| 00:06.43 | damjan | hi all, I have several channels that are prepaid gsm lines, how can I set them to be round-robin acording to their credit |
| 00:17.02 | *** join/#asterisk adeel (~adeel@72.53.72.133) |
| 00:25.45 | *** join/#asterisk Russ (~russ@conference/linaro-connect/x-lnkkhwxybtmudwbw) |
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| 00:43.51 | mahables | Is anyone aware of an issue with Asterisk segfaulting after authentication to a prosody or ejabberd XMPP server? |
| 00:47.35 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 00:49.02 | Cubber | I have an asterisk 10 server behind a linksys wrt54g router running dd-wrt. SIP clients on the local network work fine with the asterisk server, however external SIP clients can access features and call PSTN using a google voice trunk but if I call an internal SIP phone I get one way or no audio issues. |
| 00:49.20 | Cubber | I have UDP 5060 and 10000-20000 forwarded to the asterisk server |
| 00:50.48 | Cubber | sip.conf http://pastebin.com/awdfPY9x |
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| 00:59.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:30.47 | leifmadsen | Cubber: sounds like you don't have localnet option setup correctly in sip.conf |
| 01:30.53 | tmrhmdv | Hi folks, has anyone had success testing Asterisk on Amazon's EC2 "Amazon Linux |
| 01:31.04 | tmrhmdv | "/ |
| 01:33.03 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 01:33.26 | tmrhmdv | I am having problem hearing sound, I have enabled ulaw and alaw in sip.conf, softphone 1, and softphone 2, they can call each other but can't hear |
| 01:33.27 | Cubber | leifmadsen: thanks I am really going through the default sip.conf right now and reading all of the NAT stuff in detail |
| 01:33.46 | leifmadsen | you just need to deal with externip and localnet |
| 01:33.49 | Cubber | leifmadsen: btw I have the definitive guide book, great job on that! |
| 01:33.54 | leifmadsen | that pretty much handles all the asterisk-behind-nat stuff |
| 01:34.27 | Cubber | so if I use externhost I dont need externip right? |
| 01:34.46 | Cubber | localnet=192.168.42.0/24 |
| 01:34.58 | Cubber | nat=yes |
| 01:35.38 | leifmadsen | right |
| 01:36.01 | leifmadsen | nat=yes is for devices behind nat, not asterisk itself |
| 01:36.12 | leifmadsen | if you set that in the general section, it'll assume all devices are behind nat |
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| 01:36.39 | Cubber | well I have devices on the local lan with teh asterisk server and am trying to get external devices working as well |
| 01:36.45 | Cubber | some may be behind nat some may not |
| 01:37.01 | leifmadsen | nat=yes in the general section won't hurt anything (in most cases) |
| 01:37.08 | Cubber | ok |
| 01:37.18 | Cubber | whatabout canreinvite is that depricated? |
| 01:37.30 | leifmadsen | I don't know what version of asterisk you're on |
| 01:37.36 | leifmadsen | but it's been renamed to directmedia I believe |
| 01:37.45 | Cubber | 10 |
| 01:38.15 | Cubber | so is that something I would need to explore? |
| 01:39.56 | leifmadsen | just set it to no |
| 01:41.22 | Cubber | I think that just did it |
| 01:41.28 | Cubber | directmedia=no |
| 01:44.20 | Cubber | yup I can now use a softphone on my laptop that is teathered to my cellphone to connect to my hardphone on my desk that is on the same lan as the asterisk server and have 2 way audio! |
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| 01:47.24 | Cubber | leifmadsen thanks for the help! Just in time too gotta go put my son to bed. Have a great evening! |
| 01:48.34 | dijib | so have we found anybody that wants this Sangoma A-200 for $100? |
| 01:49.02 | dijib | p3nguin: around? |
| 01:49.10 | dijib | SeRi |
| 01:49.11 | dijib | ? |
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| 03:08.51 | dijib | nobody in here this eveening? |
| 03:13.01 | *** join/#asterisk sacitec (~newbie@189.251.77.229) |
| 03:17.44 | sacitec | good afternoon people, i'm working with asterisk 1.8.7.0 with compiled videosupport and cisco 8941 videophones with SIP firmware; at the time i'm able to register the phones with sip extensions and work properly, but no video; anyone has success story with this phones and videocalls ? Also i'm trying with chan_sccp-b V4 designed for asterisk 1.8.x but still i'm unable to stablish videocalls. Any suggest would be great to hear |
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| 03:31.31 | sacitec | <PROTECTED> |
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| 05:00.39 | p3nguin | What a long day. |
| 05:01.13 | WIMPy | Yes. Linux is evil. |
| 05:01.54 | p3nguin | I actually didn't touch any Linux all day. |
| 05:02.36 | WIMPy | I had a nice issue with it mounting the root fs with the wrong type. |
| 05:03.36 | p3nguin | The fstab had it specified correctly? |
| 05:04.08 | WIMPy | The fstab is on the root fs. |
| 05:04.29 | p3nguin | Then what is the point of having an entry for / in the fstab? |
| 05:04.51 | p3nguin | Sounds like a chicken/egg problem. |
| 05:05.06 | WIMPy | Not sure. Cosmetics? |
| 05:05.21 | p3nguin | I doubt it's as trivial as that. |
| 05:05.41 | p3nguin | I bet if I delete the line for it and try to reboot it will fail. |
| 05:05.42 | WIMPy | Well, it certainly can;t be used for mounting. |
| 05:05.44 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 05:05.59 | WIMPy | It might be used on remount. |
| 05:06.01 | p3nguin | Maybe it's only used for remounting. |
| 05:06.32 | WIMPy | Anyway that was exactely the part that failed. |
| 05:07.39 | p3nguin | So if that is the case, the kernel has to detect the fs somehow prior to that. GRUB doesn't have any values for fs in menu.lst, so I don't know how you'd overcome that. |
| 05:08.01 | WIMPy | Neither do I. |
| 05:08.17 | p3nguin | Did you ever get past it or give up on it? |
| 05:08.21 | WIMPy | I ended up buliding a kernel that only has support for the right type. |
| 05:08.34 | p3nguin | That would have been crappy if it still didn't work. |
| 05:09.09 | WIMPy | It was crappy enought the way it was. |
| 05:10.42 | WIMPy | That way , changing disks too me a whole 5.5 hours. |
| 05:10.56 | WIMPy | incl. sceqing that is. |
| 05:15.37 | WIMPy | Oh wow. After 6h14m of uptime there's still parts of the screen blank in the firewall log. |
| 05:23.01 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
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| 05:45.01 | dijib | p3nguin: did you see what i found in a dumpster? |
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| 05:48.38 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 05:50.30 | p3nguin | No I didn't. |
| 05:50.58 | p3nguin | I guess I need to talk to someone who knows cable TV stuff to be able to finish up this project. |
| 05:52.01 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 05:55.05 | dijib | my brother does that |
| 05:55.07 | dijib | whats the project? |
| 05:55.40 | dijib | i found a core duo 1.66@800mhz machine with 512@667 ram and 2x80gb sata drives today |
| 05:55.56 | dijib | in it was a sangoma a200 card with an fxo dauter module |
| 05:56.21 | dijib | plugged it in and it was running centos4 and was in a kernel panic |
| 05:56.36 | dijib | ints now running 6.2 & asterisk 10.x |
| 05:58.14 | dijib | i dont have a use for this sangoma card.. what should i do with it? |
| 05:59.50 | p3nguin | You could donate it to me. |
| 06:00.14 | p3nguin | Then I can learn about analog cards and help people with that topic. |
| 06:00.22 | p3nguin | So the cable TV thing... |
| 06:01.26 | *** part/#asterisk techwerkz (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 06:01.29 | ChannelZ | It costs too much. What about it? |
| 06:02.13 | p3nguin | I'm installing what we call a "single channel server." It's a PC which connects to a Kramer scan converter which connects to a Channel Plus modulator. |
| 06:02.32 | p3nguin | I output on channel 120. |
| 06:03.05 | p3nguin | I don't know what the frequency is because the modulator only tunes by US channels and not frequency. |
| 06:03.12 | ChannelZ | Kickin it old-school |
| 06:03.42 | p3nguin | The facility has 23 TVs. They are all connected to Mediacom cable. |
| 06:03.51 | dijib | i would trade it for a couple ATA's |
| 06:04.33 | p3nguin | I had Mediacom install a combiner on the trunk line where it comes into the building so I can insert my new channel into the system. |
| 06:04.49 | p3nguin | The combiner is about 300 feet from the server. |
| 06:05.51 | p3nguin | The owner was going to get a young kid to come in and run the coax so he didn't have to pay $98 per hour to have it ran for him... |
| 06:06.04 | dijib | heh |
| 06:06.43 | dijib | p3nguin: what do you got to trade? |
| 06:07.36 | ChannelZ | try 769.2500mhz |
| 06:08.03 | ChannelZ | http://www.csgnetwork.com/tvfreqtable.html |
| 06:08.07 | p3nguin | So I told him to install RG-11 coax and not RG-6. RG-6 has something like 5 db loss per 100 feet, so we're looking at 15 db loss if he uses RG-6. RG-11 should only have ~7 db loss over the 300-foot run. |
| 06:08.41 | p3nguin | The modulator only has a 25 db output. |
| 06:08.59 | dijib | so guy cheaps out? |
| 06:09.09 | p3nguin | So I get a call saying the coax was pulled... and that he installed RG-6! |
| 06:09.37 | p3nguin | Okay, fine, so I'll see what happens. |
| 06:10.03 | coppice | RG6 == 75 ohms |
| 06:10.05 | coppice | RG11 == 50 ohms |
| 06:10.28 | coppice | or am I misremembering the numebrs? |
| 06:10.46 | ChannelZ | I'm already bored. Analog.. bleh |
| 06:10.50 | coppice | ah, maybe I am |
| 06:12.18 | p3nguin | RG-11 is 75 Ohms... which is why I told him to use it. |
| 06:13.12 | p3nguin | I hooked up a 35 db amplifier in line and turned the gain all the way up, but my new channel is still fuzzy. |
| 06:13.35 | p3nguin | With the gain all the way up, it causes interference on other channels. |
| 06:16.13 | p3nguin | I figured maybe I can find someone with the appropriate test equipment to get it straightened out. |
| 06:16.31 | coppice | oh, yeah. RG11 is the fat one that's really hard to get around corners |
| 06:16.51 | p3nguin | It's a good 1/2 inch diameter. |
| 06:17.38 | coppice | why do you have a problem? you really don't need much signal. how do you think TV work over the air? |
| 06:18.15 | p3nguin | There has to be enough signal to feed 23 TVs over hundreds of feet of coax. |
