| 00:02.05 | joaovianna | What is the main different between 1.2.xxx and 1.4.xxx ? |
| 00:02.23 | hardwire | 1.3.xxx |
| 00:03.37 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 00:04.03 | [TK]D-Fender | joaovianna, 0.2.xxx |
| 00:05.15 | toerkeium | hello people |
| 00:05.16 | toerkeium | I am trying to Originate a call from the API Manager, and I need the first call to be a mobile phone.. does anyone know how should I specify the outgoing trunk? for * extensiones I use: fputs($socket, "Channel: 1558\r\n"); |
| 00:05.40 | toerkeium | but if I out my home number instead of 1558 it won't work |
| 00:06.16 | [TK]D-Fender | toerkeium, that is not a channel, that is just a NUMBER |
| 00:06.26 | *** join/#asterisk kairu0 (n=kairu@softbank221108010089.bbtec.net) |
| 00:06.41 | [TK]D-Fender | toerkeium, you would enter a channelt he exact same way you would call it in app_dial |
| 00:06.56 | *** join/#asterisk saftsack (n=saftsack@pD9E0612C.dip.t-dialin.net) |
| 00:07.20 | toerkeium | how should I enter the number? I am confused |
| 00:08.23 | [TK]D-Fender | toerkeium, You need to put what you would do in Dial() there |
| 00:09.25 | toerkeium | ohh.. let me try |
| 00:13.43 | toerkeium | working!! thanks |
| 00:14.24 | *** part/#asterisk kairu0 (n=kairu@softbank221108010089.bbtec.net) |
| 00:15.24 | robin_sz | bah, going to have to find an active BRI card ... misdn and an hfc card is just not doing it for me |
| 00:15.47 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 00:17.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:26.01 | *** join/#asterisk Here_And_There (n=Here_And@pool-71-244-103-43.phlapa.fios.verizon.net) |
| 00:29.47 | *** part/#asterisk kiscokid (n=Ron@adsl-66-122-34-11.dsl.sntc01.pacbell.net) |
| 00:32.55 | *** join/#asterisk Fieldy (i=gW1Mnbub@gentoo/contributor/Fieldy) |
| 00:34.21 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:41.11 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:41.30 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 00:42.31 | toerkeium | [TK]D-Fender, do you know if it's possible to check the status of a Channel (the number which was called first, before calling to the * exten) ? |
| 00:43.50 | toerkeium | for example, I am calling to the number 555 6699, and then when this number pickups the phone, I call to exten 1558 |
| 00:44.23 | toerkeium | what I want to do, is to show to the person who is triggering this using a web broser, the state of the comunication. lets say: |
| 00:45.00 | toerkeium | 1.- Pick up your phone - 2.- Thank you, now we are connecting you with exten 1558 - 3.- Hangup |
| 00:45.05 | toerkeium | understand what I mean? |
| 00:47.10 | toerkeium | I found ExtensionStatus for Extensions, but can't find how to check a Channel status |
| 00:55.17 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 00:57.41 | MrWup | say if i do: exten => 2,4,Dial(${RECEPTION},20,Tt) |
| 00:57.53 | MrWup | how do i tell it to do something after call to RECEPTION has been hung up? |
| 00:58.18 | blitzrage | check ${DIALSTATUS} |
| 00:58.23 | blitzrage | or use the 'h' extension |
| 00:58.36 | blitzrage | if the call was answered, then 'h' is your only option |
| 00:58.51 | MrWup | so |
| 00:59.06 | MrWup | erm |
| 00:59.10 | MrWup | oh ok |
| 01:09.19 | *** join/#asterisk NEURALWIND (n=root@201.222.93.233) |
| 01:09.24 | *** join/#asterisk red9012 (n=marc3234@206-248-160-251.dsl.teksavvy.com) |
| 01:09.53 | NEURALWIND | [root@localhost zaptel-1.2.17.1]# make install |
| 01:09.53 | NEURALWIND | grep: /include/linux/autoconf.h: No such file or directory |
| 01:09.53 | NEURALWIND | You do not appear to have the sources for the 2.6.9-42.EL kernel installed. |
| 01:10.08 | NEURALWIND | how can i fix that? |
| 01:10.31 | Nivex | NEURALWIND: do you have kernel-devel installed? |
| 01:11.02 | NEURALWIND | yes |
| 01:11.15 | NEURALWIND | [root@localhost zaptel-1.2.17.1]# rpm -q kernel-devel |
| 01:11.15 | NEURALWIND | kernel-devel-2.6.9-42.0.10.EL |
| 01:11.22 | blitzrage | you don't have the right kernel-devel |
| 01:11.39 | blitzrage | does that match what is in /etc/redhat-version ? |
| 01:11.45 | blitzrage | (or whatever that file is) |
| 01:11.47 | MrWup | hmm. i have a slight problem... PSTN user calls SIP1. SIP1 then transfers the PSTN user using the asterisk attended transfer (*2) to SIP2. When PSTN user dialled SIP1 i added a record to MySQL saying call begun. but how do i get when the SIP2 hangs up? if i use the h extension all i get is when the zombie channel hangs up (i.e. when SIP1 is cut off) |
| 01:11.52 | NEURALWIND | let me check |
| 01:11.58 | blitzrage | do you have SMP kernel and non-SMP kernel sources installed? |
| 01:12.04 | blitzrage | there is a mis-match there you need to resolve |
| 01:12.14 | NEURALWIND | blitzrage how |
| 01:12.19 | MrWup | (so that i can add a mysql record which says "call ended") |
| 01:13.03 | NEURALWIND | theres nothing on my /etc folder |
| 01:13.23 | key2 | http://www.youtube.com/watch?v=o1O7ZvDFP_8&NR=1 |
| 01:14.07 | NEURALWIND | blitzrage how do i fix that? |
| 01:16.35 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 01:32.41 | *** join/#asterisk dakmatt (n=dakmatt@60.50.56.210) |
| 01:32.43 | dakmatt | hi |
| 01:35.40 | dakmatt | have anyone trying configure using the LinkSys(SIPURA) 3102A ? |
| 01:36.54 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:48.50 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:52.19 | LeddyHM | man, nufone sucks |
| 01:52.24 | LeddyHM | still not able to make a call |
| 01:53.29 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:54.31 | etfonhomey | wow |
| 01:54.40 | etfonhomey | Are you talking to anyone there? |
| 01:54.54 | LeddyHM | to the person not returning my email, yes |
| 01:55.32 | etfonhomey | No phone number? |
| 01:55.44 | LeddyHM | calling now |
| 01:55.53 | LeddyHM | since they supposedly close in 5 mins |
| 01:56.24 | LeddyHM | "nobody is available to take your call at the moment" |
| 01:56.33 | LeddyHM | I just need to test dtmf |
| 01:56.38 | LeddyHM | SWEARS |
| 01:56.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:57.04 | LeddyHM | speak of the devil |
| 01:57.10 | LeddyHM | shido: still at nufone? |
| 01:57.23 | shido6 | :) |
| 01:57.26 | shido6 | whats up? |
| 01:57.32 | LeddyHM | my shit ain't working |
| 01:57.37 | LeddyHM | and greg hasn't called me back |
| 01:57.41 | LeddyHM | <-- new customer |
| 01:57.49 | shido6 | who are you? |
| 01:57.55 | shido6 | username [?] |
| 01:58.02 | blitzrage | shido6 is Greg :) |
| 01:58.06 | LeddyHM | haha |
| 01:58.21 | LeddyHM | pm |
| 01:58.21 | shido6 | 603 ? |
| 01:58.22 | blitzrage | shido6: I wouldn't help him... you should have seen the smack he was talking before you got here! :) |
| 01:58.51 | LeddyHM | yeah |
| 01:59.34 | shido6 | I'll wake the dead. |
| 01:59.43 | LeddyHM | please do |
| 01:59.55 | LeddyHM | or if you have a working account I can use temporarily |
| 01:59.58 | dakmatt | have anyone trying configure using the LinkSys(SIPURA) 3102A ? |
| 02:00.04 | LeddyHM | I just need to call an 877 number |
| 02:00.35 | shido6 | I suggested using IAX when you called earlier. |
| 02:00.42 | shido6 | temporarily |
| 02:01.02 | LeddyHM | I am not able to completely change my dialplan |
| 02:01.09 | LeddyHM | I need sip |
| 02:05.34 | dakmatt | can i used SPA3102 to have asterisk server dial-out to PSTN ? |
| 02:12.23 | LeddyHM | hmm |
| 02:12.28 | LeddyHM | iax works |
| 02:12.34 | LeddyHM | wth could be wrong with sip :) |
| 02:12.48 | LeddyHM | dropkick the proxy |
| 02:14.34 | LeddyHM | hmm it's rejecting ulaw |
| 02:14.42 | LeddyHM | that appears to be the common thread |
| 02:15.05 | LeddyHM | May 5 21:14:08 WARNING[5748]: chan_iax2.c:7126 socket_read: Call rejected by 66.225.202.80: Unable to negotiate codec |
| 02:15.05 | LeddyHM | <PROTECTED> |
| 02:15.05 | LeddyHM | <PROTECTED> |
| 02:15.18 | LeddyHM | now that I am talking to myself |
| 02:17.12 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:17.58 | tzafrir | I managed to find two live CDs, and both we quite broken. None of them even booted in qemu |
| 02:19.09 | tzafrir | thus the zaptel build problem on centos is still not fixed |
| 02:19.25 | tzafrir | painful platform |
| 02:19.46 | LeddyHM | I had no problems with it |
| 02:19.51 | LeddyHM | unless you're talking about 5 |
| 02:23.05 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 02:23.08 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 02:23.13 | mattwj2005 | hey guys |
| 02:23.31 | mattwj2005 | this isn't an asterisk question but rather a telephony question |
| 02:23.51 | mattwj2005 | do you guys know any numbers to use for testing out a fax machine? |
| 02:23.58 | mattwj2005 | USA please :) |
| 02:25.21 | Putzz | send or receive? |
| 02:25.50 | mattwj2005 | both would be nice |
| 02:26.10 | dakmatt | hi, can SPA3102A device allow asterisk servers make an out-bound call to PSTN ? |
| 02:26.10 | mattwj2005 | basically I want to set up a modem for faxing purposes |
| 02:26.14 | Putzz | u can find a fax on demand somewhere |
| 02:26.25 | Putzz | or u can use the fax online trials |
| 02:26.30 | mattwj2005 | fax on demand |
| 02:26.30 | LeddyHM | or fax to email service |
| 02:26.36 | mattwj2005 | okay |
| 02:26.49 | mattwj2005 | well I'll check out the wiki and google |
| 02:27.08 | mattwj2005 | I got a POTS line....I thought it would be great to fax for a change :) |
| 02:27.28 | LeddyHM | we use hylafax/iaxmodem(s) |
| 02:28.02 | mattwj2005 | hmmm |
| 02:28.13 | mattwj2005 | that would be cool too...interface it will asterisk |
| 02:28.37 | LeddyHM | works great |
| 02:28.43 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-49-79.socal.res.rr.com) |
| 02:29.27 | Putzz | hylafax is the best |
| 02:29.42 | LeddyHM | yeah it works great |
| 02:30.19 | mattwj2005 | everyone once and a while someone is like please fax me |
| 02:30.40 | mattwj2005 | this will save me from running up town |
| 02:31.02 | LeddyHM | just setup hylafax |
| 02:31.05 | tzafrir | anybody here with centos5 and willing to spend some time on building zaptel? |
| 02:31.12 | mattwj2005 | plus I am using this computer for folding@home anyways |
| 02:31.21 | LeddyHM | you can setup a printer and print to it which in turn causes a fax to be sent |
| 02:31.35 | mattwj2005 | nice :) |
| 02:31.36 | LeddyHM | tza: use c4.4 ;) |
| 02:32.01 | LeddyHM | I'm in the process of converting from gentoo to c4.4 |
| 02:32.05 | tzafrir | I have that already in my vserver... |
| 02:32.05 | mattwj2005 | man I am being a wuss right now :( |
| 02:32.10 | LeddyHM | and am amazed how easy it was |
| 02:32.12 | mattwj2005 | I am install AsteriskNOW |
| 02:32.17 | tzafrir | LeddyHM, why would you want to do that? |
| 02:32.30 | LeddyHM | c4 > gencrap |
| 02:32.34 | tzafrir | And why Centos4.4 when 4.5 and 5.0 were released? |
| 02:32.48 | LeddyHM | centos 4.5 isn't relased |
| 02:32.49 | mattwj2005 | I am being lazy....I should be building it from scratch |
| 02:32.52 | LeddyHM | only rhel 4.5 |
| 02:33.11 | LeddyHM | it will be a few weeks before c4.5 is out |
| 02:33.12 | tzafrir | Centos5 is. |
| 02:33.18 | LeddyHM | and when it is I will upgrade |
| 02:33.34 | LeddyHM | not going to touch c5 for a production environment |
| 02:33.47 | LeddyHM | your comments just solidy my point ;) |
| 02:33.52 | LeddyHM | solidify |
| 02:34.07 | mattwj2005 | 5.0 is out |
| 02:34.09 | mattwj2005 | http://en.wikipedia.org/w/index.php?title=CentOS&oldid=128158890 |
| 02:34.13 | LeddyHM | I know c5 is |
| 02:34.16 | LeddyHM | but not 4.5 |
| 02:34.22 | mattwj2005 | oh :P |
| 02:34.39 | tzafrir | This is so much unlike the transision in Debian |
| 02:34.44 | LeddyHM | I'm like smart and stuff |
| 02:35.03 | mattwj2005 | like smart? lol :) |
| 02:35.13 | LeddyHM | yeah |
| 02:35.15 | mattwj2005 | is that like kind of smart ;) |
| 02:35.43 | tzafrir | "like smart" is what yum hopes to be |
| 02:35.51 | mattwj2005 | lol |
| 02:35.55 | mattwj2005 | :) |
| 02:36.25 | mattwj2005 | just teasing ya....it is Saturday night after all |
| 02:36.36 | LeddyHM | been a rough week :) |
| 02:36.43 | mattwj2005 | not too bad |
| 02:36.58 | mattwj2005 | fix this do that |
| 02:37.10 | LeddyHM | no, mine. |
| 02:37.11 | mattwj2005 | nothing too stressful |
| 02:37.22 | mattwj2005 | oh why has it been rough? |
| 02:37.47 | LeddyHM | cbeyond is pissing me off |
| 02:37.54 | LeddyHM | dtmf issues |
| 02:38.06 | mattwj2005 | sorry to hear that |
| 02:38.52 | mattwj2005 | I am sure you'll get it figured out shortly |
| 02:39.30 | LeddyHM | new provider ;) |
| 02:39.34 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-92.plaicmtc01.tx.dh.suddenlink.net) |
| 02:40.04 | mattwj2005 | don't they confirm to standards or what is the deal? |
| 02:40.27 | LeddyHM | they are blaming everything |
| 02:40.34 | LeddyHM | they say we need to upgrade to 1.4 |
| 02:40.44 | LeddyHM | to be able to change our tone timing |
| 02:41.13 | mattwj2005 | bummer |
| 02:41.35 | mattwj2005 | or as Bill and Ted would say Bogus ;) |
| 02:41.42 | LeddyHM | I can see the tones going in from phone, and going out to them |
| 02:41.53 | LeddyHM | so somewhere something is getting lost in translation |
| 02:42.17 | mattwj2005 | anyone have any ideas? |
| 02:42.26 | mattwj2005 | I don't know much about dtmf |
| 02:43.49 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:44.04 | mattwj2005 | are you finding any help on the wiki? |
| 02:44.16 | LeddyHM | I was able to make a call with nufone (finally) and ad no issue at all |
| 02:44.28 | LeddyHM | but it was iax and gsm, vs sip and ulaw |
| 02:45.10 | mattwj2005 | hmmm |
| 02:46.11 | LeddyHM | not sure if that makes any difference |
| 02:47.01 | mattwj2005 | yeah I really don't know |
| 02:47.06 | shido6 | grr |
| 02:47.13 | LeddyHM | shido: exactly! |
| 02:47.37 | mattwj2005 | I recommend drinking a nice Diet Dew and relax :) |
| 02:47.48 | shido6 | jer isnt answering the batphone :) |
| 02:48.14 | LeddyHM | he was here earlier |
| 02:49.43 | jer | shido6, ? |
| 02:49.54 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 02:50.12 | LeddyHM | I think he wants you to fix sip |
| 02:50.16 | LeddyHM | it no workie |
| 02:52.17 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 02:56.07 | DocHolliday | any wholesale providers in here? |
| 02:57.12 | mattwj2005 | http://www.voip-info.org/wiki/view/Asterisk+DTMF |
| 02:57.30 | mattwj2005 | http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode |
| 02:57.44 | mattwj2005 | http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode |
| 02:57.52 | mattwj2005 | have you looked at any of these pages? |
| 02:58.07 | LeddyHM | I've looked at a lot :| |
| 02:58.18 | LeddyHM | apparantely firefox needs to update |
| 02:58.28 | LeddyHM | so I'll look in a sec and see if they look familiar |
| 02:58.37 | mattwj2005 | oh I am not a lot of help...but I figured it was worth a shot |
| 02:58.39 | mattwj2005 | :) |
| 02:59.30 | mattwj2005 | here is a good note too |
| 02:59.34 | mattwj2005 | Inband DTMF won't work unless the codec is ulaw or alaw (G711). Use out of band DTMF aka rfc2833 or info. |
| 02:59.34 | mattwj2005 | Also, dtmfmode=info does not work with Asterisks voicemail system (this statement is most likely incorrect). |
| 03:01.34 | LeddyHM | we use rfc2833 |
| 03:01.36 | *** join/#asterisk raptorra1 (n=rathomps@cpe-66-25-25-138.houston.res.rr.com) |
| 03:01.49 | LeddyHM | and ulaw |
| 03:02.11 | LeddyHM | which according to the first document is the best case ;) |
| 03:04.58 | mattwj2005 | okay |
| 03:05.29 | mattwj2005 | if I learned anything from school....RTFM = read the freckin' manual |
| 03:05.33 | mattwj2005 | :) |
| 03:05.43 | LeddyHM | I learned that from irc ;) |
| 03:05.54 | mattwj2005 | lol |
| 03:06.29 | JT | error |
| 03:06.32 | JT | :P |
| 03:09.05 | LeddyHM | I just need to compare apples to apples with nufone |
| 03:09.16 | LeddyHM | however the iax portion made it look promising |
| 03:14.37 | LeddyHM | any difference in ulaw and gsm that could affect dtmf detection? |
| 03:14.53 | shido6 | dtmfmode |
| 03:15.28 | LeddyHM | that didn't change |
| 03:17.25 | JT | LeddyHM: if using inband, yes |
| 03:17.35 | JT | dtmf signalliing inband only works with g.711 |
| 03:17.37 | LeddyHM | rfc2833 |
| 03:18.00 | LeddyHM | how about sip vs iax? |
| 03:18.41 | JT | iax has its own out of band signalling method |
| 03:20.02 | mattwj2005 | ;) |
| 03:20.05 | LeddyHM | ok |
| 03:20.13 | LeddyHM | so I need sip to work then |
| 03:20.26 | LeddyHM | I'm gonna try gsm with nufone |
| 03:20.39 | mattwj2005 | so is nufone sip only? |
| 03:20.58 | shido6 | no |
| 03:21.09 | shido6 | SIP & IAX2 |
| 03:21.27 | mattwj2005 | so is IAX2 working? |
| 03:21.32 | JT | LeddyHM: why do you need sip to work? |
| 03:21.33 | mattwj2005 | if so why not just use that? |
| 03:21.47 | shido6 | iax2 is working but if you signed up in the last 3 days for sip the proxy doesnt recognize you yet |
| 03:22.58 | mattwj2005 | iax2 punches through firewalls better anyways |
| 03:23.15 | LeddyHM | I'm testing our * box |
| 03:23.22 | LeddyHM | I need an apples to apples comparison |
| 03:23.50 | LeddyHM | expecially if iax2 uses a different methond for dtmf |
| 03:23.55 | LeddyHM | -x+s |
| 03:24.18 | LeddyHM | I was able to get iax+gsm to work with nufone, but nothing with sip |
| 03:25.55 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.34 | LeddyHM | shido: still no ideas yet? |
| 03:27.41 | kiscokid | potentially silly question: can a sip phone be connected over an IAX trunk? |
| 03:27.48 | shido6 | no. When Jer gets back he'll take a peek |
| 03:28.12 | LeddyHM | k |
| 03:28.21 | LeddyHM | I'm getting ready for bed |
| 03:28.29 | LeddyHM | so just email me with any updates |
| 03:28.33 | shido6 | sure |
| 03:28.51 | LeddyHM | fyi, I can get iax+gsm to work, but not iax+ulaw, or sip |
| 03:29.31 | shido6 | IAX+ ulaw works |
| 03:29.41 | LeddyHM | as of? |
| 03:29.59 | LeddyHM | I tried it recently |
| 03:31.24 | shido6 | :) mind if i shorten your password? |
| 03:31.45 | LeddyHM | to? |
| 03:31.51 | LeddyHM | 8? |
| 03:31.52 | shido6 | im going to flip if its just a password issue. |
| 03:32.04 | shido6 | LOL hold on. |
| 03:36.03 | LeddyHM | same pwd for the iax connection though |
| 03:36.09 | LeddyHM | thinking sip is a little pickier? |
| 03:36.15 | shido6 | two different systems |
| 03:40.34 | LeddyHM | cli needs a "clear" |
| 03:40.44 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 03:41.46 | raptorra1 | doesn't <ctrl+l> that a lowercase L clear it |
| 03:41.59 | LeddyHM | it does |
| 03:42.00 | LeddyHM | cool |
| 03:42.22 | raptorra1 | LeddyHM: that seems to work just about anywhere in linux |
| 03:42.55 | LeddyHM | I've always used "clear" |
| 03:43.07 | LeddyHM | <-- old school |
| 03:43.58 | raptorra1 | LeddyHM: can't laugh b/c I used to do the same... over time started learning/picking up the short cuts and using those instead |
| 03:44.13 | raptorra1 | I sure hit "k" and "j" to navigate around in a lot of editors.... |
| 03:51.58 | dc3aes | heres something I have found little documentation on is the asterisk process.. I am running it in interactive console mode right now.. but eventually I would like to have it run unattended (for lack of better term) and simply log to a file.. but then how do i reload dialplan, etc? |
| 03:52.16 | JT | extensions reload |
| 03:52.21 | shido6 | asterisk -rx 'releoad' works |
| 03:52.27 | shido6 | if you spell it right |
| 03:52.32 | dc3aes | hehee |
| 03:52.56 | JT | you don't need to reload the whole of asterisk just for a dialplan change |
| 03:53.15 | dc3aes | ok ill poke around.. howcome the documentation says "old documentation" and then everything says refer to the o'reilly book? i mean.. not that the book isnt great it doesnt go into detail with some things :P |
| 03:53.40 | dc3aes | JT: so if its running, not attached to a console.. how do we send it a command? like shido6 said? |
| 03:53.50 | shido6 | blitz is writing a new book |
| 03:53.59 | dc3aes | oh! |
| 03:54.07 | hads | dc3aes: Umm, like shido6 said |
