| 00:04.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:27.13 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 01:25.41 | *** join/#asterisk shootbird (~quassel@205.185.122.214) |
| 01:28.52 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-ymcamgcxdoypwdzl) |
| 02:10.45 | *** join/#asterisk F2Knight (~F2Knight@wsip-98-189-15-166.oc.oc.cox.net) |
| 02:13.05 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 02:16.09 | *** join/#asterisk raspberrypifan (~raspberry@cpe-66-108-70-5.nyc.res.rr.com) |
| 02:17.02 | raspberrypifan | Samot: |
| 02:17.04 | raspberrypifan | hello |
| 02:26.52 | Samot | Yes? |
| 02:34.08 | drmessano | One Pi Per GSM Modem |
| 02:34.11 | drmessano | Its the only way |
| 02:34.36 | Samot | Lol |
| 02:34.46 | drmessano | Name them after Dwarves |
| 02:34.53 | raspberrypifan | lol |
| 02:34.56 | raspberrypifan | im moving to a digital cloud |
| 02:35.02 | drmessano | A digital one? |
| 02:35.03 | WIMPy | What flavour Pi? |
| 02:35.07 | raspberrypifan | not for pis |
| 02:35.11 | raspberrypifan | for commercial traffic |
| 02:35.13 | Samot | Vs an analog cloud |
| 02:35.17 | drmessano | Whoa.. Digital.. Sounds fancy |
| 02:35.22 | raspberrypifan | i know gotta be fancy |
| 02:35.27 | drmessano | 100101010101000101110010101 |
| 02:35.49 | drmessano | Wait, thats not a cloud |
| 02:35.54 | drmessano | (100101010101000101110010101) |
| 02:35.58 | drmessano | There |
| 02:36.37 | raspberrypifan | so should i add one ip per user |
| 02:36.47 | drmessano | Oh at least |
| 02:36.55 | drmessano | Wait, what? |
| 02:37.21 | drmessano | one ip per user? |
| 02:37.29 | Samot | What? |
| 02:37.36 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 02:37.57 | Samot | Can that come with context? |
| 02:38.22 | WIMPy | suggests an additional /30 transfer net per user. |
| 02:38.32 | WIMPy | Contexts go to extensions.conf. |
| 02:38.47 | raspberrypifan | for the deployment sheet |
| 02:38.55 | drmessano | lol |
| 02:39.06 | Samot | Oh you mean that thing from like 8 hours ago? |
| 02:39.14 | drmessano | Asterisk Config ????? IP Address ==> Deployment Sheet |
| 02:39.16 | drmessano | I get it now |
| 02:39.22 | WIMPy | Why be so negative? What about protexts? |
| 02:39.45 | Samot | Its very simple. |
| 02:39.51 | drmessano | subtexts as well, for lower volume users |
| 02:40.00 | Samot | Those are the ips you need to peer to. |
| 02:40.22 | Samot | If you want origination or termination traffic with them. |
| 02:42.09 | Samot | Heres a bit of advice, when your upstream provider says "these are the ips you need to peer with us" you use them. |
| 02:43.53 | raspberrypifan | yea i mean i get it |
| 02:43.56 | raspberrypifan | but i mean the configuration details |
| 02:44.08 | raspberrypifan | one of these |
| 02:44.11 | raspberrypifan | [TheITSP-In] |
| 02:44.11 | raspberrypifan | type=peer |
| 02:44.11 | raspberrypifan | host=inbound.theitsp.com |
| 02:44.13 | raspberrypifan | insecure=port,invite |
| 02:44.15 | raspberrypifan | context=incoming |
| 02:44.17 | raspberrypifan | per ip? |
| 02:44.18 | Samot | IUhm. |
| 02:44.20 | Samot | Yes. |
| 02:44.28 | Samot | Only if you want to accept traffic from them |
| 02:44.32 | Samot | Or send traffic to them. |
| 02:44.42 | Samot | I mean you could just use one or two... |
| 02:44.50 | Samot | But I'm sure you won't get all your origination. |
| 02:45.14 | raspberrypifan | so 5 sections for origination because there are 5 ips from them |
| 02:45.15 | Penguin | How many do they have? |
| 02:45.21 | raspberrypifan | http://pastebin.com/9mv1M3Lw |
| 02:45.41 | Penguin | If they have 5, you'd need 5 sections. I recommend templating 4 of them. |
| 02:45.55 | raspberrypifan | the only change would the ip though right? |
| 02:46.07 | WIMPy | Only 4 of 5? |
| 02:46.16 | Samot | First off there's only 4 |
| 02:46.25 | Samot | Second off, 2 of them do both orig/term |
| 02:46.29 | Samot | So that's just two trunks. |
| 02:46.31 | raspberrypifan | gues your right lol |
| 02:46.32 | Penguin | Then I'd set up one and template the other 3. |
| 02:46.47 | raspberrypifan | ok cool got it |
| 02:46.51 | Penguin | Seems like we're narrowing down. |
| 02:47.08 | WIMPy | I'd make a template and use it in all 4 peers. |
| 02:47.11 | Samot | But most importantly, if you can't figure out how to peer to your upstream provider how are you even thinking about selling people voice service? |
| 02:47.47 | drmessano | Do you have your MagicJack cluster in place for backup? |
| 02:48.01 | Samot | Ooma. |
| 02:48.11 | drmessano | Ooma God |
| 02:49.10 | raspberrypifan | well im just getting started in doing commercial work |
| 02:49.19 | Samot | Vs? |
| 02:49.23 | Samot | Hobbist work? |
