| 00:09.18 | Katty | hi kids |
| 00:11.00 | [TK]D-Fender | Katty: Mew. |
| 00:11.08 | WIMPy | Hi mum. |
| 00:11.22 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 00:12.14 | *** join/#asterisk lukerobi (~lukerobi@rrcs-97-79-163-146.sw.biz.rr.com) |
| 00:15.14 | Katty | hugs [TK]D-Fender |
| 00:15.31 | lukerobi | I want to write a custom extension where a user dials 1234 followed by 3 digits (1234999) and asterisk would dial to an external phone number and punch in that extension once the remote IVR anwsers. I would prefer for the dialer to listen to hold music until after the extension is dialed remotely... anyone have any idea how I might accomplish this? Google has been no help :( |
| 00:15.48 | Katty | WIMPy: i hereby claim the matriarch position in this channel. |
| 00:15.56 | [TK]D-Fender | lukerobi: "core show application dial" <--- all there |
| 00:16.07 | [TK]D-Fender | lukerobi: Music... digits after answer, the works |
| 00:16.15 | WIMPy | Katty: I think you did so a long time ago. |
| 00:16.19 | Katty | this is true. |
| 00:16.22 | [TK]D-Fender | lukerobi: So go make your pattern to match and chop it up |
| 00:16.45 | Katty | nickserv claims i've been here since 05 |
| 00:16.48 | Katty | that seems accurate. |
| 00:16.55 | [TK]D-Fender | WIMPy: that's is questionable claim nowadays.... |
| 00:17.02 | lukerobi | i tried toying with the d option, which really didnt work out... but ill give this a read and see if i can figure it out. Any hints to what i might want to be looking for? |
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| 00:17.18 | [TK]D-Fender | Katty: You certainly win on history... but not quite "seniority" |
| 00:17.23 | [TK]D-Fender | Speaking of whom :) |
| 00:17.27 | WIMPy | Katty: Yes, you've had the position before I was around. Like a real mum :-) |
| 00:17.28 | [TK]D-Fender | </timing> |
| 00:17.52 | [TK]D-Fender | I couldn't have planned that better... |
| 00:17.57 | Katty | [TK]D-Fender: i read that as Senorita-y |
| 00:18.13 | [TK]D-Fender | Katty: Simultaneously funny & accurate! |
| 00:19.03 | Katty | i still claim Matriarch position |
| 00:20.22 | Katty | who else is going to keep you lot in line?! |
| 00:21.01 | WIMPy | Katty: core show application queue |
| 00:21.13 | Katty | i'll queue you in a minute. |
| 00:21.30 | WIMPy | A queue for the queue? |
| 00:21.39 | Katty | a queue for queuing queues. |
| 00:21.46 | Katty | suddenly, star trek. |
| 00:25.20 | carrar | Thats called a call center manager |
| 00:26.18 | lukerobi | [TK]D-Fender: That did help a little... I was able to figure out the D option was dialing too soon. Is there a way to slow it down or delay it? |
| 00:26.46 | WIMPy | Add some commas. |
| 00:27.00 | [TK]D-Fender | lukerobi: p = 250ms |
| 00:27.05 | [TK]D-Fender | not commas |
| 00:27.11 | lukerobi | ah |
| 00:27.27 | WIMPy | Right. they are hard to write in applications. |
| 00:27.28 | [TK]D-Fender | p=pause |
| 00:27.29 | lukerobi | so like D(p100) ? |
| 00:27.35 | [TK]D-Fender | yes |
| 00:27.39 | lukerobi | :D |
| 00:27.48 | [TK]D-Fender | I'd recommend maybe 3-4 to be sure |
| 00:27.52 | Katty | carrar: hello cupcake! |
| 00:27.54 | [TK]D-Fender | ppp100 |
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| 00:28.49 | carrar | w00t!! |
| 00:28.52 | carrar | Hi Katty! |
| 00:28.57 | Katty | how are you dear |
| 00:34.47 | lukerobi | [TK]D-Fender: Hmm... p seems to have no effect at all? I used 3,4, and then 10 of them and the result is the same. :/ |
| 00:36.30 | lukerobi | ah... i missed the error. Illegal DTMF character 'p' in string. (0-9*#aAbBcCdD allowed) |
| 00:38.14 | [TK]D-Fender | ah, thought it should pause like all the other methods... |
| 00:38.23 | lukerobi | maybe w? lemme try that |
| 00:38.38 | [TK]D-Fender | lukerobi: us a macro or gosub in there, and call Wait() to pause and SendDTMF() to do the send |
| 00:38.51 | [TK]D-Fender | wait.. it was w, not p |
| 00:38.52 | [TK]D-Fender | my bad |
| 00:39.01 | [TK]D-Fender | though I'm still not sure if D() allows it. |
| 00:39.02 | lukerobi | haha |
| 00:39.11 | [TK]D-Fender | (w)ait |
| 00:43.12 | lukerobi | success... |
| 00:43.13 | lukerobi | ! |
| 00:43.41 | [TK]D-Fender | even better |
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| 00:45.31 | lukerobi | ill figure out the hold music for the caller tomorrow... been here nearly 12 hours :/ I tried using playback, which left me listening to elevator music for a few minutes. |
| 00:45.31 | Katty | likes success |
| 00:45.57 | carrar | Did you make those creamecheese filled chocolate cupcakes? |
| 00:46.17 | carrar | Those are the YUM |
| 00:46.36 | Katty | they're on my list, but the boyfriend requested lemon merangue |
| 00:46.41 | Katty | and i can't tell him no |
| 00:46.45 | [TK]D-Fender | lukerobi: Hold music is annother dial option |
| 00:46.54 | [TK]D-Fender | lukerobi: as I said, it's all there |
| 00:47.22 | dorphalsig | hello, I'm having a weird iax error. I have two different IAX trunks to the same server (and different contexts), both of them are authenticated by ip address. However when I dial the other end seems to get mixed up and recieves the calls through the other trunk. |
| 00:47.47 | dorphalsig | Hrmmm that didnt come out so well |
| 00:47.51 | dorphalsig | I have this: http://pastebin.ca/2434977 |
| 00:48.03 | WIMPy | You know that you can specify the context from remote? |
| 00:48.10 | dorphalsig | How? |
| 00:48.22 | dorphalsig | Dial(IAX2/connextion/extension@context) ??? |
| 00:48.42 | WIMPy | Looks sensible. |
| 00:48.52 | Katty | nothing is sensible around here. |
| 00:49.01 | WIMPy | YOu must not specify the context in the peer definition. |
| 00:49.03 | dorphalsig | Im sensitive if that counts... |
| 00:49.15 | WIMPy | There is quite a lot of truth there. |
| 00:49.17 | Katty | sensitive always counts. |
| 00:49.17 | dorphalsig | I'm defining them as friends |
| 00:49.27 | Katty | aww, being friends. how sensitive. |
| 00:49.30 | Katty | ^_^ |
| 00:49.39 | lukerobi | I'm guessing its the r option (tone) but ill mess with it tomorrow... |
| 00:49.53 | dorphalsig | Now that we are friends, how about helping a friend in distress :P |
| 00:50.07 | Penguin | lukerobi: Dial's option r? That is for a ringing sound. |
| 00:50.31 | Katty | hi Penguin |
| 00:50.36 | Penguin | Hello! |
| 00:50.52 | Katty | how're you dear |
| 00:51.01 | lukerobi | penguin: i was trying to play hold music to the caller until the bridge |
| 00:51.03 | dorphalsig | WIMPy --> please take a look at the pastebin. I think thats is more clear than my babbling |
| 00:51.14 | Penguin | I'm feeling rather typical right now. |
| 00:51.31 | Katty | well...typical is better than downright awful, yes? |
| 00:51.56 | dorphalsig | Katty, you're in a good moood today. |
| 00:52.09 | Katty | pats dorphalsig |
| 00:52.11 | Penguin | Considering all circumstances, yes. |
| 00:52.13 | Katty | dorphalsig: i'm like this every day. |
| 00:52.22 | Katty | Penguin: well i'm happy to hear you're not downright awful! |
| 00:52.34 | Penguin | And I'm happy that you're happy. |
| 00:53.12 | Katty | next come rainbows and sunshine? and barfing unicorns? |
| 00:53.28 | igcewieling | last week I was downright awful, but this week is a bit better. |
| 00:53.30 | dorphalsig | No no. just white rabbits coming out of the bushes |
| 00:53.31 | Penguin | lukerobi: I would have to know more about the entire scenario to know what you're doing. |
| 00:53.48 | Katty | Penguin: lukerobi is going home and drinking beer after working a 12 hour day |
| 00:53.57 | Katty | Penguin: and that's all that really matters at this point :P |
| 00:54.03 | Katty | igcewieling: oh? what happened? |
| 00:54.34 | Penguin | lukerobi: But if someone is calling in and getting the digital receptionist, you can start playing music when the caller dials any extension. |
| 00:54.45 | igcewieling | Katty: had to euthanize my cat |
| 00:54.52 | Katty | igcewieling: :< :< |
| 00:54.58 | Katty | igcewieling: did you have a proper burial service? |
| 00:55.30 | igcewieling | heh, no. once he is dead what is left is just meat. |
| 00:55.38 | igcewieling | I let the vet "dispose of the body" |
| 00:55.44 | Katty | nods sadly |
| 00:56.12 | igcewieling | How have you been? |
| 00:56.36 | Penguin | Meat? I think in Thailand, they call that "supper." |
| 00:56.53 | Katty | igcewieling: not too shabby, i suppose. could be worse. could be better. |
| 00:56.58 | igcewieling | Penguin: hush you |
| 00:57.01 | dorphalsig | I thought everywhere else that was 'sausage' |
| 00:57.20 | dorphalsig | Anyway. I'm outta here. |
| 00:57.20 | Katty | it's too soon to talk about igcewieling's cat, gentlemen. |
| 00:57.34 | igcewieling | bbiaw |
| 00:57.48 | Penguin | week? |
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| 03:57.56 | LonghornWork | were there any known issues with asterisk regarding sip 400 errors on version ~1.6? |
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| 04:55.46 | [TK]D-Fender | go search the tracker |
| 04:56.00 | [TK]D-Fender | All 1.6 branches were EOL a long while ago |
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| 05:30.50 | _abc_ | Good morning (here). I asked yesterday several times, does anyone know if attendant consoles CP-7914 and 7915 have their own MAC and/or IP on an ethernet when connected to a CP-7960 etc phone? |
| 05:30.58 | _abc_ | (I asked in #cisco) |
| 05:31.01 | _abc_ | Does anyone know? |
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| 05:34.20 | kaldemar | doubt it. they don't even have an ethernet interface, do they? |
| 05:45.33 | _abc_ | They plug into a 7960 or 7970 iirc with a RJ connector. I don't know if it's ethernet or not. I have a network problem and a nonworking 7914, and possibly an unidentified MAC floating on the ehternet. |
| 05:46.17 | _abc_ | Also I found one asterisk reference where a 7914 was declared addon to a 7960 and also had its own [SEPxxx] entry and config in sccp.conf |
| 05:46.59 | _abc_ | Which raises some legitimate questions, I think, kaldemar? |
| 05:48.16 | _abc_ | Anyway, is there a known working config for sccp.conf for asterisk, for CP-7960+CP-7914 addon? Default commented out lines in sccp.conf indicate that such a config must exist. |
| 05:48.26 | _abc_ | But I can't find one. I am using chan_sccp2 |
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| 06:52.14 | _abc_ | http://article.gmane.org/gmane.comp.voip.cisco/21661/match=cp+7914+cisco weird one |
| 06:52.24 | _abc_ | so it does NOT work with 7960?! |
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| 06:54.11 | ChannelZ | All hail Crisco! |
| 06:54.56 | _abc_ | and THEN you get the pdf from cisco.com on 7914, which is discontinued, which shows a nice pic of a 7960 working with TWO 7914s |
| 06:55.15 | _abc_ | All hail the heap of crap formerly known as internet collective brain |
| 06:55.48 | _abc_ | Seriously, since everyone is online, the average IQ of material online is the average IQ of the average slum gutter |
