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| 01:40.00 | tengulre | hi,all |
| 01:40.13 | tengulre | where have free g72x codecs? |
| 01:40.56 | *** join/#asterisk unspin (~unspin@S010600031d02196a.vc.shawcable.net) |
| 01:42.47 | ectospasm | tengulre: you'll have to be more specific |
| 01:43.32 | tengulre | ectospasm, why? |
| 01:44.46 | ectospasm | tengulre: Which specific codec are you looking for? I'm sure most G.72x codecs have a free reference implementation, but in order to use them commercially (and legally) you'll need to purchase a license. I don't know much about anything other than G.729, but I know it's true for that codec. |
| 01:45.26 | tengulre | ectospasm, OK, I see. |
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| 01:55.24 | carrar | tengulre: http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
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| 02:46.15 | Juggie | there is a free g729 codec avail which in some countries is legal |
| 02:46.19 | Juggie | you can find it if you look |
| 02:46.31 | Juggie | that said its nice to suport digium and buy their g729 codec |
| 02:47.30 | tengulre | Juggie, give me the URL for download . |
| 02:48.52 | Juggie | google it |
| 02:49.42 | Juggie | took me 10 seconds to find. |
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| 03:00.16 | *** join/#asterisk rift0r (~rift@ip70-162-172-202.ph.ph.cox.net) |
| 03:01.00 | rift0r | What would be the syntax for an if statement if I have a registered peer at ext 500, and whenever he dials out I want him to use trunk2 instead of the default sip trunk? |
| 03:06.21 | *** join/#asterisk rift0r (~rift@ip70-162-172-202.ph.ph.cox.net) |
| 03:07.56 | ChannelZ | well firstly a SIP peer and 'ext 500' are two totally independent unrelated things |
| 03:08.42 | rift0r | ok it is a sip peer, and the phone is registered as username 500 |
| 03:09.57 | rift0r | whenever this peer dials out, i want it to use trunk2 |
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| 03:31.23 | ChannelZ | Set(dialtrunk=${IF($["${CHANNEL(peername)}"="Bob-Softphone"]?BobsTrunk:NormalTrunk)}) |
| 03:31.56 | ChannelZ | then ${dialtrunk} would either be BobsTrunk if the peer was called 'Bob-Softphone' or it would be NormalTrunk if anything else. |
| 03:32.07 | ChannelZ | Dial(SIP/${dialtrunk}) |
| 03:32.11 | ChannelZ | or whatever |
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| 04:08.58 | *** join/#asterisk seanjohn (~admin@gateways.sheltoncomputers.com) |
| 04:09.03 | seanjohn | what is different in 1.8? |
| 04:10.51 | seanjohn | and why not wait until most are in 1.6.x before making a 1.8 |
| 04:11.02 | seanjohn | or merge the changes in 1.6 |
| 04:12.06 | ChannelZ | What? |
| 04:12.38 | ChannelZ | For the same reason it's not still called 1.0 |
| 04:12.41 | seanjohn | I'm still using 1.4, there is a 1.8 release candidate now and most aren't even in a 1.6.x branch |
| 04:13.09 | ChannelZ | Most what? most people using you mean? |
| 04:13.34 | seanjohn | yeah but the same reason people won't go to 1.6 is because of changes needed to be made on their systems for their config to keep working. |
| 04:13.48 | ChannelZ | So what? That's their failing |
| 04:14.02 | seanjohn | and now people in 1.6.1 or lower will probably have a problem going to 1.8 |
| 04:14.07 | ChannelZ | We're supposed to halt all progress until people wake up? |
| 04:14.28 | seanjohn | their failing that digium changes dialplan syntaxes or deprecates dialplan functions/applications |
| 04:14.30 | seanjohn | ?? |
| 04:15.00 | ChannelZ | Their failing that they're apparently too lazy to read a CHANGES.txt file to figure out what did. It's not that bad. |
| 04:15.09 | seanjohn | knowing them, they'll change, for instance, voicemailmain() to voicemail(main,etc..) |
| 04:15.22 | ChannelZ | So stay at 1.4 where you are. Nobody cares. |
| 04:15.55 | seanjohn | yes, I could, but to get anticipated features, I must go to 1.6, 1.8, or w/e supplies it |
| 04:16.04 | ChannelZ | If people don't upgrade then whatever they have is working well for them and they apparently don't need whatever new functionality has been put in since. It's like welfare. No motivation to change |
| 04:16.04 | seanjohn | just for one feature |
| 04:17.03 | ChannelZ | so you want new features without anything changing. Yeah that makes perfect sense. |
| 04:17.29 | seanjohn | there should be an EASY way to merge one application from another version, minded that it doesn't require other things from that version to work properly, in an older version. |
| 04:17.47 | seanjohn | I know I manually can edit the source code |
| 04:18.22 | ChannelZ | Progress isn't always EASY |
| 04:18.26 | ChannelZ | Deal with it |
| 04:18.55 | seanjohn | i didn't know there was a changes, I am reading the changelog.txt |
| 04:20.11 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 04:20.20 | seanjohn | I use swift from cepstral with 10 ports. To go to 1.6.2.x I would have to find the correct app_swift for 1.6.2 to compile |
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| 04:22.21 | ChannelZ | 3rd and 4th party software is a bitch |
| 04:23.01 | seanjohn | thanks for letting me understand channelz |
| 04:23.10 | *** part/#asterisk seanjohn (~admin@gateways.sheltoncomputers.com) |
| 04:24.18 | ChannelZ | shrugs |
| 04:25.53 | ChannelZ | I don't want the world, I just want your half. |
| 04:30.46 | carrar | I like 10th party software |
| 04:32.36 | *** join/#asterisk Faithful (~Faithful@180.194.2.70) |
| 04:48.40 | drmessano | Yeah, that seanjohn dude is annoying |
| 04:49.50 | drmessano | Sadly, if he's using app_swift and not app_cepstral, he can google and find there's been a 1.6.2 compatible release for some time |
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| 05:13.59 | ChannelZ | Yeah I noticed that too. |
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| 05:30.19 | [TK]D-Fender | checkout time, later all |
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| 05:34.11 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
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| 06:27.23 | Russ | what does "'caller ABODE AIR' is not a verboser number" mean? |
| 06:29.24 | Russ | what the heck is a verboser? |
| 06:29.34 | ectospasm | heh |
| 06:32.44 | kaldemar | Russ: sounds like you have an invalid verbosity level for app Verbose |
| 06:33.23 | Russ | ah, I'm probably not escaping what I'm passing |
| 06:34.13 | kaldemar | that may occur if the message has an unescaped comma. |
| 06:36.53 | Russ | that would be why there is an LLC on the next line |
| 06:37.23 | Russ | Verbose(caller ${CALLERID(name)}); |
| 06:37.32 | Russ | do I just change that to Verbose("caller ${CALLERID(name)}"); |
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| 06:41.02 | deonv | is there any "free" "OpenSource" multi-tenenat like App available for Asterisk? |
| 06:41.43 | kaldemar | Russ: what does ${CALLERID(name)} look like? that may not be enough, but go ahead and try. |
| 06:42.23 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 06:43.01 | Russ | I think it was "ABODE AIR, LLC" |
| 06:43.14 | Russ | but since it comes from an outside source, it could really be anything |
| 06:43.40 | ectospasm | deonv: you can configure Asterisk to be multi-tenant, and you shouldn't need anything more than the appropriate application of dialplan |
| 06:45.54 | ectospasm | basically, each tenant has their own contexts (sections) of dialplan, you can create separate voicemail domains, and each tenant can have a different SIP peer/user that they connect through. |
| 06:47.04 | deonv | ectospasm: Excellent. Thanks for the Info. Can I do this via the FreePBX gui? Or is there another GUI you'd recommend? |
| 06:47.11 | kaldemar | Russ: you could try defining the level as 0 for the command, or using app NoOp instead, as you seem to always want to print the text. |
| 06:48.13 | deonv | ectospasm: Here is a tool I came across. http://www.vecsector.com/phonecall/demo/ |
| 06:48.33 | ectospasm | deonv: I wouldn't use any GUI to set it up, it will be far too limited. |
| 06:48.53 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 06:49.01 | ectospasm | Get a copy of the book, and piece it together yourself. It may take longer, but you'll have a more robust solution in place |
| 06:49.05 | ectospasm | ~thebook |
| 06:49.06 | infobot | [thebook] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 06:49.12 | deonv | ectospasm: There is no documentation and source code is not available for download |
| 06:49.27 | ectospasm | deonv: then don't use that |
| 06:49.36 | deonv | ectospasm: Excellent. Thanks. |
| 06:49.54 | deonv | ectospasm: Thanks again for the info. Appreciate. |
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| 06:58.25 | ectospasm | deonv: no problem |
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| 06:59.49 | Fruchthoernschen | A Newbie must say realy cool software, helps me save my money ;) |
| 07:00.42 | Russ | btw, I don't know if it actually makes a difference, but it seems like early audio helps get rid of telemarketers quicker |
| 07:00.45 | Russ | Playback(custom/bell/disconnected-number,noanswer); |
| 07:01.02 | Russ | twice, then if they are still ringing, answer, play it again, then hang up |
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| 07:04.23 | TobSnyder | Hello! |
| 07:04.38 | ectospasm | *that* was weird. I had open a list of international country codes, made some notes because I couldn't get in contact with someone, and I looked back at that window looking for the next person's county, and the page was already on the right section, and my eyes went directly to the right dialing code before they looked at anything else. |
| 07:04.50 | ectospasm | yeah, it's late |
| 07:05.50 | TobSnyder | When using Asterisk 1.4.33, what would you prefer to control asterisk via PHP scripts - AMI or Call Files (or any other way?)? I want to originate calls, e.g. like Click2Dial, Make outoging calls from MeetMe conference rooms and hang them up again etc. |
| 07:06.48 | ectospasm | TobSnyder: call files, or AMI. It's your choice. |
| 07:06.56 | ectospasm | Whichever seems easiest to you. |
| 07:07.44 | TobSnyder | for my click2dial I am currently using AMI - but it seems that mostly it works, but sometimes it doesn't (like I have to click twice to get a call) |
