| 00:01.58 | jpsharp | Chances are that the ./configure script didn't pick the bluetooth libraries. Look through config.log to see what it is complaining about. |
| 00:02.26 | WIMPy | Did you run configure again after installing bluetooth? |
| 00:02.38 | atan | WIMPy, yes |
| 00:02.46 | atan | jpsharp, checking that out for bluetooth stuff |
| 00:02.46 | WIMPy | Or probably some buetooth-dev. |
| 00:03.08 | atan | "/usr/bin/ld: cannot find -lbluetooth" |
| 00:03.13 | atan | Hmm, wonder what that is |
| 00:03.39 | WIMPy | No "libbluetooth" found. |
| 00:04.08 | atan | Wonder which one it wants, libbluetooth3 or libbluetooth-dev |
| 00:04.49 | atan | Ahh the second one :D w00t |
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| 00:21.05 | Velluto | Hi Everyone - I was just wondering if someone could help me out with an error I'm receiving with CDR (related to ODBC & Realtime) |
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| 00:47.41 | Velluto | Can anyone help me with an error I'm receiving with CDR? ("CDR Direct execute failed") Connecting through ODBC to MSSQL 2008 R2, Realtime is working correctly through the same connection |
| 00:48.29 | WIMPy | can't but I'm open to suggestion on what to break today. |
| 00:51.49 | Velluto | it seems like Realtime is broken with CDR through cdr_alternative_odbc (realtime only works for the first call) - and CDR is not reporint anything using cdr_odbc |
| 00:52.08 | Velluto | reporting* |
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| 01:03.55 | sarthor | Hi, How can I get a voip or sip dialer with my own sip.example.com etc setting, The dialer that just need to enter login/password. Any guidance ? |
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| 02:13.32 | Curs0r | Hey folks. Popped in to say I got it all working and asterisk is so... I could hug it. |
| 02:27.18 | ChrisInSydney | Curs0r: awwwww :D |
| 02:28.09 | ChrisInSydney | now you can go back to your family and the ones that really love you ;) |
| 02:29.15 | Curs0r | When a friend of mine asked for a phone system I looked into ip phones. People came out of the woodwork trying to ram a hosted pbx down my throat. So many, that I knew there must be readily available software for the task |
| 02:30.01 | WIMPy | Now I'm finally back on TRUNK again. |
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| 03:57.09 | amessina | good evening. would anyone be able to help troubleshoot inbound PSTN CallerID issues with an Astribank FXO port? I would appreciate the assistance. |
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| 04:21.37 | OldMonk | please remind me again: if i have two matches for a number in a dialplan, will the call go to the more specific match or the first match? |
| 04:22.19 | OldMonk | i.e. number is 12345, i have rules for _1. , _123. and 12345 (in that order) |
| 04:23.54 | OldMonk | gs/number/extension/ |
| 04:27.51 | OldMonk | well, http://www.the-asterisk-book.com/unstable/einleitung-regex.html says: The reason for this is simple: more than one pattern might match a dialled number. If two extensions match a dialled number, Asterisk will always choose the better match. Before deciding which extension matches best, it processes the entire context. |
| 04:28.10 | OldMonk | i guess that answers my question (though the page is talking about asterisk 1.2 and 1.4) |
| 04:31.55 | ChrisInSydney | OldMonk: I'm sure you are correct in your assumtions. Its simple enough to test. I can say from experience that it seems to process the last rule first if you have identical dial rules |
| 04:32.54 | OldMonk | ChrisInSydney: "last rule first" or "last rule only"? |
| 04:33.07 | ChrisInSydney | last rule only |
| 04:33.29 | OldMonk | OTOH, i'm not planning to fsck up my dialplan with identicat rules, so i guess i should be ok |
| 04:33.39 | OldMonk | s/identicat/identical/ |
| 04:33.53 | ChrisInSydney | whch sort of makes sense as the first rule is overwritten by the second as its loaded into memory |
| 04:33.57 | OldMonk | now that is a bit irritating |
| 04:34.11 | OldMonk | ChrisInSydney: not you... infobot |
| 04:34.26 | ChrisInSydney | that was cool |
| 04:34.33 | ChrisInSydney | :D |
| 04:34.46 | ChrisInSydney | i knew what you meant |
| 04:35.12 | OldMonk | precisely, if you're familiar with the paradigm it's just creating noise |
| 04:35.13 | ChrisInSydney | I'd be keen to try some other words with it, but I'd probably get kick/banned |
| 04:35.23 | OldMonk | haahaa! |
| 04:35.32 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 04:35.48 | OldMonk | hmm, need to test something |
| 04:35.56 | atan | What's the name of that bluetooth device that let's you connect your cellular device and use it similar to an ATA? |
| 04:37.26 | ChrisInSydney | atan: http://www.voip-info.org/wiki/view/Asterisk+Bluetooth+channels |
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| 04:37.38 | ChrisInSydney | chan_bluetooth ?? |
| 04:38.01 | atan | Yeah, chan_mobile... I've been playing around with it, I just had high hopes of something a bit easier to put together :D |
| 04:38.02 | ChrisInSydney | or maybe chan_mobile |
| 04:38.10 | ChrisInSydney | GSM gateway |
| 04:38.23 | atan | Without removing the sim from the device, only use via bluetooth |
| 04:38.29 | ChrisInSydney | ahh |
| 04:38.37 | atan | I'm sure it's workable I've just never set it up. Going to be a good day playing in the docs I'm sure hah :D |
| 04:38.43 | atan | Have you done any such thing with it? |
| 04:38.44 | ChrisInSydney | actually, is there an ATA app for 'droid or iPhone ? |
| 04:38.58 | atan | Chris, I don't believe so I've been looking for Android |
| 04:39.03 | atan | And god only knows Apple won't allow |
| 04:39.13 | ChrisInSydney | there is an idea for an app |
| 04:39.14 | atan | Go make one :D |
| 04:39.28 | ChrisInSydney | main(){ |
| 04:39.29 | ChrisInSydney | } |
| 04:39.34 | ChrisInSydney | now what to pout in betweeen |
| 04:39.42 | atan | I believe the problem was in that the audio streams are protected or something like that =\ |
| 04:40.02 | ChrisInSydney | you can CTI the phone to make calls |
| 04:40.16 | ChrisInSydney | and change the caller ID on onbound |
| 04:40.21 | ChrisInSydney | on ian iPhone |
| 04:40.23 | ChrisInSydney | an |
| 04:40.36 | ChrisInSydney | ian had nothing to do with it |
| 04:40.42 | ChrisInSydney | :/ |
| 04:49.20 | drmessano | chan_mobile is actually relatively easy to get working |
| 04:54.28 | atan | drmessano, I made it as far as getting a pairing request on the phone but was working on finding out where the passkey bit is set |
| 04:54.39 | atan | Do you happen to know if that's something the os manages or do you set that in Asterisk? |
| 04:55.26 | drmessano | /etc/bluetooth/hcid.conf on CentOS |
| 04:56.48 | drmessano | One of the options { is passkey= |
| 04:57.42 | atan | I assumed I might find something like that on debian but there's no hcid.conf in there |
| 04:58.44 | atan | I find options for audio.conf input.conf main.conf network.conf rfcomm.conf serial.conf, but no hcid. The adapter happens to get picked up as hci0 of course, and hcitool lets me scan and all that good stuff O_O |
| 04:59.11 | drmessano | Did you grep all those for options and device options? |
| 05:01.06 | drmessano | Im looking now on Ubuntu. |
| 05:02.20 | atan | cat * | grep passkey gives me nothing |
| 05:04.13 | drmessano | I guess a few things changed in bluez... try this.. http://www.linuxquestions.org/questions/linux-wireless-networking-41/setting-up-bluez-with-a-passkey-pin-to-be-used-as-headset-for-iphone-816003/ |
| 05:05.58 | atan | "You can no longer configure PINs in plain-text files as you could under BlueZ 3.x, as it was seen as a security liability. All pairing must now be done through D-BUS calls, so you need a helper application." interesting |
| 05:07.33 | atan | Thank you kindly for the link :) I'm going to spend some time there in a bit :) will be great to get the cellular devices ringing hard phones while in the office... true love. |
| 05:08.08 | drmessano | Everything I have done was on CentOS 5, which would have been 3.x It worked very well, on Samsung, Motorola Razr, and the iPhone |
| 05:08.19 | drmessano | I am updating my docs now |
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| 05:37.23 | drmessano | atan, you there? |
| 05:37.39 | atan | I am :) |
| 05:37.46 | atan | Well, off in the distance but I am |
| 05:38.18 | drmessano | Ok, heres a few things |
| 05:39.01 | drmessano | First off, bluez-tools provides bluetooth-agent... Running sudo bluetooth-agent 1235 should help you pair |
| 05:39.02 | drmessano | However |
| 05:39.45 | drmessano | I am not getting far on this machine because hcitool dev is not showing my device.. seems to be a common problem with ebay chinacom dongles. I am still checking on that. |
| 05:39.53 | drmessano | the rest is pretty well documented here: |
| 05:40.23 | drmessano | http://en.gentoo-wiki.com/wiki/Bluetooth_Network_Aggregation_Point <-- ignoring the network setup at the end |
| 05:43.38 | drmessano | I know these devices work, because I have been using them for years. I can pair just fine on Ubuntu desktop. Strange |
| 05:43.59 | atan | I'll get into that here in a few just need to finish up some paperwork :) |
| 05:44.11 | atan | Seems like a neat module I can't wait use it :D |
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| 05:50.36 | drmessano | What I loved, back in the pre-google voice days, was that when I got scatter brained and couldnt find my phone, I could always count on calls ringing through the phones at home. Never missed a call from work |
| 05:50.52 | atan | :D |
| 05:51.19 | drmessano | Even worked when I left the phone in the car, with the server on the second floor, at the far end of the apartment |
| 05:51.26 | drmessano | Amazing |
| 05:52.02 | atan | :) I can only dream of this |
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| 05:53.23 | amessina | tzafrir: Can I ask for some pointers on debugging FXO callerid on an Astribank? I've followed http://docs.tzafrir.org.il/dahdi-tools/README.Astribank.html "to a t" and am unable to receive callerid from the PSTN. |
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| 06:02.13 | kaldemar | amessina: how have you configured the channels in chan_dahdi.conf? |
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| 06:02.32 | v0lZy | lo |
