| 00:02.21 | hardwire | WiretapWork: haha.. it does when I need DID in hawaii. |
| 00:02.47 | hardwire | I hate getting kicked out of this channel via chanserv |
| 00:02.51 | hardwire | err nickserv |
| 00:02.57 | hardwire | everytime I come back I'm a newb. |
| 00:04.07 | *** join/#asterisk CyberPony (~CyberPony@cpe-075-181-142-115.carolina.res.rr.com) |
| 00:08.06 | tzafrir | WiretapWork, the timezone does (err, supposed to) |
| 00:09.03 | hardwire | indeed |
| 00:09.21 | hardwire | by the time hawaii peeps want help.. everybody only speaks japanese! |
| 00:15.01 | WiretapWork | lol |
| 00:15.13 | WiretapWork | hardwire: not true, you're almost in the same TZ as .nz |
| 00:15.24 | hardwire | WiretapWork: right.. like I can understand them either. |
| 00:15.37 | WiretapWork | you're talking to me aren't you? |
| 00:15.42 | hardwire | actually IIII am in Alaska |
| 00:15.53 | hardwire | WiretapWork: yeh well.. everything is "wonky" and stuff over there. |
| 00:16.21 | hardwire | My parent corp needs inbound DIDs in Hawaii.. We have them, but through a rather difficult to please provider. |
| 00:16.29 | WiretapWork | I see |
| 00:17.00 | hardwire | wonky... I tell you. |
| 00:17.40 | WiretapWork | Iunno... you're the ones tethered to a country you're massively distant from :P |
| 00:18.07 | hardwire | The provider we're using that has been banned from mention in this channel is having problems with us registering again after a sip reload. |
| 00:18.12 | hardwire | Too many registrations.. DEAD |
| 00:18.24 | WiretapWork | hardwire: which provider? |
| 00:18.30 | hardwire | Broadvoice |
| 00:18.42 | WiretapWork | never erd of em |
| 00:18.47 | hardwire | I have to change the IP we use with them in /etc/hosts every time I run sip reload |
| 00:19.05 | WiretapWork | jesus, not nice |
| 00:19.07 | hardwire | so I'm setting up a dedicated little machine to simply do nothing more than register to them. |
| 00:19.25 | WiretapWork | I <3 non-registering providers :P |
| 00:19.29 | hardwire | on top of that.. the provider they use in Hawaii likes to go offline for days at a time. |
| 00:19.32 | hardwire | WiretapWork: indeed. |
| 00:19.44 | WiretapWork | I just set my IP address in the provider's control panel and its all done |
| 00:43.09 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:519b:5472:b8b7:b9d) |
| 00:50.11 | luckman212 | guys I've been asked to submit a possible modification to Asterisk source code as a patch file. Do I just use the command 'svn diff include/asterisk/logger.h > logger_h.patch' --- for example? |
| 00:52.24 | WIMPy | yes |
| 00:58.00 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 00:58.18 | JunK-Y | NoOp(mooo) |
| 00:58.55 | hardwire | moooo |
| 01:03.16 | JunK-Y | whats up? |
| 01:04.57 | WiretapWork | well.. if you add an 'n' to the end of that, you have the answer |
| 01:05.04 | hardwire | not here. |
| 01:05.16 | hardwire | at the top of the earth |
| 01:05.20 | hardwire | where the moon doesn't exist. |
| 01:05.40 | WiretapWork | moon? I'm talking about the moooon |
| 01:05.47 | hardwire | makes a face |
| 01:07.05 | *** join/#asterisk coppice (~chatzilla@116.92.27.206) |
| 01:11.44 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 01:11.48 | luckman212 | arg. license agreements. |
| 01:11.51 | jasoncarter | Hello |
| 01:12.10 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 01:12.20 | jasoncarter | This is slightly off topic but does anyone know how to forward Google Voice numbers directly to a SIP URI? |
| 01:12.31 | puzzled | hi |
| 01:15.52 | ChannelZ | jasoncarter: I don't think Google can do it |
| 01:20.45 | jasoncarter | grear |
| 01:20.52 | jasoncarter | great* thanks |
| 01:21.10 | ChannelZ | You could use Asterisk's GTalk support and get it in that way |
| 01:21.48 | WIMPy | And hope they don't change the API. |
| 01:21.50 | *** join/#asterisk kumbang (~tux@180.245.137.5) |
| 01:22.00 | ChannelZ | Yeah they're still doing that |
| 01:22.28 | jasoncarter | Too risky |
| 01:22.49 | jasoncarter | I'll just sign up for a free did provider |
| 01:23.45 | ChannelZ | whatever that is |
| 01:25.53 | jasoncarter | DID number provider. Sorry I have bad hayfever |
| 01:34.19 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 01:38.51 | *** join/#asterisk d-_-b- (~d-_-b-@66.113.32.206) |
| 01:50.50 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 01:51.53 | cutlass | I'm using ARA and in troubleshooting an issue, I found that I can completely blow away the "fullcontact" value in the sip users table for a user but somehow I am still able to route calls to it...that made me wonder, exactly what is that value being used for? Does anyone know? |
| 01:59.58 | *** join/#asterisk adolfomaltez (~taro@190.86.144.223) |
| 02:02.14 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:519b:5472:b8b7:b9d) |
| 02:09.14 | *** join/#asterisk galaxywatcher_ (~galaxywat@pdpc/supporter/active/galaxywatcher) |
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| 02:24.09 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 02:24.13 | mattwj2002 | hey all |
| 02:27.13 | luckman212 | anybody know about the Transfer() cmd? |
| 02:30.28 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:31.34 | *** join/#asterisk coppice (~coppice@m121-202-39-215.smartone-vodafone.com) |
| 02:31.45 | WiretapWork | luckman212: |
| 02:31.47 | WiretapWork | ~ask |
| 02:31.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:31.55 | *** join/#asterisk BeeBuu (714706ae@gateway/web/freenode/ip.113.71.6.174) |
| 02:32.02 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 02:32.08 | BeeBuu | hi,all |
| 02:33.25 | BeeBuu | i get a question about hardware requires,please tell me if anyone know that |
| 02:34.03 | BeeBuu | can i setup a machine over 12 E1 port ? |
| 02:34.08 | WiretapWork | ~ask |
| 02:34.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:34.20 | luckman212 | alright. I've set up a context for click2dial on my pbx. The process starts with a php script that triggers AMI to originate a call into my special click2dial context. I want Asterisk to dial my extension first, and once I answer, set up the B leg of the call (outbound via SIP trunk) and then use a SIP REFER so that my phone's display reads: "To: Bob (555-1212)" as it is now, everything works but the phone says "From: Bob (555- |
| 02:34.20 | luckman212 | 1212" |
| 02:34.51 | WiretapWork | luckman212: sounds like you want to originate? |
| 02:34.54 | luckman212 | I'm on Asterisk 1.8.5 |
| 02:35.01 | luckman212 | yes I want to Originate() |
| 02:35.09 | WIMPy | BeeBuu: It really depends on what you plan to do with those calls. |
| 02:35.37 | BeeBuu | WIMPy: that's a big callcenter |
| 02:35.44 | WiretapWork | and luckman212, with regards to the 'from'/'to' thing, I'm guessing you're using a Cisco? |
| 02:35.49 | BeeBuu | about 1,000 agent |
| 02:35.50 | WiretapWork | as far as I know there's no way to change it |
| 02:35.55 | luckman212 | WiretapWork: no I have a Polycom 331 |
| 02:36.00 | WiretapWork | ah |
| 02:36.40 | WIMPy | BeeBuu: What do these 1000 agents do if only 360 can have a call? |
| 02:37.09 | BeeBuu | not that mean, all agents need call out |
| 02:37.11 | luckman212 | I know it works, b/c I've got a TAPI plugin (SIPTAPI) that sends the right commands and when I dial with that, the phone reads "To: xxxxx" while dialing. I traced it with Wireshark and I see that after the initial INVITE there is a SIP REFER |
| 02:37.16 | WIMPy | Anyway. Just forwarding voice doesn't generate much load. Things like transcoding does. |
| 02:38.28 | BeeBuu | WIMPy: i only done with 4 E1 system,so i had no idea about the bigger system |
| 02:38.36 | *** join/#asterisk freakazoid0223 (~matt@pool-173-49-209-91.phlapa.fios.verizon.net) |
| 02:38.47 | BeeBuu | now ,the new case is too big to me |
| 02:38.59 | WIMPy | BeeBuu: So what is your question? |
| 02:39.21 | WIMPy | If you have done 4 E1, you should know what the load situation is. |
| 02:40.04 | BeeBuu | i want to know how many pri channels can handle by asterisk in a machine(4 cores 4GB RAM)? |
| 02:40.42 | WIMPy | If you have done 4 E1, you should know what the load situation is. |
| 02:41.14 | luckman212 | WiretapWork: any ideas? |
| 02:41.51 | WiretapWork | luckman212: no idea, I just put up with the From: |
| 02:42.18 | luckman212 | WiretapWork: lol. I would too but since I know it "can work" I'm determined |
| 02:42.37 | WiretapWork | lol |
| 02:42.38 | WIMPy | luckman212: You could try to set up connectedline information. |
| 02:42.38 | WiretapWork | oh dear |
| 02:42.47 | luckman212 | I can't find any concrete dialplan examples using the TRANSFER or ORIGINATE commands |
| 02:43.01 | BeeBuu | WIMPy: Oh,your mean i can calculate it out? |
| 02:43.25 | luckman212 | WIMPy: yes I've got CONNECTEDLINE working but that just sets the CID name/num it doesn't affect the call disposition (outgoing/incoming) |
| 02:43.28 | WIMPy | BeeBuu: You should be able to. |
| 02:43.56 | luckman212 | I can make the Polycom say "From: President Obama" but not "To: President Obama" |
| 02:44.06 | WIMPy | luckman212: Where? On the phone? |
| 02:44.31 | BeeBuu | WIMPy: oh, i just wonder that result is close the real? |
| 02:44.33 | WIMPy | Hmm. That sounds very special. |
| 02:44.34 | luckman212 | WIMPy: yes |
| 02:45.21 | luckman212 | WIMPy: do you know what the 'tag' datatype in CONNECTEDLINE does? |
| 02:45.30 | luckman212 | https://wiki.asterisk.org/wiki/display/AST/Function_CONNECTEDLINE |
| 02:45.48 | WIMPy | BeeBuu: You get more IRQs with multiple cards. Otherwise it sould be the usual scaling calculation. |
| 02:47.11 | BeeBuu | WIMPy: if my machine get enough IRQs to use,multiple cards can run? |
| 02:47.28 | WIMPy | luckman212: Looks exactely liek the internal structures. -valid is just a flag saying the other parameters with that name contain sensible data. |
| 02:47.54 | BeeBuu | WIMPy: one E1 port need 1 IRQ or more? |
| 02:47.59 | WIMPy | -charset should be obvious and -pre is the same as in CALLERID. |
| 02:48.11 | WIMPy | BeeBuu: One per crad, usually. |
| 02:48.16 | WIMPy | card |
| 02:48.22 | BeeBuu | thanks. |
| 02:48.52 | WIMPy | luckman212: Oh, you mean the "tag". Hmm. Let's see if I can find something. |
| 02:49.01 | luckman212 | WIMPy: yeah I meant the 'tag' |
| 02:49.05 | BeeBuu | WIMPy: As you know, what's the MAX cards had installed? |
| 02:50.58 | WIMPy | BeeBuu: As many as you have slots? I only tried the IRQ load on rather old hardware. There you want less cards than CPUs. But on modern hardware the number of slots will be the limit, I guess. |
| 02:51.28 | WIMPy | But how many ports do you really want? Wouldn't that be a case fo a channelized E3? |
| 02:51.31 | BeeBuu | My machine get 4 PIC slots |
| 02:51.57 | BeeBuu | and i want to use them all |
| 02:52.57 | BeeBuu | so 4X4 ,16 E1 ports, 16X30=480 channels in A machine, that's great for save money to me :D |
| 02:53.01 | WIMPy | 4x4? 4x8? I don;t have any idea what the IRQ situation is like with Samgoma cards. |
| 02:53.22 | WIMPy | BeeBuu: I think you rather want 1 E3. |
| 02:54.24 | BeeBuu | Oh,what's that? i don't know that? any URL? thanks. |
