| 00:06.29 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 00:12.12 | *** join/#asterisk sourcode (~code@ppp-58-8-85-239.revip2.asianet.co.th) |
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| 00:26.10 | f2Knight | leifmadsen, Thanks, I think the Set(CDR(X-Tag)=${SIP_HEADER(X-Tag)}) is what I was looking for, will try it out and find out. |
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| 00:49.24 | Elijah_ | Anyone here have experience getting asterisk 1.8 to work with google voice via gtalk? |
| 00:51.14 | *** join/#asterisk guilhermebr (~Guilherme@189.63.55.25) |
| 00:53.51 | Elijah_ | I'm connected, and the calls go through for a little bit... but then fail |
| 00:53.53 | Elijah_ | http://pastebin.com/bLcVC8WQ |
| 00:54.02 | Elijah_ | Log of the asterisk console during a call |
| 01:00.04 | *** part/#asterisk high-rez (~gus@2001:470:8115::2) |
| 01:00.07 | Elijah_ | Little higher verbosity if it's helpful: http://pastebin.com/m4g9rx1n |
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| 01:10.31 | *** join/#asterisk benklop (~ben@c-24-8-170-86.hsd1.co.comcast.net) |
| 01:10.56 | SirSquidness | Greetings! I am seeking advice/help on getting a Cisco 7906G running SIP formware to register with Asterisk. The configuration examples found on voip-info do not work |
| 01:11.05 | benklop | hello, i'm trying to set up ipkall, but i'm getting 1 way outgoing audio and a dropped connection after 30 seconds |
| 01:11.22 | benklop | i wish I had the experience to know what this means, but I don't |
| 01:12.41 | benklop | i'm also getting [Feb 2 20:08:54] WARNING[1292]: chan_sip.c:3383 retrans_pkt: Retransmission timeout reached on transmission 299d809c5b3943e65a9f186724ccfe05@66.54.140.46 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. |
| 01:12.54 | benklop | didnt see that till just now, i'll look at that doc |
| 01:13.50 | Elijah_ | One-way audio is usually a firewall issue... transversing NATs |
| 01:14.21 | benklop | right. my server is behind a nat, but I have udp 5060, 10000-20000 forwarded to it.. |
| 01:15.05 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 01:15.26 | Elijah_ | I usually forward tcp 5060 too, doubt that's the issue though |
| 01:15.53 | benklop | actually I did forward that too, but didnt mention it cause i thought it was unused |
| 01:16.22 | *** part/#asterisk bvierra (~bvierra@route1.lax.office.netdna.com) |
| 01:17.21 | Elijah_ | and externip is set to your external address? |
| 01:18.20 | benklop | i'll check that.. is it possible to use a hostname? |
| 01:19.32 | benklop | which file is that in? |
| 01:19.38 | Elijah_ | I think so... it's in sip.conf |
| 01:19.49 | Elijah_ | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 01:19.54 | Elijah_ | Look for externip, and exterhost |
| 01:20.17 | benklop | looks like i hadnt se t it! that might do it |
| 01:20.20 | Elijah_ | *externhost |
| 01:20.37 | Elijah_ | I think so, I seem to recall having to do that when I had an asterisk box behind a NAT |
| 01:21.54 | benklop | would make sense.. |
| 01:23.25 | benklop | well, now i donrt have audio either way :-P |
| 01:24.06 | Elijah_ | Is the phone local there I assume? |
| 01:24.14 | Elijah_ | Look at this, http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 01:24.26 | Elijah_ | You need to set localnet too I think to get local phones to work |
| 01:27.40 | benklop | cool, well that gets me 1 way audio again.. |
| 01:28.25 | Elijah_ | What are you using for a phone? Aside from that, I don't know... maybe someone here more experienced has an idea? |
| 01:28.30 | benklop | ah darn. the issue now is my dynamic dns name is not getting updated. |
| 01:28.44 | Elijah_ | Setting the debug/verbose levels higher may provide some insight too |
| 01:29.01 | Elijah_ | Get it working with externip first, that rules out DNS as being a problem |
| 01:30.44 | benklop | well, updated to use the correct IP and still nothing. i'll have to retuyurn to this at another time. |
| 01:30.51 | benklop | thanks for the help! |
| 01:31.02 | Elijah_ | Ok, no prob... good luck... |
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| 01:37.07 | benklop | haha! canreinvite=no |
| 01:37.42 | benklop | since my sip phones are also behind the nat and arent being forwarded those udp ports, now everything makes sense |
| 01:38.21 | Tozz_ | SirSquidness: Those phones do not work behind NAT |
| 01:38.38 | Tozz_ | perhaps that is your issue? |
| 01:38.48 | Tozz_ | We use Cisco 7940's and 7970's without issues |
| 01:39.03 | Tozz_ | but behind NAT they will not register |
| 01:40.13 | SirSquidness | Tozz_: Both the phones and the asterisk server are on the same LAN/subnet |
| 01:40.19 | Tozz_ | then it should work |
| 01:40.25 | Tozz_ | what is your problem? |
| 01:41.03 | Tozz_ | must say I never actually worked with a7906, but I guess its all the same |
| 01:41.14 | SirSquidness | The phones are not registering; When I make a call on the phones, after it times out I get (in Asterisk verbose log) "handle_request_invite: Call from '' to extension '10' rejected because extension not found" |
| 01:41.35 | SirSquidness | And Asterisk does not list any registerd SIP clients |
| 01:42.02 | Tozz_ | ok, well the log you paste is because your phone dials out, but because it is not registered asterisk assumes default context |
| 01:42.08 | Tozz_ | which does not have extension 10 |
| 01:42.21 | Tozz_ | so your problem is getting the phone to register |
| 01:42.57 | SirSquidness | I changed the default context to the same context as the phone accounts are in, and both phones have their defaultip set in the asterisk config |
| 01:43.00 | Tozz_ | do you have the XML config type or the key: value config type? |
| 01:43.02 | SirSquidness | but yes, the problem is that they're not registering |
| 01:43.03 | benklop | Elijah_: thanks for the help! you got me pointed in the right direction |
| 01:43.06 | SirSquidness | XML |
| 01:43.33 | SirSquidness | Watching the TFTPD log, the phoens don't appear to ask for the non-XML files (which is unfortunate, since they seem infintitely simpler) |
| 01:43.47 | Tozz_ | yes they are simpler;) |
| 01:43.50 | Tozz_ | I can send u a copy of mine |
| 01:43.54 | Tozz_ | maybe you can spot some differences |
| 01:44.02 | SirSquidness | That would be apreciated |
| 01:44.19 | SirSquidness | Do you know if it's possible to tell the CIsco phones to use the var: value format files instead of the XML one? |
| 01:44.32 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 01:44.57 | Tozz_ | well, we have the same phones, 1 uses XML 1 uses key value format, so I guess u can. But dont ask me how ;) |
| 01:45.33 | SirSquidness | DO you have identical firmware loaded on to them? |
| 01:47.33 | CoffeeIV | I am using Ubuntu server with the asterisk from the ubuntu repositories, and I've been having problems with segfaults and disconnections from people calling into ConfBridge. Is the asterisk in the Ubuntu repositories reliable for production use ? Should I recompile a stable version from scratch ? |
| 01:50.41 | Tozz_ | apparantly its not ;) |
| 01:52.51 | CoffeeIV | What server and asterisk do you find sufficiently reliable ? |
| 01:54.03 | Tozz_ | i would recompile from scratch |
| 01:54.15 | Tozz_ | precompiled packages are usually pretty outdated |
| 01:54.27 | Tozz_ | so perhaps there as a bug in ConfBridge that was fixed ages ago |
| 01:56.10 | Elijah_ | benklop: no problem |
| 01:56.31 | Elijah_ | Any of you guys have experience with asterisk1.8 and gtalk? |
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| 02:03.47 | CoffeeIV | is there an apt-get repository I can had that has newer, more reliable builds ? |
| 02:06.48 | KavanS | build from source! |
| 02:22.00 | Elijah_ | I just keep getting "><pho:recipient-unavailable xmlns:pho="http://www.google.com/session/phone">Session timed out" |
| 02:27.19 | CoffeeIV | Am I supposed to put the asterisk-addons inside the asterisk source directory or otherwise link them in some way ? |
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| 03:02.58 | CoffeeIV | It appears from initial testing, that installing asterisk 1.8 from the stable tarball made a huge difference. I'm going to have to file a bug report on the asterisk package in ubuntu, and tell them that shit is not up to snuff. |
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| 03:55.57 | Chewie71 | evening...trying to setup and test an outgoing route. Have 2xfxo ports setup in a trunk, but only one is connected to an outgoing line. How can I tell if my test calls are trying to use that one connected line? I make calls, and I can here a ring, but it never connects to the number I am calling externally... |
| 04:26.54 | ChannelZ | What are you Dial()ing? |
| 04:27.38 | ChannelZ | If you're dialing a group (DAHDI/G1 etc) then remove the channel from the group. Or it sounds like you only have 2 lines, so just dial the active channel since there's only one (DAHDI/1 etc) |
| 04:31.50 | Chewie71 | ChannelZ: I created 2 groups in the trunk, one for each FXO port....and that solved it. The calls are going out now. Now trying to figure out why my Ring Group isn't ringing any extensions. |
| 04:32.43 | ChannelZ | ring groups don't do what you probably thing they are supposed to |
| 04:32.52 | ChannelZ | s/thing/think/ |
| 04:34.03 | ChannelZ | if you're wanting multiple extensions to ring for a call, dial them all with & -- Dial(SIP/Foo&SIP/Bar&SIP/Woot) and such |
| 04:35.22 | Chewie71 | Not sure I understand. When a call comes in from the phone company, I want all extensions to ring. But in the Inbound Route definition, I can only pick one extension unless I setup a Ring Group. |
| 04:35.34 | ChannelZ | oh.. you must be using FreePBX |
| 04:35.37 | Chewie71 | yes |
| 04:35.59 | ChannelZ | nevermind then, no idea how to help you with that thing |
| 04:37.01 | Chewie71 | hehe...I *think* the FreePBX ring group is what I want, but it doesn't seem to work... |
| 04:37.44 | WIMPy | -> #freepbx |
| 04:38.19 | Chewie71 | oh....ok I'll try there. tks |
| 04:38.23 | ChannelZ | well ring groups and pickup groups in Asterisk allow you to control who can pickup a ringing extension other than their own. Whether or not that's what you're configuring in FreePBX or if it's hotwired the meaning and should do what you are actually talking about, I have no idea. |
