| 00:00.40 | p3nguin | iPhone 5 pres? |
| 00:00.50 | dym | 4s |
| 00:01.02 | p3nguin | oh |
| 00:01.11 | p3nguin | Was it bad? |
| 00:01.26 | tm1000 | it wasnt what everyone expected |
| 00:01.28 | tm1000 | aka new phone |
| 00:01.29 | tm1000 | i mean |
| 00:01.32 | tm1000 | it has new hardware |
| 00:01.34 | tm1000 | but still |
| 00:01.34 | dym | it was shit in bags |
| 00:01.37 | dym | to be frank |
| 00:01.49 | dym | hence the fall |
| 00:01.57 | WIMPy | That hasn't stopped anyone so far. |
| 00:02.03 | dym | has stopped me |
| 00:02.08 | p3nguin | seri: Maybe you could give us something useful pretty soon. |
| 00:02.31 | SeRi | p3nguin, what do you need? |
| 00:02.33 | p3nguin | as in "sip set debug on" |
| 00:02.39 | p3nguin | Make a call which fails. |
| 00:02.44 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 00:02.46 | SeRi | Thats all I am getting from the error logs :( |
| 00:02.49 | SeRi | p3nguin, ok |
| 00:02.50 | p3nguin | Then "sip set debug off" |
| 00:02.54 | SeRi | on it |
| 00:03.02 | p3nguin | Then copy EVERYTHING and paste it all in the pastebin. |
| 00:03.20 | p3nguin | Set it to expire, because it will undoubtedly contain your phone number. |
| 00:03.39 | p3nguin | I usually go private with expiration for things like that. |
| 00:06.26 | SeRi | p3nguin, http://pastebin.com/NvLU2wmQ its set to priv for 1hr |
| 00:06.41 | navaismo | see you SeRi you are in p3nguin hands |
| 00:06.46 | navaismo | se you all |
| 00:07.29 | SeRi | Thanks for the help! |
| 00:08.45 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 00:09.53 | p3nguin | You're either using the wrong password or the wrong user id. |
| 00:09.56 | p3nguin | (or both) |
| 00:10.48 | p3nguin | And you should put nat=no in your ITSP peer entries, because they are not behind NATs. |
| 00:10.59 | *** join/#asterisk Janos (~Janos@186.32.185.5) |
| 00:11.36 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 00:12.03 | p3nguin | And with both your callcentric peer and your phone peer sharing a context, that tells me that you have both incoming and outgoing extensions in the same context, which is a no-no. |
| 00:12.23 | SeRi | p3nguin, thats correct |
| 00:12.28 | SeRi | so thats bad. |
| 00:12.33 | SeRi | so should I split them? |
| 00:12.42 | p3nguin | And what does family have to do with having to use callcentric? |
| 00:12.57 | SeRi | they can call me for free |
| 00:13.14 | SeRi | ip to ip |
| 00:13.16 | p3nguin | CallCentric sucks, and there's nothing my family could do to get me to rely on it. |
| 00:13.30 | p3nguin | If you can call IP to IP, you don't need CallCentric. |
| 00:13.42 | SeRi | I agree. but what can I say... |
| 00:13.52 | SeRi | Is that right? |
| 00:13.59 | SeRi | can I use voip.ms? |
| 00:14.03 | p3nguin | If you want. |
| 00:14.11 | SeRi | how? |
| 00:14.53 | p3nguin | What kind of setup does your family have for use with CallCentric? |
| 00:15.39 | SeRi | they have an asterix. though they are not setup yet. I was supose to do it as soon as I was able to get mine to work. |
| 00:15.46 | SeRi | asterisk* |
| 00:16.06 | SeRi | my brother owns a business and need it a small trunk |
| 00:16.17 | SeRi | so I was going to help him and thats when he mention CC |
| 00:16.28 | p3nguin | If they have asterisk and you have asterisk, you don't need callcentric or any other ITSP. |
| 00:17.28 | p3nguin | I would have told him no as soon as he said "call cen..." |
| 00:17.50 | SeRi | man thats cool. the trick is that I dont know how to do that :/ lol... |
| 00:18.05 | p3nguin | Maybe you don't, but I do. |
| 00:19.24 | SeRi | can you help me? |
| 00:19.40 | p3nguin | You create a peer for his asterisk. He creates a peer for your asterisk. You put in a register statement for his asterisk. He puts in a register statement for your asterisk. |
| 00:20.17 | p3nguin | Remember, asterisk is not a server. Asterisk is a back-to-back user agent. It's really just another device, like a phone. |
| 00:20.50 | SeRi | can you give me some examples in a pb please? |
| 00:20.57 | p3nguin | If you both have dynamic DNS host names, it's an easy task. |
| 00:21.05 | p3nguin | Yes, just a minute. |
| 00:21.31 | *** join/#asterisk jhmichelle352 (~jhmichell@41.130.172.112) |
| 00:22.04 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 00:22.12 | SeRi | we both do :) |
| 00:24.14 | luckman212 | p3nguin: sorry I'm coming into this conversation late -- but why do you hate callcentric? just curious because I use them at home too :) |
| 00:24.55 | p3nguin | They have a really jacked up design that is impossible to use without sacrificing normal operations and best practices. |
| 00:25.20 | SeRi | p3nguin, ^^ |
| 00:25.30 | SeRi | I am a live example |
| 00:25.54 | luckman212 | i know they have problems with DID routing ... they do some crazy stuff with that (hate that myself) |
| 00:26.04 | luckman212 | but their call quality is generally pretty good I have found |
| 00:27.38 | *** join/#asterisk brunner (~chris@pdpc/supporter/gold/brunner) |
| 00:28.18 | p3nguin | seri: Is the family asterisk behind NAT? |
| 00:28.31 | SeRi | p3nguin, Yes sr. |
| 00:33.20 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 00:34.03 | eduzimrs | im in trouble to configure digium TE205P in * 1.8 with E1 trunk... nothing happens |
| 00:34.44 | p3nguin | seri: http://pastebin.com/QGNj4n1T |
| 00:35.35 | p3nguin | And that takes care of peering the two systems together. |
| 00:35.47 | p3nguin | The only thing left is dial plan on each side. |
| 00:36.09 | SeRi | ah! sweet |
| 00:36.14 | SeRi | I can give this a try |
| 00:36.17 | SeRi | brb! |
| 00:36.19 | SeRi | Thanks |
| 00:36.38 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 00:36.43 | p3nguin | As for dial plan... |
| 00:36.56 | p3nguin | Do you have a lot of phones on each of the systems? |
| 00:38.49 | p3nguin | This is similar to what I do with my accounts (clients). If my client's account number is 187, then I would dial extension 187 403 on my phone to reach 403 on their system. |
| 00:39.26 | p3nguin | I use the client's accout number as a dialing prefix to reach their phones. |
| 00:39.51 | p3nguin | You could use a dialing prefix, or if you don't have any overlapping extensions, just dial directly. |
| 00:39.55 | p3nguin | Your call. |
| 00:42.22 | jhmichelle352 | hello |
| 00:42.51 | SeRi | my extensions are very simple I Rather do what you are saying like If I dial 10005 it will dial my brother. Is that possible? |
| 00:43.24 | p3nguin | You want that to be a special extension to reach him, or that's a prefix where you dial that + his extension? |
| 00:43.53 | jhmichelle352 | is there away that i can add some prefixs to exisiting dailplan automaticlly my country is going to change all the prefixs tommorow and that is a lot of work to be done on many servers |
| 00:44.30 | p3nguin | For a small system, we can set up extension 10005 on your system to call any given extension on his system which Dial()s his phone. For larger systems where there are lots of phones, extensions, contexts... that might not be the best route to take from the start. |
| 00:44.30 | SeRi | just a special extention |
| 00:44.47 | p3nguin | Sure, easy stuff. |
| 00:44.51 | SeRi | no this is a small system in both areas mine and his |
| 00:45.37 | p3nguin | Your phone has a context of phones? In phones, do you include other contexts? |
| 00:45.53 | *** join/#asterisk garnarsson (~garnarsso@96.49.96.81) |
| 00:45.58 | garnarsson | hi there |
| 00:46.17 | *** join/#asterisk wesphillips (~wphill04@99.161.156.160) |
| 00:46.21 | garnarsson | any chance of getting help with connecting a cisco phone to my asterisk box? |
| 00:46.26 | p3nguin | For just one extension, I'd add it to my internal context. For lots, I'd create a new context called something like family-asterisk. |
| 00:46.33 | p3nguin | ~ask |
| 00:46.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:46.35 | p3nguin | garnarsson: ^^ |
| 00:46.42 | garnarsson | been fighting it all day and have gone from bad to wors |
| 00:46.59 | garnarsson | e |
| 00:47.08 | garnarsson | I know... just being polite... |
| 00:47.16 | p3nguin | Don't ask to ask. |
| 00:47.41 | garnarsson | I have a cisco 7970 and a freepbx |
| 00:47.49 | p3nguin | ~freepbx |
| 00:47.49 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 00:47.57 | jhmichelle352 | p3nguin: is it possible? |
| 00:48.26 | garnarsson | hence the question "can I get help with this here" |
| 00:48.28 | p3nguin | jhmichelle352: You can do anything you want if you have control over your system. |
| 00:48.43 | p3nguin | garnarsson: If you have an asterisk question, you're in the right place. |
| 00:49.00 | p3nguin | <garnarsson> any chance of getting help with connecting a cisco phone to my asterisk box? <-- you said asterisk |
| 00:49.11 | p3nguin | If you meant FreePBX, this is the wrong place. |
| 00:49.39 | garnarsson | freepbx is only a gui on top of asterisk!? |
| 00:49.57 | p3nguin | is not new here |
| 00:50.08 | jhmichelle352 | i mean automatically within asterisk so i dont have to write a script for it there are hunderds of extension for example extension 015 will be 0115 so can i do this? |
| 00:50.08 | p3nguin | I've seen your kind before. |
| 00:50.10 | garnarsson | and the question is about getting the phone to register with the server |
| 00:50.29 | p3nguin | If it pertains to set up of a peer, that's a FreePBX question. |
| 00:50.39 | SeRi | p3nguin, I have one extension for my office and one for my family |
| 00:50.40 | garnarsson | ok, head over to freepbx channel then? |
| 00:51.01 | p3nguin | If it involves pointing and clicking, that's the place to go. |
| 00:51.20 | garnarsson | nope, has to do with config files and firmware |
| 00:51.27 | p3nguin | seri: You have one extension for an entire office phone system? You sure about that? |
| 00:51.45 | p3nguin | Just ask your question already, please. |
| 00:52.09 | WIMPy | jhmichelle352: sed? |
| 00:52.16 | garnarsson | so I have the cisco 7970 and the freepbx/asterisk server |
| 00:52.21 | p3nguin | vim |
| 00:52.28 | garnarsson | I 'm trying to get the phone to register with the server |
| 00:52.39 | p3nguin | :%s/015/0115/gc |
| 00:52.53 | dym | garnarsson: feels like a freepbx question |
| 00:52.53 | p3nguin | Press y for the ones you want to change, n for those you do not want to change. |
| 00:53.03 | garnarsson | it started with it not bein able to authenticate |
| 00:53.13 | garnarsson | got a 401 error |
| 00:53.17 | dym | garnarsson: you will have to add the phone via freepbx |
| 00:53.24 | dym | if thats what you're using |
| 00:53.25 | SeRi | p3nguin, its just one phone for one office in my home. |
| 00:53.36 | p3nguin | Does "sip show peers" show the phone? |
| 00:53.40 | p3nguin | garnarsson: ^ |
| 00:54.00 | p3nguin | seri: Okay, so what context does your phone have assigned? |
| 00:54.06 | garnarsson | <authPassword> and <authUser> were correct |
| 00:54.09 | SeRi | one sec pb coming up |
| 00:54.12 | jhmichelle352 | thanks WIMPy & p3nguin but that mean i will have to do that on all servers ? |
| 00:54.23 | p3nguin | If you say something about call centric, I'll tell you you're doing it wrong. |
| 00:54.52 | jhmichelle352 | they are like 15 boxs |
| 00:55.08 | p3nguin | So you're run vim and that command 15 times. |
| 00:55.10 | WIMPy | jhmichelle352: Unless you share the same configuration, you will obviousely have to touch them all, yes. |
| 00:55.13 | p3nguin | Better get started. |
| 00:55.34 | jhmichelle352 | the sad truth |
| 00:56.29 | WIMPy | If you have some sort of lookup running (e.g. dundi/enum) you might abuse that. |
| 00:57.09 | p3nguin | seri: I typically recommend a context such as "phones" for all phones. Then the "phones" context includes any other context that the phones need access to -- usually "internal" and "outbound". |
| 00:57.52 | p3nguin | Either one of those could be a good place for your new special extension to reach your family member. |
| 00:58.01 | jhmichelle352 | it seems that it's going to be a long night however i will think about adding dundi in the future thanks for help guys |
| 00:58.08 | SeRi | p3nguin, http://pastebin.com/JVE2TQV4 |
| 00:58.26 | p3nguin | # |
| 00:58.31 | p3nguin | in sip.conf here is my ext: <--- no |
| 00:58.39 | p3nguin | extensions go in extensions.conf, not sip.conf |
| 00:58.57 | p3nguin | [trunk-provider] |
| 00:58.57 | p3nguin | include => voipms-inbound |
| 00:58.58 | p3nguin | include => voipms-outbound |
| 00:59.08 | p3nguin | Do not ever do this ^^ |
| 01:00.08 | p3nguin | exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@voipms) <--- Dial(SIP/voipms/${EXTEN} |
| 01:00.11 | p3nguin | ) |
| 01:00.26 | SeRi | p3nguin, I mean my pear for my ext. |
| 01:00.40 | p3nguin | You mean the peer for your phone. |
| 01:00.48 | SeRi | lol yea ops |
| 01:01.33 | p3nguin | http://pastebin.com/Piqv4Egj see lines 118-130 |
| 01:02.03 | *** part/#asterisk jhmichelle352 (~jhmichell@41.130.172.112) |
| 01:02.09 | p3nguin | reduction of duplicate code. |
| 01:02.42 | SeRi | ah I see |
| 01:04.33 | *** join/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
| 01:04.45 | p3nguin | So anyway, pick one of your contexts that your phone has access to, and put in your new extension. |
| 01:05.29 | p3nguin | exten => 10005,1,Dial(SIP/family-asterisk/jack) |
| 01:05.45 | SeRi | cool!!!!! |
| 01:05.52 | p3nguin | "jack" will match whatever extension is used on the other side to dial someone's phone. |
| 01:06.12 | SeRi | "1000" in his case |
| 01:06.23 | SeRi | :) cool I am going to give this a try now |
| 01:06.43 | p3nguin | Sorry, family_asterisk, not family-asterisk. |
| 01:06.51 | SeRi | :) |
| 01:06.54 | *** part/#asterisk wesphillips (~wphill04@adsl-99-161-156-160.dsl.hstntx.sbcglobal.net) |
| 01:06.57 | p3nguin | What does his extension 1000 do? |
| 01:07.12 | p3nguin | Dial his phone, I guess. |
| 01:07.46 | SeRi | yes thats his office ext that recives all the incoming calls |
| 01:08.00 | p3nguin | exten => 1000,1,Dial(SIP/654321) |
| 01:08.27 | p3nguin | Extension 1000 dials SIP phone named 654321. |
| 01:09.30 | p3nguin | You can do the same type of thing on the other side. Add in an extension over there to send a call to your system, where an extension does something useful such as dial your phone. |
| 01:10.09 | p3nguin | Another way would have been to use SIP URIs. |
| 01:10.19 | p3nguin | The setup would be similar. |
| 01:11.45 | SeRi | cool |
| 01:17.10 | p3nguin | Think you can do it? |
| 01:17.33 | SeRi | I am trying now :) |
| 01:17.41 | SeRi | will report in a few min |
| 01:21.19 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 01:41.19 | *** join/#asterisk coppice (~chatzilla@m121-203-229-167.smartone-vodafone.com) |
| 01:46.00 | SeRi | almost done |
| 01:48.07 | *** join/#asterisk irroot (~irroot@197.108.200.71) |
| 01:50.56 | Kobaz | http://www.apple.com/ |
| 01:51.18 | p3nguin | OH MY GOD NO!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 01:52.09 | SeRi | LOL |
| 01:52.20 | Kobaz | why is that funny? |
| 01:52.23 | WIMPy | No more products |
| 01:52.37 | p3nguin | I think it was my reaction that was funny. |
| 01:52.48 | Kobaz | apple will still come up with new stuff |
| 01:52.53 | Kobaz | but it probably won't be as whizbang |
| 01:58.27 | *** join/#asterisk irroot (~irroot@197.107.26.37) |
| 02:12.13 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 02:19.04 | SeRi | p3nguin, I guess I failed. |
