| 00:00.36 | Bonte | http://pastebin.com/bELdkUkH - Yeah, I see my flaw now |
| 00:00.48 | Bonte | I think I can take it from here. Thanks [TK]D-Fender! |
| 00:02.02 | [TK]D-Fender | dialplan SNAFU |
| 00:02.30 | Bonte | This won't be my first and certainly not last :D |
| 00:14.20 | mattwj2002 | g729 |
| 00:14.35 | mattwj2002 | what is the deal with the licensing? |
| 00:16.59 | *** join/#asterisk u0m3 (~u0m3@92.80.86.213) |
| 00:17.50 | mattwj2002 | it is so freaking expensive |
| 00:17.51 | mattwj2002 | :) |
| 00:18.54 | [TK]D-Fender | GUess it depends what your idea of "expensive" is. |
| 00:19.11 | mattwj2002 | $10 per channel |
| 00:19.12 | mattwj2002 | :) |
| 00:19.43 | mattwj2002 | for a company I am sure it is worth it |
| 00:19.55 | [TK]D-Fender | I consider > $4 for a coffee at Starbucks "expensive". For less that 3 coffee's I can support another channel.... |
| 00:20.11 | [TK]D-Fender | And how many does an individual need? |
| 00:20.36 | mattwj2002 | not many |
| 00:20.36 | *** join/#asterisk navaismo (~navaismo@201.124.146.234) |
| 00:20.38 | mattwj2002 | :) |
| 00:21.28 | [TK]D-Fender | So you're done wining now... right? :) |
| 00:21.33 | [TK]D-Fender | whining* |
| 00:21.37 | mattwj2002 | yes sir |
| 00:21.38 | mattwj2002 | :) |
| 00:22.08 | mattwj2002 | [TK]D-Fender: last question about g729 and I'll drop the subject |
| 00:22.13 | mattwj2002 | how is the call quality? |
| 00:22.31 | [TK]D-Fender | Good enough |
| 00:22.48 | [TK]D-Fender | Do you NEED the bandwidth? |
| 00:22.48 | mattwj2002 | cool |
| 00:23.59 | mattwj2002 | the whole reason I brought up the subject is I have been playing around with zoiper for android |
| 00:24.27 | [TK]D-Fender | it's not like you need a license to find out... |
| 00:26.48 | mattwj2002 | good point |
| 00:31.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 01:10.39 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 01:21.59 | *** join/#asterisk serafie (~erin@24.96.64.240) |
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| 01:23.25 | *** join/#asterisk TBotNik (~quassel@webmail.tabanigroup.com) |
| 01:31.34 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 01:33.24 | TBotNik | All, Have 2ea. PolyCom IP4000 conf phone and tried connecting them to both Fonality PBXtra and Elastix and always get error: "application not present"! Any suggestions. Have copied most of the files in /tfptboot per docs on this, but find I'm on ver 3.2.3 on most phones and most sure what version supports IP4000. Some things I read say only 3.1.7/8 are only support version for IP4000. |
| 01:34.56 | TBotNik | can I get some help here? |
| 01:37.40 | *** join/#asterisk Entelin (~Paruza@c-71-63-229-101.hsd1.mn.comcast.net) |
| 01:38.42 | Entelin | I am new to voip and asterisk, can someone recommend good starting documentation that explains the general logical flow of a call and how it ties in with the config files? |
| 01:42.11 | WIMPy | ~book |
| 01:42.11 | 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 |
| 01:43.12 | TBotNik | Repeating: All, Have 2ea. PolyCom IP4000 conf phone and tried connecting them to both Fonality PBXtra and Elastix and always get error: "application not present"! Any suggestions. Have copied most of the files in /tfptboot per docs on this, but find I'm on ver 3.2.3 on most phones and most sure what version supports IP4000. Some things I read say only 3.1.7/8 are only support version for IP4000. |
| 01:43.13 | TBotNik | can I get some help here? |
| 01:43.21 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 01:43.45 | Entelin | WIMPy, thanks ill have a look |
| 01:53.45 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 02:00.10 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 02:08.38 | TBotNik | Repeating: All, Have 2ea. PolyCom IP4000 conf phone and tried connecting them to both Fonality PBXtra and Elastix and always get error: "application not present"! Any suggestions. Have copied most of the files in /tfptboot per docs on this, but find I'm on ver 3.2.3 on most phones and most sure what version supports IP4000. Some things I read say only 3.1.7/8 are only support version for IP4000. |
| 02:12.32 | *** join/#asterisk D30 (~deo@222.127.13.226) |
| 02:14.39 | k-man | do you guys like freepbx? |
| 02:15.14 | Entelin | no everyone here hates it |
| 02:15.19 | Entelin | j/k |
| 02:15.45 | WIMPy | Sut sure if that helps in any way, but |
| 02:15.48 | WIMPy | ~freepbx |
| 02:15.48 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 02:15.57 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 02:16.02 | mattwj2002 | hi all |
| 02:16.02 | mattwj2002 | :D |
| 02:16.15 | k-man | WIMPy, i know - i was just wondering |
| 02:16.15 | mattwj2002 | googletts is awesome |
| 02:17.10 | k-man | i just tried installing freepbx and was surprised to discover that the systemadmin module is apparently commercial now, which makes it virtually unuseable without it imho |
| 02:17.36 | mattwj2002 | systemadminn module what does it do? |
| 02:18.01 | k-man | mattwj2002, dunno - i can't tell you as I don't have it but i presume its used for configuring? |
| 02:18.29 | mattwj2002 | that sucks |
| 02:18.52 | k-man | well, i could be wrong but i'm finding it hard to find concrete information on it |
| 02:22.26 | *** join/#asterisk D30 (~deo@222.127.13.226) |
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| 02:36.36 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 02:48.43 | TBotNik | mattwj2002, k-man: Are you referring to the "EndPoint Manager" that FreePBX uses to manage/install phone devices. That is a sweet tool and sets FreePBX ahead of the other Asterisk forks as it solves all the device install issues, such as the one I'm having with my PolyCom IP4000 conf phones. |
| 02:49.21 | WIMPy | Freepbx is not an Asterisk fork. |
| 02:49.38 | Penguin | Asterisk forks? |
| 02:49.44 | Penguin | FreePBX is a configuration tool. |
| 02:49.58 | TBotNik | There is a free "EndPoint Manager" but does not support "ALL" devices. The paid version supports them all. |
| 02:50.32 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 02:51.07 | ChannelZ | FreePBX is a steaming pile of bling that sits on top of asterisk so you can't see it. |
| 02:51.38 | TBotNik | ChannelZ: Can tell you are an Asterisk purist! |
| 02:51.56 | WIMPy | The last days were ones where I whished I didn't have to seee Asterisk, either. |
| 02:52.29 | TBotNik | All: Does anyone know/understand my PolyCom IP4000 issue, or should I repost? |
| 02:52.39 | Penguin | I would venture to say that anything FreePBX can do, someone else could do it without FreePBX. |
| 02:52.54 | ChannelZ | It's fine if that's what you want, it just gets boring watching people come in here wondering what exploded on their system |
