| 00:01.33 | ManxPower | After literally 5 years of running just fine, my personal mail server just up and stopped working. |
| 00:01.43 | ManxPower | well the SMTP did |
| 00:01.48 | SublimE-wk | What MTA ? |
| 00:01.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:02.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:03.02 | nvrpunk_ | ManxPower, I am getting " Attempting to exceed Jitterbuf max 600 timeslots" yet I have maxjitterbuffer=500, any idea why? |
| 00:03.23 | ManxPower | nvrpunk_: no idea. |
| 00:03.41 | ManxPower | I don't really have to worry about jitter on my network |
| 00:03.43 | nvrpunk_ | ManxPower, btw no trunking made it work well |
| 00:03.56 | *** join/#asterisk joez212 (n=jhart@CPE001c101b40b5-CM0018c0d91624.cpe.net.cable.rogers.com) |
| 00:03.59 | joez212 | hello |
| 00:04.04 | nvrpunk_ | hello |
| 00:06.15 | jeffspeff2 | i'm looking for suggestions on a voip service provider. I'm wanting to setup asterisk systems for small business, and myself. I currently use Teliax. Somebody else had mentions Route 5060 http://route5060.com while their prices are much cheaper than the $0.02 i'm paying at the moment, i'm not sure if they are what I need. Any suggestions? Thanks for the feedback. |
| 00:07.20 | SublimE-wk | Anyone at all know if those log messages are normal? I have googled, and I have found people that say they are and others that say they are not. |
| 00:09.23 | Qwell | jeffspeff2: never heard of them.. |
| 00:10.00 | jeffspeff2 | Qwell: who, Teliax or the Route guys? |
| 00:10.08 | Qwell | route5060 |
| 00:10.11 | Qwell | I know teliax |
| 00:10.19 | joez212 | anybody heard of the ata sip by grandstream? |
| 00:10.34 | Qwell | ~gs |
| 00:10.34 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 00:10.40 | jeffspeff2 | I hadn't either, but their site boasts rates at like 0.005 and lower |
| 00:10.48 | Qwell | jeffspeff2: steer clear, IMO |
| 00:11.00 | jeffspeff2 | ok |
| 00:11.10 | Qwell | there are generally 2 types of companies that can afford that price |
| 00:11.22 | Qwell | 1) telcos that interconnect themselves |
| 00:11.30 | joez212 | jbot: well i didnt know that |
| 00:11.31 | jbot | You didnt know that? |
| 00:11.39 | Qwell | 2) companies I've never heard of, and won't hear of in a few weeks/months |
| 00:11.49 | *** part/#asterisk _polto_ (n=polto@elphelut.fttp.xmission.com) |
| 00:12.00 | jeffspeff2 | true |
| 00:12.08 | Qwell | (ie; fly-by-night providers) |
| 00:12.14 | jeffspeff2 | yeah |
| 00:12.19 | joez212 | jbot: i guess that explains why i am having some much trouble setting it up lol |
| 00:12.20 | jbot | You guess that explains why i am having some much trouble setting it up lol? |
| 00:12.25 | Qwell | that is, of course, just my opinion |
| 00:12.28 | ManxPower | I honestly barely look at the per min cost |
| 00:12.33 | joez212 | oh its a bot lol |
| 00:12.47 | jeffspeff2 | Qwell, who would you suggest? does it get any cheaper than 2 cents a minute? |
| 00:13.00 | Qwell | 2c/min is about the norm |
| 00:13.01 | jeffspeff2 | joez212, yes, that's a bot. :P |
| 00:13.14 | ManxPower | Teliax has been around for a while. |
| 00:13.17 | jeffspeff2 | hmm |
| 00:13.20 | Qwell | yesm that they have |
| 00:13.50 | ManxPower | I don't know about vitelity.net but I use them for some things, I don't know how long they have been around |
| 00:14.04 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 00:14.14 | jeffspeff2 | they seem like a good company and all, but for business that place lots of calls, that seems like it could rack up a big bill quickly |
| 00:14.15 | joez212 | qwell: do you have any suggestions about the grandstream? |
| 00:14.46 | Qwell | joez212: abandon all hope |
| 00:14.53 | jeffspeff2 | lol |
| 00:15.12 | SublimE-wk | Can anyone offer a reference to a site that may help? |
| 00:15.21 | joez212 | the first issue i have, is that its a router!! |
| 00:15.35 | joez212 | it can authenticate pppoe |
| 00:15.42 | joez212 | how on earth do i disable it |
| 00:15.46 | Qwell | the ata is a router? |
| 00:15.57 | jeffspeff2 | ~ata |
| 00:15.58 | jbot | from memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 00:16.00 | Qwell | wtf wo..nevermind |
| 00:16.24 | SublimE-wk | One that has authors that can give a deffinite answer on those logs. "Auto destroying call..." and "Stopping retransmission on..." |
| 00:17.28 | jeffspeff2 | ~fxs |
| 00:17.29 | jbot | from memory, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 00:17.37 | joez212 | ~linksys |
| 00:17.38 | jbot | linksys is probably a tool of satan |
| 00:17.44 | jeffspeff2 | lmao |
| 00:17.44 | joez212 | ~sipura |
| 00:17.44 | jbot | Sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 00:17.54 | joez212 | interesting |
| 00:18.03 | joez212 | i should have got a sipura :( |
| 00:18.15 | SublimE-wk | Is my text going to this channel? |
| 00:18.24 | jeffspeff2 | yes |
| 00:18.26 | joez212 | sub yes |
| 00:18.31 | SublimE-wk | ok. Just checking. |
| 00:18.39 | jeffspeff2 | ~fxo |
| 00:18.40 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 00:18.43 | joez212 | i want to write a book about the non-existant info about grandstreams on the web |
| 00:19.08 | Juggie | ~grandstream |
| 00:19.09 | jbot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 00:19.42 | jeffspeff2 | i like the linksys one the best |
| 00:19.45 | jeffspeff2 | ~linksys |
| 00:19.46 | jbot | well, linksys is a tool of satan |
| 00:19.49 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 00:19.59 | jeffspeff2 | lmfao |
| 00:20.14 | jblack | So that's why some of my hardware has been growing horns.... |
| 00:20.15 | Juggie | ~cisco |
| 00:20.15 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 00:20.30 | SublimE-wk | Is there another help channel I should be in for answers to these kind of questions? |
| 00:20.55 | joez212 | sub: hopefully you dont have grandstream equipment |
| 00:21.12 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:21.31 | jjshoe | offtopic: anyone great with postfix? |
| 00:21.35 | SublimE-wk | I dont even know what that is... so I will have to say no. |
| 00:21.43 | jblack | jjshoe: I'm ok with it. Why? |
| 00:21.53 | SublimE-wk | I am fairly decent with postfix as well jjshoe. |
| 00:22.01 | jblack | Are you trying to send phone calls through smtp? |
| 00:22.37 | jeffspeff2 | can somebody verify the following statement of my limited understanding? You connect a RJ-11 phone device to the FXS ports on the PBX, and your lines coming in from the phone company connect to the FXO ports. ?? |
| 00:22.38 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:22.45 | SublimE-wk | Funny thing is this phone system has been setup for months and months and only 2 weeks ago did calls start not coming through. |
| 00:22.50 | outtolunc | ~thebook |
| 00:22.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:24.18 | SublimE-wk | It has a 4 port sangoma card in it too. So I seriosuly doubt it's the hardware, and the software config hasnt changed. I guess I will just have to wait for Qwest to quality test the line to their box... and go from there. |
| 00:24.40 | SublimE-wk | I really want to find out if those log messages are normal though. |
| 00:25.08 | Qwell | jeffspeff2: that is correct |
| 00:25.12 | Qwell | ~fxofxs |
| 00:25.13 | jbot | [fxofxs] An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 00:25.44 | jeffspeff2 | :) thanks. |
| 00:25.50 | jeffspeff2 | i'm learning so much! lol |
| 00:26.29 | SublimE-wk | Ok. I guess I will try this channel again tomorrow, maybe more people will be on. I have to leave the office in a few minutes. |
| 00:32.46 | jjshoe | jblack SublimE-wk I'm trying to get a script to run when I email log@mydomain, so I edt /etc/postfix/aliases and add: log |wall hi i then run postalises /etc/postaliases. followed with postaliases -q log /etc/postfix/alises and i get my command back. however when i send an email to log@mydomain i get Recipient address rejected: User unknown in virtual mailbox table; |
| 00:39.57 | husimon | I just happen to have an adtran atlas 550 with t1 interfaces cards in it. |
| 00:39.58 | husimon | what use is it? |
| 00:40.16 | husimon | it just converts t1 pri to frame relay? which can then be read in via a frame relay card to asterisk? |
| 00:40.59 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:41.35 | jeffspeff2 | ~record |
| 00:41.36 | jbot | record is probably like a big cd, but with cool skipping effects... |
| 00:43.36 | jjshoe | husimon ditch it if you can |
| 00:43.46 | husimon | jjshoe, ya i was just looking at ebay prices for them |
| 00:43.46 | husimon | hehe |
| 00:56.47 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:59.15 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 00:59.56 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 01:00.13 | *** join/#asterisk nvrpunk (n=zomgobli@81.90.21.227) |
| 01:00.38 | nvrpunk | conclusion to testing, no jitterbuffer at the midpoint seems to always be better than having one |
| 01:00.41 | *** join/#asterisk hunmonk (n=hunmonk@drupal.org/user/22079/view) |
| 01:01.46 | nvrpunk | is there a way to do includes with the extension.conf ? |
| 01:01.59 | nvrpunk | so that I can create something less verbose and more readable |
| 01:02.37 | ManxPower | nvrpunk: yes |
| 01:02.55 | nvrpunk | covered in the ast* book? |
| 01:02.59 | nvrpunk | by oreilly |
| 01:04.07 | lmadsen | nvrpunk: yes |
| 01:04.22 | lmadsen | nvrpunk: and probably covered in the extensions.conf.sample as well |
| 01:04.31 | lmadsen | you might even be able to guess the syntax... :) |
| 01:05.13 | husimon | what is the suggested type of shared storage to use for an asterisk heartbeat config? Someone mentioned to me earlier that nfs was bad because asterisk has problems with file locking |
| 01:05.30 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:05.51 | ManxPower | Well, it looks like the entire day of messages for asterisk-users has started arriving (my mail server was down today) |
| 01:06.36 | husimon | or is rsync the best idea |
| 01:07.04 | ManxPower | husimon: Isn't heartbeat outside of Asterisk |
| 01:07.22 | husimon | yes it is i just mentioned that because it would be the thing turning asterisk on and off |
| 01:07.36 | husimon | one the up or down server in the pair. |
| 01:07.38 | ManxPower | Ah, of course rsync would be easier/better/more reliable thatn NFS |
| 01:08.16 | husimon | have you played with rsyncing the asterisk db ? |
| 01:08.22 | husimon | i guess the question is, does it work |
| 01:08.23 | plik | <3 rsync |
| 01:08.29 | ManxPower | husimon: I wouldn't dream of it. |
| 01:08.47 | husimon | ManxPower, well how do you share that then ;P |
| 01:09.15 | ManxPower | I don't. rsync is just better than NFS for pretty much any task of syncing files |
| 01:09.26 | husimon | ManxPower, but I have phone forwarding in the astdb |
| 01:10.00 | ManxPower | husimon: hint: the "asterisk db" is really "Berkeley DB" (an older one) |
| 01:11.05 | ManxPower | so read up on that |
| 01:11.42 | husimon | up backing up berkely db ? |
| 01:12.42 | ftp3 | so.. if i have a asterisk server.. and i want someone to be able to call my server extension.. but they do not have a paid sip account anywhere.. is it possible? (i mean, allowing the general public to call my sip (that rings to my phone) for free.. is this possible? |
| 01:13.08 | Qwell | yes |
| 01:14.00 | husimon | they would have to register their sip client to the asterisk server eh? |
| 01:14.12 | Qwell | no |
| 01:14.41 | husimon | nm a sip call you can just connect with any sip client |
| 01:15.11 | *** join/#asterisk Zipper_32 (n=None@142.232.219.94) |
| 01:15.38 | joez212 | oh this piece of shooot grandstream |
| 01:15.47 | joez212 | why wont it see my nat router |
| 01:15.49 | joez212 | grr |
| 01:15.53 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:16.43 | ftp3 | well, what software would i use, and can someone point me at some instructions to do it? :-) i do not know how thats done.. my x-lite asks for a sip username, pw etc |
| 01:16.43 | husimon | ManxPower, btw the wiki mentions you can just copy the astdb file between servers |
| 01:25.47 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:26.00 | ftp3 | even just a clue could get me started ;-) |
| 01:26.19 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-152-9.red.bezeqint.net) |
| 01:26.38 | Zipper_32 | ftp3: What is it you want to do? Connect to an existing Asterisk server? |
| 01:26.51 | Dovid | hi guys. anyone know the "standard" channel for 1 half T1 in the US ? (trying to get ahold of my carrier) |
| 01:27.26 | Strom_C | Dovid: huh? |
| 01:27.43 | Dovid | Strom_C: I know for a full T the D-Chan is 24 |
| 01:27.54 | Dovid | is it still the same if my carrier is giving me a half T1 ? |
| 01:28.02 | *** part/#asterisk timothy_jones (n=chatzill@207.90.4.189) |
| 01:28.02 | Strom_C | that's PRI, not "T1" |
| 01:28.09 | Dovid | PRI |
| 01:28.09 | Strom_C | do you have PRI service, or partial PRI service? |
| 01:28.11 | Dovid | sorry |
| 01:28.13 | Strom_C | ok |
| 01:28.15 | Dovid | Partial pRI |
| 01:28.15 | *** join/#asterisk erojasv (n=erojasv@190.40.76.144) |
| 01:28.28 | Strom_C | usually the D-channel is still on 24 for ease of adding more channels in the future |
| 01:28.37 | *** join/#asterisk Katty (n=The@adsl-71-153-54-82.dsl.stlsmo.sbcglobal.net) |
| 01:28.44 | Dovid | ok |
| 01:28.46 | Katty | hihi |
| 01:28.53 | Katty | i have question |
| 01:29.04 | Katty | how to use astrisk?????????????? |
| 01:29.14 | Strom_C | Katty: you press the buttons |
| 01:29.21 | Strom_C | and then you do the thing |
| 01:29.24 | Strom_C | and then you're using asterisk |
| 01:29.26 | Dovid | Strom_C: if the box is set up for a full PRI (1-23 for VOICE) will it make an issue now that I have a partial ? |
| 01:29.28 | Katty | giggles |
| 01:29.45 | Strom_C | Dovid: if it tries to select unassigned channels, then perhaps. |
| 01:30.01 | Zipper_32 | Dovid: You will have to specify only to use the active channels. |
| 01:30.14 | Zipper_32 | erm, assigned channels. |
| 01:30.33 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:30.34 | Dovid | I Stom_C: It seems to be going up and down |
| 01:30.44 | Strom_C | what do you mean? |
| 01:30.52 | Dovid | i get in the CLI: Primary D-Channel on Span 1 is up |
| 01:30.54 | Dovid | then i get down then up |
| 01:30.59 | Dovid | back and forth every few seconds |
| 01:31.28 | Strom_C | that could be one of a number of issues |
| 01:31.40 | Dovid | like ? |
| 01:31.45 | Dovid | btw i have a sangoma card |
| 01:32.00 | Strom_C | circuit problems, flaky hardware, flaky card drivers |
| 01:32.03 | *** join/#asterisk tobias (n=tobias@user-0c998nt.cable.mindspring.com) |
| 01:32.07 | Strom_C | among others |
| 01:32.17 | Dovid | now i keep getting messages that it is up |
| 01:32.22 | Strom_C | does the pri still bounce if you hard-loop the interface? |
| 01:32.25 | Dovid | any setting ? |
| 01:32.37 | ManxPower | Dovid: in my experience you need to recompile everything with the latest versions -- I've only seen that where I had an asterisk/zaptel mismatch in versions |
| 01:32.37 | Dovid | hard-loop ? i am not there :( |
| 01:32.53 | Dovid | i didnt change anything on the box. the just changed the kind of PRI that they have |
| 01:33.35 | ManxPower | It could be any of the things Strom_C suggested, but I would expect alarms, at least yellow, maybe red alarms would happen around the same times at the D Chan going down |
| 01:33.40 | *** join/#asterisk hi365_w (n=hi365@mail.pcgeula.co.il) |
| 01:34.01 | kyron | so...guess no one uses HBB1 shapers... :/ anyone can suggest a traffic shaper in that case? |
| 01:34.08 | ManxPower | perhaps your switchtype is wrong |
| 01:34.24 | jeffspeff2 | what is the recommended method for allowing users a "Record" button or extension on their phone? This way if they need to record a call before or during the conversation, they can press or dial the record button/extension. |
| 01:34.25 | Katty | traffic shaper? |
| 01:34.30 | Katty | maybe some... orange cones. |
| 01:34.32 | ManxPower | kyron: what little traffic shaping we do is all on the cisco routers |
| 01:34.33 | Katty | and road blocks |
| 01:35.08 | ManxPower | jeffspeff2: do a search on "one touch recording" in The Book |
| 01:35.17 | ManxPower | and The Wiki, and may The Google |
| 01:35.19 | ManxPower | ~book |
| 01:35.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 01:35.26 | Katty | mmm, one touch recording |
| 01:35.46 | jeffspeff2 | manxpower: thanks, i didn't know what to call it. |
| 01:36.21 | ManxPower | chances are it's also mentioned in features.conf.sample, but I don't know for sure. |
| 01:36.52 | florz | kyron: what do you wanna shape? |
| 01:37.21 | *** join/#asterisk BeeBuu (n=beebuu@218.13.83.163) |
| 01:37.33 | Katty | ManxPower: you know what asterisk needs |
| 01:37.35 | Katty | ManxPower: tapi. |
| 01:39.22 | ManxPower | Speak not to me of Windows, for today (and most days) I *loathe* it. |
| 01:40.35 | ManxPower | Asterisk+TAPI: now your viruses can CALL people and voice synth all your porn stories, or bank account numbers, or passwords, or e-mails with specific key words |
| 01:40.36 | drmessano | tapi? |
| 01:40.45 | Katty | drmessano: tapi. windows api for dialing stuff |
| 01:40.46 | drmessano | Heathen |
| 01:40.52 | drmessano | I know what it is |
| 01:40.55 | Katty | k |
| 01:42.13 | ManxPower | I have a lot of trouble trying to understand why you would want a dedicated traffic shaping box unless you are shaping significant amounts of data. |
| 01:42.27 | ftp3 | zipper32 - my goal is this.. if i setup an asterisk server, and then people that live in england could call my sip without paying anything and get connected to one of my asterisk extensions |
| 01:42.51 | ftp3 | i want them to be able to just connect to my sip box for free. without an account anywhere |
| 01:43.02 | ManxPower | T-1 <-> Router <-> T-1 Where would you put the shaper? Do these things support T-1/DS-3, etc interfaces? |
| 01:43.50 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 01:43.59 | ManxPower | ftp3: seems like a lot of trouble for basically IAXtex/pulver.com system and those are free |
| 01:44.15 | grandpapadot | Hi all. In Asterisk 1.2, is there an in-call MeetMe command to mute-all exept admin? |
| 01:45.50 | jeffspeff2 | is one-touch monitor where you can listen to other calls? |
| 01:46.37 | ftp3 | manxpower: url for info about iaxtex/pulver.com system? |
| 01:47.20 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:47.33 | ManxPower | Pulver does http://www.freeworlddialup.com/ |
| 01:48.07 | ftp3 | thank you |
| 01:48.30 | ManxPower | He's a rabble rouser in the VoIP regulatory areas, doing all sorts of stuff to help get the FCC kick the telcos and make them play fair |
| 01:55.12 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:55.35 | kyron | florz, VoIP traffic going onto 1MB/s cablemodem connection... don't want the conversations to get choppy because "big e-mail coming in". Obviously, I'm using a VoIP to PSTN provider.. |
| 01:57.19 | ManxPower | kyron: 1) You can only traffic shape outgoing connection, many modern consumer (cheap) routers have primitive packet priority, and it seems to be enough for most people. |
| 01:57.27 | kyron | ManxPower, I want to make sure VoIP is prioritized and that all else is simply delayed as non-RT |
| 01:57.40 | ManxPower | I've seen several Linksys with that feature, as well as 3rd party firmware for some models of Linksys |
| 01:57.57 | florz | kyron: have you tried the linux kernel tc stuff? |
| 01:58.15 | ManxPower | kyron: If you want "sure" you don't want a cable modem |
| 01:59.02 | kyron | florz, ManxPower ...problem is I have an IPSec tunnel to keep up and don't want to add _yet another firewall/router_ in the link... I was hoping for a traffic shaper that does only that.. |
| 01:59.07 | ManxPower | If you want "works good enough I don't notice audio problems %99 of the time" then a cheap linksys router would do it |
| 01:59.45 | _ShrikE | kyron: http://www.kentrox.com/products/Q1300 |
| 01:59.52 | kyron | ManxPower, re cable modem, you're insinuating that it can be quite unreliable?.. |
| 01:59.59 | _ShrikE | works great, but not cheap |
| 02:00.29 | florz | kyron: well, why shouldn't a single box be able to so ipsec and prioritization at the same time?! |
| 02:00.47 | ManxPower | kyron: No, I'm saying it is impossible to do 100% total control of packet priorities, latency, jitter, queueing for both incoming and outgoing data for TCP and UDP. |
| 02:00.50 | kyron | well oto, I'm opened to another router solution that can do decent inter-site tunneling... presently doing this using m0n0wall but I've never been able to get their traffic shaper to work correctly |
| 02:00.57 | ManxPower | on a cable modem, yourself |
| 02:01.36 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584477.dsl.bell.ca) |
| 02:01.43 | ManxPower | But if you want something that's "good enough for almost everyone" then you can use pretty much anything. |
| 02:02.05 | kyron | ManxPower, yes, totally, I know that. You only get such luxury with a $$$ contract with supplier that grants you guaranteed bandwidth and such (QoS) over the entire path... |
| 02:02.13 | kyron | might as well rent T1 lines at that point.. |
| 02:02.55 | kyron | ManxPower, yes, of course on cable modem. Just switched because the latency was horrible over ADSL (40ms to start with) |
| 02:03.01 | florz | kyron: the bandwidth of your line is fixed? |
| 02:03.13 | ManxPower | kyron: that is not normal for DSL |
| 02:03.47 | florz | ManxPower: Well, that depends on the provider, I'd say ;-) |
| 02:04.07 | ManxPower | Unless you are at like 384K "DSL" |
| 02:04.33 | kyron | florz, I get a good 1MByte/s down and 800/900kByte up. |
| 02:04.49 | florz | nope, that really doesn't have much to do with bandwidth, it mostly depends on the interleaving depth on the line |
| 02:04.51 | kyron | ManxPower, I was supposed to be 5MB DSL...which bombed on me.. |
| 02:05.32 | ManxPower | *nod* That's a provider issue, not really a technology issue. |
| 02:05.35 | kyron | florz, I'd think ManxPower knows the difference between latency and BW ;) |
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| 02:06.01 | kyron | at any rate, got static IP, non-blocked, 5x faster Internet access.. |
| 02:06.25 | ManxPower | Trust me, a 56k modem has much more latency than a T-1 |
| 02:06.25 | kyron | (oh...and found out one of my router cards was either defective or really badly supported under BSD...but shhhhhhhhhhh) |
| 02:07.18 | ManxPower | Any time you look at strict prioritized packet queueing, you realize how interrelated bandwidth and latency are. |
| 02:07.24 | drmessano | Having a big VoIP switch for people to use is great, or you can just use FWD or Sipphone and not waste b/w |
