| 00:03.46 | terrapen | hrm maybe i need a SetCallerIDNAme |
| 00:03.49 | terrapen | err CIDNAME |
| 00:05.06 | terrapen | bingo. |
| 00:05.20 | ManxPower | terrapen, Don't use quotes in callerid stuff |
| 00:05.44 | terrapen | thx, i got it working with a SetCIDName() |
| 00:05.51 | ManxPower | terrapen, NuFone (like most VoIP providers) does not provide callerid name information on their toll free numbers |
| 00:06.05 | terrapen | i guess the IP500 does its own CallerID Name based on the CallerID number |
| 00:06.29 | ManxPower | terrapen, if that number is in the caller list or the directory, yes, it will. |
| 00:07.01 | terrapen | the manual doesn't mention it but i'm all good now |
| 00:07.05 | terrapen | maybe i'll add it to the wiki |
| 00:11.32 | xkev | you can configure that behavior, I believe |
| 00:15.52 | Moc | hi all |
| 00:17.34 | *** join/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 00:17.52 | Moc | ip500 show caller only, then once pickup will show caller id |
| 00:18.00 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 00:18.38 | xkev | you mean caller name only, then shows name and number? |
| 00:18.52 | ctooley | is there a way to cluster linux so that Asterisk, using SIP peering to a provider, can assign 10,000 DID's to one SIP account and have the load spread over 10 servers? |
| 00:19.01 | xkev | SER |
| 00:19.06 | Tall-guy | So, cause I'm a cheap bugger, I replaced a Ati 9000 video card in my Asterisk box with a cheap Geforce 2 card......now Asterisk is freaking out with an Error code 1 - Died...restarting......can anyone tell me "Why" |
| 00:20.20 | xkev | ctooley, so a call would come to your ser proxy, that would dish them out to the backend servers |
| 00:20.34 | ctooley | xkev, perfect |
| 00:20.36 | xkev | ser is challenging to configure at first, but once you understand, it's not so bad |
| 00:21.22 | terrapen | strangely, the IP500 only displays CallerID |
| 00:21.24 | terrapen | err |
| 00:21.29 | terrapen | CallerID Name |
| 00:21.35 | terrapen | not CallerID Name + Number |
| 00:22.29 | Moc | terrapen, it does once the call is answered |
| 00:22.35 | terrapen | yup |
| 00:22.51 | terrapen | im going to monkey around with the SetCIDName application |
| 00:23.09 | terrapen | and see if i can get that to set it to Name <Number> |
| 00:23.13 | Moc | if the Caller Name is empty, it should show the number |
| 00:24.03 | terrapen | i want it to show number and, if name != null, name |
| 00:25.34 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 00:26.04 | wwalker | xkev So would ser "handoff the call" or does all the call traffic go thru the ser from start to finish, while the computation occurs on one of a group of back end servers? |
| 00:26.46 | terrapen | exten => 8668177667,1,SetCIDName(${CALLERID}) |
| 00:26.51 | terrapen | that *should* do it |
| 00:27.48 | terrapen | would someone with working callerid name+number mind calling me? 1-866-817-7667 |
| 00:29.37 | *** part/#asterisk Tall-guy (tall-guy@hssxrg207-195-103-110.sasknet.sk.ca) |
| 00:31.27 | *** join/#asterisk hacim (micah@micha.hampshire.edu) |
| 00:31.28 | harryvv | Tall, thats just weird ;) Why did you change the vidio card? |
| 00:31.47 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 00:32.05 | hacim | when I try to compile zaptel, i get this error: |
| 00:32.05 | hacim | make -C /lib/modules/`uname -r`/build SUBDIRS=/home/micah/working/asterisk/zaptel modules |
| 00:32.08 | hacim | make[1]: Entering directory `/usr/src/kernel-headers-2.6.7-1-k7' |
| 00:32.11 | hacim | make[1]: Makefile: No such file or directory |
| 00:32.14 | hacim | make[1]: *** No rule to make target `Makefile'. Stop. |
| 00:32.24 | guugmember | hello guys, is there a way I can make my asterisk box a sniffer for all the incoming calls that are handled by another pbx? |
| 00:32.28 | hacim | this is a debian machine |
| 00:32.57 | terrapen | sure |
| 00:33.29 | ctooley | ok so I'll ask the dumb question, where do I get a SER server |
| 00:34.23 | *** join/#asterisk Kumbang (~ecvs@dsp.paume.itb.ac.id) |
| 00:34.46 | terrapen | http://lists.digium.com/pipermail/asterisk-biz/2004-November/001287.html |
| 00:34.49 | terrapen | there's my answer |
| 00:34.55 | xkev | ctooley iptel.org |
| 00:35.03 | guugmember | terrapen, to my question? |
| 00:35.11 | xkev | iptel.org/ser/ to be exact |
| 00:35.13 | terrapen | http://lists.digium.com/pipermail/asterisk-biz/2004-November/001288.html |
| 00:35.14 | terrapen | no |
| 00:35.16 | terrapen | to my question |
| 00:35.41 | guugmember | ahh, ok |
| 00:35.46 | harryvv | This problem keeps comming up and is preventing my zap from dialing wcfxo: out of space to write register 05 with 08 anyone seen this before? I thought I had this fixed. |
| 00:36.57 | xkev | ctooley, SER is your typical SIP proxy. you can also look at the vocal suite: www.vovida.org/vocal/ |
| 00:37.22 | xkev | ctooley, I use SER in front of asterisk to divert presence subscriptions to the phones and send everything else to asterisk (unless it comes from asterisk, then it just routes it along) |
| 00:37.47 | netsurfer | hey harryvv |
| 00:37.48 | harryvv | I found a partial solution to this problem. |
| 00:37.50 | ctooley | xkev, we're going the wrong way, we're not going to have any SIP phones |
| 00:38.07 | xkev | no you don't need sip phones for ser, I'm just spouting an example |
| 00:38.11 | harryvv | netsurfer, I need to force the vidio and zap there own irq adresses. |
| 00:38.16 | ctooley | the only SIP we've got is one connection to our provider, all of our calls come in on that SIP channel and go back out on it. |
| 00:38.24 | xkev | http://www.voip-info.org/wiki-SER+load+balancing |
| 00:38.30 | netsurfer | see priv msg harry |
| 00:38.31 | ctooley | I'm reading that now. |
| 00:38.40 | xkev | erm wait, not a good page :) |
| 00:39.32 | xkev | http://lists.digium.com/pipermail/asterisk-dev/2004-May/004438.html |
| 00:40.36 | fa | hu? |
| 00:41.43 | ctooley | xkev, so round robin is going to be my best bet? |
| 00:42.18 | xkev | well hrm. a can of worms opens up... |
| 00:42.54 | xkev | let's say you get an INVITE and you use a t_relay() to send it to one of 10 servers. if a BYE comes in, SER won't know where it was at |
| 00:45.04 | xkev | what is asterisk's job in this configuration? |
| 00:45.29 | ctooley | to accept the call, run an AGI, make some decisions, and route the call back out the SIP channel. |
| 00:45.53 | xkev | define route the call back out |
| 00:46.03 | ctooley | bridge the call. |
| 00:46.05 | xkev | btw, you might want to look at res_perl for high load stuff |
| 00:46.28 | ctooley | Ok, so forget 10,000, lets call it 1,000,000 |
| 00:46.50 | eKo1 | ctooley: Looks like you need a Session Border Controller. |
| 00:46.51 | ctooley | res_perl only buys me what I can get on one server, I need to spread it out. |
| 00:47.00 | xkev | right, I'm just making a side note |
| 00:47.17 | xkev | lousy udp complicating things |
| 00:47.20 | ctooley | xkev, thanks, I do appreciate that, bandwidth is a bigger issue though, not just CPU |
| 00:48.31 | *** join/#asterisk itnomad (~jackal@199.89.146.110) |
| 00:48.57 | *** join/#asterisk kaitseb (~sadie@aaf66.warszawa.sdi.tpnet.pl) |
| 00:49.23 | xkev | http://www.voip-info.org/wiki-Vovida.org+load+balancer |
| 00:49.52 | *** join/#asterisk guugmember (~nachoramo@168.234.226.39) |
| 00:49.54 | xkev | do you really need asterisk or just a complex SIP mangler? |
| 00:50.02 | tzanger | ewrd, y'all |
| 00:50.04 | tzanger | uh |
| 00:50.05 | tzanger | werd even |
| 00:50.11 | xkev | w0rd ^ |
| 00:50.54 | guugmember | how can i integrate an asterisk box to an avaya pbx so my asterisk will record all the calls? |
| 00:51.04 | xkev | Stickyness is computed by hashing one of either the CallID, To, |
| 00:51.05 | xkev | From, or SIP URI. |
| 00:51.34 | xkev | http://www.vovida.org/downloads/loadbalancer/README_lb-1.0.0.txt\ |
| 00:51.37 | xkev | sans-\ |
| 00:51.42 | ctooley | that actually looks pretty good (Vovida) |
| 00:52.00 | ctooley | xkev, both |
| 00:52.02 | xkev | the vovida vocal suite is very robust. it's basically a sip soft switch |
| 00:52.17 | ctooley | a SIP mangler with some other Asterisk things |
| 00:53.00 | ManxPower | guugmember, I don't think anyone has ever done that. Welcome, trailblazer! |
| 00:53.32 | ManxPower | guugmember, Be sure to write up your experience when you figure it out! |
| 00:53.43 | *** join/#asterisk ROM_Man (rom_man@mike.netrom.com) |
| 00:54.27 | xkev | guugmember, sounds like you need calls to come into asterisk, then dial the same on the avaya pbx (and vice versa).. ie: put it between the PRI lines feeding the pbx |
| 00:54.55 | xkev | ..that handles outside calls. otherwise, we're getting into avaya land and not many here will be lords of the avaya :) |
| 00:55.05 | *** join/#asterisk Hmmhesays (~Hmmmhesay@66.173.103.108) |
| 00:56.09 | guugmember | ManxPower, nobody, weird, we are working with an Avaya distributor and want to test asterisk as a solution for some define jobs |
| 00:56.14 | Hmmhesays | I'm having trouble setting my callerid in info blank, should this work exten => _X.,1,setglobalvar(CALLERID=) |
| 00:56.14 | Hmmhesays | ? |
| 00:56.28 | xkev | SetCallerID() |
| 00:56.52 | Hmmhesays | _X.,1,SetCallerID() ? |
| 00:57.07 | xkev | ..herm but blanking it? /me tries |
| 00:57.39 | Hmmhesays | yeah, i'm trying to test out this farking fxs unit, it's supposed to not let anonymous calling, but I got no joy, sending no callerID |
| 00:57.43 | Hmmhesays | over SIP |
| 00:58.08 | *** join/#asterisk Legend` (~legend@24.244.142.133) |
| 00:58.32 | guugmember | if i want to display the caller id on my telephones, i need my telephone provider to send me that information within the call right? |
| 00:58.49 | xkev | hmmmhesays, yes SetCallerId() ; just like that with no args, will clear it |
| 00:59.05 | Hmmhesays | indeed i just tried it xkev, thanks for the info |
| 00:59.11 | xkev | hehe |
| 00:59.14 | Hmmhesays | this box doesn't block unavailables either |
| 00:59.22 | Hmmhesays | POS |
| 00:59.42 | Hmmhesays | guugmember: yes |
| 01:00.17 | guugmember | anyone here with avaya experience interested in giving professional services |
| 01:00.18 | hacim | I got a Analog Line/FXO pci card I just installed and am wondering how I can get it to work with asterisk? |
| 01:00.25 | guugmember | integrating avaya with asterisk? |
| 01:01.41 | guugmember | Hmmhesays, just bein a consultant |
| 01:01.54 | Hmmhesays | Nothing sexy, like a hooker? |
| 01:02.01 | guugmember | Hmmhesays, when we have questions about integrating Asterisk with other PBXs |
| 01:02.10 | Hmmhesays | what kinda pbx's? |
| 01:02.12 | Hmmhesays | legacy? |
| 01:02.33 | guugmember | Hmmhesays, not legacy, Avaya IPOffice 2.1 |
| 01:02.37 | itnomad | quit |
| 01:02.42 | Hmmhesays | ahhhh |
| 01:02.44 | *** part/#asterisk itnomad (~jackal@199.89.146.110) |
| 01:03.13 | guugmember | Hmmhesays, we are working with an Avaya distributor and they want to test asterisk as a solution for some define jobs |
| 01:03.29 | guugmember | for some especific tasks |
| 01:03.33 | Hmmhesays | i see |
| 01:03.34 | silik0n | guugmember check your messages |
| 01:03.45 | Hmmhesays | i've done some integration with legacy stuff |
| 01:03.49 | silik0n | PMs that is |
| 01:04.26 | Hmmhesays | oooooohhh I think i just figured out what "block anonymous callers" does on this pile of hell |
| 01:05.31 | Hmmhesays | grrrr, they need to rename that to .... "i'm a useless feature, do not check me" |
| 01:07.36 | bjohnson | guugmember: you should post to the bsuiness mailing list |
| 01:10.12 | silik0n | ewww yuck |
| 01:10.30 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 01:13.38 | ManxPower | hacim, What specific card? |
| 01:13.52 | *** join/#asterisk mr_zack (zack@adsl-70-241-27-240.dsl.hstntx.swbell.net) |
| 01:14.01 | mr_zack | <PROTECTED> |
| 01:14.11 | mr_zack | the sound is distinguishable, but i think the pitch is too high, any suggestions? \ |
| 01:14.18 | hacim | ManxPower: I just found that I need zaptel, and the wcfxo driver, so I am putting that stuff together |
| 01:14.19 | *** join/#asterisk boukensya (~boukensya@amtech.arach.net.au) |
| 01:14.33 | tzanger | mr_zack: pitch shift? Are you playing back at 8kHz? |
| 01:14.39 | mr_zack | yeah |
| 01:14.47 | xkev | ctooley, you may also want to look at vocal's redirect server. it sends 302 redirects instead of trying to track everything. your sip provider should be able to observe those, and contact each asterisk box directly |
| 01:14.50 | mr_zack | is ulaw linearly sampled? |
| 01:14.56 | tzanger | mr_zack: IIRC, yes |
| 01:15.06 | tzanger | mr_zack: you're not asking about interleaving are you? |
| 01:15.13 | boukensya | hey guys... I am trying to figure out what port asterisk uses so I can foward it on the router. |
| 01:15.13 | mr_zack | channel interleaving? |
| 01:15.22 | mr_zack | no, i just have a straight ulaw stream that i'm trying to decode |
| 01:15.25 | boukensya | 680? |
| 01:15.37 | boukensya | or 1720 |
| 01:15.38 | tzanger | mr_zack: right ... or did you mean linear vs logarithmic sampling |
| 01:15.41 | mr_zack | it might be the random java codec i downloaded off the web that's broken, but i'm leaning on something ithat i'm doing |
| 01:16.05 | mr_zack | linear vs. logarathmic |
| 01:16.12 | tzanger | IIRC ulaw/alaw and slinear are all more or less the same, except for the handling of sign |
| 01:16.23 | Hmmhesays | what's the syntax for setcallerid() ? setcallerid("name" <number>) ? |
| 01:17.38 | xkev | you can use setcidnum and setcidname, they are the "new way (tm)" |
| 01:17.55 | Hmmhesays | k |
| 01:17.57 | xkev | otherwise no quote needed SetCallerID(Foo <1345>) is fine |
| 01:18.24 | xkev | in cvs, num and name are separate members of a struct, instead of having to parse them out every time it needs to be split into its parts |
| 01:18.29 | Hmmhesays | I was just about to look it up.... you saved from like 3 words and a couple mouse clicks |
| 01:18.35 | kpfleming | tzanger: that's not quite true... slinear is 16-bit uncompressed audio... ulaw and alaw are both compressed to 8 bits |
| 01:18.44 | hardwire | hmmphm |
| 01:18.47 | tzanger | kpfleming: ahh |
| 01:19.05 | ManxPower | Quotes in Caller*ID Name can cause some versions of Cisco SIP firmware to reject the call. |
| 01:19.06 | Hmmhesays | thanks xkev |
| 01:19.16 | ManxPower | Since quotes are obvously not valid callerid |
| 01:19.16 | tzanger | kpfleming: so ulaw/alaw are 8bit 8kHz and slinear is 16bit 8kHz (so double the bandwidth of ulaw/alaw?) |
| 01:19.35 | mr_zack | actually |
| 01:19.40 | mr_zack | ulaw has an effective 12-bit bandwidth |
| 01:19.44 | mr_zack | and alaw 14-bit if iirc |
| 01:19.46 | kpfleming | double the bandwidth consumption, yes... not double the audio bandwidth though, ulaw is approx equiv to 13-bit audio, and alaw is 14-bit |
| 01:20.02 | mr_zack | er, well, kpfleming is probably right :) |
| 01:20.09 | tzanger | I'm not up on my codecs obviously. :-S |
| 01:20.22 | mr_zack | kpfleming: any ideas as to why my pitch would be off? |
| 01:20.45 | kpfleming | nope, sorry, i'm wondering how to convince chatzilla to let me connect to two IRC nets athe same time :-) |
| 01:20.45 | *** part/#asterisk qwerp (~abc@219.93.57.58) |
| 01:21.22 | `Sauron | kpfleming: /server <new server> |
| 01:21.27 | `Sauron | IIRC |
| 01:21.36 | kpfleming | i'll try it... |
| 01:21.49 | tzanger | `Sauron: typically that disconnects you form the current server |
| 01:21.50 | kpfleming | yep, thanks, that worked |
| 01:22.15 | `Sauron | tzanger: Only when NOVICE is set true |
| 01:24.09 | Hmmhesays | hmm if i setcidname() and setcidnum() Asterisk sends From: "asterisk" <sip:asterisk@x.x.x.x> |
| 01:24.57 | Hmmhesays | shouldn't the from field be From: <sip:x.x.x.x> ? |
| 01:25.03 | tzanger | `Sauron: ah |
| 01:26.05 | kpfleming | Hmmhesays: it's hard to get asterisk to send NULL CLID and CNAM... it has hardcoded defaults in chan_sip |
| 01:26.14 | Hmmhesays | or am i missing something completely obvious |
| 01:26.26 | Hmmhesays | gotcha |
| 01:27.30 | ManxPower | Hmmhesays, I don't know about the SIP headers, but setting the info using the SetCID* functions always work if I use them correctly. |
| 01:27.52 | ManxPower | Unless, of course, I'm trying to send the call to the PSTN, then CID Name does not work, and it's not expected to. |
| 01:28.05 | *** join/#asterisk Godsey (lanny@207-229-102-125.cortland.com) |
| 01:28.23 | Hmmhesays | yeah it shows up unavailable if I use those.... I'm trying to test out the "block anonymous caller" feature on this POS fxs box |
| 01:28.32 | Hmmhesays | so far it's useless |
| 01:28.51 | harryvv | http://pastebin.ca/5255 manx seen a case where out of the blue another device grabs the zaptel irq? |
| 01:29.03 | ManxPower | Hmmhesays, PASTE the ACTUAL lines from extensions.conf that you using to set the CLID |
| 01:29.43 | Hmmhesays | I'm good.... I just got it to clear out any callerid information |
| 01:29.46 | Hmmhesays | in the sip header |
| 01:30.27 | Hmmhesays | thanks tho ManxPower |
| 01:30.34 | ManxPower | Hmmhesays, Every person that I've seen that tries to fuck with the headers for something as simple and trivial as callerid has falied. that doesn't mean that some have not worked, I just don't know about htem |
| 01:31.03 | Hmmhesays | yeah I just have to make sure i've done everything before I declare this a useless function to my boss |
| 01:31.17 | ManxPower | Hmmhesays, PASTE the ACTUAL lines from extensions.conf that you using to set the CLID |
| 01:31.33 | ManxPower | harryvv, no. |
| 01:31.38 | Hmmhesays | heh, well if you really want to see them |
| 01:31.43 | Hmmhesays | hold on |
| 01:31.43 | *** join/#asterisk Guest^DJ (some@211.24.146.10) |
| 01:31.50 | ManxPower | there should only be 2 lines, of course. |
| 01:32.02 | Hmmhesays | yeah hold on i gotta sign into xchat |
| 01:32.24 | *** join/#asterisk nullogic (~nullogic@216.24.172.242) |
| 01:32.51 | *** join/#asterisk hmmhesays (~root@66.173.103.106) |
| 01:33.26 | hmmhesays | exten => 5001,1,setcidname() |
| 01:33.26 | hmmhesays | exten => 5001,2,setcidnum() |
| 01:33.32 | znoG | is NuFone down?? |
| 01:33.49 | ManxPower | hmmhesays, So you are trying to set them to empty? |
| 01:33.54 | `Sauron | setcidname("Name") |
| 01:33.59 | hmmhesays | yes |
| 01:34.02 | `Sauron | setcidnum("12345") |
| 01:34.08 | ManxPower | Why do you want them empty? |
| 01:34.25 | hmmhesays | just testing out features on an fxs unit |
| 01:34.36 | ManxPower | `Sauron, don't use quotes |
| 01:34.42 | `Sauron | hum, right |
| 01:34.43 | bjohnson | can I kill a sip call from the cli? |
| 01:34.44 | hmmhesays | I got it empty though, like i said before |
| 01:34.47 | `Sauron | that was always weird |
| 01:34.54 | `Sauron | bjohnson: soft hangup <whatever> |
| 01:35.00 | nullogic | how do I selectively set my ANI for outbound calls but not for internal calls? |
| 01:35.08 | `Sauron | Usage: soft hangup <channel> |
| 01:35.08 | `Sauron | <PROTECTED> |
| 01:35.08 | `Sauron | <PROTECTED> |
| 01:35.12 | ManxPower | bjohnson, no, but you can hangup a channel, which will do the same thing. |
| 01:35.12 | bjohnson | I started an outgoing call to a cell phone and hung up .. I don't think * hung up |
| 01:35.19 | `Sauron | show channels |
| 01:35.30 | `Sauron | if it's there, you can do soft hangup on it |
| 01:35.35 | `Sauron | if it's not there, it already hung up |
| 01:35.50 | bjohnson | ip, exten, or callid? |
| 01:35.54 | nullogic | anyone? |
| 01:35.54 | hmmhesays | exten => 5001,1,setcidname() |
| 01:35.54 | hmmhesays | exten => 5001,2,setcidnum() that doesn't set it empty in the debugs though |
| 01:36.01 | `Sauron | channel number/thingy |
| 01:36.04 | `Sauron | I dunno :p |
| 01:36.07 | hmmhesays | i just erased all identifying information out of the orginating unit |
| 01:36.08 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca) |
| 01:36.33 | `Sauron | bjohnson: do a "show channels" first |
| 01:37.24 | ManxPower | nullogic, set it before the exten lines that dial outside. |
| 01:37.39 | bjohnson | ahh I was doing sip show channels |
| 01:37.44 | bjohnson | thnx |
| 01:38.52 | *** join/#asterisk Frantic (~ab@68-175-94-179.nyc.rr.com) |
| 01:40.31 | harryvv | im at a loss on this problem. I removed the sound card which was using irq 5 and then assigned that to the zapcard in bios. Well getting lots charicters when doing asterisk -c Getting rrros of unable to handle kernel paging request at virtual address ...then alot of hex codes scroll by half a page long. |
| 01:42.26 | nullogic | ManxPower: I tried that. sip.conf's context it outgoing. outgoing includes extentions and then setcallerid. When I place calls between sip phones, I still see Setcallerid called... |
| 01:43.21 | ManxPower | nullogic, EXTENSIONS.CONF |
| 01:43.54 | ManxPower | Unless you are using the same pattern for dialing phone to phone as you use to dial outside then you have a simple task. |
| 01:44.45 | *** join/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
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| 01:48.17 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) [NETSPLIT VICTIM] |
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| 01:49.50 | *** join/#asterisk gopinsurg (cashmoney@dialup-4.225.5.3.Dial1.Cincinnati1.Level3.net) |
| 01:50.22 | `Sauron | Anyone tried to use spandsp with your own application? |
| 01:50.37 | gopinsurg | Any get a Intel 536EP working as an FXO ? |
| 01:51.04 | ManxPower | gopinsurg, nobody will until someone writes a driver or updates wcfxo to support it |
| 01:51.17 | gopinsurg | thx |
| 01:52.08 | *** part/#asterisk chetan (freetibet@24-193-188-21.nyc.rr.com) |
| 01:53.19 | ROM_Man | Just updated my system from using x100p's to the tdm 4 port card. Now distinctive ring detection is broken. Any ideas? Google searches are not getting me anywhere. |
| 01:54.12 | nullogic | ManxPower: Here is a snip of my setup, can you take a look? http://www.pastebin.com/237404 |
| 01:54.15 | *** join/#asterisk Godsey (lanny@207-229-102-125.cortland.com) [NETSPLIT VICTIM] |
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| 01:58.00 | ManxPower | nullogic, http://www.pastebin.com/237406 |
| 02:01.55 | *** join/#asterisk crash3m (crash3m@crash3m.user) [NETSPLIT VICTIM] |
| 02:03.15 | ariel_ | argh network split. |
| 02:04.31 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:07.14 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 02:08.39 | znoG | how can all my VoIP providers that I'm with be down |
| 02:09.24 | znoG | NuFone says bad context or extension (bullshit, it used to work), VoipJet says its congested/busy, Freshtel says it can't bridge channels.. ok so they're not down, just broken |
| 02:09.26 | WilliamK | are they outsourcing from each other? =) |
| 02:09.31 | nullogic | ManxPower: Thanks much.. the simple answer is almost the hardest to find |
| 02:09.52 | Mike | hey guys i have my voicemail setup but when i get incoming calls from the x101p they leave the message and they hangup and asterisk records like 1 minute of fastbusy |
| 02:10.02 | Mike | anyone knows what can i do |
| 02:10.23 | ManxPower | nullogic, NEVER EVER use a different pattern in an extenson on different priorities |
| 02:12.28 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 02:13.39 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 02:13.39 | *** mode/#asterisk [+o twisted] by ChanServ |
| 02:15.05 | harryvv | what irq is recomended for the x100p/ |
| 02:15.05 | harryvv | ? |
| 02:17.22 | *** part/#asterisk yogurt2ungue (~yogurt2un@host15.201-252-158.telecom.net.ar) |
| 02:17.35 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 02:19.44 | *** join/#asterisk lalalala (~lalalala@netblock-66-245-217-43.dslextreme.com) |
| 02:19.52 | lalalala | hello everyone |
| 02:19.53 | *** join/#asterisk jterrero (~jterrero@ool-43576e0d.dyn.optonline.net) |
| 02:20.23 | jterrero | someone wanna help me out? when i load0 |
| 02:20.23 | jterrero | . |
| 02:20.40 | jterrero | oops. sorry, |
| 02:21.02 | jterrero | when i load asterisk i get a error saying "Unable to get IP address for asterisk, SIP disabled |
| 02:21.22 | jterrero | i do have an IP address, anyone know whats the problem ? |
| 02:21.28 | jterrero | iax sucesfully registers with a remote server |
| 02:22.54 | *** join/#asterisk jterrero (~jterrero@ool-43576e0d.dyn.optonline.net) |
| 02:23.19 | *** join/#asterisk jesse_132 (~chatzilla@12-203-179-57.client.insightBB.com) |
| 02:24.00 | jterrero | dont know if anyone got my last message, client crashed.. when i launch asterisk i get an error saying "unable to get an IP address, SIP disabled", my server is getting an IP address and IAX sucessfully registers with a remote server |
| 02:24.01 | lalalala | anyone here? |
| 02:24.04 | jterrero | anyone know whats up ? |
| 02:25.14 | cypromis | probably things that are not down are up |
| 02:25.43 | cypromis | your reverse dns is screwed |
| 02:25.55 | cypromis | so best put the ip address with the proper hostname into /etc/hosts |
| 02:25.58 | cypromis | and sip will woek |
| 02:26.05 | lalalala | how do i get a real world phone number that i can connect/associate to/with a sip/iax client |
| 02:26.46 | jterrero | www.vonage.com, contact them and they will give you infos on IAX providors |
| 02:27.02 | jterrero | www.ziacom.us is where i get my DIDs from |
| 02:27.51 | lalalala | jterrero : i want phone numbers for my sip or IAX client ... not an IAX provider |
| 02:28.05 | harryvv | vonage is really pushing there marketing hard. seem them advertised almost everywhere |
| 02:28.53 | jterrero | they are getting real big in the nyc area |
| 02:29.06 | jterrero | lalalala: not following you |
| 02:29.08 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 02:29.30 | toddf | heh, just as they're trying to get big, I'm trying to get away from them |
| 02:29.42 | *** join/#asterisk skeeziks (~skeeziks@66-23-208-2.clients.speedfactory.net) |
| 02:30.05 | lalalala | jterrero : i want to get a new phone number that i can recieve calls on for my contact center voip network |
| 02:30.29 | lalalala | how do i get this phone number and connection that'll work with my contact center voip network |
| 02:30.32 | skeeziks | Is the Broadvoice patch in Asterisk 1.0.5? |
| 02:34.17 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp201195.qc.sympatico.ca) |
| 02:34.50 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 02:36.07 | lalalala | ? |
| 02:37.38 | hacim | skeeziks: broadvoice patch? |
| 02:40.23 | jterrero | anyone know what "asked to transmit frame type 4, while native format is 1 |
| 02:40.26 | jterrero | " |
| 02:42.04 | lalalala | where do these telecom service providers get their number blocks from |
| 02:43.01 | skeeziks | hacim: Broadvoice sponsored a patch in November sometime to get Asterisk to work better with their service - I see reports that it's in CVS HEAD, but I'm trying to find out if it's also in release 1.0.5. |
| 02:43.05 | *** join/#asterisk wm (~wm501@ip70-177-80-170.ok.ok.cox.net) |
| 02:43.23 | wm | I can't find any good examples on # transfering from a queue. Anyone? |
| 02:43.38 | hacim | skeeziks: what kind of interaction with broadvoice is possible with this patch? |
| 02:43.54 | skeeziks | hacim: All the normal stuff, really |
| 02:44.06 | skeeziks | hacim: I've been using BV in production since about November, with minimal trouble |
| 02:44.21 | lalalala | hello |
| 02:44.24 | hacim | skeeziks: in what way? I've got a BV ATA-100, but how do you use it with asterisk? |
| 02:44.27 | skeeziks | hacim: It's my main line. |
| 02:44.28 | lalalala | anyone? |
| 02:44.32 | lalalala | where do these telecom service providers get their number blocks from |
| 02:44.55 | skeeziks | hacim: http://www.broadvoice.com/support_install_asterisk.html |
| 02:45.24 | lalalala | also, how do i get this phone number and connection that'll work with my contact center voip network (do i need a special hardware, or can i just plug my phone cord in my modem slot and it'll start working? |
| 02:45.49 | hacim | skeeziks: so you save the money on the ATA by using asterisk, basically? |
| 02:46.10 | *** join/#asterisk jumpingin (~asd@10.knoxville-04-05rs.tn.dial-access.att.net) |
| 02:46.24 | lalalala | come on, dont tell me you dont know the answer to it |
| 02:46.35 | skeeziks | hacim: Yeah, and I still have internal extensions |
| 02:46.56 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 02:47.29 | hacim | interesting |
| 02:48.05 | *** join/#asterisk wm501 (~wm501@ip70-177-80-170.ok.ok.cox.net) |
| 02:48.13 | wm501 | grr, stupid weak cable modem signal |
| 02:48.15 | wm501 | I can't find any good examples on # transfering from a queue. Anyone? |
| 02:48.26 | wm501 | my ghost is still hanging around :? |
| 02:48.26 | lalalala | ? |
| 02:48.45 | lalalala | where do these telecom service providers get their number blocks from |
| 02:51.04 | jumpingin | Has anyone converted a 7960 or 7940 to SIP? |
| 02:52.01 | wolfson | jumpingin: ensure you reset to system defaults at each step |
| 02:52.07 | wolfson | other than that just follow the cisco docs |
| 02:52.33 | jumpingin | I can't get cisco to take my money... |
| 02:52.34 | jterrero | yeah i did recently |
| 02:53.08 | jterrero | i used www.solarwinds.net tftp |
| 02:53.22 | jterrero | and i started with POS300600 |
| 02:53.26 | jterrero | or smoething like that |
| 02:54.38 | jumpingin | i'll try that, thanks |
| 02:56.01 | wm501 | does anyone here using pound transfering? |
| 03:05.12 | lalalala | bastards! |
| 03:06.13 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 03:06.27 | Gronker | test |
| 03:07.07 | Legend | failed |
| 03:08.24 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 03:08.27 | MrEntropy | yo |
| 03:08.37 | Gronker | heh...Trying to get trillian to talk IRC...easier to build a free BSD box and run IRC me thinks |
| 03:08.40 | MrEntropy | does anyone know a tool i can use to convert G723.1 IVR's to WAV? |
| 03:08.43 | wm501 | oh, |
| 03:08.46 | wm501 | it worked by default |
| 03:08.47 | wm501 | cool |
| 03:09.35 | gambolputty | Can you have something like _* in a dialplan to match the asterisk character? |
| 03:11.20 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 03:11.44 | *** join/#asterisk WiFiGuy (~bob@d079.max6-2.mke2.ticon.net) |
| 03:12.06 | *** part/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 03:12.19 | *** join/#asterisk three55ml (~who@cs666898-229.austin.rr.com) |
| 03:14.29 | WiFiGuy | will I need any special hardware to have a VoIP connection over the internet or a lan using asterisk? |
| 03:14.47 | bjohnson | WifiFred: no |
| 03:14.57 | wm501 | okay, in call queuing, if my queue is full, how do i give a busy signal? |
| 03:15.13 | bjohnson | WifiFred: but most people WANT special hardware |
| 03:15.20 | three55ml | http://pastebin.ca/5258 - This IAX registration is driving me crazy. Exact same config registers fine on another server. |
| 03:15.37 | WiFiGuy | where did fred come in? |
| 03:16.20 | three55ml | There's a guy named Wififred in here too, probably auto-completion |
| 03:16.23 | bjohnson | WiFiGuy: tab completion shortcut in my irc client .. someone else named that |
| 03:16.33 | WiFiGuy | ahh |
| 03:16.59 | bjohnson | three55ml: ug . .I can never follow that stuff |
| 03:17.24 | three55ml | bjohnson: If I change the username to an invalid one, I get the registration failed message. Switch back to the valid one - I get no message at all. |
| 03:17.32 | WiFiGuy | will it make a difference in voice quality over a lan, whether I use special hardware? |
| 03:17.42 | bjohnson | WiFiGuy: eg. you could use softphones, with * on a machine you already have, and use a voip provider for incoming and/or outgoing calls |
| 03:17.53 | three55ml | WiFiGuy: Not really. Softphone and hardware phones used the same codecs. |
| 03:18.22 | bjohnson | WiFiGuy: most people prefer hardware phones .. eg analog phones in fxs ports (hardware) or voip phones |
| 03:19.10 | bjohnson | WiFiGuy: also, if you want local phone lines from the telcoo tied in, you will need hardware for that (but it IS possible just to use a voip provider if one provides DIDs in suitable areas for you) |
| 03:19.50 | bjohnson | WiFiGuy: it is pretty much universal here that hardware phones have better sound quality than softphones |
| 03:19.59 | WiFiGuy | I'm probably only going to by using it in my lan |
| 03:20.31 | bjohnson | WiFiGuy: I have no idea what that means .. but you should start reading these docs |
| 03:20.33 | bjohnson | ~docs |
| 03:20.34 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 03:20.55 | *** join/#asterisk Q-At-Home (~Queue@S0106000c41bb87af.ed.shawcable.net) |
| 03:21.03 | WiFiGuy | lan = local area network |
| 03:21.12 | WiFiGuy | basicly a small and fast network |
| 03:21.21 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 03:21.29 | Q-At-Home | whee-- long time now see all |
| 03:21.45 | WiFiGuy | thanks for the links |
| 03:21.48 | *** part/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 03:22.23 | ROM_Man | Just updated my system from using x100p's to the tdm 4 port card. Now distinctive ring detection is broken. Any ideas? Google searches are not getting me anywhere. |
| 03:22.49 | *** join/#asterisk simon_ca (sedgett@S01060004e23873e1.vc.shawcable.net) |
| 03:25.50 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 03:27.00 | WiFiGuy | the digium.com link is down, dns error |
| 03:27.09 | *** join/#asterisk santiago (~santiago@63.245.86.104) |
| 03:32.32 | bjohnson | WiFiGuy: you're slaying me |
| 03:32.54 | WiFiGuy | lol |
| 03:33.00 | bjohnson | ROM_Man: sorry, I don't use that hardware .. sounds like a specific problem |
| 03:33.33 | bjohnson | ROM_Man: since noone answering .. search wiki and mailing list for info .. or try back later |
| 03:34.40 | bjohnson | WiFiGuy: what does "I'm probably only going to by using it in my lan" mean? You're not going to have any connection to the outside world? You don't want hardware phones? I have no idea what that statement means to you. |
| 03:35.43 | WiFiGuy | by = be |
| 03:36.01 | WiFiGuy | it just means I won't be going through the internet |
| 03:36.04 | *** join/#asterisk juice (~juice@mo-65-40-248-28.dyn.sprint-hsd.net) |
| 03:36.14 | WiFiGuy | I just want to see what I can do for free |
| 03:36.28 | WiFiGuy | thats why I'm interested in only using software |
| 03:36.55 | hacim | i just got a x100p installed, got zaptel compiled and loading the zaptel and wcfxo modules. I've configured my zaptel.conf extentions.conf and my iax.conf |
| 03:37.34 | hacim | I thought i'd give bellster a shot... i plugged a phone into the x100p and a phone line in... but I get no dialtone on the phone and am not sure what I am missing |
| 03:38.08 | wolfson | -you can't plug a phone into the x100p |
| 03:38.13 | three55ml | X100P is an FXO device, you can't plug a phone into i |
| 03:38.13 | three55ml | it |
| 03:38.33 | three55ml | You need an FXS like a Sipura, Grandstream, or one of many others |