| 02:53.35 | ChannelZ | And as demonstrated, some people seem to think FreePBX IS Asterisk. |
| 02:53.45 | TBotNik | All: I post on forum at: http://forums.digium.com/viewtopic.php?f=1&t=89494&p=197108#p197108 |
| 02:54.03 | ChannelZ | and to be fair AsteriskNOW doesn't help that much |
| 02:54.42 | TBotNik | ChannelZ: The Fonality PBXtra is like AsteriskNOW, and a pile of crap! |
| 02:55.41 | Penguin | I use AsteriskNOW to ease installation of underlying OS and asterisk as a unit. Pop in the CD, start the computer, select "Asterisk, No GUI," and wait about 15 minutes. Then I take out the CD and reboot into CentOS with asterisk ready for me to configure without FreePBX. |
| 02:56.17 | WIMPy | 15 Minutes? |
| 02:56.24 | Penguin | Maybe 20 on a slow computer. |
| 02:56.26 | TBotNik | ChannelZ: I'm on customer site now with PBXtra and PolyCom IP 4000 will not recognize in the system. Hand adding/editing .cfg files in /tfptboot has not fixed the issue, so stuck until I get some insight! |
| 02:56.32 | WIMPy | Sounds like it might be faster to install from source. |
| 02:56.43 | Penguin | That involves work. |
| 02:57.03 | Penguin | And I can't imagine trying to install CentOS from sources. |
| 02:57.14 | Penguin | That would take far longer than 15 minutes. |
| 02:57.17 | Kobaz | hah |
| 02:57.19 | Kobaz | yeah |
| 02:57.29 | Kobaz | might as well get gentoo |
| 02:58.09 | TBotNik | Kobaz: Yeah GenToo! Not! |
| 02:59.08 | TBotNik | Kobaz: GenToo is real Linux for people who understand all the internal moving parts of Linux. If you don't stay away, you will be quickly confused. |
| 02:59.40 | TBotNik | Penguin: That's why there are LiveCDs! |
| 02:59.54 | Penguin | What is why? |
| 03:00.22 | *** join/#asterisk D30 (~deo@58.71.19.178) |
| 03:00.40 | TBotNik | All: OK repeating my issue for all to understand: Have 2ea. PolyCom IP4000 conf phone and tried connecting them to both Fonality PBXtra and Elastix and always get error: "application not present"! Any suggestions. Have copied most of the files in /tfptboot per docs on this, but find I'm on ver 3.2.3 on most phones and most sure what version supports IP4000. Some things I read say only 3.1.7/8 are only support version for IP4000. |
| 03:01.21 | TBotNik | All: Also I posted this on forum at: http://forums.digium.com/viewtopic.php?f=1&t=89494&p=197108#p197108 |
| 03:01.41 | Penguin | I'm talking about AsteriskNOW being a nice platform to build a PBX. It provides CentOS and asterisk. I put in the CD, push a few buttons to get it started, wait for it to install CentOS and asterisk, then reboot into a new system which is ready for me to configure. It doesn't get much easier than that. |
| 03:02.47 | TBotNik | All: I think I'm down to just a driver version problem as almost all write ups I see say support stopped on IP4000 but not sure about the right/last supported version. |
| 03:04.33 | Kobaz | TBotNik: kinda |
| 03:04.51 | Kobaz | TBotNik: gentoo is for people who want to spend all their time optimizing compile options and not actually getting any real work done |
| 03:05.24 | WIMPy | Sounds like the ideal OS for Asterisk Users. |
| 03:06.02 | WIMPy | Spend your days configuring your phone system instead of using it. |
| 03:06.37 | TBotNik | Penguin: A "LiveCD" or .iso (for flash boot) has the OS and the apps needed for a clean install, such as AsteriskNOW, or Elastix, FreePBX, etc. Most Asterisk forks have CentOS liveCDs, but there is version of Asterisk and Elastix for Ubuntu and ViciDial runs on OpenSuse. |
| 03:06.50 | Kobaz | i spend my days fixing bugs mainly |
| 03:06.59 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 03:07.08 | Kobaz | and then when i run out of bugs to fix, i add new ones |
| 03:07.09 | Penguin | I didn't know asterisk forked. |
| 03:07.25 | Kobaz | when did asterisk fork? |
| 03:07.31 | Penguin | Last never? |
| 03:07.51 | TBotNik | Kobaz: I run Ubuntu Linux, very stable, few bugs on Laptop and 4 home servers. Also have CentOS and OpenSuse boxes. |
| 03:07.53 | Penguin | Some of the devs forked off some time ago. |
| 03:08.04 | Kobaz | into what project? |
| 03:08.05 | Penguin | They didn't take any parts of asterisk with them, though. |
| 03:08.09 | Kobaz | haha |
| 03:08.13 | Kobaz | doesn't sound like a fork |
| 03:08.18 | Kobaz | you mean freeswitch? |
| 03:08.20 | Penguin | They built FreeSWITCH, which is not a fork of asterisk. |
| 03:08.33 | WIMPy | It isn't? |
| 03:08.41 | Kobaz | no it's not a fork |
| 03:08.55 | Penguin | It's a fork of asterisk just like Linux is a fork of UNIX. |
| 03:09.00 | Kobaz | it's built ground up with all the failings of early asterisk considered and avoided |
| 03:09.02 | WIMPy | Then why do asterisk channels compile for freeswitch as well? |
| 03:09.06 | WIMPy | Or used to do. |
| 03:09.18 | Kobaz | compat layers |
| 03:09.19 | TBotNik | Penguin: Not so, most still have Asterisk as base and mostly in same folders, but some moved the folders around, so then you think Asterisk is not there but it is! |
| 03:09.49 | Penguin | I don't know of one asterisk fork. Not a single one. |
| 03:10.05 | Kobaz | "avoiding deadlock" |
| 03:10.16 | Kobaz | that was some good steak |
| 03:11.06 | Kobaz | the whole premise was having each channel run by a state machine so everything is very predictable and in sync |
| 03:11.10 | Kobaz | which sounds like a good idea |
| 03:11.33 | TBotNik | All: So what version of the PolyCom driver support my IP4000 for Asterisk? |
| 03:11.52 | TBotNik | SP drivers |
| 03:13.23 | TBotNik | Aldo posted my issues at: http://community.polycom.com/t5/VoIP/Error-Application-Not-Found/m-p/20468/highlight/false#M2946 |
| 03:13.31 | Kobaz | yay grub installed |
| 03:14.02 | Kobaz | reinstalling my backup server |
| 03:14.05 | Kobaz | with 64bits |
| 03:14.13 | TBotNik | is there a polycom channel out here? PolyCom support is useless since QualCom bought them. |
| 03:14.28 | Kobaz | i'm in the process of getting polycom certified |
| 03:14.37 | Kobaz | so i can call up and say, yo, your firmware is broken, fix it |
| 03:18.29 | TBotNik | All: Use to be PolyCom gave response to forum posts in uder 24 hours, now you are lucky if you get response in a week. |
| 03:18.56 | Kobaz | and generally the response is |
| 03:19.07 | Kobaz | "we don't directly support end users, look at the docs, k, pls, tks" |
| 03:20.28 | TBotNik | Kobaz: They don't respond to tickets either. |
| 03:20.58 | TBotNik | Kobaz: Have a ticket open on this now for 2 month and nothing. |
| 03:22.16 | TBotNik | Kobaz: Qualcom gutted the support staff, to save money. Think there used to be 20 support engineers, now there are 2. |
| 03:22.38 | Kobaz | oh wow |
| 03:22.48 | Kobaz | are you certified? |