| 02:19.10 | SeRi | the registration fails |
| 02:19.13 | p3nguin | Great job! |
| 02:19.21 | SeRi | and the host is unrechable |
| 02:19.22 | SeRi | lol |
| 02:19.47 | p3nguin | Did you create a peer on his system for your system? |
| 02:19.56 | p3nguin | We'll do it one step at a time. |
| 02:20.21 | SeRi | ok one sec |
| 02:20.56 | dijib | just user/register in either iax or sip.cong |
| 02:21.00 | dijib | conf? |
| 02:21.23 | p3nguin | I gave you DROP IN examples. |
| 02:21.28 | p3nguin | All you have to do is copy and paste. |
| 02:21.46 | SeRi | yes sr and its what I did. the issue I see as that they are unable to talk to each other. |
| 02:21.55 | dijib | some of us p3nguin are copypasta inclined you know |
| 02:22.08 | dijib | behind a NAT? |
| 02:22.14 | p3nguin | The ports are open on the firewall, forwarded as needed? |
| 02:22.41 | *** join/#asterisk loconut (~loconut@173-16-61-8.client.mchsi.com) |
| 02:23.00 | dijib | SeRi, are you chekcing the ports? |
| 02:23.07 | loconut | is there a way to make QueueStatus or CoreShowChannels give the Max/Min Penalty for a channel? |
| 02:23.14 | loconut | in AMI |
| 02:23.17 | SeRi | p3nguin, yes and dijib yes |
| 02:23.23 | SeRi | one sec loading it all over again |
| 02:23.59 | dijib | 10000-20000 udp & 5060 udp SeRi to your box |
| 02:25.35 | loconut | would code need to be modified to make asterisk include Max/Min Penalty in QueueMember entries? |
| 02:25.49 | loconut | er QueueEntry entries |
| 02:28.01 | SeRi | yes |
| 02:29.27 | dijib | so forwarding is good? |
| 02:32.41 | SeRi | yes |
| 02:33.30 | SeRi | no mater what i do the port 5060 is reported unreachable by asterisk |
| 02:33.39 | SeRi | ask well as the host when registartion is going threw |
| 02:33.42 | p3nguin | How did you determine that? |
| 02:34.10 | SeRi | -- Registration for 'seri_asterisk@IP' timed out, trying again |
| 02:34.36 | p3nguin | Maybe the IP address is wrong. Use the dynamic host name. |
| 02:34.50 | SeRi | I did. its the same issue |
| 02:34.57 | SeRi | I think my ISP blocks 5060 |
| 02:35.00 | p3nguin | (2119.46) <p3nguin> Did you create a peer on his system for your system? |
| 02:35.12 | dijib | will someone have mercy on me and tell me why this line doesnt work. |
| 02:35.14 | p3nguin | Still unanswered. |
| 02:35.14 | SeRi | I have never able to use 5060 for any carrier |
| 02:35.14 | dijib | same => n,System(/usr/bin/text2wave -otype ulaw -o /var/lib/asterisk/sounds/forecast.ulaw < /tmp/forecast.txt); |
| 02:35.20 | SeRi | p3nguin, Yes I did |
| 02:35.30 | p3nguin | You can't use 5060 ever? |
| 02:35.51 | dijib | who is your ISP? |
| 02:35.58 | dijib | i doubt its blocked. |
| 02:36.05 | SeRi | so far I have not been able to use 5060 not even if God told them to do so. |
| 02:36.05 | dijib | but maybe |
| 02:36.07 | SeRi | comcast |
| 02:36.20 | SeRi | I all go threw 5080 |
| 02:36.26 | p3nguin | Business or residential? |
| 02:36.31 | SeRi | and voip.ms custom port |
| 02:36.39 | SeRi | p3nguin, residential |
| 02:37.05 | SeRi | can I use alternate port to do this? |
| 02:37.10 | SeRi | like 5075? |
| 02:37.20 | p3nguin | Maybe, but I'm interested in finding out about 5060. |
| 02:37.35 | SeRi | I have not been able to find an answer my self. |
| 02:37.50 | SeRi | I use to have comcast phone |
| 02:37.55 | p3nguin | Do you have nc/netcat handy? |
| 02:38.01 | SeRi | yes |
| 02:38.09 | p3nguin | Does it support udp mode? |
| 02:38.21 | SeRi | Mhhh never tried. |
| 02:38.43 | p3nguin | I'm not sure if all of them do or not, so that's why I ask. |
| 02:38.45 | dijib | SeRi, whats the router your trying to do this through? |
| 02:39.19 | SeRi | dijib, My side works. with other carriers is just from peer to peer that I am having issues |
| 02:39.33 | SeRi | p3nguin, tell me what you want me to do with netcat ill try it |
| 02:39.43 | dijib | so someone on the internet logged into asterisk? |
| 02:41.35 | dijib | has this setup ever worked? or is this a new build? |
| 02:47.40 | p3nguin | See what happens if you set port=5080 in the peer entry on both sides and :5080 on the DNS name in both register statements. Also forward port 5080 to both systems. |
| 02:47.55 | SeRi | doing it now |
| 02:48.59 | p3nguin | I don't have to use alternate ports, so I don't know if that's all that is needed or not. |
| 02:52.27 | SeRi | yea I think I am fuck lol |
| 02:54.48 | p3nguin | Did you forward port 5080 to port 5060? |
| 02:55.32 | SeRi | Thats port mapping. and piece of shit pfsense does not do that :) |
| 02:55.44 | SeRi | I can port forward |
| 02:55.57 | p3nguin | That's a basic NAT feature. |
| 02:56.06 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:56.13 | SeRi | actually I am wrong. I sure can |
| 02:56.24 | SeRi | lol I was thinking about application triggering |
| 02:57.13 | SeRi | one sec |
| 02:59.04 | p3nguin | I know you set port in a peer when you change the sip port on a phone, but I'm not sure if it'll work the way you need it to work for this case. |
| 02:59.35 | dijib | can i use cat in a system() app? |
| 02:59.47 | p3nguin | Sure, but don't quote it. |
| 03:07.31 | dijib | still doesnt seem to work in that command im trying to get going |
| 03:07.38 | dijib | am i just retarted or what? |
| 03:19.27 | *** join/#asterisk radic (~radic@dslb-178-002-234-160.pools.arcor-ip.net) |
| 03:29.27 | *** join/#asterisk vinhdizzo (~vinh@173.60.112.55) |
| 03:50.40 | p3nguin | Do you want me to answer that? |
| 03:51.42 | p3nguin | System(cat /path/to/file1 /path/to/file2 > /path/to/newfile) |
| 04:21.33 | kaldemar | dijib: if you're trying to get command output to CLI or a variable, use func SHELL. |
| 04:27.46 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 04:33.09 | dijib | kaldemar, im trying to output text from a file into text2wave |
| 04:33.57 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 04:40.35 | SeRi | p3nguin, I got my brother up and running with a voip.ms account |
| 04:47.17 | p3nguin | That'll work, too. calling between accounts is free. |
| 04:47.41 | dijib | im getting my whole family one them |
| 04:48.01 | dijib | even with voip.ms services and an ATA is good enough for simple home use |
| 05:11.34 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 05:12.37 | SeRi | p3nguin, I didnt know that |
| 05:12.58 | p3nguin | Now you do! |
| 05:13.13 | SeRi | I just dial hes number? or hes uid? |
| 05:13.49 | p3nguin | You can dial his phone number that he has with voipms, as long as you are dialing it via your voipms account. |
| 05:13.58 | p3nguin | It won't cost you nor him. |
| 05:14.04 | SeRi | sweet! |
| 05:15.35 | SeRi | brb |
| 05:21.48 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
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| 06:01.22 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:01.25 | schmidts | good morning |
| 06:05.23 | kaldemar | dijib: if you're just reading a file, see the FILE function. |
| 06:05.55 | dijib | can i read into a variable? |
| 06:06.45 | dijib | yes i can |
| 06:11.21 | dijib | this is what im trying to accomplish but this fails |
| 06:11.22 | dijib | exten => *61,1,NoOp(); |
| 06:11.22 | dijib | same => n,Read(forcast,/tmp/forecast,256); |
| 06:11.22 | dijib | same => n,System(echo ${forecast} > text2wave -F 8000 -o /var/lib/asterisk/sounds/en/forecast.ulaw -otype ulaw); |
| 06:11.23 | dijib | same => n,Playback(forecast); |
| 06:11.26 | dijib | crap flood |
| 06:12.28 | kaldemar | dijib: i said function READ. now you're using application Read. |
| 06:13.49 | kaldemar | Set(forecast=${READ(/tmp/forecast)}) |
| 06:15.43 | kaldemar | your example looks like it has a dozen other issues too |
| 06:15.51 | dijib | that still doesnt get it working but i think its on the roa to accomplishment |
| 06:16.10 | dijib | im thinking my ech ${variable} is the problem |
| 06:16.17 | dijib | echo |
| 06:16.28 | kaldemar | usage of text2wave is a problem. |
| 06:16.34 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 06:16.47 | dijib | if i do |
| 06:16.59 | kaldemar | replace > with a pipe |
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| 06:17.04 | dijib | echo this is a test | text2wave it works |
| 06:17.14 | dijib | oh do i have > |
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| 06:17.43 | dijib | still doesnt work |
| 06:17.54 | dijib | anywhere where there is ${variable} |
| 06:18.05 | dijib | in the verbose cli output i dont see the variable |
| 06:18.10 | dijib | is that normal? |
| 06:18.52 | kaldemar | pastebin |
| 06:19.34 | dijib | http://pastebin.com/SXCS6LLW |
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| 06:23.53 | kaldemar | dijib: and the dialplan? |
| 06:24.43 | kaldemar | does text2wave read from stdin? |
| 06:24.57 | dijib | uhm? |
| 06:25.32 | dijib | i believe so |
| 06:25.49 | kaldemar | can you pipe text to it? i know you can redirect a file to it, which makes reading a file to a channel variable just nonsense. |
| 06:26.11 | kaldemar | func READ should be replaced by FILE. |
| 06:26.18 | dijib | i can echo text | to the rest of the command |
| 06:26.28 | dijib | just cant get the variable to read the file |
| 06:26.59 | kaldemar | but to do that all in a single line, just use System(text2wave < /tmp/forecast -F 8000 -otype ulaw /var/lib/asterisk/sounds/en/forecast.ulaw) |
| 06:27.37 | kaldemar | reading the file to a variable is just unneeded extra in the dialplan. |
| 06:27.55 | kaldemar | my System is missing "-o". |
| 06:30.49 | dijib | so that should do it in one line? |
| 06:31.18 | dijib | the command options and output location and type is what was messing me up.. let me try that |
| 06:33.05 | dijib | no action |
| 06:33.19 | dijib | "text2wave < /tmp/forecast -F 8000 -otype ulaw -o /var/lib/asterisk/sounds/en/forecast.ulaw" |
| 06:33.53 | dijib | same => n,System(text2wave < /tmp/forecast -F 8000 -otype ulaw -o /var/lib/asterisk/sounds/en/forecast.ulaw); |
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| 06:35.42 | kaldemar | works here.. maybe you have a path issue, put full path fo text2wave. |
| 06:38.18 | dijib | /usr/sbin/ ? is it in? |
| 06:38.43 | SunTsu | which text2wave |
| 06:39.00 | dijib | yes k its in there testing.... |
| 06:39.28 | dijib | do i have that -o in the right place? |
| 06:39.39 | dijib | should be. |
| 06:39.44 | dijib | wtf |
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| 07:00.14 | *** join/#asterisk bis0n (~56416@78.238.86.105) |
| 07:00.17 | bis0n | hello |
| 07:00.47 | bis0n | have an problem, channel.c:4201 ast_request: No channel type registered for 'DAHDI' / Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) |
| 07:01.03 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-129-148.chyn.qwest.net) |
| 07:01.05 | bis0n | this is my conf: http://pastebin.com/KDRCke3g |
| 07:01.12 | bis0n | I don't understand.... |
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| 07:04.13 | kaldemar | bis0n: for some reason, you don't have chan_dahdi.so module loaded. what does "module load chan_dahdi.so" in CLI tell you (with verbosity enabled)? |
| 07:04.46 | bis0n | load_resource: Module 'chan_dahdi.so' already exists. |
| 07:05.20 | kaldemar | "module unload chan_dahdi.so" and try again |
| 07:05.34 | bis0n | no reload? |
| 07:07.28 | kaldemar | reload just unloads and loads the modules again. just unload and load. |
| 07:08.07 | bis0n | ok, after load module: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 07:08.18 | bis0n | not same error, but error :/ |
| 07:08.40 | kaldemar | and you got no errors on the module load command? |
| 07:09.05 | kaldemar | what does "dahdi show channels" say? |
| 07:09.35 | bis0n | <PROTECTED> |
| 07:09.35 | bis0n | <PROTECTED> |
| 07:09.45 | bis0n | soryy for c/c |
| 07:10.06 | bis0n | sorry for c/p * |
| 07:10.42 | bis0n | that is correct, no? |
| 07:11.10 | bis0n | http://pastebin.com/cjYM5AFz |
| 07:13.02 | bis0n | :) |
| 07:13.07 | bis0n | that's ok now! |
| 07:13.13 | bis0n | thx u very much |
| 07:21.11 | bis0n | http://pastebin.com/UW0cn8W4 |
| 07:21.30 | bis0n | but no ring... |
| 07:22.25 | bis0n | just one "tuuut" at first when asterisk hangup, but no more... like dahdi no compose... |
| 07:25.32 | *** join/#asterisk like_a_horse (~like_a_ho@firesl.saao.ac.za) |
| 07:26.20 | like_a_horse | hi all, anyone know a way for authing against another mechanism instead of the passwords in voicemail.conf when checking voicemail? |
| 07:26.34 | like_a_horse | i'd like my pins for dialing out and my voicemail passes the same |
| 07:27.06 | like_a_horse | any elegant way of doing this? |
| 07:34.48 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 07:36.35 | bis0n | how to remove an red alarm on dahdi |
| 07:36.37 | bis0n | ? |
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| 07:40.10 | kaldemar | bis0n: connect the cable. |
| 07:40.37 | bis0n | yes it is ok... now ^^ |
| 07:41.34 | kaldemar | like_a_horse: use option s for VoiceMailMain to disable password checking and do it with app Authenticate or build your own with something else. |
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| 07:45.13 | like_a_horse | kaldemar, thanks! |
| 07:45.37 | like_a_horse | you wouldn't have an example config laying around would you? |
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| 07:53.38 | kaldemar | like_a_horse: no. how exactly do you want to do it? |
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| 07:55.12 | like_a_horse | ok, just tested and it doesn't ask for a password now |
| 07:55.41 | like_a_horse | so i guess i need to shove in the Authenticate before the VoiceMailMain app |
| 07:55.41 | *** join/#asterisk TobSnyder (~schneider@91-64-124-121-dynip.superkabel.de) |
| 07:55.48 | like_a_horse | think i got it now |
| 07:55.52 | like_a_horse | will let u know |
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| 08:01.05 | moobius | hello! Does anyone know if it somehow possible to find out on asterisk which side terminated a call? |
| 08:01.37 | *** part/#asterisk TobSnyder (~schneider@91-64-124-121-dynip.superkabel.de) |
| 08:06.53 | bis0n | when I'm not on local network, I get this error with my sip phone: __sip_xmit: sip_xmit of 0x2ccd300 (len 565) to 10.42.182.128:56315 returned -1: Operation not perm |
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| 08:08.38 | moobius | bis0n: the ip you are binding on is not available? wrong routes ? |
| 08:08.59 | bis0n | on sip.conf i have bindaddr=0.0.0.0 |
| 08:10.44 | moobius | bis0n: most likely there is something wrong with your network. can you ping/traceroute to 10.42.182.128 from the asterisk box? |
| 08:12.44 | bis0n | :/ ping not seems working |
| 08:12.56 | wdoekes2 | bis0n: is that your softphones IP or is it a completely random IP? (do you have a username 170571392?) |
| 08:13.15 | bis0n | 10.42.182.128 is an 3G connexion... |
| 08:13.18 | wdoekes2 | moobius: you can have Dial() continue after the other end hangs up |
| 08:14.17 | wdoekes2 | bis0n: then asterisk doesn't know it's behind NAT, I suppose? are you running a recent asterisk? |
| 08:15.01 | wdoekes2 | .. and do you have the nat= setting set to something reasonable? |
| 08:15.02 | moobius | wdoekes2: what good will do that? (continue on after Dial()) |
| 08:15.20 | *** join/#asterisk jpsharp (~jsharp@postman.fivecats.org) |
| 08:15.22 | kaldemar | moobius: options g/F and a hangup extension. |