| 02:07.36 | florz | kyron: yeah, but the latency obviously is influenced by the bandwidth as well, for any non-zero-sized packet - it just happens that there is another parameter that influences the latency on DSLs |
| 02:07.49 | kyron | ManxPower, ok, there is a frequency issue here and also some packet size...bet let's not debate on stuff we already know ;) |
| 02:08.03 | ManxPower | florz: there are many things that impact latency on DSL. |
| 02:08.27 | florz | ManxPower: well, let's rather say: a configurable parameter ;-) |
| 02:08.53 | ManxPower | high latency to the first hop I would expect to be an oversubscription of the link between your ISP and the physical plant provider |
| 02:10.05 | kyron | okokokok...so...so... /me needs a router that does good traffic shaping _and_ can set up at least 2 VPN connection simultaneously |
| 02:10.35 | florz | well, yeah, also could be - but at least Deutsche Telekom by default configured lines with pretty high interleaving depths, leading to latencies > 40 ms even though links most certainly are not overloaded |
| 02:10.47 | kyron | the latency was between my router and the ISP's router (subletting BELL's infra...) |
| 02:11.29 | florz | (and the latency really is to the DSLAM - even PPPoE is affected by the latency) |
| 02:13.51 | florz | kyron: well, I guess the linux kernel tc stuff might be useful - maybe http://mshaper.florz.de/ could be useful, too - however, if you don't have any of the DSL specific problems, you might want to stay with the kernel stuff for ease of use |
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| 02:17.43 | kyron | is starting to think crazy thoughts...use VoIP machine as router...wooOOOoooOOOOooo! |
| 02:18.00 | kyron | is trying to bring down the machine count... |
| 02:18.14 | kyron | doh!...that doesn't fix the VPN problem... |
| 02:19.15 | ManxPower | sveasoft will do both, IIRC |
| 02:19.35 | ManxPower | Granted, if you use their software you are going to hell, but I never let that concern me too much. |
| 02:19.56 | ManxPower | hums a Grateful Dead song to himself. |
| 02:20.58 | drmessano | At least you're enjoying the ride? |
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| 02:21.32 | kyron | another note... if a 60$ box can do the trick, I won't spend 4 days figuring out how to do it and then try to find the hardware to implement it on with the risk that said hardware will fail... |
| 02:22.10 | drmessano | A cheap router for VPN will suck ass |
| 02:22.14 | drmessano | Trust me |
| 02:22.42 | drmessano | I have a customer with 3 locations that has cheap routers |
| 02:22.46 | drmessano | We reboot them once a week |
| 02:22.57 | kyron | drmessano, yes, I lost much time and $$$ with the damned WRT54G (or what was it's name) |
| 02:23.32 | drmessano | A $100 Linksys is gonna be the same way |
| 02:23.37 | drmessano | and so will a $200 linksys |
| 02:23.43 | kyron | that one cost 250$ at the time.. |
| 02:23.47 | drmessano | You need a sonicwall or a decent Cisco |
| 02:24.13 | kyron | well, the m0n0wall on CD I used kicked the linksys solution out, kept them for wifi but now I need QoS... |
| 02:24.24 | kyron | ouch...cisco.. |
| 02:24.33 | keith4__ | someone refresh my memory... I'm trying to setup Digium hardware on a new asterisk box. built zaptel modules, using the sample zaptel.conf with just 2 FXO modules in a TDM410. when I run ztcfg |
| 02:24.50 | keith4__ | ... I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" |
| 02:25.14 | drmessano | kyron: You either want it to work, or you dont |
| 02:26.14 | kyron | drmessano, I gage wether it's worth buying or worth my time... and those solutions make my time more interesting sine I'll be getting the $$$ |
| 02:26.23 | ManxPower | modprobe wctdm (or whatever driver the Zaptel README suggests) |
| 02:26.51 | keith4__ | it's loaded |
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| 02:27.06 | ManxPower | then your config does not match you hardware. |
| 02:27.17 | keith4__ | ugh |
| 02:27.17 | drmessano | So you're looking to screw your customer by buying something usable, but high maintainence so you're all set for return trips? |
| 02:27.25 | ManxPower | For example, are you SURE the modules are in ports 1 and 2 and not ports 3 and 4? |
| 02:27.46 | keith4__ | they're numbered 1 and 2 on the board |
| 02:27.57 | keith4__ | let me make sure |
| 02:28.45 | keith4__ | hmm. maybe I didn't reboot with the new kernel...? |
| 02:29.25 | keith4__ | wtf. udev loaded "wctdm24xxp" |
| 02:29.41 | keith4__ | does the TDM410 use a different module than the TDM400? |
| 02:30.07 | ManxPower | the Zaptel README would tell you that. |
| 02:30.24 | ManxPower | type "less READE", go ahead, it doesn't bite. |
| 02:30.30 | keith4__ | is scared |
| 02:30.58 | keith4__ | yay! it works |
| 02:31.00 | keith4__ | does a jig |
| 02:31.46 | Juggie | yay |
| 02:31.58 | ManxPower | keith4__: On Mandriva (Redhat-like) "make config" installs the zaptel init script, as well as /etc/sysconfig/zapata which lists what modules to load |
| 02:32.17 | keith4__ | too bad I'm a debian man |
| 02:32.26 | drmessano | That is too bad |
| 02:32.58 | ManxPower | Oh, I'm sure debian is atuodetected and "make config" should to the right thing. |
| 02:33.04 | keith4__ | correct |
| 02:33.14 | keith4__ | it's just been too long since I had to setup asterisk |
| 02:33.21 | keith4__ | (with analog hardware, anyway) |
| 02:33.23 | ManxPower | But I have no idea what that right thing is or how to make it do a more-right thing |
| 02:33.48 | ManxPower | business must be pretty bad to go back to installing analog. |
| 02:34.42 | keith4__ | well, an office of 3 people with only 2 lines... no need to make it complicated |
| 02:34.58 | keith4__ | existing wiring, old office, yadda yadda yadda |
| 02:34.59 | keith4__ | not my call |
| 02:34.59 | ManxPower | no need to even have a PBX |
| 02:35.19 | keith4__ | well, they want IVR, voicemail, etc. |
| 02:35.22 | keith4__ | shrugs |
| 02:35.45 | ManxPower | As, I said, business must be pretty bad to accept a project like that. |
| 02:36.53 | keith4__ | yah, well. we're their "computer people" |
| 02:37.09 | keith4__ | this is but a small part of the ridiculousness |
| 02:38.52 | ManxPower | I think the term is "techbitch" |
| 02:39.21 | ManxPower | 8-) |
| 02:39.52 | keith4__ | bingo |
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| 03:06.24 | husimon | kyron just as a thought I'd probably just setup the cheap shit and then keep cold spares that are configured the same and then you just drop them in when they fail. |
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| 03:18.39 | jeffspeff2 | ~chanspy |
| 03:18.39 | jbot | well, chanspy is an application that adds the ability to spy on any bridged call, this includes VoIP only calls where ZapScan/ZapBarge couldn't this can. As of october 19 2004, ChanSpy is not included in the standard Asterisk distribution or the development CVS tree. |
| 03:19.03 | jeffspeff2 | ~automon |
| 03:19.15 | jeffspeff2 | ~monitor |
| 03:19.16 | jbot | hmm... monitor is A device for viewing the output from a computer, traditionally a much more precise TV set. |
| 03:19.51 | jeffspeff2 | ~extenspy |
| 03:22.02 | jeffspeff2 | I'm trying to set up one-touch recording and hopefully monitoring like chanspy or something. But from what I've read, chanspy won't work when also using monitor. I've researched these on voip-info, but everything seems really spotty on where to enable it and the syntax to use. There are several different pages on the use of monitor, and none of them say the same way to implement. Could somebody please give me some pointers, or a sol |
| 03:22.02 | jeffspeff2 | id source of info to get this done? And I couldn't find anything relevant in The Book either. |
| 03:22.45 | acxty | hey guys, My ITSP says I have 50 simultaneous call allow to my line, but the problem is that I can only have one call at the time and the other call sounds busy. I set CLI to verbose 9 and sip set debug and never see the other call arriving. Reading on some forums and pages it says that maybe is a NAT issue? may someone give me a glue on this |
| 03:23.46 | Strom_C | acxty: well, is it busy tone or is it reorder tone you're hearing? |
| 03:23.56 | acxty | busy tone |
| 03:24.14 | Strom_C | regular busy tone? not "fast busy"? |
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| 03:25.58 | moy | Any MFC/R2 user around from Argentina? I need Argentina testers for R2 support in chan_zap |
| 03:26.38 | Mahmoud | how to pass DTMF from Google Talk client on my PC to my Asterisk that's running chan_gtalk.so ? |
| 03:27.21 | acxty | Strom_C, I verify is not a busy tone, is a reorder tone, faster than a busy tone right |
| 03:27.58 | palichis | hi, I need help, but my english is bad |
| 03:28.52 | Strom_C | acxty: yes, reorder tone is faster than busy tone |
| 03:29.09 | palichis | I need conect a celular base in my asterisk by fxo, but the call external is not reconosing |
| 03:29.36 | acxty | Strom_C, well, I compare tones right now with a different telephone, and the itsp line sounds with a faster busy tone |
| 03:29.45 | acxty | what might be the problem? |
| 03:29.56 | Strom_C | acxty: error condition, not a busy condition |
| 03:30.14 | acxty | sip.conf issue or ITSP issue? |
| 03:30.31 | Strom_C | I can't tell based on the information you've given me |
| 03:30.50 | acxty | Strom_C, what else would you need to know |
| 03:31.18 | Strom_C | which ITSP? what are you running on the PBX? what does your network look like? |
| 03:33.38 | acxty | well the ITSP is non US or some know. I live in Honduras. The PBK is does not have anything right now. you just make a call and a Background() start. My Network, right now Is alone. I have a public ip address assing to that machine |
| 03:34.31 | acxty | no nat |
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| 03:38.34 | acxty | Strom_C, are you still there? |
| 03:39.04 | Strom_C | acxty: make sure the ITSP is doing everything correctly |
| 03:39.46 | acxty | thanks, will call them then |
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| 03:46.27 | palichis | hi, I need help, but my english is bad |
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| 03:46.41 | drmessano | My help is bad too |
| 03:46.46 | nvrpunk | :) |
| 03:46.59 | drmessano | I wonder what "Inpatient little bitch" is in his native tongue |
| 03:47.12 | nvrpunk | impatient? |
| 03:48.01 | drmessano | ah yes |
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| 03:59.59 | Mahmoud | aghh disapointing. this feature is canceld due to lack of testing.. http://bugs.digium.com/view.php?id=8659 |
| 04:01.06 | Mahmoud | it was exactly what I needed |
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| 04:11.31 | Juggie | Mahmoud, feel free to test it and update it to latest trunk |
| 04:12.35 | Mahmoud | Juggie, where to download the patch to begin with? |
| 04:13.01 | Mahmoud | hmm attached files, nvm |
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| 04:58.57 | acxty | Strom_C, may yoy have a look at this http://dpaste.com/47863/ |
| 05:00.17 | acxty | that appear on cli |
| 05:02.54 | acxty | take a look on line 25, shouldn't it appear myuser@myipaddress? |
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| 05:05.12 | jeffspeff2 | I'm trying to set up one-touch recording and hopefully monitoring like chanspy or something. But from what I've read, chanspy won't work when also using monitor. I've researched these on voip-info, but everything seems really spotty on where to enable it and the syntax to use. There are several different pages on the use of monitor, and none of them say the same way to implement. Could somebody please give me some pointers, or a sol |
| 05:05.12 | jeffspeff2 | id source of info to get this done? And I couldn't find anything relevant in The Book either. |
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| 05:12.02 | [TK]D-Fender | jeffspeff2, lookup "features.conf" on the WIKI. |
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| 05:31.00 | acxty | hey guys, I can I define in register => to use context incoming instead of default? |
| 05:32.14 | Juggie | register doesnt define a context because its not for incoming calls |
| 05:32.22 | Juggie | its to register to a remote sip provider |
| 05:33.07 | Juggie | you would do that in your peer |
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| 05:36.28 | acxty | Juggie, I am having problem with simultaneous call. My ITSP says that I can have 50 simultaneous call using there line at the same time, but when I am handling one call, the other call receive a reorder tone |
| 05:36.42 | acxty | and I never see it on the cli |
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| 05:40.38 | Juggie | sounds like your itsp |
| 05:40.55 | Juggie | what end point is terminating the call? |
| 05:41.08 | Juggie | where are you routing the calls when they enter asterisk |
| 05:42.29 | HaMYaI | anyone knows why cdr_addon_mysql doesn't compile in addons 1.4.6? |
| 05:42.39 | Juggie | make menuconfig |
| 05:43.16 | acxty | I never receive the new call on asterisk, and the only think I do with asterisk right now when a new call arrive is to send them to defaul. Answer(), Background(), Hangup(), I am tyring to send them to an [incoming] context on extensions and not to default |
| 05:43.20 | HaMYaI | Juggie: ohh, thanks |
| 05:43.50 | Juggie | acxty, then you need to set context=incoming in your peer for your itsp in sip.conf |
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| 05:46.12 | acxty | Juggie, context works fine. but still not receiving anything from the other call |
| 05:46.50 | Juggie | acxty, can you pastebin your sip.conf |
| 05:46.55 | Juggie | (remove any passwords) |
| 05:47.01 | Juggie | eg in your peers, or register |
| 05:49.03 | acxty | http://dpaste.com/47867/ |
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| 05:50.02 | Juggie | theres no reason why your asterisk config would only accept one call |
| 05:50.51 | Juggie | only thing i can suggest is checking sip debug |
| 05:50.56 | Juggie | to see if you get another call comming in |
| 05:52.23 | acxty | what should I see in the sip debug. I have been searching all over but don't find anything that tell me that a new call is incoming |
| 05:52.40 | acxty | all that I can see is invites |
| 05:53.50 | Juggie | should be an invite |
| 05:54.13 | the_5th_wheel | Does anyone here use the NCID package with the sip2ncid? |
| 05:54.13 | acxty | will make some calls, and pastebin the result |
| 05:58.44 | Juggie | you should do rfc2833 signaling too |
| 05:58.49 | Juggie | if your itsp supports it |
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| 06:07.06 | acxty | Juggie, I sip set debug make some calls, and verify invites and the second call never arrive. I get From asterisk to itsp, from UNKNOW@itsp to asterisk, From phone1@itsp to asterisk but phone2 never appear |
| 06:07.23 | acxty | also I got a form user@itsp to user@itsp |
| 06:09.16 | acxty | Juggie, so wait you say, is for sure a itsp problem? |
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| 06:27.03 | Juggie | acxty, i'm no sip expert, but it sure seems that way |
| 06:27.09 | Juggie | is it outgoing calls only? |
| 06:27.42 | acxty | Juggie, no, that are incoming calls |
| 06:27.55 | Juggie | i would contact your itsp and ask |
| 06:28.02 | Juggie | it sure seems like their issue so far as i can tell |
| 06:28.04 | Juggie | if you want to test it |
| 06:28.07 | acxty | from itsp to asterisk |
| 06:28.14 | Juggie | try registering like a soft sip client to them |
| 06:28.16 | Juggie | like xten |
| 06:28.22 | Juggie | (disable asterisk first of course) |
| 06:28.28 | Juggie | and see if you can make multiple calls to it |
| 06:30.24 | acxty | thanks, I will do thta |
| 06:30.28 | acxty | *that |
| 06:32.54 | Daviey | Hi, i'm having a problem with a Sangoma card.. I believe the card is setup correctly, but getting: |
| 06:32.57 | Daviey | *CLI> module load chan_zap.so |
| 06:32.59 | Daviey | any ideas? |
| 06:33.02 | Daviey | [May 2 07:31:35] ERROR[11603]: chan_zap.c:10937 process_zap: Unknown signalling method 'pri_cpe' |
| 06:33.39 | Daviey | (Asterisk SVN-branch-1.4-r115102) |
| 06:34.57 | Juggie | did you install lib pri? |
| 06:35.10 | Daviey | I'm pretty sure, let me check |
| 06:35.55 | Juggie | you needed to install libpri, then zaptel, then asterisk |
| 06:36.07 | Juggie | i believe sangoma has some sort of install script, though i have never used it myself |
| 06:36.17 | dominic1 | Do you know any alternatives to flash operator panel? |
| 06:36.28 | Juggie | no |
| 06:36.50 | Daviey | Juggie: seems i removed it :(.. /me recompiles - thanks |
| 06:39.28 | Daviey | dominic1: FoP is quite powerful, and has lots of custom options.. if a little hacky |
| 06:39.37 | Daviey | It's the only webbased panel that i know of |
| 06:43.54 | dominic1 | yes, but my main proble is, that it's only possible to manage about 100 extensions. We need a lot more |
| 06:45.27 | Daviey | yeah, i'm using it on a site with ~170, and it is pretty crappy for that :( |
| 06:46.16 | dominic1 | Yes, we will need it for about 300 users + conference rooms |
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| 06:47.52 | Defraz | Hudlite won't work? |
| 06:49.18 | dominic1 | @Defraz oh, didn't see that yet. I will have a look on it. If anybody has further ideas, please write me! Thanks a lot- |
| 06:50.01 | Defraz | I love it and it works pretty well. |
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| 06:59.26 | dominic1 | Where can I get the hudlite server? |
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| 07:08.14 | dominic1 | anybody using hudlite without trixbox? |
| 07:09.04 | dominic1 | Is it possible to customize trixbox like asterisk? |
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| 07:17.12 | mort_gib | -No |
| 07:17.36 | mort_gib | ~trixbox |
| 07:17.39 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
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| 07:53.43 | dominic1 | Hello guys. I am currently testing some phones. I like the snom 370 and the polycom 650 most. The snom has the better software and the poly the better mic and speaker. Is it possible to costomize the poly. Like setting buttons to an url or stuff like that? |
| 07:53.57 | dominic1 | With the snom this is pretty easy |
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| 08:26.16 | E-bola | Hello |
| 08:28.23 | E-bola | Anybody here who uses snom phones (320 and 360) with asterisk and their blf buttons/led's? I cant seem to get the lights going |
| 08:29.29 | s0ck | E-bola: i tried using blf on a 360 about 6 months ago |
| 08:29.40 | s0ck | the lights would flash etc but it would never pick the bloody call up lol |
| 08:29.56 | s0ck | i would hope it has been addressed in a firmware update by now |
| 08:29.56 | mort_gib | Yeah, I use them |
| 08:30.15 | mort_gib | I use the latest firmware, I get buttons working fine |
| 08:30.20 | E-bola | mort_gib: i got it working on another asterisk version 6 motnhs ago |
| 08:30.24 | E-bola | with a 320 though |
| 08:30.31 | E-bola | but this new 360 doesnt wann light up at all |
| 08:30.39 | E-bola | newest firmware |
| 08:30.41 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 08:30.43 | mort_gib | With 300 and 370, I'm not using any in between |
| 08:30.55 | mort_gib | Eh, save the M3, but no buttons there |
| 08:31.13 | E-bola | mort_gib: Know if there have been any code changes in asterisk 1.4.x thats addressed blf stuff? |
| 08:31.18 | E-bola | i can find a german aptch and howto |
| 08:31.26 | E-bola | but my german isnt good enough to understand it hehe |
| 08:31.27 | mort_gib | I'm on 1.4.19 |
| 08:31.46 | mort_gib | No, It worked more or less straight off, using hints |
| 08:31.55 | E-bola | i got it working on a 1.2.13 many months ago |
| 08:32.18 | E-bola | the system i acnt get working is 1.4.11 |
| 08:32.38 | mort_gib | E-bola: I would be tempted to update |
| 08:32.44 | mort_gib | Pronto |
| 08:32.47 | E-bola | mort_gib: so setup hints for each phone u wanna monitor, and setup function keys in snom to use Extension"type" and then the number of the hint? |
| 08:33.00 | E-bola | mort: i will update today, im just afraid it wont solve this problem |
| 08:33.07 | mort_gib | Eh, no the number of the extension |
| 08:33.18 | E-bola | well in my case the number is the same |
| 08:33.36 | mort_gib | So you start the extension you want to monitor with exten => 100,Hint |
| 08:33.43 | E-bola | i think the problem is that: core show hints |
| 08:33.46 | E-bola | always reports state as idle |
| 08:33.51 | E-bola | no matter what the phones are doing |
| 08:34.01 | mort_gib | Because you haven't put in hints |
| 08:34.14 | E-bola | i have... maybe its a context/security issue |
| 08:34.27 | *** join/#asterisk nix_rbi (n=nix@193.198.162.14) |
| 08:34.28 | E-bola | exten => 100,hint,SIP/100 |
| 08:34.28 | E-bola | exten => 101,hint,SIP/101 |
| 08:34.31 | E-bola | thats how my hints look |
| 08:35.19 | mort_gib | Well, you have to use the right context in sip.conf |
| 08:36.13 | E-bola | i put the hints in the "Internal" context and thats what the sip accounts are in as well |
| 08:36.59 | mort_gib | in sip.conf |
| 08:37.15 | E-bola | ya its the same |
| 08:37.31 | mort_gib | and still nothing?? |
| 08:37.44 | E-bola | no only from the 360 phone |
| 08:37.49 | E-bola | i have 18 320's and 1 360 |
| 08:38.04 | mort_gib | So it only work from the 360, not the 320?? |
| 08:38.05 | E-bola | the asterisk reports the status of the 360 correct, but not the 320's |
| 08:38.25 | E-bola | i havent tried monitoring the extension of the 360 on one of the 320's but thats not what i want anyway |
| 08:38.26 | Mahmoud | what happen to dundi? http://www.dundi.com/dundi.txt |
| 08:38.32 | pukkita | I'm using asterisk 1.2.x, using thomson ST2030 (2 line) and want to return a different tone (busy for example) or play a warning if someone calls a ST2030 with 1 active line, right now it simply looksl ike the phone not being picked up |
| 08:38.41 | E-bola | the 360 is the receptionist phone which shuld monitor the state of all the employees |
| 08:40.19 | mort_gib | pukkita: Have a look at ChanIsAvail |
| 08:41.19 | mort_gib | E-bola: Well it looks like you are doing it right, it works for my Snom 300 using exten => XXX,hint,${sipvar} |
| 08:41.44 | E-bola | mort_gib: what pussles me is that it works for the 360's extension but not for any of the 320's |
| 08:42.22 | E-bola | Hmm in my 1.2 system where it works |
| 08:42.28 | E-bola | show hints, lists the extensions without a context |
| 08:42.28 | mort_gib | Well... I have it working with the more basic model 300 so, you have to go over the config again! Are you using the latest Firmware?? |
