| 00:03.25 | *** join/#asterisk techie (n=gus@antibala.net) |
| 00:07.16 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
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| 00:26.25 | *** join/#asterisk NOT_guru (n=chatzill@209.145.181.55) |
| 00:27.19 | NOT_guru | hello I have gotten a 4 port digium card and have never setup an FXS port before |
| 00:27.54 | NOT_guru | should I be alarmed when I run genzaptelconf if dmesg echos "Module 0: FAILED FXS (FCC)" |
| 00:28.18 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:28.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:28.18 | NOT_guru | or is that saying it sees it but I need to conf it correctly |
| 00:28.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:29.22 | NOT_guru | "wctdm: probe of 0000:01:01.0 failed with error -5" |
| 00:29.32 | NOT_guru | think this is a dead FXS module? |
| 00:30.16 | mutilator | .. |
| 00:30.21 | mutilator | anyone wanna but a lightly used te110p? |
| 00:31.08 | *** join/#asterisk purserj (n=purserj@k-sit.com) |
| 00:34.03 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:35.06 | NOT_guru | anyone with some digium experiance here that could help me? |
| 00:43.40 | [TK]D-Fender | NOT_guru, pastebin "ztcfg -vvvv" , zaptel.conf, zapata.conf and "dmesg" |
| 00:44.00 | NOT_guru | Fender thank you... gathering now |
| 00:48.06 | NOT_guru | Fender... Thank you again for looking |
| 00:48.08 | NOT_guru | http://pastebin.ca/557539 |
| 00:51.54 | [TK]D-Fender | Module 0: FAILED FXS (FCC) |
| 00:51.58 | [TK]D-Fender | Yeah, not good... |
| 00:52.15 | [TK]D-Fender | NOT_guru, I'd suggest you try the module on port 4 of the card just to test. |
| 00:52.39 | NOT_guru | wonderfull ... I will power down now and move it |
| 00:52.48 | NOT_guru | thank you for the suggestion I will brb |
| 00:53.03 | [TK]D-Fender | k |
| 01:01.24 | NOT_guru | same result |
| 01:01.32 | NOT_guru | guess I need to try the -DNO_CALIBRATION in Makefile |
| 01:04.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:04.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:07.46 | [TK]D-Fender | could jsut be dead... |
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| 01:11.27 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:12.14 | NOT_guru | thats what I am hoping against |
| 01:12.25 | NOT_guru | just how to determine if its the card or daughter board |
| 01:15.40 | *** join/#asterisk casimir (n=casimir@25-125.34-65.tampabay.res.rr.com) |
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| 02:08.45 | Tomasu | How can I reduce the amount of silence I hear before an outgoing call starts ringing? Even when I press # to make it skip the second or so wait, it still has a few seconds of silence (up to 4-6) |
| 02:23.46 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 02:28.06 | Corydon76-home | Tomasu: get a PRI circuit from your carrier |
| 02:28.39 | Corydon76-home | Digital circuits tend to be faster than analog |
| 02:32.49 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:36.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
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| 02:49.03 | *** part/#asterisk tictoc (i=tabac@gateway/tor/x-804ad25994a35ee2) |
| 02:59.37 | iBuMp | g729a doesnt cure my bandwidth problems! arggg |
| 02:59.53 | Tomasu | Corydon76-home: thanks, but its not analog or a pri style setup. pure voip |
| 03:01.53 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 03:02.04 | Corydon76-home | Tomasu: then you need to talk to your provider about the time |
| 03:02.24 | Tomasu | ah |
| 03:03.22 | Tomasu | is there a way to enable "fake" ringing while its conncting? |
| 03:03.37 | Corydon76-home | Yes, you can use the 'r' option to Dial |
| 03:03.57 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.18 | iBuMp | anyone here can help me with trimming down my bandwidth use. |
| 03:04.53 | Corydon76-home | Tomasu: it may be indicative that your provider doesn't actually have any lines, but merely routes your call to other voip providers |
| 03:05.14 | Corydon76-home | Tomasu: in which case, you might want to find a more direct route and cut out the middle man |
| 03:05.24 | Tomasu | I _think_ they have lines. |
| 03:05.56 | Corydon76-home | 6 seconds is insanely long to connect digital lines |
| 03:06.24 | Corydon76-home | If they really have lines, they have major infrastructure issues |
| 03:07.06 | Tomasu | ok, it seems more like 3-4 atm. 6 when asterisk was reconnecting... |
| 03:07.39 | Tomasu | at least incomming |
| 03:08.20 | Tomasu | ah, since I'm calling my cell, I think its 2-3 seconds voip, and 2-3 seconds cell... |
| 03:08.31 | Tomasu | stupid cell networks taking too long to connect |
| 03:09.57 | Corydon76-home | voip doesn't take that long |
| 03:10.24 | Tomasu | I'll contact my provider see if this is "normal" for them. |
| 03:11.09 | Corydon76-home | voip connection time is, at most, the time it takes for two round trips between you and your provider |
| 03:11.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:11.50 | Tomasu | I'm not certian if its maybe my asterisk config being funky |
| 03:12.04 | Corydon76-home | Doubtful |
| 03:13.43 | Corydon76-home | but you can run a 'sip debug'. As soon as you get back a 100 level response from your carrier, voip has connected |
| 03:14.05 | Corydon76-home | That's usually within a split second. |
| 03:14.40 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 03:15.18 | *** join/#asterisk luckyone (n=jordan@CPE-65-28-7-102.kc.res.rr.com) |
| 03:16.00 | luckyone | can anyone help me with this error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" when trying to connect to a running * process? |
| 03:16.16 | *** join/#asterisk iBuMp (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 03:16.18 | Tomasu | with that, I get a response from my provider within a split second. it takes a moment for it to start ringing my cell phone though. |
| 03:17.00 | Corydon76-home | There you go. The problem is either at your carrier or between your carrier and your cell carrier |
| 03:17.01 | Tomasu | I have a feeling most of the time is related to the cell network my cell is on. |
| 03:17.16 | Tomasu | I recall having issues with my cell a while back. |
| 03:17.38 | Tomasu | last time I tested asterisk half the time my cell would say my local number (assigned to asterisk) was longdistance. |
| 03:17.40 | Tomasu | fun stuff. |
| 03:17.41 | Corydon76-home | in any case, I think that resolves the issue as not being in Asterisk |
| 03:17.45 | Tomasu | thanks :) |
| 03:17.48 | Tomasu | yup |
| 03:19.31 | iBuMp | ok is it me or does diaplan save screw up your extensions.conf file with a bunch of )&(*&()*&(*& |
| 03:19.56 | luckyone | I am trying to connect to * with this: sudo asterisk -rv -U asterisk -G asterisk and it is failing... it used to work, what am I doing wrong? |
| 03:20.22 | Corydon76-home | luckyone: you shouldn't be trying to set the user with a remote connection |
| 03:20.30 | Corydon76-home | luckyone: just 'asterisk -Rc' |
| 03:21.41 | jql | yeah, I have call connect times all over the map when it comes to voip providers |
| 03:22.41 | luckyone | Corydon76-home: same error |
| 03:22.54 | luckyone | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 03:23.40 | Corydon76-home | luckyone: is Asterisk running? |
| 03:23.45 | luckyone | yes |
| 03:24.00 | Corydon76-home | Does /var/run/asterisk exist? |
| 03:24.26 | luckyone | Corydon76-home: no |
| 03:24.36 | luckyone | root 4915 1 0 Jun06 ? 00:00:00 /bin/bash /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 03:24.36 | luckyone | asterisk 4928 4915 0 Jun06 ? 00:00:03 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c |
| 03:24.43 | Corydon76-home | Well, there's your problem |
| 03:25.00 | Corydon76-home | See, the error message told you exactly what you needed to know. |
| 03:25.09 | luckyone | but the server IS up |
| 03:25.35 | Corydon76-home | Yes, but the socket file needs to exist for you to connect to Asterisk remotely |
| 03:25.39 | luckyone | I have two sip phones connected to them |
| 03:25.49 | luckyone | why didn't the socket file start? |
| 03:26.01 | Corydon76-home | Because the directory didn't exist |
| 03:26.25 | Corydon76-home | You should probably ensure that the directory exists in your startup script |
| 03:27.21 | Corydon76-home | or possibly, an adminstrator deleted the socket and the directory, because they didn't know it was needed. |
| 03:28.16 | luckyone | Corydon76-home: hmm, I am the only "admin" |
| 03:28.39 | De_Mon | beware of the grimlins |
| 03:28.43 | luckyone | Corydon76-home: I created the directory, is it normal for people to use safe_asterisk startup script when starting *? |
| 03:28.52 | Corydon76-home | Yes |
| 03:29.37 | Corydon76-home | luckyone: have you rebooted recently? Some system adminstration scripts are known to wipe all files in /var/run on reboot |
| 03:29.54 | luckyone | Corydon76-home: could be, I am going to restart now |
| 03:30.00 | luckyone | Corydon76-home: could be, I am going to restart * now |
| 03:30.08 | iBuMp | when i do dialplan save it add a ton of garble in my extensions lines is this normal |
| 03:30.25 | Corydon76-home | iBuMp: most people don't use that command |
| 03:30.47 | Corydon76-home | iBuMp: in fact, I suspect most people disable it |
| 03:30.59 | iBuMp | DOH! |
| 03:31.06 | iBuMp | are you serious |
| 03:31.24 | Corydon76-home | iBuMp: the only reason for you to use it is if you use the command line to add and remove extensions. |
| 03:31.26 | iBuMp | so something like this is bad right! exten => _011XXXXXX!,1����/i����^A�^N,Macro(trunkdial,${trunk_1}/${EXTEN:0}) |
| 03:31.29 | Corydon76-home | Do you do that? |
| 03:31.37 | Corydon76-home | Yep, looks bad |
| 03:31.49 | Corydon76-home | Report it as a bug, please |
| 03:31.59 | iBuMp | crap now im screwed! |
| 03:32.10 | iBuMp | gotta find my backup file and replace on fly |
| 03:32.28 | Corydon76-home | iBuMp: what version? |
| 03:32.33 | iBuMp | 1.4 |
| 03:32.40 | iBuMp | latest installed this week by digium |
| 03:32.42 | Corydon76-home | Okay, I'll check it |
| 03:32.43 | iBuMp | tech support |
| 03:33.19 | iBuMp | want access to see yo self |
| 03:33.25 | iBuMp | or you want me to post file? |
| 03:34.54 | Corydon76-home | No, it's easily enough replicable |
| 03:35.41 | luckyone | Corydon76-home: ok, I recreated the directory |
| 03:36.29 | luckyone | Corydon76-home: then I tried to stop *, it wouldn't listen, so I killed it, then I started it again. It is running, but /var/run/asterisk/asterisk.ctl still doesn't exist |
| 03:37.42 | Corydon76-home | luckyone: is the directory owned by the 'asterisk' user? |
| 03:38.40 | Corydon76-home | If you're going to change the user, you need to ensure the directory is owned by that user |
| 03:39.48 | luckyone | Corydon76-home: you fixed it |
| 03:40.06 | De_Mon | Corydon76-home rawks |
| 03:43.49 | *** join/#asterisk nervlord (n=peter@titanserver.titan.net.au) |
| 03:44.09 | nervlord | guys, quick question, trying to get real time asterisk working with voicemail... my sql connections show up in my loog file as "connected" |
| 03:44.13 | nervlord | so theres no worries there |
| 03:44.18 | nervlord | but for some reason i cant see them using voicemail show users |
| 03:44.34 | nervlord | under asterisk -r |
| 03:44.45 | nervlord | ive setup my extconfig |
| 03:48.06 | Corydon76-home | nervlord: correct, that command won't show realtime users |
| 03:49.36 | luckyone | how do I now setup * to send caller id information? Or is that something that my provider would have to configure |
| 03:50.03 | Corydon76-home | Asterisk sends cid by default. |
| 03:50.20 | Corydon76-home | Your provider is free to accept or reject it, though |
| 03:50.26 | luckyone | interesting |
| 03:50.38 | luckyone | where would I populate this information? |
| 03:50.48 | luckyone | Perhaps I didn't set it up for each of my clients? |
| 03:50.54 | luckyone | is it a client side config? |
| 03:51.02 | luckyone | or something in a dialplan? |
| 03:51.07 | Corydon76-home | It can be |
| 03:52.03 | *** join/#asterisk bmg505 (n=leon@196.209.182.35) |
| 03:56.49 | nervlord | whats naother good way to test my voicemail is correctly looking in realtime for its info then... |
| 03:59.03 | Corydon76-home | Check voicemail for a user configured with realtime |
| 03:59.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.01 | nervlord | is that the only way really? |
| 04:01.05 | Corydon76-home | Sorry, it was complicated when I wrote the command, so I never bothered with writing it for realtime |
| 04:03.38 | nervlord | hehehee |
| 04:03.38 | nervlord | bugger |
| 04:03.42 | nervlord | thanks anyway.. |
| 04:03.45 | nervlord | i just hope ive done it right |
| 04:03.48 | nervlord | i suspect ive done it wrong |
| 04:03.53 | nervlord | same with sip show peers, i cant see any database |
| 04:05.50 | nervlord | surely its not that hard |
| 04:05.53 | nervlord | surely its a sql statement |
| 04:05.53 | nervlord | oh wait |
| 04:05.54 | nervlord | lol |
| 04:06.00 | nervlord | just cause we write a sql statement doe snot mean asterisk is using it |
| 04:09.43 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:12.00 | *** join/#asterisk Delta_Offset (n=abrahamc@201.226.130.55) |
| 04:12.06 | Delta_Offset | hello |
| 04:12.16 | nervlord | a hoy hoy |
| 04:12.18 | [[blah]asfd | if anyone is interested I have a digium te410p i am selling. I is nearly brand new. I removed it from a system that I turned into a SIP gateway connected to global crossing. was in service for about 3 months before I took it out and placed it into a static bag. digium is asking about $1400 for one brand new. I will give this one up for $700 if anyone wants it. PM me if you are interested. |
| 04:12.31 | Delta_Offset | im having this error here when tring to install zaptel modules |
| 04:12.46 | Delta_Offset | ./lib/modules/2.6.20-1.2952.fc6xen/build: No such file or directory. Stop. |
| 04:13.00 | Delta_Offset | thats when i do make clean |
| 04:13.01 | [[blah]asfd | you dont have the kernel headers installed is my guess |
| 04:13.40 | Delta_Offset | i have 2 hours on google trying to find out how to fix this |
| 04:13.58 | [[blah]asfd | Delta_Offset: what OS? |
| 04:14.16 | Delta_Offset | Fedora Core 6 |
| 04:14.29 | Delta_Offset | Linux ZZ 2.6.20-1.2952.fc6xen #1 SMP Wed May 16 19:19:04 EDT 2007 i686 i686 i386 GNU/Linux |
| 04:14.33 | [[blah]asfd | do a "yum install kernel kernel-devel" as root |
| 04:14.40 | Delta_Offset | i did |
| 04:15.02 | tzafrir_laptop | what's the output of: uname -r |
| 04:15.02 | [[blah]asfd | k- that was my guess... sorry. |
| 04:15.15 | tzafrir_laptop | ah |
| 04:15.20 | Delta_Offset | 2.6.20-1.2952.fc6xen |
| 04:15.20 | [[blah]asfd | did you reboot after the yum install kernel kernel-devel? |
| 04:15.27 | Delta_Offset | yes |
| 04:15.28 | tzafrir_laptop | no need for that |
| 04:15.44 | tzafrir_laptop | rpm -qa | grep kernel |
| 04:17.06 | Delta_Offset | i know but.. just in case |
| 04:17.07 | [[blah]asfd | just checking to make sure you were using the new kernel. i have seen a few people not do that. |
| 04:17.07 | nervlord | modprobe |
| 04:17.07 | nervlord | or |
| 04:17.07 | nervlord | lsmod |
| 04:17.07 | Delta_Offset | tzafrir_laptop done |
| 04:17.07 | nervlord | send us the output |
| 04:17.07 | Delta_Offset | one momennt |
| 04:17.11 | Delta_Offset | http://www.pastebin.ca/557873 |
| 04:17.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:18.22 | Delta_Offset | let me know |
| 04:20.28 | Delta_Offset | <tzafrir_laptop> i did the rpm -qa | grep kernel... then what? |
| 04:20.43 | tzafrir_laptop | what is the output of that? |
| 04:20.52 | tzafrir_laptop | rpm -q is 'query' |
| 04:21.00 | Delta_Offset | kernel-headers-2.6.18-1.2798.fc6 |
| 04:21.00 | Delta_Offset | kernel-devel-2.6.20-1.2952.fc6 |
| 04:21.00 | Delta_Offset | kernel-xen-2.6.20-1.2952.fc6 |
| 04:21.00 | Delta_Offset | kernel-xen-2.6.18-1.2798.fc6 |
| 04:21.11 | Delta_Offset | sorry bout that |
| 04:22.11 | tzafrir_laptop | right. yum-install kernel-xen-devl |
| 04:22.23 | tzafrir_laptop | or a similar name. No need to reboot |
| 04:23.01 | Corydon76-home | iBuMp: Upgrade to 1.4 as of revision 68595. |
| 04:23.11 | iBuMp | whats teh best way |
| 04:23.22 | [[blah]asfd | yeah... sorry, i didnt see the xen in the error... sorry i missed that. |
| 04:23.28 | Corydon76-home | Do a svn checkout of the 1.4 repo |
| 04:24.11 | Delta_Offset | should i do yum install kernel-xen-devel |
| 04:24.12 | Delta_Offset | ? |
| 04:24.23 | *** part/#asterisk nervlord (n=peter@titanserver.titan.net.au) |
| 04:24.23 | Corydon76-home | iBuMp: or wait until 1.4.5 is released |
| 04:24.41 | Corydon76-home | iBuMp: either way, that revision is where the bug was fixed |
| 04:25.27 | [[blah]asfd | BuMp: i singed on after you posted your bug.. .what are you running into? |
| 04:25.34 | [[blah]asfd | just interested. |
| 04:26.24 | iBuMp | i ran dialplan save and it FOOBARED my conf |
| 04:26.32 | [[blah]asfd | oh... |
| 04:26.35 | Delta_Offset | Installed: kernel-xen-devel.i686 0:2.6.20-1.2952.fc6 |
| 04:26.46 | [[blah]asfd | lol., that sucks. did you have a backup copy? |
| 04:26.48 | JT | tzafrir_laptop: if you need me, please message me, i will be generally afk for next few days |
| 04:26.55 | iBuMp | yeap! |
| 04:27.00 | [[blah]asfd | whew! |
| 04:27.08 | iBuMp | its not like my extensions.conf sucks anyway |
| 04:27.17 | Delta_Offset | now what should i do? |
| 04:27.20 | [[blah]asfd | i could see that costing a person his job ;-) |
| 04:27.27 | iBuMp | yes in deed |
| 04:27.28 | Delta_Offset | nice |
| 04:27.38 | iBuMp | but really i wouldnt mind losing this contract right about now |
| 04:27.39 | [[blah]asfd | mine is pretty complex, it would smoke me completely with out a backup |
| 04:27.46 | Delta_Offset | seems to be fixed |
| 04:27.55 | [[blah]asfd | yeah, i have a few like that too. |
| 04:27.59 | Delta_Offset | ohhh i hate this fedora |
| 04:28.19 | iBuMp | hey Corydon76-home |
| 04:28.21 | Corydon76-home | iBuMp: wasn't difficult to fix; there was just an uninitialized buffer |
| 04:28.29 | [[blah]asfd | Delta_Offset: have you considered CentOS 4.4? |
| 04:28.37 | iBuMp | so you said to svn then run make make install right |
| 04:28.39 | Delta_Offset | tzafrir_laptop thanks |
| 04:28.46 | Delta_Offset | well... i have this server in costa rica |
| 04:28.47 | Corydon76-home | iBuMp: yep |
| 04:28.49 | iBuMp | never upgraded in place from tar |
| 04:28.56 | iBuMp | forgive me.. old rpm guy |
| 04:28.59 | tzafrir_laptop | centos 4.5, you mean |
| 04:29.07 | Delta_Offset | and im doing the installs remotely |
| 04:29.28 | [[blah]asfd | theres a 4.5? i have not kept up on it. |
| 04:29.32 | [[blah]asfd | i know it went to 5.0 |
| 04:29.43 | [[blah]asfd | but i ran into some php conflicts there. |
| 04:29.44 | iBuMp | wheres the connect info for svn |
| 04:29.45 | tzafrir_laptop | Fedora is a fast-changing beast while it is maintained. And then it is no-longer maintained and you're on your own |
| 04:29.49 | Delta_Offset | i had centos on another pbx |
| 04:29.58 | Delta_Offset | im running with freepbx |
| 04:30.05 | iBuMp | runnin debian on this one |
| 04:30.10 | iBuMp | rusty told me to do it |
| 04:30.15 | Delta_Offset | and FC6 was the only OS they had there |
| 04:30.17 | tzafrir_laptop | I figure 5.0 woul be nicer at this stage. If only it had the server CD |
| 04:30.38 | tzafrir_laptop | (though I naturally prefer something else) |
| 04:30.44 | [[blah]asfd | so long as you dont need php you should be fine with 5.0 |
| 04:30.47 | [[blah]asfd | that was my only issue |
| 04:30.55 | [[blah]asfd | php:mysql to be exact. |
| 04:31.03 | tzafrir_laptop | [[blah]asfd, what issue with php? |
| 04:31.31 | iBuMp | hey Corydon76-home you said 1.4 repos? what ver again |
| 04:31.38 | [[blah]asfd | just a bunch of bogus errors. i could not get any of my scripts to run. i reverted my OS back and it all worked just fine. |
| 04:31.48 | [[blah]asfd | i did not pursue it much more than that. |
| 04:32.03 | [[blah]asfd | figured I would rather spend more time of my asterisk work |
| 04:32.22 | [[blah]asfd | not broken, dont fix it kind of a thing for me ;-) |
| 04:32.23 | iBuMp | Corydon76-home: use trunk or branh |
| 04:32.43 | Corydon76-home | http://svn.digium.com/svn/asterisk/branches/1.4/ |
| 04:32.52 | iBuMp | mucho gracias senor |
| 04:33.54 | Corydon76-home | de nada |
| 04:33.56 | iBuMp | DOH |
| 04:34.04 | iBuMp | what is the apt-get command to get svn |
| 04:34.04 | iBuMp | hahaha |
| 04:34.13 | iBuMp | damnit.. first debian install |
| 04:34.21 | iBuMp | been a redhat person |
| 04:34.28 | Corydon76-home | apt-get install subversion |
| 04:34.34 | iBuMp | rusty told me it was my redhat causing my original problem |
| 04:35.27 | iBuMp | so i stopped asterisk, svn this file, make && make install, then reboot all is well? |
| 04:36.54 | iBuMp | is the gui good enough to make dialplans yet? |
| 04:39.45 | iBuMp | do i need to upgrade my zaptel driver |
| 04:40.09 | Corydon76-home | nope |
| 04:40.13 | iBuMp | coo |
| 04:40.31 | iBuMp | and ill take the no anwer on gui as a no |
| 04:40.47 | Corydon76-home | My answer is I've never tried it |
| 04:40.54 | iBuMp | k |
| 04:41.07 | iBuMp | im just horrible @ diaplans... never been a programmer |
| 04:41.20 | iBuMp | and i think you need to have the mentality of a coder to do them |
| 04:41.38 | Corydon76-home | I won't disagree, but I'm a coder |
| 04:41.51 | iBuMp | <--carries a handicap plaquer |
| 04:42.06 | iBuMp | <--mentally challenged |
| 04:42.30 | iBuMp | my head too many times. |
| 04:42.53 | Nugget | your head a splode. |
| 04:42.57 | iBuMp | hahaha |
| 04:43.09 | iBuMp | keep it wrapped just to hold in whats left |
| 04:43.56 | iBuMp | so many peops in channel so little said |
| 04:45.33 | iBuMp | so i need a little guidance on my codec |
| 04:45.52 | iBuMp | the g729 doesnt seem to help my bandwidth problem as well as I planned |
| 04:46.26 | iBuMp | or, probably the reality of the situation is I havnet a clue on the best configuration for it all |
| 04:47.19 | iBuMp | is my chan_features.so and codec_g729a.so compat wth this version? |
| 04:48.33 | iBuMp | Generating docs for file app_skel.c... |
| 04:48.42 | iBuMp | .. /usr/src/asterisk/apps/app_skel.c:19 Warning: Unsupported xml/html tag <Your> found |
