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| 01:53.15 | aarobc | Hello, so I'm just getting started with asterisk, and I'm trying to figure out the best approach for allowing sip connections for phones while still maintining security. I've done a bit of googling, but there seems to be no clear answer. Can anyone point me to some documentation? |
| 01:55.21 | Katty | arapaho: you could use a VPN connection |
| 01:55.41 | igcewieling1 | define "maintaining security" |
| 01:56.11 | Katty | arapaho: please disregard, message intended for aarobc |
| 01:56.24 | Katty | aarobc: or you could use a firewall to limit it from certain public static IP addresses |
| 01:58.28 | aarobc | Katty: Is that what is typically done? |
| 02:02.16 | igcewieling1 | aarobc: depends on your needs. for example we block all SIP access from outside our network address block. |
| 02:02.22 | igcewieling1 | This is not workable for most people. |
| 02:02.45 | *** join/#asterisk andrewyager (~andrewyag@114.141.105.250) |
| 02:03.11 | igcewieling1 | Others leave SIP open to anywhee, but use fail2ban to block people trying to hack in |
| 02:03.52 | igcewieling1 | Others use a combination of the two |
| 02:04.10 | igcewieling1 | You should also read The Book |
| 02:04.12 | igcewieling1 | ~book |
| 02:04.12 | 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 |
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| 03:06.01 | mae_tae | hello people |
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| 03:29.39 | ChannelZ | people hi |
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| 06:36.11 | BeeBuu | hello,all |
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| 07:00.06 | BeeBuu | i get a problem following: my asterisk server had a sip register,and i use PC using another sip user on the asterisk and call out,the callee can get voice,but caller get sliense,anyone help me please.... |
| 07:09.46 | hebber | Beebuu: codecs or NAT problems I assume |
| 07:14.49 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 07:16.01 | ChannelZ | Probably the callee's RTP either isn't making it off their network/through their firewall, or isn't making it through yours. |
| 07:16.33 | BeeBuu | Oh... |
| 07:17.15 | ChannelZ | but there are other possibilities, not knowing anything about your config. IS your Asterisk behind a NAT firewall? |
| 07:19.22 | floren | hi all |
| 07:19.42 | floren | wokeup early... 3.19am here |
| 07:23.16 | ChannelZ | I haven't gone to bed yet |
| 07:25.05 | floren | ChannelZ: i made the config straight, finally http://codepad.org/gTLQlCOD |
| 07:25.06 | BeeBuu | ChannelZ: yes,it's behind a gate |
| 07:26.15 | floren | it works proper, except that when i dial in, the phone picks rightaway after first ring. i have to set the voicemail locally. is better imo than using voip.ms one |
| 07:26.39 | floren | if i chnage servers with them, i will not lose my voice recordings |
| 07:27.05 | ChannelZ | BeeBuu: are you properly configured in sip.conf with externaddr and localnet? Do you have a range of UDP ports port-forwarded (or allowed in the firewall if your Asterisk is the same machine as the firewall) as specified in rtp.conf? |
| 07:27.33 | floren | ChannelZ: what do you use as database, native asterisk or mysql, prstgre? |
| 07:27.40 | floren | postgre* |
| 07:28.09 | ChannelZ | floren: you can erase your entire [default] context in extensions.conf because it isn't being used |
| 07:28.44 | floren | is used into inbound |
| 07:28.46 | ChannelZ | oh nevermind you are including it |
| 07:28.55 | floren | ya |
| 07:29.11 | ChannelZ | although I don't know what any of it actually accomplishes |
| 07:29.22 | floren | i only need it for local (inbound) for cisco phone |
| 07:29.40 | floren | is part of cisco patch, it performs a forward from a line to another |
| 07:29.48 | floren | for local extensions |
| 07:30.51 | BeeBuu | rtp.conf i put rtpstart=10000 rtpend=11000 |
| 07:30.58 | BeeBuu | in it |
| 07:31.31 | ChannelZ | and are those ports forwarded or allowed in (as the case may be?) |
| 07:31.47 | ChannelZ | and all my other questions |
| 07:31.50 | floren | ChannelZ: any extra tips for me related to mp3 and database storage? i plan to build the mpg123 codec, do i really needed if i compiled mp3 support in asterisk? |
| 07:32.11 | ChannelZ | That depends on what you think you're going to do with it |
| 07:32.28 | floren | i need it to play moh |
| 07:32.33 | floren | i guess :) |
| 07:32.49 | BeeBuu | Thanks,let me try. |
| 07:32.53 | ChannelZ | chances are you will be unhappy |
| 07:33.28 | floren | i should probably convert the mp3 to sln with sox |
| 07:33.43 | floren | the package is prebuilt in centos6 |
| 07:33.51 | floren | so less work to do on that area |
| 07:34.15 | floren | sox sound.mp3 -t raw -r 8000 -s -2 -c 1 sound.sln |
| 07:34.31 | floren | this should work i think |
| 07:35.44 | floren | i still have to look into it how asterisk supports sln, if you have any tips, please let me know |
| 07:36.13 | floren | i did compile asterisk with mp3 support |
| 07:36.38 | ChannelZ | sln is built in. It's basically headerless, uncompressed |
| 07:36.53 | floren | perfect, did not know |
| 07:36.59 | ChannelZ | signed linear, at 16-bit and then there are various samplerates you can use with different sln extensions for wideband support |
| 07:37.29 | ChannelZ | wav works too, format_wav should be built standard too |
| 07:37.41 | floren | what do you use? |
| 07:37.54 | floren | might as well follow you :) |
| 07:38.05 | floren | since you know asterisk in and out |
| 07:38.10 | ChannelZ | wav because it's easy to save out of any audio app |
| 07:38.17 | floren | aha |
| 07:38.43 | ChannelZ | mp3 is convenient for just dumping shit you already have in your MOH folder, but you waste CPU transcoding it |
| 07:38.59 | floren | i'm reading the asterisk 4th edition book to see how i can set properly the voicemail |
| 07:39.18 | floren | a bit priecy but worth it |
| 07:39.22 | ChannelZ | and I don't remember exactly, but I think with format_vorbis it still has to be at the right samplerate anyway.. not sure if mp3 was the same or not |
| 07:39.33 | ChannelZ | what's pricey? |
| 07:39.35 | *** join/#asterisk PLMg (PLMg@78.96.148.151) |
| 07:39.41 | floren | ya i'll stick with wav so i have no issues |
| 07:39.52 | floren | i have plenty of space on my little tiny NUC |
| 07:40.09 | ChannelZ | not that 8khz is huge |
| 07:40.14 | ChannelZ | mono |
| 07:40.25 | floren | 60gb ssd should be ok for one voicemail box |
| 07:40.30 | ChannelZ | maybe that was it. format_ogg_vorbis barfed on stereo. I don't remember. |
| 07:40.36 | floren | aha |
| 07:41.23 | ChannelZ | anyway if you're going to run it on one of those lunchpail computers anyway, might as well not make it harder on it. |
| 07:41.46 | floren | ChannelZ: you mentioned earlier that you use the dialplan to dial your long distance calls routed to a provider, right? |
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| 07:42.39 | ChannelZ | well sort of, I have that separated at work and I use POTS there but basically yes |
| 07:43.04 | ChannelZ | at home it's just me so I don't have it broken out into local/LD and it all costs the same anyway |
| 07:43.26 | floren | anyways, i wanted to ask you about your voicemail setup. i read into book about it and they dont talk at all about the actual setup but rather the asterisk config files |
| 07:43.51 | ChannelZ | what do you mean |
| 07:44.00 | floren | did you created manually the dirs inside the folder? |
| 07:44.14 | floren | or asterisk built them automatically |
| 07:44.27 | ChannelZ | no it does it |
| 07:44.33 | ChannelZ | you just configure the mailboxes in voicemail.conf |
| 07:44.34 | floren | ok |
| 07:44.40 | floren | ya |
| 07:45.05 | floren | i thought it would because is logical |
| 07:45.56 | floren | what do you use for storage, sqlite mysql or postgre? |
| 07:46.06 | floren | i'm tuhinking to move everything to mysql |
| 07:46.06 | ChannelZ | none |
| 07:46.12 | floren | ok |
| 07:46.44 | ChannelZ | just disk-based mailboxes |
| 07:46.52 | floren | ok will do the same |
| 07:47.10 | ChannelZ | I'm not doing anything weird/custom so there's no point |
| 07:47.18 | floren | ideed |
| 07:47.24 | floren | why reinvent the wheel |
| 07:47.47 | ChannelZ | although some day I might setup IMAP support so they can just wind up automagically in a folder in someone's mailbox, but meh |
| 07:48.12 | floren | i do have a quick question for you: because i have nothing setup after Answer(), is normal that the phone picks after one ring, right? |
| 07:48.58 | ChannelZ | depends on how slow your telco switch is I suppose, I'd figure you'd near no ring at all |
| 07:49.13 | floren | ya no ring at all |
| 07:49.30 | floren | if i dial from my mobile the DID, the device picks instantly |
| 07:49.31 | ChannelZ | but according to your last paste, you're not doing the Answer anymore, you're dialing your SIP/101 |
| 07:49.51 | floren | ya, i was wondering if you could help me with that last setup |
| 07:50.13 | floren | so i get the phone working like: 5 rings then voicemail |
| 07:50.54 | ChannelZ | looks like what you've already got, except your voicemail line is commented out |
