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| 03:52.16 | iprouteth0 | anyone here successfully using sip-tls and srtp? |
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| 05:01.30 | gnudna | Hi people ;) |
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| 05:46.31 | ChannelZ | ahoy |
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| 06:03.06 | gnudna | hello |
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| 07:11.52 | fling | [TK]D-Fender: sorry me if I did it wrong, now it is working |
| 07:12.04 | fling | ChannelZ: hey :] |
| 07:12.23 | fling | Is it good idea to 'noload' all the modules I do not use? |
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| 09:31.59 | bretolius | can Asterisk be set up as a light wheight voip server that SIP clients connected to it can make calls to each other? |
| 09:32.37 | bretolius | i dont care about landlines |
| 09:32.41 | ChannelZ | I guess that depends on your definition of "light weight" |
| 09:33.02 | bretolius | could it be run on a cheapish VPS? |
| 09:33.40 | bretolius | IE 768MB RAM 2 CPU Cores, 20Mbps |
| 09:33.58 | bretolius | for like 3 or for clients |
| 09:34.02 | ChannelZ | yes |
| 09:34.39 | ChannelZ | but it seems like a bit much for 3 people to talk to each other |
| 09:36.37 | bretolius | do you have a better recomendation for making voice calls over the internet? I wanted to avoid skype and either go the p2p route, or self hosted, but also retain the ability to use mobile clients |
| 09:37.46 | bretolius | Asterisk came to mind as a self hosted solution, but it seems kind of overkill |
| 09:38.02 | bretolius | the SRTP support looks neat though |
| 09:38.09 | ChannelZ | Well if you're not making calls to actual phone numbers, why NOT use Skype? Otherwise just get clients and use ekiga.net or something |
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| 09:38.38 | bretolius | i use skype already, I was looking into not using it though |
| 09:41.06 | bretolius | ekiga.net looks usefull though thanks |
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| 12:35.27 | DND | hi guys im getting a hangup cause 27. i just wanted to know if this is a problem with telco or from our side. in general wanted to know about this error |
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| 12:37.07 | WIMPy | The answer to that question is in always the location field. |
| 12:37.43 | WIMPy | And 1B usually appears if the U interface at the called end is down. |
| 12:38.38 | WIMPy | I think I've also seen it from mobile phones out of coverage. |
| 12:40.25 | DND | WIMPy: this stupid telco cant even provide me their setting. just double framing and clocksource 16. |
| 12:40.46 | WIMPy | Whatever that means. |
| 12:40.47 | DND | and they are blaming me about not working the isdn work |
| 12:41.34 | DND | WIMPy: so its clearly a telco issue |
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| 12:43.48 | WIMPy | Are you trying to call yourself from somewhere else? |
| 12:43.58 | puzzled | hi |
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| 12:45.08 | DND | telco said isdn is capable of outgoing calls only.. from my pbx i can see dahdi_scan link is OK but when i call it gives hangup cause 27. im pastebin the log hold on |
| 12:46.26 | WIMPy | So you got that when trying to call out via dahdi? IIRC dahdi gives that cause when it's unable to call out. |
| 12:48.10 | DND | yes and based from this site: http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=293 it seems a telco issue |
| 12:48.47 | phix | I mean day/month/year and year/month/day makes sense but who the hell thought month/day/year was even a good idea let alone useful |
| 12:50.24 | WIMPy | DND: I think you're misinterpreting that. |
| 12:50.38 | WIMPy | It is about the called party. |
| 12:56.32 | DND | WIMPy: they said we just call normally without restrictions |
| 12:56.43 | DND | so i tried calling local and international |
