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| 00:59.58 | dorphalsig | Hello, I have two * boxes connected via IAX, however I would like to route the incoming calls to multiple contexts |
| 01:00.32 | dorphalsig | Must I define a connection per context, or can I just specify which context I want when DIALing |
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| 01:00.34 | dorphalsig | ? |
| 01:01.33 | WIMPy | If you don't specify a context in the peer, you can specify iy in the Dial (if my memory serves me right). |
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| 02:42.45 | dj_hamsta | whats the diff between a sip and a did? |
| 02:42.56 | ChannelZ | SIP is a protocol, a DID is a thing |
| 02:43.24 | ChannelZ | DID is a phone number |
| 02:43.25 | dj_hamsta | DID is a physical extention on a phone? |
| 02:43.51 | ChannelZ | Not really. It's just a public phone number. |
| 02:43.55 | dj_hamsta | ahhh |
| 02:44.05 | dj_hamsta | and sip is how the did gets to one place to another? |
| 02:44.24 | ChannelZ | That can get routed to you by any number of different means (PRI, POTS, terminated at an ITSP and sent to you as a SIP call...) |
| 02:45.28 | ChannelZ | SIP is a protocol, much like HTTP, used for setting up/tearing down a "call" |
| 02:45.35 | ChannelZ | (among other things) |
| 02:45.41 | dj_hamsta | ahhhhhhhhhhh |
| 02:46.06 | dj_hamsta | i thank you sir. |
| 02:46.25 | dj_hamsta | i kept thinking a did was an extention |
| 02:46.25 | ChannelZ | What that call is can be any number of things. SIP-to-SIP even, not involving the telephone network at all. |
| 02:47.08 | ChannelZ | Nope, really just a phone number. |
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| 02:56.11 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 02:57.11 | UnixDev | hi everyone.. I have a question about dealing with realtime users/peers… how can I have asterisk keep the state of the peers in memory when a reload is executed? for some reason, when a reload is done, all registered peers show as unavailable |
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| 03:33.06 | ChannelZ | dunno but I'd suggest only reloading what is necessary |
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| 03:50.50 | jero | hi |
| 03:54.08 | UnixDev | ChannelZ: I have an include that includes a directory, what if I needed only to reload 1 file in that directory? |
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| 06:20.54 | din3sh | anybody deployed a polycom vvx 1500 with asterisk ? |
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| 06:54.56 | jero | yes |
| 07:07.40 | din3sh | works fine with video? |
| 07:16.40 | vader-- | do any of you guys run a codec other than 711 on your phones? |
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| 16:50.55 | catphish_ | what is he best free tts for asterisk? |
| 16:56.18 | catphish_ | festival seems to be the standard solution, will look at that |
| 17:11.11 | [TK]D-Fender | Cepstral is fairly inexpensive and much better |
| 17:12.21 | file | falls over |
| 17:15.29 | [TK]D-Fender | runs in circles |
| 17:18.46 | file | I'm reading a diff of code I wrote long ago and giggled when I saw "threeway" |
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| 17:30.02 | catphish_ | i need something free as i'll be distributing it |
| 17:31.02 | catphish_ | festival seems ok, though not very good at punctuation |
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| 18:17.27 | mcf3782 | I've been reading the source code for app_meetme.c for version 1.8.12.0. It appears that if a MeetMe conference is a dynamic one (created with either the 'd' or 'D' option); that there is only a way to specify a single PIN — while non-dynamic conferences can have both an Admin PIN and a User PIN. Is there a technical reason that dynamic conferences can't have both types of PIN? |
| 18:19.52 | mcf3782 | By technical reason, I mean as opposed to a design philosophy reason. |
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| 19:47.29 | clintlawwco | Need a bit of advice, if anyone's game: |
| 19:48.24 | clintlawwco | How do I ask asterisk 1.8 to stop regenerating touchtones on a Dahdi to Dahdi bridged call? |
| 19:48.38 | clintlawwco | The Google has been wholly unhelpful. |
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| 19:55.07 | clintlawwco | (also, this is with features enabled - disabling them "fixes" the issue, as asterisk doesn't molest the tones) |
| 19:55.25 | clintlawwco | I need to disable on a per call or per port basis. |
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| 20:11.43 | phaidros | does anyone know if the linksys sipura 1001 support pulse dial? |
| 20:47.15 | clintlawwco | phaidros - The linksys does not. The only low cost ATA I've heard does is the Grandstream ATA502. |
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| 21:10.39 | F|shie | hello all, need to know how to compile asterisk for external header and source files i have included in a custom module |
| 21:23.55 | vader-- | do any of you guys run a codec other than 711 on your phones? |
| 21:32.53 | [TK]D-Fender | probably the ton of people using G.729 |
| 21:35.58 | F|shie | [TK]D-Fender: need to know how to compile asterisk for external header and source files i have included in a custom module, any idea? |
| 21:39.01 | [TK]D-Fender | no idea |
| 21:44.25 | vader-- | any benifits to running G.729 over g.711?? |
| 21:44.41 | vader-- | i have a user complaining of a "far away" sound from the callers |
| 21:44.56 | vader-- | they are describing the calls as sounding "far away" |
| 21:45.04 | vader-- | when i make calls on it, it sounds crystal clear |
| 21:49.17 | [TK]D-Fender | th codec has nothign to do with sounding "far away" |
| 21:49.45 | [TK]D-Fender | And what codec a phone is using isn't describing the big picture" |
| 22:03.16 | [TK]D-Fender | There is a weak A/D conversion somewhere in the chain. Maybe multiple |
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| 22:39.53 | gusto | hm |
| 22:39.55 | ChannelZ | Is 'far away' a description of sound quality, or of time? |
| 22:40.00 | ChannelZ | (distance) |
| 22:40.01 | gusto | hey, how's going? |
| 22:40.35 | ChannelZ | Great! I just pooped. |
| 22:40.36 | gusto | far away would be latence, right? |
| 22:40.50 | ChannelZ | Dunno. |
| 22:40.58 | ChannelZ | <vader--> i have a user complaining of a "far away" sound from the callers |
| 22:41.17 | gusto | maybe he has an echo |
| 22:42.04 | leifmadsen | or low volume |
| 22:42.17 | ChannelZ | Thus my clarification request |
| 22:42.21 | leifmadsen | indeed |
| 22:42.32 | leifmadsen | "more information required" is appropriate here |
| 22:42.37 | leifmadsen | return after you have clarified |
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| 22:51.16 | batfastad | Hi everyone. Is there a way to find out the time when an extension last registered? A remote worker's phone hasn't been working for over a week, he was away and something's screwed his internet connection |
| 22:52.12 | batfastad | Will I just have to trawl a log looking for that extension? Or is there an easier way? |
| 22:52.43 | ChannelZ | Trawl away |
| 22:53.19 | ChannelZ | But you might not even be logging such things |
| 22:53.36 | batfastad | Urrrg yeah. I was hoping asterisk might keep a timestamp somewhere. What log should I start with? |
| 22:53.54 | ChannelZ | /var/log/asterisk/messages or such if you have it |
| 22:54.26 | ChannelZ | Logging is controlled in /etc/asterisk/logger.conf |
| 22:54.57 | batfastad | No messages. I have log files called "full" |
| 22:55.13 | ChannelZ | That'll work |
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| 23:04.19 | batfastad | Sorted... tac full | grep -m 1 "chan_sip.c: Peer '32' is now UNREACHABLE" |
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