| 00:01.06 | ChannelZ | yes |
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| 00:35.39 | lordvadr | In chan_dahdi.conf, how to I specify the D channel for a PRI? I see a dchannel for trunkgroups, but not in channels. |
| 00:36.05 | lordvadr | My guess is that it's dchannel=x but I don't see that in the comments anywhere |
| 00:42.34 | tzafrir_laptop | lordvadr, just don't touch that. You normally don't need to have trunkgroup settings |
| 00:43.03 | tzafrir_laptop | lordvadr, unless you tie together several spans |
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| 00:44.52 | lordvadr | txafrir: I'm aware of that. This is more academic. I have a 16 channel PRI to a cisco box (limited due to the number of DSP's in the cisco) and I want to change the D channel. How do I do that? |
| 00:45.14 | lordvadr | tzafrir_laptop: ... |
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| 00:46.36 | tzafrir_laptop | trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...] |
| 00:46.54 | tzafrir_laptop | lordvadr, see the sample chan_dahdi.conf |
| 00:47.48 | tzafrir_laptop | But I figure that the D channel would still be the same time slot even in your case |
| 00:48.38 | lordvadr | I saw that, but I'm running a production PRI and I never set neither a trunk group, or a span map. Can I not do that at the channel level? Or does it assume a "trunkgroup => 0,24"? And at that point, what's the point of group=0 under the channel definition? |
| 00:50.12 | lordvadr | I dial using TRUNK=g0 in my extensions.conf |
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| 00:52.06 | lordvadr | tzafrir_laptop: ... |
| 00:52.52 | tzafrir_laptop | lordvadr, 'trunkgroup' and 'group' are completely unrelated |
| 00:53.50 | lordvadr | tzafrir_laptop: I would imagine so, but the question still remains, how do I set the D channel outside of a trunk group? |
| 00:54.14 | lordvadr | or do I have to put it in a trunk group to do that? |
| 00:54.45 | tzafrir_laptop | Again, why would you need to set the D channel? |
| 00:54.52 | tzafrir_laptop | Is this T1 or E1? |
| 00:56.06 | lordvadr | T1, but again, say I wanted to, how would I do that? |
| 00:58.42 | tzafrir_laptop | lordvadr, before actually making calls, make sure that the span is up |
| 00:58.44 | lordvadr | tzafrir_laptop: This is mostly a curiosity/documentation question. On a cisco, you can specify a PRI as "pri-group timeslots 1-5,7,9,21" which means it's a 7 channel pri using 1-5, 7, and 9 with D channe at 21. I'm looking for the asterisk equivalent, mostly as a function of writing an interface. |
| 00:59.07 | tzafrir_laptop | Both sides have to actually communicat ethrough the D channel to do that |
| 00:59.13 | lordvadr | I know. |
| 00:59.34 | tzafrir_laptop | also, have you restarted dahdi (at least 'dahdi restart' ) to apply the changes? |
| 01:00.21 | lordvadr | Everything works just fine. I have a dozen PRI's, but simply as an academic question, if I had a customer that wanted a PRI with D-channel at 11, how could I pull it off with asterisk? |
| 01:00.50 | lordvadr | I know how to do it in Toshiba, Cisco, and Shortel. I'm looking for the asterisk equivalent. |
| 01:01.42 | tzafrir_laptop | trunkgroup, basically |
| 01:01.47 | tzafrir_laptop | I suppose |
| 01:02.19 | tzafrir_laptop | You'll still need to configure the (b)channels as usual. |
| 01:02.23 | tzafrir_laptop | Anyway, I'm off |
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| 01:05.50 | lordvadr | thanks for the help |
| 01:05.54 | lordvadr | have a good evening |
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| 01:16.07 | tzafrir_laptop | Evening? I wish |
| 01:17.54 | ChannelZ | Good? I wish |
| 01:20.01 | florz | have a? I wish |
| 01:23.11 | ChannelZ | Stupid collection agencies have the wrong number |
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| 02:23.40 | ruben23 | hi are there any affordable asterisk training in the philippines.. |
| 02:25.37 | coppice | affordable trainin == read the book :-) |
| 02:26.20 | ruben23 | coppice::-D yeah... |
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| 02:45.47 | devdvd | hey all, im taking a poll to choose what new ip phones i should get for my office. What ip phones is everyone using and what do you think about them? |
| 02:46.55 | [TK]D-Fender | Polycom > All |
| 02:47.06 | devdvd | thats kinda my thought as well, what model you using? |
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| 02:47.48 | [TK]D-Fender | devdvd: Many. 301,320,335,430, 501, 600,601, |
| 02:48.15 | devdvd | nice |
| 02:48.33 | devdvd | have you heard anything about the 670? |
| 02:48.42 | devdvd | as far as quality, reliability, etc |
| 02:49.34 | [TK]D-Fender | devdvd: Just like any other barring the obvious difference in features. Frankly I don't see a point for this. Overkill |
| 02:50.05 | [TK]D-Fender | Polcyom is tops on quality and the same on stability. |
| 02:51.25 | [TK]D-Fender | devdvd: Why are you looking at the 670 in particular? |
| 02:52.19 | devdvd | my boss likes pretty things? |
| 02:53.01 | [TK]D-Fender | devdvd: sad waste |
| 02:53.03 | devdvd | no real reason, just from reading that looks like the top of the soundpoint line right now. |
| 02:53.41 | [TK]D-Fender | devdvd: It is... but still unless you have something special to take advantage of the difference between in and lower models its a huge price hike to swallow |
| 02:53.48 | devdvd | yea |
| 02:54.30 | devdvd | honestly id be happy with a 301, ive used those, the sound quality rocks, dont look like you can purchse those retail anymore (at least it's not listed on polycom's site) |
| 02:56.00 | [TK]D-Fender | 30X is Disco'd, as are 50X, 60X. |
| 02:56.28 | [TK]D-Fender | devdvd: All were good, but when looking new those old models aren't what you should aim for. |
| 02:56.36 | devdvd | yea |
| 02:56.51 | devdvd | reading up on a 321 atm |
| 02:57.12 | [TK]D-Fender | devdvd: 321 is the standard entry model where you don't need pass-through |
| 02:57.27 | [TK]D-Fender | devdvd: this is 90% of users |
| 02:57.32 | [TK]D-Fender | (IMO) |
| 02:57.51 | devdvd | yea |
| 02:58.36 | devdvd | wait..forgive me for sounding stupid here..your refering to pstn pass-through right? |
| 02:59.21 | [TK]D-Fender | devdvd: No, none of these do that. I'm referring to a switch network port |
| 02:59.35 | [TK]D-Fender | devto plug a computer in-line |
| 02:59.59 | devdvd | oh oh |
| 02:59.59 | devdvd | no |
| 03:00.04 | devdvd | none of that dumb stuff |
| 03:03.02 | [TK]D-Fender | devdvd: Will your users be on headsets? |
| 03:03.10 | devdvd | that could be useful |
| 03:03.34 | devdvd | they aren't currently but its one of the things they have been wanting |
| 03:04.05 | [TK]D-Fender | devdvd: What kind of call volume and usage? |
| 03:04.49 | devdvd | pretty low, each phone will be lucky to see 30 calls / day |
| 03:05.36 | [TK]D-Fender | devdvd: it's looking like 321 is your standard then. |
| 03:06.10 | [TK]D-Fender | devdvd: Are they in poor lighting or frequently needing to read the display? |
| 03:07.27 | devdvd | no, the only thing they would need the display for is to see the CID and maybe a phone book if i choose to put one in there, its a pretty small setup and internally we communicate through IM anyway |
| 03:15.59 | [TK]D-Fender | devdvd: 321 is really looking like it. If you're in poor light, the 335 is a nice phone. I used to use a 600 at my desk. I gave up the big-screen, extra contextual keys for the IP 335 mostly for the backlight. Don't feel the loss much. |
| 03:16.52 | devdvd | cool. I'll submit the options let my boss decide how much he wants to spend. but it looks like polycom is it |
| 03:19.07 | [TK]D-Fender | devdvd: Unless you have rather special needs, Polycom wins hands down in North America |
| 03:22.03 | devdvd | yea, that was kinda my thought as well before i came in tonight...but i wanted to ask just to make sure that was still the standard..boss was asking about cisco phones... |
| 03:22.14 | devdvd | flat told him not worth the price |
| 03:24.04 | [TK]D-Fender | devdvd: And the trouble. Licensing, the fact that Polycom beats the on updates functionality and stability as well |
