| 00:00.28 | lvlinux | you don't know? uhoh, you haven't been in here long then hehe :-D |
| 00:00.32 | lvlinux | ~book |
| 00:00.32 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:01.05 | ahklerner | no i am really new, thank you |
| 00:01.15 | lvlinux | well then you are doing great then... |
| 00:01.33 | lvlinux | yes check out that book---it's the best to start with, and probably the best * book overall. |
| 00:01.56 | lvlinux | you can skip the part at the beginning about installation, as obviously you have that covered. |
| 00:02.19 | lvlinux | but read all the dialplan basics part especially. |
| 00:02.47 | lvlinux | It's very well written and covers a lot of important stuff. |
| 00:03.18 | ahklerner | thank you very much |
| 00:03.32 | lvlinux | np. I'm not an * expert myself but getting there someday :-) |
| 00:05.45 | ahklerner | i was able to copypasta some codes yesterday and set up an extension that downloads and playback the weather forecast, i thought that was neat |
| 00:06.38 | lvlinux | hehe yeah that's cool. |
| 00:06.46 | lvlinux | ur doing this on a pi? |
| 00:06.57 | ahklerner | yeah |
| 00:07.05 | ahklerner | freaking amazing huh |
| 00:07.19 | ahklerner | it is to me anyway |
| 00:07.28 | lvlinux | yep i love those things |
| 00:08.11 | lvlinux | i have one set up in a church which i integrated with the security system---when the alarm is triggered, it emails and calls people and plays a prerecorded message telling them to check the CCTV system. |
| 00:08.59 | ahklerner | awesome |
| 00:09.00 | lvlinux | it's amazing we have those little things now---i started on a Tandy 1000TL with 640k of RAM... |
| 00:09.41 | ahklerner | i had a small white tandy with little tiny keys that did not have storage, had to type in the programs from the book every time |
| 00:09.51 | ahklerner | at least i think it was a tandy |
| 00:10.18 | ahklerner | that was like 30 years ago |
| 00:10.22 | lvlinux | Timex Sinclair maybe? |
| 00:10.44 | ahklerner | i am pretty sure it was a tandy i think my dad got at radio shack |
| 00:10.55 | ahklerner | maybe i am wrong on it being white but i think so |
| 00:11.03 | lvlinux | TRS-80? |
| 00:11.28 | lvlinux | did it have the screen on it or plug into external video? |
| 00:11.52 | ahklerner | http://oldcomputers.net/mc-10.html |
| 00:11.55 | ahklerner | looked like that |
| 00:12.23 | ahklerner | external plugged in to tv |
| 00:12.46 | ahklerner | we did not have the thing on the back |
| 00:12.52 | ahklerner | that is shown in that pic |
| 00:13.04 | lvlinux | cool yeah that's one of the TRS-80 or "Trash 80s" as they were called lol. |
| 00:13.23 | lvlinux | I hadn't seen that particular styled one |
| 00:13.51 | ahklerner | i have told the story wuite a few times and never found a pic of the one we had until now |
| 00:13.58 | ahklerner | funny |
| 00:14.08 | lvlinux | haha yep |
| 00:14.47 | lvlinux | yeah---4k of RAM on that one---yeah you have me beat :-) |
| 00:15.07 | lvlinux | still have a Sinclair and a Commodore 64 but never used either of them. |
| 00:15.38 | lvlinux | well good to chat w u and glad u got it working---i better get back to work here... |
| 00:19.08 | ahklerner | thanks for your help |
| 00:21.09 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
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| 00:27.58 | snadge | whats the default registration interval for a sip trunk? |
| 00:29.13 | snadge | thats kind of like how long is a piece of string.. but a customers pbx is sending options packets.. and no register for a while.. and they're in serbia.. so.. just curious i guess |
| 00:30.05 | snadge | kinda curious how that works.. does asterisk send a registration interval every n interval.. or does it only do it, if it thinks its currently unregistered |
| 00:30.52 | lvlinux | i think the default is 1hr but i could be wrong. you can set it to whatever you want on the register string in sip.conf |
| 00:31.15 | snadge | cool :) .. that means in theory, i might get another one within the hour.. thats enough for me to leave the packet trace open |
| 00:31.25 | snadge | and not bother to send an email or whatever |