| 02:50.01 | raspberrypifan | yea hobbist on the pi |
| 02:50.28 | Samot | #triggered |
| 02:50.47 | raspberrypifan | why triggered |
| 02:50.52 | drmessano | Probably best to get concepts down before going pro |
| 02:51.27 | Samot | Because before going freelance I worked for an OC. |
| 02:51.45 | drmessano | If you dont know how it works, you're probably going to kill someone |
| 02:53.41 | Samot | And not rely on #asterisk and #freepbx as your tech support dept. |
| 02:54.29 | drmessano | Youre talking about providing a public utility to someone |
| 02:54.37 | drmessano | Not just hosting the website for their bingo club |
| 02:54.40 | Samot | VoIP != web hosting |
| 02:55.38 | raspberrypifan | well ill tell u what |
| 02:55.47 | raspberrypifan | im the most experienced and professional guy in this rag tag team |
| 02:56.21 | Penguin | That's supposed to be impressive? |
| 02:56.25 | raspberrypifan | no |
| 02:56.27 | raspberrypifan | its supposed to be sad |
| 02:56.29 | raspberrypifan | very very sad |
| 02:56.37 | drmessano | Oh |
| 02:56.37 | Penguin | Oh, thanks for clarification. |
| 02:56.41 | drmessano | We were already sad |
| 02:56.42 | Samot | Then why is anyone toying with this idea? |
| 02:57.42 | drmessano | $$$$ |
| 02:57.46 | raspberrypifan | exactly |
| 02:57.48 | drmessano | Asterisk is a money printing machine |
| 02:57.50 | raspberrypifan | and im doing it just for the sport of it |
| 02:57.56 | raspberrypifan | im not much into money |
| 02:57.57 | Samot | OMFG. |
| 02:58.48 | Samot | This isn't a game. |
| 02:58.54 | Samot | You are not LeBron James. |
| 02:58.59 | raspberrypifan | i dont wnat money |
| 02:59.02 | drmessano | Sure it is |
| 02:59.07 | raspberrypifan | they can keep the money on their side |
| 02:59.09 | raspberrypifan | im just using it to learn |
| 02:59.09 | drmessano | Telephone services = LOL |
| 03:00.17 | drmessano | So build an Asterisk box.. buy a bunch of ATAs for your family members, and set them all up a "Bat Phone" to play around with to call each other |
| 03:00.28 | drmessano | Thats one way to learn |
| 03:00.49 | drmessano | Not in the actual process of providing someone vital telephone service |
| 03:00.51 | Samot | Don't learn while Grandma Smith needs to call 911 and can't because you can't peer to a carrier. |
| 03:01.09 | Penguin | I bought a PAP2T on ebay. |
| 03:01.14 | raspberrypifan | n one is gonna be making 911 on this |
| 03:01.15 | Penguin | It was a counterfeit. |
| 03:01.18 | Penguin | It's junk. |
| 03:01.23 | raspberrypifan | it wont be doing anything in the us actually |
| 03:01.27 | raspberrypifan | not on my end anyway |
| 03:01.44 | drmessano | Wow |
| 03:02.02 | drmessano | So since no one is actually going to be dialing actual 911 |
| 03:02.03 | drmessano | Its legit |
| 03:02.08 | raspberrypifan | no its not legit |
| 03:02.38 | Samot | So you don't plan on having business or residential customers? |
| 03:03.01 | drmessano | Let me explain this weird sort of paradox |
| 03:03.04 | raspberrypifan | not directly |
| 03:03.15 | drmessano | In the old days, we all had copper phones |
| 03:03.17 | drmessano | and it was good |
| 03:03.23 | drmessano | Then we all moved to cell phones |
| 03:03.24 | drmessano | and it was good |
| 03:03.33 | drmessano | But in some places, cell phones are not so good |
| 03:03.50 | Samot | Either because of service or GPS issues. |
| 03:04.06 | drmessano | So you find some people migrating their home phone to VoIP or going back to something they consider a backup for their cell phone |
| 03:04.33 | drmessano | Which in a lot of ways make it more important now than before |
| 03:04.44 | drmessano | Thats a crazy thing to take as a game |
| 03:05.29 | drmessano | Even if we're not talking about LITERAL 911 |
| 03:06.47 | Penguin | Maybe like 611 instead. |
| 03:07.04 | drmessano | or calling your doctor to get a medication refilled |
| 03:07.05 | Samot | Or whatever country the emergency service is. |
| 03:07.08 | Penguin | No, wait. That's customer service or something. |
| 03:07.11 | Penguin | 311? |
| 03:07.18 | Penguin | Yeah, 311 I think. |
| 03:07.24 | drmessano | or calling your kid because you feel dizzy and feel stupid calling 911 |
| 03:07.36 | drmessano | or anything short of 911 |
| 03:08.57 | drmessano | I have elderly parents.. I could go through lots of scenarios that involved "This was important, I didnt want to use my cell phone for it" |
| 03:16.36 | *** part/#asterisk beek (~klinebl@96.227.183.9) |
| 03:51.06 | *** join/#asterisk F2Knight_ (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 04:19.02 | raspberrypifan | so im trying to get g729 to load |
