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| 00:01.21 | OneFix_Work | Qwell: KEwl |
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| 03:25.21 | dlynes | Are there a lot of authentication issues with Asterisk 1.8.1.1? I seem to be having rudimentary authentication problems with two boxes that worked on 1.4 just fine |
| 03:25.43 | dlynes | One of these boxes is an asterisk 1.8.1.1 box, and the other is a sipura 2000 |
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| 03:33.46 | pabelanger | dlynes: I know of one issue that needs fixed |
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| 03:37.25 | dlynes | pabelanger, do you know the issue number? |
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| 03:47.03 | russ | http://code.google.com/p/android/issues/detail?id=9392 |
| 03:47.08 | russ | er, sorry |
| 03:47.13 | russ | wrong channel, ironic |
| 03:48.30 | dlynes | russ, it's still phone-related :p |
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| 08:08.24 | asterisk-learner | hello |
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| 08:37.21 | tuxx- | ola |
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| 08:45.13 | ChannelZ | ole' |
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| 09:35.14 | hurdman | hi |
| 09:35.58 | hurdman | i have got a problem using odbc and asterisk, i don't understand why my odbc (only with asterisk) register in iso instead of utf 8 |
| 09:36.01 | hurdman | any idea ? |
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| 09:57.55 | skrusty | hipitihop, anyone know why i wouldn't be getting any QueueEntry events from * after doing a QueueStatus request? Works fine on 1.6, but on a server with 1.4, it doesn't give me the callers in a queue... |
| 09:58.08 | skrusty | oops, not sure why it addressed that to hipitihop :/ |
| 10:03.35 | tuxx- | hi<tab>, |
| 10:03.36 | tuxx- | :D |
| 10:03.53 | skrusty | indeed :) |
| 10:04.12 | skrusty | does 1.4 have some additional option that needs turning on? :/ |
| 10:04.32 | skrusty | i get member events, but nothing about callers entering queues :/ |
| 10:04.41 | skrusty | or existing callers in queues... |
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| 10:16.09 | Sheeplet | lo all |
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| 11:27.55 | klashniv | hullo all, quick que here: using asterisk 1.6.2.9, need to know whether increasing value of trunkfreq in iax.conf will increase or decrease bandwidth use |
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| 11:34.27 | klashniv | does increasing value of trunkfreq in iax.conf will increase or decrease bandwidth use? |
| 11:34.35 | klashniv | anyone know? |
| 11:35.57 | klashniv | am trying to get the calls to be as uncompressed as possible, am troubleshooting call quality issues |
| 11:36.09 | klashniv | average concurrent calls is 120 |
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| 11:38.49 | ruyo | What can be causing DISA to capture 2 DTMF for each key press? |
| 11:39.00 | ruyo | Via SIP - inband DTMF. |
| 11:40.10 | ruyo | More exacly, Asterisk calls a mobile phone via VoipCheap and presents it the DISA app. |
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| 12:27.15 | dlyneswork | ruyo, incorrect translation between dtmf encodings (inband/sip info/rfc2833), if it's a sipura/linksys unit (and maybe for other atas as well) it could be incorrect dtmf recognition (need to force a certain type in that case) |
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| 12:29.06 | dlyneswork | ruyo, also, are you using ulaw or alaw along with the inband dtmf? |
| 12:29.30 | black187 | hello - one question, is it possible to patch asterisk 1.6.x with the new connectedline feature of 1.8 - for call pick-up and call tranfer notifications... |
| 12:29.53 | dlyneswork | ruyo, apparently you need to specify 'dtmfmode=inband' and 'dtmf=inband' (http://www.freepbx.org/forum/dtmf-problems-with-callback), although that's not my experience |
| 12:30.39 | WIMPy | black187: That is not going to be easy at all. So in general I'd say no, even if that technically warong off course. |
| 12:31.16 | black187 | WIMPy: Ok, thanks :( |
| 12:32.37 | black187 | is it possible to notify the phone which makes the call pickup of the caller id of the pick-up person - via SIP Invite? |
| 12:32.44 | black187 | SIP reinvite... |
| 12:33.17 | WIMPy | That could work the exact same way. |
| 12:33.31 | WIMPy | But I have no idea if that has been done. |
| 12:33.58 | black187 | So this could be done with dialplan macros - (to initiate a reinvite with different caller id information)? |
| 12:34.16 | black187 | just so I know where to start looking |
| 12:35.05 | WIMPy | I don't think you can do it via the dialplan. |
| 12:36.28 | black187 | damn - any idea where i can do it? |
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| 13:03.12 | *** join/#asterisk moe_adam_345 (5d32d572@gateway/web/freenode/ip.93.50.213.114) |
| 13:04.17 | moe_adam_345 | Hi. I am working for a small company in Canada trying to launch a vo-ip service. We are looking into Asterisk for our PBX needs. Does anyone have a few minutes to answer some basic questions? |
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| 13:05.58 | WIMPy | ~ask |
| 13:05.58 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:06.21 | JAMMAN2110 | I love the "or against our will" part of that |
| 13:06.46 | JAMMAN2110 | Also, moe_adam_345: You're trying to launch a VoIP service and need to ask basic questions? |
| 13:07.14 | moe_adam_345 | Does anyone have any experience with VoIPJet? |
| 13:07.22 | WIMPy | Yes, not a good starting position. |
| 13:07.56 | WIMPy | moe_adam_345: Read infobot again. |
| 13:07.56 | JAMMAN2110 | VoIP Jet the company? |
| 13:08.00 | JAMMAN2110 | What part of the company? |
| 13:08.12 | JAMMAN2110 | Their services? What part of the services? The support? What part of the support? |
| 13:08.20 | JAMMAN2110 | Their website? Well its pretty shit, but what part of the website? |
| 13:09.24 | drmessano | First off, what is "SIP" and do I need one to sell VoIP? |
| 13:09.43 | moe_adam_345 | I am planning on using them for call termination of our beta set-up. I set up an account, but it looks like their website has not been updated since 2006, and their mail-list is a desert. Am I wasting my time? Or can they be relied on? |
| 13:10.34 | JAMMAN2110 | SIP stands is the acronym for the Stupid Invertebrate Protocol, it defines people that come into IRC channels and behave like dicks |
| 13:11.11 | drmessano | That's just awful |
| 13:11.18 | JAMMAN2110 | Best I could come up with at 2am |
| 13:11.28 | JAMMAN2110 | Thats Rev 2 |
| 13:11.32 | JAMMAN2110 | Rev 1 was Stupid Idiot P |
| 13:12.18 | drmessano | VoIPJet US48 termination is 1 cent per minute |
| 13:12.55 | drmessano | Considering I can get .98 cents per minute as end user from a few places, how much are you going to resell? |
| 13:13.22 | drmessano | Sounds a bit like starting an ITSP with Vonage ATAs, TBBH |
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| 13:14.21 | moe_adam_345 | Would Asterisk+VoIP jet be a suitable solution for a simple service that makes two outgoing calls (terminated through VoIPJet), then bridges the channels? I have been reading a good amount through the asterisk documentation. It is impressive, but somewhat overkill for what I am looking for. All I need is to build a system that can scale, make outgoing calls, and bridge the channels. |
| 13:14.59 | hipitihop | skrusty, been away from keyboard and indeed wondered why there was something for me unprompted while I lurked :-) |
| 13:15.08 | WIMPy | Maybe you are searching for a SIP proxy instead? |
| 13:15.20 | JAMMAN2110 | VoIP Jet are IAX |
| 13:15.21 | drmessano | Asterisk would work fine for an ITSP, though I question your provider choice.. but whatev |
| 13:16.37 | moe_adam_345 | It was really the first provider I found, this is just for a proof of concept system. Do you have any specific reasons to avoid VoIPJet from a reliability standpoint? |
| 13:19.03 | JAMMAN2110 | I suggest using Skype |
| 13:19.11 | moe_adam_345 | :) |
| 13:19.28 | drmessano | Nope, just never heard anyone mention them until now.. Doesn't sound like it's widely used... and again, they have a different focus than what you ultimately have |
| 13:19.30 | WIMPy | What's wrong with google? |
| 13:19.56 | JAMMAN2110 | Or telnet, I hear good things about telnet |
| 13:19.56 | Nugget | telnet is eeeeeeevil! |
| 13:20.15 | JAMMAN2110 | Snap |
| 13:20.28 | moe_adam_345 | The goal is to allow mobile users in Europe (where incomming calls are free) to instigate two outgoing calls (using an SMS gateway). Calling in-between european countries is very expensive. The cost of terminating two outgoing calls through VOIP is frequently less than the cost of making one outgoing call. |
