| 00:00.54 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 00:01.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 00:07.58 | *** join/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
| 00:08.14 | John_Clay | Hello |
| 00:08.15 | John_Clay | Hopefully someone is around, else I'm talking to myself. |
| 00:09.20 | John_Clay | I'm looking to have a linksys SPA3102 bridge between a PSTN line and Asterisk. I've been going by a guide that I found, here, http://www.sunrisetel.net/software/asterisk/HOWTOs/SPA3K-FXO-and-Asterisk.shtml but I've hit an impasse. The adapter seems to detect the incoming call, but does nothing with it. |
| 00:09.34 | John_Clay | AsteriskNOW is saying that the peer (adapter) is available in the logs |
| 00:09.51 | John_Clay | any clues would be much appreciated. |
| 00:11.02 | jaytee | John_Clay, checkout this site: http://forum.voxilla.com/linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html |
| 00:11.18 | John_Clay | ahahaha |
| 00:11.19 | John_Clay | awesome |
| 00:11.35 | John_Clay | I googled for about an hour and got nothing that concise. I'll run through it now. |
| 00:11.55 | jaytee | it has an autoconfig but first you have to enable web access from the WAN link side |
| 00:12.14 | jaytee | and it doesn't cost anything to sign up on voxilla |
| 00:12.40 | jaytee | it worked for me, but I'm using Asterisk, not AsteriskNOW. Shouldn't matter though. |
| 00:13.00 | John_Clay | I enabled the web access, but it's deep inside my LAN ;) |
| 00:13.05 | John_Clay | some port forwarding should fix that. |
| 00:13.14 | iamthelostboy | hi.. we are running * on a single server at the moment, with configuration done on a command line though config file.. it looks like we will be expanding the system out to 3 or 4 servers around the world, though the CFO has spoken to another company using Trixbox and wants to head down that path, instead of vanilla *. Is there any downside to this? |
| 00:13.17 | jaytee | I set mine to bridge instead of nat |
| 00:13.39 | John_Clay | Forgive the ignorance, but where's the autoconfig? |
| 00:13.44 | John_Clay | or were you referring to the option on the linksys page |
| 00:13.55 | *** join/#asterisk propellerhead (n=yogurt2u@host209.200-82-99.telecom.net.ar) |
| 00:14.16 | WimpMan | iamthelostboy: Yes, you won't get support here. |
| 00:14.26 | iamthelostboy | :) |
| 00:15.25 | iamthelostboy | everything it can do is possible under *? graphical configuration is also possible under vanilla *? I havent really looked into it too far |
| 00:15.48 | jaytee | John_Clay, http://voxilla.com/tools/device-configuration-wizard/linksys-spa-3xxx-configuration-wizard-for-asterisk-807.html |
| 00:16.28 | LiNeTuX_Home | iamthelostboy: you can install FreePBX on top of 'vanilla' Asterisk |
| 00:16.30 | WimpMan | * does not come with gui. |
| 00:16.55 | LiNeTuX_Home | iamthelostboy: but then you'll have to jump over to #FreePBX |
| 00:17.07 | jaytee | iamthelostboy, * doesn't have a gui but you can install the asterisk-gui used in AsteriskNOW on vanilla asterisk. |
| 00:17.26 | jaytee | either way, you'll end up needing to rework your dialplan probably. |
| 00:17.40 | *** join/#asterisk moy (n=moy@189.169.68.109) |
| 00:17.58 | iamthelostboy | im not too worried about that.. i dont think ive done a spectacular job of setting it up in the first place... |
| 00:18.20 | jaytee | I think you have too! in fact, here's a raise! |
| 00:19.32 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 00:20.59 | [TK]D-Fender | iamthelostboy: With just about every GUI if you don't like the little box they shove you in, forget about changing that situation much. |
| 00:28.30 | John_Clay | jaytee: sadly it didn't work right away ;) |
| 00:29.08 | jaytee | John_Clay, really? it worked ok for me |
| 00:29.29 | John_Clay | mind if I PM? |
| 00:29.31 | jaytee | are you having it register to Asterisk? |
| 00:29.36 | John_Clay | not afaik |
| 00:29.42 | John_Clay | I used the auto config |
| 00:30.21 | jaytee | but there were configuration pieces you had to manually add to Asterisk at the bottom of the config screen |
| 00:30.24 | [TK]D-Fender | John_Clay: Screw config tools and just follow the guides in their forums |
| 00:30.47 | John_Clay | That's what I'm doing now... |
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| 00:39.24 | John_Clay | :o |
| 00:39.41 | John_Clay | it works |
| 00:39.41 | John_Clay | sort of |
| 00:40.57 | *** join/#asterisk chendy (n=chatzill@219.134.230.203) |
| 00:43.17 | John_Clay | hm, ok |
| 00:43.32 | John_Clay | incoming calls now get a "Please wait while I connect your call", followed by a busy signal |
| 00:47.10 | [TK]D-Fender | John_Clay: You should clearly be PASTEBIN-ing your CLI output with SIP debug included.... |
| 00:47.12 | [TK]D-Fender | ~pb |
| 00:47.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:47.14 | [TK]D-Fender | ^^^ |
| 00:47.41 | John_Clay | If only I knew what you meant by the cli output. Referring to logs? |
| 00:48.05 | [TK]D-Fender | John_Clay: No, I mean * CLI. Like how you connect to the running daemon to actually SEE whats going on. Logs = useless |
| 00:48.23 | jaytee | AsteriskNOW = semi-useless |
| 00:48.29 | John_Clay | haha |
| 00:48.41 | [TK]D-Fender | jaytee: Not at all. |
| 00:48.59 | jaytee | the best gui for * is vim with nano a close second |
| 00:49.27 | jaytee | reaching for the rodent just slows ya down |
| 00:49.31 | John_Clay | here we go |
| 00:49.31 | John_Clay | http://pastebin.ca/1221283 |
| 00:57.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:01.56 | _ShrikE | Is there an equivalent for "show g729" that will show the transcoder usage for the TC400B? |
| 01:02.39 | *** join/#asterisk tvirus (i=TheVirus@c-69-243-46-240.hsd1.va.comcast.net) |
| 01:03.18 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 01:03.54 | tvirus | exten = s,2,ExecIf($[${LEN(${CALLERIDNUM})} = 12],AGI,email.sh) <-- How come that never executes my script even though the length is 12? The script itself is fine, if I run it outside of * (It's in /var/lib/asterisk/agi-bin) |
| 01:04.04 | John_Clay | jaytee: Got it working :D |
| 01:04.14 | John_Clay | now answers PSTN calls and tosses them in voicemail |
| 01:04.17 | John_Clay | (as desired) |
| 01:05.39 | John_Clay | Final question, however, is how to get Asterisk to email voice messages |
| 01:06.13 | WimpMan | Just enable it :-) |
| 01:06.41 | John_Clay | How? :) |
| 01:06.46 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 01:06.48 | John_Clay | This is my first time with *, ever. |
| 01:07.20 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 01:07.22 | WimpMan | look into voicemail.conf |
| 01:08.44 | [TK]D-Fender | John_Clay: the sample configs show you the options to enable this |
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| 01:09.42 | John_Clay | Sample configs in asterisk, or in voxilla? |
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| 01:14.55 | [TK]D-Fender | John_Clay: * samples |
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| 01:26.42 | John_Clay | [TK]D-Fender: the voicemail setting I have is 1234 => 4242,Example Mailbox,EMAIL |
| 01:26.53 | John_Clay | the console makes no mention of the emailed message though |
| 01:26.58 | drbrown | I am reading a white paper written by Malcolm Davenport explaining that digium cards are superior to sangoma and rhino as it relates to irq's, don't digium cards still require the system not share irq's? |
| 01:27.26 | carrar | you have sendmail or whatever configured? |
| 01:27.35 | iamthelostboy | cheers for your help... will tell them we should stick to * and they will just have to learn to configure it... |
| 01:28.15 | [TK]D-Fender | John_Clay: there are MORE parameters to read in that config file. Don't just stop because you read the first one you came across. Read the ENTIRE file |
| 01:28.34 | John_Clay | I did, and I altered the second instance of that extension |
| 01:29.12 | [TK]D-Fender | drbrown: Actually Sangoma was the one far ahead of the game. For Rhino cards, just read up on RMA requests. |
| 01:29.24 | [TK]D-Fender | John_Clay: there are MORE parameters to set. GO READ THEM ALL |
| 01:30.13 | drbrown | [TK]D-Fender: are you very familier with sangoma cards? |
| 01:31.40 | [TK]D-Fender | drbrown: Somewhat. |
| 01:31.47 | drbrown | [TK]D-Fender: I am having problems getting one to work with dahdi |
| 01:31.49 | [TK]D-Fender | drmessano: I've installed several |
| 01:32.18 | [TK]D-Fender | drbrown: You will no doubt need to use the very latest wanpipe for this |
| 01:32.27 | [TK]D-Fender | drbrown: As everything was renamed from Zaptel |
| 01:33.51 | drbrown | [TK]D-Fender: I am using the latest stable version august |
| 01:35.22 | drbrown | [TK]D-Fender: It doesn't seen to accept the dahdi source directory |
| 01:36.42 | [TK]D-Fender | drbrown: Check their WIKI for this |
| 01:37.01 | [TK]D-Fender | drbrown: I do not have any Dahdi-specific experience with them |
| 01:37.44 | drbrown | [TK]D-Fender: I am going to try a different version off of their ftp site |
| 01:37.50 | drbrown | [TK]D-Fender: Thanks |
| 01:40.39 | RypPn | drbrown: keep me pasted if you suceed plz |
| 01:41.05 | drbrown | RypPn: will do |
| 01:41.07 | RypPn | posted also |
| 01:41.13 | RypPn | lol |
| 01:42.29 | [TK]D-Fender | http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi |
| 01:42.52 | [TK]D-Fender | Explicit instructions |
| 01:43.23 | [TK]D-Fender | Wow, and an even more explicit direct linked download! |
| 01:43.38 | [TK]D-Fender | And to think I wasted a whole 10 seconds looking for it! |
| 01:43.45 | [TK]D-Fender | runs in circles |
| 01:46.18 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
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| 01:59.55 | adr3nalin3 | Shoud I be using wcte12xp or wctdm24xxp for a PRI T1? Or am I just wrong all together? I am having a hell of a time getting my PRI to talk to * |
| 02:06.02 | jaytee | adr3nalin3, the wcte12xp is for a Digium T1, the wctdm24xxp is for their analog FXO/FXS cards |
| 02:11.21 | adr3nalin3 | jaytee: thank you. that would be why my t1 is failing miserably constantly. |
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| 02:25.47 | riddlebox | has anyone else upgrade to zapte-1.4.12.1 and now their fxs ports on a tdm card dont work? |
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| 02:46.53 | LARefugee | [TK]D-Fender: Hah! I got chan_alsa to work Asterisk 1.6.0 on Ubuntu Intrepid |
| 03:00.40 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:00.40 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:07.31 | John_Clay | Hm, still having issues. I'll try again tomorrow.... |
| 03:07.31 | *** part/#asterisk John_Clay (n=J@unaffiliated/johnclay) |
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| 04:23.02 | scooby2 | hrm hrm hrm |
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| 04:41.22 | cli-work | hi |
| 04:41.31 | andrewyager | hi |
| 04:42.16 | cli-work | i'm having an issue with a TE122 - when we run ztcfg it throws back invalid argument (22) |
| 04:42.28 | cli-work | it's configured as an E1 span |
| 04:47.54 | pputman | cli-work, try modprobe'ing the driver with t1e1override=1 |
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| 04:56.11 | scooby2 | is there a way to use one queue for multiple numbers/companies? different queue announcement/hold music for each |
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| 04:59.19 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 05:00.28 | rrrobert | Has anybody got the experience of working with asterisk and oracle connectivity? or can point me to any tutorial |
| 05:03.09 | cli-work | pputman: where are we best to specify this? Should we add it in /etc/sysconfig/zaptel ? |
| 05:04.40 | cli-work | it's a TE122 |
| 05:06.42 | [TK]D-Fender | rrrobert: To do what? |
| 05:07.11 | rrrobert | through an asterisk application, i just want to query oracle DB |
| 05:07.50 | [TK]D-Fender | rrrobert: Thats something usually best done in AGI |
| 05:08.08 | [TK]D-Fender | rrrobert: In which you can whatever you want in whatever language you want |
| 05:09.41 | rrrobert | [TK]D-Fender, Can I just add the oracle libs and add the oracle headers in the asterisk application, and call the queries |
| 05:09.41 | rrrobert | ? |
| 05:10.25 | [TK]D-Fender | rrrobert: You can setup ODBC and use func_odbc as is described in the BOOK, but I still say for most things, AGI is best |
| 05:12.02 | rrrobert | [TK]D-Fender, hmm you looks a big fan for AGI, I will also try it, Hope it would solve my problem. What book you recommend? |
| 05:12.11 | [TK]D-Fender | ~book |
| 05:12.12 | jbot | somebody said book was 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 |
| 05:12.38 | [TK]D-Fender | rrrobert: in AGI you can do whatever you want because you are in a script outside of * |
| 05:12.52 | [TK]D-Fender | rrrobert: And your description says abosultely nothing abuot the scope of your plans |
| 05:13.07 | [TK]D-Fender | rrrobert: You would have us advise you nearly blind it would seem |
| 05:13.26 | [TK]D-Fender | rrrobert: So naturally I would aim for the tool that limits you the least |
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| 05:15.21 | scooby2 | [TK]D-Fender: our old * consultant mentioned we could combine queues and use different hold music/queue announcements/etc per number instead of having one queue in queues.conf for each number. Is that possible? |
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| 05:16.10 | [TK]D-Fender | scooby2: Might be a way, certainly messy. |
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| 05:17.51 | scooby2 | the issue with multiple queues is * will not always grab the longest waiting call. It seems to jump around queues and if a new call comes in it will sometimes get that before getting around to the longest waiting. |
| 05:18.38 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 05:19.14 | [TK]D-Fender | scooby2: Is your system multi-lingual? |
| 05:19.36 | scooby2 | no |
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| 05:20.07 | [TK]D-Fender | scooby2: then you could make all of your recordings in the format used for alternate language versions, and set the language before entering the queue |
| 05:20.08 | rrrobert | [TK]D-Fender, My plans are quite simple, When the user presses the 1 from the soft phone the asterisk application will make an oracle DB query, that it. I hope my situation is not quite comeplex |
| 05:20.35 | [TK]D-Fender | rrrobert: Do an Oracle query and do WHAT with it? |
| 05:21.06 | rrrobert | [TK]D-Fender, through the AGI ;) |
| 05:21.36 | [TK]D-Fender | rrrobert: .... not HOW, WHAT is ift going to do following this "query"? |
