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| 01:05.08 | s7r | <PROTECTED> |
| 01:07.23 | WIMPy | You need some hardware interface. |
| 01:07.34 | s7r | like what? |
| 01:07.40 | s7r | what is the easiest way to do it |
| 01:07.52 | WIMPy | An ATA |
| 01:08.34 | s7r | ATA? |
| 01:08.44 | WIMPy | ~ata |
| 01:08.45 | infobot | i heard ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 01:10.29 | s7r | that's all? |
| 01:11.05 | WIMPy | Yes. An ATA and some peer (server) to connect it to. |
| 01:11.16 | WIMPy | Like e.g. a RPi. |
| 01:12.22 | s7r | WIMPy how do i connect the APA to the RPi? at the ethernet port? |
| 01:13.08 | nix8n82 | Use a cable |
| 01:13.18 | WIMPy | yes |
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| 01:13.45 | s7r | WIMPy and could i use an ATA with 2 telephone landline ports in order to connect 2 phones and ring each other on different numbers? |
| 01:13.57 | s7r | i mean both of them to be able to call between them, not outside this small circuit |
| 01:15.30 | WIMPy | yes |
| 01:16.06 | WIMPy | You might be able to do that with only the ATA, depending on th model. |
| 01:16.19 | s7r | I want the other phone to respond to a particular number |
| 01:18.16 | nix8n82 | s7r what does this has to do with asterisk? |
| 01:18.57 | s7r | I googled what i am trying to do and found lots of links to asterisk, think asterisk is the software i need to run on the server side (RPi) ? |
| 01:19.48 | nix8n82 | Could be.. depending on what you want to accomplish |
| 01:20.42 | s7r | I have 2 fax machines which i want to connect each other using a phone line simulator, so each of them can respond to a custom number |
| 01:22.04 | WIMPy | If you do it with only the ATA, the numbers are most probably fixed. |
| 01:22.25 | s7r | fixed? |
| 01:22.30 | nix8n82 | You probably could get away with just using an ata. |
| 01:22.33 | s7r | but can i assign them? |
| 01:23.13 | WIMPy | No |
| 01:23.21 | WIMPy | Only if you use external software. |
| 01:23.38 | s7r | WIMPy like a raspberry pi and some kind of server app |
| 01:23.43 | s7r | that server app can be asterisk? |
| 01:23.47 | s7r | or i need something else? |
| 01:23.47 | WIMPy | yes |
| 01:23.54 | nix8n82 | Are these fax macines going to talk to the pstn? |
| 01:24.01 | s7r | pstn? |
| 01:24.07 | s7r | they are going only to talk with each other |
| 01:24.11 | s7r | there are 2 building corps |
| 01:24.16 | WIMPy | Ther is other software available, liek SIP proxies. |
| 01:27.30 | WIMPy | And BTW, I haven't touched anything analog in decades. |
| 01:27.34 | nix8n82 | Seems strange but you could do it with just an ata or 2 and some dialplan manipulation if they are only ever going to fax each other |
| 01:28.21 | WIMPy | Yeh, do you really want specific numbers, not just whatever some ATA has pre-assigned to its ports? |
| 01:29.13 | WIMPy | And if you really just want to connect two fax machines to each other, why don't you get some old soho plastic PBX from the skip to do it? |
| 01:29.52 | s7r | i want specific numbers yeah |
| 01:30.05 | nix8n82 | He could specify any numbers he wanted..dont know how well it would work though |
| 01:30.23 | WIMPy | That makes it a lot more complixcaed. |
| 01:30.47 | WIMPy | On some software, yes, but on an ata? |
| 01:31.23 | WIMPy | Ok, you can set some to PBX mode. That might allow you to assign any 2-digit number. |
| 01:31.30 | WIMPy | the rest would just be ignored then. |
| 01:32.18 | s7r | and just use an ata? |
| 01:32.22 | nix8n82 | Yeah he just uses the user id as the number he wants on each |
| 01:33.04 | nix8n82 | If you are not going to handle any other calls...yes just an ata would work |
| 01:34.29 | s7r | yup no other calls |
| 01:34.34 | s7r | just between these 2 machines |
| 01:34.47 | nix8n82 | Although it might be a lot of fun and educational to build an asterisk server... |
| 01:34.55 | s7r | so using the ata control panel i assign custom user id for each machine, and they talk to each other via that UID? |
| 01:34.58 | WIMPy | So you just have to find the right ata that will do what you want. |