| 03:54.31 | dc3aes | ok.. i think i just wrapped my head around this.. i can connect to the running process and attach at will |
| 03:54.59 | dc3aes | sorry guys.. gotta admit this is a different way of doing things.. but definitely more interesting |
| 03:55.56 | dc3aes | its funny how i can read and read and try.. and not figure something out.. then break down to whining on irc and i figure it out on my own lol |
| 03:56.15 | shido6 | connecting to the collective helps |
| 03:56.39 | shido6 | a "." properly placed can begin a whole new way of stinking |
| 03:56.55 | shido6 | is that a period a dot or a rat dropping... |
| 03:57.04 | shido6 | ahh the possibilities. |
| 03:57.24 | dc3aes | i like to sprinkle periods in my sentences on IRC, never figured out why |
| 03:57.31 | threat | I prefer CTRL+l |
| 03:58.53 | LeddyHM | dc3: are you using gsm? |
| 03:58.58 | threat | ok so blitzrage reckons I need to recompile asterisk to change the rtp / rfc2833 payload type to 96. Can any one else confirm that? |
| 03:59.14 | dc3aes | LeddyHM: it appears I am.. I have everything set to auto but I see in the logs many of the calls are opting to gsm |
| 03:59.39 | LeddyHM | I am able to get iax+gsm, not ulaw though |
| 04:00.10 | dc3aes | hmmm |
| 04:00.23 | LeddyHM | they are still working on the sip issue |
| 04:00.53 | LeddyHM | it's been a long day, and I have to get up early so I go sleep |
| 04:04.59 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:06.33 | *** join/#asterisk Dovid (n=Dovid@ool-43511bec.dyn.optonline.net) |
| 04:06.56 | Dovid | anyone knows the latest version of wanpipe for zaptel 1.2.17.1 ? |
| 04:17.31 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-92.plaicmtc01.tx.dh.suddenlink.net) |
| 04:24.13 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 04:54.32 | *** join/#asterisk ESCulapio__ (n=The@186stb68.codetel.net.do) |
| 05:01.16 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 05:01.35 | Sargun | Does anyone here have access to a recent version of Polycom's SIP/BootROM software? |
| 05:03.14 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 05:03.18 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:05.49 | hads | Sargun: You reseller will |
| 05:06.01 | hads | *Your |
| 05:07.01 | Sargun | Voip-supply does, but they won't talk to me |
| 05:07.06 | Sargun | I've called them a dozen times |
| 05:07.12 | Sargun | and e-mailed them three or four times |
| 05:09.25 | [TK]D-Fender | And why not? |
| 05:11.50 | *** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 05:11.52 | hads | I'm sure you can find it with google fairly easily if you really can't get it fromyour reseller. |
| 05:11.53 | dracosilv | okay.. |
| 05:12.09 | dracosilv | finally found out where this channel is. |
| 05:12.13 | dracosilv | lemme test somethin |
| 05:12.24 | dracosilv | WIERD |
| 05:12.35 | dracosilv | i can only hear what's going on here, over there. |
| 05:12.47 | dracosilv | over there=#asterisk on irc.2600.net |
| 05:13.01 | dracosilv | okay... here's my question. |
| 05:13.20 | dracosilv | <dracosilv> i downloaded the 2.0 version of trixbox from the link specified on the trixbox website |
| 05:13.20 | dracosilv | <dracosilv> and i try to burn it as slow as i can (4x or 8x) |
| 05:13.20 | dracosilv | <dracosilv> and when i boot the machine up.. |
| 05:13.27 | [TK]D-Fender | ~trixbox |
| 05:13.39 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:13.39 | dracosilv | <dracosilv> the cd loads up.. |
| 05:13.39 | dracosilv | <dracosilv> and all is well until i press enter at the 'boot options' screen |
| 05:13.39 | dracosilv | <dracosilv> it loads the first 2 files (can't remember what they are offhand |
| 05:13.39 | dracosilv | <dracosilv> but then it reboots |
| 05:13.40 | dracosilv | <dracosilv> what's going on? |
| 05:13.42 | [TK]D-Fender | dracosilv, and stop spamming the channel |
| 05:13.52 | dracosilv | hey.. i'm not trying to spam the channel |
| 05:13.58 | hads | But you are. |
| 05:14.02 | dracosilv | i had just posted my question (line by line) |
| 05:14.06 | dracosilv | e.g. typed it out |
| 05:14.08 | [TK]D-Fender | dracosilv, And doing a bang-up job of it |
| 05:14.15 | dracosilv | and i didn't feel like retyping it. |
| 05:14.23 | dracosilv | and i tried to make sure i didn't post TOO much all at once. |
| 05:14.28 | [TK]D-Fender | dracosilv, Yeah.. you did it one by one for THEM, and you're spamming US |
| 05:14.35 | dracosilv | (and get a flood kick) |
| 05:14.51 | [TK]D-Fender | dracosilv, Regardless, please heed jbot. |
| 05:16.30 | dracosilv | okay.. but i don't see how hard a simple cd-boot problem would be.. i'm just wondering if my problem lies in the speed i'm burning the cd at.. or the download... |
| 05:17.18 | hads | Yes, but that has nothing to do with Asterisk |
| 05:17.20 | [TK]D-Fender | ~trixbox |
| 05:17.30 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:17.50 | [TK]D-Fender | indeed. Nothing at all to do with Asterisk. |
| 05:19.17 | dracosilv | so you're saying that you can't help me get it running... |
| 05:19.25 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 05:19.36 | dracosilv | too bad you guys don't have an MD5 sum or something.. |
| 05:19.37 | dracosilv | for the isos |
| 05:19.47 | dracosilv | so i know if the dl is corrupt or not.. |
| 05:20.08 | dracosilv | that WOULD be something helpful and related to the trixbox release |
| 05:20.15 | hads | Trixbox != Asterisk |
| 05:20.17 | [TK]D-Fender | dracosilv, You aren't getting it. WE are not TRIXBOX. |
| 05:20.32 | Dovid | anyone have a script that looks up CID from the internet ? |
| 05:20.33 | [TK]D-Fender | dracosilv, it is NOT supported here. |
| 05:20.48 | hads | Read what jbot said, twice. |
| 05:21.00 | Dovid | ~trixbox |
| 05:21.22 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:21.22 | dracosilv | ~trixbox |
| 05:21.31 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:21.31 | [TK]D-Fender | ... thrice! |
| 05:21.32 | dracosilv | ~trixbox |
| 05:21.34 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:21.34 | [TK]D-Fender | <jbot> Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 05:21.35 | hads | FFS |
| 05:21.51 | dracosilv | hehe |
| 05:22.37 | Sargun | Anyone have firmware-polycom |
| 05:22.52 | hads | Sargun: Your reseller will |
| 05:23.04 | Sargun | no |
| 05:23.05 | Sargun | the package |
| 05:23.13 | Sargun | and I know my reseller will |
| 05:23.17 | Sargun | but I need it now |
| 05:23.50 | Sargun | or tonight |
| 05:23.50 | Sargun | why does polycom hide their firmware? |
| 05:23.50 | hads | Then look for it. It's not that difficuly. |
| 05:23.51 | hads | s/difficuly/difficult/ |
| 05:23.51 | dracosilv | okay.. another thing.. |
| 05:23.59 | Dovid | anyone have a script that looks up CID from the internet ? |
| 05:24.04 | dracosilv | why is there a relay for this channel on irc.2600.net |
| 05:24.17 | dracosilv | yet it's not bi-directional? |
| 05:24.23 | dracosilv | what's up with that |
| 05:24.28 | Sargun | hads, I'm not that great at finding illegal software, do you have it? |
| 05:24.58 | [TK]D-Fender | dracosilv, probably someone running a listening bot here, and just another thing we don't support or care about here./ |
| 05:25.54 | dracosilv | well the username's ircrly (the relay's name) if you care about it... |
| 05:27.05 | [TK]D-Fender | Can I really be any more explicit? *sigh* |
| 05:27.18 | Sargun | hads, Can you find it for me? |
| 05:27.59 | [TK]D-Fender | Sargun, Yes, we are all here to do your dirty work for you. Now you know why Polycom doesn't want to deal with consumers directly. |
| 05:28.19 | Sargun | [TK]D-Fender, Heh |
| 05:28.53 | hads | Now, why was it I don't lurk here that much anymore again? |
| 05:29.16 | Sargun | [TK]D-Fender, A) I have no idea where to find illegal things B) Uh, why would it be any different to offer a newer firmware? |
| 05:30.48 | [TK]D-Fender | Sargun, since you completely lack the foresight to figure it out, think on this : They don't want to hear about bugs from 1,000,000 whiny idiot installing the most untested version released. |
| 05:31.39 | [TK]D-Fender | Sargun, and as for your "A"..... I don't even need to say it I hope.... |
| 05:31.55 | Sargun | I'm googling right now |
| 05:32.03 | Sargun | I have been for 20-30 minutes. |
| 05:32.11 | hads | It amuses me the way the word illegal is thrown around. |
| 05:32.22 | Sargun | hads, It has it in the license agreement |
| 05:32.42 | Sargun | they aren't supposed to release it publically |
| 05:32.47 | Sargun | but polycom doesn't enforce it |
| 05:34.11 | [TK]D-Fender | Sargun, So you come in here asking US to do "illegal" things for you. BRILLIANT. |
| 05:34.24 | Sargun | [TK]D-Fender, Ok? |
| 05:34.41 | Sargun | And, technically it isn't illegal to find it. |
| 05:34.51 | [TK]D-Fender | Sargun, Wait to hear back from your reseller. its their job to provide it to you. |
| 05:34.59 | Sargun | Ok |
| 05:35.00 | Sargun | Sure. |
| 05:36.52 | Sargun | supply |
| 05:39.20 | Dovid | i use telephonydepot.com |
| 05:40.10 | Sargun | Ok, I found bootrom.ld |
| 05:40.10 | Sargun | now I wonder |
| 05:40.11 | Sargun | where bootrom.ver is |
| 05:41.49 | Sargun | Dovid, Thanks |
| 05:44.21 | *** join/#asterisk Dovid[laptop] (n=Dovid@ool-43511bec.dyn.optonline.net) |
| 05:44.34 | Dovid[laptop] | i know this is OT but does anyone here use sqlite ? |
| 05:45.44 | Sargun | Dovid, I used to. |
| 05:47.19 | Dovid[laptop] | Sargun: whats the service name to start it ? |
| 05:47.22 | Dovid[laptop] | sqlite ? |
| 05:47.32 | Dovid[laptop] | i tried service sqlite start and it didnt work |
| 05:48.05 | Sargun | uh |
| 05:48.06 | hads | There is a #sqlite channel. Also sqlite isn't a daemon. |
| 05:48.08 | Sargun | its not service based |
| 05:48.12 | Sargun | its file based |
| 05:49.15 | Dovid[laptop] | Shargun: may i PM u ? |
| 05:49.35 | Dovid[laptop] | i am in the channel now |
| 05:50.56 | Sargun | Why not |
| 06:01.17 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:10.06 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-209-201.socal.res.rr.com) |
| 06:18.36 | *** join/#asterisk Dovid (n=Dovid@ool-43511bec.dyn.optonline.net) |
| 06:20.12 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 06:21.37 | *** join/#asterisk uppal (n=uppal@2001:618:400:888c:218:deff:fe9f:a77f) |
| 06:31.51 | Zeeek | hello |
| 06:33.50 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 06:41.52 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 06:47.44 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 06:57.43 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:58.18 | *** join/#asterisk Black-Kakugane-1 (n=lonewolf@c-71-206-157-165.hsd1.va.comcast.net) |
| 07:05.57 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 07:08.17 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:09.27 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 07:14.45 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 07:20.20 | MrWup | hey guys |
| 07:20.25 | MrWup | need help again =] |
| 07:20.32 | MrWup | if RECEPTION => SIP/101&SIP/105 |
| 07:20.44 | MrWup | and i do a dial(${RECEPTION}) |
| 07:20.57 | MrWup | how do i actually find out which phone answered? |
| 07:33.35 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:36.52 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 07:44.56 | *** join/#asterisk Dovid (n=Dovid@ool-43511bec.dyn.optonline.net) |
| 07:45.16 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:46.45 | Dovid | hi guys |
| 07:47.12 | Dovid | i am using mysql for cdr logs. how can i do a query to see what calls were not bridged ? |
| 07:51.57 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 08:04.25 | robin_sz | sigh .. I need an active single BRI isdn card ... |
| 08:04.41 | tzafrir | Is there such a beast? |
| 08:04.55 | robin_sz | theres the Eicon Diva stuff |
| 08:05.29 | robin_sz | the Digium card looks nice but its 4 times more than I need and more expensive than I can afford |
| 08:06.08 | robin_sz | per channel, its in the right ballpark though :) |
| 08:06.48 | robin_sz | or, I need someone to fix the DTMF detection algo in the mISDN_dsp |
| 08:07.23 | rue_mohr | why dont you do it? |
| 08:07.47 | robin_sz | if I could understand how it worked, I'd have a go |
| 08:07.57 | robin_sz | rue_mohr, I think its a fft |
| 08:08.07 | rue_mohr | hmm |
| 08:08.38 | rue_mohr | I'd help, but I'm too busy trying to understand how to fix the state tables to make my mainstreet 3624 work |
| 08:09.14 | robin_sz | the algo used suffers from a defect of not detecting when the two tones are of different levels |
| 08:09.22 | robin_sz | its a known shortcoming of that algo |
| 08:09.39 | rue_mohr | maybe redo it with the source from... |
| 08:09.51 | robin_sz | ... ? |
| 08:11.25 | rue_mohr | hmm |
| 08:12.46 | rue_mohr | multimon |
| 08:13.02 | rue_mohr | yea, thats it |
| 08:13.15 | rue_mohr | demod_dtmf.c |
| 08:13.33 | robin_sz | looking |
| 08:13.43 | Sargun | Does anyone here have access to a recent version of Polycom's SIP/BootROM software? |
| 08:14.34 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 08:14.44 | MrWup | hi |
| 08:18.09 | Sargun | hi |
| 08:20.33 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 08:26.23 | JT | robin_sz: i always hear you complaining about misdn every week |
| 08:26.26 | JT | just use bristuff already |
| 08:26.28 | JT | misdn is rubbish |
| 08:29.34 | robin_sz | thats not so, I havent complaoined in months |
| 08:30.22 | robin_sz | I did try with bristuff originally, failed to get it to work |
| 08:32.31 | robin_sz | weirdly .. the cologne HFC chips seem to have hardware DTMF detection on all B channels |
| 08:32.58 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 08:34.47 | JT | so buy one? |
| 08:35.36 | coppice | when people complain about FAX and are using mISDN, mISDN is always the answer |
| 08:36.30 | tzafrir | robin_sz, but zaptel normally doesn't bother using it |
| 08:36.38 | tzafrir | (hardware dtmf detection) |
| 08:36.41 | robin_sz | oh |
| 08:37.03 | tzafrir | zaptel can actually use it. Though drivers simply don't implement it |
| 08:37.05 | tzafrir | sadly |
| 08:37.08 | coppice | tzafrir: he's not using zaptel. he's using mISDN |
| 08:37.13 | robin_sz | JT, I already have one, but for some reasonthe SW is trying to do it the hard way |
| 08:37.32 | robin_sz | seems zaptel will try and do it the hard way too |
| 08:38.23 | robin_sz | coppice, this is just a DTMF detection issue, im quite happy with mISDN in all other respects, but I'll happily go to bristuff IFF the dtmf detection is better |
| 08:38.27 | JT | coppice: the answer to what? |
| 08:39.16 | coppice | Q: what is the problem when I use FAX with mISDN |
| 08:39.18 | coppice | A: mISDN |
| 08:39.31 | robin_sz | I dont use fax .. |
| 08:39.45 | JT | coppice: ah |
| 08:39.56 | coppice | the DTMF software in isdn4linux is junk |
| 08:40.00 | JT | the answer to the problem, not the solution ;) |
| 08:40.04 | robin_sz | so ... |
| 08:40.23 | robin_sz | is the DTMF sw in bristuff any better? |
| 08:40.51 | coppice | bristuff uses the decoder in asterisk, which is a high quality decoder |
| 08:41.02 | robin_sz | ok |
| 08:41.35 | robin_sz | I just need to archive everything here and then build an * with the bristuff patches then |
| 08:41.39 | *** join/#asterisk yenno (i=dfgdfgdg@80-219-243-201.dclient.hispeed.ch) |
| 08:42.26 | yenno | hi, if my client receives a 401, he retries with auth infos and gets a 401 again - is that ok according to rfc or shouldn't the pbx return a 403? |
| 08:42.57 | robin_sz | oh dear .. now I have to make the 1.2.17 or 1.4.4 decision :( |
| 08:44.57 | JT | robin_sz: no decision to make with bristuff |
| 08:45.05 | JT | robin_sz: download the latest bristuff |
| 08:45.08 | JT | and run the script |
| 08:45.18 | JT | it downloads and installs the correct version of zaptel and asterisk |
| 08:46.02 | robin_sz | OK |
| 08:46.30 | robin_sz | its running ... |
| 08:47.18 | robin_sz | err .. asterisk 1.0.10? |
| 08:47.53 | JT | clearly not the latest bristuff |
| 08:48.08 | robin_sz | straight from the Junhanns site ... |
| 08:48.42 | JT | junghanns.net/downloads |
| 08:48.54 | robin_sz | yep 0.2.0 rc8s |
| 08:49.15 | JT | bristuff-0.3.0-PRE-1y-e.tar.gz 03-Apr-2007 11:57 279K |
| 08:50.18 | robin_sz | oh, thats the experimental version |
| 08:50.34 | JT | make sure you wipe your asterisk modules directory before installing it |
| 08:50.53 | JT | it uses 1.2, i've found the recent releases work fine |
| 08:50.59 | robin_sz | ok |
| 08:51.36 | JT | the 0.2 bristuff branch isn't maintained anymore |
| 08:52.45 | robin_sz | looks like a downgrade in terms of * version from what I run now |
| 08:53.04 | robin_sz | 1.2.10 compared to 1.2.14 |
| 08:53.18 | JT | you're focusing on stupid things like minor release numbers |
| 08:53.27 | JT | is there anything specific you're missing out on? |
| 08:53.36 | JT | and i think it's newer than 1.2.10 |
| 08:53.48 | robin_sz | its not |
| 08:53.48 | JT | the point is it works, and misdn doesn't |
| 08:54.52 | robin_sz | I just watched it d/l it .. I do remembe some issues i had that were solved by keeping up to date (ish) as we went along throughthe minor releases, maybe they are not important when used with zap, but were important when used with misdn .. shrug |
| 08:56.41 | robin_sz | ah, it is 1.2.14, my mistake |
| 08:57.00 | tzafrir | it uses asterisk 1.2.17 and zaptel 1.2.16 |
| 08:57.15 | JT | tzafrir: which version? |
| 08:57.25 | robin_sz | http://www.asterisk-support.de/mirror/asterisk-1.2.14/asterisk-1.2.14.tar.gz |
| 08:57.25 | robin_sz | <PROTECTED> |
| 08:57.32 | tzafrir | 1y-e ( the latest 0.3) |
| 08:57.33 | hads | mISDN works. |
| 08:57.48 | JT | for small amounts of "works" |
| 08:57.51 | robin_sz | hads, doesnt do dtmf detectionat all well |
| 08:58.09 | JT | it is useless in NT mode too |
| 08:58.14 | JT | misdn... |
| 08:58.43 | robin_sz | `zaptel-1.2.10.tar.gz' |
| 08:58.55 | mvanbaak | we run the 0.3.0-PRE-1y-e at work in production |
| 08:58.58 | mvanbaak | works fine |
| 08:59.00 | yenno | nobody knows an answer to my question? :( |
| 08:59.06 | robin_sz | tzafrir, im watcing it d/l different versions to what you have |
| 09:01.05 | tzafrir | from download.sh: zaptel-1.2.16.tar.gz , libpri-1.2.4.tar.gz , asterisk-1.2.17.tar.gz |
| 09:01.26 | robin_sz | shrug .. must be a different bristuff |
| 09:01.28 | tzafrir | robin_sz, do you refer to 1y or to 1y-e ? |
| 09:01.45 | robin_sz | bristuff-0.3.0-PRE-1y |
| 09:02.04 | tzafrir | well, the latest 0.3 is 1y*-e* |
| 09:02.11 | robin_sz | sigh .. |
| 09:02.35 | JT | robin_sz: don't sigh at us. |
| 09:02.47 | JT | robin_sz: i told you the exact version and you downloaded the wrong one |
| 09:02.48 | robin_sz | i was sighing at junhanns ... |
| 09:02.59 | robin_sz | i d'ld the one from their download page |
| 09:03.00 | tzafrir | http://junghanns.net/downloads/?C=M;O=D # with the right sorting |
| 09:03.03 | JT | because you don't understand alphabetical order? |
| 09:03.15 | JT | and datestamps |
| 09:03.28 | robin_sz | I understand http://www.junghanns.net/en/download.html |
| 09:03.35 | robin_sz | and the only two versions listed ... |
| 09:03.36 | tzafrir | because the versioning scheme has changed... |
| 09:03.49 | JT | cool |
| 09:03.54 | JT | so you went to the wrong web page too |
| 09:03.58 | robin_sz | or is going to their website, clicking on downlaods and folloewing the instructions not what I should be doing |
| 09:03.59 | JT | not the one i said to go to |
| 09:04.16 | JT | indeed, you should go straight to the downloads directory |
| 09:04.25 | JT | junghanns.net/downloads |
| 09:05.20 | robin_sz | OK |
| 09:08.09 | robin_sz | it would probably be easier if they linked it from some publically accessible link, but yeah, OK, thanks, its d#l ing |
| 09:08.42 | JT | yeah the web page doesn't always have the software up to date |
| 09:08.50 | JT | displayed on the page anyway |
| 09:08.53 | robin_sz | right |
| 09:08.58 | robin_sz | an easy mistake to make |
| 09:09.41 | JT | basically the only thing we hear from klaus junghanns these days is his tarballs, or new products |
