| 00:20.51 | *** join/#asterisk matrix1233 (~matrix123@197.0.213.162) |
| 01:10.08 | raspberrypifan | come on someone sell me a gsm gateway and get some commision |
| 01:41.26 | *** join/#asterisk mcf3782 (~Adium@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 01:42.03 | drmessano | I think thats the point |
| 01:42.08 | drmessano | No one wants to recommend one |
| 01:44.05 | raspberrypifan | why is it a dangerous thing to do? |
| 01:48.09 | [TK]D-Fender | I'm sure lots of STORES will sell you one. |
| 01:48.14 | [TK]D-Fender | Have you gone looking yet? |
| 01:49.08 | raspberrypifan | yes |
| 01:49.13 | raspberrypifan | the point is there are soooo many |
| 01:49.21 | [TK]D-Fender | So what's the problem? Hit the magic "buy" button |
| 01:49.32 | [TK]D-Fender | DO IT |
| 01:49.41 | [TK]D-Fender | #thevoicesmademedoit |
| 01:50.09 | mcf3782 | goes to buy one of whatever it is |
| 01:51.18 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 01:53.32 | raspberrypifan | but how do i know |
| 01:53.35 | raspberrypifan | what is good and what is not |
| 01:54.23 | [TK]D-Fender | Considered any other brands we throw around here with any regularity? |
| 01:54.26 | [TK]D-Fender | Digium? |
| 01:54.28 | [TK]D-Fender | Sangoma? |
| 01:54.30 | [TK]D-Fender | Media5? |
| 01:54.35 | [TK]D-Fender | AudioCodes? |
| 01:54.51 | [TK]D-Fender | Googled any of the one's up to see what other's experience has been? |
| 01:55.19 | [TK]D-Fender | Have you done anything that we'd qualify as "actually researching"? |
| 01:55.24 | [TK]D-Fender | Think hard on that one.... |
| 01:56.32 | [TK]D-Fender | heads out for a while... |
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| 02:02.33 | mcf3782 | Does anyone in here have Google Voice working on a 11.23.0 box? I can't get past an error about "ICE support not available" |
| 02:08.01 | raspberrypifan | none of those companies have standalone gsm gateways |
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| 05:57.24 | Vamp898 | Is there an easy way to get the Asterisk config without secrets and so on? |
| 06:01.23 | Vamp898 | We have one device which registers fine and can execute calls, but doesnt receive calls (it always shows as UNREACHABLE). If we set qualify=no and keeplive=yes the device still doesn't receive calls :/ i want to post configs but its tiresome to clean alle the configs and logs every time |
| 06:07.06 | bhans | What is the best cloud server host that can be hosted with Asterisk? |
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| 07:56.58 | lorsungcu | bhans: cat /etc/asterisk/sip*.conf | awk -vRS= -vFS="\n" '/<extension or blank for anything>/' | sed -re 's/(secret|callerid)=[^=]*$/\1=/' |
| 07:59.30 | wyoung | bhans: define best |
| 08:00.06 | wyoung | Do you want a person allocated to you to hold your hand and configure it all for you? or are you broke and can't afford such a serviec? |
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| 08:00.41 | lorsungcu | er, sorry, that wass for Vamp898 ^ |
| 08:01.03 | lorsungcu | although for real quit being lazy |
| 08:04.15 | bhans | I am asking you guys so that I could have an idea in which I could go to.. I am referring to AWS, Linode, SalesForce, etc... Which is good to have my server hosted to. |
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| 08:16.12 | wyoung | bhans: SalesForce does virtualisation? |
| 08:16.18 | wyoung | bhans: well it depends on your country |
| 08:16.30 | wyoung | bhans: in AU AWS is better, in US linode is probably better |
| 08:17.13 | pawiecki | Hi! I have a problem with randomly 'Lagging' peers. I've changed network switch, looked at cables, but it seems that 'hardware' side is ok. As a result of this problem i get a lot of retransmission errors, and today asterisk just stopped making calls. It's a bit critical env. so i tried to restart it, and it helped. But now I want to investigate and fix the issue for good. Have you encounter similar |
| 08:17.19 | pawiecki | problems? I'm looking for some hints as to why it happened. |
| 08:18.05 | wyoung | pawiecki: where are these peers located? |
| 08:18.15 | wyoung | in a LAN or VPN/ Internets /? |
| 08:18.39 | wyoung | also what are these peers? softphones? SIP phones? can on a string? |
| 08:20.31 | pawiecki | wyoung: it's all inside one building, simple LAN network (* server, switch, PoE-switch, phones). Peers are at Venus SIP-Analog GW, Gigaset phones and our 'proprietary' POS linux PC, that acts like a touchscreen phone basically. |
| 08:20.56 | wyoung | hmmmmm |
| 08:21.36 | wyoung | and what happens when you get "lag"? the sound gets choppy / cuts in and out? volume changes? |
| 08:21.44 | wyoung | pawiecki: are all peers using the same codec? |
| 08:22.00 | wyoung | I have had issues with transcoding on setups before |
| 08:22.20 | wyoung | setups / deployments |
| 08:22.32 | pawiecki | i've had this "asterisk not making calls" problem like a week ago, and restarting * helped, but i couldn't find any general problems (aside lagging/unreachable peers), and now it's happening again. |
