00:00.45 | SeRi | dijib: I replyed to you |
00:00.51 | *** join/#asterisk itamarjp (~itamar@fedora/itamarjp) |
00:01.54 | itamarjp | anyone have setup a polycom music on hold using uri ? voIpProt.SIP.musicOnHold.uri ? |
00:06.13 | Chainsaw | I just leave that to my regular Asterisk MOH. |
00:06.31 | Chainsaw | (I do have Polycom handsets, yes) |
00:08.00 | itamarjp | Chainsaw, the phone is not using asterisk, I just need music on hold from asterisk or other place. |
00:09.42 | [TK]D-Fender | itamarjp: We don't do that. Guess you'll just have to try for yourself |
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00:24.15 | paulc | tosses ghost75 a loonie |
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00:31.32 | jraddin | I am using ALSA to tie into a ConfBridge. I can dial in to the bridge via a SIP trunk and merge the ALSA in with "console dial 6300@conferences". Over time, however, the audio quality degrades and begins to motorboat. After about 30 minutes, the audio is unusable. If I do a console hangup and redial, everything is fine. Any ideas? |
00:36.32 | ChrisInSydney | jraddin: Ouch ! :-/ |
00:37.47 | ChrisInSydney | jraddin: Small question from someone who really has no idea, but, are you using an internal sound card ? do you have a USB one you could try ? |
00:37.54 | WIMPy | I haven't tried for a while, but when I did, i had the issue that I got an ever increasing delay. |
00:38.32 | jraddin | I have tried the onboard soundcard and 2 different PCI cards |
00:39.12 | ChrisInSydney | :-/ |
00:39.18 | jraddin | havent tried USB, but a lot of the USB cards don't support multiple sample rates which I have found asterisk requires since it outputs at 8khz |
00:39.26 | ChrisInSydney | ahh |
00:39.41 | WIMPy | Known issue :-( |
00:40.12 | ChrisInSydney | no specific reason for USB, just wonderin' if you had tried a different device |
00:40.19 | jraddin | sample rate issue or motorboating? |
00:40.29 | ChrisInSydney | but if WIMPy is right :-/ |
00:41.28 | ChrisInSydney | Motorboating usuallly power caps, but probably not in this instance |
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00:46.30 | jraddin | would meetme have the same issue? |
00:48.32 | WIMPy | I wouldn't be surprised if just a normal call would have it. |
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00:55.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
00:56.35 | slav3_kitten | is 100ms of jitter correctable on iax? |
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00:58.47 | jraddin | I'm also getting a chan_alsa.c:479 alsa_read: Read error: Resource temporarily unavailable when I dial the call from the console |
00:59.32 | WIMPy | slav3_kitten: Sure, but it might be a bit hard for the conversation. |
01:00.52 | slav3_kitten | hmmm |
01:01.30 | slav3_kitten | this wisp sucks dick... i seem to get jitter and such on my IAX2 voip.ms incoming calls. sip out to flowroute is supposedly nice an solid |
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01:14.31 | jraddin | WIMPy: Is there any timeline for a fix on this issue? |
01:15.15 | WIMPy | I don;t know if anyone is interested in fixing it. |
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01:15.39 | jraddin | So ALSA just doesn't work? |
01:16.30 | WIMPy | I don't know. As I said, I heven;t tried it for quite a while. |
01:16.42 | jraddin | Gotcha |
01:16.43 | jraddin | sorry |
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01:33.22 | slav3_kitten | huzzah i have a UK number |
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02:10.49 | dijib | anybody in here know if asterisk has the ability to accept collect calls? |
02:10.53 | dijib | automaticall? |
02:10.54 | dijib | y |
02:11.23 | Maliuta | Would depend on how the call is to be accepted |
02:11.38 | dijib | hmm ok then |
02:12.09 | Maliuta | If you get an operator asking a question you could use a small IVR (Collect calls accepted, press 1 to connect) |
02:13.05 | Maliuta | Otherwise you'd have to have something recognise a particular phrase and then have it send the DTFM response |
02:13.30 | Maliuta | do you know what happens when you receive a collect call? |
02:13.48 | dijib | no |
02:13.51 | dijib | i should try it |
02:14.38 | Maliuta | Another option would be to set up a dialback (assuming the other end is sending CID) ... if the rings X times and hangs up then you dial the number back and connect it to a local SIP/IAX/DAHDI channel at the same time |
02:14.59 | dijib | and i could be different from a jail. im just wanting to setup a system in case one would need to call out from a cell block through asterisk |
02:15.11 | Maliuta | there are many options, you need to know exactly what you want/need |
02:15.13 | dijib | callback is not an option |
02:15.29 | dijib | ok then thanks Maliuta |
02:15.47 | Maliuta | unless you know the procedure that occurs when the call comes in, you're screwed |
02:15.49 | Maliuta | :) |
02:16.19 | dijib | that cant be that hard to attain |
02:16.34 | dijib | just mixmonitor a call |
02:21.10 | ChrisInSydney | hi all |
02:21.43 | ChrisInSydney | quick q?? Does anyone have a list of SIP addresses that "do things" |
02:21.51 | ChrisInSydney | monkeys, |
02:22.00 | ChrisInSydney | tell me has etc |
02:22.04 | ChrisInSydney | tell me has gone |
02:22.22 | ChrisInSydney | just want to do some quick tests / demos for trainnig |
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02:27.46 | WIMPy | What do you need? |
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02:32.52 | ChrisInSydney | WIMPy: Maybe something like tellme or something where we can use a Dial() command to connect directly to a SIP URI and have it do somthing like tell the time, play some file, echo test |
02:33.10 | ChrisInSydney | got a list with ekiga I'm testing but they are all a bit crap |
02:35.34 | Maliuta | ChrisInSydney: with which provider? most of them have echo test numbers and talking clocks |
02:36.26 | Maliuta | if you want to do it on your own machine then just create extensions that play sound files |
02:36.39 | Maliuta | or do echo |
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02:46.44 | WIMPy | Is there a way to set the language for sip guest calls only? |
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02:48.17 | WIMPy | If I get it right, you can only set defaults and not anything for only guests. |
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02:49.41 | linocisco | hi all, I have three PSTN lines and want to share it among VOIP phones extensions, which grandstream devices should I buy? |
02:50.02 | linocisco | or any cheaper products? |
02:50.36 | WIMPy | waits for someone to argue that this isn't possible. |
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02:57.30 | dfgas-cr48 | trying to compile asterisk 11.0.1 |
02:58.04 | dfgas-cr48 | and its complaining about missing ccart |
02:58.07 | dfgas-cr48 | ccar |
02:58.12 | dfgas-cr48 | not ccart |
02:58.34 | dfgas-cr48 | what is that? i put in apt-get install ccar and no |
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03:04.52 | ChrisInSydney | Maliuta: Nahh. Looking for an anon SIP URI to call |
03:04.59 | ChrisInSydney | WIMPy fixed me up offline |
03:05.02 | ChrisInSydney | :-) |
03:07.14 | dijib | dfgas-cr48: i got that when i tried to compile the rc1 or two whatever it was |
03:07.52 | dfgas-cr48 | dijib, i am downloading what ever is on the asterisk page |
03:07.56 | ChrisInSydney | WIMPy: Re Guests, I'm pretty sure or at least i had vivid haloucinations of, a way of setting the language from within the dialplan |
03:08.16 | dfgas-cr48 | dijib, how did you get past it? |
03:08.34 | WIMPy | ChrisInSydney: yes. That's possible. |
03:08.36 | slav3_kitten | anyone in here good with cisco ios routers? |
03:09.09 | WIMPy | Actually that would work as guest have their own context. |
03:09.16 | ChrisInSydney | Correct |
03:09.33 | ChrisInSydney | I always have a separate context for guest connects |
03:09.40 | slav3_kitten | http://pastie.org/5385221 < which is better and will they both work? |
03:10.02 | ChrisInSydney | Other SIP services get their own too |
03:10.03 | dijib | i got the offical release and not the review candidate. |
03:10.05 | dijib | dfgas-cr48: |
03:11.05 | WIMPy | Ok. That should help. |
03:11.42 | ChrisInSydney | slav3_kitten: The one that requires less typing ;-) |
03:11.52 | ChrisInSydney | unless you need to fix it, |
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03:11.56 | slav3_kitten | ChrisInSydney, but will they both accomplish the same thing? |
03:12.00 | ChrisInSydney | then the one that is most descriptive |
03:12.24 | dfgas-cr48 | dijib, hmmmm |
03:12.24 | ChrisInSydney | not too good on Cisco, I only wear their shirts so I can charge more |
03:12.29 | ChrisInSydney | but |