| 04:48.46 | iBuMp | doh |
| 04:48.48 | iBuMp | wrong windo |
| 04:51.31 | iBuMp | so far about 4 errors in progdocs |
| 04:51.44 | iBuMp | wohhh make it about 15 |
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| 05:44.01 | IOscanner | I am trying to upgrade asterisk a machine I just build it on a few weeks ago. It seems the svn build I used had a bug with no playing the selected voice mail options. Instead it just played the default greeting. |
| 05:44.06 | IOscanner | I am getting error: codec_zap.c:833: warning: unused variable `info' |
| 05:44.07 | IOscanner | make[1]: *** [codec_zap.o] Error 1 |
| 05:44.41 | IOscanner | I tried stable and even the code I used before. Not sure why I can't get it to compile? Any ideas? |
| 05:45.41 | IOscanner | starts erroring with the VAR ZT_TCOP_..... |
| 05:50.42 | IOscanner | I have upgrade zaptel to the latest build. Not sure what else it could be |
| 05:52.34 | IOscanner | I think I found it. damn zaptel never updated due to /usr/share/man/man8 path. You would think that would be last or not required to install the kernel modules. |
| 05:57.40 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 05:58.35 | IOscanner | damn that didn't fix it either. What would cause voicemail not to use the new recorded messages instead of the default message. |
| 06:05.12 | Daejeo1 | I want to buy a polycom IP phone. which one should I buy " Polycom IP PHONE 320 or 301?" |
| 06:05.31 | Daejeo1 | 320 is better? |
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| 06:10.51 | zeeesh | hi |
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| 06:19.52 | tzafrir | IOscanner, what version of Asterisk is it? What version of zaptel? |
| 06:20.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:25.42 | ido | hi tzafrir_laptop :) |
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| 06:33.37 | tzafrir_laptop | hi |
| 06:36.54 | lsodi | hi I'm looking for asterisk manager GUI for voip service providers commercial/GPL, http://www.voip-info.org/wiki-Asterisk+GUI seems to be out of date, didnt find suitable. Any suggestions? |
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| 06:48.23 | angryuser | good day i have some error messages "RTCP SR transmission error, rtcp halted" what is that? |
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| 07:06.56 | zeeesh | <tzafrir>: simply dialing out ... i was using .. WaitExten(15) before sending dtmf "now want to change it by pressing # key " where to set "#" rather than .. digit timeout or specific seconds ... ?? |
| 07:10.28 | tzafrir_laptop | zeeesh, sorry. I lost the context |
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| 07:24.50 | samarora | can anybody guide me to connect * to Database |
| 07:24.52 | samarora | ? |
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| 07:28.51 | angryuser | <samarora> which one? |
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| 08:24.49 | *** join/#asterisk ramindia (n=ramindia@202.63.96.133) |
| 08:25.22 | ramindia | hey can some one tell me asterisk as end point how can i make them trust ? |
| 08:30.03 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 08:31.41 | kakarot | hi people, is there any way to make fxo or fxs with modems ? |
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| 08:32.09 | tzafrir_laptop | kakarot, if you have the exact spec, and time to write a driver |
| 08:32.21 | tzafrir_laptop | and it will be an FXO, not an FXS |
| 08:32.32 | tzafrir_laptop | somewhat like the X100P |
| 08:32.33 | angryuser | <kakarot> yea, hylafax with sipmodem, or asterfax ;) |
| 08:32.44 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 08:33.00 | tzafrir_laptop | iaxmodem? |
| 08:33.20 | tzafrir_laptop | that's a fax. Not an FXO device |
| 08:33.27 | angryuser | i am using mp3 files for * ivr , but sometimes they bug, what format isthe best for quality with asterisk? |
| 08:33.50 | tzafrir_laptop | why ue mp3 and not transcode it to something nicer? |
| 08:34.17 | matsk | The best is to have all the formats that you have allowed codes |
| 08:34.18 | angryuser | <tzafrir_laptop> i saw * user wab 8000 16bit mono;( |
| 08:34.27 | angryuser | *wav |
| 08:35.08 | tzafrir_laptop | right. And as you'll be downsampling it, chances are the file will not be even larger (or much larger) |
| 08:35.17 | angryuser | what format should i use, coz the voice is far to be great, too saturated |
| 08:35.41 | angryuser | i cant rerecord if needed |
| 08:35.57 | angryuser | pff i Can |
| 08:35.58 | tzafrir_laptop | wav is nice, and trivial to transcode to slin, and hence simple to transcode to any other format |
| 08:36.34 | angryuser | i have transcoders, give me the name of format, frequency bitrate whatever |
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| 08:37.23 | angryuser | i use mp3 128, but the sound is bad, too saturated ;) |
| 08:37.36 | angryuser | sound like ****** really |
| 08:37.45 | tzafrir_laptop | again, the 128bits won't help you. asterisk will downsample it |
| 08:37.56 | tzafrir_laptop | so better downsample it yourself to wav |
| 08:38.06 | tzafrir_laptop | and provide it to asterisk as wav |
| 08:38.24 | angryuser | <tzafrir_laptop> so 8000 16 bit mono? |
| 08:38.30 | tzafrir_laptop | yes |
| 08:39.00 | angryuser | <tzafrir_laptop> tested it 5 min ago, it is crappy, worse than downgraded |
| 08:39.43 | ramindia | can some tell me, how can i configure ip authentication in sip.conf ( end point asterisk sending ip only) |
| 08:40.53 | tzafrir_laptop | that's the quality of a phone line. Is the quality any difference than listening to the downsampled file ? |
| 08:41.21 | tzafrir_laptop | if so: do you use native moh? |
| 08:42.41 | angryuser | <tzafrir_laptop> yes the quality is different, voice is more saturated when listening resampled 8000 16bit mono |
| 08:42.58 | angryuser | <tzafrir_laptop> native moh? |
| 08:43.38 | tzafrir_laptop | asterisk playing files from a directory, and not using an external program |
| 08:45.22 | angryuser | <tzafrir_laptop> i have my ivr files under /var/lib/asterisk/sounds/menu , and yes i think i have it |
| 08:46.32 | angryuser | <tzafrir_laptop> i have changed the default dir if onhold music, and changed it to some elevator samples works :) |
| 08:47.02 | angryuser | "RTCP SR transmission error, rtcp halted" , what does it mean? |
| 08:53.02 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:53.29 | angryuser | is is possible to bring down the volume on files played in * ? (ivr) |
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| 09:00.56 | [pyro] | hey guys, i have an aastra 53i, but cant get it to register with my pbx. I cant find any info in the logs (full or others) the phone just says "no service" when it boots up. Any suggestions on where else i can look? |
| 09:04.37 | *** part/#asterisk ramindia (n=ramindia@202.63.96.133) |
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| 09:48.14 | tzafrir_laptop | angryuser, with sox :-( |
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| 09:49.06 | kakarot | i want to build an asterisk pbx, where can i find fxo and fxs cards? i mean cheap cards |
| 09:50.00 | Dovid | trying to build zaptel on a centos 5 box. can anyone tell me what is causing this error? |
| 09:50.00 | Dovid | http://pastebin.ca/558287 |
| 09:50.56 | Dovid | using ztdummy |
| 09:52.32 | tzafrir_laptop | fixed in 1.2.18 |
| 09:52.56 | tzafrir_laptop | alternatively: |
| 09:53.13 | tzafrir_laptop | edit xpp/Makefile and remove the line with DEBUGFS |
| 09:53.39 | tzafrir_laptop | (or remove xpp and all other drivers and build just ztdummy. It will also build faster) |
| 09:53.54 | Dovid | tzafrir: i am installing 1.2.18 - thought that may be the issue |
| 09:54.12 | Dovid | tzafrir: are you at VON now ? |
| 09:55.34 | Dovid | 1.2.18 built like a charm |
| 09:56.07 | tzafrir_laptop | kakarot, can't think of any. cheapest I can think of is an ATA. alternatively, check out: http://www.rowetel.com/blog/?p=26 |
| 09:56.32 | tzafrir_laptop | Dovid, no. |
| 09:57.29 | kakarot | tzafrir_laptop thnaks |
| 09:57.34 | kakarot | thanks |
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| 11:03.21 | *** join/#asterisk ramindia (n=ramindia@202.63.96.133) |
| 11:03.53 | ramindia | hi, how can authenticate with ip address rather from user and password, and bill the same with cdr ? |
| 11:03.57 | ramindia | any one suggest me |
| 11:06.24 | Daejeo1 | ramindia: I am using IP auth. let me see what I did |
| 11:06.34 | Daejeo1 | hold on |
| 11:06.44 | ramindia | Daejeo1: thanks |
| 11:14.44 | ramindia | Daejeo1: what happend.. did u got any configs |
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| 11:20.04 | *** join/#asterisk davidcsi (n=davidcsi@21.Red-83-50-58.dynamicIP.rima-tde.net) |
| 11:20.42 | davidcsi | hello all, i configured 1.2.18 for realtime, but voicemail doesn't work.. is there any special thing i gotta do? |
| 11:21.25 | ramindia | davidcsi: no not much, add the things in extconfig should do the job |
| 11:22.20 | davidcsi | doesn't work... i'm trying to configure it with openser though... |
| 11:22.37 | davidcsi | apparently the engine connects fine and all but... |
| 11:22.43 | ramindia | are u integrating with Openser ? |
| 11:23.50 | sergee | anybody experienced in chan_h323 configuration? |
| 11:24.03 | sergee | i have a strange issue with it, |
| 11:24.39 | sergee | my scheme is: SIP phone -> asterisk -> h323 GW -> my cellphone |
| 11:25.12 | sergee | call setup is successful and 2 phones are connected, but no audio at all (in both directions) |
| 11:25.34 | sergee | although rtp debug on asterisk shows tons of rtp packets, |
| 11:25.54 | sergee | anybody have an idea how to fix it? :) |
| 11:26.06 | *** join/#asterisk davidcsi (n=davidcsi@21.Red-83-50-58.dynamicIP.rima-tde.net) |
| 11:26.13 | davidcsi | sorry, wifi went down.. |
| 11:26.33 | davidcsi | sergee: i wouldn't recommend asterisk/h323 for production. |
| 11:26.58 | davidcsi | ramidia: yes, i'm trying to integrate it with openser |
| 11:28.45 | ramindia | same server ? |
| 11:29.37 | sergee | davidcsi: unfortunately, i don't have a choice |
| 11:29.43 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:30.21 | davidcsi | sergee: i understad, but you'll run into a lot of problems... it eats up the memory |
| 11:30.28 | davidcsi | RAM that is. |
| 11:30.42 | ramindia | davidcsi: are u runninng both openser and asteriks in same server ? |
| 11:30.47 | davidcsi | sergee: you should try some protocol comverter. |
| 11:30.55 | davidcsi | ramidia: no. |
| 11:31.24 | *** join/#asterisk volker__ (i=volker@uhrig.eu.org) |
| 11:31.25 | volker__ | hi |
| 11:31.44 | ramindia | davidcsi: are u using central database for both ? |
| 11:31.53 | davidcsi | ramidia: yes |
| 11:32.07 | ramindia | then should not be problem |
| 11:32.09 | ramindia | what is the problem |
| 11:32.37 | Daejeo1 | <PROTECTED> |
| 11:32.52 | ramindia | Daejeo1: its ok.did u got the config |
| 11:32.53 | davidcsi | asterisk says that it can't find the user... I can't see the realtime doing the query though... |
| 11:33.11 | ramindia | may be if u post here will be problem, can i private message u |
| 11:33.11 | Daejeo1 | do you have a Static IP address? |
| 11:33.30 | ramindia | iam serving for my other department, |
| 11:33.35 | ramindia | both have static IP |
| 11:33.45 | Daejeo1 | good |
| 11:33.47 | Daejeo1 | then |
| 11:33.56 | Daejeo1 | what you need to do |
| 11:34.05 | ramindia | davidcsi: i dont see any problem iam running realtime only, its picup all messages from ser/openser |
| 11:34.23 | volker__ | i have problems to find in google and a book the option how to handle calls depending on the caller ID. someone has a hint? |
| 11:34.55 | Daejeo1 | in your configuration file sip/iax specify host- client ip address |
| 11:35.04 | *** join/#asterisk kakarot (n=kakarot@41.Red-88-15-39.dynamicIP.rima-tde.net) |
| 11:35.04 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 11:35.07 | casix | hello |
| 11:35.15 | ramindia | yes thats what i did, but still guy able to make calls |
| 11:35.20 | *** join/#asterisk Zdrulio (n=sux_@82.119.72.130) |
| 11:35.22 | Zdrulio | hello all |
| 11:35.40 | Zdrulio | i have little questin about asterisk + voip gate. |
| 11:36.20 | casix | I'm making a migration to realtime. Can I use labels in realtime?? |
| 11:36.21 | davidcsi | ramidia: can you send me yor view? you are getting the voicemail users from a view right? |
| 11:36.38 | ramindia | what version of mysql ? |
| 11:36.44 | davidcsi | 5 |
| 11:38.21 | davidcsi | ramindia: and you voicemail.conf please |
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| 11:39.01 | ramindia | i dont have in that |
| 11:39.02 | ramindia | voicemail => mysql,asterisk,voicemail |
| 11:39.38 | davidcsi | ok, you view? |
| 11:42.38 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 11:42.50 | casix | anyone know how can I put labels in asterisk realtime?? |
| 11:43.09 | Dovid | casix: labels = ? |
| 11:44.16 | casix | yes in the priority |
| 11:44.38 | Dovid | what is a label ? |
| 11:44.43 | Dovid | setting a variable ? |
| 11:44.45 | casix | exten => _33XX.,1(label),NoOp() |
| 11:44.48 | Dovid | noop'ing something ? |
| 11:44.59 | Zdrulio | i try to connect gsm gate voiceblue lite to asterisk . but asterisk: Username/auth name mismatch. But pass and user are OK! any ideas ? |
| 11:45.43 | Dovid | Zdrulio: you must have something wrong. Something must be spelled wrong. do a sip debug and see if the user name is coming in right |
| 11:46.00 | Dovid | casix: what are you trying to accomplish ? |
| 11:46.32 | casix | i'm migrating to realtime but i don't know how to use "labels" in realtime |
| 11:46.40 | casix | if it is possible |
| 11:47.17 | Dovid | http://www.voip-info.org/wiki/index.php?page=Asterisk%20priorities |
| 11:47.21 | Dovid | did u try it yet ? |
| 11:48.37 | Dovid | mysql> describe extensions; |
| 11:48.37 | Dovid | +----------+--------------+------+-----+---------+----------------+ |
| 11:48.37 | Dovid | | Field | Type | Null | Key | Default | Extra | |
| 11:48.37 | Dovid | +----------+--------------+------+-----+---------+----------------+ |
| 11:48.37 | Dovid | | id | int(11) | | MUL | NULL | auto_increment | |
| 11:48.37 | Dovid | | context | varchar(32) | | PRI | | | |
| 11:48.37 | Dovid | | exten | varchar(20) | | PRI | | | |
| 11:48.37 | Dovid | | priority | int(11) | | PRI | 0 | | |
| 11:48.38 | Dovid | | app | varchar(20) | | | | | |
| 11:48.39 | Dovid | | appdata | varchar(128) | | | | | |
| 11:48.41 | Dovid | sorry about that |
| 11:49.06 | Dovid | I would chnage prioroty from int(11) to varchar(11) and see if it works |
| 11:49.14 | Dovid | but do it on a live machine !!!!! |
| 11:49.50 | casix | no it don't work |
| 11:50.15 | casix | because when astersik queryes de db it makes a priority="1" |
| 11:50.17 | casix | for example |
| 11:50.54 | Dovid | do u have the query ? |
| 11:52.02 | Dovid | ah ok |
| 11:52.22 | Dovid | i see what ur saying I would swtich the query from: SELECT * FROM extensions WHERE exten = 's' AND context = 'notrio-closed-ivr' AND priority = '3' |
| 11:52.22 | Dovid | to |
| 11:52.35 | Dovid | SELECT * FROM extensions WHERE exten = 's' AND context = 'notrio-closed-ivr' AND priority = '3' or '3(%' |
| 11:52.41 | casix | Jun 11 13:52:24 DEBUG[4114]: res_config_mysql.c:141 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_realtime WHERE exten = '1111' AND context = 'prefix_a2bill' AND priority = '1' |
| 11:52.44 | casix | yes it is |
| 11:53.02 | casix | thx :) |
| 11:53.04 | Dovid | i would email the asterisk users list and see if anyone has any ideas |
| 11:53.14 | Dovid | or u can email the dev list and see if some one can create a patch for u |
| 11:53.22 | casix | ok |
| 11:53.47 | casix | and I have another feature can I explain it to you and ggive me your opinion |
| 11:53.51 | casix | ? |
| 11:53.59 | Dovid | go for it |
| 11:54.18 | casix | in a realtime to put a position field |
| 11:54.41 | casix | it can be used to order each app in a extension |
| 11:54.49 | casix | then you can use n in the priority field |
| 11:55.05 | casix | my english is no so good, sorry |
| 11:55.23 | casix | do you understand what i want to say? |
| 11:56.00 | casix | you thing it's a good idea? or you know if it is hard to implement? |
| 11:58.16 | Dovid | position field = ? (prioroty?) |
| 11:58.16 | casix | not is the priority... is the order |
| 11:58.16 | casix | only |
| 11:58.20 | casix | it is the order asterisk have to read it |
| 11:58.27 | Dovid | i am lost |
| 11:58.30 | casix | like in a text |
| 11:58.30 | Dovid | what do u want to do ? |
| 11:58.39 | casix | to use n in the priority |
| 11:58.39 | *** join/#asterisk Falcon-x (i=Rodrigo@201.86.187.221.adsl.gvt.net.br) |
| 11:58.54 | casix | but in realtime if you use n it don't have order |
| 11:59.00 | casix | in the text file is from top to down |
| 11:59.04 | Dovid | pb an example of what ur trying to do |
| 11:59.05 | casix | i want to simulate it |
| 11:59.07 | *** part/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 11:59.11 | casix | in the db |
| 11:59.12 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 11:59.24 | casix | pb? |
| 11:59.55 | Falcon-x | where downloading zaptel package? |
| 12:00.53 | casix | the goal is not to put the priority but put the order of each app, like you do in the text file configuration when using "n" in the priority or n+101 |
| 12:01.34 | casix | then you can keep on using n in the priority |
| 12:03.49 | casix | Dovid: I have to go, thx for your help. I'll post a feature bug in asterisk to see if someone can implement it. thx a lot |
| 12:07.48 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 12:08.31 | Dovid | Falcon-x: http://www.asterisk.org/downloads |
| 12:09.27 | Dovid | casix: I dont think u can use n in real time |
| 12:10.35 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 12:11.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 12:15.47 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-193-11.d4.club-internet.fr) |
| 12:15.52 | angryuser | re |
| 12:16.37 | angryuser | i have a strange problem, i have agent present in queue support, but when i call teh queue it goes directly n+1, no timeout waiting |
| 12:16.49 | angryuser | http://www.pastebin.ca/553262 my config |
| 12:17.11 | *** join/#asterisk ReDNeQ (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 12:18.19 | *** join/#asterisk EvilGreen (n=Miranda@ppp85-140-139-132.pppoe.mtu-net.ru) |
| 12:23.46 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 12:24.41 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
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| 12:35.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:35.37 | waKKu | morning folks |
| 12:36.33 | waKKu | dudes... can somebody recommends me a ATA to 8 lines ? |
| 12:37.43 | keith4 | digium makes one, don't they? |
| 12:38.10 | keith4 | TDM800P? |
| 12:40.10 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 12:40.18 | waKKu | hm ?.. I want 8 SIP lines |
| 12:40.40 | bobbytux | hello everybody |
| 12:41.57 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 12:42.11 | mosty | tdm800p or an equivalent sangoma card |
| 12:42.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:48.12 | keith4 | waKKu: then you asked the wrong question ;-) |
| 12:48.14 | waKKu | mosty keith4 hm.. i was confuse.. i'm reading about tdm800p now.. very thanx ;) |
| 12:48.31 | waKKu | may it can help me |
| 12:48.35 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 12:48.47 | keith4 | waKKu: what do you want to do? give an overview |
| 12:50.28 | *** join/#asterisk Mulvane_ (i=Mulvane@76.206.23.191) |
| 12:50.54 | Mulvane_ | With VoIP, is it possible to recieve 2-3 calls at once on same line? |
| 12:51.06 | penguinFunk | of course |
| 12:51.10 | Mulvane_ | With multiple calls routed via one phone number? |
| 12:51.29 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 12:51.29 | penguinFunk | if they are purely digital? |
| 12:51.33 | penguinFunk | with analogue you are limited to the number of lines of course |
| 12:51.56 | waKKu | keith4 i'll use 6 phones (hand phones) without computers on the table... so, i want an ATA (or other solution) to do it... but i dont want buy 4 ATA's with 2 channels each |
| 12:51.56 | Mulvane_ | Well, the VoIP is pure dig..And the phones are SIP adapters. |
| 12:52.55 | penguinFunk | yes but you mean, you have 1 phone number (analogue) being called multiple times? |
| 12:53.10 | Mulvane_ | yes |
| 12:53.21 | penguinFunk | unless you have a multi-line you will not get > 1 call going through there |
| 12:53.31 | Mulvane_ | I didn't think so.. |
| 12:53.37 | keith4 | waKKu: analog phones? you mentioned someting about SIP phones before |
| 12:53.44 | mosty | waKKu, the only thing that has more than 2 fxs ports that i know of is a pci card like the tdm800p or a sangoma equivalent, or the asterisk appliance from digium which is way way too expensive in my opinion |
| 12:53.45 | Mulvane_ | I'll have to have it transfer to the other number |
| 12:54.39 | Mulvane_ | I only want to give out the one phone number..i can have asterisk transfer incoming to the second phone number and free up the primary.. |
| 12:54.54 | EvilGreen | waKKu: check Audiocodes MP-118, Addpac AP1200 or new device from Linksys which is coming |
| 12:54.59 | waKKu | keith4 yes.. SIP or IAX... i'll get one FXO and need 8 FXS (2 sipphones on computer) and 6 analog phones |
| 12:55.16 | waKKu | EvilGreen hm.. let me c |
| 12:55.42 | penguinFunk | Mulvane_: we have a multi-line (where there are 6 analogue lines all linked to 1 phone number) these can all me mapped internally to different voip phones then |
| 12:55.54 | penguinFunk | me/be* |
| 12:56.02 | waKKu | mosty i found some gateway with 8 port.. |
| 12:56.09 | waKKu | but a gateway is redundat |
| 12:56.11 | keith4 | waKKu: the digium boards are modular... you can mix and match FXS's and FXO's on the same card |
| 12:56.13 | penguinFunk | therefore: 1 phone number. up to 6 concurrent calls |
| 12:56.19 | Mulvane_ | penguinFunk How did you go about doing that setup? |
| 12:56.26 | penguinFunk | with BT, what country you in? |
| 12:56.29 | mosty | waKKu, do you have an asterisk PBX? |
| 12:56.32 | Mulvane_ | US |
| 12:56.32 | penguinFunk | ask for a multi-line |
| 12:56.36 | waKKu | mosty yes |