| 05:22.35 | rrrobert | [TK]D-Fender, "query" just inserts a record in the DB |
| 05:23.13 | [TK]D-Fender | rrrobert: That is All it will do? Then func_odbc may very well be enough for this. Both are well documented in the book |
| 05:24.58 | rrrobert | [TK]D-Fender, the most I can do is calling an oracle stored procedure |
| 05:25.43 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 05:25.56 | [TK]D-Fender | rrrobert: Are you trying to tell me the most you WANT to do is that? |
| 05:26.23 | rrrobert | yeah |
| 05:26.45 | [TK]D-Fender | rrrobert: Ok, well you have your answer then |
| 05:27.33 | rrrobert | thanks for the answer |
| 05:27.34 | rrrobert | [TK]D-Fender, Have you worked with oracle instant client? I have configured it and oracle c applications are working fine. Now I will be including the oracle part in the asterisk application. |
| 05:28.04 | rrrobert | through the func_odbc, am I right? |
| 05:28.17 | [TK]D-Fender | rrrobert: I have answered this repeatedly. |
| 05:28.36 | rrrobert | Sorry, thanks for yr time. |
| 05:29.10 | rrrobert | trying what [TK]D-Fender told me to do. ;) |
| 05:29.36 | [TK]D-Fender | ok, I'm done for the night. |
| 05:29.38 | [TK]D-Fender | Later all |
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| 05:45.15 | patrick-- | hey all! is it possible to write a dialplan based on time? like if time laster XX:XX do YY else do ZZ |
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| 05:54.15 | rrrobert | I am connecting to a sip server from sip api, the problem that i am having is, It just connects and automatically disconnects from the sip server. I can see the connect and dissconect messages on the asterisk CLI. Any suggestions? |
| 05:58.33 | LARefugee | patrick--: You betcha. Checkout extensions.conf in the samples. |
| 05:59.23 | patrick-- | gnah samples :D |
| 06:00.04 | patrick-- | doubt i sill have them |
| 06:00.21 | LARefugee | patrick--: It's the first place I go. Just about everything you can do with Asterisk is in there. |
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| 06:05.04 | patrick-- | LARefugee: im trying to write an IVR |
| 06:05.43 | patrick-- | LARefugee: im trying to write an IVR |
| 06:05.54 | patrick-- | on the very first context ive got to: http://pastebin.ca/1221483 |
| 06:06.06 | patrick-- | do i have to re-do the timeouts when jumping into another context? |
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| 06:08.42 | LARefugee | patrick--: No I don't think so |
| 06:09.45 | LARefugee | patrick--: What distro do you use. You need to have the demo from the samples. It's required reading. |
| 06:10.07 | *** join/#asterisk oilinki (n=oil@ppp-124-120-23-100.revip2.asianet.co.th) |
| 06:13.42 | LARefugee | I'm shutting down. Been real... |
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| 06:30.37 | drmessano^ | beta1 FTW |
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| 06:54.22 | Cutlass | is anyone aware of the issues with the "directrtpsetup" option? I've been told that there are known issues with it, but I'm not sure what they are... |
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| 07:12.01 | Madkiss | hi folks, I am left with one single problem with my mISDN-setup. People that call from the outside don't hear a dialtone ... the phone on my side rings but they don't hear the "beep beep" .. |
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| 07:35.04 | dandre | How can I issue a manager redirect command from cli interface? |
| 07:38.40 | *** join/#asterisk robin_sz (n=Robin@212.243.40.130) |
| 07:38.46 | robin_sz | morning .. |
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| 07:39.24 | robin_sz | hi there, if I wanted asterisk to trigger an event in an external application, (you can assume the external app has TCP sockets, MySql access etc available) .. what would be the simplest way, use an agi-bin script for example to insert an event into the MySQL table? or can you trigger MySQL code directly from asterisk? |
| 07:40.03 | robin_sz | would it be hardto get astersik to conect directly the external apps TCP socket and talk directly? |
| 07:40.17 | creativx | robin_sz: why not make the external app monitor asterisk |
| 07:40.25 | hi365 | robin_sz: im not sure i understand your question: do you want asterisk to connect to mysql? |
| 07:40.27 | creativx | that way your external app could misbehave |
| 07:40.32 | creativx | without dragging down asterisk |
| 07:40.55 | kaldemar | dandre: i doubt there is a smart way to use manager commands via the CLI. |
| 07:41.16 | creativx | dandre: use the AMI, not cli. |
| 07:41.42 | robin_sz | hi365, i want to connect * to an external app, im not fussed how I do it, the app already reads MySQL tbales, so I could have * make an entry there if thats going to be easiest ... |
| 07:42.07 | robin_sz | creativx, have a thread in the external app connect to the * AMI port? |
| 07:42.17 | hi365 | robin_sz: asterisk can connect directly to mysql using the mysql command |
| 07:42.23 | dandre | ok |
| 07:42.41 | creativx | robin_sz: yeah, that could be one way to do it. or use an intermediate mysql table |
| 07:42.49 | creativx | depends on how much information/control you require |
| 07:42.54 | robin_sz | h1365, that must be newer than last time I fiddled, I'll go check it out |
| 07:43.16 | dandre | how can I put a channel on hold using the AMI? |
| 07:43.31 | robin_sz | creativx, not much, bascially guy on phone presses *, some event tied to that user happens in external app |
| 07:43.32 | creativx | dandre: redirect it |
| 07:43.52 | dandre | ok but to which extension? |
| 07:44.11 | creativx | dandre: depends, i made my own ghetto parking extension |
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| 07:44.30 | dandre | I don't know how to do that |
| 07:44.31 | creativx | dandre: so when we park calls via our cms app the calls get placed to an extension that only plays music on hold forever |
| 07:46.19 | dandre | do you have an example? |
| 07:48.05 | dandre | How do you unhold the channel? |
| 07:48.28 | mandh | Hi all |
| 07:49.27 | mandh | i am trying to setup asterisk , i installed zaptel and asterisk , when trying to load zap channels "load_resource: Module 'chan_zap' could not be loaded." |
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| 07:49.31 | Dovid | was this issue ever fixed or there will be a fix for it ? |
| 07:49.31 | Dovid | http://bugs.digium.com/view.php?id=13042 |
| 07:50.05 | Dovid | mandh; zap built fine ? |
| 07:50.10 | mandh | i search about it "chan_zap.so" i cant see it |
| 07:50.15 | Dovid | r u using ZTDUMMY ? |
| 07:50.16 | mandh | Dovid, yes and i can load it |
| 07:50.20 | mandh | no |
| 07:50.44 | mandh | Dovid, Wildcard TDM2400P |
| 07:50.53 | Dovid | had the issue while back forgot what I did to fix it |
| 07:51.08 | Dovid | i think i just disabled it in modules.conf |
| 07:51.17 | Dovid | but i was using ZTdummy |
| 07:51.22 | mandh | why to disable it |
| 07:51.59 | Dovid | i idabled cause i didnt need it. never looked in to what was causing it |
| 07:52.27 | mandh | but i need it |
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| 07:53.51 | tzafrir_laptop | mandh, any message right before that? |
| 07:54.07 | tzafrir_laptop | mandh, it's chan_zap*.so* |
| 07:54.33 | mandh | tzafrir_laptop, yes i cant find the file cham_zap* |
| 07:54.42 | mandh | tzafrir_laptop, so it cant load it |
| 07:55.27 | tzafrir_laptop | mandh, 'core show modules like chan_z' |
| 07:55.34 | tzafrir_laptop | what version of asterisk? |
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| 07:55.57 | tzafrir_laptop | Dovid, the report says that it is fixed ... |
| 07:55.59 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 07:56.25 | mandh | tzafrir_laptop, is that command or what |
| 07:56.47 | tzafrir_laptop | yes, a command in the CLI of asterisk |
| 07:56.57 | mandh | tzafrir_laptop, No such command 'core show modules like chan_z' |
| 07:57.02 | tzafrir_laptop | asterisk -rx 'core show modules like chan_z' |
| 07:57.11 | tzafrir_laptop | is it asterisk 1.2? |
| 07:57.56 | mandh | Asterisk 1.6.0 |
| 07:59.05 | creativx | dandre: just think of it as taking the current call and transferring it to a different extension.. which happens to be playing music |
| 07:59.59 | dandre | How do you recall the call? |
| 08:02.29 | creativx | you mean pick it uå? |
| 08:02.29 | creativx | up |
| 08:08.06 | mandh | also i can't find chan_zap enable at menuconfig also at source code related to that , the source is asterisk-1.6.0 |
| 08:16.52 | tzafrir_laptop | mandh, you have chan_dahdi rather than chan_zap |
| 08:17.10 | tzafrir_laptop | do you have 'dahdi' commands in the asterisk CLI? |
| 08:17.14 | mandh | tzafrir_laptop, u mean the newer one |
| 08:17.19 | tzafrir_laptop | yes |
| 08:18.45 | mandh | how i can check it |
| 08:22.52 | mandh | tzafrir_laptop, it is like XXX chan_dahdi |
| 08:22.52 | mandh | <PROTECTED> |
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| 08:24.08 | tzafrir_laptop | mandh, egrep 'TONEZONE|DAHDI' build_tools/menuselect-deps |
| 08:24.40 | tzafrir_laptop | you probably need dahdi-linux and dahdi-tools installed |
| 08:24.44 | mandh | DAHDI=0 |
| 08:24.44 | mandh | TONEZONE=0 |
| 08:25.51 | mandh | tzafrir_laptop, is that an package? or what |
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| 08:31.08 | tzafrir_laptop | mandh, yes |
| 08:31.35 | mandh | tzafrir_laptop, yes i am trying to install it but i that new way is stable? |
| 08:31.41 | tzafrir_laptop | see: http://svn.digium.com/svn/asterisk/branches/1.6.0/Zaptel-to-DAHDI.txt |
| 08:32.08 | tzafrir_laptop | or the same file in the source tree |
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| 08:54.22 | mandh | how to install it with Zaptel instead of DAHDI |
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| 08:58.53 | casix | hello |
| 09:01.37 | petererer | hmm, on a fresh install of asterisk, there's a lot of default config and samples. |
| 09:01.58 | petererer | should i be moving them out of the way to start from scratch, using them as examples? |
| 09:02.24 | casix | I have a dial with the 'r' option, but it doesn't make the dial tone it make the tone of the phone i'm calling. In the debug there is no error... how can I make asterisk to make the ring? |
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| 09:09.31 | festr_ | anyone using openser/kamailio? I've problem with redirecting PSTN calls through openser and back to the same asterisk. it seems that the same callid is the problem. I dont want to use redirect because openser must stay in the middle (CDR) |
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| 09:36.23 | patrick-- | <PROTECTED> |
| 09:38.00 | phpboy | I'm using apllication Pickup(); it seems to be answering other channels and not the channel I specified, why would this be? |
| 09:41.01 | mort_gib | phpboy: pastebin |
| 09:41.39 | mort_gib | sip.conf extensions.conf |
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| 09:47.02 | hi365 | tzafrir_laptop: ping, interested in hellping someone set up an astribank? in #freepbx (or should i send him here?) |
| 09:48.07 | tzafrir_laptop | Who? |
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| 09:50.45 | phpboy | mort_gib: I fixed it |
| 09:51.04 | mort_gib | :-) ok |
| 09:51.36 | phpboy | just put two steps for Pickup(), 1. EXTEN@external, 2. EXTEN@internal |
| 09:51.41 | phpboy | works like a charm :D |
| 09:51.54 | mort_gib | ?? |
| 09:52.32 | mort_gib | You REALLY only need one pickup... the ability to pick up a selected extension (internally) |
| 09:52.53 | mort_gib | But it will of course have to be in the same context :-) |
| 10:02.21 | patrick-- | <PROTECTED> |
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| 10:22.33 | fantasticmrfox | I'm not a user of Skype, in fact I discourage people using it... but humour me - is there a way of having Skype as a Trunk in asterisk (i.e. without extra hardware) ? |
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| 11:05.35 | U-238 | hello |
| 11:05.42 | U-238 | I've got an asterisk appliance |
| 11:05.54 | U-238 | which quite happily boots from the internal drive or whatever it has |
| 11:06.00 | U-238 | but ignores the compact flash card |
| 11:06.09 | U-238 | can anyone help? |
| 11:06.13 | U-238 | thanks in advance |
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| 11:34.57 | write_erase | Hi... when I Call my misdn phone number , misdn create 2 communications channels to my sip phone ... any idea how to solve that ? |
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| 11:35.15 | redax | hi |
| 11:35.43 | write_erase | http://pastebin.com/m5d84b6e |
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| 11:36.56 | write_erase | Looks like misdn create 2 channels one on each ports, and then execute the same exten. |
| 11:37.18 | redax | is it possible to make a transfer which is started as attended to be BlindXfer on a PolyCom phone, if transferer hangs up before the ringed party answers? |
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| 12:11.35 | riddlebox | why does asterisk-1.4.22 stop my zap channels from working? as soon as I install it my fxs ports no longer work? |
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| 12:23.25 | tvirus | Is there a difference between exten = and exten => ? |
| 12:24.13 | [TK]D-Fender | tvirus: No |
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| 12:35.26 | Blackvel | hi all. when using Record and Playback, can I use de/welcome and en/welcome instead of just "welcome"? I really want to store the prompts in the subfolders de and en and not in the default directory /var/lib/asterisk/sounds |
| 12:35.43 | riddlebox | anyone else having problems with asterisk-1.4.22 and fxs? |
| 12:35.54 | riddlebox | Blackvel, yes you can |
| 12:36.25 | Blackvel | probably there is no easier way available as to change all the 30 prompts? |
| 12:36.57 | Blackvel | is there any way to dynamically change the recording directory one time for all 30 prompts? |
| 12:39.00 | riddlebox | well you can download the de prompts, then create /var/lib/asterisk/sounds/de, and untar them inside that dir |
| 12:39.27 | Blackvel | recording my own ivr prompts |
| 12:39.37 | Blackvel | well |
| 12:39.44 | *** join/#asterisk wonderworld (n=ww@ip-62-143-38-55.unitymediagroup.de) |
| 12:40.12 | Blackvel | i could record and move all prompts from sounds directory to sub dir "de" and after that move the en prompts to "en" directory |
| 12:40.35 | Blackvel | well, I don't really like the option to record both of them to the default |
| 12:40.41 | Blackvel | directory |