| 07:51.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 07:51.19 | floren | ya i coment it because i did not worked on voicemail.conf |
| 07:51.21 | ChannelZ | although your timeout in the Dial is too high for 5 rings |
| 07:51.33 | floren | i'll put it to 25 |
| 07:52.01 | floren | should be about 5 rings? |
| 07:52.22 | ChannelZ | maybe. I use 20 which I think is 4 to 5 |
| 07:52.43 | floren | aha no prob |
| 07:53.17 | ChannelZ | I think the ring cadence in the US is like 6 seconds total? |
| 07:53.25 | floren | the part i dont understand is: why the device picks rightaway |
| 07:53.36 | floren | it should not. it should ring and ring |
| 07:53.43 | ChannelZ | what do you mean it picks up right away |
| 07:53.49 | ChannelZ | it's answering the call? |
| 07:53.55 | ChannelZ | on its own? |
| 07:54.03 | floren | let say i call now my did, it answers rightaway |
| 07:54.04 | floren | ya |
| 07:54.38 | ChannelZ | it's got to be something in your screwball phone's config |
| 07:54.50 | floren | i see |
| 07:55.16 | floren | but in real live, with my config the device will just ring |
| 07:55.54 | ChannelZ | well that's what it should do |
| 07:56.03 | floren | :) |
| 07:56.42 | ChannelZ | the 5xx phones don't have a config option for auto-answer I don't think, but you can do it by adding a SIP header before dialing.. |
| 07:56.46 | floren | i thought i need to device Answer() somewhere |
| 07:57.38 | ChannelZ | well it's complicated |
| 07:58.05 | floren | something like: exten => 101,1,Answer() > do something > Hangup() |
| 07:58.24 | ChannelZ | You have to answer a channel to do certain things with it.. like play sounds for an IVR, etc (partial lie) but some applications will automatically answer the channel for you if it hasn't been already... like Playback |
| 07:58.42 | floren | i see |
| 07:58.45 | ChannelZ | but if you're just sending a call to a device for instance, you generally don't want to answer it |
| 08:02.28 | ChannelZ | like, if your ITSP charges you for inbound, if you have asterisk Answer and then Dial your local device, the billing starts from the second asterisk Answered the incoming channel, not when you picked up your phone. |
| 08:02.48 | floren | i see |
| 08:04.40 | ChannelZ | higher level systems like SIP and PRI (compared to POTS) track the state of a call, its 'progress' |
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| 08:24.35 | ChannelZ | bed time |
| 08:24.44 | floren | have a good night ChannelZ |
| 08:25.14 | ChannelZ | ty you too |
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| 09:14.37 | bamsefar | How would I go about installing asterisk via the centos6 "asterisk-11" yum repository? A large number of packages are missing from it. |
| 09:14.38 | LieutPants | [ASTERISK-11] [Status: Closed] AGI channel_status failure - https://issues.asterisk.org/jira/browse/ASTERISK-11 |
| 09:15.00 | floren | bamsefar: use axivo repo |
| 09:15.06 | floren | rpm.axivo.com |
| 09:15.25 | floren | keep in mind that this is a set of rpm's set for cisco phones |
| 09:15.37 | floren | yum --enablerepo=axivoplus install asterisk |
| 09:16.28 | bamsefar | Okay, why is the official repository broken? |
| 09:16.49 | floren | i still have to build the dahdi package, well fax, jabber and radius |
| 09:16.57 | floren | default repo? centos6 does not have asterisk |
| 09:17.28 | floren | only fedora |
| 09:19.04 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 09:19.05 | mirela666 | hi |
| 09:19.06 | floren | which explains why i build the rpm's |
| 09:19.20 | mirela666 | what can cause this call end http://pastebin.com/ZZ98uERn |
| 09:19.20 | mirela666 | <PROTECTED> |
| 09:19.28 | floren | the funny thing is, centos selinux has the asterisk rules built-in |
| 09:20.38 | bamsefar | floren: Yes, but http://packages.asterisk.org/centos/6/asterisk-11/ is not complete. Why is this repo not working? |
| 09:20.38 | LieutPants | [ASTERISK-11] [Status: Closed] AGI channel_status failure - https://issues.asterisk.org/jira/browse/ASTERISK-11 |
| 09:21.53 | floren | no idea, i use mines because i need a patch for cusco 99xx 89xx |
| 09:21.59 | floren | cisco* |
| 09:22.23 | bamsefar | Okay |
| 09:22.41 | bamsefar | I'm currently running asterisk 1.6 and would like to upgrade, hopefully without building too much myself |
| 09:23.17 | bamsefar | the asterisk-current repository has some asterisknow-version package which installs a number of repositories and some shellscripts. |
| 09:25.04 | floren | do you need asterisknow? |
| 09:25.25 | floren | personally i settled for plain asterisk |
| 09:25.46 | floren | centos6 has all packages you need to get it working |
| 09:25.57 | floren | for any missing ones i'm building my own |
| 09:26.31 | floren | but out of the box, the most important one missing is dahdi, in my case i dont use it |
| 09:26.52 | floren | i'll have it done this week though :) |
| 09:27.12 | floren | so i have a complete asterisk set on hand for eberyone to use |
| 09:27.27 | floren | i'll have 2 versions, one with cisco patches and one without |
| 09:27.55 | floren | 11.5 is in rc1 anyways |
| 09:28.10 | floren | so i kind of wait for the 11.5 release in order to do everything clean |
| 09:28.29 | bamsefar | floren: No, I just want standard asterisk. |
| 09:28.51 | floren | if i understand correctly, aspterisknow is a complete distro |
| 09:28.53 | bamsefar | it would seem you need asterisk-11, asterisk-current and digium-current repositories to install asterisk. |
| 09:28.54 | LieutPants | [ASTERISK-11] [Status: Closed] AGI channel_status failure - https://issues.asterisk.org/jira/browse/ASTERISK-11 |
| 09:28.58 | floren | set it and forget it |
| 09:29.39 | floren | well, all i did is install the asterisk package myself. that is all you need |
| 09:29.54 | floren | i means that is all i needed :) |
| 09:31.05 | floren | what phone models do you plan to use? |
| 09:31.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:31.16 | floren | myself i got into asterisk game very recently |
| 09:31.26 | floren | i switched my land line to voip |
| 09:31.48 | floren | but i wanted a elegant phone so i got 2 cisco 9971 and 8961 |
| 09:32.49 | floren | it was a pain to setup everything but is doable. took me one week to dig into settings.many thanks to ChannelZ for his help with the asterisk configs |
| 09:34.25 | floren | so to get back to your remark, all you need is asterisk.x86_64 rpm |
| 09:36.36 | bamsefar | floren: Cisco 7940s |
| 09:49.40 | *** join/#asterisk MrQuist (~rob@83.232.96.217) |
| 09:49.47 | MrQuist | hey guys, small question |
| 09:49.53 | MrQuist | What are the standards for sip passwords? |
| 09:50.00 | MrQuist | secrets* |
| 09:50.05 | MrQuist | max. 16 characters? |
| 09:50.14 | MrQuist | I know for example that % is allowed as special character |
| 09:50.23 | MrQuist | but some (!, ^) aren't. |
| 09:50.40 | MrQuist | I can't seem to find any pages, does anybody here knows the standard? |
| 09:51.16 | floren | i use 16 chars with 4 groups: upper, lower, digits and minus |
| 09:51.38 | floren | keepass should help you generate easy those :) |
| 09:52.22 | MrQuist | i'm using a PHP script for that |
| 09:52.35 | MrQuist | upper lower digits and %, 16 chars is what i have now |
| 09:52.43 | MrQuist | but that should suffice wouldn't it? |
| 09:52.50 | floren | personally i use keepass to store and my passwords |
| 09:53.08 | floren | all* |
| 09:53.17 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-040-142.pools.arcor-ip.net) |
| 09:53.44 | floren | is portable and easy to sync the datanase with google drive |
| 09:53.49 | floren | database* |
| 09:54.33 | MrQuist | easy to steal |
| 09:54.38 | MrQuist | hehe |
| 09:54.55 | MrQuist | i have a custom built phone provision script that works with CURL and PHP |
| 09:55.00 | MrQuist | so i have one array with all phones |
| 09:55.01 | MrQuist | works great |
| 09:55.02 | floren | cant if you dont know the master |
| 09:55.17 | floren | plus you can bind it to a device so it will work only on your computer :) |
| 09:55.23 | MrQuist | even then i'd need to copy 1000 secrets |
| 09:55.26 | MrQuist | do not want |
| 09:55.36 | floren | i see :) |
| 09:55.57 | floren | i was reffering to my simple setup, in your case is totally diff |
| 09:56.08 | floren | i only have 2 phones :D |
| 09:56.36 | MrQuist | hahah yeah then i understand |
| 09:56.38 | MrQuist | but thanks :) |
| 09:57.01 | floren | related to your original question, the 4 groups i used combined in 16chars are working for me |
| 09:57.09 | MrQuist | great |
| 09:57.28 | MrQuist | yeah i just provisioned some phones with secrets like J3&(@k,{@p}><2!@o? |
| 09:57.28 | floren | should be pretty secure imo |
| 09:57.31 | MrQuist | that didn't work |
| 09:57.35 | floren | ya |
| 09:57.45 | MrQuist | and i love the CLI debug information |
| 09:57.46 | floren | i remember trying also |
| 09:57.50 | MrQuist | "Not authenticated" |
| 09:57.50 | floren | the brackets group |
| 09:58.00 | floren | it broke everything |
| 09:58.07 | floren | so i stick with teh groups i mentioned earlier |
| 09:58.13 | MrQuist | yes |
| 09:58.27 | MrQuist | but, from the point of sharing knowledge |
| 09:58.30 | MrQuist | you can also use % |