| 03:38.40 | hacim | really? there is a jack on it that has a picture of a phone and says "phone" and a jack that has a picture of a rj-11 and says "line" |
| 03:38.51 | *** join/#asterisk PyroSteve (~steve@ip68-227-149-247.no.no.cox.net) |
| 03:38.59 | PyroSteve | hey guys |
| 03:39.08 | wolfson | hacim: when acting as a modem thats a pass through port |
| 03:39.09 | three55ml | It might work as a pass-through, I'm not sure...but you won't be able to ring it to Asterisk or anything. |
| 03:39.16 | three55ml | from Asterisk |
| 03:39.20 | wolfson | just bridges them, but its not an actuall interface |
| 03:39.21 | PyroSteve | somoene helped me out the other day by pointing out the random command |
| 03:39.22 | Q-At-Home | hey guys, I'm banging my head against a callerid problem... I have a vonage line coming into a TDM fxo 4 port card, and no matter what I do, I cant get the caller ID to pass to * on port 1 of the vonage ata... port 2 works fine |
| 03:40.01 | three55ml | Q-At-Home: I've actually heard of several people having misc problems with port 1 |
| 03:40.35 | hacim | wolfson: why does the description of the x100p say "the device is fully supported by asterisk for both incoming and outgoing calls" if you can't use your phone on it? |
| 03:40.51 | Q-At-Home | three55ml: DOH |
| 03:41.13 | Q-At-Home | I switched the ports around, and it follows port 1 on the ATA |
| 03:41.17 | three55ml | hacim: Not sure, where does it say that? |
| 03:41.35 | PyroSteve | how does random work ? |
| 03:41.39 | three55ml | Q-At-Home: I think I've read on the DSL Reports forums about Vonage actually having to move peoples main line to port 2 |
| 03:41.51 | wolfson | hacim: it does clearly support a call in and out |
| 03:41.53 | PyroSteve | not the syntax |
| 03:41.59 | PyroSteve | but the inner workings |
| 03:42.11 | hacim | three55ml: in what way? |
| 03:42.14 | wolfson | hacim: i see no where, where it says its an FXS interface, clearly labeled FXO |
| 03:42.15 | Q-At-Home | then my fax will break :) |
| 03:42.27 | PyroSteve | Does asterisk keep track of the last result of random |
| 03:42.29 | Q-At-Home | and believe it or not, I have faxing working :) |
| 03:42.33 | three55ml | Q-At-Home: :) |
| 03:42.38 | hacim | wolfson: how do you do call out with it then? |
| 03:42.41 | Q-At-Home | that took a bunch of time to fix |
| 03:42.44 | three55ml | Q-At-Home: How reliably? I have problems with it all the time |
| 03:42.52 | Q-At-Home | heres how I get it working 100% |
| 03:43.01 | Q-At-Home | vonage ata to tdm fxo module |
| 03:43.11 | wolfson | hacim: however you want, soft phone, fxs card, etc... |
| 03:43.14 | Q-At-Home | then tdm fxs module to external modem |
| 03:43.22 | Q-At-Home | to hylafax LOCKED at 2400 baud |
| 03:43.28 | wolfson | hacim: same way an incomming call works |
| 03:43.32 | three55ml | Q-At-Home: Ah |
| 03:43.33 | Q-At-Home | ran thru 5 different modems, |
| 03:43.38 | PyroSteve | does random have to keep track of its last decision in order to make the next descision ? |
| 03:43.43 | Q-At-Home | USR 56k serial works the best |
| 03:43.48 | Q-At-Home | external |
| 03:43.49 | hacim | wolfson: ahhh, so if I plugged in a PTSN line in the line port, then used a softphone to call my asterisk instance, I could call out on the phone line |
| 03:43.58 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 03:44.12 | wolfson | hacim: if you set up your dialplan correctly, yes |
| 03:44.15 | Q-At-Home | I harped on vonage for hours trying to get them to give the info on the ata, so I can use asterisk as the ATA |
| 03:44.18 | Q-At-Home | no joy :( |
| 03:44.23 | Q-At-Home | the soft phone works fine |
| 03:44.31 | hacim | wolfson: hmm ok |
| 03:44.39 | three55ml | Q-At-Home: Yeah, I'm ditching them this bill cycle |
| 03:44.57 | PyroSteve | Q-At-Home: couldn't you sniff your sip traffic to Vonage ? |
| 03:45.21 | three55ml | PyroSteve: I think it's more an an issue with the ATA being locked |
| 03:45.38 | three55ml | I'm sure someone will eventually figure out a firmware hack, but in the end you end up having to send it back to Vonage anyways. |
| 03:45.57 | PyroSteve | yeah, but the goal is too get the login credentials, correct ? |
| 03:46.17 | Gronker | FXS ports are what would be used to drive normal POTS-like handsets...correct? |
| 03:46.27 | wolfson | yes |
| 03:46.30 | PyroSteve | Gronker: yes |
| 03:46.44 | Gronker | whats a good 2 and 4 port FXS card...cheap? |
| 03:46.51 | PyroSteve | FXS ports provide dialtone and listen for DTMF tones |
| 03:46.52 | Q-At-Home | I have no intention of defrauding vonage |
| 03:46.59 | Q-At-Home | I just hate using up 2 FXO ports |
| 03:47.04 | Q-At-Home | when direct SIP is reliable |
| 03:47.11 | Q-At-Home | just.. much more expensive to use |
| 03:47.30 | PyroSteve | Gronker: FXO port recv. dialtone and provide dtmf tones |
| 03:47.58 | MrEntropy | anyone know a tool to decode g723.1 to wav? |
| 03:48.08 | Gronker | doh...typo...need 2 and 4 port FXS port boards...hehe...I have a long day ahead of me |
| 03:48.57 | three55ml | Gronker: I don't think the Digiums are all too expensive |
| 03:49.35 | Gronker | didnt say they were...havent done much of any research yet...just decided to start here...so I know what to compare |
| 03:50.07 | Gronker | (too be honest, Id like to buy some of my stuff for this project from Digium to support them for the asterisk dev, but not my call) |
| 03:50.12 | three55ml | Gronker: The Digiums are the only ones I know of. Plus they have the upside of supporting the project. |
| 03:50.14 | three55ml | yeah |
| 03:50.51 | Gronker | thats cool...just need to be able to tell the client that I got best price per port...would be best if Icould find 2-3 vendors |
| 03:51.04 | three55ml | Yeah, I know what you mean |
| 03:51.55 | Q-At-Home | hrmm |
| 03:52.05 | Q-At-Home | "Freeworld tel" has dids in edmonton now |
| 03:52.07 | Gronker | what am I looking at per extension? (err...FXS port) |
| 03:53.10 | Gronker | ahhh...nite...daughter boards for digium cards...cool...reading more |
| 03:53.10 | bjohnson | Q-At-Home: the vonage ATA's are fxs ports .. not fxo |
| 03:53.40 | Gronker | ahhh...crap...75$ per FXS module? |
| 03:53.45 | three55ml | Gronker: Yeah, 4 total. Any combo of FXS and FXO |
| 03:53.51 | Q-At-Home | yes... thats what I said, I plug my ATA into the FXO port of my * |
| 03:53.59 | Q-At-Home | and then the modem into the FXS |
| 03:54.02 | bjohnson | Gronker: best price per port for fxs .. channel bank |
| 03:54.16 | bjohnson | Gronker: (but provides 23 ports) |
| 03:54.18 | Q-At-Home | I have 2 fully populated tdm cards, and love em |
| 03:54.27 | Q-At-Home | one full of fxo, one full of fxs |
| 03:54.28 | bjohnson | Gronker: second best Sipura SPA 2000 |
| 03:54.39 | bjohnson | Gronker: third best Digium |
| 03:54.59 | bjohnson | Gronker: there are pros and cons to each that have nothing to do with price |
| 03:55.44 | bjohnson | Q-At-Home: ewww |
| 03:55.56 | Gronker | great info here guys...thanks... |
| 03:55.59 | bjohnson | ATA<->fxo |
| 03:56.02 | bjohnson | ewww |
| 03:56.08 | Q-At-Home | heh |
| 03:56.15 | three55ml | Haha, I did that for about a week |
| 03:56.15 | Q-At-Home | its only 2 lines :) |
| 03:56.25 | Q-At-Home | I have 3 other IP based providers |
| 03:56.30 | three55ml | Then I saw the light :) |
| 03:56.36 | Gronker | Im prolly gonna need about 7 stations (extensions) and interface for 2 POTS lines...and VoIP long distance and 800 services...in the end |
| 03:56.37 | Q-At-Home | trying like hell to ditch the ATA |
| 03:56.57 | Q-At-Home | Caller*ID failed check is killing me |
| 03:57.05 | Q-At-Home | (dont bother googling it) |
| 03:57.16 | bjohnson | Gronker: none of the voip stuff needs any hardware .. it's connecting to the pstn and the people that could need hardware |
| 03:57.17 | Q-At-Home | I think its the ATA... yet a phone plugged into it works |
| 03:57.36 | Q-At-Home | wonder if its not rx/tx gains |
| 03:57.42 | Q-At-Home | hrm.. all set to 0 |
| 03:58.12 | Gronker | aye...I get the VoIP stuff (have it working in other solutions) but this is first day with asterisk...so trying to price out the hardware needed to support the extensions and POTS |
| 03:58.16 | bjohnson | Gronker: you could also look at voip phones for the extensions instead of fxs+analog phone |
| 03:58.26 | Q-At-Home | voip phones are great |
| 03:59.23 | Gronker | Price is the key here for this proj...will be dropped into (potentially) dozens of remote offices...so have to break this down using existing POTS service and phones...if possible |
| 03:59.36 | bjohnson | Gronker: 2 examples to check .. 4 port digium cards with mixed fxo and fxs AND Sipura SPA 2000s and 3000s |
| 03:59.38 | Gronker | Building the "great Merlin killer" ...snicker |
| 04:00.57 | bjohnson | SPA 3000 are $100USD for 1 fxo and 1 fxs and have 2 pros .. plugs into lan, not pci slot (no drivers needed) AND do auto-failover connect between fxs and fxo in power outage |
| 04:01.03 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 04:01.17 | Q-At-Home | do the spa3000's have fxo addressable from * ? |
| 04:01.30 | Q-At-Home | i.e inbound calls to the spa3000 could terminate to the * box? |
| 04:01.57 | bjohnson | SPA 2000 are 2 port fxs for $80 USD .. same lan connection .. can be used anywhere (including a connection in from a remote office) |
| 04:02.02 | bjohnson | Q-At-Home: definitely |
| 04:02.05 | Q-At-Home | last time I was here, there was some confusion |
| 04:02.07 | Q-At-Home | REALY>?@?#@ |
| 04:02.10 | Q-At-Home | GAH |
| 04:02.10 | bjohnson | I have 3 |
| 04:02.30 | bjohnson | it would be crap otherwise |
| 04:02.34 | Q-At-Home | and you can plug PSTN lines into the SPA, and plop it onto your * box like a FXO port |
| 04:02.43 | bjohnson | exactly |
| 04:02.45 | Q-At-Home | well woo... I can order some up |
| 04:02.47 | bjohnson | it IS a fxo port |
| 04:02.51 | Q-At-Home | enough of these PCI cards |
| 04:02.56 | Q-At-Home | I can go small form factor! |
| 04:03.05 | bjohnson | well .. the digim cards have a couple of pros too |
| 04:03.08 | Q-At-Home | wife will love me |
| 04:03.15 | Gronker | looking at the sipura stuff...rather have PCI cards and have everything plug into the back of the system...these converters tend to get screwed with in the field |
| 04:03.19 | Chuji | Dammit I can't get cdr_odbc to work for shit |
| 04:03.23 | Q-At-Home | oh don't get me wrong, I love the digium cards |
| 04:03.26 | bjohnson | 1. they are THE way to get proper timing for meetme conferencing |
| 04:03.31 | Chuji | When is bkw coming back? |
| 04:03.37 | Q-At-Home | I'll leave one in |
| 04:03.40 | Chuji | ~seen bkw_ |
| 04:03.41 | jbot | bkw_ <~brian@65.38.28.146> was last seen on IRC in channel #asterisk, 10h 20m 6s ago, saying: 'bbl'. |
| 04:03.42 | bjohnson | 2. they support the guys who worte the software |
| 04:03.46 | Q-At-Home | my box is a small form factor with one PCI |
| 04:03.55 | Q-At-Home | I own 4x digium cards |
| 04:04.01 | bjohnson | 3. price per port hey are pretty good is you do 4 ports at a time |
| 04:04.02 | Q-At-Home | for well over $2000 :) |
| 04:04.15 | Q-At-Home | they have a great warranty too |
| 04:04.42 | Gronker | guess I was just hoping to get extensions down to the $25-33 per port...looks like that is a pipe dream |
| 04:04.45 | Q-At-Home | does the wikki have a sample config for using the FXO's as inbound? |
| 04:04.46 | bjohnson | Gronker: you could always screw them to the side of the cpu box |
| 04:04.52 | Chuji | ~seen anthm |
| 04:04.52 | jbot | anthm <~anthmct@CPE-69-76-83-52.wi.rr.com> was last seen on IRC in channel #asterisk, 13d 13h 3m 53s ago, saying: 'if i get into conflicting situations i usualy pick one and try and get it committed so i can eliminate the conflict'. |
| 04:05.02 | bjohnson | Gronker: basically, there are all sorts of hardware configs you can do |
| 04:05.32 | bjohnson | Gronker: you could do it with a channel bank .. but then you're buying in bulk |
| 04:05.39 | Gronker | This bundle includes: |
| 04:05.40 | Gronker | One (1) TDM400P |
| 04:05.40 | Gronker | Four (4) FXS Modules (green) |
| 04:05.40 | Gronker | = $300+ |
| 04:06.09 | bjohnson | Gronker: you could shop ebay .. think I saw a 1 port thing that did iax go for $25 and shipping .. try getting a bunch from that supplier? |
| 04:06.24 | Q-At-Home | atacomm still a regular here? |
| 04:06.32 | bjohnson | yes |
| 04:06.42 | Gronker | Channel bank for the extension (FXS) side? I can see T1 for the FXO side...but what is the deal for bulk on the FXS side? |
| 04:06.46 | Chuji | ~seen atacomm |
| 04:06.47 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 1d 2h 9m 54s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
| 04:07.00 | Q-At-Home | unless someone in Canada shipping from Canada has the spa3000 |
| 04:07.14 | bjohnson | Gronker: t1 to channel bank .. channel banks can be mixed fxo or fxs |
| 04:07.16 | Q-At-Home | woo pstn fail over! |
| 04:07.20 | Q-At-Home | wife will be happy |
| 04:07.26 | bjohnson | Gronker: search ebay for adit600 for examples |
| 04:07.38 | bjohnson | Q-At-Home: ? |
| 04:07.47 | Gronker | I will get my develpoment stuff from eBay prolly...but I have to price this for something like 50 systems, over 12 months... |
| 04:07.58 | Gronker | can rely on eBay stuff to supply it |
| 04:08.05 | bjohnson | Gronker: what country |
| 04:08.14 | Gronker | US |
| 04:08.25 | Q-At-Home | my asterisk box is huge |
| 04:08.25 | bjohnson | Q-At-Home: "unless someone in Canada shipping from Canada has the spa3000"? |
| 04:08.31 | Q-At-Home | oh, was gunna order it |
| 04:08.42 | Q-At-Home | online from atacomm, but he's US based |
| 04:08.45 | Gronker | err...well...Im sure that 95% of the offices will be |
| 04:08.54 | Q-At-Home | I'm cheap, and would like to save duty/shipping/brokerage/etc |
| 04:09.05 | bjohnson | Q-At-Home: there's a couple of cdn dealers that carry spas |
| 04:09.17 | Q-At-Home | speaking of cheap, anyone seen the vaporware vonage wifi phone? |
| 04:09.18 | bjohnson | one I remember seeing on wiki is syonex |
| 04:09.31 | Q-At-Home | or any wifi sip phone for under $200 :) |
| 04:09.43 | Gronker | I do love this TDM400P module system though...anyone have multiple of these in a system working? |
| 04:10.05 | bjohnson | Gronker: get digium hardware direct from digium in US I think (unless you need a local supplier for speed) |
| 04:10.17 | Q-At-Home | syonex: Our apologies: We are currently unable to supply Sipura or Grandstream products |
| 04:10.26 | Q-At-Home | Gronker: yes |
| 04:10.28 | bjohnson | Gronker: usually by the time they get over 2 pci cards .. they run into other limitations |
| 04:10.42 | Q-At-Home | Gronker: I have 2 PCI cards, one with 4 ports FXO, one with 4 ports FXS |
| 04:10.47 | bjohnson | Gronker: saw something on the wiki about it under asterisk sizing or something |
| 04:11.07 | Q-At-Home | I had a x101 in there a while ago too |
| 04:11.11 | Q-At-Home | all at once |
| 04:11.20 | bjohnson | Q-At-Home: check the wiki for canadian asterisk users for other possibilities |
| 04:11.41 | Q-At-Home | yep |
| 04:11.51 | bjohnson | Q-At-Home: I bought 2 from voxilla and got screwed by UPS for customs broker fees (20% of order value) |
| 04:12.01 | Q-At-Home | yep |
| 04:12.03 | BoRiS | ugh |
| 04:12.07 | *** join/#asterisk chrisf0rd (~chris@cvg-165-100-203.cinci.rr.com) |
| 04:12.11 | chrisf0rd | Hello |
| 04:12.11 | Q-At-Home | ~how can brown screw you today~ |
| 04:12.15 | bjohnson | plus the expected 15% taxes |
| 04:12.17 | chrisf0rd | How do you fix |
| 04:12.24 | Q-At-Home | chrisf0rd: with duct tape |
| 04:12.27 | bjohnson | with a hammer |
| 04:12.31 | *** join/#asterisk thetalon (~Ari@pcp05736786pcs.norstn01.pa.comcast.net) |
| 04:12.31 | Q-At-Home | mmm hammer |
| 04:12.33 | sivana | or bubble gum |
| 04:12.37 | bjohnson | the bigger the better |
| 04:12.48 | chrisf0rd | Everyone is busy/congested at this time what does that mean |
| 04:12.49 | sivana | a mars bar and a rubber band... that's it |
| 04:12.50 | bjohnson | even better if it plugs into the wall |
| 04:13.00 | chrisf0rd | it is just me on the server |
| 04:13.14 | sivana | chrisf0rd: what are you interfacing? |
| 04:13.20 | chrisf0rd | Broadvoice |
| 04:13.23 | bjohnson | chrisf0rd: could be a bunch of things |
| 04:13.37 | bjohnson | I got it when I has a line that was wired in reverse |
| 04:13.56 | chrisf0rd | on my end it rings and rings |
| 04:14.02 | Q-At-Home | bjohnson: your spa3000 does it support incoming caller id? |
| 04:14.06 | Q-At-Home | as an fxo I mean |
| 04:14.08 | chrisf0rd | but on the CLI> it says <sivana> chrisf0rd: what are you interfacing? |
| 04:14.08 | chrisf0rd | <chrisf0rd> Broadvoice |
| 04:14.08 | chrisf0rd | <bjohnson> chrisf0rd: could be a bunch of things |
| 04:14.10 | chrisf0rd | ops |
| 04:14.12 | bjohnson | do you have an 'r' in the dial command? |
| 04:14.24 | bjohnson | Q-At-Home: yes |
| 04:14.31 | Q-At-Home | sweet |
| 04:14.36 | chrisf0rd | on the CLI> it says Everyone is busy congested at this time |
| 04:14.51 | Gronker | hmmm...anyone else have * on a free/BSD machine? |
| 04:15.38 | chrisf0rd | I have been trying to intergrate Broad voice into this machine for 6 hours |
| 04:16.00 | chrisf0rd | it is driving me nuts especially my episode with Customer support earlier today |
| 04:16.04 | thetalon | i had a bad asterisk day. My TE410P was not parsing DNIS property. All calls went to the s|1 extension inside default. Then in 2 hours the problem started the problem went away. This happened to 4 T-1's from 2 different carriers on two servers. This started about 15 minutes after we successfully made calls from 2 new PRI lines. Any ideaWe were adding the new PRI's |
| 04:16.28 | thetalon | that's 2 hours after the problem started it magicaly went away |
| 04:16.40 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 04:16.41 | thetalon | all we did was turn more debuggin on in logger.conf |
| 04:17.01 | thetalon | the Q.931 trace had the DNIS/ANI in there |
| 04:17.12 | Q-At-Home | bed time, l8z all |
| 04:17.13 | chrisf0rd | Am I out in Left field on this one |
| 04:17.16 | Q-At-Home | thanks for the info |
| 04:17.32 | sivana | chrisf0rd: what's your register line look like? |
| 04:17.35 | thetalon | i'm tempted to trash my digium inventment and go cisco at the gateway |
| 04:17.37 | sivana | blank out the pwd |
| 04:17.44 | three55ml | Man, FINALLY solved my IAX issues. Had to do with my VLAN. |
| 04:18.00 | hacim | can I run a softphone and an asterisk server on the same machine? |
| 04:18.06 | hacim | they all seem to want to use port 5060 |
| 04:18.14 | thetalon | hacim sure |
| 04:18.18 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 04:18.19 | bjohnson | chrisf0rd: something on the mailing list about needing a certain version of * |
| 04:18.22 | three55ml | Try an IAX phone |
| 04:18.26 | bjohnson | chrisf0rd: for bv |
| 04:18.35 | hacim | thetalon: so far all the ones I've tried are competing for ports |
| 04:18.41 | thetalon | yep |
| 04:18.42 | chrisf0rd | I have DIAX |
| 04:18.52 | thetalon | it told me that 0/1 was in use already |
| 04:19.15 | thetalon | like * did not tell the CO that the channel was i nuse |
| 04:19.23 | hacim | thetalon: I've tried kphone, sjphone and linphone, all no good so far :p |
| 04:19.29 | bjohnson | chrisf0rd: does diax connect to bv? |
| 04:19.32 | thetalon | iaxcomm |
| 04:19.37 | chrisf0rd | egister => <accountid>@sip.broadvoice.com password> account id>@sip.broadvoice.com |
| 04:19.40 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 04:19.42 | *** join/#asterisk {sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 04:19.50 | sivana | chrisf0rd: that doesn't work for me either |
| 04:19.54 | kFuQ | http://solastyear.com/owned.htm?photo=36 |
| 04:20.05 | bjohnson | hacim: all of those are sip phones .. not sure if that would really be a problem though |
| 04:20.08 | {sean} | does anyone have MOH working with asterisk running as a service? |
| 04:20.11 | sivana | you need to: register => username:password@sip.broadvoice.com/exten |
| 04:20.28 | Gronker | OMG...Im just laughed so hard I spilled my drink |
| 04:20.29 | hacim | thetalon: thanks |
| 04:20.52 | chrisf0rd | I want to be able to dial 9 then get an outside line |
| 04:21.05 | chrisf0rd | not to have it bound to an extension |
| 04:21.15 | thetalon | exten => _9N.,1,Dial( |
| 04:21.17 | sivana | ok |
| 04:21.35 | {Sean} | asterisk running as a service anyone? |
| 04:21.36 | {Sean} | :) |
| 04:21.39 | sivana | having the double username/pass combo that BV says to use never worked for me |
| 04:21.52 | sivana | I mean in the register line |
| 04:22.06 | thetalon | sean see safe_asterisk |
| 04:22.07 | chrisf0rd | example |
| 04:22.22 | chrisf0rd | us:pas#sip.*.* |
| 04:22.24 | sivana | register => username:password@sip.broadvoice.com |
| 04:22.29 | {Sean} | i have it working as a service, just not MOH |
| 04:22.38 | sivana | Sean.. look at mpg123 |
| 04:22.43 | sivana | and the wiki |
| 04:22.50 | {Sean} | yeah mpg123 is fine |
| 04:22.55 | {Sean} | config is fine |
| 04:22.59 | {Sean} | it just doesn't start mpg123 |
| 04:23.11 | {Sean} | i found a thread about it w/ no resolution |
| 04:23.17 | `Sauron | I think the concept of a working MoH is nothing but a tease thing. |
| 04:23.27 | `Sauron | Everybody says it's there, but nobody can get it working :) |
| 04:23.29 | chrisf0rd | Sivana I am goon throw you soem text in a chat box so you can see my CLI> scroll off is that ok |
| 04:23.31 | {Sean} | i've had it working when * runs as root regularly |
| 04:23.33 | sivana | I dunno.. mine works |
| 04:23.37 | sivana | chrisf0rd: www.pastebin.ca |
| 04:24.07 | {Sean} | sivana your MOH works w/ * running as a service? |
| 04:24.11 | sivana | yes |
| 04:24.17 | *** join/#asterisk beto75 (~beto75@201.128.177.84) |
| 04:24.17 | {Sean} | weird |
| 04:24.27 | beto75 | hello Guys |
| 04:24.31 | sivana | what line do you have in the musiconhold.conf? |
| 04:24.40 | {Sean} | standard default |
| 04:24.43 | beto75 | excuse me is JerJer Online? |
| 04:24.53 | sivana | chrisf0rd: don't use the proxies either |
| 04:24.54 | sivana | :) |
| 04:25.06 | sivana | ~seen JerJer |
| 04:25.08 | jbot | jerjer is currently on #asterisk |
| 04:25.13 | {Sean} | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 04:25.16 | `Sauron | Hehe |
| 04:25.22 | chrisf0rd | dont use the proxies |
| 04:25.29 | {Sean} | mpg123 is in the right path |
| 04:25.29 | chrisf0rd | what do you use then |
| 04:25.34 | sivana | chrisf0rd: no, use sip.broadvoice.com |
| 04:25.44 | sivana | sean: doesn't look like your using mpg123 |
| 04:25.49 | sivana | let me check mine |
| 04:25.58 | toddf | Sean what version of asterisk are you using? |
| 04:26.04 | `Sauron | sivana: the double user/pass thing works fine now |
| 04:26.07 | `Sauron | with broadvoice |
| 04:26.12 | chrisf0rd | use chrisf0rd: no, use sip.broadvoice.com instead of proxy.dca.broadvoice.com |
| 04:26.17 | hacim | thetalon: dangit, iaxcomm doesn't work either :P |
| 04:26.30 | sivana | {Sean}: default => mp3:/var/lib/asterisk/mohmp3 |
| 04:26.38 | `Sauron | chrisford: Look at the voip-info.org example for getting * to talk to broadvoice |
| 04:26.44 | sivana | `Sauron: I could never get it to work |
| 04:26.57 | sivana | `Sauron: BV works fine for me without the double thing |
| 04:27.00 | toddf | sivana I am having music on hold not working for me too, used 1.0.2, 1.0.5, stable, now compiling current cvs |
| 04:27.01 | beto75 | guys I am looking for jerjer becaus eI am using H323 and I have a real mess the calls are suddenly cut off without a pattern |
| 04:27.10 | beto75 | someone here has any idea why? |
| 04:27.17 | `Sauron | sivana: shrug, dunno - it worked for me and randu and a couple others |
| 04:27.21 | sivana | beto75: he's here |
| 04:27.26 | `Sauron | which is why I updated the voip-info page |
| 04:27.37 | sivana | what's the link on wiki page? |
| 04:27.48 | `Sauron | Don't remember the url :) |
| 04:27.53 | sivana | `Sauron: that's using the info on BV site? |
| 04:27.54 | hacim | thetalon: both asterisk and iaxcomm want to use port 4569 |
| 04:28.10 | `Sauron | sivana: It's using the CORRECT info, not neccesarily what BV's site says. |
| 04:28.12 | {Sean} | Asterisk CVS-05/20/04-12:23:42 |
| 04:28.21 | `Sauron | I already told BV support that their site's info isn't completely accurate |
| 04:28.22 | sivana | `Sauron: ok |
| 04:28.33 | chrisf0rd | sivana I can call out now |
| 04:28.36 | `Sauron | I'll find the page |
| 04:28.40 | toddf | Sean, so basically, you're using quite an older version of asterisk, nothing ''recent'' |
| 04:28.44 | chrisf0rd | but when I try to call back in I get teh party is busy recording |
| 04:28.46 | sivana | chrisf0rd: the proxies? |
| 04:29.01 | chrisf0rd | I cahnged them |
| 04:29.03 | `Sauron | http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broadvoice |
| 04:29.04 | chrisf0rd | changed |
| 04:29.13 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:29.15 | chrisf0rd | Tried it Sauron |
| 04:29.19 | chrisf0rd | no help |
| 04:29.23 | {Sean} | its recent as of a couple months ago :) |
| 04:29.23 | `Sauron | But you have to call support for the sip password |
| 04:29.37 | `Sauron | chrisford: It works, I know it does. |
| 04:29.45 | beto75 | JerJer: Please Help me !!!!!!! |
| 04:29.50 | sivana | `Sauron: I see.. you removed the proxies |
| 04:29.50 | chrisf0rd | I have a password |
| 04:29.51 | {Sean} | so was there a bugfix or something in the last couple of months for this problem? |
| 04:30.04 | `Sauron | I configured it less than a week ago, and the page is an exact copy of my currently running configuration |
| 04:30.07 | chrisf0rd | the one they gave me when I bought the service |
| 04:30.14 | chrisf0rd | ok |
| 04:30.16 | chrisf0rd | just asec |
| 04:30.20 | `Sauron | chrisford: did you get a BYoD account? |
| 04:30.47 | chrisf0rd | yes |
| 04:31.17 | `Sauron | then the example config should work |
| 04:31.46 | chrisf0rd | for inbound calling |
| 04:31.46 | sivana | `Sauron: looks good to me |
| 04:32.14 | `Sauron | They even support dtmfmode=rfc2388 now |
| 04:32.58 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 04:33.23 | chrisf0rd | let me make sure I understand |
| 04:33.43 | chrisf0rd | this config will let inbound calls come in and goto the auto attendant |
| 04:34.04 | `Sauron | You'll have to modify the extensions.conf for that |
| 04:34.23 | `Sauron | but the example should be enough to get you going with the BV config |
| 04:35.06 | chrisf0rd | I have been at this since 3pm today and am going to have a nervious breakdown (-; hahaha |
| 04:35.43 | *** join/#asterisk {Sean} (~sean@adsl-69-214-130-169.dsl.lgtpmi.ameritech.net) |
| 04:35.48 | {Sean} | actually got some error output now |
| 04:35.49 | {Sean} | res_musiconhold.c:305 moh0_exec: Unable to start music on hold (class '') on channel SIP/sean-4cee |
| 04:36.20 | `Sauron | chrisford: I spent 3 days straight. You won't have a nervous breakdown anytime soon |
| 04:36.39 | chrisf0rd | THANK YOU sivana and Sauron you guys ROCK!!!!!!!!!!!!!!!!!! |
| 04:36.43 | chrisf0rd | it works |
| 04:36.47 | chrisf0rd | YEAHHHHHHHHHHHHHHHHHH |
| 04:38.44 | `Sauron | I told you it'll work. |
| 04:38.46 | `Sauron | ;) |
| 04:41.27 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 04:41.35 | hacim | arg |
| 04:44.42 | chrisf0rd | Yes you did |
| 04:45.05 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
| 04:46.46 | niZon | will asterisk run on a 64 bit OS? |
| 04:49.10 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 04:49.55 | zimdog | Do I just need port 5060 forwarded through nat firewall for an outside client to connect? |
| 04:50.09 | zimdog | 5060 UDP |
| 04:51.22 | Grooby | yes |
| 04:51.33 | niZon | you need 10000 to something as well |
| 04:51.41 | Grooby | RTP ports |
| 04:51.45 | Grooby | 10000-20000 by default |
| 04:51.51 | niZon | thats it :P |
| 04:51.54 | Grooby | or change it in the rtp.conf in /etc/asterisk |
| 04:52.25 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 04:52.47 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 04:52.47 | zimdog | My firewall mentions forwardign udp or tcp what does rtp use? |
| 04:52.52 | Grooby | udp |
| 04:53.05 | dontmsgme | What does this mean: Feb 3 20:45:05 NOTICE[-245572688]: chan_iax2.c:1311 iax2_destroy: Avoiding IAX destroy deadlock |
| 04:53.32 | Grooby | no clue...haven't touch IAX yet |
| 04:54.31 | zimdog | Thanks grooby and niZon |
| 04:54.50 | Grooby | no problem |
| 04:56.51 | *** part/#asterisk santiago (~santiago@63.245.86.104) |
| 04:57.04 | *** join/#asterisk Tough_Nuts (~chatzilla@26.137.202.68.cfl.rr.com) |
| 05:01.54 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 05:03.35 | chrisf0rd | <`Sauron> are you still around |
| 05:03.43 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 05:05.59 | chrisf0rd | is anyone here |
| 05:06.02 | chrisf0rd | ??? |
| 05:06.05 | chrisf0rd | yoo hooo |
| 05:06.18 | NormAst | Any Debian Pro's on tonight? |
| 05:06.47 | Grooby | i am around |
| 05:06.48 | Grooby | sorta |
| 05:07.23 | NormAst | Grooby: Trying to get zaptel drivers to compile on Debian Sarge... NetInstall... all kinds of mod dep errors. |
| 05:08.14 | Grooby | Norm, wish I can help you |
| 05:08.18 | Grooby | i don't even have zaptel hardware |
| 05:08.23 | NormAst | hmmm.... :) |
| 05:08.33 | NormAst | you can still compile them... :) |
| 05:08.44 | NormAst | you might want to get ztdummy going. |
| 05:08.53 | Grooby | i got ztdummy |
| 05:08.55 | Grooby | hehehe |
| 05:09.07 | NormAst | Did you compile it? |
| 05:09.12 | Grooby | but i cheated my way around via asterisk at home |
| 05:10.07 | Grooby | are you using the newest zaptel? |
| 05:11.52 | postel | NormAst: pastebin your errors |
| 05:12.15 | NormAst | I see that * @ home compilies asterisk on first boot.. |
| 05:12.55 | NormAst | postel: Just compiling the kernel to see if it will fix some of the dep. errors. |
| 05:14.02 | postel | NormAst: well, thats nice, i still want your errors tho |
| 05:15.05 | NormAst | need to wait for the kernel to finish... |
| 05:15.48 | postel | ... |
| 05:16.00 | NormAst | ................ |
| 05:16.08 | Tough_Nuts | Hello all.. Just wanted to see if anyone here knew some things about using SJPhone and * thru a NAT.. |
| 05:16.12 | NormAst | P4 2.4 512megs.. |
| 05:16.12 | Grooby | ......zzzzZZZZZZZ |
| 05:16.35 | Grooby | tough_nuts...i run sjphone w/ * thru nat |
| 05:16.54 | Grooby | we... * behind nat and sjphone outside when I am on client sites |
| 05:17.49 | Tough_Nuts | cool.. I run it thru a nat as well.. and it seems to work. what I want to do is have a buddy of mine using sjphone, and he is also behind a nat... |
| 05:17.51 | Tough_Nuts | I want to make his sjphone, an ext off my *.. |
| 05:18.13 | Grooby | is your router forwarding ports? |
| 05:18.25 | Grooby | 5060 udp, 10000-20000 udp to the * box? |
| 05:18.53 | Tough_Nuts | his phone cant see me.. if I have the router configed, it SHOULD be showing the whole dam box thru the nat.. :) |
| 05:19.12 | Tough_Nuts | its a dlink, and I have the * in the dmz.. |
| 05:20.00 | Grooby | and is he shown up when you type sip show peers? |
| 05:20.01 | Tough_Nuts | I can hit other ports of the box from outside.. and they are not setup as forwarded.. |
| 05:20.33 | Tough_Nuts | didnt know to type that.. but his sjphone says it cant reg anyway.. |
| 05:20.49 | Grooby | and he's hitting your extern IP right? |
| 05:20.50 | *** join/#asterisk Andrey_Kirov (~Andrey_Ki@16-51-customer.kirov.mtsnet.ru) |
| 05:21.08 | Tough_Nuts | he SAYS he is.. :)) so I hope its true.. |
| 05:21.47 | Grooby | lol |
| 05:22.02 | Tough_Nuts | I mean he has the correct IP for my outside IP f the router... |
| 05:22.11 | Grooby | ok |
| 05:22.11 | Andrey_Kirov | Hi everybody! |
| 05:22.17 | Grooby | hello |
| 05:22.46 | Grooby | tough: in your sip.conf do you have localnet=internalip (i.e. 192.168.1.0) and localmask=255.255.255.0 |
| 05:22.53 | Grooby | and externip=your external ip? |
| 05:23.29 | Tough_Nuts | nope... and the gw here should be 192.168.01 |
| 05:23.45 | Andrey_Kirov | Somebody can help me with h323 addon? |
| 05:24.18 | Nivex | any idea if FWD's IAX server is ever going to come back up? |
| 05:24.53 | Grooby | add those in the sip.conf |
| 05:24.59 | Grooby | and reload |
| 05:24.59 | postel | Tough_Nuts: 192.168.01 aint a valid RFC 1918 address |
| 05:25.01 | Grooby | see if that helps |
| 05:25.19 | Grooby | yean..192.168.0.1 is your internal ip |
| 05:25.28 | Tough_Nuts | sry.. 192.168.0.1 |
| 05:25.32 | Grooby | if you have a dlink router, check the status page |
| 05:25.39 | beto75 | any h323 guru here? |
| 05:25.40 | Grooby | it should shown your WAN IP address |
| 05:25.59 | Tough_Nuts | yep.. got that.. where is sip.conf do I add this ? |
| 05:26.00 | beto75 | asterisk h323 guru sorry |
| 05:26.48 | Andrey_Kirov | beto75: what's problem? |
| 05:26.55 | Grooby | under [general] |
| 05:27.12 | Grooby | if your friend is gonna be using this long term |
| 05:27.21 | Grooby | i suggest you sign up a dyndns domain |
| 05:27.22 | beto75 | Andrey: some calls are cutt off but can be at 10 seconds, 40 seconds 3 minutes last |
| 05:27.28 | Tough_Nuts | i dunno, just messing for now.. |
| 05:27.36 | Tough_Nuts | got one of those already... |
| 05:27.47 | Grooby | ok |
| 05:27.56 | Grooby | then externip=your dydns domain |
| 05:27.58 | Grooby | makes your live easier |
| 05:27.58 | beto75 | Andrey I have also a gnugk with radius (for billing in the same box) |
| 05:28.04 | Grooby | and just have your friend hit your dyndns |
| 05:28.11 | Andrey_Kirov | beto75: sorry, but i can't help you :( |
| 05:28.16 | beto75 | Andrey : this issue is new since a couple of weeks |
| 05:29.19 | Tough_Nuts | for localnet=internalip do I need to put in the netmask some where as well ? |
| 05:29.33 | Grooby | that's what localmask=255.255.255.0 is for |
| 05:29.34 | Grooby | ;) |
| 05:29.56 | Tough_Nuts | oh... duh.. sry.. didnt see that.. |
| 05:29.59 | Andrey_Kirov | beto75: may be you can help me to compile h323 addon on fc3? |
| 05:30.01 | Grooby | it's all good |
| 05:30.05 | beto75 | Andrey , what you recomend to see what the heck happens in here? |
| 05:30.52 | Tough_Nuts | just to make sure I know I got it right... 192.168.0.1 is mask 255.255.0.255 ? |
| 05:30.58 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) [NETSPLIT VICTIM] |
| 05:30.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) [NETSPLIT VICTIM] |
| 05:30.58 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 05:30.58 | *** join/#asterisk angler_ (~angler@suid.digium.com) [NETSPLIT VICTIM] |
| 05:31.12 | Grooby | localnet=192.168.0.0 |
| 05:31.16 | Grooby | localmask=255.255.255.0 |
| 05:31.20 | Grooby | use those 2 |
| 05:31.27 | Grooby | and externip=your dyndns domain name |
| 05:31.32 | Tough_Nuts | ah.. ok.. not the gw, but the network.. |
| 05:31.35 | niZon | who said something about FWD's IAX being down? |
| 05:31.40 | Grooby | yup |
| 05:31.41 | beto75 | Andrey, well I dindnt do it , there is a pal that can do it ,, max litnisky litnimax@asterisk-support.ru |