| 12:58.22 | WIMPy | So far I assume you're not even talking to your telco. |
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| 13:41.12 | ujjain | SIP registry shows - "No Authentication", does this mean the phone number wouldn't work? |
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| 17:57.29 | learath | Anyone use Broadvoice? |
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| 18:10.11 | _zoom_ | need a sip client with gsm opensource? |
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| 18:12.39 | [TK]D-Fender | _zoom_: No thanks, we already have several :) |
| 18:13.00 | [TK]D-Fender | learath: Someone must or I imagine they'd have run themselves out of business long ago... |
| 18:15.29 | _zoom_ | [TK]D-Fender: may i borrow one of them |
| 18:15.31 | _zoom_ | :) |
| 18:16.53 | [TK]D-Fender | They're non-returnable.... |
| 18:18.02 | [TK]D-Fender | ~softphone |
| 18:18.03 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 18:18.33 | [TK]D-Fender | twinkle, ekiga and many others are OSS |
| 18:19.15 | [TK]D-Fender | SFLphone ... |
| 18:19.29 | carrar | Blink supports gsm |
| 18:19.40 | carrar | and you can hackXor it to change the user-agent too :) |
| 18:19.59 | carrar | (Blink Pro) that is |
| 18:21.35 | _zoom_ | <PROTECTED> |
| 18:22.24 | _zoom_ | carrar: not gsm the codec, I meant gsm voice calls :D |
| 18:22.35 | carrar | that is codec |
| 18:22.43 | [TK]D-Fender | _zoom_: http://www.callcentric.com/img/ekiga_06.png |
| 18:22.48 | [TK]D-Fender | _zoom_: Right there at the top :) |
| 18:23.04 | igcewieling | _zoom_: correct. You'll find no SIP softphone with native CELLULAR GSM support. |
| 18:23.17 | carrar | hahaha |
| 18:23.24 | [TK]D-Fender | _zoom_: Go find one that integrates with your phone... because there is virtually nothing to wil hybridize that functions. |
| 18:23.29 | carrar | I get what he is asking now |
| 18:24.10 | carrar | Just use the cell phone as is, that supports GSM :) |
| 18:28.04 | igcewieling | carrar: the idea is so totally crazy and impossible that is not surprising. |
| 18:28.29 | carrar | heh |
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| 18:31.51 | _zoom_ | carrar: maybe i should try that lool |
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| 19:39.13 | GreatSUN | hi all |
| 19:39.58 | GreatSUN | Does someone of you have know-how onto asterisk + iaxmodem + hylafax configuration? |
| 19:40.24 | GreatSUN | I think I configured everything properly but somehow I dont get asterisk to talk with iaxmodem |
| 19:40.46 | GreatSUN | if qualify is set to yes it says unreachable |
| 19:41.04 | GreatSUN | but port in iaxmodem and asterisk configuration are same |
| 19:41.26 | GreatSUN | ulaw is configured for iaxmodem and allow is set to ulaw in asterisk, too |
| 19:42.08 | GreatSUN | anyone alive who might be able to help? |
| 19:47.01 | [TK]D-Fender | Look at your iaxmodem log file |
| 20:11.39 | GreatSUN | [TK]D-Fender: [2013-04-14 22:10:50] Modem started |
| 20:18.32 | GreatSUN | I can also see iaxmodem listening on 4569 |
| 20:18.54 | GreatSUN | but asterisk is still mentioning it is not reachable |
| 20:19.10 | [TK]D-Fender | It should not be. |
| 20:19.19 | [TK]D-Fender | ASTERISk should be bound to 4569 <- |
| 20:19.32 | [TK]D-Fender | You have misconfigured things |
| 20:19.39 | GreatSUN | oh well |
| 20:19.41 | [TK]D-Fender | IAXModem needs to be on a completely different port |
| 20:19.42 | GreatSUN | then... |
| 20:19.52 | WIMPy | Are Asterisk and iaxmodem running on the same machine? |
| 20:19.53 | GreatSUN | I will reconfigure that and restart |
| 20:19.54 | [TK]D-Fender | And thta's what you set the peer to |
| 20:20.01 | GreatSUN | WIMPy: yeah |
| 20:20.17 | [TK]D-Fender | Go fix your IAXmodem settings |
| 20:20.29 | WIMPy | That's basic networking then. You can't have multiple tasks listening on the same port. |
| 20:21.07 | WIMPy | Well, you can, but the results will not be what you want. |