| 08:15.41 | wdoekes2 | moobius: (a) Dial() (b) Set(PEERHUNGUP=1) (c) Hangup (d) h .. ${PEERHUNGUP} or not |
| 08:15.42 | kaldemar | moobius: core show application Dial |
| 08:16.24 | jpsharp | I'm having issues with the Digium FAX stuff + T38. Anyone around who can help? |
| 08:16.49 | wdoekes2 | ~ask |
| 08:16.49 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:17.07 | moobius | ah, so If the other side hangs up the dial() will go directly to h? |
| 08:17.20 | wdoekes2 | almost |
| 08:17.33 | wdoekes2 | the other way around |
| 08:18.02 | like_a_horse | kaldemar, do you know of a mysql plugin for the Authenticate() app ? |
| 08:18.06 | like_a_horse | like to auth via mysql |
| 08:18.09 | moobius | jpsharp: I have written a quite good (i believe) tutorial on setting up t38 via t38modems and asterisk. I don't know if you are interested or just want to use the app_fax |
| 08:18.28 | moobius | wdoekes2: ehehe, ok fair enough. I will give it a try :) thanks |
| 08:18.44 | kaldemar | like_a_horse: there is no mysql plugin for it. you could define a function through ODBC though. |
| 08:19.08 | jpsharp | moobius: I'm using the Digium free fax stuff. It works mostly okay (as to be expected across the internet) with G711, but I want to use T38 for reliability. |
| 08:19.14 | like_a_horse | and Authenticate() uses ODBC? |
| 08:20.11 | jpsharp | The call gets placed, the far end answers, Asterisk waits & then sends a reinvite to initiate T.38, and then just seems to completely ignore the incoming T38 data. |
| 08:20.26 | kaldemar | like_a_horse: it doesn't use anything. it just takes an option. see http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html and especially func_odbc. |
| 08:21.05 | like_a_horse | cool |
| 08:21.10 | like_a_horse | thanks |
| 08:21.19 | schmidts | anyone using chan_ss7? i need some help for the initial setup, i can even load chan_ss7.so cause i get some errors :( |
| 08:21.41 | moobius | jpsharp: have you traced it? does the payload indeed containt t38? |
| 08:21.47 | moobius | s/containt/contain |
| 08:21.54 | jpsharp | Yep. Wireshark sez it is T38. |
| 08:22.22 | *** join/#asterisk bis0n (~56416@78.238.86.105) |
| 08:22.25 | bis0n | re |
| 08:23.04 | bis0n | sorry, kernel error Oo :/ |
| 08:23.52 | jpsharp | I see bidirectional communications with G711, then the reinvite to T38 goes through, and the far end gateway spits out T38 datagrams until Asterisk times out and drops the call. |
| 08:25.12 | moobius | jpsharp: hm, weird. does asterisk print out anything (I guess you have corrently enabled the t38 udtpl options?) |
| 08:26.54 | jpsharp | FAX session started, "STAT_EVT_STRT_TX", "STAT_EVT_TX_HW_RDY", "P30EVN_SEND_STARTED", then "fax session timeout". |
| 08:28.09 | jpsharp | That's if I send SendFAX with the "z" option to initiate the T38 reinvite (which googling says I have to do with Gafachi). If I don't use that option, Asterisk just squawks about not being able to negotiate T38. |
| 08:31.38 | moobius | jpsharp: it is an outgoing fax? |
| 08:34.21 | jpsharp | yes |
| 08:34.22 | like_a_horse | kaldemar, i'm using mysql directly from my dialplan to do lookups on which CID to use on outbound calls,etc |
| 08:34.39 | like_a_horse | odbc will do the same thing it seems |
| 08:34.48 | like_a_horse | just need to setup a few more config files |
| 08:34.51 | *** join/#asterisk AlecTaylor (~alec.tayl@unaffiliated/alectaylor) |
| 08:34.53 | AlecTaylor | hi |
| 08:35.53 | like_a_horse | so if have a clear text pin in my mysql db and then write a few lines in my dialplan to check that the pin entered matches the pin in the mysql db it would work nicely |
| 08:36.00 | like_a_horse | wondering if anything like this has been done already |
| 08:36.15 | AlecTaylor | sqlite might be better |
| 08:36.19 | AlecTaylor | oh |
| 08:36.22 | AlecTaylor | hmm |
| 08:36.27 | AlecTaylor | I've seen a script like that before |
| 08:37.01 | like_a_horse | AlecTaylor, i'm all ears ;) |
| 08:37.23 | AlecTaylor | hmm |
| 08:37.31 | AlecTaylor | It was on sourceforge |
| 08:37.34 | AlecTaylor | One of the GUIs |
| 08:37.41 | AlecTaylor | umm... |
| 08:40.01 | moobius | jpsharp: have you trying setting d on SendFax to see what it prints out? |
| 08:45.41 | jpsharp | moobius: I did. It prints out only what I showed earlier. |
| 08:45.47 | jpsharp | Not especially chatty. |
| 08:46.42 | jpsharp | I gots to head off to bed, though. I'll beat on it tomorrow. Thanks for your help. |
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| 08:54.06 | kaldemar | like_a_horse: how are you using mysql from the dialplan? |
| 08:54.34 | kaldemar | like_a_horse: how as in with which application/function? |
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| 09:00.09 | like_a_horse | kaldemar, http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
| 09:00.09 | like_a_horse | pretty much that |
| 09:00.09 | like_a_horse | with the AGI stuff |
| 09:00.10 | like_a_horse | so i have a couple exten =>... MYSQL(.....) lines in a context called setcid |
| 09:00.12 | like_a_horse | which i use to setcid based extension numbers |
| 09:00.54 | like_a_horse | which are in mysql |
| 09:02.51 | moobius | like_a_horse: this might sound complicated but I usually seperate the application logic (mysql queries etc) from the dialplan, and so I have a php script being called by ${CURL()} to do all the job. |
| 09:03.17 | moobius | like_a_horse: I usually prefork the apache to wait for multiple connections and it has never failed me so far |
| 09:03.36 | moobius | like_a_horse: therefore no agi, no mysql/odbc/etc |
| 09:04.00 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:05.37 | like_a_horse | moobius, so u use curl to connect to apache which auth's against a db(mysql?) ? |
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| 09:07.27 | *** part/#asterisk TobSnyder (~schneider@91-64-124-121-dynip.superkabel.de) |
| 09:07.48 | kaldemar | like_a_horse: just get the PIN into a variable and then Authenticate(${variable}) |
| 09:08.23 | kaldemar | like_a_horse: don't mix apache into this, especially when scripts can be called directly from the dialplan. |
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| 09:08.53 | moobius | kaldemar: i don't agree. I believe curl is a much safer option for a production environment with hundred of calls that using mysql directly |
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| 09:10.32 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:12.02 | moobius | especially on latest asterisk versions where you can define the timeout as well |
| 09:12.07 | Gugge | moobius: so using curl+webserver+php+mysql libs, is safer than using mysql libs? :) |
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| 09:12.10 | like_a_horse | kaldemar, we have very small setup here. maybe 100 phones |
| 09:12.33 | like_a_horse | kaldemar, thanks for auth variable tip |
| 09:13.10 | kaldemar | moobius: i like to keep things simple when there is no need to make them complex. |
| 09:14.11 | Gugge | i prefer to use odbc though |
| 09:14.16 | moobius | kaldemar: true I agree. when it's something simple |
| 09:14.35 | moobius | Gugge: well yes. |
| 09:14.50 | moobius | Gugge: I've had many problems in the past the mysql support in asterisk. |
| 09:15.25 | Gugge | sure, but mixing in a webserver and php is not something i would recommend |
| 09:15.35 | Gugge | switching to odbc is much better :) |
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| 09:18.09 | moobius | Gugge: safer in comparison to mysql I would say. :) |
| 09:18.46 | moobius | Gugge: I am thinking of using node.js to host my applications in the future. I think it could keep up to more requests with less load |
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| 09:53.41 | like_a_horse | kaldemar, this is getting interesting.. So each user is a given a 4 digit pin to dial out with. Their voicemail pins are different (also 4 digits). There is a good chance the user will dial out from different locations so I can't use cid to lookup what the password should be and set it for Authenticate(${variable}) to evaluate. |
| 09:54.19 | like_a_horse | I want to be able to auth and against mysql for voicemail and outgoing calls with the same pin |
| 09:55.20 | *** join/#asterisk TobSnyder (~schneider@91-64-124-121-dynip.superkabel.de) |
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| 09:58.20 | kaldemar | like_a_horse: then just ask for a user id. |
| 10:00.08 | like_a_horse | nods |
| 10:00.10 | like_a_horse | got ya |
| 10:01.21 | *** join/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 10:02.02 | zealiod | Can any one help me with this error… started to get it when I changed from sip gate to voiptalk. WARNING[1147]: channel.c:3201 ast_channel_make_compatible: No path to translate from SIP/5003-00a73300(256) to SIP/217.14.138.131-00a776b0(8) |
| 10:02.08 | *** part/#asterisk TobSnyder (~schneider@91-64-124-121-dynip.superkabel.de) |
| 10:04.21 | schmidts | zealiod looks you dont have the right codecs they want |
| 10:05.48 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 10:05.59 | kaldemar | zealiod: 256 is G.729A and 8 is G.711A. looks like you don't have a G.729A codec. |
| 10:06.00 | zealiod | schmidts: How can i tell what they are using> |
| 10:06.24 | zealiod | kaldemar: OK, I'll dig around - thanks |
| 10:06.37 | kaldemar | zealiod: in that call, the leg to the provider was using G.711A. |
| 10:07.22 | zealiod | my sip.conf does have allow = g729 |
| 10:08.57 | zealiod | Im a bit lost - what do i need to do? |
| 10:11.03 | kaldemar | zealiod: do you have the g729 codec? |
| 10:11.52 | zealiod | kaldemar: how can i tell> |
| 10:11.56 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 10:12.37 | kaldemar | zealiod: if you can't tell, then you don't have it. you'd have to purchase a lisence for it. just remove or comment out the line that allows g729. |
| 10:13.07 | kaldemar | license that is. |
| 10:19.16 | zealiod | kaldemar: I have commented out the line - reloaded, but still the same thing |
| 10:20.05 | zealiod | kaldemar: I also get this error: channel.c:2811 set_format: Unable to find a codec translation path from ulaw to g729 |
| 10:21.28 | like_a_horse | kaldemar, Application for wait for user input? |
| 10:22.16 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 10:28.48 | kaldemar | like_a_horse: app Read |
| 10:29.24 | kaldemar | zealiod: you might have it configured in many places ot the phone might be forcing g729. |
| 10:30.48 | like_a_horse | thanks |
| 11:01.45 | zealiod | kaldemar: So I'm downloading the .so binaries - I'm confused as to what processor I have… is this important? |
| 11:04.08 | wdoekes2 | zealiod: hardly, but you can run the benchmark tool |
| 11:04.24 | zealiod | what is that> |
| 11:04.53 | *** join/#asterisk AlecTaylor (~alec.tayl@unaffiliated/alectaylor) |
| 11:05.50 | catphish | i'm using monitor with /b, is there any way i can note in the CDR whether a call was recorded or not? |
| 11:08.00 | kaldemar | you can add new fields to CDR. |
| 11:08.21 | catphish | yes i do that, in fact i save the call recording path |
| 11:08.35 | catphish | but i have no way to determine if the recording actually happened or not |
| 11:08.39 | kaldemar | after the call, do a check and add the result to CDR. |
| 11:11.25 | catphish | that could work |
| 11:11.37 | catphish | not sure where to set the post-call code to run |
| 11:12.05 | kaldemar | in the extension where the Dial is made or in a hangup extension. |
| 11:13.15 | catphish | that's fine, just need to actually relate that to my excessively complicated dialplan :) |
| 11:14.25 | catphish | which context's 'h' is executed at the end of a call? |
| 11:14.49 | catphish | the final one? |
| 11:16.00 | Gugge | the one the call is in at the end |
| 11:16.14 | catphish | thanks |
| 11:17.53 | catphish | actually... i think i can use the dstchannel field |
| 11:18.16 | catphish | if the call isn't bridged it won't have a dstchannel |
| 11:19.02 | catphish | that seems to work :) |
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| 11:43.36 | zealiod | trying to build this mod is a pain in the ass… I'm getting this error: configure: error: IPP not found - use --with-ip=/path/to/ipp/version/arch to specify - even though I do specify it |
| 11:43.37 | zealiod | ./configure --with-ip=/opt/intel/ipp |
| 11:43.42 | zealiod | any help appreciated |
| 11:44.48 | Dovid | zealiod: What OS ? |
| 11:44.57 | zealiod | debian |
| 11:45.03 | zealiod | Dovid: debian |
| 11:45.34 | Dovid | argh. I use CentOS |
| 11:45.52 | Dovid | ooh. i have deb here in office. leme see |
| 11:47.39 | Dovid | maybe aptitude search ipp >\ |
| 11:47.41 | Dovid | ?* |
| 11:47.54 | wdoekes2 | zealiod: there are precompiled g729 .so's out there |
| 11:48.23 | zealiod | wdoekes2: Im using a rackspace vm - wasn't sure what processor to go for, etc... |
| 11:48.29 | zealiod | wdoekes2: thought best to try and compile em |
| 11:49.58 | *** join/#asterisk cusco_ (~tralala@88.157.128.26) |
| 11:50.02 | cusco_ | hello |
| 11:52.49 | *** join/#asterisk reber (~reber@178.33.192.26) |
| 11:53.45 | cusco_ | im using a file .call to generate a callm right. the Channel is some local dialplan where I set some vars, and then context/exten is another bit of dialplan where I would like to re-use variable that I just set in previous dialplan |
| 11:53.50 | cusco_ | is this possible? |
| 11:55.39 | wdoekes2 | you can manipulate globals |
| 11:55.47 | wdoekes2 | but if it's wise is another question |
| 11:56.15 | wdoekes2 | you're probably better off using one of the DB methods to store/fetch data |
| 11:56.34 | wdoekes2 | (dbget/dbput or realtime) |
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| 11:57.00 | cusco_ | yes I thought about that and forgot. thanks |
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| 12:01.56 | grEvenX | possible to log custom CDR fields to ODBC/PGSQL storage now? |
| 12:06.05 | Dovid | cusco_: Two sepeate leg calls or one going to another (like local channel) ? |
| 12:10.24 | carrar | Y*A*W*N |
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| 12:27.38 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:33.29 | cusco_ | Dovid: im really not sure... |
| 12:33.57 | cusco_ | Dovid: a call file. I specify the Channel at the top, then the Context/exten that it goes to when answered |
| 12:34.07 | cusco_ | but the channel is a loca, that goes to queue() |
| 12:34.28 | cusco_ | so when agent answers goes the the specified Context/exten ... in there I don't have the agent peer number |
| 12:35.20 | cusco_ | wdoekes2: about db |
| 12:35.29 | cusco_ | how do I delete a value from DB? using 1.6.2 here |
| 12:35.56 | cusco_ | pbx.c: No application 'DB_DELETE' for extension |
| 12:36.51 | [TK]D-Fender | cusco_, DBDel |
| 12:37.02 | cusco_ | ah, thanks |
| 12:39.29 | kaldemar | cusco_: DB_DELETE is a function, not an application. |
| 12:41.23 | *** join/#asterisk fobus912 (~fobus912@41.141.248.71) |
| 12:41.36 | zealiod | Im getting status is 'CHANUNAVAIL' when making an outgoing call - what am i doing wrong? |
| 12:42.06 | [TK]D-Fender | zealiod, First would be believing that message was meaningful :) |
| 12:42.22 | zealiod | heh |
| 12:42.28 | [TK]D-Fender | zealiod, lets look at the actual complete call. |
| 12:42.45 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:42.51 | zealiod | [TK]D-Fender: what do i need - I'm in -v (x14) |
