| 00:05.46 | datruth | what ports are needed to be open for asterisk to function properly? |
| 00:05.56 | Kobaz | depends what you want it to do |
| 00:06.01 | Kobaz | and what protocols you're using |
| 00:06.17 | datruth | ahh I see |
| 00:06.27 | Kobaz | you don't need any ports open if you're using only isdn and analog |
| 00:06.57 | Kobaz | if you want to use it for voip, you'll probably want to use sip, which is port 5060 and media ports (generally something like 20000-21000 |
| 00:07.01 | Kobaz | ) |
| 00:07.21 | datruth | I have asterisk on a server I dont have physical phone I was going to do everything via the internet. |
| 00:07.40 | Kobaz | sip most likely then |
| 00:08.46 | datruth | I wanted to use this as a internet conference i'm not sure if it'll work how I want but worth a shot |
| 00:10.06 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:10.30 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 00:13.18 | *** join/#asterisk pguillem (~pguillem@190.9.253.185) |
| 00:13.23 | pguillem | hi all |
| 00:13.26 | *** join/#asterisk mythicalbox (~mitchlovi@99.99.151.110) |
| 00:13.44 | pguillem | anybody with audiohook API knowledge? |
| 00:14.14 | mythicalbox | is there a trick to get Set to save a variable that's pushed out via AMI (using the VarSet packet) for a new channel? the docs say to put two underscores "__" in front of the var name, but that is not working |
| 00:14.34 | mythicalbox | I'm wondering if this might just be an obscure bug/use case |
| 00:14.46 | pguillem | The __ means the cariable inheritance.. |
| 00:14.48 | pguillem | variable |
| 00:15.10 | pguillem | so it will be available in macros and subs in the dialplan.. |
| 00:15.21 | pguillem | what are you trying to do? |
| 00:16.34 | mythicalbox | pguillem, here is my AMI packet capture: http://pastebin.com/0QduyYJS |
| 00:16.55 | mythicalbox | i have an app monitoring AMI for a crmpath to load a webpage when the callee answers the phone |
| 00:17.15 | mythicalbox | my call to set is being assigned to the caller channel, not the callee |
| 00:17.29 | pguillem | i understand |
| 00:18.26 | pguillem | you could code a small socket in javascript to plug the web into the AMI.. a simple event from the AMI could trigger a page change |
| 00:18.46 | pguillem | you would have to build a parser thou... |
| 00:19.00 | pguillem | i have one written in AS3 in case you need it |
| 00:19.11 | pguillem | i'm not familiar with CRMPATH |
| 00:19.18 | mythicalbox | i'd be interested in looking |
| 00:19.22 | mythicalbox | that's my var i'm looking for |
| 00:19.28 | pguillem | cool |
| 00:19.38 | pguillem | i totally undertand what you are trying to acomplish |
| 00:19.55 | mythicalbox | I obviously see that data in the AMI, but it's not directly tied to the called channel |
| 00:20.20 | pguillem | give me your email and i will send you the AS3 library i wrote to plug Flash into the AMI |
| 00:20.44 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 00:20.50 | mythicalbox | sent |
| 00:28.04 | *** join/#asterisk kessius (~cassio@201.21.173.58) |
| 00:30.37 | p3nguin | datruth: If your phones have to pass through a firewall to get into asterisk, you'll want to allow ports 5060 UDP and whatever range is configured in rtp.conf (usually 10000-20000, also UDP) to reach asterisk. |
| 00:31.05 | datruth | ahh gotcha |
| 00:31.50 | p3nguin | If your phones are on the same network as the server, there probably isn't a firewall in between. |
| 00:33.29 | *** part/#asterisk linuxishawt (~kevin@adsl-99-96-193-115.dsl.lsan03.sbcglobal.net) |
| 00:35.33 | pguillem | You will have |
| 00:35.40 | pguillem | Sorry |
| 00:45.36 | kessius | hi, what is the difference between install asterisk, with (./configure make and make install ...) and simply yum install asterisk16 |
| 00:48.27 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:48.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:49.59 | pguillem | Guys.. when i Dial() with the L option.. i need warning messages to be listened by both bridged channels at the same time... this is not happening |
| 00:50.46 | pguillem | basically, the channel.c function that plays the audio file reads the warning one channel at a time... making a huge silence for the other party. |
| 00:51.23 | pguillem | Anybody knows how should i fork a new thread to play the warning messages simultaneously (or almost) ????? |
| 00:53.19 | pguillem | channel.c already includes utils.c and audiohook.c ... i know i could use ast_pthread_create() to do that, but i'm not an expert in handling a live channel. So what considerations should i take when firing threaded events from channel.c ??? |
| 00:53.31 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.154.87.dsl.dyn.forthnet.gr) |
| 00:58.31 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 01:00.11 | kessius | difference in which to install simply yum install asterisk16 / and compile the source |
| 01:01.12 | [TK]D-Fender | kessius: Tink about the obvious nature of what YUM implies you'll get. |
| 01:01.15 | [TK]D-Fender | Think* |
| 01:07.23 | pguillem | Anybody? |
| 01:10.23 | [TK]D-Fender | pguillem: I would just call dial with M() then have that lauch a backgouned script passing it the 2 channel names. That script would AMI Originate() local channels that would bridge to each end to play the messages |
| 01:22.32 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:22.48 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 01:24.20 | kessius | but case , we need to install any module - case need recompile ? - I will not be able because yum install asterisk18 ? |
| 01:27.56 | p3nguin | You'll either use yum to install everything, or you'll compile from source. Don't try to do both. |
| 01:28.20 | p3nguin | If you insist on compiling from source, use checkinstall to make your own package so you can manage it properly with rpm. |
| 01:33.31 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 01:39.42 | pguillem | thanks TD-Fender |
| 01:39.48 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:40.46 | pguillem | [TK]D-Fender: What cmd should i use to launch the background script?, do you mean an AGI? |
| 01:41.54 | [TK]D-Fender | pguillem: Doesn't have to be AGI. That script would want to check that the channel(s) are still up before launching the new channel to bridge in the announcement as it's process will be independant and will need to know when to terminate |
| 01:42.33 | p3nguin | its |
| 01:46.01 | pguillem | [TK]D-Fender: Good advise. I hadn´t think about it that way. I had tried to spawn a macro on answer, but the channels won´t bridge untill the macro is finished. I have no idea how to launch the macro in the background and let the call bridge. |
| 01:46.40 | [TK]D-Fender | pguillem: No, the macro launches immedaite. you use System() to launch a completely independant script with & |
| 01:47.41 | pguillem | [TK]D-Fender: Totally got the idea. I'll give it a try :) |
| 01:47.46 | [TK]D-Fender | p3nguin: sig heil mein herr! |
| 01:48.06 | pguillem | [TK]D-Fender: Thanks for the light ;) |
| 01:48.33 | [TK]D-Fender | pguillem: You're welcome... |
| 01:50.07 | pguillem | [TK]D-Fender: Would you rather use ChanSpy() to play on whisper mode ? or Background() on each channel? |
| 01:54.08 | pguillem | I think they coded the B option in ChanSpy() to |
| 01:54.10 | [TK]D-Fender | pguillem: As you are starting a new channel you need to hook into the other ones. This will have Chanspy on one end, and Playback on the other. |
| 01:54.42 | kessius | but when I try to compile dahdi-tools - Centos appears erro kernel - then send command - yum -y install kernel-devel _because keep getting erro |
| 01:57.19 | kessius | but when I try to compile dahdi-tools - Centos appears erro kernel - then send command - yum -y install kernel-devel _because i getting erro |
| 01:57.45 | pguillem | [TK]D-Fender: Ok. So from the async script i originate() a Local channel hooked to the bridge, and exec a context where ChanSpy is called with the Barge option and then Playfile? |
| 01:58.05 | [TK]D-Fender | Playback. Yes |
| 01:58.49 | pguillem | [TK]D-Fender: I owe you a beer! |
| 01:59.22 | [TK]D-Fender | pguillem: You're welcome. |
| 02:00.16 | [TK]D-Fender | pguillem: it is kinda hackish, but releives you from having to mod C directly and complicate upgrades, etc |
| 02:04.56 | *** join/#asterisk jdoe (jdoe@falseprophet.ca) |
| 02:05.42 | jdoe | anyone here logging rtcp data to for call quality stats? (and if so, what are you logging?) |
| 02:15.44 | kessius | RTCP is a good question - how to it works |
| 02:21.16 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:26.24 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 02:33.57 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:33.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:49.50 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 03:11.03 | ChannelZ | So how does one work around a remote peer whose behind NAT and giving out its LAN IP, when that IP happens to match the same LAN network of the Asterisk box it's connecting to which is also behind a firewall? |
