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| 04:13.08 | linocisco | who are using grandstream GXP1405? |
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| 06:14.48 | linocisco | hi all. what is the different between ip phones with one sip lines and two or more sip lines? what can we do that? |
| 06:14.56 | linocisco | hi all. what is the difference between ip phones with one sip lines and two or more sip lines? what can we do that? |
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| 06:16.30 | Penguin | linocisco: Usually if you have more than one line, that allows more than one SIP account name on the phone. This is often not necessary. |
| 06:17.00 | linocisco | Penguin, yes. we can only talk or handle only one call at a time |
| 06:17.34 | Penguin | Most phones allow more than one call at a time. Especially in the case of call waiting. |
| 06:29.12 | linocisco | Penguin, how can I configure call waiting? |
| 06:29.33 | Penguin | I wouldn't know, since you didn't tell me what phone you have. |
| 06:31.01 | linocisco | Penguin, mine is GXP1405 grandstream |
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| 07:25.29 | hos7ein | hi |
| 07:25.39 | hos7ein | i am create this IVR : |
| 07:25.41 | hos7ein | http://paste.fedoraproject.org/51996/72271713/ |
| 07:26.05 | Penguin | That isn't IVR. |
| 07:26.17 | Penguin | That's an auto attendant. |
| 07:26.24 | Penguin | (digital receptionist) |
| 07:27.01 | Penguin | linocisco: Call waiting is probably enabled by default. If not, look in the menu. |
| 07:27.27 | hos7ein | Penguin: this is for test and i learning to create IVR |
| 07:27.34 | Penguin | That isn't IVR. |
| 07:28.03 | Penguin | Your learning has failed. |
| 07:29.09 | hos7ein | Penguin: how to can i playing Twice this line : |
| 07:29.14 | hos7ein | same => n(loop),Background(custom-menu) |
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| 07:29.35 | Penguin | Would you like me to show you the proper way to make that auto attendant? |
| 07:30.21 | hos7ein | Penguin: yeah ,tnx |
| 07:32.46 | Penguin | You only want to play two times and then disconnect? |
| 07:34.26 | hos7ein | Penguin: yeah, play Twice , if user not press digit, go hangup |
| 07:35.23 | Penguin | hos7ein: http://pastebin.com/Nwy3YQLh |
| 07:35.46 | Penguin | This is not IVR, though. It is an auto attendant. |
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| 07:42.00 | hos7ein | Penguin: tnx,but not worked! after calling show this msg : http://paste.fedoraproject.org/52003/72365913/ |
| 07:43.18 | Penguin | I see that you do not have the sound file sorry-youre-having-problems. |
| 07:43.35 | Penguin | Feel free to remove that part of the AA. |
| 07:44.25 | Penguin | I did make one mistake. I'll fix it now. |
| 07:45.01 | Penguin | I forgot to set the variable to 0 to start. |
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| 07:45.43 | Penguin | hos7ein: http://pastebin.com/tiwq5fRf |
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| 07:48.45 | hos7ein | Penguin: yeah its good :) it's worked |
| 07:48.55 | hos7ein | Penguin: tnx |
| 07:49.09 | Penguin | No problem. |
| 07:50.36 | hos7ein | Penguin: for learnig to create IVR ,are you give me refrence? |
| 07:51.02 | Penguin | ~book |
| 07:51.02 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:51.22 | Penguin | Both auto attendant and IVR can be found in the asterisk book. |
| 07:54.07 | hos7ein | tnx |
| 07:55.35 | hos7ein | Penguin: what Different Between attendant and IVR ? |
| 08:07.14 | Penguin | Auto attendant is a digital receptionist. |
| 08:07.39 | Penguin | IVR takes user input to process elsewhere. |
| 08:08.26 | Penguin | If you call a number and it says something like "Thanks for calling today. To reach Sales, press 1," that is an auto attendant. |
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| 08:10.06 | Penguin | If you call a number and it says something like "Please say or enter your account number," And you enter it, then says "Thank you. Please enter your phone number," that is an IVR. |
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| 08:16.31 | Penguin | hos7ein: Can you understand that difference? |
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| 08:21.05 | hos7ein | Penguin: huuummm, Slightly |
| 08:21.09 | hos7ein | tnx |
| 08:21.58 | linocisco | hi all |
| 08:23.28 | linocisco | Hi all, i would like to put callme button like skype on companywebsite, and when customers click to call via webpage, it should reach or ring at operator phone. how can I configure it? |
| 08:27.52 | Penguin | hos7ein: If it only asks for extension number, it's an AA. |
| 08:28.09 | Penguin | If the dialplan is using WaitExten(), it is an AA. |
| 08:28.56 | Penguin | If you have to enter bits of information either by speaking or keying digits, it is probably IVR. If your dialplan uses Read(), it might be an IVR. |
| 08:32.58 | hos7ein | Penguin: aha, tnx,vrer tnx |
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| 09:39.02 | linocisco | how to configure skype gateway on asterisk so that click to call via skype will reach my PBX and extensions at office? |
| 09:39.27 | linocisco | do we need special software addon or hardware card for skype gateway? |
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| 09:49.12 | jploh | skype gateway is software based? |
| 09:49.30 | jkroon | hi guys, last I looked it this it seemed that T.38 over IAX/2 was a problem. I was wondering whether anybody is aware of any progress having been made in this respect, or whether someone would be willing to assist me in seeing if a patch can be created? |
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| 10:04.48 | hebber | Could someone help me with a WHILE statement - can't get it to work - doesn't exit the loop |
| 10:04.50 | hebber | same => n,While($[${LEN(${string}) > 2}]) |
| 10:05.16 | hebber | My first while statement :) |
| 10:06.34 | hebber | The string ends up less than two chars - still the while doesn't stop |
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| 10:11.20 | hebber | Think I got it - should be: same => n,While($[${LEN(${string})} > 2}]) |
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| 10:13.28 | hebber | Think that worked :) thanks for listening - normally help to explain to others |
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| 10:24.40 | kaldemar | hebber: check again. |
| 10:25.23 | kaldemar | hebber: you have an extra } in there. |
| 10:25.56 | hebber | kaldemar: oh - thanks - hard to spot :) |
| 10:30.05 | hebber | kaldemar: yes - finally the While loop works |
| 10:30.33 | linocisco | jploh, whatever good and reliable? |
| 10:31.42 | jploh | linocisco: oh i don't use one. i have chan_skype licenses that have long been deprecated |
| 10:32.11 | linocisco | jploh, so what should I use if deprecated? |
| 10:33.13 | jploh | linocisco: you might be better off using Skype Connect and use SIP. http://www.skype.com/en/features/skype-connect/ |
| 10:34.08 | linocisco | jploh, is it free or paid? |
| 10:34.31 | jploh | linocisco: paid. i haven't used it but you can open a Skype Manager account for free |
| 10:35.23 | linocisco | jploh, skype manager account is free. but to use it is not fre |