| 08:42.39 | E-bola | but in my 1.4 where it doesnt work they are listed as extension@Internal |
| 08:42.47 | E-bola | mort_gib_ how are urs listed? |
| 08:43.56 | *** join/#asterisk af_ (n=getsmart@88-149-230-0.dynamic.ngi.it) |
| 08:43.57 | mort_gib | like: 120@internal :SIp/120 State:busy (or idle or... |
| 08:44.32 | mort_gib | The important part is watchers though... |
| 08:44.41 | E-bola | watchers register correctly |
| 08:44.48 | mort_gib | Yes |
| 08:44.51 | E-bola | i setup the 360 to monitor all extensions, and each hint has 1 watcher |
| 08:44.54 | E-bola | so that part apears to work |
| 08:45.03 | E-bola | its jsut the state never changes |
| 08:45.17 | mort_gib | I think you have a context issue! |
| 08:45.41 | mort_gib | Do you have a context=internal under every sip user |
| 08:46.50 | E-bola | Yes |
| 08:48.29 | E-bola | Should the console report the state of the hints? |
| 08:49.19 | mort_gib | Yes |
| 08:49.43 | mort_gib | Are you on the latest firmware?? |
| 08:49.54 | E-bola | not on the 320's, i am on the 360 though |
| 08:50.03 | E-bola | or well i think one of the 320's are the newest |
| 08:50.21 | E-bola | but i mean shouldnt asterisk be able to monitor an extension independently of what sip device is using that extension? |
| 08:50.38 | mort_gib | It should yes |
| 08:51.04 | mort_gib | All devices are in the same context?? |
| 08:51.10 | E-bola | Yes |
| 08:51.14 | E-bola | Internal |
| 08:52.02 | mort_gib | -Sorry, that's all I can do then... |
| 08:52.23 | mort_gib | 1. Start ANY extension with a hint statement |
| 08:52.41 | mort_gib | 2. Use the SAME context in extensions.conf and sip.conf |
| 08:53.01 | E-bola | mort_gib: my hints are all grouped together, should it make a difference if they are at the top of the normal extension stuff? |
| 08:53.42 | mort_gib | 3. Program the button to type "extension" ans make sure that the Snom completes the extension like <sip:ext@*> |
| 08:54.08 | mort_gib | ANY hint has to come just before the exten => ZZZ,1,... |
| 08:54.21 | mort_gib | At least that's what they do on my working install! |
| 08:54.44 | mort_gib | so exten => 120,hint,${shane} |
| 08:55.01 | E-bola | hmm i can try to move them there, thats not how it looks on my working 1.2 install though, But cant hurt |
| 08:55.05 | mort_gib | exten => 120,1,Dial(${shane}, tT... |
| 08:55.24 | mort_gib | I can't say for sure, I never did any 1.2 installs |
| 08:57.46 | E-bola | Alright, well thanks for your help. I hope i can get it working |
| 08:58.03 | E-bola | Btw. upgrading a 1.4 to newest 1.4 versino should be relatively painless right? Its just running the builds and then make install? |
| 08:58.23 | mort_gib | Yes, but remove modules first! |
| 08:59.06 | mort_gib | I had/have issues when just compiling. And shut down Zaptel and * first |
| 08:59.13 | mort_gib | So |
| 08:59.26 | mort_gib | 1. shut down *, Zaptel |
| 08:59.44 | mort_gib | 2. rm -f /usr/lib/asterisk/modules/* |
| 09:00.15 | mort_gib | 3. Do a normal install, zaptel first and DON'T do make config in the asterisk dir ;-) |
| 09:00.32 | E-bola | hehe i got nitely config backups no worries :) |
| 09:00.45 | mort_gib | AS do I, but still |
| 09:01.05 | E-bola | hates that he has to do this during the day |
| 09:01.11 | E-bola | Getting all new modules scares me a bit hehe |
| 09:02.03 | mort_gib | Well... if you don't you will have no end of complaints that "this modules was compiled for a prior version of * and might not be compatible" crap |
| 09:02.29 | E-bola | i know |
| 09:12.09 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 09:12.35 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
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| 09:33.06 | dominic1 | anybody using CTI with asterisk? |
| 09:34.47 | _gm | dominic1: CTI? |
| 09:37.17 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-89-187.w90-54.abo.wanadoo.fr) |
| 09:37.32 | pukkita | I'm using asterisk 1.2.x, using thomson ST2030 (2 line) and want to return a different tone (busy for example) or play a warning if someone calls a ST2030 with 1 active line, right now it simply looksl ike the phone not being picked up. ChanIsAvail doesn't seem to work, it always return 0 |
| 09:45.59 | MaartenB | hi guys |
| 09:46.22 | MaartenB | is it possible to show the callerid with 'show queue supportqueue' ? |
| 09:59.14 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 10:01.16 | mort_gib | Any idea what "WARNING[2270] chan_sip.c: Remote host can't match request BYE to call" means |
| 10:01.21 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 10:01.42 | E-bola | is upgrading to newest 1.4..... |
| 10:04.31 | *** join/#asterisk tompaw (n=tompaw@gw.krakow.tompaw.net) |
| 10:04.37 | tompaw | hello |
| 10:05.07 | tompaw | do you guys know any simple web-based sip client which I can use with custom provider (asterisk-based obviously?) |
| 10:05.08 | *** join/#asterisk gabriolo68 (n=gabriele@host177-70-dynamic.11-87-r.retail.telecomitalia.it) |
| 10:05.15 | tompaw | working on java and/or flash? |
| 10:05.21 | gabriolo68 | Hi to ev'ribody! |
| 10:05.55 | gabriolo68 | I'm here desperately searching for help with updateconfig!!! |
| 10:06.34 | gabriolo68 | does anybody could try to help? |
| 10:07.25 | gabriolo68 | basically, when I send the command to update, I receive "Success" message,.... |
| 10:08.04 | gabriolo68 | but the file remain the same, even if it looks readed and writed over |
| 10:10.35 | E-bola | mort_gib: Installing newest 1.4 didnt solve the hints problem :( |
| 10:10.46 | E-bola | the only states i see are unavaileble and idle :/ |
| 10:11.22 | mort_gib | That's better I guess.... |
| 10:11.45 | E-bola | Nah thats whats its been from the start |
| 10:11.52 | E-bola | Onyl idle and unavaileble states |
| 10:11.59 | E-bola | and no lights anywhere on the phones |
| 10:12.27 | mort_gib | And you have hints before the 1 extension statement |
| 10:13.13 | mort_gib | WARNING[2270] chan_sip.c: Remote host can't match request BYE to call -HAve anyone had this one before?? |
| 10:13.18 | E-bola | i now put hints infront of every extensions normal dialplan stuff |
| 10:13.18 | mort_gib | ??? |
| 10:13.42 | mort_gib | Ok, now is when you enable sip debug on your fancy Snom phones |
| 10:13.47 | mort_gib | :-) |
| 10:14.58 | E-bola | mort_gib: but isnt the problem asterisk? I mean who is responsible for changing the state? Asterisk or the sip device? |
| 10:15.39 | E-bola | I mean, because asterisk itself cant see whats going on on the extensions, except that they are idle or unavaileble |
| 10:16.38 | E-bola | http://wiki.snom.com/Talk:Snom360 hmm that sugests blf is broken... |
| 10:16.41 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 10:21.09 | *** join/#asterisk fmueller (n=user@p548F58EF.dip.t-dialin.net) |
| 10:21.15 | Guggemand | E-bola: do you have call-limit = something in sip.conf on the phone setup ? |
| 10:22.22 | Guggemand | or busy-level=1 in sup.conf |
| 10:22.23 | Guggemand | sip.conf |
| 10:22.23 | E-bola | Guggemand: No |
| 10:22.26 | mort_gib | Guggemand: That's right, you need that too |
| 10:22.28 | E-bola | No to both |
| 10:22.33 | E-bola | Thats necesary? |
| 10:22.35 | Guggemand | you need one of those |
| 10:22.45 | mort_gib | He is right, -Sorry! |
| 10:22.48 | E-bola | Ile try it straight away... |
| 10:23.07 | mort_gib | WARNING[2270] chan_sip.c: Remote host can't match request BYE to call -Nobody seen this?? |
| 10:23.21 | skyphyr | hi all - I've got a strange issue. Well a couple. I've put some extensions that directly dial some test numbers SIP/blah@something - and the work perfectly - sound in both directions etc |
| 10:23.42 | Guggemand | mort_gib i saw it once ... it seemed to disapper when i set canreinvite=no |
| 10:23.53 | skyphyr | but a few weeks ago my sipgate (DID) and voipstunt (PSTN termination) started behaving strangely |
| 10:24.03 | skyphyr | they were working and config hadn't changed |
| 10:24.14 | skyphyr | you'd get calls with audio only in one direction or neither |
| 10:24.26 | skyphyr | and there is no ringing sound on the caller end for DID |
| 10:24.39 | skyphyr | I didn't have any indications etc and the module wasn't loaded |
| 10:24.58 | skyphyr | so I added those (even though I had no idea why it would have worked before) and it still doesn't work |
| 10:25.42 | E-bola | Guggemand: What are the consequences of setting call-limit to 1 in terms of transfering calls? |
| 10:25.53 | E-bola | I mean I guess u got 2 calls when u make an attended transfer? |
| 10:25.56 | Guggemand | just set ot to 10 :) |
| 10:26.12 | skyphyr | though the ones not working properly are in the authentication section of my sip.conf so I guess they're used differently from when I just dial a SIP number directly |
| 10:26.39 | E-bola | Guggemand: well in terms of snom's, will it show it as being busy, even if it hasnt used up its calllimit? |
| 10:26.45 | Guggemand | yes |
| 10:27.00 | Guggemand | asterisk shows it as busy when 1 line is used (when call-limit is set to something) |
| 10:27.11 | Guggemand | all mine is set to 10 |
| 10:27.29 | E-bola | ahh perfect |
| 10:28.15 | E-bola | Ahh we got progress :) |
| 10:28.28 | E-bola | Now the 360 is reporting the state correctly, both ringing and busy |
| 10:28.42 | E-bola | But i dialed the 360 from a 320, shouldnt both be busy when they have a call going? |
| 10:28.53 | E-bola | Only the called extension is showing busy, not the one i called from.... |
| 10:29.06 | mort_gib | E-bola: Try core sho whints |
| 10:29.22 | E-bola | mort_gib: thats what im using |
| 10:29.27 | E-bola | show hints/core show hints |
| 10:29.32 | mort_gib | Guggemand: Only I have caninvite=no always |
| 10:29.43 | Guggemand | mort_gib: then i dont know |
| 10:29.56 | Guggemand | E-bola: what type do you have on the phones in sip.conf ? |
| 10:30.03 | mort_gib | ?? Calls are dropped, not very frequently, and I think it has to do with this |
| 10:30.17 | E-bola | guggemand: friend on all of them |
| 10:30.20 | Guggemand | try peer |
| 10:31.26 | E-bola | guggemand: trying... |
| 10:32.51 | E-bola | hugs guggemand |
| 10:32.55 | E-bola | that made the difference :) |
| 10:33.07 | E-bola | How come my asterisk wont monitor the extensions of friends? |
| 10:35.15 | Guggemand | dont know |
| 10:35.59 | E-bola | I cant belive i didnt spot that difference between my working 1.2 and this 1.4 |
| 10:36.05 | E-bola | Guggemand: your danish btw right? :) |
| 10:36.10 | Guggemand | yeps |
| 10:36.19 | E-bola | hehe thought so |
| 10:36.49 | Guggemand | im pretty sure 1.2 didnt require the call-limit ... and worked on friends |
| 10:37.12 | mort_gib | A lot of Danes in here |
| 10:37.14 | mort_gib | :-) |
| 10:37.15 | Guggemand | i had the same problems when i upgraded some time ago :) |
| 10:37.35 | mort_gib | I got a bit pissed of with the Snoms because of that |
| 10:37.51 | Guggemand | i like snoms :) |
| 10:37.57 | E-bola | Guggemand: Which sip phone are you using? |
| 10:38.00 | mort_gib | -So my clients wanted to do assisted transfers |
| 10:38.03 | Guggemand | snom 320/260 |
| 10:38.08 | mort_gib | no problem *2 |
| 10:38.25 | mort_gib | But when you dial a SIP url it fails! |
| 10:38.32 | E-bola | Ya we generally go for those as well. SAetup a system with the snom M#'s recently |
| 10:38.33 | E-bola | worked fairly ok |
| 10:38.40 | E-bola | M3's even |
| 10:38.52 | mort_gib | So programming all the extensions, but can't use them with *2 |
| 10:39.06 | mort_gib | Yes, I have a few M3's running, they are nice too |
| 10:39.11 | Guggemand | i havent tried the M3's yet ... but as far as i know they are the same as some dualphone phones i have |
| 10:39.24 | mort_gib | Use one at home with Voipstunt, over Vodafone 3G |
| 10:39.27 | mort_gib | HSDPA |
| 10:39.37 | Guggemand | mort_gib, cant you just call the other phone, and press transfer ? |
| 10:39.44 | mort_gib | NO! |
| 10:40.04 | mort_gib | It's a feature, It needs DMTF |
| 10:40.13 | Guggemand | i know, but transfer works too |
| 10:40.28 | Guggemand | you dont _have_ to use the feature to transfer calls |
| 10:40.44 | mort_gib | -As far as I know! If you have a fix I would treat you to a beer anywhere in Copenhagen |
| 10:41.02 | mort_gib | I know, but they want MOH and xtfer |
| 10:41.15 | Guggemand | in my setup i can just press one of the keys to another user, then the caller is on hold, with music |
| 10:41.19 | Guggemand | and the other phone dials |
| 10:41.37 | mort_gib | Yeah?? With the transfer T_TRANSFER |
| 10:41.39 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:41.42 | dominic1 | http://en.wikipedia.org/wiki/Computer_Telephony_Integration |
| 10:41.46 | Guggemand | when its picked up i talk to the other phone, and then i just hang up, or press transfer |
| 10:41.49 | Guggemand | and they are talking |
| 10:42.02 | E-bola | We have the same setup |
| 10:42.10 | dominic1 | I found a nice solution from http://www.phonesuite.de, but it's in german. |
| 10:42.17 | Guggemand | or i can press transfer, and then the key |
| 10:42.27 | Guggemand | then its directly transfered |
| 10:42.57 | dominic1 | anybody using astmanproxy? |
| 10:43.09 | mort_gib | Ok, let me have a look, only the normal transfer didn't work like that |
| 10:43.18 | E-bola | guggemand / mort_gib: Any of you doing mroe fancy things with the snom's blf's? It looks like it just indicates busy or not busy (led on or off). I had some grandstreams which could do a bit more advanced stuff |
| 10:43.44 | E-bola | Like how can you tell another co-worker that DND is on? |
| 10:43.54 | Guggemand | with app_devstate ported from 1.6 i make some dnd stuff |
| 10:43.58 | mort_gib | ChanIsAvail |
| 10:44.22 | E-bola | Ya i looked at devstate |
| 10:44.31 | E-bola | But what to do with it? |
| 10:44.36 | E-bola | all you can do is blink? |
| 10:44.50 | Guggemand | yes, i just blink both on dnd and calls |
| 10:44.59 | mort_gib | I have some rather strange setups where a user wants the call, only if he is not in a call, if he is unavail he wants forward to his mobile |
| 10:45.01 | Guggemand | so we cant really see if they are talking or dnd |
| 10:45.13 | E-bola | Ya thats what i'd like a solution for |
| 10:45.20 | E-bola | the grandstreams could show 3 colors in the led' |
| 10:45.25 | E-bola | that was kinda neat :) |
| 10:45.29 | mort_gib | if he is in a call he wants the call forwarded to his PA, if SHE is in a call he wants the call in his VM |
| 10:45.37 | Guggemand | i wish snom had that :) |
| 10:45.50 | mort_gib | I use the manager for that |
| 10:45.56 | Guggemand | actually snom has 3 modes, on/off/blink :) |
| 10:46.25 | E-bola | Guggemand: you make it so all lights are on unless ppl are on dnd or talking |
| 10:46.34 | E-bola | so if light is off phone is on dnd |
| 10:46.41 | E-bola | and if its blinking then person is talking |
| 10:46.44 | E-bola | would work |
| 10:46.49 | Guggemand | i think i have it so they are off they are normal, on dnd and blink talking |
| 10:46.53 | Guggemand | i cant remember actually :) |
| 10:47.04 | Guggemand | i never use the lights myselv :P |
| 10:47.05 | E-bola | but you had to make it manualy with devstate? |
| 10:47.21 | Guggemand | i had to get the snom phone to call an extension on dnd on and dnd off |
| 10:47.29 | E-bola | ahhh hehe |
| 10:47.30 | Guggemand | and set a custom state when they do with devstate |
| 10:47.50 | mort_gib | Have you seen AsteriskGuru Switch board?? |
| 10:47.51 | E-bola | Since its mainly gonna be the receptionist needing the lights |
| 10:47.58 | Guggemand | its not pretty, but it works |
| 10:47.59 | E-bola | i think ile prefer a webpage if they want extended pressence info |
| 10:48.15 | Guggemand | yep, a webpage or some manager app |
| 10:48.18 | Guggemand | much better :) |
| 10:48.20 | *** join/#asterisk ziro_axis (n=ziro_axi@41.208.79.111) |
| 10:48.28 | E-bola | nope hadnt seen that |
| 10:48.35 | E-bola | but been plying with the flash operator thingie |
| 10:48.39 | E-bola | which is pretty much the same |
| 10:49.11 | ziro_axis | dears i need help |
| 10:49.36 | E-bola | Anyway thanks for helping me out guggemand and mort_gib :) Må sige til hvis i skal ha en øl :P |
| 10:49.52 | mort_gib | Jeg er paa :-) |
| 10:49.52 | Guggemand | np :) |
| 10:50.31 | creativx | mm øl |
| 10:51.25 | ziro_axis | my doctor in the Univ requested me to build a softPBX and i found the astersk(R) as more convinent for me |
| 10:51.45 | ziro_axis | i have just started |
| 10:52.28 | ziro_axis | so can i find any PDF which discribing every thing about it, i.e. detailed user guid |
| 10:53.25 | ziro_axis | this is the 1st step of course you will have a great hand on supporting me in my miniProject |
| 10:53.49 | ziro_axis | thus i would like to thank any help/support in advance! |
| 10:55.12 | creativx | asterisk the guide |
| 10:55.31 | creativx | ~tfot |
| 10:55.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:55.33 | creativx | perhaps |
| 10:58.52 | dominic1 | ~tfot |
| 10:58.52 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:59.12 | dominic1 | anybody using astmanproxy? |
| 11:07.48 | E-bola | Anybody have any cool tips/tweaks for the snom 360's xml functionality |
| 11:09.23 | ziro_axis | fine it is on process of downloading |
| 11:09.47 | ziro_axis | now let us considert the problem of nating over SIP |
| 11:10.43 | ziro_axis | lets suppose the Astersk(R) server is behind one Public IP |
| 11:11.07 | ziro_axis | and i want to set up 3 SIP usert for example |
| 11:11.40 | Guggemand | add the phone to asterisk, config the phones, start calling |
| 11:11.42 | ziro_axis | almost of us are behind a NAT thus how can this problem solved |
| 11:11.55 | Guggemand | i see no problem |
| 11:12.15 | ziro_axis | let's go more detailed |
| 11:12.35 | Guggemand | off cause the asterisk needs to be reachable on the public internet |
| 11:12.38 | ziro_axis | what id you are behind local are network |
| 11:12.38 | Guggemand | like any other internet service |
| 11:13.04 | ziro_axis | what if you are behind local are network |
| 11:13.13 | Guggemand | read the docs, follow some guides, try it out |
| 11:13.16 | hwt | hi |
| 11:13.24 | hwt | i can't get video to work with canreinvite=no |
| 11:13.26 | hwt | on 1.4 |
| 11:13.29 | Guggemand | dont try to figure out what problems you are gonna get |
| 11:13.33 | hwt | i only get one-way video |
| 11:13.39 | hwt | anyone experienced this? |
| 11:14.01 | hwt | oh, and also nat=yes on both clients |
| 11:14.10 | Guggemand | ive never tried setting up video |
| 11:14.27 | Guggemand | does audio work ? |
| 11:14.32 | ziro_axis | ok |
| 11:14.40 | ziro_axis | i would like to excuse now |
| 11:14.41 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:14.53 | hwt | Guggemand: yes. |
| 11:14.59 | ziro_axis | 2h abd i will be back again |
| 11:15.04 | ziro_axis | issue still open |
| 11:15.06 | ziro_axis | C U |
| 11:15.10 | *** part/#asterisk ziro_axis (n=ziro_axi@41.208.79.111) |
| 11:15.36 | hwt | videosupport=yes and everything of course |
| 11:16.37 | Guggemand | does it work if the one not seeing video puts the other one on old and take them back ? |
| 11:17.52 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 11:20.52 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584477.dsl.bell.ca) |
| 11:21.31 | hwt | Guggemand: i can pause video, yes. |
| 11:21.39 | hwt | Guggemand: but only on one side. |
| 11:21.46 | hwt | might be codec negotiationg bug. |
| 11:21.55 | Guggemand | http://www.trixbox.org/forums/trixbox-forums/help/problem-trixbox-and-x-lite-3-one-way-video-transmission |
| 11:22.02 | Guggemand | then i guess you problem is not like that ? |
| 11:24.22 | hwt | it is. |
| 11:24.29 | hwt | when i put on hold and back it works. |
| 11:24.42 | Guggemand | you just said it was still only on one side |
| 11:25.05 | Guggemand | and that page is near the top on google when you search "asterisk one way video" |
| 11:25.08 | hwt | Guggemand: i first thought you mean pause the video, not place the actual call on hold. |
| 11:25.26 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 11:25.31 | Guggemand | does the fix on the last post work for you then too ? |
| 11:26.39 | hwt | Guggemand: that is not the actual fix, i suspect. |
| 11:26.47 | Guggemand | "last post" |
| 11:26.53 | hwt | Guggemand: there is no such codec h263+ (but there is h263p). |
| 11:27.15 | Guggemand | so you get an error when you type that in your config ? |
| 11:27.15 | hwt | Guggemand: i suspect why it is working in that case is that it is then not so many codecs. |
| 11:27.24 | hwt | Guggemand: probably a codec negotiation issue. |
| 11:29.49 | Guggemand | then just remove the p codecs and see if that works :) |
| 11:30.27 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 11:33.08 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 11:39.40 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 11:42.19 | skyphyr | ok - well I got the ringing on the inbound sipgate working again, but there's still no sound in either direction |
| 11:42.32 | skyphyr | outbound on voipcheap I can hear the person I'm calling, but they can't hear me |
| 11:42.42 | skyphyr | anyone got some suggestions? |
| 11:44.05 | hwt | Guggemand: did that, and it works! |
| 11:44.12 | hwt | Guggemand: now i only have h263 |
| 11:44.19 | hwt | Guggemand: so apparently it was a codec neg. bug. |
| 11:46.37 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 11:55.42 | hwt | does the hint extension work from mysql now? |
| 11:55.43 | hwt | realtime |
| 11:58.45 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 12:00.54 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
| 12:04.19 | Mahmoud | what happen to DUNDi? |
| 12:11.45 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 12:11.56 | *** join/#asterisk shido6 (n=shido6@74-130-57-24.dhcp.insightbb.com) |
| 12:12.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.06 | *** join/#asterisk guillote_GNU (n=guillote@host7.201-253-11.telecom.net.ar) |
| 12:18.17 | *** join/#asterisk artz22 (n=artz@190.189.0.33) |
| 12:18.19 | artz22 | hi all again |
| 12:19.08 | artz22 | I'm trying to use wengo as a peer with asterisk. Everything works alright except that the only codec used is alaw. I set it up to use gsm and ilbc if possible but it never uses them, any suggestions? The phone on the other side uses gsm |
| 12:20.38 | artz22 | maybe wengophone doesn't support gsm or ilbc |
| 12:20.54 | *** join/#asterisk ^shark_ (n=^shark_@217.194.147.193) |
| 12:22.23 | skyphyr | ok - I'm now wondering if it's when I upgraded from shorewall 3 -> 4 that it broke |
| 12:22.31 | skyphyr | my asterisk server is behind the firewall |
| 12:22.51 | skyphyr | I've got masquerading on and didn't change my config when I upgraded shorewall, but perhaps a default has changed. |
