| 00:00.01 | C4colo | see if I can isolate the bug |
| 00:00.44 | jaytee | ManxPower, you're right. All I'm seeing in the sip.conf.sample for 1.4.15 and 1.4.20 is mohinterpret and mohsuggest, no musicclass except for a reference to using the musicclass function with SET in the dialplan |
| 00:00.55 | ManxPower | jaytee: EXACTLY! |
| 00:01.04 | jaytee | but it's in the book! |
| 00:01.13 | UnixDog | documentation for 1.4 sucks |
| 00:01.21 | UnixDog | it needs to be rewriten |
| 00:01.28 | jaytee | documentation sucks period |
| 00:02.16 | CanWood | OK, I'm trying macros for the first time today. and I'm finding it's not returning to the next line in a dialplan when it completes. Can anyone see what I'm missing? |
| 00:02.18 | CanWood | http://pastebin.com/m24fd59cc |
| 00:02.22 | jaytee | if you think the * documenation is bad wait till you need to connect * to Exchange UM through sipXecs as a udp/tcp router |
| 00:02.37 | TrentCreek | ManxPower: chan_iax2.c:7540 socket_process: Rejected connect attempt from 64.34.181.47, who was trying to reach |
| 00:05.12 | TrentCreek | no other errors |
| 00:05.22 | TrentCreek | that is not very useful |
| 00:07.54 | CanWood | ne'r mind. got it |
| 00:09.31 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) [NETSPLIT VICTIM] |
| 00:09.31 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-151-207.neo.res.rr.com) [NETSPLIT VICTIM] |
| 00:09.31 | *** join/#asterisk jks (i=jks@193.189.93.254) [NETSPLIT VICTIM] |
| 00:09.31 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.77) [NETSPLIT VICTIM] |
| 00:09.32 | *** join/#asterisk SanityIO_ (n=SanityIO@77.242.106.77) [NETSPLIT VICTIM] |
| 00:09.32 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) [NETSPLIT VICTIM] |
| 00:09.32 | *** join/#asterisk implicit (n=bayan@72.211.213.26) [NETSPLIT VICTIM] |
| 00:09.32 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 00:10.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:10.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:10.21 | *** join/#asterisk bmg505 (n=leon@196.209.78.139) [NETSPLIT VICTIM] |
| 00:10.21 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 00:10.21 | *** join/#asterisk pa (n=pa@unaffiliated/pa) [NETSPLIT VICTIM] |
| 00:10.21 | *** join/#asterisk defswork (n=andy@77.44.54.26) [NETSPLIT VICTIM] |
| 00:10.21 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) [NETSPLIT VICTIM] |
| 00:10.21 | *** join/#asterisk ManxPower (n=manxpowe@195.sub-75-201-209.myvzw.com) [NETSPLIT VICTIM] |
| 00:10.23 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) [NETSPLIT VICTIM] |
| 00:10.23 | *** join/#asterisk DigitalIrony (n=eric@216.207.245.1) [NETSPLIT VICTIM] |
| 00:10.23 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) [NETSPLIT VICTIM] |
| 00:10.23 | C4colo | from what I can tell musicclass=foo under the user details in sip.conf does not seem to work |
| 00:10.23 | *** join/#asterisk Aaton (i=yazz@ghost.230volts.net) [NETSPLIT VICTIM] |
| 00:10.23 | C4colo | has it been deprecated? |
| 00:10.23 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) [NETSPLIT VICTIM] |
| 00:10.43 | jaytee | C4colo, hang on I'm checking |
| 00:10.48 | *** join/#asterisk damjan (n=damjan@217.16.95.15) [NETSPLIT VICTIM] |
| 00:10.48 | *** join/#asterisk moy (n=moy@189.169.83.78) |
| 00:10.51 | MatBoy | anyone a good idea why an imcoming sip trunk could give an busy tone ? |
| 00:10.59 | C4colo | ok, just letting you know that I have even restarted asterisk |
| 00:11.11 | MatBoy | *incoming |
| 00:11.28 | C4colo | fast or normal busy? |
| 00:11.41 | MatBoy | normal |
| 00:11.53 | C4colo | is it hitting the CLI on the box? |
| 00:11.54 | jaytee | if you're using 1.4.x you can use Set(musicclass()=classical) in your dialplan. |
| 00:12.02 | MatBoy | first it gave a " not available" tone |
| 00:12.08 | MatBoy | that's gone :) |
| 00:12.18 | C4colo | SetMusicOnHold(blah) in the dialplan works too |
| 00:12.23 | C4colo | I have tested that |
| 00:14.42 | jaytee | C4colo, I think musicclass=something in sip.conf has been deprecated but someone like russellb or Qwell would have to confirm that. It's is in Asterisk 1.2.x |
| 00:15.56 | C4colo | hmm |
| 00:16.32 | C4colo | let me test setting the alternate music on hold class in the general section |
| 00:17.45 | C4colo | yes, in the [general] section musicclass=rock works |
| 00:18.14 | C4colo | but not for individual users |
| 00:19.08 | C4colo | let me double-check everything |
| 00:20.34 | jaytee | ok, cuz I didn't see anything that indicated musicclass=something had been deprecated in the upgrade.txt file or anywhere else I looked. |
| 00:21.09 | jaytee | if you need to do it per user you'd need to use the musicclass() function with Set. |
| 00:21.19 | jaytee | in the dialplan |
| 00:21.27 | nephfl | how can i tell why this system is hanging up on me in the ivr? |
| 00:21.33 | C4colo | ok, that will be more work, but if that's what I gotta do |
| 00:21.46 | vader-- | ok so i am trying to do realtime voicemail and im running into an issue |
| 00:21.51 | vader-- | i am getting this error in my log |
| 00:22.00 | vader-- | Jul 9 20:19:58 ERROR[20813] res_config_mysql.c: MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. |
| 00:22.00 | vader-- | Jul 9 20:19:58 WARNING[20813] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. |
| 00:22.07 | vader-- | i have tried localhost as well |
| 00:22.23 | vader-- | i use CDR logging to mysql on the same box |
| 00:22.28 | vader-- | and it works with no problem |
| 00:29.46 | nephfl | looks like after awhile of playback i get "chan_sip.c: Maximum retries exceeded" |
| 00:31.22 | ManxPower | C4colo: I suspect you are not matching the peers you think you are matching. |
| 00:31.31 | MatBoy | C4colo: any idea about that busy tone ? |
| 00:33.16 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:34.04 | *** join/#asterisk fgravato (n=frank@office-nat.choopa.net) |
| 00:34.22 | fgravato | ~centos52bug |
| 00:34.22 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 00:36.04 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:37.41 | nephfl | im wondering if this is a router issue causing it to max out retransmit on my sip trunk |
| 00:38.54 | ManxPower | retransmits means "network problems" |
| 00:39.04 | wwalker | ManxPower: jaytee: thanks with regard to 3 versus 4 digiits |
| 00:39.17 | jaytee | yw |
| 00:39.43 | C4colo | MatBoy: I'm changing the setting on the sip user for the phone on my desk then calling it and putting the call on hold |
| 00:39.51 | C4colo | well, one of the phones on my desk, I have four of them |
| 00:39.53 | ManxPower | wwalker: I consult for an office that "never ever will go over 50 people". We just had to order our 2nd batch of 100 DIDs. |
| 00:40.15 | C4colo | sorry, not MatBoy ... ManxPower |
| 00:40.29 | C4colo | and MatBoy: I would watch the CLI to see what kind of failure you are getting ... |
| 00:41.20 | C4colo | if you don't have any traffic on the CLI check to make sure that a) firewall is not blocking the traffic (check iptables rules) b) the connection is either registered or using an ip address for auth |
| 00:41.34 | wwalker | ManxPower: it's a realty office, they implode, sell, rebuild, implode sell rebuild, never getting to 20 phones. but I found that their legacy system was using 28xx as the extensions. Thanks for the feedback though. |
| 00:41.43 | jaytee | ManxPower, I worked in a call center that when we were setting it up before we opened I questioned the logic of choosing a Nortel Option 11C. "Shouldn't we just get an Option 61C? We'll outgrow this in no time." "No, it'll last for 2 years before we have that many people." |
| 00:41.57 | wwalker | anyone using adhearsion? |
| 00:42.47 | jaytee | ManxPower, less than 6 months later I was on the phone to my boss in Chicago going, "Um, Chris? I need to add 10 more ACD agents and 10 more phones and I don't have any open slots in the pbx and I only have 4 available digital TN ports left." |
| 00:43.13 | jaytee | fastest upgrade and cutover I've ever lived through |
| 00:43.23 | C4colo | MatBoy: also make sure you have some sort of route for the call to take, such as a 1NXXNXXXXXX,1,Dial(SIP/somepeer) |
| 00:43.41 | MatBoy | C4colo: for incomming ? |
| 00:43.44 | MatBoy | incoming ? |
| 00:44.05 | C4colo | (that is just an example it won't probably match unless your ITSP sends the call with a 1 in front of it, as ViaTalk does/did) |
| 00:44.14 | C4colo | yea, in whatever incoming context set a route for that DID |
| 00:44.50 | C4colo | like if it were 303-303-3003 add a line that says exten => 3033033003,1,Dial(SIP/someuser) |
| 00:44.59 | C4colo | whatever your user for the phone you want to ring is set to |
| 00:45.38 | C4colo | or Dial(ZAP/3) ... etc |
| 00:45.46 | C4colo | make sure there is a path for the call to follow |
| 00:46.00 | C4colo | if asterisk cannot find a path for the call to follow it will go busy |
| 00:46.28 | C4colo | check all of those things as any of them can cause the call to fail |
| 00:48.49 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.44) |
| 00:52.54 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:01.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:01.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:05.45 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:06.19 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 01:09.05 | jaytee | does anyone here know of a way to block the ability of a user to transfer a call to an external number? I want users to be able to make outgoing calls directly but I want to block external transfers and external call forwarding based on a user by user basis. |
| 01:11.48 | unpaidbill | let me know when you figure it out |
| 01:11.57 | unpaidbill | <PROTECTED> |
| 01:12.19 | jaytee | sure thing |
| 01:12.34 | unpaidbill | If you set the variable __TRANSFER_CONTEXT, then that context will be used (note the two leading underscores). |
| 01:12.36 | unpaidbill | that's how |
| 01:13.10 | jaytee | is it a global variable? |
| 01:13.23 | MatBoy | woei ! |
| 01:13.34 | unpaidbill | not sure if that works with SIP, it's listed in the features.conf information |
| 01:13.44 | MatBoy | C4colo: okido ! |
| 01:14.15 | unpaidbill | i dont know jaytee, i'd have to mess with it |
| 01:14.44 | unpaidbill | and it does work for SIP apparently |
| 01:14.52 | jaytee | unpaidbill, I'm using the softkey features on Polycom phones, not the * features so I don't think it will work with that. |
| 01:15.07 | unpaidbill | it should |
| 01:15.08 | unpaidbill | read doc/channelvariables.txt |
| 01:15.13 | jaytee | I'll check it out. |
| 01:15.31 | unpaidbill | set it and test it. it should work with sip and zap cahnnels |
| 01:18.29 | jaytee | I'll try it out and see. I'm at home now so I'll have to wait till tomorrow. This is what's listed in my channelvariables.txt file for __TRANSFER_CONTEXT; ${TRANSFER_CONTEXT} Context for transferred calls |
| 01:18.43 | unpaidbill | yup |
| 01:18.59 | unpaidbill | set it and rock out with your transfer context out |
| 01:19.03 | MatBoy | why do we IT people have to work actually these days... oh we need 1999 again :P |
| 01:19.26 | unpaidbill | i browse a lot of porn bteween menial tasks |
| 01:20.34 | MatBoy | unpaidbill: hehe, you have to get rid of your stuff in a way indeed, I prefer a real hottie for this under my desk |
| 01:20.47 | jaytee | unpaidbill, I'm not sure how I'd set that variable. Would it be Set(${TRANSFER_CONTEXT}="somecontext" and then in that context do a pattern match and any number more than 4 digits send to Hangup or a some message? |
| 01:25.45 | unpaidbill | ~astbook |
| 01:26.09 | unpaidbill | that would be my guess jaytee |
| 01:27.09 | unpaidbill | i havent tried it so i couldnt say for sure |
| 01:28.33 | unpaidbill | ~asteriskbook |
| 01:28.38 | unpaidbill | ~thebook |
| 01:28.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:28.55 | tzanger | you're not supposed to file your unpaidbills |
| 01:29.03 | unpaidbill | i am going to save that damn pdf this time |
| 01:29.49 | unpaidbill | what is this cachefly business, terrible |
| 01:29.53 | jaytee | unpaidbill, thanks but I've got the book open in my lap, the channelvariables.txt file open on one screen and this chat on another. |
| 01:30.20 | unpaidbill | grats, but i wasnt doign the book for you, i neeeded to check something in it |
| 01:30.37 | unpaidbill | you're just gonna have to test the transfer crap out tomorrow :P |
| 01:31.53 | jaytee | unpaidbill, yeah I know. I just don't understand the two leading underscores |
| 01:32.10 | unpaidbill | well that was from voip-wiki so it could be wrong |
| 01:32.19 | unpaidbill | i would ignore that and go wit hchannelvariables.txt |
| 01:32.26 | jaytee | ok |
| 01:32.36 | unpaidbill | i dunno what the two underscores are about |
| 01:33.38 | jaytee | well, I'll just mess with it tomorrow and experiment. |
| 01:33.45 | russellb | underscores are for inheritance |
| 01:34.11 | russellb | set a variable with two underscores and it will be inherited by all channels created as children of the one you set it on |
| 01:34.19 | russellb | for example, the outbound channel on Dial() |
| 01:34.38 | unpaidbill | ahh |
| 01:34.40 | russellb | a single underscore means inherit one level, and not forever |
| 01:34.48 | russellb | which is very rarely used |
| 01:35.24 | C4colo | russellb: do you know if musicclass=foo has been deprecated as an option in the user definitions in sip.conf? |
| 01:35.38 | C4colo | it works in the general section, but not per user |
| 01:35.43 | C4colo | 1.4 |
| 01:36.19 | russellb | it has been deprecated, yes |
| 01:36.27 | russellb | but it should still work |
| 01:36.28 | jaytee | russellb, so if I wanted to block certain users from doing external transfers I'd have a seperate context for outbound calls that matches any external number to Hangup or a message and set the TRANSFER_CONTEXT to that context? |
| 01:36.40 | C4colo | it is currently ignored |
| 01:36.53 | C4colo | if I SetMusicOnHold(foo) in the dialplan it works |
| 01:37.10 | russellb | jaytee: that would work, yeah |
| 01:37.18 | C4colo | but musicclass=foo under [user00] doesn't |
| 01:37.39 | russellb | if you change that to mohinterpret, does it behave the same way or different? |
| 01:37.40 | C4colo | what is the preferred method under 1.4+ for setting a music class for a user? |
| 01:37.48 | C4colo | let me try |
| 01:37.59 | russellb | look at the sip.conf.sample entries for mohinterpret and mohsuggest |
| 01:38.22 | jaytee | russellb, so I'd do something like Set(${TRANSFER_CONTEXT}="outboundblocked") or do I need to put an underscore in front of the T? |
| 01:39.00 | jaytee | because channelvariables doesn't list any underscore in it for 1.4 |
| 01:39.07 | russellb | that's not valid syntax for Set ... |
| 01:39.15 | *** part/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167059025.pppoe-dynamic.nb.aliant.net) |
| 01:39.23 | russellb | if you're setting it on the channel you want to control, then no, it's not needed |
| 01:39.27 | unpaidbill | he said it already, __ for inheritance! |
| 01:39.57 | unpaidbill | if you didnt use __ the only thing it would effect is the Transfer command, i believe |
| 01:39.58 | C4colo | mohsuggest=foo works under a [user] in sip.conf |
| 01:40.03 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 01:40.10 | unpaidbill | although im probably wrong. |
| 01:40.32 | russellb | C4colo: ok, then you're confusing what musicclass means |
| 01:40.44 | russellb | C4colo: musicclass is the same as mohinterpret. mohsuggest is a new option in 1.4 |
| 01:41.06 | russellb | C4colo: that's the whole reason we made it so we have 2 options now, because the old behavior was confusing. people actually expected it to work like mohsuggest |
| 01:41.31 | russellb | reworked MOH handling for 1.4 |
| 01:41.43 | jaytee | russellb, ok. So I drop the ${} and just use Set(TRANSFER_CONTEXT=MYCONTEXT) ? |
| 01:41.46 | russellb | yes |
| 01:41.53 | jaytee | thanks!!! |
| 01:42.05 | jaytee | I'll mess with it tomorrow |
| 01:42.19 | russellb | k |
| 01:42.31 | jaytee | earlier tonight I used Acronis to make an image backup of my * box. I can sleep easier nights now :-) |
| 01:44.29 | C4colo | hmm, I see |
| 01:45.12 | jaytee | I figure I've cut the restore time down to less than 2 hours if I have a major disaster |
| 01:45.28 | jaytee | until I get a second server as a hot spare |
| 01:48.26 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-75f7262394db23bf) |
| 01:53.15 | *** join/#asterisk JCJC (n=JCJC@72.25.115.165) |
| 01:54.59 | *** join/#asterisk ta^3 (n=tacvbo@189.146.188.205) |
| 01:58.28 | jaytee | nite everyone |
| 01:58.59 | jaytee | and thanks again, russellb |
| 01:59.14 | russellb | np |
| 01:59.55 | jaytee | unpaidbill, and thank you for pointing out the variable setting for that |
| 02:00.14 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:00.15 | C4colo | ok, the mohsuggest seems to do exactly what I was looking for, thanks russellb |
| 02:00.26 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:1) |
| 02:00.28 | russellb | you're welcome :) |
| 02:00.33 | russellb | that will be $5.95 |
| 02:00.43 | C4colo | wow, that's cheap |
| 02:00.46 | russellb | i know! |
| 02:00.52 | C4colo | no minimum 1hr at $175.00 ? |
| 02:01.13 | C4colo | are you sure you are an asterisk consultant? |
| 02:01.45 | russellb | i mean, that will be $175.00 |
| 02:01.45 | C4colo | lol |
| 02:02.11 | russellb | no, i'm not a consultant .. |
| 02:05.59 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:09.10 | vader-- | ok i moved my voicemail.conf over to mysql |
| 02:09.12 | vader-- | kinda cool |
| 02:09.33 | vader-- | now i was able to attach an employee id with each mailbox |
| 02:10.26 | vader-- | not sure if i am ready to temp doing my sip.conf or my dialplan over to mysql |
| 02:10.54 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 02:11.33 | MatBoy | mhh, I hate this busy tone :) |
| 02:12.23 | vader-- | i wonder if i can build a db for access codes |
| 02:12.32 | MatBoy | vader--: why not ? |
| 02:12.40 | vader-- | i have a text file right now i load the numbers into |
| 02:12.50 | vader-- | we do an authenicate against it in the dial plan |
| 02:13.23 | pcrane | vader--: nice to see its going well ;) |
| 02:13.46 | vader-- | exten => _9011XXXXXXXXXXX,n,Authenticate(/etc/asterisk/account_codes|a) |
| 02:13.51 | vader-- | thats what i call in my dialplan |
| 02:13.57 | MatBoy | when I have an incoming trunk, does that number needs to be an extention also ? |
| 02:14.03 | vader-- | pcrane i figure if i get all my sources of information semi synced up |
| 02:14.14 | vader-- | im trying to use our employee id as a reference number in everything |
| 02:14.34 | pcrane | mmm |
| 02:14.35 | vader-- | i just wrote a vbscript that connects to mysql database and retrieves the voicemail information |
| 02:14.39 | vader-- | which is kinda cool |
| 02:14.40 | pcrane | that sounds v. useful |
| 02:15.04 | vader-- | like active directory has an employeeID field that is hidden |
| 02:15.16 | vader-- | you can't adjust it from any screens other than adsiedit or scripting |
| 02:15.26 | vader-- | so im going to start putting that value in |
| 02:15.44 | vader-- | so i can call records from active directory based off that |
| 02:17.12 | MatBoy | mhh, elastix is great, but incomming sip calls are a pain |
| 02:18.53 | MatBoy | someone still some ideas for busy tone ? |
| 02:19.10 | vader-- | pcrane do you know if authenticate command would pull from mysql? |
| 02:19.41 | pcrane | nope, sorry... |
| 02:19.53 | pcrane | I don't know |
| 02:23.18 | JT | this isn't #elastix |
| 02:25.34 | *** part/#asterisk chigambamukoko (n=junk@71.55.10.211) |
| 02:25.50 | *** join/#asterisk quah (i=kilgore_@cpe-075-181-036-081.carolina.res.rr.com) |
| 02:26.13 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 02:27.41 | vader-- | exten => _9011XXXXXXXXXXX,n,Authenticate(Realtime(command options)) |
| 02:27.43 | vader-- | would that work? |
| 02:28.27 | quah | I'm looking for some suggestions on basic IP phones. I currently have 2 * systems running in my pizza shops, I am using Polycom 320's at both stores - and they are just barely loud enough when turned up full blast. Any suggestions for basic phones that can be turned up very loud for noisy environments? |
| 02:29.59 | jblack | use the headset port to plug in a stereo speaker? |
| 02:30.58 | quah | there are banks of 5 phones for order takers. I just need a phone that is loud. These polycoms I guess were made with an office environment in mind. |
| 02:31.24 | quah | my old nortel phones were more than loud enough in the same environment. |
| 02:31.44 | jblack | Is thre some reason you're getting defensive? |
| 02:31.53 | andrewy | do T1/E1 cards show up as Zap/1..n ? |
| 02:32.11 | quah | sorry, not meaning to sound defensive. I'm not feeling that way.. <G> |
| 02:32.13 | jblack | If you're using zapata (that would be PRIs, T1s, and such), you can look at rxgain and txgain, quah |
| 02:33.00 | quah | thanks, yes, I've done that actually. I'm pretty sure this is just a characteristic of these specific phones and their handsets |
| 02:33.18 | JT | i don't see how txgain or rxgain will have any bearing on ringer volume |
| 02:33.23 | quah | the polcom guy on the trixbox forum basically said they were not going to be loud |
| 02:33.38 | jblack | Oh, the ringer. I thought you meant call volume |
| 02:33.40 | JT | get a remote ringer |
| 02:33.53 | quah | no, not the ringer. call volume is correct |
| 02:34.01 | JT | oh ok |
| 02:34.09 | quah | the rigner on these is also low, but that's isn't a big problem. |
| 02:34.12 | JT | headsets with amps |