| 09:59.13 | floren | probably ! also |
| 09:59.31 | MrQuist | not so sure about that |
| 09:59.35 | floren | but those are conditionals so it might break something |
| 09:59.43 | MrQuist | indeed |
| 09:59.46 | floren | != == && etc |
| 09:59.52 | MrQuist | i tried putting "" around the secrets |
| 09:59.55 | MrQuist | that doesn't work either |
| 09:59.59 | floren | stick with teh simple things :) |
| 10:00.22 | floren | ChannelZ said that and it works really good :) |
| 10:00.55 | floren | how are your storing the secrets |
| 10:01.12 | floren | if you store them into a db is easy to validate if another secret exists already |
| 10:01.54 | floren | do a call in your php script to validate against the db and bingo |
| 10:02.08 | floren | a simple loop should take care of that |
| 10:04.29 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 10:06.56 | MrQuist | oh, the chances of a secret being the same are 1000 : 26^26^10^2 |
| 10:07.08 | floren | :) |
| 10:07.10 | MrQuist | i am willing to take that rist |
| 10:07.11 | MrQuist | risk* |
| 10:07.19 | floren | you have 1000 phones though :) |
| 10:07.40 | floren | a db is a must to manage them :) |
| 10:08.10 | floren | you use some fancy interface to manage them? like freepbx etc? |
| 10:09.12 | mirela666 | 11:19:22 AM - mirela666: what can cause this call end http://pastebin.com/ZZ98uERn |
| 10:09.12 | mirela666 | 11:19:22 AM - mirela666: missing beep file would be noticed with file.c:663 ast_openstream_full: File beep does not exist in any format I guess |
| 10:10.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.245) |
| 10:15.07 | floren | mirela666: i'm not experienced with asterisk, but most of the issues i had them solved with asterisk -rvvvv > sip set debug on |
| 10:15.18 | floren | you should rightaway what is wrong |
| 10:15.41 | floren | should see* |
| 10:17.39 | *** join/#asterisk LooserOuting (~LooserOut@ip-176-198-132-85.unitymediagroup.de) |
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| 10:23.27 | FlashDeluxe | hi @all! Is there a nice web gui tool (or sth else) which can handle queues? I would like to have a simple solution, that i see my queue and get connected to somebody with one click :-) Does anybody know such a tool? |
| 10:42.49 | *** join/#asterisk CeBe (~CeBe@port-92-206-90-229.dynamic.qsc.de) |
| 10:44.56 | bamsefar | floren: I got it working now, with digium and asterisk repositories. |
| 10:45.07 | floren | bravo :)\ |
| 10:45.18 | bamsefar | Puppet installs without issues now. :) |
| 10:46.14 | CeBe | Hi, I am trying to set up asterisk with SIP failover via SRV record. so when one server in SRV list fails it should take another one. |
| 10:46.14 | CeBe | as far as I could figure out asterisk only takes the first entry of SRV record and does not support failover yet. |
| 10:46.22 | mirela666 | floren: it is on |
| 10:46.42 | CeBe | Is this a feature you would like to have implemented? I could not find a feature request in bug tracker |
| 10:48.04 | CeBe | As far as I see SRV is used for failover in most cases so asterisk should use it this way too or at least be able to via config option |
| 10:48.38 | CeBe | I am not yet familiar with asterisk code but could try to work on it if you point me to the code I have to look at |
| 10:50.07 | bamsefar | I have two asterisk servers (one old, 1.6) and one new (11.4.0), I would like to be able to place calls from the old one to the confbridge app on the new one for some testing before I move phones to the new asterisk box. How would this be configured? |
| 10:50.33 | *** join/#asterisk geeksteve (~geeksteve@emh-nat.poundbury.com) |
| 10:52.02 | bamsefar | Just configure a "newasterisk-peer" on the old asterisk box, and 666 => { Dial(SIP/newasterisk/${EXTEN}); } in my extensions.ael? |
| 10:52.56 | CeBe | bamsefar: thats how I would do it too. |
| 10:56.48 | bamsefar | CeBe: Thanks! :) |
| 10:57.47 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 11:01.51 | bamsefar | Hrrm, why is the asterisk package depending on asterisknow-version, which installs a number of yum-repositories along with a script to update /etc/issue? Shouldn't this be separate packages? |
| 11:06.43 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:27.49 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 11:32.06 | CeBe | btw: is there maybe another way to configure SIP failover with or without SRV records? |
| 11:33.47 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:34.28 | CeBe | I'd like to avoid programming this behavior inside of the dialplan |
| 11:35.58 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 11:37.38 | cneb3000 | CeBe: i've never used SRV failover for VoIP, purely because a SRV isn't reliable instant. Remember VoIP is a realtime thing, so you ideally want the failover to be as instant as possible |
| 11:38.01 | cneb3000 | there's a load of different things you can use. I personally use OpenSIPS for something like that |
| 11:38.42 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:39.32 | cneb3000 | unfortunatly, it's one of those things thats often documented in a way that, if you're not very familiar with networking and linux, you may struggle to understand it :( |
| 11:40.15 | CeBe | cneb3000: thanks for your answer! Why isn't SRV reliable instant? As far as I understood you got a list of servers and try one after the other until you succeed |
| 11:40.53 | cneb3000 | I'm being picky :) let me try and explain.. |
| 11:40.53 | CeBe | the time here is milliseconds which is okay for creating a call imo |
| 11:40.53 | CeBe | k :) |
| 11:40.53 | cneb3000 | oh ok - if its that fast no problem |
| 11:40.57 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:41.33 | cneb3000 | sometimes SRV isn't always that fast is all. With OpenSIPS for example, the call will hit opensips, which will already know that one of the two servers is down, so won't even take the time to try it. if that makes sense? so it's quicker. |
| 11:41.44 | cneb3000 | i'm biased, I've never actually used SRV. I'm going oiff what I read |
| 11:42.44 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 11:43.15 | cneb3000 | I'm not an SRV expert, but would it be miliseconds? If it checks one IP, and then fails over if that doesn't work, surely it has to wait maybe a second or two for that first IP to respond? Or is that not the case? |
| 11:44.06 | CeBe | cneb3000: okay, maybe if the server is down you are right, I did not think about that case. |
| 11:44.32 | CeBe | current situation I encoutered was that I got error 500 internal server error back imidiately |
| 11:44.55 | CeBe | will definitively look at OpenSIPS, thanks for the tip |
| 11:44.58 | cneb3000 | ahh ok. yes I guess in that case the failover would be immediate |
| 11:45.18 | cneb3000 | its OK :) I highly recommend watching the videos for OpenSIPs, because the documentation can be a bit overwhelming, and is sometimes a little out of date |
| 11:48.59 | CeBe | cneb3000: thanks, can you tell me where I find them? They don't seem to be on documentation page |
| 11:50.33 | cneb3000 | try http://www.opensips.org/Documentation/Webinars would be a good start. but before you go down a winding trail, maybe trying looking at 'heartbeat'. As opensips might be a bit overkill. http://www.open-voip.org/index.php?title=Asterisk_High_availablity#Configure_heartbeat |
| 11:53.39 | CeBe | hm... guess we misunderstood here :) I am trying to send outgoing calls to my telephone provider who has 3 Servers and I want to fail over when one of them is not reachable |
| 11:53.57 | CeBe | I currently only have one server |
| 11:54.33 | cneb3000 | oooohh! |
| 11:55.46 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 11:55.58 | cneb3000 | I think you'll probably be able to achieve what you want with a clever dialplan |
| 11:56.51 | CeBe | okay, was not sure if dialplan was the right place to implement this... |
| 11:57.07 | CeBe | but seems to be the only possible thing right now |
| 11:57.13 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 11:57.16 | cneb3000 | well, SRV just goes to one server, if it fails, the next right? well a dialplan will do that same |
| 11:57.24 | *** join/#asterisk mintos (~mvaliyav@14.99.31.86) |
| 11:57.55 | CeBe | got it. SRV would be handy so that I do not need to configure the list of servers manually |
| 11:58.08 | cneb3000 | are you ok with dialplans? have a look into, the 'dial' command, the 'gotoif' command and the channel variable 'dialstatus'. |
| 11:58.10 | cneb3000 | ahh i see |
| 11:59.13 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 11:59.16 | CeBe | sure, I think I know how to do it in dialplan. |
| 12:00.11 | CeBe | thanks cneb3000! will try implementing dialplan now |
| 12:00.26 | cneb3000 | enjoy! |
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| 13:27.06 | bamsefar | I get this error: Dropping incompatible voice frame on SIP/pbx1-00000002 of format gsm since our native format has changed to (ulaw) When trying to call to a ConfBridge application. How would I fix this? |
| 13:27.45 | cneb3000 | basefar: can you change the verbosity of the logging and run that again? |
| 13:28.00 | cneb3000 | sorry... bamsefar* |
