| 00:01.53 | onlyI | tzanger you there |
| 00:01.53 | tzanger | nope |
| 00:01.57 | onlyI | good |
| 00:02.00 | onlyI | ;) |
| 00:02.03 | onlyI | can i PM |
| 00:02.20 | tzanger | bajanman: if you're taking in calls and you need them to go to a different context (or machine) ... that's the big one |
| 00:03.04 | onlyI | tzanger did you get your tdm11b |
| 00:03.24 | bajanman | no. I'm trying to figure out why, one minute I can call, and the next its busy (outside to my *), but when I dial from * to outside, it works, then right afterwards, I can call to my * |
| 00:03.26 | bajanman | that's what's weird |
| 00:03.38 | tzanger | onlyI: yup but they're sold |
| 00:03.45 | onlyI | god |
| 00:04.10 | tzanger | bajanman: sounds like a registration or NAT timeout |
| 00:04.13 | miguellinux | opus_, hi, yes I cant register to any sip provider I get SIP 483 error Too many hops |
| 00:04.45 | bajanman | tzanger: I think youre right |
| 00:05.18 | tzanger | I'm always right. |
| 00:06.38 | bajanman | I'm thinking it has to do with the externip =hostname issue... |
| 00:06.45 | tzanger | it saves a lot of guesswork :-) |
| 00:08.14 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 00:08.57 | neopher | whats a good "free" network anaylizer |
| 00:09.18 | neopher | plan on running it on neolinux |
| 00:10.17 | Nugget | 99 flavors of linux on the wall... 99 flavors of linux... take one down and pass it around... 118 flavors of linux on the wall... |
| 00:11.18 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:13.30 | opus_ | <miguellinux> what is your traceroute ? |
| 00:15.40 | DrukenHME | anyone here done an agi in php ? |
| 00:19.48 | Nugget | php has always seemed to me like a strange language to develop non-web apps in. |
| 00:20.39 | Sato1 | havent seen yet any php on non-web apps |
| 00:20.43 | DrukenHME | Nugget: perhaps... but if it's the only language you know.... |
| 00:21.18 | Nugget | I know a better solution to that dilemma than "use php for everything" :) |
| 00:21.34 | Sato1 | DrukenHME, you may try based in the samples of perl, if you know php, you will catch perl |
| 00:22.00 | *** join/#asterisk folsson (~filip@h100n2fls35o985.telia.com) |
| 00:22.09 | DrukenHME | oh probably... i've played with perl before.... it pisses me off too :) |
| 00:23.33 | Sato1 | so.. there you have a sample so you can start doing somethings with php, it took me about 20 minutes to understand the perl scripts and do my very first agi in TCL, just saying digits |
| 00:24.07 | Sato1 | and look for agi commands in voip-info.org |
| 00:25.40 | *** join/#asterisk BSDnewbie (hehe@202.179.26.180) |
| 00:25.52 | BSDnewbie | hi ppl, |
| 00:26.10 | BSDnewbie | Has asterisk web interface or graphical GUI? |
| 00:26.18 | Nugget | no |
| 00:26.29 | Sato1 | those are separated projects |
| 00:26.41 | Nugget | there are third party web interfaces, but I can almost guarantee that they don't do what you're hoping they do. |
| 00:26.53 | BSDnewbie | aanha |
| 00:26.57 | BSDnewbie | what projects? |
| 00:27.17 | Sato1 | BSDnewbie, search in voip-info.org, there are a list of interfaces |
| 00:27.31 | BSDnewbie | Okay tnx Sato1, Nugget |
| 00:27.34 | Sato1 | mine is not yet posted there, need to translate it to english |
| 00:27.35 | Nugget | if you are looking for a gui to asterisk because you think it will allow you to avoid learning how to set up asterisk, you should expect to be disappointed. |
| 00:27.53 | Sato1 | good point |
| 00:28.01 | blitzrage | aye |
| 00:28.05 | Nugget | there are web and gui interfaces which can allow you to automate or delegate the routine maintenance of a working asterisk server, but not much more. |
| 00:28.16 | BSDnewbie | oww |
| 00:28.19 | blitzrage | I started creating a GUI for Asterisk and quickly realized documentation was more important |
| 00:28.32 | BSDnewbie | can i get billing and record results via GUI? |
| 00:28.38 | Nugget | yes, that's possible |
| 00:28.54 | Sato1 | now, if you are looking for an easy way to configure your asterisk, i would recomend you to look for Asterisk@home, still, you will need to understand the basics of asterisk |
| 00:29.10 | BSDnewbie | that is what i need :D. I don't need installation, configuration GUI |
| 00:29.31 | Sato1 | BSCnewbie, you can enable cdr_mysql then create your own GUI |
| 00:29.51 | Nugget | http://areski.net/asterisk-stat-v2/about.php might do what you want. |
| 00:29.56 | *** join/#asterisk jtodd (~jtodd@blob.fox-den.com) |
| 00:30.31 | BSDnewbie | aahan, |
| 00:30.58 | miguellinux | opus_, only 9 hops from the * to sip.simpletelecom.com |
| 00:32.08 | miguellinux | opus_, IAX2 rules.. but SIP registration with this providers (simpletelecom and Broadvoice) |
| 00:35.04 | miguellinux | which other ports I have to redirect , because it seems to be a problem of NAT |
| 00:35.09 | christo | I'm using the manager interface to setup SIP calls, but I get the error "Got SIP response 482 "Loop Detected" back". Has anybody seen this one before? |
| 00:40.08 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:40.36 | bajanman | would NAT cause this: |
| 00:40.36 | bajanman | NOTICE[2702]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 00:40.36 | bajanman | <PROTECTED> |
| 00:41.01 | bajanman | I'm trying to figure out why, when I do a transfer to my sip phone I get a busy... when its available |
| 00:43.33 | onlyI | Nugget what happen to meetme2 page |
| 00:45.52 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 00:45.52 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:47.20 | Weezey | bajanman: is it registered? |
| 00:49.09 | bajanman | weezey: yes |
| 00:49.51 | Weezey | are you doing Dial(SIP/context|20|r) ? |
| 00:50.07 | bajanman | weezey: I had problems dialing to the *, and out. it would work, then not, then busy, so I decided to setup greetings. and transfers |
| 00:50.27 | bajanman | that I did try: was good/bad |
| 00:50.36 | bajanman | so now i'm doing tansfers instead |
| 00:50.49 | Weezey | what do you mean by transfers? |
| 00:50.58 | bajanman | so, I call to *, I get my board, and I try an extension to a sip phone: but its always busy |
| 00:51.02 | bajanman | oh |
| 00:51.10 | bajanman | I dial into the switchboard |
| 00:51.13 | bajanman | and do a transfer |
| 00:51.24 | Weezey | i see. |
| 00:51.27 | bajanman | but every time I do, i get the message my sip phone is busy |
| 00:51.32 | bajanman | which it isn't |
| 00:51.42 | Weezey | what's the context of your phone? |
| 00:52.21 | miguellinux | Help, I get SIP 483 error "Too many hops" when I try to call by a registered sip provider, please help |
| 00:52.38 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 00:52.38 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:53.15 | Nethab | marantz |
| 00:54.40 | *** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net) |
| 00:56.44 | drbrown | Weezey: How do you apply the auto-answer support patch to asterisk? |
| 01:01.13 | Weezey | I would assume: patch -p1 < filename |
| 01:01.55 | tzanger | I always use -Np1 --dry-run to make sure it applies clean; otherwise it's a pain cleaning up after it shits all over the source dir |
| 01:02.35 | Weezey | tzanger: good point. |
| 01:03.19 | drbrown | Weezey: I will try, thank you |
| 01:03.50 | Weezey | can anyone tell me how to make chan_h323.so load? I'm getting no error, it just dies. |
| 01:04.09 | Nethab | don't use the one that comes with asterisk |
| 01:04.13 | Weezey | (also is there a way besides -vvvvvvvvvvvvvvvvvvvvvvvvvv to make errors show up?) It seems |
| 01:04.22 | Weezey | Nethab: I can't compile oh323 |
| 01:04.45 | Weezey | chan_pvt has been removed from CVS since early april, so oh323 won't compile. |
| 01:07.30 | *** join/#asterisk Rick_Hunter (~rhunter@01-196.008.popsite.net) |
| 01:13.57 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 01:18.55 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 01:18.58 | jskcr|lappy | hy all |
| 01:20.14 | Nethab | hello |
| 01:20.22 | jskcr|lappy | Im starting to make a live asterisk cd |
| 01:20.47 | Nethab | congratulations |
| 01:21.14 | jskcr|lappy | No spent about a year at a company working on custom live cd's |
| 01:21.19 | jskcr|lappy | s/No/i lol |
| 01:21.42 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:21.58 | *** part/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:22.14 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:22.21 | Sato1 | wrong botton :S |
| 01:27.21 | *** part/#asterisk T-Squared (~ted@hidden.serreyn.com) |
| 01:27.22 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 01:27.33 | shmaltz | hi everybody |
| 01:28.48 | shmaltz | tzanger, you around? |
| 01:31.20 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
| 01:34.08 | *** join/#asterisk Rick_Hunter (~rhunter@04-055.008.popsite.net) |
| 01:35.28 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:40.34 | *** join/#asterisk Entegrity (~Entegrity@c-65-96-119-254.hsd1.ma.comcast.net) |
| 01:41.56 | *** join/#asterisk PBXtech (nik@70-58-41-173.slkc.qwest.net) |
| 01:43.41 | PBXtech | why is my irq CPU load like 30% ? is that normal? or is it because im using ztdummy |
| 01:44.09 | jskcr|lappy | yup |
| 01:44.21 | *** join/#asterisk Taadow (Taadow@70.70.36.6) |
| 01:44.33 | PBXtech | or real |
| 01:44.41 | Taadow | Jas->Not sure if you're around. |
| 01:45.37 | Taadow | Anyone know why (with h.323 debug on) under External RTP Session Starting RTP channel paramaters, it says ExternalIpAddress: 127.0.0.1. |
| 01:45.51 | Taadow | In call. |
| 01:46.00 | Taadow | Well, trying to establish I mena. |
| 01:46.02 | Taadow | mean |
| 01:46.08 | PBXtech | sounds like nat prob eh |
| 01:46.27 | Taadow | No nat. |
| 01:46.59 | Taadow | gateway to gateway |
| 01:47.57 | Entegrity | Hello, I'm trying to use asterisk to route all calls from a callmanager sip trunk to a vonage sip trunk and vice versa. What would my extensions.conf need to have. The sip pieces are done. [callman01] & [Vonage] are sip contexts. |
| 01:47.57 | Taadow | External RTP Session Starting |
| 01:47.57 | Taadow | RTP channel id 1 parameters: |
| 01:47.57 | Taadow | -- remoteIpAddress: xx.xx.xx.xx |
| 01:47.57 | Taadow | -- remotePort: 10084 |
| 01:47.57 | Taadow | -- ExternalIpAddress: 127.0.0.1 |
| 01:47.58 | Taadow | -- ExternalPort: 18128 |
| 01:48.06 | Taadow | Very strange. |
| 01:48.31 | Taadow | I'm assuming that would most likely be the reason I'm not getting audio. |
| 01:49.21 | *** join/#asterisk HeadachesAbound (~mirc@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 01:51.37 | HeadachesAbound | Gee, it sure is quiet in here. |
| 01:51.38 | Entegrity | Could anyone help me with extensions.conf (I'm new to asterisk) |
| 01:51.48 | HeadachesAbound | What kind of help you need? |
| 01:51.52 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 01:51.52 | Entegrity | Hello, I'm trying to use asterisk to route all calls from a callmanager sip trunk to a vonage sip trunk and vice versa. What would my extensions.conf need to have. The sip pieces are done. [callman01] & [Vonage] are sip contexts. |
| 01:51.58 | Entegrity | ^ :op |
| 01:52.32 | Entegrity | I may have somethings missing on callmanager but I see this in asterisk |
| 01:52.44 | Entegrity | I'm just too new to asterisk to understand the extensions.conf |
| 01:52.49 | Entegrity | and what it should look like |
| 01:53.06 | Sato1 | Entegrity, what do you see in the console? |
| 01:53.10 | Entegrity | Vonage/17813536 216.115.25.198 255.255.255.255 5061 Unmonitored |
| 01:53.11 | Entegrity | callman01 192.168.200.21 255.255.255.255 5060 OK (6 ms) |
| 01:53.33 | Sato1 | no, you should have some extra messages anytime a call is generated |
| 01:53.39 | Sato1 | or in any attempt |
| 01:53.41 | *** join/#asterisk suma (~suma@81-86-77-235.dsl.pipex.com) |
| 01:53.46 | Entegrity | hmm |
| 01:53.59 | Entegrity | my dialplan is royally screw up right now |
| 01:54.09 | Entegrity | I used asterisk@home :o |
| 01:54.17 | Qwell | Thats why its all screwed up... |
| 01:54.21 | Entegrity | I renamed all the stuff they did |
| 01:54.29 | Sato1 | start your asterisk using some "v" at it, if you do "asterisk -cg" add some "v"'s |
| 01:54.35 | Sato1 | or asterisk -r, then asterisk -rvvvvvvvvv |
| 01:54.35 | Entegrity | trying to create my own extensions.conf |
| 01:54.49 | Entegrity | v? |
| 01:54.54 | Sato1 | vervouse |
| 01:55.10 | Entegrity | whats the default extensions.conf |
| 01:55.59 | Sato1 | Entegrity, you may try www.voip-info.org, extensions.conf is a wide subject to explain, get as much information you can, then you can ask specific questions |
| 01:56.47 | jskcr|lappy | If that does not work I take paypal :) |
| 01:56.54 | Sato1 | lol |
| 01:57.23 | Entegrity | I'm close to building ser |
| 01:57.26 | Entegrity | I just need sip proxy |
| 02:00.12 | Sato1 | how many devices you have with your asterisk? |
| 02:00.16 | jskcr|lappy | Asterisk is *not* a SIP proxy. A SIP proxy handles call control on behalf of other user agents (UA) and usually does not maintain state during a call and therefore is never the endpoint of a call. |
| 02:00.22 | Entegrity | none |
| 02:00.34 | Entegrity | I just want to learn callmanager |
| 02:00.39 | Entegrity | and use a voip provider |
| 02:00.41 | Entegrity | :\ |
| 02:00.56 | Entegrity | but vonage sucks |
| 02:01.00 | Entegrity | and its unreachable now |
| 02:01.10 | Sato1 | so, all you got is an asterisk, the callmanager and a provider link? |
| 02:01.12 | Entegrity | it was working at one point |
| 02:01.19 | Entegrity | yes |
| 02:01.24 | Sato1 | hmmm... |
| 02:01.25 | jskcr|lappy | What are you using vonage soft phone as you sip connection |
| 02:01.28 | Entegrity | and a softphone to test if needed |
| 02:01.35 | Sato1 | ah! |
| 02:01.37 | Entegrity | yes jsharp |
| 02:01.38 | Entegrity | err |
| 02:01.44 | Entegrity | yes vonage sphone |
| 02:01.50 | Entegrity | for sip provider |
| 02:01.55 | Entegrity | for now anyways... |
| 02:02.00 | jskcr|lappy | And now all of a sudden vonage stoped working |
| 02:02.00 | Entegrity | looks broke again |
| 02:02.11 | Entegrity | ya |
| 02:02.16 | Entegrity | sip connection broke |
| 02:02.25 | Entegrity | I had it working at one point :| |
| 02:02.28 | jskcr|lappy | wait 120 seconds before trying to reestablish it |
| 02:02.31 | jskcr|lappy | its still working |
| 02:02.51 | jskcr|lappy | This is common to vonage I had the same problem its not you its vonage |
| 02:03.17 | Entegrity | how do I reestablish it? |
| 02:03.32 | Entegrity | I put everything back to when it worked... |
| 02:03.35 | jskcr|lappy | wait two minutes before trying |
| 02:04.01 | jskcr|lappy | if you try to establish to fast to vonage it will block you automaticly for like 2-5 minutes |
| 02:04.17 | Entegrity | thats problably whats giving me the headache |
| 02:04.34 | jskcr|lappy | yea I sat for like 2 hours and finnaly packet sniffed it to see its not me its vonage |
| 02:05.06 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:05.46 | jskcr|lappy | Quoth their TOS: " c. Unauthorized Usage Customer may not program the Number into any equipment or infrastructure in or on which the number is intended or used as the origination or destination of a communication other than the Device that was provided by Vonage." |
| 02:05.46 | jskcr|lappy | <PROTECTED> |
| 02:06.17 | Entegrity | oh brother |
| 02:06.22 | Entegrity | I'm getting new service tomorrow |
| 02:06.26 | Entegrity | so |
| 02:06.28 | jskcr|lappy | In other words they can terminate your account if they feel like it because you use asterisk on the soft fone |
| 02:06.30 | Entegrity | I need advice |
| 02:06.32 | Qwell | Vonage? |
| 02:06.36 | Qwell | yeah... |
| 02:06.56 | Entegrity | this is what I want to do: Use Callmanager to VOIP provider |
| 02:07.07 | Entegrity | it supports sip, but not very well |
| 02:07.15 | Entegrity | needs a proxy or some sort to do the authentication etc |
| 02:07.30 | Entegrity | supports h323 as well |
| 02:07.40 | Entegrity | what should I use to interface with the provider? |
| 02:07.42 | Entegrity | SER? |
| 02:07.49 | jskcr|lappy | Broadvoice |
| 02:07.53 | Entegrity | yep |
| 02:07.59 | Entegrity | they use sip right? |
| 02:08.15 | jskcr|lappy | yea |
| 02:08.25 | Entegrity | so Callmanager -> h323orSIPproxy -> SIPprovider(broadvoice) |
| 02:08.37 | Entegrity | I've been battling what to use for the h323orSIPproxy |
| 02:08.56 | *** join/#asterisk steven_ (optimist@eurocompton.net) |
| 02:09.03 | Entegrity | I haven't tried SER yet... |
| 02:09.25 | Entegrity | but I figured asterisk would be a good option. |
| 02:09.46 | steven_ | I have an analog phone connected to a linksys pap2-na which talks to asterisk via SIP. I'd like to be able to get new voicemail notification alerts sent to this phone. |
| 02:09.55 | steven_ | what do I need to do to make this possible? |
| 02:10.18 | wildcard0 | try mailbox=xxx in your sip.conf |
| 02:10.47 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-53-145.dsl.pltn13.pacbell.net) |
| 02:11.05 | steven_ | wild, but I have that defined. |
| 02:11.07 | Himeko | MWI |
| 02:11.37 | wildcard0 | ya. mwi is the next thing to try if that doesn't work |
| 02:12.01 | steven_ | what's MWI? |
| 02:12.07 | Himeko | something like it is broken if you are reading your cfg out of a db |
| 02:12.32 | Himeko | ask slePP or LOT maybe |
| 02:12.40 | wildcard0 | mwi = message waiting indicator |
| 02:12.45 | steven_ | oh |
| 02:12.47 | slePP | ? |
| 02:12.48 | wildcard0 | the device needs to have it on as well |
| 02:12.50 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-102.modem.logical.net) |
| 02:12.51 | slePP | what'd i do? |
| 02:13.06 | Himeko | slePP didn't you get mwi working |
| 02:13.06 | wildcard0 | you made the baby jesus cry! |
| 02:13.09 | Entegrity | anyone have a broadvoice Retail Activation Code or Promo Code? |
| 02:13.10 | steven_ | the device has it on, where do you set the mwi in asterisk? |
| 02:13.11 | Carp1 | Anyone in here use PHPAGI? |
| 02:13.23 | steven_ | I had MWI working with Asterisk@Home |
| 02:13.27 | slePP | Himeko: yeh, by getting rid of realtime sipfriends :> |
| 02:13.45 | Entegrity | Retail Activation Code or Promo Code? I'll send a referal |
| 02:13.52 | *** join/#asterisk iq|laptop (~iq@65-103-164-23.omah.qwest.net) |
| 02:14.08 | steven_ | slepp, what did you have to do to get MWI working? |
| 02:14.17 | Carp1 | Whats the difference between AGI and DeadAGI? |
| 02:14.20 | slePP | i stopped using sipfriends from a db |
| 02:14.25 | slePP | and just generate a sip.conf from the database instead |
| 02:14.31 | Carp1 | Nevermind. |
| 02:14.34 | Carp1 | Just foudn the answer. |
| 02:14.48 | steven_ | slepp, there's no setting to enable? |
| 02:14.53 | steven_ | I'm not using a a db |
| 02:15.06 | iswm | How can I get incoming calls to wait for me to pick up? |
| 02:15.08 | slePP | you're using sip.conf? |
| 02:15.12 | steven_ | yes |
| 02:15.17 | slePP | if you are, then you need to set: mailbox=mailbox@context |
| 02:15.22 | slePP | ie: mailbox=1001@default |
| 02:15.23 | slePP | in the sip entry |
| 02:15.27 | steven_ | I have that setting. |
| 02:15.41 | slePP | then it should work if the device supports it, and the mailbox exists |
| 02:15.42 | steven_ | strange it's not working. |
| 02:16.59 | suma | anybody has latest version of Cisco 7960 IP Phone firmware |
| 02:17.03 | suma | thanks in advance |
| 02:17.09 | mDuff | are current CVS builds expected to work under valgrind? "valgrind --tool=addrcheck asterisk -c -f" is aborting for me with "stack smashing attack in function read()" |
| 02:17.15 | Carp1 | Anyone in here use PHPAGI? |
| 02:17.27 | *** join/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 02:17.38 | mbishop | how do I set up asterisk to wait for me to answer incoming calls? |
| 02:17.44 | ariel_ | Carp1, that is a loaded question. |
| 02:17.49 | suma | mbishop: Wait |
| 02:18.06 | ariel_ | exten => s,1,Wait(20) |
| 02:18.07 | kimo_sabe | mbishop: how do you mean? |
| 02:18.07 | Carp1 | ariel_: ? |
| 02:18.33 | suma | anybody has latest version of Cisco 7960 IP Phone firmware |
| 02:18.49 | ariel_ | Carp1, I use php and most of the people use it in some way or another. |
| 02:19.30 | Carp1 | No... |
| 02:19.32 | Carp1 | PHPAGI |
| 02:20.56 | Carp1 | http://pastebin.ca/10792 |
| 02:21.05 | Carp1 | Completes, but doesnt stream the file. |
| 02:21.17 | ariel_ | Carp1, your looking at version 1.12 or the newer one 2.0? |
| 02:21.30 | Carp1 | 1.12 |
| 02:21.33 | Carp1 | I didnt know there was newer |
| 02:22.18 | steven_ | slepp, should type = peer or friend? |
| 02:22.32 | ariel_ | Carp1, first check path and then rights |
| 02:23.04 | slePP | steven_: i think mailbox only works for type=user |
| 02:23.05 | slePP | or type=friend |
| 02:23.18 | Carp1 | ariel_: What do you mean the path? |
| 02:23.29 | steven_ | it's type=friend right now.. not working. perhaps I'll try user |
| 02:23.35 | Carp1 | # |
| 02:23.35 | Carp1 | require "/root/downloads/phpagi-1.12/phpagi.php"; |
| 02:23.35 | ariel_ | peer is for outgoing calls user is for inbound and yes mailbox is for user or friend |
| 02:23.37 | Carp1 | that line? |
| 02:24.06 | ariel_ | yes |
| 02:24.21 | *** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com) |
| 02:25.02 | Carp1 | chmod to 755, right? |
| 02:25.20 | bkw_ | php for AGI....ewwwwwwww |
| 02:25.28 | steven_ | corp, no |
| 02:25.35 | steven_ | 755 is executable. |
| 02:25.39 | steven_ | 644 |
| 02:25.44 | bkw_ | php needs to stay on the web where it belongs |
| 02:25.45 | drbrown | Weezy: are you only able to use the intercom with the SPA-841 with the development version of asterisk? |
| 02:25.57 | bkw_ | :P |
| 02:26.12 | sean | bah. PHP is primarily for web, but is a fine general-purpose scripting language. |
| 02:26.14 | steven_ | I think I may buy one of these polycom phones |
| 02:26.15 | Carp1 | It says permission denied |
| 02:26.21 | *** join/#asterisk voip0 (~orwall@ottawa-hs-209-217-83-86.d-ip.magma.ca) |
| 02:26.23 | Carp1 | with 664 |
| 02:26.38 | steven_ | are you executing it from teh cmdn line? |
| 02:26.48 | steven_ | bleh |
| 02:26.52 | steven_ | ok, then use 755 |
| 02:26.59 | voip0 | Good morning / good evening |
| 02:27.05 | Carp1 | in command line it still says permission denied with 755 |
| 02:27.22 | Carp1 | but when called from extensions.conf it doesnt, it just doesnt stream the file. |
| 02:27.23 | steven_ | look at the location of the script in the first line |
| 02:27.29 | steven_ | it's probably incorrect |
| 02:27.41 | steven_ | mountie: the path to the PHP binary |
| 02:28.01 | Carp1 | Oh, wait....its not denied from command line |
| 02:28.18 | *** join/#asterisk santiago (~santiago@63.245.86.187) |
| 02:28.29 | Carp1 | # |
| 02:28.29 | Carp1 | #!/usr/bin/php -q |
| 02:28.32 | Carp1 | thats in my scrit |
| 02:28.34 | Carp1 | script* |
| 02:28.47 | Carp1 | [root@VOICE agi-bin]# whereis php |
| 02:28.48 | Carp1 | php: /usr/bin/php /etc/php.d /etc/php.ini |
| 02:29.09 | Carp1 | You know what |
| 02:29.19 | Carp1 | phpagi.php isnt chmod right |
| 02:29.47 | steven_ | heh |
| 02:30.03 | steven_ | I'M OUT! |
| 02:30.04 | Carp1 | But it still isnt working in asterisk :-\ |
| 02:30.08 | Carp1 | it says its completed |
| 02:30.12 | Carp1 | returned 0 |
| 02:30.16 | voip0 | hello, I've been able to connect to Digium using 500, how do I dial 17009999613 using a SIP phone? |
| 02:30.18 | Carp1 | but it doesnt stream the file |
| 02:30.58 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 02:31.54 | sean | Carp1: become the user that asterisk is running as: su - asterisk |
| 02:32.08 | drbrown | can anyone assist with setting up the intercom on the spa-841? |
| 02:32.21 | Carp1 | Right now asterisk is running as root |
| 02:32.27 | voip0 | Is there a test number that can be phone that after hangup will return a call to test my dialplan? |
| 02:32.28 | Carp1 | and I am on the other terminal also |
| 02:32.36 | sean | then try to execute the script: ./scriptname.php |
| 02:32.52 | Carp1 | I did. |
| 02:33.15 | Carp1 | http://pastebin.ca/10791 |
| 02:34.03 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 02:35.22 | puowvip | whomp |
| 02:36.18 | mbishop | can kphone be used for iax to fwd? |
| 02:36.45 | mbishop | if not, any good softphones that use iax? or can do iax to fwd? |
| 02:37.02 | sean | mbishop: don't see why kphone wouldn't work. |
| 02:37.16 | sean | but I've never used it (-: |
| 02:37.23 | sean | it's plain IAX, AFAIK |
| 02:37.35 | Carp1 | http://gmvs.pastebin.ca/10875 |
| 02:38.13 | *** join/#asterisk voip0 (~orwall@ottawa-hs-209-217-83-86.d-ip.magma.ca) |
| 02:38.14 | Carp1 | no stream played. |
| 02:38.21 | mbishop | sean: nah kphone wants sip |
| 02:38.29 | sean | ah..sorry. |
| 02:38.40 | mbishop | sean: but it may be able to configure it like fwd to use the iax2 proxy |
| 02:40.00 | Carp1 | I guess no one knows why my AGI isn't working. |
| 02:40.23 | sean | Carp1: if you Playback(demo-thanks) in your dialplan, does it work? |
| 02:40.30 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 02:41.33 | Carp1 | Hold on |
| 02:42.37 | Carp1 | May 3 23:55:15 WARNING[26218]: pbx.c:1646 pbx_extension_helper: No application ') Playback' for extension (local, 101, 1) |
| 02:42.37 | Carp1 | <PROTECTED> |
| 02:42.38 | Carp1 | Nope |
| 02:43.09 | sean | looks like a syntax problem.. what does your Playback line look like? |
| 02:43.14 | sean | (in extensions.conf) |
| 02:43.21 | Carp1 | Whoops |
| 02:43.24 | Carp1 | I messed it up |
| 02:43.48 | Carp1 | Yeah, it works when I do this. |
| 02:43.59 | sean | hmm.. well, that's all I got. (-: |
| 02:44.06 | Carp1 | :( |
| 02:44.11 | sean | but it seems the AGI script is working properly. |
| 02:44.18 | sean | and you know that the file is playable |
| 02:44.45 | Carp1 | Yeah, it just isnt playing back the file |
| 02:44.55 | Carp1 | I wish i was still using the first version, I think it was better. |
| 02:51.44 | Carp1 | I got it working! |
| 02:51.59 | kimo_sabe | Carp1: what was the trick? |
| 02:52.04 | Carp1 | phpagi.conf |
| 02:52.17 | Carp1 | I had to make that file in /etc/asterisk |
| 02:52.22 | sean | ah |
| 02:52.24 | Carp1 | More more question. |
| 02:52.30 | sean | nice work (-: |
| 02:52.31 | Carp1 | when I call a file to be streamed |
| 02:52.45 | Carp1 | I jsut do agi_play("file") |
| 02:52.53 | blitzrage | "The Dial() application consists mostly of a bird, similar to the simpson's episode where the bird 'pecks' the keys on a telephone. The order of the keystrokes is predefined by the 'speeddial()' application and can be preconfigured using a 'mash' sequence if so desired".... |
| 02:52.55 | Carp1 | but I downloaded hte extra sounds from CVS |
| 02:53.05 | Carp1 | should I move them all to the asterisk sound dir? |
| 02:53.36 | kimo_sabe | Carp1: the ones you want to use at least |
| 02:53.43 | Carp1 | Ok. |
| 02:53.44 | Carp1 | Thans |
| 02:53.46 | Carp1 | Thanks* |
| 02:54.16 | Carp1 | comments in config files are with a ; or a # |
| 02:54.17 | Carp1 | ? |
| 02:55.20 | Carp1 | wait |
| 02:55.22 | Nethab | Anyone hear about Uniden's UIP1868 |
| 02:55.25 | Carp1 | That may have not worked |
| 02:55.38 | Carp1 | Becuase I didnt change my extensions file back to the agi script |
| 02:55.41 | Carp1 | Its still on Playback |
| 02:55.42 | Carp1 | damnit |
| 02:56.13 | Carp1 | Nope. |
| 02:56.15 | Carp1 | Doesnt work. |
| 02:56.22 | Weezey | Carp1: ; |
| 02:56.46 | *** join/#asterisk KristinG (~KristinG@muppet.geekgirls.us) |
| 02:56.51 | Carp1 | I wish someone used PHPAGI so they could tlel me the trick. |
| 02:57.02 | voip0 | hello I've setup Asterisk but I've not been able to make a call yet other then 500 asterisk demo? I have this in my extension.conf "exten => _1700NXXXXXX,1,Dial(IAX2/orwall:secret@iaxtel.com/${EXTEN}@iaxtel)" I would like to phone 17009999613 an echo test number? do I have to add an extra digit or something? |
| 02:57.03 | KristinG | hi |
| 02:57.08 | voip0 | hello |
| 02:57.14 | KristinG | any extensions gurus here? |
| 02:57.48 | Weezey | voip0: get rid of that last @iaxtel and see what happens. |
| 02:57.57 | Weezey | KristenG: what's your problem? |
| 02:58.18 | voip0 | thanks Weezey |
| 02:58.20 | *** join/#asterisk salviadud (~dude@201.129.86.120) |
| 02:58.30 | KristinG | I have a few carriers |
| 02:58.39 | Weezey | as you should. |
| 02:58.50 | KristinG | and I want to be able to try them in order if one cannot complete a call |
| 02:58.59 | Weezey | no problem. |
| 02:59.02 | wildcard0 | so just do one and then the next |
| 02:59.15 | salviadud | hey, im using a sipura 3000, do i need to do a zapata.conf file if i want to get my pstn line calling from line 1? |
| 02:59.29 | Weezey | salviadd: nope |
| 02:59.35 | KristinG | weezy there needs to be some sort of if rule |
| 02:59.35 | kimo_sabe | salviadud: no, it's a SIP device |
| 02:59.37 | wildcard0 | salviadud, no |
| 02:59.37 | KristinG | ? |
| 02:59.40 | Weezey | Sipura 3000 is only sip. |
| 02:59.56 | salviadud | but, its got a fxs and fxo port |
| 03:00.00 | kimo_sabe | nope |
| 03:00.05 | Weezey | Kristin: for routing based on the number dialed. |
| 03:00.13 | wildcard0 | KristinG, no, if a dial statement fails, it goes to the next priority. just add the next dial on the next line |
| 03:00.14 | *** part/#asterisk santiago (~santiago@63.245.86.187) |
| 03:00.15 | kimo_sabe | salviadud: and they're both SIP enpoints (UA's I think) |
| 03:00.16 | Weezey | salivadd: they register seperately in sip.conf |
| 03:00.32 | Weezey | kimo: no, one's FXO one FXS |
| 03:01.02 | kimo_sabe | Weezey: yes, SIP doesn't care |
| 03:01.07 | salviadud | so, i can get calls from the pstn line, and maybe transfer them over IAX to someone in who knows where |
| 03:01.08 | KristinG | i tried that today and it failed |
| 03:01.12 | kimo_sabe | Weezey: I have one of them |
| 03:01.23 | wildcard0 | KristinG, what was wrong with it? |
| 03:01.26 | salviadud | but i can't use my sipura and asterisk to dial to the pstn line? |
| 03:01.32 | salviadud | it should be possible... |
| 03:01.35 | Weezey | it is. |
| 03:01.41 | Weezey | I have three set up right now. |
| 03:01.43 | wildcard0 | salviadud, ya you can do that |
| 03:01.59 | salviadud | ah... that makes me feel better |
| 03:02.01 | KristinG | it fails on the first carrier and goes to reorder |
| 03:02.22 | salviadud | im reading the vol1 documentation, i just registered to iaxtel |
| 03:02.34 | salviadud | im a newbie, but im learning! |
| 03:03.03 | salviadud | well, thanx for the info, im gonna read some more |
| 03:03.08 | wildcard0 | KristinG, how does it fail? what does the carrier return? |
| 03:03.55 | Weezey | salviadud: the forums on http://voxilla.com were a lot of help |
| 03:04.00 | syle | what is iaxtel |
| 03:04.10 | Weezey | a provider |
| 03:04.17 | KristinG | brb |
| 03:05.04 | stormfr | is anybody update to last cvs ? seems pri is now don't working |
| 03:09.21 | kb1_kanobe | last nights libpri and zaptel from head are working fine for me. |
| 03:10.07 | stormfr | i move back to 04/30/05 and work again |
| 03:16.58 | Hogie | has anybody ever seen a cisco 7960 act funny running sip 6.3 in the regard that it starts doing random numbers when you dial (it happens when we dial 98 [817 is a local area code]). It happens on 3 diff phones I put at the same location, but dont happen at other locations... changed out power blocks, network cables, etc, I dunno:( |
| 03:18.24 | stormfr | hogie : i have around 50 and use long time 6.3 and never see this. now running 7.3 or 7.4. I have see funny thing with 7912 but it's not the same sip firmware |
| 03:19.41 | Hogie | so you ran 6.3 for a long time? I dont get it, its at our 2nd office, and it only happens to 1 person, I can even switchout phones to that person, and it stays there, but *I* caused the problem to happen today, when she wasn't there |
| 03:19.54 | Hogie | so I know it isn't PEBCAK |
| 03:20.55 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:20.57 | stormfr | do you have check full log ? maybe a bad sip message |
| 03:21.11 | niZon | http://www.rafb.net/paste/ <- pastebin alternative |
| 03:21.17 | niZon | has syntax highlighting too |
| 03:21.19 | drbrown | can anyone assist in setting up an spa-841 on an intercom setup? |
| 03:21.33 | Hogie | its before it does SIp messages, it isn't timing out yet |
| 03:26.43 | *** part/#asterisk KristinG (~KristinG@muppet.geekgirls.us) |
| 03:34.26 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 03:34.57 | *** join/#asterisk candyz_u8 (~candyz_u8@220-132-176-126.HINET-IP.hinet.net) |
| 03:35.59 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 03:36.21 | *** join/#asterisk Synapse-_ (~pnats@c211-30-74-249.belrs2.nsw.optusnet.com.au) |
| 03:36.34 | Carp1 | My AGI not offically works. |
| 03:36.37 | Carp1 | now* |
| 03:39.33 | Sato1 | what it does? |
| 03:40.26 | sean | wow, call files are cool (-: |
| 03:43.20 | outtolunc | call files are NOT cool <G> |
| 03:43.36 | sean | hmmph. |
| 03:43.37 | outtolunc | that is if they never get 'answer' or 'failed' <G> |
| 03:44.00 | sean | 'me laughs at file and his later-than-everyone-else timezone. |
| 03:44.11 | file | yeah, almost 1 |
| 03:44.12 | file | I should sleep |
| 03:44.18 | outtolunc | (meaning spooled files that never get 'answer'd or 'failed' never get DIALSTATUS <G> |
| 03:44.21 | outtolunc | ) |
| 03:45.01 | sean | ah, I was wondering what happens in that case. |
| 03:45.03 | nestAr | file is in n00b runswick? |
| 03:45.22 | sean | file: you have VOIP termination in 506? |
| 03:45.25 | outtolunc | all you get is the spool 'reason' code |
| 03:45.29 | file | sean: I wish |
| 03:45.34 | nestAr | :ohsnaps: |
| 03:45.35 | outtolunc | which isn't a global |
| 03:45.41 | file | which reminds me, I should ask Aliant how much for a PRI! |
| 03:46.00 | file | ten bucks says their answer is, "what's a PRI?" |
| 03:46.21 | file | darn |
| 03:46.27 | outtolunc | takes that bet |
| 03:46.38 | file | outtolunc: you're just desperate for money |
| 03:46.51 | outtolunc | (might get my new chair after alL <G>) |
| 03:46.54 | file | haha |
| 03:46.55 | outtolunc | er all |
| 03:47.02 | sean | when I left, in 2000, a data T1 was >$1500/month (anecdotally). |
| 03:47.36 | outtolunc | data T1's are priced seriously different from voice t1s and pri's |
| 03:47.49 | nestAr | yeah, they're generally cheaper |
| 03:47.51 | nestAr | :) |
| 03:47.56 | nestAr | at least in my region |
| 03:47.57 | sean | .. I had NBNet dialup service in the early-mid 90s. $6/minute at peak hours. |
| 03:48.06 | file | ah nbnet, hehe |
| 03:48.11 | file | I remember fundy stuff |
| 03:48.14 | outtolunc | the only time they 'appear' similar is if you live in bum-f-e..... |
| 03:48.16 | sean | ah fundy. |
| 03:48.17 | file | fundy cable systems, then shaw got it, then rogers |
| 03:48.23 | sean | cable modem early adopters |
| 03:48.28 | nestAr | mmmm.. |
| 03:48.38 | sean | with the upstream on a modem that may or may not sync with your downstream |
| 03:48.41 | outtolunc | and that's due to the 'distance' factor |
| 03:48.55 | nestAr | actually, my pri's cost about the same as most of our PtP t1's |
| 03:48.59 | drbrown | Has anyone installed the webmin module on the ftp site? |
| 03:49.10 | file | drbrown: hasn't been updated in ages, doesn't work, give up now |
| 03:49.13 | nestAr | we pay $400/mo per pri |
| 03:49.24 | drbrown | ok |
| 03:49.32 | drbrown | thought it would be nice for my customers |
| 03:49.51 | outtolunc | (nestAr would shit if he knew what we pay for 'most' of the loops on our voice t1's) |
| 03:49.54 | drbrown | file: have you used the spa-841 in an intercom setup? |
| 03:50.06 | file | drbrown: nah, I'm not an SPA-841 user |
| 03:50.21 | drbrown | file: do you use snom? |
| 03:50.34 | outtolunc | the exception is the one that is long hauled for over 500 miles on mci <G> |
| 03:50.55 | wvbroadband | I know they're going through some big upgrade, but anyone been able to get ahold of nufone support lately? |
| 03:51.05 | wvbroadband | I've tried, I'd say 20 times |
| 03:51.14 | file | I have a Cisco myself |
| 03:51.25 | outtolunc | new account, or existing? |
| 03:51.31 | outtolunc | (nufone that is) |