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| 07:42.50 | _abc_ | http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/60269/match=addon+7914+asterisk+sccp+conf this config seems wrong. How can it by type=7914 and addon=1 ? |
| 07:43.02 | _abc_ | I think the current syntax is not like that at all?! |
| 07:43.26 | HzD_Byte | hello. I have a question: how to configure Asterisk/FreePBX to notify originating side of a call (trunk) about new line number on transfer? |
| 07:43.31 | _abc_ | I just read the source, I am seriously confused. 7914 does not have a MAC address, how can it have a [SEPxxxx] header? |
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| 07:46.11 | HzD_Byte | does anybody know the solution? |
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| 07:48.11 | _abc_ | Wow I went and read the source of chan_sccp. The post in the link above on gmane is wrong, do not use. Addon must be like 'addon = 7914' (can be 7914, 15, 16). If more than one is used, use more than one addon= line in the config section one after the other. |
| 07:48.38 | _abc_ | To get the buttons assigned to the right one, there should be a way to sort the speeddial entries but I can't see how it is done |
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| 07:59.31 | tea_ | Hi, on Asterisk 11.5 my driver channel chan_sip is XXXed out , " Session Initiation Protocol (SIP) Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) where can i read which dependencies i need for it? |
| 08:00.08 | _abc_ | And these is a way to sort them, the speeddial position parameter in speeddial = , the 6th parameter |
| 08:00.29 | _abc_ | Okay done here for now. Still, if someone has a working 7914 config, please show it? |
| 08:00.32 | _abc_ | sccp onl |
| 08:00.32 | _abc_ | y |
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| 08:09.53 | tea_ | well , the libssl-dev was missing :) |
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| 08:46.30 | aberrios | got a funny issue in the UK with EE(orange/t-mobile), when ringing via DAHDI *sometimes* ringing is detected some times not. Landlines, 3 and Vodafone always detect ringing. |
| 08:48.07 | aberrios | can anyone confirm this? to replicate ring a EE(orange) mobile via a dahdi (isdn30) trunk. Should get DAHDI call is proceeding message, then sometimes a DAHDI is ringing message in verbose cli output |
| 08:48.31 | aberrios | half the time it works half the time it doesn't. |
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| 08:50.30 | lkthomas | hey guys, is it possible to display unknown caller ID? |
| 08:52.21 | aberrios | lkthomas: depends on the carrier and connection to the carrier afaik. |
| 08:52.43 | lkthomas | hmm ? |
| 08:53.41 | aberrios | lkthomas: Are you meaning an incoming call from the PSTN? |
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| 08:57.36 | lkthomas | yes |
| 08:57.49 | aberrios | lkthomas: what type of lines do you have? |
| 08:58.06 | lkthomas | probably T1 |
| 08:59.21 | aberrios | lkthomas: afaik if the carrier on T1 isn't sending the cli there's nothing you can do, if you were using ss7 to connect to the carrier you may be able to do it. |
| 09:00.24 | lkthomas | ss7 is not a general connection here |
| 09:01.44 | aberrios | lkthomas: then I'm pretty sure the answer is no. HOwever try contacting the carrier to see if they will override the setting on the connection to display cli no matter what |
| 09:02.20 | lkthomas | I doubt it |
| 09:03.08 | aberrios | lkthomas: some SIP providers in the UK display cli no matter what, you could try finding a provider that does the same. |
| 09:03.29 | lkthomas | actually if I try to get a SIP number, would it work ? |
| 09:03.46 | aberrios | lkthomas: depends on the provider. |
| 09:03.52 | aberrios | lkthomas: ask around |
| 09:03.58 | lkthomas | I am thinking skype :P |
| 09:04.27 | aberrios | lkthomas: dont know about skype |
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| 09:09.36 | KOPRajs | hi, how do I play a file to multiple SIP clients from command line? |
| 09:09.52 | KOPRajs | I can make a call do those devices using Page |
| 09:10.24 | KOPRajs | and I can play a file to one device from console using originate and Playback |
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| 09:10.47 | KOPRajs | but how can I call Page and then Playback from console |
| 09:14.54 | KOPRajs | or can I use originate and call an extension instead of device |
| 09:14.55 | KOPRajs | ? |
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| 09:36.50 | NET||abuse | hi folks, i've been asked to spend tonight, tomorrow and friday night learning what i can about asterisk features, taking on a little side project for a favour i owe. what's my best bet for learning what i can do in code with an asterisk box? Need to learn how to pass commands and parameters to the api's asterisk exposes to automate something ( I have a feeling they have some high level things they want |
| 09:36.53 | NET||abuse | me to answer over the weekend) |
| 09:38.27 | NET||abuse | So i don't actually know what, but i have some basic points i want to hit on, I want to setup a box, generate a bunch of fake in and out bound calls across 2 or or 3 trunks(simulate multiple outgoing lines) |
| 09:39.15 | NET||abuse | also to create some kind of voicemail automation fun, like deliver a web interface to visualise mailboxes ( I imagine there's actually something like this out there already, but just to learn the api) |
| 09:40.43 | NET||abuse | if i wantto install this in a vm and do this learning, what's the starting points, which linux distro, which packaged version of asterisk should i use, what is a 3 nights worth of reading/practical doing resource I could get started on? |
| 09:41.38 | NET||abuse | ooh, would be cool to be able to generate calls to internal lines and playback a wav/mp3 or something like a voice reminder service |
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| 10:28.50 | HzD_Byte | i've found the solution |
| 10:38.21 | Erwon | NET||abuse just grab the asterisk book (or pdf) and try stuff from there, just read the chapters you think you might need (book link is http://cdn.oreilly.com/books/9780596510480.pdf for example) |
| 10:38.44 | bulkorok | are there any asterisk gui's with an additional user-interface?! |
| 10:39.05 | bulkorok | I see many "admin" gui's but no user-gui'S... |
| 10:41.21 | pazifi | when i use to call "Dial(SIP/account@proxy)" with an peer from DB (sips) it loose the NAT=yes information and asterisk (1.6.1) send the RTP-stream to the private-ip instead to the public-ip.... any ideas? :( |
| 10:41.43 | pazifi | so, if i change the proxy. |
| 10:44.39 | pazifi | bulkorok: I think that there is not... but i don't know it. |
| 10:45.25 | bulkorok | pazifi: I'm afraid of that too... |
| 10:49.41 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 11:21.57 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 11:24.53 | *** join/#asterisk duchman (~paulo@41.203.69.1) |
| 11:25.55 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 11:26.03 | WIMPy | lkthomas: Ask for CLIRO or CLIR override. |
| 11:26.49 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 11:30.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:32.02 | pazifi | ..or is this a asterisk 1.6-problem? |
| 11:32.44 | WIMPy | ~upgrade asterisk |
| 11:32.44 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 11:33.25 | pazifi | yes... but i can't it is a larger environment |
| 11:36.17 | pazifi | 12 servers with asterisk... |
| 11:38.10 | pazifi | 16, with acc |
| 11:39.41 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:40.06 | WIMPy | And you don;t want to provide good service to your users? |
| 11:40.32 | pazifi | ... |
| 11:41.37 | nam3l3zz | bonjour everyone |
| 11:41.57 | pazifi | a bit ignorant response. |
| 11:43.30 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 12:26.37 | nam3l3zz | how to distinct rtp packets in iptables ? |
| 12:37.17 | *** join/#asterisk saxa (~saxa@201-50-103-98.user.veloxzone.com.br) |
| 12:37.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:44.17 | Greenlight | nam3l3zz: UDP to your RTP ports? |
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| 12:53.37 | nam3l3zz | Greenlight: UDP are more general, i have few udp using apps on the same unit, so i would like iptables to be able to distinct rtp from all of the udp ones, i read about nf_conntrack_sip, is there anything else ? |
| 12:54.36 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 12:54.38 | Greenlight | I dind't just say UDP. I said UDP to your RTP ports. |
| 12:55.51 | Greenlight | You define specifically which ports to use for rtp in rtp.conf |
| 12:56.20 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:57.20 | joram | Hello, I have a question about my Asterisk configuration: I want to change the outgoing display number. So if I forward an incoming call from a customer to an employees cell phone the employee sees the number of the customer. I've tried "Set(CALLERID(num)=xxx) and SipAddHeader(...) but nothing works. Any suggestions? |
| 12:57.42 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:57.55 | Greenlight | Your SIP provider may not allow you to present *any* number. |
| 12:58.04 | Greenlight | Check with them that they allow it and will support it. |
| 12:58.14 | *** join/#asterisk [Outcast] (~anonymous@50-200-130-22-static.hfc.comcastbusiness.net) |
| 13:00.03 | joram | Greenlight: I've asked it and they say it is possible, I have four options: static, danymic in FROM HEADER, dynamic in REMOTE-PARTY-ID HEADER, dynamic in P-ASSERTED or P-PREFFERED-IDENTITY HEADER |
| 13:00.41 | [TK]D-Fender | joram: then your peer isn't right |
| 13:00.50 | Greenlight | Ok, so make a normal call and try Set(CALLERID(num)) *before* the Dial |
| 13:01.02 | [TK]D-Fender | joram: You need : sendrpid=yes , trustrpid=yes |
| 13:01.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:01.39 | joram | what option do i have to chose? dynamic in FROM HEADER? |
| 13:02.12 | Greenlight | dynamic in REMOTE-PARTY-ID HEADER I'd think |
| 13:03.28 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:04.40 | joram | ok, sendrpid=yes and trusrpid=yes, got the set caller id. I'l give it a try |
| 13:06.08 | *** join/#asterisk pi3r (~pi3r@86.188.229.227) |
| 13:06.27 | joram | nope, number displayed on the cell phone is still my number from the provider |
| 13:07.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:07.14 | Greenlight | Lets see your dialplan just to check |
| 13:07.25 | Greenlight | ~pb |
| 13:07.26 | 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. |
| 13:08.21 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2cc1:f390:a2d:7be1) |
| 13:08.52 | [TK]D-Fender | joram: pastebin your peer masking ONLY the password and the call output at verbose 10, sip debug enabled |
| 13:09.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zrxrjuxlitzcbmfn) |