03:12.52 | slav3_kitten | ChrisInSydney, i'm not great myself but i think they'd both work an option B is prettier |
03:13.34 | dfgas-cr48 | dijib, this is the link for the source i am using http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz |
03:14.11 | ChrisInSydney | looking at it, if you're just doing IAX stuff, B. Less ACLs and probably easier to read |
03:14.16 | ChrisInSydney | assuming it works |
03:14.32 | ChrisInSydney | WIMPy. Glad to return the favour |
03:14.43 | slav3_kitten | ChrisInSydney, as far as i can test it appears to function |
03:14.55 | ChrisInSydney | cool |
03:15.18 | ChrisInSydney | copy run start. Then reload. Just to check :-) |
03:15.24 | WIMPy | ChrisInSydney: So you should be able to understand the talking clock now :-) |
03:15.34 | dijib | that should do it |
03:15.44 | ChrisInSydney | WIMPy: I'll give it a go |
03:16.54 | ChrisInSydney | 1191 Zeitansage ? |
03:17.08 | WIMPy | google fail? |
03:17.10 | WIMPy | yes |
03:17.34 | WIMPy | Oops. Typo. |
03:17.49 | ChrisInSydney | :-) |
03:18.24 | WIMPy | I guess CHANNEl works better than CAHNNEL. |
03:18.42 | dijib | dfgas-cr48: try 'make distclean' from /asterisk-11.0.xxx/res/pjproject then go back to root and remake |
03:18.53 | dijib | it was something to do with that res/pjproject |
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03:20.33 | dfgas-cr48 | dijib, alrighty |
03:21.18 | slav3_kitten | any easy cli to show the version on 1.8? |
03:21.20 | dijib | dfgas-cr48: http://lists.digium.com/pipermail/asterisk-dev/2012-September/056899.html is where i got the res/pjproject |
03:21.33 | slav3_kitten | ChrisInSydney, it's been running and i get the iax in from what i can tell |
03:21.46 | WIMPy | core show version |
03:21.52 | dijib | i think it was through a make clean , ./configure , make menuselect , make , make install |
03:21.56 | dijib | possibly |
03:22.06 | dijib | after the pjprojects |
03:22.07 | slav3_kitten | thanks WIMPy |
03:22.10 | WIMPy | Interesting how google changes the numbers. |
03:23.12 | WIMPy | After translation **61*xxx**sek# becomes **61*xxx#sec** |
03:24.31 | ChrisInSydney | WIMPy: Im getting a mixture of Alison Smith and some German chick |
03:24.34 | slav3_kitten | svn checkout http://svn.asterisk.org/svn/asterisk/trunk asterisk < should get version 11 right? |
03:24.55 | WIMPy | Sounds interesting. |
03:25.01 | ChrisInSydney | anyway WIMPy. Thanks heaps. I'll keep going |
03:25.15 | WIMPy | slav3_kitten: More than that. |
03:25.26 | ChrisInSydney | WIMPy. Yep. I get the "Im sorry thats not a valid extension" |
03:25.30 | ChrisInSydney | anyway |
03:25.36 | ChrisInSydney | Thanks heaps |
03:25.38 | slav3_kitten | WIMPy, more then that? |
03:25.38 | ChrisInSydney | :-) |
03:26.03 | WIMPy | I can see that SayDigits() won't succeed with your caller ID. |
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03:50.18 | rue_mohr | if I needed a 6 or 8 channel "channelbank" of mixed fxo fxs what would you suggest |
03:50.26 | rue_mohr | box or card |
04:00.11 | slav3_kitten | WIMPy, upgrading from 1.8 to 11, just make and make install everything right? |
04:11.49 | ChrisInSydney | WIMPy: It skips the non numeric characters |
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04:13.47 | slav3_kitten | i think oyu killed him ChrisInSydney |
04:14.28 | ChrisInSydney | oops |
04:15.06 | rue_mohr | is there a way to check the voip audio levels? |
04:15.17 | rue_mohr | say, in the rtp stream? |
04:15.51 | rue_mohr | _realtime_ |
04:16.24 | rue_mohr | instead of hours of recording 1mw tones, loading them into an app, and readjusting gains |
04:26.24 | Nivex | rue_mohr: http://www.voip-info.org/wiki/view/Asterisk+cmd+Milliwatt ? |
04:26.39 | dijib | dfgas-cr48: whats going on over there |
04:28.45 | rue_mohr | I know I c an generate a 1mw tone |
04:29.06 | rue_mohr | I use to use the providers tone and check the levels between the gain amps on the rtp stream |
04:29.14 | rue_mohr | I have evil polycom phones |
04:33.38 | ChrisInSydney | c yaz all later |
04:33.47 | dijib | gday |
04:38.46 | dijib | i think i killed you slav3_kitten |
04:39.07 | slav3_kitten | i am dead |
04:40.07 | dijib | i knew it |
04:40.18 | dijib | are you good in linux? |
04:40.25 | dijib | or have you just been toying with asterisk |
04:40.31 | dijib | + asterisk related issue |
04:40.46 | dijib | ISP blocks port 25 |
04:41.05 | dijib | have a intranet smtp server and allow relay. |
04:41.14 | dijib | the relay does not allow .wav |
04:41.22 | dijib | to be send as attachment |
04:41.26 | dijib | sent |
04:41.31 | slav3_kitten | dijib, i'm decent with linux but i'm no good with smtp server crap |
04:41.38 | dijib | :D |
04:41.55 | dijib | welcoe to the world of home asterisk |
04:51.35 | dfgas-cr48 | dijib, well i got it farther, i am now installing freepbx |
04:52.07 | dijib | DO NOT INSTALL FREEBOX |
04:52.11 | dijib | effin hell |
04:52.26 | dfgas-cr48 | how do i know if asterisk compiled with sip support, because typing core show help does not show anything with sip |
04:52.27 | dijib | wait or is that pbx in a flash |
04:52.37 | dijib | sip show registry |
04:52.52 | dfgas-cr48 | sip show registery does not work either |
04:53.00 | dijib | here people if you want my help join the conf if you can, & teaviewer me |
04:53.39 | dfgas-cr48 | Laptop*CLI> sip show registry |
04:53.41 | dfgas-cr48 | No such command 'sip show registry' (type 'core show help sip show' for other possible commands) |
04:53.41 | dfgas-cr48 | Laptop*CLI> |
04:57.14 | dijib | well then youve got.... possibly permission isues |
04:57.43 | dfgas-cr48 | ughhhh |
04:58.02 | dijib | what did you do with make menuselect |
04:58.15 | dijib | do anything funny in there or just eep the basic + mp3? |
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05:03.45 | dfgas-cr48 | basic with mp3 |
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05:23.43 | slav3_kitten | is there anyway to forbid dialing out on an interface? |
05:23.58 | slav3_kitten | err channel |
05:24.01 | slav3_kitten | thing |
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05:40.11 | Kobaz | so is it bad when ast_db_put deadlocks |
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05:52.52 | navaismo | ~nat |
05:52.52 | infobot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
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06:57.23 | slav3_kitten | Lucent 8411D anyone ever use? |
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07:18.57 | bulkorok | hi |
07:20.45 | v0lZy | Hi. |
07:20.56 | dijib | no |
07:21.08 | dijib | im working on dfgas-cr48 asterisk deployment |
07:21.22 | dijib | and i think im going to barf |
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07:32.14 | ectospasm | looks like you already did |
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08:08.27 | dijib | all over you ectospasm |
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08:51.56 | unicron | dijib: still awake? |
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09:04.32 | mirela666 | Can a channel be hanged if i know SIPCALLID or BRIDGEDPEER or any other var than CHANNEL(name) |
09:04.34 | mirela666 | ? |
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09:16.01 | freckle | mirela666: CHANNEL REQUEST HANGUP <SIP CHANNEL> |
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09:21.52 | hwt | good morning. if I were to implement a failover scheme (to the outgoing partner) using an AGI script, what would be the best way to proceed? |
09:22.04 | hwt | fetching dialstatus and dialing over, or is there a smarter way to do this? |
09:22.19 | hwt | (basically this, but in AGI: http://mikepultz.com/2010/05/automatic-dial-resource-fail-over-in-asterisk/) |
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09:24.14 | luneff | hey guys. i'm trying to use ARA here with mysql and i can't get calls to ARA SIP users. I can make a call as a ARA user, but when i try to dial ARA number, i get bad-number message. i guess, the trouble lies within nonconfigured realtime switch, but i can't get the thing right :-( |
09:27.14 | R1ck | hey. I've used Trixbox CE in the past to get an "Out of the Box" Asterisk PBX running but it seems that Trixbox CE won't be maintained anymore. Can anybody recommend a different Asterisk PBX solution? |
09:28.28 | ghost75 | gemeinschaft |
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09:34.05 | kaldemar | R1ck: try asterisknow |
09:35.02 | hwt | is it perhaps easier to just dump the call into a macro from AGI and let that one sort it out? i would of course need to supply it with the two (or more) SIP peers it should try |
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09:38.35 | kaldemar | hwt: why are you using AGI for that? |
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09:44.34 | WIMPy | ghost75: That's dead as well. At least as far as Asterisk is concerned. |