| 01:35.24 | s7r | oh, good since i need yet to purchase it. what specifications am I looking for? how is this UID function called? |
| 01:36.07 | WIMPy | I fear you will have to read manuals. |
| 01:36.21 | WIMPy | Unless somone can recomment a model. |
| 01:40.16 | s7r | i think i found one , 32 eur value |
| 01:41.20 | s7r | http://www.manualslib.com/manual/616318/Planet-Technology-Ata-150.html |
| 01:47.11 | nix8n82 | I have an idea of how to do it with cisco ata but it's not going to be easy and nobody here is going to do it for you...for free anyway |
| 01:49.17 | s7r | didn't expect so, what's free in life. |
| 01:49.23 | s7r | but this Planet ata 150 can't do the job? |
| 01:49.49 | nix8n82 | Dont know never used one |
| 01:55.34 | nix8n82 | Wimpy's suggestion of getting some plastic PBX might be the easiest solution for you. Either way you will have to learn by trial and error |
| 01:56.01 | s7r | yeah. i am just totally new to it and reading many pages. |
| 01:56.03 | s7r | just one q |
| 01:56.24 | s7r | the file /etc/asterisk/sip.conf can contain UIDs starting with 0? like user 00440989445 whatever? |
| 01:57.25 | nix8n82 | Yeah |
| 01:57.33 | WIMPy | Yes, but that's not what you call. The callable extensions are found in extensions.conf. |
| 01:57.43 | WIMPy | If you want to dig in to it, try |
| 01:57.50 | WIMPy | ~primer |
| 01:57.50 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 01:57.55 | WIMPy | or the |
| 01:57.57 | WIMPy | ~book |
| 01:57.57 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:03.09 | nix8n82 | ~buybook |
| 02:03.10 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/0636920025894 so go buy it SERIOUSLY |
| 02:05.56 | nix8n82 | Anyone know if safari books online gives money to the authors of the book? |
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| 02:06.40 | nix8n82 | That's where I read it and it is a must read |
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| 02:09.38 | s7r | wow |
| 02:09.43 | s7r | everything is so complex |
| 02:10.35 | s7r | by default, asterisk treats the connections different if 2 phones are exchanging voice (calls) or fax (data)?/ |
| 02:13.23 | [1]Kiwi | s7r just be aware that using ATA's for alarm systems can be a bit hit and miss if your wanting to get it to talk to a monitoring station |
| 02:14.15 | [1]Kiwi | I normally get ip modules for the alarms to adapt them to talk to the station over the net |
| 02:14.31 | [1]Kiwi | much more reliable |
| 02:20.52 | [1]Kiwi | what do you do if you think you have found a bug in asterisk ? |
| 02:21.19 | WIMPy | Go to http://issues.asterisk.org/ |
| 02:21.30 | [1]Kiwi | thanks |
| 02:21.41 | WIMPy | Check if you can find it there and if not and you are reasonably confident it's a bug, open a ticket there. |
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| 03:42.27 | [1]Kiwi | can someone please tell me why this is a warning WARNING[1975][C-00000001]: chan_oss.c:853 oss_request: oss_request ty <console> data 0x0x7f84b8001bb0 <dsp> |
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| 08:43.21 | wasanzy | how do I do file test in asterisk? to check if a file exist or not |
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| 09:38.44 | stevenm | Anyone know a way of getting into SPA devices (full admin preferably) when they're locked down *without* factory resetting - or perhaps just get them to reveal all settings they have in them? |
| 09:39.32 | stevenm | I need CPC Duration set to 1 (for CPC Disconnect signal) but the stupid provisioning server (which we don't control) doesn't allow access to it |
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| 10:40.35 | wasanzy | this condition is not working as expected and I don't know why: same => n,GotoIf($["${filexist}" = "1" & ${filesize} > 1]?start:error) |
| 10:41.53 | wasanzy | more details: http://pastebin.com/MbjrtZ5v |
| 10:42.31 | wasanzy | so am checking to see if the file exist, if it exist, it should go to lable start, else, error. is there a problem with my logic? |