| 13:09.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:10.39 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 13:11.06 | joram | http://pastebin.com/4pEKGrDN |
| 13:11.55 | *** join/#asterisk troyt (~troyt@2001:1938:240:2000::3) |
| 13:12.59 | joram | [TK]D-Fender: how to change the verbose level? |
| 13:13.16 | [TK]D-Fender | "core set verbose 10 |
| 13:13.24 | *** join/#asterisk sidus (~abracadab@37-5-74-93-dynip.superkabel.de) |
| 13:13.40 | [TK]D-Fender | joram: and that isn't call debug OR your peer either |
| 13:13.48 | [TK]D-Fender | joram: That was not at all what I requested |
| 13:16.13 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 13:18.33 | *** join/#asterisk bulkorok (~chatzilla@85.183.61.47) |
| 13:20.13 | joram | [TK]D-Fender: call log http://pastebin.com/8VP2aFQz |
| 13:21.15 | [TK]D-Fender | Remote-Party-ID: "0681337xxx" <sip:0857xxx178@141.136.124.2>;party=calling;privacy=off;screen=no |
| 13:21.27 | [TK]D-Fender | From: "0681337xxx" <sip:0681337xxx@89.184.172.55:5080>;tag=as5f32833e |
| 13:21.43 | [TK]D-Fender | rpid matches the FROM CID |
| 13:21.52 | [TK]D-Fender | So you have done what they said they support |
| 13:21.57 | [TK]D-Fender | so your side it right |
| 13:22.00 | [TK]D-Fender | is* |
| 13:22.03 | joram | ok |
| 13:22.17 | *** join/#asterisk serafie (~erin@nat/digium/x-jjoazdzyuutdenxb) |
| 13:22.28 | [TK]D-Fender | go show the debug to your provider and ask them why they don't accept it like they say they should |
| 13:22.47 | joram | ok |
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| 13:28.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:36.31 | Katty | morning |
| 13:36.40 | [TK]D-Fender | <PROTECTED> |
| 13:36.58 | [TK]D-Fender | joram: Wrong line before... sorry, matches the callerID you set in the dialplan prior |
| 13:37.10 | *** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131) |
| 13:37.24 | Katty | hows things? |
| 13:37.32 | [TK]D-Fender | joram: or.. hmm.. I'm not 100% sure on the read of that... let your provider look it over... |
| 13:37.54 | [TK]D-Fender | Katty: Am alive(-ish) |
| 13:37.55 | joram | ok, i've send an email with the full log |
| 13:38.02 | aurs | what is the secret recipe for "softphones - zyxel usg 300 - internets - asterisk" to make it work without having to restart the zyxel router every X days? |
| 13:38.06 | Katty | [TK]D-Fender: well that's better than dead! |
| 13:38.11 | Katty | [TK]D-Fender: are you still caffeinating |
| 13:38.28 | [TK]D-Fender | aurs: Get a new router |
| 13:38.53 | aurs | [TK]D-Fender, that is my gut feeling too! (but my customers won't listen) |
| 13:39.02 | [TK]D-Fender | aurs: That is what seems to have the issue since restarting it appears to be a requirement |
| 13:39.09 | aurs | I always cringe when they say they're using zyxel :D |
| 13:39.32 | aurs | there might be some secret configuration that the Internet haven't heard about, but |
| 13:39.56 | AllanButton | aurs: put it on a christmas tree timer to restart every day ;) |
| 13:40.03 | Greenlight | Try turning off any fancy features, and grab the latest firmware. |
| 13:41.43 | aurs | they refuse restarting it, because they're hosting some other services on the same connection |
| 13:42.39 | aurs | so I don't KNOW that a restart will fix it, but it has happened before ;) |
| 13:42.43 | joram | I have another question: I have two numbers from my provider but only one trunk. How can I know what number the customer has dialed when a call comes in? |
| 13:42.59 | Greenlight | joram: By matching it in your dialplan. |
| 13:43.18 | Greenlight | exten => <NUMBER1>,1,.... |
| 13:43.26 | Greenlight | exten => <NUMBER2>,1,..... |
| 13:44.37 | joram | Greenlight: but both numbers seem to come on one extension |
| 13:45.22 | Greenlight | So change it. |
| 13:45.58 | joram | Greenlight: my configuration is: register => username:password@vcproxy.voiceworks.nl |
| 13:45.58 | joram | is that good? |
| 13:46.08 | Greenlight | Looks okay. |
| 13:46.19 | Greenlight | Lets see the context that calls are hitting when they come in |
| 13:48.43 | [TK]D-Fender | joram: some providers don't sent the DID in the INVITE line at the start of the packet and instead use the "To:" header at which point you have to process that in your dialplan |
| 13:50.17 | *** join/#asterisk [J_M_C] (~J_M_C]@unaffiliated/j-m-c/x-8000295) |
| 13:50.43 | joram | my incomming configuration is: exten => 0857600177,1,Playback(tt-weasels) and exten => 0857600178,1,Playback(tt-monkeys) do not work, when i call i receive a "number is not in service" message |
| 13:50.53 | [J_M_C] | GS UCM6100 webinar in about 10 minutes |
| 13:52.33 | joram | I can change some DID settings from my provider. "to header" on/off, "uri header" on/off, "CALLERID IN DISPLAYNAME HEADER" on/off, "length" 10/9 numbers |
| 13:53.58 | [TK]D-Fender | look at the actual call |
| 13:54.10 | [TK]D-Fender | Forget what you hear as the caller |
| 13:54.15 | [TK]D-Fender | pay asttention to the debug |
| 13:54.29 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 13:56.59 | nam3l3zz | how to have Read app not wait for the # key, just read every entered dtmf within allocated time frame (i.e. 3 seconds) ? |
| 13:57.11 | nam3l3zz | *asterisk 1.8 |
| 13:57.39 | jmetro | what you mean like waitexten? |
| 13:57.53 | jmetro | auto attendant kinda? |
| 13:58.35 | WIMPy | nam3l3zz: You did see the timeout parameter? |
| 13:58.58 | nam3l3zz | WIMPy: i did, but it is ignoring it somehow, i'll pastebin, one sec |
| 13:59.25 | joram | [TK]D-Fender: In the debug I only see the dialed number in the To field |
| 13:59.31 | WIMPy | works for me (tm) |
| 14:00.02 | [TK]D-Fender | joram: that doesn't sound like "message played to caller" or "call wan't accepted" yet |
| 14:00.05 | nam3l3zz | WIMPy: one line, no need for pastebin, here she is : exten => s,n,Read(dst_num,,,,3) |
| 14:00.26 | [TK]D-Fender | joram: and the fact you see it in the "To:"confirms what I mentioned earlier |
| 14:00.59 | WIMPy | nam3l3zz: one comma too little. |
| 14:01.19 | nam3l3zz | tried with 5 commas, will retry again |
| 14:02.14 | WIMPy | Maybe you want to specify the number of digits to read? |
| 14:02.22 | pi3r | Hi guys, is there a way to record call as mp3 instead of wav? |
| 14:02.57 | WIMPy | pi3r: no |
| 14:03.13 | nam3l3zz | WIMPy: from 7~8 to 15 , an international number excluding preffix (00 / +) |
| 14:03.37 | pi3r | WIMPy: should I run a script to do it with lame or somerthing? |
| 14:03.55 | kaldemar | pi3r: you could use the command option for MixMonitor to convert recordings after they are finished. |
| 14:04.04 | Greenlight | Record as wav, convert to mp3 overnight. |
| 14:04.21 | [TK]D-Fender | or immediately after it's done |
| 14:04.22 | Greenlight | You *can* convert to mp3 after the call completes, but it can cause CPU spikes |
| 14:04.22 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 14:04.25 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 14:04.28 | WIMPy | nam3l3zz: I'm not sure I tried that without maxdigits. WaitExten would be another option to use. |
| 14:04.30 | Greenlight | Depends on server load and how many users |
| 14:04.49 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 14:04.56 | pi3r | kaldemar: Greenlight: Thanks guys |
| 14:05.00 | [TK]D-Fender | CPU spike is probably pretty low overall unless you have a lot of calls ending at the same time |
| 14:05.08 | [TK]D-Fender | real-time encoding would be much greater |
| 14:05.14 | joram | [TK]D-Fender: I've enabled some DID options from the provider and now i can extract the number from the TO field |
| 14:05.22 | pi3r | [TK]D-Fender: it's for a POC, not a production server |
| 14:05.49 | Greenlight | I agree, but with say 150 users you risk say 25 calls just happen to end at once, and cause CPU load for a minute |
| 14:06.21 | Greenlight | I feel safer keeping as wav till that evening, and the database just references the wavs and gets updated as well, so users don't notice |
| 14:06.27 | [TK]D-Fender | Greenlight: at which point ... queue them |
| 14:06.51 | [TK]D-Fender | Greenlight: easy to mitigate |
| 14:06.56 | pi3r | Greenlight: the thing is, I need to upload the file immediatly after the file |
| 14:07.00 | pi3r | s/file/call/ |
| 14:07.13 | Greenlight | That case just do it on the fly :) |
| 14:07.17 | pi3r | arf |
| 14:07.27 | pi3r | Greenlight: so with lame or somethinbg? |
| 14:07.28 | [TK]D-Fender | pi3r: the do it right after |
| 14:07.33 | [TK]D-Fender | pi3r: yes |
| 14:07.39 | Greenlight | [TK]D-Fender: We did look at having then queued to be convered and then process strictly X at a time |
| 14:07.43 | pi3r | [TK]D-Fender: Greenlight: cheers guys |
| 14:07.49 | Greenlight | Yes, with lame |
| 14:08.21 | pi3r | Do there is a example for Asterisk? Right now I'm doing it from the adhearsion ruby server |
| 14:08.21 | Greenlight | Also, you should "nice" it. |
| 14:08.57 | Greenlight | Look at MIXMONPOST |
| 14:09.04 | [TK]D-Fender | pi3r: lame is lame. There is no asterisk example for running lame |
| 14:09.18 | tuxx- | thats so lame [TK]D-Fender |
| 14:09.21 | tuxx- | ^_^ |
| 14:09.22 | [TK]D-Fender | pi3r: "core show application monitor" <- |
| 14:09.22 | pi3r | ^^ |
| 14:09.56 | [TK]D-Fender | pi3r: read the instructions for calling a script post-recording |
| 14:10.22 | pi3r | [TK]D-Fender: on it! Thanks! |
| 14:10.40 | Greenlight | I could give you the syntax that we use for lame but it's horrible. |
| 14:10.46 | nam3l3zz | WIMPy: is the delay for one digit to be read or for the whole lot ? |
| 14:11.15 | Greenlight | nice /usr/local/bin/lame -b 16 -m m -q 8 --resample 8 "$i" "${i%.wav}.mp3" |
| 14:12.11 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-lanvxygcntjuomzc) |
| 14:12.12 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:12.28 | WIMPy | nam3l3zz: There are timeouts for both. |
| 14:15.21 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.238.132) |
| 14:17.21 | pi3r | Thanks Greenlight |
| 14:20.57 | *** join/#asterisk GOrozco (~GOrozco@189.169.102.60) |
| 14:21.06 | GOrozco | hi |
| 14:22.09 | nam3l3zz | strange thing timeout 3 , gets " User entered nothing." and moves on after around 3 seconds, timeout 4 reads an 11 digit number and moves on only after 14 seconds, seems that timeout 3,5 cannot be assigned :) |
| 14:22.27 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:22.27 | GOrozco | hope anyone can help me... our Asterisk server is not sending sound to external calls... it started yesterday. we can receive calls and the other party can hear, but when we call, the other party rings, answer but no sound either way |
| 14:23.05 | nam3l3zz | check firewall, if rtp packets are being dropped/blocked |
| 14:23.10 | GOrozco | we have a TDM410P card |
| 14:23.26 | GOrozco | it is only when we call outside using dahdi |
| 14:23.40 | GOrozco | I can connect through SIP without issues |
| 14:23.47 | GOrozco | and call other extensions |
