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| 00:29.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:27.36 | mavrix | how to tell the audio codecs supported on my mobile carrier end? |
| 01:35.51 | mavrix | I'm having audio issues |
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| 01:42.02 | mavrix | ? |
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| 03:19.10 | linagee | is there a channel for AGI related questions or is that in here? |
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| 03:24.20 | ChannelZ | Here is a good a place as any |
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| 04:20.56 | mjordan | waits for the AGI question... |
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| 04:59.52 | Fwny | Hey all, I am trying to send a crafted INVITE to asterisk, but I find it is getting completely ignored. http://pastebin.com/eNDqiv5S Anyone have any ideas? |
| 05:00.45 | Fwny | sip.conf http://pastebin.com/059iPD8c |
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| 05:24.07 | linagee | is there a way on AGI to get an unspecified number of digits followed by the pound sign? |
| 05:24.34 | linagee | get_data might work...... |
| 05:24.44 | linagee | (if I ignore the pound sign part and rely on timeouts) |
| 05:25.06 | WIMPy | exec Read() |
| 05:27.28 | linagee | hrm..... # might already be a get_data "terminator".... |
| 05:27.42 | linagee | and get_data is *kind* of like Read().... |
| 05:27.55 | linagee | (referring to the Asterisk::AGI get_data |
| 05:29.02 | linagee | it does. yay! :) |
| 05:29.46 | WIMPy | never did any actual call processing in AGI |
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| 05:40.47 | Fwny | solved my problem, was missing a tag on the From |
| 05:45.50 | WIMPy | Wow. Someone trying to scan SIP user names once again. |
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| 05:52.51 | ChannelZ | How unusual! |
| 05:55.04 | WIMPy | Yes. Haven't seen that for many months. |
| 05:56.44 | WIMPy | I almost thought they had given up. |
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| 11:58.56 | matthew-moretalk | Hi Anyone got any ideas how to intergrate asterisk Voicemail with Kamailio Presence for MWI to work? one article I read said you need to force asterisk to generate PUBLISH Responses when voicemail boxes change? any ideas? |
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| 14:02.04 | leifmadsen | I have an international dialing pattern question. If I was setting up a dialplan in the Netherlands, the country code is 31. My understanding is that you prefix with a 0 when dialing in country, and 00 when dialing internationally. From the viewpoint of being in NL, does it even make sense to allow 0031 ? |
| 14:02.33 | nicknam1232 | Nice to have you in the irc room. |
| 14:02.41 | nicknam1232 | On the side. : - ) |
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| 15:07.38 | matthew-moretalk | Hi Anyone got any ideas how to intergrate asterisk Voicemail with Kamailio Presence for MWI to work? one article I read said you need to force asterisk to generate PUBLISH Responses when voicemail boxes change? any ideas? |
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| 15:40.15 | gusto | so |
| 15:40.26 | gusto | somehow few ppl ere? what happend? |
| 15:40.59 | gusto | WIMPy: are you here? |
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| 15:45.55 | Zogot | ahoy all |
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| 16:17.24 | WIMPy | leifmadsen: That's the way standardized dialplans work. And it makes as much sense as using the area code you're in. |
| 16:17.32 | WIMPy | gusto: no |
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| 16:31.28 | pabelanger | leifmadsen: likely not, it would be like dialing 0116135551234 in Canada. You likely get a message |
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| 16:34.06 | WIMPy | Depends on your telco. It should usually work. Known exception is Deutsche telekom who won't allow you to call Germany from within. |
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| 16:36.18 | WIMPy | That's only from the PSTN, off course, from T-Mobile it works as expected. |
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| 17:03.09 | gusto | anyone saw penguin yet? did someone break his mirrors again? |
| 17:06.44 | pabelanger | ~seen penguin |
| 17:06.48 | infobot | penguin <~xwQ5kwYl6@20264.odci.gov.united-states.rltk.us> was last seen on IRC in channel #asterisk, 18d 17h 2m 39s ago, saying: 'They're taking away astdb?'. |
| 17:06.58 | pabelanger | gusto: ^ |
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| 17:15.13 | gusto | yes, he was talking to me because i had forgotten to put that (!) away in my asterisk.conf and that's why the astdb adjustment did not work |
| 17:16.30 | gusto | WIMPy: you mean you can not use 0049 on DT PSTN? i ll have to try that out someday! |
| 17:18.19 | gusto | lol, i am just noticing that my beaglebone clock seems to be synchronized with my mechanical wall clock :-D they are ticking exactly at the same time |
| 17:18.59 | gusto | eh, now it desynced again, lol |
| 17:19.12 | *** join/#asterisk dingle (dingle@162.243.8.166) |
| 17:19.19 | dingle | hello good people |
| 17:20.04 | dingle | does asterisk come with an httpd interface or will i have to install switchvox (which i believe is payware) |
