| 00:01.18 | knarfly | wxactly what does sip show channels report? |
| 00:01.54 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:02.01 | p3nguin | existing sip channels |
| 00:02.33 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 00:02.49 | knarfly | so iif no one is on any of my extensions and I do sip show channels at the cli and a channel shows up what does that mean? |
| 00:03.09 | p3nguin | You've confused extensions and sip channels. Again. |
| 00:03.57 | p3nguin | sip show channels shows SIP CHANNELS, not extensions. |
| 00:04.14 | knarfly | okay but if there are no users on the phone why would a strange IP address (not my provider's) show up in sip show channels? |
| 00:04.47 | p3nguin | Someone might be making an anonymous call inbound, or maybe there was an phone registering. |
| 00:05.20 | p3nguin | There should be a message to indicate the reason the channel is active. |
| 00:06.18 | knarfly | how would I see the message, all it says is the channel and a strange callerID |
| 00:06.31 | p3nguin | You can also use "sip show channel <channel>" to see more info on the channel. |
| 00:06.48 | p3nguin | sip show channels does not show "callerID." |
| 00:06.58 | knarfly | okay thanks. |
| 00:07.41 | p3nguin | It does, however, show a "Call ID," which is a unique identification for the channel. |
| 00:07.44 | knarfly | excuse me...Call ID not callerID ... my bad |
| 00:08.54 | p3nguin | That's the abbreviated name of the channel. |
| 00:08.54 | p3nguin | If, for example, the Call ID is 69b0359a0d5, you could try "sip show channel 69b0359a0d5<TABKEY>" and it should complete the full channel ID and then you can press Enter. |
| 00:10.26 | knarfly | cool...it looks like it has something to do with my VOIP provider |
| 00:10.56 | p3nguin | There's no "last message" showing when you did sip show channels? |
| 00:11.09 | p3nguin | like Rx: INVITE |
| 00:11.15 | p3nguin | or Rx: REGISTER |
| 00:12.05 | knarfly | ok it's showing up as an incoming call...which is strange because this is a private server and no one has the DID |
| 00:12.28 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:12.47 | knarfly | the IP address is in Santa Clara, CA |
| 00:13.09 | p3nguin | Check "core show channels" to see if there is an active call. |
| 00:14.20 | p3nguin | If there is, it will show you the location in the dialplan where the call is. |
| 00:14.23 | knarfly | shows 0 active calls but this things is coming and going like some kiddie on this channels is trying to hack in or gain access to the service |
| 00:14.46 | p3nguin | If the IP address is of no use to you, block it at the firewall. |
| 00:15.03 | p3nguin | The IP address is in the sip channel information. |
| 00:15.05 | knarfly | done |
| 00:15.09 | vader-- | i know im going to get flack for asking this hehe.. I am running make menuselect right now and im looking for unneccessary modules i can remove. I am going to be running freepbx 2.8 B2 on top of asterisk 1.6.2.8 |
| 00:21.19 | p3nguin | Now, before we get too far away from strange sip channels being active... how can I "close" a sip channel that is active? There's no active call, so I can't do a soft hangup on it. |
| 00:23.08 | knarfly | p3nguin: can't say from my end...that IP turned out to be my VOIP provider...although I don't understand...they usually only show up as a sip registry |
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| 00:23.28 | p3nguin | Heh, so you blocked your ITSP! |
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| 00:24.57 | carrar | block everything |
| 00:25.04 | knarfly | no I just took a closer look at the IP again and then pinged the information I knew about my provider |
| 00:25.24 | carrar | lessen your change of getting h4X0r3D |
| 00:25.27 | carrar | chance |
| 00:28.58 | knarfly | carrar: h4X0r3D ??? |
| 00:33.52 | grungies1138 | Hacked in queer speak |
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| 00:39.11 | p3nguin | lol |
| 00:40.06 | knarfly | takes another Percocet and drifts off into the jazz music and into the darkness |
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| 01:02.47 | miamiseb | Night all. |
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| 01:09.27 | netpro25_ | Hello, does anyone know what the going rate is for setting up an asterisk server and phones? |
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| 01:28.17 | carrar | $1 |
| 01:35.52 | rjk` | I need an outbound calling system |
| 01:36.11 | rjk` | and do you think I can find one for skype ? |
| 01:36.29 | rjk` | rhetorical question |
| 01:36.35 | rjk` | bleh :( |
| 01:36.47 | netpro25_ | $1 eh |
| 01:37.00 | netpro25_ | thats good, I should sub out the work to you |
| 01:37.01 | pabelanger_ | netpro25_: $40 - $250 per hour? |
| 01:37.39 | netpro25_ | pabelanger_, I mean for a complete system with lets say 3 phones |
| 01:37.46 | netpro25_ | not including hardware |
| 01:39.25 | pabelanger_ | netpro25_: like I said, anywhere from $40 to $250 per hour. You would need to spec out what you needed. |
| 01:39.32 | pabelanger_ | programming wise |
| 01:39.59 | netpro25_ | pabelanger_, k. Guess I will just charge per hour, wanted to charge per phone and per server. |
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| 01:44.56 | p3nguin | Most people have an hourly rate for services, but you can certainly specify a total price for the entire job if that's what you want to do. |
| 01:50.48 | ChannelZ | Hourly ho, or contract ho? Choose wisely! |
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| 01:52.31 | coppice | for some, a 5 minutely ho offers better value |
| 01:53.51 | vader-- | how can i test connectivity to a remote sip server? |
| 01:54.34 | ChannelZ | connect to it? |
| 01:55.16 | ChannelZ | or you could play with sipsak |
| 01:55.22 | ChannelZ | which sounds totally dirty but it's not |
| 01:56.18 | vader-- | ya i asked this guy to configure his firewall but im not sure if he did or not |
| 01:59.08 | vader-- | when i do a netstat -an on the box i don't see an port 5060 listening |
| 01:59.09 | vader-- | :-( |
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| 02:07.48 | p3nguin | Are you ON the server when you check netstat? |
| 02:08.15 | p3nguin | You indicated a remote server, so I'll need some clarification. |
| 02:08.41 | vader-- | yes |
| 02:08.48 | vader-- | chan_sip.so isn't loading |
| 02:09.50 | p3nguin | Do you know why? |
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| 02:12.21 | vader-- | it wasn't loading because there is no sip.conf file |
| 02:12.25 | vader-- | im trying to install freepbx |
| 02:15.06 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
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| 02:20.40 | vader-- | ok getting further |
| 02:20.47 | vader-- | now im getting no matching peer found |
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| 02:21.04 | *** part/#asterisk coil (platypus@unaffiliated/coil) |
| 02:21.44 | p3nguin | Failure to create the necessary peer definition will do that. |
| 02:22.25 | vader-- | i did though |
| 02:22.28 | vader-- | hmmm |
| 02:22.56 | p3nguin | Failure to configured it correctly will also result in failure. :) |
| 02:23.29 | vader-- | hehe |
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| 02:24.50 | Carlos_Tico | i have an issue with the dtmf .. whenever i call an ivr cant detect the tones or detects the wrong ones ? |
| 02:25.24 | p3nguin | What dtmfmode are you using? |
| 02:25.46 | Carlos_Tico | on the outbound trunk dtmfmode=rfc2833 |
| 02:27.13 | vader-- | p3nguin ive created an extenstion in freepbx: 2000 with a secret of test |
| 02:27.25 | p3nguin | I'm sorry to hear that. |
| 02:27.27 | vader-- | when i pop that info into zoiper soft phone |
| 02:27.43 | vader-- | username: 2000 password: test i get that no matching peer |
| 02:27.49 | p3nguin | Subsequent references to FreePBX will not gain you any brownie points. |
| 02:28.07 | vader-- | hehe i know :-) |
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| 02:29.09 | Carlos_Tico | <p3nguin> What dtmfmode are you using? --> dtmfmode=rfc2833 |
| 02:30.54 | Carlos_Tico | p3nguin ? |
| 02:31.44 | grungies1138 | so I tried to 'file convert' in asterisk to get my MOH files to work, but it can't read either my coverted wav or my mp3 |
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| 02:39.01 | spenguin[work] | hey |
| 02:43.36 | spenguin[work] | how do I figure what codec is currently in use |
| 02:43.41 | spenguin[work] | when we make sip calls |
| 02:44.19 | p3nguin | sip show channels |
| 02:44.37 | spenguin[work] | will try it out p3nguin |
| 02:45.01 | p3nguin | It'll work. |
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| 02:45.43 | spenguin[work] | p3nguin: Im just wondering in general what codec is being used |
| 02:45.54 | spenguin[work] | this would show active calls |
| 02:46.01 | carrar | split personality? |
| 02:46.45 | spenguin[work] | and I dont really get anything under the "Format" section |
| 02:46.50 | p3nguin | "in general what codec is being used" doesn't make sense to me. |
| 02:46.51 | spenguin[work] | just 0x0 (nothing) |
| 02:46.59 | p3nguin | No codec is used if there is no call. |
| 02:47.13 | p3nguin | Make a call and a codec will show up there. |
| 02:47.19 | Carlos_Tico | <p3nguin> What dtmfmode are you using? --> dtmfmode=rfc2833 |
| 02:47.27 | russellb | you can see it in "core show channel <foo>", as well |
| 02:47.30 | p3nguin | That's three times you've said that, now, carlos_tico. |
| 02:47.32 | spenguin[work] | p3nguin: what about incomming calls? |
| 02:47.43 | p3nguin | If it's a sip call, there will be a codec. |
| 02:47.44 | spenguin[work] | incomming sip* |
| 02:47.54 | Carlos_Tico | yes bcz i didnt hear your advice ? |
| 02:47.57 | Carlos_Tico | :) |
| 02:47.58 | spenguin[work] | I dont see anything, will paste |
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| 02:48.31 | spenguin[work] | 204.11.xxx.xxx 17772043xxx 0b6b0bxxxed 00759/00000 0x0 (nothing) No |
| 02:48.46 | p3nguin | I guess it's not an active call, then. |
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| 02:48.56 | spenguin[work] | eh, ill confirm |
| 02:49.55 | jesselang|laptop | Hello. kaldemar, remember me from yesterday? I was asking about whether you could see which call leg signaled the call tear-down from AMI events. It doesn't seem that you can. |
| 02:54.20 | spenguin[work] | p3nguin: ok I got "alaw" under format |
| 02:54.28 | spenguin[work] | so that should be the code? |
| 02:54.31 | spenguin[work] | codec* |
| 02:54.36 | p3nguin | Then you know what codec that leg of the call is using. |
| 02:54.42 | spenguin[work] | ok sir |
| 02:54.51 | p3nguin | alaw is g.711a |
| 02:55.46 | spenguin[work] | which end decides what codec should be used, what if the codec is not supported on the other end |
| 02:55.54 | p3nguin | both |
| 02:56.07 | spenguin[work] | hrm, so sip handles that? |
| 02:56.30 | p3nguin | If you configure only g729 on your side, but the other side only supports alaw, no coced will be chosen and the call will fail. |
| 02:56.37 | p3nguin | Yes, sip handles that. |
| 02:56.51 | spenguin[work] | ok p3nguin |
| 02:56.55 | p3nguin | erm, codec |
| 02:58.36 | p3nguin | If you look in your sip debug, you'll see where the codec is negotiated. |
| 02:59.00 | spenguin[work] | ok, well I want to set a higher quality codec |
| 02:59.06 | spenguin[work] | and ensure itll fallback |
| 02:59.09 | p3nguin | There isn't one. |
| 02:59.11 | spenguin[work] | incase its not supported |
| 02:59.18 | p3nguin | alaw is the best you have. |
| 02:59.22 | spenguin[work] | hrm |
| 03:00.01 | spenguin[work] | well these guys are planning to move all sip calls to over skype |
| 03:00.04 | spenguin[work] | for quality |
| 03:00.13 | spenguin[work] | which I think isnt really a good idea |
| 03:00.16 | p3nguin | I'd fire them. |
| 03:01.33 | coppice | i'd fire anyone saying alaw is the best you can get |
| 03:02.09 | p3nguin | What codec do you think he has that is of better quality than alaw? |
| 03:02.28 | spenguin[work] | by quality I meant, audio - its ok if its a bit bandwight consuming |
| 03:02.34 | coppice | practically any wideband codec |
| 03:02.57 | p3nguin | You would have to buy a wideband codec if you wanted something better than alaw. |
| 03:03.54 | p3nguin | If he says he has a wideband codec, I'll retract my statement. |
| 03:04.15 | coppice | skype is wideband |
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| 03:04.44 | spenguin[work] | so its either skype or any other commercial codec |
| 03:04.51 | spenguin[work] | if thats the right word |
| 03:05.30 | p3nguin | Is your provider giving you a skype channel driver when they change or will you have to buy your own? |
| 03:05.53 | spenguin[work] | p3nguin: we have to buy our own |
| 03:06.02 | p3nguin | I think it costs something like $60 USD per channel. |
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| 03:06.06 | spenguin[work] | yeah |
