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| 04:36.52 | diegoviola | hi |
| 04:36.56 | diegoviola | is asterisk supporting pjsip these days? |
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| 04:40.46 | [TK]D-Fender | for the lsat 2 versions... |
| 04:41.35 | diegoviola | is it stable? |
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| 04:50.08 | [TK]D-Fender | Why wouldn't it be? |
| 04:53.54 | diegoviola | [TK]D-Fender: just trying to see what changed in asterisk in the last few years, it does seem like it has improved in many ways |
| 04:54.17 | [TK]D-Fender | There is a doc that explains all these changes |
| 04:54.26 | [TK]D-Fender | Go read them for each branch you missed |
| 04:54.32 | diegoviola | okay |
| 04:54.39 | diegoviola | I recall trolling this channel a few times, sorry about that |
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| 06:58.29 | gustopn | lol, did I tell you folks what was causing my audio to become sluggish? |
| 07:10.23 | drmessano | ? |
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| 08:33.21 | gustopn | well |
| 08:33.29 | gustopn | my router runing openwrt |
| 08:33.37 | gustopn | it hit the top of the cpu load |
| 08:34.17 | gustopn | however, i can not understand it that he is packed with transcoding one or two channels from alaw to g711 and reverse |
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| 08:34.43 | gustopn | the second thing i can not understand is why he does that transocding at all, when there is alaw on both sides |
| 08:35.22 | gustopn | the problem is that you have first g711a -> g722 and g722 -> g711a in the other direction |
| 08:35.39 | gustopn | now he sees that he can not do g722 because an old telephone picked up |
| 08:36.13 | gustopn | first he transcodes g711a -> sln -> sln16 and out sln16 -> g722 |
| 08:36.37 | gustopn | after the change he transcodes g711a -> sln -> sln16 and out sln16 -> sln -> g711a |
| 08:37.16 | gustopn | and the same bullshit backwards ... all the time going over sln16 for transcoding g711a to g711a WTF? |
| 08:37.26 | gustopn | it is asterisk 11 |
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| 08:57.41 | gustopn | btw. I also use now another cofig_dir and since then my ACL.conf is ignored, someone an idea? maybe it wants to use the default dir for ACL? I havent tried to change the old dir's ACL.conf if then something show up yet |
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| 09:13.32 | WIMPy | gustopn: Future jersions of chan_pjsip will probably avoid transcoding inthese cases. |
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| 09:44.47 | TyronhMaximus | hello |
| 09:45.22 | TyronhMaximus | I've some questions related the integration of google hangout with the asterisk system |
| 09:45.54 | TyronhMaximus | is there anyone that can help or have the experience to do this? |
| 09:47.15 | WIMPy | As far as I know there's no such thing. |
| 09:48.28 | sekil | alaw and g711 is the same thing |
| 09:48.46 | WIMPy | No |
| 09:48.54 | WIMPy | The can be the same thing. |
| 09:49.03 | sekil | g711a that is |
| 09:49.23 | WIMPy | Yes. G.711 describes both a-law and µ-law. |
| 09:50.02 | sekil | but basically u-law and alaw are the same thing...transcoding in between can't be all that bad |
| 09:50.08 | sekil | resource wise |
| 09:50.42 | WIMPy | It can be done with a simple table. |
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| 11:12.04 | gustopn | I meant g722 and alaw of course |
| 11:12.12 | gustopn | my mistake i wrote g711 there |
| 11:13.13 | gustopn | the transcodes from alaw to alaw over sln and sln16 |
| 11:13.30 | gustopn | I do not use µ-law at all |
| 11:14.11 | gustopn | and for that one has that a-mu-law codec module for transcoding between that two |
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| 11:34.47 | russlevy | hi, is it possible to get a status update on a specific JIRA bug that references an internal bug? |
| 11:37.53 | file | any updates or information would be on the public JIRA issue |
| 11:39.44 | russlevy | so if there's no public updates then can't find out status, correct? |