| 03:11.40 | ChannelZ | Asterisk sees the 192.168.x.x IP and says 'Hey that's MY local net' and happily sends RTP to it. |
| 03:12.19 | [TK]D-Fender | you set the peer to nat=yes and * will send to the IP it is sending from, not what it claims |
| 03:12.36 | ChannelZ | But it doesn't. |
| 03:13.09 | ChannelZ | Unless it's an oddity with it being an anonymous call as opposed a specifically defined peer. |
| 03:13.25 | ChannelZ | nat=yes is in [general] |
| 03:14.07 | [TK]D-Fender | And details are only slowly emerging... without actual configs and debug. |
| 03:18.04 | ChannelZ | http://pastebin.com/3Xg2FYK8 |
| 03:18.26 | BuenGenio | Just found the best hold music |
| 03:19.09 | ChannelZ | Sent RTP packet to 192.168.1.5:8222 (type 08, seq 062547, ts 004800, len 000160) |
| 03:19.17 | ChannelZ | etc. Forgot that. |
| 03:23.51 | ChannelZ | BuenGenio: Morning zoo show on the mexican radio station? |
| 03:23.59 | BuenGenio | lol |
| 03:24.06 | BuenGenio | http://www.youtube.com/watch?v=83F9ek8J25M |
| 03:24.59 | BuenGenio | need at least SPEEX for it though |
| 03:25.02 | BuenGenio | check out that bassline |
| 03:25.16 | ChannelZ | Ah. Malfunctioning CD player. |
| 03:25.21 | [TK]D-Fender | ChannelZ: Odd indeed.. because it responds to the SIP packet properly as NAT'd. You are several versions behind. This may have been fixed and I think I saw something similar a few weeks ago... |
| 03:25.33 | ChannelZ | Exactly. |
| 03:25.55 | ChannelZ | I'll try the other direction tomorrow from work to my home which is running *10 |
| 03:40.21 | kessius | <PROTECTED> |
| 03:42.50 | [TK]D-Fender | kessius: First he wants guests. Second his IP is set correctly for 1.8. Allowguest=NOT.... "not" is not a valid value. And finally he isn't matching a peer at all.... so this has nothing to do with them. |
| 03:46.23 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 03:52.56 | woleium | Has anyone successfully paid license fees for g729 without using Digium's (factor of 10 too expensive) option? |
| 03:53.19 | [TK]D-Fender | woleium: Where do you come up with "factor of 10"? |
| 03:53.34 | woleium | it's a secret |
| 03:53.37 | woleium | :p |
| 03:54.33 | woleium | I know someone that works for an OEM. it was their comment :-) |
| 03:54.34 | [TK]D-Fender | woleium: Oh.. you mean "delusionally derived". Gotcha! |
| 03:55.25 | woleium | But to all intents and purposes it could be "fat bloke down the pub told me" |
| 03:56.17 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 03:56.50 | [TK]D-Fender | woleium: Just found out Sipro is actually located right where I live... maybe I'll drop by one day ;) |
| 03:57.35 | woleium | it kinda makes sense though. I'd not expect handset manufacturers to include a license if it adds $6 or so to the cost - but then they are probably just paying a flat fee for unlimited use. |
| 03:57.50 | [TK]D-Fender | woleium: http://www.sipro.com/g729_licterms.php |
| 03:58.27 | woleium | that looked good until i saw "Initial Fee: $15,000" |
| 03:58.32 | [TK]D-Fender | woleium: woleium Note the "initial fee" and "minimum prepaid" sections. These are the "gotchas" |
| 03:58.41 | [TK]D-Fender | Yup :p |
| 03:59.17 | [TK]D-Fender | woleium: So if you're rolling out a ton... I suppose it could be cheaper. Till then "good luck". |
| 03:59.28 | woleium | I guess digium's offering is the most logical then |
| 04:01.32 | woleium | maybe if we all go stand outside Polycom's HQ with speex banners… |
| 04:03.06 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 04:07.02 | [TK]D-Fender | woleium: G.711, G.729, G.722, and iLBC aren't enough? |
| 04:08.16 | woleium | It's not a bad list |
| 04:10.58 | woleium | but 711 is to big, 729 is too expensive. I guess 722 & iLBC are OK, but my current SIP provider doesn't talk either :-( |
| 04:12.20 | [TK]D-Fender | G.722 would be too big then |
| 04:13.33 | woleium | ?? i thought polycoms did 722.1c @~32k |
| 04:13.42 | kessius | [TK]D-Fender you're correct - externip was used * 1.4 - |
| 04:14.15 | [TK]D-Fender | woleium: Last I heard it was 64kbps standard |
| 04:16.47 | woleium | http://www.polycom.com/company/about_us/technology/siren14_g7221c/index.html |
| 04:17.04 | woleium | 24, 32, and 48 kbps |
| 04:18.11 | woleium | but it's probably proprietary. Looks like it's based in their SIREN codec. |
| 04:19.48 | [TK]D-Fender | woleium: I'm referring to *'s support |
| 04:19.52 | [TK]D-Fender | ^^ |
| 04:21.06 | woleium | indeed. |
| 04:22.13 | woleium | wonders if there is a way to customise the "unable to connect" to say "out of 729 licenses" when appropriate... |
| 04:22.40 | [TK]D-Fender | woleium: I'd sooner have * fallback on SDP offering G.729 once you've run out.... |
| 04:22.56 | [TK]D-Fender | woleium: Because it still lets you make promises you can't keep |
| 04:23.40 | woleium | you are probably right. Maybe something in the log then |
| 04:24.09 | woleium | at the moment i don't think it logs anything when it runs out. |
| 04:28.58 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 04:55.50 | *** join/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 04:56.04 | *** part/#asterisk gonewage (~gonewage@c-68-54-124-223.hsd1.il.comcast.net) |
| 04:58.51 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-rwmhyufqdotxbyzz) |
| 05:23.47 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 05:25.55 | [TK]D-Fender | Alrighty... bed time, back tomorrow.... |
| 05:27.17 | *** join/#asterisk jetlag (~jetlag@pool-71-168-201-186.cmdnnj.east.verizon.net) |
| 05:29.17 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 05:51.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:54.12 | *** join/#asterisk bmg505 (~leon@196-209-72-124.dynamic.isadsl.co.za) |
| 06:15.31 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:24.41 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:25.10 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 06:25.12 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:25.41 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:26.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:27.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:27.30 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 06:28.11 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 06:30.50 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 06:34.01 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 06:36.02 | *** join/#asterisk cyborg-one (1000@212-178-7-243.broadband.tenet.odessa.ua) |
| 06:36.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:36.51 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 06:38.28 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 06:38.43 | *** join/#asterisk plundra (1000@v0.article.se) |
| 06:40.16 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 06:40.27 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 06:40.31 | *** join/#asterisk ketas (~ketas@195.20.191.90.dyn.estpak.ee) |
| 06:42.14 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 06:42.22 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 06:42.55 | *** join/#asterisk Who-m3 (john@my.virt-host.com) |
| 06:54.14 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 06:54.15 | *** join/#asterisk ectospasm (~ectospasm@66.172.33.249) |
| 07:06.57 | *** join/#asterisk RypPn_OuT (~RypPn@rosscom.co.uk) |
| 07:08.15 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-gdrxbukjfwrhzuie) |
| 07:13.43 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 07:13.49 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 07:14.32 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 07:18.48 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:27.29 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 07:34.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:34.44 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:37.47 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 07:39.20 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-omoopqtuzgvftdmq) |
| 07:42.19 | *** join/#asterisk bmg505 (~leon@196-209-72-124.dynamic.isadsl.co.za) |
| 07:58.39 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:58.43 | schmidts | good morning |
| 07:58.59 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 08:07.22 | *** join/#asterisk dtrich (~qwe@athedsl-4365051.home.otenet.gr) |
| 08:07.48 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:09.54 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 08:12.08 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 08:15.49 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:22.11 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:26.43 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 08:29.43 | *** join/#asterisk aurs (~aurs@84.49.69.109) |
| 08:48.11 | *** join/#asterisk ollii (~risker@81.14.178.55) |