| 06:18.54 | p3nguin | I also installed the same setup at another place which had maybe 30 feet of RG-6, and that one worked perfectly. I still needed the inline amplifier, but at least I didn't have to turn it all the way up. |
| 06:19.12 | ChannelZ | Are the 23 TVs not being fed by a DA? |
| 06:19.30 | p3nguin | I mean 30 feet between the amp and the combiner, of course. |
| 06:20.03 | p3nguin | As far as I know, there are no other amplifiers anywhere in the system. |
| 06:20.32 | ChannelZ | What are they all daisy chained to each other? |
| 06:20.34 | p3nguin | The entire building has the huge trunk line and taps like you see outside on the pole feeding a neighborhood. |
| 06:21.43 | p3nguin | Trunk line comes in, hits a tap, feeds four TVs. Trunk line goes out of that tap and hits another tap which feeds eight TVs, and then to another tap which feeds another eight TVs. Then that trunk line heads off to another place which has more taps, and so on. |
| 06:22.24 | p3nguin | It's all set up like a city cable TV distribution is, but it is within a building. |
| 06:22.59 | p3nguin | I'd imagine there is one amplifier on the outside of the building which takes care of it all. |
| 06:23.54 | p3nguin | I expected to see a large distribution amplifier and enough ports on a tap to feed everything, but that wasn't what I found when I started looking around. |
| 06:27.30 | p3nguin | All this so the company can have their own channel to show their own crap on it. |
| 06:27.52 | p3nguin | I doubt most of the customers will even care to watch the company channel. |
| 06:33.42 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 06:35.41 | dijib | its good for marketing no? |
| 06:36.22 | dijib | who wants it? |
| 06:36.26 | dijib | http://i.imgur.com/USml9.jpg |
| 06:36.32 | dijib | http://i.imgur.com/XjDiO.jpg |
| 06:37.29 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:37.31 | schmidts | good morning |
| 06:38.03 | WIMPy | That's funny somehow. |
| 06:38.21 | *** part/#asterisk gajini (~root@61.12.17.171) |
| 06:38.32 | WIMPy | A modern port on one end and antique one on the other. |
| 06:39.47 | p3nguin | I don't really have anything to trade, but you can still donate it to me. |
| 06:40.13 | dijib | :D |
| 06:40.21 | dijib | i need some ATAs |
| 06:40.31 | dijib | i could really do some good with them |
| 06:41.07 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 06:50.18 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wfewheeahaovazex) |
| 06:51.58 | *** join/#asterisk jeffgus (~jeffgus@2001:470:f2eb:1::4) |
| 06:53.32 | p3nguin | What would you do with them? |
| 06:56.25 | *** join/#asterisk its_jeremy_ (~omghax@24-119-28-62.cpe.cableone.net) |
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| 07:02.31 | anny__ | i want to use a VOIP protocol over the internet, does IAX offer benefits over SIP, because I read that SIP suffers from NATing problems |
| 07:07.50 | WIMPy | TheSIP NAT issue is usually not a problem with Asterisk, but IAX only uses a single port, avoiding any issues. |
| 07:09.38 | anny__ | does Asterisk have mechanisms to cope for the NAT traversal issue? |
| 07:09.40 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 07:09.42 | anny__ | in SIP i mean |
| 07:10.07 | WIMPy | Yes |
| 07:10.37 | dijib | crap |
| 07:10.41 | WIMPy | And it usually works, but some routers are to bad. |
| 07:10.54 | dijib | spaced mountained out |
| 07:11.49 | p3nguin | Spaced out mountains? |
| 07:12.10 | dijib | i space mountained out. and i can say for one i have ridin space mountain |
| 07:16.10 | anny__ | WIMPy: thx |
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| 07:18.41 | dijib | there is not enough space left on your desktop |
| 07:18.44 | dijib | uhhh |
| 07:19.35 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
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| 07:26.28 | dijib | anybody got any trades for this sangoma card? |
| 07:27.34 | p3nguin | I'll trade you the satisfaction of getting rid of it... for the card. |
| 07:27.53 | WIMPy | I'd love to try out a Sangoma card, but I'd need a time machine to use that one. |
| 07:27.53 | dijib | see i just dont have any anolog lines i can play with |
| 07:28.35 | dijib | and i want else |
| 07:33.13 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:39.09 | schmidts | dijib i dont know about analog lines but i use sangoma octo Pri cards |
| 07:40.22 | dijib | i have one i want to get rid of |
| 07:44.22 | schmidts | what card? |
| 07:56.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 08:09.47 | saxa | ChannelZ: but why before the upgrade i had no this kind of error ? The setup is the same as it was and the sip.conf too. |
| 08:19.01 | ChannelZ | Maybe you had all the error/warning logging suppressed... or your device wasn't registering before and now is. I don't know. |
| 08:19.26 | ChannelZ | but if the peer has a host=something.com or host=1.2.3.4 it shouldn't be trying to register, and that's why you're getting that. |
| 08:19.29 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:22.55 | saxa | ChannelZ: in sip the peer don't have a host=dynamic option. Anyway now it seems that after restarting is back to normal. |
| 08:25.19 | saxa | s/sip/sip.conf |
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| 09:45.03 | the_tux | hi all |
| 09:45.46 | the_tux | having issue, "No such host" for incoming calls, outgoing calls are ok |
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| 10:03.13 | prasanthprabalan | hi all |
| 10:06.44 | prasanthprabalan | Just now i tried to register a softphone with asterisk 1.8.9.2 for a sip connection. |
| 10:06.50 | prasanthprabalan | But not regitering. |
| 10:07.23 | prasanthprabalan | do i have to do any other setup other than sip and extension configuration for asterisk 1.8.9.2 |
| 10:08.58 | prasanthprabalan | This is my SIP and EXTENSION confs |
| 10:09.05 | prasanthprabalan | http://pastebin.com/AFm4WUrv |
| 10:09.28 | prasanthprabalan | Softphone used is "Zoiper" |
| 10:11.55 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 10:13.00 | WIMPy | A peer is enough. Try to increase debug and verbose and see if anythig shows up. |
| 10:13.21 | WIMPy | If not, try sip debug to see if anything arrives at all. |
| 10:14.01 | ziz212 | dear all, I am finding out a asterisk application to terminate the call. Hangup is good but after Hangup dial plan wont continue. I need a application to to terminate the current call and go to next line in dial plan. Pls help me on this. |
| 10:14.09 | prasanthprabalan | No actually. When am registering am getting no response from asterisk |
| 10:14.33 | Donper | try to set nat=no in your peer configuration |
| 10:15.19 | prasanthprabalan | @Donper: Is this nat=no for me.?? |
| 10:15.22 | WIMPy | ziz212: No go. You might be able to use the h extensions to do whatever you need. |
| 10:15.24 | kaldemar | ziz212: no reponse to a register message from asterisk in asterisk's sip debug? |
| 10:16.14 | Donper | @prasanthprabalan : yes it is |
| 10:16.54 | Donper | @prasanthprabalan : i've got the same problems to register a phone with 1.8.9.2, and you can also specify in general section |
| 10:18.57 | prasanthprabalan | @Donper: I put "nat=no" in general session of sip.conf.. But no change |
| 10:19.16 | Donper | What do you see in the CLI ? |
| 10:19.26 | prasanthprabalan | @Donper: Still no response from asterisl CLI |
| 10:19.50 | Donper | and if you activate debug ? in CLI : sip set bdebug on |
| 10:21.09 | prasanthprabalan | @Donper: I enabled. But still no response from asterisk |
| 10:21.29 | prasanthprabalan | @Donper: Not showing any cli o/p.. |
| 10:21.53 | prasanthprabalan | @Donper: My verbose level is 20 |
| 10:22.21 | Donper | how did you configure your softphone ? |
| 10:23.24 | Donper | in peer session you can add "defaultuser=test" (before, this parameter called username) |
| 10:24.17 | kaldemar | the zoiper is not sending anything to asterisk if there is nothing in sip debug. |
| 10:24.32 | kaldemar | it is misconfigured or there is a network problem. |
| 10:25.01 | prasanthprabalan | @Donper: My asterisk and softphone is in same machine. |
| 10:25.44 | prasanthprabalan | @kaldemar: No n/w probs since bothe in same machine |
| 10:26.07 | gladier | does anyone have a practical example where RDNIS would be used |
| 10:26.28 | gladier | DNIS and ANI make perfect sense... RDNIS is just a mindfsck |
| 10:26.31 | Donper | @prasanthprabalan: what do you specify about outbound on your softphone? Domain=127.0.0.1 or Proxy=127.0.0.1 ? |
| 10:26.50 | prasanthprabalan | In Zoiper this is my setup: Domain: <IP>, Username: test; password: test; Caller ID: 6000 |
| 10:27.43 | WIMPy | prasanthprabalan: You have to change the port on one of the applications. Twice the same port on the same machine doesn't work. |
| 10:30.09 | prasanthprabalan | @Donper: Domain i have given as my asterisk IP ie 192.168.39.14; Not the loop back |
| 10:30.58 | prasanthprabalan | @WIMPy: Do i have to change the bindport in SIP.conf |
| 10:31.03 | the_tux | how I can change the username and passwords of users from sip.conf? username=somename is showing same results as username= ... |
| 10:31.32 | kaldemar | prasanthprabalan: running asterisk and a soft phone on the same machine is a common cause for network issues. |
| 10:31.37 | WIMPy | prasanthprabalan: Either that or the one in zoiper, but I guess that's not possible. |
| 10:32.19 | prasanthprabalan | kaldemar: But i have done it a 100 times with asterisk 1.6. |
| 10:32.45 | prasanthprabalan | kaldemar: Problem seems to occur only with this asterisk version 1.8.9 |
| 10:33.36 | kaldemar | prasanthprabalan: and they are not trying to use the same port? |
| 10:35.33 | ziz212 | kaldemar: I couldnt get what you said? |
| 10:35.49 | prasanthprabalan | My "sip reload" is giving this warning |
| 10:35.50 | prasanthprabalan | http://pastebin.com/sBscFfka |
| 10:36.15 | WIMPy | That's what I said. |
| 10:36.22 | prasanthprabalan | @ kaldemar: My "sip reload" o/p |
| 10:36.41 | prasanthprabalan | @kaldemar: http://pastebin.com/sBscFfka |
| 10:37.08 | WIMPy | You can't use the same port twice. |
| 10:37.58 | kaldemar | prasanthprabalan: why would you think that is? zoiper is already listening on port 5060. |
| 10:38.22 | kaldemar | ziz212: sorry, that wasn't meant for you. |