| 02:34.40 | quah | cant use headsets as people are walking around the store. needs to be a normal handset |
| 02:35.20 | macros73_ | anyone here with hands on experience with a digium aa50? |
| 02:35.21 | JT | maybe you can use a headset amp to amplify a handset, i'm not sure |
| 02:35.52 | quah | im going to be converting 2 more stores soon, figured I just go with a different phone |
| 02:36.07 | quah | but I'm not sure what others are good as I went with the polycom thinking it was the "best" |
| 02:36.26 | jblack | you can have more choice if you use an ATI. |
| 02:36.45 | jblack | Pardon, ATA. That will let you hook up traditional phones, some of which you probably already like |
| 02:38.00 | quah | right, I could do that, but I like some of the functions available on the ip phones. Generally, what other brands are considered good besides ploycom? |
| 02:39.10 | vader-- | i guess i could write an authenticate function that takes a number from a person on the phone then does a realtime lookup against the sql database for it, if it returns null then it's invalid |
| 02:39.16 | vader-- | is that possible? |
| 02:39.42 | jblack | Somethign close to that, yeah |
| 02:40.15 | macros73_ | quah: Pretty happy with the Aastra 480i that I have. |
| 02:40.23 | macros73_ | quah: I'm not using it in a pizza shop, though |
| 02:41.07 | quah | the other thing is there get beat up somewhat, I've had to replace a couple of handsets that couldn't take the abuse of dropping on the floor every once in a while. |
| 02:42.04 | JT | quah: iirc there might be provisioning options that allow you to increase the gain |
| 02:43.29 | quah | yes, there are. I've tried that and it still isn't very loud. The polycom guy says these are just not loud phones. |
| 02:56.04 | jblack | I get freaked out when I go an entire day without email. I've now gone two days with no mail. |
| 02:57.46 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 03:00.22 | MatBoy | weird, my imcoming call is on going to the asterisk box and it's not routed or whatever |
| 03:01.53 | JT | routed? |
| 03:02.48 | MatBoy | yeah, to a extention |
| 03:02.52 | MatBoy | *an |
| 03:04.56 | MatBoy | JT: I get a busy tone |
| 03:06.51 | MatBoy | and I cannot figure out why |
| 03:17.44 | unpaidbill | hmmm, i think i have finally completed this damn firefox 'click to dial' extension for firefox! |
| 03:17.56 | unpaidbill | hooray |
| 03:18.34 | unpaidbill | i feel like i accomplished something today |
| 03:21.05 | lowlevel | http://www.youtube.com/watch?v=q22QRO1uz6c |
| 03:28.27 | JerJer | unpaidbill: you wrote your own? |
| 03:28.38 | unpaidbill | yea |
| 03:28.48 | unpaidbill | i couldnt find one that already existed |
| 03:29.59 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 03:30.05 | unpaidbill | is there one already out there? |
| 03:30.39 | MatBoy | wow, 15 min without internet... that was a strange feeling.. |
| 03:35.43 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:35.56 | unpaidbill | i only see softphones built into firefox |
| 03:37.56 | JerJer | unpaidbill: there are a few - prolly tied to someone's service though |
| 03:38.45 | unpaidbill | yeah, the one i wrote is really simple.. hilight a number on a web page, right click, click dial, it calls your desk phone then connects you to the # you hilighted |
| 03:39.38 | *** join/#asterisk JCJC (n=JCJC@netblock-72-25-115-165.dslextreme.com) |
| 03:40.44 | JerJer | i think junction networks has one |
| 03:41.00 | unpaidbill | eh, i'll stick with mine :P |
| 03:42.59 | *** join/#asterisk lucky|aba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 03:50.59 | JerJer | www.junctionnetworks.com/knowledgebase/onsip/web-services/click-to-call-firefox-add-on |
| 03:52.43 | lucky|aba | thats sick |
| 03:53.58 | MatBoy | lucky|aba: I hope it's not sick, maybe your PC will become infected with it |
| 03:55.59 | lucky|aba | haha |
| 03:56.02 | lucky|aba | lets hope not |
| 03:57.00 | lucky|aba | any recommendations for easy to configure softphones in windows? |
| 03:58.24 | fakhir | zoiper |
| 03:58.28 | unpaidbill | yeah jerjer that uses some other service bs |
| 03:59.10 | unpaidbill | the one i wrote connects directly to the local asterisk server via a cgi request, it also formats the number correctly |
| 03:59.37 | unpaidbill | i use the same cgi for a few different click/dial applications too so it works well for me |
| 03:59.37 | JerJer | yuup |
| 04:00.00 | *** join/#asterisk NeoSkedar (n=NeoSkeda@ip68-3-76-146.ph.ph.cox.net) |
| 04:00.27 | NeoSkedar | anyone avail to answer a question? |
| 04:01.33 | NeoSkedar | didnt think there would be =[ |
| 04:03.19 | JerJer | just ask |
| 04:03.37 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:03.40 | *** join/#asterisk oilinki (n=oil@124.120.7.174) |
| 04:04.13 | *** part/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:05.09 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:05.27 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:05.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:06.58 | *** part/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:07.28 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:08.44 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:10.08 | MatBoy | yes, zoiper is good ! |
| 04:10.22 | MatBoy | lucky|aba: maybe there is twinkle for windows ? |
| 04:10.38 | lucky|aba | yeah i am using that |
| 04:10.44 | lucky|aba | now anyway. |
| 04:10.44 | MatBoy | nice one |
| 04:10.51 | lucky|aba | I can't get the 2 phones to connect though |
| 04:11.43 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 04:13.19 | MatBoy | mhh |
| 04:13.24 | MatBoy | I use it on linux |
| 04:13.37 | MatBoy | but I can't get my inbound working, so useless atm :P |
| 04:13.42 | lucky|aba | asterisk is but the phones are on windows |
| 04:13.53 | MatBoy | yes understand |
| 04:13.56 | lucky|aba | I am just trying to get a call to go across the LAN |
| 04:14.07 | MatBoy | that is simple |
| 04:14.15 | lucky|aba | and it isn't working. Mind you this is an hour into my first install |
| 04:14.23 | lucky|aba | still quite new to it |
| 04:14.37 | MatBoy | try elastix |
| 04:15.15 | MatBoy | wow, damn win98 is fast |
| 04:17.39 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 04:22.55 | *** join/#asterisk b0lt (n=b0lt@pool-71-126-39-32.pitbpa.east.verizon.net) |
| 04:23.35 | NeoSkedar | if im using asterisk now, do i still have to manually edit the extensions and sip.conf files? or does the gui do it |
| 04:24.20 | b0lt | is there any way for me to have asterisk serve as a proxy for an SIP account, except run a script when an outgoing call is made instead of using the associated SIP connection? |
| 04:26.16 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 04:29.47 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 04:30.09 | Robba | Hi Guys |
| 04:31.44 | Robba | I'm having trouble getting a Polycom phone to talk to an Asterisk Server in another network. |
| 04:32.30 | Robba | The phone is in on an ADSL connection behind a NAT trying to talk to an Asterisk server with a public IP |
| 04:33.23 | Robba | The odd thing is when we configure a Linksys Phone it works no worries... |
| 04:34.09 | Robba | any clues? |
| 04:35.32 | Robba | ~centos52bug |
| 04:35.33 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 05:02.05 | *** join/#asterisk PaulAviles (n=Miranda@dsl-7-36.cofs.net) |
| 05:02.34 | PaulAviles | hello all, hope everyone is doing ok |
| 05:03.01 | PaulAviles | does anyone know what a "Calling (Out INV)" means on a cisco phone? |
| 05:03.41 | PaulAviles | this is a remote phone from a nat device over the internet to a nat pbx |
| 05:04.25 | PaulAviles | when I do sip show peers the port for the remote phone is 49874, which I cannot figure out why.... |
| 05:05.00 | PaulAviles | any ideas anyone? |
| 05:08.35 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 05:08.58 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 05:11.15 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 05:13.37 | bijit | anyone has set correctly multiple ip on the same nic? |
| 05:13.56 | denon | you mean, aliases? |
| 05:13.58 | denon | yes, everyone has |
| 05:14.17 | bijit | i have but i get an error |
| 05:20.18 | ManxPower | PaulAviles: that port number is what the nat router translated the source port to when it did the NAT, it is harmless. You should study up on some TCP/UDP/IP networking |
| 05:22.24 | PaulAviles | Manx, well I also have a SIP Trunk provider and that shows as 5060, should I also study that or that is ok? |
| 05:23.21 | ManxPower | the port listed in "sip show peers" is generally totally meaningless. |
| 05:23.52 | PaulAviles | ok what about the Calling (Out INV)? |
| 05:24.09 | ManxPower | I don't use Cisco Phones, check their docs. |
| 05:24.26 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:25.03 | PaulAviles | and the unreachable part? |
| 05:25.22 | PaulAviles | under status for sip show peers |
| 05:25.24 | ManxPower | what unreachable part? |
| 05:26.42 | ManxPower | unreachable means "I can't reach it" |
| 05:26.53 | ManxPower | but it is time for me to go to bed. |
| 05:27.20 | *** part/#asterisk PaulAviles (n=Miranda@dsl-7-36.cofs.net) |
| 05:31.26 | pcrane | urgh... gotta go to gym in about 30mins |
| 05:31.40 | pcrane | been ages since I've done *anything* like it... |
| 05:34.41 | *** join/#asterisk _khan (n=shariq@116.71.6.15) |
| 05:40.08 | *** join/#asterisk eharris_ (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 05:47.31 | *** join/#asterisk Mcooper (n=kmoore@64.110.169.173) |
| 05:49.25 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 06:09.07 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 06:11.34 | lucky|aba | hey hey! its working |
| 06:18.57 | _khan | can we use asterisk as sms gateway |
| 06:20.20 | b0lt | is there any way for me to have asterisk serve as a proxy for an SIP account, except run a script when an outgoing call is made instead of using the associated SIP connection? |
| 06:21.28 | b0lt | i want to use grandcentral's call feature to make free phone calls, but that requires you to query a page and then pick up a call. is there any way to have asterisk run a script and then forward an incoming call to the current line? |
| 06:26.40 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 06:36.05 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 06:36.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:38.19 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 06:38.38 | jblack | b0lt: Anything can be done with sufficient effort. Unless there's a program out there that already does it, that would take a lot. |
| 06:46.23 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:47.51 | *** part/#asterisk Scrye (i=1127@216.81.48.50.epikip.net) |
| 06:47.59 | b0lt | jblack, well, is there any way to patch in an incoming call to a line without any input from the user? |
| 06:55.50 | NeoSkedar | i change my extensions.conf file, do a dialplan reload, it shows the right info, but a dialplan show shows the old info |
| 06:55.55 | NeoSkedar | any ideas |
| 06:57.45 | [netman] | did u change the wrong file (wrong path) ? |
| 06:58.20 | NeoSkedar | the sip file updates worked |
| 06:58.32 | NeoSkedar | "/etc/asterisk/extensions.conf" |
| 06:58.35 | [netman] | sip settings are in another file |
| 06:58.43 | NeoSkedar | yeah sip.conf |
| 06:58.47 | NeoSkedar | same directory |
| 06:59.26 | NeoSkedar | i can register my phones now |
| 06:59.30 | NeoSkedar | just cant make calls |
| 07:04.46 | NeoSkedar | any ideas? |
| 07:05.27 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 07:06.01 | NeoSkedar | is there a way to just completely delted the loaded diaplan |
| 07:14.58 | Mcooper | Hello, trying to figure out a dialplan issue... I am trying to dial an extension when the code 999 is entered. Asterisk is not originating the call, but is in the middle ... We have a Cisco 3745 with an E1 and CCME installed. |
| 07:15.40 | Mcooper | I can pick up the phone, dial 999 and get a dial tone and dail 130.. but I need for it to be automatic... any thoughts? |
| 07:23.23 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 07:23.49 | J4zen | Has anyone ever recorded all the base-audio from Asterisk.. not including the additional/extra's and useless junk |
| 07:24.12 | J4zen | how long do you reckon the the total play-time would be? |
| 07:34.49 | Mcooper | What is the easiest way to set a permanent call forward on an extension? |
| 07:38.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:41.17 | *** join/#asterisk pputman (n=centrex@216.207.245.1) |
| 07:41.36 | Mcooper | pputman, greetings... |
| 07:42.18 | NeoSkedar | omg its actually working! |
| 07:42.44 | pputman | hi Mcooper |
| 07:48.10 | *** join/#asterisk adrin_ (n=adrin@chello089078212153.chello.pl) |
| 07:48.14 | adrin_ | hello gys |
| 07:48.16 | adrin_ | guys |
| 07:48.18 | adrin_ | ;-p |
| 07:48.39 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:48.44 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 07:50.06 | adrin_ | well i have a question... is it possible to invoke a custom command on incoming call? let's say a number A calls a number B, i would like to invoke a command myscript A B which would connect to B's computer and open website with A's details |
| 07:52.11 | adrin_ | System()? |
| 07:56.14 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:57.37 | Mcooper | ~pastebin |
| 07:57.38 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 07:58.39 | Mcooper | http://pastebin.ca/1067798 |
| 07:58.49 | Mcooper | The pastebin.com appears to be down. |
| 07:59.59 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 08:07.47 | ThoMe | good morning! |
| 08:07.53 | NeoSkedar | morning |
| 08:07.58 | ThoMe | +good |
| 08:08.07 | NeoSkedar | well its 1am here |
| 08:08.08 | NeoSkedar | lol |
| 08:08.16 | ThoMe | hoho |
| 08:08.20 | ThoMe | 10am :P |
| 08:08.23 | NeoSkedar | hehe |
| 08:08.34 | NeoSkedar | ive been up since 10am trying to get this thing working |
| 08:08.35 | NeoSkedar | lol |
| 08:08.40 | NeoSkedar | i finally got it |
| 08:11.41 | *** join/#asterisk matrix1233 (n=Administ@41.227.249.101) |
| 08:11.50 | matrix1233 | hello |
| 08:13.36 | matrix1233 | any one can help me about asterisk-video ?? |
| 08:17.39 | matrix1233 | aloo |
| 08:17.46 | NeoSkedar | not many active people |
| 08:17.51 | NeoSkedar | and i barely got my voice to work |
| 08:17.53 | NeoSkedar | so i cant help |
| 08:17.54 | NeoSkedar | lol |
| 08:18.39 | matrix1233 | lol |
| 08:18.57 | NeoSkedar | i got two softphones w/ voicemail setup |
| 08:18.59 | NeoSkedar | im happy |
| 08:19.42 | matrix1233 | i wanna to use my softphones with my camerera |
| 08:19.56 | NeoSkedar | i might get there eventually lol |
| 08:20.07 | matrix1233 | lol |
| 08:20.31 | matrix1233 | ye but i d'ont know how can i install the video on my asterisk |
| 08:20.41 | matrix1233 | no documentation in the net |
| 08:21.05 | matrix1233 | no codec |
| 08:21.05 | NeoSkedar | is it possible? |
| 08:21.05 | matrix1233 | nothing !! |
| 08:21.05 | matrix1233 | yes it's possible |
| 08:21.07 | NeoSkedar | id assume it is, cisco makes phones with built in video |
| 08:21.18 | matrix1233 | it's the h263 codec |
| 08:21.40 | NeoSkedar | standard video |
| 08:22.31 | NeoSkedar | i assume i can have softphones in other places connect to my asterisk? |
| 08:22.51 | matrix1233 | yes |
| 08:22.59 | matrix1233 | it can bé |
| 08:23.16 | matrix1233 | i have a server in france connected |
| 08:23.18 | NeoSkedar | what port is the server on |
| 08:23.26 | matrix1233 | and i can work |
| 08:23.47 | *** join/#asterisk HarmoniousChaos (n=Harmonio@unaffiliated/harmoniouschaos) |
| 08:30.02 | *** join/#asterisk voltagex (n=voltagex@dsl-202-45-99-40.ACT.netspace.net.au) |
| 08:30.29 | voltagex | hi, is DISA available on Asterisk 1.4.21? |
| 08:30.38 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:30.56 | NeoSkedar | anyone have a voip softphone installed that wants to help me |
| 08:46.19 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:48.49 | *** join/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 08:48.54 | bootc | morning folks |
| 08:49.12 | bootc | I've got a reproducible crash in pbx_ael, anyone interested? |
| 08:49.58 | voltagex | bootc: probably not in irc, try submitting a bug report |
| 08:50.03 | tzafrir_laptop | bootc, http://bugs.digium.com is |
| 08:51.05 | voltagex | is there a trick to enabling DISA? it's not shown in the list from 'show applications' |
| 08:51.06 | bootc | I guess I need an account for this? |
| 08:55.19 | matrix1233 | h.263 |
| 08:55.23 | matrix1233 | and h264 |
| 08:55.38 | matrix1233 | any one can help me how can i install it ? |
| 09:07.46 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:08.55 | Mcooper | anyone interested in helping with an issue... |
| 09:11.14 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 09:16.38 | *** join/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 09:17.17 | Ast001 | Hi I have problem I can not force openvox d110p card to work on my system |
| 09:17.54 | *** join/#asterisk genin (n=phrame@ANice-252-1-10-177.w82-122.abo.wanadoo.fr) |
| 09:18.01 | Ast001 | Manual said it works with zaptel driver with no patch but it does not obviosly |
| 09:18.11 | genin | mornin folks |
| 09:18.39 | genin | does anyone know if asterisk can be configured to allow prack mode |
| 09:21.30 | Ast001 | and when I downloaded driver from manufacturer site (for d110pg but works with d110p too) it said ERROR during make of zaptel! |
| 09:22.09 | Ast001 | os: fedora 9 zaptel 1-4-10 |
| 09:22.16 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 09:23.17 | genin | how yah goin mate |
| 09:23.26 | genin | qld sweet |
| 09:23.30 | genin | whereabouts |
| 09:23.57 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 09:31.43 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:32.58 | *** join/#asterisk InsolentDreams (n=Insolent@cpe-76-175-181-103.socal.res.rr.com) |
| 09:34.28 | voltagex | I have no DISA in my build of Asterisk 1.4. Have I missed a config option? |
| 09:34.56 | *** join/#asterisk oussema-harbi (n=harbi_ou@41.226.241.51) |
| 09:37.12 | xnixan | i have centos 5.2, when making zaptel i have these errors |
| 09:37.16 | xnixan | make[4]: *** [/home/xnixan/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 |
| 09:37.16 | xnixan | make[3]: *** [/home/xnixan/asterisk/zaptel-1.4.11/kernel/xpp] Error 2 |
| 09:37.16 | xnixan | make[2]: *** [_module_/home/xnixan/asterisk/zaptel-1.4.11/kernel] Error 2 |
| 09:37.16 | xnixan | make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.6.el5-i686' |
| 09:37.16 | xnixan | make[1]: *** [modules] Error 2 |
| 09:37.16 | xnixan | make[1]: Leaving directory `/home/xnixan/asterisk/zaptel-1.4.11' |
| 09:37.20 | xnixan | make: *** [all] Error 2 |
| 09:37.45 | xnixan | any ideas? |
| 09:39.20 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:40.23 | Mcooper | Go to the xdefs,h file - line 117 and change the 19 to a 17 - recompile. |
| 09:41.08 | Mcooper | Its kernel version mismatch of some sort.. I have done that to get around the error, and there has been no errors reported. |
| 09:41.30 | Mcooper | Its located in the zaptel/kernel directory. |
| 09:41.51 | HarmoniousChaos | Mcooper, have done it, and it, and it did do make, but not make install |
| 09:42.15 | tzafrir_laptop | ~centos52bug |
| 09:42.17 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 09:42.27 | Mcooper | HarmoniousChaos, I did it several times .. an no problem. |
| 09:42.33 | tzafrir_laptop | (should be fixed withthe next zaptel release) |
| 09:42.47 | Mcooper | tzafrir_laptop, That is what I have heard... |
| 09:42.53 | genin | ~prack |
| 09:43.07 | Mcooper | tzafrir_laptop, Thank you for the additional info .... |
| 09:43.20 | genin | hey folks |
| 09:43.33 | genin | i have clients that complain that calls are taking too long to connect |
| 09:43.53 | genin | but i think it has to do with the clients limited upstream and the fact we have to use g711 |
| 09:43.56 | genin | for the moment |
| 09:44.22 | mikkel | Anyone know if you can get SIP to ISDN convert. Need to use a ISDN phone and have a lot of problems connecting it to my sangoma card. Thought mayby I could connect a SIP box and to that box connect by ISDN phone ? |
| 09:44.29 | genin | is there a way to configure asterisk to allow pra&ck mode |
| 09:45.05 | Mcooper | HarmoniousChaos, what was the issue? |
| 09:45.30 | xnixan | Mcooper, do you mean in make install? |
| 09:45.57 | tzafrir_laptop | genin, you mean to allow you to make typos? yes, Asterisk can be configured to do so, if that is what you want |
| 09:46.15 | tzafrir_laptop | But I suggest you be more specific |
| 09:46.21 | genin | HEH |
| 09:47.14 | genin | im just the junior admin so if i say something retarded feel free to give me shit |
| 09:47.25 | tzafrir_laptop | "too long to connect" is probably unrelated to the codec used in the actual call |
| 09:47.43 | Mcooper | xnixan, No in the xdefs files... Hold on... |
| 09:47.47 | genin | hrm |
| 09:47.53 | genin | maybe the route |
| 09:48.00 | *** part/#asterisk oussema-harbi (n=harbi_ou@41.226.241.51) |
| 09:48.18 | voltagex | genin: do you use QoS? |
| 09:48.44 | genin | at the colocation we have they were using Qos and clients were complaining |
| 09:48.55 | genin | but they removed it and ive had less complaints of quality |
| 09:49.12 | genin | i would prefer to use gsm or g729 |
| 09:49.14 | voltagex | genin: whats the upstream limit? |
| 09:49.33 | genin | 10 mega with a 30 meg burst |
| 09:49.39 | voltagex | what's the ping time to your trunk |
| 09:50.00 | voltagex | might be latency? |
| 09:50.24 | genin | does that mean from our asterisk to the provider |
| 09:50.53 | genin | if so its like 17 ms |