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09:48.25 | ghost75 | dead? |
09:50.13 | hwt | kaldemar: it is a generic SIP router for SIP trunking. all routing decisions are fetched from a RESTful API. |
09:50.20 | hwt | kaldemar: also call state, actually |
09:50.59 | hwt | kaldemar: 1. get incoming call, 2. check if source is authenticated and tell where to route it, 3. try primary IP address, and if it fails 4. try secondary IP address |
09:51.03 | hwt | something like that |
09:53.52 | kaldemar | hwt: still, why are you using AGI? |
09:54.38 | bulkorok | ghost75: Macro() is depricated; Use GoSub() instead with RETURN() |
09:55.06 | kaldemar | checking DIALSTATUS is the way to approach this. |
09:55.23 | ghost75 | dont tell me |
09:56.06 | kaldemar | ghost75: anything that still uses old versions like 1.6.X should not be recommended to anyone. |
09:57.10 | ghost75 | what is freepbx using |
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10:02.45 | kaldemar | freepbx is originally just the GUI, but they do have a full linux distro with asterisk and freepbx nowadays. |
10:03.50 | ghost75 | i see gemeinschaft changed to freeswitch |
10:04.09 | bulkorok | don't use gemeinschaft |
10:04.42 | bulkorok | there is only one main programmer... |
10:05.04 | kaldemar | the freepbx distro has a version of 1.8 branch, i guess. |
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10:44.37 | WIMPy | Interesting to see how many people are looking for hfcs-usb support in dahdi. |
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11:04.19 | hwt | kaldemar: because i don't want to embed scripts and stuff in the dialplan? |
11:04.27 | hwt | kaldemar: i need to talk to this restful api |
11:04.35 | hwt | kaldemar: and you can't do that from the dialplan |
11:05.14 | hwt | kaldemar: from this IP address i get the request uri user, the hostname, transport type, port, etc. |
11:05.27 | hwt | kaldemar: so i don't have any peers defined (allowguest=yes) |
11:05.38 | hwt | kaldemar: if there is an uknown IP or user trying to call, i just deny the call |
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11:09.12 | asteriskATmarmuD | any idea on how to reducing the volume of the ringtone heard by the caller? |
11:09.43 | ghost75 | turn volume down on phone? |
11:10.01 | kaldemar | asteriskATmarmuD: what kind of a phone? |
11:10.10 | asteriskATmarmuD | :) this is not an option, since it would reduce the overall volume |
11:10.20 | WIMPy | Assuming you're talking SIP it's probably coming from the phone itself. |
11:10.39 | WIMPy | So that's the only option. |
11:10.45 | asteriskATmarmuD | <PROTECTED> |
11:10.59 | freckle | anyone know if it is possible for another channel to PauseMonitor a different monitored channel? |
11:11.05 | WIMPy | Ok, the gateway then. |
11:11.18 | kaldemar | asteriskATmarmuD: configuring the gateway is pretty much your only choice. |
11:11.49 | asteriskATmarmuD | WIMPy: ok, I guess I can't do anything about it, since the gateway only allows to tune the general volume |
11:12.01 | asteriskATmarmuD | thx guys for reassuring |
11:12.34 | WIMPy | You can do the hack of answering the call before passing it on. |
11:13.32 | kaldemar | freckle: you'd have to do that via AMI. |
11:14.09 | hwt | kaldemar: let me turn the question around, how would you design it? |
11:15.54 | kaldemar | hwt: depends on how your API is used. |
11:16.45 | hwt | kaldemar: it's an HTTP service with JSON data. i query it when an INVITE comes in, and Dial() based on data I get in the response to the API |
11:17.07 | hwt | kaldemar: if it's a known IP and user, I forward it to the PSTN. if not I deny it. if it comes from PSTN, I forward it to the correct user. |
11:17.12 | kaldemar | nothing stops your from using dialplan for that. |
11:17.23 | WIMPy | hwt: How many calls/time? |
11:17.26 | freckle | kaldemar: thanks |
11:18.27 | kaldemar | using func CURL is one possibility. |
11:18.37 | hwt | WIMPy: nothing frightening. perhaps 5-10 calls per second |
11:19.18 | WIMPy | I wouldn't use anything that forks anyway. |
11:19.49 | hwt | kaldemar: yes, i know it's possible. it would just be very complicated to change the logic. i need a lot of if/elses, and might need to add more advanced options later on. also, i'm more comfortable in python. you haven't really convinced me it's a problem using AGI for this. ;) |
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11:20.06 | hwt | WIMPy: agree, it's ugly, but i don't have that many options if i want to use ast. |
11:20.24 | hwt | WIMPy: i wanted to use sippy for this, actually. but the decision wasn't mine. |
11:20.37 | WIMPy | So you don't get an result from your http request but have to do lots of processing still? |
11:20.57 | kaldemar | hwt: i'm not convincing you that it's a problem using AGI for that. it's just that AGI is often used even when there is no need whatsoever for it. |
11:21.19 | hwt | WIMPy: if the API is down, i handle this in a different way. |
11:21.51 | hwt | WIMPy: advantage here is that i can use an absolute timeout (from urllib2), which i guess will be hard with func CURL. |
11:22.13 | WIMPy | is AMI fan |
11:22.26 | hwt | kaldemar: ok. |
11:22.33 | hwt | WIMPy: would AMI fit here? |
11:22.51 | WIMPy | You can do anything with it. |
11:23.19 | hwt | *reading* |
11:23.33 | hwt | WIMPy: because i too am worried about the scaling of forking a script for every call |
11:23.47 | hwt | WIMPy: plus, i have experience with scripts not terminating properly |
11:25.56 | hwt | WIMPy: so basically you can set it up to send an event over the manager interface to my program when a call enters, then pass back Command statements to forward that particular call? |
11:26.44 | WIMPy | You don;t even have to send the event yourself. You just have to enable them. |
11:27.00 | kaldemar | hwt: btw, you can configure curl usage with CURLOPT. |
11:27.20 | WIMPy | You just have to make sure your call waits in the dialplan for your application to direct it elsewhere. |
11:27.59 | hwt | kaldemar: okay, cool. however, i think the logic will be to complex to make it maintainable in extensions.conf. AEL could possibly work, but then I need to learn that first. ;) |
11:28.30 | kaldemar | hwt: that is most likely just a matter of opinion. :) |
11:28.47 | hwt | WIMPy: and you can also add headers, change transport, port, host, etc, and catch dial status through it? |
11:29.08 | hwt | if anyone has some good literature on getting started with AMI (voip-info sucks) it would be appreciated |
11:29.25 | WIMPy | I haved never tried to change transport in realtime. |
11:29.47 | gusto | hi WIMPy |
11:30.01 | hwt | WIMPy: you can do that with Dial(). if you get an incoming call over UDP, it's not a problem to forward it using TCP or even TLS |
11:30.04 | WIMPy | But getting info about dialstatus works very well that way. |
11:30.13 | freckle | kaldemar: I am sending this "Action: PauseMonitor SIP/mychan-0002475f" but it says invalid command. Any ideas? |
11:30.22 | kaldemar | hwt: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-AMI.html |
11:30.25 | WIMPy | hwt: Then you're fine. |
11:30.32 | WIMPy | Moin gusto |
11:30.46 | hwt | WIMPy: any limitations I should be aware of? |
11:30.51 | kaldemar | freckle: that is not a valid command. :P |
11:31.19 | kaldemar | freckle: see what "manager show command PauseMonitor" tells you. |
11:31.20 | WIMPy | hwt: Don't think so. |
11:31.50 | freckle | kaldemar: ty |
11:32.13 | hwt | thanks, guys! |
11:42.06 | gusto | well |
11:42.13 | gusto | WIMPy: what's new? |
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11:58.10 | mirela666 | Can I hangup channel if I don't know CHANNEL(name), if i have something else valuble, maybe bridgedpeer,... |
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12:01.19 | kaldemar | mirela666: how are you trying to perform the hangup? in dialplan? |
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12:05.48 | fabsoft | hi all |
12:07.59 | fabsoft | is there a way to fix called number in a macro routine? i need to call this macro with EXTEN which has some pattern and return to caller stack at new fixed EXTEN |
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12:08.23 | WIMPy | Goto |
12:12.06 | fabsoft | WIMPy: how i'll able to return to the caller context ? |
12:12.34 | fabsoft | i'd pass it as argument ? |
12:13.13 | WIMPy | Goto(${CONTEXT},newexten,1) |
12:13.16 | fabsoft | eg: Macro(fixDID,thiscontextname) |
12:14.31 | fabsoft | he calling extension, context, and priority are stored in ${MACRO_EXTEN}, ${MACRO_CONTEXT} and ${MACRO_PRIORITY} respectively. .. ok :) |
12:14.38 | mirela666 | kaldemar: yes |
12:15.59 | mirela666 | kaldemar: basicly idea is that i create channel to another pbx with originate, and i would like to hang up that Originated call on my hangup |