| 10:38.35 | ziz212 | let say if I am dialing from out side through sip and i need to write a call back dial plan for that. First need to save the caller id, disconnect the call and dial back to the perticular caller id. I did alll and to disconnect I used Hangup(). But there after dial plan stop execute to next line. |
| 10:38.48 | ziz212 | ok it ok |
| 10:38.55 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 10:39.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:39.39 | ziz212 | Busy() is also not suitable, because it stop execute dial plan after that |
| 10:40.22 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:40.40 | prasanthprabalan | @kaldemar: WARNING[2414]: chan_sip.c:28526 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use |
| 10:40.44 | WIMPy | ziz212: You need to start a new call. 'channel originate', AMI originate or a call file. |
| 10:41.06 | prasanthprabalan | <PROTECTED> |
| 10:41.10 | WIMPy | prasanthprabalan: What's your question? Asterisk tells you exactely what we are telling you. |
| 10:41.50 | ziz212 | thanks but how could i stop the call. If i stop using hangup then it wont continoue |
| 10:41.54 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:42.17 | WIMPy | ziz212: You start the new call and then hangup. |
| 10:43.00 | ziz212 | Thanks but my try is to dial same caller id. so first of all I need to terminate the call. |
| 10:43.51 | WIMPy | It's game over then. |
| 10:43.52 | ziz212 | let say I am dial to pabx and my caller id get stored. then call need terminate. last pabx will call me by using my caller id. |
| 10:44.26 | prasanthprabalan | <PROTECTED> |
| 10:44.27 | WIMPy | Start a new call in any of the ways I told you. |
| 10:45.32 | WIMPy | prasanthprabalan: If you want to do yourself a favour, don't try to run a soft phone on the same machine as Asterisk. |
| 10:46.11 | ziz212 | Can pabx originate a call to same caller when the caller is there online with pabx |
| 10:46.37 | *** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net) |
| 10:46.39 | WIMPy | It can do whatever you like. And if you want to add a delay, do it in the new call. |
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| 10:48.39 | ziz212 | ok first of all the caller need to be termiante. then he is free to receive a call. my prob is if I terminate the call using hangup in the dial plan, then dial plan wont continue to next line. because of Hangup command |
| 10:49.10 | WIMPy | Read again. |
| 10:49.29 | kaldemar | prasanthprabalan: zoiper and asterisk cannot listen on the same port, if that's what you mean. i don't know if you can configure a different port in your zoiper. so do it in asterisk. |
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| 10:52.10 | kaldemar | prasanthprabalan: looks like you can configure the SIP port zoiper listens on, but not in the zoiper GUI. modify sip_options/port in ~/.zoiper/zoiper.conf |
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| 11:15.29 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 11:15.36 | Tuju | [Feb 17 07:57:50] NOTICE[13462]: chan_sip.c:22081 handle_request_invite: Call from '' (66.226.74.249:5060) to extension '1337442032988634' rejected because extension not found in context 'default'. |
| 11:15.49 | Tuju | how is that possible that someone tries to call via '' ? |
| 11:16.05 | Tuju | i've tens of those lines to different numbers. |
| 11:16.08 | WIMPy | Because you allowed it? |
| 11:16.16 | Tuju | i didin't. |
| 11:16.29 | WIMPy | So how is it possible then? |
| 11:16.38 | Tuju | i'm asking, not answering. |
| 11:17.01 | WIMPy | I gave the answer, even if technically was written as a question. |
| 11:17.22 | WIMPy | Don't allowguests. |
| 11:17.26 | Tuju | you asked if i allowed it, i didn't. |
| 11:17.31 | Tuju | hmmm... |
| 11:17.44 | WIMPy | is sure, you did. |
| 11:18.42 | Tuju | ;allowguest=no <---- that setting? |
| 11:18.53 | WIMPy | yes |
| 11:19.05 | Tuju | ack. i didn't touch that one. |
| 11:19.10 | Tuju | stupid default. |
| 11:19.24 | WIMPy | well... |
| 11:20.08 | Tuju | is there something like postconf for postfix? |
| 11:20.18 | Tuju | to show all active settings from cli |
| 11:20.44 | WIMPy | Not just one. |
| 11:20.58 | WIMPy | 'sip show settings' might be the one you like. |
| 11:20.59 | schmidts | Tuju core show settings for example ;) |
| 11:21.09 | Tuju | ack, i'll try it. |
| 11:21.52 | Tuju | and yes, is there some other setting that should be checked from initial installation? |
| 11:23.20 | schmidts | Tuju there is a file i dont know its especial name but you will see it after make install, its like a startup guide for security stuff |
| 11:23.22 | WIMPy | Look at the READMEs, esp. the README-SERIOUSELY.bestpractices.txt |
| 11:23.53 | schmidts | WIMPy thats what i meant :D |
| 11:23.56 | Tuju | ack, thanks. I need to go through them before i attach and pstn lines to that box. :) |
| 11:24.11 | schmidts | Tuju thats why its called README-SERIOSELY ;) |
| 11:24.27 | Tuju | yeah, kind of sounds like it. :) |
| 11:25.07 | Tuju | i just wonder what's the point of creating internet technology that would need such tuning by default. |
| 11:25.10 | schmidts | hmm maybe we should something like asterisk will not start until you have read this file. |
| 11:25.20 | Tuju | like internet wouldn't have evil people. |
| 11:25.30 | schmidts | like this old copyright stuff, read until the end and enter the last three characters of the last line |
| 11:26.27 | kaldemar | schmidts: sounds as nasty as those EULA pop-ups that need to be scrolled down and a box checked before you can install a program. |
| 11:26.33 | Tuju | schmidts: imo better defaults would be better. |
| 11:26.52 | Tuju | be conservative with them. like apache httpd does. |
| 11:27.12 | WIMPy | That has been discussed many times. |
| 11:27.32 | WIMPy | It's to make it usable without configuration. To make it less scary or something. |
| 11:27.38 | Tuju | to make asterisk work, you need to open 5060 to internet. you can assume that shit happens if it's allowed to. |
| 11:28.53 | *** join/#asterisk RickCogley_ (~anonymous@114.179.8.185) |
| 11:29.14 | Tuju | what is that guest functionality? |
| 11:29.26 | Tuju | allow proxying without configured lines? |
| 11:29.34 | Tuju | you just claim who you are? |
| 11:30.27 | kaldemar | to allow unauthenticated calls that land into the context that is defined under [general], or "default" if omitted. |
| 11:30.57 | kaldemar | not proxying anything, since asterisk is not a SIP proxy. dialplan defines what asterisk does with calls. |
| 11:32.35 | Tuju | ack |
| 11:33.35 | Tuju | so it's like allowing apache to write web pages over http protocol for anyone to ease up web deployment. |
| 11:34.18 | Tuju | except that if some poor missed that SERIOUSLY file and hooked some real life phone lines to that box, you loose real life money on those prank calls. |
| 11:34.30 | kaldemar | or to communicate with people you do not know and devices that are not configured on your end. |
| 11:35.12 | Tuju | yep, less damaging, only possible dent in your reputation. |
| 11:35.20 | WIMPy | There are more possibilities to pay for other ppls calls when using voip. |
| 11:35.25 | kaldemar | loosin money would require making real calls available in the default context that. so not only default settings would allow that. |
| 11:35.56 | Tuju | kaldemar: would that be so rare situation to allow making calles in default? |
| 11:35.59 | Tuju | calls |
| 11:36.19 | WIMPy | Hopefully |
| 11:36.59 | kaldemar | Tuju: it should be rare and it almost never happens. nothing outgoing should be allowed in the default context at all. |
| 11:37.21 | Tuju | imo user community should write their own README-SERIOUSLY for different projects. :) |
| 11:38.03 | Tuju | and wave that when they haven't read it :) |
| 11:38.04 | kaldemar | well this channel is close to that kind of "ask us questions and take us seriously". :) |
| 11:39.13 | Tuju | you make it sound like other end of your ecosystem wouldn't be for taken seriously (with no disrespect from my side however ) |
| 11:40.20 | Tuju | i guess my point is - SIP, like IPv6 for example, are still new technology for most even technical people out there. You just can't assume anything from their knowledge level about it. |
| 11:42.51 | Tuju | real life example - i consider myself completely newbie on this. regardless of that, one sw company asked me to get them started with asterisk. |
| 11:42.56 | kaldemar | agreed. that's why people usually try to point out that the sample configs that come with asterisk are not meant for real usage. and there are docs on the subject also. |
| 11:43.39 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 11:43.52 | Tuju | kaldemar: i've heard that before. only after i had some issues with them. the fact is that learning curve for creating a working getting-started configs is way too steep to do so. |
| 11:44.07 | Tuju | secondly, everyone uses those shipped files with apache httpd. |
| 11:44.33 | Tuju | you're fighting against old behaviour culture of opensource unix world here. |
| 11:45.14 | Tuju | why don't you just provide some good, conservative default configs for distribution packagers? |
| 11:45.52 | Tuju | now it seems obvious that all those that are shipped are more or less done by people who didn't have competence to write them. |
| 11:50.34 | WIMPy | With traditional telephony you also have a chance to pay for foreign calls, so that's not really a new thing. |
| 11:51.19 | kaldemar | i'm not fighting anything. the thing is that comparing asterisk to apache for example is kind of troublesome. apache is meant for serving web pages, where asterisk can be used to build many different kinds of applications, a PBX being just one purpose. |
| 11:52.17 | Tuju | kaldemar: you know that, we don't. for majority of people getting knowing it, it's just the thing that runs phones. |
| 11:53.00 | kaldemar | of course there are default settings that should be changed, and there always will be. depends on who one asks. |
| 11:53.14 | Tuju | if you assume outsiders having same level of knowledge as asterisk insiders have, we end up having problems like currently. |
| 11:53.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:54.06 | kaldemar | you got me wrong. i don't and neither does anyone else. on the contrary, there's more like an assumption of zero knowledge until proven otherwise. |