| 20:23.40 | GreatSUN | ok, moved port |
| 20:23.51 | GreatSUN | now I got status UNKNOWN |
| 20:24.22 | GreatSUN | and in log when call arrives, I see app_dial.c:2196 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 20:25.28 | WIMPy | What did you configure as host=? |
| 20:25.53 | GreatSUN | dynamic |
| 20:26.00 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 20:26.01 | GreatSUN | should that be 127.0.0.1 ? |
| 20:26.07 | WIMPy | Then you need to register. |
| 20:26.19 | WIMPy | And don;t forget the port. |
| 20:27.18 | GreatSUN | the port in the extension should be the port of iaxmodem, right? |
| 20:27.50 | WIMPy | yes |
| 20:28.49 | [TK]D-Fender | [16:23]GreatSUNnow I got status UNKNOWN <- Don't bother calling it... it hasn't registered |
| 20:29.21 | GreatSUN | how do I either eliminate registering or activate it? |
| 20:30.40 | [TK]D-Fender | You don't. |
| 20:30.48 | [TK]D-Fender | Go set it up properly |
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| 20:34.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 20:43.51 | *** join/#asterisk blackvipe (~blackvipe@23.31.140.107) |
| 20:43.59 | blackvipe | Good evening I have a question |
| 20:44.10 | blackvipe | I am working on a Cisco ATA-187 |
| 20:44.30 | blackvipe | I can dail the device |
| 20:44.35 | blackvipe | and pickup it works |
| 20:44.41 | blackvipe | but when I dial out bound on it |
| 20:45.11 | blackvipe | its like the dial plan is missing or something |
| 20:47.11 | blackvipe | anyone with some ideas on how to solve it? |
| 20:52.47 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 20:52.51 | ChannelZ | well it probably does have a dialplan of its own |
| 20:53.17 | ChannelZ | And the default is probably really generic and is sitting there waiting for more digits or something |
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| 20:54.35 | ChannelZ | for fun you can try dialing a number and hitting # and see if it then puts it through (don't know about your ATA specifically but many will treat the # key as a sort of "send now" button) |
| 20:55.19 | blackvipe | well it only allows me to send one digit then I am seeing asterisk dial |
| 20:55.36 | blackvipe | I think one way of possibly sovling it is creating a DISA for that extenion |
| 20:56.54 | blackvipe | maybe create a contact for the ATA adaptor |
| 20:57.36 | [TK]D-Fender | Or..... you could just fix the dialplan on it. |
| 20:57.52 | blackvipe | ya if I could get to the dialplan |
| 20:58.02 | blackvipe | I don't know how to get to the dialplan on it |
| 20:58.07 | blackvipe | I know there is a way |
| 20:58.16 | blackvipe | but I have no xml detail for it |
| 20:58.32 | blackvipe | the dial plan would be a optumal way if I knew the dial plan xml |
| 20:59.01 | blackvipe | or were in the ATA<MAC>.xml file I should put it |
| 20:59.30 | blackvipe | what line needs to be added |
| 21:02.10 | ChannelZ | I'm sure you can hunt around cisco's site for the administration/configuration guide |
| 21:02.21 | blackvipe | looked there already hahahah |
| 21:02.35 | blackvipe | these adaptors are like Ssssssssssssssssssssshhhhhhhhhhhhhhhhhhhhhhhhhhh we aint telling hahhaha |
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| 21:04.55 | ChannelZ | is it SIP? |
| 21:05.35 | blackvipe | yes |
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| 21:14.40 | blackvipe | any ideas on adjusting the dialplan? |
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| 21:54.10 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:54.31 | danfromuk | Hi. How can I test if an extension exists? |
| 21:54.44 | danfromuk | Or even if a SIP Peer exists? |
| 21:55.01 | WIMPy | core show function DIALPLAN_EXISTS |
| 21:55.29 | WIMPy | core show function SIPPEER |
| 21:55.54 | danfromuk | Thanks. DPE is perfect. I knew if existed, just couldnt remember the name. |
| 21:56.05 | danfromuk | it* |
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