| 12:56.42 | penguinFunk | with your call provider |
| 12:57.06 | Mulvane_ | Cool.. |
| 12:57.09 | penguinFunk | :] |
| 12:57.15 | mosty | waKKu, then get a sangoma A200 |
| 12:57.28 | Mulvane_ | Now if I could find a hardware phone that could dial through a computer via USB or something |
| 12:57.33 | waKKu | hm.. i'll check it |
| 12:57.37 | keith4 | as for your question about cost... you can't beat ebay ;-) |
| 12:57.51 | waKKu | hehe... i'm from brazil.. |
| 12:59.11 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 12:59.16 | waKKu | mosty hmm... with this board i wont need other, correct ? i can do 1 FXO and 7 FXS on the same board.. |
| 13:00.27 | mosty | waKKu, the a200 cards do up to 4 ports (in groups of 2), but you can get a remora add-on card for more ports. or an a400 card will do more ports per card |
| 13:01.29 | waKKu | nice.. but I need check for it here, in brazil ;/ |
| 13:01.42 | *** join/#asterisk saftsack (n=saftsack@pD9E06EA7.dip.t-dialin.net) |
| 13:02.45 | tzafrir_laptop | 1 and 7? I don't think so. You'll probably need something of the sort of 2 and 8 |
| 13:02.50 | Mulvane_ | My goal.. I have 6 computers that server as media client...I'd like those to also serve phone purposes via software or a hardware based phone via usb to the specific computer.. |
| 13:03.17 | keith4 | there are usb phones |
| 13:03.32 | Mulvane_ | I'm looking at a couple now.. |
| 13:03.40 | Mulvane_ | trying to find nice wireless ones |
| 13:04.02 | keith4 | there are wireless sip phones, too |
| 13:04.08 | keith4 | they are a bit pricey, imho |
| 13:04.50 | Mulvane_ | A lan phone with wireless handset would be ideal |
| 13:05.13 | keith4 | hmm |
| 13:05.28 | waKKu | yeah.. tzafrir_laptop ... ;/ |
| 13:05.38 | waKKu | maybe and tdm800p is a better choice |
| 13:05.49 | waKKu | s/and/a |
| 13:05.50 | keith4 | Mulvane_: http://www.voip-info.org/wiki-VOIP+Phones see "Cordless Hard Phones" |
| 13:06.03 | keith4 | waKKu: i don't think tdm800p lets you do 1 and 7 |
| 13:06.13 | mosty | waKKu, the sangoma cards are better than the digium cards, better quality all around |
| 13:06.44 | keith4 | either way, he would need 2 four-port cards to get 1 FXO and 7 FXS, right? |
| 13:07.24 | tzafrir_laptop | or 1 TDM800P/8FXS and one TDM400P/1FXO :-( |
| 13:07.46 | [TK]D-Fender | PCI FXS = ass. Get a SIP gateway / ATAs |
| 13:07.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:08.16 | waKKu | [TK]D-Fender yeah.. this was my first idea.. but i need 6 FXS (ATA ports) ... |
| 13:08.40 | waKKu | i dont found some ATA with 6 ports.. so, i would need 3 ATA's ;/ |
| 13:08.43 | [TK]D-Fender | waKKu: ATA's are much more cost effective, more flexible, and reduce server load. |
| 13:09.22 | waKKu | hm.. server load is a good thing to worry.. those machine is poor |
| 13:09.58 | tzanger | I don't like ATAs |
| 13:10.02 | [TK]D-Fender | waKKu: And there are SIP gateways with 4/8/16/24 ports as well. The 8 port range is not very cost effective right now however, but the Linksys SPA 8000 would be exactly waht you want http://www.digiumcards.com/linksys_spa8000 |
| 13:10.07 | tzanger | Katty: you were looking for me earlier? |
| 13:10.35 | tzanger | more wall-warts, heat, cabling... ugh |
| 13:10.47 | tzanger | the only ATA I'd consider is a channel bank. :-) |
| 13:11.01 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 13:11.39 | Katty | tzanger: yes, yes i was. |
| 13:11.47 | Katty | tzanger: but, i can't for the life of me, remember why. |
| 13:12.04 | waKKu | [TK]D-Fender man.. thanks again :) .. u always give me nice alternative ;) |
| 13:12.08 | Katty | tzanger: going to be around most of today? |
| 13:12.42 | [TK]D-Fender | tzanger: overkill is for PANSIES! |
| 13:12.50 | tzanger | Katty: yep should be |
| 13:12.53 | jkiff | How do I kick all the remote connections to asterisk besides me? I had some open, but the ssh sessions died, but they're still hanging around. |
| 13:13.11 | Katty | tzanger: i'm sure i'll remember later (= |
| 13:13.29 | tzanger | heh np |
| 13:13.32 | [TK]D-Fender | jkiff: restart * :) |
| 13:14.18 | [TK]D-Fender | waKKu: the SP is opening to retail at 300$ USD. Excellent value. |
| 13:14.29 | jkiff | [TK]D-Fender: Hehe, *besides* me! I suppose that'll do. |
| 13:14.32 | [TK]D-Fender | waKKu: Assuming they don't screw something up it'll be a start option for * |
| 13:14.56 | [TK]D-Fender | jkiff: I'm sure you'll cry for having to spend the extra 3 keystrrokes to log back in... |
| 13:15.21 | waKKu | [TK]D-Fender unavailable in brazil :/ |
| 13:15.26 | waKKu | damn |
| 13:15.36 | [TK]D-Fender | waKKu: Thats what SHIPPING is for. |
| 13:15.42 | jkiff | It's cool, I'm just pulling your chain. :) |
| 13:15.55 | waKKu | heheh.. |
| 13:16.36 | waKKu | [TK]D-Fender however.. a solution with 3 ATA's is better than use one board tdm800p/fxs right ? |
| 13:16.48 | [TK]D-Fender | waKKu: IMO definately. |
| 13:17.07 | waKKu | ta ;) |
| 13:17.20 | [TK]D-Fender | waKKu: You don't have to worry about getting the RJ11 driectly into the back of your server, PCI hassles, redundancy options, etc. |
| 13:18.20 | tzanger | [TK]D-Fender: heh |
| 13:18.45 | tzanger | well if it's < 8 lines, I could be happy with a SINGLE ATA that didn't fuck around |
| 13:19.12 | *** join/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 13:19.31 | QbY_ | Does anyone know of a Canada Toll Free DID provider? |
| 13:20.19 | jkiff | [TK]D-Fender: Clean potty-humor... I like it! |
| 13:20.29 | waKKu | yeah.. and if I get one ATA blow up, is only 30% of my structure.. |
| 13:20.52 | [TK]D-Fender | tzanger: Thats why I figure the SPA-8000 covers it all.... its price point is blow all other decent 4-port models. |
| 13:21.01 | [TK]D-Fender | waKKu: that too. |
| 13:21.07 | tzanger | I may have to evaluate it |
| 13:21.18 | [TK]D-Fender | waKKu: They can be reused in a variety of ways |
| 13:22.59 | file | [TK]D-Fender: I got my SPA3102 integrated into my setup lastnight... it's working oddly well |
| 13:23.24 | [TK]D-Fender | file: Nifty little box isn't it? |
| 13:23.39 | file | disconnect supervision works :D |
| 13:23.55 | file | I dunno whether Aliant is doing a polarity reversal or what, but it works great |
| 13:24.44 | *** join/#asterisk BouYYY (n=BouYYY@81-86-77-70.dsl.pipex.com) |
| 13:31.52 | [TK]D-Fender | QbY_: www.unlimitel.ca |
| 13:32.02 | *** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net) |
| 13:33.12 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:35.13 | waKKu | [TK]D-Fender what do u think about: AddPac AP1200 ? http://www.addpac.com/addpac_eng/addpac_product_view_detail.php?class_id=19&item_id=66 |
| 13:36.35 | [TK]D-Fender | waKKu: I don't know the brand, so I distrust it from the start... Might do the job fine, or be a total piece of garbage. Hard to say |
| 13:36.48 | angryuser | http://www.pastebin.ca/553262 fender can you help me out, i have 3 queues but the behaviour is not the same, in queue usine if agent is busy, it goes tp N+1 directly without a holdtime, in queue magasin, it goes to n+1 after the timeout |
| 13:36.51 | [TK]D-Fender | waKKu: It says it does what you need basically... |
| 13:37.10 | angryuser | really strange |
| 13:38.29 | waKKu | yeah.. i'll check this brand.. |
| 13:38.37 | [TK]D-Fender | angryuser: do "show queues" |
| 13:38.47 | angryuser | k |
| 13:38.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:38.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:39.30 | angryuser | http://www.pastebin.ca/558671 [TK]D-Fender: |
| 13:39.40 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:40.29 | saftsack | tzafrir_laptop: hi, i tried to patch oslec against the bristuffed zaptel but i get a runtime error while loading zaphfc. are you interested in the background? |
| 13:40.31 | angryuser | when agent/1005 is busy, it goes to n+1, i need it to wait the hold time at least |
| 13:41.07 | tzafrir_laptop | saftsack, hmm, yes |
| 13:41.33 | tzafrir_laptop | what error? |
| 13:41.44 | [TK]D-Fender | angryuser: NoOp the QUEUESTATUS on exit to see what its says |
| 13:42.17 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 13:42.21 | angryuser | [TK]D-Fender: another thing it dont call the agent at all now ;( n+1 directly |
| 13:42.31 | angryuser | [TK]D-Fender: ok |
| 13:42.37 | [TK]D-Fender | angryuser: Actually... it DOESN'T got to n+1 |
| 13:42.43 | [TK]D-Fender | angryuser: don't bother |
| 13:42.51 | [TK]D-Fender | angryuser: You cans ee the call just completely DROPS. |
| 13:43.02 | angryuser | [TK]D-Fender: it goes to n+1 because another phone dials |
| 13:43.04 | [TK]D-Fender | angryuser: And I'd suspect your MoH may be at fault. |
| 13:43.19 | [TK]D-Fender | <PROTECTED> |
| 13:43.21 | [TK]D-Fender | <PROTECTED> |
| 13:43.23 | saftsack | tzafrir_laptop: http://pastebin.ca/558678 |
| 13:43.24 | [TK]D-Fender | no, it HUNG UP |
| 13:43.25 | angryuser | [TK]D-Fender: yes! i had some problems with Moh |
| 13:43.46 | [TK]D-Fender | angryuser: Fix your MP3's. No VBR, make sure 128kbit. no ID3 tags |
| 13:44.13 | ramindia | can any one provide me some URL, where i can look ip authentication examples |
| 13:44.14 | angryuser | [TK]D-Fender: i dont use mp3 only wav 8000 16 bit mono |
| 13:44.33 | angryuser | [TK]D-Fender: because my server crashed a lot if i had some mp3's |
| 13:44.33 | [TK]D-Fender | angryuser: Change you MoH for this queue and retest |
| 13:44.44 | angryuser | [TK]D-Fender: ok |
| 13:44.48 | [TK]D-Fender | angryuser: I'm guessing your WAV's are not much better... |
| 13:44.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:45.14 | angryuser | [TK]D-Fender: hm , used forge8 ;) |
| 13:45.45 | angryuser | where can i find a decent waiting music * compatible |
| 13:46.01 | [TK]D-Fender | angryuser: just use the 3 freeplay samples that came with * |
| 13:46.16 | [TK]D-Fender | angryuser: We are doing a "sanity check" test, so use something we all know works. |
| 13:46.25 | angryuser | [TK]D-Fender: deleted them ;( |
| 13:46.39 | [TK]D-Fender | angryuser: Re-install them ;) |
| 13:46.52 | angryuser | [TK]D-Fender: moh come with addons? |
| 13:47.29 | [TK]D-Fender | angryuser: * base IIRC |
| 13:48.38 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:50.04 | tzafrir_laptop | saftsack, hmm... I wonder how to get that working with a newer zaptel. I patched several newer zaptel versions for it |
| 13:53.25 | ghenry | nice |
| 13:53.35 | ghenry | fax via zaptel over SIP worked fine |
| 13:53.46 | ghenry | fax all through zaptel all works fine |
| 13:53.54 | ghenry | defaulting to analogue though ;-) |
| 13:56.20 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 13:56.53 | irule | what is the default password to cosmopbx? |
| 13:57.28 | tzanger | what the fuck is cosmopbx? |
| 13:59.01 | *** part/#asterisk ramindia (n=ramindia@202.63.96.133) |
| 14:00.46 | keith4 | cosmopbx = asterisk + knoppix? |
| 14:00.53 | *** join/#asterisk CelticSoul (n=CelticSo@bne75-1-81-57-10-55.fbx.proxad.net) |
| 14:02.03 | angryuser | [TK]D-Fender: ok tested with default music same thing, agent/1002 is present, but ig goes to n+1 |
| 14:02.41 | [TK]D-Fender | angryuser: Pastebin it... |
| 14:02.51 | [TK]D-Fender | angryuser: after afdding the NoOp |
| 14:03.03 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:04.45 | angryuser | http://www.pastebin.ca/558730 forgot to ad noop ;) 1 min |
| 14:05.10 | angryuser | it's NoOp(${QUEUESTATUS}) ? |
| 14:06.19 | [TK]D-Fender | yup |
| 14:06.43 | [TK]D-Fender | angryuser: You do not appeat to have changed your MoH |
| 14:07.30 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:07.50 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:07.53 | angryuser | [TK]D-Fender: i did deleted the old onholdmusic , and reload, working |
| 14:08.11 | angryuser | i have new music on hol it is sure |
| 14:08.24 | [TK]D-Fender | angryuser: Something is clearly very wrong... |
| 14:08.29 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 14:08.38 | [TK]D-Fender | angryuser: If it says junk, you've screwed something up... |
| 14:09.56 | angryuser | [TK]D-Fender: i have only 2 line in my mysiconhold.conf [default] mode=files directory=blabla |
| 14:10.30 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:10.39 | [TK]D-Fender | angryuser: Thats fine... you folder has crap in it or your format_mp3 is messed up bad. |
| 14:10.56 | msetim | Hi :-D |
| 14:10.59 | msetim | good morning |
| 14:11.07 | [TK]D-Fender | angryuser: it doesn't which for nothing. I'm also not sure if a "reload" will do the job or if you have to reload the whole module. |
| 14:11.34 | angryuser | [TK]D-Fender: hm, ok let's try |
| 14:11.45 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:11.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:12.34 | angryuser | RTCP SR transmission error, rtcp halted [TK]D-Fender: i have sometime this also, but i have set canreinvite=no to everybody , maybe also moh related? |
| 14:12.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:12.50 | [TK]D-Fender | angryuser: nope |
| 14:14.11 | blitzrage | RTCP is call stats |
| 14:15.27 | angryuser | [TK]D-Fender: i have reloaded musiconhold, working fine for one agent, doing test's to others |
| 14:16.49 | angryuser | [TK]D-Fender: ok working fine, thank you |
| 14:16.57 | [TK]D-Fender | angryuser: np |
| 14:17.08 | CelticSoul | hi guys, please help, do I have to setup RTP or UDP for asterisk to work with SIP? |
| 14:17.25 | angryuser | CelticSoul: rtp.conf range is there |
| 14:17.27 | anonymouz666 | PAP2 sucks balls... it's incredible that there is no butto to reset the device |
| 14:17.28 | [TK]D-Fender | CelticSoul: RTP is a protocol that USES UDP. |
| 14:17.54 | [TK]D-Fender | CelticSoul: and * SIP uses only UDP, not TCP |
| 14:17.59 | EvilGreen | PAP2 is perfect ;) |
| 14:18.21 | [TK]D-Fender | anonymouz666: Who needs a button? thats wht the PHONE is for... |
| 14:18.29 | CelticSoul | I'm newbie here, so pleqse bear with me ;) |
| 14:18.30 | EvilGreen | anonymouz666: is it frozen ? |
| 14:18.38 | anonymouz666 | I need to reset the damn device |
| 14:18.45 | [TK]D-Fender | anonymouz666: UNPLUG IT |
| 14:19.03 | EvilGreen | phone or url (probably) |
| 14:19.08 | [TK]D-Fender | anonymouz666: Highly complex idea... you may want to get some help with this ;) |
| 14:19.09 | anonymouz666 | ok, maybe you can do that in another country |
| 14:19.50 | [TK]D-Fender | I con move my bowel just by using my mind! |
| 14:19.57 | angryuser | what about my strange "RTCP SR transmission error, rtcp halted" message, i google it first , it was written it is canreinvite=yes related, so i sat it to no in sip.conf ;) no effect |
| 14:20.27 | [TK]D-Fender | angryuser: No clue.... get googling! |
| 14:20.39 | [TK]D-Fender | angryuser: But at least now we fixed your primary error |
| 14:20.39 | angryuser | it does not bother the use of the systems, but it bother my console |
| 14:21.50 | angryuser | [TK]D-Fender: yes, really thanks , asterisk crashed sometimes of that crappy mp3's , but i dont understand why does it crashed from converted to standart by forge8 |
| 14:22.02 | CelticSoul | I have an OpenSuse machine run asterisk, 2 windows XP machine run X-lite, I can make call between them but there's no sound |
| 14:22.16 | CelticSoul | when I pick up the phone |
| 14:22.37 | angryuser | CelticSoul: they are in local network? |
| 14:22.40 | [TK]D-Fender | CelticSoul: I presume there is NAT involved somewhere between these 3 systems? |
| 14:22.46 | CelticSoul | yes |
| 14:22.52 | [TK]D-Fender | ~sipnat |
| 14:23.04 | jbot | i heard sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:23.05 | angryuser | CelticSoul: set nat=yes and route ports |
| 14:23.05 | angryuser | in sip.conf |
| 14:23.20 | [TK]D-Fender | CelticSoul: There is a wholoe pile of setting you need to do for * to work with NAT depending on which sides are behind NAT's relative to each other. |
| 14:24.06 | *** join/#asterisk cellphone (i=lysol@monoperative.net) |
| 14:24.42 | CelticSoul | I did enable nat=yes in sip.conf |
| 14:24.53 | rob0 | WOW! Cool!! He did it! |
| 14:24.55 | [TK]D-Fender | CelticSoul: I said a PILE. |
| 14:24.59 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 14:25.10 | [TK]D-Fender | CelticSoul: Go read the like |
| 14:25.12 | [TK]D-Fender | link* |
| 14:25.21 | CelticSoul | OK |
| 14:25.36 | CelticSoul | thanks ;) |
| 14:26.12 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 14:26.13 | zeeesh | hi |
| 14:26.20 | angryuser | http://forums.digium.com/viewtopic.php?p=45547&sid=b13a25e6a529c718986c25da3dcf5134 [TK]D-Fender: hm, i have found this, |
| 14:26.56 | zeeesh | using redhat enterprise ... i enterd agi script but not working by giving this error??? |
| 14:27.18 | zeeesh | BEGIN failed--compilation aborted at /var/lib/asterisk/agi-bin/vopium.agi line 3. |
| 14:27.27 | CelticSoul | so this is NAT problem right? |
| 14:27.28 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 14:27.44 | zeeesh | *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains: |
| 14:27.59 | [TK]D-Fender | CelticSoul: You say there are NAT's involved between these system, so I'd say YES.... |
| 14:28.53 | CelticSoul | [TK]D-Fender: OK, thank you |
| 14:34.20 | *** join/#asterisk Conductor (n=Conducto@port-83-236-229-212.static.qsc.de) |
| 14:34.32 | Conductor | hi all! |
| 14:34.48 | Conductor | is there a amr_codec available? |
| 14:38.35 | coppice | yeah. I have one in my pocket :-) |
| 14:38.55 | Conductor | cool |
| 14:39.01 | Conductor | can i have it? |
| 14:39.11 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f) |
| 14:39.17 | coppice | nah! buy your own phone |
| 14:39.20 | Conductor | coppice, i cant find too much on this topic |
| 14:39.41 | Conductor | coppice, ... |
| 14:40.05 | Conductor | coppice, why? can i take the codec out of the phone and put it into the asterisk box? |
| 14:40.15 | Conductor | coppice, is there a howto ;) |
| 14:40.40 | Conductor | no honestly - i cant find too much on this topic. do you? |
| 14:41.57 | saftsack | tzafrir_laptop: bristuff with zaptel .13 works. but if oslec is compiled in then the load of zaphfc fails in the described segfault |
| 14:42.30 | tzafrir_laptop | just a segfault? an oops? a panic? |
| 14:43.31 | angryuser | [TK]D-Fender: crap, asterisk just quitted, grrrrr |
| 14:44.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:44.18 | CelticSoul | [TK]D-Fender: I don't have sip_rtp.conf file, should I create one? |
| 14:45.17 | *** join/#asterisk logyati (n=suporte@201.29.73.49) |
| 14:46.13 | logyati | hello |
| 14:46.24 | logyati | i have a tdm400p with 1 fxo and 1 fxs |
| 14:46.58 | logyati | i've done all steps to put asterisk to work but i got a problem with the fxs port |
| 14:47.19 | logyati | dmesg |grep -i module |
| 14:47.28 | logyati | Module 0: FAILED FXS (FCC) Module 1: Installed -- AUTO FXO (FCC mode) |
| 14:47.40 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:47.42 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 14:48.04 | logyati | what should i do? i've searched google but i didnt find anything to help |
| 14:48.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:48.27 | [TK]D-Fender | CelticSoul: this is no such thing in * as sip_rtp.conf |
| 14:49.11 | logyati | btw im using debian 3.0 sarge |
| 14:50.03 | CelticSoul | [TK]D-Fender: someone said about it in the link you gave me: .....your sip.conf general settings into sip_nat.conf |
| 14:50.25 | CelticSoul | oops, it's sip_nat.conf |
| 14:50.34 | CelticSoul | sorry |
| 14:50.37 | CelticSoul | ;) |
| 14:50.55 | angryuser | P[ 2] --> Didn't find BC so temporarly creating dummy BC (l3id:40049) on this port. what is that ? misdn message |
| 14:51.21 | [TK]D-Fender | CelticSoul: And that is not an * file... that is a BS FreePBX config file that gets included. |
| 14:51.56 | [TK]D-Fender | CelticSoul: Ignore his comment (or at least that it belongs in a separate file". |
| 14:52.11 | logyati | can anyone help me? |
| 14:52.13 | CelticSoul | OK, thank you |
| 14:52.22 | [TK]D-Fender | logyati: Did youplug in the molex connector? |
| 14:52.41 | logyati | you mean the energy cable to the board? |
| 14:52.41 | logyati | yes |
| 14:52.45 | file | [TK]D-Fender: are you having a fun day with the #asterisk crowd? |
| 14:53.18 | [TK]D-Fender | logyati: pastebin your zaptel & zapata.conf. |
| 14:53.20 | [TK]D-Fender | ~pb |
| 14:53.31 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 14:53.32 | logyati | ok |
| 14:53.32 | logyati | w8 |
| 14:53.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 14:53.43 | [TK]D-Fender | file: ooohhh yeah |
| 14:53.59 | angryuser | http://www.pastebin.ca/558834 fender can you look at log, just after this my server died |
| 14:54.02 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 14:54.35 | *** join/#asterisk kay2 (n=two@gob75-7-82-247-113-230.fbx.proxad.net) |
| 14:54.36 | *** join/#asterisk dreamind (n=dreamind@p54A7A071.dip0.t-ipconnect.de) |
| 14:54.39 | dreamind | Hi folks |
| 14:54.43 | casix | anyone know where can I find documentation about realtime? |
| 14:54.54 | dreamind | can anybody tell me how to do any interactive stuff in a macro? |
| 14:54.55 | drako | im having problem with Monitoring calls, when a call is recieved from a pstn and is transfered to internal extensions it crash asterisk saying it can't merge or re-record |
| 14:54.56 | casix | about the source code of realtime? |
| 14:55.11 | dreamind | I need to do it in a macro because I'm calling that macro for the callee channel in Dial() |
| 14:55.27 | dreamind | and WaitExten does not work as expected in a macro :( |
| 14:55.43 | logyati | pastebin has errors, i cant paste |
| 14:55.44 | dreamind | moreover, it seems ${EXTEN} "jumping" for dtmf signals does not work in macros |
| 14:55.53 | logyati | did you know another website? |
| 14:56.09 | [TK]D-Fender | dreamind: "show application read" |
| 14:56.34 | drako | logyati, http://pastie.caboo.se/ |
| 14:56.38 | [TK]D-Fender | logyati: it jsut worked for these 2 people. You need to try again... |
| 14:56.42 | dreamind | [TK]D-Fender: thanks a lot! :D |
| 14:57.05 | Qwell | logyati: uuoc.com |
| 14:57.26 | Qwell | or any of the above referenced ones... |
| 14:57.40 | [TK]D-Fender | angryuser: that debug doesn't tell me much without CLI output. |
| 14:58.03 | angryuser | i have no messages before crash nothing at all |
| 14:58.10 | *** join/#asterisk tparcina (n=tparcina@89-172-116-117.adsl.net.t-com.hr) |
| 14:58.26 | tparcina | anybody from Canada? |