| 03:06.11 | spenguin[work] | I think its a stupid move |
| 03:07.15 | spenguin[work] | if we did purchase a wideband codec, we just have to ensure our voip provider supports it? |
| 03:07.42 | p3nguin | Pretty much, yes. |
| 03:08.00 | coppice | most people who support wideband support G.722 |
| 03:08.11 | p3nguin | Both ends need to allow the codec in order to be able to use it. |
| 03:08.53 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 03:08.59 | k-man | im having some problems with sip registration |
| 03:09.16 | p3nguin | As I mentioned before, the sip debug will reveal what codecs the other side has available, so you can see what they are supporting and choose one of those. |
| 03:10.01 | spenguin[work] | p3nguin: ok |
| 03:10.16 | k-man | i keep getting Registration for '0xxxxx@sip.mynetfone.com.au@sip.mynetfone.com.au' timed out, trying again (Attempt #6) |
| 03:10.30 | k-man | it was working fine until today |
| 03:10.36 | k-man | could it be the router somehow? |
| 03:12.02 | grungies1138 | For those that care, I figured out my MOH issue. I'd still like it to just use MP3s but I'm happy it works. |
| 03:12.44 | p3nguin | moh happily plays mp3s. |
| 03:16.34 | grungies1138 | caouldn't get it to. |
| 03:17.02 | p3nguin | Did you build asterisk with mp3 support? |
| 03:17.05 | grungies1138 | tried with the mpg123 application set up listed in musiconhold.conf sample stuff. no errors but no audio |
| 03:17.23 | grungies1138 | I didn't compile it. did apt-get install asterisk |
| 03:17.48 | p3nguin | That's weird that it wouldn't have been included. Did you install asterisk-addons, too? |
| 03:18.27 | p3nguin | You also don't need to do anything special in musiconhold.conf to make it play mp3s. |
| 03:18.38 | p3nguin | Here's my mp3 class: |
| 03:18.39 | p3nguin | [mp3s] |
| 03:18.40 | p3nguin | mode=quietmp3 |
| 03:18.40 | p3nguin | directory=/var/lib/asterisk/mohmp3 |
| 03:18.55 | p3nguin | Then I just put some mp3 files in the directory specified. |
| 03:19.04 | p3nguin | That's all there is to it. |
| 03:19.51 | grungies1138 | do you know what the ubuntu package would be for addons? I tried, but couldn't fin dit |
| 03:19.55 | grungies1138 | find it* |
| 03:19.57 | spenguin[work] | coppice: skype uses the silk codec? |
| 03:20.08 | spenguin[work] | havent they open sourced it? |
| 03:20.24 | p3nguin | apt-cache search asterisk |
| 03:20.43 | p3nguin | It should literally be called asterisk-addons. |
| 03:22.03 | grungies1138 | ahh asterisk-mp3 |
| 03:23.18 | k-man | how can i test if a sip register connection is failing due to my router? |
| 03:23.25 | coppice | silk seems to be in limbo. you can get the source, but the attached strings aren't entirely clear |
| 03:25.45 | coppice | there is IETF work now to blend SILK and CELT and other good stuff into a possible new IETF codec |
| 03:26.03 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 03:27.04 | grungies1138 | ok in MOH.conf - [default] mode=quietmp3 directory=/var/lib/asterisk/moh random=yes |
| 03:27.09 | grungies1138 | p no errors and no audio |
| 03:27.34 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 03:29.23 | p3nguin | Do you have ONLY mp3s in that directory? My guess is that you do not. |
| 03:29.39 | p3nguin | There's a good reason to use /var/lib/asterisk/mohmp3. |
| 03:29.40 | grungies1138 | hmm. no. |
| 03:29.46 | grungies1138 | ok let me shift them |
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| 03:38.04 | grungies1138 | O'Doyle rules! |
| 03:38.14 | grungies1138 | and so does p3nguin |
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| 04:04.12 | p3nguin | grungies1138: I take it you have mp3s on hold, now. |
| 04:07.15 | spenguin[work] | coppice: hrm |
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| 05:58.10 | vincems | #trixbox |
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| 05:59.59 | kruemeltee | greets all in the channel |
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| 07:02.19 | nix8n82 | I want to have 3 or 4 asterisk servers on a local network dial through another server on that network that has a public ip and connects to one or more sip providers, all rtp traffic must pass through the server with the public ip and then to the other servers on the network, what software other than asterisk would I need and what would be the right technical name for the server with the ip? |
| 07:06.26 | kaldemar | nix8n82: asterisk can do that. if you don't want asterisk for it, you could probably use a SIP proxy and an RTP proxy. |
| 07:10.17 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:11.44 | nix8n82 | Thanks kaldemar, so if I want the other servers handle recordings,voicemail,ivr's,and maybe other cpu intensive task I just dial the main server and then have that server dial again without answering, because I don't want the other machines to think a call has been answered until my sip provider connects the call to the pstn |
| 07:13.02 | nix8n82 | I would like to stick with asterisk |
| 07:21.54 | *** join/#asterisk dD3 (~gvhuysste@196-215-8-99.dynamic.isadsl.co.za) |
| 07:34.06 | dD3 | hey. i'm completely new to this. how do i connect Yealink Usb phones to the asterisk server ? |
| 07:34.24 | dD3 | some documentation would be appreciated |
| 07:36.36 | drmessano | dD3: You don't |
| 07:37.09 | drmessano | The USB is meant to be used in conjunction with something like Skype |
| 07:37.28 | dD3 | oh ? |
| 07:38.25 | drmessano | ? |
| 07:39.04 | dD3 | so i cant use the usb phones ? |
| 07:39.13 | dD3 | that is connected to a computer |
| 07:39.53 | drmessano | You don't connect the phone to Asterisk. You use it in conjunction with a desktop client, of which, Skype seems to be the most noted on their website |
| 07:39.57 | NiugeS | guys.. i'm also quite new and installed asterisknow.. having a few error messages pop up in freepbx.. Could not reload FOP server, Failed to copy from module agi-bin.. any suggestions.. |
| 07:40.04 | *** join/#asterisk Tim_Toady (~moi@77.49.21.138.dsl.dyn.forthnet.gr) |
| 07:40.15 | NiugeS | I had a search on the net but answers are vary different on solutions |
| 07:40.18 | mianos | can anyone suggest the best codec for an IAX trunk between servers on a LAN? |
| 07:40.21 | drmessano | It looks as though perhaps one or two of the devices may work with OTHER softphones, but they do specifically mention Skype quite a bit |
| 07:40.31 | mianos | ulaw, alaw? or use the g729? |
| 07:40.34 | mianos | not a lot of calls |
| 07:40.42 | mianos | and tons of CPU |
| 07:40.43 | drmessano | mianos: I would stick with ulaw |
| 07:40.57 | mianos | yer, thanks, thought so |
| 07:41.06 | dD3 | drmessano: would i need to get the pbx system then ? |
| 07:41.14 | dD3 | and connect that to pstn network |
| 07:41.29 | drmessano | dD3: PBX System? |
| 07:41.52 | dD3 | ok. i think i'm completely confused |
| 07:43.16 | drmessano | The Yealink *USB* phones work with a COMPUTER... You don't "connect them" to Asterisk in the sense you would a traditional stand alone SIP phone. MOST of the phones seems to mention SKYPE as a target for the phones usage |
| 07:44.35 | drmessano | If some of them work with a SIP SOFTPHONE that will work with Asterisk, you're set.. but the issue is not "How do I connect the phone to asterisk", it's "can I use these phones with SOME SIP softphone that I can then use with Asterisk" |
| 07:45.07 | dD3 | so i need to look for a SIP softphone ? |
| 07:45.36 | drmessano | *That works with the YEALINK PHONE* |
| 07:45.38 | dD3 | oh wait |
| 07:45.45 | dD3 | yes ok |
| 07:45.56 | mianos | <PROTECTED> |
| 07:45.57 | mianos | <PROTECTED> |
| 07:45.57 | mianos | <PROTECTED> |
| 07:45.58 | mianos | <PROTECTED> |
| 07:45.58 | mianos | <PROTECTED> |
| 07:46.56 | dD3 | drmessano: so i get the yealink phone. with a softphone(the pc software) that connects it to the asterisk server. |
| 07:47.25 | drmessano | If you dont have the USB phone, dont bother getting one.. Get a real SIP phone |
| 07:47.58 | dD3 | thanks |
| 07:48.08 | drmessano | A USB phone is only a glorified headset.. Don't waste your time. Get a real phone |
| 07:48.31 | dD3 | what else do i need to create the connection between softphone and the server ? |
| 07:49.14 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-jzzgiumlpyozqpwj) |
| 07:49.17 | kaldemar | NiugeS: try in #asterisknow and #freepbx |
| 07:50.05 | drmessano | dD3: That's it |
| 07:50.06 | dD3 | you see.. i dont need to make outside calls.. only phones that is connected to the server should be able to phone |
| 07:50.19 | dD3 | so only the IP of the server.. and then i'm set ? |
| 07:50.23 | drmessano | dD3: Ok, that's irrelevant |
| 07:50.32 | kaldemar | mianos: if bandwidth is no issue, use alaw or ulaw. if the calls are going out of the LAN and bandwidth is an issue and you want to avoid transcoding, consider using something else. |
| 07:50.42 | drmessano | dD3: The client really has nothing to do with your intended usage |
| 07:51.01 | drmessano | dD3: A phone doesn't care if you're making inside or outside calls |
| 07:51.05 | dD3 | oh ok. |
| 07:51.21 | dD3 | thanks alot drmessano you cleared up lots of things |
| 07:51.23 | dD3 | :) |
| 07:56.23 | dD3 | lol.. sorry. and you can transfer calls.. and block the phone so that you can only dial certain numbers ? |
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| 08:31.51 | mianos | is IAX trunking always GSM? |
| 08:33.53 | coppice | no |
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| 08:39.39 | mianos | oh ok, yer my iax trunk was provisioned from users.conf and was not relaoded |
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| 08:55.08 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 08:56.31 | sawgood | I have a remote extension which will not register to an Asterisk 1.6.2.7 box (the remote extension has a public IP with no firewall concern) ... the Asterisk box has no firewall concern (I have local phones registered to the box) sip set debug from the CLI keeps saying "Bad Auth" |
| 08:56.40 | sawgood | I know for sure 100% the secret is correct on both sides |
| 08:56.46 | sawgood | I've restarted the Asterisk box, etc. |
| 08:56.58 | sawgood | there is no NAT concern here ... |
| 08:57.01 | sawgood | any tips? |
| 08:58.48 | kaldemar | look at sip debug to know what peer it matches. |
| 08:59.39 | sawgood | I have sip set debug on |
| 08:59.43 | sawgood | not sure what you mean? |
| 09:00.27 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 09:00.42 | sawgood | sip set debug peer 1281 (says no IP found) |
| 09:01.37 | ChannelZ | Is it a dynamic peer? |
| 09:01.45 | kaldemar | you can't set sip debug on a peer if it's not registered. either "sip set debug on" or "sip set debug ip <ip>" |
| 09:01.47 | sawgood | The remote phone has a static IP |
| 09:02.04 | sawgood | right ... the remote phone will not register that is the concern |
| 09:02.09 | sawgood | I keep getting "bad auth" |
| 09:02.16 | sawgood | I know for sure the password is dialed in |
| 09:02.18 | ChannelZ | yeah but does asterisk know that? As kaldemar says you can't debug by peer if it hasn't registered |
| 09:02.45 | sawgood | I'm trying to see why it will not register |
| 09:02.53 | sawgood | the secret is simply 11aa11 |
| 09:02.58 | sawgood | very simple secret |
| 09:03.08 | kaldemar | sawgood: asterisk doesn't see the registering phone as you expect it to, that's the issue here. |
| 09:03.31 | sawgood | I understand that part ... |
| 09:03.38 | kaldemar | sawgood: now enable sip debug to see what device definition in sip.conf it matches. |
| 09:03.43 | *** join/#asterisk Gokee2 (~gokee2@24-113-159-168.wavecable.com) |
| 09:04.09 | kaldemar | and pastebin it, if you can't figure it out yourself. |
| 09:05.53 | sawgood | I see the extension in sip.conf ... all is correct |
| 09:06.03 | sawgood | I have the right extension and secret in sip.conf |
| 09:06.26 | sawgood | I see packets come in from the remote phone to the Asterisk box (when sip set debug on) is set |
| 09:06.38 | sawgood | the sip invite request says, "Bad Auth" |
| 09:06.52 | ChannelZ | what does the console say (not the sip debug)? |
| 09:07.22 | ChannelZ | Like 'wrong password', or 'no matching peer found', or 'username mismatch'.... |
| 09:07.24 | sawgood | without sip debug on set at the CLI (if it is off) ... nothing poplulates in the CLI |
| 09:07.31 | ChannelZ | core set verbose 5 |
| 09:07.47 | sawgood | definitely at 5 |
| 09:07.52 | sawgood | core set verbose 5 |
| 09:08.09 | ChannelZ | and it says *nothing*? It should be saying something |
| 09:08.24 | sawgood | nothing (not a single frame of information) comes across the CLI ... |
| 09:08.35 | kaldemar | i'd wait for a pastebin from this point on... |
| 09:09.30 | ChannelZ | did you disable warn/notice in your logger.conf or something for console? |
| 09:10.22 | sawgood | MVLA20*CLI> |
| 09:10.33 | sawgood | nothing ever populates on the screen |