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| 10:37.23 | jploh | linocisco: that's right |
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| 10:40.11 | linocisco | jploh, what is other free alternative? |
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| 10:54.52 | cdr1 | hi |
| 10:55.52 | cdr1 | somebody her? |
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| 11:05.41 | jploh | yes |
| 11:06.08 | cdr1 | thanks |
| 11:06.29 | cdr1 | I im looking for some explanation on the asterisk 12 pjsip conf |
| 11:06.53 | cdr1 | more specificaly, aor |
| 11:08.08 | jploh | sorry, i haven't migrated yet (*gasp*) but maybe someone else here can help you |
| 11:08.24 | cdr1 | thanks anyway |
| 11:08.41 | jploh | have you followed the steps here: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip ? |
| 11:08.52 | cdr1 | yes |
| 11:09.21 | cdr1 | but i get a message in the log : aor not found for 6000 (endpoint) |
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| 11:17.52 | phix | Any good GUI's for user to set vaction messages or redirects or what not? |
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| 11:21.51 | danfromuk | Hi, is anyone here from Israel? |
| 11:23.22 | phix | Probably, this is a very globalised channel |
| 11:23.58 | danfromuk | Can you recommend a router that is suitable for use with SIP and a place to purchase it? We've got a client with a bezeq supplied router and regardless of what i do, it blocks SIP. The device can register but the call drops after 30 seconds due to one of the SIP packets not being received. |
| 11:24.47 | danfromuk | That question is only for someone in Israel. I'm not familiar with the router/pc market in israel. |
| 11:25.36 | WIMPy | Are you just talking about a simple router or something with integrated SIP client? |
| 11:25.58 | phix | I probably need to turn themselves off on Saturday, that would be the only difference |
| 11:26.00 | WIMPy | Or is the later what you have an want to get rid of? |
| 11:26.10 | phix | I = They |
| 11:27.24 | danfromuk | simple router |
| 11:27.38 | jploh | are you sure it's not your isp dropping it? |
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| 11:34.37 | danfromuk | jploh: I'm not sure but SIP is heavily used in Israel so i doubt its being blocked. |
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| 11:38.45 | Greenlight | Isn't Background() supposed to direct to the "t" extension if no input is given from the caller within the timeout? |
| 11:39.43 | WIMPy | ... after the message finished. |
| 11:39.53 | Greenlight | Yes |
| 11:40.04 | Greenlight | Hmm |
| 11:40.15 | Greenlight | And that behaviour hasn't been changed ? |
| 11:40.26 | kaldemar | no, it goes to the next priority. |
| 11:40.48 | Greenlight | That is what appears to be happening, but I was expecting it to goto "t" |
| 11:41.06 | kaldemar | Background doesn't even have a timeout. |
| 11:41.08 | Greenlight | From https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions -> "When the caller waits too long before entering a response to the Background() or WaitExten() applications, and there are no more priorities in the current extension, the call is sent to the t extension." |
| 11:41.47 | Greenlight | I assumed it used the builtin timeout |
| 11:42.06 | kaldemar | that's likely to be wrong on the part of Background. would need to test it. |
| 11:42.19 | Greenlight | Yes, I can certainly use WaitExten |
| 11:42.19 | WIMPy | Uh, a dynamic one. |
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| 13:40.16 | bulkorok | anyone knows a useable dialer-software that works with SugarCRM?! |
| 13:49.49 | Greenlight | Any ideas what the cause of this error would be: [Nov 6 13:49:18] ERROR[4243]: res_rtp_asterisk.c:2248 ast_rtcp_write_rr: RTCP RR transmission error, rtcp halted: Operation not permitted |
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| 13:50.41 | Greenlight | There's 30 or so calls in progress on this box, and I'm not getting any reports of problems, but still seeing that error a few times a minute |
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| 13:58.33 | anonymouz666 | dialer software redirects me to AMD. AMD just does not work and there are companies that makes tons of money selling it :-) |
| 13:59.02 | Greenlight | AMD does work, just don't expect 100% accuracy |
| 13:59.53 | Greenlight | Very often though the "delay" in connecting the call caused by AMD, is undersirable. |
| 14:00.09 | anonymouz666 | I don't expect nothing to work 100% |
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| 14:01.30 | anonymouz666 | Greenlight: mobile phones in general the accuracy is very very low. |
| 14:02.08 | Greenlight | There are "other methods" (TM) of detecting mobile calls which go straight to voicemail |
| 14:02.57 | Greenlight | AMD may well need some tweaking but generally we fine accuracy between 50% and 90% |
| 14:03.22 | Greenlight | Yes, there can be issues on mobiles where there's a lot of background noise if your thresolds are not set right |
| 14:04.41 | anonymouz666 | We have 30-40% that's very very low. Almost tweak every configuration existent. |
| 14:05.36 | Greenlight | That does seem on the low side, but if you've adjusted to air on the side of caution and avoid false positives, then 40% isn't too bad. |
| 14:05.37 | anonymouz666 | I can't sell that. you can't guarentee anything. that's unnacceptable for most contact centers. |
| 14:05.59 | Greenlight | Do you play a very short audio file *before* doing your AMD |
| 14:10.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:10.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.05 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 14:13.06 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 14:16.48 | WIMPy | I have a very simple call queueing issue. It's just a periodic announcement with some IVR options. But the issue is that I need to ensure calls are delivered in the order they called. |
| 14:17.15 | WIMPy | Any suggestions how to sensibly do that part? |
| 14:18.44 | *** join/#asterisk jhlavacek (~jirka@80.215.9.33) |
| 14:19.13 | [TK]D-Fender | Sounds like Queue() already... |
| 14:20.10 | WIMPy | But that doesn't esure order, does it? |
| 14:20.34 | Greenlight | SUre does |
| 14:20.46 | WIMPy | As far as I understood it has the same issue that another call can cut in line while another hears the announcement. |
| 14:21.49 | Greenlight | Hmm I'd always presumed it would just pull the caller mid-announcement, if it doesn't then, yea, that's another problem with Queue |
| 14:22.02 | [TK]D-Fender | WIMPy: Queue is an ordered line |
| 14:22.08 | [TK]D-Fender | WIMPy: they are delivered one by one |
| 14:22.26 | [TK]D-Fender | WIMPy: So far it fit what you seem to need |
| 14:23.05 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:23.24 | WIMPy | Ok, I guess I have to try that then and see if it fits. |
| 14:23.40 | *** join/#asterisk Syco (d9494f22@gateway/web/freenode/ip.217.73.79.34) |
