actual call routing in asterisk |
| 21:59.37 | Ryushin | Strom_C: Okay. As soon as it hits 10 digits it goes to a busy signal right now. However, if there is a 1 in front of it, I can dial more. I'll have to go through that file in detail. |
| 21:59.41 | knobo | tzafrir_laptop: finaly... It was not under [general] I had it in the wrong plase |
| 21:59.46 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 21:59.54 | Ryushin | I have 10 digit local dialing here. |
| 22:00.05 | knobo | tzafrir_laptop: thank you so much |
| 22:00.14 | Strom_C | Ryushin: it should be on the first screen of sip.cfg |
| 22:00.51 | Strom_C | look for a line that has something like "1[2-9]xxxxxxxxx|0|[2-9]11" |
| 22:01.01 | knobo | and it works! |
| 22:01.12 | knobo | Finally I can go to sleep :) |
| 22:03.34 | Ryushin | I should change it to 2-8 instead? |
| 22:17.09 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:17.41 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 22:20.32 | Ryushin | I'm trying to dial out. I have a entry like this: exten => 9,1,Dial(Zap/g1/). How do I tell what is my correct Zap dial out my pri. I think it's group1 so that would be g1, right? |
| 22:22.01 | hotroot | I think you shoud tell * which number you'd like to dial |
| 22:22.53 | hotroot | you do this by appending the number behind g1/ |
| 22:24.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:25.31 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 22:27.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-1761.bb.online.no) |
| 22:28.51 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:30.38 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:31.27 | Ryushin | Should I see in the asterisk log when a outgoing pri call is initiated? The polycom phone just gives me a instant busy when I type in the number and then hit the line number. |
| 22:31.59 | Ryushin | I don't know if it's the phones fault, or my asterisk config. |
| 22:32.06 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.10) |
| 22:38.29 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 22:40.33 | *** join/#asterisk [Outcast] (n=outcast@222-154-72-242.jetstream.xtra.co.nz) |
| 22:41.07 | dlynes_ | Ryushin: you probably don't have a dialplan defining that extension in your context for the polycom |
| 22:41.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 22:41.28 | *** join/#asterisk budairc (n=chatzill@200.215.57.174) |
| 22:43.33 | Ryushin | Well, it should bypass the dialplan if I type the number in and then send it. |
| 22:43.48 | dlynes_ | why would it bypass it? |
| 22:44.18 | Ryushin | My dialplan looks like this: dialplan.digitmap="[2-9]11|0T|011xxx.T|9011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|9[2-9]xxxxxxxxx|9[0-1][2-9]xxxxxxxxx|[2-9]xxxT" |
| 22:44.48 | dlynes_ | Ryushin: i'm talking about your asterisk dialplan, not your polycom dialplan |
| 22:45.29 | Ryushin | Oh, well I have a entry like this: 9,1,Dial(Zap/g1/1) |
| 22:45.41 | Strom_C | you're just dialing "1" out the PRI |
| 22:45.45 | Strom_C | that's not going to go anywhere |
| 22:45.50 | Strom_C | you need to dial a full number |
| 22:46.37 | Ryushin | What if I get rid of the last 1. So it looks like (Zap/g1) |
| 22:47.04 | Strom_C | then you're not dialing /anything/ out the PRI |
| 22:47.14 | Ryushin | This would explain lots. |
| 22:47.24 | Strom_C | Ryushin: you understand how ISDN call setup works, right? |
| 22:47.47 | Ryushin | So I have to make it like (Zap/g1/XXXXXXXXXX) |