| 09:10.46 | sawgood | just sits there with core set verbose 5 and core set debug 5 |
| 09:10.58 | ChannelZ | hmm I guess you must have then |
| 09:11.30 | ChannelZ | look in /etc/asterisk/logger.conf for a line starting with console => |
| 09:11.53 | sawgood | ;console => notice,warning,error,debug |
| 09:12.04 | ChannelZ | and that's it? there's not one that isn't commented out? |
| 09:12.21 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 09:12.47 | sawgood | four or five instantes of console, and each has a ; at the start of the line |
| 09:13.16 | ChannelZ | ok.. so remove the ; from the head of that one line, save, and then 'reload' on your asterisk. You might want to turn debug off because it's going to vomit up a lot of stuff you won't care about unless you have big problems you're trying to solve. |
| 09:13.40 | ChannelZ | (it's in the [logfiles] section) |
| 09:14.10 | sawgood | ok done ... |
| 09:14.14 | sawgood | core set debug 0 |
| 09:14.18 | sawgood | I'll reboot my phone now |
| 09:14.35 | mianos | back to iax trunking, can I trunk from 1.4 to 1.6? |
| 09:14.56 | ChannelZ | mianos: probably.. I don't think the protocol has changed in any meaningful way |
| 09:14.58 | mianos | i get this: chan_iax2.c:10147 socket_process: Rejected connect attempt from 131.84.1.1, requested/capability 0x8/0xe00c incompatible with our capability 0x703. |
| 09:15.04 | mianos | yet |
| 09:15.14 | mianos | iax2 show peer firewall6iax |
| 09:15.14 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 09:15.15 | mianos | shows: |
| 09:15.21 | mianos | <PROTECTED> |
| 09:15.22 | mianos | <PROTECTED> |
| 09:15.26 | mianos | abd the other end |
| 09:15.34 | mianos | <PROTECTED> |
| 09:15.35 | mianos | <PROTECTED> |
| 09:15.40 | sawgood | [Jun 3 00:36:14] NOTICE[5495]: chan_sip.c:21549 handle_request_register: Registration from '<sip:1281@172.16.150.44>' failed for '173.13.158. 29' - Wrong password |
| 09:15.45 | sawgood | finally something! |
| 09:15.54 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 09:16.31 | mianos | <PROTECTED> |
| 09:17.14 | mianos | they are the next two bits |
| 09:17.30 | mianos | bit 1 is 723 and bit 2 is gsm |
| 09:18.21 | sawgood | wow ... keeps saying 'wrong password" (I have 11aa11) its in sip.conf and the Grandstream phone GUI |
| 09:18.24 | sawgood | very strange |
| 09:18.53 | ChannelZ | are you sure you don't have two peers defined that could match that? Like one with a static IP set in sip.conf |
| 09:19.51 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:20.02 | sawgood | nothing else comes from the network I am on to the Asterisk box ... |
| 09:20.18 | sawgood | its just a lab GXP2010 phone with a live static IP |
| 09:20.43 | sawgood | host=dynamic in sip.conf for this phone, but i've never seen that be a concern |
| 09:20.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 09:21.29 | ChannelZ | pastebin your sip.conf |
| 09:22.38 | mianos | hmm, ok, the global settings mask off the allowed settings in the iax trunk sections of users.conf |
| 09:23.22 | dD3 | what is good hardware products for PSTN networks ? |
| 09:23.43 | sawgood | http://pastebin.org/302518 |
| 09:23.45 | ChannelZ | Digium TDM.. |
| 09:24.00 | mianos | good old TDM400P |
| 09:24.08 | dD3 | thanks :) |
| 09:24.31 | mianos | I used to have a X100 card, shite |
| 09:24.44 | mianos | then used a sipura3000 as an FXS / FXO |
| 09:24.49 | mianos | pretty good |
| 09:24.54 | mianos | then got a TDM400 |
| 09:24.56 | ChannelZ | sawgood: that's the only thing in there? |
| 09:24.56 | mianos | awsome |
| 09:25.19 | sawgood | I have about 4 other working local extensions ... but thats about it |
| 09:25.19 | mianos | don't worry about the echo cancelloe, oslec works better than the hardware |
| 09:27.46 | ChannelZ | well either you maybe have a duplicate which you didn't show me, or perhaps you've got the wrong password typed into the phone.. a space or some other junk character that you're not seeing perhaps |
| 09:28.04 | sawgood | oh good point ... thanks! |
| 09:30.25 | ChannelZ | bed time |
| 09:30.41 | sawgood | me too |
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| 09:42.06 | dD3 | cheap SIP phones ? |
| 09:42.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 09:43.08 | mianos | grandstream? |
| 09:43.55 | mianos | mitel or snoms |
| 09:44.04 | dD3 | ty |
| 09:44.14 | mianos | I have 4 different ones from ebay |
| 09:44.15 | mianos | all work |
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| 09:51.10 | dD3 | with this Digium TMP400P can one connect it to a office telephone network ? |
| 09:51.16 | dD3 | and still transfer calls |
| 09:51.18 | dD3 | etc |
| 09:53.19 | domi | is there a better method of deny outgoing calls than answer with Busy()? ie: exten => _0900X.,1,Busy() |
| 09:53.56 | domi | i also tried SendText() but the client do not display it |
| 09:54.05 | mianos | depends on the transfers |
| 09:54.33 | domi | with SIP-Clients |
| 09:54.40 | mianos | it will do hook flash |
| 09:55.52 | kaldemar | domi: Hangup, Congestion, Playback... depends on what you mean by better. |
| 09:56.44 | domi | hmm... sending a 403? |
| 09:57.07 | domi | or 488 |
| 10:01.02 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 10:04.45 | domi | and how can i use the function keys on a snom phone to show the status of an other extension? |
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| 10:19.02 | dD3 | i can use this with asterisk ? http://www.yealink.com/en/view.asp?ClassLayer=36 |
| 10:29.10 | tzafrir_laptop | dD3, you surely can use it as a low-quality sound card |
| 10:29.36 | tzafrir_laptop | But also look for a kernel module called 'yealink'. I suppose it supports those |
| 10:34.46 | dD3 | hmm. ok.. and one more thing. is it possible to connect a existing telephone network(RJ11) to a TDM400P ? |
| 10:35.09 | *** join/#asterisk ChkDigit (~mike@24.72.71.175) |
| 10:35.11 | dD3 | so that we still can transfer calls etc |
| 10:36.04 | tzafrir_laptop | dD3, you need one with an FXO module |
| 10:36.35 | dD3 | so if it supports it. it will be able to work |
| 10:36.44 | dD3 | this one supports it |
| 10:43.32 | kruemeltee | if I want to use * within a hotel I need to summarize how much a person who uses his telephone has to pay for his calls ... may anybody give me a hint how to realize this issue with asterisk? |
| 10:44.06 | Chainsaw | kruemeltee: Generally you'll want to use CDR (Call Dispatch Records) to record this information. |
| 10:44.26 | Chainsaw | kruemeltee: And then a package like A2Billing can read this CDR information (in a file or a database) and produce bills. |
| 10:45.29 | kruemeltee | oh ... there's already a package, that collects this information? CDR ist already collecting the time and duration of every call ... I just needed to figure out the costs ... |
| 10:45.44 | kruemeltee | thanks a lot for this hint ... I'll try to search for this one ... |
| 10:45.56 | Chainsaw | kruemeltee: A2Billing might be able to do what you want, yes. |
| 10:46.09 | Chainsaw | kruemeltee: Perhaps the others have competing software to suggest as well :) |
| 10:47.13 | kruemeltee | I'll try it at first with a2billings ... hope I'm able to reset the running costs if the person leaves the hotel :-) |
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| 11:08.15 | emora | Name resolution failure causes asterisk to bog down. Can we configure asterisk to not resolve dns names? |
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| 11:48.34 | ujjain | How can I troubleshoot bad telephony at some times from an office in Suriname that uses satellite? |
| 11:48.41 | *** part/#asterisk ming_zym (~ming_zym@116.69.107.241) |
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| 11:57.10 | ujjain | Do people here use Asterisk with 500ms+ latency? |
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| 12:04.55 | Polis_ttt | ujjain: no, 500 isn't so good :) |
| 12:06.11 | ujjain | I know :D |
| 12:06.16 | ujjain | But it' s the best stable you can get |
| 12:06.18 | ujjain | it' s satellite internet |
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| 12:29.38 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 12:35.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:36.21 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 12:37.59 | cusco | whats the most common cause for audio to go only one way? |
| 12:38.17 | [TK]D-Fender | cusco: NAT/firewall |
| 12:38.31 | cusco | what if we are on the same network? |
| 12:38.44 | cusco | well it is a VPN, we can treat it as the same network... |
| 12:38.49 | cusco | (I guess) |
| 12:39.29 | [TK]D-Fender | cusco: Perhaps you should look at the call |
| 12:41.38 | cusco | [TK]D-Fender: http://paste.debian.net/75981/ |
| 12:41.40 | cusco | <--- Reliably Transmitting (no NAT) to 192.168.2.150:5065 ---> |
| 12:43.09 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 12:44.24 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 12:45.03 | cusco | I am also capturing with wireshark at the end that I can hear, but not talk... |
| 12:45.18 | [TK]D-Fender | cusco: You're only looking at one end |
| 12:45.20 | cusco | I don't really know what to look for |
| 12:45.22 | cusco | yes |
| 12:45.32 | [TK]D-Fender | cusco: and look in * CLI, not an external tool |
| 12:45.44 | cusco | ok, Im looking in * cli |
| 12:46.00 | cusco | wireshar was only to check if RTP was set with * or the other softphone |
| 12:46.24 | cusco | ok so this is asterisk box2 and asterisk box3 lol |
| 12:46.36 | cusco | a sofptphone registered on each end |
| 12:48.04 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:48.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:48.25 | cusco | peer information http://paste.debian.net/75983/ |
| 12:49.26 | [TK]D-Fender | cusco: Looks like the same subnet. Why is that? |
| 12:50.05 | cusco | it is the same subnet... |
| 12:50.10 | cusco | shouldn't it be? |
| 12:50.19 | [TK]D-Fender | cusco: Multiple boxes, same LAN? |
| 12:50.23 | cusco | err |
| 12:50.25 | cusco | its a VPN |
| 12:50.36 | cusco | so we are phisically far away |
| 12:50.44 | cusco | but same LAN |
| 12:50.49 | [TK]D-Fender | cusco: That is not normal. Normally you mix multiple different subnets together with direct routing |
| 12:52.01 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 12:52.18 | cusco | err.. Im confused |
| 12:59.13 | cusco | [TK]D-Fender: do you mean the subnet mask? |
| 12:59.51 | [TK]D-Fender | cusco: What are yours? |
| 13:00.23 | [TK]D-Fender | cusco: because you are showing a standard class C which would normally be 255.255.255.0 |
| 13:00.45 | cusco | ok you mean the mask |
| 13:01.02 | [TK]D-Fender | cusco: or 255.255.0.0 though that runs the risk of overlap. Anything else is not standard and not considered healthy |
| 13:02.18 | cusco | [TK]D-Fender: but we are using 255.255.255.0 |
| 13:02.51 | cusco | wich is pretty normal |
| 13:03.06 | [TK]D-Fender | cusco: That makes no sense. How does anything know what is on each side? How do you deal with all the excess broadcast traffic? |
| 13:03.28 | [TK]D-Fender | cusco: Or are you referring to a single endpoint being VPN'd? |
| 13:03.45 | cusco | we don't, we are the same network. so we can route between 10.100.100.0/24 and 192.168.2.0/24 |
| 13:03.55 | [TK]D-Fender | cusco: And not a site-site VPN |
| 13:04.20 | cusco | it is a site-site ip2sec |
| 13:04.25 | cusco | router to router |
| 13:06.33 | cusco | [TK]D-Fender: the networks are not bridged, only routed |
| 13:07.28 | [TK]D-Fender | Hard to route when teh ermote doesn't have their own mask for your gateways to know whose traffic is whose |
| 13:07.39 | [TK]D-Fender | cusco: anyway pastbin a COMLPETE call end to end. |
| 13:08.36 | cusco | ok |
| 13:09.13 | cusco | [TK]D-Fender: when I trace connection to 10.100.100.3 it shos the packet going first to the router (192.168.2.254) |
| 13:09.41 | cusco | its just that box2 has a lot going on (live) and its hard to filter exactly what I need |
| 13:09.45 | cusco | but let me try |
| 13:09.46 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 13:09.52 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:09.55 | [TK]D-Fender | cusco: Your 2 peers for 2 boxes were both 192.168.2.X. This is the first mention of 10.X.X.X |
| 13:10.00 | *** join/#asterisk henk (~henk@213.133.111.59) |
| 13:10.14 | cusco | err... |
| 13:10.19 | [TK]D-Fender | cusco: http://paste.debian.net/75983/ |
| 13:10.21 | [TK]D-Fender | ^^^ |
| 13:10.34 | cusco | let me explain |
| 13:10.42 | [TK]D-Fender | cusco: Either your paste was bad or your description that folloed extremely misleading |
| 13:10.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:10.49 | [TK]D-Fender | followed* |
| 13:11.15 | cusco | box2 is 10.100.100.3 but as I am phisically here, I registered myself (192.168.2.150) at 10.100.100.3 |
| 13:11.39 | cusco | then called a phone registered on box3 (192.168.2.5). |