| 14:58.54 | logyati | [tk]d-fender: zapata.conf http://pastie.caboo.se/69467 |
| 14:59.35 | logyati | [tk]d-fender: zaptel.conf http://pastie.caboo.se/69468 |
| 14:59.52 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:00.17 | *** join/#asterisk orn (n=orn@staff.iphive.is) |
| 15:01.11 | ccesario | morning.... |
| 15:01.34 | logyati | gm ccesario |
| 15:01.37 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 15:01.40 | tzafrir_laptop | logyati, you use debian 3.1 (Sarge), I guess |
| 15:01.53 | ccesario | I'm try compile app_rxfax and app_fxfax, and I get this error |
| 15:01.55 | ccesario | http://pastebin.ca/558851 |
| 15:01.56 | logyati | tzafrir_laptop: yes |
| 15:01.59 | tzafrir_laptop | If you got that error, you zapata.conf still isn't relevant, and your zaptel.conf isn't either |
| 15:02.18 | logyati | tzafrir_laptop: what you mean/] |
| 15:02.26 | ccesario | any idea ? |
| 15:03.00 | tzafrir_laptop | either the module is bad, soemthing is wrong in the PCI , no power cable connected to the card, or whatever |
| 15:05.02 | orn | Hi. Any ideas on how I can prefix the outgoing CID on certain outbound routes? |
| 15:05.18 | ccesario | tzafrir_laptop, idea abou this ? |
| 15:05.26 | neverblue | morning |
| 15:05.35 | neverblue | tparcina, I am |
| 15:06.19 | tzafrir_laptop | warning: implicit declaration of function ‘t30_set_ecm_capability’ |
| 15:06.40 | tzafrir_laptop | ccesario, that should come from spandsp, right? |
| 15:07.14 | tzafrir_laptop | how is spandsp installed? |
| 15:07.14 | *** join/#asterisk adker (n=chatzill@74-33-221-202.br1.glv.ny.frontiernet.net) |
| 15:07.23 | ccesario | yes, and I have spamdsp installed and compiled |
| 15:07.40 | ccesario | s/spandsp |
| 15:07.54 | logyati | [tk]d-fender: and you? any ideas? |
| 15:08.06 | casix | anyone know where can I find doxygen information about realtime?? |
| 15:08.23 | ccesario | tzafrir_laptop, /usr/local/include/ I have spandsp.h and spandsp/ |
| 15:08.24 | [TK]D-Fender | logyati: I agree |
| 15:08.30 | tparcina | neverblue: do you work as network VoIP/tehnican? |
| 15:08.38 | neverblue | negative |
| 15:08.43 | neverblue | i am just a wanna-be |
| 15:09.12 | tzafrir_laptop | sorry, I don't know |
| 15:09.12 | [TK]D-Fender | orn: "show function CALLERID" |
| 15:09.16 | tparcina | neverblue: I was offered a job as netwok VoIP/tehnician in NWT and I'm thinking of moving there |
| 15:09.31 | neverblue | good luck :) |
| 15:09.55 | tparcina | I know it's small town , but I would like to see how it would be to live there |
| 15:10.26 | neverblue | well, you might save some money |
| 15:10.37 | neverblue | unless your an alcoholic |
| 15:10.44 | tparcina | anyway, i waned to ask, what is the avarage salary forr this kind of job at NWT? |
| 15:10.57 | neverblue | NWT pays more, i am positive |
| 15:11.08 | tparcina | no, I'm not alcoholic :) how do you think - I could save money? |
| 15:11.20 | dreamind | [TK]D-Fender: hm, seems that doesn't work... I always get "user entered nothing"... :( |
| 15:11.22 | neverblue | well, you have nothing to spend money on... |
| 15:11.23 | neverblue | lol |
| 15:11.50 | tparcina | well, can you tell me aproximatly, how much can I expect for month or year? |
| 15:11.58 | neverblue | negative |
| 15:12.06 | neverblue | ask your employer |
| 15:12.19 | [TK]D-Fender | dreamind: Veryfy that your DTMF is functional outside your usage of this macro. |
| 15:12.26 | tparcina | neverblue: :) that doesn't sound promising, I like to spend my money :) |
| 15:12.35 | neverblue | ebay! |
| 15:13.30 | dreamind | [TK]D-Fender: hm, maybe my iax2 connection is somehow a problem to those dtmf signals? |
| 15:13.44 | tparcina | neverblue: there has to be some avarage salary... or you don't wana to tell me(don't know? |
| 15:14.13 | [TK]D-Fender | dreamind: go test... |
| 15:14.32 | neverblue | why do you think I would know? |
| 15:14.37 | neverblue | cause I am from Canada? |
| 15:14.43 | neverblue | give me a break |
| 15:14.45 | neverblue | lmao |
| 15:14.45 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:14.46 | tparcina | yes, but I prefer spending money by going out, enyojing myself and not buying things over net :) |
| 15:14.55 | zeeesh | *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains: |
| 15:15.17 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 15:15.19 | *** join/#asterisk ManxPower (n=manxpowe@135.sub-75-201-87.myvzw.com) |
| 15:16.18 | mocker | zeeesh: Do you have AGI.pm in the @INC path? |
| 15:16.23 | neverblue | what would a twisted pair transceiver be used for? |
| 15:16.32 | neverblue | just happened to find one laying around |
| 15:18.02 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 15:18.31 | logyati | guys please take a look at my /var/log/messages -> http://pastie.caboo.se/69471 |
| 15:18.35 | drako | Jun 11 17:15:11 WARNING[30044]: res_monitor.c:319 ast_monitor_change_fname: Cannot change monitor filename of channel Zap/3-1 to null |
| 15:19.16 | logyati | i think it could be driver problem, so, please look that and tell me if im right :) |
| 15:22.45 | *** join/#asterisk EvilGreen_ (n=Miranda@ppp85-141-152-123.pppoe.mtu-net.ru) |
| 15:22.57 | *** join/#asterisk saftsack (n=saftsack@pD9E06EA7.dip.t-dialin.net) |
| 15:23.38 | saftsack | tzafrir_laptop: do you have any ideas? |
| 15:24.49 | ccesario | tzafrir_laptop, idea ? |
| 15:25.47 | tzafrir_laptop | ccesario, make sure that CFLAGS include -I/usr/local/include ? |
| 15:26.08 | tzafrir_laptop | saftsack, not right now |
| 15:26.29 | saftsack | ok ;) s******** junghanns ^^ |
| 15:27.00 | Katty | is 9 the generally accepted To Dial Longdistance number? |
| 15:27.46 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:29.08 | tzafrir_laptop | saftsack, use vzaphfc? |
| 15:29.21 | tzafrir_laptop | hmmm? |
| 15:29.41 | saftsack | im using zaphfc at the moment but i can try vzaphfc |
| 15:29.58 | saftsack | but i dont think that this is a complete rewrite |
| 15:30.24 | [TK]D-Fender | logyati: Your module might simply be dead. Try it in another module's slot and if it fails there, then RMA it. |
| 15:30.41 | ccesario | tzafrir_laptop, see http://pastebin.ca/558913 |
| 15:30.55 | ccesario | -I/usr/local/include is ok |
| 15:30.55 | tzafrir_laptop | saftsack, where was that pastebin again? |
| 15:31.19 | saftsack | uuuuh that are questions ^^ if im honorous i dont know |
| 15:31.40 | logyati | [tk]d-fender: sry to ask, but, what is RMA? hehehe |
| 15:31.41 | [TK]D-Fender | Katty: Nothing to do with LD. ( as a prefix is jsut a stupid holdover from old annoying PBX's that did the same. Prefixes are unnecessary and I avoid them if at all possible. |
| 15:31.54 | tzafrir_laptop | ccesario, so maybe the version of spandsp doesn't match? |
| 15:32.00 | tzafrir_laptop | what version do you use? |
| 15:32.06 | logyati | [tk]d-fender: and i already changed slots |
| 15:32.07 | [TK]D-Fender | logyati: Return Merchandise Authorization : AKA send it BACk to get a new one |
| 15:32.21 | [TK]D-Fender | logyati: Not talking PCI slots, I'm talking position ont he card itselrf. |
| 15:32.50 | [TK]D-Fender | logyati: And if you've doe that (esp BOTH), then definitely call up to return it. |
| 15:32.56 | ccesario | tzafrir, spandsp-0.0.3pre22.gz |
| 15:33.41 | zeeesh | <mocker>: how to check Asterisk/AGI.pm in @INC path? |
| 15:34.10 | saftsack | tzafrir_laptop: do you have the link in your cache? |
| 15:34.11 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 15:35.10 | mocker | zeeesh: Does it print your @INC path after that error? |
| 15:35.36 | tzafrir_laptop | saftsack, http://pastebin.ca/558678 |
| 15:36.15 | saftsack | yes so i wouldnt suggest that vzaphfc would help in this case ... |
| 15:36.32 | zeeesh | yes its showing more than 8 lines "like" /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi |
| 15:37.29 | ccesario | tzafrir_laptop, spandsp-0.0.3pre22.gz |
| 15:38.07 | mocker | zeeesh: So you need to make sure that you copy AGI.pm to one of those directories. |
| 15:38.26 | tzafrir_laptop | Asterisk/AGI.pb |
| 15:38.29 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:38.31 | tzafrir_laptop | Asterisk/AGI.pm , that is |
| 15:38.47 | tzafrir_laptop | and it has to be in the standard perl include path |
| 15:39.14 | *** join/#asterisk astguy (n=astguy@c-24-8-95-194.hsd1.co.comcast.net) |
| 15:39.19 | tzafrir_laptop | (or check 'use lib') |
| 15:40.05 | tzafrir_laptop | it is part of the asterisk-perl distribution |
| 15:41.05 | *** join/#asterisk aiksa[LV] (i=THUG@87.226.106.195) |
| 15:43.13 | tzafrir_laptop | saftsack, I need to go now, bt looking at zt_chan_reg(), I can't really figure out why the condition if (!chans[x]) { |
| 15:43.20 | tzafrir_laptop | was met for any channel |
| 15:43.26 | tzafrir_laptop | (at registration time) |
| 15:44.53 | penguinFunk | anyone here know much about isdn30? |
| 15:45.05 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-79.ph.ph.cox.net) |
| 15:45.26 | penguinFunk | is there a device that you can manage what channels are to be used for what? |
| 15:45.41 | penguinFunk | i.e use 15 channels for voip and 15 channels for broadband etc |
| 15:46.44 | penguinFunk | we're hopign that isdn lines will give perfect clear sound quality compared to the analogue lines here that do not have a very good signal to noise ration when analyzed with fxotune |
| 15:46.51 | dreamind | [TK]D-Fender: hm, just tested, with the asterisk directly connected to my zaptel card, dtmf signals work there, but there is the same problem - read does not read any value... it always reports "user entered nothing" |
| 15:46.55 | penguinFunk | ratio* |
| 15:47.01 | dreamind | :( |
| 15:47.56 | [TK]D-Fender | dreamind: First you are talking about IAX2 then Zaptel. APPLES & ORANGES. test he device you are supposed to be working with using a different bit of dilaplan. |
| 15:48.00 | [TK]D-Fender | dialplan |
| 15:48.34 | dreamind | [TK]D-Fender: ok... will test it... |
| 15:49.23 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.10) |
| 15:49.30 | Inkubot | hi :) |
| 15:49.40 | killfill | Inkubot: ! |
| 15:49.44 | Inkubot | i'm getting this message from the CLI WARNING[12044]: acl.c:378 ast_ouraddrfor: Cannot create socket |
| 15:49.57 | Inkubot | hi killfill, how are you ? HAHA |
| 15:50.01 | killfill | jeje |
| 15:50.19 | killfill | Inkubot: your not running asterisk inside a freebsd jail arent ya?... |
| 15:50.29 | Inkubot | when this message appears, incoming calls are rejected.. |
| 15:50.32 | zeeesh | guys actually i copied agi file from my older redhat server and my server has latest AGI.pm so... is perl module making conflict?? |
| 15:50.33 | Inkubot | nops |
| 15:50.38 | killfill | if so check the socket sysctl thingy.. :P |
| 15:50.40 | killfill | ah |
| 15:50.50 | Inkubot | it's Debian |
| 15:51.40 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 15:51.40 | zeeesh | <tzafrir_laptop>:actually i copied agi file from my older redhat server and my server has latest AGI.pm so... is perl module making conflict?? |
| 15:51.40 | Inkubot | it's not my system :) |
| 15:51.44 | angryuser | tzafrir: you here? |
| 15:52.20 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:52.20 | killfill | werent you a bsd buddy?.. :P |
| 15:52.24 | killfill | ah.. |
| 15:53.39 | Inkubot | but for Asterisk i think (for support) is better linux. Generally i use Slackware... the only BSD-Like linux |
| 15:53.48 | angryuser | i have astribank alarm received (only astribank) then in "zap show status" it's state goes "unconfigured" had to restart asterisk/load firm/start |
| 15:54.01 | angryuser | what the hell is that? |
| 15:54.40 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:59.41 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-fe74fe70049e094f) |
| 15:59.45 | killfill | Inkubot: i just broth a Sangoma A101d card.. i hope the fbsd support is better than the digium te cards... |
| 16:02.12 | ManxPower | I don't suppose anyone can help me via /msg with a problem I have with the unix "find" utility. |
| 16:04.26 | *** part/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 16:04.35 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:05.05 | orn | here's a line from my extensions.conf: exten => s,n(normcid),Set(USEROUTCID=${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)}). |
| 16:05.34 | orn | can i somehow add a prefix to that |
| 16:05.51 | orn | that is, do something like exten => s,n(normcid),Set(USEROUTCID=354${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)}) |
| 16:06.31 | orn | I tried that, but when I do that the outbound CID is only 354... why does it not interpret the variable? is there some method of doing it like in perl, like... 354 . $variable ? |
| 16:12.27 | Hmmhesays | so has anyone done sms with asterisk? |
| 16:12.35 | Qwell[] | Hmmhesays: with chan_cellphone |
| 16:12.44 | Qwell[] | that's kinda cheating though |
| 16:14.17 | Hmmhesays | how so? |
| 16:14.17 | Qwell[] | well, it's not something you'd use to send tons of messages out |
| 16:14.17 | Uatec | oops |
| 16:14.17 | Hmmhesays | define "tons" |
| 16:14.18 | [TK]D-Fender | orn: your DB lookup is returning nothing and that is the fault of your dialplan. welcome to FreePBX, which is NOT supported here... |
| 16:15.06 | Uatec | oops, i just called someone up thinking they were Digium support, and explained my entire problem to them |
| 16:15.11 | Uatec | the poor woman was so confused |
| 16:15.18 | Uatec | she was a receptionist at an autioneers |
| 16:15.48 | Qwell[] | Uatec: that just may have made my day :p |
| 16:16.05 | Hmmhesays | nice |
| 16:16.52 | mog | lol |
| 16:17.26 | mocker | <PROTECTED> |
| 16:19.47 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 16:21.11 | rob0 | To call Digium, reenable the [demo] and dial 500. :) |
| 16:22.01 | orn | [TK]D-Fender: So doing somthing like 354${DB(random/db/lookup)} should work? |
| 16:22.55 | mog | iax2/guest@misery.digium.com <- works to |
| 16:23.58 | *** part/#asterisk aiksa[LV] (i=THUG@87.226.106.195) |
| 16:24.04 | file | mog: I'm not sick but I'm not well... |
| 16:24.08 | file | and I'm so hot cause I'm in hell |
| 16:24.16 | logyati | [tk]d-fender: well, im gonna keep going without FXS then... now please tell me, is this "zaptel: no version for "struct_module" found: kernel tainted." a relevant error? |
| 16:24.46 | mog | I'm not sick but I'm not well, And it's a sin to live so well? |
| 16:24.47 | rob0 | Hey, I have what's probably a stupid question. I've got my Sipura SPA-2000 working through an X100P clone. But dialing anything seems to have a long delay, which shows nothing in my -vvvc console. What could that be? Reverse DNS failure? |
| 16:25.39 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-39-83.dsl.tul2ok.sbcglobal.net) |
| 16:25.58 | [TK]D-Fender | orn: It'll work, just like the other does. Its a question of there being a VALUE to find there or not. |
| 16:26.25 | [TK]D-Fender | logyati: Thats just a "warning" IIRC |
| 16:27.57 | orn | [TK]D-Fender: ok, thanks. |
| 16:29.15 | logyati | [tk]d-fender: ok ty |
| 16:30.27 | tzanger | can you do batphones with SIP phones? |
| 16:30.29 | tzanger | with zaptel it's simple |
| 16:30.56 | file | if they have an option to do it. |
| 16:31.16 | [TK]D-Fender | tzanger: Depends on the phone clearly. |
| 16:31.29 | tzanger | [TK]D-Fender: hmm |
| 16:31.32 | tzanger | that kind of blows |
| 16:31.51 | s0ck | [TK]D-Fender: lspci... is there a similar command for isa cards? |
| 16:32.33 | [TK]D-Fender | s0ck: Wouldn't know |
| 16:32.45 | s0ck | trying to configure bri with some nondescript isa card |
| 16:32.52 | s0ck | can't see anything on dmesg output either ;/ |
| 16:33.10 | coppice | s0ck: no. there is no way to scan ISA cards. |
| 16:33.11 | file | s0ck: I wouldn't hold your breath... |
| 16:33.51 | coppice | i wish I had a machine with an ISA slot. I don't have anywhere to plug the adaptor card for my device programmer any more :-( |
| 16:34.04 | s0ck | the card looks older than me, to be fair |
| 16:34.16 | Hmmhesays | around here, you can get one in about a week if you're willing to go dumpster diving |
| 16:34.18 | Hmmhesays | ;) |
| 16:34.43 | s0ck | sposed to be some kind of eicon isdn card |
| 16:34.50 | s0ck | beginning to wonder if it's completely dead |
| 16:37.32 | coppice | if its dead a local museum might like it :-) |
| 16:38.21 | s0ck | hehe |
| 16:38.27 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 16:38.58 | *** join/#asterisk guillote_GNU (n=guillote@190.7.23.192) |
| 16:39.03 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 16:39.22 | s0ck | anyone used a snom 360 or ip501? |
| 16:39.31 | s0ck | both seem to be around the same price |
| 16:40.40 | *** part/#asterisk orn (n=orn@staff.iphive.is) |
| 16:41.04 | thekidrio | polycom ip 501? |
| 16:41.39 | thekidrio | that polycom 501 is probably one of the most common phones, at least as far as I have seen |
| 16:42.07 | [TK]D-Fender | s0ck: Polycom > all |
| 16:42.20 | thekidrio | yeah, i would have to agree |
| 16:42.26 | thekidrio | with cisco after that |
| 16:42.32 | thekidrio | I love the new cisco phones heh |
| 16:43.38 | s0ck | [TK]D-Fender: that seems to be the general opinion |
| 16:43.53 | s0ck | what's so special about them though... |
| 16:44.02 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:44.03 | thekidrio | rock solid, predictable |
| 16:44.14 | thekidrio | and widely used/supported |
| 16:44.15 | rob0 | Hmmm, reverse DNS is fixed for the Sipura, but still takes 8-10 seconds to place a call. |
| 16:44.22 | [TK]D-Fender | Cisco's SIP functionailty & stability is a bit lacking. Physically nice, but their software needs work they are unwilling to put into it (for obvious reasons) |
| 16:44.24 | *** part/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 16:44.43 | s0ck | thekidrio: you could argue that's a function of firmware etc |
| 16:44.51 | thekidrio | yeah, I agree, its the look that gives me the warm and fuzzies |
| 16:44.54 | s0ck | are there any features that stand out amongst the general ip phones riff raff? |
| 16:45.26 | thekidrio | s0ck, well yeah it's a function of firmware, but that is a sub-function of properly focused engineering |
| 16:45.35 | [TK]D-Fender | s0ck: Where are you located? |
| 16:45.37 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 16:45.37 | s0ck | i thought the 501 looks a bit daft compared to the snom |
| 16:45.39 | rene- | hello |
| 16:45.41 | s0ck | maybe it's a uk thing |
| 16:45.43 | s0ck | ^ |
| 16:45.47 | [TK]D-Fender | :/ |
| 16:45.47 | rene- | <PROTECTED> |
| 16:46.05 | thekidrio | they are a bit boring looking i suppose heh |
| 16:46.09 | [TK]D-Fender | s0ck: Polycom is definately a better choice, but in your case not a CHEAPER one... |
| 16:46.09 | rene- | how come when i do chan_spy(channel) the spyed on channel always can hear me? |
| 16:46.19 | [TK]D-Fender | s0ck: Depends what you want out of a phone. |
| 16:46.22 | s0ck | i am not necessarily worried about cost |
| 16:46.26 | thekidrio | I have a giant yellow pingtel |
| 16:46.35 | s0ck | however, i do not want the customer ringing me to whinge about this/that/the other |
| 16:46.35 | [TK]D-Fender | s0ck: Oh... in that case screw the rest and go Polycom ;) |
| 16:46.37 | thekidrio | its friggin massive |
| 16:46.50 | s0ck | is the handset nice and heavy on the poly? |
| 16:46.57 | thekidrio | s0ck, go polycom for sure |
| 16:47.04 | s0ck | i've only tried a gxp2000 and some aastra phone so far |
| 16:47.08 | thekidrio | its tried and true for customer satisfaction |
| 16:47.09 | s0ck | and they felt terribly cheap |
| 16:47.09 | [TK]D-Fender | s0ck: Model to choose depends on whereth you need the microbrowser, have/will have PoE, etc. |
| 16:47.12 | [TK]D-Fender | ~gs |
| 16:47.25 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:47.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 16:47.28 | s0ck | have no immediate need for poe |
| 16:47.32 | Inkubot | . |
| 16:47.48 | s0ck | im also familiar with the general opinion on gs |
| 16:47.49 | [TK]D-Fender | s0ck: IP 501 is a fine choise then. If you INTEND to go PoE at some point, IP430. |
| 16:47.56 | s0ck | my experience of them doesn't necessarily reflect that tho |
| 16:48.25 | rob0 | Purity of Essence = Peace on Earth |
| 16:48.42 | *** part/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 16:48.45 | rob0 | You can't fight in here! This is the War Room!! |
| 16:49.06 | [TK]D-Fender | s0ck: many do just fine with the GS's, but they are flakey, their firmware is about as stable as Andy Kaufman, and are cheap flimsy crap in feel. |
| 16:49.25 | s0ck | they do feel cheap ;/ |
| 16:49.33 | s0ck | it's immediately noticeable imo |
| 16:50.22 | [TK]D-Fender | s0ck: if you don't need a pass-through port for PC, the IP 320 + power brick might be a thought. |
| 16:53.01 | CunningPike | s0ck: The 501 is the best choice, in our experience - even if you want PoE |
| 16:54.26 | [TK]D-Fender | s0ck: Only reason I recommend against IP 501 for PoE is that you need to buy an extra special bulky adapter cable for it to work with PoE. Adds a noticable amount to the cost and is a bit bulky. Makes the other choices much more cost effective. |
| 16:55.02 | [TK]D-Fender | s0ck: As a phone all the polycoms with the a speakerphone do the job geat and pretty much equally. the frills are rather superfluous. |
| 16:55.15 | ManxPower | GS has had YEARS to make stable firmware, they still have not done so. They have had YEARS to make good hardware -- they have not done so. |
| 16:55.25 | ManxPower | Not much they can do now -- it is too late |
| 16:55.29 | s0ck | i'd quite like the option for the user to log into the phone ala hot desk |
| 16:55.35 | thekidrio | I still miss my pingtels |
| 16:55.36 | s0ck | can the 501's do that |
| 16:55.44 | ManxPower | s0ck: ANY PHONE can do that. |
| 16:55.46 | thekidrio | loved those giant ugly phones |
| 16:55.53 | ManxPower | you write it in the dialplan |
| 16:56.23 | s0ck | is that a work around or the normal way to do it |
| 16:57.53 | ManxPower | s0ck: asterisk has no built in app to do hotdesking |
| 16:59.11 | ManxPower | Asterisk is NOT a full built PBX. It is a toolkit for building a PBX |
| 17:01.38 | Hmmhesays | toolkits are cool |
| 17:01.40 | thekidrio | that's why we get paid the big bucks for supporting it... |
| 17:02.00 | rene- | full built pbx: tixbox |
| 17:02.11 | Hmmhesays | trixbox |