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| 04:43.01 | Chewie71 | opinions on VOIP phones? I'm thinking of getting the Polycom Soundpoint IP phones...this is for a small non-profit...could be anywhere from 5-10 internal extensions depending on who they want to configure it. |
| 04:43.18 | Chewie71 | s/who/how |
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| 04:45.25 | ChannelZ | I think others here seem to like Polycom in general |
| 04:48.43 | nestAr | ive used polycom a lot, and they work well. |
| 04:49.04 | nestAr | good build quality, sound quality, great speakerphones |
| 04:51.24 | Chewie71 | easy to program and setup? They don' |
| 04:52.02 | Chewie71 | don't need anything fancy so I'm looking at the IP 335 model... |
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| 05:04.11 | Chewie71 | any thoughts on why no CID is coming in to extensions from external calls? how can I tell if the system is even getting any CID from the phone co? |
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| 05:24.33 | ChannelZ | with freepbx? who knows! |
| 05:25.13 | ChannelZ | If you were just running asterisk I'd say modify your dialplan to do a NoOp(${CALLERID(all)}) and see |
| 05:25.36 | ChannelZ | your DAHDI should be configured to pay attention to CallerID |
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| 05:29.57 | WIMPy | Analog is evil. |
| 05:32.23 | Chewie71 | meh...so I've heard... :) |
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| 05:33.52 | Chewie71 | ChannelZ: what file does that NoOp command go in? Maybe I can track down FreePBX's way to do it that way? |
| 05:35.44 | ChannelZ | well normally extensions.conf but 90% of what FreePBX does is take over the dialplan so good luck |
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| 05:55.18 | Chewie71 | maybe the telco has CID disabled on this line? |
| 05:55.35 | Chewie71 | have to check that tomorrow...tks for the help |
| 06:06.35 | ChannelZ | plug a POTS phone in and see what happens |
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| 07:39.30 | schmidts | good morning |
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| 07:55.36 | wdoekes2 | morning |
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| 08:00.28 | ChannelZ | woot |
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| 09:07.59 | Leentje | good morning |
| 09:08.55 | Leentje | i'm trying to compile asterisk so chan_dahdi gets w/PRI, installed libpri first, but its not working, can someone assist me? |
| 09:09.31 | Leentje | strings chan_dahdi.so |grep '^DAHDI Telephony' gives me "DAHDI Telephony Driver" and not "DAHDI Telephony Driver w/PRI" |
| 09:10.36 | WIMPy | You need to install libpri before making dahdi. |
| 09:12.07 | Leentje | the dahdi-linux you mean? |
| 09:13.07 | WIMPy | yes |
| 09:13.58 | WIMPy | And thereafter you need to configure and make Asterisk. |
| 09:14.11 | Leentje | ok i'll try recompiling dahdi first |
| 09:14.24 | WIMPy | Wait. Only the later. |
| 09:14.48 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 09:14.58 | WIMPy | libpri goes to chan_dahdi only. |
| 09:15.42 | Leentje | yeah thats what i thought, only need to recompile asterisk right? but i did that, to no avail yet |
| 09:16.21 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85cc.bcn.adamo.es) |
| 09:16.24 | WIMPy | Did you re-./configure it? |
| 09:16.30 | Leentje | yes |
| 09:17.35 | Dovid | helo all |
| 09:24.49 | Leentje | checking libpri.h usability... yes |
| 09:24.49 | Leentje | checking libpri.h presence... yes |
| 09:24.51 | Leentje | checking for libpri.h... yes |
| 09:25.00 | Leentje | WIMPy: guessing that is ok though... |
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| 09:25.43 | WIMPy | It's certainly not bad :-) |
| 09:26.32 | Leentje | hehe |
| 09:26.43 | Leentje | but why not w/PRI then :S |
| 09:27.29 | WIMPy | I've never looked at the title. |
| 09:28.08 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-51.nayatel.pk) |
| 09:29.26 | WIMPy | Now I have :-) |
| 09:29.39 | Leentje | :) |
| 09:31.27 | mechbangirc | ok i dont see /var/lib/asterisk/mohmp3 folder and no such file like fpm-calm-river.wav. asterisk-1.4.38 |
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| 09:40.19 | Leentje | so WIMPy , you do have w/pri ? |
| 09:40.51 | WIMPy | yes |
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| 10:18.04 | J4zen | Hi guys, has anyone had any issues with a Linksys RV082 (dual wan router) and SNOM (or other brand) SIP phones? I'm randomly losing registrations for anywhere between 5 minutes and 48 hours. This issue occurs on every phone on the site (approximatly 30 phones on site). |
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| 10:22.25 | J4zen | I've read on http://www.voip-info.org/wiki/view/Linksys+RV082 that changing the MTU will improve the situation, does that make sence to anyone? |
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| 10:31.27 | Tozz_ | J4zen: i've had the RV082 |
| 10:31.32 | Tozz_ | the device is just crap |
| 10:31.35 | Tozz_ | I threw it away |
| 10:31.42 | Tozz_ | that MTU issue didn't fix anything here |
| 10:32.01 | Tozz_ | the 'dual wan' feature doesnt work that well either |
| 10:35.32 | J4zen | You've tried the MTU setting? |
| 10:36.48 | J4zen | Were you expierencing the same issues as i am? |
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| 10:46.20 | hrhrhr | WARNING[3199]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) <-- does this look like a provider issue? |
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| 10:51.45 | schmidts | hrhrhr cause 34 could be a normal busy from the destination |
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| 11:04.19 | joachim_- | Any1 here using csync2 with asterisk ? |
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| 11:09.04 | Tozz_ | J4zen: No, my device would reboot |
| 11:09.15 | Tozz_ | or switch WAN, even if there was no problem on the primary WAN |
| 11:09.25 | J4zen | Heh, alright |
| 11:09.34 | Tozz_ | or not switch WAN when there was |
| 11:09.45 | Tozz_ | but are you using load balancing? |
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| 11:10.21 | J4zen | no, smart-link |
| 11:10.21 | Tozz_ | because that randomly uses WAN1 or WAN2. And I very much doubt that the device keeps track of everything that comes in play with SIP and RTP |
| 11:10.30 | schmidts | j4zen the MTU size will only be a problem if the registratrion from your snom phones are bigger than the mtu size + headers |
| 11:10.50 | Tozz_ | yes, usually |
| 11:10.52 | Tozz_ | but not with the RV082 ;) |
| 11:11.24 | J4zen | yeah that´s what would make sence, but the topic suggests that the default value of 1500 has issues with SIP, changing this to 1492 will fix it. |
| 11:11.30 | J4zen | I don´t see how that could improve anything |
| 11:12.12 | Tozz_ | the MTU thing is also recommended with poor WAN speeds |
| 11:12.21 | Tozz_ | really, the device is just crap. Replace it ;) |
| 11:12.34 | Tozz_ | you cannot contact Cisco, all they have is a crappy forum |
| 11:12.43 | J4zen | Hard to tell my client they just bought a crappy device from us ;-) |
| 11:13.07 | *** join/#asterisk awclin (~alinford@bdv75-5-82-230-151-206.fbx.proxad.net) |
| 11:13.15 | Tozz_ | yeah I know how that feels ;) |
| 11:13.24 | J4zen | poor WAN speed shouldn't be the issue, it's on a 20/2mbit connection with stable latency |
| 11:13.34 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 11:13.36 | Tozz_ | I sold 80 ATS devices which are all defective ;) |
| 11:13.44 | J4zen | heh |
| 11:19.25 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 11:21.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:23.04 | joachim_- | Where can I find a list of changes been made to the Asterisk Manager Interface for Asterisk 1.8 ? |
| 11:26.11 | Tim_Toady | joachim_- in the UPGRADE-1.6.txt file in he source tree (for changes since 1.6) |
| 11:26.25 | Tim_Toady | same goes for UPGRADE-1.4 ect |
| 11:26.48 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 11:28.03 | joachim_- | Tim_Toady: Thank you |
| 11:29.25 | joachim_- | Im getting errors: "Ignoring uknown event 'RTCPsent'" when connecting with astman |
| 11:30.54 | *** join/#asterisk capitan (~captain@76.91.206.32) |
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| 11:39.25 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-jtnywpbprhmtyozn) |
| 11:41.49 | J4zen | Has anyone ever had any issues with SPI (Stateful Packet Inspection) causing SIP registrations to randomly fail? Or perhaps a firewall interpeting SIP-packets (perhaps keep-alive INVITE's from the PBX) as a DoS attempt? |
| 11:42.31 | J4zen | I realise that's easy to find out by checking the firewall's logs, unfortunatly i'm unable to access the firewall at the moment. |
| 11:44.47 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
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| 12:44.53 | a_m_y | hi, anyone tried asterisk 1.4.x or 1.6.x in a 64 bit OS? tnx in advance |
| 12:45.12 | leifmadsen | all the time |
| 12:45.25 | leifmadsen | I just compile and install as normal |
| 12:45.27 | path | yes, Ive tried |
| 12:45.30 | a_m_y | or which asterisk version better for a 64bit? tnx in advance |
| 12:45.35 | leifmadsen | neither is better |
| 12:45.48 | leifmadsen | they both work fine on 64-bit operating systems |
| 12:47.05 | a_m_y | tnx so much, appreciate it |
| 12:47.36 | a_m_y | also whick linux distro is better for asterisk, for CentOS and Ubuntu |
| 12:48.08 | leifmadsen | neither |
| 12:48.12 | hrhrhr | they're all much of a muchness |
| 12:48.16 | leifmadsen | use whatever one you're comfortable with |
| 12:48.31 | luckman212_phone | muchness? eh |
| 12:48.33 | leifmadsen | I use both all the time |
| 12:48.34 | a_m_y | tnx again. |
| 12:48.35 | leifmadsen | ~newbook |
| 12:48.35 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 12:48.39 | leifmadsen | both are documented there |
| 12:48.46 | leifmadsen | I suggest you read the introductory chapters |
| 12:49.03 | a_m_y | am kinda new in irc, what do you mean by ~newbook? |
| 12:49.07 | path | well |
| 12:49.07 | luckman212_phone | cant wait 4 that book to come out |
| 12:49.09 | path | I'd go for 1.8 a_m_y |
| 12:49.19 | path | or 1.4 since both have lts support |
| 12:50.29 | luckman212_phone | is madplay or mpg321 any better than mpg123 w/ respect to zombie'ing out? i am having my mpg123 go 99% cpu a lot lately. |
| 12:50.32 | leifmadsen | a_m_y: ~newbook was a method to get the bot to show you links |
| 12:50.53 | a_m_y | ~newbook |
