| 00:23.31 | xochilpili | [TK]D-Fender, still ? |
| 00:23.37 | xochilpili | http://pastebin.com/esmjYU1b << server_A |
| 00:24.30 | xochilpili | http://pastebin.com/Kh5U1Ls0 << server_B |
| 00:31.31 | xochilpili | http://pastebin.com/q8VSBRfi < server_A [NAT] |
| 00:32.23 | xochilpili | http://pastebin.com/L6aufJgN < server_B [NAT] |
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| 00:45.23 | Katty | Peanut butter pretzel time!!! |
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| 01:01.09 | drab | any clue about that issue with ringtone not happening after dialing an extension from an AR system? |
| 01:01.20 | drab | I can hear the ringtone just fine if I dial the number from scratch |
| 01:01.53 | drab | by if I dial the main number with the auto attendant and then press the same extension number the call happens (I can see that in the logs), but I don't hear any ringing |
| 01:02.17 | drab | so if someone didn't know they may think something's broken and hungup |
| 01:05.59 | Katty | What does cli say at verbose 10 |
| 01:28.39 | drab | Katty: when it comes to dial that extension it says the same exact thing, Executing [499@from-internal:1] Dial("SIP/498-00000025", "SIP/499,20,t") in new stack |
| 01:28.53 | drab | the only difference is the Goto statement that sends the dialplan there |
| 01:29.00 | drab | after hitting the DTFM for that extension |
| 01:29.33 | drab | I'm wondering if it's my test phone (soft phone), I need to get another physical phone to try on |
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| 01:30.27 | drab | zapier on iphone |
| 01:30.36 | drab | but it rings fine if I call directly |
| 01:32.06 | drab | I also just tried to use the option "r", was doing some googling, to force ringing, but nothing happens |
| 01:32.13 | drab | same silence until I pick up |
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| 02:06.15 | drab | mmmh, how does this suggestion make any sense? http://lists.digium.com/pipermail/asterisk-users/2011-February/259292.html |
| 02:06.25 | drab | why would you Answer() before a Dial()? |
| 02:06.47 | drab | I can understand something else I ready about answering before playing a MoH or something, but an Answer() before a Dial() seems just odd |
| 02:07.01 | drab | and when I tested that I actually got no sound/ringtone at all |
| 02:07.31 | drab | I get ringtone as expected when I just have the Dial() as priority one |
| 02:07.47 | drab | altho http://www.voip-info.org/wiki/view/Asterisk+cmd+Answer |
| 02:07.59 | drab | By the same token, if you Dial() out before Answer()ing, the ringback |
| 02:07.59 | drab | generated will also be via early media (or, if applicable, |
| 02:08.04 | drab | whups |
| 02:08.21 | drab | but that seems to suggest Answering() before a Dial() makes sense |
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| 02:20.52 | drab | I'm mighty confused by this. the phone is on the local subnet calling another phone on the local subnet |
| 02:21.06 | drab | it's just that one time it's calling it directly and another through an auto attendant |
| 02:21.36 | drab | so in one case it goes straight to the extension while in the other is goes through a Goto first |
| 02:21.57 | K0HAX | Hey, how do I send a variable from a Macro called by Dial() back to the original thread that sent the Dial()? |
| 02:22.15 | K0HAX | I tried Set(key=value,g) but it isn't working, this is on the latest build from Git |
| 02:26.10 | drab | and actually it doesn't even matter, if I take to goto out and just do a Dial, same problem, no ringtone |
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| 02:34.04 | drab | urm, now it's even weirder, if I add rR I still get no ringtone... |
| 02:58.08 | drab | and I just tried a different softphone (linphone) from my mac and some thing, no audio if I go through the autoattendant |
| 03:40.18 | wyoung | I thought macros were deprecated |
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| 03:55.53 | [TK]D-Fender | drab, You have an INDICATIONS issue |
| 03:56.09 | wyoung | hi [TK]D-Fender! |
| 03:56.12 | [TK]D-Fender | drab, Which is either because you can't generate them, or you don't have them defined |
| 04:03.39 | drab | doh, that was it... I'd love to claim I have at least learned something besides patiance through the last 5hrs, but I'm not sure I can come up with anything |
| 04:03.51 | drab | except maybe some Rr stuff while I was looking at Dial |
| 04:04.23 | drab | [TK]D-Fender: for my own erudition, how come it works just fine without indications.conf when I call extensions directly or any other number for that matters? |
| 04:04.41 | drab | people seem to have similar problems with outgoing or incoming calls, I didn't, just the autoattendant thing |
| 04:04.46 | drab | I can't tell what's different about it |
| 04:04.55 | drab | that would trigger this |
| 04:05.12 | [TK]D-Fender | it works when you dial direct because * isn't generating AUDIO |
| 04:05.37 | [TK]D-Fender | It is TELLING the calling phone that it is "ringing" and your phone is generating its own sound |
| 04:06.41 | [TK]D-Fender | And in general you should never be using "r" to force it as audio |
| 04:06.58 | drab | yeah, I read it's bad, I was just testing ot see if it made any difference, and it didn't |
| 04:07.14 | [TK]D-Fender | Because of a lack of indications period |
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| 04:08.14 | drab | I'll have to go back to look at the wiki, but it's the first time this indication thing comes up, I didn't see it through the setup guide, but maybe I missed it |
| 04:08.38 | drab | it's certainly generated by sample-config tho, just didn't see it called out as a necessary piece for a basic setup |
| 04:08.41 | drab | anyway |
| 04:08.51 | drab | thank you very much for the help |
| 04:09.08 | [TK]D-Fender | That's the problem with trying to add things from zero when you don't knwo what's what |
| 04:09.21 | [TK]D-Fender | It's an uphill battle of "oh shit, I should have known that" |
| 04:10.59 | drab | well, in a sense that's exactly why I tried to tho, because I did go with the stock to begin with and there were so many configs I couldn't figure out what was what |
| 04:11.22 | drab | starting from zero the dialplan had so much I was getting lost trying to even figure out which route a simple call was taking |
| 04:11.42 | drab | so I tried to go with the wiki and bare configus and in a sense it's helped a lot ot learn how things work |
| 04:11.52 | drab | but yeah, then stuff like this happens, I hear you |
| 04:12.37 | [TK]D-Fender | You leave the basics there and tweak what needs tweaking |
| 04:12.42 | [TK]D-Fender | indications is stock |
| 04:12.46 | [TK]D-Fender | * also comes wityh MoH. |
| 04:12.50 | [TK]D-Fender | Enough to get you off the ground |
| 04:12.57 | [TK]D-Fender | As well as several others like Asterisk.conf, etc |
| 04:13.12 | [TK]D-Fender | So the idea of starting from Zero means you're lining up a series of failures |
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| 05:17.07 | xochilpili | [TK]D-Fender, still here? |
| 05:17.29 | xochilpili | http://pastebin.com/q8VSBRfi < server_A [NAT] |
| 05:17.35 | xochilpili | http://pastebin.com/L6aufJgN < server_B [NAT] |
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| 09:52.26 | Samot | Anyone work with the Asterisk Call files? |
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| 10:16.06 | BarthezZ | Does anyone has a good idea to manipulate the context of call-forward-handling in ast1.8?? I have a peer sending 302 Moved temporarilyies, which result in asterisk outputting "Now forwarding SIP/SomeSipPeer-00016ec4 to 'Local/SomeDestinationNumber@default' (thanks to SIP/SomeSipPeer-00016ec5)"... But it would be awsome if I could get the local channel in a different context to do the handling apart from |
| 10:16.12 | BarthezZ | <PROTECTED> |
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| 10:23.30 | wdoekes | setvar=TRANSFER_CONTEXT=othercontext on your sip peer? |
| 10:24.15 | wdoekes | (from the top of my head) |
| 10:31.10 | BarthezZ | Ah great wdoekes :) A search on that results in the documentation I couldn't find. Thanks! |
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| 10:49.11 | octo675 | hi all |
| 10:49.19 | octo675 | can any help please |
| 10:49.29 | octo675 | i have a warning flood |
| 10:49.53 | octo675 | chan_sip.c:7350 sip_write: Can't send 10 type frames with SIP write |
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| 10:49.58 | octo675 | thanks ;) |
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| 11:08.51 | BarthezZ | hmm wdoekes, do you have any experience with the transfer_context? I've set the variable on the peer being dialed, and even set it explicitly (tested with __ prefix too) and still my logging reports: http://pastebin.com/5LMpuGnF |
| 11:15.06 | wdoekes | "I've set the variable on the peer being dialed". it should be set on the dialing channel (which could be through the dialing peer) |
| 11:15.40 | wdoekes | i.e. not the dialED peer |
| 11:16.11 | wdoekes | you could set it in the dialplan instead, using Set(__TRANSFER_CONTEXT=xyz) |
| 11:16.16 | wdoekes | before the Dial() |
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| 11:26.26 | _0x5eb_ | hi, after upgrading from Asterisk 1.8 to Asterisk 1.11, I found that the blindxfer function no longer supports multiple digits (when I do #1 123, it tries to xfer to 11 instead of 123), is there a way to remove the leading 1 and support multiple digits again? |
| 11:27.08 | _0x5eb_ | (I guess that some default parameters changed but I don't find which ones) |
| 11:40.02 | octo675 | warning flood chan_sip.c:7350 sip_write: Can't send 10 type frames with SIP write |
| 11:40.05 | octo675 | any idea? |
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| 12:25.06 | BarthezZ | yes wdoekes, I've tried that by setting it explicitly in the dialplan, but the dial statement just seems to ignore it. |
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| 14:38.56 | wdoekes | BarthezZ: hehe; forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT") |
| 14:39.01 | wdoekes | try that one |
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| 15:10.04 | Guest8939 | Howdy, I changed the bindport to 5064 now I do not seem to be able to register any softphones. |
| 15:10.11 | Guest8939 | Are there other settings that must be changed? |
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| 15:20.34 | TandyUK | Guest8939 and your specifiying port 5064 on the softphone |
| 15:21.21 | Guest8939 | Yeah TandyUK, I am suspecting there is a FW issue. |
| 15:21.29 | Guest8939 | Getting nothing on tcpdump, so I close this issue for now |
| 15:25.39 | TandyUK | if the router in front of your softphnes has an ALG turn it off |
| 15:25.54 | TandyUK | on drayteks for example you have to do this from the CLI |
| 15:26.02 | TandyUK | its not in the web gui anywhere |
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| 16:06.23 | grim_ | any news on astricon 2016? |
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| 16:17.25 | dar123 | Hey guyz, is there any tutorial out there to build a scalable service like viber using asterisk? |
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| 16:24.59 | [TK]D-Fender | dar123: There shouldn't be anything specific |
| 16:25.20 | [TK]D-Fender | dar123: * is a telephony toolkit. It is whatever you make it to be and scales depending on how you use it and what you use it with |
| 16:25.44 | [TK]D-Fender | If you're looking at being an ITSP then you should alrady have a backgroud with some of the larger proxy solutions out there |
| 16:26.00 | [TK]D-Fender | And have an idea as to how much * itself is going to do for you. |
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| 16:46.31 | kunwon1 | hi- if i push the same hangup handler onto the same channel twice, what will happen? will it execute twice? will there be an error? |
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| 16:46.48 | [TK]D-Fender | yes twice |
| 16:47.16 | kunwon1 | thanks! |
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| 16:47.43 | jpastore | hi, I'm fairly new to asterisk, but not linux. I have about 140 servers. is there a good log parser for asterisk? I was thinking of setting up a syslog server and dumping everything there and writing a parser to notify |
| 16:48.33 | cresl1n | jpastore: log parser, i.e. logstash/ELK? |
| 16:48.40 | jpastore | some servers getting a little wonky and throw a ton of warnings as the first sign of a death spiral. |