| 07:09.07 | TobSnyder | and as I've read at some firefox addons they will not support AMI in future because of it's "unreliability" or something like that - that made me unsure |
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| 07:16.10 | the_weard | can somone plz help me with this irretating error im getting on my snom phones on asterisk |
| 07:16.13 | the_weard | http://pastebin.com/NF6iNRxd |
| 07:18.09 | henk | 317 lines of paste - find the irretating error |
| 07:18.28 | henk | feels like easter :) |
| 07:19.04 | the_weard | <PROTECTED> |
| 07:19.14 | the_weard | im not able to make outgoing calls |
| 07:20.24 | the_weard | henk one is my xlite soft phone that works fine but the other is my snom 320 that is not working on outgoing calls but is receving calls fine |
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| 07:28.18 | TobSnyder | line 287Â == Everyone is busy/congested at this time (1:0/0/1) |
| 07:28.48 | TobSnyder | Cause No. 58 - bearer capability not presently available. |
| 07:28.48 | TobSnyder | <PROTECTED> |
| 07:29.22 | TobSnyder | try to enable pri debug span X |
| 07:32.13 | TobSnyder | the_weard: if you're using ISDN, than try "pri show spans" to check what spans are available, "pri debug span X" to debug this span and than make a call again and check what Bearer Capability is sent out through ISDN / DAHDI |
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| 07:57.21 | the_weard | ok TobSnyder im using a sangoma a220 analog card |
| 07:57.37 | the_weard | that command doesent show me anything |
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| 08:04.19 | TobSnyder | ah ok |
| 08:04.54 | TobSnyder | than try to compare the logs between your SNOM and your X-Lite call |
| 08:07.03 | WIMPy | goes for a codec issue. |
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| 08:15.05 | the_weard | yes i did and there is no clear indication of what is wrong? |
| 08:15.31 | ectospasm | the_weard: did you observe the failure with SIP debug enabled ("sip set debug on")? |
| 08:32.13 | *** join/#asterisk DND (~clarencec@94.200.7.26) |
| 08:32.28 | DND | guys is there any other reporting tool aside from a2billing? |
| 08:32.44 | DND | it seems a2billing is too much for us |
| 08:32.47 | the_weard | ok setted it now |
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| 08:53.30 | jeremy_g1 | Hi, I have ordered g729 codec licenses and I have got the keys in the mailbox. I have followed the readme file but after /root/register writes the licenses file to disk, i get stuck on the next step: bech_g729 utility execution. I see this message No valid license key found. |
| 08:53.38 | jeremy_g1 | Can digium support me on this? |
| 08:54.04 | jeremy_g1 | I have ordered like 300 lics. |
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| 09:17.04 | beardy | jeremy_g1: Sounds like something an email to them would answer best? |
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| 09:25.47 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 09:27.27 | dandate2 | Hi everyone, so I found out the local ISP offers 1/2 MB T1 lines (512kb). I calculated 6 people on the phone with no computers using ulaw will use 504kb. Is the throughput really this accurate or will I need the 1MB line? Are there any other overheads I'm not factoring? |
| 09:29.42 | petern_ | so is it 0.5mbps, or a T1? |
| 09:30.19 | petern_ | a T1 is 1.544mbps |
| 09:30.21 | dandate2 | its a T1 line set at 512kb |
| 09:30.48 | dandate2 | the ISP here in the philippines offers 1/2 MB, 1MB, 2MB, 3 MB, in that increment |
| 09:31.01 | dandate2 | but 1/2 MB i'm told is only 512kb |
| 09:31.28 | dandate2 | really worried about that 8kb difference heh |
| 09:31.57 | dandate2 | but it comes in on a PRI and connects to the router via serial port |
| 09:33.33 | kaldemar | dandate2: there are some overheads you forgot since your total is that small. |
| 09:33.45 | dandate2 | =@ |
| 09:33.51 | petern_ | ulaw is 64kbps, 6 * 64 = 384kbps |
| 09:34.08 | dandate2 | i'm told that ulaw can be up to 84kbps though factoring tcpip overhead |
| 09:34.16 | petern_ | so 384 + overhead... |
| 09:34.31 | kaldemar | IP, UDP and RTP overheads |
| 09:34.37 | dandate2 | so i figured 504kb maximum if the PRI connects directly to the cisco ATA |
| 09:34.54 | kaldemar | and SIP of course causes some traffic. and RTCP. |
| 09:35.12 | kaldemar | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 09:36.22 | dandate2 | how to know if I need to check RTCP on this? |
| 09:36.48 | kaldemar | if you use RTP, you use RTCP. but RTCP overhead is minor. |
| 09:37.21 | dandate2 | mmm how to know if I use rtcp heh |
| 09:37.54 | dandate2 | I just got a basic freepbx install at a colocation and SIP trunks |
| 09:38.56 | kaldemar | SIP uses RTP, so there you go. |
| 09:39.51 | dandate2 | k |
| 09:40.49 | dandate2 | Total bandwidth (incoming and outgoing): 1003.28 Kbps |
| 09:41.05 | *** join/#asterisk timahvo1 (~rogue@196.201.218.252) |
| 09:41.43 | dandate2 | so when they give me a 1/2 MB T1. Does this provide me full streaming of that speed up and down or is it a pool that if I'm downloading at 1/2 MB I can't upload at that speed? |
| 09:42.12 | ectospasm | dandate2: that's something the ISP should answer |
| 09:42.24 | ectospasm | ...because our answer is "it depends" |
| 09:42.32 | kaldemar | dandate2: you should probably ask the ISP about the characteristics of their product. |
| 09:43.14 | dandate2 | oh man if it werent the case... I'd need more than 1MB just to cover the 3.28 kpbs!!! |
| 09:43.29 | dandate2 | thats like an extra $350/mo =0 |
| 09:44.02 | dandate2 | i think i'm going to pull my hair out or order an array of cheap DSLs heh |
| 09:44.22 | ectospasm | I've seen some services touted as "synchronous", meaning same up and down, but that term isn't pedantically correct |
| 09:44.40 | ectospasm | dandate2: that would be disastrous to use on a SIP trunk. |
| 09:44.56 | ectospasm | typically DSL has very low upload rates |
| 09:45.14 | ectospasm | ...you need something with same up and down |
| 09:45.33 | dandate2 | right but enough to handle at least 1 person using ulaw, so for 6 IP phones I could get 6 DSLs at $20/mo each heh |
| 09:46.19 | ectospasm | yeah, but are you going to set up the routing table so a call doesn't get denied if there's already one up? |
| 09:46.24 | *** join/#asterisk timahvo1 (~rogue@196.201.217.252) |
| 09:46.27 | dandate2 | assuming they want me to pay $1300/mo for a 2MB T1 if their bandwidth was pooled up and down |
| 09:46.45 | dandate2 | the calls go into a queue is that what you mean? |
| 09:47.07 | dandate2 | pbx is at a datacenter |
| 09:47.21 | ectospasm | no... if there is already a call up (doesn't matter where in the PBX it is), what happens when another call comes in? How is it going to go over the right DSL line? |
| 09:47.46 | ectospasm | ...assuming you can only get one or two concurrent calls on one DSL trunk |
| 09:48.10 | dandate2 | each DSL line will connect to a seperate ATA device |
| 09:48.40 | ectospasm | Oh, so these are remote to the Asterisk system? |
| 09:48.55 | dandate2 | no silly we're chilling in the noisy ass data center =) |
| 09:49.00 | dandate2 | yes all remote heh |
| 09:49.04 | ectospasm | ... |
| 09:49.37 | ectospasm | That would work, then, but it may be rather inflexible. |
| 09:49.49 | dandate2 | right |
| 09:50.01 | dandate2 | because of lack of redundancy i imagine |
| 09:50.05 | ectospasm | ...are these DSL trunks gonna be shared with normal Internet traffic (web, e-mail, etc) |
| 09:50.12 | dandate2 | oh hell no |
| 09:50.20 | dandate2 | this is for industrial application, there will not even be computers in the room |
| 09:50.21 | ectospasm | OK, good |
| 09:50.28 | dandate2 | going to give these call center agents manual type-writers |
| 09:51.24 | dandate2 | but uhhh..spending an extra $350 for a MB to cover 4kpbs...this is just bad business |
| 09:51.35 | ectospasm | Looks like the solution could work, then. But you'd have to get another phone line in for expansion, or if one of the extensions (ATAs) needs to move to another site, I could see it being problematic |
| 09:51.58 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 09:52.17 | ectospasm | actually, if you used a better codec than G.711 you'd probably be better off. |
| 09:53.02 | dandate2 | i'm just worried about delay times due to compression at some end |
| 09:53.16 | ectospasm | G.729 (or GSM even) would be MUCH better. Compression times are negligible. |
| 09:53.22 | dandate2 | its a hard selling campaign real fast paced. trying to get that toll quality in a 3rd world country as much as poissible |
| 09:53.44 | ectospasm | G.729 guarantees toll quality at 8Kbps |
| 09:54.16 | dandate2 | hmm how costly is the licensing? I'm seein up to 25 people calling in and waiting in queue with all 6 agents on the phone also |
| 09:54.21 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:54.28 | ectospasm | Digium sells licenses for $10/channel |
| 09:54.38 | ectospasm | (disclaimer) I work for Digium |
| 09:54.59 | ectospasm | are you planning on doing any call recording? |
| 09:55.03 | dandate2 | nah |
| 09:55.29 | ectospasm | are the agents going to be doing any meetme or three-way calls? |
| 09:55.35 | dandate2 | negative |
| 09:56.01 | ectospasm | you could probably get away with six channels for these ATAs (assuming they supported G.729) |
| 09:56.29 | ectospasm | if they don't, they should support GSM which is slightly less efficient (and slightly lower quality) at 13Kbps |
| 09:56.56 | dandate2 | 6 channels? I thought a phone call used 2 channels? or is this only seen at the pbx end |
| 09:57.27 | ectospasm | well, you'd only need one channel per ATA (one channel includes both up and down) |
| 09:57.34 | dandate2 | ok |
| 09:57.36 | ectospasm | this is only seen at the PBX |
| 09:57.47 | dandate2 | right because its sending to the IP phone and trunk at the same time |
| 09:57.57 | ectospasm | what kind of trunk does the Asterisk system have? |
| 09:58.04 | dandate2 | SIP trunk |
| 09:58.21 | ectospasm | what codec does it use? |
| 09:58.46 | dandate2 | its set to ulaw but they also offer g729 transmission from the trunk level but would have to get a lot of licensing for 25 callers i imagine |
| 09:59.47 | dandate2 | 25 licenses for the callers and 6 for the reps = 31 g729 licenses. am I calculating this correctly? |