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| 01:39.06 | ahklerner | well snadge did you get a registration packet |
| 01:39.38 | *** join/#asterisk diametric (~diametric@2604:3400:dc1:43:216:3eff:fe27:bf9d) |
| 01:40.27 | diametric | So it's been almost 10 years since I've played around with Asterisk. Whats a good SIP provider these days? I want to get back into this stuff. |
| 01:46.21 | ChannelZ | What country? |
| 01:46.56 | *** join/#asterisk gnudna (~gnudna@unaffiliated/sklav) |
| 01:47.42 | *** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net) |
| 01:47.53 | gnudna | Hi Guys |
| 01:48.29 | *** join/#asterisk saint_ (~saint@216.155.131.74) |
| 01:48.59 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:08.12 | diametric | ChannelZ: USA |
| 02:13.13 | fling | diametric: ipkall |
| 02:13.27 | fling | diametric: (free did for incoming calls) |
| 02:13.36 | diametric | oh nice. |
| 02:13.57 | fling | diametric: I use zadarma.com for outgoing to usa (free outgoing calls) |
| 02:14.23 | fling | diametric: you can set your zadarma caller id to be the ipkall number |
| 02:14.33 | fling | diametric: this is what I used few months ago :> |
| 02:14.36 | diametric | perfect |
| 02:17.24 | gnudna | can a did be forwarded to a sip uri in order to get free calls going? |
| 02:17.31 | gnudna | DID^ |
| 02:17.39 | fling | yes |
| 02:18.26 | gnudna | need to look at getting this setup for a greece or florida number |
| 02:18.49 | fling | gnudna: tell me then about your progress after a while ;> |
| 02:19.12 | gnudna | progress will be slow ;) im just starting to get the hang of asterisk |
| 02:19.20 | fling | ohh |
| 02:20.00 | gnudna | baby steps i just officially moved my DID to asterisk 2 weeks ago to reduce my phone bill from 50$ to maybe 4$ |
| 02:20.30 | fling | great! |
| 02:21.11 | gnudna | my next step is getting some # where i have family and friends so they can call those numbers to avoid charges |
| 02:22.02 | fling | gnudna: you can install asterisk on their routers and/or install softphones on their boxen |
| 02:22.18 | fling | gnudna: so they will call directly using your pbx network :> |
| 02:22.34 | gnudna | that means i need to support them |
| 02:22.40 | gnudna | i do that for a living |
| 02:23.03 | gnudna | quick search shows ipkall |
| 02:24.53 | *** join/#asterisk ahklerner (~nkruzan@unaffiliated/ahklerner) |
| 02:24.56 | gnudna | is IPKall anygood |
| 02:26.28 | fling | gnudna: try it |
| 02:26.39 | ahklerner | i recently signed up seems they only offered washington state or new york numbers |
| 02:26.56 | fling | ahklerner: are these numbers bad? ;> |
| 02:27.10 | gnudna | ahklerner i noticed that |
| 02:27.19 | gnudna | im looking for florida or california |
| 02:27.22 | *** join/#asterisk pii3 (~pii3@unaffiliated/pii3) |
| 02:27.23 | ahklerner | no but just sayin....he was looking for a place with numbers local to his peeps |
| 02:27.42 | ahklerner | what you CAN do though |
| 02:27.55 | ahklerner | is create new gmail accounts 1 for each |
| 02:28.06 | ahklerner | then go to google.com/voice |
| 02:28.12 | ahklerner | and get a number local to them |
| 02:28.26 | gnudna | they do not offer numbers in canada |
| 02:28.44 | gnudna | i tried a while back should check again in case they have changed it since |
| 02:28.46 | ahklerner | then with those addresses sign up with ipkall |
| 02:28.59 | ahklerner | and forward the google voice numbers to the ipkall numbers |
| 02:29.00 | fling | gnudna: http://zadarma.com/en/intertel/dirnum/?country=United%20States |
| 02:29.28 | fling | gnudna: http://zadarma.com/en/intertel/dirnum/?country=Canada |
| 02:29.49 | gnudna | i currently pay 1$ |
| 02:29.57 | fling | gnudna: which provider? |
| 02:29.58 | gnudna | actually 0.99$ |
| 02:30.01 | gnudna | voip.ms |
| 02:30.10 | fling | gnudna: ok, thanks :> |
| 02:30.26 | fling | ahklerner: why to forward google voice to ipkall? |
| 02:30.44 | ahklerner | so that his family has a local number to call..... |
| 02:31.11 | fling | ahklerner: can't google voice be forwarded directly to your pbx? |
| 02:31.24 | gnudna | yes but it is flaky at times |
| 02:31.29 | ahklerner | ? |
| 02:31.38 | fling | hmm hmmm |