| 09:10.05 | robin_sz | he has a 2 ch BRI card ... |
| 09:10.29 | JT | 2 port, yeah |
| 09:10.33 | JT | 4ch to be technical :P |
| 09:11.04 | robin_sz | 4B and 2D to be pedantic |
| 09:11.38 | mvanbaak | anyone tested his GSM cards ? |
| 09:12.32 | robin_sz | £245 ... not too bad |
| 09:14.13 | coppice | That's a whole lot of dollars these days |
| 09:15.28 | robin_sz | still uses the same cologne chips as my generic card though |
| 09:15.40 | robin_sz | pick them up on ebay for £5 |
| 09:16.03 | coppice | people try to charge huge sums for the multi-channels cards |
| 09:16.09 | robin_sz | yeah |
| 09:17.05 | robin_sz | the basic single channel hfc cards were only £50 new, find them on ebay, boxecd for £5 now |
| 09:19.56 | robin_sz | bri was pushed as a broad-ish band intenrent connection by the telcos here before adsl rolled out |
| 09:20.08 | robin_sz | so lots and lots of cards around |
| 09:21.25 | coppice | 50 was a pretty high price for such a simple card. the UK always gets ripped off :-) |
| 09:22.53 | robin_sz | that was for a good one :) not a fritz . .thery were 15 |
| 09:23.20 | coppice | they are all pretty much the same cologne chip reference card |
| 09:25.37 | robin_sz | yep |
| 09:26.00 | robin_sz | and now they sell them for £200 as a "isdn telephony solutions card" |
| 09:27.11 | *** join/#asterisk friedrich| (n=friedric@e177255060.adsl.alicedsl.de) |
| 09:28.00 | coppice | the ones I have here are new, and I think they were about $10 |
| 09:28.10 | robin_sz | nice |
| 09:28.24 | robin_sz | they have much BRI in the US? |
| 09:28.28 | JT | no |
| 09:28.55 | coppice | BRI is rare almost anywhere except the EU |
| 09:28.57 | robin_sz | come to think of it, ive visted NYC, |
| 09:29.09 | robin_sz | I saw the infrastructure ;) |
| 09:29.14 | coppice | and Japan. they seem to use it quite a bit |
| 09:29.23 | JT | coppice: it's also available a fair bit in .au |
| 09:29.24 | JT | .nz too |
| 09:29.36 | robin_sz | I didn't realise morse keys were still popular ;) |
| 09:30.25 | coppice | I've only got these cards because someone wants me to do some development with them |
| 09:31.14 | coppice | most places made sure BRI would never catch on |
| 09:32.41 | robin_sz | this looks well broken ... seems to be a pile of errrors during compile |
| 09:34.00 | robin_sz | paste? |
| 09:35.21 | MrWup | is there any way to keep some sort of channel variable in a channel when it goes zombie? |
| 09:35.57 | MrWup | its really crap that im trying to do some logic on the hangup of a channel (when it goes zombie) and so i have the event there, but no variables to do anything with |
| 09:36.56 | robin_sz | http://nopaste.snit.ch:8001/10104 |
| 09:41.10 | robin_sz | ok wait ... |
| 09:41.12 | robin_sz | http://nopaste.snit.ch:8001/10105 |
| 09:41.17 | robin_sz | thats a more useful paste |
| 09:41.31 | robin_sz | could someone have look and point me in the right direction? |
| 09:41.49 | tzafrir | make[1]: Entering directory `/usr/src/kernel-source-2.6.8 |
| 09:42.31 | tzafrir | why is that? is that the place pointed to by /lib/modules/`uname -r`/build ? |
| 09:43.23 | tzafrir | include/linux/config.h:4:28: linux/autoconf.h: No such file or directory <- that kernel source tree is not configured, right? |
| 09:43.41 | coppice | robin_sz: I think this is Linux versions issue. files like autoconf.h changed names at one point |
| 09:43.55 | robin_sz | It first complaine that the kerneal source was not linked to /usr/src/linux-2.6 |
| 09:44.00 | robin_sz | so I linked it ... |
| 09:44.48 | tzafrir | do you have a distro kernel or a self-built one? |
| 09:45.10 | robin_sz | /lib/modules/2.6.8-2-686/build: is a directory# |
| 09:45.12 | robin_sz | distro |
| 09:45.28 | robin_sz | I d/l the source to keep it happy earlier |
| 09:45.35 | robin_sz | debian sarge |
| 09:45.47 | robin_sz | well, it started of life as a Xorcom rapid box once |
| 09:46.02 | tzafrir | apt-get install kernel-headers-`uname -r` |
| 09:46.09 | robin_sz | got them |
| 09:46.40 | robin_sz | kernel-headers-2.6.8-2-686 is already the newest version. |
| 09:46.46 | tzafrir | alternatively: |
| 09:46.53 | robin_sz | heh .. yes? |
| 09:47.22 | tzafrir | echo "deb http://updates.xorcom.com/rapid sarge main" >>/etc/apt/sources.list |
| 09:47.34 | robin_sz | uh |
| 09:47.49 | tzafrir | apt-get update; apt-get install zaptel-modules-`uname -r` asterisk asterisk-bristuff |
| 09:47.56 | tzafrir | apt-get update; apt-get install zaptel-modules-`uname -r` asterisk asterisk-bristuffed |
| 09:48.16 | robin_sz | # Updates from xorcom |
| 09:48.16 | robin_sz | deb http://updates.xorcom.com/rapid sarge main |
| 09:48.16 | robin_sz | # deb-src http://updates.xorcom.com/rapid sarge main |
| 09:48.23 | robin_sz | way ahead of you ... next? |
| 09:48.55 | tzafrir | sorry, the asterisk version there is dated. I forgot |
| 09:49.05 | tzafrir | (only zaptel there is very up-to-date) |
| 09:49.13 | robin_sz | this is back to where i started .. |
| 09:49.58 | robin_sz | I installed xorcom rapid in the first place ... it ran, zaptel showed the card etc, I plugged in the ISDN, and ...the alarms cleared |
| 09:50.09 | robin_sz | but calls did not pass to * |
| 09:50.26 | robin_sz | which is how I ended up on mISDN |
| 09:58.32 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:58.32 | *** join/#asterisk potsboy (n=jsoap@c2-388-1.vic.dial.mweb.co.za) |
| 10:06.28 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 10:07.51 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 10:07.55 | sasch | hi all |
| 10:09.19 | crochat | sasch: Hi |
| 10:10.08 | *** join/#asterisk heanol (i=heanol@88.80.13.121) |
| 10:13.14 | MrWup | anyone know how to stop inheritedvariables being dumped on the h extension when a channel hangs up and goes zombie? |
| 10:24.11 | JT | robin_sz: yes, you need to understand how to compile software |
| 10:24.19 | JT | i have bristuff working under debian no probs |
| 10:24.34 | sasch | anyone know a html page that a list of global var of asterisk 1.4 (for example ${CALLERID} and ....) |
| 10:24.34 | robin_sz | etch I presume? |
| 10:24.41 | JT | no |
| 10:24.50 | JT | sarge or something |
| 10:24.58 | JT | not that relevant since i compiled it myself |
| 10:25.01 | pipwerk | JT: www.voip-info.org |
| 10:25.09 | JT | pipwerk: ? |
| 10:25.20 | pipwerk | sri, wrong nick |
| 10:25.35 | pipwerk | sasch: www.voip-info.org |
| 10:25.44 | robin_sz | well, it fails for soem reason I dont understand ... presumably some mistake in the makefile |
| 10:25.59 | JT | sasch: README.variables |
| 10:26.03 | JT | in the docs dir |
| 10:26.28 | JT | robin_sz: i doubt it's a mistake in the makefile |
| 10:26.40 | JT | probably a mistake in your kernel sources and headers config |
| 10:26.48 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 10:26.52 | robin_sz | JT: im just someone who wants to use * ... I didn't claim to be a kernel developer or have an in depth understanding of the way things are linked up by the compiler ... |
| 10:27.19 | JT | i'm not a kernel developer, but a lot of software requires your kernel sources and headers be setup right |
| 10:27.47 | robin_sz | well, I wasnt aware I set them up at all, apart from doing the apt-get thing ... |
| 10:27.57 | EmleyMoor | I am having a few problems setting CALLERID(name) - I can do it with LookupCIDName but if I want to set it to something arbitrary, it appears as blank when the call rings on X-Lite |
| 10:28.01 | robin_sz | and the ln -s that bristuff asked me to do |
| 10:28.44 | robin_sz | JT where should I look, is there a file where these are set up ? |
| 10:28.46 | sasch | <pipwerk> if I want to check a caller number .... if the number is 1223 i want to send in a context ... how made it ?? |
| 10:28.57 | sasch | excusme for my english :-P I'm italian :-P |
| 10:29.53 | JT | robin_sz: i'm thinking it's a headers issue |
| 10:29.57 | JT | is it a default kernel? |
| 10:30.06 | robin_sz | I think so |
| 10:30.22 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 10:30.26 | Aces1Up | hey all. |
| 10:30.33 | EmleyMoor | I'd like to be able, on calls to two of my incoming numbers, to set CALLERID(name) to something special before sending the call to the phones - I have tried and it in fact goes blank |
| 10:30.43 | robin_sz | the first error seems to be: |
| 10:30.44 | robin_sz | include/linux/config.h:4:28: linux/autoconf.h: No such file or directory |
| 10:31.16 | EmleyMoor | Why would it go blank?# |
| 10:31.33 | robin_sz | I have: |
| 10:31.35 | robin_sz | rapid:~/bristuff/bristuff-0.3.0-PRE-1y-e/zaphfc# locate autoconf.h |
| 10:31.35 | robin_sz | /usr/include/linux/autoconf.h |
| 10:31.35 | robin_sz | /usr/src/kernel-headers-2.6.8-2-686/include/linux/autoconf.h |
| 10:31.36 | robin_sz | /usr/src/kernel-headers-2.6.8-2/include/linux/autoconf.h |
| 10:31.37 | sasch | I recompile my asterisk with svn .... now when start say this error |
| 10:31.38 | sasch | http://pastebin.ca/473627 |
| 10:31.40 | sasch | why ?? |
| 10:31.46 | heanol | i've set up a registry with my sip provider, mapping it to an extension that goes to voice mail.. that works and i can call my number with a real phone and leave a message. but now i'd like to be able to dial in to asterisk with a local voip softphone to listen to my messages, but i don't really know where to start. can anyone give me any pointers? |
| 10:32.18 | robin_sz | heanol, set up an extension in sip.conf |
| 10:32.26 | heanol | robin_sz, i've done that.. |
| 10:32.30 | heanol | but how do i reach it? |
| 10:33.02 | heanol | i mean, i added another extension for the VoicemailMain() |
| 10:33.06 | robin_sz | make sure that the context yset in sip.conf matches a context that has access to voice mail |
| 10:33.07 | heanol | but i don't know how to reach it |
| 10:33.10 | JT | ~thebook |
| 10:33.21 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:33.21 | JT | heanol: |
| 10:33.21 | robin_sz | then dial 8501 usuall |
| 10:33.38 | tzafrir | robin_sz, the bristuff build script seems to do some strange things |
| 10:33.38 | robin_sz | or 8500, I forget |
| 10:34.35 | robin_sz | I installed the latest xorcom zaptel, ran genzaptelconf |
| 10:34.46 | robin_sz | but it failed to find my card :( |
| 10:35.10 | tzafrir | you don't see its span loaded? |
| 10:35.16 | robin_sz | no |
| 10:35.22 | tzafrir | under /proc/zaptel ? |
| 10:35.36 | robin_sz | I rmmoded the misdn stuff first, to make sure it was free |
| 10:35.54 | tzafrir | did you run 'genzaptelconf -d' (the hardware detection: unloads and then loads all zaptel modules) |
| 10:36.03 | robin_sz | ahh, ok, wait :) |
| 10:36.08 | JT | his bristuff didn't even compile properly... |
| 10:37.52 | robin_sz | hmm ... it found the card twice :) |
| 10:39.42 | sasch | in CLI for look hints what is the command ?? |
| 10:40.07 | EmleyMoor | When I do an arbitrary Set(CALLERID(name) on an incoming call, the name goes blank, instead of going to what I tried to set. Why would that happen? |
| 10:40.56 | JT | do you set it after the call has come in or what? |
| 10:41.30 | EmleyMoor | JT: Yes - in priority 1 of that extension... with priority 2 being the Dial command |
| 10:43.23 | JT | when are you trying to access the var? |
| 10:43.42 | sasch | ok I found core show hints |
| 10:43.46 | EmleyMoor | Umm... what possible answer is there? |
| 10:44.29 | EmleyMoor | (when the call is in, immediately before sending it to phones) |
| 10:44.56 | JT | you say it's blank |
| 10:45.05 | JT | when are you seeing it blank? |
| 10:45.14 | EmleyMoor | Yes - it displays as a blank line on X-Lite |
| 10:45.33 | JT | that doesn't mean the var is blank |
| 10:45.51 | EmleyMoor | It does mean it's not what I set, though# |
| 10:46.46 | EmleyMoor | I will add a NoOp to help me see |
| 10:46.50 | JT | do Verbose(CallerID Name is:${CALLERID(name)}) before Dial |
| 10:49.54 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 10:50.16 | EmleyMoor | Ah, getting there |
| 10:50.28 | EmleyMoor | X-lite doesn't like " marks |
| 10:52.07 | EmleyMoor | Can I set it to something with a space in without using them? |
| 10:52.55 | EmleyMoor | Great, done |
| 10:53.14 | EmleyMoor | I can tell those two numbers are ringing now when out of earshot of my Zap phone |
| 10:54.24 | EmleyMoor | Damn, it doesn't work when the incoming number is withheld |
| 11:00.36 | sasch | what is the command busy-level in sip.conf ?? |
| 11:09.08 | *** join/#asterisk Vec (n=Vec@dsl-244-217-119.telkomadsl.co.za) |
| 11:11.22 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 11:11.23 | redax | hi. |
| 11:21.21 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 11:21.48 | *** join/#asterisk asdx (n=diego@200.26.177.4) |
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| 11:25.33 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 11:37.35 | *** join/#asterisk notnyt (i=nyt@countercultured.net) |
| 11:37.52 | notnyt | hey |
| 11:39.16 | notnyt | so, is there any reason that the cdr behavior was changed to not update with agent info? |
| 11:41.40 | robin_sz | OK, so I got zaptel working, zttest returns sane figures ... I configured zapata.conf I think .. |
| 11:42.49 | robin_sz | but when I dial in .. nada. call dropped, no activity in * |
| 11:44.47 | notnyt | did you make sure you specified a context for incoming calls? |
| 11:44.47 | notnyt | and that you have channels associated with it |
| 11:44.48 | *** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 11:46.04 | robin_sz | ummm |
| 11:46.10 | robin_sz | dunno .. wait i check |
| 11:52.07 | robin_sz | ahh, I think this is where I have to do the signalling method for ISDN/bri right? |
| 11:52.17 | robin_sz | signalling is ? ami? |
| 11:52.21 | robin_sz | ccs? |
| 11:57.18 | tzafrir | robin_sz, was there anywhere with zap/bri with different signalling? |
| 11:57.31 | robin_sz | mmm? |
| 11:57.36 | robin_sz | not sure I understand |
| 11:57.37 | tzafrir | ccs,ami |
| 11:57.46 | tzafrir | seems to always work |
| 11:57.58 | robin_sz | so in zapat.conf ... |
| 11:58.07 | tzafrir | At least Australia, Italy, Israel, France and Germany |
| 11:58.39 | robin_sz | i put ignalling=ami |
| 11:58.41 | robin_sz | ? |
| 11:59.13 | tzafrir | sorry, no |
| 11:59.34 | tzafrir | with bristuff: normally signalling=bri_cpe_ptmp |
| 12:01.03 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:03.43 | robin_sz | OK, chan_zap loaded |
| 12:11.26 | robin_sz | <PROTECTED> |
| 12:11.26 | robin_sz | May 6 13:10:27 WARNING[6369]: chan_zap.c:7708 zt_pri_error: PRI: received SETUP message for call that is not a new call (retransmission). |
| 12:11.32 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 12:12.42 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:17.53 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 12:22.54 | *** join/#asterisk henryv2 (n=henry_vo@82-39-113-10.cable.ubr03.newy.blueyonder.co.uk) |
| 12:24.11 | henryv2 | I'm having a problem with an x100p when making outbound calls. The inbound calls work fine and asterisk picks up the call and handles it as expected. On outbound calls the call is initiated and rings the external number, however, when the external number answers the call asterisk hangs up immediately |
| 12:24.26 | henryv2 | any ideas what might cause that? I can't find any solutions when searching on google! |
| 12:33.22 | robin_sz | wahey .. seems to rx incoming calls .. AND seems to grok my DTMF :) |
| 12:33.38 | JT | :) |
| 12:35.47 | robin_sz | just outgoing to figure out now |
| 12:40.11 | robin_sz | OK, I changed TRUNK=mISDN to TRUNK=Zap/1 right? |
| 12:40.39 | robin_sz | and I see Executing Dial("SIP/home-fd8d", "Zap/1/01xxxxxxxxx") |
| 12:41.10 | robin_sz | but then it just fails .. I get "forbidden" onthe handset |
| 12:41.27 | robin_sz | <PROTECTED> |
| 12:41.27 | robin_sz | <PROTECTED> |
| 12:41.27 | robin_sz | <PROTECTED> |
| 12:41.27 | robin_sz | May 6 13:38:37 WARNING[6599]: app_dial.c:451 wait_for_answer: Unable to forward voice |
| 12:41.27 | robin_sz | <PROTECTED> |
| 12:41.28 | robin_sz | <PROTECTED> |
| 12:41.36 | robin_sz | sorry, I know, I pasted. |
| 12:42.35 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:43.12 | tzafrir | something is fishy around here |
| 12:43.22 | robin_sz | this is fun! |
| 12:43.31 | henryv2 | what kind of card are you using robin_sz? |
| 12:43.48 | robin_sz | hfc |
| 12:44.15 | robin_sz | cologne chipset blah |
| 12:45.07 | henryv2 | can you put a bri card in the same box as a tdm card? something like the digium tdm800p? |
| 12:45.25 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-171-123.buff.east.verizon.net) |
| 12:46.16 | robin_sz | I would expect so, I think you can mix and match digium cards as you like |
| 12:47.02 | henryv2 | just i read something about using bri cards and pri cards on the same machine. does bristuff still cause problems for pri? |
| 12:47.40 | tzafrir | heard rumors. Never got to test them |
| 12:52.21 | robin_sz | is Zap/1 the correct thing to use as a trunk entry? |
| 12:52.37 | robin_sz | changed TRUNK=mISDN to TRUNK=Zap/1 |
| 13:00.04 | JT | if you want to only ever call out on channel 1 |
| 13:00.35 | JT | henryv2: i don't think there's truth to that |
| 13:00.39 | JT | but i havent tested that |
| 13:03.55 | robin_sz | JT, I'll set up groups when I have it working on at least 1 channel |
| 13:04.16 | robin_sz | wonder if I need to add a leading 9 ... |
| 13:04.25 | JT | pri intense debug |
| 13:04.49 | JT | see exactly what the error is |
| 13:06.24 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) |
| 13:09.18 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 13:09.40 | henryv2 | any ideas on what might cause a zap channel to hangup on outbound calling when the remote party answers the call? |
| 13:09.42 | robin_sz | > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) |
| 13:09.42 | robin_sz | > Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] |
| 13:09.46 | robin_sz | ?? |
| 13:09.51 | robin_sz | is that it? |
| 13:10.19 | robin_sz | JT, shall I paste the debug? |
| 13:10.42 | JT | no |
| 13:10.55 | JT | maybe you are dialling the number wrong |
| 13:12.15 | robin_sz | mmm ... I think I know my home number :) |
| 13:12.18 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 13:12.31 | JT | you wondered about the leading number |
| 13:12.35 | JT | shrug |
| 13:12.44 | robin_sz | the 1/0xxxxxxx |
| 13:14.14 | notnyt | so any idea why the cdr/callerid only gets updated with the agent name now? it used to get updated with agent #, and src channel as the agent |
| 13:15.56 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 13:18.50 | robin_sz | im just wondering if its not liking my outgoing CID which is peresentign as 5004 ... my locla extension number |
| 13:19.11 | robin_sz | where in zapata.conf do I set the outgoing CID? |
| 13:20.54 | pipwerk | robin_sz: you set it in your dialplan Set($CALLERID(num)=...); |
| 13:23.27 | robin_sz | nah |
| 13:23.54 | robin_sz | No application Set for ... blah |
| 13:25.35 | EmleyMoor | Is there a way I can set the caller ID for the diverted leg of a call to that with which it came in?n (assuming, that is, it goes out on a channel that allows) |
| 13:25.58 | EmleyMoor | robin_sz: There is a callerid field in the Zap configs... |
| 13:26.30 | robin_sz | EmleyMoor, yeah tried that ... set it from asreceived to 01565656565 .. still went out as 5004 |
| 13:26.37 | EmleyMoor | callerid="Name" <number>, before the channel => |
| 13:26.49 | robin_sz | oh, in "" |
| 13:26.57 | robin_sz | Name? |
| 13:27.03 | robin_sz | nah, on the PSTN? |
| 13:27.11 | robin_sz | just numbers sureley? |
| 13:27.20 | EmleyMoor | Oh, for an outgoing call over the PSTN, you can't set it |
| 13:27.26 | robin_sz | eh? |
| 13:27.38 | robin_sz | you SURE about that? |
| 13:27.39 | EmleyMoor | It's fixed by your PSTN provider |
| 13:28.05 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 13:28.10 | robin_sz | you SURE about that? |
| 13:28.16 | EmleyMoor | robin_sz: Are you on ISDN? |
| 13:28.22 | robin_sz | yes |
| 13:28.35 | EmleyMoor | Ah, then the same should work, in that case |
| 13:28.44 | robin_sz | here are two lines from the PRI debug: |
