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| 02:13.28 | esaym153 | I have a strange issue with asterisk set up. I have a home phone plugged into an ATA, going to asterisk over SIP, that is fed to PBX via IAX link to voip.ms. If I call my cell phone from VOIP, I can talk in either phone and hear fine. However while I am talking in my cell phone, I cannot hear my voip phone. So outgoing audio from asterisk, is like one way/halfduplex. |
| 02:14.33 | esaym153 | it is had been like that for years, people would always complain that I am breaking up, but when I would call a test phone it was always fine because I never tried to play something through voip and talk into the receiving phone... |
| 02:14.38 | esaym153 | asterisk 1.8 |
| 02:15.29 | WIMPy | That's usually a phone (mis)feature. Are you using speakerphone? |
| 02:15.29 | esaym153 | I am using monitor in my call plan, even the audio files show that outgoing audio cuts out when a bunch of incoming audio is coming in |
| 02:16.13 | esaym153 | WIMPy: so are you saying it could be an echo canceler on the hardware phone? (because I was thinking that) |
| 02:16.26 | WIMPy | yes |
| 02:16.27 | esaym153 | WIMPy: but now, not on speaker phone |
| 02:17.00 | esaym153 | no* |
| 02:17.32 | esaym153 | I guess i could try turning on the software echo canceler on the ata |
| 02:17.39 | esaym153 | but sometimes, that makes bad echo |
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| 02:24.58 | esaym153 | so the same issue happens on both my cordless phone, different brand too. I'll try turning on echo canceler |
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| 02:30.00 | WIMPy | On? Sounds more like some EC should be DISabled. |
| 02:32.01 | esaym153 | WIMPy: agreed, just checking though :) |
| 02:32.31 | esaym153 | WIMPy: turning it on didn't help :(, but I got another idea for a test. |
| 02:34.22 | esaym153 | While I had one cordless connected to my cell phone and playing music (laid it on top of a radio), I muted the cell phone, put up the other cordless phone and hit Talk so I could listen in on the line, I could hear the music fine, until I started talking, then it would cut out. So even cordless to cordless (by passing all VOIP) had the same issue... :( |
| 02:34.32 | esaym153 | is this just the norm for cordless? |
| 02:34.36 | esaym153 | kinda suxx |
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| 02:36.28 | WIMPy | no |
| 02:36.55 | WIMPy | Maybe for analog cordless. I don't know. |
| 02:37.36 | WIMPy | Analog is evil! |
| 02:37.52 | WIMPy | Sun is rising. Time to go to bed. |
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| 03:07.19 | esaym153 | well it is not 'pure' analog. DECT 5.0 or something |
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| 03:09.32 | [Outcast] | anyone know if pjsip has a channel on here? |
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| 03:13.10 | aandrew | whirred |
| 03:13.56 | aandrew | I scored a pile of phones today. something like 20 SPA942s for $100 |
| 03:14.46 | aandrew | I was only looking for a couple for now, my old polycom ip650 is starting to act up (probably internal supply issue) |
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| 03:59.08 | carrar | 942's are decent cheap phones |
| 03:59.32 | carrar | especially at $5 each |
| 04:11.50 | drmessano | Yep |
| 04:12.07 | drmessano | That was a great deal.. Still good phones |
| 04:12.33 | drmessano | The 942's especially, as they got some later firmware upgrades that the 921/941 did not |
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| 12:21.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:20.52 | esaym153 | are there settings anywhere in sip.conf or iax.conf that would enable some kind of echo suppression? Because I am getting half-duplex audio on my ATA and softphone clients |
| 15:21.25 | esaym153 | I have echo cancellation turned off on all my softphones and ATAs |
| 15:21.26 | robmal | nat=yes |
| 15:21.40 | [TK]D-Fender | no. |
| 15:22.08 | esaym153 | robmal: this isn't one way audio, this is if someone is talking on one phone, they can't hear me until they stop talking |
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| 15:22.12 | [TK]D-Fender | You cannot doe EC over VoIP. It has to be done at the conversion point |
| 15:23.18 | esaym153 | [TK]D-Fender: so asterisk is not interfering here? |
| 15:23.50 | WIMPy | You can do EC everywhere, but if VOIP is onvolved it will likely fail. |
| 15:23.55 | esaym153 | how is it I still get half duplex audio even though all my software had echo suppression set to off ? |
| 15:24.02 | [TK]D-Fender | There is no such thing as EC over VoIP period |