| 02:31.40 | gnudna | i used to route long distance threw google voice but it was sketchy |
| 02:31.54 | gnudna | not bad for free |
| 02:32.04 | fling | I need to try google voice, never used it⦠|
| 02:32.16 | gnudna | to be honest but when the number on the display says 123-456-7890 |
| 02:32.23 | gnudna | people in general do not answer |
| 02:32.24 | gnudna | ;) |
| 02:32.36 | fling | what is 123-456-7890? |
| 02:32.44 | gnudna | nothing ;) |
| 02:32.48 | fling | doh! |
| 02:33.04 | ahklerner | when i make a call on google voice it displays the google voice number |
| 02:33.04 | fling | what are you talking about? :D |
| 02:33.35 | fling | not possible to set your caller id with google voice? |
| 02:33.44 | gnudna | ahklerner same for me but sometimes my friednw ould tell me it displayed either 0000000 or 123-xxx |
| 02:34.03 | ahklerner | hmm, |
| 02:34.12 | fling | one more reason to use zadarma.com for outgoing calls |
| 02:34.27 | gnudna | which is funny cause he told me about the service and then would never answer cause well he was being a prick |
| 02:34.59 | ahklerner | were you using the xmpp or whatever it is |
| 02:35.11 | gnudna | yes i believe i was |
| 02:35.20 | gnudna | let me check the config i still have it lying around |
| 02:35.31 | gnudna | pretty sure it was xmpp |
| 02:35.44 | ahklerner | i had read that was supposed to be shut off |
| 02:35.48 | fling | will you guys help me registering google voice account? I can't get it from where I am. The account needs to be registered from USA for google voice to work ;> |
| 02:35.49 | ahklerner | but i do not know |
| 02:36.01 | gnudna | well it stopped working one day so i guess it is |
| 02:36.14 | gnudna | i just assumed my asterisk upgrade broke it |
| 02:36.44 | ahklerner | do you have an ipad fling ? |
| 02:36.55 | fling | ahklerner: no, I'm not using proprietary things |
| 02:37.06 | fling | ahklerner: I have gta04 :D |
| 02:37.07 | ahklerner | gotcha |
| 02:37.34 | gnudna | i use my android when im out of town threw wifi |
| 02:37.49 | gnudna | no need to pay the providers ther ridiculous sums |
| 02:39.07 | ahklerner | well if you have a device and can download ericom access to go rdp |
| 02:39.21 | ahklerner | there is a cloud demo thing |
| 02:39.21 | fling | umm? |
| 02:39.29 | ahklerner | that is like a proxy |
| 02:40.01 | gnudna | weird i see my google options but missing is my login |
| 02:40.12 | gnudna | need too look into this again |
| 02:40.58 | gnudna | explains why it no longer worked |
| 02:41.00 | fling | ahklerner: what is the best way to call skype users and to receive calls from them? |
| 02:41.00 | gnudna | ;) |
| 02:41.10 | ahklerner | oh i have no idea |
| 02:41.13 | ahklerner | i am real noob |
| 02:41.20 | ahklerner | with asterisk |
| 02:41.22 | fling | siptosis gives me bad sound quality |
| 02:41.37 | fling | I'm in the middle of freeswitch + skypopen install |
| 02:41.43 | fling | skypopen fails to build |
| 02:42.07 | ahklerner | i am just happy to get free calls from google |
| 02:42.12 | fling | ahklerner: skype can forward incoming calls to a number hmm hmmm |
| 02:42.23 | gnudna | http://community.skype.com/t5/Skype-for-business/Connecting-Asterisk-SIP-PBX-to-Skype/td-p/8584 |
| 02:42.33 | gnudna | fling check that out if you haven't already |
| 02:42.35 | fling | gnudna: skype connect is not wat I want |
| 02:42.38 | fling | gnudna: too expensive |
| 02:42.39 | gnudna | might be a bit dated though |
| 02:44.10 | gnudna | fling the example shows how to register threw sip |
| 02:44.52 | fling | gnudna: and you pay $6/line/month |
| 02:45.23 | gnudna | ii figured you can setup the sip aspect with userid and register |
| 02:45.25 | fling | just to call skype users⦠too expensive |
| 02:45.33 | fling | umm? |
| 02:45.41 | gnudna | no need for a DID |
| 02:45.50 | fling | yes no need for a did I know |
| 02:45.51 | gnudna | free skype to skype unless that has changed |
| 02:46.22 | fling | skype <--- free ---> skype <--- $6/month ---> sip |
| 02:47.16 | gnudna | ah ok |
| 02:47.39 | gnudna | disregard more of a noob than both of you guys |
| 02:50.10 | ahklerner | i just started friday lol |