| 12:43.00 | [TK]D-Fender | zealiod, lets look at the actual complete call. <- |
| 12:43.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:44.17 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:45.04 | zealiod | [TK]D-Fender: How |
| 12:45.26 | [TK]D-Fender | zealiod, You're already in CLI. Show us the call |
| 12:46.26 | zealiod | [TK]D-Fender: WARNING[25227]: chan_iax2.c:7832 socket_process: Call rejected by 217.14.138.130: No authority found |
| 12:46.27 | zealiod | <PROTECTED> |
| 12:46.27 | zealiod | <PROTECTED> |
| 12:46.27 | zealiod | <PROTECTED> |
| 12:46.32 | [TK]D-Fender | ~pb |
| 12:46.32 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
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| 12:49.01 | [TK]D-Fender | zealiod, add "requirecalltoken=no" to your peer |
| 12:50.23 | zealiod | [TK]D-Fender: No joy |
| 12:50.45 | [TK]D-Fender | zealiod, try on "auto" and also add to [general] |
| 12:51.25 | [TK]D-Fender | zealiod, What ver of * are you on? |
| 12:51.33 | zealiod | 1.4 |
| 12:52.27 | zealiod | [TK]D-Fender: Nope - no joy |
| 12:54.55 | [TK]D-Fender | ok, those don't apply top you then. check the rest of your auth setup. Odds are you've got something else off... |
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| 13:00.35 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 13:03.02 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 13:03.06 | kareena | hi |
| 13:03.26 | kareena | it is possible to link asterik with carte digium to vocalcom? |
| 13:04.38 | [TK]D-Fender | What is "vocalcom"? |
| 13:05.24 | *** join/#asterisk serafie (~erin@nat/digium/x-tzhhctfwfagshqlm) |
| 13:07.56 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 13:08.20 | kuku | can I use patters for the ex-girlfriend function ? |
| 13:09.08 | SunTsu | kuku: Too many ex-girlfriends at the same time? |
| 13:09.37 | [TK]D-Fender | kuku, Yes |
| 13:15.15 | zealiod | [TK]D-Fender: I think this might be the issue - but Im not sure how to correct it. Any help appreciated. "The 'No authority found' error is generated when an iax entry [example] matching the username sent can not be found." |
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| 13:16.05 | [TK]D-Fender | zealiod, That'd be "wrong auth info" |
| 13:16.58 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 13:17.14 | zealiod | [TK]D-Fender: do I need to explicitly register => |
| 13:18.20 | [TK]D-Fender | zealiod, technically not for the fact you're placing a call. |
| 13:18.36 | zealiod | [TK]D-Fender: Crap, OK |
| 13:25.00 | kareena | www.vocalcom.com |
| 13:25.11 | kareena | system for callcenter |
| 13:27.22 | *** join/#asterisk arekm (~arekm@pld-linux/arekm) |
| 13:27.53 | arekm | hi, my asterisk sends via E1: NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '521228840##' |
| 13:28.03 | [TK]D-Fender | kareena, What CC functionality are you looking at them for? |
| 13:28.15 | arekm | which looks incorrect to me since IMO it should be 5212288## - which options are responsible for this? |
| 13:28.22 | [TK]D-Fender | arekm, Show us the complete call. |
| 13:28.24 | [TK]D-Fender | ~pb |
| 13:28.24 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:28.25 | [TK]D-Fender | ^^^ |
| 13:28.41 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:29.43 | CrossWired | kareena: do you have vocalcom? |
| 13:29.54 | *** join/#asterisk chazzam (~chazz@173.24.236.90) |
| 13:30.27 | arekm | [TK]D-Fender: http://pastebin.com/Dh0rSGs8 |
| 13:30.45 | kareena | calldistribution and recorder |
| 13:30.55 | kareena | yes i have vocalcom |
| 13:31.29 | [TK]D-Fender | kareena, Asterisk already has these features. What are your "pros" for looking at an external piece like this? |
| 13:31.30 | CrossWired | what need would asterisk fill? the IVR portion? |
| 13:31.49 | [TK]D-Fender | arekm, the complete call...... |
| 13:32.14 | zealiod | [TK]D-Fender: All sorted |
| 13:32.14 | CrossWired | [TK]D-Fender: vocalcom is a full fledged skill based routing call center application, has the whole enchilada, reporting, supervision, tons of stuff |
| 13:32.53 | [TK]D-Fender | zealiod, How far off were your peer settings? |
| 13:33.00 | Katty | i love you like a love song baby and i keep hittin repeat-peat-peat-peat-peat-peat! |
| 13:33.27 | kareena | i explaine |
| 13:33.33 | zealiod | [TK]D-Fender: Annoyingly it was provider… account frozen up |
| 13:33.48 | kareena | now i have vocalcom with internet satellite provider that give me minute to france |
| 13:33.57 | kareena | the internet satellite is very expensive |
| 13:33.58 | [TK]D-Fender | zealiod, Yup, you'd be kind of blind till you call them up and here "WTF"... glad its resolved... |
| 13:34.03 | [TK]D-Fender | hear* |
| 13:34.21 | kareena | i want to stop it and get a sip account to make the calls |
| 13:34.30 | kareena | just vocalcom those not support sip |
| 13:34.44 | CrossWired | kareena; yes they do, which version are you running? |
| 13:34.55 | kareena | i have to get a voip gateway E1/T1 to configure the sip account on it |
| 13:35.02 | kareena | 9.5 |
| 13:35.06 | kareena | i have 9.5 |
| 13:35.18 | kareena | the latest version |
| 13:36.06 | CrossWired | kareena: that might be the difference as I have both their Hermes 4.5 Avaya based and their Hermes 4.5 Sip based, I'm not familiar with Pro Agent 9.5 |
| 13:36.21 | CrossWired | kareena: sorry 4.1 not 4.5 |
| 13:37.12 | kareena | yes i have hermes pro |
| 13:37.20 | arekm | [TK]D-Fender: now should be complete. http://pastebin.com/JRY29yGp . The thing is that my caller id is ...7740 while the guy on the other side sees 7700 (and now I'm not sure if the bug is on my side, operator side ..) |
| 13:37.25 | CrossWired | kareena: have you talked to support about your request? which country are you in btw? |
| 13:37.29 | kareena | CrossWired can i private msg you? |
| 13:39.22 | [TK]D-Fender | arekm, your callerID isn't 7740, that is the number you are calling |
| 13:39.53 | [TK]D-Fender | arekm, > Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '523667741' ] <- this is your callerid |
| 13:40.15 | [TK]D-Fender | arekm, Now if they see something else, then it is the telco's fault. |
| 13:40.18 | [TK]D-Fender | arekm, call them |
| 13:40.19 | arekm | [TK]D-Fender: right. I meant calling from 41 to 40 |
| 13:40.22 | [TK]D-Fender | (if that's the case) |
| 13:40.54 | arekm | [TK]D-Fender: is this the indication of that? " [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '3667740' ]" |
| 13:41.34 | arekm | already notified telco but trying to make sure it's not local fault here |
| 13:42.23 | [TK]D-Fender | arekm, No, everything that is identifying you is consistent |
| 13:49.24 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-9.dsl.net.pk) |
| 13:50.25 | mechbangirc | If I make some changes in apps/app_read.c file do I have to recompile whole asterisk or is there a way I can recompile this file alone and copy paste app_read.so in modules dir? |
| 13:51.14 | leifmadsen | mechbangirc: just run 'make' again -- it'll just recompile stuff that's changed |
| 13:51.27 | leifmadsen | (assuming you've already compiled before) |
| 13:51.30 | mechbangirc | cool |
| 13:51.39 | mechbangirc | yes it is compiled already, thanks |
| 13:52.03 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 13:53.21 | arekm | [TK]D-Fender: telco fault - they just fixed it. Thanks for help! |
| 13:54.28 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:54.48 | *** join/#asterisk alexisvv (~alexis@csdsl-1.prestashop.com) |
| 13:54.52 | alexisvv | hi all |
| 13:55.39 | alexisvv | i don't know if it's possible to ask to digium to revoke our current licence key in order to make a new order with more channels for g.729 codec |
| 13:55.56 | alexisvv | is there a way to do it ? :) |
| 13:56.49 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 13:57.15 | *** join/#asterisk darkdrgn2k (~darkdrgn@67.211.114.226) |
| 13:57.17 | darkdrgn2k | Morning everyone |
| 13:57.25 | [TK]D-Fender | arekm, You're welcome. |
| 13:57.49 | timeshell | Greetings. I have been having issues in 1.8.x where a blind transfer to a Bria phone using TCP transport causes the Bria phone to ring but sends the call directly to voicemail. |
| 13:57.56 | timeshell | Any thoughts on this? |
| 13:58.26 | darkdrgn2k | quick question about CNAMs and CLID, i have a client that usses an outsourced NORTEL telephone system, the provider is telling me that onace a "domain" clid is set, they cannot update the CNAM database for an individual user |
| 13:58.28 | timeshell | Specific asterisk version at present is 1.8.6 but the issue also occured in 1.8.4 |
| 13:58.38 | timeshell | Hey TK |
| 13:58.39 | _Corey_ | alexisvv: You can have multiple g729 licenses stacked on a machine |
| 13:59.03 | darkdrgn2k | I was under the impression that VOIP lines do NOT pass cname information to pots, just clid? |
| 13:59.09 | *** join/#asterisk celord (~celord@201.202.104.220) |
| 13:59.28 | darkdrgn2k | in that case are they incorrent in saying they cannot update the cnam database for one user? |
| 14:00.10 | [TK]D-Fender | timeshell, y0 |
| 14:00.57 | _Corey_ | darkdrgn2k: Sounds like a provider limitation, everyone I've ever used was able to set CNAM per DID |
| 14:01.12 | [TK]D-Fender | darkdrgn2k, your scale is too vague. |
| 14:01.13 | darkdrgn2k | they CAN set cnams per DID |
| 14:01.25 | alexisvv | _Corey_: we can use 2 or more licence under one asterisk server ? |
| 14:01.35 | darkdrgn2k | but i have two options, one option to use the VOIP "account" name (so my firstname lastname) or a blanket name for the company (COMNPANY XYZ) |
| 14:01.36 | [TK]D-Fender | darkdrgn2k, They can do general whatever they want. |
| 14:01.45 | alexisvv | hum strange i don't read it that way but it's great if it's possible |
| 14:02.23 | alexisvv | do you have any documentation about that _Corey_ ? :) |
| 14:02.41 | darkdrgn2k | i gues im trying to understand how cnam databases get updated? |
| 14:02.49 | Katty | soo much sleepy |
| 14:03.01 | _Corey_ | alexisvv: I can confirm that I have a system with two licenses, one 250 channel lic. and one 20 channel license |
| 14:05.37 | alexisvv | ok _Corey_, good to know :) |
| 14:06.15 | alexisvv | _Corey_: digium can refund old licence or not ? (i would like to confirm if it's possible :)) |
| 14:06.45 | _Corey_ | alexisvv: I couldn't confirm, as I'd say it's unnecessary |
| 14:07.17 | alexisvv | ok thank you so much :) |
| 14:08.05 | *** join/#asterisk master_of_master (~master_of@87.181.75.61) |
| 14:10.22 | *** join/#asterisk mtltemplar (~mtltempla@204.134.45.21) |
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| 14:11.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:12.35 | mtltemplar | Hello. I am trying to allow International in my dialplan. I have _9011. in my plan. when a user dials 9011XXXXXXXXX it will ring. if they try to dial any numbers beyond that it drops them to a release and try again busy signal. |
| 14:13.02 | mtltemplar | how to i allow 4 more digits than it seems to be allowing? i though the . represented *any* number of digits after that being allowed? |
| 14:13.11 | mtltemplar | is that the phone device itself stopping it? |
| 14:16.23 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:16.34 | leifmadsen | mtltemplar: that pattern match isn't the problem -- . is unlimited |
| 14:16.51 | leifmadsen | sounds like the service provider could be refusing it for some reason |
| 14:17.14 | leifmadsen | just a guess -- would need to see more dialplan, but _9011. has no length limit |
| 14:17.26 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 14:17.52 | *** join/#asterisk timahvo1 (~rogue@41.215.65.50) |
| 14:18.26 | mtltemplar | Okay. I will check with our trunk provider |
| 14:18.27 | mtltemplar | thanks |
| 14:18.35 | mtltemplar | figured that it wasnt that dialplan |
| 14:19.34 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 14:20.08 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:20.47 | darkdrgn2k | whats the app that allows a ping of all points along the line? |
| 14:21.06 | [TK]D-Fender | traceroute |
| 14:21.30 | darkdrgn2k | yes but theres one that will coninuaslly ping and show deltas between ping |
| 14:22.05 | p3nguin | mtr |
| 14:22.09 | darkdrgn2k | thats it thank you |
| 14:25.35 | darkdrgn2k | is 60ms to high of a ping for voip to work well on |
| 14:27.02 | zealiod | Is it possible to change the outgoing caller-id from astericks |
| 14:27.39 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
| 14:27.47 | [TK]D-Fender | darkdrgn2k, No, |
| 14:28.00 | darkdrgn2k | ok thank you |
| 14:28.01 | [TK]D-Fender | zealiod, depends on your provider |
| 14:28.07 | *** join/#asterisk n0h0pe (~tester@237-41.187-72.tampabay.res.rr.com) |
| 14:28.17 | zealiod | [TK]D-Fender: what would be the set up to test |
| 14:28.45 | [TK]D-Fender | zealiod, Chenge your callerid. Dial. See. Confirm with provider what they allow. |
| 14:29.07 | luckman212 | is it possible to "dump" a call into a voicemail box without answering the call first? e.g. call from x101 --------> x501 which is a ring group that rings 201 & 202 & 203. User 201's phone starts ringing and he wants to divert that call to his voicemail box --- is there any way to do that without Answering the call first (if he just does a SIP REJECT then x202 and 203 continue to ring...) |
| 14:29.07 | [TK]D-Fender | zealiod, Many won't allow at all. Some will allow you to set to any number they provide to you (DID), otehrs are unrestricted |
| 14:30.00 | luckman212 | i'm trying to make a 'dump to voicemail' button on my polycom phone |
| 14:30.23 | [TK]D-Fender | luckman212, check the DIALSTATUS after your dial. You'll need to dial multiple Local channels that will spawn the actual calls. |
| 14:30.54 | n0h0pe | can someone give me some hints, privately, why I am not receiving VM on an older v1.4.21.4 system ? |
| 14:31.35 | luckman212 | [TK]D-Fender: i think that is already happening ... how would I check that? |
| 14:31.49 | [TK]D-Fender | luckman212, "think"? |
| 14:32.05 | [TK]D-Fender | luckman212, this is your dialplan. you're either dialing local channels or not. |
| 14:32.07 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:32.39 | [TK]D-Fender | n0h0pe, clarify "not receiving vm" please. |
| 14:32.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:34.30 | luckman212 | [TK]D-Fender: I'm dialing SIP channels e.g. SIP/201 |
| 14:35.06 | [TK]D-Fender | luckman212, the try what I just told you to do. Dial Local channel and have them call the individual devices and chack the dialstatus for the reject. |
| 14:35.47 | n0h0pe | I see /var/spool/asterisk/voicemail/default/202/tmp/xxxx then I see /usr/pluto/bin/SendVoiceMailEvent.sh 202 then nothing. I get no VM and it dosen't appear in the /var/spool/asterisk/voicemail/default/202/INBOX |
| 14:36.03 | zealiod | [TK]D-Fender: Can I change the caller-id at a user level? |
| 14:36.18 | [TK]D-Fender | zealiod, yes, and in the dialplan |
| 14:36.51 | *** join/#asterisk jimbo_uk (~IceChat77@84.12.253.146) |
| 14:36.54 | jimbo_uk | hola |
| 14:36.57 | jimbo_uk | ok |
| 14:37.13 | jimbo_uk | pattern matching a + in an exten -> _+X.. |
| 14:37.15 | jimbo_uk | for E164 |
| 14:37.19 | jimbo_uk | any ideas |
| 14:37.27 | [TK]D-Fender | n0h0pe, check your permissions and pastebina complete call attempt along with a complete folder dump from the root of the VM box |
| 14:37.27 | luckman212 | [TK]D-Fender: ok thanks. I'm using Freepbx so I dont have direct control over the dialplan :( |