| 11:54.38 | Tuju | imo the question was not about defaults, but some default configs that override what ever was default and those are files that gets every time here judgement: "don't use them." |
| 11:55.32 | Tuju | i for example would have been very happy with very very basic files instead of those i ended up using then. |
| 11:55.38 | kaldemar | yes, also consider that fact that i don't represent digium or the development team for asterisk in any way. |
| 11:55.51 | Tuju | kaldemar: ack |
| 11:56.44 | Tuju | i've a feeling that i also share my distributin packager opionion of those configs. those exist when there's nothing better. |
| 11:59.17 | WIMPy | It's abviousely a package issue. If you build Asterisk from source it should be clear that the supllied configs are samples, not defaults. |
| 11:59.58 | Tuju | that message got lost somewhere until i got it. :) |
| 12:00.29 | Tuju | yum install -y asterisk doesn't contain any 1h introduction lesson. :) |
| 12:01.37 | WIMPy | That's why the samples allow everything: So you have a chance to get anyhting going in an hour. |
| 12:02.03 | WIMPy | But they really should only be used as a reference. |
| 12:03.13 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 12:03.53 | Dovid | are there any stats where Asterisk is downloaded from by country? |
| 12:09.59 | Tuju | WIMPy: i'm not trying to be nuisance here. finally just point out the controversy that "starting from full example to cut it down to needed" approach doesn't work if you don't know anything about it. |
| 12:10.39 | WIMPy | finds the comments perfectly valid. |
| 12:11.09 | WIMPy | But that's a dilemma. You can't understand Asterisk in an hour. |
| 12:11.14 | Tuju | it's easier to add bit by bit by reading docs "okay, that's how callerid looks like and i need it". |
| 12:11.26 | Tuju | WIMPy: certainly true. |
| 12:11.59 | WIMPy | I think the real issue is that the sample files a placed on the configuration directory. |
| 12:12.09 | Tuju | imo it would be good to have very bare config so that it would start. No lines/accounts whatever. Just basic config and then steps how to make it functional. |
| 12:12.22 | Tuju | WIMPy: certainly adds the mess. |
| 12:12.28 | WIMPy | And delivering a package like that is indeed questionable. |
| 12:12.41 | Tuju | those should reside under /usr/share/docs/asterisk-something |
| 12:13.14 | Tuju | but imo that's the result of nothing having default configs from upstream. |
| 12:13.22 | Tuju | but imo that's the result of nothaving default configs from upstream. |
| 12:13.27 | Tuju | argh. :) |
| 12:13.29 | WIMPy | Probably. |
| 12:14.21 | Tuju | i've wondered why some calls have ip address in the right side of @ char. |
| 12:14.21 | WIMPy | But as kaldemar already pointed out, Asterisk can be used for very different things. So a default might indeed be impossible. |
| 12:15.17 | Tuju | that's probably true as you say it. new people installing it are doing it however for voip server. having such default configs would serve 100% of them. |
| 12:15.31 | Tuju | since it was news to me too and i already know something about it. |
| 12:16.48 | WIMPy | So what is a "voip server"? |
| 12:17.20 | Tuju | you're again assuming that we new comers would know something about this. :) |
| 12:17.39 | Tuju | voice == phones, server - that's it. |
| 12:17.56 | WIMPy | You said newcommers are doing it for voip server. But what is that? |
| 12:18.07 | WIMPy | That can still mean a lot of things. |
| 12:18.09 | Tuju | running phones. isn't that obvious? |
| 12:18.22 | Tuju | phones don't work by themselves. |
| 12:18.42 | WIMPy | Actually voip phones usually do. |
| 12:18.58 | WIMPy | SIP is a P2P thing. |
| 12:19.01 | Tuju | you're making this to academic question by raising irrelevant details on table. |
| 12:19.12 | Tuju | i know it. in practise it's not. |
| 12:19.36 | Tuju | thanks to firewalls, NAT and all asswipes out there willing to break things. |
| 12:19.40 | WIMPy | Why not? Most of them allow you to just enter an IP. |
| 12:20.08 | Tuju | WIMPy: in practise? i think our discussion is over already. it's not fruitful. |
| 12:20.09 | WIMPy | See, there we go, so you assume that not all phones are on site? |
| 12:20.11 | Tuju | but thanks anyway. |
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| 13:02.48 | *** join/#asterisk enoch (~arturo@unaffiliated/enoch) |
| 13:02.56 | enoch | guys i have a question |
| 13:03.18 | enoch | what's the good driver for hfc-s based boards? mISDN or vzaphfc? |
| 13:05.37 | leifmadsen | I imagine it's mISDN now |
| 13:05.47 | enoch | ok |
| 13:09.06 | WIMPy | Depends on what features you want. |
| 13:09.21 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/3-3 |
| 13:11.20 | enoch | i need to use it with a 2 bri isdn line |
| 13:11.30 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:11.50 | enoch | using software sip and a tdm410p with 2 fxo |
| 13:11.58 | enoch | fxs* sorry |
| 13:12.09 | enoch | the hfc-s board will be the LINE IN |
| 13:14.43 | WIMPy | You were the guy with only one slot? |
| 13:15.11 | enoch | lol yep |
| 13:15.28 | enoch | switched to a micro-atx board |
| 13:15.33 | WIMPy | Sou you fond a 2nd one? |
| 13:16.22 | WIMPy | You didn't like ATAs or terminal adapters? |
| 13:16.34 | enoch | naa |
| 13:16.45 | enoch | btw misdn seems to be simpler to install |
| 13:17.09 | WIMPy | Nothing is simple here. Not until you done it :-) |
| 13:18.44 | [TK]D-Fender | enoch, And last time it was FXO you needed for analog lines... not FXS for phones... |
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| 13:23.57 | cusco_ | hi folks |
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| 13:46.35 | rgsteele | So, I have users' vm passwords defined in voicemail.conf. Users occasionally change these passwords, but when Asterisk is restarted, they revert back to what's in voicemail.conf. Isn't Asterisk supposed to use its internal database before consulting the files, or do I have that backwards? |
| 13:47.27 | [TK]D-Fender | rgsteele, * mods voicemail.conf |
| 13:48.58 | rgsteele | [TK]D-Fender: Does that require any options to be set? E.g., writeprotect or static? |
| 13:49.15 | [TK]D-Fender | rgsteele, Don't believe so. |
| 13:51.00 | *** join/#asterisk latz (~frank@85.93.194.177) |
| 13:51.25 | leifmadsen | rgsteele: file needs to be writable by asterisk, and you can't use an #include file to store the voicemail information |
| 13:52.56 | rgsteele | Hm, yeah, it does change it. I recalled them reverting... oh, I know what the problem is. Configuration mgmt engine overwriting the file.... |
| 13:52.58 | rgsteele | D'oh |
| 13:53.17 | rgsteele | Sorry for wasting your keystrokes :P |
| 13:53.19 | latz | hey guys, I have this really strange provider from South Africa, from where I have to receive calls in my asterisk setup, and this provider does't usse any username nor password, so I figure I can't use the register => thing ... so I added the provider as a peer = friend with host = ipofprovider ... when I do sip show peers I see the host online, the question that I have is where and how do I need to specify the telephone number ... thanks for any help |
| 13:54.22 | *** part/#asterisk enoch (~arturo@unaffiliated/enoch) |
| 13:54.23 | leifmadsen | rgsteele: update the database at the same time using the script hooks in voicemail.conf |
| 13:54.44 | [TK]D-Fender | latz, What do you mean "specify the telephone number" ? |
| 13:55.11 | leifmadsen | latz: it depends entirely on how the provider sends you the call. If they request a number that happens to match the DID, then you match that in the dialplan |
| 13:56.23 | latz | o.key but I should be able to see that in the sip debug right? |
| 13:56.39 | [TK]D-Fender | latz, yes |
| 13:56.44 | latz | if I place a call I don't see any connection coming in ... nor via tcpdump nor in sip debug |
| 13:57.14 | [TK]D-Fender | latz, Then packets aren't arriving. Since you're not registering I presume you gave them you IP... right? |
| 13:57.38 | latz | [TK]D-Fender: yes exactly |
| 13:57.54 | [TK]D-Fender | latz, Go check all your routing, firewalls, etc |
| 13:57.55 | latz | so but just to be right .. I don't need the register thing then with this provider? |
| 13:58.01 | latz | thanks ! |
| 13:59.23 | [TK]D-Fender | latz, Register just tells them where you are. |
| 13:59.30 | [TK]D-Fender | ltYou gave them the IP. |
| 14:01.23 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 14:01.50 | *** join/#asterisk Assid (~Assid@unaffiliated/assid) |
| 14:01.53 | Assid | heya |
| 14:03.32 | Assid | i just re-setup a asterisk box after ages.. but i seem to be facing the whole nat issue... i have enabled directmedia=no for every sip peer/friend which is behind a nat.. and also set nat=yes |
| 14:04.20 | Assid | however, i am still facing issues.. sip debug shows it keeps trying to register |
| 14:04.58 | Assid | when i use a stun server from ekiga.. the registering stops going all nutty. however, i would like to avoid the use of a stun server if possible |
| 14:05.29 | leifmadsen | Assid: you don't need a stun server, and it's not really used with asterisk is almost all situations |
| 14:05.34 | leifmadsen | Assid: is asterisk behind nat too? |
| 14:05.50 | *** join/#asterisk serafie (~erin@nat/digium/x-nxempkbgjjkputje) |
| 14:05.51 | Assid | leifmadsen no its on the piublic internet |
| 14:06.01 | leifmadsen | then you've done everything you need really |
| 14:06.22 | leifmadsen | directmedia=no and nat=yes can just be set in the general section |
| 14:06.35 | ziz212 | deal all, now i am trying a dial plan which will dial back to the calling party using his calling number after terminating his call from pabx. As an ex if a caller called to a pabx, his num get stored and his call get terminated. then pabx generat a call to the caller using stored calling number. But it is not working. Dial plan and out put is here http://pastebin.com/jdr6prqN pls help... |
| 14:06.37 | ziz212 | ...me for this. |
| 14:07.19 | *** join/#asterisk domi (domi@manzano131.server4you.de) |
| 14:08.00 | leifmadsen | ziz212: callerid matches, so it goes to 'yes', which is Hangup() |
| 14:08.28 | WIMPy | ziz212: You still didn't read what I told you. |
| 14:08.38 | leifmadsen | and a no would also be a hangup() |
| 14:08.45 | WIMPy | Scroll up. I'm not going to write it again. |