| 15:24.39 | esaym153 | [TK]D-Fender: I think it is called echo suppression, which mutes your line if there is incoming audio detected |
| 15:24.45 | esaym153 | which sucks |
| 15:25.10 | file | Acoustic Echo Cancellation. |
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| 15:53.21 | esaym153 | so if this issue is from my ATA, what is a good ATA to get? |
| 15:59.50 | esaym153 | ok, so yea. with more testing, the issue is not found with my sip softphone. It is only an issue with my cordless phone plugged into a grandstream ATA |
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| 16:23.53 | gusto | so |
| 16:23.55 | gusto | WIMPy, hi |
| 16:23.59 | gusto | Penguin, hi |
| 16:24.40 | esaym153 | hmm, looks like OBi200 is best ATA? |
| 16:24.58 | gusto | ATA's are shitty all of them |
| 16:25.11 | gusto | I gave up on them |
| 16:26.17 | gusto | this device you mentioned I just looked it up |
| 16:26.24 | gusto | I does not seem to be better than any ATA |
| 16:26.53 | gusto | and with only one port, one can be curious at what cost it is, because for <40EUR you get the cisco spa112 |
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| 16:39.12 | esaym153 | gusto: so no recommendations? |
| 16:39.26 | gusto | for an ATA? SPA112 |
| 16:39.57 | esaym153 | gusto: I would like a cordless phone for the house, but all wifi based sip phones are expensive and low battery life (and reviews show them to be crap too) |
| 16:40.15 | gusto | OK |
| 16:40.28 | gusto | then you need a VoIP DECT Base from Gigaset |
| 16:46.04 | cmendes0101 | I like the panasonic KX-TGPxx. You can get extra headsets to go around the house wireless. I picked some used off ebay but in the 100-200 range w/ extra headsets |
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| 17:06.12 | esaym153 | hmm, expensive |
| 17:06.24 | esaym153 | gusto: so why do you say all ATAs are crap? |
| 17:06.48 | gusto | of course they are |
| 17:06.54 | gusto | but sometimes you need them |
| 17:07.04 | gusto | I have also a lot of SPA112 and |
| 17:07.07 | gusto | PAP2T's |
| 17:07.07 | esaym153 | gusto: so what is wrong with SPA112 ? |
| 17:08.25 | gusto | unstable |
| 17:09.21 | gusto | one would need some kind of alternate firmware, like openwrt is for routers one would need openpap or openspa or something like that (openata) LOL! |
| 17:13.08 | Navion | Has anyone seen the "error, application is not present" error on a Polycom IP335? How do I fix it? Can't connect to the web interface on the phone. it just reboots over and over. |
| 17:14.13 | [TK]D-Fender | reprovision it |
| 17:16.10 | Navion | [TK]D-Fender: Where's a good reference for that? I've tried a "factory reset" but that didn't help. |
| 17:16.31 | [TK]D-Fender | Factory reset resets settings |
| 17:16.36 | [TK]D-Fender | it doesn't install firmware |
| 17:16.39 | [TK]D-Fender | REPROVISION IT |
| 17:16.55 | [TK]D-Fender | download the SIP firmware and setup up a place for it to pick it up from |
| 17:20.11 | gusto | LOL |
| 17:31.13 | Navion | [TK]D-Fender: What should that file look like? Like the file suffix? or filename? Is there a Polycom provisioning server I can just point the phone to? |
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| 17:45.07 | [TK]D-Fender | Navion, You set that up |
| 17:45.13 | [TK]D-Fender | grab firmware extract |
| 17:45.19 | [TK]D-Fender | point phone to it. |
| 17:45.19 | [TK]D-Fender | done |
| 17:46.51 | Navion | [TK]D-Fender: I'm trying to locate it online. I have a tftpd running on the switch. I assume I just have to put the right file in the /tftpboot directory and the phone will find it. Does it look for a file with the mac address prefixing it? |
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| 17:50.39 | [TK]D-Fender | Navion, in it's current shape it's likely to be different than what's in there and will basically be a factory reset |
| 18:00.14 | Navion | [TK]D-Fender" Sp, I have Polycom_UC_Software_4_0_8_Rev_C_release_sig_combined.zip Do I put that whole thing in the tftpboot directory or just a piece of it or... |
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| 18:04.38 | [TK]D-Fender | all of it |
| 18:05.19 | Navion | [TK]D-Fender: as a zip file or as the unzipped directory structure? |
| 18:05.42 | [TK]D-Fender | <[TK]D-Fender> grab firmware extract |
| 18:09.21 | Navion | [TK]D-Fender: Is that the sip.ld file? |
| 18:11.15 | [TK]D-Fender | Ok, lets try to make this clear |
| 18:11.16 | [TK]D-Fender | EXTRACT THE WHOLE DAMN THING |
| 18:11.20 | [TK]D-Fender | ALL OF IT |
| 18:11.48 | [TK]D-Fender | You put a firmware file in there with mismatched or otherwise broken configs and you'll end up in another dead-end boot-loop |
| 18:11.50 | Navion | Done that |