| 12:24.30 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 12:28.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:30.33 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:30.33 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:30.44 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:31.01 | [TK]D-Fender | skyphyr: * is behind it? |
| 12:36.11 | ^shark_ | hi friends, someone should take a look at the http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz its not in gzip format |
| 12:37.34 | ^shark_ | i have already downloaded asterisk 1.4-current but i cant download zaptel 1.4-current cos the file is not gzip, i wonder if zaptel 1.2-current and asterisk 1.4-current are compartible? |
| 12:38.16 | [TK]D-Fender | ^shark_: http://downloads.digium.com/pub/zaptel/zaptel-1.4.10.1.tar.gz |
| 12:38.41 | lmadsen | ^shark_: I'll have someone look into that file -- it's not gzip because it seems the symlink might be bad |
| 12:38.45 | [TK]D-Fender | ^shark_: No, you can't switch 1.2 for 1.4 like that |
| 12:38.57 | [TK]D-Fender | lmadsen: Seems to be the case |
| 12:45.00 | *** join/#asterisk nirz (n=nir@89-138-43-14.bb.netvision.net.il) |
| 12:52.44 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 12:52.53 | tinkerghost | Good morning |
| 12:58.13 | *** join/#asterisk tomfmason (n=tom@unaffiliated/tomfmason) |
| 12:59.28 | skyphyr | D-Fender - yeah - running on a different box - I have a wctdm card and didn't want the phone sitting in a closet with the server |
| 12:59.47 | skyphyr | though I'm thinking I may shift my main asterisk sans zaptel to the server box |
| 12:59.59 | skyphyr | and run an IAX client on the box with the phone |
| 13:00.39 | skyphyr | well run asterisk as an IAX client on the box with the phone and then just open up the SIP port on the server and not need worry about masq working |
| 13:01.09 | dominic1 | I used channelredirect to transfer a communcationpartner to a meetme conference. Is it possible to transfer me too? |
| 13:01.28 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 13:03.28 | [TK]D-Fender | skyphyr: masq isn't the problem, only your firewall / WAN IP settings |
| 13:03.53 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 13:03.59 | nephfl | hello |
| 13:04.00 | skyphyr | D-Fender - not sure what you mean? |
| 13:04.09 | [TK]D-Fender | dominic1: Last I read it takes the first leg and throws it somewhere else, but the other end just dies and you have no chance. |
| 13:04.25 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:04.40 | skyphyr | firewall was working, settings are the same from upgrade from shorewall 3->4 |
| 13:04.43 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-66-31.bstnma.east.verizon.net) |
| 13:04.52 | skyphyr | never put anything special in place to say this is the IP address to use |
| 13:04.58 | [TK]D-Fender | skyphyr: in * <- |
| 13:04.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.33 | dominic1 | and there is no other command? |
| 13:06.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:06.31 | *** join/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 13:09.34 | [TK]D-Fender | dominic1: Doesn't look like |
| 13:10.19 | pukkita | E-bola shoud FOP blink the leds when an extension is ringing? |
| 13:10.30 | BrianR___ | SetCIDName(Rick Astley) |
| 13:11.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.11 | rpyne | Does anyone know how to allow asterisk to return a device ID (not the callerID) |
| 13:13.35 | BrianR___ | Now I just need a rick astley audio file... |
| 13:13.38 | rpyne | Anyone has the problem where the called phone keep ringing after the caller phone hangs up if the call is not answered before the timeout period? How did you fix it in Asterisk 1.4 |
| 13:14.15 | tomfmason | I am having some trouble getting my server registered with my provider. Any one mind looking over the configs and seeing if anything sticks out http://pastie.caboo.se/190459 |
| 13:14.33 | tomfmason | it should be simple |
| 13:15.51 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net) |
| 13:17.24 | tomfmason | the odd part is that I have bindport=5060 but in sip show peers from the cli it shows 5080 |
| 13:17.45 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:17.49 | Katty | morning |
| 13:18.01 | ZaVoid | why bindport? |
| 13:18.02 | ZaVoid | not port? |
| 13:18.06 | [TK]D-Fender | tomfmason: Read up : |
| 13:18.07 | [TK]D-Fender | ~sipnat |
| 13:18.08 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:18.11 | [TK]D-Fender | ^^^^^^^^ |
| 13:18.39 | ZaVoid | morning fender |
| 13:19.37 | [TK]D-Fender | tomfmason: And stop comparing your peer to your register. They have nothing to do with each other. Second, don't mess with bindport, and jsut leave your peer "as-is" |
| 13:19.42 | [TK]D-Fender | ZaVoid: Morning. |
| 13:20.26 | [TK]D-Fender | Katty: Mew. |
| 13:20.51 | errr | Is there a long term support version of asterisk? |
| 13:21.09 | tomfmason | [TK]D-Fender: thanks |
| 13:23.03 | [TK]D-Fender | errr: ABE |
| 13:23.18 | errr | I see |
| 13:27.02 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:28.21 | tzafrir_home | errr, I believe that at least the SuSE and Debian packages try to include important fixes for a period that may or may not be suffciently long enough for oyu |
| 13:29.24 | *** join/#asterisk Tourinho (n=tourinho@201.37.118.16) |
| 13:30.09 | Tourinho | hello people, does anyone know if it is possible to user "#" insted of "*" to hangup when using "H" option on Dial command? Thanks |
| 13:30.36 | zoid99 | resfeatures.conf? |
| 13:30.55 | zoid99 | er.. features.conf |
| 13:31.37 | nephfl | anybody know why polycoms default digit map doesnt like 10x extensions? they are sent as 10 |
| 13:32.19 | zoid99 | Tourinho: in the [featuremap] section |
| 13:35.29 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:35.34 | Tourinho | I though that it was only for transfer |
| 13:35.36 | Tourinho | thanks |
| 13:35.52 | mort_gib | Guggemand: How did you setup that transfer you mentioned earlier?? |
| 13:36.31 | zoid99 | Tourunho: disconnect => * ; Disconnect |
| 13:36.47 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:36.48 | dominic1 | thank you fender |
| 13:37.39 | Guggemand | mort_gib i didnt, it just works |
| 13:37.54 | mort_gib | Not on my setup... |
| 13:39.35 | mort_gib | I get RTCP SR transmission error, rtcp halted on * |
| 13:41.12 | Katty | anyone had problems with polycoms getting their directory when set on DHCP? |
| 13:41.53 | [TK]D-Fender | Katty: How does DHCP set anything regarding the directory? |
| 13:42.15 | [TK]D-Fender | Katty: Should only look for tis file on its provisioning server. |
| 13:42.40 | Katty | [TK]D-Fender: well...if i put my polycom on dhcp, it can't find boot server. |
| 13:42.48 | Katty | [TK]D-Fender: if i put it on static (the ftp settings are still the same) it works. |
| 13:43.05 | Katty | it's very weird. |
| 13:43.54 | s0ck | Katty: have you set dhcp option 66? |
| 13:43.55 | _ShrikE | Katty: Is your dhcp server setting option 66? |
| 13:44.48 | [TK]D-Fender | Katty: Never seen it. What & where have you set the boot server? |
| 13:44.59 | tinkerghost | is anyone using Vitelity as a DID provider? IPcommunications & les.net can't provide me with a DID |
| 13:45.16 | Katty | [TK]D-Fender: the boot server is vsftpd on the same machine as asterisk |
| 13:45.21 | Katty | _ShrikE: i've never heard of option 66. |
| 13:45.39 | Katty | [TK]D-Fender: it's on the same lan... |
| 13:45.42 | Katty | [TK]D-Fender: same subnet |
| 13:45.45 | [TK]D-Fender | Katty: Where did you tell your phone to look for your provisioning server? |
| 13:45.53 | [TK]D-Fender | Katty: And what values did you set? |
| 13:46.00 | Katty | to the IP of the asterisk server. |
| 13:46.06 | Katty | 192.168.0.whatever |
| 13:46.19 | Katty | i put my lappy on that switch, and confirmed i could ftp into the server |
| 13:46.47 | Katty | server ip, user and pass were the only ones i set. |
| 13:46.48 | s0ck | what dhcp server you use |
| 13:47.04 | Katty | s0ck: that changes per client. some have linksys, some have cisco... |
| 13:47.21 | s0ck | it's usually one of the 'scope options' |
| 13:47.31 | s0ck | if you set it, they will pick that up via dhcp |
| 13:47.33 | Katty | s0ck: some have windows server doing it... others have bind or tinydns |
| 13:47.39 | s0ck | windows is easy |
| 13:47.41 | s0ck | scope option 66 |
| 13:47.58 | Katty | i don't understand what option 66 is. |
| 13:48.10 | s0ck | eg |
| 13:48.17 | s0ck | option 003 is the default gateway |
| 13:48.27 | s0ck | option 006 is the dns servers you pick up |
| 13:48.37 | s0ck | option 066 happens to be 'boot server' |
| 13:48.42 | Katty | ahhh. |
| 13:48.45 | Katty | ponders |
| 13:48.51 | Katty | so polycoms ignore their settings... |
| 13:48.54 | Katty | on dhcp? |
| 13:49.15 | s0ck | it's prolly one or the other |
| 13:49.16 | *** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp) |
| 13:49.19 | s0ck | soundsl ike you're trying to mix/match |
| 13:49.21 | Katty | that seems odd. i would think manual settings would override that |
| 13:49.59 | [TK]D-Fender | Katty: If you hard-coded the provisioning server (not 0.0.0.0) in the BootROM, then DHCP has nothingt o do with it. |
| 13:50.17 | [TK]D-Fender | Katty: I'd say go reenter your login info onto the phone, and check your file permissions. |
| 13:51.11 | Katty | well it can't be login permissions... |
| 13:51.17 | Katty | because it does work when set staticlly |
| 13:51.21 | Katty | and that has nothing to do with the login info. |
| 13:51.32 | s0ck | in that particular instance, what is running dhcp? |
| 13:51.41 | Katty | windows 2003 server/ |
| 13:51.47 | s0ck | do you have access to it? |
| 13:51.54 | Katty | no... |
| 13:52.11 | Katty | well... |
| 13:52.12 | s0ck | email the admin to set dhcp option 66 to the ip of your boot server/pbx/whatever |
| 13:52.15 | dominic1 | did anybody implement the CCBS Isdn feature with asterisk? Call back when busy |
| 13:52.18 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:52.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.24 | Katty | s0ck: they don't have an admin there. they outsource their IT (= |
| 13:52.32 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 13:52.40 | s0ck | someone will have access to it though...? |
| 13:52.44 | Katty | s0ck: it's just easier to set them all static out of the dhcp range |
| 13:52.55 | Katty | s0ck: heh, no |
| 13:53.03 | Katty | s0ck: no one ever remembers the server password |
| 13:53.10 | s0ck | lol |
| 13:53.14 | Katty | s0ck: i did redo the ip of the boot server tho |
| 13:53.15 | vlt | Hello. I'm trying to initiate SIP calls fro mbehind NAT. How can I make asterisk using a certain external host IP in the INVITE packets? I tried "externhost" in sip.conf but that didn't work. |
| 13:53.22 | Katty | s0ck: i noticed it did 198*168 |
| 13:53.26 | Katty | instead of 192.168 |
| 13:53.36 | Katty | erm * being the difference. not my typo |
| 13:53.42 | Katty | at the same time, i set them all static too |
| 13:53.51 | Katty | could be that it just didn't like 192*168*1*202 |
| 13:53.54 | s0ck | a lot of handsets to do? |
| 13:53.59 | Katty | not really |
| 13:54.04 | Katty | it's a little office |
| 13:54.23 | Katty | i'll try it when my next phone system i setup. see what happens. |
| 13:54.31 | Katty | i'd really prefer to keep it on dhcp |
| 13:54.36 | s0ck | definitely |
| 13:54.50 | s0ck | it's rather cool turning them on for the first time and everything magically working :D |
| 13:55.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:55.00 | Katty | yeah. |
| 13:55.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:55.07 | Katty | i think i'm going to keep the original login info all the polycoms have too |
| 13:55.16 | Katty | right now i make a /home/polycom directory |
| 13:55.23 | Katty | and run around changing all the phones login info |
| 13:55.44 | Katty | it would be easier to just change the boot server IP address, and nothing else |
| 13:55.53 | Katty | hugs russellb |
| 13:56.24 | Katty | but even then, i still have to put all the login sip.conf info into the polycom's IP |
| 13:56.36 | Katty | that'll be a pain for the larger installs. |
| 13:57.10 | [TK]D-Fender | vlt: Read up : |
| 13:57.12 | [TK]D-Fender | ~sipnat |
| 13:57.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:58.25 | vlt | [TK]D-Fender: Thank you. |
| 13:58.45 | *** join/#asterisk nirz (n=nir@89-138-89-252.bb.netvision.net.il) |
| 13:59.11 | *** join/#asterisk gr0mit (n=tim@dhcp4.zuk40.mot-tools.co.uk) |
| 14:00.17 | mort_gib | How do I see what modules are loaded?? |
| 14:00.25 | *** join/#asterisk chammiya (n=chamms@ip-208-109-254-83.ip.secureserver.net) |
| 14:00.27 | ManxPower | lsmod |
| 14:00.34 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:00.43 | mort_gib | Well, yes, but in * |
| 14:01.01 | ManxPower | Katty: Even easier, you can specify the boot server address in the config of your DHCP server. |
| 14:01.36 | ManxPower | mort_gib: you didn't say "asterisk modules" that would be "show modules" or "core show modules", depending on your Asterisk version |
| 14:02.50 | Katty | ManxPower: yeah, but i don't always have access to the dhcp server... |
| 14:03.05 | ManxPower | Katty: It sucks to be you. |
| 14:03.15 | Katty | ManxPower: you have no idea. |
| 14:03.20 | Katty | ManxPower: i only make 27k a year |
| 14:03.33 | Katty | ManxPower: and southern missouri is eating my soul |
| 14:04.07 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 14:04.14 | ManxPower | Katty: That is not unexpected. Move to a real place 8-) |
| 14:04.28 | Katty | ManxPower: if i move, then i can't take care of my mother. |
| 14:04.50 | ManxPower | Katty: drag her along, I bet she wants to live in a real place too 8-) |
| 14:04.56 | Katty | lol... |
| 14:04.59 | Katty | you don't know my mom (= |
| 14:04.59 | s0ck | :P |
| 14:06.10 | *** join/#asterisk zzack (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 14:07.20 | vlt | [TK]D-Fender: "localnet=..." was missing, thanks again. |
| 14:07.28 | [TK]D-Fender | vlt: You're welcome |
| 14:09.13 | ManxPower | It looks like I might be starting to sell pre-configured, pre-wired, ready to plug in Tellabs echo cancelers |
| 14:10.14 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 14:12.18 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 14:12.26 | zzack | anyone know any tricks on getting SIP/Asterisk to work through a VPN tunnel? |
| 14:12.59 | zzack | the tunnel seems to be working fine except when I call someone, i can hear them but they cant hear me, which sounds like a NAT problem, but that hsouldn't exsist in a VPN tunnel |
| 14:13.50 | *** join/#asterisk HuntsvilleAL (n=ccvp@66.0.46.210) |
| 14:13.57 | HuntsvilleAL | hello |
| 14:16.20 | [TK]D-Fender | zzack: if you are behind NAT as well you'll need to add a "localnet=" line for the remote subnet |
| 14:17.20 | [TK]D-Fender | zzack: So * doesn't think they are on the internet and require your WAN IP. |
| 14:17.23 | *** join/#asterisk Weezey (n=ohno@206.210.111.16) |
| 14:17.26 | zzack | ooo ok |
| 14:18.12 | *** join/#asterisk willianmazzardo (n=willianm@201-66-189-123.smace701.dsl.brasiltelecom.net.br) |
| 14:19.11 | Weezey | I'm trying to write some logic into my dial plan, I wanted to use phpAGI, but I think I'm missing something. If I do an EXEC NVFaxDetect how do I create an exten => fax,1, in the AGI itself, or would that have to fall back to the extensions.conf ? |
| 14:19.43 | HuntsvilleAL | ThakillaZ |
| 14:20.30 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 14:21.15 | zzack | Fedner: so a |
| 14:21.21 | zzack | sterisk tries to figure out nat? |
| 14:21.24 | [TK]D-Fender | Weezey: has to already exist |
| 14:21.38 | [TK]D-Fender | zzack: if it doesn't know that a subnet is considered local, then yes |
| 14:22.22 | mort_gib | Just installed G.729, how do I make the SIP devices use it?? I use allow=g729 in sip.conf and on each phone I have set the deault to G.729A |
| 14:22.35 | zzack | Fedner: that didn't work :( |
| 14:22.48 | Weezey | [TK]D-Fender: technically it could just call the agi again from that extension (which may be kinda hacky), so it's not a big deal, thanks |
| 14:23.04 | zzack | i dont have to restart * to load sip.conf do i |
| 14:23.13 | zzack | i just did a sip reload |
| 14:23.32 | [TK]D-Fender | zzack: thats fine. |
| 14:23.42 | [TK]D-Fender | zzack: pastebin your sip.conf masking only passwords |
| 14:23.49 | zzack | ok |
| 14:26.16 | zzack | http://pastebin.com/m69a8541d |
| 14:27.20 | zzack | 702 is just a Polycom phone behind a firewall going through a VPN tunnel |
| 14:28.08 | [TK]D-Fender | zzack: looks ok if the #'s are legit |
| 14:28.22 | [TK]D-Fender | zzack: pastebin a failed call with SIP debug enabled. |
| 14:28.54 | [TK]D-Fender | zzack: And * does have a route that knows hot to reach them, right? |
| 14:29.23 | JayTee52 | is it normal for one of my PRI spans to restart the zap channels every so often? |
| 14:30.01 | *** join/#asterisk xenonex (n=xenonex@89.218.236.171) |
| 14:30.03 | JT | JayTee52: yes |
| 14:30.16 | JayTee52 | JT, thanks. Like the nick, btw :-) |
| 14:30.21 | JT | :P |
| 14:30.22 | zzack | fender: what do you by it has a route? you mean configured in * itself or through the router? i can ping the phone, get to its web interface, etc from the * server |
| 14:31.33 | JayTee52 | JT, it's always Span 1 which I have set to do the clocking and it's currently looped to Span 2 using a crossover so I can just test calls out and back in and over IAX2 trunk to another * box. |
| 14:32.03 | JayTee52 | I figured it was a built in reset feature because I didn't see any errors in the system logs or the asterisk logs. |
| 14:32.06 | [TK]D-Fender | JayTee52: Yes. you can "prirestart=never" to avoid this if you wish |
| 14:32.17 | JayTee52 | in zapata.conf? |
| 14:32.25 | [TK]D-Fender | zzack: then it has a route to it. PB the call |
| 14:32.36 | [TK]D-Fender | JayTee52: yes |
| 14:32.44 | JayTee52 | [TK]D-Fender, thanks! |
| 14:33.10 | JayTee52 | I'm still digging through all the docs I can find on this. :-) |
| 14:34.32 | zzack | fender: this is on a production system so I tee'd the output , but the file is pretty huge... |
| 14:38.01 | *** join/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 14:38.47 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 14:41.21 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:42.38 | tinkerghost | any recommendations for a DID provider? IPcommunications & les.net can't get me numbers |
| 14:42.45 | *** join/#asterisk gitguy (n=diego@adsl-132-58.click.com.py) |
| 14:42.46 | tinkerghost | (us) |
| 14:42.53 | gitguy | hi |
| 14:43.16 | gitguy | can i have a endpoint using IAX and another endpoint using SIP? |
| 14:43.31 | gitguy | and make them both communicate together? |
| 14:43.51 | tinkerghost | git: not really - they both need to accept at least 1 common protocol |
| 14:44.15 | gitguy | ok |
| 14:44.17 | tinkerghost | git: you could use a proxy in the middle to translate |
| 14:44.17 | zzack | [TK]D-Fender: here is the pastebin http://pastebin.com/m34b19a0f |
| 14:44.25 | zzack | i cut out the blocks regarding the IP of the remote phone |
| 14:44.43 | [TK]D-Fender | gitguy: Yes, * will sit in the middle and translate |
| 14:45.11 | gitguy | [TK]D-Fender: really? |
| 14:46.10 | [TK]D-Fender | gitguy: Thats the point of *. |
| 14:48.42 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:48.58 | gitguy | [TK]D-Fender: cool, I didn't know that |
| 14:49.39 | [TK]D-Fender | gitguy: Did you think that you had to use IAX phones just because you want to use an IAX ITSP? |
| 14:50.11 | gitguy | [TK]D-Fender: uhm, no |
| 14:50.24 | [TK]D-Fender | gitguy: well you just asked pretty much the same thing |
| 14:50.37 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:50.41 | gitguy | [TK]D-Fender: I didn't asked that, maybe I didn't expressed myself correctly. |
| 14:50.55 | *** join/#asterisk LemensTS (i=CustomGT@adsl-70-238-154-213.dsl.stlsmo.sbcglobal.net) |
| 14:51.15 | gitguy | [TK]D-Fender: let me reprease |
| 14:51.31 | LemensTS | I cant find an adapter that will convert Analoug line into a sip line, so i can set it up in my asterisk. Ive seen them before... |
| 14:52.20 | gitguy | [TK]D-Fender: I have an asterisk server in chicago, I have like 22 endpoints, some endpoints are in different countries, all of them use SIP and some of them can't really connect... |
| 14:52.37 | gitguy | [TK]D-Fender: I think is probably their telcos blocking the packets |
| 14:53.48 | LemensTS | like a tdm card but external |
| 14:53.55 | [TK]D-Fender | gitguy: entirely possible |
| 14:54.09 | [TK]D-Fender | LemensTS: Linksys SPA-3102 |
| 14:54.11 | grandpapadot | gitguy: Any in China? |
| 14:54.19 | [TK]D-Fender | LemensTS: www.telephonydepot,com |
| 14:55.17 | gitguy | grandpapadot: pakistan, india... |
| 14:55.25 | gitguy | grandpapadot: india is reported to work |
| 14:55.30 | gitguy | grandpapadot: pakistan is not working |
| 14:55.33 | *** join/#asterisk ziro_axis (n=ziro_axi@41.208.72.70) |
| 14:55.52 | gitguy | I just googled "pakistan voip problems" and I foud some weird things |
| 14:56.03 | [TK]D-Fender | gitguy: Yeah I've heard of all sorts of blocakges due to telcom regulations |
| 14:56.11 | gitguy | no, "pakistan sip block problems" |
| 14:56.29 | gitguy | yeah, some governments are nasty |
| 14:56.46 | gitguy | here in Paraguay the government were blocking too |
| 14:56.57 | gitguy | not until a long time ago |
| 14:57.27 | gitguy | I'm going to try with IAX |
| 15:01.29 | *** join/#asterisk zacwolfe (n=zxwolfe@c-67-171-236-13.hsd1.or.comcast.net) |
| 15:01.34 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:01.46 | *** join/#asterisk gitguy (n=diego@adsl-132-58.click.com.py) |
| 15:02.29 | jeffspeff2 | <[TK]D-Fender> i'm looking at http://www.voip-info.org/wiki-Asterisk+config+features.conf for the one-touch recording thing, but they list two different ways; with extremely different code. Which way is the "proper" method? thanks |
| 15:04.37 | *** join/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 15:05.43 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 15:06.57 | grandpapadot | Anyone have a favorite SIP/RTP prioritizing firewall? |
| 15:07.32 | jeffspeff2 | or better yet, which way will work? |
| 15:09.00 | HuntsvilleAL | ThakillaZ |
| 15:10.34 | eric2 | grandpapadot iptables |
| 15:13.10 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 15:13.23 | HuntsvilleAL | Time to have my 2,000 calorie lunch |
| 15:13.31 | HuntsvilleAL | Flaxseeds sprinkled on chopped green peppers, and a glass of water |
| 15:15.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:18.14 | b11d` | any of you guys ever work with an NEC ElectraElite IPK? |
| 15:18.22 | grandpapadot | What the hell is a "Flaxseed"? |
| 15:18.25 | b11d` | i've got a client that wants to switch his out in favor of an astersk setup, but keep his NEC phones.. |
| 15:18.54 | Qwell | grandpapadot: seed from flax |
| 15:18.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 15:19.34 | grandpapadot | lol, nss |
| 15:19.49 | Qwell | they use it in making linens |
| 15:20.25 | grandpapadot | HuntsvilleAL: Sucks to be you I guess. I'm going to see IronMan for lunch and eating Nachoes and probably a slushy. |
| 15:20.31 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 15:21.40 | *** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com) |
| 15:22.04 | grandpapadot | HuntsvilleAL: Although i should be eating a tuna sandwhich and going to the gym ... |
| 15:23.29 | *** join/#asterisk jduggan (i=thom@s.tankengine.co.uk) |
| 15:24.18 | jduggan | hey guys, im sure you get these type of questions every day, but does asterisk or any solutiosn built around asterisk support handsets from say an Avaya PBX system? |
| 15:28.30 | grandpapadot | Google said this: http://www.voip-info.org/wiki/view/Avaya |
| 15:28.45 | grandpapadot | Looks like the 4602 is compatable, but I didn't read the whole thing. |
| 15:31.55 | gitguy | when I create a IAX2 account on my asterisk server I can't connect to it |
| 15:32.03 | gitguy | I don't even see activity on the cli |
| 15:32.07 | gitguy | with iax2 set debut |
| 15:32.09 | gitguy | debug* |
| 15:32.14 | gitguy | but I see with SIP |
| 15:32.35 | gitguy | should I open a port or something? |
| 15:33.02 | grandpapadot | Anyone ever seen this? It started about a week ago on one of our servers. It's not affecting anything, but I can't find it's source, it just repeats 5-10 times every 20-30 seconds: chan_sip.c:3967 copy_header: No field 'Call-ID' present to copy |
| 15:33.10 | grandpapadot | This is asterisk 1.2.24 |
| 15:33.56 | grandpapadot | It has to be a phone as we have a lot of identical servers not experiencing the message ... |
| 15:34.25 | grandpapadot | But I can't find the root, sip debug isn't helpful here |
| 15:34.38 | grandpapadot | And there's 300 peers .. |
| 15:35.20 | coppice | sounds like the House of Lords when there's free beer |
| 15:36.22 | gitguy | which port should i open for IAX? |
| 15:36.25 | gitguy | on the server |
| 15:37.43 | JayTee52 | gitguy, port 4569 |
| 15:38.16 | JayTee52 | iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT |
| 15:38.46 | gitguy | cool, thanks |
| 15:38.58 | jeffspeff2 | can somebody assist me with getting one-touch calling to work? I'm following the instructions at http://www.voip-info.org/wiki-Asterisk+config+features.conf for [macro-apprecord] but when I make a test call and dial *9 then hangup after a moment, there aren't any recordings; and nothing shows up in the CLI when i press *9 either. |
| 15:39.03 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 15:39.05 | gitguy | JayTee52: OUTPUT too? |
| 15:39.23 | JayTee52 | gitguy, I've never had to set it on my servers |
| 15:39.27 | gitguy | ok |
| 15:39.38 | gitguy | thank you |
| 15:39.42 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:39.43 | JayTee52 | no problem |
| 15:39.50 | b11d` | a cunning array of stunts |
| 15:40.01 | JayTee52 | lol |
| 15:40.29 | gitguy | JayTee52: nope, it doesn't register |
| 15:40.39 | jsmith-busy | jeffspeff2: First things first... is DYNAMIC_FEATURES set? (You can have your dialplan call the DumpChan() application to see) |
| 15:40.47 | JayTee52 | are you using the examples in the book? |
| 15:41.27 | JayTee52 | and is this for an IAX trunk or using an IAX softphone? |
| 15:41.36 | jeffspeff2 | i followed the example for [macro-apprecord] listed below the Dynamic_Features example |
| 15:41.39 | gitguy | JayTee52: IAX softphone |
| 15:41.48 | jeffspeff2 | JayTee52, SIP softphone |
| 15:41.59 | gitguy | JayTee52: yes, i'm using a very simple config |
| 15:42.01 | JayTee52 | jeff, sorry I was speaking to gitguy |
| 15:42.10 | jeffspeff2 | oh, sorry. |
| 15:42.46 | jeffspeff2 | jsmith-busy: are you supposed to do both examples? or just one of the two? |
| 15:43.03 | JayTee52 | gitguy, sorry I've not used Zoiper or other IAX phones, just X-Lite and Eyebeam for SIP. I use IAX for trunking between servers only. |
| 15:43.13 | HuntsvilleAL | I love this, my cisco forum is displaying ads |
| 15:43.21 | HuntsvilleAL | for linux routers, and trixbox ads |
| 15:43.25 | HuntsvilleAL | ...... |
| 15:43.26 | jsmith-busy | jeffspeff2: I looked at the page, and didn't see two separate examples... |
| 15:43.36 | JayTee52 | but you might try that same line for port 4569 and just change the INPUT to OUTPUT to make sure it's both ways. |
| 15:43.37 | jsmith-busy | HuntsvilleAL: Beautiful |
| 15:44.09 | gitguy | JayTee52: oops ok |
| 15:44.14 | gitguy | this is what i have, http://pastebin.com/m69ba3088 |
| 15:44.21 | jeffspeff2 | jsmith-busy: It says Example, then shows the Dynamic Featuers thing, then below that it says Example One Touch Recording (applicationmap) with WAV to MP3 Conversion Macro. |
| 15:44.22 | gitguy | i use SIP, it works well |
| 15:44.35 | gitguy | but i need IAX for some phones |
| 15:44.44 | jeffspeff2 | jsmith-busy: i followed the bottom one, should i have done the top one? |
| 15:44.46 | jsmith-busy | jeffspeff2: OK, I see what you're saying |
| 15:44.58 | jeffspeff2 | :) |
| 15:45.18 | JayTee52 | gitguy, is that your sip.conf or iax.conf file? |
| 15:45.34 | HuntsvilleAL | I'm not that good with website design, but does this look like an "aesthetically" pretty forum? |
| 15:45.36 | jsmith-busy | jeffspeff2: Set DYNAMIC_FEATURES, make sure you have a code defined for automon in [featuremap] (not in applicationmap) |
| 15:45.41 | HuntsvilleAL | http://theciscocommunity.com/forums/ |
| 15:45.45 | gitguy | JayTee52: iax.conf |
| 15:45.52 | jeffspeff2 | ok |
| 15:46.58 | JayTee52 | gitguy, you need to set your bindport=4569 and bindaddress=0.0.0.0 in the [general] section |
| 15:47.08 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 15:47.28 | jeffspeff2 | jsmith-busy: just so i know i'm doing this right.... add "DYNAMIC_FEATURES=>automon" to featuremap |
| 15:47.41 | jeffspeff2 | and how do i define a codec to that? |
| 15:48.08 | jsmith-busy | jeffspeff2: No, you have to define a variable in your dialplan called DYNAMIC_FEATURES |
| 15:48.25 | jeffspeff2 | ahh |
| 15:48.35 | jsmith-busy | jeffspeff2: Either a global variable (if you want everyone to be able to do it), or a channel variable |
| 15:48.38 | jeffspeff2 | so add that to the gloabals |
| 15:48.44 | gitguy | JayTee52: ok |
| 15:48.52 | jeffspeff2 | ok |
| 15:49.24 | gitguy | JayTee52: it doesn't register |
| 15:49.49 | JayTee52 | gitguy, and if you're using authentication you need to add the secret="whatever" line for your phone in iax.conf and in the softphone too. |
| 15:50.57 | jeffspeff2 | ok, i have "DYNAMIC_FEATURES=>automon" in my global, and "apps => *9,caller,Macro,apprecord" defined in my featuremap. with that macro in it's own context in the dialplan. is this correct? |
| 15:51.01 | JayTee52 | and did you do an iax2 reload after you edited your iax.conf file? |
| 15:51.16 | gitguy | JayTee52: ok, i think the problem really is on the server, because i don't see any output on the * cli |
| 15:51.24 | JayTee52 | lunchtime for me, back in a few |
| 15:51.27 | gitguy | some firewall or something |
| 15:51.49 | gitguy | iptables -L -v shows nothing, except that the rule that i added |
| 15:52.10 | JayTee52 | ok |
| 15:52.15 | HuntsvilleAL | can asterisk be integrated with exchange 2007, so I can give employees voicemail as mp3 attachments in their inbox? |
| 15:52.31 | *** join/#asterisk wichitalineman (n=svf@67.106.246.227.ptr.us.xo.net) |
| 15:52.35 | [TK]D-Fender | HuntsvilleAL: Yes, * can e-mail your VM's. |
| 15:52.46 | gitguy | [TK]D-Fender: any ideas on this issue? |
| 15:53.06 | jeffspeff2 | jsmith-busy: still not recording or showing any output in the CLI |
| 15:53.08 | HuntsvilleAL | we have some cheap ITC Deltacom IPK II pbx now |
| 15:53.11 | [TK]D-Fender | gitguy: You're showing us nothing, so no, I don't |
| 15:53.13 | HuntsvilleAL | its gimp, and can't do what we need |
| 15:53.36 | HuntsvilleAL | NEC |
| 15:53.48 | JayTee52 | gitguy, this may be of some help http://www.voip-info.org/wiki-Asterisk+firewall+rules |
| 15:54.17 | gitguy | [TK]D-Fender: i have SIP working, some accounts working... very well with SIP, but I tried to add a simple iax.conf http://pastebin.com/m69ba3088 and is not even registering, I have iax2 set debug and I don't see any output |
| 15:54.22 | gitguy | JayTee52: thx |
| 15:54.57 | gitguy | theres nothing really to show because i don't get any output |
| 15:55.06 | jsmith-busy | jeffspeff2: And you've verified that DYNAMIC_FEATURES is set by calling the DumpChan() application? |
| 15:55.22 | HuntsvilleAL | D-fender, there's this company in town that said, base 20k for labor for call manager express, excluding hardware/software prices, and customization. |
| 15:55.27 | [TK]D-Fender | gitguy: if you can't get debug on CLI then you have a networking issue |
| 15:55.34 | gitguy | yeah |
| 15:55.39 | jeffspeff2 | jsmith-busy: no.... how do i do that? |
| 15:55.42 | HuntsvilleAL | So now I'm tasked to find come cheaper path, and I stumbled across this channel via google searching |
| 15:55.46 | gitguy | [TK]D-Fender: this server has public ip address |
| 15:55.51 | BCS-Satori | is there a way to tell if asterisk has successfully connected to mysql? |
| 15:56.07 | adeel | HuntsvilleAL, if all you want is to just get voicemail to email, then it should be pretty easy to setup...although you'll need to find a way to route your calls to the asterisk box |
| 15:56.18 | jsmith-busy | jeffspeff2: exten => 12345,n,DumpChan() (where 12345 is your own extension, obviously) |
| 15:56.18 | gitguy | [TK]D-Fender: is not behind nat I think |
| 15:56.19 | [TK]D-Fender | gitguy: Doesn't matter |
| 15:56.29 | [TK]D-Fender | gitguy: Networking issue it is. |
| 15:56.33 | jsmith-busy | jeffspeff2: Then watch the CLI and see what gets printed |
| 15:58.06 | jeffspeff2 | jsmith-busy: sorry for the idiocricy, but... where do i need to define the dumpchan string? |
| 15:58.28 | HuntsvilleAL | What range of compression can the mp3's be encoded in, I wouldnt want to see a 30-45 seocnd voicemail exceed 2+ megs? |
| 15:58.29 | jeffspeff2 | in the macro context? |
| 15:58.49 | jsmith-busy | jeffspeff2: See private message |
| 15:59.52 | gitguy | JayTee52: i tried adding these iptables rules but nothing |
| 15:59.57 | gitguy | [TK]D-Fender: yep, i think is a network issue |
| 16:00.03 | gitguy | [TK]D-Fender: maybe on my side? |
| 16:00.18 | gitguy | i have a dlink router |
| 16:00.55 | gitguy | i'm going to get rid of that |
| 16:01.53 | [TK]D-Fender | HuntsvilleAL: no need for MP3, and nativ is far smaller |
| 16:01.54 | adeel | HuntsvilleAL, most of the * voicemail's are done in either wav49, wav, or gsm....and a 30 second vmail doesn't take more than 20-30kb in wav49 iirc....keep in mind it's mono encoded at 8khz |
| 16:02.25 | HuntsvilleAL | heh, guess it wont contribue to the employees 8gig email inboxes then |
| 16:03.00 | *** join/#asterisk acxty (n=acxty@201.220.132.138) |
| 16:03.06 | adeel | HuntsvilleAL, no, not reall |
| 16:03.15 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 16:03.33 | adeel | HuntsvilleAL, you could also give them a web interface to hear their emails through the browser using ARI |
| 16:04.36 | HuntsvilleAL | well, we have an existing NEC phone system, with service through a provider called ITC Deltacom. . . . its an IPK II system, am wondering how to take the voice mail portion of that system, and somehow redirect it to a possible asterisk voicemail forwarding system, to store (for web access to VM as you say), and/or to their inboxes |
| 16:05.18 | HuntsvilleAL | would existing system have to be yanked out? cant do partial parallel systems? |
| 16:06.20 | [TK]D-Fender | HuntsvilleAL: R&R |
| 16:06.43 | zerohalo | HuntsvilleAL: Anything is possible w/ *, including running external voicemail systems... all depends on how much time you have to configure it all. |
| 16:06.55 | Katty | oh boy. so, me and two managers are going to lunch to talk about $person who isn't doing their job... and $person just asked me what i was doing for lunch. i can't say i'm going to lunch with two managers to talk about you!! |
| 16:07.13 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 16:07.27 | adeel | HuntsvilleAL, asterisk can, without too much of a problem, but i've found that most PBX-in a box type stuff (nec, toshiba, cisco) require work to play nicely with anything outside of their own brand |
| 16:08.29 | HuntsvilleAL | executives have such a desire to just sit at their desk, double click an email, see icon for voicemail, click, and play, without wreaking havoc to existing phone system |
| 16:08.51 | HuntsvilleAL | since they have OWA email access, it would be nice to access voicemail from any computer in the world, via web interface |
| 16:09.05 | grandpapadot | Yea, voicemail to email rocks. |
| 16:09.20 | zerohalo | HuntsvilleAL: voicemail to email w/ wav attachments would solve this, no? |
| 16:09.26 | HuntsvilleAL | yes |
| 16:09.37 | errr | anyone ever setup nagios and used the /usr/lib/nagios/plugins/contrib/check_asterisk.pl plugin to monitor asterisk servers? |
| 16:09.38 | HuntsvilleAL | but can asterisk tie into OU's in active directory? |
| 16:09.43 | HuntsvilleAL | ldap hooks |
| 16:09.52 | adeel | i believe theres and ldap module |
| 16:09.54 | grandpapadot | HuntsvilleAL: For what? |
| 16:09.55 | adeel | at least for 1.2 |
| 16:09.59 | [TK]D-Fender | HuntsvilleAL: to do what exactly? |
| 16:10.27 | HuntsvilleAL | seamless ldap integration, instead of a lot of hacking around, so voicemail can tie in with active directory for ease of wave file redirection (inbox) |
| 16:10.39 | adeel | errr, i monitor asterisk through a different mechanism other than nagios, but yes, it's possible |
| 16:11.04 | HuntsvilleAL | we're a mixed shop here, half linux, half windows 2k3 servers |
| 16:11.06 | [TK]D-Fender | HuntsvilleAL: ?! * e-mails VM's. Thats it. |
| 16:11.24 | errr | adeel: I knew it was possible, but Im just having trouble getting it working in nagios, and was looking for some help if anyone knew anything about it |
| 16:11.27 | grandpapadot | HuntsvilleAL: yea, don't overcomplicate what could be a ridiculously simple solution |
| 16:11.59 | zerohalo | HuntsvilleAL: You may be overthinking this. If you have an email address, you have voicemail -> email |
| 16:12.05 | adeel | errr, ah, i've never used nagios....i had found a decent howto monitor * using snmp & cacti....you should be able to find what you're looking for in the nagios forums though |
| 16:12.14 | gitguy | damn, i don't know how to start fixing this network issue |
| 16:12.27 | HuntsvilleAL | so whats a good thought path to start at, existing phone system (NEC IPKII)(existing provider), exchange 2003 mail system |
| 16:12.37 | HuntsvilleAL | the goal is to do this all for free, if possible. |
| 16:12.43 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:12.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:12.59 | zerohalo | Grab a server, install *, and tweak until it's what you want. |
| 16:13.25 | adeel | HuntsvilleAL, the exchange server doesn't matter...as long as you can create a user on there, you can setup * to authetnticate, because it's just an email system |
| 16:13.57 | gitguy | [TK]D-Fender: i see this server has like 2 virtual address in the same interface, could that make problems to *? http://pastebin.com/m44ca2e55 |
| 16:14.01 | [TK]D-Fender | HuntsvilleAL: R&R <---- |
| 16:14.47 | [TK]D-Fender | gitguy: If you don't even see an incoming packet then it doesn't matter |
| 16:15.52 | gitguy | http://pastebin.com/m76360f5b |
| 16:16.19 | [TK]D-Fender | gitguy: my statement stands. |
| 16:16.20 | gitguy | i don't see anything when i try to connect with iax |
| 16:16.45 | [TK]D-Fender | gitguy: maybe the other side is screwed up, maybe the device is bad. take your pick. |
| 16:17.03 | gitguy | [TK]D-Fender: i'm using softphone |
| 16:17.07 | gitguy | zoiper |
| 16:17.08 | gitguy | for iax |
| 16:17.16 | [TK]D-Fender | gitguy: My answer is not changing... |
| 16:19.40 | gitguy | ok |
| 16:20.54 | *** join/#asterisk CCFL_Man2 (i=32f1a794@pool-70-105-204-227.scr.east.verizon.net) |
| 16:22.14 | adeel | gitguy, run tcpdump on the machine you're running the soft phone on |
| 16:22.28 | adeel | more than likely you're having network issues |
| 16:22.56 | gitguy | adeel: ok |
| 16:23.19 | adeel | i had a client call and complain about call quality sucking, and after a little investigation, turns out he's sharing his T1 with 2 other people and was averaging 1.2 mbit loads |
| 16:23.42 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 16:23.42 | grandpapadot | adeel: Looks like a job for QoS and g729a! |
| 16:23.42 | adeel | he refused to believe that was the problem, until i yanked out the other 2 guys and made some calls |
| 16:23.51 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 16:23.57 | adeel | grandpapadot, yeah, i put in QoS as well as g729 |
| 16:24.09 | adeel | while it's better, i can't fully resolve it as it spikes |
| 16:24.29 | grandpapadot | adeel: Looks like a job for a bandwidth controlling firewall? |
| 16:24.39 | adeel | grandpapadot, i'm using iptables with tc |
| 16:25.24 | adeel | grandpapadot, what becomes difficult is that the two other offices want public ip's with no firewalls (as they have their own) while the first client doesn't want them to be able to access his internal network |
| 16:25.33 | adeel | so right now i have a multi-vlan setup going on |
| 16:25.38 | grandpapadot | Put the firewall between your T1 and *everybody* else. |
| 16:25.52 | adeel | that's what i've done |
| 16:26.12 | grandpapadot | Call a network engineer? |
| 16:26.25 | adeel | i'm just thinking of hard limiting all the other traffic |
| 16:27.46 | gitguy | adeel: i don't see nothing bad on tcpdump |
| 16:28.15 | adeel | gitguy, now run tcpdump on both machines to see if the iax trafic is reaching |
| 16:28.24 | gitguy | ok |
| 16:30.25 | *** join/#asterisk NRich (n=NRich@72.37.252.50) |
| 16:32.12 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 16:32.51 | gitguy | adeel: i see my ip and all that on the server... i think is reaching, but i don't understand the output very well |
| 16:33.23 | gitguy | http://pastebin.com/m45be3c43 |
| 16:33.53 | gitguy | adsl-132-58.click.com.py <-- thats me |
| 16:33.53 | adeel | i can't recall, iax is udp correct? |
| 16:34.00 | Qwell | yes |
| 16:34.05 | adeel | what port? |
| 16:34.14 | jsmith-busy | gitguy: You're in Paraguay? |
| 16:34.24 | gitguy | yes |
| 16:34.32 | jsmith-busy | spent two years in Paraguay |
| 16:34.39 | zoid99 | bindport=4569 |
| 16:34.39 | gitguy | really, in asuncion? |
| 16:34.50 | zoid99 | look at iax.conf |
| 16:35.12 | gitguy | i have that |
| 16:35.12 | jsmith-busy | gitguy: Asuncion, Loma Pyta', Caaguazu... all over the place really |
| 16:35.19 | gitguy | jsmith-busy: nice |
| 16:35.26 | BCS-Satori | [TK]D-Fender: I went off your advice to use an SQL table for parsing a list of phone numbers for CID routing. What kind of asterisk code would I need to tell the system to pasre a sql statement followed by an accept/reject? |
| 16:36.02 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 16:36.13 | *** join/#asterisk _Hug[o]_ (n=pectic@unaffiliated/existenz) |
| 16:36.40 | adeel | gitguy, try running ' tcpdump udp port 4569' on both boxes and pastebin that please |
| 16:36.48 | gitguy | adeel: ok, thx |
| 16:36.54 | gitguy | jsmith-busy: where you from? |
| 16:37.01 | jsmith-busy | gitguy: Wyoming, USA |
| 16:37.09 | gitguy | jsmith-busy: nice, what did you come to do in py? |
| 16:37.18 | jsmith-busy | gitguy: I was a missionary for my church |
| 16:37.26 | gitguy | hehe nice |
| 16:37.43 | adeel | gitguy, and try connecting to iax and all during the output |
| 16:37.47 | gitguy | jsmith-busy: i live in asuncion |
| 16:38.00 | adeel | gitguy, you can also write the output to a file by using the -w foo flag |
| 16:38.05 | jsmith-busy | gitguy: Which part of asuncion? Just curious.... and how are things now that the elections are over |
| 16:38.27 | gitguy | jsmith-busy: do you know "club centenario" ? |
| 16:38.33 | gitguy | jsmith-busy: villa morra |
| 16:38.38 | jsmith-busy | Yes, I know it |
| 16:38.46 | gitguy | well in that zone |
| 16:38.47 | adeel | brb |
| 16:38.51 | jsmith-busy | gitguy: Very nice! |
| 16:38.56 | gitguy | :) |
| 16:39.03 | gitguy | it's a nice place |
| 16:39.22 | jsmith-busy | Sounds like it... |
| 16:39.55 | gitguy | jsmith-busy: lugo is cool, we are waiting for him to do good things... |
| 16:40.20 | gitguy | well i shouldnt call him cool yet |
| 16:40.29 | rupa | my asterisk is behind nat. In my sip.conf's [general] section, I have externip=myipaddress. However, all SIP traffic still has my 192.168.1.x address. am I missing something? |
| 16:40.33 | jsmith-busy | gitguy: Glad to hear it... I was there in '95 and '96, when things were still a little tense |
| 16:40.46 | gitguy | jsmith-busy: yeah things are starting to get better now |
| 16:40.54 | [TK]D-Fender | rupa: ..... |
| 16:40.56 | [TK]D-Fender | ~sipnat |
| 16:40.57 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:40.58 | [TK]D-Fender | ^^^^ |
| 16:41.12 | rupa | [TK]D-Fender, ok |
| 16:41.32 | *** join/#asterisk _Hug[o]_ (i=pectic@unaffiliated/existenz) |
| 16:43.03 | rupa | [TK]D-Fender, beautiful -- looks right now |
| 16:43.30 | [TK]D-Fender | rupa: You're welcome. |
| 16:43.34 | gitguy | adeel: http://pastebin.com/mfed72df |
| 16:43.41 | *** join/#asterisk CasseTrop (n=cast@64.254.255.99) |
| 16:43.49 | rpyne | <PROTECTED> |
| 16:44.04 | JayTee52 | anyone know if I need to use a straight through cable or a crossover cable when connecting the spans on my TE212P card to Adtran CSUs? |
| 16:44.06 | rupa | [TK]D-Fender, hrrm.. except problems with voicepulse. looking closer at sip debug. thanks |
| 16:44.33 | jsmith-busy | JayTee52: I'm not sure... but I'd guess it's a T1 cross-over (not an ethernet cross-over) |
| 16:44.46 | [TK]D-Fender | rupa: make sure their entry has "nat=no" |
| 16:44.57 | [TK]D-Fender | JayTee52: I second taht |
| 16:45.32 | JayTee52 | I know it has to be pinned for T1 and not ethernet, I'm just not sure if I need straight or x-over and I hate breaking the existing circuit to look at the cable ends. Tried Adtrans' site but they have scant info |
| 16:45.44 | [TK]D-Fender | JayTee52: Typically crossover |
| 16:46.55 | rpyne | <PROTECTED> |
| 16:47.07 | JayTee52 | that's what I thought but it's been soooooooo long (1998) since I had to cable up a T1 to a demarc and that was going from an Adtran CSU to a Kentrox T1 add/drop mux. |
| 16:47.23 | rupa | [TK]D-Fender, ah, yeah, setting it in [general] means I have to turn it off in the voicepulse templates. |
| 16:47.52 | [TK]D-Fender | rupa: Correct |
| 16:51.58 | *** join/#asterisk Trapa (n=no@207.230.238.94) |
| 16:52.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:53.17 | gitguy | adeel: any ideas? i tried to connect to register my softphone to another asterisk server (using IAX) and it registers |
| 16:53.58 | gitguy | the network on that server is screwed up i think |
| 16:54.01 | *** join/#asterisk nvrpunk (n=zomgobli@81.90.21.227) |
| 16:54.15 | nvrpunk | good evening :) |
| 16:54.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.41 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 17:06.12 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:06.31 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 17:08.12 | *** join/#asterisk Delvar (n=Delvar@77.240.56.22) |
| 17:08.17 | acxty | how can I record a dial. User call asterisk box, and asterisk Dial() to a channel, I want to record what is being say between them |
| 17:13.53 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 17:15.51 | Primer | How does one make asterisk NOT do "proxy authentication required" when a SIP call comes in whose CID matches a peer I have on my system? |
| 17:16.13 | *** part/#asterisk artz22 (n=artz@190.189.0.33) |
| 17:16.18 | jsmith-busy | Primer: As far as I know, there's no way to disable authentication based on CallerID |
| 17:16.51 | Primer | this seems like a major barrier to asterisk to asterisk calling over the internet using SIP |
| 17:17.04 | adeel | gitguy, it doesn't seem like your packets are getting there |
| 17:17.15 | Primer | un-registered calling, that is |
| 17:17.19 | adeel | gitguy, the timestamps aren't even anywhere near close |
| 17:18.49 | jsmith-busy | Primer: You can allow un-authenticated calling -- you just can't have it based on CallerID |
| 17:19.28 | Primer | that's what I'm trying to do |
| 17:20.14 | Primer | the extension of the CALLER, when that extension number matches a configured peer on my system, causes the call to fail |
| 17:21.04 | Primer | but if that callers extension does not exist, the call happily goes through |
| 17:21.22 | Primer | I fail to see why the caller's extension makes any difference |
| 17:22.35 | adeel | Primer, sounds like you have some dialplan problems |
| 17:22.43 | Primer | I don't see how |
| 17:22.52 | Primer | perhaps I haven't made it clear what's happening |
| 17:23.08 | adeel | when the caller extension matches a peer, the call fails right? |
| 17:23.09 | gitguy | adeel: yep, I tried with another server and IAX works perfectly... seems like it's giganet.com problems |
| 17:23.11 | NRich | where's a good tutorial for just the basics of asterisk's commands |
| 17:23.20 | gitguy | adeel: I also already tried from multiples endpoints (locations) |
| 17:23.30 | NRich | I don't want to master it, just know how t odo a few things for now |
| 17:23.31 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 17:23.32 | gitguy | different locations* |
| 17:23.46 | adeel | gitguy, yeah i'm pretty sure its a network related issue...try filing a bug? |
| 17:24.03 | Primer | a remote asterisk box that I have no control over wants to call my asterisk box. The caller on that remote box has extension 1234. It calls extension 6942 on my box. Thing is, I have a peer on my box with extension 1234. This call fails, asking for "proxy authentication" |
| 17:24.20 | gitguy | adeel: a bug to their network admins? |
| 17:24.25 | adeel | gitguy, yep |
| 17:24.31 | Primer | but if that same remote asterisk box calls from extension 2345, which is NOT a peer on my box, the call goes through |
| 17:24.42 | adeel | Primer, you might have overlapping contexts for your incoming & peers |
| 17:25.06 | talntid | anyone here have a Polycom IP301? |
| 17:25.17 | gitguy | adeel: I will, I hope they are helpful |
| 17:25.18 | Primer | adeel: not sure what you mean |
| 17:25.35 | adeel | Primer, can you pastebin the entire CLI output of a call with high verbosity? |
| 17:25.45 | Primer | I'd like to know what the extension (the 1234 part of 1234@remotesite.com) matters on my box |
| 17:25.54 | adeel | Primer, of both cases, 1 where the extension matches, and 1 where the extension doesn't? |
| 17:25.55 | Primer | s/what/why/ |
| 17:26.18 | Trapa | I'm having a problem with a outbound call, I have exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@pbx.somedomain.com,60) but whenever i try to make the call i get the peer saying "failed to authenticate" |
| 17:26.35 | Primer | adeel: that'll take me some time, but I'll get on it |
| 17:26.59 | Primer | Trapa: haha sounds like the same exact problem I'm having! |
| 17:27.37 | Trapa | Primer: Just was reading that and thinking it was too. |
| 17:27.48 | adeel | Trapa, they need to allow anonymous incoming calls |
| 17:28.05 | talntid | anyone here have a Polycom IP301? Can you confirm 100% that it has a dedicated headset jack? not handset, but headset |
| 17:28.19 | Trapa | adeel: But ... what if they don't want to allow anonymous incoming calls? And ... How do they do that? |
| 17:28.55 | Trapa | adeel: can i send authentication information that remains static with that dial command? |
| 17:29.08 | adeel | Trapa, sip.conf, set allowguest=yes should do the trick...and if they don't want to, then you need to bs setup as an extension on their box, and have your box register |
| 17:29.23 | adeel | Trapa, what you're trying to do is pretty much link 2 * boxes |
| 17:30.34 | Trapa | adeel: We are set up as a extension of their box i belive. But unsure since i don't have any control over the guy setting up their boxes. The guy and my boss think they know a lot mroe about this than me, but yet deem me the task of figuring it out. Heh .. anyway .. I think he has it set up right because this same config works for another remote office ... |
| 17:30.57 | Trapa | But that being said .. ours is supposedly the same as the remote office and it's not working so there must be somthing wrong |
| 17:31.03 | Primer | adeel: not sure if this is enough, but: http://sh.nu/p/24359 |
| 17:31.12 | Primer | that's the output on my box from sip debug ip remotebox |
| 17:31.35 | *** join/#asterisk l2cache (n=chatzill@117.178.101.97.cfl.res.rr.com) |
| 17:32.03 | Primer | now I'll remove the 3205 peer from sip.conf and reload |
| 17:32.42 | adeel | Trapa, pastebin the CLI output of a call...that's the easiest way to debug |
| 17:33.03 | adeel | Primer, i'm looking for Asterisk's CLI output, not just the sip messages |
| 17:33.10 | Primer | http://sh.nu/p/24360 |
| 17:33.23 | Primer | is without the 3205 peer |
| 17:33.40 | adeel | Primer, can you repaste without the sip debug? |
| 17:33.43 | Primer | can you specify the CLI command line I should use? core verbose 10? |
| 17:33.50 | Primer | that is the entire CLI output |
| 17:34.00 | Trapa | adee: http://pastebin.com/d73349230 |
| 17:34.11 | Trapa | Adeel: Sorry thats the sip.conf :P |
| 17:34.49 | adeel | Trapa, having the type as peer requires the extension to register...try setting it to friend |
| 17:35.03 | Primer | core set verbose 10? |
| 17:35.14 | adeel | Primer, sure, but disable the sip debug |
| 17:35.25 | Primer | that shows nothing |
| 17:35.28 | Primer | absolutely nothing |
| 17:35.47 | Trapa | adeel: http://pastebin.com/d6440d693 |
| 17:36.17 | Trapa | adeel: whats the cli to reload the sip.conf ? |
| 17:36.24 | adeel | sip reload |
| 17:36.38 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 17:36.57 | Trapa | Same result with type=friend |
| 17:36.59 | *** part/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 17:37.36 | Trapa | and my register command is register => user:pass@pbx.somedomain.com |
| 17:37.36 | Primer | yeah, that seems to make no difference, friend or peer |
| 17:37.42 | Primer | as I've tried both |
| 17:38.06 | adeel | so i wonder if it's actually registering |
| 17:38.26 | Trapa | adeel: Ship show registery shows that it's registered |
| 17:38.53 | adeel | Primer, in the second paste is....Found no matching peer or user for '216.239.128.31:5060' on line 42 |
| 17:39.57 | adeel | Primer, and on the first paste, line 60... Found user '3205' |
| 17:40.31 | adeel | pretty much what seems to be happening is that * finds the peer and thinks it's local, so it tries to get it to authenticate |
| 17:41.35 | acxty | hey guys, how can I record a channel call? |
| 17:42.08 | Primer | adeel: why is there no "realm" ever considered? |
| 17:42.12 | Primer | extension@realm? |
| 17:42.26 | Primer | why does it only consider the extension part of the incoming call? |
| 17:42.45 | Primer | and shouldn't peer calls come in via registrations? |
| 17:43.02 | Primer | or does SIP simply not differentiate between the 2? |
| 17:43.09 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
| 17:44.04 | adeel | Primer, unless you specify a realm explicitly, i believe, asterisk doesn't know look at it |
| 17:44.08 | acxty | I want to record what its say on an specific extension |
| 17:44.08 | acxty | sorry dial(channel) |
| 17:44.18 | AlienPenguin | hi, i read on voip-info.org that the $HANGUPCAUSE is defined even for SIP channels but i still get 0 on every case i tried |
| 17:44.24 | AlienPenguin | (using asterisk 1.4.19) |
| 17:44.37 | *** join/#asterisk Ubluzok (n=ubluzok@79.173.65.61) |
| 17:44.43 | Primer | adeel: I'm guessing this is the problem then. I'm just not sure how this will affect exiting, connected peers |
| 17:44.56 | adeel | Primer, but your realm is defined as this....Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="686c9158" |
| 17:45.12 | Primer | lovely |
| 17:45.19 | Primer | I'm guessing that's the same on the remote box too |
| 17:45.24 | acxty | can that be done? |
| 17:45.26 | adeel | Primer, its the default |
| 17:45.29 | Primer | presumably the default |
| 17:45.31 | Primer | heh |
| 17:45.47 | adeel | Primer, so i'm assuming this is what's happening.... |
| 17:46.39 | adeel | Primer, asterisk gets the call, see's an extension that matches the peer, asks for authentication because the SIP realms the same (Default) and fails because the expected hash is different |
| 17:46.55 | Primer | that makes sense |
| 17:47.11 | adeel | which might be the same reason why trapa is having problems....but i'm not entirely sure |
| 17:47.21 | adeel | this is a little beyond my knowledge of asterisk |
| 17:47.22 | acxty | may someone help me with that? I want to run a dial() to a cellphone and record that channel |
| 17:47.35 | Primer | except I've now added the realm, the SIP packet refelcts this, and the problem remains |
| 17:47.36 | adeel | acxty, you should be able to do that |
| 17:47.59 | Primer | reflects, even |
| 17:48.11 | Primer | Proxy-Authenticate: Digest algorithm=MD5, realm="sh.nu", nonce="1bcae7a8" |
| 17:48.14 | acxty | adeel, what should I do record then dial? |
| 17:48.16 | adeel | Primer, what are your contexts |
| 17:48.51 | acxty | I make the test using an extension, but with a dial() don't get how |
| 17:48.56 | *** join/#asterisk tinman_gz (n=tinman_g@66.193.134.74) |
| 17:49.31 | tinman_gz | is it possible to fork within a dial plan? (probably the wrong term) |
| 17:49.50 | tinman_gz | i want inbound callers to be given a ton of choices so they feel they are doing something |
| 17:49.53 | Trapa | adeel: What if we got rid of our default contexts? |
| 17:50.12 | [TK]D-Fender | tinman_gz: Yeah, its called an IVR. |
| 17:50.19 | tinman_gz | at the same time i want an outbound call to call me and give me the option of answering the call when the inbound caller is done with the maze |
| 17:50.21 | adeel | acxty, check out the RECORD command |
| 17:50.22 | [TK]D-Fender | tinman_gz: basic stuff detailed in THE BOOK. |
| 17:50.24 | [TK]D-Fender | ~book |
| 17:50.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:50.24 | adeel | or Monitor |
| 17:50.37 | adeel | Trapa, what do you mean? |
| 17:51.09 | tinman_gz | what i mean by fork is have 2 different actions happening at the same time |
| 17:51.23 | Primer | adeel: not sure what you mean |
| 17:51.31 | adeel | Primer, -- Executing [12345@default:1] Macro("SIP/216.239.128.31-b7926238", "dialsip|1006") in new stack ....on the call that completed...12345@default....default is a fairly common context |
| 17:51.31 | Trapa | well I have the context named default, if the calls are being thought local because they have the same context names, maybe you have to have different context names |
| 17:51.33 | tinman_gz | i didnt type fast enough :) |
| 17:51.52 | adeel | Primer, typically, you have incoming and outgoing using different contexts in your dialplan |
| 17:52.05 | Primer | adeel: sure...I've setup all my known extensions in that context |
| 17:52.09 | [TK]D-Fender | tinman_gz: like? |
| 17:52.25 | adeel | Trapa, that's exactly what i'm saying |
| 17:52.38 | gitguy | adeel: I already wrote them, I hope they will give me some helpful answer, thanks for your help :) |
| 17:52.55 | Primer | adeel: all my sip peers have their own contexts based on trust, but each of those contexts include [default] |
| 17:53.14 | Primer | but in [general] in sip.conf, the context=default |
| 17:53.23 | adeel | Primer, i think you need to have more unique context names, it could very well be possible that the incoming call is being routed to a local context |
| 17:53.41 | adeel | Primer, i think that's your problem right there |
| 17:53.47 | Primer | err, how else do I get those calls to my extensions? |
| 17:54.02 | Primer | surely you don't expect me to duplicate all the extensions in a different context? |
| 17:54.23 | adeel | no, incoming calls should be handled by an incoming context which does some logic |
| 17:54.59 | tinman_gz | i dont think i am explaining it right... inbound call fires 2 processes 1 is for the inbound caller (maze of ivr prompts) and 2 is it calls me and festivals thier caller id if i hit "3" or something then it connects the 2 calls |
| 17:55.11 | adeel | i think you need to re-read the book...the section for incoming calls |
| 17:55.20 | Primer | just for kicks, I setup a new context |
| 17:55.43 | Primer | and made this the default context in [general] in sip.conf. Same problem happens |
| 17:55.53 | Primer | context=test |
| 17:55.57 | Primer | [test] |
| 17:56.01 | Primer | etc... |
| 17:56.24 | adeel | Primer, how are your calls coming in? |
| 17:56.31 | adeel | Primer, though a SIP provider? |
| 17:56.43 | adeel | or direct from another box? |
| 17:56.44 | Primer | polycom phone -> asterisk -> internet -> asterisk |
| 17:57.33 | adeel | so the problem is the incoming call is being matched against local extensions because of an improper dialplan setup |
| 17:57.49 | Primer | no, it's being matched against an existing peer in sip.conf |
| 17:58.06 | Primer | polycom phone (whose extension is 3205) -> asterisk #1 -> internet -> asterisk #2 |
| 17:58.10 | adeel | because your context is throwing it there |
| 17:58.18 | Primer | if I remove the 3205 peer on asterisk #2, the problem goes away |
| 17:59.08 | Primer | so the mere fact that this 3205 peer uses that context makes it want to be a peer |
| 17:59.13 | tinman_gz | if i dial in the incomming context the calls are instantly connected... |
| 17:59.42 | Primer | ok, let me change the context of my 3205 peer on asterisk #2 to test this |
| 17:59.42 | adeel | Primer, scroll down to example 4 on this link......http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 18:00.24 | adeel | Primer, you should have 1 'master' context for outbound, and 1 'master' context for incoming....not just one global master context |
| 18:00.32 | Primer | same thing happens |
| 18:01.11 | [TK]D-Fender | tinman_gz: Again jsut sounds liek a basic IVR... nothing is simultaneous... |
| 18:02.12 | tinman_gz | [TK]D-Fender the incomming call and the call to me should both be happening at the same time... if i hit a certain key then the calls are connected or i can ignore the call and have it go to vm |
| 18:02.45 | [TK]D-Fender | tinman_gz: how are YOU involved with this "incoming call"? |
| 18:02.57 | [TK]D-Fender | tinman_gz: when a call comes in, you are doing nothing. |
| 18:03.15 | adeel | Primer, http://www.siliconvp.us/modules.php?name=News&file=article&sid=6 |
| 18:03.24 | adeel | Primer, you're basically trying to link 2 asterisk boxes |
| 18:04.09 | *** join/#asterisk RoyK (n=roy@ip-86-12-149-91.dialup.ice.no) |
| 18:04.12 | Primer | except I don't want to register |
| 18:04.21 | tinman_gz | [TK]D-Fender i suck at explaining things... call comes in its is answered automatically by asterisk meanwhile asterisk calls my line... gives me the callerID via festival and gives me the option of answering the call or sending it to vm (or another extension) |
| 18:04.31 | Primer | I want to be able to receive arbitrary calls over the internet to my asterisk using SIP |
| 18:04.47 | Primer | how the hell is emun supposed to work if everything requires registration? |
| 18:05.27 | adeel | the problem lies in your diaplan |
| 18:05.40 | Primer | http://sh.nu/p/24361 |
| 18:05.40 | adeel | if you don't want to believe me, that's fine |
| 18:05.55 | Primer | well, my tests show that what you're saying is wrong |
| 18:06.02 | rupa | tinman_gz, the second part has to be done in a Macro of the Dial() -- but beware, if you hang up in the macro prior to having hte macro return then you'll also hang up on the caller |
| 18:06.13 | rupa | M() option to Dial |
| 18:06.21 | [TK]D-Fender | ^^^ |
| 18:07.40 | jeev | hi Fender.. |
| 18:07.40 | tinman_gz | fart,..,. |
| 18:07.40 | tinman_gz | ok |
| 18:07.40 | jeev | my stuff is on a stand still.. havne't ordered the additional phones yet |
| 18:07.52 | kingsob | any ideas whats wrong here?? http://rafb.net/p/y1RYXQ43.html |
| 18:07.55 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:08.02 | rupa | tinman_gz, I have a hack to 1.4 which add a new option to Dial() that doesn't hangup on the caller but it is not very much tested... |
| 18:08.11 | tinman_gz | its like i need the incomming call to spawn a thread or another process.... completely unrelated the the primary thread other than somehow they can be connected later |
| 18:08.35 | rupa | tinman_gz, then create a .call file with the right options? |
| 18:08.52 | [TK]D-Fender | tinman_gz: thats what the "M()" option ont he Dial that calls you is for. |
| 18:09.32 | [TK]D-Fender | tinman_gz: To let you go through a little dialplan to see if you want to accept the call or not. The other side (while you're processing) can't do anything, they're just sitting waiting for you to conclude what you're doing. |
| 18:10.17 | rupa | so if you want to be totally async, create a call file with the right options and then dump the caller into the IVR. If you decide to talk to the caller, umm.. what, bridge you in? |
| 18:10.19 | tinman_gz | i think rupa might be on the right track with the .call file... |
| 18:10.51 | tinman_gz | rupa sounds like it |
| 18:11.12 | *** part/#asterisk korihor (n=humberto@190.39.161.145) |
| 18:11.20 | tinman_gz | .call+bridge... i will try some stuff thx |
| 18:13.12 | unpaidbill | have any of you guys used linksys poe switches? im looking for some cheap 8/16/24 porters |
| 18:15.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:15.58 | jsmith-busy | unpaidbill: I have... they're OK. |
| 18:16.26 | unpaidbill | yeah they at least do vlan and qos |
| 18:16.30 | unpaidbill | and are cheap... |
| 18:16.47 | unpaidbill | i wish cisco wasnt so expensive |
| 18:17.57 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 18:18.19 | tinman_gz | rupa when you spoke of bridge did you mean a meetme? |
| 18:18.31 | jsmith-busy | unpaidbill: Just make sure it'll power the total number of phones you'd like to power |
| 18:18.42 | rupa | tinman_gz, not sure, maybe. didn't think that far. was speaking of bridge in a generic sense |
| 18:19.02 | tinman_gz | ic i ws just word searching the oriely book |
| 18:19.52 | unpaidbill | jsmith haha yeah, that would be embarrasing :) |
| 18:20.51 | Primer | adeel: I appreciate your help, btw |
| 18:21.01 | *** join/#asterisk aricon (i=nick@bordergatewayprotocol.net) |
| 18:21.05 | aricon | oi |
| 18:21.34 | Primer | adeel: after more testing, I've found something interesting |
| 18:22.16 | aricon | Can anyone tell me if there are any call analysis comparisons for digital circuits (robbed bit and PRI t-1s) on cards that work with asterisk? |
| 18:22.34 | aricon | digium vs. dialogic Diva series and such |
| 18:25.19 | Primer | if I change this: callerid="Daniel Ceregatti" <3205> to this: callerid="Daniel Ceregatii" <32055>, on the asterisk machine making the call, I can get through |
| 18:25.31 | *** join/#asterisk ZBRA (n=ZB@189.100.120.132) |
| 18:25.50 | HuntsvilleAL | ./window close 7 |
| 18:26.06 | adeel | Primer, it's matching against the CID number...the xtension it gets |
| 18:26.30 | adeel | Primer, i'm about 90% sure it's a dial plan issue.... |
| 18:27.16 | Primer | adeel: ok, I'm still hacking at it. Thanks again for your help and interest in this, I really appreciate it |
| 18:27.39 | *** part/#asterisk aricon (i=nick@bordergatewayprotocol.net) |
| 18:27.40 | adeel | Primer, no problem....take a look at the book again, it should provide some insight |
| 18:27.51 | ZBRA | hi adeel, Primer |
| 18:28.06 | HuntsvilleAL | heh, theonion.com |
| 18:28.09 | HuntsvilleAL | (Breaking News)(AP/Reuters) - Iran has just launched a nuclear assault on Tel Aviv Israel, causing an estimated 5.9 million deaths Reuters has learned. Details Soon. |
| 18:29.53 | [TK]D-Fender | HuntsvilleAL: Normally I like theonion... but sorry, that is NOT ^@$ING FUNNY. |
| 18:30.21 | ZBRA | its really not funny |
| 18:31.50 | adeel | i really think asterisk needs an overhaul on it's logging stuff.... |
| 18:32.24 | HuntsvilleAL | can asterisk send logs to a Kiwi daemon, with local0,local1,local2,local3, etc......? |
| 18:32.34 | adeel | it can send logs to syslog |
| 18:32.38 | HuntsvilleAL | well thats what i meant. |
| 18:33.05 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 18:33.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:33.45 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 18:33.45 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:33.46 | adeel | i personally like having all the logs in separate files, and one master log with everything dumped |
| 18:33.50 | HuntsvilleAL | I like kiwi, but its on one of the windows 2003 servers. Is there a linux equivalent? |
| 18:34.01 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 18:34.04 | HuntsvilleAL | Our pix 515 sends about 100megs a day of syslog to it |
| 18:34.04 | adeel | syslog? |
| 18:34.08 | HuntsvilleAL | yep |