| 08:51.04 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:54.19 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:54.29 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:01.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:12.56 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 09:13.39 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 09:16.05 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 09:29.11 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 09:30.18 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 09:37.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:39.25 | *** join/#asterisk faithlove (proxyuser@188.27.133.9) |
| 09:40.42 | faithlove | hi. i'm currently developing a project based on asterisk and i'm wondering if there's any tool which can simulate calls so I can test the functionality of my tool. |
| 09:45.31 | *** join/#asterisk Freeaqingme_ (~Freeaqing@91.214.168.110) |
| 09:55.24 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 09:55.24 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-rwmhyufqdotxbyzz) |
| 09:55.24 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.154.87.dsl.dyn.forthnet.gr) |
| 09:55.24 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 09:55.24 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 09:55.24 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 09:55.25 | *** join/#asterisk PhiBrick (~phil@212.80.245.102) |
| 09:55.25 | *** join/#asterisk tuxx- (tuxx@pantoff0l.nl) |
| 09:55.25 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 09:55.25 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 09:55.25 | *** join/#asterisk bindi (~bindi@f430.ip15.netikka.fi) |
| 09:58.14 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 09:58.25 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:07.56 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-143.telkomadsl.co.za) |
| 10:40.00 | *** join/#asterisk dxd828 (~dxd828@dxdservers-3-pt.tunnel.tserv10.par1.ipv6.he.net) |
| 10:40.27 | dxd828 | Does anyone know if any of the Cisco video hardphones could work with Asterisk? |
| 10:48.54 | vlt | Hello. When connecting two asterisks via IAX is it possible to use two different voice codecs for each direction? I want the data sent in g711 and the data received in gsm? Possible? |
| 10:52.59 | kaldemar | vlt: no. |
| 10:53.52 | bulkorok | vlt: you have to put a third machine in the middle for transcoding |
| 10:55.23 | *** part/#asterisk phr3ak (~noreply@gnet.hu) |
| 10:58.05 | vlt | bulkorok: A third machine? What exactly would that machine "C" do that not "A" or "B" can do? |
| 11:00.55 | kaldemar | enable a session between A and B with A using gsm and B using G.711. |
| 11:01.11 | kaldemar | but what's the point in using different codecs anyway? |
| 11:01.40 | bulkorok | a calls c with g711 ---- c calls b with gsm so you have a translator... |
| 11:01.49 | vlt | kaldemar: limited bandwidth in only _one_ direction. |
| 11:02.41 | vlt | bulkorok: Hmm, I don't understand why A can't call B directly using gsm. |
| 11:04.06 | vlt | I want A to call B, send all voice data in gsm but receive B's voice in alaw/ulaw. |
| 11:04.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 11:05.10 | vlt | I'd understand "impossible", but fail to see how a third box in the middle could solve this. |
| 11:06.11 | bulkorok | I have to check, but a SIP-session can only use one codec for both UAs. So you can not say UA 1 uses codec A and UA 2 uses codec B in the same session... |
| 11:07.30 | fakhir | ~itsp |
| 11:07.30 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 11:07.44 | fakhir | ~itsplist-us |
| 11:07.44 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 11:09.59 | kaldemar | vlt: it would not solve it. it would only enable the A and B to use different codecs, but not two codecs at the same time in a single session. |
| 11:12.05 | bulkorok | vlt: right... it's only one codec per session possible. The codec is handled in the SDP-part of INVITE and the response-pakets from the other UA. And there it's only one possible per session... |
| 11:12.35 | *** join/#asterisk pjm (~pjm@109.104.96.45) |
| 11:14.11 | *** join/#asterisk jkroon (~jkroon@dsl-244-28-128.telkomadsl.co.za) |
| 11:15.14 | bulkorok | so again: set up one session from a to c with g711 and then a second session with GSM from c to b ... two sessions but two different codecs... |
| 11:19.06 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 11:19.34 | vlt | kaldemar: Ok, thank you. |
| 11:19.52 | vlt | bulkorok: Ok, only one codec per session. |
| 11:20.09 | vlt | bulkorok: So machine C in the middle doesn't make sense at all ;-) |
| 11:22.39 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 11:27.21 | faithlove | hi. i'm currently developing a project based on asterisk and i'm wondering if there's any tool which can simulate calls so I can test the functionality of my tool. |
| 11:28.50 | *** join/#asterisk MarKsaitis (~MarKsaiti@88.96.60.78) |
| 11:29.50 | kaldemar | faithlove: what kind of calls? |
| 11:30.02 | faithlove | inbound / outbound |
| 11:30.10 | faithlove | anything which would help |
| 11:30.53 | kaldemar | as in technology |
| 11:31.28 | kaldemar | use a soft phone, another asterisk box or sipp for example |
| 11:38.39 | bowzak | i'm having problems with some SPA525 phones running over a VPN client. They will work for a while, but occasionally they fall into a loop where they cannot complete registration. If I change the SIP port, they come right back up.. but eventually the problem recurrs. Ideas? |
| 11:38.40 | faithlove | aha. well, i throught that there might be something like a simulator app. thanks anyway :) |
| 11:47.29 | kaldemar | faithlove: sipp can be thought as a simulator app. |
| 11:49.39 | *** join/#asterisk BuenGenio (~Gene@n058152141099.netvigator.com) |
| 11:52.18 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 11:53.32 | pjm | Hi, I wonder if someone might be able to give me a pointer re voicemail. I have a few SIP phones on internal LAN, all using ulaw, voice mail is fine, I can hear all the user recorded announcements etc. Dialling in via dahdi or SIP peer, the VM announcement starts to play but then stops after a few seconds, call remains but there is just silence, what can I check? |
| 11:54.16 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:54.16 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:55.07 | *** join/#asterisk Rewris (~Tontones@186.215.253.82) |
| 11:58.28 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 11:59.34 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 12:07.24 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:21.56 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 12:25.34 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 12:31.58 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 12:33.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:35.37 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-143.telkomadsl.co.za) |
| 12:36.17 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:42.13 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 12:43.19 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:43.28 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:44.12 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:46.36 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:47.06 | *** join/#asterisk FainaUkraina (~Gene@n058152141099.netvigator.com) |
| 12:49.50 | *** join/#asterisk mahiti-irc (~mahiti@115.109.178.168) |
| 12:50.08 | mahiti-irc | hi |
| 12:50.53 | mahiti-irc | is it possible to setup a single asterisk box with a STM-1 connection ? |
| 12:51.09 | mahiti-irc | if yes how many concurrent calls can this box handle?? |
| 12:53.24 | mahiti-irc | i was thinking of using STM-1 OC3 PCI card like this http://www.iphase.com/products/product.cfm/PCI%20Express/472 |
| 12:56.48 | *** part/#asterisk gajini (~root@61.12.17.171) |
| 12:58.47 | *** join/#asterisk matsk (~matsk@83.172.89.35) |
| 13:01.16 | *** join/#asterisk remuda (~Tim@173.243.243.172.rhinocommunications.net) |
| 13:01.42 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 13:04.59 | [TK]D-Fender | mahiti-irc, That card does not mention any support of Asterisk |
| 13:07.20 | [TK]D-Fender | mahiti-irc, I also can't imagine the system surviving the EC load that many channels could place.... |
| 13:07.48 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-143.telkomadsl.co.za) |
| 13:08.23 | [TK]D-Fender | mahiti-irc, And their "Applications" doesn't seem to list telephony too directly in any way. |
| 13:08.29 | mahiti-irc | [TK]D-Fender, EC ? |
| 13:08.36 | [TK]D-Fender | Echo Cancellation |
| 13:08.42 | mahiti-irc | ok |
| 13:09.03 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 13:09.10 | [TK]D-Fender | mahiti-irc, As it is there still aren't any DS3 cards out there that are compatible with * yet |
| 13:09.14 | qakhan | hi all |
| 13:09.20 | mahiti-irc | ok |
| 13:09.43 | mahiti-irc | [TK]D-Fender, actually i am setting up a project where the client requires around 1200 calls simultaneously |