| 02:54.56 | WIMPy | A cahnnelized E3 containing 16 E1. |
| 02:55.11 | BeeBuu | it can use by asterisk? |
| 02:55.20 | WIMPy | Well, trechnically 4 E2 containing 4 E1. |
| 02:55.55 | WIMPy | Sangoma do a E3 card. But I don't knwo any specifics. Try to check with them. |
| 02:56.30 | WIMPy | But I doubt they do it for data only. |
| 02:57.13 | BeeBuu | WIMPy: pri can run thought E2 or E3? |
| 02:57.35 | WIMPy | No, it's always E1. |
| 02:58.02 | BeeBuu | so poor,i only get pri line... |
| 02:58.26 | WIMPy | But you can have 4 virtual E1s in 4 virtual E2, giving you 16 virtial E1 in one E3. |
| 02:58.54 | BeeBuu | and it can used by asterisk? |
| 02:59.03 | WIMPy | luckman212: I can't find any reference to that "tag" field. |
| 02:59.42 | WIMPy | BeeBuu: Check that with Sangoma. They do an E3 card. They will tell you what it can do. |
| 03:00.37 | WIMPy | luckman212: It doesn't seem to exist in the internal structure. Must be something "else". |
| 03:01.04 | WIMPy | has a trace. |
| 03:01.07 | BeeBuu | WIMPy: thanks |
| 03:01.44 | WIMPy | luckman212: Seems to be Asterisk internal and not transmitted anywhere. |
| 03:01.53 | WIMPy | At least if I got that right. |
| 03:02.46 | luckman212 | WIMPy: hmm.. ok. I am trying to find dialplan examples where a local extension is dialed and then a REFER command is issued to another SIP call that has been placed over a trunk |
| 03:03.49 | *** join/#asterisk galaxywatcher (~galaxywat@pdpc/supporter/active/galaxywatcher) |
| 03:05.08 | mattwj2002 | hi guys |
| 03:05.56 | mattwj2002 | anyone know how to change outgoing phone number with google voice? |
| 03:09.35 | WIMPy | How do I compare a variable against a string containgin a +? |
| 03:10.53 | WIMPy | It doesn't even work whe I assign that to a temporary variable. |
| 03:12.00 | *** join/#asterisk moy (~moy@69.157.46.221) |
| 03:15.45 | mattwj2002 | how do you change caller id in Asterisk 1.8? |
| 03:16.10 | kaldemar | mattwj2002: core show function CALLERID |
| 03:17.46 | mattwj2002 | so I must just set that variable |
| 03:17.47 | mattwj2002 | :) |
| 03:17.55 | kaldemar | WIMPy: what exactly does not work? how are you comparing it? |
| 03:18.22 | kaldemar | mattwj2002: CALLERID is not a variable, it's a function. write that into CLI. |
| 03:19.08 | mattwj2002 | I want to change my outgoing calls caller id |
| 03:19.09 | WIMPy | $[${var}=+123] - That complains about non-numeric argument. \+ or .+ didn;t work, either. |
| 03:19.41 | kaldemar | WIMPy: use some "'s around them. |
| 03:20.11 | ChannelZ | Set(CALLERID(num)=123456) |
| 03:20.21 | mattwj2002 | thank you |
| 03:20.37 | ChannelZ | whether or not it actually does anything is a different story |
| 03:21.11 | kaldemar | $["${VAR}" = "+123"] works fine. |
| 03:21.49 | *** join/#asterisk Peaceful (~Peaceful@63-235-131-194.dia.static.qwest.net) |
| 03:22.20 | WIMPy | Didn't try ", as that's usually a no go. But Quotes on both sides will indeed do. Thanks. |
| 03:22.45 | WIMPy | Need to be careful about that when using callerids or the like. |
| 03:23.02 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:23.39 | Peaceful | Does asterisk yet support sending/receiving SIP RTP traffic to/from a different IP address than the SIP signalling? |
| 03:24.05 | Peaceful | My new SIP provider (unexpectedly) sends signalling on one IP, but RTP by another. |
| 03:24.15 | mattwj2002 | I think google voice blocks it |
| 03:24.22 | mattwj2002 | thanks anyways |
| 03:24.24 | mattwj2002 | :) |
| 03:24.51 | WIMPy | Peaceful: If you have nat enabled it will send the traffic, where it received from unless you set strictrtp. |
| 03:24.52 | Peaceful | I've been googling and found many old posts that said asterisk didn't (at the time of the posts) support SIP via different signalling and rtp ip addresses. |
| 03:25.30 | Peaceful | WIMPy: Oh, so just pretend I'm using NAT? |
| 03:25.31 | WIMPy | But that will make you vulnerable agains attacks. |
| 03:25.43 | Peaceful | It's a private dedicated link to the provider |
| 03:25.50 | Peaceful | Not over the Internet. |
| 03:26.07 | *** join/#asterisk PwrSurge (~nrg@206-248-139-212.dsl.teksavvy.com) |
| 03:26.31 | BuenGenio | does Asterisk do video yet? |
| 03:26.33 | WIMPy | The question is, if Asterisk or rather the RTP ports can be reached from the internet as well. |
| 03:26.59 | Peaceful | WIMPy: No, asterisk is in a private network, not reachable from the Internet at all. |
| 03:27.16 | WIMPy | Then that might work. |
| 03:28.53 | kaldemar | BuenGenio: not conferencing, but single calls, yes. |
| 03:29.25 | BuenGenio | was actually looking at conferencing |
| 03:29.39 | BuenGenio | I spotted a plugin called app_konference, but haven't been able to compile it |
| 03:29.45 | WIMPy | has heard 1.10 will do it. |
| 03:29.49 | BuenGenio | there had been no changes since 2009 |
| 03:34.06 | Peaceful | WIMPy: nat=yes and nat=route don't seem to help. :-( |
| 03:35.12 | WIMPy | Peaceful: Maybe they aren't sending befor they receive anything? |
| 03:35.40 | WIMPy | Do you have multiple destinations? |
| 03:36.17 | Peaceful | multiple destinations? you mean like multiple entries on the host= line in the sip.conf entry? |
| 03:36.43 | WIMPy | More than one IP. |
| 03:37.09 | Peaceful | no, I don't! Can host= accept comma-separated IPs? |
| 03:38.08 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 03:38.17 | WIMPy | no |
| 03:39.04 | WIMPy | So you have exactely one IP for SIP and exactely one IP for RTP? |
| 03:39.11 | Peaceful | yes |
| 03:39.23 | WIMPy | Do some nasty shit with iptables. |
| 03:39.30 | Peaceful | ahhh |
| 03:39.33 | Peaceful | that could work |
| 03:40.00 | Peaceful | pretend that packets from seconds, actually come from first, and vice-versa on the way back out |
| 03:40.04 | WIMPy | ist pretty sure, it will. |
| 03:40.56 | Peaceful | hmmmm |
| 03:41.04 | Peaceful | that's going to take serious iptables-fu |
| 03:41.24 | WIMPy | Two lines? |
| 03:42.18 | WIMPy | if dstip=sipip and dport!=sip set dstip to rtpip. |
| 03:42.20 | Peaceful | You know the options? I've mostly just denied or accepted packets on the filter table |
| 03:42.37 | Peaceful | what targets do you use? |
| 03:42.42 | Peaceful | I'll read the rtfm |
| 03:42.48 | WIMPy | mangle |
| 03:42.48 | Peaceful | lol, that was redundant |
| 03:42.52 | Peaceful | ok |
| 03:43.07 | WIMPy | It might even be possible via the routing table, without iptables. |
| 03:43.36 | WIMPy | On the outgoing side. |
| 03:45.27 | Peaceful | routing doesn't usually change the IP destination, though, AFAIK, just the MAC addr destination. |
| 03:45.35 | Peaceful | is reading 'man iptables' |
| 03:45.48 | WIMPy | is too tired. That's a simple SNAT thing. |
| 03:46.14 | WIMPy | Oh, you can do a lot with iproute2. |
| 03:46.20 | Peaceful | heads over to the SNAT section... |
| 03:47.00 | WIMPy | Err. DNAT. |
| 03:47.00 | Peaceful | SNAT one way, DNAT the other? |
| 03:47.17 | Peaceful | er, no, I don't want to change mine ever, eh? |
| 03:47.38 | WIMPy | If it is symmetic RTP only DNAT should be enough. |
| 03:48.07 | WIMPy | No, but you might have to change source address on incoming RTP. |
| 03:51.34 | Peaceful | like this? iptables -t nat -d 10.2.2.10 -I PREROUTING -i eth1-p udp -m udp -j DNAT --to-destination 10.2.2.11 <- for outbound rtp? |
| 03:51.44 | Peaceful | 10.2.2.10 is the signalling |
| 03:51.54 | Peaceful | 10.2.2.11 is the rtp |
| 03:52.45 | WIMPy | OUTPUT. PREROUTING will only catch packets received from an interface. |
| 03:53.16 | WIMPy | And you need to check for dport!=5060. |
| 03:54.19 | Peaceful | iptables -t nat -d 10.2.2.10 -I OUTPUT -i eth1 -p udp -m udp -j --dport != 5060 DNAT --to-destination 10.2.2.11 |
| 03:54.42 | Peaceful | is "!=" actually usable in iptables? I've never done a 'not' in iptables, not sure of the syntax |
| 03:55.14 | WIMPy | -o eth1. it is oging out. |
| 03:55.29 | Peaceful | doh <facepalm> |
| 03:55.36 | WIMPy | And the -j needs to be after all conditions. |
| 03:55.50 | Peaceful | iptables -t nat -d 10.2.2.10 -I OUTPUT -o eth1 -p udp -m udp --dport != 5060 -j DNAT --to-destination 10.2.2.11 |
| 03:55.59 | Peaceful | sheesh, I'm tired. How's that? |
| 03:56.54 | Peaceful | "not" syntax is wrong |
| 03:56.58 | Peaceful | no '=' i think |
| 03:57.07 | Peaceful | iptables -t nat -d 10.2.2.10 -I OUTPUT -o eth1 -p udp -m udp --dport ! 5060 -j DNAT --to-destination 10.2.2.11 |
| 03:57.26 | WIMPy | Yes. Had to look that up as well. it is ! -- dport |
| 03:57.46 | Peaceful | oh, "! --dport 5060"? |
| 03:57.53 | WIMPy | jepp |
| 03:58.16 | Peaceful | iptables -t nat -d 10.2.2.10 -I OUTPUT -o eth1 -p udp -m udp ! --dport 5060 -j DNAT --to-destination 10.2.2.11 |
| 03:58.36 | WIMPy | I think that's it. |
| 03:59.01 | Peaceful | okay, now with my newfound iptables-fu-awesomeness, I will create a SNAT rule that will be syntactically correct the first time! Maybe. |
| 03:59.16 | WIMPy | :-) |
| 03:59.28 | WIMPy | I'm already falling asleep. |
| 03:59.50 | WIMPy | Need to go somewhere where I get a soft landing. |
| 04:00.02 | WIMPy | Good luck! |
| 04:00.28 | Peaceful | iptables -t nat -s 10.2.2.11 -I PREROUTING -i eth1 -p udp -m udp -j SNAT --to-source 10.2.2.10 |
| 04:00.41 | Peaceful | wait! before your soft landing, you must see the awesomeness! |
| 04:01.10 | Peaceful | I'm tired too. this is 13.5 hours at work so far today. |
| 04:04.29 | Peaceful | Oops, had the dest/src before the chain. |
| 04:04.34 | Peaceful | Think maybe this will work... |
| 04:04.49 | Peaceful | iptables -t nat -I PREROUTING -s 10.2.2.11 -i eth1 -p udp -m udp -j SNAT --to-source 10.2.2.10 for IN |
| 04:04.58 | Peaceful | iptables -t nat -I OUTPUT -d 10.2.2.10 -o eth1 -p udp -m udp ! --dport 5060 -j DNAT --to-destination 10.2.2.11 for OUT |
| 04:06.34 | *** join/#asterisk cyborg-one (1000@188-115-189-233.broadband.tenet.odessa.UA) |
| 04:12.37 | Peaceful | nope |
| 04:12.48 | Peaceful | no worky :( No audio either way, still. |
| 04:13.19 | cutlass | I'm using ARA and in troubleshooting an issue, I found that I can completely blow away the "fullcontact" value in the sip users table for a user but somehow I am still able to route calls to it...that made me wonder, exactly what is that value being used for? Does anyone know? |
| 04:14.16 | ChannelZ | is your FORWARD table open? |
| 04:14.57 | ChannelZ | chain rather |
| 04:36.39 | kaldemar | cutlass: Contact header of a registration message is saved to the fullcontact field. |
| 04:44.21 | ChannelZ | ARGH f@#ing Secured Servers |
| 04:44.36 | ChannelZ | It's unbelievable how much spam I get from IPs owned by these fools |
| 04:45.59 | carrar | Anyone tried CentOS 6.0 x86_64 with 1TE420BF cards? |
| 04:46.09 | carrar | pci-x card |
| 04:47.54 | carrar | good times |
| 04:49.25 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-nzxzxdbewkezgcuq) |
| 04:51.12 | WiretapWork | ChannelZ: just /drop them? |
| 05:04.30 | *** join/#asterisk lucasb (~lucasb@S0106000c42710923.ok.shawcable.net) |
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| 05:44.14 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 05:53.54 | ChannelZ | yes I have a bunch of their netblocks blocked |