| 02:50.44 | ahklerner | i have outbound calls with google |
| 02:50.54 | ahklerner | well not technicly i guess |
| 02:51.06 | gnudna | hehe ok that is what i had done last year around may |
| 02:51.15 | gnudna | and i was happy for a long time |
| 02:51.27 | gnudna | then i decided to move my home phone to voip |
| 02:51.37 | gnudna | this was 3 weeks ago |
| 02:52.14 | ahklerner | ic, we did not have a home phone, i decided to try and set one up on a raspberry pi |
| 02:52.19 | gnudna | now im learning that half the how-to's on the internet were outdated and have been re-writing with lots of hand holding the extensions.conf |
| 02:52.34 | gnudna | im running asterisk in lxc |
| 02:52.38 | gnudna | ;) |
| 02:53.18 | ahklerner | cool |
| 02:54.20 | gnudna | actually asterisk and 10 other containers on a dual core pc |
| 02:54.55 | gnudna | life is good dreading the day this all crashes on me aka loosing a power supply MB |
| 02:55.05 | gnudna | data is on raid so not worried |
| 02:55.20 | gnudna | actually less worried ;) |
| 02:59.09 | snadge | ahklerner, i did get a registration packet.. but the result was auth failure.. so at that point, i just gave up .. emailed the customer the correct settings, told them they were unblocked.. and to contact us if they have any problems ;) |
| 02:59.32 | snadge | you cant trust serbians anyway |
| 02:59.41 | snadge | hehe |
| 03:00.02 | gnudna | i guess depends which part of the world your in |
| 03:01.55 | fling | ahklerner: I want to get one of these numbers -> https://support.skype.com/en/faq/FA24/how-do-i-dial-toll-free-numbers-for-example-1-800-1-866-and-1-877-number-series |
| 03:02.06 | fling | then forward skype incoming to the number |
| 03:02.10 | fling | and it should be free :> |
| 03:02.20 | gnudna | brb in a few |
| 03:20.58 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 03:21.42 | spicyramen_ | hi people anyone have worked with DTLS-SRTP to RTP calls with *? Call comes from a Kamailio Server |
| 03:21.56 | spicyramen_ | should this work ? |
| 03:22.47 | spicyramen_ | webrtc client -audio g711/dtls-> kamailio -> asterisk -g711/rtp-> Sip provider |
| 03:31.23 | *** join/#asterisk u0m3__ (~u0m3@92.80.68.194) |
| 03:32.22 | gnudna | later fling |
| 03:32.24 | gnudna | im out |
| 03:32.40 | fling | cya |
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| 03:35.57 | diametric | holy crap.. 10 years of not using asterisk, and coming back to it the config directory is ridiculous |
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| 03:46.32 | fling | Found it in my logs! -> http://markusgoebel.blogspot.ru/2008/03/free-bridge-from-skype-to-phone.html |
| 03:46.37 | *** join/#asterisk smellis_werk (~smellis@2001:470:c452:2:224:e8ff:fe45:ee53) |
| 03:46.52 | fling | I remember I have not figured out how to use voxeo two years ago :D |
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| 03:47.55 | Penguin | diametric: It has a few more features now. |
| 03:48.25 | diametric | serious understatement right there hehe |
| 03:49.40 | fling | this is what I will use -> http://www.kv.by/content/kak-raskulachit-skype |
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| 05:56.04 | fling | How to output peer ip address for a sip peers connecting to my default context? |
| 05:56.20 | fling | there are few bots trying to call 810972592338145 which is India or something |
| 05:57.25 | fling | Sorry it is a Japan number. |
| 05:58.13 | eirirs_ | that was a long number. |
| 05:58.18 | eirirs_ | Glad I'm in Norway. |
| 05:58.34 | fling | eirirs_: are you using short dids there? |
| 05:58.39 | eirirs_ | 8 |
| 05:58.55 | fling | nice |
| 05:59.28 | eirirs_ | 2 are for area, then 6 for the actual number |
| 05:59.31 | eirirs_ | heh |
| 06:01.37 | Penguin | Are you using asterisk 10 or higher? |
| 06:01.59 | fling | no, sip over nat stops working with any recent version |
| 06:02.09 | Penguin | So you're using 1.8? |
| 06:02.15 | fling | so I downgraded to 1.8 |
| 06:02.43 | fling | I'm about to try 12.5 after a while |
| 06:02.58 | Penguin | Okay, you'll have to allow anonymous calls (allowguest=yes) and then capture the calls with an extension that matches what they are calling. |