| 14:24.17 | *** join/#asterisk navaismo (~navaismo@189.241.68.101) |
| 14:24.24 | nam3l3zz | GOrozco: u will need to ask others, i am new enough to asterisk |
| 14:25.52 | GOrozco | thanks nam3l3zz for trying to help me :-) |
| 14:26.50 | GOrozco | as i said there were no changes to our server and suddenly it started not transferring sound... I was suspecting defective hardware but I can call from outside and hear the IVR |
| 14:26.52 | WIMPy | GOrozco: Have you tried turning it off an on again? |
| 14:27.30 | GOrozco | thanks WIMPy, I tried that, then also while I do not like that, tried restarting the server... no joy, its doing the same |
| 14:27.41 | newtonr | GOrozco, read up on what the various fields in SDP mean, get a pcap look at your SIP dialog, check out the SDP in the invites, and pay attention to the RTP streams |
| 14:27.59 | GOrozco | when I dial outside, my cel number rings, but then no sound either way |
| 14:28.06 | newtonr | GOrozco, then you can come point out specific oddities to people here and they can point to specific Asterisk settings that may relate |
| 14:29.00 | GOrozco | thanks newtonr! sip calls between internal and external work.. it is only when dialing to the PSTN when there is no sound |
| 14:29.12 | *** join/#asterisk SGjunior (~sgjunior@out-on-228.wireless.telus.com) |
| 14:29.46 | igcewieling | GOrozco: You have an older Digium card? If so, reboot. |
| 14:30.02 | GOrozco | yes it is an old card... but I did reboot |
| 14:30.22 | newtonr | GOrozco, ahh i just now saw that you are using an analog PSTN card :) |
| 14:30.36 | WIMPy | Rebooting is not the same as turning it off and on again. |
| 14:30.47 | igcewieling | GOrozco: a reboot has solve that issue for me every single time when using an older Digium card. |
| 14:31.40 | newtonr | GOrozco, If you are unable to figure it out - Digium provides free support on those cards: http://www.digium.com/en/support/contact |
| 14:31.55 | WIMPy | Just like a reboot is not the same as physically pressing the reset key. |
| 14:32.52 | GOrozco | I am rebooting again... hope I do not need to go there, its three hours by bus (remote office) |
| 14:32.56 | GOrozco | [Aug 21 09:29:46] VERBOSE[28656][C-00000008] app_dial.c: -- DAHDI/2-1 answered SIP/190-0000000e |
| 14:33.59 | GOrozco | oh... this is what it says when I dial outside. it says answered, then nothing |
| 14:38.30 | *** join/#asterisk powerunits (b6b2bb80@gateway/web/freenode/ip.182.178.187.128) |
| 14:39.00 | newtonr | GOrozco, try to split your call path in half to see which side has the problem. For example get rid of the SIP side for testing, Originate a call over the card using dialplan/originate/callfiles, etc maybe play a sound file and then record. |
| 14:41.04 | newtonr | Right now you are doing PSTN <-> CARD <-> DAHDI <-> ASTERISK <-> SIP <-> SIPPHONE , try to cut it down to just PSTN <-> CARD <-> DAHDI <-> ASTERISK |
| 14:41.27 | *** join/#asterisk felipealmeida (~user@177.41.21.40) |
| 14:42.36 | newtonr | GOrozco, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files , https://wiki.asterisk.org/wiki/display/AST/Application_Originate |
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| 14:46.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:48.42 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 14:52.53 | igcewieling | You could just call into the port and have asterisk play a sound file. |
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| 14:59.09 | newtonr | igcewieling, he said inbound calls over the card always had two way audio. |
| 14:59.21 | newtonr | but he should probably still try that too |
| 15:00.15 | GOrozco | hi... I was trying on the command line :-) |
| 15:00.44 | GOrozco | I am now trying using SIP to my own Asterisk, then dialing using PSTN to the other Asterisk. I can hear the IVR |
| 15:00.53 | [TK]D-Fender | No, he say "internal" which means nothing |
| 15:01.00 | [TK]D-Fender | And I don't trust the SIP end at all |
| 15:01.15 | [TK]D-Fender | And we are getting only vague terms, not details, and no debug |
| 15:03.45 | igcewieling | *nod* sounds like a SIP issue to me. |
| 15:07.52 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
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| 15:11.03 | GOrozco | well I dialed from PSTN and was answered from a SIP telephone inside, and we had a chat |
| 15:11.23 | GOrozco | so incoming calls work correctly without issues |
| 15:11.47 | GOrozco | it is just outgoing calls to PSTN which get connected but no sound both ways |
| 15:12.49 | GOrozco | I can paste the log of an outgoing call to a pastebin.. |
| 15:15.00 | igcewieling | sounds like a SIP issue to me |
| 15:15.23 | *** part/#asterisk orn (~orn@2a01:8280:101:0:193a:e99f:30ef:9801) |
| 15:16.24 | *** join/#asterisk GOrozco (~GOrozco@189.169.102.60) |
| 15:16.36 | GOrozco | closed the wrong window... sorry |
| 15:16.56 | GOrozco | we are shuting down the server to get the serial number of the digium card |
| 15:18.01 | igcewieling | GOrozco: yes, pastebin a failed call without sip debug and pastebin a failed call with sip debug |
| 15:18.15 | AllanButton | GOrozco: sorry, I am joining this late, but you say it an analog card and inbound works? How many lines do you have connected? |
| 15:18.16 | newtonr | GOrozco, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information following those instructions. |
| 15:18.57 | AllanButton | GOrozco: because if have line 1 dead ( shorted ), and you are dialing out line 1 first, you'd get dead air. inbound your ring group would put you on line two |
| 15:19.20 | newtonr | thats a great point |
| 15:20.03 | GOrozco | I have an analog TDM410P and inbound works perfectly |
| 15:20.05 | aberrios | anyone in the UK have ISDN30 and a phone on EE/Orange handy? |
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| 15:20.15 | GOrozco | just outbound there is no sound both ways |
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| 15:20.29 | newtonr | GOrozco, AllanButton asked how many lines you have connected to it.. |
| 15:20.31 | GOrozco | the card has 2 FXO's only |
| 15:20.44 | GOrozco | both connected to a line on PSTN |
| 15:20.50 | newtonr | GOrozco, swap the lines in their ports and try calling out again |
| 15:21.22 | GOrozco | I did swap on the dialing plan... does that work? |
| 15:21.30 | GOrozco | let me ask them to swap the lines |
| 15:21.44 | newtonr | GOrozco, yeah as long as you specified which port to dial out, and then tried both ports one at a time |
| 15:22.19 | newtonr | GOrozco, but swapping physically could help troubleshoot if you had a failure of one of the cards ports |
| 15:22.26 | AllanButton | newtonr: unless the port is dead, in which case swapping the lines would still yeild valuable information |
| 15:22.31 | GOrozco | I put port 2 first usually.. it is not working. then I did with port 1, no joy either. |
| 15:22.52 | GOrozco | they are swapping the lines |
| 15:23.09 | GOrozco | testing |
| 15:23.20 | AllanButton | GOrozco: I am on the edge of my seat |
| 15:23.43 | igcewieling | I'm not. The problem is too weird to be a simple analog issue. |
| 15:23.54 | newtonr | GOrozco, like igcewieling I suspect an issue on the SIP side despite your other SIP scenarios working |
| 15:23.55 | igcewieling | I still think a screwed up localnet= setting |
| 15:24.07 | *** part/#asterisk joram (~joram@141.136.124.2) |
| 15:24.08 | igcewieling | newtonr: but we won't know until he PASTEBINS THE INFO |
| 15:24.51 | ChannelZ | does the console show "Called DAHDI/x/xxxx" and "DAHDI/x-x answered XXXXX" when you call out? |
| 15:24.53 | newtonr | give him time |
| 15:25.14 | AllanButton | igcewieling: newtonr: I am sorry, I may have created this diversion. |
| 15:25.26 | GOrozco | http://pastebin.com/MJCRJ8gu |
| 15:25.39 | GOrozco | Hi ChannelZ, it does |
| 15:25.53 | GOrozco | then no sound.. only some noise on the line |
| 15:26.00 | newtonr | AllanButton, no no, thanks for chiming in to help |
| 15:26.55 | GOrozco | thanks guys for trying to help me |
| 15:26.59 | newtonr | GOrozco, looks like you got only VERBOSE in your log.. try those instructions again.. make sure you get DEBUG, etc |
| 15:27.24 | GOrozco | oh.. increasing log and restarting my asterisk then |
| 15:28.15 | ChannelZ | well I'd tell you to make a test extension that does a Playback and an Echo and then use originate on the console to make a test call to it to a remote number, but it looks like you're using FreePBX and probably don't know what I"m talking about |
| 15:28.38 | ChannelZ | and I need to leave for work |
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| 15:31.40 | GOrozco | debug set to 4... |
| 15:31.41 | GOrozco | http://pastebin.com/2eHrZgYw |
| 15:32.40 | GOrozco | and you are right... using FreePBX, but I can learn.. looking on how to do the originate part |
| 15:33.47 | igcewieling | FreePBX? Thanks for wasting our time! |
| 15:34.36 | *** join/#asterisk Unchained (~Unchained@surft.nackt-im-garten.de) |
| 15:35.50 | igcewieling | I can't even read that past paste. |
| 15:35.58 | igcewieling | set debug to 0, and enable sip debug |
| 15:36.31 | igcewieling | GOrozco: set directmedia=no |
| 15:38.37 | newtonr | GOrozco, most of us here are not using FreePBX and are unfamiliar with the dialplan written by it. If you are using FreePBX, it's typically best to ask for help in #FreePBX. It's still likely an issue with configuration in Asterisk regardless of what you are using to write configuration. |
| 15:39.48 | AllanButton | GOrozco: I only remember the basics of analog card here myself, having not touched one in a few years. |
| 15:40.11 | AllanButton | GOrozco: but I would suggest testing your lines outside of the phone system with a butt-set perhaps. |
| 15:41.50 | newtonr | GOrozco, I'll also repeat that you are using a Digium card, you likely want to call support at the link I sent you earlier. |
| 15:43.52 | igcewieling | I hate how FreePBX does queues |
| 15:44.14 | Greenlight | With Local channels looping back ? |
| 15:44.29 | igcewieling | Greenlight: yeah, constant looping. |
| 15:44.34 | Greenlight | mhmm |
| 15:44.36 | AllanButton | igcewieling: is there a better gui? or is that a loaded question. |
| 15:44.50 | igcewieling | AllanButton: no. |
| 15:45.25 | igcewieling | All Asterisk GUIs SUCK. FreePBX sucks least. |
| 15:45.27 | AllanButton | igcewieling: didn't think so. I am not a fan of the way it handles some things, but we need something for customers and teir 1 support to access |
| 15:45.37 | newtonr | AllanButton, not that I've seen. FreePBX is pretty amazing |
| 15:46.02 | AllanButton | newtonr: are you @digium? |
| 15:46.13 | Greenlight | It's quite clever *what* it does, but quite often the *how* can make things difficult to diagnose |
| 15:46.26 | newtonr | AllanButton, yes I'm Digium's community support manager |
| 15:46.31 | igcewieling | When I say FreePBX sucks least, you should consider it a ringing endorsement. |