| 06:02.44 | kaldemar | amessina: there are other parameters besides callerid=asreceived. |
| 06:04.05 | kaldemar | amessina: usecallerid, cidsignalling, cidstart, dtmfcidlevel |
| 06:04.20 | amessina | kaldemar: http://pastebin.com/URe5P543 |
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| 06:05.27 | amessina | kaldemar: I was previously using a TDM400p card with this configuration --except the channel numbers have changed. I am in the US so I should be getting the default: cidsignalling=bell, didstart=ring. |
| 06:06.29 | amessina | kaldemar: the callerid works perfectly on the FXS modules --the internal phones. I can't get the incoming callerid from the PSTN: AT&T. |
| 06:07.00 | amessina | kaldemar: I have plugged in a regular phone to the line from the telco to see if that phone receives callerid and it does. |
| 06:07.32 | kaldemar | sounds like the astribank is to blame then. |
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| 06:09.53 | amessina | kaldemar: either that or I don't have Asterisk configg'd properly, but again, it's the same config that works with the TDM400p |
| 06:12.22 | amessina | kaldemar: Do you know of any debug settings for Asterisk that would let me see the caller id detection in the console? I've tried debug at 10 and verbose at 10, but they don't seem to indicate anything about caller id detection errors. I'd like to narrow down the debug output to callerid, if possible. |
| 06:20.16 | kaldemar | adding dtmf to logger would show incoming caller id if you used dtmf, but you have bell... i guess core debug is your best choice to see anything. |
| 06:21.48 | amessina | kaldemar: ok. i'll keep plugging away before i resort to getting a hold of Xorcom. this unit is brand new. |
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| 06:36.33 | gajini | Hi, I need to install wct2xxp driver for Digium TE210P card. Please tell me how to install wc2xxp module only not wct4xxp |
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| 06:38.35 | schmidts | good morning |
| 06:51.33 | tzafrir | amessina, what country is it? |
| 06:52.00 | amessina | tzafrir: thanks for responding. it's the US. |
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| 07:02.43 | tzafrir | IIRC you should explicitly state 'usecallerid = yes' |
| 07:04.45 | amessina | tzafrir: I think I have that: http://pastebin.com/URe5P543 Do you mean I should do that just before "channel => 13"? |
| 07:06.35 | tzafrir | no. it's ok. I missed it |
| 07:07.16 | tzafrir | If you replace the FXO port with a simple caller-id capable analog phone, does it detect the caller ID on the line? |
| 07:08.08 | amessina | tzafrir: yes, I did replace the FXO port with a regular phone and that phone was able to detect the caller id on the line properly |
| 07:11.37 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3df8:fbe9:fdfd:9e98) |
| 07:15.27 | _val_ | Hello everyone. I've the following line in extension.conf exten=>_0X.,5,Dial(SIP/${EXTEN:0}@mycompany.com,30,tr). Now When I call a number in Germany 0049221... it says: == Using SIP RTP CoS mark 5 |
| 07:15.47 | _val_ | This is all the information I get. I've debugging 10 verbosity 10 |
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| 07:26.30 | amessina | tzafrir: if a regular phone can detect the callerid, but the Astribank FXO cannot, what is generally a person's next step? |
| 07:26.31 | AdvoWork | morning, does anyone know anything about ISDN2/ISDN30? |
| 07:28.07 | tzafrir | amessina, any reason to believe it's not a standard Bell style caller ID? |
| 07:28.57 | _val_ | When I dial this number 00439.. I get this: http://pastie.org/4124933 |
| 07:29.00 | _val_ | any idea? |
| 07:29.23 | amessina | tzafrir: Not sure how to find out. I don't think so. I'm using almost exactly the same config as I did with a TDM400p, which worked perfectly detecting callerid. The only thing that has changed is the FXO channel number (from 3 to 13). |
| 07:33.33 | kaldemar | _val_: what you say you dialed does not match your paste. |
| 07:36.41 | kaldemar | _val_: sip debug may show you some additional information. what is mycompany.com? |
| 07:42.27 | _val_ | kaldemar: where my sip account is hosted |
| 07:42.43 | _val_ | sorry for the late response. I'm trying to debug. I'll issue sip debug |
| 07:43.10 | ChannelZ | amessina: what hardware is it now? |
| 07:43.19 | ChannelZ | oh nm you said Astribank |
| 07:43.42 | amessina | ChannelZ: Astribank 6FXS/2FXO |
| 07:43.44 | _val_ | kaldemar: sip show debug doesn't help |
| 07:44.07 | kaldemar | _val_: you have three different factors there. i'll assume that mycompany.com is fabricated, and the real address is the one below. looks like you use an outbound proxy. is that correct? |
| 07:44.18 | kaldemar | _val_: are you sure you can judge if it helps or not? |
| 07:44.18 | ChannelZ | I know nothing of those, do they show as DAHDI channels or is it a SIP bridge? |
| 07:44.54 | _val_ | kaldemar: now I meant "command is not recognized" |
| 07:45.26 | _val_ | I'm the one who's is trying to get help remember. I can't judge anything at this moment. |
| 07:45.37 | _val_ | kaldemar: true. vncproxy... |
| 07:45.51 | _val_ | I changed the paste for ethical issues |
| 07:47.34 | kaldemar | _val_: "sip set debug on" is the correct command. |
| 07:47.46 | amessina | ChannelZ: The channels show up as DAHDI channels. |
| 07:48.11 | kaldemar | _val_: if you're going to change something, change all of it. otherwise you're just screwing up the debug. |
| 07:50.01 | kaldemar | _val_: have you configured a peer for the provider in sip.conf? |
| 07:50.36 | _val_ | kaldemar: yes. That's all done. I can make calls within holland but not outside |
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| 07:52.35 | kaldemar | _val_: "within holland" as in using the same peer? |
| 07:53.54 | _val_ | kaldemar: yes. Give me a second please. I'll post the sip.conf account details. |
| 07:54.31 | kaldemar | _val_: you should rather post CLI output of a call that does not fail. |
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| 07:57.36 | kaldemar | _val_: however, the call seems to be going to spain since that's where the 500 respones comes from. check the dialed number first. |
| 08:00.15 | _val_ | kaldemar: http://pastie.org/4125053 some information |
| 08:00.20 | _val_ | I'll check the number |
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| 08:08.39 | amessina | tzafrir: is there any special debugging I should use to figure out this callerid issue before I move forward to contact Xorcom? |
| 08:10.55 | _val_ | kaldemar: here is the debug http://pastie.org/4125095 |
| 08:11.20 | _val_ | This is a german number, but I won't even get a tone! |
| 08:11.29 | tzafrir | amessina, Can't think of anything |
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| 08:12.31 | amessina | tzafrir: ok, well thanks for your help. |
| 08:13.20 | kaldemar | _val_: that tells that you are behind a nat but have not configured your asterisk according to it. |
| 08:13.24 | kaldemar | ~sipnat |
| 08:13.24 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 08:14.01 | _val_ | kaldemar: how does it explain that I can call numbers of different cities in holland but not outside? |
| 08:14.06 | _val_ | How is this related to NAT? |
| 08:15.49 | _val_ | Eg. When I call from Groningen to Amsterdam it works. Calling from Groningen to Germany it doesn.t |
| 08:16.51 | kaldemar | _val_: the information you're giving is not sufficient to say anything about the calls working to holland numbers. |
| 08:17.38 | v0lZy | maybe its your dial patteren or somethin |
| 08:17.40 | v0lZy | g |
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| 08:18.07 | kaldemar | _val_: the nat config is not necessarily to blaim with the other calls not working. however, your asterisk seems to send a private address in the SIP messages, and that is most likely wrong. |
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| 08:19.26 | *** join/#asterisk jsjc (~Adium@175.43.62.126) |
| 08:20.15 | beaver_rrr | do sip_reregister breaks active calls on this peer? |
| 08:20.51 | _val_ | kaldemar: fair enough. What I'm worried about is. This. if a number in Belgium is 10 digits it will work because I put there exten_0XXXXXXXXX.4,set(.. |
| 08:21.23 | *** join/#asterisk iscsi (~light@cafe.zesdid.com) |
| 08:21.47 | _val_ | if I need to make a call to germany the number is 15 digits. So I need to put _0XXXXXXXXXXXXXX,3,Set(CALLER... |
| 08:22.07 | _val_ | this is annoying as I have to specify for each account the exten... |
| 08:22.41 | _val_ | exten=_0X. won't work nor will exten=_!X nor exten=_X. |
| 08:22.47 | iscsi | moin |
| 08:24.51 | sekil | _val_: eh? |
| 08:26.53 | kaldemar | _val_: _0X. will work when configured properly. _X. is also valid. _!X is invalid. |
| 08:28.33 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 08:28.56 | _val_ | kaldemar: define properly please? I don't think there is anything wrong with the configuration as I can make any call anywhere. The only problem is that I've to define.. _08XXXX , _032XXXX... _0XXXXXXX... for every lands code everytime |
| 08:29.29 | _val_ | if Tomorrow one of the employees calls to china.. I've to define it manually.. |
| 08:29.50 | nicola_pav | hello. could digium fax for asterisk send a confirmation message to the sending side? |
| 08:30.04 | *** join/#asterisk jsjc (~Adium@175.43.62.126) |
| 08:31.06 | kaldemar | _val_: _0X. will match any dialed string that starts with 0, is followed by 0-9 and one or mode digits or characters. |
| 08:31.35 | v0lZy | uh |
| 08:31.52 | kaldemar | _val_: what is the problem? |
| 08:31.54 | v0lZy | _val_ what does your provider require? |
| 08:32.05 | v0lZy | what kind of patteren does it want? |
| 08:33.38 | v0lZy | 3861+0|XX. |
| 08:33.40 | v0lZy | 386+00|XX. |
| 08:33.41 | v0lZy | 000|X. |
| 08:33.43 | v0lZy | thats my patteren |
| 08:33.53 | v0lZy | 0 for outgoing outside my pbx |
| 08:35.10 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 08:36.17 | v0lZy | so in case of 000|X. i dial 0 to get outside my pbx, 00 to go international and X. is the phone number im dialing together with the country code |
| 08:36.21 | _val_ | v0lZy: I don't know I think I've lost it :<. I've gone through tutorials yesterday |
| 08:36.31 | _val_ | germany is e.g 0049 |
| 08:36.42 | v0lZy | your voip provider should give you the patteren |