| 14:26.50 | Syco | hello, I'm having trouble with an atxfer from ami, when i send the command "Action: Atxfer\r\nChannel: Agent/7001\r\nContext: atxfer\r\nExten: a7002\r\n\r\n" in the cli I get "WARNING[3608][C-00000053]: features.c:2728 builtin_atxfer: Extension '70' does not exist in context 'atxfer'". That's right cause context atxfer in my dialplan accept only _aXXXX, but why the 'a' is not sent with the rest of the extension?? could this be a bug? |
| 14:29.21 | Greenlight | I've got a feeling (could well be totally wrong) that the Atxfer command works by mimicing DTMF tones being sent |
| 14:29.24 | [TK]D-Fender | Syco: I don't see the 2 digits after 70 either... so I don't think it's executing what you think it's executing... |
| 14:31.27 | Greenlight | Lets see the CLI and the AMI of an attempt. tcpdump for the AMI to verify exactly what's being sent |
| 14:31.47 | Syco | ok, is coming tnx |
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| 14:46.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 14:56.12 | Syco | here it is, http://pastebin.com/NRR9YMfn |
| 14:58.33 | [TK]D-Fender | pcap might help |
| 15:00.39 | Syco | https://db.tt/rH91vddQ |
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| 15:04.07 | Syco | capture is on the loopback iface, that's the full dump |
| 15:05.51 | Qwell | weird |
| 15:06.28 | [TK]D-Fender | Syco: What are you runinng? |
| 15:06.28 | [TK]D-Fender | running* |
| 15:08.00 | Syco | *11.6 on centos 6.4 |
| 15:08.07 | Syco | 64 bit |
| 15:08.42 | gerritfromsa | your definitely passing the full a7002 ... |
| 15:09.22 | boom^time | What's the best way of turning asterisk into a service on ubuntu 12.04? |
| 15:10.14 | *** join/#asterisk smellis (~stephan@24.49.219.131) |
| 15:10.21 | gerritfromsa | wonder what happened to SequenceNumber: 43762 ? |
| 15:10.44 | Qwell | okay, that manager action is screwy |
| 15:11.06 | Qwell | Syco: pastebin your features.conf? |
| 15:12.25 | gerritfromsa | Syco, did you see the Message: Missing action in request ? |
| 15:12.48 | Qwell | gerritfromsa: ignore all of that - the atxfer was done properly early on |
| 15:13.10 | boom^time | nm make config |
| 15:13.33 | Syco | http://pastebin.com/3JDWUdea |
| 15:13.45 | gerritfromsa | got it |
| 15:14.19 | Syco | no, never seen the missing action message |
| 15:15.18 | Qwell | Syco: with the way this works, I'm betting that you can't use non-numeric extensions. |
| 15:16.22 | Syco | in other point of the software i dial Local/q7002 and it works fine |
| 15:16.35 | gerritfromsa | it's at the end of the trace , usingwireshark -right click an the 1st packet and select FOLLOW TCP STEAM you'll see the response from the server(Blue) with this message |
| 15:17.40 | Qwell | Syco: dialing is fine - but this manager action is weired |
| 15:17.42 | Qwell | weird* |
| 15:18.01 | Syco | @gerritfromsa: that happens when I press enter in the telnet console without any command |
| 15:18.01 | Nugget | telnet is eeeeeeevil! |
| 15:18.22 | gerritfromsa | gotcha |
| 15:18.45 | gerritfromsa | did you notice the missing sequence no? |
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| 15:21.04 | gerritfromsa | brb |
| 15:21.41 | *** part/#asterisk gerritfromsa (~gerritfro@8ta-229-221-142.telkomadsl.co.za) |
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| 15:24.16 | Syco | i noticed the missing number, but is not in the dump, nor in the console |
| 15:24.24 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 15:30.14 | Syco | https://db.tt/7qLJDYxN another dump, as you can see it skips a few sequence numbers |
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| 15:40.38 | plut0 | when making an outgoing phone call, with certain phone numbers, when the other side answers, the call ends seconds later. with other numbers it works fine. is this a codec issue or something else? |
| 15:43.40 | [TK]D-Fender | if you get audio at all, no |
| 15:43.45 | *** part/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 15:43.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
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| 15:46.44 | Syco | ami atxfer just sends the digit on the channel, in manager.c:3903 the extension is appended as AST_FRAME_DTMF |
| 15:47.14 | Syco | is it possible that AST_FRAME_DTMF discard non digit chars? |
| 15:47.52 | plut0 | [TK]D-Fender: depends on who you call, they each use different codecs, no? |
| 15:49.56 | [TK]D-Fender | plut0: "phone numbers" don't have codecs |
| 15:50.13 | plut0 | [TK]D-Fender: ok i'm probably not explaining it well then |
| 15:50.23 | [TK]D-Fender | plut0: The service or device you are calling to uses codecs |
| 15:50.31 | [TK]D-Fender | plut0: Any call that doesn't terminate instantly on answer is not a codec issue |
| 15:50.50 | [TK]D-Fender | And the SIP debug would tell you that |
| 15:50.54 | plut0 | [TK]D-Fender: ok. what would it be then? |
| 15:50.55 | [TK]D-Fender | Are you looking at it? |
| 15:51.01 | [TK]D-Fender | I'd suggest looking |
| 15:51.06 | [TK]D-Fender | that's usualy a good tip-off |
| 15:53.49 | Katty | hi kids. |
| 15:54.55 | Katty | [TK]D-Fender: thanks for the links |
| 15:54.59 | Katty | i still need to look at them |
| 15:55.03 | file | Greenlight, you were excited awhile ago about my comments re: spying/whispering in the new REST interface weren't you? |
| 15:55.10 | Katty | HI FILE |
| 15:55.20 | file | hihi |
| 15:56.09 | *** join/#asterisk navaismo (~navaismo@189.191.206.207) |
| 15:56.23 | Katty | this trainer needs to come in pink. |
| 15:56.27 | Katty | why does it not come in pink?! |
| 15:56.41 | Katty | dear kinetic. plz to provide products in pink. kthx, katty |
| 15:57.07 | Katty | [TK]D-Fender: should i get this Floor Mat thing? |
| 15:58.37 | [TK]D-Fender | ~8ball should Katty get this Floor Mat thing? |
| 15:58.37 | infobot | Negative. |
| 15:58.45 | Katty | righto. |
| 15:58.49 | Katty | infobot: good bot. |
| 15:58.49 | infobot | Katty: thanks |
| 15:59.22 | Katty | Authentic Road Feel! how exciting. |
| 15:59.40 | Katty | i hope it doesn't include the big trucks that like making me feel nervous with their side view mirrors getting uncomfortably close. |
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| 16:59.47 | zemmali-voip | hello, please i need help to <<Development in JAVA code and configuration control of VoIP ainsique web services routing server.>> |
| 17:01.15 | [TK]D-Fender | ~ask |
| 17:01.15 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 17:05.38 | eirirs | I am here against my will. |
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| 17:17.35 | plut0 | can someone review this log? i can make outgoing calls but receiving is having issues. the phone rings, i answer but there is no audio either way. http://pastebin.com/hfYVFZGZ |
| 17:18.32 | prasha | I am trying to do simple conference call but, I'm getting following msg |
| 17:18.42 | prasha | Parsing '/etc/asterisk/meetme.conf': Found == Spawn extension (default, 5001, 2) exited non-zero on 'SIP/1003-00000007' |
| 17:19.35 | [TK]D-Fender | prasha: Show the actual complete call. Those messages say nthing |
| 17:19.48 | *** join/#asterisk eric_hill (eric_hill@wsip-184-180-163-58.ks.ks.cox.net) |
| 17:19.49 | [TK]D-Fender | plut0: what have you set up on your router? |
| 17:20.41 | plut0 | [TK]D-Fender: for firewall rules? |
| 17:20.48 | eric_hill | Telmex in Mexico only supports E1/CAS/HDB3/MFC/R2 - does anyone know of an IP SIP gateway that supports that signalling type? The Digium G100/200 don't. :( |
| 17:21.13 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 17:21.21 | prasha | D-Fender : Thanks for reply..following is complete log |
| 17:21.22 | prasha | == Using SIP RTP CoS mark 5 |
| 17:21.24 | prasha | <PROTECTED> |
| 17:21.24 | prasha | <PROTECTED> |
| 17:21.24 | prasha | <PROTECTED> |
| 17:21.24 | prasha | <PROTECTED> |
| 17:21.24 | prasha | <PROTECTED> |
| 17:22.48 | [TK]D-Fender | ~PASTEBIN |
| 17:22.48 | 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. |
| 17:22.50 | [TK]D-Fender | ~pb |
| 17:22.50 | 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. |
| 17:22.52 | [TK]D-Fender | ^^^ |
| 17:23.25 | plut0 | [TK]D-Fender: here are my firewall rules, http://pastebin.com/2SDixmaa |
| 17:23.49 | plut0 | [TK]D-Fender: the phone adapter is a pap2t, i don't see any rules for that |
| 17:24.55 | [TK]D-Fender | plut0: JABBER: asterisk OUTGOING: <iq from='myusername@gmail.com/TalkF4441770' to='+15558675309@voice.google.com/srvenc-8yEJj93YHPvht62ZsTTbhOzLWuLwc8kU' type='set' id='aaaaw'><session type='candidates' id='SIP1944430449@10.48.62.237' initiator='+15558675309@voice.google.com/srvenc-8yEJj93YHPvht62ZsTTbhOzLWuLwc8kU' xmlns='http://www.google.com/session'><candidate name='rtp' address='71.244.121.220 |
| 17:24.56 | [TK]D-Fender | ' port='19218' username='733fce9b30525ebc' password='32d42d39314af9bb' preference='0.90' protocol='udp' type='stun' network='0' generation='0'/><transport xmlns='http://www.google.com/transport/p2p'/></session></iq> |
| 17:25.15 | prasha | D-Fender : sorry, about that |
| 17:25.30 | prasha | D-Fender : here is the log .. http://pastebin.com/SwnkwySb |
| 17:26.26 | [TK]D-Fender | 19218 is no in the range you forwarded |
| 17:26.26 | [TK]D-Fender | $IPTABLES -t nat -A PREROUTING -i $EXTIF -p udp --dport 19900:20000 -j DNAT --to 192.168.0.231 |
| 17:26.26 | [TK]D-Fender | from here |
| 17:26.37 | [TK]D-Fender | prasha: "sip set debug on" <- lets see who ends the call. |
| 17:27.04 | plut0 | [TK]D-Fender: ok let me check that |
| 17:28.19 | Penguin | You forwarded 18000-20000, but only NATted 19900-20000. |
| 17:28.27 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 17:29.09 | [TK]D-Fender | yup, inconsistency is a killer |
| 17:31.22 | plut0 | [TK]D-Fender: ok i updated the firewall rules, both are using 18000:20000 now |
| 17:31.30 | plut0 | i tried again but it still failed |
| 17:32.56 | plut0 | [TK]D-Fender: new log, http://pastebin.com/iV2ra2KC |
| 17:34.58 | prasha | D-Fender: these are my sip debug logs ..http://pastebin.com/3xbBB5Gi |
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| 17:37.45 | [TK]D-Fender | JABBER: asterisk OUTGOING: <iq from='myusername@gmail.com/TalkF4441770' to='+15558675309@voice.google.com/srvenc-8yEJj93YHPsWSXFnVz2MAsq4jvymFjix' type='set' id='aaabb'><session type='candidates' id='SIP368748051@10.68.253.74' initiator='+15558675309@voice.google.com/srvenc-8yEJj93YHPsWSXFnVz2MAsq4jvymFjix' xmlns='http://www.google.com/session'><candidate name='rtp' address='192.168.0.231'... |
| 17:37.46 | [TK]D-Fender | ...port='18050' username='32356ce87103fac5' password='secret' preference='1.00' protocol='udp' type='local' network='0' generation='0'/><transport xmlns='http://www.google.com/transport/p2p'/></session></iq> |
| 17:38.01 | [TK]D-Fender | Looks like your * is advertising it's PRIVATE address for RTP, and not the PUBLIC on. |
| 17:38.06 | Penguin | its? |
| 17:38.08 | [TK]D-Fender | Seems lie an improper NAT setup |
| 17:38.19 | [TK]D-Fender | make sure your sip.conf is set up properly for this. |
| 17:38.27 | [TK]D-Fender | It should cover both for settings..... |
| 17:38.33 | [TK]D-Fender | (IIRC) |
| 17:38.42 | file | it's okay for that to be in the jingle transport |
| 17:38.48 | file | Google Voice relays use symmetric RTP |
| 17:39.09 | file | so it'll start sending to your public IP address/port as soon as the first RTP packet reaches 'em |
| 17:43.57 | [TK]D-Fender | ok, this is falling outside my experience to assist much more... |
| 17:44.03 | [TK]D-Fender | perhaps others can carry this onward... |
| 17:46.11 | Penguin | plut0: Did you ensure that any fancy SIP handling stuff is disabled on the router? |
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| 17:48.34 | moy | eric_hill: Sangoma Vega gateways do that |
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| 18:03.31 | prasha | I was trying to implement meetme using wiki but I am getting following type of error .No database handle available with the name of 'general' (check res_odbc.conf) |
| 18:03.45 | prasha | why its searching for general |
| 18:04.08 | prasha | I was able to setup 2 softphone and make them call each other |
| 18:04.45 | [TK]D-Fender | That has nothing to do with meetme. |
| 18:05.08 | [TK]D-Fender | And your paste expired and I never got to see the debug |
| 18:05.10 | [TK]D-Fender | do it again |
| 18:09.16 | prasha | these are general logs what i got...http://pastebin.com/80sRTRQF |
| 18:10.37 | prasha | D-Fender: these are the logs after debug ... http://pastebin.com/bQf94uQ5 |
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| 18:33.10 | plut0 | [TK]D-Fender: sorry i had to step away but i'm back now |
| 18:33.15 | plut0 | [TK]D-Fender: so you think it's a nat issue? |
| 18:33.39 | plut0 | i had a suspicion it would be |
| 18:35.01 | plut0 | how do you setup nat properly? i guess this is a mystery to me |
| 18:36.00 | Penguin | (1146.04) <Penguin> plut0: Did you ensure that any fancy SIP handling stuff is disabled on the router? |
| 18:37.47 | plut0 | Penguin: what fancy sip handling are you referring to? i'm using iptables for my router |
| 18:38.27 | Penguin | If I set the port numbers on the sip conntrack module in my router, it breaks audio. Make sure you are not doing that. |
| 18:39.10 | Penguin | I guess setting those things are actually creating ALG crap, which we do not ever use with asterisk. |
| 18:41.24 | plut0 | Penguin: can you show me documentation for setting up nat properly with asterisk? |
| 18:41.52 | Penguin | Your NAT rules that I saw looked okay. Your forwarding rules looked okay. |
| 18:41.58 | Penguin | My only concern is the conntrack modules. |
| 18:42.34 | Penguin | Oh, I did think of one other thing. |
| 18:42.45 | plut0 | conntrack hmm |
| 18:43.09 | Penguin | You were forwarding ports 18000-20000. Is that the port range you set in asterisk's rtp.conf? |
| 18:45.16 | plut0 | Penguin: yes |
| 18:46.32 | plut0 | Penguin: i checked, nf_conntrack_sip module is not loaded on my router |
| 18:46.37 | plut0 | Penguin: should i load it? |
| 18:47.10 | Penguin | I probably wouldn't. |
| 18:47.50 | plut0 | what about the sip nat configuration for asterisk? |
| 18:48.30 | Penguin | You have to configure it. You need to set the localnet, externaddr or externhost, directmedia=no, and nat=yes. |
| 18:53.22 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 18:53.36 | plut0 | Penguin: i didn't have directmedia=no set so let me try that again |
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| 18:58.53 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-opjufijvmtoojpzn) |
| 19:00.38 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-mgplocikbsfpzeah) |
| 19:01.43 | plut0 | Penguin: |
| 19:01.48 | plut0 | Penguin: i tried again, no luck |
| 19:02.11 | plut0 | Penguin: looks like its still using the local address |
| 19:03.23 | Penguin | For jabber? file said that was okay. I thought your problem was getting the audio to/from your phone. |
| 19:03.47 | *** join/#asterisk ickmund (uid11004@gateway/web/irccloud.com/x-rtvxxxlcxjxzumum) |
| 19:03.52 | Penguin | Is your phone on the same LAN as asterisk? |
| 19:03.54 | plut0 | one sec i see a problem |
| 19:03.58 | plut0 | yes |
| 19:07.04 | plut0 | ok |
| 19:07.11 | plut0 | logs show the public address now |
| 19:07.16 | plut0 | but still not working |
| 19:07.43 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-pgncfcyhiyjrjzsx) |
| 19:13.49 | plut0 | here is the new log, http://pastebin.com/59b3vgDV |
| 19:14.25 | plut0 | Penguin: ideas? |
| 19:14.55 | *** join/#asterisk X-Rob_ (sid14615@gateway/web/irccloud.com/x-lydmfugmwqlwadyk) |
| 19:16.26 | plut0 | or anyone else? |
| 19:17.12 | Penguin | I may be missing the big picture. |
| 19:17.45 | Penguin | Are you using a SIP phone, calling to/from your google voice account? |
| 19:17.50 | plut0 | Penguin: issue is when receiving a call. i answer the phone and the call doesn't go through |
| 19:18.17 | plut0 | Penguin: its an analog phone connected to a pap2t device connected to asterisk |
| 19:18.25 | Penguin | Did you tell asterisk to answer and accept the call before sending it to the phone? |
| 19:18.41 | plut0 | Penguin: let me check |
| 19:20.08 | plut0 | Penguin: no, doesn't look like it |
| 19:20.11 | plut0 | Penguin: should i? |
| 19:20.34 | Penguin | You must. |
| 19:20.38 | plut0 | ok |
| 19:20.56 | Penguin | Call screening is enabled, so you have to accept the call by sending a 1 to it. |
| 19:21.08 | plut0 | yeah |
| 19:23.31 | plut0 | Penguin: asterisk answered but still didn't work |
| 19:24.11 | *** join/#asterisk vlad_starkov (~vlad_star@171.25.189.14) |
| 19:24.40 | Penguin | The next thing I would do in troubleshooting this is to eliminate one leg of the call. I would send the inbound call directly to the echo test. |
| 19:24.58 | Penguin | If the echo test works when you call your google voice number, you can stop concentrating on that side of the call. |
| 19:25.07 | *** join/#asterisk vlad_sta_ (~vlad_star@87.117.139.75) |
| 19:25.11 | plut0 | Penguin: new log, http://pastebin.com/rcircm20 |
| 19:26.20 | plut0 | Penguin: does the pap2t device need to nated also or is that handled by asterisk? |
| 19:26.45 | Penguin | If the pap2t is on the same LAN as asterisk, there is no nat involved. |
| 19:27.22 | plut0 | ok |
| 19:27.36 | plut0 | Penguin: how do i do the echo test? |
| 19:28.03 | Penguin | Answer(), Echo() |
| 19:28.10 | plut0 | Penguin: and what does echo do? |
| 19:28.36 | Penguin | It listens to the audio and echoes it back "just as soon as it is received." |
| 19:28.42 | plut0 | hmm |
| 19:28.55 | Penguin | If you can hear yourself talk, you know that side of the call is working correctly. |
| 19:32.01 | plut0 | Penguin: hmm interesting |
| 19:32.20 | plut0 | Penguin: asterisk shows it answered and echoed but it kept ringing on the other end |
| 19:32.46 | Penguin | Then you didn't answer and properly accept the call by sending the 1 like I mentioned. |
| 19:33.02 | plut0 | Penguin: i turned off call screening on google voice |
| 19:33.10 | Penguin | You can't do that. |
| 19:33.16 | Penguin | It is on, no matter what. |
| 19:33.23 | plut0 | really? why? |
| 19:33.35 | Penguin | You'd have to ask Google. |
| 19:33.46 | plut0 | weird, so use Send(1) ? |
| 19:33.50 | Penguin | No. |
| 19:34.46 | plut0 | SendDTMF(1) ? |
| 19:35.05 | Penguin | Yes. But wait a few seconds after answring before you send it. |
| 19:35.09 | plut0 | ok |
| 19:35.18 | Penguin | Here's what I do: |
| 19:35.20 | Penguin | same => n,Playback(silence/1&hello&silence/6); |
| 19:35.20 | Penguin | same => n,SendDTMF(1); |
| 19:36.34 | plut0 | Penguin: ok so that worked, i can hear myself |
| 19:36.50 | Penguin | Good. Now change the echo test to the Dial you were using earlier. |
| 19:38.23 | plut0 | Penguin: doesn't work. |
| 19:39.31 | plut0 | Penguin: where is the problem now? |
| 19:39.31 | Penguin | But now you can stop concentrating on the gv side of the call, since you know the call makes it to asterisk and audio is two-way. |
| 19:39.44 | plut0 | yeah |
| 19:39.53 | plut0 | Penguin: did i mention i can make outgoing calls? |
| 19:39.57 | Penguin | So now it is on the ATA side. |
| 19:40.05 | Penguin | I don't know if you did or not. |
| 19:40.15 | plut0 | just the incoming isn't working |
| 19:40.26 | Penguin | That's a NAT problem. |
| 19:40.39 | plut0 | ok |
| 19:40.46 | Penguin | When it happened to me, I was because of those conntract modules. |
| 19:40.59 | plut0 | Penguin: yeah i don't have conntrack loaded for sip though |
| 19:41.10 | Penguin | You said you didn't have them loaded though. lsmod|grep _sip |
| 19:41.27 | plut0 | yeah nothing... |
| 19:42.05 | Penguin | Did you correctly set the localnet? Did you correctly set externhost or externaddr? How about directmedia and nat? |
| 19:42.17 | plut0 | Penguin: externhost is set |
| 19:42.29 | Penguin | You're using dynamic DNS? |
| 19:42.30 | plut0 | Penguin: localnet is set |
| 19:42.41 | plut0 | Penguin: sort of |
| 19:43.09 | Penguin | If you don't have a host name that correctly resolves to the public IP address of your asterisk network, externhost is not going to help. |
| 19:43.22 | *** join/#asterisk n0taBug (~quassel@CPE002401e92f95-CM185933426b1c.cpe.net.cable.rogers.com) |
| 19:43.28 | plut0 | Penguin: it is resolving correctly because the logs show my public ip address |
| 19:43.30 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:43.49 | plut0 | Contact: <sip:asterisk@108.44.40.125:5060> |
| 19:43.51 | Penguin | Check it manually on the asterisk box. |
| 19:44.05 | plut0 | Penguin: how? |
| 19:44.13 | Penguin | From the asterisk CLI: !ping -c1 your.host.name |
| 19:45.01 | plut0 | Penguin: yup it works |
| 19:45.21 | Penguin | Okay, if you are accurate, we'll mark that off the list. |
| 19:45.34 | Penguin | What values did you set for localnet? |
| 19:45.49 | plut0 | localnet=192.168.0.0/255.255.255.0 |
| 19:47.01 | Penguin | You've set nat=yes and directmedia=no in the general section? You set nat=no in the phone's section? |
| 19:47.26 | plut0 | Penguin: nat=yes in both general and phone, directmedia set in the phone section |
| 19:47.34 | Penguin | Stop doing that. |
| 19:47.43 | n0taBug | Hi Room! I was wondering if anyone could help me out with an Asterisk Realtime issue. I have exhausted searching google to no avail. I would like to create realtime contexts with out having to create a static proxy context in extensions.conf, I have found thru google that there is an un-documented option called ovverrideswitch=yes for the [general] section of extensions.conf, the way I read it is this should |
| 19:47.43 | n0taBug | let me do what I am after. Has anyone here used this option? Or can any of you let me know how to create a context in the realtime db with out having to create a static proxy context in extensions.conf? I am using Asterisk v11. |
| 19:48.08 | plut0 | Penguin: phone section should have nat=no then? |
| 19:48.54 | Penguin | The phone is not behind a nat in relation to asterisk, so set nat=no and directmedia needs to be either no or nonat and set directmediadeny and directmediapermit. |
| 19:49.53 | plut0 | directmediapermit should be local net and directmediadeny should be 0.0.0.0/0 ? |
| 19:51.21 | Penguin | I like to allow directmedia between phones on the same LAN, but always deny directmedia for any call that has to pass the nat. I would use directmediadeny=0.0.0.0/0 and directmediapermit=192.168.0.0/24 |
| 19:51.53 | plut0 | ok thats all set now |
| 19:52.09 | Penguin | Or use directmedia=nonat to try to guess at it. |
| 19:52.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:53.31 | plut0 | Penguin: hmm still not working |
| 19:53.49 | Penguin | Be sure you "sip reload" after saving the changes. |
| 19:54.03 | plut0 | Penguin: i did a whole restart |
| 19:55.49 | plut0 | hmm |
| 19:55.55 | plut0 | it stopped nat'ing now |
| 19:56.11 | plut0 | its using the internal ip again |
| 19:56.20 | Penguin | file said that is not a problem. |
| 19:56.47 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 19:56.57 | Penguin | If the echo test works, forget what IP address is showing. Test the echo again to make sure it works with the priv IP address, then move on. |
| 19:57.08 | file | I saw a Google Voice XMPP transport, that part is fine to have a private IP in |
| 19:58.43 | plut0 | good idea, let me try echo test again |
| 19:59.29 | plut0 | ok echo test works |
| 19:59.59 | n0taBug | can any of you guys here help point me in the right direction regarding my question above? I am trying to create submenu's on the fly dynamically without having to dial plan reload asterisk each time.....anyone? |
| 20:00.56 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 20:00.58 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:06.47 | *** join/#asterisk BlackDex_ (~BlackDex@ori.vyus.nl) |
| 20:07.17 | *** join/#asterisk mirela666 (~mirko.bra@95.180.109.92) |
| 20:08.10 | ChannelZ-Wk | This sounds like you should just write an AGI for this dynamic menu system rather than trying to build and destroy dialplan contexts and extensions on the fly |
| 20:09.07 | n0taBug | how would you handle using an AGI for the sub menu's though? |
| 20:09.28 | *** join/#asterisk apb1963 (~quassel@174.135.116.22) |
| 20:10.05 | ChannelZ-Wk | That would be up to the logic of your AGI. Use Read to wait for people to type things, then react accordingly. |
| 20:10.34 | *** join/#asterisk krapper (~krapper@23.21.191.223) |
| 20:10.55 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 20:13.04 | n0taBug | I hadent actually thought about using an AGI, I may end up having to look further into that, however is there no way to add a context to asterisk realtime without doing a 'reload'? It seems to defeat the purpose of using the database for this if you have to create a static proxy context in extensions.conf and reload each time you do it......do you know anything about the overrideswitch=yes option for extensions. |
| 20:13.05 | n0taBug | conf? I read the bug tracker and this was merged into versions after 1.6 to address this matter to the best of my reading anyway.....it just seems there is absolutly no documentation on how to use it... |
| 20:14.30 | plut0 | Penguin: i got it to work if i pass the dmtf (1) through asterisk, otherwise i don't hear anything and pressing 1 from the phone doesn't work, ideas? |
| 20:14.59 | Penguin | plut0: I told you that you have to make asterisk answer and send 1. What part of that are you having trouble with? |
| 20:15.19 | plut0 | Penguin: so you can't do that from the analog phone? |
| 20:15.34 | Penguin | I specifically said to take out the echo test and replace it with the Dial. I never said take out the answer or the senddtmf. |
| 20:15.46 | plut0 | i see |
| 20:15.53 | plut0 | well, if thats what it is then, i have it working now |
| 20:16.34 | Penguin | You accept the call on asterisk and then pass it along to a phone. |
| 20:17.08 | Penguin | Accepting the call means you have to "press 1" by sending dtmf with SendDTMF(). |
| 20:18.14 | plut0 | Penguin: well its working now, thank you for all the help! |
| 20:26.11 | *** join/#asterisk dxd828 (~dxd828@host-92-24-127-29.ppp.as43234.net) |
| 20:30.46 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:33.02 | *** join/#asterisk dar123 (~DAR@216.131.72.59) |
| 20:33.54 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 20:36.35 | dar123 | i installed asterisk 12 on centos, once i start the service i get "Asterisk died with code 127. |
| 20:36.36 | dar123 | cat: /var/run/asterisk/asterisk.pid: No such file or directory" |
| 20:36.49 | dar123 | Anyone faced similar issue? |
| 20:37.18 | [TK]D-Fender | Start it NOT as a service an ssee what you get |
| 20:37.24 | [TK]D-Fender | asterisk -gvvvvvvvvvvvvvvvvvvc |
| 20:38.12 | n0taBug | hrm...anyone else know the answer to my question? |
| 20:38.33 | [TK]D-Fender | n0taBug: No, that is how it works. |
| 20:39.17 | Penguin | No place for realtime extensions here? |
| 20:39.32 | dar123 | error while loading shared libraries: libjansson.so.4 |
| 20:40.11 | n0taBug | [TK]D-Fender: do you know what the purpose of the overrideswitch=yes option in extensions.conf is for? From what i've read online its supposed to allow for what I am trying to do, unless I totally mis-understand this, is posting a link to the asterisk forums allowed in here? If so I can show you what i've read |
| 20:41.30 | [TK]D-Fender | n0taBug: I'm not seeing any reference to that.... |
| 20:41.50 | n0taBug | [TK]D-Fender: is posting a link allowed in the channel? |
| 20:41.56 | [TK]D-Fender | absolutely |
| 20:42.14 | n0taBug | [TK]D-Fender: http://forums.asterisk.org/viewtopic.php?p=160660 |
| 20:43.01 | n0taBug | When I use that option, i get the exact same error as the poster of that forum got No such switch 'yes' - the option does impact asterisk but I can find no documentation for it |
| 20:43.26 | [TK]D-Fender | n0taBug: Doesn't appear to confirm that it actually exists... |
| 20:43.40 | boom^time | Does anyone know of a good place to find an up-to-date list of originateresponse failure codes with their description? |
| 20:43.47 | [TK]D-Fender | n0taBug: This was 4 years ago and how knows if it ever got merged? |
| 20:43.51 | boom^time | the reason codes that is |
| 20:45.02 | n0taBug | [TK]D-Fender: I read that it did get merged and i can see that it is infact affecting v11 - when I use that option I get the warning 'No such switch yes' - so i've been on a mission to understand how to use it but I cant find documentation on it, on that link I sent you 'malcolmd' said the patch was merged but not documented. |
| 20:51.04 | *** join/#asterisk dar123 (~DAR@2.88.59.198) |
| 20:56.56 | n0taBug | [TK]D-Fender: do you think this would be a better question suited to the -dev channel? I know its not l2 support but as it appears to be an undocumented option not sure how else I can get the info I need on how to use this or if it even works at all for my purpose...as you say its a 4 year old issue, but it looks like it was merged and is active in versions after 1.6....thoughts? |
| 20:58.08 | *** join/#asterisk bulkorok (~Benjamin@053d9311.dynamic.tele-ag.de) |
| 20:59.09 | [TK]D-Fender | n0taBug: Its not documented, and dubious existence... so yes, I'd say it's "-dev" material |
| 20:59.32 | n0taBug | [TK]D-Fender: thanks i'll try over there |
| 21:05.21 | anonymouz666 | hehe reading the history of the mayor of toronto... remember that existed something here with the bot like "put down the crack pipe" |
| 21:05.23 | anonymouz666 | lol. |
| 21:05.51 | *** join/#asterisk vlad_starkov (~vlad_star@171.25.189.14) |
| 21:07.37 | *** join/#asterisk oscalation (~Family@unaffiliated/oscalation) |
| 21:07.56 | oscalation | can someone help me understand, or link me to an explaination of what a dialplan is ? |
| 21:08.22 | anonymouz666 | !book |
| 21:08.26 | anonymouz666 | ~book |
| 21:08.26 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:08.34 | anonymouz666 | here it goes |
| 21:09.34 | oscalation | does this sound right? |
| 21:09.47 | oscalation | "The dialplan is the heart of Asterisk. All channels that arrive in the system will be passed through the dialplan" |
| 21:09.47 | [TK]D-Fender | oscalation: "Dialplan" (as relates to Asterisk) is the term for the configuration that determines how Asterisk processes calls. ie : all of the instruction s on what to do, step-by-step. |
| 21:10.06 | [TK]D-Fender | oscalation: It is all the programming steps the call will go through. |
| 21:10.06 | oscalation | [TK]D-Fender: awesome, thanks anonymouz666 |
| 21:13.10 | *** part/#asterisk bulkorok (~Benjamin@053d9311.dynamic.tele-ag.de) |
| 21:38.48 | *** join/#asterisk markit (~marco@88-149-177-66.v4.ngi.it) |
| 21:39.59 | *** join/#asterisk endemic (~endemic@lynx.onvox.net) |
| 21:40.00 | markit | hi, does anybody know if Digium "Switchbox" platform is a sort of "closed" platform (i.e. you can't access the AMI, or add as many extensions as you want, etc) or it's asterisk core is open? |
| 21:40.42 | markit | sorry, Switchvox |
| 21:43.01 | pabelanger | markit, http://developers.digium.com/switchvox/ |
| 21:45.17 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 21:46.02 | markit | pabelanger: ehm, the answer is "dig all that site"? :) |
| 21:46.42 | markit | I don't want to develop "upon" switchvox, just be able to use asterisk features in addition to switchvox |
| 21:46.55 | pabelanger | right |
| 21:47.02 | pabelanger | not many people will know |
| 21:47.05 | pabelanger | this is #asterisk |
| 21:47.12 | pabelanger | you'll likely need to talk to digium |
| 21:47.17 | pabelanger | or read documentation |
| 21:47.53 | markit | well, I thought that many people here would have had some knowledge of digium stuff |
| 21:48.07 | pabelanger | Well, I used to work at digium |
| 21:48.12 | pabelanger | but didn't touch switchvox |
| 21:48.17 | pabelanger | different team then asterisk |
| 21:48.27 | markit | I see, thanks a lot |
| 21:48.31 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:48.33 | pabelanger | either way, call digium support / sales, they should give you a quick answer |
| 21:48.36 | pabelanger | they are friendly |
| 21:48.51 | markit | I don't speak english fluently enough, I'll try by email |
| 21:49.01 | rrittgarn | i was a switchvox reseller for a while.. as i recall you will have to use their API for doing stuff, but direct asterisk manipulation is limited / restricted |
| 21:49.05 | markit | but chatting with someone would have been better |
| 21:49.24 | markit | rrittgarn: ah, I see :( |
| 21:49.26 | rrittgarn | depending on what the exact need is they may be able to address it with their API |
| 21:49.26 | pabelanger | Sounds about right |
| 21:49.46 | markit | rrittgarn: but their api is something locked at multiple level? |
| 21:49.56 | markit | I mean, if you want to do A |
| 21:50.16 | markit | you need API extension X and costs 3.000 dollars? |
| 21:50.46 | rrittgarn | I think they had some custom stuff for that type of event where you could store account codes with the CDR like regular asterisk |
| 21:52.09 | *** join/#asterisk hugobulnes1 (~Thunderbi@189.215.47.77.cable.dyn.cableonline.com.mx) |
| 21:55.22 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:03.18 | *** join/#asterisk moy (~moy@unvlon55-1176057127.sdsl.bell.ca) |
| 22:17.59 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 22:21.02 | ipengineer | file: Are you around? Having trouble getting PJSIP to register to a trunk.. I am getting a 401 Unauthorized response. The credentials work from my soft phone so Im sure it has something to do with my PJSIP config. Does anything look out of wack here? http://paste.laravel.com/15dW |
| 22:27.08 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:33.54 | cusco | hey folks |
| 22:34.06 | cusco | cheap working hardphones |
| 22:34.10 | cusco | any suggestions? |
| 22:34.18 | Qwell | ~cheap |
| 22:34.18 | infobot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:34.24 | Penguin | What's your budget for cheap? |
| 22:34.44 | cusco | less than 50EUR |
| 22:34.48 | cusco | per hard phone |
| 22:35.00 | Penguin | infobot: convert 50 EUR to USD |
| 22:35.27 | cusco | lol |
| 22:35.33 | cusco | found http://www.onedirect.pt/produtos/grandstream/grandstream-gxp1100 |
| 22:35.40 | cusco | err (yes its pt, sorry) |
| 22:35.43 | [TK]D-Fender | ~gs |
| 22:35.43 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:35.52 | cusco | m |
| 22:35.52 | [TK]D-Fender | ~grandstream |
| 22:35.52 | infobot | somebody said grandstream was the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 22:35.56 | Penguin | You can get Polycom, Cisco/Linksys, GrandStream, maybe a used Aastra, probably used Yealink, etc. |
| 22:39.47 | rrittgarn | from what i've heard Aastra is coming out with a phone cheaper than the 6731 come January... |
| 22:40.03 | rrittgarn | the 6731 is a good cheap phone (in my opinion of cheap) |
| 22:41.17 | Penguin | Polycom 321, perhaps? |
| 22:41.46 | Penguin | They're like $50 USD. |
| 22:42.11 | cusco | ok thanks |
| 22:43.45 | Penguin | Actually, there's the 450 for around the same price. |
| 22:44.06 | Penguin | 335, too. |
| 22:44.20 | file | ipengineer, did you get it working? nothing looked out of the ordinary and my flowroute registers fine |
| 22:47.12 | ipengineer | file: No never got it going.. Wondering if it has something to do with me being NAT'd... This is what I am getting.. "Temporal response '401' received from 'sip:12345@sip.flowroute.com'" |
| 22:47.27 | file | I'm behind NAT and it's fine |
| 22:47.38 | ipengineer | my config looks just like yours? |
| 22:47.39 | Penguin | I'm behind NAT and I can use flowroute. |
| 22:47.56 | file | I'm not specifying a transport explicitly and I'm using my own user/pass |
| 22:48.01 | Penguin | I just have to configure asterisk correctly for being behind nat. |
| 22:48.41 | *** join/#asterisk onizo (~onizo@cpe-98-155-117-134.san.res.rr.com) |
| 22:48.42 | Penguin | And make sure there is no SIP ALG on the router. |
| 22:48.49 | file | this amuses me: [Nov 6 15:29:04] NOTICE[41946]: res_pjsip/pjsip_distributor.c:241 log_unidentified_request: Request from '<sip:ping@invalid>' failed for '216.115.69.144:5060' (callid: 8230e4e2-5cac4215-7011508@216.115.69.131) - No matching endpoint found |
| 22:49.18 | ipengineer | Yea.. There is no ALG I have a ton of other SIP devices working across it as well.. |
| 22:49.59 | ipengineer | I wonder if it has something to do with me having endpoint set to realtime in sorcery and this is in my pjsip.conf file? I tried by putting "endpoint=config,pjsip.conf" one line above realtime though |
| 22:50.30 | file | should work, haven't tested it myself |
| 22:50.55 | ipengineer | K |
| 22:51.08 | file | well actually that wouldn't work... |
| 22:52.15 | file | outbound registration has nothing to do with an endpoint so that doesn't matter |
| 22:52.22 | file | did you map anything else to realtime? |
| 22:52.55 | ipengineer | Yes. endpoint, auth, air, domain_alias, identify |
| 22:53.17 | ipengineer | That would make sense.. Its picking up registration from the file but the rest is set to realtime so it doesn't see it |
| 22:53.33 | file | those are fine |
| 22:54.01 | file | oh, did you specify auth in pjsip.conf for the trunk? |
| 22:54.06 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 22:54.06 | ipengineer | Yes |
| 22:54.11 | ipengineer | outbound_auth=mytrunk |
| 22:54.19 | file | did you specify in the sorcery.conf to also allow pjsip.conf for that? |
| 22:54.48 | ipengineer | No, that is what I am doing now. Given the way I was doing it is correct |
| 22:55.26 | file | auth=config,pjsip.conf,criteria=type=auth |
| 22:56.57 | file | I don't think we have a log message for that situation... if auth is specified but it's not found |
| 22:58.02 | file | ah there is |
| 22:58.13 | ipengineer | Yea I think that makes sense as to the problem.. I am not seeing that registration error anymore. Going to test everything real quick.. |
| 22:58.45 | file | flowroute has a list in the interconnection tab too which shows registered contacts |
| 22:59.26 | ipengineer | Yea so we are registered.. According to that. Now to fix the dial plan |
| 23:00.05 | file | I like it when stuff I write does what I want it to |
| 23:01.16 | ipengineer | file: Yep thanks I went ahead and applied that to the rest of the realtime configs. |
| 23:01.33 | file | yeah just change the type=<value> accordingly |
| 23:01.43 | file | it tells the sorcery configuration file wizard to filter |
| 23:02.06 | file | or else it may throw an object consistency fit when it tries to treat an aor as an endpoint and etc |
| 23:02.34 | ipengineer | file: Are you registering for inbound calls as well or just outbound? |
| 23:03.50 | ipengineer | file: Makes sense now.. Went ahead and added them all with appropriate filters so hopefully won't have to revisit it for a while. |
| 23:03.58 | file | inbound |
| 23:04.20 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
| 23:04.24 | file | registration isn't used for outbound, they auth |
| 23:05.16 | Penguin | I think they require you to be registered before you can send calls outbound to them. I know other ITSPs do. |
| 23:05.24 | ipengineer | Based on IP? I was thinking inbound worked without AUTH and then outbound required registration.. Or at least thats how I am setup in another scenario |
| 23:05.57 | Penguin | Registration is used to be able to receive calls inbound to your asterisk, and they do not auth those calls to you. |
| 23:06.17 | Penguin | But some ITSPs require you to be registered before they will accept your calls and auth on outbound calls. |
| 23:06.49 | Penguin | It's just a security mechanism they use. Registration isn't really related to outgoing calls. |
| 23:06.59 | ipengineer | Ok I will change my settings in flow route then.. They are set to send to host now instead of "SIP Registration" |
| 23:07.24 | Penguin | If you are not using SIP registration, I don't know how they deal with the outgoing calls. |
| 23:07.44 | ipengineer | Well.. I say that. When I had it set to host I would see the invite coming into asterisk but being declined by me as unauthorized |
| 23:07.44 | *** join/#asterisk Jamuel (~Adium@70-90-173-246-California.hfc.comcastbusiness.net) |
| 23:07.44 | Penguin | Perhaps they disable that "register before dialing out" mechanism. |
| 23:08.24 | Penguin | Don't require them to auth against your asterisk on incoming calls. They won't do it. |
| 23:08.48 | ipengineer | Right.. I just need to allow calls based on sip.flowroute.com correct? |
| 23:09.37 | Penguin | Pretty much. I use the host line to set the ITSP's SIP address and I use remotesecret so that I send auth to them but do not require them to auth against the peer entry. |
| 23:09.53 | file | basically an identify with flowroute's IP address pointing to an inbound endpoint |
| 23:10.00 | file | (speaking chan_pjsip lingo) |
| 23:10.19 | ipengineer | Ok.. |
| 23:10.38 | Penguin | If you set secret, the incoming calls are going to require auth as well. I'm talking chan_sip, since I use 1.8. |
| 23:11.25 | Penguin | I hope I don't have to learn a new sip.conf format when I upgrade. :/ |
| 23:11.38 | mjordan | heh |
| 23:11.56 | mjordan | You will not have to learn a new sip.conf schema. |
| 23:12.07 | mjordan | You will have to learn a new pjsip.conf schema, if you want to use chan_pjsip. |
| 23:12.23 | file | or use the conversion script and see if it works |
| 23:12.34 | Penguin | If chan_sip is still there, I probably won't have any reason to change to pjsip. |
| 23:12.39 | mjordan | well, ostensibly, you'd still want to understand what it spits out |
| 23:12.42 | mjordan | fair enough |
| 23:12.43 | ipengineer | SO if I take a step back I have a [mytrunk] with type=registration and a [mytrunk] with type=endpoint. Do I need both or just the one with registration |
| 23:12.46 | file | and on that bombshell I am going to Costco where I aim to spend only a few dollars but will end up spending much more because that's what Costco does |
| 23:12.53 | mjordan | file: Costco rules |
| 23:13.10 | Penguin | My *only* reason to use pjsip would be so I can help support others who use pjsip. |
| 23:13.15 | mjordan | ipengineer: off the cuff, you'll need both |
| 23:13.59 | *** join/#asterisk Sjors (~sgielen@foo.kassala.de) |
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| 23:18.30 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-aeysmiscnxfvofmq) |
| 23:26.29 | ipengineer | So in trying to send an outbound call I get no response.. So if I am clear the [mytrunk] type=registration is for inbound calls correct? Because I know definitively that I do not have to auth with them to send them calls.. I am sending calls now based on IP address from opensips. |
| 23:27.47 | phix | I can register fine to my SIP Provider, can make calls but cannot receive calls. Do I need to open / forward any ports on my router? |
| 23:29.30 | cusco | tried register => in sip.conf ? |
| 23:31.02 | Penguin | If you are already registered, they know how to reach you... if the call never makes it to you, consider the ports are not available. Yes, forward the ports. |
| 23:34.03 | phix | forward what ports? |
| 23:34.15 | phix | 5060? |
| 23:34.45 | Penguin | UDP 5060 (the SIP port) and UDP 10000-20000 (the RTP ports; or as configured in rtp.conf) |
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