| 22:48.11 | Strom_C | something like that, yes...depending on what your telco is expecting you to send |
| 22:48.37 | Ryushin | Strom_C: Somewhat. The calls go out over the b channels with d being the signal channel. |
| 22:49.00 | Strom_C | yes, but you understand that you must send the complete number in the SETUP message, right? |
| 22:49.22 | Strom_C | it's not like POTS where the process of requesting a channel and signaling the number is a series of discrete steps |
| 22:49.31 | Ryushin | The astrisk handbook needs to be fixed then. It specifies it to be 9,1,Dial(Zap/g2/) |
| 22:49.49 | Strom_C | point me at the relevant part of the document |
| 22:50.15 | file | Strom_C: strike a pose! |
| 22:50.36 | Strom_C | shit, now it's bleeding |
| 22:50.43 | Ryushin | Page 36 |
| 22:50.51 | Ryushin | [directdial] |
| 22:50.59 | Strom_C | URL please |
| 22:51.20 | Ryushin | It's the asterisk handbook pdf. |
| 22:51.26 | Ryushin | I'll try and find the url |
| 22:52.05 | Strom_C | no need |
| 22:52.06 | RoyK | ~book |
| 22:52.08 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:52.08 | Strom_C | one moment |
| 22:52.14 | RoyK | that's better |
| 22:52.20 | RoyK | the handbook is old and battered |
| 22:52.59 | Strom_C | Ryushin: that example assumes an analog line |
| 22:53.10 | Ryushin | Oh, that's nice. |
| 22:53.15 | Strom_C | where you can pick up the trunk and get a real CO dial tone and do whatever you like with it |
| 22:53.33 | Strom_C | which is completely not how ISDN works :) |
| 22:53.45 | Ryushin | So much for this book then. |
| 22:54.04 | Strom_C | Ryushin: step one: understand the technology you're dealing with |
| 22:55.01 | *** part/#asterisk hotroot (n=michael@pD9E96D37.dip.t-dialin.net) |
| 22:55.07 | Strom_C | http://www.ralphb.net/ISDN/proto_l3.html |
| 22:55.47 | Ryushin | reading..... |
| 22:56.09 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 22:56.09 | *** join/#asterisk daysmen3 (n=primus@host86-140-208-167.range86-140.btcentralplus.com) |
| 22:58.49 | Ryushin | Okay I see that. Since the outgoing lines has to handle 911, international, longdistane, and local, what would a outgoing extension look like for me? |
| 22:58.57 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 22:59.08 | Strom_C | Ryushin: who is your telco |
| 22:59.39 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 23:00.07 | Ryushin | Qwest. |
| 23:00.16 | Strom_C | which city are you in |
| 23:00.21 | Ryushin | Denver |
| 23:00.39 | Strom_C | ok, so ten digit intra-NPA, eleven digit inter-NPA |
| 23:00.42 | Strom_C | something like this: |
| 23:00.55 | Ciber311 | so what business-grade voip providers do you guys recommend? |
| 23:01.17 | Strom_C | exten => _NXXNXXXXXX,1,Dial(ZAP/G1/${EXTEN}) ; 303/720 calls |
| 23:01.38 | Strom_C | exten => _1NXXNXXXXXX,1,Dial(ZAP/G1/${EXTEN}) ; outside 303/720 area |
| 23:01.55 | Strom_C | exten => _011.,1,Dial(ZAP/G1/${EXTEN}) ; international |
| 23:02.26 | Strom_C | exten => _[2-9]11,1,Dial(ZAP/G1/${EXTEN}) ; Services (DA, repair, business office, emergency) |
| 23:02.48 | [Outcast] | has anyone seen anthm? |
| 23:03.40 | Qwell | ~seen anthm |
| 23:03.56 | jbot | anthm <n=anthmct@CPE-69-76-83-52.wi.res.rr.com> was last seen on IRC in channel #asterisk, 13d 7h 10m 42s ago, saying: 'nothing is proper in chan_sip it's ass from head to toe'. |
| 23:04.29 | Ryushin | So the ignorepat => 9, remove the 9, and I just try and process the extension directly then. |
| 23:04.30 | [Outcast] | that is not right i just talk to him 2 days ago on here |
