| 00:00.24 | FastFeet | yes, I can even register |
| 00:00.33 | ManxPower | FastFeet: Putzz is using bad terms -- maybe he was brain damaged from using Trixbox or something |
| 00:00.34 | FastFeet | just happens when I try to place a call |
| 00:00.41 | FastFeet | lol |
| 00:01.16 | FastFeet | Could it be a firewall issue? |
| 00:01.29 | FastFeet | I am using IAX protocol with callwithus.com |
| 00:01.43 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-28-58.hsd1.pa.comcast.net) |
| 00:02.01 | *** join/#asterisk darkmug (n=Dennis@201.82.82.94) |
| 00:02.31 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
| 00:03.14 | ManxPower | FastFeet: it can always be a firewall issue |
| 00:03.21 | FastFeet | true.. |
| 00:03.46 | ManxPower | port 4569/UDP is what iax2 uses |
| 00:04.05 | FastFeet | I feel that it the case, as the same dial plan works here at home, but when I use it at work not so good. |
| 00:04.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:04.50 | FastFeet | Yes, I will have the administrator(s) open the UDP port 4569, I should also get them to open 5060 and 5061, while I am at it |
| 00:04.52 | FastFeet | thanks. |
| 00:06.42 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:08.27 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 00:08.28 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.203.59) |
| 00:09.44 | tuan_modulis | oooof, anyone here had problems with RxFax restarting asterisk (with segmentation fault in safe_asterisk)? |
| 00:10.17 | tuan_modulis | I made it work on a virtualised server (both with QEMU and vserver)... but dang production server doesnt want to play nice |
| 00:12.29 | tuan_modulis | now im musing with the idea of having the production server run one virtual server... totally inefficient |
| 00:20.39 | *** join/#asterisk gammah (n=gammah@cpe-70-113-92-93.austin.res.rr.com) |
| 00:29.52 | tuan_modulis | hmmm, well I guess i can upgrade the kernel |
| 00:33.22 | *** join/#asterisk SophWise (n=SophWise@72.169.74.29) |
| 00:34.17 | *** join/#asterisk hydan (n=Others@c-71-207-247-195.hsd1.al.comcast.net) |
| 00:35.22 | SophWise | Anyone know if there is some means of an email to fax gateway that can send faxes out an outbound route? |
| 00:35.33 | *** join/#asterisk jpon (n=jpniane@neu67-3-82-239-80-181.fbx.proxad.net) |
| 00:41.35 | *** part/#asterisk hydan (n=Others@c-71-207-247-195.hsd1.al.comcast.net) |
| 00:43.32 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 00:44.17 | SophWise | Anyone know if AsterFax will work with a default install of SpanDSP? |
| 00:44.46 | SophWise | and how can I tell if txfax and rxfax modules are installed for SpanDSP? |
| 00:45.17 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 00:53.40 | ManxPower | show application rxfax |
| 00:53.54 | SophWise | cool thanks |
| 00:54.46 | SophWise | so ifit returns Your application(s) is (are) not registered |
| 00:54.52 | SophWise | for both txfax and rxfax, |
| 00:54.55 | SophWise | I'm hosed right? |
| 01:09.17 | mocker | Getting flooded wtih Got SUBSCRIBE but there is no hint messages, even after I added hints to the extensions.conf. |
| 01:10.04 | mocker | Anyone have any suggestions? |
| 01:12.31 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:14.25 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 01:16.15 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:16.28 | DocHolliday | anyone here have experience with flashing cisco 7940/60G? |
| 01:16.38 | [TK]D-Fender | mocker, stop your users from subscribing to things they shouldn't be |
| 01:18.29 | mocker | [TK]D-Fender: These are Polycom phones, they have presense information that should be transmitted.. |
| 01:18.38 | DocHolliday | anyone? |
| 01:19.37 | [TK]D-Fender | mocker, odds are your users are adding 5,000,000 directory entries to lord knows where and jsut shoving "buddy watch" by default without thinking |
| 01:22.16 | mocker | [TK]D-Fender: It's weird, I'm migrating this off of a trixbox setup to a straight asterisk box.. |
| 01:22.24 | mocker | On trixbox it doesn't do this at all. |
| 01:22.37 | DocHolliday | nobody here has experience with cisco 7940/7960? |
| 01:23.12 | [TK]D-Fender | mocker, your phones don't behave any different because of trixbox building your configs. |
| 01:23.20 | mocker | right. |
| 01:23.32 | mocker | But 'show hints' on both systems matches up.. |
| 01:23.40 | mocker | So I must be missing something on the config side. |
| 01:23.46 | [TK]D-Fender | mocker, So if you've screwed up your hints, go fix them. if they are subscribing to things they shouldn't be, stop them |
| 01:23.53 | mocker | Or they've compiled it to squelch those messages.. |
| 01:23.59 | [TK]D-Fender | mocker, probably not int he right PLACE. |
| 01:24.27 | mocker | The hints? |
| 01:24.49 | [TK]D-Fender | yes |
| 01:24.58 | mocker | Actually.. |
| 01:25.10 | mocker | Does #include automatically do a standard include? |
| 01:25.39 | mocker | I bet that's my problem. |
| 01:26.37 | [TK]D-Fender | yes.. the idea that anything "standard" happens. |
| 01:26.43 | [TK]D-Fender | time to think EXPLICIT |
| 01:26.57 | LeddyHM | oh baby |
| 01:27.05 | LeddyHM | show me the explicit! |
| 01:28.39 | mocker | guh. |
| 01:28.50 | mocker | I've dumped the hints in multiple contexts now to no avail. |
| 01:29.37 | [TK]D-Fender | LeddyHM, I've told you before...... I wasn't hired for THOSE services ;) |
| 01:30.03 | *** join/#asterisk raptorrat (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 01:30.11 | [TK]D-Fender | mocker, every heard of www.pastebin.ca ? Now's a good time to familiarize yourself ;) |
| 01:32.44 | raptorrat | rtfm question can an ael dial plan have variables on the left hand side of => ? So instead of _9XX => ... I could have ${SOME_VAR} => on the left hand side? |
| 01:34.07 | [TK]D-Fender | raptorrat, No. |
| 01:34.25 | mocker | [TK]D-Fender: Actually I think I may have got it. |
| 01:34.27 | [TK]D-Fender | raptorrat, You can have a CONSTANT as defined in [globals] but thats it |
| 01:34.44 | mocker | Dumping it to multiple contexts worked, I just left out a couple.. |
| 01:34.51 | mocker | So the stream went from a flood to a trickle. |
| 01:35.09 | mocker | Thanks for your help, I wasn't aware that hints were context dependent. |
| 01:35.34 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 01:36.31 | SophWise | anyone know ifthere is an easy way to install rxfax and txfax? |
| 01:37.08 | [TK]D-Fender | mocker, indeed they are. think if you were running a multi-company PBX and one division could spy on the other's status, just because everyone has hints... |
| 01:37.36 | mocker | [TK]D-Fender: I don't see a problem with that. :) |
| 01:37.47 | raptorrat | [TK]D-Fender: thanks, I'm digging on that now, but can the globals be dynamically defined from other apps/functions. In short I'm trying to define a generic dial plan htat can be used at several sites but it would dynamically pulls what pattern/number they use for things like local extensions so instead of having things like _4xxx => in site a's dial plan and _5xxx => at site b's dial plan they would have the same dial plan and |
| 01:38.53 | [TK]D-Fender | raptorrat, No, not variables. Not GLOABL variables. CONSTANTS. that part of the extensions parsing is completely static |
| 01:40.39 | raptorrat | so there isn't a real way to have the lh side be looked up in some dynamic fashion with out writting my own app to do it |
| 01:43.17 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
| 01:43.18 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:43.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:51.54 | mocker | Sweet, presense is working like a champ. |
| 01:52.07 | mocker | I didn't want to have angry calls from the secretary tomorrow morning. :) |
| 01:52.56 | SophWise | Anyone know if rxfax and txfax can be installed without recompiling spandsp/asterisk? Any easy way? |
| 01:53.53 | mocker | SophWise: They require spandsp to be installed. |
| 01:54.19 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 01:54.21 | SophWise | yeah,spandsp was compiled and installed when I installed |
| 01:54.25 | *** part/#asterisk fysa (i=fysa@653414hfc175.tampabay.res.rr.com) |
| 01:54.31 | SophWise | astersik, but I can't for the life of me undersatnd |
| 01:54.35 | mocker | SophWise: But you can't get it to compile w/ Asterisk? |
| 01:54.54 | SophWise | well, not sure, I don't even know where to look to |
| 01:55.03 | SophWise | find the rx and tx files |
| 01:55.11 | SophWise | just reading the install faq on asterfax |
| 01:55.24 | mocker | You have to download those seperately. |
| 01:55.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:55.35 | SophWise | .c or .so? |
| 01:55.38 | mocker | .c |
| 01:55.41 | mocker | So you can compile them. |
| 01:55.52 | mocker | I think asteriskguru has a pretty good walkthrough. |
| 01:55.52 | SophWise | bummer, and I have to run a patch too right? |
| 01:55.55 | mocker | Yeah. |
| 01:56.14 | mocker | It actually bombed on my compile and I cheated and used precompiled from another box. |
| 01:56.17 | mocker | :) |
| 01:56.19 | SophWise | great, is it worth my time or should I just continue to pay Send2Fax $8 per month |
| 01:56.44 | mocker | Eh, it's nice to have it in Asterisk but I think it's kind of a hacky addon. |
| 01:57.01 | SophWise | so is there really a solid way to provide the |
| 01:57.07 | SophWise | send/receive fax functionality |
| 01:57.16 | SophWise | without farting around with experimental stuff? |
| 01:57.29 | mocker | I don't know if AsterFax uses rxfax/txfax, you might check into that.. |
| 01:57.41 | SophWise | yes, they do, that is the install I am reeading |
| 01:57.45 | mocker | It's for $ after your first line or some silly amount. |
| 01:57.53 | mocker | Oh, that's lame. |
| 01:57.55 | mocker | ;)( |
| 01:58.19 | groogs[h] | SophWise: get a real fax/modem, install hylafax |
| 01:58.22 | mocker | I can put up my precompiled versions for you to dl, but I don't make any guarentees.. |
| 01:58.23 | SophWise | well, what I am hoping to do is to add asterfax on to the PBX and then get an extra DID |
| 01:58.25 | SophWise | for the fax |
| 01:58.44 | SophWise | problem is this is in the datacenter |
| 01:58.44 | groogs[h] | fax+voip without t.38 is a crapshoot |
| 01:58.48 | SophWise | no lines available |
| 01:59.03 | SophWise | I have access to a T.38 DID |
| 01:59.27 | SophWise | well, Send2Fax works fine |
| 01:59.44 | SophWise | but its only 2.99 per month for another DID |
| 01:59.54 | SophWise | so I could be saving a measly 6 bucks |
| 02:00.08 | SophWise | if I just had my own service configged into our server |
| 02:00.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:00.26 | SophWise | but if its a PITA then I just pay Send2Fax and continue using :-) |
| 02:01.02 | mocker | SophWise: http://mocker.org/app_rxfax.so and app_txfax.so |
| 02:01.08 | mocker | But it might totally crash your asterisk box. |
| 02:01.12 | shmaltz | anybody here using Act? |
| 02:01.13 | mocker | :) |
| 02:01.20 | SophWise | ugh, no thanks then :-) |
| 02:01.23 | SophWise | what OS? |
| 02:01.27 | SophWise | version etc |
| 02:02.10 | SophWise | if you're on standard CentOS 4.4 then all should be fine right, LOL |
| 02:02.14 | groogs[h] | mocker: lol, yeah.. most likely it will not work |
| 02:02.19 | LeddyHM | mmm centos |
| 02:03.05 | groogs[h] | or worse, it may work until it makes a certain system call several days from now |
| 02:03.11 | SophWise | mocker: are you able to send/receive faxes np? |
| 02:03.17 | SophWise | LOL |
| 02:03.26 | SophWise | yeah, I think I am liking Send2Fax for now |
| 02:03.32 | SophWise | too much to add to my plate right now |
| 02:03.35 | SophWise | would be nice, |
| 02:03.45 | SophWise | but I don't need more headaches in my life |
| 02:03.53 | SophWise | enough to go around for now |
| 02:08.04 | LeddyHM | we use hylafax/iaxmodems |
| 02:08.38 | SophWise | if I make clean, make and make install asterisk, does it wipe my confs? |
| 02:08.53 | LeddyHM | make samples does |
| 02:09.17 | LeddyHM | a wiseman makes backups |
| 02:09.17 | SophWise | but make install just installs the binaries? |
| 02:09.39 | SophWise | yes, I could just backup the /etc/asterisk and restore it after the compile right? |
| 02:09.47 | SophWise | would maybe to a merge |
| 02:09.49 | CCFL_Man2 | the stun server, in a cisco ip phone with sip firmware, do i put the stun server in the NAT IP field? |
| 02:09.49 | SophWise | or something |
| 02:09.54 | LeddyHM | you don't need to restore |
| 02:10.05 | LeddyHM | backup for safe keeping just in case something happens |
| 02:10.09 | SophWise | ok, so no fear of having my config wiped |
| 02:10.11 | SophWise | yes of course |
| 02:10.12 | LeddyHM | a good admin knows the power of backups |
| 02:10.19 | SophWise | I do backups before anything major like that anyhow |
| 02:10.35 | SophWise | yes, I have the crontabbed to a tape |
| 02:10.36 | [TK]D-Fender | Jesus saves! |
| 02:10.58 | [TK]D-Fender | (Real admins use rsync for incremental live backups) |
| 02:11.05 | mog | Buddha does incramental backups |
| 02:11.09 | mog | damn beat me to it |
| 02:11.13 | [TK]D-Fender | :D |
| 02:11.15 | CCFL_Man2 | is stun required for sip? |
| 02:11.16 | SophWise | lol |
| 02:11.23 | [TK]D-Fender | CCFL_Man2, no |
| 02:11.35 | LeddyHM | real admins know backups are overrated |
| 02:11.35 | SophWise | just helps with firewalls |
| 02:11.37 | LeddyHM | ;) |
| 02:11.40 | SophWise | but not required |
| 02:13.12 | CCFL_Man2 | SophWise: with a cisco ip phone with sip firmware, do i put a stun server in the NAP IP field or doesn't it belong there? |
| 02:13.39 | SophWise | No a clue CCFL |
| 02:13.45 | CCFL_Man2 | hmm.. |
| 02:13.49 | SophWise | I mostly work with cheap phones and softphones |
| 02:13.56 | CCFL_Man2 | admin guide isn't really clear |
| 02:14.11 | CCFL_Man2 | well, sip firmware is limited |
| 02:14.12 | SophWise | well, you're behind a firewall? |
| 02:15.42 | CCFL_Man2 | no |
| 02:15.46 | CCFL_Man2 | just nat |
| 02:15.52 | CCFL_Man2 | The Cisco phone (at least with SIP v7.5) has no support for STUN and its support of DynDNS is pretty weak. |
| 02:16.19 | SophWise | hmm, yeah, I would imagine you would have to have support for STUN on the phone itself since its |
| 02:16.29 | SophWise | used as a service to help with NAT routing |
| 02:16.47 | SophWise | but I can't say for sure on that |
| 02:16.56 | SophWise | as I'm not an expert in such matters |
| 02:17.28 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 02:17.47 | CCFL_Man2 | the nat ip field says it's the wan ip so other sip user agents to communicate with the phone |
| 02:18.16 | SophWise | yes, and STUN would allow you to find out that IP |
| 02:18.27 | SophWise | but you wouldn't put the STUN server in that place |
| 02:18.46 | CCFL_Man2 | you put your own nat ip there? |
| 02:19.01 | SophWise | its the external internet facing address your ISP assigns to your router, assuming you're |
| 02:19.04 | SophWise | home based? |
| 02:19.18 | SophWise | areyou on a dynamic IP? |
| 02:19.27 | CCFL_Man2 | dynamic ip, yes |
| 02:19.55 | *** join/#asterisk ChkDigit (n=mrw@static24-72-71-175.regina.accesscomm.ca) |
| 02:19.57 | SophWise | bummer, if no support for STUN and dyndns, then not sure what other options exists short of reconfigging the phone everytime your |
| 02:19.59 | SophWise | ISP |
| 02:20.01 | SophWise | changes things |
| 02:20.11 | SophWise | does the cisco phone have any remote administration |
| 02:20.13 | SophWise | options? |
| 02:20.15 | CCFL_Man2 | i think thats what i'll do |
| 02:20.23 | CCFL_Man2 | oh yes |
| 02:20.33 | SophWise | maybe write a small application to get your main IP |
| 02:20.38 | SophWise | and update the phone periodically |
| 02:20.43 | SophWise | by remote |
| 02:21.08 | CCFL_Man2 | the service works fine without anything in the nat ip field |
| 02:21.34 | SophWise | I don't know Cisco phones so this is all pure conjecture on my part |
| 02:21.51 | SophWise | what model phone? |
| 02:22.02 | CCFL_Man2 | ahh |
| 02:22.03 | CCFL_Man2 | 7912 |
| 02:22.12 | CCFL_Man2 | sip 8.0.0 firmware |
| 02:22.29 | CCFL_Man2 | Software Version8.0.0(060111A) |
| 02:22.49 | SophWise | yeah, looks like those have zilch support for STUN |
| 02:23.35 | SophWise | is it a 7912G? |
| 02:23.40 | CCFL_Man2 | ahh, no big deal then |
| 02:23.43 | CCFL_Man2 | yes |
| 02:24.01 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 02:24.36 | SophWise | Yeah, looks like from googling it, that its pretty bad on STUN support |
| 02:25.04 | SophWise | and doesn't do any DNS resolution, so any DynDNS is out of the question |
| 02:25.44 | CCFL_Man2 | ahh |
| 02:25.54 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:26.01 | CCFL_Man2 | it does do dns resolution with the sip server though |
| 02:26.33 | SophWise | well, are you able to setup a local proxy server to handle all of the |
| 02:26.35 | SophWise | NAT fun? |
| 02:27.04 | CCFL_Man2 | the phone works fine without anything in the nat ip field |
| 02:27.15 | SophWise | So then what is the problem? |
| 02:27.23 | CCFL_Man2 | SophWise: i was using the 7912 with vonage, but i'm dumping vonage |
| 02:27.25 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:27.31 | DrukenLPY | evening everyone.... |
| 02:27.36 | SophWise | yes, good move |
| 02:27.38 | CCFL_Man2 | well, my new provider gave me a stun server |
| 02:27.40 | SophWise | its a sinking ship |
| 02:27.50 | CCFL_Man2 | and i got nervious |
| 02:27.57 | DrukenLPY | what would cause asterisk to register to a diffrent asterisk and change the username to s? |
| 02:27.57 | SophWise | ifit works without the STUN server then all |
| 02:27.58 | SophWise | is well |
| 02:28.19 | CCFL_Man2 | even if not a sinking ship, it still sucks ass, quantumvoice kicks ass |
| 02:28.20 | CCFL_Man2 | ahh |
| 02:28.43 | SophWise | well, I had Vonage for a couple of years, and had no problems, but I typically like to do it myself if I can, and got into Asterisk |
| 02:28.49 | SophWise | and havenever looked back |
| 02:29.02 | SophWise | just added a voip server to our datacneter and its all happy days from there |
| 02:29.14 | shmaltz | DrukenLPY, pb your configs |
| 02:29.15 | CCFL_Man2 | SophWise: also the latest sip firmware for the 7912 does not support multiple ringtones if anyone asks |
| 02:29.16 | shmaltz | ~pb |
| 02:29.19 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 02:29.32 | CCFL_Man2 | SophWise: what adapter you have? |
| 02:29.47 | SophWise | nothing, pure IAX for me |
| 02:29.52 | SophWise | and grandstream phone |
| 02:29.54 | SophWise | phones |
| 02:30.01 | SophWise | trunks are IAX |
| 02:30.05 | CCFL_Man2 | SophWise: the vonage adapter |
| 02:30.22 | SophWise | ok, it was the , not the PAP |
| 02:30.24 | SophWise | but the other one |
| 02:30.27 | SophWise | can't remember now |
| 02:30.34 | CCFL_Man2 | RT31P2? |
| 02:30.38 | SophWise | yes |
| 02:30.40 | SophWise | thats it |
| 02:30.49 | SophWise | used it when I lived in Mexico |
| 02:30.52 | CCFL_Man2 | you booted vonage then? |
| 02:30.53 | SophWise | and it worked great |
| 02:31.01 | SophWise | gave me a nice US number without probs |
| 02:31.04 | SophWise | yeah, |
| 02:31.07 | SophWise | at drop of a hat, |
| 02:31.20 | SophWise | nice and all, but I am a tech savvy person that would prefer to have my own |
| 02:31.25 | SophWise | box in our datacenter |
| 02:31.37 | SophWise | and have all the IVR stuff for my busniesses |
| 02:32.09 | SophWise | then just use VoiceVector for all of my prompts |
| 02:32.18 | SophWise | pretty good recordings |
| 02:32.29 | CCFL_Man2 | SophWise: i would to, but i have no way to connect to the pstn |
| 02:32.38 | SophWise | and gives me an easily maintained professional system and I only pay $2.99 per month, per DID |
| 02:32.44 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 02:32.50 | SophWise | no, go get a SIP provider |
| 02:32.54 | SophWise | or IAX provider |
| 02:32.59 | SophWise | like VoipStreet.com |
| 02:33.00 | CCFL_Man2 | ahh |
| 02:33.22 | SophWise | then connect your asterisk box right to the VoIP provider |
| 02:33.25 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:33.28 | SophWise | with all the lines youwant |
| 02:33.47 | CCFL_Man2 | you can do that with vonage if you hax your adapter |
| 02:33.57 | SophWise | for 2.99 I get unlimited simultaneous inbound/outbound channels |
| 02:34.04 | *** join/#asterisk dan__t (i=dan@neener.neener.org) |
| 02:34.06 | CCFL_Man2 | sweet |
| 02:34.13 | dan__t | 'evening. |
| 02:34.13 | SophWise | 1.6c per minute |
| 02:34.17 | SophWise | which for business is fine |
| 02:34.22 | SophWise | howdy dan |
| 02:34.31 | dan__t | How goes it? |
| 02:34.40 | SophWise | buano mate, qu pasa |
| 02:34.44 | dan__t | werd |
| 02:34.48 | dan__t | not bad. |
| 02:34.49 | dan__t | thanks. |
| 02:34.52 | CCFL_Man2 | so it's unlimited incomming and outgoing through your sip trunk? |
| 02:34.59 | *** join/#asterisk mactimes (n=mactimes@201.29.1.14) |
| 02:35.07 | mactimes | Hello everyone! |
| 02:35.12 | SophWise | well unlimited simultaneous channels, but 1.6c per minute |
| 02:35.14 | dan__t | Was wondering right quick if anyone has any suggestions for a nice VoIP home system. I was looking at this little Vonage setup by Vtech, looks kinda neat. |
| 02:35.20 | CCFL_Man2 | SophWise: ahh |
| 02:35.26 | dan__t | But I'd like to extend it a lot more than just what Vonage is offering. |
| 02:35.26 | SophWise | meaning I can have 20 lines coming in at the same time |
| 02:35.30 | SophWise | without a busy signal |
| 02:35.37 | CCFL_Man2 | dan__t: vonage sucks |
| 02:35.54 | dan__t | I've got it at the ex's house, never did me bad for simple home use, I've got no gripes there. |
| 02:35.55 | CCFL_Man2 | SophWise: ahh, wonder how thats done at the telco |
| 02:35.58 | SophWise | agreed, if you have the tech savvy, then setup an asterisk box and connect it through VoipStreet |
| 02:36.11 | SophWise | don't even go Vonage |
| 02:36.14 | dan__t | I've played with Teliax before as an IAX provider when I was doing Asterisk stuff a long time ago, but its been quite a bit. |
| 02:36.23 | SophWise | asterisk with IAX provider outstrips vonage hands down |
| 02:36.28 | dan__t | I'd be comfortable going that way again. |
| 02:36.34 | DrukenLPY | anyone know what would cause an account to change it's username? |
| 02:36.55 | CCFL_Man2 | dan__t: vonage hardware is crap |
| 02:36.59 | DrukenLPY | sip show peers shows <username>/s instead of <username>/<username> |
| 02:37.16 | dan__t | For fear of asking a dumb question... I'll just fire it off - what are my options when I want to connect a traditional digital keyset, such as a wireless phone kit, to an Asterisk setup |
| 02:37.18 | shido6 | wow |
| 02:37.27 | CCFL_Man2 | dan__t: like the vonage vtech thing? |
| 02:37.30 | SophWise | hmm, lots of options |
| 02:37.44 | dan__t | Well. With my POTS system I've got a base system, and four phones. |
| 02:37.50 | SophWise | well, keep in mind that you can get anything you could possibly want done phone wise in the |
| 02:37.52 | SophWise | asterisk box |
| 02:37.57 | SophWise | then just add sip phones as extensions |
| 02:38.06 | dan__t | hrm |
| 02:38.11 | dan__t | Hardware-wise, what should I be looking for. |
| 02:38.13 | [TK]D-Fender | DrukenLPY, check your register line |
| 02:38.30 | CCFL_Man2 | dan__t: cisco ip phones |
| 02:38.40 | dan__t | How about something a bit less expensive heh. |
| 02:38.50 | [TK]D-Fender | ~phones |
| 02:38.52 | jbot | phones is, like, http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 02:38.57 | SophWise | are you wanting to connect to your analog PSTN? |
| 02:39.27 | dan__t | No, I want to use my IAX provider. |
| 02:39.29 | SophWise | or are you going to go with a SIP/IAX provider? |
| 02:39.32 | SophWise | gotcha |
| 02:39.40 | CCFL_Man2 | [TK]D-Fender: cisco 7912s are really nice, 7940s are too big |
| 02:39.58 | SophWise | are you planning on putting togheter an asterisk box |
| 02:40.05 | [TK]D-Fender | CCFL_Man2, Cisco's are lacking in WAY to many ways, and overpriced |
| 02:40.11 | SophWise | for handling IVR, etc |
| 02:40.25 | dan__t | I am, yes. |
| 02:40.30 | CCFL_Man2 | [TK]D-Fender: as a basic phone, the 7912 is great |
| 02:40.37 | SophWise | middle-range grandstreamswork fine |
| 02:40.54 | SophWise | Snom too |
| 02:40.56 | SophWise | both good |
| 02:40.57 | [TK]D-Fender | CCFL_Man2, Again, nothing Polycom doesn't do better & cheaper... |
| 02:41.07 | dan__t | How about some stuff that I can go buy at Best Buy or something tonight. |
| 02:41.15 | CCFL_Man2 | [TK]D-Fender: but prettier lcd screen :P |
| 02:41.15 | dan__t | I want to get this rolling without any specialty equipment. |
| 02:41.26 | [TK]D-Fender | Snom has a history of flakey firmware, poor screen usage, etc. |
| 02:41.28 | SophWise | hmm, impatience in the SIP.VoIP arena leads to your foot kicking your arse |
| 02:41.30 | SophWise | later |
| 02:41.51 | dan__t | hahaha. |
| 02:42.00 | SophWise | Fender: but grandstreams are fine |
| 02:42.12 | [TK]D-Fender | dan__t, Not happening. best buy doesn't carry telecom gear. they carry Vonage adapters and other locked stuff. You will have to buy REAL hardware and would be advised that... |
| 02:42.14 | [TK]D-Fender | ~ygwypf |
| 02:42.27 | jbot | [ygwypf] You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 02:42.27 | [TK]D-Fender | ~gs |
| 02:42.43 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 02:42.43 | dan__t | And I've got no problem doing that. |
| 02:42.45 | DrukenLPY | [TK]D-Fender: lend a hand for a sec? |
| 02:42.45 | [TK]D-Fender | ^^^^^^ |
| 02:43.12 | CCFL_Man2 | [TK]D-Fender: the 7912 sip firmware is lacking big time though, all sip firmware for cisco phones are lacking, unfortunately |
| 02:43.20 | [TK]D-Fender | dan__t, You say you have some "digital" phones already you wanted to sue. what are they EXACTLY? |
| 02:43.24 | DrukenLPY | [TK]D-Fender: register line is using proper username... |
| 02:43.32 | CCFL_Man2 | though you can have cute little logos with 7912 |
| 02:43.34 | [TK]D-Fender | DrukenLPY, PM it to me |
| 02:43.34 | DrukenLPY | wouldn't be able to register otherwise... |
| 02:43.52 | CCFL_Man2 | we have some old srx phones here :P |
| 02:43.59 | [TK]D-Fender | CCFL_Man2, Same with Polycoms..... again, the ONLY thing they do a bit better is the XML push stuff. Every other way.. they lose |
| 02:44.00 | SophWise | jbot info can bite me, GS work fine. This isn't an opinion, I have plenty of them in the field working perfectly fine |
| 02:44.08 | [TK]D-Fender | DrukenLPY, PM it. |
| 02:44.50 | CCFL_Man2 | polycom's don't allow lcd logos :P |
| 02:44.59 | [TK]D-Fender | SophWise, they are hit/miss, have had nasty firmware problems, inferior feel, call handling, LCD usage. |
| 02:45.06 | [TK]D-Fender | CCFL_Man2, I beg to diff. |
| 02:45.18 | [TK]D-Fender | diff beg > you :O |
| 02:45.39 | [TK]D-Fender | DrukenLPY, guess people here tonight don't know me very well :D |
| 02:46.02 | DrukenLPY | :) |
| 02:46.03 | SophWise | yes, I will give that one, they have a cheap, non-solid feel |
| 02:46.16 | SophWise | but when everything is on one network segment |
| 02:46.19 | SophWise | everything is goo |
| 02:46.28 | SophWise | lol, good |
| 02:47.17 | [TK]D-Fender | Polycom > All |
| 02:47.34 | SophWise | sound quality wise, I would definately have to agree there |
| 02:47.46 | SophWise | polycom kicks arse on the sound q |
| 02:48.34 | dan__t | [TK]D-Fender, sorry, just using big words. POTS phones, I'd like to of course use my wireless base and accompanying handsets behind an Asterisk machine. |
| 02:48.40 | dan__t | Ideally that's what I'm looking for. |
| 02:49.30 | [TK]D-Fender | dan__t, ok, for 4 phones, get 2 x Linksys SPA-2102's, and how many lines? |
| 02:50.33 | dan__t | Ideally, 2. The phone set is capable. |
| 02:50.53 | [TK]D-Fender | dan__t, first thing to learn... you phones have nothing to do with your LINES. |
| 02:51.29 | dan__t | Right. |
| 02:51.57 | [TK]D-Fender | dan__t, Now because of the terminology you used we're going to have to be clear on something. are these 4 completely independent wireless phones, or one of those multi-handset home "systems"? |
| 02:52.27 | dan__t | As I said, one of those multi-handset home "systems". In that case, yes, the phones do have everything to do with the LINES. |
| 02:52.33 | dan__t | I don't see how they wouldn't otherwise. |
