| 00:00.25 | phix | Penguin: ah yes, I was refering to the worst case |
| 00:00.35 | phix | TCP over TCP is something you want to advoid |
| 00:00.39 | phix | avoid* |
| 00:01.19 | phix | monsterco: ok so you want secure phone calls, then use SRTP |
| 00:01.54 | phix | Does the phones you are using support that? |
| 00:03.01 | phix | also since now your asterisk server will be exposed to the Internet (as aposed to your private network via VPN) you are now open to direct DDOS attacks / exploits to your asterisk service |
| 00:03.09 | phix | (as aposed to just OpenVPN) |
| 00:05.55 | monsterco | phix - not secure phone calls - past NSA who cares - I don't care. I just don't want any attacks on our Asterisk servers |
| 00:06.39 | monsterco | phix - if I can block everything but known IP address based on dynamic DNS at firewall level then Asterisk won't be exposed to DDoS or SIP attacks |
| 00:08.38 | monsterco | another question in between talks....is there a SIP Global option on Polycom IP 335 like there is on Aastra phones? I usually set global SIP line on Aastra phones and it populates all other lines on that phone - In case of Polycom I see Line1 and Line2 and I only set Line1. I heard someone complained that all calls now comeon same line - what config am I missing? |
| 00:09.44 | phix | monsterco: Well get all of your clients to use static addresses, or allow their entire ISP range in |
| 00:10.04 | phix | You can't use DNS as it is cached when the rule is added |
| 00:10.55 | *** join/#asterisk fulcan (joe@2600:3c01::f03c:91ff:feae:9ef3) |
| 00:11.39 | phix | unless you want to clear and re-add your firewall rules every x minutes or hours |
| 00:11.50 | fulcan | I have a trixbox that is getting 'network congested' on most IVR functions and degraded calls. http://paste.ee/p/oGDeL |
| 00:12.01 | Penguin | I wouldn't want to do that, but I would do it if I had to. |
| 00:12.23 | fulcan | http://sprunge.us/eWbd |
| 00:12.24 | Penguin | I would script a name/address checker. |
| 00:12.57 | phix | Yeah or what |
| 00:12.57 | Penguin | Query the name, check against the IP address. If it is different, update the firewall. |
| 00:13.01 | phix | what = that |
| 00:13.10 | phix | Use python <3 |
| 00:13.11 | phix | :) |
| 00:13.53 | Penguin | I would use bash and cron, since I don't know python. |
| 00:15.23 | monsterco | phix - yeah static IP would be a brilliant resolution but in cases that it's not possible I still would like to know why dynamic DNS won't be good. |
| 00:15.29 | phix | import socket; ip = socket.gethostbyname('hostname') |
| 00:15.41 | Penguin | I already told you why dynamic DNS is a bad idea. |
| 00:15.44 | Penguin | Clearly told you. |
| 00:15.54 | monsterco | fulcan - you still using Trixbox? lol |
| 00:16.03 | phix | if ip not in firewallList: addIpToFirewallAllow(ip) |
| 00:16.24 | Penguin | Also write something to clear the old entry. |
| 00:17.05 | phix | monsterco: because dynamic DNS isn't exact, and requires polling at intevals, between the intevals your clients wont have phone service |
| 00:17.21 | phix | Penguin: <3 python |
| 00:17.49 | fulcan | monsterco I inherited this beast. |
| 00:17.49 | Penguin | That's even exactly how the updater clients work. They have to perform the exact same type of tests to see if the IP has changed. |
| 00:17.55 | monsterco | and phones don't support any ICMP pings or any other methodes to inform me who they are - so I will end up with adding another little box - but I guess that would be so for dynamic dns too |
| 00:18.34 | Penguin | Many plastic routers have ddns update clients built in. |
| 00:19.03 | monsterco | but not the dyn dns providers I want who are more stable that dynamic dns |
| 00:19.34 | phix | ? |
| 00:19.40 | phix | dyn dns is stable |
| 00:19.49 | phix | you just have to log in every month now |
| 00:19.58 | phix | or subscribe to their basic level service |
| 00:20.35 | monsterco | they have limits on free services - |
| 00:21.03 | phix | so sign up with them then, or just use OpenVPN :) |
| 00:21.23 | monsterco | will sign up - but I wish there was wider support for other types of dynamic dns providers |
| 00:21.30 | monsterco | dyndns is not stable - I know from experience |
| 00:21.57 | Penguin | How many end points will you be supporting? |
| 00:22.34 | monsterco | around 100 |
| 00:23.15 | Penguin | I think you should pay for dynamic DNS services if you cannot create your own in-house. |
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| 00:23.22 | monsterco | problem is that endpoints do not always behave same way with updating dynamic dns - some are better than others - so I was thinking of putting a little box there for updates - raspberry pi? |
| 00:23.23 | monsterco | ! |
| 00:24.06 | monsterco | but then again this whole things gets complicated - if I were to put a raspberry pi then I would simply have it to connect to my pfSense server and then create a script of pfSense server to change IP based on incoming traffic every minute |
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| 00:25.09 | Penguin | Here's what I might do: |
| 00:25.23 | Penguin | Set up a private name server in my network to use for dynamic clients. |
| 00:25.48 | Penguin | Write my script to query the private name server and update the firewall as necessary. |
| 00:26.16 | Penguin | You'll have to allow all global hosts to access that nameserver, but set up tokens to use for updating the client addresses. |
| 00:26.39 | phix | monsterco: it is stable if you pay for it |
| 00:29.08 | monsterco | Penguin - like a simple linux server that acts as DNS Server - and then on the client end use what? an updater client? can you think of something very stable and low cost? |
| 00:29.14 | fulcan | anyone know what this error is about? FONALITY: This thread has already held the conlock, skip locking |
| 00:29.24 | monsterco | phix - I can also afford my own name server for this so why not |
| 00:29.44 | phix | It is sounding more simplier to just use OpenVPN :P |
| 00:30.09 | monsterco | wish all phones came with dns updater clients |
| 00:30.10 | phix | or IPsec |
| 00:30.23 | monsterco | it's really stupid that they don't have vpn or something else built in when SIP is so risky |
| 00:30.28 | phix | monsterco: Most routers do |
| 00:30.59 | phix | or are the phones connected directly to a gateway / modem? |
| 00:31.05 | monsterco | phix - like I said not all modem/routers do a good job of updating dyn dns and also users tend to reset routers easily |
| 00:31.16 | phix | true |
| 00:31.46 | phix | but who cares if you are getting paid to reset it up for them :) |
| 00:32.07 | monsterco | probably a better system would be to recieve the first SIP packet on a separate server, analyze it and match ther username/password received in first packet against a database of allowed users and then if it is good add firewall rule and direct all subsequent SIP request to Asterisk |
| 00:32.38 | monsterco | and to avoid renumeration use long random username and long random password |
| 00:32.43 | phix | hmmmm |
| 00:33.03 | monsterco | phix - I just don't want to receive phone calls because IT idiots who mess up the network for me too |
| 00:33.48 | monsterco | I actually purposed above changes for Asterisk but was told that renumeration of SIP users exploits this - but there is no exploitation if long usernames are used instead of extension 200, 201, 202..... |
| 00:39.55 | monsterco | Penguin - problem with my own nameserver is that not all modems/routers support a customer dynamic dns provider......arggggg |
| 00:39.59 | Penguin | User names have nothing at all to do with extensions. |
| 00:40.09 | monsterco | seems like dyn.com is the winner in this market |