| 12:50.53 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 12:50.59 | luckman212_phone | makes my pbx overheat and generally sucks |
| 12:51.11 | a_m_y | kewl, sorry for being naive. But am learning. tnx |
| 12:51.16 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 12:51.25 | leifmadsen | path: 1.4 goes into security only mode come April per https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 12:51.42 | leifmadsen | a_m_y: that's what the documentation is for :) please read the links I gave you. |
| 12:52.05 | leifmadsen | path: I'm working on the review today, all of which needs to be done by February 14, at which point the book should be going to print and distribution. |
| 12:53.23 | a_m_y | tnx again. it has been bookmarked |
| 12:57.09 | Dovid | morning all |
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| 13:05.22 | *** join/#asterisk mzahariev (adminimini@piem-vsichko-v.unixsol.org) |
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| 13:21.37 | schmidts | have anybody of you ever used a software called axon virtual pbx? a customer of me use this for recording phone calls but the problem is this axon pbx doesnt send a refer out, or not even try to do a transfer |
| 13:26.03 | Tozz_ | I have a client who uses that |
| 13:26.16 | Tozz_ | works okay on 1.4, had tons of problems on 1.6 |
| 13:26.39 | Tozz_ | Useragent : NCH Software Axon Virtual PBX 2.01 |
| 13:27.17 | schmidts | tozz_ yes thats it ;) |
| 13:27.43 | Tozz_ | I dont exactly know what issues he had with 1,6. Its been a while since we've tried that |
| 13:28.31 | schmidts | tozz_ they want to use this with my hosted pbx cause they only want to record the calls from two phones but they cannot transfer calls with this two phones and i dont see anything incoming via the axon when they try a transfer |
| 13:28.35 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:28.48 | schmidts | only starting MOH and stoping MOH invites but nothing else |
| 13:28.58 | Tozz_ | all I can say is.. it works here on 1.4 ;) |
| 13:29.02 | Tozz_ | I havent used NCH myself |
| 13:29.14 | schmidts | tozz_ i am using 1.2 :D but thx |
| 13:29.22 | Tozz_ | np ;) |
| 13:29.59 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
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| 13:43.24 | BlackBishop | any idea on how do I se "my own number" in a datacard ? |
| 13:43.51 | BlackBishop | I don't like seeing that "unknown" thing... |
| 13:45.18 | BlackBishop | found at+CPBW=1,.... |
| 13:47.21 | *** join/#asterisk Arcopix (~Arcopix@78.90.85.118) |
| 13:47.29 | Arcopix | Hello |
| 13:48.28 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:48.30 | Arcopix | I am having a small but critical issue with Asterisk 1.6.2.16.1 regarding reINVITE for t38 |
| 13:48.54 | Arcopix | The asterisk should stay out of the way and the initial call setup is just ok |
| 13:49.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:49.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:49.16 | Arcopix | Though on the re-INVITE the Asterisk is changing the port in the SDP |
| 13:50.27 | Arcopix | hello? |
| 13:51.34 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 13:59.17 | Arcopix | w=/whois joachim_-, |
| 14:00.14 | *** join/#asterisk makafre (45a5c755@gateway/web/freenode/ip.69.165.199.85) |
| 14:02.52 | makafre | good morning guys, is there somebody kind enough to take a quick look at my SIP trace for a registration issue? it's here: http://pastebin.com/yEBP9SWX |
| 14:03.16 | makafre | I am using a SPA-942 |
| 14:03.39 | makafre | and it generally works, it just stops regietering every two days or so.... |
| 14:03.57 | makafre | and the status light become orange |
| 14:05.00 | *** join/#asterisk fofware (~fofware@wdctf.siup.gov.ar) |
| 14:05.09 | schmidts | makafre i dont see any repsonse from asterisk to your phone |
| 14:05.32 | makafre | schmidts: ok.... |
| 14:05.55 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:06.00 | makafre | schmidts: what would be my next step in order to debug this |
| 14:06.05 | *** join/#asterisk ELBunce (~erik@kde/developer/bunce) |
| 14:07.56 | makafre | i am using the realtime db with cache |
| 14:11.50 | schmidts | makafre is this reproduceable every time? |
| 14:20.30 | Katty | gooooooooooooooooooood morning asterisk!!!!! |
| 14:24.59 | makafre | schmidts: yes, every couple of days or so |
| 14:26.34 | makafre | i am using 1.4.26.1 |
| 14:28.57 | drmessano | 1.4.26.1 is quite old.. Is it just the one phone you are having a problem with, and others are working? |
| 14:29.52 | makafre | I think its a problem on all accounts, but not necessary all at the same time |
| 14:30.30 | makafre | I alos have noticed that if I reboot my dlink dir-655 and the phone it sometimes resolve the problem, even if I keep my dynamic IP |
| 14:31.06 | makafre | presently, other phones are working |
| 14:31.12 | makafre | but not mine |
| 14:32.21 | chuckf | gooooooooooooooooooooooooooooooood morning Katty!!!!!!!!! |
| 14:33.59 | devmod | Is there a way to make asterisk log sip to a file? |
| 14:34.01 | wdoekes2 | makafre: a case of notify packets timing out every once and a while? |
| 14:34.15 | wdoekes2 | devmod: logger.conf and sip set debug on |
| 14:34.35 | wdoekes2 | s/and a/in a/ |
| 14:35.00 | makafre | here is a sample: http://pastebin.com/yEBP9SWX |
| 14:35.02 | devmod | wdoekes2, so I need to have enabled on my console sip debug on, nothing can be done behind the scenes? |
| 14:35.14 | wdoekes2 | devmod: not console, look at file logging |
| 14:35.29 | wdoekes2 | but yes, the sip debug will have to be on |
| 14:35.41 | wdoekes2 | if you want to log separately, use e.g. ngrep(1) or tcpdump(1) |
| 14:35.52 | devmod | wdoekes2, ok thanks |
| 14:36.10 | *** join/#asterisk mgbowman (~mgbowman@78.96.148.233) |
| 14:36.24 | Katty | hugs chuckf |
| 14:36.57 | chuckf | hugs Katty |
| 14:37.06 | makafre | I also have noticed that when I use a domain name instead of an IP, that it reproduces more frequently, like once or twice a day |
| 14:37.13 | wdoekes2 | makafre: are you debugging that IP only and is asterisk trying to write to 192.168.0.199? |
| 14:37.15 | mgbowman | is there anyway to announce *both* callers upon a call being established? I'm aware of the A(x) and M(x) options of Dial, but I would like to playback an audio file to *both* callers. |
| 14:37.33 | makafre | wdoekes2: yes, i am loggin ony that IP |
| 14:37.34 | Arcopix | Hello again. Anyone has idea if it possible to make Asterisk (1.6.2) to never ever touch the SDP, to make it completely ignore it |
| 14:38.00 | makafre | wdoekes2: good point, let me check the others |
| 14:38.13 | wdoekes2 | mgbowman: can't you use both? |
| 14:38.24 | mgbowman | well both play to the called party |
| 14:38.48 | wdoekes2 | oh, okay |
| 14:39.10 | leifmadsen | Arcopix: probably not as Asterisk is a B2BUA and not a SIP Proxy |
| 14:39.23 | leifmadsen | it's going to try and translate and such where necessary |
| 14:39.35 | Arcopix | Well it is making a mess with udptl :( |
| 14:39.43 | Arcopix | I need it as b2bua |
| 14:40.01 | Arcopix | though it should never touch the sdp anyway |
| 14:40.08 | wdoekes2 | mgbowman: the G option perhaps? although I don't know you tie the calls together |
| 14:40.21 | Arcopix | since it should be always off the path of the media |
| 14:41.00 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-51.nayatel.pk) |
| 14:41.20 | mechbangirc | how can i install dahdi_dummy and none other modules. |
| 14:41.47 | mgbowman | wdoekes2: yeah I saw that but like you said ... how do you tie the calls together ... you would think there would be an option to playback to the calling user |
| 14:44.19 | mechbangirc | and also i am not able to download this: http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fwload-vpmadt032-1.25.0.tar.gz |
| 14:44.44 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 14:45.43 | makafre | ok, I think it really is trying to communicate with my LAN ip, while it's on the internet: http://pastebin.com/rfMNgrPc |
| 14:46.08 | drmessano | ~sipnat |
| 14:46.08 | infobot | [~sipnat] 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 14:46.12 | drmessano | makafre, ^^ |
| 14:49.44 | makafre | drmessano, thanks a lot, I indeed change the nat=no DB field to yes and prune my peer, it recovered its registration instantly |
| 14:50.36 | makafre | I was so sure it was already set to yes... |
| 14:54.06 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 14:54.35 | devmod | What does "8 - Congestion " equate to in a SIP call? |
| 14:54.46 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 14:57.07 | makafre | devmod: this page might help you: http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 14:57.35 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-lxhjkxpvzsryygsr) |
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| 15:00.11 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:05.10 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:08.45 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:08.54 | SaiSoma | quick question, is there a cli to see all calls currently on hold? |
| 15:09.51 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 15:10.55 | *** join/#asterisk mechbangirc (~mechbangi@115-186-140-105.nayatel.pk) |
| 15:11.06 | Kobaz | you can kind of figure it out from sip show channels |
| 15:11.08 | mechbangirc | how can i compile dahdi without internet |
| 15:12.07 | sysreq | SaiSoma: as Kobaz said, if from the linux CLI you do a asterisk -rx 'sip show channels' | grep ' Yes ' | wc -l .. you'd kind of know how many SIP channels are currently on hold. |
| 15:13.04 | SaiSoma | sweet, thanks sysreq and Kobaz |
| 15:13.19 | Kobaz | yeap |
| 15:13.19 | sysreq | or just a grep -c (count), for that matter.. nevermind the wc -l |
| 15:13.35 | SaiSoma | *nod* |
| 15:22.36 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 15:26.26 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:30.51 | *** join/#asterisk Andrew__M (62712e9c@gateway/web/freenode/ip.98.113.46.156) |
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| 15:42.47 | BlackBishop | I have an extension 123 defined in extensions.conf .. any way I could make it not answer 'till I verify some stuff ? |
| 15:43.30 | BlackBishop | I want it to keep ringing 'till I check what datacards are free |
| 15:43.41 | BlackBishop | and if none are free I want it to reject the call |
| 15:43.48 | *** part/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld) |