| 13:09.55 | mahiti-irc | what architecture would u suggest for it |
| 13:09.57 | qakhan | i have a queue with 5 agents. i want to setup a alarm on queue after 3 mins if not agent answer the call |
| 13:10.05 | mahiti-irc | i thought STM-1 handles it |
| 13:10.12 | [TK]D-Fender | mahiti-irc, * could do that if you use a gateway to do the conversion to a voip protocol |
| 13:10.50 | mahiti-irc | [TK]D-Fender, can u please elaborate it? |
| 13:10.52 | [TK]D-Fender | mahiti-irc, They mention no support of *, no drivers, and no application layer to handle EC, etc. Looks dead to me. |
| 13:11.32 | [TK]D-Fender | mahGet a gatway appliance that can convert to SIP. They have these for DS1, and I think I saw a DS3 one once (never implemented by anyone I know) |
| 13:13.00 | mahiti-irc | [TK]D-Fender, You mean to say the STM-1 will be terminated in the gateway appliance and the asterisk speaks to this appliance |
| 13:13.15 | [TK]D-Fender | mahiti-irc, yes |
| 13:13.32 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 13:13.54 | mahiti-irc | [TK]D-Fender, ok can you suggest me one such appliance, would help in the right direction |
| 13:14.09 | [TK]D-Fender | mahiti-irc, I imagine this would have to have a gigabit interface to handle the packet overhead, and would probably be VERY expensive... but conceivable |
| 13:14.27 | [TK]D-Fender | mahiti-irc, As I said... I'm not sure I've ever even seen one this big before... |
| 13:14.31 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 13:14.36 | [TK]D-Fender | mahiti-irc, get googleing.... |
| 13:15.13 | mahiti-irc | [TK]D-Fender, oh but the question is wht do i google for ? "Gateway appliance for STM-1" |
| 13:15.22 | fprior | Hi, http://pastebin.com/YKSHkztS : what cause this messages when debug is on ? |
| 13:15.25 | eduzimrs | anyboody knows this issue > WARNING[5328] chan_sip.c: Matched device setup to use SRTP, but request was not! im trying to use MizuPhone with SRTP and this appears i cant make calls |
| 13:15.31 | [TK]D-Fender | SIP STM-1 OC3 |
| 13:15.34 | [TK]D-Fender | ^ |
| 13:15.51 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ncdgfxunqmlqzqlw) |
| 13:15.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:16.08 | [TK]D-Fender | eduzimrs, As it says, the device didn't request SRTP.... and the perr says it's supposed to. |
| 13:16.12 | [TK]D-Fender | eduzimrs, Fix your device |
| 13:17.26 | eduzimrs | [TK]D-Fender: im using SRTP option at the device, (Encrypt Media [SRTP]) |
| 13:17.39 | qakhan | i have a queue with 5 agents. i want to setup a alarm on queue after 3 mins if not agent answer the call |
| 13:17.40 | [TK]D-Fender | eduzimrs, * doesn't seem to see that in the incoming request.... |
| 13:18.32 | eduzimrs | [TK]D-Fender: maybe a problem with MizuPhone? |
| 13:18.48 | eduzimrs | [TK]D-Fender: do u konow this softPhone? |
| 13:19.00 | [TK]D-Fender | qakhan, * has no mechanism for this. Before dumping the caller into the queue, launch some background external process to watch the caller's channel and act on a timeout. |
| 13:19.07 | [TK]D-Fender | eduzimrs, No, I do not. |
| 13:19.35 | eduzimrs | [TK]D-Fender: do u know other that uses TLS and SRTP ? |
| 13:19.59 | [TK]D-Fender | eduzimrs, Not offhand .... I try to never use softphones... |
| 13:20.19 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 13:21.19 | mahiti-irc | [TK]D-Fender, ok i got a media gateway link which takes care of all what we wanted to do only on asterisk. |
| 13:21.26 | mahiti-irc | is that what you mention? |
| 13:22.15 | qakhan | [TK]D-Fender you mean some agi script? |
| 13:33.14 | fprior | Hi, what cause this boring messages http://pastebin.com/YKSHkztS ? |
| 13:34.51 | jkroon | hi guys, is it a problem to call ast_timer_set_rate multiple times on the same (active) timer? |
| 13:35.22 | jkroon | in chan_iax2 there is an issue where trunkfreq from the config file does not take effect, and the simplest fix would be to just call ast_timer_set_rate on the existing timer. |
| 13:35.47 | jkroon | not sure if this can break other things |
| 13:41.42 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:46.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:47.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:51.41 | mahiti-irc | [TK]D-Fender, or anyone, can you tell me if this can be used with asterisk box ? |
| 13:51.42 | mahiti-irc | http://www.teleprime.com/whatis/ss7sip.html |
| 13:54.51 | jkroon | mahiti-irc, looks sane |
| 13:56.50 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 13:56.57 | mahiti-irc | thanks jkroon , but can i know what u found in there? |
| 13:57.14 | jkroon | it talks SIP. |
| 13:57.33 | jkroon | so as long as you can get it to bridge properly between SS7 or C7 and SIP you should be fine. |
| 13:57.55 | McBoingBo | Looking at the SNTP section of my phone config, do I always need to manually adjust the daylight savings or is there some automagical way of setting this up? |
| 13:59.44 | *** part/#asterisk kl4m (~kl4m@gw2.noc1.sys-tech.net) |
| 14:00.23 | jkroon | should be able to deduce it based on the timezone which you will need to set manually. |
| 14:01.11 | [TK]D-Fender | qakhan, No, not an AGI. I said BACKGROUND process. |
| 14:01.48 | [TK]D-Fender | mahiti-irc, That says it talks SIP. that is what I'm suggesting already... |
| 14:02.39 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 14:02.45 | mahiti-irc | ok thanks a lot jkroon and [TK]D-Fender :) |
| 14:02.59 | mahiti-irc | btw |
| 14:03.32 | mahiti-irc | have u guys any idea of a USSD gateway that works with asterisk? |
| 14:04.37 | jkroon | mahiti-irc, i've seen the question before, I still don't understand what USSD has to do with voice. it's a textual protocol from what I understand. |
| 14:07.33 | mahiti-irc | ok, but cant asterisk be used to redirect a USSD transaction to a USSD gateway?? |
| 14:12.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:12.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.21 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 14:18.26 | leifmadsen | what is USSD? |
| 14:18.48 | leifmadsen | and since Asterisk is a B2BUA I suspect Asterisk would have to somewhat understand how to handle USSD |
| 14:19.32 | leifmadsen | oh ya, this doesnt' seem like anything Asterisk should really handle |
| 14:20.37 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 14:20.38 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 14:20.38 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 14:30.54 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 14:31.20 | Dovid | with libss7 is there any variable that will let me know if an incoming call is a national or international call? |
| 14:31.39 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 14:47.47 | mahiti-irc | thanks for your support :) |
| 14:47.51 | *** part/#asterisk mahiti-irc (~mahiti@115.109.178.168) |
| 14:56.39 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 15:02.16 | *** join/#asterisk af_ (~getsmart@78.134.18.236) |
| 15:03.46 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:04.01 | *** join/#asterisk Vince-0 (~AndChat@41-135-218-35.dsl.mweb.co.za) |
| 15:05.10 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 15:06.24 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:14.28 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 15:14.38 | *** join/#asterisk libryder (~libryder@libryder.com) |
| 15:15.41 | *** join/#asterisk sgtpepper (~sgtpepper@190.48.114.81) |
| 15:16.09 | sgtpepper | hi there... quick question.. I'm trying to mask an early media message with a ringing tone since I might retry the call through another route |
| 15:16.22 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 15:16.29 | sgtpepper | and I don't want the customer the hear a messge like "The subscriber you're trying to reach blah..." |
| 15:16.39 | sgtpepper | I've tried with R,r in the dial command |
| 15:16.50 | sgtpepper | switching progressinband in the peeer in sip.conf |
| 15:16.58 | sgtpepper | still, I cannot find a way to mask that message |
| 15:17.09 | sgtpepper | I'm using asterisk 1.8.3.2 |
| 15:17.21 | sgtpepper | (one of the latest) |
| 15:18.11 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 15:18.32 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 15:23.08 | kaldemar | sgtpepper: 1.8.3.2 is actually quite old for its branch. it was released a year ago. have you tried setting prematuremedia=yes in addition to progressinband=never? |
| 15:24.07 | sgtpepper | let me try that combination specifically kaldemar |
| 15:27.01 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:27.13 | sgtpepper | kaldemar: I'm trying those on the caller side... not on the trunk side... is that correct? |
| 15:31.31 | kaldemar | prematuremedia is a general option. |
| 15:38.47 | *** join/#asterisk sgtpepper (~sgtpepper@190.48.108.153) |
| 15:39.23 | sgtpepper | grrr I hate early media |