| 17:20.20 | dingle | i just started this expedition today so i apologize for lame questions |
| 17:22.03 | WIMPy | gusto: Yes. They call it reserve for future additional features. Don't know if they believe that themselves. |
| 17:23.07 | WIMPy | dingle: There is not configuration tool included. |
| 17:23.24 | WIMPy | There are many different ones available, all with their restrictions. |
| 17:23.46 | WIMPy | FreePBX seems to be the most active one. |
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| 17:32.38 | dingle | so there are addons i can get to make asterisk gui friendly? |
| 17:33.05 | WIMPy | There are GUIs available. |
| 17:33.24 | WIMPy | Not sure what "GUI friendly" might be. |
| 17:35.57 | dingle | could you please recommend the most easily setup and versatile one? i know those two things are a bit of an oxymoron |
| 17:36.12 | WIMPy | They are. |
| 17:36.46 | WIMPy | But I can't recommend one. All I can say is that FreePBX seems to be the most active project. |
| 17:37.07 | WIMPy | We don't support the use of GUIs in here. |
| 17:37.10 | dingle | i would try asterisknow but i am also attempting this on a vps so thats out |
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| 17:37.25 | dingle | ok, thanks |
| 17:37.34 | WIMPy | Why? |
| 17:38.25 | dingle | why what? |
| 17:38.44 | WIMPy | Why is it out because of using a vps? |
| 17:39.25 | dingle | they dont allow custom iso installs at this time |
| 17:39.40 | gusto | lol |
| 17:39.52 | WIMPy | What a useless service. |
| 17:39.54 | dingle | if i had another machine i would just use that |
| 17:39.59 | gusto | i would hit them with something, my provider allows custom iso installs |
| 17:40.03 | dingle | its very inexpensive |
| 17:40.16 | dingle | how much is your provider per month? |
| 17:43.37 | gusto | i think 9 eur |
| 17:44.51 | gusto | but mine is an overkill i am also thinking about moving to the cheapsest product for 5 eur |
| 17:45.13 | gusto | i have there like 2 GB of RAM and 2 cores and 80 GB of hdd |
| 17:45.57 | dingle | i'd rather not have it based in eu if i am going to be using it for voip |
| 17:46.26 | gusto | actually that is not such a big problem if you enable directmedia=yes |
| 17:47.00 | gusto | but then of course you need to make sure that your provider does not block foreign IP's .. mine does |
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| 17:47.25 | gusto | so being able to take phone calls in slovakia needed me to disable directmedia ... within germany everything works |
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| 19:51.20 | i_am_good | Hi. I want to make a web app which sends a number to call and sound file name to play to Asterisk. But I don't know where to start. What should I know or use? |
| 19:58.48 | [TK]D-Fender | ~book |
| 19:58.49 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:58.58 | [TK]D-Fender | AMI , Call Files. |
| 19:59.02 | [TK]D-Fender | Look 'em up.... |
| 19:59.10 | [TK]D-Fender | "AMI Originate" |
| 20:01.04 | i_am_good | Thanks! |
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| 20:07.10 | oscalation | I have installed asterisk and free pbx, the freepbx/asterisk machine has internet access. I have three ip phones pluged into the same switch, however im stuck now. Not sure what to do to get these phones working |
| 20:07.28 | oscalation | looks like I have to configure routes and trunks? confused |
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| 20:07.54 | WIMPy | Ask in #freepbx |
| 20:08.09 | oscalation | WIMPy: does asterisk have any sort of gui out of the box? |
| 20:08.22 | WIMPy | no |
| 20:08.37 | WIMPy | And we don't support them here, either. |
| 20:08.51 | oscalation | thanks |
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| 20:55.28 | Sean-Der | Has anyone ever seen an ATA that also has a VPN client? |
| 20:56.07 | Sean-Der | I don't want to open up my asterisk box to the outside, but it would be cool if I could just have on device that would handle the VPN and the ATA |
| 20:56.22 | Sean-Der | it is not a huge deal if I had two small devices, but I would rather pay more to have them both combined |
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| 21:01.20 | WIMPy | A router with ATA. |
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| 21:05.30 | Sean-Der | WIMPy: Ok thank you |
| 21:05.40 | Sean-Der | The first thing that turns up is the Cisco SPA122, ever used it? |
| 21:06.06 | WIMPy | nope |
| 21:06.19 | Sean-Der | Do you have any suggestions? |
| 21:07.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.247) |
| 21:08.37 | WIMPy | There are many, but I'm not sure which ones support VPN setups, except for newer FritzBoxen. But they are expensive. |
| 21:09.11 | WIMPy | Draytek should also do it, but they aren't cheap, either. |
| 21:10.01 | WIMPy | And both would probably do a lot more than you need. |
| 21:12.29 | Sean-Der | Yea a little bit in the pricier side, but that beats having two devices |
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| 21:47.00 | Zogot | Ahoyhoy |
| 21:50.51 | [TK]D-Fender | Chips |
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| 22:16.51 | *** join/#asterisk AlphaErr0r^ (~error@186.44.154.208) |