| 23:04.38 | Ryushin | maybe that's been my problem. |
| 23:04.47 | Strom_C | Ryushin: you're to modify those to match your internal dialplan |
| 23:04.50 | Ryushin | Also, not understand isdn doesn't help. |
| 23:05.05 | file | Ryushin: ignorepat is used for analog phones connected to Asterisk |
| 23:05.17 | Strom_C | Ryushin: but the end result should be that you send exactly the same things out the PRI as in those examples |
| 23:05.48 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 23:05.54 | Ryushin | Okay, let me try it for awhile and go from there. Thank you very, very much for you help. |
| 23:06.42 | Ryushin | Success. Thank you so much!!!! |
| 23:08.19 | Strom_C | this is why I usually charge $150 per hour |
| 23:08.43 | file | $150? $49! |
| 23:09.00 | Strom_C | $800km |
| 23:09.42 | file | excellent |
| 23:11.14 | RoyK | http://karlsbakk.net/fun/ugly-hint.txt |
| 23:11.15 | blitzrage | file: no.... you don't |
| 23:11.50 | Ryushin | Well, Strom, even you had to learn somewhere for the first time. I plan on learning asterisk as much as possible. And to do consulting is my end goal as well for my current business. I already do Security, IDS, Networks, Unix, Linux, FreeBSD, etc. |
| 23:12.26 | Strom_C | Ryushin: awesome :D |
| 23:12.33 | [Outcast] | do you need to have a disclaimer on file to contribute addons? |
| 23:14.19 | eKo1 | Ryushin: what does "I already do" mean exactly? |
| 23:15.52 | Ryushin | I already run my own business. I do security auditing for unix/networks/windows, Solaris, Linux, FreeBSD support and design, etc. |
| 23:16.48 | Ryushin | Asterisk was one of the things that has always been on my back burner to learn. I have a customer that was moving, and they wanted a new phone system. So this was the perfect chance to get in and swim or drown with it. |
| 23:17.24 | Ryushin | Strom_C, actually just gave me the breath of fresh air that I needed, and I really appreciate it. |
| 23:18.38 | Strom_C | :) |
| 23:19.04 | Strom_C | i came into asterisk from a telephony background, so usually the problems I run into are all linux-related :) |
| 23:19.54 | *** part/#asterisk andresmujica (n=andresmu@201.244.243.150) |
| 23:20.47 | shmaltz | does the SPA42 have backlight lcd? |
| 23:21.12 | shmaltz | SPA942* |
| 23:21.12 | Ryushin | Strom_C: Well, I've been doing Linux since '93, if you need help with any of that, I'll be glad to help out. |
| 23:21.26 | Strom_C | sweet |
| 23:22.09 | JT | bit from column A, bit from column B here, does that make me a fence sitter? |
| 23:22.15 | Ryushin | Well, couple of things. Each channel on the PRI has a phone number associated with it correct? |
| 23:22.24 | Strom_C | no |
| 23:22.46 | Strom_C | the circuit itself has a DID block assigned to it, but calls can come in on any channel for any number |
| 23:23.02 | Strom_C | the number that the calling party dialed is transmitted to the PBX in the SETUP message |
| 23:23.07 | Ryushin | Oh, okay. |
| 23:23.26 | Ryushin | I have a 100 DID's, so I'll have to look at doing that next. |
| 23:23.30 | Strom_C | so you can, for example, have a single PRI with 100 DID numbers assigned to it |
| 23:23.33 | Strom_C | haha |
| 23:23.38 | Strom_C | i picked 100 arbitrarily |
| 23:23.48 | Ryushin | Oh, that is funny. |
| 23:23.59 | Vco | shmaltz yea, it has a backlight, and yes, it will flicker slightly when there is ringing when yo uhave it on speakerphone |
| 23:24.12 | Vco | kinda lame |
| 23:24.13 | shmaltz | Vco, and the SPA941? |