| 08:36.49 | _val_ | Koln city in Germany is 0049221 |
| 08:37.04 | _val_ | Ah. I'll have to contact the voip provider than. |
| 08:37.09 | v0lZy | _val_: this is probably about how you need to pass a number to your provider |
| 08:37.11 | _val_ | v0lZy: how can I see the patterns? |
| 08:37.57 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 08:38.44 | v0lZy | check configuration files |
| 08:38.50 | v0lZy | sec, let me see which one its in |
| 08:41.27 | v0lZy | (i use askoziapbx so im not that afmiliar with config files, this will take a while) |
| 08:41.44 | _val_ | v0lZy: I've all the time in the world. |
| 08:41.52 | _val_ | Thanks in advance mate. |
| 08:42.17 | sekil | _val_: what are you trying to achieve? |
| 08:44.06 | _val_ | sekil: simple thing. Getting exten=_0X.[work_with_all_numbers] instead of defining myself. exten=_08XXXX..... or exten=_09XXXX... etc.. |
| 08:44.38 | sekil | _val_: how does _0X. exten not work? |
| 08:45.01 | v0lZy | extensions.conf |
| 08:45.13 | v0lZy | in my case its in extensions conf where i have an outgoing provider defined. |
| 08:45.38 | v0lZy | i think |
| 08:45.39 | v0lZy | w8 |
| 08:46.06 | v0lZy | ah ok |
| 08:46.08 | v0lZy | i think in my case |
| 08:46.13 | v0lZy | in my extensions.conf which is really messy |
| 08:46.20 | v0lZy | its defined on a per-phone basis |
| 08:46.28 | iscsi | use ael :) |
| 08:46.47 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:48.22 | v0lZy | _val_: in my configuration, each phone includes the patteren |
| 08:48.28 | v0lZy | and sets outgoing caller id etc |
| 08:48.40 | v0lZy | thats where the patteren is configured in my case |
| 08:48.56 | v0lZy | lots of gotos and stuff |
| 08:53.11 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 08:55.14 | v0lZy | _val_: just call your service provider and ask them for the pattern |
| 08:59.36 | *** join/#asterisk beebeeep (~beebeeep@46.182.24.147) |
| 08:59.57 | _val_ | sekil: it does not call 15 digit number unless I specify it using exten=_0XXXXXXXXXXXXX |
| 09:00.02 | _val_ | 15-times an X |
| 09:00.18 | _val_ | err. 14-times X including 0 = 15 |
| 09:01.05 | sekil | _val_: 15 digit number w/o zero is what you need? |
| 09:01.30 | sekil | anyhow..._0X. should work if you don't have any other extensions |
| 09:01.34 | v0lZy | DOT |
| 09:01.41 | v0lZy | _0X. |
| 09:01.44 | v0lZy | with a dot. |
| 09:02.27 | v0lZy | _val_: this really depends on what your provider assigns you |
| 09:02.40 | v0lZy | _val_: for example, my provider wants me to dial a full country code every time |
| 09:03.16 | v0lZy | _val_: so even to dial locally, my provider wants to receive a countryode formated number |
| 09:03.28 | v0lZy | im from slovenia (country code 386) |
| 09:03.50 | v0lZy | so i have to send 386<city code><number> |
| 09:03.59 | v0lZy | im from the capital so my city code is 1 |
| 09:04.03 | _val_ | v0lZy: you mean no other extension should be there? |
| 09:04.19 | v0lZy | so if i want to dial my neighbour... i need to send 3861555555 for example |
| 09:04.22 | _val_ | I've _0X. and I have _08XXXXXXXXX.... and I have _0X999999 |
| 09:04.24 | v0lZy | if i want to call to germany |
| 09:04.25 | sekil | _0X. will match everything starting with 0digit and digits afterwards |
| 09:04.26 | v0lZy | i have to call |
| 09:04.31 | sekil | nothing to do with any provider |
| 09:04.39 | v0lZy | 49<city code><number> |
| 09:04.49 | _val_ | yes |
| 09:04.58 | v0lZy | but this is not waht people are used to calling |
| 09:05.05 | v0lZy | peopel are used to calling a number without city prefix |
| 09:05.09 | _val_ | but v0lZy do you use 0049221... when you make the call or without 00? |
| 09:05.10 | v0lZy | to call their neighbour |
| 09:05.14 | v0lZy | so they just go |
| 09:05.17 | v0lZy | 555 555 |
| 09:05.36 | v0lZy | for that reason, i need a patteren that will send 3861555555 when i dial 555 555 |
| 09:05.57 | v0lZy | if i want to dial to another city, people are used to dial 0<city code><number> like 02 444 444 |
| 09:06.01 | v0lZy | so i need a patteren to send |
| 09:06.15 | v0lZy | 3862444444 to my provider |
| 09:06.22 | v0lZy | when i dial 02 444 444 |
| 09:06.39 | v0lZy | and for dialing abroad, people here are used to dial 00<country code><city code><number> |
| 09:07.23 | *** join/#asterisk Jo_Bell (sasishiju@115.249.39.106) |
| 09:07.34 | v0lZy | so if i want to dial to germany i need to send a pattern like this: 493888888 when i input 00493888888 |
| 09:07.46 | v0lZy | and because i have internal number on my pbx for other internal users |
| 09:08.18 | _val_ | sekil: [Jun 21 11:07:54] NOTICE[27636]: chan_sip.c:20163 handle_request_invite: Call from 'jeroenmunt' to extension '4922129930343' rejected because extension not found. |
| 09:08.20 | v0lZy | i add a 0 infront of everything (0 means dialing outside my pbx, 00 means dialing outside my pbx and outside my city code, and 000 means go out of my pbx, out of my city code and out of my country) |
| 09:11.49 | v0lZy | _val_: well your PBX has to 'know' where to send numbers.... |
| 09:12.17 | v0lZy | you need a pattern to connect |
| 09:12.26 | v0lZy | your PBX knows nothing of country codes or who to call |
| 09:12.54 | v0lZy | it is simply 'see a number route it' |
| 09:13.23 | v0lZy | it needs to know that 00 means dial outside my country |
| 09:14.00 | v0lZy | and it needs to know that when u press 00 it needs to send the number you are dialing to your voip provider in a format that the provider accepts. You have to tell it this format by using a pattern |
| 09:14.49 | _val_ | v0lZy: http://www.pyrosoft.co.uk/blog/2008/05/02/asterisk-outgoing-calls-rejected-because-extension-not-found/ |
| 09:15.14 | _val_ | This is what the guy did to solve it. include=numberplan-custom-1. Where does it get it from? |
| 09:19.14 | sekil | bangs his head on the wall |
| 09:20.14 | wdoekes | ~book |
| 09:20.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:23.34 | v0lZy | _val_: read his whole post |
| 09:26.19 | v0lZy | and i mean... read it in a way where you will understand what the problem was |
| 09:27.04 | jol | Is there some way to stop the current AGI command, making it end early, and let the AGI continue? Such as STREAM FILE on a long sound file, and you want to stop the playback and let the AGI continue. |
| 09:27.46 | jol | (Or, what user email group is best for sending such a question?) |
| 09:32.04 | _val_ | v0lZy: so I did modify the [internal] clauss. I called it [internal]. Error is this. http://pastie.org/4125407 |
| 09:33.37 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 09:35.59 | _val_ | gives up. |
| 09:36.14 | _val_ | Thanks everyone for the help. |
| 09:36.22 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:36.27 | v0lZy | w8 |
| 09:36.33 | v0lZy | sec |
| 09:38.09 | v0lZy | _val_: query |
| 09:47.51 | *** join/#asterisk danfromuk (~IceChat77@2.27.7.133) |
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| 09:58.38 | *** join/#asterisk gusto (~gusto@ip-109-84-0-101.web.vodafone.de) |
| 09:58.39 | gusto | so so |
| 09:59.02 | gusto | here i am |
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| 10:13.48 | *** join/#asterisk Node_403 (cbdf5c48@gateway/web/freenode/ip.203.223.92.72) |
| 10:13.56 | Node_403 | I need some one who has fair knowledge about asterisk and VPN and who can do a project on Hire |
| 10:18.22 | Node_403 | I need some one who has fair knowledge about asterisk and VPN and who can do a project on Hire. Please mail me |
| 10:18.39 | zamba | node_403@internet? :) |
| 10:19.07 | Node_403 | :) |
| 10:26.48 | WIMPy | Node_403: Have you been reading maps? ;-) |
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| 11:01.29 | v0lZy | Odesk. |
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| 12:03.30 | carrar | *Y*A*W*N* |
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| 13:01.49 | thecardsmith | I'm looking to read up about fail-over between two asterisk boxes, especially the feasibility of having Asterisk stay in the media path keep a call up / not dropping a call |
| 13:02.12 | *** join/#asterisk Neptu (~Neptu@41.251.186.164) |
| 13:02.18 | thecardsmith | Does anyone know if, A. this is possible, and B. what direction i should look? a google hint, or maybe a point to some docs? |
| 13:02.56 | thecardsmith | For what it's worth, that would be during a SIP call. |
| 13:03.17 | EvilGrin|3D | I don't believe you can do a hot failover |
| 13:03.26 | thecardsmith | I heard in Asterisk Advanced training that this was possible, and I also read the manual for the R850, thinking that might have some pointers, but, as far as I can tell the solution is only to bring the service back up, not keep the calls up |
| 13:03.39 | EvilGrin|3D | so you'd lose all your ongoing calls |
| 13:03.43 | thecardsmith | (I may have heard "happily" in that case, too) |
| 13:03.55 | thecardsmith | ahhh, yeah, that's kinda what I was afraid of |
| 13:04.07 | EvilGrin|3D | but you'd maintain service |
| 13:04.44 | thecardsmith | true, better than one box that's just down :) |
| 13:04.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-vdfpeoilblcmxbxf) |
| 13:04.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:05.21 | EvilGrin|3D | well it depends, how robust is your carrier and underlying layer 3 network for example? :) |
| 13:06.07 | EvilGrin|3D | doesn't matter how many boxes you have to fail over to if your telco's exchange is underwater :P |
| 13:06.43 | thecardsmith | hahah, truth. i'm kinda lucky in that regard, i have my asterisk boxes sitting behind a pretty hefty TDM gateway, it's a lucent 5ess switch with PRIs and viop gateway capability |
| 13:07.08 | EvilGrin|3D | very nice |
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| 14:05.26 | *** join/#asterisk Rolandow (~rolandow@92.68.81.83) |
| 14:05.53 | Rolandow | good afternoon! i have a question that has been discussed many times, but still i cannot find the right answer :( |
| 14:06.05 | Rolandow | it's about the caller-id that does not update when i do an attended transfer |
| 14:06.55 | Rolandow | the thing is that when the receptionist transfers a call, i see the display of reception in my display .. then when she hangs up, the display doesn't change. |
| 14:07.05 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 14:07.17 | Rolandow | i tried sendrpid=yes and trustrpid=yes in sip.conf for my peer and for my phone sip accounts, that didn't help |