| 09:59.54 | ectospasm | Well, if you set everyone to G.729, you can do passthrough and not use any codecs (since no transcoding will take place). The moment you go from one codec to another, though, you'll need the codec to transcode |
| 10:00.08 | dandate2 | yes thats what i was thinking |
| 10:00.34 | dandate2 | just dont want to be short licenses and have people get a busy signal though =/ |
| 10:01.02 | ectospasm | it'd be safe then to get 31 channel licenses. Also, the beefier the PBX machine the better. |
| 10:01.28 | ectospasm | ...although I haven't seen G.729 be a performance hog in a while. |
| 10:01.34 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 10:02.19 | dandate2 | PBX is pretty nice, dual-core u-server. though I'm thinking if I use G729 I could run a PBX out of the philippines and save $1,000/mo on the server rental and colocation |
| 10:03.04 | ectospasm | OK, that should be plenty (especially if all it's doing is acting as a PBX) |
| 10:03.30 | dandate2 | though I dont know where i would buy a u-server in the philippines and probably not safe to ship even through fedex heh |
| 10:03.38 | dandate2 | would i beable to run an industrial quality pbx on say a desktop computer? |
| 10:03.55 | ectospasm | I wouldn't trust a desktop machine for that purpose. |
| 10:04.07 | ectospasm | ...though it is possible |
| 10:04.37 | ectospasm | though since it's software only it would probably be OK |
| 10:05.12 | dandate2 | would be different if i were using cards>? |
| 10:06.13 | ectospasm | scenario would be different. Desktop class machines don't fit well in real-time scenarios very well. |
| 10:06.25 | dandate2 | k thats what i thought |
| 10:06.29 | ectospasm | ...whether you're using software or hardware. |
| 10:06.38 | dandate2 | probably give me the occasional processing lag |
| 10:07.05 | ectospasm | I've seen customers install our interface cards into desktop-class machines and have nothing but problems. |
| 10:07.17 | ectospasm | ...but it also happens where it... just... works.... |
| 10:07.21 | dandate2 | the electricity flow isnt very redundant here either. just turning a u-server on would probably cause brownout |
| 10:07.44 | ectospasm | yeah, that's always a concern. |
| 10:08.22 | petern_ | what's a u-server? |
| 10:08.29 | dandate2 | rack server |
| 10:08.37 | dandate2 | flatbed |
| 10:08.40 | ectospasm | I assume you mean 1U |
| 10:08.41 | dandate2 | hella noisy |
| 10:08.44 | dandate2 | right |
| 10:08.54 | petern_ | ah, 1U server |
| 10:09.07 | petern_ | size doesn't really indicate power consumption though :) |
| 10:09.14 | ectospasm | true dat |
| 10:09.26 | petern_ | intel atom 1U servers, for example... |
| 10:09.50 | ectospasm | ...and just because it has a loud fan doesn't mean it consumes a lot of power. |
| 10:10.03 | dandate2 | alright so ill talk to the isp and find out if their t1 bandwidth is pooled up/down. consider running 6 basic DSL lines and test it with ulaw, go with g729 if its crap |
| 10:10.22 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 10:10.42 | ectospasm | dandate2: do you already have the six analog/DSL lines at the call center site? |
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| 10:10.46 | dandate2 | *whew* all this while marketing supervising serving and hustling (flex) |
| 10:11.01 | dandate2 | no right now we just have 1 business class DSL and it sucks |
| 10:11.07 | dandate2 | every day is quality problems |
| 10:11.24 | dandate2 | and the reps have computers and are opening youtube |
| 10:11.27 | ectospasm | yeah, but I assume the provider isn't gonna give you six lines with no installation fees |
| 10:11.38 | ectospasm | dandate2: install squid |
| 10:11.41 | dandate2 | oh its the philippines, everything here costs in pennies heh |
| 10:11.43 | Russ | dandate2, autotune the news increases productivity at least 2x! |
| 10:12.14 | ectospasm | ...only allow sites necessary for their job, no others. |
| 10:12.15 | dandate2 | like $25 installation fee |
| 10:12.29 | dandate2 | right thats what i realized |
| 10:12.40 | dandate2 | got to get rid of these damn computers and go back to manual type-writer |
| 10:13.05 | ectospasm | um, if you installed squid and set it up right that wouldn't be necessary |
| 10:13.26 | dandate2 | well its a waste of electricity, i dont want to run a bunch of computers just so they can use notepad and skype |
| 10:13.35 | dandate2 | skype chatting with people in the same room lol |
| 10:13.40 | fors1 | I'm currently setting up asterisk in one of our satellite offices. This one is located in US, and to be honest I don't know much about the US numbering plan. Is there any good documentation on how outgoing routing should be configured, and an explanation on how the numbering system works? (area codes, long distance, short distance etc etc). |
| 10:13.42 | ectospasm | ouch |
| 10:14.11 | ectospasm | fors1: http://en.wikipedia.org/wiki/NANP |
| 10:14.32 | fors1 | ectospasm: that should get me started, thanks :) |
| 10:14.38 | *** join/#asterisk Failrar (~Failrar@5ED66E6D.cm-7-7b.dynamic.ziggo.nl) |
| 10:15.16 | ectospasm | fors1: all countries in NANP begin with 1 (as you know). Depends on the telco whether you have 7 digit or 10 digit local numbers |
| 10:16.22 | ectospasm | 1(NXX)-NXX-XXXX is general form, where (NXX) is the area code, -NXX- is the exchange, and -XXXX is the extension. (This is using Asterisk extension pattern notation) |
| 10:17.06 | ectospasm | there is no way to tell by looking at a number whether it is landline, VoIP, or mobile. |
| 10:17.33 | fors1 | yeah. so, basically, if you want to call the neighbor building, the extension is sufficient? Or do you always need to include the exchange? |
| 10:17.36 | dandate2 | unless its really ugly looking you can assume its voip |
| 10:17.53 | ectospasm | fors1: always need the exchange, unless you don't go through the PSTN |
| 10:18.23 | ectospasm | fors1: if it's through the PSTN, you'll need to consult the local telco's documentation whether the three-digit area code is required. |
| 10:18.46 | ectospasm | fors1: also, some locales require the leading 1 when dialing local, others don't. |
| 10:18.59 | fors1 | ok. how about special (national) numbers? like 911, is there any 4 / 5 / 6 digit numbers that is reachable without area / exchange number ? |
| 10:19.08 | ectospasm | usually the telco will respond with a message if you dial incorrectly |
| 10:19.36 | Russ | the number of rules and exceptions is scary |
| 10:20.08 | ectospasm | X11 numbers can be passed to the telco with no extra digits. I don't know of any telco that will route 4-6 digits. |
| 10:20.25 | Russ | they used to, some rural areas might still do it |
| 10:20.30 | ectospasm | Essentially, it's 7 digits in same calling area (not necessarily same area code) |
| 10:20.41 | ectospasm | Russ: right, but that's becoming less common. |
| 10:20.52 | ectospasm | ...or 10 digits in same calling area |
| 10:21.26 | fors1 | thanks for all the input. I feel like I'm stepping into a minefield here. |
| 10:21.43 | Russ | there are even some places where international calls are "local calls" |
| 10:22.12 | Russ | that's why I like dialing out on a VoIP provider |
| 10:22.20 | ectospasm | ...and some international calls will be NANP numbers, so they seem to be US |
| 10:22.20 | fors1 | one last question. here in norway it's common to use the "0" to get an external line (when you're behind a PBX). Is it the same in US? Don't want to confuse the employees over there. |
| 10:22.26 | Russ | I hate that telco's give you the "it is not necessary to dial a 1..." |
| 10:22.34 | ectospasm | fors1: usueally it's '9' here |
| 10:22.41 | ectospasm | fors1: but that's configurable by you |
| 10:23.27 | Russ | ectospasm, and sat phone calls |
| 10:23.34 | fors1 | yeah, I know. Just don't want to create extra confusion on the employees, want to do the "common" thing. So they would expect to dial 9 991 for emergencies? or is some numbers excluded from "get an external line first" |
| 10:24.00 | Russ | http://en.wikipedia.org/wiki/North_American_Numbering_Plan is a good place to scan |
| 10:24.01 | fors1 | s/991/911 |
| 10:24.15 | ectospasm | fors1: you should probably route 911! to emergency services immediately |
| 10:24.27 | ectospasm | put both 911! and 9911! in your dialplan |
| 10:24.32 | ectospasm | ...AND TEST IT! |
| 10:24.47 | Russ | I usually test by adding an identical rule for 511 |
| 10:25.01 | ectospasm | you should really test by calling the actual 911 |
| 10:25.17 | ectospasm | they don't mind, if you keep the testing at a minimum |
| 10:25.20 | fors1 | ok. and the operator don't mind me testing it? |
| 10:25.32 | ectospasm | they actually want you to test |
| 10:25.53 | ectospasm | ...if someone really needs it, and it *doesn't* work, the company running the PBX is liable. |
| 10:26.07 | ectospasm | ...or can be |
| 10:26.08 | fors1 | great! thanks a lot for all your help. will read through the wiki article and make sure at least the emergency number works :) |
| 10:26.41 | ectospasm | fors1: you can always dial 18005551212 to test long distance |
| 10:28.45 | ectospasm | fors1: and then find the number of a local movie theater to test local |
| 10:28.59 | ectospasm | (typically they play recordings, but ymmv) |
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| 11:28.07 | jeffik | is it possible to retrieve a voice mail after user deleted it? |
| 11:28.45 | *** join/#asterisk jmkgreen (~chatzilla@wish-hq3.gotadsl.co.uk) |
| 11:29.41 | jmkgreen | we're getting reports that some dtmf tones are not being registered; we're running a pretty old asterisk connecting to a voip provider but since it's a random problem we're wondering if there are particular places to look? |
| 11:30.18 | jmkgreen | unfortunately when we switch on debugging the volume of calls means the service stutters badly which doesn't help matter |
| 11:31.28 | joobie | jeffik, negative.. but check out the folder where ur voicemal is stored.. just do a ls -R in there to be sure |
| 11:31.39 | joobie | im not sure if there's a folder it mvoes to when the user deletes.. but that will tell u |
| 11:32.31 | joobie | jmkgreen, u need to record the call to hear it |
| 11:32.34 | joobie | it could be packetloss |
| 11:32.45 | jeffik | joobie: tks, I will look there |