| 15:46.32 | pabelanger | GUI's are hard to do, since everybodys requirement is different |
| 15:46.56 | pabelanger | Askozia had a decent one, until they closed source the code |
| 15:47.17 | igcewieling | We use PBXware for our hosted service. That is about as closed as you can get 8-( |
| 15:47.35 | jmetro | Speaking of closed source shenanigans, I wish ExQuilla wasnt going paid. Those guys could have been open-source allstars. |
| 15:47.38 | igcewieling | they even "encrypt" their PHP scripts. |
| 15:47.40 | AllanButton | igcewieling: I am so sorry, I manage 3 of those bicom servers myself |
| 15:47.50 | AllanButton | igcewieling: i've considered sending the php off to be decoded |
| 15:48.20 | igcewieling | AllanButton: I would love to be able to intercept outgoing trunk calls. |
| 15:48.49 | AllanButton | igcewieling: same. I'd like to be able to set sip headers for accounting reasons. |
| 15:49.20 | igcewieling | PBXware does stupid things like remove the calling number when a user disabled outgoing callerid, so we can't bill for those calls. 8-( |
| 15:49.37 | jmetro | disallow outbound calls with no callerid =D |
| 15:49.51 | igcewieling | jmetro: that is what we do, but customers get annoyed. |
| 15:50.04 | AllanButton | igcewieling: same problem, except for call forwarded calls. |
| 15:50.07 | jmetro | "Sorry but you cant make anonymous telemarketer calls on our network" |
| 15:50.34 | igcewieling | jmetro: but you can. Just not from our hosted service. |
| 15:50.56 | AllanButton | igcewieling: but honestly, even after burning it to disk just so I drive over it, its still one of the best multi-tenant platforms |
| 15:51.14 | igcewieling | AllanButton: it is, we replaced FreePBX with it for our hosted service |
| 15:51.25 | AllanButton | let me refrash before flames, by platforms I mean gui for admin and client |
| 15:51.30 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 15:51.44 | AllanButton | igcewieling: we are going the other route. using vmware with freepbx per customer |
| 15:52.26 | AllanButton | newtonr: I forgot to say something back after I asked my question. Hello! |
| 15:52.27 | igcewieling | AllanButton: we install a server on-site for customers who want a local PBX. Only hosted customers (i.e. cheap bastards) which go with hosted. |
| 15:52.46 | newtonr | AllanButton, howdy! |
| 15:52.59 | AllanButton | igcewieling: Same here. But with FreePBX on Centos 6, and a centralize MySQL repo each VM is down to about 512ram usage |
| 15:53.14 | igcewieling | AllanButton: Why would you install the VM server on the customer site? |
| 15:53.20 | AllanButton | igcewieling: put on a IBM Blade with 96GB of ram, you get a lot out of a little. |
| 15:53.38 | AllanButton | igcewieling: oh, sorry, for customers we use hardware, with no vmware. |
| 15:53.53 | jmetro | VMware player runs it pretty good |
| 15:54.16 | AllanButton | jmetro: it does, but I would not run production on vmware player. |
| 15:54.19 | igcewieling | Anyone who knows anything discourages hosted. |
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| 15:55.06 | AllanButton | I never said I encourage it. lol. |
| 15:55.19 | jmetro | "hosted" to me means that the IT company hosts it and the user just has phones on their desk |
| 15:55.25 | jmetro | in which case, it works pretty awesome for us =D |
| 15:55.36 | igcewieling | I call our hosted clients "soon to be former clients" |
| 15:56.11 | igcewieling | jmetro: until there is a cable failure with a ETR 6 weeks away |
| 15:56.17 | AllanButton | I am at an ISP, so we can build circuits out to customers. So really, a PBX on site or hosted will give the same experience ( mostly ). Difference is capacity. |
| 15:57.01 | jmetro | igcewieling: Only cable issues that affect us are Comcast, etc.. in which case our clients are usually smart enough to know its not us to be mad at. |
| 15:57.21 | AllanButton | We have some customers who pretend to be big. 1 man show with a full auto attendant, multiple voicemails, ect. |
| 15:57.24 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:57.25 | igcewieling | jmetro: we are the ISP for almost all of our VoIP customers. |
| 15:57.42 | jmetro | Ah. never get into ISP business |
| 15:57.45 | jmetro | =p |
| 15:57.47 | AllanButton | igcewieling: same here. wait, are you in the next room? |
| 15:58.02 | igcewieling | AllanButton: unless you are in an apartment building in Pensacola, no. |
| 15:58.13 | AllanButton | igcewieling: now thats just freaky. |
| 15:58.19 | AllanButton | just kidding of course. hah. |
| 15:58.46 | jmetro | [hes not kidding and is actually in an apt building in pensacola. PLOT TWIST] |
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| 16:00.39 | AllanButton | hey, whats the best way to diagnose t38 issues? The carrier I am using says they support t38, unfortunately, even on my least busy system I have over 200 concurrent calls going to them. tshark becomes a pain in the ass with that many calls. |
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| 16:02.58 | AllanButton | no easy way huh. |
| 16:03.19 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:03.19 | WIMPy | Wireshark IS the easy way. |
| 16:03.28 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:03.32 | jmetro | wireshark & filters yo |
| 16:03.51 | AllanButton | that's what I was afraid off. |
| 16:04.08 | igcewieling | AllanButton: T.38 is not SIP |
| 16:04.09 | AllanButton | with rtp I get a couple gigs of pcap very quickly. |
| 16:04.13 | igcewieling | and does not use RTP |
| 16:04.23 | AllanButton | igcewieling: I honstly don't know much about t38. |
| 16:04.25 | igcewieling | it uses UDPTL and the ports are (IIRC) 4000 - 4000 |
| 16:04.34 | AllanButton | thats a small port range ;) |
| 16:05.00 | AllanButton | my problem is on the reinvite I beleive. Never seems to make it one way or another |
| 16:05.49 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:08.10 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:10.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:10.56 | igcewieling | AllanButton: Asterisk's UDPTL does not support directmedia. |
| 16:10.59 | igcewieling | you MUST disable it |
| 16:11.42 | igcewieling | also I meant 4000-4999, but check udptl.conf to be sure. |
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| 16:15.34 | AllanButton | igcewieling: son of a bitch I thought I had it turned off yesterday |
| 16:15.39 | AllanButton | to many cooks in the kitchen, lol |
| 16:15.49 | AllanButton | or just one cook who only got a few hours a sleep |
| 16:16.14 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:16.15 | WIMPy | One cook with split personality? |
| 16:16.48 | AllanButton | lol shutup. |
| 16:16.55 | AllanButton | ;) |
| 16:17.07 | WIMPy | Says who? |
| 16:17.13 | AllanButton | both of us. |
| 16:17.19 | jmetro | if you ever pay attention to my errors, the problem is the cook is sleeping while cooking |
| 16:18.37 | AllanButton | off topic, internet explorer 10 is hitting my office like a bad virus |
| 16:18.49 | jmetro | f12, browser mode ie9 |
| 16:18.51 | jmetro | fixd |
| 16:18.54 | AllanButton | nope |
| 16:18.56 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:19.06 | AllanButton | we use a CRM that does not work in IE10, no compatible mode will fix it |
| 16:19.15 | jmetro | oh thats nasty. |
| 16:19.21 | AllanButton | ironicaly, its microsoft dynamics crm |
| 16:19.33 | jmetro | funny how that works |
| 16:19.40 | coppice | microsoft doesn't understand irony |
| 16:19.49 | jmetro | developers developers developers developers |
| 16:19.55 | WIMPy | Nothing new |
| 16:20.00 | AllanButton | on my linux laptop I ran multiple versionof ie |
| 16:20.36 | AllanButton | something that microsoft can't even figure out on their own os. |
| 16:20.42 | coppice | when he said "developers developers develop", I wonder how many in the audience invested in Kodak? |
| 16:21.05 | WIMPy | *cough* |
| 16:21.11 | jmetro | and lost everything. |
| 16:21.25 | AllanButton | nothing like a joke that needs to be explained... |
| 16:21.38 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:22.36 | jmetro | the word developers sounds fake in my head now. thanks AllanButton |
| 16:22.44 | AllanButton | oh the note of corny jokes, two termintes walk in to a bar, they sit down and ask "Where's the bar tender" |
| 16:23.09 | AllanButton | s/termintes/termites/ |
| 16:23.39 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
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| 16:24.13 | coppice | that one takes me back to my childhood |
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| 16:27.52 | igcewieling | must be one of my friends: http://xkcd.com/ |
| 16:28.42 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
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| 16:33.56 | AllanButton | I hope italorossi comes in and asks why his voip sounds so choppy. |
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| 17:00.05 | pJok | has anyone in here managed to connect asterisk to facebook chat through res_xmpp? |
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| 17:01.07 | Faustov | AllanButton: lol |
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| 17:08.30 | neonerz | does anyone know an app that support click to dial from outlook 2013 (and hopefully support outlook versions older)? It doesn't seem like TAPI works on 2013 |
| 17:09.25 | [TK]D-Fender | https://www.google.ca/#fp=6320079e7b593ad9&q=outlook+asterisk+click+to+dial |
| 17:09.36 | [TK]D-Fender | 2 second search seems to turn up a bunch... |
| 17:09.53 | neonerz | yea, been there, none of those results support outlook 2013 |
| 17:09.56 | [TK]D-Fender | Have you tried all the ones on the first page of results? |
| 17:11.17 | neonerz | everyone I tried uses TAPI, which isn't supported by outlook 2013 |
| 17:12.31 | neonerz | the best I found was TeleTrigger which uses Polycom |
| 17:12.38 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 17:12.57 | neonerz | ''s push capability, but since phones and computers are on different VLANs with not inter-VLAN routing, it doesn't help |
| 17:14.29 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:16.29 | Greenlight | I'm seeing a few of these in my logs "ERROR[17688][C-000081bc]: chan_sip.c:32788 setup_srtp: No SRTP module loaded, can't setup SRTP session.". Does this mean that an endpoint is attemtping secure RTP ? |
| 17:16.50 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:17.59 | *** join/#asterisk vogong (~vogong@d24-141-181-199.home.cgocable.net) |
| 17:19.25 | [TK]D-Fender | yours or theirs |
| 17:19.27 | igcewieling | Greenlight: that would be my guess. |
| 17:20.21 | Greenlight | I don't like seeing "ERROR" in there, but it seems to not be that severe if it's attemtping SRTP then presumably falling back |
| 17:20.38 | Greenlight | Now to track down which users been playing with their phone settings |