| 05:31.41 | beto75 | I hope he can help me when he gets online later |
| 05:31.41 | *** part/#asterisk thetalon (~Ari@pcp05736786pcs.norstn01.pa.comcast.net) |
| 05:31.50 | Tough_Nuts | thanks much.. |
| 05:32.40 | Grooby | don't thank us yet |
| 05:32.42 | Grooby | til it's working |
| 05:32.43 | Grooby | hehehe |
| 05:33.31 | Tough_Nuts | you know what i mean.. hehe :) |
| 05:33.36 | NormAst | .....zzzzZZZZZ |
| 05:34.13 | Grooby | ..zzzZZZZZ |
| 05:34.24 | Grooby | or in the UO world "oooOoOOoooOOOO" |
| 05:34.48 | Andrey_Kirov | Beto, i am a not professional in asterix. I just hoped that at you the same problem |
| 05:35.50 | Andrey_Kirov | Beto: thank you for help |
| 05:40.58 | brc__ | what is asterix? |
| 05:41.22 | Grooby | * |
| 05:41.30 | beto75 | brc: asterix is Obelissc partner |
| 05:41.37 | beto75 | here is Asterisk :) |
| 05:41.59 | brc__ | beto75, I believe you mean obelix |
| 05:42.14 | beto75 | sorry for typo |
| 05:42.21 | brc__ | sallgood |
| 05:42.24 | beto75 | I think you also Typo asterisk |
| 05:43.25 | beto75 | asterisk is a soft pbx with VOIP options, muticodec , multy protocol |
| 05:43.48 | beto75 | tdm, h323, sip , mgcp , IAX (any other forgotten?) |
| 05:43.59 | cypromis | sccp |
| 05:44.04 | beto75 | sorry |
| 05:44.08 | cypromis | :D |
| 05:44.14 | cypromis | you forgot the headaches |
| 05:44.15 | beto75 | giack cisco |
| 05:44.18 | cypromis | I'd say |
| 05:44.27 | cypromis | they come free |
| 05:44.28 | cypromis | :D |
| 05:45.07 | beto75 | cypromis: what can be this world without headaches |
| 05:45.08 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-82-48.sentechsa.net) |
| 05:45.14 | brc__ | cypromis, and the migraines... |
| 05:45.57 | beto75 | cypromis: if yu have headaches with a <5k legacy brand pbx ,, I prefer headaches with asterisk :) |
| 05:46.13 | beto75 | by far |
| 05:46.34 | brc__ | the ivr scripting induced, gotoif riddled, priority hell that is asterisk configuration |
| 05:46.40 | *** join/#asterisk unixgeek (~unixgeek@216-220-234-197.exploremaine.com) |
| 05:46.46 | brc__ | hm |
| 05:46.51 | brc__ | I can do better then that |
| 05:46.53 | brc__ | ah well |
| 05:46.54 | brc__ | tata |
| 05:47.33 | Tough_Nuts | Anyone here using the Asterisk@home iso load with CentOS ? |
| 05:47.44 | Grooby | oh |
| 05:47.45 | Grooby | mememe |
| 05:47.46 | brc__ | no |
| 05:47.53 | Grooby | lol |
| 05:48.06 | Grooby | what's up TN? |
| 05:48.12 | Tough_Nuts | its NOT al that bad... is it ? :) |
| 05:48.20 | Grooby | i like it |
| 05:48.33 | Grooby | it's a start that I can go around and modify |
| 05:48.34 | Grooby | hehehehe |
| 05:48.59 | datareactor | anyone using go2call with asterisk i am trying from manys days but no successful |
| 05:49.05 | Tough_Nuts | I just wish I could the Digital Asst to work.. cant figure out howto get that going.. |
| 05:49.14 | *** join/#asterisk |^Angel^| (~funschrip@port336.ds1-alb.adsl.cybercity.dk) |
| 05:49.22 | Grooby | digital receptionist? |
| 05:49.27 | Tough_Nuts | yes.. |
| 05:49.36 | Grooby | what are you trying to do w/ it? |
| 05:50.08 | Tough_Nuts | well, without it all I can do is have it answer the phone auto transfer to a preset ext... |
| 05:50.16 | *** join/#asterisk pranav (dawda_pran@203.115.89.185) |
| 05:50.38 | Grooby | i basically dial *77 |
| 05:50.42 | Tough_Nuts | I just want alittle IVR, press 1 for this, 2 for that.. and so foth.. |
| 05:50.42 | Grooby | record my message |
| 05:50.56 | Grooby | and continue the setup in AMP |
| 05:51.07 | pranav | hi grooby |
| 05:51.12 | Grooby | hi pranav |
| 05:51.17 | Tough_Nuts | what kinda of hw you using, TDM or X100 ? |
| 05:51.28 | pranav | hi i have a problem in astguiclient |
| 05:51.29 | *** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net) |
| 05:51.39 | Grooby | analog phone => sipura 2k => * |
| 05:51.44 | Grooby | no X100 |
| 05:52.02 | jdv79 | i just bought a budgetone 100 and the manual says the default http passwd is admin but that's not working |
| 05:52.02 | Grooby | pranav, what's the problem? |
| 05:52.07 | jdv79 | anyone know what it could be? |
| 05:52.13 | Tough_Nuts | great... well, I only got the x100p, so i cant dial *77 from inside the pbx... |
| 05:52.14 | BoRiS | jdv: just hit enter |
| 05:52.29 | pranav | Grooby I am also using sipura 200 , at one point in the astguiclient site it says insert the phone values |
| 05:52.29 | Grooby | sure you can |
| 05:52.35 | Grooby | that's what AMP is designed for |
| 05:52.43 | jdv79 | not working either |
| 05:52.48 | Grooby | phone values? |
| 05:52.53 | Grooby | would that be extension of the phone? |
| 05:53.02 | Grooby | i've actually never use astguiclient.... |
| 05:53.06 | Grooby | i sorta went with AMP |
| 05:53.16 | Tough_Nuts | AMP is really nice... |
| 05:53.29 | pranav | i am using the site http://astguiclient.sourceforge.net/scratch_install.html |
| 05:53.31 | Grooby | jdv79, try a factory reset if there's such thing ont hose phone |
| 05:53.43 | *** join/#asterisk doushanes (~doushanes@c-67-184-189-220.client.comcast.net) |
| 05:53.47 | brc__ | apple pepsi itunes ad leaked http://www.appleinsider.com/article.php?id=866 |
| 05:53.50 | *** join/#asterisk Corydon76-home (three@pcp08665860pcs.500ash01.tn.comcast.net) |
| 05:53.57 | Tough_Nuts | can I do the *77 from my sjphone ? |
| 05:54.26 | pranav | in this if you can check out point number 6.1 wheree it asks to enter the phone values |
| 05:54.28 | Grooby | yes you can |
| 05:54.31 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 05:54.40 | Grooby | tough nuts, you have your analog phone connect to x100 right? |
| 05:54.48 | Grooby | pranav, looking now |
| 05:55.06 | SexyKen | Hey guys -- I have SoundPoint IP 600 VOIP Phones that multiple people are going to be using. Is there anyway to have them be able to login to the asterisk system through the phone before they can make calls? |
| 05:55.08 | Tough_Nuts | thats the wrong end... :) the x100p is FXO not FXS... |
| 05:55.17 | Grooby | ah ok |
| 05:55.19 | Grooby | gotcha |
| 05:56.21 | cypromis | SexyKen: authenticate is your friend |
| 05:56.53 | SexyKen | cypromis What do you mean |
| 05:56.55 | doushanes | anyne have a recommendation for a sip phone in the $100-150 range? |
| 05:57.01 | moonwick | I've got a question that someone here might know the answer to... let's say I've got a voice t1 run into an asterisk server. If all the channels are full and another call comes in, will I know about it? or will the PSTN just shoot back a busy signal without telling me? |
| 05:59.16 | SexyKen | So is it possible to have users enter their extension and password and then calls will be routed to that specific phone? |
| 05:59.28 | datareactor | moonwick you will dialing person will get busy tone |
| 06:01.17 | jdv79 | ah, resetting did it |
| 06:01.18 | jdv79 | thanks |
| 06:01.22 | moonwick | datareactor: so I'll still know about calls that get rejected due to busy? |
| 06:01.28 | jdv79 | PITA to reset, probably a good thing:) |
| 06:01.44 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 06:02.38 | *** join/#asterisk wasim_ (~wasim@203.81.200.8) |
| 06:03.32 | jdv79 | how does one config an IAXy? |
| 06:03.47 | jdv79 | does it require DHCP or something? |
| 06:04.06 | wasim_ | iax2-provision |
| 06:04.21 | jdv79 | i mean how does the thing itself get an IP |
| 06:04.50 | JerJer | jdv79: DHCP |
| 06:04.59 | jdv79 | no static? |
| 06:05.04 | JerJer | you can static it, sure |
| 06:05.11 | jdv79 | what does it have now? |
| 06:05.19 | jdv79 | i just got one and don't know how to get it workin |
| 06:05.26 | JerJer | if you just received it, then it has nothing |
| 06:05.28 | JerJer | you have to provision it |
| 06:05.43 | jdv79 | ok, will do |
| 06:05.43 | JerJer | use iaxyprov |
| 06:05.51 | JerJer | cvs co iaxyprov |
| 06:05.57 | jdv79 | ah |
| 06:06.04 | denon | why would you own an iaxy without having the nufone service to go with it? |
| 06:08.47 | datareactor | can someone help setting up * with go2call |
| 06:09.21 | chipig | hmm. I don't suppose anyone round here has had luck getting intercom to work with snom 190 phones? |
| 06:09.26 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 06:10.39 | WilliamK | chipig, hadn't tried yet |
| 06:11.11 | chipig | some mailing list talk about the snom200 from august.. |
| 06:11.44 | chipig | but, nothing recently, and I when I send it, the snom sends back requiring Digest Authentication |
| 06:12.06 | PyroSteve | hey guys |
| 06:12.19 | PyroSteve | can i post a short amount of my dial plan here |
| 06:12.30 | PyroSteve | im trying to load balence my outgoing calls |
| 06:12.32 | sivana | ~pastebin |
| 06:12.33 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 06:12.53 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 06:13.08 | *** join/#asterisk nullogic (~nullogic@c-24-98-72-110.atl.client2.attbi.com) |
| 06:13.42 | PyroSteve | http://pastebin.ca/5264 |
| 06:13.56 | PyroSteve | take a loon at that sample random useage |
| 06:14.17 | PyroSteve | i want to spread my usage across 4 sip lines for outgoing calls |
| 06:14.32 | PyroSteve | is that a correct way of implementing that |
| 06:15.13 | PyroSteve | the say digits 1-4 is just a subsution for what line will be used |
| 06:15.18 | SexyKen | Yea well I want users to be able to go to a phone that is already logged into the Asterisk system, but it isn't allowed to do anything...it's just logged in. Then users have to dial a number and are prompted for their Extension and a password, then the phone recieves calls for that person and they can make calls |
| 06:15.19 | SexyKen | Possible? |
| 06:15.36 | PyroSteve | and the goto statement is so i can see how good random works |
| 06:15.42 | PyroSteve | am i on the right track ? |
| 06:16.08 | nullogic | any reason to record all three : format=wav49|gsm|wav ? |
| 06:16.12 | SexyKen | Is what I want to do possible? |
| 06:16.35 | JerJer | nullogic: no |
| 06:16.43 | JerJer | unless you like to burn HD space |
| 06:17.20 | JerJer | SexyKen: you are not making much sense |
| 06:17.22 | JerJer | at least to me |
| 06:17.32 | postel | he does to me |
| 06:17.55 | JerJer | then answer his question |
| 06:18.17 | PyroSteve | what about me !?! |
| 06:18.31 | postel | JerJer: he want registered clients but not in a context that allows incoming/outgoing, then they authenticate against a context that CAN do the above and dial out/in |
| 06:18.37 | JerJer | PyroSteve: i don't see anything about SIP in that post |
| 06:19.07 | JerJer | postel: ok |
| 06:19.18 | JerJer | so make a context with some exten in it |
| 06:19.18 | PyroSteve | JerJer: dont see SIP in what post ? |
| 06:19.29 | PyroSteve | my pastebin post ? |
| 06:19.38 | JerJer | make that exten do the authentication process |
| 06:19.43 | JerJer | next |
| 06:19.50 | nullogic | JerJer: .wav files are almost 10x the gsm files... most of my end users use windows so will the gsm play on media player |
| 06:20.02 | postel | JerJer: do it right, its NEXT! ;-) |
| 06:20.10 | JerJer | if you have the gsm codec installed |
| 06:20.19 | JerJer | you = end lusers |
| 06:21.04 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 06:21.13 | nullogic | will asterisk allways playback the .gsm files? |
| 06:21.15 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 06:21.32 | shaZwaz | morning room |
| 06:21.55 | JerJer | nullogic: if you have the gsm codec installed |
| 06:22.01 | JerJer | is there an echo in here? |
| 06:22.19 | postel | SexyKen: well, i think what JerJer said along with some kind of callcard app would do what you want, drop everyone on an ext that does auth and have them jump to a context that allows in/out after they submit auth info |
| 06:22.54 | nullogic | JerJer: that was a different question.. |
| 06:22.59 | SexyKen | postel Interesting...and then have an extension that logs them out as well? |
| 06:23.05 | JerJer | doesn't even have to be a seperate context |
| 06:23.18 | JerJer | the authentication process can make the call happen |
| 06:23.25 | JerJer | or nto |
| 06:23.26 | JerJer | not |
| 06:23.47 | SexyKen | JerJer, do you understand the point for this? |
| 06:24.00 | Qwell | hmm |
| 06:24.03 | JerJer | i don't even really understand what you are asking |
| 06:24.07 | JerJer | so, no |
| 06:24.09 | Qwell | How common is a setup with no hitches? |
| 06:24.14 | shaZwaz | Q is there some way to run some part of dialplan or script at * startup/reload to fetch vales from astdb ? |
| 06:24.22 | postel | Qwell: 39.67% |
| 06:24.26 | Qwell | postel: :p |
| 06:24.52 | Qwell | That was...simple. Installed an x100p, worked straight away |
| 06:25.14 | SexyKen | Lets assume I have 4 VOIP phones in an Office. I have 8 techs. Only 4 are on shift at a time. So I only need 4 phones. But I dont want techs to share extensions because A: I want them to have their own voicemail, B: Their own extensions and C: I just dont. But I'm not going to buy 8 phones if they're never gonna be in use all at once. |
| 06:25.36 | SexyKen | It's a pain in the ass to have people change the actual login on the phone everytime they use it, that's not very efficient. |
| 06:26.07 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 06:26.08 | SexyKen | So I want the phone to be logged in at all times. It just doens't do anything unless a user 'logs in' to the system. Once they login, the phone should ring for that extension and make calls from that extension |
| 06:26.11 | JerJer | so then build dialplan logic that authenticates them and keeps track of who is online |
| 06:26.38 | SexyKen | JerJer Does that mean program the phone itself or program something within Asterisk? |
| 06:26.42 | silik0n | or buy a commercial system that does it hah |
| 06:26.48 | JerJer | smells like a perfect job for the asterisk db |
| 06:27.03 | silik0n | program a dialplan where the user can login and then log out... |
| 06:27.25 | ta[i]nted | do you guys just colo your * box or host it in house? |
| 06:27.25 | SexyKen | Interesting. |
| 06:27.25 | JerJer | ta[i]nted: all of the above |
| 06:27.25 | SexyKen | Colo. |
| 06:27.36 | SexyKen | I am going to pay someone to do it. |
| 06:27.39 | Qwell | ta[i]nted: in-livingroom |
| 06:27.46 | *** join/#asterisk Juggie (agony@24.114.136.55) |
| 06:27.54 | JerJer | my new house has Asterisk wired into the house |
| 06:28.02 | JerJer | complete with a TA-750 |
| 06:28.04 | Qwell | JerJer: really? |
| 06:28.14 | JerJer | hell yes |
| 06:28.19 | silik0n | the 8 techs each have "virtual" extensions that are routed to one of the 4 real live SIP/whatever extensions based on where they logged in/out from... |
| 06:28.23 | Qwell | You go in before they were finished? |
| 06:28.30 | JerJer | i built it |
| 06:28.40 | Qwell | before, or after it was built? |
| 06:28.42 | clive- | jerjer, I can imagine your house wired up with more bandwidth than we have in africa |
| 06:28.42 | ta[i]nted | is it cheaper to colo than to bring in a T1 line? |
| 06:28.51 | JerJer | i didn't even have copper brought in from the local copper farmer |
| 06:28.59 | SexyKen | silik0n, I understand the concept I just dont have the knowledge to do it. |
| 06:29.08 | JerJer | clive-: i have 27 megabits to the house on a redline link |
| 06:29.17 | Qwell | Thats just sick |
| 06:29.20 | JerJer | 27 or 24 can't remember... a couple megs |
| 06:29.34 | silik0n | well support asterisk and hire one of the dev guys to set it up for you |
| 06:29.36 | Qwell | Thats gotta cost a pretty penny, eh? |
| 06:29.50 | JerJer | about 3 grand for the redline link |
| 06:29.56 | JerJer | another grand for the tower |
| 06:30.04 | Qwell | monthly? |
| 06:30.08 | JerJer | um no |
| 06:30.11 | Qwell | oh, heh |
| 06:30.18 | SexyKen | silik0n I will definatley hire someone. How much? |
| 06:30.21 | znoG | JerJer: any ideas why i get invalid context when i try to dial out on NuFone? |
| 06:30.33 | JerJer | znoG don't sepcificy a context on a type=peer |
| 06:30.46 | znoG | i don't |
| 06:30.47 | JerJer | and make sure you are sending a 1 or 011 |
| 06:30.50 | clive- | jerjer sounds awesome |
| 06:30.53 | znoG | ah |
| 06:30.56 | znoG | might be the 011 |
| 06:31.07 | JerJer | and send a valid caller*id |
| 06:31.10 | JerJer | value |
| 06:31.19 | znoG | is that new? cause it used to work fine |
| 06:31.29 | JerJer | these lameass sip clients out there like to send "bob" as the caller*id number and my switch gets pissed off at that |
| 06:31.56 | sivana | hehe |
| 06:32.49 | *** join/#asterisk DrmC (drmc@66.150.13.18) |
| 06:33.04 | ta[i]nted | i'm trying to set up a voice mail system using asterisk.. and was wondering if it'd be cheaper to bring in a T1 line or just to colocate it somewhere? |
| 06:33.24 | JerJer | depends |
| 06:34.21 | JerJer | hell why do you need a T-1? |
| 06:34.30 | *** join/#asterisk Guest^DJ (some@211.24.146.10) |
| 06:34.38 | DrmC | channel volume im sure |
| 06:35.00 | JerJer | just turn the gain up then |
| 06:35.12 | DrmC | not amplitude |
| 06:35.28 | DrmC | quantity |
| 06:35.32 | Guest^DJ | hi, there is an article on wiki that * can work with old Panasonic, i just could not figure out how they connect to each other? any one ? |
| 06:35.37 | znoG | all good now JerJer, thanks |
| 06:35.55 | znoG | no idea why i get 480ms pings to you though |
| 06:35.57 | JerJer | please pull forward to pay your bill |
| 06:35.59 | znoG | thats the sort of pings i get to AU |
| 06:36.26 | JerJer | try switch-2.nufone.net |
| 06:36.30 | znoG | oh wait, its me |
| 06:36.40 | znoG | pinging at 800ms to AU... something funny going on |
| 06:36.54 | DrmC | 800 ms! |
| 06:36.56 | JerJer | icmp likes to lie from time to time |
| 06:37.35 | DrmC | lie? |
| 06:37.45 | SexyKen | Seriously lol. |
| 06:37.46 | SexyKen | Lie? |
| 06:37.46 | denon | icmp has a very low priority on most routers |
| 06:37.57 | SexyKen | Well that makes sense. |
| 06:38.00 | SexyKen | It's a truth though. |
| 06:38.00 | JerJer | sure routers can be configured to hold on to icmp packets |
| 06:38.01 | denon | you prioritize whats important.. to heck with the rest |
| 06:38.08 | DrmC | its true ... |
| 06:38.10 | denon | especially when its often used to DDoS |
| 06:38.10 | JerJer | HELL |
| 06:38.17 | JerJer | to hell with the rest! |
| 06:38.23 | DrmC | not hold on to them |
| 06:38.27 | SexyKen | denon You a developer of Asterisk? |
| 06:38.28 | DrmC | just drop them |
| 06:38.28 | denon | keepin' it clean, family channel .. |
| 06:38.34 | denon | SexyKen: no, I just play one on tv |
| 06:38.40 | JerJer | we are in safe harbor hours |
| 06:38.42 | SexyKen | Oh stop. |
| 06:38.47 | Qwell | denon: Thanks, my wife and kids like to hang out in here :p |
| 06:38.59 | DrmC | processing them 'held' would take far more cpu time than a simple drop |
| 06:39.06 | denon | SexyKen: JerJer wrote all of asterisk single-handedly last week .. privmsg him with any questions |
| 06:39.13 | SexyKen | denon, I need a little bit of a custom configuration for my asterisk setup. |
| 06:39.18 | denon | that's nice |
| 06:39.24 | DrmC | anyways you can do a tcp ping and get around all that |
| 06:39.24 | SexyKen | denon, interested in a litte side money or no? |
| 06:39.28 | denon | nah |
| 06:39.39 | JerJer | nor am I |
| 06:39.45 | SexyKen | Know of any other developers who'd be interested? |
| 06:39.52 | Qwell | I'd love to, but I do shoddy work :p |
| 06:39.53 | denon | SexyKen: i dunno, msg me what kinda work .. |
| 06:41.14 | *** join/#asterisk Tough_Nuts (~chatzilla@26.137.202.68.cfl.rr.com) |
| 06:42.02 | DrmC | heh |
| 06:43.58 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 06:45.19 | Tough_Nuts | still trying to get an sjphone outside my nat working.. any takers ? :) |
| 06:45.54 | Primer | sjphone Just WorksâĒ for me |
| 06:46.29 | Tough_Nuts | i think we have the settings on my buddies phone wrong... |
| 06:47.40 | Primer | is either behind NAT? |
| 06:47.51 | Tough_Nuts | both are.. :( |
| 06:47.55 | Primer | that's why |
| 06:48.10 | Primer | you need to either DMZ each box or redirect the appropriate ports |
| 06:48.26 | Primer | as I'm sure you know |
| 06:49.45 | Tough_Nuts | ok.. here is the setup.. 1 sjphone and an * behind a nat. But in dmz.. the other sjphone behind a nat. |
| 06:49.47 | Tough_Nuts | I dont know that I got all the ports right, and I am not sure the settings in his sjphone are correct.. |
| 06:50.20 | pranav | ghfgh |
| 06:59.01 | simon_ca | how do you set calling name on a zap fxo (tdm400p)? i tried callerid="name <1000>" in zapata.conf w/ no luck... |
| 07:01.12 | wasim_ | simon_ca: setcallerid() in the dialplan |
| 07:03.14 | simon_ca | wasim: i do that when i go out a trunk (i.e. x100p), but trying to get the tdmp400p to have name associated w/ it when it calls sip extensions... |
| 07:04.30 | cypromis | it is callerid = "name" <number> |
| 07:04.45 | cypromis | that is probablyyour problem |
| 07:05.13 | simon_ca | thanks |
| 07:05.27 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 07:05.37 | cypromis | try, I don't use analog stuff but on my trunks it works |
| 07:08.32 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 07:08.40 | simon_ca | cypromis: Thanks! that fixed it. |
| 07:08.45 | cypromis | np |
| 07:08.46 | wasim_ | poor aussie tail didn't wag |
| 07:09.14 | shaZwaz | wasim score ? |
| 07:09.34 | wasim | 230 all out |
| 07:09.58 | shaZwaz | looks like the bowlers clicked |
| 07:10.11 | shaZwaz | at last |
| 07:12.59 | *** join/#asterisk Deggy (~Deggy@217.14.136.2) |
| 07:13.25 | datareactor | i get this error SIP Status: 405 Method Not Allowed |
| 07:15.40 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 07:15.41 | *** mode/#asterisk [+o kram] by ChanServ |
| 07:16.11 | kram | greetings |
| 07:16.34 | wasim | happy basant astmaster |
| 07:16.56 | wasim | basant is the spring festival in indo-pak |
| 07:17.06 | wasim | lahore is celebrating it this weekend |
| 07:17.20 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:18.59 | shaZwaz | wasim you in lhr ? |
| 07:20.54 | wasim | shaZwaz: affirmative |
| 07:21.14 | shaZwaz | no going back to isloo ? |
| 07:22.28 | wasim | shaZwaz: not unless the wife kicks us out |
| 07:22.35 | shaZwaz | ha ha |
| 07:29.16 | *** join/#asterisk shaZwaz (~adnans@203.81.196.167) |
| 07:35.16 | *** join/#asterisk r0d3nt|m (~RatMan@4.19.77.194) |
| 07:40.46 | wasim | stupid paki openers |
| 07:42.37 | wasim | 2/7 238 to win ... |
| 07:44.35 | shaZwaz | here we go again |
| 07:45.23 | wasim | 3/9 |
| 07:46.06 | wasim | lee's 150 km/h yorker to clean bowl joe |
| 07:46.08 | shaZwaz | eew |
| 07:47.14 | wasim | only inzi can same us now, and maybe razzaq/afridi (but only we get to the end) |
| 07:48.22 | shaZwaz | damn.. |
| 07:52.02 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 07:57.53 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 08:20.06 | Makenshi | morning |
| 08:20.15 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:20.22 | *** join/#asterisk Hobbit (~kdyson@humdrum.messagelabs.net) |
| 08:20.37 | *** part/#asterisk Hobbit (~kdyson@humdrum.messagelabs.net) |
| 08:23.06 | *** join/#asterisk Wireless (~bad@220.233.77.87) |
| 08:26.44 | *** join/#asterisk peted20 (~pete@d3-207.rb.gh.centurytel.net) |
| 08:27.21 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 08:28.21 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:34.23 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 08:35.48 | Zeeek | who has Siemens Gigaset phones connected to asterisk |
| 08:35.52 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 08:35.57 | mAsH` | mornign all |
| 08:36.03 | Zeeek | mornin |
| 08:48.56 | *** join/#asterisk schurig (~schurig@p5080BCB9.dip0.t-ipconnect.de) |
| 08:49.17 | *** join/#asterisk meppl (~mephisto@pD9542384.dip.t-dialin.net) |
| 08:49.22 | meppl | guten morgen |
| 08:58.13 | *** join/#asterisk Kraven (kraven@217.115.141.163) |
| 09:01.24 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:02.05 | christo | can anybody please point me to some examples for setting out an IVR mapping in * ? |
| 09:02.44 | oej | christo: read the extensions.conf example in your asterisk installation |
| 09:03.37 | christo | oej - ok thanks |
| 09:03.45 | christo | I also found this http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+ivr+menu |
| 09:04.42 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:05.24 | MrEntropy | playing an IVR i get "g729_read: Short read (10) (No such file or directory)!" dumped to the CLI, anyone got any idea what this means? the file plays, but it's only supposed to play one, and plays twice for some reason. |
| 09:15.36 | pashah | morning |
| 09:17.58 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:18.46 | MrEntropy | ok, i figured out the short read |
| 09:19.58 | MrEntropy | but why does the sound file play twice? |
| 09:21.17 | MrEntropy | ~pastebin |
| 09:21.18 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 09:22.11 | christo | How do I record a file in GSM format for my IVR prompts? Or is there an easy way to convert from wav? |
| 09:22.28 | MrEntropy | christo: google 'sox' |
| 09:22.41 | christo | ta |
| 09:23.00 | MrEntropy | http://pastebin.ca/5266 <-- a call arrives on this context, and tt-weasles plays twice, anyone know why? |
| 09:25.39 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 09:27.26 | MrEntropy | anyone? this is really boggling |
| 09:28.08 | *** join/#asterisk Tornad (~Tornad@81.255.65.249) |
| 09:28.19 | WilliamK | strange |
| 09:28.43 | Tornad | hi |
| 09:29.18 | Tornad | I'm looking for a good ratio for price / quality for SIP hardphone. |
| 09:29.29 | Tornad | It seems that grandstream are the best ? |
| 09:30.16 | MrEntropy | any ideas on stuff i can try |
| 09:30.18 | WilliamK | you want something that's business or consumer grade? |
| 09:31.16 | Tornad | In the middle :o) |
| 09:31.29 | WilliamK | probably a Sipura then |
| 09:31.41 | Tornad | what's business grandstream problem ? |
| 09:31.49 | Tornad | quality or functionality ? |
| 09:32.32 | WilliamK | I've always known grandstreams to be almost at the bottom of the line (consumer), someone else though may disagree or agree with me |
| 09:32.38 | WilliamK | Sipura's work well though |
| 09:32.58 | clive- | william how do they compare to the pa168 phones? |
| 09:33.27 | MrEntropy | clive-: a lot better imho =) |
| 09:33.46 | WilliamK | no idea, Moc would be able to answer better probably on that |
| 09:33.59 | Tornad | William : Which sipura model you like ? |
| 09:34.16 | WilliamK | I'm using the 2000 2 port FXS right now |
| 09:35.11 | WilliamK | very user friendly, but has alot of advanced features and someone with barely any skills on voip can set one up |
| 09:36.02 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:36.07 | clive- | Mrentropy no, not pa186 to sipura, ...thats no competition...I mean pa186 to grandstream |
| 09:36.37 | MrEntropy | clive-: that's what i meant, i got 1 grandstream, it's a lot more 'solid' than the pa* |
| 09:38.25 | clive- | lol..ok, |
| 09:38.37 | MrEntropy | clive-: i'm a cisco fan though and i'd get one if they weren't so damn expensive |
| 09:38.43 | clive- | do you disable the auto update firmware option? |
| 09:39.01 | MrEntropy | clive-: i'm not aware of that option |
| 09:39.18 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 09:39.20 | WilliamK | MrEntropy, if you're a cisco fan you'd love to know that Cisco has a contract with Sipura for OEM stuff |
| 09:39.28 | WilliamK | =) |
| 09:39.45 | SeaForth | how can someone be a 'fan' of Cisco? |
| 09:39.59 | Tornad | thanks guy, I think I will try sipura SPA 841 and grandstream b102. |
| 09:39.59 | MrEntropy | they make solid gear, why not? |
| 09:40.13 | WilliamK | cisco is good on some things, but they've been getting more rude and rude lately |
| 09:40.13 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 09:40.53 | MrEntropy | WilliamK: which way do you mean, cisco provides sipura with hardware or the other way around? |
| 09:40.54 | WilliamK | and also, Cisco Call Mgr is a P.O.S. package |
| 09:40.58 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 09:40.58 | clive- | cisco does make great stuff...just way expensive |
| 09:41.05 | WilliamK | sipura provides hardware to cisco |
| 09:41.06 | MrEntropy | clive-: amen |
| 09:41.40 | clive- | actually , astersik can do IVR better than cisco |
| 09:41.44 | MrEntropy | i got an MC3810 on my desk i still haven't touched. it looks neat-o |
| 09:42.02 | WilliamK | clive, Asterisk doesn't take up 5 boxes to load basic features either |
| 09:42.13 | WilliamK | and I guarantee Asterisk can handle calls 500% better |
| 09:42.38 | Wireless | can anyone suggest a site for resolving asterisk audio issues? |
| 09:42.49 | Wireless | it seems that my * box cannot encode audio |
| 09:42.57 | clive- | william, true, but cisco handles voip better than asterisk, aspecially for us guys in africa who cant use g711 |
| 09:42.58 | SeaForth | http://www.cisco.com/warp/public/cc/pd/rt/mc3810/index.shtml |
| 09:43.24 | *** join/#asterisk los415 (~los415@ssf-office.corp.race.com) |
| 09:43.29 | WilliamK | clive, why can't you use g729a? |
| 09:43.37 | SeaForth | did you get it cheapo MrEntropy |
| 09:44.02 | MrEntropy | SeaForth: I did, with a codec dsp card an an E1 module |
| 09:44.41 | MrEntropy | "and an" |
| 09:44.52 | WilliamK | MC3810s are usually cheap fully loaded, not sure about the v3s but the older models were crapola |
| 09:45.18 | MrEntropy | i'm looking for a 12.3 IOS for it though, 12.2 has some SIP bugs =/ |
| 09:45.55 | MrEntropy | i haven't experienced them myself, but that's what i've read |
| 09:46.35 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:47.15 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 09:47.49 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 09:48.17 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-82-48.sentechsa.net) |
| 09:48.42 | kks | if i have existing channel h323, can i still able to use oh323? |
| 09:49.17 | MrEntropy | so anyway, does anyone have ANY suggestions about my above problem? I'd really appreciate any ideas you can throw at me |
| 09:49.46 | clive- | sorry, I missed your post there |
| 09:49.59 | MrEntropy | http://pastebin.ca/5266 <-- a call arrives on this context, and tt-weasles plays twice, anyone know why? |
| 09:50.49 | WilliamK | clive, why can't you use g729a? |
| 09:52.00 | clive- | william , are you referring to voip cisco vs asterisk...well, asterisk doesnt do good jitter buffering, and vad |
| 09:52.06 | clive- | for a start |
| 09:52.58 | clive- | 729 is fine, but the jitter buffering is important to us |
| 09:53.09 | WilliamK | reason for my curiosity is I've never ran into a codec handling issue with Asterisk yet |
| 09:53.13 | clive- | It will be in asterisk I am sure, but at the moment its not good |
| 09:53.17 | Delvar | morning all |
| 09:53.30 | RoyK | mrnng ll |
| 09:53.32 | christo | morning |
| 09:53.36 | WilliamK | I've used g729 and 711 constantly for the last 3-6 months now |
| 09:53.37 | oej | MrEntropy: _. is a very dangerous wildcard. Asterisk will match this also for the "h" extension, so you will get the first round on the extension, then a second round on hangup. Try _X. instead |
| 09:53.49 | Delvar | RoyK: dont like vowels? |
| 09:53.57 | RoyK | nt rll |
| 09:54.00 | Delvar | lol |
| 09:54.06 | MrEntropy | oej: but that won't match an 8 digit number? |
| 09:54.19 | MrEntropy | oej: and then there are other lenght ones |
| 09:54.29 | RoyK | MrEntropy: _XXXXXXXX will match eight digits |
| 09:54.31 | RoyK | :P |
| 09:54.41 | oej | MrEntropy: _X. would match any number - the dot just say "any number of something, at least one" |
| 09:54.58 | MrEntropy | oh...good idea |
| 09:55.01 | MrEntropy | ok |
| 09:55.03 | oej | MrEntropy: You do not want to match extensions like "h", "i" etc |
| 09:56.13 | MrEntropy | oej: haha, ok...that fixed it, thanks...i never thought it would cycle like that |
| 09:56.40 | oej | MrEntropy: No problem, just remember that _. matching is dangerous! :-) |
| 09:56.56 | Delvar | oej: i didnt know that either, thanks |
| 09:59.52 | MrEntropy | oh and one more thing, i was getting a short read error a while ago, i fixed it with a hack and reading the asterisk source. when i would play my IVR encoded in g729 it would spit out "g729_read: Short read (10) (No such file or directory)!" so in the source code it had the rule "if ((res = read(s->fd, s->g729, 20)) != 20)" my IVR was 1832 bytes, when I padded it using a hex editor with 0's to 1840 (an equal div of 20). the err |
| 09:59.52 | MrEntropy | ors stopped. but isn't it presumptuous on asterisks behalf to assume that every file is a multiple of 20? is this a bug? |
| 10:00.35 | oej | That's the default sound format of Asterisk 20ms - you need to reconfigure the phones... |
| 10:01.16 | MrEntropy | what do you mean 20ms? 20 bytes, it reads 20 bytes at a time and spits an error if it's not |
| 10:03.06 | MrEntropy | which just looks weird. i code C myself, and i never expect a file to be a multiple of anything when read()ing. if read() doesn't return however many bytes you wanted to read all that means is you're at EOF, nothings 'wrong'... |
| 10:03.54 | pashah | what could this be: rtp.c:452 ast_rtp_read: RTP: Received packet with bad UDP checksum |
| 10:04.35 | MrEntropy | anyone got an idea why they'd do that? |
| 10:05.18 | MrEntropy | is it a written standard that g729 files are to be in multiples of 20 bytes? if so, i'm content. |
| 10:15.42 | *** join/#asterisk adnans (~adnans@noterik2.demon.nl) |
| 10:21.01 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 10:24.33 | mAsH` | how i debug zap channel? |
| 10:24.34 | mAsH` | :/ |
| 10:29.19 | *** join/#asterisk yogurt2ungue (~yogurt2un@host71.201-252-158.telecom.net.ar) |
| 10:33.11 | *** join/#asterisk Leeky (Leeky@dustpuppy.keme.net) |
| 10:33.45 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 10:34.01 | heka | Hello, Does the Sipura 3000 support Early Dial? |
| 10:36.23 | *** join/#asterisk pranav (dawda_pran@203.115.89.185) |
| 10:37.30 | *** join/#asterisk slamm (~post@213.184.212.218) |
| 10:37.40 | pranav | does anyone know how to configure the astguiclient? |
| 10:41.31 | slamm | I have finally gotten CLI dial to work. It it possible to ute a normal extension setup as if the number I call was the party calling in? I have tried to create an context [default] s,1,Dial(Zap/g2/360) s,2,Playback(testsound) s,3,Hangup , but this just makes the call.. it dows not playback the file... Any idea? |
| 10:41.40 | slamm | ute = use |
| 10:44.16 | pranav | hello slam do you have anyidea about astguiclient |
| 10:45.02 | slamm | nope.. sorry.. |
| 10:45.47 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 10:48.24 | pranav | slam: do you know someone who knows about the astguicloient |
| 10:48.35 | pranav | sorry astguiclient |
| 10:52.18 | slamm | don't know.. |
| 10:55.25 | pranav | hello anyone there |
| 10:56.24 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 10:59.44 | *** join/#asterisk Martohtar (Martohtar@194.19.32.100) |
| 11:13.41 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-83-135.sentechsa.net) |