| 23:24.17 | Strom_C | so in the context your calls come into, for example, you'll have a separate extension for each DID |
| 23:24.23 | Strom_C | so you'll have something like: |
| 23:24.30 | Vco | i dont' believe there was a backlight.. |
| 23:24.35 | Vco | only seen the 942 |
| 23:24.38 | Ryushin | Now I have to figure out how to handle incoming calls. I saw that 7666 came in, so I made an extension for that to announce the date and time, and that worked fine. |
| 23:24.46 | Strom_C | exten => 3033333333,1,Goto(taxi,s,1) |
| 23:24.56 | Strom_C | exten => 3037777777,1,Goto(taxi2,s,1) |
| 23:25.11 | Strom_C | depending on how Qwest is sending DNIS, of course |
| 23:25.21 | Strom_C | (this example assumes they're sending the full ten digits) |
| 23:25.33 | Ryushin | I think it's just the last 4. |
| 23:25.40 | Strom_C | in a ten digit environment? yech |
| 23:25.55 | Strom_C | are your DIDs contiguous? |
| 23:26.35 | Ryushin | Yea, 2710-2809 |
| 23:26.49 | Strom_C | that's...an odd block to assign :) |
| 23:27.01 | Ryushin | Interesting that I can't call my main number. |
| 23:27.11 | Strom_C | what happens when you call your main number? |
| 23:27.15 | Ryushin | Yea, well, CBeyond hasn't always been the quickest on the draw. |
| 23:27.19 | Ryushin | It goes busy. |
| 23:27.23 | Strom_C | who is CBeyond? |
| 23:30.01 | Ryushin | Well, they are a voice and data provider. What they do is provide a data T1, and then hand of a partial PRI to me (in my case), and a ethernet drop for data. For every voice call, it gets translated in their router to voip and sent to their center to be processed out to a PSTN. When there are no call in place, I have full T1 speeds for data, and every call takes 80K off the top. |
| 23:30.21 | Strom_C | Ryushin: I thought you said you have a PRI from Qwest |
| 23:31.05 | Ryushin | Well, no one knows who CBeyond is. It is a Qwest T1 Circuit, but I guess in reality, it's a CBeyond PRI. |
| 23:31.23 | JT | slight technicality |
| 23:31.26 | Ryushin | Coming off their propritory cisco 2600 router. |
| 23:31.28 | JT | worth mentioning |
| 23:31.43 | Strom_C | so...wait. you've got only a partial PRI? |
| 23:31.47 | Ryushin | JT: True. |
| 23:31.52 | Ryushin | Yea. |
| 23:31.55 | Strom_C | i thought you said you had b-channels on channels 1-23 |
| 23:31.58 | Strom_C | that's a full PRI |
| 23:32.00 | Ryushin | 9 channels. |
| 23:32.12 | Strom_C | ok; i might be confusing you with someone else then :) |
| 23:32.14 | Ryushin | 1-9 with 24 being the dchannel. |
| 23:32.34 | Ryushin | But it works right now. My name won't be mud tomorrow. |
| 23:32.40 | Strom_C | cool |
| 23:33.05 | Ryushin | I'll have to fix the fax lines later. After I talk with Sangoma. |
| 23:33.20 | Strom_C | so it's asterisk <-PRI-> Cisco <-VoIP-> {their network} <-> {PSTN} ? |
| 23:33.44 | Strom_C | Ryushin: what kind of module do you have in your analog card? |
| 23:34.24 | Ryushin | FXS |
| 23:34.39 | Strom_C | hmm, ok |
| 23:35.02 | Strom_C | my only experiences with configuring sangoma hardware have all ended with me stating "boy, what a pain in the ass that was" |
| 23:36.21 | *** join/#asterisk Samoied (n=Samoied@201.21.232.88) |
| 23:36.30 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 23:38.33 | Ryushin | Well, the T1 by itself, and the analog by itself seems to work fine. I think I have to figure out how to correctly set up the zapata.conf file. |
| 23:39.41 | Strom_C | well there's multiple things that can go horribly wrong with such a setup; you've got to do the wanpipe jiggerypokery and then you've got to do the zapata jiggerypokery and then wave the appropriate dead chicken and say the appropriate incantation and then maybe it will work :) |