| 13:20.53 | drmessano | You're buying donuts from the bakery and selling them at your bakery... You should be looking for a wholesale joint |
| 13:21.06 | WIMPy | moe_adam_345: Where is that expensive? |
| 13:21.39 | WIMPy | Free calls to half of europe aren't that special any more. |
| 13:21.48 | drmessano | finds a list of countries where VoIP is banned ---> There |
| 13:22.02 | drmessano | lol |
| 13:22.28 | drmessano | Sounds like another one of those "I am setting up a server in Dubai. Before they chop my hands off, can you look at this Macro" scenarios |
| 13:22.59 | WIMPy | Dubai is not in Europe. |
| 13:23.04 | drmessano | I am well aware |
| 13:23.52 | moe_adam_345 | Switzerland and Italy are two big culprits. |
| 13:24.14 | moe_adam_345 | It costs around 80c to make a local outgoing call on Swisscom pre-pay. |
| 13:24.26 | JAMMAN2110 | Is that all? |
| 13:24.29 | JAMMAN2110 | Hell |
| 13:24.34 | moe_adam_345 | that is per minute |
| 13:24.58 | WIMPy | moe_adam_345: Are you talking about mobiles? |
| 13:25.03 | moe_adam_345 | yes |
| 13:25.23 | JAMMAN2110 | "There is a standard rate of just 89 cents per minute to call NZ mobiles, NZ landlines and mobiles" |
| 13:25.23 | drmessano | I wasn't aware they had pre-pay home phones |
| 13:25.31 | JAMMAN2110 | Source: http://www.vodafone.co.nz/prepay/supa-prepay.jsp |
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| 13:28.07 | drmessano | moe_adam_345, people do this all the time.. There's even an application called "A2Billing" that makes all this calling card stuff quite easy, and it runs on top of Asterisk |
| 13:28.12 | moe_adam_345 | I am not really hear to discuss the economics of this. I am trying to build a prototype. I think I will just hack around a bit more. I feel like you guys are looking for questions somewhat more technical. Thank you. |
| 13:28.22 | WIMPy | moe_adam_345: Well, then you have the problem of VOIP being bannd on many networks. |
| 13:28.59 | JerJer | A2Billing is a joke - yet nobody gets the punch line |
| 13:29.01 | moe_adam_345 | In what sense? Are you saying they ban VoIP companies terminating on their lines? |
| 13:30.03 | WIMPy | moe_adam_345: No they are blocking voip traffic. Maybe you should take a look in to the terms and conditions of the providers in question. |
| 13:30.11 | drmessano | JerJer, the rest of this conversation has been as well.. including paying non-wholesale rates to some mystery provider |
| 13:30.22 | drmessano | inflates a dot com bubble and grabs a pin |
| 13:30.26 | drmessano | *BOOM* |
| 13:32.43 | drmessano | Toll-Free termination, such as 1800/1888/1877/1866 numbers, cost us money to terminate and thus are charged at the same rate as regular US calls. There are some providers who can terminate some, but not all, 1800 numbers for free. (If they could terminate all 1800 numbers for free, then we'd use them!) <-- Comical |
| 13:33.13 | drmessano | Standard US rates to terminate a TOLL FREE call |
| 13:33.26 | drmessano | Red Flag.. |
| 13:33.40 | moe_adam_345 | Yes, I understand they are blocking voIP traffic, but that is on the Mobile Data channel. These calls are still going over the voice channel. They are terminated through the voice channel. Mobile internet use is not the goal here. The calls are instigated though an SMS. |
| 13:33.47 | WIMPy | Some mobile operators tried that here as well. But that was found to be illega. |
| 13:34.25 | moe_adam_345 | drmessano, could you be so kind as to point me to a more suitable VoIP termination service that uses IAX2? Perferrably something quick to set up. |
| 13:34.30 | WIMPy | moe_adam_345: What is an SMS? So yu are talking call-through? |
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| 13:35.25 | WIMPy | . o O ( the more I read, the less sense it makes ) |
| 13:36.14 | moe_adam_345 | 1. caller sends sms to me (sms contains recipients number) 2. I make two outgoing calls (to caller and recipient) 3. I bridge the calls 4. caller talks to recipent. |
| 13:37.09 | WIMPy | Sounds like in the days of USSD callbacks. |
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| 13:38.38 | JAMMAN2110 | My VoIP provider offers that, for nothing, bar the cost of the call |
| 13:38.56 | JAMMAN2110 | Why would I pay moe_adam_345? |
| 13:38.58 | moe_adam_345 | May I ask who your VoIP provider is? |
| 13:39.03 | JAMMAN2110 | 2talk.co.nz |
| 13:39.39 | JAMMAN2110 | I can even request the callback online, via email, etc |
| 13:39.55 | moe_adam_345 | Can you do it through an SMS? |
| 13:40.02 | JAMMAN2110 | Yes |
| 13:40.41 | WIMPy | A SM, BTW. You're not sending the whole service. |
| 13:41.08 | WIMPy | But I'm pretty sure that the use of some calling card would be a lot cheaper. |
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| 13:42.05 | JAMMAN2110 | moe_adam_345: 2talk even offer free accounts which give you two phone numbers and an amount of minutes |
| 13:42.18 | JAMMAN2110 | And their entire featureset |
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| 13:52.58 | JerJer | drmessano: (late response) very large red flag when they charge for toll-free calls |
| 13:55.29 | sivang | re all |
| 13:55.30 | sivang | mornang |
| 13:55.33 | sivang | mornange |
| 13:55.49 | sivang | any sip conf gurus around? |
| 13:56.05 | sivang | I'm still unable to route my call into my designated extesnuio |
| 13:56.19 | sivang | from inbound that is, and afaik I am matching against username |
| 13:56.41 | JerJer | context=foo in the sip peer stanza? |
| 13:56.51 | sivang | JerJer: yep |
| 13:57.02 | sivang | context=incoming_calls, to be more precise |
| 13:57.08 | JerJer | do you have more than one peer for this same sip trunk ? |
| 13:57.11 | WIMPy | Sounds like a peer/friend/user thing. |
| 13:57.19 | JerJer | (ie where does the call go) |
| 13:57.30 | sivang | JerJer: the weasels are answering it :) |
| 13:57.54 | sivang | JerJer: the default context, that is |
| 13:58.02 | JerJer | WIMPy: i have never found a need for anything more than type=peer |
| 13:58.17 | sivang | JerJer: yep that's not it, alerady tried eitehr |
| 13:58.58 | JerJer | sivang: add insecure=port,invite |
| 13:59.08 | JerJer | and sip reload |
| 13:59.28 | sivang | JerJer: port? |
| 13:59.37 | sivang | JerJer: as in the literal word 'port' ? |
| 13:59.44 | WIMPy | yes |
| 14:00.12 | JerJer | exactly like that in your sip stanza for this trunk |
| 14:00.31 | sivang | you mean, have ti in the trunk and all the extensions? |
| 14:00.38 | sivang | err, |
| 14:00.40 | sivang | sip peers |
| 14:00.44 | JerJer | [blah] |
| 14:00.47 | JerJer | type=peer |
| 14:00.48 | sivang | I don't have it anywhere |
| 14:00.53 | JerJer | host=whatever |
| 14:00.59 | JerJer | insecure=port,invite |
| 14:01.02 | *** part/#asterisk mintee (1000@static-64-115-220-213.isp.broadviewnet.net) |
| 14:01.07 | sivang | for all the entries in sip.conf? |
| 14:01.16 | sivang | not just for the trunk? |
| 14:01.20 | JerJer | no, just for this particular trunk (ie see sip.conf.sample) |
| 14:01.25 | sivang | k |
| 14:01.28 | jkroon | sivang, just ask the actual question. |
| 14:02.21 | sivang | JerJer: same thing |
| 14:02.33 | sivang | e.g. no go, I still hit the default context |
| 14:03.06 | JerJer | did you reload? |
| 14:03.55 | JerJer | take the default context out - then you should get a notice of it being denied (which might help you match what's not matching) |
| 14:04.26 | sivang | JerJer: good idea, yes I realoded but i found out that in some cases it must go through a restart |
| 14:04.26 | JerJer | might need to add defaultuser=[your real username] |
| 14:05.00 | JerJer | anything sip related can be changed with a simple reload |
| 14:06.01 | JerJer | sivang: is this between two asterisk boxes or some sip provider and your asterisk box ? |
| 14:06.39 | JerJer | i have had serious difficulties making two asterisk boxes talk SIP to each other (depending on the network situation) |
| 14:08.33 | sivang | JerJer: me and the provider |
| 14:08.54 | sivang | JerJer: well, more accurately, an asterisk that is at/connected to the provider |
| 14:09.00 | sivang | JerJer: I'm not into those details |
| 14:09.41 | JerJer | is there NAT? |
| 14:10.13 | sivang | JerJer: so while we speak I change the register line to use the internal address (it has an internal address as well) |
| 14:10.35 | JerJer | do you have a host=[internal.address] ? |
| 14:10.42 | JerJer | in the sip peer |
| 14:10.44 | sivang | JerJer: changing it right now |
| 14:10.51 | *** join/#asterisk rrb3942 (~rbullock@208.49.79.66) |
| 14:11.31 | sivang | JerJer: what is the significance of the register => in relation to the [..] stanza? |
| 14:11.42 | ruyo | dlyneswork, Thanks for the tips, I'll look into it. |
| 14:11.53 | sivang | JerJer: e.g. , the regsiter in in [general] and I have my provider defined in [myproivdier] |
| 14:12.03 | JerJer | the register is simply the registration |
| 14:12.16 | JerJer | you still need a matching [foo] type=peer entry for the provider |
| 14:12.39 | sivang | okay |
| 14:12.50 | sivang | so register to the iternal address gives me 944 |
| 14:12.56 | sivang | err |
| 14:12.58 | sivang | 404 |
| 14:13.06 | sivang | switches back to external address |
| 14:13.08 | JerJer | that's a provider then than |
| 14:13.13 | sivang | yes |
| 14:13.15 | JerJer | thing * |
| 14:13.25 | sivang | why ddoes it have two ips? |
| 14:13.39 | sivang | or maybe the internal is the NAT device? |
| 14:13.47 | sivang | 192.168. something |
| 14:14.18 | JerJer | if its colocated with the provider, you really should use the internal network |
| 14:14.30 | sivang | I wish I knew |
| 14:14.38 | sivang | I will have this asnwer only on sunay |
| 14:14.45 | JerJer | kind of important details to know |
| 14:14.59 | sivang | I understand, but I'mn not the network admin |
| 14:15.47 | sivang | can I put 2 hsot entries inthe stanza? |
| 14:16.02 | JerJer | i doubt it |
| 14:16.05 | WIMPy | no |
| 14:17.54 | sivang | and I should remove the hos tline if I'm matching based on user right? |
| 14:18.05 | leifmadsen | yes |
| 14:18.11 | WIMPy | yes |
| 14:18.16 | leifmadsen | in that case you just need type=user |
| 14:18.19 | sivang | so that does not work, as well |
| 14:18.19 | leifmadsen | not type=peer |
| 14:18.34 | sivang | so no friend no peer just user? |
| 14:18.38 | sivang | what's the significance of that? |
| 14:19.16 | WIMPy | leifmadsen: I read the dundi chapter yesterday. There is some unfortunate negation right at the start. I hope caching isn't used to _increase_ response times. |
| 14:19.52 | leifmadsen | WIMPy: oh really! then I should probably fix that ;) |
| 14:20.09 | leifmadsen | thanks for the feedback! |
| 14:20.34 | leifmadsen | haha wow, and I get it was like that in the other book too |
| 14:20.38 | leifmadsen | I don't think I changed that section |
| 14:20.46 | JerJer | sivang: the user/peer/friend is an unfortunate take over from IAX |
| 14:20.53 | leifmadsen | fixed |
| 14:20.55 | JerJer | which does not work very well with SIP concepts |
| 14:21.45 | sivang | I don't mind. I wish it just worked so I can move along with my coding task |
| 14:21.54 | sivang | as I did with freeswitch |
| 14:22.06 | sivang | (I need to setup both) |
| 14:22.12 | JerJer | there is a very specific reason why its not matching |
| 14:22.20 | sivang | I'm sure |
| 14:22.25 | sivang | JerJer: I've removed the default context |
| 14:22.26 | JerJer | does the invites come from a different address than the registration ? |
| 14:22.30 | sivang | JerJer: and the call is now rejected |
| 14:22.42 | sivang | JerJer: so apparentyl yes, but matchin on user hsould have fxied that |
| 14:22.55 | JerJer | not in my experience |
| 14:23.19 | leifmadsen | WIMPy: if that was the only issue you found, then that'd be swell :) |
| 14:23.22 | sivang | isn't asterisk deterministic? I would think experience shoudl not be rlated to that |
| 14:23.39 | sivang | the book also mentioned user mathicn is the solution to that. |
| 14:23.42 | JerJer | i cannot recall one instance where i used type=user (or worse friend) in sip.conf |
| 14:24.16 | sivang | not type user, but set user matchin as in fromuser= and [userid] in the stanza |
| 14:24.28 | WIMPy | leifmadsen: No, it tells you about the dbsecret, but there is no exaple if using that in iax.conf. I think that should be added. |
| 14:24.37 | sivang | JerJer: what do I look for in a failed sip trace? |
| 14:24.38 | JerJer | add defaultuser= |
| 14:24.45 | sivang | JerJer: E.g.where the call got reject? |
| 14:24.50 | Katty | yellow. |
| 14:24.56 | JerJer | purple |
| 14:25.06 | WIMPy | rain |
| 14:25.22 | sivang | JerJer: defaultuser= ? |
| 14:25.23 | JerJer | sivang: asterisk should spew out a message on the cli |
| 14:25.28 | sivang | JerJer: eq. what? |
| 14:25.32 | leifmadsen | WIMPy: can you email me anything you can think of to leif.madsen -at- gmail -dot- bomb so I can work on that tomorrow? I'm done working on the book for this morning |
| 14:25.39 | JerJer | sivang: to the sip trunk |
| 14:25.44 | leifmadsen | even just a copy-paste of what you just wrote is fine |
| 14:25.46 | JerJer | ie see sip.conf.sample |
| 14:25.59 | WIMPy | leifmadsen: ok |
| 14:26.10 | sivang | [Jan 3 18:45:03] NOTICE[27969]: chan_sip.c:15503 handle_request_invite: Call from '' to extension 's' rejected because extension not found. |
| 14:26.22 | JerJer | ok thats a whole different ball game |
| 14:26.28 | sivang | Katty: I thought it was more than just Yello |
| 14:26.38 | JerJer | your sip provider is not sending an extension to dial |
| 14:26.42 | sivang | Katty: henc eit wook me more time, since tere are sometimes other failing things not colord |
| 14:27.06 | sivang | JerJer: can it work in freeswitch if it does not do that? |
| 14:27.24 | sivang | JerJer: I mean, that would probably make freeswitch cripple as well? |
| 14:27.47 | JerJer | tweak your register line: register => username:pass@host/extension |
| 14:28.02 | *** join/#asterisk tftech (~tftech@ip65-47-56-98.z56-47-65.customer.algx.net) |
| 14:28.08 | JerJer | but that is going to seriously limit your provider's usefulness |
| 14:28.14 | sivang | exactly |
| 14:28.27 | sivang | I would not be able to move it to a queue and have a bunch of agents responding no? |
| 14:28.40 | JerJer | you could, but that's all its gonna do |
| 14:28.48 | JerJer | ie - no DIDs |
| 14:29.14 | sivang | okay |
| 14:29.44 | sivang | so the provider does not pass the extension that was dialed to him, so asterisk has got nothing to match on? |
| 14:29.52 | JerJer | yup |
| 14:29.54 | sivang | bad provider! |
| 14:30.28 | sivang | JerJer: but this is still enough to pass it into default context? |
| 14:30.37 | sivang | JerJer: where have you been for the last 3 days? :) |
| 14:30.59 | JerJer | coding |
| 14:31.16 | sivang | JerJer: thanks a lot. |
| 14:32.06 | JerJer | no problemo |
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| 14:58.51 | SuPrSluG | having an issue with sip subscriptions. most phones ok, but a few aren't showing their subscriptions to an extension. What settings, timing etc would asterisk use? |
| 15:02.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:02.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:08.34 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 15:08.42 | IsUp | hello |
| 15:10.52 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
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| 15:16.32 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 15:19.47 | AdvoWork | hi there, using trixbox, ive just set up a new extension. and the phone is SIP snom 300. I'm wondering if I can somehow set it so if that extension/phone picks up the handset, it would auto dial a ring group? |
| 15:20.55 | WIMPy | AdvoWork: You are lucky: That has nothing to do with trixbox, wich we don't support here. You need to tell your phone. |
| 15:21.50 | IsUp | yeah actually it must be a phone feature |
| 15:24.33 | AdvoWork | i can setup a function key so i press a number ie L2 and it dials an extension |
| 15:25.47 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:27.44 | AdvoWork | hmm, can you have extension > ring group > multiple extensions? |
| 15:27.45 | angryuser | Hello, is that normal that Asterisk ignore TO tag when invite comes ? |
| 15:27.52 | angryuser | asterisk 1.6 |
| 15:28.28 | WIMPy | AdvoWork: That is something you should ask in #trixbox. |
| 15:28.32 | angryuser | I.E i have an account 1234546, invite comes INVITE:1234546, TO 99999999 |
| 15:28.49 | angryuser | why asterisk does not ring 9999999 ? |
| 15:29.25 | IsUp | angryuser: can you paste your 'sip debug' @ pastebin? |
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| 15:32.14 | *** part/#asterisk WonTu (~WonTu@p57B5425B.dip.t-dialin.net) |
| 15:32.52 | angryuser | IsUp, sure here it is: http://pastebin.ca/2039578 i have added the dialplan i use (really simple) |
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| 15:33.40 | angryuser | IsUp, i just set Set(NUMBER=${EXTEN}) |
| 15:34.01 | angryuser | and as you see i get the account number, instead of To tag number |
| 15:34.51 | leifmadsen | angryuser: well it's doing exactly what I would expect it to be doing |
| 15:35.03 | IsUp | leifmadsen: right |
| 15:35.05 | leifmadsen | INVITE sip:99000000000123@83.167.156.46 SIP/2.0 |
| 15:35.36 | angryuser | how can i extract the To field, as it is contain dialed number ? |
| 15:36.14 | leifmadsen | I'd like to know why your invite is constructed like that more so |
| 15:36.39 | angryuser | leifmadsen, sure i can telll you more, what exactly do you want to know ? |
| 15:36.42 | leifmadsen | anyways, you can try SIP_HEADER() |
| 15:37.09 | leifmadsen | sorry, I have to head off to do some work |
| 15:37.15 | leifmadsen | give that function a try though |
| 15:37.35 | angryuser | By the way is it 'rfc' doing invite like this ? i have for example X DID's at that account + username |
| 15:37.54 | angryuser | My provider change To field to tell which did is called |
| 15:38.00 | angryuser | leifmadsen, ok, thank you |
| 15:38.32 | IsUp | angryuser: yeah, its 'rfc' standart, i think |
| 15:38.52 | angryuser | IsUp, damn |
| 15:39.11 | IsUp | angryuser: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header |
| 15:39.52 | IsUp | ${SIP_HEADER(TO)} |
| 15:40.08 | angryuser | IsUp, yes i see it |
| 15:40.29 | angryuser | It will work, i suppose |
| 15:40.53 | angryuser | IsUp, the problem is that it will require me to modify 30+ servers ;( |
| 15:41.24 | IsUp | angryuser: well i dont have another solution :P sorry |
| 15:42.42 | angryuser | IsUp, damn standarts |
| 15:44.10 | IsUp | in this case, it's "damn provider" |
| 15:44.20 | angryuser | exten => _.,3,Set(NUMBER=${SIP_HEADER(TO)}) gives me empty $Var |
| 15:44.39 | angryuser | IsUp, yes damn me, i am the provider xD |
| 15:44.39 | *** join/#asterisk dlyneswork (~dlynes@216.185.79.50) |
| 15:45.09 | IsUp | replace "TO" -> "To" |
| 15:45.19 | IsUp | and see if its make any difference |
| 15:45.46 | angryuser | IsUp, same |
| 15:45.58 | angryuser | i will try to read man |
| 15:47.11 | angryuser | IsUp, sure |
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| 16:05.19 | neurosys_ | Quick question for anyone who may know: Do the Aastras require a special software update (like polys) to use directories with LDAP? |
| 16:05.29 | z4nD4R | hi all...i have voicemail.conf defined as 71001 => 1001,1001 .. but if i put user 1000 and passwd 1000 astaerisk say that login incorect... somebody know way? |
| 16:05.51 | *** part/#asterisk sivang (~sivang@unaffiliated/sivang) |
| 16:09.21 | IsUp | z4nD4R: "Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options>" |
| 16:09.39 | IsUp | so user is 71001 and password is 1001 |
| 16:09.44 | IsUp | as i understand |
| 16:11.18 | z4nD4R | IsUp: - Incorrect password '1000' for user '71000' (context = default) |
| 16:11.31 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 16:11.39 | IsUp | try 1001 on password |
| 16:11.43 | IsUp | not 1000 |
| 16:11.48 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:11.58 | z4nD4R | i have booth users 71000 => 1000,1000 |
| 16:12.51 | IsUp | can you paste output of "voicemail show users" on CLI? |
| 16:13.01 | z4nD4R | w8 |
| 16:13.46 | z4nD4R | http://pastebin.com/pavD3gug |
| 16:15.39 | z4nD4R | i know.... (context = default) ... but i have another context |
| 16:16.32 | z4nD4R | or |
| 16:16.35 | z4nD4R | ? |
| 16:17.21 | IsUp | can you paste your voicemail.conf please? |
| 16:20.12 | ChannelZ | Your context is wrong |
| 16:20.19 | ChannelZ | at least based on your config and based on the error you pasted |
| 16:20.51 | z4nD4R | http://pastebin.com/v9nXa76t |
| 16:21.12 | ChannelZ | You'd need to be calling VoiceMailMain(71001@skuska_voicemail) |
| 16:21.53 | IsUp | yes, your users are in "skuska_voicemail" context |
| 16:22.56 | z4nD4R | IsUp: so? i muss set this context in extensions.conf? |
| 16:23.15 | z4nD4R | ChannelZ: only(71001) |
| 16:23.16 | z4nD4R | ? |
| 16:23.20 | IsUp | <ChannelZ> You'd need to be calling VoiceMailMain(71001@skuska_voicemail) |
| 16:23.45 | IsUp | or define your users in "default" context |
| 16:24.01 | z4nD4R | ok w8 |
| 16:25.41 | z4nD4R | i have in sip per each user |
| 16:25.59 | z4nD4R | mailbox=71000@skuska_voicemail |
| 16:27.18 | ChannelZ | That just informs it what mailbox to monitor for MWI |
| 16:27.46 | ChannelZ | You still have to access the right voicemail context when you send people to voicemail. |
| 16:30.15 | z4nD4R | i am confused ... with context muss to set in wich file...so |
| 16:30.27 | z4nD4R | in sip i have -> mailbox=71000@skuska_voicemail |
| 16:30.51 | z4nD4R | in voicemail.conf -> [skuska_mailbox] |
| 16:30.52 | z4nD4R | 71000 => 1000,1000 |
| 16:30.58 | ChannelZ | As I just said, in sip.conf that statement only tells * what voicemail box/context to monitor for message-waiting indications |
| 16:31.03 | IsUp | z4nD4R: go to extensions.conf |
| 16:31.15 | IsUp | look for the "VoiceMailMain" |
| 16:31.26 | ChannelZ | When you call VoiceMail or VoiceMailMain you need to specify the right mailbox and context as well. It has nothing to do with anything in sip.conf |
| 16:31.28 | z4nD4R | IsUp: yes |
| 16:32.04 | IsUp | as ChannelZ said, you should specify context in VoiceMailMain, like VoiceMailMain(71001@skuska_voicemail) |
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| 16:33.41 | highvoltz | Any idea when 1.4.39 will be released? |
| 16:35.16 | z4nD4R | ChannelZ: IsUp thx.. :) |
| 16:36.06 | IsUp | np :p |
| 16:36.53 | OneFix_Work | What is the best softphone for use with Asterisk? I would like one that works on both Windows and Linux |
| 16:37.18 | *** join/#asterisk rrb3942 (~rbullock@208.49.79.66) |
| 16:38.49 | IsUp | i am using eyeBeam on Windows |
| 16:38.58 | IsUp | its my favorite |
| 16:39.22 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 16:39.59 | kalimc | If we have "national" for our switch type, and things are working... and the switch is actually "dms100"... would there be any problems? |
| 16:40.20 | IsUp | kalimc: if its working, dont touch to it :p |
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| 16:45.33 | kalimc | well, the company says DMS100 - and I have national... |
| 16:45.40 | kalimc | we have some fax problems here and there |
| 16:47.20 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:54.52 | z4nD4R | IsUp: i have next question... if 1001 call 1000 ( exten => 1000,n,Dial(SIP/1000,10) |
| 16:54.52 | z4nD4R | exten => 1000,n,VoiceMailMain(71000@skuska_voicemail) |
| 16:54.52 | z4nD4R | ) |
| 16:55.10 | z4nD4R | i am asking abou password... way? |
| 16:56.21 | IsUp | exten => 1000,n,VoiceMail(71000@skuska_voicemail) |
| 16:56.47 | IsUp | use VoiceMail, instead of VoiceMailMain |
| 16:56.54 | z4nD4R | ok w8 |
| 16:58.36 | z4nD4R | IsUp: but ... when i change.... - Incorrect password '1001' for user '71001' (context = default) |
| 17:01.18 | IsUp | you are trying to do "leave a message" stuff? |
| 17:01.49 | angryuser | Can someone confirm me i understood this SIP RFC phrase correctly please |
| 17:01.51 | angryuser | The initial Request-URI of the message SHOULD be set to the value of |
| 17:01.51 | angryuser | <PROTECTED> |
| 17:02.05 | z4nD4R | IsUp: .. fix it... thx :) |
| 17:02.16 | IsUp | z4nD4R, np |
| 17:02.23 | angryuser | This means initial INVITE has to mach TO tag in uri ? |
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| 17:41.51 | *** join/#asterisk anita_voip (7aa13952@gateway/web/freenode/ip.122.161.57.82) |
| 17:41.59 | anita_voip | Hi all |
| 17:42.03 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 17:42.15 | IsUp | hello anita_voip |
| 17:42.21 | anita_voip | Update is that my ss7 link is working with outbound calling |
| 17:42.21 | WIMPy | anita_voip: Lo you |
| 17:42.46 | anita_voip | i am using chan_ss7 |
| 17:43.04 | anita_voip | libss7 still giving the same iAM, ACM error |
| 17:43.12 | anita_voip | could there be a bug in libss7 ? |
| 17:43.17 | anita_voip | Hi IsUp |
| 17:43.17 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:43.28 | anita_voip | Hellp WIMPy |
| 17:43.30 | anita_voip | howdy |
| 17:44.00 | diemos | Wish I could namedrop for attention.... |
| 17:44.01 | IsUp | libss7 was so buggy, i am talking about 1 year ago, maybe its fixed now but i am using chan_ss7 too |
| 17:44.35 | anita_voip | yes IsUp, i have a bad feeling about libss7 |
| 17:44.51 | anita_voip | this was the latest version 1.0.2 |
| 17:45.03 | *** join/#asterisk golikwid|mac_ (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 17:45.05 | tf87 | Hi, can the digium TE220B be used to route the data connection in my T1? |
| 17:45.09 | diemos | anita_voip: Is there another process in which you can call upon libss7? Maybe you can force the problem through other means. |
| 17:45.30 | anita_voip | IsUp: do you have SS7 links working with chan_ss7 ? I mean today ? |
| 17:45.38 | IsUp | yes anita_voip |
| 17:46.16 | anita_voip | IsUp: Cool ! is your CDR working correctly ? mine always gives the status as NO ANSWER irrespective |
| 17:46.24 | anita_voip | this is for outgoing calls |
| 17:46.46 | IsUp | its working correctly as far i know |
| 17:46.57 | anita_voip | diemos: do you have libss7 working on sangoma ? |
| 17:46.57 | IsUp | because we are generating bill reports every month |
| 17:47.06 | anita_voip | IsUp: cool |
| 17:47.13 | IsUp | you got sangoma a104? |
| 17:47.23 | anita_voip | IsUp: you are using TDM calling or SIP ? |
| 17:47.30 | IsUp | TDM |
| 17:47.31 | WIMPy | anita_voip: Do the channels come up if you look at them with core show channels? |
| 17:47.32 | anita_voip | no, i got Sangoma A108 |
| 17:48.10 | anita_voip | WIMPy: about libss7 ? it says all inactive channels |
| 17:48.11 | diemos | anita_voip: I couldn't tell you, I have at any given time 3-5 different voip systems running and always setting up new ones. |
| 17:48.53 | WIMPy | anita_voip: The one that gives you wrong CDRs. |
| 17:49.26 | anita_voip | WIMPy: oh ok :) |
| 17:49.32 | anita_voip | WIMPy: yes it does |
| 17:49.43 | anita_voip | WIMPy: ss7 show channels gives more info though |
| 17:50.12 | IsUp | ss7 show channels its just for CIC information |
| 17:50.38 | WIMPy | wonders how that is possible then. |
| 17:52.29 | anita_voip | wait |
| 17:52.32 | anita_voip | this is what i got |
| 17:52.33 | anita_voip | debian*CLI> core show channels Channel Location State Application(Data) SS7/siuc/30 9415000417@ss7:1 Ringing (None) 1 active channel 0 active calls |
| 17:52.42 | anita_voip | while it is rining |
| 17:52.42 | anita_voip | and |
| 17:52.44 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 17:52.55 | anita_voip | debian*CLI> core show channels Channel Location State Application(Data) SS7/siuc/30 s@demo:5 Up BackGround(demo-congrats) 1 active channel 1 active call |
| 17:52.58 | anita_voip | while the call is on |
| 17:53.02 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 17:53.08 | anita_voip | and then it gives an error |
| 17:53.22 | anita_voip | -- SS7 hangup 'SS7/siuc/30' CIC=30 Cause=16 (state=7) [Jan 6 23:22:09] WARNING[13077]: func_strings.c:778 csv_quote: No argument specified! |
| 17:53.32 | *** part/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 17:53.37 | anita_voip | WARNING[13077]: func_strings.c:778 csv_quote: No argument specified! |
| 17:53.55 | WIMPy | Immediately after answering? |
| 17:54.00 | WIMPy | Or later? |
| 17:54.09 | anita_voip | later |
| 17:54.13 | anita_voip | when the call is finished |
| 17:54.25 | IsUp | maybe a problem with your cdr_*.conf files? |
| 17:54.30 | anita_voip | and in the CDR it shows NO ANSWER |
| 17:54.33 | anita_voip | oh ok |
| 17:54.41 | anita_voip | the conf files are all the deault mostly |
| 17:54.52 | IsUp | paste cdr.conf and cdr_custom.conf to pastebin |
| 17:54.57 | IsUp | also ss7.conf is useful too |
| 17:55.40 | anita_voip | Hey ! I am new to IRC :) and new to SS7, yesterday i was thinking wireshark is a support guy at sangoma |
| 17:55.46 | anita_voip | what is a pastebin ? |
| 17:56.05 | WIMPy | Looks fishy to me. How can a call that is definitely up be logged as NO ANSWER? I'd even exprect that to not be related to the channeltype. |
| 17:56.17 | WIMPy | ~pb |
| 17:56.17 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:56.25 | anita_voip | WIMPy: you are right |
| 17:57.25 | IsUp | who is your technician at Sangoma? Marc? |
| 17:57.39 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:57.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:58.12 | IsUp | wireshark is not needed ATM |
| 17:58.25 | IsUp | probably something wrong with your setup |
| 17:58.33 | IsUp | lets see your files first |
| 18:00.10 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 18:00.34 | anita_voip | i have put them on pastebin http://asterisk.pastey.net/144609-vrgj |
| 18:01.09 | anita_voip | IsUp: Yes, Marc took a look at my system with libss7 but he didn't have much experience with libss7 |
| 18:02.41 | IsUp | yeah he is better on chan_ss7, lemme check your files |
| 18:03.37 | anita_voip | IsUp: ok thanks |
| 18:04.16 | IsUp | can you try with 'unanswered = no' in cdr.conf? |
| 18:05.23 | anita_voip | sure, let me do it. But in that case will i not miss the unanswered calls altogether from my cdr ? |
| 18:05.38 | IsUp | lets try it first :> |
| 18:07.40 | IsUp | dont forget the restart your Asterisk |
| 18:07.48 | IsUp | if you are in test environment |
| 18:08.20 | anita_voip | IsUp: yes i did restart |
| 18:08.29 | anita_voip | funny thing happened ! no CDR !!! |
| 18:08.48 | anita_voip | because now there will be no entry with unanswered=no :) |
| 18:08.49 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 18:09.26 | IsUp | i dont think you need unanswered=no, i never used it |
| 18:09.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:09.38 | anita_voip | WIMPy: Hey, what are you doing here ? are you also stuck on something ? |
| 18:09.53 | IsUp | anita_voip: http://asterisk.pastey.net/144610 |
| 18:10.02 | anita_voip | IsUp: exactly |
| 18:10.04 | IsUp | you can see what is unanswered stands for |
| 18:10.40 | anita_voip | ok |
| 18:10.46 | WIMPy | anita_voip: Oh, tehre's a lot of things. |
| 18:11.13 | *** join/#asterisk doolph (~doolph@unaffiliated/doolph) |
| 18:11.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-011.mobile.uci.edu) |
| 18:12.10 | doolph | hi |
| 18:12.16 | IsUp | hello doolph |
| 18:13.10 | doolph | testing asterisk |
| 18:13.12 | doolph | :P~ |
| 18:13.19 | IsUp | anita_voip: any error on CLI? |
| 18:13.40 | anita_voip | IsUp: yes plenty of errors |
| 18:14.13 | IsUp | did u tested other channels? like SIP? IAX2? |
| 18:14.18 | anita_voip | but these are related to SS7 |
| 18:14.24 | anita_voip | not yet |
| 18:14.29 | anita_voip | only TDM over SS7 |
| 18:14.46 | anita_voip | yes the same thing |
| 18:14.50 | anita_voip | no CDR at all :) |
| 18:15.25 | anita_voip | by default unanswered=no, so it wont make any difference whether you have it or not |
| 18:16.07 | IsUp | i can take look into your system if you want |
| 18:16.18 | doolph | what codec binary is faster GCC, GCC4 or ICC |
| 18:16.44 | doolph | I am using ubuntu linux |
| 18:17.16 | Qwell | Use the compiler that comes with your system |
| 18:17.57 | IsUp | probably its gcc 4.x.x on Ubuntu |
| 18:18.08 | doolph | ok |
| 18:19.15 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 18:20.20 | *** part/#asterisk anita_voip (7aa13952@gateway/web/freenode/ip.122.161.57.82) |
| 18:21.55 | doolph | ack |
| 18:21.59 | doolph | the new xlite is so ugly |
| 18:25.21 | doolph | [Jan 6 18:24:30] NOTICE[1064]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.126' |
| 18:25.24 | doolph | ugh |
| 18:26.00 | IsUp | it happens to me too, i dont know why |
| 18:29.13 | tzanger | xlite was always ugly |
| 18:29.14 | tzanger | always |
| 18:30.18 | Qwell | who decided that softphones need to look like cellphones, anyways? |
| 18:30.21 | IsUp | can someone recommend me a hardphone? i was using grandstream before, i am planning to buy a phone for my home |
| 18:35.03 | Qwell | ~phones |
| 18:35.04 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 18:36.08 | IsUp | thank you Qwell and which one you are using? |
| 18:39.42 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
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| 18:49.07 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:58.43 | SuPrSluG | wow phone from bot lookin pretty dated |
| 18:58.44 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:59.15 | SuPrSluG | s/phone/phones/ |
| 19:07.07 | *** join/#asterisk jblack (~jblack@pool-71-181-222-126.sctnpa.east.verizon.net) |
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| 19:10.54 | OneFix_Work | PBXIAF or AsteriskNow? |
| 19:11.30 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 19:11.33 | Qwell | Depends. Do you want a distro that actually uses packages and has a record of stability? |
| 19:11.56 | OneFix_Work | Qwell: Well, I suppose so :) |
| 19:12.03 | Qwell | AsteriskNOW then :p |
| 19:12.58 | OneFix_Work | Qwell: What about the new one that I've been reading about...Elastix |
| 19:13.16 | Qwell | I know very little about Elastix. |
| 19:13.36 | Qwell | I've not heard much bad (or good, really) about it. I don't know how widely it's used. |
| 19:13.52 | Qwell | let me point out that I'm exceptionally biased towards AsteriskNOW, since I'm the author. |
| 19:14.38 | OneFix_Work | Qwell: I also noticed that a lot of the config options for TrixBox are kept in the MySQL database, and not in the /etc/asterisk config directory as suggested by Asterisk |
| 19:15.01 | pabelanger | pabelanger |
| 19:15.03 | Qwell | that's mostly a FreePBX thing |
| 19:15.09 | pabelanger | <PROTECTED> |
| 19:15.41 | OneFix_Work | Which seems like it would be more problematic when you go changing config options if you have to manually update the database every time |
| 19:16.20 | OneFix_Work | Does AsteriskNow use CentOS as a base OS? |
| 19:16.22 | Qwell | OneFix_Work: there are some like "override" config files that can be used |
| 19:16.23 | Qwell | yes |
| 19:16.35 | OneFix_Work | That's good |
| 19:18.05 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 19:24.30 | *** join/#asterisk errr_ (~errr@fedora/errr) |
| 19:30.51 | OneFix_Work | Qwell: After playing with TrixBox, I have come to the conclusion that AsteriskNow is probably the better option, if not simply because it's made by Digium |
| 19:45.55 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
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| 20:06.03 | Kobaz | so where's the next astricon |
| 20:06.05 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23.wls.msr03bkc3.adsl.static.edpnet.net) |
| 20:09.10 | russellb | Kobaz: it has not yet been decided. |
| 20:09.16 | russellb | in the US, though :-) |
| 20:09.46 | Kobaz | heh, cool |
| 20:09.51 | thehar | i demand to know |
| 20:09.57 | thehar | tell marketing to announce |
| 20:09.59 | Kobaz | thehar: it's in my basement |
| 20:10.07 | thehar | runs behind russellb |
| 20:10.07 | russellb | O.O |
| 20:10.08 | thehar | scary man! |
| 20:10.17 | Kobaz | i have space for one vendor and 1/2 of a talk |
| 20:10.19 | russellb | I know what places they are considering, but I'm not telling. |
| 20:10.22 | russellb | *tease* |
| 20:10.56 | thehar | please say california |
| 20:10.56 | thehar | please |
| 20:11.15 | russellb | !California |
| 20:11.17 | russellb | there, I said it |
| 20:11.33 | thehar | :| |
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| 20:15.42 | Katty | hmm |
| 20:15.48 | Katty | so asteriskaddons is built into 1.8 now? |
| 20:15.48 | NightMonkey | PBXIAF isn't stable? Comparative metrics, please. :) |
| 20:15.55 | Katty | i no longer need to set that up |
| 20:17.03 | russellb | Katty: yes. |
| 20:20.34 | *** join/#asterisk _Garry_ (~chatzilla@2001:4cd8:2:0:dcde:998d:ee44:2f26) |
| 20:20.38 | _Garry_ | Hi * |
| 20:21.00 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 20:22.01 | _Garry_ | Looking for some hint - is there some feature in Asterisk (agi w/ php) to continously play some sound while still reading dtmf? |
| 20:22.39 | Katty | russellb: fancy |
| 20:23.14 | OneFix_Work | Does AsteriskNow have Fesitval installed? |
| 20:25.27 | OneFix_Work | _Garry_: I'm pretty sure, since a lot of companies do it with their IVR systems |
| 20:26.25 | _Garry_ | to quote the wiki, "If the user presses a key while the message is playing, the message stops playing" (on Get Data) - I'd like to have the sound to keep on playing while the dtmf is collected ... |
| 20:27.36 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 20:29.25 | OneFix_Work | _Garry_: I assume that is done to recognize the tones better |
| 20:29.28 | OneFix_Work | ls -l |
| 20:29.30 | OneFix_Work | oops |
| 20:32.03 | _Garry_ | well, guess I can make do with it ending once the first digit is entered ... |
| 20:33.03 | n3hxs | Personally, I would think that there was a problem if my keypress didn't stop the sound and would likely press it again. |
| 20:33.24 | n3hxs | Just as I expect dialtone to stop at the same keypress. |
| 20:33.34 | n3hxs | But that is just me. |
| 20:33.52 | _Garry_ | well, the sound I was planning to play is more or less just a short "beep" or something with shorter intervals in between, until the actual time is up |
| 20:33.53 | OneFix_Work | _Garry_: Any reason you are wanting to do this? |
| 20:36.59 | OneFix_Work | _Garry_: I still don't understand why you would want to do that |
| 20:37.29 | _Garry_ | The user has "x" seconds to enter a code, just trying to add to the suspense / stress ... :) |
| 20:38.24 | OneFix_Work | _Garry_: The best way to do that is to play another recording if the input times out. |
| 20:38.46 | Kobaz | stopping a track is a pretty standard way to signify that it got a digit |
| 20:38.54 | Kobaz | it's pretty much *the* standard |
| 20:39.26 | _Garry_ | yeah, will do that anyway (as the timeout restarts after every digit), so if the input isn't finished in time, the caller loses ... |
| 20:54.02 | atan | When I try to pass a call to a SIP phone it says all busy/conge... yet sip show peer shows it as connectes & the whole bit. |
| 20:54.12 | atan | Does port 5060 on the phone end need to be open? =\ |
| 20:54.37 | p3nguin | That depends on the entire configuration. |
| 20:55.02 | atan | p3nguin, well when I had the device locally it worked fine. The server is remote. |
| 20:55.11 | OneFix_Work | Actually, Elastix looks like a lot of people are moving to it over AsteriskNow |
| 20:55.21 | atan | I dropped it off at the new location, same cable modem network and it won't ring goes to voicemail. |
| 20:55.21 | p3nguin | Does the phone register out to Asterisk? Is Asterisk configured with qualify for the phone peer? Did you set the nat setting accordingly for the phone peer? |
| 20:55.24 | *** join/#asterisk jetlag (jetlag@pool-173-61-245-217.cmdnnj.east.verizon.net) |
| 20:55.35 | atan | p3nguin, qualify=yes, nat=yes is in the sip.conf |
| 20:55.53 | p3nguin | But is it within the peer definition for the phone? |
| 20:55.57 | atan | The only thing different as far as I can tell is they have some strange no name router from like Routerz Inc. or somethingt. |
| 20:56.02 | p3nguin | And it Asterisk also behind a NAT? |
| 20:56.16 | atan | p3nguin, it's under the [4423] user, which is the phone who is connected |
| 20:56.19 | p3nguin | s/it/is/ |
| 20:56.35 | atan | Asterisk shouldn't be behind a nat, no. |
| 20:57.01 | p3nguin | Shouldn't? I don't care if it should or should not be... I want to know IS IT. |
| 20:57.01 | atan | Other phones of the same kind work fine on other networks without the no-name router. =\ I suppose I should drive over there and remove the router then eh? =\ |
| 20:57.12 | atan | Sorry, the box isn't behind a nat, no. |
| 20:57.23 | atan | Asterisk is sitting open to the world. *wimper* |
| 20:57.34 | p3nguin | It could be a jacked up router trying to fix SIP. |
| 20:57.46 | p3nguin | Start with sip debug. |
| 20:58.41 | atan | p3nguin, is that the same as -vvvvvvvvvvvvv? |
| 20:58.47 | p3nguin | no |
| 20:59.03 | atan | How do I 'sip debug'? =) |
| 20:59.19 | atan | err, set debug on |
| 20:59.22 | p3nguin | What you've described is the same as "core set verbose 13" |
| 20:59.47 | p3nguin | What you need to do is "sip set debug on" |
| 20:59.48 | Kobaz | does anything even use verbose 13? |
| 20:59.54 | p3nguin | no |
| 20:59.59 | Kobaz | the max i've seen in the code is like 3 |
| 21:00.07 | p3nguin | As far as I know, nothing is changed above verbose 4. |
| 21:00.32 | p3nguin | So core set verbose 4 gives the exact same output as core set verbose 100000000. |
| 21:00.33 | Kobaz | i would like to see a more granular debugging system at some point |
| 21:00.46 | Kobaz | add's that to the todo |
| 21:00.51 | p3nguin | add's? |
| 21:01.01 | p3nguin | Maybe you mean "adds." |
| 21:01.06 | Kobaz | perhaps |
| 21:01.10 | p3nguin | probably |
| 21:01.18 | Kobaz | you never know |
| 21:01.24 | p3nguin | I'm pretty sure, though. |
| 21:01.35 | Kobaz | you might be right |
| 21:01.44 | p3nguin | I'm very likely right. |
| 21:01.56 | Kobaz | i think you're close |
| 21:01.57 | atan | How do I set a sip debug for just one peer? |
| 21:02.18 | Kobaz | you have an extra '.' in there |
| 21:02.28 | russellb | atan: *CLI> sip set debug peer <peer> |
| 21:02.33 | p3nguin | "sip set debug <TAB>" should give you a hint. |
| 21:02.40 | russellb | my answer was more helpful :-p |
| 21:02.50 | OneFix_Work | BTW, is everyone using Cepstral with Asterisk just buying a license for their voices, or is there a version available for free that I'm not seeing? If so, why isn't everyone just using Festival + CMU Arctic voices? |
| 21:02.52 | atan | You both rock. |
| 21:03.27 | Kobaz | p3nguin: i would appreciate if you didn't complain about every little typo |
| 21:03.28 | russellb | Cepstral isn't free. People use it because it is generally high quality than Festival. |
| 21:03.38 | russellb | I'm not familiar with the festival voices you refer to, though. |
| 21:03.50 | russellb | If you can find a way to make festival produce a comparable result, by all means, go for it |
| 21:04.03 | Kobaz | Cepstral is pretty cool |
| 21:04.10 | OneFix_Work | russellb: But, the CMU Arctic voices are as good as or better than the Cepstral voices |
| 21:04.41 | *** join/#asterisk aut (~aut@c-174-48-60-12.hsd1.fl.comcast.net) |
| 21:04.57 | Kobaz | p3nguin: i'm fully aware of how to write proper sentences, but my fingers don't always hit the right keys, nor do i care most of the time when I'm typing quickly on irc |
| 21:05.14 | aut | could anyone give me suggestions on how to troubleshoot call quality issues? i use onsip hosted pbx, but im considering running a local asterisk install if it could solve our quality issues |
| 21:05.43 | Kobaz | aut: measure packet loss and jitter from your location to your hosted provider... use something like mtr |
| 21:05.50 | OneFix_Work | russellb: As a matter of fact, search for CMU Arctic on this page ... http://www.cepstral.com/publications/ |
| 21:06.34 | russellb | k, go for it |
| 21:06.49 | OneFix_Work | russellb: http://festvox.org/cmu_arctic/ <- this is their site ... |
| 21:07.01 | atan | http://pastie.org/private/potextiv7sgdztqfk590yq is the server log =\ |
| 21:07.05 | p3nguin | kobaz: That's the problem with society nowadays -- people don't care -- and it affects me in a way that makes me let people know. |
| 21:07.11 | OneFix_Work | russellb: Well, I already have it working, I was just interested in the reasoning... |
| 21:08.26 | Kobaz | p3nguin: i write emails with utmost care, irc with random other techies, I give it a best-effort. Chat and text messages to business associates get high levels of care as well |
| 21:08.29 | Kobaz | irc... not so much |
| 21:08.54 | russellb | on IRC you can be l33t l4wl |
| 21:08.58 | Kobaz | yeap |
| 21:09.41 | Kobaz | p3nguin: but i see where you're coming from... the kids these days... it's just bad |
| 21:11.45 | atan | p3nguin, you wouldnt happen to have any ideas what I setup wrong based on the pastebin would you? |
| 21:12.04 | p3nguin | atan: Was the paste a sip debug? |
| 21:13.28 | atan | p3nguin, that it was |
| 21:13.41 | p3nguin | atan: I'll help you out: No, it wasn't. |
| 21:13.54 | p3nguin | atan: It was a mutilated PARTIAL debug. |
| 21:14.20 | p3nguin | Neither of which I am interested in (not mutilated, nor partial). |
| 21:16.29 | atan | p3nguin, although I don't know you would you be open to me passing you a private link which isn't turnicated or masked? =) |
| 21:18.16 | p3nguin | There is really nothing secretive in a sip debug that can't be shown in this channel. If you don't want a lot of people seeing the debug, mark the paste as private so outsiders can't stumble upon it looking in the pastes. |
| 21:18.30 | p3nguin | I don't do the "super secret debug" crap. |
| 21:19.46 | atan | Well it's nothing overly secret I suppose. It's just I don't know much about it and would hate to paste some secret without knowing it. |
| 21:21.52 | p3nguin | There are no plain-text passwords in the debug. |
| 21:22.10 | atan | Looking at my debug there are plain-text mysql passwords being thrown around. Hmm. |
| 21:22.13 | p3nguin | And since you change passwords every 2-3 months anyway... |
| 21:22.46 | p3nguin | mysql passwords? in a debug? |
| 21:23.00 | atan | p3nguin, they're flying across my screen right now. |
| 21:23.01 | p3nguin | That's not very nice, if true. |
| 21:23.18 | atan | My dialplan queries a database on each call to lookup rates. |
| 21:23.42 | p3nguin | So your dialplan has passwords exposed. |
| 21:23.53 | p3nguin | That's a very rare case. |
| 21:24.48 | p3nguin | I'm still not sure how a sip debug would have THAT info in it. |
| 21:24.49 | *** join/#asterisk joobie (~joobie@CPE-124-176-179-1.lns3.win.bigpond.net.au) |
| 21:24.52 | atan | Holy monkeys. SIP debug is showing me bloody tons of data here. I have several peers using it. Can I limit my debug to just the phone that isn't ringing? |
| 21:25.21 | p3nguin | Oh, maybe you didn't bother turning the core verbose back down to 0 where it really should be. |
| 21:26.02 | p3nguin | Like I mentioned before, "sip set debug <TAB>" will give you a clue. |
| 21:26.23 | atan | p3nguin, I've figured out how to do it but I'm wondering how much of this data you actually want. |
| 21:27.41 | p3nguin | To find out what is going on for a specific peer, I would want the debug for the peer. If the peer has not registered, you probably cannot debug by peer name; you'll have to debug by IP address if the peer is not recognized. |
| 21:30.43 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.12) |
| 21:31.51 | atan | p3nguin, http://pastie.org/private/80zms4i8d0fpfh6f9idpq |
| 21:33.01 | p3nguin | Still just a partial. |
| 21:33.49 | atan | p3nguin, well I am confused then. I said sip set debug on peer 4423 and then called it. That is the only output. Where exactly did I go wrong? |
| 21:35.13 | Katty | pants |
| 21:36.20 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:36.35 | p3nguin | atan: Maybe I'm just expecting to see more than what is available. Tell me what is happening in this debug. |
| 21:37.01 | p3nguin | You call from SIP/what to extension something, which Dial()s SIP/something. |
| 21:37.17 | atan | Well I'm trying to call the phone and the thing doesn't even ring. Verbose 5 says it's busy |
| 21:37.51 | p3nguin | What are you calling from? |
| 21:37.55 | atan | Yep. It did once work fine without any changes when the phone was registered on my home network. I since moved it out to a friend's house who has some funny brand router, and now it won't ring. Outbound calls from it are fine, but I can not ring it. |
| 21:38.14 | atan | It's a Cisco IP phone, 7940 I believe. |
| 21:38.18 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.12) |
| 21:38.37 | p3nguin | From a Cisco... named SIP/what? |
| 21:38.53 | p3nguin | to what ... named SIP/4432? |
| 21:39.02 | p3nguin | 4423, I meant. |
| 21:39.07 | atan | The calling phone is SIP/7960 calling the SIP/4423 |
| 21:39.34 | atan | But it has a number in the dialplan to accept calls from voip.ms, which also just goes to voicemail |
| 21:39.51 | atan | The dialplan has Dial(SIP/4423, 25) in it, then voicemail on the next line. |
| 21:41.21 | atan | As far as I can tell when I look at the CLI it shows it trying to call the SIP/4423 without any issue, but it just comes back busy |
| 21:41.53 | atan | I'm really wondering about this router they have. Would I need to port forward 5060 to the phone for any reason? |
| 21:42.08 | p3nguin | Are you calling from the Cisco phone out to VoIP.ms and then back in to Asterisk hitting the extension that dials SIP/4423, or are you calling from the phone to an extension which dials SIP/4423? |
| 21:42.33 | atan | p3nguin, call it going out to voipms then back in to the dialplan to the phone |
| 21:42.42 | p3nguin | Typically you don't need to forward the ports because the NAT settings on Asterisk help keep the NAT open for calls to the phone. |
| 21:43.17 | p3nguin | Is there any reason why you can't call an extension that dials SIP/4423? |
| 21:43.21 | atan | It's a wireless N "airlink" router if that's any help. |
| 21:43.24 | p3nguin | Make this as simple as possible. |
| 21:43.28 | atan | p3nguin, nope. Let me add one in the dialplan! =) |
| 21:44.10 | p3nguin | It probably won't solve this issue, but at least it will take a bit of the complication out of it. |
| 21:46.00 | p3nguin | If the phone at SIP/4423 can make calls out to asterisk and beyond, the phone can communicate correctly and authenticate accordingly. Check "sip show peer 4423" and see if everything is good. I'm worried about the registration. |
| 21:47.12 | atan | Sip show peer 4423 shows it as connected, ip, whatnot |
| 21:47.34 | aut | how do small call centers (say 10 agents) deal with limits on inbound calls? do companies exist to queue the calls remotely so that every call gets answered regardless of the small office's bandwidth availability? |
| 21:47.51 | atan | err, phone call, one sec. Bloody =X =) |
| 21:48.05 | aut | does it make sense to colo the pbx at a local datacenter? |
| 21:51.32 | carrar | always makes sense to do that |
| 21:51.34 | p3nguin | aut: Most people get enough bandwidth and/or available channels to take the number of calls they need to take. |
| 21:52.16 | p3nguin | If they don't have enough channels to take calls, I guess a busy signal usually indicates to a person to call back soon. |
| 21:52.17 | carrar | or roll over your busrt of calls to some answering service |
| 21:52.22 | carrar | burst |
| 21:52.45 | carrar | or roll over to a message to call back later |
| 21:53.01 | carrar | typical of some gov services numbers |
| 21:53.14 | p3nguin | On an ITSP, you can usually configure things like that... but on a telco, do they do that stuff for you? |
| 21:53.30 | carrar | they can |
| 21:53.35 | carrar | everything is for a FEE |
| 21:53.45 | carrar | it's only $$ |
| 21:53.53 | p3nguin | Some things they just won't do, no matter how much you offer. |
| 21:54.05 | carrar | find a different provider |
| 21:54.29 | p3nguin | Sometimes possible, other times it's not. |
| 21:54.34 | brainiac | Does anyone know about why a call becomes irretrievable when parked to 701? You get a busy signal after you hit Transfer the 2nd time. I'm running 1.4.21 and it happens sporadically. |
| 21:54.51 | carrar | I'm on 701 |
| 21:54.55 | carrar | thats why |
| 21:55.20 | p3nguin | :) |
| 21:55.29 | russellb | brainiac: 1.4.21 is pretty old. |
| 21:56.27 | russellb | We have made 1693 changes to Asterisk 1.4 since the release of 1.4.21. |
| 21:56.36 | brainiac | Yeah, I know but I can only perform an upgrade to 1.4.31 |
| 21:56.39 | russellb | One of them could have been a fix for that bug. :-) |
| 21:56.47 | OneFix_Work | Actually, Elastix includes the CMU Arctic voices by default |
| 21:56.50 | aut | how dumb is it to use hosted pbx services like onsip vs colocating your own box locally? (specifically with regards to jitter/latency/quality issues, not maintenance workload) |
| 21:56.51 | carrar | 1693!! |
| 21:57.06 | russellb | carrar: yup. |
| 21:57.11 | carrar | not 1694? |
| 21:57.24 | russellb | not yet. |
| 21:57.36 | carrar | Thats Awesome |
| 21:57.42 | OneFix_Work | carrar: they know that because they are using a change management system |
| 21:58.08 | russellb | another fun fact ... over 10,000 changes to the development tree that became asterisk 1.8 since we created 1.4 |
| 21:58.11 | carrar | Mt Etna erupts, in Sicily in 1693 |
| 21:59.07 | carrar | Dom Perignon invents champagne in 1693 |
| 22:01.32 | leifmadsen | russellb: yay! |
| 22:02.26 | carrar | Where can one purchase the game of Asterisk Trivia! |
| 22:02.57 | carrar | That should be in the BOOK |
| 22:03.27 | carrar | Russell's Fun Facts and Trivia Chapter |
| 22:05.11 | *** join/#asterisk SuPrSluG__ (~SuPrSluG@8.22.96.106) |
| 22:06.22 | russellb | carrar: heh |
| 22:06.29 | russellb | Appendix D: Fun Facts |
| 22:07.50 | russellb | the data is all public ... you just have to know how to mine it! |
| 22:12.30 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 22:12.35 | *** join/#asterisk bjhaid (~IceChat7@41.206.12.9.vgccl.net) |
| 22:15.59 | *** join/#asterisk Da-Geek (~Da-Geek@2001:8b0:ff85:0:223:6cff:fe87:e49c) |
| 22:19.34 | raden_work | russellb, 10,000 changes since 1.4 ? |
| 22:19.43 | russellb | yup |
| 22:19.47 | raden_work | wow |
| 22:19.57 | leifmadsen | wow indeed |
| 22:19.59 | raden_work | how many people contribute to all this insanity ? |
| 22:20.10 | raden_work | just amazes me what can be accomplished when people work together |
| 22:20.11 | russellb | i counted by getting the revision when 1.4 was branched off, then I counted how many changes were made to trunk since then |
| 22:20.22 | russellb | it was just over 11,000 i think |
| 22:20.30 | raden_work | wow |
| 22:20.40 | raden_work | how amny people actually develop asterisk ? |
| 22:20.44 | russellb | how many people? ever? |
| 22:20.58 | russellb | I counted how many people contributed to 1.8 (since the release of 1.6.2) |
| 22:21.03 | russellb | where was that number ... in a tweet I thin |
| 22:21.06 | raden_work | how many is there |
| 22:22.14 | Katty | hellllllllllllooooo nurse! |
| 22:22.15 | russellb | http://twitter.com/#!/russellbryant/status/28031995158 |
| 22:22.20 | Katty | and russell dear |
| 22:22.21 | thehar | twatter |
| 22:22.22 | Katty | hello to you too |
| 22:22.28 | Katty | ooh and thehar |
| 22:22.32 | thehar | hellos! |
| 22:22.34 | Katty | hugs thehar |
| 22:22.38 | thehar | hugs Katty |
| 22:22.44 | Katty | thehar: how're you dear |
| 22:22.54 | russellb | so, > 500 reporters, > 300 testers, > 200 programmers in just over a year |
| 22:22.54 | thehar | sooooo busy |
| 22:22.58 | leifmadsen | damn yo :) |
| 22:23.08 | russellb | those that I could count (that sent something back to us) |
| 22:23.13 | Katty | mister leif (= |
| 22:23.18 | Katty | how's the lady friend. |
| 22:23.19 | thehar | wow russellb |
| 22:23.23 | russellb | :-) |
| 22:23.26 | russellb | Asterisk has grown up. |
| 22:23.36 | thehar | you've grown up! |
| 22:23.39 | *** join/#asterisk RypPn (~TuMbL@unaffiliated/ryppn) |
| 22:23.41 | Katty | russellb: and the developers have not ;P |
| 22:23.42 | russellb | thehar: hehe, true |
| 22:23.47 | russellb | Katty: lies! |
| 22:23.50 | Katty | *hee* |
| 22:23.51 | leifmadsen | ohai! |
| 22:23.53 | leifmadsen | she is well :) |
| 22:23.59 | leifmadsen | runs off to the mailbox to see if he got any cheques |
| 22:23.59 | Katty | leifmadsen: most excellent. |
| 22:24.08 | russellb | leifmadsen: learn how to spell |
| 22:24.12 | russellb | you call yourself an author ... |
| 22:24.17 | Katty | ahhh i remember the first cluecon |
| 22:24.21 | Katty | it was such a ...fun disaster |
| 22:24.33 | aut | so i'm using mtr to measure jitter. what's acceptable? my route appears to be comcast -> level3 -> arbinet |
| 22:24.36 | leifmadsen | I remember speaking at the first astricon in 2004 :) |
| 22:24.44 | leifmadsen | I remember when there was no documentation! ;) |
| 22:24.46 | aut | one of the level3 hops shows a max jitter of 249 |
| 22:24.47 | Katty | leifmadsen: i had /just/ graduated high school |
| 22:24.56 | russellb | me too |
| 22:25.02 | russellb | but got commit access anyway somehow |
| 22:25.02 | *** join/#asterisk bjhaid (~IceChat7@41.220.69.11) |
| 22:25.19 | Katty | hmm. cracker snack pack sitting in front of me. |
| 22:25.27 | Katty | not really hungry, but it looks so yummy |
| 22:25.29 | NightMonkey | aut: Acceptable for what purpose? |
| 22:25.51 | aut | nightmonkey: having a decent-sounding voip call? :) |
| 22:25.56 | Katty | puts crackers out of sight |
| 22:26.06 | aut | nightmonkey: and then, having several of those... :) |
| 22:26.26 | carrar | puts on his cracka XRAY glasses |
| 22:26.31 | NightMonkey | aut: Right. So, you need to be less general for us to help you. |
| 22:26.50 | Katty | carrar: are you disrespectin me |
| 22:26.54 | aut | nightmonkey: okay, in what way? |
| 22:27.09 | Katty | carrar: dpm |
| 22:27.17 | Katty | carrar: don't you EVEN be goin there hunny |
| 22:27.42 | NightMonkey | aut: What protocols are you using? What are you connecting across your links? |
| 22:27.51 | NightMonkey | aut: Asterisk can do LOTS of different things. |
| 22:27.55 | carrar | heh |
| 22:28.08 | Katty | Asterisk can do lots of fun things too. |
| 22:28.10 | NightMonkey | aut: And VoIP encompases LOTS of different components. |
| 22:28.14 | Katty | Like dial carrar repeatedly, over and over and over |
| 22:28.16 | Katty | and over. |
| 22:28.43 | Katty | sadly asterisk also does blacklisting )= |
| 22:29.56 | carrar | exten => carrar/katty,1,Hangup |
| 22:30.04 | carrar | err |
| 22:30.23 | carrar | exten => carrar/katty,1,Playback(Please_Hold_I_Will_Be_Right_There) |
| 22:30.57 | Katty | :< |
| 22:31.07 | Katty | how dare you hangup on me sir! |
| 22:31.30 | Katty | that's IT |
| 22:31.34 | Katty | you are going to the 404 context |
| 22:31.47 | *** join/#asterisk joobie (~joobie@CPE-124-176-179-1.lns3.win.bigpond.net.au) |
| 22:31.51 | carrar | WOAH! |
| 22:31.55 | carrar | Not the 404!! |
| 22:32.03 | Katty | teehee |
| 22:33.50 | raden_work | anyone use Twinkle ? |
| 22:34.20 | carrar | I thought that was something you take |
| 22:35.47 | Katty | i read that was Twinkie |
| 22:35.56 | Katty | i swear i'm not hungry |
| 22:36.44 | atan | p3nguin, are you still around here buddy? |
| 22:37.00 | atan | p3nguin, sorry about dropping off like that the phone caught me off guard. Blah! |
| 22:37.17 | Katty | hi atan |
| 22:37.40 | atan | Hey Katty |
| 22:37.49 | atan | Help me fix my peer who won't accept calls? =) |
| 22:41.40 | carrar | Did you read the BOOK? |
| 22:42.22 | carrar | -=>#*"!" The Book "!"*#<=- |
| 22:42.33 | thehar | The Book of Destiny |
| 22:43.10 | carrar | The Internet HitchHikers Guide to Asterisk and Fun Facts |
| 22:43.39 | Katty | The Book of Pretty Girl Pictures |
| 22:44.10 | carrar | Thats http://www.japansugoi.com/ |
| 22:49.25 | raden_work | is there a Linux SIP phone that worth a shit ? |
| 22:50.15 | OneFix_Work | raden: I think Ekiga is about the best one |
| 22:51.22 | OneFix_Work | s/raden/raden_work |
| 22:51.28 | raden_work | OneFix_Work, that thing Sucks |
| 22:51.32 | raden_work | twinkle is a little better |
| 22:51.37 | OneFix_Work | raden: You think? |
| 22:51.41 | raden_work | X-lite is sweet but only ubuntu version |
| 22:51.52 | raden_work | OneFix_Work, more features better UI |
| 22:52.01 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:53.15 | OneFix_Work | raden_work: But, I don't wanna pay for a softphone and X-Lite serves ads to me |
| 22:53.22 | NightMonkey | raden_work: Worth a shit? No. But you can write your own better one, after you're done slagging existing ones. |
| 22:53.54 | raden_work | NightMonkey, I might just do that |
| 22:54.15 | NightMonkey | sets his stopwatch. |
| 22:55.56 | raden_work | NightMonkey, It might be a while |
| 22:56.17 | raden_work | NightMonkey, I mean thoose ones work it just no BLF etc for a office enviroment |
| 23:00.06 | NightMonkey | raden_work: F/OSS only works when people dedicate time and work to make it better. If no one from your "office environment" helps, well, your "office environment" gets no help. |
| 23:00.21 | raden_work | very truee |
| 23:00.39 | *** join/#asterisk leejohn (ljohn@58.69.35.120) |
| 23:02.00 | NightMonkey | raden_work: One bad move it to slag the work of other F/OSS volunteers/coders. Makes for an awkward relationship going forward.. |
| 23:02.03 | leejohn | hi! good day guys, i just want to ask Does Queue(quename,tT,,,60) is compatible with res_feature (blind transfer) ?, please bare with me if this is a simple one, but i can't figure it out, normal transfer works |
| 23:02.28 | raden_work | NightMonkey, how on each does a person even get involved in a project like this ? |
| 23:02.33 | leejohn | by the way Asterisk SVN-branch-1.8-r300798 built |
| 23:03.18 | NightMonkey | raden_work: Like this? Or writing a new SIP client? |
| 23:03.48 | raden_work | Like writing a SIP client |
| 23:04.31 | NightMonkey | raden_work: Look up the RFCs, pick your interface (GUI, text, etc.), determine your design, and start coding. |
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