| 09:51.07 | genin | i think the bottle neck is clients that are making like up to 8 calls at a time |
| 09:51.10 | genin | with adsl |
| 09:51.14 | genin | using g711 |
| 09:51.35 | genin | so that is 512 kbps up and adsl is up to 786k but rarly that |
| 09:52.07 | genin | i tried using g726 but the quality is really bad |
| 09:52.19 | genin | 723 and 729 i need a trancoding card |
| 09:52.24 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 09:52.26 | voltagex | !!!!!! no wonder |
| 09:52.29 | genin | heh |
| 09:52.29 | Strom_M | g726 should sound almost like 711 |
| 09:52.31 | genin | yeah right |
| 09:52.37 | genin | oh yeah? |
| 09:52.46 | genin | 711 alaw |
| 09:52.50 | genin | is much more clear i think |
| 09:52.57 | Mcooper | xnixan, http://pastebin.ca/1067860 |
| 09:53.04 | genin | i could only get asterisk to work with G726-32 |
| 09:53.12 | genin | and it was really bad |
| 09:53.15 | Strom_M | g711 is clearer than g726, but the quality of g726 is going to be much better than anything else |
| 09:53.18 | Mcooper | That is located in /usr/src/zaptel-1.4.11/kernel/xpp |
| 09:53.26 | voltagex | genin: 512kbit up is like 50kilobytes a second. g711 is 8-10kilobytes a second |
| 09:53.28 | Mcooper | xdefs.h - line 117 |
| 09:53.32 | Strom_M | genin: how was g726 "bad"? |
| 09:53.51 | Mcooper | the 2, 6, 17 was originally 2, 6, 19 |
| 09:53.58 | genin | the ringing and everything sounded like inside a tin can |
| 09:53.59 | Strom_M | voltagex: don't convert shit to kilobytes...it just makes the calculations more confusing |
| 09:54.05 | genin | HEH |
| 09:54.10 | voltagex | Strom_M: brainfart |
| 09:54.15 | genin | 711 is 64k per call right |
| 09:54.25 | Mcooper | How would you break dial tone to send an extension back. |
| 09:54.33 | Mcooper | ISDN PRI rate line? |
| 09:54.38 | Strom_M | Mcooper: that question makes no sense |
| 09:54.52 | genin | someone told me to use gsm |
| 09:54.58 | voltagex | I thought it was 128k both ways |
| 09:54.58 | genin | but audiocodes dont support it |
| 09:55.07 | genin | ah right |
| 09:55.08 | Strom_M | voltagex: no no no. |
| 09:55.11 | genin | there are two call legs |
| 09:55.13 | Strom_M | G711 is 64kbps |
| 09:55.20 | genin | but i mean from the client to asterisk |
| 09:55.21 | Strom_M | G726 is 32kbps |
| 09:55.27 | Strom_M | GSM is 13kbps |
| 09:55.34 | genin | but there are diff g726s |
| 09:55.45 | Strom_M | well, asterisk uses G726-32 |
| 09:55.56 | genin | basically we need to use a codec that takes less bandwidth |
| 09:55.58 | genin | ah right |
| 09:55.59 | voltagex | Strom_M: what the? Then what the hell am i using to use 128kbit both ways for one call? |
| 09:56.06 | genin | payload type 2 |
| 09:56.07 | genin | heh |
| 09:56.11 | Strom_M | voltagex: slin perhaps? |
| 09:56.18 | Strom_M | and if G726 sounds like crap, it's not asterisk's fault |
| 09:56.20 | genin | from asterisk it is double |
| 09:56.26 | genin | because it is a back to back user agent |
| 09:56.36 | genin | so for one call there are two call legs |
| 09:56.57 | Strom_M | genin: you're only partially right |
| 09:57.06 | genin | that is better than not right at all |
| 09:57.13 | genin | i didnt know anything about asterisk in feb |
| 09:57.14 | genin | heh |
| 09:57.42 | genin | hrm |
| 09:57.44 | Mcooper | Strom_M, We have a Cisco 3745 with an PRI ISDN E1 with in asterisk sitting on the E1. When a extension is dialed 999 we want to send that call to extension 130 - The phones are off the Cisco CCME |
| 09:58.19 | Strom_M | Mcooper: the question still doesnt make a whole lot of sense |
| 09:58.22 | Mcooper | I can get asterisk to do the play back, we can get it to do just about anything.. but we cannot get it to dial 130 |
| 09:58.35 | genin | so there isnt anyway to have asterisk encode in a codec that uses less bandwidth |
| 09:58.38 | Strom_M | where is "extension 130" located? |
| 09:58.40 | genin | with putting a card in it right |
| 09:58.41 | Strom_M | genin: yes there is |
| 09:58.46 | genin | ah |
| 09:58.52 | Mcooper | On the Cisco Call manager... |
| 09:58.54 | genin | i only got 726 to work |
| 09:58.57 | Strom_M | GSM, ILBC, SPEEX are all low bandwidth |
| 09:59.05 | genin | ah okay |
| 09:59.10 | Strom_M | g729 can use either the TC400B or a software codec |
| 09:59.13 | genin | i dont think our clients audiocodes |
| 09:59.13 | Mcooper | we just need to send the tones back to Cisco.. so it deals with it. |
| 09:59.17 | genin | can it? |
| 09:59.23 | genin | software |
| 09:59.25 | Strom_M | genin: huh? |
| 09:59.29 | genin | 7729 is the ideal choice |
| 09:59.36 | genin | 729 |
| 09:59.44 | Strom_M | genin: if you think g726 sounds bad, boy are you in for a shock with g729 |
| 09:59.45 | *** join/#asterisk Alibb (n=Alibb@41.226.241.51) |
| 09:59.52 | genin | HEH |
| 09:59.59 | Strom_M | it sounds 10x worse :) |
| 10:00.01 | voltagex | lol |
| 10:00.05 | genin | 8 kb |
| 10:00.08 | genin | per call |
| 10:00.19 | Strom_M | Mcooper: there should be no "tones" that get sent at all |
| 10:00.19 | genin | someone told me that gsm is a telco standard |
| 10:00.27 | Strom_M | GSM is a mobile phone standard, yes |
| 10:00.35 | Mcooper | True... |
| 10:00.40 | genin | hmm |
| 10:00.56 | genin | and how about 723.1 |
| 10:01.12 | Strom_M | also a telco standard, but that sounds even worse than 729 |
| 10:01.17 | genin | HEH |
| 10:01.19 | genin | ouch |
| 10:01.26 | Mcooper | how would you suggest that I get the Cisco to handle the 130.. What we are getting now.. we dial 999 and asterisk picks up... and gives us another dial tone.. |
| 10:01.55 | Strom_M | Mcooper: it sounds like you fucked up your configuration somewhere then |
| 10:02.09 | Strom_M | also, i have zero idea how to handle CCM |
| 10:02.14 | Strom_M | ive never touched it |
| 10:02.51 | genin | what software/configuration do i needto try and encode various codecs to try |
| 10:02.51 | genin | do i need to patch the asterisk or something |
| 10:03.04 | Strom_M | genin: no |
| 10:03.06 | Strom_M | go read the book |
| 10:03.07 | Strom_M | ~book |
| 10:03.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 10:03.11 | Strom_M | also, read this |
| 10:03.13 | Strom_M | ~101 |
| 10:03.14 | jbot | somebody said 101 was Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 10:03.18 | genin | cool |
| 10:03.19 | genin | thanks |
| 10:04.04 | Mcooper | Strom_M, I would assume that I have messed it up some where.. |
| 10:04.22 | Strom_M | Mcooper: where is the extension you're dialing from? |
| 10:04.44 | genin | and one last question |
| 10:04.49 | genin | or is it in the book |
| 10:04.49 | Mcooper | The phones for now come off the CCME |
| 10:04.50 | genin | heh |
| 10:04.54 | genin | allowing prack mode |
| 10:05.05 | genin | is it just configuration |
| 10:05.09 | Strom_M | wtf is "prack mode"? |
| 10:05.12 | genin | HEH |
| 10:05.14 | Mcooper | All the CCME is doing is providing Dial tone. |
| 10:05.23 | genin | well in the sip trace |
| 10:05.29 | genin | normally it sends an ACK |
| 10:05.57 | genin | but for some reason i was lead to believe PRACK is better which the clients gateways support |
| 10:06.07 | Strom_M | Mcooper: pastebin your extensions.conf |
| 10:06.09 | Strom_M | ~pb |
| 10:06.10 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:06.29 | Strom_M | genin: i have no idea where you heard that. I've never heard of such a thing. |
| 10:06.37 | genin | something about our current setup there is one udp packet sent and if it gets lost then sometimes the client might not here the ringing |
| 10:06.42 | genin | hrm |
| 10:07.47 | xnixan | Mcooper, i had deselected the xpp and it worked ok for make, but with make install this problem appears http://www.pastebin.org/50195 |
| 10:08.05 | Mcooper | xnixan, Ok... |
| 10:08.39 | Mcooper | Strom_M, Here it is.. it very simple.. I got tossed into this project after the initital people left... |
| 10:09.00 | Mcooper | http://pastebin.ca/1067881 |
| 10:09.15 | Mcooper | I am using the uncommented [incoming] |
| 10:09.34 | Strom_M | Mcooper: that's....ugh, that's very wrong. |
| 10:09.39 | Strom_M | i suggest you read the book. |
| 10:09.41 | Strom_M | ~book |
| 10:09.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 10:10.42 | Mcooper | Strom_M, I figured it was wrong.. but I am working from having to rebuild the system from scratch... I have been using the o'reilly book... |
| 10:11.06 | Mcooper | xnixan, This is a different error than what I was seeing.. which version of zaptel are you using? |
| 10:11.19 | Strom_M | Mcooper: I'd like to kindly suggest that you learn how to configure asterisk :) |
| 10:11.48 | Mcooper | Strom_M, So what would you suggest.. oh great one... |
| 10:11.51 | genin | rtfm |
| 10:11.52 | genin | heh |
| 10:11.59 | Mcooper | genin, have been. |
| 10:12.01 | xnixan | Mcooper, zaptel-1.4.11 |
| 10:12.08 | genin | no i need to |
| 10:12.11 | Mcooper | xnixan, same version... |
| 10:12.13 | genin | i was thinking out loud |
| 10:12.13 | genin | heh |
| 10:12.18 | Mcooper | genin, it is very good... |
| 10:12.19 | xnixan | Mcooper, yup |
| 10:12.55 | Mcooper | xnixan, interesting.. I do not know what is wrong.. I just ran into the one I experienced.. and that took care of it.. |
| 10:13.01 | genin | <genin> is there a way to configure asterisk to allow pra&ck mode |
| 10:13.02 | genin | <Mcooper> HarmoniousChaos, what was the issue? |
| 10:13.02 | genin | <xnixan> Mcooper, do you mean in make install? |
| 10:13.02 | genin | <tzafrir_laptop> genin, you mean to allow you to make typos? yes, Asterisk can be configured |
| 10:13.06 | genin | <PROTECTED> |
| 10:13.07 | genin | fuck osrry |
| 10:13.09 | genin | heh |
| 10:13.24 | genin | anyways someone else told me i coudl configure asterisk to allow prack mode |
| 10:13.30 | xnixan | genin, yup |
| 10:26.52 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:26.52 | *** join/#asterisk THE-DON (n=gg@86.108.82.44) |
| 10:27.04 | THE-DON | hello guys |
| 10:27.36 | *** join/#asterisk my007ms (i=master@botmaster.x86.be) |
| 10:27.40 | THE-DON | i have an question about asterisk system |
| 10:27.41 | my007ms | hello |
| 10:28.28 | my007ms | i have truck use GSM 12 units to call mobiles sometime 1 or 2 of this units stop working |
| 10:29.17 | THE-DON | asterisk can handle ip protocoles perfectly, but can it handle land phones as well |
| 10:29.19 | my007ms | so trunk as wall start not working coz he can not identified with zap port that have not working units |
| 10:29.41 | my007ms | is there method make auto mark bad zap port to be out of trunk ?? |
| 10:30.25 | THE-DON | i want to know if i wanna build up a system for 25 phones connected to 10 landlines and 2 voip can i do that? |
| 10:31.27 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d369af8a2b1ff7cc) |
| 10:31.29 | my007ms | THE-DON yes |
| 10:31.32 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:31.43 | tzafrir_laptop | my007ms, the problem is that Asterisk (or chan_zap) fails to start? |
| 10:32.04 | tzafrir_laptop | What type of gsm units? How are they connected to Asterisk? |
| 10:32.25 | my007ms | tzafrir_laptop it's normal one unknown sometime :) |
| 10:32.29 | THE-DON | my007ms and they wouldnt interfere anything in the system |
| 10:32.34 | my007ms | and connect with zap port yes |
| 10:32.49 | my007ms | thy work exact like land line |
| 10:33.22 | tzafrir_laptop | so asterisk runs fine and you want Asterisk not to call through those ports? |
| 10:33.29 | my007ms | yes |
| 10:33.55 | tzafrir_laptop | remove them from the group you use and reload config |
| 10:34.09 | my007ms | but i don't know how asterisk will identifies this |
| 10:34.15 | tzafrir_laptop | or maybe it's enough that you don't get power there |
| 10:34.25 | my007ms | tzafrir_laptop ya this what i do for now i need to automated this task |
| 10:35.01 | *** join/#asterisk Segnale007 (n=Segnale0@host189-253-dynamic.36-79-r.retail.telecomitalia.it) |
| 10:35.01 | tzafrir_laptop | do the ports connected to bad ports/units appear as "RED" in /proc/zaptel/1 ? |
| 10:35.23 | my007ms | i don't check this b4 |
| 10:35.28 | tzafrir_laptop | What version of zaptel is it? |
| 10:35.42 | tzafrir_laptop | And what device is it? |
| 10:35.55 | THE-DON | my007ms so what kinda of device should i get to make the 10 main landlines enter the asterisk systm |
| 10:36.46 | *** join/#asterisk gfather1 (n=enforcer@86.108.82.44) |
| 10:36.46 | THE-DON | before i devide them to 25 ip phones |
| 10:37.24 | my007ms | THE-DON u will find many options in http://www.digium.com/en/ |
| 10:37.39 | my007ms | there is other Co but digium is the 1st |
| 10:37.54 | my007ms | tzafrir_laptop i use zaptel-1.2.19-3 |
| 10:37.56 | gfather1 | <THE-DON> how many land lines do u want to connect to the asterisk ? |
| 10:38.12 | THE-DON | <gfather1> 10 |
| 10:38.27 | THE-DON | and i wanna bring 2 voip lines |
| 10:38.41 | THE-DON | to connect them to a 25 seperate ip phones |
| 10:38.51 | my007ms | tzafrir_laptop i have 3 card every card have 4 port |
| 10:39.07 | THE-DON | <my007ms> ill check it out |
| 10:39.10 | gfather1 | well then u have to get a diguim that will have 8 + ports |
| 10:39.15 | my007ms | THE-DON 10 zaptel or pri ?? |
| 10:39.30 | tzafrir_laptop | zaptel 1.2 does not support channel alarms |
| 10:39.55 | THE-DON | zaptel old phone lines |
| 10:40.14 | my007ms | THE-DON then all u see fxo prots |
| 10:40.18 | THE-DON | the one that comes with only 2 wires |
| 10:40.24 | THE-DON | without data |
| 10:40.44 | THE-DON | yup |
| 10:40.57 | gfather1 | <my007ms> so hee needs only fxo ports , right no need for fxs |
| 10:43.14 | tzafrir_laptop | gfather1, they also have a larger card with up to 24 ports |
| 10:43.33 | tzafrir_laptop | the TDM24xx card |
| 10:44.01 | my007ms | i think 1 card 8 and 1 card 4 port will be what he need |
| 10:44.20 | my007ms | to keep 2 ports if he need any upgrade and save some money :) |
| 10:44.26 | THE-DON | aha |
| 10:44.27 | gfather1 | <tzafrir_laptop> but if he have like 10 lines , i think hell need 1 or 2 lines for fax or stuff |
| 10:44.42 | gfather1 | wich is better if it was not linked to asterisk , right |
| 10:44.48 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 10:44.51 | tzafrir_laptop | gfather1, I didn't say you'll need it. I said that this is what they sell |
| 10:44.52 | Uatec | Evening |
| 10:45.07 | THE-DON | no problem ill connect them seperatly :( cause im not that good in configuiring |
| 10:45.21 | my007ms | tzafrir_laptop so can zaptel 1.4 help me solve my problem ?? |
| 10:45.21 | gfather1 | <tzafrir_laptop yes ofcource , they have really good stuff with allot of ports :) |
| 10:45.33 | tzafrir_laptop | my007ms, latest zaptel 1.4, yes |
| 10:45.51 | THE-DON | zalataa |
| 10:46.03 | my007ms | i have the TE card it's 2 span can take 64 line PRI ;) |
| 10:47.01 | tzafrir_laptop | 60 lines (if you use E1/PRI) actually |
| 10:47.26 | tzafrir_laptop | But I agree that PRI is nice |
| 10:50.11 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 10:51.26 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:52.28 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:55.20 | Mcooper | tzafrir_laptop, What do you think of my issue (besides extension file is screwed - which I knew anyways) |
| 10:55.44 | Mcooper | I am stumped and could use a couple of ideas.. reading the book.. is not doing much good either... |
| 10:56.05 | tzafrir_laptop | Mcooper, what issue, exactly? |
| 10:56.28 | THE-DON | guys is it better to buy digium products from digium direct or a reseller |
| 10:56.52 | Mcooper | I need to have asterisk "dial" an extension through a call manager express.. |
| 10:57.50 | Mcooper | if a person picks up a phone.. and dials 999 we want asterisks to dial extension 130 - located over on the Cisco CCME |
| 10:58.18 | Mcooper | Asterisk sees the 999 come in.. ... |
| 10:59.02 | *** join/#asterisk coppice (n=chatzill@122.193.17.210.dyn.pacific.net.hk) |
| 10:59.05 | Mcooper | http://pastebin.ca/1067915 |
| 10:59.37 | Mcooper | that is what we are seeing in asterisk.. The call manager register the "number" of the phone, and the fact that we pressed 999 |
| 11:01.59 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 11:06.34 | tzafrir_laptop | Mcooper, how is that CCME connected? PRI? SIP? |
| 11:07.33 | tzafrir_laptop | exten => 130,1,Dial(SIP/ccme/999); ? |
| 11:08.34 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 11:08.39 | tzafrir_laptop | ah, PRI. can you enable pri debug on that span? |
| 11:10.09 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 11:10.29 | gnorbert | Hi for all |
| 11:11.42 | gnorbert | Does sb know a free softphone for unix, that can be handle-d via console? |
| 11:12.31 | *** join/#asterisk pootle (n=pootle@adsl.ntsols.com) |
| 11:12.36 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:15.01 | Mcooper | PRI |
| 11:15.27 | Mcooper | tzafrir_laptop, Its connected PRI. |
| 11:16.27 | Mcooper | Got it... |
| 11:19.48 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 11:20.10 | Stese | Hi all :) |
| 11:20.23 | Mcooper | tzafrir_laptop, http://pastebin.ca/1067935 |
| 11:22.08 | *** part/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 11:23.40 | *** part/#asterisk voltagex (n=voltagex@dsl-202-45-99-40.ACT.netspace.net.au) |
| 11:23.57 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:24.18 | Mcooper | tzafrir_laptop, You there? |
| 11:25.47 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:30.41 | *** join/#asterisk pputman (n=centrex@216.207.245.1) |
| 11:32.38 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-185.hsi.ish.de) |
| 11:34.23 | *** join/#asterisk joelwells (n=joel@cpe-76-175-176-77.socal.res.rr.com) |
| 11:34.25 | *** join/#asterisk lotho (n=lotho@static.69.46.46.78.clients.your-server.de) |
| 11:35.35 | *** part/#asterisk InsolentDreams (n=Insolent@cpe-76-175-181-103.socal.res.rr.com) |
| 11:36.16 | joelwells | hello, can anyone recommend a fairly inexpensive (9.99USD) SIP provider for residential in the US that has "all the features"? |
| 11:37.10 | joelwells | I'd go for a yearly fee too if it's cheaper, but still as reliable of course |
| 11:37.49 | joelwells | I'm ok with a little downtime...doesn't have to be bandwidth.com |
| 11:38.23 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-154-224.red.bezeqint.net) |
| 11:40.42 | Dovid | hi. when transfering a call from one phone to another the h extension is called. why is it if the call is still on going ? http://pastebin.ca/1067961 |
| 11:50.50 | tzafrir_laptop | Mcooper, I don't really understand this. Is the call originated from an analog phone, BTW? |
| 11:52.42 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 11:54.40 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 11:59.18 | *** join/#asterisk macros73_ (n=cs@c-67-165-65-27.hsd1.pa.comcast.net) |
| 12:03.20 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 12:07.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:07.50 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 12:08.37 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 12:08.37 | *** join/#asterisk _ys (i=yuri@91.151.196.254) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk genin (n=phrame@ANice-252-1-10-177.w82-122.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk Aaton (i=yazz@ghost.230volts.net) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk krdian (i=krdian@killer.radom.net) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk SplasPood (i=jwb@69.9.183.119) [NETSPLIT VICTIM] |
| 12:08.37 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 12:08.37 | *** mode/#asterisk [+o file] by irc.freenode.net |
| 12:11.11 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:11.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:14.35 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582770.dsl.bell.ca) |
| 12:15.49 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:19.16 | *** join/#asterisk PepOSX (n=angeldav@190.72.145.54) |
| 12:23.24 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 12:25.53 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 12:26.42 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:30.30 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 12:32.50 | iamhrh | can anyone tell me how sound files are sorted for music on hold when random=no in the configuration? I have a bunch of files named 001-011, but they seem to sort in a seemingly random, but consistently repeatable order. |
| 12:33.14 | *** join/#asterisk CVirus (n=Burzum@196.218.41.31) |
| 12:34.23 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 12:35.41 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:36.01 | Stese | is it 010,011,001,002,003 etc? |
| 12:36.14 | iamhrh | 001, 002, 003, 004, ... 010, 011 |
| 12:36.33 | *** join/#asterisk iobug (n=gustav@212.124.71.46) |
| 12:36.40 | iamhrh | i just found an option on http://bugs.digium.com/view.php?id=10855 ... about to try it |
| 12:36.44 | iamhrh | sort=alpha |
| 12:36.45 | iobug | ~book |
| 12:36.46 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:36.58 | iamhrh | (read it cover to cover :-)) |
| 12:37.00 | Stese | I meant the play order, but never mind |
| 12:37.14 | iamhrh | it starts with 007 |
| 12:37.16 | iamhrh | atm |
| 12:37.58 | *** join/#asterisk Bonix-BR (n=Bonix-BR@cisco-rt6.isimples.com.br) |
| 12:38.17 | iamhrh | yeah didn't seem to make a darn bit of difference |
| 12:40.57 | macros73 | joelwells: Vitelity. |
| 12:41.42 | *** join/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 12:42.37 | Ast001 | notice chan_zap.c pri d channel : pri got event hdlc bad fcs (8) on primary D channel of span 1 What does it mean ? |
| 12:43.20 | Ast001 | zaptel conf is soan=1,1,0,ccs,hdb3 |
| 12:43.26 | Ast001 | *span |
| 12:43.57 | Ast001 | with bchan=1-15,17-31 |