| 02:52.49 | dan__t | And I would not have mentioned the "capability" |
| 02:52.59 | [TK]D-Fender | dan__t, ok, this poses an issue because they were not really meant to operate independently |
| 02:53.14 | dan__t | Those SPA-2102s look pretty neat. |
| 02:53.23 | [TK]D-Fender | dan__t, they "share" the lines connected to the base. These kinds of phones aren't ideal for the concept of independance |
| 02:53.31 | dan__t | Yes, I understand. |
| 02:53.51 | [TK]D-Fender | dan__t, so you may ave 2 lines going into the bse, but 1 guy could be on a call and another handset can but in, etc. |
| 02:54.04 | dan__t | Yep. |
| 02:54.17 | [TK]D-Fender | dan__t, also 2 could be on their own calls and the 3rd & 4th have no way to dial out |
| 02:54.25 | SophWise | UTStarcom's F3000 does a good job too for a wireless phone |
| 02:54.29 | dan__t | Yep. |
| 02:54.33 | [TK]D-Fender | ~wifisip |
| 02:54.55 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 02:55.52 | SophWise | I am having less and less respect for this jbot rubbish, sounds like opinionated uninformed vacuosity |
| 02:56.01 | SophWise | myexperience with all ofthat has been perfect |
| 02:56.35 | SophWise | again, I'm not speaking from opnion, its been my experience that I have had zero problems |
| 02:57.19 | SophWise | its like jbot is pulling its crap from some editorial rubbish wiki that is sponsored by cisco and polycom |
| 02:57.43 | SophWise | ~jbotisapos |
| 02:57.55 | SophWise | nope, no entry, bummer |
| 02:57.55 | [TK]D-Fender | SophWise, Ironic.. you are telling EXACTLY what your opinion is :) Of course its an opinion, what else would it be? :) |
| 02:58.13 | SophWise | silly me, i said I had NO problems with it, that sounds more like FACT not OPINION |
| 02:58.23 | [TK]D-Fender | SophWise, I've had clients cursing the UTSC, I loath the WIP300 I have personally, and the rest.... rarely worth thinking about |
| 02:58.25 | SophWise | 3rd grade english taught me the diff |
| 02:58.58 | SophWise | the f3000 I have in the hands of several travelling exec that have used them without a single hitch |
| 02:58.59 | [TK]D-Fender | SophWise, You are clearly blessed... others in my years of being here have not been so fortunate. |
| 02:59.04 | SophWise | in Australia, Switzerland |
| 02:59.06 | SophWise | and US |
| 02:59.14 | SophWise | with even one incident |
| 02:59.31 | SophWise | there was a bit of a latency issue in ozzieland |
| 02:59.40 | SophWise | but other than that, they performed great |
| 02:59.56 | SophWise | of course that doesn't include public hotspots |
| 03:00.05 | SophWise | its all controlled wireless infrastructure |
| 03:00.29 | [TK]D-Fender | Linksys doesn't neal with NAT well, has no transfer etc. Most are lacking in too many ways. poor batter life / range (the F1000 was bad for 1 client of mine) |
| 03:00.54 | [TK]D-Fender | SophWise, But unless you are truely mobile (out of office), WIFI is not needed, nor suggested |
| 03:01.23 | SophWise | its more for office environments where we have numerous repeaters |
| 03:01.30 | SophWise | all well saturated in wifi signal |
| 03:01.59 | *** join/#asterisk pifiu (n=someone@c-65-34-152-249.hsd1.fl.comcast.net) |
| 03:02.38 | irule | a customer of mine, sent the delivery guy to a computer store to buy a WIRELESS CABLE, the poor delivery guy called his boss to tell them that they dont carry that produtct, so he calls me, asking me to call the store, so that I can explain to them what he wants, heh |
| 03:02.39 | [TK]D-Fender | SophWise, the specifics of a particular instal will determine the rules naturally |
| 03:02.45 | SophWise | like I said, not tested with public hotpots, so surely that could be patchy |
| 03:02.47 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 03:03.25 | _Sam-- | how can i find out what the possible values in SIP_HEADER are? |
| 03:03.43 | _Sam-- | im trying this, but getting nothing ;exten => 212202XXXX,1,Noop(Alert-Info -> '${SIP_HEADER(Alert-Info)}' |
| 03:03.46 | DrukenLPY | irule: uhmm... a wirless cable eh? |
| 03:03.55 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 03:03.56 | [TK]D-Fender | BBIAB..... |
| 03:03.56 | ohadz | why can't i dial a single number? i get a dialtone but each ext i call i get a bz signal.. |
| 03:03.58 | ohadz | http://pastebin.ca/480150 |
| 03:04.07 | _Sam-- | (trying to distinguish which DID was dialed from broadvoice) |
| 03:04.46 | [TK]D-Fender | ohadz, SIP/2.0 404 Not Found |
| 03:04.54 | [TK]D-Fender | Looking for 1000 in other (domain 10.1.7.17) |
| 03:04.54 | dan__t | Welp, thanks for the tips. |
| 03:05.14 | irule | DrukenLPY I went to visit my customer with a bunch of hardware to explain it all |
| 03:05.22 | ohadz | [TK]D-Fender, so i need to install sip2.0? |
| 03:05.25 | SophWise | sure thing, hasta dan |
| 03:05.27 | _Sam-- | i have about 5 different itsps....just trying to setup a specific BV account on my asterisk server for a friend. |
| 03:05.31 | dan__t | Much appreciated. |
| 03:05.36 | dan__t | You too, [TK]D-Fender. |
| 03:05.40 | [TK]D-Fender | ohadz, No, the number you dialed is NOT FOUND |
| 03:05.43 | _Sam-- | but broadvoice sends all calls over the same DID...and you cant tell what number was called without examing the sip_header |
| 03:06.35 | ohadz | [TK]D-Fender, Really destroying SIP dialog 'd0052632-abd3c0cc@10.1.7.167' Method: ACK |
| 03:08.31 | ohadz | [TK]D-Fender, http://pastebin.ca/480155 my sip.conf file.. |
| 03:08.44 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 03:10.26 | _Sam-- | anyone got any info about the SIP_HEADER function? the wiki page leaves alot to be desired....http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header |
| 03:11.27 | ohadz | [TK]D-Fender, my extensions.conf file -- http://pastebin.ca/480161 |
| 03:12.23 | _Sam-- | seen zoa |
| 03:14.40 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.44 | ohadz | [TK]D-Fender, i tried to call 1234, 8500, 500 (demo ext), 777 etc.. i am trying to get to the demo ext, voicemail etc..nothing.. everytime i try i get to bz single |
| 03:16.23 | *** join/#asterisk joebob777as7 (n=corn13re@yoda.peacefulescape.com) |
| 03:16.58 | *** join/#asterisk Trevor_b (n=tbenson@64-142-72-37.dsl.static.sonic.net) |
| 03:17.25 | joebob777as7 | hey guys. i'm trying to set up an asterisk box for our office and had some questions. I am wondering what i will need. we currently have 3 phone lines in our office one is a fax. |
| 03:17.41 | Trevor_b | Any developers who work on Digium GUI here? |
| 03:18.08 | Trevor_b | joebob, tdm400 sounds about right |
| 03:18.28 | Trevor_b | 4 line interface |
| 03:18.28 | Trevor_b | unless that is you wanted to run analog phone on the system too. |
| 03:18.35 | Trevor_b | inside extensions. |
| 03:18.47 | joebob777as7 | what do you mean Trevor_b ? |
| 03:19.06 | joebob777as7 | sorry still just getting into this |
| 03:19.12 | Trevor_b | Were you going to use softphones or VoIP phones, or POTS (house style) phones |
| 03:20.01 | joebob777as7 | pots probably i guess i could use viop phones... |
| 03:20.09 | joebob777as7 | probably not softphones |
| 03:20.13 | Trevor_b | ok |
| 03:20.45 | Trevor_b | VoIP phones cost a decent amount, depending on the model and features. POTS (analog) phones are inexpensive, but now you have to get hardware to plug them into on the system. |
| 03:20.52 | Trevor_b | How many desks you have inside your office |
| 03:20.59 | joebob777as7 | I was looking into broadvoice.com and it looked like they offered dirt cheap voip for around $40 a month. we are currently paying $150 or so |
| 03:21.06 | Trevor_b | UGH |
| 03:21.10 | joebob777as7 | we have 5 desks looking to add 3 more or so |
| 03:21.10 | Trevor_b | i hate BV |
| 03:21.15 | dan__t | What about Teliax? |
| 03:21.18 | Trevor_b | i have 3 broadbvoice lines, dropping them |
| 03:21.19 | dan__t | What's the take on them? |
| 03:21.19 | Trevor_b | YES! |
| 03:21.21 | Trevor_b | Teliax rocks |
| 03:21.24 | dan__t | Good stuff huh |
| 03:21.35 | dan__t | I've had Teliax service, but have noto used it in well, well over a year |
| 03:21.36 | Trevor_b | Corporate 44 is nice, otherwise their pay as you go or just any plan is good. |
| 03:21.40 | Trevor_b | email support is decent |
| 03:21.45 | Trevor_b | and they actually respond |
| 03:21.55 | Trevor_b | BV drops calls, has RTP issues, and ignores inbound support line. |
| 03:22.00 | dan__t | I want unlimited service, with the possibility of a toll free number. |
| 03:22.10 | Trevor_b | i am yanking customers i have off BV andmoving to teliax. |
| 03:22.17 | Trevor_b | well |
| 03:22.42 | Trevor_b | BV 'unlimited' is fairly unlimited, but word is that if you REALLY push the calls then they will backcharge and do weird things. |
| 03:22.54 | dan__t | Welp, ok, I guess I need to find a set of VoIP phones then. |
| 03:23.02 | joebob777as7 | ok so would teliax benifit me Trevor_b ? |
| 03:23.04 | dan__t | By doing that, I'd be set. |
| 03:23.14 | Trevor_b | Teliax is up front,. they softcap 2500 minutes, and then charge $.02 per minute after that (or at least corporate is 2500 and 4 lines simultanious) |
| 03:23.20 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.203.59) |
| 03:23.31 | *** join/#asterisk tenzind (n=tenzind@202.144.144.77) |
| 03:23.39 | Trevor_b | ok |
| 03:23.43 | Trevor_b | about 8 desks |
| 03:23.46 | Trevor_b | 3 inbound lines |
| 03:23.56 | Trevor_b | that would take a tdm2400 |
| 03:24.00 | Trevor_b | 24 possible ports |
| 03:24.21 | dan__t | tdm2400, assuming one would not want to use real VoIP phones? |
| 03:24.32 | Trevor_b | FXS=Station (inside desks, power to make the phone ring) FXO=office (inbound service provider line, provides power for ring) |
| 03:24.36 | Trevor_b | so |
| 03:24.45 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:24.59 | dan__t | Sorry, just trying to relate all of this to the Inter-Tel setup that we have at work heh. |
| 03:25.06 | Trevor_b | TDM2400 with 2FXS modules (each module is 4 ports, making 8 internal phones) and 1 FXO module (4 possible outside lines). |
| 03:25.32 | Trevor_b | DAN yes, joebob was talking about using POTS inside the office |
| 03:25.46 | *** part/#asterisk chino (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 03:26.22 | Trevor_b | joebob777as7: yes teliax would be goood instead of broadvoice, plus if your on west coast they have a new EXTREMELY low latent CA server. |
| 03:26.41 | joebob777as7 | well i can run all voip i guess would that be a lot better? I am in Oregon so that's good |
| 03:27.02 | dan__t | Ah alright, sorry |
| 03:27.05 | joebob777as7 | ok so that looks to be a lot cheaper |
| 03:27.09 | Trevor_b | its less hardware lines to desks to wire, and more just rely on network connections at desks. |
| 03:27.17 | Trevor_b | ok |
| 03:27.22 | Trevor_b | BRB gotta go deal with my son :) |
| 03:27.26 | pifiu | trebor wow thanks for the input on teliax |
| 03:27.38 | pifiu | i was actually thinking about calling them tomorrow sinc ei might switch from IP Communications |
| 03:27.59 | Trevor_b | try pinging voip-ca1.teliax.com for millisecond ratings to their new beta server |
| 03:28.03 | Trevor_b | BRB |
| 03:28.04 | joebob777as7 | I can do all the wiring. so what am i looking like i'm in for? |
| 03:28.24 | joebob777as7 | 85ms |
| 03:30.36 | pifiu | are you guys using teliax with IAX or SIP? |
| 03:32.08 | joebob777as7 | ok Trevor_b i'm really interested this could save our company lots of money! |
| 03:33.27 | joebob777as7 | so Trevor_b if i have one line i can just set up park lines right? and push ppl on hold to park and not pay for additional lines? how does this work? BTW take your time |
| 03:33.42 | Trevor_b | pifiu: using teliax with IAX, every so often i get a weird call on it, but i havent even tried them on SIP. So many RTP issues with broadvoice that i got sick of thinking about SIP issues. |
| 03:34.06 | Trevor_b | you mean 1 inbound VoIP line? |
| 03:34.15 | pifiu | well i was gonna go to broadvoice but i keep hearing conflictng stories |
| 03:34.19 | pifiu | some people swear by it some dont |
| 03:34.24 | pifiu | what do you mean a "weird call"? |
| 03:34.36 | pifiu | the issue im having with IP Communications is that i think they are over selling their services |
| 03:34.39 | joebob777as7 | yes 1 inbound voip line? or would we need 3? |
| 03:34.44 | pifiu | randomly i cant dial out or calls dont come in |
| 03:34.46 | Trevor_b | yeah, i have 4 broadvoice accounts myself for 2 companies, i have 2 clients with each 2 accounts. CLosing all that crap down for teliax. |
| 03:34.50 | pifiu | i think they are out of channels |
| 03:35.40 | joebob777as7 | Trevor_b, was that a yes to my question? or were you talking to pifiu ? |
| 03:36.21 | Trevor_b | a call where the other end seems like they here me great, but it sounds a bit funny to me (1 in many calls, not very often). This is much more acceptable than having a call drop from the middle of a conversation, or getting the random error on their proxy where it sends you their voice, but yours doesnt go back, or vice versa. There are many issues like that to resolve in asterisk itself, but when BV has this randomly happen when i |
| 03:37.04 | Trevor_b | joebob777as7: depends on provider. Broadvoice basic lines (1 number) give you 2 channels. So 2 simultanious calls going at one time, on one account. |
| 03:37.53 | Trevor_b | teliax basic i cant remember, corporate is 4 lines simultanious. OR pay as you go $10 a month, plus $.02 per minute (billed in 1/10 minute increments, or 6 seconds). |
| 03:37.56 | pifiu | hmm trevor, well i think i might try them |
| 03:38.02 | Trevor_b | pay as you go is unlimited channel |
| 03:38.05 | pifiu | i wonder if they can add more channels? i need 5 |
| 03:38.29 | Trevor_b | pifiu: how many minutes a month on current plan you use? |
| 03:39.14 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:39.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:40.07 | Trevor_b | joe: parking a call is like putting a call on hold. MOST voip lines you can consider to be like SBC/PacificBell/AT&T house line + call waiting. Except with asterisk call waiting doesnt mean you have to put one on hold to take the other, as long as you have another employee to answer at same time. |
| 03:40.31 | _Sam-- | is there a way to echo the "INVITE" from the SIP header via noop or anything? |
| 03:40.33 | pifiu | i can tell you from my past invoice in a sec |
| 03:40.41 | pifiu | i think the 2500 shoudl be fine |
| 03:42.04 | dan__t | Trevor_b, just got off the phone with a friend who uses a Polycom 301 for most of his installs... looks to be a decent phone, may I ask your take on it? |
| 03:42.09 | pifiu | We did 409 minutes inbound and 940 outbound |
| 03:42.13 | Trevor_b | pifiu: i have used corporate on the call center we support to dial up to 5, even though they say its 4 lines simultanious. Not to say they wont tighten up. But if you need more outbound lines then 4, pay as you go is good, especially if minutes are low and you need only a single number. |
| 03:42.24 | pifiu | but thats very weird, normally its about even at 1000 each |
| 03:42.31 | Trevor_b | danalien: poly is my FAVORITE |
| 03:42.38 | Trevor_b | dan__t: test |
| 03:42.41 | pifiu | so even at 2500, i have 500 overhead and .02 over doesnt seem expensive |
| 03:42.44 | dan__t | test |
| 03:42.46 | pifiu | i have polycom 501's and they are graet |
| 03:42.47 | Trevor_b | dan__t: there didnt realize we had another dan in here ;) |
| 03:42.51 | dan__t | heheh |
| 03:43.01 | joebob777as7 | so what's the next step? Trevor_b |
| 03:43.10 | dan__t | I think I can use two of those 301's, look to be decent setups. I think they will work well. |
| 03:43.32 | Trevor_b | yeah 501s are very nice, 601's are REALLY nice, never noticed how much the display upgrade was to the look of the phone.. plus i like the smaller line buttons, and the LED mute, speaker, headset buttons. |
| 03:43.51 | Trevor_b | but that costs too ;) |
| 03:44.02 | dan__t | Eh I think the 301 is in my price and needed features range |
| 03:44.22 | Trevor_b | dan__t: yes 301's (two lines?) are good, if they have speakerphone (cant remember) then you have the best speakerphone in industry... |
| 03:44.36 | JT | 301s don't have speakerphone afaik |
| 03:44.39 | dan__t | I want a headset, too, hoping I can find one. |
| 03:45.07 | Trevor_b | polycom makes the speaker for the Cisco phones, and POLY has a MUCH easier ftp method to upgrade and setup then Cisco TFTP with phones that decide to stop asking for firmware and shit. |
| 03:45.34 | Trevor_b | dan__t: voipsupply.com they have the 301 bundled with a headset for ya ;) |
| 03:45.37 | dan__t | hah! |
| 03:45.43 | joebob777as7 | Trevor_b, any suggestions on good cheap voip phones? |
| 03:45.55 | Trevor_b | grandstream budgettones |
| 03:46.13 | Trevor_b | $37 on voipsupply.com, decent quality, amazing at sub $40 per phone |
| 03:47.04 | Trevor_b | if you want to run your computer from the built in jack because you dont have lots of ethernet to each desk, then upgrade to the $45 or $50 version with the 100Mbps jack though, NEVER slow down your network to 10Mbps unless you only print and use internet, if you do file access on local network get the 100Mbps! |
| 03:47.35 | Trevor_b | GS-100 through GS-102 i think |
| 03:47.36 | JT | ~gs |
| 03:47.50 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 03:47.50 | pifiu | trevor so ill give teliax a call |
| 03:47.50 | pifiu | tomorrow |
| 03:47.50 | pifiu | see if they can do 5 ports |
| 03:47.51 | joebob777as7 | those work nice with asterisk? |
| 03:47.54 | Trevor_b | yeah, i just sign up online ;) |
| 03:48.13 | pifiu | their additional numbers are expensive though! |
| 03:48.21 | pifiu | 9.99 each? |
| 03:48.26 | pifiu | i think i pay 1.99 each right now |
| 03:48.30 | pifiu | that might kill me |
| 03:48.35 | pifiu | i have like 23 did's |
| 03:48.53 | Trevor_b | Do you NEED 23 DID's or you just had them with a PRI? |
| 03:49.15 | JT | do not buy grandstream for business purposes |
| 03:49.28 | pifiu | i got them as a current package i have. i cant beat that. its 120 a month roughly for 23 of them, 5 channels at .02 outbound |
| 03:49.38 | pifiu | i dont need tem all maybe, like 5 or 6 |
| 03:49.54 | joebob777as7 | JT, why not grandstream for business purposes? |
| 03:50.01 | dan__t | I don't quite know what I'm looking for when it comes to a bundle like that, Trevor_b. I found a few Polycom keysets with a headset for ~189, didn't say anything about speaker, though. |
| 03:50.18 | Trevor_b | jbot: what GOOD.. and CHEAP.. voip phones do you suggest? if they want quality i suggest polycom, i hear snom are ok, and lots of business i see run GXP-2000 and never complained, but i dont support them or use them either. All we do is polycom, although i have some test units from the GS-102 series and have not found an issues yet, but they are lab phones, not our desk lines. |
| 03:50.32 | jbot | Trevor_b: what are you talking about? |
| 03:50.34 | Trevor_b | I dont think the 301 has speakerphone |
| 03:50.42 | Trevor_b | think you have to get 500 series for speaker. |
| 03:50.49 | dan__t | d'oh. |
| 03:51.26 | JT | Trevor_b: or 430 |
| 03:51.31 | *** join/#asterisk orn (n=orn@85.197.193.137) |
| 03:51.48 | JT | Trevor_b: usually "good" and "cheap" don't go together |
| 03:51.53 | orn | Hi! Any life saver here want to take a gander at this and tell me what's wrong?: |
| 03:51.54 | orn | Reliably Transmitting (no NAT) to xx.xxx.xxx.xx:5060: |
| 03:51.54 | orn | SIP/2.0 404 Not Found |
| 03:51.54 | orn | Via: SIP/2.0/UDP RYJKICIP01C:5060;maddr=xx.xxx.xxx.xx;branch=z9hG4bK-7808-1d4dfd6-a8cb36;received=xx.xxx.xxx.xx |
| 03:51.59 | SwK | the 300s dont have a full duple speaker they only have listen on... 430 and up polycoms ahve the full duplex speaker |
| 03:52.01 | Trevor_b | jbot: joebob wanted to know about good cheap voip, I just mentioned the GS-102 as something i had seen and was amazingly cheap. Just wondering since you said they were suck and fail, and i only have them in a lab not production, what you suggest on cheap voip hardphone. |
| 03:52.11 | JT | joebob777as7: they're cheap rubbish that is inferior to most other stuff on the market |
| 03:52.11 | pifiu | get the 500 series at leat |
| 03:52.16 | pifiu | the 300 series feels cheap |
| 03:52.20 | pifiu | and the display sucks |
| 03:52.27 | pifiu | correct swk |
| 03:52.33 | pifiu | get the 500 |
| 03:52.37 | SwK | Trevor_b, ahhh jbot is exactly what the title implies... hes a bot... |
| 03:52.38 | pifiu | or 600 if you can |
| 03:52.43 | SwK | 650 y0 |
| 03:52.47 | *** join/#asterisk bmg505 (n=leon@196.209.179.167) |
| 03:52.47 | dan__t | Trevor_b, Polycom IP 320 - http://www.voipsupply.com/product_info.php?products_id=2907?utm_source=polycomcompare&utm_medium=blog |
| 03:52.49 | dan__t | Looks perfect. |
| 03:53.09 | Trevor_b | pifiu: i thought 300 and 500 had same display, just different buttons? Or was display size differnt but res was not high until 600?? |
| 03:53.13 | joebob777as7 | JT, know anything about gxp-2000? |
| 03:53.31 | JT | yes, it's not recommended for business use |
| 03:53.41 | JT | grandstreams should always be regarded as toyhs |
| 03:53.43 | Trevor_b | anyone actually have Snom's |
| 03:53.44 | Trevor_b | ? |
| 03:53.44 | JT | toys |
| 03:54.10 | Trevor_b | JT you use poly or something else? |
| 03:54.16 | JT | yes |
| 03:54.59 | Trevor_b | yeah, the only thing i SUGGEST for business at this point is Polycom series. Hate supporting Cisco, and havent tried snom's. Had grandstreams as cheap tossouts for when softphones went nuts on callcenters. |
| 03:54.59 | pifiu | the resolution is different |
| 03:55.19 | pifiu | ciscos are a pain to deploy unless you use all cisco nice call manager stuff |
| 03:55.22 | joebob777as7 | another question... if we have one line through teliax and want to be able to park 3 calls do I need a 4 line phone or can i program the parkin buttons? |
| 03:55.26 | joebob777as7 | sorry i'm a newb |
| 03:55.31 | Trevor_b | ahh, dont think i have a 301, think mine are all 501 and a couple 601'ds. |
| 03:55.33 | ohadz | why is it that every number i call gives me a busy signal? anyone? (i just finished install asterisk again.. am i missing anything - downloaded svn from asterisk, libpri, zaptel. i installed all and then started *. setup my phone to have the user and pass auth with * in addition to the proxy. setup sip.conf and ext.conf - i have a dialtone but all calls are bz. |
| 03:56.19 | Trevor_b | joe: park is like hold, if you have 1 line, and they give you 2 simlutanious calls (like call waiting) you can only park 1 call in your PBX and answer the other (consider park like hold for GROUPS, not for your personal extension). |
| 03:56.43 | Trevor_b | ohadz: you setup a dialplan that matches your dialstring? |
| 03:56.48 | dan__t | Thanks for the tips, Trevor_b. Very, very much appreciated. |
| 03:57.19 | Trevor_b | NP, glad someone here who had seen GS in use commented other then me. |
| 03:57.24 | joebob777as7 | Trevor_b, so if i get the teliax corporate i can have 3 parks and be on the phone? do i need a four line voip phone o do that? |
| 03:57.43 | Trevor_b | park is to the system, so you wont need the lines on the phone |
| 03:57.54 | Trevor_b | a single line phone could system park 20 calls, and be on 1 |
| 03:57.58 | pifiu | oh trebor do you know of any other test teliax servers? |
| 03:58.01 | pifiu | im in miami |
| 03:58.01 | dan__t | Bah, looks like they're all on backorder, regardless of where you buy it. |
| 03:58.04 | JT | fwiw, polycom line appearances allow multiple calls |
| 03:58.04 | Trevor_b | as long as you didnt limit your parking in some way. |
| 03:58.06 | pifiu | routing to cali aint close! |
| 03:58.14 | Trevor_b | pifiu: you mean to ping? |
| 03:58.16 | JT | their 2 line phone can do 8 simultaneous calls or something |
| 03:58.17 | pifiu | yeah |
| 03:58.19 | Trevor_b | yeah |
| 03:58.27 | pifiu | actually i would be doing from houston to the provider |
| 03:58.28 | pifiu | and then to miami |
| 03:58.41 | pifiu | i have a colocated server that connects to the provider and then to our 2 locations |
| 03:58.41 | joebob777as7 | Trevor_b, so how on the voip phone do you recover let's say park 7? do you program a button to park 7? |
| 03:58.54 | Trevor_b | voip-co1.teliax.com (co1 through co3) voip-ny1.teliax.com and a couple others, dont remember off the top of my head |
| 03:59.31 | Trevor_b | you can program a button dependin on the phone, or you could make it a dial code (like *69) |
| 03:59.34 | Trevor_b | lets sayy |
| 03:59.53 | Trevor_b | randomly you setup Parking with *58 |
| 03:59.54 | pifiu | hmm |
| 03:59.57 | ohadz | Trevor_b, hmm.. the dialplan on my phone is this : (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) -- i don't know how to machien that with * |
| 03:59.58 | pifiu | do you know any closer ones? |
| 04:00.04 | pifiu | actually colorado isnt far |
| 04:00.05 | joebob777as7 | so what phones should i get for my business that i can program some buttons and make it easy for my very unsavvy people... |
| 04:00.08 | Trevor_b | or pickup parked with *58 that is |
| 04:00.30 | Trevor_b | you get a clear line, dial, and then i beleive dial which park to grab. I never do it that way, so not really thinking about it. |
| 04:00.57 | Trevor_b | I like polycom services buttons where you code a xml webpage appliucation to jump into that, list the calls, and then highlight one and grab it from park. |
| 04:01.11 | Trevor_b | on the LCD of the phone that is. |
| 04:01.14 | pifiu | oh wow 19 ping to voip-co1.teliax.com |
| 04:01.27 | Trevor_b | co is colorado (or so i beleive) |
| 04:01.39 | pifiu | yeah thats where they are based off |
| 04:01.43 | Trevor_b | yeah i get like 14-17 to ca1, which is amazing |
| 04:01.49 | pifiu | can you pick which one you connect to? |
| 04:02.21 | Trevor_b | yeah, in the setup window you set the proxy to have inbound from, then just set that for your registration and peer in iax.conf so you dial out through it and register with it for the inbound.... |
| 04:02.36 | joebob777as7 | Trevor_b, so do you have a specific phone with programmable buttons that you'd suggest for our business? |
| 04:02.40 | Trevor_b | dont set 1 proxy in the web page and another in asterisk though, outbound is good but inbound hits a server your not speaking with. |
| 04:04.19 | Trevor_b | Polycom is the way to go really. Best quality and features. 301 is the baseline (or 320 as others suggested as the newer model), 430's are supposed to be nice as well from what i hear. I use 501's for every desk we do and clients, unless they want more then 3 physical lines on the phone (each phone can have a call waiting instance on lines, so you can actaull make a 1 line polycom have like 8 'hold' lines on that one line). |
| 04:04.54 | ohadz | Trevor_b, how do i go about doing that?> |
| 04:05.04 | Trevor_b | if they want more then 3 physical buttons for lines, then i go 601's. The higher res display is also very nice for reading, and web pages if you code any custom apps for the phone. |
| 04:05.17 | Trevor_b | ohadz: the multiple lines per phone? |
| 04:05.25 | Trevor_b | sorry per line |
| 04:05.51 | ohadz | i have a linksys spa921 |
| 04:05.57 | Trevor_b | IS THERE AN ASTERISK DEVELOPERS IN HERE WHO WORK ON THE GUI? |
| 04:06.45 | Trevor_b | ohadz: should be in the web page configuration for "line instances" or thats how polycom refferences it, i think cisco did too, but i think the 921 is before linksys looked like knockoff ciscos?? |
| 04:08.19 | Trevor_b | ohadz: just glancing at the dialplan line. Hmmm, thats alot of OR's. You in a country with different standards then US arent you? |
| 04:08.22 | pifiu | trevor are you going to be online during the day tomorrow? |
| 04:08.27 | joebob777as7 | Trevor_b, how many buttons could i customize with the 501's? |
| 04:08.33 | ohadz | Trevor_b, nope. it still looks like a cisco knockoff |
| 04:08.34 | ohadz | :) |
| 04:08.50 | joebob777as7 | Trevor_b, or am i going to have to teach a bunch of my agents to use keycodes |
| 04:09.02 | Trevor_b | pifiu: maybe, trying to track me down a digium developer to talk to them about tyhe GUI supporting Zaptel. probably sign in from work and hang out until i get a response... |
| 04:09.11 | ohadz | Trevor_b, nope. i'm in the us |
| 04:09.43 | Trevor_b | joebob777as7: how many things you intend to code? Polycoms come with hold, redial, all sorts of things, so it sort of depends on what you think you need the phones to do, that they dont already. |