| 12:41.01 | Blackvel | as I may overwrite some of them in a view month |
| 12:41.04 | Blackvel | months |
| 12:42.13 | riddlebox | true |
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| 12:56.13 | LiNeTuX_Home | Anyone have some tips on IVR's being over-sensitive? We have a conferencing system that seems to be echoing the DTMF tones, which the IVR is 'reading' and acting on... any way to fine-tune that sensativity? |
| 12:56.42 | LiNeTuX_Home | This doesn't happen with other conference systems (Infinate, Raindance, etc) |
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| 13:10.59 | ibm2 | hello, i would like install video in my asterisk,someone have an idea |
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| 13:28.11 | jasonwoot | Qwell, agentcallbacklogin question |
| 13:29.50 | wonderworld | hi. i redirect calls from several source numbers to a single target number. is there a way to play a short sound right after i pick up the phone to find out from which source number the call originated from? |
| 13:34.29 | [TK]D-Fender | wonderworld: Your description is so vague that it can't be reliably answered. |
| 13:35.10 | wonderworld | i already found out. the A option in dial is what i wanted. thanks anyway..... |
| 13:35.19 | [TK]D-Fender | wonderworld: Please be very detailed in exactly what device & tech the call comes in on, what answers it, what is meant by "redirect", etc. |
| 13:35.52 | [TK]D-Fender | wonderworld: Yes, A() can prefix a message. Doing it conditional to the evaluated state of the call may be another matter. |
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| 13:40.34 | adr3nalin3 | Good morning everyone, I am having an issue where when I make a call from phones on another subnet I get --> [Oct 7 08:38:19] WARNING[14445]: chan_sip.c:5095 process_sdp: Unable to lookup host in c= line, 'IN IP4 192.168.050.118' someone yesterday suggested that I have a corrupt sdp. Does anybody know a way to fix this? |
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| 13:41.00 | oej | the 050 is bad |
| 13:41.11 | oej | why have a zero in front of the 50? |
| 13:41.23 | oej | Your library treats that as a host name, not an ip address |
| 13:41.31 | oej | and tries to find it in DNS |
| 13:41.50 | adr3nalin3 | oej: ok, on the hostname, and I think the snom wanted 3 digits input |
| 13:42.43 | adr3nalin3 | so it was either 192.168.500.118 or 192.168.050.118 so I opted for the latter. |
| 13:42.45 | [TK]D-Fender | adr3nalin3 : pastebin the ENTIRE call please. |
| 13:43.21 | fukz | no, snom work fine with one digit in IP address |
| 13:43.22 | [TK]D-Fender | adr3nalin3 : we should not be commenting on such a tiny snippet with no information about relevant subnets |
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| 13:44.12 | Tili | is it possible to do only ip auth for incoming sip calls. no user pass |
| 13:44.45 | adr3nalin3 | [TK]D-Fender: at debug 5 that is all I get |
| 13:45.03 | adr3nalin3 | let me try sip debug |
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| 13:47.21 | adr3nalin3 | Here is a sip debug: http://pastebin.com/m2bcece6 |
| 13:47.54 | LiNeTuX_Home | why does a call placed on a FXS channel fail if someone puts a "#" at the end of the dial string? |
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| 13:50.16 | fskrotzki | 19s63 |
| 13:50.49 | [TK]D-Fender | adr3nalin3 : Lets try this again... pastebin the ENTIRE CALL PLEASE. |
| 13:51.03 | adr3nalin3 | [TK]D-Fender: There is no call. Nothing happens. |
| 13:51.19 | [TK]D-Fender | adr3nalin3 No, it is not the entire call. |
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| 13:52.30 | [TK]D-Fender | LiNeTuX_Home: You should pastebin the CLI at debug 10, verbose 10, and include your zapata.conf & inbound dialplan contexts.... |
| 13:52.57 | adr3nalin3 | [TK]D-Fender: Here is debug 12: http://pastebin.com/m188313c3 from the time I press dial until I hangup. |
| 13:54.38 | [TK]D-Fender | adr3nalin3 : Better. Where is that subnet relative to *'s? |
| 13:55.13 | [TK]D-Fender | adr3nalin3 : And does * have any reason to directly talk to the internet? |
| 13:55.39 | adr3nalin3 | We are vpn'ed together, that phone is on .50 asterisk is on .100, I don't believe so.l |
| 13:57.07 | [TK]D-Fender | adr3nalin3 : If * doesn't need to sue the internet directly, set its externip to the internal IP, and set it up as if it were in a NAT scenario, and add "nat=yes" to the peer entry. Then test |
| 13:57.50 | [TK]D-Fender | adr3nalin3 : And I do hope you've tested that your * server can, through other protocols,fully comunicate with that subnet. |
| 13:58.01 | adr3nalin3 | Yes I have |
| 13:59.05 | [TK]D-Fender | adr3nalin3 : Then try what I've suggested in the meantime. It does seem like a config for firmware bug, but you may be able to compensate for it as I described if necessary |
| 13:59.34 | [TK]D-Fender | adr3nalin3 : Of course you should investigate on the phone intensely |
| 13:59.42 | adr3nalin3 | ok thank you, I will give it a try |
| 14:03.20 | *** part/#asterisk dna (n=dna@85.93.10.9) |
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| 14:03.29 | *** join/#asterisk pluesch0r (n=pluesch0@91.186.159.120) |
| 14:04.54 | pluesch0r | evening! i'm using asterisk in conjunction with voiceone. i've got a working setup with sipgate. i recently got a shiny new ENUM number, so now i should be able to work with multiple extensions. unfortunately, every call to the new number gets kicked on the asterisk side with a nasty "Call from '' to extension '100' rejected because extension not found." error. |
| 14:05.15 | pluesch0r | what could i be doing wrong? i'm able to call the 100 extension internally and vice versa. |
| 14:06.05 | [TK]D-Fender | pluesch0r: Because it is not looking in the context you think it is. Go look at your peer setup and SIP debug to see what peer its matching, and in which context it is looking for "100" |
| 14:07.04 | pluesch0r | [TK]D-Fender: i already did a sip set debug but couldn't make any meaning of the output. i'll have a look at the peer setup .. |
| 14:07.27 | [TK]D-Fender | pluesch0r: www.pastebin.com <- Show us all of what I jsut mentioned and we may be able to tell you |
| 14:08.15 | *** join/#asterisk moy (n=moy@nat/ibm/x-4a6ad10ac7d90161) |
| 14:08.44 | v4mp | hmm suppose should finish my setup, i have it setup so the agent has to login which works fine but when theres an incoming call its trying to find the agent thats logged in but its not ringing the agents phone so the agent cant aswer any idea what this could be ? |
| 14:09.48 | pluesch0r | [TK]D-Fender: thanks for caring. one sec. |
| 14:10.26 | *** join/#asterisk panerai_go (n=neo@88-149-176-89.static.ngi.it) |
| 14:10.27 | pluesch0r | http://pastebin.com/d8e70abd <= sip show peers |
| 14:11.37 | panerai_go | Hi, I need to create a prepaid card application with asterisk. Has anyone sorted out the concurrent calls on single account problem? |
| 14:12.24 | [TK]D-Fender | pluesch0r: No, I mean your sip.conf peer. And all the debug info as well... |
| 14:12.33 | Madkiss | hi pluesch0r |
| 14:12.48 | [TK]D-Fender | panerai_go: Problem? What problem? |
| 14:13.01 | pluesch0r | hey Madkiss |
| 14:14.01 | pluesch0r | http://pastebin.com/d7ebe639d <= sip debug info when trying to call the enum number |
| 14:14.38 | Madkiss | well, obviously it does not find the 100-extension ;) |
| 14:15.20 | pluesch0r | [TK]D-Fender: the sip configuration for the sipgate peer seems to be saved in the database. |
| 14:15.26 | [TK]D-Fender | pluesch0r: found no matching peer or user for '212.183.31.134:5060' <- It does not know who the call is coming from |
| 14:15.31 | pluesch0r | Madkiss: yeah .. question is: why. |
| 14:15.46 | [TK]D-Fender | pluesch0r: Looking for 100 in default (domain sip.mydomain.com) <- Here's the context its looking in |
| 14:15.57 | pluesch0r | [TK]D-Fender: aha! so .. how do i make it accept the call? |
| 14:16.01 | [TK]D-Fender | pluesch0r: SIP/2.0 404 Not Found <- and here is tragic failure |
| 14:16.23 | panerai_go | [TK]D-Fender: you have one customer with 60 seconds left, he makes the call and you can set a limit time (60 seconds ) through a dial parameter so that when 60 seconds are reached asterisk hangs up the channel. With single calls it works great but if a call arrives before the 1st call is ended it will recieve a time limit of 60 seconds again so at the end the balance will be -60 seconds |
| 14:16.27 | pluesch0r | [TK]D-Fender: unfortunately, i'm able to call 100@sip.mydomain.com from my softphone. |
| 14:16.33 | pluesch0r | (which has extension 301) |
| 14:16.43 | pluesch0r | i'm just not able to call from the outside .. |
| 14:17.10 | [TK]D-Fender | pluesch0r: Yes because your softphon IS recognized and does not USE [default] |
| 14:17.13 | pluesch0r | how do i make asterisk accept incoming calls from unknown peers (as i think that's what i want with the whole ENUM stuff) |
| 14:17.35 | Madkiss | add a context for 100 in the default-context |
| 14:17.40 | [TK]D-Fender | pluesch0r: This is YOUR dialplan. You see what context it points to. CHANGE IT. |
| 14:17.42 | Madkiss | err, an extension for 100, i mean. |
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| 14:18.32 | *** join/#asterisk zydoon (n=zydoon@41.225.153.134) |
| 14:18.58 | [TK]D-Fender | panerai_go: No, you cannot simultaneously downgrade consecutive calls without a massive time-keeper process that will monitor all calls. |
| 14:19.21 | *** part/#asterisk zydoon (n=zydoon@41.225.153.134) |
| 14:19.26 | jasonwoot | anyone know the license fee to add a queue in isymphony? |
| 14:19.41 | panerai_go | [TK]D-Fender: and this hasn't been implemented yet, right? |
| 14:19.42 | [TK]D-Fender | panerai_go: Because you COULD see how long the first call was in call for and subtract, but then you'd have to subtract on both for the duration. Real pain. |
| 14:19.52 | [TK]D-Fender | panerai_go: This isn't something for * to implement. |
| 14:20.17 | [TK]D-Fender | panerai_go: You would have to write a process that would look at the history and evaluate all calls in progress and then force terminate, etc |
| 14:20.23 | [TK]D-Fender | panerai_go: Indeed a lot fo work |
| 14:20.25 | [TK]D-Fender | of* |
| 14:21.22 | panerai_go | [TK]D-Fender: I'll write an agi, but it'll be a lot resource consuming |
| 14:21.55 | [TK]D-Fender | panerai_go: Oh, this timekeeper wouldn't even be AGI. It would ahve to be AMI and monitor EVERYTHING in real-time. |
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| 14:22.06 | [TK]D-Fender | panerai_go: As I said, a LOT of work. |
| 14:22.23 | [TK]D-Fender | panerai_go: Because AGI won't process in the background. |
| 14:23.17 | [TK]D-Fender | panerai_go: on a minute basis you'd have to "add up call durations for active calls", "Check CDR balance", "check overrun", "gracefully terminate calls in progress" |
| 14:23.26 | the_5th_wheel | has anyone here used a standard ISDN modem in asterisk? Or is it so bad i shouldnt even bother? |
| 14:23.41 | Madkiss | the_5th_wheel: you would better shoot yourself in the head |
| 14:23.48 | Madkiss | that's less painful |
| 14:24.01 | tzafrir_laptop | the_5th_wheel, HFC modems work quite nicely |
| 14:24.27 | tzafrir_laptop | naturally you can also shoot Madkiss , which would be less painful |
| 14:24.34 | panerai_go | I hate AMI, i'll keep it outside asterisk, just write a db record with begin date and accountcode at the exten 1 . Then i'll run a daemon to check critcal calls and use ami to hangup them |
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| 14:25.19 | the_5th_wheel | tzafrir_laptop: can you reccommend a specific model of these modems? |
| 14:27.09 | tzafrir_laptop | the_5th_wheel, I'm a bit biased, as I work for a hardware vendor |
| 14:27.42 | the_5th_wheel | may i enquire who? |
| 14:28.00 | tzafrir_laptop | points to the address |
| 14:28.05 | mltlnx | Hello, I want to be able to jump out of voicemail by pressing *. I added exten => a,1,Goto(somewhere) yet it playback invalid |
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| 14:46.51 | [TK]D-Fender | mltlnx: Feel free to show us the complete situation because otherwise we're running blind |
| 14:47.38 | mltlnx | [TK]D-Fender: Thanks, I actually had it correct except that reload were not fully parsing extensions.conf because I closed a macro with a ) instead of a ] |
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| 14:49.28 | hi365 | how can i tell if asterisk 'forked' |
| 14:49.30 | hi365 | ? |
| 14:50.45 | ManxPower | hi365: is Asterisk running? Then it forked at some point. |
| 14:50.57 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 14:51.06 | ManxPower | For regular stuff asterisk just fires off another thread rather than fork a process. |
| 14:51.08 | *** join/#asterisk festr_ (n=festr@ns.hiro.cz) |
| 14:51.51 | ManxPower | Ask a better question and you'll get a better answer. |
| 14:52.00 | hi365 | ManxPower: i see. i have a problem that sometimes there seems to be two instances of asterisk running. if i do service asterisk stop it says [OK] |
| 14:52.13 | hi365 | if i run that command again it still says OK |
| 14:52.27 | hi365 | only n the theird time does it say failed |
| 14:52.38 | hi365 | so im assuming its running twice somehow |
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| 14:53.51 | hi365 | question is how/why is it running twice? |
| 14:54.35 | ManxPower | hi365: it's not running twice, it's failing to kill asterisk the first time. |
| 14:54.49 | ManxPower | Asterisk refuses to run twice |
| 14:54.50 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 14:54.53 | edwin_quijada | Hi! |
| 14:54.57 | *** join/#asterisk ToTo (n=ToTo@207.176.6.110) |
| 14:55.13 | edwin_quijada | somebody here use Dell SC440 server with openvox card? |
| 14:55.18 | ManxPower | Back in the Good Old Days mpg123 was what messed up when asterisk exited. |
| 14:55.25 | hi365 | ManxPower: any specific reason that it would fail to kill it the first time? |
| 14:56.11 | ManxPower | hi365: I can't think of any. Maybe you are running the service asterisk stop before Asterisk fully exits. Have you updated your init script with the one from the currently installed Asterisk (make config will do it) |
| 14:56.21 | petererer | hmm, pgsql does not like some of the sql that is used :( |
| 14:56.54 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 14:56.56 | hi365 | ManxPower: er... its the one that came with asterisk (1.4.18) |
| 14:59.17 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:59.22 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:01.06 | ManxPower | hi365: how long are you waiting before running the stop script a 2nd time? |
| 15:01.14 | hi365 | a second or two |