| 17:02.19 | Hmmhesays | or asterisk 1.4 with the gui |
| 17:02.32 | Hmmhesays | use at your own risk though |
| 17:02.34 | rene- | yeah |
| 17:02.58 | Hmmhesays | the new asterisk gui is really nice |
| 17:03.06 | Hmmhesays | but I don't know if i'd put 1.4 into a business using it y et |
| 17:03.09 | Hmmhesays | *yet |
| 17:03.31 | rene- | i am using 1.4 |
| 17:03.41 | Hmmhesays | what type of setup? |
| 17:03.49 | mvanbaak | full blown configfile maddness virus: freepbx/trixbox |
| 17:03.53 | rene- | 30 sip lan customers |
| 17:04.00 | rene- | 2 isdn e1 pri |
| 17:04.00 | Hmmhesays | using the gui? |
| 17:04.04 | rene- | not at all |
| 17:04.09 | rene- | the gui i dont like it at all |
| 17:04.20 | rene- | well tho i havent checked out the new one |
| 17:04.25 | rene- | the old one was horrible |
| 17:04.38 | Hmmhesays | I checked it out and installed friday for testing |
| 17:04.42 | Hmmhesays | I like it |
| 17:05.30 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:06.01 | *** join/#asterisk Maxxed (i=foobar@65.59.245.122) |
| 17:06.18 | Maxxed | i have a quick ? |
| 17:06.31 | Hmmhesays | no quickies here |
| 17:06.38 | Maxxed | right now the telco is only sending the last 4 digits of a dialed number |
| 17:06.46 | Maxxed | and im going to increase it to the full 10 |
| 17:07.08 | Maxxed | will i need to change anything other than my extentions.conf to acomodate the change? |
| 17:07.12 | Hmmhesays | um yes |
| 17:07.15 | rene- | i still dont like it |
| 17:07.16 | Maxxed | the DNIS |
| 17:07.27 | Hmmhesays | wherever you have matching for the 4 digits, you'll have to now match 10 |
| 17:07.30 | Maxxed | Hmmhesays, yes? |
| 17:07.32 | Maxxed | well yeah |
| 17:07.32 | *** part/#asterisk rene- (n=rene@200.34.66.137) |
| 17:07.41 | Maxxed | in the extentions.conf |
| 17:07.41 | Hmmhesays | unless you are matching it with something like _X. which you shouldn't be |
| 17:07.53 | Maxxed | righty'o |
| 17:07.54 | Hmmhesays | other than that, no |
| 17:08.06 | Maxxed | but i dont have to like jack with anything in the zap configs |
| 17:08.08 | Maxxed | ok ok, cool :) |
| 17:08.12 | Maxxed | i was just making sure |
| 17:08.16 | Hmmhesays | you shouldn't |
| 17:08.28 | Maxxed | i had them change it the other day, and then nobody could dial in |
| 17:08.29 | Hmmhesays | I had one telco guy tell me that he had never had anyone request 10 digits before |
| 17:08.37 | Maxxed | wuah!? |
| 17:08.44 | Hmmhesays | change it without changing extensions.conf? |
| 17:08.48 | Maxxed | i would think just the oposite |
| 17:09.24 | logyati | step two hehe: where can i find a very good tutorial about extensions.conf ? cos i cant understand this file! |
| 17:09.32 | Maxxed | good luck ;) |
| 17:09.56 | logyati | im trying to configure asterisk here... the phone line needs "0" before dialing any external number |
| 17:10.07 | logyati | i dont know how to do this |
| 17:11.57 | [TK]D-Fender | logyati: "show application dial" . what you dial and how is entirely up to you. |
| 17:12.18 | [TK]D-Fender | logyati: Just shove the "0" in front of whatever you want. |
| 17:13.14 | EvilGreen_ | exten => _9Z.,1,Macro(trunkdial,${TRUNK1}/0${EXTEN:1}) |
| 17:13.24 | [TK]D-Fender | logyati: Lets say uou're using a Zaptel analog channel to dial and had a pattern like _NXXXXXX , you would do something like "exten => _NXXXXXX,1,Dial(ZAP/1/0${EXTEN}) |
| 17:14.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 17:15.32 | logyati | hmmm |
| 17:15.32 | s0ck | ManxPower: i was thinking more along the lines of this |
| 17:15.57 | s0ck | at the moment, the phone registers with whatever details i put into the account. you power cycle it, it auto reg's |
| 17:16.19 | s0ck | alternatively, the phone powers up but waits for extension/pass before it will proceed |
| 17:16.39 | thekidrio | anyone here use the astaro security gateway 220? |
| 17:16.40 | s0ck | hence my confusion about the dialplan because auth should happen way before it gets anywhere else |
| 17:16.43 | s0ck | if that makes sense |
| 17:16.50 | ManxPower | s0ck: that does not look like hotdesking to me. |
| 17:17.12 | ManxPower | hot desk. walk up to a phone, dial a code, your call now go to that phone. |
| 17:17.24 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 17:17.35 | s0ck | agreed |
| 17:17.45 | s0ck | prior to dialling the 'code' was the phone usable? |
| 17:18.07 | logyati | omg its so complicated to me |
| 17:18.11 | logyati | like a NASA couse |
| 17:18.17 | logyati | course |
| 17:18.48 | [TK]D-Fender | logyati: .... |
| 17:18.50 | [TK]D-Fender | ~book |
| 17:19.02 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:19.03 | [TK]D-Fender | ~wikis |
| 17:19.05 | jbot | i guess wikis is http://www.voip-info.org |
| 17:19.11 | [TK]D-Fender | logyati: Or hire a consultant / go to a conference, etc. |
| 17:19.29 | s0ck | i'd like to be able to report on what time a user logged into the phone and started taking calls kinda thing |
| 17:19.37 | [TK]D-Fender | logyati: Do you have any programming experience beyond setting up a VCR? |
| 17:19.48 | s0ck | perhaps im approaching this from the wrong angle |
| 17:19.58 | [TK]D-Fender | s0ck: Are you referring to call-center type scenarios? |
| 17:20.00 | s0ck | but if so, you have both a phone and a user authing? |
| 17:20.01 | logyati | extensions.conf is no |
| 17:20.03 | s0ck | i guess so |
| 17:20.11 | logyati | ops, sry i was typing |
| 17:20.15 | logyati | i ment no, i dont have |
| 17:20.18 | logyati | experience |
| 17:20.25 | [TK]D-Fender | s0ck: So you mean for them to act like roaming agents to log in and take suppotr calls, etc? |
| 17:20.37 | coppice | setting up most VCRs would be the assembly language programming of consumer applicances :-) |
| 17:20.47 | [TK]D-Fender | logyati: So you've never done ANY programming before? no BASIC, LOGL, ANYTHING? |
| 17:20.51 | logyati | i never did nothing like this before, im trying to learn building a voip server, with openser radius ldap mediaproxy and asterisk |
| 17:20.59 | logyati | ah, programming yes |
| 17:21.03 | [TK]D-Fender | coppice: Assembly with 4 instructions ;) |
| 17:21.11 | logyati | i know perl hehe |
| 17:21.17 | Hmmhesays | does asterisk 1.4 support instant messaging yet? |
| 17:21.23 | [TK]D-Fender | logyati: Ok, well extensions.conf should be a BREEZE then. |
| 17:21.24 | Hmmhesays | like x-lite instant messaging? |
| 17:21.32 | [TK]D-Fender | Hmmhesays: N,o. |
| 17:21.48 | logyati | and is there something like "extensions.conf" for dummies? lol |
| 17:21.49 | coppice | [TK]D-Fender: its RISC |
| 17:22.00 | [TK]D-Fender | coppice: High-risc indeed ;) |
| 17:22.02 | s0ck | [TK]D-Fender: roaming agent = hot desking? |
| 17:22.09 | logyati | cos im learning from the beggining, i dont know nothing about asterisk |
| 17:22.40 | EvilGreen_ | logyati: start with the book, one was already mentioned |
| 17:22.48 | [TK]D-Fender | s0ck: if we're talking about having people log in from a phone for the purpose of call queues, this is already quite normal and documented. |
| 17:22.57 | EvilGreen_ | I'll find another name in a sec |
| 17:23.13 | logyati | evilgreen, could you give me the name of a nice book? |
| 17:23.18 | [TK]D-Fender | s0ck: If you mean that someone goes over and logs into a phone and their directed personal calls end up there... well thats a pile of dialplan logic. |
| 17:23.23 | [TK]D-Fender | ~book |
| 17:23.25 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:23.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 17:23.28 | s0ck | [TK]D-Fender: ok great, as long as it's not phone specific that's great |
| 17:23.44 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:24.49 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 17:25.11 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 17:25.22 | EvilGreen_ | logyati: I really like this one Building Telephony Systems With Asterisk http://www.packtpub.com/asterisk/book |
| 17:25.31 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 17:26.06 | jlewis | anyone ever seen / know why asterisk would respond to an AGI doing a GET VARIABLE FOO with "AGI Tx >> 200 result=0CLI>" |
| 17:26.27 | jlewis | when result should have been a dialed phone number |
| 17:26.54 | jlewis | same agi on other servers works properly |
| 17:27.10 | NOT_guru | question about support |
| 17:27.23 | NOT_guru | will a 24 port digium card take a X100M module? |
| 17:27.31 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 17:27.44 | Qwell[] | NOT_guru: No, I do not believe so |
| 17:27.53 | Qwell[] | the TDM800p does though |
| 17:28.13 | NOT_guru | bummer only reason I ask is I have a tdm400 and a buddy has the 24XX |
| 17:28.26 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 17:28.59 | ManxPower | Huh? I thought the X100M was an FXO module for tdm400P |
| 17:30.05 | NOT_guru | and I think I have a bad FXS module ( sorry meant FXS not FXO earlier ) |
| 17:31.45 | NOT_guru | so I meant S110M module |
| 17:31.55 | NOT_guru | I have some X100M's on the way |
| 17:32.02 | NOT_guru | thats why I was confused |
| 17:32.45 | NOT_guru | but I had heard the TDM8XX took single or quad ports daughter boards and was hoping the TDM20XX did as well |
| 17:32.52 | NOT_guru | then I could test it |
| 17:33.09 | NOT_guru | anyways this is why you don't buy on teh ebay |
| 17:34.02 | NOT_guru | and has the TDM4XX always registered with the zaptel driver as a "Freshmaker" or did I buy a clone ? |
| 17:34.29 | NOT_guru | the card is the digium blue PCB |
| 17:35.25 | NOT_guru | and Qwell I have a real retail TDM8XX for one of my offices and I love it |
| 17:36.01 | NOT_guru | would get one of those for myself at home... except I own a home therefore I am broke LOL |
| 17:38.25 | NOT_guru | Module 0: FAILED FXS (FCC) |
| 17:38.37 | NOT_guru | wctdm: probe of 0000:01:01.0 failed with error -5 |
| 17:39.07 | NOT_guru | would I get those errors if it was just misconfigured / unconfigured? |
| 17:39.25 | NOT_guru | or pretty much with an FXS only if the module was shot? |
| 17:39.58 | NOT_guru | seems the card pass' test |
| 17:40.00 | NOT_guru | Freshmaker passed register test |
| 17:40.17 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 17:41.50 | thekidrio | nah i get freshmaker on my direct from digium card |
| 17:42.01 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 17:42.04 | thekidrio | I think thats a reg utility |
| 17:42.25 | rob0 | NOT_guru: did you try genzaptelconf(8) to configure channels? |
| 17:42.32 | NOT_guru | yes sir |
| 17:42.42 | NOT_guru | here is some output if anyone cares to look |
| 17:42.43 | wunderkin | in addition to the digium cards, the manufacturer also makes mentos |
| 17:42.44 | NOT_guru | http://pastebin.ca/557539 |
| 17:43.36 | NOT_guru | and thank you for the feedbak kidrio |
| 17:43.41 | NOT_guru | on the Freshmaker bit |
| 17:44.01 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-300c4b6701c04981) |
| 17:44.19 | NOT_guru | but seems to be its JUST the FXS module thats bad |
| 17:44.26 | NOT_guru | err thats my hopes |
| 17:44.52 | NOT_guru | as I don't want to have to return the whole thing as I mentioned earlier that I have FXO nodules for it already ordered |
| 17:45.13 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:47.11 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:48.29 | rob0 | Looks like a bad FXS, yes. But I'm NOT_guru too. :) |
| 17:48.45 | NOT_guru | awsome rob0 |
| 17:48.52 | NOT_guru | well not awsome that its bad |
| 17:48.58 | wunderkin | call digium already |
| 17:49.06 | NOT_guru | but someone agrees that it APPREARS to be a module only issue |
| 17:49.25 | NOT_guru | FYI I have tried this module in port 1 and 4 just to make sure |
| 17:49.30 | NOT_guru | and same results |
| 17:50.02 | NOT_guru | wunderkid were you telling me to call digium? |
| 17:50.14 | NOT_guru | I didn't buy direct or a retail package |
| 17:50.24 | NOT_guru | I would not feel right calling for support |
| 17:50.30 | NOT_guru | thats why I am here =) |
| 17:50.40 | NOT_guru | I paid too little |
| 17:51.01 | NOT_guru | I dunno maybe they wouldn't mind |
| 17:51.05 | wunderkin | doesnt matter, they support the hardware not the sale |
| 17:51.11 | NOT_guru | sure |
| 17:51.13 | rob0 | I never had a module do that on me. I've had the whole card flake out with the interrupts, but it comes back after stopping asterisk, rmmod, modprobe, restart. |
| 17:51.24 | NOT_guru | well also before today I wasn't sure it was a real card or a fake |
| 17:51.30 | wunderkin | try to support at least, sometimes |
| 17:51.36 | NOT_guru | as I have always heard the card refered to as a freshmaker |
| 17:51.45 | NOT_guru | but mine was labeled err |
| 17:51.48 | NOT_guru | wild card |
| 17:51.59 | NOT_guru | refered to as wildcard |
| 17:52.03 | NOT_guru | mine is a freshmaker |
| 17:52.04 | rob0 | My dmesg is similar (less the module error of course.) |
| 17:52.10 | NOT_guru | indeed |
| 17:52.14 | *** join/#asterisk NirS (n=hamiflat@62-90-47-117.barak.net.il) |
| 17:52.23 | NOT_guru | did you do ANY special conf rob0? |
| 17:52.30 | NOT_guru | to make it work |
| 17:52.37 | NOT_guru | newer zaptel driver |
| 17:52.43 | thekidrio | NOT_guru, freshmaker and wild card are not the same thing |
| 17:53.01 | thekidrio | my wild card TDM400p still uses the freshmaker test |
| 17:53.09 | thekidrio | I think freshmaker is a registration test |
| 17:53.16 | NOT_guru | well my card |
| 17:53.23 | NOT_guru | has on the silkscreen "FRESHMAKER" |
| 17:53.26 | thekidrio | i had mine failing when it tried to register with octveqe soft echo |
| 17:53.39 | thekidrio | what do you see in an lspci? |
| 17:53.59 | NOT_guru | checking 1 sec |
| 17:55.22 | logyati | guys i have the same problem |
| 17:55.34 | logyati | i have a tdm400p with 1 fxo and 1 fxs |
| 17:55.44 | thekidrio | Should be something like 00:02:0 Communication Controller: Tiger Jet.... or something similar |
| 17:55.46 | NOT_guru | hrm |
| 17:55.56 | NOT_guru | well under its pci number |
| 17:55.57 | logyati | fxs give me FAILED FXS (FCC) in /var/log/messages |
| 17:55.58 | NOT_guru | I get |
| 17:55.59 | NOT_guru | 01:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 17:56.00 | tzanger | NOT_guru: your nick reminds me of the "I AM NOT PAUL AVERY" buttons from the zodiac movie |
| 17:56.12 | *** part/#asterisk karlm (n=karl@unaffiliated/karlhaines) |
| 17:56.21 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 17:56.29 | NOT_guru | well.... I'm not Paul Avery either LOL |
| 17:56.55 | logyati | tk-defender said that i should plug the power cable to the borad, and i did |
| 17:56.55 | NOT_guru | and I will have to rent / download zodiac and see what your talking about =P |
| 17:57.15 | logyati | so, we concluded that my module isnt working, i should RMA |
| 17:57.21 | NOT_guru | yes |
| 17:57.26 | rob0 | NOT_guru: http://pastebin.ca/559219 my dmesg |
| 17:57.27 | NOT_guru | and FXS REQUIRES power |
| 17:57.32 | NOT_guru | fxo does not |
| 17:57.35 | logyati | yes |
| 17:57.37 | NOT_guru | if I understand correctly |
| 17:57.37 | logyati | so, i did |
| 17:57.49 | logyati | but i still have fxs failed |
| 17:57.53 | NOT_guru | yes rob0 |
| 17:58.06 | logyati | someone here fixed this problem? |
| 17:59.22 | irule | has anybody ever done a replica of the voicemail application in dialplan? the idea is to avoid some bugs |
| 17:59.22 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 17:59.22 | ManxPower | WHAT you should do is CALL DIGIUM for support for your card. |
| 17:59.22 | NOT_guru | sorry phone rang brb |
| 18:00.02 | rob0 | My /etc/zaptel.conf has 4 lines: "echo -e 'fxoks=1,2\nfxsks=4\nloadzone=us\ndefaultzone=us'" |
| 18:00.38 | rob0 | Call Digium or the vendor, I'd ask to return it. |
| 18:00.46 | NOT_guru | yes |
| 18:00.46 | *** join/#asterisk Delta_Offset (n=abrahamc@201.226.130.55) |
| 18:00.50 | Delta_Offset | hello |
| 18:00.57 | NOT_guru | well I am just waiting for my FXO modules to show up |
| 18:01.02 | Delta_Offset | how do i fix an issue with firewall... |
| 18:01.10 | NOT_guru | so I can see if I am just returning the FXS module or the whole card |
| 18:01.12 | Delta_Offset | my sip extension gets an error 408 |
| 18:01.39 | NOT_guru | you see I am really hoping some portion of this works |
| 18:01.49 | NOT_guru | as I got the card and FXS module for $70 |
| 18:01.53 | NOT_guru | and he has another |
| 18:02.07 | NOT_guru | which is an incredible deal |
| 18:02.19 | NOT_guru | and as I stated before I own a house so I am broke |
| 18:02.23 | rob0 | Yes, that's a module-only price. |
| 18:02.27 | Katty | so |
| 18:02.29 | logyati | =/ bad luck |
| 18:02.34 | Delta_Offset | hello |
| 18:02.40 | logyati | can i buy another fxs separated? |
| 18:02.41 | NOT_guru | and I use the TDM8XX at work |
| 18:02.47 | Delta_Offset | my sip extension gets an error 408 |
| 18:02.49 | NOT_guru | and just wanted a card for home to "test" on |
| 18:03.32 | *** join/#asterisk WindBack (n=Administ@host9.190-30-210.telecom.net.ar) |
| 18:05.30 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:05.31 | Delta_Offset | mmmm |
| 18:05.42 | Delta_Offset | i already put the externip and the localnet on sip.conf |
| 18:05.56 | WindBack | I need a good softphone for windows that can handle sip, free or open source, and use low resources. Can anybody recomendme one?? |
| 18:05.59 | Delta_Offset | but for some reason it won't conect |
| 18:06.18 | WindBack | please |
| 18:06.18 | Nugget | "good" and "softphone" don't really fit together. |
| 18:06.24 | Delta_Offset | im getting an error 408 request timeout |
| 18:07.11 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 18:07.39 | WindBack | I read thath idefisk need at least 256 mb. Are there any who need lees?? |
| 18:07.51 | [TK]D-Fender | WindBack: Go try X-Lite |
| 18:08.32 | WindBack | [TK]D-Fender, Do you prefer x-lite than idefisk??? |
| 18:08.34 | *** join/#asterisk Tako-san (n=Tako-san@216.232.147.102) |
| 18:08.38 | NOT_guru | WindBack I have also used Kphone ( KDE ) and it worked well enough for testing purposes |
| 18:08.47 | [TK]D-Fender | WindBack: I'm mixed. Just go TRY IT. |
| 18:09.10 | [TK]D-Fender | NOT_guru: And you apparently missed the brig print of "Windows". |
| 18:09.19 | d00gster | Are there IAX2 or SIP providers out there I can use to do long distance pay per use without getting a DiD? |
| 18:09.23 | NOT_guru | yes I did =( sorry |
| 18:09.27 | NOT_guru | =) |
| 18:09.31 | Tako-san | Is there a way to check for a dialtone on a specific ZAP channel using the Asterisk CLI? |
| 18:09.36 | WindBack | [TK]D-Fender, I'm using X-lite, but I want a softphone more ligth than it |
| 18:09.51 | [TK]D-Fender | WindBack: Go write one yourself then. |
| 18:10.31 | NOT_guru | oh yas and Fender thanks for your help with my FXS Debugging yesterday appreciated |
| 18:11.43 | WindBack | [TK]D-Fender, ok, thank you. I'm only asking you because I'm not an expert |
| 18:12.53 | [TK]D-Fender | WindBack: You're just goping to have to get off your ass and try stuff out. Go to the WIKI and download everything on the soft-phone list and try for yourself. |
| 18:12.57 | [TK]D-Fender | NOT_guru: np |
| 18:18.05 | WindBack | [TK]D-Fender, OK |
| 18:18.16 | *** part/#asterisk WindBack (n=Administ@host9.190-30-210.telecom.net.ar) |
| 18:20.37 | thekidrio | hrmm, something is constantly trying to authenticate via manager.conf at the localhost ip, anywhere I can see what/who is trying to authenticate? |
| 18:24.14 | ManxPower | thekidrio: 127.0.0.1 ? |
| 18:24.31 | thekidrio | yeah |
| 18:24.36 | wunderkin | it's a virus! |
| 18:24.43 | thekidrio | heh |
| 18:25.00 | thekidrio | i am guessing i messed up a password somewhere in a conf |
| 18:25.05 | ManxPower | thekidrio: it is a process on the local machine. Flash Operator Panel or anything else that uses the manager interface |
| 18:25.07 | thekidrio | but everything seems to be working heh |
| 18:25.19 | thekidrio | ManxPower, yeah I just can't find out which one heh |
| 18:25.22 | thekidrio | they all seem to be working |
| 18:25.52 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 18:26.25 | ManxPower | thekidrio: are you using one of those sissyass GUIS? |
| 18:27.08 | thekidrio | haha well asterisk gui is installed |
| 18:27.13 | thekidrio | but I have not fired it up yet |
| 18:27.42 | thekidrio | only have a terminal and I hate terminal based webbrowsers heh |
| 18:27.45 | ManxPower | what makes you think it did not start automatically? |
| 18:27.45 | rob0 | sissyass.com is taken :( |
| 18:28.01 | ManxPower | rob0: how about sissyassgui.com |
| 18:28.17 | rob0 | No match for "SISSYASSGUI.COM". |
| 18:28.28 | thekidrio | ManxPower, my guess is the fact that its still sitting in a tarball |
| 18:28.30 | rob0 | (So try a lighter instead.) |
| 18:28.56 | ManxPower | that would be a good reason |
| 18:28.57 | thekidrio | i just removed gui installation :) |
| 18:29.13 | thekidrio | and still constant auth requests |
| 18:29.20 | ManxPower | you don't really need our help to find the rogue process |
| 18:29.27 | ManxPower | that's just linux stuff |
| 18:29.55 | thekidrio | indeed, just hoping that it was a common issue |
| 18:30.11 | ManxPower | thekidrio: it is not. |
| 18:33.44 | DarKnesS_WolF | what is the best billing systems for asterisk ? |
| 18:34.44 | DarKnesS_WolF | asterisk2billing , astbill , astpp or what ? |
| 18:34.49 | DarKnesS_WolF | sure billing and calling cards |
| 18:36.05 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 18:36.07 | ManxPower | DarKnesS_WolF: the all suck as far as I know. |
| 18:36.15 | Daejeo1 | ? iax2 port |
| 18:36.28 | Daejeo1 | ? port |
| 18:36.31 | *** join/#asterisk EmleyMoor (i=phil@62.49.246.92) |
| 18:36.33 | ManxPower | Any time you ask "What is the best...." The answer is "read the mailing list archives, you lazy bastard." |
| 18:36.39 | DarKnesS_WolF | ManxPower: okay any recommendation ? |
| 18:36.47 | DarKnesS_WolF | or the best of the worest :D? |
| 18:36.51 | DarKnesS_WolF | 4569 |
| 18:36.55 | ManxPower | Daejeo1: Perhaps if you used SENTENCES people would understand what the hell you are saying. |
| 18:36.55 | DarKnesS_WolF | udp Daejeo1 |
| 18:37.07 | Daejeo1 | it is blocked |
| 18:37.09 | EmleyMoor | I am getting the occasional drop of my PSTN line - using a TDM400P - with no obvious cause. What should I check? |
| 18:37.10 | ManxPower | DarKnesS_WolF: don't reward people like Daejeo1 |