| 13:28.41 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 13:31.15 | bamsefar | Yes |
| 13:31.28 | cneb3000 | ok, then put the logging here - http://pastebin.com/ |
| 13:33.14 | Katty | morning |
| 13:34.25 | bamsefar | cneb3000: Which debug level would you want? |
| 13:34.35 | cneb3000 | set it to 23 |
| 13:35.07 | bamsefar | Right |
| 13:35.12 | bamsefar | http://pastebin.com/ZPMGSY1g <-- Theres with level 10. |
| 13:35.16 | cneb3000 | ahh thats cool |
| 13:35.28 | cneb3000 | i meant to say 3 anyway lol. not much more extra after 10 |
| 13:35.32 | cneb3000 | i meant to say 3 anyway lol. not much more extra after 3*. |
| 13:35.56 | bamsefar | No, I saw that. :) |
| 13:35.58 | cneb3000 | whats the symptom? you cant hear the messages being played no? |
| 13:36.05 | bamsefar | Yeah |
| 13:36.08 | bamsefar | It's dead silent. |
| 13:36.17 | cneb3000 | ok, it's because for some reason, asterisk isn't letting ytou play 'ulaw files'. |
| 13:36.26 | bamsefar | Okay |
| 13:36.28 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 13:36.33 | cneb3000 | you see the line.. <Bridge/0x7f06cc002f18-input> Playing 'confbridge-join.gsm' (language 'en'). how theres no error |
| 13:36.41 | cneb3000 | but then the line <SIP/pbx1-00000005> Playing 'conf-onlyperson.ulaw' (language 'en') there is one |
| 13:36.48 | bamsefar | Yup |
| 13:37.08 | bamsefar | I don't hear the gsm one either tho |
| 13:37.25 | cneb3000 | i think the gsm is just like a small beep isnt it? |
| 13:38.16 | cneb3000 | basically, either your phone or asterisk is saying "i cant do ulaw". First step would be to check your sip.conf file, find the account you're using. there should be some lines like.. disallow=all, allow=gsm, etc.? |
| 13:38.29 | bamsefar | Yes |
| 13:38.51 | cneb3000 | the first line, disallow=all, is a blanket config that says "dont allow any codecs". but then anything you follow up with allow=gsm for example, it allows that. |
| 13:39.12 | bamsefar | Okay |
| 13:39.14 | cneb3000 | you need to make sure allow=ulaw is on there |
| 13:39.22 | *** join/#asterisk serafie (~erin@nat/digium/x-rtdvfvnyzqobgkgk) |
| 13:39.30 | Katty | hugs serafie |
| 13:39.42 | bamsefar | Hrrm, my topology is cisco 7940G -> asterisk 1.6 -> asterisk 11. (migration project) |
| 13:40.01 | cneb3000 | so where is this log being generated? 1.6 or 11? |
| 13:40.10 | bamsefar | 11 |
| 13:40.23 | cneb3000 | ok, so you must have an account in sip.conf on 11 for 1.6 to connect to right? |
| 13:40.38 | cneb3000 | could you paste bin that sip.conf account onto pastebin. make sure you remove the password :) |
| 13:41.10 | bamsefar | cneb3000: Just for that phone and the peers between asterisks, right |
| 13:41.33 | cneb3000 | just the sip.conf from the asterisk 11 box |
| 13:43.01 | bamsefar | http://pastebin.com/DpE9Rhmc |
| 13:43.20 | cneb3000 | oh wait.. i didn't spot the line "> 0x7f06c0021f10 -- Probation passed - setting RTP source address to 192.168.204.11:12216" |
| 13:43.21 | bamsefar | I should probably just remove allow=gsm from the asterisk peer definitions, right? |
| 13:43.41 | cneb3000 | yes try that, but i'm not sure if that's the issue.. i've never spotted the > 0x7f06c0021f10 -- Probation passed - setting RTP source address to 192.168.204.11:12216 before. |
| 13:43.55 | *** join/#asterisk guitarHester (~guitarHes@nat/digium/x-ppsqlfwodtfsoqzn) |
| 13:43.56 | cneb3000 | i think that's something to do with moving the place the media gets sent to? |
| 13:44.03 | cneb3000 | is 192.168.204.11 your 1.6 box? |
| 13:45.10 | bamsefar | Yes |
| 13:45.14 | cneb3000 | if none of this works, I suggest running a tcpdump/wireshark trace on each box, making the call. and seeing if RTP/media is being trasmitted correctly. |
| 13:45.36 | cneb3000 | hope that helps :( |
| 13:45.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ffhggasaexmqsdzt) |
| 13:45.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:48.22 | Katty | mjordan: morning |
| 13:48.46 | WIMPy | waves to Katty |
| 13:48.54 | Katty | hugs WIMPy |
| 13:49.18 | bamsefar | cneb3000: The asterisk 11 box would not send any data directly to the phone, would it? |
| 13:49.52 | bamsefar | Yes it does. :) |
| 13:49.57 | bamsefar | Firewall eats the packets. :) |
| 13:50.08 | cneb3000 | Oh is that the problem? lol |
| 13:50.39 | cneb3000 | as soon as I saw that line in the logging I thought of that. doh. that's why I suggested the tcpdump thing. |
| 13:58.15 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:58.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:05.38 | *** join/#asterisk LemensTS (~t15@mail.zuckerfeather.com) |
| 14:06.25 | LooserOuting | is it Ok if a client uses ilbc mode20 and ptime20 and asterisk answers with ilbc mode 30 and ptime 30 ? |
| 14:07.08 | LemensTS | What do you guys recommend for hosted pbx? I have looked at ring central, virtualpbx, and vocalcity. I need 32 iphones, half of them are used just dial interoffice. I guess I would like to know which company is the most solid... |
| 14:07.19 | LemensTS | ip phones not iphones...lol |
| 14:08.01 | Katty | i recommend vmware and linux. |
| 14:08.12 | Katty | and a hippa certified data center |
| 14:08.54 | cneb3000 | LemensTS: everyone has a favourite, if they use one. but most people in here probably have their own 'hosted pbx' of sort :) |
| 14:09.04 | LemensTS | Katty: so do I, I wrote easyitsp but my boss wants to go hosted with a company other than me :/ |
| 14:09.14 | Katty | LemensTS: oh i see :< |
| 14:09.20 | Katty | shakes fist at LemensTS's bossman |
| 14:09.39 | cneb3000 | lol |
| 14:09.43 | cneb3000 | what a bummer! |
| 14:09.52 | Katty | oh well. |
| 14:09.59 | Katty | they sign the paychecks, we do what they say |
| 14:10.10 | Katty | and glare. a lot. |
| 14:10.17 | cneb3000 | why don't you choose the worst hostedpbx you can find |
| 14:10.22 | cneb3000 | and he'll come crying back to you |
| 14:10.23 | LemensTS | Hmmm...its hard to judge by reviews on the net hard to say if its the companies writing them or actual customers |
| 14:10.39 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 14:10.41 | Katty | LemensTS: you just never know about that. |
| 14:10.56 | Katty | LemensTS: it could even be an 'actual' customer, but the company is just pushing a front |
| 14:11.05 | Katty | LemensTS: i vote on a trial |
| 14:11.22 | LemensTS | Yea that is probably what I'll end up doing |
| 14:11.41 | Katty | maybe you'll even get some swag out of the deal! |
| 14:12.23 | cneb3000 | i do know someone in the uk who used sipgate.co.uk for theres.. i can't attest to how good they are myself, but he never had a complaint |
| 14:12.29 | cneb3000 | theirs* |
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| 14:35.59 | k610 | how can i install in unattended mode ? i get a menu when i run : |
| 14:36.00 | k610 | <PROTECTED> |
| 14:37.23 | k610 | nvm |
| 14:37.35 | k610 | i just have to skip the make menuselect part ... |
| 14:42.07 | carrar | https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options |
| 14:42.10 | carrar | menuselect/menuselect --list-options |
| 14:44.18 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:44.22 | igcewieling1 | k610: also see http://pastebin.ca/2425350 it shows the menuselect options |
| 14:45.30 | *** part/#asterisk LemensTS (~t15@mail.zuckerfeather.com) |
| 14:45.40 | carrar | yeah google has lots of answers |
| 14:47.35 | k610 | thx |
| 14:47.45 | k610 | i wonder if i really need anything extra |
| 14:49.42 | k610 | is it easier in asterisk 11 to to set a db so i don't have to do the : edit sip.conf / extensions.conf -> reload asterisk |
| 14:49.52 | Katty | carrar: YOU |
| 14:50.03 | carrar | Katty: US!!! |
| 14:50.11 | Katty | carrar: we should so hug right now. |
| 14:50.17 | carrar | We should |
| 14:50.21 | Katty | let's do it! |
| 14:50.21 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 14:50.24 | Katty | hugs carrar |
| 14:50.30 | carrar | hugs Katty!! |
| 14:50.35 | Katty | you good? |
| 14:50.38 | carrar | Oh yeah |
| 14:50.40 | Katty | everything peachy? |
| 14:50.47 | Katty | egg-cellent. |
| 14:50.48 | carrar | better then a room full of kittens |
| 14:50.54 | Katty | ^___________^ |
| 14:51.13 | carrar | I just want to hug every cat |
| 14:51.22 | Katty | do you also like to run? |
| 14:51.29 | carrar | YES!! |
| 14:51.34 | carrar | sorry I was thinking about cats |
| 14:51.52 | cneb3000 | k610: yes, config in a database does make things a little easier to manage for some :) |
| 14:52.08 | Katty | carrar: *hee* |
| 14:52.11 | Katty | listens to song |
| 14:56.09 | igcewieling1 | Running is for people in a hurry. Being in a hurry is bad for you. Therefore running is bad for you. |
| 14:56.38 | igcewieling1 | This is why I don't run. |
| 15:04.10 | Katty | igcewieling1: it's also pretty hard on your knees. |
| 15:04.18 | Katty | igcewieling1: and if you trip, it's a real pain |
| 15:10.13 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:10.48 | *** join/#asterisk Draecos (~Draecos@124-149-94-184.dyn.iinet.net.au) |
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| 15:36.09 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:36.27 | Katty | hello cupcake |