| 15:01.54 | hi365 | ManxPower: there are also other funny things that happen. for example core show channels wont show the channel count lines... |
| 15:02.18 | *** join/#asterisk [gnubie] (n=[gnubie]@cm141.omega112.maxonline.com.sg) |
| 15:02.18 | hi365 | also other cli command wont exectue properly - then the is no more verbose... |
| 15:02.32 | *** part/#asterisk fiddur (n=fiddur@c042.rit.se) |
| 15:03.01 | petererer | WARNING: nonstandard use of \\ in a string literal |
| 15:03.05 | petererer | LINE 1: SELECT * FROM extensions WHERE exten LIKE '\\_%' AND context... |
| 15:03.06 | petererer | :o |
| 15:03.06 | *** join/#asterisk shinao1 (n=shinao1@193.238.30.36) |
| 15:03.12 | petererer | is that a mysql thing? hhe |
| 15:03.20 | petererer | i'm using postgresql |
| 15:03.23 | *** join/#asterisk sack (n=sack@106.Red-88-24-156.staticIP.rima-tde.net) |
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| 15:04.45 | edwin_quijada | petererer: what do u trying to do? |
| 15:06.40 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
| 15:07.16 | ManxPower | hi365: fix the other problems first |
| 15:07.28 | hi365 | what other problems? |
| 15:07.29 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:07.35 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:07.38 | ManxPower | (10:01:55 AM) hi365: ManxPower: there are also other funny things that happen. for example core show channels wont show the channel count lines... |
| 15:07.41 | ManxPower | those problems |
| 15:07.56 | ManxPower | petererer: try removing the extra / |
| 15:07.59 | hi365 | hmm, so your saying there not related? how do i go about ficing them then? |
| 15:08.12 | ManxPower | hi365: Oh I'm sure they are related. |
| 15:08.32 | hi365 | regardless, how do i go about fixing them? |
| 15:08.40 | ManxPower | fix the other problems and I bet the stop problem will go away. |
| 15:08.55 | hi365 | how? |
| 15:08.57 | ManxPower | hi365: since I've seen a system that screwed up I would not be able to help you with those issues. |
| 15:09.19 | ManxPower | ..er.. since I have NEVER seen a system as screwed up as your system |
| 15:10.23 | hi365 | yup - same issues on another system. |
| 15:10.42 | ManxPower | then I suspect it's a config issue either either with asterisk.conf or modules.con |
| 15:10.44 | ManxPower | modules.conf |
| 15:11.04 | hi365 | service...stop: ok; service..status: running :( |
| 15:12.42 | hi365 | ManxPower: anything wron here? http://pastebin.ca/1221775 |
| 15:12.45 | hi365 | *wrong |
| 15:13.16 | hi365 | also, after stop asterisk just once - the cli issues go away |
| 15:13.23 | mort_gib | Is there a list for what modules are now loaded when autoload in on and maybe what the all ldo?? |
| 15:14.05 | ManxPower | There's an mp3 entry: load => format_mp3.so |
| 15:14.12 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 15:15.00 | XnOSX | anybody here know about the site for download music on hold audio files? |
| 15:15.08 | petererer | ManxPower: where is it specified? |
| 15:17.07 | hi365 | ManxPower: i have AA installed, so i doubt thats an issue... AND: after stop asterisk just once - the cli issues go away |
| 15:19.23 | petererer | ManxPower: oh, it's part of pbx_realtime.c :o |
| 15:20.47 | ManxPower | AA? |
| 15:20.53 | hi365 | asterisk addons |
| 15:20.59 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-e0e676a8d9f9f503) |
| 15:20.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:22.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:29.26 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 15:29.29 | freezey | i have my 7940G phone unlocked on the SIP POS-08-02-00 software and i am wondering how i can change the IP and set it statically |
| 15:30.07 | UnixDawg | threw the phne gui |
| 15:30.24 | UnixDawg | there is a manual you can read also |
| 15:31.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:31.37 | freezey | UnixDawg: you have internet copy? |
| 15:35.16 | UnixDawg | google it .. its out there |
| 15:35.30 | UnixDawg | heading for lunch meeting bbl |
| 15:38.31 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
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| 15:48.34 | *** part/#asterisk dmg123 (n=mechanix@89-28-113-143.starnet.md) |
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| 15:51.54 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:54.02 | Blackvel | anyone using ldapget? trying to set it up with openldap and using in parallel csv2ldiff perl script. what do you use as base? cn=x,cn=com or ou=xxx,o=xxx? |
| 15:54.05 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 15:54.31 | Blackvel | I don't understand too much about ldap and how to exactly create tree elements (even what all this ou/o flags mean) |
| 15:55.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
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| 16:07.49 | *** join/#asterisk marc7 (n=marc@S0106001ff33f8523.vc.shawcable.net) |
| 16:07.59 | marc7 | what's the 1.6 equivalent to sip set debug? |
| 16:08.17 | russellb | sip set debug on? |
| 16:08.17 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:08.21 | Qwell | yes |
| 16:08.30 | marc7 | arg... that was painful. thanks. |
| 16:08.42 | [T]ank | anyone here use online fax service? looking for a provider that will port local us DID numbers to them. |
| 16:09.18 | marc7 | by the way guys, congrads on getting 1.6 out the door. |
| 16:09.33 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:09.51 | *** join/#asterisk geraint_ (n=geraint@81.145.128.115) |
| 16:10.08 | Qwell | marc7: thanks |
| 16:14.04 | malaiwah | I have a question about asetrisk 1.6, i'm wondering if handling of calls using SIP INFO (dtmfmode) was improved. In asterisk 1.2 and 1.4 (from what I experimented), even if the general dtmfmode parameters tells to use SIP INFO for dtmf signaling, Asterisk will make two extensions connect by itself if using the "t" or "w" (features) dial optinos, instead of making them talk to each other directly. Is this something that has changed in 1.6 |
| 16:14.04 | malaiwah | <PROTECTED> |
| 16:18.20 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:19.58 | *** join/#asterisk sky_blue (n=ayates@cpc4-bexl4-0-0-cust504.bmly.cable.ntl.com) |
| 16:21.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:22.29 | sky_blue | i'm having problems with direct dialling into a meetme room, i get alison asking for pin. enter it then a male voice announcing transfer then call gets binned. any ideas? |
| 16:24.15 | sky_blue | btw... meetme works ok from extensions, it's just direct dial in that fails |
| 16:26.19 | *** join/#asterisk sircco (n=sircco@dh207-70-239.xnet.hr) |
| 16:27.24 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-25-109.hsd1.nj.comcast.net) |
| 16:27.36 | sircco | i have 3 bri channels connected to asterisk, using misdn. how can i set outgoing cid for each extension? Problem is that each bri line has 3 numbers assigned to it and i always get numbers from first free bri |
| 16:29.14 | *** join/#asterisk DJGg (n=IceChat7@c95162db.virtua.com.br) |
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| 16:37.55 | cesar_CR | hello guys... can somebody explain me the SAY DATE AGI command ?? |
| 16:39.44 | cesar_CR | I need to have an extention that say the date |
| 16:41.12 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 16:41.23 | *** join/#asterisk moy (n=moy@nat/ibm/x-9c1f93455295b3bb) |
| 16:43.17 | hardwire | hai |
| 16:43.40 | hardwire | cesar_CR: still using .net? :) |
| 16:44.17 | cesar_CR | hardwire, hi , .net ??? no way :) |
| 16:45.02 | Blackvel | sircco: I dont run it...but can't you use misdn/MSN:{EXTEN}...? |
| 16:45.02 | hardwire | oh.. wrong cesar :) |
| 16:45.09 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) |
| 16:45.13 | hardwire | cesar_CR: what do you have set up now? |
| 16:45.35 | sircco | Blackvel: yeah but problem is that i wont get that extension unless im at right bri that owns those extensions |
| 16:45.36 | cesar_CR | hardwire, yes wrong cesar :) |
| 16:45.37 | Blackvel | sircco: what happens when you set the callerid by Set(CALLERID(num)=MSN? |
| 16:45.59 | sky_blue | i'm having problems with direct dialling into a meetme room, i get alison asking for pin. enter it then a male voice announcing transfer then call gets binned. any ideas? btw... meetme works ok from extensions, it's just direct dial in that fails |
| 16:46.07 | hardwire | Blackvel: nothing? |
| 16:46.13 | sircco | Blackvel: i get it if im at right bri.. otherwise i just get first free number form free bri |
| 16:46.31 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:46.34 | hardwire | sky_blue: weird.. what PBX system are you using? |
| 16:46.54 | sky_blue | hardwire: asterisk 1.4.22 |
| 16:46.55 | sircco | Blackvel: wonder how come it worked before asterisk on plain old pbx |
| 16:46.57 | Blackvel | when I run my zap devices with bristuff....I used to set any MSN which belongs to telco line connect |
| 16:47.13 | Blackvel | sircco: do you mean that you have three telco bri lines? |
| 16:47.19 | hardwire | sky_blue: not using any management addon or anything? |
| 16:47.22 | sircco | Blackvel: exactly |
| 16:47.42 | hardwire | cause I don' remember there being any male voices, unless maybe you are using a different language than en_US |
| 16:47.49 | Blackvel | sircco: how do you control what line (1-3) to use for outbound call? |
| 16:47.51 | *** join/#asterisk ManxPower (n=manxpowe@194.sub-70-220-56.myvzw.com) |
| 16:47.53 | sircco | Blackvel: i have 4 port bri card, 3 lines connected |
| 16:47.54 | sky_blue | hardwire: no nothing, very basic setup, this system is just for conferencing |
| 16:48.08 | sky_blue | i'm in uk |
| 16:48.16 | sircco | Blackvel: i don't .. i just connect to first free on mISDN channel |
| 16:48.18 | ManxPower | Anyone have any idea what would cause this "linux/drivers/dahdi/dahdi_dynamic_eth.c:104: error: too many arguments to function âskb_linearizeâ" |
| 16:48.19 | Blackvel | sircco: is that for outbound dailing? |
| 16:48.24 | sircco | Blackvel: yes and incoming |
| 16:48.48 | hardwire | wonders if there are male prompts anywhere in the asterisk base sounds for english |
| 16:49.05 | *** part/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:49.17 | Blackvel | is there any group g1 on misdn like zap? |
| 16:49.35 | sircco | similar... everything is here in one group, 3 x 3 numbers |
| 16:49.39 | Blackvel | do you bundle all 6 b channels into one group? |
| 16:49.47 | sircco | 3 numbers for each bri.. 3 bri |
| 16:49.50 | Blackvel | oh okay |
| 16:49.50 | hardwire | sky_blue: whats your dialplan look like? |
| 16:49.51 | sircco | yes |
| 16:50.15 | sky_blue | hardwire, i set my default options to uk, and may have d/l additional sound packages, i can't remember exactly.... |
| 16:50.28 | ajohnson | jtodd: Ping |
| 16:50.37 | sky_blue | i've tested this on 2 different * boxes... both the same |
| 16:50.41 | hardwire | sky_blue: weird.. I just don't remember them, I guess. |
| 16:50.47 | hardwire | either way.. dialplan me |
| 16:52.34 | ajohnson | Anyone here have information on how to configure a Cisco 7942? I'm trying to upgrade the firmware and some of the documentation is pretty scarce |
| 16:52.47 | sky_blue | <PROTECTED> |
| 16:52.47 | sky_blue | <PROTECTED> |
| 16:52.47 | sky_blue | <PROTECTED> |
| 16:52.50 | sircco | Blackvel: here is how it looks like http://pastebin.com/d340aa459 |
| 16:53.00 | Blackvel | sircco: good question. there are no options in capi.conf / misdn.conf like zapata.conf? how about splitting into three groups and testing before if the channels are busy so you know what line you use...then you could probably even set the correct callerid manually |
| 16:53.31 | sircco | can you do this with zapata.conf? |
| 16:54.44 | hardwire | sky_blue: does it act normally (play MoH) when just one user is in the conference? |
| 16:55.10 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:56.13 | sky_blue | if the user(s) are on extensions either hard/soft phones and dialing xtn 3000 all works perfect, it's just when i direct dial from the ddi there is a problem |
| 16:56.53 | hardwire | maybe there is a hang when "introducing" a funky caller id |
| 16:57.01 | hardwire | kill the c and i flags and try again |
| 16:58.51 | *** join/#asterisk ew01f (n=chatzill@201.170.36.149) |
| 16:59.31 | Blackvel | hmm.... can't find old examples anymore....callerid option for zapata.conf is not described like this on voip-info.org. it seems to be for incoming. not outgoing |
| 16:59.53 | Blackvel | so probably I did it manually in exensions.conf (but didn't have 3 telco lines) |
| 17:00.57 | Blackvel | ChanIsAvail |
| 17:01.00 | sircco | Blackvel: this is probably something common, and i guess people solved it somehow.. |
| 17:01.03 | Blackvel | I would go for this |
| 17:01.08 | Blackvel | think so too |
| 17:01.09 | sircco | Blackvel: i think i'll split in groups and use as you said |
| 17:01.17 | sircco | no other idea |
| 17:01.21 | sky_blue | hardwire: that time i got alison announcing transfer, then dial tone, then call binned with c and i flags removed |
| 17:01.36 | hardwire | what does alison say? |
| 17:01.39 | Blackvel | not me. but one using multiple T1/E1 probably has the same problem |
| 17:01.40 | Blackvel | as you |
| 17:01.48 | sky_blue | "transfer" |
| 17:01.52 | hardwire | I'm wondering if you have multiple of the same extensions in your dialplan |
| 17:02.02 | hardwire | cause meetme doesn't ever say "transfer" |
| 17:02.04 | hardwire | afaik |
| 17:02.13 | hardwire | it just poops you into a meetme :) |
| 17:02.20 | Blackvel | sircco: does it work any good? what card? no echos? |
| 17:02.28 | hardwire | they are incredibly handy due to the insta-poop feature |
| 17:02.36 | sircco | Blackvel: openvox b400p works great and it's cheap |
| 17:02.42 | sircco | Blackvel: no problems for month so far |
| 17:03.18 | sircco | Blackvel: first here were some problems with echo but i raised echocancel to max and now it works great, customer aint complaining :) |
| 17:03.30 | Blackvel | so softecho cancellation? |
| 17:03.47 | Blackvel | mg2 / oslec / octasic / hpec ? |
| 17:03.55 | sircco | oslec |
| 17:04.12 | sircco | and i use grandstream phones |
| 17:04.14 | sircco | some 20 phones |
| 17:04.16 | Blackvel | tried to get junghanns duobri working before with mg2 and octasic...no way |
| 17:04.40 | sircco | you can try this openvox, people said it's good so i went with the flow |
| 17:04.41 | Blackvel | only patton gateway resolved my problems with snom phones |
| 17:04.44 | sircco | and bought one |
| 17:04.53 | Blackvel | nice to hear |
| 17:04.54 | sircco | what problems you had with snom? |
