| 00:01.06 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 00:04.23 | TJNII | Old rotary phones used to have a 3rd ground wire, but that was for party line ringing, not the voice circuit. |
| 00:04.50 | *** join/#asterisk aurix (~aurelio@81.174.13.196) |
| 00:07.28 | *** join/#asterisk seanjohn (~admin@gateways.sheltoncomputers.com) |
| 00:07.41 | seanjohn | originate sip/201 application macro connect argument1,argument2 |
| 00:07.59 | seanjohn | these are originated but the arguments don't get passed |
| 00:08.13 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:09.06 | seanjohn | <PROTECTED> |
| 00:10.17 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:11.01 | Mark22 | seanjohn: could you give an example in a pastebin of your dialplan we can look at to help you? |
| 00:11.59 | seanjohn | the dialplan is [macro-connect] and the arguments are used as ${ARG1} ${ARG2} and so on |
| 00:12.20 | Mark22 | that sounds correct |
| 00:12.30 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 00:12.34 | seanjohn | its a bug in asterisk's orginate |
| 00:13.44 | seanjohn | with other apps, you would do arg1,arg2,arg3 and so on and they would pass to the application correcly but originate is looking at the arguments concatenated together with ',' as contexts and priority |
| 00:14.14 | seanjohn | originate sip/201 application macro connect,argument1,argument2 |
| 00:14.44 | seanjohn | it tries to do argument1@connect |
| 00:15.32 | seanjohn | it thinks macro is a dial command |
| 00:17.10 | Mark22 | sorry, I don't know the solution to your problem :S |
| 00:17.49 | seanjohn | i figured out the difference. With using macro as the originate application, you have to do macroname|arg1|arg2|arg3 instead of macroname,arg1,arg2,arg3 |
| 00:18.09 | seanjohn | other applications will take commas |
| 00:18.25 | Mark22 | with asterisk 1.6? |
| 00:18.47 | seanjohn | 1.4; i'm sure with 1.6 too. remember, this is using the originate application command, not in a dialplan |
| 00:19.59 | Mark22 | that is probably the difference (I do everything if possible using the dialplan) |
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| 00:32.49 | SaiSoma|AtHome | hey guys, can anyone point me to a tutorial of asterisk/mysql and handling an array of data (i need to loop through the array in asterisk) |
| 00:41.23 | kc8pxy | !debug |
| 00:41.54 | kc8pxy | how do i get the info i need to have you guys help diagnose connection issues? |
| 00:48.59 | *** join/#asterisk rlankfo (~areohbee@hahainyourface.com) |
| 00:49.26 | *** join/#asterisk nitram (foo@superblob.com) |
| 00:58.35 | ChannelZ | well you could start by explaining what the issue is |
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| 01:06.44 | kc8pxy | !help |
| 01:10.16 | pabelanger-lap | ~ask |
| 01:10.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:13.36 | kc8pxy | i have 2 sip peers trying to call each other. we seem to be able to connect the call reliably, but only once could we hear each other. I've tried everything i know, and need to know how to assemble the proper data to get diagnostic help here so connecting is more reliable. |
| 01:14.55 | Bendbanks | hi I'm trying to install asteriskforskype and I seem to be missing the chan_skype.so file after the install, does anyone have any thoughts |
| 01:29.16 | kerframil | kc8pxy: rtp problems are commonplace. where are the peers in relation to one another? is nat employed in the network path between them and/or the server? if so, what if anything, have you done to address it? you can use set sip debug to obtain diagnostic information. |
| 01:29.33 | kerframil | kc8pxy: it's also worth noting that one can force asterisk to stay in the media path by setting canreinvite=no |
| 01:33.30 | *** part/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 01:35.15 | kc8pxy | kerframil: both peers are set to canreinvite=no, and the server is on the gateway for lan it's on. one client is on the lan, one is in the wild, on the internet. the firewall on the gw has 10000-20000 set to accept for rtp, neither are set to nat. |
| 01:42.44 | *** join/#asterisk zerohalo (~zerohalo@cambridge.zerohalo.com) |
| 01:43.00 | kerframil | kc8pxy: do you have wgetpaste? |
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| 02:17.00 | _structz | Gugge, there? |
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| 04:33.06 | boodu | ciao |
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| 05:36.15 | redax | hi |
| 05:37.20 | redax | what's this extensions registers at my sip server with unknown user.... from 1.1.1.1 ? |
| 05:37.22 | redax | logger.c: -- Registered SIP '3706' at 1.1.1.1 port 5060 |
| 05:37.38 | redax | asterisk 1.6.0.9 |
| 05:37.44 | redax | some kind of security hole? |
| 05:38.57 | redax | <PROTECTED> |
| 05:38.59 | redax | hm. |
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| 05:52.02 | sawgood | By default, does a Diguim (Asterisk Appliance) AA50 run the same Asterisk (1.4 or 1.6) as other items, or is the build of software on these appliances more like SwitchVOX SMB stuff? |
| 05:52.25 | redax | packets coming from 173.1.76.11 |
| 05:55.34 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:59.15 | redax | http://honeynet.org.au/?q=phoneynet_part2 |
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| 07:38.18 | _zoom_ | hi, to specify rtp port range for certain sip channel? |
| 07:39.10 | tzafrir | _zoom_, you were given some answers (basically: change the source, and it's not going to be easy) |
| 07:39.25 | tzafrir | Why would you need that? |
| 07:39.38 | tzafrir | For a specific channel? For a group of them? |
| 07:40.26 | _zoom_ | may be, |
| 07:41.25 | _zoom_ | i have two nodes behind nat f/w, i cannt change there port number (5060) |
| 07:42.07 | _zoom_ | n i should drop calls from outside to f/w and dnat sip request to one of those two node |
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| 08:19.09 | tzafrir | _zoom_, so you have a problem with NAT |
| 08:19.14 | tzafrir | Ask about that |
| 08:19.56 | tzafrir | For starters: |
| 08:19.57 | tzafrir | ~nat |
| 08:19.58 | infobot | hmm... nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 08:21.09 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 08:22.14 | [sr] | hi |
| 08:30.26 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 08:32.29 | [sr] | hi portuga |
| 08:33.46 | [sr] | ruyo: it's for you |
| 08:33.46 | [sr] | :p |
| 08:34.00 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
| 08:34.40 | ruyo | Hi, sir. :> |
| 08:37.13 | _zoom_ | tzafrir: exten => _XXXXX,1,dial(${EXTEN}@1.2.3.4:8060) |
| 08:37.23 | _zoom_ | i do dnat in f/w |
| 08:37.35 | _zoom_ | but internal sip info is the problem |
| 08:38.01 | tzafrir | _zoom_, for starters, externip and localnet |
| 08:40.16 | tzafrir | _zoom_, also: the NAT router is the asterisk server? Or a different box? |
| 08:40.29 | _zoom_ | different box |
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| 09:07.49 | tzafrir | _zoom_, so you want any rtp port that is related to a call through the externip to be allocated from a different pool of ports? |
| 09:08.32 | tzafrir | I suspect having more than one pool of ports is doable. What version of Asterisk is it? |
| 09:15.44 | _zoom_ | 1.6.2 |
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| 09:35.27 | [sr] | _zoom_: be more exact about the version, 1.6.2.? |
| 09:47.45 | _zoom_ | 1.6.2.10 |
| 09:52.57 | [sr] | that way its easier for them to help you |
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| 10:32.13 | m_tadeu | hi...I just noticed that the cdr fields in the csv file are not the ones in the mysql database |
| 10:32.35 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 10:33.07 | m_tadeu | I need some fields that exist in the csv file to be in the mysql cdr log...how do I do that? |
| 10:34.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:36.47 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 10:40.16 | m_tadeu | ah I think I got it :) going to test it now |
| 10:40.36 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:55.45 | m_tadeu | no luck...how do I get asterisk to insert 'start', 'answer', 'end' firlds? |
| 10:55.55 | m_tadeu | in the cdr table, I mean |
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| 11:26.39 | m_tadeu | ah I got it :) |
| 11:30.36 | *** join/#asterisk [netman] (~netman@83.52.208.225) |
| 11:33.25 | m_tadeu | why is start time and answer time the same all the time? |
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| 11:56.28 | Jinxed- | what is the recommended install for asterisknow |
| 11:56.46 | Jinxed- | nvm.... lol it timed out and picked the default |
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| 12:03.18 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 12:03.40 | ruben23 | hi guys any idea on how to set dial pla for calling paris..anyone can share.. |
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| 12:07.54 | drmessano | ruben23: http://www.countrycallingcodes.com |
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| 12:22.46 | hariom | I want to start asterisk in non root user mode so that no body can modify anything in tty 9. Can somebody suggest a robust way to do that? |
| 12:24.55 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 12:26.59 | Jinxed- | the 32 bit version of *now should be able to work on a 64 bit machine correct? |
| 12:27.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:29.49 | fauxalliance | http://www.voip-info.org/wiki/view/Asterisk+non-root @ hariom |
| 12:30.03 | fauxalliance | chroot = robust |
| 12:31.09 | hariom | fauxalliance: what do you mean by chroot = robust? |
| 12:31.47 | [TK]D-Fender | robust = full featured |
| 12:31.50 | fauxalliance | chroot is capable of coping well with variations |
| 12:31.52 | [TK]D-Fender | "thorough |
| 12:32.07 | [TK]D-Fender | ~asterisk-non-root |
| 12:32.08 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 12:32.10 | [TK]D-Fender | ^^^ |
| 12:33.57 | fauxalliance | ~selinux |
| 12:33.58 | infobot | hmm... selinux is the NSA's port to Linux of the FLASK security Architectur, called Security Enhanced Linux. |