| 12:44.12 | Ast001 | dchan=16 default and loadzone=hu |
| 12:47.15 | Ast001 | it is interesting that after install of debian linux i got message /dev/hda losing interrupts |
| 12:47.41 | Ast001 | and after I installed Fedora 9 and it looks ok |
| 12:48.04 | tzafrir_laptop | Ast001, non crc4? |
| 12:48.06 | *** part/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 12:48.09 | Ast001 | motherboard is GA-EP35-DS3R/DS3 |
| 12:48.13 | Ast001 | no crc4 |
| 12:48.27 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 12:48.27 | Ast001 | with crc4 i got rec/blue signal |
| 12:48.34 | Ast001 | in zttool |
| 12:48.38 | gnorbert | How can I give an extension by linphone via console? |
| 12:48.49 | tzafrir_laptop | gnorbert, libphonec? |
| 12:49.00 | tzafrir_laptop | Ast001, a bad line? |
| 12:49.06 | gnorbert | linphonec |
| 12:49.07 | Ast001 | wihout i dunno |
| 12:49.21 | Ast001 | People from telco come yesterday and they say everything is ok |
| 12:49.34 | Ast001 | they tested line with tester and no problems |
| 12:49.49 | Ast001 | and said it is my equipment |
| 12:49.56 | Ast001 | which cause a problem |
| 12:50.05 | gnorbert | tzafrir_laptop: I tried >linphonec -s sip:localhost |
| 12:50.26 | tzafrir_laptop | gnorbert, sorry, never really used it |
| 12:50.47 | Ast001 | its openvox d110p card |
| 12:50.48 | gnorbert | No problem.:) |
| 12:51.28 | [TK]D-Fender | Ast001: pastebin "cat /proc/interrupts" |
| 12:51.30 | [TK]D-Fender | ~pb |
| 12:51.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:51.44 | Ast001 | ok just a sec I need to reconnect |
| 12:56.50 | gnorbert | Does sb know, how can I call an extension via console? |
| 12:57.26 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 12:58.03 | Stese | dial? |
| 12:58.46 | gnorbert | No, and not even call |
| 12:59.34 | gnorbert | Or DTMF (In ekiga you can dial an extension typing DTMF <extension>) |
| 12:59.48 | gnorbert | I mean Nor.:) |
| 13:00.00 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 13:00.40 | gnorbert | I use linphone, iff |
| 13:00.47 | gnorbert | <PROTECTED> |
| 13:01.20 | [TK]D-Fender | gnorbert: "man libphonec" |
| 13:02.08 | *** join/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 13:02.39 | Ast001 | Here are my interrupts http://pastebin.com/m7d2e1965 |
| 13:02.56 | gnorbert | [TK]D-Fender: I use linphone, not libphone, it's a free softphone under unix...:) |
| 13:02.58 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:02.58 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:03.02 | *** join/#asterisk kippi (n=kippi@untrust-gct.equinoxit.net) |
| 13:03.10 | Ast001 | pri got event hdlc bad fcs(8) on primary Dchannel of span 1 |
| 13:03.12 | [TK]D-Fender | gnorbert: "man linphonec" |
| 13:03.21 | *** part/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 13:03.22 | [TK]D-Fender | gnorbert: Was a typo. now READ |
| 13:03.28 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 13:03.34 | l2cache | hello |
| 13:03.41 | gnorbert | I've already read. |
| 13:03.53 | [TK]D-Fender | gnorbert: the man page tells you how to dial. |
| 13:04.31 | [TK]D-Fender | Ast001: 19: 5719 5718 2214964 5757 IO-APIC-fasteoi uhci_hcd:usb7, ata_piix, wcte11xp, ahci <--- its sharing IRQ's with a LOT of devices. This is extremely bad. |
| 13:04.34 | kippi | hey |
| 13:04.48 | gnorbert | Only a number, not an extension... |
| 13:04.56 | Ast001 | how can I solve this situation ? |
| 13:05.11 | Ast001 | I can turn off cdrom and usb in bios |
| 13:05.11 | [TK]D-Fender | Ast001: Get that device on its own IRQ. It is also based on a poorer PCI design. What */Zaptel version are you running |
| 13:05.25 | Ast001 | 1-4-10 |
| 13:05.31 | kippi | I have been looking at loading sip.conf from a database, I have setup the tables etc, it is working fine for the dialplan, I have this setup so far: http://pastebin.ca/1068030 could someone point me in the right direction? |
| 13:05.55 | Ast001 | zaptel 1-4-10 |
| 13:06.06 | Ast001 | but how can I move it to its own irq ? |
| 13:06.13 | *** join/#asterisk Peri (n=redanti@surf99.net.rss.rogers.com) |
| 13:06.14 | [TK]D-Fender | gnorbert: The most important commands are : call <sip url> : to call someone. A sip url is in the form sip:user@host |
| 13:06.28 | Peri | morning |
| 13:06.46 | lmadsen | Ast001: you won't see users until a phone registers with the system |
| 13:06.55 | [TK]D-Fender | Ast001: Either do it in your MB's BIOS or hope your OS can help manage it. |
| 13:07.01 | lmadsen | asterisk doesn't pre-cache the sippeers and sipusers prior to that |
| 13:07.11 | lmadsen | oops! |
| 13:07.16 | lmadsen | s/Ast001/kippi |
| 13:07.20 | lmadsen | kippi: see above |
| 13:07.43 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 13:07.51 | Ast001 | kippi ? whats that ? |
| 13:08.27 | Ast001 | s/Ast001/kippi |
| 13:08.45 | Ast001 | i can only disable cdrom and usb devices in bios |
| 13:08.46 | dominic1 | what do you think about setting the accountnames to a full e.164 number? |
| 13:09.10 | Ast001 | how can I set irqs in Fedora 9 ? |
| 13:09.15 | kippi | when the handset registers I get - No matching peer found |
| 13:09.15 | gnorbert | [TK]D-Fender: Thanks, but again: I can call that way any phone number, but not an extension. |
| 13:09.38 | [TK]D-Fender | gnorbert: You don't seem to understand SIP at all. EXTENSION = NUMBER |
| 13:09.39 | lmadsen | Ast001: that is a function of the motherboard and BIOS, not the OS |
| 13:09.44 | *** join/#asterisk browser (n=afg_ch@213.189.154.108) |
| 13:09.53 | lmadsen | gnorbert: well technically it is alphanumeric :) |
| 13:10.01 | gnorbert | I guess, you don't really understand the problem.:) |
| 13:10.16 | lmadsen | s/gnobert/[TK]D-Fender |
| 13:10.18 | Ast001 | so I need to go in bios and set |
| 13:10.19 | lmadsen | I gotta stop doing that |
| 13:10.25 | gnorbert | The server waits for an extension.:) |
| 13:10.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:10.43 | [TK]D-Fender | gnorbert: user@host <- user = the extension to dial |
| 13:10.46 | Ast001 | and you are 100% sure this annoying hdlc bad fcs(8) on primary D channel is coz of irq ? |
| 13:11.07 | gnorbert | [TK]D-Fender: Sry, stone me.:D |
| 13:11.31 | [TK]D-Fender | Ast001: as close to 100% as I can get |
| 13:11.36 | Ast001 | ok |
| 13:11.45 | [TK]D-Fender | Ast001: sharing IRQ's on an old card like that... hell yeah |
| 13:11.46 | Ast001 | thank you I will try to do it |
| 13:12.13 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:12.39 | *** part/#asterisk Ast001 (n=Administ@cable-89-216-185-58.dynamic.sbb.rs) |
| 13:14.14 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:14.51 | jaytee | I'm having a senior moment this morning. What does _**XXXX match against? |
| 13:15.06 | *** join/#asterisk flokuehn (i=flokuehn@globalways/developer/flokuehn) |
| 13:15.21 | [TK]D-Fender | jaytee: 2 *'s and any 4 digits |
| 13:15.37 | [TK]D-Fender | (followed by) |
| 13:16.40 | jaytee | [TK]D-Fender, that's what I figured but I thought I remembered there was something special about the 2 *'s and I ended up using it in one place in my dialplan but forgot to put comments there to remind me. |
| 13:17.18 | [TK]D-Fender | jaytee: nothing inherently special. |
| 13:18.38 | dominic1 | must the username macht the name? |
| 13:18.43 | jaytee | Aha!!! I remember now! it's for when someone presses a BLF on a Grungestream |
| 13:21.27 | dominic1 | ? |
| 13:24.38 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:25.26 | [TK]D-Fender | jaytee: Only if you set it up that way... |
| 13:27.20 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-85-204.vif.net) |
| 13:27.52 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 13:29.08 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:30.42 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 13:30.58 | jaytee | [TK]D-Fender, yeah, I was messing around with it awhile back and it would work but for some reason not consistently, I have no idea why....it couldn't possibly be the Grandstreams that are to blame ;-S |
| 13:31.32 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 13:31.32 | hsv-al | . |
| 13:31.48 | [TK]D-Fender | jaytee: it'd be the phone's fault. perhaps its BLF keys are state-based and reserve certain patterns, etc. |
| 13:32.23 | *** join/#asterisk xezz (n=asdasd@trust-it.gr) |
| 13:32.33 | dvdevel | greetings, all. i see a note on voip-info wiki regarding Macro "Macro isn't very good at going several levels deep. It is by design limited to 7 levels deep..." can somebody explain that? by "level" did he mean "self recursion" or so? |
| 13:33.05 | jaytee | dvdevel, it means calling one macro from another drops you down a level. think nesting |
| 13:33.07 | *** join/#asterisk moy (n=moy@nat/ibm/x-38702e3175386c7c) |
| 13:33.38 | xezz | hello, im having an issue compiling sangoma a500d : http://www.pastebin.ca/1068055 |
| 13:33.38 | *** join/#asterisk ManxPower (n=manxpowe@212.sub-75-202-228.myvzw.com) |
| 13:33.41 | xezz | any idea ? |
| 13:33.55 | dvdevel | ah, excellent. if you "goto" out of a macro, does it "break" the nesting (i.e. resets level back to zero)? |
| 13:34.10 | [TK]D-Fender | dvdevel: No |
| 13:34.47 | dvdevel | hmm... mildly troubling, but may never be a problem in the real world. |
| 13:34.57 | *** join/#asterisk salmander (n=omi@CPE0002b323839e-CM001225024984.cpe.net.cable.rogers.com) |
| 13:34.59 | lmadsen | [TK]D-Fender: actually I think it does because it exists the Macro() as far as I'm aware |
| 13:35.14 | lmadsen | exits* |
| 13:35.37 | [TK]D-Fender | lmadsen: I would figure you had to either explicitly call "macroexit" or run out of "s". |
| 13:35.49 | ManxPower | [TK]D-Fender: I think that applies to Gosubs |
| 13:35.55 | lmadsen | [TK]D-Fender: I think going a Goto() out of the Macro() also causes it to end |
| 13:36.03 | lmadsen | doign* |
| 13:36.06 | lmadsen | doing* |
| 13:36.07 | lmadsen | damnit! |
| 13:36.08 | lmadsen | lol |
| 13:36.17 | [TK]D-Fender | boing! |
| 13:36.28 | lmadsen | ManxPower: that's right -- because a GoSub() is really just a Goto() that remembers where it came from |
| 13:36.54 | hsv-al | d-fender, i gave up with sendmail sending notices for asterisk |
| 13:36.59 | hsv-al | gonna use ssmtp instead |
| 13:37.08 | ManxPower | hugs postfix |
| 13:39.01 | dvdevel | i'm thinking of it in terms of a "hunt" macro, you pass it who to ring, where to go on busy, where to go on no answer - all situations would 'goto' out, and whether it's a "level" or not would affect how long my hunt lists could be. as i said, may never really be an issue in the real world. |
| 13:39.49 | lmadsen | dvdevel: I don't understand where a Macro() would call another Macro() in that situation |
| 13:40.03 | dvdevel | it wouldn't |
| 13:40.09 | ManxPower | dvdevel: the better way is for the macro to set a variable, then exit, the line after the Macro() would be a Goto() based in the info in the set variable |
| 13:40.23 | lmadsen | dvdevel: then you don't need to worry about nesting Macro()'s |
| 13:40.45 | ManxPower | jumping out of subroutines is just bad programming, even if you can do it. |
| 13:41.00 | dvdevel | yes, i know.... |
| 13:41.01 | lmadsen | it's referring to when you have a call to a Macro() inside of another Macro() |
| 13:41.11 | lmadsen | ManxPower: agreed |
| 13:41.35 | dominic1 | What do you think about e.164 username - schemes? like 4420154654564 |
| 13:41.35 | dvdevel | however, i was trying to minimize duplication of code. |
| 13:41.45 | *** join/#asterisk scampbell (n=scampbel@64-109-140-65.ded.ameritech.net) |
| 13:42.06 | dominic1 | will that not be positive in large scaled environments? |
| 13:42.13 | ManxPower | dominic1: no. |
| 13:42.39 | ManxPower | many phone numbers ring on multiple devices. We use the MAC address for the SIP userid |
| 13:43.25 | salmander | How would you guys rate handytone 286 vs 386 vs linksys pap 2 nt? |
| 13:43.35 | ManxPower | ~gs |
| 13:43.36 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:43.51 | ManxPower | salmander: and it's the PAP2-NA, not NT |
| 13:44.03 | salmander | you're right, my bad |
| 13:44.04 | dvdevel | another angle, i suppose, would to make heavy use of the database to keep my lists, and exten states, do the developers discourage that? |
| 13:44.05 | salmander | -NA |
| 13:44.26 | dominic1 | okay that's an option too. But I have the problem that I have to use the siemens deployment server for phone deployment and it's only possible to use numbers and hex will not be possible |
| 13:44.44 | salmander | ManxPower what's wrong with GS line? |
| 13:44.46 | ManxPower | dominic1: it sucks to be you |
| 13:45.44 | ManxPower | salmander: Their hardware is nothing special, but the firmware sucks, you basically just keep trying different versions of the GS firmware until you find one that does not crash when used in the ways you use it. |
| 13:45.50 | dominic1 | manx: you can't imagine :-D |
| 13:45.59 | ManxPower | GS has had YEARS to correct these issues and they have not done so. |
| 13:47.12 | ManxPower | ~phones |
| 13:47.13 | jbot | well, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 13:47.24 | salmander | I just need to test a few people connecting to a server and need the cheapest solution, and they seem to be the cheapest. If I get moeny for the project, then we could get better kit :( |
| 13:47.56 | ManxPower | salmander: We tried GS phones when we first started working with Asterisk. They worked so badly the entire project was almost canceled. |
| 13:48.07 | ManxPower | And ours is not an isolated experience |
| 13:48.14 | salmander | You have a point. |
| 13:48.34 | anonymouz666 | qualify= just doesn't work and UAc without keep-alive mechanism make things fun. :) |
| 13:48.34 | anonymouz666 | weee |
| 13:48.34 | ManxPower | salmander: Stick with Linksys/SIPura stuff if you want to go cheap. |
| 13:49.04 | ManxPower | go with polycom if you want business quality stuff. |
| 13:49.41 | salmander | I'm looking at voipdepot linksys pap2t-NA sells for $60 CAD can I get a better price somewhere? |
| 13:49.42 | ManxPower | salmander: Linksys/SIPura (linksys bought sipura) are mostly considered here to be low end business / mid range residential hardware. |
| 13:49.48 | salmander | (voipdepot.ca) |
| 13:49.59 | ManxPower | salmander: are you in Canada? |
| 13:50.03 | salmander | yes |
| 13:50.11 | *** join/#asterisk sack (n=sack@42.Red-83-44-211.dynamicIP.rima-tde.net) |
| 13:51.17 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 13:53.24 | gr0mit | ManxPower, i like the idea of mac addresses for sip ID |
| 13:53.56 | ManxPower | gr0mit: We also add a -a, -b, -c, etc for each line appearance. |
| 13:54.20 | ManxPower | most phones let you have all lines register as a single user, but we found that too limiting for how we wanted the calls to hunt between lines on the phone. |
| 13:54.38 | ManxPower | gr0mit: http://www.fnords.org/~eric/polycom-config-examples/ |
| 13:54.58 | salmander | So would anyone know a place that would offer better price than $60 on linksys ata? |
| 13:55.04 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-85-204.vif.net) |
| 13:55.32 | ManxPower | salmander: even with import duities, it may be cheaper to buy from an USA site, because of the USD/CAD exchange rate. |
| 13:56.05 | ManxPower | gr0mit: we use all hard phones, if you use a lot of softphones the usernames might be more complicated |
| 13:56.22 | salmander | usd is close to par and you pay roughly 1/6 of the price of what you import in duties :( |
| 13:57.03 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 14:00.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 14:02.20 | DavidR2008 | I had a very troubling experience this morning and was hoping there might be a master available who could help me shed some light on my problem: |
| 14:02.22 | DavidR2008 | I have two asterisk servers each with a Sangoma two port and four port card. One server is always a hot standby spare. We have two PRIs from the phone company that come in to asterisk and then calls are bridged to one three places: legacy telephony system (via T1), legacy PBX (via T1), new telephony system (via IAX) This morning I received an urgent call because we weren't getting calls in... |
| 14:02.23 | DavidR2008 | ...to our system. I started to troubleshoot and it looked like we were getting nothing on one of the PRIs, but the other one seemed to be fine. I called the telco ... long story short: I switched to the hot standby and everything started working again. The problems: no calls at all, calls that did come in were only able to hear, not talk (on the one T that seemed to be working) |
| 14:02.26 | DavidR2008 | My question: Any suggestions on what to start looking at on my server? It's Asterisk 1.4.20 running on CentOS 5.1 |
| 14:03.47 | ManxPower | DavidR2008: recent 1.4.x had a bug where it would randomly stop working. The bug was fixed in 1.4.21.1 |
| 14:03.47 | ManxPower | ~centos5bug |
| 14:03.58 | ManxPower | also notice that centos issue (minor) |
| 14:04.07 | ManxPower | I dunno what the jbot factoid name is |
| 14:04.50 | [TK]D-Fender | ManxPower: thats a compile issue |
| 14:04.57 | [TK]D-Fender | ~centos52bug |
| 14:04.58 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 14:05.12 | DavidR2008 | ManxPower: Thx, I thought the "randomly stop working" bug was introduced in 1.4.21 |
| 14:05.29 | x86 | hmm... now all of my users are saying that they are hearing a "robotic grinding" sound randomly during calls, sometimes even when they pick up their phone, and it only lasts for a second or two, but enough to make it impossible to hear a few words the customer is saying while in a conversation with them |
| 14:05.30 | DavidR2008 | which is my people said to stay away from it |
| 14:05.40 | DavidR2008 | *why |
| 14:05.47 | ManxPower | DavidR2008: It was at LEAST in 1.4.21, I don't know about others |
| 14:05.59 | ManxPower | ~ecfo |
| 14:06.00 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly realises its a crappy design based on a half baked 20 year old apps note. |
| 14:06.09 | ManxPower | x86: you don't have ECFO, do you? |
| 14:07.08 | ManxPower | I do NOT know if ECFO is an issue with the more recent EC available in Asterisk |
| 14:07.17 | x86 | don't think so... using analog stations to channel bank to T1 into asterisk... somewhere in that has to be the problem, as they can just go off-hook sometimes and experience it |
| 14:07.21 | *** join/#asterisk Bazy (n=bazy@79.114.18.115) |
| 14:07.31 | x86 | I use hardware EC with Sangoma cards though |
| 14:07.54 | ManxPower | x86: then you don't have ECFO. Are you SURE the Sangoma EC is enabled and working? |
| 14:08.05 | x86 | ManxPower: positive it's enabled... |
| 14:08.16 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:08.21 | x86 | we don't get any echo or feedback, just the "robotic grinding" noise |
| 14:08.41 | x86 | ManxPower: also, rxgain and txgain are default, not specified in zapata.conf anywhere |
| 14:09.16 | ManxPower | x86: "robotic grinding noise" is a unhelpful term. |
| 14:09.38 | x86 | well it kinda sounds like a squelch or something... |
| 14:09.49 | x86 | electronic distortion |
| 14:09.51 | x86 | it's not static |
| 14:10.25 | x86 | I'm running fxo_ks to my channel bank, and my channel bank only supports fxo_ls, would that be a problem? |
| 14:10.28 | ManxPower | ECFO is not static either, BTW, ignorant users just describe it that way 8-) If you have hardware EC, you don't have ECFO |
| 14:10.36 | x86 | I thought ks would work fine with ls equipment |
| 14:11.04 | x86 | should I change asterisk's signalling to ls? |
| 14:11.58 | x86 | I have errors on the interfaces, but they haven't increased in days |
| 14:12.49 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 14:14.24 | CVirus | Why isn't "zap show channels" command available in one of my asterisk installations ? |
| 14:14.40 | x86 | I wonder if I could record all of one of the users calls and get an example of the noise |
| 14:15.31 | ManxPower | CVirus: because you did not have zaptel installed when you installed Asterisk? |
| 14:15.47 | CVirus | it was working !! |
| 14:15.57 | CVirus | and suddenly disappeared |
| 14:15.59 | tzafrir_laptop | CVirus, because chan_zap failed to load? |
| 14:16.08 | tzafrir_laptop | module show like chan_zap |
| 14:16.09 | ManxPower | CVirus: when did you change? /etc/asterisk/modules.conf ? |
| 14:16.26 | ManxPower | CVirus: "core load chan_zap.so" |
| 14:16.29 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
| 14:16.33 | tzafrir_laptop | module unload chan_zap.so |
| 14:16.37 | tzafrir_laptop | module load chan_zap.so |
| 14:17.05 | ManxPower | I;m sure tzafrir_laptop is using the correct command for 1.4.x |
| 14:17.25 | CVirus | can I paste 3 lines ? |
| 14:17.29 | *** join/#asterisk arcanine (n=saxon_m2@202.138.162.67) |
| 14:17.39 | ManxPower | why not put it on pastebin? |
| 14:17.42 | CVirus | sure |
| 14:17.53 | CVirus | http://rafb.net/p/wu2Icb33.html |
| 14:17.53 | ManxPower | but yes, you should not be /kick'd for 3 lines |
| 14:18.20 | ManxPower | You broke your /etc/asterisk/zapata.conf [Jul 10 17:16:50] ERROR[3461]: chan_zap.c:10458 build_channels: Signalling must be specified before any channels are. |
| 14:18.49 | ManxPower | <allison>Naughty, naughty, naughty!</allison> |
| 14:18.54 | CVirus | let me paste it |
| 14:19.11 | ManxPower | no need to paste it, the error message tells you exactly what is wrong |
| 14:19.26 | CVirus | my zapata.conf is not broken I guess |
| 14:19.31 | CVirus | take a look please http://rafb.net/p/K4eir387.html |
| 14:20.02 | ManxPower | signaling = fxs_ks how many l's in signalling? |
| 14:20.16 | CVirus | my bad english |
| 14:20.18 | ManxPower | "Signalling must be specified before any channels are." |