| 04:19.05 | raspberrypifan | and al i get is Command 'module load codec_g729.so' failed. |
| 04:20.44 | raspberrypifan | nvm dumb me |
| 04:25.14 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-woheekhnzkomprqq) |
| 04:26.17 | Samot | Considering it requires a license per channel. |
| 04:27.59 | raspberrypifan | well more so then that i still cant get it to load |
| 04:28.09 | raspberrypifan | it seems to load fine but when i go to show translations its not there |
| 04:28.19 | raspberrypifan | but it is listed in module show |
| 04:32.28 | raspberrypifan | copy protection, so there is some issue there yet im just copying from an old asterisk to a new one and as far as i can remember on the old they did make any changes |
| 04:34.21 | *** join/#asterisk beachdog (~beachdog@ip72-197-114-232.sd.sd.cox.net) |
| 04:35.24 | Samot | You dont have a license for it. |
| 04:35.43 | Samot | Its not going to work. |
| 04:36.08 | raspberrypifan | but on the other box it does work |
| 04:36.20 | raspberrypifan | and there arent any licenses installed as far as i can tell |
| 04:36.39 | raspberrypifan | i had someone help me do the install on the other box and he jsut downloaded it and installed it |
| 04:38.32 | raspberrypifan | i wouldnt have set it up that way if i knew about licensing |
| 04:40.36 | Samot | G729 has always required one. |
| 04:40.45 | raspberrypifan | idk how he did it then |
| 04:40.53 | raspberrypifan | everywhere in the conf its set up for g729 |
| 04:41.13 | raspberrypifan | where would the license file be |
| 04:41.24 | raspberrypifan | or how would it know |
| 04:42.18 | WIMPy | You should know if you bought any. |
| 04:42.30 | raspberrypifan | i didnt but its clearly using g729 |
| 04:42.48 | Samot | So tell it not to use g729. |
| 04:43.33 | raspberrypifan | well i got g723 g726 and g711 |
| 04:43.36 | raspberrypifan | which is the best among those |
| 04:43.43 | raspberrypifan | besdies 729 of course |
| 04:44.01 | WIMPy | G.711 |
| 04:44.45 | raspberrypifan | mu or a? |
| 04:45.18 | Samot | Depends on were you are and what supported. |
| 04:45.32 | WIMPy | Almost the same. Usage depends on geographic region. |
| 04:45.47 | Samot | er where. |
| 04:45.50 | raspberrypifan | the gateway is in south america but ill be selling to international customers |
| 04:46.24 | Samot | Then you should start a-Googlin'. |
| 04:47.14 | raspberrypifan | alirght hten |
| 04:47.20 | Samot | We weren't joking. |
| 04:47.23 | Samot | Learn the basics. |
| 04:47.33 | Samot | While you might be able to sell a service that you don't understand.. |
| 04:47.39 | Samot | You can't support a service you don't understand. |
| 04:48.52 | raspberrypifan | i did read the book and ill reread it i guess |
| 04:49.07 | Samot | Did you understand it? |
| 04:49.28 | Samot | Reading without comprehension is not learning. |
| 04:49.47 | raspberrypifan | i did understand but i did not pay detailed attention |
| 04:49.50 | raspberrypifan | ill just re read |
| 04:50.18 | Samot | And this is the most technical and professional of the group. |
| 04:50.32 | raspberrypifan | yup |
| 04:50.53 | Samot | No one's really invested money in this, right? |
| 04:51.06 | raspberrypifan | theyve invested plenty |
| 04:52.55 | Samot | Typical. I'm digging a wireless carrier out of basically the same hole. |
| 04:54.13 | raspberrypifan | well im actually willing to ask |
| 04:54.22 | raspberrypifan | online where to find information and how to google |
| 04:54.27 | raspberrypifan | the next one down the line doesnt know how to do that |
| 04:55.11 | Samot | Yup and that's why this wireless carrier is out a couple hundred thousand. |
| 04:55.38 | Samot | Because the company they hired for their SIP/VoIP expansion read a book and sometimes know how to goolge. |
| 04:55.43 | Samot | Because the company they hired for their SIP/VoIP expansion read a book and sometimes know how to google. |
| 04:56.08 | raspberrypifan | i know how to google all |
| 04:56.59 | Samot | How long they been doing this? |
| 04:57.01 | WIMPy | Well, doesn;t seem to have worked for the simple peer setup within a day. |
| 04:57.41 | Samot | Even more. He was asking about it Saturday night as well. |
| 04:57.51 | WIMPy | In how many years do you plan to go to production? |
| 04:58.00 | raspberrypifan | 12 hours |
| 04:58.04 | raspberrypifan | dont ask me |
| 04:58.11 | raspberrypifan | im not the one responsible for the business |
| 04:58.15 | raspberrypifan | go ask my uncles |
| 04:58.24 | Samot | Oh great. It's a family business. |
| 04:58.29 | raspberrypifan | its not even that |
| 04:58.35 | Samot | This wasn't your idea was it? |
| 04:58.39 | raspberrypifan | no |