| 11:25.18 | mAsH` | i'm connecting * to a PABX |
| 11:25.51 | mAsH` | anyone can help me? |
| 11:33.36 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 11:34.18 | shadebob | Asterisk can manage s0 cards? |
| 11:34.37 | tih | ~docs |
| 11:34.38 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 11:37.47 | Makenshi | those of you running a large number of ip phones, do you bother to put them into a separate vlan? |
| 11:44.06 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-80-54.sentechsa.net) |
| 11:50.05 | pashah | Makenshi: I separeted them with physical switch =) |
| 11:50.44 | Makenshi | pashah, presumably you only have one building then :> |
| 11:51.51 | pashah | Makenshi: right. but all the rest phones, in other locations are not separeted. why would you want to make a vlan for them? |
| 11:59.38 | *** join/#asterisk njafo (~njafo@user155.bwa.etnet.fo) |
| 12:02.03 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-80-54.sentechsa.net) |
| 12:08.00 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 12:10.04 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:17.03 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 12:18.07 | Wireless | hi. my * will not receive audio from registered clients. can anyone help? |
| 12:19.07 | RaYmAn-Bx | is your server behind nat? |
| 12:19.19 | Wireless | yes, but that's the issue. |
| 12:19.28 | Wireless | i set up two local extensions and it still plays up. |
| 12:19.38 | Wireless | but that's NOt the issue i meant |
| 12:20.01 | MrEntropy | can i add/replace the 'searchpath' that asterisk tooks for IVRs in when you Playback()? |
| 12:20.16 | MrEntropy | s/tooks/looks/ |
| 12:20.31 | *** join/#asterisk jackflash (~jf@cpc2-rdng8-4-0-cust187.winn.cable.ntl.com) |
| 12:20.38 | Wireless | it can make and receive calls to an external (beyond the firewall) IAX server, but will not send audio from the local useragent, but will receive incoming audio from external IAX server alright |
| 12:20.51 | Wireless | if a local user calls a local user, nothing will work. |
| 12:21.40 | Wireless | after from that external server, everything else is running on SIP |
| 12:22.06 | RaYmAn-Bx | have you tried with an IAX client? |
| 12:22.08 | Wireless | apart from, i mean |
| 12:22.37 | Wireless | what exactly do you suggest i do with that? |
| 12:22.59 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 12:23.49 | RaYmAn-Bx | have you set your asterisk up for nat? I.e. nat=yes as well as externip and such? |
| 12:23.54 | Wireless | one question: do you need a soundcard in the box to make it work? |
| 12:24.20 | Wireless | nat=yes for external (ie beyond firewall connections), nat=no for local devices. |
| 12:24.44 | Wireless | i did set externalip= my public IP... hmm... maybe that's what's confusing my local clients. |
| 12:25.49 | RaYmAn-Bx | no, you don't need a soundcard..Have you setup localnet as well? |
| 12:25.58 | Wireless | that's it |
| 12:26.01 | Wireless | that's it!!! |
| 12:26.04 | Wireless | fixed it!!! |
| 12:26.20 | Wireless | it's because i set my externalip=210.blah that made it break. |
| 12:26.43 | Wireless | is that because * was trying to annouce itself as 210.xxx to the local clients? |
| 12:26.43 | RaYmAn-Bx | I doubt that...well..partially..you prolly broke external clients now ;) |
| 12:26.52 | Wireless | yes, my concerns too. |
| 12:26.56 | Wireless | let me try. |
| 12:27.07 | RaYmAn-Bx | if you setup localnet it should be able to tell whether it's a local or a remote (natted) client |
| 12:27.17 | Wireless | sorry, but what's localnet? |
| 12:27.44 | RaYmAn-Bx | my localnet : localnet=192.168.0.0/255.255.255.0 |
| 12:27.55 | Wireless | ah... |
| 12:27.57 | RaYmAn-Bx | i.e. a definition of what the local non-nat network is |
| 12:28.20 | Wireless | strangely enough, my external connections are not broken... |
| 12:29.06 | Wireless | but that's when i'm try to place a call using my sip client via asterisk to an external sip server onwards to my cell phone. |
| 12:29.51 | Wireless | i'm not sure if it'll work if a sip client beyond the firewall tries to register to me. |
| 12:30.02 | Wireless | wait... toilet trip. |
| 12:31.02 | Wireless | back. |
| 12:31.45 | Wireless | is anyone interested in logging into my box to help me test it? |
| 12:31.47 | Wireless | SIP |
| 12:38.13 | Makenshi | pashah, various reasons - security, performance |
| 12:38.40 | Wireless | anyone want a sip account on my box? |
| 12:46.44 | *** join/#asterisk visik7 (~ciao@host149-36.pool80182.interbusiness.it) |
| 12:50.27 | heka | Hello, Does the Sipura 3000 support Early Dial? |
| 12:54.14 | *** join/#asterisk zoa (~zoa@ip-212-239-162-34.dsl.scarlet.be) |
| 12:57.33 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 12:58.40 | jackflash | Microsoft will never live up to HURD. Everybody knows it takes many Longhorns to make one HURD. |
| 13:01.10 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 13:02.40 | *** join/#asterisk tomtom- (~tomtom@dD5761F16.access.telenet.be) |
| 13:02.46 | tomtom- | hi |
| 13:07.31 | *** join/#asterisk meppl (~mephisto@pD9542384.dip.t-dialin.net) |
| 13:12.13 | *** join/#asterisk hmodes (hmodes@pcp0010853935pcs.potshe01.pa.comcast.net) |
| 13:13.40 | *** join/#asterisk ltostain (~ltostain@i01v-30-28.d4.club-internet.fr) |
| 13:14.04 | ltostain | Hi all |
| 13:14.23 | tomtom- | hi |
| 13:14.32 | ltostain | Nice day :) |
| 13:16.01 | RoyK | bad day |
| 13:17.33 | ltostain | Ooo |
| 13:21.41 | slamm | I have finally gotten CLI dial to work. It it possible to ute a normal extension setup as if the number I call was the party calling in? I have tried to create an context [default] s,1,Dial(Zap/g2/360) s,2,Playback(testsound) s,3,Hangup , but this just makes the call.. it dows not playback the file... Any idea? |
| 13:22.49 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 13:25.59 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:26.39 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:28.52 | *** join/#asterisk lung (~lung@24-148-96-186.ip.mhcable.com) |
| 13:30.09 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 13:32.16 | jerlique | hi - do facilities like hold and transfer rely on the underlying protocol eg sip,iax etc or the capabilities of the pbx? |
| 13:37.49 | *** join/#asterisk jointe79 (~huot@212.247.174.226) |
| 13:38.13 | Delvar | jerlique: its the pbx |
| 13:38.25 | jointe79 | First I freely admit that while I can figure out most of what is happening |
| 13:38.25 | jointe79 | in the .conf files I still don't fully understand how to set up something |
| 13:38.25 | jointe79 | new. |
| 13:38.25 | jointe79 | I am trying to use SIPp to do some testing of stuff with asterisk but I am |
| 13:38.25 | jointe79 | not sure how to set up asterisk and especailly the .conf files to do this. |
| 13:38.26 | jointe79 | I saw some information on the wiki but did not see how to set up the |
| 13:38.28 | jointe79 | sip.conf and extensions.conf. Can anyone give me any ideas of how to |
| 13:38.30 | jointe79 | procceed please? |
| 13:39.17 | jerlique | delvar: thanks. so does that mean using any channel bank that is sip compatible will be fine with asterisk? |
| 13:40.20 | mAsH` | i'm connecting * to a PABX |
| 13:40.22 | mAsH` | anyone can help me? |
| 13:40.42 | *** join/#asterisk D1ng0 (~dingo@3.217.8.67.cfl.rr.com) |
| 13:41.21 | vaewyn | ok... who wants to bid on replacing a Meridian Option 11e with * ;} |
| 13:42.49 | *** join/#asterisk jointe12 (~huot@212.247.174.226) |
| 13:43.18 | jointe12 | sorry my computer hang up so i ask again |
| 13:43.22 | jointe12 | First I freely admit that while I can figure out most of what is happening |
| 13:43.22 | jointe12 | in the .conf files I still don't fully understand how to set up something |
| 13:43.22 | jointe12 | new. |
| 13:43.22 | jointe12 | I am trying to use SIPp to do some testing of stuff with asterisk but I am |
| 13:43.22 | jointe12 | not sure how to set up asterisk and especailly the .conf files to do this. |
| 13:43.23 | jointe12 | I saw some information on the wiki but did not see how to set up the |
| 13:43.25 | jointe12 | sip.conf and extensions.conf. Can anyone give me any ideas of how to |
| 13:43.27 | jointe12 | procceed please? |
| 13:45.22 | vaewyn | jointe12: http://www.voip-info.org/wiki-Asterisk+config+sip.conf and http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 13:45.37 | wankel | wow. way to paste your email into the channel. |
| 13:45.43 | vaewyn | plus use the sample configs that some with the source |
| 13:45.51 | vaewyn | s/some/come/ |
| 13:46.00 | RoyK | ~lart jointe12 |
| 13:54.42 | bjohnson | mAsH`: you're going to have to give WAY more info |
| 13:55.15 | bjohnson | mAsH`: did you read the wiki pages abouting connecting to a pbx? some specfiic examples about Nortel and Panasonic |
| 13:55.39 | RaYmAn-Bx | bjohnson: actually, as far as I can see, in dvorak keymap l and r are right next to eachother :P |
| 13:55.47 | bjohnson | ahhhh |
| 13:56.07 | RaYmAn-Bx | roughly o & p places (according to a quick google) |
| 13:56.41 | vaewyn | bjohnson: it's that engrish thing :P |
| 13:56.50 | slamm | has anyone tried the new atxfer feature? if yes, is it working? |
| 13:57.39 | mAsH` | bjohnson yes |
| 13:57.58 | mAsH` | i follewed the examples about panasonic |
| 13:58.06 | mAsH` | but it doesn't work |
| 13:58.19 | bjohnson | vaewyn: outline the replacement details and post to bus mailing list |
| 13:58.21 | mAsH` | the link with the PABX go up |
| 13:58.52 | vaewyn | bjohnson: yeah... am going to... :} |
| 13:59.01 | bjohnson | vaewyn: don't forget to say what city it is in |
| 13:59.51 | vaewyn | city? :} village baby... village :} |
| 14:00.21 | bjohnson | vaewyn: why aren't you doing it? |
| 14:01.14 | vaewyn | bjohnson: manpower... 3 telco jockies and 2 programmers ain't enough to get this done in a timely fashion |
| 14:01.58 | vaewyn | (by telco jockies I mean wire pullers... they arn't good for much else) |
| 14:05.18 | bjohnson | so you need a team with more than that? You should put that in the email too |
| 14:05.58 | fafnir | okay i saw the last few lines.... and vaewyn, i jus thave to know, what are you doing? |
| 14:07.06 | vaewyn | fafnir: looking at replacing a Meridian Option 11e with * boxes |
| 14:08.24 | vaewyn | Funny thing... I think the * will cost less than upgrading the 11e from 2.4 to 4.x |
| 14:08.37 | vaewyn | even with all brand new hardware |
| 14:08.54 | tzanger | vaewyn: yup |
| 14:09.02 | tzanger | Norstar has "bend over and take it" pricing for everything |
| 14:09.27 | fafnir | wont it be more flexible as well, and easier to upgrade in the future? |
| 14:09.28 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 14:09.34 | vaewyn | yes and yes |
| 14:09.44 | vaewyn | and we will FINALLY be able to backup voicemail |
| 14:09.49 | vaewyn | :} |
| 14:09.49 | tzanger | fafnir: the only thing is that I've not met a sip phone I like as much as teh norstar phone |
| 14:09.57 | fafnir | wow, with all these compelling options, why havnt you switched to gieco already? |
| 14:10.24 | tzanger | hahah |
| 14:10.37 | bjohnson | nortel handsets are noce |
| 14:10.38 | bjohnson | nice |
| 14:10.39 | fafnir | tzafrir: shoudlnt there be a way to use your existing phones with an asterisk setup? |
| 14:10.42 | _Brian | tzanger: do you have a url for the phone you are referring to? |
| 14:10.45 | tzanger | yes you can |
| 14:10.50 | vaewyn | tzanger: Heh... well the Polycom IP600s are nice enough for me :P |
| 14:10.50 | bjohnson | too bad they don't work with anything else |
| 14:10.53 | tzanger | I am doing it but it's not all that great yet |
| 14:11.06 | _Brian | I like my ip500 :) who needs a microbrowser anyway :) |
| 14:11.09 | tzanger | you can get an ATA module and plug that into an FXS port |
| 14:11.13 | vaewyn | you can't use the nortel digitals with * |
| 14:11.16 | fafnir | i need money |
| 14:11.17 | tzanger | or an 8-port ATA rack module and 8 FXS ports |
| 14:11.20 | vaewyn | which is most of what we have |
| 14:11.25 | tzanger | that will let * have "extensions" inside the MICS |
| 14:11.36 | tzanger | and I already have a PRI between * and the norstar |
| 14:11.37 | vaewyn | screw ata... PRI it |
| 14:11.40 | vaewyn | :} |
| 14:11.44 | tzanger | unfortunately to do anything really nice you need the SL1 protocol |
| 14:11.45 | bjohnson | tzanger: not the best solution |
| 14:12.04 | tzanger | because you can do Public, Tie or FX trunks |
| 14:12.09 | bjohnson | tzanger: only suitable in a migration situation .. I wouldn't do a new one that way |
| 14:12.13 | tzanger | but Public and Tie don't seem to let you have DNs on the PRI |
| 14:12.19 | tzanger | and FX doesn't let you dial enough digits to be useful :-) |
| 14:12.41 | tzanger | if I can reverse engineer SL1 (in the works) I can do remote voicemail, the whole shebang |
| 14:12.50 | Grooby | really wierd |
| 14:12.51 | tzanger | and if I can reverse engineer the optical port protocol I can do even more |
| 14:12.54 | Grooby | i do sip show channels |
| 14:13.07 | Grooby | i get 2 channels that i know no one is using |
| 14:13.21 | Grooby | then sip show channels again |
| 14:13.23 | Grooby | and they are gone |
| 14:14.40 | vaewyn | If ISDN ports for * weren't so @#$@#$ expensive I would love it if someone reverse engineered the nortel digital phones |
| 14:14.52 | vaewyn | but as is we might as well just replace them |
| 14:18.36 | bjohnson | what is the goal of this job? more handsets? cheaper ld? integration of offices? |
| 14:19.06 | vaewyn | replacing a $$$hole from hell called a 17 year old Option11e |
| 14:19.37 | fafnir | if anyone wants to hire marginally skilled manual labor person that learns quickly with hands on experience, message me |
| 14:19.40 | fafnir | please? |
| 14:19.55 | Grooby | lots of vtech phones? |
| 14:20.18 | tzanger | vaewyn: ISDN isn't all that expensive on * -- it's bloody expensive on Norstar |
| 14:20.41 | tzanger | you need a DTI and either a clocking/services module or a 6 port fiber/services combo card |
| 14:20.44 | tzanger | that gets you CT1 |
| 14:20.48 | tzanger | to do PRI you need a software enabler key |
| 14:20.58 | tzanger | and to do MCDN (SL1) you need that and another key |
| 14:21.07 | vaewyn | tzanger: we've already got PRI up the wazooo |
| 14:21.12 | tzanger | about $3000 to get it on MICS and that's using USED equipment |
| 14:23.25 | vaewyn | running ISDN ports on * is expensive because it takes a lot of machines... they don't have a good dense ISDN channel bank yet (at least that I have seen) |
| 14:23.39 | tomtom- | anyone here experience with Junghanns quadBRI cards and faxing? |
| 14:24.34 | tomtom- | fax layout is completely messed up even though we have zaprtc running for synchronization |
| 14:25.36 | florz | tomtom-: You probably better don't use zaprtc. |
| 14:25.53 | tomtom- | florz, if i use the kernel's rtc it's the same |
| 14:26.06 | bjohnson | vaewyn: arbitrary replacement of a working system? |
| 14:26.19 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 14:26.23 | florz | tomtom-: Erm, you have a quadBRI, right? |
| 14:26.28 | tomtom- | yes |
| 14:27.02 | florz | tomtom-: Then the only thing you achieve by using any "auxilary timing driver" is that you _de_sychnronize things. |
| 14:27.38 | tomtom- | florz, so you would recommend not using zaprtc _and_ not using the kernel's rtc either? |
| 14:27.55 | florz | tomtom-: exactly. Neither of them is likely to be in sync with the PSTN. |
| 14:28.20 | vaewyn | bjohnson: would be arbitrary... but they have EOL'd 75% of the existing structure... and we are looking at 1.5-2million to remedy that... and they have pretty much told us we will be spending 5-7million in 4-5 years to upgrade frames cause they will be EOL as well |
| 14:28.24 | tomtom- | florz, and what would be the recommended way to get them properly in sync? |
| 14:29.01 | vaewyn | bjohnson: for a 4800 student university that is a big $$$$ amount to swallow... let alone twice |
| 14:29.19 | blitzrage | damnit... cygwin's site is down |
| 14:29.32 | tomtom- | florz, even in that configuration the faxes are not properly rendered |
| 14:29.46 | *** join/#asterisk coolschooluk (~coolschoo@server1.pointnet.co.uk) |
| 14:29.49 | tzanger | blitzrage: get to school |
| 14:29.50 | florz | tomtom-: I don't have any experience with the quadBRI cards, but I guess that in some way you can chose some TE port to provide the zap timing. |
| 14:29.58 | tzanger | and make me a pie while you're at it |
| 14:30.04 | florz | tomtom-: TE ports synchronize to the PSTN. |
| 14:30.22 | blitzrage | tzanger: only got school on Tuesdays :) |
| 14:30.30 | tzanger | blitzrage: then why is my pie not ready yet |
| 14:30.31 | blitzrage | tzanger: you make me a pie first |
| 14:30.38 | florz | tomtom-: using * for fax passthrough or using spandsp? |
| 14:30.39 | tzanger | don't make me come over there and beat you |
| 14:30.49 | blitzrage | tzanger: I can make brownies... thats all I've got for ingredients |
| 14:30.55 | tzanger | I'll give you FAQ questions in a way you did not anticipate them to be delivered |
| 14:31.00 | bjohnson | I know a university student with linux background and some voip experience if nedding another "hand" .. couldn't run the show though |
| 14:31.03 | blitzrage | tzanger: lol |
| 14:31.12 | tzanger | hmm... in univeristy and makes brownies... count me in! |
| 14:31.23 | tzanger | bjohnson: eh? |
| 14:31.31 | bjohnson | for vaewyn |
| 14:31.33 | blitzrage | tzanger: now... just need to find some people to give me work and pay me |
| 14:31.39 | tzanger | hehe |
| 14:31.42 | blitzrage | tzanger: then I can afford multiple brownies |
| 14:31.45 | vaewyn | bjohnson: :} |
| 14:31.49 | bjohnson | not me |
| 14:31.56 | tzanger | well if you got cheaper hash you'd be able to eat more brownies |
| 14:31.57 | tomtom- | florz, with spandsp |
| 14:32.06 | *** join/#asterisk sjaak538 (~sjaaknabu@bmr-d8e8.mxs.adsl.euronet.nl) |
| 14:32.10 | florz | tomtom-: Sure that your libtiff is ok? |
| 14:32.26 | tomtom- | yes, libtiff is ok |
| 14:32.32 | tomtom- | and we are using TE ports |
| 14:32.40 | jluk | OT: anyone using plesk with FC2 ? |
| 14:32.48 | florz | tomtom-: How you know that libtiff is OK? |
| 14:32.57 | tzanger | wtf... smokers families' suing tobacco... as if |
| 14:33.03 | tzanger | the U.S. legal system is completely fucked up |
| 14:33.59 | tomtom- | florz, because before it was fscked up, and now it seems to render stuff properly |
| 14:34.28 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
| 14:34.29 | tomtom- | would it be useful if i put a sample .tif online somewhere? |
| 14:34.42 | florz | tomtom-: maybe |
| 14:34.50 | tomtom- | we're using 3.5.7-7 |
| 14:34.55 | pbxjunkie | I got AVM C4 active controller but I still get echo.. what do I do you guys?:) |
| 14:34.58 | tomtom- | let me just quickly throw one online |
| 14:35.28 | *** join/#asterisk sabre (sabre@69.149.209.83) |
| 14:38.23 | florz | tomtom-: Uh, that looks really weird :-) |
| 14:38.44 | florz | tomtom-: I don't think that's a timing issue |
| 14:38.50 | bjohnson | tzanger: I'm thinking of suing Hersheys because I ate a lot of chocolate and got a fat ass .. they should put warnings on their packages |
| 14:39.05 | tomtom- | no? sometimes it shows a third of the actual page |
| 14:39.05 | florz | tomtom-: what's the output of spandsp while receiving the fax? |
| 14:39.11 | tomtom- | sec |
| 14:39.27 | florz | tomtom-: Well, then again. |
| 14:39.37 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 14:39.53 | vaewyn | well.. that would be an option if choclate was actively addictive... |
| 14:40.09 | florz | tomtom-: You said you had TE ports only? |
| 14:40.11 | tzanger | bjohnson: heh |
| 14:40.11 | sjaak538 | Anybody knows a little * with ISDN |
| 14:40.29 | tomtom- | florz, they're switchable with jumpers |
| 14:40.38 | florz | tomtom-: Yep, but using them as TE only? |
| 14:40.46 | tzanger | sjaak538: yup what's up |
| 14:40.56 | tomtom- | florz, yes i think so |
| 14:41.00 | sjaak538 | I use it as TE as well |
| 14:41.03 | tomtom- | all ports are set to TE |
| 14:41.10 | tomtom- | except for one, but i'm not using that one |
| 14:41.13 | sjaak538 | I've configured everything well (i guess) |
| 14:41.22 | sjaak538 | but ISDN seems to bee dead |
| 14:41.26 | tzanger | hmm this is ISDN BRI then |
| 14:41.31 | sjaak538 | I'ts remote |
| 14:41.33 | tzanger | I don't know anything about chan_capi or anything |
| 14:41.39 | sjaak538 | hfc |
| 14:41.46 | *** join/#asterisk mflorell (~mattf@rrcs-24-173-158-34.se.biz.rr.com) |
| 14:42.10 | tzanger | I know PRI fairly well through the zap interfaces |
| 14:42.19 | sjaak538 | ISDN Bri zap okay |
| 14:42.43 | tomtom- | sjaak538, r u using quadBRI cards? |
| 14:42.43 | sjaak538 | I'm not sure if my provider pluged in the wire |
| 14:42.52 | sjaak538 | single |
| 14:43.00 | sjaak538 | how to test this |
| 14:43.29 | florz | sjaak538: Is the NT1's LED lit? |
| 14:44.03 | *** join/#asterisk ToyMan (~konversat@204.8.82.238) |
| 14:44.03 | sjaak538 | it's remote |
| 14:44.27 | sjaak538 | 200 Km away from my home |
| 14:44.37 | tomtom- | sjaak538, do you have qozap.o in your modules? |
| 14:44.38 | *** join/#asterisk tangel (tangel@66.232.52.252) |
| 14:44.51 | tangel | can one do VAR3 => ${VAR1}&${VAR2} ? |
| 14:45.10 | sjaak538 | zap show channel shows onhook |
| 14:45.24 | tangel | when i try it, i see it trying to Dial(SIP/1001&&SIP/1002) |
| 14:45.30 | tangel | i think the && is messing it up |
| 14:45.35 | *** part/#asterisk coolschooluk (~coolschoo@server1.pointnet.co.uk) |
| 14:45.50 | tangel | oh.. should it be: VAR3 => ${VAR1}${VAR2} ? |
| 14:46.07 | sjaak538 | tomtom no qozap.0 |
| 14:46.10 | tangel | no & needed since your not passing it to a function/method |
| 14:46.13 | florz | sjaak538: IC. Is it PtP or PtMP? |
| 14:46.35 | florz | sjaak538: It's a HFC-S PCI A, isn't it? |
| 14:46.40 | tangel | also, is there free software that can use the h.261 video through asterisk stuff? |
| 14:46.42 | sjaak538 | ?? what does this means |
| 14:46.49 | sjaak538 | Yes florz |
| 14:46.55 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 14:47.19 | sjaak538 | HFC PCI |
| 14:47.27 | sjaak538 | lspci show okay |
| 14:47.36 | sjaak538 | ztcfg shows okay |
| 14:47.48 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc1aq.dialup.mindspring.com) |
| 14:47.52 | *** join/#asterisk randu (~randu@dsl-pppoe-pool1-37.d02.ncshlt.infoave.net) |
| 14:48.40 | florz | sjaak538: PtP == Point-to-Point, PtMP == Point-to-Multi-Point. "Anlagenanschluss" and "Mehrgeraeteanschluss", respectively, in German, if that helps you =:-) |
| 14:49.18 | `Sauron | Ick. |
| 14:49.34 | `Sauron | Why can't the Germans just adopt the real (english) names for it? :) |
| 14:50.34 | tomtom- | sjaak538, you should have the qozap.o if you want to use a quidbri |
| 14:50.38 | tomtom- | quadbri |
| 14:51.02 | *** join/#asterisk multrix (~chatzilla@ADijon-109-1-25-246.w80-11.abo.wanadoo.fr) |
| 14:51.42 | tangel | is anyone familiar w/ extensions.conf? |
| 14:52.02 | florz | tomtom-: It is no quadbri :-) |
| 14:52.05 | tangel | should i be able to combine variables like: VAR3 => ${VAR1}${VAR2} ? |
| 14:52.14 | *** join/#asterisk randu (~randu@dsl-pppoe-pool1-37.d02.ncshlt.infoave.net) |
| 14:52.16 | bjohnson | yes |
| 14:52.18 | tangel | the handbook doesn't give syntax for that sort of thing |
| 14:52.30 | bjohnson | but I think VAR3=${VAR1}${VAR2} |
| 14:52.41 | tangel | ok.. will try that out.. thanks |
| 14:52.48 | tangel | how can one manage extensions/dialplan dynamically |
| 14:52.50 | tomtom- | oh ic ;p |
| 14:52.57 | *** join/#asterisk hans (fugalh@falcon.fugal.net) |
| 14:52.57 | tangel | do some of the db driven thingies let you make changes on the fly? |
| 14:53.01 | bjohnson | if that doesn't work .. use a setvar() .. not in the globals section |
| 14:53.03 | `Sauron | yes |
| 14:53.03 | tangel | ...or do you always have to reload extensions? |
| 14:53.11 | `Sauron | they don't all work as well as you'd hope, though. |
| 14:53.17 | bjohnson | tangel: you have to reload |
| 14:53.32 | tangel | bjohnson, setvar applies globally? |
| 14:53.39 | bjohnson | tangel: I think the developers are working on a realtime system that doesn't require reloads |
| 14:53.43 | hans | I want some to talk IAX2, which is more mature: iaxclient or libiax2? |
| 14:53.52 | `Sauron | bjohnson: You can use ast_data for dynamic dialplans |
| 14:53.58 | tangel | i'm trying to do stuff like: tangel1=SIP/1001; tangel2=SIP/1002; tangel-all=${tangel1}${tangel2} |
| 14:54.04 | tangel | and then use that throughout extensions.conf |
| 14:54.10 | bjohnson | tangel: normally you set variables in the globals section .. the format you want to use may work there |
| 14:54.23 | bjohnson | tangel: if it is not in the globals section .. use the setvar command |
| 14:54.24 | tangel | ok.. i tried it a couple ways and couldn't get it to work |
| 14:54.46 | tangel | use setvar in globals? or anywhere and the var will be global? |
| 14:54.50 | SeaForth | http://www.rafb.net/paste/results/X3eglQ88.html |
| 14:54.55 | tangel | (at least for things that come below it) |
| 14:55.00 | SeaForth | enjoy, asterisk while looking at space images :) |
| 14:55.14 | tangel | is DUNDi worth using? |
| 14:55.32 | bjohnson | depends on what you want to do .. people use it |
| 14:56.38 | sjaak538 | My ISP told me they have plugged in the wire into 2 ISDN cars on my 2 servers is that allowed !!! |
| 14:56.38 | tangel | in a practical sort of way? |
| 14:56.52 | tangel | i'm all for joining some p2p network that would allow people to use my analog line for free calls |
| 14:57.20 | tangel | hopefully it would allow me to make free calls in other areas but at this point i just want to put the work i put into setting up * to good use |
| 14:59.46 | *** join/#asterisk YoYo (~gunk@pool-141-152-71-178.roa.east.verizon.net) |
| 15:00.23 | florz | sjaak538: That depends on whether it's a PtP or a PtMP line. |
| 15:00.37 | florz | sjaak538: For PtMP it's OK. |
| 15:00.42 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 15:00.42 | *** join/#asterisk kwaldo_aloft (~waldo@dsl7.rbh1.pppoe.execulink.com) |
| 15:01.30 | kwaldo_aloft | hello all, i am new to asterisk and this group |
| 15:01.48 | kwaldo_aloft | 1st can ne1 read this to confirm connectivity via IRC? |
| 15:02.01 | YoYo | nope, we can't see you |
| 15:02.03 | YoYo | try again |
| 15:02.08 | kwaldo_aloft | <g> |
| 15:02.47 | kwaldo_aloft | I would like to set up an asterisk system (as a deployment model) |
| 15:02.49 | randu | lol |
| 15:03.22 | kwaldo_aloft | question: can someone tell me what kind of equipment you can use after the asterisk software? |
| 15:03.50 | sivana | kwaldo_aloft: lots of different kinds, depends on what you want to do |
| 15:04.04 | sivana | kwaldo_aloft: asterisk is very versatile |
| 15:04.11 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 15:04.15 | *** join/#asterisk Dalion (anon@Toronto-HSE-ppp3884464.sympatico.ca) |
| 15:04.59 | kwaldo_aloft | well, I am wondering what would be required to duplicate the functionality of a common Nortel system (multiple lines, call manipulation, voicemail etc) |
| 15:05.05 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:05.57 | kwaldo_aloft | I understand that the Asterisk software running on *IX will match the capabilities of the Nortel "brain box" |
| 15:06.20 | kwaldo_aloft | but then after the "brain box" what kind of handset units are supported>?' |
| 15:06.33 | eKo1 | Anything you want. |
| 15:07.52 | kwaldo_aloft | ok, so with a nortel system that I am thinking of duplicating, the handsets are digital, and comunicate with the central processing unit, can I use that type of phone with asterisk? |
| 15:08.00 | kwaldo_aloft | i.e. a nortel fone? |
| 15:08.36 | jero | if you want to, you need to link your nortel pbx to asterisk |
| 15:08.43 | kwaldo_aloft | <PROTECTED> |
| 15:09.02 | kwaldo_aloft | I want to replace Nortel systems completely if possible |
| 15:09.10 | jero | because the phones probably only work with the pbx family they're shipped with. |
| 15:09.22 | kwaldo_aloft | but maintain as much of the same functionality |
| 15:09.25 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 15:09.26 | florz | kwaldo_aloft: A nortel phone is a "nortel system", too, isn't it? =:-) |
| 15:10.23 | bjohnson | kwaldo_aloft: you can use analog phones or voip phones .. analog phones needs fxo hardware to talk to * |
| 15:10.42 | bjohnson | kwaldo_aloft: you can use a voip provider for incoming and/or outgoing |
| 15:11.06 | bjohnson | kwaldo_aloft: you can also buy hardware to use telco supplied lines |
| 15:11.20 | bjohnson | kwaldo_aloft: now is the time to start reading |
| 15:11.22 | bjohnson | ~docs |
| 15:11.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:11.23 | kwaldo_aloft | i find this challenging at best. |
| 15:11.49 | *** join/#asterisk mildenhall (~jmd@194.114-84-212.ippool.ndo.com) |
| 15:11.58 | bjohnson | kwaldo_aloft: perhaps you should hire someone .. post to the bus mailing list |
| 15:11.58 | kwaldo_aloft | for this model implimentation, we have a vonage connection (with converter to analog line, and a POTS line |
| 15:12.01 | florz | BTW anyone in here who has an idea why this kind of message starts appearing regularly after some time of continuous operation: Feb 4 15:45:30 WARNING[31124]: chan_zap.c:7409 zt_pri_error: PRI: !! Got reject for frame 98, but we have nothing -- resetting! |
| 15:12.06 | florz | ? |
| 15:12.12 | randu | see ya I'llback in a bit :-) |
| 15:12.18 | bjohnson | kwaldo_aloft: scrp the vonage connection and get a voip provider that is * friendly |
| 15:13.22 | Makenshi | anyone using xten x-web, is there a way to have asterisk send the user's ip address as CLI? |
| 15:13.43 | sivana | ~norm |
| 15:13.44 | jbot | it has been said that norm is server to client, but client to client is also possible. |
| 15:13.50 | kwaldo_aloft | ok, firstly, I plan to model this setup so that my company can be hired to install these systems, maybe I am in the wrong group, I was hoping to get some detailed help on how to install and configure these asterisk systems, but, perhaps someone can suggest the proper group for that? |
| 15:13.51 | sivana | ~seen normast |
| 15:13.53 | jbot | normast is currently on #asterisk (13h 37m 45s). Has said a total of 12 messages. Is idling for 9h 40m 17s |
| 15:14.06 | Makenshi | or just, set the CLI to the ip address of the caller |
| 15:14.47 | mildenhall | Have anyone used the Microsoft Real-Time Client API SDK? I'm trying to write an XML profile that will register with Asterisk, and can't seem to get one to work. |
| 15:14.50 | bjohnson | kwaldo_aloft: read the docs |
| 15:15.11 | kwaldo_aloft | thx, but: |
| 15:15.14 | bjohnson | come back with specific question |
| 15:15.25 | kwaldo_aloft | specific question: |
| 15:16.01 | nestAr | i get a lot of B-channel retarts.. is that normal? like every hour or so... |
| 15:16.19 | kwaldo_aloft | what are a list of handsets both digital and/or analog that work with asterisk that will match the capability of a common Nortel phone system |
| 15:16.21 | kwaldo_aloft | ? |
| 15:16.31 | eKo1 | mildenhall: First of all, why M$? |
| 15:16.35 | bjohnson | kwaldo_aloft: you can use analog phones or voip phones .. analog phones needs fxo hardware to talk to * |
| 15:16.35 | mutilator | ~seen mutilator |
| 15:16.36 | jbot | mutilator is currently on #asterisk. Has said a total of 1 messages. Is idling for 1s |
| 15:16.36 | *** join/#asterisk zno (~zeno@ip-160-79-174-101.autorev.intellispace.net) |
| 15:16.51 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 15:16.52 | bjohnson | kwaldo_aloft: you cn use ANY analog phones |
| 15:17.01 | mildenhall | eKo1 : Because my boss decided so! |
| 15:17.14 | eKo1 | mildenhall: Tell your boss to rethink. |
| 15:17.19 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 15:17.33 | bjohnson | kwaldo_aloft: you can use ANY voip phones that use SIP or IAX protocol (other combination are possible but get harder) |
| 15:17.50 | tangel | has anyone experienced floating point exceptions when calling from iax to zap? |
| 15:17.54 | kwaldo_aloft | so can an analog phone initiate all features or, do I need a specific digital handset? |
| 15:18.11 | mildenhall | eKo1 : Tried and he stuck his feet in. Every used it? |
| 15:18.26 | mildenhall | eko1 : Sorry - Ever used it? |
| 15:18.38 | bjohnson | an analog phone can initiate all features (or you can limit what they do) |
| 15:18.39 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 15:18.47 | eKo1 | mildenhall: I've read articles about it. Wouldn't touch it though.... |
| 15:19.11 | *** join/#asterisk JohnAB (~johna@host-212-158-200-158.bulldogdsl.com) |
| 15:19.24 | mildenhall | eko1 : It seems OK and works if not registered with anything |
| 15:19.34 | kwaldo_aloft | are there phones that work with asterisk that have many of the similar key shortcuts and LCD feedback that one would find in a common Nortel system? |
| 15:19.37 | bjohnson | kwaldo_aloft: phone hardware selection is more about extra buttons and handset features than pbx features |
| 15:20.23 | bjohnson | kwaldo_aloft: Polycom IP500s are popular voip phones that are about the same idea .. about $180 each |
| 15:20.24 | kwaldo_aloft | ok, well now I am learning something, thx, is there a HCL somewhere for Asterisk compat handsets? |
| 15:20.44 | kwaldo_aloft | this idea of VOIP fones I don't get |
| 15:20.45 | Uther_P | are there any api's that would allow me to write an application that would initiate commands, scripts or parts of a dial plan? (i.e. could I write an app that could tell asterisk to dial a phone and playback or initiate other plans on that line as it the call came in to the server instead of the server making the call)? |
| 15:21.00 | bjohnson | kwaldo_aloft: new models of handsets keep coming out .. you can start at the wiki |
| 15:21.14 | florz | Uther_P: Look for call files and manager api |
| 15:21.23 | Uther_P | ok thanks |
| 15:21.25 | florz | Uther_P: And AGI maybe |
| 15:21.30 | kwaldo_aloft | I just want a system to manage my phone lines after they enter our builing, but not IP phones.. |
| 15:21.30 | bjohnson | kwaldo_aloft: again .. ANY analog phone would work .. from the $10 wall phone at walmart to the $400 multihandset cordless phone |
| 15:21.34 | kwaldo_aloft | they are too costly |
| 15:21.51 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 15:21.57 | Slainte | Good Morning |
| 15:22.00 | Uther_P | agi from what I understand is for calls made to xfer to an application, I want something to do just the opposate |
| 15:22.16 | bjohnson | kwaldo_aloft: analog phones require fxs hardware ports .. about $50 each |
| 15:22.25 | kwaldo_aloft | OK |
| 15:22.36 | bjohnson | plus the phone cost |
| 15:23.06 | florz | Uther_P: yep, but after a call has been established via a call file, you might want to control it using an AGI script, for example |