| 05:58.40 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:09.24 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:11.02 | *** join/#asterisk cerberus_za (~coert@41.134.110.162) |
| 06:11.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 06:17.19 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:17.23 | schmidts | good morning |
| 06:18.17 | ChannelZ | Ahoyhoy. |
| 06:21.09 | *** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:21.19 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:21.45 | drmessano | Ah, that's better |
| 06:24.11 | ChannelZ | slipped into something more comfortable? |
| 06:24.29 | schmidts | http://www.bonkersworld.net/wp-content/uploads/2011/06/2011.06.27_organizational_charts.png near the truth :D |
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| 06:39.12 | irroot | lo there folks |
| 06:44.45 | ChannelZ | ahoy |
| 06:45.47 | irroot | at home today so may get som stuff done :P |
| 06:46.48 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 06:48.06 | ChannelZ | I wish I could take the day off. Not motivated to work |
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| 06:57.36 | irroot | im doing a experiment to determine the volume of mucus required to drown one ... |
| 06:59.37 | *** join/#asterisk drdru (~Adium@76.77.182.195) |
| 06:59.57 | drdru | hola |
| 07:02.54 | kaii | i totally agree to ChannelZ |
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| 07:29.50 | Bipul | p3nguin, ping ? there |
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| 08:06.05 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:06.07 | jacc0 | hi all |
| 08:06.18 | sawgood | hi |
| 08:06.30 | jacc0 | the first results of my automated debug info gathering script are here |
| 08:07.35 | jacc0 | asterisk deadlock |
| 08:07.36 | jacc0 | http://pastebin.com/WwrfRUx7 |
| 08:07.53 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:08.50 | jacc0 | so asterisk is still running ; netstat shows it's listening on the sip port but asterisk isn't responding to any sip packeges |
| 08:09.35 | jacc0 | I also have an wireshark trace of the past 30 minites before the deadlock |
| 08:09.37 | sawgood | why not stop and restart Asterisk? |
| 08:09.51 | jacc0 | it also does that |
| 08:10.16 | jacc0 | But I want the deadlocks , segfaults and other crashes fixed |
| 08:10.33 | jacc0 | they arfe making me sick |
| 08:10.42 | jacc0 | I mean realy sick |
| 08:10.48 | jacc0 | I'm not getting any sleep |
| 08:11.04 | jacc0 | with 20+ production env. choking all the time |
| 08:11.41 | jacc0 | this is my last option before abondoning asterisk |
| 08:11.56 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 08:12.00 | jacc0 | I'm providing as much debug info as i can |
| 08:30.23 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 08:30.31 | jacc0 | jira report created |
| 08:30.44 | jacc0 | uploading wireshark trace |
| 08:31.02 | jacc0 | I clasified it as a blocker |
| 08:31.19 | *** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7) |
| 08:31.58 | jacc0 | https://issues.asterisk.org/jira/browse/ASTERISK-18166 |
| 08:32.55 | *** join/#asterisk fyrfaktry (~dread@adsl-070-145-120-039.sip.pns.bellsouth.net) |
| 08:34.26 | kaii | jacc0: did you compile asterisk with debugging options? an output of "core show locks" would definitely help |
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| 09:02.53 | *** join/#asterisk cervajs2 (~cervajs@178.148.broadband4.iol.cz) |
| 09:03.54 | cervajs2 | hello, someone with asterisk rpm building skills? |
| 09:06.18 | jacc0 | thanks kaii: will add that to the debug gathering script |
| 09:07.01 | kaii | jacc0: support for this command must be compiled in at build time. |
| 09:07.09 | jacc0 | it's in there |
| 09:07.32 | jacc0 | I've set all the debugging flags in the compiler optionis of menuselect |
| 09:07.33 | kaii | k |
| 09:07.53 | kaii | then core show threads is also relevant, maybe |
| 09:08.10 | jacc0 | ty ty |
| 09:08.28 | kaii | also, "bt full" may give more information than just a "bt" .. but the output is huge.. |
| 09:09.01 | jacc0 | I'm also adding that |
| 09:09.37 | jacc0 | I'll be releasing the debug nifo gathering script later one |
| 09:09.41 | jacc0 | *info |
| 09:10.07 | jacc0 | maybe it could be added as a menuselect option |
| 09:10.43 | jacc0 | gdb -p `pidof asterisk` -ex 'thread apply all bt full' -ex detach -ex quit |
| 09:10.46 | jacc0 | will that do? |
| 09:11.47 | kaii | yes |
| 09:13.47 | aiksa[LV] | anyone here using git for maintaining configuration for a number of pbxes? |
| 09:15.33 | kaii | jacc0: hmm .. forget the bt full thing .. i guess its more information than necessary |
| 09:15.49 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 09:21.55 | *** join/#asterisk sidh (~tinom@intellitec2.net) |
| 09:22.07 | sidh | Greetigns ladies and Gentlemen |
| 09:23.07 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 09:23.36 | sidh | I would need your advices, I would like to set a SOHO (small office/home office) AsterisK platform, on a soekris net4801 (FreeBSD OS based), do you think 128 MB ram is enough ? |
| 09:24.57 | sidh | (there would be max 3-4 phones, with max 2-3 concurent call) |
| 09:25.37 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 09:34.07 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 09:37.36 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 09:42.11 | irroot | sidh should be ok but more is better :P |
| 09:42.36 | irroot | get more ram if you can |
| 09:42.36 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 09:47.23 | irroot | ping jacc0 |
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| 09:56.44 | jacc0 | pong irroot |
| 09:57.09 | irroot | some feedback on originate timeout |
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| 10:27.17 | Bipul | humm.. there is few issue i can here in coming calls but the caller cant hear my voice when i am calling it through my asterisk |
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| 10:37.31 | jacc0 | @Bipul:look at your NAT settings in sip.conf |
| 10:38.19 | Bipul | ok |
| 10:38.54 | Bipul | ;nat=yes |
| 10:39.29 | Bipul | nat:yes <-- it shud be ? is that ';' a problem ! |
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| 10:50.34 | *** join/#asterisk cbf (~c1@unaffiliated/hrhrhr) |
| 10:50.41 | cbf | hi guys |
| 10:50.43 | kaldemar | Bipul: ";" is a comment character in asterisk configuration files. |
| 10:50.50 | cbf | does asterisk 1.8 support sip_simple ? |
| 10:50.54 | cbf | for IM presence |
| 10:50.54 | Bipul | yes |
| 10:51.01 | Bipul | i think i shud remove it |
| 10:56.51 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:57.09 | cbf | maybe i should re-phrase |
| 10:57.38 | cbf | we use bria and the instant message feature does not work. do any of the current asterisk versions support this feature natively, if not, is it on the roadmap? |
| 10:58.02 | cbf | counterpath tell me it is the SIP_SIMPLE feature |
| 10:58.51 | *** join/#asterisk coppice (~coppice@m121-202-25-194.smartone-vodafone.com) |
| 10:59.23 | kaldemar | cbf: asterisk does not support MESSAGE outside a call. |
| 11:00.16 | cbf | ok, thx |
| 11:01.02 | irroot | kaldemar asterisk 10 will :P cbf |
| 11:01.45 | *** part/#asterisk cbf (~c1@unaffiliated/hrhrhr) |
| 11:03.44 | kaldemar | irroot: well, it'll take some time before it arrives in a stable form. :) |
| 11:04.08 | irroot | rummor has it that the same applies to 8 |
| 11:04.37 | irroot | betas will be out soon ... |
| 11:21.34 | puzzled | hi |
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| 11:28.47 | sidh | irroot: I found the net 4801 version with 256 MB ram, I go to ask the vendor if it is still available |
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| 12:27.27 | *** join/#asterisk DocAwesome (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:27.27 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 12:30.04 | *** join/#asterisk echinos (~echinos@67.196.136.211) |
| 12:31.10 | echinos | anyone else installed asterisk 1.8 from the repos on debian squeeze? when I try apt-get install asterisk-1.8, I get "selecting 'asterisk' instead of 'asterisk-1.8' |
| 12:33.00 | DocAwesome | pabelanger: ^^^^ |
| 12:35.04 | jacc0 | @echinos: yes |
| 12:35.32 | echinos | any ideas? |
| 12:36.04 | echinos | the plain asterisk package is 1.4 |
| 12:36.07 | echinos | :( |
| 12:38.05 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-eavyxkomtyascdfu) |
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| 12:38.30 | jacc0 | add sources ? |
| 12:39.31 | jacc0 | http://pastebin.com/Ghka1AwL |
| 12:39.34 | jacc0 | like that |
| 12:40.26 | jacc0 | make sure you remove older versions before "apt-get install asetrisk-1.8" |
| 12:40.38 | jacc0 | even better: do a clean install of debian |
| 12:41.14 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 12:41.18 | echinos | I'd like to, but it's handling other stuff |
| 12:41.26 | echinos | thx |
| 12:42.46 | jacc0 | good luck |
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| 12:59.31 | azv4 | any KX-TF500 pro's available for a couple quick questions? |
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| 13:03.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:22.02 | tuxxie | what is the best method in for tracking call quality issues on three way calls? |
| 13:23.24 | *** part/#asterisk kumbang (~tux@180.245.137.5) |
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| 13:30.02 | psilikon | So let's say I have 1000 calls to originate using AMI but I only have 100 SIP channels available and I want to burn through my list of 1000 programmatically. Do I need to check for available channels before I attempt a call? |
| 13:30.42 | WIMPy | Or you just count how many you set up. |
| 13:31.09 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 13:31.38 | DocAwesome | psilikon: seems reasonable unless you start using party lines |
| 13:32.52 | psilikon | What if I just let asterisk sort out the details and I just keep trying to dial without keeping track of available channels? Will asterisk magically figure it out for me? |
| 13:33.51 | WIMPy | Your provider will. |
| 13:35.04 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 13:36.35 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:36.35 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:37.11 | robl^laptop | does anyone know of a good tool for automated testing of an asterisk server? Basically, at a given interval, check that a SIP trunk is registered, send a test call to each (SIP, DAHDI, etc) trunk.. and send a test call back to the asterisk server.. if anything fails, send an email alert to an admin? Automate test calls in and out of asterisk? |
| 13:38.05 | psilikon | WIMPy, right so I'll get congestion but then I would need some logic that attempts the calls again right? Which ami originate can do with the 'retry' option. |
| 13:38.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:39.15 | WIMPy | I never noticed it has a 'retry'. But that might work. |
| 13:39.44 | WIMPy | Don't you want to know the results of your calls otherwise? |
| 13:40.07 | echinos | jacc0: had to apt-get remove --purge the old version |
| 13:40.21 | jacc0 | ;) |