| 23:40.48 | Strom_C | then cube the voodoo if you're doing fax :) |
| 23:40.50 | file | maybe? nay, never will work |
| 23:41.23 | *** join/#asterisk psilo (i=1000@user-12lmv5n.cable.mindspring.com) |
| 23:41.53 | Strom_C | anyway, I should go fix my client's other consultant's nubbing-up of the network |
| 23:43.46 | Strom_C | you fix one thing, and then you break something brain-damaged that the other guy set up |
| 23:44.27 | Strom_C | note to my clients: the pimply-faced gamer kiddie next door is not a qualified IT professional |
| 23:44.51 | Strom_C | back later :) |
| 23:45.14 | JT | qualifications aren't always everything though |
| 23:45.23 | Strom_C | no, qualifications are not everything |
| 23:45.36 | Strom_C | but understanding what you're doing certainly counts for something |
| 23:46.05 | Strom_C | and trying to do VLANning on a layer-2 switch does not count as "understanding what you're doing" |
| 23:48.01 | JT | heh |
| 23:48.12 | Strom_C | I can't totally blame the consultant though; the client is as much to blame for being ridiculously cheap |
| 23:48.34 | JT | i've seen layer 2 switches that can do vlaning? |
| 23:49.00 | Ryushin | Strom_C: I can't even begin to count how many times when my clients have gone cheap, and paid a lot more in the long run. |
| 23:49.15 | Strom_C | well, this one can kind of sort of do vlanning, but its implementation is really totally 100% stupid |
| 23:49.24 | JT | ah ok |
| 23:49.57 | Strom_C | the impression I get is that the vlan features of the switch are there to convince the nub that buys it that they spent a lot of money on something really exciting when in reality they spent a lot of money on something completely useless |
| 23:50.15 | JT | is it a no-name managed switch? |
| 23:50.20 | Strom_C | it's an Asus |
| 23:50.26 | JT | close enough |
| 23:50.29 | Strom_C | :D |
| 23:50.35 | tzanger | hmm |
| 23:50.54 | Strom_C | I'm going to tell this client "yeah, you need to replace this piece of junk with a Cisco or something similar" but I doubt they'll ever change it |
| 23:51.19 | tzanger | T100P to channel bank with FXO interfaces. relaxdtmf is *off*, busydetect is *off*, I'm getting quite frequent double-dtmf detection on the first digit when Asterisk answers and dumps the caller into the IVR. Any ideas? |
| 23:51.21 | JT | the lifetime warranty of the procurve stuff really impresses me |
| 23:51.27 | JT | that and they're reliable |
| 23:51.59 | Strom_C | the other problem is that each group of six desks at this client's place shares a single driop |
| 23:52.01 | Strom_C | er, drop |
| 23:52.17 | Strom_C | so they've got some cheapo eight-port switch at each group of desks |
| 23:52.23 | JT | ciscos are nice, but they're also arseholes about support |
| 23:52.27 | JT | yeah that's stupid |
| 23:53.00 | Strom_C | so i can't physically segregate voice from data |
| 23:53.14 | JT | unmanaged hell |
| 23:53.18 | Strom_C | yeah |
| 23:53.44 | Strom_C | I brought a friend of mine in to look at it and give me his professional opinion, and his professional opinion was "Tardnet" |
| 23:54.01 | JT | heh |
| 23:54.06 | JT | slightly up from sneakernet |
| 23:54.22 | Strom_C | "and you can tell the client that's what I said." |
| 23:54.54 | Strom_C | this is why I love using him as a subcontractor :D |
| 23:55.39 | JT | fair enough |
| 23:56.30 | Strom_C | ok, time to go |