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| 11:12.33 | nextime | hi all, have anyone an idea on how to debug/fix a one-way audio issue on a B410p card ( 4x bri ), where the issue is on the BRI and NOT on the voip side? ( when i try to call the ISDN number from my mobile with just an Echo() on the dialplan, i have a one way audio issue, no voip side involved at all ) |
| 11:13.35 | nextime | i was using the very same card with misdn drivers in an old asterisk ( 1.4.2 ) some time ago, now i have an updated asterisk and dahdi, and i have this issue |
| 11:15.53 | WIMPy | How do you know you have one-way audio when using the echo application? |
| 11:16.39 | nextime | WIMPy : cause if i do a playback it work, if i do a Record it doesn't work, with echo it doesn't work |
| 11:16.53 | WIMPy | Anyway: If you don't fail to answer te call in the first place, there's no way to get one-way audio unless the driver is broken. |
| 11:17.15 | nextime | WIMPy : so, the drivers seems to be broken :| |
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| 11:17.31 | rhineheart_m | hello.. what is this for.. answeronpolarityswitch |
| 11:17.39 | WIMPy | Sounds strange. |
| 11:18.07 | nextime | WIMPy : heh, same to me, but at this point i have no other idea |
| 11:18.51 | WIMPy | You are sure the call is anwered? |
| 11:18.56 | nextime | WIMPy : yes |
| 11:19.25 | nextime | i can see it get an "Answer" in the CLI, and playback is working |
| 11:19.36 | WIMPy | Oh an btw (for all): Does someone by chance have access to a Cisco AS and could provide me with a file containing q931 debug form it? |
| 11:19.47 | rhineheart_m | hwllo.. what is this for answeronpolarityswitch? |
| 11:20.07 | rhineheart_m | that's in chan_dahdi.conf |
| 11:20.20 | WIMPy | rhineheart_m: It's something used on POTS interfaces. |
| 11:20.40 | rhineheart_m | yeah.. WIMPy.. that's why I asked.. |
| 11:22.05 | doop | some kinds of POTS lines will switch polarity when the called party answers |
| 11:22.15 | doop | that's what that's for |
| 11:24.52 | rhineheart_m | doop, thanks for the idea.. |
| 11:25.05 | doop | what idea |
| 11:26.42 | rhineheart_m | m just in a situation where my calls to mobile phones are dropped when I comment it.. but when I uncomment it.. the incoming lines are not terminated when I ended the calls (testing) |
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| 11:27.17 | rhineheart_m | incoming calls though ended already..asterisk won't terminate it. |
| 11:27.25 | doop | time to get out the buttset and see what your line is actually doing |
| 11:29.46 | rhineheart_m | doop, meaning? |
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| 11:31.19 | doop | is your line reversing polarity on answer? are you getting CPC on disconnect? these are the things the right piece of test equipment will tell you |
| 11:33.25 | rhineheart_m | where to get that info doop? |
| 11:33.54 | doop | rhineheart_m plug a buttset into your telephone line and use it |
| 11:35.37 | rhineheart_m | doop, sorry but I don't have like that. Is there another way to test it? |
| 11:36.42 | rhineheart_m | when I uncomment "answeronpolarityswitch=no busycount=6 busydetect=yes " no problem with incoming calls... but outgoing calls to mobile devices are dropped |
| 11:37.12 | doop | rhineheart_m without knowing what your line is actually doing, flipping switches in the dark isnt going to help much |
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| 12:02.10 | madax | hi guys, i can't find the command PlayDTMF in asterisk 11 through the AMI, do i need to add something to have it ? |
| 12:05.48 | WIMPy | Well, AMI doesn't handle calls, so that makes sense. |
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| 13:43.28 | zaf | so.. my yum updated to asterisk 12.7 from 12.5 last night, and SIP phones became unable to register. rolled back the yum updates this morning and they started working again |
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| 13:48.38 | marlinc | Is it possible to pronounce the name configured in the voice mail on its own? |
| 13:49.16 | marlinc | In some cases I don't want a voicemail but just pronounce the name and then let Asterisk say something |