| 11:40.05 | russlevy | and if i am reading this bug correctly https://issues.asterisk.org/jira/browse/ASTERISK-24602 there are no updates in 6 months |
| 11:40.21 | file | the status is that the issue remains open |
| 11:40.44 | file | no work has been done by anyone, no patches are present |
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| 11:41.39 | afournier | hi |
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| 11:49.46 | russlevy | thanks @file |
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| 11:52.59 | russlevy | @file is there a way to see the internal issue so my team can work on a patch? |
| 11:56.15 | file | russlevy, nope |
| 11:56.17 | file | russlevy, there is nothing of use/additional on it |
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| 11:59.20 | russlevy | ok thanks. would this issue probably go away if we move back to chan_sip? |
| 11:59.32 | file | it's possible |
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| 12:00.27 | russlevy | ok -- so to confirm, the problem is probably somewhere in chan_pjsip, and moving to chan_sip probably would fix it |
| 12:01.24 | file | without looking at the issue and testing things, I don't know |
| 12:01.48 | file | it is possible that it is only specific to chan_pjsip and chan_sip would not have the problem |
| 12:02.26 | russlevy | ok that's all i want to know :) not holding you to anything |
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| 12:20.17 | qakhan | is there any way we can check which carrier a caller is using who is calling on dahdi (PRI line), like T-Mobile, AT&T or SIP trunking..... |
| 12:21.25 | [TK]D-Fender | No |
| 12:26.23 | qakhan | [TK]D-Fender i am getting almote 15 call from same phone number in 1 mins |
| 12:26.30 | qakhan | how to stop it? |
| 12:27.09 | WIMPy | Dialplan? |
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| 12:27.29 | qakhan | it seems like someone attacking on my PRI |
| 12:27.41 | qakhan | WIMPy i am getting calls in queue |
| 12:28.09 | WIMPy | What does that mean? |
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| 12:32.50 | qakhan | WIMPy here http://pastebin.com/pWsg4bcx |
| 12:33.42 | WIMPy | So? |
| 12:33.46 | [TK]D-Fender | qakhan: That was pointless to show... |
| 12:33.52 | [TK]D-Fender | qakhan: just ignore them |
| 12:34.05 | qakhan | what do you need from me? |
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| 12:34.09 | [TK]D-Fender | nothing |
| 12:34.14 | WIMPy | nothing |
| 12:34.27 | [TK]D-Fender | ignore that caller |
| 12:34.35 | WIMPy | Extend your dialplan to ignore the calls you don't want. |
| 12:34.44 | qakhan | [TK]D-Fender i cannot, it is happening daily |
| 12:35.05 | [TK]D-Fender | qakhan: If it's the same caller IGNORE that caller |
| 12:35.09 | qakhan | i think he is using sip truning and keep changing the caller id |
| 12:35.14 | [TK]D-Fender | qakhan: And don't send them into your queue |
| 12:35.36 | qakhan | he keep changing the caller id |
| 12:36.01 | [TK]D-Fender | Go get PRI debug to compare for the calls and see if there is something in common for them |
| 12:36.14 | WIMPy | You said it were calls from the same phone number. |
| 12:36.21 | WIMPy | So what IS you question? |
| 12:37.16 | qakhan | i call back on those numbers and those people said they did not call. |
| 12:37.40 | qakhan | WIMPy i did not say that i am gettig call from same number |
| 12:38.16 | WIMPy | <qakhan> [TK]D-Fender i am getting almote 15 call from same phone number in 1 mins |
| 12:39.31 | c0ldg0ld | q |
| 12:39.35 | WIMPy | So if you can't identify those calls, you will obviousely have to ignore them manually. |
| 12:39.38 | c0ldg0ld | *sorry |
| 12:40.00 | qakhan | same number in 1 min next time some other caller id |
| 12:40.42 | WIMPy | Ok, that's something to work with. |
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| 14:14.24 | drathir | wonder of any working recomendation of gmail calls tutorials? |
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| 14:15.37 | drathir | mostly old versions availables... ofc with latest version... |
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| 15:04.43 | afournier | is there a way to check a device state from the cli (not custom device device state) ? |