| 03:23.21 | Penguin | They probably dropped the certification program, too. |
| 03:23.23 | TBotNik | Not on Asterisk, my Certs are in PHP. Have Level III cert there. |
| 03:23.40 | TBotNik | Also SEO and Security certified. |
| 03:24.08 | TBotNik | 6 level of PHP certs, so halfway now! |
| 03:24.12 | Kobaz | no i mean polycom certified |
| 03:24.21 | Kobaz | like if you get polycom cert they'll supposidly answer your tickets |
| 03:24.39 | Kobaz | and if you pay them some $$, then you supposedly get access to level3 techs |
| 03:25.31 | TBotNik | No and only touch PC if customer specs on system. My choice either GrandStream or Aastera as much much cheaper and GS is better quality than PC. |
| 03:25.42 | Kobaz | what? |
| 03:25.43 | Kobaz | no wa |
| 03:25.51 | Kobaz | grandstream has terrible call quality |
| 03:25.56 | Penguin | Someone has some shitty PCs. |
| 03:27.18 | TBotNik | I never have call quality issues. That all originates in the T1 provider and correct provisioning of the line. Get the right line and all QoS issues are gone. Who do you use? |
| 03:27.21 | Kobaz | these are older phones but i remember the gxp2000's were TERRIBLE |
| 03:27.32 | Kobaz | like, if you picked up the handset |
| 03:27.33 | Kobaz | it was all crackley and really bad |
| 03:27.39 | Kobaz | the speakerphone was actually better than the handset |
| 03:28.32 | TBotNik | I remember those, but was just a bad batch, they still make them and are solid now, least I'm not having issues. |
| 03:30.58 | TBotNik | If you want the best it is the UK company that actually own all the technology Cisco licensed from them. Yes Cisco owns absolutely nothing SIP/VOIP, they licensed it all from UK Company. Forgot the name and so close to ShoreTel that I get them mixed up. Something like CenTel, but forgot. |
| 03:31.43 | Penguin | I thought Polycom used to make the Cisco stuff. |
| 03:32.09 | MaliutaLap | What about the stuff that Cisco took while it owned Linksys? |
| 03:32.15 | WIMPy | And I thought Ciscos SIP support sucked? |
| 03:32.20 | Kobaz | the spa phones aren't that bad actually |
| 03:32.29 | Kobaz | cisco sip on the skinny phones is terrible |
| 03:32.31 | WIMPy | Not bad? |
| 03:32.32 | MaliutaLap | WIMPy: I use it, it's not that bad at all |
| 03:32.36 | TBotNik | Penguin: Don't think so, but with all the inter-supplier relations, who knows? |
| 03:32.39 | Kobaz | yeah spa's not bad |
| 03:33.04 | Penguin | Like SIP on a 7960... not very good at all. |
| 03:33.08 | WIMPy | The Linksys stuff was taken over from Sipura and I haven't heard from anyone who liked them. |
| 03:33.22 | Kobaz | like the newest spa is actually decent |
| 03:33.51 | MaliutaLap | Penguin: 7940 - NP |
| 03:34.03 | WIMPy | Can you hande multiple calls now in a half way acceptable way? |
| 03:34.08 | Penguin | Like SIP on a 7940... not very good at all. |
| 03:34.10 | MaliutaLap | yes |
| 03:34.18 | Penguin | 7940 and 7960 are the same, by the way. |
| 03:34.19 | MaliutaLap | Penguin: wrong |
| 03:34.28 | MaliutaLap | Sameish |
| 03:34.33 | Penguin | Maybe your opinion is wrong. |
| 03:34.55 | Penguin | The SIP firmware provides almost no useful features on the 7900 series phones. |
| 03:34.58 | MaliutaLap | Penguin: the functioning 7940 with SIP firmware on my desktop says otherwise |
| 03:35.07 | Penguin | Yes they can make calls. Yes they can receive calls. |
| 03:35.10 | Kobaz | haha |
| 03:35.10 | Kobaz | yeah |
| 03:35.31 | WIMPy | Not something to be taken for granted these days. |
| 03:35.33 | Penguin | But where are the features?! Not in the SIP image, that's where. |
| 03:35.36 | MaliutaLap | Penguin: are you using reinvite? |
| 03:35.58 | Penguin | I don't use the SIP images because they aren't featureful. And yes I do use directmedia. |
| 03:36.02 | MaliutaLap | I leave * in the loop - mainly out of paranoia than anything |
| 03:36.38 | Penguin | When I first got into the 7900 series phones I put SIP on them because I was told I had to use SIP when using those phones with asterisk. |
| 03:36.53 | Penguin | They made calls and they got calls, but there wasn't a whole lot more they did. |
| 03:37.07 | Penguin | If that's all you are after, you're set! |
| 03:37.27 | Penguin | If you want more, consider not using SIP or consider different phones. |
| 03:37.50 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 03:37.55 | Kobaz | chan_skinny is nice and crashy |
| 03:38.00 | Kobaz | at least it was the last time i tried it |
| 03:38.02 | Penguin | It was useless. |
| 03:38.05 | Penguin | totally useless. |
| 03:38.22 | Kobaz | like, make a phone call |
| 03:38.24 | Kobaz | it works |
| 03:38.25 | Penguin | I had to use chan_sccp-b to get anything useful out of skinny/asterisk. |
| 03:38.27 | Kobaz | put someone on hold |
| 03:38.29 | Kobaz | crash |
| 03:38.50 | Penguin | I heard that asterisk 10+ got a rewrite to chan_skinny, but I haven't tested that out yet. |
| 03:38.54 | Kobaz | yeah |
| 03:38.56 | Kobaz | i heard it's good |
| 03:39.00 | Penguin | That would be nice. |
| 03:39.16 | MaliutaLap | there was supposedly some *+mgcp stuff that worked |
| 03:39.18 | Penguin | I still have some trouble with chan_sccp, but nothing like chan_skinny. |
| 03:39.21 | MaliutaLap | never bothered though |
| 03:39.39 | Penguin | I didn't have much luck setting up MGCP on asterisk. I gave up after a few minutes of trying. |
| 03:40.04 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 03:40.25 | Penguin | I can't rate MGCP on asterisk since I never actually used it. |
| 03:41.36 | Penguin | I only even considered it because I had this one 7960 that would not convert from my SIP version to the SCCP version I wanted to use on it. I was able to get it to go to MGCP as an intermediate step, and I thought I would stop there for a minute to see how it was. |
| 03:42.04 | *** part/#asterisk k-man (~jason@unaffiliated/k-man) |
| 03:42.09 | Penguin | It didn't work out for me, so I went on to the SCCP image, which I still use today. |
| 03:45.32 | TBotNik | All: OK repeating my issue for all to understand: |
| 03:45.32 | TBotNik | Have 2ea. PolyCom IP4000 conf phone and tried connecting them to both a.) Fonality PBXtra |
| 03:45.32 | TBotNik | and b.) Elastix and always get error: "application not present"! |
| 03:45.32 | TBotNik | Any suggestions? |
| 03:45.32 | TBotNik | Have copied most of the files in /tfptboot per docs on this. All other PolyCom phones |
| 03:45.33 | TBotNik | on this system are on ver 3.2.3 and not sure what version supports IP4000. |
| 03:45.34 | TBotNik | Some things I read say only 3.1.7/8 are only support version for IP4000. |
| 03:45.37 | TBotNik | The "README" file is 3.1.7 says it supports IP4000 but still not working. |
