| 00:06.56 | josefig | i'm trying to setup an outgoing call rule : http://pastie.org/private/ailid4pi7xkeylwejsyuhq but no luck, someone can help me please? |
| 00:10.54 | X-Rob | So, file. This -is- your fault. You don't have an asterisk-sounds rpm for .sln16. |
| 00:11.10 | X-Rob | How am I meant to have wideband audio, if I've got crappity ulaw soundfiles? |
| 00:11.49 | X-Rob | ps: crappity. It's a word. |
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| 03:14.58 | txt23 | Hey guys. I have a very small website and when my customers call I want them to press different numbers for different things. And also want to record conversations, whats my best option for it? |
| 03:15.36 | litn | asterisk |
| 03:16.04 | txt23 | Right but how much of an investment am I looking here? |
| 03:16.25 | litn | money wise? |
| 03:30.33 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 03:30.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 06:18.37 | txt23 | I just installed asterisk on a VPS. Does it have a web UI to access and configure it from? |
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| 06:29.13 | _omer | hello |
| 06:29.51 | _omer | Is there a way to turn off Queue "announce-position" for a specific caller through dialplan? |
| 06:30.49 | _omer | Is there a way to turn off Queue "announce-position" for a specific caller through dialplan? .... Turning off "announce-position" in queue settings, turns it off for all the callers. any suggestions? |
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| 09:39.35 | Blashyrkh | i have a problem with an old asterisk version that used to be connected to PSTN lines, but was disconnected now and handles only sip calls |
| 09:39.44 | Blashyrkh | but when using playback it sounds really awful |
| 09:39.58 | Blashyrkh | i guess its a timing related issue, but iam completely lost here |
| 09:40.48 | Blashyrkh | it still uses zaptel drivers |
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| 10:26.11 | alexw | What is the cleanest way to install asterisk? |
| 10:26.20 | alexw | Does asterisk have it's own config console |
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| 10:56.34 | Dpunkt | if i divide a call using dial(x,y,G(z)), how can i reconnect this call later in the dialplan |
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| 14:17.11 | WIMPy | Cool questions so far today. |
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| 14:46.57 | *** join/#asterisk elenril (~elenril@lain.khirnov.net) |
| 14:47.36 | elenril | hi |
| 14:48.07 | elenril | is there any documentation on how codec negotiation in asterisk (and sip in general) works |
| 14:50.05 | mjordan | elenril: There is a plethora of information on how codec negotiation is achieved using SDP offer/answer inside SIP requests/responses. |
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| 14:51.20 | elenril | anything about how it works in asterisk specifically? |
| 14:51.31 | elenril | google is rather vague on the subject |
| 14:52.01 | mjordan | elenril: as for Asterisk, you're most likely best served by reading the book. |
| 14:52.10 | mjordan | elenril: a specific question would probably be appropriate here |
| 14:52.31 | mjordan | elenril: Depending on what you're trying to find out, the answer may be very straight forward, or very complex. |
| 14:53.13 | elenril | i want to understand how it works so i can configure it to use other codecs than *law pcm automagically |
| 14:53.28 | mjordan | Define "automagically" |
| 14:53.46 | mjordan | Do you want Asterisk to Offer codecs other than ulaw/alaw? |
| 14:54.03 | mjordan | The chapter on SIP configuration in A:TDG describes the disallow/allow options |
| 14:54.03 | mjordan | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html#DeviceConfig_id291081 |
| 14:54.55 | elenril | the problem is that i don't have pretty much any idea how are codecs negotiated; so i cannot really ask any specific questions |
| 14:55.10 | mjordan | Do you understand SDP offer/answer? |
| 14:55.11 | elenril | i suppose each client connected to asterisk has its own set of supported codecs |
| 14:55.32 | elenril | and "automagically" for me means that the best codec will be selected for each connection |
| 14:55.53 | elenril | no, not really |
| 14:56.05 | mjordan | Go read up on SDP Offer/Answer. Google will return many hits for you there. |
| 14:56.35 | elenril | like https://www.ietf.org/rfc/rfc3264.txt? |
| 14:56.42 | mjordan | That would be the definitive answer, yes |
| 14:56.51 | mjordan | The first codec listed in an SDP offer is the preferred codec. |
| 14:57.00 | mjordan | What a device offers is up to the specific devices. |
| 14:57.18 | mjordan | When Asterisk calls a device, the order of your codecs in your allow settings determines the preference order |
| 14:57.48 | mjordan | When negotiation happens between an offer/answer, Asterisk picks the best matching codec between what we offered/they offered and what we believe their capabilities to be (based on what you configure in sip.conf/pjsip.conf/iax.conf/etc.) |