| 14:20.37 | [TK]D-Fender | hiz speling iz perfekt! |
| 14:20.49 | CVirus | lol |
| 14:20.50 | CVirus | sorry |
| 14:20.50 | [TK]D-Fender | hukt on fonix werkt 4 u! |
| 14:21.17 | CVirus | come on |
| 14:24.10 | [TK]D-Fender | CVirus: s'ok, just roll with it :) |
| 14:24.33 | gr0mit | signalling has two l |
| 14:24.36 | tzafrir_laptop | both "signaling" and "signalling" should work. |
| 14:24.53 | gr0mit | unless you live in the USA |
| 14:24.54 | ManxPower | tzafrir_laptop: but one l does not work |
| 14:26.40 | x86 | ManxPower: once I get the user to call me back and tell me it did it again, I'll chomp the call recording so that you can hear what I'm talking about... perhaps you've got another word for it |
| 14:27.45 | *** join/#asterisk Peri (n=redanti@66.185.87.56) |
| 14:28.32 | ManxPower | x86: those sorts of things don't always show up in a recording |
| 14:28.55 | x86 | hmm |
| 14:28.57 | arcanine | wer can i do free testing of outbound calls for my server |
| 14:28.58 | x86 | I'm hoping it will |
| 14:29.14 | ManxPower | x86: it might, it depends on what it is. |
| 14:30.50 | gnorbert | What can be the problem, if Zoiper shows, that it's connected to a meetme conference, but the members of the conference don't hear it? |
| 14:31.17 | gnorbert | All the clients are zoipers and it's in a local network. |
| 14:31.43 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 14:32.27 | gnorbert | Asterisk shows also, that there's a new member of the conversation. |
| 14:32.52 | x86 | wtf is zoiper? |
| 14:32.55 | ManxPower | codec issue, transcoding issue, packet filtering filrewall issue, nat issue |
| 14:33.10 | coppice | zoiper is a very buggy softphone |
| 14:33.12 | ManxPower | (especially if the Asterisk server is doing NAT, the packets could be using the wrong interface) |
| 14:33.14 | gnorbert | Zoiper is a softphone... |
| 14:33.16 | hsv-al | x86 |
| 14:33.20 | hsv-al | i think its an iax soft client |
| 14:33.24 | hsv-al | standalone binary |
| 14:33.28 | coppice | a softphone with T.38 |
| 14:33.36 | x86 | x-lite ftw |
| 14:33.46 | gnorbert | (I tried it also with Linphone and ekiga.) |
| 14:33.51 | hsv-al | ekiga is gimp |
| 14:34.26 | hsv-al | its the rusty ford pinto that sits next to an SL55 AMG(xlite) |
| 14:34.31 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:34.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:34.34 | gnorbert | Some noise can be heard, but not the speak. |
| 14:34.38 | Peri | hsv-al i disagree, you can't do any sort of image manipulation in ekiga ;P |
| 14:35.30 | hsv-al | ekiga is just clumsy looking |
| 14:35.32 | DavidR2008 | is upgrading asterisk as simple as downloading the lastest source and ./configure, make, make install? |
| 14:35.35 | hsv-al | no aesthetics |
| 14:36.06 | hsv-al | I just found out the joys of compiz-cube today |
| 14:36.18 | hsv-al | im now rotating a 15 sided orthagonal cube for my desktop :) |
| 14:36.39 | hsv-al | each plane is a DT :) |
| 14:37.15 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 14:38.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:38.57 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:39.02 | Dr-Linux|home | i'm using 2 digium 4 port FXO cards, when i dial to PSTN, there are lot of noise, but if i dial through 2nd card, all works just fine |
| 14:39.08 | Dr-Linux|home | what should be checked? |
| 14:39.45 | M1s3ry | Dr-Linux|home, run fxotune on the card. change modules around, and change the lines around too... |
| 14:39.54 | M1s3ry | if none works call digium support |
| 14:39.57 | *** join/#asterisk huey23 (n=yea@65.111.241.212) |
| 14:40.36 | ManxPower | Dr-Linux|home: any IRQ sharing happening? |
| 14:40.51 | ManxPower | cat /proc/interrupts |
| 14:40.52 | x86 | hsv-al: I'm still running Beryl myself... it can only do a 6-sided cube (while only 4 sides are actually usable) |
| 14:41.06 | Dr-Linux|home | M1s3ry: well, i've no backup server, and all is up, however if i dial-in from the same line it works just fine |
| 14:41.11 | x86 | hsv-al: I've gotta upgrade to compiz-fusion one of these days ;) |
| 14:41.47 | Dr-Linux|home | M1s3ry: i unplugged the lines and checked with an analog phone, lines are just fine .. looks like something went wrong with FXO card/ports |
| 14:41.53 | kippi | I have been looking at loading sip.conf from a database, I have setup the tables etc, it is working fine for the dialplan, I have this setup so far: http://pastebin.ca/1068030 could someone point me in the right direction? |
| 14:42.05 | ManxPower | kippi: Not I. |
| 14:42.15 | hsv-al | you can have up to a 60 plane/face object |
| 14:42.19 | ManxPower | Databases are just used for the Coolness Factor unless you have a zillion users. |
| 14:42.20 | hsv-al | dont know if there is a mathematical term for it |
| 14:42.31 | M1s3ry | Dr-Linux|home, have you gone with ManxPower's suggestion? |
| 14:42.39 | lmadsen | kippi: usually it has something to do with the tables either be setup wrong, or no data in the table. You haven't really provided anything that we can debug from |
| 14:42.44 | Dr-Linux|home | ManxPower: same car/machine setup was just working fine since 2 years |
| 14:43.04 | Dr-Linux|home | ManxPower: IRQ's can be shared automatically? |
| 14:43.19 | ManxPower | Dr-Linux|home: they are always shared automatically |
| 14:43.24 | ManxPower | now, answer my question |
| 14:43.24 | Kobaz | ManxPower: or want to do dynamic updates from the outside: (ie: web interface for voicemail) |
| 14:43.39 | ManxPower | Kobaz: a database is not required for that |
| 14:43.47 | Kobaz | not required but makes it a hell of alot easier |
| 14:43.58 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:44.00 | Kobaz | you dont have to regenerate a config file and merge in differences |
| 14:44.01 | DavidR2008 | *ping* is upgrading asterisk as simple as downloading the lastest source and ./configure, make, make install? |
| 14:44.06 | Dr-Linux|home | ManxPower: here: http://phpfi.com/330885 |
| 14:44.17 | ManxPower | DavidR2008: Yes. No. |
| 14:44.42 | Dr-Linux|home | i've 2 cards installed, 4 port each |
| 14:45.10 | ManxPower | Dr-Linux|home: you're not sharing IRQs. Is this a real 2-CPU system, or is that a Hyperthreading CPU? |
| 14:45.18 | [TK]D-Fender | DavidR2008: Withing the same major version, typically yes. |
| 14:45.24 | ManxPower | Dr-Linux|home: did you try rebooting the system? |
| 14:45.52 | Dr-Linux|home | ManxPower: hhm.. not sure with your first question |
| 14:45.52 | ManxPower | In my experience Analog cards just stop working occasionally until a reboot happens. That's why we don't use analog cards anymore and have removed them from all of our server. |
| 14:46.17 | Kobaz | external gateways are always nice |
| 14:46.23 | ManxPower | Dr-Linux|home: the /proc/interrupts info shows you have two CPUs in the system (i.e. SMP). Is this really true? |
| 14:46.29 | Kobaz | you dont have to bring down the server to add lines |
| 14:46.34 | Dr-Linux|home | ManxPower: rebooting will be difficult this time :S |
| 14:46.42 | ManxPower | Dr-Linux|home: it always is. |
| 14:46.43 | Dr-Linux|home | lemme see |
| 14:46.57 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-68-151.vif.net) |
| 14:46.57 | ManxPower | If you have to reboot, I recommend a power cycle |
| 14:47.31 | Dr-Linux|home | ManxPower: yeah, two CPU's |
| 14:47.59 | ManxPower | OK. Good. Asterisk has in the past had major issues with Hyperthreading being enabled. |
| 14:48.01 | Dr-Linux|home | hhm.. |
| 14:48.17 | ManxPower | Hyperthreading is a way to "fake" 2 CPUs using only one actual CPU. |
| 14:49.16 | Dr-Linux|home | ManxPower: i'm using Dell powerage 840 |
| 14:49.23 | JerJer | Asterisk used to have a problem with HT on dual processor sitautions |
| 14:51.58 | Dr-Linux|home | before resarting what possible things i can verify .. |
| 14:52.07 | Dr-Linux|home | fxotune .. what does it show |
| 14:52.24 | arcanine | hello guys |
| 14:52.40 | arcanine | what is the diff of g729 and g711 codec? |
| 14:52.44 | Dr-Linux|home | M1s3ry: how can i try fxotune ? |
| 14:53.13 | ManxPower | Dr-Linux|home: when you solve the issue, you might consider installing the irqbalance utility to better use your 2-CPUs |
| 14:53.26 | ManxPower | Dr-Linux|home: fxotune is for helping with echo |
| 14:53.40 | M1s3ry | ManxPower, it's not just for that |
| 14:54.00 | Kobaz | [Jul 10 10:52:56] WARNING[12202]: chan_zap.c:6410 ss_thread: CallerID feed failed: Success |
| 14:54.03 | Kobaz | [Jul 10 10:52:56] WARNING[12202]: chan_zap.c:6510 ss_thread: CallerID returned with error on channel 'Zap/1-1' |
| 14:54.07 | Kobaz | so hmm |
| 14:54.14 | M1s3ry | ManxPower, actually It's not specifically for that, even though in switchvox it says so :p |
| 14:54.16 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:54.17 | ManxPower | Kobaz: if this happens often play with your rxgain |
| 14:54.19 | Kobaz | we're not getting callerid on this particular line |
| 14:54.24 | Kobaz | ah |
| 14:54.43 | M1s3ry | Dr-Linux|home, stop asterisk, then run "fxotune -i 8" to see if that helps any |
| 14:54.49 | ManxPower | Kobaz: remember Caller*ID (in USA/Canada) is a 1200 baud FSK modem tone burst. |
| 14:55.52 | Dr-Linux|home | hhm.. |
| 14:56.00 | Dr-Linux|home | M1s3ry: all lines like: |
| 14:56.05 | Dr-Linux|home | /dev/zap/10 absent: No such file or directory |
| 14:56.15 | M1s3ry | did you stop asterisk? |
| 14:57.06 | Dr-Linux|home | M1s3ry: hhm... 10 calls are bridged right now ... so i didn't |
| 14:57.28 | M1s3ry | stop gracefully, and then run fxotune -i 8 |
| 14:58.00 | Dr-Linux|home | stopped |
| 14:58.01 | Dr-Linux|home | -bash-3.1# fxotune -i 8 |
| 14:58.02 | Dr-Linux|home | Tuning module /dev/zap/1 |
| 14:58.13 | Dr-Linux|home | should i wait? |
| 14:58.14 | jaytee | if my extension is 300 and I've got a context called 300 in extensions.conf and my sip phone has the context=300 set in sip.conf isn't that where calls I make will enter the dialplan? |
| 14:58.14 | ManxPower | Dr-Linux|home: the reason M1s3ry told you to stop asterisk is because you can't run fxotune when Asterisk is running |
| 14:58.20 | M1s3ry | there ya go... this will take a few minutes |
| 14:58.22 | ManxPower | Dr-Linux|home: oh, this could take 10 mins or more. |
| 14:58.26 | anonymouz666 | no |
| 14:58.30 | anonymouz666 | less than that |
| 14:58.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:58.39 | anonymouz666 | if you use the latest version of fxotune |
| 14:58.39 | ManxPower | depends on the number of ports. |
| 14:58.40 | anonymouz666 | like zaptel 1.4. |
| 14:58.47 | M1s3ry | meh... for 8 lines I would think about 5 mins or so |
| 14:58.49 | Dr-Linux|home | Tuning module /dev/zap/1 |
| 14:58.49 | Dr-Linux|home | Done! |
| 14:59.05 | M1s3ry | Dr-Linux|home, then it should run on the other ports |
| 14:59.23 | Dr-Linux|home | M1s3ry: what it does BTW? |
| 14:59.33 | Dr-Linux|home | how does it tune the ports? |
| 15:00.18 | jasonwoot | hi sports fans |
| 15:00.18 | M1s3ry | Dr-Linux|home, it's tuning the lines... specifically what is it doing, and how is it doing this? hehe... I'm not sure. I just know to use it in cases like this |
| 15:00.20 | CVirus | Why would a call go to the default context instead of the context I defined it for the specific channel inside my zapata.conf ? |
| 15:00.43 | Dr-Linux|home | M1s3ry: :) |
| 15:00.46 | CVirus | http://rafb.net/p/u5IFrl45.html |
| 15:01.26 | jasonwoot | any suggestions when faxdetect=incoming isn't working? |
| 15:01.50 | jasonwoot | in 2 years, never tried faxing with asterisk... |
| 15:02.09 | Dr-Linux|home | M1s3ry: all done |
| 15:02.14 | Dr-Linux|home | should i start asterisk? |
| 15:02.22 | M1s3ry | Dr-Linux|home, yes, and test a call |
| 15:02.27 | M1s3ry | wait |
| 15:02.36 | M1s3ry | restart zaptel, then start * |
| 15:03.10 | M1s3ry | Dr-Linux|home, do you have an echo cancellation module on board as well? |
| 15:03.11 | ManxPower | jasonwoot: usually your idea of incoming is wrong. Remember once you set an option it is applied to all the following channels unless unset. |
| 15:03.34 | ManxPower | so if you have ports 1-4 as telco lines and ports 5-8 as phones..... |
| 15:03.40 | Dr-Linux|home | M1s3ry: no, i'm using TDM04B |
| 15:04.00 | Dr-Linux|home | ManxPower: all telco lines .. i mean all fxo lines |
| 15:04.15 | Dr-Linux|home | 5-8 are just working fine |
| 15:04.30 | ManxPower | Dr-Linux|home: I was talking to jasonwoot |
| 15:04.37 | Dr-Linux|home | i mean the 2nd card is working fine for dial-in/dial-out |
| 15:04.42 | Dr-Linux|home | oppsss okey sorry |
| 15:05.00 | ManxPower | jasonwoot: we use nvfaxdetect, then we can enable fax detection on the fly in the dialplan. |
| 15:05.33 | M1s3ry | Dr-Linux|home, how has the test call gone? |
| 15:05.56 | Dr-Linux|home | no luck, dialin is okay, but still dial-out has noise |
| 15:05.56 | [TK]D-Fender | CVirus: Pastebin your extensions.conf |
| 15:06.21 | flokuehn | i have the problem that if call waiting disabled callers dont get an busy signal. first they get the normal signal and then fly out of line. any hints ? |
| 15:06.36 | M1s3ry | Dr-Linux|home, call Digium support. we'll test it |
| 15:06.43 | ManxPower | flokuehn: without information on your phones, dialplan design, etc we can't help you. |
| 15:06.44 | DavidR2008 | [TK]D-fender: I'd be going from 1.4.20 to 1.4.21.1 |
| 15:06.57 | Dr-Linux|home | M1s3ry: what's the number for them? |
| 15:06.57 | [TK]D-Fender | DavidR2008: then no issu |
| 15:07.22 | CVirus | [TK]D-Fender: http://rafb.net/p/aUxY3d11.html |
| 15:07.33 | flokuehn | ManxPower: k. the telephone we are using siemens gigaset 475 ip. |
| 15:08.02 | [TK]D-Fender | CVirus: And your zapata.conf |
| 15:08.18 | flokuehn | ManxPower: dialplan follows |
| 15:08.45 | CVirus | [TK]D-Fender: http://rafb.net/p/fYxCnp78.html |
| 15:09.01 | [TK]D-Fender | CVirus: And now the fun part.... users.conf |
| 15:09.43 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:10.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:10.35 | CVirus | [TK]D-Fender: looks like asteriskNOW altered this file .. I never touched it though http://rafb.net/p/YnBZIJ26.html |
| 15:11.18 | [TK]D-Fender | CVirus: zapchan = 1 <- 94+ |
| 15:11.22 | rwaite | is it bad to have multiple cdr modules loaded? |
| 15:11.31 | CVirus | [TK]D-Fender: my zapata.conf and extensions.conf are correct ? |
| 15:11.40 | [TK]D-Fender | CVirus: to summaraize : usres.config = fresh steamy pile of SHIT. |
| 15:11.46 | [TK]D-Fender | users.conf* |
| 15:11.50 | CVirus | i see |
| 15:11.55 | [TK]D-Fender | CVirus: its overriding your zapata |
| 15:11.56 | jaytee | I so just want to walk away from all this crap |
| 15:13.02 | ManxPower | CVirus: almost nobody here uses users.conf. |
| 15:13.22 | CVirus | [TK]D-Fender: thanks alot .. works now |
| 15:14.27 | CVirus | it works but the echo test is not replying back |
| 15:14.34 | CVirus | it's not echo'ing back I mean |
| 15:18.58 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:19.09 | *** join/#asterisk funkyhippy (n=funkyhip@78-32-92-91.no-dns-yet.enta.net) |
| 15:19.52 | funkyhippy | Hi does anyone know where asterisk logs cdr-odbc messages? |
| 15:20.33 | UnixDog | ./var/log/asterisk |
| 15:21.00 | UnixDog | should be in the full log |
| 15:25.31 | funkyhippy | I've found a mention in /var/log/asterisk/messages just says Unable to connect to datasource |
| 15:25.42 | funkyhippy | but isql and tsql both work fine |
| 15:25.53 | funkyhippy | anyone got any ideas what else it could be? |
| 15:25.59 | bkw_ | where did you define your dsn? |
| 15:26.20 | funkyhippy | using the odbcinst command and a template file |
| 15:26.42 | bkw_ | did you know if it put it in your home directory or in the global files in /etc? |
| 15:27.08 | bkw_ | and I can guess Asterisk is running as a different user than you installed the templates with ? |
| 15:27.39 | funkyhippy | ah yeah good point didn't think of that |
| 15:27.58 | bkw_ | wrote cdr_odbc so he knows a few things here and there bout the sillyness you can run into |
| 15:28.19 | funkyhippy | if I put it in /etc/odbc.ini will asterisk pick it up? |
| 15:28.24 | bkw_ | yes |
| 15:28.28 | bkw_ | those are system wide |
| 15:30.29 | flokuehn | ManxPower: because iam new and learning. for what do you need the dial plan > |
| 15:30.31 | flokuehn | ? |
| 15:31.24 | funkyhippy | bkw_ put the settings into /etc/odbc.ini but still gettin uable to connect to datasource |
| 15:31.31 | *** join/#asterisk stansmith (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:31.43 | bkw_ | are you sure its /etc/odbc.ini and not /etc/odbc/odbc.ini? |
| 15:31.55 | bkw_ | also make sure all your odbcinst.ini files are updated too |
| 15:32.09 | funkyhippy | ok doing it now |
| 15:32.14 | *** part/#asterisk stansmith (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:33.15 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:33.28 | mgroman | Hi, what is the difference between a TDM411B and a TDM411E ? |
| 15:33.35 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 15:34.20 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 15:34.34 | funkyhippy | bkw_ I didn't have a /etc/odbc/odbc.ini so I've ln -s it to /etc/odbc.ini also the same for /etc/odbcinst.ini still the same :( |
| 15:34.41 | *** join/#asterisk J4zen (n=Jeroen@213-84-194-178.adsl.xs4all.nl) |
| 15:34.48 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:34.51 | bkw_ | wierd |
| 15:34.58 | bkw_ | strace it and see where its opening the files |
| 15:34.59 | funkyhippy | do i need to restart asterisk or will a reload be ok? |
| 15:35.12 | Kobaz | you dont have to do anything to asterisk to update odbc.ini |
| 15:36.21 | funkyhippy | I'm having to use a domain user account to connect to mssql so username is DOMAIN\username could that be a problem? |
| 15:38.25 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-19-198.revip2.asianet.co.th) |
| 15:39.15 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 15:39.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:40.00 | [TK]D-Fender | flokuehn: pastebin the complete CLI output of a failed call at verbose 10. |
| 15:40.02 | [TK]D-Fender | ~pb |
| 15:40.03 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:40.04 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:40.05 | Peri | quick question. Anyone ever set up asterisk to create a user and voicemail etc on first call in to the voicemail service? |
| 15:40.24 | [TK]D-Fender | Peri: .... HUH? |
| 15:41.02 | Kobaz | [TK]D-Fender: user doesn't exist in voicemail... dial the voicemail feature code and asterisk will write out a new user entry in voicemail, and then send the user to that voicemail |
| 15:41.03 | rwaite | Peri: Your ideas intrigue me and I would like to subscribe to your newsletter |
| 15:41.09 | mog | you want to create users based on them calling into voicemail? |
| 15:41.35 | [TK]D-Fender | Peri: * doesn't configure itself. You can code this yourself however |
| 15:41.49 | Kobaz | yeah |
| 15:41.52 | mog | ya i would think what he wants is possible just a bit tricky |
| 15:41.57 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 15:42.06 | flokuehn | mom |
| 15:42.06 | mog | you could do it all in dialplan |
| 15:43.33 | flokuehn | [TK]D-Fender: http://rafb.net/p/v6ikev33.html |
| 15:44.25 | flokuehn | [TK]D-Fender: what i found out is that the porblem occures together with the call waiting option. this is the output if disabled. |
| 15:44.45 | *** join/#asterisk routerweasel (n=4stfed@core.spokanecomputing.com) |
| 15:45.46 | [TK]D-Fender | flokuehn: FreePBX is NOT supported here. Go ask in their channel. |
| 15:45.56 | flokuehn | k |
| 15:46.04 | flokuehn | thank you anyway |
| 15:48.22 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 15:48.44 | mgroman | Hi, what is the difference between a TDM411B and a TDM411E ? |
| 15:48.56 | M1s3ry | mgroman, echo cancellation module |
| 15:49.14 | Peri | [TK]D-Fender: oh i know i'd have to code it myself, just wondering if anyone had done anything of the sort before |
| 15:49.24 | Kobaz | peri: i'm sure they have |
| 15:50.02 | Peri | Kobaz: just an idle thought to make this demo platform a little smoother, i'll look into it i guess |
| 15:50.11 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 15:50.19 | [TK]D-Fender | Peri: it doesn't sound common to have "extensions" decide for themselves that they are entitled to VM and that its configured jsut by trying to use it. |
| 15:50.23 | Kobaz | probably the easiest way is to launch an external script via agi |
| 15:51.11 | [TK]D-Fender | Peri: Sounds more like you should set up the box itself when you configure the user, and just not utilize it until they go to log in for the first time. That would very basic dialplan work. |
| 15:51.17 | mgroman | M1s3ry: thanks.. i assume the E stands for echo cancellation |
| 15:51.30 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-151-207.neo.res.rr.com) |
| 15:51.33 | M1s3ry | mgroman, np |
| 15:51.38 | Peri | Kobaz: that's what i was thinking. [TK]D-Fender: Oh i know it's not common, believe me :) Normally I'd just have the * use the user db on whatever switch i'm connecting to, but the switch i'm connecting to doesn't actually allow external connections to it's user db except by specific "safe" products |
| 15:52.30 | Peri | [TK]D-Fender: Yeah, i'll just provision all the mailboxes to work with the DIDs and have it ready whether they use it or not |
| 15:52.49 | Peri | it is just a demo server, so it's only 24 DIDs to take care of |
| 15:53.31 | [TK]D-Fender | hands Peri an elephant gun and a box of ammo |
| 15:53.48 | Peri | lol |
| 15:54.46 | *** join/#asterisk BBHoss_ (n=hoss@68.159.148.214) |
| 15:55.20 | *** part/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 15:55.28 | funkyhippy | quit |