| 15:41.48 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:42.49 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:42.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:43.45 | sgtpepper | any other way you can think of to mask that early media message? |
| 15:46.48 | Qwell | tzafrir_laptop: I'm sad that nobody ever recommends using JPAH. I bet it's the least expensive echo can there is. |
| 15:47.16 | jacc0 | sgtpepper: let the callee pers a dtmf to confirm his/he pickup and then bridge maybe? |
| 15:47.24 | jacc0 | *pres |
| 15:48.07 | tzafrir_laptop | Qwell, it can also quarntee the same quality regardless of the jitter and such, which is also important |
| 15:48.23 | Qwell | exactly |
| 15:48.30 | Qwell | guaranteed results - that's the Qwell way |
| 15:49.19 | Qwell | tzafrir_laptop: I couldn't contain my laughter when committing that. There was actually a useful reason for it, too. |
| 15:49.29 | Qwell | I'm sure nobody has actually used it since, though |
| 15:54.35 | *** join/#asterisk r0gers (rogers@bling.bling.org) |
| 15:55.02 | sgtpepper | jacc0: mmm... thats not going to work |
| 15:57.07 | jacc0 | ok, it was just a suggestion |
| 15:57.29 | dxd828 | are there asterisk repo's for centos 6.2? |
| 15:57.33 | Qwell | dxd828: no |
| 15:58.39 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:58.44 | dxd828 | Qwell, are there for Debian? |
| 15:58.54 | Qwell | there are a bunch for debian |
| 16:00.37 | *** join/#asterisk its_jeremy_ (~omghax@96.18.45.74) |
| 16:02.04 | p3nguin | Yay for stupid people! Another set of faxes went out today from that place...with my number on it again. |
| 16:02.16 | p3nguin | FCC Form 1088A has been filed. |
| 16:05.50 | mcrownover | i know this will most likely be a difficult question but for a novice building a new PBX today would anyone recommend going with Asterisk 10 over Asterisk 1.8.10.0 |
| 16:06.10 | p3nguin | Depends on your use and expectations. |
| 16:06.12 | Qwell | mcrownover: Do you need features in 10? |
| 16:06.31 | p3nguin | I only deploy LTS branches in business production. |
| 16:06.58 | mcrownover | this would be for business use - about 150 extensions |
| 16:07.17 | mcrownover | and 5 locations in the enterprise |
| 16:07.18 | p3nguin | Do you mean 150 phones? |
| 16:07.33 | mcrownover | yes 150 phones |
| 16:07.43 | p3nguin | That's a lot different from 150 extensions. |
| 16:08.04 | mcrownover | <----"novice" |
| 16:08.29 | p3nguin | Learn the difference early. |
| 16:08.59 | mcrownover | i'm trying |
| 16:09.10 | p3nguin | Take this for comparison: on a system I have with six phones, I have 63 extensions. |
| 16:11.12 | p3nguin | In Asterisk, extensions are the dialing rules which match numbers called. |
| 16:11.37 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:12.01 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 16:13.25 | mcrownover | well we have about 150 desk phones with unique DIDs and about 75 or more that will route calls out via followme - plus 10 or so unique numbers inbound |
| 16:13.56 | p3nguin | Okay, so that's 235 extensions right there. |
| 16:14.08 | p3nguin | Plus you'll have more with subroutines and whatnot. |
| 16:14.36 | p3nguin | ~book |
| 16:14.37 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:15.30 | woleium | ~istplist-ca |
| 16:15.54 | woleium | ~itsplist-ca |
| 16:15.54 | infobot | it has been said that itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 16:16.02 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 16:16.59 | mcrownover | i guess i am most confused on afer implementation on what to upgrade, what not to, how long the life expetancy is for the pbx |
| 16:17.29 | p3nguin | Life for the system as a whole, or life of the software? |
| 16:18.09 | mcrownover | for the software |
| 16:18.11 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:18.35 | p3nguin | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 16:19.14 | mcrownover | looked at that earlier - so my current 1.6.2.20 is going to EOL in April |
| 16:19.47 | p3nguin | <p3nguin> I only deploy LTS branches in business production. <---- important |
| 16:20.15 | [TK]D-Fender | ^ For p3nguin |
| 16:20.30 | mcrownover | so there is my easy answer to go with the 1.8.x vein |
| 16:20.42 | [TK]D-Fender | mcrownover, Not a bad choice |
| 16:20.55 | [TK]D-Fender | mcrI'd recommend it unless there is something special in 10 you need |
| 16:21.56 | mcrownover | of course all of this would be easier if I were lucky enough in my organization to only work on one item like the PBX - (i'm sure you guys know how it is to be responsible for EVERYTHING in the building) |
| 16:22.57 | [TK]D-Fender | mcrownover, Yes... I'm in the middle of moving 1/3 of my IT infrastructure around this week... |
| 16:23.02 | bulkorok | hi... where can I see all the status strings for FAXOPT(status) and FAXOPT(statusstr)!? |
| 16:23.18 | p3nguin | If you are responsible for everything else, too, maybe you don't have time to learn Asterisk. Is outsourcing an option? |
| 16:26.04 | mcrownover | outsourcing would be great if the cost weren't an issue - we used to host our pbx with a local source but it was upwards of $50k a year |
| 16:26.54 | drfreeze | Is there a way to specify a moh file for callers? |
| 16:27.06 | p3nguin | What kind of PSTN connectivity are you going to have on your in-house PBX? |
| 16:27.23 | drfreeze | I have callers dialing in - in context inbound |
| 16:27.28 | p3nguin | drfreeze: musiconhold.conf and extensions.conf |
| 16:27.30 | mcrownover | our pbx is working, but i know it is not optimal - and i need to fix little issues like xml phone directories on the desk phones |
| 16:27.45 | drfreeze | There are two offices - each with a DiD |
| 16:28.02 | drfreeze | so, some callers dial in 1000 and others 2000 |
| 16:28.06 | mcrownover | we only have SIP from our provider |
| 16:28.34 | drfreeze | I would like to have the callers who dialed 1000 to hear a different moh that the callers who dialed 2000 |
| 16:28.53 | p3nguin | drfreeze: Define two classes in musiconhold.conf. |
| 16:28.53 | drfreeze | I'm guessing, that I will need to create a context for each |
| 16:29.08 | p3nguin | No, you can use a single context for two extensions. |
| 16:29.13 | drfreeze | p3nguin: can you point to some examples? |
| 16:29.27 | mcrownover | back in a bit - small issue down the hall |
| 16:29.28 | p3nguin | The sample musiconhold.conf has examples. |
| 16:31.07 | drfreeze | p3nguin: didn't see anything that would work for this scenario |
| 16:31.33 | p3nguin | After you have defined TWO classes, one for each moh you want, then you specify which moh to use in each extension (extensions.conf). |
| 16:31.50 | p3nguin | Let's call them class1 and class2. |
| 16:32.42 | p3nguin | In one extension, you use Set(CHANNEL(musicclass)=class1) |
| 16:32.50 | p3nguin | In the other, use Set(CHANNEL(musicclass)=class2) |
| 16:33.01 | p3nguin | That's all there is to it. |
| 16:33.35 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:33.37 | p3nguin | I have to take care of something for about 15 minutes. I'll be back to help if you aren't done with it. |
| 16:34.24 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 16:34.32 | *** join/#asterisk DanFromUK (~IceChat77@2.30.234.49) |
| 16:35.24 | drfreeze | p3nguin: cool. I'll give that a try. Thanks |
| 16:38.40 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 16:38.44 | *** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 16:40.09 | DanFromUK | hi, if a user has an analog sip adapter, whats the best way for allowing the user to place a call on hold, and make a second code, and then allow the user to return to the original call? |
| 16:41.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:49.20 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:51.26 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 16:53.56 | p3nguin | drfreeze: Back. Did you get it? |
| 16:56.09 | *** join/#asterisk cadmium (1000@adsl-99-121-236-244.dsl.sfldmi.sbcglobal.net) |
| 16:57.07 | cadmium | Hi, i'm using exten => _1NXXNXXXXXX,n,Set(FROM_DID=12142421782) but its not setting that as the caller iD |
| 16:57.18 | cadmium | any idea what statment I need to use to sed the CID properly? |
| 16:58.00 | navaismo | Set(CALLERID(num)=) |
| 16:58.24 | cadmium | ;exten => _1NXXNXXXXXX,n,SetCallerID(12142421782) |
| 16:58.30 | cadmium | this causes an error |
| 16:58.45 | Andee | 16:58 < navaismo> Set(CALLERID(num)=) |
| 16:58.52 | Andee | Set(CALLERID(num)=12142421782) |
| 17:00.00 | cadmium | k pbx.c:4232 pbx_extension_helper: No application 'SetCallerID' for extension (MITUS, 12149013232, |
| 17:00.19 | navaismo | ¬¬' |
| 17:00.59 | *** join/#asterisk macroevolve (~macroevol@c-98-234-125-202.hsd1.ca.comcast.net) |
| 17:01.10 | Qwell | cadmium: Don't use SetCallerID. Set(CALLERID(num)=1234) |
| 17:01.35 | cadmium | Asterisk 10.0.0-rc2, Copyright (C) 1999 - 2011 Digium, Inc. and others. |