| 15:39.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:39.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:41.23 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:42.32 | *** join/#asterisk mr_claus (~crosenbe@ltea-047-065-097-249.pools.arcor-ip.net) |
| 15:47.52 | *** part/#asterisk mr_claus (~crosenbe@ltea-047-065-097-249.pools.arcor-ip.net) |
| 15:51.44 | *** join/#asterisk Voicu (~quassel@80.97.64.4) |
| 15:51.50 | *** join/#asterisk tzanger (andrew@wallace.mixdown.ca) |
| 15:52.33 | Voicu | hello, I'm going to ask a noobish question |
| 15:52.48 | Katty | braces self |
| 15:52.53 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 15:52.57 | Katty | hai dwayne! |
| 15:53.16 | Voicu | is asterisk viable for RTMP <-> h.323 communication? |
| 15:53.26 | Voicu | I know there is rtmp and h323 capability |
| 15:53.32 | dwayne | Gamarjoba Katty! |
| 15:53.34 | Voicu | but how difficult is it to link them up? |
| 15:53.43 | Katty | dwayne: gamarjoba? |
| 15:53.50 | Voicu | I've never installed asterisk so I don't know what I'm up against |
| 15:53.52 | dwayne | Hello in Georgian |
| 15:53.59 | Katty | oh. |
| 15:54.16 | Katty | re-reads, carefully |
| 15:54.35 | Katty | my brain refuses to parse that into any sort of accented word or phrase |
| 15:55.10 | Voicu | I just read it like it's written Gah-mahr-joa-bah |
| 15:55.30 | Voicu | sorry for buttin' in |
| 15:55.36 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 15:55.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.184) |
| 15:56.53 | cneb3000 | Voicu: why don't you try install asterisk? it's not to long a process. |
| 15:56.58 | pabelanger | Voicu: IIRC there was a RTMP patch up on the issue track some time ago, but don't know what ever happen to it |
| 15:57.06 | pabelanger | you could try getting it to work again for asterisk |
| 15:57.24 | Voicu | yeah, that was my next step |
| 15:57.31 | Voicu | but... I thought someone maybe tried it already |
| 15:58.09 | Voicu | I don't have a lot of experience in this area so I'm not sure what is the best solution |
| 15:58.21 | pabelanger | Voicu: https://issues.asterisk.org/jira/browse/ASTERISK-14451 |
| 15:58.23 | LieutPants | [ASTERISK-14451] [Status: Open] [Assigned: phsultan] [patch] [branch] RTMP support in Asterisk - https://issues.asterisk.org/jira/browse/ASTERISK-14451 |
| 15:58.33 | Voicu | yes |
| 15:58.35 | Voicu | I found that |
| 15:58.45 | pabelanger | that is the best you are going to get |
| 15:58.58 | Voicu | but it says it will connect to a RTMP server |
| 15:59.32 | Voicu | I (think) I need a standalone solution |
| 16:00.51 | *** join/#asterisk navaismo (~navaismo@189.191.224.201) |
| 16:00.53 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.184) |
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| 16:55.33 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 16:55.38 | qakhan | hi all, |
| 16:56.03 | *** join/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 16:56.12 | *** part/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 16:56.36 | qakhan | how i can change CallerID on dahdi channel calls. tried Set(CALLERID(num)) but it didnt work |
| 16:57.45 | Greenlight | Your carrier may not be allowing it |
| 16:58.56 | robl^ | and if they allow it, you can usually only change the number, not the name associated with it. |
| 16:59.37 | qakhan | i have PRI from verizon |
| 17:03.31 | igcewieling1 | qakhan: Again, your carrier may not allow it. |
| 17:04.04 | igcewieling1 | qakhan: show us the Set line from the CLI (JUST THE SET LINE) |
| 17:04.20 | igcewieling1 | We can at least verify you are sending valid callerid before you contact your Carrier. |
| 17:04.55 | Greenlight | Pri trace will further show what you're sending |
| 17:05.03 | Greenlight | The carrier may want it in specific format |
| 17:05.13 | qakhan | Set("SIP/3288-00080fd1", "CALLERID(num)="301208XXXX"") in new stack |
| 17:05.14 | Greenlight | I know we had to alter settings for Virgin Media here in UK |
| 17:05.20 | WIMPy | qakhan: Did you set the correct TON (pri*dialplan)? |
| 17:05.32 | *** join/#asterisk mnathani (~mnathani@198.84.231.11) |
| 17:05.33 | Greenlight | Yup, that's the one TON |
| 17:05.39 | igcewieling1 | qakhan: that looks valid, assumeing the X is you just trying to hide the number |
| 17:05.54 | qakhan | yes |
| 17:05.59 | Greenlight | Call your carrier, and ask firstly if they accept ANY CLI, and secondly the specific format |
| 17:06.07 | igcewieling1 | you are not adding a leading 1 or putting dashes, which is what many people do |
| 17:06.17 | igcewieling1 | This is good. |
| 17:06.37 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 17:07.02 | qakhan | WIMPy what do you mean by TON (pri*dialplan) |
| 17:09.05 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 17:21.19 | *** join/#asterisk g_r_eek (~g_r_eek@78-22-218.adsl.cyta.gr) |
| 17:23.52 | [TK]D-Fender | [13:05]qakhanSet("SIP/3288-00080fd1", "CALLERID(num)="301208XXXX"") in new stack <- NO QUOTES ALLOWED |
| 17:24.57 | Kobaz | depends on the asterisk version, but yeah 1.8 treats quotes as a literal part of the value |
| 17:25.30 | qakhan | [TK]D-Fender Set("SIP/3288-000811ae", "CALLERID(num)=3012082222") |
| 17:25.33 | qakhan | same result |
| 17:25.55 | [TK]D-Fender | qakhan: Not as far as I see |
| 17:26.38 | qakhan | i am not getting this number as callerid |
| 17:27.06 | [TK]D-Fender | qakhan: And we have no proof about what is going out and what else you've configured |
| 17:27.51 | qakhan | what you want me to show you |
| 17:28.10 | [TK]D-Fender | pakPrrof & configs. What else would you think? |
| 17:28.21 | [TK]D-Fender | qakhan: Proof & configs. What else would you think? |
| 17:28.27 | WIMPy | qakhan: http://voice.yeti.dk/Asterisk_vs_ISDN/5 |
| 17:29.06 | [TK]D-Fender | WIMPy: I wouldn't waste time on guesses till he actually shows something sane |
| 17:30.05 | WIMPy | [TK]D-Fender: That was just to answer the question and calarify what "TON" I was talking about. |
| 17:30.33 | qakhan | exten => 3900,1,Set(CALLERID(num)=3012082222) |
| 17:30.34 | qakhan | exten => 3900,2,Dial(dahdi/g1/${EXTEN}) |
| 17:31.04 | [TK]D-Fender | qakhan: Do you think that is proof? |
| 17:31.08 | WIMPy | qakhan: What kind of number is it your're setting there? |
| 17:31.10 | [TK]D-Fender | qakhan: Where is THE CALL? |
| 17:31.23 | [TK]D-Fender | qakhan: Where are the CONFIGS? |
| 17:32.24 | qakhan | i have PRI from verizon and i want to send this number as callerid instead of our main number |
| 17:32.51 | carrar | You are calling toa 4 digit number? |
| 17:32.54 | WIMPy | Is that a number that is routed to that line? |
| 17:33.58 | qakhan | sorry i was test through 3900 |
| 17:34.09 | *** join/#asterisk CyberBob (~Networks@2.49.71.26) |
| 17:34.33 | qakhan | i used _X.,1,Dial(dahdi/g1/${EXTEN}) |
| 17:34.39 | igcewieling1 | qakhan: they want CLI output, not DIALPLAMN |
| 17:34.43 | carrar | ambiguous response |
| 17:34.54 | CyberBob | I am able to dial but cannot hear from both sides using elastix please advise. |
| 17:35.07 | igcewieling1 | ~gui |
| 17:35.07 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 17:35.10 | igcewieling1 | ~elastix |
| 17:35.10 | infobot | i heard elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 17:35.54 | CyberBob | Oh thanks. |
| 17:36.41 | *** join/#asterisk peetaur2 (~peter@x2f0f151.dyn.telefonica.de) |
| 17:37.23 | igcewieling1 | CyberBob: likely a nat/localnet/externip/directmedia issue but if you change it in the standard config files your settings will be overwritten by the GUI |
| 17:39.11 | CyberBob | igcewieling1 But on LAN my calls are going and fine, I've checked NAT but nothing effects.. |
| 17:39.13 | qakhan | Dial("SIP/3288-00000000", "dahdi/g1/347761XXXX") |
| 17:39.38 | igcewieling1 | qakhan: chances are you need a leading 1 |
| 17:39.45 | [TK]D-Fender | Means nothing |
| 17:39.54 | [TK]D-Fender | moves on to more productive matters |
| 17:40.08 | igcewieling1 | [TK]D-Fender: I'm sure it is a carrier issue. |
| 17:40.30 | [TK]D-Fender | igcewieling1: I'm not. And I'm also not going to ask for it a dozen times. |
| 17:40.36 | WIMPy | Where's the debug? |
| 17:40.40 | igcewieling1 | qakhan: if you want to CONFIRM your pbx is sending the number, use pri debug settings in Asteirsk and slog through the output |
| 17:46.12 | *** part/#asterisk Voicu (~quassel@80.97.64.4) |
| 17:49.44 | qakhan | here is pri debug |
| 17:49.45 | qakhan | http://pastebin.com/VZc6ndsZ |
| 17:50.33 | WIMPy | should mention that he prefers hex or intense debug. |
| 17:51.05 | WIMPy | But that callerID looks correct to me. |
| 17:51.53 | WIMPy | But I also see that you won't be able to make international calls with your config. |
| 17:52.07 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:52.26 | qakhan | i dont need to make internation calls |
| 17:56.38 | igcewieling1 | > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)> Presentation: Presentation permitted, user number not screened (0) '3012082222' ] |
| 17:56.46 | igcewieling1 | the number is being sent. Now CONTACT YOUR CARRIER. |
| 17:57.05 | igcewieling1 | qakhan: set your pridialplan=unknown and you'll be able to make local, toll, AND interational calls. |
| 17:57.36 | igcewieling1 | qakhan: it is pointless to hide TNs. Trust us, nothing you have to say is interesting. 8-| |
| 17:58.08 | *** join/#asterisk guitarHester (~guitarHes@nat/digium/x-sdncxbdtemgqxsgi) |
| 17:59.03 | *** join/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 17:59.31 | *** part/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 17:59.45 | igcewieling1 | qakhan: I was under the impression calls were working. Based on your debug they are not working. |
| 17:59.59 | igcewieling1 | Ext: 1 Cause: Interworking, unspecified (127), class = Interworking (7) ] |
| 18:00.53 | WIMPy | Read the Message Type. |
| 18:04.11 | igcewieling1 | read the end cause |
| 18:04.31 | igcewieling1 | he looks to be gone now so it is moot |
| 18:04.38 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:05.01 | *** join/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 18:05.05 | *** part/#asterisk russum (~russum@83-131-91-45.adsl.net.t-com.hr) |
| 18:09.14 | *** join/#asterisk jmls1 (~julian@77.107.171.82) |
| 18:11.51 | *** join/#asterisk aelliott22 (~aelliott@198.41.29.45) |
| 18:14.50 | qakhan | i m here guys. lil busy with other work |
| 18:17.47 | igcewieling1 | qakhan: come back when you have the time to dedicate to solving your issue. |
| 18:38.42 | raden | Katty, :D :D :D |
| 18:48.05 | *** join/#asterisk c|oneman (cloneman@2605:6400:2:fed5:22:0:3b06:3913) |
| 18:49.36 | CyberBob | Unable to hear, tried everything almost. |
| 18:49.39 | CyberBob | :( |
| 18:50.01 | igcewieling1 | CyberBob: start by listing what you tried. |
| 18:50.19 | igcewieling1 | you set your nat= localnet= externip= and directmedia= settings to the correct values? |
| 18:52.54 | *** join/#asterisk random_cat (~random_ca@gateway/tor-sasl/randomcat/x-49498005) |
| 18:53.20 | CyberBob | igcewieling1 Done with nat localnet externip but can you please explain directmedia? |
| 18:53.36 | igcewieling1 | directmedia should be no. |
| 18:53.53 | igcewieling1 | you can't do direct media with NAT unless you want to spent a lot of time and complicated proxy stuff |
| 18:54.21 | igcewieling1 | directmedia was called "canreinvite" in older Asterisk versions. |
| 18:54.39 | CyberBob | Ah, canreinvite=no |
| 18:56.05 | CyberBob | But still we're not be able to hear each others. |
| 18:56.54 | igcewieling1 | CyberBob: I'd suggest putting the CLI output if a failed call on a pastebin, but I have no interest in slogging through 500 lines of Elastix/FreePBX crap. |
| 19:00.19 | CyberBob | Here are the logs, http://pastebin.com/khvbGg81 |
| 19:02.35 | [TK]D-Fender | "sip set debug on" <---- |
| 19:02.47 | [TK]D-Fender | If you don't have SIP DEBUG enabled.. you're not really looking. |
| 19:06.41 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-104-246-231.ph.ph.cox.net) |
| 19:10.49 | CyberBob | http://pastebin.com/ugPY9nyH here is the logs after sip set debug on |
| 19:10.54 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 19:16.01 | navaismo | incomplete! |
| 19:19.16 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 19:20.07 | [TK]D-Fender | Very incomplete |
| 19:20.11 | *** join/#asterisk lanning_ (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 19:20.21 | *** join/#asterisk peter_ (~peter@x2f0f151.dyn.telefonica.de) |
| 19:23.30 | CyberBob | Is there anyway to get the output in a file? |
| 19:24.25 | igcewieling1 | CyberBob: it should already be in /var/log/asterisk/whateveryourguiconfiguredforthefilename |
| 19:24.40 | *** join/#asterisk cjm_ (~cjm@tclc.org) |
| 19:25.19 | *** join/#asterisk wolfmitchell_ (~wolfmitch@botters/wolfmitchell) |
| 19:25.20 | *** join/#asterisk g_r_eek (~g_r_eek@78-132-136.adsl.cyta.gr) |
| 19:26.03 | Katty | hugs raden |
| 19:26.16 | Katty | raden: hello cupcake! what's doin? |
| 19:28.14 | navaismo | CyberBob, asterisk -rnvvvvdd | tee sipdebug.txt |
| 19:31.28 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 19:32.38 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 19:32.50 | CyberBob | Thanks navaismo. |
| 19:32.57 | CyberBob | Here are the logs http://pastebin.com/URsycH4X |
| 19:44.06 | CyberBob | Still no voice.. :S |
| 19:44.07 | *** join/#asterisk fischli (~fischli@static-31-25-152-181.ewacom.ropa.net) |
| 19:48.48 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 19:51.26 | CyberBob | Any suggesstion GURU's? |
| 19:51.45 | *** join/#asterisk JonnyWhatshisfac (~rain@99-17-60-154.lightspeed.nworla.sbcglobal.net) |
| 19:52.00 | JonnyWhatshisfac | hey guys - anyone around that has a spare minute to let me pick their brain? |
| 19:53.34 | navaismo | CyberBob, your debug show a call from 3cx to asterisk to AVAYA trunk?* then the avaya stuff its bridged to 1012, your avaya stuff is properly configured? |
| 19:54.25 | navaismo | ~ask |
| 19:54.26 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:54.30 | navaismo | JonnyWhatshisfac, ^ |
| 19:55.50 | CyberBob | navaismo Yes, it's configured and working properly I've checked it twice. |
| 19:57.55 | navaismo | well you should start a tcpdump captuing and also see if rtp is going and coming |
| 19:59.37 | JonnyWhatshisfac | I have an issue between two asterisk boxes... One holds a PRI and there's a SIP trunk between the two of them. All outgoing calls from location two go through the SIP trunk then out the PRI |
| 19:59.53 | *** join/#asterisk zamba (marius@flage.org) |
| 19:59.56 | JonnyWhatshisfac | everything was working fine until someone apparently rebooted the boxes this morning and swear they changed nothing - now this is my result: |
| 20:00.10 | JonnyWhatshisfac | Jul 11 14:35:54] NOTICE[5695]: chan_sip.c:15049 handle_request_invite: Failed to authenticate user "Danine Ricereto" <sip:8504739223@12.250.111.154:5060>;tag=as3de0e8d6 |
| 20:00.11 | JonnyWhatshisfac | [Jul 11 14:35:56] WARNING[5852]: channel.c:1958 ast_waitfordigit_full: Unexpected control subclass '9' |
| 20:00.11 | JonnyWhatshisfac | <PROTECTED> |
| 20:00.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:00.37 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:01.41 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 20:01.47 | CyberBob | navaismo here are the RTP debug http://pastebin.com/zhyZ6wWq |
| 20:02.01 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 20:03.49 | navaismo | JonnyWhatshisfac, its like your peer cant athenticate correctly |
| 20:04.03 | CyberBob | RTP packets got and sent on ringing, but one the other party receive calls, it just sent the RTP packets Sent RTP P2P packet to 192.168.3.17:2102 (type 00, len 000160) |
| 20:04.41 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 20:04.48 | igcewieling1 | packets look like they are being sent to a NAT address, not a public address |
| 20:05.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.51) |
| 20:05.19 | JonnyWhatshisfac | navaismo - what's odd is there are no users configured for that trunk... |
| 20:05.29 | JonnyWhatshisfac | im not familiar enough with SIP between asterisk boxes |
| 20:05.41 | CyberBob | igcewieling1 Yes, we've a private address. |
| 20:05.42 | JonnyWhatshisfac | but IF i understand it correctly... |
| 20:05.52 | *** part/#asterisk fischli (~fischli@static-31-25-152-181.ewacom.ropa.net) |
| 20:05.53 | WIMPy | Users for a trunk? What does that even mean? |
| 20:06.12 | JonnyWhatshisfac | WIMPy - no clue... it's giving me this error: |
| 20:06.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.51) |
| 20:06.17 | JonnyWhatshisfac | Jul 11 14:35:54] NOTICE[5695]: chan_sip.c:15049 handle_request_invite: Failed to authenticate user "Danine Ricereto" <sip:8504739223@12.250.111.154:5060>;tag=as3de0e8d6 |
| 20:06.17 | JonnyWhatshisfac | [Jul 11 14:35:56] WARNING[5852]: channel.c:1958 ast_waitfordigit_full: Unexpected control subclass '9' |
| 20:06.17 | JonnyWhatshisfac | <PROTECTED> |
| 20:06.22 | JonnyWhatshisfac | failed to authenticate... |
| 20:06.31 | JonnyWhatshisfac | why would it be trying to authenticate? |
| 20:06.36 | JonnyWhatshisfac | that's on the side the PRI is on |
| 20:07.04 | JonnyWhatshisfac | side the pri is not on will give a similar message: |
| 20:07.05 | JonnyWhatshisfac | WARNING[3542]: chan_sip.c:12774 handle_response_invite: Received response: "Forbidden" from '"Mikel Chano" <sip:8508582008@172.16.0.200>;tag=as2b080c16' |
| 20:07.13 | JonnyWhatshisfac | (different user tried making the call there) |
| 20:07.15 | igcewieling1 | CyberBob: I thought there was NAT involved. |
| 20:07.41 | JonnyWhatshisfac | but one side tells me forbidden... the side with the pri tells me it failed to authenticate the user from the originating side of the sip trunk |
| 20:07.44 | igcewieling1 | JonnyWhatshisfac: rell your users to use the correct passwords |
| 20:07.56 | igcewieling1 | s/rell/tell/ |
| 20:08.11 | JonnyWhatshisfac | igce: those users are on the other system... |
| 20:08.16 | CyberBob | igcewieling1 No, my nat option is set to no. |
| 20:08.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.51) |
| 20:08.29 | JonnyWhatshisfac | and they work for the SIP phones |
| 20:08.40 | JonnyWhatshisfac | they're connected and able to call locally within that asterisk box |
| 20:08.48 | JonnyWhatshisfac | but i cant reach anything on the remote asterisk box |
| 20:08.54 | igcewieling1 | CyberBob: There is no nat between the 172.x,x,x and 10.x.x.x networks? |