| 06:03.08 | fling | Penguin: already capturing |
| 06:03.16 | fling | and I see hostnames in tcpdump output |
| 06:03.17 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 06:03.24 | fling | s3.codejawa.co.5074: SIP, length: 899 |
| 06:03.24 | Penguin | In that extension, you can get their IP address using CHANNEL(peerip) or CHANNEL(recvip). |
| 06:03.35 | fling | great! thanks |
| 06:03.43 | fling | what about it's hostname? |
| 06:03.53 | Penguin | it is hostname what? |
| 06:04.05 | fling | Penguin: hostname of the calling bot? |
| 06:04.15 | Penguin | Who cares about the name. |
| 06:05.08 | fling | Penguin: thanks! gtg cya later |
| 06:05.34 | Penguin | Now what I do is create a log notice from dialplan and then have fail2ban match said notice string, banning the offending IP address with iptables. |
| 06:06.09 | Penguin | Using asterisk 10+, there's a logger level for that. |
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| 07:50.51 | makmak78 | Hi all. |
| 07:51.00 | makmak78 | could anybody help me with an annoying issue |
| 07:51.13 | makmak78 | 0x0000003cd773356f in __strlen_sse42 () from /lib64/libc.so.6 |
| 07:51.44 | makmak78 | im getting this 1-4 times a day. mostly when around 100 concurrent calls |
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| 07:52.39 | makmak78 | im using vmware esxi |
| 07:53.01 | makmak78 | i have checked hardware and increased ulimit |
| 07:53.15 | makmak78 | i have no idea where to look now |
| 07:53.31 | makmak78 | im grateful if anyone got any ideas regarding this |
| 07:53.43 | makmak78 | running asterisk 1.4.44 |
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| 08:31.19 | ChannelZ | it's probably dying of old age |
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| 09:03.44 | makmak78 | I would prefer an answer that i can do something with, though |
| 09:29.37 | fling | Is it possible to connect to Lync server using just client access? |
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| 09:50.52 | fling | I have no sound with linphone when calling 600 |
| 09:51.01 | fling | but I can hear linphone ringing |
| 09:51.13 | fling | and I hear dftm when click keypad |
| 09:51.23 | fling | btw mpd is playing fine |
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| 09:51.49 | fling | no nat, both asterisk and linphone addresses are in the same vpn network |
| 09:51.49 | andy09usa | hi |
| 09:51.53 | fling | andy09usa: Hello. |
| 09:51.59 | andy09usa | help pls |
| 09:52.02 | andy09usa | [2014-09-09 13:50:58] WARNING[3263]: chan_sip.c:3892 __sip_xmit: sip_xmit of 0xc98970 (len 392) to (null) returned -1: Invalid argument |
| 09:52.37 | andy09usa | d/not register all sip and other trunks :( |
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| 12:55.37 | ghost75 | somebody here uses ghostscript to convert pdf to tiff? I get crap quality as output |
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| 15:27.47 | Sgjunior | Hi, I am looking for a tool somewhat resembling chan_mobile.so but with mms capabilities. Has anyone encountered a solution to this? I am from Canada so I am very limited as far as providers go |
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| 15:56.48 | marceloamorim | guys, I wish to know when I do CLI> unload module, my optoins here are lots of modules and some " services" like dsp, plc, logger, indications and others |
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| 15:58.17 | marceloamorim | anyone knows if I set in my modules.conf autoload=no and loading just modules I wanted, how I suppose to load those services that doesn`t have .so at the end |
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| 16:05.37 | Sgjunior | Have you tried specifying them in modules.conf ex : chan_mobile.so load=no ( syntax should be in the template file or the /config dir of the source code) |
| 16:06.45 | marceloamorim | http://pastebin.com/RfsVCcwQ, this is all modules that I could unload from my asterisk, maybe I`ll put some lines explain all modules for my documentation , maybe this is good for you that is new like me. But it isn`t done yet, probably I`ll remove those load => ACL because it isn`t right yet |