| 11:32.54 | joobie | but given ur box shits itself when it records |
| 11:32.58 | joobie | that's not a good sign |
| 11:33.04 | joobie | i mean it's not much i/o to record?? |
| 11:33.06 | jmkgreen | joobie: hm, not good then |
| 11:33.28 | joobie | i duno man |
| 11:33.34 | joobie | i had a similar problem about a week ago |
| 11:33.54 | joobie | but it was that a user tries to access voicemail and puts in the correct pass, but they get a prompt saying incorrect pass |
| 11:33.57 | jeffik | joobie: can you tell me where folder is? |
| 11:34.04 | joobie | works everytime OK.. just from this international caller dialling in |
| 11:34.18 | joobie | so i setup call recording to actually hear the call to see if the tones come through correctly |
| 11:34.28 | joobie | duno how else u would debug that scenario |
| 11:34.55 | joobie | jeffik, duno man.. just search ur asterisk tree for one of the voicemail extensions |
| 11:35.02 | joobie | it creates a dir with the voicemail extension |
| 11:35.06 | joobie | and dumps the crap below that tree |
| 11:35.23 | joobie | sorry, dont have access to an asterisk box atm to check |
| 11:35.33 | jeffik | i think i found it var/spool/asterisk/voicemail |
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| 11:41.03 | jeffik | joobie: tks, but seems it's really gone |
| 11:41.19 | joobie | yea |
| 11:41.30 | joobie | that sounds like the location |
| 11:41.35 | joobie | and sounds like it would delete it:P |
| 11:41.46 | joobie | i mean the user did ask it to be deleted - this aint imap with purge ;P |
| 11:43.09 | jeffik | understand, I told him probably not but I would check, it's the CFO (pays my bill) so wanted to give it a try |
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| 12:12.31 | ectospasm | jeffik: it's gone jim. Once a user marks it for deletion and hangs up, it's gone. |
| 12:13.02 | ectospasm | ...unless you make backups of that directory on a very regular basis |
| 12:13.55 | ectospasm | I could see setting up a backup tool to run every 10 minutes or so, and backup voicemail messages, but that requires foresight. |
| 12:18.33 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
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| 12:20.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:23.00 | c0rnoTa | Hello all |
| 12:24.02 | c0rnoTa | why after "channel.c: Got a FRAME_CONTROL ( 8 ) frame on channel DAHDI/5-1" asterisk hangups the call. (send disconnect frame)? |
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| 12:25.52 | dacm_work | Hi guys, |
| 12:26.32 | dacm_work | When trying to make or receive calls through a SIP trunk I get messages about not having an audio codec available. |
| 12:26.49 | dacm_work | What steps should I take to debug this issue? |
| 12:26.50 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 12:29.52 | c0rnoTa | dacm_work: try to 'sip set debug ip ...' |
| 12:31.22 | c0rnoTa | dacm_work: and take a loot at Audio codecs compatable.. There should be line like "we have: .. they have:.... combined: ..." |
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| 12:45.34 | *** join/#asterisk xoveruk (~xover@193.220.59.2) |
| 12:45.42 | xoveruk | hi all |
| 12:46.27 | xoveruk | I have a phone that will not register on asterisk. I have enabled sip debugging and checked the error logs. I can log into the phone but there is no record of it attempting registration |
| 12:46.49 | [TK]D-Fender | xoveruk: then packets aren't reaching * |
| 12:47.10 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:48.46 | xoveruk | TK, where do I go from here? |
| 12:49.08 | xoveruk | if i enable debug does it log this to /var/logs/? |
| 12:49.59 | [TK]D-Fender | xoveruk: If you see nothing in sip debug then there is nothing to look for in logs |
| 12:50.04 | [TK]D-Fender | xoveruk: Packets aren't arriving |
| 12:50.21 | [TK]D-Fender | xoveruk: Check your firewalls, networking and where you have the phone pointing to |
| 12:51.13 | *** join/#asterisk dacm_work (~dan@host109-156-247-158.range109-156.btcentralplus.com) |
| 12:51.43 | dacm_work | Hmm bit worried about this internet connection if I want to run voip... |
| 12:52.39 | dacm_work | Hi guys, |
| 12:52.42 | dacm_work | When trying to make or receive calls through a SIP trunk I get messages about not having an audio codec available. |
| 12:52.45 | dacm_work | What steps should I take to debug this issue? |
| 12:53.11 | dacm_work | I have data from trying to receive a call with sip debug on. |
| 12:53.27 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:54.48 | dacm_work | http://fpaste.org/pdrK/raw/ |
| 12:58.32 | c0rnoTa | dacm_work: take a look on it: Capabilities: us - 0x0 (nothing), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 12:59.33 | c0rnoTa | dacm_work: "sip show peer Orbtalk" output? |
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| 13:04.07 | TobSnyder | does automon => *1 work without wW in dial options? |
| 13:05.08 | krion | damn you "channel.c: Avoiding deadlock for channel" |
| 13:05.39 | xoveruk | thanks TK |
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| 13:06.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:11.31 | krion | what could be the cause of a flood of avoiding deadlock for channel 0x |
| 13:14.18 | the_weard | http://pastebin.com/YHfi8CDB @ ectospasm |
| 13:15.10 | the_weard | [TK]D-Fender can u plz assit me with this problemo? |
| 13:15.56 | the_weard | i can phone into my home but not out? with my xlite softphone i able to phone out but not my snom phones? |
| 13:16.19 | krion | http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging#HowToDebugaDeadLockinAsterisk |
| 13:16.22 | krion | fiuu |
| 13:16.38 | krion | i don't earn enough money in order to even try to debug the matter :) |
| 13:18.40 | c0rnoTa | krion: bad source code is the main cause of deadlock issue in my opinion :) |
| 13:18.55 | dacm_work | c0rnoTa: us - 0x0 (nothing) would that mean somewhere in my config I'm disallowing all codecs? |
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| 13:20.06 | c0rnoTa | dacm_work: it means that for peer Orbtalk all codecs disabled. allow alaw, ulaw or g729, or all of them |
| 13:20.13 | dacm_work | c0rnoTa: http://fpaste.org/V2Vy/raw/ |
| 13:20.32 | dacm_work | That says nothing too. |
| 13:20.39 | dacm_work | I wonder why they are being disabled.. |
| 13:20.56 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:22.12 | c0rnoTa | dacm_work: sip.conf ar your's disposal |
| 13:22.23 | *** join/#asterisk Faithful (~Faithful@180.194.3.183) |
| 13:22.23 | krion | c0rnoTa: i'm running stock debian's asterisk |
| 13:22.41 | dacm_work | c0rnoTa: Seems I was stupid and put disallow all after my allow lines. |
| 13:22.52 | dacm_work | c0rnoTa: Thank you very much for your help. |
| 13:24.25 | c0rnoTa | dacm_work: no thanks for me. It's household problem :) |
| 13:25.48 | c0rnoTa | krion: low data transfer with hdd could be the cause too. Take a look at your system state. There shouldn't be errors in `dmesg`. |
| 13:26.15 | Katty | GOOOOOOOOOOOD MORNING ALL YOU WONDERFUL PEOPLES!!!!!!!! |
| 13:26.22 | Katty | distributes hugs and muffins. |
| 13:26.36 | c0rnoTa | krion: and as I said, I solved a lot of deadlock issues by upgrading asterisk |
| 13:26.44 | c0rnoTa | Katty: good morning |
| 13:26.52 | c0rnoTa | and thx for hugs :)) |
| 13:29.28 | krion | c0rnoTa: for sure it would fix it i guess |
| 13:30.08 | krion | i'm running 1.6.2.10 |
| 13:30.21 | krion | should i try a rc in production :-D |
| 13:32.37 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 13:32.38 | Katty | c0rnoTa: <3 |
| 13:32.40 | Katty | SuPrSluG: ohai |
| 13:33.00 | SuPrSluG | Katty: ciao |
| 13:33.29 | SuPrSluG | Katty: dobry rano |
| 13:33.54 | SuPrSluG | ur pie looked delicious |
| 13:35.48 | Katty | thank you. |
| 13:38.26 | c0rnoTa | krion: I don't think so :)) |
| 13:38.46 | russellb | http://www.digium.com/en/mediacenter/viewpress/Digium-to-Announce-Future-of-Open-Source-Communications-During-AstriCon-2010-Keynote |
| 13:38.50 | c0rnoTa | krion: You shouldn't try 1.6.2 in production :)) |
| 13:41.22 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-202-120.w83-203.abo.wanadoo.fr) |
| 13:41.51 | merlin8282 | Hi all. Does someone know if a softphone exists, that supports the BLF feature ? |
| 13:42.23 | krion | du -hs /var/log/asterisk/messages |
| 13:42.26 | krion | 5.6G |
| 13:42.28 | krion | :-D |
| 13:42.44 | krion | merlin8282: hum, interesting question |
| 13:42.49 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 13:43.34 | krion | have you tryed ekiga ? |
| 13:44.16 | krion | c0rnoTa: really ? why not ? |
| 13:44.20 | merlin8282 | yes, I didn't find this feature |
| 13:44.44 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:44.49 | krion | i got 1.6.2.10 and looks fine, except deadlock from time to time and some "on hold" problem (related to rtp but not sure) |
| 13:45.37 | krion | i got asterisk 1.4.7 on my sbc trough |
| 13:47.36 | Katty | so. i might getting a baby squirrel. |
| 13:47.46 | Katty | a neighbor of mine found one knocked out of a nest. |
| 13:47.55 | coppice | do they taste good? |
| 13:49.06 | Katty | this is my frowny face |
| 13:49.08 | Katty | and it's frowning at you |
| 13:49.34 | c0rnoTa | krion: I don't trust 1.6 branch :)) |
| 13:50.43 | xoveruk | what is the maximum level of verbosity when connecting to asterisk using -rv? |
| 13:51.11 | *** join/#asterisk Faithful (~Faithful@180.194.3.183) |
| 13:51.30 | krion | i was thinking, like, release are sort of a debian way, 1.4 old stable , 1.6 stable, 1.8 testing |
| 13:54.16 | [TK]D-Fender | xoveruk: 6 is the largest effective value IIRC |
| 13:54.30 | [TK]D-Fender | xoveruk: And you can set it in CLI regardless fo what leve it was logging in |
| 13:55.41 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:57.05 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-tfxyfuxiqbxuaogu) |
| 13:57.11 | xoveruk | thanks |
| 13:57.43 | *** join/#asterisk timeshell_atwork (~chatzilla@206.248.136.108) |
| 13:58.01 | timeshell_atwork | Shouldn't ${CALLERID(num)::3} return the first 3 digits of the callerid? |
| 13:58.44 | wdoekes2 | 0:3 should, dunno about just :3 |
| 13:59.50 | wdoekes2 | merlin8282: twinkle |
| 14:00.04 | *** join/#asterisk mpe_ (~mpe@gate.ipvision.dk) |