| 03:51.40 | sean | (sortof (-; ) |
| 03:51.41 | wvbroadband | existing |
| 03:51.41 | jakepdev | wvbroadband - there are nufone guys on here most of the time and #nufone |
| 03:51.45 | file | I can't keep my eyes open |
| 03:51.59 | sean | 'night file. Enjoy your muddy river. |
| 03:52.08 | outtolunc | strange, existing accounts (i had an issue last week) he helped me with |
| 03:52.18 | sean | speaking of.. go have a pint of Muddy River Stout at Pumphouse.. |
| 03:52.23 | wvbroadband | oh, I've been waiting for the elusive JerJer to sign on |
| 03:52.38 | jakepdev | ~seen JerJer |
| 03:52.43 | jbot | jerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 5d 22h 28m 33s ago, saying: 'try ulaw or gsm'. |
| 03:52.57 | file | sean: I'm not old enough to drink yet lol |
| 03:52.57 | jakepdev | did you try ulaw or gsm? |
| 03:53.07 | sean | file: oh yeah.. heh.. forgot it's 19 in NB |
| 03:53.20 | file | yeah don't remind me |
| 03:53.20 | sean | well, when you turn, that's the place to go, if you like good beer. |
| 03:53.27 | outtolunc | file, let the glow of the lcd fade <G> you are getting SLEEPY.. SLEEPy.. SLEepy.. SLeepy.. etc |
| 03:53.39 | wvbroadband | anyone else here work for nufone |
| 03:54.00 | syle | whats the best cordless phone |
| 03:54.03 | file | gah goodnight |
| 03:54.29 | outtolunc | hehe |
| 03:54.30 | syle | i think only uniden makes cordless's right? |
| 03:54.55 | outtolunc | syle, you are joking right? |
| 03:55.04 | wvbroadband | syle: one of those illegal (in the US, and FCC regulated areas) imports that work for miles |
| 03:55.15 | sean | so, back to call files.. what happens when the retries expire without a connection? |
| 03:55.39 | wvbroadband | I once knew a guy who lived on a mountaintop and realistically could travel 20 miles away and still place calls |
| 03:55.46 | outtolunc | obviously, the call file disapprears |
| 03:55.59 | sean | and is there any indication of the result? |
| 03:56.07 | outtolunc | so, either trap the 'reason' and regen, or let it go |
| 03:56.15 | syle | i haven;t seen any voip cordless phones cept uniden |
| 03:56.20 | sean | how would I trap the reason? |
| 03:56.26 | sean | logs/console only? |
| 03:56.36 | outtolunc | look at pbx_spool.c you will see a res = ... |
| 03:57.39 | outtolunc | remember what i said.. unless xyz happens.. (it doesn't get to ast_pbx_run, and wasn't an 'answer' or 'failed' it doesn't get DIALSTATUS) |
| 03:57.52 | outtolunc | so that's out |
| 03:58.03 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 03:58.06 | vpp | hi guys |
| 03:58.07 | outtolunc | meaning you will have to trap the res in pbx_spool |
| 03:58.16 | vpp | anyone use oh323? |
| 03:58.17 | outtolunc | enough said |
| 03:58.31 | sean | outtolunc: ah. I see your point, now. That DOES suck. |
| 03:58.50 | outtolunc | sean, not much if you can code |
| 03:59.02 | sean | my C skills are.. lacking. |
| 03:59.12 | outtolunc | but, those updates will not be 'core' |
| 03:59.30 | sean | are there plans for a more robust call API? |
| 03:59.46 | *** join/#asterisk matthewa (~matthewa@S010600121701a8b2.vc.shawcable.net) |
| 04:01.56 | sean | maybe this is too naive, but souldn't it make sense to have a /var/spool/asterisk/outgoing/results ? |
| 04:02.15 | sean | s/souldn't/wouldn't/ |
| 04:02.58 | Juggie | sean, what asterisk needs mostly is a better event system |
| 04:03.01 | Juggie | and its in the works |
| 04:03.05 | sean | cool. |
| 04:03.05 | Juggie | that ould solve some of these issues |
| 04:03.20 | Juggie | the guy who was working on it was just hired by digium. |
| 04:03.27 | sean | .. I'm really new to this. |
| 04:03.48 | Hogie | wow, nufone has saved us $11 this week so far |
| 04:03.54 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 04:03.54 | *** mode/#asterisk [+o twisted] by ChanServ |
| 04:04.26 | drbrown | Does anyone know how to use the CALL_INFO variable? |
| 04:05.58 | *** part/#asterisk matthewa (~matthewa@S010600121701a8b2.vc.shawcable.net) |
| 04:12.54 | outtolunc | a /var/spool/asterisk/results is just plain wrong, more filehandles and resources is not that way |
| 04:13.24 | outtolunc | i'd suggested an ast_state in channel stuct for it |
| 04:13.53 | outtolunc | (as noted previously, HELL there is already an moh_state) <G> |
| 04:15.29 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 04:15.55 | outtolunc | but then, that's just me <G> |
| 04:16.12 | outtolunc | some people think the world exists in 'dial' |
| 04:22.49 | niZon | eep |
| 04:22.49 | kb1_kanobe | huh... there goes the neighbourhood. |
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| 04:25.05 | blankman | Hey all. I don't suppose any of the guys from NuFone are lurking? |
| 04:31.08 | outtolunc | so WilliamK: were you attempting to ask about a certain 'type' of traffic TO or FROM mexico? |
| 04:31.17 | Nethab | hey WilliamK the new sipura firmware isn't available for the 3000 i'm so dissapointed |
| 04:31.55 | outtolunc | you mentioned 'to' just asking since the rest was vague |
| 04:32.49 | Sato1 | WilliamK, i guess iconnect |
| 04:32.58 | Sato1 | i use it even being in mexico |
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| 04:36.04 | Sato1 | and i anwered an our later, *smirsk* |
| 04:36.11 | Sato1 | s/our/hour/ |
| 04:37.53 | outtolunc | that or i just an being too general |
| 04:40.14 | outtolunc | regardless, of previous, i'd still like to help people with real questions |
| 04:40.37 | vpp | anyone here use oh232? |
| 04:40.51 | outtolunc | never heard of it |
| 04:40.57 | kb1_kanobe | I'm just trying out kernel 2.6.11 with Ingo Molinars realtime patches applied in a quest to consistent 100% score from zttest... but it doesn't seem to have made any material difference. This surprises me. Any suggestions? |
| 04:41.29 | outtolunc | kb1: you meant the zttest-mod ? |
| 04:41.42 | kb1_kanobe | I was just using zttest for now. |
| 04:42.10 | outtolunc | the tests they were 'referencing' was a mod that was named 'zttest-mod' |
| 04:42.32 | outtolunc | so if you are using zttest responses they will NOT equate |
| 04:43.05 | kb1_kanobe | errr...? I was only comparing between my own machine. Not to the discussion on -users. There seems to be some consunsion with the math in that. |
| 04:43.14 | outtolunc | ah |
| 04:43.16 | kb1_kanobe | s/consunsion/confusion. |
| 04:43.30 | outtolunc | ok elaborate |
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| 04:44.03 | kb1_kanobe | Well, I got to thinking that if there are jitters in the expected delivery of data to/from the t1 card then perhaps the kernel is part of the issue. |
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| 04:44.18 | outtolunc | basically you mentioned you patched, but it didn't help.. it didn't help how? |
| 04:44.20 | kb1_kanobe | there has been much made recently of 2.6.11 for real time audio work. |
| 04:45.01 | kb1_kanobe | well, I still get 'misses' intermittantly and I'm still seeing plenty of EAGAINs in the console, accompanied by pops&clicks on the PRI. |
| 04:45.03 | outtolunc | the whole point of the zttest-mod was too determine MB/kernel |
| 04:45.14 | outtolunc | so you are doing a 'side test' |
| 04:45.26 | kb1_kanobe | correct. I'm ignoring zttest-mod for now. |
| 04:45.33 | outtolunc | using existing methods that are already found to be 'non-effective' |
| 04:45.45 | kb1_kanobe | non effective how? |
| 04:45.47 | outtolunc | (as a real test) |
| 04:46.11 | outtolunc | because the base code looks at it 1024 wide |
| 04:46.18 | outtolunc | not 1000 wide |
| 04:46.37 | outtolunc | as noted in the emails regarding the zttest-mod |
| 04:47.08 | outtolunc | are you sure you don't want to join 'the other' and test as they are? |
| 04:47.14 | outtolunc | er the others |
| 04:47.21 | kb1_kanobe | does it really matter in the sense of zttest? It simply compares the expected samples in a sample window so, so long as it goes for 8192 and is expecting 8192 in slightly > 1 second, all is well, no? |
| 04:47.58 | kb1_kanobe | ta. |
| 04:47.59 | kb1_kanobe | :-) |
| 04:48.10 | kb1_kanobe | alrighty then. I'm running zttest-mod |
| 04:48.48 | outtolunc | simply put, for a test to mean anything, there must be other data, that other data must be from the same spec |
| 04:49.01 | outtolunc | IF NOT IT"S WORTHLESS |
| 04:49.16 | kb1_kanobe | For comparison purposes, yes. |
| 04:49.44 | outtolunc | so, since 'all the coders' are doing zttest-mod.. if you want to participate.. i'd suggest using that |
| 04:49.54 | kb1_kanobe | point taken. |
| 04:50.39 | outtolunc | ty |
| 04:51.29 | kb1_kanobe | which gets me back to my original point - isn't zttest-mod itself part of the possible issue if it's not running as a realtime process? |
| 04:52.11 | outtolunc | it's an 'issue in progress' they already know 'something is amiss' in the quantification |
| 04:52.48 | outtolunc | so, even if the 'current test' is amiss, at least 'that' is the one they are using to deal with the issue |
| 04:53.12 | outtolunc | so any mods are to 'it' and not something not even being used |
| 04:54.04 | outtolunc | would you try and send speed tests to a company that said we are using 'xyz' for speed tests.. when yours was not |
| 04:54.22 | outtolunc | some might.. |
| 04:54.36 | kb1_kanobe | nope. however i'm just comparing between my various * boxes and don't intent to share results from a modified benchmark. :-) |
| 04:54.37 | outtolunc | but those would be dealt with similarly <G> |
| 04:55.03 | outtolunc | ok, lets go on just zttest |
| 04:55.20 | kb1_kanobe | ? |
| 04:55.21 | outtolunc | and the differerces you have within your own boxes |
| 04:55.31 | outtolunc | provide the data |
| 04:55.41 | outtolunc | i'll wait |
| 04:56.29 | outtolunc | (i'm gonna go get a drink, but i'll be right back) |
| 04:57.16 | *** join/#asterisk tomassia (~user@203.131.164.162) |
| 04:57.50 | kb1_kanobe | no worries. |
| 04:57.52 | kb1_kanobe | Box A (2.6.8-2-686 from debian testing) 17 active channels, 15 are t100p, 3 IAX2 --- Results after 356 passes --- Best: 1.026179 -- Worst: 1.022098 -- Average: 1.024011 real 6m5.079s user 0m0.012s sys 0m0.044s |
| 04:58.08 | kb1_kanobe | Box A (2.6.8-2-686 from debian testing) 17 active channels, 15 are t100p, 3 IAX2 --- Results after 356 passes --- Best: 1.026179 -- Worst: 1.022098 -- Average: 1.024011 real 6m5.079s user 0m0.012s sys 0m0.044s |
| 04:58.33 | tomassia | my xlite cannot register to the asterisk server, what should i do? |
| 04:58.35 | kb1_kanobe | Box B (2.6.11-realtime) 18 active channels, 9 are t100p, 9 IAX2 --- Results after 164 passes --- Best: 1.024175 -- Worst: 1.023850 -- Average: 1.024017 real 2m48.867s user 0m0.009s sys 0m0.036s |
| 04:58.40 | kb1_kanobe | ,even. |
| 05:00.17 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 05:00.32 | kb1_kanobe | but it's difficult to compare as the results aren't displayed in terms of frequency, however I'm surprised box B is not bang on all the time given the kernel it's runing in. |
| 05:02.21 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
| 05:02.55 | tomassia | how can i register my xlite to the asterisk server |
| 05:03.05 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
| 05:04.33 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 05:04.48 | menger | anyone had a snom phone which goes to music on hold but can't get it back again? |
| 05:09.21 | outtolunc | scrolling back, hey menger |
| 05:09.27 | menger | hi otl |
| 05:10.07 | outtolunc | 17 active channels, 15 are t100p, 3 IAX2 ?? |
| 05:10.16 | menger | huh? |
| 05:10.33 | outtolunc | that was from kb1's paste before you got here |
| 05:10.52 | kb1_kanobe | Sorry, I've got load running on the machine, otherwise the test results tend to be spot on (at least from the old zttest perspective) |
| 05:11.14 | outtolunc | i was just talking about the # |
| 05:11.16 | outtolunc | <G> |
| 05:11.44 | kb1_kanobe | forget that realtime influence consideration - I just patched it up to see and it makes no difference. |
| 05:11.56 | outtolunc | i'm 'assuming' that meant there was 1 'inactive' iax channel that for 'some' reason was pulled into the equation <G> |
| 05:12.35 | kb1_kanobe | there is one pstn b channel left open just incase. Someone must have been on it. |
| 05:12.57 | kb1_kanobe | errr.. forget I said that. |
| 05:13.01 | outtolunc | <G> |
| 05:13.30 | kb1_kanobe | it's that beer you gave me talking. |
| 05:14.04 | outtolunc | yeah, but i didn't expect you to feed it to your * box <G> |
| 05:14.22 | kb1_kanobe | I try to keep it happy - that way it keeps me happy! |
| 05:14.47 | outtolunc | obviously the test (if that is actual output) is flawed <G> |
| 05:15.23 | kb1_kanobe | how so? you mean the fact it's > 1.0 seconds? |
| 05:15.34 | outtolunc | no it's inability to count |
| 05:15.45 | kb1_kanobe | ah. |
| 05:15.45 | outtolunc | 17 active channels, 15 are t100p, 3 IAX2 |
| 05:15.52 | outtolunc | knock knock |
| 05:16.12 | kb1_kanobe | I'll just write a quick patch for that... brb. |
| 05:16.15 | outtolunc | not only once, but twice <G> |
| 05:16.18 | kb1_kanobe | <duh> |
| 05:16.37 | outtolunc | k |
| 05:16.48 | *** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com) |
| 05:17.18 | outtolunc | (notes: i really want to help, and i'm sorry if my backward-country-ass brain just can't get around it) <G> |
| 05:17.48 | outtolunc | i grewup on a farm |
| 05:18.12 | outtolunc | if the tractor in not infront of the plow, it does no good <G> |
| 05:18.54 | outtolunc | (lord knows we used it for many other things) <G> |
| 05:20.35 | *** join/#asterisk nixter (~nick@OCI-19-41.OneCall.Net) |
| 05:21.00 | drbrown | does anyone know if they are going to port the SIPaddheader app to the current stable release anytime soon? |
| 05:21.13 | drbrown | or at all? |
| 05:21.59 | outtolunc | that would be a question to post to the bugtracker, that way the 'maintainer' can inform you/us of it's travels |
| 05:22.58 | outtolunc | all, backports are the joy/bliss of one man, he does come out to 'chat' much <G> |
| 05:23.26 | outtolunc | bugtracker is your best bet for a response is what i meant |
| 05:23.42 | drbrown | ok |
| 05:23.45 | drbrown | thanks |
| 05:23.47 | outtolunc | np |
| 05:24.09 | drbrown | these spa-841's are pissin me off |
| 05:24.22 | drbrown | so close to complete, but so far away |
| 05:24.24 | outtolunc | that sounds like a personal issue <G> |
| 05:24.47 | outtolunc | can you form it in a fashion that is 'bug' based <G> |
| 05:24.50 | drbrown | I just can't get the intercom function to work |
| 05:24.54 | *** join/#asterisk pbx123 (~nbrand@adsl-69-230-199-226.dsl.irvnca.pacbell.net) |
| 05:25.52 | outtolunc | side features of MANY phones (hard or soft) are 'usually' the issue of the phone.. buton the off chance it's proto based why not start there |
| 05:26.14 | pbx123 | Hi. I'm new to iRC as well as asterisk. Can anyone lend a hand? |
| 05:26.15 | drbrown | I wish the 480i was finished |
| 05:26.26 | outtolunc | you have a phone, it uses, proto x, it is/isn't doing y |
| 05:26.57 | outtolunc | pbx123, hold a few |
| 05:27.04 | pbx123 | thanks |
| 05:27.08 | outtolunc | (i'll get to you) |
| 05:27.35 | outtolunc | drbrown, are you going to elaborate? |
| 05:28.19 | outtolunc | if not, i'm sure pbx123 wants his turn |
| 05:28.27 | drbrown | the spa-841 is a sip based phone, and I cannot get it to auto answer |
| 05:29.02 | outtolunc | and you have obviously talked to them about why it doesn't autoanswer? |
| 05:29.19 | drbrown | it has to do with the sip header |
| 05:29.23 | outtolunc | (that IS a phone (self) issue) |
| 05:29.48 | outtolunc | ok so that phone autoanswer is tied to the proto |
| 05:29.57 | drbrown | yes |
| 05:30.06 | outtolunc | and what does that phone expect? |
| 05:30.25 | drbrown | asterisk in it's current state (stable) does not support setting the sip header |
| 05:30.41 | outtolunc | (and the following question will be what is asterisk sending/not sending that falls short) |
| 05:31.14 | drbrown | here is an example of the "hack" that's supposed to take care of the issue |
| 05:31.31 | outtolunc | well if you can... try shifting to cvs-head and see if that helps |
| 05:31.59 | outtolunc | there are 'so many' things in head that aren't in stable.. (it as they say, is not funny) |
| 05:32.02 | drbrown | xten => 80,2,SIPaddheader(Call-Info: \;answer-after=0) |
| 05:32.12 | drbrown | "exten => 80,2,SIPaddheader(Call-Info: \;answer-after=0)" |
| 05:32.37 | outtolunc | thats is beyond the proto, that is called force-feeding |
| 05:32.38 | drbrown | "exten = 80,2,SIPaddheader(Call-Info: \;answer-after=0)" |
| 05:33.27 | outtolunc | at a proto level (if it 'part of the norm') all that is not needed |
| 05:33.29 | drbrown | I need a better answer than snom is what it comes down to |
| 05:33.40 | outtolunc | no |
| 05:34.02 | drbrown | I can't use the sipura spa-841 either |
| 05:34.19 | outtolunc | i'll never say mnfr x is better than y, what i will say is that x had better interaction with the app, than y <G> |
| 05:34.25 | menger | no snom users here? |
| 05:34.33 | outtolunc | did you see the slight of hand <G> |
| 05:34.53 | drbrown | price isn't the issue, when it comes to my customers it is the looks of the phone |
| 05:34.54 | outtolunc | menger you having snom issues? |
| 05:35.06 | menger | outtolunc, yep |
| 05:35.14 | outtolunc | pbx123 i've not forgetten <G> really |
| 05:35.28 | outtolunc | pbx123 priv me. .i'll help same time |
| 05:35.29 | pbx123 | no problem, just let me know |
| 05:35.33 | drbrown | thanks take her easy |
| 05:35.37 | menger | outtolunc, i have a snom 220, i have set the music on hold server to be 197 (extension which goes into music on hold) |
| 05:36.03 | outtolunc | snom's require a set moh channel? |
| 05:36.03 | menger | however we can't get the call back after, the snom handset can here them, but you can't here the snom user |
| 05:36.11 | menger | outtolunc, seems so |
| 05:36.17 | outtolunc | not the * channel aspect? |
| 05:36.27 | outtolunc | that is outthere |
| 05:36.30 | outtolunc | but ok |
| 05:36.38 | outtolunc | so, what happens |
| 05:36.48 | pbx123 | I'm trying to see if my TE110P is DOA. I just got it. After configurations, the light on the card just doens't light up at all. |
| 05:37.14 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 05:37.22 | outtolunc | pbx123 what drivers did you load? |
| 05:37.24 | menger | the call goes on hold, remote hears music on hold, but when the operator picks up again, you still here music on hold and operator can here you |
| 05:37.29 | menger | but you can't here operator |
| 05:37.37 | outtolunc | afk a sec |
| 05:37.40 | *** join/#asterisk leibniz_ (~leibniz@200.122.157.91) |
| 05:38.25 | leibniz_ | if i have an ip phone doing 3-way calling with two callers on the pstn, will the ip phone see one or two RTP streams ? |
| 05:39.02 | menger | lecram, i suspect 2 |
| 05:39.08 | menger | leibniz_, i suspect 2 |
| 05:39.22 | pbx123 | Do I have it right, that all correct modules have to be loaded before the led can light up? |
| 05:40.29 | leibniz_ | menger: does * support dsp boards for transcoding? |
| 05:40.49 | remmo | the spa3000 seems like they are feature packed |
| 05:41.37 | menger | leibniz_, not sure |
| 05:46.04 | remmo | anyone has any hints to conf a spa3000 through a nat firewall, so i dont have to change the nat |
| 05:47.12 | outtolunc | sorry back, scrolling back |
| 05:48.27 | outtolunc | pbx123, you need the zaptel, and specific card driver loaded before it lights up |
| 05:48.56 | outtolunc | menger, are you using another else? agentlogin/callbacklogin? |
| 05:49.04 | outtolunc | er anything |
| 05:49.04 | menger | nope |
| 05:49.11 | menger | just a dial(SIP/101) |
| 05:49.23 | outtolunc | straignt 1to1 'dial'? |
| 05:49.32 | menger | yep |
| 05:49.34 | menger | no queues |
| 05:49.38 | outtolunc | compatible codecs? |
| 05:49.46 | menger | is all ulaw |
| 05:49.51 | menger | and it works before you put them on hold |
| 05:49.56 | pbx123 | great. basically, I just want to make sure the card isn't DOA just cause it doesn't light up at all. I'll keep trying, thanks. |
| 05:49.56 | outtolunc | you are sure? |
| 05:50.00 | menger | yep |
| 05:50.06 | menger | they would have had my head by now otherwise |
| 05:50.07 | outtolunc | and no firewall issues? |
| 05:50.13 | menger | and i have spoken to them on the phone too |
| 05:50.16 | menger | nope |
| 05:50.19 | menger | all local lan |
| 05:50.30 | menger | ISDN BRI <--> Asterisk <--> Snom 220 |
| 05:50.41 | outtolunc | pbx123 'cat /proc/modules' see zaptel and one for the card |
| 05:51.02 | outtolunc | if to do.. then ztcfg -vvv |
| 05:51.06 | outtolunc | THEN try |
| 05:51.48 | outtolunc | menger and yo got debug wide open, and verbose at 5 |
| 05:51.51 | pbx123 | I'm not at work where the server is, but i'll do so when I get back to work. thanks again for your help. |
| 05:52.13 | pbx123 | just want to beat the RMA deadline. good to know there is still hope for the card. |
| 05:52.13 | outtolunc | np, i'll be here tomorrow, just priv me |
| 05:52.34 | menger | outtolunc, i put sip debug on, i need to go on site and look as testing this from here is driving me nuts |
| 05:52.51 | outtolunc | 9 times out of 10 it's the lack of something that 'can' be fixed |
| 05:53.40 | outtolunc | menger i figured since it was ulaw you were using a card to get out |
| 05:53.47 | outtolunc | yes/no? |
| 05:53.49 | menger | outtolunc, yep, i am |
| 05:53.53 | menger | bri card |
| 05:53.58 | menger | with zaptel driver |
| 05:53.59 | outtolunc | ah |
| 05:54.17 | outtolunc | then hit the console 'set verbose 5' |
| 05:54.33 | outtolunc | make sure debug is set in logger.conf |
| 05:54.42 | outtolunc | and run |
| 05:54.59 | outtolunc | watch the console and /var/log/asterisk/debug |
| 05:55.30 | outtolunc | see if ANY thing if outof place, especially 'incompatable codec' |
| 05:55.51 | outtolunc | and it transcodes on the fly regardless |
| 05:56.32 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 05:56.38 | outtolunc | you have been around this long enough i feel like i suggesting the obvious |
| 05:56.47 | outtolunc | in which case, i don't mean too |
| 05:57.11 | menger | asterisk*CLI> sip show channels |
| 05:57.11 | menger | Peer User/ANR Call ID Seq (Tx/Rx) Format |
| 05:57.11 | menger | 192.168.25.248 101 3c2680fb337 00101/00002 ulaw |
| 05:57.11 | menger | 192.168.25.248 101 3c267ab6114 00101/00006 ulaw |
| 05:57.11 | menger | 192.168.25.248 101 3c26729883e 00101/00002 ulaw |
| 05:57.11 | menger | 192.168.25.248 101 3c2672020eb 00101/00002 ulaw |
| 05:57.13 | menger | 192.168.25.248 (None) 3c2cf53c081 00103/00005 unknow |
| 05:57.38 | outtolunc | and junior on the end is what? |
| 05:57.59 | outtolunc | hell he doesn't even know how to spell <G> |
| 05:58.20 | outtolunc | (assuming a chop, i was being funny) <G> |
| 05:59.14 | outtolunc | and as such i'm assuming you are using mac as 'user' |
| 06:00.16 | outtolunc | what device is a '3c2cf5'. no i don't feel like querying the db |
| 06:01.17 | outtolunc | simple test, if you remove that device, do things go back to norm |
| 06:01.34 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 06:01.36 | FuriousGeorge | hey all. i just installed a home intercom type system using asterisk softphone and a few fxs. not hard, it has a simple dialplan for outbound calls. |
| 06:01.56 | FuriousGeorge | tomarrow i wanna install an "asterisk bridge" at an office at work |
| 06:02.37 | FuriousGeorge | they spend 300/mo using the telco sercvice to foreward calls. so i got a few fxo for that |
| 06:03.22 | lehel | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 06:03.22 | lehel | Did you forget that FXS interfaces are configured with FXO signalling |
| 06:03.23 | lehel | and that FXO interfaces use FXS signalling? |
| 06:03.40 | lehel | << tells me the "ztcfg"... |
| 06:03.45 | Nethab | ~seen kram |
| 06:03.47 | jbot | kram is currently on #asterisk (3d 6h 23m 28s). Has said a total of 139 messages. Is idling for 1d 2h 10m 54s |
| 06:03.53 | lehel | where's the problem? |
| 06:04.04 | FuriousGeorge | the point is: do i need to do advanced things like call parking, or will it be as simple as: answer, and call the office with people in it |
| 06:04.13 | outtolunc | lehel, that is a very good message, can you re-read it |
| 06:04.30 | outtolunc | (too yourself) |
| 06:04.47 | lehel | ok.. but i don't know wich .conf file i need to change! |
| 06:05.19 | lehel | zapata.conf? .. zapata-channels.conf ?? |
| 06:05.19 | outtolunc | well both zaptel.conf and zapata.conf have 'signalling' right? |
| 06:05.22 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 06:05.31 | outtolunc | look closer |
| 06:06.16 | lehel | the problem is that i don't know wich Signalling method i should use |
| 06:06.33 | outtolunc | well what card do you have |
| 06:06.35 | lehel | in zapata.conf > fxs_ks (now) |
| 06:06.48 | lehel | Wildcard TDM400 |
| 06:07.19 | outtolunc | and you most certainly looked at the info on digium right? |
| 06:08.31 | outtolunc | http://www.digium.com/index.php?menu=configuration |
| 06:09.05 | outtolunc | (feels like saying 'next' <G>) |
| 06:09.29 | outtolunc | so FG, what is your boggle? |
| 06:09.51 | FuriousGeorge | i just have a theoretical question |
| 06:10.05 | outtolunc | you installed 'asterisk intercom' and want to get 'asterisk bridge' going.. right? |
| 06:10.19 | FuriousGeorge | its wierd to ask. i wanna install an asterisk bridge to avoid using telco pay-per-use forewarding service |
| 06:10.20 | outtolunc | sadly, i've never heard of them |
| 06:11.11 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 06:11.17 | outtolunc | i'm 'assuming' you simply want to forward you calls 'throught' an account (pstn/term provider) |
| 06:11.26 | outtolunc | er -t |
| 06:11.36 | FuriousGeorge | i just want it to answer a call after a given amount of time (not the hard part), then use the opposite line to call another local office |
| 06:11.58 | outtolunc | well here is you first lesson |
| 06:12.05 | FuriousGeorge | i dont need call parking or nothing right? i can just wait, answer the call on s, then call the office from the opposite line |
| 06:12.07 | outtolunc | there are 2 sides to telecom |
| 06:12.13 | outtolunc | inbound, and outbound |
| 06:12.17 | FuriousGeorge | uhuh |
| 06:12.21 | outtolunc | they ARE separate |
| 06:12.35 | FuriousGeorge | thats what i was afraid of ;) |
| 06:12.39 | outtolunc | if you want to 'join' them you have to get fancy |
| 06:12.51 | FuriousGeorge | no fancy! |
| 06:12.55 | outtolunc | fancy is with asterisk simple |
| 06:13.06 | FuriousGeorge | phew |
| 06:13.33 | FuriousGeorge | the point of my mentioning the home intercom thing. it only involves fxs though |
| 06:13.34 | outtolunc | you simply need to 'answer' the inbound (on one channel) and dial on aother (channel) |
| 06:14.02 | outtolunc | your best bet... |
| 06:14.21 | outtolunc | www.digium.com/handbook-draft.pdf |
| 06:14.25 | FuriousGeorge | so priority 1, would be the wait 20 sconds; 2, Answer on 's';, call on other line |
| 06:14.28 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 06:14.37 | outtolunc | that will give you a basic overview |
| 06:15.13 | outtolunc | the point being, once you 'have' the inbound (answered it), you can do anything you want with it |
| 06:15.36 | FuriousGeorge | its been about a month since i read that one. i got the book since then and read that. |
| 06:15.53 | outtolunc | playback, please wait, playback ringing, dump it, whatever |
| 06:16.06 | outtolunc | you really need to read it again |
| 06:16.17 | outtolunc | there are examples that DO APPLY |
| 06:16.32 | FuriousGeorge | i was just trying to get a feel for how complex it would be, having no landlines at my appartment, i have no need for FXO |
| 06:16.41 | FuriousGeorge | outtolunc: i will definately take a look at it tomarrow |
| 06:16.46 | outtolunc | there are none |
| 06:16.48 | FuriousGeorge | just wanted to go to work prepared |
| 06:17.25 | outtolunc | "once you answer" a channel, you can "DO anything with it" |
| 06:17.46 | outtolunc | anything is defined (not in whole) with that document |
| 06:18.04 | FuriousGeorge | sweet ;) ill uise that as a reference while i hammer it out |
| 06:18.24 | FuriousGeorge | and by hammer i mean trial and error-ate |
| 06:18.31 | outtolunc | just remember, answer, playback/dial/whatever |
| 06:18.53 | FuriousGeorge | outtolunc: ive been trying to visualize it in my head for a while, and thats a lot what it looked like |
| 06:18.56 | outtolunc | once you answer, you are in control of that channel |
| 06:19.15 | outtolunc | you just have to decide what to do with it |
| 06:19.37 | FuriousGeorge | outtolunc: i guess after that ill do voice mail, got a good reference for htat one? |
| 06:19.52 | FuriousGeorge | on the wiki perhaps? |
| 06:19.58 | outtolunc | you could feed it a background/prompt, playback 'gf go away', etc |
| 06:20.14 | FuriousGeorge | lol |
| 06:20.28 | outtolunc | which leads too, of what you decide to send it too isn't 'available' send it elsewhere |
| 06:20.40 | FuriousGeorge | its actually for my two bosses with the same name. "for maria press 1 for maria press 2" is what i had in mind |
| 06:20.41 | outtolunc | if if if if else |
| 06:21.13 | outtolunc | thats common and explained.. there are 'real'examples on various sites |
| 06:21.21 | FuriousGeorge | outtolunc: so thats when i would use conditionsals, i get that. that pascal class i took in HS in 98 will come in handy |
| 06:21.29 | outtolunc | http://www.digium.com/index.php?menu=documentation |
| 06:21.34 | outtolunc | look for dialplan |
| 06:21.39 | outtolunc | hit the wiki |
| 06:22.04 | outtolunc | (first on 'unoffical links') |
| 06:22.10 | lehel | or: http://www.voip-info.org/wiki-Asterisk+dial+plan+-+working+example |
| 06:22.14 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:22.24 | outtolunc | ty lehel |
| 06:22.38 | lehel | outtolunc: i'm looking now the extensions.conf |
| 06:22.45 | outtolunc | says.. "it's that easy" |
| 06:23.20 | lehel | and on the "digium" says me to add on the [default] context.. but i haven't |
| 06:23.23 | FuriousGeorge | yeah, i think im getting the hang of this. |
| 06:23.31 | *** join/#asterisk three55ml_laptop (~three55ml@cpe-66-25-85-88.satx.res.rr.com) |
| 06:23.40 | lehel | i should create [default]? |
| 06:23.44 | FuriousGeorge | one lass ? is there anything like a "counter" that can be used in the dialplan |
| 06:23.45 | brookshire | you almost have to think of dialplan as it's own programming language |
| 06:23.46 | brookshire | heh |
| 06:24.33 | FuriousGeorge | brookshire: i never did much coding but thats instantly what comes to mind here |
| 06:24.34 | outtolunc | examples are example are examples |
| 06:25.02 | brookshire | it's pretty basic though.. not many functions it can do |
| 06:25.04 | FuriousGeorge | outtolunc: you know of anything like a "counter" that could keep track of minutes |
| 06:25.20 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 06:25.43 | brookshire | furious: you can read the logs |
| 06:26.04 | brookshire | but there is also other stuff out there |
| 06:26.22 | lehel | outtolunc: so it is [outgoing] <10x |
| 06:26.23 | FuriousGeorge | brookshire: i thought i heard something about * being able to keep track of billing |
| 06:26.32 | outtolunc | tracking 'minutes' are the aspect of the cdr's (or if if app based, within each app) |
| 06:26.55 | outtolunc | suggestion to each of you |
| 06:27.09 | FuriousGeorge | cdr? |
| 06:27.26 | outtolunc | if you have inbound, set a context for it, if you have more that one, you should have more that ONE inbound context |
| 06:27.34 | outtolunc | same with outbound |
| 06:27.48 | outtolunc | put calls where they should be |
| 06:28.15 | outtolunc | if they 'fall thru' to [default] tell them (nicely) 'you shouldn't be here' |
| 06:28.33 | FuriousGeorge | gotcha |
| 06:28.41 | outtolunc | then if you do this, you will understand/control your pbx |
| 06:29.00 | outtolunc | it really is, 'that simple' |
| 06:29.22 | FuriousGeorge | you've armed me with knowledge :) |
| 06:29.34 | *** join/#asterisk fabioFVZ (~fabio@213-92-104-168.f5.ngi.it) |
| 06:29.53 | outtolunc | as is my mission in life, please don't tell me it's time to die happy <G> |
| 06:30.19 | FuriousGeorge | lol, gonna hit the sack. thanks for the info |
| 06:30.39 | outtolunc | (usually i have to 'fight tooth and nail' and people thing i'm a total .... |
| 06:30.45 | *** join/#asterisk oej (~oej@213.204.186.40) |
| 06:30.52 | outtolunc | er think |
| 06:30.54 | outtolunc | ) |
| 06:31.31 | outtolunc | after 25 years, i've gotten used to it <G> |
| 06:32.07 | FuriousGeorge | the techy is oft' misunderstood |
| 06:32.22 | outtolunc | if that were only it |
| 06:33.26 | outtolunc | even if one is told what to 'push' step by step, there are still people that cannot do it |
| 06:33.42 | outtolunc | even today! |
| 06:33.51 | FuriousGeorge | b/c they think they cant |
| 06:33.58 | outtolunc | (as they all work for the company i work for.. ) <G> |
| 06:34.12 | outtolunc | and i'm reminded, everyday <G> |
| 06:34.53 | outtolunc | so if some of you think i'm abrupt, you don't know the half of it |
| 06:35.34 | FuriousGeorge | im out, g'night |
| 06:39.20 | outtolunc | damn, i guess i am scary <G> |
| 06:41.30 | outtolunc | if there is one thing i have learn in all these years, if that *usually* this issue is already known, it's the ability of there person with the issue, and the person 'trying' to help to DIG it out of them |
| 06:41.45 | outtolunc | er is that |
| 06:42.33 | dudes | what's up outtolunc |
| 06:42.47 | outtolunc | just waiting for you <G> |
| 06:43.01 | outtolunc | it's all 'his' fault <G> |
| 06:43.09 | dudes | we didn't see you in gnudialer, so we figured you weren't around |
| 06:43.19 | outtolunc | i was earlier |
| 06:43.50 | dudes | I woke up later than normal. Heath also rode with his boss and was later than normal |
| 06:44.08 | outtolunc | np |
| 06:54.48 | *** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 06:59.00 | vpp | does anyone know if there's a way to set the codec frame size in asterisk? |
| 07:00.02 | outtolunc | why would you want to adjust a codec frame size? |
| 07:00.37 | outtolunc | (noting the algorithm is kinda designed with it in mind) |
| 07:00.53 | vpp | because different frame sizes have different overhead |
| 07:01.01 | outtolunc | why not just adjust your mtu and let your routers deal with packet size |
| 07:01.40 | vpp | because i have limited bandwidth and would like to limit the codec frame size |
| 07:01.43 | outtolunc | if you want to 'max' a proto.. design a new one <G> |
| 07:01.48 | vpp | so that i can handle more calls |
| 07:02.02 | Qwell | Here's a novel idea...why not use a different codec? |
| 07:02.05 | outtolunc | like iax and trunking <G> |
| 07:02.17 | outtolunc | hehe |
| 07:02.18 | vpp | because i can't control the codec the end user will use |
| 07:02.38 | vpp | only that it will be g729 or g723.1 with varying frame size |
| 07:02.39 | outtolunc | but SOME codec have intermediaries |
| 07:02.53 | vpp | every 'real' gateway lets you choose it |
| 07:02.59 | vpp | not sure why atserisk doesnt |
| 07:03.18 | outtolunc | because its not designed that way? |
| 07:03.33 | outtolunc | it's designed to traverse, trunk. |
| 07:03.40 | vpp | hmm |
| 07:03.58 | outtolunc | and this isn't to your liking because you want to trim it more? |
| 07:04.09 | outtolunc | it's oNLY as big as it needs to be |
| 07:04.32 | vpp | well i'm thinking to replace my gateways with a bunch of asterisk's |
| 07:04.39 | outtolunc | if you MTU is adjusted to your 'usage' your network will 'react' efficiently |
| 07:04.52 | vpp | since i only have 512/512 at each site |
| 07:05.32 | vpp | and need 24 calls.. the max i can do is.. G729 @ 30ms |
| 07:05.41 | outtolunc | have you ever delt with determining the 'block size' you should design a database to use? or a filesystem? |
| 07:05.49 | vpp | or G723.1r6.3 @ 30ms |