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| 14:00.15 | eduzimrs | im running ./configure into asterisk1.8 folder and i got this erro: "configure: error: in `/usr/src/asterisk-1.8.5.0':" ideas ? |
| 14:01.29 | eduzimrs | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 14:01.40 | DocAwesome | you're missing cpp |
| 14:02.04 | irroot | do you have a dev toolchain ?? |
| 14:02.09 | eduzimrs | but the file is there |
| 14:02.21 | DocAwesome | apt-get install build-essential |
| 14:02.29 | eduzimrs | ok |
| 14:02.46 | eduzimrs | im running CentOS x86_64 |
| 14:02.49 | Naikrovek | anyone know what formats (other than 8000hz ulaw) polycoms can play as ringtones, if any? |
| 14:03.46 | DocAwesome | Naikrovek: I doubt anything else is supported |
| 14:03.55 | eduzimrs | @DocAwesome humm /lib/cpp is missing really |
| 14:03.57 | DocAwesome | Naikrovek: although the administration manual should list what you can use |
| 14:04.12 | DocAwesome | eduzimrs: check the Asterisk book for a list of dependencies |
| 14:04.17 | Naikrovek | I read somewhere else that others were supported, but now I can't remember what I was reading or what it said. Yah, good idea, Dr. |
| 14:04.31 | DocAwesome | Naikrovek: my guess is some wav formats are also supported |
| 14:04.43 | DocAwesome | I looked actually at those a bit the other day now that I think about it |
| 14:04.50 | DocAwesome | so not just ulaw is supported at least :) |
| 14:04.55 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 14:05.03 | Naikrovek | parks & recreation theme doesn't sound as good as I'd like at 8000hz. |
| 14:05.10 | DocAwesome | ah gotcha |
| 14:05.15 | Naikrovek | pranking a coworker |
| 14:05.16 | DocAwesome | not sure if it'll do 16k or not... would be nice |
| 14:05.19 | Naikrovek | yes |
| 14:05.22 | DocAwesome | Naikrovek: sounds like a great use of time |
| 14:05.22 | Naikrovek | i'm sure some will |
| 14:05.27 | Naikrovek | lol :) day off |
| 14:05.29 | DocAwesome | nice |
| 14:05.36 | DocAwesome | I approve of this prank |
| 14:05.42 | Naikrovek | yeah, but even if it wasn't, I'd still do it. |
| 14:05.47 | DocAwesome | :) |
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| 14:14.59 | JonathanRose | [ls |
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| 14:19.07 | DocAwesome | JonathanRose: keep it in your terminal |
| 14:19.41 | JonathanRose | Crap, I've been found out. |
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| 14:28.49 | eduzimrs | @DocAwesome i did, the following dependencies were missing > newt-devel ncurses-devel libtermcap-devel libxml2-devel < now it works! Tks |
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| 14:39.49 | WIMPy | Is there any configuration option to suppress the warnings about the use of _! or _. ? |
| 14:43.48 | anonymouz666 | irroot: where I can download the T.38 gw patch for 1.8.5.0? |
| 14:43.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:44.13 | irroot | anonymouz666 there is a patch on JIRA |
| 14:44.48 | anonymouz666 | this is patch is the same that is present on 1.10? |
| 14:45.50 | irroot | yes or better get it from svn team/irroot/t38gateway-1.8 |
| 14:47.59 | irroot | its missing the v17 detection in dsp.c |
| 14:48.16 | irroot | im planing on backporting it soon |
| 14:48.29 | anonymouz666 | this in version 1.8.5.0? |
| 14:48.49 | anonymouz666 | you are backporting the v17 dectection from 1.10 to 1.8.5.0? |
| 14:50.21 | *** join/#asterisk receptor (~Miranda@188-223-90-113.zone14.bethere.co.uk) |
| 14:50.41 | irroot | yes |
| 14:51.03 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:51.11 | WIMPy | Does anyone have any experience using Incomplete()? It seems to be synonymous to Goto(i), which is not what I had hoped for (and doesn't seem to make too much sense). |
| 14:51.34 | Bipul | p3nguin, ping |
| 14:51.38 | anonymouz666 | irroot: can't find the ID, there are two many issues if you search for T.38 gw |
| 14:52.05 | receptor | hi, is there a way to launch a test call in asterisk without a phone? i have a sangoma A108e card and i connected it as loop. log says E1 connected but i need to test if it's actually making calls |
| 14:53.14 | irroot | ASTERISK-12667 |
| 14:53.36 | irroot | receptor "originate" |
| 14:53.52 | WIMPy | receptor: channel originate |
| 14:54.13 | irroot | https://issues.asterisk.org/jira/secure/attachment/40623/faxgateway-1.8.patch <- anonymouz666 |
| 14:54.23 | receptor | ok thanks, will try that |
| 14:55.38 | anonymouz666 | irroot: what's the framehook patch? |
| 14:56.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:56.08 | irroot | that is pre 1.8.5 there is a bug in older asterisk |
| 14:56.57 | anonymouz666 | alright |
| 14:57.07 | anonymouz666 | going to install 1.8.5.0 with this patch |
| 14:57.10 | anonymouz666 | thanks |
| 14:57.34 | irroot | buzz me if you want |
| 15:12.34 | *** join/#asterisk cutlass (~cutlass@c-67-165-132-67.hsd1.il.comcast.net) |
| 15:12.35 | *** join/#asterisk hfb (~hfb@pool-96-247-53-174.lsanca.dsl-w.verizon.net) |
| 15:12.38 | cutlass | does anyone know how to view the current SIP registry that asterisk uses to route calls? the "sip show peer" command does not see like the answer because when you perform "sip reload" that data is wiped out. Also, the /SIP/registration family in AstDB can be deleted and Asterisk is still able to route calls...any ideas? |
| 15:14.15 | cutlass | the reason I'm asking is because I've observed some corruption in my DB and I'm trying to understand how/if it might affect call delivery for SIP endpoints |
| 15:24.14 | *** join/#asterisk luminforce (~luminforc@rrcs-71-42-183-54.sw.biz.rr.com) |
| 15:31.53 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 15:33.39 | jacc0 | @cutlass: asterisk realtime ? |
| 15:34.05 | jacc0 | otherwise try sip show users |
| 15:34.21 | jacc0 | I'm leaving: weekend!! |
| 15:34.24 | jacc0 | :) |
| 15:34.30 | jacc0 | havea good one all!!! |
| 15:35.05 | carrar | It's only Friday morning! |
| 15:35.20 | carrar | Weekend is so far away |
| 15:35.45 | irroot | 17:35 ... carrar its on like donkey kong |
| 15:35.51 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:35.55 | Katty | gmorning |
| 15:35.56 | carrar | Fri Jul 22 08:35:56 PDT 2011 |
| 15:36.02 | carrar | Morning KattyWoooo |
| 15:36.34 | irroot | Fri Jul 22 17:36:27 SAST 2011 |
| 15:36.46 | luckyaba | I am with ya carrar |
| 15:36.49 | Katty | hugs carrar |
| 15:36.52 | luckyaba | just got to work |
| 15:37.00 | Katty | i haz some killer sinus pressure tday |
| 15:37.08 | carrar | Had to look that one up, SAST = South African Standard Time |
| 15:37.25 | irroot | katty lo there i was home today same reason |
| 15:37.33 | carrar | Katty, just blow it all out! |
| 15:37.37 | carrar | Wilderness style |
| 15:37.38 | Katty | can't |
| 15:37.39 | irroot | carrar lol was wondering who would get it |
| 15:37.40 | Katty | nose is fine |
| 15:37.46 | Katty | it's above my eyebrows |
| 15:37.51 | Katty | and the back of my head |
| 15:37.53 | Katty | also the side of my temples |
| 15:37.59 | Katty | err right temple, at least |
| 15:38.23 | carrar | Sounds like a excuse to use POT!! |
| 15:38.27 | carrar | :) |
| 15:38.30 | Katty | i think not. |
| 15:38.34 | carrar | haha |
| 15:38.41 | irroot | mmm good plan indeed |
| 15:38.49 | Katty | SO |
| 15:38.51 | irroot | have got some fine weed in stock |
| 15:38.54 | Katty | been watching dr who. |
| 15:39.01 | Katty | lots of dr who infact. 5 seasons worth |
| 15:39.07 | carrar | woah |
| 15:39.12 | carrar | should go outside or something |
| 15:39.14 | Katty | yes. it's been lovely |
| 15:39.17 | Katty | oh i have |
| 15:39.19 | carrar | TV is bad mmmkay |
| 15:39.22 | Katty | work's been ....slow |
| 15:39.55 | Katty | dr who helps pass the time. |
| 15:39.59 | Katty | but this last doctor... |
| 15:40.05 | Katty | from season 5. just don't know about him. |
| 15:40.23 | Katty | he seems to lack the ENTHUSIASM |
| 15:40.31 | Katty | and RECKLESS ABANDON that the previous doctors had. |
| 15:40.42 | Katty | he's not quite Mad enough. |
| 15:40.45 | irroot | http://twicsy.com/i/7NDNv http://twicsy.com/i/wy9Mv katty carrar |
| 15:41.00 | irroot | pix taken from gf phone |
| 15:41.04 | Katty | what is that. |
| 15:41.05 | Qwell | Katty: I hated him too. The previous doctor was awesome. He grows on you though. |
| 15:41.07 | *** join/#asterisk m_tadeu (~quassel@89-180-230-215.net.novis.pt) |
| 15:41.29 | Qwell | I think they realized that like mid-season, and now he's pretty mad. |
| 15:41.30 | carrar | irroot, yeah I'm not really into drugs, just like to make fun of them |
| 15:41.43 | Qwell | still missing "something" though.. |
| 15:42.14 | Katty | i don't know Qwell |
| 15:42.18 | Katty | i just finished season 5. |
| 15:42.25 | Katty | still not very impressed. |
| 15:42.38 | Katty | tho, i will admit, it IS better |
| 15:42.49 | Qwell | I'm not saying he's better than the previous doctor, by any means. |
| 15:42.54 | Katty | still not enough mad, bouncy, excited, running around |
| 15:43.05 | Katty | waving sonic screwdrivers at things. |
| 15:43.13 | Katty | head smacking |
| 15:43.22 | Katty | and certainly not enough random AH HAS! |
| 15:43.35 | Katty | (david tennant)++ |
| 15:43.50 | Katty | someone needs to kill the doctor so we can get a new actor. |
| 15:43.57 | Qwell | yesplease |
| 15:44.13 | Katty | best moment ever. |
| 15:44.17 | Katty | the dr and martha jones. |
| 15:44.19 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 15:44.24 | Katty | meeting the face of bo |
| 15:44.32 | KavanS | irroot, I just checked out your gf's rack pics fyi |
| 15:44.42 | Katty | SO |
| 15:44.44 | Katty | Qwell: torchwood? |
| 15:44.47 | KavanS | cheers! |
| 15:44.49 | Katty | Qwell: i've been recommended. |
| 15:44.51 | Qwell | Katty: meh |
| 15:44.51 | irroot | she has the goods :P |
| 15:44.53 | Katty | Qwell: should i watch it? |
| 15:45.03 | KavanS | indeed, lucky man! |
| 15:45.03 | Qwell | Katty: file would say yes. I say meh. |
| 15:45.23 | file | Qwell is just lame |
| 15:45.32 | Katty | where does torchwood fit into the time line of thigns |
| 15:45.40 | Qwell | present day |
| 15:45.41 | Katty | bubble of timey wimey |
| 15:45.51 | Katty | oh well that's no fun |
| 15:45.51 | Qwell | well, the show is present, anyways |
| 15:46.20 | echinos | More aliens, less wibly-wobbly timey-wimey stuff |
| 15:46.22 | Katty | any crossovers? |
| 15:46.24 | jaytee | I get tired of all the ginger bashing on Dr Who |
| 15:46.28 | Qwell | Katty: only 1 |
| 15:46.30 | Qwell | well, 2 |
| 15:46.37 | Katty | then it's worth watching! |
| 15:46.41 | echinos | River is supposed to be in it IIRC |
| 15:46.43 | Qwell | but it's only captain jack :p |
| 15:46.44 | Katty | ,...if it's david tennant |
| 15:46.58 | Katty | but captain jack is the face of BO! |
| 15:46.59 | *** join/#asterisk jsemar (~joel@office.appiction.com) |
| 15:47.04 | echinos | SPOILERS! ;) |
| 15:47.12 | Katty | oh it's not a spoiler |
| 15:47.15 | Katty | it happened in season...3 |
| 15:47.18 | Katty | no, wait, 4? |
| 15:47.29 | Katty | 3! |
| 15:47.48 | Katty | i had a case of the monty pythons. |
| 15:48.07 | jaytee | "it's only a model" |