| 14:08.47 | leifmadsen | your dialplan logic (or lack thereof) is bogus |
| 14:08.59 | leifmadsen | and it looks like WIMPy has already told you what to do, so my work here is done |
| 14:09.02 | WIMPy | leifmadsen: He wants a callback. |
| 14:09.20 | leifmadsen | WIMPy: ah ya that's what I figured... he needs Originate() or something like that (which I'm sure you've already told him) |
| 14:09.33 | WIMPy | And I'm sure google has some examples as well. |
| 14:09.45 | WIMPy | yes |
| 14:11.16 | ziz212 | Thanks guys ... I think i might misunderstood what you all are saying .. I dont know how to resolve this issue. How ever I will try my best to understand. |
| 14:11.18 | WIMPy | But the Dialplan originate wouldn't work as far as I know. |
| 14:11.53 | Assid | leifmadsen i thought you can set nat=yes per friend/context |
| 14:12.09 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-90.telkomadsl.co.za) |
| 14:12.24 | leifmadsen | Assid: you can, but it's not necessary in nearly all cases |
| 14:12.31 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 14:12.44 | *** join/#asterisk vassilux (~vassilux@LPuteaux-156-14-48-156.w82-127.abo.wanadoo.fr) |
| 14:12.49 | leifmadsen | WIMPy: right, the channel would need to remain active, so he needs to use a call to the AMI or a callfile likely |
| 14:13.02 | [TK]D-Fender | ziz212, You jump to a Hangup() in both cases. You clearly don't want to do this. so REMOVE one. |
| 14:13.09 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-awuwuehhfcyfqqba) |
| 14:13.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:13.41 | Assid | leifmadsen but im still facing that issue |
| 14:13.54 | leifmadsen | Assid: then you're going to have to look at the SIP debug and determine what is going on |
| 14:14.05 | WIMPy | A non-blocking Originate() would be good for things like that. |
| 14:14.05 | leifmadsen | there are no additional options that need to be set on the Asterisk side |
| 14:14.20 | leifmadsen | WIMPy: indeed |
| 14:15.02 | Assid | also i get the following : [Feb 17 19:44:35] WARNING[31068]: chan_sip.c:3389 __sip_xmit: sip_xmit of 0x8a28b88 (len 748) to x.x.x.x:36071 returned -2: Success |
| 14:15.22 | Assid | when i try and connect without stun |
| 14:15.45 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:15.48 | vassilux | hi, I make cross post into asterisk-gui :-) I use asterisk-gui and want to receive fax. I put faxdetect=both into chan_dahdi.conf and users.conf but I can't get fax. Can someone indicate me what is right way to configure my asterisk box for fax receiving. |
| 14:15.51 | latz | [TK]D-Fender: so I have this in sip show peers it should be o.k for the configuration ---> africa 196.34.250.62 5060 OK (188 ms) |
| 14:16.16 | Assid | leifmadsen but google doesnt show me anything for that xmit issue.. |
| 14:16.20 | [TK]D-Fender | latz, Still has nothing to do with not seeing an inbound attempt |
| 14:16.30 | Assid | leifmadsen apparently goes away IF i use stun |
| 14:16.53 | leifmadsen | Assid: I don't know, you haven't provided any further information that allows me to provide new suggestions |
| 14:17.09 | Assid | let me get you a sip debug |
| 14:17.16 | latz | [TK]D-Fender: sure ... which ports does the provider try to connect to my server via 5060/tcp 5060/udp ? |
| 14:17.17 | leifmadsen | sip transmit issue is due to other end not responding |
| 14:17.18 | [TK]D-Fender | vassilux, that doesn't receive a fax... that just sets it to detect and jump to the "fax" exten. You are still responsible for the dialplan code in it |
| 14:17.31 | leifmadsen | work time, sorry kids |
| 14:17.33 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:17.33 | latz | [TK]D-Fender: sorry I am really a noob in terms of sip and stuff |
| 14:17.50 | vassilux | Yes; I add fax to my inbound route |
| 14:17.56 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 14:20.48 | [TK]D-Fender | vassilux, Doesn't tell us what it actually added, or show that it detected it in a call. |
| 14:22.04 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 14:22.10 | Assid | [TK]D-Fender care to help me with this please? http://pastebin.com/2pGxh97D ; sip client doesnt reigster correctly |
| 14:23.05 | Assid | if i use a stun server; like stun.ekiga.net works fine.. |
| 14:23.42 | [TK]D-Fender | Assid, Looks like it registers fine |
| 14:23.50 | ziz212 | Dear guys, I have figure out that if the call is terminated from the pabx there is no process is going on. so no dial plan rule is applied. so how could i accomplish my target? |
| 14:23.55 | [TK]D-Fender | Assid, comes in, 401, acks with auth, 200's... |
| 14:24.14 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:24.18 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:24.18 | Assid | yeah.. but it keeps doing that.. also gives me that also i get the following : [Feb 17 19:44:35] WARNING[31068]: chan_sip.c:3389 __sip_xmit: sip_xmit of 0x8a28b88 (len 748) to x.x.x.x:36071 returned -2: Success |
| 14:24.35 | [TK]D-Fender | "Success" |
| 14:24.39 | [TK]D-Fender | Dunno... |
| 14:25.03 | [TK]D-Fender | ziz212, We can't understand what you're asking for. Please rephrase. |
| 14:25.29 | Assid | everything said anddont .. it should be working.. i dont see why it shouldnt |
| 14:25.50 | kaldemar | ziz212: use hangup extension that originates a new call. |
| 14:26.33 | Katty | drags in |
| 14:27.20 | [TK]D-Fender | conversation drags on ... |
| 14:27.55 | ziz212 | ok pls corrent my dial plan http://pastebin.com/1Wm7RNXi |
| 14:28.04 | [TK]D-Fender | exten => s,1,GotoIf([${CALLERID(num)}=112021081]?yes:no) <------ missing $ |
| 14:28.14 | ziz212 | @kaldemar http://pastebin.com/1Wm7RNXi |
| 14:28.15 | [TK]D-Fender | ziz212, fix your expressions in there |
| 14:28.47 | [TK]D-Fender | ziz212, and your Goto has BOTH sides leading to a HANGUP. the call is OVER. Dead. |
| 14:29.05 | kaldemar | ziz212: https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 14:29.38 | [TK]D-Fender | kaldemar, Not what he's looking for...I think he's trying to make a "call me back so I can use free incoming calls" type scenario |
| 14:29.38 | ziz212 | yes what i need is to terminate the call from pbx and generate a new call to the same number |
| 14:29.54 | [TK]D-Fender | ziz212, Totally wrong way to do this |
| 14:30.11 | [TK]D-Fender | ziz212, you need to Originate() a completely new call. |
| 14:30.34 | ziz212 | once the call is hangup then no process is there |
| 14:31.01 | kaldemar | [TK]D-Fender: which can be done using a hangup extension. app Originate is a pain because it does not support retries. |
| 14:31.13 | ziz212 | it is giving "Spawn extension (test, s, 5) exited non-zero on 'SIP/SIP-2-out-00000085'" |
| 14:31.21 | vassilux | Sorry , I made a mistake in my inbound route |
| 14:31.37 | [TK]D-Fender | ziz212, Actually... that alone won't do it right because it locks the current call. You'll need to use a "call file", use System to call * CLI "Originate" (not the dialplan app), or issue an AMI Originate. All are documented in THE BOOK |
| 14:31.37 | Assid | wow.. i think it started working |
| 14:31.38 | [TK]D-Fender | ~book |
| 14:31.38 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:31.41 | [TK]D-Fender | ^^^ |
| 14:31.45 | Assid | i had to setup externhost and localnet |
| 14:31.45 | kaldemar | [TK]D-Fender: if the call comes when the caller is still listening to the wonderful tone of congestion, the call back will fail. |
| 14:31.53 | Assid | not sure if thats what did the job |
| 14:32.00 | [TK]D-Fender | kaldemar, That's the "blocking' part I was mentioning... |
| 14:32.12 | [TK]D-Fender | kaldemar, Well aware :) |
| 14:32.19 | ziz212 | let me know the book |
| 14:32.28 | [TK]D-Fender | ziz212, I already linked it |
| 14:32.32 | ziz212 | ok |
| 14:32.33 | [TK]D-Fender | ziz212, look up |
| 14:32.35 | ziz212 | thnks |
| 14:32.57 | [TK]D-Fender | Assid, You were giving them the wrong ip's server side then... |
| 14:33.08 | ziz212 | thanks for the help guys .... |
| 14:33.46 | Assid | [TK]D-Fender any idea what this means ? [Feb 17 20:03:16] WARNING[31218]: chan_sip.c:3389 __sip_xmit: sip_xmit of 0xa4986c0 (len 733) to x.x.x.x:37526 returned -2: Interrupted system call |
| 14:33.53 | [TK]D-Fender | EEK |
| 14:33.58 | [TK]D-Fender | Nope... |
| 14:34.47 | *** join/#asterisk mpe_ (~mpe@office.ipvision.dk) |
| 14:34.58 | Assid | anyone managed to use openvpn + sip ? |
| 14:35.47 | [TK]D-Fender | Assid, I don't see why it shouldn't work... |
| 14:35.58 | Assid | man i need to call someone to confirm this works right... |
| 14:36.12 | Assid | [TK]D-Fender openvpn seems to slightly lag.. |
| 14:36.29 | [TK]D-Fender | Assid, Well.. yeah there is overhead, so I supose that could happen.. |
| 14:36.46 | Assid | specially when i use it for my terminal services |
| 14:41.34 | Assid | interesting.. if i use GSM codec, sipdroid crashes.. the call does nothing.. and wham.. dead |
| 14:47.36 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 14:49.17 | *** join/#asterisk SteveWilliams (~chatzilla@121.246.188.11) |
| 14:49.54 | Assid | anyone know a free sip provider that lets you send all your toll free traffic tyhrough |
| 14:50.07 | Assid | rapidvox is charging me for 800 calls also |
| 14:51.02 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 14:52.18 | [TK]D-Fender | Assid, I heard that IPKALL did this from someone recently, but never tested it personally |
| 14:52.39 | Assid | i thought ipkall is incoming only |
| 14:53.52 | Defraz | can you specify a range with sip notify? |
| 14:56.12 | *** join/#asterisk myyrdin (~textual@74-93-34-252-Sarasota.hfc.comcastbusiness.net) |
| 14:57.40 | [TK]D-Fender | Assid, So did I, but apparently they do TF outbound too... |
| 14:58.09 | [TK]D-Fender | Assid, which because they can charge their fraction of a cent for crossing their network onward it pays for it |
| 14:59.54 | zkn | ummÂ…how to invoke the list of columns in CLI required by realtime integration? when I do for example module unload/reload then nothing happens but as I understand I should get a list of warnings about missing columns.. |
| 15:01.29 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:01.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:02.55 | zkn | or is there some other/better way finding out what columns a module will try to access in a realtime table? |