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| 13:52.10 | JerJer | marlinc: i haven't done it in a while, but one could just play the recorded name, then tell voicemail to skip the prompt |
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| 13:56.07 | [TK]D-Fender | [08:49]marlincIn some cases I don't want a voicemail but just pronounce the name and then let Asterisk say something <- then don't use voicemail. |
| 13:57.52 | marlinc | Well we do use the Voicemail but in some contexts I just want to pronounce the name |
| 13:58.08 | marlinc | And the Voicemail has a handy feature that allows someone to say their name |
| 13:58.47 | [TK]D-Fender | then use that recording in a placyback |
| 13:58.51 | [TK]D-Fender | And don't call voicemail |
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| 14:00.25 | marlinc | Well I was wondering if there was a build in function that could do that |
| 14:00.30 | marlinc | I guess thats a no :) |
| 14:01.03 | [TK]D-Fender | "core show application playback" <-- there is |
| 14:01.12 | [TK]D-Fender | that is an application to "just play a sound file" |
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| 14:12.34 | [TK]D-Fender | Playback is as built-in as any other standard dialplan app. |
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| 14:31.05 | Katty | goooooooooooooooooood morning!!!! |
| 14:31.39 | Chainsaw | Good afternoon Katty :) |
| 14:38.18 | Katty | hugs Chainsaw to bits! |
| 14:38.21 | Katty | how're you dear? |
| 14:39.02 | Chainsaw | Doing good :) |
| 14:39.13 | Chainsaw | 11.13.1 has been running for a week. |
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| 15:05.16 | net_ | can I use https post in curl function? |
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| 15:17.42 | n3tctrl | following this wiki p0age on asterisk.org https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages there appears to be some misinformation |
| 15:18.44 | n3tctrl | under 2.2 Adding the repository, the second method is incorrect, as the rpm file does not exist at the stated url. can anyone tell me where it might be located now, or what the name might have changed to? |
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| 16:22.29 | mdhas | Does anyone know why I keep getting NOTICE[22944] chan_sip.c -- Registration for 'telephonenumber@providerdomain' timed out, trying again |
| 16:22.33 | mdhas | repeatedly? |
| 16:22.45 | Qwell | My guess is because it timed out. |
| 16:24.52 | mdhas | Qwell: Yeah...but the question is if anyone has any idea why... |
| 16:25.51 | Qwell | The other side isn't responding, or the response isn't getting to you. If I had to guess (and I do, since you provided no information), I'd say there's a firewall getting in the way. |
| 16:28.25 | net_ | https post can be used in curl function? |
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| 16:30.08 | mdhas | would the log report incorrect username/password if the password was wrong? well I guess since it can't reach the remote server that is moot |
| 16:31.54 | mdhas | is it the sip device that is having problems or the server? |
| 16:32.29 | Qwell | mdhas: It is impossible to tell with the information that we have. |
| 16:34.02 | mdhas | Qwell: okay what information would help or can you think of somewhere where I can go to get more information? |
| 16:34.12 | mdhas | as to the cause of the issue? |
| 16:35.28 | *** join/#asterisk kgunnIT (~kgunnIT@198-243-23-225.dia.static.qwest.net) |
| 16:35.29 | Qwell | A SIP debug would be a start. pcap trace of the server side would also be useful. Also making sure that your firewall is properly configured. |
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| 18:09.09 | mdhas | Does anyone have any ideas why an inbound call says that it failed to authenticate device? |
| 18:13.03 | kgunnIT | @mdhas - if it is over IAX or SIP trunk, have you verified username/password with your provider? |
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| 18:59.49 | mdhas | Is there a way that you can see the remote ip of the sip device that's trying to connect to the server? |