| 03:45.39 | TBotNik | I posted this on the Asterisk forum at: |
| 03:45.41 | TBotNik | http://forums.digium.com/viewtopic.php?f=1&t=89494&p=197108#p197108 |
| 03:45.43 | TBotNik | and on the PolyCom forum at: |
| 03:45.46 | TBotNik | http://community.polycom.com/t5/VoIP/Error-Application-Not-Found/m-p/20468/highlight/false#M2946 |
| 03:45.48 | TBotNik | All help appreciated! |
| 03:45.50 | TBotNik | Thanks! |
| 03:45.52 | TBotNik | TBNK |
| 03:47.26 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-wliaonepmydahniw) |
| 03:53.10 | TBotNik | All: Is my issue with the sig or cfg files? Do I need to PasteBin them? |
| 03:55.19 | Kobaz | application not found means the application is not found |
| 03:55.24 | Kobaz | your firmware is missing for that phone |
| 04:01.10 | *** join/#asterisk AceFrahm (~SG2W03n4m@75-162-242-88.slkc.qwest.net) |
| 04:04.46 | *** join/#asterisk D30_ (~deo@222.127.13.226) |
| 04:06.04 | TBotNik | All:My files for these are: |
| 04:06.04 | TBotNik | phone1macid-appFlash.log |
| 04:06.04 | TBotNik | phone1macid-boot.log |
| 04:06.04 | TBotNik | phone1macid.cfg |
| 04:06.04 | TBotNik | phone1macid-phone.cfg |
| 04:06.05 | TBotNik | phone-phone1macid.cfg |
| 04:06.07 | TBotNik | phone2macid-appFlash.log |
| 04:06.09 | TBotNik | phone2macid-boot.log |
| 04:06.11 | TBotNik | phone2macid.cfg |
| 04:06.12 | Penguin | ~pb |
| 04:06.13 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:06.13 | TBotNik | phone2macid-phone.cfg |
| 04:06.15 | TBotNik | phone-phone2macid.cfg |
| 04:06.18 | TBotNik | sip.SPIP4000.3.2.3.revc.ld |
| 04:06.20 | TBotNik | I can PasteBin these, if that will help determine my issue, which I think is in these files. |
| 04:06.22 | TBotNik | I may need the sip.SPIP4000.3.2.3.revc.ld replaced with sip.SPIP4000.3.1.7.revc.ld but not sure. |
| 04:06.24 | TBotNik | All help appreciated! |
| 04:06.25 | Penguin | ~pb |
| 04:06.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:06.26 | TBotNik | Thanks! |
| 04:06.29 | TBotNik | TBNK |
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| 04:10.11 | TBotNik | Kobaz: <Kobaz> your firmware is missing for that phone. Yes but after searching for it now for 3 weeks, not sure what/where it is. Version 3.1.7 and 3.1.8 README files say they support IP4000, but give you generic files to copy over and hand configure. Not sure which files to copy, and edit due to non specific nature of the instructions on these. |
| 04:13.15 | TBotNik | Kobaz the files I copied and edited were of existing IP500 phones on the system, which I edited for 1.) Display Name, 2.) Ext, 3.) Config file location. Could not see anything else to edit. |
| 04:14.18 | TBotNik | Kobaz: Also edit MAC address in those for the IP4000 MAC addresses. |
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| 05:05.09 | TBotNik | Hello anyone still there? |
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| 06:27.55 | ChannelZ | kinda |
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| 06:54.32 | bsdice | Good morning |
| 07:00.11 | MaliutaLap | It's 5pm :P |
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| 08:02.35 | Entelin | I can call into an extension from the outside, but when I press a key I cannot hear anything, and it doesnt recognize that I have pressed something (this is inside of a WaitExten) would this likely be a port issue? |
| 08:03.08 | Entelin | though my sip phone seems to work fine |
| 08:04.41 | Entelin | outside in this case from my sip trunk from nexvortex |
| 08:04.48 | Entelin | from a cellphone |
| 08:05.34 | Entelin | http://codepad.org/D1VzsorJ |
| 08:05.46 | Entelin | thats my little hello world dialplan |
| 08:06.30 | Entelin | i can hear my phone system, like if I put a background or playback, but I cannot seem to send keypresses |
| 08:10.23 | Entelin | actually that paste was messed up, http://codepad.org/4zNZgSZn |
| 08:10.41 | Entelin | its still an issue though |
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| 08:49.38 | bsdice | MaliutaLap you must be an Aussie then |
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| 09:49.15 | boratynskikamil | Would you recommend to play WAV files or MP3 files? |
| 09:49.42 | boratynskikamil | As MoH, ofc. |
| 09:50.34 | Chainsaw | I use S3M files. |
| 09:50.58 | Chainsaw | But assuming G711 uLaw/aLaw, MP3 will suffice. The additional quality of a WAV is unlikely to be audible. |
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| 11:39.41 | fornax | hi, have a one-way voice problem with dahdi where i need help, is there someone who can give me a hint? |
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| 12:01.59 | boratynskikamil | Question. I have such as extension: |
| 12:02.00 | boratynskikamil | http://wklej.org/id/1284722/ |
| 12:02.16 | boratynskikamil | When I hangup phone, it phones one more time, what's wrong? |
| 12:11.54 | mirela666 | Print out the DIALSTATUS afer the frst Dial and see what happens (should be Dialing again only on busy signal) |
| 12:23.03 | boratynskikamil | mirela666: Verbose? |
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| 12:28.44 | mirela666 | Verbose (3,..), Log(ERROR,...), Nooop(..) any |
| 12:29.04 | mirela666 | Noop* |
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| 12:59.05 | MaliutaLap | bsdice: yes I am :) |
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| 13:06.44 | fornax | probably now someone here who can help me debugging a dahdi one-way voice problem? |
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| 13:12.58 | [TK]D-Fender | What backup have you provided? What are you running now? |
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| 14:35.43 | maruen | Hi All...does anyone here could get to work asterisk on Max OS X Mavericks ? |
| 14:42.20 | Katty | morning |
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| 14:51.15 | TBotNik | All: Hey finally got my Polycom IP4000 phones to boot but still no outside lines or intercom calls (ext to ext). These are hosted on Fonality PBXtra box, which has no interface for outside lines or other Asterisk level configs. It only allows you to set the external direct dial number, nothing else. Have not had to manhandle Asterisk before, so can use some help in this process. Any step-by-step config instructions would be great. Been here |
| 14:51.15 | TBotNik | <PROTECTED> |
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| 14:52.49 | [TK]D-Fender | TBotNik: If you have no access to Asterisk itself on that box we cannot help you. That is a closed commercial GUI |
| 14:53.26 | TBotNik | All: OK to start with, where in Asterisk do I find the files where I set the outside lines? |
| 14:54.56 | TBotNik | [TK]D-Fender: Hey I have both root and admin on the box at OS level, so can and will by-pass the "Web Enabled" GUI |
| 14:55.45 | [TK]D-Fender | The GUI builds those things and we do not know how. Also based on its heritage any changes you attempt to make directly to it will get blown away once you apply any other change in the GUI |
| 14:55.53 | [TK]D-Fender | So you have to obey their rules |
| 14:55.55 | Katty | waves to sruffell |
| 14:56.02 | [TK]D-Fender | Your are not configuring Asterisk ... they are |
| 14:56.14 | sruffell | mornin' |
| 14:56.25 | Katty | how're you dear |
| 14:59.43 | MaliutaLap | waves at Katty |
| 14:59.49 | TBotNik | [TK]D-Fender: We are in the process of replacing the Fonality with Elastix box. Right now have the Elastix box as backup. Need T1 card to complete. Then will be off that junk. Fonality contract expired last Sept and have not renewed. Only have to get us off the integrated T1 plan to a real SIP trunk plan to complete the change, but need the new phones working ASAP |
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| 15:00.55 | [TK]D-Fender | TBotNik: Elastix is another mess. They do a hatchet-job hacking FreePBX GUI elements up and make supporting it a real pain.. |
| 15:01.01 | MaliutaLap | Katty: you should have come to the ballet with me - if you flew supersonic (like "borrowing" an F-16) you could still make the performance that's in 17 hours :) |
| 15:01.41 | MaliutaLap | Katty: although I'm not sure the RAAF would let you land at Amberley :) |
| 15:02.17 | [TK]D-Fender | TBotNik: Also no need to wait for a T1 card to get off your hosted PBX solution if you were using a SIP ITSP connection with them... jsut bring it over |
| 15:02.35 | TBotNik | [TK]D-Fender: I have been working with Elastix for over 5 years and perfer it over others because of the addons that give me the extra features my customers want. |
| 15:02.49 | MaliutaLap | [TK]D-Fender: lets be honest, pretty much every GUI/Web-UI - of anything - is a hatchet job that doesn't let you control jack shite properly :) |
| 15:03.31 | [TK]D-Fender | MaliutaLap: Not really all that bad. Stock FreePBX is ok, and supportable |
| 15:04.19 | TBotNik | [TK]D-Fender, MaliutaLap: Yes Web GUIs can usually never give you all the features, some are better, but if you want it all, back to hand editing files and cli commands. |
| 15:04.39 | MaliutaLap | [TK]D-Fender: I'm adverse to them. It always seems that the thing I want to tweak isn't available for tweaking. That and having being forced to work with a couple that did nasty web hosty/vm shite ... |
| 15:05.04 | MaliutaLap | TBotNik: ahh, that's what I do for a living |
| 15:05.32 | MaliutaLap | TBotNik: that and breaking support communities because things fell out of the docs 10 years ago |
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| 15:06.17 | TBotNik | <[TK]D-Fender>,<MaliutaLap>: So where in Asterisk do I find the files or DBs to set the outside lines? |
| 15:06.34 | Katty | hugs MaliutaLap |
| 15:06.41 | Katty | MaliutaLap: ballet is always lovely :> i would have enjoyed it! |
| 15:06.53 | Katty | MaliutaLap: can't today tho, testing backups :< |
| 15:07.06 | Katty | it's Serious Important Business |
| 15:08.00 | [TK]D-Fender | TBotNik: You don't |
| 15:08.39 | [TK]D-Fender | TBotNik: There is no "asterisk database" for such things. They are CONCEPTS. How your GUI builds those in Asterisk for you we cannot say. PBXtra is a CLOSED system and we don't know |
| 15:08.50 | TBotNik | [TK]D-Fender:, MaliutaLap: Should I be looking inthe /pbx or the /configs directory? |
| 15:09.13 | [TK]D-Fender | TBotNik: You should be looking "nowhere" in raw configs". Your GUI owns your setup. You need to learn it |
| 15:09.37 | MaliutaLap | <PROTECTED> |
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| 15:56.38 | TBotNik | All: OK found the extension to line assignments in the /etc/asterisk/sip.conf file and added my 2 extensions there. Reboot of server and now get an "Your Call can not be completed" message from the system, but still no connect to the actual outside lines. |
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| 15:58.04 | [TK]D-Fender | TBotNik: We cannot support your GUI. You'll have to contact Fonality |
| 15:58.19 | TBotNik | All: Also call in to that extension from outside and extension is ignore, saying "Go to another mailbox" |
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| 16:10.12 | _omer | Hello, I want to save Asterisk Manager Events in MySQL......Any Suggestions please? |
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| 16:13.38 | [TK]D-Fender | _omer: Get coding.... |
| 16:14.09 | _omer | I thought if asterisk have something builtin .... |
| 16:15.05 | [TK]D-Fender | _omer: nope. |
| 16:15.44 | [TK]D-Fender | _omer: Did you see a config file stating it? entry in the official wiki? In the book? |
| 16:17.08 | _omer | According to my knowledge, Asterisk does not have any builtin feature to save AMI events in MYSQL ... but my knowledge about asterisk is still limited so I thought to have experts advise before a big NO :) |
| 16:17.41 | _omer | Alright. I want to see if a Realtime-Peer is online or offline without AMI ... |
| 16:21.13 | [TK]D-Fender | _omer: I don't know if qualify/reg stuff is regularly pushed back to the DB or not. |
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| 16:24.46 | ddadminman | Hello |
| 16:25.16 | ddadminman | We have 8 pots lines coming in and was curious was is the recommended hardware to integrate those lines into a asterisk server we have? |
| 16:25.22 | ddadminman | what are my options? |
| 16:27.05 | [TK]D-Fender | PCI(e) cards, SIP gateways, USB gateways, etc |
| 16:28.35 | _omer | yes. this is what i was thinking about .... qualify/qualifyfrequency |
| 16:29.16 | _omer | thanks |
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| 17:57.35 | xaristax | i know this is probably simple and more related to linux but im getting this error when i try to use mix monitor st_writefile: Unable to open file |
| 17:57.48 | xaristax | i set the permisions to the folder and make asterisk the owner |
| 17:57.56 | xaristax | can anyone give me a light on this? |
| 17:58.33 | xaristax | if i set the monitor folder as defaults works great but only on this folder |
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| 18:03.28 | Entelin | http://codepad.org/kM2rdo7r could someone |
| 18:03.30 | Entelin | er |
| 18:04.06 | Entelin | * have a look at that and tell me if theres anything in there that would cause one side not to hear anything over the call? |
| 18:05.15 | [TK]D-Fender | extensions.conf has nothing to do with the audio of your devices |
| 18:05.18 | Entelin | basically if I call in from the nexvortex rule from my cell, to exten 10, I can |