| 20:09.15 | igcewieling1 | JonnyWhatshisfac: your two boxes trunks are not set up correct.y |
| 20:09.24 | igcewieling1 | perhaps you need a fromuser= something in the calling side? |
| 20:09.59 | robl^ | igcewieling1: they could be not NATed-- but routed over private network/vpn |
| 20:10.04 | JonnyWhatshisfac | it passes that... :/ |
| 20:10.18 | CyberBob | igcewieling1 10.x.x.x is my VPN, and there is no nat 172.x.x.x |
| 20:10.18 | igcewieling1 | robl^: they could be natted with no localnet |
| 20:10.48 | igcewieling1 | CyberBob: can you ping from the client device to the server as well as the other way around? |
| 20:10.59 | CyberBob | Yes, I can. |
| 20:11.23 | robl^ | igcewieling1: that too. too many possibilities. demand a nettwork diagram |
| 20:11.26 | igcewieling1 | JonnyWhatshisfac: your remote server appears to think sip.conf user [8508582008] is making the call, not your server. |
| 20:11.38 | igcewieling1 | robl^: IIRC this one is using Elastix |
| 20:12.29 | igcewieling1 | try setting fromuser=thesip.conf entry on the remote side and on the remote side set type=friend |
| 20:15.11 | *** join/#asterisk jmls (~julian@77.107.171.82) |
| 20:17.03 | JonnyWhatshisfac | still the same result |
| 20:17.09 | JonnyWhatshisfac | failed to authenticate.. |
| 20:17.17 | JonnyWhatshisfac | it's bewildering. everything has been working for over a year |
| 20:17.23 | JonnyWhatshisfac | and a reboot made it stop functioning... |
| 20:17.26 | JonnyWhatshisfac | lol |
| 20:26.49 | JonnyWhatshisfac | hm |
| 20:26.56 | JonnyWhatshisfac | i dunno, i'm stuck here... |
| 20:27.11 | JonnyWhatshisfac | if i set a sip trunk between two asterisk boxes with type=friend |
| 20:27.15 | igcewieling1 | show us the sip.conf entries for the calling server and the called server on a pastebin |
| 20:27.17 | igcewieling1 | ~pb |
| 20:27.17 | 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. |
| 20:27.20 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 20:27.54 | JonnyWhatshisfac | [meadowrun01] |
| 20:27.54 | JonnyWhatshisfac | type=friend |
| 20:27.54 | JonnyWhatshisfac | context=hosted-trusted |
| 20:27.55 | JonnyWhatshisfac | host=12.23.37.5 |
| 20:27.57 | JonnyWhatshisfac | dtmfmode=auto |
| 20:27.59 | JonnyWhatshisfac | disallow=all |
| 20:28.01 | JonnyWhatshisfac | allow=ulaw |
| 20:28.03 | JonnyWhatshisfac | qualify=yes |
| 20:28.05 | JonnyWhatshisfac | canreinvite=yes |
| 20:28.06 | robl^ | PASTEBIN! |
| 20:28.07 | JonnyWhatshisfac | setvar=calltype=outgoing |
| 20:28.09 | JonnyWhatshisfac | deny=0.0.0.0/0.0.0.0 |
| 20:28.11 | JonnyWhatshisfac | permit=12.23.37.5/255.255.255.255 |
| 20:28.13 | JonnyWhatshisfac | pastebin? |
| 20:28.18 | robl^ | ~pb |
| 20:28.18 | 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. |
| 20:28.19 | igcewieling1 | Can't follow directions either, apparently. |
| 20:28.47 | JonnyWhatshisfac | ahh, sorry. |
| 20:30.58 | JonnyWhatshisfac | http://pastebin.com/PpVVJPiv |
| 20:32.24 | JonnyWhatshisfac | pastebin is a pretty nifty concept... |
| 20:32.33 | JonnyWhatshisfac | beats the hell out of having a thousand lines flooding the channel |
| 20:37.28 | igcewieling1 | I see not fromuser= item like I suggested |
| 20:37.52 | JonnyWhatshisfac | icge: i apologize - i set that and reloaded and had no luck, so i had taken it out... |
| 20:38.04 | igcewieling1 | fromuser=marcuspointe01 on the marcuspointe01 side |
| 20:38.32 | igcewieling1 | also try setting type=peer on that same side |
| 20:40.04 | JonnyWhatshisfac | kk, set type=peer and fromuser=marcuspoint01 on the marcuspointe01 side... still same result |
| 20:45.48 | igcewieling1 | that is what we use, don't know why it isn't working |
| 20:46.32 | *** join/#asterisk joesmoe (~joesmoe@admins.phreefilez.com) |
| 20:46.39 | igcewieling1 | does "sip show peer marcuspointe01 show the correct type and fromuser |
| 20:47.29 | JonnyWhatshisfac | it tells me the peer isn't found... |
| 20:48.13 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 20:48.22 | igcewieling1 | when you have a serious problem. |
| 20:48.32 | igcewieling1 | "sip show peers" will list your peers |
| 20:48.58 | igcewieling1 | qall [something] entries in sip.conf show show in sip show peers, if they don't then they don't exist |
| 20:51.59 | *** join/#asterisk |PiP| (~PiP@CPE68efbdb817c9-CMbc140129a570.cpe.net.cable.rogers.com) |
| 20:52.16 | |PiP| | i know this is off topic, but anyone here familiar with the old school nortel phone systems? |
| 20:54.18 | *** part/#asterisk aelliott22 (~aelliott@198.41.29.45) |
| 20:59.51 | JonnyWhatshisfac | http://pastebin.com/9nZcfR9j |
| 21:00.16 | JonnyWhatshisfac | if im not mistaken, the fromuser= should make it use IP based auth instead of user auth, correct? |
| 21:00.17 | pabelanger | |PiP|: likely a lot of people, just ask your question |
| 21:00.52 | JonnyWhatshisfac | because even still, it fails authentication showing me the caller ID name of the remote SIP peer as the failed auth |
| 21:16.04 | |PiP| | how can i have my line automatically dial *67 (private call) when i pick up a line? |
| 21:18.53 | igcewieling1 | |PiP|: Fisrt set immediate=yes in chan_dahdi.conf (I'm assuming you are running DAHDI, since you didn't say) |
| 21:19.06 | |PiP| | i meant on the norstar system |
| 21:19.24 | igcewieling1 | |PiP|: Oh. Sorry, I should have put you on /gnore. |
| 21:21.49 | JonnyWhatshisfac | lol... |
| 21:22.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:22.48 | igcewieling1 | No point in seeing messages from people who asking off-topic questions which I'll never ever answer. |
| 21:26.01 | drmessano | This isn't "VoIP system support, also maybe Asterisk" |
| 21:26.10 | drmessano | Sorry, left off the hash |
| 21:26.39 | drmessano | Thats my one, federally mandated troll for the day |
| 21:27.57 | artyx | <PROTECTED> |
| 21:28.13 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 21:28.19 | artyx | Nope, it doesn't like that channel, something about the spaces |
| 21:28.35 | artyx | Maybe we should settle for VoIP_system_support_also_maybe_asterisk ? |
| 21:28.35 | drmessano | s/something/everything/ |
| 21:28.59 | artyx | and it'd be TWO hashes |
| 21:29.24 | drmessano | yes |
| 21:29.42 | drmessano | Because nothing is more pretentious than a double-hash channel |
| 21:30.13 | drmessano | Like "Why dont you just /join ##OffTopicDUMBASS" |
| 21:30.43 | artyx | one hash is officially endorsed/support, two hash means its amateur hour right? |
| 21:31.15 | drmessano | Two hashes means what you want to talk about isn't important enough for one hash. As a matter of fact, die in a fire. |
| 21:31.55 | drmessano | If there was a ##asterisk, only a bot would reside there.. telling people they are the next caller in line... and they're not |
| 21:32.14 | artyx | Topical or reference channel names, formatted with two leading hash marks (##), are allocated on a first-come, first-served basis to unofficial groups wishing to discuss a project, group or general topic area. For example, the channel ##linux has been reserved for an unofficial group which uses it to provide GNU/Linux support to visiting users. |
| 21:32.32 | artyx | These channels should not be used as platforms for flaming and trolling. By registering such a channel, you indicate that you assert no legal or informal right to the channel name used. Topical or reference channels should not be registered directly or indirectly by their corresponding projects. You may not register such a channel on freenode without accepting these policies. |
| 21:32.56 | artyx | Note that's for ## ... not the # you can flame and troll away on # according to that clause |
| 21:33.07 | drmessano | You read clauses like I do |
| 21:33.09 | artyx | so, as you were |
| 21:33.28 | artyx | DO you mean read them at all? or group it together or what |
| 21:34.24 | artyx | Uh oh... or any other behaviour meant to deliberately bring upon a person harassment, alarm or distress. We do NOT tolerate discrimination on the grounds of race, religion, gender, sexual preference or other lifestyle choices and run with a zero-tolerance policy for libel and defamation. |
| 21:34.26 | drmessano | you can flame and troll away on # according to that clause <---- |
| 21:34.36 | artyx | Do you think failing to read the directions is a lifestyle choice? |
| 21:34.48 | artyx | grins |
| 21:41.53 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 21:43.10 | JonnyWhatshisfac | so.. does specifying type=friend in a sip trunk force it to use IP authentication as apposed to user authentication? I'm reading this from a source that I'm not certain the reliability of. |
| 21:43.23 | igcewieling1 | this is the discussuon I was trying NOT to start. |
| 21:43.42 | JonnyWhatshisfac | my question? |
| 21:43.47 | JonnyWhatshisfac | or the other discussion? lol |
| 21:45.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.169) |
| 21:58.03 | artyx | Was he trying not to answer your question, or get involved in the lifestyle choice debate (on whether or not "Choosing to not read the directions" is a lifestyle choice |
| 22:01.06 | WIMPy | TGIF! |