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| 19:01.21 | [TK]D-Fender | "sip set debug on" <- yes |
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| 19:20.52 | marlinc | Is it possible two have multiple SIP devices register on one SIP entry in the config? |
| 19:22.47 | [TK]D-Fender | in sip.conf they will fight for registration. * will only take the last one when sending it calls |
| 19:22.58 | [TK]D-Fender | PJSIP allows for multiple registrations |
| 19:23.16 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 19:25.28 | ChannelZ-Wk | Anyone pay much attention to their sip channelstats ? Seeing what seem like a lot of lost packets received.. |
| 19:27.52 | ChannelZ-Wk | 1% average at the moment but it's still one or two every couple seconds. I think it's something with intertubes as a whole today (or at least my ISP) |
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| 19:42.20 | pa | Hi guys, i have a little problem with a voip phone.. Essentially it works when i call from it, but when i call to it, i can't hear anything. Even within the LAN. I'm not sure if it's the phone, asterisk conf, or the firewall. So this phone is in a separate subnet, 192.168.3.220, and here's what happens when i call from and to it: http://codepad.org/C0U9DspA (the first is from it and the second is to it) |
| 19:42.30 | pa | there's something strange with that Probation line |
| 19:42.43 | pa | first time the ip changes to the right phone, second time it doesnt |
| 19:43.43 | pa | extensions are here: http://codepad.org/x5s75LDS |
| 19:43.46 | doop | pm: what happens when you try it with the phone on the same subnet |
| 19:44.07 | pa | this is a good question.. and i can't answer because i can't have it on the same subnet |
| 19:44.18 | doop | "can't"? |
| 19:44.24 | pa | well this phone is wired |
| 19:44.29 | pa | and that's a wireless subnet |
| 19:44.42 | doop | wireless and wired can be a flat network amigo |
| 19:44.44 | pa | but i can try to plug it to a different network |
| 19:45.08 | lorenzo | I have a odd question about SIP protocol, I have a cisco CME registering to asterisk via SIP. So I call phones "behind" the CME with SIP/user/extension |
| 19:45.26 | lorenzo | what is that called? I mean, how does it work? |
| 19:45.42 | [TK]D-Fender | pa: that isn't SIP debug and as such is useless |
| 19:45.55 | [TK]D-Fender | pa:Go dget a properly debugged call. |
| 19:46.01 | pa | [TK]D-Fender, how do i turn sip debug on? |
| 19:46.12 | [TK]D-Fender | "sip set debug on" |
| 19:46.48 | lorenzo | like, I can call a user that is directly registerd on my server with SIP/user .. but the CME user is acting as proxy for the others |
| 19:46.59 | lorenzo | so a user can be a sip proxy too? |
| 19:47.57 | lorenzo | ChannelZ-Wk: my Lost % is < 0.05% |
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| 19:49.41 | lorenzo | ChannelZ-Wk: my Lost % is < 0.05% |
| 19:49.43 | lorenzo | whops |
| 19:51.12 | [TK]D-Fender | lorenzo: If there is a proxy to reach them then specify that. how you call them via the CME depends |
| 19:51.48 | lorenzo | [TK]D-Fender: no I mean, the CME is registering with a single user agent, the phones "behind" the CME (the dial-peer, destination-pattern stuff) |
| 19:51.58 | lorenzo | is called via Asterisk in SIP/cmeuser/123 for example |
| 19:52.04 | lorenzo | I'm curious... how is that behaviour called? |
| 19:52.15 | [TK]D-Fender | lorenzo: if it's an ACTUAL SIP proxy then you'd set the outboundproxy on your peer for that particular device. If you call it THROUGH the CME as a B2BUA then you'd just call CME targeting the # it knows to use to reach them |
| 19:52.27 | [TK]D-Fender | There is no name that behaviour |
| 19:53.15 | lorenzo | ah, I'm reading on that |
| 19:53.24 | lorenzo | looks like CME is acting as B2BUA :) |
| 19:53.52 | [TK]D-Fender | So i is not directly routing the call packets to the actual phone, but rather acting as an intermediary. |