| 14:08.09 | Rolandow | we're using tiptel 280 (or yealink t-20, they're practically the same) |
| 14:08.12 | Rolandow | any ideas? |
| 14:09.33 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:09.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:09.58 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 14:10.01 | cusco | hello.... |
| 14:10.13 | malcolmd | maybe the yealink supports P-Asserted Identity instead of Remote Party ID. try sendrpid=pai |
| 14:10.19 | malcolmd | if that doesn't work, talk to Yealink |
| 14:10.39 | cusco | in case I accidentally deleted a AEL, while it is still loaded in asterisk's memory, can I still get it back? |
| 14:10.42 | Rolandow | should i do that for the sip account of the phone? |
| 14:10.51 | Rolandow | probably not for the sip account of my voip provider right? |
| 14:11.41 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 14:11.46 | malcolmd | for the phone, yes; some providers don't like it, so you may need to disable it for the provider |
| 14:12.46 | Rolandow | hm.. doesn't work eigher :( |
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| 14:17.06 | kaldemar | cusco: you may be able to save the dialplan as extensions.conf with "dialplan save" if you didn't have writeprotect=yes under [general]. |
| 14:17.17 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 14:17.37 | kaldemar | cusco: if you have something relevant in extensions.conf, back it up before issuing that command. |
| 14:20.54 | *** join/#asterisk Samuel_CTX (~svanderp@62-213-207-59.colo.housingcenter.be) |
| 14:25.48 | AdvoWork | anyone knows what this actually means? Defines the expiration time (in seconds) of an subscription. The value is used in a SIP SUBSCRIBE message "Subscription Expiry" |
| 14:25.56 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 14:26.57 | leifmadsen | means how long until a new subscribe should be sent |
| 14:28.30 | cusco | thanks kaldemar |
| 14:29.05 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:29.13 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 14:29.39 | drmessano | Start > Settings > Control Panel |
| 14:29.46 | Katty | daaaaaaaannnnnyy! |
| 14:29.56 | Katty | hugs drmessano |
| 14:29.59 | drmessano | Hallo! |
| 14:30.03 | Katty | how's you and the mrs. |
| 14:30.07 | drmessano | hugs Katty |
| 14:30.19 | drmessano | Doing great.. Howz you? |
| 14:30.28 | Katty | ehn. life's complicated at the moment |
| 14:30.31 | Katty | but i bought a boat! |
| 14:30.39 | drmessano | ohhhhhhhh |
| 14:31.00 | drmessano | Im jealous |
| 14:31.00 | leifmadsen | yay boats |
| 14:31.56 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 14:32.05 | Katty | it's not a very fancy one. just one of those inflatible ones off amazon |
| 14:32.09 | Katty | but i can go exploring in it! |
| 14:32.26 | drmessano | Just remember what happened to Marshall, Will, and Holly |
| 14:32.27 | Katty | it should hold me until i have some place to store a For Real boat. |
| 14:33.44 | drmessano | Sounds awesome |
| 14:34.33 | Rolandow | those botes are fast! |
| 14:34.39 | Rolandow | boats |
| 14:34.42 | *** join/#asterisk Velluto (~stephen@dhcp-0-13-c4-f4-76-22.cpe.quickclic.net) |
| 14:34.51 | Katty | there are photos on my fb if you wanna go look at it danny |
| 14:34.58 | Rolandow | zodiac? |
| 14:35.57 | drmessano | Holy crap, thats pretty cool |
| 14:36.24 | *** join/#asterisk minotaur01 (minotaur01@S01060018e7f9c7df.hm.shawcable.net) |
| 14:37.02 | Katty | i thought so (= |
| 14:37.13 | Katty | my next purrchase is the sea eagle 370 |
| 14:37.53 | *** join/#asterisk zerohalo (~zerohalo@67.103.230.22) |
| 14:38.18 | Katty | can never have enough water toys. |
| 14:39.37 | drmessano | That looks like a lot of fun. i wish I had the cash to get on the water this year. I think bikes and all the accessories are going to be our summer treat |
| 14:41.10 | drmessano | bbiaf, headed to the office |
| 14:41.32 | *** join/#asterisk lunaphyte_ (~lunaphyte@unaffiliated/lunaphyte) |
| 14:41.49 | *** part/#asterisk lunaphyte_ (~lunaphyte@unaffiliated/lunaphyte) |
| 14:42.24 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
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| 15:12.00 | *** join/#asterisk Alex_Bkash (b4ea1593@gateway/web/freenode/ip.180.234.21.147) |
| 15:13.17 | *** join/#asterisk macjunkie (~paul@2607:f380:80c:995:c50:1020:c2b9:1532) |
| 15:13.39 | macjunkie | trying to figure out how to setup voicemail so when a user dials the extension from outside or from another extension they actually enter their extension and login |
| 15:13.46 | macjunkie | they just get prompted for a password |
| 15:15.50 | [TK]D-Fender | macjunkie, "core show application voicemailmain" |
| 15:16.11 | *** join/#asterisk Samuel_CTX (~svanderp@62-213-207-59.colo.housingcenter.be) |
| 15:16.39 | macjunkie | thanks just stumbled onto it on google too |
| 15:16.40 | macjunkie | thanks!!!! |
| 15:16.55 | macjunkie | is doing happy dance about probably being able to kill Audix soon |
| 15:17.22 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-dctunntsowztidta) |
| 15:17.38 | Alex_Bkash | I want to hire someone in asterisk+VPN project. If anyone is interested plz mail me. alex.jhon110@gmail.com |
| 15:18.46 | newtonr | Alex_Bkash: You may want to post that on http://lists.digium.com/mailman/listinfo/asterisk-biz |
| 15:24.25 | *** join/#asterisk savenger (~savenger@p5496E6A0.dip.t-dialin.net) |
| 15:25.39 | savenger | Hi guys, my asterisk is no opening port 5063 as specified in my config for tls. res_crypto is installed. I can see no SIP messages when setting "sip set debug on". UDP and TCP is working on different ports. Does TLS have to be the same port as TCP? |
| 15:25.49 | Katty | drmessano: why are you txtign me?! |
| 15:25.52 | Katty | drmessano: i'm right here! |
| 15:26.47 | malcolmd | tls should default to 5061 and tcp should default to 5060. you can't run them on the same port |
| 15:27.26 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-bftbjfnkepdwinnh) |
| 15:27.55 | savenger | malcolmd: I have set "tlsbindaddr=0.0.0.0:5063" that should do it, right? |
| 15:28.08 | savenger | there's already another process using TCP 5061 |
| 15:28.31 | malcolmd | should; you'll also need tlsenable=yes |
| 15:29.01 | savenger | malcolmd: I have so... |
| 15:29.04 | malcolmd | and look at the TLS settings section of the sample sip.conf file for other certificate settings |
| 15:30.56 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 15:33.18 | savenger | I followed: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial using an Asterisk 1.8.11-cert2 |
| 15:34.59 | thecardsmith | did you do a pcap to see if you're rejecting the packets from iptables, or something like that? |
| 15:35.02 | thecardsmith | savenger: ^^ |
| 15:35.35 | savenger | thecardsmith: I looked with netstat for an open port 5063, but it's not |
| 15:35.59 | savenger | And I found this in the log: "[Jun 21 16:58:15] WARNING[20879] chan_sip.c: No valid transports available, falling back to 'udp'. " But TCP is working. |
| 15:36.59 | savenger | and then something like this: PLEASE NOTE: Setting 'nat' for a peer/user that differs from the global setting can make the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users |
| 15:37.09 | savenger | will be sent to a different port than replies for an existing peer/user. If at all possible, use the global 'nat' setting and do not set 'nat' per peer/user. |
| 15:37.43 | *** join/#asterisk mintos (~mvaliyav@114.143.165.116) |
| 15:37.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v029-208.mobile.uci.edu) |
| 15:38.21 | thecardsmith | granted, i haven't used TLS on a non-standard port, but, in my setup, I do allow both plain sip on 5060 and tls on 5061, and in my general section i have a transport=udp |
| 15:38.27 | thecardsmith | but! in the [user_device] section i have: |
| 15:38.31 | thecardsmith | transport=tls |
| 15:40.00 | savenger | In the user section I have: transport=udp,tcp,tls. But shouldn't I see 5063 with "netstat -nap | grep 5063" ? |
| 15:40.47 | savenger | I can see udp and tcp ports in netstat and both is working |
| 15:40.51 | savenger | in asterisk |
| 15:43.35 | Qwell | What makes you think 5063 would be used? |
| 15:44.34 | savenger | <PROTECTED> |
| 15:46.02 | [TK]D-Fender | ~itsplist-us |
| 15:46.02 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 15:52.08 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:346e:1744:1493:9560) |
| 15:53.50 | drmessano | savenger: Which version Asterisk are you running? |
| 15:54.04 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:55.57 | savenger | drmessano: Asterisk 1.8.11-cert2 |
| 15:55.58 | thecardsmith | he's using 1.8.11-cert2 |
| 15:59.51 | drmessano | I was having some problems setting up TLS yesterday, and I do remember seeing a few bug reports for the port not working as part of the tlsbindaddr option. I cant even tell if those were current, but may be worth googling |
| 16:01.19 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
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| 16:04.16 | *** part/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 16:04.49 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 16:07.07 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 16:10.24 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-168.bois.qwest.net) |
| 16:10.27 | Hive | is there a setting in sip.conf to make a SIP trunk timeout if it is receiving dead air from the provider? I see 'rtptimeout' but im not exactly sure what 'RTP traffic' is :x |
| 16:10.47 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 16:11.54 | Qwell | Hive: Yes. Enable that. |
| 16:12.48 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:14.20 | Katty | QWELL |
| 16:14.22 | Katty | WHY I OUTTA |
| 16:14.23 | Katty | JUST |
| 16:14.27 | Katty | hug you. |
| 16:14.43 | Katty | hugs Qwell |
| 16:16.01 | *** join/#asterisk Roadblock_RVA (~Roadblock@static-72-84-192-5.rcmdva.fios.verizon.net) |
| 16:16.20 | Roadblock_RVA | hey all. I have an Asterisk 1.2 system (trixbox) and have a problem with the dialplan |
| 16:16.44 | Roadblock_RVA | If I only dial 7 digits it doesn't prepend 804 or 1804 and I'm not clear on why |
| 16:16.50 | drmessano | Thats not your only problem |
| 16:16.57 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:17.09 | Qwell | You could have stopped a "and have a problem" |
| 16:17.20 | Roadblock_RVA | lol |