| 18:05.22 | Entelin | ok |
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| 18:05.33 | Entelin | i'm thinking its got something to do with nat |
| 18:07.37 | [TK]D-Fender | It certainly could if it's involved. |
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| 18:09.34 | Entelin | is externip= supposed to appear only in sip_nat.conf ? i'm reading on a wiki that its supposed to be in general of sip.conf? |
| 18:10.08 | file | depends on the configuration layout, as you can include other files |
| 18:10.52 | Entelin | is sip_nat.conf a standard asterisk file? |
| 18:10.52 | [TK]D-Fender | that is not a base asterisk config file... |
| 18:10.55 | Entelin | ok |
| 18:11.01 | Entelin | so it should be in general then |
| 18:11.18 | [TK]D-Fender | there are general settings, and there are device-specific settings |
| 18:12.40 | Entelin | ok sweet that was the issue |
| 18:12.57 | Entelin | i was following some asterisk docs that were for some specific configuration |
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| 18:48.08 | jeffspeff | I'm having an issue with the call from 1506 to 2113. For some reason 2113 is not able to hear any audio on any calls (not just from 1506). I'm not seeing anything wrong in the debug log, but here it is... any suggestions? http://pastebin.com/x8XEQjRt |
| 18:58.21 | *** join/#asterisk jgowdy (~bt@cpe-76-176-91-127.san.res.rr.com) |
| 18:58.40 | jgowdy | is there some reason I can't symlink extensions.conf into /etc/asterisk/ rather than it being a real file? |
| 19:01.17 | pabelanger | jgowdy, nope |
| 19:01.57 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:02.50 | *** join/#asterisk vlad_sta_ (~vlad_star@77.41.82.191) |
| 19:04.35 | jeffspeff | I'm having an issue with the call from 1506 to 2113. For some reason 2113 is not able to hear any audio on any calls (not just from 1506). I'm not seeing anything wrong in the debug log, but here it is... any suggestions? http://pastebin.com/x8XEQjRt |
| 19:07.33 | mjordan | ~book |
| 19:07.33 | 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 |
| 19:08.34 | jeffspeff | mjordan, by suggesting the book, are you implying it's a nat issue? nat and rtp settings haven't changed. the phone was working fine yesterday and days previous. |
| 19:08.46 | mjordan | jeffspeff: I'm sorry, but I wasn't actually |
| 19:08.52 | mjordan | I needed the infobot blurb for another channel :-D |
| 19:09.01 | jeffspeff | LOL, ok |
| 19:17.29 | *** join/#asterisk navaismo (~navaismo@189.146.21.208) |
| 19:19.05 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:22.14 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:22.40 | newtonr | jeffspeff, I'll take a look |
| 19:22.53 | MaliutaLap | jeffspeff: so it worked up until now - no changes on the * box, phone, or NAT point? == network problems ... tcpdump is your friend :) |
| 19:24.46 | jeffspeff | the phone is a conference phone (polycom ip 7000) it's not in the same office as i am. the users i've spoken to say they haven't changed anything, but i swear this issue is like the volume is turned all the way down or something. no changes were made to the network, asterisk or the phones config files. i have rebooted the phone a few times to no avail. |
| 19:26.02 | jeffspeff | i'm about to have a user down there reset it to defaults and start it from scratch. hopefully if it's a user error (like volume muted) then that'll get rid of it. and yes, i've asked them to make sure the volume is turned all the way up; but you know how users are, he may have been looking at his cell phone. |
| 19:27.49 | *** join/#asterisk chapo (~chapo@unaffiliated/chapo) |
| 19:27.52 | chapo | Hi |
| 19:29.41 | chapo | I was wondering if somebody can help me, I´m testing asterisk, I did all the configuration from the source and its going well, I was able to do the hello world, im able to make and receive calls. Server is running on my LAN, softphones has been configured and Im able to reach them, dialing sound is fine, but when I answered the call I,m unable to hear the voice sound, any advice? |
| 19:30.18 | *** join/#asterisk rcaskey (~Rob@dumbledore.athenshousing.org) |
| 19:30.58 | Entelin | check to ensure the spi and voice ports are open and if its behind nat that you have that enabled and configured |
| 19:31.30 | Entelin | *sip |
| 19:31.37 | chapo | I´m just workign in the LAN, do I have to configure NAT? |
| 19:31.47 | chapo | files config can find here http://pastebin.com/xAK1J51N |
| 19:32.02 | newtonr | jeffspeff, SDP looks fine, nothing obviously wrong to me |
| 19:32.03 | Entelin | you arent calling in or out to a sip provider? |
| 19:32.17 | newtonr | jeffspeff, with rtp debug on what do you see when the call is up? |
| 19:32.32 | chapo | No I´ not using SIP provider, just local extension comunnication |
| 19:32.37 | jeffspeff | chapo, you need to set a localnet in your sip.conf |
| 19:32.38 | Entelin | ah |
| 19:32.38 | chapo | Ill show you the output |
| 19:33.23 | chapo | http://pastebin.com/bxbDx1HJ |
| 19:33.55 | jeffspeff | newtonr, I have a user in the office resetting the phone and reconfiguring it. If it still doesn't work at that point i'll try the rtp debug. thanks. i'll let you know my findings. |
| 19:34.26 | chapo | I see jeffspeff |
| 19:34.31 | chapo | you mean 192.168.0.0/24 |
| 19:34.32 | chapo | isnt? |
| 19:34.37 | chapo | variable localnet? |
| 19:35.16 | chapo | Ill check |
| 19:35.33 | jeffspeff | in the general context of sip.conf you need to define a localnet and you also need to set your bindaddr to the systems LAN ip |
| 19:35.50 | jeffspeff | then do a sip reload |
| 19:36.06 | chapo | ok |
| 19:37.08 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:39.59 | chapo | jeffspeff: this is what I did http://pastebin.com/27FMuGYM |
| 19:40.05 | chapo | still no voice sound |
| 19:40.13 | newtonr | jeffspeff, cool, yeah if things look good in SDP, i go to rtp debug and a pcap at various points in the network to verify RTP is actually making it to the phone. |
| 19:40.59 | jeffspeff | chapo the syntax is localnet=192.168.0.0/255.255.255.0 |
| 19:43.21 | jeffspeff | chapo, here's an example of my sip.conf you may find useful http://pastebin.com/gU9AjfVH |
| 19:43.47 | chapo | thanks man |
| 19:45.02 | chapo | i did the config with local net and bind addr ut still no voice let me see the file u sent me |
| 19:45.02 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 19:49.33 | *** join/#asterisk foamz (~foamy@pool-108-41-222-138.nycmny.fios.verizon.net) |
| 19:49.41 | chapo | new sip file http://pastebin.com/S2dQALke |
| 19:52.22 | *** join/#asterisk matrix1233 (~matrix123@41.228.40.6) |
| 19:52.25 | matrix1233 | hello |
| 19:52.26 | *** join/#asterisk retentiveboy_ (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net) |
| 19:54.59 | chapo | cant hear the voice sound in my lan, localnet and bind configured, im using zoiper softphone |
| 19:55.57 | matrix1233 | i have asterisk 11.7 and i have a problem with echo .. i have 300MB of band and its Intel(R) Core(TM) i7 CPU 950 @ 3.07GHz |
| 19:57.07 | matrix1233 | chapo: if its connected to provider see the nat of 10000:20000 udp |
| 19:57.18 | [TK]D-Fender | Echo clearly has nothing to do with CPU |
| 19:58.05 | chapo | Im not using sip provider |
| 19:58.17 | chapo | Im just testing in LAN enviroment |
| 19:58.28 | matrix1233 | [TK]D-Fender: justo for info .. so what can't be a search all solution and i dont know what i will do |
| 19:58.29 | chapo | Softphones are in the same LAN with local IP address |
| 19:59.00 | matrix1233 | have a firewall enabled ? |
| 19:59.15 | chapo | im running in debian my asterisk server, iptables -F |
| 19:59.23 | chapo | still no sound voice |
| 19:59.33 | matrix1233 | service iptables stop |
| 19:59.38 | matrix1233 | and try |
| 19:59.41 | chapo | already did |
| 19:59.44 | chapo | no luck |
| 20:00.05 | matrix1233 | stop the firewall in your pc |
| 20:00.22 | chapo | debug in asterisk cli its fine , wireshark everything looks great |
| 20:00.32 | matrix1233 | the antivirus can block also |
| 20:00.56 | chapo | I was able to hear messages like hello world but when I configured the extensions, no voice :s |
| 20:01.52 | chapo | other softphones clientes are running under debian where iptalbes isnt install |
| 20:02.28 | matrix1233 | [TK]D-Fender: how can i do a test procedure to found the problem of echo .. i have a provider with premium quality |
| 20:03.02 | [TK]D-Fender | matrix1233: You haven't actually said what you're using. |
| 20:03.40 | chapo | any advice matrix1233 ? |
| 20:04.04 | matrix1233 | [TK]D-Fender: i have asterisk 11.7 connected to provider |
| 20:04.36 | chapo | I was thinkign that could be codecs issues, idk |
| 20:04.37 | matrix1233 | [TK]D-Fender: the server have 300 Mega of band |
| 20:04.48 | [TK]D-Fender | matrix1233: you keep saying "provider" and not mentioning anything about the technology involved. You are also not saying what is on the OTHER end of the call. |
| 20:05.08 | [TK]D-Fender | matrix1233: You need to rethink your definition of "details" |
| 20:05.15 | matrix1233 | chapo: have u anadher box with asterisk where u know thats work 100%? |
| 20:05.34 | chapo | this is my first time doing the config |
| 20:06.29 | chapo | if i use the hold button im listening the music, but still no voice |
| 20:08.04 | matrix1233 | [TK]D-Fender: so the provider is an account sip installed on my asterisk like a trunk .. and i have a client that call via my server |
| 20:08.39 | [TK]D-Fender | matrix1233: So I have a SIP ITSP on one side of the call... and on the other? |
| 20:10.07 | chapo | is it weird huh listening the hold´ music but no voice matrix1233 |
| 20:10.34 | matrix1233 | [TK]D-Fender: softphone --SIP-> myserver --SIP--> provider |
| 20:11.04 | *** join/#asterisk tedstriker (~tedstrike@host-135-196-33-208.lines.viateldsl.com) |
| 20:11.16 | matrix1233 | chapo: you try to call a second local sip account ? |
| 20:11.16 | chapo | its working the hold in one pc, default coe GSM |
| 20:11.21 | chapo | yeah |
| 20:11.23 | chapo | i have 3 |
| 20:11.27 | [TK]D-Fender | Most softphones suck at echo-cancellation right at that point. Go test it against other things |
| 20:11.58 | chapo | It´s weird cause only in one pc the hold works, but in the other ones isnt letting me |
| 20:14.33 | matrix1233 | [TK]D-Fender: the problem that the softphone have echocancellation actived and also its not really one softphone ...it cant be the more detail ( Client:100 softphone--->servr of client-SIP->Myserver--->Provider) |
| 20:15.28 | chapo | voice mail its working now |
| 20:15.33 | chapo | no luck just the voice whle call |
| 20:16.05 | matrix1233 | chapo: when u call the second softphone the call is connected but no audio ? |
| 20:16.11 | chapo | yes |
| 20:16.17 | chapo | connection established i can hold |
| 20:16.19 | chapo | answer |
| 20:16.22 | chapo | but just no audio |
| 20:18.07 | matrix1233 | tos_sip=cs3 |
| 20:18.07 | matrix1233 | tos_audio=ef |
| 20:18.09 | matrix1233 | [TK]D-Fender: i actived the Qos the default parameter |
| 20:18.25 | matrix1233 | [TK]D-Fender: but its the same |
| 20:18.37 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:18.52 | matrix1233 | [TK]D-Fender: 30% of call haeve echo |
| 20:19.08 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 20:19.14 | *** join/#asterisk skrusty (~support@168.63.14.171) |
| 20:19.29 | skrusty | evening |
| 20:19.53 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 20:20.06 | skrusty | does anyone know if there is a way to use microsoft speech platform from asterisk? |
| 20:23.15 | matrix1233 | [TK]D-Fender: any idea ? |
| 20:24.48 | maruen | does anyone now how to solve this issue: aruen@Mehanas-Mac/usr/local/Cellar/asterisk/12.0.0/sbin$ sudo ./asterisk -r |
| 20:24.48 | maruen | Unable to connect to remote asterisk (does /usr/local/Cellar/asterisk/12.0.0/var/run/asterisk/asterisk.ctl exist?) |
| 20:25.11 | maruen | Im using Mac OS X Mavericks |
| 20:25.13 | matrix1233 | maruen: start asterisk |
| 20:26.18 | rrittgarn | @skrusty http://zaf.github.io/asterisk-googletts/ its google TTY, but depending on how microsoft's audio is delivered you should be able to figure something out |
| 20:26.30 | rrittgarn | /s/TTY/TTS |
| 20:26.35 | maruen | thanks matrix1233 |
| 20:29.34 | [TK]D-Fender | matrix1233: nothing networking related causes echo. ENDPOINTS cause echo. In your case your SOFTPHONE could be at fault, the headset (or whatever) you use with it, or however you terminate to the PSTN. |
| 20:31.40 | matrix1233 | [TK]D-Fender: the pstn can be on the provider but i dont have any access and the provider have tested and say that it have no problem |
| 20:32.24 | matrix1233 | [TK]D-Fender: there is a way to test that or install a module that correct the audio .. |
| 20:33.02 | [TK]D-Fender | matrix1233: No. It was to be corrected at the endpoint. |
| 20:33.18 | [TK]D-Fender | matrix1233: And what kind of provider sells you a product and says that it sucks? |
| 20:33.52 | *** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0) |
| 20:34.34 | [TK]D-Fender | matrix1233: Test all of your softphones, etc INTERNAL. If those are good 100% of the time, then it's your provider |
| 20:35.49 | matrix1233 | [TK]D-Fender: on the internal its ok ... |
| 20:36.26 | [TK]D-Fender | matrix1233: Then you either have acoustic echo due to too much volume on your headsets.... or your provider sucks |
| 20:37.15 | maruen | I got this message after try to run asterisk on mac os : "No 'modules.conf' found, no modules will be loaded |
| 20:37.23 | matrix1233 | [TK]D-Fender: the provider say alway thats is ok and i don't have nothing to do beacause it a a telecom |
| 20:37.34 | maruen | my asterisk installation is in /usr/local/Cellar/asterisk/12.0.0 |
| 20:37.39 | [TK]D-Fender | matrix1233: Changes nothing.... |
| 20:37.58 | maruen | where can I get this file and where should I put it? |
| 20:38.16 | matrix1233 | [TK]D-Fender: yeahh .. |
| 20:38.18 | [TK]D-Fender | maruen: "make samples" <------------- |
| 20:38.50 | matrix1233 | [TK]D-Fender: so i am searching a way to correct that ... |
| 20:39.06 | [TK]D-Fender | matrix1233: Change providers |
| 20:39.10 | maruen | [TK]D-Fender, I installed it with brew |
| 20:39.12 | matrix1233 | [TK]D-Fender: no solution with telecom so i have to found solution |
| 20:39.30 | jeffspeff | matrix1233, i'll be your provider |
| 20:39.40 | [TK]D-Fender | maruen: Go download the * tarball and grab the sample configs. |
| 20:39.40 | jeffspeff | i'll mail you a can with some string next week. :) |
| 20:39.47 | matrix1233 | jeffspeff: what u offer |
| 20:39.56 | jeffspeff | a can with strings |
| 20:40.00 | jeffspeff | only 1 can though |
| 20:40.09 | jeffspeff | more cans cost extra |
| 20:42.05 | maruen | [TK]D-Fender, ok, after it where do I put them? |
| 20:42.12 | maruen | I could installed it on linux |
| 20:42.21 | maruen | my question is in Max OS X |
| 20:42.31 | maruen | as it installs in a different place |
| 20:42.41 | maruen | and it seems with different structure |
| 20:42.52 | [TK]D-Fender | proportionately extremely few people use * on OSX. |
| 20:43.02 | [TK]D-Fender | You'd better know how your own system works |
| 20:43.25 | [TK]D-Fender | on linux std it's /etc/asterisk. |
| 20:43.34 | maruen | yeap |
| 20:44.00 | [TK]D-Fender | shove them whever you feel is appropriate and make sure that init scripts, etc know where to find it. You'll also have to tweak asterisk.conf, etc to account for this |
| 20:44.11 | maruen | in max os X I have the files in /usr/local/Cellar/asterisk/12.0.0 |
| 20:44.16 | maruen | but I dont have etc in it |
| 20:44.30 | matrix1233 | [TK]D-Fender: thanks |
| 20:44.31 | maruen | I created etc file inside it |
| 20:44.43 | maruen | but it stills does not load |
| 20:45.19 | [TK]D-Fender | We have no idea how you're trying to run it. Or where you configured it to find your modules in the first place. |
| 20:48.13 | skrusty | rrittgarn: i was hoping more for an mrcp connector :) |
| 20:49.56 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:55.41 | rrittgarn | Skrusty: sorry... I'm working less and less with windows these days and have not had any experience with that. Best of luck |
| 20:59.38 | skrusty | thanks :) |
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| 21:08.39 | matrix1233 | [TK]D-Fender: one think ..its a dedicated server from OVH (myserverasterisk ) it can't be the OVH that have a filter sip or that cause that ?? |
| 21:09.26 | [TK]D-Fender | matrix1233: No. |
| 21:09.34 | [TK]D-Fender | matrix1233: It's your provider, or your phone. |
| 21:10.12 | matrix1233 | [TK]D-Fender: ok thanks |
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| 21:17.51 | matrix1233 | [TK]D-Fender: sorry for my 1000 question ..but i need to know .. how can i configure a good configuration of Qos in my asterisk |
| 21:29.57 | [TK]D-Fender | qos options are in the sample configs... |
| 21:30.57 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 21:30.59 | matrix1233 | [TK]D-Fender: its the default setting and its alrady set but there is a way to configure an optimal configuration |
| 21:33.27 | [TK]D-Fender | If there was one magical setting why do you think that itt would be an OPTION? |
| 21:36.41 | matrix1233 | [TK]D-Fender: i found this .. not really an option but more http://www.linuxexpert.ro/VoIP-Tutorials/asterisk-qos.html |
| 21:37.18 | MaliutaLap | matrix1233: you _could_ do it that way ... or you could do it properly with iptables and iproute |
| 21:37.55 | MaliutaLap | matrix1233: it also depends on how your switches are conf'd and some other stuffs |
| 21:38.02 | [TK]D-Fender | checkout time, heading home. |
| 21:38.30 | matrix1233 | MaliutaLap: its a dedicated server on OVH |
| 21:39.08 | MaliutaLap | matrix1233: so you have no control over the switches or the outgoing 'net connection? |
| 21:40.45 | matrix1233 | MaliutaLap: no |
| 21:40.47 | MaliutaLap | matrix1233: in that case it wouldn't be 100% effective, look at DSCP marking in iptables |
| 21:41.14 | matrix1233 | MaliutaLap: how can i see that |
| 21:41.21 | MaliutaLap | in general things _should_ honour DSCP - the TOS stuff was deprecated |
| 21:41.39 | matrix1233 | MaliutaLap: am using ufw o simplify iptables |
| 21:43.36 | MaliutaLap | matrix1233: read http://en.wikipedia.org/wiki/Differentiated_services and http://www.linuxtopia.org/Linux_Firewall_iptables/x4172.html |
| 21:44.33 | MaliutaLap | matrix1233: can't help with ufw - I use ip[6]tables, and am trying to migrate to nftables |
| 21:45.12 | matrix1233 | MaliutaLap: ok thanks |
| 21:45.30 | MaliutaLap | matrix1233: you have to realise that doing QoS on linux means tagging, then using tc to set queues and filters |
| 21:46.20 | MaliutaLap | matrix1233: those two are a good start - there is also one out there about how you should set DSCP numbers for VoIP content |
| 21:48.24 | matrix1233 | MaliutaLap: ok thanks am reading |
| 22:00.20 | kruftindustries | Anyone know how to hang up the called party with Dial() command? |
| 22:05.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:20.33 | navaismo | uh |
| 22:20.47 | navaismo | so the g option didnt work? |
| 22:24.12 | MaliutaLap | navaismo: the g-string option? thought that was for strip clubs ;P |
| 22:34.32 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 22:39.56 | mjordan | kruftindustries: I'm not sure I understand the question. You don't hang up the called party using Dial, you call them. Can you describe in more detail how you want the called party to be hung up? |
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| 23:27.33 | jeffspeff | newtonr, here's an excerpt of my rtp debug you asked for http://pastebin.com/p4qfyMci i'm still not seeing where this problem is |
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| 23:40.30 | igcewieling | Is there an easy way to figure out which side hungup when using hangup handlers? The old way, if "g" was on dial and the destination hangs up the dialplan continues for source. If the source hangs up then exten "h" is run. I'm not how to translate that to use hangup handlers. |
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