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| 17:35.36 | SuperNull | i think my event cache for MWI is jammed.. seems stuck at 346 items. |
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| 17:48.52 | boom^time | Has anyone any tips for the AMD application (my first time really using it). Are the stock values pretty good? I've read that adding a second of background silence before calling AMD can be useful. |
| 17:56.49 | WIMPy | A working AMD is the best kept secret of the trade. |
| 18:01.38 | cusco | hi |
| 18:01.53 | cusco | haha |
| 18:03.37 | boom^time | WIMPy, so in other words I'm on my own? |
| 18:05.42 | navaismo | yea |
| 18:05.44 | navaismo | haha |
| 18:06.21 | boom^time | Just looking for some tips not your private key |
| 18:06.29 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 18:07.35 | navaismo | so far in my test and production enviroments I need to Use an Answer Background(beep) then the AMD and the worst part is the adjustment of the amd.conf |
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| 18:08.30 | navaismo | my best result are 70% of effectiveness |
| 18:08.37 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 18:08.41 | boom^time | Right, tweaking that thing and then attempting testing |
| 18:09.05 | boom^time | What sample do you use for testing? |
| 18:09.21 | boom^time | And how do you verify accuracy? Listen to the calls yourself? |
| 18:09.29 | *** join/#asterisk module000 (~module000@ec2-54-225-236-107.compute-1.amazonaws.com) |
| 18:09.45 | navaismo | yes I call to my phone, cellphone and familiars phones and checkit |
| 18:09.48 | *** join/#asterisk SGjunior (~sgjunior@out-on-228.wireless.telus.com) |
| 18:09.54 | jmetro | what about different answering machines with different tones? |
| 18:09.55 | navaismo | let me pb the stuff |
| 18:10.45 | boom^time | Seems like a good idea would be to make a change, call 1000 random local numbers, listen to each, compare. tweak another setting, rinse repeat. |
| 18:12.00 | navaismo | well i not call 100 times but its up to you |
| 18:12.10 | navaismo | here play with those values-->http://pastebin.com/S1W1uZgy |
| 18:12.49 | boom^time | These are yours? Thank you very much. |
| 18:12.50 | nam3l3zz | how to become aware, that congestion has been "result code" of Dial() ? |
| 18:13.54 | boom^time | navaismo, I've read other places that a second of background silence, or in your case the beep is a good idea. I'm just not sure why? |
| 18:14.12 | navaismo | nam3l3zz, maybe using ${DIALSTATUS} |
| 18:15.00 | navaismo | as WIMPy said AMD is a obscure case of study, a myth... |
| 18:15.08 | navaismo | so you need to start playing with it |
| 18:15.17 | navaismo | and find your best scenario based on your test |
| 18:15.21 | nam3l3zz | navaismo: yep, thanx |
| 18:15.23 | AllanButton | My guess on the silence is it trains the AMD to know what silence is. |
| 18:18.00 | AllanButton | this is how other software packages detect answering machines, fyi: http://pastebin.com/hepudR3M |
| 18:18.32 | AllanButton | which is really just based off this: http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect |
| 18:18.58 | boom^time | Right I was looking at that before. AMD looks a bit more refined. |
| 18:19.36 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 18:22.36 | boom^time | That one second of silence is no good, I can answer and say "hello" in less than a second and it feels natural |
| 18:23.05 | boom^time | it's not for training AMD since it doesn't roll to AMD until it's done playing the silence |
| 18:25.25 | navaismo | welcome to the amd strange non-working world |
| 18:26.19 | boom^time | Thanks! |
| 18:28.05 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:28.25 | *** join/#asterisk SGjunior (~sgjunior@out-pq-187.wireless.telus.com) |
| 18:28.49 | navaismo | if you are looking for a trusted solution IRC sangoma has one |
| 18:31.15 | jmetro | the one second of silence is to delay any delays on the answering machine |
| 18:31.17 | jmetro | not to train amd |
| 18:31.49 | boom^time | jmetro, but what about the fact that I say "hello" within a second of picking up my phone? |
| 18:32.02 | boom^time | it explains why so many robot calls to my house hang up on me I suppose. |
| 18:32.04 | jmetro | thats why AMD is sill. |
| 18:32.07 | jmetro | y |
| 18:32.36 | jmetro | "many call centers prefer to not use AMD and actually have agents leave a voicemail as the customer was more likely to respond to the personalized message" |
| 18:33.07 | navaismo | boom^time, http://www.sangoma.com/products/lyra-amd/ |
| 18:34.38 | boom^time | navaismo, thanks |
| 18:36.13 | *** join/#asterisk bambu (4382a6af@gateway/web/freenode/ip.67.130.166.175) |
| 18:36.16 | bambu | Hi |
| 18:36.36 | bambu | I'm not sure about the details of wiring, but I had a question. |
| 18:36.37 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:37.19 | bambu | I have two buildings, the phone technician spliced the phone line for one building (555-6666) so that the other building also has (555-6666) |
| 18:37.54 | bambu | Could I setup a system that directs call from building to building? |
| 18:38.24 | bambu | The wires that he crossed were in the street junction box (forget the accuratename) |
| 18:39.55 | [TK]D-Fender | bambu: that is a SINGLE line to the telco |
| 18:40.06 | [TK]D-Fender | bambu: the same as having 5 jacks in your house |
| 18:40.10 | [TK]D-Fender | they are not separate |
| 18:40.18 | bambu | Right.... |
| 18:40.20 | [TK]D-Fender | that is also not a means of calling from one end to the other. |
| 18:40.23 | [TK]D-Fender | it is the same line |
| 18:40.40 | bambu | so what would be the best option? |
| 18:40.55 | [TK]D-Fender | get actual separate wiring |
| 18:41.15 | bambu | So tell me if I got this right, I HAVE TO: telco to system, phone (building 1) to system, (building 2) to system... |
| 18:41.19 | [TK]D-Fender | that is simply a "party line" as-is |
| 18:41.28 | [TK]D-Fender | that line doesn't cut it |
| 18:41.28 | bambu | right? |
| 18:41.34 | [TK]D-Fender | how you want to connect them is up to you |
| 18:41.40 | boom^time | jmetro, what delay in an answering machine would require a 1 sec pause? I can't imagine it. |
| 18:41.51 | [TK]D-Fender | bambu: What do you actually WANT to have? |
| 18:42.01 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:42.08 | bambu | An auto attendant that directs calls to each seperare building (business) |
| 18:42.34 | [TK]D-Fender | And what PBX do you have now? |
| 18:42.47 | [TK]D-Fender | I presume "none" based on that shared-line you've described |
| 18:42.48 | bambu | None, so looking for new solution. |
| 18:43.01 | bambu | Good presumption :) |
| 18:43.08 | [TK]D-Fender | Well you could set up Asterisk as that PBX. You'd then connect phones to it as well as your lines |
| 18:43.29 | bambu | I'd have telco line to asterisk. |
| 18:43.35 | [TK]D-Fender | So you should figure out how many lines you need, and how many phones, and what kind |
| 18:43.36 | bambu | Then I'd connect each phone to asterisk? |
| 18:43.50 | [TK]D-Fender | also consider what you are going to use to connect to the other building |
| 18:43.56 | bambu | I want two phones, one in building a , the other building b |
| 18:44.11 | bambu | So whats my best option? |
| 18:44.12 | [TK]D-Fender | normally you'd NETWORK them together so you could use VoIP phones |
| 18:46.04 | bambu | Elaborate on the term "network" |
| 18:46.24 | WIMPy | IP |
| 18:46.56 | WIMPy | But unless ou get rid of that line and change it for something else, I'd recommend not to change anything at all. |
| 18:47.06 | bambu | So there is nothing i can build on the existing part line? |
| 18:47.15 | WIMPy | Using POTS connections is a PITA. |
| 18:47.17 | [TK]D-Fender | bambu: you don'derstand basic LAN's and computer networking? |
| 18:47.30 | [TK]D-Fender | bambu: you don't understand basic LAN's and computer networking? |
| 18:47.45 | bambu | LAN's, but buying voip phones is up there |
| 18:48.20 | bambu | I wanted POTS |
| 18:48.27 | [TK]D-Fender | ba[14:47]bambuSo there is nothing i can build on the existing part line? <- no, that is a dumb line. yoyu can'[t be on a call and transfer it or anything. You can't plug 10 phones on there and "transfer" calls. It's a single dumb line just like at home |
| 18:48.55 | bambu | Thats what i feared. |
| 18:48.59 | [TK]D-Fender | bambu: Feel free to use POTS phones.. but consider how those are going to talk to your SERVER. |
| 18:49.06 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:49.15 | SuperNull | Any idea when the event cache gets iterated ? (how often or is it triggered to happen or what..) ? |
| 18:49.17 | [TK]D-Fender | Are you going to a ton of extra wires between your buildings? |
| 18:49.25 | [TK]D-Fender | I doubt it. |
| 18:49.37 | bambu | OK. |
| 18:49.41 | [TK]D-Fender | Which is when you'd be suggested to use VoIP devices. |
| 18:49.55 | [TK]D-Fender | Now you CAN use analog phones with ADAPTERS to get them to connect to your network |
| 18:49.57 | [TK]D-Fender | ~ata |
| 18:49.57 | infobot | somebody said ata was Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:50.07 | bambu | ty |
| 18:50.09 | [TK]D-Fender | Instead of actual nice dedicated hard-phones... |
| 18:50.13 | bambu | INTERNET FALLING OUT, BBL |
| 18:50.13 | [TK]D-Fender | but it is suckier |
| 18:51.35 | boom^time | any idea why the answer application is ignoring my required 50000 millisecond delay before moving on with the dialplan? |
| 18:51.55 | WIMPy | It's the maximum time. |
| 18:52.31 | boom^time | hmm the asteriman page didn't mention a limit |
| 18:53.09 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:53.20 | boom^time | even 1000 or 1 second is ignored. |
| 18:53.32 | boom^time | I feel like that argument quit being acknowledged at some point along development. |
| 18:53.39 | WIMPy | I have no idea in what situations it makes a difference or not. It you want a dealy use Wait or some silence in the first Plaback. |
| 18:54.01 | WIMPy | s/It/If/ |
| 18:54.46 | boom^time | I agree with you, it's just that it is documented as an argument of Answer that simply doesn't work. But not everyone is as OCD as me. Definitely not in the * community anyhow. |
| 18:55.36 | *** join/#asterisk skorzen (~skorzen@89.214.74.82) |
| 18:55.37 | [TK]D-Fender | boom^time: just call WAit() yourself |
| 18:55.46 | WIMPy | Right. The documentation says after answering. |
| 18:56.04 | SuperNull | is #asterisk-dev a bad channel to ask source related questions ? event dispatching related. |
| 18:56.23 | [TK]D-Fender | no, it is the right place |
| 18:56.29 | WIMPy | IIRC there was something about it being the maximum time it wats for confirmation that the channel is up. |
| 18:56.29 | boom^time | [TK]D-Fender, I'm going to |
| 18:56.45 | boom^time | Oh I see. the documentation is misleading |
| 18:57.20 | boom^time | or just flat out wrong "Asterisk will wait this number of milliseconds before returning to |