| 13:11.42 | *** join/#asterisk clintc (~clintc@n128-227-179-127.xlate.ufl.edu) |
| 13:12.01 | cusco | and that phone is 192.168.2.130 |
| 13:12.14 | cusco | is that ok? |
| 13:12.34 | cusco | so I will try to get a log of the entire call |
| 13:12.42 | [TK]D-Fender | cusco: yes |
| 13:12.48 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:14.02 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:14.20 | orn | Whenever I try to use the featurecode attended transfer (as configured in features.conf), I get the playback "Transfer" and a dial tone, but as soon as I enter the first digits, I get "WARNING[11165]: features.c:1436 builtin_atxfer: Did not read data.", which is the same error as occurs in a timeout when no digits are entered. Blind transfer works like a charm. Any ideas? |
| 13:14.46 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 13:15.01 | tuxx- | Hi guys, could anyone tell me what kind of bugs i could encounter if i disabled the break in the following piece of code in apps/app_dial.c ? http://pastebin.org/302940 |
| 13:15.32 | *** join/#asterisk KnickLighter (~meh@node55-fbi-gw.research.nlsecurity.org) |
| 13:15.52 | cusco | [TK]D-Fender: so that is sip debug on the other ent: http://paste.debian.net/75988/ |
| 13:15.57 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 13:15.58 | cusco | and dialplan |
| 13:17.44 | [TK]D-Fender | cusco: again, why am I seeing only ONE subnet here? You said that each then had their own completely different class. |
| 13:18.20 | cusco | dialplan here http://paste.debian.net/75989/ |
| 13:18.43 | cusco | [TK]D-Fender: well like I said both end-phones have the same subnet |
| 13:18.54 | cusco | one of the asterisk box is on another subnet |
| 13:19.15 | henk | subnet != subnet mask |
| 13:19.21 | cusco | yep |
| 13:19.24 | henk | i'm not sure if you are confusing those... |
| 13:19.26 | [TK]D-Fender | cusco: Either your network concept is whacked or your description is. |
| 13:19.37 | cusco | 14:11 < cusco> box2 is 10.100.100.3 but as I am phisically here, I registered myself (192.168.2.150) at 10.100.100.3 |
| 13:19.40 | [TK]D-Fender | henk: I have no confirmation of a different mask. |
| 13:19.41 | cusco | 14:11 < cusco> then called a phone registered on box3 (192.168.2.5). |
| 13:19.54 | cusco | the mast is the same |
| 13:20.10 | cusco | mask |
| 13:20.12 | [TK]D-Fender | cusco: Where do I see box2 traffic with that IP on it? |
| 13:20.35 | henk | [TK]D-Fender: afaict he has the same mask but once for 192.168.2.0 and once for 10.100.100.0. more i have not understood yet 'g' |
| 13:20.38 | cusco | well my peer (699) is registered on it |
| 13:20.48 | cusco | sip.conf from 192.168.2.5 also registers in 10.100.100.3 |
| 13:20.53 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:24.04 | domi | how can i react on the response of Authenticate() without priority jumping? |
| 13:24.32 | [TK]D-Fender | cusco: ok, disable reinvites across the board and retest. I can't see or prove anythign and this descrption is turning into a circular mess where I never actually see IPs from the otther range. |
| 13:25.40 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 13:27.03 | cusco | http://paste.debian.net/75989/ ok I made this up on paint |
| 13:27.07 | cusco | lol |
| 13:27.13 | cusco | (I have no drawing tools here) |
| 13:27.37 | cusco | ok I will disable re-invites (tho the whole point is to make RTP audio go trough endpoints only) |
| 13:29.47 | tuxx- | Hi guys, could anyone tell me what kind of bugs i could encounter if i disabled the break in the following piece of code in apps/app_dial.c ? http://pastebin.org/302940 |
| 13:30.28 | cusco | [TK]D-Fender: audio works like that. let me paste |
| 13:31.06 | cusco | one end: http://paste.debian.net/75993/ |
| 13:31.49 | pabelanger | tuxx-: Why are you commenting it out? |
| 13:32.17 | [TK]D-Fender | cusco: Your phones don't know what is local and things go bad from there. Leave them without reinvites |
| 13:32.21 | cusco | I lost the other end buffer |
| 13:32.38 | cusco | [TK]D-Fender: but the whole point is to have re-invites |
| 13:32.45 | cusco | phones don't know what is local? |
| 13:32.53 | cusco | what do you mean? |
| 13:33.09 | [TK]D-Fender | cusco: Reinvites require them to send their own contact headers. |
| 13:33.22 | cusco | [TK]D-Fender: ok what is wrong with that? |
| 13:33.31 | [TK]D-Fender | cusco: they phones aren't that smart <- |
| 13:33.38 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:33.42 | [TK]D-Fender | cusco: Jsut... don't |
| 13:33.55 | cusco | [TK]D-Fender: here is the big problem |
| 13:34.02 | cusco | so we are office1 and office2 |
| 13:34.13 | cusco | all PRI lines are comming in office1 |
| 13:34.25 | cusco | now we have some PRI as well on office2 |
| 13:34.51 | cusco | and we will take a large amount of calls from that PRI line in office2, so operators on office2 will pick them up |
| 13:35.02 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:35.27 | cusco | but wen they are all busy, office2 will pass the calls down to office1 as well |
| 13:35.31 | cusco | so far so good |
| 13:36.27 | [TK]D-Fender | cusco: If your point is close.... please take aflying leap at it... |
| 13:36.30 | cusco | now our queuing box is here at office1 so incomming calls come trough office2 come in office1 for queuing and then go back to office2 to be picked up |
| 13:36.54 | henk | it's a pity lines on irc must be so short... |
| 13:36.56 | *** join/#asterisk Baylink1 (~jra@cerberus.vicimarketing.com) |
| 13:37.17 | henk | </sarcasm> |
| 13:37.42 | cusco | to save bandwidth we would like to route calls comming in office2, queuing in office1 and picked up in office2, stay inside office2 |
| 13:37.50 | tuxx- | pabelanger: were having some problems with a switchboard, and this seems to fix the problem. |
| 13:37.53 | cusco | maitain the traffice inside office2 |
| 13:37.59 | cusco | not sure if I was clear |
| 13:38.23 | cusco | was I? |
| 13:39.00 | tuxx- | i know its not the best way to fix it, but we need a quick workaround / fix before tomorrow, and was wondering if i comment the break out would that cause any more problems. pabelanger |
| 13:39.24 | [TK]D-Fender | cusco: ugh |
| 13:39.31 | cusco | :( |
| 13:39.45 | cusco | my boss doesn't aprove of having different asterisks for different queues |
| 13:40.24 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:40.52 | cusco | so I thought canreinvite would help |
| 13:41.37 | pabelanger | tuxx-: Its usually best to QA patches before putting them into production. |
| 13:42.11 | [TK]D-Fender | cusco: Your networking is not playing well into this. |
| 13:42.58 | tuxx- | i know pabelanger, problem is, we need it fast xD |
| 13:43.46 | BarthezZ | Hmm, I'm looking for a way to limit the number of active calls going over a SIP uplink, but it consists of 5 registrations... Would it be a smart move to put an astdb counter before and after the Dial commands for the in-and outgoing calls to increment and descend the number? and check if it's too high? |
| 13:44.16 | cusco | [TK]D-Fender: what modifications should I do to my network? |
| 13:44.21 | cusco | have different netmasks ?? |
| 13:45.44 | [TK]D-Fender | BarthezZ: GROUP() + GROUP_COUNT() |
| 13:46.47 | BarthezZ | ${VOIPMAX} is a user-set Global which limits the concurrent number of outbound VOIP calls |
| 13:46.53 | BarthezZ | awsome :) thanks [TK]D-Fender |
| 13:47.01 | pabelanger | tuxx-: then you have your answer. Expect problems. |
| 13:47.46 | tuxx- | tnx for the help pabelanger :) |
| 13:48.04 | cusco | [TK]D-Fender: talking of GROUP() We are using that to limit only one call for each peer but in the dialplan for outbounds there is no group() nor group_count |
| 13:48.19 | cusco | is it ok to put them there as well, and dialplan on inbounds will know about them?? |
| 13:49.05 | [TK]D-Fender | <PROTECTED> |
| 13:49.19 | cusco | lol |
| 13:49.26 | cusco | and what modifications on my network would be needed? |
| 13:51.59 | [TK]D-Fender | cusco: I'm not certain what point breaks this. |
| 13:52.04 | *** join/#asterisk lowtek (~lowtek@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 13:53.51 | BarthezZ | lol Fender, you trekkie! :+ |
| 13:54.59 | [TK]D-Fender | BarthezZ: Star WARS.... </fail> |
| 13:55.23 | BarthezZ | they both have stars? |
| 13:55.33 | BarthezZ | I only watched star trek voyager because 7of9 was hot |
| 13:55.36 | lowtek | Ok, this is cool - http://www.youtube.com/watch?v=7bleA88_muI |
| 13:55.47 | henk | only the wars have yoda! |
| 13:56.40 | [TK]D-Fender | BarthezZ: You mean 38ofD |
| 13:56.48 | Baylink1 | "2 of 38". |
| 13:56.56 | BarthezZ | 2 of D^2 |
| 13:57.07 | Baylink1 | And in fact, I think Jeri Ryan was a 36DD. |
| 13:57.23 | [TK]D-Fender | Baylink1: Borg enhanced ;) |
| 13:57.27 | Baylink1 | Mr Bra says 36D, but I think he's wrong. |
| 13:57.35 | Baylink1 | And I'm a professional at this. |
| 13:57.45 | Baylink1 | No, really; I once went 8 for 9 at a Hooters. |
| 13:57.47 | [TK]D-Fender | FBI? |
| 13:58.08 | *** join/#asterisk af_ (~getsmart@78.134.22.35) |
| 13:58.20 | Baylink1 | Our waitress friend was so surprised that she kept calling over her cow-orkers (a pun which, on reflection, is much more appropriate here than usually). |
| 14:00.08 | [TK]D-Fender | GOT MILK? |
| 14:00.38 | KavanS | relevance? |
| 14:00.40 | BarthezZ | ok... |
| 14:00.41 | [TK]D-Fender | "I'd asy about 2 large D-cups worth!" |
| 14:00.54 | BarthezZ | This is the moment my boss in me is saying "get back to work" |
| 14:01.21 | *** join/#asterisk lirakis (~irssi@ool-ad024dab.dyn.optonline.net) |
| 14:01.22 | KavanS | BarthezZ, agreed - shall we start announcing the size of our male members? |
| 14:01.41 | BarthezZ | please don't, you would lie anyway |
| 14:02.07 | KavanS | BarthezZ, was making a point ;) |
| 14:02.21 | BarthezZ | just lay it on the table and keep it there |
| 14:02.43 | cusco | [TK]D-Fender: 10.100.100 |
| 14:02.55 | Baylink1 | No, really; I hit on 8 of 9, and on the 9th one, the girl was actually not certain she was wearing the proper size. :-) |
| 14:03.02 | Baylink1 | "What do you call that?" |
| 14:03.12 | Baylink1 | "It's something like a man's penis, only smaller." |
| 14:04.46 | [TK]D-Fender | "For a long time - in fact, from the beginning if my memory serves correctly - Skype has "promised" that Skype-to-Skype calls will "always be free". Well, it turns out that promise is worth exactly as much as any other promise Skype has ever made, or will ever make for that matter. A big, fat nothing." |
| 14:04.59 | [TK]D-Fender | "Skype calls made on the iPhone via 3G will be "free" only until the end of this year. They originally tried to make the cutoff even earlier - August 2010 - but have apparently retreated to a later date as a result of a huge protest from Skype users. They don't mention it specifically, but you can bet your last dollar that the same "Whoops, no longer free" policy will apply to other... |
| 14:05.00 | [TK]D-Fender | ...mobile/3G versions." |
| 14:05.02 | [TK]D-Fender | \o/ |
| 14:08.24 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:09.38 | lowtek | No comments on my iPhone Asterisk Manager app? http://www.youtube.com/watch?v=7bleA88_muI |
| 14:10.13 | pabelanger | lowtek: Is it open source? |
| 14:11.07 | *** join/#asterisk centoslinux (~centoslin@212.17.132.238) |
| 14:13.39 | Baylink1 | [TK]D-Fender: Not Skype's fault: *every carrier*'s data plan ToS says that you can't make phone calls over it -- though I expect Verizon's LTE700 to differ when it rolls out, as the native telephony app will be VoIP; LTE is a data only air-interface -- so I've been surprised that they allowed this at all, much less encouraged it. |
| 14:13.44 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:13.51 | Baylink1 | Carriers Do Not Want flat-rate telephony, unless they're getting the money. |
| 14:15.37 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:16.22 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:17.14 | BarthezZ | [Jun 3 16:17:24] WARNING[14948]: pbx.c:1854 pbx_extension_helper: No application 'Group' for extension (macro-dialOut, s, 4) |
| 14:17.17 | BarthezZ | hmm, |
| 14:19.44 | [TK]D-Fender | BarthezZ: FUNCTION <--------- |
| 14:19.51 | [TK]D-Fender | BarthezZ: pay attention |
| 14:20.53 | BarthezZ | oh ofcourse |
| 14:21.01 | BarthezZ | damn... I need vacation |
| 14:22.18 | *** join/#asterisk dajhorn (~dajhorn@adsl-71-158-161-171.dsl.rcsntx.sbcglobal.net) |
| 14:23.11 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:23.48 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca) |
| 14:24.21 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 14:24.45 | cusco | [TK]D-Fender: so if we changed 10.100.100.0/24 to 192.168.3.0/24 - would that make any difference? |
| 14:25.18 | cusco | I am trying to understand what kind of network set up would work |