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| 15:11.29 | anonymouz666 | node.js or python with Asterisk AMI? |
| 15:17.13 | newtonr | afournier, a direct CLI command? I don't think so. |
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| 15:29.22 | afournier | ok |
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| 15:39.37 | pa | WIMPy, hello :) you are surely more informed than me on LCR.. has there been any news on it lately? or is it still in the same semi-abandoned state? |
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| 16:02.07 | dimitry7 | hi guys, the quality of my voice is degraded, what can I do to mesure the Quality? |
| 16:02.22 | dimitry7 | I have only wan R720 server with 16 GB ram 24 cores as PBX |
| 16:02.30 | dimitry7 | I have 35 sip clients and 10 IAX clients |
| 16:02.35 | dimitry7 | I use E1 |
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| 16:33.59 | anonymouz666 | dimitry7: watch the rtcp reports |
| 16:34.21 | dimitry7 | anonymouz666, aparently it is just an extension :S |
| 16:34.32 | dimitry7 | but the users make it appear as the world's gonna end |
| 16:34.50 | anonymouz666 | watch this specific rtcp report then |
| 16:35.11 | anonymouz666 | all the users are like that |
| 16:35.30 | anonymouz666 | but remember, they make the company possible |
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| 16:42.52 | WIMPy | pa: has it been abandoned? |
| 16:45.21 | pa | WIMPy, well i don't think so |
| 16:45.28 | pa | but development seemed kind of stopped |
| 16:45.53 | WIMPy | Well, I did have an issue contacting Jolly lately. |
| 16:46.12 | WIMPy | After I discovered a nasty locking issue. |
| 16:46.21 | WIMPy | Or rhater matt did. |
| 16:47.02 | pa | WIMPy, last commit 8 months ago |
| 16:47.04 | pa | :) |
| 16:47.53 | WIMPy | I'm still on the last pre 1.13 version with some cherry picking and some own additions. |
| 16:48.03 | pa | yes me too, on your version :) |
| 16:48.07 | pa | but sometimes i have some weird problems |
| 16:48.12 | WIMPy | I don't like the new config. |
| 16:48.13 | pa | and i don't know if it's LCR or what |
| 16:48.28 | WIMPy | Well, I do, but in some situations the old one is superior. |
| 16:48.41 | WIMPy | What kind? |
| 16:48.46 | qakhan | what could be the prossiblility of delay of voice on call coming from the queue. agent can hear the caller but caller cannot hear the agent for 5 seconds |
| 16:48.57 | [TK]D-Fender | Slow RTP setup |
| 16:49.37 | qakhan | [TK]D-Fender RTP setup in sip.conf? |
| 16:50.10 | pa | WIMPy, well sometimes i hear noises such as a button dialed |
| 16:50.23 | pa | and with different people |
| 16:50.31 | pa | but only on LCR |
| 16:50.39 | pa | it doesnt happen when i call via google voice |
| 16:50.40 | [TK]D-Fender | <PROTECTED> |
| 16:50.44 | pa | or other virtual operators |
| 16:50.47 | [TK]D-Fender | qakhan: This is not a setting |
| 16:50.50 | WIMPy | By defauilt the DTMF decoder is voersesitive. |
| 16:50.52 | pa | so i dont know what it can be |
| 16:50.56 | pa | aah |
| 16:51.08 | pa | so you think it's some noise that sound like a button |
| 16:51.12 | pa | and is interpreted as such? |
| 16:51.43 | WIMPy | Try to load the mISDN_dsp module with dtmfthreshold=200. |
| 16:51.56 | pa | ah great, thanks i will try |
| 16:51.57 | pa | :) |
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| 16:52.49 | coppice | the DTMF decoder in mISDN is garbage |
| 16:52.49 | qakhan | [TK]D-Fender how can i check current RTP setup/ |
| 16:53.18 | [TK]D-Fender | watch the negotiation, watch when it starts coming in. |
| 16:53.43 | WIMPy | Yes, the level calculation seems to be wrong. |
| 16:54.26 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-xlheebytydxapxus) |
| 16:56.36 | anonymouz666 | sends a DTMF to coppice |
| 16:57.05 | coppice | I *will* decode it reliably |
| 16:57.33 | WIMPy | takes coppice and sticks him in to the dsp module. |
| 16:57.41 | WIMPy | Fixed. |
| 16:57.46 | anonymouz666 | coppice: what's the minimum duration required? |
| 16:58.14 | coppice | depends on the particular spec. its usually around 40ms |
| 16:58.41 | anonymouz666 | then asterisk emulates to 80ms to match the default duration? |
| 17:01.41 | WIMPy | pa: What Asterisk version are you running? |