| 17:04.55 | sircco | echo? |
| 17:05.00 | Blackvel | on pstn side yes |
| 17:05.04 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:05.08 | hardwire | sky_blue: I'm wondering if dialing in uses a different context than the local extensions |
| 17:05.09 | sky_blue | hardwire: i also have [conf] |
| 17:05.09 | sky_blue | <PROTECTED> |
| 17:05.09 | sky_blue | <PROTECTED> |
| 17:05.09 | sky_blue | <PROTECTED> |
| 17:05.09 | sky_blue | <PROTECTED> |
| 17:05.10 | sky_blue | <PROTECTED> |
| 17:05.12 | sircco | i guess i was lucky :) |
| 17:05.13 | Blackvel | shouldn't normally happen with bristuff |
| 17:05.18 | *** join/#asterisk fukz (n=fukz@p5B062080.dip.t-dialin.net) |
| 17:05.20 | hardwire | sky_blue: pastebin ftw :) |
| 17:05.26 | Blackvel | maybe its my via epia 1 gig cpu (only had 15%) |
| 17:05.35 | hardwire | sky_blue: yeh.. none of that is an announcement. |
| 17:05.36 | sircco | Blackvel: here is... |
| 17:05.48 | sircco | model name: Intel(R) Core(TM)2 Duo CPU E4500 @ 2.20GHz |
| 17:05.59 | hardwire | sky_blue: so check the context that inbound external calls is hitting before trying the context that extension is in. you dig? |
| 17:07.04 | sky_blue | hardwire: they hit default context, it's just for a conf system, and it must be corect otherwise alison wouldn't ask for the pin? right? |
| 17:07.23 | hardwire | oooh.. sorry |
| 17:07.53 | hardwire | sky_blue: I forgot you had a pin |
| 17:08.03 | sky_blue | hardwire: whats pastebin ? |
| 17:08.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:08.18 | hardwire | sky_blue: this wonderful safehaven for your text. |
| 17:08.23 | hardwire | and our sanity. |
| 17:08.35 | hardwire | www.pastebin.ca or .com or whatever. |
| 17:08.42 | sky_blue | ok sorry, never used irc before now |
| 17:08.43 | hardwire | you can paste your stuff there, then it gives you a URL you can paste here |
| 17:08.57 | hardwire | sky_blue: it's not IRC, but it helps IRC stay clean :) |
| 17:09.18 | sky_blue | ok thanks for the tip off! :) |
| 17:10.17 | sky_blue | hardwire: odd thing is nothing is showing up in the cli, core set verbose 999!!!! |
| 17:10.37 | hardwire | sky_blue: I hate to ask .. but |
| 17:10.42 | hardwire | are you even dialing into the right system? |
| 17:11.32 | sky_blue | :-D yes sorry i'm meant the transfer announcement, i can see the call hit the system |
| 17:11.56 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 17:12.13 | hardwire | I don't think meetme displays what sounds it's playing |
| 17:12.22 | hardwire | which would be nice |
| 17:12.53 | sky_blue | hardwire: Playing 'conf-getpin' (language 'en') |
| 17:13.26 | hardwire | why doesn't it show it playing 'transfering' I wonder |
| 17:14.47 | sky_blue | hardwire: i wonder exactly the same, and why is it ok when just using "internal" extensions... the server is remote, all the extensions are ip remote, but you get my meaning? |
| 17:15.29 | hardwire | and calls inbound from the PSTN don't go to the right place eh? |
| 17:17.33 | malaiwah | does anyone know if asterisk honours "a=direction:passive" in a sdp payload ? |
| 17:17.48 | sky_blue | i have a sip trunk with 1 number registered to the * box, when dialling that number i get alison asking for the pin |
| 17:18.30 | sky_blue | if i changed the dial plan to point to a couple of sip phones instead of the meetme room, they work ok |
| 17:19.54 | *** join/#asterisk reallost1 (n=reallost@12-215-208-156.client.mchsi.com) |
| 17:23.24 | reallost1 | Is there and easy way to forward a call to voicemail on a different asterisk server? |
| 17:24.08 | denon | sure, set up an extension and forward to it |
| 17:24.25 | [TK]D-Fender | high-5's denon |
| 17:24.38 | denon | hehe |
| 17:25.05 | reallost1 | hmm... |
| 17:25.47 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 17:26.19 | sky_blue | hardwire: is it normal to hear the dial tone on a xfer? |
| 17:26.29 | hardwire | sky_blue: no |
| 17:26.38 | hardwire | how is your system connected to PSTN? |
| 17:27.25 | sky_blue | it isn't, it's sip to the provider then to their pstn g/w |
| 17:28.11 | hardwire | interesting |
| 17:30.51 | sky_blue | i have 2 * boxes in 2 different locations, one is ubuntu, one is cent os, same problem, the only common denominator is it's the same sip provider, however the calls still hit the * boxes so the calls should just progress normally once * picks up the call (you would have thought) |
| 17:30.51 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 17:31.48 | *** join/#asterisk ManxPower (n=manxpowe@158.sub-75-251-215.myvzw.com) |
| 17:33.10 | talntid | Anyone need a Rhino R1T1 card? |
| 17:33.58 | jameswf | has a warhouse :) |
| 17:34.13 | reallost1 | denon, I can dial the extension@ the voicemail server, but I need to make some sort of different extensions to dial, or it will loop back to myself. |
| 17:34.28 | talntid | jameswf, want to add one more to the warehouse? :P |
| 17:35.14 | hardwire | sky_blue: and you are using Answer() |
| 17:35.16 | hardwire | so I'm lost |
| 17:35.43 | reallost1 | denon, how do I pass in the vmail box number with out creating a new exten for each user? |
| 17:35.48 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 17:36.26 | [TK]D-Fender | reallost1: Here's a though... if you're passing the call over to go to vm, the exten being DIALED should be the VM box number |
| 17:36.29 | ManxPower | reallost1: where is the mailbox number stored? |
| 17:36.37 | blitzrage | reallost1: setvar=VOICEMAIL_BOX=8000 in sip.conf for the peer |
| 17:36.43 | blitzrage | or use AstDB |
| 17:36.47 | [TK]D-Fender | OMG... |
| 17:36.58 | blitzrage | omghi! |
| 17:37.07 | blitzrage | drugsaregoodmmmkthxbye |
| 17:37.10 | [TK]D-Fender | omgrunsforthehills |
| 17:38.28 | drwelby | Any insights on Sangoma vs. Digium for a 4-port FXO card? |
| 17:38.41 | sky_blue | hardwire: i've tried that before, i'll put it back in and see |
| 17:38.44 | jameswf | drwelby: buy (new) digium |
| 17:38.48 | hardwire | runs away |
| 17:40.02 | blitzrage | analog lines suck in general :) |
| 17:40.51 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-92-169.w86-215.abo.wanadoo.fr) |
| 17:40.55 | ManxPower | drwelby: both cards IF NEW should work fine |
| 17:41.03 | ManxPower | OLDER digium cards had issues |
| 17:41.19 | drwelby | Yup, buying new |
| 17:41.26 | blitzrage | just about TigerJet based stuff -- which is the old TDM400P, no longer sold by Digium |
| 17:41.36 | blitzrage | s/about/avoid |
| 17:41.55 | blitzrage | I believe the new card is the TDM410P which doesn't use it |
| 17:42.57 | sky_blue | :) Answer() makes no difference, thinking laterally .... if the conf works with xtens connected to the system, can i make the calls forward to an xten, then frwd to the conf? |
| 17:47.27 | sky_blue | thanks for you help hardwire, anyone else want to jump in on this? |
| 17:50.50 | reallost1 | hmm... |
| 17:51.18 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 17:51.54 | reallost1 | Ah, adding a prefix to the number will do the trick. |
| 17:52.10 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 17:52.17 | reallost1 | then I can just strip off the prefix and have the vm box number, which is the same as the DID, normally. |
| 17:53.17 | reallost1 | thanks for the ideas. |
| 17:54.39 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 17:58.11 | jdnWEST | Anyone using Fring for iphone? |
| 17:58.28 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 17:59.24 | sacitec | i do |
| 17:59.32 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 18:00.27 | adr3nalin3 | is there a way to grab the extension number of someone calling from a sip trunk? This is what I have so far. http://pastebin.com/m1acb9c03 |
| 18:01.25 | adr3nalin3 | Also for some reason it always says asterisk KC Office on the phone. |
| 18:02.04 | reallost1 | it says "asterisk" when there is no caller id and asterisk forwarded the call. |
| 18:02.26 | bkruse | Or when it can't grab callerid |
| 18:02.46 | reallost1 | you need to set ${CALLERID(num)} |
| 18:03.06 | reallost1 | that is why it is saying "asterisk" that should be the digits. |
| 18:03.59 | malaiwah | sacitec: on fring over iphone; does it work flawlessly with asterisk? i used it on S60 only. |
| 18:06.47 | Carlos_PHX | Anyone know off-hand what the filename is for the Allison recording that says "Is that a phone in your pocket, or are you just happy to see me?" |
| 18:07.19 | malaiwah | Carlos_PHX: funny, i didn't even know there was such prompt ;-) |
| 18:07.27 | Carlos_PHX | There are many funny ones. |
| 18:07.32 | Carlos_PHX | So many I forget the names. |
| 18:07.59 | Carlos_PHX | There must be a list somewhere. |
| 18:08.03 | mahlon | Carlos_PHX: http://www.voip-info.org/wiki/view/Asterisk+sound+files+additional |
| 18:08.14 | malaiwah | telephone-in-your-pocket.gsm :Oooo! Is that a telephone in your pocket, or are you just happy to see me? |
| 18:08.27 | Carlos_PHX | Thanks! |
| 18:08.27 | malaiwah | i was about to paste the same url ;-) |
| 18:08.49 | Carlos_PHX | I must be lacking coffee. Searched phone but not telephone |
| 18:09.11 | *** join/#asterisk setkeh (n=setkeh@CPE-124-180-146-148.vic.bigpond.net.au) |
| 18:11.20 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:15.33 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:16.19 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:18.36 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
| 18:20.32 | jameswf | look live asterisk :) http://trixbox.org/devblog/introducing-new-trixbox-ce-livecd |
| 18:21.16 | rwaite | would it be a stupid idea to run the latest svn of asterisk on a production server? |
| 18:21.26 | seanbright | yes |
| 18:21.27 | jameswf | rwaite: yes |
| 18:21.43 | rwaite | intuitively i know it is, but when bugs are found and fixed |
| 18:22.06 | seanbright | do you mean the latest SVN of a branch? or of trunk? |
| 18:22.14 | jameswf | rwaite: if you expierience a fixed bug it is better to patch |
| 18:22.37 | rwaite | well, do they branch for 1.4.22 or is there just a 1.4 branch |
| 18:23.00 | jameswf | 1.4 branch 1.4.22 tag |
| 18:23.38 | rwaite | so when they fix a bug, they fix it on the 1.4 branch, 1.4.22 is frozen |
| 18:23.40 | rwaite | right? |
| 18:23.45 | *** join/#asterisk oej (n=olle@81.193.129.50) |
| 18:24.25 | jameswf | tags dont change they are locked as revusions unless someone commits to a tag.. I am pretty sure that is bad juju |
| 18:25.27 | rwaite | so in the case of 1.21.1, that was probably a new tag with backported changes from 1.4 applied to the 1.4.21 tag? |
| 18:25.40 | blitzrage | 1.21.1 eh? wow! |
| 18:25.47 | rwaite | lol |
| 18:25.53 | rwaite | sorry 1.4.21.1 |
| 18:26.00 | blitzrage | 1.6.0 released, and now 1.21.1... shocking development! :) |
| 18:26.17 | [TK]D-Fender | hordes his copy of chan_fluxcapacitor.so |
| 18:26.41 | jameswf | someone in version control was playing a drinking game durring the vp debae and released 1.21 |
| 18:26.48 | blitzrage | lol |
| 18:26.51 | rwaite | maverick! |
| 18:26.59 | Qwell | looks at putnopvut |
| 18:27.21 | jameswf | thought the SNL veepee debate was beter than the real one |
| 18:28.09 | *** join/#asterisk mltlnx (n=mltlnx@w058.z066088094.nyc-ny.dsl.cnc.net) |
| 18:28.28 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 18:28.33 | Blackvel | anyone here with openldap and ldapget experience? |
| 18:28.59 | Blackvel | looks like I missing some configuration and setup knowledge :( |
| 18:33.13 | *** join/#asterisk mltlnx (n=mltlnx@w058.z066088094.nyc-ny.dsl.cnc.net) |
| 18:36.35 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 18:37.12 | gsiener | hi all - in asterisk 1.4.21.2, setting up users. When I dial from one user extension to another, the other extension doesn't ring. Where would this logic live? |
| 18:37.50 | [TK]D-Fender | gsiener: extensions.conf |
| 18:38.23 | [TK]D-Fender | gsiener: this is the most important part of *. |
| 18:38.33 | [TK]D-Fender | gsiener: Time to go read the book... |
| 18:38.35 | [TK]D-Fender | ~book |
| 18:38.35 | jbot | methinks book is 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 |
| 18:38.39 | putnopvut | Qwell: why'd you look at me... |
| 18:38.41 | gsiener | I have the book :) |
| 18:38.43 | putnopvut | reads up |
| 18:39.06 | gsiener | thanks |
| 18:39.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 18:39.21 | putnopvut | oh, lol |
| 18:39.25 | jameswf | has a fresh copy from amazon on my desk to give away thursday |
| 18:39.26 | anonymouz666 | hey |
| 18:40.11 | talntid | to give away how? |
| 18:40.25 | talntid | and to whom? |
| 18:40.26 | anonymouz666 | anyone know anything related to a bug in chan_sip.c? chan_sip says that there's a channel in use therefore app_queue understand that the member is in use where it isn't. |
| 18:40.42 | jameswf | at a linux user group meeting.. I am presenting on Open Source Telephony |
| 18:41.05 | anonymouz666 | call-limit configured etc etc. |
| 18:41.50 | anonymouz666 | it seems there's a SIP dialog stuck or something like that. |
| 18:42.10 | anonymouz666 | asterisk 1.4.21.1 |
| 18:42.45 | anonymouz666 | only with a "restart now" the member get a correct state |
| 18:44.46 | *** join/#asterisk gsiener_ (n=gsiener@63.245.116.75) |
| 18:47.20 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:47.49 | *** join/#asterisk angryuser (n=Miranda@lns-bzn-54-82-251-66-8.adsl.proxad.net) |
| 18:48.48 | angryuser | hi, while building asterisk i have "cpu clock changed compilation maybe incomplete" what is the problem ? |
| 18:49.21 | *** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 18:50.45 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 18:52.57 | rwaite | would gsm be any more "reliable" than g729 over an iax2 connection? |
| 18:53.43 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 18:56.44 | *** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 18:57.29 | [TK]D-Fender | rwaite: Codec has nothing to do with reliability of its carrier protocol |
| 18:58.40 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 18:58.58 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 18:59.07 | Dovid | where can I get asterisk 1.4 trunk from ? |
| 18:59.42 | blitzrage | it's called a branch -- not a trunk |
| 18:59.55 | blitzrage | svn co http://svn.digium.com/svn/asterisk/branches/1.4 |
| 19:00.30 | jaytee | what? no twigs? |
| 19:00.57 | Dovid | ;) |
| 19:01.35 | blitzrage | svn co http://svn.digium.com/svn/asterisk/trunk <-- that is trunk |
| 19:01.39 | Dovid | how would I go about testing http://bugs.digium.com/view.php?id=10934 |
| 19:01.53 | Dovid | do i need 1.6.X ? |
| 19:02.12 | hardwire | sky_blue, did you fix your issue? |
| 19:02.20 | Greek-Boy | is 1.6 ready for production? |