| 15:44.35 | leifmadsen | BlackBishop: just don't Answer() the call |
| 15:44.45 | leifmadsen | or play back a prompt |
| 15:44.52 | leifmadsen | you don't have to do anything special |
| 15:45.41 | BlackBishop | I don't have any Answer() so far .. |
| 15:45.47 | BlackBishop | yet it answers.. |
| 15:45.49 | pabelanger | Progress() |
| 15:46.22 | pabelanger | Oh, no Progress(). If you want the phone to keep ringing |
| 15:46.35 | leifmadsen | would have to see the dialplan and such to know what is going on |
| 15:46.39 | leifmadsen | not enough information |
| 15:47.00 | BlackBishop | ok |
| 15:47.01 | BlackBishop | one sec |
| 15:49.33 | BlackBishop | http://d3xt3r01.pastebin.com/k3z8rxmJ |
| 15:49.40 | BlackBishop | I think it answers when I call Read()... right ? |
| 15:50.43 | Skrusty | quick question, is it possible via an AGI to do a dial, while maintaining the existing audio a caller is hearing... i.e. they are hearing onhold music lets say, while the AGI is dailling another extension, but the caller doesn't hear ringing? |
| 15:51.20 | Tozz_ | Yes |
| 15:51.29 | Tozz_ | Dial() has a 'play moh' option |
| 15:51.34 | Skrusty | ah |
| 15:51.42 | Skrusty | yeah, aleady done that... |
| 15:51.51 | Tozz_ | but? :) |
| 15:52.20 | Skrusty | it assumes you are using mpg123 for MOH, because otherwise it's a new stram of audio (and the music would keep repeating) |
| 15:52.27 | Skrusty | am i right? |
| 15:52.32 | Skrusty | it seemed that way when i tried it |
| 15:52.50 | Skrusty | is it possible not to progress until the channels are bridged? |
| 15:53.46 | Tozz_ | that I dont know |
| 15:53.49 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 15:53.51 | jshriver | Greetings |
| 15:54.30 | jshriver | What is the command to send an outbound call from within the cli that just says a hello message or some pre-recording? Have done it before but forgot the command and syntax. |
| 15:55.51 | *** join/#asterisk mzahariev (adminimini@piem-skrishno-v.unixsol.org) |
| 15:56.05 | Skrusty | hmm |
| 15:56.10 | Skrusty | would be nice if you could :) |
| 15:56.14 | *** join/#asterisk tobi- (~tobi@85.220.135.170) |
| 15:56.19 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 15:56.20 | jshriver | you can just don't remember it. |
| 15:56.37 | Skrusty | heh :) |
| 15:56.52 | jshriver | basically fake a sIP id, tell it to spawn a call to either another SIP or real #, and play a gsm file or built-in test |
| 15:57.47 | BlackBishop | leifmadsen: I guess that was it .. I just have to do everything I need before Read() ... |
| 15:57.53 | jshriver | now that I think about it I think I had to create a plain text file, and just dump it in a specific directory |
| 15:57.54 | BlackBishop | which answers the call... |
| 15:58.03 | jshriver | googlign callfile now |
| 15:58.29 | *** join/#asterisk esing (~core@dslb-084-058-044-092.pools.arcor-ip.net) |
| 15:58.30 | tobi- | asterisk and queue members are behind the same nat, problem is he caller can't hear the callee but other way round it works, rtp ports specified are forwarded |
| 15:59.12 | esing | hi, I wonder if asterisk supports the original dosch&amand type III card in addition to osmocom/dedected |
| 16:00.32 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:07.40 | jshriver | is there another place to find out where callfiles go? tried putting one in /var/spool/asterisk/outgoing and the file just sits there * doesn't appear to be picking it up. |
| 16:09.04 | leifmadsen | BlackBishop: right -- anytime you play audio to an end point you're going to answer the call |
| 16:09.44 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 16:12.52 | jshriver | Is there a command within asterisk that lets you see all environment variables? Such as spool directory |
| 16:13.38 | *** join/#asterisk niekniek (~niekniek@92.70.112.34) |
| 16:13.41 | niekniek | hello! |
| 16:13.52 | leifmadsen | jshriver: ENV() |
| 16:13.55 | jshriver | ty |
| 16:14.00 | niekniek | what is the difference between TE220PEX & TE220BF if any? |
| 16:14.10 | jshriver | pbx*CLI> ENV() |
| 16:14.10 | jshriver | No such command 'ENV()' (type 'help ENV()' for other possible commands) |
| 16:14.21 | leifmadsen | niekie: the digium.com website should tell you -- probably one is PCI-X and the other isn't |
| 16:14.31 | leifmadsen | could be different configurations for echo cans etc |
| 16:14.37 | b0gatyr | morning everyone, im trying to allow another "remote" pbx to have access to my dialplan, I have created a context and assigned this context to the sip trunk on sip.conf but when someone tries to make a call it says "extension not found" can someone shed some light? |
| 16:14.39 | leifmadsen | jshriver: that's a dialplan function |
| 16:15.06 | leifmadsen | jshriver: if you want to look at environment variables from the console, just look at them from linux -- same command can be run if you prepend it with a bang (!) |
| 16:15.56 | jshriver | ty, was trying to find out what dir it's using for the spooler, /var/spool/asterisk/outgoing doesnt seem to work for call files |
| 16:16.03 | leifmadsen | look at asterisk.conf |
| 16:16.22 | niekniek | anyone? what is the difference between TE220PEX & TE220BF if any? |
| 16:16.25 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 16:16.52 | jshriver | hrm it's set correctly, maybe my callfile is broken then, use to work though |
| 16:17.21 | niekniek | TE220PEX could mean PCI Express, but B means including the echo cancellation module |
| 16:17.48 | niekniek | So what about the F? |
| 16:18.46 | jshriver | do you guys have a recommended site to post quick text snippets rather than spam channel? |
| 16:19.47 | Skrusty | pastebin |
| 16:19.57 | jshriver | Here is my call file, only changed the number but it's a phone # with area code |
| 16:20.00 | jshriver | http://codepad.org/ohXc1LcR |
| 16:21.05 | esing | jshriver do most people use asterisk in addition to GSM ? |
| 16:21.11 | esing | (less DECT related?) |
| 16:21.18 | jshriver | no clue not even sure what GSM is. |
| 16:21.26 | leifmadsen | ~pb |
| 16:21.26 | 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. |
| 16:21.44 | esing | jshriver you dont know what GSM is ??? |
| 16:22.00 | esing | Please educate http://en.wikipedia.org/wiki/GSM |
| 16:22.18 | jshriver | http://pastebin.com/0psZChQ7 |
| 16:22.47 | esing | jshriver what do people mostly use Asterisk in this channel mostly then? |
| 16:23.01 | jshriver | pbx I would guess |
| 16:23.14 | jshriver | didnt know * could be used with cell phones |
| 16:23.20 | esing | ah I see |
| 16:24.26 | esing | ye you can use asterisk for checking the vulnerability of GSM or dect |
| 16:24.51 | jshriver | what is dect? |
| 16:25.00 | esing | you pretend being a "base station" (dect) or being a "provider" (GSM) |
| 16:25.08 | esing | dect is cordless phone |
| 16:25.11 | jshriver | oh, interesting |
| 16:25.16 | esing | yus |
| 16:25.48 | esing | see here http://dect.osmocom.org/trac/dect |
| 16:25.51 | jshriver | Well good luck with your project, I need to get this call file working so I can have a method of testing outbound calling on a remote site. |
| 16:26.01 | esing | ah Okay |
| 16:26.05 | esing | Good luck you too |
| 16:26.07 | esing | :) |
| 16:26.26 | jshriver | ty |
| 16:27.49 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
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| 16:32.43 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
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| 16:47.05 | cusco | hi |
| 16:49.35 | The_Boy_Wonder | automobile |
| 16:54.03 | *** part/#asterisk esing (~core@dslb-084-058-044-092.pools.arcor-ip.net) |
| 16:55.50 | drift- | i have a question question is it possible to have a phone number on asterisk , with internal extensions foward to a cell phone so lets say , you call 954 999 9999 and asterisk picks up and says press 1 for mike press 2 for john press 3 for jake ............... if you press 1 the asterisk uses the 954 did to call my 954 cell phone |
| 16:55.53 | drift- | 954 being celll phoen |
| 16:55.55 | drift- | area code |
| 16:57.41 | leifmadsen | yes it is certainly possible |
| 17:03.01 | drift- | what should my ping be to my asterisk server |
| 17:03.14 | leifmadsen | as low as possible? |
| 17:05.59 | drift- | well i have a ping of 200 |
| 17:06.04 | drift- | does that mean 2 second delay |
| 17:06.59 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 17:07.11 | leifmadsen | that means 200ms delay (0.2 seconds) |
| 17:08.03 | drift- | hrm so thats kinda slow right? |
| 17:08.07 | drift- | 200 ms delay |
| 17:09.16 | leifmadsen | it's not awesome |
| 17:09.25 | leifmadsen | over 150ms gets noticeable |
| 17:14.23 | *** join/#asterisk awclin (~alinford@bdv75-5-82-230-151-206.fbx.proxad.net) |
| 17:15.51 | mgbowman | yay! |
| 17:16.18 | mgbowman | using Dial(...,L(x)) and some trickery I can playback an audio file to both parties |
| 17:16.43 | leifmadsen | :) |
| 17:16.56 | mgbowman | it's hackish ... but I really which there was a convenient way to announce *both* users |
| 17:17.24 | citywok | for reference our call center in the philippines is right around 200ms and it's not toooo bad. |
| 17:17.52 | citywok | that's just phone to asterisk, that doesn't count the next 20-50ms to connect the call to the pstn |
| 17:21.01 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 17:22.48 | drift- | okay |
| 17:23.03 | drift- | beacuse what i'm trying to do is create 2 asterisk servers in 2 parts of world |
| 17:23.26 | drift- | when you call 1 phone number you can speak to somone like its local device in diffrent part of world |
| 17:23.30 | drift- | avoid international fee's |
| 17:23.37 | drift- | so you call in like a conference i guess it would be.... |
| 17:24.15 | drift- | so if i'm in europe and i call this europe asterisk i can talk to somone in usa like its local call |
| 17:24.55 | citywok | yea i avoid doing that b/c i don't want to have to deal with call quality complaints over public internet across continents. it's bad enough in state. |
| 17:25.28 | WIMPy | Not sure about the current situation, but the have been times when a call to the USA was cheaper than a local call. |
| 17:26.42 | citywok | i think it's funny that it's more expensive for me to call alaska than sri lanka. lol |
| 17:27.40 | drift- | i'm sure if sarah palin was president it would be free to call alaska |
| 17:28.47 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 17:28.49 | Qwell | drift-: nah, they'd charge extra, and use that to pump oil |