| 18:57.20 | boom^time | <PROTECTED> |
| 18:57.25 | WIMPy | No. Either what I remember or the documentation is wrong. |
| 18:57.55 | WIMPy | Yes, that's pretty clear. |
| 18:59.07 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:59.31 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:00.54 | boom^time | any ideas why half the time a call file will just do nothing? I'm doing a cp blah.call tmp.call && mv tmp.call /var/spool/asterisk/outgoing/ |
| 19:00.58 | boom^time | it's extremely hit or miss. |
| 19:01.12 | WIMPy | You're doing it wrong. |
| 19:01.17 | boom^time | that's what I wanted to hear. |
| 19:01.38 | WIMPy | You must create the call file in another directory on the same FS and then MOVE it to .../outgoing. |
| 19:01.48 | boom^time | this is on the same fs |
| 19:01.57 | boom^time | in a different directory |
| 19:01.58 | WIMPy | Or link if you use an application. |
| 19:02.12 | WIMPy | cp doesn't work. |
| 19:02.16 | boom^time | wimpy you didn't read my whole command |
| 19:02.25 | boom^time | please see the && mv tmp.call |
| 19:02.50 | boom^time | that command is run from a different dir on the same partition |
| 19:02.58 | WIMPy | Err, what do you cp and what do you mv? |
| 19:03.20 | boom^time | I cp the file to tmp first and then mv it so I don't have to recreate the file every time I want to test. |
| 19:03.24 | WIMPy | Do they come from another FS? |
| 19:03.36 | WIMPy | Ah, ok. |
| 19:03.53 | WIMPy | That's right then. |
| 19:03.58 | boom^time | no all ext4 same fs |
| 19:04.01 | boom^time | and same partition |
| 19:04.53 | navaismo | and what happen to the call file it disappear? |
| 19:05.18 | boom^time | good question, I've just been rerunning that command until the call finally is made |
| 19:05.30 | boom^time | i'll reproduce and check |
| 19:05.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 19:05.55 | WIMPy | It's all done locally? |
| 19:06.25 | navaismo | try enabling the option Archive: Yes and then look at the stored file |
| 19:06.49 | boom^time | yes locally, yes it disappears |
| 19:06.58 | boom^time | navaismo, good idea |
| 19:06.59 | *** part/#asterisk Unchained (~Unchained@surft.nackt-im-garten.de) |
| 19:07.12 | WIMPy | If it disappears it must have been processed. |
| 19:08.50 | boom^time | they are coming through I think. I think I'm just testing so rapidly that it's causing it to hiccup and queue them |
| 19:09.08 | boom^time | Since all of a sudden I'm having difficulty reproducing |
| 19:09.55 | WIMPy | Are you always using the same name? |
| 19:10.00 | boom^time | yes |
| 19:10.10 | Qwell | adds a comment to his out of context quotes list |
| 19:10.43 | igcewieling | you do not want to push lots of call files into asterisk at the same time. |
| 19:10.49 | igcewieling | it will try dialing them all at once. |
| 19:10.55 | WIMPy | Well, then you might be writing to a new file when Asterisk tries to delete the previous one. |
| 19:10.58 | igcewieling | hilarity will ensue. |
| 19:11.06 | jmetro | Qwell: i'd be interested in this Out of context quotes list. Reminds me of bash.org |
| 19:11.09 | AllanButton | igcewieling: can't you set a start time? |
| 19:11.34 | WIMPy | You can |
| 19:11.41 | boom^time | you can set the timestamp for the file right? |
| 19:11.46 | boom^time | or is there a better way |
| 19:12.19 | AllanButton | I don't think there is a better way outside of Local channels and crazy dialplans |
| 19:12.34 | WIMPy | No, you have to use the timestamp. |
| 19:13.33 | boom^time | damnit it just did it but I didn't have an outgoing_done dir created |
| 19:14.31 | WIMPy | Use uniqe names. |
| 19:15.44 | igcewieling | AllanButton: yes. set the ctime (maybe mtime) on the file to the future. |
| 19:15.55 | boom^time | Okay it just happened again, once about 2 seconds after a completed call and another a two mins with zero activity |
| 19:16.11 | boom^time | here is the bottom of the outgoing_done archive StartRetry: 11568 1 (1377112802) |
| 19:16.11 | boom^time | Status: Expired |
| 19:16.50 | *** join/#asterisk LonghornWork (~kvirc@74.115.99.1) |
| 19:17.18 | AllanButton | asterisk wiki does not go in to detail on what the status codes mean. :( https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files |
| 19:17.29 | WIMPy | Unless you make sure the file you try to write doesn't exist, this is not going to get you anywhere. |
| 19:19.01 | [TK]D-Fender | AllanButton: Status: <exitstatus> - Can be "Expired", "Completed" or "Failed" |
| 19:19.08 | [TK]D-Fender | AllanButton: Seems pretty clear to me |
| 19:19.33 | AllanButton | [TK]D-Fender: for somebody who does not use .call files, I don't know what status expired means |
| 19:19.33 | boom^time | WIMPy, it didn't exist |
| 19:19.36 | boom^time | I checked |
| 19:19.58 | [TK]D-Fender | AllanButton: MaxRetries: <number> - Number of retries before failing, not including the initial attempt. Default = 0 e.g. don't retry if fails. |
| 19:20.04 | [TK]D-Fender | AllanButton: is that not a big enough hint? |
| 19:20.12 | WIMPy | That could be a question of the definition of "exists". |
| 19:20.30 | boom^time | the inode no longer existed |
| 19:20.40 | AllanButton | [TK]D-Fender: I need a corner to nap in. |
| 19:20.51 | WIMPy | That's the point. The file might still have existed. |
| 19:21.00 | [TK]D-Fender | AllanButton: This room is round... sorry |
| 19:21.17 | boom^time | [TK]D-Fender, so it attempted, reattempted, and then gave up? |
| 19:21.27 | WIMPy | If Asterisk still has the old file open and then closes it and deletes it, it might delete the new copy. Or it might write status information to the new file or whatever. |
| 19:21.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.191) |
| 19:22.11 | boom^time | I waited a long period of time before retrying the call, I checked with ls and the file was not there. I then copied it over and it vanished into the abyss |
| 19:22.28 | Qwell | You used cp? Don't use cp. |
| 19:22.37 | AllanButton | boom^time: cp blah.call tmp_`date "+%s"`.call && mv tmp_`date "+%s"`.call /var/spool/asterisk/outgoing/ |
| 19:22.49 | boom^time | Qwell, reread the whole command |
| 19:22.49 | Qwell | Never copy call files into place. Always move. |
| 19:23.04 | boom^time | I did say copy didn't I. I'm doing this |
| 19:23.10 | [TK]D-Fender | And only withing the same filesystem |
| 19:23.15 | WIMPy | boom^time: The command you gave initially was incomplete. |
| 19:23.16 | boom^time | cp blah.call tmp.call && mv tmp.call /var/spool/asterisk/outgoing/ |
| 19:23.29 | AllanButton | boom^time: you can try my above command to make the filenames 'unique' |
| 19:23.49 | boom^time | AllanButton, thanks I'm going to try it. |
| 19:24.08 | AllanButton | just puts a unix timestamp on the filename |
| 19:24.30 | AllanButton | so the number will change every second, good enough for testing |
| 19:26.17 | WIMPy | Yu can add %N if you want to try more than once per second. |
| 19:26.51 | WIMPy | Or just the good old $$. |
| 19:30.09 | boom^time | with a unique name I still got StartRetry: 11568 1 (1377113668) |
| 19:30.09 | boom^time | Status: Expired |
| 19:31.03 | navaismo | cli console show someting |
| 19:31.18 | boom^time | No that's the worst part |
| 19:31.19 | navaismo | enabled the debug and search for it |
| 19:31.30 | igcewieling | have you shown us your .call file? |
| 19:31.35 | boom^time | debug on what? |
| 19:31.40 | navaismo | debug on asterisk |
| 19:31.52 | navaismo | core set debug 10 |
| 19:33.00 | boom^time | call file: http://pastebin.com/gKwJRnu4 |
| 19:34.24 | igcewieling | try removing lines 11 and 12 |
| 19:34.26 | igcewieling | 8-| |
| 19:35.26 | igcewieling | you also have maxtries 0 |
| 19:35.26 | boom^time | those are outputted by asterisk after it's finished |
| 19:35.50 | igcewieling | I was looking for YOUR call file |
| 19:35.59 | AllanButton | blah.call |
| 19:36.35 | boom^time | http://pastebin.com/jqC6DmLF |
| 19:36.47 | boom^time | I posted the other since it's the same thing only with the extra two lines |
| 19:37.10 | jmetro | i keep following pastebin links thinking there are treats at the other end and am always disappointed. |
| 19:37.38 | igcewieling | other than the maxtries I don't see anything odd, try using Local/ channels so you can see what is happening |
| 19:37.55 | WIMPy | Are you sure pastebin.com doesn;t send cookies? |
| 19:37.56 | igcewieling | use /n on the local channel |
| 19:38.02 | navaismo | and enable the asterisk debug you will see somthing like: DEBUG[11511] pbx_spool.c: Filename: /var/spool/asterisk/outgoing/2258397900_to_58397900.call, Retries: 0, max: 1 |
| 19:38.02 | navaismo | [Aug 21 14:35:48] DEBUG[11511] pbx_spool.c: Outgoing Local/58397900@localcall: StartRetry |
| 19:38.08 | AllanButton | jmetro: http://pastebin.com/Jp18TmbT |
| 19:38.16 | igcewieling | MY server sends out bananna bread, not cookies. |
| 19:38.34 | WIMPy | igcewieling: URI? |
| 19:38.35 | boom^time | I did a core set debug 10 and I'm seing nothing |
| 19:38.56 | WIMPy | logger.conf |
| 19:46.37 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:46.58 | AllanButton | guys. I need some help with t38. I have a carrier who says they support it, and an ATA that says it works with it ( spa122 ). I can't for the life of me make them work together. |
| 19:47.01 | AllanButton | http://pastebin.com/BXPBN3BR |
| 19:47.35 | AllanButton | I have the ATA on a public IP ( aka no nat, directly public ) and the asterisk box is as well. I've been toying will all the configuration options, and so far no love. |
| 19:50.21 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 19:53.25 | igcewieling | AllanButton: what about udptl.conf |
| 19:53.53 | igcewieling | AllanButton: register your ATA directly to the provider. If it doesn't work go ask somewhere else 8-) |
| 19:54.25 | igcewieling | atas and providers often lie about t.38 support. |
| 19:54.33 | AllanButton | udptl.conf http://pastebin.com/fHEBBhpG |
| 19:55.19 | AllanButton | I'd love to ditch faxing completely, but sadly its not an option :( |
| 19:55.59 | jmetro | remote faxxxxxxxx |
| 19:58.23 | igcewieling | AllanButton: if you ever make it work, let us know! You'd be the first I've heard of. |
| 19:59.05 | AllanButton | igcewieling: Great! :( So why all the work in Asterisk for fax and t38 support but nobody is using it? |
| 19:59.05 | navaismo | igcewieling, with FFA and flowroute you can receive(at least) Fax via T38 |
| 20:00.46 | *** join/#asterisk boom^time (~boom^time@75-151-20-174-Michigan.hfc.comcastbusiness.net) |
| 20:02.06 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 20:02.19 | boom^time | So what does the expired exitstatus mean for a call file? |
| 20:02.32 | navaismo | expired |
| 20:02.45 | navaismo | caduco, muerto |
| 20:03.23 | boom^time | how does that differ from failed? |
| 20:03.27 | navaismo | manged to enabled the debug? |
| 20:03.56 | boom^time | yes but haven't tracked down anything good yet |