| 04:10.15 | Trevor_b | like 6-8 buttons or something, none in front of me (had to take my 501 to work for my brother) so no more deskline at home anymore. |
| 04:10.25 | Trevor_b | ANYONE HAVE A GOOD IAX HARDPHONE SUGGESTION? |
| 04:10.51 | *** join/#asterisk friedrich| (n=friedric@e177246168.adsl.alicedsl.de) |
| 04:10.53 | sevard | Trevor_b: PROBABLY BUT IN CAPS? |
| 04:11.03 | Strom_M | Trevor_b: must you fucking yell? |
| 04:11.13 | SwK | SPEAK UP I CANT HEAR YOU |
| 04:11.22 | *** join/#asterisk BBhoss (n=hoss@adsl-230-28-185.hsv.bellsouth.net) |
| 04:11.32 | joebob777as7 | Trevor_b, so should i look into iax phones since i'm running asterisk? |
| 04:11.34 | Trevor_b | considering i cant get anyone to answer besides the guys im replying to figured maybe someone would notice if it was CAPS.......... |
| 04:11.48 | Trevor_b | joebob777as7: im just always on the lookout for good iax hardware./ |
| 04:12.01 | Strom_M | Trevor_b: I just came in here, but since you're being irritating I'm going to refuse to answer your questions on principle |
| 04:12.23 | sevard | Trevor_b: I'm following suit behind Strom_M on that decision. |
| 04:12.28 | joebob777as7 | so i would be fine with getting some 501's with my asterisk system if going through teliax? |
| 04:12.50 | ohadz | Strom_M, can you answer my questions?:) |
| 04:13.17 | SwK | Trevor_b, if you want people to answer you maybe you should try hiring a consultant... bitching and whining on IRC for answers will get you no where |
| 04:13.24 | Trevor_b | . o O (gee how mature, feel like im in highschool again) |
| 04:13.39 | Strom_M | ohadz: i don't see your question in the last two screens; can you repeat it please? |
| 04:13.40 | Trevor_b | SwK what? |
| 04:13.40 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:13.50 | sevard | Trevor_b: yelling for attention should make you feel like you're in elementry school, not highschool. |
| 04:13.51 | Trevor_b | OMFG people dont even know shit about what they are saying now. |
| 04:14.40 | SwK | this is irc people arent required to answer your questions.. we do this becuse we want to... so excuse my being a dick and saying if you dont like the speed at which we reply (if we even feel like replying in the first place) then go hire a consultant |
| 04:14.53 | Strom_M | someone better call the waaaaaahhhhhhmbulance for Trevor_b |
| 04:14.57 | Trevor_b | nobodies yelling, i just caps'd the line. get real. One reason i hate trying to track developers in IRC is the whining that goes on in here. I am helping spending my time explainging things to noobs while im trying to track a dev to get a question answered and this is the crap i get for typing a caps line? Whatever. |
| 04:15.05 | sevard | Trevor_b: I'd like to extend my services as a VoIP consultant to you at this time. |
| 04:15.15 | ohadz | why is it that every number i call gives me a busy signal? (i just finished installing asterisk again.. am i missing anything? - downloaded svn from asterisk - made asterisk libpri, zaptel. i installed all and then started *. setup my phone to have the user and pass auth with * in addition to the proxy. setup sip.conf and ext.conf - i have a dialtone but all calls are get a bz signal.. |
| 04:15.18 | SwK | see there you go a consultant |
| 04:15.38 | Strom_M | ohadz: a busy signal, or a "fast busy"? |
| 04:15.44 | Trevor_b | SURE dude, get a life. I build Call Centers and VoIP business systems with asterisk, i asked if someone KNEW of a good IAX hardphone instead of softphone, WOW i need your help now, please help me....... |
| 04:15.58 | ohadz | Strom_M, i guess it's a fast busy... |
| 04:16.07 | Strom_M | ohadz: that's a reorder tone, not a busy signal |
| 04:16.13 | Strom_M | it means your call never went anywhere in the first place |
| 04:16.15 | remmo | Trevor_B: there are some YUMCHA phones but not good IAX phones |
| 04:16.23 | sevard | Trevor_b: No problem, sir. Does paypal work for you? |
| 04:16.24 | Strom_M | have you configured extensions.conf yet? |
| 04:16.24 | SwK | Trevor_b, if you atually built callcenters and VoIP businesses for a living (liek I have for the past several years) you would know the answer to those questions... |
| 04:16.36 | remmo | Trevor_B: i have about 10 here and they work but your better off with a real SIP phone |
| 04:16.47 | ohadz | Strom_M, yes |
| 04:16.58 | SwK | *yawn* |
| 04:17.05 | ohadz | Strom_M, my extensions.conf file -- http://pastebin.ca/480161 |
| 04:17.56 | Strom_M | what context do your sip phones live in? |
| 04:18.13 | Trevor_b | SwK get real. Knowing that poeople in china or other countries are having a new release that someone else may have seen before me is something I would ONLY know if i had been building systems for a living? Thats a pretty stupid statement, i know people who know nothing about building the system and configuration inside asterisk or creating custom gateway interfaces that will know more then you or i about new voip hardware on the ma |
| 04:19.15 | sevard | on the machines that I put my penis into for a living. I'm a big jerk who likes to yell at people, blah blah all day long |
| 04:19.18 | ohadz | Strom_M, i have a sip linksys spa921. - i am not using any digium cards, just my internet connection.. though i'm not sure if i answered your question..:/(newbiez) |
| 04:19.32 | Strom_M | ohadz: no, that tells me nothing |
| 04:19.41 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.169.195) |
| 04:19.55 | Strom_M | ohadz: in your sip.conf entries, what does your context= line say? |
| 04:20.15 | pifiu | ok trevor im out |
| 04:20.17 | pifiu | thanks for the help |
| 04:20.23 | Trevor_b | your welcome pif |
| 04:21.23 | ohadz | Strom_M, tried both default and other |
| 04:21.37 | Strom_M | it should be set to "default" |
| 04:22.48 | ohadz | Strom_M, ok.let me try that.. |
| 04:23.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:25.28 | Trevor_b | Any developers who work on the Digium GUI around> |
| 04:26.02 | Strom_M | not that I know what I'm talking about, but you might have more success in #asterisk-gui as per the topic |
| 04:26.17 | hacim | I've added an extention number to my extentions.conf in a sip context, is there a way I can reload asterisk to get that new config, without stopping and starting asterisk completely? |
| 04:26.30 | Strom_M | hacim: "extensions reload" at the CLI |
| 04:26.35 | hacim | Strom_M: thanks |
| 04:26.40 | Trevor_b | last time i spent time in there it was less to do iwth the gui then i got in here. not that i wont try again, but dead as for keeping on topic last timel |
| 04:27.35 | joebob777as7 | do you guys think i'd be fine with the polycom ip 330's? does the second ethernet go to the computer? |
| 04:28.27 | sevard | joebob777as7: yeah, it's a switch. |
| 04:28.38 | Strom_M | the 330 looks like a sexy phone |
| 04:28.40 | Strom_M | I want one |
| 04:29.13 | sevard | Strom_M: that would make the number of phones in your house to a total of... 584? |
| 04:29.26 | Strom_M | 2,368 |
| 04:29.30 | sevard | s/make/bring |
| 04:29.46 | sevard | I only have 42. |
| 04:32.12 | ohadz | Strom_M, that worked! thank you. now i get the following error -- Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/888-b6a8bd40' Really destroying SIP dialog 'f93c84db-299dcf69@10.1.7.167' Method: BYE |
| 04:35.31 | Strom_M | that just means the call is over |
| 04:35.38 | Trevor_b | hopping over in that gui channel |
| 04:36.06 | Trevor_b | Anyone here played with Digium GUI and got the zaptel components to work? |
| 04:36.14 | BBhoss | whats the most friendly codec to high latency or pakcet loss |
| 04:37.54 | sevard | Strom_M: somebody needs to beat him with the idiot stick. |
| 04:39.58 | joebob777as7 | I'm pretty sure the polycom 501's will be too expensive for my starting office... anything cheaper that'll work well with asterisk? |
| 04:40.00 | Trevor_b | sevard: glad to know people didnt loose the im so bold behind my keyboard mentality. |
| 04:40.21 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:40.38 | sevard | I'm bold while naked in k-mart, what does that prove? |
| 04:42.01 | sevard | <PROTECTED> |
| 04:42.40 | Trevor_b | not much, who wouldnt be bold inside a kmart |
| 04:43.01 | Trevor_b | not like those peopple should intimidate or anything, such power wielding! |
| 04:44.04 | ohadz | Strom_M, what does this mean -- Make sure you set a valid, 10-digit US48 CallerID value for this SIP device. where do i set that number? on my sip phone web interface? |
| 04:44.46 | Strom_M | brb, on phone |
| 04:46.52 | Strom_M | ok |
| 04:47.20 | Strom_M | you can either set that in the phone (pain in the ass), in sip.conf (easy), or on a per-call basis in extensions.conf (also easy) |
| 04:47.48 | ohadz | Strom_M, how do i set it up in sip.conf or extension.conf? |
| 04:47.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:48.41 | Strom_M | in sip.conf: callerid=Name Here<3115552368> |
| 04:48.45 | Strom_M | er |
| 04:48.50 | Strom_M | in sip.conf: callerid="Name Here"<3115552368> |
| 04:49.09 | Strom_M | in extensions.conf, you'd use Set() in conjunction with the CALLERID() function |
| 04:49.54 | Trevor_b | anyone here run zapscan without it 'Skipping Zap Scanning'? |
| 04:50.41 | ohadz | Strom_M, thanks! in the sip.conf - i'll include it under the ext block or under the voip provider block? |
| 04:50.58 | Strom_M | sip phone |
| 04:51.10 | ohadz | Strom_M, thanks! |
| 04:51.12 | *** part/#asterisk BBhoss (n=hoss@adsl-230-28-185.hsv.bellsouth.net) |
| 04:52.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:52.47 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 04:54.05 | ohadz | Strom_M, two questions -- 1. i am trying to setup nufone. but when i dial the outside number i get a fast bz signal rather then the call i dialed 12122222111 |
| 04:54.28 | Strom_M | ohadz: what does the console say |
| 04:54.33 | ohadz | Got SIP response 603 "Declined" back from 66.225.202.76 -- SIP/NuFone-0821aea0 is busy |
| 04:54.39 | ohadz | could nufone really b bz? |
| 04:54.55 | Strom_M | they have IAX2 you know |
| 04:55.44 | ohadz | Strom_M, yes..is there a different way i should do this? |
| 04:55.49 | ido | can asterisk handle ×™×•× ×™×§×•×“? |
| 04:56.24 | aptura | hebrew? |
| 04:56.43 | ohadz | Strom_M, i chose the Asterisk using SIP option |
| 04:57.09 | ido | unicode, actually, but ostensibly hebrew |
| 04:57.24 | ido | (that was unicode spelled in hebrew) |
| 04:57.40 | aptura | I should goto isreal some time. |
| 04:57.54 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.147.43) |
| 04:58.47 | Strom_M | is that different from isfake? |
| 04:58.54 | Strom_M | or did you mean israel? |
| 04:59.02 | CCFL_Man2 | does cisco sip firmware support multiple ringtones? |
| 04:59.06 | ohadz | Strom_M, should i've used a different method of connecting? |
| 04:59.08 | CCFL_Man2 | does anyone here know |
| 04:59.33 | Strom_M | ohadz: i think you should try calling a known working number |
| 04:59.44 | Strom_M | 213-853-1212 works nicely |
| 05:00.31 | irule | how can I have the pbs tell me a number? |
| 05:00.32 | ohadz | Strom_M, this really sucks -- Called 12138531212@NuFone |
| 05:00.32 | ohadz | <PROTECTED> |
| 05:00.33 | ohadz | <PROTECTED> |
| 05:00.33 | ohadz | <PROTECTED> |
| 05:00.36 | ohadz | oops... |
| 05:00.43 | ohadz | sorry for the flood.. |
| 05:00.48 | irule | 1.4 saynumber(500) or something? |
| 05:01.51 | hacim | I set the "M" flag on Meetme() and I see in the console it saying, "Started music on hold, class 'default'" but nothing plays and almost immediately after, "Stopped music on hold " but I dont know why? |
| 05:02.19 | sevard | Great, another skunk. |
| 05:02.36 | CBU[^_^]M`` | hello... can i use SPA 3102 on asteriks? |
| 05:02.43 | ido | yes aptura |
| 05:02.45 | ido | you should |
| 05:03.55 | hacim | another skunk? |
| 05:04.51 | sevard | hacim: it's summer in minnesota, and the skunks come out. |
| 05:05.00 | sevard | you can smell that crap inside your house |
| 05:05.13 | hacim | sevard: oh yeah, I used to live there :) |
| 05:06.51 | sevard | hacim: where? |
| 05:07.57 | ido | scavhunt.uchicago.edu |
| 05:09.28 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:10.02 | hacim | sevard: somewhere halfway between duluth and minneapolis |
| 05:10.47 | *** join/#asterisk IronMan2000 (n=kent@r12h8.dixie-net.com) |
| 05:11.11 | hacim | anyone know where meetme looks for musiconhold files to play? |
| 05:11.55 | IronMan2000 | ff |
| 05:12.27 | IronMan2000 | I have a blacklist in my extenstions.conf to block unwanted callers to my office. Example: |
| 05:12.31 | IronMan2000 | exten => _X./330659XXXX,1,Goto(default,3000,1) ;RMS Collections |
| 05:12.31 | IronMan2000 | exten => _X./8006666994,1,Goto(default,3000,1) ;RMS Collections |
| 05:12.32 | sevard | hacim: neat, i like the north shore area |
| 05:12.43 | IronMan2000 | but sometimes callers block their caller Id |
| 05:12.49 | IronMan2000 | how can I block these as well? |
| 05:14.19 | IronMan2000 | if I were to put "n/a" in place of the number, would asterisk block anyone that calls and their caller Id is blocked? |
| 05:14.55 | Strom_M | GotoIf($["${CALLERID(num)}" = ""]?blocked) |
| 05:16.09 | Strom_M | or somesuch |
| 05:16.09 | blitzrage | damn... I love when you spent an hour writing up a large dialplan that it practically works first try |
| 05:16.10 | CCFL_Man2 | ahh shit |
| 05:16.10 | CCFL_Man2 | in cisco v1 firmware you need the tftp server and dhcp server the same address |
| 05:17.08 | *** part/#asterisk IronMan2000 (n=kent@r12h8.dixie-net.com) |
| 05:17.53 | blitzrage | http://www.pastebin.ca/480242 |
| 05:18.07 | blitzrage | ringall strategy implemented into my huntgroup logic! |
| 05:19.21 | blitzrage | I think I should write a cookbook... :) |
| 05:21.49 | *** join/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:22.05 | asteriskguy | hello |
| 05:23.10 | aptura | blitz want to see what my life was like in the late 80s when I started my working carrer? http://www.youtube.com/watch?v=gOFF4MEB1_E I was a crewchief on these. |
| 05:24.14 | aptura | Alot more exciting then the IT world :) |
| 05:25.32 | CCFL_Man2 | i can't get my cisco ip phone to change the tftp ip address |
| 05:25.39 | CCFL_Man2 | i cannot upgrade the phone |
| 05:26.35 | CCFL_Man2 | wtf |
| 05:26.40 | CCFL_Man2 | i'm styuck |
| 05:27.50 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 05:29.27 | DarKnesS_WolF | guys i have a quesion .. now i want to integrate asterisk with a panasionc TDA100 PBX.. the idea is i want to recorde all the calls incoming and outgoing .. using analog interfaces what will be the best senairo ? |
| 05:30.07 | DarKnesS_WolF | what i had in mind 8 ports in the asterisk server ... 4 for trunks " pstn " and 4 to connect to the PBX .. so the analog phones can call using the asterisk machine and all phones get recoreded.. |
| 05:33.00 | CCFL_Man2 | dammit the hell |
| 05:36.32 | DarKnesS_WolF | CCFL_Man2: what ? |
| 05:40.57 | CCFL_Man2 | DarKnesS_WolF: i downgraded to sip firmware 1.01 for my cisco 7912 phone and can't set the tftp server address now |
| 05:41.18 | DarKnesS_WolF | :-s |
| 05:42.02 | hacim | someone from brasil called my asterisk server, the echo test was fine, but when they connected to the meetme, it was terrible |
| 05:42.22 | DarKnesS_WolF | hacim: do u have a timing source? |
| 05:46.01 | FuriousGeorge | i set up a sangoma a200 today for the 1st time, and i did the 2fxo2fxs backwards of what they were on my tdm400p |
| 05:46.11 | FuriousGeorge | at least relative to my dialplan |
| 05:47.51 | FuriousGeorge | took me 2 hours to figure out what i was doing wrong. asterisk started and didnt complain, then by chance i plugged one of the fxo into the wall (to see if incoming calls worked), so next time i tried to call i got a dial tone and it clicked |
| 05:48.01 | FuriousGeorge | my brain, not the dial tone |
| 05:48.12 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:48.14 | *** part/#asterisk asteriskguy (n=learnast@cpe-75-80-111-113.socal.res.rr.com) |
| 05:51.04 | hacim | DarKnesS_WolF: I've got ztdummy |
| 05:53.31 | aptura | how did the sangoma work out? |
| 05:58.04 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 05:59.50 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 06:09.39 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.43) |
| 06:11.20 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:13.24 | JerJer | what is wrong with: exten => _XXXX,n,GotoIf($["${pin}" != "${pass}"]?30:40) ??? |
| 06:13.55 | JerJer | if i NoOP(${pin}) i get the expected value |
| 06:13.56 | *** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net) |
| 06:14.11 | JerJer | but in the GotoIf it is 0 |
| 06:16.09 | blitzrage | Verbose(1|${pin} == ${pass}) ? |
| 06:16.32 | blitzrage | I don't see anything syntactically wrong |
| 06:18.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:26.37 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 06:26.49 | blitzrage | hrmmm... I should probably go to bed.... |
| 06:27.12 | *** join/#asterisk primatus2 (n=solapus@66.90.121.129) |
| 06:27.21 | blitzrage | ya, I might as well do that -- night all! |
| 06:28.02 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 06:28.02 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:29.30 | DarKnesS_WolF | what i had in mind 8 ports in the asterisk server ... 4 for trunks " pstn " and 4 to connect to the PBX .. so the analog phones can call using the asterisk machine and all phones get recoreded.. |
| 06:29.43 | DarKnesS_WolF | guys i have a quesion .. now i want to integrate asterisk with a panasionc TDA100 PBX.. the idea is i want to recorde all the calls incoming and outgoing .. using analog interfaces what will be the best senairo ? |
| 06:29.47 | DarKnesS_WolF | what i had in mind 8 ports in the asterisk server ... 4 for trunks " pstn " and 4 to connect to the PBX .. so the analog phones can call using the asterisk machine and all phones get recoreded.. |
| 06:29.48 | *** join/#asterisk oej (n=olle@213.115.215.55) |
| 06:29.51 | DarKnesS_WolF | what do u think guys ? |
| 06:33.33 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 06:37.00 | *** join/#asterisk chrismac (n=chris@87-194-75-247.bethere.co.uk) |
| 06:48.14 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
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| 06:51.19 | *** join/#asterisk reydasmarinas (n=rbdasmar@213.184.163.194) |
| 06:56.54 | *** join/#asterisk my007ms (n=my007ms@217.139.224.221) |
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| 07:04.50 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 07:05.02 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 07:08.58 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:10.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:11.18 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
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| 07:17.35 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 07:17.39 | zeeesh | hi |
| 07:17.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:18.58 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 07:18.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:25.08 | *** join/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 07:25.49 | SoloFlyer | anyone noticed if kb1ec is much better than mec2? |
| 07:29.16 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 07:33.43 | *** join/#asterisk lorinc (n=ang@pool-4344.adsl.interware.hu) |
| 07:40.06 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 07:42.21 | *** join/#asterisk Arc_Tangent (n=Arc_Tang@connerdata.plus.com) |
| 07:42.22 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:43.15 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 07:46.20 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 07:48.53 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:53.02 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 07:58.49 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:02.57 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 08:03.01 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 08:03.03 | Nobbie | heya = |
| 08:03.18 | Nobbie | who here runs their SIP Trunks with keepalive=yes ? |
| 08:05.13 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:05.15 | *** join/#asterisk agreen (n=agreen@cpc1-nthc5-0-0-cust289.nrth.cable.ntl.com) |
| 08:22.24 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:22.54 | *** part/#asterisk agreen (n=agreen@cpc1-nthc5-0-0-cust289.nrth.cable.ntl.com) |
| 08:31.26 | FreezeS | hey guys |
| 08:31.54 | FreezeS | is it possible to have a generic user in sip.conf ? |
| 08:32.27 | FreezeS | like 009999XXXX so everyone can register with an extension that starts with 009999 |
| 08:32.42 | FreezeS | or in realtime |
| 08:36.10 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:42.54 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 08:43.42 | Aces1Up | hey all i'm have some fun with the asterisk api manager, and am trying to debug a script of mine. how to i turn on events and or debug messages for the api manager to see if the requestes i'm making of it are even getting to the asterisk box? |
| 08:49.50 | Nobbie | FreezeS: i'm guessing no |
| 08:50.01 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 08:50.02 | MrWup | hey |
| 08:50.11 | MrWup | having problems with asterisk dialling out |
| 08:50.20 | MrWup | it dials but doesnt register the call as answered |
| 08:50.26 | MrWup | dialling out on normal pots line to the pstn |
| 08:50.29 | MrWup | through 4fxo card |
| 08:50.36 | MrWup | hangups seem to be registered fine |
| 08:50.38 | MrWup | but answers dont |
| 08:50.46 | MrWup | im assuming that my line might not have answering attendant? |
| 08:52.16 | Nobbie | MrWup: you mean incoming calls aren't being processed ? |
| 08:53.31 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:54.16 | MrWup | nope |
| 08:54.20 | MrWup | incoming calls are fine |
| 08:54.48 | MrWup | sip phone -> asterisk -> digium analogue FXO card -> PSTN |
| 08:55.09 | Nobbie | so the Dial() works ? |
| 08:55.18 | MrWup | when i make a call from sip phone to a normal PSTN number, asterisk says dialling... the call is connected, but asterisk never registers that the PSTN user has picked up the call |
| 08:55.21 | MrWup | yeah dial works |
| 08:55.25 | MrWup | the call is even connected |
| 08:55.31 | MrWup | but then it drops after 20 seconds |
| 08:55.35 | MrWup | cause the timeout triggers |
| 08:55.40 | MrWup | cause it thinks the call hasnt been answered |
| 08:55.41 | Nobbie | BRI or PRI ? |
| 08:55.47 | MrWup | no its not isdn |
| 08:55.52 | MrWup | its normal analogue pots line |
| 08:55.54 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:56.00 | MrWup | im assuming this has something to do with answering attendant |
| 08:57.09 | Nobbie | your ringcadence selection using country= in indications.conf may be wrong ? |
| 08:57.44 | MrWup | well hangups seem to be detected fine |
| 08:57.51 | MrWup | and indications.conf is definitely UK as it should be |
| 09:00.16 | Nobbie | *shrug* |
| 09:02.47 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 09:07.23 | redax | hi, |
| 09:11.05 | redax | how can I implement Call forward on response 302 "Moved Temporary" ? |
| 09:11.11 | *** join/#asterisk kippi (n=cowen@host81-154-165-233.range81-154.btcentralplus.com) |
| 09:11.14 | kippi | hey |
| 09:12.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:12.33 | sumasuma | redax, you have SIP Programming experience ? |
| 09:13.04 | kippi | I am having a problem with asterisk, when I do the echo test very thing sounds fine, but when I have a call from outside across the ISDN it sounds fine, but it sounds really bitty, as if its losing packets, has anyone got anyideas? |
| 09:14.14 | redax | sumasuma: nothing serious |
| 09:14.22 | redax | just learning... |
| 09:15.59 | sumasuma | redax, did asterisk supported 302 on sip ?? |
| 09:17.39 | redax | chan_sip.c handles 302 responses |
| 09:19.37 | sumasuma | redax, you can go from there, what is the problem ? |
| 09:20.48 | redax | on linksys spa922 if you press DnD; it will reply 302;. |
| 09:21.07 | kaldemar | sumasuma: take a look at sip parameter promiscredir. |
| 09:21.14 | redax | and I'd like to use it as a fast way to forward the calls to my mobile... |
| 09:23.38 | redax | but according to chan_sip.c it will result BUSY |
| 09:27.09 | *** part/#asterisk s-ndh-c (n=michi@85.93.11.18) |
| 09:38.24 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 09:40.14 | Zeeek | a chilling silence ensued |
| 09:43.29 | Zeeek | lunch is near |
| 09:43.43 | Zeeek | microwave is warming up |
| 09:44.19 | Zeeek | install the asterisk software on microwave so I can get calls during lunch |
| 09:46.09 | *** join/#asterisk agile (n=mike@68-189-222-201.dhcp.ftwo.tx.charter.com) |
| 09:48.43 | Zeeek | hunger approaches |
| 09:55.01 | Zeeek | Tomorrow at 12:30 PM EDT you can talk to other asterisk users by following the instructions at http://x2z.eu |
| 09:55.09 | Zeeek | ~AUC |
| 09:55.45 | jbot | hmm... auc is Asterisk Users Conference, Friday at 12:30 PM Eastern Time |
| 09:55.58 | Zeeek | ~AUC Fridays at 12:30 PM EDT you can talk to other asterisk users by following the instructions at http://x2z.eu |
| 09:56.10 | Zeeek | ~AUC |
| 09:56.25 | jbot | auc is, like, Asterisk Users Conference, Friday at 12:30 PM Eastern Time |
| 09:56.50 | Zeeek | what is this, a Jack Keroac imitation? |
| 10:00.06 | sergee | can anybody share a link to some manual or howto, describing how to enable T.38 in scheme linksys 2102 <-> * <-> linksys 2102 ? |
| 10:00.06 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:02.45 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:08.39 | mosty | i'm trying to figure out how to mass update the firmware on a network of snom phones of various models, and i can't figure out how to separate them (each model needs to load a different firmware settings file), anyone figured that out already? |
| 10:11.47 | Zeeek | mosty isn't that done by Mac number in the file name? |
| 10:12.45 | mosty | zeeek: i can identify individual phones by the mac address (which i do), but i don't have a list mappings of mac -> model |
| 10:14.33 | *** join/#asterisk dseeb_ (n=dcb@124.177.38.240) |
| 10:14.57 | *** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za) |
| 10:19.25 | sergee | [May 10 14:18:28] ERROR[27440]: chan_sip.c:12559 handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled. |
| 10:19.32 | sergee | what does it mean? |
| 10:20.43 | mosty | sergee, asterisk only supports t38 in passthrough mode to my knowledge, are you aware of that? |
| 10:21.08 | Zeeek | ~T.38 |
| 10:21.22 | jbot | i guess t.38 is see t38 |
| 10:21.22 | sergee | mosty: yes, i'm trying to pass fax between 2 sipuras |
| 10:21.33 | hads | ~t38 |
| 10:22.04 | jbot | i heard t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 10:22.05 | Zeeek | thanks, my hands were covered in carrot purée |
| 10:22.13 | hads | jbot must be sleepy |
| 10:22.44 | Zeeek | jbot really should answer "too lazy to google?" to every query |
| 10:22.59 | hads | :) |
| 10:23.28 | Zeeek | ~jbot is a lazy software construct from a past millenium |
| 10:23.51 | Nobbie | When a RTP stream is setup between * and a SIP handset, and the handset is rebooted without sending SIP BYE, the RTP Stream remains active. Isn't there a mechanism to detect that stale RTP stream and close it ? |
| 10:24.37 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:24.42 | skirmisha | guys |
| 10:26.07 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583012.dsl.bell.ca) |
| 10:26.39 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:26.43 | skirmisha | back again |
| 10:29.22 | Zeeek | welcome back! |
| 10:29.31 | Zeeek | nice trip? |
| 10:30.06 | skirmisha | hehe |
| 10:30.22 | skirmisha | how many concurrent calls 1 asterisk can handle |
| 10:30.31 | hads | 49 |
| 10:31.01 | Zeeek | wrong. 43 |
| 10:31.29 | hads | Sorry, 49 must be planned for 1.6 |
| 10:31.30 | kaldemar | 42? |
| 10:31.50 | Zeeek | 43 plus one zombie |
| 10:32.06 | Zeeek | 43 including one zombie |
| 10:32.22 | skirmisha | is this a joke? |
| 10:32.26 | kaldemar | mine does only ~10 on a wrt54g. |
| 10:32.59 | mosty | skirmisha, you didn't give enough info to get a good answer |
| 10:33.02 | HarryR | skirmisha, generally you can push it upto ~400+ |