| 12:46.46 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 12:49.11 | *** part/#asterisk kleofas (~kleofas@router.dir.pl) |
| 12:50.03 | ruben23 | <PROTECTED> |
| 12:50.16 | ruben23 | but how would be my dial plan look like |
| 12:52.16 | drmessano | No different than any other pattern matching, and now you have the prefixes |
| 12:54.02 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 12:57.38 | ruben23 | drmessano:can you give me sample like..please |
| 12:58.19 | [TK]D-Fender | ruben23: Dial(sip/PROVIDER/00331234567) |
| 12:58.20 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 12:58.27 | [TK]D-Fender | ~BOOK |
| 12:58.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:58.29 | [TK]D-Fender | ^^^ |
| 12:58.34 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 12:58.36 | Jinxed- | So I just installed *now, and Im stuck at the terminal... how do I get to the gui? |
| 12:58.55 | drmessano | http://ipaddress |
| 12:59.00 | [TK]D-Fender | Jinxed-: Via a WEB BROWSER on another computer |
| 12:59.43 | Maliuta | I thought xterm+screen+vim _was_ a gui |
| 12:59.58 | Jinxed- | [TK]D-Fender, didn't work |
| 13:00.24 | Jinxed- | [TK]D-Fender, I did an ifconfig on my virtual machine running *now |
| 13:00.28 | Jinxed- | got the ip address |
| 13:00.36 | Jinxed- | and typed it in my browser and got nothing |
| 13:00.54 | [TK]D-Fender | Jinxed-: maybe your VM networking is bad |
| 13:01.02 | [TK]D-Fender | Jinxed-: Is it listening on the port? |
| 13:01.17 | Jinxed- | [TK]D-Fender, how would i check for that? |
| 13:01.47 | [TK]D-Fender | Jinxed-: netstat -an|grep 80 |
| 13:02.39 | Jinxed- | [TK]D-Fender, should I do that in the VM or on my actual ubuntu desktop |
| 13:02.54 | [TK]D-Fender | Jinxed-: on the SERVER |
| 13:03.42 | Jinxed- | [TK]D-Fender, it says stream connected |
| 13:04.09 | [TK]D-Fender | Jinxed-: Show us.,.. |
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| 13:07.36 | phretor | could you recommend a good ISDN card for a linux-based Asterisk installation? It doesn't necessarily have to be an officially-suported card: as long as it's know to work robustly, I'm fine. |
| 13:08.37 | phretor | and also, I've been given an ISDN PRI, is there anything special that I need to know in order to set up an Asterisk box for making outbound calls? |
| 13:09.33 | *** join/#asterisk b0ot (~b0ot@198.99.129.129) |
| 13:09.52 | b0ot | [TK]D-Fender, http://i.imgur.com/ipkU3.png (This is Jinxed- ) |
| 13:10.02 | b0ot | sorry it took me a sec to get my other laptop online |
| 13:11.00 | [TK]D-Fender | b0ot: the 1st line, not the 2nd |
| 13:11.12 | [TK]D-Fender | b00it is listening. Your vm networking seems to eb at fault |
| 13:11.12 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:11.14 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 13:11.36 | b0ot | hmm |
| 13:12.02 | *** join/#asterisk kleofas (~kleofas@router.dir.pl) |
| 13:12.08 | b0ot | you don't happen to know what I might need to do to fix it :/ ? |
| 13:12.25 | b0ot | should the network adapter be set to NAT? |
| 13:12.48 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:13.19 | [TK]D-Fender | b0ot: Go ask in whatever channel supports your VM environment. That would not be "here" |
| 13:13.29 | b0ot | ok |
| 13:14.42 | b0ot | thanks |
| 13:18.07 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 13:22.17 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 13:31.01 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:38.10 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:40.10 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:46.04 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:46.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:46.32 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 13:48.25 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 13:49.35 | *** join/#asterisk iamy_china (~iamy_chin@221.223.53.171) |
| 13:52.40 | phretor | is there any special hardware requirement for making calls from an ISDN PRI? |
| 13:54.22 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 13:55.41 | Chainsaw | phretor: Aside from a PRI ISDN adapter or PRI ISDN-to-SIP gateway? No. |
| 13:56.50 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:58.38 | kc8pxy | ~help |
| 14:00.03 | *** join/#asterisk adyn (~adyn@onu-hq.onenetusa.net) |
| 14:00.13 | *** join/#asterisk methodvon (~methodvon@108.18.246.223) |
| 14:03.56 | kc8pxy | i know I've asked this again in recent memory, but my brain has been on teh fritz. where is a link on how to properly perpare debug output for offering here? |
| 14:04.27 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 14:05.56 | Chainsaw | ~debug |
| 14:05.57 | infobot | ACTION DeBuggers $1 |
| 14:06.05 | Chainsaw | Ah well. It was worth a go. |
| 14:06.09 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:06.11 | Chainsaw | Only Fender knows all of the ~annoyances |
| 14:06.16 | Goshen | pastebin.ca |
| 14:06.41 | Chainsaw | Goshen: That's where we want it. But there's an explanation of *what* we want. |
| 14:07.27 | Goshen | you don't want to comb through 40 pages? :) |
| 14:09.00 | eppigy | win 15 |
| 14:09.01 | kaldemar | kc8pxy: http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 14:09.26 | kc8pxy | i think i'm having rtp issues.. i've gotten ONE successful voice call out of probably 20 tries, and 80% of those actually connect the sip channels. |
| 14:09.36 | kc8pxy | kaldemar: bookmarking |
| 14:10.04 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 14:10.26 | Deeewayne | ~sipnat |
| 14:10.27 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:10.50 | kaldemar | kc8pxy: that's a file in the source package. |
| 14:11.20 | Goshen | kc8pxy, if you can, use IAX |
| 14:13.50 | spiceycurry | Just restarted my asterisk server, which has been running seemingly fine for a couple months. , and getting the new error "FATAL: Module dahdi not found". When I run - find /lib/modules -name dahdi - I am getting /lib/modules/2.6.18-164.15.1.el5/dahdi - however when I run modinfo dahdi - I am getting modinfo: could not find module dahdi. Any ideas? |
| 14:15.02 | russellb | spiceycurry: is that the kernel version you are actually running? |
| 14:15.12 | spiceycurry | yes |
| 14:15.29 | spiceycurry | [root@f2e6 asterisk]# uname -a |
| 14:15.29 | spiceycurry | Linux f2e6.hostopia.com 2.6.18-194.8.1.el5 #1 SMP Thu Jul 1 19:04:48 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux |
| 14:16.19 | russellb | well I think the dahdi you see from find is a directory |
| 14:16.23 | russellb | you should see dahdi.ko |
| 14:16.41 | russellb | see what's in that dir |
| 14:16.46 | spiceycurry | k |
| 14:17.14 | spiceycurry | there are .ko files there |
| 14:17.27 | spiceycurry | is there a way to reinstall? |
| 14:17.40 | russellb | sure, download dahdi and install .. |
| 14:17.58 | russellb | you can try running insmod directly on dahdi.ko and see if it's complaining about something |
| 14:18.42 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:19.10 | [TK]D-Fender | [10:14]<russellb>spiceycurry: is that the kernel version you are actually running? <- NO |
| 14:19.35 | [TK]D-Fender | spiceycurry: /lib/modules/2.6.18-164.15.1.el5/dahdi != Linux f2e6.hostopia.com 2.6.18-194.8.1.el5 |
| 14:19.39 | [TK]D-Fender | 194 != 164 |
| 14:19.50 | ruben23 | hi guys how do i verify if my call are suing g729 codec or g711..? are there way i can see it.. |
| 14:20.00 | spiceycurry | shit, someone must have upgraded my kernel |
| 14:20.19 | [TK]D-Fender | ruben23: Look at the SIP DEBUG |
| 14:20.33 | kc8pxy | Goshen: both channels are phones, on on the internet from my android phone (sipdroid), and one from my pc, using ekiga. |
| 14:20.36 | spiceycurry | I suppose I am going to have to reinstall dahdi now |
| 14:20.48 | [TK]D-Fender | [10:17]<russellb>sure, download dahdi and install .. |
| 14:20.55 | kc8pxy | Goshen: i don't think either of those does iax :) |
| 14:21.26 | b0ot | [TK]D-Fender, I got it WORKING!!!! |
| 14:21.42 | *** join/#asterisk dacm_work (~dan@host86-182-228-210.range86-182.btcentralplus.com) |
| 14:22.55 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 14:23.07 | [TK]D-Fender | b0ot: Congratulations. |
| 14:23.07 | Goshen | kc8pxy, looks like there are iax clients for droid |
| 14:23.39 | drmessano | IAX is never the answer to "I have a SIP problem" |
| 14:23.47 | drmessano | If it's NAT, fix it |
| 14:23.50 | kc8pxy | Goshen: what is the name? |
| 14:23.51 | drmessano | ~sipnat |
| 14:23.52 | infobot | sipnat is, like, Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:24.25 | Goshen | kc8pxy, looks like iaxagent |
| 14:24.47 | Goshen | http://www.androidzoom.com/android_applications/communication/iaxagent-beta_fkwc.html |
| 14:24.55 | b0ot | [TK]D-Fender, When I loaded freepbx it says that Critical Error: retrieve_conf failed, config not applied. Crtical Error * Manger connection Failure.... is this normal or something I should be concerned with |
| 14:25.24 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 14:25.24 | [TK]D-Fender | b0ot: NOT SUPPORTED HERE. |
| 14:25.30 | [TK]D-Fender | b0ot: #freepbx |
| 14:26.41 | dacm_work | This probably isn't the right place to ask this (a pointer on where to ask this would be almost as welcome as an answer) but I have a question regarding what a PBX can actually do. (I'm a telephony n00b.) - Let's say I have one line connected to a PBX with many handsets connected to the PBX. If someone calls, is the line still tied up for the duration of the call, or can another handset pick up a second incoming call? (C |
| 14:26.41 | dacm_work | an a second handset make any outgoing calls?) If it is tied up, then how do people have single phone numbers that can take multiple calls? (Are incoming calls redirected to other lines?) |
| 14:27.12 | Goshen | kc8pxy,Zopier does IAX....two IAX clients..problem solved ;) and you only have to open one port for IAX |
| 14:28.33 | kc8pxy | Goshen: that's if my problem ins a non-nat, rtp issue with sip. I'm still working on getting the debug info for you guys. |
| 14:30.13 | telnettech | dacm_work......you only have 1 call either inbound or outbound.....you can get services from your telo provider that will allow you to get multiple channels depending on your needs |
