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02:56.59 | rvhi | does anyone use polycom vvx 1500 with asterisk 1.4? |
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05:07.40 | JerJer | someone is having a bad night…. amazon cloud has a wide spread outage http://www.engadget.com/2012/06/30/netflix-is-down/ |
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07:42.33 | jpsharp | Huh, I didn't think JerJer existed any more. |
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08:06.15 | Psychon4ut | Hi, I am searching a good Howto Configuring my Asterisk Server on debian to work with ekiga. |
08:10.12 | ChannelZ | ~thebook |
08:10.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
08:10.48 | Psychon4ut | nice thanks I will take a look |
08:12.43 | beardy | Psychon4ut: Self-advertising is bad, but some people find this useful. http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx read the book though. |
08:13.21 | Psychon4ut | I don´t have problem with that :-) |
08:15.15 | beardy | Psychon4ut: Consider ekiga being a provider in that case. When I had it set up to call out via ekiga, and another line, I used a one digit prefix to route through ekiga. Worked nicely, when ekiga itself was working at all that is. |
08:15.35 | Psychon4ut | Oh now ekiga is SIP Client for PC |
08:15.45 | Psychon4ut | for PC UNIX, MAX and Windows |
08:16.05 | Psychon4ut | ekiga.org |
08:16.19 | beardy | Yes. |
08:16.38 | beardy | So you just want to use the ekiga client as a softphone, and connect to your Asterisk? |
08:16.50 | Psychon4ut | correct |
08:17.14 | Psychon4ut | I got an root server and want to provide it for me an my fiends/family/consumers |
08:21.23 | beardy | You just create the necessary peers/friends and extensions and configure ekiga to connect. http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client search the web. ;) |
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11:00.47 | *** join/#asterisk Steve31 (ce75f082@gateway/web/freenode/ip.206.117.240.130) |
11:00.53 | Steve31 | Hello. |
11:02.05 | Steve31 | Does anyone know how much it costs Julie's (US) TTS voice from here ? http://www.asteriskexchange.com/listings/1001 & www.speechtechnologygroup.com . I can't see any prices or a checkout page on that website. |
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11:05.40 | Steve31 | Nobody alive? |
11:07.03 | Steve31 | Hello fishie. Do you know maybe...another asterisk channels? |
11:11.59 | Steve31 | ... |
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11:20.08 | Steve31 | hello vlad |
11:20.20 | Steve31 | DO u know another channel for asterisk? |
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11:21.15 | Steve31 | everyone is away here... |
11:22.07 | WIMPy | Hmm. A list of channels might be a good thing for infobot. |
11:23.50 | Steve31 | Might be, not that I know to use that bot...It's first time when I'm using mirc :) |
11:25.27 | Steve31 | However, for me it's very strange that a company offers something for sale but they don't give a link where people can see the price or checkout the product |
11:25.47 | Steve31 | Google doesn't help either, so I'm wondering if that voice really exist or it's just a beta project |
11:26.17 | WIMPy | Did you look in the shop? |
11:26.51 | Steve31 | I can't find a shop page. I would really appreciate if you could take a look for 2 seconds. |
11:26.54 | Steve31 | http://www.speechtechnologygroup.com/ |
11:27.24 | Steve31 | Just can't find anything related to the words shop, buy, checkout, add to cart or anything else |
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11:42.46 | Steve31 | WIMPy , any lyck...? |
11:43.07 | F|shie | Steve31: asterisk-dev |
11:43.11 | Steve31 | luck* |
11:43.36 | Steve31 | Thank you fishie |
11:46.39 | WIMPy | That's not a -dev question. |
11:47.15 | WIMPy | You will have to ask those guys if they care to sell their stuff. |
11:47.41 | Steve31 | I completed that form and sent them an email 3 days ago |
11:48.37 | Steve31 | I'm pretty sure if I was a bank I woudn`t have to wait, but heh, I`m not. |
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11:49.01 | WIMPy | So maybe that is the answer. |
11:49.03 | F|shie | check some other providers, acapella/neo |
11:49.28 | Steve31 | Fishie, I consider all voice from Acapella, Neo, Cepstal so bad |
11:49.34 | Steve31 | They all sound as robots |
11:49.38 | Steve31 | voices* |
11:50.16 | Steve31 | WIMPy: might be. Not everyone is a professional. Otherwise they could give an answer, I guess, that's normal. |
11:50.38 | F|shie | you checked nuance? |
11:50.44 | Steve31 | Yes |
11:51.07 | F|shie | and..? |
11:51.09 | Steve31 | They're all the same. Julie from the site I gave you sounds fantastic |
11:51.20 | Steve31 | The rest are really poor quality voices. |
11:52.17 | F|shie | what sampling rate are you checking for? |
11:53.57 | Steve31 | 16 |
11:54.15 | Steve31 | There are plenty of voice on the web, but you can't compare them with that |
11:54.53 | Steve31 | Then I've seen that the price for all 26 languages is 210, that means only Julie might be under 75$ which would be great |
11:55.47 | Steve31 | Well, thank you for your time guys. |
11:56.12 | Steve31 | I guess I have no choice but to wait for their reply, no matter how long it will take. |
11:56.34 | F|shie | gluck with that |
11:57.07 | Steve31 | Thank you very much. |
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12:40.20 | darrenlooby | Watching AMI events, is there something for IVRs, to know what button was pressed? |