| 10:33.14 | HarryR | but it's a how long is a piece of string question :) |
| 10:35.11 | skirmisha | 400 sounds good |
| 10:35.30 | skirmisha | HarryR any software that can do load balancing |
| 10:35.43 | skirmisha | or hardware |
| 10:35.47 | Zeeek | http://www.urbandictionary.com/define.php?term=meaning+of+life&defid=1115478 |
| 10:36.10 | HarryR | generally frontend it with ser and use a Linux VServer based load balancing package like RedHat's Piranha |
| 10:36.48 | skirmisha | let me check |
| 10:37.02 | HarryR | what kind of machines are you putting Asterisk on? |
| 10:37.34 | skirmisha | hp dual xeon |
| 10:37.50 | HarryR | with hyperthreading, fast disks & lots of memory? |
| 10:37.54 | skirmisha | yes |
| 10:38.10 | HarryR | should be able to handle 400 fairly easily then |
| 10:38.21 | HarryR | although I don't really trust astersk not to segfault on it'self |
| 10:38.44 | skirmisha | so u prefer ser/openser |
| 10:38.54 | HarryR | Ser isn't a PBX :) |
| 10:39.04 | skirmisha | i know |
| 10:39.12 | skirmisha | i don't need pbx |
| 10:39.21 | skirmisha | but i don;t think ser can handle rtp |
| 10:39.21 | HarryR | oh well then use OpenSER |
| 10:39.28 | HarryR | OpenSER can with mediaproxy |
| 10:39.32 | HarryR | but it can't do transcoding |
| 10:39.36 | hads | So you are asking about Asterisk why? |
| 10:39.54 | skirmisha | because i plan to use asterisk |
| 10:40.05 | skirmisha | i know openser can handle 10k calls |
| 10:40.16 | skirmisha | but it is hard to configure it |
| 10:40.20 | hads | But you just said you don't need a PBX? |
| 10:40.26 | skirmisha | with all these c/c++ style of config |
| 10:40.41 | HarryR | so you're severly crippling your service by using Asterisk instead of OpenSER for SIP routing/registration? |
| 10:40.56 | skirmisha | there is no registration |
| 10:40.58 | skirmisha | only routing |
| 10:41.06 | mosty | skirmisha, is it sip only? |
| 10:41.28 | skirmisha | yes sip only, but not so sure |
| 10:41.50 | HarryR | you might consider looking at YATE |
| 10:42.00 | skirmisha | i mean sip for now, but for future we can use iax also |
| 10:42.25 | mosty | skirmisha, dealing with phones or pbxs? |
| 10:42.29 | skirmisha | most important thing for me to do load balance |
| 10:42.43 | skirmisha | mosty just pure proxy |
| 10:42.52 | HarryR | no transcoding at all? |
| 10:44.07 | HarryR | i'd still say go with OpenSER |
| 10:44.17 | mosty | i agree with harryr |
| 10:44.23 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 10:44.53 | HarryR | Asterisk is great for PBXs and custom applications (e.g. the heavy part of your infrastructure) but really sucks for bulk sip proxying |
| 10:45.13 | skirmisha | i tried openser could not make it to do simple routing, probably config was wrong |
| 10:45.24 | hads | probably |
| 10:45.30 | HarryR | you need to do dynamic routing or least cost routing? |
| 10:45.42 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:45.50 | *** part/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 10:46.01 | skirmisha | probably least cost routing |
| 10:46.09 | skirmisha | but then i need openser with mysql |
| 10:46.31 | skirmisha | what about yate |
| 10:46.46 | skirmisha | anyone tried openser with load balancing |
| 10:47.13 | HarryR | you'd rarely need load balancing other than basic roundrobin dns with openser |
| 10:48.09 | skirmisha | is dns rounrobin stable enough |
| 10:48.14 | HarryR | yes |
| 10:48.26 | HarryR | if it wasn't, the whole internet would fail instantly |
| 10:48.31 | HarryR | :) |
| 10:48.46 | HarryR | well not literally, but it'd cause some serious issues |
| 10:48.51 | skirmisha | but wasn't only for registrations or also for calls |
| 10:49.07 | HarryR | for round robin dns, it should work with both |
| 10:49.31 | HarryR | because each time a connection is made (and it resolves the ip address of the domain name) you'll get a different ip address in your cluster |
| 10:50.06 | skirmisha | i need to test this |
| 10:50.17 | skirmisha | but then i can use asterisk behind dns |
| 10:50.46 | HarryR | if you want, yes |
| 10:51.35 | skirmisha | another issue is radius, most probably i'll need to make sip proxy to work with radius |
| 10:51.45 | HarryR | why do you need radius? |
| 10:51.51 | jacq | skirmisha: tried sipwise.com? |
| 10:52.14 | mosty | skirmisha, though you said no registrations? |
| 10:52.27 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 10:52.31 | skirmisha | i need radius for accounting |
| 10:52.35 | skirmisha | and auth calls |
| 10:52.36 | Stephnie | hello |
| 10:52.44 | skirmisha | but no registrations |
| 10:52.55 | HarryR | cant you just use a RDBMS like MySQL or PostgreSQL for everything like _normal_ people? |
| 10:52.56 | Stephnie | does asterisk support SIP/SIMPLE protocol for instant messaging? |
| 10:53.06 | HarryR | e.g. personally i've found that Radius is way too restricting |
| 10:53.18 | jacq | i agree with the HarryR |
| 10:53.24 | skirmisha | i see |
| 10:53.35 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 10:53.35 | jacq | mysql backend for Openser and Asterisk cluster , much easier to deploy and maintain |
| 10:54.12 | HarryR | e.g. we built our own rating system which bills customers per-minute based on custom criteria - all being done live as the call is in progress |
| 10:54.34 | mosty | postgresql is better than mysql in terms of licensing and consistency |
| 10:54.46 | HarryR | yes :) |
| 10:55.01 | Stephnie | jacq: openser or ser supports media? I think it is only for signalling...right? |
| 10:55.03 | HarryR | but MySQL and NDB kicks it's ass for high-update stuff |
| 10:55.58 | mosty | i've never used ndb, postgresql can handle more connections than mysql- i haven't hit its limits yet |
| 10:56.09 | hads | Postgres has got some serious multi CPU performance lately. |
| 10:56.48 | HarryR | yah agreed, but with NDB we're talking about clustering over 4-16 machines :\ |
| 10:57.08 | jacq | Stephnie: you can run rtp proxy or media proxy |
| 10:57.31 | jacq | Stephnie: but only to relay packets (to bypass NAT for example) |
| 10:57.46 | jacq | Stephnie: openser is not a back to back UA like Asterisk |
| 10:57.51 | skirmisha | so openser + DNS SRV + mysql/postgre will do my job |
| 10:57.55 | Stephnie | I see... |
| 10:57.59 | hads | http://tweakers.net/reviews/657/5 |
| 10:58.19 | Stephnie | whats the difference between Openser or ser? and which is better? |
| 10:58.33 | jacq | does PostgreSQL supports replication natively? |
| 10:58.40 | mosty | Stephnie, openser is a fork of ser |
| 10:58.56 | jacq | we found openser more up to date, but thats our opinion |
| 10:59.03 | mosty | jacq, i believe there are addon packages for it, i've never investigated them though |
| 10:59.51 | Stephnie | do they support SIP/SIMPLE protocol for instant messaging? |
| 10:59.59 | jacq | i havent looked back at SER for more than a year now honestly ;) |
| 11:00.17 | jacq | i know openser supports jabber almost natively, through jabber module, not sure about SIMPLE. |
| 11:02.29 | Stephnie | what about asterisk? |
| 11:02.36 | Stephnie | does it support SIMPLE? |
| 11:03.11 | mosty | Stephnie, sounds like a job for google |
| 11:03.15 | skirmisha | is there some free billing system for asterisk or openser |
| 11:03.41 | jacq | skirmisha: tons for asterisk |
| 11:04.36 | skirmisha | jacq i know but need some that is already tested and works so so |
| 11:05.24 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 11:05.53 | jacq | i run mine on openser, so i'm not the best one to ask for specifics |
| 11:06.54 | skirmisha | tell me for openser |
| 11:06.58 | skirmisha | what do u run |
| 11:07.19 | jacq | in house module developped around the bb module of openser |
| 11:07.43 | skirmisha | i see |
| 11:07.45 | Zeeek | bjr |
| 11:07.50 | skirmisha | so it is your own develop |
| 11:08.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:09.22 | puzzled | hi |
| 11:12.43 | skirmisha | hi |
| 11:13.33 | HarryR | most people either develop their own, buy one of the big commercial billing systems, or get something like AstBill and customize it lots |
| 11:19.49 | Dr-Linux | Cepstral always crashes my server |
| 11:20.13 | Dr-Linux | i installed cepstral on different asterisk servers, but same happend :S |
| 11:20.23 | Dr-Linux | any clue? |
| 11:24.08 | sergee | seems like nobody managed to use t38 at all? |
| 11:28.40 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:28.54 | drako | Im having problem when transfering calls |
| 11:29.02 | drako | it hangs up |
| 11:29.15 | drako | May 10 13:27:41 WARNING[20950]: res_monitor.c:319 ast_monitor_change_fname: Cannot change monitor filename of channel Zap/3-1 to null |
| 11:29.26 | drako | <PROTECTED> |
| 11:29.26 | drako | <PROTECTED> |
| 11:29.36 | phearless | hello, is it possible to transfert a call via the "asterisk manager"? For example, I receive a call, and I use a software/webpage connected to the asterisk manager, and I tranfert the call to somebody in one click. |
| 11:36.01 | phearless | guys? |
| 11:37.40 | mosty | yes |
| 11:38.48 | skirmisha | anyone tried cisco sip proxy? |
| 11:41.30 | phearless | mosty ? |
| 11:41.50 | mosty | phearless ? |
| 11:41.58 | phearless | are you answering me ? |
| 11:42.03 | mosty | yes |
| 11:42.53 | phearless | I have to use "Redirect" ? |
| 11:43.49 | phearless | this post says that ZapTransfer does not work http://lists.digium.com/pipermail/asterisk-dev/2006-July/021925.html |
| 11:44.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:44.50 | mosty | http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect |
| 11:45.13 | DrukenLPY | anyone use sip between asterisk servers? |
| 11:46.11 | penguinFunk | yep |
| 11:46.40 | DrukenLPY | i'm havin some auth issues... and my brain can't get around it.... |
| 11:47.22 | DrukenLPY | i have a client with a <shudders> a@h box... he registers to me fine, and has outgoing calls.... |
| 11:48.08 | DrukenLPY | but when i try to send him calls, i get an error, chan_sip.c:9761 handle_response_invite: Forbidden - wrong password on authentication for INVITE to |
| 11:49.09 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 11:49.19 | mosty | sounds like the a@h box is requiring authentication |
| 11:49.53 | DrukenLPY | yeah, but why? afaik it shouldn't be requiring auth |
| 11:52.09 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 11:53.34 | mosty | what type is the sip entity? |
| 11:56.12 | DrukenLPY | user/peer combo |
| 11:56.46 | jacq | do you know what setting the A@H box has for its user ? |
| 11:56.59 | jacq | user trunk, err |
| 11:57.23 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 11:58.08 | DrukenLPY | ?? no idea lost me there |
| 11:58.26 | chrismac | any idea how to disable logging CDR data to .csv files. Don't need (want) it as I am using cdr_mysql? |
| 11:59.09 | mosty | disable cdr_csv.so in modules.conf |
| 11:59.47 | jacq | if you send from your box peer to the A@H user, and A@H has auth info in its etting, you will need those in your peer. |
| 12:00.02 | jacq | let me get you a Ast to Ast via SIP howto link. |
| 12:00.23 | defswork | I have my mobile phone (n95) connected to my asterisk box at home via wifi/VPN. I can dial out as an extension from it but for incoming calls on the PTSN side that are call grouped to ring the extension of the mobile return -- SIP/220-081d12b0 is circuit-busy - any pointers ? |
| 12:01.01 | DrukenLPY | jacq: i've done ast to ast sip without a problem... i just don't know what the peice of shit a@h box is doing.. hehehe |
| 12:01.16 | chrismac | ahh, good point - thanks for help |
| 12:01.32 | chrismac | so all I need is cdr_addon_mysql.so only? |
| 12:02.18 | chrismac | or app_cdr.so + cdr_addon_mysql.so ? |
| 12:03.07 | *** join/#asterisk doolittlework (n=davidh@196.209.244.243) |
| 12:03.09 | doolittlework | hi there |
| 12:03.27 | jacq | DrukenLPY: you have a peer declared in SIP that sends traffic to the A@H right? |
| 12:04.07 | doolittlework | need some help with trixbox fax please |
| 12:05.48 | doolittlework | i can receive the fax no problem but it gets converted to pdf before it mails the fax, in my /var/spool/asterisk/fax folder it receives the fax as tif format i need it remain a tif file when it gets emailed. |
| 12:06.29 | DrukenLPY | jacq: the a@h box registers to my server, and i dial it by name |
| 12:07.07 | mosty | doolittlework, see the topic |
| 12:08.46 | jacq | DrukenLPY: in your sip.conf you should have a user entry for your customer with A@H |
| 12:09.11 | doolittlework | thx sorry |
| 12:09.14 | jacq | DrukenLPY: in that entry, do you specify fromuser, username and secret |
| 12:09.47 | DrukenLPY | fromuser no, but username and secret of course... |
| 12:10.39 | skirmisha | guys is there radius client for asterisk released from portaone |
| 12:10.44 | skirmisha | for ver 1.4 |
| 12:12.36 | defswork | this is odd - I can dial the SIP/220 extension ok from manager - just not getting there when it's on a ring group |
| 12:13.16 | jacq | fromuser, changes your from tag in sip invite, if i remember correctly... try it out, but ultimately, you have to adjust your user entry according to your customer config |
| 12:15.35 | *** join/#asterisk ManxPower (n=manxpowe@145.sub-70-216-67.myvzw.com) |
| 12:15.49 | ManxPower | 'morning peeps |
| 12:16.46 | DrukenLPY | argh... |
| 12:17.57 | *** join/#asterisk grEvenX (n=even@ti500720a080-7918.bb.online.no) |
| 12:18.58 | ManxPower | DrukenLPY: Your problem is pretty obvious. |
| 12:19.37 | DrukenLPY | aside that it's a@h? :) |
| 12:19.41 | DrukenLPY | peice of shit... |
| 12:20.02 | ManxPower | DrukenLPY: then why are you using it? |
| 12:20.21 | DrukenLPY | I'M NOT using it.... a customer is.... |
| 12:20.42 | DrukenLPY | i'm just trying to figure out why the peice of crap won't accept calls |
| 12:21.09 | DrukenLPY | all the other users and peers on my system work fine.... just his... |
| 12:21.24 | ManxPower | Ah. |
| 12:21.32 | ManxPower | I just don't accept customers that are stupid. |
| 12:22.21 | DrukenLPY | yeah i'm real close to telling the guy that it's not my problem and either he figures it out on his own, or well... i'm still charging ya for the service :) |
| 12:23.10 | penguinFunk | a@h ? |
| 12:23.44 | DrukenLPY | asterisk at home/freepbx/trixbox |
| 12:23.45 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@85-18-66-28.ip.fastwebnet.it) |
| 12:23.57 | d4rkst4r75 | hi |
| 12:24.19 | d4rkst4r75 | someone can tell me what kind of pciexpress slot is required for Sangom A200X |
| 12:24.20 | d4rkst4r75 | = |
| 12:24.22 | d4rkst4r75 | ? |
| 12:24.53 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:27.35 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-17-37.w81-251.abo.wanadoo.fr) |
| 12:28.08 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:31.28 | ManxPower | d4rkst4r75: I guess I could go to sangoma.com and read the specs, but I usually require dinner and drinks before holding someone's hands. |
| 12:31.37 | ManxPower | And cash of course. |
| 12:32.00 | defswork | d4rkst4r75: I hard the same query once :) |
| 12:32.26 | d4rkst4r75 | it should be pci-express x1 |
| 12:32.45 | defswork | d4rkst4r75: there are 2 models |
| 12:32.53 | defswork | PCIx and PCI Express |
| 12:33.55 | defswork | http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html?osCsid=69fb74c7eeb71997e51de526e47ff12f |
| 12:37.46 | DrukenLPY | uhg.... |
| 12:37.51 | DrukenLPY | i finally got it working |
| 12:38.52 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 12:40.58 | ManxPower | "What's a nice geek like you doing in a place like this?" |
| 12:42.46 | DrukenLPY | just sittin around, watching the text scroll.... and scaring off any potential newbs :) |
| 12:43.22 | phearless | <mosty> http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect <--- it is for a meetup, not for a transfer, right ? |
| 12:43.44 | *** join/#asterisk oej (n=olle@213.115.215.55) |
| 12:51.25 | shido6 | coming this way? |
| 12:51.26 | shido6 | :) |
| 12:51.54 | ManxPower | shido6: Yes, actually. |
| 12:52.03 | ManxPower | but not very far your way. |
| 12:52.11 | ManxPower | New Orleans -> Birmingham |
| 12:52.17 | shido6 | ahh. |
| 12:52.21 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-161-200.red.bezeqint.net) |
| 12:52.48 | Dovid | hello ev1. just got a polycom 501 (only used 601 b4). dont see any settings in there to grab new configs via ftp. am i missing something ? |
| 12:52.58 | *** join/#asterisk ocgltd (n=support@CPE004063e0ee74-CM00159a010632.cpe.net.cable.rogers.com) |
| 12:53.15 | Dovid | nm |
| 12:53.18 | Dovid | i found it |
| 12:54.23 | ocgltd | Does anyone know if Asterisk uses UDP port 3000 ? In diagnosing a remote SIP phone problem we are being told by our ISP that our asterisk box is sending 1000 UDP packets/second on port 3000. (This seems like neither SIP nor RTP). |
| 12:54.38 | ManxPower | ocgltd: no, it does not. |
| 12:54.45 | ManxPower | unless you specify it in some config file |
| 12:57.11 | *** join/#asterisk ltdwk- (n=z@203-173-10-9.perm.iinet.net.au) |
| 12:58.21 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 12:58.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 12:59.51 | ManxPower | MY cell signal will drop in about 30 mins and will be unavail until about 2pm DET |
| 12:59.56 | ManxPower | ..er...CDT |
| 13:01.56 | LeddyHM | CDT > * |
| 13:03.13 | iCEBrkr | Can't it be Friday?!?!@#@#@#% |
| 13:03.40 | ManxPower | Central Daylight Time (USA) |
| 13:04.01 | ocgltd | Another question (another system) - my Asterisk box appears to register with the ITSIP, but I regularly see "REGISTER attempt 1 to 999@pbx.itsp.com" and "REGISTER attempt 2 to 999@pbx.itsp.com". Why does this happen? (I haven't seen this on other asterisk installs) |
| 13:04.15 | DrukenLPY | ManxPower: aircard or the train have a working wifi network? |
| 13:04.29 | ManxPower | DrukenLPY: I got another aircard, not in my name 8-) |
| 13:04.47 | DrukenLPY | hehe |
| 13:05.02 | DrukenLPY | ie why your ircing with it... :) |
| 13:05.05 | ManxPower | Verizon does not have service on I-59 from about Picayunne MS to about Tuscaloosa, AL |
| 13:05.48 | ManxPower | because Verizon is the ONLY provider that works where I live. |
| 13:06.19 | LeddyHM | sounds like you need to move out of hickville usa |
| 13:06.30 | ManxPower | LeddyHM: I have my reasons for where I live. |
| 13:06.39 | skirmisha | ManxPower what about verizon |
| 13:06.53 | ManxPower | skirmisha: what about verizon? |
| 13:06.57 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 13:07.10 | skirmisha | what problem do u have with them |
| 13:07.50 | ManxPower | skirmisha: Their Unlimited Data service has a cap for 5GB/month. If you exceed that cap they will send you a letter, cancel your account. No warning, no appeal. |
| 13:08.24 | skirmisha | that's for internet? |
| 13:08.32 | LeddyHM | "data" |
| 13:08.39 | ManxPower | My other problems are that they have a giant dead zone for part of my trips for New Orleans / Burmingham. |
| 13:08.44 | [TK]D-Fender | ManxPower: Thats a very harsh limit for something "unlimited" |
| 13:08.50 | ManxPower | skirmisha: PCMCIA cell internet card. |
| 13:08.54 | ManxPower | [TK]D-Fender: Yup. |
| 13:09.24 | ManxPower | You will know when I am leaving Picayunne because I'll lose my connection 8-) |
| 13:09.42 | DrukenLPY | how the hell do you pull 5gb on an aircard anyways? hehehe |
| 13:09.50 | DrukenLPY | that's a shitload of data for dialup speeds |
| 13:10.28 | ManxPower | DrukenLPY: If the software is running and you are connected it takes over the default route. I had Bittorrent running and forgot I still had the verizon acct running |
| 13:11.13 | ManxPower | DrukenLPY: Dialup speeds? "Low speed" is 114Kbs, When I am in Birmingham, Tuscaloosa,or near New Orleans I get about 1Mbps |
| 13:11.37 | DrukenLPY | oh... evdo :) |
| 13:11.39 | DrukenLPY | nice.... |
| 13:12.02 | ManxPower | At home I'm at 114Kbps as well, but I also have Satellite Internet there. |
| 13:12.49 | ManxPower | So I used the satellite for everything except SSH and Verizon for SSH when I'm at home |
| 13:13.10 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:14.30 | DrukenLPY | too much lag on the sat for ssh ? |
| 13:14.51 | ManxPower | 250ms-350ms latency on the verizon, 900-3000ms latency on the satellite |
| 13:15.03 | DrukenLPY | ewhh..... nasty |
| 13:15.17 | ManxPower | and that verizon latency is for the lower speed 114Kbps aka 1xRTT service. |
| 13:15.43 | ManxPower | overall it works pretty good. |
| 13:16.31 | ManxPower | [TK]D-Fender: I sort of assumed that Unlimited means Unlimited. |
| 13:16.33 | DrukenLPY | i have an aircard somewhere... |
| 13:16.54 | *** join/#asterisk sob0l (n=sobol@emi250.internetdsl.tpnet.pl) |
| 13:17.06 | ManxPower | There is some fine print that basically says "If you use more than 5GB we are sure you are violating your ToS, so we'll cancel your ass." |
| 13:17.13 | [TK]D-Fender | ManxPower: Unlimited(tm) |
| 13:19.26 | ManxPower | [TK]D-Fender: Sptrint apparently does not do this, but sprint does not work where I live. |
| 13:19.26 | DrukenLPY | what's verizons price for the unlimited? |
| 13:19.26 | ManxPower | DrukenLPY: $60/month |
| 13:19.26 | DrukenLPY | oh that's not bad at all |
| 13:19.26 | DrukenLPY | i'd pay that... |
| 13:19.26 | ManxPower | Well until you hit the cap |
| 13:19.28 | DrukenLPY | i'd prolly never hit 5gig... |
| 13:19.41 | ManxPower | I've had the service since about 2 weeks after Katrina |
| 13:20.27 | ocgltd | FYI - in Canada $60 buys you 60MB of download...must be nice |
| 13:21.20 | [TK]D-Fender | YUCK |
| 13:21.37 | ManxPower | ocgltd: I would almost rather have the cheap internet service in Canada |
| 13:22.51 | DrukenLPY | cheap internet service? |
| 13:23.37 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:23.46 | DrukenLPY | yeah i don't think that exsists in canada.... |
| 13:23.57 | ManxPower | ocgltd: how much does your internet service cost and what is the speed (DSL or cable)? |
| 13:24.20 | DrukenLPY | cept if your in toronto... then you can piggy off toronto hydro... it's cheap... wifi network spanning a GOOD portion of the city |
| 13:24.31 | ManxPower | Ugh Wifi |
| 13:24.54 | blitzrage | what was the question? I live in Toronto |
| 13:25.12 | DrukenLPY | blitzrage: sorry to hear that :) |
| 13:25.20 | ManxPower | blitzrage: how much is internet access (dsl or cable) and how fast is it |
| 13:25.29 | blitzrage | DrukenLPY: you must live in Vancouver |
| 13:25.44 | DrukenLPY | blitzrage: barrie... only 45 mins up the hiway |
| 13:25.51 | blitzrage | barrie? ewwww |
| 13:25.58 | DrukenLPY | hehe |
| 13:26.16 | DrukenLPY | i operate a courier that runs between daily |
| 13:27.49 | blitzrage | ManxPower: I think I pay $45/mth for 5mbit/800kbps |
| 13:28.11 | ManxPower | I hate you |
| 13:28.14 | blitzrage | that's Roger's cable -- I also have DSL for like... $30/mth which is like 3mbit/400kbps I think |
| 13:28.18 | DrukenLPY | that's what i pay.. 45/mnt for 5mbit/800kbps |
| 13:29.13 | [TK]D-Fender | DrukenLPY: Same here, but add fixed IP, no caps, and dry loop |
| 13:29.26 | *** join/#asterisk Gringo_ (n=N3TW4LK3@83.30-245-81.adsl-dyn.isp.belgacom.be) |
| 13:29.42 | DrukenLPY | yeah.. i'm cable... |
| 13:29.45 | Gringo_ | anyone using Thomson ST2030's here? |
| 13:29.55 | DrukenLPY | as long as i don't reboot the modem... the ip don't change... |
| 13:30.25 | ManxPower | DrukenLPY: So you have the "Cheap Bastard Static IP" |
| 13:30.41 | DrukenLPY | only at home... |
| 13:31.00 | DrukenLPY | block of 32 and $200 a month at the office... |
| 13:32.01 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:32.06 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 13:33.16 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 13:33.23 | [TK]D-Fender | DrukenLPY: If I knew what I was doing I could have had a blok of 8 for the same price... but I suck at iptables :/ |
| 13:33.44 | [TK]D-Fender | ManxPower: Go read up on the "caffeine nap" |
| 13:33.46 | DrukenLPY | nap :) |
| 13:33.58 | LeddyHM | yay for sbc |
| 13:34.07 | ManxPower | [TK]D-Fender: is that anything like the "smoke a bowl nap"? |
| 13:34.10 | LeddyHM | we have fiber to our neighborhood :) |
| 13:34.16 | LeddyHM | 6down/768 up |
| 13:34.20 | LeddyHM | 5 statics |
| 13:35.06 | ocgltd | My cable is approx $60/mo, with UL of 500k (actual), DL of 4.5M (actual) |
| 13:35.10 | LeddyHM | it's only a uhaul trip away |
| 13:35.17 | ManxPower | The only really bad thing about living where I live is lack of broadband |
| 13:35.31 | LeddyHM | I max out my dl, 650KB/s |
| 13:35.55 | LeddyHM | manx: satellite? |
| 13:35.55 | LeddyHM | yes it's laggy, but better than nothin |
| 13:36.09 | ManxPower | LeddyHM: The only thing worse than satellite is...dialip |
| 13:36.29 | ManxPower | LeddyHM: For SSH, satellite is worse than nothing. |
| 13:38.08 | blitzrage | wow I'm sooooooooo groggy |
| 13:38.18 | blitzrage | these 9:30am -> 3am shifts are..... weird |
| 13:39.12 | ManxPower | looks like I'm in Picayunne now. |
| 13:39.31 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 13:39.32 | ManxPower | Soon it will be "So long, cruel world!" until I reach Tuscaloosa |
| 13:39.54 | LeddyHM | BFE! |
| 13:40.11 | DrukenLPY | your addicted to the internet when.... you belive it is a real world |
| 13:40.19 | [TK]D-Fender | ManxPower: not sure about your "smoke a bowlnap" reference, but a cafeine nap is when you grabe a coffe quick, and take a 15 minute nap immediately. You wake up refreshed and able to go one a long time. |
| 13:40.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:40.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:41.37 | ManxPower | Sounds similar to when I would take an ephedrine then take a nap; |
| 13:42.22 | [TK]D-Fender | ManxPower: Both are stimulants, so I'd guess its a reasonable idea |
| 13:42.44 | ManxPower | [TK]D-Fender: I just had to get to sleep before it kicked in. |
| 13:43.07 | irule | what is the simplest method I can use to tell asterisk that I am not home so that it will fw calls somewhere else besides local sip phones? thanks |
| 13:43.46 | [TK]D-Fender | irule: ifs your dialplan, how are we to know how you set anything up. Just change your dial. |
| 13:43.50 | ManxPower | irule: none |
| 13:44.59 | *** join/#asterisk pavlicek (n=jpavlice@mail.genevakc.com) |
| 13:44.59 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 13:46.03 | DrukenLPY | just call the sip phone and your cell all the time |
| 13:47.05 | DrukenLPY | at the SAME time... |
| 13:47.19 | *** join/#asterisk tsdragon (i=481d8101@gateway/web/cgi-irc/ircatwork.com/x-09fa3cb5bdaae5b2) |
| 13:47.54 | tsdragon | hello...i have a channel development question...anyone here do channel development? |
| 13:48.37 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
| 13:48.50 | unixlike | whats up |
| 13:48.55 | *** join/#asterisk vicentealencar (i=MyIdent@200.234.208.152) |
| 13:49.13 | tsdragon | anyone with experiance with the requester callback? |
| 13:49.19 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:49.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:50.15 | *** join/#asterisk imapfool (n=edhorton@216.23.111.98.nw.nuvox.net) |
| 13:50.16 | unixlike | i'm looking for a way to dial *67 before the number.. like dial(Zap/2/*675554545) anyone has an idea ? |
| 13:50.16 | vicentealencar | Hello Everyone! I would like to know if the execution of an AGI script could affect the "flow" of the dial plan. I mean, if I put "exit 1" would the dial plan consider that an error occurred and go to the n+101 rule? |
| 13:51.57 | mosty | vicentealencar, you can set context/extension/priority from agi programs |
| 13:52.08 | mosty | that's equivalent to doing a goto |
| 13:52.45 | *** join/#asterisk myiagy (i=myiagy@201.31.20.47) |
| 13:52.45 | DrukenLPY | unixlike: *67,,number like back in the old dialup days? |
| 13:52.54 | vicentealencar | mosty, Do you know a tutorial where I can learn about it |
| 13:53.02 | imapfool | unixlike: Can't yor just add it to the dial string like "Dial(Zap/2/*67${num-to-call}) Should work |
| 13:53.15 | mosty | vicentealencar, http://www.voip-info.org/wiki/index.php?page=Asterisk+AGI |
| 13:53.29 | vicentealencar | thanks!!! |
| 13:53.38 | unixlike | no it doesn't |
| 13:53.49 | unixlike | hold on 2 sec |
| 13:53.54 | *** join/#asterisk [hC] (n=hardcore@110.59.19.209.transedge.com) |