| 19:02.40 | [TK]D-Fender | Dovid: What VERSION does it very clearly say its for in the bug entry? |
| 19:02.43 | hardwire | Greek-Boy, it's officially released. Not good enough for you? :) |
| 19:02.43 | blitzrage | Greek-Boy: does it pass the tests you've done to determine if anything is ready for production? |
| 19:03.14 | Greek-Boy | hardwire: thats good enough I guess |
| 19:03.19 | Greek-Boy | blitzrage: Haven't tested yet |
| 19:03.48 | blitzrage | Dovid: svn co -r 139771 http://svn.digium.com/svn/asterisk/trunk ; cd trunk ; patch -p0 < 10934.patch ; ./configure ; make install |
| 19:05.51 | Dovid | blitzrage: is that considerd 1.6 or 1.4 ? |
| 19:05.57 | blitzrage | NEITHER |
| 19:06.04 | blitzrage | 1.6 and 1.4 are branches off of trunk |
| 19:06.30 | blitzrage | it is considered 'trunk' |
| 19:06.31 | Dovid | ok. trunk is considerd non stable ? |
| 19:06.38 | blitzrage | trunk is development |
| 19:06.59 | Dovid | thanks. |
| 19:07.03 | Dovid | gona find me a test mashine |
| 19:07.04 | blitzrage | 1.6 branches are pulled off the trunk, then 1.6.x releases are made off the branch |
| 19:07.07 | Dovid | machine* |
| 19:07.24 | Dovid | ok. i learn new things every day |
| 19:08.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:10.08 | *** join/#asterisk datacompboy (n=datacomp@l64-89-221.cn.ru) |
| 19:11.30 | datacompboy | Hi all! What's wrong? I'm pass to * via AGI 'SAY DATETIME <time> "" "dB"' and it plays first HOUR, next month. While shoud play DAY, then month. |
| 19:12.50 | datacompboy | http://pastebin.ca/1221962 |
| 19:13.50 | sky_blue | hardwire: not yet, had to go and pick up my missus from the train station. i've got a sipgate account so i'm going to add that to the box and see if the problem remains |
| 19:14.38 | hardwire | what codec is your sip provider using? can your force it to ulaw? |
| 19:16.12 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 19:16.20 | sky_blue | hardwire: i'm not sure to be totally honest, but i wouldn't have thought the codec would matter if i'm hearing alison ok and the dtmf is accepted |
| 19:17.09 | hardwire | I agree |
| 19:17.17 | hardwire | but I love making people do weird things |
| 19:17.47 | sky_blue | hardwire: i was thinking it's either the caller id or ..... well something else! |
| 19:18.53 | hardwire | what codec are the local phones using? |
| 19:19.00 | hardwire | and what codec is the sip trunk using? |
| 19:19.48 | sky_blue | i have heard there are problems with conf without any zap hardware, but i thought ztdummy solved that? the local phones are using gsm or ulaw |
| 19:22.16 | datacompboy | use app_conference instead of meetme, and no problem without zap hardware since * 1.0 :) |
| 19:22.29 | datacompboy | since 1.2 no problem even with sip silence suppression |
| 19:22.53 | hardwire | asterisk 1.6 solves a lot.. btw |
| 19:23.03 | hardwire | meetme will work much better w/o app_conference |
| 19:23.27 | datacompboy | i'm use app_conference in production, and much happier, than with meetme:) |
| 19:23.59 | sky_blue | even on sip trunks? |
| 19:24.04 | datacompboy | yep. |
| 19:25.06 | sky_blue | i did initially install 1.6 on this latest server, but found myself getting lost in the dahdi, so rolled back to 1.4.22 |
| 19:25.56 | datacompboy | i'm use separated hardware * (asterisk connected to ZAP) and service * (asterisk controlled via agi+ami) |
| 19:26.01 | datacompboy | they connected with sip |
| 19:26.09 | JoseBravo | Any one know a good page where I can download backgrund sounds for my asterisk? |
| 19:26.13 | datacompboy | service * have no hardware. and conference mixed on software * |
| 19:26.25 | smth | <PROTECTED> |
| 19:28.14 | *** part/#asterisk fukz (n=fukz@p5B062080.dip.t-dialin.net) |
| 19:28.26 | sky_blue | i was having problems with ztdummy on 1.6, and for speed rolled back, as the customer needs this done by .. well er... yesterday! |
| 19:28.45 | datacompboy | sky_blue: never use latest asterisk in production :DDD |
| 19:28.54 | datacompboy | i have found that about 0.86 version... |
| 19:29.30 | datacompboy | every new version need to be tested on separate test server... otherwise, you can get stuck in inntresssting problems. where you don't want to find them |
| 19:30.03 | sfire | I never ever upgrade anything that works except for security concerns |
| 19:30.11 | sky_blue | it's odd though as i explained to hardwired earlier, conf works perfectly with hard/softphones, it's just direct dialling into the conf room that is giving me a headache |
| 19:30.52 | datacompboy | sfire: good solution too:) |
| 19:32.53 | datacompboy | AGI Rx << SAY DATETIME 1224268596 "" "eB" -=- "File digits/h-17 does not exist in any format" -=- why it playing hour?!! while it should play daY! |
| 19:33.21 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:35.24 | *** join/#asterisk CrazyTux (n=brandon@nmd.sbx05741.irvinca.wayport.net) |
| 19:35.46 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 19:37.55 | the_5th_wheel | I have just installed the latest bristuff. and it didnt actually install chan_zap. Has anyone experienced this? |
| 19:38.15 | *** join/#asterisk knobo` (n=user@77.241.96.163) |
| 19:39.26 | [TK]D-Fender | smtInband dtmf over GSM? Are you on crack? NEVER supposed to do that. next you didn't show my your configs to backup the call. |
| 19:39.40 | knobo` | I have always installed libpri as a reflex when installing TE410P cards, but I realy don't know what it is. |
| 19:39.44 | knobo` | When do I need libpri? |
| 19:39.52 | knobo` | And what does it doe? |
| 19:40.03 | knobo` | s/doe/do/ |
| 19:40.09 | *** join/#asterisk [netman] (n=netman@171.Red-83-45-38.dynamicIP.rima-tde.net) |
| 19:40.28 | [TK]D-Fender | knobo`: Well... do you want PRI signalling over your T1 or not? |
| 19:41.23 | knobo` | [TK]D-Fender: so, it is used asterisk uses it to signalling. |
| 19:42.16 | JoseBravo | Where I found the music on hold of my asterisk? its in /var/lib/asterisk ? |
| 19:42.47 | rwaite | /var/lib/asterisk/moh |
| 19:43.03 | JoseBravo | rwaite tahnk |
| 19:43.05 | JoseBravo | thanks |
| 19:43.42 | JoseBravo | rwaite do you know where can I donwload more? |
| 19:44.08 | rwaite | search google for "free music on hold" or "creative commons music on hold" |
| 19:44.56 | rwaite | beware that just because some music may be creative commons does not mean you can use it as music on hold for a company as some CC licenses forbid commercial use |
| 19:46.24 | [TK]D-Fender | knobo`: ....huh |
| 19:47.00 | knobo` | Ok, yes I want it. |
| 19:50.35 | Kobaz | hmmmmm |
| 19:50.58 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 19:51.06 | Kobaz | do features.conf feature codes require the device (if sip) to send dtmf's in sip info packets? |
| 19:51.17 | [TK]D-Fender | Kobaz: No |
| 19:51.29 | Kobaz | i have a device that isn't sending any sip packets and * is not picking up my feature code |
| 19:51.38 | Kobaz | er, not sending sip packets for dtmf |
| 19:51.45 | Kobaz | sip debug shows nodda |
| 19:51.47 | [TK]D-Fender | Kobaz: And maybe its not supposed to |
| 19:52.04 | sky_blue | datacompboy: do you use a sip provider to have customers dial into your conf rooms? |
| 19:52.08 | [TK]D-Fender | Kobaz: And right now YOU'RE showing "nada", but I'm getting used to that too... |
| 19:52.18 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:52.19 | Kobaz | hehe |
| 19:52.24 | Kobaz | well, i'll show you my features.conf |
| 19:52.30 | datacompboy | sky_blue: yes, voipfone |
| 19:52.34 | [TK]D-Fender | Kobaz: No, go prove to me that DTMF works at all |
| 19:52.41 | Kobaz | it works with other iax and sip phones |
| 19:52.50 | Kobaz | [TK]D-Fender: i can dial 1800tellme and it gets my dtmf |
| 19:53.13 | [TK]D-Fender | Kobaz: Who cares about features.conf when your dialplan defines if its relevant, and your peer says it * will even HEAR it if you tried |
| 19:53.25 | [TK]D-Fender | Kobaz: Still talking and not showing... |
| 19:53.33 | Kobaz | what do you want to see? |
| 19:53.47 | Kobaz | you just said you dont care about features.conf... that's the only thing i can think to show you |
| 19:53.49 | [TK]D-Fender | Kobaz: From what I just said, you should already know |
| 19:54.02 | Kobaz | the feature is enabled in the dialplan |
| 19:54.08 | Kobaz | i can show you that bit |
| 19:55.00 | sky_blue | datacomp: ok so you are in the uk, that gives me hope :) |
| 19:55.41 | datacompboy | sky_blue: nope:) i'm in russia, and server in germany :D |
| 19:56.17 | Kobaz | [TK]D-Fender: http://pastebin.ca/1222023 |
| 19:56.39 | Kobaz | if i dial out with say, a polycom |
| 19:56.40 | *** join/#asterisk Jabroni (n=asskick@201.170.242.193) |
| 19:57.00 | Kobaz | the feature code works fine |
| 19:57.16 | [TK]D-Fender | Kobaz: I see no point where features.conf could be relevant in there. |
| 19:57.20 | Kobaz | if i dial out with this phone connected to this sip gateway, it's not so fine |
| 19:57.35 | Kobaz | [TK]D-Fender: well, if features.conf was botched, then *1 wouldn't work no matter what |
| 19:58.23 | datacompboy | hmmm... that's really strange!! |
| 19:58.38 | [TK]D-Fender | Kobaz: Show me the call where I should see a successful trigger. |
| 19:58.44 | Kobaz | okay |
| 19:58.58 | [TK]D-Fender | Kobaz: Because that last bit wasn't it. |
| 19:59.12 | edwin_quijada | there is a difference between use Centos or Debian for asterisk |
| 19:59.17 | datacompboy | http://pastebin.ca/1222027 -- looks like bug in asterisk? |
| 19:59.43 | edwin_quijada | I have a big problem with Dell SC440 , Debian and OpenVox card |
| 20:00.07 | edwin_quijada | somebody told me that use centos |
| 20:00.18 | datacompboy | edwin_quijada: yes, there difference, since Debian apply its own patches. so, somethimes that differs |
| 20:00.28 | edwin_quijada | i am testing everything |
| 20:00.29 | jeev | fender, i got 3 phones here that after i clean settings, reset everything, even format filesystem, they come back with the name but dont try to register. or at least dont register, they're polycom 330's. |
| 20:00.30 | Jabroni | guys i have trouble using the L() parameter on the dial() app.. on the console it does show that the call will be time limited, but the call isnt hanged after that time.. heres the log http://pastebin.ca/1222031 |
| 20:00.34 | datacompboy | but you can download and compile it from scratch without debian patches |
| 20:00.54 | *** join/#asterisk shriven (n=shriven@rdu.crosscomm.net) |
| 20:01.03 | edwin_quijada | i use asterisk from scratch |
| 20:01.04 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 20:01.29 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 20:01.33 | waverly360 | For those wanting to know (if you don't already) Sangoma does have a beta release available that works with dahdi. |
| 20:01.36 | shriven | Hello. I'm trying to get asterisk jabber to connect with my jabber server. All seems to be setup ok but when I do "jabber show connected" I get " User: asterisk@crosscomm.net/asterisk - Disconnected" for my user. |
| 20:01.45 | shriven | Does anyone know that that means and how to make it connect? |
| 20:02.13 | [TK]D-Fender | Jabroni: Where is the rest of the call? I see no ringing, no answer, nothing... |
| 20:02.36 | [TK]D-Fender | waverly360: Yeah, you'd almost think it was on their WIKI or something.... |
| 20:03.05 | waverly360 | [TK]D-Fender: Well I haven't checked today, but it wasn't on their wiki on Friday. |
| 20:04.01 | sky_blue | datacompboy: http://www.pastebin.ca/1222033 could you take a look and tell me if this could be causing the problem? |
| 20:04.41 | waverly360 | Someone asked about it in here last week, I got an email from them late friday night. They must have added it since then. |
| 20:05.00 | Kobaz | [TK]D-Fender: http://pastebin.ca/1222036 |
| 20:05.32 | Kobaz | [TK]D-Fender: working and non-working |
| 20:05.32 | datacompboy | sky_blue: i'm use app_conference everywhere:) since have only negative exp with meetme |
| 20:05.47 | edwin_quijada | I hearing th e voice like a robot when somebody call me from ptsn |
| 20:06.03 | datacompboy | sky_blue: so, can't help you there:( |
| 20:06.24 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-246-86-237.nycmny.east.verizon.net) |
| 20:06.29 | sky_blue | datacomp: ok thanks, up to 1.6 then? |
| 20:07.17 | datacompboy | sky_blue: nope, on latests builds of service -- 1.4.18-debian-addpatched used |
| 20:07.51 | sky_blue | ok i'll take a look, thanks |
| 20:07.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:09.15 | [TK]D-Fender | Kobaz: Now go prove to me his DTMF works elsewhere |
| 20:09.19 | Jabroni | [TK]D-Fender i just updated the log |
| 20:09.30 | [TK]D-Fender | Jabroni: Which requires a NEW link... |
| 20:09.51 | Jabroni | oh sorry thought it updated the same :p |
| 20:09.58 | Jabroni | http://pastebin.ca/1222041 |
| 20:10.09 | Kobaz | [TK]D-Fender: i guess tellme isn't a good test? |
| 20:10.16 | datacompboy | o! have found problem. it use "h-..." as numbers for days 1..20 |
| 20:10.25 | Jabroni | wait i copies the wrong test |
| 20:10.26 | datacompboy | :D very strange, but let be so |
| 20:10.43 | [TK]D-Fender | Kobaz: If I don't see it, I don't trust it. It really is that simple. |
| 20:11.23 | Jabroni | [TK]D-Fender do u see something wierd on the log ? |
| 20:12.02 | *** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:12.52 | shriven | does anyone know what this means? |
| 20:12.53 | shriven | User: asterisk@crosscomm.net/asterisk - Disconnected" for my user. |
| 20:13.00 | shriven | how can I make my user connect? |
| 20:13.24 | shriven | my apologies, bad copy... "User: asterisk@crosscomm.net/asterisk - Disconnected" |
| 20:13.43 | [TK]D-Fender | Jabroni: You seem to limit it to 10s but you don't let it last at all... |
| 20:14.03 | datacompboy | pfffrrr... everybody: thanks! :) i have fixed now problem. going to doing next things. |
| 20:14.08 | [TK]D-Fender | Jabroni: looks like its disconnected immediately. Also you formatted your zap channel wrong. |
| 20:14.23 | Jabroni | well ive tried up to 60seconds.. and call still continues |
| 20:14.33 | [TK]D-Fender | smth: Inband dtmf over GSM? Are you on crack? NEVER supposed to do that. next you didn't show my your configs to backup the call. |
| 20:15.05 | Kobaz | [TK]D-Fender: so umm... i can call tellme, or i can call one of my other boxes, and dtmf works fine... should i build some example to show you? |
| 20:15.39 | [TK]D-Fender | Kobaz: If you expect any kind of help. You know I wouldn't take your word for it... |
| 20:16.09 | Kobaz | heh |
| 20:16.11 | [TK]D-Fender | Jabroni: Continues? It seems to show me you hung up almost instantly |