| 03:24.36 | [TK]D-Fender | devdvd: Only reason to use Cisco is if you find a really great deal and have people to help you with the rest. |
| 03:25.16 | [TK]D-Fender | devdvd: Also there are several lower models whose PoE is not the current 802.3af standard and thus a larger problem, don't support SIP, etc |
| 03:25.31 | devdvd | yea |
| 03:25.36 | [TK]D-Fender | devdvd: 7960/7940 for instance are Cisco PoE. |
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| 03:28.24 | [TK]D-Fender | devdvd: Cisco for home use where you get a great deal, sure. But I wouldn't push it for a company |
| 03:28.35 | devdvd | yea, i never was a huge fan of the cisco phones |
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| 03:29.08 | devdvd | eventhough my first ip phone was a cisco 20 :) |
| 03:29.35 | devdvd | i learned the long hard way that sccp/skinny kinda sucks |
| 03:30.01 | devdvd | anyway |
| 03:30.28 | devdvd | [TK]D-Fender: as always, thank you. Everytime i come in here with my dumbass questions (and yes, i realize some of them are pretty dumb. you have always helped and not given me crap. |
| 03:35.18 | [TK]D-Fender | devdvd: I.... haven't? Don't worry, I'll get caught up soon no doubt ;) |
| 03:36.11 | [TK]D-Fender | devdvd: I tend to dish when people are twits for extended periods of time. Those that don't understand where they are, how to get help, follow instructions, etc. |
| 03:36.12 | devdvd | lol |
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| 03:37.16 | spenguin[work] | test |
| 03:37.19 | devdvd | usually my issues aren't so much lazyness (at least i dont think they are) its more so that i dont know how to word what i want and need someone to grab me by the head and go this way dumbass |
| 03:38.24 | [TK]D-Fender | devdvd: Well.. you don't seem retarded, so you've got decent odds :) |
| 03:39.01 | devdvd | no, thankfully i got the brains from my fathers family :) |
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| 03:52.20 | ruben23 | hi nayone used and install asterisk with opensuse 11.2, i cant compile dahdi and also tried zaptel, got error on compilation process, any help.. |
| 03:54.33 | devdvd | the first step would be to tell us what error you are getting ruben23 |
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| 03:57.34 | ChannelZ | I suggest you fix the problem |
| 03:58.05 | [TK]D-Fender | suggests going for the salad |
| 03:58.12 | ruben23 | devdvd: it seems i resovle it with the current version of dahdi, i compile..i tink its with the kernel-pae with my opensuse 11.2 |
| 04:02.13 | [TK]D-Fender | Thats the problem. Too much unqualified "thought". |
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| 04:42.33 | doolittlework | hi there anyone active |
| 04:43.22 | doolittlework | i have a e1 dual, primary d channels keeps going up and down on span 1, what might cause this? |
| 05:01.54 | doug | wrong signaling protocol |
| 05:01.57 | doug | on either end |
| 05:02.13 | doug | bad config on the e1's in the first place, like wrong encoding |
| 05:02.27 | doug | or maybe jesus hates you |
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| 05:11.35 | doolittlework | lol |
| 05:11.38 | doolittlework | douglas |
| 05:11.53 | doolittlework | down the "lane" |
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| 05:27.04 | proserion | good morning from GErmany |
| 05:27.21 | ChannelZ | DA! |
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| 05:28.12 | ChannelZ | or wait that's russian |
| 05:28.33 | p3nguin | Guten Morgen. |
| 05:28.43 | p3nguin | aus den Vereinigten Staaten. |
| 05:28.52 | ChannelZ | Fahrvergnugen |
| 05:29.01 | p3nguin | Volkswaagen! |
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| 05:29.49 | proserion | I was wondering, I am on the verge to install Asterisk, but need counselling, which IP service provider I should use. We use it for commercial (obviously) and would like to have a lokal number (in Germany of course) to call in. Any suggestions? |
| 05:30.08 | p3nguin | ~itsp-germany |
| 05:30.56 | proserion | chekking |
| 05:30.56 | p3nguin | infobot fails us again. |
| 05:31.07 | proserion | Ah ~ is the bot's key, I suppose |
| 05:31.22 | p3nguin | yes |
| 05:31.30 | p3nguin | such as: |
| 05:31.31 | p3nguin | ~itsp |
| 05:31.32 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 05:31.57 | p3nguin | ~itsplist-germany |
| 05:32.04 | proserion | ~itsp-de |
| 05:32.14 | proserion | ~itsp-ger |
| 05:32.19 | proserion | hmm |
| 05:32.26 | p3nguin | I guess no one ever told it any ITSPs in Germany. |
| 05:32.42 | proserion | <PROTECTED> |
| 05:32.47 | proserion | <PROTECTED> |
| 05:32.52 | ChannelZ | Try Google |
| 05:32.57 | proserion | sorry for spamming, gotta try |
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| 06:20.24 | F_Madison | ~itsplist-us |
| 06:20.47 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 07:16.01 | DND | hi guys may i know which brand of ip phones that has arounf $100 to $150 price range? aside from snom |
| 07:16.35 | DND | *around |
| 07:20.02 | jblack | You might be able to find a couple polycoms in that range. |
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| 07:28.11 | ChannelZ | is happy enough with his Linksys SPA 9xx phones |
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| 07:59.37 | DND | guys need help. i get weird errors like this: chan_dahdi.c: !! Got S-frame while link down |
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| 08:02.16 | WIMPy | So why is the link down in the first place? |
| 08:02.52 | DND | <PROTECTED> |
| 08:02.59 | DND | then we can dial again |
| 08:03.09 | DND | happened so fast |
| 08:03.32 | DND | also showing up "Junk at the beginning of frame" |
| 08:04.35 | WIMPy | Let me have a guess: The problem lasted for about seven seconds? |
| 08:05.19 | DND | the S-Frame error? |
| 08:05.30 | DND | hmm probably... |
| 08:05.35 | DND | a couple of seconds |
| 08:05.50 | DND | do you know the problem? |
| 08:05.52 | WIMPy | You've got a PRI thats delivered over DSL? |
| 08:05.58 | DND | yes |
| 08:06.08 | DND | oh sorry |
| 08:06.17 | DND | hmm its isdn |
| 08:07.09 | WIMPy | Jes, but the physical line is over DSL? |
| 08:07.25 | DND | yes |
| 08:08.06 | WIMPy | Get used to it or tell your telco you want a classic line. |
| 08:09.16 | WIMPy | The DSLAMs are sometimes being reset. That will make your line unavailable for about 7s and kill all active connections. A real PITA :-( |
| 08:10.34 | DND | i see. |
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| 08:11.20 | DND | so the S-Frame problem is connected to this |
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| 08:11.59 | WIMPy | Probably just a small glitch while trying te re-activate the line. |
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| 08:12.49 | WIMPy | Usually you get a short red alert during that time, but that depends on the modem. |
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| 08:14.17 | DND | im just checking on my modem |
| 08:14.20 | DND | its elcon |
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| 08:17.03 | DND | WIMPy, i have elcon SDSL-NT2M |
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| 08:18.11 | WIMPy | Don't know that thing, but if you don't get a red alert that might be the normal procedure in such a case. |
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| 08:20.47 | DND | is sdsl same as dslam? |
| 08:21.17 | DND | or sdsl part of dslam or vice versa? |
| 08:21.48 | DND | also what kind of classic line are we talking about? |
| 08:21.54 | WIMPy | The DSL Access Multiplexer is (at) the other end of the line. |
| 08:22.35 | WIMPy | The have been differen line types, before someone had the idea that DSL might be used. |
| 08:23.42 | WIMPy | Hasn't been of concern for customers so far. |
| 08:26.03 | DND | hmm i see. ok now time to fix the iax issue |
| 08:26.18 | DND | keeps disconnecting via vpn |
| 08:27.03 | DND | is it reliable to use vpn or we should do it via normal http connection? |
| 08:27.26 | WIMPy | http??? |
| 08:27.52 | WIMPy | is using dundi and IAX2 over openvpn without any issues. |
| 08:28.34 | DND | hmm |
| 08:28.41 | DND | need to think what's the prob |