| 00:41.07 | monsterco | Penguin - yes, and that is why I said use long usernames so you can simply drop the first packet and not respond at all so the sipvicious kiddie can move on - otherwise if you respond with "authentication refused" or send me token etc then they know it's a SIP server or Asterisk server and RED-LIST for future attacks |
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| 01:50.31 | Penguin | What is the right way to have asterisk listening on the regular sip port as well as an alternate sip port for when clients' port 5060 is blocked? |
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| 02:08.36 | pabelanger | Penguin, setup a proxy |
| 02:08.54 | Penguin | That's what I thought. Is kamailio or opensips going to be overkill? |
| 02:09.17 | pabelanger | depends if you have every used it before |
| 02:09.26 | pabelanger | IMO, no |
| 02:09.34 | pabelanger | but we drop it infront of all our asterisk boxes |
| 02:10.05 | Penguin | Which one would you recommend using? |
| 02:10.43 | pabelanger | I think kamailio |
| 02:10.45 | pabelanger | err |
| 02:10.47 | pabelanger | like* |
| 02:10.59 | pabelanger | but both are very similar |
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| 02:18.22 | Penguin | Is there any simpler proxy to handle only providing an alternate port for SIP? The easier, the better. |
| 02:21.42 | Penguin | Is there a simple explanation for why a person cannot use iptables to redirect port numbers and make SIP still work? It works with many other services, but we've all found out at some point that if you redirect SIP with the firewall, it breaks it. |
| 02:29.11 | pabelanger | because otherwise you need to rewrite patch headers, they will contain the wrong information if you forward them |
| 02:29.19 | pabelanger | proxy is your best be |
| 02:29.31 | pabelanger | or run another instance of asterisk on the other port |
| 02:29.40 | pabelanger | then interconnect asterisk boxes |
| 02:35.32 | Penguin | patch headers? |
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| 02:41.48 | monsterco | patch headers and then don't patch once white listed |
| 02:42.28 | monsterco | IS there a global SIP option on Polycom phone like there is on Aastra phones? |
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| 03:10.40 | pabelanger | s/patch/sip |
| 03:10.44 | pabelanger | sorry, tired |
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| 05:06.19 | fulcan | I have a trixbox macro that is failing. calls are supposed to be sent to an agent that is logged in. the agent is there, but the call goes to que, and just lays there for a few minutes, then gets booted off. http://paste.ee/p/XyeZP complete extensions -> http://sprunge.us/WMdM |
| 05:09.37 | fulcan | http://paste.ee/p/jTCHM <- asterisk cli output |
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| 05:23.34 | fulcan | this is my fail -> -- Executing [1@ivr-2:1] DBdel("SIP/vitel-inbound-00000035", "") in new stack |
| 05:25.49 | kaldemar | how is that a failure? |
| 05:28.02 | kaldemar | also, why do you think the call should do anything else but go to the queue once it hits 777@ext-queues? |
| 05:30.04 | fulcan | i can't figure out why the free extension in not pulling the ques. |
| 05:30.54 | fulcan | these guis are so modular, it's difficult to back track through the code. :( |
| 05:35.59 | kaldemar | the queue should have the device as a member. check that with "queue show". |
| 05:38.11 | fulcan | http://paste.ee/p/flpYm |
| 05:38.17 | fulcan | users are waiting |
| 05:41.37 | fulcan | I tried changing ivr 0 to ring groups on a whim. got it to ring, but still the same issue. |
| 05:41.46 | fulcan | no connect |
| 05:42.25 | kaldemar | all the members are listed as Local/XXX@from-internal/n, and you don't even have a context called from-internal in your dialplan. |
| 05:43.20 | fulcan | kaldemar then how tf was this working before? I inherited this mess, forgive me. |
| 05:44.05 | kaldemar | are you sure it has been working? |
| 05:44.53 | fulcan | kaldemar my buddy swares by it. I'm with you, my jaw is on the ground screaming 'how?' |
| 05:46.10 | kaldemar | with those configs, it does not work. |
| 05:46.50 | kaldemar | if that all is generated by trixbox, i afraid i can't help you any further since i'm not familiar with it. you can try asking in #trixbox if anyone is there. |
| 05:47.24 | fulcan | kaldemar thank you kindly. My head is in the same place. |
| 05:47.31 | kaldemar | or maybe someone at #freepbx would know about that. |
| 05:48.09 | fulcan | that's right, isn't trixbox now morphed to freepbx? |
| 05:48.52 | kaldemar | trixbox is a linux distro with asterisk, freepbx and a bunch of other stuff. |
| 05:49.29 | kaldemar | freepbx is the GUI in trixbox (although freepbx has a full distro too nowadays). |
| 05:54.22 | fulcan | Mr. Spencer my be jizzing in his pants. He created a monster. :) |
| 05:55.20 | fulcan | two phucking fork in his name, geezus. |
| 05:55.29 | kaldemar | i don't think he's to be held responsible for trixbox in any way. :P |
| 05:56.21 | kaldemar | and those are not asterisk forks, just other entities that use asterisk and/or co-operate with it. |
| 05:59.30 | fulcan | kaldemar granted, but without the spencer code, those forks wouldn't be there. Asterisk is a very interesting revolution. Very interesting to watch it progress. |
| 06:00.08 | fulcan | 2002 was my first asterisk server. |
| 06:00.55 | fulcan | that was the same year I dumped windows. |
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| 09:16.14 | linocisco | asterisk vs freeswitch? |
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| 10:03.18 | linocisco | asterisk vs freeswitch? |
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| 10:19.38 | kaldemar | linocisco: what's your goal with that? |
| 10:22.34 | linocisco | just want to know for what they are used |
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| 13:10.38 | carrar | *Y*A*W*N* |
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| 13:22.07 | niplo | Need some help with a dial plan i can permit 69ZXXXXXXX on out goin routes but no 3069ZXXXXXXX |
| 13:22.16 | niplo | What should i do :/ |
| 13:22.41 | [TK]D-Fender | those are already 2 complete different patterns |
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| 13:22.52 | [TK]D-Fender | they don't start the same. There is no overlap |
| 13:23.08 | boom^time | ? just exten=>_69ZXXXXXXX |
| 13:23.14 | niplo | :O |
| 13:23.17 | [TK]D-Fender | yup |
| 13:23.40 | [TK]D-Fender | 69ZXXXXXXX <- this doesn't permit the other... because this DOESN'T start with 30 at all like the other |
| 13:24.29 | [TK]D-Fender | Now if you wanted specifically not to allow 6955XXXXXX that would be another matter |
| 13:26.20 | niplo | sec |
| 13:26.23 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 13:26.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:26.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:28.23 | niplo | Your call cannot be completed as diealed |
| 13:28.26 | niplo | *dialed |
| 13:29.16 | [TK]D-Fender | I be looking very closely at that call right about now... |
| 13:29.22 | [TK]D-Fender | and showing us your configs to match.. |
| 13:30.53 | niplo | Well, i would love so, but its an elastix system, So the configs will not be..... |
| 13:31.44 | [TK]D-Fender | niplo: You should not be in here for support of it. #elastix , #freepbx |
| 13:32.00 | [TK]D-Fender | niplo: Where you have to fill things in to configure that has nothing to do with configuring Asterisk for yourself |
| 13:33.10 | *** join/#asterisk danjenkins (~danjenkin@62.254.236.250) |
| 13:33.19 | niplo | I know but elastix has, few people. |
| 13:34.50 | [TK]D-Fender | niplo: this is not a GUI support channel. I also list TWO. |