| 18:37.15 | Daejeo1 | want to use any other port |
| 18:37.16 | ManxPower | Daejeo1: It sucks to be you |
| 18:37.38 | DarKnesS_WolF | Daejeo1: read in iax.conf http://voip-info.org |
| 18:37.44 | ManxPower | EmleyMoor: busydetect and callprogress will randomly disconnect your calls. They are NOT set by default. |
| 18:37.44 | DarKnesS_WolF | ManxPower: okay roger that sir ! |
| 18:37.55 | Tako-san | Hey, is there some way to check if a ZAP line has dialtone from the Asterisk CLI? |
| 18:38.00 | Daejeo1 | Manx : mind ur fucking language |
| 18:38.06 | Daejeo1 | ok? |
| 18:38.12 | EmleyMoor | ManxPower: OK - I wll check those first |
| 18:38.17 | ManxPower | Tako-san: Asterisk does not support checking for dialtone at all in any way |
| 18:38.24 | Tako-san | Ok, thanks. |
| 18:38.33 | DarKnesS_WolF | ManxPower: callprogress works with me " show progress , but the SIP no voice with PSTN ":-D " will debug that later i need it for AutoRedial fucntion with .call filesi 'm working on. |
| 18:38.43 | ManxPower | Well, I guess that does sort of qualify as a sentence. |
| 18:39.34 | ManxPower | If you type like a caveman or a 4 yr old, we will treat you like one. |
| 18:39.36 | EricL | Does anyone have any idea why the incoming callerID would come up correctly in the CDR and in comedian mail, but won't show up on the phones or follow me or email sent by comedian mail as a result of getting the voicemail? |
| 18:39.37 | EmleyMoor | ManxPower: Hmmm... I have busydetect enabled - should I up busycount or disable it? What are the pros and cons? |
| 18:39.51 | ManxPower | EmleyMoor: disable it |
| 18:39.54 | EmleyMoor | EricL: What kind of phones? |
| 18:40.13 | EricL | EmleyMoor: Cisco 7961G and Polycomm SIP Soundpoint 400s. |
| 18:40.31 | Tako-san | Is there some way to initiate a phone call on a specific channel using the CLI? |
| 18:40.31 | ManxPower | EmleyMoor: the only time you might want to consider busytdetect is if you have analog FXO ports and your telco does not provide far end disconnect supervision (almost all telcos in the USA support this feature) |
| 18:40.39 | EmleyMoor | EricL: Does it show up on SIP softphones? |
| 18:40.45 | EmleyMoor | ManxPower: I am in the UK |
| 18:40.52 | ManxPower | Tako-san: you can't do much of any thing with CALLS i the CLI |
| 18:40.59 | ManxPower | EmleyMoor: I cannot help you. |
| 18:41.04 | EricL | EmleyMoor: I don't know, I don't have any setup. |
| 18:41.09 | DarKnesS_WolF | EmleyMoor: ManxPower i have busy detect and busycount and never got disconnected from my PSTN |
| 18:41.11 | Tako-san | Ah, I see. I didn't know that. Thanks for the info. |
| 18:41.27 | ManxPower | Tako-san: Asterisk is a PBX, not a softphone. For the most part if you want to do call related things with Asterisk you need a phone. |
| 18:41.29 | DarKnesS_WolF | Tako-san: yes DIAL command |
| 18:41.50 | EricL | Its Asterisk 1.4.4 |
| 18:42.09 | Tako-san | manxpower: Yes, I realize the difference between the PBX and the softphone. I am sure unaware of all of the Asterisk CLI options. |
| 18:42.15 | ManxPower | DarKnesS_WolF: you poor innocent boy. DIAL on the CLI only works if you have ALSA/OSS installed, configured, the alsa/oss devel libs installed all before Asterisk. |
| 18:42.30 | DarKnesS_WolF | ManxPower: yes :-) i know |
| 18:42.32 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:42.32 | ManxPower | one also assumes that you have soundcard and speaker |
| 18:42.38 | DarKnesS_WolF | ManxPower: i always have this installed ;-) |
| 18:42.46 | DarKnesS_WolF | yes and mic too :-D |
| 18:42.49 | ManxPower | Tako-san: "help" is a good place to start 8-) |
| 18:42.49 | EmleyMoor | (that would include a supported headset of course) |
| 18:43.08 | Tako-san | manxpower: Yep. I have been slowly making my way through that. |
| 18:43.14 | ManxPower | DarKnesS_WolF: the first thing we do on a server is disable everything we can |
| 18:43.28 | ManxPower | Tako-san: The Book is also helpful (it is specific to 1.2) |
| 18:43.29 | ManxPower | ~thebook |
| 18:43.40 | 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 |
| 18:43.40 | DarKnesS_WolF | ManxPower: my servers dosn't have that much of load.. |
| 18:43.40 | Tako-san | manxpower: Just got tired of wading through it all and figured a quick question on IRC would be worth saving a bunch of time. 8-) |
| 18:43.40 | DarKnesS_WolF | be back later. |
| 18:43.54 | Tako-san | manxpower: Yep, got it all printed out and mostly read. |
| 18:43.56 | ManxPower | DarKnesS_WolF: neither do mine, but having 5,000 kernel modules loaded is something I find distracting |
| 18:44.32 | EmleyMoor | If it hadn't been for the book, I would probably never have tried asterisk - now I enthuse about it constantly |
| 18:45.13 | EmleyMoor | Any of you lot going to Edinburgh on Saturday? |
| 18:45.15 | EricL | Any ideas on the CallerID problem? |
| 18:45.50 | EmleyMoor | EricL: Sadly, I have no SIP hardphones so cannot comment specifically |
| 18:46.31 | *** join/#asterisk mtaht4 (n=m@55-85-62-200.enitel.net.ni) |
| 18:46.50 | EricL | The other problem I am having is that internal calls give the proper SIP information but the caller ID reads the destination extension. |
| 18:47.02 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 18:47.33 | EricL | ie the name of the incoming caller gets displayed on the phone, but not the phone number. |
| 18:48.27 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:48.30 | EmleyMoor | EricL: What happens when there is no CALLERID(name)? |
| 18:48.37 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:48.47 | EricL | EmleyMoor: What do you mean? |
| 18:49.10 | EmleyMoor | EricL: If there's just a CALLERID(num), does the phone display it? |
| 18:49.39 | EricL | EmleyMoor:The only phone number/name combo I have ever seen the phone display properly is from CDW. |
| 18:50.09 | EricL | Other than that, it usually just displays the CALLERID(name) and then my extension for number. |
| 18:50.14 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 18:50.55 | EmleyMoor | EricL: Hmmm... well, without one of the phones you mentioned, I can't comment |
| 18:51.42 | EricL | It used to work just fine in Asterisk 1.2.12. Then when I upgraded to 1.4.4, it stopped working correctly and I am at the setup I am at now. |
| 18:51.57 | EricL | Something must have changed in what's required to pass along. |
| 18:54.20 | *** join/#asterisk ikey (i=ikey@219.64.78.253) |
| 18:55.42 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 18:59.50 | *** join/#asterisk dennisharrison (n=dennisha@68-114-124-171.dhcp.slid.la.charter.com) |
| 19:00.25 | ikey | hi |
| 19:01.25 | *** part/#asterisk mtaht4 (n=m@55-85-62-200.enitel.net.ni) |
| 19:04.30 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 19:12.24 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 19:19.17 | Mercestes | <PROTECTED> |
| 19:23.33 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 19:33.32 | *** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243) |
| 19:33.46 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 19:34.11 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 19:35.16 | sahafeez | question - anyone every have an issue with a polycom phone just random turning the speaker on and dialing... |
| 19:35.57 | blitzrage | sounds like you have ghosts |
| 19:36.20 | [TK]D-Fender | unload chan_gremlins.so |
| 19:37.43 | sahafeez | hum. it is one phone. i have formated the flash, etc. it just pickups and dials then hangs up. |
| 19:38.54 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 19:40.13 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 19:40.57 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 19:43.53 | mocker | sahafeez: I don't think that's a feature of Asterisk. :) |
| 19:44.14 | mocker | load polycom_prankcall.so |
| 19:47.33 | sahafeez | hum. |
| 19:47.50 | sahafeez | well i am at a loss. it is very strange. sun spots maybe |
| 19:48.21 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:48.31 | thekidrio | haha sunspots |
| 19:48.36 | thekidrio | thats an old excuse :) |
| 19:49.00 | sahafeez | well unless someone has a better idea i think this phone is going in the trash |
| 19:49.05 | Waverly360 | sahafeez: Do you have dhcp setup to autoconfigure the phone? |
| 19:49.21 | sahafeez | yes. it is no different from the other 12 i have. |
| 19:49.30 | [TK]D-Fender | sahafeez: What BR/SIP is it using? What model? |
| 19:49.58 | Waverly360 | sahafeez: check the mac address and find the corresponding config file on your server. Make sure it looks like the others. |
| 19:50.21 | sahafeez | its a 601. any i have checked it 6 ways to sideways. |
| 19:50.53 | sahafeez | i copied the file from another 601 and just did a s/g for the ext # |
| 19:50.53 | rob0 | s/trash/mail to rob0/ :) |
| 19:51.35 | sahafeez | sip ver is 2.0.2.0076 |
| 19:51.38 | [TK]D-Fender | sahafeez: s/g? |
| 19:51.50 | sahafeez | :1,$s/4201/4224/g |
| 19:51.52 | sahafeez | vi |
| 19:52.13 | [TK]D-Fender | sahafeez: Ok, and if you just rename the mac file for the other its just acts flakey regardless? |
| 19:52.37 | sahafeez | yah. just this phone. sunspots or a ghost |
| 19:53.16 | sahafeez | i will be sitting at the desk. it goes off hook and dials something. it is gen the last few numbers dialed. |
| 19:53.19 | Waverly360 | sahafeez: well, you don't have another dhcp server running somewhere do you? Maybe another pbx with that phone setup in it? |
| 19:53.25 | sahafeez | nope. |
| 19:53.35 | [TK]D-Fender | sahafeez: Ok, that sucker is just fried. RMA it. |
| 19:53.37 | Waverly360 | sahafeez: What numbers are being dialed? |
| 19:54.03 | sahafeez | the last 2 entries from the recent list |
| 19:54.03 | Waverly360 | sahafeez: nevermind...missed it :) |
| 19:54.48 | Waverly360 | sahafeez: Yeah, I think I agree with [TK]D-Fender...I've never heard of one doing that...not even sure you can program that 'functionality' :) |
| 19:55.09 | sahafeez | hum, i could call a priest |
| 19:55.35 | *** join/#asterisk Tako-san (n=Tako-san@24.68.129.29) |
| 19:55.40 | Waverly360 | sahafeez: You could... |
| 19:56.27 | *** join/#asterisk guillote_GNU (n=guillote@host176.201-252-205.telecom.net.ar) |
| 19:56.27 | sahafeez | out demon. leave this phone. wonder if the Holly H2O would fry it |
| 19:57.39 | Waverly360 | If I remove entries in my sip.conf file, and reload asterisk, any phone still logged in will stay that way until I restart asterisk..right? |
| 19:58.54 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 20:00.24 | ^majik^ | hi all. I was considering using the iLBC codec with an ATA.. any suggestions for an IAX2 ATA? |
| 20:02.01 | *** join/#asterisk saftsack (n=saftsack@pD9E06EA7.dip.t-dialin.net) |
| 20:05.05 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:07.37 | Mercestes | :P @ [TK]D-fender |
| 20:08.30 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 20:08.53 | coolbeans | Hi all. Which is better, g729a or gsm as a codec between my asterisk 1.2 box and a provider such as Vitelity? |
| 20:09.16 | [TK]D-Fender | ^majik^: Nothing decent uses ilbc, and pretty much the same goes for IAX2. |
| 20:09.38 | [TK]D-Fender | coolbeans: there is a mixed opinion on that. |
| 20:10.13 | [TK]D-Fender | coolbeans: most would say that G729 is better for basic voice, but if there is MoH playing, etc that it sounds pretty bad, as though underwater |
| 20:10.55 | *** join/#asterisk grey (n=grey@bas3-sudbury98-1168048322.dsl.bell.ca) |
| 20:11.01 | [TK]D-Fender | coolbeans: GSm is a bit grainy, but consistant. GSM is a free codec and is low in transcode load, G729 is not. |
| 20:11.09 | coolbeans | Thanks, [TK]D-Fender. I'm trying to fight a DTMF issue. I keep getting double digits in some cases and I'm assuming it's bandwidth related. |
| 20:11.56 | coolbeans | I have a handful of people successfully faxing through asterisk over the Internet and I know if I go gsm or g729 I'll essentially kill that ability. |
| 20:12.12 | grey | I'm tinkering with Asterisk for the first time, currently just messing around with voice menus, when I use Background(enter-ext-of-person) it doesn't delay for the 10 seconds described in the doc's, it just keeps looping over and over (due to the goto line beneath it that goes back to the start of the menu) |
| 20:12.15 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 20:12.24 | bapril | g729 is also know bad an DTMF |
| 20:12.42 | [TK]D-Fender | coolbeans: Codec has NO impact on DTMF |
| 20:12.43 | coolbeans | Right now it's just ulaw and I'm having some DTMF issues. |
| 20:14.21 | Mercestes | grey, You either have to waitexten or set a digit timeout isntead, or play silence/10 or something. |
| 20:14.21 | [TK]D-Fender | coolbeans: You should be using rfc2833 |
| 20:14.21 | coolbeans | [TK]D-Fender: Is't that the sip channel default? |
| 20:14.22 | grey | Mercestes: it wouldn't pick up the dialed extension after that would it? |
| 20:14.22 | Mercestes | grey, I suggest not having that goto line, setting timeout(digit) and just trailing off at the end of yoru IVR, and then specifying code under exten => t,1 to handle yoru time out event' |
| 20:14.22 | Mercestes | grey: No, it won't. |
| 20:14.22 | [TK]D-Fender | coolbeans: Never assume some sort of magical "default". SET THINGS YOURSELF |
| 20:14.22 | coolbeans | lol, [TK]D-Fender, I'll try and see if it helps. Hey, I appreciate your help :) |
| 20:14.22 | mocker | Anyone have any recommendations on dual-FXS adapaters? |
| 20:14.27 | mocker | I know the PAP2-NA was popular awhile back.. |
| 20:14.31 | [TK]D-Fender | mocker: SPA-2102 |
| 20:14.37 | grey | hey removing the goto did it |
| 20:14.45 | Mercestes | There ya go |
| 20:14.49 | grey | thanks :) |
| 20:14.52 | Mercestes | np |
| 20:14.57 | [TK]D-Fender | PAP2 cost a few bucks less but has a weaker CPU and doesn't support T.38, doesn't have the router option, etc. |
| 20:15.01 | mocker | [TK]D-Fender: You have to pick the most expensive. ;) |
| 20:15.09 | [TK]D-Fender | I highly recommend spending the extra 10$ |
| 20:15.39 | [TK]D-Fender | mocker: 2102 +/- $70 USd. If I wanted to go deluxe it'd be a Mediatrix that would be at least 50% more ;) |
| 20:18.11 | mocker | [TK]D-Fender: Alright, I'll try to get the extra money in the budget. :) |
| 20:19.51 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:21.58 | *** join/#asterisk mtaht4 (n=m@55-85-62-200.enitel.net.ni) |
| 20:22.21 | ^majik^ | [TK]D-Fender: what codec would you recommend? our VoIP customers say its skippy sometimes.. we're currently using ulaw |
| 20:22.51 | [TK]D-Fender | ^majik^: are you actually low on bandwidth for the number of channels that you use? |
| 20:23.14 | *** join/#asterisk RageMax (n=RageMax@static-69-95-128-160.pit.choiceone.net) |
| 20:23.57 | *** join/#asterisk pejo_ (n=pete-joh@triton.dsv.su.se) |
| 20:24.46 | ^majik^ | [TK]D-Fender: for local calls, bandwidth shouldn't be a problem. I'm thinking that since we're capping customers at 64 - 128K up (we're the ISP too), that we need to use a lower bitrate codec |
| 20:25.56 | [TK]D-Fender | ^majik^: OUCH |
| 20:26.07 | ^majik^ | yeah ;) |
| 20:26.23 | [TK]D-Fender | ^majik^: that is nowhere enough to justify ULAW, and makes VoIP look bad altogether. |
| 20:26.57 | ^majik^ | right :) looks like we'll be ditching the IAXy's we have |
| 20:27.05 | [TK]D-Fender | ^majik^: You need the most compression you can get. That says G.729 (I'd say G.723.... but you don't wanna go there), and change your frame size from 20 to 40ms |
| 20:28.10 | ^majik^ | [TK]D-Fender: could we buy and use multiple G729 licenses (ie, per VoIP customer) ? |
| 20:28.35 | *** join/#asterisk aptura (n=sales@d64-180-62-58.bchsia.telus.net) |
| 20:28.40 | [TK]D-Fender | ^majik^: this part you'll have to research based on your overall deployment... |
| 20:28.47 | [TK]D-Fender | but I'm off home... BBIAB |
| 20:28.50 | sergee | Are there any europian users here? (DE, NL preferable :) ) |
| 20:28.53 | aptura | see ya |
| 20:29.04 | sergee | Are there any european users here? (DE, NL preferable :) ) |
| 20:30.04 | aptura | Anyone here tested or know of a good reputable blue tooth enables headset that uses a usb reciver? to noisy in resteraunt establishments to use mic/speaker for wifi voip. |
| 20:31.24 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 20:33.05 | *** join/#asterisk y7n (n=na@81-179-100-200.dsl.pipex.com) |
| 20:33.16 | m4rkl4r | I have done quite a bit of research and have so far been unable to find satisfactory answers: How can one detect that a blind REFER-based call transfer has ocurred? |
| 20:34.17 | y7n | I want to run the command "core show channels" in the command line interface every X number seconds, any ideas? |
| 20:35.15 | *** part/#asterisk EvilGreen_ (n=Miranda@ppp85-141-152-123.pppoe.mtu-net.ru) |
| 20:35.52 | m4rkl4r | I.e. if a calls b and be transfers to c, the CDR makes it look as if a called c directly |
| 20:37.48 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 20:37.51 | Tagor | Hi |
| 20:37.56 | Tagor | Can someone tell me what's wrong with this: |
| 20:37.58 | Tagor | exten => 1,11,Background(${BGSBGS}) |
| 20:38.05 | Tagor | I get the BGSBGS variable with AGI |
| 20:38.36 | Tagor | For some reason it doesn't play anything |
| 20:38.46 | Tagor | -- Executing BackGround("SIP/212-ddde", "") in new stack |
| 20:39.11 | ^majik^ | y7n: I'm not sure if you can pass commands to the asterisk CLI from, say, bash, but if its possible, you can write a shell script and then run it every 5 seconds or so from crontab |
| 20:40.05 | y7n | ok, thanks :) |
| 20:40.18 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-feb61f267f53edc3) |
| 20:40.28 | rob0 | cron can't do less than one minute increments. Just sleep(1) and loop. |
| 20:42.03 | Tagor | Anybody? :( |
| 20:42.11 | ^majik^ | yeah, sleep(1) and loop, and just run it in the background (or foreground, depending on how you're viewing the output) |
| 20:42.28 | y7n | thanks |
| 20:43.00 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 20:43.08 | sergee | i wonder, which country has most liberal telecom laws: no lincensing (can sell voip services without any special allowence from goverment), no CALEA etc... ? |
| 20:44.08 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 20:44.12 | *** part/#asterisk tparcina (n=tparcina@89-172-116-117.adsl.net.t-com.hr) |
| 20:44.22 | *** join/#asterisk juancri (n=juancri@65.111.169.15) |
| 20:44.25 | sergee | where i can place server and start selling voip services without any problems, just paying taxes annualy? |
| 20:45.11 | lee_is_me | hey all. I have an installation question... |
| 20:45.47 | lee_is_me | assuming a small office that has an existing small network. Is it better to use existing network typically or run new network along side existing network? |
| 20:46.08 | thekidrio | better to run new for pure QoS |
| 20:46.26 | thekidrio | but if its really small and/or they have the bandwidth its not much difference |
| 20:46.30 | thekidrio | in my exp. |
| 20:46.55 | thekidrio | I did not notice the quality drop on a shared network until maybe 20 or so on near constant dial |
| 20:47.04 | lee_is_me | I just did one where we ran all new network for them. I was a bit curious what other would have done.l |
| 20:47.42 | lee_is_me | what about sudden congestion from some employee downloading large files? |
| 20:47.52 | sergee | lee_is_me: it all depends on how much money do you have, and how much of them you are ready to spend :) |
| 20:48.12 | lee_is_me | Well, small offices typically are budget conscious |
| 20:48.25 | thekidrio | lee_is_me, i would have run the new network if they plan to grow |
| 20:49.00 | thekidrio | and the bandwidth issue, about employee leeching it with a big download, that is a mangement issue as well |
| 20:49.03 | sergee | new neworks are the most expensive solutions |
| 20:49.05 | thekidrio | management even |
| 20:49.34 | lee_is_me | I guess like anything it's a case by case basis... |
| 20:49.36 | thekidrio | sergee, most expensive in the short term, but rebuilding a large network later on is more expensive thenplanning ahead |
| 20:50.01 | lee_is_me | personally, I just like the idea of insulating voice from data.... |
| 20:50.16 | lee_is_me | separated and conquered. |
| 20:50.21 | sergee | thekidrio: not exactly, |
| 20:50.21 | lee_is_me | (hopefull) |
| 20:50.23 | thekidrio | lee_is_me, yeah set up a few cases, example (Ideal SoHo 0-10 users, Cost Balance SoHo 0-10 users, and Cheap SoHo 0-10) |
| 20:50.43 | thekidrio | sergee, well it does depend on how fast/large the network grows |
| 20:50.57 | thekidrio | but I have never found it cheaper to rebuild a network then to extend and existing one |
| 20:51.26 | *** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42) |
| 20:51.30 | sergee | thekidrio: maintaining 2 networks is more expensive in mid/long term too, so 2 networks are always more expensive then 1 |
| 20:51.41 | lee_is_me | Of course, a large company with an already embedded infrastructure might change things... |
| 20:51.58 | lee_is_me | For now, I'm still cutting my teeth on small installs |
| 20:52.03 | Greek-Boy | any call termination operators here? |
| 20:52.33 | thekidrio | sergee, hrmm i have not founnd that to be true at all. Once the infrastructure is up 1 network/10 networks all about the same for me to manage |
| 20:54.45 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 20:54.48 | sergee | bringing infrastructure up = building/maintaining |
| 20:55.36 | thekidrio | ahh I see :) I separate building and maint |
| 20:56.02 | thekidrio | as building is normally a larger up front cost |
| 20:56.08 | thekidrio | and maint is a monthly/yearly cost |
| 20:56.35 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.65) |
| 20:56.37 | zeeesh | hi |
| 20:56.41 | sergee | ofcause, if there are 10 sip phones, and you don't expect any more in 5 years, then put some cable, cheap chinese switch and you'll even don't need to maintain that network at all |
| 20:56.57 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:57.03 | *** join/#asterisk mutilator (n=WebChat@the.drinkproject.com) |
| 20:57.07 | mutilator | anyone wanna but a lightly used te110p? |
| 20:57.23 | thekidrio | hehe yeah with small networks everything gets changed really |
| 20:57.47 | thekidrio | you don't require the QoS of a higher end router/firewall |