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| 09:02.38 | stix | is Skype for SIP Open Beta for Asterisk users? |
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| 09:23.50 | drmessano | stix: It's for any application using SIP |
| 09:24.03 | ChannelZ | stix: Skype for SIP is by Skype, not digium |
| 09:24.35 | stix | yes, but what should I use then? Digium Skype module or skype for SIP? |
| 09:24.47 | drmessano | Though, it's pretty dumb compared to SkypeForAsterisk.. Doesn't have any of the integration |
| 09:26.01 | stix | okay, but it's still beta. Maybe it will replace skypeforasterisk some day |
| 09:26.06 | *** part/#asterisk memph (~memph@sd1438.sivit.org) |
| 09:26.10 | drmessano | Ewww |
| 09:26.24 | drmessano | No, it will never replace SFA |
| 09:26.29 | drmessano | Why would you even want it to ? |
| 09:26.45 | stix | I don't want to, I just want to know which one to use |
| 09:26.59 | drmessano | SkypeForAsterisk |
| 09:27.26 | stix | okay |
| 09:27.28 | drmessano | You didn't ask which one to use.. You asked if Skype for SIP was open for Asterisk users |
| 09:28.12 | drmessano | Of course it is.. But do you want a dumb SIP gateway to Skype or a native Skype connector that not only speaks Skype's protocol, but has buddy list integration, etc? |
| 09:28.15 | drmessano | I chose 2 |
| 09:28.27 | stix | yes, because Asterisk is not here: http://www.skype.com/intl/en-us/business/sip/pbx-solutions/ |
| 09:32.46 | drmessano | If I was Digium or Skype, I wouldn't be promoting it as a solution, because a REAL one exists. |
| 09:34.14 | drmessano | That's a bit like setting up a MS Exchange server and setting up all your Outlook clients with SMTP/POP3 |
| 09:34.21 | drmessano | Defeats the purpose entirely |
| 09:38.44 | ChannelZ | plus Skype for SIP will cost per month and I don't think they've said what the price will be after the beta |
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| 10:26.39 | MAbbas | Hi All, how can I do Consult transfer in asterisk? |
| 10:30.29 | MAbbas | For blind transfer I am using Dial() |
| 10:30.54 | MAbbas | But I have no idea about consult transfer? |
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| 10:40.54 | DND | what's consult transfer? |
| 10:41.02 | DND | attended transfer? |
| 10:41.23 | DND | you ask first the permission of the user before transferring? |
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| 10:49.09 | residentgrey | apparently 'blind' and 'consult' are synonymous dnd |
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| 10:51.25 | residentgrey | ok i am confused I an getting two different definitions here |
| 10:52.01 | residentgrey | http://www.uccx.net/scripting/call-consult-transfer-external-numbers.html refers it to supervised transfer |
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| 10:58.24 | shamelessn00b | Hello |
| 10:59.33 | shamelessn00b | I have 60 dahdi channels on my asterisk box, is it possible to tell asterisk to dial through specific channel ranges, lets say channel 16-30 and 45-60 |
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| 11:16.03 | seba11223 | hi all |
| 11:16.26 | DND | halow |
| 11:16.29 | seba11223 | does anyone happen to know an up to date documentation for various applications? |
| 11:16.39 | seba11223 | I'm currently loooking into Queue |
| 11:16.46 | seba11223 | http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 11:17.04 | seba11223 | I don't know what macro, gosub and rule params are |
| 11:17.15 | seba11223 | thank you |
| 11:19.50 | WIMPy | core show application|function whateva |
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| 11:22.45 | seba11223 | thank you |
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| 11:58.37 | artur__ | hello everybody. I use trixbox software. Let's assume user 1 calls to user 2, is there any way to check if the user 2 hang up his phone (better through php-api)? |
| 11:59.20 | artur__ | The idea is if user did not hang the phone, then server sends e-mail notification "missed call". |
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| 12:09.30 | artur__ | or through asterisk |
| 12:10.40 | tzafrir_laptop | artur__, send the call to voicemail |
| 12:10.58 | tzafrir_laptop | send voicemail messages to users |
| 12:11.24 | tzafrir_laptop | Or send the call to a custom extension that does System(something) |
| 12:14.33 | artur__ | yes, i know about voicemail - it is recorded message. Our users don't usually use it, so I want to notify the user 2 that someone called, just using usual Outlook |
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| 12:22.44 | shamelessn00b | tzafrir_laptop: question |
| 12:23.01 | tzafrir_laptop | shamelessn00b, answer |
| 12:23.08 | shamelessn00b | :D |
| 12:23.24 | shamelessn00b | can I give the dial() application a channel range to dial from |
| 12:23.45 | shamelessn00b | zap/channel(15-30) |
| 12:23.53 | shamelessn00b | something like that |
| 12:24.20 | shamelessn00b | or would I need to write an agi, to calculate whichever channel is free |
| 12:24.37 | shamelessn00b | and call that agi first, to get channel num to call from |
| 12:24.41 | tzafrir_laptop | shamelessn00b, no. But you can use a group (g, G, r, R) if it's dahdi/zap |
| 12:24.59 | shamelessn00b | its ss7 |
| 12:25.13 | shamelessn00b | so can't realy use groups |
| 12:25.44 | shamelessn00b | would most probably have to write an agi to maintain a list of available channel numbers |
| 12:27.54 | [TK]D-Fender | shamelessn00b: I don't see why you can't use groups |
| 12:30.37 | shamelessn00b | [TK]D-Fender: cuz you need a seperate point code for each trunk group you define |
| 12:30.43 | shamelessn00b | and we only have 1 point code |
| 12:32.27 | tzafrir_laptop | shamelessn00b, 'group' as in 'group=', not as in trungroup |
| 12:32.30 | [TK]D-Fender | shamelessn00b: You acn limit DAHDI even on PRI, so SS7 shouldn't be any different |
| 12:33.11 | shamelessn00b | Im using the netfors ss7 lib not the chanss7 lib |
| 12:35.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:35.28 | shamelessn00b | I see what you guys men, I think I will be able to set it up the way you guys are saying, still have to try taht out |
| 12:41.08 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 12:41.56 | Martinblr | when I terminate a call in PSTN and try to bridge with IP softphone extension using G711 alaw codec the line seems to be very disturbance.. |
| 12:42.08 | Martinblr | If i use G711 ulaw it is perfect why is it so... |
| 12:42.46 | Martinblr | also if it answers in G711 ulaw and transfer to G711 alaw it seems to be disturbance in the line |
| 12:44.23 | [TK]D-Fender | Martinblr: Have 2 SIP phones all each other (no PSTN). 1 using ALAW, the other ULAW. Force * to transcode. Does it still distort? |
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| 12:58.06 | DND | guys, anyone using cisco phone here for sip? |
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| 13:03.46 | Martinblr | [TK]D-Fender: let me try to configure... |
| 13:14.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:22.45 | leifmadsen | is there a function I'm not seeing that would tell me if a string is a valid number? |
| 13:22.54 | leifmadsen | (i.e. doesn't contain alpha characters" |
| 13:22.55 | leifmadsen | ) |
| 13:23.49 | leifmadsen | actually nevermind, I don't think I need to check that. I can just use DIALPLAN_EXISTS() |
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| 13:40.40 | AndyML | tzafrir_laptop: are you around? |
| 13:40.56 | tzafrir_laptop | AndyML, yes |
| 13:41.09 | AndyML | mind if i pm you? |
| 13:42.27 | AndyML | err tzafrir_laptop ^ |
| 13:43.03 | tzafrir_laptop | AndyML, NP |
| 13:43.08 | AndyML | tnx! |
| 13:44.03 | *** join/#asterisk MindTheGap (~caio@201.80.176.188) |
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| 13:56.52 | MindTheGap | hello ppl. Ive just changed E1 providers but now I cannot dial. I can receive calls on every channel what makes me believe dahdi config is ok. during "Dial(dahdi/g1/${EXTEN},40)" no calls hit dahdi. here is the paste: http://pastebin.ca/1874843 |
| 13:57.33 | [TK]D-Fender | mindI see no PRI debug or configs |
| 13:57.53 | [TK]D-Fender | MindTheGap: I see no PRI debug or configs |
| 13:59.57 | MindTheGap | [TK]D-Fender, hes is the updated paste... pri debug at the bottom |
| 13:59.58 | MindTheGap | http://pastebin.ca/1874846 |
| 14:01.44 | [TK]D-Fender | MindTheGap: Looks like the disconnect starts locally. Include SIP debug for the complete call. |
| 14:02.21 | MindTheGap | [TK]D-Fender i hang up because it was not calling |
| 14:02.38 | *** join/#asterisk s519 (~steve@87-194-151-213.bethere.co.uk) |
| 14:02.38 | MindTheGap | will get sip debug though, just a sec |
| 14:02.41 | [TK]D-Fender | MindTheGap: Let it run ALL the way through. |
| 14:02.54 | [TK]D-Fender | MindTheGap: You've polluted the test by cutting it short |
| 14:06.05 | MindTheGap | [TK]D-Fender, http://pastebin.ca/1874850 |
| 14:06.26 | MindTheGap | complete w pri and sip debug. left the call go till dialplan timeout |
| 14:09.58 | [TK]D-Fender | MindTheGap: Check with your telco as to what they see on thier end. They show progress. Have you tried other #'s? |
| 14:10.31 | MindTheGap | [TK]D-Fender yes, same behaviour |
| 14:12.25 | [TK]D-Fender | MindTheGap: Check with your telco then |
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| 14:17.46 | seba11223 | hi again |
| 14:18.14 | seba11223 | is there a way to inject a sip header in a different message than INVITE? |
| 14:18.51 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 14:22.11 | seba11223 | or, how can I generate SIP MESSAGE communication between 2 asterisk? this would be a solution as well |
| 14:24.23 | [TK]D-Fender | seba11223: You don't. * doesn't support SIP messaging |
| 14:25.02 | [TK]D-Fender | seba11223: Also if you have 2 * servers there are plenty of other protocols to get the boxes talking to each other |
| 14:28.30 | seba11223 | [TK]D-Fender like SendSMS? |
| 14:28.48 | seba11223 | [TK]D-Fender sorry SendText |
| 14:29.12 | [TK]D-Fender | seba11223: Like "anything else" Why would you use SMS as a comminucation method between PC's? That's very limited for no good reason. |
| 14:30.18 | seba11223 | i would like to use something via SIP/IAX, no external protocol |
| 14:30.22 | seba11223 | communication is very simple |
| 14:30.31 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 14:32.38 | [TK]D-Fender | seba11223: To do what? |
| 14:33.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:34.21 | seba11223 | to ask a logical question (yes/no) |
| 14:34.29 | leifmadsen | seba11223: use DUNDI? |
| 14:34.43 | leifmadsen | I still don't understand what you're trying to accomplish though |
| 14:34.45 | seba11223 | are you available right now? answer is yes/no |
| 14:34.54 | leifmadsen | use DUNDi for that then |
| 14:35.25 | leifmadsen | you could setup a mapping that returned a value using the GLOBAL() function which would read a variable that you set that says yes/no |
| 14:35.31 | [TK]D-Fender | seba11223: To ask WHAT? When? |
| 14:35.32 | leifmadsen | no response also means no |
| 14:35.32 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733492.dsl.bell.ca) |
| 14:36.05 | [TK]D-Fender | [10:34]<seba11223>are you available right now? answer is yes/no <- What is "available" |
| 14:36.26 | seba11223 | let's say that I want to know how many peers the other asterisk has connected right now |
| 14:36.34 | leifmadsen | DUNDi |
| 14:36.41 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:36.42 | seba11223 | I inject an X-Peers-nmber sip header in the first invite |
| 14:36.44 | jblack | So you basically want dynamic routing. that's what dundi is for. =) |
| 14:36.51 | leifmadsen | taps the mic |
| 14:37.07 | leifmadsen | testing... testing... 1 2. 3.... test.icles. |
| 14:37.08 | seba11223 | ok, thank you |
| 14:37.19 | seba11223 | :D |
| 14:37.37 | [TK]D-Fender | seba11223: when is this request triggered? |
| 14:37.40 | jblack | dundi is really, really cool. You'll love it. |
| 14:37.45 | leifmadsen | hells ya :) |
| 14:38.02 | seba11223 | before entering a queue |
| 14:38.19 | leifmadsen | yes, you want to use DUNDi to request and pass messages between servers |
| 14:38.27 | leifmadsen | that's the way to do it with asterisk |
| 14:38.35 | seba11223 | it can be done using an outside web service or so, but that introduces more points of failure |
| 14:38.37 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-188-099-208-231.pools.arcor-ip.net) |
| 14:38.37 | leifmadsen | the handling via SIP messages is not going to work the way you want likely |
| 14:38.56 | seba11223 | @leifmadsen I saw |
| 14:38.58 | seba11223 | :( |
| 14:39.14 | leifmadsen | if you want to inject messages into SIP then you need to use an external proxy like OpenSIPS or something |
| 14:39.17 | Polysics | still wrestling with the totally-firewalled users situation... |
| 14:39.49 | leifmadsen | something that handles SIP messages at a more fundamental level. Asterisk does not do that. |
| 14:40.06 | seba11223 | I have a very simple communication, I will look into dundi |
| 14:40.14 | Polysics | i wonder if IAX over HTTP is actually doable or if i am just looking for trouble |
| 14:40.20 | leifmadsen | yes, DUNDi is for very simple communications like what you want |
| 14:40.47 | p3nguin | wonders what leifmadsen would use in this case |
| 14:40.53 | leifmadsen | in what case? |
| 14:41.07 | p3nguin | messaging between the two Asterisk systems. |
| 14:41.19 | leifmadsen | your sarcasm is thick :) |
| 14:41.28 | leifmadsen | cuts it with a knife |
| 14:41.37 | leifmadsen | that's not a knife! THIS is a knife! |
| 14:43.47 | seba11223 | [TK]D-Fender to answer you last question, I want ot know if another system has agents available on a queue, in order to call it, or skip it to the next queue |
| 14:44.01 | seba11223 | [TK]D-Fender I have to use queue because of the stats system |
| 14:44.12 | [TK]D-Fender | seba11223: Use another means of communication. |
| 14:44.26 | seba11223 | like web? |
| 14:44.39 | [TK]D-Fender | seba11223: Like SOMETHING ELSE. |
| 14:44.43 | seba11223 | ok :) |
| 14:44.57 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
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| 14:48.23 | seba11223 | 10x for your help |
| 14:48.25 | seba11223 | bye |
| 14:48.50 | russellb | 10x what |
| 14:52.35 | [TK]D-Fender | russellb: it's Twit for "thanks" |
| 14:52.37 | Polysics | is IAX traffic always through *, not peer-to-peer? |
| 14:52.53 | [TK]D-Fender | Polysics: "nope, can be both |
| 14:52.56 | Polysics | [TK]D-Fender, do we have a Twitter language too now? |
| 14:53.06 | Polysics | [TK]D-Fender, can or has to be? |
| 14:53.22 | [TK]D-Fender | Polysics: Twitter is one who speaks Twit. Well conjugated! |
| 14:53.29 | [TK]D-Fender | Polysics: Can. |
| 14:53.38 | Polysics | because i just got the admin of the closed-off network to agree to open a single port towards a single IP |
| 14:53.39 | *** join/#asterisk smooth_penguin (~smoove@triband-mum-120.61.136.112.mtnl.net.in) |
| 14:53.49 | [TK]D-Fender | Polysics: Twitter is also a very for one convering in Twit :p |
| 14:53.53 | [TK]D-Fender | conversing* |
| 14:54.07 | Polysics | that will obviously get a heavier load on the server, but if it makes the app sellable, i am ok with it :-) |
| 14:54.58 | Polysics | one thing i think i already asked, but i didn't get an answer: if i do have access to a DMZ, can't i put a sip proxy in there? |
| 14:55.04 | [TK]D-Fender | Polysics: Twitter is also a verb for one conversing in Twit* |
| 14:55.15 | [TK]D-Fender | Damn I'm all kinds of dyslexic today... |
| 14:55.29 | Polysics | i think Twit is also some sort of an insult in some English dialects, but i might be wrong :-) |
| 14:55.30 | russellb | why is 10x better than thx |
| 14:55.32 | russellb | that's stupid |
| 14:55.35 | [TK]D-Fender | Polysics: packets are packets... |
| 14:55.46 | [TK]D-Fender | russellb: See: Twit |
| 14:55.55 | russellb | :-p |
| 14:56.21 | Polysics | i do like twitter, it's still better than facebook |
| 14:57.41 | *** join/#asterisk robsonfelix (~robsonfel@187.56.7.163) |
| 14:57.51 | robsonfelix | hey guys, good morning |
| 14:57.56 | [TK]D-Fender | Twitter: When you acre to share every retarded thought that jsut crosses your mind with the world at large |
| 14:58.11 | robsonfelix | has anyone here used the " speech * " AGI commands? |
| 14:58.20 | [TK]D-Fender | care* |
| 14:59.02 | Polysics | [TK]D-Fender, i sort of agree, though many ppl do publish meaningful things |
| 14:59.08 | Kyosh | twitter is for those who wish to share the pain of their excrement. facebook is for those who wish to show the pics of it |
| 14:59.18 | Polysics | i agree about the "i am currently breathing" posts |
| 14:59.40 | Polysics | [TK]D-Fender, i didn't get your "packets are packets" above |
| 14:59.46 | [TK]D-Fender | I have 11 friends on Facebook... haven't found how to ditch the last of them properly yet ;) |
| 14:59.52 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:00.02 | Polysics | at least Facebook tells you about birthdays |
| 15:00.12 | [TK]D-Fender | Polysics: asking if you can run a proxy beihnd a DMZ. Well.. if the packets flow, then waht does that tell you? |
| 15:00.28 | Polysics | which proxy would you recommend? |
| 15:00.37 | [TK]D-Fender | Polysics: You need to know the IP to tell others to respond to you at, and the packets have to flow both ways. |
| 15:00.53 | [TK]D-Fender | Polysics: Who says you need aproxy. you just said you'd be DMZ'd. |
| 15:01.14 | Polysics | the proxy can sit in this "open" area, and the local network would connect to it |
| 15:01.30 | Polysics | does such a thing even exist? i am truly ignorant in the area |
| 15:01.56 | Polysics | the same thing as Squid, but for SIP or IAX :-) |
| 15:02.23 | leifmadsen | Polysics: sure... have a local Asterisk box :) |
| 15:02.55 | Polysics | leifmadsen, can i tell the local box to just be a proxy for the remote one? |
| 15:03.04 | Polysics | that would be just great :-) |
| 15:03.32 | leifmadsen | I don't understand why not... just have the phones connect to the local box, then have external connections come into that Asterisk on another interface |
| 15:04.19 | Polysics | point is, the "central" asterisk has the Adhearsion app that manages billing, and the related real-time webpages, all sort of stuff |
| 15:04.32 | leifmadsen | "central" means nothing |
| 15:04.39 | Polysics | my problem is "just" not being able to connect to the main server |
| 15:04.47 | leifmadsen | it's like saying Toronto is the centre of the universe |
| 15:04.58 | riddlebox | hrmm trying to send calls from one asterisk box to another, I can get one to the other fine, for some reason my second one is sending calls to the first as asterisk@IP |
| 15:05.00 | leifmadsen | (which it is, of course) |
| 15:05.26 | Polysics | i respectfully disagree, Des Moines had that title before |
| 15:05.39 | Polysics | oh, i am sorry, i read "hell" instead of "world" |
| 15:05.40 | Polysics | :-) |
| 15:07.14 | Polysics | leifmadsen, anyway, i would need all registration, calls and activity to go through the main server, the proxy would just be, uhm ,a proxy :-) |
| 15:07.25 | Polysics | i don't even know if it is possible/recommended |
| 15:07.26 | [TK]D-Fender | [11:04]<leifmadsen>it's like saying Toronto is the centre of the universe <- like the dark whole in the middle of a doughnut ;) |
| 15:11.00 | Polysics | does the above make any sense? |
| 15:11.25 | leifmadsen | just broke a tooth -- off to the dentish |
| 15:12.55 | [TK]D-Fender | EEK |
| 15:13.18 | Polysics | argh |
| 15:13.29 | Polysics | did he bite a softphone in anger? |
| 15:13.52 | Polysics | *phone |
| 15:14.11 | Polysics | i am so used to softphones i use the term even when i am not supposed to |
| 15:14.16 | [TK]D-Fender | Polysics: No, it was clearly a hard-phone :) |
| 15:14.25 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 15:26.06 | robsonfelix | has anyone used the "speech" AGI commands? somebody? |
| 15:26.12 | pwntang | is there anyone here that has tried the Pirelli WLAN phone on Asterisk 1.4 pls? |
| 15:26.42 | pwntang | Pirelli DP-L10 |
| 15:28.00 | [TK]D-Fender | pwntang: Looks like a cheap import junk |
| 15:29.16 | pwntang | I don't think they're particularly expensive |
| 15:29.49 | [TK]D-Fender | pwntang: http://www.voip-info.org/wiki/view/DP-L10 |
| 15:29.56 | [TK]D-Fender | pwntang: All sorts of lovely issues |
| 15:30.00 | [TK]D-Fender | ~wifivoip |
| 15:30.01 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 15:30.12 | [TK]D-Fender | pwntang: What are you r actual needs? |
| 15:32.01 | pwntang | I get a 404 unauthorised error when it tries to register |
| 15:32.15 | [TK]D-Fender | pwntang: Then go fix your auth |
| 15:32.16 | pwntang | there isn't much to change on the SIP settings within the phone itself |
| 15:33.38 | pwntang | it doesn't seem to be as simple as that. I've been trying to get it to register for a while now. It's not important. I've already replaced it with a Cisco ATA186 and a cordless phone |
| 15:33.57 | pwntang | I just thought I'd ask here as someone may have 'played' with one themselves |
| 15:34.12 | pwntang | I think perhaps it's something within its firmware |
| 15:34.20 | pwntang | that perhaps isn't compatible |
| 15:34.33 | *** join/#asterisk Gokee2 (~gokee2@24-113-159-168.wavecable.com) |
| 15:35.39 | robsonfelix | is there somebody here that have used TTS in an AGI Script? There are no AGI commands that performs TTS... |
| 15:37.56 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:38.53 | Tim_Toady | robsonfelix you can intergate in ur script sytem commands like espeak, flite etc or use app_festival in the dialplan |
| 15:39.11 | robsonfelix | i want to use either Nuance or Loquendo... is it possible? |
| 15:39.22 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:39.28 | robsonfelix | so you mean by using the EXEC agi command? |
| 15:39.45 | [TK]D-Fender | pwntang: No. Clearly others get it working fine. |
| 15:40.06 | [TK]D-Fender | robsonfelix: AGI is just a way to call DIALPLAN apps |
| 15:40.34 | [TK]D-Fender | robsonfelix: there is very little call-control wise that is special to AGI |
| 15:40.42 | robsonfelix | D-Fender, I know... I have been using it for a while, but with plain vanilla audio files... |
| 15:41.21 | robsonfelix | The only trick part is to get the TTS and Speech Recognition working.... Ast 1.6 now has the SPEECH commands for the ASR end |
| 15:41.25 | robsonfelix | but no TTS feature |
| 15:42.25 | robsonfelix | well, the EXEC agi command would be able to call a dialplan application |
| 15:42.29 | robsonfelix | like Tim_Today said |
| 15:42.36 | robsonfelix | flite or festival |
| 15:42.40 | [TK]D-Fender | robsonfelix: * has the same TTS features as always. Cepstral, Sphinx, etc. The app names are all well known |
| 15:42.51 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 15:43.27 | robsonfelix | so it would work from within my AGI script by issuing a EXEC app_festival(here comes my text), for exemple... |
| 15:44.03 | [TK]D-Fender | robsonfelix: Just like any other dialplan app call |
| 15:44.10 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-74-70.vif.net) |
| 15:44.12 | robsonfelix | ok then... I will try :) |
| 15:44.15 | robsonfelix | thanks guys |
| 15:44.43 | pwntang | d-fender: clearly. |
| 15:46.15 | ZeXr0 | Is it possible to catch a Number Not In Service with Asterisk ? I get an error while dialing : " NoOp("OutgoingSpoolFailed", "ATTENTION! OUTBOUND CALL FAILED WITH REASON: 0") in new stack" |
| 15:46.28 | ZeXr0 | But is it possible to determine if the error was because the number was not in service ? |
| 15:48.03 | pwntang | fender: if it's that simple, then perhaps you could help me get it to register? |
| 15:48.05 | Tim_Toady | ZeXr0 maybe using chanisavail() before dialing? |
| 15:49.48 | ickmund | Are there any negative effects from reloading? I mean, are there any chance of calls being dropped or anything like that? |
| 15:50.00 | [TK]D-Fender | pwntang: Perhaps you could show your configs and teh SIP debug of your failed attempts |
| 15:50.06 | [TK]D-Fender | pwntang: PASTEBIN is your friend. |
| 15:50.08 | [TK]D-Fender | ~pb |
| 15:50.09 | 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. |
| 15:50.11 | pwntang | of course |
| 15:50.15 | pwntang | one sec |
| 15:50.22 | [TK]D-Fender | ickmund: "reload" will not drop calls |
| 15:50.43 | pwntang | I'm guessing it's not the done thing to paste debugs into this channel? |
| 15:50.48 | ZeXr0 | Tim_Toady : Wouldn't it works only for my channels ? I mean, I dial to a number, and it tells me that the number is not available. Would it be possible to catch it with chanisavail() ? |
| 15:51.24 | Tim_Toady | i dont think so |
| 15:53.21 | [TK]D-Fender | pwntang: Correct |
| 15:53.45 | [TK]D-Fender | ZeXr0: Depends how it tells you |
| 15:54.04 | [TK]D-Fender | ZeXr0: Of course you haven't given us any details so... who knows |
| 15:55.15 | [TK]D-Fender | [11:49]<[TK]D-Fender>pwntang: Perhaps you could show your configs and teh SIP debug of your failed attempts |
| 15:55.22 | [TK]D-Fender | pwntang: And please do not PM me |
| 15:56.06 | pwntang | if you don't want me to post them here, and I am not allowed to PM you, can you suggest where else I should put them? |
| 15:56.29 | ickmund | ~pb |
| 15:56.30 | 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. |
| 15:57.53 | pwntang | http://pastebin.com/0gy9T8uc |
| 15:58.38 | pwntang | sip.conf as well? |
| 15:59.56 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 16:01.35 | TJNII | pwntang: Well, that's the error, but you left out everything else. Please pastebin the COMPLETE debug. |
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| 16:02.45 | pwntang | will do.... |
| 16:03.32 | ZeXr0 | [TK]D-Fender : In my callflow, it redirect to extension failed using the ${REASON} = 0 |
| 16:03.42 | TJNII | And by complete debug I mean start with the INVITE and then stop with that error. |
| 16:03.51 | ZeXr0 | Which is supposed to mean 0 - Failed (not busy or congested) |
| 16:04.21 | [TK]D-Fender | ZeXr0: and it tells me that the number is not available <- how? Got something to SHOW us? |
| 16:05.16 | ZeXr0 | Oh, well I get my phone, and I call that number, the number is not in service, but I do not get it in the callflow (Sorry : English is not my native language) |
| 16:07.05 | pwntang | this is what I get from the moment I switch the phone on until about 10 seconds later..... |
| 16:07.09 | pwntang | http://pastebin.com/d7XqsF9t |
| 16:11.36 | pwntang | apologies, I missed this from the start |
| 16:11.38 | pwntang | http://pastebin.com/uWmgX4ra |
| 16:14.25 | pwntang | here is a part of my sip.conf that may be relavent http://pastebin.com/kfKWbchm |
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| 18:06.27 | seanjohn | I know I can replace musiconhold command with mp3player command but can I define mp3player as a custom application in musiconhold.conf? |
| 18:06.59 | p3nguin | If you want to play mp3s in musiconhold, you don't need to screw with mp3player to do it. |
| 18:07.15 | seanjohn | i know i'm already using mpg123 |
| 18:07.27 | p3nguin | You don't need to do that, either. |
| 18:07.31 | p3nguin | [mp3s] |
| 18:07.31 | p3nguin | mode=quietmp3 |
| 18:07.31 | p3nguin | directory=/var/lib/asterisk/mohmp3 |
| 18:07.48 | seanjohn | mpg123 is executed 4 times, in ps -aux, for the same STREAM |
| 18:07.50 | p3nguin | Then put your mp3s in /var/lib/asterisk/mohmp3/ directory. |
| 18:07.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:08.02 | seanjohn | i know how to do that but i'm using streams |
| 18:08.02 | p3nguin | ps aux, not ps -aux |
| 18:08.44 | seanjohn | it only needs to play one stream for all calls of the same class |
| 18:08.48 | p3nguin | I use a stream as well and I don't have such issues. |
| 18:08.52 | seanjohn | but it plays 3 to 4 |
| 18:09.00 | seanjohn | how are you configuring it? |
| 18:09.08 | p3nguin | application=/var/lib/asterisk/scripts/mohstream.sh |
| 18:09.40 | seanjohn | what is mohstream.sh, just the same as application=/usr/bin/mpg123 except its in the file? |
| 18:09.41 | p3nguin | Then the script contains: /usr/bin/mpg123 $mp3_opts $stream |
| 18:09.48 | seanjohn | ok |
| 18:10.25 | p3nguin | The mp3_opts I use are: -q -b 128 --preload 32 -r 8000 -f 4096 -m -s |
| 18:10.48 | p3nguin | Then the stream I use is a regular shoutcast mp3 stream. |
| 18:10.53 | seanjohn | i only buffer 64 as I only pick streams with 56k or lower |
| 18:13.51 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 18:27.28 | seanjohn | what do you set as format= |
| 18:28.03 | seanjohn | i did the exact thing you said, with asterisk 1.4.31, and there is no music, just silence and the scripts aren't executed by asterisk |
| 18:28.15 | seanjohn | in px -aux |
| 18:28.36 | seanjohn | asterisk crashes on a reload by freepb |
| 18:29.50 | seanjohn | correction: asterisk crashes on any reload |
| 18:30.09 | p3nguin | ps aux, not ps -aux |
| 18:30.46 | seanjohn | [May 31 14:30:08] WARNING[2170] res_musiconhold.c: Unknown format '' -- defaulting to SLIN |
| 18:30.47 | seanjohn | [May 31 14:30:08] DEBUG[2170] res_musiconhold.c: killing 2753! |
| 18:34.19 | seanjohn | hello? p3nguin. asterisk crashes |
| 18:34.27 | p3nguin | sounds painful. |
| 18:34.43 | seanjohn | you said this would work |
| 18:34.53 | p3nguin | [default] |
| 18:34.59 | p3nguin | mode=custom |
| 18:34.59 | p3nguin | application=/var/lib/asterisk/scripts/mohstream.sh |
| 18:35.08 | p3nguin | That's all it takes. |
| 18:35.21 | seanjohn | [default] |
| 18:35.21 | seanjohn | mode=custom |
| 18:35.35 | seanjohn | application=/etc/asterisk/common.sh |
| 18:36.06 | seanjohn | | /usr/bin/mpg123 -q -r 8000 -b 56 -f 4096 --preload 15 --mono -m -s /usr/bin/mpg123 -q -r 8000 -f 8192 http://scfire-dtc-aa01.stream.aol.com:80/stream/1076 |
| 18:36.11 | seanjohn | that's common.sh |
| 18:36.20 | p3nguin | If /etc/asterisk/common.sh contains the right commands, I don't see what the problem would be. |
| 18:36.24 | seanjohn | common.sh made executable chmod +x |
| 18:37.05 | seanjohn | what asterisk are you using? |
| 18:37.07 | p3nguin | It seems that your command in the script is all jacked up. |
| 18:37.19 | p3nguin | 1.4.30 |
| 18:37.35 | TJNII | stream.aol.com? _really?_ |
| 18:37.51 | p3nguin | Lots of shoutcast streams are on aol.com. |
| 18:38.31 | TJNII | sings "Tonight we're gonna party like its Nine-teen-nienty-Nine!" |
| 18:38.46 | p3nguin | I could go look in the shoutcast directory, pick any random stream name, and it probably has an aol.com server name. |
| 18:40.14 | p3nguin | I guess I could test that stream to make sure it's working on Asterisk. |
| 18:40.18 | *** join/#asterisk kazaa_lite (~eddie@212.183.140.52) |
| 18:40.22 | kazaa_lite | hi all |
| 18:40.39 | kazaa_lite | what is difference between narrowband and wideband? |
| 18:41.03 | kazaa_lite | like there are wideband codecs... narrowband codecs... ultrawideband... |
| 18:41.17 | p3nguin | Works just fine here. |
| 18:42.01 | p3nguin | "signing 'I love rock and roll... put another dime in the juke box, baby...'" |
| 18:43.13 | p3nguin | [root@asterisk ~]# ps aux|grep mpg123 |
| 18:43.13 | p3nguin | asterisk 25335 0.5 0.2 4460 1316 ? S 13:41 0:00 /usr/bin mpg123 -q -b 128 --preload 32 -r 8000 -f 4096 -m -s http://scfire-dtc-aa01.stream.aol.com:80/stream/1076 |
| 18:43.41 | coppice | kazaa_lite: the band of the narrowband codecs is not as wide as the band of the wideband ones :-\ |
| 18:43.53 | voxter | call me crazy, but has anyone considered having asterisk auto-convert sound files into their native format on first-play so that subsequent plays dont need to be transcoded? like as an option |
| 18:44.53 | kazaa_lite | coppice: is the sinewave of g722 much wider than of g711? |
| 18:45.49 | coppice | well, a 7kHz sinewave comes through well with G.722, but is completely blocked by G.711 |
| 18:45.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:47.36 | seanjohn | p3nguin: can you please pastebin both your musiconhold.conf and your musiconhold.sh or w/e |
| 18:47.51 | p3nguin | You've already seen both. |
| 18:48.17 | seanjohn | did you chmod it to executable |
| 18:48.33 | p3nguin | of course |
| 18:48.37 | kazaa_lite | oka so if someone asks me whats the difference between wideband and narrow band codecs..... so I will be correct to say that ... narrowband codecs have shorter sine wave |
| 18:48.57 | seanjohn | where are mp3_opts defined? |
| 18:49.16 | kazaa_lite | or what defines whether some codec is narrowband or wideband codec? |
| 18:51.27 | p3nguin | It's really this hard for you? Write the following in the shell script: /usr/bin/mpg123 -q -b 128 --preload 32 -r 8000 -f 4096 -m -s http://scfire-dtc-aa01.stream.aol.com:80/stream/1076 |
| 18:51.40 | [TK]D-Fender | voxter: Easy enough to script that already |
| 18:51.45 | p3nguin | save, exit, make sure it is +x |
| 18:51.48 | [TK]D-Fender | voxter: Oh.. and "You're crazy" |
| 18:52.02 | voxter | [TK]D-Fender: yeah, its scriptable., lots of stuff is scriptable that is now a core feature. |
| 18:52.15 | voxter | [TK]D-Fender: Hah, it only seems to make sense to me. |
| 18:52.21 | p3nguin | If you already have some mpg123 processes running, kill them. |
| 18:52.32 | p3nguin | Then call musiconhold. |
| 18:52.36 | coppice | kazaa_lite: http://www.polycom.com/common/documents/whitepapers/effect_of_bandwidth_on_speech_intelligibility_2.pdf |
| 18:52.38 | [TK]D-Fender | voxter: the rest of us have either a ) done it already, or b ) don't ahve to care |
| 18:55.51 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:56.32 | seanjohn | what version of mpg123 are you using, p3nguin |
| 18:56.52 | seanjohn | [buffer.c:170] debug: Prepare for waiting; draining command queue. (There's a lot of wakeup commands pending, usually.) |
| 18:56.52 | seanjohn | [buffer.c:213] debug: bytes: 0 |
| 18:56.52 | seanjohn | [buffer.c:225] debug: got -1 |
| 18:57.04 | seanjohn | that's what I get when I execute the common.sh |
| 18:57.10 | p3nguin | Currently 1.10.0, but I'm sure they all work. |
| 18:57.15 | seanjohn | well, its alot more than that and keeps scrolling |
| 18:57.47 | seanjohn | can you try executing your script and see if you get garbled or readable text |
| 18:57.58 | p3nguin | Execute it on the command line? |
| 18:58.01 | seanjohn | yeah |
| 18:58.24 | p3nguin | It outputs the music stream's data onto stdout. |
| 18:58.51 | seanjohn | version 1.12.1 |
| 18:59.15 | p3nguin | I'll see if I have a newer version available. |
| 18:59.34 | p3nguin | I have that version, so I'll upgrade to it. |
| 18:59.55 | p3nguin | Still works. |
| 19:00.16 | p3nguin | mpg123-1.12.1-1-i68... |
| 19:00.36 | seanjohn | | /usr/bin/mpg123 -q -b 128 --preload 32 -r 8000 -f 4096 -m -s http://scfire-dtc-aa01.stream.aol.com:80/stream/1076 |
| 19:00.41 | seanjohn | that's common.sh |
| 19:00.45 | p3nguin | What's with the pipeline? |
| 19:01.08 | seanjohn | the | is to keep irc from thinking /usr/ is a command |
| 19:01.12 | p3nguin | oh |
| 19:01.49 | seanjohn | you use variables on yours but where are you defining the variables? |
| 19:01.54 | kazaa_lite | coppice: are you drunk? |
| 19:02.09 | coppice | eh? |
| 19:02.17 | ZeXr0 | seanjohn : You can also press Ctrl when pressing enter |
| 19:02.20 | ZeXr0 | /hellp |
| 19:02.24 | ZeXr0 | ;) |
| 19:02.38 | seanjohn | /usr/bin/mpg123 -q -b 128 --preload 32 -r 8000 -f 4096 -m -s http://scfire-dtc-aa01.stream.aol.com:80/stream/1076 |
| 19:02.40 | seanjohn | ok |
| 19:03.12 | p3nguin | Variables should be defined in the script. But it doesn't make any difference if you use a variable or not; you've got the same options as what I have, using the same version of mpg123 that I am now using. |
| 19:03.14 | kazaa_lite | coppice: i went through the whole pdf and it was no where DEFINING the difference betweeb narrowband and wideband codecs...... though it was a good story |
| 19:03.30 | seanjohn | pretty much same version of asterisk |
| 19:03.43 | seanjohn | i compiled mpg123 and used dummy audio output |
| 19:04.04 | seanjohn | the pc does have a sound card in it but is never used |
| 19:04.09 | p3nguin | I use the mpg123 package provided by my distro. |
| 19:04.10 | coppice | kazaa_lite: bye, troll |
| 19:04.17 | seanjohn | distro? |
| 19:04.21 | seanjohn | not centos? |
| 19:04.39 | p3nguin | CentOS is a distro, but I'm not using CentOS. |
| 19:04.47 | p3nguin | yum -y install mpg123 |
| 19:05.15 | kazaa_lite | coppice: My pleasure. It will at least save my time than hitting my head in wrong directions. |
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| 19:11.36 | seanjohn | i know how to use yum i was asking if you was using centos |
| 19:11.48 | seanjohn | that's your difference and why it won't work |
| 19:13.49 | p3nguin | lol, no, that's not why it doesn't work. |
| 19:14.08 | p3nguin | The fact that you didn't use your package manager is probably why it doesn't work. |
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| 20:12.27 | Micc | I see these rtcp messages every week or so: rtp.c:3397 ast_rtcp_write_sr: RTCP SR transmission error to x.x.x.x:13079, rtcp halted Operation not permitted |
| 20:12.55 | Micc | I ignored it on most errors because it was a provider that maybe didn't support RTCP. |
| 20:14.23 | Micc | But the one I just got was another asterisk server with full firewall open between the two. The one I got the error on is running 1.6.1.20 and the one it says the error came from or is about from port 13079 is running 1.6.2.7 |
| 20:15.10 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 20:24.06 | *** join/#asterisk acxty (~acxty@201.220.136.118) |
| 20:24.08 | *** join/#asterisk residentgrey (~residentg@108.96.23.64) |
| 20:24.09 | acxty | Hi guys |
| 20:25.18 | *** join/#asterisk bn-7bc (bjarne@lan-gw.noare-1.holmedal.net) |
| 20:25.22 | Trixboxer | hi |
| 20:26.19 | acxty | Right now when I want to call outside I need to call extension 2, It answer automatically then I press 9 and it give me tone to dial the number. I am trying to do that on extension.conf whant I am doing is this Dial(SIP/2/96021212,2) being 6021212 the number I want to dial |
| 20:26.44 | acxty | but it is not working |
| 20:27.22 | acxty | what is the correct way so asterisk can understand what to do? |
| 20:28.07 | Trixboxer | what does your CLI shows ? |
| 20:28.14 | Trixboxer | pastebin from the begining |
| 20:28.24 | Trixboxer | like when u initiate the call to extension2 |
| 20:28.30 | acxty | right now |
| 20:32.52 | acxty | http://dpaste.com/201570/ |
| 20:33.06 | acxty | I am using a .call file |
| 20:33.21 | acxty | to call to an extension 1 and then bridge to the cellphone using extension 2 |
| 20:41.17 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 20:41.19 | cusco | hi |
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| 20:41.44 | cusco | two asterisks, peer dials asterisk1, asterisk1 dials asterisk2, asterisk2 dials peer2 |
| 20:42.04 | cusco | now peer2 can hear peer1, but peer1 cannot hear peer2 |
| 20:42.06 | cusco | why?? |
| 20:43.34 | Trixboxer | acxty: it is saying that "SIP/2-081c1df8 answered SIP/1-081c0868" |
| 20:53.08 | cusco | ok disabling directmedia it works |
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| 21:09.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:10.17 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 21:11.34 | ujjain | Hi, I want to purchase 2 VOIP phones. |
| 21:11.39 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 21:11.48 | ujjain | GXP2000 was available for â¬40,00 including shipping costs. |
| 21:12.16 | ujjain | but I would like to give my parents a solution with wireless and speakerphone capabilities. |
| 21:13.33 | Chainsaw | ujjain: I've found that wireless networks & VoIP don't always cooperate well. |
| 21:13.35 | Chainsaw | ujjain: However... |
| 21:13.42 | ujjain | I do not mean wi-fi. |
| 21:13.46 | Chainsaw | ujjain: You can get some Siemens handsets which are DECT wireless and talk SIP. |
| 21:14.06 | ujjain | I have a wireless phone at the moment, but it works via PAP2T. |
| 21:14.10 | Chainsaw | ujjain: I use a few in the production network and one at home. While they are by no means perfect, I'm happy to recommend them. |
| 21:14.20 | ujjain | Also there is always a huge delay in calling and when the phone finally rings. |
| 21:14.30 | ujjain | Chainsaw: You use PAP2T a lot? |
| 21:14.41 | Chainsaw | ujjain: What is this PAP2T of which you speak? |
| 21:14.43 | stix | What do I do when no extension can register and I have no sip-command in the asterisk cli? |
| 21:14.47 | ujjain | Linksys PAP2T Adapter |
| 21:14.57 | ujjain | allows you to use normal phones with VOIP. |
| 21:15.01 | Chainsaw | ujjain: No, I use Patton gateways. |
| 21:15.02 | p3nguin | stix: Extensions don't register, devices do. |
| 21:15.19 | ujjain | Analog phone ->PAP2T -> router. |
| 21:15.25 | Chainsaw | ujjain: To make the emergency phone ring: http://www.vroon.org/ringring.jpg |
| 21:15.35 | stix | p3nguin: okat we will call my sip-phones devices from now on |
| 21:15.45 | cusco | ujjain: we have a semens... let me check the model.. with wireless 8radio) and speaker |
| 21:15.47 | p3nguin | sip phones are devices. |
| 21:15.53 | p3nguin | ~a580 |
| 21:15.54 | infobot | [~A580] The Siemens Gigaset A580 IP with ECO DECT technology is multi-line so you are free to register up to 6 handsets for 6 SIP accounts from different providers and make up to 3 calls in parallel: 2 VoIP calls and 1 fixed-line call. See http://gigaset.com/hq/en/product/GIGASETA580IP.html for details. Cost is about $70USD for one handset with base. Extra A58H handsets are around $40USD each. |
| 21:15.57 | p3nguin | ujjain: ^^^ |
| 21:16.04 | cusco | siemens gigaset a58h |
| 21:16.04 | stix | sure they are but thats not the point |
| 21:16.16 | cusco | thats it!! |
| 21:16.20 | Chainsaw | It doesn't seem to have a colour screen like mine. |
| 21:16.22 | cusco | yes up to 6 handsets per base |
| 21:16.31 | ujjain | hi p3nguin |
| 21:16.48 | ujjain | Also, my parents have 2 phones for 1 connection atm :P |
| 21:16.57 | ujjain | Not sure I even want to give them a VOIP birthday present :P |
| 21:17.23 | ujjain | 2 phones -> 1 analog telephone line, costs ⬠12,00 monthly I think. They call 30 minutes monthly. |
| 21:17.26 | p3nguin | stix: Make sure you have sip.conf configured correctly and that you have chan_sip.so loaded. |
| 21:17.40 | stix | okay |
| 21:18.00 | p3nguin | stix: If you are not sure if sip.conf is correct, put it in pastebin.com. |
| 21:18.11 | stix | oki, will have a look |
| 21:19.11 | ujjain | I am sad that my PAP2T adapter seems to cause a 10 second delay in the phone actually ringing |
| 21:19.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:19.19 | Chainsaw | (It looks like I have a C470) |
| 21:19.21 | ujjain | hey Bender |
| 21:19.26 | stix | the module was not loaded, wondering why it wasn't |
| 21:19.29 | ujjain | Fender* |
| 21:19.32 | p3nguin | ujjain: What kind of network latency are you seeing? |
| 21:19.43 | p3nguin | stix: Do you have sip.conf configured? |
| 21:19.48 | Chainsaw | C475 even. |
| 21:19.51 | ujjain | p3nguin: Good question, no idea. It's all wired, but cannot be high. |
| 21:19.55 | Chainsaw | Mine has the answering machine. |
| 21:20.01 | p3nguin | stix: Usually not having that is why chan_sip doesn't load automatically. |
| 21:20.08 | stix | yes, and it loads just fine when I load it manually |
| 21:20.23 | p3nguin | stix: Restart asterisk and see if it is loaded automatically now. |
| 21:20.27 | stix | yes |
| 21:20.47 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.2) |
| 21:23.02 | bjhaid | i want to call my asterisk box from mobile i have a sip provider account, how do i put the call through/ |
| 21:24.02 | p3nguin | bjhaid: Configure sip.conf so that your provider is a peer. Configure extensions.conf to accept the phone number from your provider; make the extension do something useful. |
| 21:24.46 | *** join/#asterisk DrDamnit (~michael@65.14.113.178) |
| 21:25.39 | DrDamnit | How do I send SIP hints over an IAX trunk so that BLF shows up properly two networks with two asterisk servers that are connected by said IAX trunk? |
| 21:26.05 | carrar | heh |
| 21:26.16 | bjhaid | i am not sure i know how to configure my sip for my provider correctly my account is 2233406561@sip2sip.info how do i include it in my register section and in my extensions |
| 21:26.53 | p3nguin | bjhaid: Have you read The Book? |
| 21:26.56 | p3nguin | ~book |
| 21:26.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:27.25 | bjhaid | yes, its not well detaile there |
| 21:27.27 | DrDamnit | bjhaid: you would not need to include it in the extensions. Just configure it in sip.conf. Read the book p3nguin just referenced. |
| 21:27.49 | p3nguin | bjhaid: It IS detailed there. |
| 21:28.12 | DrDamnit | bjhaid: It won't have specifics for your ITSP, but what you need is in there. You just need the register string for registration, and then the trunk setup as detailed in the book. |
| 21:28.56 | [TK]D-Fender | bjhaid: register => user:pass@host/extentohavethemsendcallsto |
| 21:29.14 | p3nguin | And you do have to include your phone number in extensions.conf, or at least some pattern that matches. |
| 21:30.26 | DrDamnit | bjhaid: See also: http://bit.ly/bGReAs |
| 21:32.33 | DrDamnit | How do I send SIP hints over an IAX trunk so that BLF shows up properly two networks with two asterisk servers that are connected by said IAX trunk? |
| 21:35.31 | [TK]D-Fender | DrDamnit: http://snapvoip.blogspot.com/2008/06/distributed-presence-for-asterisk-in.html |
| 21:35.34 | [TK]D-Fender | DrDamnit: Not over IAX |
| 21:40.58 | DrDamnit | TKD-Fender: that's what I figured. I know you can do it in switchvox, but not so much in asterisk with IAX. |
| 21:42.00 | DrDamnit | Thanks! |
| 21:43.00 | *** join/#asterisk stix_ (~stix@exchange2003.corporate.billetkontoret.dk) |
| 21:45.01 | [TK]D-Fender | DrDamnit: you just named aGUI that USES asterisk. The says absolutely nothing about the "hows" |
| 21:45.03 | cusco | a question... |
| 21:45.13 | cusco | what are the dial options that I can't use to have directmedia working |
| 21:45.14 | cusco | with sip |
| 21:45.38 | [TK]D-Fender | cusco: any recording, transfers, call features, etc |
| 21:46.36 | cusco | call featurs liek atxfer *8 in features.conf |
| 21:53.04 | cusco | [TK]D-Fender: peer1 dials to peer2. Asterisk1 dialplan has: http://paste.debian.net/75613/ where 10.100.100.3 is asterisk2 where peer2 is registered |
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| 22:00.40 | [TK]D-Fender | cusco: No dialplan in the way. |
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| 22:25.19 | cusco | [TK]D-Fender: ok I found that realtime configuration on mysql had canreinvite=no |
| 22:25.32 | cusco | is it safe to update the field to directmedia |
| 22:25.35 | cusco | instead? |
| 22:30.33 | cusco | [TK]D-Fender: I can't figure why I can't have directmedia between peer1 and peer2 |
| 22:30.46 | cusco | but if peer3 is on asterisk1 (as well as peer1) I have directmedia |
| 22:31.32 | cusco | so if call goes trough asterisk2 |
| 22:31.42 | cusco | there is no directmedia |
| 22:32.15 | cusco | altho if I register a peer in asterisk2 and dial a peer in the same asterisk, I will have directmedia |
| 22:32.19 | cusco | :( |
| 22:32.28 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 22:33.49 | cusco | [TK]D-Fender: any hints? |
| 22:34.36 | [TK]D-Fender | cusco: Allow reinvites and stop using the term "directmedia" |
| 22:35.59 | cusco | well I read that in newer asterisk versions canreinvite is directmedia |
| 22:36.14 | cusco | but reinvites are alowed |
| 22:37.31 | cusco | they are allowed as I can in each asterisk dial to a peer on the same asterisk, and have re-invites |
| 22:37.41 | cusco | but asterisk to asterisk that doens't seem to happen |
| 22:38.01 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 22:40.29 | cusco | how can I diagnose that? |
| 22:41.54 | *** part/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
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| 22:47.15 | cusco | [TK]D-Fender: asterisk1 is on ip 192.168.2.5 and asterisk2 is on 10.100.100.3 ... both our networks are in a vpn |
| 22:47.16 | *** join/#asterisk leif[mobile] (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:47.16 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 22:47.25 | cusco | so we can access each other with no nat |
| 22:47.49 | cusco | does that make a difference on how asterisk recognises if it allows the reinvite? |
| 22:49.41 | leif[mobile] | what was the question? |
| 22:49.58 | cusco | well... |
| 22:52.08 | leif[mobile] | welp time is up. Off to volleyball. |
| 22:54.54 | cusco | sorry |
| 22:54.56 | cusco | ouch |
| 22:54.57 | cusco | : |
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