| 16:07.24 | marceloamorim | I didn`t get yet Sgjunior |
| 16:08.24 | marceloamorim | oh, do you mean the sample files? |
| 16:09.32 | Sgjunior | ya sample files sorry |
| 16:11.41 | marceloamorim | if you get the indications for example, we could use CLI> module unload indications, but this " module " don`t have .so |
| 16:12.21 | marceloamorim | because of that, how could load this if I use the option autoload=no |
| 16:12.27 | Sgjunior | Here is mine, http://pastebin.com/rMJ7V7vi |
| 16:12.57 | marceloamorim | you use autoload yes and just unload all you don`t want |
| 16:13.10 | Sgjunior | Is that not what your are looking for? |
| 16:13.30 | marceloamorim | not exactly, I have some asterisk configured like that |
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| 16:13.57 | Raden | Katty, HI !!!!!! |
| 16:14.03 | Sgjunior | So you are trying to load modules at start and then unload them at a later moment? |
| 16:14.13 | Sgjunior | "services" |
| 16:14.50 | Raden | How would I go about enabling and disabling a attended menu by dialing like *200 ? |
| 16:15.07 | marceloamorim | well, I put load in all modules, but I put this just because I want the asterisk works the way it works now |
| 16:15.18 | marceloamorim | but I`ll probably test removing the load for unload |
| 16:15.34 | marceloamorim | at this moment there is 208modules up on my configuration |
| 16:15.56 | marceloamorim | so all .so files are ok |
| 16:16.05 | marceloamorim | the problem is those that don`t have .so |
| 16:16.12 | marceloamorim | like acl and indications for example |
| 16:16.18 | [TK]D-Fender | Raden: There is no concept of "enabling" Everything is dialplan flow |
| 16:16.24 | Sgjunior | Raden, it's pretty simple. Add a test against a flag inside a database to your dialplan. When you dial *200 it turns the flag off and the menu is unreachable Execif("SOME_odbf_func" = true/false |
| 16:17.14 | [TK]D-Fender | Raden: You would have to check for something to indicate your desired processing like Sgjunior suggested and change how you act accordingly |
| 16:17.37 | Sgjunior | Pretty sure I dont have .acl files in my system. I would have to double check. I'll tell you if I find anythingon that |
| 16:17.47 | [TK]D-Fender | Raden: Pick whatever method for storing that state you feel like: AstDB, ODBC, global variable, etc |
| 16:17.56 | Sgjunior | ^ |
| 16:18.11 | Raden | [TK]D-Fender, exactly what i was looking for thanks Sgjunior ... I want to be able to dial like *200 if our ISP is experiancing issues so there is a message ready to go so we done get 50+ voicemails a hour |
| 16:18.40 | jeev | [TK]D-Fender, you think tm1000 got your request abuot accepting calls in inbound routes? |
| 16:18.48 | [TK]D-Fender | Raden: Go make exten(s) to set/toggle that state, and go check for it in your dialplan |
| 16:19.05 | [TK]D-Fender | jeev: Highly doubt it. Go post a request yourself on the tracker |
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| 16:21.51 | Dovid | hi all |
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| 16:33.12 | jeev | k |
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| 17:58.40 | Katty | Raden: howdy (= sorry busy morning! |
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| 18:22.46 | lvlinux | Hello! |
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| 20:52.56 | marceloamorim | well guys, I finish this file, I thing is very helpful http://pastebin.com/pDF9dE3m |
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| 20:53.26 | ovoshlool | Hello. We want to emplementate queue with ARI at our service. Sowe use channels and bridge for it. But we need to call a queue member through Local channels.So At first we call Stasis to call Our app Queue. It creates a local channel, and when channel creates we Dial real device with SIP channel (We use Endpoint: to call this) . When device answers- it creates real channel that sended to us from another Stasis (from macro) and w |
| 20:53.43 | marceloamorim | ops |
| 20:53.48 | ovoshlool | So we have problem, couse when device called - it is just ringing one time... |
| 20:53.50 | marceloamorim | autonoload=yes is wrong, sorry =( |