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| 16:49.09 | jpastore | cresl1n, let me check those out. not familiar with them |
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| 17:22.16 | jpastore | cresl1n, logstash looks neat, but I'm looking for something to warn me when events happen |
| 17:24.23 | jpastore | elk certainly is pretty |
| 17:25.20 | cresl1n | jpastore: you can do log related email triggers |
| 17:25.24 | cresl1n | with logstash |
| 17:25.49 | cresl1n | for system health checks, sensu is something that seems to be used in a lot of places |
| 17:25.50 | jpastore | oh yea? I'll check that out! |
| 17:26.05 | jpastore | thanks for that tip as well! |
| 17:26.59 | jpastore | cresl1n, so sensu would be a replacement for nagios? someone recommended zabbix recently. I was going to check that |
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| 18:01.06 | saratogga | So I have sip-cli hitting with unauthenticated INVITEs to dial out but these are never recorded in the log so fail2ban is not picking them up. The From header comes in with the PBX's own IP, so that's what gets written in the log - Never the source IP where the request actually came from. |
| 18:01.08 | saratogga | http://pastebin.com/kvEXtC28 and 27.50.86.100 is the PBX IP |
| 18:01.15 | saratogga | This results in this in the log: [2016-02-09 23:00:47] NOTICE[22407] chan_sip.c: Sending fake auth rejection for device 2310<sip:2310@27.50.86.100>;tag=fd83190e |
| 18:01.31 | saratogga | the offending IP 74.208.77.139 never gets recorded anywhere |
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| 19:02.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 19:40.08 | jybab | ahh it's a wonderful day |
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| 20:14.05 | Kobaz | soooo |
| 20:14.18 | Kobaz | [2016-02-09 15:13:28.436] WARNING[3168]: chan_sip.c:3371 __sip_xmit: sip_xmit of 0x9f0a6a70 (len 485) to 18.11.7.7:55024 returned -2: No such file or directory |
| 20:14.35 | Kobaz | [2016-02-09 15:12:37.235] WARNING[3168]: chan_sip.c:3371 __sip_xmit: sip_xmit of 0x9f7d5ff0 (len 485) to 18.11.7.7:55024 returned -2: Interrupted system cal |
| 20:14.38 | Kobaz | and stuff like that |
| 20:14.54 | Kobaz | i'm wondering a good way to debug this |
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| 20:36.58 | [TK]D-Fender | Check your firewalls |
| 20:46.25 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 20:48.30 | qakhan | i am recording calls using Monitor(gsm,/filename,bm) . some calls record fine but some recording has voice breaking while there is no voice issue in actual call. |
| 20:55.40 | [TK]D-Fender | Difference is due to your phones compensating with PLC and Jitter buffers |
| 20:55.49 | [TK]D-Fender | The files are the way it comes in and that is what you are stuck with |
| 20:57.05 | qakhan | 1 end is softphone and other end is cell phone or landline |
| 20:58.16 | qakhan | do you think it is due to softphone which is insalled on device and call generate over 4G network |
| 20:58.32 | qakhan | plus i am using G.729 codec |
| 21:04.43 | [TK]D-Fender | I just gave you the 2 reasons |
| 21:05.51 | qakhan | i did not understand your second reason "the files are the way it comes in and that is what you are stuck with" |
| 21:06.10 | [TK]D-Fender | That isn't a reason |
| 21:06.15 | [TK]D-Fender | That is a statement |
| 21:06.33 | [TK]D-Fender | there is no setting to adjecut. Your files are the way they are. |
| 21:06.38 | [TK]D-Fender | adjust* |
| 21:06.45 | [TK]D-Fender | [15:55][TK]D-FenderDifference is due to your phones compensating with PLC and Jitter buffers |
| 21:06.47 | [TK]D-Fender | ^^^^^ |
| 21:06.51 | [TK]D-Fender | PLC |
| 21:06.53 | [TK]D-Fender | Jitter Buffers |
| 21:06.55 | [TK]D-Fender | THOSE |
| 21:07.11 | [TK]D-Fender | You are LOSING packets, or they are arriving out of order. |
| 21:07.21 | [TK]D-Fender | The recordings do not have these applied to them |
| 21:07.47 | qakhan | but PLC and jitter buffer should also effect voice |
| 21:09.05 | [TK]D-Fender | it does on the PHONE |
| 21:09.11 | [TK]D-Fender | that's why the PHONE sounds good |
| 21:09.19 | [TK]D-Fender | the recoding is NOT PROCESSED THIS WAY |