| 15:04.20 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 15:17.55 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 15:19.40 | wdoekes2 | zkn: unfortunately the best resource is the source |
| 15:19.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:19.57 | zkn | okayÂ…the .c files then, yes? |
| 15:20.04 | wdoekes2 | correct |
| 15:21.02 | zkn | although the "book" explains in dynamic realtime example that the console should output stuff like WARNING: Realtime table ast_sipfriends@asterisk requires column 'name', but that column does not exist! |
| 15:21.52 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.153.86) |
| 15:21.56 | zkn | :s |
| 15:24.21 | wdoekes2 | that may depend on your logger.conf settings |
| 15:25.05 | zkn | aha a..okay, let's have a look.. |
| 15:25.14 | wdoekes2 | the chan_sip sippeer realtime dynamic table does indeed produce such warnings, at startup |
| 15:25.52 | wdoekes2 | (are you starting asterisk with -c? if not, you should look at your logs)' |
| 15:27.38 | zkn | right now i'm starting with -vvvvvvvvvvvvc or smth equivalent |
| 15:27.54 | *** join/#asterisk rjvvliet (~rjvvliet@217.21.249.170) |
| 15:28.53 | kaldemar | zkn: you'll find some table structures in contrib/realtime in the source package. |
| 15:29.23 | zkn | kaldemar, thanks, i'll have a look |
| 15:32.56 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:34.03 | joshaidan | Quick question. Would I get better performance from Asterisk if forking is enabled? |
| 15:54.46 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:54.46 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:08.43 | *** join/#asterisk myyrdin (~textual@74-93-34-252-Sarasota.hfc.comcastbusiness.net) |
| 16:11.10 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:12.49 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
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| 16:14.15 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 16:14.43 | fornax | I would like to start / stop a system service using my cisco scup phone by using a speed dial button and let the corresponding light indicate if it is started or stopped. Does someone have an idea how to do this? Maybe I could create an extension that toggles the service (/etc/init.d/hostapd) but how to do this the best way? |
| 16:17.21 | [TK]D-Fender | fornax, "core show application System" |
| 16:17.41 | [TK]D-Fender | fornax, "core show function DEVICE_STATE" |
| 16:18.58 | fornax | I'm not sure if I expained my question correctly. I have a service that enables my local WLAN, it is /etc/init.d/hostapd and I want to start and stop the service using my telephone |
| 16:19.34 | [TK]D-Fender | fornax, You did. And you would call System() to execute your script to do that |
| 16:20.53 | fornax | okay, I will have a look. And is it possible to set the light for the light in some way for my cisco phone? I want to enable / disable my WLAN using a speed dial button |
| 16:21.12 | [TK]D-Fender | fornax, that is the OTHER function I pointed you to the directions for |
| 16:21.27 | [TK]D-Fender | ? book |
| 16:21.58 | [TK]D-Fender | fornax, set a custom device state for the ON/OFF as you toggle through and execute your script to start it up,/bring it down |
| 16:22.01 | [TK]D-Fender | ~book |
| 16:22.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:22.03 | [TK]D-Fender | ^^^ |
| 16:23.47 | fornax | I have the book from Wintermeyer and the Oreilly and work with it for several years with Asterisk but I asked me if it would be possible with a cisco over sccp but I think this should be. now the next question is if it also works without picking up the phone |
| 16:23.57 | tzafrir_laptop | malcolmd, any issue in the bug tracker about http://forums.digium.com/viewtopic.php?f=1&t=76909 ? |
| 16:24.22 | tzafrir_laptop | See also https://issues.asterisk.org/jira/browse/ASTERISK-17310 |
| 16:25.01 | [TK]D-Fender | fornax, "without picking up the phone"? that would be with an XML script then as you're now stating that you're using Cisco phones. |
| 16:25.33 | *** join/#asterisk atan3 (~atan@unaffiliated/atan) |
| 16:26.41 | fornax | [TK]D-Fender: Okay, so this works too? Good to know. I have had the System or AGI approach in my mind but when this can be directly using the XML config, this would be far better. I googled for some time but did not find a hint where to look for such a feature. Do you have an idea what I can search for? |
| 16:27.07 | atan3 | How would one go about setting up their systems so it was high availability but also could handle a call continuously during the downtime? Is this even possible? Idea being when one server blows up the next one just continues on without dropping any activce calls. |
| 16:27.14 | [TK]D-Fender | fornax, You dno't use System at all with this new idea. You aren't calling from the phone |
| 16:27.25 | [TK]D-Fender | fornax, same with AGI |
| 16:27.54 | *** join/#asterisk Russ (~russ@108.60.121.130) |
| 16:27.55 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-sttrfqzjridorgyr) |
| 16:28.23 | [TK]D-Fender | fornax, XML = web script = nothing to do with *. For the light... you can ORIGINATE or use a CALL FILE to trigger a Local channel to do the dialplan to flip your custom devicestate flag if you'd like |
| 16:29.31 | *** join/#asterisk dandate2 (~dan@222.127.49.142) |
| 16:29.49 | dandate2 | is it possible to tftp flash old cisco ip phones from a router with dd-wrt rather than a dedicated machine? |
| 16:30.19 | fornax | [TK]D-Fender. Yes, sure these are two approaches, I was searching the cisco sites and the web for an approach using the cisco XML files but did not find anything so thought about a fallback with System / AGI but when you say it works with the first solution this is the best…. so do I need to rewrite the xml file or can i call a script from the xml file or what is the way, sorry I'm a newbie to these cisco configs and happy that I got my phones to work |
| 16:30.40 | [TK]D-Fender | dandate2, TFTP is TFTP. |
| 16:30.47 | atan3 | One would reason since dd-wrt allows you to connect to it and run software one could run a little tftp on it. O_O |
| 16:30.57 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 16:31.07 | [TK]D-Fender | fornax, I did not say the XML provisioning files. |
| 16:31.28 | [TK]D-Fender | fornax, I'm talking about the XML BROWSER on the higher end models |
| 16:32.06 | fornax | [TK]D-Fender: Oh, okay, so do you maybe have a link with information about that? I have a CISCO 7970 with two 7914 and want to use one button on my second 7914 |
| 16:32.22 | [TK]D-Fender | fornax, www.cisco.com |
| 16:35.08 | fornax | [TK]D-Fender: I search cisco.com with a friend for 3 days but they have so much information text on their sites that you are unable to find anything when you have no good keyword or idea where you are searching for. I just don't want to waste your time with questions that I can read somewhere on my own but I really have the problem that I have no real starting point. And I do not understand how I can use the XML Browser to set a hardware speed dial... |
| 16:37.06 | [TK]D-Fender | fornax, It's their docs. Has nothing to Asterisk... |
| 16:37.28 | [TK]D-Fender | fornax, Go get the programming manual for the XML browser |
| 16:37.53 | rjvvliet | fornax: http://lmgtfy.com/?q=cisco+7970+XML+browser |
| 16:38.58 | fornax | Ok, this indeed helped me, thank you very much, I will start reading and hope I will find a solution. Thank you very much. |
| 16:43.47 | *** join/#asterisk vale-wrk (~vale@i-83-67-17-227.freedom2surf.net) |
| 16:44.19 | vale-wrk | hey there guys |
| 16:44.39 | vale-wrk | I am having an issue starting the vboxdrv deamon for virtual box at boot on my system |
| 16:45.01 | vale-wrk | i have added it to the rc.conf file in the daemons section but it isnt starting |
| 16:45.21 | vale-wrk | virtualbox runs fine if I manually start it using modprobe however |
| 16:45.30 | vale-wrk | anyone had this problem? |
| 16:46.17 | kaldemar | vale-wrk: is this really the place to ask? |
| 16:46.29 | vale-wrk | your right |
| 16:46.37 | vale-wrk | i have totally just posted in the wrong channel |
| 16:46.38 | vale-wrk | my bad |
| 16:46.45 | vale-wrk | its a friday can you let me off :3 |
| 16:46.55 | kaldemar | no problem. :) |
| 16:47.09 | anonymouz666 | vale-wrk: tomorrow starts the carnival, so don't worry |
| 16:47.27 | vale-wrk | phew :P |
| 16:47.30 | anonymouz666 | everybody should be tripping right now |
| 16:49.15 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 17:00.54 | *** part/#asterisk ILIyahoo (~Iliya@84.94.76.109.cable.012.net.il) |
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| 17:38.50 | adeel|work | is there a logging channel for recording only the SIP history for a particular call? |
| 17:40.29 | leifmadsen | there is not |
| 17:40.45 | adeel|work | that would probably a pretty sweet feature |
| 17:40.55 | leifmadsen | sure would; patches welcome |
| 17:41.03 | adeel|work | leifmadsen, heh, if i knew how, i would =p |
| 17:43.18 | adeel|work | does * even maintain a transaction history in memory for a particular call id? |
| 17:44.40 | *** part/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
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| 17:52.31 | pigpen | I am asking this as I am 99.9% sure of the answer. It is that .1% I want to clarify. Is T.38 possible on IAX? |
| 17:55.04 | Qwell | pigpen: no |
| 17:55.21 | p3nguin | It doesn't even work for me in SIP. :( |
| 17:55.29 | Qwell | p3nguin: you're doing it wrong then :p |
| 17:56.03 | pigpen | Qwell, tks, what I thought. |
| 17:56.05 | Qwell | pigpen: to be more specific - T.38 is defined as being for use with SDP |
| 17:56.27 | Qwell | so in theory, it could be used with anything that uses RTP |
| 17:57.20 | Qwell | (disclaimer: blah blah udptl, blah blah) |
| 17:57.46 | pigpen | thanks! get busy on it! |
| 17:57.47 | pigpen | haha |
| 17:59.47 | adeel|work | for 'sip show history' is the call-id, the call-id in the 'sip show channels' ? |
| 18:00.09 | *** join/#asterisk FilipeM (~FilipeM@81.84.247.129) |
| 18:00.23 | pigpen | yeah, my business partner was inquiring why not IAX. He had many valid points of why it "could" work. So he is partially right. |
| 18:00.32 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-prudbmvhyqegekbl) |
| 18:01.08 | p3nguin | How is he right in even the tiniest way? |