| 07:08.45 | *** join/#asterisk julianjm_ (~juianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 07:11.25 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 07:11.41 | outtolunc | my happy pill is in the form of a 'few' glasses of whiskey <G> |
| 07:11.56 | outtolunc | which, i've already had, mind you <G> |
| 07:12.09 | outtolunc | so this IS, as happy as i get5 |
| 07:12.12 | outtolunc | er -5 |
| 07:21.39 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 07:23.23 | JamesDotCom | i'm sure there must be more documents about debugging this kind of stuff, but I cant seem to find any... when making a SIP -> PSTN call through a quad-span card, a faint crackle can be heard on the PSTN side |
| 07:23.58 | JamesDotCom | we found the card was sharing an IRQ but changed motherboard and that seemed to minimise the crackle problems a little |
| 07:24.06 | JamesDotCom | but these ones are persistant |
| 07:24.16 | JamesDotCom | Zap -> Zap is fine |
| 07:24.29 | JamesDotCom | just !Zap -> Zap is the problem |
| 07:24.47 | Qwell | are you using a crappy codec? |
| 07:25.00 | JamesDotCom | nup, g.711 |
| 07:25.18 | kb1_kanobe | JamesDotCom: pops and clicks, no particular pattern? |
| 07:25.43 | JamesDotCom | doesnt seem to be a pattern, just persistent |
| 07:25.53 | JamesDotCom | they show up audibly on the pstn side of a ztmonitor -v |
| 07:26.11 | kb1_kanobe | interesting, haven't tried monitoring mine. |
| 07:26.15 | kb1_kanobe | head or stable? |
| 07:26.29 | JamesDotCom | if there is a pattern, i'm not sure what it correlates to yet |
| 07:26.30 | JamesDotCom | stable |
| 07:26.32 | JamesDotCom | 1.0.7 |
| 07:26.38 | kb1_kanobe | kernel 2.4? |
| 07:26.41 | JamesDotCom | 2.6 |
| 07:26.45 | kb1_kanobe | hmm... |
| 07:27.01 | kb1_kanobe | have you taken a look at the IRQ mapping in 'lspci -vvb' ? |
| 07:27.08 | JamesDotCom | the thing that's got me, is SIP -> SIP is fine, Zap -> Zap is fine |
| 07:27.14 | JamesDotCom | and yes, i'll make extra sure now :P |
| 07:27.55 | JamesDotCom | lspci -vvb | grep IRQ |
| 07:27.56 | JamesDotCom | <PROTECTED> |
| 07:27.56 | JamesDotCom | <PROTECTED> |
| 07:27.56 | JamesDotCom | <PROTECTED> |
| 07:27.56 | JamesDotCom | <PROTECTED> |
| 07:28.00 | JamesDotCom | on 12, all by itself |
| 07:28.03 | JamesDotCom | sorry bout the paste |
| 07:28.13 | kb1_kanobe | which 2.6? |
| 07:28.27 | JamesDotCom | 2.6.8 |
| 07:29.12 | JamesDotCom | i just have nothing to go on atm :< |
| 07:30.15 | JamesDotCom | nor can i really find any documentation on how you would go about debugging the problem |
| 07:30.37 | kb1_kanobe | there is a patch in head that raises one of the debug messages to warning level which may give you something to correlate to: search chan_zap.c for 'Write returned %d (%s) on channel %d' and upgrade it to a WARNING and recompile and see what pops up. otherwise you'll need to run w/set debug 1 and read really fast. |
| 07:31.01 | JamesDotCom | haha |
| 07:31.04 | JamesDotCom | i'll give that a look |
| 07:32.03 | kb1_kanobe | its something i've been exploring the last few days myself. When I have more than about 26 channels up the write() in that function starts to fail and bits and peices of audio are dropped from random zaptel channels. |
| 07:34.19 | JamesDotCom | alright, well unfortunately this is on a production machine atm |
| 07:34.25 | JamesDotCom | so i'll give it a recompile tonight and test |
| 07:34.36 | JamesDotCom | see how i go, thanks for the help :D |
| 07:34.52 | kb1_kanobe | np, it might be a red-herring after all. :-) |
| 07:36.02 | three55ml_laptop | msg PTG123 How's it going? |
| 07:36.37 | Qwell | msg three55ml_laptop You need a slash |
| 07:36.43 | Qwell | :p |
| 07:37.53 | three55ml_laptop | Ooops :) |
| 07:37.55 | three55ml_laptop | It's late |
| 07:41.00 | three55ml_laptop | It's quiet in here tonight |
| 07:42.30 | tzafrir | what night? |
| 07:43.14 | three55ml_laptop | I guess it is almost 3AM even here |
| 07:57.06 | PTG123 | yo |
| 08:03.16 | *** join/#asterisk opsys (~aa@adsl-065-006-173-010.sip.mia.bellsouth.net) |
| 08:03.51 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 08:04.01 | nrc | hi all |
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| 08:04.09 | opsys | Hi |
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| 08:28.49 | *** mode/#asterisk [+o denon] by ChanServ |
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| 08:32.21 | Blackvel | where is the difference between agi_dnid and agi_extension? |
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| 08:37.22 | bonez39 | any other PC Magazine subscribers here? |
| 08:39.23 | bonez39 | Dvorak had a rant in current issue about VOIP getting absorbed by the telcos...the ILEC's...did anyone else see it? |
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| 09:20.47 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 09:20.52 | lehel | hello |
| 09:21.17 | *** join/#asterisk Deryl (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 09:22.34 | lehel | i changed the zaptel.conf and zapata.conf ... as it is http://www.digium.com/index.php?menu=configuration#DevKitTDM |
| 09:22.48 | lehel | still not working |
| 09:22.58 | lehel | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 09:23.11 | tzafrir | lehel, tried genzaptelconf? |
| 09:23.42 | lehel | i'll try again now |
| 09:27.25 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 09:35.13 | lehel | tzafrir: ok!.. that problem is solved |
| 09:35.27 | Blackvel | what is agi_dnid? is there any difference to agi_extension? |
| 09:35.37 | Blackvel | is there only a different for ZAP devices? |
| 09:36.05 | Blackvel | when i call with x-lite into * (exten 200), both variables have the value 200 |
| 09:36.29 | lehel | Asterisk Gateway Interface |
| 09:38.07 | Blackvel | in fact |
| 09:38.20 | lehel | now: [chan_capi.so] => (Common ISDN API for Asterisk) |
| 09:38.20 | lehel | May 4 12:37:54 NOTICE[4276]: chan_capi.c:2635 load_module: CAPI not installed! |
| 09:38.32 | lehel | May 4 12:37:54 WARNING[4276]: loader.c:440 load_modules: Loading module chan_capi.so failed! |
| 09:38.43 | Blackvel | lsmod |
| 09:39.09 | lehel | Blackvel: capi 19584 0 |
| 09:39.09 | lehel | kernelcapi 31908 1 [capi] |
| 09:39.09 | lehel | capiutil 22112 0 [kernelcapi] |
| 09:39.10 | lehel | capifs 3888 1 [capi] |
| 09:39.17 | Blackvel | weird |
| 09:39.20 | lehel | yes |
| 09:39.42 | lehel | used: 0 |
| 09:40.06 | lehel | i hope this is my last problems.. pls help |
| 09:41.09 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
| 09:42.45 | lehel | it is nothing to do? |
| 09:43.18 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 09:50.18 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 09:50.24 | tengulre | Hi,all |
| 09:51.05 | lehel | i have downloaded asterisk-chan_capi-0.3.5 |
| 09:51.24 | lehel | when "make && make install" >> error error.... ?? |
| 09:52.24 | tengulre | what error? |
| 09:53.00 | lehel | i think there are just errors ;] ... :( |
| 09:53.09 | lehel | example: |
| 09:53.09 | Blackvel | maybe you need to check your kernel headers |
| 09:53.30 | lehel | chan_capi.c:2716: error: dereferencing pointer to incomplete type |
| 09:53.43 | lehel | chan_capi.c:2133: error: parse error before ')' token |
| 09:53.46 | lehel | a lot of this... |
| 09:54.22 | Blackvel | make sure your kernel path matche the one from Makefile |
| 09:54.31 | Blackvel | matches |
| 09:55.39 | lehel | but where to extract? anywhere is ok? |
| 09:56.31 | lehel | there is no path in the Makefile |
| 09:57.54 | Blackvel | prolly for kernel header the chan_capi Makefile references some directory, with the -I option in make |
| 09:58.11 | Blackvel | maybe you dont have it |
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| 10:11.32 | lehel | Blackvel: could you be more specific pls.. |
| 10:13.16 | lehel | should i paste in pastebin.ca my Makefile? |
| 10:13.28 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 10:14.24 | langals | Hi there...I am assuming that one can only use IAX trunking between 2 asterisk boxes, and not between an asterisk box and a NAT...could someone confirm? |
| 10:16.13 | *** join/#asterisk DeeJayTwo (~deejay2@207.134.166.34) |
| 10:16.14 | Sato1 | a NAT? |
| 10:16.23 | Sato1 | or thru a NAT? |
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| 10:21.09 | langals | Sato1 - this is my setup - I have an Asterisk box on a public IP, with users on different LANs signing into a meetme conference. There will be quite a few people on certain LANs at the same time, so I am trying to conserve bandwidth |
| 10:21.36 | langals | Sato - so those on a LAN will be communicating through the same NAT |
| 10:22.00 | langals | Sato1 - is there any way to to have to duplicate voice packets? |
| 10:23.24 | *** join/#asterisk Fabe (~spamhere@p54B21DE2.dip0.t-ipconnect.de) |
| 10:23.28 | Sato1 | depending on the device, there is an option from which you specify that every device uses your asterisk as a bridge, so, no matter from your LAN or internet, everybody can connect, if that option is not specified, when you call from internet to a device in your LAN, connection wont happend due the NAT |
| 10:23.42 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 10:24.59 | langals | Sato1 - ok, thanks for the help |
| 10:25.23 | Sato1 | and that option, at least in iax.conf is called "notransfer" and has to be set as "yes" |
| 10:25.28 | Sato1 | not sure for sip |
| 10:27.31 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:28.45 | Delvar | sip is canreinvote=no |
| 10:28.57 | Delvar | canreinvite=no * |
| 10:29.08 | Sato1 | there it is |
| 10:30.15 | lehel | ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include ?? |
| 10:30.23 | lehel | wich is the Asterisk header dir? |
| 10:32.28 | Sato1 | in /usr/include/asterisk |
| 10:32.32 | Sato1 | at least in my linux |
| 10:32.43 | lehel | what distro? |
| 10:32.47 | Sato1 | rh |
| 10:32.58 | Sato1 | rh, fedora, centoo |
| 10:33.03 | *** part/#asterisk RobinReliant (~mozrat@80.68.89.215) |
| 10:33.23 | lehel | i have no asterisk in /usr/include! |
| 10:33.36 | Sato1 | what OS/distro? |
| 10:33.40 | lehel | Debian |
| 10:34.02 | Sato1 | you can use the source tree |
| 10:34.33 | Sato1 | it is inside your source tree at ./include/asterisk |
| 10:34.46 | lehel | this is why i can't install asterisk-chan_capi |
| 10:34.52 | Martohtar | /usr/local/include ? |
| 10:35.00 | Sato1 | that one too |
| 10:35.21 | lehel | .usr/local/include < empty |
| 10:36.22 | Sato1 | if you downloaded the source from cvs, and installed in the normal way in a linux, it should reside at /usr/include/asterisk/ |
| 10:36.56 | lehel | ./usr/src/asterisk/asterisk/include/asterisk/ |
| 10:37.22 | lehel | ./usr/src/asterisk/include/asterisk |
| 10:37.31 | lehel | the second is ok? |
| 10:37.31 | *** join/#asterisk pokui (~pokui@193.108.252.162) |
| 10:37.34 | Sato1 | is your asterisk up and running? |
| 10:37.55 | Sato1 | dont know, i cant see your screen |
| 10:37.56 | lehel | it is up.. and not running.. i'm trying to configure now |
| 10:38.05 | lehel | i have problems with chan_capi |
| 10:38.26 | Sato1 | up and not running, ok... did you made "make install"? |
| 10:39.49 | lehel | command: asterisk -vvvgc |
| 10:41.04 | Sato1 | dont msg me, try here, if i dont know, someone else may help |
| 10:42.37 | lehel | 6 lines coming... |
| 10:42.39 | lehel | Parsing '/etc/asterisk/capi.conf': Found |
| 10:42.39 | lehel | May 4 13:39:38 NOTICE[5271]: chan_capi.c:2635 load_module: CAPI not installed! |
| 10:42.39 | lehel | May 4 13:39:38 WARNING[5271]: loader.c:345 ast_load_resource: chan_capi.so: loa d_module failed, returning -1 |
| 10:42.40 | lehel | May 4 13:39:38 WARNING[5271]: chan_capi.c:2811 unload_module: Unable to unregis ter from CAPI! |
| 10:42.41 | lehel | <PROTECTED> |
| 10:42.41 | lehel | May 4 13:39:38 WARNING[5271]: loader.c:440 load_modules: Loading module chan_ca pi.so failed! |
| 10:43.02 | Sato1 | and.. i dont recognice that module from the normal asterisk instalation, you may have more success contacting the group or author that made that module |
| 10:44.14 | lehel | [chan_capi.so] => (Common ISDN API for Asterisk) |
| 10:44.34 | lehel | maybe i should just disable in the modules.conf ? |
| 10:45.15 | Sato1 | or erase it from the modules dir |
| 10:45.31 | newl | erm..that's definately not a standard asterisk library. |
| 10:46.00 | *** join/#asterisk wiz8291 (~dang@kay.arcbox.com) |
| 10:46.19 | wiz8291 | hi guys, i have a problem with incoming caller id... anyone about that could give me a hand? |
| 10:46.56 | fenlander | wiz8291L what is the problem? |
| 10:47.01 | Sato1 | wiz8291, provide more information |
| 10:47.09 | wiz8291 | basically, caller id is not presented... |
| 10:47.15 | wiz8291 | no reason it shouldn't be |
| 10:47.31 | Sato1 | caller id from what? to what? |
| 10:47.37 | wiz8291 | i'm calling from my cell which DEFINITELY sends out the caller ID |
| 10:47.50 | Sato1 | still the same, need more information |
| 10:47.52 | wiz8291 | and i'm calling a Zap channel via an ISDN30e circuit |
| 10:48.13 | RoyK | anyone that knows what sort of sound file this is? phrase0101: raw G3 data, byte-padded |
| 10:48.14 | Sato1 | oh, thats beyond my knowledge |
| 10:48.31 | wiz8291 | so its incoming caller id on an E1/T1 i guess |
| 10:49.51 | Sato1 | as far as i know, an ISDN is a digital 3 channels line (2 for voice and one for handling or control) |
| 10:50.07 | Sato1 | but never used that before on asterisk |
| 10:50.36 | wiz8291 | nah, this is an ISDN30 |
| 10:50.44 | wiz8291 | 31 B channels, 1 D |
| 10:51.01 | wiz8291 | much bigger |
| 10:51.03 | wiz8291 | :/ |
| 10:51.11 | Sato1 | i c |
| 10:51.28 | Sato1 | well, lets wait for anotherone for that question, sorry |
| 10:51.51 | wiz8291 | cheers |
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| 10:57.18 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 10:59.18 | pokui | hi all, I'm having problems with pthreads on a 2.4.19 kernel ... includes/asterisk/utils.h has an #ifdef LINUX that tries to use pthread_create but pthread.h defines this as __use_ast_pthreads__ etc... any way I can sort this out? |
| 11:00.44 | Blackvel | lehel: check your chan_capi Makefile against -I/usr/src/linux-headers or something |
| 11:03.18 | tld | I'm trying to make a stream available through Astrisk, and I'm using MP3Playback to do it. This almost works, except that a) mpg123 sucks mountains through drinking straws, and b) I need a buffer in front of the MP3 player. |
| 11:03.22 | tld | Any suggestions on either? |
| 11:05.45 | ManxPower | tld: mpg123 buffers |
| 11:08.55 | *** join/#asterisk clive- (~pirch@rndf-146-52-213.telkomadsl.co.za) |
| 11:11.16 | tld | ManxPower: Yeah, but not enough it seems. |
| 11:11.24 | tld | ManxPower: I'm playing a Norwegian stream on a US Asterisk. |
| 11:13.38 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 11:14.29 | ManxPower | buffering only applies between the mp3 application and the transmitted RTP. Sounds like you have network issues. |
| 11:15.20 | *** join/#asterisk eper-werk (~eperdeme@telkom.gotadsl.co.uk) |
| 11:16.24 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 11:16.55 | tld | No, what I mean is that I need a buffer between the incomming MP3 player and the playback routine. |
| 11:17.09 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 11:17.18 | tld | Because mpg123 starts playing at once, which means it can't have anything buffered in case of network delays. |
| 11:18.38 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 11:18.46 | lehel | hey! |
| 11:19.02 | lehel | Asterisk Ready. |
| 11:19.21 | lehel | but!: |
| 11:19.32 | lehel | *CLI> May 4 14:08:13 ERROR[5939]: rate_engine.c:697 poster_worker: Failed to connect to MySQL database 'rating': Unknown MySQL Server Host 'server.sigmasoft.com' (1) |
| 11:19.46 | lehel | pls HELP |
| 11:20.06 | Sato1 | [root@gateway rauleli]# ping server.sigmasoft.com |
| 11:20.07 | Sato1 | ping: unknown host server.sigmasoft.com |
| 11:20.08 | ManxPower | tld: network delays would be buffered using the jitter buffer. |
| 11:20.15 | pokui | err is there a particular version of gcc I should use with asterisk? |
| 11:20.27 | pokui | to get it to compile? |
| 11:20.44 | Sato1 | lehel, that server does not exist, check if its the right name |
| 11:21.32 | tld | ManxPower: You're thinking about it with a asterisk-only view. My point is that asterisk can have as good handling it wants, and it won't help a bit if mpg123 cant' supply it with a stready stream. |
| 11:21.37 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 11:21.41 | *** join/#asterisk OloBola (~casper_sp@adsl-69-110-121-26.dsl.pltn13.pacbell.net) |
| 11:22.14 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 11:22.59 | pokui | ok.. lemme start at the beginning. downloading and compiling asterisk gives me errors like chan_modem.c:690: warning: implicit declaration of function `__use_ast_pthread_create_instead__' |
| 11:24.02 | tld | slink sak: http://www.mpx.no/aspx/prdinfo.aspx?plid=15125 |
| 11:24.06 | tld | Oops, sorry. |
| 11:24.27 | Blackvel | lehel: what the heck. what version did you install? :) |
| 11:24.44 | OloBola | nuuuuuuuuuuuuuuuuufffffffffffooooooooooooooooooonnnnnnnneeeeeeeeeeeeeeeeeeeeee? |
| 11:24.49 | pokui | looking at the source gives lines like if (ast_pthread_create(&monitor_thread, NULL, do_monitor, NULL) < 0) { |
| 11:26.22 | pokui | but utils.h has an #ifdef LINUX ... #define ast_pthread_create pthread_create |
| 11:26.31 | pokui | any way round this loop? |
| 11:26.43 | RoyK | ~lart OloBola |
| 11:26.58 | Wonka | ~lart jbot |
| 11:28.53 | RoyK | :) |
| 11:28.57 | RoyK | ~lart himself |
| 11:31.18 | Wonka | ~lart itself |
| 11:31.26 | Wonka | ~lart the bot |
| 11:32.00 | eper-werk | so i have an ISDN with 8 voice channels coming into the office, and this old PBX doesn't record voice, so can i get some card to put in the server+asterisk that will use the 8 voice channels from the ISDN line for incoming/outgoing calls? any idea of what card |
| 11:32.41 | Blackvel | eper-werk: you need a isdn bri card |
| 11:32.45 | Blackvel | err quad card |
| 11:33.02 | Blackvel | an even..typo |
| 11:33.20 | eper-werk | ok |
| 11:48.55 | *** join/#asterisk daork (~daork@202.89.128.251) |
| 11:49.26 | daork | so, does anyone here have any tips on increasing the buffers that oh323 uses? I'm running h.323 over a 600-800 ms link... |
| 11:50.01 | pokui | hmm... fixing utils.h compiles with no warnings, but running asterisk -vvc gives undefined symbol: _Z18ast_pthread_createPmP16__pthread_attr_sPFPvS2_ES2_ |
| 11:50.11 | pokui | on chan_vpp.so any ideas? |
| 11:51.32 | *** join/#asterisk key2 (~key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 11:52.01 | key2 | hey |
| 11:52.54 | key2 | how do u tell asterisk to call a number from callerid with changing the first digit ? for a callback, ex: if the number 0123456789 calls i want it to callback on 9123456789 changin the 0 ot 9 |
| 11:57.12 | masonc | 9${EXTEN} |
| 11:57.19 | masonc | 9${EXTEN:1} |
| 12:00.33 | key2 | what's the :1 for |
| 12:00.33 | key2 | ? |
| 12:00.57 | daork | drop the first digit |
| 12:01.03 | key2 | ok |
| 12:01.05 | daork | :2 is drop the first two digits |
| 12:01.06 | daork | etc |
| 12:01.28 | key2 | so basically, if I want my asterisk to callback someone that calls without answering when he calls |
| 12:01.32 | key2 | how should I do that / |
| 12:01.32 | key2 | ? |
| 12:01.48 | daork | what are you trying to accomplish? |
| 12:01.50 | key2 | like I wanna set up a callback that uses the callerid to call |
| 12:01.55 | daork | ok |
| 12:02.17 | key2 | daork: I basically want to know for example how I could call my asterisk with a mobile phone |
| 12:02.22 | daork | and you want to strip the first digit of the number and prefix it with a 9? |
| 12:02.29 | key2 | and without asterisk answer after 1 ring, it calls back the mobile phone |
| 12:03.04 | key2 | daork: first I wanna know how to callback without answering, is that doable ? |
| 12:03.14 | daork | i dont see why not |
| 12:03.47 | key2 | like I call from a phone my FXO card, and as soon as i hang up it calls back this phone number |
| 12:03.47 | daork | you'd probably need to make an AGI or something |
| 12:03.47 | key2 | AFI |
| 12:03.47 | key2 | AGI? |
| 12:03.51 | daork | ok |
| 12:03.51 | daork | so |
| 12:03.57 | daork | did you read the asterisk documentation? |
| 12:04.01 | key2 | yeah |
| 12:04.07 | key2 | i read the stuff about the dialplan |
| 12:04.08 | key2 | .. |
| 12:04.13 | daork | and you dont know what AGI is? |
| 12:04.14 | daork | oh |
| 12:04.15 | daork | yeah |
| 12:04.17 | daork | read the lot |
| 12:04.33 | daork | and have a look at voip-info.org |
| 12:04.40 | key2 | the asterisk gateway? |
| 12:04.51 | daork | im sorry? |
| 12:04.55 | key2 | agi is the asterisk gateway interface? |
| 12:04.59 | daork | yeah |
| 12:05.11 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 12:05.13 | key2 | ok |
| 12:05.46 | daork | I believe * has some kind of builtin callback stuff, but I dont know how to drive it |
| 12:06.04 | key2 | okb |
| 12:06.06 | key2 | ok |
| 12:06.09 | key2 | but basically |
| 12:06.25 | key2 | how would the agi helps? |
| 12:06.37 | daork | do you know what AGI is? |
| 12:06.44 | daork | not what it stands for, but what it /is/ |
| 12:08.56 | key2 | yeah |
| 12:09.02 | key2 | like u send command to asterisk trhough a language |
| 12:09.14 | key2 | like u do a printf("a command") |
| 12:09.22 | key2 | and since stdou is set to asterisk |
| 12:09.29 | key2 | asterisk takes the command |
| 12:09.30 | key2 | right? |
| 12:09.54 | Ahrimanes | like cgi is a way to do stuff on a webserver, agi is a way to do stuff in asterisk |
| 12:09.59 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 12:10.05 | key2 | ok |
| 12:10.06 | key2 | got it |
| 12:10.07 | key2 | :) |
| 12:10.22 | key2 | and how do u run an agi ? |
| 12:10.28 | daork | ok, so, the way i'd implement callback is, take a call, send it to an AGI that writes it to disk, and have something that reads that call from disk and makes a .call file in asterisk's call spool directory to make the outgoing call |
| 12:10.38 | daork | key2: you've got some reading to do |
| 12:11.01 | key2 | ok |
| 12:11.03 | key2 | i start to get it |
| 12:11.11 | Ahrimanes | key2: http://www.voip-info.org/wiki-Asterisk+AGI |
| 12:12.30 | *** join/#asterisk ethogeek (~ethogeek@CPE-24-209-154-94.wi.res.rr.com) |
| 12:12.37 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:13.17 | daork | key2: here you go |
| 12:13.17 | daork | http://lists.digium.com/pipermail/asterisk-users/2004-October/065406.html |
| 12:13.37 | ethogeek | so is nufone dead for anyone else? |
| 12:14.34 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:16.21 | OloBola | ethogeek: yep |
| 12:19.32 | *** part/#asterisk daork (~daork@202.89.128.251) |
| 12:20.29 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 12:21.03 | dca[laptop] | morning all, anyone have any hints on modprobing a wct1xxp? |
| 12:22.49 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 12:28.59 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 12:32.01 | ManxPower | ~docs |
| 12:32.02 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:37.54 | *** join/#asterisk Dovid (~hirisk@pool-151-198-15-84.mad.east.verizon.net) |
| 12:44.44 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:45.02 | OloBola | ~Lart nufone |
| 12:47.12 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 12:47.24 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 12:50.07 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 12:51.10 | Othello | lol |
| 12:51.12 | Othello | quake3!!!! |
| 12:51.20 | newbien | hi, will kphone OSS sound work if i setup fwd for iax authentication, etc? |
| 12:53.03 | bjohnson | wtf is Antispam UOL? |
| 12:53.36 | bjohnson | I just sent an email to the biz list and some putz has their system set up to auto-reply asking for confirmation |
| 12:54.42 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:55.08 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:55.30 | *** join/#asterisk shaonss (~shaon@acc8-ppp241.hay.dialup.connect.net.au) |
| 12:55.39 | bjohnson | how kphone works has nothing to do with fwd |
| 12:55.48 | bjohnson | (when using asterisk as a gateway) |
| 12:57.47 | Blackvel | who has tried new xten eyebeam? is it better in quality than xlite? |
| 12:58.05 | Blackvel | my soundcard must be crap, I always had an echo with xlite |
| 12:58.12 | gambolputty | hi |
| 12:58.21 | shaonss | which is the most compact linux(smallest) to use asterisk? |
| 12:58.37 | gambolputty | I can't connect * to a mysql database of mine. CDR works fine, but realtime doesn't connect. |
| 12:58.59 | gambolputty | current CVS version of * is being used. |
| 12:59.20 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 13:00.11 | tengulre | Hi,all |
| 13:02.22 | newbien | bjohnson: is my soundcard borked if i get a warning that alsa sound was not install and asterisk is using OSS sound? |
| 13:02.37 | tzafrir | shaonss, smallest is probably astlinux. (Naturally you can always roll your own) |
| 13:02.39 | *** join/#asterisk marlowe (~marlowe@marlowe.active.supporter.pdpc) |
| 13:03.13 | ManxPower | Blackvel: eyebeam is a video phone |
| 13:03.24 | ManxPower | ~docs |
| 13:03.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:03.38 | tzafrir | gambolputty, I can't connect my * to your database either. However if you'll give some error messages and such, some folks here may be able to help you |
| 13:04.06 | DeeJayTwo | suppose a client A is connecting to a Server 1. The Server 1 is in fact a gateway and is sending everything to server 2, what happens on the network side? Is everything still passing by server 1 or is the client connecting to server 2? Well... Is there a way to avoid any unncesseray traffic? |
| 13:04.51 | tzafrir | ManxPower, is there a similar stanza from the bot for "do a,b, c to give us your error/debug info"? |
| 13:05.07 | ManxPower | tzafrir: no idea |
| 13:05.22 | ManxPower | But you can tell jbot things |
| 13:05.29 | ManxPower | jbot, tzafrir is a drag queen |
| 13:05.30 | jbot | ManxPower: okay |
| 13:05.35 | ManxPower | ~tzafrir |
| 13:05.36 | jbot | tzafrir is probably a drag queen |
| 13:05.39 | ManxPower | see? |
| 13:05.54 | DeeJayTwo | Here I'm talking about iax connections... where the client A could be another gateway.. |
| 13:05.57 | shaonss | tzafrir: the link does not work to download |
| 13:05.58 | tzafrir | jbot, tzafrir is http://tzafrir.org.il/ |
| 13:05.59 | jbot | ...but tzafrir is already something else... |
| 13:06.14 | ManxPower | jbot, no tzafrir is http://tzafrir.org.il/ |
| 13:06.15 | jbot | okay, ManxPower |
| 13:06.19 | ManxPower | ~tzafrir |
| 13:06.20 | jbot | rumour has it, tzafrir is http://tzafrir.org.il/ |
| 13:06.41 | gambolputty | <PROTECTED> |
| 13:06.41 | tzafrir | Let's try some phrasing before we harase the poor bot |
| 13:06.59 | ManxPower | tzafrir: you can /msg the bot to do that sort of stuff too |
| 13:07.03 | gambolputty | found a mailing list letter that said to use res_odbc.conf, not res_mysql.conf |
| 13:07.09 | *** join/#asterisk chip_tmc (~chip@60-240-145-3.tpgi.com.au) |
| 13:07.10 | gambolputty | will try that now |
| 13:09.05 | chip_tmc | anyone know if MWI works with res_data? |
| 13:09.05 | *** join/#asterisk vpp (~noone@host-83-146-50-131.bulldogdsl.com) |
| 13:09.09 | vpp | hi! |
| 13:09.20 | vpp | is there a way to change the rx/tx gain in asterisk? |
| 13:09.24 | kajtzu | yes |
| 13:09.28 | kajtzu | see zapata.conf |
| 13:09.43 | vpp | but will that work if there isn't a physical device... |
| 13:09.45 | durex | Asterisks... |
| 13:09.49 | vpp | its just h323 in, h323 out |
| 13:09.54 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 13:09.57 | durex | I'm having some problem to compile asterisk-addons on FreeBSD 5.3-STABLE... |
| 13:10.01 | durex | look what I got: |
| 13:10.06 | durex | su-2.05b# cd /usr/src/asterisk-addons/ |
| 13:10.07 | durex | su-2.05b# make clean |
| 13:10.07 | durex | "Makefile", line 56: Missing dependency operator |
| 13:10.07 | durex | "Makefile", line 57: Could not find .depend |
| 13:10.07 | durex | "Makefile", line 58: Need an operator |
| 13:10.08 | durex | Makenshi: fatal errors encountered -- cannot continue |
| 13:10.14 | kajtzu | vpp: hmm wouldn't you set gain on your h.323 gateway |
| 13:10.14 | cpatry | durex: use pastebin! |
| 13:10.22 | kajtzu | vpp: (where the call enters the h.323 realm) |
| 13:10.23 | durex | cpatry sorry, but just a few lines... |
| 13:10.26 | *** join/#asterisk wmoran (~wmoran@24-53-250-148.pittpa.adelphia.net) |
| 13:10.31 | wmoran | Hello all |
| 13:10.39 | durex | does anybody knows what it should be? |
| 13:10.50 | gonzo- | durex: use gmake |
| 13:10.50 | wmoran | Has anyone dreamed up a way to make Merlin phones work with an * box? |
| 13:11.00 | vpp | yeah, but its normal on my gateway when its going to other devices... |
| 13:11.03 | vpp | ont he asterisk its louddddddd |
| 13:11.17 | tzafrir | durex, try using gmake instead of the bsd make? |
| 13:11.28 | *** part/#asterisk onlyI (~hisemail@gate.idsnetguard.net) |
| 13:11.40 | durex | yes... now it works ;-))) gmake! |
| 13:11.41 | durex | thank u |
| 13:11.42 | *** join/#asterisk onlyI (~hisemail@gate.idsnetguard.net) |
| 13:11.56 | newbien | is my soundcard borked if i get a warning that alsa sound was not install and asterisk is using OSS sound? |
| 13:12.01 | durex | but now I have a lot of compilation errors... |
| 13:12.47 | shaonss | my sound card does not load how can i make it to load? |
| 13:13.15 | durex | well... now I think I must compile /usr/src/asterisk first... and then gmake in /usr/src/asterisk-addons... |
| 13:13.41 | tzafrir | http://www.onlamp.com/pub/a/onlamp/2005/04/28/packaging2.html talks about handling such make issues more gracefully |
| 13:13.43 | durex | but the place where addons must go isn't /usr/src/asterisk... my real path isn't it... I'm under FreeBSD |
| 13:14.22 | durex | tzafrir thank u |
| 13:14.35 | tzafrir | durex, you don't need the include of asterisk.h in the cdr_mysql app |
| 13:14.38 | bjohnson | shaonss: openwrt |
| 13:14.41 | tzafrir | uncomment it |
| 13:15.35 | durex | tzafrir let me first end the gmake process of /usr/src/asterisk |
| 13:15.45 | tzafrir | durex, my debian deb of asterisk-addons (built out-of-tree) can be found under http://tzafrir.org.il/rapid/ |
| 13:17.08 | tzafrir | Generally depends on asterisk-dev , which is some headers under /usr/include/asterisk/ |
| 13:17.55 | tzafrir | the diff is the part that may interest you. debian/rules is the makefile bits |
| 13:19.07 | shaonss | bjohnson:[root@Mypbx root]# openwrt |
| 13:19.07 | shaonss | -bash: openwrt: command not found |
| 13:19.28 | durex | tzafrir plz take a look: http://pastebin.ca/10899 |
| 13:20.00 | vpp | why does the asterisk send ring before the far end rings? |
| 13:20.10 | *** join/#asterisk Veto (mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 13:20.11 | tzafrir | dudes, dr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory |
| 13:20.30 | tzafrir | durex, where are the asterisk include files? |
| 13:20.33 | shaonss | how to use music on hold in extensions.conf? |
| 13:21.08 | puowvip | asterisk is yummy. |
| 13:21.13 | durex | tzafrir I have to found it in FreeBSD.... |
| 13:21.22 | bjohnson | shaonss: ?? openwrt is the smallest linux distro I've seen that runs asterisk |
| 13:21.26 | durex | to find.. |
| 13:21.42 | bjohnson | shaonss: it is not a command |
| 13:21.52 | shaonss | bjohnson: ohhhh |
| 13:22.40 | bjohnson | it only runs on mips based hardware I think .. but you didn't specify what hardware you were asking about |
| 13:22.51 | *** join/#asterisk scubasteve (~steve@office65.neonova.net) |
| 13:23.21 | durex | tzafrir /usr/src/asterisk/include/asterisk/config.h |
| 13:23.22 | tzanger | scuba steve! |
| 13:23.23 | durex | is it? |
| 13:23.27 | scubasteve | Hey TZ! |
| 13:23.58 | tzafrir | durex, so you need to get -I/usr/src/asterisk/include into CFLAGS |
| 13:24.09 | shaonss | bjohnson: i have pIImmx-264MB pc and i want to run just asterisk and its some components |
| 13:24.15 | gonzo- | durex: why don't you use ports? |
| 13:24.47 | durex | gonzo- AMP aren't on ports... |
| 13:25.08 | wmoran | Has anyone dreamed up a way to make Merlin phones work with an * box? |
| 13:25.18 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:25.19 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:25.28 | djMax | so I know you can't send CID name over most PRIs, but can you get the telco's to associate names with CIDs? |
| 13:25.29 | tzafrir | durex, amp is a complete asterisk distribution. However with some careful action you could package its separate components separately |
| 13:26.00 | vpp | why does the asterisk send ring before the far end rings? |
| 13:26.05 | tzafrir | durex, for stanrters, take a look again at http://tzafrir.org.il/rapid/ and look for amportal |
| 13:26.48 | shaonss | <PROTECTED> |
| 13:28.31 | durex | wow... does anybody here running freebsd? |
| 13:28.45 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 13:28.55 | durex | tzanger It seems that I have to specify a lot of -I in cflags.... |
| 13:29.00 | fantomax1 | hi all |
| 13:29.10 | tzanger | durex: what are you trying to do |
| 13:29.44 | fantomax1 | can anyone suggest me how to solve the prob "Unable to allocate socket: Too many open files "? |
| 13:29.50 | fantomax1 | thanks in advance |
| 13:30.00 | ManxPower | durex: every single person that I know of that was new to Asterisk and tried running it on *BSD switched to running it on Linux |
| 13:30.02 | fantomax1 | I have 1.0.7 running |
| 13:30.18 | ManxPower | Some of them switched back to *BSD once they gained experience with Asterisk |
| 13:30.33 | tzanger | fantomax1: are you ulimiting files? |
| 13:30.42 | durex | ManxPower I don't use Linux on my servers, just BSD |
| 13:30.46 | fantomax1 | i have the normal configuration |
| 13:30.52 | fantomax1 | i can check wait |
| 13:30.59 | ManxPower | durex: Asterisk is only well supported on Asterisk |
| 13:31.02 | fantomax1 | in /proc/sys ? |
| 13:31.48 | fantomax1 | ulimit -n = 1024 |
| 13:32.01 | fantomax1 | i believe is the default |
| 13:32.08 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
| 13:32.18 | tzanger | yes it is |
| 13:32.25 | tzanger | fantomax1: are you running hundreds of calls? |
| 13:32.36 | fantomax1 | uhmm not so few |
| 13:32.48 | fantomax1 | i believe I have more that 200 sip channels opened |
| 13:32.54 | newbien | is my soundcard borked if i get a warning that alsa sound was not install and asterisk is using OSS sound? |
| 13:33.06 | fantomax1 | anyway .. the load .. is heavy |
| 13:33.18 | fantomax1 | for asterisk I mean |
| 13:33.25 | fantomax1 | cpu and memory is ok |
| 13:33.34 | tzanger | fantomax1: well jeez |
| 13:33.38 | tzanger | try unlimit -n 2048 |
| 13:34.02 | Blackvel | hey, who has the ultimate solution for zaphfc audio problems? maybe I only get them because i have connected my * to my isdn/analog pbx where all my telephones are connected to |
| 13:34.04 | fantomax1 | but i tried ... and after a while the system lose this setting |
| 13:34.08 | ManxPower | fantomax1: you can assume that each call uses at LEAST 4 file descriptors |
| 13:34.09 | *** join/#asterisk dmccollum (~dmccollum@eycb01-00-cntnga-69-164-245-72.atlaga.adelphia.net) |
| 13:34.31 | fantomax1 | uhmmm I see |
| 13:34.33 | Blackvel | either that or another way, these are the dmesg error messages I receive from time to time (and now I had tested asterisk to asterisk IAX2 calling): |
| 13:34.41 | fantomax1 | so I need to extedn the limit ... enough |
| 13:34.44 | Blackvel | zaphfc: empty HDLC frame or bad CRC received (framelen = 4, stat = 0xff). |
| 13:35.02 | ManxPower | newbien: you need to have alsa or oss installed, running, and working outside of Asterisk. You also need the devel package for alsa or oss |
| 13:35.14 | Blackvel | zaphfc: dropped audio (z1=6075, z2=6058, wanted 8 got 17, dropped 9). |
| 13:35.28 | Blackvel | is there any miracle how to turn errors off? :) |
| 13:35.34 | fantomax1 | i have the same error with RTP channel, but I believe is the same |