| 17:01.58 | macroevolve | hi i was wondering. when customer calls a customer service number. when the agent picks up and decides he needs to transfer the customer to a more experienced agent, my thoughts are there are two ways to do this. Either the experienced agent is already waiting on some sort of queue line and can be hot transferred in to or the agent can initiate a separate call to the experienced agent's |
| 17:01.58 | macroevolve | extension and when connected can bridge the calls (experienced agent wasn't waiting on queue line). If in the latter case, the experienced agent's extension had auto-answer turned on, is there a difference in connection speed between the two approaches, or would both approaches take < 1 s to connect the experienced agent into the call? |
| 17:02.07 | p3nguin | You don't set things FROM DIDs, anyway. DIDs are for inward dialing, hence "Direct Inward Dialing." |
| 17:02.59 | cadmium | Qwell thanks! |
| 17:03.00 | cadmium | working now! |
| 17:03.32 | cadmium | and thanks also Andee, i had already tried that one |
| 17:03.36 | cadmium | i guess i don't have that application? |
| 17:04.06 | [TK]D-Fender | cadmium, that app was deprecated something like FIVE YEARS ago. |
| 17:04.18 | [TK]D-Fender | in * 1.2 |
| 17:04.36 | [TK]D-Fender | 6 branches ago |
| 17:05.09 | cadmium | ahh |
| 17:06.21 | cadmium | can you send text messages with asterisk ? |
| 17:06.53 | [TK]D-Fender | No |
| 17:07.57 | cadmium | any recomendations for wholesale sms carrier gw ? |
| 17:08.38 | *** join/#asterisk flan (~flan@23.17.142.228) |
| 17:09.36 | flan | Does anyone know how, inside of Bridge(), to suppress DTMF pass-through, so that the other party doesn't get keypress data? |
| 17:10.06 | *** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909) |
| 17:12.58 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-207.mobile.uci.edu) |
| 17:13.22 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 17:15.10 | cadmium | flan maybee a sip peer setting? |
| 17:16.44 | flan | cadmium, any idea what it might look like? I haven't noticed anything helpful-looking in sip.conf's documentation, but I have noticed that combinations of settings make things work in unexpected ways. |
| 17:17.19 | cadmium | dtmfmode=inband. |
| 17:18.09 | mcrownover | If $$ wasn't an issue would any of you choose a Cisco pbx over an asterisk one?? |
| 17:18.12 | flan | I've currently got it at 2833 (and have both parties configured the same way). I want to take the event out of the stream, not keep it in there. |
| 17:18.52 | flan | That would depend on what you need it to do, mcrownover. Asterisk is probably a lot more flexible, if more involved to configure. |
| 17:19.32 | cadmium | i don't see an option to disable it completely for a peer |
| 17:19.45 | p3nguin | mcrownover: No. |
| 17:20.22 | cadmium | mcrownover never |
| 17:20.25 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 17:20.45 | drfreeze | p3nguin: yes, got it working |
| 17:20.45 | drfreeze | thanks |
| 17:20.49 | cadmium | i thought call managers were for suckers :P |
| 17:21.45 | p3nguin | mcrownover: For agents waiting in queue, the agent calls the login number and logs in. He/she then sits in the queue listening to either silence or moh. Calls entering the queue are then connected to the agent who is waiting. The time to connect can be adjusted with dial plan. |
| 17:23.42 | flan | cadmium, thanks for confirming that. I'll probably have to weigh the cost of hacking RTP (or doing aggressive firewall-filtering) versus the possible annoyance factor. |
| 17:23.51 | p3nguin | The time can also be adjusted with the member delay value in the queue. |
| 17:24.53 | cadmium | flan I would think you could disable it ... or chose which channel was the receiver in a bridge call |
| 17:24.59 | cadmium | somtiems i lose dtmf when i conference |
| 17:25.07 | cadmium | and dtmf only responds to the person i conference |
| 17:25.12 | cadmium | and they have to do all the number pressing |
| 17:25.52 | flan | I know ConfBridge hides it. Looking at the Asterisk debug log, AMI events, tcpdump, and some debug lines I added to the code, it's doing an explicit retransmission of the event. |
| 17:26.31 | flan | But there doesn't seem to be a simple flag or channel variable that can be set (documented, at least) to turn off that behaviour. |
| 17:26.48 | cadmium | right.. you can always hack the source :P |
| 17:27.15 | cadmium | wish i knew |
| 17:27.20 | cadmium | sure some guys could tell you more |
| 17:27.29 | flan | Yeah, which is what I'm considering doing. I just don't want to redo someone else's work and possibly provide two implementations for the exact same thing if I publish the patch, though. |
| 17:27.57 | cadmium | oh .. if i hacked the source i wouldn't publish the patch i'd just be happy it was working :P |
| 17:28.49 | flan | Sure, until someone comes along after you and tries to maintain your fix. =P |
| 17:29.04 | flan | Hence weighing the costs. |
| 17:29.45 | *** join/#asterisk Reenigne_Esrever (~reenignee@mail.trmcopycenters.com) |
| 17:30.54 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 17:31.13 | Reenigne_Esrever | Hello, I'm thinking of registering for the Digium class coming up in April in Vegas but was hoping to get some feedback from someone who has attended the class whether or not they think it is worth it. I'm paying for it out of my personal account to further my knowledge and opportunities with asterisk |
| 17:41.07 | *** join/#asterisk mintos (~mvaliyav@114.143.162.200) |
| 17:42.19 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:46.20 | [TK]D-Fender | Reenigne_Esrever, Depends what topics they are going to cover that yuo care about, how much you already know, how long it would take you to acquire any knowledge you'd get from there by other means & times... |
| 17:46.39 | [TK]D-Fender | Reenigne_Esrever, This is a key reason I've never left town for one fo these before... |
| 17:46.58 | [TK]D-Fender | Reenigne_Esrever, Your needs may be different Judge accordingly |
| 17:49.47 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 17:51.50 | joesuffceren | I have two fractional T1s (pri_cpe,esf,b8zs) connected to a single box (two digium single span T1 cards). When I do dahdi show status, but cards show up as ok (no alarm). When I do PRI show span 1, it is up, provisioned, and active. PRI show span 2 show down, provisioned, active. Where should I start troubleshooting why span 2 is down? Span 2 is new and has not been functional previously.... |
| 17:51.52 | joesuffceren | ...Span 1 is existing and is still functioning properly |
| 17:53.10 | Reenigne_Esrever | [TK]D-Fender, I was going to sign up for both classes they offer if I'm going to take the time to fly out of town, I have minimal knowledge on asterisk, I didn't setup our last system only maintained the configuration etc... after it was installed |
| 17:53.49 | Reenigne_Esrever | I have since then installed asterisk from source on my home pc and setup a simple dial plan, I want to take it to the next level and build a solution that would be reliable in an enterprise environment |
| 17:54.27 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 17:54.31 | Reenigne_Esrever | the $5k price is a little high and that is why I was hoping to hear from someone who has taken any class from them if they thought it was worth it |
| 17:54.52 | jaytee | The Advanced class is worth it just for the swag. The backpack laptop bag is awesome. I also got a Polycom 331, a TDM410 card and a TE110 card along with an orange * Geek t-shirt. |
| 17:55.52 | Reenigne_Esrever | jaytee, did you have any asterisk knowledge before taking the advanced class? |
| 17:57.51 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 18:01.51 | mcrownover | i am considering going the same classes in May in Huntsville |
| 18:10.25 | [TK]D-Fender | Reenigne_Esrever, Well you're in "general" territory... unless they really target your needs across the board I'd save up on it and just spend the time getting your hands dirty and drilling specific bits towards your goal. |
| 18:14.38 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 18:15.43 | bmoraca_work | Asterisk 10 should be able to detect faxes over a sip channel by default, correct? |
| 18:16.31 | p3nguin | I wouldn't think so -- you have to enable it with the faxdetect setting. |
| 18:17.22 | bmoraca_work | well i understand that...but i mean if i have that enabled, there shouldn't be anything else I have to do (any other modules or anything) |
| 18:18.35 | p3nguin | I don't know of anything else. Just enable it and create the necessary fax extension within the context where the call lands. |
| 18:19.47 | mjordan | bmoraca_work: there are other modules, but it depends on your configuration. Are you using Fax For Asterisk? Or res_fax_spandsp? |
| 18:20.06 | p3nguin | For fax detection, no other modules are needed. |
| 18:20.07 | bmoraca_work | no, i just want to use app_fax and sip fax detection to redirect |