| 17:03.01 | pa | i think 1.12 |
| 17:03.03 | pa | moment |
| 17:03.18 | pa | 1.11.7 |
| 17:03.36 | WIMPy | 11.7? |
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| 17:07.39 | pa | yes, i think so |
| 17:07.47 | pa | it's the stock version in ubuntu 14.04 |
| 17:08.02 | WIMPy | 'core show version' |
| 17:08.19 | WIMPy | But you only get some DTMF shit, no missing audio? |
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| 17:45.59 | pankid | can I install astdb2sqlite3 on an asterisk server with 1.8 so I can migrate the data to an asterisk server running 11? |
| 17:47.04 | [TK]D-Fender | pankid: * will automatically import the DB2 and convert |
| 17:47.09 | [TK]D-Fender | on start |
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| 17:48.09 | qakhan | [TK]D-Fender it takes 3 secs to setup audio i get this > 0x7f45940c1d70 -- Probation passed - setting RTP source address to 192.168.9.61:2254 |
| 17:48.17 | qakhan | after call answer |
| 17:48.33 | qakhan | but this happens sometimes |
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| 17:54.00 | Mango45 | When I use WaitForSilence, Asterisk does not send RTP, and causes the other end to disconnect the call thinking it has dropped. |
| 17:54.20 | pankid | [TK]D-Fender: thank you, so what is the best way to get my astdb over to my asterisk 11 server? |
| 17:54.20 | Mango45 | Is there any way to send empty RTP in between prompts? |
| 17:55.28 | [TK]D-Fender | pankid: Delete the AstDB on the 11 server. copy over the db2 |
| 17:55.31 | [TK]D-Fender | restart * |
| 17:56.06 | pankid | I think I already tried that, db2 is the astdb file in /var/lib/asterisk right? |
| 17:56.11 | pa | WIMPy, well i miss the audio when i hear the bip of some pseudo pressed button |
| 17:56.12 | pa | :) |
| 17:56.26 | pa | but i think the audio in the background is not missed, no |
| 17:56.33 | [TK]D-Fender | pankid: Depensd on asterisk.conf |
| 17:56.48 | pa | 11.7.0 (core version) |
| 17:59.28 | Mango45 | never mind, I had rtpkeepalive too high. |
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| 18:14.10 | Scunizi | Hi All.. when making an outbound call on a did line, can I make/force the outbound caller id my google voice number? |
| 18:14.43 | [TK]D-Fender | Scunizi: Depends on your tech & provider |
| 18:15.00 | [TK]D-Fender | Scunizi: Also, there is no such term as "DID line" |
| 18:15.44 | Scunizi | [TK]D-Fender: what "tech" are you referring to? The Asterisk software or hardware and provider it's attached to? |
| 18:17.21 | [TK]D-Fender | Scunizi: the TECHNOLOGY you are dialing out using, and the provider of the service for that technology |
| 18:18.29 | Scunizi | Essentially I'm thinking of 2 DiD's with my GV number forwarded to them (easy). As for "Tech".. would a 2 line Grandstream qualify in that category? |
| 18:18.49 | [TK]D-Fender | Scunizi: A 2-line Grandstream WHAT? |
| 18:19.04 | Scunizi | hangon and I'll give you a reference. |
| 18:19.06 | [TK]D-Fender | Scunizi: What are you calling OUT on? |
| 18:19.31 | [TK]D-Fender | Scunizi: Grandstream is a "brand", not a "technology" |
| 18:20.15 | Scunizi | ok.. voip phone.. as in http://goo.gl/lPMNxr |
| 18:20.38 | [TK]D-Fender | Scunizi: You just said "force outbound callerid". |
| 18:20.44 | robmal | Scunizi: [TK]D-Fender is asking about something related to elephants. Just don't use the 't' word. |
| 18:21.03 | [TK]D-Fender | Scunizi: How is calling a voip phone .... outbound? |
| 18:21.13 | Scunizi | [TK]D-Fender: probably because I'm under the miss-perception that you can dail out on a DiD as well as receive a call. |
| 18:22.41 | [TK]D-Fender | Scunizi: A DID is a dumb phone number. It is not a technology. It is not a service provider that delivers service on a technology. |
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| 18:22.59 | [TK]D-Fender | Scunizi: Give a better description of the entire call-chain |
| 18:25.32 | qakhan | [TK]D-Fender here is rtp debug |
| 18:25.33 | qakhan | http://pastebin.com/PCwBPTCL |
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| 18:27.11 | qakhan | it keep sending RTP packet to answered phone and after 3 sec phone responded to RTP packets |