| 13:35.53 | fantomax1 | problem I mean |
| 13:36.39 | fantomax1 | max-files in /proc/sys/fs has anything to do with this limitation? |
| 13:36.42 | *** join/#asterisk eluizbr (~eluizbr@201.19.59.119) |
| 13:36.54 | bkw_ | ulimit can |
| 13:37.02 | fantomax1 | ok ... |
| 13:37.05 | newbien | ManxPower: alsa and oss work fine before asterisk install; i need to install the alsa devel libs now? OSS develop libs also? |
| 13:37.06 | bkw_ | bet you're default to 1024? |
| 13:37.08 | fantomax1 | i put 4096 |
| 13:37.11 | bjohnson | wmoran: is merlin nortel? |
| 13:37.13 | fantomax1 | i was on 1024 |
| 13:37.19 | bkw_ | I put 1000000 |
| 13:37.30 | ManxPower | newbien: Yes |
| 13:37.36 | jsharp | merlin is not nortel. |
| 13:37.54 | *** part/#asterisk eluizbr (~eluizbr@201.19.59.119) |
| 13:38.00 | ManxPower | Merlin is Lucent. Meridian is Nortel |
| 13:38.06 | jsharp | merlin == lucent/at&t/avaya/whatever they're going by this week. |
| 13:38.34 | ManxPower | jsharp: I just call them "The Red Coffee Stain Company" |
| 13:38.39 | wmoran | Merlin are the old AT&T phones |
| 13:38.41 | newbien | ManxPower: thanks, so asterisk or kphone will be mute with only OSS and no develop libs? |
| 13:38.43 | jsharp | Heh. |
| 13:38.43 | wmoran | It's Lucent now |
| 13:39.22 | wmoran | I know they're a proprietary phone system (neither VoIP nor analog), but I just wanted to make sure nobody had figured out how to use them before I told this client that his old phones were useless. |
| 13:39.23 | fantomax1 | in your opinion , with a Dual Xeon and 2 GB ram , how many concurrent call can I manage in GSM ? |
| 13:39.28 | mutilator | hmm |
| 13:39.34 | jsharp | We need someone in a non DMCAized country to reverse engineer the nortel and merlin protocols. |
| 13:39.46 | ManxPower | wmoran: the only way is to put asterisk between the PBX and the telco. |
| 13:39.54 | wmoran | Are they still keeping those big secret? |
| 13:39.59 | jsharp | Yup. |
| 13:40.08 | wmoran | It's amazing to me how many people still use those ancient Merlin systems. |
| 13:40.08 | ManxPower | wmoran: Of course. It's their ONLY advantage. |
| 13:40.24 | wmoran | OK, I know what to tell the client |
| 13:40.41 | wmoran | Just didn't want to tell him something and find out there was newly developed technology available. |
| 13:40.50 | tzanger | jsharp: I am working on that |
| 13:40.54 | tzanger | <-- Canada |
| 13:41.03 | jsharp | Exxxxxxxcellent. |
| 13:41.08 | tzanger | Norstar MICS and hopefully O11 |
| 13:41.12 | tzanger | I think the protocol's the same |
| 13:41.21 | tzanger | I am waiting on another TE405P so I can hack it up |
| 13:41.35 | wmoran | Anything expected in the near future, tzanger ? |
| 13:41.42 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 13:41.47 | tzanger | wmoran: near future as in next few weeks? no. probably 30-60 days off |
| 13:41.54 | tzanger | depending on business of my day job |
| 13:42.03 | wmoran | Hmmm ... but it will still be beta support at that point ... |
| 13:42.13 | jsharp | If I could solve this conferencing feature request that the CEO wants, I could rip this Merlin out and replace it with *. |
| 13:42.18 | tzanger | wmoran: of course |
| 13:42.25 | wmoran | And you're doing it on your time, so interruptions/distractions are likely |
| 13:42.27 | tzanger | besides |
| 13:42.31 | tzanger | DMCA does not apply |
| 13:42.41 | tzanger | DMCA gives *specific* exception for interoperabiliyt |
| 13:42.47 | tzanger | which is exactly what this is for |
| 13:42.47 | wmoran | IC |
| 13:42.52 | wmoran | I didn't know that |
| 13:42.54 | jsharp | I'm sure they'd take a whack at it, though. |
| 13:43.02 | tzanger | at least that is my opinion, and yes they are likely to take a whack at it |
| 13:43.04 | tzanger | :-) |
| 13:43.05 | *** join/#asterisk jmacz (~jmacz@63.245.86.185) |
| 13:43.36 | ManxPower | the problem is that most people that will hack a closed protocol for interoperability don't have the money to defend themselves against lawsuits. |
| 13:43.41 | ManxPower | Even if they would win. |
| 13:43.46 | tzanger | ManxPower: correct |
| 13:43.46 | wmoran | As long as you don't mind the whacking ... |
| 13:43.47 | jsharp | yup. |
| 13:44.07 | tzanger | now while Canada's not exactly the safest haven from U.S. influence it should be relatively save |
| 13:44.10 | tzanger | er safe |
| 13:44.13 | jsharp | Its the proverbial "Who does a 800 pound gorilla sue?" "Anyone he wants to". |
| 13:44.18 | tzanger | and so long as I keep my development 100% open it can't get shut down |
| 13:44.26 | tzanger | they can make ME stop working on it but my work to date is already out there |
| 13:44.35 | *** join/#asterisk eluizbr (~eluizbr@201.19.59.119) |
| 13:44.43 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 13:45.01 | eluizbr | how use kphone 4.1.1? |
| 13:45.25 | eluizbr | someone here use kphone? |
| 13:45.44 | nrc | yes |
| 13:45.58 | eluizbr | nrc: you use kphone? |
| 13:46.10 | tzanger | hahahaha |
| 13:46.14 | tzanger | 09:48 < eluizbr> someone here use kphone? |
| 13:46.14 | tzanger | 09:48 < nrc> yes |
| 13:46.14 | tzanger | 09:49 < eluizbr> nrc: you use kphone? |
| 13:46.16 | tzanger | DUH |
| 13:46.36 | bjohnson | tzanger: I'm waiting for a call from Bell. Do you happen to know data T1 costs and if it would be available to a farm house that can't get DSL? |
| 13:46.45 | tzanger | T1 is available anywhere |
| 13:46.50 | tzanger | be prepared for sticker shock |
| 13:46.54 | bjohnson | that's what I tought |
| 13:46.55 | ManxPower | It's just a matter of money. |
| 13:47.00 | bjohnson | thought |
| 13:47.14 | tzanger | it'd be better to find someone who has a tall structure in town and wireless link it to there |
| 13:47.25 | onlyI | anygood iax hardphone on the market |
| 13:47.43 | bjohnson | local wireless ISP is too full to serve them .. they've been on waiting list a year |
| 13:47.47 | tzanger | no |
| 13:47.48 | ManxPower | onlyI: You read the review on the mailing list of an IAX hardphone, right? |
| 13:47.51 | tzanger | not a wireless ISP |
| 13:48.03 | tzanger | find a guy with a tall structure and just point-to-point, and get DSL there and share it with him |
| 13:48.08 | tzanger | in exchange for the use of his tall structure |
| 13:48.14 | onlyI | ManxPower not really i'm on iaxtalk |
| 13:48.35 | ManxPower | onlyI: too bad you missed it. |
| 13:48.41 | onlyI | ManxPower looking at at320ed |
| 13:48.43 | burbankmarc | I keep getting this error "Unable to create channel of type 'SIP' " and from what i've read it means my lines are busy, but they're not, it's a config issue, i'm just too much of a newb to know where to look |
| 13:48.53 | tzanger | burbankmarc: sip debug |
| 13:48.58 | onlyI | ManxPower what mailling list |
| 13:49.00 | ManxPower | burbankmarc: "sip show peers" should show your SIP phone |
| 13:49.05 | ManxPower | s IP address. |
| 13:49.15 | ManxPower | If it doesn't, then the phone isn't registering with Asterisk |
| 13:49.20 | nrc | tzafrir: yes.. sorry, at work |
| 13:49.30 | ManxPower | onlyI: asterisk-users, I think |
| 13:50.08 | bjohnson | they're looking at satellite now |
| 13:50.15 | onlyI | ManxPower k thanks |
| 13:50.20 | zoa | what would you guys think |
| 13:50.31 | zoa | of me releasing the sip jitter buffer stable version today ? :) |
| 13:50.35 | burbankmarc | my phones are all registered and show up in the sip show peers |
| 13:50.58 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 13:52.18 | *** join/#asterisk olivier_ (~olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
| 13:52.45 | clive- | zoa, woooohooo,,,great news |
| 13:54.04 | jief- | is there a special config you need in * for your phone to be able to pick up the voice messages when you press the voice message button? or is that a standard? |
| 13:54.30 | zoa | http://astertest.com/forum/viewtopic.php?t=22 |
| 13:54.39 | ManxPower | zoa: stable jitter buffer for -HEAD for jitter buffer for 1.0.x? |
| 13:55.22 | *** join/#asterisk rkioko (~kiokorobe@196.200.26.42) |
| 13:56.11 | OloBola | a system status page is always nice, my lord |
| 13:56.53 | zoa | http://astertest.com/forum/viewtopic.php?p=61 |
| 13:57.00 | zoa | stable jitter buffer for -head |
| 13:57.05 | oej | zoa:! |
| 13:57.10 | zoa | olle! |
| 13:57.16 | zoa | how was the jitter buffer for you so far ? |
| 13:57.25 | oej | zoa: Great stuff. |
| 13:57.36 | tzanger | hmm |
| 13:57.41 | oej | zoa: That was what indicated to me that we were allocating way to many RTP channels |
| 13:57.44 | tzanger | there is no trouble with DTMF on SIP? |
| 13:57.47 | tzanger | with jitter buffer? |
| 13:57.52 | zoa | we did around a million calls on it so far without glitches, so guess its ready to be tested |
| 13:57.55 | oej | zoa: So I removed them at the same time as kpfleming did... |
| 13:58.10 | oej | zoa: So now, the JB buffer is not initialized as often |
| 13:58.15 | zoa | super! |
| 13:58.21 | tzanger | oej: eh? |
| 13:58.24 | tzanger | what did you change? |
| 13:58.28 | zoa | olle had a small preview |
| 13:58.35 | oej | zoa: But there must be something wrong with NAT support, since I got RTP errors with the patched rtp.c |
| 13:58.40 | tzanger | the jitter buffer has some issues with IAX2 but I think it has to do with a couple of gotchas chan_iax2 has |
| 13:58.47 | zoa | hmm we also used it with NAT i think |
| 13:59.00 | fantomax1 | is there anyone here that used SIPP for testing ? |
| 13:59.07 | zoa | fantomax1: i did |
| 13:59.10 | oej | zoa: Haven't had time to figure that one out, just saw it, replaced rtp.c with the old stuff and it works (I hope) |
| 13:59.14 | fantomax1 | hi zoa |
| 13:59.23 | clive- | tzanger what issues do you mean? |
| 13:59.27 | zoa | at least i did a while ago |
| 13:59.45 | zoa | how do you mean you replaced it with the old stuff ? |
| 13:59.59 | fantomax1 | can you give me some hint in how to setup a call including RTP , and possibly using an user not called sipp |
| 14:00.10 | fantomax1 | I tried by myself .. but no luck |
| 14:00.16 | fantomax1 | just the sintax |
| 14:00.19 | tzanger | clive-: if you hav asterisk boxes A and B, with JB on A and you make a call from A to B (even just to B's echo() app) and you hit DTMF digits, A's jitter buffer will go apeshit |
| 14:00.20 | zoa | a call including rtp, works with playback on the asterisk server |
| 14:00.24 | jief- | i have Gnet sip phones, when i leave a message, the light flash to notify me there's a message for me, but when i press the button to get the messages, nothing happens |
| 14:00.26 | oej | zoa: rm rtp.c / gmake update |
| 14:00.32 | burbankmarc | debugging the sip didn't help me out too much, well it might of, just i don't understand it too much |
| 14:00.39 | *** join/#asterisk frood (~frood@213.228.232.61) |
| 14:00.41 | frood | hey all |
| 14:00.48 | ManxPower | jief-: nothing works magically. you have to configure the button on the phone. |
| 14:00.58 | zoa | but not with for example app_milliwatt like some people on the mailinglist think they could do |
| 14:01.00 | fantomax1 | yes ... i terminate the call on an extension with a prercorded audio |
| 14:01.18 | frood | at the moment, asterisk bridges all calls directly. how do i get them to go via asterisk so i can record them, etc |
| 14:01.22 | frood | ? |
| 14:01.49 | oej | frood: Read the docs :-) Look at canreinvite=no |
| 14:01.59 | bkw_ | zoa |
| 14:02.00 | bkw_ | oh zoa |
| 14:02.02 | bkw_ | where art thou |
| 14:02.03 | frood | ok, thanks for your help |
| 14:02.20 | jief- | ManxPower: there's no where i can setup that in my phone settings |
| 14:02.31 | oej | bkw! |
| 14:02.37 | clive- | tzanger does anyone know about this bug? |
| 14:03.04 | tzanger | clive-: yes, stevek and kpflemming, although kpfleming probably doesnt' care at this point :-) |
| 14:03.04 | oej | bkw_: Question for you: If I get an AST_CONTROL_HOLD indication from pbx to chan_sip - would it be okey to start music on hold on that channel? |
| 14:03.17 | bkw_ | I think so |
| 14:03.21 | tzanger | stevek is busy and I've bene busy too but it's just something I'm trying to figure out |
| 14:03.26 | bkw_ | PCadach, added that if I recall |
| 14:03.27 | clive- | tzanger...why doesnt he care ?..:) |
| 14:03.37 | bkw_ | he would be more able to answer that question for sure |
| 14:03.41 | oej | bkw_: The other option is to send an SDP to 0.0.0.0 and let the phone sort out what to do |
| 14:03.42 | tzanger | clive-: because he's got a ton of tohter stuff to do and stevek and I are on it |
| 14:03.46 | *** join/#asterisk shaonss (~shaon@61.68.26.241) |
| 14:03.52 | durex | folks... does anybody have cdr_addon_mysql.so compiled for FreeBSD 5-STABLE ???? |
| 14:03.59 | bkw_ | oej hrm how would phones react? |
| 14:04.06 | clive- | tzanger well good luck in getting it fixed |
| 14:04.21 | tzanger | clive-: :-) oh it'll get fixed |
| 14:04.22 | durex | tzafrir should your cdr_addon_mysql.so would work on FreeBSD? |
| 14:04.39 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 14:04.48 | oej | Well, (guessing) Snom and advanced phones have a setting for music on hold URI, so they will propably know that they're on hold and do something about it. Other, simple, phone will propably be confused. |
| 14:04.50 | *** part/#asterisk eluizbr (~eluizbr@201.19.59.119) |
| 14:04.54 | clive- | tzanger out of interest, would an option for a fixed length jitter buffer be a good idea, to overcome dtmf funnies? |
| 14:04.56 | tzafrir | durex, assuming asterinsk and mysql are binary compatible: probably yes |
| 14:05.13 | tzanger | nope |
| 14:05.23 | tzanger | clive-: it's already got a limiter in it |
| 14:05.25 | tzanger | that's not the issue |
| 14:05.53 | durex | tzafrir could u send it to me? |
| 14:05.59 | tzafrir | durex, mysql used there is version 12 of the interface (the newest for 4.0/4.1). As for Asterisk: I doubt it. |
| 14:06.04 | shaonss | how to put user on hold in extensions.conf? |
| 14:06.13 | tzafrir | durex, debs are ar-ed tarballs |
| 14:06.32 | *** join/#asterisk opsys (~aa@adsl-065-006-173-010.sip.mia.bellsouth.net) |
| 14:06.35 | durex | plz gimme the URL again |
| 14:06.47 | tzafrir | http://tzafrir.org.il/rapid/ |
| 14:06.55 | clive- | tzanger, well we hope to hear good news about a fixed jitter buffer soon:) |
| 14:07.13 | tzanger | clive-: you and me both |
| 14:07.29 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 14:07.30 | tzanger | I'm getting "10s one-way audio" probelms even with jitterbuffer=no |
| 14:07.38 | bkw_ | tzanger, with/ |
| 14:07.39 | zoa | ah yes |
| 14:07.43 | zoa | thats a problem |
| 14:07.45 | zoa | we all know about |
| 14:07.48 | zoa | use stable |
| 14:07.50 | tzanger | it's infrequent and the dead audio is one-way only which means it's NOT internet related, at least not in a normal sense |
| 14:07.51 | zoa | it helps |
| 14:07.57 | tzanger | zoa: with IAX2?? |
| 14:07.58 | tzafrir | durex, I don't know bsd, but could a linux .so link with a freebsd binary/.so ? |
| 14:08.01 | tzanger | it's a know problem? |
| 14:08.12 | zoa | yes its a know problem with native bridging |
| 14:08.32 | tzanger | damn I didn't know that |
| 14:08.35 | kajtzu | tzanger: mysql 4.1 uses major version 14 for the .so:s |
| 14:08.36 | durex | I have linux binary emulation enabled... maybe it works... |
| 14:08.36 | tzanger | where's the discussion? |
| 14:08.45 | zoa | on mantis there is some stuff about it |
| 14:08.49 | tzanger | damn |
| 14:08.51 | tzanger | I had no idea |
| 14:08.53 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 14:08.54 | tzanger | I can disable native bridging |
| 14:09.11 | tzanger | and in fact I have commented out IAX2_BRIDGE_OPTIMIZATION since it fucks up the new jb pretty good |
| 14:09.24 | durex | sh!t... |
| 14:09.26 | tzanger | and submitted a patch to mantis that ocmpletely rips out the code that that enables as per kram's request |
| 14:09.41 | durex | does somebody have cdr_addon_mysql.so compiled for FreeBSD ???? |
| 14:09.47 | tzanger | so if I kill native bridging entirely it will make that go away |
| 14:09.48 | tzanger | ?? |
| 14:09.55 | zoa | no probably not |
| 14:09.58 | tzanger | oh dammit |
| 14:10.05 | zoa | iax2 has some pretty weird timestamp things |
| 14:10.12 | tzanger | zoa: yes I am working on those with stevek |
| 14:10.29 | zoa | its the timestamps fucking everything up |
| 14:10.36 | tzanger | yes I know |
| 14:10.52 | tzanger | 3007? |
| 14:10.55 | tzanger | is that what you're talking about? |
| 14:11.06 | clive- | there is someone called "grollo" I think, who has apparently fixed up iax timestamping |
| 14:11.29 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 14:12.13 | tzanger | http://bugs.digium.com/view.php?id=3007 |
| 14:12.36 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 14:12.48 | djMax | anybody done anything to speed up polycom sip phone boot time? Seems quite long. |
| 14:12.48 | tzanger | oh wait |
| 14:12.55 | zoa | yes that was one of them |
| 14:13.00 | tzanger | native bridge == IAX_BRIDGE_OPTIMIZATION |
| 14:13.01 | zoa | but there is more similar stuff |
| 14:13.06 | tzanger | which is commented out on my systems |
| 14:13.21 | tzanger | which helped it a great deal but did not eliminate the periodic 10s one-way dead auido |
| 14:13.38 | *** join/#asterisk loick (~loick@APuteaux-151-1-38-85.w82-124.abo.wanadoo.fr) |
| 14:13.55 | zoa | tzanger we only had that with iaxclient old jitter buffer i think |
| 14:14.05 | zoa | that 10s one way dead audio |
| 14:14.06 | tzanger | zoa: this is just * to * |
| 14:14.08 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 14:14.09 | tzanger | no iaxclient anywhere |
| 14:14.15 | zoa | well its the same thing |
| 14:14.22 | tzanger | yes |
| 14:14.25 | zoa | you do have jitter buffers right ? |
| 14:14.28 | tzanger | your comment on april 4 is incorrect |
| 14:14.37 | tzanger | "only native bridging doesn't work" -- I think :-) |
| 14:14.45 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:14.49 | tzanger | I have newjb defined but I have at the moment jitterbuffer=no |
| 14:14.52 | tzanger | in iax.conf on both sides |
| 14:14.55 | tzanger | since it causes issues |
| 14:14.58 | tzanger | with DTMF |
| 14:15.22 | tzanger | actually wait |
| 14:15.33 | tzanger | PRI->A->B->PRI |
| 14:15.34 | tzanger | I have not seen it |
| 14:15.39 | tzanger | neither A nor B are native bridging |
| 14:15.43 | tzanger | but PRI->A->B->C->PRI |
| 14:15.53 | tzanger | B is native bridging, even if BRIDGE_OPTIMIZATION is turned off |
| 14:15.59 | tzanger | and that's where I see it infrequently |
| 14:16.13 | zoa | yes |
| 14:16.17 | tzanger | aha |
| 14:16.17 | blitzrage | morning all |
| 14:16.17 | zoa | 3 asterisk servers |
| 14:16.28 | zoa | but without jb i dont know |
| 14:16.31 | burbankmarc | when i call this is what i get from the debug output: CSeq: 102 BYE |
| 14:16.32 | zoa | didnt see that yet |
| 14:16.38 | tzanger | A, B, and C are with jitterbuffer=no |
| 14:16.43 | *** part/#asterisk Madkiss (madkiss@madkiss.staff.freenode) |
| 14:16.46 | zoa | no idea then |
| 14:16.47 | tzanger | C I think has NEWJB *UN* defined even |
| 14:16.49 | zoa | what could be causing it |
| 14:16.50 | tzanger | C = nufone |
| 14:16.56 | zoa | but we did see something similar before |
| 14:17.06 | adjacent | anyone familiar with dundi? |
| 14:17.09 | blitzrage | whats all this smart talk doing in #asterisk? |
| 14:17.16 | blitzrage | dundi? |
| 14:17.20 | tzanger | blitzrage: shut up and go fetch me a coffee |
| 14:17.30 | blitzrage | tzanger: only those who drink coffee, need coffee |
| 14:17.49 | adjacent | http://www.dundi.info/ |
| 14:17.55 | tzanger | I haven't been sleeping well, I need it |
| 14:18.06 | tzanger | I drank a coffee+hot chocolate this morning |
| 14:18.10 | tzanger | time for a regular cofeee |
| 14:18.10 | blitzrage | tzanger: doesn't matter - you shouldn't need it and shouldn't drink it :) |
| 14:18.15 | tzanger | blitzrage: :-) |
| 14:18.46 | blitzrage | tzanger: that, and you're slightly too far away for me to get you a hot coffee, so it'd be useless |
| 14:18.54 | tzanger | blitzrage: hop to it |
| 14:18.58 | tzanger | you're young and you have a bike |
| 14:19.03 | blitzrage | tzanger: I don't "hop" |
| 14:19.22 | blitzrage | tzanger: I wiped out on a wet streetcar rail the other day and took out my rear brakes :) |
| 14:19.38 | blitzrage | tzanger: snapped the brake handle right off |
| 14:19.46 | burbankmarc | do i need anything else here? exten => _2002,1,Answer exten => _2002,2,Dial(SIP/200x,10) |
| 14:19.46 | jief- | how do you retrieve messages in your mailbox with *? |
| 14:20.13 | blitzrage | burbankmarc: I don't think you udnerstand pattern matching... |
| 14:20.18 | zoa | somebody go try the new sip jitter buffer :p |
| 14:20.29 | tzanger | blitzrage: that'll teach you |
| 14:20.35 | blitzrage | exten => _200X,2,Dial(SIP/${EXTEN},10) |
| 14:20.38 | blitzrage | makes more sense |
| 14:21.03 | blitzrage | zoa: I don't get stats for jitter buffer or latency on CVS HEAD for IAX anymore? |
| 14:21.11 | burbankmarc | ok, that's what i have for the voicemail portion.... |
| 14:21.13 | *** join/#asterisk Donuil (~fpatria@217.9.64.234) |
| 14:21.36 | blitzrage | burbankmarc: the extension part has to be THE SAME for every priority. |
| 14:21.48 | tzanger | blitzrage: is NEWJB defined? |
| 14:22.01 | blitzrage | tzanger: I don't know where that gets defined :) |
| 14:22.15 | frood | that works great |
| 14:22.20 | frood | thanks guys |
| 14:22.43 | *** part/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 14:23.07 | burbankmarc | nice job blitz...you're the man |
| 14:23.25 | bkw_ | "the" man? |
| 14:23.27 | bkw_ | I knew it |
| 14:23.52 | blitzrage | *shakey fist* |
| 14:23.58 | bkw_ | haha |
| 14:24.03 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 14:24.15 | bkw_ | Patrick from canada? |
| 14:24.16 | blitzrage | "shake harda boi!" - Simpsons. |
| 14:24.19 | bkw_ | har har har |
| 14:24.28 | tzanger | bah |
| 14:24.31 | tzanger | bugs' filters dont' work |
| 14:24.37 | Patrick^ | bkw_: yep |
| 14:24.37 | tzanger | I am monitoring a half dozen bugs but the filter doesn't show it |
| 14:24.37 | bkw_ | bugs has bu gs |
| 14:24.41 | bkw_ | ironic eh? |
| 14:24.43 | blitzrage | tzanger: you don't work |
| 14:24.58 | tzafrir | gee, nobody has yet changed the topic? I thought worse about the mental capabilities of folks here |
| 14:25.13 | blitzrage | tzanger: so where is this crazy NEWJB defined? Is it in the Makefile or some .c file or something? |
| 14:25.28 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - Chicago || h.323 rules!!! || bugs has bugs... ironic eh? |
| 14:25.35 | blitzrage | hehehe |
| 14:25.40 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || ClueCon Dev Conf Aug 3rd - 5th - Chicago || h.323 rules!!! || bugs has bugs... ironic eh? (flea dip anyone?) |
| 14:25.58 | blitzrage | mmmmm... flea dip... |
| 14:26.11 | newbien | do i need to install the alsa devel libs before asterisk will use alsa instead of OSS? |
| 14:27.24 | blitzrage | I think ALSA just needs to be working... |
| 14:27.28 | blitzrage | and enabled in modules.conf |
| 14:27.36 | jief- | so, for someone to be able to retrieve their messages, i need to create an extention that uses Voicemailmain2()? |
| 14:27.39 | blitzrage | noload => chan_oss.so |
| 14:27.46 | *** join/#asterisk Corydon76-home (black@pcp08665860pcs.500ash01.tn.comcast.net) |
| 14:27.46 | blitzrage | load => chan_alsa.so |
| 14:28.01 | tzanger | dammit |
| 14:28.14 | blitzrage | jief-: anything which has a '2' on the end I think is the same thing for the non-'2' version - but yes |
| 14:28.42 | *** join/#asterisk kioko (~kiokorobe@196.200.26.42) |
| 14:28.46 | blitzrage | OH PROOFREADING - HOW I LOVE THEE! |
| 14:29.01 | blitzrage | bkw_: more chapters going up for review today |
| 14:29.07 | bkw_ | w00000000000000000000000t |
| 14:29.15 | blitzrage | bkw_: I'm hoping like... 3 more |
| 14:29.38 | blitzrage | bkw_: just finishing up the last 'technical' chapter - the one after that is more fluffy :) |
| 14:29.42 | tzanger | http://bugs.digium.com/view.php?id=3961 |
| 14:29.44 | tzanger | there it is |
| 14:30.00 | tzanger | dammit if I've sumbitted a patch or made a comment on a bug I should automatically be added to hte monitoring list |
| 14:30.10 | tzanger | it's not possible to search for anything I've personally seen or worked on otherwise |
| 14:30.24 | blitzrage | tzanger: true - I thought thats how it worked in bugs-old |
| 14:30.25 | newbien | blitzrage: i get a asterisk warning that alsa was not used and OSS is the default |
| 14:30.32 | tzanger | I'm on bugs-old |
| 14:30.37 | tzanger | well bugs.digium.com not bugs2 |
| 14:30.40 | blitzrage | tzanger: bugs-old == bugs2 |
| 14:30.48 | tzanger | oh they moved it |
| 14:30.48 | blitzrage | afaik |
| 14:30.50 | blitzrage | yah |
| 14:30.58 | blitzrage | you didn't notice the "My View" at the top? |
| 14:31.00 | blitzrage | dead give away :) |
| 14:31.26 | tzanger | blitzrage: my view doesn't show shit |
| 14:31.43 | tzanger | bug 3961 is not in "my view" |
| 14:31.50 | tzanger | oh wait |
| 14:31.51 | tzanger | it si |
| 14:31.53 | tzanger | er is |
| 14:31.56 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 14:31.56 | Nugget | heh |
| 14:32.01 | carbon60 | Morning all. |
| 14:32.01 | tzanger | but under "recently modified" |
| 14:32.22 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 14:32.35 | carbon60 | What is simplest/safest web app to use that will simply display SIP users's status? All I need is something that users can look at to see who is on the phone. |
| 14:32.43 | blitzrage | tzanger: yah - should be under Monitored By Me |
| 14:32.48 | tzanger | it's not |
| 14:32.52 | zoa | flash operator panel probably |
| 14:32.53 | tzanger | there is nothing under "monitored by me" |
| 14:33.03 | blitzrage | tzanger: oh reeeeeeeeally |
| 14:33.13 | blitzrage | tzanger: probably because you weren't using bugs2? :) |
| 14:33.22 | blitzrage | tzanger: but I agree with you |
| 14:34.01 | *** join/#asterisk Corydon76-home (pink@pcp08665860pcs.500ash01.tn.comcast.net) |
| 14:34.03 | blitzrage | anyone ever get a SIP NOTIFY to reboot a 7960? |
| 14:34.04 | tzanger | blitzrage: as I said I'm using bugs.digium.com |
| 14:34.10 | tzanger | bugs2 wouldn't let me log in or do anything about 2 weeks ago |
| 14:34.12 | blitzrage | tzanger: yah, but its been moved over... |
| 14:34.32 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:34.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:34.41 | blitzrage | tzanger: not sure what issues its caused - I think most of the other main devs have been using it for weeks now? |
| 14:34.52 | *** join/#asterisk djMax (~chatzilla@artsalliancelabs.com) |
| 14:34.55 | blitzrage | tzanger: so you might be one of only a handful of active devs who weren't using it... ? |
| 14:35.15 | newbien | blitzrage: do i: load => chan_alsa.so from the asterisk CLI? |
| 14:35.53 | tzanger | blitzrage: huh? |
| 14:35.53 | djMax | if I have "conf => 100,1234" in meetme.conf, and I do "meetme,100||1234" in ext.conf, shouldn't it work? |
| 14:36.44 | blitzrage | newbien: modules.conf |
| 14:37.58 | blitzrage | tzanger: I'm just thinking that other people have been using bugs2, so they didn't notice any change during the rollover, but you did since you weren't actively using it... |
| 14:38.19 | tzanger | dammit |
| 14:38.24 | tzanger | bugs2 should have fixed the patch problem |
| 14:38.37 | newbien | blitzrage: alsa works fine long before i installed asterisk; works before and after asterisk is run |
| 14:38.51 | RoyK | anyone that knows what I can use for IAX2/H.323 gatewaying? |
| 14:39.04 | tzanger | I want to be able to copy a link and wget it in another window and get the right fucking filename, not 'http://bugs.digium.com/file_download.php?file_id=5725&type=bug' |
| 14:39.35 | blitzrage | tzanger: I agree!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 14:39.51 | anthm | needs a mime header to tell you the disposition |
| 14:39.55 | tzanger | uuuuuuuuuuuuuuuuuuugh |
| 14:39.59 | tzanger | now they added a fucking cookie too |
| 14:40.01 | blitzrage | newbien: load => chan_also.so in modules.conf and verify the module loads at startup and it exists in /var/lib/asterisk/modules/ |
| 14:40.02 | tzanger | dammit |
| 14:41.11 | newbien | blitzrage: k, will try it, thanks |
| 14:42.22 | OloBola | I built an asterisk voicemail to mp3 to IIS to SQL Server streaming flash thinga majiggy |
| 14:42.29 | blitzrage | newbien: you didn't try it after the first two times I said it? |
| 14:43.13 | newbien | blitzrage: only now do i understand what you meant ;) |
| 14:43.48 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 14:44.51 | djMax | in theory, would it be possible to have a meetme option that allowed an individual participant to increase or decrease their volume? |
| 14:45.34 | bkw_ | djMax, you got about 7k on you? |
| 14:45.35 | blitzrage | djMax: I know a certain someone who has it working |
| 14:45.48 | djMax | is that a bounty or something? |
| 14:45.58 | bkw_ | well i'm sure 7k or so would make it be released |
| 14:46.03 | bkw_ | ;) |
| 14:46.39 | djMax | heh. Well, I guess that would make it theoretically possible then. |
| 14:46.45 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 14:46.57 | djMax | I'll pay $500. :) |
| 14:47.51 | djMax | (by the way, I got around the meetme pwd problem by just using Authenticate() but not sure why it wasn't working with normal meetme.conf) |
| 14:48.03 | blitzrage | djMax: did you reload the meetme app? |
| 14:48.13 | djMax | yeah, even restarted the server. |
| 14:48.19 | anthm | you wanna pay 500 bux to change the vol in meetme ? |
| 14:48.38 | djMax | I would pay $500 if there was a feature that let an individual caller in a meetme control their gain. |
| 14:48.43 | Juggie | ohh anthm, thats a start to your 2k pot for app_confcall :) |
| 14:49.04 | djMax | (I would pay $500 to bkw_ because he's been very helpful to me in the past and I've definitely gotten more than $500 worth of value from *) |
| 14:49.09 | tzanger | trying grolloj's 3965 patches |
| 14:49.18 | tzanger | see if they fix dtmf and 10s audio drops |
| 14:49.22 | tzanger | trunking would be nice to try too :-) |
| 14:49.30 | anthm | oh then you're out of luck cos I'm the one who can do it =D |
| 14:49.35 | tzanger | dammit where's jerjer |
| 14:49.36 | anthm | bye |
| 14:49.41 | djMax | ok, well then I'd pay you instead. :) |
| 14:49.45 | anthm | ok im back |
| 14:50.13 | djMax | I remember the other thing I wanted... Stealing calls from VM. That one I just think we need for home use, so I got less money there. :) |
| 14:50.25 | blitzrage | lol |
| 14:50.31 | djMax | but seriously, our company would happily pay $500 to have that feature in *. |
| 14:50.37 | blitzrage | tzanger: need me to try so patches to help you test? |
| 14:50.42 | blitzrage | s/so/some |
| 14:51.56 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 14:52.27 | ilium007 | hi - first time in here |
| 14:53.10 | ilium007 | just wanted to say hello - i have just stumbled across asterisk !!!! WOW |
| 14:53.11 | eKo1 | be wary of the Asterisk de-virginazers |
| 14:53.16 | blitzrage | lol |
| 14:53.19 | ilium007 | heheh |
| 14:53.20 | blitzrage | their rough :) |
| 14:53.26 | blitzrage | they're* |
| 14:53.29 | ilium007 | this is awesome |
| 14:53.34 | puowvip | condoms $5 |
| 14:53.50 | Nugget | without ME this channel is just AWESO. |
| 14:53.50 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 14:53.50 | puowvip | get 'em while they're ....NEVERMIND |
| 14:53.51 | eKo1 | ask durex for condoms |
| 14:54.04 | ilium007 | especially considering we are looking at spending $100k on a phone system |
| 14:54.19 | blitzrage | hi, I'd like to introduce myself |
| 14:54.21 | blitzrage | :) |
| 14:54.27 | Nugget | hah |
| 14:54.33 | RoyK | ~lart blitzrage |
| 14:55.04 | sivana | does 'reload' also reload the zapata? |
| 14:55.11 | ilium007 | question - i have seen some of the digium hardware, what do you use to connect traditional phones to asterisk ? |
| 14:55.12 | blitzrage | you guys are just jealous of my l33t sk1llz |
| 14:55.14 | Nugget | apple delayed shipment on my powermac by nine more days. you may all point at me and laugh cruelly now. |
| 14:55.19 | RoyK | sivana: yes |
| 14:55.23 | RoyK | but not zaptel :P |
| 14:55.48 | blitzrage | ilium007: TDM400P or a T1 card attached to a channel bank (Adit or Adtran are good choices) |
| 14:56.03 | ilium007 | channel bank ?? |
| 14:56.29 | sivana | ah ok |
| 14:56.33 | sivana | I knew it was one of those |
| 14:56.50 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 14:56.57 | zoa | royk, the sip jitter buffer is declared quite stable |
| 14:57.34 | *** join/#asterisk kisu (~Snake@218.237.126.163) |
| 14:57.35 | blitzrage | ilium007: a channel bank allows you to have a higher density of FXS and FXO channels than what you could get with multiple TDM400P cards in a single machine |
| 14:57.49 | RoyK | zoa: nice |
| 14:57.51 | ilium007 | oh ok cool |
| 14:57.53 | RoyK | zoa: where is it? |
| 14:58.17 | ilium007 | blitzrage: can you give me a model number ? |
| 14:58.26 | blitzrage | ilium007: Adit 600 or Adtran 750 |
| 14:58.37 | zoa | http://www.astertest.com/forum/viewtopic.php?t=22 |
| 14:58.44 | blitzrage | ilium007: adtran 750 off of eBay is cheaper than new |
| 14:59.04 | puowvip | is the only way to get dialed number information to use PRI? |
| 14:59.05 | blitzrage | ilium007: I think netxusa sells them new |
| 14:59.19 | ilium007 | ok cool |
| 15:00.08 | bjohnson | puowvip: no |
| 15:00.12 | felipex | exten => s,2,Dial(SIP/201,,tT) |
| 15:00.21 | felipex | in case of SIP/201 is busy |
| 15:00.22 | bjohnson | puowvip: most voip providers will feed through dialed number |
| 15:00.31 | felipex | how can i call other sip ? |
| 15:00.44 | felipex | which extension i have to use? |
| 15:00.48 | bjohnson | puowvip: and if you use a fxo to connect an analogue line, you will know which fxo it came from |
| 15:01.14 | bjohnson | felipex: use the superdial macro |
| 15:01.17 | bjohnson | on the wiki |
| 15:01.29 | christo | when somebody calls SIP extension 3901 it drops to Voicemail after 10 secs. Great. When somebody calls XXXXX810597 I want * to forward to the same sip extension. The sip extension rule is in context [sip-local] so what would the Goto() command look like to pass through to sip-local/3901 ? |
| 15:01.35 | PCadach | oej & bkw_: AST_CONTROL_HOLD is just notification, not real control. So, remote party placed call on hold should start MOH or something else. |
| 15:01.46 | ilium007 | blitzrage: very interesting - i am literally lying here in bed - 1.00am i cant sleep !!!! this is the best stuff i have seen in ages ! |
| 15:02.58 | blitzrage | ilium007: welcome to the addiction |
| 15:03.10 | blitzrage | ilium007: I've been dealing with it for just over 2 years now... :) |
| 15:03.28 | puowvip | bjohnson, I must be using the wrong voip provider (broadvoice) then. |
| 15:03.40 | blitzrage | the 2 unemployed months I had 2 years ago was the best thing that ever happened to me :) |
| 15:03.50 | blitzrage | puowvip: yah... use mixnetworks.com :) |
| 15:04.17 | puowvip | hopefully they have my area code |
| 15:04.21 | puowvip | looking |
| 15:04.28 | HeadachesAbound | * is not an addiction. The last thing I need is another addiction. And besides, I can stop if I want to. I just don't want to. |
| 15:05.26 | blitzrage | puowvip: we have most (all?) area codes |
| 15:05.32 | Donuil | hi to all... I have installed an h.323 gatekeeper on a pc with oh323 and phonejack card correctly... I try to call h323 phone with a Sjphone trough asterisk but I have an error... asterisk says that the no channel type registred for 'H323'.. someone can help me? |
| 15:05.35 | jsharp | Just one more IAXing. Just one more. Then I'll quit. I swear. |
| 15:05.49 | jsharp | Donuil: Did you go through the procedure of building chan_h323? |
| 15:05.59 | jsharp | Its entirely separate from building the main asterisk tree. |