| 04:59.08 | Samot | How long have they been working on this little venture? |
| 04:59.34 | raspberrypifan | well somehow they managed to get it work many years ago |
| 04:59.37 | raspberrypifan | with landlines |
| 04:59.53 | Samot | That has nothing to do with this. |
| 05:00.01 | Samot | It is completely different beasts. |
| 05:00.11 | raspberrypifan | well they did the same grey routing |
| 05:00.21 | raspberrypifan | and they had some kind of softswitch |
| 05:00.25 | raspberrypifan | idk what it was |
| 05:00.59 | Samot | So basically this is bleeding money. |
| 05:01.29 | raspberrypifan | small ammounts |
| 05:01.35 | raspberrypifan | im only stepping because its bleeding |
| 05:01.40 | raspberrypifan | and i at least know how to google |
| 05:01.43 | Samot | You're not stopping it. |
| 05:01.49 | raspberrypifan | im doing what i can with the tools i have |
| 05:01.50 | Samot | You will most likely add to it. |
| 05:02.01 | raspberrypifan | i doubt it |
| 05:02.09 | Samot | Uhm. |
| 05:02.13 | Samot | You will. |
| 05:02.17 | raspberrypifan | how |
| 05:02.18 | Samot | This is what I do. |
| 05:02.23 | Samot | By not knowing anything. |
| 05:02.32 | Samot | Right now you've wasted two days on a simple peering issue. |
| 05:02.33 | raspberrypifan | if its working its working |
| 05:02.37 | raspberrypifan | oh plz |
| 05:02.37 | Samot | That's bleeding money. |
| 05:02.41 | raspberrypifan | we havent gotten that far |
| 05:02.56 | raspberrypifan | i was asking ahead of time |
| 05:03.03 | raspberrypifan | like seriously you have no idea how malformed this whole thing is |
| 05:03.04 | *** join/#asterisk F2Knight (~F2Knight@wsip-98-189-15-166.oc.oc.cox.net) |
| 05:03.12 | Samot | Then they should fold it. |
| 05:03.19 | raspberrypifan | they wont |
| 05:03.21 | raspberrypifan | there desperate for money |
| 05:03.38 | Samot | Yeah, they should stop now. |
| 05:04.26 | Samot | Or if they are really serious, they hire someone. |
| 05:04.40 | Samot | That knows what Telephony and SIP need. |
| 05:04.51 | WIMPy | Is it one of those "I heard someone got rich with that. I want that too!"-ventures? |
| 05:04.51 | raspberrypifan | Samot: whats the point in tearing me down |
| 05:05.35 | raspberrypifan | like i said they managed with landlines |
| 05:05.39 | raspberrypifan | the only change is gsm gateways now |
| 05:05.44 | Samot | Landlines are not the same as SIP |
| 05:05.56 | raspberrypifan | landline gateways |
| 05:05.59 | raspberrypifan | with sip |
| 05:06.07 | Samot | There was a reason the Operating Carrier here in Michigan bought the ITSP I managed. |
| 05:06.13 | Samot | Because landlines != SIP |
| 05:06.27 | WIMPy | That's even worse. |
| 05:07.13 | Samot | Well it's one of those that 90% of the staff and ALL of the sales staff were old school AT&T people. |
| 05:07.50 | Samot | That really believed that voip was a "tech fad" that had no real place in Telephony. |
| 05:08.15 | WIMPy | True story. |
| 05:08.23 | Samot | We came in and the first thing we did... |
| 05:08.40 | Samot | Moved their International off TDM to SIP |
| 05:09.01 | Samot | And we had to FIGHT for it. |
| 05:09.33 | Samot | Then the CFO saw the price difference and was like "Why didn't anyone ever say this was a cost saving measure?!" |
| 05:10.35 | Samot | But they took a PRI approach with SIP trunking.. |
| 05:10.41 | Samot | Which was like "WUT?!" |
| 05:11.44 | Samot | So you had to pay per voice channel and per minute outbound. |
| 05:12.00 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 05:13.11 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 05:19.37 | raspberrypifan | heres any interesting question i keep getting res_rtp_asterisk.c:4248 ast_rtp_read: RTP Read too short and the calls drop |
| 05:27.16 | *** join/#asterisk klow (~textual@c-98-247-49-57.hsd1.wa.comcast.net) |
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| 07:59.40 | *** join/#asterisk stefan27 (~stefan27@212.247.4.149) |
| 08:00.10 | stefan27 | is it possible to run two cli commands with one call to asterisk -rx "<cmds>" ? |
| 08:05.01 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 08:14.57 | nix8n82 | stefan27: yes it's possible, but one at time. |
| 08:18.12 | nix8n82 | Sorry misunderstood the question. Probably not, why would you need to run two commands with one call? |
| 08:20.04 | *** join/#asterisk F2Knight (~F2Knight@c-50-139-85-237.hsd1.or.comcast.net) |
| 08:28.12 | stefan27 | an existing script currently runs the same asterisk command on multiple remote machines |
| 08:28.27 | stefan27 | I just have to modify it then |
| 08:31.15 | nix8n82 | Just curious what two commands are you trying to run? |
| 08:32.30 | stefan27 | atm "module show like rtp" and "core show channels" |