| 14:25.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:25.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.38 | kruemeltee | say goodbye for the rest of the day :-) |
| 14:27.17 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:32.30 | *** join/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 14:32.39 | hurdman | hello folks ! |
| 14:32.55 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 14:33.42 | hurdman | if i make a gosub to a subroutine sub-A from an extension, and a goto from my sub-A to a sub-B, if i make a return into sub-B, i return back to my extension ? |
| 14:34.13 | p3nguin | That's what Gosub() does. |
| 14:35.04 | p3nguin | Except that call processing is based on a TO extension and not a FROM extension. |
| 14:37.27 | hurdman | so into sub-B , it should not know where i'm from ? |
| 14:38.25 | p3nguin | sub-B should know that you came from sub-A, so it would return to the sub-A dialplan. |
| 14:38.32 | hurdman | ok |
| 14:38.43 | p3nguin | That's what Return() is for. |
| 14:40.13 | hurdman | mmmh, thanks, i'll think hard my dialplan |
| 14:40.16 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
| 14:40.46 | sarthor | Hi, I need Help, Can we use normal pci modam as a FXO card in Tribox? Will it work, I am not so can not understand this link " http://www.voip-info.org/tiki-index.php?page=X100P+clone " very well. HELP |
| 14:41.29 | [TK]D-Fender | sarthor: No |
| 14:42.00 | [TK]D-Fender | sarthor: Only a few select chipsets are supoprted as X100P's, and even those SUCK |
| 14:42.46 | sarthor | [TK]D-Fender, I can not get FXO card here in my city, (i am in Jeddah Saudi Arabia), what shuld i do? Any idea? |
| 14:43.10 | sarthor | and also in my given link , one modem card is mentioned of motorolla. i have same card here. . |
| 14:44.33 | [TK]D-Fender | sarthor: Does Zaptel/DAHDI see it as an X100P? |
| 14:45.04 | [TK]D-Fender | sarthor: pastebint he output of "dmesg" |
| 14:45.06 | [TK]D-Fender | ~pb |
| 14:45.07 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:45.08 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 14:45.44 | sarthor | [TK]D-Fender, ] my machine is not yet installed, i am just working on preparation. |
| 14:46.04 | [TK]D-Fender | sarthor: You won't haev an answer until you do. |
| 14:46.30 | *** join/#asterisk Kyosh (~whoa@pool-71-125-3-247.nycmny.fios.verizon.net) |
| 14:46.37 | sarthor | [TK]D-Fender, hmm. i am gong to get a RAM for the new PC , and start installaton of tribox. |
| 14:46.39 | Kyosh | ack |
| 14:46.41 | Kyosh | does asterisk maintain a log file other than /var/log/asterisk/messages where sip requests (such as authentication) may be logged? otherwise is there a config param i need to specify in asterisk to store more concise information? |
| 14:46.57 | Kyosh | sorry if anyone answered, i was disconnected :( |
| 14:47.17 | sarthor | [TK]D-Fender, i will be back here, hope you not forget me to keep continue you help, if you were free. thank you. |
| 14:47.20 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 14:51.16 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:04.54 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:08.29 | *** join/#asterisk jhirley (~jhirley@adsl-4-130-148.mia.bellsouth.net) |
| 15:08.50 | *** join/#asterisk grante (~PuPic@64.235.109.58) |
| 15:08.53 | jhirley | o/ |
| 15:09.10 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-36-Atlanta.hfc.comcastbusiness.net) |
| 15:12.31 | *** join/#asterisk fourjahn (~Charmion@c-98-231-6-152.hsd1.fl.comcast.net) |
| 15:13.24 | fourjahn | Hello. Can someone point me in the right direction documentation-wise for overcoming registration timeouts? |
| 15:13.51 | fourjahn | I'm currently able to download the initial configuration to our Polycom 601 phones. However, it times out during registration. |
| 15:14.03 | *** join/#asterisk x303 (~x303@20.226.118.70.cfl.res.rr.com) |
| 15:16.20 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 15:18.12 | *** join/#asterisk centoslinux (~centoslin@212.17.132.238) |
| 15:21.31 | ChannelZ | wrong address? firewalled off? |
| 15:22.02 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 15:22.18 | Carlos_Tico | Hi |
| 15:22.45 | centoslinux | Wow, University of Oslo chose Asterisk & OpenSER |
| 15:22.47 | centoslinux | :) |
| 15:22.50 | grante | forjahn: does asterisk show the phone even trying to connect? usually that means it can't connect to the asterisk server - bad config, no dns, firewall, etc. |
| 15:24.02 | Carlos_Tico | OpenSER ? |
| 15:24.22 | centoslinux | Kamalio |
| 15:27.03 | fourjahn | grante: no it doesn't show the phone even trying to connect |
| 15:27.32 | jhirley | fourjahn: is it downloading the config from the tftp server ? |
| 15:28.00 | jhirley | what does the tftp log show, for that matter is it even getting an IP Address ? |
| 15:28.06 | grante | fourjhan: then something is wrong wtih the phone's config or network. maybe a typo in the server address? Or try using the IP address instead of hostname. |
| 15:28.22 | fourjahn | jhirley: yes it downloads the config from the server |
| 15:28.33 | fourjahn | and we're using ftp not tftp for the polycoms |
| 15:29.03 | fourjahn | grante: thanks ... i've only been using the ip address of the server. |
| 15:29.10 | *** join/#asterisk dlublink (~david@76-10-132-241.dsl.teksavvy.com) |
| 15:29.16 | jhirley | is this problem limited to just that 1 phone or all other phones ? |
| 15:29.18 | fourjahn | problem is i'm new to this company so i'm just figuring out their topography |
| 15:29.28 | fourjahn | jhirley: when i try to register my softphone, i get 408 |
| 15:29.44 | dlublink | Is it possible to set variables on individual call legs instead of on channels? This would be really useful for dynamic features. |
| 15:30.04 | p3nguin | dlublink: channels are call legs. |
| 15:30.08 | dlublink | ok |
| 15:30.13 | jhirley | is post 5060 between you and the * box open ? is it a firewall issue ? |
| 15:30.23 | jhirley | port 5060* |
| 15:30.35 | fourjahn | jhirley: i've removed every point of firewall i can think of |
| 15:30.43 | fourjahn | here's the setup |
| 15:31.14 | fourjahn | unless the POE switch is causing an issue |
| 15:31.25 | fourjahn | but this usually isn't the case as it should be 'dumb' |
| 15:31.29 | dlublink | When using the dial command, can I set a variable on the legs created by the dial command? Can I create different values for each leg that is created? Can the leg return variables back to the channel that called the dial command ? |
| 15:32.00 | fourjahn | Phones ==> POE Switch/Patch Panel ==> Linksys WRT110 ==> Asterisk |
| 15:32.11 | fourjahn | We kept the Avaya POE switches |
| 15:32.28 | fourjahn | There is another PBX on the network. Perhaps that's causing an issue? |
| 15:32.39 | fourjahn | We have an Avaya IP400 still in use. |
| 15:32.59 | fourjahn | I could build a Vyatta router to segment the network until if this is the case. |
| 15:33.33 | grante | forjahn: is it feasible to connect a computer to the phone's network cable and make sure you can ping the asterisk server? |
| 15:33.47 | p3nguin | dlublink: You can set variables on each channel, but I'm not sure what happens to the variables once the call is bridged. With hope, someone else will jump in here and enlighten us. |
| 15:33.50 | grante | fourjahn: and see if your softphone at least tries to connect |
| 15:34.04 | fourjahn | grante: i can but the phone still is able to dwonload the cfg settings |
| 15:34.21 | fourjahn | for instance, if i completely 'format' the phone |
| 15:34.33 | fourjahn | and simply enter the FTP IP and login credentials, it is still able to download |
| 15:34.37 | grante | fourjahn: that would at least eliminate switch/firewall concerns. it might be able to connect via ftp but not sip. |
| 15:34.44 | fourjahn | right |
| 15:34.59 | fourjahn | grante: are you referring to the network segementation? |
| 15:36.44 | grante | fourjahn: any other phones on that switch? can they connect? |
| 15:39.30 | Carlos_Tico | guys in need to test a sip trunk is there any number that you have that streams music or something ? |
| 15:39.51 | lordvadr | If I execute a Goto to a different context in a realtime dialplan, does that same realtime dialplan get looked at first for the context or does it breakout to the main dialplan at which point I have to send it back to the "switch => Realtime/..."? |
| 15:40.26 | fourjahn | grante: the other phones can connect but they are registering to the Avaya IP400 that is also on the network with the Asterisk box |
| 15:42.26 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 15:49.36 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:55.59 | *** join/#asterisk proute (~AnthonyCB@mail.sysun-technologies.com) |
| 15:57.24 | proute | Hello all. I use asterisk 1.4.29 with Aastra DECT 620D. Sometimes * crash randomly... I update * to 1.4.32 and I have the same issue. I meet this problem only with these DECT. I have about 300 others * (same release) and I have no problem. |
| 15:57.44 | proute | So someone have already got a problem with aastra dect ? |
| 15:57.49 | proute | thanks for your help |
| 15:57.59 | proute | to infoirmation I have this: segfault at 84 ip b7f44130 sp b6283548 error 4 in libpthread-2.7.so[b7f3c000+15000] |
| 15:59.46 | Kyosh | does asterisk maintain a log file other than /var/log/asterisk/messages where sip requests (such as authentication) may be logged? otherwise is there a config param i need to specify in asterisk to store more concise information? |
| 16:03.38 | *** join/#asterisk mweichert (~mweichert@216.16.254.34) |
| 16:04.02 | mweichert | is their a recommendation NOT to install Asterisk in an OpenVZ container? |
| 16:04.05 | *** join/#asterisk fourjahn (~Charmion@c-98-231-6-152.hsd1.fl.comcast.net) |
| 16:04.08 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:04.47 | fourjahn | grante: if you're still around, the phones aren't registering still. i haven't segmented the network yet. that's my next step. |
| 16:04.50 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:06.15 | pabelanger | proute: doc/backtrace for information about segfaults. |
| 16:06.28 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:08.45 | jhirley | anyone having follow me issues with 1.6.2 ? |
| 16:10.29 | jhirley | ~pastebin |
| 16:10.30 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:11.50 | proute | pabelanger: thanks, I will track the crash and I will come back here... |
| 16:11.54 | grungies1138 | Can Conferences (MeetMe) be the same as extension numbers? |
| 16:12.10 | KavanS | yes |
| 16:12.26 | grungies1138 | So If my extension is 208 I can assign a conference of 208 to that user? |
| 16:14.09 | idespinner | no |
| 16:14.21 | KavanS | grungies1138, what?! |
| 16:14.28 | KavanS | grungies1138, have you even used asterisk? :) |
| 16:14.31 | idespinner | unless your referring to 'meetme' pin numbers.... |
| 16:14.41 | grungies1138 | no KavanS I' |
| 16:14.48 | grungies1138 | I'm new and making sure. |
| 16:14.51 | KavanS | ahh cool |
| 16:14.55 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 16:15.00 | KavanS | yeah you can get asterisk setup pretty quickly to test and determine it's features |
| 16:15.21 | grungies1138 | yeah. I've been going through one feature at a time and setting things up to practice |
| 16:16.38 | grungies1138 | I'm reading through the meetme.conf and it's mentioning scheduling. Is it possible to schedule an actual time and/or date range for when the room is built and torn down? |
| 16:19.26 | *** join/#asterisk houms (~houms@wsip-70-167-244-115.dc.dc.cox.net) |
| 16:19.41 | houms | is intercom possible on the aastra 9133i? I can do it manually by pressing *80+ extension, but I would like to know if you can program the key like on the 6753 or 6757 series? |
| 16:19.56 | raden_work | houms, yes u can |
| 16:20.46 | *** join/#asterisk nny_1 (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 16:20.59 | houms | any idea how to achieve it? not sure what I am missing? in the 6757 the config has |
| 16:21.02 | houms | sip intercom type |
| 16:21.10 | houms | sip intercom prefix code |
| 16:21.14 | houms | sip intercom line |
| 16:21.40 | houms | <PROTECTED> |
| 16:21.45 | nny_1 | have a dial plan doing: exten => _xxxxxxx,3,Dial(DAHDI/g1/${EXTEN},,WrK) and it outputs in console as Executing [6842002@sip:3] Dial("SIP/100-00000015", "DAHDI/g1/6842002||WrK") in new stack but ZIZ gte an error WARNING[6984]: chan_dahdi.c:2289 dahdi_call: Unable to start channel: No data available any advice? |
| 16:21.58 | *** join/#asterisk Z_God (~julius@wlan235109.mobiel.utwente.nl) |
| 16:22.03 | houms | but trying this on the 9133i does not seem to work |
| 16:22.04 | nny_1 | but I get an error* |
| 16:22.15 | houms | raden any pointers you can give? |
| 16:22.44 | raden_work | houms, its in the PDF manual for the phone |
| 16:23.50 | houms | I am looking at the pdf but it seems to only refer to the 480 series for intercom setup |
| 16:24.05 | raden_work | OMG page 21 |