| 15:06.01 | blitzrage | puowvip: what do you need, I think I need to look it up in my list for you - I haven't had time to do anything with the website yet (I didn't make it) |
| 15:06.54 | ilium007 | blitzrage: would most people be looking at a VOIP / SIP solution rather than the traditional ADSI method ? |
| 15:07.19 | eKo1 | traditional? |
| 15:07.23 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 15:07.24 | blitzrage | ilium007: ADSI? |
| 15:07.38 | puowvip | blitzrage, area code 614. downtown columbus, upper arlington, grandview, hilliard, dublin are all OK |
| 15:08.03 | puowvip | and $1 says you don't have any area code 867 :-) |
| 15:08.11 | jief- | ok, in voicemail.conf, i have 501 => 1234,some name,someone@domain.com |
| 15:08.13 | ilium007 | blitzrage: newbie |
| 15:08.15 | blitzrage | ilium007: My solution includes using SIP for local phones, IAX for VoIP trunks (service providers) and a pair of backup copper lines (analog) |
| 15:08.21 | jief- | so 1234 would be the password for mailbox 501 right? |
| 15:08.27 | blitzrage | puowvip: let me look :) |
| 15:08.34 | ilium007 | ok cool |
| 15:09.00 | puowvip | broadvoice was the only provider offering 614 that didn't require me to use their !#$@ ATA |
| 15:09.55 | puowvip | no thanks Vonage, no thanks Time Warner, no thanks Speakeasy |
| 15:10.21 | Donuil | jsharp I have installef the asterisk support oh323 that include all drivers and it makes the chan_oh323.so module... |
| 15:10.29 | blitzrage | puowvip: I have lots of 614 in columbus |
| 15:10.39 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 15:11.09 | blitzrage | puowvip: you're right, I don't have 867 :) |
| 15:11.35 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 15:11.53 | puowvip | I'd *love* to have a virtual number in Inuvik, NWT just to fuck with my dad's head |
| 15:12.05 | blitzrage | puowvip: hahaha, that's where that is? :D |
| 15:12.16 | ilium007 | blitzage: can you suggest some reading for a relative newbie on the voice scene, i have been involved in purchasing and specing traditional phone systems, but i am very interested in this |
| 15:12.28 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 15:12.28 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 15:12.35 | puowvip | 867 ("TOP") is Yukon, NWT, and Nunavut, Canada |
| 15:12.40 | blitzrage | ~docs |
| 15:12.41 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:12.49 | blitzrage | ilium007: ^^^ |
| 15:13.14 | ilium007 | cool |
| 15:13.34 | blitzrage | puowvip: yah, for Canada I just have Calgary, Edmonton, Vancouver, Victoria, Winnipeg, Hamilton, Kitchener, London, Ottawa, Toronto, Windsor, Montreal |
| 15:13.42 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
| 15:13.54 | Juggie | blitzrage, gimmi one in ottawa :) |
| 15:14.00 | puowvip | the lower provinces |
| 15:14.09 | Donuil | hi to all... I have installed an h.323 gatekeeper on a pc with oh323 and phonejack card correctly... I try to call h323 phone with a Sjphone trough asterisk but I have an error... asterisk says that the no channel type registred for 'H323'.. someone can help me? |
| 15:14.26 | jsharp | Donuil: Did you go through the procedure of building chan_h323? |
| 15:14.28 | jsharp | Its entirely separate from building the main asterisk tree. |
| 15:14.44 | puowvip | blitzrage, you in here often? |
| 15:15.20 | Donuil | jsharp yes I have installef the asterisk support oh323 that include all drivers and it makes the chan_oh323.so module... |
| 15:15.20 | jsharp | is it loaded? |
| 15:15.20 | jsharp | load chan_oh323.so |
| 15:15.20 | Donuil | yes |
| 15:15.38 | blitzrage | puowvip: I'm in here nearly all the time - mostly in #asterisk-doc |
| 15:15.49 | jsharp | Ohwait. If you're using chan_oh323, you need to dial using OH323 as the technology, not H323. |
| 15:16.11 | tzafrir | how do I kill a call on a zap channel? short of hanging it up manually? |
| 15:16.24 | Blackvel | kill a call? hangup? |
| 15:16.31 | puowvip | blitzrage, one more question :-) your business plan says "1 DID and 1 extension". does that simply mean a max of two concurrent calls in any direction? |
| 15:16.34 | blitzrage | puowvip: if you go with mixnetworks, tell them Leif sent you |
| 15:16.35 | tzafrir | hangup |
| 15:16.36 | Blackvel | you can use zap as you do with sip |
| 15:16.50 | Donuil | Is it possible an exten problem? I use exten => 5552,1,Dial(H323/oh323:phonejack) |
| 15:16.58 | bjohnson | puowvip: actually, Ontario is historically referred to as Upper Canada |
| 15:17.19 | bjohnson | (and Quebec was Lower Canada) |
| 15:17.25 | carbon60 | Do I need chan_zap.so if I'm only using ztdummy for timing? |
| 15:17.26 | blitzrage | puowvip: thats something I need to clarify - I just started maintaining the backend and haven't dealt with what the really means yet :) |
| 15:17.30 | puowvip | bjohnson: thanks. Don't want to offend canadians :) |
| 15:17.39 | jsharp | Donuil: You need to dial with Dial(OH323, not Dial(H323 |
| 15:17.49 | jsharp | chan_oh323 uses a different technology name. |
| 15:18.11 | onkeltimm | if a friggin coward like me wanted to upgrade to CVS-HEAD (currently running 1.0.3), because his boss wants attended transfers, how would he go about this |
| 15:18.12 | bjohnson | yes, we'll throw balls of Pine tree sap at you |
| 15:18.16 | bjohnson | it's very sticky |
| 15:18.20 | blitzrage | puowvip: damn right - we don't want to be put into the same group as those crazy french :) |
| 15:18.28 | tzafrir | hmmm, isn't there a simple way to kill an ongoing call from the CLI? |
| 15:18.32 | puowvip | bjohnson, any hints how to get canadian companies to even consider my resume? |
| 15:18.37 | blitzrage | and we all know how uncomfortable it is to be sticky |
| 15:18.37 | jsharp | soft hangup |
| 15:18.39 | ilium007 | ok dumb ass question coming up |
| 15:18.40 | carbon60 | tzafrir: sofhangup |
| 15:18.41 | jief- | i have a login problem with my voice mailboxes. i have defined them under voicemail.conf and added them user mailbox=50X for each phones under sip.conf. but i still can't log and get my messages |
| 15:18.50 | carbon60 | Yeah, that. |
| 15:18.52 | ilium007 | digium make a tdm400p |
| 15:18.54 | bjohnson | useless fact - amber (the prized jewel) is actually fossilized tree sap |
| 15:19.12 | tzafrir | carbon60, thanks. I managed to miss that one |
| 15:19.14 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 15:19.15 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 15:19.22 | Zeeek | ilium007 read these two short articles^^^^^^^^^^ |
| 15:19.23 | ilium007 | why would you use many of these as FXO's instead of buying a chanell bank |
| 15:19.40 | bjohnson | ilium007: not enough needed to buy a channel bank |
| 15:19.42 | ilium007 | reading |
| 15:19.49 | tzafrir | ilium007, how many pci slots do you have? |
| 15:19.55 | Zeeek | you asked for a start: these are basic but a good intro |
| 15:19.58 | ilium007 | thats what im getting at |
| 15:20.06 | jief- | this is the output from *: http://pastebin.ca/10902 |
| 15:20.07 | ilium007 | say i need 40 extensions in an offive |
| 15:20.11 | ilium007 | how would you do it |
| 15:20.17 | Zeeek | FXS |
| 15:20.20 | jsharp | 2 channel banks & 2 T1 cards. |
| 15:20.23 | Zeeek | FXO are for the phone lines |
| 15:20.24 | bjohnson | yes |
| 15:20.28 | bjohnson | exactly |
| 15:20.33 | bjohnson | agreed all around |
| 15:20.38 | blitzrage | third |
| 15:20.49 | blitzrage | its the only way to scale analog lines |
| 15:20.55 | carbon60 | If I get "/usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_dump_info_str", does this mean my kernel module is out of date or libzap is out of date, or what? |
| 15:21.01 | Zeeek | so, blitz, you fell off your bike? |
| 15:21.03 | jsharp | libpri is out of date. |
| 15:21.20 | *** join/#asterisk cmk (~cmk_@p54A3ED3F.dip.t-dialin.net) |
| 15:21.26 | blitzrage | Zeeek: haha, yah, I fell correctly though - only a small road rash on the inside of my left arm |
| 15:21.32 | ilium007 | oh ok |
| 15:21.46 | blitzrage | Zeeek: unfortunately the rear brake handle sheered right off - cheap plastic I guess... :( |
| 15:22.23 | ilium007 | jsharp: what chanell banks do you recommend |
| 15:22.43 | blitzrage | ilium007: didn't I mention Adit 600 and Adtran 750? :) |
| 15:23.03 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:23.08 | ilium007 | i was looking for other options :) |
| 15:23.18 | jsharp | Carrier Access Access Bank II |
| 15:23.31 | jsharp | Practically a dime a dozen. |
| 15:23.36 | drumkilla | blitzrage: !!!! |
| 15:23.40 | ilium007 | so the product is just an external tdm device |
| 15:23.49 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 15:23.54 | jsharp | Yup. |
| 15:25.07 | jief- | ok, im gonna try one last time to get help here with this. When i try to access a mailbox, i get a login error. here is the * output: http://pastebin.ca/10902 |
| 15:25.07 | *** join/#asterisk gpuk (~me@mail.angloeuropean.com) |
| 15:25.24 | tzafrir | carbon60, incompatible version of libpri? |
| 15:25.35 | Zeeek | jief- DTMF isn't working |
| 15:25.39 | jsharp | Its not reading the password. |
| 15:25.46 | Zeeek | exactly |
| 15:25.46 | jsharp | What Zeeek said. |
| 15:25.49 | tzafrir | carbon60, what version of zaptel do you use? from where? |
| 15:25.55 | HeadachesAbound | yep, what they said. |
| 15:26.07 | Zeeek | and this can be because of the codec depending on the phone |
| 15:26.15 | Zeeek | plus what the rest say |
| 15:26.21 | Zeeek | and what I said :) |
| 15:26.25 | blitzrage | drumkilla: !!! |
| 15:26.30 | blitzrage | drumkilla: how went the exams?! |
| 15:26.42 | drumkilla | one down, one to go in an an hour and a half |
| 15:26.46 | jsharp | Make sure DTMF mode in sip.conf matches what your phone is sending. Inband/SIP INFO/RFC2883 |
| 15:26.48 | drumkilla | the first one raped me pretty bad |
| 15:27.11 | carbon60 | tzafrir: I had an old libpri1 from Debian. |
| 15:27.12 | drumkilla | I probably spent too much time with Asterisk intead of the book for the class :( |
| 15:27.12 | *** join/#asterisk Dougnaka (~Doug@207.225.223.185) |
| 15:27.20 | Dougnaka | anyone here use nufone? |
| 15:27.31 | *** join/#asterisk ethogeek (~ethogeek@mke-64-201-64-200.genevaonline.com) |
| 15:27.36 | jief- | Zeeek: well, in sip.conf, i have set my phones to inband, and the phones are set to inband also |
| 15:27.38 | carbon60 | tzafrir: I've been using the same pair of zaptel kernel modules for a few months, too lazy to rebuild them. |
| 15:27.39 | zoa | drumkilla: good luck! |
| 15:27.43 | blitzrage | drumkilla: I don't think you spent ENOUGH time with Asterisk :) |
| 15:27.45 | drumkilla | thanks! i'm off again ... |
| 15:27.47 | tzafrir | carbon60, upgrade everything together. latest versions applied bristuff, which creates some incompatibitilites. |
| 15:27.48 | zoa | thats plenty of time to check out the jitter buffer :p |
| 15:27.48 | drumkilla | blitzrage: yeah, I agree |
| 15:27.49 | jsharp | jief-: What codec are your phones using? |
| 15:27.52 | Zeeek | jief- inband won't work with mail IIRC |
| 15:27.56 | ilium007 | none of these devices i am looking at seems to be loaded up with RJ45 ports |
| 15:28.14 | bjohnson | ilium007: you need hardware for that |
| 15:28.24 | jief- | jsharp: ulaw |
| 15:28.24 | ilium007 | am i on the same wavelength ? should these tdm devices have cards in them with multiple rj45 ports |
| 15:28.24 | jsharp | ilium007: You'll need an external patch panel. Most of them spit 24 lines out on a 50 pin amphenol connector. |
| 15:28.24 | jief- | Zeeek: ok, which one should i use then? |
| 15:28.33 | Zeeek | usually RFC |
| 15:28.33 | tzafrir | carbon60, as a bonus, building the kernel modules has become very simple. module-assistant makes it a matter of one command or so |
| 15:28.35 | carbon60 | tzafrir: Yeah, would have been nice for debs to declare dependancies on newer versions. |
| 15:28.36 | ilium007 | oh ok |
| 15:28.39 | Zeeek | but I think both work |
| 15:28.45 | carbon60 | module-assistant? |
| 15:29.01 | bjohnson | ilium007: feed from channel bank comes out as one big cable with a hundred twisted pair .. you add your own rj45 |
| 15:29.04 | tzafrir | carbon60, I believe they do. If they don't, file a bug. Newer ones should. |
| 15:29.13 | carbon60 | Hrm. |
| 15:29.20 | ilium007 | i get it now :) |
| 15:29.37 | jief- | Zeeek: my phones support in/outbound and sip info |
| 15:29.38 | ilium007 | ok so back to channel banks |
| 15:29.38 | tzafrir | carbon60, I'm leaving now. Hopefully tzafrir_laptop will be here in an hour or so |
| 15:29.59 | ilium007 | are they expandable or do you just buy the whole thing in lots of say 24 or 48 |
| 15:30.21 | bjohnson | just buy them loaded .. if buying used you won't save anything buying half now |
| 15:30.23 | tzafrir | carbon60, and if you're too lazy to build those modules, http://tzafrir.org.il/rapid/APT.html |
| 15:30.29 | HeadachesAbound | jief: what make / model of phones? |
| 15:30.40 | jsharp | You can get them in 24 port models or you can get big chassis that take multiple T1s and have lots of card slots for port cards. |
| 15:30.45 | jief- | HeadachesAbound: Gnet SIP phones |
| 15:30.56 | HeadachesAbound | Gnet SIP Model? |
| 15:30.59 | carbon60 | Thanks! |
| 15:31.09 | jief- | HeadachesAbound: P104SLD |
| 15:31.37 | bjohnson | jsharp: he only needs 40 fxs iirc so he'll just be looking at the 24 port models |
| 15:31.38 | ilium007 | ok cool |
| 15:31.57 | jsharp | Ohyeah. |
| 15:32.03 | jsharp | just get 2 24 port models. |
| 15:32.27 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 15:32.33 | ilium007 | ok so when we are talkin FXS's are these what we would traditionally call an analogue extension or a digital extension - does asterisk support this notion |
| 15:32.45 | jsharp | Analogue extensions. |
| 15:32.52 | bjohnson | fxs is for analogue only |
| 15:32.54 | jsharp | Asterisk doesn't support digital phones. |
| 15:32.55 | jief- | according to voip-info.org, inband + ulaw + voicemail = no go |
| 15:33.09 | gpuk | hello all - we are looking at moving our entire company phone system over to asterisk. The only question we have is this: how do people route non-voip calls to keep costs down? |
| 15:33.10 | gpuk | Obviously we could route over the national network (in our case BT in the UK) but ideally we'd like to do something along the lines of Skype Out - i.e. use a low-cost carrier for non-voip calls. |
| 15:33.22 | bjohnson | you have 2 options for digital phones: 1. a voip phone 2. a digital phone with an adapter to make it voip |
| 15:33.34 | bjohnson | nobody does 2 |
| 15:33.54 | bjohnson | although the new nortel bcm50 looks interesting .. but I haven't asked for pricing info |
| 15:33.58 | jsharp | gpuk: Find a voip carrier that has rates that you like. |
| 15:34.00 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 15:34.05 | jief- | alright, using outband it worked |
| 15:34.05 | jsharp | Or multple carriers if you need to. |
| 15:34.06 | ilium007 | ok |
| 15:34.09 | blitzrage | bjohnson: will that phone work with unistim? |
| 15:34.21 | Deryl | (SecurityFocus Vulns) Vulns: CVS Unspecified Buffer Overflow And Memory Access Vulnerabilities < http://www.securityfocus.com/bid/13217?ref=rss > |
| 15:34.21 | bjohnson | what phone? |
| 15:34.33 | ilium007 | i am looking on ebay for chanel banks without much luck - i dont quite undrstand the terms used |
| 15:34.33 | Deryl | just an fyi. it's been updated today. |
| 15:34.59 | bjohnson | ilium007: read enough posts, you'll figure it out |
| 15:35.00 | _Vile | ilium, look for Carrier Access, FXS |
| 15:35.12 | _Vile | search title and description |
| 15:35.15 | bjohnson | I think he's referring to BCU, PSU, etc |
| 15:35.21 | _Vile | they go for ~$125-$150 |
| 15:35.31 | _Vile | or look for Mainstreet or Newbridge |
| 15:35.35 | ilium007 | jeez is that all ? |
| 15:35.37 | blitzrage | bjohnson: the Nortel phone you mentioned |
| 15:35.44 | bjohnson | I keep hearing that .. but I've never seen any actually go for ess than $250 + shipping |
| 15:35.48 | _Vile | depends on what you want to use them for |
| 15:35.55 | bjohnson | blitzrage: it's FOR Nortel phnes |
| 15:35.55 | *** part/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 15:35.58 | _Vile | if you're looking to for a channel bank to talk to the PSTN |
| 15:36.05 | _Vile | then you need to look for Carrier Access, FXO |
| 15:36.18 | _Vile | which I don't know if you'll find, and don't know what the price range is |
| 15:36.24 | bjohnson | the bcm50 is like a voip gateway .. can do 12 nortel digital phones plus some analogue plus some voip |
| 15:36.33 | bjohnson | licensing activates how many it can handle |
| 15:36.35 | _Vile | and if you're looking for that, look for AdTran 650s |
| 15:36.37 | blitzrage | bjohnson: oh, lol |
| 15:36.44 | _Vile | Adtran 650, FXO |
| 15:37.16 | wiz8291 | anyone got a second to help with some echo problems? |
| 15:37.30 | bjohnson | adit 600 are supposed to be ok for fxo/fxs mixed banks too |
| 15:37.31 | wiz8291 | on incoming calls, when the calling party speaks, they can hear a sort of distortion |
| 15:37.38 | wiz8291 | the desk phones are saysons |
| 15:38.09 | bjohnson | distortion or echo? |
| 15:38.14 | bjohnson | or both |
| 15:38.16 | wiz8291 | distortion |
| 15:38.23 | _Vile | bj, yep |
| 15:38.33 | bjohnson | what is the pstn connection? |
| 15:38.42 | bjohnson | PRI? |
| 15:38.43 | wiz8291 | ISDN30 |
| 15:38.44 | wiz8291 | yeah |
| 15:38.44 | bjohnson | or fxo |
| 15:38.45 | wiz8291 | PRI |
| 15:38.57 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 15:39.05 | bjohnson | you got me on that one .. maybe cpu overloaded on *? |
| 15:39.07 | *** join/#asterisk Fddayan (~fddayan@66.240.80.130) |
| 15:39.14 | *** part/#asterisk Fddayan (~fddayan@66.240.80.130) |
| 15:39.18 | wiz8291 | no load at all |
| 15:39.23 | wiz8291 | its a dual proc with over 1GB ram |
| 15:39.27 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 15:39.30 | wiz8291 | and this is with only one active call |
| 15:39.37 | bjohnson | it all depends on what all is running on it |
| 15:39.49 | wiz8291 | asterisk, on its own |
| 15:39.51 | wiz8291 | there is no load |
| 15:39.55 | bjohnson | but sounds like cpu and mem should b ok |
| 15:40.04 | bjohnson | hmm |
| 15:40.09 | bjohnson | what codec to phones? |
| 15:40.14 | bjohnson | sayson are SIP right |
| 15:40.26 | wiz8291 | nah, these are analogue |
| 15:40.32 | wiz8291 | on a rhino channel bank |
| 15:42.14 | jsharp | Is your T1 card sharing an interrupt or anything odd like that? |
| 15:43.45 | Seyr | Anyone know why I can play TTS at a sampling rate of 8000, but when I change it to 16000, it wont play through *? |
| 15:44.11 | Seyr | * just hangs up.... |
| 15:45.09 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 15:45.20 | ilium007 | From the Digium site: The TDM400P takes the place of an expensive channel bank and brings the system price point to a low level. |
| 15:45.36 | ilium007 | how can a 4 port FXS be a replacement for a channel bank ? |
| 15:45.49 | jsharp | It is if you only need a few ports. |
| 15:45.55 | ilium007 | i mean to get up to say 24 fxs you woudl need 6 pci cards ? |
| 15:45.55 | wiz8291 | nope |
| 15:46.05 | wiz8291 | T1 card is on its own IRQ, not sharing |
| 15:46.24 | jsharp | For a while, there was a gap...either you had a single port card or you had to get a T1+channel bank. No in between. |
| 15:46.40 | wiz8291 | any ideas on distortion? |
| 15:46.41 | ilium007 | excuse my ignorance, but how many applications would only require 4 x fxs |
| 15:46.49 | jsharp | Small office PBX. |
| 15:46.55 | ilium007 | ok cool |
| 15:47.34 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 15:47.39 | jsharp | Or a bigger PBX that used voip phones for most of the extensions, but had a few analogue ports for cordless phones or faxes. |
| 15:48.11 | eper-werk | hmm ok i'm a nub however, a quad ISDN card has 4 ports on it, however the 8 channel isdn that comes in goes into this little box on the wall and looks like 1 single rj-45 comes out into the old telephone box, so do i need a quad card ?(this is UK isdn its funky) |
| 15:48.36 | Gand_DJ | I was wondering.. has anyone tried a Sipura 3000 and the TDM400P card? Do they both have issues with echo cancellation & stuff, etc? |
| 15:49.09 | wiz8291 | eper-werk: you on 2e or 30e ? |
| 15:49.19 | eper-werk | honestly? not an idea |
| 15:49.35 | eper-werk | its a box on the wall with BT on it and two like BNC connectors on it |
| 15:49.49 | eper-werk | we have like 12 DDI's aswell if thats any help :] |
| 15:50.16 | Seyr | Can Asterisk NOT stream a wav with a sample rate of over 8000? |
| 15:50.43 | jsharp | In general, everything in * needs to be sampled at 8000. That's native telephony sample rate. |
| 15:50.50 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 15:51.15 | Seyr | ahh... so when it looks at the file, if the rate is over 8000, it just bombs out |
| 15:51.27 | HeadachesAbound | experience says so. |
| 15:52.15 | jsharp | eper-werk: Sounds like you've probably got a 30E circuit. You'll need the TE110 single port T1/E1 card, not the quad ISDN card. |
| 15:52.24 | wiz8291 | BNC? |
| 15:52.25 | wiz8291 | weird |
| 15:52.40 | wiz8291 | eper-werk: its probably ISDN2 |
| 15:52.42 | eper-werk | is there anything on the box that might be able to tell me what it is :P |
| 15:52.52 | wiz8291 | nah, call up BT and give them the main number |
| 15:52.54 | wiz8291 | they'll tell you |
| 15:52.56 | jsharp | find a model number and google it? |
| 15:53.15 | wiz8291 | i've never seen BNCs from BT though |
| 15:53.23 | wiz8291 | only on megastream circuits |
| 15:53.38 | eper-werk | ok well i say "bnc" because its a round plug and thats the closest connector i know of it :) |
| 15:53.53 | eper-werk | ok it's an ISDN 30 |
| 15:54.31 | wiz8291 | in the middle of the 2 BNCs, there should be an RJ45 plug |
| 15:54.38 | onkeltimm | guys, can I just edit the Makefiles in zaptel, libpri and asterisk setting install prefix to st like /test and get a second test install on my * box? |
| 15:55.14 | eper-werk | ISDN 30 I.421 |
| 15:55.17 | *** part/#asterisk gpuk (~me@mail.angloeuropean.com) |
| 15:55.40 | jsharp | If you want to hang an asterisk box off it, get thee a Digium TE110P card, then. |
| 15:55.55 | wiz8291 | thats what i have |
| 15:56.06 | wiz8291 | eper-werk: its a euroisdn circuit |
| 15:56.16 | DEEZED | wois h.323 better than G.729? |
| 15:56.16 | eper-werk | ah and what card have you got? |
| 15:56.17 | Godsey | might anyone be able to recomend a good customer billing software package? |
| 15:56.33 | Godsey | this isn't asterisk related really, but I'm sure someone here bills customers :) |
| 15:56.55 | HeadachesAbound | i use a guy named Guido from Jersey and he does a great job. |
| 15:56.57 | jsharp | DEEZED: h.323 is nothing like G.729. H.323 is a protocol and G.729 is an audio codec. |
| 15:57.03 | HeadachesAbound | :) |
| 15:57.23 | ilium007 | night all |
| 15:57.23 | jsharp | Narn Bat Squad Collections, Inc |
| 15:57.55 | Godsey | HeadachesAbound: thanks, I don't assess finance charges so I won't need that sort of package :) |
| 15:58.08 | *** part/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
| 15:58.29 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 15:58.29 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:58.44 | HeadachesAbound | we don't assess charges either, but we do accept donations, and Guido is very good at getting donations :) |
| 15:59.05 | *** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-211.ptr.terago.ca) |
| 15:59.57 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 16:00.44 | eper-werk | i need a remote client VoIP phone -> broadband -> cload -> broadband -> our phone server -> telephone network via ISDN channel - Do i need another card for the VoIP and can asterisk do this ( the boss wants to record agents making calls to client so needs this setup) |
| 16:00.45 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:02.30 | jsharp | You just need the ISDN card to the PSTN. Asterisk should be able to monitor & record the calls. |
| 16:02.32 | *** join/#asterisk jskcr|lappy (~jskcr@jskcr.user) |
| 16:02.42 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
| 16:03.08 | *** join/#asterisk NewSole (~dave@i216-58-91-171.avalonworks.net) |
| 16:04.20 | jsharp | Is it a bad thing that everytime I reboot the server I have, it counts up different amounts of memory? |
| 16:04.30 | jskcr|lappy | very very very very very |
| 16:04.32 | jskcr|lappy | bad |
| 16:04.46 | eKo1 | replace your memory NOW |
| 16:04.58 | jsharp | I'll probably find packing peanuts in the memory slots. |
| 16:05.23 | jskcr|lappy | youll end up corrupting your data to shit if it isnt already |
| 16:06.10 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 16:06.53 | jsharp | There's nothing on the machine yet. This is an initial install. |
| 16:07.20 | jskcr|lappy | what distro? |
| 16:07.37 | jsharp | NetBSD/i386 |
| 16:08.09 | jskcr|lappy | get a bootcd with a memory checker and run it on the machine first before you waste your time |
| 16:08.23 | denon | memtest86 |
| 16:08.57 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 16:09.19 | jskcr|lappy | It also probably means you bios does not have stop on errors setup for the memory if it supports it |
| 16:11.19 | jskcr|lappy | Im about to release a new asterisk live cd, anyone have any suggestions for third party asterisk apps to put on it? |
| 16:12.31 | Lee__ | jskcr|lappy: AMP? |
| 16:12.51 | jskcr|lappy | already put it on there anything else |
| 16:13.06 | Lee__ | jskcr|lappy: what's the URL for your livecd? |
| 16:13.19 | jskcr|lappy | It will be available next week |
| 16:13.32 | Lee__ | so you don't have a web page yet? |
| 16:13.41 | jskcr|lappy | I have a few lol |
| 16:14.22 | Lee__ | which one are you going to distribute the live cd on? |
| 16:14.22 | jskcr|lappy | bittorrent |
| 16:14.32 | Lee__ | argh. can you just give me the URL so I can bookmark it and remember to check back next week? |
| 16:14.32 | jskcr|lappy | Ill get a few http mirrors from the other gpl projects I work on too. |
| 16:14.53 | eKo1 | how about putting all the third party * apps you can find. |
| 16:14.57 | jskcr|lappy | lol |
| 16:15.08 | *** join/#asterisk slazy-jave (~jae@pk-isb-trg-sc01-019.speedcast.com) |
| 16:15.28 | slazy-jave | hi |
| 16:15.31 | slazy-jave | <PROTECTED> |
| 16:15.44 | slazy-jave | i actually want to configure modem with asterisk |
| 16:15.49 | slazy-jave | has anybody done that before |
| 16:18.03 | BoRiS | goodluck |
| 16:19.52 | *** join/#asterisk Taadow (yizo@S010600d0097b7af0.vs.shawcable.net) |
| 16:19.52 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 16:19.55 | Taadow | Good day all. |
| 16:19.58 | Juxt | good afternoon |
| 16:20.16 | Juxt | i am trying to make 1 asterisk box peer with another |
| 16:20.19 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-214-171.dsl.scarlet.be) |
| 16:20.20 | Juxt | can someone help me here |
| 16:20.46 | Juxt | my server1 is the one my server2 should authenticate with |
| 16:20.51 | Juxt | i have the following in the iax.conf on server1 |
| 16:20.51 | Taadow | A question... with h.323 debug enabled I notice that RTP's ExternalIpAddress is set to 127.0.0.1. Pretty sure this explains my lack of audio. Anyone have any idea why it would be set to localhost? |
| 16:21.05 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:21.16 | Juxt | [secura2] |
| 16:21.17 | Juxt | type=user |
| 16:21.17 | Juxt | host=dynamic |
| 16:21.17 | Juxt | username=secura2 |
| 16:21.17 | Juxt | secret=password |
| 16:21.17 | Juxt | accountcode=secura |
| 16:21.19 | Juxt | context=secura |
| 16:21.21 | Juxt | qualify=yes |
| 16:21.23 | Juxt | auth=md5,plaintext,rsa |
| 16:21.25 | Juxt | permit=64.135.20.0/255.255.255.0 |
| 16:21.34 | jakepdev | ~pastebin |
| 16:21.35 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 16:21.56 | Juxt | server2 has this in iax.conf register => secura2:password@voip.richmedium.com |
| 16:22.34 | Juxt | they refuse to authenticate |
| 16:26.31 | *** join/#asterisk key2 (~key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 16:26.33 | key2 | hey |
| 16:26.54 | key2 | can asterisk work with a normal voice modem ? |
| 16:26.59 | jief- | i was wondering, is it possible to listen to some calls with *, for example, for a call center where managers would listen to analysts? |
| 16:27.49 | eKo1 | You mean listen in on a phone conversation? |
| 16:27.49 | jskcr|lappy | jief-, ethereal works :) |
| 16:28.16 | jskcr|lappy | I wrote something to dump all the calls to timestamped files im gonna release soon |
| 16:28.19 | slazy-jave | well i also was enquiring the same does * work with voice modem |
| 16:29.18 | jskcr|lappy | the main problem was with the g.729 codec but thats about solved. |
| 16:29.59 | key2 | so, no one knows if asterisk works with a voice modem ? |
| 16:30.33 | jief- | eKo1: yes |
| 16:30.40 | jskcr|lappy | key2, I think thats been pretty much negelected because the hardware is alot cheaper than most voice modems |
| 16:30.48 | eKo1 | jief-: yes |
| 16:30.52 | jskcr|lappy | some will , some wont, some might someday. |
| 16:31.00 | jief- | eKo1: we are going to have a support center at the office soon, and my boss want me to be able to listen to the techs we're going to hire |
| 16:31.13 | wiz8291 | guys |
| 16:31.17 | eKo1 | there's a Monitor app that does just that. |
| 16:31.23 | wiz8291 | CLI, i can't make any calls!!! |
| 16:31.31 | jsharp | jief-: Yes, you can listen to them...most easily if they're on analogue zap hardware. |
| 16:31.33 | wiz8291 | i get NPI: Unknown Number Plan (0) |
| 16:31.38 | wiz8291 | Presentation: Number not available (67) |
| 16:31.41 | wiz8291 | any ideas? |
| 16:31.44 | jief- | we have a SIP system |
| 16:31.45 | key2 | jskcr|lappy: at the same time u don't find the hardware everywhere |
| 16:31.45 | key2 | ... |
| 16:31.48 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:32.20 | jsharp | There's the ChanSpy application that should work for what you need, though. |
| 16:32.42 | jief- | jsharp: ok, im gonna look it up |
| 16:33.11 | key2 | does someone know how to set up the flash hook time ? |
| 16:33.23 | Taadow | With h.323 debug enabled I notice that RTP's ExternalIpAddress is set to 127.0.0.1 when setting up a call. Anyone seen this before? |
| 16:33.41 | jief- | voip-info.org is kinda slow today |
| 16:33.57 | Juggie | its down for me |
| 16:33.58 | Seyr | yeh, im getting "Page not found" half the time |
| 16:34.00 | Juggie | or was |
| 16:34.14 | jief- | wikis tend to be slow |
| 16:34.33 | Seyr | voip-info is norm pretty good |
| 16:34.41 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-214-171.dsl.scarlet.be) |
| 16:34.42 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 16:34.46 | zoa | voip info crashes like every other day |
| 16:35.22 | FuriousGeorge | good afternoon all |
| 16:35.40 | jief- | well, now its time to order our FXO card |
| 16:35.40 | jskcr|lappy | george are you furious about something? |
| 16:36.26 | jskcr|lappy | .whois FuriousGeorge |
| 16:36.52 | FuriousGeorge | jskcr|lappy: its a pun on a popular childrens cartoon character |
| 16:36.55 | FuriousGeorge | get it? |
| 16:37.12 | jskcr|lappy | oic |
| 16:37.20 | FuriousGeorge | CuriousGeorge |
| 16:37.27 | FuriousGeorge | and sometimes i become |
| 16:37.34 | blitzrage | the man in the yellow hat |
| 16:38.50 | FuriousGeorge | holy cow, nicserv says someone owns that one, i should find out how to send him a memo |
| 16:38.52 | FuriousGeorge | anyway |
| 16:39.08 | jskcr|lappy | see how long he hasnt been on they expire in 60 days |
| 16:39.18 | FuriousGeorge | can someone recommend a good wired headset for computers (usb preferably), or a wireless one that somehow doesnt use bluetooth |
| 16:39.48 | mishehu | you don't like bluetooth? |
| 16:39.49 | mishehu | heh |
| 16:40.05 | FuriousGeorge | mishehu: its too quirky with windows |
| 16:40.12 | jskcr|lappy | logitech makes a nice usb one for 34 bucks |
| 16:40.28 | jskcr|lappy | planetronics ones suck |
| 16:40.33 | jief- | is there a place to get * voicemail default messages in french? or will i have to record them myself? |
| 16:40.33 | FuriousGeorge | dunno about with linux, but im too busy to try and make new devices work with my linux box |
| 16:40.44 | *** join/#asterisk mozrat (~mozrat@80.68.89.215) |
| 16:40.58 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:40.59 | jskcr|lappy | my logitech one works great with linux it comes right up as usb_sond |
| 16:41.04 | jskcr|lappy | oops sound lol |
| 16:41.25 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 16:41.29 | FuriousGeorge | jskcr|lappy: i had a plantronics and an "ActionTech", both had mediocre at best sound and mic quality, and both would randomly connect to the headset and transmit only static, unbeknownst to user |
| 16:41.56 | FuriousGeorge | jskcr|lappy: im the only person using linux though |
| 16:42.17 | jskcr|lappy | yea get a logitech they work well |
| 16:42.28 | FuriousGeorge | jskcr|lappy: wired? |
| 16:42.33 | jskcr|lappy | no its usb |
| 16:42.41 | FuriousGeorge | usb bluetooth? |
| 16:42.51 | jskcr|lappy | oh you want wireless. |
| 16:42.55 | FuriousGeorge | no |
| 16:43.01 | FuriousGeorge | sorry |
| 16:43.32 | FuriousGeorge | cuz i asked if it was wired and you said: <FuriousGeorge> jskcr|lappy: wired? |
| 16:43.32 | FuriousGeorge | <jskcr|lappy> no its usb |
| 16:43.43 | jskcr|lappy | oh lol Im getting tired |
| 16:44.13 | FuriousGeorge | np |
| 16:44.35 | jskcr|lappy | http://www.logitech.com/index.cfm/products/details/US/EN,CRID=103,CONTENTID=6338 |
| 16:44.38 | jskcr|lappy | thats the one I got |
| 16:44.52 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 16:46.03 | FuriousGeorge | i have the one that looks like that and is blue, but people dont like "the look" |
| 16:46.18 | FuriousGeorge | they call'em "air traffic control" headsets |
| 16:46.26 | FuriousGeorge | they sound great though |
| 16:46.48 | FuriousGeorge | its the "premium" model of that one, i just noticed |
| 16:47.05 | FuriousGeorge | i was hoping for something that goes in one ear |
| 16:47.07 | pigpen | Is there any great way to forward an extention to a different extention? IE: I have a office phone...then work out of the office for a week...forwarding my calls to a soft phone... |
| 16:48.11 | FuriousGeorge | pigpen: someone once said to me "once you have answered a call you can DO anything with it". so the priority after answer somewhere you just dial the propper extension |
| 16:48.36 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 16:48.45 | FuriousGeorge | playback a "please wait" somewhere in there |
| 16:48.53 | kioko | hi guys |
| 16:49.27 | kioko | any good billing solution for* ? |
| 16:51.08 | jskcr|lappy | Kioko you can do it with a dialplan and a mysql database pretty easily |
| 16:51.11 | durex[laptop] | kioko www.voip-info.org |
| 16:51.19 | *** join/#asterisk KristinG (~KristinG@muppet.geekgirls.us) |
| 16:51.29 | KristinG | hi |
| 16:51.43 | KristinG | can anyone help with a routing question |
| 16:51.46 | Juxt | i keep getting errors like this No registration for peer 'secura2' |
| 16:52.01 | eKo1 | Let me tell you, billing solutions for voip/pstn providers are very complex. |
| 16:52.15 | kioko | software that does invoicing based on the CDRs |
| 16:52.23 | eKo1 | So that 'pretty easily' business is bull. |
| 16:53.10 | FuriousGeorge | jskcr|lappy: does anyone make "single ear" usb headsets, i cant find any from logitech |
| 16:53.34 | sivana | KristinG: what's your question |
| 16:53.35 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:53.41 | jskcr|lappy | FuriousGeorge: I saw a gaming one that was a ear bud |
| 16:54.10 | KristinG | I have 2 few different carriers that I route calls to depending on cost/call plan |
| 16:54.26 | *** part/#asterisk pigpen (~mark@fw.seamans.cc) |
| 16:55.03 | KristinG | when I route with a one of them, I get 501 not implemented errors occasionally |
| 16:55.22 | zoa | guys |
| 16:55.24 | KristinG | I would like to try a different carrier at that point instead of getting a fast busy |
| 16:55.25 | zoa | go have a look at http://www.asteriskguru.com/tool3.html |
