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| 12:40.32 | anexit | Question, when I call outbound, it shows 1001 as the calling number. How do I set it up so I can return calls to that phone? |
| 12:44.21 | [TK]D-Fender | Return what calls? |
| 12:44.43 | [TK]D-Fender | Calls go where you tell them to. |
| 12:45.23 | [TK]D-Fender | They present the callerid you tell them to. |
| 12:47.10 | anexit | I can make outbound calls but I don't get incoming. |
| 12:47.20 | anexit | I'm using flowroute for a service |
| 12:48.26 | [TK]D-Fender | are you using PJSIP or chan_sip? |
| 12:49.41 | anexit | chansip |
| 12:50.04 | anexit | Looks like i need to purchase a did from them. |
| 12:50.08 | [TK]D-Fender | "sip show registry" <- verify that you are registered first |
| 12:50.13 | anexit | Thought the trial would include one. |
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| 12:50.39 | [TK]D-Fender | Then enable SIP debug and either watch the registration attempts fail, or place a call in and look for the inbound attempt |
| 12:51.02 | [TK]D-Fender | Never expect to get things for free. |
| 12:51.20 | [TK]D-Fender | And if you didn't pay for a DID.. then what phone number were you even dialing? |
| 12:51.29 | [TK]D-Fender | And why would you think that it'd reach you? |
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| 12:52.07 | anexit | from my lab voip system, I can call my verizon cell phone |
| 12:52.44 | anexit | I was hoping my lab pbx had a switchboard number assigned to it but not that case |
| 12:53.21 | anexit | calling voip phone extensions work awesome. |
| 12:54.23 | pawiecki | Hello. I'm trying to configure audiocodes mp-114 to work inside a VLAN but it just doesn't work. Is there anyone experienced with audiocodes? |
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| 13:11.29 | TandyUK | hey guys, whats the minimium required bandwidth, and with what codec to reliably use VOIP? |
| 13:11.40 | TandyUK | just been shafted by BT on an install I have coming up |
| 13:11.57 | TandyUK | when we ordered the line originally, FTTC (80/20mbps) available |
| 13:12.23 | TandyUK | now line is in, numbers are being ported, phones ordered etc, and suddenly FTTC isnt available as theres no spare connections on the dslam |
| 13:12.37 | TandyUK | so best we can get is ADSL with 0.9Mbps upload |
| 13:13.10 | acidfu_ | ouch |
| 13:13.31 | [TK]D-Fender | Minimum bandwidth is the bandwidth required by your calls |
| 13:13.35 | [TK]D-Fender | And that is based on the codec |
| 13:13.47 | [TK]D-Fender | And their usage is very easily googleable |
| 13:14.18 | TandyUK | i know what google says, i want answers from someone with actual experience of using them, what is the lowest 'acceptable' codec for call quality |
| 13:14.41 | TandyUK | ive played with the lowest bw codecs in the past, anbd call quality is not what id call 'acceptable' |
| 13:15.10 | [TK]D-Fender | G.729 is as low as you should go if you care about quality |
| 13:15.15 | file | acceptable is subjective |
| 13:15.33 | [TK]D-Fender | GSM is tied roughly, but less supported by ITSP's |
| 13:16.07 | TandyUK | file: i know that, but im used to being spoilt and having ~256kbps per call available |
| 13:16.07 | [TK]D-Fender | tied for quality that is.. takes a bit more BW |
| 13:16.40 | [TK]D-Fender | And if you are transcoding from something that's already below PSTN-grade where you are to another then you're doubling-down on the quality lost |
| 13:16.40 | TandyUK | with yealink phones, its crystal clear - i know were not going to get anywhere nearthat |
| 13:17.03 | [TK]D-Fender | "~256kbps per call" <- NO telephony codec uses this much |
| 13:17.14 | TandyUK | yeah i know ;) |
| 13:17.15 | [TK]D-Fender | That number is absurd |
| 13:17.22 | TandyUK | hence the never havign a shortage of bandwidth lol |
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| 13:17.49 | TandyUK | but to suddenly have to provision 7 phones on a <1mbps line worries me |
| 13:17.59 | [TK]D-Fender | G.711 as 30ms packetization over RTP ~ 85kbps |
| 13:18.21 | TandyUK | oh thats not so bad then |