| 15:55.37 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:57.34 | wwalker | which module must be loaded for call files to be picked up? |
| 15:57.45 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:58.31 | russellb | wwalker: pbx_spool |
| 15:59.28 | wwalker | russellb: thank you |
| 15:59.34 | russellb | np |
| 16:00.15 | [TK]D-Fender | russellb: darn, beat me too it. |
| 16:00.30 | [TK]D-Fender | russellb: I jsut drilled the source for the 30 sec it was worth :) |
| 16:01.13 | *** join/#asterisk Dovid (n=Dovid@77.127.126.11) |
| 16:01.24 | Dovid | hi TKL |
| 16:01.25 | Dovid | TK: |
| 16:01.53 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:02.30 | russellb | :) |
| 16:03.07 | Dovid | i have a box with ztdummy. on boot up i see on the screen mpg123: no process killed. I have never installed mpg123 on the box and I am using for musiconhold mode=files |
| 16:03.20 | Dovid | so ti should not want mpg123 at all. so y is asterisk kvetching ? |
| 16:04.18 | [TK]D-Fender | Dovid: pastebin <---- |
| 16:04.45 | Dovid | pb of musiconhold.conf ? |
| 16:04.58 | [TK]D-Fender | Dovid: Everything incriminating of course. |
| 16:06.07 | jasonwoot | :^( VP of Operations and Call Center Director just got back from a call center technology convention... |
| 16:06.26 | jasonwoot | has his tit in a wringer |
| 16:06.48 | [TK]D-Fender | jasonwoot: s'ok... not much juice in there to sqeeze out anyways :) |
| 16:07.08 | Dovid | TK: http://pastebin.ca/1068199 |
| 16:07.20 | coppice | I assume "call center technology convention" == "sleazy sales pitch" |
| 16:07.40 | [TK]D-Fender | Dovid: So you just changed this to "files" and did a "reload"? |
| 16:08.57 | Dovid | TK: I just did reboot on the box. same issue happend b4. also turned on full logging |
| 16:09.06 | DavidR2008 | does anyone have experience using hardware T1 failover equipment? (also known as: A/B switches) |
| 16:09.44 | DavidR2008 | BTW the in-place upgrade worked like a charm |
| 16:09.45 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:10.02 | russellb | DavidR2008: of asterisk? |
| 16:10.10 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 16:10.23 | browser | davevg-btwtech: If you refer to TDMoE solutions, yes I have experience but only bad ones. |
| 16:10.24 | DavidR2008 | yes, I had asked a question about it earlier today |
| 16:10.25 | russellb | yay for something working properly |
| 16:10.55 | browser | DavidR2008: If you refer to TDMoE solutions, yes I have experience but only bad ones. |
| 16:11.12 | DavidR2008 | browser: no not TDMoE, hardware T1 switches: basically an A/B switch for a T1 port |
| 16:11.38 | Dovid | TK: I rebuilt zaptel b4 the last reboot and now it works |
| 16:11.41 | Dovid | wierd |
| 16:11.51 | Dovid | cause zaptel & ztdummy were working fine |
| 16:12.26 | browser | DavidR2008: sorry the question. what do you understand as an A/B switch? |
| 16:12.44 | Qwell | a switch that switches between A and B.. |
| 16:12.50 | Qwell | it can only mean one thing :) |
| 16:12.51 | *** part/#asterisk flokuehn (i=flokuehn@globalways/developer/flokuehn) |
| 16:13.06 | DavidR2008 | I'm looking at Bero*fos, FailSafe FSV-4PFS & Dataprobe T-APS; just wanted to know if anyone has used them before |
| 16:13.58 | DavidR2008 | basically it has three T1 ports on it, you put your Telco connection in the main, plug your main asterisk box in to A and your backup * box in to B |
| 16:14.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:14.41 | DavidR2008 | something goes wrong with the main box, you can manually or automatically have it switch the T1 to port B and now your backup * box has the T1 connection |
| 16:14.57 | DavidR2008 | it's a level 1 switch so no protocols to worry about |
| 16:15.26 | Kobaz | DavidR2008: we use an A/B switch from WTI |
| 16:15.33 | Kobaz | has a nifty telnet interface and everything |
| 16:15.34 | Nugget | telnet is eeeeeeevil! |
| 16:15.35 | browser | DavidR2008: no, I can definitely not help you. We have solved it with simple LCR retries for calls we send to the Asterisk PBX but we use two PRI ports on the AXE. |
| 16:16.09 | DavidR2008 | Sorry forgot to mention WTI. |
| 16:16.18 | browser | DavidR2008: But have a look at a vendor called Patapsco. They manufacture such Port Multiplexers with some routing built-in |
| 16:16.21 | DavidR2008 | I emailed them for a quote, so |
| 16:16.58 | Kobaz | DavidR2008: what the hell, they cost $1? |
| 16:17.28 | DavidR2008 | waiting 'till I get a response before including WTI in the options |
| 16:17.50 | Kobaz | DavidR2008: |
| 16:17.52 | DavidR2008 | Kobaz: uh, yea, I'm not sure I trust the FailSafe line |
| 16:18.03 | Kobaz | DavidR2008: oh, i added it to my cart and it's 2700 |
| 16:18.13 | Kobaz | DavidR2008: the PLS-345 from WTI is around $700 |
| 16:18.21 | Kobaz | DavidR2008: and you dont need some silly usb connection |
| 16:18.46 | DavidR2008 | the other one's $399.99 |
| 16:18.55 | DavidR2008 | which seems kinda cheap |
| 16:21.08 | browser | DavidR2008: The product I had evaluated was called "Liberator D". It's a small switching core engine which allows you to switch among 4 PRI ports with good failover rules to activate and desactivate the port. I don't remember if they do SS7 on the telco side put HDLC for sure. |
| 16:23.38 | CVirus | Asterisk doesn't want to accept my Authenticate() password ... what could be wrong ? |
| 16:23.39 | DavidR2008 | Kobaz: the PLS-345 is perfect for my app, thx! $700 is a good price too |
| 16:23.54 | Qwell | CVirus: wrong password |
| 16:24.11 | CVirus | i'm using a mobile phone |
| 16:24.18 | CVirus | does it make any difference ? |
| 16:24.27 | CVirus | it's an iPhone to be more precise |
| 16:24.35 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 16:24.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:24.45 | CVirus | exten => s,n,Authenticate(1235) |
| 16:25.38 | CVirus | looks like ... it's not seeing any of the numbers I dial in |
| 16:26.04 | *** part/#asterisk browser (n=afg_ch@213.189.154.108) |
| 16:26.26 | [TK]D-Fender | CVirus: Yay, go configure a DTMF mode for it properly |
| 16:28.33 | CVirus | I don't get it .. for the phone you mean ? |
| 16:28.38 | CVirus | it's not sending proper DTMF tones ? |
| 16:29.06 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:29.32 | *** join/#asterisk Firass-z0r (n=Firass@ead224-222.housing.wwu.edu) |
| 16:29.49 | [TK]D-Fender | CVirus: If its sending any of it, you don't have * configured to accept DTMF in that manner |
| 16:30.20 | CVirus | I see |
| 16:31.10 | Qwell | how is your iPhone connecting to you Asterisk box? |
| 16:32.05 | CVirus | I'm just using it to call another GSM line that is connected to asterisk through an FXO port |
| 16:32.19 | CVirus | the sim card is plugged in some vodafone devices and the device is connected to my FXO port |
| 16:32.27 | CVirus | and I use the iPhone to call that sim card |
| 16:32.36 | Qwell | and the FXO port is...? |
| 16:32.55 | CVirus | X100P stupid-crap-sob-card-i-know |
| 16:34.01 | Kyoshi | does TDS and asterisk play nice together for realtime or will I need to use unixODBC? |
| 16:34.09 | Kyoshi | lemme rephrase |
| 16:34.22 | Kyoshi | does TDS and asterisk play nice together for realtime using MS SQL or will I need to use unixODBC? |
| 16:34.58 | Qwell | CVirus: well, there's your problem |
| 16:35.08 | CVirus | Qwell: the card ? |
| 16:35.57 | CVirus | let me test from an analog phone ... I'm suspecting it's the iPhone that is causing the problem |
| 16:35.59 | Qwell | iPhone to a crappy GSM box, do a crappy FXO card... |
| 16:36.04 | Qwell | to* |
| 16:38.53 | *** join/#asterisk coppice (n=chatzill@122.193.17.210.dyn.pacific.net.hk) |
| 16:40.10 | outtolunc | starts whistling the melody of 'motorcycles and sip' <G> |
| 16:41.58 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 16:43.49 | CVirus | I tried with another mobile phone and the problem is still there |
| 16:45.14 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 16:48.06 | [TK]D-Fender | CVirus: Go test your X100 on a real line. |
| 16:53.00 | CVirus | [TK]D-Fender: thanks for the suggestion |
| 16:55.36 | CVirus | this is the GSM box I'm using if it makes any difference http://www.huawei.com/policy/simpleres.do?id=1063&type=abouthw |
| 16:56.18 | CVirus | isn't there any other way I could authenticate a user ? |
| 16:57.02 | *** join/#asterisk joel_oliveira (n=asdas@estrela-adm.nortenet.pt) |
| 16:57.24 | [TK]D-Fender | CVirus: Stop barking up random trees and validate your card |
| 16:57.45 | joel_oliveira | ~cetos52bug |
| 16:57.46 | CVirus | okay okay .. will do |
| 16:57.51 | joel_oliveira | ~centos52bug |
| 16:57.52 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 16:58.41 | joel_oliveira | hi all! can I ask asterisk guru queue statistics questionS? |
| 17:00.01 | joel_oliveira | or can I pvt someone to ask this kind of questions? It's just that I'm getting some trouble with that application and the forums aren't a good help to my problem :\ |
| 17:04.38 | *** part/#asterisk iobug (n=gustav@212.124.71.46) |
| 17:05.07 | Qwell | joel_oliveira: it's unlikely that you'll get much help with that here |
| 17:05.41 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 17:11.31 | caio1982 | joel_oliveira: you might want to try asking it at #asterisk-br (we speak portuguese there, though I cannot guarantee that someone will help you immediately) |
| 17:11.49 | jaytee | I'm trying to figure out a way to block specific phones from doing an external transfer or external forward without blocking their ability to make outgoing calls. Has anyone here accomplished that? |
| 17:12.15 | *** join/#asterisk CVirus (n=Burzum@196.218.41.31) |
| 17:13.27 | [TK]D-Fender | jaytee: Go see if ${BLINDTRANSFER} is set or if there is a SIP header to ID the nature of the call. |
| 17:15.28 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 17:20.16 | joel_oliveira | caio1982 : thank you |
| 17:20.21 | *** part/#asterisk joel_oliveira (n=asdas@estrela-adm.nortenet.pt) |
| 17:21.28 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:26.09 | *** join/#asterisk allankardec (n=root@189.104.17.52) |
| 17:28.54 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-68-151.vif.net) |
| 17:30.55 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:31.59 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
| 17:32.10 | Wayhigh | re all.. long time no see |
| 17:32.57 | *** join/#asterisk exvito (n=exvito@mail.colours.pt) |
| 17:33.04 | Wayhigh | what's a good place that can ship me a tdm411 fast and wont gouge me for it? |
| 17:33.14 | exvito | ~centos52bug |
| 17:33.14 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 17:34.51 | exvito | i'm using the setvar "field" for sip friends so as to set a channel variable for specific phones... is there any equivalent way of doing it for zap channels ? |
| 17:36.17 | *** join/#asterisk notjohn (n=notjohn@216.68.73.132) |
| 17:36.31 | Kobaz | does rxgain and txgain get reloaded if you do a module reload chan_zap |
| 17:36.39 | Kobaz | or do you have to do a full restart |
| 17:39.52 | [TK]D-Fender | Wayhigh: www.telephonydepot.com. |
| 17:39.59 | *** join/#asterisk gramulhaozin (n=charles@c-65-34-131-58.hsd1.fl.comcast.net) |
| 17:40.00 | gramulhaozin | hey |
| 17:40.04 | [TK]D-Fender | Wayhigh: do make sure its the one based on the TDM410P base |
| 17:40.08 | gramulhaozin | Anyone there is using vonage with Asterisk ? |
| 17:40.14 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 17:40.21 | Wayhigh | fender: will do, thanks! |
| 17:40.23 | gramulhaozin | The Vonage Business Plus is not available anymore looks like |
| 17:40.40 | [TK]D-Fender | exvito: Have you tried? |
| 17:40.41 | Kobaz | gramulhaozin: i was using my regular vonage residential with asterisk |
| 17:40.55 | [TK]D-Fender | Kobaz: Just reloading the module should do |
| 17:40.55 | gramulhaozin | Kobaz: with a softphone ? |
| 17:41.03 | Kobaz | gramulhaozin: with an fxo card :P |
| 17:41.05 | gramulhaozin | Kobaz: I'm trying to set it up for a business with multiple incomming channels |
| 17:41.21 | Kobaz | gramulhaozin: use someone other than vonage and get better rates |
| 17:41.26 | gramulhaozin | Kobaz: no dude, I wanna do the right way. |
| 17:41.31 | Wayhigh | fender: how're the rhino cards these days? |
| 17:41.35 | exvito | [TK]D-Fender: ...what do you mean ? Like inserting a line setvar=MYVAR=VALUE into zapata.conf ? |
| 17:41.35 | Kobaz | Wayhigh: shitty |
| 17:41.37 | gramulhaozin | Kobaz: vonage is reliable |
| 17:41.46 | gramulhaozin | Kobaz: to the point a business can trust |
| 17:42.07 | Kobaz | gramulhaozin: how many lines? |
| 17:42.57 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 17:43.04 | Kobaz | Wayhigh: we have a brand new rcbfxx card and the volume of the audio that asterisk gets keeps going up and down randomly |
| 17:43.10 | Kobaz | Wayhigh: the drivers suck for the t1 cards |
| 17:43.11 | [TK]D-Fender | exvito: yes |
| 17:43.20 | Kobaz | Wayhigh: the drivers suck for the analog cards |
| 17:43.26 | Wayhigh | thanks for the info |
| 17:43.28 | exvito | [TK]D-Fender: honestly no... I'll give it a shot ! :) |
| 17:43.33 | Wayhigh | I'll stick with the tdm410's |
| 17:43.35 | Kobaz | heh |
| 17:43.37 | Kobaz | sangoma! |
| 17:43.52 | Kobaz | the diagnostic tools on the sangoma cards are unbeatable |
| 17:44.02 | Wayhigh | yeah? |
| 17:44.02 | [TK]D-Fender | Wayhigh: I personally recommend Sangoma A200d's anyways ;) |
| 17:44.10 | *** join/#asterisk implicit (n=bayan@64.241.37.140) |
| 17:44.10 | Wayhigh | that may be good in this case |
| 17:44.13 | Wayhigh | let me see the price |
| 17:44.20 | exvito | [TK]D-Fender: No way --> WARNING[32093]: chan_zap.c:11244 process_zap: Ignoring setvar |
| 17:44.25 | Kobaz | nothing like looking at the crc counter and telling the telco, hey your pri is fscked |
| 17:44.26 | [TK]D-Fender | Wayhigh: nearly the same |
| 17:44.53 | magic_hat | hey all. i want to add a function to my * system so that a user can begin recording a call by pressing #9. Can I do that by just creating an extension that issues the Monitor command? Or am I looking for something else? |
| 17:44.57 | [TK]D-Fender | exvito: Could be they never got around to it. besides... who uses Zaptel FXS anyways? ;) |
| 17:45.13 | exvito | [TK]D-Fender: ...silly FAX machines ?! ;) |
| 17:45.20 | *** join/#asterisk lalmia (n=cmrahman@64.129.97.77) |
| 17:45.22 | [TK]D-Fender | magic_hat: Don't reinvent the wheel. features.conf <------ |
| 17:45.28 | [TK]D-Fender | exvito: indeed! |
| 17:45.28 | exvito | [TK]D-Fender: ...+ good old DECT base stations ! |
| 17:45.30 | gramulhaozin | Kobaz: 8 |
| 17:45.52 | [TK]D-Fender | exvito: ONLY faxes validate it (even though at the point I STILL recommend leaving them on their own line). |
| 17:45.55 | Kobaz | gramulhaozin: and what does vonage business charge for 8 channels |
| 17:46.00 | [TK]D-Fender | exvito: Voice = ATA <- |
| 17:47.30 | Wayhigh | thanks for the info |
| 17:47.41 | magic_hat | [TK]D: Okay, cool. and I set the DYNAMIC_FEATURES var in the dialplan? |
| 17:48.23 | Wayhigh | anyone seen steve tutauro (sp?) |
| 17:48.26 | gramulhaozin | Kobaz: 149 for unlimited channels incoming |
| 17:48.36 | [TK]D-Fender | magic_hat: Go read the WIKI and try it. |
| 17:48.55 | [TK]D-Fender | ~seen stotaro |
| 17:49.00 | jbot | i haven't seen 'stotaro', [TK]D-Fender |
| 17:49.04 | Wayhigh | ahh that's the name |
| 17:49.15 | [TK]D-Fender | ~seen stutaro |
| 17:49.16 | jbot | [TK]D-Fender: i haven't seen 'stutaro' |
| 17:49.23 | Kobaz | gramulhaozin: if you want reliable, go with a telco, for ~$100-150 more you can get yourself a pri |
| 17:49.30 | [TK]D-Fender | nope, but maybe getting warmer :) |
| 17:49.35 | Wayhigh | hahaha |
| 17:53.12 | gramulhaozin | Kobaz: I thought vonage was a telco |
| 17:53.28 | Qwell | [TK]D-Fender: |
| 17:53.31 | Qwell | ~seen stevetotaro |
| 17:53.32 | jbot | stevetotaro <n=Administ@pool-70-17-222-90.balt.east.verizon.net> was last seen on IRC in channel #asterisk, 32d 16h 24m 45s ago, saying: 'i could see that being a huge headache'. |
| 17:54.00 | Kobaz | gramulhaozin: it's a voip provider (ie: itsp), not a telco |
| 17:54.11 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-220-rndf-esr-5.dynamic.isadsl.co.za) |
| 17:54.54 | Wayhigh | wow.. he hasn't been around in forever |
| 17:55.18 | gramulhaozin | Kobaz: it's a TELCO dude |
| 17:55.31 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 17:55.33 | Kobaz | gramulhaozin: ask vonage for a pri then |
| 17:56.05 | gramulhaozin | Kobaz: If I wanted a pri T1 I would call sprint |
| 17:56.16 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 17:56.47 | Kobaz | i'm just saying.... you said reliable a vonage in the same sentence |
| 17:56.53 | Kobaz | a=and |
| 17:56.55 | gramulhaozin | hey |
| 17:57.08 | gramulhaozin | Anyone out there uses Vonage in a Business with an IP PBX ? |
| 17:57.23 | Kobaz | i've used vonage before, it's good and all... but keep in mind... no internet = no phone |
| 17:58.16 | Kobaz | it depends how important the phones are, is what it comes down to |
| 17:58.22 | gramulhaozin | Kobaz: ooh really ? |
| 17:58.31 | gramulhaozin | Kobaz: I didn't know that |
| 17:58.36 | Kobaz | heh |
| 17:58.40 | gramulhaozin | Kobaz: what about asterisk does it work without network ? |
| 17:58.53 | Kobaz | your local network doesn't depend on the internet |
| 17:58.59 | Kobaz | my internet goes down, i still have phone service |
| 17:59.12 | gramulhaozin | Kobaz: ohh really ? |
| 17:59.21 | gramulhaozin | Kobaz: ooh you are the man. |
| 17:59.34 | gramulhaozin | Anyone out there uses Vonage in a Business with an IP PBX ? |
| 17:59.42 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 18:00.03 | Kobaz | gramulhaozin: what you really need to be doing... is asking vonage for sip credentials |
| 18:00.14 | Kobaz | gramulhaozin: and then customize your asterisk system as needed |
| 18:00.53 | gramulhaozin | Kobaz: dude I have asterisk for 4 years |
| 18:00.56 | b0lt | i want to use grandcentral's call feature to make free phone calls, but that requires you to query a page and then pick up a call. is there any way to have asterisk run a script when i dial out and then forward an incoming call to the current line? |
| 18:01.19 | Kobaz | b0lt: yeap... look up AGI |
| 18:01.59 | Kobaz | actually, no, i misread that |
| 18:02.12 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:02.20 | b0lt | er, i didn't know agi could do that |
| 18:02.27 | Kobaz | all you need is a web scraper that hits the site every x time interval, say every minute... and see if there are any lines to pick up |
| 18:02.47 | Kobaz | and then asterisk doesnt need to do anything at all, assuming you have your call comming in on sip or iax or whatever |
| 18:03.08 | Kobaz | it will just pick up the line as usual |
| 18:03.16 | b0lt | Kobaz, well, how do i pick up a call on a line that's already open? |
| 18:04.00 | Kobaz | what's the flow? |
| 18:04.01 | Kobaz | you go to a website and see your current phone calls and have an option to pick up? |
| 18:04.14 | Kobaz | or your phone rings, and then you need to go to some website to finish the call? |
| 18:04.34 | b0lt | okay |
| 18:04.36 | b0lt | when i dial out |
| 18:04.47 | b0lt | a script talks to another website, which will call me |
| 18:04.49 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 18:04.59 | b0lt | i want to forward that incoming call to the line i dial out from |
| 18:05.36 | *** part/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 18:05.41 | Kobaz | not following |
| 18:06.30 | b0lt | google grandcentral has a feature where they call both you and the reciever and link the two together |
| 18:11.45 | b0lt | i want it so that when i dial out on my asterisk phone, it uses grandcentral to make the call |
| 18:11.45 | Kobaz | ooh |
| 18:11.45 | Kobaz | the easy way would be a ringback |
| 18:11.45 | Kobaz | you dial a number, it runs the script, hangs up the line |
| 18:11.45 | Kobaz | and then your phone rings when grandcentral calls |
| 18:11.46 | b0lt | is there any way for me to forward the call to the line, though? |
| 18:11.46 | *** join/#asterisk Firass-VC22 (n=Firass-z@method.vikcomm.wwu.edu) |
| 18:11.46 | Kobaz | not easily |
| 18:11.46 | Kobaz | you would have to do either like, a call conference, or maybe send both channels into a meetme |
| 18:11.46 | Kobaz | actually, the meetme method may work nicely |
| 18:11.47 | Kobaz | make a call, asterisk uses agi to run your dial script |
| 18:11.47 | Kobaz | sends you into a meetme |
| 18:11.49 | *** join/#asterisk implicit (n=bayan@32.155.179.204) |
| 18:11.52 | Kobaz | when asterisk picks up the line, send it into the same meetme |
| 18:12.42 | b0lt | what is a meetme? |
| 18:12.50 | b0lt | is it a conference line? |
| 18:12.52 | Kobaz | conference bridge |
| 18:12.54 | Kobaz | yeah |
| 18:13.06 | Kobaz | show application Meetme |
| 18:15.21 | b0lt | thanks |
| 18:17.14 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 18:19.28 | Kobaz | b0lt: you can actually use TryExec instead of agi |
| 18:19.47 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 18:20.49 | *** part/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:21.09 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 18:21.27 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:23.26 | b0lt | Kobaz, ah |
| 18:24.09 | *** join/#asterisk implicit (n=bayan@32.155.179.204) |
| 18:25.01 | b0lt | thanks |
| 18:27.31 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:33.23 | *** join/#asterisk Ast001 (n=uros@81.18.55.102) |