| 14:30.15 | kc8pxy | it's not nat, i'm fairly sure of it, because i put the server on teh gateway this time, so no connections need nat, and the sip channels in sip.conf are not set to nat. sip show peers also shows none of the channels involved having nat. |
| 14:31.03 | drmessano | kc8pxy: Have you added any parms like externhost, localnet, etc? |
| 14:31.11 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 14:31.14 | Goshen | kc8pxy, So you have your asterisk server open to internet? |
| 14:32.29 | [TK]D-Fender | dacm_work: You can process calls sent to * by whatever you set it up with in any way you want. |
| 14:32.58 | kc8pxy | Goshen: until i can get regular calls working, yes. i can make calls from lan to vpn phones fine, no rules to limit how rtp works in that crossover. but lan and internet is buggered right now. |
| 14:33.31 | [TK]D-Fender | dacm_work: You have a mix of analog line, digital trunks, some ITSPs, a bunch of various phones? Place a call with Phone X, it can try any ersource you want. If its unavailable you can choose another, or do whatever you want |
| 14:34.05 | dacm_work | telnettech: So the line would be tied up unless I pay for multiple channels? |
| 14:34.05 | Nugget | telnet is eeeeeeevil! |
| 14:34.25 | telnettech | dacm_work: correct |
| 14:34.32 | drmessano | [TK]D-Fender: If I need to replace a _ with a space in the middle of a variable, what function would be the cleanest way to do it. Looks like there's no REPLACE until 1.8 and I see quite a few different (ugly) ways to do it |
| 14:34.32 | dacm_work | telnettech: Thanks. |
| 14:34.54 | telnettech | np |
| 14:35.12 | [TK]D-Fender | drmessano: Macro that loops char by char for the length. Old school |
| 14:35.26 | [TK]D-Fender | drmessano: Because REGEX only matches |
| 14:35.50 | dacm_work | [TK]D-Fender: I was considering the case of a single analogue line coming in to the PBX and nothing else. |
| 14:35.53 | drmessano | Ouch |
| 14:36.14 | [TK]D-Fender | dacm_work: * can only do with that analog line the same things you can do with a regular phone plugged into it |
| 14:36.25 | [TK]D-Fender | dacm_work: It doesn't make it any more capable |
| 14:37.08 | dacm_work | [TK]D-Fender: Yeah. I didn't know what was actually possible with an analogue line. (Telephony noob!) |
| 14:37.16 | dacm_work | Thanks for the help guys. |
| 14:37.25 | [TK]D-Fender | dacm_work: and people have multiple cal;ls from the same "number" because the numebr that is sent is jsut callerid. it can be made ot look like "whatever" |
| 14:38.11 | [TK]D-Fender | dacm_work: those with MULTIPLE analog lines usually set up a hunt group and have the telco cycle through them on incoming to the primary (pilot) number, and set the callerid on outbound from all of them to this same number |
| 14:38.40 | [TK]D-Fender | dacm_work: Other techs allow you to set CID yourself like PRI's, ITSPs, etc |
| 14:39.26 | dacm_work | [TK]D-Fender: I think I understand. |
| 14:39.54 | [TK]D-Fender | dacm_work: With analog its all about the telco doing this for you. Other techs give you more direct control |
| 14:40.24 | dacm_work | I see. |
| 14:41.06 | *** join/#asterisk iratik (~itariki@74-84-99-12.client.mchsi.com) |
| 14:41.14 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 14:42.26 | iratik | How can I prevent sip extensions from registering outside of the local network? |
| 14:43.04 | dacm_work | Basically we are a small business with currently 3 analogue lines that are all public and used for different things. It would be nice to be able to stop (or at least reduce the chance of) a line becoming engaged. Are there any techs that you would suggest for that? (Obviously we are still quite small so cost is a factor.) |
| 14:44.21 | [TK]D-Fender | dacm_work: If each has its own number then this poses issues. You can't choose to have them look unique when calling out for one purpose and then look like a "primary" when used for another purpose |
| 14:45.02 | [TK]D-Fender | dacm_work: So do you needs these 3 lines to each have their own unique number published for processing incoming calls differently? |
| 14:46.26 | dacm_work | [TK]D-Fender: Out-going calls don't matter so much. But incoming calls we have public unique numbers for each line/ |
| 14:46.48 | [TK]D-Fender | dacm_work: Well if you only want the outgoing to look like the same, then ask your telco |
| 14:46.50 | dacm_work | And they are processed differently. |
| 14:47.34 | [TK]D-Fender | dacm_work: But you WILL get a busy on incoming unless you set up a hunt group. But then again you'll never know what # was dialed if you get a call on a cascaded line |
| 14:48.10 | Katty | OHAI |
| 14:48.15 | dacm_work | Sure but that won't allow us to take multiple calls on the same line. Which is what I'd like ideally. So I was wondering if there is a tech better than analogue for this. |
| 14:48.50 | Katty | HOW ARE YOU LOVELIES TODAY |
| 14:49.15 | raden | morning Katty |
| 14:49.17 | raden | FML |
| 14:49.33 | [TK]D-Fender | dacm_work: ISDN PRI, or get an ITSP |
| 14:49.59 | raden | to work I go blah |
| 14:50.18 | Katty | raden: fml?! )= |
| 14:50.23 | Katty | raden: i hope your day gets better. |
| 14:50.37 | dacm_work | [TK]D-Fender: Great I'll look into those. ;-) |
| 14:50.42 | dacm_work | [TK]D-Fender: Thanks again! |
| 14:51.06 | raden | Katty, we will see ... fighting the depression |
| 14:51.21 | *** join/#asterisk s4msung (~s4msung@dice.s4msung.de) |
| 14:52.35 | anonymouz666 | iratik: deny/permit settings |
| 14:53.06 | iratik | Yep.. I can use those in [general] in sip.conf right? Then just allow my internal network and each of my trunking providers? |
| 14:53.49 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 14:54.01 | anonymouz666 | Look at the sample. It will show where can be used and its syntax |
| 14:54.33 | iratik | ty btw |
| 14:54.57 | wcselby | o/ |
| 14:57.22 | kc8pxy | [TK]D-Fender: PRI sounds a bit of overkill for his situation:) |
| 14:57.24 | Katty | hugs on wcselby |
| 14:57.32 | kc8pxy | [TK]D-Fender: but it would work :) |
| 14:57.59 | wcselby | :) hi Katty |
| 14:58.36 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:58.47 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 14:59.19 | Katty | wcselby: how are you deary |
| 14:59.46 | wcselby | Katty - good good, working in the trenches again this morning. yourself? |
| 15:00.00 | Katty | wcselby: trenches? |
| 15:00.06 | Katty | wcselby: doing good... little hungry tho |
| 15:00.35 | wcselby | heh, i grabbed some super healthy mickey d's on the way in (you know, the sausage egg and cheese mcgriddle....probably only 700 calories) |
| 15:00.41 | wcselby | lol |
| 15:00.46 | *** part/#asterisk phretor (~phretor@yummi-ng.elet.polimi.it) |
| 15:00.51 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:01.01 | Katty | yikes. |
| 15:01.18 | Katty | i had a homemade sammich. egg, cheese, pepperoni, and mushrooms...in a grilled cheese kind of setup |
| 15:01.32 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:01.39 | Katty | those mcgriddles are so bad for your health |
| 15:01.49 | Katty | hey KavanS |
| 15:02.10 | wcselby | ahh, it's only 560 calories according to their website |
| 15:02.13 | wcselby | heh |
| 15:02.15 | wcselby | "only" |
| 15:02.17 | Katty | well. |
| 15:02.21 | Katty | that's not /too/ bad. |
| 15:02.27 | Katty | i usually eat about 500 a meal |
| 15:02.30 | wcselby | that what happens when you're running late |
| 15:02.37 | Katty | but i'm sure it was packed with fat.... |
| 15:02.47 | Katty | yeah :< |
| 15:02.49 | wcselby | yeah, it's not the best thing in the world I could have eaten for breakfast |
| 15:02.56 | Katty | could have been worse. |
| 15:03.02 | wcselby | indeed |
| 15:03.06 | wcselby | i could have gotten the meal |
| 15:03.08 | Katty | could have a whopper combo at bk for breakfast. |
| 15:03.10 | wcselby | ;) |
| 15:03.13 | Katty | mhmm |
| 15:03.36 | Katty | that's okay tho. |
| 15:03.42 | Katty | every once in awhile isn't that bad |
| 15:06.09 | Katty | wcselby: did i tell you i had bloodwork done, and i'm still anemic? )= |
| 15:06.46 | wcselby | Katty - no, sorry I didn't hear about that |
| 15:06.54 | wcselby | that means you don't have enough iron, right? |
| 15:07.26 | Katty | yeah. |
| 15:07.53 | Katty | i did the vegetarian thing awhile back thinking it would help my health, but then when they did a blood test and found i was anemic i stopped. |
| 15:08.12 | Katty | i /thought/ that by eating meat again i would be okay...but apparently i'm still not getting enough iron |
| 15:08.18 | wcselby | ouch |
| 15:08.20 | prgmrchris | vegan is dangerous |
| 15:08.27 | wcselby | she didn't say vegan |
| 15:08.34 | Katty | i did vegan for about a year. and vegetarian for about a year. |
| 15:08.36 | prgmrchris | im just adding to what she said |
| 15:08.39 | wcselby | ahhh |
| 15:08.44 | wcselby | gotcha (both of you :)) |
| 15:08.50 | Katty | vegan was pretty dangerious |
| 15:08.55 | Katty | but you can do vegan if you do it right |
| 15:08.58 | n3hxs | Chew on nails Katty, |
| 15:09.09 | Chainsaw | Vegan just seems... tedious to me. |
| 15:09.13 | Katty | but it's very expensive, and difficult, to maintain those lifestyles. |
| 15:09.19 | Chainsaw | The amount of things you'd have to give up on seems extensive. |
| 15:09.40 | prgmrchris | animals eat other animals in the wild, its natural |
| 15:09.46 | prgmrchris | you shouldnt feel bad about eating meat |
| 15:09.51 | Chainsaw | (Steak & bacon are why I would never be a vegetarian, but I can understand that choice.) |
| 15:10.07 | prgmrchris | you dont see peta out in the jungle trying to save things from getting eating by lions/tigers |
| 15:10.35 | n3hxs | they would be eaten by lions/tigers ;) |
| 15:10.42 | prgmrchris | hopefully |
| 15:10.58 | iratik | Being vegan really opens you up to new creativity... Before I went vegetarian/vegan.. when i was hungry i could stop at hardees or mcdonalds ... Now i have to give it some thought ... felt like eating junk food was integrated into my "auto-pilot" ... Plus.. I love soy cheese! |
| 15:11.18 | DogBoy | soy cheese |
| 15:11.31 | coppice | you could stop at macdonalds in india |