| 08:32.46 | stefan27 | but I just started debugging |
| 08:33.16 | stefan27 | There seems to be a udp port leak that occur very seldomly |
| 08:34.02 | stefan27 | I expect the use count of res_rtp_asterisk to be proportional to the core show channel count when everything is okay |
| 08:34.26 | stefan27 | I don't know what causes the leaks; I want to be able to find out which calls causes it |
| 08:35.58 | stefan27 | but for the leaked ports as reported by netstat and lsof and ls /proc/<asterisk-pid>/fd I can't find out how long the bad sockets have been open |
| 08:36.45 | nix8n82 | Are you running out of ports? |
| 08:37.08 | stefan27 | yeah |
| 08:37.35 | stefan27 | but very slowly... 200 leaked sockets after 1 million calls |
| 08:38.23 | nix8n82 | What version of asterisk? |
| 08:39.58 | stefan27 | 13.5.0 but unfortunately there are some custom patches applied to it |
| 08:43.56 | *** join/#asterisk angryuser (~greg@LNeuilly-656-1-71-123.w92-154.abo.wanadoo.fr) |
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| 08:48.10 | nix8n82 | That is above my level. But good luck with that. |
| 08:48.36 | stefan27 | thanks i'll investigate |
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| 09:49.24 | wyoung | hi gang |
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| 10:47.01 | stefan27 | is there any way to get the creation timestamp for a sip dialog object as reported by sip show objects? |
| 10:50.47 | wyoung | ? |
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| 10:55.01 | stefan27 | CLI command "sip show objects" outputs objects like name: SDj5qjd01-f0d598da1c3fd8cde801d733e94987a8-agu9f4f type: dialog ... I want to know when they were created |
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| 11:22.43 | HCorn | Hi! I am upgrading from Asterisk 11 to Asterisk 13 (from chan_agent to app_agent_pool) and am trying to send an incoming call to a queue which has one agent. I am using realtime and I can login the agent and hear music on hold. The agent can also join the incoming queue and everything looks fine until I send an incoming call to the queue. The agent then hears a beep but then the agents |
| 11:22.43 | HCorn | connection gets disconnected and the agent gets logged out. I see this in the log when the incoming call is placed in the queue: "<SIP/agent1-00000000> Playing 'beep.alaw' (language 'en')", then "Channel SIP/agent1-00000000 left 'holding_bridge' agent_hold-bridge" and "Channel Local/agent1@agents-00000000;2 left 'simple_bridge' basic-bridge". After this the agent is disconnected and the agent |
| 11:22.43 | HCorn | logged out. Does anyone have any clues on how to keep the agents connection from disconnecting? |
| 11:24.04 | Samot | Hcorn: Show a call |
| 11:24.09 | Samot | ~pb |
| 11:24.09 | 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. |
| 11:24.55 | HCorn | Samot: not sure how you mean? complete SIP messaging? |
| 11:26.20 | Samot | A debug |
| 11:26.24 | Samot | asterisk -rvvvvvvvvvv |
| 11:26.26 | Samot | sip set debug on |
| 11:26.36 | Samot | make a call that goes to an agent. |
| 11:26.43 | Samot | Show how the agent gets logged out |
| 11:28.30 | HCorn | Alright |
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| 11:33.51 | HCorn | Samot: In the beginning of this log I am answering an outgoing call to my cell and place it into the queue, and the agent gets logged out. http://pastebin.com/cdLy1xPK |
| 11:34.59 | Samot | OK, so do it again. |
| 11:35.11 | Samot | This time SHOW the call. |
| 11:35.17 | Samot | The whole call. |
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| 11:40.11 | HCorn | Samot: Alright, here you go: http://pastebin.com/kbh3PvPq |
| 11:43.06 | Samot | So the Agent is connected via WS? |
| 11:43.14 | HCorn | Yes |
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| 11:45.21 | Samot | Why is the system trying to do T.38 when replying to your call? |
| 11:46.35 | HCorn | I am not sure.. |
| 11:47.38 | Samot | <PROTECTED> |
| 11:47.59 | Samot | Then there is this: /home/asterisk/hangup.php: Failed to execute '/home/asterisk/hangup.php': Permission denied |
| 11:48.23 | HCorn | Yes, but all of that is after the actual problem |
| 11:48.39 | HCorn | There is no problem if I make an outbound call via WS |
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| 11:49.06 | HCorn | But in this case the agent only hears the beep, then gets disconnected. |
| 12:00.15 | Samot | Can you call the Agents directly? |
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| 12:01.22 | HCorn | I haven't tried that, but I can make a quick test |
| 12:04.10 | Samot | Show that call |