| 15:48.07 | echinos | No Yelloooooow........ |
| 15:48.13 | Katty | Qwell: season 5 of just seemed...out of place |
| 15:48.32 | Katty | Qwell: kind of this floundering about |
| 15:48.45 | Katty | Qwell: no proper storyline |
| 15:50.21 | Katty | and now my car gets to see the dr. |
| 15:50.29 | Katty | oil light keeps coming on )= am worried. |
| 15:54.59 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:58.07 | chuckf | I'm not happy with season 4 of torchwood. |
| 15:58.16 | dr0ck | because its aweful? |
| 15:58.20 | dr0ck | yeah, same |
| 16:00.12 | chuckf | I like the basic story idea, but the directing and acting are just sad |
| 16:10.07 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 16:12.19 | *** join/#asterisk kfife (~Miranda@kfife.com) |
| 16:12.26 | kfife | dogs with cat dander? |
| 16:12.36 | wasanzy | hi |
| 16:12.38 | kfife | My pet fish have dinosaur dander :-) |
| 16:12.53 | kfife | DoIP - Douchbaggery over IP |
| 16:13.59 | wasanzy | thank every one that assisted me got my asterisk working in cludling Kaldemar, and eraphie |
| 16:17.15 | wasanzy | now that, people within the same network can now talk to each other using twinkle, is time for me to step forward allowing others in a different network to be able to do the same, I tried using the public IP in a different network to connect to the astrisk server but it tells me 503 service unavailable, how can I resolve it. |
| 16:17.33 | wasanzy | am not a networking guy that is why I had to ask |
| 16:20.44 | ssureshot | wasanzy: I'm not familiar with the old issue but, are you inside the same internal networking scheme? public external ip? |
| 16:22.41 | ssureshot | iee,, whats the source network and whats the destination network |
| 16:23.07 | ssureshot | you'll need to make sure the routing tables are setup |
| 16:29.54 | *** join/#asterisk Elit3 (~Elit3@41.35.239.89) |
| 16:30.48 | *** part/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 16:32.00 | *** join/#asterisk DoDaT69 (~DODAT68_1@107.0.56.212) |
| 16:33.07 | *** join/#asterisk nny1 (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 16:34.55 | nny1 | have a system that is dying in a special way. It just "stops responding", I can hit console I can do sip show peer, etc. But no calls hit console etc. A restart clears it up, any advice? |
| 16:35.07 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:35.14 | nny1 | it's not a segfault, but it's not in a working state |
| 16:35.54 | nny1 | nm |
| 16:36.00 | nny1 | found a telling situation int he logs |
| 16:41.18 | DoDaT69 | what did you see? |
| 16:44.13 | *** join/#asterisk radic (~radic@tmo-096-37.customers.d1-online.com) |
| 16:46.13 | pabelanger | ~deadlock |
| 16:46.21 | pabelanger | ~backtrace |
| 16:46.21 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 16:46.39 | pabelanger | nny1: ^ get a backtrace, sounds like a deadlock |
| 16:50.47 | *** join/#asterisk mtbf (~in9836@89-76-9-252.dynamic.chello.pl) |
| 16:51.33 | mtbf | Hi there, quick question: can i use digiums speech recognition engine to dynamically transcript speech to text |
| 16:51.48 | mtbf | and what's the average accuracy for US-english? |
| 16:52.54 | *** join/#asterisk tzafrir (~tzafrir@46-116-25-26.bb.netvision.net.il) |
| 16:53.23 | nny1 | pabelanger: ahh it's just a transcodign issue, log is full of res_rtp_asterisk.c: Oh dear... we couldn't allocate a port for RTP instance '0x203ecb78' [Jul 22 11:46:45] ERROR[14380] codec_sangoma.c: Failed to create Sangoma transcoding RTP stream |
| 16:53.44 | Qwell | 1) Digium doesn't have a speech recognition engine (you mean Lumenvox, maybe?) 2) No, it is recognition, and not transcription. |
| 16:53.45 | nny1 | pabelanger: unfortunately the client removed the transcoder card before it could be diagnosed, so that's the end of that |
| 16:55.01 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:55.07 | mtbf | Qwell: right, i mean Lumenvox, and thanks for the reply. |
| 17:10.17 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 17:12.01 | Hive | I'm trying to have a notification sent to my email if one of my trunks fails 4 times in a row. I have the logic all set up for when the trunk fails... Increment a counter and dial on another trunk, send email when counter = 4. |
| 17:12.19 | *** part/#asterisk nny1 (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 17:12.26 | Hive | But I'm want to set the counter = 0 on a successful dial on the specific trunk |
| 17:12.56 | Hive | Anyone have any ideas? I can post my current dial plan, but I dont want to be spamer McSpammy :-P |
| 17:14.21 | Bipul | p3nguin, ? |
| 17:16.07 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 17:17.43 | jeffspeff | so i just found out that my sip termination provider (aka sip trunk) only supports inbound T.38 fax support... which means I can receive faxes, but not send them. Does anybody have suggestions on maybe another termination provider or a 3rd party source that could handle this for me? |
| 17:19.28 | hardwire | efax? |
| 17:22.04 | Hive | jeffspeff, does it support G.711? |
| 17:23.47 | jeffspeff | Hive, yes, but from everything that I've read fax over g.711 is not very reliable, especially on a high traffic network... I'm doing this for an engineering company that has 3 MS active directories, 4 redundant data replication servers and 50+ plus people doing autocad and listening to pandora... not exactly low traffic, lol |
| 17:24.05 | Naikrovek | um |
| 17:24.18 | Naikrovek | g711 (ulaw) is the MOST reliable way to transmit fax |
| 17:24.23 | Naikrovek | ulaw/alaw |
| 17:25.23 | Qwell | no it isnt |
| 17:25.26 | Naikrovek | well maybe not the most, but it's definitely good enough |
| 17:25.51 | Hive | It's worth a try, I just "finished" setting up the fax system for my employer and g.711 seemed to be more reliable for me |
| 17:26.11 | Naikrovek | g711 is better than any codec that is optimized for voice |
| 17:26.19 | Naikrovek | such as g723 speex, gsm, or g729. |
| 17:26.23 | Hive | Fax over voip seems to not be very reliable in general :-P |
| 17:27.11 | jeffspeff | Qwell, any input? From what I've read T.38 was designed for fax, G.711 is designe for voice and ---> The configured voice codec is used for the fax transmission. This technique works only when the configured codec is G.711 with no voice activity detection (VAD) and no echo cancellation (EC), |
| 17:27.27 | jeffspeff | we will be using echo cancellation |
| 17:27.56 | Hive | Hmm |
| 17:29.11 | jeffspeff | Naikrovek, Hive, take a look at the 3rd question on the following page ---> http://www.digium.com/en/docs/FAX/fax_faq.php |
| 17:29.38 | Naikrovek | fine. |
| 17:29.49 | Naikrovek | you were speaking codecs, not .. nevermind |
| 17:29.54 | jeffspeff | I know I asked for everybodies input, but what you all are telling contradicts everything that I've read, and what Digium says themselves. I honestly only know what I read and what I'm told |
| 17:29.55 | Naikrovek | you fucking win |
| 17:30.21 | Hive | lol |
| 17:30.28 | jeffspeff | wow |
| 17:31.15 | Hive | I'm no expert in the voip field, but i have been working with faxes for the past 2 weeks and i dont know if "reliability can be guaranteed" for any method hehe |
| 17:31.45 | Naikrovek | if you want to get technical, there's no such thing as 100% reliability |
| 17:31.49 | Naikrovek | everything fails at some point |
| 17:31.50 | Hive | jeffspeff, probably best to listen to what digium says over me though ;-) |
| 17:31.51 | atheos | reliability cannot be guaranteed between two analog fax machines, let alone adding VOIP to the mix |
| 17:32.00 | Qwell | Naikrovek: everything, 100% reliably. |
| 17:32.20 | Naikrovek | t.38 is a protocol that layers OVER the trunk codec, i was suggesting G711 (or better) for the trunk codec |
| 17:32.20 | nightrid3r | only death is 100% reliable |
| 17:32.28 | Hive | ^ |
| 17:32.44 | Naikrovek | death is certain, death is not a piece of equipment that is reliable |
| 17:33.04 | Naikrovek | lobsters don't die of natural age, btw so even that has its uncertainty |
| 17:33.47 | Naikrovek | anyway |
| 17:33.55 | Naikrovek | t38 is very unreliable in my experience |
| 17:34.08 | Naikrovek | t38 over g729 won't work |
| 17:34.21 | Naikrovek | t38 over g711 or g722 is as good as it's going to get |
| 17:34.23 | jeffspeff | Naikrovek, I didn't know that T.38 worked over G.711... |
| 17:34.51 | Naikrovek | if you have a SIP "trunk" to a provider, and no analog lines coming in, your faxes go in and out over whatever codec is in use by that "trunk" |
| 17:34.56 | Naikrovek | if it's g729, they won't work at all |
| 17:35.09 | Naikrovek | if it's g711 or g722 you have a good platform for t38 |
| 17:35.13 | Hive | Lobsters == elves? |
| 17:35.17 | jeffspeff | Naikrovek, Hive, in your alls implementations of fax, did you have to disable echo cancelation for g.711? |
| 17:35.24 | jeffspeff | my provider uses g.711 |
| 17:35.26 | Naikrovek | i don't recall |
| 17:35.35 | Hive | I dont think we were using echo cancellation |
| 17:35.37 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:35.48 | Naikrovek | i don't think you need it, honestly |
| 17:35.53 | Naikrovek | but i'm not sure |
| 17:35.59 | Naikrovek | i'm an ignorant hothead |
| 17:36.01 | jeffspeff | Naikrovek, isn't it enabled by default? |
| 17:36.02 | Hive | Have you tested your system without it? |
| 17:36.07 | Hive | hahaha naikrovek |
| 17:36.30 | jeffspeff | Hive, no |
| 17:38.31 | Naikrovek | echo comes in where either speakerphones or analog<->digital conversion is going on |
| 17:39.11 | Naikrovek | you probably won't notice a difference with it on or off since your end is not likely to cause any echo |
| 17:39.31 | jeffspeff | ok, so i'm not doing anything with analog phones... some people use speakerphone featuer on their deskphone, but I can turn off EC and see how it works |
| 17:40.01 | Naikrovek | fax machines don't use duplex speakerphones so you're unlikely to have a problem, unless the person you're sending a fax to (or receiving one from) has a real issue with their system |
| 17:40.22 | Naikrovek | but probably best to leave it on |
| 17:40.58 | jeffspeff | Naikrovek, so is it on all the time or just enabled, only being active when required? |
| 17:41.13 | Naikrovek | are you worried about cpu use or something |
| 17:41.19 | *** join/#asterisk iscario (~div@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 17:41.19 | Naikrovek | why are you so concerned about it |
| 17:42.16 | jeffspeff | no, the physical unit is a beast... i'm just trying to make sure that when i do this, it all works properly... this company has the bitchiest/dumbest employees that whine about anything and everything related to their computer or phones |
| 17:42.36 | Naikrovek | all companies are like that |
| 17:42.40 | Naikrovek | mostly.. |
| 17:42.51 | Naikrovek | what is the physical unit, i thought you were doing IP faxing |
| 17:42.59 | Naikrovek | jumped in this conversation late, as per usual |
| 17:43.22 | jeffspeff | by physical unit, i was refering to asterisk server |
| 17:43.34 | Naikrovek | oh |
| 17:44.11 | jeffspeff | 3.x ghz i7 processor, 60gb 6gbit p/sec solid state HDD, 8gb ddr3 ram |