| 13:28.45 | robin_sz | > Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) |
| 13:28.46 | robin_sz | > Presentation: Presentation permitted, user number not screened (0) '5004' ] |
| 13:28.46 | EmleyMoor | (subject to you having permission) |
| 13:29.12 | EmleyMoor | I would check with someone who knows more about ISDN than I do |
| 13:29.51 | robin_sz | so it has to go in < > brackets then? right? |
| 13:30.35 | EmleyMoor | If you are going to try presenting both a name and a number, then yes - it won't hurt if you aren't and may be worth a try |
| 13:31.26 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 13:33.17 | robin_sz | well, its still sending 5004 |
| 13:33.33 | robin_sz | anyway .. that wasnt the initial problem |
| 13:33.49 | robin_sz | the problem was being anable to make outgoing calls |
| 13:33.57 | robin_sz | unable |
| 13:35.39 | *** join/#asterisk adorah (n=user@89-139-1-37.bb.netvision.net.il) |
| 13:36.41 | robin_sz | ah well, debugging PRI frames is not my bag ... I suspect its back to mISDN |
| 13:46.55 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 13:46.56 | MrWup | hey |
| 13:47.30 | MrWup | how can i execute an app from a PHP AGI script so that it doesn't matter whether the php script terminates before the executed program? |
| 13:47.33 | *** join/#asterisk lorinc (n=ang@pool-4344.adsl.interware.hu) |
| 13:47.44 | MrWup | cause my PBX triggers an MP3 conversion |
| 13:47.49 | MrWup | and my php needs to set it off |
| 13:47.53 | MrWup | and then die |
| 13:48.04 | MrWup | and the lame mp3 conversion carries on and terminates when its finished |
| 13:53.28 | sasch | i need help .... |
| 13:53.51 | sasch | i want if a number is more than 5 characater asterisk send it in a context.... |
| 13:53.58 | sasch | what i made it ?? |
| 13:55.26 | sasch | GotoIf(${LEN(${exten})}>5?context1:context2) |
| 13:55.29 | sasch | ?? |
| 13:56.06 | *** join/#asterisk sysko (n=sysko@modemcable113.9-203-24.mc.videotron.ca) |
| 13:59.57 | Corydon76-home | sasch: GotoIf($[${LEN(${exten})}>5]?context1,s,1:context2,s,1) |
| 14:00.39 | sasch | <Corydon76-home> thank |
| 14:00.52 | sasch | I explain my project |
| 14:01.09 | sasch | i would when i digit a number with my phone asterisk count len of exten |
| 14:01.12 | Corydon76-home | Note that you had two problems: lack of extension and priority, and lack of the expression parser syntax |
| 14:01.27 | sasch | if > 5 go to a context .... |
| 14:01.51 | sasch | i must use exten => s,1, |
| 14:02.20 | Corydon76-home | Could, yes. You could also Goto(context1,${EXTEN},1) |
| 14:03.16 | Corydon76-home | oh, and also not that EXTEN must be capitalized |
| 14:10.15 | sasch | i have in my dial plan exten => _3.,1,Dial(${GLOBAL(TRUNK)}/${EXTEN}) |
| 14:10.20 | sasch | i want to add hint |
| 14:10.30 | sasch | exten => _3.,hint, ???? |
| 14:10.43 | Corydon76-home | hints currently do not work with patterns, sorry |
| 14:10.57 | sasch | in ???? what is the variable that asterisk know what sip account i use ?? |
| 14:10.59 | Corydon76-home | You'll need to specify each hint individually |
| 14:11.15 | Corydon76-home | sasch: there isn't one |
| 14:11.29 | Corydon76-home | sasch: you canNOT use pattern matching for hints |
| 14:12.38 | sasch | <Corydon76-home> ok :-( |
| 14:12.51 | sasch | and if I want to look status of my Zap line ?? |
| 14:13.01 | MrWup | anyone know how to execute external applications so that they run separately from asterisk? |
| 14:13.10 | MrWup | and dont just terminate when AGI script has finished? |
| 14:13.24 | Corydon76-home | sasch: same thing |
| 14:13.52 | sasch | <Corydon76-home> :-( |
| 14:14.06 | sasch | but this is a features of asterisk or not ?? |
| 14:14.18 | Corydon76-home | Devicestate is supported, yes |
| 14:14.20 | MrWup | noone? |
| 14:14.40 | Corydon76-home | MrWup: to what end? |
| 14:15.05 | robin_sz | sigh ... this is getting nowhere ... debugiing PRI is more than I need right now |
| 14:15.13 | *** join/#asterisk friedrich| (n=friedric@e177251240.adsl.alicedsl.de) |
| 14:15.21 | MrWup | Corydon76-home, an event triggers my PHP script to convert some wav files to mp3. so the php script needs to launch LAME to set about converting a load of files |
| 14:15.33 | MrWup | Corydon-w, and LAME needs to carry on doing it even if the php script dies |
| 14:15.38 | MrWup | and keep going till its finished |
| 14:15.52 | Corydon76-home | MrWup: you might consider starting that from a cron job, then |
| 14:15.54 | jer | MrWup, fork() the process in your script |
| 14:16.25 | MrWup | jer, fork() ? |
| 14:16.33 | Corydon76-home | If you simply background all those jobs, you may, in fact, cause additional issues by overloading CPU/memory in your Asterisk machine |
| 14:16.33 | robin_sz | fork or exec |
| 14:16.39 | MrWup | i tried exec |
| 14:17.05 | MrWup | i did exec("lame blah.wav blah.mp3") and it gets about 3bytes through converting before the PHP AGI script dies and it dies with it |
| 14:17.13 | jer | MrWup, it's a function call; |
| 14:17.21 | Corydon76-home | The best thing to do is to stick the files to be encoded in a spool directory and use an external process (started separately from Asterisk) to convert those files as they appear |
| 14:17.35 | robin_sz | forking stuff from PHP in ana apache enviroemnt? |
| 14:17.41 | MrWup | no no apache |
| 14:17.46 | MrWup | PHP CLI (run by AGI) |
| 14:17.49 | robin_sz | then why use PHP? |
| 14:18.04 | sasch | <Corydon76-home> http://www.asterisk.org/node/48342 |
| 14:18.06 | MrWup | because php is great |
| 14:18.06 | Corydon76-home | robin_sz: when all you have is a hammer, everything looks like a nail |
| 14:18.09 | robin_sz | its a web monkey language and has no place in systems |
| 14:18.18 | robin_sz | Corydon76-home, too true |
| 14:18.35 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:21.22 | robin_sz | we sacked an admin once (long time ago) .. wrote a bunch of cron tasks in PHP ... further digging revealed the extent of his madness |
| 14:21.39 | raptorra1 | correctness question: I've been setting up asterisk phones and while going over the documentation I've seen two ways to setup the phones (http://www.voip-info.org/wiki/view/Polycom+Phones#AsterisksipconfexampleforPolycomphones [note there are two methods there back to back]) which method is more correct for an end point device (e.g. a hard phone connecting to the pbx) |
| 14:22.11 | robin_sz | OK, I'll give bristuff another hour of effort, and then if I cant dial out its back to mISDN |
| 14:23.14 | Corydon76-home | raptorra1: whichever one works better for you |
| 14:23.50 | robin_sz | pity ... the DTMF detection is better on bristuff |
| 14:23.52 | raptorra1 | Corydon76-home: is there an advantage or reason for using one over the other |
| 14:24.09 | Corydon76-home | raptorra1: it's user-contributed documentation |
| 14:25.46 | robin_sz | its just the outgoing calls that are fialing, the network responds with a 'no such number' but it looks right, outgoing |
| 14:26.44 | *** join/#asterisk last1 (n=dood@modemcable215.11-57-74.mc.videotron.ca) |
| 14:28.40 | last1 | if a call comes in, and the caller chooses an extension that in turn dials out through one of our providers. ( towards a cell phone ) |
| 14:29.05 | last1 | currently on the cell phone, I see the caller id as being the asterisk's number with that particular provider, not the number of the caller |
| 14:29.25 | last1 | any way to pass the original caller id number to the cell phone ? |
| 14:29.42 | Corydon76-home | Not without the cooperation of your provider |
| 14:29.54 | robin_sz | some providers will. others wont |
| 14:29.57 | robin_sz | it depends |
| 14:30.08 | last1 | sort of like a way to stop me from entering fake caller id's ? |
| 14:30.15 | robin_sz | yep |
| 14:30.22 | *** join/#asterisk CVirus (n=GoD@196.205.192.34) |
| 14:30.38 | robin_sz | some do, some don't |
| 14:30.47 | CVirus | Is this possible using SIP protocol http://img377.imageshack.us/my.php?image=cairoalexlibyaxj4.png ? |
| 14:31.02 | Corydon76-home | If you explain what you're trying to do, your provider is more likely to be sympathetic |
| 14:31.04 | last1 | in case they do allow this, do I have to set up anything special or the caller id will simply follow up ? |
| 14:31.15 | Corydon76-home | Nothing special, no |
| 14:36.08 | robin_sz | JT, you suggested intense debug on pri, I have the output, but it doesnt mean much to me ... I see the PRI droppinghte call because its to an unassigned number, it looks right though ... |
| 14:37.16 | MrWup | isnt there some way to launch the equivalent of a batch file in debian? |
| 14:37.24 | robin_sz | yes, many |
| 14:37.56 | robin_sz | ./<name of file> for example |
| 14:38.14 | robin_sz | assuming you put a shebang at the top of the file |
| 14:41.10 | robin_sz | or place it in /usr/local/bin .. or some other place on your patah and then just type the name on the command line |
| 14:41.18 | CVirus | Is this possible using SIP protocol http://img377.imageshack.us/my.php?image=cairoalexlibyaxj4.png ? |
| 14:42.03 | robin_sz | yes |
| 14:42.10 | robin_sz | easier with IAX though |
| 14:42.19 | robin_sz | easier still with VPN |
| 14:42.42 | CVirus | robin_sz: what's hard about configuring my sip.conf and extensions.conf ? |
| 14:42.49 | robin_sz | nothing |
| 14:42.59 | CVirus | robin_sz: anything hidden that I don't know about ? |
| 14:43.03 | CVirus | robin_sz: I mean .. I'm still new to asteris |
| 14:43.05 | CVirus | asterisk* |
| 14:43.13 | robin_sz | its persauding your router to pass the RTP ports that usually fscks it all up |
| 14:43.26 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 14:43.27 | CVirus | hmm |
| 14:43.29 | robin_sz | I assume its NAT right? |
| 14:43.43 | CVirus | robin_sz: the right most one will be NAT yeah |
| 14:44.01 | robin_sz | well, you need to have it forward all the RTP ports to the * box ... |
| 14:44.13 | CVirus | I see |
| 14:44.29 | robin_sz | the RTP port range is set in sip.conf somewhere |
| 14:44.31 | robin_sz | in general |
| 14:44.39 | robin_sz | in [general] |
| 14:44.54 | CVirus | robin_sz: for the other boxes .. will I need to forward anything ? |
| 14:45.03 | robin_sz | depends on the firewalls |
| 14:45.20 | robin_sz | essentially its an outgoing connection, so the NAT usually handles it OK on its own |
| 14:45.32 | CVirus | I see |
| 14:45.33 | robin_sz | its the incoming connections that usually screw up |
| 14:46.00 | CVirus | robin_sz: with IAX I want bother regarding NAT ? |
| 14:46.01 | robin_sz | IAX sends cotnrol and data over the same connection, much easier |
| 14:46.24 | robin_sz | try the firefly softphone on wondows |
| 14:46.56 | CVirus | robin_sz: I was planning to use Linphone |
| 14:46.59 | robin_sz | with IAX2 you need only forward the IAX port |
| 14:47.08 | robin_sz | fine, if it has iax2 supprot |
| 14:47.19 | CVirus | lemme check |
| 14:48.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:48.26 | *** join/#asterisk JoeMoes (n=JoeMoes@wolverine.vcc.de) |
| 14:49.32 | CVirus | robin_sz: I guess I'll use firefly if I'll be using IAX |
| 14:49.38 | CVirus | linphone has no IAX support |
| 14:49.54 | CVirus | robin_sz: anyways ... regarding the diagram .. do you think it's okay ? |
| 14:50.02 | CVirus | robin_sz: this has nothing to do with asterisk I guess |
| 14:51.26 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 14:51.50 | funkmaster | hi there :) |
| 14:51.54 | robin_sz | yes, its fine |
| 14:51.56 | funkmaster | i got a problem with my asterisk server |
| 14:52.03 | funkmaster | it tells me WARNING[3015] chan_sip.c: Hanging up call a9e3e28f3344de20@192.168.1.138 - no reply to our critical packet. |
| 14:52.04 | CVirus | robin_sz: Thanks alot |
| 14:52.16 | funkmaster | anyone can tell me what's going wrong? |
| 14:54.08 | robin_sz | ok, 10 minutes to go ... |
| 14:55.08 | funkmaster | no one an idea? |
| 14:55.32 | funkmaster | i had it all working then updated my distro to newest debian, now i got these problems |
| 14:55.48 | funkmaster | calls come through but are disconnected after a minute or so with that message |
| 14:56.07 | funkmaster | i alos don't get what 192.168.1.138 is supposed to be |
| 14:57.32 | robin_sz | you are using a Alcatel Speedtouch ADSL modem? |
| 15:00.47 | JoeMoes | hi... i use a te120p card in my server, but i think its broken. if i start the loop back test, the red alarm lamp still on. any idea? |
| 15:01.35 | robin_sz | does that need a loopback cable? |
| 15:02.37 | JoeMoes | my loopback plug has pin1 going to 4 and pin2 going to 5 |
| 15:03.04 | JoeMoes | but the loop back lamp is still off :( |
| 15:03.28 | *** join/#asterisk nasls_lsa (n=chatzill@87.203.68.253) |
| 15:03.50 | robin_sz | oh dear |
| 15:04.37 | JoeMoes | bad sign? |
| 15:06.20 | robin_sz | OK, here I go, back to mISDN |
| 15:06.21 | robin_sz | shame, zaptel almost works |
| 15:06.35 | funkmaster | ok i really have no clue, happens to someone else, that calls get disconnected with the same message? |
| 15:06.46 | robin_sz | is there anything else I shold try before re-installing mISDN? |
| 15:07.55 | robin_sz | funkmaster, its probably the useless VOIP 'helper' app in a Speedtouch modem .. 192.168.1.138 is where they usualy sit |
| 15:08.25 | funkmaster | i got no speedtouch modem |
| 15:08.34 | funkmaster | also no network with 192.* |
| 15:08.56 | funkmaster | i only have my debian asterisk server which is connected to my grnadstream 486 on a 10* network |
| 15:08.58 | *** join/#asterisk Fieldy (i=TOpNHWQE@gentoo/contributor/Fieldy) |
| 15:09.17 | funkmaster | i get incoming calls but they disconnect with that message after a min or so |
| 15:09.51 | funkmaster | how can i fix this? |
| 15:10.45 | robin_sz | you cant |
| 15:11.00 | funkmaster | hm why not? |
| 15:11.12 | funkmaster | or what can i do to get it back working? was all fine b4 :/ |
| 15:11.13 | robin_sz | its the other party of the call probably |
| 15:11.48 | funkmaster | nope don't think so, calling from 3 different phones to my asterisk and all the same result |
| 15:12.02 | robin_sz | all on local network? |
| 15:12.28 | funkmaster | nope |
| 15:12.33 | funkmaster | all outside |
| 15:12.33 | robin_sz | riiiight ... |
| 15:12.43 | funkmaster | differnet networks |
| 15:13.02 | MrWup | there has got to be some way of spawning a background process from php AGI |
| 15:13.09 | MrWup | how crazy that there wouldnt be |
| 15:14.07 | robin_sz | MrWup, these scripts runing as root? |
| 15:14.19 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:14.19 | MrWup | yeah |
| 15:14.35 | robin_sz | just a guess |
| 15:15.00 | MrWup | oh fuck off |
| 15:15.43 | robin_sz | you are the one crazy enough to run a web monkey language, as root, for sysadmin scripts ... not me. |
| 15:16.01 | robin_sz | its easy in Perl or bash or python or ... |
| 15:16.05 | MrWup | you're an arrogant twit and ive been awake too long to bitch with you |
| 15:16.43 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:17.28 | robin_sz | learn a real language, then come back. |
| 15:18.08 | funkmaster | hmz outgoing calls work fine and r not interrupted but the incoming still breaks up after a minute... |
| 15:18.57 | funkmaster | i don't get what the problem is |
| 15:18.59 | funkmaster | :S |
| 15:19.17 | robin_sz | funkmaster, its a firewall issue almost certainly |
| 15:19.44 | robin_sz | somewhere a firewall is closing a connection because it thinks its no longer in use |
| 15:20.09 | jer | can't we all just get along? if he wants to use php, how does that hurt you? |
| 15:20.30 | LeddyHM | jer: we can once nufone sip works for me ;) |
| 15:20.53 | LeddyHM | I signed up friday, still can't use it |
| 15:20.56 | robin_sz | jer, you realling think running PHP, as root, from the CLI is a good plan? |
| 15:21.11 | jer | LeddyHM, ? |
| 15:21.36 | LeddyHM | greg was going to have you look into it last night |
| 15:21.38 | jer | robin_sz, no |
| 15:22.00 | robin_sz | jer, secondly, he cant manage what he wants to do, and its trivial in Perl or bash ... |
| 15:22.09 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 15:22.12 | JoeMoes | hm... anything else can i check my card? |
| 15:22.15 | jer | LeddyHM, i'm probably not who you think i am |
| 15:22.33 | LeddyHM | :( |
| 15:22.34 | funkmaster | i'm not sure, both the grandsteam box and the asterisk server are behind my firewall, whicih is on a third machine, there were no changes made, so i don't think that's it.. |
| 15:22.38 | funkmaster | anz other suggestions? |
| 15:22.49 | LeddyHM | maybe so |
| 15:23.00 | jer | LeddyHM, infact, i'm extremely green when it comes to asterisk =] |
| 15:23.05 | robin_sz | funkmaster, you didnt mention it was grandstream |
| 15:23.15 | robin_sz | funkmaster, you have a hammer? |
| 15:23.22 | funkmaster | yep did sumwhere up there |
| 15:23.24 | funkmaster | lol |
| 15:23.59 | *** join/#asterisk d4rkst4r75 (n=d4rkst4r@ip-41-112.sn1.eutelia.it) |
| 15:24.21 | funkmaster | yep same here, but it was all working this morning then i did sum updates etc and now it'S screwed.. |
| 15:24.23 | d4rkst4r75 | hi, is there someone using asterisk 1.4 + sangoma A10x boards? |
| 15:25.24 | funkmaster | i still find it weird that it mentions a network, which i don't have... |
| 15:25.25 | funkmaster | WARNING[3245]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission 21dab95e3fe6970b@192.168.1.138 for seqno 59168 (Critical Response) |
| 15:25.25 | funkmaster | [May 6 19:23:58] WARNING[3245]: chan_sip.c:1898 retrans_pkt: Hanging up call 21dab95e3fe6970b@192.168.1.138 - no reply to our critical packet. |
| 15:25.55 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 15:26.06 | LeddyHM | TK! |
| 15:26.18 | [TK]D-Fender | Where?! Where?! |
| 15:26.30 | LeddyHM | you tell me |
| 15:27.40 | robin_sz | I just hope after all this, mISDN reinstalls nicely |
| 15:29.46 | MrWup | all my logic is in php, accessing mysql databases etc. it would be a nightmare to change it |
| 15:30.11 | MrWup | if what you're saying is that i can execute a bash script from php which would do what i want, what would be fine |
| 15:35.31 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 15:37.59 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 15:40.50 | robin_sz | I dont do PHP, but I would imagine it needs to do an exec() call ... in Perl its exec("<command>"); or system("<command & disown>"); |
| 15:40.55 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 15:44.44 | robin_sz | tell yo uwhat, just for you, I'll look at the PHP documentation online |
| 15:45.54 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 15:46.11 | robin_sz | yep .. straight to it |
| 15:47.06 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:47.31 | robin_sz | basically, you want to call some external program or command from a PHP script, and have it just run, finishing its run even if the PHP script terminates before its done |
| 15:48.18 | MrWup | thats right |
| 15:48.33 | robin_sz | weird .. the docs for that decalre it as a void return type, |
| 15:48.37 | MrWup | and none of php's commands such as exec() system() etc seem to achieve that |
| 15:48.50 | robin_sz | then go on to describe it as returning false on error ... |
| 15:49.13 | MrWup | even trying "lame blah blah blah" > /dev/null 2>&1 & |
| 15:49.37 | robin_sz | & disown ? |
| 15:49.41 | MrWup | yeah |
| 15:49.44 | robin_sz | but how are you calling it? |
| 15:49.50 | MrWup | Exec() in php |
| 15:49.56 | MrWup | and system() |
| 15:50.08 | robin_sz | Exec shoud work, would in Perl |
| 15:50.10 | MrWup | neither of which seem to actually make it disown |
| 15:50.23 | robin_sz | system shoudl not do that |
| 15:50.46 | robin_sz | tried nohup? |
| 15:50.51 | MrWup | nope |
| 15:51.13 | raptorra1 | robin_sz: been a long time since I did php but does the "passthru" command get you what you want? |
| 15:51.41 | robin_sz | raptorra1, it wouldnt get me what I want ... because I hate PHP in all its forms |
| 15:52.05 | robin_sz | try MrWup , hes the one writing sysadmn CLI root scripts in it :) |
| 15:52.19 | MrWup | nohup says cannot run, permission denied |