| 16:24.06 | houms | it seems to say that it is not an option on the 9133 |
| 16:24.13 | nny_1 | rofl found one damn google link about no data available, and it's an IRC log of [TK]D-Fender telling someone how useless TDMOE is... http://ibot.rikers.org/%23asterisk/20081119.html.gz |
| 16:24.22 | orn | Is it not possible to execute the command SIPAddHeader from AGI? |
| 16:24.26 | houms | admin guide? |
| 16:24.43 | raden_work | http://www.aastra.com/cps/rde/xbcr/04/9133i_41-000113-00-08_ma_en_06.pdf |
| 16:24.59 | raden_work | houms, sorry for being short boss up my ass |
| 16:25.16 | nny_1 | says "chan_dahdi asked DAHDi to do something with hook start... and it failed" from IRC log.. |
| 16:25.32 | raden_work | he can sit at his computer and master bate all day while the rest of us try to keep this place afloat . |
| 16:25.59 | grungies1138 | Sounds like a sweet gig |
| 16:26.33 | KavanS | lol |
| 16:26.40 | KavanS | at least someone's getting something done! |
| 16:26.41 | houms | lol, fair enough, no worries I understand. even a simple response is greatly appreciated on my end |
| 16:27.10 | grungies1138 | At least he's got his finger on the pulse of the company! LOL |
| 16:27.11 | houms | page 21 just basically says you can setup the auto answer feature. It does not go any further about how to program a key for it |
| 16:27.20 | houms | grungies, LOL |
| 16:27.48 | houms | not the only thing he has his fingers on... |
| 16:27.52 | grungies1138 | Houms: is there a feature to program like a Dial button? |
| 16:28.28 | houms | what do you mean a feature to program? |
| 16:28.32 | houms | its for intercom |
| 16:28.32 | nny_1 | this IRC log of jaytee and [TK]D-Fender is full of hilarity |
| 16:28.46 | grungies1138 | Can you program a button to just dial digits? |
| 16:28.54 | orn | What's the difference between the return code 200=0 and 200=1 in AGI? |
| 16:29.05 | houms | i have tried but if i program a speed dial for example it fails because it is just *80 |
| 16:29.16 | houms | it doesn't know it supposed to wait for the actual extension |
| 16:29.56 | grungies1138 | not sure. |
| 16:30.01 | houms | I was thinking using spre and then program that for *80 and then hit spre button and extension but that did not seem to work either |
| 16:30.08 | houms | though I am not sure on spre syntac |
| 16:30.14 | houms | syntax8 |
| 16:30.26 | houms | is it spre or sprecode? and is this doable? |
| 16:31.17 | jaytee | nny_1, what log? |
| 16:37.40 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:37.53 | *** join/#asterisk hfb (~hfb@pool-98-112-109-195.lsanca.dsl-w.verizon.net) |
| 16:38.14 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 16:41.14 | nny_1 | jaytee: http://ibot.rikers.org/%23asterisk/20081119.html.gz |
| 16:41.27 | nny_1 | jaytee: i have an error i am trying to figure out |
| 16:41.39 | nny_1 | : have a dial plan doing: exten => _xxxxxxx,3,Dial(DAHDI/g1/${EXTEN},,WrK) and it outputs in console as Executing [6842002@sip:3] Dial("SIP/100-00000015", "DAHDI/g1/6842002||WrK") in new stack but ZIZ gte an error WARNING[6984]: chan_dahdi.c:2289 dahdi_call: Unable to start channel: No data available any advice? |
| 16:41.58 | nny_1 | if i dial in to the channel it works fine |
| 16:45.32 | jesselang | How can I tell which end of the call is hung up? Using AMI events would be nice. |
| 16:45.43 | grungies1138 | MixMonitor() is basically call recording? |
| 16:45.54 | jesselang | grungies1138, yes. |
| 16:46.08 | grungies1138 | gotcha |
| 16:47.25 | Naikrovek | is looking at prices for new steelcase desks. result: OMFG |
| 16:48.49 | [TK]D-Fender | nny_1: that's a PRI isn't it? |
| 16:50.40 | grungies1138 | So Monitor() and MixMonitor() basically record each sode of the call seperately and either do or do not mix the recordings after? |
| 16:51.24 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 16:52.46 | grungies1138 | side* |
| 16:53.15 | [TK]D-Fender | grungies1138: MixMonitor records pre-mixed. Monitor starts separate and can merge after |
| 16:53.41 | grungies1138 | gotcha. |
| 16:54.05 | carrar | funny it says that right in the explaination of the command |
| 16:54.12 | carrar | heh |
| 16:54.23 | grungies1138 | funny that I want to make sure I /understand/ |
| 16:54.37 | carrar | might ask again to be sure |
| 16:54.45 | nny_1 | [TK]D-Fender: nah just POTS |
| 16:55.48 | nny_1 | [TK]D-Fender: TDM400P |
| 16:55.50 | [TK]D-Fender | nny_1: :/ pastebin your configs |
| 16:57.56 | grungies1138 | What I think is funny is when someone is so full of themselves that they begrudge someone who attempt to learn something that they already know. |
| 16:58.22 | KavanS | grungies1138, maybe you could try out what you are attempting to learn at the same time as asking questions? |
| 16:58.34 | nny_1 | [TK]D-Fender: do you want all of extensions.conf or just the dial plan part that dials out? |
| 16:58.45 | *** join/#asterisk hajkym (hajkym@sion.ihrisko.org) |
| 16:58.53 | hajkym | hi |
| 16:59.01 | KavanS | grungies1138, assuming that someone will give you a) software and b) support for free - somehow implies that you think that computer expertise is "like picking fruit off a tree" |
| 16:59.23 | KavanS | grungies1138, IRC = you get what you pay for...don't expect enterprise support from IRC ;) |
| 16:59.32 | KavanS | customer service in IRC - well that's another subject in itself! :) |
| 16:59.39 | [TK]D-Fender | nny_1: DAHDI only |
| 16:59.40 | hajkym | i have little problem i have establishe call between 2 UA.. and about few second these UA get BYE |
| 16:59.44 | hajkym | here is log from asterisk |
| 16:59.47 | hajkym | http://pastebin.org/303678 |
| 17:00.08 | hajkym | Scheduling destruction of SIP dialog '442b09431eadfe393287c75c36d6f8f2@10.100.0.251' in 32000 ms (Method: INVITE) |
| 17:00.15 | hajkym | what does mean... |
| 17:00.29 | nny_1 | [TK]D-Fender: ok |
| 17:00.37 | grungies1138 | at the same time, I asked a simple question. it required a simple answer. I asked for no support on MixMonitor() or anything. Yet I get berated? |
| 17:00.43 | *** join/#asterisk flapjacks (~flapjacks@wsip-72-214-208-206.ph.ph.cox.net) |
| 17:00.50 | nny_1 | [TK]D-Fender: http://pastebin.org/303686 |
| 17:00.51 | KavanS | lol grungies1138 it is IRC - this is to be expected |
| 17:00.55 | [TK]D-Fender | hajkym: You are showing us as of the start of the "h" exten... this is TOO LATE |
| 17:01.02 | [TK]D-Fender | hajkym: the call dies PRIOR to that |
| 17:01.12 | KavanS | grungies1138, I think if you are expecting people to treat you with some definition of respect - is a bit much :P |
| 17:01.15 | grungies1138 | KavanS: That is an excuse. |
| 17:01.28 | nny_1 | [TK]D-Fender: hmm chan_dahdi.conf seems wrong |
| 17:01.28 | [TK]D-Fender | grungies1138: Let it go... |
| 17:01.41 | nny_1 | [TK]D-Fender: er nm |
| 17:03.43 | grungies1138 | If we were in person, I'm pretty sure you wouldn't talk to me like that. |
| 17:03.46 | nny_1 | [TK]D-Fender: if you need anything else lemme know |
| 17:04.16 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 17:04.36 | p3nguin | grungies1138: What makes you so confident in that statement? |
| 17:04.41 | KavanS | yeah agreed |
| 17:05.01 | KavanS | internet hardasses are quite common these days... |
| 17:05.18 | KavanS | the only thing I was saying is - be patient, google and "experiment" before asking questions |
| 17:05.23 | KavanS | people will treat you with more respect |
| 17:05.30 | grungies1138 | because I would beat his ass |
| 17:05.47 | p3nguin | Maybe he would beat yours, too. |
| 17:06.00 | KavanS | instead of talking trash and acting like you are the Arnold Schwarzenegger of #asterisk |
| 17:06.41 | *** join/#asterisk centoslinux (~centoslin@212.17.132.238) |
| 17:06.52 | *** join/#asterisk ariel_ (~chatzilla@173-105-11-51.pools.spcsdns.net) |
| 17:08.14 | mtryfoss | is it normal to get these messages on a heavily used system: chan_dahdi.c: !! Got reject for frame 48, retransmitting frame 48 now, updating n_r! ? |
| 17:08.43 | citywok | while you guys are talking about mixmonitor, has anybody ever heard of mixmonitor recordings being off by around .7s? we've found that the two legs of the call are overlapping by .7s |
| 17:09.50 | nny_1 | how i envision this going down http://www.youtube.com/watch?v=cWk6RgQbPVc |
| 17:09.58 | nny_1 | <3 |
| 17:10.04 | [TK]D-Fender | nny_1: what card? |
| 17:10.08 | KavanS | nny_1, lol |
| 17:10.56 | nny_1 | [TK]D-Fender: [TK]D-Fender tdm01b |
| 17:11.47 | [TK]D-Fender | nny_1: Got another module? This one may be dead. |
| 17:11.51 | nny_1 | KavanS: also: http://www.youtube.com/watch?v=Jpoki4wBwtA |
| 17:12.03 | nny_1 | you can be peter griffon |
| 17:12.36 | nny_1 | [TK]D-Fender: i can try another. It's odd though, (i had mentioned this before, but not in our conversation) incoming calls work |
| 17:12.38 | KavanS | lol I think I'll pass |
| 17:12.41 | nny_1 | lol |
| 17:13.30 | jesselang | citywok, you mean the calls aren't mixed out of sync? |
| 17:13.43 | fourjahn | Has anyone experienced registration timeouts behind NAT? |
| 17:13.50 | citywok | yea, exactly. |
| 17:14.06 | [TK]D-Fender | grungies1138: Cool it. |
| 17:14.07 | jesselang | citywok, I haven't seen that before. Sorry. |
| 17:14.19 | [TK]D-Fender | KavanS: And stop antagonizing him |
| 17:14.27 | jesselang | fourjahn, are you using qualify=yes? |
| 17:14.47 | KavanS | understood |
| 17:14.53 | fourjahn | jesselang: yes |
| 17:15.14 | fourjahn | The phones we're using are Polycom 601. The Avaya PBX being behind the same NAT wouldn't cause an issue would it? I can't imagine .. |
| 17:16.16 | nny_1 | [TK]D-Fender: do you think a module can be bad but still accept incoming calls? |
| 17:16.44 | jesselang | fourjahn, what protocols are in use with Asterisk? And with Avaya PBX? |
| 17:17.11 | [TK]D-Fender | nny_1: Little grey on... |
| 17:17.21 | fourjahn | jesselang: sip for asterisk and h323 for Avaya |
| 17:18.59 | fourjahn | jesselang: i was goign to try and segment the network with a vyatta router but the system i threw together has a bad NIC |
| 17:19.18 | fourjahn | jesselang: so now i need to find a cheap router lying around the office to see if its truly something on the network |
| 17:19.21 | nny_1 | [TK]D-Fender: i should tell you more details, i updated dahdi and asterisk today to 1.4.32 from 1.4.24 and dahdi from 2.1.0.2 to 2.3 |
| 17:19.30 | fourjahn | jesselang: but the softphones won't reg either so |
| 17:19.31 | jesselang | fourjahn, are the phones behind NAT? Or is it Asterisk and Avaya behind NAT? |
| 17:19.39 | fourjahn | jesselang: both |
| 17:19.47 | nny_1 | [TK]D-Fender: and this error only shows up in logs after the update it seems (i just discovered this) |
| 17:20.12 | fourjahn | jesselang: i can get the asterisk box on a public ip, i just need to call the isp and request a block of IPs |
| 17:20.17 | fourjahn | but that might take a while |
| 17:20.21 | jesselang | fourjahn, so basically: phones <-> NAT <-> cloud <-> NAT <-> asterisk/avaya |
| 17:21.01 | fourjahn | jesselang: actually, i just started here ... so let me try and break down how their telco closet is setup |
| 17:21.29 | fourjahn | jesselang: ISP (NAT'd .. needs to be converted to bridge mode at some point) -<-. |
| 17:22.14 | fourjahn | jesselang: ISP <-> Linksys WRT100 <-> PoE Switches |
| 17:22.30 | nny_1 | is that double NAT? |
| 17:22.42 | fourjahn | Behind the POE switches and patch panel are the Avaya PBX and Asterisk |
| 17:23.03 | fourjahn | nny: appears to be so .. their old technician never told their ISP to change their modem over to bridge mode |
| 17:23.06 | nny_1 | ISP (NAT) <-> LINKSYS WRT100 (NAT) <-> LAN |
| 17:23.17 | fourjahn | nny_1: yes |
| 17:23.31 | jesselang | fourjahn, yeah, SIP won't play nice with that. |
| 17:23.31 | nny_1 | ouch, yeah our ISP does that by default it's annoying |
| 17:23.49 | jesselang | fourjahn, you need to get that squared away, first thing. |
| 17:23.56 | nny_1 | is it DSL? |
| 17:24.05 | fourjahn | alright .. give me 15 mins i'll call |
| 17:24.08 | fourjahn | nny_1: cable |
| 17:24.55 | hajkym | [TK]D-Fender: and by this log you can me tell what is wrong? |
| 17:24.56 | hajkym | http://pastebin.org/303762 |
| 17:25.05 | nny_1 | should be easy to add another device to the modem via switch (and tell them you need to do so) and put a public NIc on *, will work with remote SIP behind NAT at that point. But you will need to do some setup in asterisk too |
| 17:27.58 | fourjahn | nny_1: i installed the Asterisk SIP module |
| 17:28.30 | nny_1 | fourjahn: one sec, i suck with irc bot commands |
| 17:28.33 | nny_1 | ~sipnat |
| 17:28.34 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:28.40 | nny_1 | hooray for zoidberg! |
| 17:29.18 | nny_1 | actually for asterisk just put nat=yes in sip.conf under the remote peer entry |
| 17:29.27 | nny_1 | i don't think you need to do anything else if the asterisk box has a public IP |
| 17:29.45 | *** join/#asterisk MiserySoft (~elende@94.197.35.167.threembb.co.uk) |