| 13:18.34 | [TK]D-Fender | and that's at what is typically "full quality" |
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| 13:25.53 | TandyUK | ok, well im suddenly less worried, but will be getting another line installed so voip has its own dedicated line |
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| 15:16.42 | catphish | i notice that Dial() had a limit on the length of its parameters which was fixed in asterisk 1.4, however i'm seeing the same limit in Page(), would anyone be able to point me to the patch that fixed this in dial in the hope i can apply it to page? |
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| 15:21.51 | [TK]D-Fender | Limit is against all apps, not just dial |
| 15:25.06 | catphish | [TK]D-Fender: oh really? maybe i'm mistaken then, i found docs saying the limit was increased in dial, but i'm seeing my Page() params truncated somewhere |
| 15:25.18 | catphish | and i'm sure this problem doesn't happen to me on Dial() |
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| 15:28.29 | WIMPy | \o/ for multicast RTP |
| 15:30.41 | catphish | only some of the devices my my Page() call are being dialled tough :( |
| 15:37.36 | catphish | i might be mistaken, the limit might be on Dial too |
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| 16:15.29 | mafairnet | hi, does anyone have experienced this message "ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10" |
| 16:15.59 | mafairnet | I'm using Asterisk 13 + Chan_SIP + FreePBX + SIPML5 |
| 16:17.34 | mafairnet | anyone there? |
| 16:25.17 | keithf | you using a sonicwall ? |
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| 16:52.18 | mafairnet | @keithf no, i'm not using sonicwall.. |
| 16:52.44 | keithf | we have sonicwalls here and had a few RTP issues at first also and we use freepbx as well |
| 16:56.07 | mafairnet | on audio with webrtc i have no proble, the detail is with video |
| 16:56.34 | mafairnet | Im also usien Chrome |
| 16:57.01 | keithf | what about firefox ? same issue ? |
| 16:58.22 | keithf | have you checked this out also? |
| 16:58.25 | keithf | http://forums.asterisk.org/viewtopic.php?f=1&t=93987 |
| 16:59.38 | mafairnet | I've tested with firefox but the authntification is denied. |
| 16:59.48 | mafairnet | Let me check the link |
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| 17:37.25 | cusco | hi folks |
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| 19:46.10 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.8.0 (2016/03/29), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 20:52.06 | mafairnet | @keithf I've read the article but as far I have search directrtpsetup=yes doesn't work with video and the problem is nobody explains the message " SRTP unprotect failed with: authentication failure" |
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| 21:52.38 | Toerkeium | hello guys |
| 21:53.19 | Toerkeium | has anyone succeded on filtering/blocking attemps to register to asterisk from firewall? I'm not able to do it |
| 22:03.04 | [TK]D-Fender | iptables works... |
| 22:20.29 | newtonr | Toerkeium, what are you specifically doing on the firewall that isn't working? |
| 22:23.16 | Toerkeium | newtonr: I'm using mikrotik, and as UDP is a connectionless protocol, I cannot block it |
| 22:31.37 | newtonr | You should be limiting traffic to the source addresses that you expect traffic from |
| 22:33.18 | newtonr | and I'm not familiar with mikrotik |
| 22:33.29 | newtonr | and I'm leaving very soon so I won't be able to help much. Good luck! |
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| 22:41.48 | [TK]D-Fender | <Toerkeium> newtonr: I'm using mikrotik, and as UDP is a connectionless protocol, I cannot block it <- Nonsense. It's dumb packets and that is a basic Linux-based box. block the IP. YES. YOU. CAN |
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| 23:26.58 | amonk | i'm still using an old mikrotik rb-230. it doesn't have their mikrotik (linux) os on it though. it has another linux i installed. but, sure, of course one can block udp packets |
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