| 21:11.50 | qakhan | if it effect voice then why user not able to get it? |
| 21:11.59 | qakhan | user says voice is good |
| 21:12.45 | [TK]D-Fender | Read what I said another 10 times |
| 21:13.10 | [TK]D-Fender | Your PHONE is fixing the audio. ASTERISK ISN'T |
| 21:13.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:14.09 | qakhan | you mean softphone? |
| 21:14.59 | Kunsi | whatever phone you're using, yes |
| 21:17.23 | qakhan | ok, so how can we fix it on asterisk side |
| 21:18.04 | [TK]D-Fender | [15:55][TK]D-FenderThe files are the way it comes in and that is what you are stuck with |
| 21:18.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:18.09 | [TK]D-Fender | Not |
| 21:18.11 | [TK]D-Fender | Going |
| 21:18.13 | [TK]D-Fender | to |
| 21:18.16 | [TK]D-Fender | Get fixed. |
| 21:18.59 | Kunsi | qakhan: asterisk can't fix it, it's a connection problem. |
| 21:19.10 | qakhan | ok, |
| 21:19.37 | qakhan | do you think it is mostly RTP related |
| 21:19.48 | qakhan | firewall is block RTP |
| 21:19.54 | qakhan | blocking* |
| 21:20.02 | [TK]D-Fender | Obviously not |
| 21:20.18 | [TK]D-Fender | That question doesn't even make sense |
| 21:20.57 | [TK]D-Fender | Because if you had a firewall issue you'd have NO sound, not POOR sound. |
| 21:22.03 | qakhan | ok |
| 21:22.57 | qakhan | so we can say that because device has loe 4G signal therefore this problem comes up |
| 21:23.13 | [TK]D-Fender | PLC <- |
| 21:23.15 | [TK]D-Fender | Jitter <- |
| 21:23.18 | [TK]D-Fender | THE END |
| 21:26.23 | qakhan | [TK]D-Fender i got you it is PLC and Jitter but i am try to nerrow down, which end. device side or asterisk or network between both |
| 21:26.49 | [TK]D-Fender | NETWORKING |
| 21:26.56 | [TK]D-Fender | the software isn't magically bad. |
| 21:27.13 | [TK]D-Fender | PACKET LOSS. What do you think that means? Packets did not ARRIVE across your network. |
| 21:27.24 | [TK]D-Fender | JITTER. |
| 21:27.31 | [TK]D-Fender | The packets arrive.. OUT OF ORDER |
| 21:28.29 | qakhan | ok |
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| 22:01.41 | saratogga | ended up just blocking sipcli with iptables directly |
| 22:01.48 | saratogga | fixed the issue, client happy, all good |
| 22:01.53 | saratogga | asterisk 1.8 FTW |
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| 22:32.25 | xochilpili | hello all |
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| 22:44.15 | xochilpili | i have sound issues between 2 * systems |
| 22:54.02 | *** join/#asterisk Draecos (~Draecos@1.127.48.177) |
| 22:55.42 | xochilpili | i make a call from my home's number to my cid number provided by my sip provider, the sip provider report the last 4 digits so, i make a call from 2288175116 [home] to 2375659 [cid] , so in Asterisk_A catch the call and send this to Asterisk_B to the extension: 100 |
| 22:55.54 | xochilpili | here's the debug |
| 22:56.01 | xochilpili | http://pastebin.com/Ei1auMEU << Asterisk_A |
| 22:56.16 | xochilpili | http://pastebin.com/65HMNHY5 < Asterisk_B |
| 22:56.52 | xochilpili | i only have issues with incomming calls, outgoing calls works right, also from extension to extension, works |
| 22:58.45 | xochilpili | what i notice is that rtp set debu on in Asterisk_B sent RTP p2p packet to voice's gateway, dont know why |
| 22:59.40 | xochilpili | in the other hand, when i make a call for outgoing rtp debug said that sent RTP packet to ASterisk_A [which is correct] |
| 23:00.20 | xochilpili | i dont realize why does Asterisk_B when there's an incomming call send the rtp packets to voice gateway instead of Asterisk_A |
| 23:00.21 | xochilpili | ? |
| 23:13.33 | xochilpili | someone? |
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| 23:30.23 | jgowdy | So I'm trying to use SIP_CODEC_INBOUND to control the codec chosen for certain inbound calls. But it seems like Asterisk decides on which codec it's going to use for the call way in advance and thus ignores my Setvar of SIP_CODEC_INBOUND |
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| 23:45.43 | Kobaz | ever get the feeling that a polycom config line item is just stuck and won't change |
| 23:57.14 | xochilpili | any hand? |