| 13:54.20 | tsdragon | i'm trying to write a channel driver and after Asterisk calls my requester callback it always thinks there are no available channels. any ideas? |
| 13:54.24 | unixlike | <PROTECTED> |
| 13:54.33 | unixlike | May 10 09:43:59 WARNING[1676]: app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 13:57.07 | tsdragon | i keep getting "Eveyone is busy/congested at this time (0:0/0/0)...i am returning an ast_channel structure that i got from ast_channel_alloc. |
| 13:57.23 | tsdragon | anyone/ |
| 13:57.25 | tsdragon | ? |
| 14:00.12 | [TK]D-Fender | unixlike: Why do you show a sample using zap/2/ and then in your actual dial statement not have anything? |
| 14:01.09 | unixlike | well i just tried with dial(Zap/2/*675555555) |
| 14:01.47 | unixlike | asterisk shows no error, dial the number but i get "diduda the number you dialed is not correct, please try again" |
| 14:01.52 | unixlike | but it works with a cellphone and/or regular analogue phone |
| 14:02.06 | Strom_M | how about inserting a pause after *67 |
| 14:02.41 | Strom_M | also, that "diduda" is called a Special Information Tone (SIT) |
| 14:02.49 | unixlike | oh thx |
| 14:03.47 | unixlike | but how can i insert a pause after the *67, is there a way to tell asterisk to not bridge the call right after the dial *67 ? |
| 14:04.59 | [TK]D-Fender | unixlike: Dial(zap/2/ww*67wwwwww1234567890) |
| 14:05.34 | Siya | re |
| 14:05.35 | unixlike | thx i'll try this right now |
| 14:06.43 | *** join/#asterisk nasls_lsa (n=chatzill@85.73.243.95) |
| 14:07.08 | unixlike | it works good, thank you d-fender |
| 14:07.55 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 14:12.47 | tsdragon | i need help with channel development...i keep getting "Eveyone is busy/congested at this time (0:0/0/0)" after my requester function...i am returning an ast_channel structure that i got from ast_channel_alloc. |
| 14:15.08 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 14:16.06 | *** join/#asterisk runlvl (n=runlvl@200.69.219.113) |
| 14:16.13 | runlvl | Buenas |
| 14:17.41 | tsdragon | hello, i need help with channel development...i keep getting "Eveyone is busy/congested at this time (0:0/0/0)" after my requester callback function is called...i am returning an ast_channel structure that i got from ast_channel_alloc. why doesn't it see any available channels? |
| 14:18.19 | Strom_M | tsdragon: don't spam your question here |
| 14:18.26 | sob0l | hello, I have quality issues with asterisk 1.4.2 and linksys pap2 with g726-32 codec, asterisk is compiled with USE_DEPRECATED_G726 |
| 14:18.26 | Strom_M | go do it in #asterisk-dev ;) |
| 14:20.01 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:20.43 | sob0l | and I've tried to use g726nonstandard=yes |
| 14:21.24 | Strom_M | what about allow=g726 and allow=g726aal2 |
| 14:21.26 | tsdragon | it's not spam...it's a valid question...i just repost it when a few more people join...just because you can't answer it don't get testy |
| 14:21.27 | imapfool | hello - has anyone had success with IMAP voicemail storage and asterisk-1.4? |
| 14:21.28 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 14:21.29 | kippi | hey |
| 14:21.38 | kippi | anyone using the 7911 cisco handset? |
| 14:21.45 | tsdragon | oh...sorry...thanks i'll try asterisk-dev |
| 14:21.58 | sob0l | it's allowed both of them |
| 14:22.10 | *** part/#asterisk tsdragon (i=481d8101@gateway/web/cgi-irc/ircatwork.com/x-09fa3cb5bdaae5b2) |
| 14:22.16 | Strom_M | sob0l: have you tried asterisk 1.4.4? |
| 14:23.22 | sob0l | no, |
| 14:24.12 | sob0l | strange, now I get chan_sip.c:5159 process_sdp: No compatible codecs, not accepting this offer! |
| 14:26.35 | sob0l | Strom_M: it's due to g726nonstandard=yes, when it's g726nonstandard=no i can make calls |
| 14:26.45 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 14:26.50 | Strom_M | ah ok |
| 14:26.58 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:27.32 | sob0l | Strom_M: but the quality issue persist :( |
| 14:28.32 | Strom_M | what kind of "quality issues"? |
| 14:33.24 | sob0l | Strom_M: it's distorted |
| 14:33.39 | kippi | anyone using the 7911 cisco handset? |
| 14:33.56 | Strom_M | but distorted how? |
| 14:34.11 | Strom_M | clipped? jittery? packet-lossy? |
| 14:34.18 | Strom_M | extraneous noise? |
| 14:34.54 | Mercestes | echo, Mr. Roboto, one way audio, no audio. |
| 14:35.02 | Mercestes | a sip call is like a box of chocolates |
| 14:35.05 | Strom_M | etc etc etc etc etc |
| 14:35.15 | Mercestes | and that's all I have to say about that. |
| 14:35.19 | sob0l | Strom_M: clipped on pap2 on snom 190 it sounds ok |
| 14:35.45 | Strom_M | so...the other party sounds clipped when you're talking on the pap2? |
| 14:36.13 | sob0l | both, |
| 14:36.39 | Strom_M | both what |
| 14:36.48 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:37.34 | sob0l | when i say something it's clipped and I hear clipped |
| 14:37.45 | Strom_M | sounds like a gain setting on the PAP2 |
| 14:38.12 | anonymouz666 | PAP2 has some static noise that I just can't explain. |
| 14:38.21 | anonymouz666 | It does not happen all time. |
| 14:38.24 | anonymouz666 | but it happens |
| 14:38.34 | anonymouz666 | and just a reboot fixes. |
| 14:39.25 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:40.20 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 14:41.40 | sob0l | Strom_M: on other older asterisk system |
| 14:41.47 | sob0l | Strom_M: it's sounds ok |
| 14:42.34 | kippi | sob0l: I have got the same problem to |
| 14:42.42 | Strom_M | sob0l: so have you tried a newer version of asterisk? |
| 14:43.45 | sob0l | Strom_M: i have problem using 1.4.4 and sangoma |
| 14:44.10 | Strom_M | oh? |
| 14:44.43 | anonymouz666 | 1.4.4 is a good release |
| 14:44.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:45.40 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:46.01 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:47.32 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:50.22 | sob0l | Strom_M: now format is alaw but module show like codec_alaw.so show use count 0 |
| 14:50.59 | Strom_M | ? |
| 14:51.01 | dlynes_laptop | sob0l: codec_alaw.so is only if it needs to convert to alaw, or convert from alaw |
| 14:51.28 | Strom_M | yes |
| 14:51.28 | dlynes_laptop | sob0l: format_alaw is the normal module |
| 14:51.38 | dlynes_laptop | sob0l: erm format_pcm_alaw.so I mean |
| 14:52.40 | dlynes_laptop | sob0l: you'll only see use counts on those, if there's an actual active call, using one of them |
| 14:52.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 14:53.04 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 14:53.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:55.00 | Dr-Linux | anything worng with this? >> exten => 7272,2,Set(CALLERID(name)=SVC) |
| 14:55.10 | Strom_M | no |
| 14:55.37 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:55.45 | Mercestes | well, yea the >> at the front is a syntax error |
| 14:56.01 | Strom_M | yeah, apart from that |
| 14:56.42 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 14:57.14 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 14:57.32 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 14:57.59 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:58.00 | Katty | morning! (= |
| 14:58.04 | Mercestes | KATTY! |
| 14:58.09 | Mercestes | *squeeee* |
| 14:59.35 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 14:59.39 | Katty | DrukenLPY: thank you, but I have no children (= |
| 14:59.51 | DrukenLPY | oh well... i tried |
| 14:59.52 | Katty | I /do/ however, have some questions about 1.4! |
| 14:59.53 | DrukenLPY | :) |
| 15:00.05 | Katty | zaptel, particularly. |
| 15:00.20 | Katty | after ./configure and make menuselect, this zaptel module selection page thingy came up... |
| 15:00.25 | Katty | like, i presume, it's supposed to. |
| 15:00.45 | Katty | But I'm, uh, not sure what i'm supposed to do with it now since I had none of this in 1.2 |
| 15:01.03 | Strom_M | Katty: you're supposed to select the modules you want to compile |
| 15:01.15 | Katty | ah ha! |
| 15:01.25 | DrukenLPY | kinda like building a kernel :) |
| 15:01.46 | Katty | I know for a fact that I need wctdm, since right now we're using 4 port analog cards. |
| 15:01.50 | DarKnesS_WolF | Katty: why u don't just do ./configure make make install ? |
| 15:02.08 | DarKnesS_WolF | it will compile all zaptel drivers " wcdtm , ztdummy zttrancecode , etcc,,etcc, " |
| 15:04.25 | Katty | But here fairly shortly, we're going to switch over to a card that can handle a T1 |
| 15:04.25 | Strom_M | Katty: well, which card? |
| 15:04.26 | Katty | Would it be better to select all the modules, or just wctdm and then..that t1 card.. thing |
| 15:17.52 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:17.52 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 15:18.04 | afrosheen | my kingdom for an iax template.. |
| 15:18.29 | JT | afrosheen: how big is your empire? |
| 15:18.32 | Mercestes | afrosheen, how big is you rkingdom? |
| 15:18.45 | Mercestes | JT: lol |
| 15:18.46 | mocker | Woo, migrated everyone off a legacy Trixbox system to straight Asterisk. :) |
| 15:18.46 | afrosheen | it's about 8x8 |
| 15:18.56 | Mercestes | how many virgins does it include? |
| 15:19.29 | Strom_M | afrosheen: did you "make samples"? |
| 15:19.33 | afrosheen | Strom_M, nope |
| 15:19.42 | Strom_M | look in the asterisk source directory |
| 15:19.52 | afrosheen | Strom_M, it's been about 2 years since I've plugged in an iax device, please be gentle :) |
| 15:19.57 | Strom_M | look for a file called iaxprov.conf.sample |
| 15:20.03 | afrosheen | roger that |
| 15:20.05 | afrosheen | brb |
| 15:21.12 | Gringo_ | is anyone here using Thomson ST2030 SIP phones? |
| 15:21.30 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-082-085.pools.arcor-ip.net) |
| 15:25.01 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:27.36 | Katty | are there some zaptel utility module thingies that are recommended? |
| 15:27.49 | Katty | i'm framilier with ztcfg, but there are lots of other ones without an explination |
| 15:28.07 | JT | zttest is important |
| 15:28.14 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 15:28.35 | Katty | it's one of the default selected ones...along with foxtune, ztdiag, ztmonitor and ztspeed |
| 15:28.46 | Katty | s/foxtune/fxotune |
| 15:29.36 | Dr-Linux | [TK]D-Fender: around? |
| 15:29.38 | Mercestes | foxtune |
| 15:29.41 | Mercestes | LMAO |
| 15:30.00 | JT | needed by furytel |
| 15:30.11 | Katty | <PROTECTED> |
| 15:30.30 | Dr-Linux | actually i want to use some suitable MultiTech product between 2 asterisk server for VoIP from here: http://www.multitech.com/PRODUCTS/Families/MultiVOIP/ |
| 15:30.37 | Dr-Linux | any suggestions? |
| 15:30.47 | *** join/#asterisk axisys (n=axisys@ip72-205-1-67.dc.dc.cox.net) |
| 15:31.58 | pifiu | hey everyone |
| 15:32.21 | Katty | pifiu: (= |
| 15:33.02 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 15:33.34 | *** join/#asterisk Fieldy (i=Iozx7hcS@gentoo/contributor/Fieldy) |
| 15:33.47 | deeperror | any suggestions for a good firewall that will handle qos and able to handle large volume traffic? |
| 15:33.57 | file | Kattttttttttty |
| 15:35.25 | *** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik) |
| 15:35.38 | [TK]D-Fender | Dr-Linux: Why should you need ANYTHING between 2 * servers? Thats what VoIP is for. |
| 15:36.20 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 15:37.13 | Dr-Linux | [TK]D-Fender: we have some bandwidth issues here with * protocols, but multitech is equipment always works very well for us, so if their some product suits me, i gonna have it |
| 15:37.27 | Dr-Linux | but still i need suggestions for any product |
| 15:37.52 | FarrisG | We're having an issue with incoming calls over sip trunks and DTMF. Occasionally (and I can't pinpoint the conditions under which it will occur), one of the digits will be repeated. For instance, sometimes when the caller dials extensions "141", the first 1 will be repeated and they will reach extension "114". Any ideas? |
| 15:37.57 | [TK]D-Fender | Dr-Linux: IAX + G.729. Doesn't get any smaller than that. Unless you're going to go G.723.1 |
| 15:38.33 | [TK]D-Fender | Dr-Linux: And actually, IAX + G.729 will beat SIP + any other codec I can think of |
| 15:38.41 | [TK]D-Fender | Dr-Linux: past 1 channel |
| 15:38.42 | Dr-Linux | yes, with MT we are using g723.1 also that support VAD , which asterisk doesn't |
| 15:39.02 | [TK]D-Fender | Dr-Linux: If you're that screwed, just give up... |
| 15:39.09 | Dr-Linux | [TK]D-Fender: currently i'm using IAX + G.729 |
| 15:39.22 | [TK]D-Fender | Dr-Linux: its going to cost you a fortune to work around in such a petty way |
| 15:39.23 | Dr-Linux | past 1 channel? |
| 15:39.49 | [TK]D-Fender | Dr-Linux: IAX trunking is better that any savings G.723.1 could have over G.729 in SIP |
| 15:41.48 | coppice | G.723.1 saves almost nothing. set G.729 to 30ms chunks and you hardly tell the difference |
| 15:45.05 | afrosheen | wow that just scared the hell out of me...note to self, never run make samples on a production box while people are on the phone |
| 15:45.42 | Strom_M | uh |
| 15:45.53 | Strom_M | that'll wipe out your existing config files |
| 15:46.14 | afrosheen | yeah, I'm not used to the make command actually installing shit, it's a stupid design if you ask me |
| 15:46.31 | Strom_M | well..."make install" installs shit |
| 15:46.41 | afrosheen | that's precisely what I'm used to |
| 15:46.50 | afrosheen | so make samples install would make sense in that context |
| 15:46.59 | Strom_M | file a bug |
| 15:47.08 | afrosheen | oh well at least I had a backup of the entire /etc/asterisk directory from 2 days ago |
| 15:47.08 | Strom_M | or read the documentation before running things :) |
| 15:48.58 | mosty | i also find it annoying when distributions package the sample files in /etc/asterisk/ - imo samples should go in /usr/share/doc/asterisk/examples/ |
| 15:49.00 | aptura | Can anyone sugest a combined DSL/T1/ethernet tester short of a fluke tester? |
| 15:52.22 | afrosheen | mosty, that would make more sense as well |
| 15:53.01 | *** join/#asterisk amd64lover (n=amd64lov@62.117.184.29) |
| 15:53.03 | amd64lover | hi there |
| 15:53.22 | CCFL_Man2 | guys |
| 15:54.04 | CCFL_Man2 | my sip provider insists that everything is provisioned properly |
| 15:54.20 | CCFL_Man2 | but i cannot get incomming calls |
| 15:54.50 | deeperror | is setup to connect to your provider? |
| 15:55.27 | CCFL_Man2 | yep, registers every 3600 secs |
| 15:56.04 | deeperror | does the call come in and die? |
| 15:56.10 | deeperror | debug output? |
| 15:56.34 | CCFL_Man2 | nothing comes in at all |
| 15:56.38 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-675aee835e6aa5f5) |
| 15:56.38 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:56.56 | CCFL_Man2 | i'm guessing it could be a nat issue |
| 15:57.30 | deeperror | do you have the ports forwarded? |
| 15:57.37 | LeddyHM | any way to get a timestamp in the cli? |
| 15:57.40 | CCFL_Man2 | no |
| 15:57.46 | deeperror | probably nat issue |
| 15:58.40 | CCFL_Man2 | deeperror: i had a vonage sip account previously provisioned and ut never required any ports forwarded |
| 15:59.03 | deeperror | was your vonage box setup as your router as well? |
| 15:59.16 | *** join/#asterisk Vec (n=Vec@dsl-241-209-206.telkomadsl.co.za) |
| 15:59.37 | CCFL_Man2 | no, i never used a vonage box, i used the sip account from it |
| 15:59.45 | CCFL_Man2 | i haxed it |
| 16:00.00 | CCFL_Man2 | but the sip account worked perfectly |
| 16:00.39 | CCFL_Man2 | and here with quantum voiceni'm having issues |
| 16:00.41 | [TK]D-Fender | CCFL_Man2: * needs a pile of settings in SIP.conf and port forwarding. |
| 16:00.43 | [TK]D-Fender | ~sipnat |
| 16:00.50 | jbot | hmm... sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:01.19 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:01.22 | Vec | I am having a problem running sip calls over wifi I have Grandstream 200 --> Wifi AP --Bridge-- Wifi AP --> Switch --> asterisk, I am using aLaw, the problems are during the call the audio cuts out for a few seconds and then continues normally only in 1 direction, the problem does not occur when not making use of wifi link ? |
| 16:01.23 | CCFL_Man2 | hmm.. |
| 16:01.47 | MrTelephone | hey has anyone had a problem where you pick up another media channel breifly when you dial out? |
| 16:02.11 | CCFL_Man2 | why would port forwarding not be required with a vonage sip account but required with quantumvoice? |
| 16:02.30 | [TK]D-Fender | CCFL_Man2: With * acting as the UA for both? |
| 16:02.44 | *** join/#asterisk Dovid (i=HydraIRC@bzq-82-81-161-200.red.bezeqint.net) |
| 16:02.47 | CCFL_Man2 | [TK]D-Fender: yes |
| 16:03.04 | *** join/#asterisk fadey (n=fadey@176.red-82-159-215.user.auna.net) |
| 16:03.15 | MrTelephone | asterisk is picking up audio briefly from the wrong zap channel.. I'm guessing thats what the problem is?? |
| 16:03.17 | CCFL_Man2 | vonage used no stun server |
| 16:03.19 | Dovid | hi guys. I have a SNOM 360 that keeps sending back a 486 error. It keeps telling asterisk that it's busy. I made sure that the phone is not set to DND or anything of the sort |
| 16:03.29 | [TK]D-Fender | CCFL_Man2: Only way I can think of is that they forced a registration window small enough to keep the UDP mapping intact without forwarding. But that'd be insssane on their part |
| 16:03.29 | *** join/#asterisk mcf3782 (n=mfreeman@209.117.160.3) |
| 16:03.30 | Dovid | anyone know what it can be ? |
| 16:03.45 | [TK]D-Fender | CCFL_Man2: * doesn't support STUN. |
| 16:03.51 | mosty | dovid: set log level to 9 on the phone, and see what it's doing |
| 16:04.05 | coppice | a stunning omission :-) |
| 16:04.09 | CCFL_Man2 | [TK]D-Fender: i know |
| 16:04.21 | fadey | hello, everyone. Is it possible to disable DTMF detection once the voice call is established? |
| 16:04.48 | CCFL_Man2 | [TK]D-Fender: with vonage i had to register every 60s |
| 16:05.03 | [TK]D-Fender | CCFL_Man2: that might do it... |
| 16:05.04 | *** join/#asterisk DeeJayTwo (n=deejay2@148-253.sh.cgocable.ca) |
| 16:05.08 | Dovid | mosty: i set it to 9 and got nothing |
| 16:05.11 | [TK]D-Fender | CCFL_Man2: but its a bad way to rowk. |
| 16:05.11 | CCFL_Man2 | ahhh |
| 16:05.25 | [TK]D-Fender | work* |
| 16:05.27 | CCFL_Man2 | [TK]D-Fender: it's vonage |
| 16:05.37 | [TK]D-Fender | CCFL_Man2: Yeah I know.. nothing new.. |
| 16:05.45 | CCFL_Man2 | of course |
| 16:05.53 | mosty | Dovid, are you running the latest stable firmware? |
| 16:06.01 | Dovid | mosty: i just keep seeing 486 erorr in asterisk. |
| 16:06.05 | Dovid | mosty: i believe so |
| 16:06.14 | JT | regular registration to hold open nat mappings is a common nat punching technique |
| 16:06.31 | Dovid | Application-Version:snom360-SIP 6.5.8 |
| 16:06.32 | JT | you shouldn't need to port forward just to receive phone calls acting as UA |
| 16:06.36 | CCFL_Man2 | [TK]D-Fender: what port range should i use for the voice? |
| 16:06.45 | CCFL_Man2 | 16384 to whst? |
| 16:06.58 | JT | 10000-20000? |
| 16:07.20 | Dovid | mosty: v |
| 16:07.20 | Dovid | http://www.snom.com/download/snom360-3.31-r.bin |
| 16:07.31 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 16:08.17 | CCFL_Man2 | [TK]D-Fender: vonage also used port 10000 gor their sip proxy, audio was a little bit above this, like 10100 |
| 16:08.23 | CCFL_Man2 | for |
| 16:08.25 | Dovid | oops. not running that. it used to work great. did a firmware update and its acting up |
| 16:08.26 | Dovid | brb |
| 16:11.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:11.13 | JT | CCFL_Man2: port forwarding shouldn't be necessary |
| 16:11.56 | *** part/#asterisk amd64lover (n=amd64lov@62.117.184.29) |
| 16:12.49 | DeeJayTwo | in sip show peers |
| 16:12.56 | DeeJayTwo | if I have the extension 100 |
| 16:13.01 | DeeJayTwo | what does it means when I have 100/100 |
| 16:13.03 | DeeJayTwo | instead of 100 |
| 16:15.30 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:16.56 | *** join/#asterisk Fieldy (i=Q09kxh49@gentoo/contributor/Fieldy) |
| 16:17.26 | deeperror | deejaytwo: they are logged in? |
| 16:17.57 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:18.38 | DeeJayTwo | I have one which registered...and unregistered |
| 16:18.43 | DeeJayTwo | and is still 114/114 |
| 16:18.48 | deeperror | deejaytwo: or could be as the header imploys Name/username |
| 16:18.52 | DeeJayTwo | with I (Unspecified) |
| 16:18.55 | DeeJayTwo | with IP.. |
| 16:19.29 | deeperror | From sip.conf [114] username=114 |
| 16:19.50 | deeperror | [this] username=that |
| 16:19.52 | deeperror | this/that |
| 16:19.56 | DeeJayTwo | ok I see |
| 16:20.35 | DeeJayTwo | hmm... It looks like it's not this/that.. ;) |
| 16:20.41 | DeeJayTwo | i have no username= defined |
| 16:21.05 | deeperror | then it takes from [this] |
| 16:21.19 | deeperror | it says right above when you type sip show peers what that is indicating |
| 16:24.13 | *** join/#asterisk primatus2 (n=solapus@66.90.121.129) |
| 16:27.39 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-7-131.w81-251.abo.wanadoo.fr) |
| 16:28.13 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 16:29.46 | *** join/#asterisk CVirus (n=GoD@196.205.193.14) |
| 16:32.17 | *** join/#asterisk wunderkin (n=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 16:34.32 | *** join/#asterisk amd64lover (n=amd64lov@62.117.184.29) |
| 16:36.37 | amd64lover | i have a pda with internet access. I have a always on server connected to the internet. I'm thinking in a solution of making voipcalls to landlines and mobilephones through voip. I could configure a sip server, but how the server will connect to mobile devices? |
| 16:37.09 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 16:37.14 | amd64lover | a GSM modem on the server seems to be a expensive solution |
| 16:38.12 | JT | if you just need to make calls to mobile devices, than the you could just use voip to make the calls |
| 16:39.21 | amd64lover | JT> what do you mean? |
| 16:39.53 | JT | what's so hard to understand.... i assume you want to make phone calls |
| 16:39.54 | amd64lover | the destination mobile phones are in cellular lines |
| 16:40.07 | *** part/#asterisk Gringo_ (n=N3TW4LK3@83.30-245-81.adsl-dyn.isp.belgacom.be) |
| 16:40.08 | JT | sure |
| 16:40.48 | amd64lover | but I want to self service the sip server |
| 16:41.01 | JT | what? |
| 16:41.06 | JT | i have no idea what that means |
| 16:41.11 | amd64lover | I mean.. I don't want to use services like voipbuster or skype |
| 16:41.53 | JT | yes, those services are rubbish |
| 16:41.53 | amd64lover | I would like to build my own sip server |
| 16:41.53 | JT | sure... |
| 16:42.03 | amd64lover | but I dont know how to do the sip server <-> cellular mobile phones interface |
| 16:42.40 | Qwell[] | amd64lover: in the very near future, there will be a bluetooth (for cellphones) channel driver in svn trunk.. that may help with what you want |
| 16:42.43 | JT | you want to use a gsm gateway or phone of some sort? |
| 16:42.54 | amd64lover | yes |
| 16:43.02 | amd64lover | precisely |
| 16:43.45 | JT | you can use an external gsm gateway, a gsm pci card, or a gsm phone connected with bluetooth |
| 16:44.15 | amd64lover | gsm modems doesn't compensate the trouble |
| 16:44.24 | amd64lover | about gsm gateways |
| 16:44.28 | JT | that makes no sense |
| 16:44.33 | JT | i never said gsm modem |
| 16:44.51 | amd64lover | gsm pci card? isnt a modem? |
| 16:44.59 | JT | no. |
| 16:45.24 | amd64lover | it doesn MOdelate/DEModelate the radio signal? |
| 16:45.49 | JT | it would, and it'd VOCODE etc etc |
| 16:45.54 | JT | but it would not be named a modem |
| 16:46.00 | amd64lover | sorry |
| 16:46.28 | coppice | interesting. I wonder how GSM might work without modulating or demodulating a radio signal? :-\ |
| 16:46.32 | CVirus | I got a telephone line coming from the telco connected to a legacy hardware PBX ... I want a device so that I would connect the phone line to a VOIP network and to the PBX ... so that when the VOIP (Internet) is down ... I would be able to use the PSTN again |
| 16:46.53 | CVirus | suggestions ? |
| 16:46.55 | CVirus | what do I need ? |
| 16:47.16 | CVirus | Here's a diagram http://img175.imageshack.us/my.php?image=magimx2.png |
| 16:47.44 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 16:47.58 | amd64lover | JT> I did some GSM pci card cost calculations and it doesn't compensate the effort |
| 16:48.39 | JT | amd64lover: i didn't say it was the only option, feel free to not use one |
| 16:49.19 | amd64lover | JT> I searched for a gateway GSM too but I didn't found any interesting result |
| 16:49.27 | afrosheen | Strom_M, you still around? |
| 16:49.33 | amd64lover | In Portugal services like this are too expensive |
| 16:49.36 | JT | amd64lover: there are a LOT of gsm gateways out there |
| 16:49.59 | deeperror | CVirus: FXO cards? |
| 16:50.02 | JT | amd64lover: then why not use voip to an itsp? |
| 16:50.14 | CVirus | deeperror: how will I connect it ? |
| 16:50.50 | deeperror | CVirus: how many lines will need to connect to telco? |
| 16:50.51 | *** join/#asterisk poppos (n=asd@S0106004063d8e527.ed.shawcable.net) |
| 16:50.55 | CVirus | deeperror: 1 |
| 16:51.01 | amd64lover | JT> I tried to find some portuguese solutions.. |
| 16:51.10 | deeperror | a digium FXO card? |
| 16:51.13 | amd64lover | JT> but they are almost NILL |
| 16:51.19 | CVirus | deeperror: I want it to automatically switch the PSTN when the internet is down |
| 16:51.23 | poppos | I would like to know if there is a web app that will allow me to create gsm files and msg , without doing it manual wav then to gsm? |
| 16:51.24 | JT | amd64lover: ok...... |
| 16:51.34 | CVirus | deeperror: which means .. I need the line connected to the legacy PBX as well |
| 16:51.43 | JT | poppos: you can record gsm files with asterisk |
| 16:52.01 | deeperror | the line can switch as any phone line would at your home |
| 16:52.01 | amd64lover | would you make me a good itsp recomendation? |
| 16:52.01 | *** part/#asterisk theacolyte (n=theacoly@unaffiliated/theacolyte) |
| 16:52.01 | poppos | JT= How |
| 16:52.01 | amd64lover | JT> would you make me a good itsp recomendation? |
| 16:52.03 | JT | poppos: Record |
| 16:52.07 | deeperror | take red/green and run one pair to * other pair to pbx |
| 16:52.16 | poppos | Jt= i would like a web app |
| 16:52.29 | JT | amd64lover: not sure what would have good connectivity to portugal |
| 16:52.35 | *** join/#asterisk Waverly360 (n=irc@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 16:52.36 | JT | poppos: what on earth for? |
| 16:52.38 | deeperror | then use dialplan in * to determine whether to send out on pstn or net |
| 16:52.51 | deeperror | or |
| 16:52.59 | amd64lover | JT> I'm just worried about service costs |
| 16:53.27 | deeperror | CVirus: use an FXO and FXS cards and use FXO -> pstn hook FXS into legacy system? |
| 16:53.57 | amd64lover | JT> I only want to replace my cellular phone with my internet pda. Anything with higher costs than my cellular phone wont compensate |
| 16:54.32 | JT | amd64lover: use the experimental chan_cellphone to connect with bluetooth |
| 16:54.45 | CVirus | deeperror: asterisk is far away from the legaxy PBX ... they are 2 separate companies actually |
| 16:55.01 | deeperror | ok so just as your diagram is then |
| 16:55.47 | deeperror | so your just talking about the asterisk side of your diagram? |
| 16:56.02 | CVirus | deeperror: no .. the other side .. you see .. there's a gab |
| 16:56.04 | CVirus | gap* |
| 16:56.23 | CVirus | deeperror: between the network switch and the legacy PBX |
| 16:56.40 | DarKnesS_WolF | CVirus: use checkstatus function |
| 16:56.49 | CVirus | DarKnesS_WolF: what is that ? |
| 16:57.10 | CVirus | DarKnesS_WolF: the calls go from office 1 to office 2 |
| 16:57.10 | DarKnesS_WolF | and how is weather ? |
| 16:57.16 | CVirus | DarKnesS_WolF: not the other way round |
| 16:57.50 | deeperror | are you going to have another * box in office 1? |
| 16:57.56 | CVirus | deeperror: no |
| 16:58.14 | CVirus | just the legacy PBX |
| 16:58.27 | DarKnesS_WolF | CVirus: how many phone in the site 2 ? |
| 16:58.29 | *** join/#asterisk unixlike (n=spid3r@31.67.modemcable.oricom.ca) |
| 16:58.37 | CVirus | DarKnesS_WolF: about 5 |
| 16:58.41 | JT | without thinking about it too much, i think CVirus wants an SPA-3102 |