| 15:11.38 | Katty | vegan/vegetarian for me was about the animals. |
| 15:11.49 | DogBoy | think of the whales |
| 15:11.55 | Katty | it was just an attempt to try an alternative diet to see how it would affect my health |
| 15:12.02 | coppice | whales. yummy |
| 15:12.05 | Katty | i was hoping for a positive outcome, but it just didn't happen. |
| 15:12.11 | Katty | so whatever. it's fine. |
| 15:12.19 | DogBoy | so you went back to raw meat |
| 15:12.24 | n3hxs | How many calories in a humpback? |
| 15:12.33 | Katty | DogBoy: well honestly, i'm not a big fan of meat to start with. |
| 15:12.37 | Katty | DogBoy: i'd rather eat pasta. |
| 15:12.50 | Katty | DogBoy: i'd rather eat mac and cheese than a steak |
| 15:12.53 | DogBoy | I'm vegetarian also |
| 15:12.55 | wcselby | Katty - so does taking iron supplements help any? |
| 15:13.10 | wcselby | my wife used to take slowFE when she was having anemic issues |
| 15:13.15 | Katty | wcselby: i'm hoping so... they're going to run another blood test in about 3 months to see |
| 15:13.20 | wcselby | gotcha |
| 15:13.33 | Katty | wcselby: i'm making myself eat more iron rich items too |
| 15:13.36 | wcselby | just be careful not to take too much, it can mess up your stomach |
| 15:13.37 | DogBoy | I always get a kick out of people making a distinction between vegan and vegetarian |
| 15:13.46 | DogBoy | cause it's never right |
| 15:13.47 | Katty | wcselby: okay, i didn't know that...the pill they have me on is 65mg |
| 15:13.52 | iratik | I heard that IRon is really easy to overdose on .. and are there any security auditing tools to test a public IP and produce a list of issues with that public IP? |
| 15:14.07 | Katty | wcselby: and the pharmacist says that you can take 1-3 pills a day just depending on how well your body absorbs it |
| 15:14.17 | iratik | Easy to crack sip passwords etc.. |
| 15:14.21 | [TK]D-Fender | PETA : People for the Eating of Tasty Animals |
| 15:14.24 | Katty | wcselby: so i don't really think that adding a few iron rich items to my diet will really cause for an overdose. |
| 15:14.42 | wcselby | iratik - you mean like nmap? |
| 15:15.18 | *** join/#asterisk mcr_mv (~mcr_mv@239.Red-80-39-76.staticIP.rima-tde.net) |
| 15:15.26 | wcselby | Katty - hmm, i didn't really mean overdose per se - it just would hurt her stomach, and sometimes caused constipation |
| 15:15.28 | iratik | neat little utility |
| 15:15.30 | eppigy | GOOD MORNING |
| 15:15.46 | wcselby | which in turn would hurt her stomach more |
| 15:15.49 | wcselby | etc etc |
| 15:16.03 | Katty | wcselby: ah yes, the dr did tell me about that. |
| 15:16.09 | Katty | wcselby: he was all drink moar water! |
| 15:16.32 | wcselby | she's got other issues though - mostly hyper (or hypo) glycemia |
| 15:16.37 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 15:16.40 | Katty | nods. |
| 15:16.48 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:16.48 | wcselby | hypo I think (i get them mixed up) |
| 15:17.01 | Katty | they did check my blood sugar levels with another blood test....they seem to be right in the middle (90) so they're not concerned at all |
| 15:17.09 | wcselby | yeah |
| 15:17.22 | Katty | which i'm really really happy about considering i have 2 diabetics in my immediate family |
| 15:17.32 | wcselby | yeah |
| 15:17.32 | Katty | sister and dad. |
| 15:17.35 | wcselby | wow |
| 15:17.38 | wcselby | keep an eye on that |
| 15:17.39 | wcselby | obviously |
| 15:17.41 | wcselby | heh |
| 15:17.47 | Katty | most definately. i take after mom a lot. |
| 15:17.51 | wcselby | sorry, don't mean to sound like your doc / mom / whatever :) |
| 15:17.52 | Katty | a whole lot. |
| 15:18.09 | Katty | and she doesn't have any diabetic problems. good levels, including good blood pressure |
| 15:18.24 | Katty | my blood pressure has been so crazy these last few months, i was bordering hypertension. 145/75 |
| 15:18.49 | wcselby | yikes, that's high |
| 15:18.59 | Katty | after i got out of my stressful relationship, it went down to about 122/66 |
| 15:18.59 | wcselby | i'm pretty overweight, and I don't even go that high |
| 15:19.07 | wcselby | heh, yeah I can understand that |
| 15:19.10 | Katty | and i work out at curves and what not. |
| 15:19.14 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 15:19.16 | wcselby | yeah, i don't |
| 15:19.17 | wcselby | :) |
| 15:19.23 | Katty | when i went to the dr last week, it was 102/75 |
| 15:19.29 | wcselby | nice! |
| 15:19.32 | wcselby | congrats |
| 15:19.47 | Katty | so it's definately getting back down to /normal/ levels. i'm just concerned tho, because i also stopped taking my BC to see if it was having any affect on the blood pressure |
| 15:19.57 | Katty | now i have to get back on the BC and see if it's going to spike the levels on me |
| 15:20.02 | wcselby | yeah |
| 15:20.19 | wcselby | which one do you take? they all are a little different, right? |
| 15:20.23 | Katty | stress can do a lot of things, and this has been one hell of a month |
| 15:20.27 | wcselby | my wife hasn't found any she likes |
| 15:20.39 | Katty | yeah they're all a little different. i'm on Yaz just because it seems to help with my mood swings. |
| 15:20.42 | wcselby | they all have some kind of negative side effect for her |
| 15:20.56 | wcselby | yeah, that's the one everyone seems to be on these days |
| 15:21.03 | Katty | oh yeah, mine kills my sex drive |
| 15:21.08 | Katty | it's terrible |
| 15:21.26 | Katty | but i'm way too squeamish to get the depo shot..soo.... |
| 15:21.39 | Katty | they might change it up, but..idk |
| 15:21.51 | Katty | i guess i don't really care at this point whether or not i have a sex drive lol |
| 15:23.21 | Katty | wcselby: do you know what they have your wife on? |
| 15:24.06 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:24.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.20 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:24.20 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:24.27 | wcselby | Katty - she's not on anything at the moment |
| 15:24.37 | Katty | ah. |
| 15:24.43 | Katty | hey Qwell |
| 15:25.00 | Katty | wcselby: well if she finds somethin that works well, i'd be interested in knowing what it is |
| 15:25.15 | wcselby | i'll let you know |
| 15:25.19 | Katty | :> |
| 15:25.26 | wcselby | the problem with those is they take like a month or two before they start to work, right? |
| 15:25.29 | *** part/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:25.37 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:25.37 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 15:27.35 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 15:28.04 | Katty | wcselby: yeah they asked me to wait a full month |
| 15:28.23 | *** part/#asterisk kleofas (~kleofas@router.dir.pl) |
| 15:28.33 | Katty | wcselby: so it does take a little bit to start working |
| 15:28.39 | wcselby | yeah |
| 15:28.48 | ijpalmer | hi, Is there a command to logout all users from all queues, I'm running * 1.4.27 |
| 15:29.41 | [TK]D-Fender | ijpalmer: No. |
| 15:29.53 | [TK]D-Fender | ijpalmer: You'll have to script this yourself |
| 15:30.21 | ijpalmer | D-Fender: OK thanks |
| 15:30.46 | *** part/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 15:30.53 | wcselby | wouldn't restarting the queue service with "persistentmembers=no" work? |
| 15:31.06 | eppigy | another satisfied chatter |
| 15:31.12 | *** join/#asterisk Tim_Toady (~moi@178.128.17.211.dsl.dyn.forthnet.gr) |
| 15:33.08 | [TK]D-Fender | wcselby: taht would only affect on complete restart |
| 15:33.14 | [TK]D-Fender | wcselby: Which I doubt he wants to do |
| 15:33.17 | wcselby | [TK]D-Fender - ahhh, gotcha |
| 15:33.32 | wcselby | just throwing out an idea, had no idea if it would work or not |
| 15:34.56 | Katty | HELLO DAVE |
| 15:35.07 | Katty | hugs eppigy |
| 15:35.16 | Katty | eppigy: i decided no btw. |
| 15:35.37 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 15:35.48 | wcselby | do iMac's display Flash okay? I know there's like a rift between Apple / Adobe recently..... |
| 15:36.18 | [TK]D-Fender | s/recently/most of the last decade/ |
| 15:36.23 | Katty | :P |
| 15:36.37 | wcselby | lol |
| 15:36.44 | wcselby | hence my question |
| 15:37.04 | wcselby | i've got a client that has an office full of imacs. i'm wondering if fop2 will display okay for them |
| 15:37.42 | Katty | would take long to setup a demo and just check it |
| 15:37.50 | wcselby | heh |
| 15:37.53 | Katty | you'd only have to add like...1 extension |
| 15:38.24 | wcselby | yeah, but then the client would ask "why is there only one extension on there, I thought you said I could see everything..." blah blah blah...difficult client |
| 15:38.35 | wcselby | now, if I had an imac handy............ |
| 15:38.41 | wcselby | which I might, now that I think about it |
| 15:38.59 | wcselby | searches client main office for an imac |
| 15:40.37 | Katty | wcselby: lol |
| 15:40.40 | Katty | wcselby: yeah i know how that goes. |
| 15:43.01 | eppigy | Katty: rude |
| 15:43.12 | *** join/#asterisk iamy_china (~iamy_chin@221.223.53.171) |
| 15:43.27 | eppigy | that is a shame |
| 15:43.28 | *** part/#asterisk iamy_china (~iamy_chin@221.223.53.171) |
| 15:44.03 | Katty | eppigy: yes it is a shame. |
| 15:44.58 | eppigy | pretty much what i expected though i guess |
| 15:46.57 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 15:48.39 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
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| 15:56.25 | telnettech | Katty: Have you talked with jaytee in a while? I have seen him on here forever and a day |
| 15:57.59 | *** join/#asterisk ajm0716 (~ajm0716@c-76-97-74-68.hsd1.ga.comcast.net) |
| 15:58.08 | Qwell | Deeewayne: ping! |
| 15:58.30 | Katty | telnettech: hmm, no. i haven't talked to him in awhile |
| 15:58.44 | Deeewayne | Qwell, hey, what's up ? |
| 15:59.01 | Qwell | http://failblog.org/2010/08/01/epic-fail-photos-this-should-be-happening-right-now-win/ |
| 15:59.03 | Qwell | thought you might enjoy that |
| 15:59.51 | telnettech | katty: im gonna have to dig into my classroom folder and find his number and check on him |
| 16:00.21 | Katty | telnettech: i think he's on my fb, let me check. |
