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| 01:47.34 | zekoZeko | i'm playing some early media to notify the caller if their operation was successful (without answering the line and burning precious minutes for the user :). The action that gets done is called through System() and takes a few seconds. Is it possible to play early media while this is done? Currently Background(file,noanswer) blocks for me... |
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| 05:18.14 | wowi42 | Good morning |
| 05:19.34 | wowi42 | little question: is that possible, through the ARI, to receive all the events for the entire asterisk ? or do I need to use the AMI ? (for example, to calculate the hold time) |
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| 06:04.41 | jzu_ | do I need seperate VoIP number for fax or can I use my existing number? |
| 06:05.27 | ChannelZ | only if you want a "fax only" number for some reason |
| 06:05.38 | jzu_ | nah, no need for fax only :) |
| 06:05.53 | jzu_ | seems like Localphone = Voxbone = T.38 support, so it should work. |
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| 06:07.00 | jzu_ | by the way, my CID issue was resolved by switching from Sonetel to Localphone \o/ |
| 06:07.13 | jzu_ | I immediately got CID showing up on outgoing calls |
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| 07:29.56 | krefik | does anyone got problems with asterisk in docker (centos 6.6 + asterisk 11.16.0 builds from digium)? |
| 07:30.10 | krefik | I've got regular segfaults (3 to 8 daily) |
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| 07:58.18 | imihaylov | hi |
| 07:58.35 | imihaylov | here i post the sip debug http://pastebin.com/rKG4zwwW |
| 07:59.32 | imihaylov | the problem was that a call was initiated again and again instead of going to voicemail when there is a Ð(call-macro) in the Dial option |
| 08:00.52 | sarvik | I donât undestand the Dialplan syntaks. |
| 08:00.52 | sarvik | Example: exten => _5X.,1,Dial(SIP/${EXTEN}@provider) |
| 08:00.52 | sarvik | Firts number _5X. Is the incomming call. And in parentheses, where comming the ${EXTEN} variable value. |
| 08:02.36 | imihaylov | output from the cli http://pastebin.com/qWi2ygXF |
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| 08:03.07 | imihaylov | here is the dialplan itself: http://pastebin.com/wgZwpTcY |
| 08:05.54 | wdoekes | imihaylov: that SIP log ( http://pastebin.com/rKG4zwwW ) contains 0 calls |
| 08:06.43 | imihaylov | I just rang the phone for over 30 seconds |
| 08:06.51 | imihaylov | and that was the debug info |
| 08:07.03 | imihaylov | (allong my call there were some other) |
| 08:08.20 | wdoekes | imihaylov: those OPTIONS are ping/keepalive packets |
| 08:08.33 | ChannelZ | sarvik: _ means "this is a pattern." It then matches the number 5 followed by anything (X) any number of times (.) ${EXTEN} represents whatever extension was dialed that matched the pattern |
| 08:09.03 | wdoekes | "followed by anything (X) any number of times (.)" <-- no |
| 08:09.19 | wdoekes | X = any digit, . = 1 or more "something" |
| 08:10.24 | imihaylov | thats strange... isn't it supposed to show info from ringing ? As I debugged before it does. |
| 08:10.48 | wdoekes | _X. would match "55", "9abc" and "123", but not "9", "a" or "a9" |
| 08:11.47 | wdoekes | <PROTECTED> |
| 08:11.47 | wdoekes | <PROTECTED> |
| 08:11.47 | wdoekes | <PROTECTED> |
| 08:11.50 | wdoekes | <PROTECTED> |
| 08:11.52 | wdoekes | <PROTECTED> |
| 08:11.53 | wdoekes | <PROTECTED> |
| 08:11.56 | wdoekes | ... |
| 08:12.40 | wdoekes | it's not regex |
| 08:13.20 | wdoekes | "_X." (dialplan) == "^[0-9].*$" (regex) |
| 08:14.32 | wdoekes | imihaylov: poke around some more to get the needed info. if it says "CSeq: .. OPTIONS" or "CSeq: .. NOTIFY", it's not of interest to us right now |
| 08:20.07 | sarvik | OK Thanks |
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| 08:31.10 | sarvik | And when and where this pattern is _X5. used. I've undestand that dialplan extension is registered in sip.conf files. Is it meane that _5X. is registered in sip.con like 5somethink. Example: 56789 |
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| 08:44.58 | wdoekes | I can't parse that last sentence of yours, but _5X. will indeed match 56789 |
| 08:46.48 | sarvik | hmm. OK |
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| 09:32.40 | sarvik | What is wrong in thes sentenence? I read the Asterisc dock (http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-3.html) and try undestand :-) |
| 09:54.23 | sarvik | Example: I was registered SIP phone in provider (123456) and I need to receive incomming call to another phone, like mobile phone (56789). |
| 09:54.24 | sarvik | Is right when i wrote exten dialplan like: |
| 09:54.24 | sarvik | Exten => 123456,1,dial(SIP/56789) |
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| 10:16.45 | mic_ | hello |
| 10:17.15 | zekoZeko | sarvik: you probably want Dial(SIP/provider/56789) if you want this to go out the SIP trunk. |
| 10:17.36 | mic_ | did anyone go through asterisk11 -> asterisk13? |
| 10:18.22 | mic_ | features in asterisk13 are great - I am just thinking whether asterisk11 is more suitable for small-enterprise deployment. |
| 10:20.30 | Chainsaw | I haven't tried 13 yet. I generally wait until X.4.0 before trying a new branch. |
| 10:20.46 | Chainsaw | Since 13 is only on 13.2.0 I will leave it to the early adopters for the moment. |
| 10:25.58 | wdoekes | I can tell you that 13.1 gave me a pjsip module ordering headache; that may have been fixed in 13.2 |
| 10:34.25 | mic_ | so in general |
| 10:34.51 | mic_ | some balanced skepticism is pretty much in place. |
| 10:35.30 | mic_ | or in other words - for serious adoptions -> be conservative and run with asterisk11. |
| 10:36.16 | mic_ | one more option would be to go for 1.8, but its EOL is approaching. |
| 10:54.43 | Chainsaw | 11 is a safe bet. |
| 10:56.13 | mic_ | thanks a lot. |
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| 11:52.43 | elitas | hello, how to increase the size of my history in bash |
| 11:53.40 | zekoZeko | not really an asterisk question, is it? HISTSIZE variable is your friend. |
| 11:54.21 | elitas | in asterisk 13 |
| 11:56.16 | elitas | yeah, ok thanks |
| 11:58.05 | freemanls | elitas: nice joke man |
| 11:58.10 | freemanls | HISTFILESIZE |
| 11:58.15 | freemanls | is what u'r looking 4 |
| 11:59.02 | freemanls | or as zekoZeko suggested could be HISTSIZE :D |
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| 12:01.50 | zekoZeko | it seems he wants a longer command history in asterisk console. |
| 12:03.29 | zekoZeko | don't think bash settings will change it, but i might be wrong. Would have to check readline docs (i guess it uses readline, but am not sure) |
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| 12:06.23 | elitas | yes zekoZeko, i wanna longer command history in asterisk console |
| 12:13.55 | zekoZeko | elitas: don't know how to set in it asterisk console, sorry. quick search also didn't return anythin useful. |
| 12:19.37 | freemanls | http://doxygen.asterisk.org/trunk/d3/d83/readline_8c.html |
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| 14:24.41 | anonymouz666 | master-of-ari has joined asterisk |
| 14:25.28 | file | hrm? |
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| 14:28.16 | anonymouz666 | in the past was master-of-queues |
| 14:31.28 | anonymouz666 | i am really impressed about what whatsapp did. even in poor 3g networks the voip calls seems to be at least reasonable |
| 14:35.23 | coppice | whatsapp does VoIP? |
| 14:35.48 | anonymouz666 | to work in brazil 3g networks it does magic. |
| 14:36.45 | anonymouz666 | some people here still talk about g729... this codec is some years out-of-date |
| 14:37.48 | coppice | when did G.729 become out of date? |
| 14:37.55 | drmessano | <PROTECTED> |
| 14:38.09 | drmessano | checks the expiration date on his ATAs |
| 14:38.15 | anonymouz666 | hehe |
| 14:38.18 | anonymouz666 | nice joke |
| 14:38.38 | anonymouz666 | coppice: when there are better choices to make |
| 14:38.44 | drmessano | Like? |
| 14:39.15 | anonymouz666 | OPUS? SILK? |
| 14:39.44 | anonymouz666 | OPUS is the right way to go |
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| 14:40.31 | coppice | when most phones support OPUS it will be a good choice. today its niche. SILK is dead |
| 14:40.56 | anonymouz666 | coppice: I hope we got this support soon |
| 14:41.00 | coppice | but at 8kbps OPUS is really no better than G,729 |
| 14:41.17 | anonymouz666 | you mean MOS? |
| 14:42.09 | coppice | for anything other than MOS G.729 wins easily on technical grounds. Less memory. Less compute |
| 14:42.59 | coppice | OPUS isn't really about replacing G.729. its about pushing things to wideband |
| 14:43.12 | anonymouz666 | what about the quality in poor networks? |
| 14:43.18 | anonymouz666 | G729 sucks at that |
| 14:43.30 | coppice | OPUS is worse |
| 14:44.44 | Chainsaw | anonymouz666: If your network is terrible enough to make G729 unable to cope... |
| 14:44.48 | coppice | when you talked about VoIP over whatsapp, were you talking about the push to talk stuff? |
| 14:44.50 | Chainsaw | anonymouz666: Why bother running SIP over it at all? |
| 14:45.01 | anonymouz666 | coppice: no, whatsapp can make voip calls. |
| 14:45.09 | anonymouz666 | not ptt. |
| 14:45.17 | WIMPy | Well, with G.729 low packet loss has high impact. |
| 14:45.43 | Chainsaw | WIMPy: Because of high compression, yes. Something's gotta give. |
| 14:46.02 | anonymouz666 | coppice: they are thinking to develop something that could work even in 2G networks. |
| 14:46.05 | WIMPy | Because the packets aren't self-contained. |
| 14:46.20 | anonymouz666 | 3G here in Brazil sucks big time and most of whatsapp call just works. |
| 14:46.21 | Chainsaw | WIMPy: I can still get legible G729 down circuits that wouldn't cope with uLaw/aLaw stutter-free. |
| 14:46.34 | anonymouz666 | ok, it can jitter buffer a little bit but people are getting used to this |
| 14:46.39 | WIMPy | That doesn't make sense. |
| 14:47.15 | anonymouz666 | coppice: could you think about the impact in the big telcos? |
| 14:47.17 | Chainsaw | WIMPy: It's about bandwidth. The 3G network doesn't have very much of it. |
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| 14:47.37 | anonymouz666 | telcos these days don't talk about minutes |
| 14:47.43 | anonymouz666 | they talk about bytes |
| 14:47.45 | coppice | how do you get whatsapp to do VoIP? |
| 14:48.17 | WIMPy | Sure, you're less likely to loose a packet, but you will loose more that one packet due to the CODEC, not due to the network. |
| 14:48.20 | anonymouz666 | coppice: I got you in my contact list and I call will (assuming you have the latest whatsapp) and then will active the voip feature |
| 14:48.41 | drmessano | coppice, they just pushed it out for Android.. IOS to follow. It's new |
| 14:49.11 | anonymouz666 | coppice does not seem to use IOS, heh |
| 14:49.21 | anonymouz666 | drmessano: do you? |
| 14:49.30 | drmessano | I do |
| 14:49.45 | drmessano | I was just being thorough. Doesnt matter to me what anyone uses |
| 14:50.20 | drmessano | I never found much use in WhatsApp. |
| 14:50.34 | drmessano | It seems redundant |
| 14:50.52 | Chainsaw | drmessano: It's very popular here. On the Underground, you get WiFi coverage about every other station. |
| 14:51.17 | anonymouz666 | People seem to know what WhatsApp is, but doesn't know the economy minister name |
| 14:51.18 | Chainsaw | drmessano: But no mobile signal whatsoever for most of your journey. So it wins over SMS. |
| 14:51.26 | coppice | why whatsapp appears to be up to date, and I can't find any VoIP stuff? |
| 14:51.46 | Chainsaw | drmessano: That's just the messaging feature though, can't speak for any voice options. |
| 14:51.53 | coppice | s/why/my |
| 14:51.58 | anonymouz666 | coppice: you have to receive a voip call through whatsapp to enable it |
| 14:52.27 | coppice | that's weird |
| 14:52.28 | drmessano | Most of my friends have iPhones, and we use them exclusively at my workplace. So we all have iMessage. I guess thats part of the redundancy |
| 14:52.49 | anonymouz666 | drmessano: what is great about IOS is the facetime, very straight forward |
| 14:53.10 | anonymouz666 | coppice: that's simple |
| 14:53.50 | drmessano | With iMessage, I can message from multiple devices. Put down my main iPhone, and pick up the one I keep plugged in next to the bed and continue messaging. I hated that WhatsApp could only be signed in on one device |
| 14:54.11 | drmessano | Even Facebook messenger does that better |
| 14:55.10 | coppice | ooh, a little telephone handset has appeared in whatsapp. i'm not sure what triggered it to appear |
| 14:55.36 | drmessano | Granted, iMessage is a single platform closed island, but you would think something as simple as messaging from multiple devices would be a no-brainer |
| 14:55.52 | anonymouz666 | coppice: yeah that's it |
| 14:57.32 | anonymouz666 | coppice: I know you love SIP but they don't use it |
| 14:57.41 | anonymouz666 | heh |
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| 14:59.32 | anonymouz666 | SIP is too complex to make something simple |
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| 15:11.23 | anonymouz666 | coppice: are you having fun with whatsapp calls? |
| 15:13.08 | anonymouz666 | for those still interested in integration with telephone systems and whatsapp, there's a lib python called "yowsup". |