| 15:23.10 | kwaldo_aloft | so for 2 incoming POTS lines, to be distributed to our office with full pbx features, what cards would be required? |
| 15:23.19 | Uther_P | kwaldo_aloft: I use Sipura 2000, which has 2 fxs ports... very configurable and I've had good performance from them |
| 15:23.34 | JohnAB | TDM400P with 2 FXO modules |
| 15:23.35 | bjohnson | 2 fxo ports and whatever you want for the user interface |
| 15:23.40 | JohnAB | or equiv. |
| 15:24.13 | kwaldo_aloft | so that would cover the incomming lines right?> |
| 15:24.15 | bjohnson | kwaldo_aloft: the problem you are running into is that these systems are extremely configurable |
| 15:24.31 | kwaldo_aloft | how about the internal distribution? |
| 15:24.39 | eKo1 | I have * running as user asterisk. When * starts, it spawns some mpg123 procs. that are owned by root. |
| 15:24.40 | kwaldo_aloft | I agree |
| 15:25.32 | JohnAB | can someone perhaps help me with a problem. I have a working SIP extension that works fine with a SIP softphone; however when I configure my Cisco 7960 to use that SIP login, it can make outgoing calls but doesn't seem to work for incoming calls |
| 15:25.33 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:25.43 | eKo1 | I've also noticed that * has troubles (sometimes) shuting down until I personally kill these mpg123 processes. |
| 15:25.46 | JohnAB | i get the following error when I try to dial the 7960: app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 15:25.46 | Uther_P | JohnAB: in the cli, type "sip show peers" and see if it is registering |
| 15:25.47 | bjohnson | kwaldo_aloft: as JohnAB said .. a TDM400P with 2 FXO modules would handle the pstn lines .. that is not the only option .. but is one option .. however not a good one if you have a SIDN telco line (common in europe I believe) |
| 15:25.47 | bjohnson | ISDN .. not SIDN |
| 15:26.05 | kwaldo_aloft | however, I think it would help if I could find a deployment diagram or summary, I find alot of the information is very specific to the asterisk software, but the hardware deployment, seems somewhat convoluted |
| 15:26.10 | zno | anyone else have problems with sipura spa-841's not obtaining a dhcp address? |
| 15:26.21 | kwaldo_aloft | I am in NA |
| 15:26.25 | JohnAB | TDM400P works "ok" in europe, agreed it's not great, for a start the line impedence is wrong which annoys my telco but hey |
| 15:26.33 | kwaldo_aloft | so standard lines i think |
| 15:26.33 | *** join/#asterisk sabre (sabre@69.149.209.83) [NETSPLIT VICTIM] |
| 15:26.33 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) [NETSPLIT VICTIM] |
| 15:26.33 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) [NETSPLIT VICTIM] |
| 15:26.33 | *** join/#asterisk angler_ (~angler@suid.digium.com) [NETSPLIT VICTIM] |
| 15:26.40 | bjohnson | kwaldo_aloft: you've said you daon't want voip phones .. and you want physical handsets like the Nortels so softphones are out .. only leaves fxs ports and whatever analog phone you desire |
| 15:26.52 | *** join/#asterisk bannerman (~bannerman@dpc6682105089.direcpc.com) |
| 15:27.03 | Dalion | why not jus tchange line impedence |
| 15:27.05 | Zeeek | siemens phones on TDM400 FXS anyone has mit experience? |
| 15:27.09 | bannerman | Anyone have experience using Asterisk with Covad's VoIP service? |
| 15:27.11 | Dalion | making a device or something |
| 15:27.12 | bjohnson | kwaldo_aloft: hundreds of deployments examples in those docs I referenced |
| 15:27.14 | bjohnson | ~docs |
| 15:27.15 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:27.18 | kwaldo_aloft | ok, so i need 2 incomming ports, and 2 outgoing analog fxs ports right? |
| 15:27.30 | JohnAB | you need as many fxs ports as you want handsets |
| 15:27.34 | bjohnson | 2 fxs would give 2 lines to phones |
| 15:27.51 | bjohnson | you could have 2 handsets .. or 20 handsets with access to 2 line |
| 15:27.56 | Dalion | now what kind of title is "Vice President Industry Solutions" |
| 15:27.57 | Dalion | lol |
| 15:28.06 | bjohnson | most offices do not want to share lines the way a house does |
| 15:28.52 | JohnAB | yeah i think it's fair to say that you need 1 handset per fxs port in most situations |
| 15:29.04 | bjohnson | kwaldo_aloft: you haven't yet said how many phones you want |
| 15:29.44 | bjohnson | JohnAB: depends on use .. that is accurate for business situations .. not necessary in home deployments |
| 15:30.17 | Zeeek | "Honey, I'm baaaack" = 1 FXS -> many handsets |
| 15:30.32 | JohnAB | true, but presumably there's a limit to how many handsets you can have off one FXS |
| 15:30.40 | Zeeek | "Wilson get in here, NOW!" = 1 fxs per phone |
| 15:30.40 | bjohnson | I do 1 fxs to many handsets at home |
| 15:30.42 | kwaldo_aloft | how about using another digital base station after the asterisk box, that manages 2 line functionality, then distributes the signal to many wireless handsets? |
| 15:30.52 | JohnAB | especially in europe, with the ringing capacitors we use |
| 15:31.12 | bjohnson | JohnAB: yes .. depends on hardware .. I have 6 off a SPA 3000 fxs port |
| 15:31.34 | *** join/#asterisk sabre (sabre@69.149.209.83) |
| 15:31.39 | JohnAB | your home must be a headache with so many phones ringing :) |
| 15:31.48 | ManxPower | ~docs |
| 15:31.49 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:31.58 | bjohnson | kwaldo_aloft: you can do whatever you want .. that is the beauty of it (and what confuses most people) |
| 15:32.03 | *** join/#asterisk opticalcarrier (~bo@optical.neteng.earthlink.net) |
| 15:32.25 | *** join/#asterisk Phirsk (~Phirsk@northpole.globalvc.co.uk) |
| 15:32.25 | bjohnson | JohnAB: just like every house in NA |
| 15:32.28 | opticalcarrier | is there a way to make asterisk periodically go clear MWI on phones |
| 15:32.30 | bjohnson | JohnAB: besides .. I would want all the phones to ring on an incoming cll anyway |
| 15:32.54 | JohnAB | i'd settle for one phone ringing at this moment in time |
| 15:33.02 | JohnAB | but i hear what you say |
| 15:33.22 | Phirsk | hi all... quick question... i know its a long shot, but can i get asterisk to connect to the skype network? |
| 15:33.31 | bjohnson | in fact .. I have one handset that connects in front of the voip system that can also be used |
| 15:33.37 | *** join/#asterisk adjacent (~scott@office.bftwave.com) |
| 15:33.40 | bjohnson | Phirsk: no |
| 15:33.51 | JohnAB | yeah i've been known to do that when my * is playing up |
| 15:33.59 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 15:34.10 | zno | Phirsk: is skype SIP based? |
| 15:34.16 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 15:34.29 | bjohnson | no |
| 15:34.29 | JohnAB | not afaik, although it uses iLBC |
| 15:34.29 | Phirsk | zno: no, but was hoping for a channel driver or similar |
| 15:34.32 | Jas_Williams | zno No |
| 15:34.36 | Zeeek | skype is proprietary and hooks to nothing |
| 15:34.43 | bjohnson | skype uses proprietary protocol |
| 15:34.46 | zno | then Skype is useless |
| 15:34.49 | bjohnson | yes |
| 15:34.55 | opticalcarrier | why is skype so sccessful? |
| 15:34.58 | Zeeek | too bad too cause it works better than anythiong else |
| 15:35.06 | JohnAB | because it's simple to use, it has no NAT problems |
| 15:35.12 | bjohnson | their phones are also usually USB which just nails the coffin lid shut |
| 15:35.12 | Zeeek | and excellent sound |
| 15:35.24 | Phirsk | its popular and effective, so thats why i wanted to hook between the 2 worlds |
| 15:35.34 | ManxPower | skype threw out all the existing VoIP protocols and designed their own from the ground up. |
| 15:35.50 | Zeeek | and it has text chat so you can say "Turn your mic on!" |
| 15:35.50 | JohnAB | i'm saying nothing, i have a USB headset on right now |
| 15:35.52 | adjacent | i have a situation maybe someone can help me resolve. I have been reselling voip service now for almost 2 years through a SIP provider. we have a local PRI, and cisco voice gateway for local calling. now i no loger have a ASIP provider. if i build an asterisk box for long distance routing can i continue to sell voip service? |
| 15:35.56 | Phirsk | any way to build a peer between the 2 ? |
| 15:36.03 | bjohnson | Zeeek: depends on the OS used |
| 15:36.14 | Zeeek | what does? |
| 15:36.19 | ManxPower | adjacent, Yes, in theory. |
| 15:36.30 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 15:36.30 | opticalcarrier | is there any way to manually clear MWI on a cisco 7960? |
| 15:36.47 | Phirsk | ive got messenger to plug into asterisk, so can interconnect with msn clients, but wanted to open the door to a whole lot more - i.e. skype |
| 15:36.49 | adjacent | ManxPower: heh. i lose dialtone on monday... i need more than theory, do you have any reference to point me towards? |
| 15:36.57 | bjohnson | JohnAB: USB = drivers required (for current and all future OSs) + computer on required |
| 15:37.09 | adjacent | im getting $.025 a minute now |
| 15:37.35 | ManxPower | adjacent, You cannot do what you want to do quickly. |
| 15:37.38 | JohnAB | the usb headset has its uses |
| 15:37.40 | bjohnson | JohnAB: not a great future for a limited deployment system dependant on one company |
| 15:38.00 | ManxPower | adjacent, I assume you have all the billing, etc stuff already handled using the Cisco? |
| 15:38.01 | adjacent | ManxPower: why? should i look for another partner? |
| 15:38.09 | JohnAB | oh sure with skype, i don't have a skype usb phone, just a plain old headset |
| 15:38.32 | adjacent | ManxPower: ehh. i dont, i left the voip stuff almost completely hands-off to my old partner. maybe a mistake now |
| 15:39.00 | *** join/#asterisk mildenhall (~jmd@194.114-84-212.ippool.ndo.com) |
| 15:39.02 | adjacent | i dont know. actually |
| 15:39.16 | ManxPower | adjacent, Imagine deploying Cisco Call Manager quickly. Or imagine deploying a Windows2k server quickly, but you had never used Windows before. The system and setup and testing is just too complex. Most people spend at least a coupole of months with Asterisk before deploying in production. |
| 15:39.21 | JohnAB | does the 7960 have to register to a context, does anyone know? |
| 15:39.55 | ManxPower | JohnAB, SIP does not have the concept of a context. |
| 15:40.00 | *** join/#asterisk Ridgeback (~Ridgeback@ppp54-145.lns1.adl2.internode.on.net) |
| 15:40.11 | ManxPower | Asterisk adds the idea of context, but the SIP client doens't know about it. |
| 15:40.14 | Ridgeback | morning all |
| 15:40.24 | JohnAB | doesnt' that map to the sip concept of a realm? |
| 15:40.32 | ManxPower | JohnAB, NormAst |
| 15:40.39 | ManxPower | ..er... JohnAB no |
| 15:40.40 | adjacent | ManxPower: ok. i have 90% of what i need. an existing county wide wireless network, 50 cisco ata 186's, a cisco voice gateway, PRI access, the abilty to port numbers. and now im without long distance routing... |
| 15:40.44 | JohnAB | just grasping at straws here :) |
| 15:41.15 | ManxPower | adjacent, You keep everything the same and use a SIP service provider. |
| 15:41.25 | adjacent | so i need a call manager for accounting, and some way (i dont know or understand this yet) to route LD |
| 15:41.30 | Ridgeback | anyone here ever use of those IAX phones from iaxtalk? |
| 15:41.36 | ManxPower | adjacent, Asterisk has no billing software |
| 15:41.41 | *** join/#asterisk felipex (~dsfdsf@host179-130.pool8172.interbusiness.it) |
| 15:41.59 | adjacent | ManxPower: ok. any free service? or should i still look to a paid SIP service? |
| 15:42.01 | *** join/#asterisk eivindtr (~Eivind@193.91.146.34) |
| 15:42.41 | bjohnson | JohnAB: the 7960 would register as one of the devices in your sip.conf |
| 15:42.48 | ManxPower | adjacent, all the free services suck for business use (at least in my opinion) |
| 15:43.04 | bjohnson | that sip.conf entry would send incoming calls from the 7960 to a specific context |
| 15:43.07 | adjacent | im sorry. i know im uneducated about this, lots of reading to do, but with a limited time frame any general help is greatly appreciated |
| 15:43.16 | JohnAB | the 7960 appears to register because it works fine for outgoing calls, i.e. incoming calls from the 7960 |
| 15:43.21 | adjacent | ManxPower: ok. i will look for a paid service then |
| 15:43.24 | JohnAB | so i'm guessing sip.conf is sending it to the right context |
| 15:43.27 | ManxPower | adjacent, You need to hire an Asterisk consultant, but don't expect them to work miricles. |
| 15:43.44 | adjacent | ManxPower: heh. i know. 2 years and calls still suck. |
| 15:43.50 | JohnAB | well in any case that SIP entry works fine with a softphone |
| 15:43.50 | ManxPower | JohnAB, registration only is an issue with calls Asterisk -> Phone |
| 15:44.03 | bjohnson | JohnAB: to get calls to go to the 7960 .. you have to define an extension for it in extensions.conf and dial(SIP/7960name) |
| 15:44.04 | adjacent | voip inho is much beer suited for in-office calling |
| 15:44.11 | ManxPower | Phone -> Asteirsk does not require any registation at all |
| 15:44.32 | JohnAB | bjohnson: yes, but surely that's no different as for a softphone, which worked fine on the same SIP extension |
| 15:44.42 | bjohnson | correct |
| 15:45.05 | JohnAB | and yet calls going to the the 7960 don't get there |
| 15:45.09 | JohnAB | that's random then |
| 15:45.29 | bjohnson | JohnAB: check 7960 config .. check codecs .. check registering |
| 15:45.35 | ManxPower | JohnAB, "sip show peers" will show the IP address of the phone, if it's registering |
| 15:45.45 | ManxPower | If it ways (Unspecified) then the phone is NOT registering. |
| 15:45.51 | JohnAB | the 7960 config is the thing i can't find much on |
| 15:45.55 | labo | JohnAB, can you telnet to the 7960 ? |
| 15:46.04 | JohnAB | telnet, not tried it, just rebooting it |
| 15:46.13 | bjohnson | JohnAB: likely the 7960 config is the problem |
| 15:46.21 | JohnAB | yeah sounds that way |
| 15:46.24 | JohnAB | it's coming up as unspecified |
| 15:46.36 | labo | Type: debug sip-reg & debug sip-reg-state when you telnet to it in mode 2. |
| 15:46.47 | bjohnson | get that going and sounds like rest will "magically" work |
| 15:47.15 | bjohnson | also sounds like labo has first hand experience with that phone .. you're in luck |
| 15:47.25 | JohnAB | doesn't seem i can telnet to it anyhow, probably disabled by default |
| 15:47.42 | labo | you are using the sip image i suppose |
| 15:47.57 | JohnAB | yes, 7.3, like i said i can call out fine |
| 15:48.01 | *** join/#asterisk Wireless (~bad@220.233.77.87) |
| 15:48.03 | labo | and no application loader errors, etc etc |
| 15:48.17 | JohnAB | no errors at all on the unit that i can see |
| 15:48.41 | JohnAB | it knows what its extension is |
| 15:48.53 | labo | can you pastebin your 7960 config @ sip.conf, ill compare it to mines. |
| 15:49.11 | labo | mabe its a codec issue, are you using g729 |
| 15:49.16 | *** join/#asterisk Juggie (agony@24.114.136.55) |
| 15:49.35 | JohnAB | checking |
| 15:49.51 | blitzrage | ManxPower: do you know what purpose specifically loading res_musiconhold.so in the modules.conf sample is for? |
| 15:50.10 | blitzrage | Seems to load fine with it commented out (just not first) |
| 15:50.32 | labo | JohnAB, are you using smoe specific codec in your SIPD****.cnf ? |
| 15:50.35 | labo | some* |
| 15:50.36 | Juggie | i think there may be a reason to laod modules in a certain order |
| 15:51.02 | Zeeek | why would I be seeing -- Registered to '69.73.19.178', who sees us as 192.168.1.5:4569 followed by the same line with my ext ip ? |
| 15:51.18 | *** join/#asterisk dalabera (~Dalabera@146.82.190.162) |
| 15:51.25 | dalabera | hello everyone |
| 15:51.50 | Zeeek | someone here once said you could use externip=domain.com - WRONG. It screwed up SIP |
| 15:52.08 | Zeeek | you can use it on a fixed ip |
| 15:52.22 | blitzrage | Zeeek: I've used it on a domain name |
| 15:52.42 | Zeeek | but on a server that changes ip ? because it broke sip as soon as I changed addresses |
| 15:52.56 | Zeeek | SIP was working fine - just no audio :) |
| 15:53.01 | blitzrage | probably because the DNS didn't update |
| 15:53.02 | `Sauron | as long as the domain points at your IP, you're good to go |
| 15:53.10 | Zeeek | as soon as I put the ip adr in, it was back to normal |
| 15:53.28 | JohnAB | http://pastebin.ca/5278 <-- my sip.conf and SIP......cnf |
| 15:53.28 | Zeeek | pinging the domain name from the box was right ip |
| 15:53.43 | Zeeek | and this was after a reboot btw |
| 15:53.59 | Slainte | I am open to suggestions on how to setup a user (polycom IP600) so that all oeprator traffic goes down one line, and all internal or direct traffic goes to another line. I have it working but want it efficient. |
| 15:54.53 | Zeeek | ok question 2: why do my newest phones get no CID? huh? huh? |
| 15:54.57 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 15:54.58 | JohnAB | just enabling telnet so i can debug it, seems there's a config for it |
| 15:55.00 | labo | JohnAB, what about codec in the 7960 config, which one are you using, i doubt you are doing ilbc |
| 15:55.23 | labo | yes, sip debug messages on 7960's tell a lot. |
| 15:55.27 | Zeeek | while SIP was broken I had to make a call on a real phone! It was awful! |
| 15:55.34 | JohnAB | i'm trying to figure what codec it's using, not sure how i actually tell :) |
| 15:55.48 | Zeeek | there was no breaking up - I couldn't handle it, it made me nervous |
| 15:55.58 | *** join/#asterisk jcollie (~jcollie@12-216-136-56.client.mchsi.com) |
| 15:55.59 | Zeeek | JohnAB sip show channels |
| 15:56.22 | labo | JohnAB, you should have a preferred_codec: in your SIPDefault.cnf |
| 15:56.51 | JohnAB | nope, i just copied the 2 line one from the cisco docs |
| 15:56.58 | JohnAB | image_version:P0S3-07-3-00 ; |
| 15:57.01 | `Sauron | hum |
| 15:57.02 | JohnAB | proxy1_address: 192.168.254.20 ; |
| 15:57.09 | JohnAB | i'm guessing that may be not helping my problem |
| 15:57.32 | `Sauron | anyone here have the zyxel p2000w? |
| 15:57.46 | JohnAB | app_dial.c:884 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) <-- when i try to call the phone |
| 15:57.48 | Jas_Williams | JohnAB: Do you have this line in SIPDefault.cnf proxy_register: 1 |
| 15:57.59 | JohnAB | i do now :) |
| 15:58.11 | dalabera | Hi, I Lost my Extension.conf configuration, however the configuration is loaded on the Asterisk Box, on the wiki it says there is a command to wrdite the current configuration of the extension.con to file however my release does not have the command "save dialplan" does anyone please explain me what's going on? |
| 15:58.22 | labo | Do this JohnAB , check those loligo.com 7960 conf files and try them |
| 15:58.29 | wwalker | ManxPower Don't do it!!! :) |
| 15:58.34 | JohnAB | on my way over |
| 15:58.37 | labo | :) |
| 15:58.43 | `Sauron | ManxPower: Depends on what you want to do with it? |
| 15:58.48 | *** join/#asterisk tull (~sdk@wwwcache2.livjm.ac.uk) |
| 15:58.58 | ManxPower | wwalker, My last dell workstation was rock solid stable, reasonably fast, and reasonably priced. |
| 15:59.20 | JohnAB | as if by magic, it's registered |
| 15:59.39 | ManxPower | I gave it to my boyfriend because his machine started crashing. We swapped computers and it's apparently some motherboard related issue that I cannot resolve. |
| 15:59.46 | JohnAB | and more magic, it actually rings, although there's hellish static, which is probably codec related |
| 15:59.58 | bjohnson | they had a sweet deal on a desktop with a 17" lcd that had me tempted |
| 16:00.13 | Jas_Williams | Without proxy_register: 1 |
| 16:00.13 | Jas_Williams | <PROTECTED> |
| 16:00.21 | tull | Hello |
| 16:00.30 | JohnAB | i don't mind dell, my account manager gives me good deals so that's ok by my books |
| 16:00.53 | JohnAB | yeah thanks jas, it's incredibly obvious when you see it, but boy does cisco's sip documentation suck |
| 16:01.03 | tull | i wanted some help to come up with a minimum spec list to make a demonstration in uni. |
| 16:01.12 | `Sauron | Manx: What's your opinion on the Zyxel P2000W? |
| 16:01.25 | tull | just a small pbx and a line and 3 users. |
| 16:01.34 | bjohnson | tull: a sipura SPA3000, a linux laptop, and a voip phone |
| 16:01.45 | bjohnson | 2 voip phones then |
| 16:01.55 | JohnAB | the P2000W has had very mixed reviews, i've been umming about buying one |
| 16:02.19 | `Sauron | Humm. |
| 16:02.22 | JohnAB | i've heard that the CPU is perhaps a bit slow, because with WEP enabled it breaks up a lot |
| 16:02.30 | `Sauron | Cuz, the WiSIP hasn't had very good reviews |
| 16:02.41 | `Sauron | everybody says to get the p2000w |
| 16:02.41 | JohnAB | but i don't know what the conditions of that, the codecs etc. |
| 16:02.52 | `Sauron | And I don't know of any other non-cisco wifi sip phones |
| 16:02.54 | mAsH` | can i connect * to a PABX by an HFC card? |
| 16:02.56 | Jas_Williams | Its the same hardware as a WiSIP |
| 16:03.02 | bjohnson | `Sauron: do you really need a wifi phone? |
| 16:03.12 | tull | bjohnson if i need more users just get a switch and add the voip phones right? |
| 16:03.17 | `Sauron | Yes. I hate being chained to my desk |
| 16:03.33 | bjohnson | tull: yes .. you could also show a softphone with that rig |
| 16:03.36 | Jas_Williams | `Sauron: I use a WiSip with p200w firmware as it supports stun |
| 16:03.44 | labo | sad to say, but i regret of buying a p2000 |
| 16:03.50 | bjohnson | tull: you'll need a switch anyway for the SPA3000 |
| 16:03.55 | `Sauron | labo: why? |
| 16:04.03 | tull | bjohnson right ok. |
| 16:04.06 | tzanger | hmm |
| 16:04.09 | JohnAB | labo: what's your experience of it? |
| 16:04.11 | tzanger | zapscan doesn't work right iwth groups |
| 16:04.13 | labo | because it sucks, it has poor battery, sometimes wep encryption does not work |
| 16:04.19 | Delvar | anyone know of any tools that i can use to stress my connection for testing my qos? |
| 16:04.20 | bjohnson | tull: actually .. use a router .. then you could plug the WAN port into the LAN if desired and have your own subnet |
| 16:04.23 | tzanger | zapscan(1) or zapscan(g1) both just "hang" (no channels) |
| 16:04.29 | `Sauron | Hmm. |
| 16:04.41 | `Sauron | Jas: What's your experience with the wisip? |
| 16:04.52 | tzanger | wisip sucks ass |
| 16:04.52 | bjohnson | tull: don't try to connect back through the router from the lan until you've prepared suitable port forwarding |
| 16:05.04 | labo | wisip ir worse than p200 ive been told :) |
| 16:05.08 | labo | is* |
| 16:05.22 | bjohnson | `Sauron: there are other wireless options that are not wifi sip phones |
| 16:05.36 | JohnAB | i've been tempted to buy one so i can have a phone upstairs, that's all though |
| 16:05.50 | shido6 | cordless iaxy |
| 16:05.53 | tzanger | don't go wisip |
| 16:05.54 | bjohnson | JohnAB: just use a cordless phone on a fxs |
| 16:05.58 | shido6 | cordless and a iaxy |
| 16:05.58 | `Sauron | bjohnson: I am quite aware of that. |
| 16:05.58 | JohnAB | bjohnson, yes, ata186 with dect |
| 16:06.02 | tzanger | all around poor device in my opinion |
| 16:06.28 | JohnAB | i have fxs with a cordless phone, but i have issues with irq sharing and the fxs |
| 16:06.37 | vaewyn | if you go wisip get teh zyxel firmware... but I recommend the Hitachi IP-5000 it's my new toy :P |
| 16:06.52 | bjohnson | JohnAB: get a non-pci fxs |
| 16:07.02 | vaewyn | although an awful lot of Japanese on this thing :P |
| 16:07.05 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 16:07.16 | chipig | but so purty.. |
| 16:07.21 | tzanger | JohnAB: I have a panasonic cordless which hass issues with everything :-) |
| 16:07.51 | tull | bjohnson ok! thanks a lot. |
| 16:08.15 | JohnAB | just a cruddy motherboard on my * server really, it thinks that everything should be on irq11 |
| 16:08.26 | JohnAB | which seems to make fxs-fxo bridging unhappy |
| 16:08.30 | jcollie | can anyone tell me... is the "conferencing" on the polycom ip300... is the conferencing handled by asterisk or the phone? |
| 16:09.25 | bjohnson | tull: maybe drop the second voip phone and get a SPA 2000 instead .. gives 2 more fxs ports |
| 16:09.48 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 16:09.59 | bjohnson | tull: really wow them with a bluetooth headset/softphone combo |
| 16:10.09 | JohnAB | bjohnson, yeah i was thinking of probably getting an ata186 or equivalent, besides i have all my network wiring in the walls and phone wires get in the way |
| 16:10.20 | tzanger | http://www.hitachi.co.jp/Div/omika/prdcts/h-ip/ |
| 16:10.27 | tzanger | not the ip-5000 you are talking about I imagine. :-) |
| 16:11.17 | JohnAB | google is clearly not your friend :) |
| 16:11.39 | tzanger | vaewyn: |
| 16:11.39 | tzanger | Hitachi Cable, Ltd. will start marketing the Wireless IP phone "Wireless IP-5000" on 1 April. |
| 16:11.43 | JohnAB | http://products.wi-fiplanet.com/wifi/phone_handset/1105849969.html |
| 16:11.44 | tzanger | so it's not an april fool's joke? |
| 16:12.00 | JohnAB | http://www.abptech.com/mainpages/products/HCL-WirelessIP5000.html better page |
| 16:13.06 | tzanger | not a bad looking phone |
| 16:13.16 | tzanger | vaewyn: how much controld o you have over the display/soft buttons from the SIP side? |
| 16:13.21 | Zeeek | anyone using siemens DECT ? |
| 16:13.25 | bannerman | Anyone have experience with simpletelecom's service? Any history of trouble? |
| 16:13.42 | JohnAB | yeah i have siemens dect |
| 16:13.53 | Zeeek | JohnAB hooked to an FXS? |
| 16:14.07 | JohnAB | yeah, but i have problems, see above, but it's not siemens related afaik |
| 16:14.18 | Zeeek | any callerid issues? I'm not seeing it on the phone |
| 16:14.29 | Zeeek | the other phones do it |
| 16:14.38 | Zeeek | shitty Alcatel does it |
| 16:14.46 | JohnAB | dunno, my callerid doesn't work on my FXO |
| 16:14.50 | Zeeek | Siemens C200/C2 no cid |
| 16:14.52 | Juggie | has anyone here used a mitel 5055/w asterisk |
| 16:14.56 | Juggie | mine keeps locking up |
| 16:15.00 | Zeeek | well even internal calls |
| 16:15.14 | blitzrage | I suppose no one could explain to me in plain english what the [globals] section of modules.conf does? |
| 16:15.20 | Zeeek | If 2006 calls 2007 it should work and doesn't |
| 16:15.25 | blitzrage | and when I might have to add a line to it? |
| 16:15.26 | JohnAB | not sure which model i have but i don't recall it being a problem |
| 16:15.40 | JohnAB | i had great problems getting it to ring but welcome to the uk |
| 16:15.44 | Zeeek | JohnAB are you in EuropeN |
| 16:15.47 | JohnAB | yes |
| 16:15.48 | Zeeek | ya ok |
| 16:16.00 | Zeeek | You know there's a patch to change the ring frequency? |
| 16:16.08 | Zeeek | It works on the continent |
| 16:16.14 | Zeeek | but prolly not with BT :( |
| 16:16.23 | JohnAB | our CLI works really weirdly |
| 16:16.34 | Zeeek | none of my older phiones ring on FXS without the patch |
| 16:16.36 | JohnAB | IIRC, they reverse the polarity of the line before it rings |
| 16:16.46 | Zeeek | ya those brits are nutz :) |
| 16:16.54 | JohnAB | and then put it over the line |
| 16:17.09 | JohnAB | so unless you can detect the polarity change, detecting cli is hard |
| 16:17.22 | *** join/#asterisk Tough_Nuts (~chatzilla@26.137.202.68.cfl.rr.com) |
| 16:17.46 | Zeeek | ya |
| 16:17.48 | JohnAB | the ringing i got working with a capacitor |
| 16:18.07 | Zeeek | I screwed around for a half day with capacitors |
| 16:18.11 | Zeeek | never worked |
| 16:18.18 | JohnAB | put a PSTN master socket in the way |
| 16:18.25 | Zeeek | one line to change in wcfxs.c |
| 16:18.36 | JohnAB | =) |
| 16:19.11 | JohnAB | i've been meaning to get cli to work with a modem |
| 16:19.28 | JohnAB | i bought a modem but i got ripped off and sent the wrong one |
| 16:19.50 | Zeeek | here's the line just in cqase: |
| 16:19.52 | Zeeek | {21,"RING_X",0x023A}, // new value for 25Hz |
| 16:19.59 | JohnAB | thanks |
| 16:20.12 | Zeeek | maybe save some capacitors :) |
| 16:20.27 | Zeeek | just comment out the old {21,... |
| 16:21.29 | _-Jon-_ | Has anyone noticed that BroadVoices phone support sucks? |
| 16:21.46 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-99.ags.bellsouth.net) |
| 16:21.55 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 16:21.55 | junky[work] | Feb 4 11:20:29 WARNING[21509]: chan_zap.c:785 zt_open: Unable to open '/dev/zap/pseudo': No such device or address |
| 16:21.55 | junky[work] | Feb 4 11:20:29 ERROR[21509]: chan_zap.c:6848 chandup: Unable to dup channel: No such device or address |
| 16:22.06 | `Sauron | Jon: define "sucks" |
| 16:22.12 | junky[work] | i really need a digium card to run MeetMe, wtf? |
| 16:22.31 | ManxPower | Wow! All my PRI channels are in use. |
| 16:22.33 | _-Jon-_ | Sauron, you call at 3am and apparently they're "busy", and you call during the day and get the same crap. They never answer the phone |
| 16:22.42 | Zeeek | $sucks = $SUPPORT_EXISTS || $SUPPORT_INCOMPETENT |
| 16:22.47 | `Sauron | They've answered everytime I called |
| 16:22.49 | ManxPower | Of course that's because some idiot dialed an extension by dialing their DID, so the call is using 2 channels |
| 16:22.55 | _-Jon-_ | Really? Try calling at 3am |
| 16:23.00 | `Sauron | It might take a while on hold, but they answered... |
| 16:23.04 | _-Jon-_ | Hmmm |
| 16:23.07 | `Sauron | I'm sleeping at 3am. As should you. |
| 16:23.23 | _-Jon-_ | Heh |
| 16:23.36 | bjohnson | has anyone got one of the SPA1001 ? They look pretty slick |
| 16:23.39 | _-Jon-_ | Oh and what's up with a telephone company not having a tollfree number? |
| 16:23.47 | _-Jon-_ | That's like my ISP not having a website |
| 16:23.48 | `Sauron | bjohnson: I do. |
| 16:23.59 | `Sauron | Umm, they do have a tollfree number. |
| 16:24.02 | bjohnson | supposed to be half the size of the SPA 2000 and provided for free from Broadvoice |
| 16:24.20 | _-Jon-_ | Really? |
| 16:24.23 | bjohnson | `Sauron: how'd you get it? They selling them now? |
| 16:24.27 | _-Jon-_ | This is news to me |
| 16:24.31 | `Sauron | bjohnson: Yep. |
| 16:24.33 | `Sauron | Umm. |
| 16:24.42 | `Sauron | I got it with my BV signup |
| 16:25.02 | bjohnson | what activation fees? |
| 16:25.11 | Zeeek | Can anyone speak of anything at all that DOESN'T suck ? I can wait :) |
| 16:25.34 | `Sauron | bjohnson: From my bill: Device Fee $49.95 |
| 16:25.43 | `Sauron | Device Discount ($49.95) |
| 16:25.45 | `Sauron | :) |
| 16:25.54 | bjohnson | Zeeek: my spa 2000 and 3000s are awesome |
| 16:26.19 | Zeeek | maybe when you call the tollfree it's like here where they say they can only help you with changing your credit card number? |
| 16:26.20 | _-Jon-_ | Hmmm no support # provided at all |
| 16:26.22 | `Sauron | bjohnson: I ended up with $76.99 in activation |
| 16:26.23 | bjohnson | `Sauron: do they have another activation fee too? Do they have account closing fees? |
| 16:26.55 | `Sauron | IF you return all equipment, there's no account closing fee |
| 16:27.07 | bjohnson | no .. I want to keep the unit :) |
| 16:27.26 | `Sauron | bjohnson: Then they don't waive the $49.95 account closing fee |
| 16:27.34 | `Sauron | so basically, you'll get the spa1001 for $49.95 |
| 16:27.38 | _-Jon-_ | Account closing fee? with who? |
| 16:27.47 | JohnAB | do you need a sound card for MoH to work? |
| 16:27.55 | _-Jon-_ | JohnAB, no |
| 16:28.14 | `Sauron | Jon: bv |
| 16:28.16 | `Sauron | duh |
| 16:28.18 | JohnAB | didn't think so, hmm |
| 16:28.23 | `Sauron | read your contract when you sign up for things |
| 16:28.43 | `Sauron | Figures |
| 16:28.45 | `Sauron | idjit |
| 16:28.56 | _-Jon-_ | did you just call me an idiot? |
| 16:29.16 | `Sauron | Want me to spell it out? |
| 16:29.23 | _-Jon-_ | Hah |
| 16:29.38 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:29.38 | _-Jon-_ | what's that url? :) |
| 16:30.07 | `Sauron | bjohnson: I'm considering keeping my spa1001 and just giving them the $49.95 when/if I close the account |
| 16:30.17 | `Sauron | the thing is sexy |
| 16:30.30 | bjohnson | I was close to signing up with vonage to get the linksys pap2. Even after the close out cost .. it was a good deal. But I couldn't crack it. |
| 16:30.33 | _-Jon-_ | JohnAB, mpg321 maybe? |
| 16:30.45 | JohnAB | Version 0.59s-r9 (2000/Oct/27) |
| 16:30.46 | bjohnson | `Sauron: $65 at voipsullply |
| 16:30.50 | JohnAB | is it 0.59r that works? |
| 16:30.53 | bjohnson | voipsupply |
| 16:31.02 | `Sauron | yep |
| 16:31.17 | `Sauron | $69 vs. $49 |
| 16:31.19 | `Sauron | dum di dum |
| 16:31.20 | _-Jon-_ | I hav mpg123-0.59r |
| 16:31.28 | `Sauron | well, there's the other signup costs |
| 16:31.34 | _-Jon-_ | 0.59s wasn't working for me |
| 16:31.46 | JohnAB | yeah i remember someone mentioning that |
| 16:31.48 | JohnAB | get lots of static? |
| 16:32.03 | bjohnson | `Sauron: doh .. $70 at voxilla includes free activation at bv and a month free |
| 16:32.03 | _-Jon-_ | No actually it was completely silent on the line |
| 16:32.09 | JohnAB | i had that earlier |
| 16:32.20 | JohnAB | oh yeah that was because i had the moh file ;ed ut |
| 16:32.23 | `Sauron | bjohnson: Their activation fee is $39.95, $15 for S&H |
| 16:32.38 | _-Jon-_ | Hmmm |
| 16:32.49 | _-Jon-_ | JohnAB, try 0.59r though, see if that helps |
| 16:33.04 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 16:33.13 | *** part/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 16:33.24 | bjohnson | <`Sauron> bjohnson: I ended up with $76.99 in activation ?? |
| 16:33.29 | JohnAB | yep just compiling it |
| 16:33.47 | _-Jon-_ | Hey has anyone in here had problems with BroadVoice billing them recently? |
| 16:34.37 | nestAr | anyone got a Polycom registering to two different * systems via boot server? I cant' seem to get my 2nd line to configure right.. |
| 16:35.10 | pointer-gaim | nestAr: yes |
| 16:35.23 | JohnAB | works sweet with 0.59r |
| 16:35.29 | pointer-gaim | line = home |
| 16:35.32 | pointer-gaim | line2 == work |
| 16:35.35 | JohnAB | damn the default moh mp3 is comedy |
| 16:35.46 | JohnAB | remind me to change it before anyone calls |
| 16:35.49 | nestAr | pointer-gaim: yeah, that's what i'm trying to do.. mind sharing config example? |
| 16:35.54 | _-Jon-_ | JohnAB, it came like that? |
| 16:36.05 | pointer-gaim | nestAr: can you email me? on a conference call right now |
| 16:36.18 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 16:36.35 | JohnAB | yeah supermegasprokets inc or something |
| 16:36.46 | nestAr | pointer-gaim: sure, thanks! |
| 16:36.55 | _-Jon-_ | Hmm my defautl was classical i believe |
| 16:39.43 | *** join/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 16:39.56 | pointer-gaim | anyone care to recommend any of the VoIP monitoring tools listed in the wiki -> http://www.voip-info.org/wiki-How+to+debug+and+troubleshoot+VoIP |
| 16:40.09 | pointer-gaim | mainly to get metrics on call quality/jitter and the like |
| 16:43.20 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 16:44.16 | tull | sorry guys, is this book worth it?? "VoIP Telephony with Asterisk by Paul Mahler" |
| 16:45.51 | Zeeek | wasim I need my daily fix of fantasy |
| 16:45.57 | Zeeek | and a stiff drink |
| 16:47.12 | *** part/#asterisk schurig (~schurig@p5080BCB9.dip0.t-ipconnect.de) |