| 15:52.37 | robin_sz | i think hes doing some sort of voice mail to email thing |
| 15:52.48 | robin_sz | encoding to mp3 |
| 15:52.51 | raptorra1 | robin_sz: sorry... only half reading the channel... I'm actually RTFM'ing at the moment (literally) |
| 15:53.24 | raptorra1 | square peg / round hole? |
| 15:53.43 | robin_sz | raptorra1, no problem, mmm .. more like turd shaped peg, round hole |
| 15:55.21 | robin_sz | OK, chan_mISDN reinstalled ... back to where I was .. another day of my life slips by |
| 15:55.33 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:55.56 | robin_sz | bristuff oh so nearly worked too |
| 15:57.32 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:58.52 | MrWup | hah |
| 15:58.59 | MrWup | well i think ive found the problem |
| 15:59.03 | MrWup | not dear old php... but silly me |
| 15:59.23 | robin_sz | forgot to set exec flag on file? |
| 15:59.24 | MrWup | the lame mp3 app is executed just before the MixMonitor stream is closed and saved to file |
| 15:59.37 | blitzrage | morning all |
| 15:59.41 | MrWup | as the php script is called on hangup |
| 15:59.45 | robin_sz | evening |
| 15:59.49 | blitzrage | in the http.conf file, what does enablestatic=yes do exactly? |
| 16:00.02 | MrWup | thing is... im not sure how i'd get the php script to be called only after the mixmonitor stream has closed |
| 16:00.27 | MrWup | unless i can end the recording myself somehow from within the php script |
| 16:00.32 | MrWup | by sending an AGI command |
| 16:00.49 | blitzrage | MrWup: that's the issue I had too -- what I ended up having to do was actually make my DeadAGI() check for the existance of the unmixed files, and mix them myself if they were not already mixed |
| 16:01.04 | uwe | hello, im getting results in zttest as law as 99.475098 , 99.157715 and 99.780273 with a TE410p, even though i 1) disabled as much hardware as i could 2) set the irqs so that the digium card doesnt share it with any other device 3)distrubuted the inturreupts using smp_affinity and tried to disable the HT cpu ... what else can be done ? |
| 16:01.06 | blitzrage | MrWup: that's not a bad idea if you can figure out how to force Asterisk to end the recording |
| 16:01.33 | MrWup | yeah its a tricky one |
| 16:01.58 | blitzrage | basically, I had to do like I said above... |
| 16:02.08 | blitzrage | sometimes it mixed, and sometimes it doesn't, depending on who hung up |
| 16:02.09 | MrWup | well mine is all premixed with mixmonitor |
| 16:02.29 | blitzrage | well, mixmonitor mixes it automatically after it ends -- it doesn't mix it during the call |
| 16:02.33 | blitzrage | so you still have separate files |
| 16:02.37 | MrWup | oh really |
| 16:02.38 | *** join/#asterisk atca_pres (n=rick@modemcable200.50-201-24.mc.videotron.ca) |
| 16:02.39 | blitzrage | yes -- I was using mixmonitor to |
| 16:02.46 | MrWup | where do they keep these sep files? |
| 16:02.50 | blitzrage | MrWup: ya, actually look in the /var/spool/asterisk/monitor directory |
| 16:02.52 | MrWup | theyre not in the var spool asterisk monitor dir |
| 16:02.58 | blitzrage | that's where they should be |
| 16:02.59 | MrWup | huh |
| 16:03.00 | MrWup | well |
| 16:03.03 | blitzrage | that's where they are for me |
| 16:03.06 | blitzrage | look during the call |
| 16:03.11 | MrWup | for me there is only one file put there during the call |
| 16:03.23 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
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| 16:03.36 | MrWup | yeah |
| 16:03.37 | MrWup | one single wav |
| 16:03.45 | robin_sz | are there sound files in english for * somewhere? |
| 16:03.59 | MrWup | of 0 bytes, until the call ends, and then it turns into 250kb or whatever |
| 16:04.09 | robin_sz | rtaher than american ... |
| 16:05.24 | atca_pres | hi everyone |
| 16:05.51 | atca_pres | I got a little setup here that doesn't work with my iax |
| 16:06.25 | atca_pres | anyone know why the first get me a No Authority and the second works ? |
| 16:06.26 | atca_pres | switch => IAX2/User@server/context |
| 16:06.26 | atca_pres | exten => _1XXXXXXXXXX,1,Dial(IAX2/User@server/${EXTEN},20,R) |
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| 16:18.33 | MrWup | blitzrage, how did you mix them if they were still open for writing? |
| 16:18.55 | MrWup | that seems to be my problem. at the time the php script is running, the files are all locked for writing |
| 16:20.10 | blitzrage | hrmmm |
| 16:20.14 | blitzrage | it just seemed to work... |
| 16:20.30 | blitzrage | they may not have been mixed, but the call was ended, so they must have been unlocked |
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| 16:22.54 | MrWup | mm |
| 16:24.04 | MrWup | just wish i could spend a day doing productive things instead of wasting hours over such simple things as converting a wav to an mp3 |
| 16:24.05 | rue_mohr | wav to mp3? use bladeenc |
| 16:24.05 | MrWup | im using lame, its fine |
| 16:24.05 | MrWup | its asterisk thats the problem |
| 16:24.08 | MrWup | having things in a cocked up order as usual |
| 16:24.25 | MrWup | like the inherited variables are all dumped at hangup when a channel goes zombie |
| 16:24.27 | MrWup | fat lot of use that is |
| 16:24.37 | rue_mohr | hmm |
| 16:24.47 | MrWup | before the hangup logic is executed |
| 16:24.55 | MrWup | would have made sense to zombify the channel |
| 16:24.58 | MrWup | execute all the logic |
| 16:25.02 | MrWup | and then clear it out |
| 16:25.05 | rue_mohr | I have to figure out how to add machine states for my newbridge channelbank |
| 16:25.08 | MrWup | gah i dont know |
| 16:25.21 | MrWup | theres got to be some PBX thats less strife |
| 16:25.54 | rue_mohr | ? |
| 16:26.14 | rue_mohr | you just after background music? |
| 16:26.35 | MrWup | no |
| 16:26.43 | MrWup | im trying to record calls and then convert them to mp3 |
| 16:26.56 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 16:26.56 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:26.56 | rue_mohr | erm |
| 16:27.01 | MrWup | it looks like im going to have to do some sort of cron job that scours the monitor folder for wavs |
| 16:27.08 | MrWup | and then converts them to mp3s and then moves them |
| 16:27.14 | MrWup | what a thoroughly dirty way of doing it |
| 16:27.15 | rue_mohr | its recording it to wav? or ...(asf)? |
| 16:27.19 | MrWup | wav |
| 16:27.23 | rue_mohr | k |
| 16:27.32 | MrWup | that side of things is all fine... records it all to wav as it should |
| 16:27.35 | rue_mohr | so, how could anything be giving you trouble there? |
| 16:27.53 | MrWup | just that asterisk doesnt release the wav from writing before your AGI script is executed at hangup |
| 16:28.05 | MrWup | so when it comes to doing anything with the recording the AGI script is powerless |
| 16:28.07 | rue_mohr | ah, that could be complex |
| 16:28.25 | MrWup | its not complex its just fucked up |
| 16:28.43 | rue_mohr | well, sounds like moving a function call up by few lines to me |
| 16:28.54 | rue_mohr | idealy |
| 16:29.24 | rue_mohr | but I'm lost tying to figure out where the demo ivr is :) |
| 16:29.44 | MrWup | yeah thats what it sound like to me too |
| 16:29.52 | MrWup | would be interesting to find out where in the source that is |
| 16:29.59 | MrWup | and then i can just edit the damn thing |
| 16:30.05 | MrWup | but that will take years of digging |
| 16:32.40 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 16:33.12 | MrWup | "<command> will be executed when the recording is over\n" |
| 16:33.15 | MrWup | OH |
| 16:33.46 | rue_mohr | ? |
| 16:33.55 | rue_mohr | is the file closed by then? |
| 16:34.03 | MrWup | i think you can define a command in mixmonitor to be executed as soon as the recording is over |
| 16:34.36 | MrWup | id hope so |
| 16:34.53 | *** part/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:35.37 | rue_mohr | aha, found the ivr |
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| 16:43.01 | d4rkst4r75 | hi to all |
| 16:43.12 | Zeeek | indeed |
| 16:43.37 | d4rkst4r75 | someone can give me some hints on what versions are better for an asterisk + sangoma A102 board? |
| 16:44.12 | errr | d4rkst4r75: versions of what? |
| 16:44.23 | d4rkst4r75 | wanpipe & asterisk |
| 16:44.54 | errr | d4rkst4r75: get the latest wanpipe from sangoma and I use 1.2.18 with our setup |
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| 16:46.03 | errr | wanpipe will need to patch zaptel so make sure you dont install zaptel first cause you will have to rebuild it again if you do |
| 16:47.20 | d4rkst4r75 | errr you use asterisk 1.2 and what zaptel verison? |
| 16:47.21 | errr | but the wanpipe installer kinda automates all that anywaay |
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| 16:47.25 | errr | d4rkst4r75: 1.2.x asterisk is used with 1.2.x zaptel |
| 16:48.23 | d4rkst4r75 | errr i've made a setup with asterisk 1.4 + zaptel 1.4.2.1 ? |
| 16:48.23 | errr | well thats your choice :) |
| 16:48.23 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 16:48.23 | d4rkst4r75 | but i've experienced some problems with PRI |
| 16:48.53 | d4rkst4r75 | after a while all the channels go blocked |
| 16:48.56 | d4rkst4r75 | so i think i'm not using stable versions |
| 16:49.31 | d4rkst4r75 | errr, what kernel version you use for your production systems? |
| 16:49.43 | errr | call sangoma, ask them what they reccomend, they provide awesome tech support. and as much as you pay for the hardware you might as well use it |
| 16:50.19 | errr | d4rkst4r75: I use centos 5 I dont recall the kernel version.. i think its 2.6.18 |
| 16:50.34 | d4rkst4r75 | thank you errr |
| 16:50.48 | d4rkst4r75 | i found a102d board is a great hardware |
| 16:51.01 | d4rkst4r75 | but i'm at the first experience with digital boards |
| 16:51.12 | d4rkst4r75 | and the first with sangoma : |
| 16:51.15 | errr | yeah we have 8 of them, we use them in our routers I built and in the pbx systems for our pri's |
| 16:51.42 | rue_mohr | anyone know if I can get asterisk to tell me all the state changes in a channels status? |
| 16:51.45 | [TK]D-Fender | errr, Cost is on par with Digium.... |
| 16:52.15 | d4rkst4r75 | errr, just another question about PRI |
| 16:52.26 | errr | [TK]D-Fender: my boss picked the hardware I just installed it and set it up. I never even did any research on that stuff |
| 16:52.40 | d4rkst4r75 | here in Italy we have E1 so D-Chan is on chan 16 |
| 16:53.26 | d4rkst4r75 | when i load wanrouter, the channels are initialized |
| 16:53.42 | d4rkst4r75 | but is only when i start asterisk that the channels go unblocked |
| 16:53.45 | d4rkst4r75 | is this right? |
| 16:53.49 | [TK]D-Fender | d4rkst4r75, I've heard of a number of compile issues with CentOS 5, and related * bits |
| 16:54.19 | errr | the only problems I had was with zaptel |
| 16:54.33 | errr | the source is hard coded for the wrong kernel version |
| 16:55.05 | d4rkst4r75 | i've compiled all on ubuntu 6.1 server |
| 16:55.37 | d4rkst4r75 | but i find that zaptel patching is not working fine (using zaptel 1.4.x) |
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| 16:56.14 | mvanbaak | oeh. |
| 16:56.28 | mvanbaak | the new rthreads stuff in openbsd 4.1 really makes asterisk happy |
| 16:57.07 | mvanbaak | still no zaptel support. but that's ok for me |
| 16:57.10 | mvanbaak | all voip here |
| 16:57.22 | errr | d4rkst4r75: I think wanpipe is setup for 1.2.x asterisk and zaptel |
| 16:57.41 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 16:57.49 | d4rkst4r75 | yes |
| 16:57.56 | errr | d4rkst4r75: any reason you are using 1.4 instead of 1.2? |
| 16:58.11 | d4rkst4r75 | errr i'm asking all my questions for this :) |
| 16:58.36 | d4rkst4r75 | i'm trying to understand what version is stable for production purposes |
| 16:58.52 | errr | I may be wonrg but I think 1.2.x is stable and 1.4.x is more development |
| 16:58.57 | d4rkst4r75 | errr: i'm planning also to change distro |
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| 16:59.27 | robin_sz | can't seem to get it to compile on Sarge with a 2.6 kernel |
| 16:59.43 | errr | robin_sz: which kernel version is it? |
| 16:59.54 | robin_sz | 2.6.8-2-686 I think |
| 17:00.03 | errr | uname -r to be sure |
| 17:00.14 | robin_sz | there something screwed up in the makefile ... paths or somehting to includes |
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| 17:00.33 | robin_sz | 2.6.8-2-686 I think |
| 17:00.39 | errr | pastebin the build error |
| 17:00.40 | robin_sz | for sure |
| 17:01.11 | mvanbaak | robin_sz: debian etch is stable for some months now |
| 17:01.16 | mvanbaak | upgrade to it |
| 17:01.31 | mvanbaak | it comes with 2.6.18-4 |
| 17:01.41 | robin_sz | youve been using windows too long |
| 17:01.53 | d4rkst4r75 | lol |
| 17:01.57 | mvanbaak | gheh. I never used windows |
| 17:02.54 | mvanbaak | started with debian, now using more and more OpenBSD |
| 17:02.57 | robin_sz | http://nopaste.snit.ch:8001/10109 |
| 17:03.02 | robin_sz | errr, ^^ |
| 17:03.06 | d4rkst4r75 | errr: you're using wanpipe-2.3.4-8.tgz ? |
| 17:03.27 | errr | d4rkst4r75: no, they made a special version for centos 5 that I am using |
| 17:03.31 | robin_sz | mvanbaak, well, problems building part of one app are not generally a good reason for ripping out the entire distro |
| 17:04.07 | robin_sz | mvanbaak, and I need to be able to come back to where I am now, or there will be no phone calls on Tuesday morning |
| 17:04.13 | mvanbaak | robin_sz: I know. but since sarge is now oldstable and etch being stable I think it's a good idea to upgrade anywayz |
| 17:04.20 | robin_sz | why? |
| 17:04.30 | mvanbaak | because a lot have been fixed |
| 17:04.42 | errr | robin_sz: you do have the kernel source installed right? |
| 17:04.58 | robin_sz | errr, I have lots of stuff installed, yes |
| 17:05.16 | mvanbaak | my sarge was stable and all |
| 17:05.20 | mvanbaak | but etch just works better |
| 17:05.24 | errr | robin_sz: but kernel-sources-2.6.8 or what ever that package is called? |
| 17:05.30 | robin_sz | yes |
| 17:05.31 | mvanbaak | and your software will be more up-to-date |
| 17:05.46 | d4rkst4r75 | robin_sz: http://forums.vpslink.com/showthread.php?t=1123 |
| 17:05.52 | robin_sz | and symlinked to /usr/src/linux |
| 17:05.52 | d4rkst4r75 | take a look there :) |
| 17:06.48 | robin_sz | someone with a similar problem ,... and no solution |
| 17:07.01 | d4rkst4r75 | i think you've not your kernel configured |
| 17:07.08 | errr | thats not like any of the errors I have run into |
| 17:07.17 | d4rkst4r75 | robin_sz: you've the sources but your kernel sources are not configured |
| 17:07.23 | robin_sz | configured? |
| 17:07.27 | d4rkst4r75 | yes |
| 17:07.47 | errr | if you got the sources via apt-get they are |
| 17:07.54 | robin_sz | I did |
| 17:08.58 | d4rkst4r75 | but this line from your pastebin is clear: Makefile:418: .config: No such file or directory |
| 17:09.38 | robin_sz | /usr/src/kernel-headers-2.6.8-2-686/.config |
| 17:09.38 | robin_sz | /usr/src/kernel-headers-2.6.8-2/.config |
| 17:10.03 | mvanbaak | you dont need the kernel-source to compile zaptel |
| 17:10.08 | mvanbaak | kernel-headers is enough |
| 17:10.14 | robin_sz | mmm ... |
| 17:10.24 | mvanbaak | symlink to the correct kernel-headers dir |
| 17:10.33 | errr | its looking for that file in /usr/src/kernel-source-2.6.8 so you did symlink right? |
| 17:10.38 | robin_sz | it complained about not having it .. so I installed it and symlinked it like it asked me to |
| 17:10.39 | d4rkst4r75 | but the error is from dir /usr/src/kernel-source-2.6.8 |
| 17:11.00 | robin_sz | so, to the headers, NOT the kernel source? |
| 17:11.04 | mvanbaak | d4rkst4r75: yeah. debian comes with the .config file in the kernel-headers package |
| 17:11.07 | mvanbaak | robin_sz: indeed |
| 17:11.15 | robin_sz | oakey doakey |
| 17:12.49 | robin_sz | this is BRIstufffffffed |
| 17:12.55 | robin_sz | trying again |
| 17:13.01 | mvanbaak | aaaaaaaaaaaah |
| 17:13.11 | robin_sz | you are a dentist? |
| 17:13.25 | mvanbaak | dont forget the /usr/src/linux-2.6 symlink |
| 17:13.36 | robin_sz | done it |
| 17:13.36 | d4rkst4r75 | errr, a special version for centos? |
| 17:14.11 | robin_sz | hopefully this time it works ... the erlier bristuff failed miserably |
| 17:14.23 | d4rkst4r75 | robin_sz: have you successfully compiled? |
| 17:14.26 | mvanbaak | what bristuff version are you compiling |
| 17:14.33 | errr | d4rkst4r75: wanpipe-2.3.4-7.tgz but its in a special place on their ftp for centos 5 |
| 17:15.11 | robin_sz | mvanbaak, latest bristuff-0.3.0-PRE-1y-e |
| 17:15.17 | mvanbaak | ah |
| 17:15.18 | robin_sz | on Saaaaaaaarge |
| 17:15.26 | mvanbaak | that one compiles fine on my etch machines |
| 17:15.35 | mvanbaak | on sarge too btw |
| 17:15.42 | mvanbaak | oh no, sorry |
| 17:15.45 | robin_sz | the earlier one isntalled, but could not dial out |
| 17:15.52 | mvanbaak | the older machines are running testing |
| 17:15.57 | mvanbaak | 3 months old |
| 17:16.02 | mvanbaak | so that's basically etch |
| 17:16.18 | errr | d4rkst4r75: wget ftp://ftp.sangoma.com/linux/custom/Yuan/wanpipe-2.3.4-7-CentOS5.tgz |
| 17:16.40 | robin_sz | I get some warnings .. |
| 17:16.41 | robin_sz | *** Warning: "zt_register" [/root/bristuff/bristuff-0.3.0-PRE-1y-e/ztgsm/ztgsm.ko] undefined! |
| 17:16.45 | robin_sz | like that .. |
| 17:16.48 | robin_sz | quite a lot actually |
| 17:17.02 | d4rkst4r75 | errr: had you some stability problem with PRI in your configuration? |
| 17:17.21 | mvanbaak | robin_sz: I have them too |
| 17:17.29 | mvanbaak | I ignore them ;) |
| 17:17.31 | *** join/#asterisk saftsack (n=saftsack@pD9E079E4.dip.t-dialin.net) |
| 17:17.32 | errr | d4rkst4r75: so far none yet, but we just moved from fedora 6 to centos 5 last week |
| 17:17.38 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-209-201.socal.res.rr.com) |
| 17:18.04 | d4rkst4r75 | err: so you suggest a centos installation? |
| 17:18.14 | mvanbaak | I wont |
| 17:18.16 | mvanbaak | get debian |
| 17:18.23 | LeddyHM | centos |
| 17:18.29 | LeddyHM | 4.4, not 5 |
| 17:18.50 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 17:18.50 | errr | d4rkst4r75: yes but only because I am a fedora package maintainer and I am a huge redhat fanboy |
| 17:18.55 | robin_sz | debian .. the one true way |
| 17:19.10 | errr | LeddyHM: why 4.4 5 is out now and will have a 7 year shelf life |
| 17:19.37 | d4rkst4r75 | robin_sz: i work mainly with gentoo :) but for me linux is GNU/Linux and not debian or others |
| 17:19.56 | robin_sz | I got fed up with redhat, after I got a box stuck on 7.2 |
| 17:19.57 | d4rkst4r75 | robin_sz: so i prefer for some tasks a specific distro instead of another |
| 17:19.57 | LeddyHM | I'm not an early adopter |
| 17:20.05 | errr | I reccomend what ever you are most comfortable using |
| 17:20.07 | robin_sz | mever did manage to get it upgradedto anyting later |
| 17:20.10 | LeddyHM | too many packages need "assistance" |
| 17:20.32 | robin_sz | and, indeed, there it sits, still with RH7.2 on it |
| 17:21.01 | mvanbaak | if I really _HAVE_ to use linux I use debian |
| 17:21.08 | mvanbaak | but I prefer not to |
| 17:21.11 | robin_sz | anyway ... |
| 17:21.15 | robin_sz | its building |
| 17:21.25 | errr | LeddyHM: I havent had any problems yet, but I have had to build about 30 rpms for our setup so far but not really due to asterisk, mostly due to apps I wronte that work with our setup.. for faxes and various other php things we do |
| 17:21.44 | *** join/#asterisk ruied_ (n=ruied@bl7-212-239.dsl.telepac.pt) |
| 17:21.50 | mvanbaak | I use OpenBSD where I can |
| 17:21.52 | LeddyHM | hence my comment :) |
| 17:22.00 | MrWup | i just dont get it |
| 17:22.04 | MrWup | i really dont get this shit |
| 17:22.24 | robin_sz | MrWup, can I give you hint? |
| 17:22.29 | MrWup | when mixmonitor has finished, it calls a command line... |
| 17:22.32 | MrWup | this isnt even PHP now |
| 17:22.44 | MrWup | im using command line directly exec by asterisk |
| 17:23.15 | MrWup | the command line is lame 122.wav 122.mp3 |