| 22:01.20 | [TK]D-Fender | [17:42]JonnyWhatshisfacso.. does specifying type=friend in a sip trunk force it to use IP authentication as apposed to user authentication? I'm reading this from a source that I'm not certain the reliability of. <- no |
| 22:01.33 | [TK]D-Fender | JonnyWhatshisfac: peer=ip first, fried = peer+user |
| 22:09.27 | JonnyWhatshisfac | Now, when you trunk asterisk boxes together like this... Someone is claiming to me that they'll automatically know which extensions are on which end of the trunk. Is this true? |
| 22:10.23 | WIMPy | No |
| 22:10.36 | JonnyWhatshisfac | yeah, i thought that seemed a little far fetched... |
| 22:10.39 | WIMPy | But you can set up DUNDi to get the magic. |
| 22:10.57 | JonnyWhatshisfac | ahh, DUNDi is setup... is that maybe what he was referring to? |
| 22:11.10 | WIMPy | Probably. |
| 22:11.20 | WIMPy | You can also do it via IAX. |
| 22:12.05 | WIMPy | And unless you have the hope to be able to get directmedia working, that would be the better way to link Astersks together. |
| 22:13.06 | JonnyWhatshisfac | WIMPy: Right now, I have an issue where a client has two asterisk boxes put together via SIP trunks - type=friend . PRI exists at Marcus Pointe location. Meadow Run's calls, incoming and outgoing, go through the SIP trunk to Marucs Pointe and through the PRI there. |
| 22:13.14 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.181) |
| 22:13.25 | JonnyWhatshisfac | Everything has been working until a reboot this morning... (working since 2010 - first reboot) |
| 22:13.38 | JonnyWhatshisfac | and now i'm getting authentication failures when I attempt to call through the SIP trunk... |
| 22:13.51 | JonnyWhatshisfac | Should the SIP users exist at both sides? or where is it trying to authenticate that at? |
| 22:14.01 | JonnyWhatshisfac | Wondering if I might be better off moving this crap over to IAX ... |
| 22:14.21 | JonnyWhatshisfac | Only problem is the entire asterisk config is generated dynamically via perl scripts that parse CSV's and build it on the fly |
| 22:14.59 | WIMPy | Sure they need to exist on both ends. |
| 22:15.50 | WIMPy | And IAX has some advantage like using less overhead when you have simultaneous calls and you get charging information across. |
| 22:18.51 | JonnyWhatshisfac | ooohhh... in the dundi.conf... I'm noticing that the entry name appears to be the MAC address of the remote host's NIC holding the peer IP |
| 22:18.54 | JonnyWhatshisfac | is that correct? |
| 22:21.13 | WIMPy | yes |
| 22:21.21 | JonnyWhatshisfac | MAC was wrong on one end... |
| 22:21.25 | WIMPy | The entityID |
| 22:21.29 | JonnyWhatshisfac | just changed that. |
| 22:21.38 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:21.55 | WIMPy | I.e. you have to configure that manually. It doesn;t check the MAC by itself. |
| 22:22.16 | JonnyWhatshisfac | so, SIP account originating the call from the remote end is MR9998 - is that to say that MR9998 needs to also exist on the box holding the PRI? |
| 22:22.31 | WIMPy | Yes. |
| 22:23.01 | WIMPy | If you use DUNDi, yiu don;t have to use accounts however. It uuses generated passwords by default. |
| 22:23.45 | JonnyWhatshisfac | ahh, kk... |
| 22:23.49 | WIMPy | Actually I'm not sure you can make it use a predefined account. |
| 22:24.44 | JonnyWhatshisfac | hmm.. |
| 22:24.46 | JonnyWhatshisfac | Jul 11 17:24:33] NOTICE[5080]: chan_sip.c:15049 handle_request_invite: Failed to authenticate user "Mikel Chano" <sip:8508582008@12.250.111.154:5060>;tag=as59e73df9 |
| 22:24.53 | JonnyWhatshisfac | still same thing on the side holding the PRI |
| 22:25.02 | JonnyWhatshisfac | this box really hates me... :) |
| 22:25.31 | [TK]D-Fender | just using a peer between 2 boxes is easy. You have a simple auth issue |
| 22:25.59 | JonnyWhatshisfac | is there a way to get rid of the auth and just force asterisk to trust the calls coming in from the remote peer? lol |
| 22:26.14 | [TK]D-Fender | DUNDi is a nifty gimmick... but never seen a solid need for it... |
| 22:26.15 | JonnyWhatshisfac | i tried setting the type= to peer with no luck |
| 22:26.23 | JonnyWhatshisfac | i'm back to friend on both ends |
| 22:26.30 | [TK]D-Fender | boht should be : |
| 22:27.21 | [TK]D-Fender | type=peer , fromuser=[name on other side] , sendrpid=yes, trustrpid=yes , insecure=port,invite |
| 22:28.25 | *** join/#asterisk blizzow1 (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 22:29.02 | JonnyWhatshisfac | the fromuser= should be the IP of the remote end, correct? |
| 22:29.08 | JonnyWhatshisfac | or the name? |
| 22:29.17 | [TK]D-Fender | NAME |
| 22:29.35 | JonnyWhatshisfac | kk, i'm going to modify it to reflect that on both ends and give a whirl... |
| 22:30.47 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 22:32.54 | JonnyWhatshisfac | lol.. |
| 22:32.56 | JonnyWhatshisfac | same result |
| 22:33.24 | [TK]D-Fender | Show us your configs and the calls |
| 22:33.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.193) |
| 22:33.40 | [TK]D-Fender | "same result" deosn't prove anything was done right or the circumstances are sane |
| 22:34.06 | Katty | even folks |
| 22:34.09 | Katty | fender bender. |
| 22:35.07 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 22:35.18 | [TK]D-Fender | return to sender ;) |
| 22:35.27 | gusto | retoure |
| 22:36.02 | gusto | WIMPy: are you there? |
| 22:36.23 | WIMPy | No. I'm here. |
| 22:36.49 | *** join/#asterisk jonmasters (~jcm@edison.jonmasters.org) |
| 22:36.57 | Katty | hugs WIMPy |
| 22:37.16 | gusto | so |
| 22:37.31 | Katty | so, gusto> |
| 22:37.33 | gusto | now everytime i call someone on the phone i will also greet the NSA |
| 22:38.14 | WIMPy | I think GCHQ will hear you more often than the NSA. |
| 22:38.18 | JonnyWhatshisfac | http://pastebin.com/uZpSGEdY |
| 22:38.29 | gusto | is that the british one |
| 22:38.30 | JonnyWhatshisfac | i put the entries from the sip.conf as well as the debug output from the call on both ends |
| 22:38.31 | gusto | ha? |
| 22:38.40 | WIMPy | it is |
| 22:38.43 | gusto | ok |
| 22:38.45 | gusto | however |
| 22:38.53 | gusto | i just have to make fun of them |
| 22:39.06 | gusto | or i ll just greet the BND, thats fine with them too |
| 22:39.24 | JonnyWhatshisfac | garbled up in the output from meadow run, you'll see: [Jul 11 17:36:45] WARNING[4010]: chan_sip.c:12774 handle_response_invite: Received response: "Forbidden" from '"Mikel Chano" <sip:marcuspointe01@172.16.0.200>;tag=as13f952e8' |
| 22:41.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.193) |
| 22:59.57 | JonnyWhatshisfac | ahhh... |
| 22:59.59 | JonnyWhatshisfac | wait a sec |
| 23:00.05 | JonnyWhatshisfac | dundi says the peer is unreachable.. |
| 23:00.19 | JonnyWhatshisfac | if dundi stops working right, would that cause me authentication problems like the ones i'm seeing? |
| 23:00.43 | JonnyWhatshisfac | if i'm not mistaken, dundi tells the remote end that the extension is over there, and what account to use to reach it, right? |
| 23:00.51 | WIMPy | *If* you expect DUNDi to give you the route, yes. |
| 23:01.11 | WIMPy | The account is the dundi account. |
| 23:01.30 | JonnyWhatshisfac | EID Host Model AvgTime Status |
| 23:01.30 | JonnyWhatshisfac | 00:13:72:4d:78:f4 12.23.37.5 (S) Symmetric Unavail UNREACHABLE |
| 23:01.33 | WIMPy | The password is included in the dundi response. |
| 23:01.35 | JonnyWhatshisfac | so seeing that is a bad thing? |
| 23:01.38 | JonnyWhatshisfac | lol :) |
| 23:01.45 | WIMPy | Probably |
| 23:02.02 | JonnyWhatshisfac | the other end says it's OK, though |
| 23:02.06 | JonnyWhatshisfac | how can one side show unreachable.. |
| 23:02.09 | JonnyWhatshisfac | and the other okay? |
| 23:02.31 | WIMPy | Wrong hostname? |
| 23:02.55 | WIMPy | Or if it's dynamic missing registration. |
| 23:06.49 | JonnyWhatshisfac | [Unable to look up host 'meadowrun02.ppbx.bignetworks.com' |
| 23:06.49 | JonnyWhatshisfac | meadowrun02*CLI> |
| 23:07.00 | JonnyWhatshisfac | i think that'd be a bit of an issue lol |
| 23:11.42 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 23:11.47 | [TK]D-Fender | I'd forget dundi, go verify your DNS, and place the next call with SIP DEBUG enabled... |
| 23:11.54 | [TK]D-Fender | since we aren't seeing the actual comms |
| 23:15.07 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:18.42 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
| 23:21.13 | *** join/#asterisk guitarHester (~guitarHes@mobile-166-147-114-094.mycingular.net) |
| 23:28.42 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 23:29.29 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:29.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:41.48 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 23:55.31 | floren | i just had my device probed by sipvicious :) |
| 23:55.33 | floren | # dig -x 66.90.40.36 +short |
| 23:55.34 | floren | node.smartcityon-line.com. |
| 23:56.08 | WIMPy | The first time today? |
| 23:56.49 | floren | to solve this i changed the rules in iptables to allow only the voip provider and local network to access asterisk ports |
| 23:57.07 | floren | sipvicious is not vicious anymore :) |
| 23:57.58 | floren | WIMPy: several times during the day. i noticed this by accident, i left the cli with debug on |
| 23:58.12 | floren | when i came back from work i saw all the probing done |