| 18:33.35 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:33.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:33.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:34.39 | Ast001 | i still can not receive a call via post isdn pri despite i turned off everything I could in bios |
| 18:35.01 | *** join/#asterisk fogo (n=fogo@69.169.132.121) |
| 18:35.41 | [TK]D-Fender | Ast001: PASTEBIN is your friend. Use it. |
| 18:36.26 | Ast001 | does anyone use that openvox card ? Is it working for someone ? |
| 18:36.33 | Ast001 | openvox d110p |
| 18:36.43 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:37.07 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 18:37.10 | [TK]D-Fender | Ast001: Go show us your problem, and no you won't find many clone-card users here |
| 18:37.56 | *** part/#asterisk emiller (n=ed@216.207.245.1) |
| 18:40.54 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 18:41.49 | *** part/#asterisk emiller (n=ed@216.207.245.1) |
| 18:42.19 | Ast001 | Problem was that openvox card used same irq like a lot of other devices , so you sugest to turn off that in bios I did I still can not receive a call . Insted i get busy signal |
| 18:42.50 | [TK]D-Fender | Ast001: And your description does not tell me that you configured any of it properly. |
| 18:43.06 | *** join/#asterisk emiller (n=ed@216.207.245.1) |
| 18:43.08 | [TK]D-Fender | Ast001: show us your configs, and CLI output with intense PRI debug & verbose 10 in a PASTEBIN. |
| 18:43.14 | Ast001 | here is zapata.conf http://pastebin.com/m2a730a45 |
| 18:43.35 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 18:44.05 | [TK]D-Fender | Ast001: and the output of "ztcfg -vvvv" |
| 18:44.54 | Ast001 | that shows 31 channels |
| 18:45.01 | Ast001 | this error is crucial |
| 18:45.31 | Ast001 | notice chan_zap.c pri_c channel pri got event hdlc bad fc3(8) on primary d channel of span 1 |
| 18:45.45 | [TK]D-Fender | Ast001: PASTEBIN it. |
| 18:46.15 | [TK]D-Fender | Ast001: and I asked you earlier what version of Zaptel & * you were using. |
| 18:46.45 | Ast001 | 1-4-10 zaptel |
| 18:47.25 | Ast001 | which compiled well and tell me that it thinks i am using digium te110p card |
| 18:48.08 | Ast001 | I can not much pastebin at the moment coz i am not near server at the moment but i dunno what above error message means |
| 18:48.31 | Ast001 | is it openvox card problem ? motherboard problem ? or configuration problem ? |
| 18:48.55 | Ast001 | I can pastebin only zapata.conf and zaptel conf at the moment |
| 18:49.13 | [TK]D-Fender | Ast001: Sorry, we aren't psychic |
| 18:49.38 | [TK]D-Fender | Ast001: and there should be a jumper you'll need to set to E1 on that card as well. |
| 18:49.41 | Ast001 | ok i will paste bin whatever you want tomorrow |
| 18:49.49 | Ast001 | that jumper has been set |
| 18:49.51 | Ast001 | i checked that |
| 18:49.59 | Ast001 | pin on E1 pin off T1 |
| 18:50.03 | Ast001 | its pin on |
| 18:50.08 | Ast001 | jumper on |
| 18:50.30 | Ast001 | that is not problem because ztcfg -vv shows me 31 channel |
| 18:50.40 | Ast001 | zttool shows me alarm ok |
| 18:51.35 | [TK]D-Fender | Ast001: Ok, well come back tomorrow when you can actually do something useful. |
| 18:51.52 | Ast001 | ok i wll |
| 18:52.34 | jeev | Fender, so much testing gone into the system.. everything is perfect, ordering the polycom's soon1 |
| 18:52.36 | jeev | ! |
| 18:52.37 | Ast001 | you can tell me now what you want to see on pastebin |
| 18:52.44 | jeev | what's the bes source for polycom 330's now? still telephonydepot ? |
| 18:53.09 | Ast001 | ztcfg -vvvv cli with vvvvvvvvvv and something else ? |
| 18:53.26 | tzafrir_laptop | careful with asking for "ztcfg -vvv" |
| 18:54.05 | tzafrir_laptop | ztcfg may change state. Normally it doesn't. But usually simply looking at /proc/zaptel/* gives you the same information in a hrmless way |
| 18:54.17 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 18:54.18 | *** join/#asterisk ManxPower (n=manxpowe@115.sub-75-250-134.myvzw.com) |
| 18:55.04 | [TK]D-Fender | jeev: really close to he best, but good serverice to match |
| 18:55.19 | jeev | ok cool |
| 18:55.21 | jeev | cant wait. |
| 18:55.34 | [TK]D-Fender | jeev: Yeah... thats what "impulse shopper" means ;) |
| 18:56.06 | jeev | ;) |
| 18:56.08 | Assid | i think my stupid pap2 is starting to go crazy |
| 18:56.15 | jeev | i'm gonna take my friend also to the system |
| 18:56.19 | jeev | to the asterisk box |
| 18:56.23 | jeev | at another office............ |
| 18:56.47 | Assid | i cant seem to get incoming calls.. (asterisk shows alot of invites hung up) and the web interface is just messed up.. doesnt really "work" |
| 18:56.58 | *** part/#asterisk Ast001 (n=uros@81.18.55.102) |
| 18:57.09 | Assid | probably time to flash it or something |
| 18:57.27 | Assid | do the firmwares get corrupt on theser things? |
| 18:58.35 | macros73 | seeing an issue today where g729 traffic is choppy, but g711 is fine. |
| 18:59.31 | *** join/#asterisk BipBip (n=BipBip@213.58.136.163) |
| 19:00.49 | lmadsen | macros73: too much transcoding on an overloaded system perhaps? |
| 19:00.58 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 19:01.09 | Skarmeth | hi all |
| 19:01.23 | macros73 | lmadsen: Same load as yesterday, just me. Test system. |
| 19:01.51 | macros73 | only changes were updates to Core, Dashboard, and another module this morning (PiaF). |
| 19:02.11 | macros73 | If I could downgrade, I would just to see if the problem goes away. |
| 19:02.23 | Skarmeth | anyone here using two asterisk servers interconnected by IAX2 where one servers runs asterisk 1.4 and another runs asterisk 1.6? I can't make 1.4 calls to 1.6 negotiate the codecs, but calls that initiate at 1.6 to 1.4 works great. |
| 19:02.41 | Skarmeth | there is any known issue about this and any fix? |
| 19:04.11 | ManxPower | Skarmeth: If this is a known issue it would be listed in the upgrade.txt for 1.6 |
| 19:04.44 | CVirus | [TK]D-Fender: the card is working .. so it's the GSM box for sure ... any way to get that to receive DTMF ? |
| 19:05.05 | [TK]D-Fender | CVirus: Go read its manual, and find yourself a sacrifical goat. |
| 19:05.22 | CVirus | lol .. ok |
| 19:05.51 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:06.46 | Skarmeth | ManxPower, there is not such thing there |
| 19:10.32 | CVirus | [TK]D-Fender: isn't there any other way to authenticate ? |
| 19:11.17 | Skarmeth | ManxPower, but for the last two months, every asterisk (1.4 and 1.6) version combination that I used have this problem |
| 19:11.36 | Skarmeth | I have to set codec manually to GSM instead of G.711 |
| 19:11.50 | [TK]D-Fender | CVirus: Authenticate isn't the probelm. Its that you don't have any DTMF. You're pretty much fucked in that scenario. |
| 19:12.05 | Skarmeth | (I have a 1GiB link between the two servers and bandwidth it's not a problem) |
| 19:12.05 | CVirus | I see |
| 19:13.21 | macros73 | Should I try turning off silence suppression? |
| 19:14.50 | ManxPower | Skarmeth: There is no doc file for upgrades in /path/to/src/asterisk/doc/ |
| 19:14.53 | ManxPower | ?? |
| 19:15.56 | ManxPower | Skarmeth: chances are you have to set some option like 14compat=yes or something like that on the 1.6 machine. It would be in the upgrade info file. |
| 19:16.26 | Skarmeth | ManxPower, UPGRADE.txt have no info about codec incompatibility between the two versions |
| 19:17.20 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 19:17.33 | ManxPower | Skarmeth: there could be a hundred non-codec issues that could cause the problem. |
| 19:17.47 | ManxPower | is this sip, iax2, mgcp, sccp or h323? |
| 19:17.59 | Skarmeth | iax2 |
| 19:18.15 | ManxPower | Skarmeth: if I find the info when I finish downloading 1.6.0beta9 I will insult you |
| 19:20.58 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 19:21.41 | *** join/#asterisk AdamWest (n=Leif@CPE001d7e2f9574-CM0012c9db3d2e.cpe.net.cable.rogers.com) |
| 19:21.53 | ManxPower | Skarmeth: Looks like it's time to file a bug, eh? |
| 19:22.03 | ManxPower | I assume you are using beta9 and not some previous beta version? |
| 19:22.17 | Skarmeth | ManxPower, I am using a specific branch now |
| 19:24.33 | Skarmeth | ManxPower, http://pastebin.ca/1068426 |
| 19:24.54 | Skarmeth | I am using this branch http://svn.digium.com/svn/asterisk/team/moy/mfcr2 |
| 19:25.43 | ManxPower | Skarmeth: then we really can't support you here as you are not even using a released beta |
| 19:25.59 | ManxPower | you should contact the author of that branch for help |
| 19:26.28 | Skarmeth | ManxPower, that branch are in sync with normal trunk |
| 19:26.45 | ManxPower | Skarmeth: unless you are using a tarball verison you should be on #asterisk-dev |
| 19:27.53 | ManxPower | heck, if you are using anything with 1.6 you should be on asterisk-dev. |
| 19:28.19 | Corydon76-dig | ManxPower: that's not 1.6, that's a developer branch |
| 19:28.43 | Corydon76-dig | Skarmeth: who are you coordinating with, as you test that branch? |
| 19:29.40 | ManxPower | Corydon76-dig: then why is he here asking about it? 8-) |
| 19:29.44 | Skarmeth | moy are the branch coordinator, but they are working specific with channels/chan_zap.c to support MFC/R2 signalling |
| 19:31.42 | *** part/#asterisk ManxPower (n=manxpowe@115.sub-75-250-134.myvzw.com) |
| 19:33.22 | Corydon76-dig | Skarmeth: channels/chan_zap is no longer a part of trunk, so there's probably something wrong with that branch |
| 19:34.27 | xnixan | hi, when trying to compile zaptel -1.4.11 every thing goes well , but whet trying to "make install" i have this message http://www.pastebin.org/50345 |
| 19:34.34 | xnixan | any ideas? |
| 19:37.04 | Skarmeth | Corydon76-dig, chan_zap.c => chan_dahdi.c |
| 19:38.28 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 19:38.36 | UnixDog | ithought it would not be dahdi till 1.6 |
| 19:38.47 | UnixDog | who's your dahdi ........ |
| 19:40.37 | merkurie | when enabling imap voicemail storage, it creates a seperate module called app_voicemail_imap.so, right? |
| 19:40.53 | *** join/#asterisk ManxPower (n=manxpowe@115.sub-75-250-134.myvzw.com) |
| 19:41.00 | Skarmeth | xnixan, try to edit scripts/Makefile.build at line 41 and remove the / so it's read kernel/Makefile... |
| 19:41.21 | xnixan | Skarmeth, did you try it? |
| 19:43.15 | *** part/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:43.30 | ManxPower | ~centos52bug |
| 19:43.30 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 19:45.30 | Skarmeth | xnixan, no, since I don't know what you did |
| 19:45.59 | Skarmeth | but the thing that seems to be wrong are the path |
| 19:46.39 | *** join/#asterisk Op3r (n=op3r@ded-139-109.eglobalreach.net) |
| 19:46.41 | Skarmeth | it's this dahdi source from trunk? |
| 19:47.17 | Op3r | hello can anyone help me with my te110p? I was trying to use it as a timer and I get this error when dialing a meetme room |
| 19:47.36 | Op3r | primevici*CLI> dial 8600101@default |
| 19:47.36 | Op3r | <PROTECTED> |
| 19:47.36 | Op3r | <PROTECTED> |
| 19:47.36 | Op3r | <PROTECTED> |
| 19:47.36 | Op3r | Jul 10 15:46:11 WARNING[4314]: channel.c:2621 ast_request: No channel type registered for 'zap' |
| 19:47.37 | Op3r | Jul 10 15:46:11 WARNING[4314]: app_meetme.c:480 build_conf: Unable to open pseudo channel - trying device |
| 19:47.39 | Op3r | <PROTECTED> |
| 19:47.43 | Op3r | <PROTECTED> |
| 19:47.45 | Op3r | <PROTECTED> |
| 19:47.50 | Op3r | err sorry for flooding :( |
| 19:47.55 | xnixan | Skarmeth, there is nothing appear to be wrong with line 41, that is why i am asking |
| 19:47.58 | ManxPower | Op3r: you need to reinstall asterisk AFTER you install zaptel. There's a reason for the order in the install readme |
| 19:48.09 | Skarmeth | xnixan, make[1]: Entering directory `/home/iunixan/asterisk-1.4/zaptel-1.4.11' |
| 19:48.20 | Skarmeth | then |
| 19:48.21 | Skarmeth | make -C /lib/modules/2.6.25.9-76.fc9.i686/build ARCH=i386 SUBDIRS=/kernel HOTPLUG_FIRMWARE=yes KBUILD_OBJ_M="pciradio.o tor2.o torisa.o wcfxo.o wct1xxp.o wctdm.o wcte11xp.o wcusb.o zaptel.o ztd-eth.o ztd-loc.o ztdummy.o ztdynamic.o zttranscode.o wct4xxp/ wctc4xxp/ wctdm24xxp/ wcte12xp/" modules |
| 19:48.23 | Op3r | ManxPower, yep I did that :( |
| 19:48.25 | ManxPower | xnixan: what specific issue are you having? |
| 19:48.34 | Skarmeth | the SUBDIRS should be kernel not /kernel |
| 19:48.46 | xnixan | here is lines 38-41 |
| 19:48.48 | xnixan | # The filename Kbuild has precedence over Makefile |
| 19:48.48 | xnixan | kbuild-dir := $(if $(filter /%,$(src)),$(src),$(srctree)/$(src)) |
| 19:48.48 | xnixan | kbuild-file := $(if $(wildcard $(kbuild-dir)/Kbuild),$(kbuild-dir)/Kbuild,$(kbuild-dir)/Makefile) |
| 19:48.48 | xnixan | include $(kbuild-file |
| 19:49.07 | ManxPower | Op3r: What type of line to you have connected into the te110P? |
| 19:49.16 | ManxPower | xnixan: do NOT flood the channel use pastebin |
| 19:49.18 | ManxPower | ~pb |
| 19:49.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:49.26 | Op3r | ManxPower, none as I was just using it as a timer for meetme |
| 19:49.26 | Skarmeth | xnixan, the problem should be in Makefile |
| 19:49.45 | Op3r | ManxPower, none I was using it as a timer for meetme room |
| 19:50.04 | ManxPower | Op3r: I don't believe it will work without a line set up, but you can still try it. does ztcfg -vvv generate any errors? |
| 19:50.12 | Skarmeth | xnixan, looks for SUBDIRS=/kernel it should be SUBDIRS=kernel |
| 19:50.15 | Op3r | ManxPower, nope it doesnt |
| 19:50.19 | *** join/#asterisk Ebola_ (n=Ebola@unaffiliated/ebola) |
| 19:50.37 | ManxPower | put the contents of /etc/zaptel.conf and /etc/asterisk/zapata.conf on pastebin |
| 19:51.39 | Op3r | ManxPower, nope it doesnt wait let me paste my conf to pastebin |
| 19:52.24 | Op3r | ManxPower, http://pastebin.com/d1e6a087a |
| 19:52.26 | tzafrir_laptop | SUBDIRS=/kernel ? |
| 19:52.28 | Skarmeth | xnixan, http://svn.digium.com/view/zaptel/tags/1.4.11/Makefile?view=markup look at line 195 |
| 19:52.46 | tzafrir_laptop | anybody here is installing using sudo ? |
| 19:52.57 | Skarmeth | ${PWD} are not returning . and then the SUBDIRS get set to /kernel |
| 19:53.01 | tzafrir_laptop | Skarmeth, xnixan ==^ |
| 19:53.02 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 19:53.15 | Op3r | ManxPower, I used genzaptelconf to configure it as I am just using it for timing |
| 19:53.20 | ManxPower | Op3r: what country are you located in? |
| 19:53.40 | ManxPower | Op3r: none of the auto-generating utils will work because you don't have a line plugged in |
| 19:53.58 | Op3r | philippines. but im not going to use it for anything just timing |
| 19:53.58 | tzafrir_laptop | ManxPower, huh? |
| 19:54.29 | tzafrir_laptop | zapconf/genzaptelconf/dahdi_genconf only needs the lines in /proc/zaptel (/proc/dahdi) |
| 19:55.12 | unpaidbill | i love when companies come in to introduce themselves... i got 2 pens and a coffee mug! |
| 19:55.12 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 19:55.16 | unpaidbill | awesome! |
| 19:55.20 | ManxPower | tzafrir: he has no line, he's trying to use it as timing |
| 19:55.21 | tzafrir_laptop | There should be a module parameter to wctdm to use it just for timing, IIRC. Try modinfo wctdm |
| 19:55.27 | ManxPower | well mixing, at least |
| 19:55.45 | ManxPower | tzafrir: you mean modinfo wcte1xx right? |
| 19:55.59 | Op3r | hmmm |
| 19:56.04 | tzafrir_laptop | timingonly |
| 19:56.49 | tzafrir_laptop | ah, in that case set it as: span=1,0,0,<whatever>,<whatever> |
| 19:57.09 | Op3r | so Ill just set it on zaptel.conf? |
| 19:57.15 | tzafrir_laptop | I think that if it is the only module it will provide timing even when red |
| 19:57.20 | *** join/#asterisk LND (n=Lee@89.192.192.52) |
| 19:57.33 | tzafrir_laptop | But I'm not sure. Worth a shot |
| 19:58.22 | ManxPower | Op3r: try this http://pastebin.com/m49d45c1a |
| 19:58.25 | Op3r | but Im having this error http://pastebin.com/d649b32fb |
| 19:58.45 | Skarmeth | xnixan, did you get it build? |
| 19:58.51 | Op3r | ManxPower, ok let me try |
| 19:59.12 | tzafrir_laptop | sure you have a problem with that. You tell it to get timing from the remote side |
| 19:59.15 | ManxPower | Op3r: then rerun ztcfg -vvv as well as restart asterisk or unload chan_zap.so / load chan_zap.so |
| 19:59.29 | ManxPower | tzafrir: good catch |
| 19:59.30 | Op3r | ok |
| 19:59.41 | ManxPower | Op3r: change the 2nd "1" on the span= line to a 0 |
| 19:59.56 | xnixan | Skarmeth, still |
| 20:00.12 | ManxPower | tzafrir: as russel tells me all the time, MeetMe uses zaptel for audio MIXING, not timing. |
| 20:00.14 | tzafrir_laptop | Op3r, http://pastebin.com/m383da72a |
| 20:00.25 | russellb | hehe |
| 20:00.36 | Op3r | ok |
| 20:00.37 | tzafrir_laptop | Right. But it still uses pseudo channels |
| 20:00.40 | ManxPower | tzafrir: chan_zap won't load with no channels configured |
| 20:00.47 | tzafrir_laptop | It doesn't need real channels, right? |
| 20:00.58 | russellb | of course, the mixing requires timing. my point is, simply providing another timing source does not relive meetme of a zaptel/dahdi dependency |
| 20:01.12 | tzafrir_laptop | Why won't chan_zap load without channels configured? |
| 20:01.22 | ManxPower | tzafrir: Ask russel |
| 20:01.35 | russellb | looks around |
| 20:01.37 | russellb | shrugs |
| 20:01.42 | ManxPower | (2:47:36 PM) Op3r: Jul 10 15:46:11 WARNING[4314]: channel.c:2621 ast_request: No channel type registered for 'zap' |
| 20:02.01 | ManxPower | He CLAIMS he rebuilt asterisk after building zaptel -- MeetMe would not have built otherwise. |
| 20:02.26 | Op3r | as I did installed zaptel first before asterisk |
| 20:02.28 | ManxPower | (op3r claims, not russellb) |
| 20:02.46 | ManxPower | Op3r: what happens when you do "module load chan_zap.so" |
| 20:03.21 | Op3r | err Im using 1.2 ok |
| 20:03.25 | Op3r | primevici*CLI> module load chan_zap.so |
| 20:03.25 | Op3r | No such command 'module' (type 'help' for help) |
| 20:03.33 | ManxPower | ok then, "load chan_zap.so" |
| 20:03.42 | ManxPower | I hope you are using 1.2 zaptel too |
| 20:03.51 | Op3r | yep |
| 20:03.56 | tzafrir_laptop | ManxPower, my laptop loads chan_zap.so just fine with an empty [channels] section |
| 20:04.09 | ManxPower | tzafrir: must be some OTHER reason then. I hate that. |
| 20:04.13 | tzafrir_laptop | And anyway, you don't need chan_zap for meetme to have zap support |
| 20:04.13 | Op3r | here's the error |
| 20:04.15 | Op3r | http://pastebin.com/db579ff0 |
| 20:04.48 | tzafrir_laptop | process_text_line: Unknown directive '' at line 1 of /etc/asterisk/zapata.conf |
| 20:04.55 | ManxPower | Op3r: put a copy of /etc/asterisk/zapata.conf on pastebin again |
| 20:04.59 | Op3r | ok |
| 20:05.02 | tzafrir_laptop | This means you used '#' for a comment |
| 20:05.04 | ManxPower | I bet he forgot channels and trunkgroups too |
| 20:05.09 | Op3r | err |
| 20:05.16 | tzafrir_laptop | Normally harmless, but do replace it with ';' |
| 20:05.45 | ManxPower | tzafrir_laptop: thanks for helping him. I'll leave him in your hands. |
| 20:05.56 | Op3r | yeah |
| 20:06.02 | tzafrir_laptop | setup_zap: Unknown signalling method 'fxsls' |
| 20:06.04 | Op3r | I forgot to use ; |
| 20:06.06 | Op3r | :( |
| 20:06.06 | tzafrir_laptop | should be fxs_ls |
| 20:06.20 | unpaidbill | how do you become a certified asterisk consultant |
| 20:06.29 | unpaidbill | i've never seen the title until just now |
| 20:06.30 | tzafrir_laptop | But if you followed my suggested config, you wouldn't need them, anyway |
| 20:06.43 | tzafrir_laptop | (I still hope it works...) |
| 20:07.04 | Op3r | take a look at my zaptel.conf |
| 20:08.03 | Op3r | tzafrir, http://pastebin.com/m4c20a3b |
| 20:08.49 | ManxPower | unpaidbill: contact Digium |
| 20:08.53 | tzafrir_laptop | fxs_ls = 1-31; I think this is not supposed to work |
| 20:09.11 | unpaidbill | do they give you a cool plaque |
| 20:09.22 | tzafrir_laptop | channel 16 is still reserved, IIR, right? |
| 20:09.25 | ManxPower | tzafrir_laptop: I figured just set for channelized voice t-1, then you don't have to worry about Dhcannels |
| 20:09.27 | [TK]D-Fender | unpaidbill: Yes, but Crest will help remove it ;) |
| 20:09.29 | ManxPower | unpaidbill: Contact Digium |
| 20:09.32 | unpaidbill | haha |
| 20:09.43 | ManxPower | tzafrir_laptop: Even on channelized E-1? |
| 20:09.52 | tzafrir_laptop | But this may be bullshit |
| 20:09.58 | Op3r | tzafrir-laptop: i got this error when I id ztcfg -vvv line 16: Unknown keyword 'fxs_ls' |
| 20:10.23 | tzafrir_laptop | in zaptel.conf it is fxsls=<channels> |
| 20:10.29 | tzafrir_laptop | no '_' |
| 20:11.55 | Op3r | yep it did work when I remove the _ |
| 20:11.56 | ManxPower | tzafrir_laptop: that's what happens when it's been 3 years since you setup a non-PRI in Asterisk 8) |
| 20:12.07 | Op3r | 31 channels configured. |
| 20:12.08 | *** join/#asterisk los415 (n=los415@LAX-DHCP-64-201-109-217.race.com) |
| 20:12.14 | ManxPower | hugs all 8 of his PRIs |
| 20:12.18 | los415 | hey guys has anyone ever setup asterisk with smdi |
| 20:12.22 | Op3r | but Im still getting this error when I am dialing a meetme room |
| 20:12.42 | ManxPower | los415: none of the extensive mailing list discussions in the archive were helpful to you? |
| 20:12.56 | tzafrir_laptop | Op3r, what do you see in /proc/zaptel/1 |
| 20:13.01 | tzafrir_laptop | ? |
| 20:13.07 | Op3r | http://pastebin.com/d1bb1c19c |
| 20:13.19 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 20:13.30 | los415 | manxpower they where but i'm having issues with the serial side of it |
| 20:13.35 | exothermc | Can US carriers charge for numbers porting out of their network? |
| 20:13.36 | ManxPower | Op3r: "load chan_zap.so" |
| 20:13.45 | Op3r | tzafrir, [root@primevici asterisk]# cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS RED |
| 20:13.46 | los415 | dont know if i'm using the right serial cable |