| 16:17.32 | Roadblock_RVA | Yes I know. Not my system I'm working on for someone else. |
| 16:17.33 | Qwell | ~trixbox |
| 16:17.34 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 16:17.34 | drmessano | Trixbox? 1.2? Poor bastard |
| 16:17.35 | malcolmd | savenger: i've got that version of asterisk (nearly, i'm running out of branch, not a tagged release) and i followed the tutorial also (wrote it). it "works4me" |
| 16:17.40 | Qwell | ~upgradeasterisk |
| 16:17.40 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 16:17.48 | malcolmd | i've got tls running on port 5063 okay with blink registered and making calls |
| 16:18.15 | drmessano | Qwell, There needs to be a ~thatsold macro that runs all needed |
| 16:18.23 | Roadblock_RVA | Hmmm. I had hoped that a dial plan question would be generic enough to not require trixbox specifics. thanks anyway guys. |
| 16:18.32 | Qwell | It is not. |
| 16:18.59 | drmessano | Which version of freepbx is on there? |
| 16:19.07 | Qwell | none |
| 16:19.17 | drmessano | He said 1.2 |
| 16:19.18 | Qwell | trixbox doesn't use FreePBX :p |
| 16:19.22 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 16:19.23 | drmessano | Earlier versions did |
| 16:19.37 | Katty | your mom did. |
| 16:19.41 | Qwell | Before they stopped claiming it was FreePBX, maybe. |
| 16:19.41 | drmessano | I dont think they forked until 2.0 |
| 16:19.45 | Qwell | It still wasn't. |
| 16:19.55 | drmessano | Youre a picky snob |
| 16:20.04 | Qwell | just a regular snob |
| 16:20.07 | drmessano | It was as close to FreePBX as it could be with all the mods |
| 16:20.09 | Roadblock_RVA | yes it's still freepbx on their. |
| 16:20.19 | Roadblock_RVA | regular plain jane freepbx |
| 16:20.25 | drmessano | Ok, if its still "called FreePBX", its way too old to support |
| 16:20.31 | drmessano | We cant support it in #freepbx |
| 16:20.34 | drmessano | So.... |
| 16:20.59 | savenger | malcolmd: which exact revision are you using? |
| 16:21.19 | drmessano | Its not a "wont", honestly.. But I doubt many of us remember the ins and outs of FreePBX 2.5 or older. Its been a long time |
| 16:21.23 | Katty | you're both snobs! |
| 16:21.27 | Katty | Qwell didn't even say hi! |
| 16:21.31 | drmessano | lol |
| 16:21.38 | Katty | what a jerk. |
| 16:21.41 | Qwell | Katty: Did too! |
| 16:21.43 | Qwell | (hi) |
| 16:21.51 | Katty | ohai ;) |
| 16:24.01 | *** join/#asterisk tommy87it (~tommy87it@testing.assicurazione.it) |
| 16:24.45 | savenger | malcolmd: I have to go now, but I'll be back tomorrow. |
| 16:24.45 | drmessano | Ok, think I now have a working IRC client on the laptop at work. Now to rename the directory and shortcuts |
| 16:24.52 | drmessano | TPS Reports |
| 16:24.54 | drmessano | Perfect |
| 16:44.21 | Katty | danny |
| 16:44.25 | Katty | drmessano: poke |
| 16:44.26 | Katty | drmessano: --> |
| 16:44.36 | drmessano | poke |
| 16:46.41 | *** join/#asterisk Cubber (~ronny@64.8.69.51) |
| 16:47.23 | *** join/#asterisk Cubber (~ronny@64.8.69.51) |
| 16:48.51 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
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| 17:00.28 | *** join/#asterisk Jacke (jacke@valhalla.bofh.org) |
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| 17:09.02 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 17:09.34 | eXcAliBuR | i'm trying to get a rmx1500 polycom bridge to register with my asterisk, I can't find a place to change it's username... any ideas? |
| 17:10.38 | eXcAliBuR | [Jun 21 13:10:26] NOTICE[15648]: chan_sip.c:25788 handle_request_register: Registration from '"_dummy_tester1" <sip:_dummy_tester1@DomainName>' failed for '10.75.2.246:60000' - No matching peer found |
| 17:10.43 | eXcAliBuR | thats what i'm getting |
| 17:16.19 | *** join/#asterisk Alex_Bkash (b4ea257f@gateway/web/freenode/ip.180.234.37.127) |
| 17:25.07 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:43.24 | *** join/#asterisk citrusfizz (~IRC@c-24-245-46-10.hsd1.mn.comcast.net) |
| 17:44.04 | citrusfizz | i need a simple php script that will connect to the AMI and output current users and there status (on phone or available) |
| 17:44.12 | citrusfizz | i had a script like that but it broke when i went to 1.8 |
| 17:45.02 | citrusfizz | anyone have a simple script like that for 1.8? |
| 17:54.11 | *** part/#asterisk JimVanM (~JimVanM@76-10-137-116.dsl.teksavvy.com) |
| 18:02.46 | [TK]D-Fender | citrusfizz, Fix whatever minor difference caused it to break |
| 18:03.18 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 18:05.36 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 18:11.49 | *** join/#asterisk gusto (~gusto@ip-109-84-0-125.web.vodafone.de) |
| 18:14.32 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:14.32 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:18.07 | *** join/#asterisk koffel (koffel@173-167-212-105-ip-static.hfc.comcastbusiness.net) |
| 18:19.11 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 18:20.17 | koffel | i have Asterisk 1.8.11.0-1digium1~lucid and i use bandwidth voip service . does anyone would know why incoming callerid number would be unknown but cidname is correct. |
| 18:23.21 | thecardsmith | if it's a SIP provider, you can do a "sip set debug on" at the cli and read the headers in the invite -- if it's not in the headers, then it's on the far end |
| 18:23.32 | thecardsmith | then the problem* is on the far end |
| 18:24.23 | koffel | i will try the sip set |
| 18:29.39 | *** join/#asterisk McBoingBo (~McBoingBo@mail.hrsg.ca) |
| 18:30.33 | McBoingBo | if I have the full phone number that I want to redirect what do I change this line to? GotoIf($["${CALLERIDNUM:0:3}"="+55"]?s,8) - to this? GotoIf($["${CALLERIDNUM}"="18665555555"]?s,9) ? |
| 18:31.42 | koffel | thecardsmith this what it shows Call-ID: 3b798b73397a6d246f6f46340825b0f2@192.168.2.1:5060 |
| 18:32.46 | thecardsmith | McBoingBo: at a glance that looks right, the right hand expression asks if the callerid number is equal to your null number. the first expression just look at the first three digits |
| 18:33.14 | thecardsmith | koffel: lemme see what one of mine looks like, not sure which field, one sec.... |
| 18:33.15 | McBoingBo | yeah it does look right but I wanted another opinon hehe, thanks |
| 18:33.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:33.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:34.29 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:35.10 | thecardsmith | Here's what one of mine looks like: |
| 18:35.35 | thecardsmith | From: "doug" <sip:dougsecure@1.2.3.4>;tag=o1pdCk5z.6oW86TP4JFvCDeDQK3G8xTq |
| 18:36.42 | thecardsmith | it's from an incoming call from a Bria softphone, koffel ^^ |
| 18:38.51 | koffel | okay here what mine says from my cell |
| 18:38.56 | koffel | From: "WIRELESS CALLER " <sip:2157911121@192.168.2.1>;tag=as45716d50 |
| 18:39.18 | thecardsmith | there you go, looks like to number is in there |
| 18:39.21 | koffel | 21579111121 is my voip number |
| 18:39.24 | thecardsmith | s/to/the/ |
| 18:40.14 | koffel | the cardsmith is my voip number suppose to be there? |
| 18:40.24 | *** part/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 18:41.56 | thecardsmith | so you're not getting anything with a NoOp(${CALLERID(num)}) , koffel? |
| 18:42.07 | koffel | none |
| 18:43.22 | koffel | looks like i am not getting number but am getting name |
| 18:44.43 | thecardsmith | and does CALLERID(all) look like "WIRELESS CALLER" <> |
| 18:46.58 | koffel | here a log from my house land line |
| 18:47.35 | koffel | http://pastebin.com/KAXd3YX2 |
| 18:47.44 | [TK]D-Fender | McBoingBo, that is * 1.0 dialplan <- |
| 18:48.59 | thecardsmith | another thing to look at is: NoOp(${SIP_HEADER(From)}) -- you could extract it from that. But, if you're getting it there, part of me thinks it could be a typo in your extensions.conf, maybe, seems just weird (which I know you are aware of, lol) |
| 18:49.05 | McBoingBo | [TK]D-Fender, :P |
| 18:49.44 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 18:50.08 | McBoingBo | [TK]D-Fender, I inherited this old Asteisk build I am rebuilding shortly on latest and greatest Asterisk soon...sooo oooold |
| 18:54.41 | [TK]D-Fender | McBoingBo, For 1.0 that looks OK |
| 18:54.54 | [TK]D-Fender | McBoingBo, If all other circumstances are sane |
| 18:55.35 | newtonr | McBoingBo: you are moving your 1.0 installation to a more recent version of Asterisk ? |
| 18:56.08 | McBoingBo | [TK]D-Fender, oh yeah it works fine, thanks. I thought you were poking fun because of my decrepit Asterisk |
| 18:56.43 | [TK]D-Fender | Well.. it would fail on 1.4+ outright.. so it "could" have been bad depending on what you're running. |
| 18:57.28 | McBoingBo | never heard of the dialplan version being reffered to seperately |
| 18:57.50 | McBoingBo | *referred |
| 18:58.52 | [TK]D-Fender | can you clarify that? |
| 19:00.33 | newtonr | McBoingBo: By "1.0 dialplan" I believe [TK]D-Fender was meaning that your dialplan included elements deprecated in a newer version of Asterisk |
| 19:00.46 | McBoingBo | newtonr, yeah I get that now |
| 19:00.55 | [TK]D-Fender | Yes, that was the point |
| 19:01.18 | McBoingBo | ZZzzzzz |
| 19:01.44 | pigpen | I have an oddity. PolycomSoundPointIP-SPIP_450-UA/3.2.5.0508, when a call coming into the phone, the handset is picked up, but rather than answering the incoming call, it picks up a new line (ie: dialtone) |
| 19:02.01 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:02.04 | pigpen | Could someone help tell me how I foobar'ed the config to cause this? |
| 19:03.35 | newtonr | pigpen: Have you custom configured the phone itself? A default configuration for that phone shouldn't behave that way. |
| 19:04.10 | newtonr | *custom configured, beyond just SIP server, authentication, etc |
| 19:04.11 | pigpen | all phones are povisioned via the sip.cfg via the network. |
| 19:04.52 | pigpen | we have used this config for many locations, but this one is acting odd. I don't remember doing anything to cause this, but memory fades. |
| 19:05.55 | jpsharp | Could just be that the phone is broken |
| 19:06.51 | pigpen | hmm..just duplicated it. |
| 19:07.21 | pigpen | although older phones, 501's, one of my test phones does it, another does not. |
| 19:07.28 | pigpen | so at least I have something to diff against. |
| 19:07.51 | pigpen | I'll post what it as, but thanks the nudge. |