| 14:01.45 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 14:02.42 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:02.45 | kaldemar | timeshell_atwork: no, you need the 0 in there. |
| 14:02.51 | deonv | is there a User GUI Portal for Asterisk? i.e See CDRs, Setup Speed-Dials, etc? |
| 14:03.06 | timeshell_atwork | Thank you, that works |
| 14:03.16 | timeshell_atwork | I was using this as my example: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg128965.html |
| 14:05.24 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:06.32 | krion | # wc -l /var/log/asterisk/messages |
| 14:06.33 | krion | 60425101 /var/log/asterisk/messages |
| 14:06.37 | krion | hahaha :) |
| 14:06.43 | krion | i must say, i've the biggest |
| 14:07.08 | kaldemar | timeshell_atwork: well, using a nearly 5 year old email regarding a severly outdated asterisk as a reference, you get that. :) |
| 14:07.23 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 14:08.38 | kaldemar | timeshell_atwork: doc/tex/channelvariables.tex, the asterisk book or even voip-info wiki are better places to look at. |
| 14:11.33 | *** join/#asterisk Faithful (~Faithful@180.194.3.199) |
| 14:11.33 | *** join/#asterisk ritztech (~ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 14:12.22 | ritztech | <PROTECTED> |
| 14:12.30 | ritztech | when i use sip its Instant to the beep |
| 14:21.35 | *** join/#asterisk BANSAL (~bansal@117.199.124.219) |
| 14:25.10 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:31.16 | leifmadsen | Another preview of some dialplan that will be part of the next #asterisk book! http://bit.ly/cBLpJe |
| 14:32.10 | leifmadsen | http://www.digium.com/en/mediacenter/viewpress/Digium-to-Announce-Future-of-Open-Source-Communications-During-AstriCon-2010-Keynote |
| 14:32.29 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:33.12 | anonymouz666 | are you using macro? isn't this depracated? |
| 14:33.17 | leifmadsen | NO! |
| 14:33.23 | leifmadsen | Macro() is not deprecated |
| 14:33.27 | leifmadsen | I wish people would stop saying that |
| 14:33.46 | leifmadsen | GoSub() is the preferred method in most cases, but in this case, using M() is the right thing |
| 14:34.06 | anonymouz666 | yes, I use GoSub |
| 14:34.11 | leifmadsen | So do I |
| 14:34.19 | anonymouz666 | but there are macros too |
| 14:35.30 | leifmadsen | *facepalm* |
| 14:35.34 | anonymouz666 | M() save me once. So I keeping use. Was the only way to keep recording once the caller transfer the call and hangups. |
| 14:35.53 | anonymouz666 | Using MixMonitor |
| 14:36.00 | leifmadsen | Qwell |
| 14:36.56 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 14:37.06 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-190-129.pools.arcor-ip.net) |
| 14:37.30 | anonymouz666 | and native atxfer |
| 14:37.44 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:39.02 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:39.48 | anonymouz666 | leifmadsen: does the book cover SRTP? |
| 14:40.20 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:40.41 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 14:43.18 | freckle | I have 2 openser boxs both serving the same SIP domain. When asterisk gets a call for that domain I want it to send to both opensers, how do I do this? |
| 14:43.59 | leifmadsen | Dial(SIP/openser1&SIP/openser2) |
| 14:44.29 | freckle | leifmadsen: the problem is how do I setup the sip.conf to refer to both openser |
| 14:44.42 | leifmadsen | [openser1] |
| 14:44.45 | leifmadsen | [openser2] |
| 14:44.59 | leifmadsen | freckle: I think you're not providing some critical information |
| 14:45.02 | freckle | if i refer to each openser by IP then the domain doesn't get preserved in the URI |
| 14:45.48 | freckle | I already have [openser1] and [openser2] in the SIP but wu |
| 14:46.12 | freckle | ...when I dial the SIP URI get's changed to the IP of the openser box |
| 14:46.21 | anonymouz666 | freckle: yes, that's correct |
| 14:46.32 | anonymouz666 | why do you need to keep the domain of RURI? |
| 14:46.34 | freckle | I want the SIP URI to preserve the domain |
| 14:46.40 | p3nguin | Don't dial by SIP URI. |
| 14:46.45 | freckle | becuase openser is doing multi domain support |
| 14:46.56 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:46.58 | anonymouz666 | freckle: you could use fromdomain= |
| 14:47.16 | freckle | anonymouz666: i need the To: preserving not the From: |
| 14:47.20 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 14:48.43 | anonymouz666 | I didn't get you |
| 14:49.23 | anonymouz666 | explain better |
| 14:49.37 | anonymouz666 | From and To headers are cosmetics only |
| 14:49.46 | anonymouz666 | nobody should route nothing based on the To: header. |
| 14:49.49 | freckle | if you use fromdomain= it changes the From:. I need the To: changing to the domain openser is serving |
| 14:49.53 | *** join/#asterisk Mhaddog (~Mhaddog_@173-110-71-160.pools.spcsdns.net) |
| 14:49.54 | anonymouz666 | you should expect the domain of the request URI |
| 14:50.10 | freckle | but I need asterisk to send the call to 2 openser with the same INVITE |
| 14:50.44 | anonymouz666 | ok, you are doing the wrong approach. |
| 14:50.44 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:50.57 | freckle | so what should I be doing? |
| 14:51.05 | anonymouz666 | OpenSER doesn't need to expect the To: header to define the domain |
| 14:51.23 | *** join/#asterisk wwalker (~wwalker@208.92.232.27) |
| 14:51.32 | freckle | anonymouz666: it is doing multi domain support |
| 14:52.08 | anonymouz666 | the multi-domain support is done watching the domain of the request URI |
| 14:52.57 | anonymouz666 | then |
| 14:53.43 | anonymouz666 | you should use something like domain module... and the function if (is_domain_local("$rd")) ... else ... |
| 14:54.11 | freckle | anonymouz666: yes I am already doing this on a single openser with 500 + domains |
| 14:54.49 | freckle | but what happens when you have more than one openser serving the same domains and you want asterisk to send a call to one of those domains |
| 14:55.22 | freckle | you want the request to fork to both servers |
| 14:55.31 | *** join/#asterisk BANSAL (~bansal@117.199.124.219) |
| 14:56.07 | anonymouz666 | doesn't make sense to me |
| 14:56.17 | *** part/#asterisk Mhaddog (~Mhaddog_@173-110-71-160.pools.spcsdns.net) |
| 14:56.26 | freckle | the UA could be registered on either Openser... |
| 14:57.00 | wwalker | I have an app that controls making calls, so the app creates virtual CDRs. Yesterday, the calls started going out with the wrong callerid #. So, I checked the app logs, # is right there. So, I then look at the asterisk CDRs. all the CDRs show |
| 14:58.31 | wwalker | the callerid as both the callerid and the DNIS. (still confirming I'm sending the right callerid to asterisk, so I assume the VOIP provider is the problem) Anyone know how to fix the CDRs? The calls are all made via AMI Originate |
| 14:59.28 | anonymouz666 | freckle: ok, answering your question... I don't know a way to change the To: header in Asterisk. |
| 15:00.06 | anonymouz666 | of course, everything is possible when you touch chan_sip.c. |
| 15:00.19 | freckle | anonymouz666: i don't think it is possible in Asterisk... i will probably have to fork the request to the other server using openser |
| 15:00.47 | anonymouz666 | for me, the approach is totally wrong |
| 15:01.10 | freckle | well I have been doing it this way for 4 years with over 10 million calls... |
| 15:01.46 | anonymouz666 | alright then, you don't need any advice from me :P |
| 15:02.12 | anonymouz666 | if it working just leave it there. |
| 15:03.17 | ritztech | ive done this before but im stuck at Outbound SIP/Trunk it says SIP/Mitel/8123 am i missing anything its all local ? |
| 15:08.39 | *** join/#asterisk krash812 (~roni@190.196.71.206) |
| 15:08.56 | blingbling | hi everybody ... what s up |
| 15:09.52 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 15:09.56 | blingbling | is there a way to send a messange when i receive a call ? |
| 15:10.21 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.35.151) |
| 15:11.06 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 15:11.10 | blingbling | receive a message when i receive a call .. something like |
| 15:11.13 | blingbling | exten => 1,1,Set(CHANNEL(musicclass)=espera) |
| 15:11.14 | blingbling | exten => 1,2,Queue(from-pstn3) |
| 15:11.44 | p3nguin | What does "send a message" mean? |
| 15:12.07 | blingbling | a way to receive a message that says support , comercial , just a message , like receive the caller id in the softphone .. |
| 15:12.36 | p3nguin | You can change the caller ID. |
| 15:13.23 | blingbling | but there is no way to do that in exten => ? |
| 15:13.38 | anonymouz666 | change the caller ID through the exten => |
| 15:13.39 | leifmadsen | blingbling: of course there is |
| 15:13.51 | leifmadsen | blingbling: exten => Set(CALLERID(name)=FOO:${CALLERID(name)}) |
| 15:13.57 | p3nguin | You're trying to say you can't change the Caller ID in the extension? |
| 15:14.11 | blingbling | thanks .. i will try that |
| 15:14.12 | leifmadsen | well at least he's saying he doesn't know how to |
| 15:14.38 | blingbling | yes , i m saying that i dont know how |
| 15:15.07 | p3nguin | I guess I misinterpreted the "there is no way to" part. |
| 15:15.16 | leifmadsen | ESL |
| 15:15.24 | bougyman | ESL? |
| 15:15.28 | bougyman | oh, different ESL. |
| 15:15.33 | leifmadsen | English as a Second Language |
| 15:15.36 | bougyman | got it |
| 15:15.43 | leifmadsen | not ASL :) |
| 15:16.00 | bougyman | i'm used to ESL in a different context in telephony, had to context switch. |
| 15:16.05 | ritztech | ive done this before but im stuck at Outbound SIP/Trunk it says SIP/Mitel/8123 am i missing anything its all local ? i tried from-pstn from-trunk im not sure where its failing |
| 15:16.10 | ritztech | inbound worls |
| 15:16.15 | ritztech | works |
| 15:16.27 | WIMPy | Might get interesting with ETL then? |
| 15:17.29 | p3nguin | SIP/Mitel/8123 means extension 8123 on the peer named Mitel. |
| 15:18.16 | p3nguin | Does sip.conf have a [Mitel] definition? |
| 15:20.26 | p3nguin | ritztech: You... ^^^^^^^^ |
| 15:20.47 | ritztech | it should looking |
| 15:21.02 | p3nguin | sip show peer Mitel |
| 15:22.25 | p3nguin | It either exists or it doesn't. |