| 16:00.26 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:01.17 | Katty | telnettech: yeah he's still making posts. |
| 16:01.25 | Katty | telnettech: there was a post on his page from yesterday |
| 16:01.44 | Deeewayne | bear hugs Katty |
| 16:01.56 | Katty | telnettech: would you like me to let him know you were asking about him? (= |
| 16:02.13 | Katty | hugs Deeewayne |
| 16:02.25 | telnettech | yeah....tell i said hi....havent had much time to socialize with me moving from 1 city to another |
| 16:02.33 | Katty | nods |
| 16:02.34 | Katty | will do |
| 16:04.49 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 16:05.23 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:09.22 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
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| 16:16.54 | *** join/#asterisk pwntang (~pwntang@host217-41-0-107.in-addr.btopenworld.com) |
| 16:16.59 | pwntang | hi there |
| 16:17.41 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 16:17.46 | pwntang | does anyone know what "noisy feedback tells" means within the SIP debug please? |
| 16:38.23 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 16:44.03 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 16:49.34 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 16:49.36 | nextime | hello |
| 16:52.10 | nextime | after 2 days working, with asterisk 1.6.9 and latest dahdi, i have a te405p card that won't work anymore with 4 different E1 pri from 4 different providers. All 4 spans say PRI span 1/0: Provisioned, Down, Active, enabling debug on dahdi and wct4xxp modules say: [20634.579962] wct4xxp: LOF/LFA detected on span 1 but debouncing for 2500 ms, [20634.579987] wct4xxp: LOS detected on span 1 but debouncing for 2500 ms and [20634.593108] Detected loss of |
| 16:52.19 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 16:52.21 | nextime | anyone have an idea on what can be the problem? |
| 17:06.26 | *** join/#asterisk chuckz (~lechuck@93-40-114-140.ip38.fastwebnet.it) |
| 17:08.53 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
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| 17:14.28 | *** join/#asterisk p0ns (~p0ns@190.21.130.173) |
| 17:15.25 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 17:15.37 | p0ns | hi guys, i was wondering if there is any way of debugging a 401 unauthorized error on a sip client. I'm having this error with a softphone on a specific internet connection but the same softphone and config connected via another connection works without issues. |
| 17:16.18 | p0ns | is there any way of knowing why asterisk is rejecting the register request only when connected via this specific connection? |
| 17:16.47 | beardy | Because you connect to, and/or run Asterisk behind a NAT, on one or both ends. |
| 17:17.14 | p0ns | the softphone is always nated |
| 17:17.20 | p0ns | at home, with nat, it works fine |
| 17:17.31 | p0ns | via 3g connection, also nat, works fine |
| 17:17.38 | p0ns | but at work, with nat, it doesn't register |
| 17:17.50 | p0ns | the servers gets the register request but it denies it |
| 17:18.10 | beardy | The firewall at work isn't stateful then? |
| 17:18.42 | p0ns | don't know, it's just a wrt54gl router |
| 17:18.43 | beardy | It doesn't allow the reply packet back in to your phone. |
| 17:18.52 | beardy | (Perhaps.) |
| 17:19.42 | *** join/#asterisk hardwire (~spencersr@69-161-26-211.static.acsalaska.net) |
| 17:21.44 | p0ns | oh, that's a fact... just enabled my softphone's sip log and it's not getting any request back... |
| 17:22.47 | p0ns | somehow the router is changing the IP on the Contact: Header |
| 17:22.59 | p0ns | and Via |
| 17:26.04 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 17:27.47 | WIMPy | p0ns: Don't load nf_nat_sip on the router. |
| 17:28.37 | p0ns | it's using default firmware, couldn't find any voip related config :/& |
| 17:29.23 | [TK]D-Fender | p0ns: * shouldn't even be caring about the contact header |
| 17:29.43 | [TK]D-Fender | (if its eh softphone side that is being altered) |
| 17:30.03 | WIMPy | You're screwed then. You could try with nat=no for that account, but that's a rather bad situation. |
| 17:30.24 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 17:30.45 | p0ns | [TK]D-Fender, now i think it's not asterisk caring about it, nor the 401, but the softphone isn't getting any replies from * |
| 17:31.03 | [TK]D-Fender | p0ns: Where do we get to SEE this conversation to debug? |
| 17:32.13 | p0ns | i'll try to upload it |
| 17:32.45 | [TK]D-Fender | ~pb |
| 17:32.46 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:32.48 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 17:33.17 | p0ns | but the fact is that the softphone package is send with the internal, private IP on the Contact and Via headers, and on asterisk it's log as replaced with the public ip adress |
| 17:42.17 | p0ns | http://pastebin.com/0TvFtJPA |
| 17:57.56 | *** join/#asterisk hfb (~hfb@pool-98-112-109-237.lsanca.dsl-w.verizon.net) |
| 18:04.52 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 18:05.05 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 18:07.01 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 18:07.26 | spiceycurry | I did not specify any users or extensions in my sip.conf, as I read that asterisk will send all incoming calls that do not match a user or peer to the context I defined in my sip.conf. Providing that I have my sip.conf file setup, and my extensions.conf, will this allow a 'catch all' answer? |
| 18:07.31 | spiceycurry | Sotrry |
| 18:07.33 | spiceycurry | Sorry |
| 18:07.48 | spiceycurry | I have extensions setup with a catch all _X. |
| 18:08.30 | spiceycurry | I have a rule in extensions.conf that will pickup any unhandled call. |
| 18:08.40 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 18:09.13 | [TK]D-Fender | BAI BAI |
| 18:09.28 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 18:11.18 | *** join/#asterisk oej (~olle@109.58.22.139.bredband.tre.se) |
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| 18:11.57 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 18:14.34 | [TK]D-Fender | [14:07]<spiceycurry>I have extensions setup with a catch all _X. <- isn't a "catch-all". It says precisely what it will catch, and that is clearly not "all". |
| 18:15.40 | drmessano | Yeah, works fine as long as they're not calling nobody |
| 18:15.55 | spiceycurry | there is only inbound calls |
| 18:16.06 | spiceycurry | I catch every extension with _X. |
| 18:16.16 | [TK]D-Fender | spiceycurry: No, you don't |
| 18:16.23 | drmessano | ^^^ |
| 18:16.42 | [TK]D-Fender | spiceycurry: You appear to have a real misunderstanding of what that pattern covers. |
| 18:16.45 | spiceycurry | Hmm, could you please explain, as I am a bit confused |
| 18:17.01 | spiceycurry | The asterisk book says that it catches all extensions |
| 18:17.07 | spiceycurry | I've tested it also |
| 18:17.16 | spiceycurry | it indeed catches all as far as I can tell |
| 18:17.18 | [TK]D-Fender | spiceycurry: Since you think you know what it does, how about you tell US. What EXACTLY does "_X." mean? |
| 18:17.26 | [TK]D-Fender | spiceycurry: Berak it down |
| 18:17.28 | [TK]D-Fender | break* |
| 18:17.33 | spiceycurry | Ok, one sec. |
| 18:18.00 | [TK]D-Fender | (This should take time. Means you clearly are trying to actually read it for yourself again and don't actually KNOW the answer) |
| 18:18.11 | [TK]D-Fender | Shouldn't* |
| 18:18.26 | spiceycurry | Its in the asterisk book |
| 18:18.40 | [TK]D-Fender | spiceycurry: Instead of your head. |
| 18:18.50 | spiceycurry | stop being a dick, and give me a second |
| 18:19.16 | [TK]D-Fender | spiceycurry: You've had 2 minutes, and wasted much of them writing back instead of just coming back with the answer :) |
| 18:20.12 | [TK]D-Fender | notes another person easily distracted from actually finishing looking for their answers |
| 18:20.50 | spiceycurry | I believe that _X. will match any extension that starts with a number, and match anything after with the 'wildcard' . |
| 18:21.11 | spiceycurry | I did not want to use _. as it made no sense |
| 18:21.35 | spiceycurry | page 138, and its in the ~book |
| 18:21.49 | [TK]D-Fender | spiceycurry: What is this "believ" nonsense? |
| 18:22.16 | spiceycurry | read the book |
| 18:22.19 | [TK]D-Fender | spiceycurry: Tell me what EACH character in that exten means |
| 18:22.35 | spiceycurry | the '_' specifies a pattern |
| 18:22.40 | [TK]D-Fender | spiceycurry: I read the book. Both releases of it. I've also met the authors. |
| 18:23.01 | spiceycurry | the X will match any number between 0-9 |
| 18:23.22 | spiceycurry | and the '.' is wildcardmatch that matches one or more characters |
| 18:23.37 | [TK]D-Fender | spiceycurry: Therefor it will not catch ALL. |
| 18:23.55 | spiceycurry | I am only interested in real number matches |
| 18:24.08 | [TK]D-Fender | spiceycurry: Also will not catch all NUMBERS either |
| 18:24.18 | spiceycurry | How so? |
| 18:24.25 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 18:24.27 | [TK]D-Fender | spiceycurry: 5 <------------- |
| 18:24.40 | [TK]D-Fender | spiceycurry: Will. No. Match. |
| 18:24.42 | [TK]D-Fender | Not* |
| 18:25.04 | spiceycurry | Ok, well, I am only interested in numbers matching _X. |
| 18:25.13 | spiceycurry | so yes, 2 or more |
| 18:25.28 | [TK]D-Fender | spiceycurry: Better. |
| 18:25.58 | [TK]D-Fender | spiceycurry: So... now... what is you point in all of this you brought up? |
| 18:26.35 | adyn | ok so I haven't started the research so just a quick yes or no will work, is load ballancing an asterisk setup possible? |
| 18:27.06 | [TK]D-Fender | adyn: Depending on your precise definition and circumstances, yes |
| 18:27.58 | spiceycurry | I was wondering if my context=unidentified-sip (under [general]) will catch all calls where it cannot match a user or peer |
| 18:28.46 | spiceycurry | ps- I also met Jim here in Toronto. We often call him for help with our other system, but he has not messed with FFA as of yet. |
| 18:28.54 | [TK]D-Fender | spiceycurry: that is where call will go if they don't match a peer and you allow anonymous calls |