| 18:20.13 | bmoraca_work | i didn't believe so |
| 18:20.23 | p3nguin | It's built into chan_sip. |
| 18:20.23 | bmoraca_work | i think i know what the problem is, though |
| 18:20.59 | mjordan | p3nguin: true, but it isn't going to do a whole lot without something to receive the fax, act as a T.38 gateway, etc. |
| 18:21.34 | bmoraca_work | app_fax receives faxes well enough |
| 18:21.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:21.41 | bmoraca_work | and i don't need t.38 |
| 18:24.01 | *** join/#asterisk eject_ck (~eject@h82.241.159.dialup.iptcom.net) |
| 18:24.07 | eject_ck | Hi akk |
| 18:24.09 | eject_ck | all |
| 18:25.06 | eject_ck | how can I get SIP peer external IP address from cli ? When I do sip show peer 100, then I see Reg. Contact with internal IP address (peer is behind NAT) |
| 18:26.56 | p3nguin | "sip show peers" will show the peers' public addresses if the peers are behind a NAT somewhere else, or the local address if they are on the same LAN as Asterisk. |
| 18:27.43 | p3nguin | And sip show peer 100 would show Addr->IP: followed by the public address. |
| 18:28.30 | p3nguin | The Reg Contact field is the only one containing the private address. |
| 18:28.56 | jdoe | does automon/automixmon record the call from its start and delete unless the feature code is hit, or does it start recording after? |
| 18:29.42 | p3nguin | It begins recording either when the app is run or when the call goes answered, whichever is later. |
| 18:30.38 | jdoe | hmm. |
| 18:31.48 | p3nguin | If you start mixmonitor on a channel which is not up, it does not start recording until it goes "Up." |
| 18:32.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:32.18 | p3nguin | If the channel is already answered when you start mixmonitor, it starts recording immediately. |
| 18:32.56 | jdoe | p3nguin: this is the feature-code triggered version though, not MixMonitor() itself. |
| 18:33.24 | p3nguin | Is it not the exact same application? |
| 18:33.29 | jdoe | and what I was asking was more if I punch *1 in a call, does it record from that point onward, or does it record *every* call and only save flagged ones or similar |
| 18:33.42 | p3nguin | It records the current call. |
| 18:33.52 | p3nguin | from the point you press the keys to start. |
| 18:33.57 | jdoe | right, thanks. |
| 18:34.08 | p3nguin | When the call ends, the recording ends. |
| 18:34.18 | jdoe | yeah. |
| 18:34.37 | p3nguin | If you want to record every call for its entirety, use MixMonitor() in dial plan. |
| 18:35.14 | jdoe | yeah, I think that's what I'll be doing... basically I want to be able to record calls, but I may not care until something of importance has already been said. |
| 18:35.34 | jdoe | so I may need to use mixmonitor in a dial plan, and have it auto-deleted, unless someone hits a feature code or something. |
| 18:37.06 | p3nguin | That could work. Make a followup after the call where the agent has to press one key to save the recording or another to delete it. |
| 18:37.28 | jdoe | yeah that could work too. |
| 18:40.36 | p3nguin | Or have a batch job run every night to delete recordings. Anything that needs to be saved could have already been copied or moved to a safe place. |
| 18:40.50 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:44.11 | leifmadsen | I had a cronjob nightly that moved files to an ftp server remotely then deleted the files from the HD |
| 18:51.36 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:04.20 | *** part/#asterisk cadmium (1000@adsl-99-121-236-244.dsl.sfldmi.sbcglobal.net) |
| 19:04.31 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:07.32 | eject_ck | p3nguin: thank you! |
| 19:07.51 | eject_ck | I'm getting messages in console: [2012-03-13 21:58:38] NOTICE[1786]: chan_sip.c:9581 copy_header: No field 'CSeq' present to copy |
| 19:08.28 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 19:10.54 | drfreeze | Anyone have a config setup to connect a Gigaset S675 IP to asterisk? |
| 19:11.18 | drfreeze | I keep getting 'registratation failed' or 'server not accessible' |
| 19:12.17 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 19:12.43 | eject_ck | drfreeze: what about debugging problem ? |
| 19:13.28 | eject_ck | check sip traffic with wireshark to see registration info |
| 19:15.05 | *** join/#asterisk tdillon (~travis@208.85.166.100) |
| 19:15.53 | tdillon | I have asterisk version 1.4.22 and i am having dtmf out of order issuse |
| 19:15.57 | tdillon | any ideas? |
| 19:19.55 | bowzak | any good softphones that support launching an external URL when calls come in? |
| 19:19.55 | [TK]D-Fender | Upgrade. You over 20 releases behind on a branch that is 5 branches old. |
| 19:22.27 | navaismo | bowzak, zoiper |
| 19:23.29 | tdillon | If I upgrade will that fix the dtmf issue? |
| 19:25.00 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 19:28.56 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 19:31.37 | Gugge | tdillon: if it doesnt, you will at least have a better chance getting help :) |
| 19:32.07 | tdillon | Alright. That is what I was afraid of. |
| 19:34.49 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:40.40 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 19:44.56 | *** join/#asterisk salviadud (~blasko@187.162.141.20) |
| 19:46.39 | bowzak | thanks navaismo. Voiper is popping up the call log window in my smartertrack. there are query values associated with the ap... what's the stringname for incoming DID ? |
| 19:48.02 | drfreeze | eject_ck: I've got some debug data |
| 19:48.26 | drfreeze | do you have recommendations for tcpdump command to best collect SIP registration? |
| 19:49.11 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:49.28 | salviadud | quick question |
| 19:49.47 | salviadud | those digium phones, do they support IAX2? |
| 19:50.32 | salviadud | can't believe the spec sheet only mentions SIP |
| 19:51.49 | pabelanger | salviadud: no |
| 19:51.55 | pabelanger | they are SIP phones |
| 19:52.02 | salviadud | that's odd |
| 19:52.27 | eject_ck | drfreeze: sure |
| 19:52.54 | eject_ck | tcpdump -ni eth0 -s 65500 port 5060 |
| 19:53.23 | eject_ck | where eth0 is your corresponding NIC |
| 19:53.49 | eject_ck | tcpdump -w dump.pcap -ni eth0 -s 65500 port 5060 |
| 19:55.00 | eject_ck | drfreeze: check this post ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/ |
| 19:55.59 | [TK]D-Fender | salviadud, Inter Asterisk eXchange. Not Phone To Asterisk. |
| 19:56.13 | drfreeze | eject_ck: cool |
| 19:56.26 | eject_ck | and this http://wiki.wireshark.org/SIP |
| 19:56.43 | [TK]D-Fender | salviadud, IAX was not built with phones in mind and Digium wants to make sure not to scare off customers on a statistically proprietary protocol that wasn't built with phones in mind |
| 19:57.05 | eject_ck | [TK]D-Fender: AFAIK somme phones support IAX |
| 19:57.22 | [TK]D-Fender | "some". 99% or which are cheap crap |
| 19:57.29 | [TK]D-Fender | of* |
| 19:57.38 | [TK]D-Fender | The rest = meh |
| 19:57.43 | [TK]D-Fender | nothing worthwhile |
| 19:57.53 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:58.03 | eject_ck | to be honest - I've never used it |
| 19:58.09 | eject_ck | http://www.voip-info.org/wiki/view/Asterisk+IAX+clients |
| 19:58.22 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 19:58.25 | eject_ck | SNOM should be good, no ? |
| 19:58.28 | salviadud | yeah, I guess you got a point |
| 19:58.45 | salviadud | if I got 2 offices, and want to handle them with the same extensions |
| 19:58.59 | salviadud | it be easier to just trunk them through IAX and locally set a bunch of sip phones |
| 19:59.32 | eject_ck | salviadud: you can do that with sip as well :) |
| 20:00.00 | eject_ck | jusify your dialplan :) |
| 20:01.11 | salviadud | well, I think its a lot more clean to trunk with iax |
| 20:01.18 | salviadud | less ports to worry about |
| 20:04.35 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 20:19.09 | *** join/#asterisk wonderworld (~ww@dsdf-4db54bf5.pool.mediaWays.net) |
| 20:19.58 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-dajbovqorrkhydqt) |
| 20:20.33 | rrittgarn | Afternoon. Having an issue with localzed park. I have 3 separate parkinglots defined (the default, and two separate named parkinglots). I am able to dial the parking number, and then see my BLFs / pick up the call. If i do a supervised transfer to park, it works, however if I blind transfer to it, I cant pick it back up. Any ideas? |
| 20:22.50 | jdoe | ... I can make channel variables inherited by setting adding underscores... is there a way to get reverse inheritance? |
| 20:22.55 | *** part/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 20:23.16 | jdoe | ie have a channel variable set in a ... sub-channel propagate up to parents? |
| 20:26.19 | jdoe | ah, I can use SHARED. |
| 20:32.43 | *** part/#asterisk eject_ck (~eject@h82.241.159.dialup.iptcom.net) |
| 20:34.21 | bowzak | uff.. stupid trixbox. finally worked for like 4 days after a hard reboot, but now i'm getting retransmits from hell again |