| 18:34.11 | adeel | syslog-ng |
| 18:34.14 | adeel | is a good one |
| 18:34.15 | *** join/#asterisk klimonso (n=eddy@91.73.203.98) |
| 18:34.17 | keith4 | this is like abbot and costello |
| 18:34.18 | adeel | there are a few dozen syslogs |
| 18:34.22 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-b3ff99508181f36d) |
| 18:34.32 | adeel | don't know if there's a kiwi port though |
| 18:34.56 | ZBRA | can anyone help me out, i keep getting this msg on my asterisk: [May 2 15:40:30] WARNING[24319]: chan_sip.c:1784 __sip_xmit: sip_xmit of 0xb72e4484 (len 410) to 10.0.2.24:5060 returned -1: No buffer space available |
| 18:34.58 | HuntsvilleAL | adeel, kiwi is pretty nifty w/ the gui, you can in real time parse out the incoming logs |
| 18:35.05 | HuntsvilleAL | by switching the local 0,1,2,3,4 etc |
| 18:35.13 | HuntsvilleAL | back/forth in real time |
| 18:35.33 | ZBRA | i get it from all my ipphones and its an intermitent problem |
| 18:35.40 | adeel | HuntsvilleAL, yeah, you can realtime parse logs in linux too... |
| 18:35.51 | rcahilig | Hi, how do I use this in a macro dialplan Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)}_${EXTEN}) |
| 18:35.58 | adeel | HuntsvilleAL, just depends on your setup...typically, linux servers lack GUI's |
| 18:36.30 | Trapa | In several places around the internet, i see that it is somtimes a good thing to transmit silence ... |
| 18:36.54 | Trapa | Even in the config file it syas "Transmit Silence"=YES under the x-lite configuration . |
| 18:37.04 | HuntsvilleAL | Gerdazech |
| 18:37.24 | adeel | Trapa, not every one supports VAD |
| 18:37.27 | kaldemar | rcahilig: replace EXTEN with MACRO_EXTEN |
| 18:37.37 | Trapa | But every entry in the sip.conf file is one word. And i'm wondering if it's transmitsilence=yes or trasmit silence=yes |
| 18:38.18 | rcahilig | kaldimar: I have problem with this variable: ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} |
| 18:38.47 | *** join/#asterisk Confusd (n=Confusd@87-194-204-58.bethere.co.uk) |
| 18:38.52 | Confusd | hi |
| 18:39.08 | Trapa | adeel: VAD? |
| 18:39.22 | Confusd | i'm modifying chan_mobile and need some help |
| 18:39.44 | Confusd | i've got inband progress working, but how can i pass back a q931 cause code? |
| 18:39.48 | *** join/#asterisk Trapa (n=no@207.230.238.94) |
| 18:40.17 | Confusd | * i have a method to get the cause code from the mobile end, but how to pass it back to asterisk? is there a method/structure already for that? |
| 18:40.20 | Confusd | i guess there must be! |
| 18:40.44 | adeel | Trapa, variable audio detection? or something like that..it's basically not transmitting silience |
| 18:41.25 | Trapa | adeel: Yeah .. basicaially i've set up a queue, when i callt he queue i hear the music, then i hear the announcement, and then more music... |
| 18:41.36 | Confusd | vad is voice activity detection.. used to detect when someone has picked up, when you dont have a reliable mechanism to know connect |
| 18:41.48 | Trapa | adeel: When the remote party calls.. they hear silence,s then the announcement then the music .. so i thought it was just too low on teh threashold |
| 18:41.49 | rcahilig | kaldimar: It gives me this filename 20080503-023353_8647101_s.wav |
| 18:41.55 | Confusd | baground 'noise' is called 'comfort noise' or 'comfort noise injection' |
| 18:42.52 | lmadsen | hrmmm |
| 18:42.53 | lmadsen | ;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]] |
| 18:43.03 | lmadsen | how do you specify more than 1 AppArgument? |
| 18:43.31 | kaldemar | rcahilig: what is wrong with that one? |
| 18:43.51 | adeel | Trapa, it could be...or it could be something entirely different |
| 18:43.55 | jsmith-busy | lmadsen: Pipes? |
| 18:44.18 | lmadsen | 1.6.0 -- pipes are not a valid separator anymore |
| 18:44.21 | rcahilig | kaldemar: there is an "s" in hte end |
| 18:44.35 | lmadsen | unless I'm mistaken |
| 18:44.38 | Trapa | adeel: Yeah .. but unless we have an idea of what entirely differnt thing it could be .. my troubleshooting says ..lets try the things we think might be suspect. |
| 18:44.53 | kaldemar | rcahilig: as i first said, replace EXTEN with MACRO_EXTEN and you'll get the number. |
| 18:45.30 | adeel | Trapa check the CLI's with and without the sip debug, as well as rtp debug |
| 18:45.37 | adeel | bbiab, lunch |
| 18:45.44 | [hC] | Hey, if i have a "guest" sip user calling in to my system, but his From: (say 100@somehost) matches a user in my sip.conf, (100) asterisk is trying to authenticate the guest as 100. Whats the way around that? |
| 18:46.02 | rcahilig | kaldemar: thanks |
| 18:46.51 | rcahilig | kaldemar: Is it ok to use this in a macro? exten => s,n,MixMonitor(${DIR}/${FILENAME}.wav|W(2)|b) |
| 18:47.07 | *** join/#asterisk LinuxMafia (n=awatt@CPE001346a4c4cb-CM00159a642d7e.cpe.net.cable.rogers.com) |
| 18:47.13 | LinuxMafia | hello all |
| 18:47.26 | ZBRA | kaldemar do you know what this error means: chan_sip.c:1784 __sip_xmit: sip_xmit of 0xb72e4484 (len 410) to 10.0.2.24:5060 returned -1: No buffer space available ?? |
| 18:47.31 | ZBRA | hello LinuxMafia |
| 18:47.36 | LinuxMafia | i am totally noob , and i have some noob questions |
| 18:47.42 | LinuxMafia | ZBRA, hi there |
| 18:48.19 | ZBRA | maybe i can help, i was a noob, i think i am still kind of |
| 18:48.31 | LinuxMafia | ZBRA, i need to install asterisk for a company with several phones |
| 18:48.54 | LinuxMafia | ZBRA, people that work there don't know much computer |
| 18:48.56 | ZBRA | are them ip phones or analog phones ? |
| 18:49.05 | LinuxMafia | ZBRA, analog phones |
| 18:49.25 | ZBRA | you have already an analog pabx runing ? |
| 18:49.32 | LinuxMafia | no |
| 18:49.47 | LinuxMafia | i dont have any thing yet |
| 18:49.54 | ZBRA | do they want do change it all to ip phones ? |
| 18:49.58 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:50.16 | ZBRA | e ai caio |
| 18:50.19 | LinuxMafia | ZBRA, all i know they pay too much money to some company |
| 18:50.29 | Trapa | So these are all just single-line phones that are at their office? |
| 18:50.42 | LinuxMafia | i think so |
| 18:51.01 | LinuxMafia | yeah single-line phones |
| 18:51.04 | *** join/#asterisk RypPn_OuT (n=Sally@rosscom.demon.co.uk) |
| 18:51.05 | Trapa | Can they transfer a call from one desk to another? |
| 18:51.26 | ZBRA | can anyone help me out, i keep getting this msg on my asterisk: [May 2 15:40:30] WARNING[24319]: chan_sip.c:1784 __sip_xmit: sip_xmit of 0xb72e4484 (len 410) to 10.0.2.24:5060 returned -1: No buffer space available |
| 18:51.41 | LinuxMafia | Trapa, i am not sure , but they can transfer to other units |
| 18:51.54 | LinuxMafia | there are some extensions |
| 18:51.54 | ZBRA | i get this errors from all my phones at differnt moments, i use linksys spa ip phones |
| 18:52.20 | ZBRA | so i think they have an analog pabx LinuxMafia |
| 18:52.37 | Trapa | Then they are not single line phones, 99.9% likely analog or digital propritary pbx, Do you know the make/model of the phones? |
| 18:52.53 | ZBRA | do they spend a lot for international calls or calls to another prefix area ? |
| 18:52.59 | LinuxMafia | Trapa, no i dont , does it matter? |
| 18:53.19 | LinuxMafia | ZBRA, many people call them for their dogsd |
| 18:53.31 | tinman_gz | can you still get a call waiting beep while you are in a meetme conference? |
| 18:53.38 | LinuxMafia | actually it is humane society - shelter for animals |
| 18:53.44 | wichitalineman | He's just trying to get a feel for your situation, how it is configured, etc |
| 18:53.49 | [TK]D-Fender | Trapa: What do you define as "single line phones"? |
| 18:54.00 | ZBRA | dogsd ? whats that ? |
| 18:54.05 | Trapa | Linuxmafia: Only so much as, you want to make sure whatever you provide with them has the same feature set. Also depending on their phones they may be able to be used with asterisk, tho very unlikely by the sounds of it |
| 18:54.10 | ZBRA | oh |
| 18:54.17 | ZBRA | you meant dogs |
| 18:54.19 | [TK]D-Fender | tinman_gz: If your phone supports call waiting as a concepts and your phone is sent another call, sure. |
| 18:54.33 | tinman_gz | cool thx |
| 18:54.51 | Trapa | fender: non-propriatary 500set or somthing that would work with asterisk zaptel hardware |
| 18:55.07 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:55.14 | LinuxMafia | Trapa, i think they rather to change the phones if it is not compatible , cause i know they pay alot for these phone services |
| 18:55.29 | Trapa | fender: Ie grandma's home phone. |
| 18:56.51 | LinuxMafia | Trapa, let's say you installed asterix , for using it does a normal person can use it ? |
| 18:57.08 | Trapa | linuxmafia: Assuming this is a company that knows how to use their phones, you could probably make a asterisk phone system work for them. But i'd be willing to bet that they would want a to have a hardphone .. and for that you'd probably be spending 100 ish or more per phone to replace their phones with voip stations. |
| 18:57.27 | Trapa | linux : Yeah asterisk can be made to be teh same as your house phone, or be made to work the same as your office phone system |
| 18:58.02 | Trapa | linux: it's just important to know which your customers are used to dealing with because they won't take well to a change in their daily routine most likely, but the way they use it seriously affects the cost of the phones and hardware |
| 18:58.27 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 18:58.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 18:59.31 | LinuxMafia | Trapa, i did not get it sorry .do they need speciall phones for asterix ? |
| 19:00.41 | LinuxMafia | Trapa, i am sure they have regular phones |
| 19:01.11 | ZBRA | Trapa, do you know what this error means: chan_sip.c:1784 __sip_xmit: sip_xmit of 0xb72e4484 (len 410) to 10.0.2.24:5060 returned -1: No buffer space available ?? |
| 19:01.35 | Trapa | Linux: If they have phones like your granma has, or like mickeymouse phones then they can probably still use them. |
| 19:02.00 | LinuxMafia | ZBRA, using linux? |
| 19:02.03 | Trapa | ZBRA: No i don't, fender might, I think it might have to do with your system resources, check to make sure you have enough ram and cpu leftover? |
| 19:02.17 | ZBRA | yes, im using Debian |
| 19:02.22 | LinuxMafia | Trapa, so you can use any phone with astrix? |
| 19:02.54 | rupa | LinuxMafia, define "any phone" |
| 19:02.57 | ZBRA | Trapa: maybe, gotta check it |
| 19:02.58 | LinuxMafia | ZBRA, install htop , and see if you have enough ram |
| 19:03.15 | LinuxMafia | rupa, i mean those ordinary phones |
| 19:03.20 | Trapa | linux: No. You can use prettymuch any VOIP phone, and your standard telephone you buy at radioshak. But you can't for example use panasonic, or nortel, or those kinds of phones |
| 19:03.23 | ZBRA | good idea LinuxMafia |
| 19:03.48 | rupa | if you can plug the ordinary phone in to the wall's regular phone jack at home and it works, ys, it'll work fine with asterisk. If it is a PBX phone, then it "depends" |
| 19:03.59 | LinuxMafia | Trapa, oh so it depens on the phones type |
| 19:04.23 | LinuxMafia | what is PBX? |
| 19:04.33 | Trapa | linux: yeah, especially if they want to keep the phones they have. PBX: Private branch exchange |
| 19:04.34 | rupa | . o O ( google ) |
| 19:04.51 | bkruse | ~pbx |
| 19:04.52 | jbot | somebody said pbx was a Private Branch eXchange |
| 19:05.11 | LinuxMafia | let me look at the pics |
| 19:05.29 | Trapa | linux: fwd me links if you can |
| 19:05.50 | ZBRA | Trapa, there is no fender here |
| 19:06.03 | Trapa | ZBRA: Just noticed that he musta left. |
| 19:06.14 | ZBRA | ok |
| 19:06.18 | LinuxMafia | oh we have one look like this at home -->http://www.comtechphones.com/Images/nortel/m3904-pbx-phone.jpg |
| 19:06.24 | bkruse | nortel? eww |
| 19:06.47 | bkruse | We HAD one of those systems at my school. Luckily the school has a real pbx now :] |
| 19:07.01 | LinuxMafia | Trapa, so if the phone looks like this -->http://www.comtechphones.com/Images/nortel/m3904-pbx-phone.jpg it depends? |
| 19:07.49 | *** part/#asterisk rpyne (n=richard@69.77.169.14) |
| 19:08.29 | LinuxMafia | ZBRA, i usually run htop on yakuake terminal , whenever i need to check the memory or kill some process it comes handy |
| 19:08.31 | ZBRA | i just got the same No buffer space available and i am with system monitor opened, im not out of resources |
| 19:08.54 | LinuxMafia | ZBRA, you have htop now? |
| 19:08.59 | ZBRA | im using 14% of the CPU and 41% of the phisical memory |
| 19:09.38 | Trapa | linux: I would say, that unless their phones look very simple, ... like this simple http://www.ridgewayinc.com/images/2510D.jpg .. then there is some question as to whether they will work |
| 19:09.57 | Trapa | linux: if it has a TRANSFER button, or a VOICEMAIL button, then it's highly unlikely they will work. |
| 19:10.13 | ZBRA | LinuxMafia, no im using the GNOME System Monitor |
| 19:10.23 | LinuxMafia | ZBRA, use htop |
| 19:10.25 | Trapa | ZBRA: Hrmm so memory may not be the issue .. what about network .. Have you tried googling your error message? |
| 19:10.34 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:10.34 | ZBRA | but i will check out this Htop you said |
| 19:10.35 | [TK]D-Fender | ZBRA: it is highly reocmmended that you NOT run X on your server. |
| 19:10.48 | [TK]D-Fender | ZBRA: For exactly these reasons |
| 19:10.50 | LinuxMafia | Trapa, oh thanks alot i got it now |
| 19:10.55 | ZBRA | humm |
| 19:11.04 | ZBRA | im new on linux |
| 19:11.25 | Trapa | ZBRA: Is this machine only going to be doing asterisk? |
| 19:11.42 | ZBRA | no, it is a router also |
| 19:11.56 | ZBRA | i can manage to run it all on command lines |
| 19:12.23 | ZBRA | i will try to boot the box without the X |
| 19:12.46 | ZBRA | and i will see if i will continue getting those messages or not |
| 19:13.06 | LinuxMafia | any photos off asterisk set-up? |
| 19:13.08 | gitguy | damn giganet.com guys takes ages to answer their damn support |
| 19:13.16 | Trapa | fender: Do you know how i can change teh threashold for the cutoff for hold music .. So that it transmits quieter things |
| 19:13.38 | LinuxMafia | <PROTECTED> |
| 19:14.06 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:15.00 | ZBRA | i will do so |
| 19:15.23 | ZBRA | BRB gotta restart the box, its the same router i am using to access internet |
| 19:15.31 | gitguy | gigenet* |
| 19:16.17 | *** join/#asterisk dlynes (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
| 19:17.01 | dlynes | Are there any additional packages I need besides the unix odbc driver for mysql, unixodbc itself, and unixodbc-dev packages on Centos to get cdr_odbc working on asterisk 1.4? |
| 19:17.17 | dlynes | I'm having trouble getting make menuselect to recognize support for it on this box |
| 19:18.14 | adeel | LinuxMafia, screenshots of asterisks setup? most would just be cli outputs |
| 19:18.25 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 19:18.36 | keith4 | can I put sip phones in a call group? |
| 19:18.45 | adeel | yes |
| 19:18.54 | JayTee52 | [TK]D-Fender, that prirestart=never parameter you mentioned earlier, is that specific to each span? is there a particular position in the file it needs to go? |
| 19:18.55 | LinuxMafia | i found that http://img214.imageshack.us/img214/2278/asteriskhomesimplesetupdx8.jpg |
| 19:19.08 | jsmith-busy | dlynes: Yes... you need one more... let me find the name |
| 19:19.20 | dlynes | jsmith-busy: thanks |
| 19:19.40 | adeel | LinuxMafia, that's not really a screen shot...more like a topological diagram |
| 19:20.26 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:20.43 | LinuxMafia | adeel, so basically i need to connect the phone line to server and from server have to have a line to swiches? |
| 19:21.25 | adeel | LinuxMafia, yeah, pretty much....however, it depends on how you're setting up...if you use VoIP, then you just need a normal ethernet network |
| 19:21.40 | adeel | LinuxMafia, not taking into account how you're getting internet service |
| 19:22.58 | LinuxMafia | adeel, let say some one called , and you want to make him wait ? there should be some other computers for that right? |
| 19:23.20 | adeel | LinuxMafia, no, asterisk will handle all the call logic |
| 19:24.13 | LinuxMafia | adeel, but the thing i dont get it , is how ? |
| 19:24.30 | LinuxMafia | adeel, there is no need for any one to pick the phone? |
| 19:25.12 | adeel | LinuxMafia, depends on what you want to do...if you want people to answer a phone, it can be accomodated |
| 19:25.45 | LinuxMafia | adeel, let 's say you want to auto remind an appointment , then some one has to enter a date right? |
| 19:26.00 | adeel | LinuxMafia, yes, you need to program asterisk to do stuff |
| 19:26.06 | [TK]D-Fender | JayTee52: I believe its channel specific and goes back to the span to see which d-Chan to trigger it on. |
| 19:26.42 | Trapa | is there a asterisk command that returns the number of agents in queue so i can make it do somthing different if no agents are answering calls from the queue? |
| 19:26.54 | LinuxMafia | adeel, does it has client ? like some one that can tell when to call , or wait right now , put on hold ... |
| 19:27.17 | adeel | ~book |
| 19:27.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:27.32 | adeel | LinuxMafia, start there...it'll explain everything |
| 19:27.45 | JayTee52 | [TK]D-Fender, are you running PRI's with any Digium cards where you've disabled restart? I'm just wondering if I need to set it or can just let the channels reset. I'm thinking if the reset is by design then it wouldn't reset a channel that's currently in use but I hate to assume anything. |
| 19:28.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:28.43 | [TK]D-Fender | JayTee52: last I heard it shouldn't disrupt anything. If you have reason to believe it does, then feel free to disable it. |
| 19:29.01 | *** join/#asterisk ZBRA (n=ZB@189.100.120.132) |
| 19:29.16 | LinuxMafia | adeel, i have that book , can you tell me which chapter expalins these ? you know it is hard to undertsand when you it is only theory |
| 19:29.30 | adeel | LinuxMafia, it should be covered in chpater 1 |
| 19:29.43 | JayTee52 | [TK]D-Fender, cool, thanks again! I'd been digging around through docs and the book looking for more on the parameter and couldn't find anything. |
| 19:29.51 | LinuxMafia | adeel, thanks i will read it |
| 19:30.27 | jsmith-busy | JayTee52: It only restarts the idle channels, but I still disable it |
| 19:30.47 | gitguy | i got a DID number from didww.com, when I call this number I want to hear my IVR [incoming_calls], how do I do this? |
| 19:30.54 | gitguy | do I have to MAP my DID number? |
| 19:31.43 | JayTee52 | jsmith-busy , is there a particular place in the zapata.conf in the section for each span where it should go? |
| 19:32.21 | *** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com) |
| 19:32.37 | jsmith-busy | JayTee52: No, it just needs to be set *before* the channel definitions for that particular span |
| 19:32.47 | dlynes | jsmith-busy: so, I guess you weren't able to figure out what the other package was that I needed? Or, is there a search function on centos for packages? If so, I can just do that myself |
| 19:32.50 | adeel | gitguy, yes, you'd have to map it to the ivr |
| 19:32.54 | jsmith-busy | JayTee52: zapata.conf is read from top to bottom, and any setting that is set above a channel definition applies to that channel |
| 19:32.56 | JayTee52 | jsmith-busy, thanks |
| 19:33.10 | Confusd | hmm when i dial using chan_mobile, asterisk says autofallthrough after 20 seconds, even tho the other end hasnt hung up |
| 19:33.19 | Confusd | i havnt set any timers |
| 19:33.23 | Confusd | any ideas? |
| 19:33.25 | jsmith-busy | dlynes: Sorry, got distracted... still trying to log into my CentOS box |
| 19:33.37 | dlynes | jsmith-busy: ah, ok...no worries |
| 19:34.03 | jsmith-busy | dlynes: And the secret is.... "yum install libtool\*" |
| 19:34.16 | gitguy | adeel: thanks |
| 19:34.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:34.18 | dlynes | jsmith-busy: thank you kindly :) |
| 19:34.38 | gitguy | adeel: do i have to do something in my asterisk configs (server-side) or only from didww.com? |
| 19:35.22 | adeel | gitguy, on asterisk |
| 19:35.49 | gitguy | adeel: i have to put something in my configs? |
| 19:36.15 | adeel | you have to handle it in your dialplan |
| 19:36.38 | gitguy | didww.com documentation is very poor |
| 19:37.10 | adeel | gitguy, didwww has nothing to do with it....you need to be looking at howto handle incoming calls in the astersk book |
| 19:37.54 | *** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com) |
| 19:38.05 | gitguy | adeel: I know how to handle incoming calls, I already have a [incoming_calls] section for it, and the teliax DID number that I use works fine |
| 19:38.33 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:38.33 | gitguy | but I need to add another one |
| 19:38.36 | jsmith-busy | gitguy: You have to have an extension that matches the incoming DID |
| 19:39.12 | gitguy | ok |
| 19:39.23 | CasseTrop | Can someone tell me how to have a line display the " " on the next line on the display of the phone when I have this in the extensions.conf exten => s,1,Set(CALLERID(num)=${CALLERID(num)}-${CALLERID(dnid):-7}) |
| 19:39.32 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 19:39.33 | CasseTrop | sorry |
| 19:39.41 | *** join/#asterisk guillote_GNU (n=guillote@host7.201-253-11.telecom.net.ar) |
| 19:40.15 | banzaika | hello |
| 19:40.18 | CasseTrop | Can someone tell me how todisplay the "${CALLERID(dnid):-7}) " on the next line on the display of the phone when I have this in the extensions.conf "exten => s,1,Set(CALLERID(num)=${CALLERID(num)}-${CALLERID(dnid):-7})" |
| 19:40.37 | banzaika | CALLERID(name) ? |
| 19:41.06 | CasseTrop | banzaika talking to me ? |
| 19:41.44 | banzaika | there are 2 things you can set for caller id num and name |
| 19:42.04 | banzaika | depends on what your provider supports |
| 19:42.22 | CasseTrop | I already have the name and num, but i'd like the dnid to show under those 2, now is displaying on the same line. |
| 19:43.16 | banzaika | what type of phone are you using ? |
| 19:43.22 | CasseTrop | cisco 7940 |
| 19:43.33 | banzaika | does it supports 3 lines ? |
| 19:43.40 | banzaika | on display |
| 19:43.47 | CasseTrop | yesh it's like 12 lines |
| 19:44.25 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 19:44.57 | CasseTrop | http://www.3cx.com/support/images/cisco3.jpg this phone |
| 19:45.30 | banzaika | if it follows simple string rules you might be able to insert end-line char to jump to the next line |
| 19:47.01 | CasseTrop | how ? heh |
| 19:47.32 | banzaika | "\n" |
| 19:47.37 | CasseTrop | ok |
| 19:47.39 | CasseTrop | thanks |