| 18:29.13 | spiceycurry | ah ok, I must be missing the anonymous setting |
| 18:29.16 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:29.35 | ruben23 | hi guys any help how to on setting u IVR for asterisk. |
| 18:29.45 | Qwell | ~book |
| 18:29.46 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:29.46 | [TK]D-Fender | spiceycurry: Why are you even GUESSING? You also seem to have an issue actually looking at a CALL |
| 18:29.50 | Qwell | ruben23: ^ |
| 18:30.07 | [TK]D-Fender | Qwell: Whats scary is how many years he's been in here.... |
| 18:30.18 | [TK]D-Fender | Qwell: And is still miles away from 101 stuff |
| 18:30.53 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 18:30.56 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:31.04 | spiceycurry | I've been in here almost 3 months I think |
| 18:31.59 | spiceycurry | [TK]D-Fender |
| 18:32.02 | spiceycurry | woops |
| 18:32.04 | [TK]D-Fender | spiceycurry: Wasn't referring to you |
| 18:32.04 | Qwell | spiceycurry: not you |
| 18:32.08 | spiceycurry | ah ok |
| 18:32.25 | spiceycurry | Does Jim come in here? |
| 18:32.41 | [TK]D-Fender | spiceycurry: So... where do we see you actually looking at a call you think should have been processed differently? |
| 18:32.45 | [TK]D-Fender | spiceycurry: Very rarely |
| 18:33.39 | spiceycurry | [TK]D-Fender: I am getting calls via my pri fine, they get caught and processed as I'd like. What I am trying to do is also handle all incoming SIP calls and handle them much as I am my PRI calls. |
| 18:34.02 | [TK]D-Fender | spiceycurry: So... where do we see a failure to look at? |
| 18:35.55 | spiceycurry | I think it was a firewall issue, the admin just came over and told me he had not forwarded 5060 to my ip |
| 18:36.18 | spiceycurry | Is anyone able to try and make a call to anony@69.49.114.20:8191? |
| 18:36.25 | spiceycurry | (thats port 8191) |
| 18:36.58 | [TK]D-Fender | spiceycurry: So basically... you didn't even LOOK for the call. |
| 18:37.05 | spiceycurry | I was not seeing one |
| 18:37.16 | [TK]D-Fender | spiceycurry: Were you sitting at CLI with SIP DEBUG enabled? |
| 18:37.24 | spiceycurry | I am right now |
| 18:37.33 | [TK]D-Fender | spiceycurry: NOW.. oh yeah... great. |
| 18:37.46 | spiceycurry | Sorry, I have been |
| 18:38.02 | [TK]D-Fender | spiceycurry: So... do you see anything? |
| 18:38.08 | spiceycurry | no |
| 18:38.15 | spiceycurry | except a flashing prompt |
| 18:38.20 | [TK]D-Fender | spiceycurry: Then * has nothing to do with this |
| 18:38.33 | spiceycurry | I suppose I'd need a call to come through right? |
| 18:38.36 | spiceycurry | :O |
| 18:38.41 | [TK]D-Fender | SMRT |
| 18:38.42 | spiceycurry | I am waiting for one |
| 18:38.48 | spiceycurry | call em baby |
| 18:38.52 | spiceycurry | *me |
| 18:39.06 | spiceycurry | Free Sex: anony@69.49.114.20:8191 |
| 18:39.48 | [TK]D-Fender | WRONG PLUG-IN |
| 18:39.55 | spiceycurry | lol |
| 18:41.25 | spiceycurry | Anyone know of a free Sip to Sip service? |
| 18:42.04 | Qwell | uhh |
| 18:42.26 | spiceycurry | Other than my own server? |
| 18:42.29 | Qwell | why would you have to pay somebody to do a SIP to SIP call? |
| 18:43.07 | spiceycurry | I want to test my sip from outside, what services could I use? |
| 18:44.24 | hardwire | this is driving me up the wall.. I can't seem to turn off "telephone-event" media |
| 18:44.37 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:44.47 | hardwire | I just want be in the middle on these calls.. not relay progress (over g.729??!) |
| 18:44.53 | idespinner | spiceycurry, http://wiki.ekiga.org/index.php/Fun_Numbers |
| 18:45.25 | spiceycurry | Thanks a bunch |
| 18:45.45 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 18:47.31 | *** join/#asterisk ccesario (~ccesario@189-29-39-218-ac.cpe.vivax.com.br) |
| 18:48.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:48.29 | Naikrovek | eyeballs Polycom SPIP 3.3 firmware release notes... |
| 18:48.33 | Naikrovek | ooOOoohh |
| 18:50.16 | spiceycurry | Well, looks ok (errors are because I don't have the capacity to send a fax over sip) http://pastebin.com/jbgcnmLt |
| 18:50.27 | Naikrovek | this firmware is smarter - config options are smarter, i mean |
| 18:50.32 | spiceycurry | I just called, and saw that come up |
| 18:50.48 | spiceycurry | Anyone know a Fax over Sip service I could use to insure I can receive my fax? |
| 18:50.51 | spiceycurry | (test fax) |
| 18:51.19 | [TK]D-Fender | spiceycurry: Try using a fax machine. |
| 18:51.39 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:51.44 | spiceycurry | I do not have a fax machine here, that will send a fax over sip |
| 18:52.20 | [TK]D-Fender | Naikrovek: Yup... they make things sound a fair bit scarier through... doesn't feel like just another release. They word it in a way to make you feel its for a whoel other protocol or hardware series |
| 18:52.46 | spiceycurry | Anyone know of a SIP client that can send faxes? |
| 18:52.48 | [TK]D-Fender | spiceycurry: Why woul the person calling you have to use SIP? |
| 18:53.10 | [TK]D-Fender | spiceycurry: I have a fax machine. I can send faxes anywhere I can CALL |
| 18:53.12 | spiceycurry | because, that is the way we have it setup. We have a customer who sends us fax over SIP |
| 18:53.23 | Naikrovek | [TK]D-Fender: yeah it's a bit more ominous. pre-3.3 configs won't work on 3.3 and vice versa. |
| 18:53.37 | spiceycurry | We also allow customers to send faxes via pstn to our pri (it all works) |
| 18:53.44 | [TK]D-Fender | spiceycurry: You want to know if you can RECEIVE a fax. It is meaningless as to where it comes from |
| 18:54.07 | spiceycurry | I can receive a fax through the PRI just fine, what I need to test is receiving a fax via SIP |
| 18:54.21 | [TK]D-Fender | Naikrovek: Polycom X.Y releases might break beween "y"'s already. quite known. |
| 18:54.27 | spiceycurry | Do you know of a SIP client that will send a fax over sip? |
| 18:54.29 | [TK]D-Fender | Naikrovek: "z"'s are OK though |
| 18:55.02 | [TK]D-Fender | spiceycurry: Who will be sending you SIP calls for this? |
| 18:55.03 | Naikrovek | yeah i know but never has it been like "your old configs will NOT work" but rather "you may experience issues" |
| 18:55.15 | Naikrovek | just weird |
| 18:55.17 | [TK]D-Fender | Naikrovek: No.. the old stuff could stop you dead as well |
| 18:55.53 | Naikrovek | k |
| 18:55.55 | spiceycurry | Our customers customers. They receive faxes from their pstn and route them again to us. |
| 18:55.56 | Naikrovek | so no big deal then |
| 18:56.19 | spiceycurry | Our customers receive faxes, and resend them to us via SIP |
| 18:57.51 | [TK]D-Fender | spiceycurry: What are THEY using to send them to you? |
| 18:58.27 | [TK]D-Fender | spiceycurry: And why SIP? Better to receive at their side and transfer in some non-flaky manner directly. |
| 18:58.38 | spiceycurry | I believe they use Fax For Asterisk. And we |
| 18:58.44 | adyn | I've had nothing but problems with FAX'ing over SIP trunks |
| 18:58.56 | spiceycurry | They are using SIP, and there is nothing I can do to change it unfortunatly |
| 18:59.01 | spiceycurry | I have to work with what they have |
| 19:00.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:04.24 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.157) |
| 19:05.00 | spiceycurry | Customer said that they are getting faxes from pstn, and routing them through a meta switch gateway |
| 19:05.32 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:09.32 | dacm_work | Are sip-trunks far more cost effective than ISDN-PRI or am I misunderstanding something? |
| 19:12.30 | Chainsaw | dacm_work: They are likely to be cheaper for you, as ISDN-PRI tends to flow into the incumbent telco. |
| 19:13.03 | Chainsaw | dacm_work: Where-as SIP could terminate at a VoIP service provider... or even the party you wish to speak to. |
| 19:13.40 | dacm_work | Chainsaw: So if you're a telco ISDN-PRI is cheap? :-) |
| 19:14.32 | Chainsaw | dacm_work: For you, yes. For whatever poor shmuck you're billing... not so much. |
| 19:15.11 | dacm_work | Got it. |
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| 19:15.49 | *** join/#asterisk rootlinux (~rootlinux@201.143.28.20.dsl.dyn.telnor.net) |
| 19:15.58 | dacm_work | Guess I'll be going with SIP then. Can't believe how cheap it is, hope I'm not missing something. |
| 19:16.59 | dacm_work | BTW does SIP/VOIP use up much bandwidth? (Will probably bump up my bandwidth anyway.) |
| 19:18.30 | Chainsaw | dacm_work: It depends on what codec you use, and how many calls you plan on making simultaneously. |
| 19:19.22 | Chainsaw | dacm_work: Latency usually plays a bigger role than raw link speed. (An ideal link for VoIP has consistent low latency) |
| 19:19.30 | *** join/#asterisk Beltechs (~Beltechs@netblock-68-183-48-2.dslextreme.com) |
| 19:20.31 | Qwell | dacm_work: what you're missing, is that the reliability of ISDN is far higher than any protocol that goes over the public Internet. |
| 19:22.24 | WIMPy | dacm_work: It will depend heavily on your location. |
| 19:23.33 | dacm_work | Qwell: Thanks. |
| 19:24.45 | rootlinux | how can i put the caller and callee on hold before the time out with the following dial command? |
| 19:24.45 | rootlinux | exten => myext,11,Dial(${PSCHANNEL}${TOPHONE},60,gm)L(90000[:60000][:30000])M(feedback,${CALLHISTID})) |
| 19:24.57 | dacm_work | Chainsaw: Let's say that I wanted 5 concurrent calls with quality as good as an analogue line? Do you have any idea whether I would need a monster internet connection to handle that? |
| 19:26.12 | ariel_ | Anyone in South Florida area looking for work dealing with installation and configuration of asterisk system full time work? |
| 19:26.57 | Chainsaw | dacm_work: You'd need roughly 350kbit/sec symmetric. Mind your upload, consumer links tend to be rather dire in that respect. |
| 19:28.11 | rootlinux | any idea? |
| 19:28.14 | dacm_work | Chainsaw: Thanks. (I'm probably going to go for some business package with decent upload.) |
| 19:28.29 | Chainsaw | dacm_work: It would be best if the router you go for supports QoS (quality of service). |