| 18:01.24 | FilipeM | Good day. Anyone knows how to use the Alcatel-Lucent OmniPcx PBX? I'm not sure if it runs Asterisk. Thank you in advance. |
| 18:01.39 | pigpen | i.e., possible, but it isn't written into it. |
| 18:01.58 | p3nguin | There's SDP in IAX2? |
| 18:02.14 | pigpen | I thought IAX uses RTP? |
| 18:02.40 | pigpen | the theory part. not functional. |
| 18:03.05 | pigpen | p3nguin, and yes, I don't have a clue of what I am talking about. |
| 18:03.08 | Qwell | pigpen: no it does not |
| 18:03.17 | p3nguin | IAX does not use RTP at all. |
| 18:03.22 | pigpen | ah?yeah, see my note above. |
| 18:03.24 | Qwell | IAX2 is both the control and audio transport |
| 18:04.07 | pigpen | ah?.you know, I have used IAX for years and never new it did both. |
| 18:04.20 | Qwell | that's why it's only 1 port.. |
| 18:04.21 | p3nguin | That's why it only uses a single port. |
| 18:04.31 | pigpen | echo, echo... |
| 18:05.01 | FilipeM | Hi all! Could you assist me with a simple question? |
| 18:05.25 | p3nguin | Before we get too deep into it, let me make sure I understand SDP. SIP is to a phone call setup/tear-down as SDP is to RTP. Correct? SDP is RTP's setup? |
| 18:05.36 | pigpen | since I have both of your attention, any thoughts on 10.1.x stability? I am needing the enhanced T.38 functionality. |
| 18:06.00 | p3nguin | I only use LTS releases. |
| 18:06.32 | Qwell | p3nguin: sort of? |
| 18:07.18 | *** part/#asterisk FilipeM (~FilipeM@81.84.247.129) |
| 18:07.24 | pigpen | For the most part we keep to the "almost bleeding edge", but this one box is a different story?.it is still on 1.6.1 and running very stable?.and it has a decent amount of volume. |
| 18:07.38 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 18:07.58 | pigpen | You know, I need to say more stupid shit more often?..as I really learn allot. |
| 18:09.24 | Qwell | p3nguin: See 3.1. It might help to understand. http://tools.ietf.org/html/rfc4566 |
| 18:09.24 | *** join/#asterisk ILIyahoo (~Iliya@84.94.76.109.cable.012.net.il) |
| 18:11.12 | myyrdin | root |
| 18:11.19 | Qwell | Password: |
| 18:11.38 | pigpen | heh... |
| 18:11.43 | pigpen | I am the king of doing that. |
| 18:11.45 | p3nguin | Okay, so SDP is a part of the control protocol, and it is used to agree on the media type. |
| 18:11.52 | pigpen | not the password part though |
| 18:12.05 | Qwell | p3nguin: well, SDP is it's own thing. SIP just uses SDP for that |
| 18:12.10 | Qwell | other things use SDP too |
| 18:12.12 | p3nguin | And its own thing, too. |
| 18:12.41 | p3nguin | That's why I said control protocol rather than locking into only SIP. |
| 18:13.04 | Qwell | p3nguin: it's not control - it just describes sessions |
| 18:13.18 | p3nguin | SIP isn't a call control protocol? |
| 18:13.38 | Qwell | SIP is control, yes |
| 18:13.45 | p3nguin | That's what I'm saying. |
| 18:13.54 | Qwell | You said SDP :p |
| 18:14.09 | Qwell | SDP is used by the control protocol |
| 18:14.17 | Qwell | but it's not part of it in any way |
| 18:14.32 | WIMPy | And SIP isn't call control, it's session control. |
| 18:14.43 | p3nguin | That's where I made the mistake -- I said part of when I should have said used by. |
| 18:14.47 | Qwell | WIMPy: yeah I avoided using the word "call" on purpose |
| 18:15.12 | p3nguin | I avoided the word for a while, but then somehow threw it in. |
| 18:15.39 | Qwell | the end result may be a call, but SIP doesn't really know/care what type of communication it is |
| 18:16.26 | pigpen | Qwell, hence "sip" video phones? |
| 18:16.43 | Qwell | pigpen: or tts or any arbitrary data |
| 18:16.56 | Qwell | or even no media at all |
| 18:16.59 | p3nguin | When I said a part of, I should have said used by: SDP is used by the control protocol, and it is used to agree on the media type. |
| 18:17.05 | WIMPy | Or some multiplayer game. |
| 18:17.15 | Qwell | p3nguin: right, much better |
| 18:17.45 | p3nguin | Concept understood, conveyance all fouled up. |
| 18:17.48 | pigpen | dam, it even makes sense to me. |
| 18:17.53 | WIMPy | (with multi=2 probably) |
| 18:18.52 | Qwell | WIMPy: client[]/server or even server[]/server[] |
| 18:19.24 | WIMPy | More like P2P, really, isn't it? |
| 18:19.50 | p3nguin | I've never bothered to look at a debug from my phones... should SDP appear in my SCCP packets in nearly the same way it does in SIP? |
| 18:20.13 | Qwell | WIMPy: bah! Just reinvite a 1-1 combat between players. :p |
| 18:20.25 | Qwell | they'll tell the server the winner when it's over |
| 18:20.30 | WIMPy | Why not :) |
| 18:20.39 | Qwell | p3nguin: no, skinny uses something magic |
| 18:21.08 | p3nguin | Maybe it agrees on RTP forcefully. |
| 18:21.13 | Qwell | pretty much |
| 18:21.25 | Qwell | SDP is just a way to describe something. It's not required by RTP, or anything. |
| 18:22.58 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 18:24.28 | grandpapadot | Hey guys, is there a way to generate a sip voicemail notify when not using app_voicemail? We have our own scripted voicemail system but it would be nice to get the notifications to our peers so mwi lights with flash, etc. I'm about to roll my own from app_voicemail.c but there may be an easier way ... using 1.8.current |
| 18:24.52 | grandpapadot | with = will |
| 18:33.24 | dijib | ok who wants this sangoma card/?!?!! |
| 18:34.41 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 18:34.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:34.49 | [TK]D-Fender | dijib, I've got 5$ burning a hole in my pocket right now! ;) |
| 18:35.09 | glaz | Can asterisk act as a full transparent proxy SIP server? |
| 18:36.26 | Qwell | glaz: Asterisk is not a proxy, no |
| 18:36.41 | glaz | any suggestion on an opensource solution? |
| 18:36.47 | Qwell | There are a bunch |
| 18:37.05 | dijib | $5 lol |
| 18:37.20 | glaz | I've tried OpenSIPs but it is very complicated to install and not very well supported |
| 18:37.20 | dijib | i would like to get at least a papt2-na for it |
| 18:38.16 | *** join/#asterisk coppice (~coppice@123203240234.ctinets.com) |
| 18:39.35 | leifmadsen | opensource sip proxy is going to end up being openser/opensips/kamillio/etc... (which operate in a similar manner) |
| 18:39.36 | *** join/#asterisk jasonwert (~w3rt@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
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| 18:54.05 | *** join/#asterisk mahlon (mahlon@martini.nu) |
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| 19:09.29 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-fyskhhwolmtlrebu) |
| 19:15.40 | WIMPy | Wow. No more delays. My provider must have updated my IAD to use overlap sending. |
| 19:15.55 | WIMPy | So far they only made things worse. |
| 19:29.39 | *** join/#asterisk fofware (~fabian@host96.190-31-217.telecom.net.ar) |
| 19:31.18 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 19:31.41 | WIMPy | So there is finally evidence of ATAs capable of doing it. |
| 19:37.59 | *** join/#asterisk [koss] (koss@adsl-75-33-65-8.dsl.bcvloh.sbcglobal.net) |
| 19:38.24 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-183.mobile.uci.edu) |
| 19:38.48 | [koss] | whats the easiest to get going free asterisk distro with auto provisoning? |
| 19:40.49 | Qwell | [koss]: You can easily install the SIP provisioning module on AsteriskNOW |
| 19:40.58 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:46.12 | glaz | leifmadsen: isn't OpenSIPs the continuity of OpenSER |
| 19:46.30 | leifmadsen | glaz: yes, that is my point |
| 19:46.41 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 19:48.43 | glaz | it's so complicated to setup for no reason |
| 19:50.06 | F2Knight | Q: XMPP a.k.a Jabber, is supposed to be an open source messaging platform( among other things) So I was thinking... What is the easiest way to use asterisk to send "IM's" to people? Specifically to support existing IM clients, AIM/AOL, MSN, ICQ, FACEBOOK and GTalk. Ideally I see doing a quick DB lookup to find out where the message should go for a specfic user. and then am hoping that the same method can be used to send messages to al |
| 19:50.06 | F2Knight | l the networks. Any input Ideas or suggestions? What I want to avoid is for the client to have to add yet another piece of software to there computer. |
| 19:53.20 | Qwell | F2Knight: How are you getting from XMPP to AIM/MSN/Facebook? |
| 19:53.25 | Qwell | that's quite a leap |
| 19:53.42 | F2Knight | Qwell, that was the basis of the question. |
| 19:53.58 | Qwell | so then why ask in #asterisk? O.o |
| 19:54.27 | F2Knight | because asterisk has the jabber client.. res_jabber.so |
| 19:54.43 | Qwell | Asterisk doesn't support AIM/MSN/Facebook |
| 19:54.44 | F2Knight | and already commuicates with gtalk. |
| 19:55.14 | F2Knight | but Jabber is supposed to be able to communicate with MSN, and AIM/AOL, Facebook |
| 19:55.49 | Qwell | AIM and MSN are not XMPP. |
| 19:56.02 | F2Knight | so by extension asterisk speaks Jabber, asterisk 'should' be able to speak with any jabber base, regardless of carrier. just a though.... |
| 19:56.50 | F2Knight | http://www.aim.com/xmpp |
| 19:56.59 | Qwell | Are you a "select partner"? |
| 19:57.18 | F2Knight | waiting for approval |
| 19:57.56 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 19:58.42 | F2Knight | http://windowsteamblog.com/windows_live/b/windowslive/archive/2011/12/14/anyone-can-build-a-windows-live-messenger-client-with-open-standards-access-via-xmpp.aspx |
| 19:59.31 | F2Knight | https://developers.facebook.com/docs/chat/ |
| 19:59.52 | Qwell | So connect to them as a client. What is your question? |
| 19:59.53 | F2Knight | Qwell, all of that indicates that all the major players in the IM world are quietly supporting XMPP |
| 20:00.42 | glaz | I'll give Kamailio a shot |
| 20:00.55 | F2Knight | I am not well versed with Jabber, so is it possible to use one Jabber account on asterisk.. lets say an existing gtalk account to push messages out to other networks. |
| 20:01.10 | Qwell | F2Knight: it's separate channels |
| 20:08.02 | *** join/#asterisk ILIyahoo (~Iliya@84.94.76.109.cable.012.net.il) |
| 20:19.24 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:28.58 | *** join/#asterisk Rebroad (~Rebroad@31.97.211.73) |
| 20:29.28 | *** join/#asterisk jkroon (~jkroon@dsl-244-31-102.telkomadsl.co.za) |