| 15:23.23 | ChannelZ | zzzzombies |
| 15:23.33 | Katty | what does a vegan zombie say |
| 15:23.45 | ChannelZ | "please kill me" |
| 15:23.45 | Katty | grrrrrrrraaaaaaaaaainsssssssssssssssssssssssssss |
| 15:25.44 | ritztech | yes it does hav it |
| 15:26.09 | beardy | Heh |
| 15:26.12 | ritztech | not sure if i have context right |
| 15:26.13 | p3nguin | Great. Now show some type of failure. |
| 15:26.17 | ritztech | i have from-pstn haha |
| 15:27.45 | beardy | Katty: What are you eating today then? |
| 15:27.50 | Katty | lasagna. |
| 15:28.05 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:28.07 | Katty | i made thai peanut stirfry last night. |
| 15:28.20 | beardy | Oh, nice. |
| 15:28.29 | Katty | http://42ndrecipestreet.blogspot.com/2009/09/thai-peanut-chicken-and-noodles.html <- but with rice, and no meats. |
| 15:28.59 | beardy | Why no meat, are you a vegan zombie? |
| 15:29.41 | bougyman | chicken with no meat? |
| 15:29.47 | bougyman | i'm confused. |
| 15:29.47 | ritztech | where would i go look up context to make sure i can outbound dial rcorrectly |
| 15:30.22 | beardy | Chicken is included in meats.. |
| 15:31.30 | Russ | there's a zombie in your yard |
| 15:31.42 | Russ | we don't like zombies on our lawn |
| 15:32.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:33.07 | beardy | Katty: I might try that sometime. With the meat. |
| 15:33.47 | p3nguin | bougyman: Never heard of BOCA Chik'n? |
| 15:33.56 | bougyman | p3nguin: taht's not chicken. |
| 15:34.02 | p3nguin | Chicken, but without meat. |
| 15:34.13 | bougyman | that's oxymoronic. |
| 15:34.23 | bougyman | it's not even chicken-flavored. |
| 15:34.28 | bougyman | it's faux. |
| 15:34.58 | *** join/#asterisk chrism2671 (~chris@88.211.98.2) |
| 15:35.02 | p3nguin | Why would they not make it chicken flavored? |
| 15:35.18 | chrism2671 | anybody got CID lookup working over HTTP? |
| 15:35.38 | bougyman | because to be chicken flavored it would have to contain chicken! |
| 15:35.40 | chrism2671 | my Asterisk just doesn't seem to fire off the request |
| 15:36.24 | Qwell | leifmadsen: |
| 15:36.37 | leifmadsen | Qwell: hi! thanks for *facepalm* :) |
| 15:36.38 | leifmadsen | very useful |
| 15:36.57 | Qwell | huh? |
| 15:38.20 | beardy | ritztech: What do you mean go look up? Put a phone in the context, on an extension, and try calling? |
| 15:38.23 | fauxalliance | bougyman, could be worse... TVP |
| 15:38.40 | ritztech | same context as in sip/mitel ? |
| 15:38.52 | ritztech | i even tried DISA inbound and dialing outbound |
| 15:38.56 | ritztech | it just sits there ? |
| 15:40.55 | *** join/#asterisk the_weard (~arthur@41-132-111-196.dsl.mweb.co.za) |
| 15:40.56 | p3nguin | I still haven't seen any evidence from a failed/stuck call. |
| 15:41.07 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 15:42.51 | *** join/#asterisk bmoraca_work (~bmoraca@66.242.174.253) |
| 15:46.03 | ritztech | hmm in sip debug i get a From: "Unknown"<sip:Unknown@192.168.14.23>;tag=as4cc75b1d |
| 15:46.03 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
| 15:46.08 | ritztech | 14.23 is the asterisxk |
| 15:46.48 | p3nguin | At least it's a version from this century. |
| 15:46.55 | *** part/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 15:47.01 | ritztech | From: "1111"<sip:1111@192.168.14.23 THIS IS WIERD .... |
| 15:47.24 | ritztech | To: <sip:8123@192.168.14.23>;tag=as5ffff50a shouldnt it be TO 8123@192.168.14.2 |
| 15:47.28 | ritztech | 14.2 is my asterisk |
| 15:47.48 | p3nguin | oh, I misunderstood. I thought you were saying 1.4.23 was the Asterisk version. |
| 15:48.34 | Kobaz | Thank you for selecting Gaylord National\u2122 Resort & Convention Center on the Potomac. |
| 15:48.34 | ritztech | OHh haha |
| 15:48.38 | ritztech | SIP/2.0 401 Unauthorized hmm |
| 15:50.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:50.39 | carrar | h4X0r |
| 15:52.31 | Qwell | whachu talkin bout leifmadsen? |
| 15:53.12 | fracBlend | anyone got time for a rule question? |
| 15:53.40 | *** join/#asterisk jetlag (jetlag@pool-173-61-240-74.cmdnnj.east.verizon.net) |
| 15:53.53 | fracBlend | I've got a bunch of phones on CME I'm migrating over to asterisk |
| 15:54.02 | p3nguin | Ask it, and if no one answers, you'll know they have no time for it. |
| 15:54.16 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:54.18 | wcselby | o/ |
| 15:54.29 | fracBlend | I have a rule on asterisk that points back to the CME something like _0[23]XX |
| 15:54.46 | *** join/#asterisk NuclearLucifer (gavroche@gavroche.pl) |
| 15:55.01 | fracBlend | but when i move a phone over and add a rule that says _0308 or similar is now local, it still tries to go out the sip trunk |
| 15:55.21 | fracBlend | am I getting borked by sort order or seomthing? |
| 15:55.44 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 15:55.47 | ritztech | where do i find the Right Context is that where its not connecting |
| 15:55.48 | ritztech | ? |
| 15:57.01 | McBoingbo | hey guys, we are using Polycom 301's with Plantronics S12 headsets and getting a lot of complaints there is echo on the other side (poeple calling in, sales folks with headsets do not hear any echo but other end is bad) any ideas? |
| 15:57.49 | *** join/#asterisk zxvff (shahid@dev.hockingits.com) |
| 15:58.20 | Naikrovek | echo happens in places where the analog <--> digital conversion is made |
| 15:58.38 | Naikrovek | do you have telephony hardware in your asterisk server? if so, does it have echo cancellation module on it? |
| 15:58.50 | McBoingbo | no hardware, no |
| 15:59.18 | Naikrovek | so pure voip connection within your facility, up to whomever your sip provider is? |
| 16:00.05 | Naikrovek | phones > asterisk > internet > voice provider |
| 16:00.07 | Naikrovek | yes? |
| 16:01.29 | Naikrovek | McBoingbo: where'd you go |
| 16:02.41 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 16:03.12 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-202-120.w83-203.abo.wanadoo.fr) |
| 16:06.15 | *** join/#asterisk JimDickenson (~dickenson@c-24-20-160-143.hsd1.or.comcast.net) |
| 16:07.06 | wcselby | Naikrovek - there you go again, scaring off the customers |
| 16:07.25 | Naikrovek | i felt i was being helpful |
| 16:07.30 | wcselby | you were |
| 16:07.31 | wcselby | :) |
| 16:07.33 | Naikrovek | but maybe i'm psychotic and don't know it |
| 16:07.33 | wcselby | I was being silly |
| 16:07.40 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 16:07.52 | wcselby | if you were, would you know it? that's an interesting question.... |
| 16:07.59 | Naikrovek | maybe this whole IRC channel is a hallucination of mine |
| 16:08.59 | wwalker | Naikrovek: what IRC channel? |
| 16:09.10 | drmessano | Mom, is that you? |
| 16:09.14 | Naikrovek | I KNEW IT OMG WTF BBQ LOL |
| 16:09.28 | drmessano | Wait, I thought this was 4chan |
| 16:09.34 | WIMPy | puts on Imgaination -- "Just an Illusion" |
| 16:11.20 | McBoingbo | sorry Naikrovek userland pulled me away |
| 16:11.28 | Naikrovek | np man |
| 16:11.36 | Naikrovek | or woman, however you are |
| 16:11.40 | Naikrovek | dunno |
| 16:12.02 | McBoingbo | I dunno, research showed Plantronics S12 works really well with Polycom 301, but man its shit, echo no matter how low you put the amplification |
| 16:12.02 | Naikrovek | so it's voip for you from phones to asterisk to voice provider, yes? |
| 16:12.14 | McBoingbo | yes |
| 16:12.27 | Naikrovek | do you hear the echo on your side or do callers hear the echo |
| 16:12.33 | McBoingbo | callers hear it |
| 16:12.42 | McBoingbo | it sounds great from the headset side hehe |
| 16:12.43 | Naikrovek | okay, and no echo if they pick up the handset and use that? |
| 16:12.46 | Naikrovek | okay |
| 16:12.48 | Naikrovek | well |
| 16:12.55 | McBoingbo | no echo when handset is picked up yes |
| 16:13.20 | Naikrovek | sounds like the plantronics is the problem |
| 16:13.39 | Naikrovek | wonder if you can adjust the polycom's echo cancellation to work better with the headsets |
| 16:13.49 | McBoingbo | yeah, I took the headset out of the amplification system and put straight into phone, still echo |
| 16:14.07 | Naikrovek | googles. |
| 16:14.41 | Naikrovek | can you turn off noise cancellation |
| 16:14.46 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:14.50 | McBoingbo | turn off? |
| 16:15.01 | Naikrovek | sorry |
| 16:15.08 | *** join/#asterisk reallost1 (~reallost@adsl-065-015-161-123.sip.mem.bellsouth.net) |
| 16:15.27 | wcselby | get setup with a vendor with a nice return policy and test out several different headsets |
| 16:15.39 | Naikrovek | well, yeah the noise cancelling. so stuff external to the headset is muted, the headset actively mutes that stuff so your call is clearer |
| 16:15.42 | wcselby | i've honestly never heard of a headset causing echo |
| 16:15.45 | wcselby | but it's possible |
| 16:15.50 | Naikrovek | i haven't either |
| 16:15.51 | bougyman | amps do it all the time |
| 16:15.57 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 16:16.01 | sbrath | I have a call coming in from a PSTN caller, to a SIP phone number, and I get about 27 seconds of dead air before trixbox "rings" the caller. Also, I don't see trixbox "get" the call till about 15 seconds. Is there any way to get the telco to simulate ringing as the call is routed? |
| 16:16.17 | Naikrovek | the noise cancelling microphone - can you turn off the noise cancelling part, and just make it a straight mic? |
| 16:16.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 16:16.39 | sbrath | I know it's trixbox, but I think this is more a SIP call thing, I can also try routing the call to my asterisk from source server. |
| 16:16.42 | ChannelZ | yeesh, what are you on, a modem? |
| 16:17.02 | sbrath | The connection is a 6M/1M Cable |
| 16:17.14 | ChannelZ | and it takes 30 seconds to figure out where to go? |
| 16:17.23 | p3nguin | So then you are using a modem. |
| 16:17.31 | McBoingbo | dont think I can disable noise cancel Naik |
| 16:17.52 | McBoingbo | only options I see on this S12 is the ability to make amplification lower/higher |
| 16:17.59 | Naikrovek | hmm |
| 16:18.01 | sbrath | You call from my ATT cell in wisconsin, to a 800# that routes to a 920xxxxxxx number hosted by les.net SIP provider. |
| 16:18.11 | McBoingbo | yeah the S12 system uses an amp |