| 20:16.36 | [TK]D-Fender | Jabroni: [Oct 7 12:57:10] VERBOSE[23599] logger.c: == Spawn extension (custom-cell, 0446869460033, 1) exited non-zero on 'Local/0446869460033@custom-cell-48ba,2' <--- hung up |
| 20:16.40 | Jabroni | let me post another log.. that was just a log of a short test to print it |
| 20:16.52 | [TK]D-Fender | Jabroni: Next time show me something REAL |
| 20:16.58 | jdnWEST | Any chance of a not crappy sip client coming out for the iphone anytime soon |
| 20:17.18 | jdnWEST | 2 second delay is kinda annoying... |
| 20:18.43 | Jabroni | [TK]D-Fender http://pastebin.ca/1222050 |
| 20:20.10 | Jabroni | and btw.. asterisk version 1.4.21.2-2 |
| 20:22.02 | [TK]D-Fender | Jabroni: 2 thoughts : you didn't use "/" to make the call and locked end-to-end chain. Next, did it indeed last 30s? |
| 20:22.10 | [TK]D-Fender | * /n |
| 20:22.50 | Jabroni | Yeah the call lasted 30s.. and hanged by one end.. not automaticlly by asterisk |
| 20:23.16 | malaiwah | see ya, bye. |
| 20:24.18 | Kobaz | [TK]D-Fender: http://pastebin.ca/1222058 |
| 20:24.26 | Kobaz | [TK]D-Fender: i'm not sure how much that helps |
| 20:24.26 | [TK]D-Fender | Jabroni: do your local dial with the /n at the end, and fix the channel name. "Zap/3-1?" is not legal |
| 20:24.57 | [TK]D-Fender | Kobaz: In your previous sample, the IAX one worked... |
| 20:25.04 | Kobaz | [TK]D-Fender: from the phone in question... dial out to the pstn... i have a box in a colo that picks up the call, and routes the call to the local box in the office here... and i get the dtmf |
| 20:25.18 | Jabroni | k let me look onto it.. thanks for your time |
| 20:25.22 | Kobaz | [TK]D-Fender: this is a sip phone dialing out to the pstn, and then comming back in on ptsn and then to iax |
| 20:25.30 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 20:25.45 | [TK]D-Fender | Kobaz: -- Executing [2503@ivrFooBar:1] Set("IAX2/TroyIn-4530", "ARGS=dialExt,2503,no") in new stack <- this sure as hell isn't SIP |
| 20:25.46 | Kobaz | [TK]D-Fender: this is the pstn->iax side of it |
| 20:25.50 | smth | <PROTECTED> |
| 20:26.01 | l2trace99 | anyone know what format MixMonitor records to ? |
| 20:26.15 | *** join/#asterisk hi365_m (n=hi365@213.151.52.225) |
| 20:26.22 | Kobaz | [TK]D-Fender: sip -> pstn -> my box -> (iax) another box |
| 20:26.31 | l2trace99 | file gives me " ISO-8859 text, with very long lines, with no line terminators" |
| 20:26.32 | [TK]D-Fender | Kobaz: you showed me 2 samples, and the SIP one failed and now you're showing me a test with an IAX. |
| 20:26.44 | *** join/#asterisk TommyBJ (n=noosjent@77.241.96.163) |
| 20:27.00 | Kobaz | okay, let me put the whole thing together |
| 20:27.21 | Kobaz | i would have figured you would have used the previous pastebin as context |
| 20:27.23 | [TK]D-Fender | l2trace99: to whatever format you TELL it to |
| 20:28.07 | l2trace99 | [TK]D-Fender: by file ext ? |
| 20:28.20 | TommyBJ | I'm having some trouble with zaptel/sangoma cards. Anyone familiar with the driver issue where I get about 10 uknown symbol errors while trying to load the wanpipe module? |
| 20:28.23 | [TK]D-Fender | l2trace99: Go read its instructions |
| 20:29.11 | l2trace99 | http://books.google.com/books?id=vtQxJ3oSm64C&dq=asterisk+future+of+telephony&pg=PP1&ots=LVX8G_Eh19&sig=RCMW_Z7xEYx5QyDHntrLu9ooj-k&hl=en&sa=X&oi=book_result&resnum=4&ct=result#PPA412,M1 |
| 20:29.36 | l2trace99 | it doesn't state how to set format |
| 20:29.42 | l2trace99 | monitor does |
| 20:30.30 | [TK]D-Fender | l2trace99: Go read its instructions <--- |
| 20:31.38 | Kobaz | [TK]D-Fender: http://pastebin.ca/1222072 |
| 20:31.40 | Kobaz | [TK]D-Fender: is that better? |
| 20:31.49 | jaytee | wow! it's 4:30pm already |
| 20:31.55 | jaytee | day just flew by |
| 20:31.57 | Kobaz | [TK]D-Fender: on top is box 1 with the sip phone, on bottom is box2, recieving the call from a pstn->iax gateway |
| 20:32.07 | *** join/#asterisk CrazyTux (n=brandon@nmd.sbx05741.irvinca.wayport.net) |
| 20:32.14 | Kobaz | [TK]D-Fender: and going into an ivr, where 2503 is hit on the sip phone |
| 20:32.19 | [TK]D-Fender | Kobaz: No, it isn't |
| 20:32.42 | [TK]D-Fender | and I'm out of time. |
| 20:32.49 | [TK]D-Fender | back later |
| 20:33.45 | hawk | l2trace99: I'm guessing the file extension may be what is used |
| 20:36.36 | TommyBJ | I'm having some trouble with zaptel/sangoma cards. Anyone familiar with the driver issue where I get about 10 uknown symbol errors while trying to load the wanpipe module?. |
| 20:38.09 | smth | <PROTECTED> |
| 20:38.57 | *** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net) |
| 20:44.20 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 20:47.53 | Kobaz | heh |
| 20:47.57 | Kobaz | smth: he's busted out |
| 20:49.23 | smth | ic |
| 20:56.38 | *** join/#asterisk el_critter (n=el_critt@190.39.203.141) |
| 21:08.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:12.32 | *** join/#asterisk shinao1 (n=shinao1@83.229.85.138) |
| 21:14.22 | *** join/#asterisk rdgr (n=rich@82-32-1-139.cable.ubr01.aztw.blueyonder.co.uk) |
| 21:23.03 | *** join/#asterisk mintee (n=mintee@c-68-45-231-166.hsd1.nj.comcast.net) |
| 21:23.38 | mintee | by defaut if i set my CALLERID(num)=1111111111 it seems to be automagically bring up the Privacy Screening |
| 21:23.52 | *** join/#asterisk `paul (n=paul@125.252.70.126) |
| 21:24.21 | `paul | can i put SIP/<extension> as a member of a RRmemory queue? |
| 21:28.06 | *** join/#asterisk steliosk (n=Stelios@91.140.124.241) |
| 21:29.54 | *** part/#asterisk CrazyTux (n=brandon@nmd.sbx05741.irvinca.wayport.net) |
| 21:35.19 | Kobaz | `paul: yeap |
| 21:38.17 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 21:39.15 | *** join/#asterisk joobie (n=joobie@201.023.dsl.mel.iprimus.net.au) |
| 21:42.27 | `paul | Kobaz: how come the call does not jump to the next member? im using rrmemory |
| 21:42.48 | `paul | i mean it does not jump if no one answers |
| 21:46.31 | *** join/#asterisk seanmh (n=seanmh@nat/digium/x-d1565281fc5d5b40) |
| 21:47.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:47.31 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 21:47.33 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:49.22 | `paul | how come the call does not jump to the next member? im using rrmemory and (SIP/<extension>) as members |
| 21:50.14 | `paul | if member 1 does not answer... how do i set the timeout |
| 21:51.23 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 21:51.51 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 21:52.43 | jaytee | paul, what do you have timeout= set to in your queues.conf? |
| 21:55.56 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 21:56.02 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 22:01.38 | thansen | does anyone have a good suggestion for a mac softphone support h264? |
| 22:09.02 | *** join/#asterisk rasterix (n=IceChat7@80.177.176.254) |
| 22:12.11 | *** join/#asterisk hi365_m (n=hi365@213.151.52.225) |
| 22:12.45 | *** part/#asterisk sky_blue (n=ayates@cpc4-bexl4-0-0-cust504.bmly.cable.ntl.com) |
| 22:12.48 | rasterix | what is customer controlled all forwarding on an isdn30? it says check with your equipment manufacturer to see whether they support this... we have a sangoma a101 pri card |
| 22:13.24 | rasterix | call forwarding* |
| 22:13.27 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 22:13.37 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:14.52 | rasterix | does this tell the exchange to forward the call to a different number or do they mean to forward on one of our channels? |
| 22:17.39 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 22:18.18 | CGMChris | I am having a problem setting up my call queues. As soon as I call the queue extension I am directed to an agents voicemail. Any thoughts on what is going wrong? |
| 22:19.05 | hi365_m | the agents phone isnt reporting that he is busy? |
| 22:19.41 | *** join/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 22:20.07 | rasterix | are the agents on sip phones? |
| 22:20.18 | CGMChris | hi365_m: The agent is either busy or has their softphone off, but the behavior I am seeking is to keep the user in a queue until someone answers... not redirect them to voicemail. All agents are SIP phones. |
| 22:20.54 | LARefugee | chan_alsa works on 1.6 but I can't get it to work on 1.4.21 or 22 anyone have better luck? |
| 22:21.38 | rasterix | cgm: are you using 1.4? |
| 22:22.17 | CGMChris | rasterix: AsteriskNOW. Is this a bug that has been resolved? |
| 22:22.27 | rasterix | its not a bug |
| 22:22.52 | rasterix | you need to set your system so that if the agent is on a call it reports busy back |
| 22:23.35 | CGMChris | rasterix: Report busy back rather than ringing on line 2? Is that at the PBX or the phone level? |
| 22:23.54 | rasterix | i think there is a setting that you can put in sip.conf |
| 22:23.59 | rasterix | hold on |
| 22:25.10 | riddlebox | anyone else having problems with fxs ports on a tdm after upgrading to 1.2.22.1? |
| 22:25.43 | jaytee | reportinuse=yes in queues.conf |
| 22:26.01 | jaytee | CGMChris, that's for you |
| 22:26.29 | CGMChris | jaytee: Thank you, I will try that. |
| 22:26.56 | jaytee | CGMChris, it only works in 1.4 with SIP phones |
| 22:27.02 | *** part/#asterisk sircco (n=sircco@dh207-70-239.xnet.hr) |
| 22:27.40 | CGMChris | jaytee: There was no way to hold a call in a queue to wait for the next available agent pre-1.4 ? |
| 22:30.06 | rasterix | in sip.conf you can specify busy-level = number of simultaneous calls untill user / peer is busy |
| 22:31.22 | *** join/#asterisk Swabby (n=dp@74-137-56-171.dhcp.insightbb.com) |
| 22:31.38 | Swabby | Hey..need some advice |
| 22:32.09 | Swabby | i been struggling with asterisknow.. can you folks recommend a distro that is fairly easy to setup and maintain? i am not an asterisk expert but really want to help a non-profit get this going |
| 22:32.10 | LARefugee | Swabby: Ask away.. |
| 22:32.24 | CGMChris | rasterix: That works in a controlled environment, but doesnt work if a user forgets to log out as an agent and then closes their softphone. |
| 22:32.34 | sfire | Swabby: ubuntu is pretty easy.. asterisknow works too |
| 22:32.39 | LARefugee | Swabby: I use Ubuntu Linux |
| 22:32.42 | Swabby | It's like i run into issues here and there..either i can't get dialplans set correctly...or configuration..etc.. |
| 22:32.55 | Swabby | i'm fairly good with linux but i have problems with the asterisk config files..getting stuff to work properly..etc |
| 22:33.13 | Swabby | like i got asterisknow up but weird stuff started happening like ALL the phones started ringing |
| 22:33.28 | sfire | Swabby: chances are that is a config problem |
| 22:34.00 | sfire | mis configured any linux will give you the same problem |
| 22:34.22 | sfire | you would be better served to ask how to fix that problem |
| 22:35.03 | rasterix | what is customer controlled call forwarding on anisdn30? it says check with your equipment manufacturer to see whether they support this... we have a sangoma a101 pri card |
| 22:35.03 | Swabby | sfire: I'm using a group to bundle all of the phones together (Grandstream gxp 2000) so they ALL ring when a call comes in |
| 22:35.18 | Swabby | it was working fine but all of a sudden they all show up as unavailable in the CLI and it goes straight to vm |
| 22:35.32 | CGMChris | Swabby: |
| 22:35.38 | CGMChris | Swabby: did you check your log files? |
| 22:36.24 | Swabby | not yet |
| 22:36.30 | Swabby | didn't realize there was log files |
| 22:36.40 | CGMChris | You can check them from the gui, bottom left menu. Also in /var/log/asterisk/ |
| 22:42.26 | Swabby | what is the best distro for a small office in your opinion? |
| 22:42.36 | Swabby | I have a Digium TDM400 ( there's two analog lines ) |
| 22:43.01 | sfire | Swabby: the distro makes no difference |
| 22:43.38 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 22:44.04 | Swabby | which one is easier on the admin as far as user configuration? |
| 22:44.07 | Swabby | ex: the cleanest gui |
| 22:44.21 | Swabby | the goal is for me to hand this over to a non-technical person to maintain in the futre..that's my worry... |
| 22:44.28 | sfire | asterisknow |
| 22:44.28 | Swabby | i want to pick something that is easiest to maintain |
| 22:44.30 | sfire | (IMHO) |
| 22:44.40 | Swabby | IMHO = switchvox? |
| 22:44.45 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 22:44.55 | CGMChris | IMHO = In My Humble Opinion |
| 22:44.58 | Swabby | ah |
| 22:45.06 | Swabby | asterisknow livecd or custom install? |
| 22:45.08 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-10a29b725418a52c) |
| 22:45.19 | x86 | so is there not a PCIe equivalent to 1TDM413EF and 1TDM413BF? |
| 22:45.27 | CGMChris | I did express at my office. The fact is, the user is going to need *some* support. |
| 22:45.41 | sfire | Swabby: custom install |
| 22:45.52 | Swabby | cgmchris: any paticular resources you recommend as reference |
| 22:46.02 | Qwell | x86: AEX2400 |
| 22:46.11 | Qwell | what is the F at the end? |
| 22:46.11 | *** join/#asterisk arpu (n=arpu@chello084114022060.14.vie.surfer.at) |
| 22:46.19 | Qwell | oh, nm |
| 22:46.23 | Swabby | here's my other question..i have TWO analogs coming in (provider 2 and 3 ) why can't i make dialplans effective on BOTH of them?? it only lets me select one or the other |
| 22:46.26 | x86 | Qwell: *shrugs* it's a Digium part number ;) |
| 22:46.30 | Qwell | AEX410 |
| 22:46.51 | CGMChris | Swabby: I've only been using this a week... still working out a few kinks. I have used google and this room alot. Also #asterisknow and #asterisk-gui. People help. |
| 22:46.58 | x86 | my digium pricing has no AEX410 |
| 22:47.05 | Qwell | x86: it's new |
| 22:47.18 | Qwell | I think.. |
| 22:47.53 | Swabby | b/c in the book(s) it shows people selecting more than one provider... |
| 22:48.22 | Swabby | like how do i tell it to use line or two for the same rules..AND why when both lines are busy i get dead air vs no lines available |
| 22:50.39 | CGMChris | Swabby: I'm using pure VoIP...no real phone lines. Can't help. |
| 22:53.16 | *** join/#asterisk robberknight (n=gerd@HSI-KBW-085-216-023-190.hsi.kabelbw.de) |
| 22:53.26 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:55.15 | robberknight | hi, can anyone tell me how I can set the rtp payload size / length in asterisk, it seems like its always 20 msec (at least for alaw which I playing around with) |
| 23:02.39 | jaytee | robberknight, you might try this in sip.conf allow = alaw:40 |
| 23:03.14 | robberknight | jaytee: thanks. 40 means 40msecs? |
| 23:04.11 | jaytee | robberknight, I'd think so cuz like 40 secs would be an eternity in the world of VOIP. |