| 19:55.13 | ChannelZ-Wk | lorenzo: Thanks. I think Comcrap's pipes are frozen today. I was trying to watch some live streams and it was buffering like mad, not even that much bandwidth. |
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| 19:59.42 | pa | [TK]D-Fender, i managed to log the call that doesnt work, but i'm not able to decode it :) http://codepad.org/xFUgquk2 |
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| 20:13.30 | pa | but what does this line mean? > 0x7f42dc02f680 -- Probation passed - setting RTP source address to 192.168.3.220:49658 |
| 20:13.50 | pa | and if it does not appear, what could be the networking cause? some forwarding not allowed? |
| 20:13.57 | pa | it goes one way, but not the other.. |
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| 20:50.00 | pa | i start to suspect it's something with the phone |
| 20:50.15 | pa | i now plugged it to another network where softphones normally work |
| 20:50.22 | pa | and it still does not receive |
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| 20:59.31 | pa | ok, i checked with tcpdump now, and during the call that works, asterisk and the phone exchange data, during the call that does not work, there are a lot of packets going from asterisk to the phone, but nothing back |
| 21:03.30 | pa | also, during the call i see things like: IP 192.168.3.1 > 192.168.3.220: ICMP host 192.168.2.30 unreachable, length 103 . I tried to set qualify=no, but it didnt change anything |
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| 21:05.50 | pa | no, this definitely is not the reason |
| 21:05.57 | pa | i still get it on calls that work |
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| 21:27.54 | pa | doop, ok, so now i tried, on the same net, 2 different phones, the one that does not work, and an xlite softphone. xlite works.. |
| 21:28.27 | doop | pa: both connecting with the same user credentials? |
| 21:28.34 | pa | yes |
| 21:28.55 | doop | what kind of phone is the one that's giving you trouble |
| 21:30.43 | pa | doop: it's this one: http://it-pic2.ciao.com/it/4297825.jpg |
| 21:30.54 | pa | it's a little hard to give more info, because it's out of production for a while |
| 21:31.36 | doop | ok - is there no way you can perhaps read a make and model number off a label or something |
| 21:32.05 | doop | i've never seen or used one of those before |
| 21:32.32 | pa | let me start it again in conf mode, maybe in info i can find something |
| 21:33.09 | doop | no i mean like on the actual phone itself |
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| 21:37.44 | pa | on the phone they labeled it as "Aladino VoIP" that is the brand the local TelCo used to sell it |
| 21:37.48 | pa | (also below) |
| 21:37.52 | pa | i found this: |
| 21:38.00 | pa | http://codepad.org/fXtvpokg |
| 21:39.09 | doop | yeeeeeeaaaaaah maybe that phone is a lost cause |
| 21:39.28 | doop | someone here might be brave enough to help you get it working |
| 21:39.46 | pa | i found some managing to use it with commercial voip services |
| 21:39.53 | pa | (most likely running asterisk themselves) |
| 21:40.02 | doop | shrug |
| 21:40.25 | pa | i suspect maybe it's somewhat problematic on the networking, but i don't know how to make asterisk be super-understanding IP-wise |
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| 22:12.39 | pa | doop, apparently canreinvite=yes makes it work.. do you know why this is so? |
| 22:12.44 | pa | or have any idea? |
| 22:13.06 | doop | pa too many variables for me to make a guess |
| 22:13.23 | pa | okay, well thanks anyway |
| 22:13.32 | doop | sorry |
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| 22:18.07 | pa | hm.. canreinvite=yes seemingly allows the two clients to talk directly, which ideally might not be the way to go.. |
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| 22:43.02 | saint_ | hi all, can someone tell me how to fix this issue when I start asterisk: asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 22:43.20 | saint_ | i ran configure with --disable-asteriskssl option , but still get the error |
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