| 14:37.36 | jimbo_uk | X Z N are are numbers |
| 14:37.42 | leifmadsen | jimbo_uk: you just did it.... |
| 14:37.52 | jimbo_uk | [az] [AZ] cover letters |
| 14:37.52 | _Corey_ | jimbo_uk: Ideas for what? |
| 14:37.55 | [TK]D-Fender | jimbo_uk, And "+" is just a fixed char. "+" is "+" |
| 14:37.58 | jimbo_uk | really? |
| 14:38.00 | leifmadsen | yes |
| 14:38.06 | leifmadsen | + is not anything pattern matched |
| 14:38.10 | leifmadsen | means nothing |
| 14:38.10 | [TK]D-Fender | jimbo_uk, It is not symbolic, it is literal |
| 14:38.16 | jimbo_uk | huh |
| 14:38.21 | jimbo_uk | so i do _+X. |
| 14:38.25 | leifmadsen | yes |
| 14:38.27 | jimbo_uk | or _"+"X. |
| 14:38.32 | leifmadsen | _[+]X. would probably work too |
| 14:38.33 | [TK]D-Fender | You CAN. If you want to is another matter. |
| 14:38.39 | [TK]D-Fender | :) |
| 14:38.44 | leifmadsen | jimbo_uk: no -- "+" would not work |
| 14:38.53 | jimbo_uk | ok didn't think so |
| 14:38.57 | [TK]D-Fender | Are you dialing on something that can actually send a "+"? |
| 14:38.59 | leifmadsen | it would probably try to match the double quotes literally |
| 14:39.03 | jimbo_uk | yes |
| 14:39.14 | jimbo_uk | sip trunks delivering calls to us as E164 numbers |
| 14:39.28 | [TK]D-Fender | jimbo_uk, Well you can use "+" if you need to is another matter. |
| 14:39.49 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:39.50 | _Corey_ | jimbo_uk: I use _X. for that |
| 14:39.55 | _Corey_ | er |
| 14:39.59 | jimbo_uk | no Corey that won't work |
| 14:40.02 | _Corey_ | _+X. |
| 14:40.05 | jimbo_uk | X meaes 0-9 |
| 14:40.11 | jimbo_uk | ah +X. |
| 14:40.13 | _Corey_ | there was something under my + key :) |
| 14:40.16 | jimbo_uk | fine, thanks guys!! |
| 14:40.20 | jimbo_uk | lol |
| 14:40.26 | jimbo_uk | darn biscuits! |
| 14:40.31 | n0h0pe | TKJD-Fender: is atsreisk -vvvvvcr enough info ? |
| 14:40.36 | jimbo_uk | _+X. it is then |
| 14:40.38 | n0h0pe | astreisk even |
| 14:40.42 | n0h0pe | aghh |
| 14:40.59 | [TK]D-Fender | n0h0pe, core debug 10 as well |
| 14:41.04 | n0h0pe | OK |
| 14:43.38 | Faustov | where did the soft hangup channel command go? |
| 14:44.07 | WIMPy | channel request hangup |
| 14:44.08 | _Corey_ | Faustov: channel request hangup ... |
| 14:44.22 | Faustov | great, thanks |
| 14:45.54 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
| 14:47.10 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 14:48.22 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:48.26 | [TK]D-Fender | n0h0pe, ... |
| 14:48.27 | [TK]D-Fender | ~pb |
| 14:48.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:48.29 | [TK]D-Fender | ^ |
| 14:51.08 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:51.56 | *** join/#asterisk irroot (~irroot@41-134-138-158.dsl.mweb.co.za) |
| 14:53.08 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:53.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:00.29 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:03.23 | *** join/#asterisk cerberus_za (~coert@8ta-151-92-68.telkomadsl.co.za) |
| 15:05.15 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:15.07 | Katty | ohai |
| 15:18.07 | WIMPy | Who needs Popeye, if you've got Katty the sa-a-a-ailor? |
| 15:18.24 | Nugget | yay Popeye's |
| 15:19.32 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 15:21.24 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:22.12 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
| 15:22.28 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 15:26.00 | *** join/#asterisk cerberus_za (~coert@8ta-151-95-122.telkomadsl.co.za) |
| 15:27.01 | p3nguin | With regard to fax, is T.38 used for any other reason than fax on a real fax machine? |
| 15:27.29 | Qwell | whats your definition of "real fax machine"? |
| 15:27.40 | p3nguin | a real fax machine, of course. |
| 15:27.49 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:27.55 | Qwell | there's no reason at all that 2 Asterisk boxes couldn't talk to each other and send a fax. |
| 15:28.05 | [TK]D-Fender | p3nguin, fax that crosses VoIP |
| 15:28.07 | Qwell | with no "real fax machine" on either end |
| 15:28.11 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 15:28.35 | p3nguin | But that's not what I'm asking. I'm asking if T.38 is used when a fax machine is not involved. |
| 15:29.19 | p3nguin | In the case of two Asterisks faxing each other, would T.38 ever be used? |
| 15:29.28 | [TK]D-Fender | p3nguin, yes |
| 15:29.36 | [TK]D-Fender | Asterisk = fax machine at that point |
| 15:29.49 | [TK]D-Fender | sending a fax means being a fax machine |
| 15:29.56 | [TK]D-Fender | Same with receiving. |
| 15:30.15 | p3nguin | If you ask that one guy, that would make ME a fax machine. :/ |
| 15:30.15 | [TK]D-Fender | T.38 is just to ensure that network conditions don't nuke carrier |
| 15:30.32 | coppice | any real fax machine makes clunky noises. its how you tell they are real |
| 15:31.45 | p3nguin | I'm using res_fax and res_fax_digium, but I haven't configured anything specifically for T.38. Is it possible that T.38 could be used when I send or receive and I just didn't know about it? |
| 15:32.03 | p3nguin | faxing over sip |
| 15:32.07 | [TK]D-Fender | p3nguin, No. there are sip.conf settings to be made for it |
| 15:32.12 | p3nguin | okay |
| 15:32.16 | irroot | p3nguin yes T38 will be used with res_fax/spandsp |
| 15:32.21 | coppice | why would you really care? |
| 15:32.24 | [TK]D-Fender | no settings = not set = doesn't get negotiated |
| 15:32.36 | irroot | if it detects T38 it switches over |
| 15:33.03 | irroot | but the endpoint must intiate it ie be allowed in the channel config [sip/ooh323] |
| 15:35.22 | Katty | hugs WIMPy |
| 15:35.24 | Katty | hugs Nugget |
| 15:35.35 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:37.01 | p3nguin | I'm not doing any fax detection... just a dedicated phone number which runs ReceiveFAX() when a call comes to it. I wouldn't think t38pt would be useful in my case, and I don't know if using faxdetect would either. |
| 15:37.49 | *** join/#asterisk libryder (~david@stg.maculon.com) |
| 15:38.39 | p3nguin | unless "pt" doesn't actually mean "pass through," which I interpret to mean all the way through. |
| 15:38.45 | irroot | p3nguin yes faxdetect is for cases when it needs to be "diverted" ie not handled in normal dialplan |
| 15:38.56 | libryder | when setting up a queue and creating its members, how do i specify an external number for that member? |
| 15:39.06 | p3nguin | I don't see another way to negotiate T.38. |
| 15:39.06 | zealiod | So I have bought a G.729a licence and installed it. How can I make sure that Asterix uses it on all outgoing and inching calls? |
| 15:39.30 | p3nguin | zealiod: disallow=all |
| 15:39.31 | p3nguin | zealiod: allow=g729 |
| 15:39.37 | p3nguin | That is all. |
| 15:40.14 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 15:40.58 | p3nguin | You need to set that in your peer entry that takes and sends calls. |
| 15:41.46 | p3nguin | Otherwise, asteriSK can negotiate another codec if it has a chance. |
| 15:43.48 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 15:44.05 | kaldemar | libryder: external number does not mean anything is asterisk. what do you mean? an extension in the dialplan? |
| 15:44.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:44.22 | libryder | or maybe a better question is do i have to specify sip/trunk-name/+1phonenumber for the interface? |
| 15:45.00 | p3nguin | You usually do not specify the + when using SIP. |
| 15:45.17 | p3nguin | If you were using gtalk, you probably would. |
| 15:45.47 | libryder | i'm using this to create queues: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue and for the members table i'm not entirely sure what to set for the "interface" field |
| 15:46.06 | libryder | here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf it says: |
| 15:46.06 | libryder | MEMBERINTERFACE is the interface name (eg. Agent/1234) |
| 15:46.44 | libryder | so it seems that i should insert the entire dialstring but want to be sure |
| 15:46.45 | ollii | or SIP/XY |
| 15:47.22 | kaldemar | libryder: yes, the whole string. |
| 15:47.31 | libryder | woot! thanks guys |
| 15:47.41 | p3nguin | I guess interface is just the member name. |
| 15:48.25 | libryder | this is really nice because i can create queues from any framework that has access to the database |
| 15:48.58 | zealiod | p3nguin: If I just use that codec will I get congestion? |
| 15:49.10 | p3nguin | zealiod: How would I know? |
| 15:49.20 | *** join/#asterisk screenn (~screenn@users-nat.more.com.ua) |
| 15:49.22 | [TK]D-Fender | libryder, Don't just dial direct. You'll be needing a call confirmation otherwise you can get hit by falling intoVM, etc. |
| 15:49.32 | zealiod | p3nguin: Is it pretty weighty on the bandwidth> |
| 15:49.38 | [TK]D-Fender | libryder, "core show application dial" < M() |
| 15:49.44 | p3nguin | zealiod: No, g.729 is very light. |
| 15:49.58 | [TK]D-Fender | (er) |
| 15:49.58 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-129-148.chyn.qwest.net) |
| 15:50.12 | [TK]D-Fender | And of course you can hit congestion. it's a question of at what point |
| 15:50.15 | p3nguin | zealiod: It may be considered the best codec for tight bandwidth limits by many people. |
| 15:50.36 | zealiod | p3nguin: What's the best quality codec - if bandwidth wasn't a problem? |
| 15:51.10 | p3nguin | zealiod: Probably ulaw/alaw, because most ITSPs can't do anything with a better codec. |
| 15:51.12 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 15:51.15 | coppice | 48k/s 24 bit linear |
| 15:51.50 | timeshell | I have seen a couple times in 1.8.6 that sometimes a call gets established between 2 devices by itself.... anyone else seen this? |
| 15:51.51 | libryder | [TK]D-Fender: what should i be using for the memberinterface option if not the whole dialstring then? |
| 15:52.12 | [TK]D-Fender | libryder, A Local channel |
| 15:52.15 | p3nguin | (1047.40) <p3nguin> I guess interface is just the member name. |
| 15:53.16 | p3nguin | Member name is a channel tech and a device name. In the case of a local channel, the channel tech is Local and the device name is the extension@context. |
| 15:53.49 | p3nguin | Agent/1234, Local/1234@phones, et cetera |
| 15:54.10 | libryder | [TK]D-Fender: is SIP/LEVEL3TRUNK/15484584565 a local channel? |
| 15:54.19 | ollii | that is a sip channel |
| 15:54.23 | [TK]D-Fender | libryder, chan_local. No |
| 15:54.24 | p3nguin | I seriously wish people would read. |
| 15:55.03 | zealiod | p3nguin: thanks. Any idea what causes "rtp.c:1144 ast_rtp_read: RTP Read too short" whilst making a call? |
| 15:55.46 | p3nguin | zealiod: No, I don't know the cause of it. Maybe someone else here will chime in, though. |
| 15:56.41 | libryder | p3nguin: i read your comment, but i don't understand what it means |
| 15:57.00 | p3nguin | A local channel uses chan_local.so |
| 15:57.07 | p3nguin | I gave you the usage and an example. |
| 15:57.34 | p3nguin | If "Local" is not used as the channel tech, it is not a Local channel. |
| 15:57.45 | p3nguin | Local/1234@phones <-- local channel |
| 15:58.00 | p3nguin | SIP/1234 <-- SIP channel |
| 15:58.26 | p3nguin | Agent/1234 <-- Agent channel |
| 15:59.06 | libryder | so, can you specify a local channel that dials through a sip trunk? |
| 15:59.21 | p3nguin | If you mean SIP peer, yes, you could do that. |
| 15:59.29 | libryder | yes, sip peer |
| 15:59.33 | p3nguin | via dial plan, of course. |
| 15:59.42 | p3nguin | Local channels call points in the dial plan. |
| 16:00.10 | [TK]D-Fender | libryder, At which point you use Dial to call the actual indended target with the optin I handed you |
| 16:01.18 | p3nguin | If you have a peer named "LEVEN3TRUNK" in sip.conf, and you have the necessary extensions configured in a context called "outbound" to call via that peer, you would use Local/15484584565@outbound as a queue member. |
| 16:01.56 | libryder | that's badass |
| 16:02.00 | p3nguin | The Local channel would then make a phone call to that phone number in the outbound context. |
| 16:02.08 | p3nguin | If you have a matching extension, it'll go. |
| 16:02.40 | libryder | does that mean i will have to have an extension specifically configured for each number? |
| 16:02.50 | p3nguin | no, use patterns. |
| 16:03.04 | p3nguin | _1NXXNXXXXXX for NANP numbers |
| 16:04.07 | libryder | thank you so much! i'm still at the very beginning of the learning curve for asterisk |
| 16:04.13 | p3nguin | ~book |
| 16:04.13 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:04.35 | libryder | i have the book :) |
| 16:04.52 | dijib | hey all. the other day i upgraded asterisk through YUM. it broke musiconhold and MixMonitor. in verbose it looks like it writes the file. but no file ever is writen. any help would be appreciated |
| 16:05.12 | p3nguin | ps -C asterisk u |
| 16:05.18 | p3nguin | Check who is running asterisk. |
| 16:05.27 | p3nguin | Then ls -ld /var/spool/asterisk/monitor |
| 16:05.34 | p3nguin | Check who owns the directory. |
| 16:06.44 | p3nguin | If the directory is now owned by root but you're still running as user asterisk, that can cause a problem. |
| 16:06.48 | dijib | i think you solved it already |
| 16:07.08 | dijib | so have a list of directories that need to be chown'ed? |
| 16:07.44 | p3nguin | chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
| 16:11.08 | *** part/#asterisk AlecTaylor (~alec.tayl@unaffiliated/alectaylor) |
| 16:11.29 | dijib | p3nguin, fixed it already |
| 16:11.50 | dijib | thank you |
| 16:11.59 | dijib | moh seems to still be busted |
| 16:12.14 | p3nguin | Let's take a look at musiconhold.conf. |
| 16:13.03 | p3nguin | Which reminds me, I still need to know how to reopen a closed issue on jira. |
| 16:14.13 | dijib | http://pastebin.com/qy2MsRr3 |
| 16:14.31 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 16:15.14 | *** join/#asterisk Diffen (~diffen@c-f477e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 16:15.16 | p3nguin | Do /var/lib/asterisk/mohmp3 and the files within have sufficient read privs? |
| 16:15.49 | dijib | 0644 |
| 16:16.01 | dijib | owned now by asterisk |
| 16:16.03 | p3nguin | That's a fail. |
| 16:16.08 | dijib | y? |
| 16:16.13 | p3nguin | Directories need to have +x |
| 16:16.27 | dijib | no thats an mp3 file |
| 16:16.32 | p3nguin | chmod +x /var/lib/asterisk/mohmp3 |
| 16:16.32 | dijib | dir perm? |
| 16:16.37 | p3nguin | /var/lib/asterisk/mohmp3 is not a file. |
| 16:16.41 | dijib | 0755 |
| 16:16.44 | p3nguin | Well, it's a file... not a regular file. |
| 16:16.57 | p3nguin | So the dir does have +x then? |
| 16:17.01 | dijib | oui |
| 16:17.05 | p3nguin | okay |
| 16:17.17 | p3nguin | Does moh show classes show you the two classes? |
| 16:17.20 | dijib | in cli moh starts and stops one after another |
| 16:17.46 | dijib | only one. |
| 16:17.51 | dijib | default |
| 16:18.31 | p3nguin | Does moh show files show you the files in the default class's dir? |
| 16:18.47 | dijib | yes |
| 16:18.55 | dijib | oh wait |
| 16:19.02 | dijib | yes |
| 16:19.05 | dijib | there in there |
| 16:19.12 | p3nguin | module unload res_musiconhold.so |
| 16:19.48 | dijib | k |
| 16:19.56 | dijib | and i did a load. and it fails the same way |
| 16:20.03 | p3nguin | It should have indicated it was unloading. |
| 16:20.08 | dijib | it did |