| 16:58.44 | CVirus | DarKnesS_WolF: side 2 doesn't concern me |
| 16:58.48 | Strom_M | afrosheen: briefly |
| 16:58.50 | Strom_M | what's up |
| 17:00.47 | CVirus | JT: that doesn't have a rj-45 port I guess |
| 17:00.47 | CVirus | ? |
| 17:01.16 | amd64lover | JT> hum.. I think I was not clear. I mean.. I have a PDA with cellular access and Intertet access through Wi-Fi. Because on my city I have wifi everywhere, I would like to use voip through internet instead of cellular calls because my cellular calls are expensive. |
| 17:02.07 | deeperror | my buddy has a sip client on his phone that just connects up to an * box for those types of calls |
| 17:03.47 | amd64lover | JT> I have the possibility of building a SIP server. What I dont know how to do is the SIP server <-> GSM gateway interface. I though on a cellular phone connected to the SIP server, but this would lead to equivalent cellular line costs.. |
| 17:04.46 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 17:05.16 | amd64lover | I'm just hoping that your experience will give me some idea.. |
| 17:06.03 | *** join/#asterisk lavmol (n=lavmol@69.159.222.24) |
| 17:06.23 | lavmol | Hey all wondering is someon could help with a second NIC problem??? |
| 17:07.34 | *** join/#asterisk [hC] (n=hardcore@110.59.19.209.transedge.com) |
| 17:07.56 | lavmol | box has 2 nics the ethX files have 2 seperate IP's for their roles I run network restart and they end up on the same IP??? |
| 17:08.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:11.31 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 17:11.45 | syzygyBSD | would callprogress cause some calls to potentially be droped? |
| 17:12.01 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:12.24 | Delvar | does anyone know how to get the codcs used after a dial? i want to log them into CDR but cant find any way to do it without hacking the code.... |
| 17:12.46 | slmnhq | Hi, I was thinking of doing a fresh setup of Asterisk on a Ubuntu Linux system... any thoughts about the choice of OS? I'm looking for ease of installation and setup and reliability |
| 17:14.02 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 17:14.02 | syzygyBSD | ubuntu is a debian derivitave, I have the best success with management of those systems |
| 17:14.02 | sevard | you really deploy ubuntu in production environments?!? |
| 17:14.03 | syzygyBSD | me.. no.. but debian yes |
| 17:14.04 | *** join/#asterisk [hC] (n=hardcore@110.59.19.209.transedge.com) |
| 17:14.04 | sevard | phew. |
| 17:14.09 | sevard | I was about to go ape shit. |
| 17:14.12 | slmnhq | syzygyBSD i've had good success with Ubuntu in home and production environments |
| 17:14.18 | *** join/#asterisk Dimik (n=Dimik_@unaffiliated/dimik) |
| 17:14.21 | deeperror | my vote is debian |
| 17:14.22 | amd64lover | slmnhq> Gentoo is highly customisable |
| 17:14.28 | syzygyBSD | I use ubuntu at home, haven't tried it production though |
| 17:14.32 | slmnhq | sevard, haha.. relax |
| 17:14.51 | syzygyBSD | no need to.. debian makes it so easy |
| 17:15.22 | sevard | <-- slackware :) |
| 17:15.26 | syzygyBSD | lol, I used slackware in 98, then I moved on |
| 17:15.42 | slmnhq | syzygyBSD, I haven't been keeping up with Debian lately... do they have a new friendly installer now? |
| 17:15.43 | deeperror | haha |
| 17:16.05 | syzygyBSD | slmnhq: define friendly, I always understood it fine |
| 17:16.08 | sevard | That's why your slackware system built in 98 still runs |
| 17:16.43 | syzygyBSD | na, it was on a 486 with a 100mb HD. I wouldn't call that "running" |
| 17:16.46 | amd64lover | emerge asterisk selinux-asterisk -av |
| 17:17.07 | amd64lover | secure and easy.. :) |
| 17:17.47 | slmnhq | syzygyBSD, last I used debian it still had a libcurses based installer.. the new release was supposed to have a more graphical and intelligent interface... I'll give Debian a try... after all, that's how I was introduced to Linux |
| 17:18.34 | deeperror | slmnhq: i use the core netinstall it still uses curses but is very nice |
| 17:18.52 | syzygyBSD | mmm, libcurses |
| 17:19.04 | syzygyBSD | why would you ever need anything else? |
| 17:19.26 | syzygyBSD | with GUI interfaces they are too reliant on the mouse, most of my installs don't have mice |
| 17:20.36 | deeperror | or monitors haha |
| 17:20.46 | slmnhq | syzygyBSD, deeperror, et al .. I generally agree.. thanks |
| 17:20.57 | deeperror | i guess a monitorless install would be difficult |
| 17:21.10 | syzygyBSD | deeperror: I have done it before |
| 17:21.17 | deeperror | just netinstall |
| 17:21.20 | syzygyBSD | console over serial with a cyclades |
| 17:21.29 | deeperror | times are a changin |
| 17:21.52 | deeperror | i installed etch with usb stick last week was the first time doing a full usb install |
| 17:22.00 | deeperror | floppys and cd's are out |
| 17:22.18 | deeperror | no use for them |
| 17:22.29 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 17:22.46 | syzygyBSD | floppy's yes... cds.. sure, dvds are still useful |
| 17:23.12 | deeperror | yea your right |
| 17:23.12 | deeperror | dvd's will be until the usb gets cheap and over 5gig |
| 17:23.21 | syzygyBSD | I am still keeping one knoppix cd around that is 5 years old |
| 17:23.24 | deeperror | always good to have though |
| 17:23.29 | deeperror | never know when your gonna need a knoppix cd for sure |
| 17:23.32 | sevard | I still have use for floppys. |
| 17:23.37 | deeperror | yea |
| 17:23.42 | deeperror | i use them too on a few older systems ha |
| 17:23.45 | syzygyBSD | well, it is because of customizations I have.. added CHNT onto it so I can bypass any windows passwords |
| 17:24.01 | hacim | someone from brasil called my asterisk server, the echo test was fine, but when they connected to the meetme, it was terrible -- he could hear me fine, but he was a drunken robot... I've got the zaptel and ztdummy modules loaded but am otherwise lost |
| 17:24.41 | syzygyBSD | hacim: how did he call your server? SIP, Zap, ? |
| 17:24.47 | deeperror | hacim: any load issues? |
| 17:25.10 | Cresl1n | hacim, sounds almost like a codec issue |
| 17:25.14 | *** part/#asterisk CVirus (n=GoD@196.205.193.14) |
| 17:25.15 | Cresl1n | alaw vs ulaw |
| 17:25.21 | sevard | "he was a drunken robot" i love it. |
| 17:25.45 | syzygyBSD | sevard: alcohol is their fuel |
| 17:25.55 | hacim | syzygyBSD: SIP |
| 17:26.29 | syzygyBSD | hacim: Cresl1n is probably right |
| 17:26.33 | hacim | Cresl1n: yeah, but if me and two others call, with the same sip client, we sound fine |
| 17:26.34 | sevard | syzygyBSD: taking lessons from bender again, mmm? |
| 17:26.57 | hacim | deeperror: no load issues, nothing was happening on the server except asterisk |
| 17:26.57 | syzygyBSD | sevard: glad you placed it, and yes |
| 17:27.10 | sevard | syzygyBSD: bbiab, have to jack on. |
| 17:27.32 | hacim | Cresl1n: wouldn't we both have the same codecs? |
| 17:28.19 | deeperror | check debug output on the workign calls |
| 17:28.24 | deeperror | and non working calls and compare |
| 17:28.51 | syzygyBSD | I'm jealous |
| 17:30.31 | lavmol | anyone know if I run lspci and see nic a listed befor nic b then the eth0 will be assigned to nic a??? |
| 17:33.02 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:33.14 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 17:33.29 | anthony] | Hey, anyone know anything about a Cisco 2514? Is it a router or something? |
| 17:33.47 | Qwell[] | 2 seconds with google would tell you that |
| 17:34.09 | anthony] | Well, I have.. but it has no ehernet ports to plug servers into |
| 17:34.15 | *** part/#asterisk amd64lover (n=amd64lov@62.117.184.29) |
| 17:35.47 | anthony] | What kind of switch is this? |
| 17:37.38 | *** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net) |
| 17:37.46 | MrTelephone | anyone experience carrier errors on their t1 link? |
| 17:37.48 | sevard | the ports on the back of the 2514 are ethernet, they're DB-15 |
| 17:39.01 | sevard | AUI |
| 17:39.43 | sevard | http://en.wikipedia.org/wiki/Attachment_Unit_Interface |
| 17:43.37 | *** join/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 17:43.47 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 17:44.37 | naitram | how do you communicate with asterisk from a php script? Want script to run external to asterisk and do functions in the dial plan |
| 17:44.59 | JunK-Y | naitram: agi |
| 17:45.02 | JunK-Y | ~agi api |
| 17:45.22 | jbot | agi api is probably at http://home.cogeco.ca/~camstuff/agi.html |
| 17:45.24 | JunK-Y | external could be thru the manager interface |
| 17:45.25 | *** join/#asterisk asterisknerds (n=asterisk@66.7.124.15) |
| 17:45.25 | asterisknerds | <PROTECTED> |
| 17:46.29 | naitram | JunK-Y: Do you know of an example? All the examples I find look like they are executed from the dial plan? is that right? |
| 17:46.47 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:46.59 | JunK-Y | naitram: browser stuff for agi. |
| 17:47.31 | JunK-Y | http://www.voip-info.org/wiki-Asterisk+AGI |
| 17:47.50 | CCFL_Man2 | what media port range do i need to forward on my router? |
| 17:48.03 | *** join/#asterisk bawb2 (n=bawb2@ip50210.estcmp.ku.edu) |
| 17:48.30 | JunK-Y | CCFL_Man2: the same as in rtp.conf, prolly 10k-20k |
| 17:49.05 | *** join/#asterisk BB|AtWork (n=karl@38.99.18.98) |
| 17:49.07 | BB|AtWork | ~the book |
| 17:49.15 | BB|AtWork | hrm |
| 17:49.17 | naitram | JunK-Y: Looks like you just write command to stdout and that gets caught by asterisk? That seems to easy to me. Is that right? |
| 17:49.27 | JunK-Y | naitram: exactly. |
| 17:49.32 | JunK-Y | ~book |
| 17:49.34 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:49.36 | JunK-Y | ~tfot |
| 17:49.37 | jbot | tfot is probably "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 17:51.58 | BB|AtWork | thanks JunK-Y |
| 17:51.59 | naitram | JunK-Y: OK, will try it. Thanks |
| 17:51.59 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 17:51.59 | Putzz | what r u guys using for faxing these days? hylafax and iaxmodem or spandsp? |
| 17:52.41 | deeperror | Putzz: email & pdf's |
| 17:54.59 | Waverly360 | Anyone here using the new 1.2.18 version of Asterisk? |
| 17:55.07 | deeperror | yes waverly |
| 17:55.19 | Waverly360 | Any problems you've found with it yet? |
| 17:55.24 | Waverly360 | Or anyone for that matter. |
| 17:55.36 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:55.36 | deeperror | i have yet to get ring back tones on inbound calls |
| 17:55.45 | deeperror | may have something to do with my ata configurations though haven't had time to dive into that can of worms |
| 17:55.52 | Waverly360 | Hmm. |
| 17:56.03 | deeperror | everything else seemed to work fine moving from 1.2.8 - .18 |
| 17:56.09 | deeperror | in my setup that is |
| 17:56.15 | Waverly360 | I'm at 1.2.14 right now |
| 17:56.35 | Waverly360 | I'm trying to decide if the move is worth it. I'm having up upgrade my wanpipe drivers, and I figure I'll upgrade zaptel as well. |
| 17:57.00 | deeperror | isn't there some sip vuln you may want to patch up? |
| 17:57.02 | Waverly360 | Not sure I wanna change three variables at once |
| 17:57.02 | Putzz | im on 1.2.13 |
| 17:57.03 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 17:57.39 | Waverly360 | eventually, yes. But I have to be careful. If something starts crashing, it'll be harder to determine what caused it. |
| 17:57.46 | Waverly360 | If I change too many things at once. |
| 17:57.59 | pifiu | if i want to attach a PRI line to a * machine what would i need? |
| 17:58.01 | deeperror | the fair weather sysadmin haha |
| 17:58.35 | Waverly360 | deeperror: well, I'm maintaining 30+ asterisk boxes right now. |
| 17:58.49 | Waverly360 | deeperror: I can't do everything at once. :P |
| 17:58.51 | deeperror | care to share some info on how those all play with each other? |
| 17:58.58 | deeperror | ser ? |
| 17:59.03 | deeperror | load balance? |
| 17:59.12 | Waverly360 | deeperror: they don't. |
| 17:59.16 | deeperror | ohh |
| 17:59.25 | Waverly360 | deeperror: 30+ customers. |
| 17:59.29 | deeperror | makes sense |
| 17:59.41 | deeperror | yea thats a pain right there |
| 17:59.46 | deeperror | every one has different config as well ? |
| 18:00.10 | Waverly360 | deeperror: there are a few that do talk to each other. We have a few customers who have several locations, and the asterisk boxes are tied to each other so they don't have to use pstn lines to call the other offices |
| 18:00.31 | deeperror | nice |
| 18:00.33 | Waverly360 | deeperror: Yep. |
| 18:01.02 | Waverly360 | deeperror: At any rate, I'm having issues with the wanpipe drivers for sangoma. Their rpm build is broken as hell..so I have a bug in with them right now. |
| 18:01.31 | Waverly360 | deeperror: While I'm waiting on them, I'm trying to see what else I need to upgrade. I'm leary of upgrading asterisk just yet. |
| 18:01.34 | *** part/#asterisk pavlicek (n=jpavlice@mail.genevakc.com) |
| 18:01.44 | Waverly360 | deeperror: at least until I fix my crashing problem. |
| 18:02.18 | deeperror | understood |
| 18:02.28 | deeperror | i'm not familiar with |
| 18:02.34 | deeperror | the drivers there |
| 18:02.48 | Waverly360 | deeperror: I'm not as familiar as I want to be. |
| 18:02.58 | deeperror | as with most things |
| 18:03.00 | deeperror | too much change |
| 18:03.01 | deeperror | ha |
| 18:03.17 | deeperror | just got a 1TB nas in the mail |
| 18:03.20 | deeperror | so gonna go play with that here a bit |
| 18:03.39 | deeperror | more to learn |
| 18:03.43 | deeperror | secure ha |
| 18:05.18 | Waverly360 | word :) |
| 18:05.36 | deeperror | were moving a callcenter at the end of this month |
| 18:05.46 | sevard | deeperror: that's pretty sweet. |
| 18:05.49 | deeperror | not only is everything moving there is a major tech upgrade going on at the same time |
| 18:06.08 | deeperror | making the move from analog to voip |
| 18:06.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:06.52 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:06.54 | sevard | deeperror: want any help ? ;) |
| 18:07.06 | deeperror | haha always |
| 18:07.12 | deeperror | thats why i've been living in the channel ;) |
| 18:07.13 | *** join/#asterisk AMUG (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 18:07.17 | sevard | sweet, how much does the position pay? |
| 18:07.32 | deeperror | where are you @ |
| 18:07.45 | sevard | on the moon |
| 18:08.02 | deeperror | our moon base is next to googles |
| 18:08.13 | *** join/#asterisk Strom_M (n=strom@70.141.71.195) |
| 18:08.40 | sevard | google's moonbase is nice. free pop. |
| 18:08.54 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:09.50 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:10.00 | deeperror | lava lamps i hear or some type of alt fuel source very slick |
| 18:10.38 | Mercestes | <PROTECTED> |
| 18:10.42 | Mercestes | then I read the rest of the convo |
| 18:11.22 | deeperror | haha |
| 18:11.31 | deeperror | free pop haha |
| 18:11.42 | Waverly360 | lol |
| 18:12.13 | *** join/#asterisk AMUG (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 18:12.14 | sevard | there's a baby in your boddddyyy |
| 18:12.27 | Waverly360 | umm |
| 18:12.31 | Waverly360 | sure... |
| 18:13.00 | *** join/#asterisk Metfan2007 (n=metfan@189.137.99.142) |
| 18:13.56 | *** part/#asterisk daveburr (n=Miranda@66.7.124.15) |
| 18:14.31 | Metfan2007 | Hi everybody!!! I have a problem with Asterisk behind NAT, I'm trying to connect a Polycom IP301 phone from other office, but I get an chan_sip.c:11178 handle_request_register: Registration from '<sip:2101@192.168.0.8>' failed for '189.137.99.142' - Wrong password |
| 18:14.46 | Metfan2007 | could anybody helpme? |
| 18:14.59 | Waverly360 | Metfan2007: do you have a username and password assigned in your sip.conf file? |
| 18:15.05 | Mercestes | Metfan2007, I think that means that hte phone has the wrong username |
| 18:16.31 | Metfan2007 | yes, in fact that configuration works Ok if I put the phone inside the same network of Asterisk, I just change the nat=yes in the sip.conf |
| 18:16.52 | Metfan2007 | I already tryied with Aastra9133i and PolycomIP301 and the same issue |
| 18:17.12 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 18:18.20 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 18:18.39 | raidenz | hello |
| 18:19.08 | raidenz | does anyone know which sound file is played in meetme for the join/leave sound bytes? |
| 18:22.19 | Katty | right, so... |
| 18:22.31 | Katty | i took care of libpri, and now i'm doing zaptel.. |
| 18:23.59 | Katty | after the ./configure, i did make menuselect...and i'm going to save it now. |
| 18:23.59 | raidenz | any ideas? |
| 18:23.59 | Katty | what's next, or where's the page that tells me what to do next |
| 18:24.00 | afrosheen | raidenz, are there any meetme sound files in /var/lib/asterisk/sounds? |
| 18:24.00 | afrosheen | raidenz, I see alot of things that start with conf- in there |
| 18:24.16 | raidenz | I did see alot of them but I can't find the join/leave sound bytes. I am currently looking throught app_meetme.c but still no luck. |
| 18:26.18 | afrosheen | raidenz, you got me... |
| 18:28.10 | Metfan2007 | I just did some tests with an softphone and I works great, but with a hardphone no |
| 18:28.54 | Katty | do i need to do make install in the zaptel 1.4 directory after make menuselect? |
| 18:29.07 | Qwell[] | Katty: yes |
| 18:29.10 | Katty | thanks. |
| 18:29.36 | *** part/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 18:31.12 | Strom_M | when configuring polycom phone digitmaps, is there a specific order of precedence one needs to place the possible matches in? |
| 18:31.29 | Dr-Linux | Qwell[]: 7935 patch is still for only SVN version? :) |
| 18:31.34 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 18:31.37 | Qwell[] | Dr-Linux: yes, until somebody tests it |
| 18:32.00 | Strom_M | I have "*96xx|*xx" as part of the string, and *96 still sets up a call immediately upon pressing "6" |
| 18:32.34 | wunderkin | *xxT |
| 18:32.54 | Dr-Linux | Qwell[]: if i installed SVN version and configured my phone .. then if i call you using my this phone , then maybe it will in stable version? :) |
| 18:33.14 | Qwell[] | Dr-Linux: maybe, yes |
| 18:33.18 | Strom_M | wunderkin: no, I don't want to have a timeout |
| 18:33.32 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 18:33.33 | Strom_M | I want calls to go through immediately on *xx /and/ on *96xx |
| 18:33.35 | StyleWarz | Hey guys |
| 18:33.37 | Dr-Linux | Qwell[]: hhm.. ok |
| 18:33.39 | Qwell[] | Dr-Linux: 1.4 anyhow...not 1.2 |
| 18:33.43 | Qwell[] | no point in putting it in 1.2 |
| 18:33.49 | wunderkin | Strom_M, i know but i doubt it can do that |
| 18:33.52 | pifiu | how would one go about ordering a PRI from AT&T? |
| 18:33.55 | Strom_M | that's pathetic |
| 18:34.00 | Qwell[] | pifiu: call them, and say "I want a PRI!" |
| 18:34.03 | Dr-Linux | Qwell[]: opss, my all servers are on 1.2.x |
| 18:34.04 | Strom_M | pifu: call AT&T and say "I want a PRI" |
| 18:34.16 | wunderkin | i want to see starz! |
| 18:34.22 | Qwell[] | Strom_M: lack of enthusiasm you've got there |
| 18:34.24 | Dr-Linux | anybody has luck with Cepstral? |
| 18:34.27 | Strom_M | now you see starz |
| 18:34.50 | Qwell[] | mv wunderkin /dev/CA/L.A./Hollywood/ |
| 18:35.01 | Dr-Linux | i installed Cepstral on different asterisk servers "No error" but when i call it does nothing but crashes my asterisk |
| 18:35.22 | wunderkin | (that is off of an old commercial for starz) :P |
| 18:35.24 | StyleWarz | I am connecting 2 calls via the manager interface. Now if i put the whole thing into my default context, the context starts without having the "Channel" (of the caller) active. Any ideas how to tell asterisk to wait until the "Channel" i specified picks up? |
| 18:35.47 | StyleWarz | like it starts with playbacks before the caller picked up |
| 18:35.55 | Qwell[] | StyleWarz: playback with noanswer as an option |
| 18:36.00 | Qwell[] | core show application playback |
| 18:36.37 | Metfan2007 | any clue there?????? |
| 18:37.02 | naitram | so, trying to do this from php script triggered by external event outside dial plan. fputs(STDOUT,"SET CONTEXT MAST\n"), fputs(STDOUT,"SET PRIORITY 1\n"), fputs(STDOUT,"SET EXTENSION 100\n"). I thought asterisk is supposed to read standard output and parse agi commands. Help please |
| 18:37.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:38.00 | StyleWarz | Qwell[]: ah :) |
| 18:38.02 | StyleWarz | thanks |
| 18:38.09 | Qwell[] | StyleWarz: did that work? |
| 18:38.23 | naitram | shouldn't asterisk execute extension 100 on context MAST with priority 1 |
| 18:38.24 | StyleWarz | Qwell[]: it sounds logical, i will test it in 5 minutes :) |
| 18:38.29 | Qwell[] | k |
| 18:38.29 | StyleWarz | Qwell[]: i'll let you know |
| 18:39.13 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 18:40.25 | StyleWarz | Qwell[]: Yes it works (since it starts playing it on the console after i pick up) :) |
| 18:40.33 | Qwell[] | cool |
| 18:40.55 | StyleWarz | Now i got a last problem. Since i switched from OpenPBX1.2rc3 to asterisk 1.4 i don't get any sound over SIP |
| 18:41.09 | StyleWarz | there is no NAT problem since the server is in our colocation with a real ip address |
| 18:41.30 | StyleWarz | this is a real mystery to me |
| 18:41.36 | pifiu | hey qwell! |
| 18:41.38 | pifiu | long time man |
| 18:41.40 | Qwell[] | StyleWarz: probably firewall |
| 18:42.06 | StyleWarz | Qwell[]: The box is a plain freebsd with no firewall at all (since only http and ssh and sip are open) |
| 18:42.23 | StyleWarz | Qwell[]: it worked before with openpbx and that's what bothers me |
| 18:42.27 | StyleWarz | :) |
| 18:42.30 | Qwell[] | well, you were also using 1.2 |
| 18:42.46 | Qwell[] | are your clients behind nat? |
| 18:42.50 | StyleWarz | Nope |
| 18:42.55 | Qwell[] | behind firewalls? |
| 18:42.55 | StyleWarz | I interconnect 2 bellshare connections |
| 18:42.58 | StyleWarz | Nope |
| 18:42.59 | Qwell[] | ahh |
| 18:43.22 | *** join/#asterisk blepsoaf (n=pbaker@nnat-gw.adeptra.com) |
| 18:43.38 | *** join/#asterisk sung (i=sung@realfuckingnews.com) |
| 18:43.41 | StyleWarz | And i don't hear the other channel and i don't hear the Playback we just made :) |
| 18:43.59 | blepsoaf | This is likely a stupid question, but I dare to ask it anyways. Is 1.4 considered production stable? |
| 18:44.23 | StyleWarz | Qwell[]: i my opinion this is only a small switch in sip.conf and i'm done. but i don't find it. |
| 18:45.28 | [TK]D-Fender | StyleWarz: What ports exactly are open on both sides? |
| 18:45.53 | Katty | downloading kernel source takes forever :< |
| 18:45.53 | StyleWarz | [TK]D-Fender: I don't know on the Bellshare side, on mine everything. |
| 18:46.02 | StyleWarz | Katty: Take another mirror :) |
| 18:46.12 | Katty | StyleWarz: hehe, it's not the mirror...it's our horrible bandwidth |
| 18:46.25 | StyleWarz | Well then you'll need a mirror @home :P |
| 18:46.40 | Katty | probably wouldn't hurt :P |
| 18:46.47 | Metfan2007 | pls help, I'm in troubles here :( |
| 18:46.48 | Katty | sigh, 20k/s |
| 18:46.58 | Katty | Metfan2007: first, take a deep breath. |
| 18:47.03 | Qwell[] | Katty: eww |
| 18:47.04 | Katty | Metfan2007: it will get resolved. |
| 18:47.32 | Metfan2007 | ok. now???? |
| 18:47.37 | blepsoaf | lol, is my question so stupid no one will answer it :) |
| 18:47.57 | Katty | Metfan2007: breathe again, and be patient. |
| 18:48.09 | Katty | Metfan2007: if no one answers, ask again after a few minutes. |
| 18:48.16 | Metfan2007 | ok |
| 18:48.23 | Metfan2007 | thanks |
| 18:48.24 | Katty | not everyone in here is actually /here/ |
| 18:48.34 | Katty | someone new might see it and help out (= |
| 18:48.53 | Katty | don't get frustrated! |
| 18:49.06 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 18:49.07 | Katty | i get frustrated enough for the both of us :P |
| 18:49.18 | Katty | stupid 30k/s *grumble* |
| 18:49.27 | Katty | Qwell[]: make my internets faster!! |
| 18:49.53 | *** join/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 18:49.59 | *** part/#asterisk emiquelito (n=evandro@200-170-201-155.core01.spo.ifx.net.br) |
| 18:50.28 | mitcheloc | hello all, could someone help me compile zaptel? :) -> http://pastebin.ca/481252 |
| 18:50.46 | Metfan2007 | Hi everybody!!! I have a problem with Asterisk behind NAT, I'm trying to connect a Polycom IP301 phone from other office, but I get an chan_sip.c:11178 handle_request_register: Registration from '<sip:2101@192.168.0.8>' failed for '189.137.99.142' - Wrong password. I just tried with softphone Ok, PolycomIP301 BAD, Grandstream GXP2000 BAD and Aastra9133i BAD, any clue? |
| 18:50.57 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-5dfbacafb4ae114f) |
| 18:52.27 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 18:52.43 | raidenz | Hi Qwell. |
| 18:53.39 | Katty | mitcheloc: i had that problem awhile back. |
| 18:54.25 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 18:54.28 | Katty | mitcheloc: i finally found out that i did indeed have autoconf.h but it wasn't in the ..directory... it was supposed to be |
| 18:54.54 | Katty | mitcheloc: and for the life me, i never got it resolved, even after manually editing a file (don't recall which one) and removing all traces of the autoconf.h |
| 18:55.08 | Katty | mitcheloc: how i fixed it was instead of using debian etch, i used debian stable. |
| 18:55.14 | mitcheloc | Katty, from some random notes on the net i understood I had to also edit a file or two, but it wasn't clear... |
| 18:55.24 | mitcheloc | Katty, I'm using Fedora Core 6 |
| 18:55.35 | Katty | mitcheloc: but, i was thinking a lil later, that apt-get distro update or upgrade or whichever one that is supposed to fix paths and such might do it. then again.... who knows. |
| 18:55.38 | Siya | re |
| 18:56.24 | Katty | mitcheloc: see if you can locate your autoconf.h and then make a symlink to where it's asking for it |
| 18:56.44 | Katty | in the /lib/kernel/build/include/linuxdirectory |
| 18:57.03 | mitcheloc | Katty, trying it now, just a sec |
| 18:57.03 | Katty | your line 47 |
| 18:57.25 | Katty | i'm compiling zaptel too, but i already did the make menuthingy |
| 18:58.03 | mitcheloc | I'm kind of bummed, I tried this on centos and it's not working there either, apparently something to do with the new .20 kernel |
| 18:58.20 | mitcheloc | I can't locate an autoconf.h anywhere on my system... |
| 18:58.37 | Katty | can you locate autoconf.h? |
| 18:58.44 | Qwell[] | mitcheloc: update |
| 18:58.47 | Katty | after an updatedb |
| 18:59.13 | Katty | also, goshdangitanyhow, how do i get kernel source for 2.8.18-4-686 :< |
| 18:59.27 | mitcheloc | Nope, not finding it, ran "updatedb; locate autoconf.h" |
| 18:59.28 | bkruse | Katty: what os? |
| 18:59.34 | Katty | apparently apt-get installing linux-source-2.6.18 doesn't work! *sob* |
| 18:59.39 | Katty | bkruse: debian. |
| 19:00.12 | naitram | anyone know how to write command strings to asterisk using stdout and php or C |
| 19:00.16 | *** part/#asterisk redax (n=redax@mail.caracom.hu) |
| 19:00.16 | bkruse | Katty: apt-get install linux-headers-`uname -r` |
| 19:00.23 | Katty | bkruse: wait. |
| 19:00.27 | Katty | bkruse: is headers source? |
| 19:00.29 | mog | kaldemar, you dont need source |
| 19:00.32 | mog | just get headers |
| 19:00.33 | Katty | bkruse: i thought that was something else. |
| 19:00.38 | mog | that match your kernel |
| 19:00.43 | bkruse | Katty: you need the headers, assuming your trying to build zaptel? |
| 19:00.47 | bkruse | naitram: asterisk -rx 'command here' |
| 19:00.47 | vicentealencar | When I pass a variable to an AGI script it is passed by value, not by reference, right? |
| 19:00.56 | Qwell[] | mitcheloc: that's fixed in new zaptel... |
| 19:01.05 | Katty | ooh ahh |
| 19:01.11 | Katty | why can't they call it kernel headers then |
| 19:01.20 | Katty | instead of source *grumble* |
| 19:01.25 | mitcheloc | Qwell[]: i just downloaded it yesterday, should i get the svn version? |
| 19:01.35 | Qwell[] | ugh, what kernel version? |
| 19:01.41 | Qwell[] | I bet RH did something stupid - again |
| 19:01.47 | bkruse | Katty: usually the source has the headers packaged |
| 19:01.49 | bkruse | but debain is awesome. |
| 19:03.49 | StyleWarz | bkruse: Debian is the worst shit i've ever seen |
| 19:04.13 | file | mitcheloc: http://www.asterisk.org/developers/get-source |