| 17:44.16 | Naikrovek | once it's set up, it'll work well |
| 17:44.30 | Naikrovek | the pain in t38 is getting it to work in the first place |
| 17:44.40 | Naikrovek | some get lucky and have no issues, some can not get it working |
| 17:44.49 | Naikrovek | but usually once it's working, there's not much to worry about |
| 17:44.53 | Naikrovek | in my experience |
| 17:45.04 | jeffspeff | Naikrovek, do you suggest a certain fax module or just the Free Fax thing from digium? |
| 17:47.07 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:47.33 | Hive | I used Fax for Asterisk from digium, but that doesnt mean it's the best method ;-) |
| 17:48.29 | Naikrovek | yeah i don't know what to recommend there |
| 17:48.55 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 17:49.15 | Hive | (off to lunch, good luck jeff :-) ) |
| 17:49.39 | jeffspeff | ok, thanks for everybodies help |
| 17:50.37 | Naikrovek | np |
| 17:51.47 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:54.18 | McBoing | I setp peers in both my Asterisk servers, can see in the console the servera trying to connect but giving error on serverb that "Call from 'serverb' (192.168.128.244:5060) to extension rejected because extension '4000' not found in context 'incoming'." but there is an incoming context at the bottom of serverb's extension.conf...and yes I have sip reload'd, any ideas? |
| 17:55.34 | jeffspeff | McBoing, did you define an extension 4000 within the "incoming" context? |
| 17:55.47 | McBoing | yes I mentioend that |
| 17:56.15 | jeffspeff | no, you mentioned that you have the context, not the extension |
| 17:56.31 | McBoing | durr, well yes, 4000 exists there |
| 17:57.15 | McBoing | now servera is what everything is working from now, our production, what I am trying to do is setup a new Asterisk server, but couple it for now until I work out any kinks |
| 17:57.39 | McBoing | now this incoming context requirement I would imagine is expected on serverb not servera right? |
| 17:57.39 | jeffspeff | try doing a full reload instead of just a sip reload |
| 17:57.54 | McBoing | just what..."reload"? |
| 17:57.59 | jeffspeff | yep |
| 17:58.22 | McBoing | ahh different error this time |
| 17:58.25 | jeffspeff | and the dialplan depends entirely on how you're setting this up and which server you want the calls on |
| 17:58.35 | jeffspeff | McBoing, hope that helps |
| 17:59.39 | McBoing | I never setup a device to the other end of my extension, nor have I setup any voicemail options, got this error "Unable to create channel of type 'SIP' (cause 20 - Unknown)", is this because it has nowhere to go now? |
| 18:01.32 | McBoing | cool, got it working |
| 18:01.35 | *** join/#asterisk reber (~reber@212-198-54-60.rev.numericable.fr) |
| 18:03.37 | McBoing | how do I encompass extensions 0-999, I know 1000-1999 = _1xxx |
| 18:06.40 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:06.40 | Qwell | McBoing: 3 extens |
| 18:07.18 | Qwell | [0-9], [0-9][0-9], [0-9][0-9][0-9] |
| 18:07.35 | Qwell | or just Xs, heh |
| 18:07.41 | Qwell | X, XX, XXX |
| 18:08.59 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 18:11.50 | McBoing | nice thanks, so the underscore prefixing is something special I take it |
| 18:11.58 | Qwell | it means it's a pattern |
| 18:13.07 | McBoing | so it would be "exten => _xxx,1,Dial(SIP/${EXTEN})" ? |
| 18:13.27 | Qwell | X, not x |
| 18:13.35 | McBoing | oops yeah sorry I meant caps |
| 18:13.38 | paulc | Yup - that'll match any 3 digits.. including *12, 4*3, 99* (as opposed to just 000 to 999) |
| 18:13.46 | Qwell | no it won't |
| 18:13.55 | paulc | it won't? |
| 18:13.57 | Qwell | no |
| 18:16.21 | paulc | my bad.. I'm thinking device dialplan on a device that shall remain un-named |
| 18:16.32 | jeffspeff | lol |
| 18:16.36 | paulc | x = digits 0 through 9 only, in Asterisk dialplan. I stand corrected. |
| 18:17.12 | paulc | goes back to filling in his PMF form (and sighs) |
| 18:17.55 | thegoat | paulc: don't forget to use the new cover sheet |
| 18:18.38 | paulc | PMF = Performance Management Form (so yes, "PMF Form" is redundant, just like "PIN number"). It's a joke - we have a revised form every year I've been here and it's always a pointless exercise in my opinion... |
| 18:18.51 | paulc | hands thegoat some completed TPS reports |
| 18:19.22 | thegoat | paulc: lol |
| 18:19.56 | thegoat | paulc: so basically it's a performance review? |
| 18:20.48 | paulc | the goat: yup! Used to be quarterly, now we fill the form twice a year.. but it's silly because there are weekly one-on-ones, so everyone knows what's going on and where we're at.. |
| 18:22.55 | *** join/#asterisk radic (~radic@tmo-096-244.customers.d1-online.com) |
| 18:23.26 | thegoat | paulc: they had em at my last job, but we would never get any bonuses based on them. it was rediculous. |
| 18:23.56 | McBoing | it just keep HR busy for a little while |
| 18:24.04 | paulc | The way I see it, if you're meeting regularly, and having the right conversations, a formal "performance review" process or form should not be necessary. |
| 18:24.28 | paulc | McBoing: True that.. got to give them something to do right? (I honestly don't know what they do the rest of the time - it's obviously not recruiting...) |
| 18:24.40 | McBoing | planning vacation lol |
| 18:25.03 | McBoing | I wish all HR Managers were like Pam from Archer though, would definitely make life more...interesting |
| 18:26.00 | McBoing | were = was? grammar...meh |
| 18:26.57 | luminforce | it appears that Asterisk does not send a BYE when a mid-call INVITE (session-timer) isn't ackknowledge (after 3 or 4 retransmits of the OK). i can't seem to find a ticket for this, has anyone else heard of this problem? |
| 18:26.58 | McBoing | my motivation has hit a plateau for today |
| 18:28.59 | DocAwesome | McBoing: +1 |
| 18:29.25 | DocAwesome | luminforce: do you have session timers enabled in Asterisk? |
| 18:29.29 | McBoing | did I just get a golden star or are you trying to increase my motivational threshold? |
| 18:29.39 | DocAwesome | McBoing: yes |
| 18:29.51 | McBoing | head-desk lol |
| 18:35.31 | luminforce | we are accepting session timers, but not originating them. |
| 18:37.51 | *** part/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 18:42.23 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 18:53.40 | *** join/#asterisk vgster (~vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:55.34 | billmania | I've somehow managed to create a situation with asterisk 1.8.4.2 where an incoming queue call is sent to the member agent's voicemail, instead of to the next member agent, when the first agent doesn't answer the call. |
| 18:55.46 | billmania | Suggestions on which part of the dialplan could create that scenario? |
| 18:57.39 | Qwell | don't dial agents by their normal extension |
| 18:58.28 | billmania | Qwell: I'm not. :-D |
| 18:58.48 | billmania | They're being routed to the first member agent through Queue() and an IVR definition. |
| 18:58.52 | Qwell | show us dialplan |
| 18:59.00 | billmania | Standby. |
| 18:59.18 | Qwell | 14 #asterisk points says you're adding their extension to the queue. |
| 19:03.25 | ChannelZ | and the exten has macros or crap which call voicemail after a timeout. The console aught be showing you exactly what is happening |
| 19:06.52 | *** join/#asterisk giovani (~giovani@unaffiliated/giovani) |
| 19:07.10 | giovani | does anyone here use cisco voip phones via sip that are on a different subnet than their asterisk server? |
| 19:11.38 | kfife | Hey folks. MOH Question: Any way to turn down the gain on MOH without re-coding my MOH files? |
| 19:12.36 | billmania | Qwell: http://pastebin.com/raw.php?i=THBz4Dvz |
| 19:12.45 | kfife | giovani: Polycom yes, Cisco no. |
| 19:12.52 | billmania | I intentionally obfuscated the DIDs. |
| 19:14.14 | Qwell | Yep, dialing their extension. |
| 19:14.41 | billmania | Which line are you looking at? |
| 19:14.45 | Qwell | all of them |
| 19:14.53 | Qwell | macro-agent-add |
| 19:16.39 | billmania | I'm going to guess that what you are implying is that I am adding the wrong details for the agent to the queue, using AddQueueMember()? |
| 19:19.09 | ChannelZ | You are probably adding people's extensions (via a Local channel) to the queue rather than calling their device directly |
| 19:20.02 | ChannelZ | Which sometimes is what you want to do so you can make your dialplan do other things first or after.. but in this case your dialplan sends people to voicemail, etc. which you don't want |
| 19:21.06 | giovani | kfife: that's not helpful :) |
| 19:21.34 | billmania | How does app_queue decide what to do next, when the current member of the queue doesn't answer? |
| 19:21.46 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 19:22.18 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-213.telkomadsl.co.za) |
| 19:23.13 | billmania | I have "autopause" defined in queues.conf and that's no longer happening. |
| 19:23.28 | Qwell | billmania: They *are* answering. |
| 19:23.48 | Qwell | That's the point. The extension you're dialing is doing everything you told it to do. |
| 19:24.08 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:24.08 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:24.48 | billmania | Qwell: OK, voicemail is answering. |
| 19:25.19 | *** join/#asterisk MarKsaitis (~MarKsaiti@78-61-148-80.static.zebra.lt) |
| 19:25.58 | billmania | The dialplan handles an incoming call from the outside and routes it to Queue(). At that point, the dialplan has finished with that phase of the call. |
| 19:26.08 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:26.08 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 19:26.27 | billmania | The app_queue modules then retrieves a call from the queue and starts trying to find a member to deliver it to. |
| 19:26.36 | Qwell | Until it calls the queue member, and dials the extension that you gave it. |
| 19:26.37 | billmania | Am I close so far? |
| 19:26.47 | Qwell | which, in this case, is the extension of the queue member |
| 19:27.05 | billmania | I only "gave" it a queue member by having called AddQueueMember() at some point in the past. |
| 19:27.29 | billmania | How should I specify the member for AddQueueMember() differently than the way I'm doing it? |
| 19:27.53 | Qwell | Use a device other than local/theirextension. |
| 19:28.08 | Qwell | Either create a separate extension, or dial the real device directly |
| 19:28.32 | billmania | OK, I understand. Define yet another macro which never routes to voicemail or anything else. |
| 19:28.55 | Qwell | or don't use a macro |
| 19:29.03 | billmania | Here's the rub. I'm using the same macro-agent-add with two other systems, both of whose queues do exactly what I want. |
| 19:29.50 | billmania | This dialplan also did exactly what I want, until I made some change somewhere else (not to macro-agent-add). Now the behavior of the queue is different. |
| 19:30.08 | Qwell | So you made a change, and now something doesn't work? Shocking. |
| 19:30.23 | jaytee | changelogs are nice |
| 19:30.30 | jaytee | in case you have to backtrack |