| 16:55.29 | zoa | tell me what you think |
| 16:55.54 | sivana | KristinG: what's the dialstatus at the point when you get 501? |
| 16:56.25 | jskcr|lappy | zoa: it looks like a mac app |
| 16:56.34 | zoa | its a windows app |
| 16:56.35 | zoa | :) |
| 16:56.41 | KristinG | sivana not sure |
| 16:56.53 | KristinG | i am in debug now |
| 16:56.59 | KristinG | and i can reproduce it |
| 16:57.04 | sivana | KristinG: if you NoOp(${DIALSTATUS}) |
| 16:57.17 | sivana | then you can decide what to do with it |
| 16:57.20 | jskcr|lappy | zoa: I dont see any cid information |
| 16:57.45 | KristinG | sivana for example? |
| 16:58.09 | zoa | aha |
| 16:58.12 | sivana | well, if it's CHANUNAVAIL then call your fallback carrier |
| 16:58.17 | zoa | its there i think |
| 16:58.19 | zoa | lets doublecheck |
| 16:58.40 | sivana | you'll have to do up a little macro |
| 16:59.06 | zoa | hmm link changed to http://www.asteriskguru.com/idefisk_beta.html in the mean time |
| 16:59.24 | KristinG | ok the dialstatus is "congestion" |
| 17:00.12 | sivana | after your dial, do a goto r-${DIALSTATUS}, then have r-congestion,1,Dial(fallback carrier) |
| 17:01.04 | KristinG | so create an extension called r-congestion |
| 17:01.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:01.24 | sivana | ya, it's probably best to make a macro that handles it |
| 17:01.57 | KristinG | only problem i see with that is that it only falls over to one carrier |
| 17:02.32 | ManxPower | You can pretty easily recursivly call your dial result macro |
| 17:02.32 | Taadow | Why would the RTP ExternalIpAddress be set to 127.0.0.1? |
| 17:03.19 | *** join/#asterisk Rick_Hunter (~rhunter@07-152.008.popsite.net) |
| 17:03.29 | zoa | jskcr|lappy: point taken, there will be cid in the next version |
| 17:03.37 | jskcr|lappy | :) |
| 17:03.51 | zoa | its actually there but not with a hint or so |
| 17:04.14 | jskcr|lappy | cid should be in bigger letters so you know whos callin |
| 17:04.24 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 17:04.33 | jskcr|lappy | That way if you boss calls you can just forward them to voicemail ;P |
| 17:04.49 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 17:04.57 | pjz | so I'm ordering a PRI to use with a digium TE110P |
| 17:05.07 | pjz | and the telco is asking me a bunch of questions |
| 17:05.48 | pjz | which do I want? or need? |
| 17:06.07 | jskcr|lappy | pjz, look at the bright side I connected the first pri in the state of florida. I had to goto the co to help them set it up, over ten years ago |
| 17:06.20 | pjz | jskcr|lappy: heh |
| 17:06.38 | pjz | jskcr|lappy: so is there docco I can find that will tell me what I want/need set up? |
| 17:06.50 | onlyI | was the first to have it done ( pri ) on ras in in montreal with Bell |
| 17:07.09 | pjz | is 101XXXX allowed? number of digits outpulsed? |
| 17:07.10 | jskcr|lappy | is it a e1 or a t1? |
| 17:07.14 | pjz | T1 |
| 17:07.29 | onlyI | 23b+1D |
| 17:07.51 | *** join/#asterisk burbankmarc (~djones@68.78.185.254) |
| 17:08.15 | pjz | what's the 'number of digits outpulsed' mean? |
| 17:08.16 | mishehu | Stupid Bastard Cocksuckers country |
| 17:08.41 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 17:08.49 | Juxt | ok i am pulling my hair here |
| 17:09.03 | Juxt | i have 2 locations, each has an asterisk server |
| 17:09.05 | Nugget | http://lnk.nu/slacker.com/lt <-- Bastard |
| 17:09.23 | Juxt | i want people from location1 to be able to dial extensions at location2 |
| 17:09.23 | burbankmarc | when i call my voicemailmain extension i get "comedian mail, mailbox" then nothing... |
| 17:09.53 | pjz | Nugget: hehe cute |
| 17:10.00 | pjz | Nugget: what city? |
| 17:10.14 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 17:10.23 | *** part/#asterisk bannerman (~bannerman@209.216.176.43) |
| 17:10.50 | Juxt | nuggest: what does BOFH mean? |
| 17:10.51 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 17:11.02 | pjz | Juxt: Bastard Operator From Hell |
| 17:11.04 | burbankmarc | exten => 1000,1,VoicemailMain(${CALLERIDNUM}) |
| 17:11.05 | Nugget | austin. |
| 17:11.16 | Nugget | 183 and anderson mill |
| 17:11.45 | Nugget | (which is practically south dallas to hear some people talk :) |
| 17:11.49 | pjz | heh |
| 17:11.54 | pjz | me too, but I'm down in Tarrytown |
| 17:11.58 | Nugget | cool |
| 17:12.10 | jskcr|lappy | http://lists.digium.com/pipermail/asterisk-users/2005-April/101277.html |
| 17:12.14 | Nugget | do you like spicy food? you should come to our nuclear taco night tonight |
| 17:12.16 | jskcr|lappy | thats for you pjz |
| 17:12.20 | Juxt | so can someone hold my hand and walk me thru 2 asterisks peering? |
| 17:12.22 | pjz | yeah, my company is about to move to the new Whole Foods building |
| 17:12.30 | jskcr|lappy | [Asterisk-Users] New PRI install with new te110p |
| 17:13.08 | pjz | jskcr|lappy: oh, it's wackier than that - I'm ordering the PRI from SWB and they want a bunch of questions answered |
| 17:13.10 | Taadow | Does anyone know how H323 determines the ExternalIpAddress of an RTP stream? |
| 17:13.26 | pjz | jskcr|lappy: if it were pre-existing I'd just plug stuff in and see if it worked |
| 17:13.31 | jskcr|lappy | pjz what questions? |
| 17:13.40 | pigpen | Nugget, if you want real hot food....come to San Antonio |
| 17:13.45 | jskcr|lappy | like the provisioning of it ? |
| 17:13.47 | pjz | jskcr|lappy: 5.101XXXX Allowed: Yes: No: |
| 17:13.48 | pjz | 6.Number of Digits Outpulsed:       |
| 17:13.50 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
| 17:13.54 | Nugget | if you want real hot food...come to nuclear taco night |
| 17:14.04 | pjz | jskcr|lappy: I guess so |
| 17:14.17 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 17:14.35 | JonR800 | what's in a nuclear taco? |
| 17:14.47 | *** join/#asterisk gtigene (~chatzilla@c-67-184-112-58.hsd1.il.comcast.net) |
| 17:14.55 | Nugget | red savina habanero, savina, and regular habanero. |
| 17:15.01 | Nugget | s/savina/serrano/ |
| 17:15.02 | pjz | jskcr|lappy: Protocol, Glare Control, Glare Resolution, CPN option, some Caller ID specs, is porting required? |
| 17:15.20 | pjz | jskcr|lappy: I've never heard of half this stuff |
| 17:15.28 | JonR800 | lol... that'd probably do me in, sounds like a good way to die though |
| 17:15.43 | Nugget | it's wonderful stuff. :) |
| 17:15.53 | pjz | jskcr|lappy: do I want Customor National protocol? and if national, FAS or NFAS? |
| 17:16.01 | gtigene | What mail program should I use to have email notification of voicemails in voicemail.conf? |
| 17:16.05 | Nugget | we get together once a month and eat them, then cool down with free beer. :) |
| 17:16.10 | jskcr|lappy | do it like this google for Number of Digits Outpulsed: site:lists.digium.com |
| 17:16.19 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 17:16.38 | jskcr|lappy | youll find all the questions are probably already answerd on the digium mailing list, |
| 17:17.18 | JonR800 | I'm going to have to get some habaneros now and make tacos. Though it's a bit harder to get habaneros in michigan. |
| 17:17.31 | Nugget | heh |
| 17:17.34 | pjz | jskcr|lappy: ah! good idea, thanks |
| 17:18.25 | pjz | Nugget: hrm, I'm pretty much burnt out on super-spicy food (NPI) |
| 17:19.43 | *** part/#asterisk gtigene (~chatzilla@c-67-184-112-58.hsd1.il.comcast.net) |
| 17:20.00 | pjz | jskcr|lappy: hrm, okay, no hit for 101XXXX allowed. and I'm not even sure what that means. |
| 17:20.15 | FuriousGeorge | jskcr|lappy: think these usb headsets for ps2 work with pcs? |
| 17:20.28 | pjz | jskcr|lappy: ah, nm, I grok |
| 17:20.44 | Nugget | bummer |
| 17:20.54 | Nugget | well, feel free to come for the beer. :) |
| 17:21.14 | jskcr|lappy | pjz: ive been workin on project since yesterday at 2pm to now, Im right there with ya :P |
| 17:21.48 | jskcr|lappy | I usually go 40 hours on then sleep for 5 hours. |
| 17:21.56 | pjz | Nugget: where & when? |
| 17:22.29 | Nugget | tonight, 7:30p, directions at http://www.livejournal.com/community/nucleartacos/17690.html |
| 17:22.42 | jskcr|lappy | free tacos? |
| 17:23.11 | Nugget | yah |
| 17:23.36 | jskcr|lappy | Man I miss living in dallas because of stuff like that :( |
| 17:25.32 | Nugget | cool :) |
| 17:25.47 | facek_ | anyone use PAP2 over nat? |
| 17:29.23 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 17:29.57 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 17:30.11 | Vco | re: ps2 headsets |
| 17:30.13 | Vco | ya |
| 17:30.14 | Vco | they do |
| 17:30.24 | Vco | the mile long cord is a major plus |
| 17:34.07 | pjz | if I'm getting a PRI do I want DOD? |
| 17:34.09 | pjz | or DID? |
| 17:34.10 | pjz | or both? |
| 17:34.16 | pjz | I'm guessing i want both |
| 17:34.28 | Corydon-w | wtf is DOD? |
| 17:34.34 | pjz | Outward |
| 17:34.47 | pjz | Department Of Defense |
| 17:35.01 | Corydon-w | Is that just trying to say that the PRI is bidirectional? |
| 17:35.45 | kb1_kanobe | dod is an analog signalling thing, pri is implicitly dod. |
| 17:36.07 | pjz | ah, okay |
| 17:36.08 | kb1_kanobe | ie. the remote pri switch cannot provide secondary dialtone. |
| 17:36.09 | pjz | good to know |
| 17:36.47 | pjz | do I want a back-up d channel? |
| 17:37.06 | Corydon-w | Probably not, unless you're getting multiple PRIs installed at the same time |
| 17:37.12 | pjz | okay |
| 17:37.51 | pjz | and unless I have multipe PRIs, I don[t need Trunk Group Pverflow, do I? |
| 17:38.05 | Corydon-w | Nope |
| 17:38.06 | Dovid | anyone got a moment to help me with a bandwith test |
| 17:38.06 | kb1_kanobe | backup d channel handles two conditions - that having one 64k D channel is not enough for the volume of control messages or, that the backup d channel comes on a different wire that is routed differently just incase a backhoe goes through the first cable. |
| 17:38.06 | Dovid | ? |
| 17:38.31 | pjz | kb1_kanobe: ah. neither of those should be an issue |
| 17:38.38 | Corydon-w | kb1_kanobe: yeah, but a single D channel should be enough, even for a DS3 |
| 17:38.50 | Corydon-w | in terms of the volume of call messages |
| 17:39.06 | *** join/#asterisk ooPo (~peori@blk-222-101-244.eastlink.ca) |
| 17:39.32 | Corydon-w | The maximum you'd have to wait for a call setup with 671 voice channels is half a second, and that's only if 671 calls all come in at the same time |
| 17:40.17 | burbankmarc | when i call my voicemailmain extension i get "comedian mail, mailbox" then nothing... |
| 17:40.31 | cpatry | Corydon-w: how many PRIs do you have? |
| 17:40.31 | Nugget | so enter in your mailbox. |
| 17:40.34 | Corydon-w | burbankmarc: it's prompting you to enter a mailbox |
| 17:40.45 | Corydon-w | cpatry: 1 |
| 17:41.05 | Vco | or are you expecting "voicemail" and not "voicemailmain" |
| 17:41.42 | pjz | what's Glare Resolution? |
| 17:41.50 | pjz | and Glare control? |
| 17:42.46 | *** join/#asterisk Rick_Hunter (~rhunter@02-041.008.popsite.net) |
| 17:43.20 | wiz8291 | anyone got caller presentation working in the UK? |
| 17:43.23 | pjz | do I want Calling party number or billing number? |
| 17:43.26 | wiz8291 | i am having problems |
| 17:43.30 | pjz | and do I need porting? |
| 17:43.34 | kb1_kanobe | Glare is a condition where both ends of a circuit attempt to grab it for use at the same time. They sound like vendor specific names for methods to work around this problem. |
| 17:43.48 | pjz | kb1_kanobe: hrm. okay, could be. |
| 17:43.49 | kb1_kanobe | pzj: you might want a consultant. :-) |
| 17:44.04 | Corydon-w | pjz: porting is if you need existing numbers to be 'ported' to the new PRI |
| 17:44.16 | wiz8291 | BT won't let me dial out since caller presentation was turned on on our PRI |
| 17:44.53 | pjz | kb1_kanobe: oh, i'm just trying to place an order for a PRI and SBC sent me this form to fill out |
| 17:45.25 | HeadachesAbound | glare is just what kb1 said. we are preparing to put in a ds3 and switch everything over to an asterisk box (many sleepless nights await me) |
| 17:45.37 | zoa | hehe indeed |
| 17:48.06 | ooPo | The Wildcard X100P seems to be discontinued... is there an equivalent available? |
| 17:48.14 | pjz | the TE110P |
| 17:48.37 | ooPo | Isn't that a T1 card? |
| 17:48.48 | wiz8291 | no one? |
| 17:48.49 | pjz | isn't the X100P a T1 card? |
| 17:49.00 | ooPo | I have no idea. :) |
| 17:49.10 | pjz | go to digium.com and look and find out |
| 17:49.29 | HeadachesAbound | Did i mention that my dreams are filled with alphabet soup zoa? PHP, Perl, MySQL, *, AGI, CGI, Apache, Fedora, Linux, PRI, DSP... |
| 17:49.41 | ooPo | ok |
| 17:50.03 | tzafrir_laptop | nobody wants to pick up the maintinance of its drivers? |
| 17:50.13 | HeadachesAbound | what drivers? |
| 17:50.21 | tzafrir_laptop | of X100P |
| 17:50.56 | HeadachesAbound | way above my water line there tzafrir or i might consider it. |
| 17:51.25 | pjz | oh, I'm wrong |
| 17:51.45 | pjz | the X100p is the single-POTS FXO line card |
| 17:51.51 | Vco | jah |
| 17:51.55 | tzafrir_laptop | (X100P is the original FXO card) |
| 17:51.56 | ooPo | The devkit package on digium, I don't suppose that's available to regular people? |
| 17:52.12 | Vco | now it's just getting the tdm card with however many FXO you want |
| 17:52.13 | ooPo | Yeah, I just want a single fxo line to play with. |
| 17:52.26 | Vco | the sinlge card is a bit of a waste tho.. |
| 17:52.42 | tzafrir_laptop | ooPo, then get a X100P on ebay or so. |
| 17:52.45 | pjz | ooPo: just get a TDM11B |
| 17:52.52 | Vco | like...a few more bucks.....and you have something that can expand once you've gotten "bit" from playing with the system a bit |
| 17:53.01 | HeadachesAbound | we picked up a dev kit...we aren't doing much dev at the time, but i believe it is 1 per org / person so i don't see why you couldn't get it. |
| 17:53.07 | tzafrir_laptop | In the worst case you'll lose some 20$ (s&h included) and some time. |
| 17:53.19 | pjz | ooPo: or a TDM01B |
| 17:53.20 | ooPo | Hmm. I may just pick up the devkit then. |
| 17:53.24 | Vco | if you want a clone ya |
| 17:53.54 | tzafrir_laptop | The devkit won't give you much beyond what you already have. |
| 17:53.59 | Vco | thats like buying 1024mb sticks of ram for $20...and expecting no problems.. |
| 17:54.04 | pjz | ooPo: the TDM01B is the equivalent of an X100P, but allows you to buy 3 more modules that can be either FXS or FXO and add it into it |
| 17:54.31 | ooPo | ahh, so I see |
| 17:54.40 | tzafrir_laptop | Vco, what are the possible problems of such "clones", besides not being officially blessed by digium? |
| 17:55.00 | burbankmarc | burbankmarc h |
| 17:55.04 | ooPo | $129.95 on telephonyware.com |
| 17:55.26 | tzafrir_laptop | pjz, but for that price you can get around 9 or 10 X100P cards. Or more memory for you computer, or a better CPU |
| 17:56.05 | pjz | tzafrir_laptop: ? it's only like $133 for a TDM01B |
| 17:56.23 | tzafrir_laptop | pjz, it's only like 10$ for an FXO card. |
| 17:56.26 | HeadachesAbound | i think the dev kit is a tdm11b. it allows you to add 2 additional modules later if you decide to but it comes with 1 fxo and 1 fxs. |
| 17:56.30 | onlyI | 195 for TDM11B |
| 17:56.35 | ooPo | fxo lets me hook up a normal phone line, then I can 'log in' to the server somehow from remote and use it? |
| 17:56.36 | Vco | try logging the channel for a while and do a search for "x100p" and "problem" |
| 17:56.41 | Vco | and "clone" |
| 17:56.43 | jskcr|lappy | lmao |
| 17:56.49 | kb1_kanobe | the clones are based on the original reference design that started it all. The digium cards have been revised to address problems such as bad impedance matching and so on, but the clones have not. People using the clones tend to encounter problems, point the finger at *, only to find there is no avenue of support. |
| 17:56.50 | tzafrir_laptop | pjz, how much does extra 256MB of memory cost nowadays? |
| 17:56.55 | jskcr|lappy | Vco its about 2-3 times per day |
| 17:57.04 | Vco | yup |
| 17:57.30 | jskcr|lappy | cid problems ring detection problems to list a few |
| 17:57.34 | tzafrir_laptop | Vco, because they have a bad reputation. People have problems with digium cards as well. |
| 17:57.39 | pjz | tzafrir_laptop: no one buys 256MB of RAM |
| 17:57.57 | tzafrir_laptop | Vco, buy two, in case one is bad. It'll still be much cheaper. |
| 17:58.41 | Vco | **shrug**, i generaly pay a premium to get something that is less likely to fail |
| 17:58.42 | pjz | tzafrir_laptop: I'd rather support the company that's doing real dev work and support on their cards |
| 17:58.42 | burbankmarc | how would you make it so when you call the voicemailmain extension you don't have to type the mailbox? something like this: exten => 1000,1,VoicemailMain(s${EXTEN}) ? |
| 17:58.49 | tzafrir_laptop | Vco, I'm not suggesting to run your mission critical server on such cards. But if you want a card to play with, why not? |
| 17:59.10 | wiz8291 | :( |
| 17:59.27 | jskcr|lappy | I look at it more like paying for less agrivation |
| 17:59.30 | Vco | and why not spend time working on features etc instead of trying to trace why <insert feature here> isn't working on your system... |
| 17:59.36 | HeadachesAbound | exten => 1000,1,VoiceMailMain(s${CALLERIDNUM}) - like that burbankmarc |
| 17:59.37 | tzafrir_laptop | pjz, I'd rather actually buy a card to have something to play with, so I can later convince my boss to buy a decent card. |
| 18:00.03 | burbankmarc | thanks |
| 18:00.16 | Vco | so you could spend $130 or waste $30...thats your call |
| 18:00.42 | HeadachesAbound | we started with the dev kit for our testing purposes. this was enough to convince the boss to pick up a Quad port T1 card (still for testing) and we are now preparing to put in a DS3 and another machine with 2 Quad T1 cards for production. |
| 18:00.43 | onlyI | tzafrir_laptop about 90 CDN ddr 266 |
| 18:01.34 | Blackvel | HeadachesAbound: how many telco lines do you have? |
| 18:01.48 | burbankmarc | after making the change it still says mailbox...it doesn't prompt for a passwd |
| 18:01.50 | Vco | $90 for what? |
| 18:01.52 | Vco | 512? |
| 18:02.01 | *** join/#asterisk _kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 18:02.01 | onlyI | 256 |
| 18:02.13 | tzafrir_laptop | Vco, it's not wasting. You could have actually wasted 130$. but with 30$ you'll likely get a working card. |
| 18:02.42 | tzafrir_laptop | and then you'll have some spare change for extra memory. Thanks onlyI :-) |
| 18:02.48 | HeadachesAbound | right now, we have 3 pri connections. we are about to replace 2 of those with a ds3. we are starting out with 8 active pri on the ds3 and plan to turn up the rest within the next year or so depending on business growth. |
| 18:03.31 | HeadachesAbound | burbankmarc: do you have the voicemail.conf setup for that extension? |
| 18:03.51 | burbankmarc | sure do |
| 18:04.31 | HeadachesAbound | the 3rd pri that we have now will be turned of beginning of next year (january i think) and at that point, we will actually be paying about the same price for 8 pri that we presently pay for 3. |
| 18:05.41 | HeadachesAbound | i'm using CVS from a couple days ago, but when I use the syntax I posted earlier for my extension, it goes straight thru. |
| 18:05.45 | onlyI | HeadachesAbound WOw few trunks |
| 18:05.49 | _kb1_kanobe | is anyone here successfully running multiple codecs between the same pair of peers via iax? I mean along the lines of [hosta-gsm] and [hosta-ulaw] so you can explicity specify the codec to use for teh call from the dial() command? |
| 18:05.56 | *** join/#asterisk citynet (~trillian@208.50.193.173) |
| 18:06.06 | zoa | _kb1_kanobe: ues |
| 18:06.12 | zoa | that seems to work just fine |
| 18:06.16 | zoa | i tried that before |
| 18:06.45 | _kb1_kanobe | could I trouble you to pastebin the config? when I try defining the second peer it screws with the codecs for the first... |
| 18:08.01 | ooPo | thanks guys, the TDM01B is pretty much exactly what I was looking for... |
| 18:08.15 | ooPo | *plonk* |
| 18:08.16 | *** part/#asterisk ooPo (~peori@blk-222-101-244.eastlink.ca) |
| 18:08.25 | zoa | yeah you are actually authenticating as the other user or so |
| 18:08.26 | zoa | i guess |
| 18:09.23 | _kb1_kanobe | the only thing in common is the ip address and secret. The peername is distinct... however I'm using the 'friend' declaration - perhaps that's part of the issue. |
| 18:11.01 | burbankmarc | i figured out my problem, i had the callerid feild in sip.conf set to a name, instead of a number, which is what threw it all off...hate being a newb |
| 18:12.13 | HeadachesAbound | can we get a firm definition on the term newb? I mean, I've only been using / playing with * for about 5 months, but does that make me a newb level 2 or level 3? |
| 18:12.17 | jskcr|lappy | anyone here using a sipura-3000? |
| 18:12.44 | shepherd | haha.. that means your an expert :) |
| 18:12.47 | shepherd | congrats |
| 18:12.56 | shepherd | you're also |
| 18:13.26 | HeadachesAbound | so can I start charging outrageous fees yet or is that something only expert level 4 can do? |
| 18:13.55 | shepherd | *pffft* everything should be free |
| 18:14.06 | shepherd | and you should put in 90 hours / week |
| 18:14.12 | _kb1_kanobe | have you had your first 3am nosebleed yet? you can justify level 4 fees to cover the pending medical expenses. |
| 18:14.53 | shepherd | anyways.. new nin is great :) |
| 18:14.57 | shepherd | just so you guys know |
| 18:15.01 | shepherd | in case you like nini |
| 18:15.03 | shepherd | nin |
| 18:15.13 | HeadachesAbound | i already put in at least 90 hours a week and i'm pretty sure most of that is free. i don't think i've had the 3am nosebleed but i have had the 5 days without sleep hangover, does that count? |
| 18:15.21 | Vco | bah..whatever....slap the words "consulting" and "pbx" on the invoice and slide the decimal place waaaaaaaaaaaaaay over |
| 18:16.30 | DEEZED | In your guys opinion, are IAX providers could enough for commercial use? |
| 18:16.47 | shepherd | yeah.. |
| 18:16.49 | DEEZED | as far as reliability |
| 18:16.52 | DEEZED | ok |
| 18:16.53 | shepherd | definately voicepulse |
| 18:17.07 | shepherd | and nufone.net |
| 18:17.28 | shepherd | but.. it really depends on your ping times to either |
| 18:17.47 | DEEZED | hmm looks like voice pulse doesn't have pay per minute |
| 18:18.00 | shepherd | yes they do |
| 18:18.07 | shepherd | but it's prepaid |
| 18:18.15 | *** join/#asterisk SirPrize (~blah@host-212-158-241-138.bulldogdsl.com) |
| 18:18.26 | torisa | http://connect.voicepulse.com/ |
| 18:18.36 | zoa | does somebody know of a sip phone for nokia communicators ? |
| 18:18.38 | shepherd | yes... thanks |
| 18:18.40 | DEEZED | oh thanks for the link |
| 18:18.41 | shepherd | that link :) |
| 18:19.01 | SirPrize | My Sipura 3000 has a very noticeable delay between my dialing it, and before it starts ringing |
| 18:19.05 | DEEZED | wow $11/month per phone number |
| 18:19.06 | SirPrize | any ideas why this might be? |
| 18:19.19 | shepherd | that's only incoming though |
| 18:19.30 | shepherd | you can get only outgoing if you would like |
| 18:19.53 | shepherd | at $0.024 / min |
| 18:20.00 | DEEZED | yeah im looking into running a small virtual ivr service |
| 18:20.11 | Vco | wow.. |
| 18:20.13 | Vco | at that rate.. |
| 18:20.38 | Vco | i'd have to talk 25 hours to cost the same as what a basic line costs here.. |
| 18:20.41 | _kb1_kanobe | cpatry: regarding the -dev question, check the dial() command. I doubt you meant to try to call '155@' - it'd more correctly be something like '155@incoming-context' |
| 18:21.59 | Vco | anyone know of a good source for outgoing in japan? |
| 18:22.02 | wiz8291 | anyone seen distortion on PRI channels before? |
| 18:22.10 | wiz8291 | not echo, but distortion |
| 18:22.29 | Vco | i can get a pots line at teh inlaws...but it's like $50/mo for basic line.. |
| 18:22.50 | _kb1_kanobe | wiz8291: describe 'distortion' - hiss |
| 18:23.01 | _kb1_kanobe | , pops, crackle, wheezes? |
| 18:23.10 | wiz8291 | crackle and hiss i guess |
| 18:23.18 | wiz8291 | when the person outside is talking |
| 18:23.22 | wiz8291 | only they hear it |
| 18:23.28 | _kb1_kanobe | is the zaptel echo canceller on? |
| 18:23.32 | wiz8291 | yup |
| 18:23.51 | _kb1_kanobe | is it consistent for all pstn numbers? |
| 18:23.54 | *** join/#asterisk Inv_arp (junya@adsl-3-244-116.mia.bellsouth.net) |
| 18:23.56 | wiz8291 | it is, yes |
| 18:24.00 | *** join/#asterisk bstock (~bstock@68.78.185.254) |
| 18:24.27 | _kb1_kanobe | You could try disabling the echocan and running some test calls to eliminate it - echocan implictly screws with the signal. |
| 18:24.36 | durex | folks... |
| 18:24.43 | wiz8291 | its worse without echocancel |
| 18:24.59 | durex | having problem now compiling app_addon_sql_mysql.c on FreeBSD |
| 18:25.22 | _kb1_kanobe | Try setting up a PRI DID that is answered by echo() and see if you get the same effect on that call. |
| 18:25.30 | bstock | hey, does anyone know how to get linphone to let me input numbers into asterisk? |
| 18:25.42 | wiz8291 | _kb1_kanobe: done that, and yes... i do get the effect |
| 18:25.50 | Juxt | bstock: DTFM |
| 18:26.12 | _kb1_kanobe | then the issue can be reasonable isolated to the T1 line, the zaptel card or drivers. |
| 18:26.23 | nestAr | can someone write me up an RFC for PIFoIP? |
| 18:26.26 | _kb1_kanobe | is your span timing correctly set? |
| 18:26.31 | durex | take a look: http://pastebin.ca/10923 |
| 18:26.41 | wiz8291 | installing the latest cvs drivers now |
| 18:27.15 | _kb1_kanobe | and are your interrupts behaving |
| 18:27.16 | wiz8291 | yeah |
| 18:27.16 | wiz8291 | the card has its own interrupt |
| 18:27.32 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 18:27.39 | *** join/#asterisk shepherd (~matt@207.111.174.1) |
| 18:27.54 | slePP | how do i get outgoing caller id to be proper with SIP? |
| 18:27.55 | slePP | from SIP->SIP |
| 18:27.57 | _kb1_kanobe | does the output of 'lspci -vvb' agree with the output of /proc/interrupts for the card? |
| 18:27.59 | wildgoose | anyone know of a linux app which just pops up a popup when someone rings? ie caller id on the desktop? |
| 18:28.14 | wiz8291 | _kb1_kanobe: hang 2, just checking |
| 18:29.03 | wiz8291 | yup |
| 18:29.05 | wiz8291 | they agree |
| 18:29.08 | wiz8291 | IRQ 11 |
| 18:29.16 | *** join/#asterisk ionix (~ionix@209.71.254.135) |
| 18:29.24 | ionix | Hey, how can I interface Asterisk with e911 ? |
| 18:29.35 | ionix | Do I have to populate a TCAP on SS7 or there is an other way ? |
| 18:30.07 | ManxPower | ~mailinglist site:lists.digium.com 911 OR e911 |
| 18:30.33 | ionix | ok |
| 18:30.56 | wiz8291 | :/ |
| 18:31.10 | wiz8291 | now i'm showing red alarms on everything after the cvs upgrade |
| 18:33.24 | *** join/#asterisk pbx123 (~joel@210.213.213.177) |
| 18:34.44 | ionix | it doesn'T help at all |
| 18:34.54 | ionix | only one person uses CMS MF |
| 18:34.55 | pbx123 | \nic john |
| 18:35.02 | pbx123 | \jj |
| 18:35.23 | wiz8291 | are there known issues with the zaptel drivers in the latest CVS build? |
| 18:35.59 | _kb1_kanobe | I heard someone say they had PRI problems late last night, but i've not seen anything. |
| 18:36.32 | _kb1_kanobe | your modules are loaded and the dmesg blurb from the modprobe seems correct? |
| 18:38.17 | *** join/#asterisk pbx123 (~ponaps@210.213.213.177) |
| 18:38.33 | wiz8291 | yup |
| 18:38.40 | wiz8291 | but red alarms on both spans |
| 18:38.51 | pbx123 | hi all |
| 18:38.59 | pbx123 | have a problem here |
| 18:39.22 | pbx123 | my te110p is not dtected by my motherboard |
| 18:39.41 | *** join/#asterisk vpp (~noone@83.146.58.109) |
| 18:39.45 | vpp | hi |
| 18:40.14 | _kb1_kanobe | wiz8291: that's pretty odd - try going back a few days in cvs. |
| 18:40.16 | vpp | has anyone got asterisk working with gatekeeper/gateway originated calls (H323) |
| 18:41.12 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 18:41.18 | wiz8291 | was going to go 3 days back |
| 18:42.31 | *** join/#asterisk hermie (~nick@24-236-167-53.dhcp.bycy.mi.charter.com) |
| 18:44.10 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 18:46.11 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:46.41 | shmaltz | anybody here using TDM400 FXO modules? intersted in troubleshooting a problem? |
| 18:46.59 | tzanger | shmaltz: I'll take a crack |
| 18:47.12 | johnnyb | shmaltz, I am |
| 18:47.53 | shmaltz | when doing: |
| 18:47.53 | wiz8291 | so people know |
| 18:47.55 | shmaltz | s,1,Dial(somedevice,,60) and the calling party on the PSTN hangs up after 10 seconds what happens? |
| 18:47.59 | wiz8291 | 3 days back in zaptel cvs is safe |
| 18:48.05 | wiz8291 | using the 410P card |
| 18:48.42 | shmaltz | meaning don 't do anything to the incoming call but ring something, and while it rings hanup |
| 18:48.52 | tzanger | shmaltz: ? |
| 18:49.11 | *** part/#asterisk SirPrize (~blah@host-212-158-241-138.bulldogdsl.com) |
| 18:49.14 | shmaltz | by me it takes about 4 rings untill asterisk realizes that my telco has stopped ringing the line |
| 18:49.23 | shmaltz | tzanger, yes |
| 18:49.48 | tzanger | put a real phone in parallel with the TDM400, what does it report |
| 18:50.17 | shmaltz | nothing |
| 18:50.39 | shmaltz | just stops ringing as soon as the caller hangs up (maybe a 1 second delay) |
| 18:50.47 | tzanger | shmaltz: hmm |
| 18:50.49 | johnnyb | shmaltz: have you run zttest? |
| 18:50.56 | shmaltz | nope |
| 18:50.59 | tzanger | zttest isn't gonna show you shit here |
| 18:51.02 | shmaltz | I since do answer |
| 18:51.18 | tzanger | it just looks like the DAA is taking a while to indicate stop ring (or * is taking a while to acknowlege the DAA) |
| 18:51.20 | shmaltz | that way * detects the hangup as soon as it occures |
| 18:51.25 | tzanger | you'd need to do some zaptel driver debugging to see |
| 18:51.31 | johnnyb | tzanger: when the zaptel interface loses interrupts, the TDM card has trouble knowing if a line is free or empty. I've had problems along that line. |
| 18:51.45 | shmaltz | johnnyb, I can confirm this |
| 18:51.53 | tzanger | johnnyb: true enough but I have yet to find a system that loses interrupts |
| 18:52.04 | johnnyb | tzanger: mine did. |
| 18:52.15 | tzanger | johnnyb: oh I'm not saying it can't happen |
| 18:52.18 | johnnyb | shmaltz: in your zaptel directory, run ./zttest |
| 18:52.30 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 18:52.54 | wiz8291 | upgrades haven't helped :( |
| 18:52.57 | shmaltz | I don't know exactly what the problem with 2 of my systems were, but they all stopped detecting rings alltogther untill I did a rmmod and modprobe again |
| 18:53.01 | wiz8291 | if anything the distortion is worse! |
| 18:53.11 | johnnyb | shmaltz: that's your zaptel _source_ directory |
| 18:53.25 | tzanger | shmaltz: sounds like you have older rev TDM modules |
| 18:53.28 | shmaltz | johnnyb, can I run it while asterisk is running? |
| 18:53.30 | tzanger | contact digium for a replacement |
| 18:53.33 | _kb1_kanobe | wiz8291: if it's only in one direction it sounds like hardware. |
| 18:53.33 | johnnyb | shmaltz: yes |
| 18:53.34 | tzanger | shmaltz: yes |
| 18:53.34 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 18:53.49 | gambolputty | Does the "reload" command affect calls in progress? |
| 18:53.50 | wiz8291 | it has been working fine |
| 18:53.54 | wiz8291 | it changed today |
| 18:54.04 | shmaltz | tzanger, what was the problem with the old revision? |
| 18:54.11 | tzanger | shmaltz: you just described it |
| 18:54.36 | shmaltz | oh thanks |
| 18:54.40 | shmaltz | :) |
| 18:54.58 | _kb1_kanobe | wiz8291: this is a PRI card, yes? Perhaps you or the telco have a cabling problem developing on the outward side of the link. Can they provide you CSU stats (asterisk can't sadly)? |
| 18:54.59 | shmaltz | tzanger, did you ever get echo cans? |
| 18:55.00 | johnnyb | tzanger: does the rev show up in lspci? |
| 18:55.04 | jsharp | gambolputty: Reload does not affect calls in progress. |
| 18:55.10 | tzanger | shmaltz: no |
| 18:55.16 | tzanger | johnnyb: not necessarily |
| 18:55.22 | shmaltz | johnnyb, I think so |
| 18:55.22 | tzanger | pull the card and look at it |
| 18:55.27 | tzanger | rev H or I cards are most recent |
| 18:55.42 | tzanger | and I forget what the "fixed" FXO rev is |
| 18:56.06 | tzanger | you should be able to fix it with a 220nF cap across pins 1 and 20 of any of the module sockets |
| 18:56.08 | gambolputty | Can reload handle a 500 entry sip or extensions file okay? |
| 18:56.18 | tzanger | gambolputty: why not try it? |
| 19:03.17 | shmaltz | johnnyb dmesg does report the revision= |
| 19:03.32 | *** join/#asterisk darby_t (~tom@doe237.neoplus.adsl.tpnet.pl) |
| 19:04.03 | johnnyb | shmaltz: did you try to run zttest? |
| 19:04.38 | shmaltz | nope, b/c all of the machine are currently in production, and I will not have a chance to test them until at least after business hours |
| 19:05.03 | johnnyb | shmaltz: you can run zttest on a production machine. In fact its best to. |
| 19:05.06 | shmaltz | johnnyb, thanks anyhow, I will do that as soon as I have a chance |
| 19:05.24 | shmaltz | I can't, b/c it means changing the context for the incoming lines |
| 19:06.13 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 19:06.54 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 19:09.40 | *** join/#asterisk Moc (~mochouina@64.235.210.66) |
| 19:09.51 | Moc | Hi all |
| 19:11.26 | *** join/#asterisk wiseguy_ (~chivilis@base.lt) |
| 19:11.29 | wiseguy_ | hellow |
| 19:11.42 | wiseguy_ | how to describe source based extensions in asterisk? |
| 19:13.11 | bjohnson | no idea what you mean |
| 19:14.52 | bkw_ | amplify query |
| 19:15.00 | bkw_ | because what you asked makes no sense! |
| 19:15.06 | wiseguy_ | i want |
| 19:15.23 | wiseguy_ | that ip phone |
| 19:15.29 | bkw_ | which ip phone? |
| 19:15.33 | bjohnson | that one |
| 19:16.01 | bkw_ | I get the impression that wiseguy_'s native language isn't english |
| 19:16.14 | bjohnson | sounds franglais |
| 19:16.18 | bkw_ | so cut him a little slack |
| 19:16.55 | wiseguy_ | ok, i one that ip phone with number 510 is calling 390767, and gets the call threw one gw, and another ip phone 502 calls 390767, and gets the call threw another gw. |
| 19:17.12 | wiseguy_ | yes, sorry for bad english |
| 19:17.15 | HeadachesAbound | I'm sorry, but we have no slack to spare. |
| 19:17.29 | mozrat | wiseguy_: Source based routing in effect |
| 19:17.34 | wiseguy_ | yes |
| 19:17.36 | jakepdev | i'm confused |
| 19:17.47 | wiseguy_ | source based call routing in asterisk |
| 19:17.54 | wiseguy_ | is the exact think i need |
| 19:17.59 | mozrat | route a call through a gateway depending on which extension it comes from |
| 19:18.07 | mozrat | through a channel even |
| 19:18.14 | wiseguy_ | yes |
| 19:18.24 | *** join/#asterisk ikey1 (ikey@220.226.28.86) |
| 19:18.25 | bjohnson | send them to different contexts |
| 19:18.38 | wiseguy_ | oh |
| 19:18.43 | wiseguy_ | really |
| 19:18.45 | wiseguy_ | :) |
| 19:18.50 | bjohnson | use the context= line in sip.conf or iax.conf to point them at different contexts |
| 19:19.01 | bjohnson | then those contexts could route out over different channels |
| 19:19.03 | wiseguy_ | yes |
| 19:19.21 | wiseguy_ | thanks for help |
| 19:19.22 | wiseguy_ | :) |
| 19:19.28 | bjohnson | with the dial() command |
| 19:19.46 | bjohnson | (ready for the next one) |
| 19:19.58 | mozrat | bjohnson: get ready to pull |
| 19:20.01 | mozrat | OK my turn |
| 19:20.04 | mozrat | timing |
| 19:20.04 | *** join/#asterisk crich1999 (~crich@86.56.0.135) |
| 19:20.12 | bjohnson | NOW!! |
| 19:20.17 | mozrat | doesn't work for me - I have a Digium card |
| 19:20.27 | mozrat | TE110P |
| 19:20.34 | jsharp | #asterisk/bdsm |
| 19:20.42 | *** join/#asterisk iframe (~iframe@201.144.1.165) |
| 19:20.44 | mozrat | music on hold and meetme doesn't work, I think cos of timing |
| 19:21.30 | bjohnson | I don't think moh uses timing |
| 19:21.37 | mozrat | is there a way to debug the timing source? |
| 19:21.38 | bjohnson | so might be a different problem there |