| 16:48.52 | labo | tull, ask zoa |
| 16:48.58 | labo | And don't buy the softcopy one. |
| 16:49.20 | *** join/#asterisk Xanathar (~Xanathar@pcp0010782304pcs.walngs01.pa.comcast.net) |
| 16:49.25 | *** join/#asterisk coppice (~chatzilla@205.162.17.210.dyn.pacific.net.hk) |
| 16:49.40 | tull | labo th |
| 16:49.43 | tull | labo thx |
| 16:49.45 | gambolputty | my IAXy has a recessed button on the front, what is it for? |
| 16:49.57 | Zeeek | nothing at all |
| 16:50.06 | gambolputty | why is it there then? |
| 16:50.12 | Zeeek | for sexy looks |
| 16:50.14 | Qwell | So people ask why its there |
| 16:50.16 | coppice | self destruct - that's why its recessed for safety |
| 16:50.19 | Zeeek | exactly! |
| 16:50.27 | Zeeek | betterer yet |
| 16:50.36 | ManxPower | The iaxy button is cosmetic. |
| 16:50.43 | Zeeek | too small to receive what you tried to put in it |
| 16:50.53 | Qwell | Zeeek: or is it? |
| 16:50.57 | Zeeek | actually it's for ventilation as they run a little hot |
| 16:51.34 | Zeeek | is anyone running an IAXy on a fixed ip? |
| 16:51.42 | Zeeek | I've never managed to get it working that way |
| 16:52.05 | Dalion | jbot: "VoIP Telephony with Asterisk by Paul Mahler |
| 16:52.18 | Dalion | ~jbot search "VoIP Telephony with Asterisk by Paul Mahler" |
| 16:52.19 | Zeeek | here comes the 's' word |
| 16:52.45 | coppice | sushi? |
| 16:52.57 | `Sauron | SUCKS |
| 16:53.00 | Uther_P | sunshine? |
| 16:53.10 | Zeeek | ever see that ad about the guy who goes out to eat sushi with his pals? |
| 16:53.21 | mutilator | anyone know if there is a daemon made to monitor multiple processes and restart them if they die? |
| 16:53.34 | coppice | init |
| 16:53.36 | Zeeek | he comes home to his wife, kisses her and she slaps him hard and goes out slamming the door |
| 16:53.38 | `Sauron | mutilator: IT's called "init" |
| 16:53.45 | mutilator | besides init? |
| 16:53.54 | Zeeek | the voiceover is "make sure your sushi is fresh!" |
| 16:53.58 | Qwell | Zeeek: Ever see the Sizzler commercial for shrimp cocktail? heh |
| 16:53.58 | coppice | xinit |
| 16:54.02 | mutilator | heh |
| 16:54.09 | Zeeek | This ad is a killer |
| 16:54.16 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 16:54.19 | `Sauron | mutilator: What's wrong with using init? |
| 16:54.22 | Zeeek | I think it's viral like the trunk monkey |
| 16:54.40 | shido6 | whats wrong with your iaxy ? |
| 16:54.43 | Zeeek | you guys all know about trunk monkeys, right? |
| 16:54.48 | Zeeek | http://www.trunkmonkey.com/ |
| 16:54.48 | shido6 | Zeeek what is wrong with your Iaxy? |
| 16:54.53 | *** join/#asterisk mrgoby (~mrgoby@pcp05308058pcs.wanarb01.mi.comcast.net) |
| 16:54.57 | mutilator | init doesn't constantly run? |
| 16:55.03 | Zeeek | nothing but I've never been able to set it to a fixed ip |
| 16:55.03 | `Sauron | yes it does |
| 16:55.06 | Qwell | Zeeek: There was a Sizzler commercial recently for shrimp cocktail(bad pun), and there is a shrimp with its tail in front, and a guy, and they're on a beach, classic love story, where they run to each other |
| 16:55.13 | *** part/#asterisk kwaldo_aloft (~waldo@dsl7.rbh1.pppoe.execulink.com) |
| 16:55.14 | Qwell | I saw that commercial like 5 times, laughed every time... |
| 16:55.17 | Uther_P | heh, init is the first thing to run, and the last thing to die |
| 16:55.19 | shido6 | Zeeek are you familiar with how to provision them? |
| 16:55.20 | mrgoby | hi all, having a problem with hdparm... i know this isnt asterisk specific, but i'm running asterisk on the machine , so... :-) |
| 16:55.23 | Qwell | then, they remade the commercial, and the tail is in the back now |
| 16:55.30 | mutilator | then how do i make it respawn my processes? |
| 16:55.31 | shido6 | You just put the static ip in the iaxy.conf |
| 16:55.32 | Zeeek | shido6 sure I do it all de time |
| 16:55.38 | shido6 | then what is the problem? :) |
| 16:55.38 | mrgoby | it is a SATA drive... |
| 16:55.42 | bjohnson | mutilator: inittab |
| 16:55.47 | shido6 | what does your iax.conf say? |
| 16:55.49 | Zeeek | it then goes dead |
| 16:55.51 | shido6 | pastebin it |
| 16:55.57 | mrgoby | intel 600ESB IDE controller |
| 16:55.59 | shido6 | provision, wait 5 seconds |
| 16:56.02 | shido6 | unplug power |
| 16:56.03 | bjohnson | mutilator: err .. /etc/xinitd.d |
| 16:56.04 | shido6 | plug it back in |
| 16:56.12 | shido6 | you're done if you set your iax.conf correctly |
| 16:56.13 | Zeeek | don't have iot handy but it says ip:myu ip, netmaks 255.255.255.0 |
| 16:56.15 | mutilator | um |
| 16:56.18 | `Sauron | bjohnson: /etc/inittab |
| 16:56.21 | `Sauron | not inetd |
| 16:56.25 | mutilator | ?? |
| 16:56.57 | Zeeek | shido6 what is heartbeat? |
| 16:57.04 | shido6 | u have to set ip, netmask, and gateway |
| 16:57.16 | Zeeek | hmmm gateway, maybe not |
| 16:57.19 | mrgoby | since SATA is based on scsi, can you even use DMA or UDMA ? |
| 16:57.20 | Zeeek | lemmie see |
| 16:57.22 | shido6 | never used heartbeat unless I was debugging |
| 16:57.54 | mrgoby | this whole thing is confusing... why have a scsi disk connected to an IDE bus ? |
| 16:57.57 | mrgoby | or controller |
| 16:57.59 | mrgoby | ? |
| 16:58.25 | mutilator | k what am i supposed to do with my inittab |
| 16:58.26 | Zeeek | shido6 there is currently no gateway in the file but DHCP works fine |
| 17:00.04 | Uther_P | mutilator: what application are you refering to to restart if it dies? |
| 17:00.28 | Zeeek | shido6 is gateway necessary with a fixed ip ? |
| 17:00.48 | mutilator | asterisk, or an ircd, or a prog of mine, or any number of things... |
| 17:00.56 | Uther_P | Zeeek: a voip gateway or a network gateway? |
| 17:01.11 | Uther_P | mutilator: well, asterisk does this itself if you run it with safe_asterisk |
| 17:01.14 | ManxPower | Sometimes I love my provider. I requested 2 additonal B-channels and he e-mails me back saying he can add them today if it's important. |
| 17:01.15 | `Sauron | mutilator: read man 5 inittab |
| 17:01.15 | Zeeek | talking about IAXy Uther |
| 17:01.25 | ManxPower | same day service |
| 17:01.34 | Qwell | ManxPower: "No, its not that important, it can wait." |
| 17:01.34 | Zeeek | yuo are a lucky boy Manx! |
| 17:01.45 | *** join/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 17:01.58 | *** part/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 17:02.00 | mutilator | Uther_P: was an example |
| 17:02.06 | ManxPower | Qwell, Actually I said that it's not critical. If not today then on Wed (when everyone on New Orleans returns to work) |
| 17:02.14 | Slainte | I have a security problem. When people call the autoattendent, if they enter 9 and the number they can make an outgoing call. Can someone help me organise my context to prevent this? |
| 17:02.16 | Qwell | ahh, right, Marti Gras? |
| 17:02.25 | ManxPower | It's Carnival time. Nobody will be at work on mon or tues |
| 17:02.48 | coppice | Ah, Wednesday. Holiday time |
| 17:02.53 | eKo1 | Rio here I come. |
| 17:02.54 | Zeeek | Slainte make several contexts like inhouse, privileged and outside |
| 17:02.55 | ManxPower | Slainte, Common problem. You didn't split up your contexts good. See the examples on the Wiki |
| 17:03.02 | Xanathar | Could miscabling result in an E1 being Active (no alarms) but the D Channel reporting as down ? |
| 17:03.05 | Zeeek | or |
| 17:03.06 | Zeeek | Starter tutorial: |
| 17:03.06 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 17:03.06 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 17:03.06 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 17:03.06 | Zeeek | THE reference of the moment: |
| 17:03.07 | Zeeek | http://www.asteriskdocs.org |
| 17:03.16 | ManxPower | ~docs |
| 17:03.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:03.17 | _-Jon-_ | Slainte, what's the magic #? :) |
| 17:03.17 | ManxPower | ~doc |
| 17:03.20 | Zeeek | subject i discussed at length |
| 17:03.34 | Uther_P | mutilator: I would just write a script to monitor the pid through a pid file, and restart it if the process is no longer active |
| 17:03.47 | Uther_P | mutilator: much like safe_asterisk does |
| 17:04.02 | Zeeek | I ate all the bread |
| 17:04.06 | mutilator | thats what i have now |
| 17:04.27 | Slainte | Manx, I have read in great detail, but obviously am too slow or dont know how to read |
| 17:04.28 | mutilator | i was wondering if there was somethin made that i could just add a process name into it and it'de just keep track itself |
| 17:04.35 | Uther_P | mutilator: well... take it one step further and write the script to monitor a like of pids and appilication |
| 17:04.49 | Uther_P | er, a list of pids |
| 17:04.50 | Slainte | I ahve split it all up, but ruin it all by my include statements |
| 17:04.50 | eKo1 | safe_asterisk is BS. Just start asterisk without any arguments. |
| 17:05.02 | mrgoby | eKo1 ? |
| 17:05.02 | silik0n | Comming soon to a store near you BKWs BOB |
| 17:05.09 | eKo1 | and have a mon script monitor the pid. |
| 17:05.15 | mrgoby | safe_asterisk is the way to go |
| 17:05.19 | Slainte | http://www.voip-info.org/wiki-Asterisk+security+dialplan tells me nothing |
| 17:05.23 | eKo1 | safe_asterisk sucks. |
| 17:05.23 | Uther_P | eKo1: thats what safe_asterisk does |
| 17:05.46 | *** join/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 17:05.49 | mrgoby | mmmkay |
| 17:06.49 | marc1 | does any one know's where I can find a quick guide h323 install ? |
| 17:07.07 | marc1 | for asterisk |
| 17:07.13 | Zeeek | IS THERE ANYTHING, ANYTHING AT ALL that DOESN'T SUCK? |
| 17:07.18 | mrgoby | marc1, first you are in a world of hurt |
| 17:07.22 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 17:07.24 | Qwell | Zeeek: as far as? |
| 17:07.26 | shido6 | Zeeek what is wrong ?!!? |
| 17:07.29 | mrgoby | second, check the oh323 homepage |
| 17:07.44 | Zeeek | well, the words sucks keeps coming up, I was just wondering |
| 17:07.47 | mrgoby | and the docs in /usr/src/asterisk/modules/h323 |
| 17:07.49 | coppice | we just got a new vaccum cleaner, because the old one no longer sucks |
| 17:07.55 | shido6 | Zeeek did u set your gateway in your iaxy? |
| 17:08.01 | *** join/#asterisk olivier_ (~olivier_@AMarseille-152-1-1-32.w82-127.abo.wanadoo.fr) |
| 17:08.07 | mrgoby | or, maybe that was in ~/docs/README.h323 |
| 17:08.12 | Zeeek | no I'm not experimenting at thre moment, but I did just call someone on it |
| 17:08.12 | mrgoby | cant remember |
| 17:08.20 | mrgoby | use nufone's implementation |
| 17:08.31 | mrgoby | but it will still be awful :) |
| 17:08.37 | Zeeek | gateway being the local router, yes? |
| 17:08.45 | mrgoby | cause h323 is evil |
| 17:08.48 | Zeeek | because there is no gateway in the current file |
| 17:08.55 | *** join/#asterisk freat[laptop] (~freat[lap@65.170.62.117) |
| 17:08.55 | shido6 | ok |
| 17:08.59 | shido6 | so your iaxy works now? |
| 17:09.05 | gambolputty | use the U and G options in * to start it safely |
| 17:09.07 | *** join/#asterisk ragnar (~ragnar@30.Red-80-36-33.pooles.rima-tde.net) |
| 17:09.08 | Zeeek | it always has shido6 |
| 17:09.12 | Zeeek | except when it didn't |
| 17:09.19 | Zeeek | but even then it did NOT suck |
| 17:09.22 | ragnar | hi.. i got a little isdn question.. |
| 17:09.33 | Zeeek | IDSN doesn't suck |
| 17:09.42 | ragnar | oh, just arrived in time it seems :) |
| 17:09.44 | Uther_P | haha |
| 17:09.46 | *** part/#asterisk opticalcarrier (~bo@optical.neteng.earthlink.net) |
| 17:09.47 | Uther_P | yea |
| 17:09.48 | Uther_P | you did |
| 17:09.56 | coppice | Zeek: your spelling of it does |
| 17:09.59 | Zeeek | arriving late SUCKS! |
| 17:10.02 | Uther_P | haha |
| 17:10.14 | Zeeek | I Dont Spell Names |
| 17:10.17 | Uther_P | what was refered to as sucking? iaxy? |
| 17:10.20 | Zeeek | IDSN |
| 17:10.26 | Uther_P | then spell it out Zeeek |
| 17:10.27 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 17:10.28 | Qwell | Uther_P: pretty much everything |
| 17:10.30 | Uther_P | what does it mean |
| 17:10.31 | PakiPenguin | hello everyone |
| 17:10.33 | Zeeek | everything sucks, all providers except Manxpowers' |
| 17:10.48 | Zeeek | every linux distro |
| 17:10.55 | Uther_P | heh |
| 17:10.56 | Zeeek | all politicians |
| 17:11.01 | Uther_P | I'm a FreeBSD fan myself |
| 17:11.06 | Qwell | freebsd sucks too |
| 17:11.10 | zno | me too bud asterisk on freebsd sucks |
| 17:11.12 | Uther_P | blow me |
| 17:11.13 | Zeeek | nah, it doesn't suck |
| 17:11.17 | zno | i had to switch to debian |
| 17:11.21 | Zeeek | I do,n't suck, so, no |
| 17:11.30 | Zeeek | all together... |
| 17:11.35 | Zeeek | ****** sucks! |
| 17:11.42 | ragnar | i got a isdn line with ispbx, which supports 4 voice lines.. but i cant find an isdn card for asterisk that doesnt cost an arm.. is there some adapter to convert the isdn line to 4 analog lines, so i can connect to 4 pstn fxo modules? |
| 17:11.54 | PakiPenguin | i need * to accept all incoming calls in the format 00XXXXXX in a context,any pointers? |
| 17:12.03 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 17:12.04 | *** join/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 17:12.11 | *** join/#asterisk Samoied (~samoied@200.247.141.111) |
| 17:12.14 | Zeeek | Paki^^^^^^^^^^Penguin |
| 17:12.28 | Zeeek | ooops wrong one |
| 17:12.49 | Zeeek | Pattern Matching, Making use of the ${EXTEN} channel variable, Linking Contexts with Includes, Some Other Special Extensions http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN862 |
| 17:12.53 | Zeeek | Paki^^^^^^^^^^Penguin |
| 17:12.57 | Samoied | How I use asterisk for contact a sip server as endpoint |
| 17:13.18 | ragnar | i think i saw how to do that on the wiki, Samoied |
| 17:13.22 | Samoied | Because my ITSP not permit Proxy connections |
| 17:13.31 | Samoied | ragnar: where? please |
| 17:13.36 | ragnar | i dont remember |
| 17:13.41 | Flyboy6440 | is there a way to detect if the callerid is an actual true #? been getting calls where the # is 0000000000 or *-***-***-**** |
| 17:13.46 | ragnar | i was playing with sip last weekend |
| 17:13.52 | Slainte | Manx, Ok I got my head around it now. It certainly is not the most efficient. I have two contexts. one for internal-ext, and one internal-ext-safe. the internal-ext-safe does not have an include for the outgoing, however it is an exact copy of the internal-ext. Is there a better way to do it? |
| 17:13.52 | WildPikachu[BAR] | hrmmm |
| 17:13.53 | Qwell | ~google sip asterisk site:voip-info.org |
| 17:13.57 | WildPikachu[BAR] | what does the US48 term mean? |
| 17:14.00 | ragnar | Flyboy6440: don think so |
| 17:14.10 | Qwell | WildPikachu: the lower 48 states |
| 17:14.11 | blitzrage | does LOG_VERBOSE actually do anything? |
| 17:14.15 | Qwell | excluding Hawaii and Alaska |
| 17:14.19 | Uther_P | us48? umm the contiguous united states? |
| 17:14.21 | WildPikachu[BAR] | aha |
| 17:14.33 | Flyboy6440 | bummer.. the 0's are coming from the pstn the *'s are coming from voicepulse |
| 17:14.35 | WildPikachu[BAR] | so it excludes 4 states? |
| 17:14.38 | *** join/#asterisk stevekstevek_ (~chatzilla@h-68-164-202-153.nycmny83.dynamic.covad.net) |
| 17:14.39 | Qwell | no, just 2 |
| 17:14.40 | Uther_P | umm no |
| 17:14.42 | Uther_P | it exclues 2 |
| 17:14.43 | drray | hah |
| 17:14.43 | Qwell | There are 50 states in the US now :p |
| 17:14.46 | Uther_P | there are 50 states |
| 17:14.51 | drray | canada is not a state yet |
| 17:14.53 | Uther_P | haha |
| 17:14.54 | Uther_P | yet |
| 17:14.58 | drray | yet |
| 17:15.01 | drray | I said it |
| 17:15.03 | Uther_P | :D |
| 17:15.11 | Zeeek | new york is a state of mind |
| 17:15.13 | Flyboy6440 | maybe an agi that uses regular expressions would do the trick |
| 17:15.14 | WildPikachu[BAR] | \ |
| 17:15.14 | Uther_P | DM is owned by us, but is not a state |
| 17:15.22 | Uther_P | what about Woodstock Nation? |
| 17:15.23 | WildPikachu[BAR] | ' |
| 17:15.24 | WildPikachu[BAR] | <PROTECTED> |
| 17:15.25 | Qwell | DM? Denmark? |
| 17:15.26 | Uther_P | thats a state of mind :P |
| 17:15.33 | Uther_P | Dominican Republic |
| 17:15.36 | Qwell | oh, heh |
| 17:15.36 | WildPikachu[BAR] | sorry |
| 17:15.56 | Zeeek | what BAR? Can I get a drink? |
| 17:16.11 | WildPikachu[BAR] | wife spilt wine on my laptop |
| 17:16.12 | drray | It blew me away when I saw that DM and Hati were the same island |
| 17:16.15 | Uther_P | heh, in a bar on a PDA? |
| 17:16.20 | BoRiS | Wild: No hawaii or alaska |
| 17:16.21 | Zeeek | hanging offense, that |
| 17:16.25 | BoRiS | ooops :-p |
| 17:16.36 | drray | contiguos being touching |
| 17:16.37 | Uther_P | heh boris, I think we covered that :P |
| 17:16.57 | BoRiS | Uther: Sorry, I was still reading from my scroll buffer |
| 17:16.59 | BoRiS | :-p |
| 17:17.03 | Uther_P | hehe |
| 17:17.32 | Uther_P | i'm outta here |
| 17:17.34 | Uther_P | l8rzzz |
| 17:17.58 | Zeeek | ~lart me |
| 17:18.35 | Slainte | It works, but it is messy now :) |
| 17:19.10 | Zeeek | there is no such thing as a non-messy dialplan, only meek anal retentive newbie ones |
| 17:19.47 | Zeeek | I've never seen a published dialplan example that didn't have twently lines saying :;;; must clean this up some day |
| 17:19.54 | Qwell | heh |
| 17:20.08 | Zeeek | I fixed a bug in a prog today |
| 17:20.19 | Qwell | ; 01/24/97 TODO: Fixme |
| 17:20.25 | *** join/#asterisk hmmhesays (~hmmhesays@66.173.103.108) |
| 17:20.27 | Zeeek | then added // it's either this or change the table in the database and fifty other scripts |
| 17:20.45 | Zeeek | <PROTECTED> |
| 17:21.02 | Zeeek | the classic goes back to assembler though: |
| 17:21.19 | Zeeek | 1000 ad a,10 ; add 10 to a |
| 17:21.22 | hmmhesays | callerid="name" <number> in the sip.conf is the right syntax right? |
| 17:21.32 | Qwell | Zeeek: cute |
| 17:21.34 | *** join/#asterisk stevekstevek_ (~chatzilla@h-68-164-202-153.nycmny83.dynamic.covad.net) |
| 17:21.51 | Qwell | Zeeek: I love my "//this is why Nitu doesn't work here anymore" comment |
| 17:21.54 | BoRiS | covad .. ack |
| 17:21.56 | Zeeek | HAHA |
| 17:22.06 | Zeeek | you gotta leave that |
| 17:22.12 | Zeeek | "for historical reasons" |
| 17:22.33 | junky[work] | hmmhesays: no callerid=name |
| 17:22.35 | Zeeek | <PROTECTED> |
| 17:22.50 | Zeeek | hystercal? |
| 17:23.32 | Zeeek | callerid="Bernie in Sales" <3009> |
| 17:23.37 | hmmhesays | callerid=name <number> ? |
| 17:23.49 | Zeeek | nope as above you were right |
| 17:24.30 | Qwell | Zeeek: Its been there for 2 years already. :) |
| 17:24.44 | hmmhesays | I can't get the name to show up only the number From: "5555555" sip:5555555@username |
| 17:24.54 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 17:24.55 | Zeeek | on what? |
| 17:25.12 | Zeeek | the name I mean |
| 17:25.45 | hmmhesays | I got callerid set callerid="Homer" <5555555> and the second invite comes out as stated above |
| 17:25.58 | ManxPower | Zeeek, Please stop telling people to put quotes in their callerid |
| 17:26.03 | jskcr | hy all |
| 17:26.05 | jskcr | :P |
| 17:26.11 | Zeeek | that's the way mine are and always have been |
| 17:26.13 | hmmhesays | heh, that's how the example is in sip.conf |
| 17:26.17 | Zeeek | yes |
| 17:26.21 | ManxPower | Zeeek, Yes, but some phones can't deal with it. |
| 17:26.23 | BoRiS | I have always used quotes |
| 17:26.28 | Zeeek | so if that's wrong, someone should get busy and change the 20 examples |
| 17:26.44 | hmmhesays | i'm not even going to touch that one |
| 17:26.45 | hmmhesays | lol |
| 17:26.48 | Zeeek | mine doesn't do alpha anyway :) |
| 17:27.01 | BoRiS | Zeek: budgetone? :-p |
| 17:27.07 | Zeeek | yahoo sir |
| 17:27.11 | BoRiS | hehe |
| 17:27.11 | ManxPower | Specifically some Cisco phones will totally reject calls with quotes in the CLID Name. |
| 17:27.11 | Zeeek | ya - hoo |
| 17:27.12 | BoRiS | thought so |
| 17:27.25 | Zeeek | I wouldn't talk to anyone who has a cisco phone anyway |
| 17:27.40 | Zeeek | they s**k |
| 17:27.42 | hmmhesays | mine does.... but the callerid line doesn't set the sip header correctly |
| 17:27.51 | gambolputty | zeeek: what ip phones do you use? |
| 17:28.01 | Zeeek | only the best - BT100s |
| 17:28.28 | hmmhesays | it sets the name field and the number field to whatever number you have |
| 17:28.28 | darkskiez | i hate callerid |
| 17:28.29 | Zeeek | [this is irony folks - nothing sucks or doesn't that's my point] |
| 17:28.47 | Flyboy6440 | hehe |
| 17:28.54 | Zeeek | well, military draft sucks I s'pose and a few other things |
| 17:28.57 | darkskiez | I want it to work on dialled numbers too |
| 17:28.59 | hmmhesays | so callerid is broken it seems |
| 17:29.08 | Zeeek | 1.0.5? |
| 17:29.14 | Zeeek | there are a few issues apparently |
| 17:29.33 | ManxPower | Yeah. I'm waiting for 1.0.6 before I upgrade my production systems |
| 17:29.49 | Zeeek | Boy tell me about it |
| 17:29.50 | hmmhesays | this is just on a testing machine... it's cvs head from 12/17 |
| 17:30.06 | Zeeek | there was a time when head was really stable |
| 17:30.11 | darkskiez | Is there a schedule for the next major release of asterisk, or is it just when its ready. |
| 17:30.52 | hmmhesays | usually adding features causes instability |
| 17:30.57 | Zeeek | this is where Manx jumps in with... |
| 17:31.09 | hmmhesays | until the bugs are squashed |
| 17:31.10 | Zeeek | ?mailing list? |
| 17:31.28 | darkskiez | I see |
| 17:31.57 | Zeeek | Manx but maybe that is what my problem is with the Siemens phone CID.... |
| 17:32.05 | Zeeek | so thanks for mentioning it |
| 17:32.44 | hmmhesays | setcallerid(Homer <5555555>) in extensions.conf works though |
| 17:32.53 | mAsH` | can i connect * to a PABX by an HFC card? |
| 17:34.16 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 17:34.22 | Zeeek | ok, so this doesn't suck, but it licks |
| 17:34.24 | Zeeek | http://screenclean.j1media.com/lick.html |
| 17:34.24 | ManxPower | They are adding the new B-Channels at 2pm local time. |
| 17:34.32 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 17:34.50 | *** join/#asterisk jarrod (jarrod@dipole.informationwave.net) |
| 17:35.47 | ManxPower | Zeeek, That's the first actual cool multi-media link I've seen in YEARS |
| 17:35.57 | Zeeek | Cat lovers love it |
| 17:36.15 | hmmhesays | I dunno that bruce lee animation i watched yesterday was pretty cool |
| 17:36.25 | xkev | if I 'asterisk -r' and change verbosity, does that change it globally or just for that remote session? |
| 17:36.26 | `Sauron | La di dum |
| 17:36.30 | `Sauron | I love being on hold |
| 17:37.24 | Zeeek | some of these commercials are funny if you have seen them |
| 17:37.25 | Zeeek | http://www.trunkmonkey.com/ |
| 17:37.54 | Flyboy6440 | xkev: i've noticed it stays at the highest verbosity that you last remoted in with |
| 17:37.55 | terrapen | anybody want to help me nail a phisher scammer? |
| 17:38.28 | terrapen | i'm doing some phisher vigilante-ism this morning |
| 17:38.44 | Zeeek | ManxPower does "who sees us as 192.168.1.5" mean what I think it means? |
| 17:38.57 | Zeeek | "what do you think it means?" |
| 17:39.06 | terrapen | hahah, i think i just killed their server |
| 17:39.08 | terrapen | i did. |
| 17:39.16 | terrapen | http://64.65.250.200:87/s/ |
| 17:39.20 | terrapen | can anyone reach that? |
| 17:39.28 | coppice | terrapen: which part of the scammer will the nails go through? |
| 17:39.38 | Flyboy6440 | yes |
| 17:39.59 | terrapen | http://64.65.250.200:87/s/submit.php?username=YourPHPsucks&password=yourMotherHasAPenis&submit=Submit |
| 17:40.05 | *** join/#asterisk letherglov (~letherglo@8036aa5e.resnet.ucsd.edu) |
| 17:40.05 | terrapen | put that one into http_load |
| 17:40.22 | terrapen | http_load -seconds 100000 -parallel 20 phisher_url.tt |
| 17:40.24 | terrapen | err |
| 17:40.25 | terrapen | http_load -seconds 100000 -parallel 20 phisher_url.txt |
| 17:40.31 | terrapen | coppice: the meaty parts. |
| 17:40.57 | terrapen | try that url now |
| 17:41.01 | terrapen | i think i just cooked his server |
| 17:41.06 | terrapen | probably filled his logs |
| 17:41.11 | Zeeek | or they programmed the firewall? |
| 17:41.17 | Slainte | http://pastebin.ca/5281 I am trying to have the default operator ask for an extension three times and then go to the 0 extension if there are three invalids. Its not working. Any comments on my .conf segment? |
| 17:41.21 | terrapen | or else he was watching it and just tooked his script down for maintenance |
| 17:41.30 | terrapen | zeek, probably |
| 17:41.35 | terrapen | can anyone else get to his site |
| 17:41.37 | Qwell | "Access to the port number given has been disabled for security reasons." |
| 17:41.38 | Qwell | heh |
| 17:41.41 | terrapen | http://64.65.250.200:87/s/ |
| 17:41.45 | terrapen | that's firefox, qwell |
| 17:41.54 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:41.58 | terrapen | IE/Safari do not seem to mind |
| 17:42.02 | Flyboy6440 | opening page.. |
| 17:42.13 | Flyboy6440 | but just locks there |
| 17:42.17 | terrapen | heh |
| 17:42.30 | terrapen | feel free to hit him with http_load |
| 17:42.43 | terrapen | http://www.acme.com/software/http_load/ |
| 17:42.50 | Zeeek | Slainte you need a loop |
| 17:43.15 | terrapen | a better solution would be a POE app that generated random garbage that is not so easy to filter |
| 17:43.51 | terrapen | dammit, now i'm firewalled, it seems |
| 17:44.06 | Slainte | Zeek, I cant find a loop example. can you give me a snippet to search with? |
| 17:45.02 | marc1 | who can I get the nufone h323 ? |
| 17:45.04 | Zeeek | ya just a sec I have an old onbe |
| 17:46.16 | Slainte | thanks |
| 17:47.40 | jarrod | anyone deployed ser with rtpproxy? |
| 17:48.01 | Zeeek | Slainte here's an example of a loop : http://pastebin.ca/5283 |
| 17:48.52 | Slainte | Thanks Zeeek appreciate it |
| 17:48.56 | Zeeek | np |
| 17:49.08 | Zeeek | I can't rememebr if it worked :) |
| 17:49.18 | Zeeek | just kidding |
| 17:49.21 | terrapen | heh, i guess the http flooding took him down |
| 17:49.34 | Zeeek | what scam? Paypal? |
| 17:49.37 | terrapen | either that or it was some 0wned box and the poor bastard sitting on console just rebooted |
| 17:49.45 | terrapen | SmithBarney phiser scam |
| 17:49.56 | Zeeek | you should send it to them too |
| 17:49.57 | terrapen | http://64.65.250.200:87/s/ |
| 17:50.08 | terrapen | i'm sure smithbarney knows |
| 17:50.18 | terrapen | i called the ISP and they didn't seem to care too much |
| 17:50.24 | Zeeek | they rarely do! |
| 17:50.26 | *** join/#asterisk abbas_ (nidobas@203.81.222.19) |
| 17:50.35 | terrapen | nope |
| 17:50.42 | Zeeek | they get GWF fatigue, too |
| 17:50.52 | Qwell | Tell them there's copyrighted material on there, and it'll get taken down quick, heh |
| 17:51.03 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:52.10 | Samoied | I dont find anything about use asterisk as sip endpoint |
| 17:52.20 | Samoied | anyone helpme ? |
| 17:52.50 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:52.52 | Samoied | My ITSP accept only www-authentication not proxy-authentication |
| 17:52.58 | ragnar | its there, im sure its there |
| 17:53.34 | Samoied | ragnar: in voip-info.org ? |
| 17:53.44 | terrapen | i love getting phisher scams |
| 17:53.46 | ragnar | yep |
| 17:53.47 | terrapen | i take em all down |
| 17:53.54 | ragnar | found it from there |
| 17:54.01 | *** join/#asterisk mindCrime (~mindCrime@bi01p1.nc.us.ibm.com) |
| 17:54.17 | terrapen | maybe i'll get around to writing my POE anti-phisher flood client |
| 17:54.28 | Zeeek | that would be great! |
| 17:56.56 | Zeeek | who here has written asterisk module app_ |
| 17:57.01 | Zeeek | ? |
| 17:58.14 | Zeeek | I haven't found where the functions are that handle passed variables? I've been able to mimic existing source obviously, but didn't find where those functions are kept |
| 17:58.19 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
| 17:58.35 | Flyboy6440 | have not wrote a module yet.. working on an agi though :) |
| 17:59.02 | Zeeek | I guess the modules need to be re-entrant which means you have to be careful |
| 18:00.23 | Slainte | what does this mean? ast_readaudio_callback: Failed to write frame |
| 18:00.35 | anti | terrapen: POE anti-phisher flood client? |
| 18:00.41 | Zeeek | drums stop. not good. |
| 18:01.20 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:01.23 | Samoied | ragnar: fount this: http://www.voip-info.org/tiki-index.php?page=SIP%20Authentication |
| 18:02.39 | *** join/#asterisk fizbar (~jason@dsl-232-31.speedsite.com) |
| 18:03.20 | Flyboy6440 | how does privacy manager detect if callerid was received.. just checkes for null? |
| 18:03.53 | hmmhesays | anyone have any problems with asterisk on fc3? |
| 18:04.16 | Slainte | Zeek. WARNING ast_expr.y:475 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: +1 |
| 18:04.24 | junky[work] | hmmhesays: http://bugs.digium.com/bug_view_page.php?bug_id=3507 ? |
| 18:05.04 | hmmhesays | nod |
| 18:05.49 | Samoied | ragnar: but not explain howto use www-authentication in asterisk |
| 18:06.17 | hmmhesays | hmmm |
| 18:06.42 | *** join/#asterisk coppice (~chatzilla@205.162.17.210.dyn.pacific.net.hk) |
| 18:11.37 | bannerman | Is there anything wrong with the Grandstream Budgetone 101 phones? |
| 18:12.03 | dolson | define wrong |
| 18:12.17 | bannerman | Am I going to regret having that phone instead of one $20 more? |
| 18:12.31 | bannerman | $65.95 is an awfully good price |
| 18:12.39 | *** join/#asterisk carbon60 (~adam@gw.techsupport.ca) |
| 18:12.39 | dolson | what's the other option? |
| 18:12.41 | hmmhesays | wrong with? |
| 18:12.50 | hmmhesays | <shrug> they work.. kind of ugly though |
| 18:12.55 | bannerman | I haven't picked anything specific out |
| 18:12.57 | bannerman | they are sort of ugly |
| 18:13.05 | bannerman | are they missing any features? |
| 18:13.08 | bannerman | do they sound bad? |
| 18:13.10 | dolson | we have 5 of the BudgeTone phones here |
| 18:13.44 | carbon60 | For someone looking for 400 FXS ports, what hardware would be appropriate? |
| 18:13.48 | dolson | I'm having a problem with them getting DHCP addresses and not picking up the TFTP server from the DHCP options |
| 18:13.58 | dolson | other than that, they seem alright |
| 18:14.11 | bannerman | That doesn't sound fun to deal with.. |
| 18:14.18 | bannerman | Is there another phone that you would reccommend? |
| 18:14.23 | bannerman | recommend |
| 18:14.27 | dolson | well, it's not too much of an issue, really |
| 18:15.00 | dolson | as far as I can tell, they only use TFTP for firmware upgrading, so we can just manually do that with the web interface |
| 18:15.20 | bannerman | I have some true noob questions, too -- is it difficult to have like, a voicemail light flash, things like that? |
| 18:15.25 | dolson | to be honest, I wouldn't mind a BudgeTone for at home |
| 18:16.14 | hmmhesays | i'd rather have a small fxs unit with a cordless phone personally |
| 18:16.15 | bannerman | Or more to the point, do I need to research particular phone for features like that, or are they all pretty much the same? |
| 18:16.15 | dolson | that's a good question. I don't remember trying that out, to be honest. I mainly use an Aastra 480i on my desk, so I focus on that, but that's about $290 or so |
| 18:17.03 | hmmhesays | how much do you want to spend? |
| 18:17.06 | dolson | hmmhesays is right, in my opinion. I have that at home, and just got my TDM card today, so soon I'll be using a cordless phone, and I can't see it being anything short of awesome |
| 18:17.34 | bannerman | they'll be going into an office with like 14 cubes |
| 18:17.34 | hmmhesays | yeah having a sip phone like the budgetone is pretty pointless |
| 18:17.38 | bannerman | on a very tight budget ;-) |
| 18:17.52 | dolson | bannerman: ever consider softphones w/ USB headsets? |
| 18:17.54 | bannerman | pointless in what way? |
| 18:18.15 | hmmhesays | well the budgetone is the equivalent of a 10 dollar analog phone |
| 18:18.27 | tzanger | budgettones are actually not bad SOHO phones |
| 18:18.33 | tzanger | not *great* but not bad either |
| 18:18.40 | tzanger | certainly better than the shit that a Norstar 3x8 is |
| 18:18.55 | hmmhesays | in your situation they'll probably work ok though bannerman |
| 18:19.01 | bannerman | dolson: I have, and I wouldn't mind one for myself. Most of the folks in this office will require a plain desk phone, just for simplicity's sake |
| 18:19.25 | hmmhesays | if you're looking for idiot proof, budgettone will get you by |
| 18:19.35 | hmmhesays | big buttons, speakerphone sucks |
| 18:19.42 | hmmhesays | conference call doesn't work |
| 18:19.50 | bannerman | hmmhesays: I see your point, but it's cheaper to buy the budgettone phone than it is to convert to analog |
| 18:20.02 | *** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 18:20.05 | tzanger | use a meetme room for conf |
| 18:20.11 | hmmhesays | agreed |
| 18:20.20 | hmmhesays | Use FOP for your operator panel |
| 18:20.28 | hmmhesays | makes it really easy |
| 18:20.52 | bannerman | So the budgettone isn't going to leave me missing features that I would otherwise need in order to setup my pbx and stuff, it sounds like |
| 18:20.57 | hmmhesays | no |
| 18:20.58 | terrapen | woohoo my RAM has arrived |
| 18:21.19 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 18:21.20 | hmmhesays | with a little bit of cursing and getting angry it'll do what you need to |
| 18:21.29 | bannerman | thanks, it sounds like the budgettone will be a good entry phone.. if/when the boss and/or officers decide the want something nice, we can just upgrade |
| 18:21.47 | terrapen | bbl...rebooting |
| 18:22.06 | hmmhesays | do you have a secretary that answers the phones too? |
| 18:22.10 | bannerman | yeah |
| 18:22.13 | Flyboy6440 | maybe the iaxy device will come down to $50 that would be really nice |
| 18:22.17 | hmmhesays | you need any of that keysystem functionaly? |
| 18:22.24 | bannerman | maybe |
| 18:22.25 | redder86 | the only thing that I really don't like about the BudgeTones is 1) the speakerphone sucks for the person on the other end, 2) the lack of alpha-text on the display |
| 18:22.25 | ManxPower | bannerman, If Sipura can fix the volume problems on the SPA-841 it will be a MUCH better phone for only $20 more |
| 18:23.08 | bannerman | redder86: the display won't show alpha chars? |
| 18:23.15 | hmmhesays | negative |
| 18:23.15 | *** join/#asterisk gtigene_ (~chatzilla@208.239.206.195) |
| 18:23.16 | redder86 | bannerman: no |
| 18:23.25 | bannerman | hm, that's kinda lame |
| 18:23.26 | hmmhesays | some mighty big letters though, for your blind users |
| 18:23.33 | bannerman | hah |
| 18:23.43 | redder86 | bannerman: apparently on the new versions that are soon to come out they will 1) improve the speakerphone 2) give alphatext display |
| 18:23.45 | letherglov | hellen keller jokes for telephones |
| 18:23.50 | hmmhesays | letters=numbers oops |
| 18:23.58 | dolson | bannerman: you can configure the IP, codec, etc from the display though, just the letters are pretty wonky :) |
| 18:24.19 | redder86 | dolson: yeah, but no Caller*ID Name display at all. |
