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| 01:07.12 | slav3_kitten | soo how do i force a channel to close via cli? |
| 01:07.52 | WIMPy | channel request hangup |
| 01:09.28 | p3nguin | Depends on what you mean by force, and it depends on why you want to close it. |
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| 01:13.46 | n8ideas | Anyone ever seen an issue with "ghost" calls when using a TLS trunk? |
| 01:14.05 | n8ideas | disconnect network cable, make call, wait for all timeouts to finish and all dialogs to destroy |
| 01:14.09 | *** part/#asterisk deo (~deo@58.71.19.178) |
| 01:14.16 | n8ideas | then reconnect cable, and see the call go through the far end? |
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| 01:35.13 | twitchnln | greetings everyone, I was wondering if someone might have an extensions.conf snippet that rewrites the called number with another? |
| 01:35.39 | p3nguin | Goto() |
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| 01:53.47 | slav3_kitten | thanks guys :) |
| 01:54.03 | slav3_kitten | i apparently left it up |
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| 03:03.55 | FireAndIce | Hi everyone!! |
| 03:04.17 | FireAndIce | How do I connect a regular phone line to asterisk? |
| 03:05.41 | jpsharp | Either via a Digium card or one of the handful of ATAs that have an FXO port on them. |
| 03:06.38 | jpsharp | The ATA route is going to be cheaper. |
| 03:07.27 | WIMPy | I'd ask for a definition of "regular phone line". |
| 03:08.13 | FireAndIce | WIMPy, I want to connect my phone, via my carrier, to asterisk server. |
| 03:08.39 | jpsharp | The phone line that comes out of the wall socket? |
| 03:09.00 | jpsharp | Just making sure. |
| 03:09.03 | WIMPy | That does not answer the question. |
| 03:09.59 | FireAndIce | Suppose, the user dials 1-800-****, it should connect to my asterisk server which in return will give an automated answer. |
| 03:11.22 | WIMPy | That still doesn't answer the first question, but let's try the next: Why do you want a phone line for that? |
| 03:11.46 | jpsharp | That doesn't answer the question of "regular phone line". Do you mean an actual RJ-11 phone line that would plug into a phone? Or are you just talking about routing a number to Asterisk via a random means? |
| 03:17.56 | FireAndIce | WIMPy, what I'm trying is first, the user will dial his phone number on his android phone. This number will be sent to tomcat server. Then an SMS will be sent with access code back to the user for authentication purpose. In case, the SMS is not delivered to the user, due to DND, I want to add a feature so that the user can dial a number and get his access code, in order to register. |
| 03:18.14 | FireAndIce | I'm new to asterisk. |
| 03:18.58 | WIMPy | That answers neither question, but good to know anyway. |
| 03:19.31 | jpsharp | Sounds like you need a VOIP DID that has SMS support. |
| 03:30.01 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 03:30.04 | lorsungcu | oy |
| 03:30.07 | lorsungcu | anyone up? |
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| 03:30.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 03:30.23 | jpsharp | Nyet :) |
| 03:30.30 | WIMPy | ~ask |
| 03:30.30 | 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. |
| 03:30.35 | lorsungcu | having some nasty trouble with an e&m circuit |
| 03:31.15 | lorsungcu | calls go out fine ~80% of the time, otherwise they hang up almost immediately |
| 03:34.43 | slav3_kitten | fucking flu *shivers* |
| 03:35.17 | WIMPy | That's rather vague. How's that line connected to your asterisk? Has it worked before? |
| 03:35.28 | jpsharp | lorsungcu: Any constant in the equations? Same circuit? Same channel? |
| 03:36.12 | lorsungcu | new install, all calls are leaving the same circuit. |
| 03:36.25 | FireAndIce | jpsharp, WIMPy Question changed. How do I connect a phone call through GSM to asterisk server. Please bear with me. |
| 03:36.38 | lorsungcu | the customer is complaining about it now, so i am assuming it worked prior to this |
| 03:37.03 | WIMPy | FireAndIce: The easiest way it via an ITSP. |
| 03:37.05 | jpsharp | FireAndIce: Ah, you want an Asterisk compatible GSM card, then. |
| 03:37.13 | WIMPy | ~itsp |
| 03:37.13 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 03:37.24 | _Corey_ | lorsungcu: You *sure* it's an E&M circuit? I've seen some where the customer and carrier weren't on the same page... :) |
| 03:37.35 | lorsungcu | _Corey_: yeah positive |
| 03:37.42 | jpsharp | E&M tie line to a PBX? |
| 03:37.42 | slav3_kitten | ~itsplist-us |
| 03:37.43 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 03:37.56 | lorsungcu | its a bullshit flex circuit |
| 03:37.56 | slav3_kitten | sweet the two i use are there :D |
| 03:38.11 | _Corey_ | a flex circuit as E&M? |
| 03:38.13 | lorsungcu | so CO => SIP => cisco router => E&M => PBX |
| 03:38.23 | _Corey_ | I'd have it changed over to a PRI |
| 03:38.26 | lorsungcu | right |
| 03:38.31 | slav3_kitten | googles E&M |
| 03:38.32 | lorsungcu | they want another 3 years to do that |
| 03:38.35 | _Corey_ | It was probably configured as E&M to support whatever was there before |
| 03:38.40 | lorsungcu | yes |
| 03:38.46 | carrar | cut out the cisco and T1 |
| 03:38.47 | lorsungcu | customer does not want to sign another contract |
| 03:38.54 | lorsungcu | and the telco won't let me do sip |
| 03:39.29 | carrar | get a new telco |
| 03:39.36 | lorsungcu | in 6mo, yes |
| 03:39.50 | slav3_kitten | oh nifty |
| 03:39.56 | jpsharp | Something is dropping the call, obviously. What does the Cisco say when the call drops? |
| 03:40.13 | FireAndIce | WIMPy, jpsharp thanks a lot, I'll check it out. |
| 03:40.15 | lorsungcu | I only have some debugs, but I can paste bin it... |
| 03:40.16 | lorsungcu | hold on |
| 03:40.29 | carrar | You own the cisco? |
| 03:40.46 | jpsharp | Disclaimer: Been years since I've touched E&M. |
| 03:40.49 | _Corey_ | lorsungcu: The problem with E&M is you're not going to get much useful debug |
| 03:41.43 | lorsungcu | i know |
| 03:41.50 | lorsungcu | i do not own the cisco, unfortunately |
| 03:42.08 | lorsungcu | http://pastebin.com/SGs6zNBH |
| 03:42.09 | jpsharp | The Cisco owners are the ones who want to contract you for 3 years? |
| 03:42.16 | lorsungcu | correct, that'd be the telco |
| 03:42.24 | lorsungcu | XO, in case you never want to go near them |
| 03:42.39 | jpsharp | I've never seen that kind of BS from XO. |
| 03:42.53 | lorsungcu | we see it all the time here |
| 03:43.01 | lorsungcu | they are awful |
| 03:43.19 | jpsharp | So they want to contract you for another 3 years for changing a few lines in the Cisco? Fracking awesome. |
| 03:43.34 | lorsungcu | right |
| 03:43.45 | lorsungcu | the amount of time I've had their guys on the phone |
| 03:43.50 | lorsungcu | it could have been done 10x over |
| 03:43.58 | _Corey_ | lorsungcu: I can see them being painful about going to SIP but switching it to NI2 shouldn't be too bad |
| 03:44.19 | lorsungcu | yeah, i really can't say what their problem is |
| 03:44.24 | lorsungcu | but they won't budge |
| 03:44.28 | _Corey_ | wow |
| 03:44.38 | lorsungcu | i've tried :/ |
| 03:44.41 | carrar | and you have 6 months left with them? |
| 03:44.44 | lorsungcu | right |
| 03:44.46 | lorsungcu | its up in like |
| 03:44.55 | carrar | let them know you are shopping around |
| 03:44.57 | lorsungcu | i don't know, soon, anyway.,. |
| 03:45.08 | _Corey_ | this is a customer install, right? |
| 03:45.17 | lorsungcu | yeah, it was in place when we got there, though. |
| 03:45.19 | carrar | let them know in 5 months that is |
| 03:45.23 | lorsungcu | with old vodavi |
| 03:45.29 | carrar | after you've already picked a new provider |
| 03:45.56 | jpsharp | Do you have debug logs from the Asterisk side of the world? |
| 03:46.05 | lorsungcu | sure, what do you want to see? |
| 03:46.17 | _Corey_ | lorsungcu: The only thing I noticed in your debug is that your clock seems to be way off |
| 03:46.22 | lorsungcu | yeah i saw that too |
| 03:46.29 | lorsungcu | the tech i was on with did not seem to care |
| 03:46.59 | lorsungcu | other symptoms are a ~4 second delay before any ringing |
| 03:47.23 | lorsungcu | you place a call, it sits on the cisco for that long before they see an invite get sent to their switch |
| 03:47.26 | _Corey_ | I'd sync it to an NTP source like pool.ntp.org before continuing |
| 03:47.31 | lorsungcu | i cant |
| 03:47.41 | lorsungcu | and the guy i was on with didn't give a shit |
| 03:48.10 | jpsharp | lorsungcu: Sounds like their dial peer settings are fucked up. |
| 03:48.10 | WIMPy | Have you considered hacking it? |
| 03:48.54 | lorsungcu | they just sent me the show voice port output |
| 03:48.57 | lorsungcu | hold on i will pb it |
| 03:49.21 | jpsharp | Have them send you the dial peer configurationt oo. |
| 03:49.43 | lorsungcu | alright, ill ask |
| 03:49.44 | lorsungcu | http://pastebin.com/M5zMT14w |
| 03:50.22 | _Corey_ | "dial peer configuration" ? on an E&M circuit? |
| 03:50.52 | _Corey_ | jpsharp: What are you talking about? |
| 03:51.15 | jpsharp | That matches dialed digits then shuffles the call off to the right destination. |
| 03:51.44 | lorsungcu | well |
| 03:51.47 | lorsungcu | i doubt its that |
| 03:51.57 | lorsungcu | as you can call the same number 10 times, and it'll work 6-8 times |
| 03:52.00 | jpsharp | No, but that will account for the post dial delay. |
| 03:52.04 | lorsungcu | ah |
| 03:52.41 | _Corey_ | Problems with E&M usually are mismatches on signaling, framing, coding, or timing... |
| 03:53.14 | lorsungcu | signalling/framing/coding all match |
| 03:53.25 | lorsungcu | this is the first I've seen of any timing, so I'm going to check that |
| 03:54.12 | _Corey_ | make sure you guys agree who's the master/slave |
| 03:54.19 | jpsharp | Yeah, what Corey said. |
| 03:56.14 | lorsungcu | does the wait wink duration seem extra long? |
| 03:57.13 | _Corey_ | I've NEVER changed any of those settings on an E&M before |
| 03:57.32 | lorsungcu | yeah, i've only done maybe 5 e&m and never had to |
| 04:05.27 | jpsharp | The only thing I see different is that the Cisco is sending a 200ms wink and Asterisk's default rxwink time is 300ms. |
| 04:05.43 | jpsharp | But I don't think that will cause random call drops. |
| 04:06.03 | _Corey_ | yeah, I think that's all call setup |
| 04:08.02 | SeRi | _Corey_: what is going on! :) |
| 04:08.15 | _Corey_ | hey... :) |
| 04:09.34 | jpsharp | lorsungcu: Turn on full dahdi (or zaptel) debugging and show a call trace. |
| 04:10.09 | *** part/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 04:14.09 | slav3_kitten | so anyone know if you can have a cisco 7960 point at the tftp server for services.xml, background.bmp, and directory.xml |
| 04:14.14 | slav3_kitten | or must it be a http:// |
| 04:15.24 | jpsharp | I'm pretty sure it's gotta be an http server. |
| 04:15.50 | slav3_kitten | crap |
| 04:16.07 | slav3_kitten | so, know of a simple lightweight http server for linux? |
| 04:16.23 | slav3_kitten | i'd rather not throw apache on it, seems uber overkill |
| 04:17.44 | jpsharp | TUX, the in-kernel server. |
| 04:17.54 | jpsharp | or kHTTPd. |
| 04:18.25 | slav3_kitten | thanks man :) |
| 04:19.50 | p3nguin | There's always lighttpd, too. |
| 04:20.03 | jpsharp | That's the other one I was thinking of. |
| 04:20.07 | p3nguin | But for something so small, I'd probably go another route. |
| 04:20.40 | p3nguin | such as mongoose. |
| 04:20.56 | p3nguin | or darkhttpd. |
| 04:21.55 | slav3_kitten | oh? |
| 04:22.02 | SeRi | ngix? |
| 04:22.24 | p3nguin | nginx would also be a possibility, but I wouldn't even bother. |
| 04:23.46 | SeRi | I only use light and apache |
| 04:24.27 | p3nguin | mongoose and darkhttpd require absolutely no configuration whatsoever. |
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| 04:24.37 | p3nguin | Single command, no config files. |
| 04:24.58 | slav3_kitten | oh that's got advantages |
| 04:25.27 | SeRi | havent Meh. |
| 04:25.31 | SeRi | lol |
| 04:25.35 | SeRi | Meh... |
| 04:25.37 | p3nguin | /usr/sbin/darkhttpd /srv/http/ --log /var/log/darkhttpd/log --chroot --daemon --mimetypes /etc/mime.types --uid nobody --gid nobody |
| 04:25.52 | p3nguin | Drop your files into /srv/http/ and away you go. |
| 04:26.44 | p3nguin | or /usr/sbin/mongoose -access_log /var/log/httpd/access.log -acl +192.168.0.0/16 -error_log /var/log/httpd/error.log -ports 80 -root /srv/http -uid nobody |
| 04:26.52 | p3nguin | ACL makes that one nice. |
| 04:27.43 | slav3_kitten | oh ... |
| 04:27.52 | slav3_kitten | that is nice |
| 04:29.15 | p3nguin | Of course those aren't always the right tools for every job. |
| 04:29.33 | p3nguin | Sometimes you want or need configuration files for certain things. |
| 04:29.59 | p3nguin | But for what you asked about, I'd probably use mongoose. |
| 04:34.11 | p3nguin | slaps seri around a bit with a 60 lb UNIX manual |
| 04:34.56 | slav3_kitten | yea mongoose sounds best |
| 04:35.08 | slav3_kitten | but all it needs to do is serve 3 files |
| 04:35.59 | p3nguin | I don't remember liking/disliking one of those over the other. Pick one and see if it does what you want. |
| 04:36.54 | slav3_kitten | i like the ACL bit on mongoose honestly |
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| 04:45.07 | dijib | http://www.ottawacitizen.com/news/Federal+government+cancels+fighter+purchase/7663407/story.html |
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| 04:49.05 | lorsungcu | the acronym RACF makes me giggle |
| 04:52.32 | p3nguin | How is it that it gives a page not found for me? |
| 04:52.46 | p3nguin | Page Not Found |
| 04:52.46 | p3nguin | We\u2019re sorry, the page you requested is not available. |
| 04:52.46 | p3nguin | If you typed in a URL, please check the address and spelling. |
| 04:53.12 | p3nguin | But I didn't type it -- I used the one you gave me! |
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| 05:09.41 | dijib | i didnt type it |
| 05:10.07 | dijib | i copy pasta'd |
| 05:18.18 | lorsungcu | i think i found sometihng? |
| 05:18.26 | lorsungcu | in my e&m issue |
| 05:18.29 | lorsungcu | pb coming up |
| 05:19.07 | lorsungcu | http://pastebin.com/v8b0Sgpf |
| 05:19.21 | lorsungcu | any idea what the unknown control bit is? |
| 05:20.45 | jpsharp | That's coming from the SIP side of things, though. |
| 05:22.58 | lorsungcu | blah, you're right, and its present when things go right. |
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| 07:03.18 | slav3_kitten | well hell... i don't know what i did but something with my phones is screwed up |
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| 07:05.24 | deo | hello everybody... how do we enable call recording in asterisk? both incoming and outgoing calls... |
| 07:10.07 | jpsharp | stick Monitor() in your dialplan for inbound and outbound calls. |
| 07:10.42 | jpsharp | Or mixmonitor() |
| 07:14.36 | deo | hi jpsharp .. i noticed only outgoing calls were recorded... no incoming calls recorded |
| 07:15.22 | [TK]D-Fender | doeIt's recorded if you call Monitor / Mixmonitor |
| 07:15.28 | [TK]D-Fender | deo: ^ |
| 07:15.40 | deo | hmmnnn |
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| 07:15.57 | deo | you mean everythng is recorded [TK]D-Fender ? |
| 07:16.11 | [TK]D-Fender | deo: Everything you tell it to. |
| 07:16.41 | deo | let me check this... |
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| 09:23.15 | bombev | hi all |
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| 09:45.52 | bulkorok | hi |
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| 12:24.35 | Greenlight | Hiya. Quick question, if I brdige two channels using the AMI Bridge action (so a ManagerBridge), will this cause revites at both sides and cause direct media? |
| 12:25.08 | Greenlight | (Obviosuly provided the normal rules apply - same codec, no nat issues, and I'm not recording or listening for DTMF) |
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| 13:29.42 | Greenlight | Does anyone know where abouts in the code would I find the bit that handles Manager Actions, such as Bridge? |
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| 13:37.59 | ghost75 | someone has combined cidname from astdb with mysql |
| 13:38.02 | ghost75 | ? |
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| 13:53.57 | bombev | is there easy way to create custom context because I got connected to 2 servers via IAX2 trunk, and I want to restrict calls to fixed and mobile phones |
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| 13:56.20 | [TK]D-Fender | bombev, Don't ask these kinds of questions in here and #freepbx at the same time. This is exclusively for #freepbx |
| 13:56.46 | [TK]D-Fender | <ghost75> someone has combined cidname from astdb with mysql <- this makes no sense. Please rephrase. |
| 13:57.21 | ghost75 | to store address information |
| 13:58.06 | bombev | [TK]D-Fender oki |
| 13:58.15 | [TK]D-Fender | ghost75, Since when is CID name already in AstDB? At what point is this "gombininging" you're speaking of supposed to happen? |
| 13:58.28 | [TK]D-Fender | combining* |
| 13:59.06 | ghost75 | only a query basically, i thought to use the phonenumber as "id" in both databases |
| 14:00.03 | ghost75 | in php or perl, get phonenumber from asterisk and then do query mysql and get more information about that number |
| 14:00.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:00.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 14:03.19 | [TK]D-Fender | ghost75, And when are these DB's expected to be populated? |
| 14:03.35 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 14:03.49 | drfreeze | Morning |
| 14:04.27 | drfreeze | I have an office with a PRI providing the phone lines into the office. |
| 14:04.39 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:05.10 | drfreeze | The client has a second office that is connected with a pt2pt and we use SIP to connect the two |
| 14:06.01 | drfreeze | They are selling the other office and having Toshiba provide a cloud based phone service for them. |
| 14:06.34 | drfreeze | They want to have a phone tree where callers to the new office can be transferred directly to the new office/phone number |
| 14:07.47 | drfreeze | I understand that I can just do a dial to the other office and consume two lines at the main office. I was just wondering, is there a way to do a hand-off to the new provider and completely let the call go on my end |
| 14:08.07 | cusco | hi |
| 14:08.11 | Greenlight | Not unless your ISDN carrier supports it, which most dont |
| 14:08.19 | Greenlight | afaik |
| 14:08.22 | cusco | using ael, doesn't this work? if( (${CALLERID(num)}>=218290000 && ${CALLERID(num)}<=218290029) || ${CALLERID(num)==218289951}){ |
| 14:08.46 | drfreeze | Greenlight: so, it would be conditioned on our PRI provider and not on Toshiba? Is that what you are saying? |
| 14:09.07 | Greenlight | If I understood correctly, you've have the incomming and outgoing calls both over the PRI trunk? |
| 14:09.21 | drfreeze | right now yes. |
| 14:09.40 | [TK]D-Fender | drfreeze, "phone tree" = Auto Attendant? |
| 14:09.47 | drfreeze | yes |
| 14:10.06 | Greenlight | You could always port the number over to this Toshiba cloud service and deal with it from there? |
| 14:10.32 | drfreeze | Toshiba will have the new number |
| 14:10.46 | drfreeze | But callers will be using the old number for some period of time |
| 14:11.02 | Greenlight | So, port that number over (or set a divert)? |
| 14:11.15 | drfreeze | We can definitely dial the new number (using to of our channels in the process) |
| 14:11.23 | [TK]D-Fender | drfreeze, if the other end supports a DTMF signalling on the port you're using then yes. Otherwise you'll tie up both ports on the PAP either way |
| 14:11.27 | ghost75 | (15:04:08) [TK]D-Fender: ghost75, And when are these DB's expected to be populated? <- over webbrowser |
| 14:11.44 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 14:11.45 | [TK]D-Fender | ghost75, Well you can lookup whatever you want from wherever you want. |
| 14:11.58 | [TK]D-Fender | ghost75, * has dialaplan tools for this. |
| 14:12.13 | drfreeze | [TK]D-Fender: what is the command to do the handoff? |
| 14:12.19 | [TK]D-Fender | ghost75, And AGI if you want to do more complex things while you're at it |
| 14:12.41 | [TK]D-Fender | drfreeze, that depends on your Toshiba. Read your manuals. We don't use those here |
| 14:12.58 | ghost75 | asterisk 11 has the same db structure as older versions? |
| 14:13.21 | Greenlight | For CDR's or internal database? |
| 14:13.40 | ghost75 | internal |
| 14:13.41 | [TK]D-Fender | ghost75, using *'s interfaces, yes. * 10+ moved the bacend from actual BDB to a BDB emulation layer over SQLite |
| 14:13.55 | Greenlight | Still key/value pairs though |
| 14:13.57 | drfreeze | [TK]D-Fender: so, if they have some way of accepting a handoff, asterisk would dial the new number (with some appropriate parameter) and the call gets automatically dropped when the other end answers? |
| 14:14.45 | [TK]D-Fender | drfreeze, the other side will do what the other side does. |
| 14:15.39 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:15.45 | drfreeze | Just trying to understand here. If asterisk has control of the incoming call, seems if I do a hangup, their call gets dropped |
| 14:15.49 | ghost75 | what dialplan tools? |
| 14:16.43 | [TK]D-Fender | ghost75, func_odbc, etc |
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| 14:20.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:21.19 | drfreeze | Looks like IAX is the way to do a handoff |
| 14:22.14 | [TK]D-Fender | drfreeze, You have a remote PAP2+Toshiba on one side and one server on yours. Thre is nothing "IAX2" in there. |
| 14:23.02 | [TK]D-Fender | drfreeze, This is * > PAP2 > Toshiba. No matter what, you have tell the TOSHIBA to do the handoff. Nothing outside of it will free up those ports. |
| 14:23.32 | drfreeze | ok, thank you |
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| 15:01.57 | jacekowski | hi people |
| 15:01.58 | jacekowski | [Dec 7 13:11:03] VERBOSE[13124] chan_dahdi.c: PRI Span: 1 < Ext: 1 Cause: Destination out of order (27), class = Normal Event (1) ] |
| 15:02.30 | jacekowski | that's on ISDN line when making a call |
| 15:02.55 | jacekowski | other numbers work fine, so it looks like problem with remote end |
| 15:03.17 | [TK]D-Fender | "Cause No. 27 - destination out of order. |
| 15:03.17 | [TK]D-Fender | This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term "not functioning correctly" indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line." |
| 15:03.28 | [TK]D-Fender | just like the description says |
| 15:05.24 | jacekowski | well, my problem is with message that asterisk plays to the users |
| 15:05.32 | jacekowski | which is, all circuits are busy message |
| 15:07.15 | *** part/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 15:07.28 | [TK]D-Fender | jacekowski, "Asterisk" does not play anything to the user. You do. |
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| 15:09.31 | jmetro | asterisk plays the dialtone, unless they moved that to the phones now too |
| 15:11.28 | [TK]D-Fender | jmetro, Dialplan where? |
| 15:12.02 | [TK]D-Fender | jmetro, The only time Asterisk makes dialtone is DISA and for DAHDI FXS channels |
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| 15:12.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:13.07 | [TK]D-Fender | jmetro, Even then, he's talking about a recording. The other two are synthesized based on indications.conf |
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| 15:17.02 | Guest61195 | hi, |
| 15:18.34 | *** join/#asterisk rotten777 (~rotten777@fl-67-76-134-15.sta.embarqhsd.net) |
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| 15:24.16 | jacekowski | hmm, how do i play like 1s of silence? |
| 15:25.15 | [TK]D-Fender | Playback(silence/1) |
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| 15:34.11 | fukuda76140 | hi, |
| 15:34.45 | fukuda76140 | I search documentation or tutorial to install and configure T38modem with hylafax |
| 15:36.20 | [TK]D-Fender | fukuda76140, http://tinyurl.com/bm8gt7x |
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| 15:50.27 | jmetro | [TK]D-Fender i think this one is more informative http://tinyurl.com/bhhn4ot |
| 15:50.33 | rotten777 | hey guys is there a quick and easy way to test if port forwards are working? i've got *11 behind NAT and can't seem to have the audio come over but all the ports seem to be good |
| 15:50.55 | rotten777 | and other port forwards seem to be working fine |
| 15:51.05 | jmetro | there is www.yougetsignal.com |
| 15:51.26 | [TK]D-Fender | rotten777, Testing means doing. |
| 15:51.54 | *** join/#asterisk nightrid3r (~kvirc@62.205.65.54) |
| 15:51.54 | [TK]D-Fender | rotten777, And * needs more than just port forwarding. Three a re plenty of sip.conf parameters that need to be set |
| 15:52.05 | rotten777 | perfect jmetro... i hate this friggin router |
| 15:52.05 | rotten777 | port showing closed |
| 15:52.29 | rotten777 | the configuration was working i just moved offices |
| 15:52.29 | rotten777 | the other sip parameters i wouldn't expect to change |
| 15:52.45 | rotten777 | all the did's are registering and calls go out but no udp happiness |
| 15:52.48 | [TK]D-Fender | There isno such thing as "closed" this is UDP |
| 15:52.51 | [TK]D-Fender | It is stateless |
| 15:53.21 | rotten777 | ok the router shows counters on how much traffic is forwarded on specific nat rules and nothing is showing up on any of the udp forwards setup for asterisk |
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| 16:04.40 | rotten777 | rebooted router... same. using voip.ms and flowroute... any help? |
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| 16:10.50 | lorsungcu | rotten777: whats the issue? |
| 16:11.28 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 16:12.24 | [TK]D-Fender | rotten777, Show us that you're registering with them. Show call attempts out. Check your counters on that traffic to make sure you're even looking at the right things. |
| 16:12.25 | [TK]D-Fender | ~pb |
| 16:12.25 | 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. |
| 16:12.28 | [TK]D-Fender | ^^^ your friend |
| 16:14.09 | rotten777 | http://pastebin.com/TVNHKbKu |
| 16:14.10 | rotten777 | sip.conf |
| 16:14.41 | rotten777 | lorsungcu: sip peers are registering, calls go out (testing to my cell), no audio received or sent |
| 16:14.49 | [TK]D-Fender | rotten777, sip.conf = broken |
| 16:14.59 | rotten777 | that's what i thought |
| 16:14.59 | [TK]D-Fender | rotten777, register's have to be at the END of [general] |
| 16:15.14 | [TK]D-Fender | rotten777, it cuts off everything below them |
| 16:15.40 | rotten777 | changed that.. did sip reload.. dialing now |
| 16:15.52 | [TK]D-Fender | rotten777, Also you have not configured any of the settings required for * to work from behind NAT |
| 16:16.38 | [TK]D-Fender | under [general] nat=yes , localnet=.... directmedia=no |
| 16:16.44 | rotten777 | lol i'm very confused why it was working when i had the same exact conf on a box behind nat at my old office... |
| 16:16.52 | rotten777 | k looking that up for nat settings thx |
| 16:17.02 | [TK]D-Fender | rotten777, all of your remote phones should have "qualify=yes", etc |
| 16:17.30 | rotten777 | no remote phones just the ip dids |
| 16:17.40 | rotten777 | single polycomp 600 right now |
| 16:17.51 | rotten777 | localnet=10.1.1.0/24 |
| 16:17.54 | rotten777 | is that the right syntax? |
| 16:18.31 | rotten777 | ahh nevermind looks like subnet/full mask |
| 16:18.33 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 16:21.22 | rotten777 | still same symptoms... |
| 16:22.10 | rotten777 | http://pastebin.com/icnSTH2g |
| 16:22.26 | rotten777 | counter is going up on firewall now at least |
| 16:22.30 | rotten777 | some traffic is going over udp 5060 |
| 16:22.58 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 16:23.30 | rotten777 | [TL]D-Fender, does it matter if the qualify=yes is for a phone that's on the same subnet? will that screw up the audio being routed? |
| 16:23.51 | rotten777 | on an outgoing call at least... |
| 16:24.01 | lorsungcu | no |
| 16:24.15 | rotten777 | ok |
| 16:26.23 | lorsungcu | i don't see externip anywhere in there |
| 16:26.56 | lorsungcu | rotten777: ^ |
| 16:28.16 | rotten777 | added externip=myip |
| 16:28.28 | rotten777 | a reload in the asterisk console should change that right? |
| 16:28.33 | rotten777 | or do i need to restart the service |
| 16:28.38 | lorsungcu | core reload will do fine |
| 16:28.45 | lorsungcu | or sip reload |
| 16:30.16 | rotten777 | same <.< |
| 16:30.40 | rotten777 | I'm going for a beer and pizza. maybe my mind will be refreshed when I return. thanks for the help you guys |
| 16:30.42 | rotten777 | I appreciate it |
| 16:31.49 | lorsungcu | what router |
| 16:32.04 | rotten777 | routerboard 1100 |
| 16:32.09 | lorsungcu | oh nice, thats my specialty |
| 16:32.27 | rotten777 | seriously? |
| 16:32.30 | lorsungcu | PM me if you want me to have a look |
| 16:32.31 | lorsungcu | yeah |
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| 17:59.54 | mogra | got a Digium s800i running 1.2.0.4, tells me there's an update to 1.3.0.2. Where woudl I grab that? Can't seem to find it on Digium or Asterisk sites |
| 18:00.59 | Qwell | Call them and ask? |
| 18:01.38 | mogra | heh, ironic that my current task is "Phones are broke, fix them" and you say that :) |
| 18:02.11 | Qwell | The point is, this is #asterisk. You are asking for help with a commercial PBX. |
| 18:02.41 | Qwell | There are many ways to contact support. |
| 18:02.44 | Qwell | This is not one of them. |
| 18:02.54 | mogra | ya, I was just remarking on "Call them" |
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| 18:35.23 | k3asd` | hi all |
| 18:35.57 | k3asd` | I have a problem with trunk eutelia |
| 18:36.42 | k3asd` | when I do a call, falls after 6 seconds |
| 18:37.18 | k3asd` | in the asterisk cli console I see: |
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| 18:38.18 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 18:38.20 | k3asd` | <PROTECTED> |
| 18:38.20 | k3asd` | <PROTECTED> |
| 18:38.20 | k3asd` | <PROTECTED> |
| 18:38.20 | k3asd` | <PROTECTED> |
| 18:38.20 | k3asd` | <PROTECTED> |
| 18:38.21 | k3asd` | <PROTECTED> |
| 18:38.23 | k3asd` | [Dec 7 19:37:48] WARNING[4050]: chan_sip.c:9028 process_sdp: Ignoring video media offer because port number is zero |
| 18:38.26 | k3asd` | <PROTECTED> |
| 18:38.28 | k3asd` | <PROTECTED> |
| 18:38.30 | k3asd` | [Dec 7 19:37:55] WARNING[4050]: chan_sip.c:3687 retrans_pkt: Retransmission timeout reached on transmission 741C910-3FD411E2-9B3BFF91-F392020B@62.94.71.98 for seqno 101 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 18:38.31 | WIMPy | ~pb |
| 18:38.31 | 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. |
| 18:38.36 | k3asd` | Packet timed out after 6400ms with no response |
| 18:38.38 | k3asd` | [Dec 7 19:37:55] WARNING[4050]: chan_sip.c:3716 retrans_pkt: Hanging up call 741C910-3FD411E2-9B3BFF91-F392020B@62.94.71.98 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 18:38.42 | k3asd` | <PROTECTED> |
| 18:38.44 | k3asd` | ops |
| 18:38.46 | k3asd` | sorry |
| 18:38.53 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 18:38.58 | WIMPy | Did you read the URL you just included? |
| 18:39.40 | k3asd` | yes, you do not know how to fix it |
| 18:39.53 | k3asd` | I'm* |
| 18:39.56 | k3asd` | I* |
| 18:40.48 | WIMPy | Get a working network connection. |
| 18:40.55 | WIMPy | Is your Asterisk behind NAT? |
| 18:42.22 | k3asd` | yes |
| 18:44.07 | WIMPy | Looks lik you need to configure your FW. |
| 18:44.38 | k3asd` | but I not know nat port 5060 |
| 18:44.42 | *** join/#asterisk nightrid3r (~kvirc@62.205.65.54) |
| 18:44.42 | k3asd` | It's insecure |
| 18:45.29 | WIMPy | I didn;t understand that. |
| 18:45.55 | WIMPy | You also need to configure Asterisk for the situation. But that doesn;t seem to be your issue (yet). |
| 18:47.44 | k3asd` | to solve the problem should be a nat port 5060 on my fw? |
| 18:49.48 | WIMPy | cannot parse that, either. |
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| 18:52.18 | Robotman321 | k3asd`: yes |
| 18:56.32 | k3asd` | Robotman321, but nat 1:1? |
| 18:57.14 | k3asd` | Robotman321, but so someone from the outside could connect to my PBX |
| 18:57.20 | k3asd` | it's not secure |
| 18:57.21 | k3asd` | or no? |
| 18:58.05 | WIMPy | Depends on your FW and the peer you're trying to talk to. |
| 19:00.24 | k3asd` | WIMPy, ok - now I feel |
| 19:00.30 | k3asd` | just a moment |
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| 19:08.47 | k3asd` | yes, now it works |
| 19:08.56 | k3asd` | thank you very much |
| 19:14.25 | lorsungcu | how can i verify start dial supervision on an e&m circuit with dahdi? |
| 19:14.48 | seanbright | anyone here use Sangoma's new AMD for asterisk software yet? |
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| 19:45.44 | k3asd` | bye |
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| 19:56.41 | lorsungcu | think we narrowed the issue with my E&M circuit |
| 19:56.46 | lorsungcu | the telco is saying that on failed calls |
| 19:56.58 | lorsungcu | we are sending 11 digits, or at least that is what they are seeing get passed to their network |
| 19:57.07 | lorsungcu | the last digit we send gets duplicated |
| 19:57.13 | lorsungcu | any ideas why that would happen? |
| 19:57.31 | lorsungcu | increasing tone duration to 300ms seems to have helped a little, maybe? |
| 19:57.59 | lorsungcu | I'm reading through the info on winks and such to see if theres a timing I can adjust near the end of the dtmf |
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| 20:40.04 | jmetro | okay. I'm going alitle craz with Syntax. |
| 20:40.04 | jmetro | exten => s,n,ExecIf($[${hours}=business]?Dial(SIP/501@context,20,r)) |
| 20:40.04 | jmetro | means that if Hours = Business it dials 501 with no else statement, correct? |
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| 20:41.18 | ghost75 | i think so |
| 20:41.29 | ghost75 | i used gotoiftime |
| 20:42.03 | leifmadsen | jmetro: correct |
| 20:42.28 | leifmadsen | Execute the true statement (Dial...) if $[....] evaluated to true |
| 20:42.35 | jmetro | right |
| 20:42.42 | jmetro | and if there was a : that would be the else |
| 20:42.46 | leifmadsen | if false, continue at next prioritiy |
| 20:42.57 | leifmadsen | it would be the separator for the false path, right |
| 20:43.00 | ghost75 | where is "business" defined? |
| 20:43.21 | leifmadsen | defined in the nether |
| 20:43.30 | jmetro | earlier in the code, it works, i'm just reviewing it for myself |
| 20:44.00 | leifmadsen | ExecIf($[foo = bar]?NoOp(I will not trigger):NoOp(but I will)) |
| 20:44.18 | jmetro | right. |
| 20:44.25 | ghost75 | i put like this: GoToIfTime(22:00-08:00,mon-sun,*,*,/usr/share/zoneinfo/Europe/Berlin?voiceunavail,s,1) |
| 20:44.41 | jmetro | doesnt check for holidays =) |
| 20:44.55 | ghost75 | how you do this? |
| 20:45.27 | jmetro | by building a larger block for hours checking that checks the date and the time. and setting variables [hours = business, set to false if a match is found] |
| 20:46.23 | ghost75 | sounds complicated |
| 20:46.44 | jmetro | hm...i have 5 lines of code without comments and blankspace |
| 20:46.59 | ghost75 | show |
| 20:47.05 | leifmadsen | and tell |
| 20:47.16 | jmetro | i dont know if the code writer would want me to =p |
| 20:47.48 | ghost75 | in case of holiday i can turn on DND mode |
| 21:01.10 | pabelanger | How do people usually get an overview of their queuemembers from asterisk? |
| 21:04.02 | leifmadsen | pabelanger: that's a loaded question :D |
| 21:04.31 | leifmadsen | pabelanger: I think in later versions you can ask for it from AMI, but only in like.. 10 or 11 can you get all the queue at the same time without cycling/looping yourself |
| 21:04.36 | leifmadsen | otherwise, you can get it from the CLI, but... meh |
| 21:04.54 | _Corey_ | We just parse AMI events and deal with it that way |
| 21:05.18 | pabelanger | leifmadsen, Ya, I can get it from looping queues, but I don't see another way to directly query them |
| 21:08.34 | pabelanger | I guess queue_member_table is the only other choice |
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| 22:18.24 | pabelanger | Any body ever tried to setup an apache proxy in front of asterisk http, for the purpose to add HTTP headers? |
| 22:21.31 | pabelanger | Basically my developer is asking to add http://enable-cors.org/ headers to asterisk |
| 22:21.39 | pabelanger | which, i figure I can just use a proxy |
| 22:26.15 | *** join/#asterisk rotten777 (~rotten777@fl-67-77-94-69.sta.embarqhsd.net) |
| 22:27.48 | rotten777 | hey guys with help the calls are coming and audio is as well ... iptables... meh... anyway now for some reason my callerid has changed on outgoing calls. i've got s,1,Set(CALLERID(num)=1234567890) on the outbound context |
| 22:27.51 | rotten777 | any ideas? |
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| 22:37.40 | traph | hey |
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| 22:38.56 | traph | what's the difference between installing asterisk from source or from the package manager? |
| 22:40.58 | rotten777 | package manager is precompiled with a set of options |
| 22:41.01 | rotten777 | from source you can change which options |
| 22:41.14 | rotten777 | package manager you don't need gcc tools and such |
| 22:46.10 | traph | rotten777: does the precompiled package setup hinder flexibility? |
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| 23:33.03 | tonikasch | bye |
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| 23:58.34 | P-NuT | Hi all, can anyone point me in the direction of connecting asterisk to asterisk using SIP / TLS and srtp? |
| 23:59.08 | P-NuT | There's a lot around dealing with connecting a softphone, but I cant find anything relating to asterisk to asterisk |