| 20:58.00 | thekidrio | there is more aggregate bandwidth on small networks too |
| 20:58.03 | sergee | thekidrio: but if you have office building with 200-300 phones, building separate network would be real PITA, much better to use QoS or, at least, VLANs |
| 20:58.17 | ^majik^ | [TK]D-Fender: hey, wb |
| 20:58.38 | *** part/#asterisk putnopvut (i=putnopvu@nat/digium/x-300c4b6701c04981) |
| 20:58.48 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 20:59.12 | sergee | thekidrio: hey! may be you know something about voip licensing and liberal countries? :)) |
| 20:59.28 | thekidrio | iraq is very liberal |
| 20:59.29 | thekidrio | about voip |
| 20:59.35 | thekidrio | china too |
| 20:59.52 | thekidrio | gotta look for people who do not respect other countries laws so you can circumvent them |
| 21:00.01 | ^majik^ | [TK]D-Fender: now you see why I was considering iLBC... do you think G.729 is the best for our situation? (btw, we're a WISP) |
| 21:00.28 | ManxPower | sergee: nobody is going to help you get the pirated G729 and G723.1 codec installed for Asterisk |
| 21:00.30 | [TK]D-Fender | ^majik^, you won't find any decent HARDWARE for ilbc, thats the problem. |
| 21:00.47 | thekidrio | ManxPower, hah is that what he is trying for? |
| 21:01.14 | thekidrio | yeah i don't want to encourage pirating of others IP |
| 21:01.17 | sergee | ManxPower: hmmm, very interesting, did i say something about codecs? |
| 21:01.20 | ^majik^ | [TK]D-Fender: right. I found a few, but I dunno if they'd be any good. (E-tech was one brand) |
| 21:01.37 | [TK]D-Fender | ^majik^, GSM is lighter, G729 is COMMON and lighter than either |
| 21:01.47 | sergee | ManxPower: there is no "pirated" codecs, algorythms can't be copyrighted, i'm european :-P |
| 21:02.17 | sergee | ManxPower: oh sorry, your comment wasn't for me :) |
| 21:03.40 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 21:04.06 | ^majik^ | [TK]D-Fender: ok. about the hardware, should I look for either SIP or IAX2 in an ATA? know of a good ATA that does G.729? |
| 21:04.18 | ManxPower | sergee: Talk to the ITU about that. |
| 21:04.22 | nowork | hi, I asked once , but forgot, any good suggestion for sip provider? and I need toll free as well. thanks |
| 21:04.37 | ManxPower | and it is SOFTWARE that can't be patended, not codecs |
| 21:04.46 | [TK]D-Fender | ^majik^, Linksys |
| 21:05.05 | ^majik^ | [TK]D-Fender: ok :) |
| 21:05.16 | nowork | why people said IAX better than SIP, is that true? |
| 21:05.42 | sergee | ManxPower: ITU doesn't issue national laws, so i can talk with ITU, Marsians, God etc - it won't change laws :) and laws in europe doesn't support patents for algorythms |
| 21:07.00 | ManxPower | sergee: cite your source. |
| 21:08.19 | ManxPower | I just have trouble believing that if you can't patent a codec in Europe then why are there not bunches of companies there making royalty free MP3 players, DVD players, IP phones, and video conferencing software. |
| 21:08.30 | ManxPower | I guess they could just be being nice. |
| 21:08.39 | sergee | thekidrio: seems like nobody knows laws :) or they just don't care :) i'm operating small voip service, but host country requires licensing for voip (license costs around 1-2M$) so i want to legalise my small business by moving it to some country which is more liberal then current host |
| 21:08.40 | Mercestes | Ujnless the codecs were written in europe and/or are owned by a european national, then it's not Europes place to decide whether or not that codec can be patented. It is theft, none-the-less. |
| 21:08.47 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:09.07 | *** part/#asterisk ManxPower (n=manxpowe@135.sub-75-201-87.myvzw.com) |
| 21:09.15 | *** join/#asterisk ManxPower (n=manxpowe@135.sub-75-201-87.myvzw.com) |
| 21:09.23 | sergee | ManxPower: You wouldn't believe! there are HUGE number of companies who produce TONS of royalty-free hardware |
| 21:09.38 | Mercestes | Name one. |
| 21:09.43 | Mercestes | actualy. |
| 21:09.45 | Mercestes | Better yet. |
| 21:09.49 | Mercestes | ship me some royalty free hardware. |
| 21:09.50 | *** join/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 21:09.52 | sergee | ManxPower: but there are 2 points |
| 21:10.06 | sergee | ManxPower: 1. i suppose you won't be able to buy it in the US |
| 21:10.29 | rene- | hey, how should i debug dropped zap-to-sip calls |
| 21:10.41 | rene- | i think manxpower once had an answer for that |
| 21:10.48 | sergee | ManxPower: because if anybody will try to sell such harware - he will be sued by patent's holder according to US laws |
| 21:10.49 | rene- | ~poke ManxPower |
| 21:11.02 | jbot | ACTION cuts down a small tree, sneaks up behind ManxPower, pokes ManxPower repeatedly, hilarity ensues. |
| 21:11.03 | [TK]D-Fender | rene-, crack up the verbose and pastebin the call |
| 21:11.35 | Mercestes | sergee, you stink of troll. |
| 21:11.39 | rene- | i think Manx once has a comment on what was going on when a call dropped and the callee hears strong static noise |
| 21:12.04 | sergee | 2. "royalty-free" hardware, like "royaltee-non-free" hardware manufactured in china. |
| 21:12.20 | sergee | Mercestes: OpenVox |
| 21:13.12 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 21:13.22 | sergee | Mercestes: wash your mouth every day, then smell will gone |
| 21:14.43 | sahafeez | anyway to temp block an ip phone from the cmdpromt |
| 21:14.50 | sergee | Mercestes: i'll ship you some royaltee free hardware, on a prepaid basis, name equipment, i'll find royaltee free analog and price for you |
| 21:15.15 | nowork | hi, I asked once , but forgot, any good suggestion for sip provider? and I need toll free as well. thanks |
| 21:15.51 | Mercestes | sahafeez, iptables |
| 21:16.12 | sahafeez | hum, not userfull. |
| 21:16.21 | sahafeez | do not run iptables |
| 21:16.24 | Mercestes | sergee, sounds like your selling genuine imitation bad apples. Guaranteed free for your peace of mind for only $9.95. |
| 21:16.32 | Mercestes | sahafeez, Pf then. |
| 21:16.50 | sahafeez | the phone is behind the firewall |
| 21:16.52 | thekidrio | haha Genuine Imitatin Bad Apples haha |
| 21:16.59 | Mercestes | sahafeez, Then firewall it |
| 21:17.05 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-6c375a271125cd76) |
| 21:17.14 | sahafeez | forget it. |
| 21:17.21 | sergee | Mercestes: sounds like you can't understand that US laws works only in US |
| 21:17.43 | sergee | ManxPower: here goes cite: http://www.advogato.org/article/151.html |
| 21:18.17 | sergee | ManxPower: "Currently the EU recognizes that computer software algorithms are statements of pure mathematics and are not machines that can be patented." |
| 21:18.46 | sergee | ManxPower: small hint: there are plenty of countries except US & EU |
| 21:19.18 | thekidrio | sergee, well yeah austrailia :P |
| 21:19.46 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:20.02 | *** join/#asterisk theacolyte (n=theacoly@unaffiliated/theacolyte) |
| 21:20.04 | Mercestes | sergee, Arguing on IRC is kinda like running in the special olympics. Yea, sure you might win, but in the end....you're still retarded. |
| 21:20.08 | sergee | Australia nice country, i was there once in 2000 before Olympic games :) NSW tech university - rocks :) |
| 21:20.11 | ManxPower | AU is just a lapdog for USA copyright law |
| 21:20.12 | theacolyte | What's the largest * deployment out there? |
| 21:20.19 | Mercestes | sergee: And that doesn't change the fact that you will find no help for your stolen codecs here. |
| 21:20.35 | Mercestes | sergee, Go ask your government for help, maybe they will help you. |
| 21:20.53 | thekidrio | Mercestes, his gov't wants the 1-2 mil fee |
| 21:20.59 | _VoiceMeUp_COM | a |
| 21:21.09 | Mercestes | thekidrio, Ah, must be that royalty free hardware they give away fro free. |
| 21:21.41 | Mercestes | theacolyte, Someone trying to use Asterisk for a commercial phone switch and trying to run a CLEC off of it using only one server. Why do you ask? |
| 21:21.55 | theacolyte | Curiousity more than anything |
| 21:22.08 | *** part/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 21:22.22 | thekidrio | theacolyte, hard to find that out actually |
| 21:22.31 | thekidrio | theacolyte, you can look at IAX termination minutes |
| 21:22.38 | *** join/#asterisk tabularasa (n=tabulara@64.28.197.122) |
| 21:22.45 | thekidrio | theacolyte, but that does not take in to account the asterisk boxes using sip |
| 21:22.51 | Mercestes | theacolyte, Are you looking at one server or a collection of servers? |
| 21:22.54 | theacolyte | right |
| 21:23.14 | Tagor | Hi |
| 21:23.29 | thekidrio | I read Mark Spencers article in Telephony and even he had not good knowledge of where the largest * install was |
| 21:23.34 | Tagor | I have a PHP AGI script and want to pass some variables from asterisk to the PHP script, how can I do that? |
| 21:23.37 | theacolyte | Collection... the largest deployment period... mainly just trying to see how it stacks up to large CCM deployments |
| 21:23.38 | ManxPower | http://www.soft-switch.org/patents/ar01s06.html |
| 21:23.40 | thekidrio | he said the largest he know of was in South Amrerica I think |
| 21:23.44 | Tagor | Something like AGI(test.agi?test=test) doesn't work |
| 21:23.47 | ManxPower | that is a suprizingly useful document |
| 21:23.58 | lee_is_me | Anyone know how many <server/> nodes can be defined within sip.cfg? |
| 21:24.02 | sergee | Mercestes: once more, fore some clever irc users: i don't need hel in installing codecs, i can write codecs myself if i need to (BTW i'm using "pirated" [in US terms] g729 codec, which i wrote from ITU.T reference codec) any more questions darling? |
| 21:24.29 | ManxPower | sergee: see the URL I posted. I'm looking for better ones. |
| 21:25.27 | tabularasa | Is there a good doc comparing * to CCM for features, etc ? |
| 21:25.40 | lee_is_me | Tagor: why doesn't it work? |
| 21:26.07 | Tagor | lee_is_me >> Then it doesn't execute the script |
| 21:26.13 | lee_is_me | I haven't done standard AGI in a while, but I don't think its value/pair like that. |
| 21:26.35 | lee_is_me | unless you are trying to do FastAGI (over a socket) |
| 21:26.45 | lee_is_me | which are you trying to do? |
| 21:26.49 | ManxPower | and regardless of if codecs or math or sofware CAN be patented in Europe, they ARE patented in Europe. If you don't like that then you will have to get the patent invalidated before you can legally use it. |
| 21:26.57 | *** part/#asterisk tabularasa (n=tabulara@64.28.197.122) |
| 21:27.25 | thekidrio | we just had one of our patents invalidated |
| 21:27.28 | thekidrio | by IBM |
| 21:27.31 | thekidrio | bastiches |
| 21:27.42 | Tagor | lee_is_me >> What I actually try to do is send some variables to the PHP script (like you do in a browser: test.php?test=test) and then send back some commands |
| 21:27.59 | thekidrio | we have x amount of days to say ours is significantly different then theirs or we lose the patent |
| 21:28.14 | sergee | ManxPower: well i can spend few minutes more googling for you, but i think, that i know situation a little bit better then you since i'm living on the other side of atlantic/pacific so you'd better to believe me. You can keep your opinion if you wish, it won't change reality :) |
| 21:29.30 | lee_is_me | Tagor: try place the variable like that in quoted like this: AGI(test.agi,"param1=1"|"param2=2") |
| 21:30.09 | lee_is_me | Tagor: I think your "=" may effect asterisk parser... |
| 21:30.37 | Tagor | lee_is_me >> ok, thanks, let me try :) |
| 21:30.44 | Strom_M | sergee: not that you're going to listen to me, but one of the patent holders for the codec is France Telecom |
| 21:31.18 | sergee | ManxPower: you refer to 3 different terms: software, codecs and algorythms. Look in the history i didn't say anything about software or codecs (which is particular implementation) i was speaking about ALGORYTHMS |
| 21:31.18 | ManxPower | I guess I could call up SIPPRO and ask them why only people in the USA have to pay for their stuff |
| 21:31.38 | lee_is_me | LOL |
| 21:32.01 | ManxPower | I reported the pirated the original URL of the pirated codecs to them |
| 21:33.36 | lee_is_me | I think if you know you're supposed to pay for the use of a product, it doesn't matter what local laws say about patents, etc. |
| 21:34.04 | sergee | ManxPower: so if i will use stolen Digium G.729 codec most likely it would be considered as pirated (i'm not sure though). HOWEVER, if i will implement G.729 myself or will take some free version (derivered from itu.t reference code, or from IPP) it WON'T be considered as piracy, because it's a FREE IMPLEMENTATION of ALGORYTHM, and ALGORYTHMS CAN'T BE PATENTED, that's my last words for this flame |
| 21:34.08 | lee_is_me | Now, about my my polycom config question... |
| 21:34.28 | Qwell[] | ManxPower: huh? English.. |
| 21:34.38 | lee_is_me | [TK]D-Fender: are you up for a polycom sip.cfg question? |
| 21:35.25 | [hC] | lee_is_me: just ask, its likely that someone can give you some help. |
| 21:35.34 | Qwell[] | ManxPower: you pirated the what now? |
| 21:35.43 | lee_is_me | [hC]: I did eariler |
| 21:35.44 | Qwell[] | That sentence made little sense ;/ |
| 21:35.52 | lee_is_me | let me bring it up again... |
| 21:36.00 | lee_is_me | Anyone know how many <server/> nodes can be defined within sip.cfg? |
| 21:36.15 | lee_is_me | I'm assuming depending on the phone, but... |
| 21:36.28 | [TK]D-Fender | lee_is_me, just ask |
| 21:36.37 | lee_is_me | Anyone know how many <server/> nodes can be defined within sip.cfg? |
| 21:36.41 | lee_is_me | LOL |
| 21:36.41 | [hC] | sergee: i am pretty sure you cannot code your own g729 implementation and give it away for commercial use for free. |
| 21:36.42 | ManxPower | sent |
| 21:37.09 | ManxPower | Qwell[]: I reported the URL of the pirated G729 and G723.1 codecs to SIPpro |
| 21:37.13 | [TK]D-Fender | lee_is_me, I'm quite sure it's just 1 |
| 21:37.15 | Qwell[] | ahh |
| 21:37.18 | ManxPower | like a year ago |
| 21:37.50 | sergee | [hC]: i can't give WHERE? in the Ukraine? in the China? in the Iraq? in the Thailand? on the North Pole? it all depends on NATIONAL laws |
| 21:37.51 | lee_is_me | [TK]D-Fender: OK, I thought it coincided with the 2 two servers that can be configured in the web interface... |
| 21:37.52 | [hC] | lee_is_me: yeah im pretty sure its one, you can have multiple specified proxies inside of it using the name.1.blah and name.2.blah tags (not sure the actual names) |
| 21:38.09 | lee_is_me | ah, OK I think I get it now |
| 21:38.19 | [TK]D-Fender | lee_is_me, pay attention to the CONTENT of the tag. 1 instance with multiple VALUES in there |
| 21:38.31 | [hC] | sergee: hence why there are patent offices and different patent laws for each country |
| 21:39.04 | lee_is_me | [TK]D-Fender: as [hC] said, correct? |
| 21:39.27 | sergee | [hC]: for EACH? are you sure? how about North Korea? china? ukraine? |
| 21:39.29 | [hC] | lee_is_me: yes, thats what has saying too. |
| 21:39.36 | [TK]D-Fender | lee_is_me, you can have 2 servers per reg, and multiple regs. |
| 21:40.00 | [hC] | sergee: no, i am not sure. i dont know anything about korean chines or ukrainian law. nor do i give a rats ass. |
| 21:40.07 | [hC] | sergee: it doesnt apply to me |
| 21:40.08 | sergee | [hC]: if you are interested, just type in google: "algorythms patents europe" and you will see yourself.. |
| 21:40.24 | ManxPower | Even china pays the royalties for the MPEG2 patents |
| 21:40.40 | lee_is_me | [TK]D-Fender: OK, and depending on the phone, is the number of registrations that can be made (like with 301 only allows 2)??? |
| 21:40.44 | [hC] | sergee: i will go ahead and spell algorithm's correctly. |
| 21:41.18 | sergee | [hC]: that's the point , there are plenty of people who don't know anything, but who gives a rats ass, and even trying to argue :) |
| 21:41.28 | ManxPower | [hC]: I just can't believe that companies are spending billions of dollars more around the world for telecom hardware because they are paying patent royalties when they don't have to be. |
| 21:42.04 | sergee | ManxPower: sure they do, if they want to sell harware in US, or if they owned by US companies. |
| 21:42.33 | ManxPower | sergee: you keep saying that the USA is not the world. |
| 21:42.44 | sergee | ManxPower: finaly :) |
| 21:43.19 | ManxPower | sergee: I know the USA is not the world, but my point still stands. |
| 21:43.31 | sergee | ManxPower: ok :) |
| 21:43.47 | [TK]D-Fender | lee_is_me, 301,320,330,430 = 2. 501=3, 550=4, 6XX=6 (more with sidecar) |
| 21:43.49 | ManxPower | anyway, I just sent an e-mail to SIPpro and voiceage asking for clarification |
| 21:44.22 | [hC] | ooo |
| 21:44.26 | [hC] | crazy power outage all of a sudden |
| 21:44.30 | ManxPower | I figure it can't hurt. |
| 21:44.35 | ManxPower | [hC]: new orleans area? |
| 21:44.36 | k31th | Evening. |
| 21:44.40 | [hC] | Costa rica |
| 21:44.41 | sergee | ManxPower: if you are interested: |
| 21:45.17 | [hC] | ManxPower: san jose costa rica... lightning just struck about 2 blocks from here :) |
| 21:45.37 | ManxPower | [hC]: I got a report of a customer in the new orleans area with power issues |
| 21:45.50 | lee_is_me | [TK]D-Fender: Thanks. I'm in the beginning phase of writing a GUI for polycom provisioning and I'm trying to wrap my head around some of the concepts. Not much field experience with them yet. |
| 21:45.52 | [hC] | ManxPower: ahh this is common here. its the rainy season. |
| 21:45.55 | sergee | ManxPower: Recently we tryed to buy some chip from Thomson, to build Set-Top-Box (TV-overIP, Surfing, VoIP) |
| 21:46.05 | *** join/#asterisk monstertruck (n=monstert@adsl-9-179-175.mia.bellsouth.net) |
| 21:46.06 | [hC] | I wnder how long the UPS's will carry the cable modem and access point. |
| 21:46.06 | [hC] | :) |
| 21:46.50 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 21:46.51 | sergee | ManxPower: they told us that they will sell their chip to us, only if will buy all licenses :) |
| 21:46.56 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 21:47.03 | monstertruck | quick question on a spa3102 configuration |
| 21:47.04 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 21:47.11 | *** part/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:47.23 | monstertruck | spa is behind nat, using stun server, after booting, it is able to make and receive calls |
| 21:47.40 | monstertruck | a few minutes later, it stops receiving calls |
| 21:47.41 | sergee | ManxPower: there are tons of them: mpeg2, mpeg4, g729, etc |
| 21:48.32 | ManxPower | sergee: yup. |
| 21:48.50 | ManxPower | monstertruck: you need to turn on NAT keepalive in the phone |
| 21:48.51 | sergee | ManxPower: there are chinese guys who selling similar chips and doesn't require licenses, |
| 21:48.51 | infinity1 | can someone recommend a web based sip client? |
| 21:49.01 | ManxPower | OR you need to portworward on your router |
| 21:49.01 | monstertruck | ManxPower, its set to yes |
| 21:49.26 | [TK]D-Fender | monstertruck, in the SPA's sip config in * add : "qualify=yes" and "nat=yes" |
| 21:50.06 | sergee | ManxPower: When we told about it to Thomson, they lowered the price, and shrink the list of required patents.. |
| 21:50.34 | ManxPower | sergee: I imagine they would. |
| 21:50.47 | sergee | ManxPower: trade not finished :) |
| 21:52.15 | ManxPower | Some consulting company is suggesting we use Trixbox for a network of 8 asterisk servers across 8 offices. |
| 21:52.25 | rene- | ManxPower: trixbox aint cool |
| 21:52.39 | ManxPower | rene-: trixbox is a piece of shit. |
| 21:52.55 | rene- | I have borrowed some stuff from it tho |
| 21:53.17 | rene- | the main developer posted on the web how he added mitel paging to trixbox |
| 21:53.46 | lee_is_me | rene-, you have a link for that? |
| 21:53.52 | rene- | ManxPower: i remember you once told me what to check for when one had dropped zap->sip calls that ended with ugly static noise |
| 21:53.55 | rene- | ManxPower: sure |
| 21:53.59 | ManxPower | rene-: they might consider reworking their config files so they don't require a degree from Rube Goldberg College to understand it. |
| 21:54.08 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:54.24 | rene- | http://www.freepbx.org/forums/viewtopic.php?p=4463#4463 |
| 21:54.25 | [TK]D-Fender | ManxPower, tell me the giant boot at the end isn't ENTIRELY worth it! |
| 21:54.32 | k31th | morning |
| 21:54.32 | ManxPower | I suppose nobody knows who Rube Goldberg was. |
| 21:54.37 | k31th | hows it going guys |
| 21:54.42 | [TK]D-Fender | <- I does ;) |
| 21:54.42 | rene- | i dont |
| 21:55.04 | infinity1 | can someone recommend a java or active-x based sip client? |
| 21:55.07 | [TK]D-Fender | rene-, thats because YOGURT has more culture than you! ;) |
| 21:55.09 | [TK]D-Fender | (j/k) |
| 21:55.14 | rene- | hehe |
| 21:55.18 | rene- | thats funny |
| 21:55.26 | ManxPower | He was an artist that drew things that would remind you of a Roadrunner or Tom and Jerry cartoon with the overly complex device used to catch the foe. |
| 21:55.52 | [TK]D-Fender | ManxPower, overly complex... yet SUCCESSFUL. |
| 21:56.05 | rene- | ohh i get it |
| 21:56.06 | [TK]D-Fender | rene-, go lookup "The Impossible Machine" (old video game) |
| 21:56.18 | rene- | i am looking at wikipedia entry for Rube Goldberg |
| 21:56.23 | rene- | and the Rube Goldberg machine |
| 21:56.25 | ManxPower | [TK]D-Fender: The cat never caught the mouse. |
| 21:56.58 | [TK]D-Fender | ManxPower, True, but Rube Goldberg's "inventions" DID what they were supposed to. |
| 21:57.13 | rene- | yeah trixbox is convoluted, it looks convoluted and feels that way |
| 21:57.20 | ManxPower | [TK]D-Fender: sodoes trixbox 8-) |
| 21:57.33 | [TK]D-Fender | ManxPower, Yup... doesn't mean we like it ;) |
| 21:58.02 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:58.02 | rene- | ManxPower: do you remember that advice you gave me some time ago about fugly static noise and sudden zap>sip dropped calls? |
| 21:58.07 | ManxPower | The company is trying to find a way to stop paying me for all the custom dialplan stuff and custom apps I write |
| 21:58.20 | ManxPower | rene-: no |
| 21:58.39 | Qwell[] | [TK]D-Fender: You've made the QOTD. :p |
| 21:59.20 | Tagor | lee_is_me >> Tried it, but unfortunately it doesn't work. Any other suggestions? |