| 16:19.07 | McBoingbo | I thought it was the amp causing the echo on the caller side, but I stuck the plantronics s12 just the headset into the headset plug of the polycom 301, and still echo on caller side |
| 16:20.04 | Naikrovek | i'm still leaning towards the noise cancellation feature of that mic. presumably it's an active system or they wouldn't mention it in the product literature |
| 16:20.12 | Naikrovek | but i don't know |
| 16:20.34 | McBoingbo | can someone hook me up with a simple rj9 headset that will work with polycom 301 that ships to Canada? |
| 16:20.37 | Naikrovek | if you can find a way to plug that headset into a computer or something and see if there is echo there that might help the troubleshooting process |
| 16:20.57 | fauxalliance | dyke it out, one hop at a time, then nuke the offender ;-) |
| 16:21.13 | *** join/#asterisk usjew1 (~alex@96.56.99.206) |
| 16:21.19 | usjew1 | hi |
| 16:22.18 | McBoingbo | I am grabbing some lunch, will be back though to see if you guys can help out with my echo problem, thanks for the help so far! |
| 16:22.39 | Naikrovek | you can probably go to radio shack and pick up a simple headset right now if you want |
| 16:22.50 | Qwell | radioshack? |
| 16:22.56 | usjew1 | does anybody has experience with chan_gtalk? |
| 16:22.59 | Naikrovek | they have all kinds of phone stuff like that |
| 16:23.07 | Naikrovek | it's expensive but they let you return things |
| 16:23.17 | Qwell | all they have now are RC cars and stereos |
| 16:23.22 | Qwell | oh, and cell phones |
| 16:23.28 | wcselby | i was about to say |
| 16:23.30 | Qwell | ...and batteries, obviously |
| 16:23.31 | Naikrovek | they have lots of phone stuff in there, including headsets |
| 16:23.34 | Naikrovek | maybe it's just my local one |
| 16:23.41 | Naikrovek | mine has tons of hobby electronics stuff too |
| 16:23.42 | Qwell | ever priced them? |
| 16:23.54 | Qwell | $17 for a resistor. No thanks. |
| 16:23.57 | Naikrovek | yeah they're insanely expensive but he wants something he can test with then return |
| 16:24.00 | Naikrovek | $17 lol |
| 16:24.02 | Naikrovek | not quite |
| 16:24.03 | Naikrovek | but yes |
| 16:24.05 | Naikrovek | <PROTECTED> |
| 16:24.08 | wcselby | http://www.radioshack.com/family/index.jsp?categoryId=2032097&s=null |
| 16:24.33 | Qwell | You've got questions, we've got batteries. |
| 16:25.30 | ChannelZ | and the most expensive adapters on Earth |
| 16:26.11 | usjew1 | have anybody managed to use gtalk to call PSTN for free and get calls from PSTN using the new google voice service? |
| 16:26.14 | [TK]D-Fender | Or the lowest quality shit house(ish) brand equivalent that you'd never want to touch |
| 16:26.22 | wcselby | usjew1 - no |
| 16:26.41 | usjew1 | wcselby: should it be possible? |
| 16:26.59 | wcselby | usjew1 - the last conversation I saw in here said no, but people were planning on going out there and testing |
| 16:27.10 | *** join/#asterisk Russ (~russ@149-169-247-21.nat.asu.edu) |
| 16:27.31 | wcselby | usjew1 - so we'll see. i've got either chan_gtalk or chan_jingle (i forget which) sending me IM's with call details when calls come in though |
| 16:27.41 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:28.45 | usjew1 | wcselby: u have any experience with getting gtalk to work in general,i have the setup,i can atleast "get" calls from my cell phone,however my clienet doesnt ring i just see the message in the logs |
| 16:29.32 | wcselby | you mean outside of asterisk? |
| 16:29.56 | drmessano | the Gtalk in Asterisk does not work with GtalkWeb, which means the calls from GVoice wont work |
| 16:30.08 | ritztech | can i do like Record 2 at once ? exten => s,n,Record(${PAGES}~${MonitorFile1}.wav,,,k) |
| 16:30.18 | drmessano | There is a branch out there with working code in it, but who knows if it will ever get merged |
| 16:30.24 | usjew1 | wcselby: i am having a feeling that one of 3 things is wrong either bug in the asterisk i am using 1.6 latest from asterisk web site,NAT however atleast i should get the client tell me i got the call i dont care for sound just yet,or more likely is context extentions bussnes,but i am not gettiong any errors from asterisk so its weird |
| 16:30.45 | drmessano | usjew1, ^^^ |
| 16:30.47 | ritztech | exten => s,n,Record(${PAGES}~${MonitorFile1}.wav,,,k AND {MonitorFile2}.wav,,,k) With just only 1 record |
| 16:30.48 | ritztech | ? |
| 16:31.18 | [TK]D-Fender | ritztech: No, but you can just copy the file after. |
| 16:31.27 | usjew1 | drmessano: what is GtalkWeb and how is it defferent from normal gtalk as supported by chan_gtalk |
| 16:31.34 | [TK]D-Fender | ratAnd we lost touch for planning the deployment for your project... |
| 16:31.41 | [TK]D-Fender | ritztech: And we lost touch for planning the deployment for your project... |
| 16:32.08 | ritztech | it was harsh i have another job and wokring 24 straight days haha |
| 16:32.13 | [TK]D-Fender | is off to lunch |
| 16:32.20 | drmessano | usjew1, There's a different RTP setup I believe.. I am not completely sure of the differences.. if you google for Gtalk-web or gtalkweb, you will find the associated Asterisk bugs and links to the branches where the working code exists |
| 16:32.26 | [TK]D-Fender | ritztech: Another job? As in left the one you were at? |
| 16:32.32 | [TK]D-Fender | ritor jsut MORE? |
| 16:32.42 | *** join/#asterisk Faithful (~Faithful@180.194.0.87) |
| 16:32.42 | [TK]D-Fender | ritztech: or just MORE? |
| 16:32.44 | usjew1 | i see thanx |
| 16:32.45 | ritztech | i have 2 with a bunch of side work |
| 16:33.05 | [TK]D-Fender | gotcha... BBIAB |
| 16:33.08 | drmessano | usjew1, There is indeed something different.. and it has to do with calls made from the Gmail web client, but also affects the GVoice call setup in the same way |
| 16:34.26 | drmessano | usjew1, Apparently the way jingle is implemented in a few IM clients also differs from Asterisk's implementation.. there's yet another branch addressing it, with working code, but no clue on any plans to merge it |
| 16:34.41 | *** join/#asterisk dacm_work (~dan@host109-152-190-173.range109-152.btcentralplus.com) |
| 16:34.41 | drmessano | I was hoping for 1.8.. but.. *sigh* |
| 16:35.35 | usjew1 | drmessano:yea i saw ur twit u getting famous ;-) |
| 16:35.55 | wcselby | did he just call you a twit? |
| 16:36.03 | *** join/#asterisk Faithful (~Faithful@180.194.0.87) |
| 16:36.06 | usjew1 | drmessano:so u saying their jingle codec is diffrent between web and desktop client? |
| 16:36.06 | drmessano | I am already infamous.. Fame should be right around the corner. or jail. |
| 16:36.19 | fauxalliance | wcselby, tweeter _is_ for twits ;-) |
| 16:36.20 | usjew1 | lol |
| 16:36.43 | drmessano | usjew1, not codec.. it's more in the call setup or something with the RTP stream |
| 16:36.52 | usjew1 | ok |
| 16:37.33 | usjew1 | drmessano:what definatly be awsome if they get it to work in 1.8,bye-bye phone bills |
| 16:38.24 | Naikrovek | aah |
| 16:38.32 | Naikrovek | i rediscovered smoething i had buried in the bookmarks |
| 16:38.36 | Naikrovek | hilarity |
| 16:38.37 | Naikrovek | http://sleeptalkinman.blogspot.com/ |
| 16:39.01 | drmessano | usjew1, yep.. But 1.8 is at RC already, so it won't make it. I may end up setting up a minimal FreeSWITCH install on the same box as I think it's working there |
| 16:40.09 | usjew1 | ok sounds good |
| 16:40.53 | drmessano | Just a thought.. I am not big on the idea.. but even if it's 6 months before 1.10, that's 6 more months to wait |
| 16:40.56 | leifmadsen | funny enough, we're working on that right now. chan_gtalk already exists, but appears to maybe have a bug or two that is causing Google Voice calls from work |
| 16:41.05 | leifmadsen | it will likely be working by 1.8.0 |
| 16:41.08 | usjew1 | drmessano:i will look into it,free calls is nervana which phreakers for the last 40years tried to accomplish,we can get it now without getting arrested,historical times |
| 16:41.26 | drmessano | leifmadsen: I have seen the two tickets.. and they haven't moved much in ages.. a bump by you, a test in the last month by someone else. |
| 16:41.49 | leifmadsen | drmessano: I'm not telling you I hope they get fixed, I'm telling you they will be getting fixed soon |
| 16:42.16 | *** join/#asterisk MasterXP (~IceChat7@190.166.28.14) |
| 16:42.16 | drmessano | Ok, does that mean the one GtalkWeb issue AND the other Jingle issue? |
| 16:42.23 | leifmadsen | Jingle is not related to this |
| 16:42.32 | paulc | JabberSend: I want to be able to use \n to do line breaks but it seems JabberSend passes through the literal string. The coder in me says it's got to be easy to do the substitution in there somewhere. The lack of Asterisk/C code experience says "Argghh! Go to the pub!" - anyone know the magic sauce I need? |
| 16:42.33 | leifmadsen | I'm talking about chan_gtalk, not chan_jingle |
| 16:43.23 | drmessano | Well, there is yet another bug in Jingle that makes it more or less useless as it stands now |
| 16:43.35 | drmessano | https://issues.asterisk.org/view.php?id=15634 |
| 16:43.57 | leifmadsen | possibly a true statement |
| 16:44.02 | leifmadsen | I'm just talking about chan_gtalk |
| 16:44.35 | drmessano | It would be nice to see them both working for 1.8.. then Gtalk and Jingle calls would both work |
| 16:44.50 | leifmadsen | a lot of things would be nice |
| 16:45.14 | leifmadsen | I'll take what I can get |
| 16:45.39 | Juggie | leifmadsen: nub |
| 16:45.40 | drmessano | Well, if Jingle isn't working, it should be removed from 1.8 then with the fixed gtalk support left in... |
| 16:45.48 | *** kick/#asterisk [Juggie!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (nub THIS) |
| 16:45.52 | thehar | hahahahaha |
| 16:45.56 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 16:45.59 | leifmadsen | <3 |
| 16:46.00 | Juggie | haahah :) |
| 16:46.03 | thehar | leif 1 others 0 |
| 16:46.11 | leifmadsen | leifmadsen++ |
| 16:46.13 | Juggie | i change my mind. leif is my hero. |
| 16:46.17 | leifmadsen | ~score |
| 16:46.17 | infobot | Your Personal Self-Government Score is 40%, Your Economic Self-Government Score is 90%. |
| 16:46.18 | thehar | leifmadsen++++ |
| 16:46.25 | thehar | lol |
| 16:48.48 | usjew1 | btw wanted to ask you on technical level whats difrfrent between Asterisk and FreeSwitch,like architecture,configuration,supported protocls,etc.. |