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12:17.24 | mirela666 | depending on time of my hangup, that outgoing channel might not yet be created, so i guess in that case it's impossible |
12:17.40 | mirela666 | well created but not bridged |
12:19.58 | mirela666 | SoftHangup(SIP/<resource>,a) is good but hangs all calls to that pbx |
12:20.39 | kaldemar | mirela666: func CHANNELS() might help |
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12:22.54 | mirela666 | kaldemar: thanks, that might just help :) |
12:23.58 | kaldemar | would be nice if Originate set a variable with the channel for the originated call name. |
12:24.23 | mirela666 | yes I agree, to know who created it |
12:25.04 | mirela666 | or whos child is it :) |
12:26.37 | mirela666 | I tried listening with AMI Events but I get Channel name only when originate gets response |
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12:51.24 | hwt | ok, so i have the basic manager daemon running. it's currently just displaying all events. |
12:51.48 | hwt | how should my extensions.conf look like if i want to leave all routing decisions to AMI? |
12:53.11 | WIMPy | You need to keep the call somewhere until your app reacts. |
12:54.33 | hwt | WIMPy: allright, so just do an _X.,1,Wait(10) or whatever |
12:55.21 | WIMPy | For example. |
12:55.29 | hwt | WIMPy: and then issue Command statements? |
12:55.55 | hwt | WIMPy: or call out, then bridge? |
12:56.07 | hwt | sorry for all the stupid questions |
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12:58.04 | WIMPy | The easiest way might be to set up some variables and then redirect to an extension that dials to those variables. |
12:59.11 | hwt | WIMPy: so i set some ${DIALSTRING} from AMI, then in extensions i have 1: if $dialstring, dial $dialstring, else goto 1? |
13:02.02 | WIMPy | I thought more like doing Dial({destvar},${time},${opts}) |
13:02.20 | hwt | WIMPy: yes, but i can't know when this string is set. |
13:02.50 | WIMPy | You set it befor sending the call to that exten. |
13:06.27 | hwt | WIMPy: yes, but i don't want to have an unnecessary wait(n) |
13:06.49 | hwt | i want it to trigger the Dial somehow after the {destvar} contains anything meaningful |
13:07.02 | WIMPy | You don;t have to wait. |
13:07.13 | WIMPy | You set up the vars and then do a redirect. |
13:08.09 | WIMPy | So basically what I do is a Wait() and then somethig to execute in case the script fails. |
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13:08.31 | hwt | WIMPy: oh, so Wait will just wait until you "pick it up" with a redirect? |
13:08.45 | WIMPy | That's the idea. |
13:08.52 | WIMPy | Just a timeout for the app. |
13:08.59 | hwt | aha |
13:11.09 | hwt | well, that leaves for a very compact dialplan. ;) |
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13:17.15 | hwt | hm, docs says you can't run Wait without arguments |
13:18.23 | WIMPy | Just put as meny seconds there as you think your app will need at maximum. |
13:19.04 | hwt | WIMPy: aha, then we were talking around each other. i don't find it acceptable even with a 1 second wait, if it takes 0.002 seconds for the API query to run |
13:19.45 | hwt | WIMPy: maybe a Ringing() will be slightly better, but i don't like it really |
13:20.09 | WIMPy | It doesn't matter. When you tell it to redirect, it will do so immediately. |
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13:20.26 | hwt | WIMPy: oh, it does. then it's good. |
13:20.30 | WIMPy | No, you need something that waits. |
13:21.18 | hwt | that seemed to work. i guess Uniqueid is the best to use as an identifer? |
13:21.41 | hwt | per channel |
13:21.45 | WIMPy | Or the channel name. |
13:21.54 | hwt | the uniqueid is shorter :) |
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13:52.16 | WIMPy | You need the channel name anyway. |
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13:53.42 | hwt | WIMPy: okay, good point |
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13:55.00 | Sanon | good morning all |
13:55.31 | Sanon | I'm having a bit of an issue with setting up softphones |
13:55.58 | Sanon | I have some settup just fine on my smart phones |
13:56.18 | Sanon | but the pc's are acting strange |
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14:01.34 | Sanon | I downloaded sflphone, put in all the right info |
14:01.40 | Sanon | but nothing |
14:02.38 | Sanon | i have 3cx phone and zoiper on my android devices working just fine |
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14:06.22 | Sanon | is anyone here? |
14:07.50 | WIMPy | You need to be more specific. |
14:10.01 | jmetro | i am neither here nor there |
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14:12.21 | ghost75 | some whee |
14:12.26 | ghost75 | where |
14:12.40 | jmetro | over the rainbow? |
14:13.26 | Sanon | ok. can anyone tell me how to set this sflphone thing up |
14:13.59 | WIMPy | The manual? |
14:15.40 | [TK]D-Fender | BAI BAI |
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14:21.43 | jaytee | I'm at a client site with a new Asterisk installation to replace an old 3Com VCX system that died. They have a T1 PRI and inbound and outbound calls to a landline sound clear and work fine but the system seems to have issues with inbound calls from cell phones breaking up and sometimes dropping or sometimes not accepting the DTMF tones when the caller dials an extension from the IVR menu. |
14:21.44 | jaytee | This is a very rural area but there is a Verizon 4G cell tower nearby. Not sure what I can check or test at this time to pinpoint the problem. Anyone have any suggestions? |
14:24.07 | [TK]D-Fender | jaytee, Make sure your EC is stable and that your gains aren't distorting things. Then also try "relaxdtmf=yes" for your channels and see if that loosens it up. |
14:24.46 | jmetro | i have a client who gets called on cell all the time and always complains about their inbound calls but outbound are perfect- cell calls are just awful imo and for the most part you cant be responsible if the inbound caller sets up a terrible connection. |
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14:38.56 | jaytee | [TK]D-Fender, I have the TE122B with the hardware echo cancel module. I don't have relaxdtmf set though. IIRC, that will require a restart, not just a reload. |
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14:40.57 | [TK]D-Fender | jaytee, A reload of the channel module, yes |
14:41.12 | [TK]D-Fender | Whichw ill kill calls over it |
14:41.17 | jaytee | Both txgain and rxgain are set to 0.0, wonder if kicking up the rxgain would help? |
14:41.45 | WIMPy | It surely would make things worse. |
14:42.36 | [TK]D-Fender | jaytee, Shouldn't have to. Kicking up gains tends to cause distortions... |
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14:49.54 | ghost75 | i can replace macro(bla) straight with gosub(bla) ? |
14:50.54 | [TK]D-Fender | ghost75, Depending hwhat you use to return, in general, yes |
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14:52.43 | P-NuT | Hi all, has anyone had any luck getting cisco 7940 xml configuration working? I've been googling for days but can only find references to the old configs or patchy documentation on the current examples. |
14:52.58 | P-NuT | Does anyone have a real and actual xml config example for one? |
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14:56.04 | ghost75 | does it make sense to use g.711 from ata to asterisk and g.729 on trunk ? |
14:56.47 | [TK]D-Fender | ghost75, Could be. |
14:57.13 | ghost75 | but then it would be translated |
15:01.42 | [TK]D-Fender | ghost75, Clearly. |
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15:31.31 | xoveruk | hi |
15:31.40 | xoveruk | I am having issues with my FXO card, I cannot dial out, but can dial in |
15:31.48 | xoveruk | How can I diagnose the problem? |
15:31.51 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
15:32.03 | [TK]D-Fender | xoveruk, Like everything else... * CLI |
15:32.12 | [TK]D-Fender | xoveruk, And I dunno ... looking at your configs. |
15:32.17 | xoveruk | If I run dahdi show channel 1 I get offhook |
15:32.24 | xoveruk | config should be ok, it normally works |
15:37.57 | xoveruk | can you advise? |
15:38.50 | drmessano | Look at the CLI output of a failed call, for starters |
15:42.24 | xoveruk | -- Called g0/xxxxxxxxxxxxxx |
15:42.24 | xoveruk | <PROTECTED> |
15:42.27 | xoveruk | that is it |
15:42.31 | xoveruk | nothing happens |
15:43.44 | *** join/#asterisk and7ey (~Miranda@95-25-59-202.broadband.corbina.ru) |
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15:45.58 | and7ey | hi, is something wrong with this dial plan - http://pastebin.com/uLrViPhC ? I expect 4 digit calls to be routed within network and all others - to `beeline` |