| 13:38.37 | niplo | Anyway thabks alot |
| 13:38.43 | niplo | *thanks |
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| 14:20.36 | Katty | guten morgan |
| 14:20.52 | Katty | morgen |
| 14:21.36 | tuxx- | hallo, was für ein schönes Wetter, das wir |
| 14:21.40 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cmngvmvqxujoiqmg) |
| 14:21.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:21.48 | *** join/#asterisk vagmenezes (~vagner@mail.aytycrm.com.br) |
| 14:22.18 | WIMPy | Where's the rest of that sentence? |
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| 14:22.33 | Katty | tuxx-: wie bitte? |
| 14:22.52 | tuxx- | man, writing german is harder than speaking it |
| 14:22.53 | tuxx- | :D |
| 14:22.56 | Katty | tuxx-: das Wetter schon ist? |
| 14:23.40 | Katty | ich bin verwirrt :< |
| 14:24.10 | MLNoah | I'm using Asterisk 11.5 with ODBC voicemail storage so that two load-balanced/redundant servers can "share" one database backend and the voicemail box files. A customer has been complaining of missing voicemails, and when I go look in the voicemessages table, there are many many messages in the "routing" context, including pretty much all of the messages for this customer. How do I get the... |
| 14:24.11 | MLNoah | ...messages "unstuck", and how do I troubleshoot to prevent further sticking? |
| 14:24.19 | Katty | use pan spray. |
| 14:24.22 | Katty | also hard anodized pans. |
| 14:25.05 | Maliuta | tuxx-: nien! Das ist nicht wahr! :P |
| 14:25.19 | tuxx- | i think so! |
| 14:25.29 | Katty | (nein) |
| 14:25.46 | tuxx- | but as a dutchie i should be able to write it |
| 14:25.46 | WIMPy | tuxx-: You said what a nice wether, that we ....? |
| 14:25.54 | tuxx- | i said, its nice weather, isnt it? |
| 14:26.09 | Katty | german is hard. |
| 14:26.14 | tuxx- | or maybe im completely in the wrong |
| 14:26.14 | tuxx- | hehe |
| 14:26.19 | Katty | but the doner kebab maeks it worth it |
| 14:26.23 | tuxx- | hehe |
| 14:26.25 | tuxx- | and the bratwurst |
| 14:26.28 | Katty | YES. |
| 14:26.28 | tuxx- | and 1l beer mugs |
| 14:26.30 | Katty | and kvass. |
| 14:26.31 | tuxx- | :D |
| 14:26.37 | Katty | ....wait. i don't think kvass is german |
| 14:26.37 | Maliuta | Katty - mien liebling ;) |
| 14:26.39 | Katty | i think that's russian. |
| 14:26.40 | *** join/#asterisk monsterco (~monsterco@64.231.101.21) |
| 14:26.46 | WIMPy | You can use ", nicht wahr?" |
| 14:26.53 | Katty | MEIN liebling |
| 14:26.58 | [TK]D-Fender | Irony : Germany's Oktoberfest is a competition to find the best wurst. |
| 14:26.59 | tuxx- | true, that sounds better :) |
| 14:27.09 | tuxx- | hehe |
| 14:27.15 | Katty | mmm bratwurst. |
| 14:27.18 | bacobart | "oder nicht?" |
| 14:27.22 | WIMPy | Oktoberfest isn;t german. That's bavarian. |
| 14:27.24 | Katty | i put mustard all over bratwurst. |
| 14:27.24 | Maliuta | Katty: I love you too :P |
| 14:27.32 | Katty | Maliuta: <3 |
| 14:27.37 | tuxx- | i put bratwurst all over my mustard! |
| 14:27.42 | WIMPy | That's another possibility. |
| 14:27.44 | Katty | tuxx-: yes, that's more accurate. |
| 14:27.48 | tuxx- | xD |
| 14:27.55 | Katty | needs MOAR VINEGARY BITE |
| 14:27.57 | monsterco | Hi everyone - I have just setup a Line 1 on a Polycom IP 335 - If there are two calls coming in both calls come in on Line 1. Is there a global SIP option for Polycom like there is for Aastra so both lines can take calls? |
| 14:28.17 | Katty | monsterco: it should beep at you |
| 14:28.20 | Maliuta | roll mops, sourkraut, kabana, braktehoffen - all the foods I grew up with |
| 14:28.21 | Katty | monsterco: and show the other call |
| 14:28.30 | Katty | monsterco: regardles of how many extensions are registered on the polycom |
| 14:28.48 | Katty | tho i guess you could register "lines" |
| 14:28.53 | Maliuta | my grandparents were wasted on my father. He doesn't speak any German, or Ukrainian |
| 14:28.55 | Katty | but ain't nobody got time for that |
| 14:29.17 | Katty | Maliuta: do you have...recipes?! :> :> :> |
| 14:29.33 | Katty | Maliuta: don't need the sourkraut one tho |
| 14:29.44 | Maliuta | Katty: you only used a single negative. I thought Americans would have said "ain't nobody got no time for that" ;) |
| 14:30.09 | Katty | yes. yes they would. |
| 14:30.11 | Katty | oh look. i did. |
| 14:30.17 | Maliuta | Katty: some recipes, none from my lineage though. I get most of my Ukrainian food from a friends mum |
| 14:30.31 | Katty | Maliuta: hey i'll take whatever german recipes i can get my hands on |
| 14:30.39 | Katty | except for sourkraut. and haggis. |
| 14:30.49 | Katty | or bloodsausage |
| 14:30.53 | Maliuta | Katty: and my German stuff from a friend (a yank) who lives in Germany with his /dev/gf0 |
| 14:31.01 | Katty | that works. |
| 14:31.32 | jmetro | Sauerbraten is delicious |
| 14:31.35 | Maliuta | I want to go to Germany for the pigs trotters being sold on street stalls |
| 14:31.38 | monsterco | Katty - Yes, it does but on Line 1 only - can the 2nd call not come on Line 2? this happens on Aastra phones |
| 14:31.43 | Maliuta | oh, and to get my EU passport :) |
| 14:32.17 | monsterco | can I register both Line1 and Line2 to same extension? |
| 14:32.36 | Katty | monsterco: i don't think anyone has done the line1 and line2 stuff since...a decade ago |
| 14:32.45 | Katty | monsterco: we're all extension 100. |
| 14:32.59 | Katty | monsterco: and whatever gets transfered gets transfered |
| 14:33.15 | vagmenezes | Hi Srs, someone has implemented a ambient with asterisk (queues + sip registrations ) over 1500/minute calls simultaneous? Or large ambients? |
| 14:33.18 | monsterco | Katty - so you put your credentials on Line 1 right? |
| 14:33.32 | WIMPy | I see line keys all over these sip phomes. |
| 14:33.34 | Maliuta | Katty: I once had the best sourkraut - it was served with bangers and mash. Then that restaurant went to hell. |
| 14:33.43 | Katty | WIMPy: yeah but they're not you know /lines/ anymore |
| 14:33.43 | Maliuta | It was a red cabbage sourkraut |
| 14:33.52 | Katty | WIMPy: they're more like /register/ fields |
| 14:34.12 | monsterco | I just setup Global SIP options on Aastra phones and all other lines just populate fine and calls come in order of line 1, line2, line 3,....this is not happening with Polycom when I setup Line 1 configs - should I be setting this up somewhere else? |
| 14:34.13 | WIMPy | That doesn't make it better. |
| 14:34.16 | Katty | Maliuta: i like cabbage. just not...fermented :/ |
| 14:34.30 | WIMPy | It's a shitty handling concept. |
| 14:34.35 | Maliuta | Katty: what? no kimchee for you? :) |
| 14:34.43 | Katty | NO |
| 14:35.24 | Maliuta | Katty: but you _do_ like pickled herring, right? |
| 14:35.28 | Katty | no. |
| 14:35.35 | Maliuta | especially with the right brine mix |
| 14:35.40 | Katty | i like shabushabu |
| 14:35.54 | Katty | which is totally not pickled. |
| 14:35.59 | Katty | ...or german. |
| 14:36.00 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 14:36.04 | Maliuta | no sourkraut, or rollmops? maybe I love you less ;P |
| 14:36.11 | Katty | idk what a roll mop is. |
| 14:36.19 | Katty | probably not anything like a steam mop. |
| 14:36.26 | monsterco | The receptionist is complaining that 2nd call comes on line 1 too and then she can't hangup call #1 unless she hangs up on call #2 |
| 14:36.37 | jmetro | Saeurkraut is not tasty. |
| 14:36.43 | jmetro | Sauerbraten on the other hand, delicious. |
| 14:36.49 | Katty | look what the cat dragged in! |
| 14:36.51 | Katty | it's jmetro! |
| 14:37.00 | jmetro | :> finally back in the office |
| 14:37.03 | Katty | woot! |
| 14:37.05 | Maliuta | herring rolled around a gerkin and onion, pickled in brine. |
| 14:37.15 | Katty | gerkin is a pickle, right? |
| 14:37.43 | jmetro | Sour pickle rather than sweet, if i remember. |
| 14:37.43 | Katty | yo dawg, we heard you like stuffed pickled. so we wrapped stuff around a pickle and pickled it for you |
| 14:38.03 | Maliuta | Katty: rollmops http://tinyurl.com/2ky35r |
| 14:38.22 | Katty | oh. my. |
| 14:38.28 | Katty | grumpy cat says NOPE |
| 14:38.46 | monsterco | Anyone can tell me if there is a global sip option in Polycom IP 335? |
| 14:38.47 | Maliuta | jmetro: there is more than one kind of rollmop. I don't like the Finnish ones, too sweet. |
| 14:38.52 | Katty | they have bangers in a little import store about 2 hours north of here. |
| 14:39.08 | jmetro | monsterco: i believe you just register all the lines according to what you want. |
| 14:39.11 | Katty | they're frozen. |
| 14:39.19 | Maliuta | Katty: a banger is essentially a pork sausage, it's english |
| 14:39.25 | monsterco | jmetro - can I register all lines to same username/password? |
| 14:39.30 | Maliuta | frozen? that's just bad |
| 14:39.33 | monsterco | I have only one extension for that phone |
| 14:39.37 | Katty | it's an import store |
| 14:39.42 | Katty | they have stuff from everywhere |
| 14:39.48 | jmetro | Katty> Most delicious thing in the world, is of course, tacos with ground beef. |
| 14:39.59 | Katty | ^- with taco bell lava sauce |
| 14:40.06 | Maliuta | monsterco: sure you can ... but if one line is in use * will return a busy signal for that extension |
| 14:40.17 | file | Katty, true loveeeee! true loveeeeee! it must be.... true loveeeee! nothing else can break my heart like true loveeeeeeee |
| 14:40.23 | jmetro | Katty: God yes, if you've tried it. |
| 14:40.44 | jmetro | Katty: Otherwise, at home i have real sharp cheddar and some good Cholula to sprinkle on there |
| 14:40.52 | Katty | goshdangitanyhow, file |
| 14:40.57 | file | hi |
| 14:41.07 | Maliuta | Katty: not so much. The import store near my parents in .ca changed hands and now doesn't stock Vegemite. I had to mail them a jar. |
| 14:41.07 | Katty | grumbles, listens to song |
| 14:41.19 | monsterco | Maliuta - I don't want to have two extension |
| 14:41.34 | Katty | file: also, hi. |
| 14:41.35 | Katty | hugs file |
| 14:41.41 | monsterco | I want one extension but I want Line 1 ring first and if that is busy then Line 2 ring - come on guys, this is standard phone stuff |
| 14:41.42 | file | hugs Katty |
| 14:41.56 | Katty | monsterco: just a side note... |
| 14:41.56 | Maliuta | Katty: 1 450 gram jar of vegemite: $4. Postage to .ca: $43 ... most expensive vegemite ever. |
| 14:42.02 | Katty | monsterco: if what you're wanting is Stanard Phone Stuff |
| 14:42.05 | Katty | monsterco: then why don't you know it? |
| 14:42.23 | Katty | Maliuta: and vegemite is disgusting |
| 14:42.27 | Katty | Maliuta: why would you do that?! lol |
| 14:42.30 | [TK]D-Fender | monsterco: you should configure the same on all 3 and it should roll over normally |
| 14:42.31 | monsterco | Katty - you are so smart asking some very obvious questions |
| 14:42.41 | Katty | bows |
| 14:42.47 | Katty | monsterco: we can't all be perfect. |
| 14:42.48 | monsterco | [TK]D-Fender - thank you. That was my answer! |
| 14:42.55 | monsterco | lol |
| 14:42.55 | jmetro | monsterco: i'm fetching a config from one of my polycoms now |
| 14:43.07 | monsterco | jmetro - thanks |
| 14:43.09 | [TK]D-Fender | monsterco: For Polycom it's another matter |
| 14:43.38 | [TK]D-Fender | monsterco: For Polycom's you should only configure LINE1, and set the " nukeys to the number of line-keys to use for that single reg\ |
| 14:43.47 | [TK]D-Fender | numkeys* |
| 14:43.49 | Maliuta | monsterco: then you need multiple sip registrations with different user/passwd and either a queue or invocations of Dial() with a timeout and branch statements to ensure that it gets hung up properly. |
| 14:43.55 | jmetro | monsterco: I'ts looking like you just set Line1 and then put appearances up |
| 14:44.48 | monsterco | jmetro - appearence a setting under SIP or Line 1? |
| 14:45.07 | jmetro | monsterco: my output is formatted to aastra atm, so i cant see, all i can see is the part where its 1 line |
| 14:45.13 | monsterco | [TK]D-Fender - where do I find Numkeys? |
| 14:45.26 | [TK]D-Fender | in our provisioning.... |
| 14:45.28 | [TK]D-Fender | your* |
| 14:46.14 | [TK]D-Fender | reg.1.linekeys |
| 14:46.36 | monsterco | I do see a: Number Of Line Keys |
| 14:46.39 | [TK]D-Fender | And set reg.1.callsPeLineKey="1" |
| 14:46.44 | [TK]D-Fender | +r |
| 14:46.56 | monsterco | Maliuta - what I need is not that complicated |
| 14:47.03 | *** join/#asterisk lidenbrock (liden@189-041-176-219.xd-dynamic.ctbcnetsuper.com.br) |
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| 14:50.36 | monsterco | "Calls Per Line" |
| 14:50.40 | Maliuta | monsterco: well I have one SIP user that I register to both lines on my Cisco 7942, and on other devices. If I am on a call and a second one comes in then that SIP user get returned as BUSY on any other Dial() attempts. |
| 14:50.59 | monsterco | but that's not polycom |
| 14:51.10 | monsterco | I have Aastra and works beautifully without any monkeying around |
| 14:51.28 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 14:51.36 | monsterco | I shouldn't have said "standard stuff" as this is all a big cluster f*** sometime |
| 14:52.24 | *** part/#asterisk lidenbrock (liden@189-041-176-219.xd-dynamic.ctbcnetsuper.com.br) |
| 14:53.15 | Maliuta | monsterco: but it _is_ SIP. The same thing happens with multiple registrations on soft phones (on things like my mobile - sorry "cell" -, and/or my laptop). |
| 14:53.50 | file | chan_sip does not allow multiple devices to be registered to a single account |
| 14:53.57 | monsterco | What I meant was my question was much simpler and had a quick answer rather than do things on Asterisk side....maybe you didn't get what I was looking for |
| 14:54.13 | Maliuta | monsterco: if you're looking to exploit a feature on the actual handsets, then that is not exactly "standard phone stuff". More like "vendor specific phone stuff" :) |
| 14:54.43 | Maliuta | file: tell that to my asterisk box :) |
| 14:54.57 | monsterco | in an ideal world one would expect the vendors to not be idiots - but it's never a perfect world!!!!!!! |
| 14:55.04 | file | see above. |
| 14:55.39 | Maliuta | In a perfect world we wouldn't have users or developers to mess up fine sys/net admin work :) |
| 14:55.43 | file | you can have multiple accounts and dial them all at the same time, or a single device can have multiple line appearances sharing a single account (which appears to Asterisk as one registration) |
| 14:56.27 | Maliuta | heads to beds |
| 14:57.24 | monsterco | k, i gtg see the receiptionest - thanks for the help guys |
| 15:02.41 | *** part/#asterisk monsterco (~monsterco@64.231.101.21) |
| 15:03.02 | jmetro | Katty: you made me hungry for tacos. |
| 15:04.28 | *** join/#asterisk vlad_starkov (~vlad_star@91.233.188.206) |
| 15:04.31 | Katty | jmetro: omnomtaconomnom |
| 15:10.08 | jmetro | Katty: Its settled, tacos for lunch and dinner, for another month. I havent gone a day without tacos in a while. |
| 15:10.18 | Katty | but... |
| 15:10.21 | Katty | what about spaghetti?! |
| 15:10.27 | Katty | one does not simply stop eating spaghetti |
| 15:10.45 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |
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| 15:14.40 | carrar | hi Katty! |
| 15:14.47 | Katty | hugs carrar |
| 15:14.51 | Katty | hello :> |
| 15:14.55 | carrar | w00t! |
| 15:15.35 | carrar | You ready for Christmas? |
| 15:15.53 | Katty | not at all. |
| 15:16.04 | Katty | tho my shopping list is nearly complete |
| 15:16.42 | carrar | food shopping? |
| 15:16.48 | carrar | or gift list |
| 15:16.59 | Katty | gift list |
| 15:17.11 | carrar | nice |
| 15:17.18 | Katty | mhmm |
| 15:17.29 | Katty | everytime i see the boy eye something with Great Interest, it goes on the list |
| 15:17.40 | Katty | then at the end of the year, i review the list and make decisions |
| 15:18.06 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 15:18.47 | Katty | anything from a vacuum coffee brewer to a gameboy might end up under the tree. you never know. |
| 15:19.13 | Katty | there's also this foam gun thing from Drive Clean on youtube he was eyeing |
| 15:19.58 | Katty | numerious Steam games |
| 15:20.20 | Katty | a tonneau cover |
| 15:21.58 | jmetro | I <3 steam |
| 15:22.04 | jmetro | the steam summer sale kills me |
| 15:25.34 | Katty | yesh. |
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| 15:37.54 | oryn | anyone know if its posible to run asterisk with dual stack ipv4/6 and specify one out of many ipv4 addresses plus one out of many ipv6 addresses. I have asterisk on a multohomed server |
| 15:38.18 | oryn | *multihomed* |
| 15:38.25 | WIMPy | No. One or all. |
| 15:39.07 | oryn | why not? |
| 15:39.18 | WIMPy | You can only configure one address. |
| 15:39.46 | oryn | I only want one address, one on 4 and one on 6 |
| 15:40.21 | WIMPy | That's two. |
| 15:40.28 | oryn | I'd be quite happy with one on 6 but my phones only work with 4 |
| 15:40.38 | Katty | watches with amusement |
| 15:40.44 | Katty | pulls up a chair, noms popcorn |
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| 15:57.03 | Katty | [TK]D-Fender: how much do you know about stage lighting? |
| 15:57.40 | [TK]D-Fender | Katty: Only a little... |
| 15:57.59 | Katty | [TK]D-Fender: any chance you'd have a recommendation for a fog machine/halloween party sort of thing? |
| 15:58.06 | Katty | i have a fog machine. |
| 15:58.09 | Katty | just need lights |
| 15:58.37 | [TK]D-Fender | dx.com sells these awseome laser light-show thingys for real cheap.. |
| 15:58.40 | jmetro | Purple goes well with fog. |
| 15:59.08 | [TK]D-Fender | http://dx.com/p/fd-03-mini-green-red-mixed-laser-light-show-system-deep-pink-230297 |
| 15:59.24 | [TK]D-Fender | http://dx.com/s/laser+show |
| 15:59.28 | [TK]D-Fender | there's the full group |
| 15:59.37 | [TK]D-Fender | 2 of my friend's bands use them |
| 15:59.43 | Katty | reviews |
| 16:00.06 | [TK]D-Fender | a few have use LED "spots" as well which are awesome compared to the old cannons.... |
| 16:00.07 | Katty | it's PINK! |
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| 16:28.05 | jmetro | i hear they make tacos with 12 inch bread loafs nowadays. |
| 16:28.34 | jmetro | and meats like turkey, pepperoni, ham |
| 16:29.06 | saliak | I've got a 2nd SIP trunk i've added to my asterisk server. I want calls to go to a specific context, different than the other trunk. however,r when calls come in they go to the context of the other trunk. |
| 16:29.41 | talntid | that... made.. no sense. |
| 16:29.42 | jmetro | send all calls into the same context, determine who the call is coming from, route out from there. |
| 16:30.04 | jmetro | all calls go into "incoming", if its one callerid, goto "trunk1" etc. |
| 16:31.29 | saliak | yeah, that was poorly phrased, but jmetro got it. Isn't' it possible to specify which context calls go into though? I define a sip peer for each trunk, and a default context. |
| 16:32.34 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 16:33.17 | jmetro | thats what im saying though |
| 16:33.25 | jmetro | just point both of them to one, and get rid of the problem. |
| 16:34.04 | [TK]D-Fender | saliak: Do not use the term "trunk" this way. Next, you may have created 2 PEERS in sip.conf to match the sender... but the IP's are going to be the same and most ITSP's send calls to you UNAUTHED which means you set "insecure" on them. Because of thie the first one is the one that gets matched. You will not be able to separate them this way if you do. You may not have a choice in this |
| 16:34.20 | [TK]D-Fender | saliak: At which point you'll just have to figure out which is which in the dialplan. |
| 16:35.12 | jmetro | my sip trunking providers actually send a number to me, "translate" |
| 16:35.16 | jmetro | and i just use that. |
| 16:39.05 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
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| 16:43.08 | saliak | yeah, i do get a number as the extension when it comes in. OK, so I have a single incoming context that i look at the incoming DID and make a decision based on that. so, it's just the context for the first SIP peer that it uses? |
| 16:43.34 | saliak | So it's not a trunk then? What do I call it? It's a SIP registration with my provider. |
| 16:44.27 | [TK]D-Fender | "peer" |
| 16:44.57 | [TK]D-Fender | it is used to place calls (possibly w/ auth) to them, and to match calls from them. |
| 16:45.06 | [TK]D-Fender | Registration is also completely separate |
| 16:45.36 | [TK]D-Fender | ~sipregister |
| 16:45.36 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 16:45.39 | [TK]D-Fender | ^ |
| 16:45.40 | [TK]D-Fender | I can have a provider where they know my IP and I don't have to register |
| 16:59.12 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
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| 17:45.20 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 17:50.38 | Katty | sits and stares |
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| 17:57.00 | anhvt | hello |
| 17:57.17 | anhvt | I would like to ask a question about asterisk-ss7 |
| 17:57.36 | WIMPy | Do it! |
| 17:58.02 | WIMPy | But there aren't that many users, so you will probably need patience. |
| 17:58.17 | anhvt | I want to set some parameters for outgoing caller id |
| 17:58.27 | anhvt | NAI and NI |
| 17:59.06 | anhvt | The question is: How can I change the NI for called party and calling party |
| 17:59.22 | Katty | i would recommend a shrubbery |
| 17:59.39 | Katty | and cutting down the large tree in the forest...with a herring |
| 18:00.03 | anhvt | [1] --VARIABLE LENGTH PARMS[1]-- |
| 18:00.04 | anhvt | [1] Called Party Number: |
| 18:00.04 | anhvt | [1] Nature of address: 3 |
| 18:00.04 | anhvt | [1] NI: 0 |
| 18:00.04 | anhvt | [1] Numbering plan: 1 |
| 18:00.04 | anhvt | [1] Address signals: 922388661# |
| 18:00.04 | anhvt | [1] [ 07 03 10 29 32 88 66 f1 ] |
| 18:00.05 | WIMPy | Doesn't CALLERID(ton) work? |
| 18:00.05 | anhvt | [1] --OPTIONAL PARMS-- |
| 18:00.05 | anhvt | [1] Calling Party Number: |
| 18:00.06 | anhvt | [1] Nature of address: 2 |
| 18:00.06 | anhvt | [1] NI: 0 |
| 18:00.07 | anhvt | [1] Numbering plan: 1 |
| 18:00.07 | anhvt | [1] Presentation: 0 |
| 18:00.08 | anhvt | [1] Screening: 1 |
| 18:00.31 | WIMPy | For cdpn you can use the config or the special prefixes as listed in the sample chan_dahdi.conf. |
| 18:00.36 | WIMPy | ~pb |
| 18:00.36 | 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. |
| 18:00.43 | anhvt | That's part of IAM message |
| 18:00.50 | WIMPy | Oh, already 4 users on ignore. |
| 18:01.03 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 18:01.07 | anhvt | and I want to set the NI: 2 |
| 18:01.35 | *** join/#asterisk serafie1 (~erin@nat/digium/x-iygnngsrkiigeotm) |
| 18:01.53 | WIMPy | For the NI, I have no idea. |
| 18:02.00 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:02.16 | anhvt | thanks WIMPy |
| 18:17.59 | oryn | anyone know why directmedia=no is ignored? |
| 18:18.10 | phix | yay WIMPy ! |
| 18:18.43 | oryn | is it just best to recompile without native bridging? |
| 18:18.52 | file | define ignored, and you can't |
| 18:21.16 | [TK]D-Fender | oryn: Show us |
| 18:21.28 | oryn | ok, I have 2 sip devices and despite directmedia=no they still try to native bridge resulting in one way audio |
| 18:22.17 | oryn | looks like I have a workaround as its to simply not build bridge_native_rtp.so |
| 18:22.30 | file | using Asterisk 12? |
| 18:22.33 | [TK]D-Fender | oryn: Doubt it highly |
| 18:22.54 | oryn | yup well latest svn pull |
| 18:23.03 | file | and chan_sip? |
| 18:23.18 | Katty | disables file's chan_sip |
| 18:23.22 | oryn | seems to be there |
| 18:23.27 | file | Katty, I use chan_pjsip |
| 18:23.46 | file | oryn, then file an issue on https://issues.asterisk.org/jira with the console output (including debug), sip configuration, and any other details |
| 18:24.18 | oryn | someone else already has, but they have been ignored |
| 18:24.34 | file | what's the issue number? |
| 18:25.16 | Katty | ticket 508898425 |
| 18:27.43 | oryn | https://issues.asterisk.org/jira/browse/ASTERISK-22127 |
| 18:28.21 | file | it's not ignored, it was closed |
| 18:28.51 | Katty | sometimes that's the same thing. |
| 18:28.52 | file | at that point in time there wasn't a bug in that, it was something else |
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| 18:30.51 | oryn | hmm well its defo still doing it |
| 18:31.07 | oryn | I have one way audio |
| 18:31.19 | oryn | right now to leave out rtp native |
| 18:31.58 | file | that will stop it, yes, but an issue would still be appreciated |
| 18:32.07 | Katty | i think i'd re-open that ticket! |
| 18:32.29 | file | he's not the reporter and it would just screw stuff up |
| 18:32.40 | oryn | its a real pain if yuor sip trunk is ipv6 and your phone is ipv4 |
| 18:33.41 | oryn | yup that fixes it :P |
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| 18:34.58 | oryn | no if only i could tell asterisk which ipv6 address to make outgoing calls from |
| 18:35.48 | oryn | ok so never reported a bug on asterisk before, is there as manpage or a wiki showing how? |
| 18:37.57 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |
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| 18:47.42 | jmetro | I have knights for the NI |
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| 19:16.10 | oryn | thanks file: |
| 19:19.37 | trend1 | Can anybody point me to some documentation that explains "Virtual Hold", iv googled around and have not been able to find anything |
| 19:21.09 | jmetro | park is better. |
| 19:21.38 | trend1 | how so? |
| 19:22.28 | jmetro | puts the caller in a parkinglot, provides music, able to setup for multi tennant, provides ability for any phone to pick up the hold |
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| 19:24.57 | trend1 | Yes, but i would like the caller who is sitting in a queue for a long time to have the option of leaving their number and for them to receive a call when their spot is up |
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| 19:29.52 | jmetro | that is a part of queues. |
| 19:29.58 | jmetro | automated queue callback |
| 19:32.37 | trend1 | either way i have not been able to find much documentation on it, could you point me in the right direction? |
| 19:41.42 | jmetro | if you google asterisk queue callback you should get results |
| 19:41.52 | jmetro | i dont personally have an example |
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| 19:48.16 | *** join/#asterisk Karl370 (~Karl370@12.139.131.2) |
| 19:52.22 | Karl370 | Is there a way of dynamically setting the context in the dialplan? |
| 19:53.52 | oryn | file: I have commented on the original bug #22127 do you think this will grab anyone's attention? |
| 19:54.09 | file | I really wanted a new issue... |
| 19:54.24 | jmetro | Karl370: what. |
| 19:54.24 | oryn | ok I'll do that too if you like |
| 19:54.38 | file | because that issue wasn't a bug at the time, but it's possible it was broken afterwards specifically for chan_sip |
| 19:55.02 | jmetro | Karl370: do you mean like Goto(${CONTEXT}-internal-extensions,s,1) |
| 19:55.41 | file | oryn, and bridge_native_rtp can still be active - it has two bridges... remote and local |
| 19:56.11 | Karl370 | jmetro: I want to separate some companies. I'd like to read from my odbc database the context, then set it. |
| 19:56.55 | file | oryn, remote reinvites and local just bridges in the RTP stack... so a "sip set debug on" and an "rtp set debug on" would also be useful |
| 19:57.04 | jmetro | Karl370: from which point, because at any time you could do something like Set(Var=Script(use-database-to-get-context.php)) and then Goto(var-dialplan,s,1) |
| 19:57.24 | jmetro | with formatting, and etc. |
| 19:59.51 | Karl370 | What I'm after is a way of adding multiple companies to my system & keep them separate. I looked into realtime extensions, but I don't like way you put the dialplan entirely in the db. It uses a switch => somethingoranother statement. |
| 20:01.15 | Karl370 | I'd rather put a bunch of logic in my dialplan, set variables or whatnot & then perform my own setting of the context, to keep the companies separate, but still allow a usable & dynamic system. |
| 20:01.33 | jmetro | You can use realtime extensions and have the dialplan be hard-coded in text. |
| 20:02.04 | jmetro | raeltime extensions are actually really nice <.< |
| 20:02.16 | jmetro | and what youre referring to is a multi-tennant system |
| 20:03.16 | jmetro | in which case, what i do, is numbers that call in all get dumped into incoming, and then routed out to their companys contexts |
| 20:03.27 | jmetro | all their phones sit in their [Company] context |
| 20:03.36 | Katty | ^- one stands. |
| 20:03.39 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 20:03.50 | Katty | there's ALWAYS one in every group. |
| 20:04.13 | jmetro | whatd i braek <.< |
| 20:05.52 | jmetro | Karl370: anyway, the easiest thing to do if youre looking at dynamic stuff, is to use variables, which i think you should have no problem with. |
| 20:06.37 | Karl370 | jmetro: I'll be building a website that the users will be able to make changes to their system. The different companies will be able to customize their own system. It just seemed kind of backwards to put the dialplan code directly in the db (& error prone). |
| 20:07.10 | jmetro | database driven dialing is actually a lot more flexible, but Asterisk Realtime does not force you to use database driven dialing. |
| 20:07.56 | jmetro | Dialplan reload is the only reload i use in asterisk because we still use file-driven dialplans |
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| 20:09.07 | Karl370 | I'm trying to design something that won't require manual interaction when adding a new company |
| 20:09.59 | oryn | file: sorry to bug you again, but would you say that the bug I describe is a regression? |
| 20:10.06 | file | yes |
| 20:10.10 | jmetro | so on your website, you type in a company name, context, shortname, etc. then in your dialplan, you use variables, ive said it three times now |
| 20:10.12 | file | WELL |
| 20:10.21 | file | does it work in 11? (probably) |
| 20:10.21 | Katty | ^- deep subject. |
| 20:10.46 | jmetro | if youre deliberately ignoring the part where i tell you to set a variable via script and then goto(${variable}-dialplan) then ..i dunno. ill eat more tacos. |
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| 20:11.40 | oryn | file: #22564 |
| 20:11.59 | Katty | jmetro: i agree. |
| 20:12.06 | Katty | jmetro: that's very good course of action |
| 20:12.07 | file | oryn, did you attach console output, sip debug, and rtp debug? |
| 20:12.16 | Katty | jmetro: we need to somehow integrate tacos into our regular diaplan stuffs. |
| 20:12.34 | jmetro | Katty: "SendDigit" -> "SendTacos" find replace. |
| 20:12.51 | jmetro | SipAddHeader -> StuffTacosWith find replace |
| 20:12.52 | oryn | you mean the part where is says "switching from simple_bridge technology to native_rtp" (yes) |
| 20:13.45 | Katty | jmetro: YES |
| 20:13.52 | Katty | file: integrate the taco stuff immediately. kthx |
| 20:14.17 | file | no |
| 20:14.48 | file | when you type "sip set debug on" and "rtp set debug on" it turns on SIP and RTP debugging, which outputs messages to the log about SIP and RTP traffic |
| 20:14.59 | file | that's going to be required for the issue |
| 20:15.13 | Katty | jmetro: well, we got rejected on the taco idea. |
| 20:15.26 | oryn | ok |
| 20:15.29 | Katty | jmetro: guess we'll have to eat them instead. |
| 20:15.34 | oryn | its alot to paste |
| 20:16.08 | file | you can attach files. |
| 20:16.12 | jmetro | Katty: Either that, or we re-write asterisk on our own from source. |
| 20:16.42 | Katty | considers. |
| 20:16.51 | Katty | mmm. rewrite asterisk or eat tacos |
| 20:16.52 | jmetro | Katty: or as some people say, "Make your own tacos from scratch" |
| 20:17.50 | Katty | i think eating tacos sounds more fun :< |
| 20:17.58 | Katty | can we just eat tacos instead? |
| 20:18.03 | jmetro | It does, especially crunchy ones. |
| 20:18.10 | Katty | :> |
| 20:18.18 | Katty | let's try this new proposal |
| 20:18.22 | Katty | file: can we have tacos? |
| 20:18.39 | file | Requests for tacos must be submitted 6-8 weeks ahead of consumption. |
| 20:18.41 | jmetro | Katty: A solid strategy, well said. |
| 20:18.54 | jmetro | Doh! Beaureucratic red-sauce tape. |
| 20:18.57 | Katty | we got told to put in a ticket :< |
| 20:19.31 | Katty | i know! let's use the same strategy everyone else does! we'll just start asking all of the IT people |
| 20:19.48 | Katty | surely ONE of them will give us tacos rather than telling us to put in a ticket |
| 20:19.56 | jmetro | Dear Tech Support, we have run out of tacos, fix immediately. |
| 20:19.56 | file | Katty, you get this: http://www.youtube.com/watch?v=yzC4hFK5P3g |
| 20:20.30 | Katty | watches. |
| 20:20.42 | Katty | it looks very Lisa Frank. |
| 20:21.13 | jmetro | Basically, we do this to the IT support, http://imgur.com/hkLTOr4 |
| 20:22.13 | Katty | jmetro: YES. |
| 20:22.56 | Katty | file: suddenly! kraft macaroni and cheese. |
| 20:24.25 | file | Katty, of course |
| 20:26.09 | jmetro | Ever had a pasta sandwich? |
| 20:29.07 | Karl370 | jmetro: I don't have a problem setting a variable. So are you saying then that I should use variables to act as my own context..allowing or denying calling to different extensions based upon the some criteria? Or are you saying that you can actually set the ${CONTEXT} variable via a script? Btw, I like tacos too. |
| 20:31.01 | jmetro | Karl370: I'm saying you write dialplan like... Same=> n,Set(DESTINED-CONTEXT=Shell(findDestiny.php)) |
| 20:31.19 | jmetro | which would do something like Set(DESTINED-CONTEXT = Company3) |
| 20:31.44 | jmetro | then you do Same=> n,Goto(${DESTINED-CONTEXT}-dialplan,s,1) which would execute Goto(Company3-dialplan,s,1) |
| 20:32.25 | jmetro | findDestiny.php being a script which contains your SQL query to find the context based on ???!!!?? that you provide it. |
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| 20:36.50 | Karl370 | jmetro: I don't have a problem finding the context to use with a goto. What I'm desiring is a way of doing the above without having to add a separate company3-dialplan code to my extensions.conf file. i.e. I'm trying to get it all dynamic. Under your solution it appears you would still have to create the company3-dialplan section manually. |
| 20:37.22 | jmetro | Indeed you would, as i dont believe you can match contexts the same way you can extens. |
| 20:37.30 | jmetro | though you could have a "context" be an exten and its derivatives. |
| 20:41.17 | *** join/#asterisk obnauticus (~obnauticu@unaffiliated/obnauticus) |
| 20:41.29 | obnauticus | Can anyone here reccomend a wholesale IAX trunk provider? |
| 20:41.34 | Karl370 | jmetro: I'm not following you on the "context" as an exten and it's derivates statment. Coud you explain a bit more? |
| 20:41.42 | obnauticus | I used to have a go-to provider that I forgot the name of |
| 20:43.20 | obnauticus | dammit, it was voipjet |
| 20:43.20 | jmetro | Karl370: it gets messy, you would be better off with DB-driven dialing |
| 20:43.40 | oryn | file: so there we go, lots of debugging info added |
| 20:43.50 | file | oryn, you can attach files you know... |
| 20:44.26 | oryn | ahh. didn't see that |
| 20:45.00 | file | it would also be useful to know if the same issue occurs in Asterisk 11 |
| 20:45.39 | oryn | I was too busy trying to figure out how to edit up my post with vi to not publish mycellphone number to the world |
| 20:46.28 | oryn | is it likely to work with the current version of dahdi? |
| 20:46.40 | file | sure. |
| 20:46.54 | oryn | ok I'll grab it and build it |
| 20:46.55 | file | DAHDi will work with 1.8, 11, or 12 |
| 20:47.14 | file | if you are deploying a system then right now you should be using 11 |
| 20:48.44 | oryn | wow! the server is slow |
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| 20:48.58 | oryn | only getting 300K/s |
| 20:50.07 | oryn | do you know if google voice still works? |
| 20:53.03 | Karl370 | jmetro: thanks for your help. I'll look more at the db-driven dialing. It just seemed a little kludgy putting all the code in there & dealing with priority order. |
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| 21:02.44 | trend1 | anyone here familiar with orderlyq? |
| 21:04.28 | oryn | file: ok proven that it works in 11 |
| 21:12.14 | file | it does have bridge_native_rtp, it's just part of the core and not a separate module |
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| 21:34.18 | pbxbrian | trend1: familiar yes - you having problems? |
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| 22:14.00 | P424D0X | Who can help me.. I can't generate SIP REGISTER frames and therefore 'sip show registry' tell providers are unregistred.. :-( |
| 22:16.44 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:17.32 | P424D0X | What can I do to tell asterisk send out SIP REGISTER frames to my voip provider?! |
| 22:21.59 | navaismo | add the register string in the sip.conf then sip reload |
| 22:24.01 | *** join/#asterisk tunage (joe@2600:3c01::f03c:91ff:feae:9ef3) |
| 22:25.02 | P424D0X | navaismo: "register => AAA:BBB@sip.1und1.de/AAA" <--- in sip_registrations.conf, and "#include sip_registrations" in sip.conf under [general] section.. |
| 22:25.26 | tunage | I am dragging a customer away from asterisk 1.4, about to install asterisk 12 but I will need a gui for the guy or else he will go crazy. Whats an easy gui to install on top of base asterisk? not a prepackaged cd? |
| 22:25.49 | [TK]D-Fender | P424D0X: As long as that is included in the right place it could work. |
| 22:26.03 | [TK]D-Fender | P424D0X: That file is not a direct Asterisk config file. |
| 22:26.42 | [TK]D-Fender | tunage: * 12 is not even in offical beta, let alone RC, or released |
| 22:26.58 | [TK]D-Fender | tunage: Very bad time to think that it's OK to run it in production. |
| 22:27.44 | P424D0X | [TK]D-Fender: But it doesn't work.. "sipgate.de:5060 N AAAAAAA 300 Unregistered" <--- is the output of command: sip show registry |
| 22:27.47 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:28.02 | tunage | I am dragging a customer away from asterisk 1.4, about to install asterisk 11 but I will need a gui for the guy or else he will go crazy. Whats an easy gui to install on top of base asterisk? not a prepackaged cd? |
| 22:28.23 | [TK]D-Fender | P424D0X: Then it's in the wrong place or the file isn't getting read at all, etc |
| 22:29.25 | [TK]D-Fender | tunage: FreePBX is pretty much the #1 go-to free option. Elastix (IIRC) has recently moved completely away from it to their own GUI. Then there are commercial ones |
| 22:29.43 | tunage | [TK]D-Fender but thats on a cd? |
| 22:29.52 | [TK]D-Fender | FreePBX rolls ISO's as well |
| 22:30.19 | [TK]D-Fender | AsteriskNOW is very similar as well. |
| 22:30.20 | tunage | does it have a standalone install? non-iso? |
| 22:30.26 | [TK]D-Fender | Each with their minor slants |
| 22:30.36 | P424D0X | [TK]D-Fender: No, the files are correct and correctly parsed by asterisk.. |
| 22:30.45 | [TK]D-Fender | You can install FreePBX on top of existing servers as well |
| 22:30.55 | [TK]D-Fender | P424D0X: Then location counts |
| 22:31.23 | P424D0X | [TK]D-Fender: What did you mean by location?! |
| 22:33.03 | P424D0X | [TK]D-Fender: There are going out SIP frames to my sip provider, but no REGISTER frame.. :-/ |
| 22:38.44 | [TK]D-Fender | P424D0X: You have made a mistake somewhere and you haven't shown us enough to say any more yet |
| 22:42.37 | P424D0X | [TK]D-Fender: Okay.. This is my suspicion too.. I'll post you enough information.. My asterisk system is from openwrt running on a router.. Version is 1.8.10.. |
| 22:43.13 | Katty | looks at [TK]D-Fender |
| 22:43.16 | [TK]D-Fender | P424D0X: Show us folder and config file dumps. |
| 22:43.54 | *** part/#asterisk asghar144 (~asghar144@host118-24-dynamic.8-87-r.retail.telecomitalia.it) |
| 22:45.43 | P424D0X | [TK]D-Fender: Yes.. I'am generating a file with all information (debug and verbose messages, and configs).. Upload to pastebin.com.. The link will come soon.. :-) |
| 22:57.06 | P424D0X | [TK]D-Fender: Okay.. Here are my outputs and configs: "http://pastebin.com/1GY4x476" |
| 22:58.01 | Katty | stares at [TK]D-Fender |
| 22:58.12 | Katty | [TK]D-Fender: don't you have..somewhere to be soon |
| 23:04.06 | navaismo | P424D0X: edit something in the sip conf then do a sip reload with sip debug enabled and show us the complete sip debug |
| 23:05.04 | P424D0X | navaismo: What should I edit in sip.conf?! :-/ |
| 23:05.12 | [TK]D-Fender | P424D0X: well we DO see that * tried |
| 23:05.33 | [TK]D-Fender | P424D0X: If you want to watch it again, do a "sip reload" |
| 23:05.48 | P424D0X | [TK]D-Fender: Ah, okay.. |
| 23:05.56 | [TK]D-Fender | [18:27]P424D0X[TK]D-Fender: But it doesn't work.. "sipgate.de:5060 N AAAAAAA 300 Unregistered" <--- is the output of command: sip show registry |
| 23:06.09 | [TK]D-Fender | P424D0X: this line you showed us earlier wasn't even for the same registration |
| 23:08.37 | P424D0X | ANDROMEDA*CLI> sip set debug on |
| 23:08.37 | P424D0X | SIP Debugging enabled |
| 23:08.37 | P424D0X | asterisk*CLI> sip reload |
| 23:08.37 | P424D0X | asterisk*CLI> sip show registry |
| 23:08.37 | P424D0X | asterisk*CLI> |
| 23:08.37 | P424D0X | <PROTECTED> |
| 23:08.55 | P424D0X | This is the output in CLI.. :-((( |
| 23:09.48 | P424D0X | But when I restart * and login to CLI, than the output works correctly.. |
| 23:09.52 | [TK]D-Fender | P424D0X: the fact we don't see anything is not consistent |
| 23:09.57 | [TK]D-Fender | P424D0X: Nothing is matching. |
| 23:10.47 | [TK]D-Fender | [19:09]P424D0XBut when I restart * and login to CLI, than the output works correctly.. <- and you are not telling us when this changes. |
| 23:11.56 | P424D0X | I have edited sip.conf before the above output, but no changes.. :-( |
| 23:13.57 | P424D0X | [TK]D-Fender: What do we need to solve my problem?! |
| 23:15.09 | P424D0X | [TK]D-Fender: What should I type in or what should I try in the config files?! |
| 23:16.08 | [TK]D-Fender | P424D0X: "database show" <- go look up the key holding the registry info and delete it, then "sip reload" |
| 23:16.52 | P424D0X | 0 results found |
| 23:18.05 | P424D0X | [TK]D-Fender: Should I put a new key in database?! |
| 23:19.43 | [TK]D-Fender | go restart * manually and watch |
| 23:22.10 | P424D0X | [TK]D-Fender: Same output as you can see on pastbin.com.. Or should I post a complete output?! |
| 23:22.18 | *** join/#asterisk crienzo (~crienzo@unaffiliated/crienzo) |
| 23:22.30 | [TK]D-Fender | P424D0X: Show us. |
| 23:22.48 | P424D0X | [TK]D-Fender: Is it normal, that astdb size has 0 byte?! |
| 23:22.58 | [TK]D-Fender | no, that is bad |
| 23:23.13 | [TK]D-Fender | And could be the problem |
| 23:23.20 | [TK]D-Fender | delete the file and restart * |
| 23:23.30 | P424D0X | Ah!! Okay.. |
| 23:25.28 | P424D0X | [TK]D-Fender: When I delete astdb and restart * then the file is generated by * but 0 bytes.. Okay.. I'll post you my output.. |
| 23:27.06 | P424D0X | [TK]D-Fender: http://pastebin.com/ZbPHQ0A7 |
| 23:27.21 | talntid | you have enough disk space? :) |
| 23:27.43 | [TK]D-Fender | that would also be highly important |
| 23:28.38 | P424D0X | talntid: Good idea, but: |
| 23:28.51 | P424D0X | root@ANDROMEDA:/usr# df |
| 23:28.51 | P424D0X | Filesystem 1K-blocks Used Available Use% Mounted on |
| 23:28.51 | P424D0X | rootfs 29440 13912 15528 47% / |
| 23:28.51 | P424D0X | /dev/root 1792 1792 0 100% /rom |
| 23:28.51 | P424D0X | tmpfs 63196 7112 56084 11% /tmp |
| 23:28.52 | P424D0X | /dev/mtdblock7 29440 13912 15528 47% /overlay |
| 23:28.52 | P424D0X | overlayfs:/overlay 29440 13912 15528 47% / |
| 23:28.53 | P424D0X | tmpfs 512 0 512 0% /devoice |
| 23:30.15 | navaismo | ~pb |
| 23:30.15 | 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. |
| 23:30.55 | talntid | tmpfs /devoice is interesting. heh :) |
| 23:31.18 | P424D0X | navaismo: I know.. But it was only a few lines.. ;-) Nothing matter!! :D |
| 23:32.10 | navaismo | few lines more than 15!!! ok we have different paramters about "few" |
| 23:32.18 | [TK]D-Fender | <PROTECTED> |
| 23:32.32 | [TK]D-Fender | I'm wondering if dnsmgr is coming back empty on this. |
| 23:32.42 | [TK]D-Fender | Go check your nameservers |
| 23:32.50 | [TK]D-Fender | and then disable that next if not resolved |
| 23:34.08 | P424D0X | [TK]D-Fender: Hmm.. I /etc/resolv.conf the nameserver is 127.0.0.1 |
| 23:34.37 | [TK]D-Fender | That sure doesn't look good. |
| 23:35.00 | [TK]D-Fender | maybe ... just MAYBE ... your system should have an actual name server to lookup those hosts with... |
| 23:38.08 | P424D0X | [TK]D-Fender: Hmm.. In openwrt in file resolv.conf there is a line "nameserver 127.0.0.1".. But when I look to webif of my router the nameserver is set to 195.50.140.180.. |
| 23:40.56 | P424D0X | There are two files on my openwrt system "resolv.conf.ppp" and "resolv.conf.auto" which consist the following entries: |
| 23:41.00 | P424D0X | nameserver 195.50.140.180 |
| 23:41.00 | P424D0X | nameserver 195.50.140.114 |
| 23:44.51 | [TK]D-Fender | go check it from CLI |
| 23:46.07 | P424D0X | What should I type in?! |
| 23:47.23 | P424D0X | !ping sip.1und1.de worked fine.. |
| 23:49.11 | P424D0X | [TK]D-Fender: But when I ran twice, then Asterisk ending (0) |
| 23:50.18 | [TK]D-Fender | [19:49]P424D0X[TK]D-Fender: But when I ran twice, then Asterisk ending (0) <- ? |
| 23:52.18 | talntid | P424D0X, can you delete the astdb file, clear asterisk log |
| 23:52.22 | talntid | and start asterisk |
| 23:52.27 | talntid | and pastebin the new contents of the log |
| 23:52.28 | P424D0X | !ping sip.1und1.de |
| 23:53.11 | P424D0X | talntid: Okay.. |
| 23:53.39 | talntid | obviously stop asterisk first |
| 23:56.05 | P424D0X | talntid: Yes!! Of course.. |
| 23:56.09 | P424D0X | ;) |
| 23:56.16 | talntid | ;) |
| 23:57.19 | P424D0X | talntid: http://pastebin.com/9sVzKKdH |
| 23:57.46 | talntid | did it re-make astdb? |
| 23:58.04 | talntid | nothing in the log indicates it did |
| 23:58.29 | P424D0X | talntid: Yes.. But empty.. |
| 23:58.58 | talntid | odd. it should log when it creates it |