| 19:30.34 | billmania | Yes, so is hindsight. |
| 19:30.35 | Qwell | These are probably things you should mention at some point, when troubleshooting. |
| 19:30.43 | jaytee | hindsight is 20/20 |
| 19:30.48 | billmania | Isn't that what I'm doing now? |
| 19:31.04 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:31.22 | Qwell | 36 minutes later |
| 19:33.26 | billmania | Qwell: (01:55:33 PM) billmania: I've somehow managed to create a situation |
| 19:38.59 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:50.38 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 19:54.39 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 19:59.54 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:01.11 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:02.14 | *** join/#asterisk tzafrir (~tzafrir@46-116-25-26.bb.netvision.net.il) |
| 20:02.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:09.56 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:15.12 | *** join/#asterisk edozier (~edozier@rrcs-70-61-148-242.central.biz.rr.com) |
| 20:15.25 | *** join/#asterisk pdtpatrick (~pdtpdt@mainstwan.farheap.com) |
| 20:18.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:19.06 | edozier | /usr/src |
| 20:24.01 | Hive | /etc/asterisk/ |
| 20:28.55 | edozier | oops thanks. I meant to type into terminal. |
| 20:29.24 | _Corey_ | I hate it when I type stuff in the wrong window... like 'halt' :( |
| 20:29.34 | edozier | rofl |
| 20:29.59 | edozier | New to Asterisk. Trying to figure it all out...... |
| 20:30.11 | seraphie | better "rm -rf *" here... |
| 20:30.38 | _Corey_ | ouch |
| 20:31.11 | *** join/#asterisk tzafrir (~tzafrir@93-172-185-195.bb.netvision.net.il) |
| 20:31.28 | seraphie | I actually rm -rf / home/.. on my personal computer ages ago. |
| 20:31.31 | seraphie | note the space. |
| 20:31.55 | seraphie | That was painful |
| 20:32.26 | WIMPy | Lucky ytou didn't put it in the wrong window now. |
| 20:32.33 | edozier | I tried to install the asterisknow install with the freepbx gui, but cannot get 3CX or SJphone to connect. Would I be better off doing an fresh install of Linux and then installing asterisk? |
| 20:33.11 | edozier | WIMP I agree :) |
| 20:33.38 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 20:36.09 | seraphie | edozier: try again with Asterisk GUI |
| 20:36.24 | seraphie | and if you need help, there's #asterisk-gui |
| 20:37.44 | edozier | Thank you!! Have a good one. |
| 20:42.25 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 20:49.21 | ChannelZ | Is there a Google Talk app for Android (or anything) that can do voice calls, or do they only do IM? |
| 20:50.47 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-56-250.w86-196.abo.wanadoo.fr) |
| 21:02.25 | *** join/#asterisk s14ck (~jtorres@190.202.110.236) |
| 21:03.16 | *** join/#asterisk DocAwesome (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:03.16 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 21:04.53 | hardwire | ok so.. voicemail 'sales' doesn't work with VoicemailMain |
| 21:04.57 | hardwire | voicemail 'vmsales' does. |
| 21:05.01 | hardwire | voicemail salesvm does not. |
| 21:05.02 | hardwire | sigh. |
| 21:05.47 | ChannelZ | that means nothing to any of us |
| 21:05.54 | hardwire | it does so |
| 21:05.58 | DocAwesome | nope |
| 21:06.01 | hardwire | yes. |
| 21:06.03 | DocAwesome | lies. |
| 21:06.10 | DocAwesome | see what I did that? |
| 21:06.12 | DocAwesome | there... |
| 21:06.13 | DocAwesome | eff |
| 21:06.15 | hardwire | eff |
| 21:06.18 | DocAwesome | diaf |
| 21:06.22 | hardwire | no you |
| 21:06.29 | ChannelZ | Actually it does mean something.. it means you screwed something up, but what we have no idea |
| 21:06.31 | DocAwesome | yes, I was talking to myself |
| 21:06.52 | DocAwesome | I was telling myself to diaf, so your "no you" means nothing! |
| 21:07.06 | hardwire | :| |
| 21:07.16 | hardwire | ChannelZ: yeh.. it sure feels like I may have screwed something up. |
| 21:07.37 | ChannelZ | NEXT! |
| 21:08.01 | hardwire | I have a voicemail box named 'vmsales' that works.. if I kill the 'vm' part in both the voicemail config for that box as well as in the extension that voicemailmain is being used in.. it fails. |
| 21:08.08 | hardwire | it's rather wonky |
| 21:08.48 | s14ck | hi people |
| 21:08.53 | *** join/#asterisk P-NuT (~P-NuT@5ad48b0d.bb.sky.com) |
| 21:09.20 | s14ck | any free phone number to SIP at Mexico? |
| 21:10.09 | P-NuT | Hey there all, when using MeetMe, it prompts you to record your name, which is fine, but then it says "press 1 to save this recording". How do I make it just record the users name, then press hash and put the user straight into the call? |
| 21:10.21 | hardwire | https://raw.github.com/gist/4c6e94b45be404758acf/aa23f4d81bbab845283cc5d529c679d4414da5a9/gistfile1.txt |
| 21:10.25 | hardwire | see.. very strange. |
| 21:10.53 | hardwire | vmsales and sales are identically defined.. vmsales is actually a line copy of sales. |
| 21:10.56 | hardwire | so confused. |
| 21:15.51 | WIMPy | Is there any configuration option to suppress the warnings about the use of _! or _. ? |
| 21:16.11 | hardwire | hehe |
| 21:18.10 | DocAwesome | WIMPy: don't use them? |
| 21:18.31 | DocAwesome | _. is easy to work around with _[A-Za-z0-9]! |
| 21:19.04 | hardwire | yeh |
| 21:19.50 | WIMPy | Why would I restrict the match to those characters? |
| 21:20.26 | iscario | hello, would you have good resources about the use of video in Asterisk 1.8 ? |
| 21:22.31 | WIMPy | Maybe I should use a global variable to to keep it quiet. |
| 21:23.29 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 21:23.45 | Qwell | _! ...huh |
| 21:23.47 | Qwell | would that work? |
| 21:23.53 | DocAwesome | Qwell: it does, but has warnings |
| 21:23.59 | Qwell | yes, but think of the implications |
| 21:24.10 | DocAwesome | WIMPy: I'm just showing how to do it -- put whatever chars you want there |
| 21:24.14 | Qwell | a 0 digit extension? |
| 21:24.34 | DocAwesome | in most cases, alphanumeric will match most of what you want |
| 21:24.41 | DocAwesome | and all of what you would expect |
| 21:25.07 | WIMPy | It will need a + at least. |
| 21:25.23 | DocAwesome | ok so add it :) |
| 21:25.36 | WIMPy | That's why I wanted all, not most. |
| 21:27.47 | DocAwesome | ok then modify the source to not show the WARNING |
| 21:27.51 | DocAwesome | that's what you really want :) |
| 21:28.10 | DocAwesome | beyond that the only way to suppress those messages is to not use _. |
| 21:28.19 | WIMPy | I've done that for other things. Was just asking if there's an option. |
| 21:29.08 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-eavyxkomtyascdfu) |
| 21:30.03 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 21:30.17 | WIMPy | But we can also get back to a more interesting thing: |
| 21:30.31 | WIMPy | Does anyone have any experience using Incomplete()? It seems to be synonymous to Goto(i), which is not what I had hoped for (and doesn't seem to make too much sense). |
| 21:30.32 | *** join/#asterisk nightrid3r (~kvirc@91.176.0.12) |
| 21:30.51 | DocAwesome | this is more interesting: http://www.asterisk.org/node/51667 |
| 21:31.00 | DocAwesome | Asterisk 10.0.0 Beta 1 is now available for testing! http://www.asterisk.org/node/51667 |
| 21:31.34 | *** topic/#asterisk by DocAwesome -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:32.55 | ariel_ | Just great, another way to confuse us with version numbers. |
| 21:33.15 | DoDaT69 | heh |
| 21:33.34 | WIMPy | Why? |
| 21:34.00 | WIMPy | The bad thing is that it will take some time again for some channels to get updated yo the new media handling. |
| 21:34.27 | ChannelZ | I find the number 10 highly confusing |
| 21:34.43 | DocAwesome | ariel_: totally explained on the digium blog, and it's exactly the same as 1.10.0.0-beta1 except we don't have the '1.' on the front |
| 21:34.47 | DocAwesome | that's all it is |
| 21:35.01 | DocAwesome | 1.10.0.0-beta == 10.0.0-beta1 |
| 21:35.02 | ariel_ | well lets see, version 1.0.9 then 1.2, then 14.XX, then the mess of 1.6.X.X now 1.8.X jump instead to 2.0 to 10.. |
| 21:35.04 | carrar | heh |
| 21:35.09 | carrar | 10 |
| 21:35.11 | DocAwesome | where did you get 14? |
| 21:35.19 | carrar | 1.8 to 10.0 |
| 21:35.22 | DocAwesome | please read the blog post |
| 21:35.31 | ChannelZ | Asterisk.. OF THE FUTAR! |
| 21:35.32 | ariel_ | sorry 1.4 |
| 21:35.38 | DocAwesome | 1.10 == 1.^H^H10 |
| 21:35.53 | ariel_ | in the mix it should have gone to 2.X |
| 21:36.04 | ariel_ | as 1.8 next up would have been to 2.0 |
| 21:36.13 | DocAwesome | ariel_: no, it shouldn't have got to 2.0 |
| 21:36.15 | carrar | 1 million |
| 21:36.20 | DocAwesome | 1.8 --> 1.10 |
| 21:36.21 | DocAwesome | not 2.0 |
| 21:36.25 | carrar | (pinky in teeth) |
| 21:36.43 | WIMPy | Lets discuss the leave out the odd numbers thing :-) |
| 21:36.46 | DocAwesome | infobot: asterisk10 is <reply> Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 21:36.46 | infobot | okay, DocAwesome |
| 21:36.53 | DocAwesome | wait... I always do that wrong |
| 21:36.53 | ariel_ | 1.10 is lower version then 1.8 |
| 21:36.55 | Qwell | ~10 |
| 21:36.55 | infobot | There are 10 types of people in this world. Those who understand binary and those who don't. |
| 21:36.59 | ariel_ | so there is a misstake in views |
| 21:37.00 | Qwell | ~ten |
| 21:37.01 | DocAwesome | ariel_: no it's not |
| 21:37.15 | DocAwesome | ~asterisk1 |
| 21:37.17 | DocAwesome | ~asterisk10 |
| 21:37.17 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 21:37.22 | DocAwesome | oh yay I got it right first try |
| 21:37.25 | DocAwesome | or someone fixed it :) |
| 21:37.29 | ChannelZ | It's not a decimal number |
| 21:37.33 | DocAwesome | ChannelZ: +1 |
| 21:37.42 | ChannelZ | DocAwesome: no you did it right |
| 21:37.45 | DocAwesome | the . is just a seperator |
| 21:37.55 | DocAwesome | it could have been 1-8-0-0 |
| 21:37.58 | DocAwesome | or 1-10-0-0 |
| 21:38.06 | DocAwesome | or 1*8*0*0 |
| 21:38.18 | WIMPy | Which should be obvious as numbers can only have one decimal point. |
| 21:38.19 | DocAwesome | as ChannelZ says, it's not a decimal number |
| 21:38.24 | DocAwesome | WIMPy: +1 |
| 21:38.25 | ariel_ | Oh well, that just makes no sense in my view. But it's your ball game not mine |
| 21:38.41 | DocAwesome | ariel_: it's the way most software release numbering schemes work |
| 21:38.51 | DocAwesome | Kernel: 2.6.x |
| 21:38.54 | DocAwesome | 2.4.x |
| 21:38.55 | carrar | 3.0! |
| 21:38.58 | DocAwesome | exactly :) |
| 21:40.12 | ariel_ | we were alredy there 1.0.9, 1.2.X then 1.4.X then 1.6 kinda screwed it up, then 1.8 natural view for my old brain would be 2.0 I come from way back when I started with asterisk on .50 |
| 21:40.32 | DocAwesome | your brain is working incorrectly :) |
| 21:40.34 | ChannelZ | I'm bored |
| 21:40.39 | Qwell | ariel_: Think about it this way. |
| 21:40.46 | DocAwesome | the version numbering in Asterisk (or linux) has never been a decimal number |
| 21:40.52 | Qwell | 1.4.10 is newer than 1.4.9, right? |
| 21:40.58 | Qwell | So why would the second digit be any different? |