| 17:23.31 | MrWup | basically it takes the existing .wav file which was made by mixmonitor and makes an mp3 |
| 17:23.42 | MrWup | thing is, the mp3 is always only 1.2kb in size |
| 17:23.56 | MrWup | and nothing can be heard inside it |
| 17:24.20 | MrWup | if i run the same command at the terminal, it creates a proper mp3, good size, full of sound |
| 17:24.45 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 17:24.51 | robin_sz | same command, different place |
| 17:25.00 | *** join/#asterisk wolferine (n=profx@unaffiliated/wolferine) |
| 17:25.05 | MrWup | yeah |
| 17:25.08 | wolferine | morning |
| 17:25.10 | robin_sz | paths dear boy, paths |
| 17:25.11 | errr | LeddyHM: the rpms I built were not in centos period thoough, so I would have had to build them in 4.4 not to mention I would have had to build php from source cause all my stuff depends on php5 which isnt in 4.4 |
| 17:25.17 | mvanbaak | MrWup: did you use the full path ? |
| 17:25.28 | MrWup | robin_sz, yeah i use the full path in both commands |
| 17:25.28 | robin_sz | and full paths to the source files? |
| 17:25.41 | MrWup | the command is literally |
| 17:25.59 | MrWup | lame /var/spool/asterisk/monitor/122.wav /var/spool/asterisk/monitor/122.mp3 |
| 17:26.05 | MrWup | in both cases |
| 17:26.08 | mvanbaak | use the full path to lame |
| 17:26.12 | MrWup | ok |
| 17:26.20 | d4rkst4r75 | mvanbaak: what hardware you suggest for a dual PRI (30 + 30) and 60 SIP operators? |
| 17:26.36 | mvanbaak | d4rkst4r75: sangoma A102 with EC |
| 17:26.51 | mvanbaak | d4rkst4r75: the box itself can be any normal current server |
| 17:27.06 | mvanbaak | as long as it can hold 1 pci card |
| 17:27.07 | d4rkst4r75 | mvanbaak: i mean as server hardware |
| 17:27.15 | d4rkst4r75 | i've got a sangoma a102d |
| 17:27.18 | d4rkst4r75 | :) |
| 17:27.28 | MrWup | ok... using the full path to lame in both cases now |
| 17:27.31 | mvanbaak | d4rkst4r75: we run setups like that with normal p4 3ghz and 1GB ram |
| 17:27.41 | mvanbaak | motherboard is always intel based |
| 17:27.43 | MrWup | still, when asterisk calls it, it gives a shitty junk mp3, when i call it, it works |
| 17:27.48 | mvanbaak | the via shit is very unstable |
| 17:28.06 | d4rkst4r75 | thankyou! |
| 17:28.10 | mvanbaak | ur welcome |
| 17:28.14 | errr | d4rkst4r75: our servers are dell poweredege 1950's with quad dual core xeon 3.0s |
| 17:28.15 | MrWup | /usr/local/bin/lame /var/spool/asterisk/monitor/124.wav /var/spool/asterisk/monitor/124.mp3 |
| 17:28.39 | mvanbaak | actually, most of our production servers are normal whitelabel p4 3ghz 1gbram aopen mobo boxen |
| 17:28.55 | MrWup | i type that into terminal and it works just great |
| 17:28.58 | mvanbaak | 2 or 3 intel 100mbit cards in it |
| 17:29.02 | MrWup | i dont understand whats up with * |
| 17:29.17 | mvanbaak | we are now moving to sun X2200M boxen |
| 17:29.23 | d4rkst4r75 | errr: i was thinking to use dell poweredge 1950, but i saw that there are problems with dell servers... |
| 17:29.31 | d4rkst4r75 | i saw = i read somewhere |
| 17:29.37 | mvanbaak | with dell you get what you pay for |
| 17:29.39 | mvanbaak | little |
| 17:29.48 | d4rkst4r75 | LOL |
| 17:29.54 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 17:30.11 | d4rkst4r75 | mvanbaak, dell servers are not that bad as you say:) |
| 17:30.25 | LeddyHM | mmm dell |
| 17:30.27 | mvanbaak | d4rkst4r75: I dont like them. but indeed that's just my opinion |
| 17:30.33 | errr | d4rkst4r75: we use all dells everywhere in our operation, we have 40 or so rack mounts and over 200 desktops |
| 17:30.49 | mvanbaak | I prefer HP DL380 or sun |
| 17:31.12 | errr | my boss is sold on the service contract we have.. like some kind of 24hr repair thing |
| 17:32.08 | mvanbaak | thing is: we do everything with Debian or OpenBSD |
| 17:32.14 | mvanbaak | for OpenBSD you wont get support |
| 17:32.16 | errr | we use freebsd |
| 17:32.18 | MrWup | is it possible that asterisk is not running with the same powers that i have from a terminal? |
| 17:32.23 | MrWup | if so, how can i correct this? |
| 17:32.28 | mvanbaak | but HP is the only one that gives official support for Debian Linux |
| 17:32.32 | errr | the only linux boxes we have are our pbx systems |
| 17:32.40 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 17:32.41 | mvanbaak | so IF we have to use linux we get a HP DL380 |
| 17:32.47 | d4rkst4r75 | :) |
| 17:32.48 | errr | what do you mean support for the OS?? |
| 17:32.58 | mvanbaak | all the others are running sun X2200M with debian or openbsd |
| 17:33.09 | errr | like I would ever want some dell retard helping me trouble shoot something with the OS |
| 17:33.09 | mvanbaak | and debian is only used on machines that need virtualization |
| 17:33.17 | d4rkst4r75 | what about ibm servers? |
| 17:33.35 | mvanbaak | errr: there's the difference. HP really has skilled ppl for their Debian support desk |
| 17:33.46 | d4rkst4r75 | like x3650 dual xeon? |
| 17:33.55 | mvanbaak | d4rkst4r75: didn't used them. too much power consumption for me |
| 17:33.59 | MrWup | =[ |
| 17:34.01 | errr | mvanbaak: we just require all our admins to be damn good with linux and freebsd |
| 17:34.18 | mvanbaak | errr: same here |
| 17:34.34 | mvanbaak | because sun will never help you with trouble if you are not running solaris |
| 17:34.41 | d4rkst4r75 | errr: we prefer windows for sysadmins :) |
| 17:34.45 | mvanbaak | only when the hardware breaks down |
| 17:34.50 | d4rkst4r75 | for linux we ask geeks :) |
| 17:34.59 | errr | d4rkst4r75: Id rather do manual labor than work with windows all day |
| 17:35.06 | d4rkst4r75 | lool |
| 17:35.17 | mvanbaak | I'd kill myself before working with windows |
| 17:35.44 | mvanbaak | I dont know it |
| 17:35.48 | mvanbaak | I dont wanna know it |
| 17:35.56 | errr | I wouldnt go that far but I have done and will do manual labor before taking a job working with windows all day |
| 17:36.04 | MrWup | its a lot easier than bloody linux |
| 17:36.14 | mvanbaak | MrWup: it is |
| 17:36.14 | errr | MrWup: no its not |
| 17:36.19 | MrWup | windows is great if you dont have several years to spend in a dark hole |
| 17:36.20 | mvanbaak | that's why I dont like linux neither |
| 17:36.32 | errr | psh |
| 17:36.42 | d4rkst4r75 | MrWup, i think you're wrong |
| 17:36.50 | errr | I have a lamp in my cave.. its not all dark |
| 17:37.05 | mvanbaak | I mean. try to get documentation about the bonding stuff in linux |
| 17:37.09 | mvanbaak | there isn't any |
| 17:37.25 | MrWup | im regretting asterisk and linux already |
| 17:37.29 | mvanbaak | you have to download the kernel sources and read the 10 line documentation/bonding.txt |
| 17:37.30 | MrWup | ive been awake for 2 days now |
| 17:37.39 | mvanbaak | on OpenBSD it's as easy as: man trunk |
| 17:37.41 | mvanbaak | tadaaaaaaaaa |
| 17:37.44 | MrWup | on stupid things like asterisk not converting a damed wav |
| 17:37.44 | mvanbaak | everything is there |
| 17:37.50 | robin_sz | sigh ... same problem as before |
| 17:37.53 | errr | MrWup: yeah to bad they didnt do it on windows so it would require you to have 4 or 5 servers to make it work |
| 17:38.05 | MrWup | tbh id pay for that |
| 17:38.11 | MrWup | right now |
| 17:38.18 | errr | MrWup: call intertel |
| 17:38.28 | errr | they will rape you for a telephone system |
| 17:38.36 | errr | then it will be on windows too |
| 17:38.36 | mvanbaak | MrWup: call cisco |
| 17:38.41 | mvanbaak | they have callmanager |
| 17:38.46 | mvanbaak | run it on windows |
| 17:38.48 | mvanbaak | there you go |
| 17:38.57 | MrWup | cisco he |
| 17:38.58 | MrWup | h |
| 17:39.01 | MrWup | id need to rob a bank |
| 17:39.27 | robin_sz | so |
| 17:39.34 | errr | and why should it be the job of asterisk to be a file converter? |
| 17:39.57 | MrWup | errr - it isnt. its the job of lame. but asterisk wont fucking execute it properly |
| 17:40.01 | MrWup | i wont do anything |
| 17:40.03 | MrWup | it |
| 17:40.15 | mvanbaak | create a cronjob that does that stuff for you |
| 17:40.15 | robin_sz | yes it will |
| 17:40.18 | mvanbaak | that's what I do |
| 17:40.50 | robin_sz | its probably a permissions thing |
| 17:40.51 | errr | MrWup: use the full path to lame, remember 'lame' isnt going to be in asterisks PATH |
| 17:41.00 | mvanbaak | let asterisk mixmonitor to an nfs share |
| 17:41.05 | MrWup | i did use the full path |
| 17:41.10 | mvanbaak | the cron box will convert and archive it |
| 17:41.35 | mvanbaak | my asterisk box is an asterisk box |
| 17:41.40 | mvanbaak | nothing less, nothing more |
| 17:41.52 | mvanbaak | stuff like lame etc are not to be on the asterisk box |
| 17:42.11 | MrWup | mvanbaak, could asterisk mixmonitor to an smb share? |
| 17:42.20 | robin_sz | yes |
| 17:42.31 | robin_sz | smbmount and all that |
| 17:42.37 | robin_sz | but, nfs would be better |
| 17:42.40 | MrWup | mm |
| 17:43.24 | mvanbaak | sure MrWup |
| 17:43.47 | mvanbaak | MrWup: that's the beauty of real operating systems |
| 17:43.57 | mvanbaak | they dont care where the directory comes from |
| 17:44.03 | mvanbaak | as long as it's reachable |
| 17:44.10 | mvanbaak | lol |
| 17:44.36 | mvanbaak | I remember a coworker of mine going totally out of his mind because windows could not log to a remote location |
| 17:44.57 | mvanbaak | we had to setup this win2k server for a customer |
| 17:45.02 | robin_sz | sigh ... |
| 17:45.07 | mvanbaak | in the end we ditched it and went with OpenBSD |
| 17:45.08 | mvanbaak | ;) |
| 17:45.11 | robin_sz | so i still have the same problem as before ... |
| 17:45.24 | robin_sz | * wont make outgoing calls on my euroisdn zap |
| 17:45.36 | mvanbaak | robin_sz: pastebin please |
| 17:45.46 | robin_sz | mvanbaak, OK, |
| 17:45.50 | robin_sz | wait(1) |
| 17:46.02 | mvanbaak | while (!pastebin) |
| 17:46.06 | mvanbaak | <PROTECTED> |
| 17:46.11 | DarkRift | better |
| 17:46.44 | DarkRift | Anyone here knows a bit of asterisk-gui, they are all sleeping on the other chan |
| 17:47.29 | mvanbaak | not me sorry |
| 17:47.44 | DarkRift | Not that bad I'll survive :P |
| 17:48.22 | robin_sz | mvanbaak, see pm |
| 17:49.03 | mvanbaak | going there |
| 17:49.17 | robin_sz | 2 > Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] |
| 17:49.24 | robin_sz | is where it goes tits up |
| 17:49.31 | robin_sz | thats a PRI debug |
| 17:49.56 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:50.30 | robin_sz | tzafrir_home, I got it to build from source, latest bri-stuff |
| 17:50.52 | tzafrir_home | and it works well? |
| 17:50.57 | robin_sz | but still cant dial out ... must be a config thing |
| 17:51.10 | mvanbaak | it is |
| 17:51.32 | robin_sz | mvanbaak, shoot |
| 17:51.52 | mvanbaak | robin_sz: dont know if it's the same there. but in nl I have to fiddle around with the first 0 on every install |
| 17:51.58 | mvanbaak | sometimes it's needed |
| 17:52.03 | mvanbaak | sometimes it's not |
| 17:52.13 | robin_sz | remove the leading 0 |
| 17:52.15 | robin_sz | ? |
| 17:52.16 | mvanbaak | sometimes I have to dial to the full international number |
| 17:52.22 | robin_sz | tried that |
| 17:52.25 | robin_sz | 0044 etc |
| 17:52.25 | mvanbaak | here in .nl it's fucked up |
| 17:52.45 | mvanbaak | every line is different |
| 17:52.57 | mvanbaak | no matter what's on the tech papers supplied by the telco |
| 17:53.12 | robin_sz | it worked fine with mISDN for dialout ... using those digits on that line |
| 17:53.13 | mvanbaak | robin_sz: what's your pridialplan |
| 17:53.23 | robin_sz | umm ... 'local' ? |
| 17:54.02 | mvanbaak | try 'unkown' |
| 17:54.14 | mvanbaak | that's what I always go with |
| 17:54.21 | mvanbaak | because of the same trouble |
| 17:54.22 | mvanbaak | :) |
| 17:55.19 | robin_sz | woo hoo! |
| 17:55.31 | robin_sz | well, figuretively |
| 17:55.52 | robin_sz | break out the drinks! |
| 17:56.10 | mvanbaak | :) |
| 17:56.19 | mvanbaak | lets make caipirinhas ! |
| 17:56.34 | mvanbaak | loads of them ! |
| 17:57.05 | mvanbaak | http://en.wikipedia.org/wiki/Caipirinha |
| 17:57.06 | *** join/#asterisk grndslm (n=grndslm@24-119-81-201.cpe.cableone.net) |
| 17:58.06 | d4rkst4r75 | hi to all! |
| 17:58.08 | d4rkst4r75 | bye to all |
| 17:58.28 | robin_sz | this is great :) |
| 17:58.33 | robin_sz | no echo on calls |
| 17:58.41 | robin_sz | DTMF recognised from landlines |
| 17:59.51 | robin_sz | phew ! |
| 18:01.20 | mvanbaak | there you go |
| 18:01.22 | asdx | Caipirinha, my father used to prepare/drink that lol |
| 18:02.20 | robin_sz | just to add all the cdr_mysql stuff now and we're done |
| 18:03.12 | mvanbaak | asdx: I love it |
| 18:04.39 | asdx | mvanbaak: are you brazilian? |
| 18:04.42 | asdx | mvanbaak: I should try it |
| 18:05.22 | *** join/#asterisk Fieldy (i=LgRhGy5T@gentoo/contributor/Fieldy) |
| 18:05.33 | mvanbaak | I'm from The Netherlands |
| 18:05.40 | mvanbaak | far far away from Brazillia |
| 18:05.43 | asdx | mvanbaak: cool |
| 18:06.24 | asdx | my father loved it, i wish i could drink with him now but he is dead :/ |
| 18:06.38 | *** join/#asterisk skyhawker (n=info@a62-216-22-13.adsl.cistron.nl) |
| 18:06.49 | mvanbaak | that sux :( |
| 18:07.12 | asdx | yeah but maybe he is drinking caipirinhas in heaven so is alright ;) |
| 18:07.51 | mvanbaak | well, when ur here in .nl gimme a call on [(IAX2)|(SIP)]://guest@lunteren.vanbaak.info and I'll get the stuff and make you some |
| 18:08.03 | *** join/#asterisk kn0x (n=atlantic@c-67-176-194-29.hsd1.il.comcast.net) |
| 18:11.03 | skyhawker | uhhh getting sick of DTMF and different countries |
| 18:16.17 | kn0x | does anyone know anything about LCR on (open)SER |
| 18:16.40 | robin_sz | OK, so * from the cli starts fine .. but from /etc/init.d/asterisk start .. it doesnt and leaves no footprints in the logs either ... |
| 18:16.43 | kn0x | also interested about (open)SER RADIUS support |
| 18:17.55 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:18.06 | mvanbaak | robin_sz: where did you get this /etc/init.d/asterisk script ? |
| 18:18.20 | robin_sz | wait, found it |
| 18:18.28 | robin_sz | permisions on /dev/zap/channel |
| 18:18.29 | mvanbaak | did you copy it from bristuff*/asterisk/contrib/init.d/rc.debian.asterisk ? |
| 18:18.32 | mvanbaak | ah |
| 18:18.34 | mvanbaak | yeah |
| 18:18.40 | DocHolliday | any wholesale providers in here? |
| 18:18.49 | robin_sz | of? |
| 18:19.05 | DocHolliday | origination / termination |
| 18:19.10 | robin_sz | oh, that. |
| 18:20.26 | DocHolliday | i take that as a no? :) |
| 18:21.06 | robin_sz | mvanbaak, the last thing seems to be zaptel inititilaisation ... it doesnt seem to load the zaphfc module on its own, it does load ztdummy, and then things go messy |
| 18:23.00 | mvanbaak | what i always do is add some stuff in /etc/init.d/asterisk |
| 18:23.11 | mvanbaak | that will modprobe and ztcfg -vvv |
| 18:23.21 | mvanbaak | evil, but it works |
| 18:23.56 | mvanbaak | cant be bothered to fix my modprobe or udev shit sorted |
| 18:24.42 | robin_sz | in theory zaptel shousl start before asterisk tries |
| 18:26.39 | mvanbaak | uhhuh |
| 18:26.42 | mvanbaak | I'm off |
| 18:26.43 | mvanbaak | bbl |
| 18:27.36 | robin_sz | I wonder if /dev/zap/channel will be chown asterisk:asterisk next time the box boots |
| 18:29.32 | *** part/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 18:31.43 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 18:33.28 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 18:35.24 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:39.42 | mosty | i'm trying to figure how to to safely put asterisk in parallel with a fax machine. is asterisk/zaptel able to detect different rings on an fxo port, or do i need to get something like this: http://www.mbit.com.au/product_info.php?cPath=24&products_id=127 ? |
| 18:45.00 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:45.07 | notnyt | so any idea why the cdr/callerid only gets updated with the agent name now? it used to get updated with agent #, and src channel as the agent |
| 18:49.34 | *** join/#asterisk sevard (i=chuck-th@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 18:53.04 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
| 18:55.27 | robin_sz | so, blf ... and snoms |
| 18:57.37 | *** join/#asterisk [SEP]B4t1m4m (i=Sblerght@200.181.9.45) |
| 18:57.51 | [SEP]B4t1m4m | hello |
| 18:57.54 | [SEP]B4t1m4m | please |
| 18:57.58 | [SEP]B4t1m4m | i need some help |
| 19:00.30 | [SEP]B4t1m4m | algueim vivo? |
| 19:01.46 | mosty | ask your question already |
| 19:02.01 | [SEP]B4t1m4m | <[SEP]B4t1m4m> i've downloaded trixbox 2.2 |
| 19:02.02 | [SEP]B4t1m4m | <[SEP]B4t1m4m> and installed it here in my pc |
| 19:02.02 | [SEP]B4t1m4m | <[SEP]B4t1m4m> i want to connect asterisk with my analog pbx |
| 19:02.02 | [SEP]B4t1m4m | <[SEP]B4t1m4m> than i read some txts |
| 19:02.02 | [SEP]B4t1m4m | <[SEP]B4t1m4m> to use winmodem |
| 19:02.02 | [SEP]B4t1m4m | <[SEP]B4t1m4m> i've sm56 here |
| 19:02.04 | [SEP]B4t1m4m | <[SEP]B4t1m4m> but tribox don't recognize this modem |
| 19:02.06 | [SEP]B4t1m4m | <[SEP]B4t1m4m> how i get that? |
| 19:02.08 | [SEP]B4t1m4m | <[SEP]B4t1m4m> algum brasileiro aih? |
| 19:02.34 | mosty | see the topic, we don't support trixbox here |
| 19:02.38 | [SEP]B4t1m4m | i know |
| 19:02.48 | [SEP]B4t1m4m | but i'm in that channel |
| 19:03.05 | [SEP]B4t1m4m | but nobody answer me |
| 19:03.54 | mosty | you'd be better off using a digium tdm400p or similar card |
| 19:04.10 | [SEP]B4t1m4m | but that too $$$$ |
| 19:04.43 | [SEP]B4t1m4m | i've found another txt that say to use x100p card |
| 19:04.48 | [SEP]B4t1m4m | thats $30 |
| 19:04.55 | [SEP]B4t1m4m | but i live in brazil |
| 19:05.10 | mosty | x100p's are crappy, as are winmodems |
| 19:05.24 | mosty | tdm400p's aren't even that good |
| 19:05.34 | [SEP]B4t1m4m | hmmm |
| 19:05.45 | mosty | but personally i would never recommend anything less than a tdm400p |
| 19:05.52 | [SEP]B4t1m4m | hehehehe |
| 19:06.00 | skyhawker | does anybody know if it possible to split the LCD on a telephone to top line company name and bottom line telephone number ? through the asterisk box |
| 19:06.06 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 19:09.18 | skyhawker | we using the thomson 2030 telephones |
| 19:12.34 | robin_sz | so ... BLF .. and Snoms |
| 19:13.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 19:14.51 | robin_sz | i have something like exten => 5006,1,Macro(stdexten,5001,SIP/office) |
| 19:14.51 | robin_sz | exten => 5006,hint,SIP/office |
| 19:15.04 | robin_sz | thats sorta right? |
| 19:15.37 | [TK]D-Fender | robin_sz, pretty much *'s side of things, yes |
| 19:16.00 | robin_sz | and so long as I have doen that for each extension, on * thats all I need |
| 19:16.42 | robin_sz | I want the usual "lamp lights up when person X is on the phone" thing |
| 19:16.43 | [TK]D-Fender | robin_sz, hints should be accessable to the context the phones are in. |
| 19:16.55 | robin_sz | mm, check. |
| 19:17.04 | [TK]D-Fender | robin_sz, or in a seperate context specified by "subscribecontext=" in the phone's entry |
| 19:17.33 | robin_sz | mmm .. it used to complian about "missing hint for blah" .. now it doesnt, I guess it can see them |
| 19:17.58 | *** join/#asterisk ploieel (n=ploieel@Fb3c5.f.ppp-pool.de) |
| 19:18.41 | robin_sz | mmm .. have it got it the right way round, with the same Sip/office and 5006 on both lines there? |
| 19:19.21 | mosty | robin_sz, you need hints in your dialplan |
| 19:19.25 | [TK]D-Fender | robin_sz, Yes, your phone should be now able to track the status of SIP/office via the hint on 5006 |
| 19:19.50 | robin_sz | [TK]D-Fender, ok, so then its just down to subscribes from the phoens, right? |
| 19:19.53 | [TK]D-Fender | mosty, You are going into backwards mode again, stop confusing him :) |
| 19:20.01 | [TK]D-Fender | robin_sz, Correct |
| 19:20.12 | mosty | sorry, was not paying enough attention |
| 19:22.03 | robin_sz | the options I have for fn keys are: line, destination, intercom park orbit, voice recorder, DTMF and shared line. |
| 19:22.10 | robin_sz | Snom 190's |
| 19:22.44 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@145-27.mc.cite.net) |