| 23:04.41 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 23:04.51 | talntid | [Oct 7 16:03:39] WARNING[12096]: pbx.c:1821 pbx_extension_helper: No application 'ZapBarge' for extension (staff, 8159, 1) |
| 23:05.33 | robberknight | just tried it, seems like asterisk is now sending 10msec-packets as I told it. Is there any way to influence the client to change its packet size too? |
| 23:06.18 | robberknight | I mean without using the clients config gui/files/menu but via sip during session setup? |
| 23:06.39 | jaytee | robberknight, don't know. never tried. have you googled it? |
| 23:07.16 | jaytee | talntid, if you're trying to do ZapBarge on an extension it won't work, it's only for zap CHANNELS. |
| 23:07.52 | *** join/#asterisk rene- (n=renemend@200.79.231.94.static.cableonline.com.mx) |
| 23:07.59 | talntid | i'm completely unsure what i should be doing with it... |
| 23:08.14 | jaytee | there's examples in the book |
| 23:08.20 | jaytee | ~book |
| 23:08.20 | jbot | well, book is 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 |
| 23:08.24 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 23:08.40 | jaytee | drmessano, hey ho! |
| 23:08.55 | rene- | hey, about using AMD, why do people use WaitExten after AMD has made an analysis on the called end? isnt it redundant? as per the example in http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 23:08.56 | rene- | ? |
| 23:09.15 | *** join/#asterisk NirS (n=NirS@80.250.159.240) |
| 23:09.18 | NirS | join #asterisk-dev |
| 23:09.18 | rene- | s/WaitExten/WaitForSilence |
| 23:09.28 | drmessano | Whaddup |
| 23:09.41 | jaytee | not much, just getting ready to fix some dinner |
| 23:09.57 | robberknight | jaytee: found something about Session Description Protocol rfc 4566, page 25: a=ptime but don't if this is transmitted to the client when setting alaw:nn |
| 23:11.00 | talntid | ok, so. I'm doing it right, but it's not prompting me for a zap channel |
| 23:12.17 | talntid | ooh. zapscan may be better for what i'm doing anyhow |
| 23:12.33 | jaytee | zapbarge doesn't "prompt" |
| 23:12.53 | talntid | If channel is not specified, you will be prompted for the channel number. Enter 4# for |
| 23:12.53 | talntid | Zap/4, for example. |
| 23:13.02 | rene- | sorry for posting again, people here using asterisk AMD? |
| 23:14.39 | jaytee | oh, didn't see that. so it does prompt if a channel isn't specified. |
| 23:16.18 | talntid | is there an equivelant for that, but with SIP? |
| 23:16.31 | jaytee | rene-, I've never used it but I'm sure if you google it you'll find lots of hits if you use Asterisk AMD as a search string. |
| 23:16.56 | drmessano | is having marital problems with his IS{ |
| 23:16.58 | drmessano | is having marital problems with his ISP |
| 23:17.18 | jaytee | divorce her and sue for palimony |
| 23:17.30 | jaytee | or is it malimony when it's a guy suing? |
| 23:17.45 | drmessano | Dunno, but it's Comcastic |
| 23:17.48 | drmessano | Sorry |
| 23:17.52 | drmessano | Comcastic! |
| 23:17.58 | jaytee | Comcraptastic |
| 23:18.07 | jaytee | that my ISP too |
| 23:18.08 | *** join/#asterisk ManxPower (n=manxpowe@218.sub-75-202-2.myvzw.com) |
| 23:18.19 | jaytee | they took away my Pr0n! |
| 23:18.19 | vader-- | If you have a Dell Poweredge 2950 server, and it was running Windows SBS, would you rather have 4TB of storage in the server or 4TB of storage in an external NAS solution? This being your only two choices. the server has a 5 year 24x7x4 Warranty and the NAS only has standard 2 Year warranty. |
| 23:18.21 | scooby2 | people keep telling me that * answers calls in the orders received even if they are in multiple queues but why am I not seeing this on 1.2.15 and 1.4.21.1? Most of the time it does well but at least twice a day someone that called a few seconds ago will get answered before someone waiting 5-10 minutes. |
| 23:18.41 | jaytee | vader, NAS |
| 23:18.51 | scooby2 | which nas |
| 23:18.57 | drmessano | Ugh |
| 23:18.59 | ManxPower | sorry, but I would not have a server running Windows. |
| 23:19.06 | drmessano | Put it in the server |
| 23:19.13 | ManxPower | vader--: go with the longest warrenty |
| 23:19.22 | scooby2 | in the server would mean sata for sure |
| 23:19.23 | drmessano | You wont have 4 gold support on the NAS |
| 23:19.28 | drmessano | 4-hour |
| 23:19.38 | jaytee | I'd actually rather have a 4TB SAN than either internal 4TB or NAS |
| 23:19.43 | drmessano | SATA? |
| 23:19.46 | ManxPower | use the NAS as a backup of the server |
| 23:19.49 | drmessano | Are you serious? |
| 23:19.56 | scooby2 | drmessano: yessir |
| 23:20.09 | drmessano | It's a poweredge 2950.. I doubt they even have SATA |
| 23:20.11 | jaytee | my 2950 uses SAS, not SATA |
| 23:20.12 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 23:20.15 | scooby2 | 2950's either have 6 3.5" slots or 8 2.5" |
| 23:20.15 | drmessano | Exactly |
| 23:20.18 | talntid | any alternative to zapscan for sip chans? |
| 23:20.20 | denon | if only that server is goingto access it, of course you want it on the server. lowest latency possible |
| 23:20.21 | drmessano | SATA is CONSUMER level |
| 23:20.36 | scooby2 | I have 2 2950's with SAS and 2 with SATA |
| 23:20.43 | drmessano | Yuck |
| 23:20.44 | ManxPower | I was not aware that SAS with an interface |
| 23:21.15 | scooby2 | 8 146gb 2.5" 10krpm raid 10 for database. The sata servers rarely hit disk |
| 23:21.43 | drmessano | SATA drives are low rent |
| 23:21.52 | scooby2 | yep |
| 23:21.56 | ManxPower | drmessano: and SCSI is expensive |
| 23:22.15 | scooby2 | SATA 2950's were like $2500. SAS were almost $5000 |
| 23:22.38 | drmessano | I wouldnt use SATA drives in a server |
| 23:23.07 | scooby2 | i would in anything not disk intensive. No need to waste the money. |
| 23:23.08 | ReDNeQ | drmessano: we do. |
| 23:23.13 | ReDNeQ | they have worked well |
| 23:23.27 | ReDNeQ | especially raided |
| 23:23.28 | scooby2 | my * server is two 500gb sata mirrored |
| 23:23.38 | drmessano | Somewhere along the line, someone got the idea that Just because IDE < SCSI < SATA < SAS that made it OK to use SATA for a server.. and in reality, SATA is just IDEv2 |
| 23:23.58 | drmessano | When it comes to the drives |
| 23:24.02 | ManxPower | drmessano: none of my customers use more than a fraction of their capacity -- no reason not to use SATA in that case. |
| 23:24.03 | ReDNeQ | true, but they do perform as well as SCSI |
| 23:24.09 | ManxPower | especially if you get a drive with a 5yr warrenty |
| 23:24.15 | ReDNeQ | DITTO |
| 23:24.31 | scooby2 | I did just order an external SAS array from dell |
| 23:24.32 | ReDNeQ | since they are not large DATA Transfers on the phone system |
| 23:24.45 | scooby2 | but thats for a hard hitting database server. 128gb of ram, 16 procs, etc. |
| 23:24.58 | ManxPower | ReDNeQ: I was not even thinking of asterisk. of course SATA is fine for Asterisk |
| 23:25.25 | scooby2 | asshats here before me put scsi in everything. what a waste |
| 23:25.42 | ManxPower | scooby2: they got brainwashed by the SCSI people. |
| 23:25.46 | drmessano | What was wrong with SCSI? |
| 23:25.49 | scooby2 | $$$ |
| 23:25.56 | rene- | jaytee no help so far, i would like to talk to somebody that has used Asterisk answering machine detection in depth |
| 23:26.00 | drmessano | lol |
| 23:26.01 | ManxPower | Just wait for drives to fail then move to non-scsi if the driver is low usage. |
| 23:26.23 | scooby2 | if you use disk i/o get scsi or sas. Otherwise sata is great |
| 23:26.23 | vader-- | It's a BuffaloTech Terstation Pro II 4TB model |
| 23:26.31 | scooby2 | vader--: internal |
| 23:26.34 | vader-- | they wanted it as backup and external storage |
| 23:26.42 | scooby2 | those things blow goats |
| 23:26.53 | drmessano | Ok, sorry.. but using IDE drives for a server is stupid.. and if you're talking about SCSI being too expensive, and relating to an older server, you can only be advocating IDE at this point |
| 23:27.12 | scooby2 | i've used ide in lots of servers |
| 23:27.20 | drmessano | Im sure you have |
| 23:27.30 | scooby2 | if it rarely touches disk, it doesnt matter what you use |
| 23:27.35 | scooby2 | floppy is fine |
| 23:27.44 | drmessano | .... |
| 23:28.49 | ManxPower | One of my customers tried SCSI at one point. Cost 4x what PATA would have cost, gave no performance improvement for their usage patterns, expecially since the server was limited to 100 Mbps |
| 23:29.43 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:29.45 | scooby2 | my storage server is 16x 1tb seagate 32mb seagate drives. Awesome performance with that and the new Adaptec dual core raid cards. |
| 23:30.04 | ManxPower | Of course if you have a disk intensive application and are not limited by network bandwidth you want the fastest drives you can get on the fastest interfaces |
| 23:30.54 | scooby2 | ManxPower: thats exactly why i just ordered 30 300gb 15krpm sas drives in arrays from Dell. Databases lovem otherwise complete waste. |
| 23:31.24 | ManxPower | scooby2: yes, I imagine databases DO love those drives |
| 23:31.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:32.27 | ManxPower | heck, one of my customers have servers in production running Mandriva 8.1 on 1.2Ghz CPUs with 80GB HDs. |
| 23:33.44 | vader-- | so you think it would be better in the server? |
| 23:34.00 | scooby2 | yes |
| 23:34.00 | ManxPower | vader--: what is the usage? |
| 23:34.19 | scooby2 | this card is nuts fast for sata |
| 23:34.21 | scooby2 | http://www.adaptec.com/en-US/products/Controllers/Hardware/sata/performance/SAS-51645/ |
| 23:34.29 | vader-- | file storage |
| 23:35.04 | scooby2 | go internal. better warranty |
| 23:35.15 | jaytee | SAS is going to support 6Gb/s in 2009-2010 timeframe |
| 23:35.22 | ManxPower | vader--: accessed all locally or all remotely? |
| 23:35.46 | jaytee | and you can hang SATA-II drives off an SAS backplane |
| 23:36.01 | vader-- | manx what do you mean? |
| 23:36.11 | scooby2 | some of the backplanes allow you to mix and match |
| 23:36.19 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 23:36.42 | ManxPower | vader--: I mean are the files going to be accessed locally on the server or will the files be accesses via a network link that had no chance of even coming close to the thruput your drives can handle? |
| 23:37.10 | vader-- | network link |
| 23:37.15 | ManxPower | Kind of pointless to have a kick ass fast drive if you are accessing it over a network link. |
| 23:37.37 | ManxPower | vader--: in that case I say there is no advantage to having the drives on the server except for the warrenty issue. |
| 23:39.15 | drmessano | Wait |
| 23:39.19 | drmessano | and your service contract |
| 23:39.34 | *** join/#asterisk moy (n=moy@189.169.68.109) |
| 23:39.37 | *** join/#asterisk sky_blue (n=ayates@cpc4-bexl4-0-0-cust504.bmly.cable.ntl.com) |
| 23:39.43 | vader-- | ? |
| 23:39.49 | drmessano | 4 Hr Gold support with Dell is =! RMA with NASmaster |
| 23:40.06 | drmessano | So the server may have an upside in that case |
| 23:40.25 | scooby2 | the buffalo is only 2.8tb once you raid it |
| 23:40.28 | scooby2 | or less |
| 23:42.35 | sky_blue | can somebody give me some pointers on app_conference? ie syntax. have been using meetme. exten => s,1,Conference(what goes here?) |
| 23:43.22 | ManxPower | sky_blue: have you done a "core show application conference? |
| 23:44.37 | sky_blue | yes it's installed, i was on here earlier, was conf with meet me between extensions no problem, but unable to get a sip trunk to dial direct to the conf room.......... |
| 23:44.57 | sky_blue | so was advised to install app_conference |
| 23:45.20 | *** join/#asterisk bsaxon (n=bryantsa@68.117.152.206) |
| 23:45.57 | sky_blue | [Synopsis] Channel Independent Conference [Description] Channel Independent Conference Application |
| 23:46.09 | ManxPower | sky_blue: so there is no docs there? |
| 23:46.44 | ManxPower | Because every other frickin |
| 23:46.50 | sky_blue | i read http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference |
| 23:47.10 | ManxPower | <PROTECTED> |
| 23:47.38 | drmessano | argues that functionality was removed in 1.x |
| 23:47.43 | sky_blue | but it seems a different concept to meetme, |
| 23:47.56 | scooby2 | arent we using 1.x? |
| 23:48.05 | ManxPower | sky_blue: Yup. There's docs there. They are, as always, totally out of date when it comes to Asterisk applications. |
| 23:48.07 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:48.22 | drmessano | I hope when Scoopy3 comes it, it will have a much faster CPU |
| 23:48.26 | drmessano | out* |
| 23:48.29 | ManxPower | sky_blue: so which one are you going to be using? MeetMe or Conference? |
| 23:49.27 | drmessano | Thats so not true.. the docs for the "unlimited license" g729 are current, damnit |
| 23:49.43 | drmessano | Hmm... who was that again? |
| 23:49.49 | sky_blue | using Conference, but i understand that my conf room is set up as 3000, but i don't go exten => s,1,Conference(3000) do i? |
| 23:50.15 | drmessano | I dont think anyone here uses that |
| 23:50.23 | drmessano | We all use app_meetme |
| 23:50.29 | ManxPower | drmessano: that's a codec, not an application |
| 23:50.50 | ManxPower | sky_blue: I have no idea, that's why I was trying to get you to find some docs |
| 23:50.50 | drmessano | ManxPower: Depends on your use of the word "application" ;) |
| 23:51.55 | scooby2 | thats the downside to most opensource. Either sparse documentation or its out of date |
| 23:51.56 | sky_blue | i've read the docs, that's why i'm asking for help. a catch 22 it seems. |
| 23:52.41 | drmessano | Trixbox is very well documented |
| 23:53.04 | ManxPower | scooby2: Oh, the application docs are totally up to date in Asterisk, it's just the wiki that's out of date. Someone started an automated updating of the stuff in the /path/to/src/asterisk/docs directory so at least that specific doc probem is no longer an issue. |
| 23:54.45 | sky_blue | ok thanks anyway lads, it's gone 1am here now so enough is enough, i'll call it a night. cheers |
| 23:55.11 | rasterix | <PROTECTED> |
| 23:55.19 | rasterix | whoops |
| 23:55.31 | rasterix | wrong window |
| 23:55.34 | *** part/#asterisk sky_blue (n=ayates@cpc4-bexl4-0-0-cust504.bmly.cable.ntl.com) |
| 23:55.53 | scooby2 | i'm still trying to find out what our last consultant meant by that we could combine queues. We have like 8 sales queues each having its own queue announce, hold music, and number. Same agents answer all 8. |
| 23:56.01 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:57.24 | rasterix | what is "customer controlled call forwarding" on anisdn30? it says check with your equipment manufacturer to see whether they support this... we have a sangoma a101 pri card |