| 16:20.10 | p3nguin | "fails the same way" means what? |
| 16:20.38 | dijib | moh starts/stops immediatly when calling my MOH extension |
| 16:20.42 | *** part/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 16:20.49 | p3nguin | core set verbose 3 |
| 16:20.55 | p3nguin | Call the extension that runs moh. |
| 16:20.57 | dijib | verbose is always set to 4 |
| 16:21.01 | [TK]D-Fender | dijib, ls -la /var/lib/asterisk/mohmp3 |
| 16:21.04 | [TK]D-Fender | ^ |
| 16:21.25 | p3nguin | 4 is annoying. |
| 16:21.38 | p3nguin | 3 shows you the same stuff, minus all the dns manager crap. |
| 16:21.55 | dijib | ive got debug and verbose set in asterisk.conf |
| 16:22.19 | dijib | [TK]D-Fender, permissions seem ok. 0644 on the mp3's |
| 16:22.37 | [TK]D-Fender | dijib, Go prove you have format_mp3.so loaded |
| 16:22.53 | [TK]D-Fender | dijib, Otherwise you have MP3 support in the first place |
| 16:23.32 | dijib | MP3Player: Play an MP3 file or M3U playlist file or stream. |
| 16:23.37 | dijib | in core show applications |
| 16:24.08 | *** join/#asterisk [psy] (psy@lounge.datux.nl) |
| 16:24.45 | [psy] | is it possible to let asterisk send a fax to itself using txfax and rxfax? (just for testing purposes). i tried with a .call file and localchannels but it doesnt seem to work |
| 16:25.09 | [psy] | and for that matter: is it possible to let asterisk connect to itself as sip trunk? (e.g. a loopback via sip) |
| 16:25.34 | dijib | thats another thing thats been broken res_fax_digium.so is not loading |
| 16:25.41 | dijib | but dont worry about that right now |
| 16:26.14 | coppice | use res_fax_spandsp.so instead |
| 16:27.37 | p3nguin | module show like mp3 |
| 16:27.38 | [TK]D-Fender | <dijib> MP3Player: Play an MP3 file or M3U playlist file or stream. <- that is not it |
| 16:27.48 | [TK]D-Fender | dijib, Otherwise you have MP3 support in the first place |
| 16:27.51 | [TK]D-Fender | specific file |
| 16:29.03 | dijib | [TK]D-Fender, you found it. that module is not loaded |
| 16:29.05 | [TK]D-Fender | [psy], If it didn't work then you've done something wrong. And yes * can talk SIP to itself |
| 16:29.21 | [TK]D-Fender | dijib, Fix that or you don't have MP3 support in * |
| 16:29.39 | [psy] | [TK]D-Fender thx:) |
| 16:29.40 | JonathanRose | mp3 as a file format is an addon. You need to pick it up with the contrib script before you can use it and then enable it with menuselect. |
| 16:29.52 | p3nguin | (1127.37) <p3nguin> module show like mp3 |
| 16:29.55 | JonathanRose | And I'm sorry if that's all known knowledge to you at this point. |
| 16:30.26 | JonathanRose | But once you do have it enabled and compiled, you should be able to use mp3's just like any other sound file. |
| 16:30.33 | dijib | where is thats format_mp3.so ? /usr/lib/asterisk/modules ? |
| 16:30.46 | p3nguin | (1129.52) <p3nguin> (1127.37) <p3nguin> module show like mp3 |
| 16:31.07 | [TK]D-Fender | dijib, Yes |
| 16:31.17 | p3nguin | Do you have format_mp3.so in the output? |
| 16:31.17 | *** join/#asterisk bis0n (~56416@78.238.86.105) |
| 16:31.20 | bis0n | hello |
| 16:31.28 | JonathanRose | Hi there |
| 16:32.05 | p3nguin | And I still need to know how to reopen a closed issue on jira. |
| 16:32.13 | bis0n | my little last problem, everything work with my asterisk (after many hours ^^) but i have this in message.log, http://pastebin.com/5vt4sBuz |
| 16:32.51 | dijib | p3nguin, no format_mp3.so is not there. |
| 16:33.10 | p3nguin | Then you probably didn't build it. |
| 16:33.48 | p3nguin | Actually, you didn't build any of it, now that I think about it. |
| 16:33.59 | JonathanRose | dijib: From the asterisk source directory, ./contrib/scripts/get_mp3_source.sh |
| 16:34.10 | JonathanRose | dijib: Then make menuselect and enable it in addons. |
| 16:34.11 | p3nguin | So the question is, how is a person supposed to get mp3 support when using packages from the asterisk repo? |
| 16:34.25 | JonathanRose | I don't know if there is a way. |
| 16:34.29 | dijib | yeh thats what im wondering |
| 16:34.33 | [TK]D-Fender | yum install asterisk* |
| 16:34.42 | [TK]D-Fender | they've split the packages up. |
| 16:34.47 | p3nguin | yum search asterisk ... find the mp3 package. |
| 16:34.53 | dijib | k |
| 16:35.18 | p3nguin | If there isn't one, find your format_mp3.so that is left over from your last source build, and copy it to /usr/lib/asterisk/modules with the other modules. |
| 16:35.31 | JonathanRose | asterisk-mp3 is the name of the package, at least with aptitude. |
| 16:35.47 | p3nguin | yum -y install asterisk-mp3 |
| 16:36.05 | bis0n | how I can disable messages logging / debug of dahdi? |
| 16:37.13 | dijib | asterisk18-addons ??? |
| 16:37.17 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:37.35 | p3nguin | Does it say anything about mp3? |
| 16:37.59 | dijib | nope |
| 16:38.24 | p3nguin | Install it and see what it gives you. |
| 16:39.09 | p3nguin | Once it is installed, rpm -ql asterisk18-addons |
| 16:42.03 | Janos | hey there guys, having a problem with a new E1 line and i have no idea what's wrong with it, here is the debug if someone can take a look at it http://pastebin.com/7TUnRNC0, thanks |
| 16:42.24 | dijib | where in the old /usr/src/asterisk directory would the format_mp3.so be? |
| 16:43.08 | p3nguin | Never heard of "find" before? |
| 16:43.17 | p3nguin | or "locate" for that matter. |
| 16:43.32 | dijib | nope |
| 16:43.39 | p3nguin | find /usr/src/asterisk -name format_mp3.so |
| 16:44.19 | dijib | found nothing |
| 16:44.41 | dijib | there is the get_mp3_source.sh script |
| 16:44.59 | p3nguin | Did you install asterisk18-addons yet? |
| 16:45.12 | p3nguin | Did you check it for format_mp3.so like I suggested? |
| 16:45.43 | p3nguin | If it's not there and you don't want the other stuff it provides, remove the package. It's literally as easy as that. |
| 16:45.55 | p3nguin | You've got to stop making things so complicated. |
| 16:46.31 | dijib | i wouldnt install due to missing dependencies |
| 16:46.58 | p3nguin | What dependencies? |
| 16:50.06 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-wlzorgrxwftfqytv) |
| 16:50.23 | shido6 | hi all |
| 16:50.46 | p3nguin | I haven't seen that nick for a while. |
| 16:52.42 | voipmonk | I shut that biz down a while ago |
| 16:53.14 | voipmonk | once Im able to pause time I and clone ppl I can start it up as version 2 |
| 16:53.34 | voipmonk | gets coffee |
| 16:53.38 | p3nguin | There were some people here looking for you after you took their money and did not provide services. |
| 16:55.00 | voipmonk | thanks for the update - do you know their nicks? |
| 16:55.10 | p3nguin | I'd have to go searching logs. |
| 17:00.41 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:03.35 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 17:06.24 | *** join/#asterisk Ashutto (~Ashetic@2001:b00:7c00:600:6ef0:49ff:fe7d:3d35) |
| 17:06.28 | Ashutto | Hello |
| 17:09.35 | *** join/#asterisk wonderworld (~ww@port-92-201-11-128.dynamic.qsc.de) |
| 17:10.17 | *** join/#asterisk wonderworld (~ww@port-92-201-11-128.dynamic.qsc.de) |
| 17:12.38 | voipmonk | Hello Ashutto |
| 17:20.24 | *** join/#asterisk n0h0pe (~marrandy@237-41.187-72.tampabay.res.rr.com) |
| 17:21.15 | n0h0pe | [TK]D-Fender: struggling...pastbin, finally... http://pastebin.com/VENXCRMd |
| 17:24.06 | tuxxie | I use QueueLog(queue1,${UNIQUEID},NONE,INFO,DID|${CALLERID(dnid)}) to order to track DNIS. However, I am unable to get the DNIS on call from my google call metrics accounts. Does anyone know how i can capture dnis from googlemetrics? |
| 17:26.38 | *** join/#asterisk garnarsson (~garnarsso@96.49.96.81) |
| 17:27.11 | Ashutto | Question: i have only IPv6 connectivity. Is it possible to usa asterisk? am i limited by ipv6 routed voip ? |
| 17:27.22 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 17:27.33 | garnarsson | Hi, I'm trying to use a Cisco 7970 on asterisk and need some help with firmware/config setup |
| 17:28.10 | garnarsson | I've folowed http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP and a few others with the same result |
| 17:28.19 | garnarsson | right now it's stuck at register |
| 17:28.46 | garnarsson | I have a sniff running via wireshark and all I see there is ping requests |
| 17:29.01 | SuperNull | Hey Guys im having REALLY odd behavior with ast 1.8.6.0 .. keeps sending invites without destination #s .. |
| 17:29.08 | garnarsson | I'm runnin 8-4-4s load |
| 17:30.56 | libryder | p3nguin: i got it :D https://gist.github.com/1268035 |
| 17:31.12 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 17:31.23 | libryder | ah i should the sql file to that gist |
| 17:32.12 | garnarsson | I also see that it's unable to find file: /td-sip.jar, united_States/g3-tones.xml and dialplan.xml |
| 17:33.33 | *** part/#asterisk [psy] (psy@lounge.datux.nl) |
| 17:33.42 | Madkiss | does asterisk 1.8 still include app_meetnme= |
| 17:33.45 | Madkiss | app_meetme i mean |
| 17:33.49 | p3nguin | Yes. |
| 17:34.06 | [TK]D-Fender | SuperNull, pastebin it |
| 17:35.05 | Madkiss | [Oct 6 19:34:43] WARNING[26517]: loader.c:393 load_dynamic_module: Error loading module 'app_meetme.so': /usr/lib/asterisk/modules/app_meetme.so: cannot open shared object file: No such file or directory |
| 17:35.16 | SuperNull | okay so .. its not the ATA .. |
| 17:35.30 | SuperNull | TK it seems to be asterisk too .. i do a full restart and functions being.. |
| 17:35.34 | SuperNull | i will get a sip dump hold. |
| 17:35.37 | p3nguin | madkiss: Did you install app_meetme? |
| 17:35.50 | Madkiss | p3nguin: hu? these are precompiled packages |
| 17:36.15 | p3nguin | Who build the package? |
| 17:36.42 | Madkiss | me. I just found the module. the debian asterisk maintainers put it into the "asterisk-dahdi" package. |
| 17:36.54 | SuperNull | [TK]D-Fender .. would insecure mode have anything to do with this.. (im getting unknown errors due to old database matches) |
| 17:37.25 | p3nguin | You built the package, but you couldn't tell me if you installed app_meetme? Something doesn't add up. |
| 17:38.31 | Madkiss | p3nguin: I *built* the package as in I downloaded the package sources for the testing distribution and built them on stable (this is debian, as I already mentioned) |
| 17:39.01 | p3nguin | So then you should have been able to tell me if you installed app_meetme or not instead of blaming it on someone else. |
| 17:39.23 | Madkiss | p3nguin: had a bad day, eh? |
| 17:39.27 | JonathanRose | app_meetme should be enabled by default I think. |
| 17:39.46 | Madkiss | It is, it just gets sorted into its own package now, which I didn't have installed |
| 17:40.10 | Madkiss | and apart from that, I don't think I "blamed it on someone else" |
| 17:40.23 | p3nguin | (1235.37) <p3nguin> madkiss: Did you install app_meetme? |
| 17:40.23 | p3nguin | (1235.50) <Madkiss> p3nguin: hu? these are precompiled packages |
| 17:40.31 | Madkiss | which is fully adequate. |
| 17:40.35 | garnarsson | cisco 7970 config anyone? |
| 17:40.38 | p3nguin | I don't know what else you could have possibly been trying to say there. |
| 17:40.48 | Madkiss | p3nguin: broaden your horizon, then |
| 17:40.51 | SunTsu | .o0( good to see Madkiss didn't change a bit after all these years *g* ) |
| 17:41.03 | [TK]D-Fender | Madkiss, Did you install DADHi? |
| 17:41.16 | p3nguin | If YOU built the package, it's not a "precompiled package." |
| 17:41.16 | Madkiss | [TK]D-Fender: now I did, and now app_meetme is there, so everything's fine :) |
| 17:41.31 | Madkiss | SunTsu: If only you knew ... :) |
| 17:41.33 | JonathanRose | Yeah, meetme has dahdi as a dependency, so you'll need to get dahdi in order to use it. |
| 17:41.46 | SuperNull | [TK]D-Fender ... http://pastebin.com/BA1Wb8Kx is sip dump 100.136 = Adtran TA-912 ATA (12 port analog telephone adapter) |
| 17:42.00 | JonathanRose | Once you do you'll probably need to rebuild Asterisk. |
| 17:42.17 | [TK]D-Fender | SuperNull, * SIP debug, not raw |
| 17:42.30 | Madkiss | the problem really is that app_meetme.so was part of the asterisk package in asterisk 1.6. And in asterisk 1.8, apparently the package maintainers have decided to split it into its own package, namely asterisk-dahdi. and I didn't have that one installed, because until now, I didn't have to. |
| 17:42.44 | Madkiss | good to see everything's fine, though |
| 17:43.25 | SuperNull | [TK] how does one enable that in AST 1.8.6 :-/ (im used to old 1.4 and even then i used tcpdump) |
| 17:43.42 | [TK]D-Fender | SuperNull, sip set debug on |
| 17:43.53 | p3nguin | If you knew how to enable it in 1.4, you surely know how to enable it in 1.8. |
| 17:44.05 | SuperNull | Ty but i will have to wait for this to re-create it seems ..timed .. based on something unknown |
| 17:44.51 | p3nguin | What is sending the invites? |
| 17:44.52 | Madkiss | p3nguin: it's enabled by default in the 1.6 packages provided by debian, just as it is enabled by default in the 1.8 packages provided by debian. |
| 17:44.55 | p3nguin | Asterisk or the ATA? |
| 17:45.00 | Madkiss | the interesting question just is what package it is in |
| 17:45.19 | Madkiss | I didn't know that, but I found out :) |
| 17:46.00 | Madkiss | amazing to see that the upgrade from 1.6 to 1.8 didn't actually need any manual intervention |
| 17:46.09 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:47.59 | SuperNull | Asterisk is sending the invite to the ATA the pastebin shows the raw messages.. |
| 17:48.50 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 17:49.16 | Janos | can anyone give me a hand with an E1, it's driving me crazy :P |
| 17:49.51 | WIMPy | Janos: You need to be a lot more specific, if you want help. |
| 17:50.26 | SuperNull | im going to grab the sip user line from mySQL .. maybe something is not right im comparison to 1.2/1.4 |
| 17:50.56 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 17:51.08 | Janos | ohh yeah no problem, basically it's a new line i'm trying to use, it's on RED alarm, asterisk keeps complaining it can't find the D-Channel |
| 17:52.03 | Janos | here is the pri debug http://pastebin.com/7TUnRNC0, maybe someone can make some sense out of that :P |
| 17:52.11 | SuperNull | i have a feeling fromuser has something to do with this.. |
| 17:52.49 | Janos | i keep getting, [Oct 6 11:52:23] WARNING[31846]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 17:52.51 | WIMPy | If you have a red alarm, there's no need to try and find a D-cahnnel or du any debugging. You're simply not connected. |
| 17:53.26 | SuperNull | http://pastebin.com/eGVtHv9E (secret has been XX-ed Out) .. |
| 17:53.32 | WIMPy | So either the line coding in dahdi/system.conf is wrong or the line is not up or your cabling is faulty. |
| 17:56.50 | *** join/#asterisk zealiod (~zealiod@host-78-147-39-97.as13285.net) |
| 17:57.18 | Janos | WIMPy, thanks for the info, i'm using ccs/hdb3, i guess i will contact my provider again and confirm the parameters, or start playing around with the options until i find one that works :P |
| 17:57.55 | [TK]D-Fender | Janos, CRC4/AMI is popular... |
| 17:58.43 | coppice | no its not. CRC4/HDB3 is the popular one |
| 17:58.49 | WIMPy | Janos: ccs,hdb3 sounds right. Maybe you should add crc4, but I dont thin that would cause a red alarm. |
| 17:59.08 | WIMPy | Check cabling and confirm that the line should be up. |
| 17:59.49 | Janos | i had the line installed yesterday and the provider tested it in my face :P, so cabling should be good |