| 20:13.47 | ThoMe | Op3r: yo have the channeldriver loadet? |
| 20:13.52 | ThoMe | hmm. |
| 20:13.55 | los415 | so want to know if someone knows the pinouts |
| 20:14.40 | ManxPower | los415: Yours may be the first SMDI question I've seen in my 5 years on this channel |
| 20:15.11 | los415 | yea there isnt to much info on it |
| 20:15.14 | Op3r | there you go |
| 20:15.29 | *** join/#asterisk javb (n=jvb@96santrd02.codetel.net.do) |
| 20:15.30 | los415 | i'm actully connecting a taqua switch to asterisk to do vm |
| 20:15.31 | Op3r | it works ManxPower when I did the load chan_zap.so |
| 20:15.35 | Op3r | :D |
| 20:15.36 | Op3r | thanks |
| 20:15.38 | los415 | having trouble with the smdi |
| 20:15.58 | Op3r | I was wondering why i have to load it when I just restarted asterisk by using restart now :( |
| 20:16.17 | ManxPower | Op3r: you should not have to. something screwed up in your /etc/asterisk/modules.conf |
| 20:16.31 | Op3r | hmmm |
| 20:16.36 | Op3r | let me check that conf |
| 20:16.54 | javb | hello, i have norstar with a T1 from PSTN, i put an Asterisk beetwen the Norstar and the PSTN, with 2 Ports T1, the PSTN T1 Trunk goes up (OK) but the one with the Norstar DONT, even if have it with the same info i have with th PSTN .. any idea ? ? |
| 20:17.11 | Op3r | ManxPower, amazing I have this ; Don't load zap modules |
| 20:17.11 | Op3r | noload => chan_zap.so |
| 20:17.14 | Op3r | dman |
| 20:17.16 | tzafrir_laptop | Op3r, if you want timing from zaptel, you don't really need chan_zap |
| 20:17.18 | Op3r | eeeeeeeeeeeeeeeeeerrrrrrrrrrrrrr |
| 20:17.35 | tzafrir_laptop | no, fo you actually have a good timing source? try zttest |
| 20:17.40 | ManxPower | Op3r: there's a chance that was the problem all along and everything we did was pointless |
| 20:17.41 | tzafrir_laptop | zttest -c 3 |
| 20:17.48 | Op3r | ManxPower, I think so too |
| 20:17.50 | Op3r | :( |
| 20:18.09 | javb | Any idea? |
| 20:18.16 | los415 | javb have you tried rolling the cable |
| 20:18.43 | los415 | usally from asterisk to say a pbx you use a t1 crossover |
| 20:18.59 | Op3r | tzafrir_laptop, i get this 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% |
| 20:19.28 | javb | los415, yes, rolled the cables . . . |
| 20:19.39 | ManxPower | javb: you need a T-1 crossover (NOT ETHERNET CROSSOVER) |
| 20:19.47 | tzafrir_laptop | Op3r, now, does meetme (app_meetme.so) give any errors? |
| 20:20.04 | ManxPower | telco-asterisk = straight thru and asterisk-pbxorchannelbank is crossover |
| 20:20.05 | Op3r | tzafrir, nope it doesnt anymore when I loaded the chan_zap.so |
| 20:20.07 | los415 | usally the lecs will drop an rj48x jack which rols the cable for you |
| 20:20.13 | los415 | where u can just use a straight thru |
| 20:20.52 | Op3r | is this timing any good? 100.000000% 99.987793% 100.000000% 100.000000% 99.987793% 100.000000% 99.987793% 100.000000% |
| 20:20.59 | tzafrir_laptop | Op3r, yes |
| 20:21.07 | Op3r | ok thats good |
| 20:21.08 | Op3r | thanks |
| 20:21.23 | Op3r | tzafrir_laptop, you see im using this for vicidial :D |
| 20:21.26 | tzafrir_laptop | If you had no timing source it would hold "forever" |
| 20:21.37 | Op3r | Best: 100.000000 -- Worst: 99.755859 -- Average: 99.988924 |
| 20:22.47 | Op3r | thanks guys |
| 20:22.57 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-75-51.vif.net) |
| 20:23.53 | macros73 | Wireshark is reporting 95.92% lost RTP packets for the inbound G729 stream. G711 stream shows no loss in either direction. Think my ITSP may have an issue? |
| 20:24.05 | dvdevel | people, is 'switch' only used with/for realtime? |
| 20:24.30 | ManxPower | macros73: It's a networking issue somewhere, that's for sure. |
| 20:25.21 | macros73 | It's strange, because the only variable is the codec used between my Asterisk server and the ITSP. G711 = great, G729 = crap. A few days ago, could barely tell them apart. |
| 20:25.27 | ManxPower | G711 packets will be much larger than G729 packets, perhaps the some ISP between you and them are shaping smaller packets to a lower priority? |
| 20:25.48 | ManxPower | Try using GSM if you can, see how the quality is |
| 20:26.11 | macros73 | Don't think Vitelity supports GSM |
| 20:27.13 | macros73 | I'll give it a try |
| 20:27.16 | ManxPower | macros73: Dude, the answer is the FIRST question in the vitelity user FAQ |
| 20:27.34 | macros73 | d'oh, now I feel dumb, let me check out the faq. |
| 20:28.19 | macros73 | I give up, where is the faq you mention? |
| 20:30.24 | jblack | Dude, you must be retarded. I was able to find the faq by googling "vitelity frequently asked questions", and found it on my first try. I don't even know what a vitelity is |
| 20:32.36 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 20:33.42 | macros73 | Yeah, braniac, and the "first question" in that particular faq doesn't bear on codec or sound quality issues. |
| 20:33.55 | ManxPower | I accessed it by logging into the user portal, FAQ upper right |
| 20:35.07 | macros73 | ManxPower: Thanks. That one is different and has the codec question listed first. I tried GSM. No packet loss, better sound quality, especially on outbound. |
| 20:35.15 | macros73 | Inbound, not as much, but not as bad as recent g729. |
| 20:35.40 | ManxPower | Without logging in you can see it via Main Page / Features / Supported Codecs |
| 20:36.17 | xnixan | Skarmeth, thanks, i have done it |
| 20:36.39 | ManxPower | macros73: I don't remember the exact numbers, but I believe G729 and GSM are similar packet sizes, so that would eliminate packet shaping, except for Deep Packet Inspection and it would be silly for an ISP to do that. |
| 20:36.58 | Kobaz | what's a good place to get inexpensive rj22 headsets |
| 20:37.00 | ManxPower | macros73: e-mail your ITSP and tell them they need a Digium Hardware Transcoding Card for their Asterisk server. 8-) |
| 20:37.34 | ManxPower | I'll bet they don't have enough horsepower to handle transcoding G729 in realtime |
| 20:38.10 | Kobaz | poor horses |
| 20:40.05 | macros73 | ManxPower: Strangely, though, it worked last week and most of this week. |
| 20:40.54 | ManxPower | macros73: If that's the issue, it would totally depend on exactly how many transcoded g729 calls they are handling at any one time. |
| 20:41.23 | ManxPower | maybe some call center just signed up with them and they suddenly have twice as many G729 calls happening |
| 20:41.56 | ManxPower | It's an ITSP, there's only so much over-engineering a non-telco startup can do |
| 20:42.29 | macros73 | ManxPower: True. I'll have to open a ticket and see what they say. |
| 20:42.31 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 20:43.20 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 20:45.44 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:47.03 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:48.17 | *** join/#asterisk implicit (n=bayan@ip68-5-93-237.oc.oc.cox.net) |
| 20:49.05 | *** join/#asterisk Schreiber1337 (n=Schreibe@spectrumcontrol.com) |
| 20:49.26 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 20:50.21 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:57.53 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 20:58.19 | lirakis | does anyone know of a "sip validator" for sip messages? |
| 20:58.58 | lirakis | wireshark is showing "malformed message: sip" on a trace im running.. and it all looks good to me .. im not sure what is "bad" about the message |
| 20:59.01 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 20:59.19 | spokra | are the grandstream atas ok? |
| 20:59.44 | lirakis | spokra: lots of people tear gs apart .. they are ok |
| 20:59.55 | lirakis | i mean .. for home hobby use etc. |
| 20:59.59 | lirakis | not enterprise |
| 21:00.11 | spokra | so for a home phone it's ok? |
| 21:00.21 | lirakis | yeah .. i have a gs 486 at home |
| 21:00.22 | lirakis | no probs |
| 21:00.29 | spokra | cool thanks |
| 21:03.29 | jaytee | lmadsen, you there? |
| 21:04.14 | lmadsen | depends :) |
| 21:05.43 | jaytee | got a quick one for you or at least I think it is. Is it possible in * to configure a specific extension or sip phone so it can only transfer internally and not to an external number? I've tried using TRANSFER_CONTEXT to route the call to a context that filters calls via pattern matching but the call never goes there. |
| 21:06.24 | lmadsen | if its a blind transfer, you'll never know its a transfer actually |
| 21:06.42 | jaytee | yeah, and most would be |
| 21:06.50 | lmadsen | sorry, actually other way around |
| 21:06.58 | lmadsen | I think its the attended transfer that doesn't look like a transfer |
| 21:07.12 | lmadsen | because it just looks like a 2nd call to asterisk -- the transfer happens after that is executed |
| 21:07.23 | lmadsen | so TRANSFER_CONTEXT would only work on blind transfers I believe |
| 21:07.49 | lmadsen | you might also have ot make the variable recursive by prefixing double underscore in front of the variable |
| 21:08.14 | jaytee | [TK]D-Fender suggested trying to check the value of ${BLINDTRANSER} or examine the SIP header for something I could trap on. I messed around a bit with a few ideas but I'm coming up empty now. |
| 21:08.45 | jaytee | sipXecs supposedly has more granularity of permissions for such things. |
| 21:09.40 | jaytee | I may end up having to use it as a main SIP pbx and use sip numbers registered to * to route calls to my * system's PRI's. |
| 21:09.52 | ManxPower | jaytee: not that I know of, but it is easy to prevent a specific phone from DIALING specific numbers |
| 21:10.03 | Skarmeth | ManxPower, codec problem reproduced in beta9 |
| 21:10.32 | jaytee | ManxPower, yeah but in this scenario we can't know all the possible numbers the person might want to transfer to. |
| 21:10.40 | ManxPower | Skarmeth: time to file a bug on bugs.digium.com |
| 21:10.41 | Skarmeth | ManxPower, http://pastebin.com/d161387a |
| 21:11.00 | Skarmeth | ManxPower, I will test with trunk tomorrow before reporting the bug |
| 21:11.12 | jaytee | On Nortel Meridian systems it's easy, CFXA or CFXD as a class of service to allow or deny external forwarding. |
| 21:11.21 | jaytee | not so easy in the VOIP world. |
| 21:12.12 | bkw_ | lmadsen: unless something changed there are variables that are set to let you know it was a blind transfer |
| 21:12.29 | lmadsen | bkw_: right, I think I meant attended transfer |
| 21:12.31 | bkw_ | I haven't seen the latest code but I do recall Tony and I working on that and setting those variables |
| 21:12.43 | lmadsen | I had misspoke |
| 21:12.44 | bkw_ | yah attended is a bit harder |
| 21:13.28 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:14.01 | bkw_ | TRANSFER_CONTEXT should work with attended transfers that take place by pressing # |
| 21:14.03 | bkw_ | but not via SIP |
| 21:14.09 | lmadsen | right |
| 21:14.15 | bkw_ | I can't recall 100% but I do recall adding that for that purpose |
| 21:14.19 | lmadsen | because there is no way to know until you decide to do the bridge :) |
| 21:14.29 | jaytee | I tried setting test calls so that I could use NoOp to display the value of the BLINDTRANSFER variable but it was empty for each call. I tried using TRANSFER_CONTEXT to set a context to transfer the call through and it never goes through it, it ignores it I think because I'm using the softkey feature and not the * feature for xfers. |
| 21:16.09 | jaytee | bkw, I'm thinking because I'm using the softkey SIP feature of the Polycom it's handling the redirect outside of my * dialplan and * only sees it as an outbound call. |
| 21:16.22 | bkw_ | yep thats exactly why |
| 21:17.15 | jaytee | and it doesn't look like I can set my Polycoms to only allow transfers of 4 digit numbers and block everything else but I may have missed something in the hundreds of pages of documentation. |
| 21:17.46 | jaytee | I'm tired of flipping back and forth in the PDF, I'm gonna kill me some trees and print the Polycom Admin guide out. |
| 21:18.06 | jaytee | cuz I is po! and can't afford a Kindle |
| 21:18.42 | bkw_ | jaytee: thats because your dialplan in the thing makes it send when you hit 4 digits |
| 21:18.43 | jaytee | lmadsen, bkw_, ManxPower, thanks all for your input on this. |
| 21:19.00 | bkw_ | fix the dialplan in the polycom and it'll let you transfer to any number of digits |
| 21:20.08 | jaytee | bkw, the dialplan in the Polycoms is even less intelligent than Forrest Gump. It can't distinguish between dialing a 7 digit number or transfering a call to a 7 digit number. THAT is the distinction and permission I need to enable. |
| 21:21.01 | jaytee | anyway, I'll be back later, gotta get some dinner |
| 21:28.24 | *** join/#asterisk lalmia (n=lalmia@64.129.97.83) |
| 21:29.16 | lalmia | oh gawd |
| 21:33.21 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 21:33.46 | pcrane | morning all |
| 21:33.55 | lalmia | evening |
| 21:34.28 | *** part/#asterisk lalmia (n=lalmia@64.129.97.83) |
| 21:35.53 | *** join/#asterisk geek_cl (n=geek@190.54.42.62) |
| 21:37.29 | *** join/#asterisk itakinet7 (n=itakinet@adsl-065-005-186-231.sip.asm.bellsouth.net) |
| 21:38.40 | geek_cl | hi guys, how can see http://fpaste.org/paste/3434 please ? |
| 21:42.18 | [TK]D-Fender | geek_cl: You are missing ncurses / ncurses-devel |
| 21:42.19 | CanWood | *** Install ncurses to use the menu interface! *** |
| 21:42.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 21:42.27 | CanWood | what he said |
| 21:42.30 | geek_cl | not really |
| 21:43.13 | itakinet7 | * 1.4.21.1 DTMF issue.. Per the sip trace Polycom 501 to * looks good, Asterisk out to the SIP trunk resends each DTMF event exactly 6 times in succession.. Any ideas ? |
| 21:43.27 | *** join/#asterisk BipBip (n=BipBip@bl10-109-117.dsl.telepac.pt) |
| 21:43.50 | itakinet7 | dtmfmode=rfc2833 |
| 21:44.41 | russellb | 3 begin and 3 end |
| 21:44.41 | russellb | ? |
| 21:45.17 | russellb | that would be expected behavior ... it retransmits them with the same sequence number to help make sure it gets there, heh |
| 21:45.36 | russellb | that's how asterisk rfc2833 has always worked AFAIK, and is perfectly fine standards wise |
| 21:45.58 | geek_cl | <[TK]D-Fender>: astealix:~# apt-get install libncurses5-dev |
| 21:45.58 | geek_cl | Leyendo lista de paquetes... Hecho |
| 21:45.58 | geek_cl | Creando �rbol de dependencias... Hecho |
| 21:45.58 | geek_cl | libncurses5-dev ya est� en su versi�n m�s reciente. |
| 21:47.33 | [TK]D-Fender | geek_cl: ncurses-devel |
| 21:48.51 | geek_cl | astealix:/usr/src/asterisk-addons# apt-get install ncurses-devel |
| 21:48.51 | geek_cl | Leyendo lista de paquetes... Hecho |
| 21:48.51 | geek_cl | Creando �rbol de dependencias... Hecho |
| 21:48.51 | geek_cl | E: No se pudo encontrar el paquete ncurses-devel |
| 21:49.05 | geek_cl | any repo for debian etch? |
| 21:56.39 | itakinet7 | If I call my cell phone I hear 6 beeps of dtmf, I can not navigate ivr's due to the issue |
| 21:57.00 | itakinet7 | If i press 2 I get 6 2's |
| 22:03.29 | pcrane | geek_cl: try etch main contrib non-free |
| 22:03.39 | pcrane | it should be in that lot somewhere |
| 22:04.40 | geek_cl | thanks pcrane |
| 22:05.20 | *** join/#asterisk lucky|aba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 22:06.02 | geek_cl | not appear |
| 22:06.41 | pcrane | deb http://ftp.nz.debian.org/debian/ lenny main contrib non-free |
| 22:06.42 | pcrane | deb-src http://ftp.nz.debian.org/debian/ lenny main contrib non-free |
| 22:06.42 | pcrane | deb http://security.debian.org/ lenny/updates main contrib |
| 22:06.42 | pcrane | deb-src http://security.debian.org/ lenny/updates main contrib |
| 22:06.42 | pcrane | deb-src http://http.us.debian.org/debian stable main contrib non-free |
| 22:06.54 | pcrane | that's all I've got in my sources.list |
| 22:07.36 | geek_cl | i have etch , is not problem with lenny repo ? |
| 22:07.41 | geek_cl | pcrane |
| 22:07.53 | pcrane | just substitute etch for lenny |
| 22:09.12 | geek_cl | thanks |
| 22:09.38 | itakinet7 | Seems to work fine like this: Useragent <-- dtmfmode=info ---> * <----- dtmfmode=rfc2833 ---> SIP Provider |
| 22:10.39 | hardwire | when a number is routed to your PRI, and you recieve it as the number called in your PBX.. Is that called the DNIS, typically? |
| 22:10.51 | hardwire | I thought DNIS was in-band stuff used when playing with toll-free numbers. |
| 22:11.03 | *** join/#asterisk macros73_ (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 22:12.27 | pputman- | hardwire, your description of DNIS is accurate. |
| 22:12.37 | pputman- | http://en.wikipedia.org/wiki/Dialed_Number_Identification_Service |
| 22:14.16 | hardwire | pputman: I thought DNIS could also be offered by your telco as a way of sending modulated indications once the line was answered so that PBX's could decipher it regardless of signaling type |
| 22:14.26 | hardwire | have you guys ever performed that sort of wicked stunt? |
| 22:14.47 | hardwire | I'd love to have numbers we forward to 1-800 numbers eventually be identified correctly at some point along the path |
| 22:15.42 | pputman- | hardwire, no idea man |
| 22:24.37 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 22:25.10 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 22:28.30 | tompaw | Hi guys. |
| 22:28.52 | tompaw | Basing on yesterdays suggestions, I got myself some new toys today: SPA-3102 and SPA-922 :-) |
| 22:29.01 | tompaw | Anyone here using 3102? |
| 22:29.02 | pcrane | the 922s are great |
| 22:29.25 | tompaw | pcrane: really? just unpacked it and made some sip calls to my local asterisk ;-) |
| 22:29.37 | tompaw | looks like a piece of really solid equipment. |
| 22:29.45 | pcrane | it is |
| 22:29.51 | pcrane | I |
| 22:30.14 | pcrane | I've used some cheap chinese phones... the linksys feel much more solid and more reliable than those chinese ones |
| 22:31.03 | tompaw | coolio :-) I was trying to get 942, but they didn't have it :-( |
| 22:32.23 | tompaw | now, I am trying to answer all incoming PSTN calls on -3102 automatically and forward them to my local asterisk |
| 22:32.29 | tompaw | I wonder if it's possible. |
| 22:32.32 | *** join/#asterisk hi365_m (n=hi365@213.151.61.31) |
| 22:34.49 | ManxPower | tompaw: 1) yes it is possible 2) no, it's not easy and 3) no, I don't remember how I did it when I used the SPA-3102 |
| 22:35.13 | ManxPower | there are several guides to doing it. |
| 22:37.05 | tompaw | I think I just found one ;-) |
| 22:37.09 | *** join/#asterisk hou_cj (n=chatzill@12.183.217.250) |
| 22:37.48 | hou_cj | hi is there a voip application with name similar to nu switch or gnu switch that can allow you to have sip channel and a yahoo or aim client etc |
| 22:37.59 | hou_cj | and with asterisk as well |
| 22:41.34 | andrewy | what is the asterisk 1.2 equivalent of 'dialplan reload'? i thought that worked in 1.2, but it isn't |
| 22:43.24 | tompaw | yey! I forwarded it :-) |
| 22:43.32 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 22:43.53 | tompaw | now it answers the PSTN call and gives me a signal. I wonder if it's possible to automatically CALL Something on the asterisk |
| 22:45.25 | *** join/#asterisk mandd (n=moo@bas1-toronto61-1279436026.dsl.bell.ca) |
| 22:46.30 | mandd | Hello, I set callerid="test" in sip.conf in [general], but it is not working |
| 22:46.39 | mandd | still get unknown name when call |
| 22:46.43 | mandd | anything i am missing? |
| 22:46.50 | mandd | using 2 x sip phones |
| 22:51.18 | tompaw | Is there a way to allow multiple logins for the same SIP Account? |
| 22:55.14 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:56.11 | unpaidbill | yes tompaw busy-level, http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 22:56.29 | unpaidbill | and call-limit |
| 22:58.58 | itakinet7 | I am not able to get dtmfmode=info to work with voicemail in * 1.4.21.1 am I missing something? |
| 23:08.11 | x86 | itakinet7: why not use RFC mode? |
| 23:12.28 | itakinet7 | Useragent <-- dtmfmode=rfc2833 --> ASTERISK <--- dtmfmode=rfc2833 --> SIP Provider |
| 23:13.18 | itakinet7 | DTMF is sent 6 times to SIP Provider |
| 23:13.53 | itakinet7 | If I press 2 I hear 6 2's on the other end |
| 23:14.34 | *** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com) |
| 23:14.45 | itakinet7 | Useragent <-- dtmfmode=info ---> * <----- dtmfmode=rfc2833 ---> SIP Provider works flawlessy to the SIP Provider |
| 23:15.01 | itakinet7 | but not on the local * box |
| 23:16.10 | *** join/#asterisk dmz (n=dmz@216.16.220.180.dyn-cm-pool48.pool.hargray.net) |
| 23:17.29 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:18.37 | *** join/#asterisk dieno (i=771e6a3f@gateway/web/ajax/mibbit.com/x-0c856736ddd6ed91) |
| 23:35.25 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:35.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- CentOS 5.2 Zaptel troubles:type ~centos52bug -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21.1 (2008/06/30) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX |
| 23:36.32 | *** join/#asterisk seanmh (n=johndoe@c-68-35-21-64.hsd1.nm.comcast.net) |
| 23:39.58 | tompaw | unpaidbill: thanks. |
| 23:43.24 | *** join/#asterisk IamTux (n=kman@64.5.9.121) |
| 23:45.26 | IamTux | I am trying to have a sip extension dial out a specific zap channel only (e.g. ext 101 dials out zap/1 ext 102 out zap/2... etc.) |
| 23:45.32 | IamTux | any suggestions ? |
| 23:46.39 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:52.07 | *** join/#asterisk IamTux (n=kman@64.5.9.121) |
| 23:52.59 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:54.20 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 23:59.40 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 23:59.51 | teknoprep | does asterisk 1.6-beta9 take awhile to build ? |