| 19:04.22 | StyleWarz | Everything based on Linux is stupid ;) As long as it hasn't got a touch of bsd ;) |
| 19:04.27 | Katty | strange. debian is the best i've seen. |
| 19:04.38 | Katty | ! |
| 19:04.43 | mitcheloc | ahh, thaks file, duh |
| 19:05.10 | naitram | bkruse: so that is the way to run the AGI commands from an external app? |
| 19:05.24 | Dr-Linux | i like Cepstral, but it never worked for me :S maybe for anyone? |
| 19:05.25 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 19:05.36 | Katty | here's something i don't get. why does the doc on asteriskguru say i need make linux26 for a debian 2.6 kernel (which i have) but instead i'm doing ./configure and menu select |
| 19:05.53 | Katty | whywhywhy, i don't get it. and don't say it's 1.4 cause this doc is supposedly /for/ 1.4 |
| 19:05.53 | StyleWarz | Katty: ./configure isn't for the linux kernel |
| 19:06.00 | StyleWarz | Katty: on the linuxkernel you do make menuconfig ;) |
| 19:06.14 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
| 19:06.16 | StyleWarz | (and on debian of course first apt-get install libncurses5-dev) |
| 19:06.18 | Katty | Hmmhesays: ! |
| 19:06.21 | Katty | Hmmhesays: hi snob. |
| 19:07.36 | Hmmhesays | Katty! |
| 19:07.36 | Dr-Linux | lol |
| 19:07.36 | Hmmhesays | holy cow, I drunk texted your old phone number the other day |
| 19:07.36 | Katty | Hmmhesays: oops. |
| 19:07.36 | Hmmhesays | LOL |
| 19:07.36 | bkruse | naitram: oh, i see what you mean, no |
| 19:07.37 | bkruse | thats the way to run asterisk cli commands outside of asterisk |
| 19:07.37 | file | Katty: how come I don't have your number! |
| 19:07.37 | Katty | file: you do! |
| 19:07.37 | Qwell[] | Hmmhesays: I totally did that once! |
| 19:07.43 | Qwell[] | Hmmhesays: I got a weird text message the next day, and I'm like...wtf? |
| 19:07.43 | file | bkruse: you can't fix masquerading. |
| 19:07.45 | Katty | file: or do you not have 110 as the ext? |
| 19:07.52 | file | Katty: I do, but not one I can SMS! |
| 19:07.52 | bkruse | file: to far b0rked? |
| 19:07.58 | Katty | file: oooh. |
| 19:08.02 | Qwell[] | then I responded back, something I would only say to my friend, and then she called me, all mad =x |
| 19:08.09 | Hmmhesays | I don't think I have any contact info for you anymore |
| 19:09.29 | pifiu | is 1.39 cents per minute outbound good? |
| 19:09.29 | Qwell[] | pifiu: cheap, good quality, good service |
| 19:09.29 | Qwell[] | pick up to 2 |
| 19:09.29 | Katty | yay, zaptel compiled! |
| 19:09.29 | pifiu | teliax lol |
| 19:09.29 | bkruse | Katty: told ya :D |
| 19:09.30 | Katty | now before i go to the asterisk directory and do make mpg1234 and make and make install and make samples...should there be anything else i need to know? |
| 19:09.30 | Dr-Linux | pifiu: for US? |
| 19:09.30 | pifiu | oh wait i mean vitelity |
| 19:09.30 | pifiu | yeah |
| 19:09.30 | Qwell[] | Katty: yeah, don't do that |
| 19:09.30 | Katty | oh great. |
| 19:09.32 | Qwell[] | Katty: no need to use mp3 for moh.. |
| 19:09.33 | Dr-Linux | pifiu: you can find more cheap than that |
| 19:09.35 | pifiu | im in between vitelity and teliax to switch from IP Communications |
| 19:09.44 | Qwell[] | and if so, you should use native moh with format_mp3 from -addons |
| 19:09.53 | Katty | well, addons is next |
| 19:09.57 | Katty | so i'll skip the mpg part |
| 19:10.00 | Qwell[] | then yeah, don't do that |
| 19:10.00 | Katty | anything else? |
| 19:10.01 | Dr-Linux | Qwell[]: any expereince about Cepstral? |
| 19:10.29 | bkruse | Qwell[]: say no! |
| 19:10.30 | bkruse | :P |
| 19:10.30 | bkruse | its festival, but awesome |
| 19:10.33 | bkruse | and closed source :[ |
| 19:10.34 | mitcheloc | ahh, i'm out of here, thanks guys, it compiled further |
| 19:10.40 | Hmmhesays | da da da hey |
| 19:11.36 | Dr-Linux | bkruse: festival has no good voice, but Cespstral is good, but not working for me :S |
| 19:11.38 | bkruse | ;[ |
| 19:11.38 | bkruse | explain |
| 19:11.38 | *** join/#asterisk chewthis (i=davidheu@196.211.34.2) |
| 19:11.46 | Hmmhesays | ok replacing callerid number in openser is a serious pain |
| 19:12.04 | Hmmhesays | file oh file |
| 19:12.17 | Hmmhesays | you've always helped me with ser'ish stuff |
| 19:12.23 | file | OH NO |
| 19:12.35 | bkruse | Hmmhesays: thats not the REAL file |
| 19:12.44 | Hmmhesays | lol oh? |
| 19:12.52 | Dr-Linux | bkruse: i installed Cepstral engine from http://www.cepstral.com/cgi-bin/downloads?type=1143746987 and configured, then i comple app_cepstral application with asterisk, all fine no error |
| 19:13.00 | bkruse | and. |
| 19:13.07 | chewthis | finaly got it to work receiving faxes to /var/spool/asterisk/fax/88839990.1.tif using analog fxo port on diguim hardware |
| 19:13.09 | bkruse | saytext("cepstral rocks") |
| 19:13.20 | Dr-Linux | bkruse: but now when i run cesptral from dialplan, it crashes my asterisk server |
| 19:13.20 | bkruse | eww |
| 19:13.22 | bkruse | core dump? |
| 19:13.26 | clyrrad | Any of you guys know if the 32 Button Attendant Console (Sidecar SPA932) for the SPA962 phone is working with Asterisk? Specifically I would like to know if the Busy Lamp Field stuff is working so you can see when an extension is in use. Does anyone have experience with this new device? |
| 19:13.26 | bkruse | sure its the right version? |
| 19:13.29 | Hmmhesays | I can't find any good way besides substr_body to change my CID in SER |
| 19:13.30 | Dr-Linux | bkruse: i tried that on different asterisk boxes but same happend |
| 19:13.36 | bkruse | hmm |
| 19:14.00 | Dr-Linux | bkruse: core dump? :S |
| 19:14.08 | bkruse | nvm ;[ |
| 19:14.18 | raidenz | someone needs to seriously update the cepstral application |
| 19:14.38 | Dr-Linux | bkruse: i tried different app_cepstral, never found error while installing them, but same happend when i run it from dialpan |
| 19:14.44 | Dr-Linux | it crashes my asterisk |
| 19:15.08 | chewthis | i need some help mailing the tif file received using asterisk can anyone help please? |
| 19:15.10 | Dr-Linux | when i hangup the call |
| 19:15.17 | *** join/#asterisk ManxPower (n=manxpowe@71-8-63-42.dhcp.leds.al.charter.com) |
| 19:16.03 | Dr-Linux | bkw__: is it possible, that Cesptral 4.2 doesn't support asterisk? :S |
| 19:16.06 | raidenz | I have had cepstral application work but its slooow. It took about 7 seconds just to say one sentence which some calls would hangup. Imaging if its a long weather report. |
| 19:16.40 | raidenz | Imagine |
| 19:16.46 | Dr-Linux | bkruse: ... |
| 19:17.00 | bkruse | Dr-Linux: im not sure what to tell you |
| 19:17.03 | Katty | blah, res_kattyonhold.so |
| 19:17.18 | Dr-Linux | bkruse: okey, np |
| 19:17.19 | bkruse | the ONLY thing I could suspect is something like you are using the app cepstral for 1.4 or 1.2, or an older/newer version of either |
| 19:17.28 | Dr-Linux | http://www.voip-info.org/wiki/index.php?page=App_Cepstral |
| 19:19.09 | raidenz | Does anyone know what the join/leave sound filenames for app_meetme are? I want to change the join/leave sounds but I can't seem to find the names in app_meetme.c and I heard all the conf-* files in /var/lib/asterisk/sounds directory and still nothing. Any ideas? |
| 19:19.09 | chewthis | raidenz: do you have any luck in the faxing in asterisk? |
| 19:19.32 | Dr-Linux | when my asterisk crashes, i see this before crash "Segmantation fault [core]" |
| 19:19.57 | Qwell[] | Dr-Linux: if it's crashing in app_cepstral, you need to yell at the author of that module |
| 19:20.27 | raidenz | chewthis: only using g711u and with a latency of less then < 75ms and only one page at a time. |
| 19:21.00 | Dr-Linux | Qwell[]: i tried different three author's cepstral application, |
| 19:21.11 | Dr-Linux | 1, WIKI ones that i pasted |
| 19:21.20 | chewthis | raidenz: do you know how to mail the tiff file once received? |
| 19:21.45 | raidenz | chewthis: no, sorry. |
| 19:21.47 | Dr-Linux | 2, bkw__ one at >> http://www.freeswitch.org/node/50 |
| 19:22.05 | Dr-Linux | 3, automated.de/... or something |
| 19:22.34 | chewthis | raidenz: you an expert or a biginer like me? |
| 19:23.22 | chewthis | i only wish i heard of asterisk sooner. |
| 19:26.34 | [TK]D-Fender | chewthis: This has nothing to do with Asterisk. Go google up a bash e-mail script somewhere. |
| 19:26.56 | chewthis | what is bash? |
| 19:27.09 | Dr-Linux | borne again shell :P |
| 19:27.19 | *** part/#asterisk mcf3782 (n=mfreeman@209.117.160.3) |
| 19:27.33 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:28.26 | *** join/#asterisk Math` (n=seb@modemcable234.87-70-69.static.videotron.ca) |
| 19:28.43 | [TK]D-Fender | chewthis: Yes, thats right... you actually should acquire some *nix skills.... |
| 19:29.11 | *** join/#asterisk Defraz (n=t0tal@mail.idavada.com) |
| 19:31.33 | syzygyBSD | IRC is *nix right? |
| 19:32.27 | Nugget | IRC is a protocol, not an operating system. |
| 19:32.58 | [TK]D-Fender | Dammit Jim I'm a doctor not a brick-layer! |
| 19:34.05 | Qwell[] | [TK]D-Fender: on that note.. I'm no longer able to say "What are you, a rocket scientist?" |
| 19:34.28 | Qwell[] | all the smart ass rocket scientists around here say "...yes" |
| 19:34.50 | syzygyBSD | you act like rocket science is hard |
| 19:35.13 | syzygyBSD | I mean, it isn't quantum physics |
| 19:36.54 | Qwell[] | you act like quantum physics is hard |
| 19:37.18 | Katty | yay, asterisk installed!! |
| 19:37.19 | syzygyBSD | harder than rocket science |
| 19:37.31 | syzygyBSD | at least we have one of them pretty figured out |
| 19:37.49 | Katty | so, now i'm gonna do addons... anything besides make clean and make install...for now i'm not using h323 or sql or anything |
| 19:38.10 | [TK]D-Fender | "String Theory For Dummies!" - IDC |
| 19:38.12 | syzygyBSD | Katty: you want rxfax/txfax? |
| 19:38.21 | Katty | no, we have a pots line for fax. |
| 19:38.26 | Katty | in case the power goes out. |
| 19:38.35 | Katty | or some other silly thing |
| 19:38.53 | syzygyBSD | heh, do you have a dc inverter to steal power from the phone lines to keep your servers running? |
| 19:38.55 | [TK]D-Fender | syzygyBSD: I could use a working set for 1.2.x / 1.4.x |
| 19:38.58 | Pagautas | Katty: for fax you have to patch rtp.c file |
| 19:39.10 | Katty | Pagautas: i'm uh, not using fax. |
| 19:39.23 | Katty | so, nothing else then? just make clean and make install? |
| 19:39.30 | Pagautas | addons? |
| 19:39.34 | Katty | yes. |
| 19:39.40 | Pagautas | what os? bsd? |
| 19:39.45 | Katty | debian. |
| 19:39.53 | Pagautas | then yes :) |
| 19:39.57 | Katty | [TK]D-Fender: anything besides make clean and make install in the addons directory? |
| 19:40.05 | Pagautas | for bsd it needs to be patched |
| 19:40.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:40.19 | [TK]D-Fender | Katty: Nope... the easiest part of the puzzle |
| 19:40.33 | Pagautas | make menuconfig? :) |
| 19:40.40 | *** join/#asterisk Dovid (i=dovid-la@bzq-88-155-87-189.red.bezeqint.net) |
| 19:41.15 | Katty | [TK]D-Fender: thank goodness <:) |
| 19:42.06 | Katty | [TK]D-Fender: it's saying "The configure script was just executed, so "make" needs to be restarted" |
| 19:42.10 | Katty | [TK]D-Fender: do i run make...again? |
| 19:42.19 | [TK]D-Fender | sure |
| 19:42.42 | naitram | anyone know about the AGI, help please |
| 19:43.33 | naitram | can not execute an agi command from outside asterisk, isn't that what its for? |
| 19:43.47 | Katty | meesha, this is takin awhile. |
| 19:44.32 | Katty | i execute commands at bash, but don't use agi |
| 19:44.33 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 19:44.40 | Katty | so i really don't know any agi, i guess. |
| 19:44.47 | [TK]D-Fender | naitram: No it is not. |
| 19:45.08 | Katty | [TK]D-Fender: what?! now it's telling me to run make install again. is this normal? |
| 19:45.10 | [TK]D-Fender | naitram: AGI it to be executed BY * and * passes it CHANNEL variables for it to act on. |
| 19:46.02 | [TK]D-Fender | Katty: Sure, why not. Always do the usual pile of stuff in a row : ./configure ; make clean ; make (or make menuselect) ; make install |
| 19:46.18 | [TK]D-Fender | Katty: And don't worry about swinging a bat at it a few times. |
| 19:46.34 | [TK]D-Fender | naitram: AGI is for scripted control of a call in progress. |
| 19:46.37 | Katty | i'm skeered. |
| 19:46.42 | *** join/#asterisk mpruett (n=mpruett@24-240-203-83.static.stls.mo.charter.com) |
| 19:46.43 | Katty | microsoft put the fear of...errors... into me |
| 19:47.01 | Katty | but it just made samples, and said "done" so ...dundundun |
| 19:47.08 | Katty | i'm gonna fire it up (= |
| 19:47.10 | *** join/#asterisk thoughtpolice (n=austin@75.111.139.133) |
| 19:47.36 | Katty | do i need to do the modprobe ztdummy, insmod wctdm and the /sbin/ztcfg stuff first too? |
| 19:47.44 | Katty | like on my other server |
| 19:48.13 | naitram | [TK]D-Fender: so, is there a mechanism for hooking into the dial plan from an exernal app? |
| 19:48.54 | [TK]D-Fender | naitram: perhaps you should give some details of exactly what it is you are looking to do. |
| 19:50.07 | poppos | I have a 1800 did how to i tie it to my asterisk box using voipstreet i got the outgoing working with voipstreet? |
| 19:52.12 | naitram | [TK]D-Fender: I want to be able to connect two sip devices on a sip channel by scripting asterisk from php scripts triggered from a client throgh a web server. Sort of like, hit button 1, hit button 2, hit connect and send * the right strings to set up the sip call |
| 19:52.59 | naitram | the buttons are the sip devices as defined in the sip.conf file of course |
| 19:55.21 | StyleWarz | Any ideas what it could be if my asterisk drops me a shitload of "rtp.c:885 ast_rtcp_read: RTCP Read too short" while connecting via sip? after the session gets established everything works fine and the messages stop |
| 19:56.19 | Katty | Alrighty then, so libpri, zaptel, asterisk, and asterisk-addons have all been setup... with no errors. there are no cards in the server right now... so, do i need to modprobe anything, or just fire asterisk -vvvvtoinifityc |
| 19:58.33 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 19:59.09 | StyleWarz | Katty: modprobe your zaptel driver |
| 19:59.16 | Katty | ztdummy. |
| 19:59.17 | Katty | anything else? |
| 19:59.22 | StyleWarz | zaphfc |
| 19:59.27 | StyleWarz | if you got an hfc |
| 19:59.28 | StyleWarz | :) |
| 19:59.31 | *** join/#asterisk Paavum (n=Dorphals@200.71.58.39) |
| 19:59.33 | Katty | hfc? |
| 19:59.35 | Katty | i'm going to presume no |
| 19:59.37 | Paavum | Hello |
| 19:59.38 | Katty | i don't know what that is. |
| 19:59.44 | StyleWarz | Katty: What ISDN Cards do you have? |
| 19:59.48 | Katty | none. |
| 19:59.51 | Katty | there are no cards. |
| 19:59.57 | Katty | they're all currently in the other server. |
| 20:00.00 | StyleWarz | ah "no", i misread that |
| 20:00.04 | Paavum | I'm having a problem with SIP subscriptions and hints |
| 20:00.12 | [TK]D-Fender | naitram: Then you'd be looking for the AMI:Originate command |
| 20:00.16 | Katty | so no /sbin/ztcfg |
| 20:00.20 | Katty | or insmod wctdm? |
| 20:00.45 | Paavum | I cant see the status changing when I do core show hints |
| 20:00.58 | [TK]D-Fender | Katty: ztcfg -vvvv ; asterisk -gvvvc |
| 20:01.09 | vicentealencar | How do I access the arguments passed from the dialplan to an AGI script in the code? |
| 20:01.13 | Paavum | even if the phone is offhook I see it as idle |
| 20:01.14 | Katty | [TK]D-Fender: kk |
| 20:01.20 | *** join/#asterisk Fercho (i=fer@200.63.82.225) |
| 20:01.54 | Fercho | does asterisk work with a dial-up modems? |
| 20:02.09 | Qwell[] | no |
| 20:02.16 | naitram | [TK]D-Fender: Ahh! ok will look into that. Thanks |
| 20:03.03 | Fercho | oh. I'm just looking for a caller id software. (Kallers doesn't work) |
| 20:03.23 | Paavum | Maybe someone can help me set it up properly? |
| 20:03.26 | LeddyHM | ah hah |
| 20:03.29 | LeddyHM | -T |
| 20:03.52 | Katty | does ztcfg just tell the server to load it's zaptel configurations? |
| 20:03.54 | Fercho | so asterisk is designed only for voip, right? |
| 20:03.56 | LeddyHM | yay for timestamps |
| 20:04.00 | Qwell[] | Fercho: no |
| 20:04.05 | Katty | the zaptel.conf file or whatever |
| 20:04.16 | Katty | i don't really get what it's actually doing here, besides the fact is ays 0 channels configured. |
| 20:04.55 | [TK]D-Fender | Katty: It should typically initialize whatever kernel modules are installed where applicable. |
| 20:05.43 | [TK]D-Fender | Katty: And it says 0 channels, because the "make samples" you did installed a set of sample configs which don't actually define any channels. |
| 20:06.48 | *** join/#asterisk Defraz (n=t0tal@mail.gotouchstone.com) |
| 20:06.48 | Katty | gotcha |
| 20:07.49 | Katty | [TK]D-Fender: what's the g on the vvvvc do? |
| 20:07.59 | Katty | [TK]D-Fender: yes, this is 20 questions :P |
| 20:08.12 | ManxPower | "man asterisk" |
| 20:08.22 | *** part/#asterisk Fercho (i=fer@200.63.82.225) |
| 20:08.44 | Katty | ManxPower: you're no fun. |
| 20:09.28 | vicentealencar | How do I access the arguments passed from the dialplan to an AGI script in the code? |
| 20:10.13 | Katty | ManxPower: oh, so if it segfaults, it will kill the app instead of trying to...avoid segfault or something |
| 20:11.00 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 20:12.05 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 20:12.35 | Katty | ManxPower: see, i try to read, but it doesn't sink in :< |
| 20:12.35 | Mercestes | What is the switch for "no segfaults." |
| 20:12.53 | _mm_ | --help |
| 20:12.56 | Mercestes | man asterisk |grep -i 'no segfaults" didn't return anything |
| 20:16.18 | Katty | anyway, console says asterisk is ready. |
| 20:16.26 | Katty | that's my achievement of the day |
| 20:16.38 | Katty | along with 12 pages of extensive documentation over what i've done. |
| 20:17.16 | ManxPower | Katty: -g generate a core dump on crash, IIRC |
| 20:17.40 | Mercestes | yay |
| 20:17.44 | ManxPower | Mercestes: The switch for no segfaults is Asterisk 1.2.15 |
| 20:17.47 | Mercestes | go you. and you did it all by yourself too |
| 20:17.56 | ManxPower | 17 and 18 die about once per day on my systems |
| 20:18.03 | Mercestes | ManxPower, I thought you were giong to say "CCM." LOL |
| 20:18.16 | Mercestes | I'm on .13. I don't have a .15. :( |
| 20:18.16 | ManxPower | Mercestes: no need to be vulgar |
| 20:20.39 | Mercestes | :( I'm sorry |
| 20:20.39 | Mercestes | ... |
| 20:20.39 | Katty | haha |
| 20:20.39 | sevard | ManxPower: .17 and .18 die once per day on your systems?! |
| 20:20.39 | ManxPower | sevard: Yup! |
| 20:21.11 | sevard | that's amazing |
| 20:21.38 | ManxPower | sevard: *I* thought so too. My client wasn't so amazed. |
| 20:22.27 | sevard | he could be shitting golden bricks, did you downgrade? |
| 20:22.42 | afrosheen | ManxPower, hahaha |
| 20:22.55 | ManxPower | sevard: yup! |
| 20:22.55 | afrosheen | ManxPower, tell him it's a magic trick, then he'll be amazed |
| 20:23.06 | ManxPower | back to the version that was stable for us (1.2.15) |
| 20:23.27 | afrosheen | ManxPower, if it's any consolation, asterisk business edition is stuck at a solid 1.2.12 |
| 20:23.37 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 20:23.50 | sevard | afrosheen: I thought the point of upgrading was because of the vulnerabilities |
| 20:24.02 | afrosheen | sevard, from 1.2.11 to 1.2.12, yes |
| 20:24.21 | afrosheen | there was that SIP DOS vuln in 1.2.11 but I don't know about versions higher than 1.2.12 |
| 20:24.54 | sevard | afrosheen: .18 release notes: A recently published security vulnerability in the manager interface |
| 20:25.02 | sevard | afrosheen: - Another recently published security vulnerability in the SIP channel driver |
| 20:25.06 | afrosheen | damn |
| 20:25.13 | afrosheen | .19 release notes: stability fixes |
| 20:25.16 | afrosheen | ;) |
| 20:25.18 | sevard | hahaha |
| 20:25.40 | ManxPower | none of our production Asterisk systems have any access to the internet |
| 20:26.07 | sevard | no soup for you! |
| 20:26.07 | afrosheen | ManxPower, so in production you're strictly PRI or something else right |
| 20:27.00 | afrosheen | ManxPower, yeah life has been good since we went from a bad internet connection and voip to a real PRI and 3 bonded t1's |
| 20:27.13 | afrosheen | users have never been happier |
| 20:27.20 | ManxPower | afrosheen: amazing how well things work when you Do It Right. |
| 20:28.16 | afrosheen | ManxPower, and when You Can Afford It |
| 20:28.18 | afrosheen | ;) |
| 20:28.36 | Paavum | I cant see the status changing when I do core show hints even if the phone is offhook I see it as idle Maybe someone can help me set it up properly? |
| 20:28.38 | ManxPower | If you can't afford it then get a nortel or something. 8-() |
| 20:28.43 | afrosheen | this company went from 45 people to 90 people in less than a month, it was challenging for us |
| 20:28.54 | afrosheen | 2 man IT ninja group |
| 20:28.54 | Paavum | I'm having a problem with SIP subscriptions and hints I cant see the status changing when I do core show hints even if the phone is offhook I see it as idle Maybe someone can help me set it up properly? |
| 20:28.55 | pipwerk | bad internet, bad E1's same thing, all bad ;-) |
| 20:28.58 | cpm | I'll bet |
| 20:30.19 | Mercestes | IT ninjas. ROFL nice |
| 20:31.37 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:32.04 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 20:33.21 | ManxPower | Paavum: you might start out with telling us what phone you are using |
| 20:34.29 | aptura | Manx, I am looking for a used good quality PRI/DSL/Ehternet tester and wanted to get some feedback on some respected name brands and the models you have used. |
| 20:35.12 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:35.39 | ManxPower | aptura: I've not used any. |
| 20:35.50 | aptura | ahh okay |
| 20:35.51 | ManxPower | The closest I have is an analog and T-1 tester (non-PRI) |
| 20:36.02 | aptura | k |
| 20:36.23 | aptura | which brand and model? |
| 20:36.48 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:37.19 | ManxPower | Sage Instruments |
| 20:37.27 | ManxPower | Got it off eBay dirt cheap just before Katrina |
| 20:38.11 | aptura | ohh yea know of them. |
| 20:38.21 | aptura | I need something for some POS installs. |
| 20:39.08 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 20:41.51 | Paavum | ManxPower --> I have Linksys RT31P2 ATAS and a GXP2000 Grandstream |
| 20:42.08 | nowork | hello.. I have setup asterisk and a2billing. when I use IAD for sip incoming calls, as soon as i picked up the phone, it dropped. when i use X-lite , everything is normal . |
| 20:42.17 | nowork | codec is using g711 |
| 20:42.46 | ManxPower | Paavum: I can only help with Polycom |
| 20:43.00 | *** join/#asterisk EricL (n=eric@ool-457de11c.dyn.optonline.net) |
| 20:43.36 | EricL | What do I have to add in the DialPlan to allow for someone who has "mistyped" an extension to be able to "cancel" and dial another extension. |
| 20:43.46 | Paavum | ManxPower --> I think its a problem in thec onfig |
| 20:43.58 | Paavum | ManxPower --> in the extensions.conf or sip.conf |
| 20:43.59 | ManxPower | EricL: you generally don't. |
| 20:44.12 | ManxPower | unless you are talking about an IVR |
| 20:44.33 | nowork | anyone has experence on the difference btw: X-lite & IAD? both on the same ip network |
| 20:44.38 | ManxPower | Paavum: In my experience the phone is usually not correctly configured |
| 20:44.55 | ManxPower | nowork: Integrated Access Devices don't usually run SIP |
| 20:44.56 | EricL | ManxPower: So if someone dials ext. 125 and they meant to dial 135, they have to hang up and call again and be more careful? |
| 20:44.58 | kiscokid | what is IAD? |
| 20:45.17 | EricL | Or wait for the person at ext 125 to transfer them? |
| 20:45.20 | nowork | hm, i use wrong word..sorry.. i am using atcom ag188 |
| 20:45.24 | ManxPower | EricL: from a phone on the system or from a phone on the outside and are they connecting to a voice menu? |
| 20:45.25 | nowork | somehting like linsys |
| 20:45.38 | EricL | Mans:From a phone on the outside. |
| 20:46.05 | ManxPower | EricL: exten => i,1,Goto(startlofyourmenu,1) |
| 20:46.36 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:47.23 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:47.25 | EricL | ManxPower: So I have to put that at the beginning of everyone's exten ? |
| 20:48.23 | ManxPower | EricL: It is easier to make them hangup and call back |
| 20:48.42 | JT | kiscokid: Integrated Access Device |
| 20:48.55 | kiscokid | JT: thanks |
| 20:49.11 | Paavum | ManxPower --> Just for shits and giggles would you take a look at my config? |
| 20:49.13 | nowork | JT, what should I called those small voip gateway? 1port , 2port. sip |
| 20:49.15 | Paavum | Its really small |
| 20:49.17 | *** join/#asterisk renier (n=renier@69.79.111.24) |
| 20:49.17 | JT | usually it's an ATA aimed at telcos, maybe has a couple more features :) |
| 20:49.18 | JT | ATA |
| 20:49.24 | ManxPower | kiscokid: Usually it is a box that allows you to bring a T-1 into it with both voice and data on it and split out the voice and data into different ports |
| 20:51.20 | nowork | so, my question is the inbound call works for x-lite, not for ATA..but outbound calls work for both. |
| 20:51.20 | nowork | and for inbound calls to ATA, ring is normal, just when I pickup the phone, it dropped rightaway. |
| 20:51.20 | EricL | ManxPower: Will putting that line at the beginning of every exten (should I choose to do that) allow someone to break in the middle of ringing another extension and dial someone else? |
| 20:51.30 | ManxPower | EricL: No. |
| 20:52.07 | renier | question about outgoing calls through asterisk: have a tdm02b card on the asterisk server. Both fxs ports are hooked up to lines from the telephone service. when making an outside call from a sip phone, how does asterisk chooses which line/port to use? and will it choose one if it finds the other is in use? |
| 20:52.17 | EricL | ManxPower: Oh, can they break out of voicemail and back into the IVR? |
| 20:52.52 | ManxPower | EricL: Try making the simple stuff work first |
| 20:53.34 | EricL | ManxPower: The simple stuff does work. |
| 20:54.20 | [TK]D-Fender | EricL, Go read up on the "o" and "a" standard extensions |
| 20:55.39 | EricL | [TK]D-Fender: Ok, thanks. |
| 20:55.46 | *** part/#asterisk EricL (n=eric@ool-457de11c.dyn.optonline.net) |
| 20:57.27 | *** join/#asterisk kn0x (n=atlantic@c-67-176-194-29.hsd1.il.comcast.net) |
| 20:59.34 | nowork | hi, anyone can help to use your ata register on my service and make one test call? thanks |
| 21:00.00 | *** join/#asterisk JahFarRight (n=geoffk@host81-158-240-31.range81-158.btcentralplus.com) |
| 21:04.17 | henryv2 | possibly a bit of an optimistic question here, but does anyone have any experience with the Dell 840 as a base unit for an Asterisk installation with Sangoma and Digium cards? http://www1.euro.dell.com/content/products/productdetails.aspx/pedge_840?c=uk&cs=ukbsdt1&l=en&s=bsd |
| 21:04.44 | henryv2 | it seems to have a suitable range of PCI/PCI Express and PCI-X slots to take the cards |
| 21:06.47 | *** join/#asterisk Gpl_Source (n=The_natu@unaffiliated/gplsource) |
| 21:09.28 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 21:09.28 | _VoiceMeUp_COM | 18007951004 |
| 21:09.28 | _VoiceMeUp_COM | can one dial that number ? |
| 21:09.28 | _VoiceMeUp_COM | srpint ticket system |
| 21:09.29 | _VoiceMeUp_COM | i get you entered a number htat cannot be reached |
| 21:09.35 | _VoiceMeUp_COM | from your calling area... i tried callerid 303xxxxxxx and anything i could come up with |
| 21:09.36 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:09.50 | _VoiceMeUp_COM | any one have a contact number at sprint for LOAS's ? |
| 21:14.03 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 21:14.57 | vicentealencar | hello everyone, I guess i need a litte help... I am trying to pass an argument to an AGI script... OK, I already know how to do it: AGI(script.agi|arg1|arg2) |
| 21:15.10 | vicentealencar | But I don't know how I access the values of the argument inside the code |
| 21:15.24 | bkruse | um, what language is your script in? |
| 21:15.28 | vicentealencar | python |
| 21:15.30 | bkruse | thats a script problem, not asterisk, right? |
| 21:15.33 | bkruse | eww, dont know python |