| 18:00.25 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 18:00.25 | WIMPy | Try to replace the cable from the NT to the card. |
| 18:00.48 | luckman212 | any Polycom peeps around? I have an EFK head scratcher |
| 18:01.00 | Janos | WIMPy, yup that sounds like an option, will try that |
| 18:01.06 | Naikrovek | efk makes all of our heads itch |
| 18:01.07 | WIMPy | Janos: Is the card jumpered correctly for E1? |
| 18:01.20 | Janos | WIMPy, yup |
| 18:01.21 | Naikrovek | I tried some EFK stuff a while back and gave up |
| 18:02.06 | *** join/#asterisk mpe (~mpe@31.25.23.177) |
| 18:02.25 | luckman212 | yeah it really sucks your soul right out |
| 18:02.50 | zealiod | Hey - I have a two way SIP connection testing our NAT'd network. Phone A can call the speaking clock and the speaker works, but when calling Phone B - there's no audio in the speaker. Any ideas? |
| 18:02.54 | *** join/#asterisk jkroon (~jkroon@dsl-242-3-48.telkomadsl.co.za) |
| 18:03.01 | zealiod | Phone A can hear Phone B's audio |
| 18:03.08 | luckman212 | I imagine the Polycom engineers sitting around a cauldron chanting Mwhahahahahahaa and sacrificing goats when they designed the syntax |
| 18:03.21 | Naikrovek | heh |
| 18:03.22 | p3nguin | zealiod: You probably don't have the nat settings right on asterisk. |
| 18:03.31 | Naikrovek | they do seem a bit sadistic at times, yes |
| 18:03.37 | zealiod | p3nguin: can you please advise |
| 18:03.44 | p3nguin | ~sipnat |
| 18:03.45 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 18:05.24 | luckman212 | one thing about NAT that is always shrouded in mystery... is if your main sip.conf file should have nat=yes in it if your Ast* box is behind a nat. I have found that leaving it at "no" seems to give better results |
| 18:05.55 | luckman212 | the docs say "no" = USE RFC3581 |
| 18:07.05 | Janos | WIMPy, as a side note if i enable crc4 i get YELLO and RED alarms, i have other E1 and CRC4 is not used on any, so i guess that's not the issue |
| 18:08.23 | coppice | nto using CRC4 when you use CCS is brain dead, but not too uncommon |
| 18:08.23 | *** join/#asterisk singler (~singler@91.221.145.233) |
| 18:11.14 | zealiod | p3nguin: Both my SIP phones are on the same NAT - my * server is on an external server with a public ip |
| 18:11.35 | dijib | is this the GM channel? i need to change my oil |
| 18:11.59 | p3nguin | zealiod: So then the nat setting in general should be set to no, and you should have nat=yes in the phone entries. |
| 18:12.02 | dijib | http://pastebin.com/gt8Cikbq |
| 18:12.32 | dijib | if i try & install asterisk18-addons it wants mysql installed |
| 18:12.41 | dijib | i dont want to run sql |
| 18:13.00 | Janos | there are a couple of things that i'm not so sure in the info i got, on signaling it says i can use pri etsi or ss7, is it even possible to just use either ? |
| 18:13.15 | p3nguin | Why you'd be sitting in / is beyond me, but it's a dangerous habit to be in. |
| 18:13.49 | p3nguin | Force it without mysql, then. |
| 18:14.06 | p3nguin | Or install mysql and asterisk18-addons, and don't use mysql. |
| 18:14.45 | zealiod | p3nguin: yeah, thats the case |
| 18:14.46 | p3nguin | Or just get the asterisk18-addons package and see if it contains format_mp3.so |
| 18:17.42 | n0h0pe | <PROTECTED> |
| 18:20.07 | garnarsson | can someone help me out with the config file for a cisco 7970 phone? The phone doesn't seem to register to the server. |
| 18:21.03 | garnarsson | It's calling cdp/vtp/dtp/pagp/udld and then lldp_multicast (looking at wireshark sniff) |
| 18:21.11 | WIMPy | Janos: DSS1 and SS7 are quite different things, but a red alarm is at a lower level. |
| 18:23.16 | Janos | WIMPy, ok going for the cable then |
| 18:24.21 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 18:24.31 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:25.51 | dijib | p3nguin, found it, it was in asterisk18-addon-core |
| 18:26.55 | dijib | [TK]D-Fender, p3nguin; MOH fixed. |
| 18:27.01 | dijib | not mpg123 though |
| 18:27.16 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:27.24 | p3nguin | How are you playing mp3 moh? |
| 18:29.18 | dijib | yes |
| 18:29.25 | p3nguin | Huh? |
| 18:29.36 | dijib | but i cannot still play streams using mpg1234 |
| 18:29.49 | dijib | im playing them using format_mp3.so after loading it |
| 18:29.57 | p3nguin | I'll type it slower this time so you can understand my question... |
| 18:30.07 | p3nguin | How are you playing mp3 moh? |
| 18:30.27 | navaismo | :D |
| 18:30.50 | p3nguin | format_mp3 does not play files. |
| 18:30.57 | p3nguin | mpg123 plays files. |
| 18:31.06 | dijib | mp3player |
| 18:31.17 | dijib | ? |
| 18:31.28 | dijib | where is that configured? |
| 18:31.37 | p3nguin | As far as I can tell, mp3player does not play moh. |
| 18:31.43 | dijib | no asterisk has mp3player built into it |
| 18:31.52 | dijib | it must. |
| 18:31.52 | p3nguin | MP3Player() is just an app that plays mp3 files. |
| 18:32.02 | dijib | then what plays the mp3's? |
| 18:32.07 | p3nguin | (1330.56) <p3nguin> mpg123 plays files. |
| 18:32.11 | dijib | and is that going to stdin ? |
| 18:32.29 | dijib | so if i dont have mpg123 installed moh will not play mp3's? |
| 18:32.31 | dijib | really? |
| 18:32.52 | p3nguin | lsof -u asterisk|grep -i mp3 |
| 18:33.12 | dijib | mpg123 is installed & seems to stream if you run it in bash |
| 18:33.30 | dijib | whats list of? |
| 18:33.34 | p3nguin | Why are you trying to trick me? |
| 18:33.39 | p3nguin | (1332.29) <dijib> so if i dont have mpg123 installed moh will not play mp3's? |
| 18:33.40 | p3nguin | (1333.11) <dijib> mpg123 is installed & seems to stream if you run it in bash |
| 18:34.07 | p3nguin | I'm not up for games today. |
| 18:34.12 | dijib | i was trying to understand your logic. not whitchcraft |
| 18:34.22 | dijib | good lord man. |
| 18:34.25 | [TK]D-Fender | Asterisk is playing the files in res_musiconhold |
| 18:34.26 | p3nguin | You HAVE mpg123. |
| 18:34.30 | [TK]D-Fender | not MPG123 |
| 18:34.43 | [TK]D-Fender | mode=files means no mpg123 is being used |
| 18:34.47 | p3nguin | If you use mode=mp3, it'll use mpg123. |
| 18:34.52 | [TK]D-Fender | p3nguin, He isn't |
| 18:35.12 | p3nguin | But what does that have to do with MP3Player()? |
| 18:35.57 | dijib | i just figured it was handling the mp3 playing |
| 18:36.01 | libryder | i added the db structure and some sample data for dynamic queue creation and support in adhearsion: https://gist.github.com/1268035#file_queues.sql |
| 18:36.03 | dijib | dont have to eat your hat. |
| 18:36.13 | dijib | thanks [TK]D-Fender |
| 18:36.44 | [TK]D-Fender | <p3nguin> But what does that have to do with MP3Player()? <- nothing |
| 18:37.25 | dijib | wait i've changed mode=mp3 in moh.conf and its still working,. |
| 18:38.07 | p3nguin | If you ran moh reload, now mp3s will be played with mpg123. |
| 18:38.26 | p3nguin | which can be seen with lsof -u asterisk|grep -i mp3 |
| 18:39.06 | dijib | i think i need lsof installed as its a command not found |
| 18:39.36 | *** join/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 18:39.43 | p3nguin | ps -C mpg123 u |
| 18:40.03 | Rroet | evening all. |
| 18:40.35 | Rroet | p3nguin leifmadsen [TK]D-Fender: thanks for all your helps sofar. Everything is running smoothly at the moment. |
| 18:43.16 | dijib | p3nguin, its running as asterisk. and is playing the mp3's |
| 18:43.17 | garnarsson | anyone here have a cisco 7970 running? |
| 18:44.24 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 18:45.30 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 18:50.44 | [TK]D-Fender | Rroet, Good to hear. |
| 18:53.53 | n0h0pe | Got to go - work to do. |
| 19:00.19 | *** join/#asterisk jaybee_ (~jortbloem@114.31.212.161) |
| 19:01.41 | jaybee_ | Hi all - is there any way to do string replacement in an asterisk dialplan? I want to replace slashes '/' with something else (e.g. -) in ${CHANNEL} so it's valid as a filename |
| 19:04.52 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:05.32 | leifmadsen | jaybee_: core show function REPLACE |
| 19:06.29 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 19:06.36 | jaybee_ | was unaware about core show function, and is going to go read all of them :) |
| 19:06.44 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 19:08.09 | jaybee_ | finds uriencode, which is much better for my purpose - thanks leifmadsen |
| 19:14.26 | *** join/#asterisk celord (~celord@201.191.132.177) |
| 19:14.42 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 19:17.12 | Freeaqingme | ~thebook |
| 19:17.12 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 19:18.09 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:18.19 | *** join/#asterisk hdiogenes (~hdiogenes@201.76.154.168.intranet.digi.com.br) |
| 19:21.45 | dijib | how would i debug mpg123 playing streams? |
| 19:24.07 | p3nguin | What's the problem? |
| 19:25.20 | dijib | it just refuses to play streams |
| 19:25.26 | dijib | brb folding laundry with the wife |
| 19:26.00 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-akcgkgehbyigdcud) |
| 19:29.38 | hdiogenes | dijib: error message? |
| 19:31.20 | p3nguin | Show me the command you're using. |
| 19:32.29 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:36.31 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:36.59 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053199.ics.uci.edu) |
| 19:39.30 | *** join/#asterisk celord (~celord@200.91.75.98) |
| 19:41.32 | *** part/#asterisk rafiyr (~rafi@hex.seas.UPENN.EDU) |
| 19:41.57 | dijib | back: mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 -m -s http://2953.live.streamtheworld.com:80/CIBUFM_SC |
| 19:42.00 | dijib | something like that |
| 19:43.02 | p3nguin | Here's what I use: /usr/bin/mpg123 -q -b 4096 --preload 0.2 -r 8000 -f 4096 -m -s http://streamurl:8008 |
| 19:43.26 | p3nguin | in application= in musiconhold.conf |
| 19:44.43 | p3nguin | http://2953.live.streamtheworld.com:80/CIBUFM_SC does seem to be a valid stream, so that shouldn't be the problem. I just tested it. |
| 19:44.43 | dijib | i put that exactly in with my stream and it fails still. |
| 19:44.59 | p3nguin | Describe the failure. |
| 19:45.16 | dijib | Started music on hold, class '94.5', on channel 'SIP/600-00000002' |
| 19:45.19 | dijib | with no audio |
| 19:45.51 | p3nguin | Run module unload res_musiconhold.so followed by module load res_musiconhold.so, and then call it again. |
| 19:48.28 | dijib | that seems to be my problem. |
| 19:48.29 | *** part/#asterisk hdiogenes (~hdiogenes@201.76.154.168.intranet.digi.com.br) |
| 19:48.35 | dijib | no res_musiconhold.so ???/ |
| 19:48.36 | *** join/#asterisk hdiogenes (~hdiogenes@201.76.154.168.intranet.digi.com.br) |
| 19:48.37 | p3nguin | what is? |
| 19:48.44 | p3nguin | You forgot to load it? |
| 19:49.11 | dijib | Command 'module unload res_musiconhold.so' failed. |
| 19:49.25 | p3nguin | Try to load it, then. |
| 19:49.26 | dijib | but module show like hold shows res_musiconhold.so |
| 19:49.35 | dijib | load fails also |
| 19:49.43 | p3nguin | Oh, maybe you're on the phone using it. |
| 19:49.52 | p3nguin | You won't be able to unload it while you're using moh. |
| 19:50.01 | dijib | im bounce the server |
| 19:50.05 | p3nguin | Why? |
| 19:50.09 | p3nguin | core show channels |
| 19:50.12 | dijib | might aswell |
| 19:50.24 | p3nguin | You should really consider learning how to admin a system. It's not Windows. |
| 19:50.35 | dijib | i know :/ |
| 19:50.55 | dijib | ive dissconnected channels before in tests |
| 19:50.58 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:51.06 | p3nguin | See if there are any channels in use now. |
| 19:51.26 | p3nguin | If there are, that could explain why moh won't unload. |
| 19:52.20 | dijib | ok anyways, im back in cli after the bounce |
| 19:52.29 | dijib | res_musiconhold.so is loaded |
| 19:52.36 | dijib | shall i try the moh stream |
| 19:52.38 | dijib | ? |
| 19:52.40 | p3nguin | Go for it. |
| 19:54.54 | wonderworld | is asterisk 10 rather stable or is it unusable in production atm? |
| 19:55.25 | p3nguin | There hasn't even been a single release yet, and you're already wanting to put it into production? |
| 19:56.05 | dijib | asterisk died with code 1. ??? |
| 19:56.17 | p3nguin | Fancy stuff. |
| 19:56.21 | dijib | i just did yum install asteris18-voicemail as it apparently wasnt installed either |
| 19:56.52 | dijib | nevermind. |
| 19:57.00 | wonderworld | yes, i'd love to use the new call bridge but i am unsure if it will blow up ;) |
| 19:57.15 | p3nguin | I'd assume it will blow up. |
| 19:57.17 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 19:57.21 | wonderworld | k, so i won't |
| 19:57.45 | p3nguin | Not only would I never put a beta into production, I wouldn't even put the first few released into production. |
| 19:57.54 | p3nguin | releases |
| 19:58.15 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:58.34 | wonderworld | well, the "production" i refered to isn't a real big thing. they can handle some downtime |
| 19:58.48 | p3nguin | Then you might be a great beta tester. |
| 19:58.56 | wonderworld | yes i thought so |
| 19:58.58 | wonderworld | BUT |
| 19:59.06 | wonderworld | i don't want it to blow up either :) |
| 19:59.27 | p3nguin | Beta testers have to accept blow-ups. |
| 19:59.52 | wonderworld | yes sure. i was just wondering if there are known major bugs or not |
| 19:59.57 | *** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net) |
| 20:00.59 | [TK]D-Fender | wonderworld, Join the mailing lists |
| 20:02.00 | dijib | anybody ever had a DID transfered to them? |
| 20:03.39 | eduzimrs | anyone here has REDFONE configured? |
| 20:03.57 | eduzimrs | i got a problem installing libfb |
| 20:05.07 | WIMPy | framebuffer??? |
| 20:05.27 | eduzimrs | WIMPy me? |
| 20:05.41 | WIMPy | yes |
| 20:05.41 | p3nguin | What do you mean by having a DID transferred? |
| 20:05.47 | eduzimrs | no |
| 20:05.52 | p3nguin | Like you own it now and then I'm going to own it in a few days? |
| 20:06.14 | eduzimrs | im running ./configure and its asking LIBNET to continue |
| 20:06.23 | WIMPy | likes unique names |
| 20:06.27 | eduzimrs | so a didnt found that |
| 20:08.06 | *** part/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 20:08.37 | dijib | no my cell pone provider owns it right now, and i want to transfer it to my voip.ms |
| 20:08.48 | dijib | im in chat with the guy from voip.ms he is checking |
| 20:09.33 | p3nguin | You can check it yourself. |
| 20:09.45 | p3nguin | Go to DID portability in your portal. |
| 20:10.02 | p3nguin | Click on the appropriate link for your region/number. |
| 20:10.22 | p3nguin | Well, the link is Check Availability... next to your region/number listing. |
| 20:10.30 | p3nguin | Put in your area code and exchange. |
| 20:10.41 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:10.43 | p3nguin | It'll tell you if the number is portable or not. |
| 20:11.23 | p3nguin | It's $25 per number. |
| 20:12.35 | *** join/#asterisk zealiod (~zealiod@host-78-147-39-97.as13285.net) |
| 20:12.43 | p3nguin | Remember, when you port the number away from your cell phone, you'll have to get a new cell number. |
| 20:12.54 | p3nguin | or give up the cell phone completely. |
| 20:13.49 | zealiod | Still having this issue. Phone A can not hear phone B - but phone B can hear phone A and themselves. NAT is turned on the SIP user and nat is off on the general sip.conf - any ideas? |