| 19:29.00 | Chainsaw | dacm_work: That way you can have it prioritise your telephony traffic ahead of any other packets, to make sure the latency does not start varying wildly. (That is generally what kills a call or annoys people) |
| 19:29.27 | dacm_work | Chainsaw: Thanks for the advice. :-) |
| 19:29.40 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 19:29.42 | Chainsaw | dacm_work: You're welcome :) |
| 19:38.44 | [TK]D-Fender | rootlinux: Don't use L() to limit your call. in yuor M() spawn an external watcher that will issue an AMI Redirect to split them up. |
| 19:41.58 | Beltechs | Hello. Using *1.6 Im having sound quality issues. have a digium t1 card, TWBC PRI for incoming, sip trunk for out. |
| 19:42.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:42.34 | Beltechs | out bound calls have 2 second delay for the receiver to hear the caller |
| 19:43.11 | rootlinux | [tk]D-Fender: But the M() are executed only when the callee answers ... |
| 19:43.41 | [TK]D-Fender | rootlinux: Yes, and you can deduct the ring time fron there |
| 19:46.49 | rootlinux | [TK]D-Fender: Thanks man .. i will test ir |
| 19:47.28 | [TK]D-Fender | rootlinux: If your limit is already low you can set teh RINGING time in your dial statement to that max an backtrack accordingly |
| 19:47.35 | [TK]D-Fender | rootlinux: that way you still have the same net limit |
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| 19:55.49 | rootlinux | [TK]D-Fender: Waht i need is when the maxcall time is over in this case 90000 L(90000[:60000][:30000]) .. run a macro o agi script and put the caller and calle on hold.. |
| 19:56.37 | [TK]D-Fender | rootlinux: There is no "hold". You need to actually hijack the call |
| 19:57.57 | rootlinux | [TK]D-Fender: ok :( .. you know how can i read to do that? |
| 19:58.42 | [TK]D-Fender | ~book |
| 19:58.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 20:02.28 | rootlinux | [TK]D-Fender: ok man thanks for your help.. i will read and search about hijack a call with dial command |
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| 20:28.36 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:28.38 | [sr] | hi mates |
| 20:28.57 | [sr] | i have a box that has dahdi 2.2.0, with a 4 port FXO card |
| 20:29.46 | [sr] | when i make a call using one of the FXO ports, when the person in the other side disconnects, the call on my side doesn't disconnect, who know if this is fixed on 2.3.0 ? |
| 20:30.28 | *** join/#asterisk xibalba (~reza@216.105.40.7) |
| 20:30.31 | xibalba | hey everyone |
| 20:30.43 | xibalba | i was wondering if anyone has had much luck getting voip phones to work over a linksys wrt54g |
| 20:31.01 | xibalba | it's been hit or miss for me, 1 location registers 1 phone, another location registers all 4 phones |
| 20:31.15 | xibalba | was looking to see if anyone had any insite on this router pertaining to voip |
| 20:31.17 | xibalba | and sip |
| 20:31.38 | nextime | [20634.593108] Detected loss of E1 alignment on span 0! |
| 20:31.44 | nextime | bad, bad thing. |
| 20:32.09 | [TK]D-Fender | [sr]: this isn't a DAHDI prblem, this is a "my line lacks CDS" |
| 20:32.10 | [TK]D-Fender | ~cds |
| 20:32.11 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 20:32.22 | nextime | [TK]D-Fender : yes i know |
| 20:32.49 | xibalba | ~sip |
| 20:32.50 | infobot | sip is probably Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
| 20:32.52 | xibalba | neato |
| 20:32.58 | [TK]D-Fender | nextime: And I clearly wasn't talking you :) |
| 20:32.59 | xibalba | ~wrt54g |
| 20:33.00 | infobot | well, wrt54g is a linksys WAP/Router that is very flexible if used with after-market firmware |
| 20:33.07 | [sr] | [TK]D-Fender: hum... nice, it may be 'cause its connected into my PBX analog line... and it doesn't have that info for sure, i'll try on a real POTS... second |
| 20:33.26 | [TK]D-Fender | checkout time, BBIAB |
| 20:33.29 | nextime | [TK]D-Fender : and also i know it isn't an other side issue, but a physical line issue, old and bad cables |
| 20:33.38 | nextime | ah |
| 20:33.40 | nextime | sorry :P |
| 20:34.09 | nextime | also my issue isn't on an analog line :D |
| 20:34.21 | nextime | anyway, it is a line problem and not a dahdi one :) |
| 20:35.00 | tzafrir | [sr], if none of the proper fixes work, use busydetect and the likes |
| 20:35.12 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 20:35.17 | xibalba | sorry to interupt again, but anyone know if the base model wrt54g is any good for voip? |
| 20:35.46 | nextime | xibalba : did you mean to install * on it? |
| 20:36.13 | xibalba | no i ment to use sip phones behind it. i have a hosted pbx, and am trying to get clients at a couple sites to connect back to it. |
| 20:36.18 | [sr] | tzafrir: hum tried with a direct POTS line and nothing, when the other side disconnects my side doesn't, but i still have one other line to try..tomorow i'll reask this matter |
| 20:37.00 | nextime | xibalba : if you mean just to use it as a network layer, yes, it work, and i don't see why it should'nt work as expected |
| 20:37.08 | tzafrir | [sr], the point is that the your end decides when the line is open. The the CO can only hint you to hang up |
| 20:37.28 | nextime | anyway, i have 2 clients here connected by a wifi link on a wrt54gl with openwrt on it |
| 20:37.30 | nextime | and it work. |
| 20:37.50 | xibalba | yeah i think this is just a wrt54g, i think that may be the problem. and i cannot put openwrt on it. they're 3,000 miles away |
| 20:38.07 | nextime | xibalba : i don't see any relevant difference |
| 20:38.32 | xibalba | could be the nat'ing algorithms used. |
| 20:39.33 | nextime | xibalba : of course you need to enable the nat helper on the * side and/or to use another way to bypass the nat problem |
| 20:39.37 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:39.52 | nextime | xibalba : but seriously, i don't think it can do any serious problem |
| 20:41.14 | nextime | old wrt54g use linux, so it is the same nat layer of the openwrt firmware, new one use vxworks if i rightly remember, and the vxwork nat isn't a problem in my experience |
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| 20:44.23 | xibalba | you know if there is any setting you nee dot change? |
| 20:44.30 | xibalba | at a couple of locations only 1 phone will register at a time |
| 20:44.40 | xibalba | when a 2nd phone attempts registration it kicks the first one off |
| 20:52.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:53.35 | *** join/#asterisk bbonora-vaio (~Ben_Bonor@c-67-160-101-81.hsd1.wa.comcast.net) |
| 20:54.23 | bbonora-vaio | Are there any Developers in the Seattle Area? |
| 21:04.52 | dacm_work | Could anyone here tell me the typical latency required to use a sip trunk effectively? |
| 21:05.24 | Chainsaw | dacm_work: <120ms at least. <80ms for good performance. |
| 21:05.53 | Chainsaw | dacm_work: The lower the better, as long as it is consistently that low. |
| 21:06.01 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 21:06.08 | bbonora-vaio | where would I find a consultant in the seattle area to help me setup an asterisk system for our office |
| 21:06.20 | dacm_work | Chainsaw: Thanks again. (My current connection is about 70ms, - and this connection feels sucky!) |
| 21:06.26 | defsdoor | buy me a plane ticket! |
| 21:06.54 | xibalba | i've had voip working fine under 200ms |
| 21:07.00 | xibalba | over 3g @ 70mph =P |
| 21:07.05 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 21:13.17 | bbonora-vaio | defsdoor: where do you live |
| 21:13.24 | defsdoor | uk |
| 21:13.28 | bbonora-vaio | haha |
| 21:13.46 | defsdoor | bbonora-vaio, install a minimal debian and let someone do it remotely |
| 21:14.01 | bbonora-vaio | that's fine too |
| 21:14.06 | bbonora-vaio | are you available |
| 21:14.17 | bbonora-vaio | and what is your charge |
| 21:14.37 | bbonora-vaio | I have asteriskNow already installed on a machine |
| 21:14.50 | defsdoor | I roll my own |
| 21:15.09 | bbonora-vaio | Here is what I need it to do |
| 21:15.36 | bbonora-vaio | we are getting rid of our office space and everybody will be working from home. |
| 21:16.01 | bbonora-vaio | when somebody calls our number we want the standard greeting to play with our current options |
| 21:16.31 | bbonora-vaio | if somebody dials my ext. it will ring my cell phone, skype account and home phone |
| 21:16.55 | bbonora-vaio | if I don't answer it will kick back to the system and prompt the user to leave a voicemail |
| 21:17.33 | bbonora-vaio | we don't need the ability to make calls from the server remotely |
| 21:18.04 | [TK]D-Fender | bbonora-vaio: No GUI handles Skype yet <- |
| 21:18.15 | [TK]D-Fender | bbonora-vaio: Should roll your own for this |
| 21:18.17 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 21:18.34 | [TK]D-Fender | bbonora-vaio: And of course you'll have to buy a SFA license |
| 21:19.35 | Beltechs | Should my dahdi system.conf file have |
| 21:19.43 | Beltechs | echocanceller? |
| 21:19.58 | Beltechs | echocanceller=mg2,1-23 |
| 21:24.46 | tzafrir | Beltechs, sure, why not? |
| 21:25.14 | tzafrir | It means that channels 1-23 will use the echo canceller mg2 |
| 21:30.34 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:34.04 | *** join/#asterisk ghenry (d49f3b55@pdpc/supporter/monthlybyte/ghenry) |
| 21:34.28 | ghenry | Hi, What's the default packaet size in 1.4? |
| 21:34.57 | Qwell | packet size of what? |
| 21:35.28 | ghenry | sample rate, sorry |
| 21:35.39 | Qwell | sample rate of what? |
| 21:35.39 | ghenry | 20ms or 30ms |
| 21:35.55 | Qwell | You need to be much more specific. |
| 21:36.03 | Corydon76-dig | It depends upon the codec used |
| 21:36.20 | Corydon76-dig | 20ms for most codecs, but 30ms for others |
| 21:36.22 | ghenry | Well, alaw, ulaw and g729a |
| 21:36.34 | Qwell | 20ms, 20ms, 30ms |
| 21:36.44 | ghenry | Sorry, I presume people can read my mind :-) |
| 21:36.52 | Qwell | of course, this assumes RTP |
| 21:37.13 | ghenry | Yeah, sorry. RTP, alaw and ulaw |
| 21:37.21 | ghenry | Cheers! |
| 21:38.23 | ghenry | Later. |
| 21:38.30 | *** part/#asterisk ghenry (d49f3b55@pdpc/supporter/monthlybyte/ghenry) |