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| 20:54.56 | marceloamorim | "autoload=no" if you want to fix |
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| 21:37.40 | newtonr | marceloamorim, what was your goal with the specific configuration you have in that modules.conf? |
| 21:39.21 | newtonr | ovoshlool, you might write up your question and include debug in a mail to the asterisk-app-dev mailing list http://lists.digium.com/mailman/listinfo/asterisk-app-dev |
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| 21:52.02 | mjordan | ovoshlool: what newtonr said. We've tested that scenario and not run into anything close to that, so there is probably something specific to your app/dialplan. |
| 21:55.07 | marceloamorim | well, actually I just wanna know what modules my asterisk is running, and I didn`t find another good documentation inside the docs, so I set those modules |
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| 21:56.36 | newtonr | marceloamorim, Were you just looking for a one sentence description of each module, like the annotations you created? |
| 21:57.11 | marceloamorim | kind of, I get those information at this http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-1.html |
| 21:58.08 | newtonr | marceloamorim, did you find the "module show like" CLI command? |
| 21:58.24 | newtonr | marceloamorim, that will show you what all modules are running. |
| 21:59.51 | marceloamorim | well, actually when I use this command I get just Usage: module show [like keyword] Shows Asterisk modules currently in use, and usage statistics. |
| 21:59.54 | newtonr | "module show" will give you a list of them all |
| 22:00.11 | marceloamorim | nice |
| 22:00.15 | marceloamorim | I didn`t know that |
| 22:02.09 | ovoshlool | @mjordan Before I do debug . Now I tested another scenario: /channels?endpoint=SIP/device/sip:device@my_provider.com&app=my_cool_app And Get 500 Error. I hane kamailio as auth modue (allc clients registered on it, every call goues to kamailio and koming from it). So hov I must call analog of Dial(SIP/device/trunk) From ARI ? |
| 22:02.35 | mjordan | 500 is going to be something else, generally being that you gave it something very invalid and it couldn't figure it out. |
| 22:03.15 | mjordan | The endpoint query parameter in that URL should be a valid dial string, so I'd make sure that you can dial that using dialplan first |
| 22:03.31 | newtonr | marceloamorim, sorry it was hard to find. The wiki doesn't really point it out anywhere that I can see. I'll update the wiki to make it more clear on how to list all the running and loaded modules. |
| 22:03.33 | mjordan | And hopefully, you'll get an ERROR or WARNING on the CLI when you get a 500 |
| 22:04.21 | mjordan | newtonr: The module page could use that, but eventually we should have some other task to take the CLI commands and put them on the wiki (somehow) |
| 22:04.35 | marceloamorim | well man, its awesome anyway |
| 22:05.18 | mjordan | ovoshlool: If your normal dial string is SIP/device/trunk, then you should use whatever you used in the Dial application. |
| 22:05.31 | newtonr | mjordan, agreed. |
| 22:05.51 | mjordan | newtonr: avoid the slippery slope of copy+paste all CLI commands! :-) |
| 22:06.11 | newtonr | yeah... that would be ineffective :D |
| 22:06.58 | marceloamorim | thx for the info and I`ll keep that file if someday anyone need that, because I test my modules substitute all load for noload and set autoload yes, so we can find the modules that we didn`t set manually |
| 22:07.15 | marceloamorim | I got go now, gnight, and thx for the tips |
| 22:08.13 | newtonr | marceloamorim, no problem |
| 22:09.10 | ovoshlool | Yes. I called devices as DIAL(SIP/device/trunk)... And I do nothing at CLI about 500... only ARI response. I will try now POST /channels?endpoint=SIP/device/trunk&app=my_cool_app |
| 22:09.37 | mjordan | ovoshlool: if you can get a DEBUG log with the 500 error, that'd be appreciated. I'd like to at least get an ERROR or WARNING when that happens. |
| 22:10.09 | mjordan | feel free to e-mail it to the app-dev list, or file an issue at issues.asterisk.org. The 500 is probably "okay", in that you're sending something it doesn't like, but Asterisk should tell you why |