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| 12:16.11 | Samot | ? |
| 12:24.18 | HCorn | I had to do quite a lot of modidications to do that, so I'm thinking of downgrading again instead. The setup worked fine in previous Asterisk versions. |
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| 12:24.49 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:25.43 | Samot | Is there a reason you're only updating to a non current release? |
| 12:26.05 | Samot | This is for everyone else: What release had the issues with the Queues? |
| 12:26.38 | Samot | Was it a 13.7.x or a 13.8.x release? |
| 12:26.38 | HCorn | No, it was the current release when I downloaded it to my development server.. |
| 12:26.53 | Samot | By that you mean a package. |
| 12:27.09 | HCorn | Yep |
| 12:27.49 | Samot | And I think 13.8 is the one with the queue bug. |
| 12:28.37 | HCorn | Hmm, I see, so you think there is a chance of my setup working in the latest version? |
| 12:28.54 | Samot | 13.9.1, yes. |
| 12:29.09 | HCorn | Alright, I'll give that a try! Thank you for the help! |
| 12:29.25 | Samot | Unless the issue is totally related to the websocket connection. |
| 12:29.35 | Samot | But it would be a good first step. |
| 12:30.03 | HCorn | Yes, worth a try anyway |
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| 13:31.42 | stefan27 | On a (unpatched, clean) asterisk 13.5.0 I seem to have got about 50 rtp socket leaks after ~50 000 calls. ('netstat -tupan | grep asterisk' reports 100 entries, and 'asterisk -rx "module show like rtp' reports 50 in Use Count when asterisk is idle and no channels are up. It seems these res_rtp_asterisk use count match the count of "sip show objects" | grep dialog ; which displays just over |
| 13:31.42 | stefan27 | 50 sip dialogs, ~50 of which have refcount=1 |
| 13:32.25 | [TK]D-Fender | LTS: 13.9.1 (2016/05/13) <----- |
| 13:32.28 | [TK]D-Fender | upgrade time... |
| 13:32.36 | stefan27 | How can I debug this further on that asterisk -- if i recompile with REF_DEBUG i probably won't reach necessary volume to reproduce the issue? |
| 13:33.46 | Rasputin3711 | [TK]D-Fender: Have you updated all your servers to 13.9.1? |
| 13:33.59 | stefan27 | Yes, upgrading is on our agenda; but I couldn't find any JIRA addressing rtp leaks from 13.5.0 to 13.9.1 |
| 13:34.17 | [TK]D-Fender | Rasputin3711, One to go... |
| 13:34.42 | [TK]D-Fender | stefan27, we don't tend to debug old releases, and 13.5.0 was quite a while ago |
| 13:38.19 | stefan27 | Okay, I'll make a ticket out of it after I reproduced it on the latest version |
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| 14:04.06 | HCorn | Samot: Upgraded Asterisk to 13.9.1, but still no luck unfortunately. Thanks for the help anyway! |
| 14:04.49 | Samot | Are you still having the same WS issue? |
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| 14:05.37 | HCorn | Yes |
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| 14:18.00 | HCorn | Samot: Tried the same simple thing with X-lite instead of a websocket client, still same result. The agent hears the beep and then gets logged out. |
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| 14:21.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:23.28 | Samot | What does a debug show? |
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| 14:27.53 | HCorn | Samot: http://pastebin.com/6Xt8bQ9V |
| 14:28.54 | HCorn | Samot: missed sip debug, but did the same thing with debug on if you scroll down |
| 14:35.18 | Samot | SIP/2.0 501 Unsupported Method |
| 14:35.25 | Samot | How is the agent's device setup? |
| 14:35.35 | Samot | In Asterisk? |
| 14:37.09 | HCorn | Samot: Do you mean the sip peer-config post for the user? |
| 14:37.44 | Samot | Yes, the extension/devices sip config. |
| 14:37.57 | Samot | transport, nat, etc. |
| 14:40.58 | HCorn | Samot: http://pastebin.com/eppZi7rb |
| 14:41.16 | HCorn | From the sip peer table (using realtime) |
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| 14:49.56 | Samot | X-Lite is set to UDP as it's transport? |
| 14:50.58 | Samot | Make sure that the only codec it's using is alaw. |
| 14:52.07 | HCorn | Well, it is set to automatic, but I'll try with UDP set now. |
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| 14:52.22 | HCorn | Yes, alaw is the only enabled codec. |
| 14:52.51 | Samot | OK. |
| 14:53.05 | HCorn | Nope, still same result with UDP set. |
| 14:53.16 | Samot | Are you seeing the 501 error? |
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| 14:53.34 | raspberrypifan | so my calls keep dropping. They last 10 seconds or so then drop. I have the sip debug log but i dont see anything there |
| 14:53.34 | raspberrypifan | http://pastebin.com/Xpaucg9G |