| 20:04.22 | navaismo | at least thepbx_spoll.c get the call file? |
| 20:04.23 | boom^time | expired as in the call was attempted too many times? or that it took too long to attempt the call? |
| 20:04.37 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 20:05.09 | boom^time | yes it sees it |
| 20:05.20 | boom^time | now I need to capture a bad one |
| 20:05.32 | igcewieling | AllanButton: I assume they are all sending out from Asteirsk or receiving on Asterisk, not trying to pass t.38 calls THROUGH Asterisk |
| 20:05.42 | navaismo | right^ |
| 20:06.17 | AllanButton | I thought t38 passthrough was something that was support since 1.4 days |
| 20:06.42 | igcewieling | AllanButton: Indeed. And yet I've never gotten it to work and I don't remember anyone else getting it working either. |
| 20:07.05 | AllanButton | should I take this to jira? |
| 20:07.07 | navaismo | If I remove this from res_rtp_asterisk.c: static int ast_rtp_dtmf_begin, static int ast_rtp_dtmf_end, static int ast_rtp_dtmf_end_with_duration its probably that asterisk ignored DTMF on bridged call? |
| 20:07.12 | igcewieling | AllanButton: only if you want to fight for your bug. |
| 20:07.59 | jmetro | Jira isnt so much of a Bug Tracking system so much as it is "People post bugs and they are deleted" system. |
| 20:08.17 | igcewieling | jmetro: that is why you have to fight for your bugs. |
| 20:08.47 | AllanButton | problem is, asterisk is in the middle. I am sure I'd have to prove my provider allows t38, and that my ata is capable |
| 20:08.59 | AllanButton | I know my carrier does, because I've use t38 with them before, but not with asterisk. |
| 20:09.03 | igcewieling | AllanButton: I already told you how to do that |
| 20:09.17 | igcewieling | connect your ATA direct to the carrier without Asterisk involved. |
| 20:09.59 | AllanButton | igcewieling: that involves some big networking changes I think. I'd have to think about it. |
| 20:10.18 | igcewieling | AllanButton: set up a 2nd account with the provider |
| 20:10.46 | AllanButton | my provider is a carrier, I'd have to go through test and turnup with their engineering to turn up a new sip pri |
| 20:10.58 | file | jmetro, it's rare that a bug is ever actually deleted |
| 20:11.09 | igcewieling | You are not doing test and turnup with your T.38 stuff 8-) |
| 20:12.27 | igcewieling | AllanButton: you could also use receivefax to test with your carrier and use it to test with your ATA. Then you'll at least know if the two legs work with T.38. Make sure to VERIFY the T.38 though, stuff like to fallback to ulaw when you are not looking. |
| 20:12.57 | AllanButton | oh, thats why my fax it failing right now. I don't allow it to go back to ulaw. |
| 20:13.17 | AllanButton | I only ever get one way t38 negotiation. either carrier to asterisk but not ata, or ata to asterisk but not carrier |
| 20:13.35 | jmetro | do errors happen that you see |
| 20:14.06 | AllanButton | in the sip channel I don't see errors, but I don't see ACK on the INVITE for T38 |
| 20:15.11 | jmetro | you dont see anything happening in the CLI with just core set verbose and core set debug? |
| 20:16.24 | AllanButton | I see lots in the console. See the bottom of this paste: http://pastebin.com/41EMHT5Q |
| 20:17.00 | boom^time | navaismo, When the call file is ignored the pbxspool isn't coming up on debug |
| 20:17.35 | boom^time | yet it magically ends up in outgoing_done |
| 20:19.14 | jmetro | AllanButton: i see ulaw/alaw in there. |
| 20:19.43 | jmetro | is that supposed to happen? |
| 20:20.33 | AllanButton | I beleive that is needed for the call setup, but once the ATA flips to T38, it should not come back to ulaw. A couple lines strike me as odd |
| 20:20.49 | AllanButton | Extra Color: "Transmitting (no NAT) to 216.145.101.8:5060:" |
| 20:21.04 | jmetro | Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) |
| 20:22.01 | jmetro | i think something is telling it to be ulaw |
| 20:22.11 | jmetro | and something else is going "well, ulaw sucks. BYE" |
| 20:22.44 | jmetro | because the symptoms and output look a lot like when i forget to set codecs on my system |
| 20:23.01 | jmetro | and dial a g711 only phone with a g722 only phone |
| 20:23.18 | AllanButton | this is the text of the tshark, you can see after it does some ulaw, it tries to change to T38 when it says "SIP/SDP Request: INVITE" near th ebottom, http://pastebin.com/DpJN1YGC |
| 20:24.06 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 20:24.24 | Penguin | Is the fax machine a T.38 capable device? |
| 20:25.14 | *** join/#asterisk avb (~avb@190.122.126.194) |
| 20:25.16 | AllanButton | Does the fax machine need to be a T.38 capable device? The ATA its connected to support T.38 |
| 20:25.29 | AllanButton | Cisco SPA122 ( formally Linksys SPA2102 ) |
| 20:25.53 | avb | damn, seems direct rtp media contifuration with a remote server is quite tricky |
| 20:26.06 | jmetro | the spa112 i think you mena? |
| 20:26.14 | avb | does asterisk 11/12 makes things any easier? |
| 20:26.19 | jmetro | wait nvm |
| 20:26.23 | avb | with direct rtp |
| 20:26.36 | AllanButton | jmetro: you'd be correct. I spent the past 5 years calling things 2102, and cisco had to change the number. |
| 20:26.40 | Penguin | If the fax machine is not able to handle T.38, what good will a T.38 ATA be? |
| 20:27.09 | file | it will act as a T.38 endpoint and do the translation using a DSP |
| 20:27.25 | avb | i have a pbx in the internet, any ways to make ext2ext calls direct and rest proxy via the pbx? |
| 20:27.27 | jmetro | although |
| 20:27.33 | jmetro | i think the spa112's do not go in reverse friend |
| 20:27.43 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:27.48 | jmetro | i may not be right. |
| 20:27.48 | Penguin | Is it a 112 or a 122? |
| 20:27.59 | AllanButton | SPA122 |
| 20:28.43 | AllanButton | jmetro: what do you mean by in reverse? |
| 20:29.21 | jmetro | i usually take them from Internet -> Phone, not the other way around |
| 20:30.02 | avb | jmetro: was that to me? :) |
| 20:30.19 | jmetro | No |
| 20:30.26 | avb | oh, sorry :) |
| 20:30.44 | AllanButton | know any ata that does work? |
| 20:31.35 | jmetro | AllanButton: i didnt say it didnt work, i just dont know that it does. |
| 20:31.57 | AllanButton | jmetro: ah. in that case, know any that do work? :P |
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| 20:38.27 | jmetro | AllanButton: unfrortunately no, i've only ever had to dumb down the internet for a lousy fax machine |
| 20:38.43 | jmetro | i know Sangoma ATA"s are supposed to be incredible. |
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| 20:44.06 | Katty | AHHH |
| 20:44.11 | Katty | throws rolls of toilet paper at wall |
| 20:46.20 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:46.45 | navaismo | boom^time, asterisk and your call files has the same owner? |
| 20:47.39 | jmetro | Katty: Angry at paper again? |
| 20:47.51 | boom^time | no they don't, but the majority of them work |
| 20:48.04 | AllanButton | Katty: did somebody forget to replace the empty roll? |
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| 20:49.13 | boom^time | navaismo, no they don't, but the majority of them work |
| 20:49.25 | navaismo | echo echo echo |
| 20:49.39 | navaismo | asterisk is running as root or asterisk or..? |
| 20:49.49 | boom^time | root |
| 20:50.22 | boom^time | temporarily during testing |
| 20:52.40 | boom^time | I forced about 10 calls back to back without a break. and get this.. which is self explanatory, I wonder about the calls that are offset by a few minutes and this happens to it. I feel as though it's just this terrible little linksys pap2-na |
| 20:52.47 | boom^time | pbx_spool.c:392 attempt_thread: Queued call to SIP/503 expired without completion after 0 attempts |
| 20:55.15 | navaismo | and how look your call file right now? |
| 20:56.32 | boom^time | same I changed maxretries to 5 |
| 21:07.54 | SuperNull | anyone using AIS for MWI with realtime ? |
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| 21:37.00 | navaismo | boom^time, have you consider to use AMI instead call files |
| 21:40.20 | boom^time | Yes I just liked the idea of building files and postdating them |
| 21:43.41 | jmetro | use AMI, and append an entry into a log file if thats what youre llooking for? |
| 21:44.37 | boom^time | I mean building a call file for the future |
| 21:44.46 | boom^time | is AMI more reliable/have advantages? |
| 21:53.23 | navaismo | more control |
| 21:53.24 | navaismo | yes |
| 21:53.29 | navaismo | you can see the response |
| 21:56.02 | boom^time | I see |
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| 22:14.44 | SeRi | the company I work for just switch to IP phones. They are having issues with echo. The phone installers recommended to turn on qos. If we are using pots line's for our out going... How is qos going to help us? |
| 22:14.49 | SeRi | This is a small office |
| 22:14.56 | SeRi | 50 employees at most |
| 22:15.09 | [TK]D-Fender | it's not the phone's fault |
| 22:15.17 | SeRi | so I am sure the internal communications is not saturating the switches |
| 22:15.25 | [TK]D-Fender | QoS has nothing to do withe echo |
| 22:15.32 | SeRi | +1 [TK]D-Fender |
| 22:15.37 | SeRi | I told the, |
| 22:15.43 | [TK]D-Fender | nor does lack of BW |
| 22:15.44 | SeRi | s/the,/them/ |
| 22:15.59 | SeRi | Its the hardware? |
| 22:16.52 | [TK]D-Fender | it's what the phones are talking TO |
| 22:16.59 | SeRi | ah. got cha |
| 22:17.19 | [TK]D-Fender | [18:14]SeRithe company I work for just switch to IP phones. They are having issues with echo. The phone installers recommended to turn on qos. If we are using pots line's for our out going... How is qos going to help us? <- your POTS interface is the source of echo |
| 22:17.49 | SeRi | I see the light! |
| 22:17.54 | SeRi | bows |
| 22:18.11 | SeRi | Thanks [TK]D-Fender |
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| 23:04.59 | navaismo | If I remove this from res_rtp_asterisk.c: static int ast_rtp_dtmf_begin, static int ast_rtp_dtmf_end, static int ast_rtp_dtmf_end_with_duration its probably that asterisk ignored DTMF on bridged call? |
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| 23:26.15 | ChannelZ | it's probable it would not compile.. |
| 23:27.35 | igcewieling | you could ask on.... #asterisk-dev |
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| 23:32.19 | navaismo | last time there no responses at all so.. i guess im going to trial-error |
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| 23:36.44 | igcewieling | that usually means "dumb question" 8-| |
| 23:36.54 | igcewieling | what are you trying to accomplish? |
| 23:36.59 | ChannelZ | Why do you want to swallow dtmf? |
| 23:38.11 | igcewieling | if I was trying that I'd start by deliberately forcing an inband .vs. rfc2833 mismatch |
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