| 20:16.10 | dijib | i was going to get a new number |
| 20:16.29 | p3nguin | You can check the portability of your number without waiting on someone else to do it for you. |
| 20:16.33 | dijib | but that ratecenter is not available for porting :( |
| 20:16.52 | dijib | i can only find the lists of ratecenters |
| 20:17.17 | p3nguin | Oh, they do it differently for US numbers. |
| 20:17.31 | p3nguin | https://www.voip.ms/m/lnp_us_avail.php |
| 20:17.37 | p3nguin | Please enter Area code and first 3 digits of the DID number to verify portability. |
| 20:17.39 | p3nguin | Area code: NXX: |
| 20:17.40 | dijib | damn |
| 20:17.45 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 20:17.54 | p3nguin | easy for us! |
| 20:18.05 | dijib | then i cant do it? |
| 20:18.33 | dijib | i need a local DID i guess |
| 20:18.44 | p3nguin | Probably not, if the ratecenter isn't listed. |
| 20:19.19 | p3nguin | What ratecenter is your number in? |
| 20:21.11 | SeRi | rofl @ p3nguin |
| 20:21.33 | SeRi | p3nguin, for some reasoner I read ratcenter |
| 20:21.34 | p3nguin | ? |
| 20:21.38 | p3nguin | oh |
| 20:21.40 | SeRi | :P |
| 20:21.42 | SeRi | lol |
| 20:21.52 | p3nguin | Did you send me a SIM? |
| 20:21.59 | SeRi | yes sr! |
| 20:22.09 | SeRi | today it went out. |
| 20:22.19 | SeRi | man traffic was insane |
| 20:22.27 | SeRi | ugh... city |
| 20:22.29 | SeRi | lol |
| 20:24.22 | dijib | owen sound, ON should be |
| 20:24.31 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:24.34 | SeRi | been working with a bug in 1.8.4.4 |
| 20:26.38 | p3nguin | Can you give me the area code and exchange for your number? I don't necessarily need the last four digits. |
| 20:29.26 | *** join/#asterisk screenn (~screenn@109.86.141.52) |
| 20:29.33 | Katty | i'm a CCNT, bitches! |
| 20:29.52 | Katty | woo! |
| 20:29.56 | Katty | goes back to lurking |
| 20:30.03 | zealiod | Should I use a STUN server to solve my no audio NAT issues on my SIP phones? |
| 20:41.20 | shido6 | wanna come work with me? :) |
| 20:47.31 | p3nguin | Are you threatening me? |
| 20:50.14 | SuperNull | er did TK leave? i dont have autocomplete in this client.. |
| 20:50.38 | SunTsu | 22:19:36 -!- [TK]D-Fender [~aoulton@216.191.106.162] has quit [Quit: Ex-Chat] |
| 20:50.40 | dijib | i took the ccnt track before.. never got the cert though |
| 20:50.40 | p3nguin | You have /names and /whois? |
| 20:51.00 | dijib | who is threatening who? |
| 20:51.01 | SunTsu | which according to my clock was half an hour ago |
| 20:51.13 | p3nguin | ~seen [TK]D-Fender |
| 20:51.22 | infobot | [tk]d-fender <~aoulton@216.191.106.162> was last seen on IRC in channel #asterisk, 50m 23s ago, saying: 'wonderworld, Join the mailing lists'. |
| 20:51.27 | dijib | so i just got off the phone with the cell phone provider and they have no freaking clue |
| 20:51.38 | p3nguin | Typical of a service provider. |
| 20:51.50 | p3nguin | What is the area code and exchange of your cell number? |
| 20:51.53 | p3nguin | I don't need the last four. |
| 20:54.24 | dijib | 519-379 |
| 20:55.13 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:55.28 | p3nguin | It does seem to be Owen Sound, ON, like you thought. |
| 20:56.22 | dijib | crappy |
| 20:56.44 | dijib | there is no towns in this region with transferable DID's |
| 20:56.51 | dijib | closest place is like 100km away |
| 20:57.10 | p3nguin | The term you're looking for is "portable." |
| 20:57.33 | p3nguin | Not "transferable DIDs," but portable. |
| 20:59.52 | SuperNull | anyone know if formuser/fromdomain have to be some form of data for 1.8.6 to function ? (ast realtime) |
| 20:59.57 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:00.11 | dijib | ok portable. |
| 21:00.27 | p3nguin | fromuser/fromdomain are OPTIONAL settings for SIP peers. |
| 21:00.36 | p3nguin | Not required for Asterisk to work. |
| 21:00.51 | SuperNull | Ty. |
| 21:01.19 | dijib | damn that upgrade broke meetme also |
| 21:02.42 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 21:02.52 | p3nguin | Make sure app_meetme.so is still there, and make sure you have dahdi installed still. |
| 21:03.09 | SuperNull | p3nguin you available ? i have some oddness i cannot explain.. i believe it to be asterisk .. oddly.. but the funnny thing is if i force a register from the ATA all works magically again.. |
| 21:03.50 | SuperNull | its almost like ast lost association with it. When this happens it starts sending sip invites to the ATA without any dest # info |
| 21:04.01 | dijib | dahdi wasnt installed. |
| 21:04.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:04.09 | SuperNull | TK ! :-) |
| 21:04.28 | p3nguin | I haven't seen any sip debug to support what you're saying. |
| 21:04.52 | SuperNull | ehhh i dont have the asterisk debug of it.. i have a tcpdump thats about it .. so far. |
| 21:05.22 | p3nguin | When can you get an asterisk sip debug of it? |
| 21:05.36 | SuperNull | i have a feeling it will be re-creatable when the registration time is up.. |
| 21:05.51 | SuperNull | it seems to correlate with some kind of time delay.. |
| 21:06.43 | p3nguin | sip unregister <peer name> |
| 21:07.07 | SuperNull | mmm ily let me try that. |
| 21:07.08 | SuperNull | haha |
| 21:07.28 | SuperNull | and... if its realtime ? |
| 21:07.43 | p3nguin | Now that I don't know about. |
| 21:07.47 | SuperNull | heh. |
| 21:07.55 | dijib | what package has yum-kmod in it? |
| 21:08.09 | SuperNull | well.. is there a per-user registration timeout i can set to like 5 minutes or something |
| 21:08.11 | p3nguin | I assume that people who use realtime know how realtime works. |
| 21:08.19 | SuperNull | ehhh.. |
| 21:08.21 | Janos | built a new cable with this pinout http://www.juniper.net/techpubs/hardware/m40/m40-hwguide/html/pinout4.html#1017868 but now getting BLUE + RED alarm :( |
| 21:08.26 | p3nguin | I assume that people who use realtime know how to use realtime. |
| 21:08.32 | SuperNull | i could turn caching on to allow it to work probably.. |
| 21:08.54 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:09.10 | SuperNull | p3nguin .. if i could find a great documentation source for realtime .. |
| 21:09.22 | SuperNull | even more specific .. the more up to date realtimes.. not 1.2 |
| 21:09.32 | p3nguin | I run asterisk on an embedded appliance, so realtime wouldn't necessarily be good for me to use; therefore, I do not know much about it. |
| 21:09.45 | SuperNull | understood. |
| 21:10.08 | SuperNull | i believe if i enable caching most functionality required will come back .. regarding command previously mentioned |
| 21:10.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:10.34 | p3nguin | I kind of want to mess with it just so I can help other people with problems, but I haven't gotten around to it. |
| 21:11.11 | SuperNull | wait a week or two .. we plan on moving everything to VMs .. i can 'donate' you some play space |
| 21:11.15 | SuperNull | for a little while. |
| 21:11.42 | SuperNull | is there a fairly verbose documentation of realtime on asterisk site somewhere that i cannot find ? |
| 21:13.41 | SuperNull | as expected p3nguin the command at least doesn't whine about user not existing when rtcache is on .. |
| 21:14.24 | p3nguin | I'm not sure if that's a good thing or a bad thing. |
| 21:15.05 | SuperNull | yeah not sure if its actually doing anything.. looking |
| 21:17.23 | p3nguin | Wow, must be a slow day... I haven't had an email in hours! |
| 21:17.31 | SuperNull | or email is broke.. you choose |
| 21:17.32 | SuperNull | ;-) |
| 21:17.41 | SuperNull | usually the case here when that happens lol |
| 21:17.46 | libryder | p3nguin: what do you do professionally? |
| 21:18.08 | p3nguin | I'm a Systems Engineer. |
| 21:18.41 | dijib | I'm a Systems Analyst |
| 21:18.46 | dijib | ist |
| 21:18.53 | dijib | sure i am |
| 21:18.56 | p3nguin | You analyze it, I'll engineer it. |
| 21:19.01 | dijib | these freaking dependencies are driving me madd |
| 21:19.05 | *** join/#asterisk wonderworld (~ww@port-92-201-11-128.dynamic.qsc.de) |
| 21:19.11 | p3nguin | Which one this time? |
| 21:19.17 | p3nguin | one(s) |
| 21:21.43 | dijib | yum-kmod |
| 21:21.53 | dijib | and dahdi-linux depending on it |
| 21:22.00 | dijib | and asterisk18-dahdi |
| 21:22.02 | libryder | awesome |
| 21:22.24 | dijib | its not in yum-utils apparently as thats already installed |
| 21:22.30 | dijib | yum-devel? |
| 21:23.45 | p3nguin | Why don't you install yum-kmod? |
| 21:23.52 | dijib | no package found |
| 21:23.57 | p3nguin | yum -y install yum-kmod |
| 21:24.22 | p3nguin | It should be in base. |
| 21:25.32 | p3nguin | I don't have a CentOS 6 system to check these things on for you, but I have to assume it's very similar to 4 and 5. |
| 21:26.16 | dijib | No package yum-kmod available. |
| 21:26.28 | p3nguin | Okay, give me a minute to figure out what the problem is. |
| 21:28.07 | p3nguin | i386 or x86_64? |
| 21:29.24 | *** join/#asterisk puzzled (~patrick@83.163.53.136) |
| 21:30.53 | dijib | i686 / i386 |
| 21:31.27 | SeRi | yum list|grep kmod |
| 21:33.11 | dijib | http://pastebin.com/9z7zYczG |
| 21:33.15 | p3nguin | How about yum search kmod? |
| 21:34.59 | dijib | http://pastebin.com/U4NmDgtx |
| 21:38.05 | p3nguin | I was saying that yum search kmod rather than yum and a pipe, and a grep. |
| 21:38.54 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:40.56 | dijib | oh |
| 21:40.59 | dijib | same thing |
| 21:41.08 | dijib | so kmod-dadhi-linux? |
| 21:43.43 | p3nguin | Are you trying to use a centos 5 rpm on centos 6? |
| 21:43.45 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 21:44.40 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 21:45.36 | dijib | no. |
| 21:45.45 | dijib | im trying to use the standard el6 repo's |
| 21:46.10 | p3nguin | Where are you getting the dahdi-linux and asterisk18-dahdi packages that are depending on yum-kmod? |
| 21:47.01 | dijib | http://pastebin.com/U1V72QNH |
| 21:47.39 | p3nguin | You said you're trying to install dahdi-linux and asterisk18-dahdi packages, and they are depending on yum-kmod. |
| 21:47.44 | p3nguin | Isn't that what you told me? |
| 21:48.04 | dijib | yes |
| 21:48.08 | p3nguin | Where are you getting those packages which are depending on yum-kmod? |
| 21:48.17 | dijib | yes depends on yum-kmod |
| 21:48.27 | p3nguin | Where are you getting those packages? |
| 21:48.38 | p3nguin | They are not in the "standard el6 repo" as you put it. |
| 21:48.56 | dijib | i guess from the asterisk-current repo? |
| 21:49.36 | p3nguin | I'd like to see your repo file for that. Check /etc/yum.repos.d/ for the file. |
| 21:49.43 | p3nguin | Copy the contents and pastebin it. |
| 21:51.36 | dijib | http://pastebin.com/k7nW7ZYd |
| 21:52.08 | p3nguin | baseurl=http://packages.asterisk.org/centos/5/current/$basearch/ |
| 21:52.09 | dijib | shit it is el5 |
| 21:52.15 | dijib | yep |
| 21:52.21 | dijib | there is no /6 |
| 21:52.34 | p3nguin | That's why it is depending on a package that does not exist. |
| 21:52.38 | p3nguin | It's for the wrong version. |
| 21:52.52 | dijib | ok so work around. find something with yum-kmod in it? |
| 21:53.11 | p3nguin | There is no yum-kmod for C6. |
| 21:53.23 | p3nguin | Maybe you could try to ignore it. |
| 21:54.14 | dijib | ive tried --skip-broken |
| 21:54.17 | dijib | didnt work |
| 21:56.07 | Freeaqingme | when a certain extension is dialed, I want the caller to get to hear some music, and on the background call some external numbers, wait for one of them to pick up, and press 1 (dtmf). After that, the first person pressing 1 should be connected to the called, and the other people pressing 1 should get a message that it was already taken by someone else. How is such a system called? |
| 21:57.18 | dijib | queue? |
| 21:57.38 | *** join/#asterisk mirco (~mirco@p5B283A33.dip.t-dialin.net) |
| 21:57.38 | Freeaqingme | hmmz, that could work |
| 21:57.59 | Freeaqingme | I could also use parking & paging, but I feel I would be abusing that |
| 21:58.10 | Freeaqingme | it's only one call / 2 weeks btw, so it's not like they need to queue up or something |
| 21:59.23 | mirco | hi hi, I've got a fritzbox 7570 which I use as an ATA, it's signaling the calls but doesn't transfer voice, I appreciate any help you can give... |
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| 22:04.03 | x86 | p3nguin: ipad access yet? :) |
| 22:04.59 | dijib | Freeaqingme, what kindof call is it, would a confrence work? |
| 22:05.26 | x86 | I'm wanting to test remote presence capabilities, like have a hint for SIP/user@someotherdomain... anyone have a domain-based SIP account I can try? (SRV records are fine) |
| 22:05.39 | Freeaqingme | dijib, it's our emergency number used to give support at night |
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| 22:21.46 | p3nguin | x86: If you would have reminded me at the right time, I would have checked that thing. Remind me in like six hours from now. |
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| 22:29.49 | dijib | ham and blue cheese stuffed chicken breasts for dinner.... who's in? |
| 22:29.59 | dym | meet you there! |
| 22:32.21 | Freeaqingme | when using DIAL() I want a channel to be answered only when dtmf digit 1 is pressed. Can that be done? |
| 22:32.46 | Freeaqingme | (preferably dialing multiple numbers at once) |
| 22:32.49 | dym | Freeaqingme: where is the 1 supposed to be called? |
| 22:33.05 | dym | or you mean dialout with a 1 prefix? |
| 22:33.47 | Freeaqingme | sorry. I want to dial out to several numbers simultaneously, but when answered, I want to play a file, and have the callee press 1 if he's available for a call |
| 22:34.45 | p3nguin | FollowMe() does that. |
| 22:35.02 | Freeaqingme | tnx! |
| 22:35.08 | Freeaqingme | I was thinking there had to be some function for this |
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| 23:41.11 | x86 | p3nguin: six hours I'll be long asleep ;) |
| 23:41.24 | x86 | p3nguin: can't you set an alarm on your phone or something? :P |
| 23:41.53 | x86 | p3nguin: I just need you to verify the Bria settings, and make sure you can register for inbound calls... and that two-way audio works |
| 23:42.12 | x86 | p3nguin: media5 fone works fine, but it's not designed for the iPad and the interface is sub-par |
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| 23:57.34 | root52 | Hey all. Odd problem. http://pastebin.ca/2087486 I am doing some func_odbc querys to an MS SQL server. I call the first time and it work as it should. I call back 5 second later and it seem to substitute the wrong values for variables causing the query to fail. I call back the third time and it works fine again! Maybe some one can see what is going on from the debug pasted above. Thanks!! |
| 23:58.39 | root52 | Also note the last line in both the working and non working examples is where it gets confused. The result of CIDLOCK should be 0 in both. |
| 23:59.12 | batfastad | Hi everyone. We've got a small asterisk/freepbx system of 18 extensions and I'm looking to build some sort of visual for our intranet of current SIP extension status. Kind of similar to FOP but with a manual refresh. |
| 23:59.59 | batfastad | And done in HTML of course. With AMI I understand that I can either poll for status or receive events when things change in asterisk, is that right? |