| 21:38.41 | dacm_work | Does anyone here have a lot of lines that need to be accessible to one IP phone? Most phones claim to only support up to 4 lines. I'm wondering how to work around that... |
| 21:39.03 | [TK]D-Fender | dacm_work: Doesn't work like that |
| 21:39.21 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 21:39.40 | dacm_work | [TK]D-Fender: Good. How does it work if you don't mind me asking? |
| 21:39.43 | [TK]D-Fender | dacm_work: That usually descibes how many unique identities it may have, or simultaneous calls the phone can be on. |
| 21:39.53 | dacm_work | Ok. |
| 21:40.07 | [TK]D-Fender | dacm_work: you could have 100 lines, but how many CONVERSATIONS should one user be juggling? |
| 21:40.15 | Chainsaw | dacm_work: You will generally use a dial plan to route inbound calls to the phones. So even if you have 64 lines coming in, you will generally have an extension number on the phone that you route it to. |
| 21:40.44 | dacm_work | Cool. |
| 21:42.45 | dacm_work | Is there an easy way to know which number someone has dialled other than routing the calls? And is it possible for one handset to `pick up' for another? (I've seen that done before but don't know if it was some weird proprietary system.) |
| 21:44.44 | Chainsaw | Our system has a "Pick Up" key on every phone, yes. |
| 21:44.51 | Chainsaw | If you hear a phone ringing, you can steal the call. |
| 21:45.10 | nextime | dacm_work : for the first, just look at the channel variables ( ${EXTENSION} but not only this ) |
| 21:45.21 | nextime | for the pickup... just look at the features.conf |
| 21:46.01 | nextime | and the pickup group for the account config |
| 21:46.17 | dacm_work | Awesome. |
| 21:46.29 | dacm_work | Looks like I can do everything I want. |
| 21:47.06 | dacm_work | Just need to wiggle out of out current telephone + broadband contract and buy some hardware. :-) |
| 21:47.30 | dacm_work | Thanks for being so patient with me guys, you've been a real help! |
| 21:49.55 | nextime | is patient thanks to the local phone carrier that has some E1 pri working 1 days of two from 4 months.... |
| 21:50.04 | nextime | grrr |
| 21:50.50 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
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| 21:57.47 | Hydrant | are equipment suggestions on topic here ? |
| 21:58.53 | Chainsaw | To a degree. |
| 21:58.56 | Chainsaw | Please ask your question. |
| 21:59.48 | Hydrant | I'm looking for suggestions for VOIP phone that I might use for incoming / outgoing SIP (to a provider server) that doesn't require an asterisk server locally (to boot the phone)... if such a thing exists |
| 22:00.18 | Chainsaw | Sure, that exists. Do you want an analog line on it as well? |
| 22:00.26 | Chainsaw | DECT portable or fixed line? |
| 22:00.44 | Hydrant | if it has both great... but otherwise just voip is fine |
| 22:01.08 | Hydrant | that is, just SIP |
| 22:01.15 | [TK]D-Fender | Polycom > All |
| 22:01.36 | Chainsaw | Hydrant: The Siemens "chagall" platform has most of what you're looking for :) |
| 22:01.47 | Hydrant | right now I have a bunch of Polycom soundpoint 320s, but as far as I know I can't do this kinda setup with them |
| 22:01.50 | Chainsaw | Hydrant: Any specific demands? Answering machine? |
| 22:01.54 | Hydrant | nope |
| 22:02.06 | Chainsaw | Hydrant: Then even a C450 should do nicely. |
| 22:02.21 | [TK]D-Fender | Hydrant: You don't need a local server to boot one of those |
| 22:02.30 | Hydrant | well... maybe... a friend of mine is moving across the country... and so I want to set him up with a phone to keep in touch with his friends... but also setup some extension on my own asterisk system so that we can talk for free |
| 22:03.06 | Hydrant | [TK]D-Fender: right... I can boot it... but I couldn't find out how to use it for outgoing SIP with my provider... it didn't seem to be possible |
| 22:04.36 | [TK]D-Fender | Hydrant: SIP requires the same boring things for whatever you want to to use. |
| 22:04.44 | [TK]D-Fender | Hydrant: user, pass, IP. Teh End |
| 22:05.59 | Hydrant | [TK]D-Fender: hrrm... I couldn't figure it out, I'm not yet a voip Jedi... merely a voip droid ;-) |
| 22:07.43 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 22:07.44 | *** join/#asterisk disposable (disposable@blackhole.sk) |
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| 22:08.14 | [TK]D-Fender | Hydrant: Well... you show no hints no what you've tried and hearing "but I tried" really is just wind.... |
| 22:08.23 | disposable | is there a way to limit the maximum frequency of login attempts by a single sip peer? |
| 22:10.49 | disposable | to e.g. 1 per second? i have blocked all amazon ec2 ip ranges on my firewall, but i'd like to at least slow down the attempted break-ins by IPs i don't know about. like 188.165.224.13 at this very moment. |
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| 22:14.35 | Micc | Is there a way to tell a polycom ip450 to silent ring with an alert-info header or something? |
| 22:16.13 | Micc | Where is everyone today? |
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| 22:20.55 | disposable | Micc: let me know if you find out (about the polycom, not the location of everyone), ive been looking for a way to do the opposite |
| 22:21.51 | rootlinux | everyone is on vacation |
| 22:22.17 | nextime | no vacation this summer |
| 22:22.50 | Micc | disposable, what do you mean by the opposite? I know there is a way to do different rings on an aastra, and I know you can setup different ring tones on a polycom in the cfg file and associate that with an alert-info header. |
| 22:22.58 | chuckf | I'm having a vacation this summer |
| 22:22.58 | Micc | Maybe I should try doing that and making it silent. |
| 22:23.25 | Micc | I don't get a vacation this summer. I have to work every day, even weekends. |
| 22:23.28 | chuckf | nextime: so they do exisit this year |
| 22:23.33 | disposable | Micc: i'd like to force a phone to ring even when it's on silent. but i doubt it's possible. |
| 22:23.59 | nextime | chuckf : yes, but not before october |
| 22:24.12 | Micc | disposable, ah, well the better way to do that is if you could do the silent ring with a header then you never need to put the phone itself on silent. |
| 22:24.55 | chuckf | that's alright, depending on where you're going |
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| 22:26.19 | [TK]D-Fender | [18:14]<Micc>Is there a way to tell a polycom ip450 to silent ring with an alert-info header or something? <- yes |
| 22:26.35 | [TK]D-Fender | Micc: Same means as you do for paging and other distinctive rings |
| 22:26.53 | Micc | TKD-Fender, ok, I know how to do that, I just wasn't sure if it would work as aastra's don't allow that. |
| 22:27.34 | [TK]D-Fender | Micc: Aastra isn't exactly the Gold Standard ;) |
| 22:29.15 | Micc | TKD-Fender, I know, but I really like their phones a lot except for a few really stupid things they left out. |
| 22:30.34 | [TK]D-Fender | Micc: oh well... |
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| 22:31.10 | disposable | Micc: we have a customer with a large call centre. some of the people there tened to turn their ringer volume down and pretend their phone isn't in the currently ringing group. so i set the ringer volume to 100% in the cfg files;. but if they choose the silent ringtone, i just don't know how to overcome that. |
| 22:31.33 | disposable | s/tened/tend |
| 22:32.43 | Micc | Ah, that sounds like a personel problem. |
| 22:32.52 | Chainsaw | disposable: Trying to fix a social problem with technical measures is doomed to fail. |
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| 22:33.18 | disposable | Chainsaw: sad but true... they're always a step ahead |
| 22:33.28 | [TK]D-Fender | ChaI bet to differ .... DON'T TASE ME BRO!!! |
| 22:35.14 | Micc | Lots of technology solves social problems, and creates new ones, but in this case I don't think theres a quick fix. |
| 22:35.46 | [TK]D-Fender | Micc: Printers make nice pink slips ;) |
| 22:36.43 | nextime | the fix is to remove the volume button adjustament |
| 22:36.54 | nextime | i mean physically :) |
| 22:38.29 | Micc | nextime, that could work. |
| 22:38.53 | Micc | then they wouldn't be able to change the handset/headset volume either though. |
| 22:39.08 | [TK]D-Fender | nextime: No, then you'v have to do it to all phones. Make an example of the slacker in front of his once peers.... then ensure his story lives on through furure hirings |
| 22:41.18 | disposable | nextime: i've disabled ringer volume adjustments, it's always at 100% and can't be changed. i'd just like a way to disable the silent ringtone somehow. |
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| 23:53.28 | hardwire | why is it when the link I'm using for asterisk is suddenly saturated and unusable.. it effects the local side of the network between phone and pbx (no load on pbx other than attempting to make external calls) |
| 23:53.50 | hardwire | is there some sort of blocking being done to accept new calls on port 5060? |
| 23:54.15 | hardwire | link I'm using for sip channels from asterisk.. sorry |
| 23:54.27 | hardwire | I can't seem to explain this properly enough. |
| 23:54.43 | xheliox | 5060 is for signaling.. SIP needs RTP, which the port range for your install would be in rtp.conf. |
| 23:55.17 | hardwire | xheliox: when asterisk is being told to make 20 calls to other public servers that aren't responding.. 1 call to internal services seems to be blocked. |
| 23:55.26 | hardwire | sip sessions time out |
| 23:55.28 | hardwire | etc |
| 23:56.30 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 23:57.07 | doctorray | Does anyone know a Polycom dealer on the west coast? I need to pick up a couple of IP 650s and sidecars asap but don't want to pay through the nose for shipping from NY |
| 23:58.10 | *** join/#asterisk dunhamda (~dunhamda@nat.clevermachine.com) |
| 23:58.38 | hardwire | just pay with your bosses visa. |
| 23:58.46 | hardwire | through somebody elses nose |
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| 23:59.13 | boodu | hello |
| 23:59.27 | albertAsterisk | hello boodu |
| 23:59.32 | doctorray | lol |
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| 23:59.50 | albertAsterisk | xd |