| 19:19.40 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:19.40 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:25.11 | pigpen | I think I just have screwed myself by bringing forward a sip.cfg file too many times and the 450 phone is not reading it right |
| 19:25.27 | pigpen | ie: I need to grab virgin configs and start over. |
| 19:25.57 | jpsharp | THe polycoms are extremely picky about their config file formatting. |
| 19:26.16 | *** join/#asterisk Neptu (~Neptu@41.141.24.50) |
| 19:26.28 | pigpen | yeah. I guess I have been lucky or have been more diligent about them in the years past. |
| 19:29.02 | *** join/#asterisk KNERD (~KNERD@cpe-72-178-173-150.rgv.res.rr.com) |
| 19:37.16 | pabelanger | tee hee |
| 19:37.17 | pabelanger | * anthm has quit (Quit: Avoiding Deadlock) |
| 19:40.08 | *** join/#asterisk Velluto (~stephen@dhcp-0-13-c4-f4-76-22.cpe.quickclic.net) |
| 19:42.13 | Velluto | Hi Everyone - I was wondering if someone could help me out with an error I'm getting with CDR? [Asterisk 10.5, MS SQL, FreeTDS 0.82] |
| 19:43.43 | jpsharp | What's theerror? |
| 19:44.28 | Velluto | the error is "CDR direct execute failed" - is there a way I could debug it? Also, Realtime is working with no issues using the same connection |
| 19:48.30 | *** join/#asterisk Azerus (~chatzilla@216.219.51.251) |
| 19:49.47 | *** join/#asterisk jgowdy (jgowdy@basharteg.com) |
| 19:50.10 | jgowdy | This old bug still exists somehow in Asterisk 10.5.1 |
| 19:50.12 | jgowdy | [Jun 21 12:39:18] ERROR[30608] res_timing_timerfd.c: Call to timerfd_gettime() error: Bad file descriptor |
| 19:50.32 | jgowdy | even with all the close(s->fd) fixing |
| 19:50.41 | _Corey_ | Velluto: log your SQL queries via the ODBC driver, you're probably missing something in the CDR query |
| 19:50.58 | leifmadsen | that ^^ |
| 19:51.05 | jgowdy | oh wrong channel |
| 19:51.10 | *** part/#asterisk jgowdy (jgowdy@basharteg.com) |
| 19:51.21 | Qwell | wonders what the appropriate channel for whining is? |
| 19:51.27 | leifmadsen | to debug database issues you usually need to enable the logging on the database side to see what it is getting and what it doesn't like |
| 19:51.31 | leifmadsen | Qwell: :) |
| 19:51.33 | _Corey_ | Velluto: my bet is a column is missing or something |
| 19:51.41 | leifmadsen | Qwell: you have your answer |
| 19:51.43 | _Corey_ | yeah, what leifmadsen said |
| 19:51.52 | Velluto | ok - thanks, i'll see if I can catch it |
| 20:04.43 | *** join/#asterisk bullium (~wbradshaw@216.68.250.27) |
| 20:06.57 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
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| 20:32.39 | FinboySlick | Is there a way to specify the source port on the asterisk side for connecting with a specific peer? My understanding is that port= will set the destination port and bindport= is global to all |
| 20:33.22 | pabelanger | no, asterisk should not care about the originating port |
| 20:35.10 | FinboySlick | pabelanger: Hmmm... This is mostly for a test but I think the other end has an issue with two accounts using the same source port and (obviously) the same destination as well. |
| 20:36.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
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| 21:11.20 | *** join/#asterisk General_Z0D (~Z0D@184.178.240.108) |
| 21:11.29 | General_Z0D | anyone know a way around the error WARNING[13003]: app_meetme.c:1097 build_conf: Unable to open pseudo device |
| 21:11.48 | Qwell | Do you have dahdi loaded? |
| 21:12.10 | General_Z0D | i believe so i am using debian squeeze and apt to install asterisk it says dahdi is installed |
| 21:12.27 | Qwell | lsmod | grep dahdi |
| 21:12.32 | pabelanger | WARNING != error |
| 21:12.45 | General_Z0D | no its not loaded |
| 21:12.48 | General_Z0D | its just installed i guess |
| 21:12.58 | General_Z0D | can you point me to a tutorial? |
| 21:12.59 | Qwell | General_Z0D: enable the init script to start at boot. |
| 21:13.04 | Qwell | umm... rc-update.d, I think? |
| 21:13.07 | General_Z0D | ok |
| 21:13.17 | Qwell | no, update-rc.d |
| 21:13.21 | pabelanger | well, if you installed it from that packages, it will get enabled by default |
| 21:13.29 | pabelanger | sudo service dahdi status |
| 21:13.53 | General_Z0D | /etc/dahdi/system.conf not found. Nothing to do. |
| 21:13.59 | Qwell | that's fine |
| 21:14.02 | Qwell | run the lsmod again |
| 21:14.16 | General_Z0D | still nada |
| 21:14.32 | General_Z0D | what is the syntax for update-rc.d |
| 21:14.35 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 21:15.57 | Qwell | update-rc.d dahdi enable, but pabelanger is right, in that it would've done so automatically. |
| 21:16.18 | General_Z0D | i never rebooted after i added the package if that matters |
| 21:16.30 | Qwell | General_Z0D: perhaps, but the service dahdi start would've loaded it |
| 21:16.47 | General_Z0D | lame |
| 21:16.58 | Qwell | well, should've |
| 21:17.08 | Qwell | What version of Asterisk? |
| 21:17.10 | General_Z0D | this is on a fresh install of debian too |
| 21:17.11 | General_Z0D | 1.6 |
| 21:17.16 | Qwell | ugh |
| 21:17.20 | General_Z0D | should i use a newer version 1.6 came with apt |
| 21:17.35 | General_Z0D | been so long i didnt want to mess with all the dependencies so i just used apt-get |
| 21:17.54 | General_Z0D | 1.6.2.9-2 |
| 21:18.01 | Qwell | double ugh |
| 21:18.10 | General_Z0D | what do you recomend? |
| 21:18.23 | General_Z0D | im setting up a call center |
| 21:18.31 | Qwell | Personally? Just use Asterisk 10, and use app_confbridge instead. |
| 21:18.49 | General_Z0D | is there a way to get that with package management or do i need to use source? |
| 21:19.02 | Qwell | shrugs |
| 21:19.12 | General_Z0D | ok i can do it from source no biggie |
| 21:19.17 | General_Z0D | i just want to use something that works proper |
| 21:19.45 | General_Z0D | cept when i apt-get remove asterisk --purge all the directories are still there |
| 21:20.14 | Qwell | source install will put them all back anyways |
| 21:20.29 | General_Z0D | ok |
| 21:21.00 | General_Z0D | thanks for the assistance |
| 21:21.49 | pabelanger | 1.8 is the only version packaged |
| 21:22.12 | pabelanger | since 10, is only 1 year support is won't make it into Debian / Ubuntu |
| 21:22.59 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vdfpeoilblcmxbxf) |
| 21:24.27 | General_Z0D | its all good im working on 10 now |
| 21:24.38 | General_Z0D | its bitching about sqlite3 and i already have that |
| 21:24.45 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:24.51 | [sr] | yellow |
| 21:25.24 | General_Z0D | forgot the dev package |
| 21:26.15 | General_Z0D | thanks a lot guys this looks a bit more promising menuselect didnt have issues with anything like before |
| 21:26.29 | General_Z0D | can i still use myh sip and iax confs from 1.6 |
| 21:28.21 | pabelanger | read CHANGES and UPGRADE.txt |
| 21:29.58 | General_Z0D | ok |
| 21:33.14 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-dctunntsowztidta) |
| 21:35.11 | General_Z0D | now were can i read about the confbridge app? |
| 21:35.20 | General_Z0D | with a sample |
| 21:40.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:40.47 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:41.27 | gusto | w8 w8 w8 |
| 21:41.44 | gusto | what? debian has 1.6 in its packages? on what version? |
| 21:41.49 | General_Z0D | squeeze |
| 21:41.58 | gusto | i do not believe |
| 21:42.03 | General_Z0D | i do |
| 21:42.07 | gusto | i have to look it up myself |
| 21:42.12 | gusto | squeeze is not that old |
| 21:42.15 | General_Z0D | go for it heje |
| 21:42.21 | General_Z0D | unless this host i lease from put it in there |
| 21:42.27 | General_Z0D | nfoservers.com is my provider |
| 21:42.34 | General_Z0D | got a dedicated with squeeze |
| 21:43.01 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:43.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:44.04 | gusto | ok |
| 21:44.19 | gusto | and there is not even squeeze-backports for asterisk |
| 21:44.26 | gusto | that's odd |
| 21:44.36 | General_Z0D | yup i thought so |
| 21:47.15 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 21:49.57 | General_Z0D | now were can i read about the confbridge app? |
| 21:50.00 | General_Z0D | with a sample |
| 21:50.02 | General_Z0D | ?? |
| 21:50.31 | General_Z0D | and why does oxycodone have to taste like ass |
| 21:50.42 | ChrisInSydney | http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge |
| 21:50.48 | gusto | ok, so i am with a good choice to opensuse, there is a repository for every asterisk version on every supported version of opensuse |
| 21:51.07 | gusto | http://download.opensuse.org/repositories/network:/telephony:/ |
| 21:51.11 | General_Z0D | nice but my provider is cheap its 20 bux a month they dont have many options |
| 21:51.19 | General_Z0D | i can use debian ubutnu server or centos |
| 21:51.33 | General_Z0D | i got asterisk 10.5 going now i just need a tutorial to set up confernecing |
| 21:52.04 | General_Z0D | 10.5 had less dependencies and less problems than 1.6 thats for sure |
| 21:52.11 | gusto | but there are not as much platforms as to debian, debian seems to support armel armhf mips mipsel powerpc s390 s390x sparc more |
| 21:52.19 | General_Z0D | yeah |
| 21:52.28 | General_Z0D | i like my hosting cost though |
| 21:52.30 | gusto | well |
| 21:52.34 | General_Z0D | to use anothe rOS i would need to pay a shitload more |
| 21:52.38 | gusto | i use centos for server |
| 21:53.00 | General_Z0D | im more familiar with debian and this is not my personal server this is for a community project |
| 21:53.09 | General_Z0D | so we opted for cheap vs stable |
| 21:53.32 | General_Z0D | we are setting up phone phreaking challenges |
| 21:54.02 | General_Z0D | asterisk is our internal pbx and we use freeswitch for the switching |
| 21:54.34 | General_Z0D | just need to get a partyline going with asterisk of some sort and i dont know this confbridge at all yet |
| 21:54.50 | gusto | well |
| 21:54.58 | ChrisInSydney | General_Z0D: Just jump in |
| 21:55.12 | General_Z0D | im not sure how the extension bneeds to be set is what i mean |
| 21:55.23 | General_Z0D | i dont hafve examples |
| 21:55.26 | gusto | but you have still two options to get asterisk on your debian ... either you compile it by your own, what also is an interesting experience, or you just use 1.6, that works too |
| 21:55.35 | General_Z0D | dude i already have asterisk going |
| 21:55.36 | General_Z0D | gusto |
| 21:55.39 | General_Z0D | 10.5 |
| 21:55.41 | General_Z0D | lol |
| 21:55.44 | General_Z0D | i said that twice |
| 21:55.48 | gusto | yes yes |
| 21:55.48 | General_Z0D | i just work fast |
| 21:55.57 | ChrisInSydney | Like Parlaiment / Funcadelic / George Clinton it makes a whole lot more sense once you have experienced it, rather than have somepone try to explain it to you :D |
| 21:55.57 | General_Z0D | oh you are still flogging the dead horse ok |
| 21:56.06 | citrusfizz | can i use the ami to see what number a user is connected to? |
| 21:56.11 | General_Z0D | i know asterisk well i just dont know this app |
| 21:56.15 | General_Z0D | grr |
| 21:56.24 | gusto | hey ppl |
| 21:56.58 | gusto | General_Z0D: where are you from? you seem to be far more in the west, right? |
| 21:57.09 | General_Z0D | far east depends im in florida |
| 21:57.30 | gusto | there is so hot in here, that i can barely see what is going on here, so forgive me when i do not get everything you already said |
| 21:57.31 | General_Z0D | gainesville, florida in the hospital |
| 21:57.40 | General_Z0D | its alright |
| 21:57.50 | General_Z0D | im just looking for an example extension for the confbridge app |
| 21:57.59 | General_Z0D | googling is not yielding much yet |
| 21:58.06 | gusto | General_Z0D: heh, are you in florida? |
| 21:58.26 | gusto | what hospital? |
| 21:58.45 | gusto | are you going to make a conference call? |
| 21:58.49 | FLeiXiuS | If I have an include "file.conf" in sip.conf can file.conf contain channels / registers ? |
| 21:59.21 | ChrisInSydney | FLeiXiuS: yes |
| 21:59.29 | ChrisInSydney | registers first |
| 21:59.36 | gusto | well, i am not sure, but i think it first looks up in the file and then in the included ones |
| 21:59.45 | ChrisInSydney | then peers / users / friends |
| 22:00.07 | FLeiXiuS | ChrisInSydney, that's what I thought but asterisk doesnt seem to be loading it in |
| 22:00.15 | ChrisInSydney | Check paths ? |
| 22:00.20 | gusto | ChrisInSydney: that must be followed also in the included files or have they to be included according to the file where they are included from? |
| 22:00.56 | General_Z0D | nevermind you suck at support gusto |
| 22:00.59 | *** part/#asterisk General_Z0D (~Z0D@184.178.240.108) |
| 22:01.11 | ChrisInSydney | AFAIK: If you have sip.conf include 1.conf, 2.con and 3.conf in that order, they will be loaded in that order |
| 22:01.18 | leifmadsen | true |
| 22:01.32 | gusto | heh, what does it mean "sucking at support"? |
| 22:01.51 | FLeiXiuS | ChrisInSydney, paths are right |
| 22:01.59 | ChrisInSydney | gusto: Dont worry its the oxycodone talking |
| 22:02.16 | gusto | ChrisInSydney: well, now i understnad even less |
| 22:02.43 | leifmadsen | basically he didn't like how you were helping and he threw a tantrum |
| 22:02.44 | ChrisInSydney | FLeiXiuS: unload module chan_sip; load module chan_sip |
| 22:02.49 | ChrisInSydney | then youll see what is going on |
| 22:02.51 | gusto | i do not understand english words for specific medical equipment |
| 22:02.55 | leifmadsen | module unload chan_sip.so |
| 22:02.58 | leifmadsen | module load chan_sip.so |
| 22:03.10 | ChrisInSydney | oxycodone is pain medication, he is in a hospital, in pain |
| 22:03.21 | ChrisInSydney | or he is a regisrar with a habit :-/ |
| 22:03.34 | leifmadsen | ok, dinner time |
| 22:03.36 | gusto | well, i am not that of an expert that i could help everyone, but that's not why ppl are here, right? |
| 22:03.40 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 22:03.49 | gusto | it's more about helping each other |
| 22:03.50 | ChrisInSydney | leifmadsen: Thanks, thats what meant. |
| 22:03.51 | FLeiXiuS | No equals in include...mmm |
| 22:04.05 | ChrisInSydney | Im in Aus, we are upside down to you so thats why it came out wrong |
| 22:04.17 | FLeiXiuS | include "/etc/asterisk/file.conf" |
| 22:04.25 | FLeiXiuS | Does that need to be => |
| 22:04.33 | ChrisInSydney | include => t46-extn-aliases |
| 22:04.39 | ChrisInSydney | includes the context |
| 22:04.47 | ChrisInSydney | #include t46/t46-extn-rockliffe.conf |
| 22:04.50 | ChrisInSydney | includes the file |
| 22:04.50 | FLeiXiuS | Ah! |
| 22:05.21 | gusto | so the answer to my question is, that the files have to be included in order according to the base file (sip.conf) |
| 22:05.24 | gusto | right? |
| 22:05.34 | leifmadsen | #include is different than include => |
| 22:05.41 | leifmadsen | former is for files, latter is for contexts |
| 22:05.43 | FLeiXiuS | Ah that makes sense. |
| 22:05.54 | ChrisInSydney | and you can nest |
| 22:05.59 | FLeiXiuS | So used to #'s being comments. |
| 22:06.06 | leifmadsen | ; this is a comment |
| 22:06.15 | FLeiXiuS | leifmadsen, in asterisk yes |
| 22:06.25 | gusto | well to programming languages one also can use #include |
| 22:06.34 | ChrisInSydney | if you #include 1.conf and 2.comf and 1.conf contains a #include 1.1.conf and 1.2.conf it will load 1, 1.1, 1.2, 2 |
| 22:06.40 | gusto | so no matter if it's a comment statement or not this '#' |
| 22:06.53 | ChrisInSydney | comments are ; |
| 22:07.05 | ChrisInSydney | in the asterisk scripting thingy |
| 22:07.18 | ChrisInSydney | # is a directive thingy sort of like C |
| 22:07.24 | gusto | yes, i know, i only meant generally |
| 22:07.26 | leifmadsen | ;-- this |
| 22:07.27 | gusto | yes, like in C |
| 22:07.33 | leifmadsen | is a multline comment --; |
| 22:07.40 | gusto | what is ';---'? |
| 22:07.45 | leifmadsen | ;--- is nothing |
| 22:07.49 | gusto | ah |
| 22:07.50 | leifmadsen | ;-- is the start of multiline |
| 22:07.51 | ChrisInSydney | leifmadsen: Cool, thanks |
| 22:08.00 | ChrisInSydney | like a /* */ pair |
| 22:08.03 | ChrisInSydney | ? |
| 22:08.03 | leifmadsen | notes all of this is documented :) |
| 22:08.07 | leifmadsen | ChrisInSydney: yes |
| 22:08.43 | ChrisInSydney | cool. Time to read up again. Been buried in *SER |
| 22:09.13 | ChrisInSydney | and Kamailio 3.3 is out |
| 22:10.08 | ChrisInSydney | Any way , must get back to the window I should have in focus |
| 22:10.16 | ChrisInSydney | X/ |
| 22:10.20 | gusto | well |
| 22:10.24 | ChrisInSydney | have fun all |
| 22:10.37 | citrusfizz | can i use the ami to see what number a user is connected to? |
| 22:10.41 | gusto | the goto() statement also looks like assembler |
| 22:11.52 | leifmadsen | gusto: you're welcome to write everything using an AGI() |
| 22:13.29 | gusto | leifmadsen: what is an AGI? |
| 22:14.07 | leifmadsen | Gugge: ;-- this |
| 22:14.07 | leifmadsen | <gusto> yes, like in C |
| 22:14.07 | leifmadsen | <leifmadsen> is a multline comment --; |
| 22:14.07 | leifmadsen | <gusto> what is ';---'? |
| 22:14.07 | leifmadsen | <leifmadsen> ;--- is nothing |
| 22:14.08 | leifmadsen | <gusto> ah |
| 22:14.10 | leifmadsen | oops |
| 22:14.22 | leifmadsen | gusto: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AGI.html |
| 22:15.08 | gusto | leifmadsen: i have that 3rd edition here, i just looked it up |
| 22:15.19 | leifmadsen | I'm not sure why you asked the question then :) |
| 22:15.24 | gusto | i did not get there right to interactive voice response |
| 22:15.30 | gusto | i did not |
| 22:15.45 | gusto | citrusfizz was |
| 22:16.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 22:16.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 22:16.18 | citrusfizz | gusto, ? |
| 22:16.47 | *** join/#asterisk tzafrir (~tzafrir@212.179.40.134) |
| 22:16.48 | leifmadsen | I was responding to: <gusto> the goto() statement also looks like assembler |
| 22:17.05 | gusto | ah |
| 22:17.31 | gusto | no, i was just referring to that similarities to other programming languages |
| 22:18.07 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 22:24.36 | gusto | so so |
| 22:36.35 | *** join/#asterisk General_Z0D (~Z0D@184.178.240.108) |
| 22:37.41 | citrusfizz | is there anyway to show what number a user is connected to with ami or a regular asterisk command/ |
| 22:37.43 | citrusfizz | ? |
| 22:39.46 | gusto | i do not understand that question |
| 22:40.06 | gusto | a user is connected to a "number" by dialplan, right? |
| 22:41.14 | General_Z0D | why can i not find examples of how to use confbridge out there is it that new? |
| 22:41.27 | General_Z0D | i just ewant to make a simple party line for my buddies to hop on and chat |
| 22:41.42 | citrusfizz | gusto, i am just trying to find what CID a user is connected to |
| 22:41.52 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
| 22:43.40 | gusto | General_Z0D: ok, w8 |
| 22:44.26 | gusto | chapter 10, meetme() |
| 22:44.34 | General_Z0D | is that in 10.5 still? |
| 22:44.39 | General_Z0D | i was told to use confbridge |
| 22:44.46 | General_Z0D | wish i could get a straight answer |
| 22:45.09 | WIMPy | meetme is available if you have dahdi. |
| 22:45.11 | gusto | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-7.html#meetmeConferencing |
| 22:45.33 | General_Z0D | i dont have dahdi apparently |
| 22:45.35 | General_Z0D | im on a vps |
| 22:45.52 | leifmadsen | then you'll need to install it |
| 22:45.54 | WIMPy | You could still have it. |
| 22:46.01 | leifmadsen | you don't need hardware |
| 22:46.04 | WIMPy | But you don't need it. |
| 22:46.04 | General_Z0D | there a tutorial for that you could point me to? |
| 22:46.11 | leifmadsen | for MeetMe() you do |
| 22:46.15 | leifmadsen | General_Z0D: www.asteriskdocs.org |
| 22:46.19 | leifmadsen | you just install it |
| 22:46.31 | WIMPy | Conffbridge offers more than meetme. |
| 22:46.31 | General_Z0D | ok i will try that thanks leifmadsen and WIMPy |
| 22:46.36 | General_Z0D | i just need simple conference |
| 22:46.38 | General_Z0D | thats all |
| 22:46.47 | General_Z0D | meet me worked fine for me before in the past well enough |
| 22:46.55 | General_Z0D | i just had a different server provider before |
| 22:47.09 | leifmadsen | WIMPy: that's true if you're on Asterisk 10+ |
| 22:47.26 | General_Z0D | didnt realize you were an author leifmadsen of that site |
| 22:47.30 | General_Z0D | nice work man |
| 22:47.42 | WIMPy | I'm back on trunk since yesterday. ;-) |
| 22:48.05 | leifmadsen | General_Z0D: I'm not -- just an imposter |
| 22:48.14 | leifmadsen | the real leifmadsen is way more handsome |