| 22:17.34 | AlphaErr0r^ | hey guys i'm having a problem . When i call into a trunk and dial a extension i dont get fowarded to the extension check out my pastbin showing what i do when the trunk answers http://pastebin.com/vn8riWDQ |
| 22:17.44 | AlphaErr0r^ | btw good afternoon everyone |
| 22:18.34 | AlphaErr0r^ | im tryign to figure out why when i dial a extension it does absolutely nothign except continue playign all the messages using background() |
| 22:19.01 | pabelanger | AlphaErr0r^: we need to see the console output |
| 22:19.08 | pabelanger | it will show you what is wrong |
| 22:19.16 | AlphaErr0r^ | let me add that as well |
| 22:19.41 | pabelanger | also, exten => 900,n can be replaced with same => n |
| 22:19.50 | *** join/#asterisk mjordan (~matt@75.76.55.191) |
| 22:19.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:19.50 | pabelanger | will just down on duplicate code |
| 22:20.53 | *** part/#asterisk mjordan (~matt@75.76.55.191) |
| 22:22.32 | AlphaErr0r^ | here you go |
| 22:22.33 | AlphaErr0r^ | http://pastebin.com/NVDgyFFC |
| 22:23.10 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:26.51 | [TK]D-Fender | <PROTECTED> |
| 22:26.53 | [TK]D-Fender | timeout |
| 22:26.58 | [TK]D-Fender | no choice was made |
| 22:27.43 | AlphaErr0r^ | exactly |
| 22:27.50 | AlphaErr0r^ | but i was punching away on the phone |
| 22:28.02 | AlphaErr0r^ | it didnt detect me entering any extensions |
| 22:28.15 | AlphaErr0r^ | coudl it be a mis configuration in my granstream ? |
| 22:28.20 | [TK]D-Fender | * didn't get them. Pretty strong indicator of not having set the right DTMF mode for the incominfg call |
| 22:28.48 | AlphaErr0r^ | ok so stupid question how do i fix that? |
| 22:28.58 | AlphaErr0r^ | i know its out of the asterisk realm... |
| 22:28.59 | [TK]D-Fender | pick the MATCHING mode |
| 22:30.10 | AlphaErr0r^ | thats the dtms setting in asterisk or granstream ? Not sure where that is |
| 22:30.27 | WIMPy | both |
| 22:30.27 | [TK]D-Fender | BOTH |
| 22:30.34 | AlphaErr0r^ | lol |
| 22:30.59 | AlphaErr0r^ | guess i have some reading to do |
| 22:31.04 | WIMPy | And they need to match. |
| 22:34.08 | *** join/#asterisk mjordan (~matt@nat/digium/x-bqbioiokobtvgoxq) |
| 22:34.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:34.11 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-kwgxpuacirmqrpeq) |
| 22:38.10 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-sbburieadhbnqrap) |
| 22:47.33 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-rcuoduplwejpwzxk) |
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| 22:51.39 | *** join/#asterisk mavrix (~kvirc@109.66.113.242) |
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| 23:02.53 | mavrix | need some helo with codecs negotiation |
| 23:03.12 | mavrix | can someone explain: |
| 23:03.14 | mavrix | Capabilities: us - 0x54e (gsm|ulaw|alaw|slin|g729|ilbc), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) |
| 23:03.59 | mavrix | and later on the log I see all combined codecs, one after the other, added to SDP |
| 23:04.16 | mavrix | why is that? |
| 23:07.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.75) |
| 23:10.58 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-ehvveehwngoxkewe) |
| 23:13.16 | *** join/#asterisk jpoz (~jpoz@89.sub-70-199-133.myvzw.com) |
| 23:15.52 | mjordan | mavrix: you have a set of codecs on a channel (us); the peer being dialed has another set of codecs that are configured on it (via disallow/allow). While there are exceptions to this rule, in general, Asterisk will attempt to avoid transcoding by offering only those codecs that are supported by both the inbound channel and the peer's supported codecs |
| 23:18.56 | mavrix | mjordan: but in log I see all combined codecs added to SDP - so how to tell the chosen codec? |
| 23:22.38 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-zwyrqwzeriarrtwh) |
| 23:27.06 | mavrix | any idea? |
| 23:28.56 | mjordan | mavrix: it will be the codec returned in the SDP of the response to the outbound INVITE request. In general, 'sip show channel' will show you what is being used |
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| 23:31.58 | mavrix | mjordan: thanks. Although I have alaw as my first priority - ulaw is always chosen for that call, Why do you think is that? |
| 23:32.15 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-pjevrnrfxhpcwftl) |
| 23:32.42 | mavrix | mavrix: is it a bug on Asterisk that it doesn't respect priority? |
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| 23:34.00 | *** join/#asterisk Fwny (~potato@192-0-250-120.cpe.teksavvy.com) |
| 23:34.10 | mjordan | without looking at your config - which, since I'm about to eat, I won't be around for - not sure. There are some quirks regarding ulaw/alaw; in general, if you want alaw I'd just remove ulaw from the peer in question. It'd be a little odd to need both anyway. |
| 23:34.23 | mjordan | sighs at CaptainPants |
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| 23:35.13 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-xbooytmqmjnvkedr) |
| 23:35.45 | mavrix | mjordan: thank you very much. I hope this problem is only between alaw ulaw... |
| 23:36.03 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-hdvfyhxkqnidrnub) |
| 23:50.40 | AlphaErr0r^ | thanks guys i got through |
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