| 17:30.29 | [TK]D-Fender | hajkym: Nope. Describe the setting |
| 17:33.29 | hajkym | i can't...it's too dificult...i don't know how is asterisk set...ok thx..i try make copy of asterisk settings and home try simulate situation |
| 17:43.07 | *** join/#asterisk tripps (~sean@174-148-87-88.pools.spcsdns.net) |
| 17:43.28 | *** join/#asterisk patrick^ (~patrick_@hq.clearcable.ca) |
| 17:43.53 | nny_1 | [TK]D-Fender: sorry didn't get a chance to finish the conversation, did you suggest I order another module in spite of the other circumstances (post update, works on incoming) |
| 17:44.05 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 17:44.29 | [TK]D-Fender | nny_1: Actually first chaqnge the modules POSITION on the card. Could be the card base, not just the module. That si a "freebie" |
| 17:44.36 | nny_1 | indeed |
| 17:45.08 | nny_1 | will try that today. May be easier to get them to order a new card with HWEC anyways, this one has been in service for 3 years or so |
| 17:47.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
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| 17:50.15 | fourjahn | nny_1: okay the cable cannot bridge without a dedicated IP which takes 3 days to provision |
| 17:50.18 | lirakis | random question ... any one know if you can dial non-numeric contacts on a gxp (2020) handset? |
| 17:50.25 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:50.35 | lirakis | as speed dial or from the keypad? |
| 17:50.57 | [TK]D-Fender | fourjahn: VPN or proxy |
| 17:51.03 | nny_1 | fourjahn: sounds about right |
| 17:51.14 | fourjahn | nny_1: so in the meantime, i should be able to plug the asterisk server into one of the ports on the cable modem/router and forward the appropriate ports |
| 17:51.15 | nny_1 | fourjahn: what [TK]D-Fendersaid |
| 17:51.26 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 17:51.37 | nny_1 | fourjahn: see |
| 17:51.40 | nny_1 | ~sipnat |
| 17:51.41 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 18:15.07 | tzafrir_laptop | wonders if there's any chance voip-info.org will be replaced with mediawiki or whatever |
| 18:15.26 | tzafrir_laptop | One that allows simply reverting bad commits |
| 18:16.08 | *** part/#asterisk KnickLighter (~meh@node55-fbi-gw.research.nlsecurity.org) |
| 18:16.47 | [TK]D-Fender | tzafrir_laptop: TikiWiki lets you revert to any version really easy |
| 18:17.04 | tzafrir_laptop | [TK]D-Fender, so I managed to miss that |
| 18:18.05 | tzafrir_laptop | Specifically, reverting richboy360 ("stop spamming, i am the admin. I will delete your account if you spam in the future. Please stop editing pages") |
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| 18:34.31 | fourjahn | nny_1: okay i read the docs .. it looks like my nat settings weren't being saved to my sip.conf |
| 18:34.39 | fourjahn | from the FPBX GUI |
| 18:34.47 | nny_1 | eww fpbx eh>? |
| 18:34.50 | p3nguin | :w |
| 18:34.56 | p3nguin | ~freepbx |
| 18:34.57 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 18:35.28 | fourjahn | lol |
| 18:35.37 | fourjahn | well i can do this manually |
| 18:36.10 | ChannelZ | wonders if someone has already written a doc "10 reasons why not to use FreePBX" |
| 18:36.18 | fourjahn | so what i've done for the time being is put the Asterisk server on the main Comcast modem/router |
| 18:36.28 | fourjahn | since we can't get dedicated IPs for a few days |
| 18:36.35 | nny_1 | that *should* work with a vanilla asterisk |
| 18:36.37 | fourjahn | i'm configuring the NAT now |
| 18:37.07 | lirakis | runs away |
| 18:37.19 | fourjahn | nny_1: well i'm not using a distro if that's what you mean? |
| 18:37.59 | nny_1 | fourjahn: vanilla asterisk (no gui stuff) |
| 18:39.02 | fourjahn | nny_1: looks like i need to call comcraps again .. the port forwarding never seems to take |
| 18:39.19 | fourjahn | nny_1: i can always put it in DMZ for the time being *sigh* |
| 18:39.39 | fourjahn | once i get public IPs it won't be an issue |
| 18:40.10 | fourjahn | i'm just tired and my ass is being ridden by the sales reps and the man |
| 18:40.19 | ChannelZ | why cant you get statics for days? They don't have someone who can click a few buttons until next week? |
| 18:40.50 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 18:40.59 | p3nguin | DMZ is never implemented correctly on those appliances. And if by some chance it is, the people using it don't know what DMZ actually is and does. |
| 18:41.17 | fourjahn | ChannelZ: because Comcast makes you wait for EVERYTHING |
| 18:41.31 | fourjahn | ChannelZ: I asked about expediting she said the best they could do is 24-48 hours. |
| 18:42.37 | fourjahn | this "business gateway" doesn't work .. the settings are just for decoration apparently |
| 18:43.17 | p3nguin | Tomorrow is National Donut Day -- http://www.boston.com/business/ticker/2010/05/dunkin_gears_up_1.html |
| 18:43.42 | ChannelZ | hmm. Was this a new cable install or an existing one? |
| 18:44.23 | fourjahn | ChannelZ: Existing. |
| 18:44.48 | fourjahn | ChannelZ: What I was saying earlier is I'm new to the company and their old IT guy has such a clusterbomb |
| 18:44.57 | p3nguin | Is the modem acting as a bridge or a router? |
| 18:45.11 | fourjahn | p3nguin: at the moment, router. |
| 18:45.55 | p3nguin | I would do what I could to make it a bridging modem and use my own router. |
| 18:47.05 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 18:47.10 | spiceycurry | ~book |
| 18:47.10 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:47.38 | fourjahn | p3nguin: comcast will not allow us to bridge it without a static ip |
| 18:47.56 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 18:47.57 | fourjahn | which we of course cannot get for a couple of days |
| 18:48.03 | p3nguin | How are they preventing it? |
| 18:48.29 | p3nguin | Typically, you go in the UI and change the setting. |
| 18:49.45 | fourjahn | p3nguin: bridging options in the UI are 1-to-1 NAT and static routing |
| 18:50.21 | *** join/#asterisk YellowX (radek@bsdguide.pl) |
| 18:50.23 | p3nguin | I would probably try 1:1 NAT and still use my own router. |
| 18:51.48 | p3nguin | I run a dynamic IP and NAT at home, and I have very few issues with Asterisk on that network. |
| 18:53.49 | p3nguin | My main problem is that one of my ITSPs only provides static SIP, so if my IP address changes, I have to go update my IP in their portal manually. |
| 18:54.04 | fourjahn | p3nguin: Problem is I don't know if these setting are actually 'taking'. When I attempted port forwarding earlier, it was not forwarding the ports from Comcast's router. |
| 18:54.10 | *** join/#asterisk jblack (~jblack@71.181.173.232) |
| 18:54.27 | p3nguin | Not a huge deal, though, since that one's not my primary DID. |
| 18:54.39 | jblack | any jobless friends of mine working in DC? |
| 18:56.09 | spiceycurry | I setup my phones in SIP.conf correct? |
| 18:56.14 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:56.19 | p3nguin | sip.conf, actually. |
| 18:57.58 | spiceycurry | cool |
| 18:58.32 | spiceycurry | do I need any phones or extensions setup in sip.conf if I intend to use fax only? |
| 18:58.40 | spiceycurry | I am guessing probably not |
| 18:58.46 | p3nguin | extensions go in extensions.conf, not sip.conf. |
| 18:59.09 | [TK]D-Fender | spiceycurry: And that would depend how your faxes come in |
| 19:00.10 | p3nguin | If you have a peer that sends the fax calls to you, it would be a good idea to correctly configure the peer in sip.conf. |
| 19:00.16 | spiceycurry | Sorry, I meant do I need any phones setup in sip.conf if I intend to use fax only? |
| 19:00.31 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca) |
| 19:00.39 | p3nguin | s/a peer /a sip peer / |
| 19:01.22 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 19:01.57 | spiceycurry | I am doing email to fax, and fax to email only. No real phones or faxes |
| 19:02.42 | spiceycurry | Do I need a peer to answer a line? |
| 19:03.03 | *** join/#asterisk MiserySoft (~elende@94-116-26-36.dynamic.thecloud.net) |
| 19:03.04 | spiceycurry | I mean, in order to get a fax, do I need to use sip.conf file? |
| 19:03.25 | spiceycurry | Or could I simply have the extensions.conf file handle all incoming calls |
| 19:05.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:08.10 | p3nguin | It depends on how you receive the fax, but [tk]d-fender already said this. |
| 19:08.44 | [TK]D-Fender | spiceycurry: extensions.conf doesn't make a call arrive to *. How are yuo getting your calls? |
| 19:09.13 | spiceycurry | I am getting calls through voip |
| 19:09.15 | p3nguin | If your fax comes in using SIP, then you need to have at least a minimal sip.conf to make chan_sip even work. |
| 19:09.21 | spiceycurry | sorry SIP |
| 19:09.27 | spiceycurry | ok |
| 19:09.37 | [TK]D-Fender | spiceycurry: Wel I guess you need a sip.conf if you're oing to be using SIP, now aren't you? |
| 19:09.49 | spiceycurry | yes sir :D |
| 19:09.59 | p3nguin | If you have a designated SIP peer for your faxes, you should (but don't have to) configure a peer definition for that peer. |
| 19:10.14 | spiceycurry | ok |
| 19:10.50 | p3nguin | If you want anonymous SIP calls for receiving fax, then you don't need to configure a peer definition for it, but you still have to have a minimal sip.conf to make chan_sip work. |
| 19:11.29 | spiceycurry | ok cool |
| 19:11.31 | p3nguin | Beyond that, call processing is done in extensions.conf. |
| 19:12.26 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 19:13.20 | jblack | I need to find an american with a bs that knows ASP/ajax, etc, willing to move to dc |
| 19:14.11 | spiceycurry | Can I have my callgroup and pickupgroup the same group? |
| 19:14.23 | p3nguin | That's kinda the point. |
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| 19:15.13 | spiceycurry | k |
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| 19:33.58 | scrooge-mcduck | hmm, how can i detect the + being using for a call? |
| 19:35.13 | ChannelZ | eh? |
| 19:36.10 | scrooge-mcduck | when call like +42123456 etc. |
| 19:37.00 | ChannelZ | I still don't know what you mean, how you can 'detect [it]' |
| 19:37.31 | ZeXr0 | is it even needed |
| 19:37.51 | ZeXr0 | isn't it like some sort of convention for how to dial internal number ? |
| 19:38.21 | *** join/#asterisk nortonek (~norton@pineska.nat.student.pw.edu.pl) |
| 19:38.36 | ChannelZ | well there is no + key, no.. it means "you had to have dialed something before all this probably". A country code of some sort |
| 19:39.13 | jblack | channelz: If you consider sip addresses, + is a possibility. =) |
| 19:40.12 | ZeXr0 | From wikipedia : For most countries, this is followed by an area code or city code and the subscriber number, which might consist of the code for a particular telephone exchange. ITU-T recommendation E.123 describes how to represent an international telephone number in writing or print, starting with a plus sign ("+") and the country code. When calling an international number from a fixed line phone, the + must be replaced with the international cal |
| 19:40.17 | ZeXr0 | http://en.wikipedia.org/wiki/Telephone_number |
| 19:40.22 | ChannelZ | well I suppose so but I still don't understand the question he asked |
| 19:40.49 | ZeXr0 | Let's wait :P |
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| 19:59.29 | pabelanger | must get new batteries for my keyboard |
| 19:59.46 | russellb | i have a keyboard that i spilled water on the other day that you can have |
| 20:02.04 | thehar | runs through the channel fast and furiously |
| 20:02.07 | thehar | *poof* |
| 20:02.21 | Qwell | trips thehar |
| 20:02.30 | thehar | falls onto russellb |
| 20:03.23 | ChannelZ | take off your pants! |
| 20:03.30 | thehar | oh noes |
| 20:03.42 | thehar | feels a raping about to occur |
| 20:04.32 | russellb | O.O |
| 20:04.42 | russellb | tickles thehar and then runs away |
| 20:04.47 | thehar | eeeeeeeeeeeeeeeeeeeeeeep |
| 20:10.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:12.47 | jblack | wtf? |
| 20:13.07 | jblack | Asterisk, athens style? |
| 20:13.31 | [TK]D-Fender | jblack: Know the motto of the Greek Army? |
| 20:13.47 | [TK]D-Fender | jblack: Never leave a man's behind :) |
| 20:13.49 | *** part/#asterisk lowtek (~lowtek@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 20:33.04 | *** join/#asterisk Bartockbatz (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 20:33.31 | Bartockbatz | hey all - have cdr to MySQL database question - |
| 20:34.11 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
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| 20:34.31 | Bartockbatz | the cdr_addon_mysql.so is loaded - database credentials tested manually, however I cannot get a MySQL connection |
| 20:34.54 | d4rkstar | hi, is there a place where check latest bugfixes for asterisk? |
| 20:35.28 | Qwell | d4rkstar: http://lists.digium.com/pipermail/svn-commits/ |