| 22:10.11 | ovoshlool | post here? json response |
| 22:10.18 | mjordan | pastebin |
| 22:10.23 | mjordan | ~pb |
| 22:10.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 22:11.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@kobratel.starlink.ru) |
| 22:15.57 | ovoshlool | ~pb http://pastebin.com/zVXdNUi4 |
| 22:16.01 | lvlinux | question: Is there a debug mode that will show dtmf being received by and transmitted by Asterisk??? |
| 22:17.15 | ovoshlool | lvlinux SIP Info packets may show you dtmf. |
| 22:17.42 | ovoshlool | lvlinux if you defined dtmf mode first |
| 22:20.01 | lvlinux | you mean if i set the dtmf to SIP info? I'm mainly interested in rfc2833 |
| 22:24.35 | mjordan | Hm. |
| 22:26.06 | mjordan | ovoshlool: What is the SIP peer that you are trying to dial through? |
| 22:26.43 | ovoshlool | lvlinux usually dtmf displayed at cli not depending at dtmf mode, but if you need debug you should experement with dtmfmode param |
| 22:26.45 | mjordan | also, if you aren't creating a Local channel, you really don't need the otherChannelId |
| 22:26.46 | newtonr | lvlinux, there is a DTMF logger channel |
| 22:27.00 | newtonr | lvlinux, search for dtmf in logger.conf |
| 22:27.09 | *** join/#asterisk [Outcast] (~outcast@ip-140-186-77-28.packetsurge.net) |
| 22:27.26 | newtonr | lvlinux, the sample logger.conf of course |
| 22:30.04 | *** join/#asterisk areski (~areski@80.174.128.36.dyn.user.ono.com) |
| 22:30.28 | ovoshlool | mjordan SIP peer that I trying to dial is device at kamailio/ My kamailio used as trunk for asterisk/ So scheme off call this : Client1 -> Kamailio -> Asterisk(stasis)-> Kamailio -> Client2. Clients registered at kamailio |
| 22:30.44 | ovoshlool | Yes/ I forgot to delete ItherChannelId |
| 22:30.51 | ovoshlool | Now testiong without |
| 22:33.29 | lvlinux | newtonr: k i'll check that, thanks |
| 22:33.39 | lvlinux | ovoshlool: thanks as well |
| 22:34.38 | ovoshlool | @mjordan without OtherChannelId same result... |
| 22:35.10 | mjordan | I suspect it is the dial string. What happens if you Dial using that same string? |
| 22:37.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:38.43 | ovoshlool | @mjordan If I dial it from dialplan... Oh. My bat attension... I usuallu Dial (SIP/trunk/device) Net SIP/device/Trunk... To testing with right string.. Sorry. 1 minute please |
| 22:41.27 | ovoshlool | @mjordan With this parameters as i wrote above I see at CLI calling trunk/device! And see 200 status of response. But device not called/ Going to see TCPDUMP |
| 22:41.36 | ovoshlool | thanks |
| 22:42.19 | mjordan | np. Just think of the endpoint query parameter as the dial string (as that is what it is) |
| 22:48.07 | *** join/#asterisk LemensTS (~Matt@adsl-70-238-130-27.dsl.stlsmo.sbcglobal.net) |
| 22:49.05 | LemensTS | hello, i have a itsp im connected to in sip.conf with a host=123.123.123.123 They are changing that ip address soon, can I add both of them in there and it be a smooth transition? host=123.123.123.123, 444.444.444.444 |
| 22:49.19 | [TK]D-Fender | no |
| 22:49.23 | [TK]D-Fender | 2 peers required |
| 22:49.41 | LemensTS | Ok thanks {TK]D-Fender |
| 22:50.01 | LemensTS | They have to have different context names? |
| 22:51.11 | [TK]D-Fender | yes |
| 22:52.29 | *** join/#asterisk navaismo (0c0adbe5@gateway/web/freenode/ip.12.10.219.229) |
| 22:56.35 | *** part/#asterisk LemensTS (~Matt@adsl-70-238-130-27.dsl.stlsmo.sbcglobal.net) |
| 23:00.56 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 23:09.04 | lvlinux | Hey [TK]D-Fender: why would he need different contexts? Why wouldn't it work with both peers going to the same context? |
| 23:09.36 | [TK]D-Fender | lvlinux: He was meaning the DEVICE name for the section in sip.conf |
| 23:23.24 | *** join/#asterisk [Outcast] (~outcast@64.206.121.41) |
| 23:27.01 | *** join/#asterisk Defraz (~Defraz@205.185.92.236) |
| 23:45.05 | lvlinux | [TK]D-Fender: ah ok yes then you couldn't put two of the same. thanks for clarifying :) |
| 23:46.35 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 23:47.17 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |