| 00:03.32 | linuxfool | So what front ends are you guys using for your asterisk setup? |
| 00:04.44 | ChannelZ | nano |
| 00:11.15 | Mango45 | Notepad++ |
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| 00:36.10 | [TK]D-Fender | copy con > |
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| 00:44.48 | Mango45 | You win |
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| 09:08.00 | Stefan27 | I edited <ast12>/contrib/scripts/safe_asterisk and ran make install from <ast12> folder, but that did not overwrite the old /usr/sbin/safe_asterisk, how come? |
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| 09:22.26 | RadJackson | Hello i've been using google voice recognition api on asterisk , lately google have closed this API |
| 09:22.36 | RadJackson | Do you have any other service to recommand ? |
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| 09:45.10 | linocisco | hi all , who is running call center? |
| 09:45.16 | linocisco | hi all , who is running call center business? |
| 09:54.43 | linocisco | is acceptable to ask callcenters which hardware/sofware are being used? can we expect their reply? |
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| 10:02.25 | gavimobile | is it possible that my tisp's rtp ports are below the 10,000 port range? |
| 10:02.53 | gavimobile | <PROTECTED> |
| 10:14.26 | gavimobile | can someone give me a hand here. my pbx started to use rtp ports below and above the specified rtp ports in rtp.conf |
| 10:18.31 | file | if you are referring to the probation message - that's the source of received media, which the remote side is free to use any port they want |
| 10:19.40 | gavimobile | but my firewall logs are showing that ports both above and below my set rtp ports are trying to be used when a call is initiated |
| 10:20.03 | gavimobile | no audio |
| 10:20.13 | gavimobile | unless I allow outgoing traffic on all ports |
| 10:20.34 | file | the remote side can freely choose where it wants to receive media, for both IP address and port |
| 10:20.40 | file | you can not control that unless you control the remote side |
| 10:21.10 | file | what you control is where *you* want to receive media |
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| 10:21.37 | gavimobile | file http://pastebin.com/TqUa9nq3 |
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| 10:21.55 | file | yes? |
| 10:22.27 | gavimobile | file: so does this mean I need to open all udp ports? |
| 10:22.35 | file | outgoing, yes |
| 10:22.53 | file | unless your upstream can provide a range of ports that they will use for RTP |
| 10:24.09 | gavimobile | file: weird, I blocked outgoing traffic a few days ago and it was working fine. just today I needed to reset my routers config for another issue im having with cisco support and after I reloaded my settings back to how it was this morning I am not able to get audio |
| 10:24.42 | gavimobile | I had audio this morning with only 10000 to 10200 open for outgoing |
| 10:26.10 | gavimobile | is it documented in the asterisk documentation that ALL outbound udp traffic should be allow as default? |
| 10:27.02 | file | doubt it |
| 10:27.27 | gavimobile | file: so could you explain why you think it was working? |
| 10:27.35 | gavimobile | and why its not working now |
| 10:27.40 | file | because at the time they may have chosen UDP ports within that range? |
| 10:28.20 | file | don't know otherwise |
| 10:29.17 | gavimobile | do torrents work on udp traffic, I don't want users to connect with torrents |
| 10:31.43 | gavimobile | file: wait a second, the pastebin I posted was an outbound call I made.. that means it should be using the udp ports I specified, no? |
| 10:31.54 | file | it does. |
| 10:32.01 | gavimobile | but its not |
| 10:32.03 | file | but you want bidirectional media, so there's RTP flow in BOTH directions |
| 10:32.11 | gavimobile | hrm |
| 10:32.13 | gavimobile | right |
| 10:32.34 | gavimobile | wouldn't that mean that I would need to open all inbound udp traffic instead of outbound |
| 10:32.52 | file | no. |
| 10:33.05 | gavimobile | my logs clearly show that the outbound traffic that my pbx made is clearly using ports which are not in the rtp range I specified |
| 10:33.06 | file | you specify the range of ports you are willing to accept media on, you control that, you can forward/allow only those ports |
| 10:33.17 | file | wrong |
| 10:33.23 | file | the source port is the port you've specified |
| 10:33.27 | file | or rather, the range |
| 10:33.38 | file | the destination port is NOT because that's the port the remote side has said it wants media on |
| 10:34.20 | gavimobile | alright! ill take your word for it! thanks |
| 10:34.26 | gavimobile | :-( |
| 10:34.42 | file | it's in the logging - SPT is source port, DPT is destination port |
| 10:34.43 | gavimobile | I was so sure it was working with with blocking all outbound traffic except what I've specified |
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| 10:37.20 | gavimobile | file: thanks for your help! |
| 10:37.31 | gavimobile | I only allowed all udp traffic for my pbx's ip address! |
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| 10:56.40 | cnovice | hello~~ |
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| 11:12.38 | giorgiodinapoli | hey guys i get this message: ignoring 'video' media offer because port number is zero -- Connected line update |
| 11:12.46 | giorgiodinapoli | but i dont think that my client offers this. |
| 11:12.54 | giorgiodinapoli | how can i see the incoming SDP request? |
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| 11:40.39 | file | "sip set debug on" |
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| 11:57.03 | RadJackson | Hi ,the asterisk cmd Dial , dial out all the specified channels , it there a way to ring only available devices ? |
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| 12:06.57 | WIMPy | How could it possibly ring unavailable devices? |
| 12:10.28 | giorgiodinapoli | file, thanks very much! and how can i dump this info into file or something bcs to much info for the consoel is coming, can not scroll back on top |
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| 12:45.17 | giorgiodinapoli | okay im honest. i tried to rtfrfc but i still dont understand this: fmtp:101 0-15 |
| 12:45.24 | giorgiodinapoli | what is it ^ |
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| 13:21.36 | SirLagz | [TK]D-Fender: it seems the static port mapping was what I needed. Thanks for all your help. |
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| 13:24.36 | riess82 | hi everyone |
| 13:24.54 | SirLagz | hi |
| 13:27.14 | riess82 | am i in the right place to ask about an asterisk implementation as an extension for an existing isdn system? i've been googling all across the world and can't seem to figure out if what i want to do is possible |
| 13:28.10 | WIMPy | It is possible. |
| 13:28.19 | WIMPy | what exactely is your question? |
| 13:30.10 | riess82 | i want to use asterisk as an extension behind an existing isdn-system (no-name). example: from outside i dial mynumber123, where asterisk is listening on the extension 12 and forwarding the call to whatever number 3 is (eg my soft-phone) |
| 13:30.30 | [TK]D-Fender | riess82: sure |
| 13:30.54 | WIMPy | That's not a question :) |
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| 13:31.22 | WIMPy | What's your setup? What kind of line(s)? Do you have spare ports on your PBX? |
| 13:31.40 | [TK]D-Fender | WIMPy: Clearly does since he want's to plug Asterisk onto one... |
| 13:31.46 | [TK]D-Fender | -' |
| 13:32.09 | riess82 | we have a no-name isdn system now with like 4 or 5 extensions. there are still 2 free extensions |
| 13:32.26 | WIMPy | But is it the right kind of port? |
| 13:33.10 | riess82 | don't know. i can plug in any 20⬠isdn phone to the ports. does that help as info? |
| 13:33.34 | WIMPy | Yes. And Yes, you can connect Asterisk directly to that. |
| 13:33.35 | [TK]D-Fender | riess82: Yes you can do this |
| 13:34.16 | riess82 | i asked 2 VOIP companies from around that told me NO, and offered a special deal with 3000⬠or more =) |
| 13:34.36 | WIMPy | Seems to be a pretty unusual thing. |
| 13:34.47 | riess82 | but they weren't very comvincing in the first place |
| 13:35.23 | riess82 | so i need an usb or pci-e card with a FXS? FXO? port to connect? |
| 13:35.35 | [TK]D-Fender | riess82: No, you need an ISDN card |
| 13:35.43 | [TK]D-Fender | riess82: FXO/FXS is analog. |
| 13:35.48 | riess82 | ok |
| 13:35.55 | WIMPy | A bog standard ISDN BRI card or USB dongle. |
| 13:36.12 | WIMPy | No need for expensive stuff. |
| 13:36.40 | sekil | finally ...aoc-e works |
| 13:37.05 | WIMPy | sekil: From where to where? |
| 13:37.26 | sekil | telco didn't send it at first |
| 13:37.32 | sekil | now they do.. |
| 13:37.43 | sekil | I pick it up when isdn side hangs up |
| 13:37.48 | WIMPy | Oh, I thought it was about something within Asterisk. |
| 13:37.50 | sekil | otherwise no...only last aoc-d |
| 13:39.06 | sekil | is there any chance to "catch" it when sip side hangs up? |
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| 13:39.11 | sekil | I do see it in debug |
| 13:39.23 | sekil | but can't get on h extension |
| 13:40.34 | WIMPy | Don#t think so |
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| 13:43.17 | riess82 | WIMPy: i have this in use for fax atm. is that what i need to use? http://www.usr.com/en/products/56k-dialup-modem/usr5637/ |
| 13:44.10 | sekil | WIMPy: do you have aoc sent to your bris? |
| 13:44.55 | WIMPy | riess82: I don't see anything about ISDN there. It looks analogue. |
| 13:44.57 | [TK]D-Fender | riess82: that is an ANALOG modem. |
| 13:45.11 | [TK]D-Fender | riess82: You need an ISDN card. |
| 13:45.39 | WIMPy | sekil: I think it's only AOC-D, just with a total in the last message. |
| 13:47.04 | sekil | WIMPy: yeah but AOC-E may contain different number ...usually no though |
| 13:47.17 | WIMPy | Different number? |
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| 13:48.39 | riess82 | [TK]D-Fender: so if i know that this analog modem is working as an extension to the system now, maybe the isdn-system only provides analog extensions after all? I think i'll have to find out more about the current system before i can continue to ask for help here |
| 13:48.57 | sekil | WIMPy: different number of units |
| 13:49.39 | WIMPy | riess82: Yes, you should. Earlier you said you can connect ISDN phones. |
| 13:49.58 | WIMPy | sekil: No. If you get an AOD-D with the disconnect that's the same. |
| 13:50.23 | [TK]D-Fender | riess82: You said you plugged ISDN phones into it... you don't seem to know what you have even.... |
| 13:50.27 | WIMPy | I cannot remember the exact difference. I think it oly affects information about calls that are not active on your line. |
| 13:50.50 | riess82 | WIMPy: yeah, that's what i thought. i'll have to dig into the current system |
| 13:51.15 | sekil | WIMPy: one can add a unit on AOC-E if one wants it.. |
| 13:51.43 | sekil | WIMPy: to bill for additional unit at the end of the call...hence I think AOC-E data is most important to get |
| 13:52.24 | WIMPy | sekil: No. The information in an AOC-D message after disconnect will be the same as you'd get in an AOC-E message. |
| 13:53.17 | sekil | WIMPy: I don't think AOC-D needs to come on disconnect by default..only if time period hits it |
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| 13:53.58 | WIMPy | If you don't have additional AOC-E, it will usually take that part as well. |
| 13:55.28 | sekil | WIMPy: usually yeah...but not 100% |
| 13:55.36 | sekil | WIMPy: which is why I'm pissed |
| 13:56.15 | WIMPy | Yes. It's not a must. |
| 13:56.38 | sekil | yup |
| 13:56.41 | WIMPy | But here it used to be all AOC-D. |
| 13:56.48 | sekil | Germany? |
| 13:56.58 | WIMPy | yes |
| 14:01.02 | workingcats | riess82, it could be that your system connects with ISDN to your provider, but then the "ISDN thingy" (whatever device it may be, where you then connect your actual telephony devices) only provides analogue ports. that is not uncommon in germany |
| 14:01.22 | workingcats | so the ISDN thingy converts between ISDN and analogue |
| 14:01.26 | sekil | WIMPy: used to be? you mean now they do AOC-E too? |
| 14:01.50 | WIMPy | sekil: No I mean that we don't really have phone lines any more. |
| 14:02.04 | sekil | WIMPy: hehe...all voip these days? |
| 14:02.17 | WIMPy | Almost :-( |
| 14:02.22 | workingcats | WIMPy, my grandma still pays 18â¬/month line rental for her analogue line lol. pretty sure it actually is a real line, too |
| 14:02.42 | workingcats | real as in "an actual phone line, just like the good ol' days" |
| 14:03.05 | WIMPy | Yes, there are still a lot of real lines in service, but for the last couple of years all new ones were fake. |
| 14:03.28 | WIMPy | Well most anyway. |
| 14:03.40 | workingcats | yep |
| 14:03.52 | workingcats | i think DTAG is about to abolish selling new lines and may have started mandatory conversion |
| 14:04.22 | WIMPy | They have, but they ran in to some legal issues, I think. |
| 14:04.34 | workingcats | hehe seems to be their S.O.P. |
| 14:05.10 | WIMPy | Well, considering they originally wanted to switch off the PSTN in 2010, I guess it's not exactely a success story. |
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| 14:06.37 | workingcats | well, the less success they have the less it bothers me to hear about it ;) |
| 14:08.40 | Chainsaw | workingcats: I bet they still employ someone to sit there and go "operator" when she lifts the handset. |
| 14:09.00 | workingcats | Chainsaw, nah she's all modern on the client side, DECT and everything ;) |
| 14:09.18 | workingcats | well, 1 dect base station and one handset |
| 14:17.36 | sekil | WIMPy: so ISDNs are not provisioned any more? sound weird knowing germany was all isdn? |
| 14:18.56 | WIMPy | You can still get real lines, but you have to ask very explicitely for them. |
| 14:19.29 | WIMPy | POTS will be even harder. |
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| 14:22.24 | workingcats | yep |
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| 15:02.06 | c|oneman | how do I troubleshoot a polycom phone that has no bootrom and cannot find the server? |
| 15:02.18 | c|oneman | I have pleanty of others that work with the same settings |
| 15:02.35 | sekil | no bootrom? |
| 15:03.12 | carrar | install said bootrom |
| 15:03.22 | WIMPy | I think they have special terminology. |
| 15:03.24 | c|oneman | how? it can't pull it from the server |
| 15:04.57 | carrar | So your bootrom is working |
| 15:05.05 | carrar | since it can't find the server |
| 15:05.42 | [TK]D-Fender | Go prove it's looking in the right place |
| 15:06.12 | [TK]D-Fender | and that the bootrom is actually there, that your <mac>.cfg points to it, etc |
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| 15:13.37 | c|oneman | the provisionning server is a bit of black box |
| 15:13.52 | c|oneman | but I've also tried with polycom's public one |
| 15:14.32 | [TK]D-Fender | Polycom is colour-blind... |
| 15:14.41 | [TK]D-Fender | go lok at the phone and prove where it's looking |
| 15:15.25 | c|oneman | I don't understand what you mean by prove |
| 15:16.36 | [TK]D-Fender | wlk up to the phone , drill through the menus and LOOK at the server it says its pointing to. |
| 15:16.59 | c|oneman | well, of course I checked that. |
| 15:17.11 | c|oneman | I'll try again with your local provisionning server. |
| 15:17.16 | c|oneman | our* |
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| 15:58.53 | jameswf | malcolmd any thoughts on why dpma would hate a floating IP |
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| 16:01.33 | malcolmd | replied w/ a pmsg |
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| 20:20.50 | TazzNZ | hey all |
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| 20:25.38 | carrar | haha https://i.imgur.com/4fAjB6H.jpg |
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| 20:41.29 | Micc | I have a queue with a few SIP members, and when multiple calls come in it rings multiple lines on the sip phone, is there a way to get it to only ring on one line per sip account? |
| 20:42.30 | Micc | I'm using strategy ringall, but I just want it to ring all the sip accounts, but not multiple callers at a time. |
| 20:48.05 | Micc | ah. I think its because I have the same sip account in multiple queues. |
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| 21:28.41 | tom______ | I have a strange issue. I have a very simple asterisk. 1 4port digium t1 card and only 1 line in extensions. basically I take the pri and send all calls to the same sip gateway. I see the pri info and asterisk shows a dial in the cli but I do not see the dial on the wire for 30 seconds. There is nothing else on this server. when I ping from the tty it works as expected. What would cause the delay in asterisk hitting the eth? |
| 21:30.31 | TazzNZ | DNS issues ? |
| 21:31.10 | tom______ | I send to an ip. But I just did a dig and it responded just fine |
| 21:32.45 | TazzNZ | so, you saying, the calls is coming in on the PRI and then a dial command is shown on the CLI, then 30 seconds later, you see the sip invite going onto the wire ? |
| 21:33.01 | tom______ | correct |
| 21:33.20 | TazzNZ | pastebin your extensions.conf file |
| 21:33.41 | tom______ | ok. i second |
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| 22:16.19 | Quastor | Hi can someone help me with my problem please? I have an inbound route with destination fax-dest (email). When I try to fax to it, it even doesn't show up in verbose mode. If I call the number with a phone I hear the fax sounds |
| 22:18.24 | [TK]D-Fender | "core set verbose 10" <- if you see nothing after this... then the call isn't hitting yoruserver |
| 22:18.37 | [TK]D-Fender | (dialplan that is) |
| 22:20.17 | Quastor | Ok but if you call with a phone to it, it shows up on the pbx |
| 22:20.23 | Quastor | in normal order |
| 22:20.29 | Quastor | and correct steps |
| 22:20.43 | Quastor | I'm trying with verbose 10 |
| 22:22.24 | Quastor | [TK]D-Fender: -- Channel 'SIP/SoIP1-0000e95c' FAX session '35' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
| 22:22.26 | Quastor | <PROTECTED> |
| 22:22.27 | Quastor | <PROTECTED> |
| 22:22.29 | Quastor | <PROTECTED> |
| 22:22.30 | Quastor | <PROTECTED> |
| 22:22.37 | [TK]D-Fender | PASTEBIN |
| 22:22.41 | [TK]D-Fender | Do not flood in here |
| 22:22.43 | [TK]D-Fender | ~pb |
| 22:22.43 | 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. |
| 22:22.45 | [TK]D-Fender | ^^^ |
| 22:23.03 | [TK]D-Fender | and "call with a phone is vague" |
| 22:23.41 | [TK]D-Fender | That debug was obviously a call landing on your system |
| 22:23.49 | [TK]D-Fender | And only the tail end of it |
| 22:25.31 | Quastor | Ok I pasted the fax session full, maybe there are some clues somewhere in the beginning, [TK]D-Fender: http://pastebin.com/gAT0rnYs |
| 22:26.25 | [TK]D-Fender | So taht call is being processed and attempting a fax receive which failed |
| 22:26.31 | [TK]D-Fender | What about the other kind of result? |
| 22:27.13 | Quastor | Yes now I will call with my mobile to the inbound fax route |
| 22:32.17 | Quastor | [TK]D-Fender: http://pastebin.ca/2826657 |
| 22:33.27 | [TK]D-Fender | Ok, so both calls seem to hit the dialplan and stat to try to receive a fax |
| 22:33.49 | Quastor | So I think the fax is working but the data doesnt get processed to eventually postfixµ |
| 22:34.02 | Quastor | Yes now it is indeed |
| 22:34.43 | [TK]D-Fender | <PROTECTED> |
| 22:34.51 | [TK]D-Fender | <PROTECTED> |
| 22:34.53 | [TK]D-Fender | both FAILED |
| 22:34.57 | [TK]D-Fender | it tried. |
| 22:34.58 | [TK]D-Fender | It failed |
| 22:35.13 | [TK]D-Fender | Don't look at postfix.. there is nothing to send |
| 22:35.42 | [TK]D-Fender | And on the 2nd call you stripped out all the kind of debug you had on the first call |
| 22:35.59 | Quastor | you want the full? |
| 22:36.04 | [TK]D-Fender | No real need |
| 22:36.13 | [TK]D-Fender | There is enough left to show that it failed as well |
| 22:36.17 | [TK]D-Fender | So nothing to e-mail |
| 22:36.57 | Quastor | pastebin.ca/2826659 |
| 22:37.09 | Quastor | Ok but what goes wrong then |
| 22:37.34 | Quastor | I tried a hardware fax/ our PDF2Fax software |
| 22:37.42 | Quastor | wich both work perfectly |
| 22:38.34 | [TK]D-Fender | As far as I see that is a straight SIP call. |
| 22:38.48 | [TK]D-Fender | standard SIP + fax = invitation for tragic failure |
| 22:39.25 | [TK]D-Fender | If you don't have T.38, expect failures. |
| 22:40.25 | Quastor | Oh that would suprise me, I'll check |
| 22:42.29 | Quastor | T38 Pass-Through is enabled |
| 22:43.14 | [TK]D-Fender | This is not passthrough |
| 22:43.16 | Quastor | Codecs : only ulaw is enabled |
| 22:44.52 | Quastor | Oh you mean Fax Detection type? |
| 22:45.35 | [TK]D-Fender | No. |
| 22:45.46 | [TK]D-Fender | This is NTO a pasthrough scenario |
| 22:45.48 | [TK]D-Fender | NOT* |
| 22:46.10 | [TK]D-Fender | "sip set debug on" <- pastebin a new call |
| 22:46.27 | Quastor | Oki |
| 22:49.52 | Quastor | Can i filter the debug mode on 1 extension? Because thats really flooding ma shell :) |
| 22:50.45 | [TK]D-Fender | Just pastebin the call |
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