| 14:58.18 | mjordan | So, if a UA offers g722,ulaw,alaw but you configure the UA with allow => ulaw,alaw - we'll end up with ulaw |
| 14:58.23 | elenril | i have to configure each client's capabilities in asterisk itself? |
| 14:58.33 | elenril | the client does say which codecs it can receive? |
| 14:58.36 | elenril | *does not |
| 14:58.46 | mjordan | The client says which codecs it is willing to offer |
| 14:58.55 | mjordan | some of which Asterisk may or may not support. |
| 14:59.00 | mjordan | Some you may not want Asterisk to use. |
| 14:59.16 | mjordan | What's more, when we call a UA, we have to know what to offer them |
| 14:59.26 | mjordan | And that is determined on a UA by UA basis |
| 14:59.58 | mjordan | A trunk may only support a limited subset of codecs. A more modern SIP soft phone may support celt, speex, silk, and who knows what else |
| 15:00.05 | mjordan | So yes, you have to configure Asterisk |
| 15:02.27 | *** join/#asterisk drale (~drale@unaffiliated/drale) |
| 15:03.40 | elenril | i mean can i set asterisk to "use the best of those codecs that the client supports" |
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| 15:03.58 | elenril | so e.g. i enable opus globally, but it will get used only with clients that actually support it |
| 15:03.59 | mjordan | no |
| 15:04.09 | mjordan | and doing so is not recommended. |
| 15:04.14 | elenril | why not |
| 15:04.31 | mjordan | You should know what is connecting to you, and what you want them to use. There's ways to use templates to build out configuration to use for all your endpoints |
| 15:04.41 | mjordan | And if you want to specify opus as the primary codec for all your UAs, that's fine |
| 15:04.49 | mjordan | If a UA doesn't offer it, Asterisk won't use it. |
| 15:05.07 | mjordan | Again, reading the book would help here, particularly the sip configuration chapter. |
| 15:05.09 | elenril | well that's pretty much what i was asking |
| 15:08.05 | drale | Sometimes the asterisk service just stops. I simply run cmd 'asterisk' and everything is ok again for weeks/months. Im not sure what is going on. |
| 15:08.51 | drale | Probably has something to do with this being the third server of same spec that the HDD has put in as the other server PSU failed. But never seen this symptom before |
| 15:09.43 | elenril | ok, looking at the book now |
| 15:09.46 | elenril | thanks for your help |
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| 15:27.40 | drale | i might try running safe_asterisk |
| 15:30.06 | [TK]D-Fender | [11:08]draleSometimes the asterisk service just stops. I simply run cmd 'asterisk' and everything is ok again for weeks/months. Im not sure what is going on. <- never run it direct it should be run as a daemon using a proper startup script that sets the proper user/group, etc |
| 15:30.54 | drale | ok |
| 15:31.39 | drale | like inittab used to start safe_asterisk during power up? |
| 15:31.52 | WIMPy | Err. For me Asterisk is doing that itsef just fine. |
| 15:34.17 | [TK]D-Fender | yes |
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| 15:44.11 | WIMPy | Can someone recommend phones that support transfer on onhook and (can) have at least 20 BLF buttons? |
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| 16:12.19 | drale | we have aastra 57i with M675i (560M) sidecar. i can't say on preferences because its always been here and my only other experience is a cisco system |
| 16:17.05 | drale | configuring aastra with asterisk is not difficult either. plaintext configs based on MAC filenames and a common template also. |
| 16:19.32 | drale | dhcp gives the aastra phone a boot server (asterisk) and the phone pulls the config |
| 16:21.00 | drale | it was funny when i took a phone from another location that ran on a trunked asterisk server, the old configs for the phone when it used to be in this building got picked up |
| 16:21.34 | drale | and it reconfigured itself and started ringing as part of a group of phones that ring from yet another building's number |
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| 16:34.28 | WIMPy | Would be nice if any phone would allow to push new key descriptions. |
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| 19:55.04 | gmalsack | hello all. can someone remind me how to determine which call leg disconnected the call? |
| 20:01.21 | drale | i havent done it before. have you been able to do it before? |
| 20:02.03 | drale | im googling, stuff about HANGUPCAUSE and SIP_CAUSE |
| 20:05.15 | drale | http://forums.digium.com/viewtopic.php?t=71222 |
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| 20:56.41 | [sr] | hi WIMPy |
| 20:58.33 | WIMPy | Hi [sr] |
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| 21:49.03 | gmalsack | drale: Thanks! works good... |
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| 21:53.03 | dralezero | nice |
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