| 18:24.21 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 18:24.23 | dolson | yeah |
| 18:24.29 | heath__ | Anyone know the status of ChanSpy ? |
| 18:24.44 | hmmhesays | i thought they took that out |
| 18:25.00 | bannerman | ManxPower: What's the deal with volume problems on the SPA-841? |
| 18:25.15 | redder86 | the display on the BudgeTones is like a big, old-fashioned calculator screen |
| 18:25.28 | *** join/#asterisk cjk (~cjk@80.92.75.14) |
| 18:25.30 | heath__ | it's not available in the latest cvs, anyone know if there's plans to bring it back? |
| 18:25.36 | dolson | 01134 |
| 18:25.42 | hmmhesays | you can barge in with FOP |
| 18:25.42 | redder86 | ever try spelling out your name with numbers and then look at it upside down? Well, that's close to how it looks. |
| 18:25.45 | hmmhesays | if that's what you're looking for |
| 18:25.58 | ManxPower | bannerman, I don't know. Handset and speakerphone MICROPHONE seems to be very low. i.e. you have to shout for the far end to hear you. I could not find a gain setting for the microphone. Using a headset seems fine. |
| 18:26.16 | bannerman | ManxPower: Interesting. Other users have the same problem? |
| 18:26.36 | ManxPower | bannerman, I have two of them on two different systems, both experience the problem. |
| 18:26.47 | ManxPower | The phones were bought from two different distributers. |
| 18:29.01 | gtigene_ | Is anyone using the CVS Head version? |
| 18:29.50 | outtolunc | from yesterday, in the middle of updating to current |
| 18:30.09 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 18:30.28 | heath__ | is FOP hard on resources? |
| 18:30.34 | hmmhesays | no |
| 18:30.39 | Flyboy6440 | i'm using cvs head 1/14/05 :) guess i should update |
| 18:30.46 | hmmhesays | works fine on my testing machine |
| 18:31.10 | hmmhesays | 0.20 of FOP just rocks too |
| 18:31.11 | heath__ | final question: can you barge on all channels including sip? |
| 18:31.22 | hmmhesays | zap/sip/iax |
| 18:31.32 | hmmhesays | not sure on h323 or oh323 |
| 18:31.51 | Slainte | http://pastebin.ca/5286 Unavailable message is played but then it hangsup right away with this output |
| 18:31.59 | heath__ | i lied, one more: what if asterisk is already using monitor to record a channel, can you still barge on it? |
| 18:32.06 | hmmhesays | hell if I know |
| 18:32.52 | Slainte | I can leave a message from internal, but I cant leave one from the mainline. After it plays the unail message it exits the call |
| 18:32.53 | hmmhesays | you can ask on the FOP mailing list |
| 18:33.02 | ManxPower | Slainte, Did you forget to "make datafiles" in the asterisk source? |
| 18:33.04 | hmmhesays | or you can try it out heath__ |
| 18:33.32 | Slainte | why would it work for some contexts and not for others? |
| 18:33.36 | ManxPower | Slainte, and does it play the unavail message? |
| 18:33.44 | heath__ | thx for the info, looks like a good solution for me |
| 18:34.02 | ManxPower | Slainte, I don't know. I assume you have a [ubpmail] section in voicemail.conf? |
| 18:34.12 | Slainte | Manx, yes for both internal and external. |
| 18:34.27 | Slainte | Manx, yes and both internal and external point to it. Just external crash after the message is played |
| 18:35.06 | Slainte | I created an internal extensions list and external to seperate the outgoing. Since I did that the voice mail for external will play and then die |
| 18:35.08 | jas_williams | Slainte: Are you using different voicemail contexts for each call type ? |
| 18:35.11 | ManxPower | heath__, It's not called ZapBarge for nothing! |
| 18:35.27 | Slainte | no, same context for both mail types, as both need to go to same user |
| 18:35.43 | Slainte | only one section in my mail. ubpmail |
| 18:35.47 | Flyboy6440 | do not think zapbarge will work for sip/iax.. just zap |
| 18:36.20 | hmmhesays | no, it wont' |
| 18:36.28 | tessier_ | chanspy() works for sip/iax/zap |
| 18:36.38 | ManxPower | chanspy is not supported |
| 18:36.45 | hmmhesays | chanspy is also not available in the latest |
| 18:37.02 | ManxPower | hmmhesays, I don't think it was ever included with Asterisk |
| 18:37.06 | tessier_ | asterisk in general isn't supported. |
| 18:37.20 | hmmhesays | hmm, i thought it was at one time |
| 18:37.21 | tessier_ | but chanspy might be made to work, haven't looked at how different it is from modern code |
| 18:37.31 | hmmhesays | it was in the applications list |
| 18:38.09 | hmmhesays | but... I guess I really don't care either, lol |
| 18:38.39 | jero | lol |
| 18:38.51 | Slainte | I think I figured it out |
| 18:39.06 | Slainte | in my mainline context where the external call is received |
| 18:39.13 | Slainte | I have an Absolute Timeout variable |
| 18:39.26 | Slainte | I assumed that would be reset if there is a valid transfer |
| 18:39.29 | Slainte | I dont think it is |
| 18:39.35 | `Sauron | Blah |
| 18:39.38 | hmmhesays | lunch time |
| 18:40.22 | Slainte | testing now with a 180 absolute time |
| 18:40.37 | Slainte | Yep |
| 18:40.42 | letherglov | Slainte, you got a kewlstart external line? |
| 18:40.47 | letherglov | is it getting the disconnect? |
| 18:40.50 | tessier_ | I've got 30,000 unused DID's pointed at this asterisk box and the number of worng numbers is just amazing. |
| 18:41.09 | Slainte | no, it is pri |
| 18:41.12 | tessier_ | I'm tempted to route them to my phone and answer calls with "Joe's crematorium, you kill 'em we grill 'em!" |
| 18:41.18 | Slainte | , the Absolute Timeout was the problem |
| 18:41.22 | tessier_ | Or perhaps "Joe's abortion clinic, you rape 'em, we scrape 'em!" |
| 18:41.24 | letherglov | tessier_, what business are you in? |
| 18:41.33 | Slainte | how can I get that to be ignored if a succesfull extensions is dialed |
| 18:41.40 | tessier_ | letherglov: uh...the phone business. |
| 18:41.53 | letherglov | tessier_, right, but with 30,000 DIDs? |
| 18:42.00 | letherglov | is this a CLEC or something? |
| 18:42.05 | tessier_ | letherglov: Yeah. Most phone companies have quite a few DID's. :) |
| 18:42.23 | tessier_ | No, we are not technically a CLEC> |
| 18:42.34 | *** join/#asterisk fizbar (~jason@dsl-232-31.speedsite.com) |
| 18:43.14 | tessier_ | /lib/modules/2.4.21-4.ELsmp/misc/zaptel.o: init_module: Cannot allocate memory |
| 18:43.20 | tessier_ | Now I wonder what's up with that. |
| 18:43.25 | letherglov | there's always the zapateller |
| 18:43.55 | mAsH` | can i connect * to a PABX by an HFC card? |
| 18:44.25 | junky[work] | tessier_: how many PRIs for 30 000 DIDs? |
| 18:44.41 | Flyboy6440 | i'm looking for a way to verify the callerid number is valid.... before running zapateller or privacy manger... like agi with regular expressions or something.. anyone else tried this yet? |
| 18:45.28 | ManxPower | Flyboy6440, What are the rules to determine if a callerid is valid? |
| 18:45.40 | Slainte | All fixed, changed Absolute to ResponseTimeout |
| 18:46.26 | bannerman | The Sipura SPA-841 looks like a pretty decent budget phone. I'm a little worried about the volume thing. It sounds like it has a fairly in-depth configuration system.. is it possible that there is an outbound volume level that you can adjust? |
| 18:46.42 | Flyboy6440 | something like this... |
| 18:46.42 | Flyboy6440 | ^(\d{3}-\d{3}-\d{4})*$ |
| 18:47.12 | ManxPower | bannerman, not that I could find. you can adjust the volume of the ringer, handset, headset, and speakerphone, but not the microphones. Prolly an oversite in the firmware. I reported it to them. |
| 18:47.22 | ManxPower | Flyboy6440, You mean _NXXNXXXXXX |
| 18:47.32 | Flyboy6440 | something like that.. |
| 18:47.48 | ManxPower | exten => s/_NXXNXXXXXX,1,Do The Right Stuff |
| 18:48.00 | ManxPower | exten => s,1,Do hateful stuff |
| 18:48.00 | bannerman | ManxPower: It seems like that's something that couldn't possibly be missed in testing, I mean, you try to make a call on it and nobody can hear you very well.. |
| 18:48.07 | Flyboy6440 | what i'm seeing is .. for example.. calls with no caller id from voicepulse.. have a number of literally... *-***-***-**** |
| 18:48.32 | ManxPower | Flyboy6440, CLID num is never transmitted with non-numbers, its the display device that inserts the non-numbers |
| 18:48.33 | Flyboy6440 | and since caller id is set.. zapteller, and privacymanger do not pickit up |
| 18:49.08 | tessier_ | ah, crap. This mobo has OHCO and not UHCO. |
| 18:49.09 | ManxPower | Why not just send everyone to an IVR? That stopped 99.9% of telemarketing calls. |
| 18:49.15 | ManxPower | Actually it stopped ALL of them. |
| 18:49.29 | ManxPower | Only a collection agency looking for someone I never heard of has gotten thru my basic IVR. |
| 18:49.54 | Flyboy6440 | also i've seen numbers from my pstn like 000 or 999 |
| 18:50.08 | ManxPower | I was REALLY nasty to them. Threatneing to contact the FCC, the PUC/PSC, and a lawyer. They got off the line pretty fast. |
| 18:50.44 | `Sauron | Blah. |
| 18:51.18 | `Sauron | Manx: What's your IVR? |
| 18:51.47 | ManxPower | `Sauron, "If you know the extension of the person your are trying to reach, dial it now." |
| 18:52.03 | `Sauron | Hehn. |
| 18:53.11 | ManxPower | Simple, easy, and totally confuzzles auto-dialers |
| 18:53.49 | `Sauron | what about friends, or other people - to dial you? |
| 18:53.51 | tessier_ | ManxPower: If they were a collection agency trying to collect a debt your threats to call the FCC etc were groundless. |
| 18:54.27 | ManxPower | tessier_, Yes, but I didn't actually give them a chance to tell me they were not a telemarketer before I started threatening to go to the FCC. |
| 18:54.39 | tessier_ | ah |
| 18:54.51 | Flyboy6440 | i think i pre-agi script that validates the number is truly an actual phone number should be easy enough.. |
| 18:55.04 | ManxPower | I think the conversation strarted out something like "Who the hell are you and why are you calling me!?! I am on the FCC Do not call list! I'm reporing this call to them!" |
| 18:55.12 | Flyboy6440 | lol |
| 18:55.18 | jarrod | Extension '5541111' in context 'default' from '7133442322' does not exist. Rejecting call on channel 0/1, span 1 |
| 18:55.38 | jarrod | why does the s,1, not apply from the default context |
| 18:55.41 | jarrod | on this inbound call? |
| 18:55.47 | ManxPower | jarrod, You don't have an extension 5541111 in the context named default |
| 18:55.47 | Samoied | ragnar: I found the solution: |
| 18:55.55 | jarrod | but I have an s, |
| 18:55.57 | Samoied | ragnar: add fromuser= in sip.conf |
| 18:56.01 | ManxPower | jarrod, "s" is only called when we don't KNOW the number dialed. |
| 18:56.07 | jas_williams | jarrod: s only catches no extension |
| 18:56.24 | jarrod | i see |
| 18:56.27 | jas_williams | _X. will catch everything |
| 18:56.28 | jarrod | thanks guys |
| 18:56.32 | outtolunc | '5541111' does NOT equal 's' <G> |
| 18:56.37 | ManxPower | If you have a PRI you never need exten s |
| 18:57.15 | `Sauron | Grf. Oh what I'd give for an fxo device right about now. |
| 18:57.31 | `Sauron | that came out wrong |
| 18:58.36 | Xanathar | Could miscabling result in an E1 being Active (no alarms) and showing as OK in zttool (and if I change the framing or crc checking it alarms), but not showing as Up in asterisk ? |
| 19:00.16 | jas_williams | Xanathar: No more likely shutdown at the provider, try a pri intense debug span 1 and see if you are getting anything. |
| 19:00.38 | Xanathar | Im not getting anything from the other side during the intense debug, only sending packets |
| 19:00.45 | Xanathar | well, frames |
| 19:00.48 | *** join/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 19:00.58 | adjacent | is anyone familiar with VOCAL? |
| 19:01.07 | tessier_ | Anyone here compiled zaprtc on RH9? |
| 19:01.12 | tessier_ | #error Modules should never use kernel-headers system headers ... |
| 19:01.21 | jas_williams | Xanathar: Sounds like it's shutdown on the providers end. |
| 19:01.25 | tessier_ | I keep getting the above error. I changed the makefile to get the headers from the approproate place. |
| 19:01.31 | tessier_ | But something somewhere is still pulling in the wrong ones |
| 19:02.02 | Xanathar | jas_williams, this is connected via a crossover to a NEAX 2000 PBX, and it *should* be configured right |
| 19:02.19 | Xanathar | as I saw when it went from alarm to okay when it was configured |
| 19:02.47 | Xanathar | so I was wondering if having the ring and tip swapped for the wiring would cause something like this |
| 19:03.25 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 19:03.28 | `Sauron | Humm. |
| 19:03.29 | bannerman | I can't find any reviews for the ArtDio IPF-1000 .. which leads me to believe it's probably not worth using. Anyone have experience with thme? |
| 19:05.19 | *** join/#asterisk Tornad (~Tornad@81.56.183.143) |
| 19:05.25 | jas_williams | Xanathar: What pinouts are you using on the crossover ? |
| 19:06.25 | Xanathar | jas_williams, unfortunatly that is the one thing I cant verify, I am remotly setting this up |
| 19:07.01 | Xanathar | I know it has to be at least partially right otherwise I would never get out of red alarm |
| 19:07.02 | jas_williams | Xanathar: The next thing I was coing to say was try a loop back :P oh well |
| 19:07.04 | `Sauron | Grrr. |
| 19:07.16 | `Sauron | BV is giving me a busy on my * sip connection |
| 19:07.17 | `Sauron | GRRRRRRRRRRRRRRRR |
| 19:08.09 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:09.51 | Delvar | gn all |
| 19:09.57 | Delvar | and have a good week end! |
| 19:10.33 | jas_williams | Xanathar: It could be a cable problem. Or configuration |
| 19:10.56 | sivana | `Sauron: :-) |
| 19:11.08 | hmmhesays | ahhh pizza hut buffet, it rocks |
| 19:12.46 | Xanathar | thanks jas_williams, I guess i will keep hacking at this |
| 19:14.29 | jas_williams | Xanathar: Do you have the ability to do a remote loop back on the NEAX PBX ? |
| 19:14.50 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 19:16.03 | Xanathar | jas_williams, let me see |
| 19:16.09 | Xanathar | this thing's config sucks |
| 19:16.22 | Xanathar | i thought my intertel was bad, this is 100 times worse |
| 19:16.59 | `Sauron | sivana: what? |
| 19:17.52 | Xanathar | i guess I am done testing for the day as someone turned off the machine which had the management tool and serial cable on it |
| 19:19.32 | *** join/#asterisk adjacent_ (scott@nc-65-40-81-64.sta.sprint-hsd.net) |
| 19:21.28 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 19:23.41 | *** join/#asterisk Bile_One (~TomSawyer@pcp03281999pcs.gillst01.ar.comcast.net) |
| 19:28.15 | tessier_ | hmm...I have a ton of calls going into a context with only a few legit numbers/extensions in it. I want to play a short message to all of the other calls coming in to extensions that don't exist. How do I make a wildcard extension that won't override my other existing extensions to catch these calls? |
| 19:29.39 | Bile_One | Any know why I can't make a call to an internal SIP to SIP I get the message that the person is unavailable at the extension I am calling to. |
| 19:30.22 | tessier_ | Is the other phone registered with asterisk? |
| 19:30.33 | sivana | tessier_: http://pastebin.ca/5288 |
| 19:30.41 | sivana | just change the pattern match |
| 19:30.55 | Bile_One | Yes, sip show peers says all are reachable |
| 19:31.00 | jas_williams | tessier_: Put the wild card in its own context exten => _X.,Do something and then include wild card context at the end ov you existing context |
| 19:32.12 | tessier_ | sivana: Thanks but I think jas_williams' suggestion is more along the lines of what I need. I know how to just play back a message. I want to make sure people calling my own legit extensions don't get the message though. |
| 19:32.50 | sivana | ya.. tha'ts what I have for my incoming dids.. anything not explicitly listed in that context (at the top) falls into that one |
| 19:32.51 | tessier_ | fizbar: Yeah, I guess the i extension could work also... |
| 19:34.20 | tessier_ | There, let's see how this works... |
| 19:35.08 | jas_williams | tessier_: Show dialplan will show you the pattern match order |
| 19:37.11 | Slainte | I think adding | for grep and more (like ciscos include) would be handy for the CLI |
| 19:37.13 | tessier_ | hmm...not working. |
| 19:38.13 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 19:41.07 | Slainte | Is there a way for the hold to be an intermittent beep instead of the holdmusic. my mp3 playback is garbled |
| 19:41.15 | jas_williams | tessier_: do a show dialplan context in the cli and check out the evaluation order |
| 19:41.15 | UBiQUiTY | hello.... I'm about to upgrade my Fedora Core 3 Kernel.... I know there are some problems with certain versions conflicting with the zaptel driver.... are there any redhat kernel versions i should stay away from in the 2.6.10 tree? |
| 19:42.57 | Slainte | UBiQ, yes. the ones that dont work :) |
| 19:43.21 | hmmhesays | i get sick of idiots wanting me to teach them all about the telecommunications industry |
| 19:44.23 | UBiQUiTY | haha |
| 19:44.23 | bjohnson | is there a way to get a SPA2000 to work both inside a LAN AND outsided the LAN to the Nat'd * server without changing either the SPA or the * config? |
| 19:45.22 | UBiQUiTY | Slainte: so basically, u think it's safe to try redhat's newest: kernel-smp#2.6.10-1.760_FC3 ? |
| 19:45.49 | bjohnson | UBiQUiTY: I ran a x100p at home on fc3 .. no issues |
| 19:46.06 | bjohnson | don't know what kernel though .. I've since used that card elsewhere |
| 19:46.48 | Slainte | UBiQ, the beauty about Linux is you can try mutiple kernels |
| 19:46.52 | UBiQUiTY | Bjohnson: i've had good luck with most FC3 kernels... but someone in this chat room once told me to stay away from kernel#2.6.9-1.724_FC3 |
| 19:48.02 | tessier_ | jas_williams: It is being evaluated in the right order. |
| 19:48.41 | tessier_ | It's odd...when I dial 9002 (a nonexistant extension on my system) I would expect to get the "invalid extension" message. Instead the phones gives me a fast busy and asterisk says nothing on the console. |
| 19:48.45 | *** join/#asterisk zno (~zeno@ip-160-79-174-101.autorev.intellispace.net) |
| 19:48.51 | UBiQUiTY | right now i'm on kernel-smp#2.6.9-1.681_FC3 , but they took that kernel off the apt repositories, and since i know there are some slick new scsi improvements in 2.6.10, i want to upgrade.... but the main reason why i'm asking, is because we're setting up somewhat of an asterisk cluster over here..... and i'm trying to make sure that i can have the exact same setup on each machine, same kernel version etc |
| 19:49.04 | UBiQUiTY | anyhow, i will try the newest and i'll let u know what happens in a few minutes |
| 19:49.11 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 19:49.30 | jas_williams | tessier_: Did you use i or _X. ? |
| 19:49.31 | yogurt2ungue | hello people |
| 19:50.13 | jas_williams | can you post the context to pastebin.ca for us to look at ? |
| 19:50.17 | tessier_ | jas_williams: I have tried both. |
| 19:50.20 | tessier_ | ok |
| 19:50.27 | yogurt2ungue | who had a Line Jack of Quicknet running in * with 2.6.9 or similar |
| 19:50.51 | zno | is there a way to Dial the 2nd line of a multiple line-appearance sip phone when only the first line is registered as a sip peer? |
| 19:51.24 | tessier_ | http://pastebin.ca/5289 |
| 19:51.37 | tessier_ | jas_williams: That is an abbreviated part of the relevent context |
| 19:52.04 | tessier_ | I used to have _X. in there but changed it. |
| 19:53.07 | jas_williams | tessier_: This line is incorrect exten => _1XXXXXXXXXX,1,Dial(SIP/17142639097,20) it expects a 11 digit number |
| 19:53.37 | tessier_ | Right, and we are getting 11 digit numbers |
| 19:54.19 | tessier_ | <PROTECTED> |
| 19:54.21 | jas_williams | tessier_: Starting with 1 ? |
| 19:54.24 | tessier_ | for example |
| 19:54.24 | tessier_ | Yes |
| 19:54.40 | tessier_ | The above is a call coming in from our PSTN gateway. |
| 19:57.24 | fizbar | Has anybody ever had zapata/zaptel dial incorrect numbers(seemingly randomly)? |
| 19:57.41 | tessier_ | fizbar: never heard of that |
| 19:58.05 | bjohnson | anyone config a spa2000 for nat yet? I assume someone has |
| 19:58.17 | jas_williams | tessier_: It is using the context @icci-gw not the one you posted ? |
| 19:58.40 | tessier_ | The calls are coming into the context I posted |
| 19:59.09 | tessier_ | icci-gw is in the same context which includes the catchall context |
| 19:59.52 | bannerman | So.. if I sign up for an unlimited plan at Broadvoice or TelAIX .. I have to basically pay for one account for each "line" that I need for receiving/sending calls, is that right? |
| 20:00.11 | jas_williams | tessier_: Can you post the results of show dial plan telepacket-catchall |
| 20:00.32 | bannerman | In other words, if my office is going to call 5 people at once, I've got to get 3 accounts.. |
| 20:00.34 | bannerman | er, |
| 20:00.35 | bannerman | 5 |
| 20:00.36 | bannerman | accounts |
| 20:01.53 | bjohnson | bannerman: technically yes |
| 20:02.11 | bjohnson | bannerman: some voip providers allow more that 1 concurrent call per account |
| 20:02.23 | bjohnson | pay per minute ones almost always do |
| 20:02.32 | bjohnson | all inclusive one vary |
| 20:02.37 | bjohnson | I think some allow 2 |
| 20:03.05 | bjohnson | I think bv allows more than 1 but you pay per minute for more that the first |
| 20:03.24 | bjohnson | check the terms of service |
| 20:03.43 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 20:04.06 | bjohnson | teliax commercial unlimited I think allows up to 6 concurrent .. check with them for details |
| 20:04.21 | bannerman | thanks everyone |
| 20:04.36 | bannerman | I appreciate the help with all my newbie questions :) |
| 20:04.55 | bjohnson | you need to watch bandwidth usage if a dsl connection |
| 20:05.55 | fa | HA |
| 20:05.59 | fa | I PASS THE EXAM |
| 20:07.33 | *** join/#asterisk toddf (~toddf@net-66-210-104-117.theshop.net) |
| 20:10.49 | *** join/#asterisk RoyK (~roy@8.80-203-22.nextgentel.com) |
| 20:11.29 | eKo1 | This is weird. When I call the US, the caller id show on the callees phone is their number! |
| 20:13.01 | jero | lol |
| 20:16.10 | toddf | eko1: sounds like you have some setcallerid set wrong |
| 20:16.23 | eKo1 | I'm not using setcallerid anywhere. |
| 20:16.34 | Mw3 | isdn ? |
| 20:16.41 | eKo1 | Nope, the calls are SIP. |
| 20:17.20 | Mw3 | hm, i had the same situation with isdn bri incoming lines in this morning with a panasonic kx-tda 100 |
| 20:17.34 | Mw3 | i dunno why |
| 20:17.42 | eKo1 | Hmm...If I add restrictcid=yes in the sip entry of my provider, maybe... |
| 20:21.53 | bjohnson | ~docs |
| 20:21.54 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:25.47 | *** join/#asterisk Dibbler (~Dibbler@snaddy.plus.com) |
| 20:27.46 | *** join/#asterisk WiFiGuy (WiFiGuy@CPE-69-76-99-187.wi.rr.com) |
| 20:29.20 | channan | eko1- what about calling non-US numbers or internal SIP/softphones? |
| 20:29.49 | yogurt2ungue | I have a Quicknet's Line Jack in * box, who are working with one of it? |
| 20:31.31 | eKo1 | channan: Calling internal sip phones shows the correct cid. |
| 20:31.59 | *** part/#asterisk Samoied (~samoied@200.247.141.111) |
| 20:32.20 | Bile_One | EKo1 you ever use AMP? |
| 20:32.41 | eKo1 | Bile_One: I don't know what that means. |
| 20:33.07 | Bile_One | Eko1 Asterisk Management Portal |
| 20:33.14 | channan | ek01- that's weird.. where r u calling from? |
| 20:33.31 | eKo1 | OK, I called my cellphone here in Honduras through my sip provider and I get some other cid (i.e. not mine). |
| 20:34.23 | channan | eko1- I see... I wished I could do that so I can scare my frineds :) |
| 20:34.39 | channan | friends |
| 20:34.49 | eKo1 | channan: If I find out how it works, I'll spoof it and screw everybody.... |
| 20:34.53 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 20:34.55 | eKo1 | he he |
| 20:35.17 | *** part/#asterisk Flyboy6440 (~Bobo@192.76.82.89) |
| 20:35.22 | eKo1 | It seems to only work when calling to the US. |
| 20:35.40 | Bile_One | Ek01 ever have it where you could not make internal SIP calls? |
| 20:35.58 | eKo1 | Bile_One: I had the situtation yesterday. |
| 20:36.14 | Bile_One | EKo1, What did you do to fix it |
| 20:36.22 | eKo1 | I reloaded asterisk. |
| 20:36.26 | eKo1 | I mean restarted |
| 20:36.39 | Bile_One | EKo1 I have done that 100 times today |
| 20:37.11 | eKo1 | Bile_One: so what... |
| 20:37.20 | ManxPower | ~docs |
| 20:37.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:37.23 | ManxPower | ~doc |
| 20:39.14 | Meznev | Just curious, should I be worried at all that during any phone call asterisk keeps saying "Urgent handler" over and over again? |
| 20:39.45 | Bile_One | EKo1 So what, what? I can't understand why I can make calls out through a zap and not intenral through SIP? |
| 20:41.58 | ManxPower | Meznev, no. If you don't run in debug or console mode you won't see that. |
| 20:42.26 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 20:42.38 | jas_williams | Bile_One: Have you changed you extensions.conf |
| 20:43.27 | jas_williams | Bile_One: do you see an error in the console ? |
| 20:43.58 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 20:44.02 | Bile_One | Jas_williams no it just goes to the unavailalbe menu for voice mail. |
| 20:45.21 | jas_williams | Bile_One: do a sip show peers and post the results to pastebin |
| 20:45.25 | Bile_One | Jas_williams, sip show peers shows the extenstion are availalbe. |
| 20:46.21 | jas_williams | DO the extensions have IP addresses shown ? |
| 20:46.37 | Bile_One | Jas_williams, yes |
| 20:46.47 | *** part/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 20:46.56 | Slainte | Why would * complain a sound file is not there, when in fact it is there with all the proper permissions? |
| 20:47.03 | Juggie | which module does the t405p use? |
| 20:47.14 | tzanger | wct4xxp |
| 20:47.31 | jas_williams | Bile_One: can you post the console messages for a call attempt |
| 20:47.41 | kFuQ | grrr.... does anyone have any ideas on getting callwaiting to work right on x100p |
| 20:47.42 | kFuQ | or do i need to "map" a key to the flash command or somethign like that |
| 20:47.42 | Slainte | <PROTECTED> |
| 20:47.57 | Bile_One | Jas_williams, sure thing hang tight |
| 20:49.49 | Slainte | The reason is it had no valid extensions |
| 20:49.50 | Slainte | The reason is it had no valid extension |
| 20:49.51 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 20:50.14 | terrapen | well, the phisher is back at it.... http://64.65.250.200:87/s/ |
| 20:50.25 | *** join/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 20:50.38 | harryvv | kfuq, I dont know I have the x100p and I think we dont have that option. What have you done with it? |
| 20:51.19 | *** part/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 20:51.25 | Bile_One | Jas_williams, http://pastebin.ca/5292 |
| 20:52.20 | harryvv | slainte what do you have for your start prirorty? is it background(mysoundfilenamehere) and is it listed in /var/lib/asterisk/sounds ? |
| 20:52.37 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 20:53.50 | jas_williams | Bile_One: Your AGI Script does not seem to be doing its job and calling phones ? |
| 20:54.15 | Bile_One | Jas_williams, should I post it? |
| 20:54.23 | *** join/#asterisk marc1 (~lu_anne84@node-423a7fed.mia.onnet.us.uu.net) |
| 20:55.23 | jas_williams | Bile_One: Why do you need a script to do the calling ? |
| 20:56.09 | Bile_One | Jas_williams, it is part of amp which is the Asterisk Management Portal I am playing with. |
| 20:58.48 | jas_williams | Then you need to ask the authors of amp but it looks to me that extension 601 should be called as SIP/601 not 601 as this would indicate -- Executing Macro("SIP/603-ff3c", "dial|15|tr|601") in new stack |
| 20:59.52 | jas_williams | you are currently trying to do |
| 21:00.06 | jas_williams | But I do not use AMP |
| 21:01.14 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 21:01.58 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 21:02.13 | kFuQ | harryvv: well.. the callwaiting tones come thru, it's just not clickin over to pick up other call.. |
| 21:02.48 | kFuQ | harryvv: if i hit the flash button, it just gives me dialtone like im making a 3-way call |
| 21:03.37 | terrapen | i need more people for my anti-phishing campaign |
| 21:03.42 | harryvv | fk, I dont know about that. |
| 21:03.54 | terrapen | its going to take more HTTP requests than my server can dish out |
| 21:04.02 | harryvv | need to know more about the commands I guess |
| 21:04.28 | *** join/#asterisk ToyMan (~konversat@204.8.82.238) |
| 21:07.04 | kFuQ | i think that i might work thru the transfer function |
| 21:07.55 | Bentley | Bile_One: you have permissions problems on dialparties.agi. run 'amportal chown' or /usr/src/AMP/chown_asterisk.sh |
| 21:08.02 | `Sauron | mark, do you know anything about the PPPD command that was added to *? |
| 21:08.12 | Bentley | BTW - you will get good support if you join the AMP maillist or use the forums |
| 21:10.57 | eKo1 | Fuck! I can't make internal calls anymore |
| 21:11.05 | *** join/#asterisk MooingLemur (~troy@phoenix.pinchaser.com) |
| 21:11.18 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:11.19 | harryvv | eko1, do you have any zap cards installed? |
| 21:11.37 | eKo1 | Yes. Three of them. |
| 21:11.56 | harryvv | check the asterisk and also command lines for any errors. can you check voicemail/ |
| 21:11.58 | harryvv | ? |
| 21:11.59 | eKo1 | Two X101Ps and one TDM400. |
| 21:12.04 | PBXtech | how do you install the cdr-mysql stuff? i did a make in the addons directory and it didnt compile the cdr-mysql module |
| 21:12.57 | eKo1 | No errors. Hmm...it seems that only this office isn't working. |
| 21:13.12 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 21:13.57 | harryvv | okay |
| 21:14.03 | harryvv | more then one office linked? |
| 21:15.18 | MooingLemur | I'm having trouble with nufone. I believe I have everything set up correctly and to their specifications. All other IAX accounts have been commented out. iax2 show registry shows correctly. The error comes back as: Feb 4 14:13:24 NOTICE[9346]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 198.22.67.70 when attempting a test call. |
| 21:15.29 | eKo1 | Well, there are three offices that are physically seperated by 20 miles. |
| 21:15.45 | MooingLemur | Anyone have any ideas? I don't think it's a codec issue. |
| 21:16.13 | eKo1 | Funny, I call and nothing shows on the CLI, not even with 'sip debug peer'. It's as if traffic from this office is getting lost. |
| 21:16.45 | harryvv | ek, your office phones not working mmmm are thay sip ? sip show peers |
| 21:17.23 | eKo1 | sip show peers shows them as OK. |
| 21:17.30 | harryvv | how long has it worked before the last crash? |
| 21:17.38 | harryvv | or what ever mode your in |
| 21:17.52 | harryvv | do a tcpstream |
| 21:17.54 | eKo1 | Hmm...I power cycled my Handytone and now it works. |
| 21:18.02 | eKo1 | Let me check another phone. |
| 21:18.04 | harryvv | okay |
| 21:19.11 | eKo1 | Well, it ain't working from the Cisco ATA. I'll power cycle that too and see what happens. |
| 21:19.29 | eKo1 | But it did show up in the CLI. |
| 21:19.47 | eKo1 | It says: Got SIP response 302 "Moved Temporarily" back from 192.168.124.112 |
| 21:19.50 | marc1 | eKo1 , if you are using grandstream ata's the best firmware is: 1.0.5.11 |
| 21:20.18 | eKo1 | I don't think it's a firmware problem. I think it is and * problem. |
| 21:20.30 | *** join/#asterisk BrainStormerToo (~bcbrown@c-24-20-119-153.client.comcast.net) |
| 21:20.39 | BoRiS | eko1: look at ther bug tracker... |
| 21:21.27 | harryvv | ek01, do you have any backup plan if your * desides to go south? |
| 21:21.31 | marc1 | eKo1, I use to have the same problem, since I downgraded, I have no more problems |
| 21:21.54 | hmmhesays | lol, this card looks like a glorified modem |
| 21:22.28 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 21:22.48 | kFuQ | call waiting shouldn't be this dam hard |
| 21:23.04 | buddah | i'm having a problem with a clients polycom ip500s, they are plugged into a hub which goes into their router, and the phones are bringing down the router, and their network, anyone heard of something like this? |
| 21:23.19 | jas_williams | eKo1: Have you rebooted * if yes the phones will not be registered until the registration period expires |
| 21:23.44 | buddah | its happening when they phones are not in use, but plugged in, and even sooner if they are used |
| 21:23.52 | jas_williams | eKo1: as sip show peers will have no registered IP addresses |
| 21:23.55 | marc1 | does any one know a good IAX provider for international termination ? |
| 21:24.16 | hmmhesays | i know a few fair sip providers |
| 21:24.28 | BrainStormerToo | what causes the Playback Application to timeout rather than go to next priority? |
| 21:24.30 | marc1 | sip will do it : ) |
| 21:24.42 | hmmhesays | not having a next priority |
| 21:24.57 | BrainStormerToo | I do.. hmmm |
| 21:25.07 | BrainStormerToo | [Changed] |
| 21:25.07 | BrainStormerToo | exten => 1,1,Zapateller(answer) |
| 21:25.07 | BrainStormerToo | exten => 1,2,Playback(the-number-u-dialed|skip) |
| 21:25.07 | BrainStormerToo | exten => 1.3,SayDigits(${oldnum}) |
| 21:25.07 | BrainStormerToo | exten => 1,4,Playback(has-been-disconnected|skip) |
| 21:25.07 | BrainStormerToo | exten => 1,5,Playback(the-new-number-is|skip) |
| 21:25.09 | BrainStormerToo | exten => 1,6,SayDigits(${newnum}) |
| 21:25.11 | BrainStormerToo | exten => 1,7,Hangup |
| 21:25.20 | BrainStormerToo | sdang |
| 21:25.22 | hmmhesays | take a look |
| 21:25.23 | BrainStormerToo | comma... |
| 21:25.28 | hmmhesays | good man |
| 21:25.33 | hmmhesays | that'll be 5 dollar |
| 21:25.36 | hmmhesays | or a hooker |
| 21:25.37 | BrainStormerToo | :-) |
| 21:25.40 | hmmhesays | whichever you can spare |
| 21:26.04 | zigman | or your ass |
| 21:26.06 | zigman | ;) |
| 21:26.09 | hmmhesays | lol |
| 21:26.10 | BoRiS | lol |
| 21:26.12 | BoRiS | good enuff |
| 21:26.25 | zigman | hi everyone |
| 21:26.27 | zigman | lol |
| 21:26.37 | marc1 | hmmhesays, give me a url! |
| 21:26.44 | bjohnson | anyone use Broadvox? They're supposed to have Canadian DIDs |
| 21:26.50 | hmmhesays | for what? |
| 21:26.57 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 21:27.20 | marc1 | I used broadvox long time ago |
| 21:27.30 | BrainStormerToo | nottice one other prob. put immediate=yes |
| 21:27.51 | BrainStormerToo | in zapata.conf. still get simple switch. |
| 21:28.17 | bjohnson | marc1: switched for some reason I guess? |
| 21:28.37 | hmmhesays | oh .. for sip providers? there's a million of them out there |
| 21:28.44 | hmmhesays | iconnect is *ok* |
| 21:28.50 | marc1 | yes |
| 21:28.57 | hmmhesays | gphone is really stable, but expensive |
| 21:29.07 | harryvv | what is it |
| 21:29.09 | marc1 | I got my own pri with peatec |
| 21:29.21 | redder86 | stupid user sent an e-mail with a 200MB attachment. |