| 19:21.45 | mozrat | ok |
| 19:21.53 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 19:22.05 | bjohnson | maybe start with moh .. can you playback() a file? |
| 19:23.31 | mozrat | I can indeed sir |
| 19:23.38 | iframe | are thero rpm's for fedora fc3? |
| 19:23.41 | djMax | ok, the magic meetme feature is ready says anthm! I hope we can get it through the submission process! |
| 19:23.47 | mozrat | [sir/ma'am] delete where applicable |
| 19:24.15 | djMax | and I'm making good on my impromptu bounty |
| 19:24.44 | johnnyb | How does one get an extension which will ring several phones at once, without having to have an infinitely long dial string. |
| 19:25.15 | dmccollum | Create a ring group and assing all those extensions to the ring group. |
| 19:25.30 | dmccollum | assing = assign |
| 19:25.32 | johnnyb | How does one dial a ring group? |
| 19:26.21 | johnnyb | And what file are they created in? |
| 19:26.46 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 19:26.46 | *** mode/#asterisk [+o twisted] by ChanServ |
| 19:28.08 | [TK]D-Fender | johnnyb : Create a variable containing the people (or a subset of them and combine the groups) and us that. |
| 19:28.19 | [TK]D-Fender | That'd be in extensions.conf |
| 19:28.25 | johnnyb | [TK]D-Fender: Thanks |
| 19:28.37 | *** part/#asterisk iframe (~iframe@201.144.1.165) |
| 19:28.49 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 19:28.50 | [TK]D-Fender | np. I have a "Ringall" created to ring all of my home phone and pass that to "Dial" |
| 19:29.01 | johnnyb | [TK]D-Fender: Is there a way to create a variable in sip.conf that can be accessed from extesnions.conf? |
| 19:29.07 | sivana | with queues, is there a way to have agents automatically log in ? |
| 19:30.27 | shido6 | yes |
| 19:31.02 | sivana | I need to set up a queue with 2 agents.. continously logged in |
| 19:31.08 | dmccollum | I have a ring group setup with just one voicemail box that all unanswered calls goto. Then I have *97 setup to dial that voicemail and skip the password. Works great for home. Just have to hit the mail button on the phone and goes right into voicemail. |
| 19:31.27 | dmccollum | from any phone. |
| 19:31.48 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
| 19:39.47 | tld | Cheapest best way to get people up and running against Asterisk with adapter? Anything Sipura? |
| 19:39.51 | djMax | is mantis signup broken or just damn slow? |
| 19:40.34 | bjohnson | johnnyb: I think ring groups are a zapata.conf only feature |
| 19:40.50 | mozrat | is there a way to get asterisk applications (such as music on hold) to produce more debugging? |
| 19:40.50 | bjohnson | I think [TK]D-Fender's idea is best |
| 19:41.01 | bjohnson | set verbose 5 |
| 19:41.23 | mozrat | bjohnson: as in asterisk -rvvvvvv ? |
| 19:41.24 | HeadachesAbound | you should be able to do a ring group with any Channel i think. |
| 19:41.29 | bjohnson | mozrat: make sure you're using the m option in your dial command |
| 19:41.31 | HeadachesAbound | I do it with SIP extensions. |
| 19:42.00 | johnnyb | HeadachesAbound: so how do you dial the ring group? |
| 19:42.07 | bjohnson | HeadachesAbound: how? |
| 19:42.46 | mozrat | bjohnson: tnx |
| 19:42.46 | bjohnson | how do you make a ring group? |
| 19:42.46 | *** part/#asterisk Dovid (~hirisk@pool-151-198-15-84.mad.east.verizon.net) |
| 19:42.47 | bjohnson | tld: cheapest way is softphones and pay per minute voip provider |
| 19:42.54 | HeadachesAbound | exten => 2,1,Dial(SIP/234&SIP/235,10,t) |
| 19:43.10 | bjohnson | that's not a ring group |
| 19:43.12 | bjohnson | bah |
| 19:43.23 | bjohnson | that's just ringing multiple phones |
| 19:43.27 | tld | bjohnson: Yeah, I was thinking of getting them up and running with X-Lite, but I'm looking for something to offer them if they want a 'real' phone. |
| 19:43.36 | HeadachesAbound | doh, ring group...uh yeah, right, what's a ring group? round robin sorta thing? |
| 19:43.47 | tld | bjohnson: You want them to ring one at a time, or a random free operator? |
| 19:44.02 | bjohnson | I think groups are a zapata.conf only feature |
| 19:44.22 | johnnyb | tld: we like our grandstreams, but I've heard SIPura's are better. |
| 19:44.29 | *** join/#asterisk Ferrari (~IPlexbyVe@66.64.128.142.nw.nuvox.net) |
| 19:44.38 | bjohnson | tld: no .. I was just surprised such a feature existed since I looked for it a few months ago and couldn't find anything |
| 19:44.43 | blitzrage | all hail kram! |
| 19:44.44 | Ferrari | good day everyone |
| 19:44.46 | blitzrage | :) |
| 19:44.48 | Ferrari | anyone know if there is a version of app_veletparking that works with astersisk 1.0X... I need to have a way to park a call to a specific parking slot and then retrieve it later, finally once the call is retrieved i have to be able to perform # transfers.... Is this a realistic concept. Thanks |
| 19:45.01 | bjohnson | SPA 841 voip phones are supposed to be good for <$100 phones |
| 19:45.19 | HeadachesAbound | are you thinking of being able to send a call to a group of lines, like you do with outgoing pri calls? |
| 19:45.20 | blitzrage | Ferrari: I don't think that app has been updated for quite some time - you'll either have to code it yourself or pay someone to update it for you I suspect |
| 19:45.30 | tld | neat. So Sipura SPA 841 and one of the Sipura adapters to make present a low-end offer, and Ciscos for a high end. :) |
| 19:45.34 | blitzrage | bjohnson: they are - I've used one |
| 19:45.44 | bjohnson | HeadachesAbound: I'm just referring to dmccollum's instructions to johnnyb |
| 19:45.57 | tld | Sipura phone/adapter works well with Asterisk through NAT? (* on a public, client behind nat) |
| 19:46.04 | blitzrage | tld: Cisco 7940/7960 and Polycom IP500 are my choices |
| 19:46.07 | Ferrari | what is the best way to go about finding an eager programmer to assist (for hire) |
| 19:46.08 | shido6 | damn right |
| 19:46.21 | blitzrage | Ferrari: you can make a bounty on the voip-info.org website |
| 19:46.22 | shido6 | polycomm or crisco |
| 19:46.25 | HeadachesAbound | oh, so the key is to avoid the infinitely long dial string... |
| 19:46.38 | blitzrage | shido6: oh yah... I have the IP500 on the left of my monitor, the 7960 to the right :) |
| 19:46.42 | tld | blitzrage: thanks. |
| 19:46.46 | Moonwick | Ferrari: look on the nearest streetcorner for an unshaven geek holding a "will code for food" sign? |
| 19:46.46 | sivana | how do you turn off the music for queues? |
| 19:46.50 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 19:46.53 | bjohnson | tld: currently SPA devices are all SIP .. which blows when dealing with NAT .. but you can get it to work if you try hard enough |
| 19:46.59 | Ferrari | :-) |
| 19:47.07 | [TK]D-Fender | blitzrage : I'm looking to implement * at my work and am swinging the IP500's with PoE. Have you used PoE on them yet> Also how easy is it to develop XML services for those phones? |
| 19:47.08 | bjohnson | tld: a SPA 841 does not need an adapter |
| 19:47.08 | djMax | cvs head build: "no rule to make target h323/Makefile.ast"... Thoughts? Google turned up nothing. |
| 19:47.11 | tld | thanks guys. |
| 19:47.30 | blitzrage | bjohnson: you just need to understand how NAT works :) |
| 19:47.36 | blitzrage | bjohnson: not you specifically... you generally :) |
| 19:47.45 | tld | Does the Polycoms offer a browser/menu system like the Cisco? |
| 19:47.48 | djMax | I didn't even know the Polycom's would support XML services. Are you sure they do? |
| 19:47.58 | tld | Can I set up a custom environment to allow users to sign in, access corporate info etc? |
| 19:48.00 | [TK]D-Fender | I believe so... I will verify... |
| 19:48.22 | blitzrage | [TK]D-Fender: honestly not sure, I just go the phone and haven't plugged it in yet :) I don't use PoE here unfortunately |
| 19:48.43 | blitzrage | [TK]D-Fender: yah, let me know, I'm curious about the XML :) |
| 19:48.56 | bjohnson | tld: through an IVR .. definitely .. through a screen .. depends on the hardware |
| 19:48.58 | blitzrage | rumours say Polycom used to make Cisco's phones for them |
| 19:49.07 | tld | bjohnson: I was thinking of Polycom hardware. |
| 19:49.26 | bjohnson | tld: thought you wanted cheap |
| 19:49.28 | djMax | rumours and the fact that when I pull off one of the covers on the poly I see a cisco log |
| 19:49.30 | djMax | logo |
| 19:49.42 | tld | bjohnson: Yeah, but then I started dreaming. ;) |
| 19:49.52 | bjohnson | polycom 500 = $200 each |
| 19:50.05 | tld | nice |
| 19:50.15 | bjohnson | SPA 841 = $90 each |
| 19:50.16 | ionix | check froogle |
| 19:50.20 | tld | though with shipping and norwegian taxes, that's $350. ;) |
| 19:50.20 | ionix | for the ip500 for 164 |
| 19:50.41 | Nivex | has anyone seen the grandstream hard phone offering? |
| 19:50.46 | [TK]D-Fender | blitzrage : their spec sheets say XML... |
| 19:50.52 | blitzrage | [TK]D-Fender: nice! |
| 19:51.07 | blitzrage | shit > grandstream anything |
| 19:51.07 | tld | really nice. :) |
| 19:51.14 | blitzrage | ok... <= |
| 19:51.15 | [TK]D-Fender | Have you worked with PoE? |
| 19:51.16 | blitzrage | errr |
| 19:51.18 | blitzrage | >= |
| 19:51.23 | tld | So sipura for lowend, cisco/poly for high. Simple enough. :) |
| 19:51.31 | *** part/#asterisk Ferrari (~IPlexbyVe@66.64.128.142.nw.nuvox.net) |
| 19:51.31 | tld | Now I just need to get my hands on a couple of sipuras and a poly. |
| 19:51.35 | blitzrage | tld: that seems to be what I'm looking at now |
| 19:51.45 | [TK]D-Fender | TLD : Uniden UIP-200 = great midrange |
| 19:51.50 | blitzrage | tld: Maybe even Poly over Cisco unless someone REALLY wants a "Cisco" |
| 19:51.53 | tzanger | bjohnson: I hate wall warts |
| 19:51.54 | tzanger | with a passion |
| 19:52.05 | djMax | actually, the admin guide for the IP500 says "Soundpoint IP supports an XHTML microbrowser" |
| 19:52.20 | djMax | and the config file can specify its homepage. |
| 19:52.24 | shido6 | the uniden is not bad |
| 19:52.32 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 19:52.43 | tld | bjohnson: Compared to having UPS with each phone, it's dirt cheap. |
| 19:52.51 | tld | bjohnson: You reduce to a single UPS for the switch |
| 19:52.52 | bjohnson | tzanger: passion lets you know you're alive |
| 19:53.01 | PBXtech | can you do a flash hook on an FXO card? |
| 19:53.05 | [TK]D-Fender | djMax : I'll have to follow up on that. Whatever it takes to get web-like functionality on those things... |
| 19:53.05 | tld | [TK]D-Fender: Thanks. Hadn't heard anything about the unidens, so it's good to know. :) |
| 19:53.07 | tzanger | yup and POE keeps those fucking wall warts out of my system |
| 19:53.18 | tzanger | a centralized "big" power supply is far more efficient anyway |
| 19:53.31 | djMax | the polys actually inject the wall wart into the ethernet cable that goes to the phone, so I can't imagine why they'd use something other that PoE type electrical specs. |
| 19:53.36 | bjohnson | PBXtech: supposed to be able to .. info on wiki |
| 19:53.36 | djMax | Except that they're polycom. |
| 19:53.44 | [TK]D-Fender | By wall-warts I presume you mean the transformer blocks? |
| 19:53.51 | blitzrage | shido6: the UIP200 isn't too bad - I liked the 841 better though |
| 19:54.04 | djMax | 12vdc, 400ma. Ugh. |
| 19:54.14 | bjohnson | [TK]D-Fender: yes |
| 19:54.22 | PBXtech | [bjohnson]: do you know if it works well? |
| 19:54.29 | bjohnson | PBXtech: no |
| 19:54.35 | *** join/#asterisk Dovid (~hirisk@pool-151-198-15-84.mad.east.verizon.net) |
| 19:54.35 | bjohnson | no idea |
| 19:54.39 | PBXtech | k |
| 19:55.10 | bjohnson | but someone was here about a month ago, releived that his crappy call waiting started to work |
| 19:55.36 | [TK]D-Fender | tzanger : What switch do you use to power yours? Had problems with any specific model? I'm looking at a D-Link 1526 right now |
| 19:56.04 | tzanger | I don't use SIP phones |
| 19:56.10 | tld | Off topic, but anyone know of a online fax service that's recommended? Could use something to go along my asterisk? |
| 19:57.10 | tld | tzanger: what do you use then? |
| 19:57.33 | tzanger | tld: phones. regular phones |
| 19:58.50 | Nivex | tzanger: through an ATA or an FXS card? |
| 19:59.24 | tzanger | yup |
| 19:59.58 | file | blitzrage: MEEP MEEP |
| 20:00.36 | blitzrage | meep |
| 20:01.02 | tld | tzanger: Which one? ATA og FXS? |
| 20:01.15 | tzanger | TDM400P and T1+CB |
| 20:01.27 | blitzrage | tzanger: what phones? I like the N.T. Vista series |
| 20:01.34 | tzanger | yeah they're nice |
| 20:02.05 | *** join/#asterisk UltraGra (~grahamoco@82.153.131.183) |
| 20:02.47 | sivana | tzanger: I solved the knox issue with a queue! |
| 20:02.57 | sivana | slight music at the beginning though |
| 20:03.04 | sivana | would like to replace that with a ring or nothing |
| 20:03.29 | tzanger | what was knox' issue? |
| 20:03.43 | sivana | well. I put in two Sipura 2000s |
| 20:03.47 | file | hi blitz |
| 20:03.56 | UltraGra | hi all |
| 20:04.00 | sivana | and needed to "huntgroup" the SIP users |
| 20:04.17 | sivana | so I put those channels into a knox queue |
| 20:04.27 | jskcr|lappy | sivana how do you like the sipura-2000's |
| 20:04.39 | sivana | jskcr|lappy: I don't know.. the client seems to love it now |
| 20:04.44 | sivana | I never used one |
| 20:04.46 | shido6 | heh |
| 20:04.54 | sivana | having the two lines is nice |
| 20:05.42 | UltraGra | anyone using te405 telephony card? |
| 20:05.42 | tzanger | UltraGra: yes |
| 20:06.05 | UltraGra | ever tried a span 2 span loop back test ? |
| 20:07.29 | UltraGra | anyone? |
| 20:09.38 | _kb1_kanobe | UltraGra: Can't see why it wouldn't work. |
| 20:10.27 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 20:10.30 | outtolunc | the point being, using 2 ports (crossover cable), you only want to set 1 to loopback on the net side |
| 20:11.18 | UltraGra | neither could I .. we've just commissioned an 'at-home' system .. all seems ok .. green lights on the 2 looped back E1 ports .. but when placing a call we get a 'congestion' message ... |
| 20:12.11 | _kb1_kanobe | one span is network and the other cpe? |
| 20:12.57 | UltraGra | hang on |
| 20:13.38 | UltraGra | yes |
| 20:14.53 | UltraGra | does the 'congestion' problem ring any bells? - we're new to this so config could be wrong (tho digium tell us its ok) .. |
| 20:14.54 | tld | Anyone not in the US who can comment on pros and cons about ordering SIP hardware from different online stores? Anyone to stay away from? Anyone with good service and not too high shipping costs? |
| 20:16.41 | Dovid | anyone know of a site where i can see diff. major internet routes and see how the networks are doing etc. ? |
| 20:17.41 | tld | Dovid: Which level do you want it at? High level, or low level? You might want to google 'BGP looking glass' |
| 20:18.02 | bjohnson | tld: voipsupply is a favorite |
| 20:18.04 | _kb1_kanobe | UltraGra: 'congestion' means lots of things, all of which imply 'can't get there from here'. keep diging. |
| 20:18.05 | citats | Dovid: route-views.net |
| 20:18.28 | bjohnson | tld: the owner frequents #asterisk and the mailing lists |
| 20:18.36 | UltraGra | thanks guys .. i'll repost question after some more digging! |
| 20:18.51 | tld | neat. I've noticed the site before, and always got the warm fuzzy feeling I get from a decent online merchant. |
| 20:18.51 | tld | thanks |
| 20:19.07 | tld | Unless there's a big reason not to, I'll probably give it a try with my next order. |
| 20:19.44 | UltraGra | anyone have experience with calling card 'addons' for asterisk? |
| 20:20.15 | key2 | hey |
| 20:20.43 | key2 | On what linux is it the easyest to install asterisk ? |
| 20:21.13 | mozrat | key2, if you just want a quick install have you seen asterisk@home or xorcom? |
| 20:21.29 | key2 | no |
| 20:21.30 | key2 | ? |
| 20:23.25 | eKo1 | key2: The distro. doesn't matter. Just pick one. |
| 20:23.31 | *** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com) |
| 20:24.36 | mozrat | key2: http://asteriskathome.sourceforge.net/ |
| 20:24.42 | *** join/#asterisk crich1999 (~crich@86.56.0.135) |
| 20:24.57 | mozrat | key2: http://xorcom.com/rapid/ |
| 20:26.19 | key2 | mozrat: thanks |
| 20:26.20 | key2 | i saw |
| 20:26.28 | key2 | i just installed on a redhat |
| 20:26.29 | key2 | :) |
| 20:26.42 | *** join/#asterisk Rick_Hunter (~rhunter@02-041.008.popsite.net) |
| 20:27.00 | bjohnson | if you know redhat, you might like @home which is based on Centos .. a RHEL clone |
| 20:27.08 | Nugget | Linux is poo. |
| 20:27.13 | *** join/#asterisk nitram (foo@superblob.com) |
| 20:27.38 | key2 | how do u say in a dialplan, to dial a number received from an SIP interface on the ZAP interface ? |
| 20:27.55 | key2 | i have to set route for outgoing ? |
| 20:27.59 | bjohnson | it doesn't matter how it is received |
| 20:28.10 | bjohnson | dial(zap/1/${EXTEN}) |
| 20:28.26 | bjohnson | or some variant .. plus args if desired |
| 20:28.42 | key2 | bjohnson: so basically if I want to place a call from my SIP phone, what am I supposed to do ? |
| 20:28.56 | bjohnson | pick it up and starting hitting numbers |
| 20:28.59 | Nugget | configure asterisk. |
| 20:29.21 | key2 | yeah but I have to configure it in a way that if an outgoing call is placed, so it uses zaptel right ? |
| 20:29.32 | bjohnson | make sure it is configured in sip.conf and point it at the context in extensions.conf that allows you to dial() through an outside channel |
| 20:30.32 | Nivex | [default] |
| 20:30.45 | Nivex | exten => 1234,1,Dial(Zap/1/${EXTEN}) |
| 20:31.01 | bjohnson | well .. that would dial 1234 |
| 20:31.05 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
| 20:31.07 | bjohnson | not usable for most people |
| 20:31.14 | Nivex | oh, duh |
| 20:31.40 | Nivex | exten => .9_,1,Dial(Zap/1/${EXTEN:1}) |
| 20:32.12 | key2 | Nivex: it means dial the digit received from EXTEN and drop the 1 ? |
| 20:32.28 | Nivex | key2: no, it means drop the first digit (in this case, the 9) |
| 20:32.42 | Nivex | so you'd dial 9 and then the number, and the call would get routed out through Zap/1 |
| 20:32.59 | key2 | Nivex: thanks |
| 20:33.02 | *** part/#asterisk mbishop (~martin@mbishop.user.gentoo) |
| 20:33.09 | key2 | Nivex: do u know otherwise how to call back a number ? |
| 20:33.39 | key2 | like if I call my FXO from a phone and live one ring, it calls back the number received from the callerid |
| 20:33.39 | key2 | ? |
| 20:33.39 | PBXtech | does anyone know how the RANDOM command works? |
| 20:34.37 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 20:34.37 | *** join/#asterisk sd-tux (sd@2001:6f8:1372:0:0:0:0:2) |
| 20:36.23 | machinehd | Anyone using the Cisco Conference Station 7936? If so, would you recommend it? |
| 20:37.06 | *** join/#asterisk bajanman2 (~william@cp66-203-194-32.cp.telus.net) |
| 20:38.16 | blitzrage | kram: you around? |
| 20:40.40 | key2 | what does .9 means in .9_,1,Dial(Zap/1/${EXTEN:1}) |
| 20:41.45 | torisa | any two digit number ending in 9 |
| 20:41.55 | Nivex | it should be _9. |
| 20:42.55 | bkw_ | thats backards |
| 20:42.58 | bkw_ | _9x |
| 20:43.00 | bkw_ | er _9. |
| 20:43.06 | bjohnson | key2: start with the basics before you get into callbacks, etc |
| 20:43.30 | key2 | so _9XXXXXXXXX == .9_ ? |
| 20:43.51 | jskcr|lappy | lol I just noticed digium stoped selling the x100p's |
| 20:43.58 | drumkilla | . matches anything |
| 20:44.07 | outtolunc | spin it around |
| 20:44.10 | drumkilla | . could be 1 or 9auepaisdjfo;aslnf0239023nlwkn234n2 |
| 20:44.11 | outtolunc | _9. |
| 20:44.17 | jskcr|lappy | when did that happen |
| 20:44.30 | drumkilla | jskcr|lappy: a long time ago |
| 20:44.36 | bkw_ | . is greeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeedy |
| 20:44.49 | jskcr|lappy | lol Ive only gotten the t1 cards lately |
| 20:45.07 | jskcr|lappy | I just wanted a x100p for the timing only |
| 20:45.34 | Nivex | jskcr|lappy: use the ztdummy module with a USB controller |
| 20:45.35 | jsharp | They were probably taking a beating on the folks selling the "clone" cards. |
| 20:46.07 | jskcr|lappy | Nivex, umm music on hold iiisss iiikkiii someetiimees |
| 20:46.12 | drumkilla | the tdm card is just obviously a better product |
| 20:46.16 | blitzrage | drumkilla: !!!!!!!!!!!!!! |
| 20:46.20 | blitzrage | drumkilla: how was the 2nd test? |
| 20:46.28 | drumkilla | much better than the first |
| 20:46.29 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 20:46.30 | jskcr|lappy | yea this is for a internal sip only network for the timing for music on hold. |
| 20:46.30 | citats | drumkilla: how goes your finals? |
| 20:46.39 | PBXtech | anyone use a ML110 G2 server? |
| 20:47.01 | drumkilla | citats: done now :) |
| 20:47.15 | citats | congrats |
| 20:47.24 | citats | always a good feeling to have that weight off your shoulders |
| 20:47.38 | citats | regardless of how well you did :) |
| 20:47.41 | drumkilla | yeah, it is ... thanks :) |
| 20:47.47 | *** join/#asterisk cjk (~cjk@80.92.75.4) |
| 20:47.49 | drumkilla | this morning was rough ... but, it's over |
| 20:47.50 | bjohnson | key2: no _9XXXXXXXXX != .9_ |
| 20:48.03 | citats | till the fall? or are you done done? |
| 20:48.10 | cjk | hi, rxfax iw working great on my system, but i do not find a decent doc on txfax? anyone a good link? |
| 20:48.33 | key2 | bjohnson: where do I find docs about that? |
| 20:48.35 | Meaty | There are a setting in asterisk to automaticaly change the playtone when there are one or more new message in the user voicemail ? |
| 20:49.00 | Meaty | are there* |
| 20:49.16 | bjohnson | ~docs |
| 20:49.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:49.28 | bjohnson | look at the dialplan page |
| 20:49.33 | bjohnson | or extensions.conf |
| 20:50.09 | bjohnson | Meaty: I think that it is a device feature. My SPA's play a double tone |
| 20:50.28 | Meaty | ok |
| 20:50.44 | key2 | i looked at the doc on asteriskdocs.org but it doesnt talk about .9_ format |
| 20:50.53 | blitzrage | key2: because there is NO SUCH THING! |
| 20:51.26 | PBXtech | linda like i before e |
| 20:51.26 | Meaty | _9. ? |
| 20:51.26 | blitzrage | yes |
| 20:51.26 | blitzrage | _ means you are pattern matching |
| 20:51.33 | key2 | teah |
| 20:51.34 | blitzrage | the . means anything and everything past that point |
| 20:51.34 | key2 | yeah |
| 20:51.39 | key2 | ok |
| 20:51.41 | blitzrage | .9_ means nothing |
| 20:51.50 | *** join/#asterisk EnigmaPTK (~bkwb@adsl-69-212-249-116.dsl.sfldmi.ameritech.net) |
| 20:51.51 | Meaty | means error |
| 20:51.54 | Meaty | :P |
| 20:51.55 | Nivex | key2: I'm sorry, I typed it backwards the first time. :( |
| 20:51.55 | EnigmaPTK | Afternoon everyone... |
| 20:51.58 | blitzrage | Meaty: aye! |
| 20:52.03 | key2 | ok :) |
| 20:52.06 | EnigmaPTK | Anyone Know anything about the Avaya 4630SW Phone? |
| 20:52.08 | blitzrage | Nivex: thats it... YOU DIE NOW! |
| 20:52.09 | Meaty | Whats a SPA ? |
| 20:52.10 | key2 | Nivex: that's why I didnt get it |
| 20:52.17 | blitzrage | Meaty: a place I never get to go to |
| 20:52.18 | EnigmaPTK | Available Yet? SIP Work worth a damn? |
| 20:52.33 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.167) |
| 20:52.35 | Nivex | blitzrage: I corrected myself! I guess noone saw the correction :( |
| 20:52.48 | blitzrage | Nivex: guess not :) |
| 20:53.16 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 20:55.19 | key2 | someone has tryed SJphone with asterisk |
| 20:55.19 | key2 | ? |
| 20:55.28 | CleanerX | jo |
| 20:55.33 | CleanerX | works pretty fine |
| 20:55.38 | key2 | ok |
| 20:55.48 | key2 | CleanerX: how do u declare sjphone is sip.conf |
| 20:56.02 | CleanerX | no specials |
| 20:56.07 | *** join/#asterisk Veryhot (~tho@adsl-69-109-159-210.dsl.sndg02.pacbell.net) |
| 20:56.22 | CleanerX | just the usual settings as for every phone |
| 20:57.08 | CleanerX | forgot what the dtmf was, but you just have three choices so it should be easy to find out which one worked |
| 20:57.20 | key2 | when I place a call from it, it says "no one can be reach at this time" |
| 20:57.41 | key2 | CleanerX: rfc2883 ? |
| 20:59.23 | tld | Can anyone recommend a SIP or IAX provider that will let me re-sell minutes? Preferrably one who also offers DIDs? |
| 20:59.50 | Goshen | IAXy seems to be absorbing *67... |
| 21:00.03 | CleanerX | enable sip debugging in asterisk |
| 21:00.08 | Goshen | I dial *675555555 and only the 5555555 goes to the server |
| 21:00.15 | CleanerX | you will get the problem there |
| 21:00.22 | Goshen | when I dial *67 I get a stutter and return of dialtone |
| 21:00.32 | Goshen | so it must be the IAXy that is absorbing the *67 |
| 21:00.36 | CleanerX | @key2 |
| 21:00.40 | bjohnson | Meaty: a SPA is a place where you go for a manicure |
| 21:00.51 | Goshen | Guess I have to use another code for passing to the server to pass to my provider |
| 21:01.27 | bjohnson | tld: all of them |
| 21:02.10 | Meaty | bjohnson : k thanks :) |
| 21:02.20 | pjz | what protocol does a TE110P suport? NI1? NI2? |
| 21:02.27 | jsharp | Both. |
| 21:02.29 | Meaty | I have searched on google : spa voip :P |
| 21:02.31 | pjz | if I have a choice when I'm orddering a PRI, what should I get it set to? |
| 21:02.31 | bjohnson | Meaty: it's also a Siprua product line |
| 21:02.33 | tld | bjohnson: All of them provide DIDs, or all of them would let me resell? |
| 21:02.36 | jsharp | NI-2 |
| 21:02.38 | Meaty | lol yeah |
| 21:02.50 | bjohnson | err Sipura |
| 21:02.51 | Meaty | bye ! I must leave, thx |
| 21:03.09 | Goshen | AHA! I dial 67555-5555 on my IAXy phone, then tell asterisk to append * and dial...got it :) |
| 21:03.10 | CleanerX | sipura has been bought by cisco :-) |
| 21:03.14 | bjohnson | tld: call them |
| 21:03.29 | bjohnson | sipura bought by cisco? shit |
| 21:04.04 | tld | bjohnson: d-link divition |
| 21:04.45 | Veryhot | ? |
| 21:05.07 | Veryhot | bj: yeah they will buy vonage soon :) |
| 21:05.20 | tld | Google/Cisco merger would be interesting. ;) |
| 21:05.39 | Veryhot | tld: how about M$ buy cisco? :) |
| 21:05.50 | tld | *shrug* |
| 21:05.52 | jsharp | Aieee. |
| 21:05.53 | Veryhot | tld: winIOS |
| 21:05.58 | tld | eek |
| 21:06.05 | tld | Windows Router edition |
| 21:06.30 | Veryhot | yeah. with updates |
| 21:07.08 | tld | First thing to do after upgrade is to connect VGA and keyboard to device. |
| 21:07.20 | jsharp | And your 7206VXR gets replaced with a Dell PeeCee with a T3 card in it. |
| 21:07.23 | jsharp | heh. |
| 21:07.42 | Veryhot | Didn't Sipura founder created the ATA186? |
| 21:07.55 | *** join/#asterisk fcgreco (~fcgreco@200.245.73.163) |
| 21:11.04 | bjohnson | dlink bought sipura? |
| 21:11.07 | bjohnson | crap |
| 21:11.34 | bjohnson | Veryhot: yeah |
| 21:11.46 | fcgreco | hi. I need an information. I am using two analog lines with digium TDM card. I would like to know how can I dial any number , like "0", to pick the line, and start dialing a phone number. Can anyone help?? |
| 21:12.19 | bjohnson | use a pattern match that starts with _0 |
| 21:12.34 | bjohnson | and remove that digit from the {EXTEN} |
| 21:12.44 | bjohnson | oops .. I've said too much |
| 21:13.23 | fcgreco | but I can not listen the dial tone!! I am used to dial 0 in my PBX siemens and get the line! |
| 21:14.01 | *** join/#asterisk HeadachesAbound (~mirc@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 21:17.11 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:17.29 | puowvip | sigh. |
| 21:18.42 | *** join/#asterisk at561 (~aoiahsdf@68.71.213-37.atlsfl.adelphia.net) |
| 21:19.31 | jakepdev | anyone know how do get peak port usage in asterisk? |
| 21:21.06 | pjz | jakepdev: I don't, but that would be cool! maybe get something like rrdtool to monitor port usage? |
| 21:21.47 | jakepdev | i know it can be done via a creative sql query, but was hoping not to have to go that route |
| 21:21.49 | _kb1_kanobe | if I change address bindings in iax.conf will a reload remap them or do I need to restart *? |
| 21:23.14 | jakepdev | kb1 - i think a reload will do it |
| 21:24.06 | jakepdev | pjz - yep - but i'll settle for some nice text based results at this point |
| 21:24.47 | *** join/#asterisk redG ([U2FsdGVkX@67.51.185.15) |
| 21:24.48 | redG | <PROTECTED> |
| 21:25.50 | jakepdev | fcgreco - if you want to hear the dial tone from your other switch, you need just 0 not the _0 |
| 21:28.09 | Goshen | Can't he just put in ignorepat=0 ? |
| 21:28.43 | jakepdev | that won't give dial tone from his other switch |
| 21:28.55 | *** join/#asterisk salviadud (~dude@201.129.86.120) |
| 21:28.59 | jakepdev | that'll give * dialtone |
| 21:29.02 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 21:29.07 | salviadud | anyone have a fwd numer? |
| 21:29.13 | salviadud | number |
| 21:29.21 | salviadud | i want to test if i can receive calls |
| 21:29.53 | jakepdev | salviadud - you know fwd provides a way to give you a test call |
| 21:29.59 | jakepdev | on their website |
| 21:30.04 | salviadud | i know i can dial |
| 21:30.07 | jakepdev | no |
| 21:30.12 | jakepdev | it can dial you |
| 21:30.19 | salviadud | the echo test? |
| 21:30.22 | Goshen | you go to the website and tell it to call you |
| 21:31.09 | jakepdev | it's different than an echo test |
| 21:31.26 | salviadud | all right |
| 21:31.30 | salviadud | im testing it right now |
| 21:31.34 | Goshen | I will call you, whats your number? |
| 21:31.44 | jakepdev | anyone else suddenly getting tons of spam? |
| 21:31.57 | Goshen | jakepdev: nope |
| 21:31.58 | jakepdev | within the past few days |
| 21:32.25 | salviadud | damn, seems like it don't work... |
| 21:32.26 | Goshen | salviadud: what is your number? |
| 21:32.40 | salviadud | 651692 |
| 21:33.29 | Goshen | hmm, have to edit my dialplan...you have so many digits, it goes local over 7 digit dialing :) |
| 21:33.33 | *** join/#asterisk alegh (~ag11@OL217-17.fibertel.com.ar) |
| 21:33.56 | jakepdev | loco or local? |
| 21:34.09 | salviadud | mmmm |
| 21:34.14 | *** join/#asterisk iq (~iq@65-103-167-189.omah.qwest.net) |
| 21:34.18 | alegh | hi, anyone with experience with sangoma cards with *? |
| 21:34.23 | Goshen | dials my local calling area 365-1692 |
| 21:34.24 | salviadud | i think its my sipura |
| 21:34.29 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 21:34.30 | salviadud | badly configured or something |
| 21:34.43 | jakepdev | oh |
| 21:34.44 | Veryhot | anyone using TelaSIP? |
| 21:37.06 | *** join/#asterisk Jas_Williams (~jas_willi@217.41.232.141) |
| 21:37.10 | djMax | one more attempt: anybody know why I would get errors making chan_h323.so with cvs head? |
| 21:37.11 | salviadud | does somebody understand the block anonymous call service? |
| 21:38.13 | Sato1 | for digium? |
| 21:38.27 | *** join/#asterisk pat_lehem (lehem@bzq-218-239-63.red.bezeqint.net) |
| 21:38.49 | Sato1 | that will link your asterisk with digium system so if you have questions, you can use your own asterisk to call them |
| 21:38.51 | salviadud | sipura 2000 or 3000 |
| 21:38.51 | tzafrir_laptop | djMax, presenting those errors may help folks here with the guess work |
| 21:39.54 | djMax | surely. Just wanted to see if it was a common thing first. "No rule to make target 'h323/Makefile.ast' needed by 'chan_h323.so'. Stop." |
| 21:40.07 | Jas_Williams | I'm just compiling cvs head h323 |
| 21:40.22 | Jas_Williams | That was an error a few days ago |
| 21:40.48 | Jas_Williams | got to src/asterisk and do a make update then try again |
| 21:41.24 | Sato1 | h323 does not compile if you dont do the openh323 and pwlib |
| 21:41.33 | salviadud | how do i find out if i need STUN or Outbound proxy config? |
| 21:41.49 | Jas_Williams | also you you need to make opt in channels/h323 |
| 21:42.03 | Sato1 | salviadud, just if you are behind a firewall |
| 21:42.10 | *** part/#asterisk pat_lehem (lehem@bzq-218-239-63.red.bezeqint.net) |
| 21:42.11 | Jas_Williams | follow every step in asterisk/channels/h323/README |
| 21:42.18 | salviadud | i am behind a firewall |
| 21:42.37 | *** join/#asterisk pat_lehem (lehem@bzq-218-239-63.red.bezeqint.net) |
| 21:43.01 | salviadud | i used the outbound proxy thing anyway, i dunno if thats the best thing |
| 21:43.12 | Sato1 | the outbound proxy its when you are behind a firewall and you have a sip proxy with a real ip and an ip behind the firewall |
| 21:43.30 | djMax | make opt did it. thanks. Did I miss that somewhere? |
| 21:43.40 | salviadud | well i guess im ok |
| 21:43.49 | salviadud | i still can't receive calls though... |
| 21:43.52 | pat_lehem | What is the best way to track which event relates to which action in the manager API? |
| 21:44.12 | Sato1 | if your asterisk its in your router box, then you dont need it, but you would need to set canreinvite=no in the device's sip configuration, that will make your asterisk act as a proxy with outside calls |
| 21:44.24 | alegh | anyone who tested dtmf callerid? |
| 21:45.01 | *** join/#asterisk tikkker (~tikkker@pD9580827.dip.t-dialin.net) |
| 21:45.02 | Jas_Williams | djMax, now do a make clean; make install in the asterisk directory to ensure asterisk is compiled correctly with h323 |
| 21:45.16 | *** part/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 21:45.24 | pat_lehem | I looked it up, and it seems ActionID is supposed to be a way to track actions. However, it seems most events do not send the ActionID back. |
| 21:45.30 | pat_lehem | so I was wandering about that... |
| 21:45.32 | djMax | ok, doing now. I always wondered, should I be stopping asterisk while doing "make install"? |
| 21:45.52 | Jas_Williams | no performance may be affected tho |
| 21:46.06 | djMax | k |
| 21:46.43 | djMax | guess I'll have to reread how the linux fs works, because that seems like a nifty trick. |
| 21:47.32 | Jas_Williams | djMax, asterisk is running in memory, once new version is made and installed, do a restart now in the cli |
| 21:47.47 | Sato1 | djMax, follow what channels/h323/README says, you need to point some variables to openh323 and pwlib, otherwise you wont compile it |
| 21:48.39 | Sato1 | some environment varialbes |
| 21:49.04 | tikkker | i want to use the ztdummy timer but asterisk cannot find the channel "zap" |
| 21:49.10 | tikkker | i got 2.6 kernel |
| 21:49.26 | tikkker | and modprobe zaptel and ztdummy also dont complain |
| 21:49.26 | *** join/#asterisk jonathh1 (~asd@host-84-9-23-98.bulldogdsl.com) |
| 21:49.29 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 21:49.49 | jonathh1 | hey guys i have just plugged in my x100p for the first time.. is there anyway i can tell if asterisk knows it is there |
| 21:49.49 | jonathh1 | ? |
| 21:49.59 | |Vulture| | Anyone here know a good source for a sample Telephone Usage Policy, all I can find on google are Cell Phone Policies |
| 21:52.06 | Sato1 | jonathh1, you need to compile the zaptel driver, and let your system recognize it, then do the modprove zaptel and configure your zaptel.conf and zapata.conf |
| 21:52.42 | Sato1 | |Vulture|, www.voip-info.org may help you find more specific samples |
| 21:53.54 | |Vulture| | Sato1: thanx didn't even think of checking the wiki |
| 21:54.15 | Sato1 | no problem |
| 21:54.37 | xeet2 | vulture: you can also sign up for a sip provider and look at theirs |
| 21:54.47 | xeet2 | they're all pretty much the same anyway though |
| 21:55.38 | tikkker | anybody here already done the ztdummy trick with a 2.6 kernel ? |
| 21:56.51 | BoRiS | lol Vco |
| 21:56.54 | *** part/#asterisk pat_lehem (lehem@bzq-218-239-63.red.bezeqint.net) |
| 21:58.17 | *** join/#asterisk meppl (mephisto@p54AAFA96.dip.t-dialin.net) |
| 21:59.31 | alegh | anyone with experience with sangoma cards with *? |
| 22:00.23 | meppl | guten abend |
| 22:00.46 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 22:02.04 | |Vulture| | yea I am looking for one to use in my office |
| 22:04.41 | Sato1 | |Vulture|, you mean setting extensions.conf to restrict calls to your users, or route specific calls to sip/iax providers? |
| 22:07.54 | *** join/#asterisk durex (~ironman@200.199.203.125) |
| 22:08.36 | durex | asterisks.... |
| 22:08.42 | durex | I'm having problem with cdr mysql compiled on FreeBSD... |