| 17:28.59 | drift- | hah |
| 17:33.18 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 17:34.55 | ClintGoudie-Nice | drift-: and Russia, since you can see it from Alaska ;) |
| 17:40.25 | a_m_y | did follow the installation for 64bit on http://ofps.oreilly.com/titles/9780596517342/ch03.html. My yum broke |
| 17:40.29 | a_m_y | ^_^ |
| 17:40.30 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:40.46 | Qwell | a_m_y: "broke" how? |
| 17:40.58 | a_m_y | CentOS 64-bityum remove *.i386 && yum update -y && reboot |
| 17:41.07 | a_m_y | yum remove *.i386 |
| 17:41.19 | Qwell | uhh |
| 17:41.30 | Qwell | hmm |
| 17:41.31 | p3nguin | So you removed yum, I guess. |
| 17:41.42 | citywok | lolol |
| 17:41.48 | Qwell | where is Leif |
| 17:41.54 | a_m_y | yes, its because of yum remove *.i386 |
| 17:42.53 | Qwell | a_m_y: what does it say? |
| 17:43.02 | Qwell | like, what is the actual "brokedness"? |
| 17:43.42 | a_m_y | -bash: /usr/bin/yum: No such file or directory |
| 17:43.46 | p3nguin | ;) |
| 17:43.57 | a_m_y | when issuing yum update -y |
| 17:43.58 | p3nguin | Why did you remove yum if you wanted to continue using it? |
| 17:44.07 | a_m_y | did not remove yum |
| 17:44.19 | p3nguin | Obviously you did. |
| 17:44.21 | a_m_y | issued yum remove *.i386 |
| 17:44.41 | p3nguin | yum is included in * |
| 17:44.42 | a_m_y | that what is on the http://ofps.oreilly.com/titles/9780596517342/ch03.html |
| 17:44.44 | p3nguin | * means everything |
| 17:44.46 | leifmadsen | Qwell: right here! |
| 17:44.56 | Qwell | leifmadsen: that's not gonna work :D |
| 17:45.08 | Qwell | and really, it shouldn't be an issue anymore as of like 5.4 |
| 17:45.19 | p3nguin | Now you'll probably have to get a new yum rpm and install it manually with rpm. |
| 17:45.19 | leifmadsen | missed everything that was said and has no idea why his name was said |
| 17:45.34 | Qwell | leifmadsen: `yum remove *.i386`, in the guide |
| 17:45.43 | Qwell | http://ofps.oreilly.com/titles/9780596517342/ch03.html |
| 17:45.46 | leifmadsen | yes true |
| 17:45.52 | Qwell | should just remove that bit |
| 17:45.53 | leifmadsen | is he on i386? |
| 17:45.57 | Qwell | no |
| 17:46.01 | a_m_y | 64bit |
| 17:46.26 | leifmadsen | odd, I've never run into that problem |
| 17:46.29 | Qwell | leifmadsen: it might work sometimes, but depending on the order packages get installed in, it's not so good |
| 17:46.31 | leifmadsen | I've always done it as described |
| 17:46.50 | p3nguin | Just grab the yum rpm from one of the mirrors, install it, and continue with your day. |
| 17:46.51 | leifmadsen | installed, update, remove i386 files, reboot |
| 17:47.03 | leifmadsen | if you do the installation order a different way, I guess it could die :) |
| 17:47.41 | p3nguin | What? You expect people to follow a specific order of operation? Preposterous! |
| 17:47.45 | Qwell | leifmadsen: in this case, the yum binary is in both the i386 and x86_64 packages, so if the "wrong" (it's somewhat random) one gets removed, it could remove that |
| 17:47.46 | leifmadsen | apparently |
| 17:48.08 | Qwell | RH doesn't accept that it's a bug. I've yelled at people about it, and they just don't care. |
| 17:49.19 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 17:49.24 | drmessano | 'yum install yum' should knock that right out |
| 17:49.38 | Qwell | wait, wtf |
| 17:49.42 | Qwell | yum is noarch |
| 17:49.47 | leifmadsen | rpm -Uvh http://.../yum.rpm |
| 17:49.51 | Qwell | http://mirrors.easynews.com//linux/centos/5/os/x86_64/CentOS/yum-3.2.22-26.el5.centos.noarch.rpm |
| 17:50.24 | Qwell | how the hell did that match *.i386? |
| 17:50.58 | Kobaz | hehe |
| 17:51.15 | Qwell | that doesn't touch yum at all on my box |
| 17:51.39 | citywok | fail |
| 17:51.54 | a_m_y | used CentOS 5.5 |
| 17:51.59 | a_m_y | 64bit |
| 17:52.10 | citywok | what does everyboy around here use to run their * servers? i use debian only b/c i've been using it for everything else for like 8years |
| 17:52.11 | Qwell | are you using any third-party repositories? |
| 17:52.24 | Qwell | I bet python got removed |
| 17:52.31 | a_m_y | yes python |
| 17:52.40 | Qwell | you have a third-party python repository? |
| 17:53.05 | a_m_y | let me check, sorry for the commotion. will reinstall centos 5.5 again |
| 17:53.23 | a_m_y | and redo the guide. will let you guys if this fails |
| 17:53.43 | a_m_y | again, sorry for the trouble. will simulate again |
| 17:53.56 | Qwell | a_m_y: before you do that, could you do `ls /etc/yum.repos.d/` ? |
| 17:54.03 | Qwell | how many files are in there? |
| 17:54.07 | a_m_y | yes am doing it |
| 17:54.08 | p3nguin | |
| 17:54.17 | a_m_y | removed |
| 17:54.32 | Qwell | ...removed? |
| 17:54.33 | *** join/#asterisk outtolunc (~me@c-76-21-79-73.hsd1.ca.comcast.net) |
| 17:54.47 | a_m_y | i think something on the third party probably, thats why it got removed. |
| 17:54.51 | p3nguin | Wow. Another Windows admin. |
| 17:54.54 | a_m_y | yes, no /etc/yum.repos.d |
| 17:55.02 | Qwell | oh, right, because yum got removed |
| 17:55.06 | a_m_y | i cannot yum repolist sirs |
| 17:55.09 | p3nguin | "I removed yum, so now I'm going to reinstall the entire OS." |
| 17:55.11 | Qwell | yeah I dunno. that's just plain wrong. |
| 17:55.12 | a_m_y | no /etc/yum.repos.d |
| 17:55.21 | Qwell | p3nguin: no, his system is gonna be pretty F'ed |
| 17:55.24 | drmessano | p3nguin, after rebooting, 3 or 4 times |
| 17:55.29 | a_m_y | anyone from you master when removed yum can find /etc/yum.repos.d |
| 17:55.34 | a_m_y | or reposlist instead |
| 17:56.50 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v015-112.mobile.uci.edu) |
| 17:58.03 | Qwell | leifmadsen: anyways! maybe something to look into before printing. :D that would be bad if this just sometimes happens to people |
| 18:07.25 | leifmadsen | I guess :) |
| 18:07.30 | leifmadsen | we'll see if I remember to address it |
| 18:10.17 | Qwell | when does it print, btw? |
| 18:10.22 | Qwell | or can you say? |
| 18:18.24 | leifmadsen | Qwell: I just don't know :) we're in the middle of review. Everything in by the 14th then we have no more say :) |
| 18:18.34 | Qwell | ahh |
| 18:18.42 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
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| 18:34.19 | p3nguin | On a large scale Asterisk deployment where, let's say, 1000 concurrent calls will be typical, is a single SER box in front of multiple Asterisk systems good enough to balance the calls to the Asterisk systems? |
| 18:34.29 | Qwell | sure |
| 18:34.45 | Qwell | SER is just a proxy, so it can handle much a larger volume of call setups |
| 18:35.05 | p3nguin | Do you know its limitation? 10,000? 20,000? More? |
| 18:35.11 | Qwell | no clue |
| 18:36.07 | p3nguin | Once SER is sitting there taking all the calls and sending them inbound to Asterisk(s), is there any need to interconnect the Asterisk systems that sit behind SER? |
| 18:36.29 | leifmadsen | DUNDi? |
| 18:37.08 | p3nguin | I figured DUNDi would be the way to do it, but I didn't know if they needed to interconnect or if they were fine by themselves. |
| 18:38.04 | p3nguin | I didn't know if SER was capable of doing all the work alone and then just handing off the calls to various (arbitrary?) Asterisk systems. |
| 18:54.36 | *** join/#asterisk qhartman (~qhartman@quentin.telecom.uoregon.edu) |
| 18:56.15 | qhartman | I am looking at options for replacing a ~10k extension Avaya system. Have any of you used Asterisk in that size of environment? |
| 18:58.21 | Qwell | qhartman: probably not many people have, since that's a bit larger than typical. it's not impossible though, and it's certainly been done (on a much bigger scale, even) |
| 18:59.26 | p3nguin | Asterisk can handle like 10,000 SIP registrations, right? How many concurrent calls are you looking to have? |
| 19:00.10 | Qwell | p3nguin: going by his IP address, I'd say it's for a school. I'd not want all of that on a single box. |
| 19:00.25 | qhartman | yeah, I'm with the University of Oregon |
| 19:00.35 | p3nguin | If you want to set up 10,000 phones on Asterisk, if you only have 80 calls at any given moment, I would think that would be fine. |
| 19:00.39 | p3nguin | I could be mistaken. |
| 19:01.23 | qhartman | I'm not sure what our concurrent call stats look like now, but we have ~25 PRI lines that we use |
| 19:01.36 | p3nguin | That's a lot. |
| 19:02.10 | p3nguin | That's like a 250-call capacity? |
| 19:02.17 | qhartman | yeah, I'd honestly be kinda surprised if Asterisk could do what we want, but we're fed up with Avaya, and Cisco isn't much better |
| 19:02.40 | Qwell | p3nguin: ...more like 600 |
| 19:03.03 | Qwell | qhartman: It most definitely can. It'd take a few boxes, but that's really not an unreasonable installation. |
| 19:03.10 | p3nguin | I was thinking 10 calls per T1. Did I underestimate it? |
| 19:03.10 | qhartman | yeah, total theoretical is ~630, but I know we aren't fully utilizing them |
| 19:03.29 | Qwell | p3nguin: plus inter-dorm calls |
| 19:03.36 | Qwell | which I assume there are some of.. |
| 19:03.44 | Qwell | (and surely this is the office PBX too, so much more) |
| 19:04.23 | Qwell | I think upenn has like 60k or something |
| 19:04.41 | qhartman | upenn uses Asterisk? |
| 19:04.47 | Qwell | they do |
| 19:04.51 | qhartman | Interesting |
| 19:05.04 | qhartman | I'll see if I can talk to someone over there |
| 19:05.23 | Qwell | I think I was off by like 4x, heh. This article says 15k. I thought they had more though |
| 19:05.42 | p3nguin | Always better to overprovision! :D |
| 19:05.44 | qhartman | Regardless they would be good to talk to |
| 19:05.46 | qhartman | hehe |
| 19:06.04 | Qwell | qhartman: yeah, if you've got contacts over there, I'd definitely talk to them |
| 19:06.25 | qhartman | I don't, but I'm sure I can dig someone up |
| 19:06.50 | p3nguin | I guess if you can over-sell 4:1, then 60k is a good rating for a 15k call system. |
| 19:07.12 | p3nguin | I'm just being silly. Ignore me. |
| 19:07.34 | citywok | deal |
| 19:07.45 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v015-112.mobile.uci.edu) |
| 19:08.21 | qhartman | If we migrate off Avaya, we plan on just moving our VOIP end points at first, then replacing as many of our legacy digital and analog sets as possible |
| 19:09.07 | citywok | hundreds of concurrent calls should be possible on a single box, so terminating 25 PRI's shouldn't be toooo hard. just need a lot of 4 port cards |