| 20:43.49 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:45.52 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 20:47.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.28 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:54.42 | *** join/#asterisk timahvo1 (~rogue@41.80.66.51) |
| 21:01.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:25.56 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001626.dsl.bell.ca) |
| 21:26.09 | dijib | anybody familier with shorewall? |
| 21:26.24 | *** part/#asterisk libryder (~libryder@libryder.com) |
| 21:27.56 | dym | dijib: to an extent - shoot your question and cut the metaquestions. |
| 21:29.07 | dijib | simpe question. is the network zone fw behind firewall or net facing? |
| 21:30.04 | dijib | like should i be using my 'net' zone for my internal or external NIC? |
| 21:30.37 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:30.55 | dijib | everytime i start shorewall i get locked out. i believe i have the firewall rules correct and a other basic configurations |
| 21:31.07 | dijib | 2nic system, one inside one out |
| 21:39.52 | dym | dijib: thats up to you. the zone "fw" is the machine the firewall runs on itself. |
| 21:40.03 | dym | whatever you name the other zones is completely up to you |
| 21:45.01 | mcrownover | if i were needing to deploy phones at a remote office (about 30 or more lines) would it be better to set up a second asterisk pbx in some sore of cluster or just let them register on the one via the site to site VPN? |
| 21:45.59 | dym | mcrownover: both could be done really. depends on your call capacity |
| 21:46.39 | mcrownover | dym: i am using g.729 and have 40 sip trunks at the main office |
| 21:46.56 | mcrownover | on a 12mb pipe |
| 21:47.05 | mcrownover | the second office has a 6mb pipe |
| 21:47.45 | dym | then calculate the capacity |
| 21:49.19 | mcrownover | http://www.asteriskguru.com/tools/bandwidth_calculator.php - according to this calc i need 4.6mb to support 100 alls in and out |
| 21:49.31 | mcrownover | using SIP and g.729 |
| 21:49.55 | mcrownover | sorry 100 calls |
| 21:51.47 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:52.49 | jdoe | anyone here logging rtcp data for call quality stats? (and if so, what are you logging?) |
| 21:54.11 | navaismo | jdoe, what i do is run tcpdump for a while, then open the cap file with wireshark, select voipcalls and check the completed calls with the player using the rtp time stamp |
| 21:56.47 | jdoe | navaismo: I'm looking to do something a bit more automated :P |
| 21:57.14 | *** join/#asterisk serafie (~erin@nat/digium/x-zxxxxffawgjnprom) |
| 22:04.26 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 22:07.41 | rrittgarn | anyone have any experience on how asterisk would handle a Blind Transfer differently than a transfer? Specifically on an Aastra? |
| 22:09.13 | navaismo | nope but you can enable the sip debug to see it |
| 22:14.00 | dym | dijib: and? |
| 22:16.45 | dijib | and. |
| 22:16.49 | dijib | uhm sorry i was afk |
| 22:16.50 | dym | okay then |
| 22:26.22 | dijib | i dont know whats up. anytime i enable shorewall im locked out |
| 22:28.03 | dym | dijib: locked out ssh wise? |
| 22:28.16 | dym | you must have an error in your config. |
| 22:28.32 | dijib | everything wise and 22 is defined in my firewall ruiles |
| 22:28.57 | dym | Shorewall Rule #1: Never disconnect a host after a major rulechange, without trying to connect to it on a seperate shell and testing that still works. |
| 22:29.01 | *** join/#asterisk wonderworld (~ww@dsdf-4db557b8.pool.mediaWays.net) |
| 22:29.13 | dym | dijib: There must be an error. |
| 22:29.38 | dym | Did you set shorewall to log and then try ssh connect and see what it drops? |
| 22:31.07 | dijib | no |
| 22:31.12 | dijib | /var/log/shorewall? |
| 22:31.14 | dym | No |
| 22:31.22 | dym | You need to enable logging in the policy file |
| 22:31.42 | dym | do you have four columns on there? |
| 22:31.45 | dym | or just three? |
| 22:33.06 | dijib | no clue i cant see that file right now as im on the box itself |
| 22:33.18 | dijib | on an external ssh screen instance |
| 22:33.25 | dym | you are on the box? |
| 22:33.27 | dym | connected to it? |
| 22:33.28 | dijib | yes |
| 22:33.33 | dym | well - how can you not see the file? |
| 22:33.42 | dijib | but since im playing with firewall screen irssi on another one |
| 22:34.06 | dym | Explain the situation. |
| 22:34.10 | dym | Are you locked out? |
| 22:34.16 | dym | And only connected via screen? |
| 22:34.23 | dym | (a session that runs screen) |
| 22:35.00 | dym | Lets not get on these guys nerves -> /q me |
| 22:36.07 | *** join/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net) |
| 22:36.21 | dijib | locked out |
| 22:36.37 | dijib | ive got iptables filters set to all allow |
| 22:36.43 | dym | Are you even reading me? |
| 22:36.45 | dym | Query... |
| 22:37.18 | *** part/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net) |
| 22:37.23 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:37.40 | dijib | i'll be back in a bit |
| 22:37.51 | dym | Err |
| 22:37.58 | dym | Okay, whatever. |
| 22:42.13 | *** join/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net) |
| 22:43.13 | mdiehl | Hi all. I'm looking to sign up to contribute some code to Asterisk and I have a few questions. |
| 22:43.38 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ncdgfxunqmlqzqlw) |
| 22:44.23 | jgowdy | What's the best I/O scheduler for use with Asterisk? I'm running 2.6.37 |
| 22:45.16 | jdoe | mdiehl: may have better luck in #asterisk-dev |
| 22:45.33 | mdiehl | Thanks. |
| 22:47.35 | dym | :O jdoe Are you THE J. Doe? (: |
| 22:47.43 | dym | pokes ChrisInSydney |
| 22:47.45 | dym | evening lad |
| 22:51.23 | jdoe | dym: depends on who's asking. |
| 22:51.49 | dijib | <PROTECTED> |
| 22:52.04 | dijib | let me change computers hgere |
| 22:52.32 | dijib | k bak |
| 22:52.39 | dym | jdoe: i mean "John Doe" :D |
| 22:52.47 | dym | The one on all the dummy credit cards :P |
| 22:52.51 | dijib | may i show you the confs from my simple toplogy? |
| 22:53.03 | *** part/#asterisk mdiehl (~mdiehl@173-10-242-193-Albuquerque.hfc.comcastbusiness.net) |
| 22:53.11 | dym | dijib: If you cared to read what I wrote, you'd know I asked you to query me an hour ago |
| 22:53.19 | dijib | does iptabes have an impact on shorewalls operation? its a wrapper so iptables dependent? |
| 22:53.52 | dijib | im thinking this could be fixed by adjusting my simple confs |
| 22:53.56 | dijib | its a new setup |
| 22:53.59 | dym | Are you kidding me? |
| 22:54.24 | dijib | you think logging is simpler? |
| 22:54.57 | dym | For the last time: This is an asterisk channel - if you'd like some pointers on shorewall QUERY me now. If you dont. Your loss. |
| 22:56.51 | dym | Or dont you know what a query is? |
| 23:00.56 | ChrisInSydney | hey dym |
| 23:01.51 | dym | Hey there |
| 23:02.08 | ChrisInSydney | got a whole building out. Ni lines, no providers ! |
| 23:02.28 | ChrisInSydney | panic stations |
| 23:02.40 | dym | Well now that sounds like fun... |
| 23:03.35 | dym | Dont you have PSTN fallback? |
| 23:07.13 | ChrisInSydney | no PSTN, no ISDN, no dialtone, no nothin |
| 23:07.20 | ChrisInSydney | someone has cut a line |
| 23:07.42 | dym | i can just picture the employees |
| 23:08.09 | Russ | has civilization reverted to the stone age? |
| 23:08.29 | ChrisInSydney | There is the Bat and Ball around the corner. They'll all be there |
| 23:08.37 | ChrisInSydney | pub |
| 23:08.45 | ChrisInSydney | its after 10am |
| 23:09.10 | dym | :D |
| 23:17.24 | ChrisInSydney | I wa about to rush out on site, then I said, just check next door and see, they did a door knock and no no one has a dial tone |
| 23:17.39 | ChrisInSydney | Looks like i have just sold a big GSM gateway :-) |
| 23:19.57 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:31.38 | dym | ChrisInSydney: So whats the issue? SIP provider having PSTN connectivity problems? |
| 23:32.19 | ChrisInSydney | I recon some one has put a saw through a cable |
| 23:32.29 | ChrisInSydney | or a digger |
| 23:32.51 | ChrisInSydney | http://1100.com.au/default.aspx |
| 23:32.56 | *** part/#asterisk bowzak (~bowzak@95.170.203.162) |
| 23:33.00 | dym | Sounds like fun |
| 23:33.23 | ChrisInSydney | not my fun anymore. handed it off. Now I can sell a 16 or 32 channel GSM gateway |
| 23:33.50 | ChrisInSydney | one of the clients is a small 25 seat call centre |
| 23:34.47 | dym | profit (: |
| 23:34.56 | dym | builders failure => admins win :D |
| 23:38.04 | ChrisInSydney | I really cant charge for the last hour I have spent chasing this up. |
| 23:38.11 | ChrisInSydney | comes as part of the maintenance |
| 23:38.26 | dym | But the GSM Gate? |
| 23:39.05 | ChrisInSydney | yep. I need to get a deposit first. Lets see if they will pay |
| 23:48.16 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |