| 00:11.29 | *** join/#asterisk gown (~phonetic@gateway/tor-sasl/martinphone) |
| 00:18.25 | leifmadsen | at least I think it copies stuff to a .bak file now |
| 00:18.29 | leifmadsen | but don't run it twice :) |
| 00:20.24 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:20.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:20.29 | leifmadsen | pabelanger: OHAI |
| 00:23.23 | pabelanger | leifmadsen: well hello there |
| 00:25.37 | file | ohai |
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| 01:56.27 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:56.29 | saint_ | hi all |
| 01:57.14 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 01:58.14 | saint_ | How can I Answer() a call, but then call an internal station with Dial(), and NOT pass the call through ? I would like Asterisk to call my station, and tell me "xxxx is calling" ... |
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| 01:58.28 | saint_ | Before I pass the call through. what's the best way to do that ? |
| 01:59.05 | leifmadsen | huh? |
| 01:59.17 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:59.30 | leifmadsen | what do you mean Dial() and not pass the call through? |
| 02:00.32 | saint_ | leifmadsen: I have an incoming call. I want to pick it up, and check out in a database if the caller id already exist. Let's say it does. The caller id xxxxxxx will have a wav file with the name of the caller . I want asterisk to Dial() one or more extension, say "xxx is calling" , and give me the choice to press 1 to pickup, 2 to send to voicemail or 3 to black list |
| 02:03.01 | pabelanger | app_followme |
| 02:03.13 | leifmadsen | you can do that with the M() or U() flags in Dial() easily |
| 02:05.55 | saint_ | leifmadsen: ha.. sweer.. i'll give it a shot. i forgot dial has all those options. |
| 02:06.05 | leifmadsen | never do that again |
| 02:07.46 | saint_ | leifmadsen: what ? |
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| 02:09.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:10.04 | [TK]D-Fender | saint_, Never forget apps have options. Or forget to go read them every time you think of asking "How can I XYZ?" |
| 02:10.16 | saint_ | thinking about it, i must be one of the oldest dudes on IRC nowadays.. |
| 02:27.35 | saint_ | leifmadsen: So if I use Dial with option U , how to I tranfer the caller to the callee then ? |
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| 02:34.31 | saint_ | leifmadsen: mmh...? |
| 02:39.20 | zerohalo | saint_: have you checked out the privacy manager (p option for Dial())? |
| 02:40.38 | saint_ | zerohalo: ha no.. |
| 02:40.47 | zerohalo | sounds like it's exactly what you want |
| 02:41.11 | saint_ | zerohalo: yeah. i've been reading through the lines of DIAL in order to go to the next section and finish the book. |
| 02:41.22 | saint_ | I guess i need to go back and study those options. thanks for thisone |
| 02:43.31 | zerohalo | errr - 'p' is screen and 'P' is privacy manager - you may want one or both of those. |
| 02:43.38 | zerohalo | sounds like 'p' |
| 02:51.43 | saint_ | zerohalo: i am trying p .. do I need to create the extensinos 2, 3, and 4 after the Dial ? |
| 02:54.10 | zerohalo | saint_: it's been a while - I don't believe so, but check http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial for a little more detail on the option |
| 02:54.23 | saint_ | I'm in it |
| 02:54.57 | saint_ | it says if the called party enter 2, 3, or 4 , it returns ${DIALSTATUS} with different values. |
| 02:55.12 | saint_ | I tried, and I have a Hangup() after the Dial(xx,,p) |
| 02:55.33 | saint_ | so I guess I need to do GotoIf(${DIALSTATUS)=xxx)? then, correct ? |
| 02:56.34 | zerohalo | right |
| 02:57.31 | zerohalo | ok - right - you need to check DIALSTATUS for anything but '1'. |
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| 02:58.55 | saint_ | zerohalo: thanks a ton for the trick. I'll do that tomorrow. Disconnecting now |
| 02:59.07 | zerohalo | yup - np. later |
| 03:00.12 | saint_ | dnite |
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| 03:20.52 | blizzow | I want to adjust my callerID to a different number depending on what number is dialed. EG: Dialing a number with a 347 area code will display my phone number from that area code. I currently have a context in my dialplan with hardcodes all CIDs to the same number. How do I adjust my context appropriately? Here is the current context calls are going through... http://pastebin.com/r4iB4xnt |
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| 03:25.39 | zerohalo | blizzow: I'd use AGI to run something against a db table that maps npas to callerids. |
| 03:26.39 | [TK]D-Fender | No need for AGI for a simple lookup |
| 03:28.23 | zerohalo | right - certainly lots of ways to skin that cat |
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| 03:57.37 | carrar | But only one way to fly a cat |
| 03:57.54 | carrar | http://i.imgur.com/jsgfD.jpg |
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| 04:19.47 | zerohalo | Copter cat is... creepy. |
| 04:29.45 | *** join/#asterisk Lepon (Lepon@60-241-109-208.static.tpgi.com.au) |
| 04:29.48 | Lepon | hey all |
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| 04:30.35 | ChannelZ | people hi |
| 04:31.06 | Lepon | quick question - well I hope its quick: I am dialing with a generated ringtone Dial(blah,60,r) and am wondering if there is a way to customised the generated ring tone. I tried changed indications.conf with no luck |
| 04:31.20 | Lepon | version: 1.8.11.1 |
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| 04:39.15 | ChannelZ | It should work |
| 04:40.44 | ChannelZ | You can make a test extension that answers and then does a Playtones and see if you configged it right |
| 04:49.14 | Lepon | I believe playtones is working just not the generated ring caused by the r flag in Dial |
| 04:49.21 | Lepon | ill double check though |
| 04:52.33 | Lepon | Yea, when I call a dummy channel and use Playtones(ring), wait(10) it plays it as I expected to hear it. |
| 04:52.57 | Lepon | When I used Dial(blah,x,r) I get the same generated ringtone as I have always got |
| 04:53.49 | ChannelZ | Is it a SIP channel? |
| 04:54.10 | ChannelZ | Most devices generate their own tones when told the call is making progress |
| 04:54.36 | ChannelZ | I'm not positive how it works if you haven't answered the channel yet (early media?) |
| 04:56.04 | Lepon | Yea it is a SIP channel |
| 04:56.19 | Lepon | at this stage we have answered the channel and played a message to the user |
| 04:57.09 | Lepon | we are then dialling out to connect them to another phone. our sip provider takes a very long time to connect moble calls and as such we get no feedback and lots of complaints. so we use the r flag to generate a ringing tone so they are happy something is happening |
| 04:57.26 | dpilon | it should work as long as you Answer() first |
| 04:58.05 | Lepon | must be missing something somewhere, I'll have a bit more of play. Thanks for all the help |
| 04:58.23 | ChannelZ | I'm trying over here to see what happens |
| 05:00.06 | Lepon | you are right it does work. |
| 05:00.53 | Lepon | the test system had been changed to remove the Answer() from the dial plan where as the live system has Answer() in. so when I was testing the indication changes on the test server it wasn't being answered first and hence not playing the correct tones |
| 05:01.17 | ChannelZ | yeah works here. |
| 05:01.27 | ChannelZ | Lovely british ring-ring |
| 05:02.35 | Lepon | That is great, thank you very much for all your help. we've been getting the US ring tone which sounds nothing like the AU ones (where I am) the indications file was never change to AU so nothing sounded right |
| 05:02.44 | Lepon | so much trouble for such an easy fix! |
| 05:25.39 | *** join/#asterisk jmfaubne (~jmf@123.200.155.137) |
| 05:25.45 | jmfaubne | hi all |
| 05:26.21 | jmfaubne | trying to locate a reason as to why I cannot add "incoming calling rules" via asterisk GUI |
| 05:26.47 | ChannelZ | #freepbx |
| 05:27.05 | jmfaubne | anyone know of this issue? running asterisk v10.11 and asterisk-gui.2.1 |
| 05:27.17 | tm1000 | ChannelZ: asteriskGUI != freepbx |
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| 05:34.35 | nny | o7 |
| 05:34.52 | nny | bought a vega 50 for a client, what a dissapointment :S |
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| 06:32.33 | joshuahh | hello, i am just wondering which file controls the "thank you for your patience" message? |
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| 06:33.21 | ChannelZ | thnk-u-for-patience.ulaw (or whatever) |
| 06:34.31 | joshuahh | Thanks, but where are those files stored in ? |
| 06:35.06 | joshuahh | thats the trouble i am having :/ |
| 06:35.12 | ChannelZ | Maybe but possibly not /var/lib/asterisk/sounds/en/ |
| 06:35.34 | joshuahh | i tried there :( |
| 06:35.41 | ChannelZ | try "locate thnk-u" in your shell |
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| 06:36.09 | joshuahh | hmm |
| 06:36.12 | joshuahh | locate: can not stat () `/var/lib/mlocate/mlocate.db': No such file or directory |
| 06:36.23 | ChannelZ | meh |
| 06:37.15 | joshuahh | ? |
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| 06:37.19 | joshuahh | sorry :( |
| 06:37.56 | ChannelZ | On the asterisk console, if you do "core show settings" where does it say the Data directory is? |
| 06:38.49 | joshuahh | <PROTECTED> |
| 06:39.43 | ChannelZ | ok well then that's where they should be, in sounds/en (unless you're using a different language or aren't using the localization layout in which case they'd be in a directory other than 'en' or right in the sounds directory, respectively) |
| 06:40.21 | joshuahh | ahh |
| 06:40.37 | joshuahh | it just stuck out to me (with all the 100's of files) |
| 06:40.39 | joshuahh | its called |
| 06:40.40 | joshuahh | queue-thankyou.gsm |
| 06:40.43 | joshuahh | i think :s |
| 06:41.53 | ChannelZ | that's "thank you for waiting" sounds like |
| 06:48.05 | joshuahh | ok lets see how we go |
| 06:48.07 | joshuahh | thanks! |
| 06:50.55 | nny | hmm |
| 06:51.32 | nny | I have a fresh install here, when I ran wanpipe asterisk is started via the dahdi setup script and worked. If I do asterisk -vvvvvvv it works, but the init script fails. advice? |
| 06:51.54 | nny | well |
| 06:52.11 | nny | the init script starts asterisk but /var/run/asterisk/asterisk.ctl is missing |
| 06:52.21 | nny | this is running as root, so it's not permission based afaik |
| 06:53.04 | nny | http://pastebin.com/hjz6WvJr |
| 06:53.09 | nny | only errors I can see in the log |
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| 06:58.04 | *** join/#asterisk din3sh (~din3sh@41.136.255.16) |
| 06:58.10 | din3sh | hello all |
| 06:58.29 | din3sh | anyone here been able to connect asterisk to nortel meridien? |
| 07:01.18 | nny | asterisk ctl is generated when asterisk is run with asterisk -vvvvvvvvvv but not with init script, odd |
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| 08:20.58 | din3sh | anyone here been able to connect asterisk to nortel meridian |
| 08:20.59 | din3sh | ? |
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| 08:29.51 | ChannelZ | Via what? Nortel doesn't scream "open standards" to me |
| 08:33.57 | din3sh | E1 |
| 08:34.03 | din3sh | PRI |
| 08:34.16 | din3sh | via an E1 cross over cable |
| 08:34.35 | din3sh | I am having lots of HDLC errors on the span and also line goes in alarm |
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| 08:50.11 | niklaswe | hello, any idé about this problem... I have setup confbridge and thats working create but.. when I trying to call it from my 3g iphone i dont get any playback sounds, but it works when Im connect to my wifi at home.. I can also call ppl from my phone.. |
| 08:50.17 | niklaswe | sorry for my bad english.. |
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| 09:13.57 | WIMPy | din3sh: Did you configute timing correctly? |
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| 09:47.42 | din3sh | WIMPy: sorry stepped out ealier |
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| 09:47.52 | schmidts | good morning |
| 09:48.51 | din3sh | span=2,0,0,ccs,hdb3 connected to nortel |
| 10:00.18 | kaldemar | din3sh: any particular reason for not using crc4? |
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| 10:38.17 | din3sh | kaldemar: i set the crc4 but it generated more errors |
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| 11:15.10 | Greenlight | When originating a call via the AMI is there a way to force asterisk to stay in the media path (eg don't native or remotely bridge) |
| 11:22.36 | kaldemar | Greenlight: it really has nothing to do with the origination. the call behaves in the way you have configured the participants. |
| 11:24.40 | Greenlight | But, for example with Dial, if you listen for DTMF you can prevent a native/remote bridxge |
| 11:25.27 | kaldemar | you can use all the same means if you want to. |
| 11:25.48 | Greenlight | I wasn't aware that was an avaialble option using originate |
| 11:26.08 | Greenlight | Or, do you mean, use the Dial application via |
| 11:26.09 | Greenlight | AMI |
| 11:34.15 | kaldemar | originate the call to a channel that does what you want. that might be a local channel that uses Dial as you wish. |
| 11:34.27 | Greenlight | Yea, I can't use a Local in this instance |
| 11:34.53 | Greenlight | Just wondered if there was a clever way to pass some arugment into Originate that I maybe wasn't aware of |
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| 11:49.34 | din3sh | i have a 2-span Digium card, i connect a cross-over cable from Span1 to Span2, i get |
| 11:49.35 | din3sh | [Feb 12 15:40:05] NOTICE[3478] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1 |
| 11:49.50 | din3sh | [Feb 12 15:40:05] NOTICE[3479] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 2 |
| 11:50.02 | din3sh | What am i doing wrong? |
| 11:51.33 | ariel_ | din3sh: one needs to be net the other one needs to be cpe |
| 11:52.08 | din3sh | already the case |
| 11:52.38 | Greenlight | 100% it's the correct cable ? |
| 11:53.02 | din3sh | straight cable does not work |
| 11:53.03 | ariel_ | and if it's digium cards just use a normal network cable try that |
| 11:53.47 | Greenlight | I thought it was just a normal cable too |
| 11:53.52 | din3sh | straight does not work |
| 11:53.55 | ariel_ | normal cable |
| 11:54.05 | din3sh | its a T1/E1 crossover cable |
| 11:54.10 | ariel_ | now connected to the nortel what settings is the nortel on |
| 11:54.12 | din3sh | pinouts 1245 |
| 11:54.49 | din3sh | settings you mean span 2 config for nortel or the config on the nortel itself? |
| 11:55.53 | ariel_ | settings on the nortel and on your card. |
| 11:56.25 | ariel_ | I have a Nortel with option 20 on it at the office and I am connected via 5ess settings |
| 11:56.39 | *** join/#asterisk madhatt3r (madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 11:56.43 | ariel_ | also at my other office we connect to that nortel with option 30 the same. |
| 11:56.58 | din3sh | my chan_dahdi.conf |
| 11:56.59 | din3sh | http://pastebin.com/hSKUPvBa |
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| 12:01.50 | niklaswe | which ports must i open to get asterisk to work.. I can call ppl but when the other person answer we cant hear eachother.. |
| 12:03.04 | madhatt3r | is the other person outside your lan? |
| 12:03.14 | kaldemar | ~sipnat |
| 12:03.15 | 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. |
| 12:03.18 | kaldemar | niklaswe: ^ |
| 12:03.57 | madhatt3r | as far as i've read this practice is deemed very unsafe...is it not then? |
| 12:05.09 | niklaswe | kaldemar: ah thanks I will look at it |
| 12:05.21 | longst | I have been investigating the integration of DAHDI TE110P for a while, I have a question, at moment my card jumper set to E1, it is supposed to have 32 channels, but when I do a "# dahdi_cfg -vv" it only shows 31 channels |
| 12:06.08 | *** join/#asterisk deo (~deo@122.52.128.123) |
| 12:06.27 | ariel_ | longst: one is a dchannel |
| 12:07.38 | *** join/#asterisk deo_ (~deo@203.177.9.66) |
| 12:09.02 | longst | I have tried googling this issue, I found some article saying channel 0 is D channel… then I would expect if I do a dahai_cfg -vv .it would present channel 0 |
| 12:09.04 | kaldemar | longst: the first (0) is reserved for framing. that's what you get. |
| 12:09.33 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:09.33 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:10.11 | din3sh | longst: its not really an issue, with E1, channel 16 is the D-channel |
| 12:10.38 | longst | http://pastebin.com/9eVLn7UV |
| 12:10.52 | longst | here is my my dahdi_conf -vv |
| 12:11.52 | din3sh | its not an issue |
| 12:12.13 | din3sh | i have same output on my production server |
| 12:13.40 | longst | I am following the introduction from http://www.cadvision.com/blanchas/Asterisk/DahdiDrivers.html. In this case, the card jumped to a T1, and in the bottom screen shot, it is said that 24 channels in total. But in my case if I try to use dahdi_tools it would say 31 channels in total. I am thinking if some of my hard ware configuration was wrong makes one channel missing ….. |
| 12:14.44 | Greenlight | Chan 16 is the B channel |
| 12:14.50 | Greenlight | *D |
| 12:15.06 | din3sh | 30 voice channels |
| 12:15.06 | Greenlight | It's supposed to look like that |
| 12:15.10 | Greenlight | Indeed |
| 12:15.23 | din3sh | 1 dchannel for framing |
| 12:15.39 | din3sh | One timeslot on the E1 is used for synchronization purposes and is not considered to be a B or D channel. |
| 12:15.49 | din3sh | so you cant have 32 channels |
| 12:15.59 | din3sh | stop looking for issues when there isnt one |
| 12:16.00 | din3sh | lol |
| 12:16.07 | Greenlight | E1 = 30 x 64kps chans |
| 12:16.21 | din3sh | 1-15 =B |
| 12:16.22 | longst | OK, but in this case I think the T1 card is supposed to have total 22 channels since 22 voice channels + 2 D channels = 24 channels ,what do you think ? |
| 12:16.24 | din3sh | 16=D |
| 12:16.31 | din3sh | 17-31 =B |
| 12:16.38 | *** join/#asterisk rolandow (~roland@s559700f1.adsl.online.nl) |
| 12:16.45 | *** join/#asterisk davlefouAMD (~david@197.15.73.137) |
| 12:16.51 | Greenlight | The pastebin you linked is E1, not T1 |
| 12:17.00 | Greenlight | T1 is 22 chans iirc (?) |
| 12:17.32 | Greenlight | It will also have a D channel, but you can't "use" that D channel either |
| 12:18.26 | rolandow | hi ... i transferred a cellphone to our inbound DND .. now my phone displays my number as callerid, but also displays "via", with the cellphone. |
| 12:18.37 | rolandow | what variable is this? i'd like to use this in my dialplan |
| 12:19.52 | longst | This is a screen short http://www.cadvision.com/blanchas/Asterisk/files/Dahdi_tool4.JPG from http://www.cadvision.com/blanchas/Asterisk/DahdiDrivers.html. This article introduces the way to set up T1 TE110P card. in this case. If data channel is not counted in total channels, I think this screens shot "Total/Conf/Act 24/ 6 /0 " does not make sense. |
| 12:20.42 | Greenlight | Perhaps T1 is 24 chans then |
| 12:20.57 | Greenlight | Or, perhaps that screenshot is E1 configured to use 24 chans |
| 12:21.22 | Greenlight | Remmeber, you can use *upto* the max, you don't have to. I've had E1 cicruits with 8 chans provisioned |
| 12:22.30 | longst | Greenlight: I don't think so, the context of this article is to set up TE110P with T1 in my mind, at least the writer is from North America... |
| 12:23.02 | Greenlight | T1 is 24 channels |
| 12:23.12 | Greenlight | E1 is 30 |
| 12:25.23 | longst | exactly. so according to http://www.cadvision.com/blanchas/Asterisk/files/Dahdi_tool4.JPG, data channels are counted… my question is, why my dahdi_tool does not count channel 0 data channel? |
| 12:25.41 | Greenlight | You're looking for a problem that doesn't exist |
| 12:25.59 | Greenlight | T1 would be 24 usable channels, PLUS a data/framing D-channel |
| 12:26.23 | Greenlight | Perhaps T1 has the D-channel as 0, whereas E1 it's channel 16 |
| 12:27.38 | Greenlight | I'm looking at that screenshot - where does it say data channels are counted? |
| 12:28.17 | madhatt3r | i seem to have XML Manager Event Interface missing from my asterisk web server on a elastix 2.3 installation...any clues? |
| 12:28.48 | longst | Greenlight according to this http://www.aculab.com/questions/how-many-voice-channels-does-an-e1-primary-rate-interface-pri-have/ it is said channel 0 and channel 16 are D channels. and Channel 0 is used for sync… and since I am trying to connect my Asterisk with TE110P and TE405P board to a GSM gateway. in the GSM gateway site. the GSM gateway PRI port has Lay2 lamp flash all the time. someone told me if LAY2 lamp blinking, most |
| 12:28.49 | longst | likely the connection does not set synchronisation clock properly.. |
| 12:32.15 | Greenlight | So, both are set to E1, and one is set to CPE and one NET? And the NET is set to provide timing, while the CPE is set to use timing from NET ? |
| 12:33.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:34.34 | longst | Greenlight. exactly, both E1, GSM gateway is NT and set to master (NET) , Asterisk is TE set to slave (CPE). |
| 12:35.52 | kaldemar | longst: you need to configure clocks in DAHDI aswell. |
| 12:36.28 | Greenlight | Indeed |
| 12:36.44 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 12:37.44 | Greenlight | It's in those settings against each span, the three numbers |
| 12:37.54 | Greenlight | One of them represents the timing source |
| 12:37.59 | Greenlight | iirc |
| 12:40.02 | kaldemar | span=<span_number>,<clock_setting>,<lbo>,... |
| 12:40.24 | Greenlight | That's the ones |
| 12:41.59 | longst | yes I have those configurations. actually "span=1,1,0,ccs,hdb3,crc4" in /etc/dahdi/system.conf |
| 12:44.28 | Greenlight | That looks right enough |
| 12:44.39 | Greenlight | Maybe your GSM gateway is the problem ? |
| 12:48.10 | longst | Greenlight: oh…. hopefully not because of this .. |
| 12:50.36 | longst | Greenlight: if I understand correctly, span=<span_number>,<clock_setting>,<lbo>,... if I have setting like span=1,1,0,ccs,hdb3,crc4, it means this span is going to use clock from GSM gateway in my case |
| 12:50.56 | Greenlight | Yup, exactly |
| 12:51.25 | Greenlight | Your GSM gateway is definetly setup as E1, not T1 etc ? |
| 12:51.45 | longst | Yes, I am 100% sure |
| 12:52.18 | Greenlight | You've tried another cable, just incase ? |
| 12:52.29 | longst | more over in /etc/dahdi/system.conf it has "bchan=1-15,17-31 |
| 12:52.30 | longst | dchan=16" |
| 12:52.59 | Greenlight | Yea, that's right |
| 12:53.00 | longst | so basically it means the channel 0 is not used at all... |
| 12:53.26 | kaldemar | no, it means you don't configure it. |
| 12:55.40 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.139) |
| 12:55.45 | longst | kaldemar http://pastebin.com/Gs7fTiRT here is my dahdi-channels.conf |
| 12:56.28 | kaldemar | longst: what is your issue again? |
| 12:57.27 | *** join/#asterisk rolandow (~roland@s559700f1.adsl.online.nl) |
| 12:57.52 | *** join/#asterisk barbosa2 (~juliano.b@177.157.83.193) |
| 12:57.59 | longst | I am trying to connect my Asterisk with TE110P and TE405P board to a GSM gateway. in the GSM gateway site. the GSM gateway PRI port has Lay2 lamp flash all the time. someone told me if LAY2 lamp blinking, most likely the connection does not set synchronisation clock properly.. according to http://www.aculab.com/questions/how-many-voice-channels-does-an-e1-primary-rate-interface-pri-have/ "Channel 0 - D-channel (Us |
| 12:57.59 | rolandow | hi, my asterisk 11 is missing SendDTMF, i checked with core show applications .. but when i do "make menuselect" in my source dir, i see that the app is selected |
| 12:58.01 | longst | mainly for clocking and synchronisation)" I am thinking most likely my D channel is not properly configured |
| 12:58.03 | kaldemar | longst: on your last pastebin, the context=default and group=63 have no effect whatsoever on channels 1-15,17-31. |
| 12:58.17 | pbxbrian | Hello all - is it possible to increase the last digit timeout when using overlap dialling with dahdi ? |
| 12:58.21 | rolandow | so why isn't SendDTMF included? :) |
| 12:58.30 | pbxbrian | IT seems to be hardcoded to 3 seconds (which would be odd) |
| 12:58.49 | kaldemar | longst: in chan_dahdi.conf, only the parameters that are above channel lines apply for channel(s). |
| 12:59.32 | *** join/#asterisk barbosa2 (~juliano.b@177.157.83.193) |
| 12:59.39 | longst | kaldemar: may I ask what is group=63 means in this case? |
| 12:59.44 | kaldemar | longst: is that dahdi-channels.conf of yours included in chan_dahdi.conf? |
| 13:00.03 | kaldemar | longst: it's about grouping channels so that you can use a group of channels when you dial. |
| 13:00.11 | *** join/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 13:01.58 | longst | kaldemar: thanks. and you mentioned about I didn't properly set D channel 0, I am wondering how I could set D channel 0? thanks |
| 13:02.59 | kaldemar | longst: you don't. |
| 13:03.15 | kaldemar | longst: you can't. |
| 13:03.35 | rolandow | so does anybody have an idea why i can't use SendDTMF? :) |
| 13:04.30 | longst | rolandow: what happens on CLI when "SendDTMF" is called in "dial plan" |
| 13:04.37 | kaldemar | rolandow: maybe you have not recompiled, reinstalled or both. |
| 13:04.44 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 13:05.30 | rolandow | longst: [2013-02-12 14:05:12] WARNING[11510][C-00000001]: pbx.c:4621 pbx_extension_helper: No application 'SendDTMF' for extension (voysincoming, 31202610074, 14) |
| 13:06.08 | rolandow | kaldemar: it must have been selected the very first time i compiled.. but i did a make & make install anyways |
| 13:07.47 | rolandow | is there some sort of dependency for SendDTMF ? |
| 13:07.58 | kaldemar | rolandow: do you have app_senddtmf.so under the module dir (default is /usr/lib/asterisk/modules)? if so, what does "module load app_senddtmf.so" tell you? |
| 13:08.00 | longst | rolandow: if you check "CLI> core show functions " |
| 13:08.19 | kaldemar | longst: it is not a function, but an application. |
| 13:09.20 | longst | kaldemar: oh sorry, yes, if you find SendDTMF in core show application? |
| 13:10.53 | kaldemar | i think he started by saying that the command does not list the app. |
| 13:11.37 | rolandow | kaldemar: yes, i do have the app in the folder.. i'll try to load it |
| 13:12.13 | rolandow | well it says loaded.. must have excluded it in the apps list then :s |
| 13:12.44 | rolandow | ok .. that's it .. thanks kaldemar :) |
| 13:13.44 | *** join/#asterisk Skavin (~kevins@202-180-69-27.me.tranzpeer.net) |
| 13:20.04 | Skavin | I am having troubles with GROUP_COUNT and Dial(local/ |
| 13:20.49 | Skavin | I have a set up where I have a SIP call coming in and then do a dial local then a dial SIP |
| 13:21.11 | *** part/#asterisk ariel_ (uid3533@pdpc/supporter/active/abatista) |
| 13:21.39 | leifmadsen | Skavin: that makes sense since the GROUP() is applied to the channel before the Local. If you're doing a GROUP_COUNT() in the Local channel dialplan, those are not related as the Local channel is separate of the initial channel. |
| 13:21.59 | leifmadsen | Skavin: you should start with what you're trying to accomplish |
| 13:23.04 | Skavin | I set the group in the local extension then try to clear the group in the h part of the dial plan |
| 13:23.12 | longst | kaldemar: "longst: so basically it means the channel 0 is not used at all... and then you said kaldemar: no, it means you don't configure it." but still I confused that how I could config D channel 0 |
| 13:23.35 | *** join/#asterisk bpietro (~bpietro@82.51.236.132) |
| 13:23.53 | kaldemar | Skavin: also, if /n is not used with the local channel, it does not stay on the path. |
| 13:24.11 | Skavin | this works most of the time but every so often the h for the local extension triggers after the h of the sip extension and I get a non 0 group count |
| 13:24.19 | kaldemar | longst: you can't configure it. that's what i meant. you don't configure it in any way. |
| 13:24.52 | kaldemar | longst: i also meant that your conclusion of the channel not being used at all was wrong. |
| 13:26.24 | Skavin | OK what I want to do is store the number of active calls for an extension in an SQL database. |
| 13:26.26 | leifmadsen | Skavin: not sure how you're clearing it, because the clearing is done automatically when the channel goes away |
| 13:26.27 | longst | kaldemar: according to this http://www.aculab.com/questions/how-many-voice-channels-does-an-e1-primary-rate-interface-pri-have/ it is said channel 0 will be used in " mainly for clocking and synchronisation" what do you think |
| 13:26.38 | leifmadsen | oh.. |
| 13:26.45 | leifmadsen | ya that approach is unlikely to work clean |
| 13:26.58 | leifmadsen | you're always going to run into situations where the number in the database is inaccurate |
| 13:27.14 | kaldemar | longst: i told you that it is reserved for framing over an hour ago. |
| 13:27.39 | leifmadsen | you're much better off requesting the data via the AMI when you need it |
| 13:30.42 | longst | kaldemar: so in this case, a basically question, how many channels an E1 span has on TE405P card, may I ask. (I am quite new comer in ISDN PRI) |
| 13:31.09 | Skavin | my current approach is to have a mysql table keyed with extension and server name with a count of calls then in asterisk Set(GROUP(${BPARTY})=ALLBOUND_GROUP) and update the db just before the sip dial and then in the h Set(GROUP(${BPARTY})=) and update the db |
| 13:31.32 | leifmadsen | Skavin: yes, and I'm saying that approach is flawed |
| 13:31.33 | kaldemar | longst: still 31 is what you have, including the d-channel, which is 16. 30 is what you have for calls. |
| 13:31.37 | leifmadsen | I've done it before; it doesn't work |
| 13:31.50 | leifmadsen | it's a nice idea, but it doesn't really work in practice very well |
| 13:32.18 | leifmadsen | when you need to know the values, you should just request them via AMI or some other method |
| 13:33.29 | kaldemar | you don't need to explicitly unset the group for a channel. |
| 13:33.49 | leifmadsen | kaldemar: he does since he's trying to update the database with the new value on hangup |
| 13:34.09 | longst | Oh…I see. I always made assumption that a E1 span should have 32 channels. I think my assumption was wrong at beginning.. |
| 13:36.07 | kaldemar | longst: well, it does. but your assumption on their usage in DAHDI and asterisk was wrong. |
| 13:36.55 | Skavin | AMI :/ sounds painful I currently have 8 servers in my cluster polling all of them for every call will cause major overhead |
| 13:37.53 | Skavin | and I don't think I can make an ANI connection from inside normal dial plan. |
| 13:38.01 | Greenlight | Skavin: You could always just monitor AMI and track it that way, rather than polling |
| 13:38.29 | leifmadsen | you can try the other approach, I'm just saying using the database to track calls is a surefire way to fail at accuracy |
| 13:38.57 | longst | kaldemar: oh… so it means a standard ISDN PRI has 32 channels but DAHDI has something difference only has 31 channels |
| 13:39.43 | Greenlight | longst: No |
| 13:40.25 | longst | Greenlight, kaldemar: very confused ... |
| 13:40.29 | *** join/#asterisk [TK]D-Fender (~Joe@v5208878.static2.cidc.net) |
| 13:40.40 | Greenlight | You're getting hung up on this channel 0 thing |
| 13:40.58 | Greenlight | Hmm bad choice of wording, getting confused by |
| 13:41.17 | kaldemar | longst: no. an E1 PRI gives you 30 channels for calls. |
| 13:41.30 | kaldemar | longst: channel is not necessarily a channel. |
| 13:41.30 | Greenlight | Plus a D-channel for framing. |
| 13:41.43 | leifmadsen | it's 32 channels less 2 channels for signally, leaving 30 channels for voice traffic |
| 13:41.50 | leifmadsen | signalling* |
| 13:42.04 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:42.57 | longst | kaldemar: "channel is not necessarily a channel." so is channel something else? |
| 13:43.26 | *** join/#asterisk barbosa2 (~juliano.b@177.157.83.193) |
| 13:43.33 | kaldemar | longst: in other words, a timeslot is not necessarily a voice channel. |
| 13:45.28 | longst | kaldemar: yes this I understand, signalling can be a channel. |
| 13:50.29 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 13:51.25 | Skavin | leifmadsen: Thanks. I will have a play with AMI and see if I can get it to work but not this morning as its 02:50 and I need sleep |
| 13:51.49 | SuperNull | awesome asterisk coredumps when you enter an invalid pin # for a conference |
| 13:51.57 | longst | kaldemar: but in my case connect GSM Gateway with DAHDI TE405P card. say if GSM gateway expect PRI channel 0 for sending synchronisation clock…But DAHDI TE405P card does not use channel 0, do you think it is possible to connect GSM gateway with Asterisk in my case |
| 13:53.46 | jmetro | SuperNull> old version? |
| 13:55.40 | *** join/#asterisk serafie (~erin@nat/digium/x-jnakkayubusyvauu) |
| 13:56.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:56.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:58.03 | longst | Greenlight: but in my case connect GSM Gateway with DAHDI TE405P card. say if GSM gateway expect PRI channel 0 for sending synchronisation clock…But DAHDI TE405P card does not use channel 0, do you think it is possible to connect GSM gateway with Asterisk in my case |
| 13:59.01 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:00.44 | longst | oh… another general question: if DAHDI TE405P card would be able to connect to any type of PRI terminal. |
| 14:01.26 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:7895:14d2:9d6d:6107) |
| 14:03.25 | kaldemar | longst: what gives you the idea that the card does not use timeslot 0? |
| 14:03.30 | SuperNull | jmetro yeah.. and it was a hack as in manual move from another near identical box |
| 14:04.24 | SuperNull | sux tho.. i literally cant make it work with 11 due to major upgrades being required. |
| 14:05.15 | longst | kaldemar: according to |
| 14:05.50 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:05.56 | *** join/#asterisk Sidrov (~Sidrov@ACBE2E2E.ipt.aol.com) |
| 14:06.04 | Sidrov | hello everybody |
| 14:06.13 | longst | kaldemar: according to dahdi_tool, it is said 31 channel total, and "dahdi_cfg -vv" it starts from "Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)" |
| 14:06.31 | Sidrov | is there any way in asterisk to catch Service Unavailable message with an agi script ? |
| 14:07.00 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:07.27 | kaldemar | longst: just let the channel count go already. dahdi_tool does not list it and it is not configured in system.conf nor chan_dahdi.conf. move on to your real problem. |
| 14:08.11 | longst | And I tried to alter "/etc/dahdi/system.conf" dchan=16 to dchan=0, 16. But after this change do a "service dahdi restart" it complains "Channel must be between 1 and 1023 (not '0')" |
| 14:08.47 | *** join/#asterisk bmg505 (~leon@196.209.99.237) |
| 14:09.33 | malcolmd | the synchronization timeslot isn't a delta channel. for an e1, the delta channel should be channel 16 |
| 14:12.36 | Greenlight | longst: When you try and connect them, what errors are you getting on the DAHDI/Asterisk side ? |
| 14:13.11 | longst | kaldemar: problem is I am trying to connect a GSM Gateway (NT) master PRI(E1) to a Asterisk with TE405P (TE) slave PRI (E1) with a straight cable CAT 5 cable. And my GSM Gateway PRI LAY 2 always blinking. I try to isolate the problem, and make sure the GSM Gateway well functionally, what I did is try to connect the same GSM Gateway (NT) master PRI (E1) to another GSM Gateway (TE) slave PRI (E1) with same straight cable, then LAY 2 |
| 14:13.12 | longst | keep light up … so I think something wrong with my Asterisk and TE405P problem. … |
| 14:13.20 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:13.39 | kaldemar | longst: define "straight" cable. what is the pinout? |
| 14:14.04 | *** join/#asterisk mihamina (~mihamina@25.213.74.41-ip-dyn.orange.mg) |
| 14:15.14 | Greenlight | Perhaps you need: http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 14:17.33 | malcolmd | yeah, don't use a straight through cable. 1+2, 4+5, cross them. |
| 14:18.16 | longst | straight cable means two end : 1: WHITE/orange 2: ORANGE/white 3: WHITE/green 4: BLUE/white 5: WHITE/blue 6: |
| 14:18.20 | longst | GREEN/white 7: WHITE/brown 8: BROWN/white |
| 14:19.06 | Greenlight | Did your GSM gateway not come with a suitable cable? |
| 14:20.05 | longst | Greenlight: actually, I think if you connect NT to NT (or TE to TE)device you want a 1+2 4+5 cross, but if you want to connect NT to TE you could use straight cable... |
| 14:20.33 | Sidrov | how to catch congestion with an agi script, if the dialplan doesn't get executed until answer ? |
| 14:22.09 | longst | Greenlight: malcolmd: I made a video describe my problem http://www.youtube.com/watch?v=q6spVJCv6FY |
| 14:22.29 | [TK]D-Fender | Sidrov, you don't AGI is not a background thing |
| 14:22.53 | Greenlight | I thought your issue was a blinking layer 2 light? |
| 14:23.26 | [TK]D-Fender | Sidrov, And if you have congestion, then it isn't going to answer and the dial has already aborted and you're back in the dialplan anyway |
| 14:25.04 | kaldemar | longst: find out what the pinout on your gateway is so you don't have to guess or assume. you have shown correct configs and the behavior you have described can be caused by incorrect wiring. |
| 14:25.28 | Greenlight | longst: http://asteriskfaqs.org/2013/02/11/asterisk-users/about-asterisk-with-digium-te405p-pri-isdn-cad.html <-- sounds very like your issue |
| 14:26.13 | longst | Greenlight: malcolmd: kaldemar: you will see first I try to connect top (Blue GSMGateway PRI 1 [NT E1 PRI Master]) with bottom (white GSMGateway PRI 2 [TE E1 PRI Slave]) with straight cable (two end have 1: WHITE/orange 2: ORANGE/white 3: WHITE/green 4: BLUE/white 5: WHITE/blue 6: GREEN/white 7: WHITE/brown 8: BROWN/white), after few second, LAY 2 lamp stable. But afterward, I connect another straight cable with an TE405P (TE E1 PR |
| 14:26.14 | longst | slave) , LAY 2 lamp alway blinking…. |
| 14:26.35 | Sidrov | [TK]D-Fender i get some external command to initiate the call via spool; if the call fail to connect, then i have no feedback |
| 14:26.37 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:26.59 | *** join/#asterisk denisnikulin (~dni@62.189.81.222) |
| 14:28.33 | longst | kaldemar: I am sorry, wish I could have another basic question, what do you mean by "pinout" on GSMway... |
| 14:28.43 | Katty | woo! contacts in! |
| 14:28.57 | Katty | spacial recognition is a bit wonky |
| 14:29.23 | longst | Greenlight: actually, http://asteriskfaqs.org/2013/02/11/asterisk-users/about-asterisk-with-digium-te405p-pri-isdn-cad.html this is my POST….. |
| 14:29.30 | Greenlight | Oh LOL |
| 14:29.32 | kaldemar | longst: http://www.juniper.net/techpubs/hardware/m40/m40-hwguide/html/pinout4.html |
| 14:29.54 | Greenlight | Well, no wonder he has the same problem as you! |
| 14:30.32 | Greenlight | longst: DO you have some RJ45 crimpers ? |
| 14:31.20 | *** join/#asterisk capnkooc (~capnkooc@ip25-1-15-186.ct.co.cr) |
| 14:31.34 | longst | Greenlight: yes I have. and I am 100% sure RJ 45 cable works…since I tested with tester…. |
| 14:31.53 | Greenlight | Ok, then make this cable up: http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 14:32.01 | Greenlight | Plug it in, and I bet it works |
| 14:32.14 | capnkooc | Hi I think i need some guidance maybe because I really don't know how to search for this kind of thing, I |
| 14:32.19 | capnkooc | m not a SIP expert |
| 14:32.24 | longst | kaldemar: your link is something about RJ-48... |
| 14:32.37 | Greenlight | He was describing what a pinout was |
| 14:32.59 | kaldemar | longst: still a pinout. do you get what it means? |
| 14:33.17 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pacsnnbhjzerfyrx) |
| 14:33.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:33.46 | longst | Greenlight: I made crossover cable as well.. but according to my understanding if you want to connect NT to TE device, straight cabe should be used. correct? |
| 14:33.46 | capnkooc | I have some asterisk machine but when I make a call thro a DAHDI device it reports the ringing time on the billsecs on the CDR |
| 14:34.50 | Greenlight | capnkooc: And the line defeinetly hasn't been answered? |
| 14:35.05 | capnkooc | have anyone a good advice to me to remove the ring time an only report in the CDR the correct time when the call is answer? |
| 14:35.31 | capnkooc | Greenlight, yes the line "reports" the be answered but it's only ringing |
| 14:35.34 | kaldemar | longst: don't assume that your TE405P and the gateway have the same pinout. |
| 14:36.29 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-uuwrpdezejacterl) |
| 14:36.32 | Greenlight | capnkooc: It depends what you're calling, just because you hear "ringing" doesn't mean the line's not been answered. IVR's and stuff can sometimes answere the line and then have it ring. You've tried with a normal landline desination to be sure ? |
| 14:36.36 | longst | kaldemar: Oh… I see you point, maybe it require some cross over ... |
| 14:36.37 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 14:37.50 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 14:37.54 | longst | kaldemar: if you happen to know where I could find TE405P pinpoint. |
| 14:38.45 | Greenlight | longst: https://www.google.co.uk/url?sa=t&rct=j&q=&esrc=s&source=web&cd=1&cad=rja&ved=0CEMQFjAA&url=http%3A%2F%2Fdocs.digium.com%2FTE410P%2Fte400series-user-manual.pdf&ei=0lMaUZmUOLGW0QWroIEg&usg=AFQjCNGkBWt7xiAnGbjuTwA0k3hxLF9fMA&bvm=bv.42261806,d.d2k |
| 14:38.47 | kaldemar | longst: the card manual has it. |
| 14:38.57 | Greenlight | (That's the card manual) |
| 14:39.40 | Greenlight | Appendix A "Pin Assignments" |
| 14:40.27 | denisnikulin | hi |
| 14:40.53 | denisnikulin | I am having this error when doing make codecs after getting ILBC source: |
| 14:40.59 | denisnikulin | make[2]: *** No rule to make target `all'. Stop. |
| 14:40.59 | denisnikulin | make[1]: *** [ilbc/libilbc.a] Error 2 |
| 14:41.11 | denisnikulin | asterisk 1.8 |
| 14:42.25 | denisnikulin | am I missing some package? |
| 14:42.44 | longst | kaldemar: Greenlight: actually, I think if pinpoint is not correct. PRI LAY 1 lamp would not light up what do you think? |
| 14:44.22 | Greenlight | I think for the 5 minutes it would take to make that cable up, it would be time well spent |
| 14:44.27 | Greenlight | It's not gonna blow up |
| 14:44.36 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:44.38 | *** join/#asterisk _Corey_ (~chatzilla@173.161.229.46) |
| 14:45.56 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 14:46.28 | longst | Greenlight: malcolmd: kaldemar: by the way, it is better to clear another important thing. Do you agree that connect NT to TE device should use straight cable, and connect TE to TE device should use cross cable: PS since I tried to play around two Asterisk both with TE110P, I connected them with cross cable. And I got it to work. |
| 14:47.20 | Greenlight | It depends on the pinouts |
| 14:47.35 | Greenlight | But, if it's not working, then try the other cable ... |
| 14:49.10 | *** join/#asterisk davlefouAMD (~david@197.15.73.137) |
| 14:50.42 | kaldemar | longst: it does not matter what you call the devices. the cable may still be wrong. |
| 14:51.40 | longst | Greenlight: kaldemar: thanks I will ask my colleague help me check GSMGateway PIN out. thanks for your advice. |
| 14:51.45 | capnkooc | Greenlight, thx I'm going to check all of that fist I will probably back with the results |
| 14:51.47 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 14:52.43 | longst | Greenlight: thanks the document you found. |
| 14:53.03 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-177-46.red.bezeqint.net) |
| 14:54.15 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 14:57.10 | longst | Greenlight: I am wondering how you google it… I have another two TE110P, I am planing to find out their PIN out as well. but according to https://www.digium.com/en/supportcenter/documentation/viewdocs/TE110P this page, there is no document describe the pin out information of TE110P |
| 14:58.35 | Greenlight | It's the same manual I beleive |
| 14:59.03 | Greenlight | The pinouts will be the same anyway |
| 14:59.16 | Greenlight | Just make the cable :) |
| 15:01.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:01.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:02.01 | longst | I already made the cable:-) Now I am planing pending working with GSMGateway, until I know the pinout of GSMGateway, try to connect TE405P with another Asterisk with TE110P, actually the idea here is I try to practice how to connect Asterisk with PRI. :-) |
| 15:05.44 | *** join/#asterisk muebau (~umueller@i53875E7E.versanet.de) |
| 15:09.04 | ghost75 | how can i prevent that everybody sees config files from sip phones on my tftp server (they include passwords) |
| 15:09.32 | WIMPy | longst the pin-out is (almost) always the same. |
| 15:09.50 | muebau | hi, I got an asterisk running on a fritzbox. SIP is fine. I can't make calls via ISDN. What could be wrong here? |
| 15:10.19 | longst | another general question about /etc/dahdi/system.conf on http://pastebin.com/hAqXGhej "span=4,4,0,ccs,hdb3,crc4" here we have span 4 configuration. according to span=<span num>,<timing source>,<line build out (LBO)>,<framing>,<coding>[,yellow], the second number should be timing source and it mentioned "The number used for each port should be different." If I understand it correctly , we could not have "span=4,1,0,ccs,hdb3,crc4" |
| 15:10.19 | longst | "span=1,1,0,ccs,hdb3,crc4" configuration together.. |
| 15:10.30 | WIMPy | muebau: The hardware. |
| 15:10.57 | muebau | WIMPy: what do you mean? |
| 15:11.55 | *** join/#asterisk MarKsaitis (~MarKsaiti@194.168.230.106) |
| 15:11.56 | WIMPy | muebau: The hardware is not supported. You can only connect the original phone application via SIP and use it as gateway. |
| 15:12.48 | jsmith | longst: Right... that second field is "which priority should this span be (on a multi-span card) for the timing source" |
| 15:13.05 | muebau | aaah thank you. I thought this would an option but never thought it is a need. |
| 15:13.14 | jsmith | longst: So if you have a 1 in that column for a particular span, the card will sync to the timing on that span first |
| 15:13.20 | Greenlight | longst: Generally, if I'm using network timing, I numnber them same as the span number for timing |
| 15:13.32 | jsmith | longst: If you put a 2 in that column, it'll use that span for network timing if the span with the 1 goes into alarm |
| 15:13.57 | jsmith | longst: If you put a 0 in that column, the span will be free-timing |
| 15:14.34 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 15:16.42 | muebau | Do you know a _cheap_ little "isdn modem" (usb) to build a "isdn to sip" gateway with e.g. OpenWRT and asterisk? |
| 15:16.50 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:16.50 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:17.20 | WIMPy | muebau: Yes, all the cheap ones with HFCS-USB. |
| 15:17.42 | muebau | thx |
| 15:17.51 | mjordan | denisnikulin: the iLBC source is already included in recent versions of Asterisk 1.8. There shouldn't be anything you need to do other than 'make'. |
| 15:17.53 | Greenlight | Cheap's cheap for a reason though |
| 15:18.01 | WIMPy | muebau: Or use a HorstBox. It's hardware is supported. Well, the ISDN ports, not the analog ones. |
| 15:18.31 | ghost75 | horsts box |
| 15:19.10 | WIMPy | Yes, cheap is cheap because the hardware is documented instead of coming with expensive closed source drivers that make it useless :-) |
| 15:19.35 | Greenlight | ymmv :) |
| 15:19.42 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 15:19.49 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 15:20.42 | longst | jsmith: Greenlight: Good explanation about "span configuration" thanks |
| 15:23.37 | jsmith | longst: You're most welcome... |
| 15:23.44 | longst | jsmith: what's meaning by "free-timing" |
| 15:24.10 | jsmith | longst: It's using its own timing, not syncing to the timing coming from the other end |
| 15:24.38 | jsmith | longst: (That's typically used if they other side is syncing to your timing, for example when you're the NET side of a PRI, and the other end is acting as the CPE and syncing to your timing) |
| 15:26.58 | longst | jsmith: basically, for example, if I would like to set span1 as a master. I also should set to "0" |
| 15:27.33 | Greenlight | What do you mean by master? |
| 15:27.48 | Greenlight | So, it behaves like an NET side ? |
| 15:28.42 | longst | yes |
| 15:28.52 | denisnikulin | mjordan, looks like it is opposite - it is removed from the source. |
| 15:29.10 | longst | if you do a "# dahdi_cfg -vv" it will say something like "Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)" |
| 15:29.23 | denisnikulin | mjordan, if I dont run that contrib script it gives me "ilbc/iLBC_encode.h: No such file or directory" errors |
| 15:29.58 | denisnikulin | make[1]: *** [codec_ilbc.o] Error 1 |
| 15:30.54 | mjordan | what version of 1.8? |
| 15:31.22 | denisnikulin | asterisk-1.8.13.1 |
| 15:31.38 | mjordan | k. I'd have to go back and look, but we put the codec back in at some point in time in 1.8. |
| 15:31.46 | mjordan | If you're having build problems, you should move to a version that has iLBC in it. |
| 15:32.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:32.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:32.49 | mjordan | and its in 1.8.13.1 as well |
| 15:33.02 | mjordan | so I'm not entirely sure what you're building or how :-) |
| 15:33.15 | jsmith | longst: Yes, if you want span 1 to be a timing master (and not a timing slave), set the second column to "0" |
| 15:33.24 | denisnikulin | to reproduce: |
| 15:33.24 | denisnikulin | apt-get source asterisk |
| 15:33.24 | denisnikulin | sudo apt-get install libncurses5-dev libxml2-dev |
| 15:33.24 | denisnikulin | cd asterisk* |
| 15:33.24 | denisnikulin | ./contrib/scripts/get_ilbc_source.sh |
| 15:33.25 | denisnikulin | ./configure |
| 15:33.27 | denisnikulin | make codecs |
| 15:33.38 | denisnikulin | mjordan, ^^ |
| 15:34.08 | mjordan | don't run the contrib script |
| 15:34.11 | denisnikulin | I also tried just "make", without make config. and sorry I forgot to mention "make menuselect" |
| 15:34.14 | mjordan | don't run 'make codecs' |
| 15:34.16 | mjordan | just run make |
| 15:34.23 | denisnikulin | I have just tried without contrib script |
| 15:35.03 | denisnikulin | <PROTECTED> |
| 15:35.03 | denisnikulin | codec_ilbc.c:41:30: error: ilbc/iLBC_encode.h: No such file or directory |
| 15:35.09 | denisnikulin | make[1]: *** [codec_ilbc.o] Error 1 |
| 15:35.53 | mjordan | denisnikulin: I'm not sure what the problem is, but I don't have that problem when I use a source tarball. So it's either the package, or something else that I'm not sure of. |
| 15:36.26 | mjordan | regardless, the iLBC source was added back in 1.8.10.0, so getting the source is no longer necessary. We kept the script simply as a reference to how we obtained the source. |
| 15:36.57 | denisnikulin | mjordan, ok thanks for your comment, I'll try again. |
| 15:37.08 | denisnikulin | I see same thing on ubuntu 12.10 and debian 6 |
| 15:37.40 | denisnikulin | mjordan, I'll check original tar.gz |
| 15:39.28 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.222) |
| 15:39.48 | longst | Greenlight: do you think Channel 01: Clear channel (Default) (Echo Canceler: mg2) "(Slaves: 01) " this "Slaves" matters. |
| 15:40.17 | Greenlight | Matters, in what way ? |
| 15:42.38 | longst | Greenlight: in my experience, if I change "/etc/chan_dahdi.conf" signalling = from "pri_cpe" to "pri_net" afterward this channel should display " channel (Default) (Echo Canceler: mg2) "(Master: 01) " correct? |
| 15:43.47 | WIMPy | longst: With dahdi, you alsways have to change both configs if you want to change "direction". |
| 15:44.36 | longst | OK I see. I try to change on the other end. |
| 15:45.19 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 15:45.49 | longst | Greenlight: WIMPy: do you have experience that saw something like Channel 01: Clear channel (Default) (Echo Canceler: mg2) "(Masters: 01) . This key word master appears |
| 15:45.56 | WIMPy | Err, and both ends as well, off course. |
| 15:46.15 | WIMPy | was referring to dahdi/system.conf and asterisk/chan_dahdi.conf. |
| 15:46.44 | WIMPy | It tells you which span currently provides timing for the card. |
| 15:47.35 | *** join/#asterisk din3sh (~din3sh@41.136.80.244) |
| 15:47.43 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 15:49.18 | *** join/#asterisk bigd1234567890 (~dburgess@63.96.150.226) |
| 15:49.32 | *** join/#asterisk jsjc (~Adium@149.Red-2-136-113.dynamicIP.rima-tde.net) |
| 15:50.02 | denisnikulin | mjordan, well, I downloaded asterisk-1.8.13.1.tar.gz and codec was compiled successfully, you are right! now I cannot load it as "Module 'codec_ilbc.so' was not compiled with the same compile-time options as this version of Asterisk" |
| 15:50.15 | denisnikulin | but it is another task, I'll fix that, thank you mjordan |
| 15:50.16 | denisnikulin | ! |
| 15:51.24 | capnkooc | Hi guys, I have some problems with my dahdi configuration I think, the problem is the follow: when I dial using any DAHDI channel the cdr registers all the seconds including the ringing... |
| 15:52.21 | capnkooc | so the billing information it's not usable for my at this point, and I really feel a little lost, because I really don't now how to research for this sort of things |
| 15:52.43 | capnkooc | can anyone give some advice? |
| 15:53.27 | WIMPy | capnkooc: Analog is evil |
| 15:53.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:53.43 | c0rnoTa | Hello all! |
| 15:53.48 | igcewieling | denisnikulin: have you considered using 1.8.20.1 ? |
| 15:54.26 | igcewieling | capnkooc: you MUST NOT use analog FXO if you want accurate CDR call lengths |
| 15:54.57 | denisnikulin | igcewieling, I wanted to use debian package. any 1.8 version |
| 15:55.00 | igcewieling | capnkooc: analog simply does not provide that information |
| 15:55.32 | WIMPy | I heard some people say you must not use Asterisk if you want sensible CDRs. |
| 15:55.32 | c0rnoTa | I'm getting deadlock on Asterisk 1.8.20.1 with chan_sip: |
| 15:55.32 | c0rnoTa | Lock #0 (chan_sip.c): MUTEX 25826 do_monitor &monlock 0x7f0669911020 (1) |
| 15:55.33 | c0rnoTa | Waiting for Lock #1 (chan_sip.c): MUTEX 8009 sip_pvt_lock_full pvt 0x86cc350 (1) |
| 15:55.33 | c0rnoTa | Waiting for Lock #0 (chan_sip.c): MUTEX 25845 restart_monitor &monlock 0x7f0669911020 (1) |
| 15:55.33 | c0rnoTa | Locked Here: chan_sip.c line 25826 (do_monitor) |
| 15:55.42 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 15:55.47 | pabelanger | c0rnoTa, pastebin |
| 15:56.06 | capnkooc | igcewieling, ok so If I use a pri it would work? like a E1? |
| 15:56.32 | WIMPy | capnkooc: Yes |
| 15:57.05 | c0rnoTa | pabelanger: sorry, forgot about it :) Here you are: http://pastebin.com/sRG0Hhkc |
| 15:57.06 | Greenlight | WIMPy: I hope that's not the case, I bill almost 3million minutes a month from my Asterisk CDR's ... |
| 15:57.13 | igcewieling | capnkooc: PRI would work, as would using a SIP provider (they don't use analog) |
| 15:57.25 | c0rnoTa | Same deadlock was in 1.8.12.0 |
| 15:57.35 | igcewieling | Greenlight: maybe you had 4 million mins of usage? |
| 15:57.46 | Greenlight | Don't ! |
| 15:57.52 | c0rnoTa | As I see it in chan_sip, it wired with MWI |
| 15:57.55 | igcewieling | We bill based on the CDRs we receive from the carriers, not from Asterisk |
| 15:58.24 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:58.27 | WIMPy | Greenlight: I personally find it most interesting that the reason for having CDRs is missing from them: The costs. |
| 15:58.30 | Greenlight | igcewieling: And link to your customers based on datetime and dest ? |
| 15:58.57 | igcewieling | Greenlight: no, we link based on ANI (we override the ANI if the customer sends us an ANI not on their account) |
| 15:59.03 | Greenlight | WIMPy: We've a routine that updates our CDR database every night with costs |
| 15:59.08 | *** join/#asterisk mahiti-irc (~mahiti@202.83.16.183) |
| 15:59.13 | mahiti-irc | hi |
| 15:59.19 | mahiti-irc | i have te220 card |
| 15:59.39 | Greenlight | igcewieling: Ahh .. we ruled that option out incase we had customers want to withhold number etc |
| 15:59.42 | WIMPy | He, we have PRI day today. |
| 15:59.42 | mahiti-irc | i want it to set to e1 |
| 15:59.54 | mahiti-irc | how do i connect the jumper to set it |
| 16:00.02 | c0rnoTa | any suggestions? |
| 16:00.08 | mahiti-irc | there are total 8 pins |
| 16:00.13 | mahiti-irc | which two shud i connect |
| 16:00.25 | igcewieling | mahiti-irc: Digium support can help you. |
| 16:00.36 | mahiti-irc | do they have irc? |
| 16:00.36 | Greenlight | Buuuut, in all seriousness, you're just joking about inaccuracies of CDR's... right ? |
| 16:00.44 | WIMPy | mahiti-irc: The manual tells you. And apart from that, you don;t have to, you can tell the driver when loading instead. |
| 16:00.44 | igcewieling | mahiti-irc: I believe you can also set it when the card driver loads |
| 16:01.35 | mahiti-irc | thanks WIMPy , igcewieling , you can use default_linemode for that. |
| 16:01.45 | mahiti-irc | but i would also like to know the jumper way |
| 16:02.01 | Greenlight | mahiti-irc: Iirr it tells you in the manual |
| 16:02.02 | capnkooc | WIMPy, igcewieling thanks guys |
| 16:02.10 | mahiti-irc | please can you tell me is there any IRC for dahdi |
| 16:02.22 | WIMPy | mahiti-irc: You can find the manyal on their website or using google. |
| 16:02.28 | igcewieling | mahiti-irc: I doubt anyone here knows the answer to your question |
| 16:02.38 | muebau | Ok I got it partialy running. I can make ougoing calls from SIP to ISDN. Incoming calls from ISDN are not ringing on SIP. What could be wrong? |
| 16:03.03 | mahiti-irc | WIMPy, igcewieling, the manual just mentions turn the jumper "on" to set it to E1 |
| 16:03.07 | *** join/#asterisk kart_ (~kartik_@122.179.56.12) |
| 16:03.12 | mahiti-irc | it doesnt mention how to turn on |
| 16:03.14 | denisnikulin | kart_, hi |
| 16:03.21 | kart_ | denisnikulin: heya |
| 16:03.23 | mahiti-irc | :( |
| 16:03.38 | WIMPy | muebau: Does Asterisk show as online in the FB Web interface? |
| 16:04.11 | WIMPy | mahiti-irc: Is it printed on the card, perhaps? |
| 16:04.26 | muebau | there is not enough RAM to get websrv AND asterisk run |
| 16:04.36 | muebau | so I have to kill websrv |
| 16:04.57 | WIMPy | muebau: Ugh. That's not going to make it easier. |
| 16:05.15 | mahiti-irc | WIMPy, on card its mentioned "open=T1 close=T1" |
| 16:05.22 | muebau | I have to take whats there |
| 16:05.38 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:05.42 | mahiti-irc | i am confused as to which of the 8 pins shud be closed |
| 16:05.59 | muebau | I use the nice script from http://spblinux.de/fbox/cfg_asterisk |
| 16:06.28 | WIMPy | mahiti-irc: It's one pair per port. So it seems like you can configure more ports than you have. |
| 16:06.37 | muebau | do I have to create an account in the fritzbox settings to connect to the asterisk? |
| 16:06.50 | *** join/#asterisk deo (~deo@112.198.90.93) |
| 16:06.52 | WIMPy | muebau: Sure. |
| 16:07.23 | *** join/#asterisk nightrid3r (~kvirc@62.205.64.89) |
| 16:07.32 | muebau | so I didn't jet understood the way it walks... |
| 16:07.38 | WIMPy | mahiti-irc: So unless you want to do strange things, open or close all 4. |
| 16:07.39 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 16:08.02 | WIMPy | muebau: You configure a SIP phone and then you use that Account with Asterisk. |
| 16:08.46 | longst | Greenlight: WIMPy: have two asterisk connected via PRI with following configurations http://pastebin.com/reuBxBnm |
| 16:08.55 | WIMPy | muebau: There is another way: You could connect via remotecapi and chan_capi. |
| 16:09.09 | muebau | and this is the way to build a gateway? |
| 16:09.11 | muebau | aah |
| 16:09.26 | muebau | so with usb? |
| 16:09.31 | WIMPy | SIP is the easier way. |
| 16:09.52 | WIMPy | No. That's both entirely software. |
| 16:10.40 | WIMPy | Off course you could leave the integrated telephony hardware alone and plug in a USB dongle and use that directly. But if you alread have issues with RAM that might not be an option. |
| 16:10.49 | muebau | if the box uses one of the asterisk account then the asterisk just serves 2 phones. One is the box and one is me somewhere in the world. |
| 16:10.59 | WIMPy | OTOH you could disable the telephony application. |
| 16:11.30 | longst | Greenlight: WIMPy: Asterisk 1 is NET Asterisk 2 is CPE, After I properly make the configuration like this. I restart both machine. But if I run a "dahdi_cfg -vv" it still present "Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01)" I am wondering if you see any problem for my configuration? thanks |
| 16:12.11 | *** part/#asterisk Elleni (3ec00582@gateway/web/freenode/ip.62.192.5.130) |
| 16:12.42 | lorsungcu | I've got a customer that has a small call center, and a few times a day, a recorded call gets the two channels out of sync |
| 16:12.43 | lorsungcu | ' |
| 16:12.57 | WIMPy | longst: Did you include dahdi-channels.conf? |
| 16:12.59 | Greenlight | lorsungcu: Use MixMonitor, not Monitor |
| 16:13.15 | lorsungcu | iirc, it is using mix monitor, but i will check. |
| 16:13.47 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 16:14.05 | muebau | my idea is to do something like this: ISDN<-->fritzbox(asterisk)<-->SIP<-->fritzbox<-->phone . Do I have to create an account on the fritzbox to connect to the asterisk? |
| 16:14.38 | WIMPy | muebau: What's Asterisks task there? |
| 16:14.59 | muebau | what do you mean? |
| 16:15.04 | longst | WIMPy: in both asterisks, "chan_dahdi.conf" have [global] #include dahdi-channels.conf |
| 16:15.04 | WIMPy | It would look like ISDN<>Fritzbox<>SIP<>Asterisk<>... |
| 16:15.38 | WIMPy | muebau: Why don;y you use the original telephony application? |
| 16:16.09 | muebau | it is an _old_ box |
| 16:16.43 | WIMPy | muebau: How old? Without SIP phone support? |
| 16:16.54 | muebau | so there is no option to do thinks like this. somebody told me to do it the asterisk way |
| 16:17.00 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.53.222) |
| 16:17.28 | WIMPy | muebau: That surely prevents you from connecting Asterisk that way as well. |
| 16:17.31 | muebau | its a fritzbox!fon (no version) |
| 16:18.23 | muebau | oh there is internet telephony in it. So you mean this is needed? |
| 16:18.43 | WIMPy | If it doesn't support SIP phones you will have to try the network CAPI thing. (Or use your own hardware.) |
| 16:19.07 | WIMPy | That's what I said initially. You connect Asterisk as a SIP phone. |
| 16:19.30 | nightrid3r | fritzbox contains an ata but i'm not sure it will allow routes from the ata to the lan |
| 16:19.57 | WIMPy | But in that case the question is why you need Asterisk at all. |
| 16:20.03 | lorsungcu | yeah mix monitor is being usied |
| 16:20.11 | lorsungcu | @ Greenlight ^ |
| 16:20.14 | muebau | but I got outgoing calls run without the SIP connection |
| 16:20.56 | mahiti-irc | WIMPy, thanks connecting all eight pins 1-2, 3-4, 5-6, 7-8 made it E1 |
| 16:20.59 | WIMPy | How? |
| 16:22.14 | mahiti-irc | using jumpers from HDDs |
| 16:23.04 | muebau | I have an asterisk running on the box and it does provide SIP. After I restarted telefond it is able to start outgoing calls. Just incoming calls are not ring on SIP. |
| 16:23.48 | muebau | is there a way to get DEBUG output on ISDN to see if there are calls recognised? |
| 16:24.02 | WIMPy | How yo you have Asterisk connected to telefond? |
| 16:24.11 | Greenlight | lorsungcu: If it's using MixMonitor, then I'm unsure how it would be getting out of sync; very odd. |
| 16:24.28 | WIMPy | Only if AVM supplied some debugging tool. |
| 16:25.47 | muebau | I didn't |
| 16:26.09 | WIMPy | Then how do you make calls? |
| 16:27.05 | muebau | I guess asterisk connects in some way to the ISDN bus |
| 16:27.37 | muebau | I just fallowed the howto on http://www.juerging.net/projekte/Fritzbox-Asterisk/ |
| 16:27.48 | muebau | its german |
| 16:27.55 | WIMPy | Would be new to me that the hardware is supported. |
| 16:28.50 | WIMPy | And it certainly wouldn't play along nicely with telefond if it was. |
| 16:30.33 | WIMPy | That page doesn't tell how it works. But from what is described there, I guess it uses the remote CAPI. |
| 16:30.41 | muebau | is there no way in asterisk to have a look on ISDN stuff |
| 16:31.00 | muebau | aaah ok |
| 16:31.14 | WIMPy | There's probably some capi debug command. |
| 16:31.24 | muebau | aah ok |
| 16:31.44 | longst | when I run a core show channel types from CLI have "DAHDI DAHDI Telephony Driver w/PRI (Devicestate) yes (Indications) yes (Transfer)no " I am wondering if it means the DAHDI is not really to transfer data |
| 16:32.11 | WIMPy | I once tried that setup with an Asterisk running on a PC. But I didn't do anything more than just find out that it can use the interfaces. |
| 16:32.59 | WIMPy | longst: No, it means you didn't configure it to transfer calls. |
| 16:33.44 | muebau | what does this mean: |
| 16:33.45 | muebau | <PROTECTED> |
| 16:33.57 | longst | WIMPy: thanks may I ask how I could configure it to be used for connecting calls? |
| 16:35.02 | WIMPy | muebau: I'm not much in to the CAPI stuff. Maybe the controller mappings aren't correct. Can you use both the internal and the external interface? |
| 16:35.11 | WIMPy | longst: You already have. |
| 16:35.36 | muebau | There is just one ISDN interface |
| 16:36.06 | *** join/#asterisk greenwolf (42570420@gateway/web/freenode/ip.66.87.4.32) |
| 16:36.16 | longst | when I do a test call CLI : static-host*CLI> originate DAHDI/123 extension 301@originate-context |
| 16:36.16 | WIMPy | muebau: ok, well then it seems to be ok. Maybe the configuration expects a second one. Sould be safe to ignore. |
| 16:36.26 | longst | I would receive this error message: [Feb 12 17:30:35] NOTICE[3294]: channel.c:5325 __ast_request_and_dial: Unable to request channel DAHDI/123 |
| 16:37.02 | WIMPy | longst: Do you have 123 cahnnels? |
| 16:37.26 | muebau | fine but I still have incoming calls working :-) |
| 16:38.11 | WIMPy | muebau: You probably have to configure your MSNs in chan_capi.conf or whatever it's called. |
| 16:38.22 | longst | WIMPy: I was thinking DAHDI/123 like SIP/123 in this case 123 is an extension, if I understand you correct I have to dial DAHDI/<Channel>/Extension |
| 16:38.26 | longst | correct? |
| 16:38.28 | WIMPy | Or maybe just "*". |
| 16:38.35 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 16:38.44 | muebau | aah ok |
| 16:38.47 | WIMPy | longst: No, wrong. In both cases. |
| 16:39.15 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 16:39.30 | WIMPy | longst: You should read a bit of documentation before experimenting. That will make it a lot easier. |
| 16:40.26 | longst | thanks for your advice, any quick start document you suggested? |
| 16:40.40 | jmetro | the book |
| 16:40.43 | WIMPy | ~book |
| 16:40.43 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:40.43 | igcewieling | longst: this is telecommunications there is no such thing as "quick start" |
| 16:41.06 | igcewieling | Just like there is no "quick start" guide to brain surgery or auto repair |
| 16:41.17 | *** join/#asterisk bandroidx (~bandroidx@2605:6400:2:fed5:4:0:414:11) |
| 16:41.20 | jmetro | You can quick start on asterisk pretty well, assuming you alread know basic linux and arent trying to do anything fancy |
| 16:44.09 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 16:44.25 | qakhan | hi all, i am getting this message on asterisk cli |
| 16:44.38 | qakhan | Remote UNIX connection |
| 16:44.39 | qakhan | Remote UNIX connection disconnected |
| 16:45.14 | WIMPy | Someone started asterisk -r. |
| 16:45.28 | denisnikulin | qakhan, it may be either some monitoring software or another process of "asterisk -r" |
| 16:45.34 | longst | igcewieling: I have been working with Asterisk for around 2 years but no DAHDI device at all. I know that there is no quick start for any thing. but may I ask any suggested material related Asterisk and DAHDI device I could read before I could come back again? thanks |
| 16:45.43 | jsmith | qakhan: That means that some other person (or script) is running "asterisk -r" or something like "asterisk -rx 'core show uptime'" |
| 16:47.04 | WIMPy | The best documentation for dahdi seems to be the sample config :-( But at least the general concepts should be explained in the book. |
| 16:47.38 | WIMPy | Like using groups. |
| 16:49.05 | qakhan | its mean that is not any error? |
| 16:49.28 | WIMPy | Does it say "error:"? |
| 16:51.35 | longst | WIMPy: thank I am about to read that part this evening and come back tomorrow. |
| 16:53.27 | *** join/#asterisk vlad_sta_ (~vlad_star@wn2nat52.beelinegprs.ru) |
| 16:53.49 | *** part/#asterisk longst (~longst@46-22-127-214.bredband.alvsjo.qonet.se) |
| 16:59.51 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 17:03.35 | *** part/#asterisk denisnikulin (~dni@62.189.81.222) |
| 17:05.27 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 17:09.23 | WIMPy | In what oderder are nested includes searced? First include first including includes? |
| 17:09.30 | WIMPy | +h |
| 17:12.50 | ghost75 | yes |
| 17:13.13 | ghost75 | thats why i have last include for global patterns |
| 17:14.13 | WIMPy | I was just thinking about nested includes coming before directly included contexts. |
| 17:17.37 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
| 17:23.04 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 17:24.16 | *** join/#asterisk MooJuice (41725fe2@gateway/web/freenode/ip.65.114.95.226) |
| 17:24.27 | MooJuice | hello |
| 17:25.51 | *** join/#asterisk bpietro (~bpietro@82.51.236.132) |
| 17:28.24 | MooJuice | Quazi-Offtopic question, but does anyone know of an easy/free way to test International telephone numbers? (IE: I want to originate a phone call from outside the US to a UIFN number to test functionality) |
| 17:29.03 | *** join/#asterisk vaewyn2 (root@jabber.andrews.edu) |
| 17:29.22 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 17:29.23 | Greenlight | You'll likrly struggle to get anywhere that provides free international calls. Although there's plenty sip carriers where it's very very cheap |
| 17:29.36 | Mon|A|rch | so, we're thinking about buying cepstral |
| 17:29.49 | Mon|A|rch | how much do they throttle the write to file feature? |
| 17:30.13 | Mon|A|rch | we'd record a series of prompts once, and then for each call we'd use it to record and then stream the person's name |
| 17:30.24 | MooJuice | Greenlight: yea im struggling.. im willing to pay too, if anyone has suggestions on a sip provider |
| 17:30.25 | Mon|A|rch | probably 150 calls in a day (fyi) |
| 17:31.46 | vaewyn2 | Hey all... just upgraded bigtime... using 11.2.1.... got 2 problems... but easy one first... Every sip call one of my phones initiates I see "Using SIP RTP CoS mark 5" then nothing on the console for 4-5 seconds... then the call progresses... how do I ditch that delay? |
| 17:31.58 | Greenlight | MonJoo: I mostly deal at the wholesale level, so can't recommend any end user ones personally. I did use voiptalk here in the UK a while back and they were okay |
| 17:32.03 | jmetro | hm..is it possible to play an announcement to the Agent when they pickup a call from the Queue? |
| 17:36.05 | jzaw | Greenlight, i use one of the myriad of betamax sip companies about 12€ for 120 days of zero cost land line calls to 99% of the world + you can use that 12€ for chargable calls |
| 17:36.14 | Greenlight | jmetro: Yes, however I can't remember off hand how to implement it, sorry! |
| 17:37.10 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 17:37.20 | Greenlight | MooJuice: Sounds like what jsaw is using would suit you? |
| 17:37.36 | jzaw | note im in no way related or work for them |
| 17:37.45 | Greenlight | Plus, UIFN is freephone isn't ? |
| 17:37.59 | jzaw | remember also low cost == low quality (at times) |
| 17:38.23 | Greenlight | jzaw: Sounds like he just want's it for testing, so that should be okay |
| 17:38.37 | WIMPy | Unfortunatly higher cost usually doesn't mean higher quality. |
| 17:38.42 | MooJuice | Greenlight: Yes I was looking at voiptalk just now.. but I am having a hard time finding where the call will be originated from |
| 17:38.43 | jzaw | is it ok to mention a few urls? |
| 17:38.46 | jzaw | in channel i mean |
| 17:38.50 | jzaw | or rather i can pm |
| 17:38.56 | MooJuice | Greenlight: yes UIFN is toll free |
| 17:39.08 | Greenlight | MooJuice: Voiptalk would be from UK |
| 17:39.41 | Greenlight | MooJuice: Can jzaw PM you a few urls to also look at ? |
| 17:39.48 | jzaw | just did |
| 17:39.54 | Greenlight | heh :) |
| 17:39.58 | jzaw | :) |
| 17:40.09 | MooJuice | Greenlight: Sure.. |
| 17:40.16 | *** join/#asterisk gavimobile (~gavimobil@109.67.216.82) |
| 17:41.10 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:41.25 | Mon|A|rch | has anyone used cepstral with asterisk at least? |
| 17:41.36 | Mon|A|rch | does the voice quality stay decent? |
| 17:41.50 | Mon|A|rch | I'm willing to pay for tts, since festival is awful |
| 17:42.00 | Mon|A|rch | and googletts isn't reliable |
| 17:42.13 | paulc | O |
| 17:42.33 | paulc | I've used cepstral a bunch.. we recorded a bazillion (ok, "a couple of thousand") city names.. |
| 17:42.45 | paulc | works well, using Allison's voice.. from memory I think it was $25 or $30ish |
| 17:42.56 | Mon|A|rch | something like that |
| 17:43.01 | Mon|A|rch | they've upped their prices since then |
| 17:43.06 | Mon|A|rch | 40/voice, 100/port |
| 17:43.13 | jzaw | didnt they alow you to create *samples* |
| 17:43.19 | Greenlight | "samples" :) |
| 17:43.23 | jzaw | i seem to remember doing that and using them in ivrs |
| 17:43.24 | paulc | MooJuice: When I make calls with voiptalk, it uses the same DID as I have with them (0844 number) |
| 17:43.27 | Mon|A|rch | you can buy a "save-to-file" license |
| 17:43.47 | Mon|A|rch | they've clamped down on letting people make 'samples' at their leisure |
| 17:43.55 | jzaw | ah i see |
| 17:44.01 | Mon|A|rch | still |
| 17:44.11 | Mon|A|rch | I'm willing to buy the save-to-file license |
| 17:44.18 | Mon|A|rch | just not sure if they're going to throttle the heck out of it |
| 17:44.47 | Mon|A|rch | I'd only need to synthesize about 150 customer names/day at most |
| 17:45.37 | Mon|A|rch | i definitely like their synthesis, it sounds clean |
| 17:46.11 | jzaw | when i last looked at tts there were some semi-decent free ones |
| 17:46.22 | jmetro | the googletts is probably best |
| 17:46.24 | jzaw | im sure theyve improved (was 3-4 years ago now) |
| 17:46.42 | Mon|A|rch | the googletts is actually blocking me now |
| 17:46.44 | Mon|A|rch | from what i can tell |
| 17:46.50 | Mon|A|rch | getting 503 service unavailable |
| 17:46.54 | Mon|A|rch | maybe it's something else |
| 17:47.20 | WIMPy | You killed Google! |
| 17:47.26 | Mon|A|rch | lol, basically |
| 17:47.32 | Mon|A|rch | pretty sure they're doing what maps did |
| 17:47.32 | jzaw | i think he started to download google |
| 17:47.36 | jzaw | and filled the tubes |
| 17:47.41 | Mon|A|rch | if you use it a ton, they kick you out |
| 17:47.56 | Mon|A|rch | are there any other decent free ones? |
| 17:48.10 | Mon|A|rch | had a hugely hard time getting espeak for asterisk installed |
| 17:48.37 | WIMPy | Hmm. I've been doing long journeys on google maps at high resolution. didn;t happen to me. |
| 17:49.21 | Mon|A|rch | by high usage, i mean like 4000 queries a minute |
| 17:49.32 | Mon|A|rch | i wasn't doing quite that volume |
| 17:49.36 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
| 17:49.42 | Greenlight | lol yea that'll raise an eyebrow |
| 17:50.15 | Mon|A|rch | lol, shocking right? |
| 17:50.17 | Mon|A|rch | anyway |
| 17:50.25 | Mon|A|rch | i don't think it's an option at this point |
| 17:50.37 | Mon|A|rch | i could start switching IPs, but I'd rather not mess with them |
| 17:50.58 | jzaw | Mon|A|rch, what about freeTTS ? |
| 17:51.09 | jzaw | java imsmr |
| 17:51.13 | jzaw | ducks |
| 17:51.26 | Mon|A|rch | never heard of it |
| 17:51.58 | Mon|A|rch | tell me more |
| 17:54.01 | jzaw | giybf ;p |
| 17:54.42 | Mon|A|rch | lol, i did |
| 17:54.50 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:56.36 | *** join/#asterisk macjunkie (~paul@xnat-52.csumb.edu) |
| 17:56.51 | macjunkie | anyone have suggestions for places to get good sounding TTS voices from? |
| 17:59.23 | ChannelZ | undocumented workers |
| 18:00.49 | *** join/#asterisk vedic (~V@183.82.82.82) |
| 18:01.54 | vedic | What options are available for hosting Asterisk on cloud like EC2 and get T1 line support? |
| 18:06.42 | *** part/#asterisk vedic (~V@183.82.82.82) |
| 18:07.02 | *** join/#asterisk vedic (~V@183.82.82.82) |
| 18:11.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:13.36 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 18:14.22 | vedic | Is there any option to host PRI line on cloud? |
| 18:15.00 | vedic | Any options which provides PRI line and then converts it to voip? I see all hosting providers give is voip but no access to PRI |
| 18:15.01 | *** part/#asterisk macjunkie (~paul@xnat-52.csumb.edu) |
| 18:15.04 | [TK]D-Fender | That isn't a "cloud"... that is a data-center |
| 18:15.14 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 18:15.29 | MooJuice | Greenlight: Voiptalk doesn't want to take my money.. :( |
| 18:19.19 | Mon|A|rch | [TK]D-Fender, iirc you were knowledgeable about the cepstral system |
| 18:19.43 | Mon|A|rch | does the save to file system allow you to save very many files? |
| 18:19.48 | jzaw | MooJuice, dont they do paypal? |
| 18:19.54 | Mon|A|rch | something like 100-200 short files/day |
| 18:20.07 | Mon|A|rch | they're not very clear on how much they throttle it |
| 18:20.11 | Mon|A|rch | just that they do |
| 18:20.40 | [TK]D-Fender | Are you sure? |
| 18:20.50 | [TK]D-Fender | I see a system ripe for the taking there... |
| 18:21.06 | [TK]D-Fender | I'd write up a queue system for it and a cache |
| 18:23.06 | vaewyn2 | Anyone got an idea... Sip -> DAHDI 5 second pause before I start seeing dialplan execution in the console... Just "Using SIP RTP CoS mark 5" .... 5 seconds... "Executing .... " |
| 18:23.46 | MooJuice | jzaw: Nope only Visa.. and seem to require bank statements to be faxed |
| 18:23.51 | vaewyn2 | Not happening on SIP -> internal voicemail |
| 18:25.47 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 18:26.22 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 18:27.28 | [TK]D-Fender | vaewyn2, that message is meaningless. Next we don't know what KIND of DAHDI using what hardware. And you should have SIP DEBUG enabled. |
| 18:30.00 | vaewyn2 | [TK]D-Fender: TE220 ... with debug on I see a bunch of handshaking... pause... 2 INVITE ... call progresses |
| 18:30.15 | [TK]D-Fender | ~pb |
| 18:30.15 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 18:30.17 | [TK]D-Fender | ^^^^ |
| 18:31.53 | vaewyn2 | [TK]D-Fender: http://pastebin.ca/2313169 looks like some other SIP traffic is mixed in... but not much |
| 18:32.39 | vaewyn2 | [TK]D-Fender: one I am interested in is 143.207.4.80... POLY6xxxxxxx |
| 18:33.49 | jmetro | how far is the phone from the server / your provider / etc |
| 18:33.52 | vaewyn2 | [TK]D-Fender: Incomings experience no lag at all... only outgoings have a pause... seeing it on polycoms... softphones... cisco... etc... |
| 18:33.53 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:34.06 | jmetro | do you have a different outbound provider |
| 18:34.06 | vaewyn2 | jmetro: the one in question... about .8ms ping :) |
| 18:34.43 | [TK]D-Fender | zaeSo huge wait just before the 100 Trying on inbound? |
| 18:34.53 | vaewyn2 | This is an internal phone system... if I call a sister institutions system with the same device I see no pause |
| 18:34.57 | [TK]D-Fender | vaewyn2, So huge wait just before the 100 Trying on inbound? |
| 18:35.13 | [TK]D-Fender | vaewyn2, where I see youht enter like 5 times? |
| 18:35.15 | vaewyn2 | [TK]D-Fender: 5 seconds ish |
| 18:35.17 | vaewyn2 | yep |
| 18:35.21 | jmetro | again - do you have a separate outbound provider than inbound |
| 18:35.36 | [TK]D-Fender | vaewyn2, that's * / core routing delay... nothing DAHDI-side. |
| 18:35.41 | vaewyn2 | jmetro: for PRI landing? same one |
| 18:35.42 | [TK]D-Fender | vaewyn2, And really "not good" |
| 18:36.34 | vaewyn2 | [TK]D-Fender: not getting it at all when I call to soft extensions on the same box... aka voicemail... number readback... etc |
| 18:39.03 | vaewyn2 | my asterisk foo is rusty cause I have been out of the game for about 4 years... but I got hauled in to upgrade this box and with the upgrade it suddenly pulls this crud so :) I'm willing to debug... but I am not up to the core hacking I use to do |
| 18:39.27 | vaewyn2 | you guys have changed too much ;) |
| 18:40.10 | Mon|A|rch | [TK]D-Fender, a queue system and cache for the cepstral thing? |
| 18:40.13 | Mon|A|rch | or for something else |
| 18:42.07 | Mon|A|rch | and yes, I'm sure they offer saving prompts to file |
| 18:48.04 | [TK]D-Fender | Mon|A|rch, Yes for Cepstral |
| 18:48.19 | [TK]D-Fender | Mon|A|rch, They do. It's $200 IIRC |
| 18:48.28 | Mon|A|rch | yup |
| 18:48.40 | Mon|A|rch | $200, +$100 for a concurrency port |
| 18:48.44 | Mon|A|rch | +$40 for a voice |
| 18:48.53 | [TK]D-Fender | vaewyn2, Any DB involved on the dialplan side? I'm running short on inspiration for why else it'd hit a delay... |
| 18:48.53 | Mon|A|rch | $340 for all my IVR needs I guess |
| 18:48.56 | Kobaz | heh |
| 18:48.57 | Kobaz | aww |
| 18:49.05 | Kobaz | so this bank, our new customer doesn't like our vpn :( |
| 18:49.28 | Kobaz | they want to set us up with *their* vpn |
| 18:49.31 | Kobaz | i hate that |
| 18:49.48 | vaewyn2 | [TK]D-Fender: not on the outgoing no... I do use DB for incoming dialplan... but it just uses the internal and that part is working fast as ever |
| 18:50.12 | [TK]D-Fender | vaewyn2, Hrm. |
| 18:50.24 | *** join/#asterisk vedic (~V@183.82.82.82) |
| 18:51.20 | vaewyn2 | [TK]D-Fender: :) Like I said... I would dig deeper also... but the code design as changed HUGELY since I was around last :) so my learning curve is antiquated and steep :P |
| 18:51.44 | vaewyn2 | tells all the young-ins to get off his lawn |
| 18:51.45 | vaewyn2 | :) |
| 18:52.34 | vaewyn2 | [TK]D-Fender: is there a way these days to turn on debugging on routing? |
| 18:53.11 | *** join/#asterisk lorsungcu_ (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 18:58.09 | *** join/#asterisk lorsungcu_ (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 18:58.36 | leifmadsen | Kobaz: +1 |
| 18:58.45 | leifmadsen | I've stopped giving into certain customer requests |
| 18:59.44 | vaewyn2 | [TK]D-Fender: WWOooo! found it... thanks for the hint on the database... I am NOT using a database for outgoing.... but I am using dundi... and when I comment that out of the dialplan it drops into (I don't need that path for most calls) voila... the delay goes away |
| 19:00.20 | vaewyn2 | [TK]D-Fender: so I just need to figure out why my dundi is slow |
| 19:00.23 | vaewyn2 | does a happy dance |
| 19:13.55 | [TK]D-Fender | vaewyn2, Glad to have pointed you in the right direction... |
| 19:14.46 | *** join/#asterisk iceyp (~icepick@watcher.vibecommunications.co.nz) |
| 19:16.53 | iceyp | hey guys... I have an issue with asterisk and everywhere I look it seems like a common problem since 2010 even though I'm running a new version of asterisk (11.2.0). When I try send a fax via asterisk (user -> asterisk -> sip provider) asterisk doesnt honor the T38MaxBitRate... I even tried to set it in the dialplan... Checking the sip debug messages i see this a=T38MaxBitRate:2400 |
| 19:17.13 | Kobaz | leifmadsen: this is a bank though |
| 19:18.03 | leifmadsen | Kobaz: then you're fucked :) |
| 19:18.46 | *** join/#asterisk retentiveboy_ (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:19.18 | Kobaz | heh |
| 19:19.21 | Kobaz | fscked in the a |
| 19:22.14 | pabelanger | Kobaz, Have a few clients that are banks too. Know your pain |
| 19:27.41 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 19:29.38 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.175.173) |
| 19:33.13 | din3sh | Asterisk 1.8 documentation says this on CDR :'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados. |
| 19:33.13 | din3sh | linkedid is based on uniqueID, but spreads to other channels as transfers, dials,etc are performed. |
| 19:33.42 | din3sh | I have added the 2 fields but no value is recorded |
| 19:36.31 | din3sh | how can i get this to work? |
| 19:38.20 | *** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com) |
| 19:41.57 | apb1963 | vaewyn2 Do kids ring your doorbell and run? |
| 19:43.01 | *** join/#asterisk Weezey (~ohno@206.210.111.64) |
| 19:43.12 | Weezey | MeetMe, Conference or Konference? |
| 19:43.29 | Weezey | (have a USB DAHDI timer) |
| 19:43.42 | kaldemar | there's also ConfBridge |
| 19:43.43 | Qwell | Weezey: app_confbridge |
| 19:43.48 | *** join/#asterisk jmls (~somefake@77.107.171.82) |
| 19:44.15 | vaewyn2 | apb1963: house is too far from the road :( I miss all the old fogey stuff :P |
| 19:44.45 | Weezey | kaldemar & Qwell: you guys used it? Like better than the otherse? |
| 19:45.01 | apb1963 | vaewyn2 my neighbors kids used to ring my doorbell and run... so I disconnected it. Problem solved :) |
| 19:46.06 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 19:46.18 | apb1963 | I was going to get a guardbear, but my landlord won't allow it. |
| 19:47.23 | mjordan | Weezey: we're partial to ConfBridge, and yes, we use it every day |
| 19:48.05 | vaewyn2 | apb1963: :) |
| 19:48.21 | leifmadsen | ConfBridge ftw |
| 19:54.34 | igcewieling | Does anyone have any ideas why SOME calls will not use direct media, assuming the call is not answered before being sent to the endpoint and assuming the codecs match and assuming there is no nat involved? |
| 19:55.10 | igcewieling | apb1963: 110V connected to the doorbell button? |
| 19:56.52 | apb1963 | igcewieling: legal issues. |
| 19:57.40 | igcewieling | ok, then 90V. That must be safe, it is what is used to ring telephones 8-) |
| 19:57.46 | AkkerKid | hey all.... |
| 19:58.02 | AkkerKid | I'm having trouble getting asterisk talking to a Microsoft SQL database |
| 19:58.38 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.175.173) |
| 19:58.49 | AkkerKid | asterisk shows it's connected when i do odbc show from cli |
| 19:59.12 | apb1963 | igcewieling: I don't know if it is or not... I don't know many people that stick their fingers in the phone. |
| 19:59.22 | AkkerKid | but when i use asterisk to make a query like "select 1" i don't get anything back |
| 19:59.33 | AkkerKid | MySQL works fine |
| 20:00.14 | apb1963 | igcewieling: Plus, it's not the voltage that matters so much as the current. If some kid has less resistance than other the smell could be awful. |
| 20:00.31 | apb1963 | s/other/another/ |
| 20:00.36 | igcewieling | apb1963: I guess you've never been working on repairing an analog line, outside, in the rain, when someone calls the line? 8-) It will wake you up better than a shot of espresso. |
| 20:00.41 | Weezey | kaldemar, Qwell, mjordan, leifmadsen: thanks all, glad I asked. |
| 20:01.04 | apb1963 | igcewieling: it never rains in california :) |
| 20:02.12 | AkkerKid | is there a way to run a SQL query from asterisk CLI? |
| 20:02.22 | igcewieling | apb1963: that is one of the things I miss about california |
| 20:03.41 | apb1963 | yeah but you pay up the wazoo for it |
| 20:03.49 | navaismo | AkkerKid, if you turn on the debug on the cli via the logger, console=> debug and try again can you see any info? |
| 20:04.56 | apb1963 | wonders about the proper spelling of wazoo |
| 20:05.44 | AkkerKid | navaismo: nothing add'l in debug |
| 20:06.16 | AkkerKid | nothing for any of the SQL queries... |
| 20:07.51 | AkkerKid | i have a Noop(${Do_This(${ARG1})}) |
| 20:10.02 | *** join/#asterisk BrokenArrow (~Lp@unaffiliated/brokenarrow) |
| 20:13.16 | AkkerKid | i'm putting relevant info into pastebin |
| 20:14.45 | AkkerKid | http://pastebin.com/UnT8uQWh |
| 20:15.20 | AkkerKid | when that line in the dialplan runs, I should see the output of the MSSQL query. But it comes back blank. |
| 20:28.55 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:32.13 | navaismo | AkkerKid, is there a way you can see the activity on the Sql server? |
| 20:42.57 | AkkerKid | navaismo: unfortunately no. |
| 20:45.12 | jmetro | why dont you show us the sql statement you are running |
| 20:45.15 | jmetro | maybe it returns 0 rows. |
| 20:53.31 | iceyp | anyone know why asterisk sends T38MaxBitRate 2400 even though i set 9600? |
| 20:53.33 | leifmadsen | AkkerKid: 1) your func_odbc.conf [FUNCTION_NAME] should be in all caps, 2) same for prefix, 3), the function name you reference in the dialplan then needs to be ${PREFIX_HEADER_NAME(ARG1,ARG2)}) |
| 20:53.47 | leifmadsen | less the extra ) on the end |
| 20:54.05 | leifmadsen | AkkerKid: all dialplan functions must always be full uppercase |
| 20:54.29 | *** join/#asterisk bpriddy (~bpriddy@ipv4.host.stabbyspazzout.net) |
| 20:55.58 | Weezey | <PROTECTED> |
| 20:56.11 | leifmadsen | >= 1.8 |
| 20:56.17 | leifmadsen | although 1.8 has a very minimal ConfBridge() |
| 20:56.24 | leifmadsen | so in most cases, yes, > 1.8 |
| 20:56.34 | Weezey | what should I run in production environment? |
| 20:56.44 | Weezey | It's 1.8 right now. |
| 20:56.48 | leifmadsen | if you want ConfBridge() to be useful, then Asterisk 10 or later |
| 20:57.06 | Weezey | 10 it is! |
| 20:57.07 | leifmadsen | if that is a general question, it depends on your environment and what you're trying to do |
| 20:57.14 | leifmadsen | note that 10 is not an LTS release |
| 20:57.23 | leifmadsen | if you're going to build out a new system, I would recommend Asterisk 11 over 10 |
| 20:57.38 | Weezey | is it stable? |
| 20:57.46 | leifmadsen | question does not parse |
| 20:58.02 | jmetro | asterisk 11 is in production for me, pretty stable. |
| 20:58.10 | Weezey | can I trust it with vm and conferencing? |
| 20:58.14 | [TK]D-Fender | * 10 = dead. Don't do it. Jump right on to 11. It's LTS |
| 20:58.25 | leifmadsen | stability is determined from the unique deployment and the functionality you employ |
| 20:58.30 | Weezey | LTS = long term sexy |
| 20:58.31 | Weezey | :) |
| 20:58.37 | jmetro | Weezy is LTS |
| 20:58.40 | [TK]D-Fender | Weezey, As much as you can trust VM in the first place |
| 20:58.59 | leifmadsen | Long Thick Sausage |
| 20:59.11 | jmetro | hickory farms? |
| 21:01.00 | ghost75 | wheezy lts |
| 21:07.01 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.160) |
| 21:07.25 | b0ot | Is there a good place I could get different rings? I'm trying to setup different rings based on diffeerent priority levels. |
| 21:07.46 | *** join/#asterisk lorsungcu (~anonymous@12.40.176.42) |
| 21:11.01 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.33) |
| 21:12.28 | _Corey_ | b0ot: get rings? You mean... ring tones? |
| 21:13.13 | igcewieling | b0ot: your question is not Asterisk related. The ringtines are configured on the SIP phone and is specific to the make/model |
| 21:15.15 | b0ot | igcewieling, correct, i didn't know a better channel to ask in, if you could recommend I would be interested |
| 21:15.25 | b0ot | _Corey_, correct |
| 21:15.39 | igcewieling | b0ot: phone make/model? |
| 21:15.55 | b0ot | igcewieling, I need wav files to play |
| 21:16.07 | igcewieling | b0ot: phone make/model? |
| 21:17.10 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 21:21.21 | _Corey_ | iTunes has ringtonez |
| 21:26.47 | jmetro | make your own ringtones, find a song you like, cut out 20 secs, convert to wav |
| 21:27.18 | leifmadsen | I did that with a recording of Super Mario Brothers, level 1-2 |
| 21:27.47 | _Corey_ | I have part of World Mix #1... ;-) |
| 21:29.26 | jmetro | My ringtone is the Sonic the Hedgehog 2 boss fight music. |
| 21:35.40 | *** join/#asterisk venturecommunist (~ventureco@unaffiliated/venturecommunist) |
| 21:36.36 | venturecommunist | so when i first install asterisk, if i don't do make samples then i won't have any extensions, priorities or contexts |
| 21:37.15 | jmetro | the make samples commands doesnt give you anything either, because it makes *.conf.sample instead of .conf |
| 21:37.48 | venturecommunist | i just wiped an install where i had done make samples and i had all these things from something_pbx |
| 21:37.56 | venturecommunist | in dialplan show |
| 21:38.42 | venturecommunist | whenever i would place a call to this DID i have it would touch base with asterisk but it would say there's no extension and context in 'public |
| 21:38.44 | venturecommunist | ' |
| 21:39.04 | venturecommunist | but then i did a dialplan show for public or whatever and it's provided by one of these make samples something_pbx |
| 21:39.32 | venturecommunist | but now in this time around i haven't yet done make samples (afraid to) and i just have nothing 0 extensions, 0 contexts, 0 priorities and when i place a call i get a busy signal and asterisk doesn't see it |
| 21:40.52 | venturecommunist | so i even ripped out the sip.conf and extensions.conf that were sorta working in the previous install and i replaced it with the step by step instructions from the quickstart guide |
| 21:41.00 | venturecommunist | only without make samples |
| 21:41.02 | venturecommunist | still nothing |
| 21:41.10 | jmetro | sip reload dialplan reload |
| 21:41.25 | venturecommunist | no such command sip reload |
| 21:41.38 | jmetro | module reload chan_sip.so |
| 21:42.04 | venturecommunist | no such module |
| 21:42.10 | jmetro | you need to configure modules.conf |
| 21:42.13 | igcewieling | venturecommunist: "make samples" only makes sample configs, it does NOT make configs which are designed to actually WORK |
| 21:42.16 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 21:42.31 | blizzow | I'm trying to extract the area code from a dialing string and set the callerID to our local phone number depending on the area code extracted. I was just going to go through a list of IF tests and set the CALLERID accordingly. Am I doing this right in my dialing plan? http://pastebin.com/aubUjUmw |
| 21:44.28 | venturecommunist | do i need an asterisk.conf? |
| 21:44.36 | saint_ | venturecommunist: yes |
| 21:45.03 | venturecommunist | i must have done something different than i thought i did last time |
| 21:46.25 | saint_ | blizzow: it looks right to me. although i m not an expert. this is not working ? |
| 21:46.35 | jmetro | asterisk.conf, modules.conf, sip.conf, extensions.conf will get you a working 2 phone system |
| 21:47.20 | saint_ | blizzow: although your _XXX. is dangeround |
| 21:47.25 | saint_ | *dangerous* |
| 21:47.32 | venturecommunist | and then just add confbridge.conf for a conference IIRC |
| 21:47.34 | blizzow | saint_: I haven't tried yet, but I want to have an expert look at it before I break something. |
| 21:48.21 | venturecommunist | i wish at least one tutorial would just show you idempotently "make your xyz.confs all look exactly like this" |
| 21:48.22 | saint_ | blizzow: if you know you have 7 digits dialing, you should use _xxxxxxxxxx |
| 21:48.59 | saint_ | blizzow: and what's the SETVAR for ? |
| 21:49.04 | saint_ | Whynot use SET ? |
| 21:49.10 | jmetro | The book kind of does that venture |
| 21:49.10 | blizzow | I didn't write the dialplan originally (also a n00b). |
| 21:49.44 | venturecommunist | there's a "the" (as in the one) book? |
| 21:50.00 | igcewieling | ~book |
| 21:50.00 | 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 |
| 21:50.21 | venturecommunist | okay SWEET |
| 21:50.25 | venturecommunist | thank you guys you are awesome |
| 21:51.09 | jmetro | THE asterisk book. If you alread know linux, you can get a dialing phone out of linux OS in probably an hour |
| 21:51.11 | saint_ | blizzow: and what happens if your caller dials 7 digits only ? |
| 21:51.20 | saint_ | blizzow: your area code is going to be screwed up |
| 21:51.30 | venturecommunist | i already did exactly what i'm trying to do 3 months ago in this same version of asterisk |
| 21:51.31 | blizzow | saint_: I used _XXX. because the rest of the context that I'd be inserting this into uses that style. We use 10 digit dialing. |
| 21:51.37 | venturecommunist | AND i got the conference bridge up |
| 21:51.43 | venturecommunist | so what's happening to me right now is just insanity |
| 21:51.52 | saint_ | blizzow: that's bad practice. that's letting open everything |
| 21:52.05 | saint_ | blizzow: like international |
| 21:52.34 | blizzow | saint_: But I think our dial plan prepends some numbers to get calls associated with queuemetrics queues. |
| 21:52.49 | saint_ | blizzow: what are you trying to do exactly. test the area code ? |
| 21:54.16 | blizzow | saint_: oof. We want to change the area code of any number called to reflect our regional office phone number. |
| 21:54.53 | saint_ | blizzow: because from what i see, if someone dials 347.xxx.xxxx , then you dial the number. if it's another number, then you force it all at 303.630.00100 |
| 21:55.03 | venturecommunist | okay this time i have a modules.conf with autoload=yes, i restarted asterisk, but module reload chan_sip.so still turns up No such module |
| 21:55.48 | venturecommunist | O'Reilly 4th Edition Asterisk on the open publishing system said autoload=yes should automatically find all modules in [someplace] and load them? |
| 21:55.50 | blizzow | Correct. So headquarters is in Denver CO. But we have an office in NYC. If someone at the Denver office calls a customer in NYC (347.xxx.xxxx), the caller ID shows up with our NYC office phone number. |
| 21:56.04 | jmetro | did you build your asterisk.conf as well |
| 21:56.14 | jmetro | you can also build the samples somewhere else and look at them / copy from them btw |
| 21:56.20 | venturecommunist | jmetro: ah ha! let me see what's there now |
| 21:56.45 | venturecommunist | blizzow: the other problem with that is... what would someone in NYC have to talk about with someone in Denver? |
| 21:56.50 | venturecommunist | jk |
| 21:56.55 | blizzow | ;) |
| 21:57.01 | igcewieling | blizzow: you have a lot of complex dialplan to write |
| 21:57.01 | blizzow | Snow? |
| 21:57.09 | venturecommunist | that's true |
| 21:58.23 | blizzow | igcewieling: if I can do it with the if test I posted (http://pastebin.com/aubUjUmw), I would think it's just copy pasta'ing a new line for each regional office number. |
| 21:59.12 | igcewieling | blizzow: don't you need dialplan for each area code in the USA so that when area code X is called, callerid Y is used? |
| 21:59.46 | venturecommunist | so i just copied the asterisk.conf.sample into asterisk.conf. the O'Reilly book didn't make clear why any of the options in asterisk.conf would be crucial to autoloading all the modules |
| 21:59.55 | venturecommunist | dutifully restarted asterisk |
| 21:59.58 | venturecommunist | still nothin |
| 22:00.01 | Mon|A|rch | will playback answer a channel if it's not already answered? |
| 22:00.12 | Mon|A|rch | it appears to be yes |
| 22:00.12 | jmetro | sip reload still say no such dommad? |
| 22:00.15 | jmetro | command* |
| 22:00.19 | venturecommunist | yeah |
| 22:00.28 | jmetro | check your modules folder for chan_sip.so |
| 22:00.50 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 22:01.13 | blizzow | igcewieling: If I understand the IF test correctly, it sets a value for true or a value for false. So the false case fails over to our default number. |
| 22:01.18 | igcewieling | Mon|A|rch: perhaps you should do a "core show application playback" before asking questions about the playback application? |
| 22:01.39 | jmetro | igcewieling google also works? |
| 22:01.45 | venturecommunist | is modules a top level directory? i don't see it |
| 22:02.01 | igcewieling | jmetro: yes, but it is NOT the best place to find documentation about the YOUR version of Asterisk |
| 22:02.17 | jmetro | Venture: I believe its in /usr/lib/asterisk |
| 22:02.22 | igcewieling | the best place to find documentation on applications and functions for YOUR version if asterisk is the CLI |
| 22:02.38 | venturecommunist | chan_sip.so is in there |
| 22:02.45 | venturecommunist | checking permissions now |
| 22:03.10 | jmetro | make sure your asterisk is set to use the right user too |
| 22:03.15 | jmetro | the one you think its using. |
| 22:03.31 | venturecommunist | it's all root |
| 22:03.33 | venturecommunist | -rwxr-xr-x |
| 22:03.33 | igcewieling | blizzow: what you are doing should work or close to work for simple needs |
| 22:03.44 | saint_ | blizzow: core show function if |
| 22:03.50 | igcewieling | though using quotes in CallerID is a classic Noob mistake. |
| 22:03.57 | *** join/#asterisk nny (~Scott@mail.hhhealth.com) |
| 22:04.01 | nny | o..m..g |
| 22:04.32 | igcewieling | so classic that asterisk generally silently removes them for you |
| 22:04.52 | nny | i need help badly. It's not even asterisk related, but I am in deep crap. I have 2 sangoma gateways that are SPAMMING dial outs and blowing up for no reason. Sangoma has turned into a shitty company for support in the last year or so |
| 22:05.19 | nny | asterisk sends the call to the sangoma gateway, and bam, 12000 calls |
| 22:05.55 | blizzow | saint_: , igcewieling: Thanks for the tips, I'll have to try this tonight. |
| 22:08.59 | jmetro | turn off the gateways? |
| 22:11.16 | nny | jmetro: ugh, i used these for an HA system thinking standalone gateways from sangoma would be similar to their hardware cards. They just resell these gian pieces of crap. Vega 50s |
| 22:12.19 | igcewieling | nny: my recent experiences with Sangoma have been less than great. |
| 22:12.29 | igcewieling | with sangoma SUPPORT that is. |
| 22:12.42 | navaismo | use digium r series |
| 22:13.07 | nny | igcewieling: I am returning these and getting something else |
| 22:13.32 | nny | igcewieling: they've gone down the toiler in the last year. Between this and wanpipe being a giant pile sometimes I give up |
| 22:14.05 | igcewieling | We were having a problems with the cards not generating interrupts or the driver not getting interrupts |
| 22:14.07 | navaismo | Digium R seriesDigium R seriesDigium R seriesDigium R seriesDigium R seriesDigium R series |
| 22:14.09 | navaismo | sory |
| 22:14.21 | navaismo | i need to clena my keyboard |
| 22:14.52 | jmetro | navaismo wow that was crazy, the spam from cleaning your keyboard clearly formed the words "Digium R Series" |
| 22:15.08 | nny | navaismo: I will look into it. I had to stop using digium 3 years ago when their modules kept dying and the T1 hardware echo canceller didn't perform at all |
| 22:15.29 | navaismo | jmetro, no the key paste was locked down |
| 22:16.27 | *** join/#asterisk ashd (~ashleyd@188-221-47-161.zone12.bethere.co.uk) |
| 22:17.51 | igcewieling | nny: I recommend various adtran gateways. |
| 22:19.03 | igcewieling | nny: various models of Total Access 90x and NetVanta |
| 22:19.19 | igcewieling | We have a couple hundred deployed |
| 22:21.51 | venturecommunist | so just now i was playing for 5 minutes with the chan_sip.so and still it couldn't find the module |
| 22:22.07 | venturecommunist | then i finally did make samples and restarted asterisk again |
| 22:22.13 | venturecommunist | now it sees the modules, it has dialplans, etc. |
| 22:22.40 | jmetro | if you follow the indepth installation guide from the book it helps you along with all that |
| 22:22.45 | venturecommunist | and now when i call instead of nothing in asterisk-CLI and a busy signal i get "the number you have dialed is out of service" but i see in asterisk that it rejects it because: |
| 22:22.45 | jmetro | but you can go erase what you dont want then |
| 22:22.53 | venturecommunist | call to extension '100' rejected because extension not found in context 'public'. |
| 22:25.09 | jmetro | right, the sample dialplans probably dont have that. go in there and erase the dialplan you dont want, make your own, etc |
| 22:25.25 | venturecommunist | this is the same error as i got last time when i did that repeatedly |
| 22:25.52 | venturecommunist | i got those dialplans from DIDWW.com |
| 22:26.13 | jmetro | paste your dialplan in a pastebin |
| 22:26.28 | *** join/#asterisk ^rage^ (~mts@213.21.4.41) |
| 22:26.41 | venturecommunist | http://www.didww.com/Knowledgebase/how_to_set_sip_trunk_configuration_for_didww_did_number_on_asterisk/ it's this except i put my number instead of the dummy one |
| 22:27.18 | venturecommunist | but that doesn't make the 'public' thing go away |
| 22:27.36 | venturecommunist | but when i look up that dialplan in the asterisk-CLI it comes from something_pbx not anything i made |
| 22:27.42 | venturecommunist | so it came with the make samples |
| 22:28.01 | venturecommunist | unless i'm reading that wrong |
| 22:28.53 | venturecommunist | [ Context 'public' created by 'pbx_config' ] |
| 22:29.20 | jmetro | i think i'm confused as to what exactly youre trying to do. Why dont you post your extensions.conf in a pastebin |
| 22:33.14 | venturecommunist | http://pastebin.com/LUdNzkmj |
| 22:33.19 | venturecommunist | latest incarnation |
| 22:33.46 | *** join/#asterisk dxrt (~dxrt@unaffiliated/dxrt) |
| 22:34.05 | jmetro | Youre trying to ring a phone, correct? |
| 22:35.33 | venturecommunist | i want to call using my mobile phone to my DID and then have asterisk answer it |
| 22:35.37 | venturecommunist | currently it's saying: |
| 22:35.49 | venturecommunist | Call blah blah to extension '100' rejected because extension not found in context 'public'. |
| 22:35.58 | jmetro | If you actually want asterisk to answer it, the syntax is Answer() |
| 22:36.20 | jmetro | then you can do something like same=>n,playback(allbusy-tt) to verify that you actually connected [you will hear the file play] |
| 22:38.33 | venturecommunist | i deleted the SIP/100 and added: exten => 123456789,n,playback(allbusy-tt) |
| 22:38.38 | venturecommunist | sip reload, dialplan reload |
| 22:38.52 | jmetro | first priority must be 1, it will fail |
| 22:39.05 | venturecommunist | didn't change 1 for Answer() |
| 22:39.09 | venturecommunist | the next one is n |
| 22:39.14 | jmetro | oh okay. |
| 22:39.22 | venturecommunist | i think a potentially important thing... |
| 22:39.30 | jmetro | so it now looks like exten => number,1,Answer() |
| 22:39.37 | jmetro | exten => number,n,playback(allbusy-tt) |
| 22:39.49 | venturecommunist | yes i deleted SIP/100 and added the exact second line you have |
| 22:40.09 | venturecommunist | it must be important that dialplan show public gives me crud |
| 22:40.13 | venturecommunist | from samples |
| 22:40.26 | venturecommunist | because it complains that who's calling isn't in 'public' |
| 22:40.49 | venturecommunist | Call blah blah to extension '100' rejected because extension not found in context 'public'. |
| 22:41.10 | jmetro | try it first though. if it falils we will check your conf to figure out the incomin context |
| 22:41.22 | venturecommunist | and dialplan show public in asterisk*CLI gives: [ Context 'public' created by 'pbx_config' ] |
| 22:41.36 | venturecommunist | try what first |
| 22:41.49 | venturecommunist | i've already tried what we just talked about |
| 22:41.53 | venturecommunist | it gives no change |
| 22:42.04 | venturecommunist | i sip reloaded, i dialplan reloaded |
| 22:42.05 | jmetro | oh, and it still says 100 no found, and you can see in the logs it executing the steps leading up to the playback? |
| 22:42.15 | venturecommunist | remind me where the logs are? |
| 22:42.25 | jmetro | asterisk cli, if you have debug and verbose high enough |
| 22:42.34 | jmetro | displays in realtime |
| 22:42.35 | venturecommunist | i do asterisk -rvvv |
| 22:42.49 | jmetro | asterisk -r then core set debug 1000 |
| 22:42.53 | jmetro | core set verbose 1000 |
| 22:43.25 | jmetro | [they will stay at this level as long as the server is online] |
| 22:43.34 | ChannelZ | Don't turn on debug, you're just going to see a lot of crap that doesn't matter and looks alarming |
| 22:43.37 | venturecommunist | <PROTECTED> |
| 22:43.47 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-177-46.red.bezeqint.net) |
| 22:43.52 | ChannelZ | Your problem is config. |
| 22:43.52 | venturecommunist | what is pbx_config |
| 22:44.05 | ChannelZ | That's the dialplan module (extensions.conf) |
| 22:44.05 | venturecommunist | why is it responsible for 'public' |
| 22:44.11 | venturecommunist | i see |
| 22:44.17 | ChannelZ | Because that's what context you've told your phone to be in. |
| 22:44.44 | ChannelZ | And you don't have a matching extension in that context. |
| 22:45.51 | venturecommunist | do i have to have something that says [public] in extensions.conf? |
| 22:46.15 | ChannelZ | Yes |
| 22:46.17 | ChannelZ | ~book |
| 22:46.17 | 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 |
| 22:47.42 | venturecommunist | the pbx thing wasn't very descriptive so i assumed that instead of something core to asterisk i thought it was from loading random make samples |
| 22:47.53 | ChannelZ | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-1.html |
| 22:47.53 | venturecommunist | because without make samples modules don't even load |
| 22:48.06 | venturecommunist | which doesn't sound like expected behavior |
| 22:48.26 | venturecommunist | so i searched the whole of google for the error mentioning that something isn't in the context 'public' |
| 22:48.30 | venturecommunist | only got 2 hits |
| 22:48.33 | venturecommunist | or duckduckgo rather |
| 22:48.37 | ChannelZ | Modules often don't load if they don't have config files. Make samples just put in a shit-ton of sample configs, a lot of which aren't actually meant to be functional but give examples on nearly every config option. |
| 22:48.55 | venturecommunist | yeah but sip.conf had a config file and i told it to autoload everything in modules.conf |
| 22:49.20 | ChannelZ | Right but sip.conf that comes with it is 100% useless to YOU except as documentation. |
| 22:49.33 | venturecommunist | i was using the useless asterisk.conf |
| 22:49.36 | venturecommunist | i was using my own sip.conf |
| 22:49.41 | venturecommunist | that i got from my DID provider |
| 22:49.52 | venturecommunist | http://pastebin.com/LUdNzkmj |
| 22:49.59 | ChannelZ | That's fine. |
| 22:50.00 | venturecommunist | oh wait this is extensions i think |
| 22:50.16 | venturecommunist | sip.conf is full of all these proxies / ips from the DID provider |
| 22:50.48 | ChannelZ | Well you will see a context= line in your sip.conf which directly relates to where in extensions.conf a call received by that peer will go. |
| 22:51.11 | venturecommunist | context=from-didww |
| 22:51.13 | venturecommunist | yup |
| 22:51.20 | ChannelZ | You've got a "from-didww" context in extensions.conf, but you've told your SIP peer to go to "public" |
| 22:51.24 | ChannelZ | or had, anyway. |
| 22:51.38 | venturecommunist | how do i tell it or untell it that? |
| 22:51.53 | ChannelZ | the context= line |
| 22:52.19 | ChannelZ | Also the extensions you posted you have Answer(SIP/100) which doesn't mean anything |
| 22:52.37 | jmetro | edit your sip.conf and find the line containing "public" and switch it to from-ddiww |
| 22:52.55 | ChannelZ | Assuming you have another peer called 100 in sip.conf which is a phone you want to ring, you'd do Dial(SIP/100) |
| 22:53.05 | jmetro | ChannelZ : i had him change that part alread =) |
| 22:53.10 | venturecommunist | that's cause my DID provider said Dial(SIP/100) but that was equally fail |
| 22:53.35 | venturecommunist | http://www.didww.com/Knowledgebase/how_to_set_sip_trunk_configuration_for_didww_did_number_on_asterisk/ |
| 22:53.44 | ChannelZ | They are suggestions; SIP/100 only means something if you actually have a peer called 100 |
| 22:53.46 | venturecommunist | i just added [general] to the top of their long sip stuff |
| 22:54.15 | venturecommunist | a peer is not an extension? |
| 22:54.19 | ChannelZ | No. |
| 22:54.34 | venturecommunist | okay because the extension they're kinda sorta making me use is the actual phone number i'm calling |
| 22:54.45 | ChannelZ | A peer is a peer. An extension is just an arbitrary series of digits that do things in asterisk via extensions.conf |
| 22:55.01 | ChannelZ | Well, yes. |
| 22:55.07 | jmetro | you can dial a word if your telephone supports it, then your extension is exten => someWord |
| 22:55.21 | ChannelZ | From an ITSP, they typically send you a call to an extension that matches your DID. |
| 22:55.26 | venturecommunist | yeah i remember dialing words 3 months ago when i was using zoiper et al |
| 22:55.31 | venturecommunist | linphone and what-not |
| 22:56.19 | jmetro | I want to setup URI dials for my company..tell people to call me at my email address.. |
| 22:56.41 | ChannelZ | So your extension (based on that pastebin you did a minute ago) that says "exten=> 123456789,1,Answer(SIP/100)", I assume you replaced 123456789 with whatever your actual phone number is. |
| 22:56.42 | venturecommunist | so i think my basic question is, what would i need to add to that link above's sip.conf stuff to make it not be absent from public |
| 22:56.50 | venturecommunist | ChannelZ: correct |
| 22:57.07 | ChannelZ | Ok. So then do you have a peer defined for your soft phone or whatever device it is you are using? |
| 22:57.24 | venturecommunist | i have this DID provider, DIDWW |
| 22:57.33 | venturecommunist | i call with my mobile phone to the DID number i got |
| 22:57.53 | venturecommunist | and i put almost verbatim the stuff from that article the last link i put up |
| 22:58.05 | venturecommunist | i began to add [general] to the top of my sip.conf |
| 22:58.06 | ChannelZ | OK but right now that call goes nowhere/does nothing. |
| 22:58.17 | ChannelZ | What is it you expect to happen when you dial that phone number from your mobile? |
| 22:58.29 | venturecommunist | i expect that asterisk-CLI is going to say that it answered |
| 22:58.42 | venturecommunist | even though the sip.conf they gave says Dial(SIP/100) so i changed that |
| 22:58.51 | ChannelZ | that's not sip.conf |
| 22:58.56 | venturecommunist | they say it is |
| 22:58.59 | venturecommunist | or wait |
| 22:59.02 | venturecommunist | you're right that part is extensions.conf |
| 22:59.17 | venturecommunist | so i changed it because i wanted Answer |
| 22:59.42 | venturecommunist | i just didn't remove the SIP/100 until jmetro alerted me to the issue |
| 22:59.51 | ChannelZ | Do you have a register => line in your sip.conf as well, or do they know your IP through a web interface or something? |
| 23:00.00 | venturecommunist | web interface |
| 23:00.12 | venturecommunist | the last DID provider i had briefly 3 months ago gave me a password and i did the registration thing |
| 23:00.14 | venturecommunist | i didn't see any of that here |
| 23:00.33 | venturecommunist | i gave it an IP address, it is certainly able to touch base with asterisk to be rejected regularly |
| 23:00.38 | ChannelZ | ok so assuming they are properly sending calls to you for your DID (are there any firewall issues here?) |
| 23:00.43 | jmetro | provide the full error so we can see what we need to setup as a peer |
| 23:01.07 | igcewieling | DIDWWW seems to me to be the sleasiest of all |
| 23:01.31 | jmetro | IMO the call is hitting his asterisk box fine, but the peer is not in his sip.conf and that is causing the issue. |
| 23:01.34 | ChannelZ | you should get *something* on the console when you call that number. |
| 23:01.57 | jmetro | Show us the whole error \o/ |
| 23:02.03 | venturecommunist | [Feb 12 23:01:35] NOTICE[8630][C-00000004]: chan_sip.c:25184 handle_request_invite: Call from '' (xx.xx.xxx.xxx) to extension '100' rejected because extension not found in context 'public'. |
| 23:02.26 | jmetro | that xx.xx.xxx.xxx should be didwww and that ip is what you need to setup as a peer |
| 23:02.28 | ChannelZ | post your sip.conf because what you're saying you have does not match what you are getting. |
| 23:03.10 | jmetro | Do you see how the didww webpage has all those entries like [46.19.209.11] |
| 23:03.11 | jmetro | host=46.19.209.11 etc... |
| 23:03.13 | ChannelZ | and that IP you X'd out is important (it's your providers IP, so who cares?) |
| 23:03.14 | venturecommunist | okay so peer in sip.conf |
| 23:03.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:03.33 | venturecommunist | cause i literally don't have anything other than what they gave me for sip.conf |
| 23:03.42 | jmetro | copy-paste one of those blocks they have in your sip.conf |
| 23:03.44 | venturecommunist | which is basically 80 lines of their various IP addresses |
| 23:03.50 | jmetro | and put the IP address that was "blocked" in it |
| 23:03.55 | venturecommunist | ohh i didn't even think to check if this particular ip is in it |
| 23:04.02 | jmetro | what does it end it |
| 23:04.03 | jmetro | in* |
| 23:04.08 | venturecommunist | http://www.didww.com/Knowledgebase/how_to_set_sip_trunk_configuration_for_didww_did_number_on_asterisk/ <-- just assume this is all i have |
| 23:04.15 | venturecommunist | in my sip.conf |
| 23:04.20 | jmetro | what ip address was from your error, x.x.x.what |
| 23:04.20 | venturecommunist | i put [general] up above it |
| 23:04.23 | ChannelZ | No, I'm done assuming what you have. |
| 23:04.25 | ChannelZ | Good luck. |
| 23:04.38 | venturecommunist | ChannelZ: k |
| 23:04.40 | venturecommunist | thanks for your help |
| 23:04.59 | venturecommunist | point is, i literally copy pasted that into my thing |
| 23:05.04 | venturecommunist | i can't make it more clear than that i don't think |
| 23:05.08 | jmetro | that one ip address is all you need <.< just copy paste one of those blocks and replace the IP's with the one from the CLI |
| 23:05.27 | nny | anyone who has used a vega or wants to try and help (i know this isn't an asterisk issue) here is the symptom for this horrible beast. http://pastebin.com/YgpuWp3m |
| 23:05.28 | venturecommunist | yeah yeah i'm going to check that in a second, i'm diving back and forth between a lot of pastebins and terminals and here |
| 23:05.40 | ChannelZ | If that's really all you had, then there's no way a call is going to extension 100 in the context 'public' |
| 23:05.42 | venturecommunist | been up 48 hours |
| 23:05.45 | nny | it just tried to call the same number 500 times on a line, instead of just gracefully giving up |
| 23:05.53 | venturecommunist | ChannelZ: no shit thanks for that insight |
| 23:06.18 | venturecommunist | ChannelZ: i keep asking what pbx is and why it's talking about 'public' |
| 23:06.22 | ChannelZ | So you're telling us your configs are one thing when they clearly are not. |
| 23:06.32 | jmetro | cd /etc/asterisk , vim sip.conf, /[last octet of the ip address], if not found, copy paste, replace with ip address, asterisk-r, sip reload, problem solved. |
| 23:06.36 | jmetro | I have a class to be to =) Goodnight |
| 23:07.08 | venturecommunist | i need to know right now |
| 23:07.17 | venturecommunist | how do i copy out of a remote terminal |
| 23:07.21 | venturecommunist | the EXACT same thing i linked you to |
| 23:07.23 | venturecommunist | in the link |
| 23:07.26 | venturecommunist | that i keep mentioning |
| 23:07.31 | jmetro | if its putty, highlight it, and it auto copies |
| 23:07.45 | jmetro | right click pastes in putty. |
| 23:07.46 | venturecommunist | it's kde so it's probably kterm or whatever |
| 23:07.47 | ChannelZ | does the text "context=public" appear *anywhere* in your sip.conf? |
| 23:08.07 | *** join/#asterisk jsjc (~Adium@149.Red-2-136-113.dynamicIP.rima-tde.net) |
| 23:08.31 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:08.44 | venturecommunist | just like in the link i gave you |
| 23:08.50 | venturecommunist | the actual word 'public' does not appear at all in sip.conf |
| 23:08.53 | venturecommunist | ctrl-f'd |
| 23:09.04 | ChannelZ | The word "public" does not appear anywhere on that page you gave me. |
| 23:09.09 | ChannelZ | http://www.didww.com/Knowledgebase/how_to_set_sip_trunk_configuration_for_didww_did_number_on_asterisk/ |
| 23:09.12 | venturecommunist | it may appear on the page i gave you |
| 23:09.17 | venturecommunist | but not in the section clearly marked for sip.conf |
| 23:09.33 | venturecommunist | which is verbatim what i have |
| 23:09.46 | venturecommunist | nothing more, nothing less, at times i have included [general] at the top and then removed it |
| 23:09.49 | venturecommunist | to see if anything changes |
| 23:10.19 | ChannelZ | Have you been reloading? |
| 23:10.26 | venturecommunist | sip reload, dialplan reload, module reload |
| 23:10.33 | venturecommunist | module reload less often |
| 23:10.51 | venturecommunist | more back when there was a problem with the modules prior to make samples |
| 23:11.35 | ChannelZ | The error from the console, what was the actual IP you xx'd out? |
| 23:11.53 | venturecommunist | right, that reminds me i have to cross reference that ip with this list |
| 23:12.00 | venturecommunist | the whole peer thing i got sidetracked on |
| 23:12.07 | venturecommunist | trying to figure out how to copy-paste in putty |
| 23:12.15 | ChannelZ | (and AFAIK you need the [general] section or the thing won't get parsed right.) |
| 23:12.31 | ChannelZ | In putty you just drag-select the text just like jmetro said |
| 23:12.45 | venturecommunist | well there's nothing in general it just says [general] at the top |
| 23:12.53 | ChannelZ | That's mostly fine. |
| 23:13.07 | ChannelZ | put context=fartfartfart under it for fun. |
| 23:13.16 | venturecommunist | yeah it's 46.19.209.14 matches up |
| 23:13.30 | venturecommunist | ok |
| 23:14.21 | venturecommunist | sip reload, dialplan reload, same error |
| 23:14.33 | venturecommunist | should i be restarting the whole asterisk? i don't remember having to from last time |
| 23:15.03 | ChannelZ | same error about "public"? |
| 23:15.09 | venturecommunist | and it is the same ip calling me again too which i guess is important |
| 23:15.22 | venturecommunist | same error |
| 23:15.23 | ChannelZ | Are you editing the right files? |
| 23:15.23 | venturecommunist | public |
| 23:15.37 | venturecommunist | i just go into configs |
| 23:15.47 | venturecommunist | i touch a new something.conf |
| 23:15.51 | ChannelZ | what does "sip show peer 46.19.209.14" tell you? |
| 23:16.09 | venturecommunist | not found! |
| 23:16.25 | venturecommunist | so it's a peer problem and it's like it's missing from the file but how isn't it the file? |
| 23:16.29 | ChannelZ | so you've got something broken such that sip.conf really isn't even loading |
| 23:16.29 | venturecommunist | it's sip.conf it's in configs |
| 23:16.39 | ChannelZ | what do you mean it's in configs |
| 23:16.49 | venturecommunist | asterisk*/configs |
| 23:16.53 | ChannelZ | /etc/asterisk/sip.conf |
| 23:17.02 | venturecommunist | mind = blown |
| 23:17.32 | ChannelZ | Sounds like you are editing the sample configs sitting in the source code directory, wherever that is. |
| 23:17.38 | venturecommunist | i just had a 3 month old vague recollection of /etc/asterisk |
| 23:18.15 | venturecommunist | so i guess mystery solved huh |
| 23:18.22 | venturecommunist | i'll play it through |
| 23:18.34 | venturecommunist | be back with an answer in not too long hopefully |
| 23:19.49 | venturecommunist | okay a weird warning on 30 lines but we'll see if that matters |
| 23:19.51 | venturecommunist | on sip reload |
| 23:20.13 | venturecommunist | i now have a peer at that ip |
| 23:20.32 | venturecommunist | same error |
| 23:20.40 | venturecommunist | the only two files i moved were sip.conf and extensions.conf |
| 23:20.43 | venturecommunist | not sure if that's important |
| 23:20.58 | ChannelZ | well there's a lot of things that are important. |
| 23:21.10 | nny | ok looking for someone to bounce this off of |
| 23:21.17 | ChannelZ | So the sip show peer, what does it say for Context? |
| 23:21.20 | nny | i'll paypal you for the help |
| 23:21.35 | venturecommunist | lol |
| 23:21.38 | venturecommunist | it says fartfartfart |
| 23:21.41 | venturecommunist | totally forgot |
| 23:22.13 | ChannelZ | ok but that's now a new problem because fartfartfart was under [general] which means the call you tried didn't match one of the actual peers below |
| 23:22.28 | venturecommunist | the ip was the same |
| 23:22.30 | nny | http://i.imgur.com/HnWBoQG.png |
| 23:22.38 | nny | is my current FXO hardware setup |
| 23:22.38 | ChannelZ | then you didn't put that under general.. but ok |
| 23:22.47 | venturecommunist | no i never put it under general |
| 23:22.55 | venturecommunist | i just changed the context to fartfartfart |
| 23:23.11 | ChannelZ | ok |
| 23:23.18 | nny | the cause 34 is due to the port not released cause. The sangoma gateway seems to just try over and over and over on that line. Did i miss configure the line or is this just bad hardware? |
| 23:23.38 | venturecommunist | [Feb 12 23:23:04] WARNING[8630]: chan_sip.c:29594 set_insecure_flags: Unknown insecure mode 'very' on line 10 |
| 23:23.46 | venturecommunist | and ~20 other times on various lines |
| 23:24.46 | ChannelZ | it should be insecure=port,invite |
| 23:24.54 | venturecommunist | basically the only files i guess i've changed at all now are sip.conf and extensions.conf |
| 23:24.55 | ChannelZ | very is old |
| 23:26.06 | venturecommunist | is putting that peer under general a thing to do? |
| 23:26.30 | ChannelZ | no |
| 23:26.39 | ChannelZ | peers are peers |
| 23:27.11 | ChannelZ | they only 'go under' general in the sense that they appear later in the file |
| 23:27.19 | ChannelZ | every [xxxx] is a new thing of its own. |
| 23:27.20 | venturecommunist | so i only looked at this with my own eyes just now |
| 23:27.29 | venturecommunist | but now it says that it's not 'public' that's missing it |
| 23:27.42 | venturecommunist | but from-didww |
| 23:27.45 | ChannelZ | right.. hopefully it says finw-dd or whatever the hell that was |
| 23:28.04 | ChannelZ | Is the extension it can't find still saying 100? |
| 23:28.06 | venturecommunist | so does that mean something? |
| 23:28.11 | venturecommunist | 100 yup |
| 23:28.28 | ChannelZ | Thats the part that makes no sense, that must be something you've told it to do through their website, to send your calls to extension 100 |
| 23:28.46 | venturecommunist | that's correct, this 100 thing was their instructions |
| 23:28.58 | ChannelZ | no, I'm not talking about the Dial(SIP/100) they said. |
| 23:29.05 | venturecommunist | you mean a web interface |
| 23:29.16 | venturecommunist | it says SIP/100 somewhere there too lemme show you |
| 23:29.31 | ChannelZ | Generally ITSPs send calls to an extension that matches your DID |
| 23:30.31 | venturecommunist | SIP/100@xx.xx.xxx.xxx |
| 23:30.38 | ChannelZ | so if they are doing it to 100, it's probably because you asked them to |
| 23:30.47 | venturecommunist | i did but they showed me by example someplace |
| 23:30.54 | venturecommunist | so i can change that, to whatever you think |
| 23:31.21 | venturecommunist | oh so you think i should put the phone number in there? |
| 23:31.26 | ChannelZ | it doesn't matter to me, you can make it work as 100 but it's just a little nonsensical. |
| 23:32.55 | ChannelZ | In any event, you just need to have an extension in extensions.conf, in the [from-didww] context, that does something. |
| 23:33.18 | venturecommunist | there we go |
| 23:33.21 | venturecommunist | worked |
| 23:33.24 | ChannelZ | s/have an/have a matching/ |
| 23:33.30 | venturecommunist | nods |
| 23:33.43 | venturecommunist | victory, that change did it |
| 23:34.12 | venturecommunist | thanks for your help |
| 23:34.17 | venturecommunist | this will turn into a chef script |
| 23:34.33 | ChannelZ | ? |
| 23:34.45 | venturecommunist | i already had a chef script for the install up to where i'm at |
| 23:34.51 | venturecommunist | so i'm going to be doing a write up on this |
| 23:35.07 | venturecommunist | like "here's how to get a confbridge up in 4 easy steps for real" |
| 23:35.12 | ChannelZ | oh |
| 23:35.17 | venturecommunist | with rackspace, chef |
| 23:35.45 | venturecommunist | for make glorious venture communism |
| 23:35.49 | venturecommunist | your help is much appreciated |
| 23:35.59 | ChannelZ | sure good luck |
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