| 19:22.49 | _VoiceMeUp_COM | http://www.packet8.net/equipment/residential/tango.aspx |
| 19:22.56 | _VoiceMeUp_COM | anyone know where to get this w/o packet 8 |
| 19:23.19 | robin_sz | china? |
| 19:23.28 | _VoiceMeUp_COM | yeh lol |
| 19:24.44 | _VoiceMeUp_COM | xept , cant find manuf |
| 19:24.53 | robin_sz | if you get one, take it apart to find the real model number |
| 19:25.55 | mosty | robin_sz, i don't know about the 190 model, but maybe it's similar to the 300/320/360: http://www.snom.com/wiki/index.php/Snom320/Web_Interface/Function_Keys#Extension_Monitoring_.26_Call_Pickup |
| 19:28.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:28.23 | robin_sz | yeah ... if only |
| 19:33.30 | mvanbaak | robin_sz: http://www.voip-info.org/wiki/view/Asterisk+phone+snom |
| 19:33.37 | mvanbaak | latero |
| 19:35.04 | _VoiceMeUp_COM | dones one know of Linksys RT31P2 with latest F/W does QOS ? |
| 19:36.29 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 19:36.37 | *** join/#asterisk ploieel (n=ploieel@Fb3c5.f.ppp-pool.de) |
| 19:40.01 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 19:54.32 | *** join/#asterisk wvroger (n=wvroger@216.115.192.22) |
| 19:56.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:01.03 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 20:06.53 | robin_sz | OK that almost works right ... I can see when a phone a extension I have subscribed to is ringing (steady light) ... but when the monitored extension dials out, I get no indiocation that they are engaged |
| 20:07.42 | LeddyHM | anyone recommend a pay as you go sip provider that has no waiting period to use? |
| 20:09.42 | cspot | LeddyHM: vitelity, but you have to start the account with $35 |
| 20:10.02 | robin_sz | UK based? |
| 20:10.10 | LeddyHM | someone mentioned them, they are local to us |
| 20:10.22 | LeddyHM | US preferable |
| 20:10.57 | LeddyHM | I signed up Friday with NuFone, and have yet to make a successful call with them using sip |
| 20:11.00 | cspot | LeddyHM: i'm using vitelity with g729..with nice results |
| 20:11.36 | LeddyHM | $35 is a bit much though, I just need to test a few things |
| 20:12.29 | LeddyHM | I was going to try voipjet, but they require verification |
| 20:12.35 | cspot | LeddyHM: the $35 isn't a startup fee, but an amount they wish you to deposit.....reundbale...FWIW |
| 20:12.54 | cspot | reundbale=refundable |
| 20:12.56 | LeddyHM | I don't want to wait for that process, especially if they aren't open on weekends |
| 20:13.07 | LeddyHM | refundable... hmm |
| 20:13.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:13.59 | LeddyHM | let me read through their page again |
| 20:14.18 | _VoiceMeUp_COM | no a la carte ? |
| 20:14.37 | _VoiceMeUp_COM | i can activate you an account for your test if needed |
| 20:14.38 | _VoiceMeUp_COM | lol |
| 20:15.01 | _VoiceMeUp_COM | leddy we are open 24/7/365 except on groundhog day |
| 20:15.11 | LeddyHM | that's be sweet |
| 20:15.13 | _VoiceMeUp_COM | you need a did too ? |
| 20:15.18 | LeddyHM | I just need to test an 877 number |
| 20:15.19 | _VoiceMeUp_COM | or simply term |
| 20:15.22 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:15.41 | _VoiceMeUp_COM | k then go on to my nick and ill walk you trough |
| 20:15.53 | LeddyHM | _VoiceMeUp_COM: pm me, or vice versa |
| 20:15.58 | LeddyHM | nm |
| 20:16.07 | LeddyHM | pm sent |
| 20:16.07 | _VoiceMeUp_COM | k |
| 20:18.59 | cspot | LeddyHM: if calling 877, did you try ENUM? |
| 20:21.05 | hacim | anyone have any leads on DID origination in brasil? |
| 20:21.36 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:22.51 | *** join/#asterisk af_ (n=getsmart@81-174-8-57.f5.ngi.it) |
| 20:28.24 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 20:28.31 | Defraz | Hello, I am having a heck of a time with my audiocodes gateway. Is anyone using one of them? |
| 20:34.03 | joaovianna | hacim, still there ? |
| 20:34.32 | hacim | joaovianna: i am |
| 20:35.02 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 20:35.05 | crochat | Hello |
| 20:35.27 | *** join/#asterisk zcw100 (n=zcw100@sallywhitley.com) |
| 20:36.12 | crochat | I've tried to compare my ${CALLERIDNUM} and the ${EXTEN} in my local context, so when I call from the 24 to the 24, I will to automatically go in VoiceMailMain(24) |
| 20:36.35 | crochat | But the comparison doesn't seem to work fine |
| 20:37.18 | crochat | With a GotoIf in a macro, it seems that ${CALLERIDNUM} is not the same as ${EXTEN} |
| 20:38.45 | Corydon76-home | Why would it be? |
| 20:38.54 | *** join/#asterisk JuRaDo (n=IceChat7@201-1-109-2.dsl.telesp.net.br) |
| 20:38.57 | Corydon76-home | In a macro, your extension is 's' |
| 20:39.41 | crochat | Sorry for disturbing you... it was just a f*** "$" that I missed before {ARG1} in my macro... |
| 20:39.56 | crochat | It seems that I should go to sleep for a while ;-) |
| 20:40.23 | _VoiceMeUp_COM | foudn your prob ? |
| 20:40.38 | crochat | _VoiceMeUp_COM: Yeah, it works fine now ! |
| 20:40.45 | crochat | Thanks ;-) |
| 20:41.37 | _VoiceMeUp_COM | no prob then |
| 20:42.58 | crochat | _VoiceMeUp_COM: No, no problem other than me being too sleepy :-) |
| 20:43.11 | _VoiceMeUp_COM | i feel you |
| 20:43.18 | *** part/#asterisk wvroger (n=wvroger@216.115.192.22) |
| 20:43.20 | _VoiceMeUp_COM | doing 14 hours for last 6 month here ;) |
| 20:43.56 | robin_sz | part timer |
| 20:44.20 | crochat | _VoiceMeUp_COM: It's 10:45 PM here :-) |
| 20:44.35 | robin_sz | _VoiceMeUp_COM, that is acceptable in 2 situations: |
| 20:44.52 | robin_sz | _VoiceMeUp_COM, 1) you are a contractor on $stupidMoney |
| 20:45.00 | robin_sz | _VoiceMeUp_COM, 2) you own it |
| 20:45.01 | _VoiceMeUp_COM | ? lol |
| 20:45.07 | _VoiceMeUp_COM | #2 could be a winner |
| 20:45.14 | _VoiceMeUp_COM | i own shares |
| 20:45.29 | _VoiceMeUp_COM | but im used to work 14 to 18 hours a day 7/7 |
| 20:45.35 | robin_sz | I do similar hours in my business |
| 20:45.37 | _VoiceMeUp_COM | workoholic |
| 20:45.43 | robin_sz | I own shares too ... |
| 20:45.46 | robin_sz | 100% :) |
| 20:45.53 | *** join/#asterisk ksharpe (n=test@port0059-abg-s-adsl.cwjamaica.com) |
| 20:46.05 | ksharpe | hi all |
| 20:46.16 | crochat | _VoiceMeUp_COM: Me too, 14 to 18 hours a day |
| 20:46.29 | ksharpe | Does anyone have a guide I could follow on how to install libss7? |
| 20:49.59 | ksharpe | anyone have any success with chan_ss7? |
| 20:50.00 | *** join/#asterisk menfin (n=cray@AMontpellier-152-1-64-88.w83-197.abo.wanadoo.fr) |
| 20:53.05 | *** join/#asterisk Jukei76 (n=glippoli@85-18-136-100.fastres.net) |
| 20:53.15 | Jukei76 | hi all |
| 20:53.20 | _VoiceMeUp_COM | hey |
| 20:53.30 | Jukei76 | is there someone can help me?? |
| 20:53.34 | _VoiceMeUp_COM | ask away |
| 20:53.35 | _VoiceMeUp_COM | ;) |
| 20:53.39 | Jukei76 | thanks... |
| 20:54.04 | *** join/#asterisk Caplain (n=shayne@adsl-75-46-168-105.dsl.sfldmi.sbcglobal.net) |
| 20:54.08 | Jukei76 | I would like to know only how much bandwidth usually use a VoIP client?? |
| 20:54.29 | _VoiceMeUp_COM | ~google voip calculator |
| 20:54.41 | Jukei76 | thank you! ;-) |
| 20:54.50 | _VoiceMeUp_COM | http://www.erlang.com/calculator/ |
| 20:54.52 | JuRaDo | voip calculator |
| 20:54.59 | JuRaDo | ops, wrong window |
| 20:56.17 | JuRaDo | gsm codec bandwidth |
| 20:56.25 | *** part/#asterisk JuRaDo (n=IceChat7@201-1-109-2.dsl.telesp.net.br) |
| 20:57.02 | Jukei76 | I found what I was looking for!!! thank you very much! |
| 20:57.08 | Jukei76 | see you! ;-) |
| 21:04.32 | ksharpe | I need help with setting up libss7 of chanss7. can anyone help me/ |
| 21:04.33 | ksharpe | ? |
| 21:05.44 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 21:05.45 | Corydon76-home | Given how few of us have encountered SS7 links, probably not. |
| 21:07.54 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 21:09.05 | *** part/#asterisk zcw100 (n=zcw100@sallywhitley.com) |
| 21:14.02 | robin_sz | whenever anyone says erlang I have the urge to start looking for uniselectors ans open frame relays |
| 21:14.12 | _VoiceMeUp_COM | lol |
| 21:14.22 | _VoiceMeUp_COM | how about morse |
| 21:14.52 | robin_sz | -.-. --.- |
| 21:15.32 | _VoiceMeUp_COM | ... --- ... |
| 21:15.33 | _VoiceMeUp_COM | ;) |
| 21:15.47 | robin_sz | I used to manage 8wpm on a straight key |
| 21:16.01 | robin_sz | closer to 20 on an iambic paddle |
| 21:16.20 | robin_sz | doubt I could do either these days |
| 21:16.38 | _VoiceMeUp_COM | .-. --- -... .. -. / --..-- / .- .-. . / -.-- --- ..- / --. . - - .. -. --. / - .... .. ... / - . -..- - / ..--.. |
| 21:16.39 | *** join/#asterisk Stridernzl (n=neville@125-239-190-165.jetstream.xtra.co.nz) |
| 21:16.40 | _VoiceMeUp_COM | actually |
| 21:16.41 | _VoiceMeUp_COM | ;) |
| 21:16.44 | _VoiceMeUp_COM | deciper this |
| 21:18.48 | robin_sz | .-.. --- ..- -.. .- -. -.. -.-. .-.. . .- .-. |
| 21:19.12 | _VoiceMeUp_COM | lound and clear |
| 21:19.15 | _VoiceMeUp_COM | loud |
| 21:19.21 | robin_sz | spo :) |
| 21:19.24 | robin_sz | spot on |
| 21:19.27 | _VoiceMeUp_COM | heheh back to basics.. |
| 21:19.34 | _VoiceMeUp_COM | lets make a morse module for asterisk |
| 21:19.38 | _VoiceMeUp_COM | hammering the pound key |
| 21:19.42 | robin_sz | better idea ... |
| 21:19.49 | _VoiceMeUp_COM | alisson can repeat text |
| 21:19.50 | robin_sz | lets not |
| 21:19.51 | robin_sz | :) |
| 21:19.53 | _VoiceMeUp_COM | lol |
| 21:21.21 | robin_sz | ive rarley used it ... a few times on really weak contacts .. this was at least 20 years ago |
| 21:25.14 | *** join/#asterisk jdrake (n=jdrake@CPE00045afdd0e8-CM0012254195d6.cpe.net.cable.rogers.com) |
| 21:26.15 | *** join/#asterisk VioByte (n=ircuser@cpe-69-133-27-203.cinci.res.rr.com) |
| 21:27.08 | robin_sz | or as an alternative ... pound away on alisson ans she squeals in morse ...?? you must admit, it has possibilities |
| 21:27.50 | _VoiceMeUp_COM | lol |
| 21:28.17 | _VoiceMeUp_COM | trasnport on morse codec could have less compression probs |
| 21:28.18 | _VoiceMeUp_COM | ;) |
| 21:28.20 | _VoiceMeUp_COM | if at all |
| 21:28.23 | jdrake | I have been trying to figure out how voip works exactly when you connect it upto the regular phone network. I am investigating how expensive it would be to setup my own home phone over the internet. Can anyone tell me if Asterisk or other software would work for this, and what else would be required? |
| 21:28.50 | _VoiceMeUp_COM | get a A200 i guess or a cheap clone x100p and have fun |
| 21:28.53 | LeddyHM | _VoiceMeUp_COM: do you charge for toll free calls? |
| 21:28.58 | _VoiceMeUp_COM | 1 penny |
| 21:29.07 | LeddyHM | per call or minute? |
| 21:29.09 | _VoiceMeUp_COM | call |
| 21:29.10 | _VoiceMeUp_COM | lol |
| 21:29.26 | _VoiceMeUp_COM | basicaly to dether call center on these lines |
| 21:30.24 | LeddyHM | mmkay |
| 21:30.27 | robin_sz | jdrake, if you want to use the internet to route your outbound calls, then you will make savings. you just need a phone (either a plastic thing you can touch with an ethernet connection, or a bit pf software and the mic/speakers) ... away you go, dial out and have fun, you dont need asterisk to do this |
| 21:31.10 | LeddyHM | maybe we'll just send our "problem children" your way |
| 21:31.14 | jdrake | Now how about dialin? I have read about 'DID' for north america. |
| 21:31.26 | LeddyHM | Direct Inward Dial |
| 21:31.31 | robin_sz | want to keep your existing number? |
| 21:31.31 | VioByte | who wants to help me figure out why AGI goto does not GoTO? |
| 21:31.44 | jdrake | robin_sz, It is preferable, but not required. |
| 21:32.06 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 21:32.06 | robin_sz | well, if not, then most service providers will allocate you a new one |
| 21:32.15 | robin_sz | again asterisk not required to do this |
| 21:32.22 | jdrake | brb |
| 21:32.28 | robin_sz | if you want to keep your old one, then .. |
| 21:32.29 | robin_sz | err |
| 21:32.31 | robin_sz | hes gone. |
| 21:32.42 | _VoiceMeUp_COM | ? |
| 21:32.43 | _VoiceMeUp_COM | sorry |
| 21:33.07 | jdrake | Sorry, dinner has come :p, must eat it or parents get upset (Iam visiting them now). |
| 21:33.27 | robin_sz | off you go then ,.. see _VoiceMeUp_COM later for an account |
| 21:34.21 | robin_sz | I heard of a sweet deal this week |
| 21:34.35 | robin_sz | guy I know in some rented office space |
| 21:34.47 | robin_sz | basically, you can have two deals ... |
| 21:35.00 | robin_sz | cheap calls, expensive rent |
| 21:35.01 | robin_sz | or |
| 21:35.02 | *** join/#asterisk saftsack (n=saftsack@pD9E079E4.dip.t-dialin.net) |
| 21:35.08 | robin_sz | cheap rent ... expensive calls |
| 21:35.20 | robin_sz | so, he takes the cheap rent .. expensive calls |
| 21:35.33 | bkruse_home | i want ice cream |
| 21:35.44 | robin_sz | then just uses a VOIP provider .. bypasses their switchboard and smiles :) |
| 21:37.34 | robin_sz | right ... BLF *nearly* works |
| 21:38.04 | robin_sz | when I dial to an extension, I can see it light up, and stays lit up when the callee picks up |
| 21:38.21 | robin_sz | if that extn makes an outgoing call .. .nada on the subscribed phone |
| 21:38.52 | robin_sz | I have notifyringing=yes ... do I need much else? |
| 21:39.02 | robin_sz | notifybusy? or some such? |
| 21:44.48 | VioByte | ok here it goes. |
| 21:45.11 | *** join/#asterisk Fieldy (i=keineV9v@gentoo/contributor/Fieldy) |
| 21:46.35 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:46.38 | VioByte | i've design a php daemon AGI script thats called via AGI() for a context and when the callee has picked up the phone it plays the bacgrounds fine. |
| 21:47.08 | VioByte | but when i send via the AGI to EXEC GoTO with args. |
| 21:47.11 | VioByte | it won't do it |
| 21:47.15 | VioByte | it just gives me |
| 21:47.25 | VioByte | <PROTECTED> |
| 21:47.25 | VioByte | <PROTECTED> |
| 21:47.25 | VioByte | <PROTECTED> |
| 21:47.25 | VioByte | <PROTECTED> |
| 21:47.32 | VioByte | but yet |
| 21:47.36 | VioByte | thecontext registers |
| 21:47.45 | VioByte | <PROTECTED> |
| 21:47.45 | VioByte | <PROTECTED> |
| 21:47.45 | VioByte | <PROTECTED> |
| 21:47.45 | VioByte | <PROTECTED> |
| 21:48.08 | VioByte | i'm not sure what i'm doing wrong of if goto's to another context does not work with agi or what. |
| 21:48.16 | VioByte | of - or* |
| 21:49.11 | bkruse_home | omg |
| 21:49.12 | bkruse_home | ~pb |
| 21:49.13 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 21:50.21 | VioByte | yea sorry about that. |
| 21:51.06 | bkruse_home | VioByte: that error is because your either not returning an ok before sending data, or your sending it before |
| 21:51.43 | VioByte | so i guess noone has a clue to what the prob is with a AGI daemon's exec of goto? |
| 21:52.04 | bkruse_home | ^^ |
| 21:52.16 | *** join/#asterisk flashnet (i=flashnet@gateway/tor/x-8cb909ac6f2990ef) |
| 21:55.24 | VioByte | Spawn extension (queue-collector, s, 0) <--- hmm i think i found it. |
| 21:55.55 | VioByte | forgot to svn ci |
| 21:56.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:56.34 | VioByte | so that good changes could take effect :) |
| 21:58.26 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 21:59.28 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 21:59.45 | Defraz | I am fighinting this AudioCodes gateway has any had any luck working with one of these? |
| 21:59.53 | Defraz | It is the fx0 modle. |
| 22:00.59 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:02.56 | *** join/#asterisk Hymie (i=hymie@l8r.net) |
| 22:04.51 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 22:06.02 | *** join/#asterisk burt75 (n=burt@189.157.128.236) |
| 22:06.07 | burt75 | hello Guys |
| 22:06.15 | burt75 | anyone using MIAX? |
| 22:06.21 | burt75 | need help urgent |
| 22:06.37 | burt75 | payed support |
| 22:08.04 | VioByte | burt75, have you looked at http://www.voip-info.org/wiki-Mobile+IAX |
| 22:08.40 | burt75 | there is not documentation and I am newbie :S |
| 22:09.05 | VioByte | well i've never played with it so i can't help you |
| 22:09.19 | burt75 | :( thank you |
| 22:10.01 | Hymie | hmm |
| 22:10.15 | Hymie | it seems that global variables are not working correctly for me with asterisk 1.4.3 and 1.4.4 |
| 22:11.35 | Hymie | exten => s,2,Set(GLOBAL(CELLLOOP=${EPOCH}) |
| 22:11.35 | Hymie | exten => s,3,noop(Cell no answer on. ${EPOCH} ${CELLLOOP}) |
| 22:11.45 | Hymie | <PROTECTED> |
| 22:11.50 | Hymie | so, it's not setting it... |
| 22:12.01 | Hymie | I saw docs saying I _may_ reference variables directly as global variables |
| 22:12.09 | Hymie | but not that I have to reference global variables as globals |
| 22:12.16 | Hymie | when reading inside a dialplan |
| 22:12.21 | Hymie | anyone know anything aobut this? |
| 22:13.33 | wolferine | i am writing a php script to (simple) call my phone and another, what command must I use to establish a two ended conversation? |
| 22:13.59 | [TK]D-Fender | Hymie, exten => s,2,Set(GLOBAL(CELLLOOP=${EPOCH}) <- bad |
| 22:14.17 | [TK]D-Fender | Hymie, exten => s,2,Set(GLOBAL(CELLLOOP)=${EPOCH}) <- good |
| 22:14.48 | Hymie | arg! |
| 22:14.55 | Hymie | ARG |
| 22:15.24 | *** join/#asterisk kiscokid (n=Ron@adsl-66-122-34-11.dsl.sntc01.pacbell.net) |
| 22:16.08 | [TK]D-Fender | EXFOLIATE!!!!!! |
| 22:17.05 | *** join/#asterisk jdrake (n=jdrake@CPE00045afdd0e8-CM0012254195d6.cpe.net.cable.rogers.com) |
| 22:17.29 | Hymie | hgeh |
| 22:17.33 | jdrake | I am back, apologies for the run out. |
| 22:18.54 | jdrake | What does a service provider do exactly to provide the link to the telephone network to VOIP? Is it something that one can do personally? |
| 22:20.18 | JT | jdrake: i assume you mean an ITSP |
| 22:20.29 | JT | in which case they usually have a number of PRIs that connect to the PSTN |
| 22:20.56 | jdrake | What is the relationship with that and DID? |
| 22:21.29 | JT | a did is just an incoming number |
| 22:21.45 | JT | it will come in over pri usually |
| 22:21.52 | JT | or channelised T1/E1 + SS7 |
| 22:21.54 | kiscokid | Anyone know about using SIP phone with IAX trunks? |
| 22:22.01 | jdrake | What links a DID with say my IP phone? |
| 22:22.27 | [TK]D-Fender | kiscokid, those 2 things do not belong in the same sentence typically |
| 22:22.28 | JT | jdrake: when a call comes in for that number, they send it to your ip phone account |
| 22:22.38 | [TK]D-Fender | kiscokid, a SIP phone speaks SIP. Period. |
| 22:22.49 | Hymie | [TK]D-Fender: hmm, guess I'll ask this too. ;) For some reason, whne I dial a SIP call plus a ZAP call at the same time, Asterisk now turns off my "hold music while ringing" when the SIP call starts ringing... |
| 22:22.58 | *** join/#asterisk Math` (n=seb@modemcable037.229-56-74.mc.videotron.ca) |
| 22:23.06 | Hymie | [TK]D-Fender: worked with 1.2.... there've been no changes |
| 22:23.06 | [TK]D-Fender | kiscokid, you can use * with a SIP phone and an IAX provider with * performing the translation naturally. |
| 22:23.11 | kiscokid | Fender, so if I have a Cisco SIP phone it cannot use an IAX trunk to make a call? |
| 22:23.15 | wolferine | i am writing a php script to (simple) call my phone and another, what command must I use to establish a two ended conversation? |
| 22:23.29 | jdrake | I should probably ask a more direct question: I am currently paying $25 (including taxes) for my land line. I am looking into things that may be cheaper. |
| 22:23.31 | ManxPower | kiscokid: asterisk will convert the call from SIP to/from IAX2 |
| 22:23.33 | JT | kiscokid: it needs to connect to asterisk with sip first |
| 22:23.34 | bkruse_home | wolferine: manager? |
| 22:23.34 | Math` | wolferine: you need to originate the call |
| 22:23.36 | bkruse_home | action: originate |
| 22:23.43 | [TK]D-Fender | kiscokid, the PHONE will not speak IAX, but you can have your cisco go THROUGH * and use that provider |
| 22:23.46 | bkruse_home | applcation: dial(sip/whatever) |
| 22:23.53 | Hymie | [TK]D-Fender: I noticed a bug in the tracker where someone was reporting that music on hold was staying on.. and noew I notice asterisk turning music on hold off all over the place |
| 22:23.54 | kiscokid | ok, got it thanks |