| 08:32.51 | pawiecki | when a peer gets lagged, it's unreachable for a few seconds. Codecs *should* be all set to alaw and 722, but i haven't configured Venus GW's - will check that out and make sure. What issues did you have? Was it something major? |
| 08:33.10 | wyoung | hmmmmm |
| 08:33.22 | wyoung | can you keep it the same |
| 08:33.56 | pawiecki | wyoung: yeah, i think it's a good idea to at least test it that way for some time. |
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| 08:35.31 | wyoung | I had issues with voice fading in and out plus background noise getting louder and drowning out voice and when I changed all SIP phones to alaw instead of GSM it fixed the issue |
| 08:35.47 | wyoung | (well, my country uses alaw nativally) |
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| 09:10.56 | Vamp898 | lorsungcu: thanks, thought there is something like an asterisk own feature, but in that case it have to be done that way |
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| 11:41.47 | Vamp898 | Our server have the correct time, but still the Master.csv in /var/log/asterisk/cdr-csv shows wrong timezone (GMT instead of CEST) |
| 11:41.58 | Vamp898 | can this be set somewhere? |
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| 13:39.22 | sikun | quick simple question, if the SIP trunk allows for custom/altered DID on outbound calls, can I setup an identical SIP trunk to be used only on outbound routes with the specified DID? |
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| 13:40.53 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:45.12 | sikun | mainly.. I have a trunk that I want all outbound calls to have the same DID, with the provider limiting 1 outgoing call per trunk I want to utilize the other SIP trunks to enable multiple outbound calls but configure the trunks to use the DID of the number I want |
| 13:59.29 | Samot | sikun: You can see the Caller ID based on the routes you match or on the trunks themselves. |
| 14:01.56 | sikun | yeah, I guess maybe I wasn't as clear as I thought. so I have one sip trunk inbound/outbound phone number 414-555-5555 and I can only make one outbound call at a time.. but I have several other trunks that rarely have outbound calls but all have different numbers, can I utilize those trunks and change the outbound did to match 414-555-5555 |
| 14:02.54 | [TK]D-Fender | Set your CID in your routes, not your trunks. |
| 14:03.20 | [TK]D-Fender | And you should move over tto #freepbx for support on the GUI as that is not done here... |
| 14:03.24 | sikun | in my outbound route I don't have the option |
| 14:03.34 | sikun | ah, kk, thx for the channel |
| 14:04.15 | [TK]D-Fender | Outbound route has a very clear filed for the CID to use |
| 14:04.19 | [TK]D-Fender | field* |
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| 14:05.48 | sikun | I'm using TrixBox and I can only select trunk sequence and set the dialing pattern |
| 14:05.51 | sikun | or set the route password |
| 14:06.26 | Samot | Sorry, no real help from Trixbox here. |
| 14:08.52 | [TK]D-Fender | Trixbox is ANCIENT garbage.... |
| 14:09.09 | [TK]D-Fender | dead for 5 years now... |
| 14:10.17 | sikun | I know, I'm trying to bandage it while I get a fresh copy of AsterisksNOW configured on a replacement server |
| 14:11.28 | [TK]D-Fender | Even then Outbound Routes should have a CID field |
| 14:12.09 | sikun | the only place where you can enter a cid is under the sip trunk settings |
| 14:13.04 | [TK]D-Fender | Show us the OR screens : picpaste.com |
| 14:18.39 | sikun | http://picpaste.com/2016_08_16_09_13_09_trixbox_Admin_Mode-u4wd8gfe.png <-- here is the outbound route |
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| 14:20.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:24.59 | [TK]D-Fender | Yup... that sucks |
| 14:25.02 | [TK]D-Fender | 2008 ... |
| 14:25.11 | [TK]D-Fender | *shudder* |
| 14:25.51 | sikun | lol, ok, I just need to get this new server running quicker |
| 14:25.52 | sikun | ugh |
| 14:26.09 | sikun | moving 9 companies worth of trunks/extensions |
| 14:26.11 | sikun | woo |
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| 14:28.33 | sikun | oh yeah, can't forget about all the greetings I get to recreate, ha. |
| 14:30.04 | [TK]D-Fender | Why? |
| 14:30.11 | [TK]D-Fender | You can keep all your audio files... |
| 14:30.19 | sikun | yeah, just thought of that |
| 14:30.29 | sikun | I'll just sftp them off of the existing server |
| 14:30.55 | [TK]D-Fender | Shouldn't have "just" thought about it... just like VM these are the easy bits to bring over. the PITA stuff are the structures you need to recreate, TC's, TG's, IVR's, etc... |
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| 15:05.35 | *** join/#asterisk traxx (~traxx@gate.media-ebene.de) |
| 15:05.43 | traxx | hi |
| 15:06.19 | traxx | i have a question if i go for new installation of asterisk 13 is odb still the way to go for realtime ? or should i user res_mysql ? |