| 21:15.40 | bkruse | how do you reference cli arguments from python? |
| 21:15.42 | bkruse | should be the same way |
| 21:15.47 | vicentealencar | I mean... maybe its an asterisk problem |
| 21:16.09 | vicentealencar | I thought asterisk would pass the arguments through the stdin |
| 21:16.14 | vicentealencar | but it doesn't |
| 21:16.25 | bkruse | no, think of it like a command prompt i believe |
| 21:16.38 | bkruse | so, if it was bash: |
| 21:16.38 | bkruse | #!/bin/bash |
| 21:16.38 | bkruse | echo "first argument is ${1}" |
| 21:17.20 | vicentealencar | hmmmm |
| 21:17.43 | vicentealencar | bkruse, do you think asterisk passes the arguments like it was executing the script from the command line? |
| 21:18.05 | vicentealencar | I mean like this: script.agi arg1 arg2 arg2 ? |
| 21:20.40 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 21:21.18 | raidenz | Does anyone know what the join/leave sound filenames for app_meetme are? I want to change the join/leave sounds but I can't seem to find the names in app_meetme.c and I heard all the conf-* files in /var/lib/asterisk/sounds directory and still nothing. Any ideas? |
| 21:21.42 | *** join/#asterisk andrewc (n=andrewc@dsl254-017-249.sea1.dsl.speakeasy.net) |
| 21:24.04 | bkruse | vicentealencar: yes, it hink |
| 21:24.06 | bkruse | i think* |
| 21:26.11 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:27.16 | defsdoor | my nokia n95 registers ok with asterisk and can recieve calls etc.. but a couple of seconds after it registers asterisk a 400 "Bad Request" from it. the n95 cannot make calls then - says connection error - and ideas ? |
| 21:28.34 | vicentealencar | bkruse, I guess it is not like that |
| 21:28.46 | vicentealencar | at least i've performed a test here and it did not work |
| 21:29.20 | bkruse | hmm, dunno |
| 21:29.22 | bkruse | voip-info it |
| 21:30.27 | vicentealencar | what do you mean by "voip info it" |
| 21:30.29 | vicentealencar | ask them? |
| 21:30.39 | vicentealencar | I've already searched at voip info |
| 21:31.02 | cspot | raidenz: have a look at enter.h and leave.h in same src dir |
| 21:31.09 | vicentealencar | but I didn't find anything except how to pass arguments... Never found anyone who approached the code part |
| 21:32.14 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:33.43 | aydiosmio | what are the IP phones you guys recommend? |
| 21:33.49 | Qwell[] | ~phones |
| 21:34.02 | jbot | [phones] http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom |
| 21:34.02 | aydiosmio | Sonoma? |
| 21:34.02 | Qwell[] | never heard of them.. wine country phone? |
| 21:34.17 | aydiosmio | Snom? |
| 21:34.31 | henryv2 | aydiosmio: i use a SNOM300 phone which is very good |
| 21:35.22 | aydiosmio | Qwell[]: wine country phone, you're so cute. |
| 21:35.35 | Qwell[] | umm...okay |
| 21:36.20 | defsdoor | aydiosmio: I've got aastra 9133i and 480i handset - they seem pretty solid |
| 21:36.24 | aptura | qwell to bad a wifi phone could not come close to any kind of quality rating. Is it just the nature of its highly directional and absorbent EHF or the poor engineering that goes into the phones that give them the bad rap? |
| 21:36.50 | Corydon-w | Oddly enough, I'd put snom neck-and-neck with Polycom, and everything else behind those |
| 21:37.06 | aydiosmio | we got some funky stupid Grandstreams |
| 21:37.07 | Corydon-w | and I'd put Cisco dead last, behind even Grandstream |
| 21:37.09 | Sweeper | snom up with polycom? :V |
| 21:37.10 | Qwell[] | somebody removed my ", everything else, Grandstream" |
| 21:37.17 | aydiosmio | I'm trying to convince my boss not to deploy them |
| 21:37.27 | Sweeper | aydiosmio: TRY HARDER |
| 21:37.37 | aydiosmio | it's not my business |
| 21:37.41 | aydiosmio | and I don't have to support hem |
| 21:37.42 | Waverly360 | FYI, anyone wanting to use the rpmbuild feature for the wanpipe drivers from Sangoma...you have to pass it the --split-rpms as their rpmbuild feature is currently broken. |
| 21:37.45 | aydiosmio | but one ended up on my desk |
| 21:38.03 | aptura | Qwell[] and what about wireless? |
| 21:38.05 | aydiosmio | even the gs buttons are aweful |
| 21:38.08 | Sweeper | aydiosmio: set your ringer to "ode to joy" and turn it all the way up :3 |
| 21:38.19 | Qwell[] | aptura: pretty much all wireless phones suck, for various reasons |
| 21:38.25 | Strom_M | "if you buy these phones, everything you know and love will DIE A SLOW AND AGONIZING DEAH" |
| 21:38.28 | Sweeper | you'll get polycom really fast :3 |
| 21:38.36 | henryv2 | aydiosmio: I also have a grandstream phone but don't rate it much either :-D |
| 21:39.06 | Waverly360 | Grandstreams are the greatest phones EVAR |
| 21:39.06 | Sweeper | I'd put cisco after polycom, but before snom.... |
| 21:39.06 | Waverly360 | ...lie |
| 21:39.06 | Sweeper | snom has shitty displays D: |
| 21:39.16 | henryv2 | whats the display like on the snom 360? |
| 21:39.25 | henryv2 | it looks pretty big but I dont know what it contains |
| 21:39.26 | [TK]D-Fender | <PROTECTED> |
| 21:39.30 | jbot | http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX, Snom, everything else, and finally GrandStream. For info on the latter, type ~gs |
| 21:39.31 | Waverly360 | I don't have much exp with the snoms or ciscos, but the grandstreams really do suck. |
| 21:39.43 | Waverly360 | It's like they just forget to register sometimes |
| 21:39.44 | aptura | I know thay suck :) question is can it be overcome. I can see it more of a big $ issue and engineering then anything else. |
| 21:39.45 | [TK]D-Fender | Qwell[], howwzzzaaatt? :) |
| 21:39.49 | Sweeper | gs firmware is hilarious |
| 21:39.51 | Qwell[] | [TK]D-Fender: better :p |
| 21:39.58 | Qwell[] | ~hs |
| 21:40.00 | Sweeper | every other firmware release breaks vlans |
| 21:40.00 | Qwell[] | ~gs |
| 21:40.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:40.11 | aydiosmio | \ |
| 21:40.15 | Qwell[] | :D |
| 21:40.58 | Waverly360 | anyways, I'm out guys. Peace |
| 21:41.42 | *** join/#asterisk Inkubot (n=fgutierr@200.75.4.10) |
| 21:41.47 | Inkubot | ji |
| 21:41.49 | Inkubot | hi |
| 21:41.57 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 21:42.12 | Inkubot | i finally made work SS7 signalling, with chan_ss7 and libss7 |
| 21:42.19 | Inkubot | now i'm trying to make calls with libss7 |
| 21:42.40 | Inkubot | but, mi softphone rings, but the phone that i'm calling don't ring |
| 21:42.44 | Inkubot | asterisk show me this |
| 21:42.51 | Inkubot | Received error from mtp3 layer: -1 |
| 21:42.57 | Inkubot | !! Unable to handle message of type 0x3 |
| 21:43.14 | Inkubot | with chan_ss7 is the same problem, the phone that i'm calling does not ring |
| 21:43.20 | brodiem | Anyone know if it's possible to use context= in queues.conf or something similar and to NOT take the caller out of the queue? I.e. I want a caller to be able to press a number and simpy play an audio file while still in the queue, whether it be played on top of the MOH, or to stop MOH, play the audio, and return MOH. |
| 21:44.57 | kn0x | could Asterisk handle running as a Session Border Controller w/ Least Cost Routing for approx 700 calls? |
| 21:45.02 | [TK]D-Fender | brodiem, No. Get coding.... |
| 21:45.39 | brodiem | [TK]D-Fender didn't think so.. |
| 21:45.43 | kn0x | see I'm considering using OpenSER + MediaProxy, but I'm not sure how much more effecient / stable that would be.... |
| 21:46.12 | brodiem | [TK]D-Fender periodic-announce it is, lol |
| 21:47.39 | aydiosmio | okay |
| 21:47.48 | aydiosmio | newsflash: these phones aren't even Grandstreams |
| 21:47.54 | aydiosmio | they are sub-Grand. |
| 21:48.47 | Sweeper | Grandscreams |
| 21:48.55 | Sweeper | or Ghettostreams |
| 21:49.01 | Sweeper | that's what we call them :o |
| 21:49.38 | vicentealencar | it worked!!!!! |
| 21:49.53 | renier | anyone with the s450ip phone here? |
| 21:50.50 | vicentealencar | in fact asterisk passes arguments the same way it we pass when we execute the program from the command line |
| 21:51.06 | vicentealencar | it was my fault: I forgot to reload the extensions.conf :) |
| 21:51.16 | aydiosmio | Palmmicro says the MAC address vendro prefix |
| 21:51.23 | aydiosmio | chinese hole in the wall phones |
| 21:54.32 | aydiosmio | http://www.voip-info.org/wiki/view/VanAccess |
| 21:54.44 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 21:54.59 | aydiosmio | straight outta china |
| 21:55.08 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 21:55.16 | aydiosmio | IP0020. |
| 21:55.30 | CrazyTux | Does asterisk already have the a-z sound files recorded? |
| 21:55.39 | Qwell[] | CrazyTux: yes |
| 21:55.43 | aydiosmio | yes but only the capital ones |
| 21:55.46 | Qwell[] | letters/z |
| 21:55.52 | aydiosmio | you need to record your own lowercase |
| 21:55.55 | Qwell[] | aydiosmio: other way around |
| 21:56.04 | CrazyTux | Qwell, how do I go about playing back? |
| 21:56.17 | CrazyTux | exten = xxx,n,PlayBack(letters/z) ? |
| 21:56.17 | aydiosmio | Qwell[]: does that make the joke funnier? |
| 21:56.17 | Qwell[] | Playback(letters/a) |
| 21:56.29 | Qwell[] | aydiosmio: well, the filenames are lowercase |
| 21:56.37 | Qwell[] | doesn't make sense for the sounds to be uppercase |
| 21:56.52 | aydiosmio | thanks for sucking all the fun out of that |
| 21:57.04 | CrazyTux | lol, I saw that a mile away :) |
| 21:57.15 | CrazyTux | "Capital Audio" hmmmm |
| 21:57.29 | CrazyTux | Only thing I could relate to it being true is if it actually said "Capital A" |
| 21:58.45 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 21:58.45 | clyrrad | Any of you guys know if the 32 Button Attendant Console (Sidecar SPA932) for the SPA962 phone is working with Asterisk? Specifically I would like to know if the Busy Lamp Field stuff is working so you can see when an extension is in use. Does anyone have experience with this new device? |
| 21:59.27 | CrazyTux | Qwell, exten => playback-group,1,Playback(letters/d&letters/o&letters/g) would that be legal |
| 22:01.14 | Qwell[] | legal? not sure - you should consult a lawyer |
| 22:01.24 | CrazyTux | Qwell, lol |
| 22:01.31 | aptura | CrazyTux playback-group? |
| 22:01.50 | Qwell[] | CrazyTux: should be fine |
| 22:01.52 | CrazyTux | aptura, the extension... I'm wondering if the Playback() method would be legal |
| 22:02.03 | Qwell[] | CrazyTux: next time, say "valid" :) |
| 22:02.19 | aptura | k just that seemed a little strange for a extension |
| 22:03.33 | CrazyTux | aptura, just doing a quick test. |
| 22:03.43 | CrazyTux | Qwell, what do you guys recommend for like a more true text-to-speech? |
| 22:06.18 | *** join/#asterisk mivck (i=1000@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:07.56 | kn0x | CrazyTux, Cepstral |
| 22:08.33 | kn0x | CrazyTux, it's a commercial product, but I find it really inexpensive ($30/voice) |
| 22:09.58 | Inkubot | bye |
| 22:10.36 | CrazyTux | kn0x, how easy is it to hook in with * ? |
| 22:11.17 | kn0x | CrazyTux, I believe there is a dialplan application, but I just use it with an AGI to generate a file and the play it back |
| 22:11.22 | kn0x | its really simple |
| 22:11.57 | CrazyTux | kn0x, so $30.00 per voice one time? dosen't that seem to cheap? |
| 22:12.08 | kn0x | yeah they have a dialplan app for it so you can do something like Swift("Say this text please"); |
| 22:12.29 | CrazyTux | kn0x, nice. |
| 22:12.33 | kn0x | CrazyTux, theres a demo you can try on their website, I like it www.cepstral.com |
| 22:12.36 | CrazyTux | kn0x, why don't you use it like that? |
| 22:12.47 | CrazyTux | kn0x, why do you use an agi over that? |
| 22:13.03 | CrazyTux | kn0x, nevermind, I know why :) |
| 22:13.18 | kn0x | cuz I didn't know there was a dialplan app for it when I was implementing it |
| 22:13.23 | CrazyTux | kn0x, ah |
| 22:14.41 | robin_sz | also to avoid the possibilty of resource contention deadlock when running multithreaed apps with a pre-emptive scheduler in the kernel |
| 22:15.42 | robin_sz | that was of course complete bollocks, but it sounded impressive :) |
| 22:16.27 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:18.17 | JahFarRight | does anyone know any good downloadable docs like the o'reilly pdf, but NOT pdf, i want something a bit easier to navigate through i hate the damn format? |
| 22:18.53 | kiscokid | Oreilly also has Safari |
| 22:19.09 | kiscokid | where you can see the book in a browser |
| 22:19.38 | JahFarRight | not seen that one yet, i want some docs for offline viewing, but i jsut find it virtualy impossible to work with pdf |
| 22:19.55 | kiscokid | of sorry, Safari is online only |
| 22:20.01 | kiscokid | *oh |
| 22:20.25 | JahFarRight | no problem its useful info anyway some online docs would be handy too |
| 22:20.49 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:21.26 | JahFarRight | im hust starting with artserisk before or any oher telephony software so anything is useful really |
| 22:21.28 | JahFarRight | just |
| 22:21.34 | kiscokid | I'm using the PDF of the second edition of the Oreilly book and I find it quite usable |
| 22:21.51 | JahFarRight | never used before* even |
| 22:22.08 | kiscokid | it is a "rough cut" edition made available before formal publication |
| 22:22.17 | Dimik | wtf is kisco |
| 22:22.22 | JahFarRight | i find it nad on the eyes to read for too long, i can use it but its not prefered |
| 22:22.26 | JahFarRight | bad* |
| 22:22.45 | kiscokid | Dimik: I used to live in Mt. Kisco, New York |
| 22:23.44 | kiscokid | Jah: what I sometimes do is print out one chapter at a time to read offline |
| 22:24.00 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 22:24.09 | JahFarRight | its also a pain to naviagte to different pages, spend more time scrolling that reading, but hey its free at least so won't complain too much, just wish peopel would make alternative formts available |
| 22:25.03 | JahFarRight | yeah i could print infact might end up doing so wiht it, but its a lot of paper i'd rather not |
| 22:25.40 | kiscokid | well, I print only the chapter I'm interested in at the time. Makes it seem smaller |
| 22:26.15 | JahFarRight | yeah thats true, i probalby only need to print more inportat chapter ad just reference the rest |
| 22:26.17 | nowork | what command to see the codec an active call is using... |
| 22:28.11 | nowork | hello? |
| 22:29.20 | *** join/#asterisk Defraz (n=t0tal@mail.gotouchstone.com) |
| 22:31.34 | raidenz | sip show channels |
| 22:32.04 | nowork | raidenz: thank you |
| 22:32.09 | raidenz | no problem. |
| 22:32.54 | renier | caller id question: is the only way to get caller id on an incoming call by looking up the number, or can asterisk read the caller id from the incoming call by itself? (not sure how all this works, mind you) |
| 22:36.44 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 22:38.56 | *** join/#asterisk gbilly (n=gbilly@S01060004e2caa4a2.gv.shawcable.net) |
| 22:39.00 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 22:39.05 | gbilly | Is anyone using Asterisk with Telus business lines? |
| 22:39.29 | raidenz | renier: Calling on incoming calls will be show if the source from the incoming call passes the callerid information. (usually it does) |
| 22:39.42 | KuJaX | I am getting a Buffalo router with Tomato firmware installed for QoS. My question is that I have several Linksys phones that plug into a switch of which our Asterisk server is plugged into. For the Buffalo Router for QoS, do I input the IP address (or MAC address) of the Asterisk Server or each of the phones? |
| 22:39.52 | raidenz | gbilly: no, not yet. |
| 22:39.57 | *** join/#asterisk gmfm (n=hithere@67.60.56.115) |
| 22:40.27 | renier | raidenz: so if the call comes with callerid information, asterisk will pass this on to the sip extension, right? |
| 22:40.45 | raidenz | YES |
| 22:40.51 | raidenz | err.. yes |
| 22:41.02 | renier | ok, ok :) |
| 22:41.30 | raidenz | sorry about the shouting. |
| 22:47.20 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 22:53.00 | gbilly | If I get two lines from my telco with only 1 number, is an X101P sufficient to deal with both lines? |
| 22:53.40 | JT | is it 2 lines or one? |
| 22:54.18 | gbilly | looks like a regular modem (2 rj11 - 1 telephone, 1 line) |
| 22:55.57 | JT | i'm not asking about the x101p |
| 22:56.03 | JT | i'm asking about your phone lines |
| 22:56.16 | JT | the X100P is a single line and discontinued |
| 22:56.33 | gbilly | I don't know yet - I have asked for two lines with only one number |
| 22:56.42 | JT | i don't think digium ever put out an X101P, i think that's just the name some other companies have used |
| 22:56.48 | JT | line hunt? |
| 22:56.55 | raidenz | 101P is a *clone* |
| 22:57.00 | raidenz | I have a few |
| 22:57.12 | JT | X10*P are rubbish |
| 22:57.14 | *** join/#asterisk mindCrime (i=chatzill@nat/redhat/x-09c6b6ac600755a3) |
| 22:57.17 | raidenz | haven't used them in years. |
| 22:57.31 | gbilly | what is a good interface for my situation |
| 22:57.39 | gbilly | 1 number, 2 lines |
| 22:57.53 | JT | TDM400P |
| 22:57.56 | JT | gbilly: line hunt? |
| 22:58.16 | gbilly | JT: not sure what you are asking - I am very new to this |
| 22:58.20 | raidenz | TDM400P or 2 x X10xP cards :-P |
| 22:58.40 | JT | line hunt is where incoming calls can come in over multiple physical analogue lines |
| 22:59.00 | JT | a couple of linksys/sipura ATAs would be better than an X100P |
| 22:59.03 | gbilly | I think that's what I need |
| 22:59.18 | gbilly | ability for two incoming calls to be handled simultaneously |
| 22:59.20 | JT | i try to avoid analogue :) |
| 22:59.24 | gbilly | ok |
| 22:59.25 | JT | it's evil |
| 23:00.03 | gbilly | ISDN is the next step up (digital?) |
| 23:00.22 | JT | yes |
| 23:00.58 | JT | isdn refers to a whole bunch of different digital services using out of band Q.931/ISUP signalling |
| 23:01.21 | gbilly | do I need to request multiple lines? |
| 23:02.47 | *** part/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 23:03.05 | *** join/#asterisk Math` (n=seb@modemcable037.229-56-74.mc.videotron.ca) |
| 23:03.27 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 23:03.29 | JT | gbilly: ? |
| 23:05.58 | gbilly | JT: I guess my question is - what is the best procedure for having asterisk handling one number, with two lines for simultaneous use? |
| 23:06.40 | Math` | gbilly: the "normal one" ? just ring both extensions and dialout using that did |
| 23:07.18 | gmfm | anyone have a copy of the Telemarketer Torture sound files referenced by http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture ? I would be willing to host a mirror of it so that page works correctly |
| 23:09.26 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:10.15 | JT | gbilly: best practice in what regards? you're not being very specific at all |
| 23:11.11 | CrazyTux | Hey guys I'm setting up voicemail.conf in a database on 1.4.4, do I have to use ODBC to connect to a mysql db or what are my other options? |
| 23:11.51 | Math` | use res_config_mysql |
| 23:11.54 | Math` | in asterisk-addons |
| 23:12.41 | CrazyTux | Math`, thanks :) |
| 23:14.24 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:15.23 | gbilly | JT - I want to setup a 2 line PBX - what do I order from my telco? 2 analog lines? 2 isdns? I don't really understand how this works. |
| 23:16.04 | JT | well |
| 23:16.19 | JT | for ISDN, what's you'd want is a basic rate interface, BRI |
| 23:16.31 | JT | but if you're in america, that might be hard to get for a good price |
| 23:16.36 | Math` | gbilly: get a voip provider that allows you 2 concurrent calls |
| 23:16.48 | JT | a BRI has 2 B channels (voice) and 1 D channel (signalling) |
| 23:16.55 | kiscokid | gbilly you can use Voicepulse as a SIP or IAX trunk |
| 23:17.01 | JT | Math`: that's not really a good suggestion for reliability |
| 23:17.15 | gbilly | reliability is king |
| 23:17.15 | JT | you should always have at least 1 real telco line, especially in business |
| 23:17.20 | gbilly | I'm in Canada |
| 23:17.25 | Math` | gbilly: where in canada |
| 23:17.30 | gbilly | BC |
| 23:17.32 | JT | gbilly: yeah bri is probably not cheap then |
| 23:17.45 | Math` | whats the ilec in bc |
| 23:18.00 | Math` | (main telco) |
| 23:18.05 | gbilly | Telus |
| 23:19.30 | kiscokid | JT: how would you use that single telco line? |
| 23:19.30 | CrazyTux | Math`, where is the latest SVN to grab it? |
| 23:19.31 | kiscokid | if you were going for a VOIP mostly solution |
| 23:19.33 | JT | kiscokid: gbilly wants 2 lines anyway |
| 23:19.57 | JT | kiscokid: but i don't go for solutions based primarily on VoIP over Internet |
| 23:20.23 | JT | VoIPoI should only be a supplement to real lines |
| 23:20.53 | kiscokid | JT: supplement how? |
| 23:21.20 | JT | kiscokid: cheap outbound calls maybe, connecting a branch office or telecommuter |
| 23:21.43 | JT | for most businesses, their phone lines are pretty important |
| 23:22.35 | kiscokid | JT: how reliable would you expect a purely voip solution to be? |
| 23:22.55 | JT | depends on how reliable your Internet is |
| 23:23.00 | JT | and how reliable your ITSP is |
| 23:23.27 | kiscokid | well my Internet is very reliable |
| 23:24.14 | [hC] | wow, im trying trixbox out and its decent for some things, but others are so asterisk 1.0 shitty dial plan mentality the way they're structured.. blech. |
| 23:26.40 | JT | kiscokid: so it's up and running, has low latency, no jitter, sufficient upstream bandwidth, and good routes to your itsp 24/7? |
| 23:27.04 | kiscokid | JT: so far |
| 23:27.28 | kiscokid | I suppose that could change |
| 23:28.31 | JT | also with just web browsing sort of use, you won't really notice jitter |
| 23:29.01 | kiscokid | well its an office with 20 windows machine and a bunch of servers |
| 23:29.31 | kiscokid | I can set up QOS on the internal switch and firewall |
| 23:30.47 | JT | that won't help with Internet problems |
| 23:31.03 | kiscokid | true |
| 23:31.15 | JT | just don't get rid of the PSTN, especially with that many users |
| 23:31.35 | *** join/#asterisk Black-Kakugane-1 (n=lonewolf@c-71-206-157-165.hsd1.va.comcast.net) |
| 23:32.01 | gbilly | JT - does ISDN offer the same reliability as POTS |
| 23:32.16 | kiscokid | well, we're moving and management is talking about not using PSTN at the new place |
| 23:32.26 | JT | gbilly: for the most part |
| 23:33.04 | JT | gbilly: the only thing POTS has over ISDN is that you can just plug a simple analogue phone into POTS with absolutely no mains or battery power |
| 23:33.58 | gbilly | I think I want the line(s) to connect to asterisk so this shouldn't be an issue right? |
| 23:34.01 | florz | JT: Well, you can simply plug in a "simple ISDN phone", too :-) |
| 23:34.08 | JT | kiscokid: it's your job to talk management out of making stupid mistakes |
| 23:34.18 | JT | florz: those usually require a local power feed |
| 23:34.23 | JT | and are only available for BRI |
| 23:34.34 | florz | JT: Well, _simple_ ones not :-) |
| 23:34.37 | ManxPower | I always keep a POTS line or two around for fax and credit card machines anyway |
| 23:34.39 | JT | hrm |
| 23:34.47 | JT | ManxPower: indeed |
| 23:34.52 | florz | JT: At least noch for basic functionality comparable to POTS. |
| 23:34.56 | florz | s/ch/t/ |
| 23:34.57 | ManxPower | and 911 of course |
| 23:35.20 | kiscokid | JT: I would if I could prove that a strictly voip solution would be unreliable. How can I prove such an assertion? |
| 23:35.39 | JT | gbilly: what do you mean? |
| 23:35.50 | JT | kiscokid: i dunno, it's plain obvious to tech people |
| 23:36.06 | ManxPower | JT: well obvious to NETWORKING people |
| 23:36.06 | JT | kiscokid: the multiple additional points of failure with VoIPoI |
| 23:36.19 | florz | gbilly: Erm, you should realise that there isn't really any POTS anymore - it's basically just a matter of where the A/D conversion and back is done, but apart from that, it's all ISDN |
| 23:36.19 | JT | you could summarise them in a list |
| 23:36.37 | ManxPower | kiscokid: 1) If there is a problem between you and your ITSP, who do you call. Your ISP's upstream is not going to listen |
| 23:37.13 | *** join/#asterisk mindCrime (i=chatzill@nat/redhat/x-9055919cbda5d6aa) |
| 23:37.26 | kiscokid | JT: well I'm usually a Windows Sysadmin and setting up Asterisk is a new thing for me so I don't have much experience with it. Is there a news article or some other authoritative source that I can use? |
| 23:37.30 | ManxPower | florz: in the usa/canada there is pots service at least to the CI |
| 23:37.33 | ManxPower | CO |
| 23:37.53 | florz | ManxPower: But only from the customer "to the CO", no? |
| 23:38.10 | JT | ManxPower: the rest of the network is ISDN/digital |
| 23:38.24 | ManxPower | florz: correct. Well the rest of the network is digital/ss7 |
| 23:38.36 | JT | it's only analogue on the last mile, except perhaps "bumf*ck idaho" ;) |
| 23:38.39 | ManxPower | But generally you only need to TEST to the CO |
| 23:38.43 | kiscokid | ManxPower: I could get two ITSPs for outgoing calls |
| 23:38.53 | JT | kiscokid: and 2 isps |
| 23:38.57 | ManxPower | kiscokid: POTS Just Works |
| 23:39.02 | JT | kiscokid: using totally different access technologies |
| 23:39.06 | florz | ManxPower: Well, thus the reliability of the network shouldn't make a difference. |
| 23:39.18 | JT | kiscokid: make sure you have plenty of backup power too |
| 23:39.27 | JT | kiscokid: then you'll approach the reliability of a PSTN service |
| 23:40.22 | JT | kiscokid: also, make sure your server has redundant everything |
| 23:40.40 | JT | preferably that should be serverS ;) |
| 23:40.42 | Paavum | Hello... |
| 23:41.05 | Paavum | Does anybody have any experience with SIP BLFs and a Grandstream GXP2000? |
| 23:41.16 | JT | kiscokid: now you're starting to get pretty damn close to a POTS + simple handset reliability ;) |
| 23:42.04 | Paavum | I think I missed something but I cant seem to figure out exactly what |
| 23:43.27 | Paavum | I mean... when I do a core show hints I see all phones idle |
| 23:43.34 | Paavum | even though I'm using ext 300 |
| 23:44.39 | Black-Kakugane-1 | what version of * ? do you have limitonpeers and call-limit set? |
| 23:44.48 | Paavum | I have * 1.4 |
| 23:44.57 | Paavum | 1.4.3 |
| 23:45.04 | Paavum | call-limit=20 |
| 23:45.40 | Black-Kakugane-1 | hmmm, anyone else have any ideas? |
| 23:46.10 | Paavum | Maybe I messed my configs |
| 23:46.14 | Paavum | care to look at them? |
| 23:46.24 | Paavum | there are about 9 extensions |
| 23:46.39 | Paavum | lemme pastebin it |
| 23:46.40 | Black-Kakugane-1 | i'm about 5 sec from leaving, so sorry |
| 23:46.44 | Black-Kakugane-1 | maybe someone else can |
| 23:48.26 | Paavum | Black-Kakugane-1 ... it wont take u more than 3 secs to review it ;) |
| 23:48.31 | Paavum | Reaaaly small trust me |
| 23:51.02 | Paavum | http://pastebin.ca/481776 |
| 23:52.58 | gmfm | JT: I'll have you know that Qwest provides better service here in idaho than SBC/AT&T ever did in southern california |
| 23:53.42 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 23:54.01 | JT | gmfm: but are you in the middle of nowhere? :) |
| 23:54.08 | *** join/#asterisk sysko (n=sysko@modemcable113.9-203-24.mc.videotron.ca) |
| 23:54.56 | CrazyTux | Math`, still around? :) playing with this extconfig.conf and voicemail stuff with the addon |
| 23:55.01 | Math` | yeah |
| 23:55.40 | gmfm | JT: no... and I will grant you that outside of the Boise area, Idaho has lots and lots of nowhere |
| 23:55.47 | JT | heh |
| 23:56.03 | JT | i'm not that familiar with it, but i'll take your word for it |
| 23:56.44 | Paavum | Does anybody have any experience with SIP BLFs and a Grandstream GXP2000? I think I missed something but I cant seem to figure out exactly what I mean... when I do a core show hints I see all phones idle |
| 23:57.13 | CrazyTux | Math`, find_engine: Realtime mapping for 'voicemail' found to engine 'mysql', but the engine is not available <--- I get that error? |
| 23:57.30 | CrazyTux | Math`, how can I debug why its 'not available' |
| 23:57.59 | Math` | did you compile asterisk-addons with res_config_mysql? |
| 23:58.10 | CrazyTux | Math`, yea I have the .so file |
| 23:58.26 | Math` | is it loaded? |
| 23:58.42 | Math` | *CLI> module show like res_config_mysql.so |
| 23:59.14 | CrazyTux | Math`, 0 modules loaded. |
| 23:59.28 | Math` | > module load res_config_mysql.so |
| 23:59.39 | Math` | should work better that way |