| 20:34.28 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 20:34.42 | fprior | Hi, what about free TTS service ? I've tested google from http://bit.ly/wPH1eD, another one to test ? |
| 20:35.54 | navaismo | Hi fprior, try Cepstral, Loquendo, IVONA |
| 20:36.07 | navaismo | non free sorry |
| 20:38.25 | fprior | navaismo, thanks |
| 20:40.45 | F2Knight | have you looked at http://emotion-research.net/publicnews/humaine_news_item.2006-02-14.9417432557/view |
| 20:41.31 | dijib | what to do with this sangoma, im sure i would consider a trade value of $40 |
| 20:42.34 | fprior | F2Knight, so good |
| 20:44.28 | dijib | navaismo: why is there no lumenvox free v? |
| 20:45.30 | navaismo | ?_? |
| 20:46.35 | navaismo | I just wrote that i know and im pay for it. more info in their pages |
| 20:47.30 | fprior | Festival, it's free |
| 20:49.30 | navaismo | yep |
| 20:49.52 | navaismo | i preffer google tts over festival |
| 20:56.21 | *** join/#asterisk jjulian (~jjulian@31-17-236-37-dynip.superkabel.de) |
| 20:59.30 | jjulian | hi. i have a question to asterisk (before i start into deep). we have a sip provider and a telephon system either an MSN on multi-terminal access or an extension number on a system access. Can i have a linux asterisk system which have a isdn card plugged into that telephone system? managin that stuff? |
| 21:00.56 | tzafrir_laptop | can you have? sure |
| 21:01.05 | tzafrir_laptop | managing that stuff? yes |
| 21:01.47 | tzafrir_laptop | Do you have any specific card? |
| 21:02.05 | jjulian | tzafrir_laptop: is the limit of 2 connections at a time through that telephone system then "killed"? |
| 21:02.33 | tzafrir_laptop | jjulian, that's not for asterisk to "manage". |
| 21:02.41 | jjulian | cause isdn only allows two connections at a time. |
| 21:02.48 | jjulian | we have an aastra opencome something :D |
| 21:03.30 | tzafrir_laptop | When the two lines are in use, the third call will be "busy" (well, at least with chan_dahdi. Also easy to emulate in a sip trunk to an adapter. I assume this is the same with chan_cap and chan_misdn) |
| 21:03.39 | jjulian | ive no idea if thats some kind of hard limit or if its a "oh we get an busy tone so wre not doing any more calls" |
| 21:04.07 | glaz | Anyone know a simple SIP Proxy software? I'm trying to replace a SBC device |
| 21:04.39 | tzafrir_laptop | One of SER's derivatives? OpenSIPS or Kamailio ? |
| 21:04.53 | glaz | both are very complicated |
| 21:05.06 | glaz | I tried both, can't find any simple SIP Proxy config examples |
| 21:05.15 | tzafrir_laptop | There's something called "siproxyd" |
| 21:05.30 | Qwell | jjulian: You can get multiple ISDN lines.. |
| 21:05.34 | glaz | http://siproxd.sourceforge.net/ |
| 21:06.33 | jjulian | tzafrir_laptop: what im trying to achieve is: we have an aastra opencom 120 or 130 and switch too sip, cause were now about 60 people in the office and we had 6 linues. awesome would be if we can hold that telephones and telephone system and every user could use his phone at the same time. but i guess this is a hard limit due to "cables" and how isdn works |
| 21:06.38 | glaz | I'm looking at it |
| 21:06.45 | tzafrir_laptop | http://siproxd.sourceforge.net/ , though I suspect this is too simple |
| 21:07.03 | *** join/#asterisk Gruub (~chatzilla@mail.citywholesaleelectric.com) |
| 21:07.24 | Gruub | Hi all. |
| 21:07.46 | tzafrir_laptop | jjulian, you have several ISDN (BRI) lines to your provider? |
| 21:08.06 | [TK]D-Fender | glaz, sipproxd |
| 21:08.07 | jjulian | three at the moment the system can handle 4 lines. |
| 21:08.10 | Gruub | I'm new to IP telephony, but from what I am reading about Asterisk, I can build my own Asterisk PBX machine from a linux box right? |
| 21:08.14 | [TK]D-Fender | Darn, sow again.. |
| 21:08.18 | Qwell | Gruub: sure |
| 21:08.19 | Qwell | ~book |
| 21:08.19 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:09.07 | Gruub | ok, and then if I install a specific trunk card, I ca use my existing analog lines as my outgoing copper right? |
| 21:09.16 | Qwell | sure |
| 21:09.44 | Qwell | Gruub: http://www.digium.com/en/products/hardware/ |
| 21:10.19 | Gruub | Digium is to Asterisk as Redhat is to Linux then? |
| 21:10.38 | Qwell | who currently employs Linus Torvalds? |
| 21:10.58 | Gruub | in the sense that Digium packages asterisk? |
| 21:11.16 | Gruub | Redhat packages Linux and supports for a service no? |
| 21:11.19 | Qwell | Digium is the maintainer and primary contributor. We also make hardware. |
| 21:11.34 | Gruub | oh ok |
| 21:11.50 | Qwell | pretty sure there's an "About" somewhere on that site |
| 21:11.59 | Qwell | http://www.digium.com/en/company/ |
| 21:15.37 | n3hxs | Qwell, Everyone who uses Linux, but they just don't pay him. |
| 21:16.11 | Qwell | n3hxs: I was going to compare whatever company he worked for (turns out, it's the Linux Foundation, which is a pretty apt comparison) |
| 21:16.33 | [TK]D-Fender | No .... |
| 21:16.37 | [TK]D-Fender | apt would be Debian ;) |
| 21:17.51 | n3hxs | LOL |
| 21:18.21 | *** join/#asterisk alexshell (~chatzilla@unaffiliated/alexshell) |
| 21:26.14 | [TK]D-Fender | checkout time, BBIAB |
| 21:38.22 | F2Knight | Qwell, doesn't he still work for transmittia.? |
| 21:38.31 | Qwell | F2Knight: he hasn't worked there in like 5 years.. |
| 21:39.04 | F2Knight | oh. um.. |
| 21:39.17 | F2Knight | well guess I am out dated on my linus info :) |
| 21:39.39 | *** join/#asterisk davlefou (~david@unaffiliated/davlefou) |
| 21:40.04 | F2Knight | His google+ page he list Software engineer for linux foundation, |
| 21:42.05 | *** join/#asterisk TimeRider (steve@02de05ed.bb.sky.com) |
| 21:46.38 | luckman212 | why does 'ps -C asterisk' list my running asterisk processes but 'ps -C rasterisk' shows nothing? 'ps ax | grep rasterisk' works but I wanted to avoid the |grep |
| 21:49.43 | wdoekes2 | luckman212: probably the difference between /proc/<pid>/exe and cmdline |
| 21:50.48 | wdoekes2 | luckman212: pgrep -f rasterisk |
| 21:51.44 | luckman212 | wdoekes2: I don't understand 'why' but pgrep -f rasterisk works |
| 21:53.17 | wdoekes2 | asterisk edits it's own **argv (cmdline), so the command line now reads rasterisk |
| 21:53.36 | wdoekes2 | the executable (named /usr/../asterisk) hasn't changed though |
| 21:53.48 | wdoekes2 | *its |
| 21:54.10 | luckman212 | ah... I see. so ps -C only matches on the /proc/<pid>/exe symlink |
| 21:54.22 | wdoekes2 | I don't know the internals of ps |
| 21:54.29 | luckman212 | which for 'rasterisk' is still /usr/sbin/asterisk |
| 21:54.37 | luckman212 | wdoekes2: that makes two of us |
| 21:54.40 | wdoekes2 | but, yes, that's the gist of it |
| 21:55.01 | luckman212 | pgrep is nice though, that's a neat tool |
| 21:55.09 | luckman212 | learn something new every day |
| 21:55.29 | wdoekes2 | try the -v verbose mode of pkill ;) |
| 21:55.47 | wdoekes2 | (don't) |
| 21:56.18 | luckman212 | wdoekes2: yeah not gonna do that |
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| 22:01.27 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:11.32 | pigpen | Is there anyone around that can explain the "buffers=x,immediate/half/full" setting in chan_dahdi.conf ? |
| 22:13.19 | *** join/#asterisk TimeRider (~steve@204.93.201.17) |
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| 22:34.50 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:42.09 | paulc | directmedia=no is a valid option, right? (it's not just yes/nonat/update?) |
| 22:44.25 | [TK]D-Fender | correct |
| 22:46.54 | paulc | doffs hat to [TK]D-Fender - thanks :-) |
| 22:53.23 | *** join/#asterisk NirS (~Nir@62.128.39.228.cable.012.net.il) |
| 22:53.33 | NirS | Good afternoon/Evening everybody |
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| 23:20.15 | NirS | Say, anyone had luck sending faxes with Fax For Asterisk ? |
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| 23:29.04 | *** join/#asterisk Scar-G (Scar.G@182.178.53.67) |
| 23:29.18 | Scar-G | Hello |
| 23:30.40 | Scar-G | Can someone please help me ? I need some help regarding Asterisk AMI. Need to PlayDTMF using Asterisk AMI ( Using Asterisk AMI because need to Play DTMF 'AFTER' a call is bridged ) |
| 23:30.56 | Scar-G | Is anyone here ? |
| 23:32.02 | Scar-G | aaaaeee |
| 23:33.34 | NirS | Hey Scar |
| 23:33.38 | Scar-G | Ah |
| 23:33.42 | Scar-G | hey |
| 23:33.53 | NirS | do you mean playing a Set of DTMF tones, once the call is answered ? |
| 23:34.13 | Scar-G | AFTER the call is bridged ( after audio communication starts ) |
| 23:34.39 | Scar-G | not by the D option as it plays after answer and BEFORE bridge |
| 23:39.08 | Scar-G | thanks anyway for at least replying |
| 23:40.45 | Scar-G | I have come here after a LOT of Googling my friend. |
| 23:44.25 | NirS | well |
| 23:44.30 | NirS | the answer isn't via the AMI |
| 23:44.31 | Scar-G | ? |
| 23:44.36 | Scar-G | :( |
| 23:44.37 | NirS | you need to do via the dialplan |
| 23:44.45 | Scar-G | ? |
| 23:44.54 | NirS | I guess that you are originating a call via the AMI, right ? |
| 23:44.59 | Scar-G | yes |
| 23:45.05 | NirS | well, the trick here is this |
| 23:45.11 | NirS | what you are probably doing is this |
| 23:45.37 | NirS | Originate a call to some SIP channel, then once answered, you direct it to a context, right ? |
| 23:45.57 | Scar-G | Not exactly |
| 23:46.04 | NirS | so, please elaborate |
| 23:46.21 | NirS | please detail the exact flow, so that I can fully understand it |
| 23:46.28 | Scar-G | ok. |
| 23:46.44 | Scar-G | I am using Asterisk as a trunk. |
| 23:46.56 | Scar-G | Call comes in and goes out. |
| 23:47.43 | Scar-G | I tried using the D option in the Dial command, and also with a Macro by using the M option with the dial command |
| 23:48.12 | Scar-G | but they send dtmf after the call is answered BUT 'before' the call is bridged |
| 23:49.12 | Scar-G | Using the G option, the call gets disconnected immediately after the context finishes. |
| 23:51.40 | NirS | hold on |
| 23:51.43 | NirS | be right with you |
| 23:51.49 | Scar-G | take your time |
| 23:58.37 | NirS | ok, I missed something here |
| 23:58.40 | Scar-G | http://forums.digium.com/viewtopic.php?f=13&t=79213&sid=18c50f0541b709ef018e8f6cada1999d#p160743 |
| 23:58.52 | NirS | the calls comes in on your trunk, Asterisk answers it, then what happens ? |
| 23:59.23 | Scar-G | let me post the asterisk -r CLI here |
| 23:59.55 | NirS | I'd prefer to see the dialplan |