| 21:41.07 | DocAwesome | and 1.4.9.1 can be newer than 1.4.10 |
| 21:41.50 | ChannelZ | They should be renamed as bra sizes |
| 21:41.53 | *** join/#asterisk DelphiWorld (~Miranda@openvpn/user/DelphiWorld) |
| 21:41.55 | ariel_ | in those move yes, but not from my accounting view. but I am just an old fart, I understand, just not expecting this. |
| 21:42.39 | WIMPy | Well, as I said last week, just use the current date in ISO format as version number. |
| 21:42.50 | Kobaz | oh, this is fun |
| 21:43.05 | DocAwesome | hey, what Linux distro is the best?! |
| 21:43.10 | ChannelZ | Mine! |
| 21:43.22 | SunTsu | DocAwesome: always the one you don't run |
| 21:43.28 | ariel_ | I use the one that is right for the job it needs to be |
| 21:43.44 | DelphiWorld | hello |
| 21:43.49 | DelphiWorld | WTF pabelanger |
| 21:44.00 | DocAwesome | DelphiWorld: wtf DelphiWorld |
| 21:44.07 | pabelanger | ZOMG |
| 21:44.09 | DelphiWorld | DocAwesome: :P |
| 21:44.19 | DocAwesome | DelphiWorld: you should ask a real question |
| 21:44.41 | DelphiWorld | inquire pabelanger |
| 21:44.45 | pabelanger | http://bit.ly/qi3YVM |
| 21:44.50 | DelphiWorld | DocAwesome: that's a deal bethwan me and pabelanger. |
| 21:45.04 | DocAwesome | pabelanger: you can use ~asterisk10 now too |
| 21:45.16 | pabelanger | what what! |
| 21:45.20 | pabelanger | ~asterisk10 |
| 21:45.21 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 21:45.21 | DocAwesome | omg! we're using numbers that you didn't expect! everyone freak out! |
| 21:45.28 | pabelanger | TO THE CLOUD! |
| 21:45.33 | DocAwesome | CLOUDS FTW! |
| 21:45.35 | DelphiWorld | should by 1.10! WTF. |
| 21:45.56 | DocAwesome | please read the blog! |
| 21:45.59 | DocAwesome | that's why it's there |
| 21:46.05 | DocAwesome | 1.10 == 1^H^H10 |
| 21:46.08 | DocAwesome | oops |
| 21:46.12 | DocAwesome | 1.10 == 1.^H^H10 |
| 21:46.17 | pabelanger | lol |
| 21:46.19 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:46.20 | DocAwesome | seriously though |
| 21:46.23 | WIMPy | 110.0 |
| 21:46.24 | ariel_ | DocAwesome: In your view, I don't need to freak out. There are other things we can use , we have choices in this large world |
| 21:46.25 | russellb | whoa, we didn't see this coming! |
| 21:46.38 | pabelanger | russellb: epic! |
| 21:46.39 | DocAwesome | russellb: the pie contest did NOT see this coming! |
| 21:46.47 | DelphiWorld | yo russellb |
| 21:46.51 | russellb | waves |
| 21:46.58 | DocAwesome | russellb: put your hand down |
| 21:47.04 | pabelanger | Going to try asterisk-10 packages tonight |
| 21:47.09 | russellb | DocAwesome: you say SIR when talking to me, damnit |
| 21:47.19 | DocAwesome | russellb: you shut your mouth when talking to me |
| 21:47.22 | DocAwesome | good thing we're just typing |
| 21:47.31 | pabelanger | fight, fight! |
| 21:47.38 | DocAwesome | pushes russellb into a bush |
| 21:48.02 | DelphiWorld | push back DocAwesome |
| 21:48.05 | russellb | more like DocBitchface |
| 21:48.23 | DelphiWorld | respectfully russellb |
| 21:48.39 | DocAwesome | I have no idea what is going on |
| 21:48.43 | russellb | I also do not know |
| 21:48.49 | russellb | DocAwesome: <3 |
| 21:48.59 | DocAwesome | russellb: <3 +1 |
| 21:49.28 | russellb | 10! |
| 21:49.40 | DocAwesome | nice |
| 21:49.47 | DocAwesome | ok Asterisk 10.0.0-beta1 is now available for testing |
| 21:49.59 | WIMPy | 10==2, BTW *eg* |
| 21:50.02 | DelphiWorld | lol |
| 21:50.34 | Qwell | russellb: You know what else you didn't see coming? |
| 21:50.37 | DocAwesome | russellb: that idea about using a new decimal place of Pi for each release would have been more acceptable it seems |
| 21:51.05 | Kobaz | getting illegal instruction loading up asterisk trunk |
| 21:51.14 | Qwell | Kobaz: The authorities have been alerted. |
| 21:51.18 | DelphiWorld | russellb: better to run asterisk A after 1.8 |
| 21:51.28 | Qwell | DelphiWorld: We considered 1.A briefly. |
| 21:51.37 | DelphiWorld | Qwell: readed it :P |
| 21:52.11 | WIMPy | Better change all outputs of cause codes to hex. Having them in decimal is quite annoying. |
| 21:52.30 | DelphiWorld | Qwell: any major feature ? |
| 21:52.58 | afink | Hello everyone, I am having some issues with ISDN my equipment took a power hit earlier and I am getting some messages in my cli and my PRI is not functional. Please take a look and see if you know what some of this means. http://pastebin.com/tThKSRZD |
| 21:53.47 | WIMPy | afink: Looks like someone has gone deaf. |
| 21:53.59 | WIMPy | Was something fried? |
| 21:54.32 | afink | we did have a loss of power today |
| 21:55.54 | russellb | DocAwesome: haters gonna hate |
| 21:55.55 | DelphiWorld | Qwell: may i PM you about a private (out of asterisk discution) ? |
| 21:56.10 | russellb | Qwell: it's a trap |
| 21:56.52 | Qwell | no |
| 21:57.08 | Qwell | Unless it involves sending me large sums of money, for zero effort on my part. |
| 21:57.25 | Qwell | but, a PM would be effort, so still no. |
| 21:57.26 | DelphiWorld | Qwell: no to me or to russellb ? |
| 21:57.34 | Qwell | always to you |
| 21:57.34 | DelphiWorld | Qwell: hmmm |
| 21:57.46 | Qwell | russellb already gives me moneys |
| 21:57.49 | Qwell | by proxy |
| 21:58.14 | DelphiWorld | Qwell: no joke... i am serious. |
| 21:59.19 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:01.52 | russellb | Qwell: that is a true statement |
| 22:02.07 | russellb | flexes |
| 22:02.15 | afink | WIMPy: is that the most likely option? |
| 22:02.16 | DelphiWorld | russellb: :P |
| 22:03.35 | WIMPy | afink: It looks like you are not receiving _anything_. Do you have an alarm on that port? |
| 22:03.47 | WIMPy | Or the other end? |
| 22:05.19 | afink | WIMPy: neither |
| 22:05.25 | DocAwesome | russellb: haters gonna hate hard |
| 22:06.32 | DocAwesome | runs off shortly to take his fiancee out to The Keg for their 3 year |
| 22:06.48 | DelphiWorld | enjoi DocAwesome |
| 22:07.02 | DocAwesome | no, my fiancee will enjoy DocAwesome |
| 22:07.10 | pabelanger | HIO! |
| 22:07.20 | DocAwesome | whoa |
| 22:07.28 | WIMPy | afink: Interesting. Shredded config? Or maybe some loopback got activated? |
| 22:07.52 | russellb | DocAwesome: I did NOT see that coming! |
| 22:07.57 | DelphiWorld | slaps DocAwesome around a bit with a large trout |
| 22:08.00 | DelphiWorld | ok then :P |
| 22:08.00 | Qwell | DocAwesome: Mr. and Doctor Awesome? |
| 22:08.15 | Qwell | err, wow, how the hell did that happen? I read "enjoy" as "be" |
| 22:08.31 | DelphiWorld | Qwell: :P |
| 22:08.54 | Kobaz | maybe i'll stay connected this time |
| 22:09.04 | Kobaz | so anyways... why the jump from 1.10 to 10.0 |
| 22:09.12 | russellb | ~asterisk-10 |
| 22:09.22 | russellb | ~asterisk10 |
| 22:09.22 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 22:09.27 | Kobaz | ah |
| 22:09.28 | WIMPy | -"-" |
| 22:09.34 | DelphiWorld | where's asterisk 2,3,4,5,6,7,8,9 ? dead ? |
| 22:09.42 | Kobaz | deaded |
| 22:09.45 | Qwell | reserved |
| 22:10.01 | Qwell | we can have like Asterisk Throwback, 2.0 |
| 22:10.01 | DelphiWorld | Qwell: for what? Qwell wife to eat it i think :D |
| 22:10.09 | russellb | 10 == 2.0 in binary |
| 22:10.18 | DelphiWorld | :P |
| 22:10.28 | pabelanger | Asterisk over 9000 |
| 22:10.52 | russellb | Asterisk X Signature Edition |
| 22:11.01 | WIMPy | Faster than Firefox. |
| 22:11.02 | Kobaz | Asterisk Lion XP |
| 22:11.08 | WIMPy | Oh, I like Asterisk X *ggg* |
| 22:11.24 | Kobaz | Asterisk Lion XP XXX |
| 22:11.36 | DelphiWorld | Asterisk Vista buggy edition :P |
| 22:11.37 | WIMPy | We could also increment "Asterisk" to "Plus". |
| 22:11.44 | DelphiWorld | asterisk 7 stable edition |
| 22:11.54 | DelphiWorld | asterisk 2000 |
| 22:11.57 | DelphiWorld | asterisk++ |
| 22:12.16 | DelphiWorld | WIMPy: asterisk++++++++++ |
| 22:13.32 | afink | guys here is some more debug info: http://pastebin.com/EDnzKGBq if you could take a look it would be greatly appreciated. WIMPy, I can call into the system but I cannot call out. |
| 22:13.41 | DelphiWorld | :P |
| 22:14.16 | *** topic/#asterisk by DocAwesome -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 22:14.24 | WIMPy | afink: You didn't mention that detail. |
| 22:14.52 | afink | WIMPy: just found that out myself |
| 22:16.04 | DelphiWorld | DocAwesome: good toppic:D |
| 22:16.18 | WIMPy | Looks like the upper layer is working, but something on the lower layer has a major hickup. |
| 22:16.22 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 22:16.43 | WIMPy | Have you tried to restart the whole thing again? |
| 22:17.23 | afink | WIMPy: I have several times |
| 22:17.59 | afink | I have a spare TDM 122p probably ought to try swapping |
| 22:18.12 | DelphiWorld | good night. |
| 22:18.20 | *** part/#asterisk DelphiWorld (~Miranda@openvpn/user/DelphiWorld) |
| 22:18.38 | WIMPy | afink: Unless you also have audio problem, it doesn't look that bad. |
| 22:19.00 | WIMPy | But I don't really have a clue as to what might be going on there. |
| 22:19.05 | afink | ISP? |
| 22:19.23 | WIMPy | I'd try to pull the plug, wait for at least 20s and put it in again. |
| 22:19.52 | WIMPy | No. at least 30s. |
| 22:28.54 | ChannelZ | infobot: karma for asterisk |
| 22:28.54 | infobot | for asterisk has neutral karma |
| 22:29.05 | afink | o.0 |
| 22:29.06 | ChannelZ | well you're just not paying attention then |
| 22:29.36 | ChannelZ | actually you're just broken... |
| 22:29.41 | ChannelZ | infobot: karma asterisk |
| 22:29.41 | infobot | asterisk has neutral karma |
| 22:29.51 | ChannelZ | Both really. |
| 22:32.33 | iscario | hello! I set up an Asterisk server on a lan. I only use IAX channels until now, and i'm satisfy with this configuration. Now, I would like to add a video support. Do you have resources to advice me ? |
| 22:36.09 | ChannelZ | Switch to SIP? |
| 22:36.56 | ChannelZ | (I'm not actually sure if IAX does vid or if any devices support it if it does) |
| 22:37.57 | kaldemar | iscario: iax2 does support video. allow the wanted codecs with allow lines. for SIP, add videosupport=yes in sip.conf and the codecs. |
| 22:38.34 | kaldemar | iscario: video conferencing is on the roadmap for asterisk 10 (used to be called 1.10). |
| 22:43.19 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-372608.home.otenet.gr) |
| 22:45.24 | iscario | ok kaldemar. At the present time i use an open-source iax softphone i compiled from sources, do you know if it would be easy (let's say ... possible at least ; ) to include the video support in this softphone (librairies available,...) ? |
| 22:47.27 | kaldemar | ~smop |
| 22:47.28 | infobot | extra, extra, read all about it, smop is just a simple matter of programming or encrypted floor cleaning tool |
| 23:07.44 | *** join/#asterisk jmwpc (~jeremy@c-76-103-168-194.hsd1.ca.comcast.net) |
| 23:07.47 | *** part/#asterisk jmwpc (~jeremy@c-76-103-168-194.hsd1.ca.comcast.net) |
| 23:11.56 | *** join/#asterisk hdiogenes (~humberto@201.76.154.125.intranet.digi.com.br) |
| 23:12.32 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 23:21.22 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 23:22.08 | *** part/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