| 14:54.14 | HCorn | Samot: yes |
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| 15:02.35 | Samot | HCorn: Show another test call. Just an unedited one. |
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| 15:11.27 | HCorn | Samot: http://pastebin.com/CRTZP2KB |
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| 15:12.45 | HCorn | Samot: What I am doing is 1. Start X-lite, dial 1004, get agent login, initiate an outgoing call with a call file, the cellphone rings, I answer, X-lite hears the beep and gets disconnected |
| 15:13.58 | Samot | -- Called <cellphone>@<outgoingnumber> |
| 15:14.01 | Samot | Did you do that? |
| 15:14.22 | HCorn | Nope, replaced the numbers |
| 15:14.39 | Samot | No, you didn't make it look like that? |
| 15:14.49 | HCorn | Yes, I did |
| 15:15.19 | HCorn | Don't want to paste the actual numbers there |
| 15:16.19 | [TK]D-Fender | Reliably Transmitting (no NAT) to 80.232.37.178:5060: |
| 15:16.19 | [TK]D-Fender | INVITE sip:<cellphone>@sip.phonzo.com SIP/2.0 |
| 15:16.23 | [TK]D-Fender | Contact: <sip:<outgoingnumber>@192.168.0.201:5060> |
| 15:16.27 | [TK]D-Fender | Broken SIP config |
| 15:16.43 | [TK]D-Fender | Actualy configure your WAN IP, localnmets, disable reinvites, etc like you're supposed to do |
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| 15:24.45 | HCorn | D-Fender: Thanks! I'll try fixing that! |
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| 16:00.27 | HCorn | [TK]D-Fender: I tried changing the thins you wrote but it didn't work, same result. Also tried to make an external call from the Websock client directly to my cellphone. That worked well. I am thinking the problem may be related to when the agent leaves the agent holding bridge? |
| 16:02.46 | [TK]D-Fender | Saying "tried changing" doesn't show us it was done right. |
| 16:03.01 | [TK]D-Fender | Showing > thinking. |
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| 16:09.48 | catphish | is there any way i can prevent asterisk matching incoming SIP requests against the IP of a dynamic peer? (chan_sip) |
| 16:14.31 | [TK]D-Fender | Depends on the kind of request |
| 16:15.07 | catphish | i guess making my dynamic users into INVITE specifically |
| 16:15.34 | catphish | the issue is that i have a proxy, lots of authenticated users originate from it ("friend"s) |
| 16:15.35 | [TK]D-Fender | Invites do only match against the host when registered and you are actually set to peer |
| 16:15.46 | catphish | but the proxy also has a "peer" entry for unauthenticated incoming calls |
| 16:16.15 | catphish | unfortunately, the incoming anonymous calls are matching the first authenticated user's "friend" entry |
| 16:16.21 | [TK]D-Fender | Alternative is ensure everyone is actual type=friend which creates a USER as well |
| 16:16.38 | [TK]D-Fender | type=peer will hit on IP and from that point you're screwed |
| 16:16.58 | catphish | i assume if i set my authenticated users as type=user, i wouldn't have this problem? |
| 16:17.06 | catphish | but then i also wouldn't be able to place calls to them? |
| 16:17.14 | catphish | since they couldn't register |
| 16:17.52 | [TK]D-Fender | friend <- |
| 16:18.01 | [TK]D-Fender | is creates botha peer & user |
| 16:18.08 | [TK]D-Fender | and the user gets matched first |
| 16:18.16 | [TK]D-Fender | so as long as your "from" is sane this should work |
| 16:18.42 | catphish | thanks, i'm sorry i have to dash off |
| 16:18.46 | catphish | will think more about this later |
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| 17:02.18 | Khronos | Hi ugys. |
| 17:02.20 | Khronos | guys |
| 17:02.26 | Khronos | Anyone ever seen this before? |
| 17:02.47 | Khronos | I have phones sitting behind a nat connection that work fine for the most part. |
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| 17:05.06 | Khronos | Every once in a while it seems as if one of the ip addresses goes bad after a while. |
| 17:05.50 | Khronos | What I mean is that a phone will be working fine for a while and then at some random interval the phone will not be able to register with the Asterisk server anymore. |
| 17:06.35 | Khronos | The only way to fix this is to asign a static ip address on the phone through dhcp mac reservation and when the phone gets a different inside address things work fine again. |
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| 18:03.06 | Khronos | c |
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| 19:19.18 | lvlinux | I just had a nasty power outage at a client location. Now Asterisk is complaining about "db_execute_sql: Error executing SQL (COMMIT): database is locked" |
| 19:19.24 | lvlinux | what should I do to address this? |