| 19:09.07 | Qwell | phased is definitely the way to go |
| 19:09.11 | qhartman | is there any way to run a fairly large number of analog lines via an Asterisk system? We currently have about |
| 19:09.28 | citywok | qhartman: you can use 24 port cards w/ fxs/fxo modules in them |
| 19:09.30 | qhartman | 5k analog lines, and about 3k of them will likely be phased out soon |
| 19:09.35 | Qwell | ... |
| 19:09.40 | Qwell | 5k? good. lord. |
| 19:09.42 | citywok | lolol |
| 19:09.46 | qhartman | feeding all the dorms |
| 19:09.48 | qhartman | :D |
| 19:09.48 | citywok | i feel sooo bad for you |
| 19:10.01 | citywok | i wouldn't want to deal with that on *. lol. |
| 19:10.12 | qhartman | feel bad for the guys who have to splice our 1800pr cables when they hit with a back-ho.... |
| 19:10.14 | beardy | Phase out before starting. :) |
| 19:10.17 | citywok | do students even use phones in their dorms anymore? i know i never plugged mine in and that was 5 years ago. |
| 19:10.28 | Qwell | qhartman: how are you handling it now? large channelbanks or something? |
| 19:10.33 | qhartman | not a lot, hence 3k of them going away in one fell swoop |
| 19:10.52 | citywok | i think the U.Washington stopped providing them altogether to save $. lol |
| 19:10.53 | Qwell | malcolmd: PING |
| 19:11.00 | qhartman | Our Avaya system is based on their Definity platform, it's basically a CO switch |
| 19:11.16 | citywok | what do you want to do your current system cant? |
| 19:11.23 | citywok | if it aint broke don't fix it? |
| 19:12.12 | qhartman | nothing, in terms of the system itself it's fine, but Avaya is a nightmare to deal with. They are chargin us hundreds of thousands of $$$ a year for basically nothing |
| 19:12.22 | Qwell | yeah they do that |
| 19:12.22 | qhartman | And we have some serious questions about their long term viability |
| 19:12.57 | qhartman | It's looking pretty likely that the current mngmt is going to drive the company into the ground |
| 19:13.06 | Qwell | yeah they do that too ;) |
| 19:13.16 | *** join/#asterisk Micc (~quassel@c-24-18-20-54.hsd1.wa.comcast.net) |
| 19:14.14 | Qwell | qhartman: would you mind a PM? |
| 19:14.31 | qhartman | Do the modern Asterisk management interfaces work well? Last time I looked at it closely you pretty much had to be a programmer to get it going |
| 19:14.48 | Micc | Where can I get NZ and AU DIDs? I have a customer here in the US with offices in NZ and AU and wants to get some numbers for a catelog in NZ/AU. I suggested they find a voip provider in AU or NZ and we interconnect. |
| 19:14.50 | qhartman | Qwell: not at all |
| 19:14.55 | Qwell | for a system that size you really just don't want to use any type of GUIs |
| 19:15.32 | Micc | But if its simple to get DIDs from there and its not a super long route it might not hurt to come here first. That seems like a lot of wasted travel time to come here then back. |
| 19:15.57 | Micc | Its 200+ms ping time. |
| 19:17.56 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 19:19.15 | Micc | qhartman, how big a system are you talking about? I joined in late. |
| 19:19.41 | qhartman | Micc: ultimately about 10k handsets |
| 19:20.06 | qhartman | Micc: about 25 PRI lines for PSTN connections |
| 19:20.23 | Micc | only 25 calls at a time? |
| 19:20.36 | Micc | sounds like a hotel. |
| 19:20.56 | p3nguin | 25 calls? Wouldn't THAT be nice? |
| 19:21.02 | Micc | I was going to say you won't get 10k handsets on a single asterisk box. |
| 19:21.23 | qhartman | micc: 25 PRI lines = about 630 possible calls |
| 19:21.23 | Micc | p3nguin, concurrent calls of course. smart ass |
| 19:22.41 | Micc | qhartman, oh thats more like it. yeah thats more than a single box for sure, which means using AMI would be difficult. Some custom stuff probably. But I haven't looked at the new 1.8 stuff with distributed dev state and stuff. |
| 19:24.05 | Micc | qhartman, can you break down those into departments or something? Maybe running 100 calls per box. |
| 19:24.09 | p3nguin | micc: That's what I meant, too. He was talking about call volume in the hundreds over his 25 PRIs. |
| 19:25.00 | p3nguin | I'd like to see 80-100 calls per box, if possible. |
| 19:25.18 | Micc | p3nguin, why would you want to use 25 PRIs? A good SIP provider would make more sense. I don't know if you can even get 25 PRI cards in a box. what is that 8 4 port cards? |
| 19:25.31 | Qwell | Micc: you don't use a SIP provider when you're a large univeristy |
| 19:25.36 | Qwell | -s |
| 19:25.39 | qhartman | Micc: might be possible. If we had to go that route though, I think we'd probably not go Asterisk |
| 19:25.51 | p3nguin | micc: There are 24-port cards available. |
| 19:26.47 | nestAr | Isn't there a DS3 card available as well? |
| 19:27.00 | Micc | Qwell, arent there PRI to SIP appliances they can be their own SIP provider? |
| 19:27.19 | p3nguin | Asterisk. :) |
| 19:27.20 | qhartman | Micc: those do exist |
| 19:27.32 | Micc | Qwell, at least then they could scale easier without hardware tied to specific hardware. |
| 19:28.19 | *** join/#asterisk f2Knight (~ben@c-71-236-165-148.hsd1.or.comcast.net) |
| 19:28.39 | Micc | qhartman, you might even be able to make your own with Asterisk SCF or openSIPs or something, then pass all the PRI calls to asterisk boxes over SIP/RTP. |
| 19:28.52 | f2Knight | Q: when building from SVN, how do you get the version to update correctly. So when someone looks at the CLI it shows the correct version? |
| 19:30.11 | Micc | my voicemail files keep getting out of sync lately. I never had that problem until a month or two ago. Now my customers get the voicemail files numbered wrong about once a week. |
| 19:30.20 | p3nguin | I patch my version line to correct a grammatical error... maybe you could patch yours to show whatever version you want it to show. |
| 19:30.21 | a_m_y | hi sirs |
| 19:30.22 | qhartman | Micc: Yeah, that would probably work well. We're not quite to the architecture phase of this project yet. I'm just looking to see if Asterisk could be a player. It sounds like it can, and I have a lot more reading to do.... |
| 19:31.09 | a_m_y | its was 32bit box, i have two 64bit but my collegues installed 32bit but told him to use 64bit |
| 19:31.28 | a_m_y | anyways, guide is working perfectly |
| 19:31.38 | a_m_y | again, sorry for the trouble. |
| 19:33.07 | Micc | qhartman, It absolutely can be a player I think. But I wouldn't expect to use a drag and drop call interface for that many lines. |
| 19:33.21 | Micc | you could if you broke it up into smaller departments. |
| 19:33.49 | qhartman | no, indeed not. I think the smaller departments idea wouldn't fly though. Too many moving parts. |
| 19:34.20 | Micc | qhartman, you want one switchboard for the whole thing? |
| 19:34.29 | qhartman | Our current system has several big cabinets to run our analog lines and handle the trunks, and then 4 servers to handle the VOIP stuff |
| 19:34.58 | qhartman | and, most importantly, it's all managed as one monolithic system |
| 19:36.04 | Micc | Qwell, is there anything in * SCF that might be able to provide AMI accross multiple machines sometime in the future? |
| 19:37.14 | Micc | I really don't think it would be a good idea to do that anyways. Thats a lot of data to handle in a desktop app. |
| 19:37.35 | Qwell | Micc: there is no need for AMI in Asterisk SCF. Every component is redundant/scalable like that, so it would just go "somewhere" when you send a message. |
| 19:39.20 | Micc | Qwell, it seems like AMI would still work in that environment if done properly. |
| 19:40.46 | Micc | qhartman, what type of management do you need over the whole system? You mean for transfering and monitoring calls or for creating extensions and voicemail boxes and stuff? |
| 19:40.55 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 19:42.55 | qhartman | Micc: backend management, building extensions, hunt groups, routing ,etc etc |
| 19:45.57 | Micc | qhartman, if you use realtime you can put all that stuff in a database and share from multiple machines or virtual machines. |
| 19:46.32 | Micc | qhartman, AMI is not what you want unless you want to see whats going on and transfer calls and stuff. |
| 19:46.54 | Micc | although it can create stuff too, but I would never use it for that. |
| 19:47.02 | Micc | I don't want asterisk saving my dialplan files. |
| 19:47.36 | qhartman | Micc: gotcha. We have some operators and whatnot who would be doing transfers, but they currently do that via handsets |
| 19:49.58 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 19:50.08 | Micc | qhartman, I think thats the faster way to do it anyways. dragging and dropping and typing and clicking seems like more work. |
| 19:50.24 | Micc | Although its nice to be able to lookup an extension. |
| 19:50.25 | qhartman | yeah |
| 19:51.15 | Micc | you can make a little app that ties into the extension database to lookup extensions from their desktops. Then transfer from the handset. |
| 19:51.36 | qhartman | right, that's similar to how they work now |
| 19:53.46 | Micc | qhartman, so now the question becomes, how do you register 10k phones to one monolithic system. I think you best bet there is asterisk SCF or openSER/openSIPs. |
| 19:54.37 | Micc | So now we're probably back to what you were talking with Qwell and p3nguin about. |
| 19:55.09 | qhartman | Micc: More or less. :D |
| 19:55.13 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 19:56.05 | Micc | qhartman, your problem is the same problem every ITSP using asterisk has to solve at some point. |
| 19:56.48 | *** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 19:56.48 | qhartman | Plus the added fun of university politics |
| 19:56.49 | qhartman | :D |
| 19:57.20 | Micc | So you have even more fun to be had. :) |
| 19:57.39 | Micc | When your done with this, you'll be ripe for the voip industry. |
| 19:58.14 | Dovid | hello ev1 |
| 19:58.34 | Dovid | can anyone see if they can do RDNS to: 204.145.219.1 ? |
| 19:59.06 | Qwell | Dovid: arin is the authority on that |
| 19:59.24 | Dovid | Qwell: Thanks |
| 19:59.45 | Dovid | I guess it takes time for them to propogate my NS servers |
| 19:59.56 | Micc | Dovid, nothing. |
| 20:00.07 | Dovid | as per: http://whois.arin.net/rest/net/NET-204-145-219-0-1 it should be over to me. oh well..... |
| 20:00.27 | Dovid | any one here host @ 60 Hudson in NYC ? |
| 20:00.32 | Micc | Dovid, how long ago did they change it? |