| 22:23.57 | wolferine | Math`, so on the same socket, I make both connections? |
| 22:24.05 | Hymie | [TK]D-Fender: as if to work around the bug |
| 22:24.12 | wolferine | yes bkruse_home using the manager |
| 22:24.17 | Math` | wolferine: using AMI yeah |
| 22:24.22 | Math` | read on the originate command |
| 22:24.25 | wolferine | AMI? |
| 22:24.30 | bkruse_home | wolferine: i just told ya |
| 22:24.30 | Math` | asterisk manager interface |
| 22:24.33 | wolferine | k, thanks Math |
| 22:24.38 | JT | jdrake: if you can get broadband without phone line rental, then yes, i guess you can |
| 22:24.43 | Math` | any way to make _NXXNXXXXXX match after lets say 2125125019 so calls dont go out with a provider and go in with another one? :) |
| 22:24.52 | wolferine | ah, right, thanks you two |
| 22:24.59 | kiscokid | Fender so with a SIP phone using IAX is RTP involved at all in the conversation? |
| 22:25.05 | jdrake | JT, A few providers support a dry loop setup that costs $8 for the dry loop + whatever the internet service is. |
| 22:25.05 | Math` | kinda breaks the point of including from-pstn if the pattern is used first |
| 22:25.31 | JT | kiscokid: of course, if you want to hear anything |
| 22:25.34 | ManxPower | Math`: I don't see the problem. create another pattern |
| 22:25.42 | Math` | kiscokid: SIP Phone <--- SIP+RTP --->Asterisk<--- IAX2--->whatever |
| 22:25.48 | Hymie | [TK]D-Fender: yeah, look here if you're curious -> http://bugs.digium.com/view.php?id=9052 |
| 22:25.58 | JT | jdrake: i'd never get rid of a landline myself, but up to you i guess |
| 22:26.17 | Hymie | [TK]D-Fender: so, now I see Asterisk forcing music on hold off, when sip calls start ringing (in or out).. and it's messing up my music on hold while ringing |
| 22:26.29 | jdrake | JT, I am personally just interested in lowering my phone bill, and I have it bare now. |
| 22:26.30 | kiscokid | so as long as my SIP phone and Asterisk are behind the same firewall I won't have NAT problems with RTP? |
| 22:26.34 | Hymie | [TK]D-Fender: guess I'll bug report it, just wondered if you'd heard any moo on it |
| 22:27.08 | jdrake | It does appear as though I can't find a DID in the city of Sarnia - where I am. |
| 22:27.10 | JT | jdrake: you can't get a cheaper monthly service with more expensive calls (do most calls over voip) |
| 22:27.13 | JT | ? |
| 22:27.40 | JT | kiscokid: it can work through NAT too |
| 22:27.42 | [TK]D-Fender | jdrake, not the biggest city... quite understandable |
| 22:28.18 | wolferine | what is the difference between a channel, and an exten ? |
| 22:28.19 | Math` | ManxPower: still goes outbound even if I make a "pattern" out of the number (_number)... note that Im including from-pstn from within outgoing (which has the generic patterns) |
| 22:28.27 | [TK]D-Fender | kiscokid, if they are local there is no reason to have any kind of challenge. |
| 22:28.38 | Math` | (AEL2/1.4-svn) |
| 22:28.52 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:29.12 | kiscokid | ok, thanks |
| 22:29.19 | *** join/#asterisk wacker (n=wacker@wb2flw.octothorp.org) |
| 22:30.06 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 22:30.07 | kink0 | hi |
| 22:30.52 | wolferine | one is the "out going call" and the other is the local extension by which the call is placed? |
| 22:31.46 | Math` | wolferine: you need to call a channel... |
| 22:32.01 | Math` | either SIP/phone1, IAX2/phone2... or even Local/number@outgoing |
| 22:32.17 | [TK]D-Fender | ManxPower, Hey, great timing seeing you active jsut now... I tried out that version of SpanDSP & apps you referred me to and it still craps out on me. Here's what happens at verbose 10 : http://www.pastebin.ca/474599 |
| 22:32.21 | Math` | wolferine: (outgoing being your context for outbound calls) |
| 22:32.25 | *** join/#asterisk desktoppie (n=desktopp@82-204-21-111.dsl.bbeyond.nl) |
| 22:33.48 | kink0 | I have a question, is possible to pass ( to zap ) over 60 calls from the same peer ? |
| 22:34.01 | kink0 | I am unable to do that, appears like limited to 60 |
| 22:34.47 | [TK]D-Fender | kink0, If you have bandwidth & zap channels to do it I don't see why not. Keep in mind the CPU load of transcoding where applicable |
| 22:35.53 | kink0 | [TK]D-Fender, my issue, is that I am unnable to pass over 60 from the same peer , but I can pass 50 from one peer, and 60 from other peer , without problem. |
| 22:36.39 | kink0 | the problem is I got zap cc 47, but I suspect is a fake error, may be due to timeout in zap, when sip maximum calls is reached |
| 22:36.57 | [TK]D-Fender | kink0, It is possible your PEER is limiting you? |
| 22:37.26 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 22:37.49 | kink0 | no, he sends lets say 80 calls, where I only pass 60 |
| 22:38.24 | kink0 | every time one call over 60 enters, I get 47 and returns SIP 503 |
| 22:38.26 | JT | kink0: you also seem to have difficulty remembering if the problem is with asterisk or callweaver? |
| 22:38.34 | kink0 | JT: both |
| 22:38.49 | kink0 | I try both, and is common to both , asterisk and cw |
| 22:39.11 | JT | it must be a problem with your setup |
| 22:39.40 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 22:39.45 | kink0 | JT: may be... but i tryed differents variations, in zap and in sip, and the same. |
| 22:40.14 | [TK]D-Fender | kink0, does * accept call 61 and only bomb trying to dial out Zap? |
| 22:40.37 | kink0 | I have not more ideas to try it, even I seted a username+call-limit into sip, even I found no any predefined value for peer->call_limit in the sources (may be in other part ) |
| 22:41.16 | kink0 | [TK]D-Fender, yes, I think so, because Asterisk returns sip 503 cause. |
| 22:41.50 | kink0 | while i watch hangupcause, and I see 47 when that happens |
| 22:41.52 | [TK]D-Fender | kink0, I mean if you were to answer the call and do a playback before dialing out Zap, would the call work? |
| 22:42.20 | kink0 | i did no try that |
| 22:42.34 | kink0 | you means to really issolate the problem if sip or zap, right ? |
| 22:42.44 | [TK]D-Fender | well 47 = Resource unavailable, unspecified |
| 22:43.10 | kink0 | yes, but that only happens when same peer overpass 60 calls, |
| 22:43.10 | [TK]D-Fender | kink0, have you reviewed your zap setup and independantly tested each channel manually? |
| 22:43.23 | *** part/#asterisk wacker (n=wacker@wb2flw.octothorp.org) |
| 22:43.25 | kink0 | when is happening, other peer can pass other near 60 calls here |
| 22:44.02 | kink0 | [TK]D-Fender, yes, and I tryed g, G , r ... and so in extensions.conf, to give enough time to reset the zap channel. |
| 22:44.11 | kink0 | that happens no in one defined zap channel |
| 22:44.25 | kink0 | lets say for peer 1, all channels fails ( over 60 calls ) |
| 22:44.39 | kink0 | for peer 2, any channel failled for peer 1, will be ok. |
| 22:44.43 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 22:45.27 | kink0 | I compared results, about what span was ussing every call, and the fail is not dependant of span or channel, that occurs only when the same sip peer goes over 60 simultaneus calls |
| 22:46.19 | kink0 | I also did separate groups, so I tryed Dial(Zap/g1 ... Dial(Zap/g2 ... and so |
| 22:46.38 | kink0 | but that did not changes the results, exactly the same. |
| 22:51.43 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 22:52.01 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:52.37 | Defraz | Anyone out there played with an AudioCodes MP-118 fx0 gateway. I can't seem to get it to recognize a hang up. |
| 22:53.21 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 22:53.59 | JT | FXO |
| 22:56.29 | burt75 | anyone have worked with MIAX? |
| 22:57.35 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 22:57.45 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:58.54 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 23:00.27 | jdrake | Has anyone heard of a setup where you become an extension on a 1-800 style line so it doesn't matter where you are? |
| 23:01.11 | Math` | jdrake: well you can get a 1-800 number and bind it to your asterisk... |
| 23:01.18 | jm|laptop | jdrake: are you talking enum? |
| 23:01.56 | jm|laptop | or you could bind the 1-800 to your PSTN/T1/T3 trunk and have it call extensions on your local * lan |
| 23:02.18 | jdrake | jm|laptop, I am afraid that I cannot say that I am saying specific stuff as I am unaware of them. |
| 23:02.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:02.54 | jm|laptop | jdrake: well am I right in thinking that 1-800 'route' to a regular number at the phone company? |
| 23:03.45 | jdrake | More specifically like a provider that provides the 1-800 number, you are an extension off of that line, and then it would route to an application or phone on your network. |
| 23:04.15 | jm|laptop | ah |
| 23:04.20 | jm|laptop | that's subtly different |
| 23:05.15 | *** join/#asterisk zogulus (n=zogulus@58.98.adsl.brightview.com) |
| 23:05.25 | jm|laptop | jdrake: as far as I understand - such 'proxies' often give you SIP/IAX2 forwarding |
| 23:06.02 | jdrake | So that would then work straight off of RJ45 with a SIP phone? |
| 23:06.12 | jm|laptop | so that when call comes into 1-800-you-number it generates a SIP/IAX2 call to your asterisk server [perhaps via an intermediary] |
| 23:06.31 | jm|laptop | jdrake: well that sort of depends, too :S |
| 23:06.53 | jm|laptop | jdrake: if you have an account on the host of the 1-800 number then you would register your phones with their SIP pbx |
| 23:07.24 | jdrake | I am thinking they call 1-800-some-number then dial my extension of 1234, and then it would then do SIP to my network. |
| 23:07.31 | jm|laptop | if, however, they just do termination for the PSTN line and forward the call to you via SIP, you need something like asterisk at your end to recieve the SIP call |
| 23:07.39 | *** join/#asterisk kn0x (n=atlantic@c-67-176-194-29.hsd1.il.comcast.net) |
| 23:08.17 | jm|laptop | jdrake: more likely your remote provider/proxy would have a forward from your SIP proxy e.g. 1-800-you-number --> IAX2/1234@your-asterisk-box-ip |
| 23:08.22 | jm|laptop | you see? |
| 23:08.51 | jm|laptop | s/SIP/IAX2 or the other way around :S |
| 23:09.17 | jdrake | Now before we go on, is it possible that this could be cheaper than a regular phone line? |
| 23:09.30 | jm|laptop | um ... |
| 23:09.43 | jm|laptop | yes |
| 23:09.44 | jm|laptop | feasibly. |
| 23:10.06 | jdrake | Now who could provide this sort of service? |
| 23:10.08 | jm|laptop | 1-800 numbers mean you take the cost, right? If your proxy/provider has a bulk call rate that they pass onto you, you might get a flat rate |
| 23:10.41 | jm|laptop | in the same way that if you go for call termination you can sometimes get a preferencial rate |
| 23:11.00 | jdrake | I highly doubt I could take the cost of a 1-800 number, which is why I mentioned an extension off of it. Because I am really a home phone user :p |
| 23:11.18 | jm|laptop | {sip phone} --- { your asterisk } ---- { their asterisk / sip proxy } ---- { pstn } ---- ( cheap callplan ) ---- { remote phone } |
| 23:11.54 | jm|laptop | jdrake: so what did you have in mind? |
| 23:12.40 | jdrake | Any reasonable way to lower the cost of a phone at home. I am paying $25 / month for a bare bones phone and think it is a little higher than what I would like to do. |
| 23:13.10 | jdrake | I have just so little information to be able to say much. |
| 23:13.11 | jm|laptop | jdrake: well many companies offer porting of a standard number |
| 23:13.22 | jm|laptop | it costs *you* nothing to receive a call, after all |
| 23:13.28 | jm|laptop | where are you looking to reduce you call charges? |
| 23:13.47 | jdrake | Monthly fee for the phone line itself. |
| 23:13.54 | jm|laptop | hmm |
| 23:13.56 | jm|laptop | tricky |
| 23:14.21 | jdrake | A dry loop here costs $8, internet over that can be as cheap as $30. |
| 23:14.47 | jdrake | Getting Bell Canada to admit to having them might be another thing. |
| 23:15.20 | jm|laptop | jdrake: it does depend on your domestic tariff ... |
| 23:15.33 | jm|laptop | jdrake: if you'll pardon the currency I can give you an example |
| 23:15.38 | jdrake | But there are other providers around here that do provide the service. |
| 23:15.55 | jdrake | I respect all currencies when they are not american dollars :p |
| 23:16.10 | jm|laptop | jdrake: in the UK a standard tariff might be thus: std daytime national 4p/min std evening national 1p/min international 11p/min |
| 23:16.36 | jm|laptop | jdrake: there are some termination companies who will say "okay: all national calls whenever 1.5p/min" |
| 23:16.47 | jm|laptop | because they have bought bulk carriage |
| 23:17.01 | jdrake | Looks expensive, I am presuming that 1p ~= 0.02 CDN$ |
| 23:17.12 | jm|laptop | so you route all your calls to the PSTN via this proxy and you get charged 1.5p/min whatever time of the day it is for national calls |
| 23:17.37 | jm|laptop | jdrake: if GPB->CND == GBP->USD atm |
| 23:17.43 | jm|laptop | CDN* |
| 23:17.45 | Defraz | How does a POTS line signal an hang up? |
| 23:17.57 | jm|laptop | Defraz: there is more than one way |
| 23:18.09 | jm|laptop | Defraz: sometimes it is voltage drop sometimes it is a proper signal |
| 23:18.13 | JT | Defraz: which direction, hanging up end, or other end? |
| 23:18.14 | jm|laptop | Defraz: sometimes it's just silence |
| 23:18.30 | Defraz | I call into my POTS line from my cell phone. |
| 23:18.36 | Defraz | and I hang up my cell phone |
| 23:18.47 | jm|laptop | jdrake: the fact is that you will probably be able to find a cheaper rate by going via a SIP proxy |
| 23:19.03 | JT | Defraz: the normal answer is there is NO signalling method at the far end |
| 23:19.03 | jm|laptop | jdrake: but having said that - have you investiaged other carriers on your PSTN |
| 23:19.07 | JT | except you hear silence |
| 23:19.22 | JT | Defraz: you can get the telco to setup polarity reverse on hangup though |
| 23:19.31 | jm|laptop | ^^^ |
| 23:19.48 | Defraz | OH yea does that cost more though? |
| 23:19.58 | JT | depends on telco |
| 23:20.02 | jdrake | jm|laptop, I have tried, but the options are limited. They don't quite want to get involved in cheap phone service. |
| 23:20.22 | Defraz | oh will they usually tell you to take a hight |
| 23:20.52 | jm|laptop | jdrake: which one? |
| 23:21.04 | jdrake | Rogers is an example |
| 23:21.18 | jm|laptop | you're being all foreign on me again :/ |
| 23:21.54 | Defraz | No if my FX0 gateway has a silence detection is that detecting on my pots side maybe coming from my Asterisk box or silence coming from the POTS? |
| 23:21.59 | Defraz | No - Now |
| 23:22.02 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-92.plaicmtc01.tx.dh.suddenlink.net) |
| 23:22.41 | JT | Defraz: it's FXO not FX0 |
| 23:22.50 | jm|laptop | :) |
| 23:23.03 | jdrake | Rogers starts at $30, then you need to add 14% tax |
| 23:23.13 | jm|laptop | BRB! |
| 23:23.26 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:23.32 | JT | Defraz: it's detecting for whatever way it's sending RTP traffic |
| 23:23.41 | jdrake | Has anyone used vonage in Canada? |
| 23:23.43 | Defraz | Yes FXO |
| 23:24.32 | jm|laptop | jdrake: not to over-simplify: *any* company you can find that will route your PSTN traffic will have a bulk call plan from which they can pass savings on to you |
| 23:25.18 | jdrake | I am not quite skilled in the 'finding one' area |
| 23:25.45 | jm|laptop | well my knowledge of Canadian proxies is limited too :( |
| 23:25.52 | jm|laptop | I could help you in the UK ;) |
| 23:26.09 | jdrake | heh |
| 23:26.12 | jm|laptop | so I need to speed up my RAID5 rebuild :/ |
| 23:26.32 | jdrake | I wish I had a raid5, I only have a raid1 on my home computer. |
| 23:26.49 | jm|laptop | <PROTECTED> |
| 23:26.50 | jm|laptop | <PROTECTED> |
| 23:26.50 | jm|laptop | <PROTECTED> |
| 23:26.54 | JT | RAID1 is superior to RAID5 for redundancy |
| 23:27.08 | jm|laptop | software raid :( |
| 23:27.17 | jdrake | Vonage seems to have $19.99 rates, so that is lowering the cost by about $3 or $4 |
| 23:27.21 | jm|laptop | full hw raid5 on my fileserver :) |
| 23:27.39 | jm|laptop | ar0: ATA RAID5 stripesize=128 subdisks: ad4 ad6 ad8 ad10 status: READY |
| 23:28.21 | jm|laptop | but cobbled-together sw RAID-5 on here - using an Adaptec 1200A dual channel card plus hd0 |
| 23:28.38 | *** join/#asterisk Blackthorn (n=support@w-l4.smyth.net) |
| 23:28.39 | jm|laptop | however: seems to be working so far *boggle* |
| 23:29.18 | *** join/#asterisk Curus (n=Curus@10.8.185.213.dk-amb.res.sta.perspektivbredband.net) |
| 23:29.22 | jdrake | Windows tells me that the usb drive is *in use* but won't tell me what is using it! |
| 23:29.48 | jm|laptop | jdrake: start > run > cmd > ps waux ..... no wait |
| 23:29.50 | jdrake | jm|laptop, So you are using linux on your home machine and bsd on another? |
| 23:29.56 | jm|laptop | jdrake: start > run > cmd > lsof ..... no wait |
| 23:29.58 | Blackthorn | Hello, I have a remote asterisk server with pri service with dedicated numbers. And I just setup replacement server at another location. Going to be about 30-60 days before I can get service to my new asterisk box. Is there away so I can register all my phones to the new server, and have the old * server forward all of the dedicated incoming calls? |
| 23:29.58 | Defraz | do most phone companies do polarity reversal? |
| 23:30.31 | jm|laptop | jdrake: I would have BSD all round if 1) my laptop wasn't so fucking stupid 2) FreeBSD had native Flash9 support |
| 23:30.48 | jdrake | flash9 |
| 23:30.50 | jdrake | ? |
| 23:31.01 | jm|laptop | jdrake: the HP6325 is very poor laptop |
| 23:31.19 | jm|laptop | I had a perfectly good IBM R50 which I managed to pour wine all over :/ |
| 23:31.29 | jm|laptop | the insurers sent me the HP as a 'replacement' |
| 23:31.38 | jdrake | eek |
| 23:31.42 | jm|laptop | replacement maybe; match: far far from it |
| 23:31.57 | jdrake | I am glad I use ubuntu, it is very worthy to me |
| 23:31.59 | jm|laptop | so yeah; the acpi bios is SO BROKEN |
| 23:32.12 | jdrake | I think I must restart because something thinks it is using the drive |
| 23:32.23 | jm|laptop | anyway: having use Debian on the laptop for a while I got used to be able to access flash9 sites |
| 23:32.38 | jm|laptop | so it crept onto my workstation too: but it will NEVER reach my server |
| 23:32.41 | jm|laptop | servers* |
| 23:32.46 | jm|laptop | (I hope) |
| 23:32.53 | jdrake | I suppose I can understand that. |
| 23:33.03 | jm|laptop | jdrake: you are running Asterisk in Windows? |
| 23:33.08 | jdrake | Well, I will be right back... |
| 23:35.22 | *** join/#asterisk jdrake (n=jdrake@CPE00045afdd0e8-CM0012254195d6.cpe.net.cable.rogers.com) |
| 23:36.10 | jdrake | One day there will be a linux desktop on my parent's desktop. |
| 23:36.34 | jdrake | 22MB download just for a wireless driver! |
| 23:37.26 | Blackthorn | ahh i found a doc that tells me how to do it. i'll go try that. :) |
| 23:39.36 | *** join/#asterisk thoughtpolice (n=austin@c75-111-139-92.plaicmtc01.tx.dh.suddenlink.net) |
| 23:41.35 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 23:46.48 | kink0 | anyone knows why some channels are called SIP/172.10.1.1-b51d4818 while other calls from the same peer is called SIP/5060-b323d34 ? |
| 23:47.15 | kink0 | some channels names included the IP, others the PORT, in the same section |
| 23:47.43 | kink0 | when all calls comes from the same peer and host |
| 23:47.50 | kink0 | has any explication ? |
| 23:51.17 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 23:51.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 23:52.43 | JT | kink0: pastebin sip.conf after changing the sip secrets, and also pastebin extensions.conf |
| 23:52.51 | JT | ~pb |
| 23:52.52 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:54.28 | kink0 | ok |
| 23:58.43 | Caplain | can i use a modem as a channel? |
| 23:58.57 | JT | no |
| 23:59.01 | Caplain | :( |
| 23:59.42 | Math` | you used to be able to.... but it was so buggy it was removed |