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| 15:17.53 | coppice | I just gave it a quick try talking to my children. huge delays, but it works |
| 15:18.18 | coppice | wifey has an iphone |
| 15:19.45 | anonymouz666 | coppice: what I said |
| 15:19.57 | anonymouz666 | what network did you use? |
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| 15:20.19 | coppice | I used LTE. the children were on the home wifi |
| 15:21.05 | anonymouz666 | good |
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| 18:23.45 | ipengineer | If I try to transfer to an invalid extension in asterisk the transfer fails and the active call stays on the line. This is good.. Is there a CMD in asterisk to kick a call back to the transferer? Say I transfer to exten=>1234, that extension is valid but for some reason I dont want the transfer to go through. Is there a way to cancel the transfer and allow the call to continue? |
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| 18:26.54 | mattsl | I am trying to work with a new provider who wants to verify that our server is online by sending OPTIONS messeges periodically. However, it appears that our server is not responding to them. Anyone know how to make it respond? |
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| 18:27.09 | anonymouz666 | ipengineer atxfer? |
| 18:27.17 | ipengineer | anonymouz666: Blind Transfer |
| 18:28.29 | anonymouz666 | why don't you handle the exten => i,n, and Dial() back to the source |
| 18:29.34 | ipengineer | I guess that is an option.. I just wasnt sure if there was a way to do it where the call never disconnected until the transfer was assured |
| 18:32.12 | ipengineer | anonymouz666: Maybe this will help.. I may have some errors since I just through it together but maybe it will better convey the point. |
| 18:32.12 | ipengineer | http://pastebin.com/WCtbAkDa |
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| 19:35.37 | bobbymc4_ | question about the meetme application, there is a option called G that allowes specifying the intro audio file, im not sure how to set that in the configurations, can anybody help me? thank you |
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| 22:48.36 | braden80 | hi - is anyone successfully using chan_sip and PJSIP in asterisk? I am getting a "Failed to bind to 0.0.0.0:5060: Address already in use" and wondering if one is blocking the other? |
| 22:48.56 | braden80 | trying to register my trunk wth the server. |
| 22:59.06 | smkelly | file: POOF! |
| 22:59.41 | file | smkelly, kaboom |
| 22:59.54 | file | braden80, you can't bind them both to port 5060 |
| 23:00.37 | smkelly | FUN FACT: While Frank Welker is credited with doing the voice for Fred on Scooby Doo, that was actually file. |
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| 23:06.09 | braden80 | figured as much. so if i disable the PJSIP registration the sip registration should go through right |
| 23:07.29 | file | a registration does not define what binds, those are called transports in PJSIP world |
| 23:08.04 | file | if you define no transports then PJSIP won't work, but chan_sip will continue to - the other option is to run each on different ports |
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| 23:29.32 | braden80 | @file: OK so i've reverted back to SIP only, rebooted and sip reloaded. the extensions have registered ok. sip trunk still not working but i'm seeing the following when i sip reload: "'tcp' is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used." |
| 23:29.40 | braden80 | isn't that a PJSIP setting? |
| 23:29.53 | file | that's chan_sip. |
| 23:30.26 | braden80 | where is that setting set? (i'm using freepbx too) |
| 23:30.37 | file | I don't know FreePBX |
| 23:31.07 | braden80 | ok it's somewhere in the sip.conf files |
| 23:31.18 | file | yes, but they'll just get overwritten if you change it |
| 23:31.26 | braden80 | i'll look for the gui. |
| 23:31.39 | braden80 | what should it be set to? tcpeable=yes ? |
| 23:32.25 | file | if it is using TCP, then TCP has to be enabled |
| 23:37.23 | braden80 | k i added tcpenable=yes to the conf |
| 23:37.35 | braden80 | do i also need tlsenable? (now seeing that error) |
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| 23:45.08 | cyberfab007 | what up people |
| 23:45.59 | braden80 | @file any comment? |
| 23:49.25 | cyberfab007 | hey , if I have to differnt carriers that require two differnt dialione ng out patterns, how do I make one fall over to the other |
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