| 22:01.02 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 22:01.02 | Tagor | lee_is_me >> It does execute the script now, but doesn't pass the variables. I tried $_GET/$_POST and globalvar |
| 22:01.02 | [TK]D-Fender | Qwell[], With which comment? :) |
| 22:01.03 | Qwell[] | [TK]D-Fender: That's the beauty of it... |
| 22:01.03 | lee_is_me | Tagor: You have to get the variables from stdin/stdout |
| 22:01.05 | Qwell[] | You think I'd actually TELL you? :P |
| 22:01.11 | [TK]D-Fender | Qwell[], TEASE |
| 22:01.13 | Qwell[] | heh |
| 22:01.20 | Qwell[] | yogurt ;) |
| 22:01.26 | ManxPower | I think I AM going to recommend Trixbox for the client. |
| 22:01.42 | lee_is_me | Tagor: let me check something... |
| 22:02.08 | ManxPower | Having to rewrite most of Trixbox to get them back to what they have now would prolly fund all my bills, personal projects, and travel plans for at least a year. |
| 22:02.17 | Tagor | lee_is_me >> I guess there it goes wrong. I have no idea what stdin/stdout is :P |
| 22:02.40 | lee_is_me | Tagor: what language are you writing with? |
| 22:02.53 | Tagor | lee_is_me >> PHP |
| 22:03.17 | *** join/#asterisk monstertruck_ (n=monstert@adsl-9-179-175.mia.bellsouth.net) |
| 22:03.19 | [TK]D-Fender | ManxPower, Perverse Psychology at its best. |
| 22:03.27 | ManxPower | Tagor: Why are you not using the PHP-AGI? |
| 22:03.39 | ManxPower | [TK]D-Fender: I'd hire some kid to do the hard work. |
| 22:03.41 | [TK]D-Fender | ManxPower, doing something right the first time is god business. Doing it right after 47 tries is JOB SECURITY. |
| 22:03.47 | monstertruck_ | ManxPower, [TK]D-Fender, thanks, qualify=yes seems to have done it |
| 22:04.02 | lee_is_me | Tagor: check out this link and how they use php to get from stdin/stdout |
| 22:04.04 | lee_is_me | http://www.voip-info.org/wiki/view/swift.agi |
| 22:05.38 | lee_is_me | Tagor: also I do not know how php effects the retrieval of passed in params. You need to ask on php forum. |
| 22:05.47 | Tagor | lee_is_me >> Actually this is #!/bin/sh -- shell script not PHP |
| 22:05.53 | Tagor | I see, ok, thanks :) |
| 22:06.14 | lee_is_me | If you were doing pascal or C# I might have been better help ;) |
| 22:06.15 | ManxPower | Why would anyone NOT use the PHP library? |
| 22:06.32 | *** join/#asterisk dlynes_laptop (n=dlynes@207.216.161.56) |
| 22:07.15 | *** join/#asterisk flashnet (i=fnet@gateway/tor/x-3a3e87061d0a89d8) |
| 22:08.49 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:08.59 | lee_is_me | some of us don't know php ;) |
| 22:09.22 | lee_is_me | <yet> |
| 22:09.24 | Ryushin | I have a new server that is not playing music on hold. I thought that asterisk 1.4 no longer needs mpg123. Am I right in this? |
| 22:09.54 | ManxPower | Ryushin: it does not require mpg123 if your MoH is not mp3. |
| 22:10.20 | [TK]D-Fender | Ryushin, Heck it doesn't require mp123 if it IS an MP3 ;) |
| 22:10.38 | Ryushin | Okay, that answer that. It is mp3. I thought 1.4 used its own built in mp3 decoder. |
| 22:10.38 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:10.45 | [TK]D-Fender | Ryushin, now read the big print and install asterisk-addons ;) |
| 22:10.47 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 22:10.55 | Tagor | lee_is_me >> Looks like there is just a way to send commands: http://www.voip-info.org/wiki/view/Asterisk+AGI+php |
| 22:11.00 | ManxPower | Ryushin: not until the patents on mp3 expire |
| 22:11.01 | [TK]D-Fender | Ryushin, which is SEPERATE because of legal implications ;) |
| 22:11.27 | Ryushin | Okay, so, install asterisk-addons or install mpg123? |
| 22:11.36 | ManxPower | or just convert your MoH to ANY format that Asterisk supports |
| 22:11.41 | lee_is_me | Tagor: yes, but every language has a way to grab command line parameters which is how parameters are sent from the dialplan, afaik |
| 22:11.44 | [TK]D-Fender | Ryushin, former |
| 22:11.50 | ManxPower | I suggest the format most calls will be in |
| 22:12.02 | Ryushin | OKay, thanks. I appreciate it again. |
| 22:12.17 | Tagor | lee_is_me >> But how are the vars passed from asterisk to the php script? I actually only now $_GET/$_POST |
| 22:12.34 | ManxPower | Tagor: they are passed as text lines on stdin |
| 22:12.39 | lee_is_me | ManxPower: how do you grab command line params in php? |
| 22:12.45 | ManxPower | lee_is_me: heck if I know. |
| 22:12.51 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 22:12.58 | ManxPower | but 3 seconds on google could tell me |
| 22:13.22 | lee_is_me | ManxPower: err, I was asking for Tagor here, who seems to be in the first stages of learning php... |
| 22:13.23 | Ryushin | ManxPower: I really don't want to deal with the time to convert the files right now. Thanks though. |
| 22:13.51 | ManxPower | Tagor: the interface for AGI is the same REGARDLESS of the language. Most languages have libraries available to hide all the ugly stuff like parsing the STDIN, etc. |
| 22:13.59 | lee_is_me | Tagor: ask on a php list and they will tell you how to get passed in params in php |
| 22:14.21 | Tagor | Ok, I will try in #php, thanks :) |
| 22:14.21 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 22:14.25 | ManxPower | I've NEVER had to deal with that crap in my AGIs, I just use asterisk-perl and it hides of the ugly stuff |
| 22:14.31 | Trionnis | hello all |
| 22:14.55 | ManxPower | Did they ever fix the "can only pass one command line param to AGI" problem? |
| 22:14.57 | lee_is_me | ManxPower: unfortunately, I had to write my own abstraction layer...no good ones for pascal :( |
| 22:15.21 | lee_is_me | ManxPower: I have passed more than one so it must be |
| 22:15.23 | ManxPower | lee_is_me: it sucks to be you |
| 22:15.37 | Trionnis | I'm having an issue with the Read() command, and the voip-info wiki isn't being of much help. Basically, I'm trying to place a call to a number, read a DTMF digit from that number, then run a gotoif based on that response. Is there something I might be missing in the Read() statement that's causing it not to read the number properly? |
| 22:15.52 | monstertruck_ | lee_is_me, $argv[x] |
| 22:16.06 | monstertruck_ | x is the param number |
| 22:16.12 | lee_is_me | monstertruck: hah! There it is! |
| 22:16.19 | monstertruck_ | $argv[0] is the name of the executable |
| 22:16.31 | lee_is_me | list pascal, paramstr(x) |
| 22:16.43 | lee_is_me | err, "like" pascal I meant |
| 22:16.44 | *** join/#asterisk Hymie (n=Hymie@l8r.net) |
| 22:16.48 | Trionnis | the extension config is here: http://pastebin.ca/559804 |
| 22:17.13 | monstertruck_ | wouldnt know, i havent seen pascal since last century |
| 22:17.32 | Hymie | hey guys... is there a way to call channels using and + or.. something like dial(sip/100&(zap/1||zap/4)&iax/place) |
| 22:17.32 | lee_is_me | monstertruck: pascal is like that too ParamStr(0) is the exe name |
| 22:17.32 | k31th | Guys, im trying to find a way to get clients to use voip that are worried about there Internet connection failing and them loosing there phone lines. I was thinking I could do the following... Have the lines for incoming calls still routed via PSTN and a use a hardware VoIP PBX in house and for the outgoing calls use VoIP and use the inbounc number as the caller ID. This would then work ok right? and still allow them to make use of cheap calls etc. |
| 22:17.52 | k31th | I really did type alot there :p |
| 22:20.08 | lee_is_me | Aren't companies starting to not allow setting callerid? |
| 22:20.16 | lee_is_me | for voip providers, I mean |
| 22:20.25 | _VoiceMeUp_COM | yeah |
| 22:20.33 | _VoiceMeUp_COM | some block |
| 22:20.52 | [TK]D-Fender | Trionnis, exten => test,3,GotoIf(${testtone} = "1"]?5) <- bad |
| 22:21.04 | Trionnis | ok, what am I missing? |
| 22:21.05 | Trionnis | heh |
| 22:21.06 | [TK]D-Fender | Trionnis, exten => test,3,GotoIf($[$"{testtone}"="1"]?5) <- good |
| 22:21.16 | Trionnis | hmm, I had that before |
| 22:21.22 | Trionnis | hang on, lemme try it again |
| 22:21.26 | [TK]D-Fender | Trionnis, exten => test,3,GotoIf($["${testtone}"="1"]?5) <- oopas |
| 22:22.42 | Trionnis | doesn't appear to have worked |
| 22:23.14 | [TK]D-Fender | Trionnis, pastbin your new dialplan and CLI output of its execution. |
| 22:23.49 | Trionnis | http://pastebin.ca/559819 |
| 22:23.58 | Trionnis | ah, hang on, I didn't put the dialplan in there |
| 22:24.00 | Trionnis | 1 sce |
| 22:24.01 | Trionnis | sec* |
| 22:24.50 | Trionnis | http://pastebin.ca/559821 |
| 22:24.52 | Trionnis | there |
| 22:25.20 | Trionnis | it doesn't appear to be reading the DTMF tone properly |
| 22:25.29 | [TK]D-Fender | -- User disconnected |
| 22:25.32 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 22:25.47 | Trionnis | right |
| 22:25.55 | Trionnis | the other end plays a single DTMF tone, then hangs up |
| 22:26.11 | Trionnis | I'm trying to detect that tone, and either fire off the alert script, or just drop out of the call |
| 22:26.19 | Trionnis | Read() doesn't seem to be getting the tone |
| 22:26.38 | [TK]D-Fender | Trionnis, Your read doesn't appear to be using the proper parameters either... |
| 22:26.48 | _VoiceMeUp_COM | trying to amek a call back system.. via ougoing files... call in ..rings.. never asnwer or busy() then calls you back with disa |
| 22:26.54 | [TK]D-Fender | Trionnis, perhaps you should set your DTMFMODE while you're at it... |
| 22:26.57 | _VoiceMeUp_COM | but cell co still seeing as answered |
| 22:27.10 | Trionnis | ok |
| 22:27.12 | _VoiceMeUp_COM | only doing wait -> deadagi- > hangup |
| 22:27.21 | _VoiceMeUp_COM | do i need a ringing or Congestion ? |
| 22:28.29 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:29.04 | ManxPower | you do not need a hangup in deadagi |
| 22:29.22 | _VoiceMeUp_COM | Ah |
| 22:29.36 | _VoiceMeUp_COM | ok and if i sned a congestion before it just stops there |
| 22:29.56 | _VoiceMeUp_COM | so i need to send after the agi ? |
| 22:29.56 | _VoiceMeUp_COM | or in it worst case ? |
| 22:29.56 | ManxPower | you are not going to GET to deadagi unless the channel is hungup |
| 22:29.56 | _VoiceMeUp_COM | i dont have any answers so why is cell co seing as answered |
| 22:30.01 | Trionnis | http://pastebin.ca/559831 <-- new paste |
| 22:30.22 | ManxPower | _VoiceMeUp_COM: are you using FXO ports? |
| 22:30.29 | _VoiceMeUp_COM | i tough deadagi would just make dialplan continue |
| 22:30.39 | _VoiceMeUp_COM | no its PRI -> Asterisk |
| 22:31.48 | Trionnis | still nothing |
| 22:31.50 | Trionnis | ;( |
| 22:31.54 | [TK]D-Fender | Trionnis, your read still looks bad. |
| 22:32.08 | [TK]D-Fender | Trionnis, "show application read" |
| 22:32.33 | Trionnis | exten => test,3,Read(testtone||1|||) |
| 22:32.40 | Trionnis | that didn't make it into the paste |
| 22:32.41 | Trionnis | sorry |
| 22:32.43 | Trionnis | that's the new line |
| 22:33.13 | Waverly360 | Is it possible to send a beep to a group of phones after connecting using the Page function? |
| 22:33.17 | [TK]D-Fender | Trionnis, pastebin the instructions for it. |
| 22:33.40 | ManxPower | _VoiceMeUp_COM: playback and background will answer by default unless you tell them not to |
| 22:33.52 | Trionnis | http://pastebin.ca/559842 |
| 22:33.57 | Trionnis | at he bottom |
| 22:33.58 | Trionnis | the* |
| 22:34.01 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 22:34.10 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 22:34.25 | [TK]D-Fender | Trionnis, now re-read your READ call |
| 22:34.30 | *** join/#asterisk perezmeyer (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:34.37 | CrashSys | Is there a way I can tell if a pair is a T1 pair from a terminal block? (dont have access to customer demarc) |
| 22:34.41 | _VoiceMeUp_COM | no sounds are passed |
| 22:34.50 | _VoiceMeUp_COM | no play no back no nothing just need to hangup |
| 22:34.55 | k31th | lee_is_me: you can set the caller ID if you can prove you own the number. |
| 22:35.41 | Trionnis | doh |
| 22:35.41 | [TK]D-Fender | Trionnis, nvm |
| 22:35.55 | Trionnis | yeah, the only thing I see is the brackets |
| 22:36.02 | Trionnis | and I'm 99% sure those don't apply :) |
| 22:37.03 | Trionnis | if I call the number itself, it works fine |
| 22:37.09 | Trionnis | I get the tone playback |
| 22:37.19 | Trionnis | I'll pm you the number if you wish, and you can verify that |
| 22:38.32 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:41.38 | Mercestes | Off Topic: Don't suppose anyone here can look up a license plate # for me. >:) |
| 22:41.48 | thekidrio | Anyone know a good guide to IVR scripting? Best practise stuff? |
| 22:42.18 | Waverly360 | Mercestes: Nope. Sorry :) |
| 22:42.47 | thekidrio | Mercestes, not something anyone would announce over IRC heh |
| 22:42.51 | Mercestes | ya neve rknow. |
| 22:42.56 | Mercestes | Oh, I'm checknig my /msgs. :) |
| 22:43.00 | thekidrio | haha |
| 22:43.01 | [TK]D-Fender | thekidrio, Common Sense <0 |
| 22:43.07 | Corydon76-work | ~book |
| 22:43.21 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:43.31 | thekidrio | [TK]D-Fender, I have never seen this common sense you speak up |
| 22:43.50 | [TK]D-Fender | thekidrio, I hadn't noticed.... |
| 22:43.57 | [TK]D-Fender | ~osmosis |
| 22:43.59 | jbot | i heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 22:43.59 | thekidrio | hahaha |
| 22:44.00 | [TK]D-Fender | ^^^^^^^^ |
| 22:44.04 | thekidrio | Love you too [TK]D-Fender :P |
| 22:44.17 | Trionnis | why do they call it "common" sense... ? It damn sure isn't all that common. |
| 22:44.21 | Trionnis | ^^ |
| 22:44.31 | CrashSys | If I hook a butt-set to a pair and it has a T1 circuit on it... would I get anything from the buttset? Would I destroy the repeaters/cpe? |
| 22:44.57 | Trionnis | you'd probably fry the buttset |
| 22:45.06 | CrashSys | well scratch that idea |
| 22:45.09 | Trionnis | maybe blow a fuse in the CPE |
| 22:45.17 | Trionnis | it's generally not a good idea :) |
| 22:45.37 | Trionnis | some of the sets have a limiter diode to prevent it |
| 22:45.37 | lee_is_me | k31th: I don't think so. The last company that I was aware of that allowed you to set your own callerid was Telasip and I think they stopped too. |
| 22:45.40 | Trionnis | but not the cheaper ones |
| 22:45.48 | Trionnis | lee_is_me: voipjet allows it |
| 22:45.54 | Trionnis | but that's outgoing only |
| 22:46.02 | lee_is_me | k31th: voipjet allows it. |
| 22:46.05 | lee_is_me | ;) |
| 22:46.08 | Trionnis | lol |
| 22:46.11 | Waverly360 | [TK]D-Fender: got a sec to help me out with something? |
| 22:46.13 | CrashSys | Well I have the rat's nest of lines on the outside of the building from verizon's demarc... then there's a few thousand cables all over the back of the building (ok, more like 100)... and I am trying to identify what is real and what's garbage... |
| 22:46.31 | Trionnis | ok, [TK]D-Fender, any other ideas why this thing won't read the dtmf on the call? |
| 22:46.34 | CrashSys | Trionnis: I got a Harris TS22... guess I can look up specs to see if it's got a limiter... |
| 22:46.41 | Trionnis | probably |
| 22:46.45 | Trionnis | I've not heard of it |
| 22:47.05 | [TK]D-Fender | Trionnis, Dunno about the validity of your attempt to set the mode. |
| 22:47.09 | lee_is_me | <reminds me> need to talk with axVoice about my caller id. They are using my personal name |
| 22:47.11 | Waverly360 | I'm trying to setup paging for my users by using the Page() function. It works fine, but I want to pass a beep to the users being paged, so that they'll know that their phone has picked up. |
| 22:47.17 | k31th | lee_is_me: you have to prove you own the number yeah, if it was just a control panel options it would get abused badly id imagen. |
| 22:47.33 | [TK]D-Fender | Waverly360, what are you pumping into the page? |
| 22:47.43 | lee_is_me | k3lth: agreed. |
| 22:47.55 | Trionnis | well |
| 22:48.07 | CrashSys | Trionnis: Got any good ideas for identify active mixed (analog/T1) pairs? |
| 22:48.07 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:48.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:48.17 | Trionnis | sipmedia uses inband by default (already set in sip.conf), and it's an outgoing sip call... so... ? |
| 22:48.32 | Waverly360 | [TK]D-Fender: the actual command is Page(SIP/10&SIP/11&SIP/12|d) |
| 22:49.00 | Trionnis | CrashSys: not a clue besides calling the telco, or looking at the config of the csu/dsu |
| 22:49.11 | Waverly360 | [TK]D-Fender: right after I'm trying to do a Playback(beep), but that never executes...I didn't realize Page is blocking. |
| 22:49.22 | CrashSys | Yeah... what I was trying to avoid... verizon's real good about helping the lil guys figure out their mess :) |
| 22:49.29 | ManxPower | most apps block |
| 22:49.51 | Waverly360 | ManxPower: Yeah, I hadn't really thought about it cause I'd never done anything like this before. |
| 22:49.55 | Trionnis | I have yet to deal with an ILEC that was worth anything |
| 22:50.00 | Trionnis | heh :) |
| 22:50.08 | Waverly360 | ManxPower: I was hoping there was something else I could try to workaround the issue. |
| 22:50.18 | CrashSys | Tri: That's the encouragement i'm looking for :) |
| 22:50.36 | [TK]D-Fender | Waverly360, record your page, then .call file it. w/ playback |
| 22:51.19 | Waverly360 | [TK]D-Fender: the users want it to be a live page...there's no other way? |
| 22:51.34 | Trionnis | brb |
| 22:51.36 | mutilator | anyone wanna but a lightly used te110p? $200 |
| 22:52.42 | [TK]D-Fender | Waverly360, dial a local channel to page with M |
| 22:53.18 | CrashSys | I guess you could weasel something together with meet-me... but that'd be more trouble then it's worth... |
| 22:54.37 | CrashSys | I recommend against live paging cause of feedback... but most of what I do is over-head... |
| 22:55.31 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 22:55.36 | Waverly360 | but feedback is fun. [TK]D-Fender: thanks, I"ll see if I can't make that work. |
| 22:56.51 | CrashSys | Actually there's enough lag to eliminate feedback... but the echo's are fun :) |
| 23:00.56 | Trionnis | what's the dialplan command that outputs a variable value to the cli? |
| 23:01.01 | Trionnis | I can't find it or remember it :) |
| 23:01.04 | CrashSys | NoOp |
| 23:01.06 | Trionnis | thanks |
| 23:01.09 | CrashSys | NoOp(<whatever>) |
| 23:04.38 | Trionnis | ok, found the problem |
| 23:04.47 | Trionnis | it's sipmedia messing with the dtmf somehow |
| 23:04.48 | CrashSys | Operator Error? |
| 23:04.54 | Trionnis | through voipjet it works fine |
| 23:05.01 | Trionnis | and up yours too, kthx ^^ |
| 23:05.03 | Trionnis | lol |
| 23:06.54 | [TK]D-Fender | PEBKAC |
| 23:07.55 | Trionnis | nope |
| 23:07.57 | Trionnis | not this time |
| 23:08.00 | Trionnis | most other times, yes |
| 23:08.05 | Trionnis | but not this time ;) |
| 23:09.19 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 23:10.56 | CoffeeIV | I upgraded from asterisk 1.2 to 1.4 and now my spandsp based fax receiving with the dialplan application RxFax segfaults. I've tried several different versions of the spandsp library, no luck . . . is anyone here using spandsp and asterisk 1.4 to receive faxes (successfully) ? |
| 23:12.05 | snuffy22 | i have.. |
| 23:12.23 | snuffy22 | dl'ed latest spandsp + newer rx/tx fax.. |
| 23:12.30 | *** join/#asterisk anthm (n=anthm@m010f36d0.tmodns.net) |
| 23:12.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:12.54 | CoffeeIV | snuff22: did you have to edit the code in app_rxfax.c to keep it from crashing ? Or did it just work ? |
| 23:12.54 | snuffy22 | you also have to change the * source in one section |
| 23:12.55 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 23:13.03 | snuffy22 | one of the header references |
| 23:13.12 | Mahmoud | any online service that does SIP to IAX2 conversion? |
| 23:14.13 | CoffeeIV | snuff22: you mean copy a plc.h header file into asterisk from spandsp ? I did that |
| 23:14.14 | lee_is_me | is there a way to turned down the handset mic sensitivity for polycoms? |
| 23:14.30 | Trevor_b | in the XML config file or through the menu. |
| 23:14.34 | Trevor_b | oh wait |
| 23:14.36 | Trevor_b | Mic. |
| 23:14.46 | lee_is_me | yeah |
| 23:15.05 | Trevor_b | I do beleive the xml config (sip.cnf) i beleive is in there. Grab the polycom administrators PDF and look at the volume control sections |
| 23:15.50 | lee_is_me | Trevor_b: been studying it today and didn't see anything about it. would be in phone1.cfg section right? |
| 23:16.49 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 23:17.20 | Trevor_b | i beleive so, been awhile since i looked through the file. Hmm nothing on gain in it then? Ill have to track down my PDF and look through it, ill have time in about 3 hours ;). Seriously, home for dinner then night operations tonight so ill be back with some time on my hands. |
| 23:17.39 | Trevor_b | Search on "gain" |
| 23:18.58 | lee_is_me | Trevor_b: that was it. I'll go stand in the corner now |
| 23:23.45 | _VoiceMeUp_COM | hmm exten => 12341234/45646456 needs to at least HAVE an exten => 12341234 for it to work with cid matching |
| 23:23.50 | _VoiceMeUp_COM | FYI |
| 23:26.09 | *** join/#asterisk ReDNeQ (n=iBuMp@cpe-66-68-37-190.austin.res.rr.com) |
| 23:26.28 | _VoiceMeUp_COM | anyway to DEALY the outgoing thing ? |
| 23:26.36 | _VoiceMeUp_COM | i guess i need to delay the cp |
| 23:29.59 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 23:31.29 | lee_is_me | polycom docs say "do not alter these values" for <gains/> setting. If I change them, will a giant ball come rolling out of the ceiling and chase me? |
| 23:39.02 | *** join/#asterisk iBuMp- (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 23:39.07 | irule | is it possible t just skip without hangup instead of hangup when a playback file does not exist? |
| 23:39.23 | errr | hey I was on a call for 45mins and then the call was just dropped any idea why that happened? this is the 3rd time today its happened to me. every time its right at 45mins |
| 23:43.18 | lee_is_me | errr: check AbsoluteTimeout |
| 23:44.25 | lee_is_me | errr: and rtptimeout |
| 23:47.17 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:50.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:54.23 | errr | lee_is_me: ok thanks, ill look into that |
| 23:59.12 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
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