| 22:49.00 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 22:50.18 | General_Z0D | lol |
| 22:50.57 | gusto | https://wiki.asterisk.org/wiki/display/AST/Application_ConfBridge |
| 22:52.58 | leifmadsen | Asterisk 10 version: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_ConfBridge |
| 22:59.18 | General_Z0D | ok the last make config for dahdi said no hardware found is that ok? |
| 22:59.25 | General_Z0D | no error just that text |
| 22:59.43 | General_Z0D | now i should be able to load dahdi as a module correct with service dahdi stard? |
| 23:00.25 | General_Z0D | holy shit i would suck your dick leifmadsen |
| 23:00.28 | General_Z0D | it worked mate |
| 23:00.33 | WIMPy | NFI about service. Just modprobe dahdi. |
| 23:00.35 | General_Z0D | im not gay im just that happy thanks for all the help |
| 23:00.39 | General_Z0D | both you guys |
| 23:00.47 | General_Z0D | so now meetme should work again |
| 23:01.12 | gusto | so that confbridge does not work, or what is the status of it? |
| 23:02.03 | leifmadsen | General_Z0D: uhhhhh ya not necessary :) |
| 23:02.04 | WIMPy | The old version in 1.8 had limited functionality compared to meetme. |
| 23:02.05 | General_Z0D | cept now i get this |
| 23:02.06 | General_Z0D | WARNING[27067]: pbx.c:4359 pbx_extension_helper: No application 'Meetme' for extension (default, 8600, 1) |
| 23:02.07 | leifmadsen | just buy a book :D |
| 23:02.13 | WIMPy | The current version in 10+ is better. |
| 23:02.15 | General_Z0D | i am going to order it when i get home tomorrow |
| 23:02.17 | leifmadsen | WIMPy: +1 |
| 23:02.22 | General_Z0D | im using 10.5 |
| 23:02.39 | General_Z0D | do i need to compile that app in with menuselect? |
| 23:02.53 | WIMPy | Yes. |
| 23:02.54 | leifmadsen | all apps needs to be enabled in menuselect to be compiled.... |
| 23:02.59 | WIMPy | And don't forget to ./configure first. |
| 23:03.06 | General_Z0D | ok |
| 23:03.23 | WIMPy | Otherwise it won't know you have dahdi now. |
| 23:03.30 | General_Z0D | ok thaniks |
| 23:04.00 | gusto | well |
| 23:04.17 | gusto | with confbridge you need the bridge modules |
| 23:04.40 | gusto | maybe that is a problem that some distro's do not compile it in when they do a package |
| 23:05.14 | WIMPy | Dont't trust stuff you didn't do yurself. |
| 23:05.25 | WIMPy | +o |
| 23:06.07 | General_Z0D | doing the configure now |
| 23:06.15 | General_Z0D | gusto i think you are totally lost |
| 23:06.22 | General_Z0D | you are about 20 minutes behind in my convo |
| 23:06.43 | General_Z0D | ok in make menuselect i do not see app meetme listed |
| 23:06.48 | gusto | well, 20 minutes is OK |
| 23:06.59 | leifmadsen | General_Z0D: you will under Applications |
| 23:07.01 | General_Z0D | oh its in deprecated nevermind |
| 23:07.06 | General_Z0D | i didnt scroll far enough |
| 23:07.13 | gusto | some ppl, mainly politicians, have far more latency |
| 23:07.15 | General_Z0D | do i need dahdibridge or no? |
| 23:07.22 | gusto | some are even 20 years behind |
| 23:07.24 | General_Z0D | yah gusto talk to nobama |
| 23:07.56 | gusto | i do not think that obama would catch up with my fast 20 minuts |
| 23:08.01 | WIMPy | no |
| 23:08.04 | General_Z0D | he cant read |
| 23:08.14 | General_Z0D | funny watching him do speaches with the telepromter |
| 23:08.18 | General_Z0D | when it lags he lags even more |
| 23:08.44 | gusto | well, you can read with him, his speeches are online |
| 23:09.09 | General_Z0D | lol |
| 23:09.09 | gusto | like in an elementary school |
| 23:09.18 | General_Z0D | today we say change |
| 23:09.23 | General_Z0D | we need change |
| 23:09.27 | General_Z0D | im like dude i got 50 cents for yah |
| 23:09.35 | *** join/#asterisk jsjc (~Adium@175.43.62.126) |
| 23:10.19 | gusto | what exactly is a "dude"? |
| 23:10.25 | General_Z0D | i dunno |
| 23:10.28 | General_Z0D | a cowboy |
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| 23:10.34 | gusto | you see |
| 23:11.09 | gusto | so, i am going to bed, dude |
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| 23:13.12 | General_Z0D | ok i dont need to make samples again do i lief? |
| 23:13.16 | General_Z0D | leif* |
| 23:14.11 | WIMPy | No. |
| 23:14.18 | WIMPy | That would shred your configs |
| 23:14.20 | leifmadsen | not sure why you would.... |
| 23:14.46 | General_Z0D | thanks it all works now man |
| 23:14.53 | General_Z0D | you are definatly good with support and a people person |
| 23:15.07 | General_Z0D | everyone else tried telling me to use confbridge which over complicated things |
| 23:15.45 | WIMPy | Yes, I'd also do so. |
| 23:17.21 | *** join/#asterisk zenmaster (~zenmaster@unaffiliated/auckla) |
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| 23:18.06 | zenmaster | So my asterisk server is on a excellent fat big pip on a beast of a machine over at 1willshire courtesy of some friends in digital high places. |
| 23:18.40 | zenmaster | However my office is on a Verizon Fios 20x20mbps connection running through verizon fios provided router. It seems to be choking on about 30agents all VOIP. |
| 23:18.54 | zenmaster | Other than more bandwidth, what would be the recommendation for replacement router? |
| 23:19.19 | jpsharp | Are you doing NAT through your router? |
| 23:19.23 | leifmadsen | use a codec that uses less bandwidth |
| 23:19.23 | WIMPy | One that does't suck. |
| 23:19.24 | zenmaster | Yes. |
| 23:19.34 | leifmadsen | and ya, one that can handle the packet traffic |
| 23:19.37 | zenmaster | leifmadsen: I think we are on the 8k one. |
| 23:19.45 | leifmadsen | g729 I guess |
| 23:19.47 | zenmaster | Well I heard Cisco 3600 be thrown around. |
| 23:19.57 | leifmadsen | sounds like your router can't process things fast enough |
| 23:20.03 | zenmaster | leifmadsen: Thats it. |
| 23:20.05 | leifmadsen | separate out your firewall and router |
| 23:20.07 | zenmaster | I have money to replace it. |
| 23:20.17 | jpsharp | Run a VPN to the Asterisk box, rather than having to process switch a crap ton of packets through NAT. |
| 23:20.18 | General_Z0D | yeah g729 |
| 23:20.30 | leifmadsen | ClearOS box with Intel Gb network cards |
| 23:20.33 | zenmaster | Gentlmen gentlmen. Great idea's! |
| 23:20.37 | General_Z0D | i use ulaw personaly but for bandwidth 729 is good |
| 23:20.56 | zenmaster | Bandwidth wise 20mbps up and down is good or no? |
| 23:21.03 | General_Z0D | but i only have about 4 people that would be using my box |
| 23:21.41 | leifmadsen | bandwidth does't sound to be the problem, it's the amount of pps |
| 23:22.15 | jpsharp | Yeah, the router is choking trying to keep the NAT tables straight. |
| 23:22.38 | Nugget | god I hate NAT. I hate everything about it. |
| 23:24.01 | leifmadsen | Nugget: I hate everything about you! |
| 23:24.07 | leifmadsen | doesn't mean that at all |
| 23:24.08 | zenmaster | leifmadsen: Thank you. :) |
| 23:24.20 | leifmadsen | zenmaster: clearOS is pretty spectacular to be honest |
| 23:24.28 | Nugget | heh |
| 23:24.35 | zenmaster | leifmadsen: Well currently, my agents are using five9. |
| 23:25.07 | zenmaster | leifmadsen: And that is what is choking. I want to move them to my awesome asterisk setup, so I am planning ahead beacuse I don't want to repeat the problem. |
| 23:25.08 | jpsharp | So, get a beefy router, go the VPN route, or put a local Asterisk box at the office and trunk things to your 1Willshire box over IAX. |
| 23:25.37 | zenmaster | jpsharp: Recomend a beefy router? |
| 23:25.39 | leifmadsen | just don't expect transfers and things to work on the SIP side if you go the latter :) |
| 23:25.52 | leifmadsen | still recommends a ClearOS box with a good MB and network card |
| 23:26.04 | leifmadsen | it's going to have more processing power than pretty much anything else you can buy |
| 23:26.37 | jpsharp | Cisco 2811. Or a 3600 series. |
| 23:26.39 | zenmaster | leifmadsen: I do have a Dell Power Edge 2800. 16gb Memory, 2x Xeon 3.0ghz Quad Core? |
| 23:26.43 | zenmaster | Turn that into router clear os? |
| 23:26.50 | jpsharp | That's....over kill. |
| 23:26.53 | zenmaster | I am very comfortable with BSD, and *nix. |
| 23:26.58 | zenmaster | Linux. |
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| 23:27.52 | leifmadsen | zenmaster: give it a shot -- it's an end of life server anyways, so it can't hurt since it won't cost you anything |
| 23:27.58 | zenmaster | Hmm let me draw a picture for you guys. Of my theory, correct me of I am wrong. I am new to this telephone stuff. :D |
| 23:28.02 | leifmadsen | zenmaster: if it doesn't work, then you can go the hardware route |
| 23:28.07 | leifmadsen | you're wrong |
| 23:28.10 | leifmadsen | mocks the channels |
| 23:28.50 | zenmaster | leifmadsen: What do you think about the Cisco 3600 router? |
| 23:29.05 | carrar | old router |
| 23:29.06 | leifmadsen | never used it, so no opinion |
| 23:29.15 | carrar | 2811 is newer |
| 23:29.18 | leifmadsen | I haven't deployed a router in years :) |
| 23:29.29 | leifmadsen | I let IT monkeys handle that stuff |
| 23:29.31 | zenmaster | Let me draw picture. |
| 23:29.35 | zenmaster | For you guys test my theory. |
| 23:29.37 | leifmadsen | on Drawsome? |
| 23:30.48 | zenmaster | Draw a picture of my route idea. |
| 23:32.32 | carrar | zenmaster: http://www.cisco.com/web/partners/downloads/765/tools/quickreference/routerperformance.pdf |
| 23:37.51 | zenmaster | carrar: Than you. |
| 23:38.02 | zenmaster | Can you make sense of what is going on in that pdf? |
| 23:38.30 | carrar | yes |
| 23:38.46 | zenmaster | carrar: So model numbers and performance specs of possible purchases I may need. |
| 23:39.12 | zenmaster | carrar: I need to make a descion based on my current througput and projected throughput to make a good purchase decision right? |
| 23:39.19 | zenmaster | That what you are getting at? :D |
| 23:39.21 | carrar | no idea what you are doing |
| 23:40.33 | carrar | You probably need to hire a network engineer if you are trying to plan out something important |
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| 23:41.25 | carrar | as you will no doubtly need to account for the overhead of QoS Policies, access lists and whatever else you might want |
| 23:42.11 | carrar | or perhaps not use Cisco at all |
| 23:42.18 | carrar | perhaps use Juniper |
| 23:42.50 | carrar | or use huawei |
| 23:43.30 | jpsharp | Just get a Cisco 12000 & be done :) |
| 23:43.47 | carrar | heh |
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