| 20:35.30 | Bartockbatz | besides the asterisk messages log, where would be another place to look to see where the failure is?? I don't want to have to attach to the module is gdb if I do not have to |
| 20:35.36 | d4rkstar | thank you q |
| 20:35.44 | d4rkstar | thank you Qwell |
| 20:37.13 | tzafrir_laptop | d4rkstar, #asterisk-commits :-) |
| 20:37.21 | Bartockbatz | what I am seeing is a message that cannot connect - something this simple, driving me mad : |
| 20:37.26 | Bartockbatz | :) |
| 20:37.48 | d4rkstar | i'm trying to get t.38 fax working on asterisk 1.6.2.8 :| |
| 20:38.30 | tzafrir_laptop | Bartockbatz, you connect to mysql through TCP port? unix-domain socket? |
| 20:39.26 | d4rkstar | http://pastebin.com/JSM65ns8 |
| 20:39.53 | d4rkstar | here is a log of what happens when a fax is received (t.38) |
| 20:40.31 | Bartockbatz | <tzafrir_laptop>Oh yeah - sorry - database is on the same machine |
| 20:41.12 | tzafrir_laptop | have you configured the connection through a port? Through a file (socket)? |
| 20:42.17 | Bartockbatz | yes tcp 3306 |
| 20:42.29 | Bartockbatz | tcp port |
| 20:42.57 | d4rkstar | Bartockbatz: can you pastebin your my.cnf file? |
| 20:43.04 | Bartockbatz | yes - I can |
| 20:43.26 | d4rkstar | ok! |
| 20:46.21 | *** join/#asterisk SaiSoma (~IceChat7@client105.jdcc.edu) |
| 20:56.00 | d4rkstar | Bartockbatz: did you have that pastebin? |
| 20:56.18 | Bartockbatz | I may not need to send it - |
| 20:56.30 | d4rkstar | ok |
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| 21:17.15 | lordvadr | How do I get * 1.6 to store fields other than ipaddr in the sipregs table in ODBC? Debug says it's skipping them but I can't seem to find where to turn that on. |
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| 21:45.48 | *** join/#asterisk miamiseb (~deigo@208.76.35.132) |
| 21:50.39 | miamiseb | Although it's offtopic, I'm looking for people who have experience with a2billing rating. I need to get it to rate some calls I stuck into it's cc_call database or find out weather this is possible. I DO NOT want to put a2billing in the call path, just export my cdr's from my current solution, massage em a bit and stick them into cc_call and get them rated/invoiced/billed using the a2billing portion |
| 21:51.38 | *** join/#asterisk MhaddogM1 (~MhaddogM1@z65-50-118-232.ips.direcpath.com) |
| 21:52.11 | houms | can anyone point me in the right direction to change the behavior of intercom, where intercoming a user who is already on a call autmatically puts the users call on hold? we are using aastra 9133i handsets |
| 21:52.26 | miamiseb | I've generated a test rate for the USA destination and added a couple of calls into the cc_call database with that destination id, but it's not rating the calls (not picking up any buy and sell rate) |
| 21:54.32 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:55.04 | miamiseb | Also, suggestion of other free call rating solutions that allow import of CDR's would be appreciated. I've tried DTH billing, which is good and works for what we want, but it's also 3k. |
| 21:57.02 | *** join/#asterisk plundra (404@article.se) |
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| 22:08.37 | Shaaan | has anyone in here setup vicidial before?? |
| 22:09.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:10.54 | houms | can anyone point me in the right direction to change the behavior of intercom, where intercoming a user who is already on a call autmatically puts the users call on hold? we are using aastra handsets |
| 22:13.14 | *** join/#asterisk Shaaan (Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 22:16.09 | [TK]D-Fender | houms: Not possible |
| 22:20.11 | vader-- | hey tk do you know the command for dahdi to see what ports are in use and conected? I have a TDM410P card and i want to see what the status is of each port |
| 22:20.27 | vader-- | it's a 3FXO 1FXS |
| 22:20.38 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 22:22.15 | [TK]D-Fender | vader--: dahdi show channels |
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| 22:25.09 | vader-- | umm |
| 22:25.11 | vader-- | i found it |
| 22:25.16 | vader-- | i was looking for service dahdi status |
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| 22:32.12 | houms | Fender may i ask what the allow barge in feature does in that scenario? |
| 22:33.04 | houms | also if I disable auto answer on that extension does that pretty much disable intercomming as well? |
| 22:33.39 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:34.52 | houms | is it an asterisk issue or aastra issue? I have read it is possible to have the extension return busy when the person is on the line? |
| 22:34.54 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 22:42.38 | p3nguin | vader--: That isn't an Asterisk command. |
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| 22:49.11 | *** join/#asterisk Lantizia (~lantizia@93-97-23-110.zone5.bethere.co.uk) |
| 22:49.31 | Lantizia | Hey, can anyone recommend a good but affordable supplier of PRI to SIP and BRI to SIP gateways? |
| 22:50.01 | Lantizia | I know about Patton, and that Mediatrix are rebranded Patton devices already... but wondering if theres any others I'm unaware of |
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| 22:59.17 | miamiseb | If you hadn't used the word affordable...we use ciscos for PRI to SIP. |
| 22:59.39 | Lantizia | need a supplier than can deal with both PRI to SIP and BRI to SIP |
| 23:00.17 | Lantizia | Patton look the cheapest so far... it'll never beat actual PRI/BRI cards in phone systems... but using gateways makes it alot easier to virtualise the phone system and put it in high availability |
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| 23:04.07 | tzafrir_laptop | Lantizia, that's interesting. Can you virtualize that gateway itself? |
| 23:04.26 | Lantizia | tzafrir_laptop, obviously not :) |
| 23:05.15 | Lantizia | but replacing an ISDN gateway is trivial if it's config is backed and you've another on standby. as opposed to opened up a physcial asterisk based phone system (thus taking the entire system down) and replacing a card |
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| 23:14.39 | knarfly | http://pastebin.com/ntvyeTc6 |
| 23:14.40 | knarfly | why does this alway result in unavailable instead of "on the phone" when the callee is on the phone? |
| 23:16.10 | p3nguin | I would be surprised if it works at all. |
| 23:16.58 | SaiSoma|AtHome | grrr. my xlite has decided to stop working (used for development). anyone recommend another softphone that's free? audio quality not important |
| 23:17.01 | knarfly | p3nguin: what's the deal |
| 23:17.20 | knarfly | SaiSoma|AtHome: ZoIPer |
| 23:17.24 | p3nguin | Bad application syntax, duplicate priorities... |
| 23:17.26 | knarfly | it's IAX2 |
| 23:17.31 | p3nguin | and SIP |
| 23:18.02 | p3nguin | For Windows, zoiper is by far the best softphone I have used. |
| 23:18.06 | *** join/#asterisk kilo_mike (~sysop@c-68-49-221-204.hsd1.md.comcast.net) |
| 23:18.09 | knarfly | so what you're say is that as an asterisk tech I suck.... |
| 23:18.19 | knarfly | 8-) |
| 23:19.21 | p3nguin | That's the nice way of putting it. |
| 23:19.34 | knarfly | p3nguin: I don't follow the duplicate priotrites...one is for is the callee doesn't answer, is in the can and the other is if they are on the phone, having phone sex! |
| 23:20.01 | p3nguin | exten => 101,3,Hangup() |
| 23:20.10 | p3nguin | exten => 101,3,Hangup() |
| 23:20.11 | p3nguin | same. |
| 23:20.12 | p3nguin | twice. |
| 23:20.18 | p3nguin | aka duplicates. |
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| 23:20.31 | p3nguin | That's invalid if you ask me. |
| 23:20.42 | knarfly | oh yes, I see that now...the last one should be 101, 104, Hangip() |
| 23:20.47 | p3nguin | core set verbose 4, then run dialplan reload and see if * complains. |
| 23:21.13 | p3nguin | Except that we don't do priority jumping anymore and we don't use numbered priorities. |
| 23:21.23 | SaiSoma|AtHome | gah. same thing. not the software then, something on my system, but only with softphones. too weird. no audio. period. |
| 23:21.44 | knarfly | I copied this from an old doc |
| 23:21.55 | p3nguin | That's why it's an old method. |
| 23:23.43 | knarfly | ok sau how does one get the * server to say "on the phone" instead of always unavailable...otherwise my bosses will always think I'm just not answering my phone instead of being on it.....having phone sex! |
| 23:24.03 | p3nguin | Check the DIALSTATUS. |
| 23:24.41 | p3nguin | Also, before I forget, review all the Dial() options and take out the ones that you don't need. |
| 23:26.09 | knarfly | Cool ! |
| 23:26.43 | knarfly | found ${DIALSTATUS} in TFOT Manual....RTFM now! |
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| 23:29.30 | knarfly | http://pastebin.com/PDertbNP |
| 23:29.30 | knarfly | Think I've got with this one |
| 23:31.18 | p3nguin | I think that'll work. Still need to review those Dial() options, though... I can't imagine you actually need wWFotThH. |
| 23:31.36 | knarfly | I'm anal |
| 23:31.49 | p3nguin | If you were, you wouldn't have all those options. |
| 23:32.26 | knarfly | I want to be able to record calls,, that's the most important...the others are just my testing |
| 23:32.37 | p3nguin | You know that the options you have there will allow me to call your phone, start recording, and even transfer the call, right? |
| 23:32.53 | knarfly | yes |
| 23:32.54 | SaiSoma|AtHome | i've been looking, but haven't been able to find anything on getting the last_insert_id from MYSQL. Any pointers? |
| 23:34.55 | *** join/#asterisk saftsack (~oliver@p4FF0CDF7.dip.t-dialin.net) |
| 23:35.24 | saftsack | hi, maybe wrong channel, does somebody use hylafax with capi4hylafax? |
| 23:35.34 | knarfly | nope that sux...still tells the calling party that I'm unavailable...not on the phone |
| 23:36.47 | p3nguin | http://pastebin.com/RdLzDPtW |
| 23:37.03 | p3nguin | Add the Verbose in there to output the value of DIALSTATUS so you can see what is going on. |
| 23:38.35 | knarfly | stand bu |
| 23:38.38 | knarfly | by |
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| 23:40.11 | p3nguin | There's also another way to branch out depending on DIALSTATUS. http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS |
| 23:40.27 | knarfly | nope it's still saying the callee is unavailable |
| 23:41.37 | knarfly | I make a call on 201 then while that call is gong I call it from 101...101 should hear that 201 is on the phone but it keeps saying unavailable...now my bosses will keep thinking I'm fucking off instead of being on the phone....fucking off |
| 23:42.53 | p3nguin | The dialstatus is showing what? |
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| 23:43.52 | knarfly | stand by checking again |
| 23:45.52 | knarfly | http://pastebin.com/0Z5Lr1Y2 |
| 23:45.53 | knarfly | My CLI shows this on the call |
| 23:47.35 | p3nguin | It's doing what it's supposed to be doing. |
| 23:48.11 | p3nguin | Since you won't add the Verbose(${DIALSTATUS}) to SHOW YOU the dialstatus, you don't realize it. |
| 23:49.20 | knarfly | I am using the text you posted and it has the Verbose(${DIALSTATUS}) in it...and I restarted * |
| 23:49.25 | p3nguin | GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail) <-- If "${DIALSTATUS}" does not equal "BUSY", jump to label unavail (priority 3). |
| 23:49.41 | p3nguin | You don't need to restart asterisk just to change the dialplan. |
| 23:49.47 | p3nguin | dialplan reload is sufficient. |
| 23:50.18 | knarfly | I copied and pastes. let me dounle check for typos |
| 23:50.20 | p3nguin | Goto (deluxe,101,3) |
| 23:50.33 | p3nguin | The diaplan has succeeded. |
| 23:50.59 | p3nguin | So now it comes down to knowing what the dialstatus is. |
| 23:51.11 | p3nguin | You know what it isn't. |
| 23:53.58 | knarfly | so how can I check what DIALSTATUS get's set to |
| 23:55.01 | [TK]D-Fender | [19:48]<p3nguin>Since you won't add the Verbose(${DIALSTATUS}) to SHOW YOU the dialstatus, you don't realize it. |
| 23:55.05 | p3nguin | http://pastebin.com/RdLzDPtW |
| 23:55.42 | knarfly | ok so are yous guys saying that I don't have the Verbose...in my exten....it's in there |
| 23:56.17 | p3nguin | Then one of two things is true: You either didn't run dialplan reload or DIALSTATUS is empty. |
| 23:56.35 | [TK]D-Fender | knarfly: You asked how to check it. Well holy shit THAT CHECKS IT |
| 23:56.53 | p3nguin | With core verbosity turned up, you would see where Verbose() is being ran, too. |
| 23:57.57 | knarfly | http://pastebin.com/Bs0KEVR0 |
| 23:57.58 | knarfly | this is how each extension is setup now ...is this not the way you posted? |
| 23:58.15 | knarfly | and when I start asterisk I start it with sox (6) v's |
| 23:58.41 | knarfly | I restarted asterisk |
| 23:59.56 | knarfly | so DIALSTATUS is empty ??? |