| 19:47.55 | banzaika | "bla \n bla2" |
| 19:48.03 | banzaika | see if it works |
| 19:48.58 | banzaika | anybody is using asterisk hardware from digium ? |
| 19:49.21 | CasseTrop | nope |
| 19:49.23 | CasseTrop | doesn't work |
| 19:49.55 | jsmith-busy | banzaika: I am. |
| 19:50.18 | jsmith-busy | banzaika: I've got a pretty big hardware zoo now, with all kinds of Digium cards |
| 19:50.19 | *** join/#asterisk joshaidan (n=brianj@S01060090f8009fa6.tb.shawcable.net) |
| 19:52.46 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 19:53.04 | bsdwarrior | asterisk crashed for some reason my music on hold doesnt work. Ive made no config changes. |
| 19:53.23 | bsdwarrior | the log says Started music on hold then Stopped music on hold |
| 19:54.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:55.56 | bsdwarrior | I did a reload , reload extensions, reload res_musiconhold.so, nothing works |
| 19:59.46 | gitguy | adeel: I got the answer from their support, and as I though they would give me a useless answer: "Looking at the tcpdump you supplied from the dedicated server, it looks like it's not sending any packets back to you. You would see those packets regardless of our network configuration because they are originated from your local machiane. Either something is set up wrong in the configuration on the machine or the tcpdump needs to be ran longer to |
| 19:59.56 | *** join/#asterisk nvrpunk (n=zomgobli@81.90.21.227) |
| 19:59.56 | *** join/#asterisk CCFL_Man2 (i=32f1a794@pool-70-105-204-227.scr.east.verizon.net) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk kingsob (n=kingsob@HMTNON14-1242538569.sdsl.bell.ca) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk seanbright (n=sean@c-69-251-175-43.hsd1.md.comcast.net) |
| 19:59.56 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:59.56 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) [NETSPLIT VICTIM] |
| 19:59.56 | *** join/#asterisk juso (n=adamb@216.52.163.145) [NETSPLIT VICTIM] |
| 20:00.22 | banzaika | hmmm |
| 20:00.28 | adeel | gitguy, well they're wrong...you wouldn't see them because they're UDP packets |
| 20:00.53 | gitguy | ............ |
| 20:00.57 | adeel | gitguy, and you wouldn't see them anyways unless you're sending traffic back to yourself |
| 20:00.58 | gitguy | is really pissed now |
| 20:01.21 | banzaika | hmmmmm if you capture on the server you should see them all |
| 20:01.32 | banzaika | *2 cents* |
| 20:01.55 | adeel | banzaika, but the server should be sending replies, which never was seen |
| 20:01.57 | gitguy | THEY should FIX their f**** network |
| 20:02.32 | grandpapadot | gitguy: Who are you talking about? |
| 20:02.50 | gitguy | grandpapadot: i'm talking alone... |
| 20:03.27 | adeel | gitguy, i actually have to go for a while, i'll be back later tonight if you want to take a stab again |
| 20:04.12 | gitguy | adeel: ok, thanks for your help... i will just move on from that server |
| 20:05.03 | gitguy | i learned today that support is always useless |
| 20:05.33 | bsdwarrior | moh just stopped working. makes no sense. no config changes. |
| 20:05.45 | grandpapadot | bsdwarrior: static real-time? |
| 20:06.31 | bsdwarrior | grandpaopadot - mode=files |
| 20:07.33 | grandpapadot | bsdwarrior: permissions on the music files? |
| 20:07.40 | bsdwarrior | 644 |
| 20:07.53 | grandpapadot | bsdwarrior: gremlins? |
| 20:08.06 | bsdwarrior | its flaky |
| 20:08.42 | grandpapadot | What version of asterisk and linux distro? |
| 20:09.55 | banzaika | adeel, post the settings for tcpdump |
| 20:11.46 | *** join/#asterisk MacWinner (n=chatman@74-33-175-68.dsl1-merch.roc.ny.frontiernet.net) |
| 20:13.18 | Kobaz | [May 2 16:12:14] NOTICE[4266]: chan_iax2.c:8467 __iax2_poke_noanswer: Peer 'VP03' is now UNREACHABLE! Time: 21 |
| 20:13.21 | Kobaz | [May 2 16:12:15] NOTICE[4258]: chan_iax2.c:8467 __iax2_poke_noanswer: Peer 'VP01' is now UNREACHABLE! Time: 19 |
| 20:13.25 | Kobaz | yay |
| 20:13.33 | Kobaz | voicepulse goes down way too often |
| 20:14.11 | banzaika | try uding sip |
| 20:14.14 | banzaika | using |
| 20:15.07 | vader-- | are they done making updates on the 1.2.x line of asterisk? |
| 20:16.04 | bsdwarrior | restarting asterisk, music on hold is still busted |
| 20:16.14 | [TK]D-Fender | vader--: Security fixes only. Been that way for many months now |
| 20:16.36 | banzaika | use debug mode and see what is the error(s) |
| 20:16.56 | bsdwarrior | tkd-fender can you lend me a hand |
| 20:17.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 20:18.26 | [TK]D-Fender | bsdwarrior: Not currently, sorry. |
| 20:18.42 | jsmith-busy | bsdwarrior: Can I help? |
| 20:19.24 | vader-- | tkd how difficult is an upgrade from 1.2 to 1.4? |
| 20:19.27 | bsdwarrior | jsmith-busy for some reason the music on hold stopped working. I just restarted asterisk and have the same issue. NO clue whats wrong. I haven't changed any configs |
| 20:19.30 | MacWinner | kinda off topic, but anyone know of some service providers that white label a prepaid virtual calling card service.. basically want to let my customers login to portal, charge their card, and dial a phone number to then dial out. don't want to host this piece myself |
| 20:19.31 | vader-- | does it break stuff? |
| 20:20.40 | jsmith-busy | MacWinner: Yes, ask on the asterisk-biz list and you'll have lots respond to you |
| 20:20.53 | jsmith-busy | bsdwarrior: I'm assuming you've checked for a timing device, and done "moh show classes", and tried "moh reload", right? |
| 20:20.54 | MacWinner | or if i have to host it, any turnkey asterisk project to accomplish it |
| 20:21.07 | MacWinner | jsmith-busy: thanks, will do |
| 20:21.35 | bsdwarrior | jsmith-busy I will try this. Didnt even know about those commands. |
| 20:24.00 | bsdwarrior | jsmith - moh classes show - this shows the default class. moh reload. same problem |
| 20:24.58 | bsdwarrior | moh files show is blank |
| 20:26.06 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 20:26.08 | edwin_quijada | Hi! |
| 20:26.11 | *** join/#asterisk dlynes_ (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
| 20:26.20 | dlynes_ | jsmith-busy: thanks...it worked like a charm |
| 20:26.23 | edwin_quijada | We need a gsm gateway for 4 SIM cards |
| 20:26.29 | edwin_quijada | somebody knows something? |
| 20:26.42 | dlynes_ | jsmith-busy: I'm just not used to centos...used to slackware and debian etch |
| 20:26.52 | jsmith-busy | bsdwarrior: What version of Asterisk? |
| 20:27.00 | jsmith-busy | dlynes_: Glad I could help... |
| 20:27.02 | bsdwarrior | 1.2.14 |
| 20:27.29 | jsmith-busy | bsdwarrior: Ouch... I have no idea... you could try "unload res_musiconhold.so" and then reload the module |
| 20:27.42 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 20:27.48 | *** join/#asterisk PepOSX (n=angeldav@200.90.123.15) |
| 20:28.11 | bsdwarrior | k |
| 20:28.27 | *** join/#asterisk Greek-Boy (n=email@41.221.58.14) |
| 20:28.57 | vader-- | tkd how difficult is an upgrade from 1.2 to 1.4? does it break much? |
| 20:29.13 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 20:29.15 | bsdwarrior | when mode=files in musiconhold.conf what actually plays the music ? |
| 20:29.34 | [TK]D-Fender | vader--: depends, mot much usually. |
| 20:29.39 | grandpapadot | bsdwarrior: Sorry man, I dropped out after I asked 'what asterisk version and linux distro'? |
| 20:29.52 | vader-- | have you performed any upgrades on a production system? |
| 20:30.05 | edwin_quijada | NO gsm gateway to work with asterisk |
| 20:30.06 | bsdwarrior | centos /asterisk 1.2.14 |
| 20:30.07 | edwin_quijada | ? |
| 20:30.12 | bsdwarrior | its worked fine, but now it flaked out |
| 20:30.14 | vader-- | i have a production system running 1.2.7 |
| 20:30.23 | bsdwarrior | I bet rebooting the server will fix it, but can't do that right now |
| 20:30.30 | grandpapadot | bsdwarrior: Upgrade to 1.2.2x (latest), should be an easy upgrade... |
| 20:31.02 | bsdwarrior | grandpapadot - yeah on a friday, alot of syntax has changed, If I do that, alot of stuff will break |
| 20:31.27 | grandpapadot | bsdwarrior: Nothing changed from 1.2.14 to 1.2.26 that would negatively impact you, only bug fixes ... |
| 20:31.51 | bsdwarrior | still, cant do that on a friday |
| 20:31.55 | grandpapadot | 1.2.24 is what we're running, extremely stable. |
| 20:32.36 | vader-- | ya same grandpapdaot |
| 20:32.41 | bsdwarrior | any idea what plays the music when mode=files ? |
| 20:32.41 | vader-- | we are running 1.2.7 |
| 20:32.47 | grandpapadot | bsdwarrior: asterisk |
| 20:33.02 | *** join/#asterisk keulin (n=cray@247.126.64-86.rev.gaoland.net) |
| 20:33.37 | bsdwarrior | all it says is starting music on hold, then next line stopping music on hold. |
| 20:33.55 | grandpapadot | Did you restart asterisk at any point? |
| 20:34.07 | grandpapadot | Or do a full 'reload'? |
| 20:34.23 | edwin_quijada | I need a ip phone cheap? |
| 20:34.28 | edwin_quijada | but good |
| 20:34.32 | edwin_quijada | any ideas |
| 20:34.39 | bsdwarrior | grandpapadot - I stopped asterisk, killed off any stuck processes and started it again |
| 20:34.46 | edwin_quijada | never I have used sip phone |
| 20:35.02 | grandpapadot | edwin_quijada: Polycom 330, hands down. |
| 20:35.14 | bsdwarrior | moh classes show, shows both classes I have |
| 20:35.15 | grandpapadot | bsdwarrior: hrm.. that is odd |
| 20:35.35 | bsdwarrior | moh files show - shows nothing does this show anything for anyone else ? |
| 20:36.01 | grandpapadot | Yea, shows me all my classed and their associated music files |
| 20:36.14 | grandpapadot | But I'm running 1.2.24, you're running a bug-ridden ancident build ... |
| 20:36.24 | grandpapadot | May not be related or could be. |
| 20:36.37 | bsdwarrior | I plan to upgrade in the future |
| 20:37.18 | edwin_quijada | grandpapadot: this phone works ? |
| 20:38.42 | grandpapadot | No, I recommended a broken phone. |
| 20:39.36 | kingsob | any ideas whats wrong here?? http://rafb.net/p/y1RYXQ43.html |
| 20:40.09 | grandpapadot | Yes, It's a web server that is apparently designed to capture bits of pasted text for people to review later. |
| 20:40.29 | grandpapadot | Web servers work using TCP on port 80 using the HTTP protocol. |
| 20:42.54 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 20:43.27 | bsdwarrior | would be nice if it gave me an error message |
| 20:43.43 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:44.07 | kingsob | same thing happens for incoming called.. it works when i have the DID in iax.conf... but in users.conf, which is where the web gui sticks it i get this error http://rafb.net/p/J8vYHk96.html |
| 20:44.18 | edwin_quijada | I have this phone IP Phone 7910 1 Line |
| 20:44.24 | grandpapadot | kingsob: Is that freepbx? |
| 20:44.29 | edwin_quijada | this compatible with asterisk |
| 20:44.32 | kingsob | nope |
| 20:45.07 | kingsob | thats what i see in the asterisk console when i call my number |
| 20:45.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:46.21 | tinkerghost | OK, that's just wrong - "Unlimited local DID packages .... limited to 4000 inbound minutes per month" |
| 20:46.42 | grandpapadot | tinkerghost: lol, what service is that? |
| 20:46.51 | tinkerghost | Obviously a marketing person told them what the definition of unlimited was |
| 20:47.16 | tinkerghost | grandpapadot: Vitelity, one of 2 I've found that offer service to my area |
| 20:47.30 | tinkerghost | axvoice is the other |
| 20:47.57 | *** join/#asterisk brianoak (n=chatzill@adsl-70-231-232-131.dsl.snfc21.sbcglobal.net) |
| 20:49.18 | tinkerghost | I really wanted ipcommunications, free DID & incomming calls, but no love for western MA |
| 20:49.50 | grandpapadot | tinkerghost: That link I sent you ... They offer just that ... |
| 20:50.11 | edwin_quijada | Any phone that use SIP can connect to asterisk? |
| 20:50.58 | grandpapadot | edwin_quijada: You asked for a good phone, at the top of the list is Polycom, their cheapest "good" phone is the 320/330. Don't go lower than that, you'll regret it later. |
| 20:50.59 | kingsob | ffs, i had my disallow after my allow |
| 20:52.23 | brianoak | I'm an Asterisk newbie, wondering if there's a graphical enviornment for the IVR. Something to let biz users manage the "if they hit 2, play this mp3 and put them here in the tree" a plug-in,service etc?? |
| 20:52.53 | tinkerghost | kingsob: I had a script I did that to earlier today --- pasted a big long calculation in above the default instead of below |
| 20:53.03 | brianoak | can anyone steer me to a URL/program/company/service/etc? |
| 20:54.04 | kingsob | lol |
| 20:55.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:57.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:02.15 | *** part/#asterisk fmueller (n=user@p548F58EF.dip.t-dialin.net) |
| 21:04.24 | tinkerghost | OK , time for me to head out ... been 10 hours already today |
| 21:04.31 | *** part/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 21:05.44 | *** join/#asterisk confusd (n=Confusd@87-194-204-58.bethere.co.uk) |
| 21:09.24 | tzafrir_home | brianoak, there aer a number of graphical interfaces to Asterisk that provide that for IVR. e.g: freepbx and the asterisk-gui |
| 21:09.31 | bsdwarrior | I copied an old config for musiconhold.conf and the old config which everyone said shouldnt even work, works. |
| 21:10.36 | bsdwarrior | I'd rather write config files for sendmail at this point. lol |
| 21:10.56 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 21:11.18 | tzafrir_home | bsdwarrior, with the "bad" config file, what do you see with 'moh show classes'? |
| 21:11.30 | brianoak | tzafrir: thanks, i'm looking at those now |
| 21:11.56 | brianoak | i couldn't tell if asterisk-gui was just for the basic small office set up or did IVRs |
| 21:12.03 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 21:13.07 | bsdwarrior | tzafrir_home the exact same as my old config that stopped working |
| 21:14.05 | bsdwarrior | for whatever reason moh files show displayed nothing. maybe I should read the source code and find where in the code there's no error reporting it. |
| 21:14.07 | tzafrir_home | brianoak, IVR is a very basic feature of Asterisk. You should expect it to be available even for a "small office setup" |
| 21:14.09 | grandpapadot | Hidden mal-formed characters? |
| 21:14.42 | tzafrir_home | bsdwarrior, time to add some error reporting? |
| 21:14.47 | *** join/#asterisk flo|va-nu-pied (n=flo|va-n@unaffiliated/flovanupied/x-758957) |
| 21:15.13 | flo|va-nu-pied | hi all |
| 21:20.35 | *** join/#asterisk lesouvage (n=lesouvag@s55921cce.adsl.wanadoo.nl) |
| 21:22.32 | lesouvage | I have a problem building asterisk on a new machine. Did an update of make like adviced on guru site but it doesn't help. I paste the output to http://www.pastebin.be/10956 . Anybody has a clue? |
| 21:22.47 | lesouvage | version 1.4.19.1 |
| 21:24.25 | *** join/#asterisk Confusd (n=Confusd@87-194-204-58.bethere.co.uk) |
| 21:24.36 | tzafrir_home | lesouvage, is compalins about missing "g++". Do you have gcc-c++ (or whatever) installed? |
| 21:25.01 | *** join/#asterisk UQlev (n=kvirc@ykulyev.logos.cy.net) |
| 21:27.02 | nvrpunk | I have an IAX connection between two * boxes and I am trying to get it so that inbound DIDs answer at the last box, the other box registers with the PSTN but for some reason it's getting no route to destination |
| 21:27.34 | nvrpunk | this is the dial string getting passed: IAX2/jackalteleport/2055884101 |
| 21:27.43 | nvrpunk | No such host: jackalteleport |
| 21:28.04 | nvrpunk | yet, the IAX context name is jackalteleport in the iax.conf on that box |
| 21:29.29 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 21:29.29 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:30.11 | tzafrir_home | nvrpunk, is it a peer? |
| 21:30.11 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-66-31.bstnma.east.verizon.net) |
| 21:30.24 | nvrpunk | tzafrir, nvm I realized somethign I did stupid |
| 21:30.25 | tzafrir_home | (type = peer, or: type = friend)? |
| 21:30.26 | lesouvage | Itzafrir_home: have the packages on the suse list on asterisk guru site installed and gcc-c++ was not on the list. I'm afraid not. |
| 21:30.59 | tzafrir_home | lesouvage, I still wonder how this could have gotten passed the configure script if this is indeed the issue |
| 21:31.19 | *** join/#asterisk wichitalineman (n=svf@67.106.246.227.ptr.us.xo.net) |
| 21:32.23 | nvrpunk | tzafrir, user |
| 21:35.43 | lesouvage | tzafrir: isn't the c++ part supposed to be included in gcc? (see:http://gcc.gnu.org/) btw its on a fresh basic suse enterprise installation (uname -a -> Linux DLX127 2.6.16.21-0.8-smp #1 SMP Mon Jul 3 18:25:39 UTC 2006 i686 i686 i386 GNU/Linux) |
| 21:36.50 | *** join/#asterisk geek_cl (n=lletelie@200.75.18.211) |
| 21:37.21 | wichitalineman | lesouvage some distros split it out so the c++ component is called g++ |
| 21:38.25 | wichitalineman | RH used to, the Debian derived definitely do |
| 21:39.32 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:40.03 | wichitalineman | anyway, ask yast. it should know |
| 21:40.21 | nvrpunk | tzafrir, ah, now I got it to the end box but it's saying "request '2055884101@from-pstn' does not exist" |
| 21:43.31 | HuntsvilleAL | ugh, i just swalloed a whole ice cube by accident |
| 21:44.47 | tzafrir_home | Nice. An asterisk dialer that works with a proxy on the server rather than direct manager interface access: |
| 21:44.51 | tzafrir_home | http://astitray.wiki.sourceforge.net/ |
| 21:45.25 | tzafrir_home | nvrpunk, what is the "it" that gives you this emssage? |
| 21:45.44 | tinman_gz | is there a way to bridge 2 arbitrary channels without using a meetme? |
| 21:46.05 | nvrpunk | tzafrir, I got it now |
| 21:46.39 | nvrpunk | tzafrir, however when it passes over to the end box it's adding @pstn-in after the number and I have no clue where it is getting that part |
| 21:47.07 | nvrpunk | tzafrir, i had to replace the jackalteleport part in the ext.conf to the actual IP of the endbox |
| 21:49.16 | tinman_gz | it looks like pickup will pickup any ringing extension... is there a way to put a call in 'ringing mode'? |
| 21:54.53 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:00.37 | *** join/#asterisk AsteriskRo (n=rosadesa@190.36.186.73) |
| 22:00.52 | AsteriskRo | hello all |
| 22:00.58 | nvrpunk | hola |
| 22:06.34 | lesouvage | wichitalineman/tzafrir: thanks for the help. I'm afraid they have to wait, I don't have phisical access to the server needed to put the proper cd-rom into the machine :-( |
| 22:07.36 | *** join/#asterisk xacatecas (n=jkroon@196.46.172.142) |
| 22:08.16 | xacatecas | hi guys, i'm looking for a somewhat decent soft phone. any recommendations? |
| 22:08.54 | russellb | xlite / eyebeam / zoiper |
| 22:09.15 | xacatecas | xlite doesn't want to run here. |
| 22:11.27 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:11.51 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:11.57 | *** part/#asterisk CasseTrop (n=cast@64.254.255.99) |
| 22:12.24 | xacatecas | zoiper seems to be commercial and a google on eyebeam only gets me some cartoon stuff. |
| 22:14.04 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 22:14.55 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 22:16.00 | xacatecas | wengophone? |
| 22:16.19 | xacatecas | anybody who used it? |
| 22:17.03 | AsteriskRo | anybody to ask about IAXmodem? |
| 22:17.15 | wichitalineman | lesouvage: np. save yourself a bit of trouble when you get a few minutes and see what rpms comprise a SuSE dev environment. Or you can just install things as your project breaks :) |
| 22:20.52 | *** join/#asterisk ManxPower (n=manxpowe@12.88.70.202) |
| 22:27.01 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 22:30.39 | *** part/#asterisk RoyK (n=roy@ip-86-12-149-91.dialup.ice.no) |
| 22:34.36 | *** join/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 22:38.05 | nvrpunk | strange, I have no clue why when I set an IAX jitterbuffer and set maxjitterbuffer=500 it manages to say max timeslots exceeding 600 |
| 22:38.07 | nvrpunk | :( |
| 22:38.27 | nvrpunk | this takes a while the call gets choppy then it hangs p |
| 22:38.29 | nvrpunk | up |
| 22:38.54 | *** join/#asterisk keith4__ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 22:42.34 | keith4__ | so, asterisk seems to be running fine. I can sip to it and listen to the demo, etc., but I can't dial out any zap channels. I get "Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)" |
| 22:43.03 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 22:45.03 | keith4__ | ah, I'm an asshole |
| 22:45.19 | keith4__ | zaptel.conf is set to fxs_ks, but zapata.conf is fxs_ls |
| 23:03.08 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:08.55 | *** part/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 23:09.09 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 23:14.19 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 23:21.36 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 23:24.19 | jong2 | friday beer time? |
| 23:24.42 | wichitalineman | may I have several beers? |
| 23:25.11 | jong2 | guiness? |
| 23:25.13 | JayTee52 | now there's a Glen Campbell tune I haven't heard in ages |
| 23:26.13 | wichitalineman | the guy who wrote it was set for life |
| 23:26.22 | JayTee52 | âI would like to make a toast to lying, stealing, cheating and drinking. If you're going to lie, lie for a friend. If you're going to steal, steal a heart. If your going to cheat, cheat death. And if you're going to drink, drink with me.â |
| 23:26.47 | jong2 | i will drink to that. |
| 23:29.17 | *** part/#asterisk LemensTS (i=CustomGT@adsl-70-238-154-213.dsl.stlsmo.sbcglobal.net) |
| 23:32.09 | adeel | JayTee52, i've heard multiple variations of that toast |
| 23:32.44 | JayTee52 | I didn't make it up, hence the quotes. |
| 23:33.36 | adeel | ah, didn't see the quots |
| 23:34.29 | jong2 | 20 more minutes till beer |
| 23:35.03 | jong2 | maybe i should join different channel |
| 23:36.08 | *** join/#asterisk jmesquita (n=jmesquit@200.162.210.96) |
| 23:36.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:41.01 | jong2 | is switchvox popular? |
| 23:41.22 | nvrpunk | i think trixbox is more so but sure |
| 23:41.23 | nvrpunk | :) |
| 23:42.12 | jong2 | ic |
| 23:44.13 | jong2 | do they have proprietary thing in their configuration file? |
| 23:47.18 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:48.57 | jong2 | here comes beer!!! |
| 23:51.12 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 23:51.57 | joat | just an odd question: is there any way to tell via a command or the command line if a conference is locked? |
| 23:52.05 | joat | meetme conference that is... |
| 23:57.12 | *** join/#asterisk ManxPower (n=manxpowe@adsl-150-2-141.aby.bellsouth.net) |