| 22:08.56 | Sato1 | so.. whats the problem? |
| 22:09.09 | durex | I compiled asterisk-addons and instaled, but when try to run Asterisk, I got the followin error: |
| 22:09.11 | durex | <PROTECTED> |
| 22:09.33 | durex | | /libexec/ld-elf.so.1: /usr/local/lib/asterisk/modules/res_config_mysql.so: Undefined symbol "ast_config_load" |
| 22:09.49 | xeet2 | did you recompile asterisk after asterisk-addons? |
| 22:09.56 | durex | no... |
| 22:10.15 | |Vulture| | Sato1: no something that says that "you agree your calls can be monitored or recorded... you can only use phones for business purposes.. etc." |
| 22:10.44 | Sato1 | |Vulture|, oh |
| 22:11.06 | durex | should I recompile asterisk after compile and install asterisk-addons? |
| 22:11.07 | xeet2 | durex: I've seen that suggested - and working on similar issues with the addons |
| 22:11.19 | Sato1 | i've never used that about the Policies, but i remember seeing that in voip-info |
| 22:11.51 | |Vulture| | Sato1: yea I want to make sure I get them to sign it before I start to record... and we have a message that says it when people dial in |
| 22:12.07 | durex | xeet2 are you running asterisk on FBSD? |
| 22:12.26 | xeet2 | durex: no, but that issue doesn't appear to be platform specific |
| 22:12.41 | xeet2 | seen that same error on linux |
| 22:13.19 | Sato1 | well, i just did the addon for mysql yesterday, and the only error i got was that i was not pointing to the right path for the mysql.sock |
| 22:13.28 | durex | xeet2 :( |
| 22:13.30 | durex | any idea? |
| 22:13.45 | Sato1 | durex, did you recompile your asterisk and tried again? |
| 22:13.50 | xeet2 | durex: did you try to recompile asterisk? also, did you update to the latest cvs? |
| 22:13.59 | durex | Sato1 I'll try it now... |
| 22:14.04 | *** join/#asterisk cyt0plas (~cyt0plas@masq.adoptionmedia.com) |
| 22:14.21 | cyt0plas | Hi all. |
| 22:15.28 | durex | well... |
| 22:15.31 | *** join/#asterisk harryvv (~none@S010600055d210201.vs.shawcable.net) |
| 22:15.37 | durex | I have instaled asterisk from ports tree... |
| 22:15.49 | harryvv | Where are the cheapest ip300s I can order from. |
| 22:15.50 | durex | and now, downloaded asterisk and asterisk-addons via CVS to /usr/src |
| 22:16.13 | durex | modified the follow lines of my /usr/src/asterisk-addons/Makefile: |
| 22:16.25 | cyt0plas | Anyone ever dealt with doing a man-in-the-middle setup between a T1 PRI and a T1 (non-PRI) legacy PBX? |
| 22:16.26 | durex | CFLAGS+=-I../asterisk/include -I/usr/local/include -I../asterisk |
| 22:16.33 | durex | and |
| 22:16.35 | durex | ASTLIBDIR=$(INSTALL_PREFIX)/usr/local/lib/asterisk |
| 22:16.47 | Damin | Hmmm.. |
| 22:16.48 | Damin | line 146: Unable to open master device '/dev/zap/ctl' |
| 22:16.49 | xeet2 | harryvv: what is "cheap" to you? (what price are you trying to beat) |
| 22:16.59 | harryvv | 130 dollars per phone |
| 22:17.07 | Damin | First attempt at building Zaptel under Linux 2.6 on Tao Linux 4. |
| 22:17.16 | xeet2 | hang on, I'll see what I can get them for |
| 22:17.17 | harryvv | Its my assumption this is the most requested phone by medium to large bussiness? |
| 22:17.26 | durex | If I specify -I/usr/ports/net/asterisk/work/asterisk-1.0.7 and -I-I/usr/ports/net/asterisk/work/asterisk-1.0.7/include |
| 22:17.34 | durex | I got an compilation error.... |
| 22:17.48 | harryvv | I want a phone that does not look to cheap like the spa 184 but is much more like the ip 300. |
| 22:17.54 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:17.55 | *** join/#asterisk kioko (~kiokorobe@196.200.26.42) |
| 22:19.13 | durex | xeet2 Sato1 take a look at the error at: http://pastebin.ca/10943 |
| 22:19.21 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:19.22 | durex | and I'm running |
| 22:19.34 | durex | FreeBSD weber.anpa.org.br 5.4-STABLE FreeBSD 5.4-STABLE #0: Wed May 4 18:20:31 BRT 2005 root@weber.anpa.org.br:/usr/src/sys/i386/compile/KERNEL2 i386 |
| 22:19.44 | xeet2 | harryvv: mmm, the price I can get them for is the same price everyone else can get them for |
| 22:19.55 | harryvv | and that is 130 |
| 22:19.55 | harryvv | ? |
| 22:20.01 | xeet2 | yes |
| 22:20.07 | harryvv | hehe |
| 22:20.08 | harryvv | even same on ebay |
| 22:20.16 | durex | xeet2 any idea? |
| 22:20.18 | harryvv | xeet, how many phones have you sold so far. |
| 22:20.41 | xeet2 | ip300's? maybe 2 or 3... done about 50 cisco 7960's though |
| 22:20.45 | xeet2 | they're quite popular =) |
| 22:20.51 | *** join/#asterisk psywar (psywar@rasterburn.org) |
| 22:20.56 | *** part/#asterisk psywar (psywar@rasterburn.org) |
| 22:21.08 | *** join/#asterisk psywar (psywar@rasterburn.org) |
| 22:21.16 | psywar | hrm |
| 22:21.25 | harryvv | really..was it partly because of the the clients requesting them? or you showed them the brochures? |
| 22:21.45 | xeet2 | we bought one with us and plugged it into their then-cable modems |
| 22:21.45 | durex | I'm really don't know more what to do with that.... |
| 22:21.55 | durex | app_addon_sql_mysql.c:162:36: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given |
| 22:21.56 | xeet2 | er, brought |
| 22:21.59 | durex | going crazy! |
| 22:22.01 | psywar | when a number calls me, I need to hit DTMF 9, but it doesn't appear to "go through" my SPA-2000 and out the zap interface. Any ideas how I fix this? tia |
| 22:22.17 | xeet2 | durex: looking at your pastebin, hang on |
| 22:23.01 | harryvv | and so you demonstrated it by slapping in your pbx and phone on there existing network and were impressed then? Who do you normally focus on for a client base? |
| 22:23.15 | durex | xeet2 ok |
| 22:23.58 | harryvv | Also, is there licencing requirments for these phones? anything that these phones need to work with asterisk |
| 22:24.04 | xeet2 | harryvv: yes. mostly real estate related services, but not on purpose, just word of mouth... everyone goes into their offices and sees their nice cisco ip phones and wants a phone like it |
| 22:24.36 | xeet2 | that and the 7960's have been appearing on tv shows and commercials alot lately |
| 22:24.51 | harryvv | haha thats good. Realestate here is hot though. |
| 22:25.12 | xeet2 | which is good, they all need a phone system that can do anything you tell it to |
| 22:26.10 | xeet2 | just get them to sign a contract for 1-2 years of support and maintenance... our usual rate is 20/mo/phone |
| 22:26.44 | xeet2 | durex: what version of mysql are you using? did you recently upgrade? |
| 22:27.09 | harryvv | so you charge them 20 dollars per phone then. Do you have them go though a voip carrier you own ? or though another service? |
| 22:27.09 | durex | su-2.05b# pkg_info | grep mysql |
| 22:27.12 | durex | mysql-client-4.0.24_1 Multithreaded SQL database (client) |
| 22:27.12 | durex | mysql-server-4.0.24_1 Multithreaded SQL database (server) |
| 22:27.12 | durex | p5-DBD-mysql-2.9007 MySQL driver for the Perl5 Database Interface (DBI) |
| 22:27.12 | durex | php4-mysql-4.3.11 The mysql shared extension for php |
| 22:27.19 | Nukemizer | Can anyone tell me how I might be able to test my TDM card ? I have a TDM11B and my FXO card stopped answering calls without providing any error messages. my Fax line was ringing to that port |
| 22:27.22 | durex | 4.0.24_1 |
| 22:27.40 | xeet2 | harryvv: did you recently upgrade? |
| 22:27.43 | xeet2 | er, durex |
| 22:27.49 | harryvv | upgrade to what |
| 22:27.56 | xeet2 | sorry, that was to durex |
| 22:28.01 | harryvv | k |
| 22:28.16 | Sedorox | If I buy a cisco from say... voipsupply.com... do I have to worry about firmware to get SIP.. or will they have that already? |
| 22:28.23 | durex | xeet2 yes, something about 2 weeks |
| 22:29.37 | xeet2 | sedorox: most of the time you will get a phone that does not have sip code |
| 22:29.55 | xeet2 | you have to buy the sip code/license from cisco |
| 22:30.05 | Sedorox | how much does it normally run? |
| 22:30.18 | ManxPower | Sedorox: $80 is the lowest I've seen for Cisco SIP |
| 22:30.24 | Sedorox | hmmm |
| 22:30.43 | ManxPower | That and lack of an included power supply is why we don't use Cisco |
| 22:30.48 | xeet2 | sedorox: alot cheaper with more than one license |
| 22:30.51 | Sedorox | I wonder if voipsupply pre-loads it tho.. because they advirtise SIP and asterisk tested.... |
| 22:31.11 | xeet2 | manx: yeah, thats a pain. we've had to put in poe switches on ever install |
| 22:31.12 | Sedorox | hehe, well they have a deal right now for a 7960 and a power block |
| 22:31.17 | ManxPower | they prolly do, or bundle the two togather |
| 22:31.22 | Sedorox | I'm debating between a IP600 and a 7960 |
| 22:31.33 | Sedorox | kk |
| 22:31.40 | harryvv | whats a power block |
| 22:31.41 | jonathh1 | hey guys |
| 22:31.47 | jonathh1 | i am trying to get my x100p working |
| 22:31.50 | *** join/#asterisk Weezey (~Weezey@206.210.109.226) |
| 22:31.53 | durex | xeet2 could have a problem if I just upgraded MySQL? |
| 22:32.03 | Sedorox | the power adapter for the phones |
| 22:32.07 | jonathh1 | i have downloaded nad installed Zaptel on asterisk site |
| 22:32.20 | jonathh1 | and have the card modprovbed without complaints |
| 22:32.25 | jonathh1 | bit the zfcfg -v |
| 22:32.36 | jonathh1 | reveals 0 deiveces |
| 22:32.46 | Sedorox | do you have zaptel.conf configured? |
| 22:32.50 | shmaltz | do I now have to log in to browse bugs.digium.com ? |
| 22:32.51 | jonathh1 | some online helpfiles refere to vi zapata.conf ? |
| 22:33.06 | Sedorox | thats on asterisk side... |
| 22:33.10 | Sedorox | check /etc/zaptel.conf |
| 22:33.16 | jonathh1 | i have zaptel.conf configuired in the sense i have changed the stuff to uk |
| 22:33.20 | Sedorox | you have to set that up too for it to get detexted |
| 22:33.28 | Sedorox | hmmm |
| 22:33.42 | jonathh1 | loadzone=uk |
| 22:33.45 | jonathh1 | defaultzone=uk |
| 22:33.50 | jonathh1 | only lines uncommented |
| 22:34.01 | *** join/#asterisk bajanman2 (~william@cp209-202-78-204.cp.telus.net) |
| 22:34.08 | Sedorox | fxsks=1 |
| 22:34.10 | Sedorox | is needed too |
| 22:34.16 | bajanman2 | has anyone did a firmware upgrade on a spa-2100? |
| 22:34.28 | ManxPower | bajanman2: Yes. I followed the directions. |
| 22:34.49 | bajanman2 | hmmm where can I get instuctions? |
| 22:35.00 | shmaltz | do I now have to log in to browse bugs.digium.com ? |
| 22:35.17 | ManxPower | bajanman2: http://www.sipura.com/support/index.htm |
| 22:35.21 | bajanman2 | I'm not a provider or anything, so I don't have access to their support area |
| 22:35.27 | jonathh1 | what order do i modprobe? zaptel then wsfxo? |
| 22:35.40 | Weezey | bajanman: scroll down. |
| 22:35.43 | _Vile | purple |
| 22:35.52 | ManxPower | bajanman2: anyone can access http://www.sipura.com/support/index.htm |
| 22:36.00 | bajanman2 | manxpower: I've been there, and downloaded the firmware.. maybe I didn't explain it. it won't allow me to upgrade |
| 22:36.08 | Weezey | jonathh1: I just do wsfxo and it forces zaptel |
| 22:36.30 | bajanman2 | spa2100-2.0.5(d).exe |
| 22:36.30 | Weezey | bajanman: sure it will, you just need your IP. |
| 22:36.38 | jonathh1 | ok ztcfg now see's one device |
| 22:36.38 | Weezey | it also helps to be on the same subnet |
| 22:36.46 | bajanman2 | I have the ip. |
| 22:37.27 | jonathh1 | how does asterisk know what it is called etc? |
| 22:37.27 | bajanman2 | it gives me the error: upgrade failed: Can't connect to spa |
| 22:37.52 | ManxPower | can you ping that ip? |
| 22:37.55 | bajanman2 | no |
| 22:37.57 | bajanman2 | ummm |
| 22:37.59 | bajanman2 | wait. |
| 22:38.04 | ManxPower | if you can't ping it then you can't upgrade it |
| 22:38.10 | Sedorox | jonathh1: your best bet... go read the wiki on how to do it/set it uo |
| 22:38.11 | Sedorox | up* |
| 22:38.16 | Sedorox | ~wiki |
| 22:38.18 | Weezey | is it blinking 3 fast, 2 slow, 3 fast? |
| 22:38.50 | bajanman2 | I can ping it |
| 22:38.53 | jonathh1 | does one create the zapata.conf themselves? |
| 22:38.58 | shmaltz | what do you guys think about what siparu is doing to cisco for the 2nd time in 2 years? |
| 22:39.12 | psywar | hey I know the dial plan controls placing calls, but what controls what happens when I receive a call and hit a key? anything? |
| 22:39.21 | bajanman2 | **** option 110# |
| 22:39.37 | Sato1 | jonathh1: its easy to do, check the sample itself |
| 22:39.46 | psywar | what kind of things can I do |
| 22:39.57 | jonathh1 | ok i have just seen an example comes with the asterisk source |
| 22:40.02 | bajanman2 | does it need to have the password for admin set to nothing? |
| 22:40.08 | psywar | I'd like to put people who put me on hold, on hold. |
| 22:40.29 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 22:40.37 | ManxPower | psywar: the answer to all your questions is "it depends on the phone" |
| 22:40.46 | psywar | SPA-2k |
| 22:40.49 | Weezey | How do I hide my callerID on outgoing PSTN? (how do I dial *67 (pause))? |
| 22:40.53 | *** join/#asterisk three55ml (~three55ml@cpe-24-243-30-75.satx.res.rr.com) |
| 22:41.01 | ManxPower | Weezey: analog port? |
| 22:41.02 | Sato1 | jonathh1, before editing zapata.conf, you have to edit /etc/zaptel.conf, there is where you declare the ports of your digium device |
| 22:41.10 | Weezey | Manx: yeah, Zap |
| 22:41.13 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:41.17 | three55ml | Anyone use MWI in RT with SIP? |
| 22:41.19 | Weezey | and Sipura |
| 22:41.34 | ManxPower | Weezey: You didn't see the "w" option on "show application dial"? |
| 22:42.02 | psywar | How do I figure out how to send a DTMF signal "9" when the pizza guy arrives? |
| 22:42.05 | psywar | I'm on a schedule here |
| 22:42.34 | psywar | I need to hit that to let himin the gate |
| 22:42.52 | psywar | This is on a SPA-2000 |
| 22:43.10 | bajanman2 | can someone point me to the support area (for non providers of sipura)? |
| 22:43.17 | psywar | What's blocking the 9? |
| 22:43.27 | psywar | the phone, or asterisk? |
| 22:43.35 | psywar | err.. s/phone/SPA/ |
| 22:43.51 | *** part/#asterisk Veryhot (~tho@adsl-69-109-159-210.dsl.sndg02.pacbell.net) |
| 22:44.04 | psywar | It seems intrusive for either to be blocking DTMF tones. |
| 22:45.38 | ManxPower | psywar: the phone has total control of everything in SIP. |
| 22:45.45 | ManxPower | Look at your SPA dialplan. |
| 22:46.22 | psywar | dial plan isn't for outbound calls only? |
| 22:46.38 | JunK-Y | psywar: huh? |
| 22:46.55 | psywar | JunK-Y: someone is calling me from the gate. I need to hit 9 to let them in. It doesn't work. |
| 22:47.05 | psywar | it hangs u |
| 22:47.06 | psywar | up |
| 22:47.15 | psywar | and |
| 22:47.17 | psywar | the pizza guy |
| 22:47.20 | psywar | is on his way here now |
| 22:47.26 | shmaltz | ManxPower, callpark was changed to call back the extension? (in CVS HEAD) |
| 22:47.38 | psywar | So, how do I let him in? |
| 22:47.59 | bajanman2 | manxpower: is there anything you had to change on the spa-2100 itself, before you upgraded? |
| 22:48.03 | JunK-Y | u send it a 9 ^ |
| 22:48.36 | Sato1 | how do i change the rpt range ports for iax protocol? |
| 22:48.45 | tld | I'm really struggeling to understand something. Could anyone help me with the last piece of what I'm not understanding? I'm trying to set up a call from a SIP extension (m197) to a number. Originate command (Manager API) and error message at: http://rafb.net/paste/results/cb2kjj64.html |
| 22:48.50 | psywar | WWhen I hit 9, it doesn't work. |
| 22:48.56 | tld | Would *really* appreciate a nudge in the right direction. |
| 22:48.56 | psywar | It hangs up instead. |
| 22:49.02 | psywar | How do I send it a 9? |
| 22:49.05 | psywar | If pressing 9 |
| 22:49.07 | psywar | doesn't work |
| 22:49.09 | psywar | ? |
| 22:49.19 | JunK-Y | where ya want to send a 9? |
| 22:50.15 | psywar | to the inbound cal |
| 22:50.16 | psywar | call |
| 22:50.25 | psywar | that the pizza guy makes from my apartment gate |
| 22:51.10 | psywar | where is this dial plan you speak of? |
| 22:52.06 | JunK-Y | which output do ya've ? |
| 22:52.20 | psywar | zap |
| 22:52.33 | jonathh1 | guys i cant get asterisk to recognise my x100p |
| 22:52.41 | jonathh1 | any suggestions? |
| 22:53.22 | JunK-Y | huh? your gate is connected to * ? |
| 22:53.34 | psywar | my gate places a regular analog telephone call |
| 22:53.39 | psywar | to my asterisk machine |
| 22:53.42 | Sato1 | jonathh1, see /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 22:53.46 | psywar | and it rings my SPA-200 phone |
| 22:53.51 | psywar | but when I hit 9, |
| 22:53.59 | psywar | something fucks up and doesn't transmit it to the gate |
| 22:54.05 | PatrickDK | pressing 9, has to be enabled on your spa-2000 |
| 22:54.18 | psywar | how do I do that? |
| 22:54.23 | PatrickDK | sipura doesn't do single digits by default |
| 22:54.43 | JunK-Y | psywar: not working with sipura. |
| 22:54.52 | PatrickDK | psywar, sipura manual |
| 22:55.02 | jonathh1 | ztcfg see's the card |
| 22:55.10 | jonathh1 | but show channels is asterisk doesn't |
| 22:55.11 | psywar | I got like a three-page pamphlet, wouldnt' call it a manual really |
| 22:55.11 | jonathh1 | languages=en |
| 22:55.11 | jonathh1 | context=inbound-analog |
| 22:55.12 | jonathh1 | signalling=fxs_ks |
| 22:55.19 | jonathh1 | is what i have in my zapata.conf |
| 22:55.24 | psywar | what if I need to hit arbitrary digits? |
| 22:55.25 | JunK-Y | its language, no s |
| 22:55.28 | alegh | anyone with experience with sangoma cards with asterisk? |
| 22:55.38 | psywar | SIP can't do that? |
| 22:55.41 | psywar | that's fucking stupid |
| 22:56.19 | JunK-Y | cant do what? |
| 22:56.21 | PatrickDK | sip do what? |
| 22:56.32 | PatrickDK | psywar, the manual is on their website |
| 22:56.42 | JunK-Y | PatrickDK: i think he needs more reading. |
| 22:56.49 | Jas_Williams | jonathh1, what error do you currently gwt ? |
| 22:56.50 | PatrickDK | they have it setup to only alow *?? #,*, 7 digits, 10+ digits |
| 22:56.58 | PatrickDK | more help, I will not give |
| 22:56.58 | Jas_Williams | get ? |
| 22:57.04 | jonathh1 | none |
| 22:57.40 | PatrickDK | actually, that isn't the issue |
| 22:57.42 | Jas_Williams | jonathh1, what does ztcfg -vvv give |
| 22:57.44 | PatrickDK | cause your already in a call |
| 22:57.45 | psywar | I've seen that dial plan somewhere before but can't find it now. |
| 22:57.49 | jonathh1 | well when i try and dial wit hthe device i get 'no channel registered for zap' |
| 22:57.53 | jonathh1 | pls hold |
| 22:57.59 | PatrickDK | psywar, you have tones working correctly? |
| 22:58.14 | shido6 | do you have the kernel drivers loaded ( ztcfg -vv) or lsmod |
| 22:58.14 | PatrickDK | dtmfmode= |
| 22:58.15 | psywar | I can access my pager's voice mail menus just fine |
| 22:58.19 | jonathh1 | ztcfg -vvv reveals.. |
| 22:58.20 | jonathh1 | Zaptel Configuration |
| 22:58.20 | jonathh1 | ====================== |
| 22:58.20 | jonathh1 | Channel map: |
| 22:58.20 | jonathh1 | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 22:58.20 | jonathh1 | 1 channels configured. |
| 22:58.31 | psywar | and this is important |
| 22:58.36 | psywar | b/c the piuzza guy is due any minute |
| 22:58.38 | shido6 | jonathh1, use http://pastebin.ca from now on for posts like that |
| 22:58.49 | jonathh1 | ok |
| 22:58.52 | PatrickDK | psywar, did you post your config on pastebin? |
| 22:58.53 | jonathh1 | soz guys |
| 22:59.00 | Jas_Williams | jonathh1, great, next post your zapata.conf to patebin.ca |
| 22:59.15 | psywar | and everyone is playing stupid |
| 22:59.19 | psywar | which annoys me greatly |
| 22:59.27 | PatrickDK | psywar, help you asked |
| 22:59.35 | psywar | PatrickDK: I thought it was a sipura problem, not a * problem |
| 22:59.36 | PatrickDK | we can only help you as much as you help explain the problem |
| 22:59.41 | PatrickDK | and I said PASTE THE CONFIG |
| 22:59.44 | PatrickDK | and you haven't |
| 22:59.52 | psywar | I could |
| 22:59.55 | psywar | but |
| 23:00.02 | psywar | eveyroen says it's the dial plan in sipura |
| 23:00.06 | key2 | what do u put to cancel the ECHO on asterisk |
| 23:00.06 | JunK-Y | psywar: wtf with pizza guy? |
| 23:00.07 | jonathh1 | ok zaptata |
| 23:00.07 | jonathh1 | http://pastebin.ca/10952 |
| 23:00.07 | key2 | ? |
| 23:00.10 | PatrickDK | only wen you make a call |
| 23:00.15 | psywar | JunK-Y: are you hard of thinking? |
| 23:00.21 | PatrickDK | ifyour already in a call, it's not the dialplan in sipura |
| 23:00.23 | Jas_Williams | jonathh1, Here's mine http://pastebin.ca/10951 |
| 23:00.24 | JunK-Y | go down and unlock the door? |
| 23:00.38 | psywar | HE IS GOING TO ARRIVE AT THE APARTMENT COMPLEX |
| 23:00.43 | psywar | AND HE HAS TO CALL ME TO GET IN |
| 23:00.44 | *** join/#asterisk jeffik (~jeffik@69.158.24.142) |
| 23:00.47 | JunK-Y | psywar: apparently, less then you i guess. |
| 23:00.47 | psywar | AND I HAVE TO HIT 9 |
| 23:00.50 | jonathh1 | thanks.. where in the world are you? is it region specfic? |
| 23:00.58 | jeffik | hello all |
| 23:00.59 | Jas_Williams | UK BT line :) |
| 23:01.01 | *** part/#asterisk kisu (~Snake@218.237.126.163) |
| 23:01.03 | psywar | HOW HARD IS THAT TO UNDERSTAND? |
| 23:01.07 | jonathh1 | EXCELLENT |
| 23:01.14 | jonathh1 | i'll just copy yours then ;) |
| 23:01.25 | PatrickDK | psywar, I refuse to repeat myself again |
| 23:01.34 | key2 | how do u set up the echo cancelation with asterisk ??? |
| 23:01.39 | psywar | I guess I'll go read the sipura manual |
| 23:01.49 | JunK-Y | and ya cant go down and unlock that damn door without PRESSING 9 ? |
| 23:02.02 | Jas_Williams | I have some patches installed for uk caller id so you need to comment out usecallerid = yes ; we want Caller*ID support |
| 23:02.02 | Jas_Williams | cidsignalling = v23 ; UK (BT) Caller*ID uses the V.23 std |
| 23:02.02 | Jas_Williams | cidstart = history |
| 23:02.12 | psywar | I'm tired of explaining myself too PatrickDK |
| 23:02.18 | psywar | I've only done it 3 times at least. |
| 23:03.03 | tld | I'm trying to place a call using the Manager API, but it always fails with "May 4 19:05:21 NOTICE[1180]: channel.c:1852 __ast_request_and_dial: Unable to request channel SIP/m197-d876". I'm using the same context as the SIP extension normally has, and I can call the same extension I'm trying to call manually on the phone... Any ideas? |
| 23:03.17 | tld | I've tried googling for about an hour, but I'm not getting anywhere. |
| 23:03.26 | psywar | wow, that 2 page "data sheet" didn't help at all |
| 23:03.48 | psywar | where is this manual I'm supposed to read? |
| 23:03.52 | Jas_Williams | jonathh1, use this version until you get uk callerid patches installed http://pastebin.ca/10953 |
| 23:03.59 | key2 | could someone tell me how to set up the echo cancelation with asterisk ??? |
| 23:04.42 | JunK-Y | key2: echocancel=yes ? |
| 23:04.47 | shido6 | zapata.conf, key2 |
| 23:04.51 | psywar | ah http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf |
| 23:05.05 | jonathh1 | yep just commented out.. restarting asterisk |
| 23:05.06 | key2 | JunK-Y: does it work for SIP |
| 23:05.07 | key2 | ? |
| 23:05.10 | jonathh1 | wish me luck :) |
| 23:05.16 | *** join/#asterisk kioko (~kiokorobe@196.200.26.42) |
| 23:05.45 | jonathh1 | hmm should 'show channels' show something? |
| 23:06.13 | xeet2 | key2: echo cancellation should be done at the point where tdm meets ip, a pure ip call shouldn't have any echo problems |
| 23:06.14 | Jas_Williams | zap show channels will |
| 23:06.22 | Jas_Williams | show channels will not |
| 23:07.03 | Jas_Williams | jonathh1, do you want to test outbound calls first or inbound ? |
| 23:07.12 | jonathh1 | it doesn't seem to like the command zap show channels |
| 23:07.17 | jonathh1 | outboun |
| 23:07.17 | jonathh1 | d |
| 23:07.25 | jonathh1 | i have the entry in extensions.conf |
| 23:07.41 | jonathh1 | exten => _0X.,1,Dial,Zap/1/${EXTEN:1} |
| 23:08.19 | key2 | how do I rehash with asterisk, once I changed a .conf, do I have to restart asterisk |
| 23:08.19 | key2 | ? |
| 23:08.20 | shido6 | unless you're speakerphone sux |
| 23:08.24 | shido6 | or your mic sux |
| 23:08.26 | Sato1 | you are missing a point after the X |
| 23:08.30 | xeet2 | shido6: true |
| 23:08.32 | Sato1 | jonathh1 |
| 23:08.37 | Jas_Williams | jonathh1, can you stop asterisk then do an asterisk -vvvc and post the full output to pastebin.ca it sounds like your zaptel module is not loading |
| 23:08.50 | shido6 | key2 you dont have to restart asterisk unless you're screwing with zapata.conf |
| 23:08.50 | Sato1 | oops, sorry, didnt see it, font too small |
| 23:08.51 | JunK-Y | key2: reload |
| 23:08.55 | shido6 | otherwise do a reload |
| 23:09.01 | shido6 | JunK-Y's on it.. |
| 23:09.02 | shido6 | :) |
| 23:09.27 | alegh | anyone who tested dtmf callerid with x100p? |
| 23:09.38 | jonathh1 | pls hold |
| 23:09.42 | JunK-Y | can i ask whats dtmf callerid? |
| 23:09.49 | key2 | so how comes when I use 2 SIP phone, I have a big echo ? |
| 23:09.50 | JunK-Y | whats diff from callerid standard? |
| 23:10.01 | JunK-Y | key2: what kind of sip phone? |
| 23:10.13 | xeet2 | junky: different countries/regions do caller id in different ways |
| 23:10.14 | alegh | Junk-Y: cid comes in dtmf before the first ring instead of fsk |
| 23:10.17 | key2 | sjphone |
| 23:10.42 | shido6 | thats the reason |
| 23:10.46 | shido6 | get a noise cancelling mic |
| 23:10.52 | JunK-Y | alegh: never touched that part, isnt the case in NA. |
| 23:10.59 | xeet2 | usually things are done one way in the us, and are done a better way elsewhere |
| 23:10.59 | jonathh1 | does the asterisk -vvvc dump to a file anywhere? |
| 23:11.06 | key2 | shido6: it comes from the mic ? |
| 23:11.08 | shido6 | -vvvgc does |
| 23:11.13 | Jas_Williams | jonathh1, no |
| 23:11.13 | shido6 | when it dies |
| 23:11.28 | _kb1_kanobe | jonathh1: see /etc/asterisk/logger.conf |
| 23:11.29 | alegh | junky: I found information in bug 9 and 1719 but I could not make it work reliable |
| 23:11.46 | Jas_Williams | jonathh1, why not asterisk -vvvvc | tee file.txt |
| 23:12.00 | jonathh1 | ok, im all over it :) |
| 23:12.09 | JunK-Y | which bugs? from bugs.digium.com ? |
| 23:12.16 | alegh | yes |
| 23:12.44 | *** join/#asterisk outtolunc (~me@adsl-69-110-63-171.dsl.pltn13.pacbell.net) |
| 23:12.55 | alegh | now they are closed |
| 23:13.06 | JunK-Y | it makes a long time since bug 9 and 1719 was resolved. |
| 23:13.53 | JunK-Y | i cant help ya so much about it, cause i never touched that kind of stuff. |
| 23:14.13 | jonathh1 | Jas: http://pastebin.ca/10955 |
| 23:14.14 | alegh | yes, but I believe that there are so much countries differences and that issues were partially resolved |
| 23:14.49 | JunK-Y | alegh: im sure ya can find someone in UK which can help. |
| 23:15.00 | psywar | well apprently they have removed the dial plan as a settable parameter in the newest SPA-2000 firmware, as it's not on any of the web pages it generates. |
| 23:15.04 | alegh | and I don't feel with enough knowledge to tru to continue that work |
| 23:15.10 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 23:15.14 | *** join/#asterisk santiago (~santiago@63.245.86.187) |
| 23:16.19 | alegh | junky: I believe that cid used in UK is very similar to what I need but the patches did not work for me. Maybe timings or something like that |
| 23:19.56 | jonathh1 | any comments Jas_Williams |
| 23:20.11 | Jas_Williams | no zaptel loading ? |
| 23:20.24 | Jas_Williams | do you have this file ? /usr/lib/asterisk/modules/chan_zap.so |
| 23:21.20 | alegh | I have another question: Does anybody used sangoma cards? |
| 23:21.45 | jonathh1 | firstly my path for that would be usr/local/asterisk/usr/lib/asterisk/modules/ |
| 23:22.04 | jonathh1 | and no |
| 23:22.09 | jonathh1 | there is no chan_zap |
| 23:22.20 | Jas_Williams | that is your problem then |
| 23:22.36 | jonathh1 | does the zaptel stuff need to be present when asterisk is compiled? |
| 23:23.07 | Jas_Williams | you need to re make asterisk yes you should compile zaptel and libpri before asterisk |
| 23:23.21 | jonathh1 | ok |
| 23:23.22 | Jas_Williams | make clean make install |
| 23:23.24 | jonathh1 | i'll get right on that |
| 23:23.43 | jonathh1 | do i need to tell it where zaptel is? |
| 23:24.12 | Jas_Williams | no it will find it in your kernel |
| 23:24.21 | tzafrir_laptop | jonathh1, the makefile try to detect on its own. if it has failed, you'll need to edit it |
| 23:24.54 | jonathh1 | ok. just me being the awkward sod that i am... installed aasterisk in a none standard place |
| 23:25.06 | jonathh1 | i have sym links to everything that i can see.. needs one |
| 23:25.08 | tzafrir_laptop | Jas_Williams, zaptel can be easily built outside the kernel tree |
| 23:25.54 | jonathh1 | anyway i can tell after doing a make.. if it found zaptel? |
| 23:25.54 | tzafrir_laptop | jonathh1, so look at the makefile and figure out why it won't build chan_zap |
| 23:26.01 | jonathh1 | ok |
| 23:26.23 | jonathh1 | the first time round.. zaptel wasn't there |
| 23:26.28 | jonathh1 | plain and simple |
| 23:27.05 | durex | shit... I'm having a delay of 2.300 ms to my next-hope |
| 23:27.08 | *** join/#asterisk Jas_Williams (~jas_willi@217.41.232.141) |
| 23:27.11 | durex | @¨%&¨%@ telecom |
| 23:27.39 | Jas_Williams | jonathh1, opps just disconnected how is it going ? |
| 23:28.26 | jonathh1 | i am recompiling asterisk |
| 23:28.42 | jonathh1 | i ahve vchecked the make file.. it is looking in the right places for zaptel.h |
| 23:28.48 | Jas_Williams | k |
| 23:28.56 | jonathh1 | dunno how long this will take.. tis athlon700 |
| 23:29.15 | jonathh1 | i think the bacteria living on the dust add 1 or 2 MHz's |
| 23:30.48 | jonathh1 | installing.. |
| 23:31.00 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 23:31.17 | Jas_Williams | check for the chan_zap.so |
| 23:31.25 | jonathh1 | roger |
| 23:31.31 | xeet2 | durex: 2.3ms or 2300ms? |
| 23:31.51 | jonathh1 | chan_zap.so |
| 23:31.55 | jonathh1 | present and accounted for |
| 23:32.10 | jonathh1 | lets fire this puppy up |
| 23:32.22 | Jas_Williams | ok your dial command did not look correct |
| 23:32.46 | jonathh1 | zap show channels has out put |
| 23:33.00 | Jas_Williams | remove the :1 after the extension unless you prefix all numbers with a 0 eg 00207.... |
| 23:33.17 | key2 | is a RJ45 SIP phone == USB SIP phone / |
| 23:33.18 | key2 | ? |
| 23:33.20 | Jas_Williams | great |
| 23:33.25 | key2 | or not same quality |
| 23:33.43 | jonathh1 | i was.. |
| 23:33.48 | Jas_Williams | k |
| 23:34.06 | Jas_Williams | make a test call then :) |
| 23:34.12 | jonathh1 | yay :) |
| 23:34.14 | jonathh1 | it worked! |
| 23:34.30 | xeet2 | key2: there is no single answer to your question. a comparison between two specific models would be better |
| 23:34.51 | mDuff | key2, a USB SIP phone doesn't really do SIP, generally; rather they tend to offload that to the computer they're connected to. |
| 23:35.06 | jonathh1 | now i want to process incoming |
| 23:35.10 | jonathh1 | any pointers? |
| 23:35.23 | Jas_Williams | now to fix your inbound my config specified an incomming context of from-pstn you need to create an s extension in there |
| 23:35.29 | key2 | mDuff: ok that's what I wanted to know |
| 23:35.40 | mDuff | key2, so typically they're just a pair of USB Audio devices coupled with a keypad, a LCD, an on-hook/off-hook detector, etc all controllable by USB. |
| 23:35.46 | Jas_Williams | eg [from-pstn] |
| 23:35.56 | jonathh1 | ah ha |
| 23:35.58 | jonathh1 | hang on |
| 23:36.16 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 23:36.16 | Jas_Williams | extne => s,1,dial(SIP/MYPHONE) |
| 23:36.31 | xeet2 | actually some new usb phone-like devices just use the power and network connectivity via usb but still do everything else on their own |
| 23:37.19 | mDuff | ("working on" == "one of the coworkers has been using it for the last week or so, but development is stalled 'till more hardware shows up in the mail") |
| 23:38.39 | jonathh1 | dude it works |
| 23:38.42 | jonathh1 | but for some reason |
| 23:38.47 | jonathh1 | it rings like 4 times |
| 23:38.50 | jonathh1 | before it answers |
| 23:39.24 | Jas_Williams | jonathh1, you need to add usecallerid=no to zapata.conf |
| 23:39.43 | Jas_Williams | until you patch for uk callerid |
| 23:39.51 | jonathh1 | well ti is commented out currently.. |
| 23:40.05 | Jas_Williams | is uses yes by default |
| 23:40.16 | jonathh1 | done |
| 23:40.18 | jonathh1 | lets try |
| 23:40.20 | Jas_Williams | reload |
| 23:41.20 | jonathh1 | getting May 5 00:41:05 NOTICE[22636]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... |
| 23:41.30 | jonathh1 | 4 times before my SIP/grandstream kicks in |
| 23:42.08 | Jas_Williams | jonathh1, have you restarted asterisk after the usecallerid change |
| 23:42.16 | jonathh1 | well i reloaded |
| 23:42.18 | jonathh1 | what does |
| 23:42.24 | jonathh1 | immediate=no |
| 23:42.25 | jonathh1 | mean |
| 23:42.50 | *** join/#asterisk scubasteve (~steve@cpe-066-026-046-129.nc.res.rr.com) |
| 23:43.18 | key2 | is this one OK: http://cgi.ebay.fr/ws/eBayISAPI.dll?ViewItem&category=61840&item=5771382952&rd=1 |
| 23:43.25 | Jas_Williams | jonathh1, you need to restart, |
| 23:43.28 | scubasteve | Anyone in here ever have anything bizarre happen with SixTel? I found one of my DID's I have through them is now routed to someone else. |
| 23:44.03 | Jas_Williams | immediate=no means do not pick up the line until told to do so |
| 23:44.19 | jonathh1 | ok |
| 23:44.25 | jonathh1 | now it dont work :) |
| 23:45.10 | jonathh1 | unable to create channel zap |
| 23:45.39 | Moc | scubasteve: I've lost 6 DID with some provider.. they all use broadvox and they seem to cut DID from reseller and bring them back to them.. |
| 23:45.49 | Jas_Williams | sounds like a typo in zapata.conf |
| 23:46.15 | Jas_Williams | check asterisk -vvvc it may give a clue as it tries to load chan_zap |
| 23:47.19 | shido6 | anyone else want burger slin? |
| 23:49.38 | jonathh1 | May 5 00:49:08 WARNING[28093]: chan_zap.c:9615 setup_zap: Ignoring ohannel |
| 23:49.39 | tzanger | burger slin? |
| 23:49.41 | jonathh1 | is all it says |
| 23:49.43 | tzanger | is that a new voip joint? |
| 23:49.56 | jonathh1 | got i |
| 23:49.57 | jonathh1 | t |
| 23:50.04 | scubasteve | Moc: Yeesh!! |
| 23:50.05 | jonathh1 | ( i think ) |
| 23:50.21 | *** join/#asterisk salviadud (~dude@201.129.86.120) |
| 23:50.25 | scubasteve | Moc: This DID is worth fighting for. I've got it on letterhead, business cards and corporate checks. |
| 23:50.37 | scubasteve | Moc: And it's also a very spiffy sounding number. |
| 23:50.41 | tzanger | scubasteve: ugh |
| 23:50.49 | Moc | my sixtel 1800 are all dead too |
| 23:51.04 | scubasteve | There will at the very least be a lot of visible noise about this if I don't get it back. |
| 23:51.07 | salviadud | anyone here have a sipura 2000 or 3000? |
| 23:51.19 | mDuff | salviadud, I have a few 2100s. |
| 23:51.20 | jonathh1 | Jas_Williams you are a star |
| 23:51.25 | jonathh1 | there is now little delay |
| 23:51.33 | Jas_Williams | Thats how it should be |
| 23:51.42 | salviadud | mduff im trying to get incoming calls on line 1 from FWD |
| 23:51.46 | Jas_Williams | Time for sleep now |
| 23:51.54 | jonathh1 | cheers dude |
| 23:52.01 | Jas_Williams | no problem |
| 23:52.34 | salviadud | does it matter if im on public ip? |
| 23:53.05 | salviadud | do i have to put a yes on the answer call without reg? |
| 23:53.41 | salviadud | do you have a specified sip port? |
| 23:53.46 | mDuff | salviadud, they have NAT tunneling support (via STUN), but I have no familiarity whatsoever with FWD. |
| 23:53.47 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 23:54.14 | salviadud | how you do you get incoming calls? |
| 23:54.23 | salviadud | ip dialing? |
| 23:54.54 | jonathh1 | thanks for all the help tonight guys |
| 23:55.17 | mDuff | salviadud, PRI -> * -> SIP phones |
| 23:55.51 | salviadud | ohhh |
| 23:55.59 | salviadud | that asterisk... its a beauty. |
| 23:56.14 | salviadud | still, i can't get into asterisk if i can't configure my sip |
| 23:56.25 | salviadud | thanx anyway, im gonna try the stun method |