| 15:06.26 | traxx | odbc |
| 15:09.00 | traxx | everything i found about that is really old and i dont know if its still up to date |
| 15:12.36 | *** join/#asterisk MRH2 (~Thunderbi@233.47.187.81.in-addr.arpa) |
| 15:13.35 | MRH2 | hi anyone know a SIP phone handset that also acts as a USB audio device (speakers mic) |
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| 15:18.39 | ChkDigit | MRH2: A Plantronics Savi 730 is a headset that can connect to a SIP phone, and be a USB/BT audio device. |
| 15:19.13 | ChkDigit | But that is not precisely what you're looking for... |
| 15:19.52 | MRH2 | looking for an actual phone handset - use the existing speakerphone, headset or handset as audio devices |
| 15:20.33 | WIMPy | There are USB Handsets, but they are not phones themselves. |
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| 15:21.59 | MRH2 | there are conference phones that do this switch from USB for Skype youtube etc or normal SIP over ethernet - i'm looking for a desktop handset |
| 15:22.40 | MRH2 | yeah guess not :( |
| 15:22.53 | [TK]D-Fender | There is virtually no market for such a device.... a challenging find at best.... |
| 15:23.27 | [TK]D-Fender | You'll get straight VoIP phone, or straight USB>ATA ... but the likelihood of a 2-in-1 .... |
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| 15:24.55 | MRH2 | it's a fantastic market - use office sip normally / switch phone to usb for a webex or listen to video over the speaker or headset for training or other online meeting types |
| 15:25.33 | WIMPy | Just use a softphone with it. |
| 15:25.34 | MRH2 | why have a sip handset, PC speakers, PC USB headset |
| 15:27.22 | MRH2 | lots of issues can go wrong with a desktop vs a dedicated device + it is more effective for business users to have an actual phone |
| 15:28.02 | WIMPy | Does a SIP appliance count as an "actual phone" in that context? |
| 15:28.25 | MRH2 | as in VM? |
| 15:28.36 | MRH2 | or docker |
| 15:28.45 | WIMPy | Huh? |
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| 15:29.16 | MRH2 | oh dedicated machine is the same as a hard phone ain;t it |
| 15:29.48 | WIMPy | I guess, it depends. |
| 15:29.50 | MRH2 | would u run ur dbserver on it as well - prob not |
| 15:30.58 | MRH2 | softphones are fine for certain usage cases |
| 15:34.21 | MRH2 | but if people are taking lots of calls aside from usual work flow they are distraction not to mention they don;t want that critical call when the PC is doing something CPU intensive. |
| 15:36.10 | MRH2 | closest match I have found is a switch that covers the headset only |
| 15:36.30 | MRH2 | headset to phone or USB |
| 15:37.24 | MRH2 | anyway thanks..i'll keep searching |
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| 19:47.08 | deweydb | sorry for the dumb question, but, what is the purpose of CNAME if i can spoof CID and caller name? |
| 19:50.44 | deweydb | er i meant CNAM not CNAME |
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| 21:02.05 | edong23 | Hi. I have googled.. and cant find an answer that matches what is happening on my setup. I get intermittent audio drops. So i started up some rtp debug on the pbx and on the gateway and the pbx seems to stop sending rtp randomly. nothing in sip debug shows why. it is all Sent/Got/Sent/Got then all the sudden it is just a stream of Got without a Sent. The other server is a stream of Sent/Got/Sent/Got/Sent/Got thenall the sudden, and stream |
| 21:02.05 | edong23 | of Sent with no Got. |
| 21:02.13 | edong23 | any idea why this would be? |
| 21:02.33 | edong23 | both servers are on the same network segment. One is a t1 interface to the switch, the other is a sip only pbx. |
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| 21:21.08 | traxx | hey guys can somebody help me with odbc and ubuntu ? |
| 21:27.09 | robmal | Yes. |
| 21:27.35 | traxx | cant install libmyodbc apt-get dont find it |
| 21:28.14 | traxx | freshly installed ubuntu 16 LTS |
| 21:28.16 | robmal | Impossible. |
| 21:31.02 | traxx | but it tells me that there isnt a installationcandidate fpr libmyodbc |
| 21:31.04 | traxx | for |
| 21:33.22 | traxx | http://www.ubuntuupdates.org/package/core/xenial/universe/base/libmyodbc |
| 21:35.16 | traxx | The package was removed because it's not compatible with MySQL 5.7 which ships with 16.04. fuck, any work arounds |
| 21:35.17 | traxx | ? |
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| 21:40.52 | robmal | Backports? |
| 21:42.49 | traxx | do i need odbc or can i use a native mysql driver ? |
| 21:46.16 | robmal | Depends what you need. |
| 21:48.17 | traxx | user and queue config in the database |
| 21:55.49 | traxx | robmal, is that possible ? or only with odbc ? |
| 21:56.11 | robmal | You need some sort of Sorcery(tm) |
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| 22:18.23 | drmessano | Install the ODBC package from MariaDB |
| 22:18.28 | drmessano | Works wonderfully |
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