| 19:19.44 | [TK]D-Fender | Depends what SQL that is talking aboutt... |
| 19:20.15 | lvlinux | I have cdr_adaptive_odbc talking to a MySQL db for CDRs but that's the only thing I've setup manually to do with a DB. |
| 19:20.49 | lvlinux | But there is astdb.sqlite3 in /var/lib/asterisk |
| 19:21.04 | *** join/#asterisk klow (~textual@96.81.150.137) |
| 19:22.51 | lvlinux | Also, phones don't seem to be able to register and I'm getting no output with "pjsip set logger on" |
| 19:23.28 | [TK]D-Fender | Perhaps astDB is corrupt |
| 19:23.54 | lvlinux | How would I remedy that? |
| 19:23.56 | [TK]D-Fender | if you can I'd recommend deleting it and let * recreate it. Then recreate whatever keys are needed (including resetting phones, etc) |
| 19:24.10 | lvlinux | recreate keys? |
| 19:24.34 | [TK]D-Fender | anytthing you've made in astdb yourself |
| 19:24.40 | *** join/#asterisk obelixBE (~obelix@2a02:1811:c52c:1700:b026:a58e:f445:74db) |
| 19:24.53 | lvlinux | oh no I haven't done anything to it. |
| 19:25.34 | [TK]D-Fender | Then the most you stand to lose is reg info |
| 19:25.40 | [TK]D-Fender | so kill the file and restart * |
| 19:26.21 | lvlinux | k i did, still no registration. ? |
| 19:26.37 | lvlinux | or pjsip debug output |
| 19:27.47 | [TK]D-Fender | This won'ttt MAKE them re-register |
| 19:27.50 | [TK]D-Fender | You have to make tthem |
| 19:27.57 | [TK]D-Fender | reboot the phones, etc |
| 19:28.17 | lvlinux | yes I know that, I mean when I try to register a phone, it can't register, and asterisk console is showing nothing. |
| 19:28.54 | *** join/#asterisk CeBe (~CeBe@89.204.154.235) |
| 19:30.16 | lvlinux | wait a second...odd...i just made a call and it was answered but still no pjsip debug output |
| 19:30.30 | [TK]D-Fender | You should see the packets. |
| 19:30.35 | [TK]D-Fender | if you can't then something else is wrong. |
| 19:30.38 | lvlinux | I know! |
| 19:30.39 | [TK]D-Fender | check firewalls, etc |
| 19:32.40 | lvlinux | ok looks like somehow I had two instances running. ?!?! |
| 19:32.50 | lvlinux | and the console was connected to the second one. |
| 19:33.16 | [TK]D-Fender | that... would be very bad |
| 19:33.23 | lvlinux | yeah |
| 19:33.26 | [TK]D-Fender | and would mean a contention issue over AstDB |
| 19:33.36 | [TK]D-Fender | Which explains everything really |
| 19:34.00 | lvlinux | when I kill the second one that isn't doing anything, the systemd one is still running, but it won't let me connect w the console because it says that /var/run/asterisk/asterisk.ctl doesn't exist ? |
| 19:36.32 | [TK]D-Fender | kill them both and restart |
| 19:37.10 | lvlinux | well i need to be able to connect to the console first to ensure that no calls are happening when I restart it. Any way to force it to do that? |
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| 19:43.47 | Naikrovek | ahoy |
| 19:51.01 | [TK]D-Fender | Itt already isn't letting you |
| 19:51.06 | [TK]D-Fender | Kust kill it |
| 19:51.08 | [TK]D-Fender | jsut* |
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| 20:23.02 | catphish | i apologise, i asked a question earlier, then wandered off while someone was trying to help me |
| 20:26.32 | catphish | so, i have some authenticated SIP clients (phones) configured as "friend", these connect through a proxy, so they all register with the same IP / port, this works perfectly, asterisk identifies incoming SIP requests using the From header and matches the right friend |
| 20:28.22 | catphish | however, i also have the IP of the proxy itself configured as a "peer" with a static IP address. unfortunately when the proxy sends an unauthenticated request (ie an unauthenticated incoming call) asterisk matches the first dynamic "friend" registered from the proxy, rather than the static peer |
| 20:28.46 | catphish | is there any way i can prevent asterisk IP matching against a "dynamic" peer (friend)? |
| 20:37.28 | *** join/#asterisk robmal (r@wporzo.pl) |
| 20:40.43 | catphish | unfortunately, a registered dynamic friend is also a "peer" and matches on IP, perhaps i can order the results in SQL to overcome it |
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| 23:35.53 | iheartlinux | what exactly does the underscore _X do in dialplan? |
| 23:38.19 | nix8n82 | Match any 1 digit extension 0-9 |
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| 23:40.42 | iheartlinux | nix8n82: so I have an "exten => 800,1,Answer...same => n,VoiceMailMain(800)" should it begin with _ as well? |
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| 23:41.40 | iheartlinux | Never mind I think I understand. |
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