15:50.32 | [TK]D-Fender | and7ey, exten => _X.,n,Dial(SIP/beeline/${EXTEN}) <-- you have no priority *1* |
15:50.41 | [TK]D-Fender | and7ey, You can't just start with "n" |
15:51.59 | [TK]D-Fender | (which would imply being related to a previous extensions priority which was a different pattern anyway) |
15:58.38 | and7ey | [TK]D-Fender: aah, priority shhould be 1 for new pattern.. ok, thanks! |
16:00.41 | kaldemar | and7ey: note that XXXX is a better match than X. for 4 digits. |
16:02.20 | [TK]D-Fender | kaldemar, He laready has that |
16:02.23 | [TK]D-Fender | already* |
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16:02.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
16:05.20 | and7ey | [TK]D-Fender: is that correct? http://pastebin.com/mqMvhkk8 |
16:06.04 | and7ey | looks like it tries to dial both SIP/${EXTEN} and SIP/beeline/${EXTEN} |
16:08.40 | [TK]D-Fender | and7ey, those are both priority 1. That can't both execute. |
16:09.08 | jmetro | well, anything that matches 4 digits will hit the first, and every other call with match the second, so they are independent yeah. |
16:09.09 | [TK]D-Fender | and7ey, And only one of those patterns leads to an explicit Hangup. |
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16:11.55 | and7ey | [TK]D-Fender: here is new version - http://pastebin.com/xVFzhFqn |
16:12.19 | [TK]D-Fender | and7ey, What do ouside phone #'s really look like? |
16:12.50 | and7ey | [TK]D-Fender: for ex., 79169801234 |
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16:13.11 | and7ey | [TK]D-Fender: or +79169801234 |
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16:13.54 | [TK]D-Fender | and7ey, perhaps you should make the pattern that lets you dial those out be a little more specific than the pattern you're using.... |
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16:48.53 | Sanon | not much activity going on here. |
16:49.18 | Sanon | as you all know, i'm new to this and have a lot of questions. |
16:50.04 | wdoekes | ~ask |
16:50.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
16:50.05 | Sanon | here is one. how can i get my system accept any sip or iax2 softphone |
16:50.44 | Sanon | i've downloaded many diff. ones, none work |
16:50.58 | Sanon | but all my android devices work |
16:51.27 | jmetro | if you register them properly they all should work. maybe 2 or 3 lines of code and I got my 3cx softphone online |
16:52.53 | Sanon | ok. i have Ekiga on linuxmint right now. i put the information from my sip.conf in, but it's not registering |
16:53.41 | jmetro | did you sip reload |
16:54.16 | Sanon | yes. i reload |
16:54.39 | jmetro | is the phone pointing to the correct IP address [your asterisk box] |
16:54.47 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
16:54.58 | jmetro | ala "outbound proxy" "sip registrar" etc |
16:55.14 | *** join/#asterisk jblack (~jblack@93.sub-70-192-150.myvzw.com) |
16:55.17 | Sanon | they all are correct. |
16:56.40 | navaismo | show us the cli output with the sip debug |
16:56.43 | jmetro | set your console verbose and debug levels high enough to see any incoming requests? make sure you can ping the * box from the linuxmint box |
16:56.56 | jmetro | core set verbose 9999 core set debug 9999 |
16:57.05 | jmetro | sip set debug on |
16:57.14 | navaismo | ?_? level 3 is enough |
16:58.07 | Sanon | ok |
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17:01.58 | navaismo | use pastebin to show us the debug ~pb |
17:02.02 | Sanon | ok, i tried to register the information for my computer on my android device and it rejected it |
17:04.25 | [TK]D-Fender | "sip set debug on" <---- if you aren't looking at SIP debug then you are only fooling yourself. |
17:07.17 | *** part/#asterisk freckle (~jon@firebox.enta.net) |
17:13.58 | Snivets | Is there an easy way to boost volume across a sip channel? |
17:14.05 | Snivets | Are certain codecs louder than others? |
17:14.16 | leifmadsen | Snivets: VOLUME() |
17:14.25 | Snivets | hmm |
17:14.54 | leifmadsen | codecs should not be "louder" than others no |
17:15.03 | leifmadsen | the amplitude that is recorded should be the same throughout |
17:15.07 | Snivets | well, wait a second. So I installed a snom m9 handset yesterday, and it is really damn quiet, even with the unit's volume up and the other party's mic set higher too |
17:15.13 | leifmadsen | since it's just a digital representation of analog data |
17:15.21 | Snivets | k, that makes sense |
17:15.29 | leifmadsen | Snivets: then that's a problem with the device itself driving the analog data |
17:15.48 | Snivets | Wouldn't boosting the volume to that particular client's channel be an option? |
17:15.56 | leifmadsen | hence VOLUME() |
17:16.08 | Snivets | Fair enough, now where does that go? lol |
17:16.15 | leifmadsen | in the dialplan |
17:16.22 | leifmadsen | VOLUME() is a dialplan function |
17:16.24 | Snivets | yes, but wherein? |
17:16.26 | leifmadsen | 'core show function VOLUME' |
17:16.35 | leifmadsen | Snivets: depends what you're trying to do and where it should go |
17:16.40 | Qwell | It's too bad there isn't some sort of book that has this stuff in it. |
17:16.41 | leifmadsen | suggests before Dial() at the very least |
17:16.47 | leifmadsen | Qwell: IKR?! |
17:17.04 | Qwell | I bet it would sell a HUNDRED copies. |
17:17.16 | Qwell | Think of how many nickels you'd get. |
17:17.20 | Snivets | well, realistically, I should be looking at voip info |
17:17.27 | Qwell | No you shouldn't. |
17:17.30 | Snivets | oh ya? |
17:17.31 | Qwell | ~asterisk wiki |
17:17.31 | infobot | somebody said asterisk wiki was http://wiki.asterisk.org/ |
17:17.48 | Snivets | oh snap, I see,using the set function |
17:18.00 | Qwell | application* |
17:18.20 | Snivets | "Search Confluence" |
17:18.40 | Snivets | Rusty Newton looks kinda like a dog, guys. |
17:18.58 | mjordan | Snivets: we'll let him know |
17:19.04 | Qwell | newtonr: ^ |
17:19.05 | mjordan | newtonr: you look like a dog. |
17:19.16 | Snivets | aww, yeah, we look out for each other. |
17:20.05 | *** join/#asterisk blee (~blee@72.188.117.219) |
17:20.52 | Snivets | I'm getting a Windows phone today |
17:20.55 | Snivets | what are our thoughts on that? |
17:21.00 | Qwell | Why? |
17:21.06 | Snivets | I think that about sums it up, yeah |
17:21.12 | coppice | my sympathies |
17:21.13 | Robotman321 | Congratz Snivets :p |
17:21.14 | jmetro | they look interesting. |
17:21.23 | Snivets | I'm getting the 920 |
17:21.27 | Robotman321 | I have a HTC Trophy, never had an issue |
17:21.54 | Qwell | Trophy is a fantastic name for a paperweight. |
17:22.02 | jmetro | samsungs make the best mobile devices on the market |
17:22.07 | Snivets | so, I've been telling my nerd friends that the Nokias, pre-MS takeover, were intended to run the full on linux, you know? And I keep messing up the name and calling it "sybian" instead of "symbian" |
17:22.11 | coppice | I wonder if nokia will be able to sell phones for more than $200 now |
17:22.13 | jmetro | but i dont think there are any samsung windows pones. |
17:22.19 | leifmadsen | I'm not sure what this flametopic has to do with asterisk |
17:22.27 | Sanon | ERROR[6255]: chan_iax2.c:5003 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address "0.0.0.0" in the calltokenoptional list or setting user "USER" requirecalltoken=no |
17:22.31 | leifmadsen | people use phones. it's true. |
17:22.32 | coppice | jmetro: there are some |
17:22.33 | leifmadsen | end of story. |
17:22.43 | *** join/#asterisk serafie (~erin@nat/digium/x-ygaosnwfmkayrixs) |
17:22.48 | Sanon | THIS IS WHAT I GOT THIS TIME |
17:23.03 | Sanon | ERROR[6255]: chan_iax2.c:5003 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address "0.0.0.0" in the calltokenoptional list or setting user "USER" requirecalltoken=no |
17:23.14 | Qwell | Sanon: okay, and? |
17:23.20 | Snivets | Actually, it was going to be related: |
17:23.26 | Snivets | I was wondering if there are any good mobile sip clients |
17:23.27 | jmetro | youre using iax instead of sip? |
17:23.39 | Sanon | using iax |
17:23.51 | Sanon | sflphone |
17:24.09 | Qwell | Sanon: And what happened when you did what the error message told you to do? |
17:24.31 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1005) |
17:24.47 | jmetro | need to edit iax.conf, yes? not sip.conf? |
17:24.52 | Sanon | I tried to register my sflphone to one of my users in ixa.conf |
17:24.57 | Sanon | and i got that |
17:25.00 | Qwell | Sanon: And what happened when you did what the error message told you to do? |
17:25.16 | [TK]D-Fender | <Qwell> Sanon: And what happened when you did what the error message told you to do? |
17:25.59 | Sanon | how do i do that |
17:26.05 | Sanon | is what i'm asking |
17:26.19 | navaismo | <PROTECTED> |