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| 18:28.38 | Scunizi | I'm thinking of useing a couple of lines from Vitelity or DiDlogic. |
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| 18:30.29 | Scunizi | [TK]D-Fender: by DiD I guess I mean a line from the companies I mentioned above as SIP lines.. |
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| 18:31.28 | drathir | wonder if anyone sucessfully setup/runnin gtalk at latest asterisk version? |
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| 18:43.02 | [TK]D-Fender | Scunizi: That is "SIP termination". DID is an inbound number. It has nothing to do with the outbound aspect taht may also be bundled in what you are paying for. |
| 18:43.30 | [TK]D-Fender | Scunizi: And It's up to your provider to see if they allow you to set the callerID to a number you aren't paying them for or not. |
| 18:43.52 | [TK]D-Fender | Scunizi: Some might allow you to set it, but you may have to authorize that specific number with them. Call your provider |
| 18:44.49 | Scunizi | [TK]D-Fender: I'm wrapping my head around the differences.. The two companies that I mentioned I would be paying for. Buy you're right. I'll have to contact them and see. |
| 18:45.04 | Scunizi | *buy=but |
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| 18:58.29 | Scunizi | [TK]D-Fender: thanks.. I'm off to investigate some more. DiDLogic will allow me to set my own CiD for outbound calls. |
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| 19:12.36 | Katty | STRAWBERRY LEMONADE |
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| 19:13.24 | [TK]D-Fender | STRAWBERRY BERET |
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| 20:02.31 | overyander | I get the following error sometimes when faxing over t.38... google is actually not helpful at all on this one. 3RD_FRM_CHECK_ERROR |
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| 20:32.11 | marceloamorim | guys, do you already get those "codec_gsm.c:104 gsmtolin_framein: Invalid GSM data (1) " on cli? |
| 20:32.49 | marceloamorim | I'm trying to use gsm on my phone, but I don't get the audio and getting those messages instead |
| 20:33.13 | marceloamorim | I tried to use gsm on my sip client on my cellphone and works fine |
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| 22:07.03 | Shaan | hi guys im looking for a wifi voip phone that has programmable keys or 4 line buttons anyone aware of such phone? |
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| 22:47.22 | voip | wooohooo |
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| 22:56.44 | [TK]D-Fender | Shaan, Tons of phones like that from many makers |
| 22:56.51 | [TK]D-Fender | Polycom, Aastra, Digium, etc |
| 22:57.17 | netvergent | I am trying to find west coast distrubutors and or resellers of polycom and everyone seems to be mid country or eastern us. |
| 22:58.14 | netvergent | listening to mozart MOH at the moment. |
| 22:59.44 | netvergent | no point on putting me on MOH if it slips the call into vm! |
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| 23:02.51 | Shaan | [TK]D-Fender WITH MULTIPLE LINE BUTTONS OR PROGRAMMABLE KEYS? |
| 23:03.04 | Shaan | [TK]D-Fender I WOULD LIKE IT TO BE WIFI MOBILE |
| 23:03.07 | Shaan | NOT, WIFI DESKTOP |
| 23:03.13 | Shaan | sorry about cap[s |
| 23:04.01 | netvergent | picked up vvx310 for 45 dollars each. ;) |
| 23:04.17 | netvergent | two but that is just sheer luck |
| 23:04.26 | netvergent | yea no yelling |
| 23:04.49 | netvergent | Shaan, we are not old men wearing hearing aids. |
| 23:07.39 | [TK]D-Fender | Shaan, Mobile users don't ted to have massive call handling capabilities. |
| 23:07.44 | [TK]D-Fender | They are not like receptionists |
| 23:07.52 | [TK]D-Fender | You're in "unicorn" territory |
| 23:09.20 | netvergent | Hire a receptions to take the calls. |
| 23:13.05 | netvergent | have the most important calls directed to you. |
| 23:13.09 | netvergent | Sales calls mostly |
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| 23:45.50 | jeev | is a color expansion required if you use a vvx 410? |
| 23:45.55 | jeev | or can you use bw |
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