| 20:01.19 | Qwell | Dovid: today :p |
| 20:01.29 | Micc | Dovid, do you know how to use nslookup? |
| 20:02.14 | Qwell | err, Micc ^^^ |
| 20:02.18 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 20:03.00 | Micc | ??? |
| 20:03.10 | Qwell | it was updated today |
| 20:03.58 | Dovid | Micc: Today |
| 20:04.25 | Dovid | i amn just antsy. finally got my own IP's and I want to use them |
| 20:04.54 | Micc | Dovid, wait another 12-24 hours I would think it would prop to most places by then. |
| 20:05.23 | Dovid | okies |
| 20:05.30 | Dovid | from what I hear they are out of iP's |
| 20:05.48 | Dovid | lets hope ISP's in the US start supporting IPv6 real soon. |
| 20:05.56 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 20:06.56 | Micc | anyone know where I can get NZ/AU DIDs? |
| 20:07.32 | Micc | googling for it seems like a shot in the dark. There are way too many fly by night ITSPs out there. |
| 20:07.52 | Dovid | Micc: I think you can get on Voxbone |
| 20:08.07 | nestAr | http://www.voipproviderslist.com/country/voip-new-zealand/voip-providers-new-zealand/ |
| 20:08.10 | nestAr | ? |
| 20:08.11 | Dovid | Micc: I Also have some one that I work with in AU. I will PM you |
| 20:08.55 | zerohalo | I've had decent experience w/ voxbone |
| 20:09.17 | Dovid | Voxbone is good. they dont crap out like some other carriers **count Cough** DIDWW |
| 20:09.23 | Micc | I wish vitelity did international DIDs. |
| 20:10.19 | Dovid | Micc: You can try: http://voisonics.com/ |
| 20:10.24 | Dovid | i know he is located in AU |
| 20:10.34 | Dovid | I cna get you AU and NZ but just TF |
| 20:25.14 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 20:27.11 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 20:29.29 | *** join/#asterisk diegoviola (~diego@190.23.57.22) |
| 20:29.31 | *** part/#asterisk diegoviola (~diego@190.23.57.22) |
| 20:31.14 | *** join/#asterisk Alborracho (~chatzilla@200.25.216.167) |
| 20:32.18 | Alborracho | Does anyone know how to play a 44khz wav file?, I have the 8khz but the quality is really bad |
| 20:32.34 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:32.41 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 20:35.24 | leifmadsen | Alborracho: you can't play a 44KHz file with Asterisk |
| 20:36.01 | Dovid | leifmadsen: hello |
| 20:37.02 | leifmadsen | hi |
| 20:38.09 | *** join/#asterisk ttwhy (~tekkno@p4FECF031.dip.t-dialin.net) |
| 20:38.30 | brainiac | I am happy to report that I just bought "Asterisk: The Definitive Guide 3rd Ed." |
| 20:38.43 | Freeaqingme | gratz ;) |
| 20:39.04 | brainiac | thx |
| 20:39.09 | WIMPy | Anyone here with soe Radvision H.323 gateway knowledge? I try to connect to the console port but get no reaction. |
| 20:39.22 | ttwhy | hi, if i want to send a busy signal if a person is already on phone, how do i do that? (problem is -> the softphone tells asterisk that there is still a free line and "call-limit" is not more available in iax2 with asterisk 1.6) |
| 20:40.00 | Alborracho | leifmadsen: ok thx |
| 20:41.26 | leifmadsen | brainiac: +1! :) |
| 20:47.47 | ClintGoudie-Nice | In issues.asterisk.org is there any way for me to direct message someone, or track down another way to contact them? |
| 20:48.07 | ClintGoudie-Nice | <-- trying to find cmaj |
| 20:48.54 | ClintGoudie-Nice | He and I are talking back and forth about a patch, but I think there's also some config involved and I dont want to clutter the bug further. |
| 20:49.06 | niekie | [Feb 3 20:47:52] NOTICE[7022]: chan_sip.c:23497 handle_request_register: Registration from '<sip:censoredusername@bergnetworks.com:5060>' failed for '<censored-ip>:5060' - Not a local domain |
| 20:49.06 | niekie | bergnetworks*CLI> sip show domains |
| 20:49.06 | niekie | Our local SIP domains: Context Set by |
| 20:49.06 | niekie | bergnetworks.com default [Configured] |
| 20:49.06 | niekie | sylv.bergnetworks.com default [Configured] |
| 20:49.14 | niekie | Huh? :) |
| 20:52.01 | niekie | Never mind, adding :5060 to the domain string in Asterisk worked. |
| 20:52.15 | ttwhy | is there a way to get the number of active calls of a iax user in the dialplan? |
| 20:52.24 | brainiac | I look fwd to reading it (and learning from it |
| 21:01.59 | *** join/#asterisk FLeiXiuS (~asdasd@unaffiliated/fleixius) |
| 21:02.12 | FLeiXiuS | In asterisk sip.conf how do I set my peer's extension? |
| 21:02.21 | citywok | [extension] |
| 21:02.39 | citywok | http://wiki.asterisk.org has lots of good information |
| 21:02.57 | FLeiXiuS | citywok, sure but I tried setting the username to be something different than the extension. |
| 21:03.06 | FLeiXiuS | username= did not work as expeceted. |
| 21:03.23 | p3nguin_ | You put the phone's user id in the [] in sip.conf. |
| 21:03.27 | citywok | i've always made everything match up, no erason to mess with it. |
| 21:03.32 | p3nguin_ | [stanley] |
| 21:03.32 | FLeiXiuS | gotcha. |
| 21:03.52 | p3nguin_ | exten => 2000,1,Dial(SIP/stanley) |
| 21:04.00 | p3nguin_ | Now stanley's extension is 2000. |
| 21:04.11 | p3nguin_ | basic example |
| 21:04.47 | citywok | i imagine that's useful when you have multiple different users using the same * pbx. |
| 21:04.56 | citywok | that way you can have 5 different 2000's |
| 21:05.24 | FLeiXiuS | That's exactly what i was looking for |
| 21:05.26 | FLeiXiuS | Thanks |
| 21:10.53 | *** join/#asterisk mickecarlsson (~Micke@h238n1c1o1101.bredband.skanova.com) |
| 21:11.11 | ttwhy | no one any idea how to check if a iax user is already on the phone? |
| 21:11.17 | *** join/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 21:11.28 | bougyman | ugh, dahdi is going to be the death of me. |
| 21:15.56 | Dovid | ? |
| 21:17.26 | bougyman | i can't get this astribank to come up |
| 21:17.36 | bougyman | i have this problem every time this box reboots. |
| 21:17.43 | bougyman | it's like xpp_loader isn't doing the right things. |
| 21:17.50 | tzafrir_laptop | bougyman, what problem? |
| 21:17.50 | bougyman | usb:003/003 xpp_usb- e4e4:1151 Astribank-multi USB-firmware |
| 21:18.02 | bougyman | shows up, but i get that damn TIMEOUT when dahdi tries to start. |
| 21:18.15 | bougyman | tzafrir_laptop: you got me b y this last time, but I can't remember what the magic sauce was. |
| 21:18.56 | tzafrir_laptop | bougyman, I suspect that either you connect it to a USB1.1 controller, |
| 21:19.11 | bougyman | there are no usb 1.1 controllers. |
| 21:19.14 | tzafrir_laptop | or you built dahdi-tools with no libusb support |
| 21:19.27 | bougyman | INFO-xpp_usb: revision Unknown |
| 21:19.34 | p3nguin_ | ttwhy: iax2 show channels |
| 21:19.39 | bougyman | tzafrir_laptop: haven't changed a thing on this box, it was working before reboot. |
| 21:19.48 | bougyman | haven't rebooted since the last time you fixed it. |
| 21:20.13 | tzafrir_laptop | bougyman, so check the logs and see what error you got |
| 21:20.30 | bougyman | no errors in dmesg. |
| 21:20.50 | bougyman | just that silly TIMEOUT |
| 21:20.52 | ttwhy | p3nguin_, thanks, but iam searching for a way inside the extension.conf ... need to react on the count |
| 21:21.23 | bougyman | it's the revision Unknown thing that has to be right for this thing to operate. |
| 21:21.26 | bougyman | i just don't know how to get that right. |
| 21:21.32 | bougyman | i think we did some manual fw loading. |
| 21:22.04 | bougyman | but it keeps saying + logger -i -t ''\''xpp_fxloader'\''' -s -- '--------- FIRMWARE IS LOADED' |
| 21:22.17 | bougyman | when I do xpp_fxloader usb |
| 21:22.43 | bougyman | ok, just manual loaded USB |
| 21:22.55 | bougyman | that was it. |
| 21:23.06 | bougyman | yep, initializing spans. |
| 21:23.09 | bougyman | BLAH |
| 21:23.35 | tzafrir_laptop | hmm... so automatic firmware loading is not configured? grep xpp_fxloader /etc/udev/rules.d/* |
| 21:23.43 | bougyman | it's there. |
| 21:23.52 | bougyman | but xpp_fwloader usb doesn't do anything |
| 21:24.00 | bougyman | i have to do xpp_fwloader load FILE.hex |
| 21:24.26 | bougyman | anyway, it's up, thanks for the assistance. |
| 21:24.30 | *** part/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 21:26.22 | *** join/#asterisk Alborracho (~chatzilla@200.25.216.167) |
| 21:30.13 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:31.46 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 21:32.27 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 21:32.50 | i_heart_fs | hello |
| 21:32.56 | i_heart_asterisk | hey |
| 21:33.14 | i_heart_fs | what's up |
| 21:33.22 | i_heart_asterisk | not much |
| 21:34.35 | i_heart_asterisk | anybody have issues w/ polycom and transfering? when you use the softkey to transfer, asterisk does not notify AGI of the transfer |
| 21:35.01 | i_heart_asterisk | so the program using AGI thinks the polycom is on phone until that transfered call is terminated |
| 21:35.51 | i_heart_asterisk | interesting enough when you use # to transfer asterisk notifies AGI or xfer |
| 21:36.00 | i_heart_asterisk | *of xfer |
| 21:36.38 | i_heart_asterisk | i've been told this is a bug in asterisk yet digium does not consider it one so they will not fix it... |
| 21:36.45 | i_heart_asterisk | anybody know of this issue? |
| 21:38.04 | *** kick/#asterisk [i_hear_asterisk!~north@pdpc/sponsor/digium/Qwell] by Qwell (Stop switching nicks repeatedly.) |
| 21:38.54 | citywok | yea why doesn't he go to #freeswitch for help if he loves it so much |
| 21:38.56 | *** join/#asterisk i_hear_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 21:39.15 | Qwell | i_hear_asterisk: what's the issue # on mantis? |
| 21:39.17 | citywok | and that's not a bug in *, that's the phone doing the transfer itself and just bridging the two calls tgoether rather than teling * to do a redirect |
| 21:40.06 | i_hear_asterisk | citywok, i see in asterisk log:[2011-02-03 16:18:05] VERBOSE[17753] chan_sip.c: Call 00fd6b0e596439c96b2bf668193c7a6b@8.224.32.16 got a SIP call transfer from caller: (REFER)! |
| 21:40.08 | i_hear_asterisk | [2011-02-03 16:18:05] VERBOSE[17753] chan_sip.c: SIP transfer to extension 8522222@from-internal by 01163956@betaprov.i-evolve.com |
| 21:40.22 | i_hear_asterisk | so asterisk knows it's a call transfer |
| 21:40.48 | i_hear_asterisk | Qwell, i'm not certain what you are talking about |
| 21:41.11 | Qwell | If it isn't on mantis, then how are we refusing to fix it? |
| 21:41.16 | citywok | mantis = issues.asterisk.org |
| 21:41.37 | i_hear_asterisk | citywok, i understand what mantis is |
| 21:41.52 | i_hear_asterisk | Qwell, i've just been told that this is a known issue |
| 21:42.08 | Qwell | by who? if it's known, there will be an issue on mantis. |
| 21:42.09 | i_hear_asterisk | Qwell, are you saying if i post it on mantis, it will be addressed? |
| 21:42.11 | citywok | who told you that? |
| 21:42.24 | Qwell | I'm saying that it WON'T be addressed if it isn't on mantis. |
| 21:42.51 | Qwell | How could we possibly fix it if we don't even know about it? |
| 21:43.05 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 21:43.11 | i_hear_asterisk | i'm certain that there are many pple using asterisk and polycoms...how could you not know about it? |
| 21:43.23 | Qwell | I give up |
| 21:43.47 | i_hear_asterisk | citywok, someone who has many years of experience w/ asterisk told me that |
| 21:43.52 | citywok | i_hear_asterisk: i use asterisk & polycoms and don't have issues with it |
| 21:44.04 | i_hear_asterisk | citywok, do you use AGI? |
| 21:44.10 | citywok | and i_hear_asterisk considering qwell works for digium i thik you should listen to him |
| 21:44.46 | citywok | no, i use dialplan & AMI. never had to write any AGI scripts. and the phone wouldn't tell the AGI anything... |
| 21:44.53 | i_hear_asterisk | Qwell, if you work for digium, i suggest you look into this issue as it is definitely a real bug |
| 21:45.01 | Qwell | http://issues.asterisk.org |
| 21:45.06 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:45.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:45.21 | citywok | i_hear_asterisk: thats why he was asking you to submit a report, or point him to to the existing one |
| 21:45.44 | citywok | so he COULD help you. but you're not providing enough information to be helpful. |
| 21:45.58 | i_hear_asterisk | i have call traces... what else do you need |
| 21:46.07 | citywok | go to teh freaking tracker, it tells you. |
| 21:46.39 | FLeiXiuS | Does MeetMe require a dahdi now? |
| 21:46.44 | citywok | yes |
| 21:46.45 | FLeiXiuS | pseudo device* |
| 21:46.58 | citywok | yes |
| 21:47.00 | citywok | confbridge doesn't |
| 21:47.31 | FLeiXiuS | Mmm as long as I can set who can talk and who can just receive |
| 21:47.34 | FLeiXiuS | I'll be fine. |
| 21:48.19 | citywok | i'm pretty sure you can mute participants |
| 21:48.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:49.16 | FLeiXiuS | Sure, but that just disables transmits, I need to disable receives on the transmitting end. And disable transmitting audio on the receiving end. |
| 21:49.43 | FLeiXiuS | Listen and not talk - Talk and not listen. |
| 21:49.47 | FLeiXiuS | If that makes sense |
| 21:50.12 | citywok | core show application confbridge |
| 21:52.21 | FLeiXiuS | Yep just looks like you can mute mic on either side. |
| 21:52.55 | Katty | MARCO |
| 21:53.37 | Kobaz | polo |
| 21:53.49 | i_hear_asterisk | https://issues.asterisk.org/view.php?id=18745 |
| 21:53.51 | i_hear_asterisk | thanks guys |
| 21:55.36 | citywok | roflmao |
| 21:55.42 | Katty | do do be do do |
| 21:55.47 | citywok | that's like the worst bug report i've ever seen |
| 22:01.47 | Tozz_ | wow ;) |
| 22:09.21 | *** join/#asterisk nix8n82-phone (~AndChat@21.sub-174-253-166.myvzw.com) |
| 22:14.56 | devmod | Uhm, my dtmf tones only work when I call some phone numbers. Any suggestions on how to debug this? |
| 22:16.06 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:18.58 | nix8n82-phone | Are you using different codecs for different calls? |
| 22:21.26 | luckman212 | guys is there a new version of res_fax_digium.so for Ast 1.8.2.3 ?? |
| 22:22.15 | luckman212 | this page is vague: http://www.digium.com/en/docs/FAX/faa-download.php |
| 22:22.32 | Tozz_ | i can read it clearly |
| 22:22.34 | luckman212 | it says "For Asterisk version(s): 1.8.0 and later (EXCEPT 1.8.2.0, 1.8.2.1, or 1.8.2.2)" |
| 22:22.40 | Micc | is 1.8 pretty stable now? |
| 22:22.53 | Tozz_ | luckman212: what part of that dont u understand? |
| 22:22.53 | luckman212 | so does that exclude 1.8.2.3 or did they just forget to add it to the end? |
| 22:23.01 | Micc | does 1.8 have SRTP now? |
| 22:23.11 | Tozz_ | 1.8.2.3 should be supported |
| 22:23.15 | Tozz_ | since its not excepted ;) |
| 22:23.57 | luckman212 | so whatever bug was fixed in 1.8.2.0 was RE-introduced in 1.8.2.3?? |
| 22:24.33 | Tozz_ | it doesnt say it has anything to do with a bug? |
| 22:25.02 | luckman212 | ok I should have said 'change' |
| 22:25.11 | luckman212 | whatever change was made |
| 22:25.24 | Tozz_ | 1.8 is under development. Maybe they tested some fax things in the exception versions |
| 22:25.29 | Tozz_ | who knows |
| 22:25.58 | luckman212 | hmm ok, thanks |
| 22:26.34 | Tozz_ | or perhaps there was a bug introduced in 1.8.2.0 which is fixed in 1.8.2.2 |
| 22:30.45 | *** join/#asterisk Defraz (~Defraz@63.226.95.152) |
| 22:30.46 | luckman212 | I think I understand the meaning now. I interpret it as "don't use the module with 1.8.2.0-1.8.2.2 because it's broken. But it works with 1.8.0 or 1.8.2.3 because we fixed the issue" |
| 22:31.17 | Tozz_ | yes |
| 22:36.48 | leifmadsen | luckman212: 1.8.2.3 fixed an issue with res_fax_digium (latest version) |
| 22:37.12 | luckman212 | leifmadsen: thank you! |
| 22:37.17 | leifmadsen | the download page is saying, "don't use this version with 1.8.2.0 or .1 or .2 because there was a change that caused regression" |
| 22:37.28 | leifmadsen | .3 resolved that regression |
| 22:37.33 | leifmadsen | per the ChangeLog |
| 22:37.41 | luckman212 | I just re-downloaded & diff'ed the res_fax_digium.so module, it hasn't changed |
| 22:37.53 | leifmadsen | exactly |
| 22:37.56 | leifmadsen | the change was in Asterisk |
| 22:37.59 | leifmadsen | not res_fax_digium |
| 22:38.14 | luckman212 | I see. On my pbx I have /etc/asterisk/res_fax.conf and also res_fax_digium.conf |
| 22:38.39 | luckman212 | are both needed? google seems to say that settings have moved to res_fax.conf. e.g. the "ecm" setting |
| 22:44.04 | Kobaz | oh |
| 22:46.52 | *** join/#asterisk luckman212 (~quassel@pool-173-77-253-145.nycmny.fios.verizon.net) |
| 22:47.39 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 22:47.40 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 23:12.30 | voxter | anyone know if theres a way I can determine which channels the g729 transcoder is actually applying itself to? I have a particularly high number of translators in use and im not sure which channels are responsible. too many calls going by to eyeball it |
| 23:14.11 | frigidzephyr | if you would expect that only channels in g729 format would be using it, you could just grep the output of "asterisk -rx "sip show channels" |
| 23:14.35 | frigidzephyr | or "core show channels" , i forget this second which one shows a list of the formats in use |
| 23:15.31 | *** join/#asterisk philfine (~philfine@a213-22-45-215.cpe.netcabo.pt) |
| 23:16.27 | voxter | the problem is, they report as in g729 format wether they are doing passthru or a translation path |
| 23:16.33 | voxter | so you cant tell translated apart from bridged. |
| 23:17.27 | JerJer | anyone happen to know exactly what would produce a TRANSFER in the queue_log? voip-info says to use the 'built-in transfer functionalities of asterisk' but an AMI Redirect does not produce a TRANSFER in queue_log |
| 23:32.51 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 23:33.05 | *** join/#asterisk grkblood (~grkblood1@c-68-58-177-90.hsd1.sc.comcast.net) |
| 23:33.12 | grkblood | something really bad is going on right now |
| 23:33.21 | grkblood | i cant call in or out on my softphones |
| 23:33.28 | grkblood | i get a busy signal calling in |
| 23:33.53 | grkblood | and when i dial out it gets stuck on calling |
| 23:33.56 | grkblood | never rings |
| 23:35.04 | voxter | ah. I found my issue with transcoders being used.. When an incoming sip call comes into asterisk, if you have two entries for the same host, how does asterisk pick which one comes first? |
| 23:36.13 | grkblood | this keeps getting spit out in the terminal: http://pastebin.com/Kk0e3qmu |
| 23:37.41 | FLeiXiuS | Anyone use any sip clients for HTTP? |
| 23:37.59 | voxter | FLeiXiuS: only when I FTP over DNS. |
| 23:38.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:38.53 | FLeiXiuS | voxter, meh i'd prefer SSH over ICMP |
| 23:39.10 | FLeiXiuS | Im looking for a HTTP SIP Client. |
| 23:39.17 | FLeiXiuS | Something I can embed a "call" button in the web. |
| 23:39.24 | voxter | FLeiXiuS: so, what are yout alking about... sip clients for HTTP? you mean a sip client that works via a web browser over HTTP? |
| 23:39.27 | voxter | Ah. |
| 23:39.35 | voxter | InVox just launched one, i dont believe its free, but you can check that out |
| 23:39.45 | voxter | Zoiper Web works, but not for every browser, |
| 23:39.50 | FLeiXiuS | Ahh fre and open source would be great |
| 23:39.56 | voxter | theres' a few, i havent found one thats free, open, and great. |
| 23:43.05 | FLeiXiuS | Mmm, I'd love to easily adapt the source for my particular environment. |
| 23:45.24 | grkblood | anyone know about this? |
| 23:46.04 | grkblood | its taking forever for me to log in as well |
| 23:47.57 | grkblood | when i call out it says "all circuits are busy now" |
| 23:48.31 | ttwhy | maybe your softphone is blocked? |
| 23:48.39 | voxter | grkblood: presumably you are using freepbx or trixbox or something? |
| 23:48.51 | voxter | grkblood: it sounds like whatever you are dialing is not matching an outbound route rule. |
| 23:49.18 | grkblood | freepbx |
| 23:49.21 | grkblood | and its never done this |
| 23:49.31 | grkblood | im dialing right |
| 23:49.43 | grkblood | calls arnt getting in or out |
| 23:49.57 | voxter | grkblood: then... your trunk is down? |
| 23:50.04 | voxter | grkblood: the console should give you a good indication of whats going wrong. |
| 23:50.22 | grkblood | trunks are green colored |
| 23:50.30 | grkblood | im not getting any error messages |
| 23:55.32 | grkblood | when i call all i get is a busy signal, console is showing no signs of a call coming in |
| 23:55.53 | *** join/#asterisk The-Kernel (~Thomas@unaffiliated/the-kernel) |
| 23:56.23 | The-Kernel | hi, I have a SIP trunk configured, and for some reason asterisk isn't trying to register with it. Any idea why or what I missed? |
| 23:59.07 | Tozz_ | Asterisk needs to register with the SIP provider? |
| 23:59.12 | The-Kernel | yes |
| 23:59.20 | Tozz_ | do you have a register => line in sip.conf ? |