| 21:29.24 | hmmhesays | anchor is so/so |
| 21:29.33 | redder86 | base64 encoded that's a whopping big e-mail |
| 21:29.42 | harryvv | gphone is a hardphone? |
| 21:29.50 | hmmhesays | gphone is a service provider |
| 21:29.54 | harryvv | k |
| 21:30.08 | hmmhesays | expensive, but you get what you pay for |
| 21:30.19 | harryvv | reeder 200mb vm? |
| 21:30.20 | harryvv | ;) |
| 21:30.28 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 21:30.37 | redder86 | harryvv: not a voicemail - scanned payroll files |
| 21:30.52 | hmmhesays | ok, why is someone sending payroll files across email? |
| 21:31.07 | redder86 | didn't I say it was a stupid person? |
| 21:31.10 | jas_williams | Cus they think it is secure |
| 21:31.14 | hmmhesays | lol |
| 21:31.34 | harryvv | reeder, sounds like thay are a talker |
| 21:31.35 | harryvv | :) |
| 21:31.51 | harryvv | reeder, put a timmout on that cid next time |
| 21:31.52 | harryvv | ;) |
| 21:31.58 | bjohnson | marc1: so no issues with broadvox then? |
| 21:32.05 | bjohnson | looks like good prices to me |
| 21:32.07 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
| 21:33.20 | marc1 | bjohnso: Broadvox is too expencef |
| 21:33.31 | *** join/#asterisk coppice (~chatzilla@205.162.17.210.dyn.pacific.net.hk) |
| 21:35.32 | Bile_One | Bentley, I did run the atrisk_chown shutdown restarted... I tried amportal chown restarted still cannot make internal sip to sip calls. |
| 21:36.48 | bjohnson | marc1: too expensive? |
| 21:37.01 | marc1 | bjohnson :yes |
| 21:37.21 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 21:37.41 | marc1 | bjohnson : they ask me for 3500.00 garanty per month |
| 21:37.49 | blitzrage | I don't think anything bothers me more than Out of Office replies every single time I reply to a message on the mailing lists |
| 21:37.59 | terrapen | hahah i killed the phisher again |
| 21:38.06 | zigman | blitzrage the german guy ? |
| 21:38.13 | blitzrage | yep... and someone else too |
| 21:38.27 | zigman | remove them from the ml |
| 21:38.28 | zigman | ;) |
| 21:38.32 | blitzrage | wish I could |
| 21:38.32 | zigman | onyl chance |
| 21:38.36 | terrapen | http://64.65.250.200:87/s/ |
| 21:38.38 | terrapen | can anybody reach that? |
| 21:38.39 | *** join/#asterisk Caede (~chatzilla@204.94.248.81) |
| 21:39.11 | zigman | terrapen firefox denied the port for security reasons ;) |
| 21:39.34 | harryvv | terrapen no |
| 21:40.00 | terrapen | harry, good |
| 21:40.04 | terrapen | its a dumb phisher |
| 21:40.05 | BoRiS | terrapen: no |
| 21:40.07 | terrapen | i took him down once |
| 21:40.18 | terrapen | and then he changed his form to point to a different PHP script |
| 21:40.26 | terrapen | so i changed my attacker |
| 21:40.34 | eKo1 | I don't get it. I can call my internal phones fine but nobody can call me. |
| 21:40.42 | blitzrage | terrapen: yah, my FF blocks it too |
| 21:40.53 | *** join/#asterisk CCDAS (~spears_da@65.163.100.254) |
| 21:40.54 | eKo1 | It always says: Got SIP response 302 "Moved Temporarily" back from 192.168.124.112 |
| 21:40.55 | terrapen | try from a non-FF browser |
| 21:40.58 | terrapen | or try telnet |
| 21:41.04 | eKo1 | And I get a 403 on the phone. |
| 21:41.07 | terrapen | i just need to see if he's really down or just filtering me |
| 21:41.16 | blitzrage | checking |
| 21:41.18 | terrapen | i doubt he's filtering me because, i think, he's on a windows box |
| 21:41.35 | terrapen | http_load is a neat utility |
| 21:41.38 | blitzrage | unable to connect |
| 21:41.44 | terrapen | sweet. |
| 21:41.45 | blitzrage | from lynx |
| 21:42.13 | terrapen | many of these guys use "pre-boxed" phisher sites |
| 21:42.14 | BoRiS | eko1: What firmware version? |
| 21:42.17 | terrapen | with very lame PHP |
| 21:42.25 | terrapen | i'd love to get ahold of the actual code they run |
| 21:42.35 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 21:42.54 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 21:43.26 | zigman | lynx: Can't access startfile http://64.65.250.200:87/s/ |
| 21:43.53 | PBXtech | how do you install cdr-mysql? i did a make install in the addons directory and it only compiles format_mp3 |
| 21:44.18 | zigman | PBXtech you got mysql installed? |
| 21:44.21 | PBXtech | yes |
| 21:44.22 | zigman | and the headers ? |
| 21:44.27 | PBXtech | hmm |
| 21:44.33 | zigman | fc2? |
| 21:44.36 | PBXtech | fc3 |
| 21:44.45 | zigman | rpm -qa | grep mysql |
| 21:44.53 | zigman | look for a mysql-devel package |
| 21:45.12 | PBXtech | yea i installed the devel rpm |
| 21:45.33 | terrapen | ang |
| 21:45.45 | PBXtech | it doesnt even look like its trying to install cdr-mysql |
| 21:45.47 | terrapen | darn |
| 21:45.49 | terrapen | he's back up |
| 21:46.06 | terrapen | i think i need more drones |
| 21:46.55 | PBXtech | am i suppose to do 'make cdr_addon_mysql' ? |
| 21:47.19 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 21:47.23 | blitzrage | ~seen zx81 |
| 21:47.24 | jbot | zx81 <~ZX81@222-153-18-2.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 1d 9h 11s ago, saying: ':)'. |
| 21:47.34 | PBXtech | : undefined reference to `main' |
| 21:47.34 | PBXtech | cdr_addon_mysql.o(.text+0x236): In function `handle_cdr_mysql_status': |
| 21:47.36 | blitzrage | sheesh... just can't seem to get a hold of that guy on IRC |
| 21:47.39 | terrapen | im thinking i need random hex usernames and passwords |
| 21:49.19 | eKo1 | Whats the format of the clid I pass to the SetCallerID app.? Is it SetCallerID("Foo" <12345>)? |
| 21:49.30 | zigman | blitzrage he is out of office |
| 21:49.32 | zigman | for 2 weeks |
| 21:49.41 | zigman | thats what his msg says |
| 21:49.42 | blitzrage | :) |
| 21:49.44 | blitzrage | really? |
| 21:49.46 | blitzrage | he has a message? |
| 21:49.46 | zigman | yes |
| 21:49.54 | zigman | the autoresponder |
| 21:50.00 | blitzrage | interesting |
| 21:50.06 | blitzrage | must be on vacation |
| 21:50.15 | zigman | yes |
| 21:50.16 | zigman | he is |
| 21:50.22 | eKo1 | "Foo" or Foo |
| 21:50.26 | eKo1 | Which is it? |
| 21:50.34 | *** join/#asterisk guugmember (~nachoramo@mail.epa.com.gt) |
| 21:50.35 | zigman | for what ? |
| 21:50.49 | zigman | Foo |
| 21:50.51 | zigman | sorry |
| 21:50.57 | zigman | didn't read your first msg |
| 21:51.08 | zigman | withouth "" |
| 21:53.47 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 21:53.47 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 21:53.59 | jas_williams | eKo1: Both are valid |
| 21:54.05 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 21:54.25 | terrapen | bingo. now i have randomized requests |
| 21:54.33 | terrapen | i'll trash his logs for sure now |
| 21:54.43 | eKo1 | jas_williams: Thanks. |
| 21:54.55 | terrapen | i'm such an asshole. or a hero, depending... |
| 21:55.13 | eKo1 | maybe both... |
| 21:55.16 | terrapen | yep :) |
| 21:55.28 | jas_williams | terrapen: Who's systems are you breaking along the way |
| 21:55.37 | terrapen | not breaking into anything |
| 21:55.40 | terrapen | using my personal box |
| 21:55.53 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 21:55.53 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 21:55.57 | terrapen | this phishing scam is most likely running on some poor bastard's 0wned win32 box |
| 21:55.57 | bkw_ | ~seen mrunix |
| 21:55.59 | jbot | mrunix <bwann@chiba.tessier.com> was last seen on IRC in channel #asterisk, 11d 4h 56m 2s ago, saying: 'it's a little unclear'. |
| 21:56.03 | Darwin35 | BKW whaz up hommie |
| 21:56.09 | terrapen | and i'm filling up his capture script |
| 21:56.12 | blitzrage | bkw_: yo |
| 21:56.14 | terrapen | with useless junmk |
| 21:56.15 | bkw_ | about to leave this channel |
| 21:56.20 | blitzrage | bkw_: I thought you quit here |
| 21:56.23 | blitzrage | ahhh, that makes sense ;) |
| 21:56.23 | bkw_ | its the life force sucking black hole |
| 21:56.29 | bkw_ | trust me guys |
| 21:56.32 | bkw_ | its sucking you dry |
| 21:56.33 | blitzrage | bkw_: agreed. #asterisk-doc is way cooler ;) |
| 21:56.34 | bkw_ | LEAVE NOW |
| 21:56.56 | guugmember | we are playing with asterisk, but want to change the messages of the autoreponder to spanish |
| 21:57.03 | guugmember | we are in central america |
| 21:57.06 | jas_williams | bkw_ I will one day ;-) #asterisk-stable is the coolest |
| 21:57.10 | bkw_ | haha |
| 21:57.29 | bkw_ | seeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee ya |
| 21:57.31 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 21:57.37 | eKo1 | guugmember: Where abouts? |
| 21:57.52 | outtolunc | missed damnit |
| 21:57.54 | *** part/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:58.04 | eKo1 | Guatemala I see. |
| 21:58.10 | guugmember | eKo1, yeah |
| 21:58.23 | eKo1 | Where in Guatemala? |
| 21:58.49 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 21:59.16 | guugmember | do you know where we can find the files that are in /var/lib/asterisk/sounds but in spanish |
| 21:59.21 | guugmember | we are in Antigua Guatemala |
| 21:59.26 | jarrod | in ser if i accept sip:1111 and I have an alias setup for 1111 to point to sip:me@my_sip_realm will it ring my phone? |
| 21:59.33 | *** join/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.rr.com) |
| 21:59.33 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:59.50 | eKo1 | guugmember: There aren't any. You have to make your own. |
| 22:00.27 | redder86 | what got bkw_ so anti #asterisk ? |
| 22:00.31 | eKo1 | jarrod: Try it and find out. |
| 22:00.47 | jarrod | i did :( |
| 22:00.57 | jarrod | i was hoping for more along the lines of.. no do it this way |
| 22:00.57 | jarrod | hah |
| 22:01.04 | outtolunc | probably the constant asking of the same ol questions for 2+ years <G> |
| 22:01.30 | labo | eKo1, http://www.telecomabmex.com/asterisksounds/AsteriskSounds_ES.tar.gz |
| 22:01.51 | labo | poor quality though. |
| 22:02.30 | lohelle | has anyone patched/merged bristuff with a new cvs with atxfer support? |
| 22:02.41 | labo | and she's not as sexy as allison smith |
| 22:02.48 | zigman | atxfer ? |
| 22:02.53 | lohelle | and does anyone have norwegian sounds |
| 22:03.03 | zigman | b-channel bridging? |
| 22:03.03 | lohelle | atxfer in features.conf |
| 22:03.11 | lohelle | attended transfer |
| 22:03.44 | guugmember | thanks labo |
| 22:03.48 | lohelle | availible in cvs (january), but not yet in stable.. and not in bristuff.. |
| 22:04.06 | lohelle | and I need zaphfc (from bristuff) |
| 22:04.28 | CCDAS | Excuse me, does anyone know how to make Queues.conf save the monitored files to another location? |
| 22:05.15 | *** join/#asterisk Blackvel (~blackvel@dsl-082-082-059-221.arcor-ip.net) |
| 22:06.18 | zigman | lohelle my bet is.. it won't be in there till the cvs stuff gets stable |
| 22:06.24 | zigman | and a new asterisk is released |
| 22:06.45 | Blackvel | something unstable? |
| 22:06.53 | eKo1 | labo-rat: I'll get my own sexy latina babe for the sounds. |
| 22:07.07 | eKo1 | And a good mic. can't forget that. |
| 22:07.47 | netsurfer | wiki seems slow tonight, anyone else finding that ? |
| 22:08.48 | nestAr | i've obviously been using the wrong logic for my include's |
| 22:08.51 | Dalion | anyone who can get into a vonage while still subb'ed pm me |
| 22:09.07 | nestAr | because i can call in, and make outgoing calls |
| 22:09.11 | nestAr | ugh |
| 22:09.20 | *** join/#asterisk PBXtech (~nik@67.107.241.9.ptr.us.xo.net) |
| 22:09.50 | nestAr | [incoming] -> [local] -> [outgoing] |
| 22:12.00 | labo-rat | mm, there's this software .. which might give you nic equality, Audacity |
| 22:12.19 | labo-rat | my wife is brazilian, so her spanish is like messy |
| 22:12.35 | lohelle | is it possible to connect asterisk to skype? |
| 22:12.51 | *** part/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.rr.com) |
| 22:15.23 | eKo1 | skype uses a weird codec so no. |
| 22:16.31 | toddf | not only that, skype is encrypted |
| 22:16.51 | labo | they use ilbc. |
| 22:17.25 | coppice | they use wideband ilbc mostly |
| 22:17.47 | coppice | that's why people say it sounds good |
| 22:18.14 | tzanger | I can't for the life of me get ilbc to sound good |
| 22:18.19 | tzanger | and this is on good hardware too |
| 22:18.26 | tzanger | Xeon 2.6 running *nothing* but asterisk |
| 22:18.29 | labo | http://www.ilbcfreeware.org/ |
| 22:18.40 | coppice | ilbc sounds pretty good |
| 22:18.41 | tzanger | I use gsm for everything... every time I move to ilbc I get complaints about audio quality |
| 22:18.56 | coppice | the wideband version isn't freely available, though |
| 22:19.02 | tzanger | coppice: in my experience it can't hold a candle to gsm on asterisk |
| 22:19.12 | *** join/#asterisk Tornad (~Tornad@81.56.183.143) |
| 22:19.26 | coppice | there's something wrong in the setup then. the codec is good |
| 22:19.33 | tzanger | coppice: I'd *love* to figure out what |
| 22:19.48 | tzanger | I've tried trunkfreq=30 too but even with trunking turned off the quality's the same |
| 22:20.06 | coppice | dunno. maybe something screwed up in *, as a number have the same complaint as you |
| 22:20.12 | tzanger | coppice: ahh |
| 22:20.22 | tzanger | I think there's something odd in the wctdm driver too |
| 22:20.31 | tzanger | I *cannot* receive a fax through it reliably, but sending works great |
| 22:20.42 | tzanger | and app_rxfax on the same box works perfectly |
| 22:20.56 | coppice | I think there are various weird things with the TDM card and its driver |
| 22:20.58 | tzanger | with the exception of the odd sigfpe I haven't been able to nail down yet... maybe one per 150 faxes or so |
| 22:22.02 | tzanger | coppice: wanna know what's weird? TDM430P... fax on ports 1 and 2. faxing back and forth between ports is fine. faxing out to a PRI is fine. faxing in from a PRI is not, but faxing in from a PRI to app_rxfax on the same machine is good |
| 22:23.26 | coppice | you are not the only one saying that. a couple of people have said faxes from a machine plugged into a TDM card are Ok, but faxes to the machine fail most/all the time |
| 22:24.15 | Blackvel | ehm |
| 22:24.16 | tzanger | yeah -- the funny thing is -- exact same machine with a t100p and channel bank -- I can send and receive faxes 100% |
| 22:24.25 | tzanger | so it's something with the tdm card IMO, I just have no clue what |
| 22:24.25 | Blackvel | you cant receive faxes with a tdm400p? |
| 22:24.43 | Blackvel | with one or more fax machine connected to it? |
| 22:24.47 | tzanger | Blackvel: not reliably. same box with a t100p+channel bank, no issue whatsoever. same box with app_rxfax, no problem |
| 22:25.04 | tzanger | but that box can *send* faxes from a fax on a tdm430p without issue |
| 22:25.20 | letherglov | tzanger, intel box? |
| 22:25.25 | letherglov | hey coppice |
| 22:25.45 | coppice | hi |
| 22:25.45 | tzanger | letherglov: yes. P3/733 but also Xeon 2.8 |
| 22:25.47 | letherglov | I got your latest spandsp and ported top_bit to ppc |
| 22:25.55 | tzanger | top_bit? |
| 22:26.00 | Blackvel | whats app_rxfax? spandsp? |
| 22:26.02 | letherglov | however, I couldn't figure out what function (mathematically) bottom_bit is |
| 22:26.04 | letherglov | spandsp |
| 22:26.25 | letherglov | I got a copy of hacker's delight, which seemed to cover it for me |
| 22:26.31 | letherglov | I think it ended up being nlog2 or something like that |
| 22:26.36 | coppice | bottom_bit is the opposite of top_bit. |
| 22:26.50 | letherglov | it was log base 2 of x |
| 22:26.56 | letherglov | 31 - number of leading zeros (x) |
| 22:27.03 | letherglov | now, maybe that was just endianess crap? |
| 22:27.05 | letherglov | :-\ |
| 22:27.16 | eKo1 | Could be. |
| 22:27.45 | tzanger | what's top_bit and bottom_bit? |
| 22:27.47 | coppice | ppc should have special instructions to find the top and bottom 1 in a word, just like the x86. most modern CPUs do. |
| 22:27.52 | letherglov | it ended up being 31 - the result of cntlz |
| 22:27.53 | letherglov | yeah |
| 22:27.55 | letherglov | cntlzw |
| 22:27.57 | letherglov | or cntlz, I forget |
| 22:28.04 | letherglov | honestly, I just lost the disk it was on |
| 22:28.07 | letherglov | I think it took a dump |
| 22:28.09 | Blackvel | when you cant receive a fax over your fax machine which is connected to tdm400 |
| 22:28.23 | letherglov | the problem I had, is that there's only a leading zero function |
| 22:28.26 | Blackvel | what can you do else? e.g if you have 4 machines? |
| 22:28.26 | letherglov | not a trailing zero function |
| 22:28.43 | letherglov | so then the question becomes, what transform do I preform before I call the leading zero function? |
| 22:28.48 | tzanger | Blackvel: eh? |
| 22:28.52 | letherglov | one option is to load it into the register backwards |
| 22:28.53 | Blackvel | replacing tdm400 for fax machines may be not the way you can go? |
| 22:28.57 | letherglov | :-\ |
| 22:29.04 | letherglov | either way, the C code worked perfectly ;-) |
| 22:29.07 | tzanger | Blackvel: it's not hte fax machine, and it's not the port on the TDM400 |
| 22:29.09 | letherglov | danka |
| 22:29.19 | Blackvel | so its the driver |
| 22:29.26 | Blackvel | ? |
| 22:29.33 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 22:29.38 | tzanger | Blackvel: something in it, yes |
| 22:30.02 | dan2 | kram: ping |
| 22:30.18 | Blackvel | are there any other ways? different cards with different drivers? is there a well-known workaround (e.g you would have tdm400p + 4 fax maschines) |
| 22:30.29 | outtolunc | tzanger: where in the fax does it get screwed? |
| 22:30.29 | ixx | dose anyone know where to get DIDs in argentina? |
| 22:30.36 | ixx | s/dose/does/ |
| 22:31.03 | tzanger | outtolunc: anywhere really ... most times about 1/4" down or about 1/3 of the way down the page it will look like someone got the paper stuck in the other end... long "run" of the same line over and over |
| 22:31.08 | tzanger | ixx: no, sorry |
| 22:31.09 | letherglov | hmm |
| 22:31.18 | letherglov | tzanger, you really need to ask lee howard about that |
| 22:31.21 | tzanger | Blackvel: t100P+channel bank works wonders :-) |
| 22:31.24 | letherglov | he's the t.30 error master |
| 22:31.27 | outtolunc | that would seem to be a rx buffer issue then |
| 22:31.41 | ixx | I thought I had seen some a few months ago, but I can not seem to find any providers now |
| 22:31.42 | letherglov | are you sending it from a real fax machine? |
| 22:31.46 | tzanger | lee howard? does he show up here at all? |
| 22:31.50 | letherglov | hylafax |
| 22:31.50 | tzanger | letherglov: yes |
| 22:31.52 | letherglov | he knows his fax |
| 22:31.57 | letherglov | no, he doesn't though |
| 22:32.02 | tzanger | ohhhh okay I hear ya |
| 22:32.05 | letherglov | on the hylafax mailing list you could ask about the error |
| 22:32.12 | letherglov | and he can diagnose the HDLC from hearing the symptoms |
| 22:32.16 | tzanger | *nod* I want to get the test data down |
| 22:32.23 | letherglov | so ok |
| 22:32.31 | letherglov | I've got hylafax going into an adtran channel bank |
| 22:32.35 | letherglov | via a multitech modem |
| 22:32.37 | letherglov | into a t100p |
| 22:32.40 | letherglov | it works perfectly. |
| 22:32.53 | letherglov | I'm still a little curious about the whole gain issue though |
| 22:33.00 | letherglov | that's always got me a little bit worried |
| 22:33.04 | tzanger | yeah I have an Ascend Max I'd love to shove all faxes to and from |
| 22:33.14 | tzanger | you can telnet to port 9000 and get a modem |
| 22:33.14 | Blackvel | tzanger: t100p + channel bank? t100p is how many ports? |
| 22:33.17 | coppice | "and he can diagnose the HDLC from hearing the symptoms" you mean he can decode V.21 FSK by ear? :-) |
| 22:33.19 | letherglov | you think maybe the gain on the CB into the t400p or whatever is fucked? |
| 22:33.28 | letherglov | coppice, yes, he uses a harmonica |
| 22:33.29 | Blackvel | channelbank = t1 21 channels? |
| 22:33.33 | tzanger | just need to write a serial telnet driver that hylafax'll like... what I've found so far works backward to what I want |
| 22:33.36 | tzanger | Blackvel: 24 |
| 22:33.40 | Blackvel | right |
| 22:33.45 | tzanger | coppice: I thought you were able to do that already |
| 22:33.46 | letherglov | no, but more seriously |
| 22:33.51 | tzanger | or is that steve underwood, I always get you two mixed up |
| 22:33.54 | letherglov | it depends on v.34/vs v.17 |
| 22:33.55 | Blackvel | t100p was that t1 thing? |
| 22:33.58 | tzanger | Blackvel: yes |
| 22:33.59 | letherglov | and then ecm |
| 22:34.01 | letherglov | etc etc |
| 22:34.06 | Blackvel | uhm |
| 22:34.10 | tzanger | T100P is a T1 card. 24 DS0s |
| 22:34.28 | Blackvel | too bad to buy it, when you have no channel bank handy |
| 22:34.32 | letherglov | tzanger, coppice = underwood |
| 22:34.32 | Blackvel | hehe |
| 22:34.43 | tzanger | see I'm already confused by you two :-p |
| 22:34.47 | letherglov | haha |
| 22:35.04 | Blackvel | not sure why I have written down in my quote : buy voip telephones, so you can safe t100p + channel bank :)) |
| 22:35.18 | letherglov | Blackvel, yes, and no |
| 22:35.29 | letherglov | I still think voip phones, for the most part, are too immature |
| 22:35.37 | letherglov | at least with a standard old telephone |
| 22:35.39 | jarrod | dcd |
| 22:35.42 | letherglov | you won't sink too much $$$ into it |
| 22:36.04 | letherglov | Cisco, for example |
| 22:36.11 | letherglov | denies the existence of their 1st gen gear |
| 22:36.17 | Blackvel | hm |
| 22:36.24 | Blackvel | but your are more scalable |
| 22:36.28 | coppice | letherglov: its amazing how many native english speakers don't realise that underwood means coppice |
| 22:36.46 | Blackvel | if i would go the old german isdn way, i have max of 16 channels for 8 ports |
| 22:36.51 | Blackvel | that is 16 telephones |
| 22:37.01 | Blackvel | for the 17. telephone I buy a new cards |
| 22:37.07 | Blackvel | too bad :) |
| 22:37.25 | Blackvel | god thank you tzanger you told me, fax + tdm400p doesn't work really good |
| 22:37.28 | outtolunc | my question would be why in wctdm.c is readchunk defined twice <G> |
| 22:37.37 | tzanger | Blackvel: well it does and doesn't |
| 22:37.40 | tzanger | my particular setup doesn't |
| 22:37.44 | letherglov | coppice, yeah |
| 22:37.50 | outtolunc | <PROTECTED> |
| 22:37.51 | letherglov | coppice, how's your cantonese, btw? |
| 22:38.04 | Blackvel | have you connected asterisk box to telco then ? With some other card? |
| 22:38.23 | coppice | ngoh gwong dung wah m'ho |
| 22:38.37 | tzanger | coppice: I'm hoping to learn a lot from the zaptel dsp code |
| 22:38.42 | letherglov | hehe |
| 22:38.45 | tzanger | porting it to the OMAP architecture (C55x DSP) |
| 22:38.51 | tzanger | it won't be efficient for a while but it'll be a start |
| 22:39.04 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 22:39.13 | modulus_ | jbot babelfish de en du werdest eine krankenschwester brauchen |
| 22:39.13 | tzanger | basic things like echo cancel, tone detection/generation and some simple codec stuff |
| 22:39.19 | letherglov | talking about the the dsp code |
| 22:39.20 | tzanger | nothing like jumping right into the fire :-) |
| 22:39.25 | coppice | tzanger: most of spandsp uses floats, as they are faster than ints on a pentium. |
| 22:39.30 | letherglov | anyone know why sgi got involved with digium? |
| 22:39.46 | tzanger | coppice: I see |
| 22:39.55 | tzanger | I think the C55x is an integer DSP only |
| 22:39.59 | letherglov | eh? since when did an intel floatig point unit not suck? |
| 22:40.02 | tzanger | have to take another look, I honestly don't remember |
| 22:40.11 | coppice | in what way are sgi involvedwith digium? |
| 22:40.15 | Blackvel | hm, whats best to tell a customer when something like this happens, e.g you cant receive faxes? :) |
| 22:40.18 | letherglov | they're doing the VON show with them |
| 22:40.22 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
| 22:40.33 | SuPrSluG | helo |
| 22:40.42 | letherglov | seems odd |
| 22:40.52 | letherglov | unless they want to build a super altix based asterisk box |
| 22:40.54 | epoch | man, this is so weird... this polycom IP500 keeps freezing up on me ;/ |
| 22:40.55 | modulus_ | jbot babelfish de en du bist eine, kleine, dummer schweinehund |
| 22:41.09 | letherglov | epoch, get the latest firmware? |
| 22:41.12 | coppice | letherglov: you can do more floating point than integer operations on a pentium. also, there is no saturation logic for integer |
| 22:41.12 | SuPrSluG | my moh on my mepis box sounds like a jet engine. what up wit dat? |
| 22:41.29 | coppice | tzanger: what are you trying to use OMAP for? |
| 22:41.34 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 22:41.42 | ManxPower | SuPrSluG, usually happens if you are not running ,pg123 0.59r |
| 22:41.43 | epoch | letherglov: not yet -- 1.3.1 |
| 22:41.48 | epoch | letherglov: I'm about to upgrade to 1.4.1 though |
| 22:41.52 | letherglov | epoch, good |
| 22:42.00 | letherglov | so anyway, the dsp stuff |
| 22:42.09 | epoch | boy I hope it fixes it... this phone's the CEO's |
| 22:42.10 | letherglov | I was looking into Cg and Sh for GPU calculations |
| 22:42.12 | letherglov | for the heck of it |
| 22:42.13 | epoch | and he's _pissed_ |
| 22:42.18 | SuPrSluG | ManxPower: i'll check the version |
| 22:42.19 | letherglov | it's interesting... |
| 22:42.28 | letherglov | but I think it needs to be PCI or PCI-X for it to work, not AGP |
| 22:42.40 | letherglov | and, it's crazy ass where it treats it like a shader in the PBuffer |
| 22:42.58 | letherglov | I got confused writing a blue screen software video app--let alone getting the data back. |
| 22:43.37 | letherglov | epoch, hmm |
| 22:43.44 | coppice | I've never tried, but I hear its very slow getting the data back from the GPU. it wasn't designed for that |
| 22:43.53 | letherglov | sounds like you need to tape a dilbert to it |
| 22:43.57 | epoch | hahaha |
| 22:43.59 | letherglov | just to avoid getting fired |
| 22:43.59 | letherglov | ;-) |
| 22:44.00 | epoch | indeed. |
| 22:44.13 | letherglov | or better yet |
| 22:44.16 | letherglov | go get some string |
| 22:44.17 | letherglov | a bell |
| 22:44.18 | letherglov | and a tin can |
| 22:44.33 | letherglov | and hook it up on his desk |
| 22:44.35 | letherglov | and see what he says |
| 22:44.37 | *** join/#asterisk aspworld (~sivana@165.154.13.35) |
| 22:44.44 | aspworld | heh |
| 22:45.10 | letherglov | coppice, yeah, I hear the AGP bus isn't so great for GPU->CPU |
| 22:45.24 | letherglov | but I see the PCI-X cards that steal main system ram for video memory |
| 22:45.34 | letherglov | so that's gotta be somewhat ok...I mean, the i810 did that too, but it wasn't 3-d |
| 22:45.44 | aspworld | ^A |
| 22:46.29 | *** join/#asterisk florz (nobody@odnb-d9baa5eb.pool.mediaWays.net) |
| 22:46.36 | coppice | I wish the SIMD in pentiums didn't suck so badly at DSP |
| 22:47.01 | florz | coppice: Why does it? |
| 22:47.37 | coppice | alignment is the nastiest part. it really screws adaptive things like EC |
| 22:47.52 | letherglov | even worse--- |
| 22:47.53 | aspworld | hrm |
| 22:48.01 | letherglov | intel's only chip to move to on the desktop is the lv pentium m |
| 22:48.11 | letherglov | and it's missing half the instructions for mmx, sse, sse2, simd, etc. |
| 22:48.42 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 22:48.42 | fa | cypromis are you alive? |
| 22:48.46 | florz | coppice: IC |
| 22:49.40 | coppice | pentium M does mmx, sse and sse2 |
| 22:51.26 | fa | florz Hi. Did you test Zaphfc with more then 3/4 zaphfc cards (one port for one card) with BRI lines. |
| 22:51.34 | Bile_One | Does anyone in here know how to make perl install a module or how to reconfigure all the ftp locations? |
| 22:54.20 | zno | Bile_One: I think #perl is a safe bet |
| 22:54.31 | Bile_One | I think I found too! |
| 22:54.37 | Bile_One | Thanx zno |
| 22:55.43 | florz | fa: Nope. But I think in a system that has no trouble with four cards, five or six usually will do fine as well. More won't fit into a single box anyway =:-) |
| 22:55.58 | ManxPower | Pentium M: the "M" stands for "money"! |
| 22:56.03 | fa | florz Did you test four? |
| 22:56.21 | florz | fa: yep, as described on the web page =:-) |
| 22:56.42 | florz | fa: More exactly: It's running productively with four cards. |
| 22:56.51 | fa | ;] |
| 22:57.15 | fa | florz How Can i check if zap channel if free. and if not to send playback to IAX user |
| 22:57.40 | fa | In moment when he (iax user) is trying co connect with cellular phone by zap |
| 22:59.48 | *** join/#asterisk robf (~robf@208.188.247.3) |
| 23:00.30 | *** join/#asterisk __a (user@193.140.215.2) |
| 23:04.34 | Qwell | stupid question, can Goto take you to a different context? |
| 23:04.49 | epoch | awesome. the polycom 1.4.1 SIP firmware fixed my problems :) |
| 23:04.52 | toddf | Goto(context,extension,level) |
| 23:04.58 | fa | Qwell of course |
| 23:05.10 | Qwell | yeah...show application goto, I should have done that first |
| 23:05.11 | Qwell | Thanks |
| 23:05.58 | florz | fa: Sorry, I don't get it. What do you want to do? |
| 23:08.25 | SuPrSluG | ManxPower: installed source. worked after i restarted. doh! |
| 23:08.30 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 23:08.41 | *** part/#asterisk __a (user@193.140.215.2) |
| 23:10.22 | florz | now anyone in here who has an idea why this kind of message starts appearing regularly after some time of continuous operation?: Feb 4 15:45:30 WARNING[31124]: chan_zap.c:7409 zt_pri_error: PRI: !! Got reject for frame 98, but we have nothing -- resetting! |
| 23:11.08 | silik0n | afds |
| 23:11.19 | florz | lhjk |
| 23:11.20 | BoRiS | For anyone doing realtime stuff, please check out http://bugs.digium.com/bug_view_page.php?bug_id=0003509 |
| 23:11.55 | BoRiS | MWI with Realtime! |
| 23:12.47 | JerJer | bleh |
| 23:12.52 | JerJer | realtime is not the answer people |
| 23:14.14 | BoRiS | JerJer: it is :) |
| 23:17.35 | *** join/#asterisk techie (gus@38.119.236.18) |
| 23:18.09 | sivana | aspworld: test? |
| 23:18.30 | terrapen | holy shit, my nufone toll-free rules |
| 23:18.30 | aspworld | siv: ya right |
| 23:18.51 | aspworld | help |
| 23:20.10 | Blackvel | n8 all |
| 23:21.32 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 23:23.29 | *** join/#asterisk HenryTheBIG (~chariga@160.79.172.147) |
| 23:23.29 | HenryTheBIG | Hi |
| 23:24.07 | sivana | shoots.. how do I detach from a screen but leave it going? |
| 23:24.18 | *** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 23:24.43 | RaYmAn-Bx | sivana: ctrl+a then d |
| 23:24.56 | sivana | thank you |
| 23:28.46 | lohelle | norwegian sounds anyone? need them, but I am not the right person to make them myself... |
| 23:29.00 | HenryTheBIG | I have a hard time with one * box |
| 23:29.26 | HenryTheBIG | I make the new cvs upgrade |
| 23:29.26 | HenryTheBIG | and now doesn't start at all |
| 23:29.45 | HenryTheBIG | I try to install the v1 version but I get the same error |
| 23:29.53 | toddf | here's a novel concept. test on a non production box before updating a production box. |
| 23:30.48 | HenryTheBIG | :( |
| 23:30.50 | HenryTheBIG | Thanks |
| 23:30.53 | heath__ | what does a t3 line run per month these days? |
| 23:32.07 | jas_williams | HenryTheBIG: If you do an asterisk -vvvc where does it fall over ? |
| 23:32.37 | jas_williams | HenryTheBIG: And what is the error ? |
| 23:32.49 | HenryTheBIG | one sec |
| 23:32.52 | HenryTheBIG | I will pasrw |
| 23:32.54 | HenryTheBIG | I will paste |
| 23:33.51 | HenryTheBIG | <PROTECTED> |
| 23:33.51 | HenryTheBIG | Warning, flexibel rate not heavily tested! |
| 23:33.51 | HenryTheBIG | Segmentation fault |
| 23:33.51 | HenryTheBIG | [root@localhost root]# Ouch ... error while writing audio data: : Broken pipe |
| 23:33.51 | jas_williams | wuse pastebin |
| 23:34.16 | zimdog | Does anyone use vonage with * ? |
| 23:34.18 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 23:34.36 | zimdog | for a trunk |
| 23:35.10 | jas_williams | Ok looks like a problem with Music on Hold may be, what version of * are you now attempting to run CVS or Stable ? |
| 23:36.04 | jas_williams | try mpg123 and see what version is returned ? |
| 23:36.10 | HenryTheBIG | stable |
| 23:36.21 | lohelle | I have a "problem" when compiling asterisk/zaptel. I always have to compile both of them twice to get meetme to work.. (and mpg123) |
| 23:36.39 | *** join/#asterisk RoyKa (~roy@83.80-203-29.nextgentel.com) |
| 23:36.40 | ManxPower | lohelle, Always install Zaptel FIRST |
| 23:36.48 | HenryTheBIG | mpg123 Version 0.59r |
| 23:36.53 | jas_williams | Compile and install zaptel first |
| 23:37.35 | jas_williams | HenryTheBIG: Can you post your musiconhold.conf to pastebin.ca |
| 23:38.11 | lohelle | hmm.. I think that is what I use to do.. hmm.. yes.. but no problem of course.. I just run make on zaptel first.. then on asterisk.. then zaptel and then asterisk.. strange.. :) |
| 23:38.13 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 23:38.36 | HenryTheBIG | [classes] |
| 23:38.36 | HenryTheBIG | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 23:39.15 | HenryTheBIG | this is all I have in musiconhold.conf |
| 23:40.16 | lohelle | I'm getting more and more impressed with asterisk.. I actually use it to dial in to the office (from home) and run external scripts to reboot servers etc.. |
| 23:40.24 | jas_williams | looks ok comment out the default to ;default => quietmp3:/var/lib/asterisk/mohmp3 and see if asterisk will start |
| 23:41.12 | HenryTheBIG | nope, the same :( |
| 23:41.16 | jas_williams | ok |
| 23:41.43 | lohelle | the thing I'm not getting to work is to make a call from outside asterisk (script) and make it go to [context] s,1, -> s,2 etc.. |
| 23:42.07 | lohelle | that is auto dial and play sound when answer.. |
| 23:42.08 | dca[laptop] | anyone know if the g729 register utility works on FreeBSD yet? |
| 23:42.36 | tessier_ | Linux only afaik |
| 23:43.54 | dca[laptop] | digium was working on it, something about the MAC, wonder if it's done yet... |
| 23:45.10 | kFuQ | ~docs |
| 23:45.13 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 23:45.30 | jas_williams | HenryTheBIG: Delete all of the files in /usr/lib/asterisk/modules/ and then run make install for asterisk |
| 23:46.42 | jas_williams | lohelle: look at http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 23:47.04 | HenryTheBIG | ok |
| 23:49.14 | Qwell | HenryTheBIG: If that doesn't work, run asterisk with -g, let it crash, run gdb /path/to/asterisk /path/to/corefile, within gdb run a bt |
| 23:49.59 | HenryTheBIG | Qwell: I will do it. Now I recompile the asterisk |
| 23:50.45 | kFuQ | what's the difference between loopstart, koolstart and groundstart protocols ? |
| 23:52.32 | jas_williams | ~help |
| 23:55.25 | ctooley | Feb 4 17:54:37 DEBUG[27423]: chan_sip.c:1580 update_user_counter: Call from user 'CxH96CfE39' is 1 out of 0 |
| 23:55.38 | ctooley | anyone got a clue stick to beat me with? |
| 23:59.44 | kFuQ | ctooley: time for someone to smack ya upside the head with the clue-by-fourŪ ? |