| 18:12.07 | [TK]D-Fender | get it back to NORMAL to start |
| 18:13.02 | Navion | It seems to be in a "NORMAL" start, rebooting. |
| 18:22.49 | Navion | [TK]D-Fender: It seems to be much better now. |
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| 18:31.19 | [TK]D-Fender | heads out for a bit |
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| 20:16.02 | jmordica | Why does the adaptive jitter buffer in 13.4 adjust up much faster than it adjusts back down? |
| 20:16.54 | jmordica | adjusting up takes a second or so but adjusting back down after the connection isn't contested anymore takes up to 1m |
| 20:18.19 | jmordica | i ask because during the time it is adjusting back down, the audio is garbled somewhat so that is what I'm trying to avoid |
| 20:18.57 | jmordica | jbmaxsize=3000 and resync is the same value. |
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| 20:56.51 | dtrainor | Hi. I'm having difficulty getting CDR data via AGI. I read that some variables are not available in AGI. Are CDR variables some of those variables not available in AGI? |
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| 21:12.17 | [TK]D-Fender | not that I've seen |
| 21:12.24 | [TK]D-Fender | Where di yo see this quoted? |
| 21:12.28 | [TK]D-Fender | And show us what you're doing |
| 21:13.02 | ChannelZ | AGI can call any diaplan application/function so if it's something accessable via the dialplan it's accessable via AGI |
| 21:14.39 | mjordan | for that matter, anything that is available via a dialplan function is available through any of Asterisk's APIs. Dialplan applications are only available through AGI. |
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| 21:24.22 | dtrainor | Sorry, had to leave for a bit. Let me see if I can find the docu that describes variables via AGI |
| 21:28.39 | dtrainor | I can't find it. I swear I read it from a pretty good source, though. |
| 21:28.48 | dtrainor | Might have been old docu, there's a lot of that floating around... |
| 21:30.25 | dtrainor | I may have accidentally fibbed. Looks like I am getting CDR data, though it's being returned as an Array according to PHP https://gist.github.com/dantrainor/9793552334711cc0752c |
| 21:30.45 | dtrainor | Just toying around with phpagi today, nothing fancy. |
| 21:33.46 | WIMPy | Well, that's be phpagi, not AGI. |
| 21:34.01 | WIMPy | s/s/d/ |
| 21:35.13 | dtrainor | Okay. |
| 21:35.19 | dtrainor | Yes. You are right. |
| 21:35.34 | dtrainor | Completely different I suppose |
| 21:35.53 | WIMPy | Somehow |
| 21:41.29 | dtrainor | is phpagi still the way to go? |
| 21:47.34 | WIMPy | If you like php... |
| 21:48.04 | dtrainor | haha |
| 21:48.28 | dtrainor | Sorry, should have been more specific. Unfortunately I'm not a great developer. Of the little experience I do have, it's in PHP. |
| 21:49.36 | WIMPy | Then you probably should use php. Although that still doesn't answer whether you should use phpagi. |
| 21:53.05 | dtrainor | I'm a poor developer. Of the little development work I've done in the past, it's mostly been in PHP. Is phpagi still the way to go if someone wanted to develop some AGI applications using a PHP abstraction library for AGI? |
| 21:59.10 | ChannelZ | I just do it myself. |
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| 22:07.09 | mjordan | dtrainor: it's probably the one I've heard the most about personally with PHP. Really, AGI is a pretty simple protocol, so any wrapper library is probably going to be relatively okay |
| 22:14.06 | dtrainor | It is pretty simple, you're right. It's a learning process, and I'd like to get experience with some more structured development. Figured I'd make it as difficult as I could for myself to learn something. |
| 22:37.46 | dtrainor | Ah, from the phpagi docs: Does not work with global variables. Does not work with some variables that are generated by modules. |
| 22:38.02 | dtrainor | Which I've found to not be entirely true... you can create channel variables that contain data that come from global variables |
| 22:38.56 | WIMPy | Yes, sometimes libraries make things more complicated instead of what they claim. |
| 22:41.27 | dtrainor | What DO you like? How do you use and interface with AGI? I get the feeling that I'm "doing it wrong". |
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| 22:43.06 | WIMPy | I don't use any 3rd party stuff. |
| 22:45.31 | mjordan | I wouldn't say you're doing anything wrong. If I were going to write an application that built on top of Asterisk, the first thing I would do would be to either look for a library that wraps up a protocol, or write one myself. |
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