17:27.25 | [TK]D-Fender | Sanon, it just GAVE you the exact parameters to put in your iax.conf peer entry. What do you mean "how"? |
17:27.33 | [TK]D-Fender | Sanon, PUT THOSE INTO YOUR PEER |
17:27.47 | Sanon | got it. |
17:29.01 | Sanon | thanks guys, for being patient and helping me |
17:29.07 | Sanon | it's working |
17:30.30 | slav3_kitten | son of a bitch... |
17:31.05 | slav3_kitten | got a filling a week ago, eating cold pizza pulled it out. |
17:31.14 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
17:31.46 | slav3_kitten | i'm going to be getting a free filling |
17:34.32 | n3hxs | Sounds like you set the first one free, slav3_kitten |
17:35.18 | [TK]D-Fender | slav3_kitten, Go for Boston creme..... |
17:36.38 | Snivets | that is unfortunate |
17:37.07 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
17:38.16 | jmetro | get the ceramic fillings, they dont come out. I cant even remember what tooth I got that in. |
17:38.26 | Qwell | jmetro: maybe it came out |
17:39.30 | [TK]D-Fender | Don't ask, don't tell |
17:39.38 | Qwell | [TK]D-Fender: was abolished |
17:39.45 | slav3_kitten | jmetro, i'll check itout |
17:39.58 | [TK]D-Fender | Qwell, Not for FILLINGS yet... progress is a slow wheel |
17:47.17 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
17:53.01 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
17:53.57 | jeffspeff | anybody having any issues with confbridge causing an error making asterisk restart? |
17:53.59 | *** join/#asterisk navaismo (~navaismo@189.144.200.37) |
17:54.22 | jeffspeff | just a moment ago, i did confbridge list |
17:54.33 | jeffspeff | it showed 2 bridges open |
17:54.50 | jeffspeff | i did confbridge list bridge1 and it showed a few users in the bridge |
17:55.05 | jeffspeff | i did confbridge list bridge2 and asterisk died |
17:55.37 | Qwell | Did you search JIRA? |
17:56.12 | jeffspeff | no |
17:56.14 | jeffspeff | i'll go there now |
17:57.22 | mjordan | jeffspeff: what version? |
17:57.27 | jeffspeff | 11.0 |
17:57.50 | mjordan | kk. If you can reproduce it and get a backtrace, please open a new issue. I'm pretty sure I haven't seen any crashes reported from CLI commands interacting with ConfBridge |
17:58.00 | mjordan | a quick peek in Jira would be appreciated however :-) |
17:58.52 | ghost75 | i had once zement filling for like 15 years |
17:58.53 | jeffspeff | yes, looking in jira now |
17:59.04 | jeffspeff | it's a production box, so i'll try and replicate it on a test box |
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18:38.09 | oneadvent | hi i'm trying to do a phpagi script but my log says: Starting SIP/vitel-outbound-00000023 at makeCall,s,1 failed so falling back to exten 's' |
18:38.23 | oneadvent | does that mean it cannot find makeCall context? |
18:38.32 | oneadvent | (generated by call files) |
18:38.50 | [TK]D-Fender | It cannot find a match for that exten in that context at priority 1. |
18:38.53 | [TK]D-Fender | jsut like it says |
18:40.12 | oneadvent | [groupCall] |
18:40.12 | oneadvent | exten => s,1,Answer |
18:40.12 | oneadvent | exten => s,2,AGI(makeCall.php) |
18:40.13 | oneadvent | exten => s,3,Hangup |
18:40.19 | oneadvent | is at the bottom of extensions.conf |
18:40.29 | oneadvent | i'm not sure what else i need for it :( |
18:40.48 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
18:44.31 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
18:44.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
18:46.56 | oneadvent | anywhere i should read up on this [TK]D-Fender ? I'm lost, i have everything in the agi-php directory and the call file calls out...it is like it can't find my script but the error says it can't find the context, and i'm not sure how to make it find it anymore than putting it in extensions.conf. |
18:47.09 | slav3_kitten | ok maybe i missed it in the book, but when having all phones dial *7 to get to voicemail main. how do i pull the caller id number out to use for automatically throwing them into the correct mailbox? |
18:47.19 | [TK]D-Fender | oneadvent, PASTEBIN your dialplan and call file.... |
18:47.35 | slav3_kitten | wait... |
18:47.39 | [TK]D-Fender | <oneadvent> hi i'm trying to do a phpagi script but my log says: Starting SIP/vitel-outbound-00000023 at makeCall,s,1 failed so falling back to exten 's' |
18:47.43 | slav3_kitten | nvrmind |
18:47.44 | [TK]D-Fender | <oneadvent> [groupCall] |
18:47.59 | [TK]D-Fender | makeCall != groupCall |
18:48.15 | slav3_kitten | i had it right, apparently my saving of my voicemail password didn't work |
18:48.57 | oneadvent | omg if that was it [TK]D-Fender.... |
18:49.05 | [TK]D-Fender | if? |
18:49.09 | oneadvent | lol |
18:49.20 | oneadvent | well i fixed that but i still just get hung up on :( |
18:49.34 | [TK]D-Fender | ok/fine/sure |
18:51.07 | oneadvent | http://paste2.org/p/2486964 |
18:51.08 | oneadvent | there |
18:51.11 | oneadvent | that is the pastebin |
18:51.24 | oneadvent | probably me doing something stupid with phpagi |
18:51.31 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
18:53.46 | slav3_kitten | helps to have proper file owner/group |
18:55.21 | [TK]D-Fender | Also helps to enable AGI debug |
18:55.41 | [TK]D-Fender | Also helps to show us that the files are even where * is trying to call them from... |
18:55.54 | [TK]D-Fender | ...with the right permissions... |
18:57.27 | oneadvent | slav3_kitten: that was the problem |
18:57.33 | oneadvent | thanks [TK]D-Fender and slav3_kitten |
18:57.48 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
18:58.22 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
18:58.39 | slav3_kitten | wait it was file permissions oneadvent ? |
18:59.04 | oneadvent | yes sir slav3_kitten i had it owned by root and it should have been asterisk:asterisk, it didn't even occur to me to check |
18:59.59 | slav3_kitten | holy crap i inadvertently solved that |
19:00.11 | slav3_kitten | that was the solution to my voicemail password not getting changed |
19:00.12 | oneadvent | lol slav3_kitten good job doing it though! |
19:00.45 | slav3_kitten | huzzah |
19:04.50 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
19:06.35 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-41-221.user.veloxzone.com.br) |
19:21.29 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
19:22.15 | AkkerKid | Anyone know how to pass channel variables to hylafax? |
19:22.41 | AkkerKid | Or at least where Hylafax sources it's callerID info... |
19:27.49 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
19:32.55 | p3nguin | its |
19:33.59 | AkkerKid | Corrention: Or at least where Hylafax sources it is callerID info... |
19:34.05 | AkkerKid | :) |
19:34.21 | jmetro | correction |
19:34.52 | AkkerKid | Correctshun: Correction |
19:35.10 | AkkerKid | So i guess thats a no then? |
19:35.30 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
19:35.30 | *** mode/#asterisk [+o Qwell] by ChanServ |
19:38.52 | ghost75 | hmm gosub is not a own context? |
19:39.11 | slav3_kitten | huzzah, voicemail setup and mwi correctly triggered |
19:41.19 | [TK]D-Fender | AkkerKid, you channel has its callerid |
19:41.35 | [TK]D-Fender | your* |
19:42.27 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
19:44.35 | ghost75 | is this valid under 1.6: Gosub(bla,s,1) |
19:47.25 | Qwell | ~upgrade asterisk |
19:47.25 | 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. |
19:47.42 | ghost75 | haha |
19:48.28 | jmetro | slav3_kitten the hard part is getting blf's to work. |
19:48.39 | dijib | slav3_kitten: now tell me how to disable mwi |
19:49.11 | dijib | hey p3nguin you would be proud of me I pretty much did an entire * deployment for a guy last night |
19:49.25 | dijib | 5/4 in the bag to boot |
19:50.25 | ghost75 | in incoming context i use gosub and when sub is finished i get: Auto fallthrough, channel 'SIP/arcor_in-00000004' status is 'UNKNOWN' |
19:50.25 | [TK]D-Fender | ghost75, Does it work? |
19:50.40 | ghost75 | wtf |
19:50.49 | [TK]D-Fender | ghost75, Why would we assume that has anything to do with your Gosub? |
19:51.10 | ghost75 | i changed from macro to gosub |
19:51.35 | [TK]D-Fender | ghost75, You are asking for an autopsy without providing the body..... |
19:51.41 | ghost75 | one sec |
19:54.35 | ghost75 | http://pastebin.com/hTvaiFvC |
19:54.46 | ghost75 | do i need another gosub from spamcheck to incoming arcor? |
20:03.36 | *** join/#asterisk k610 (~Instantbi@host-78-129-3-116.brutele.be) |
20:11.06 | [TK]D-Fender | ghost75, .... and the full call.... |
20:12.07 | [TK]D-Fender | ghost75, And you are Goto-ing OUT of that context. I also see no use of the retun app |
20:12.14 | [TK]D-Fender | return |
20:15.28 | ghost75 | how i return to incoming-arcor |
20:16.04 | [TK]D-Fender | ghost75, You are goto-ing into no-mans-land, we do not see the full picture. |
20:16.15 | [TK]D-Fender | ghost75, Your layout is messay and warrants a redesign |
20:16.25 | ghost75 | ok one sec |
20:17.02 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
20:18.59 | ghost75 | http://pastebin.com/8QKHqy80 |
20:20.18 | ghost75 | http://pastebin.com/a7bdaEL5 |
20:21.03 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+cmd+Return |
20:23.25 | ghost75 | thx |
20:29.28 | AkkerKid | If I move a fax call through asterisk into hylafax and change the callerid variables, the hylafax still sees the original CID number... How can I inject MY CID info? |
20:30.53 | [TK]D-Fender | AkkerKid, Show us |
20:34.14 | AkkerKid | Right before I put a fax call into a ring group full of virtual fax machines, I Set(CallerID(number)=${CALLERID(number)}.${LocationCode}) |
20:35.06 | AkkerKid | basically, this should addend the LocationCode (which is a variable I set based on the did a call comes in on) to the end of the callerid number. |
20:35.39 | [TK]D-Fender | AkkerKid, Set(CallerID(number)=${CALLERID(number)}.${LocationCode}) <-- you should be setting the function here..... |
20:35.44 | AkkerKid | but when faxrcvd.conf gets the callerid variables on the other side of hylafax, I get the original CID number |
20:35.56 | [TK]D-Fender | AkkerKid, But seem to have forgotten that it has to be UPPERCASE |
20:36.13 | AkkerKid | function? |
20:36.24 | [TK]D-Fender | Set(CallerID(number) <------------------------- |
20:36.30 | [TK]D-Fender | Your set is bad. |
20:36.35 | AkkerKid | how so? |
20:36.41 | [TK]D-Fender | HAS TO BE UPPERCASE |
20:36.51 | AkkerKid | I disagree. |
20:37.33 | AkkerKid | I have success with CallerID(name) in lowercase. Why should CallerID(number) be any different? |
20:37.55 | [TK]D-Fender | ..... |
20:38.45 | AkkerKid | I'm certain my case is not the issue. I have a few instances of the variable name formatted exactly like that and it works fine. |
20:38.47 | [TK]D-Fender | You are wrong. |
20:39.05 | AkkerKid | I'm just thinking the hylafax gets its CID info from some other variable. |
20:39.15 | AkkerKid | I'll try it. |
20:39.33 | [TK]D-Fender | There is no fact to back up your "thought". |
20:40.14 | AkkerKid | none beside the fact that other instances of that variable work fine. |
20:40.25 | [TK]D-Fender | I'm not seeing backup. |
20:40.46 | AkkerKid | BRB testing. |
20:45.21 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
20:45.57 | ghost75 | there is an predefined extension "fax" in asterisk? |
20:48.36 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
20:50.50 | AkkerKid | thats another thing i'm wondering about... sometimes i'll get a call into my main call sorting code from exten "fax"... |
20:51.00 | AkkerKid | no other CID info |
20:51.24 | ghost75 | then maybe someone calls you with a t.38 fax |
20:52.28 | AkkerKid | i can't even trace the call to see who it may have come from |
20:52.58 | AkkerKid | i was worried that that maybe some rogue function was intercepting my outbound faxes and redirecting them back into the dialplan... |
20:53.12 | AkkerKid | but i haven't seen hard evidence of that |
20:53.31 | ghost75 | no CALLERID ? |
20:53.37 | AkkerKid | nil. |
20:54.04 | AkkerKid | [TK]D-fender: no change between cases. |
20:56.12 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
20:57.43 | AkkerKid | the no callerid on inbound call from "fax" thing may have something to do with elastix rearing it's ugly head. |
21:03.07 | ghost75 | exten => fax,n,System('/usr/bin/fax2mail ${CALLERIDNUM} "${CALLERIDNAME}" FaxNum RecipName email@address.com ${FAXFILENOEXT} p') |
21:03.14 | ghost75 | i couldnt find any fax2mail on linux |
21:03.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
21:09.52 | *** join/#asterisk mdg (6c5df7d9@gateway/web/freenode/ip.108.93.247.217) |
21:10.25 | mdg | Greetings fellow telephony enthusiasts, is there a way to have a pattern matching extension of variable length ? |
21:10.57 | mdg | I know you can use !, but that would not work in coordination with "enter the amount followed by the pound sign" |
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21:12.57 | artyx | mdg . i think |
21:13.25 | artyx | X. for a dialplan could be any # of digits .. but its generally a bad idea |
21:14.25 | mdg | artyx: ah yea, forgot about that from the book. thanks for the reminder |
21:14.57 | artyx | Worsst case you could try a fixed 5 digit #, and say in the script "prefix with 0's , for 100 use 00100" tacky but |
21:16.17 | mdg | true |
21:16.32 | Qwell | err |
21:16.37 | Qwell | Why not use Read? |
21:17.09 | mdg | Qwell: because Cepstral 6 wants 200 bucks to generate sound files :-/ |
21:18.07 | Qwell | How is using Read any different from what you're doing? |
21:18.33 | artyx | i didnt realize he was talking ivr until i scrolle ddown |
21:19.03 | mdg | artyx: I wasnt clear enough in my original question, sorry |
21:19.16 | mdg | Qwell: Im not sure I understand? |
21:19.52 | mdg | On Cepstral 5, I could generate the wav files and use Read, but my hand has been forced in upgrading an IVR to use Cepstral 6 |
21:22.16 | AkkerKid | what's wrong with _X.# |
21:22.40 | AkkerKid | that would at least require a single digit followed by a pound, wouldn't it? |
21:23.42 | mdg | AkkerKid: yea.. I just tested it and it is working the same as _X. - Both work fine, but dont seem to go to the extension untill the swift command times out |
21:23.57 | mdg | which im about to confirm (was just counting in my head) |
21:24.28 | ghost75 | does $NumberCalled still exist in h context ? |
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21:24.47 | Qwell | ghost75: What? Such a thing never existed. |
21:25.21 | ghost75 | CALLERID(num) |
21:25.31 | ghost75 | no wait |
21:25.47 | ghost75 | forget it |
21:26.43 | AkkerKid | what about _X.# followed by something the takes the pount off the end like ${Exten:0:-1} |
21:29.34 | mdg | AkkerKid: thats not the issue (although I still need to do that), I just confirmed that the dialplan wont goto _X.# extension until my swift call times out (i have it set for 10 seconds) |
21:29.52 | mdg | So it does work.. but not ideal having a 10 second delay after hitting # |
21:31.09 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
21:31.28 | AkkerKid | oh i see. |
21:33.22 | mdg | not sure... but it looks like i can modify https://github.com/awayment/app_swift/blob/master/Asterisk_1.8/app_swift.c#L261 to return if '#' is encountered.. since the '#' is always terminating on this ivr |
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21:34.20 | mdg | wonders if cepstral did that on purpose |
21:35.26 | artyx | How do you turn a phone number into an address? |
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21:38.38 | slav3_kitten | if i'm not mistaken if i have a variable with 04 in it an i go VAR:0:1 i'll get 0 an :1:1 i'll get 4 right? |
21:39.58 | mdg | ha... it works |
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21:45.39 | leifmadsen | slav3_kitten: VAR:<offset>:<length> |
21:45.51 | mdg | https://github.com/awayment/app_swift/blob/master/Asterisk_1.8/app_swift.c#L278 right after the digit gets concatenated, check if it == '#' then break |
21:45.54 | leifmadsen | you don't even need the second :1 in your 2nd example |
21:46.08 | leifmadsen | since your length is only 2, and an offset of 1 with no length specified will go to the end |
21:49.19 | slav3_kitten | isn't it better to specify a length however even if it's not needed in in the event somehow there is extra data? |
21:52.47 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
21:54.32 | ghost75 | what is stored in /var/spool/asterisk/fax/${CALLERIDNUM} |
21:54.40 | ghost75 | header from fax ? |
21:56.49 | *** join/#asterisk Sean-Der (~sean@NW1-DSL-74-215-64-154.fuse.net) |
21:56.54 | slav3_kitten | ghos i think calleridnum is depreciated a should be CALLERID(num) |
21:57.01 | slav3_kitten | ghost75, * |
21:57.10 | ghost75 | yes, i just copypasted it |
21:57.17 | Sean-Der | Does anyone know of a service that does 'area code -> current time OR zip code' |
21:57.37 | Sean-Der | I know it won't be perfect, but seems like a cool thing to do |
21:58.37 | slav3_kitten | Sean-Der, you could make a database with area code, and local time . then search it? |
21:59.43 | ghost75 | http://www.voip-info.org/wiki/view/Asterisk+fax#SpanDSPSendingandReceivingFaxeswithAster |
22:00.08 | Sean-Der | slav3_kitten: Yea that would be a lot smarter, just store it in SQLite. Hell it would be small enough I could put it the stack in what ever lang I use to process the data |
22:01.23 | slav3_kitten | store local time as an offset to UTC |
22:01.29 | slav3_kitten | it'd work well |
22:04.49 | ghost75 | dahdi is for t.38 ? |
22:05.34 | *** join/#asterisk navaismo (~navaismo@187.187.96.51) |
22:05.46 | Sean-Der | Ok thanks for the advice! Is there an easy way to change the time in Asterisk per call thread? Or can I only go off server time |
22:08.35 | slav3_kitten | now that's a question i have no idea about |
22:14.43 | ghost75 | why so many commands are deprecated i dont get it |
22:15.05 | Chainsaw | ghost75: This is to generate demand for consultancy and sustain the Asterisk eco-system. |
22:15.52 | Sean-Der | Just wait till 11 comes out.... Dialplan can only be written in whitespace |
22:16.46 | ghost75 | lol |
22:16.55 | slav3_kitten | Sean-Der, you could set the caller name to be like LocalTime: hh:mm |
22:18.21 | slav3_kitten | i have it set the name of my friends based on their number |
22:19.00 | Sean-Der | slav3_kitten: What I am thinking of doing is pipeing out to a program to returns a var of the users current 24 hour time, then (if $time < 700 Good morning) |
22:19.20 | slav3_kitten | ah, that's cool |
22:19.32 | slav3_kitten | i have a shit auto attendant for my system |
22:20.07 | slav3_kitten | i might recode it so the extensions like parents room an such are disabled past certain hours |
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22:23.43 | Sean-Der | I actually have something like you are describing! Teachers set 'no-disturb' hours in a web |
22:23.48 | Sean-Der | interface |
22:24.22 | slav3_kitten | my interface is the old busted ass network hardware interface |
22:24.38 | slav3_kitten | and i'm the admin of it so only i change things |
22:25.01 | Sean-Der | Hahha keep it that way, supreme ruler is the best way to be |
22:25.45 | slav3_kitten | yea my equipment is all old cico |
22:25.46 | slav3_kitten | cisco even |
22:26.59 | Sean-Der | At least you have nice hardware! I know everyone makes duds, but you always hear good things about Cisco. I am more of a software developer than a networking guy though :( I am proud of myself that I got my routing set up for OpenVPN |
22:27.43 | slav3_kitten | :D |
22:27.54 | slav3_kitten | yea it's decent stuff but all very old |
22:28.03 | jmetro | cisco stuff is okay. |
22:28.07 | slav3_kitten | like i have a 7911 as the living room phone, it has 1 way speaker phone :| |
22:28.34 | slav3_kitten | no mic so yea that makes it useless, still got no idea how to transfer calls |
22:29.10 | *** join/#asterisk xnt14 (~xnt14@xceleo.us) |
22:29.13 | slav3_kitten | i have no idea how to do transfers actually |
22:29.36 | Sean-Der | Asterisk CLI only :D |
22:30.07 | slav3_kitten | yea don't even knowhow to do that |
22:30.20 | slav3_kitten | i'm sure it's in my giant book i've had a hard time getting through |
22:31.02 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
22:32.14 | Sean-Der | the 'Asterisk: The Definitive Guide' is a great book |
22:33.07 | Sean-Der | I enjoy flipping around in it, and learn random stuff from it all the time |
22:33.17 | slav3_kitten | it's like getting through Ulysses though |
22:34.36 | Sean-Der | Its not that big! I have actually never read Ulysses :/ I really should though |
22:35.26 | slav3_kitten | it's not bad, just not engaging |
22:39.16 | *** join/#asterisk Alex25 (~kvirc@109.67.191.133) |
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22:48.54 | ghost75 | i need to allow anything in trunk if i want t.38 over sip? |
22:49.44 | Alex25 | I'm an new here, so I have a newbie general question about codecs |
22:50.31 | Alex25 | What's the point of allowing some codecs and disallowing others, or putting codecs in specific order? |
22:50.35 | Alex25 | Won't it be better to allow all codecs by default for all channels, and let Asterisk chose the best one for each call? |
22:50.54 | ghost75 | you tell asterisk the order |
22:51.01 | Alex25 | Maybe you knwo some link which explains this point |
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22:52.21 | jpsharp | Asterisk doesn't know the "best" for a call. |
22:52.35 | [TK]D-Fender | Alex25: because maybe you want to prioritize ULAW for internal calls, but allow G.729 to be native bridged for outbound |
22:52.51 | [TK]D-Fender | Alex25: "best" is based on YOUR priorities. |
22:52.54 | Alex25 | yea |
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22:54.03 | Alex25 | so what if I allow all codecs? |
22:54.40 | Alex25 | won't asterisk chose a good codec by itself? |
22:54.52 | Alex25 | will the call fail? |
22:55.10 | jpsharp | If you allow all codecs, Asterisk will choose the first one that it and the client can agree upon. |
22:55.35 | Alex25 | and what's wrong with that? |
22:56.05 | ghost75 | depends which codecs you want to use |
22:56.13 | [TK]D-Fender | Alex25: You keep assuming "good". It will pick the FIRST thing it agrees on in order, which may not be good for YOU. |
22:56.20 | jpsharp | Nothing, assuming you don't have a codec preference. |
22:56.23 | [TK]D-Fender | [17:52][TK]D-FenderAlex25: "best" is based on YOUR priorities. |
22:56.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
22:58.02 | [TK]D-Fender | Alex25: If you pick your owder wrong * could negotiate each side differently and in the case of licenced codecs like G.729, if you don't have any, or are full, then your call dies. |
22:58.25 | jpsharp | But you can allow all and then someone comes along with a client that decides that it wants to present a SIP call and LPC10 is the first codec in its list, even though it supports ULAW, and your calls will sound like crap. |
22:58.32 | [TK]D-Fender | 'aleStop thinking you can get away without configuring your system and that it'll all just work out. You are asking for your system to bite you in the ass. |
22:58.37 | [TK]D-Fender | Alex25: ^ |
22:58.57 | Alex25 | so why people list a 'priority' for a peer, instead of chosing the BEST for them?? |
22:59.14 | [TK]D-Fender | Alex25: best is BASED on your priorities. |
22:59.15 | Alex25 | the best one, i mean |
22:59.18 | [TK]D-Fender | Alex25: YOUR <------------- |
22:59.21 | Alex25 | instead of a list |
23:00.57 | ghost75 | the best is the first you list |
23:01.18 | Alex25 | ok |
23:01.28 | Alex25 | i'm reading your comments |
23:01.40 | Alex25 | thanks very much |
23:02.11 | Alex25 | i wonder if you know some link which puts it simple |
23:02.19 | [TK]D-Fender | This IS simple. |
23:02.24 | [TK]D-Fender | What's so hard to understand? |
23:02.34 | [TK]D-Fender | YOUR circumstances determine what is "best" |
23:02.35 | ghost75 | just dont put codecs which you dont want, easy as that |
23:02.40 | [TK]D-Fender | There is no formula. |
23:02.43 | [TK]D-Fender | There is no magic. |
23:02.47 | [TK]D-Fender | IT DEPENDS ON YOU |
23:02.54 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
23:03.10 | Alex25 | sure |
23:03.23 | [TK]D-Fender | Alex25: If you have shitty bandwidth then if your calls all use high-BW codecs they'll CHOKE out and your calls will sound like crap and may just drop entirely. |
23:03.33 | [TK]D-Fender | So the best quality will SCREW YOU |
23:04.03 | [TK]D-Fender | If you have lots of BW, then maybe you can afford to use higher quality codecs. |
23:04.38 | Alex25 | is g722 the best codec today for high bw? |
23:04.41 | [TK]D-Fender | It also matters for things like CPU load when recording depending on the format you choose to write in. |
23:04.56 | [TK]D-Fender | Alex25: There are times when "best quality" don't even matter. |
23:05.40 | [TK]D-Fender | Alex25: Each codec also has a transcoding performane hit. This will really add up when recording, or in a conference, etc |
23:05.53 | [TK]D-Fender | Alex25: So time to really look at what you're doing. |
23:06.49 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-spdhnzgaqhrkvxwe) |
23:06.53 | Alex25 | sure. My system is up and working, I just want to optimize it with best codecs. so i must learn this terminilogy.. |
23:06.56 | ghost75 | t.38 goes over alaw/ulaw ? |
23:07.20 | Chainsaw | ghost75: No, normally you switch away *to* T.38 from alaw/ulaw. |
23:07.46 | Chainsaw | ghost75: You do need something that's clear enough to hear the fax CNG tone so you can renegotiate for T.38 |
23:07.58 | ghost75 | but i need to have alaw/ulaw in my trunk enabled then |
23:08.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
23:09.21 | ghost75 | will this passthrough work also over nat ? |
23:10.23 | [TK]D-Fender | ghost75: Same as voice |
23:12.45 | ghost75 | crap if i need to enable g711 in trunk then all calls go over g711 instead g729 |
23:21.05 | p3nguin | Doesn't T.38 still use the exact same RTP stream that a voice call will use? |
23:24.03 | ghost75 | http://what-when-how.com/voip/fax-over-ip-overview-voip/ |
23:24.36 | ghost75 | so no g.711 at all i guess |
23:25.39 | ghost75 | asterisk has udptl |
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23:35.12 | jpsharp | T38 does not use a voice codec. |
23:35.24 | p3nguin | We covered that. |
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