12:40.53 | WIMPy | Where? |
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12:41.53 | WIMPy | Thinking about it, I don't think you get any information, befor action is taken anyway. |
12:48.12 | darrenlooby | What I'm really looking for is to know when a call finally makes it to a person, what route they came through... |
12:49.53 | WIMPy | That's easy. |
12:50.07 | WIMPy | You get events whenever the call moves in the dialplan. |
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15:46.44 | slav3_kitten | god i hate myself sometimes *looks at credit card statements* |
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16:45.22 | fulcan | when my carrier gets my SIP MESSAGE it asks my server to authenticate by replying "SIP/2.0 407 Proxy Authentication Required", however, my server does not send authenticated SIP MESSAGE back? http://pastie.org/4177737when Anveo gets your SIP MESSAGE it asks your server authenticate it by replying "SIP/2.0 407 Proxy Authentication Required", however, your server does not send authenticated SIP MESSAGE back |
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16:47.44 | fulcan | http://pastie.org/4177758 |
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17:42.41 | fulcan | . |
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18:40.33 | phaidros | finally, I have asterisk 1.6 & freepbx 2.9.0 .. now: how do I set up a sip server (google wasnt very friendly to me so far) |
18:42.47 | fulcan | . |
18:45.08 | fulcan | When my carrier gets my SIP MESSAGE it asks my server to authenticate by replying "SIP/2.0 407 Proxy Authentication Required", however, my server does not send authenticated SIP MESSAGE back? http://pastie.org/4177737 when Anveo gets your SIP MESSAGE it asks your server authenticate it by replying "SIP/2.0 407 Proxy Authentication Required", however, my server does not send authenticated SIP MESSAGE back? |
18:45.14 | fulcan | http://pastie.org/4177758 |
18:46.11 | phaidros | fulcan: your question is already in the backlog ;) |
18:46.52 | fulcan | phaidros I was hoping to get lucky :) |
18:49.03 | fulcan | ? |
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20:25.43 | [TK]D-Fender | phaidros: Depernds on your definition of "sip server" |
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21:20.54 | fling | 5060/tcp closed sip http://dpaste.com/765817/ |
21:21.27 | WIMPy | SIP usually uses UDP. |
21:21.36 | fling | oh |
21:21.51 | Chainsaw | fling: But if you want your nmap report to look "proper", allow TCP as a transport. |
21:22.05 | Chainsaw | uses TCP as a transport because some of the office networks show packet loss |
21:22.16 | fling | I do not want TCP |
21:22.21 | Chainsaw | (And it's better to have transfers and other call setup features to at least work!) |
21:22.34 | Chainsaw | fling: Then nmap will never show TCP 5060 as open. Sorry. Can't have it both ways. |
21:22.34 | WIMPy | Is TCP faster than Asterisk? |
21:22.57 | [TK]D-Fender | ... |
21:23.04 | Chainsaw | WIMPy: Are you trying to crash our brains? |
21:23.06 | [TK]D-Fender | pardon? |
21:23.08 | drmessano | lol |
21:23.09 | Chainsaw | WIMPy: Rooting out androids? |
21:23.10 | *** join/#asterisk debuggerboy (~phr34k@117.196.163.49) |
21:23.34 | WIMPy | No that was a serious question. |
21:23.34 | drmessano | Only if Spinach is faster than a Porsche |
21:24.12 | Chainsaw | drmessano: I'm just glad I'm not the only one that found the question impossible to answer. |
21:24.19 | WIMPy | Asterisk would retry on UDP. |
21:25.13 | fling | I'm following this > http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
21:25.28 | fling | but ekiga says can' connect |
21:25.33 | fling | cat't |
21:29.12 | debuggerboy | fling: Ekiga and Asterisk are on the same host? |
21:29.18 | fling | debuggerboy: no |
21:30.04 | fling | 5060/udp closed sip |
21:32.00 | debuggerboy | checkd your iptables? |
21:32.27 | fling | do not have iptables, wait I will emerge |
21:34.23 | fling | -P INPUT ACCEPT -P FORWARD ACCEPT -P OUTPUT ACCEPT |
21:34.36 | fling | same if I nmap locally |
21:34.45 | debuggerboy | try something like : iptables -A INPUT -p udp --dport 5060 -j ACCEPT |
21:36.18 | debuggerboy | is your server behind NAT/FireWall |
21:36.31 | fling | no |
21:36.47 | fling | http://dpaste.com/765827/ |
21:40.05 | WIMPy | If you want to make a difference in case of other rules, you need to use -I instead of -A. |
21:40.24 | fling | WIMPy: I have a single rule |
21:42.12 | fling | don't I need any configuration options for this? like caps or something? |
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21:43.36 | debuggerboy | fling: what is the output of your iptables -L |
21:44.04 | fling | https://gist.github.com/3025634 |
21:44.48 | WIMPy | Do you have a bindaddress in your sip.conf? |
21:45.25 | fling | http://dpaste.com/765827/ |
21:46.15 | WIMPy | Looks ok to me. |
21:46.46 | WIMPy | sip show settings |
21:46.50 | fling | bus 5060/udp should be open? |
21:46.58 | fling | but* |
21:48.36 | fling | sip show settings? |
21:57.37 | fling | Am I doing something wrong? |
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21:58.02 | fling | reemerged asterisk with samples and syslog, now I have 5060/udp open|filtered sip |
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22:42.28 | fling | looks like I did not had needed files in /etc/asterisk |
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