| 16:49.07 | paulc | backs away slowly |
| 16:49.29 | Qwell | rolls his eyes |
| 16:49.55 | fauxalliance | smirks |
| 16:50.03 | sbrath | anyone have a how-to on using DIDforSale with a nat'd asterisk behind an iptables firewall? I guess I need to proxy in udp 5060. |
| 16:50.34 | Naikrovek | usjew1: read about that on the freeswitch site. he lays it all out. |
| 16:50.50 | Qwell | With absolutely no bias or incorrect information AT ALL. |
| 16:50.52 | Naikrovek | i don't know much about either so i can't tell you anything regarding the differences |
| 16:50.53 | Naikrovek | yeah |
| 16:51.02 | Naikrovek | he's clearly biased but i dunno how much |
| 16:51.06 | Qwell | (and certainly no outright lies.) |
| 16:51.25 | *** join/#asterisk d-tech (~d-dtech@72.245.233.107) |
| 16:53.33 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 16:53.35 | bmoraca_work | <cleveland voice>erverybody lies</cleveland voice> |
| 16:54.27 | Naikrovek | i thought that was the House tagline |
| 16:54.32 | Naikrovek | they say it in every single episode |
| 16:54.51 | bmoraca_work | yeah, but House doesn't sound as funny as Cleveland when he says "erverybody" |
| 16:54.56 | Naikrovek | true |
| 16:55.15 | Naikrovek | though when you hear that actor's normal speech you may be surprised to learn that he's british |
| 16:55.17 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 16:55.24 | Naikrovek | in here i'm sure that's common knowledge but my wife was spooked by it |
| 16:55.27 | bmoraca_work | i'm not surprised |
| 16:57.11 | fullstop | I'm using the macro-stdexten on voip-info.org, and something is baffling me. Here's the log: http://pastebin.com/33vu0RuY |
| 16:58.27 | fullstop | s,n,Goto(s-${DIALSTATUS},1) should go to s-NOANSWER.. but it doesn't match for some reason. It ends up matching the "_s-." pattern. |
| 16:58.29 | leifmadsen | what is baffling you? |
| 16:59.09 | Kobaz | fullstop: probably because you have a _s-. defined before s-NOANSWER |
| 16:59.27 | fullstop | Nope.. It's the last line in the macro |
| 16:59.40 | Kobaz | do a dialplan show on that context |
| 16:59.48 | Kobaz | and pastebin |
| 17:00.37 | fullstop | http://pastebin.com/4mkSQVsi |
| 17:01.11 | leifmadsen | the order defined in extensions.conf has no bearing on the order of matching |
| 17:01.59 | fullstop | leifmadsen: if I remove _s-., it just bails on the channel |
| 17:02.01 | Kobaz | leifmadsen: i never looked at the internals of the matching... but i found the defined order to be very important |
| 17:02.24 | leifmadsen | Kobaz: the defined order in extensions.conf does not matter -- it is sorted when loaded into memory based on rules |
| 17:02.41 | leifmadsen | fullstop: then something is wrong with the first priority of s-NOANSWER |
| 17:02.44 | leifmadsen | like Voicemail() doesn't exist |
| 17:02.46 | Kobaz | fullstop: it's not going to match s-NOANSWER |
| 17:02.46 | fullstop | does it match the most specific pattern |
| 17:03.01 | leifmadsen | fullstop: yes |
| 17:03.20 | Kobaz | you need to do this: [s][-][N][O][A][N][S][W][E][R] |
| 17:03.26 | fullstop | silly question, but how can I verify that Voicemail() exists ? |
| 17:03.42 | Kobaz | fullstop: it's nothing to do with voicemail |
| 17:03.48 | Kobaz | fullstop: your extension is not matching |
| 17:04.00 | leifmadsen | Kobaz: no he doesn't |
| 17:04.06 | leifmadsen | s-NOANSWER is a literal, not a pattern match |
| 17:04.31 | Kobaz | leifmadsen: i've had to do character set matching like that, when the text wasn't matching for me |
| 17:04.33 | leifmadsen | fullstop: core show application voicemail -or- module show like voicemail |
| 17:04.38 | fullstop | voicemail show users gives output.. so I assume that I have 'Voicemail()' |
| 17:04.43 | leifmadsen | Kobaz: only if you prefix with a _ do you need that |
| 17:04.53 | Kobaz | hmm, k |
| 17:05.04 | Naikrovek | _ means "everything that follows is a pattern" |
| 17:05.06 | Kobaz | well in any case, there's something wrong with the exten |
| 17:05.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 17:05.09 | Kobaz | Naikrovek: yeah |
| 17:05.32 | fullstop | even if I did not have voicemail, wouldn't it step into that part of the dialplan and spew some sort of error message? |
| 17:05.45 | leifmadsen | yes, in which case, you need to do: _[n]ever-XXX if you want to match: never-123 |
| 17:05.47 | Kobaz | yeah it would |
| 17:05.55 | leifmadsen | yes |
| 17:06.08 | *** join/#asterisk Faithful (~Faithful@180.194.3.112) |
| 17:06.17 | Kobaz | leifmadsen: oh, just the first one needs to be []'d ? |
| 17:06.58 | Kobaz | that's just a weird problem |
| 17:07.04 | Kobaz | pastebin your console output? |
| 17:07.21 | leifmadsen | with debug enabled |
| 17:07.57 | leifmadsen | I'd also like to see "dialplan show macro-stdexten" |
| 17:08.08 | wcselby | Kobaz - only certain, reserved characters, need to be escaped like that |
| 17:08.11 | leifmadsen | oh you already showed that |
| 17:08.20 | wcselby | Kobaz - n, i, s, t, .... one or two others I think |
| 17:08.26 | Kobaz | wcselby: oh okay |
| 17:08.27 | leifmadsen | N, Z, X, ., ! |
| 17:08.31 | wcselby | yeah |
| 17:08.33 | wcselby | there you go |
| 17:08.43 | leifmadsen | i, s, t, are extensions, not pattern match characters |
| 17:08.54 | fullstop | leifmadsen: which debug do you want on? |
| 17:09.02 | fullstop | verbosity is at 20 right now |
| 17:09.10 | leifmadsen | fullstop: actually, I need to go to lunch now, so someone else will need to help you |
| 17:09.16 | fullstop | okay |
| 17:09.19 | fullstop | enjoy your food! |
| 17:09.20 | wcselby | leifmadsen - ahh, it's been a while since I had to deal with those, so yeah... |
| 17:10.17 | wcselby | fullstop - do a "dialplan show macro-stdexten" in the CLI and pastebin that |
| 17:10.56 | fullstop | wait |
| 17:11.01 | fullstop | might have figured out what it is |
| 17:11.12 | Naikrovek | what is it |
| 17:11.13 | Naikrovek | erm |
| 17:11.14 | Naikrovek | what was it |
| 17:13.21 | fullstop | It doesn't work with extenpatternmatchnew=yes set. |
| 17:14.21 | fullstop | probably a bug |
| 17:14.32 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 17:15.06 | [TK]D-Fender | No, clearly a FEATURE |
| 17:15.57 | fullstop | I don't want any voicemail anyway. :) |
| 17:16.33 | Kobaz | yes, all the bugs in asterisk have been squashed long ago |
| 17:16.52 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 17:17.16 | wcselby | fullstop, if you comment out that line in your dialplan (the _s-.,), does it do what you'd expect it to? |
| 17:18.38 | wcselby | meh |
| 17:18.39 | wcselby | must eat now |
| 17:19.09 | Katty | hhhhhhhhhhheeeeeelllllllllllooooooooooooooooooooooooo nurse. |
| 17:21.48 | fullstop | wcselby: nope. it just bails on the channel |
| 17:21.57 | fullstop | I am not a nurse. Just Dr. Dad at times. |
| 17:22.20 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 17:22.35 | [TK]D-Fender | fullstop: PASETBIN |
| 17:22.39 | [TK]D-Fender | PASTEBIN even |
| 17:23.08 | fullstop | [TK]D-Fender: I have, but is there anything else you want to see? |
| 17:23.20 | [TK]D-Fender | fullstop: Full currnet and the call |
| 17:23.28 | fullstop | It never matches with extenpatternmatchnew=yes, but does match with extenpatternmatchnew=no |
| 17:24.51 | *** part/#asterisk fracBlend (~fracBlend@unaffiliated/fracblend) |
| 17:25.19 | [TK]D-Fender | PASTEBIN |
| 17:26.55 | fullstop | http://pastebin.com/pEsZRXfY |
| 17:27.41 | fullstop | * version 1.6.2.11 |
| 17:27.54 | [TK]D-Fender | fullstop: '_s-.' => 1. Hangup() [pbx_config] <-- he said to REMOVE that and test |
| 17:28.21 | [TK]D-Fender | fullstop: So remove it and test with extenpatternmatchnew=yes |
| 17:28.37 | fullstop | I have already presented that pastebin.. let me find it. |
| 17:29.02 | Letoric | hi [TK]D-Fender - any chance you know where the coloring for vi is coming from, when I edit an asterisk config file as a regular user? |
| 17:29.08 | Letoric | I would love to get that in place for root ;p |
| 17:29.24 | [TK]D-Fender | Letoric: Don't use vi... |
| 17:29.29 | Letoric | :/ |
| 17:29.40 | fullstop | http://pastebin.com/UemaEf0r |
| 17:29.49 | Letoric | my boss has a hardon for vi, I'm kind of stuck with it |
| 17:30.23 | Letoric | I see that vim has it the coloring, which is nice |
| 17:31.18 | p3nguin | letoric: It's vim, and it is syntaxing that provides the color. :syntax on |
| 17:31.43 | fullstop | [TK]D-Fender: amended with dialplan show macro-stdexten -- http://pastebin.com/f3N4yWG5 |
| 17:31.55 | Letoric | p3nguin: Thank you ;) |
| 17:32.14 | fullstop | Letoric: :colorscheme elf-lord is nice |
| 17:32.20 | p3nguin | A lot of systems have a link from vim to vi, so you'll never know you're really using vim. |
| 17:32.48 | fullstop | elflord, that is |
| 17:33.45 | ChannelZ | You'll know when every key you hit does something retarded |
| 17:33.54 | fullstop | I actually like vim.. but I hate when I stumble upon a solaris system and they have the ancient vi which does not allow you to move the cursor while in insert mode. |
| 17:34.19 | fullstop | and you end up with [B[B[B[U all over your screen |
| 17:34.27 | Letoric | Heh, I've grown used to that, but it has taken a lot of painful undo's to do so ;> |
| 17:34.41 | [TK]D-Fender | fullstop: that is looking like a bug now... |
| 17:35.02 | fullstop | I use busybox's vi implementation on some systems.. there is no undo! It just prints "not implemented!" |
| 17:35.19 | [TK]D-Fender | fullstop: I can understand a prioritization issue between a sparse pattern vs specific, but not a failur to find a match with a speciific alone that exists |
| 17:35.24 | fullstop | [TK]D-Fender: is there any way I can help debug it further? |
| 17:35.51 | [TK]D-Fender | fullstop: No, I think that las one pretty much says it. You have only 1 possible match, and it looks 100% accurate yet doesn't |
| 17:36.00 | fullstop | okay |
| 17:36.37 | fullstop | I will turn off the new matching for now. Should I submit a bug report? |
| 17:36.42 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 17:38.16 | [TK]D-Fender | fullstop: I'd say yes |
| 17:39.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:41.12 | fullstop | [TK]D-Fender: What category would you put it in? Core/general ? |
| 17:42.08 | [TK]D-Fender | fullstop: Nore sure of what better palce, but it sounds about right |
| 17:47.27 | bmoraca_work | nano > vi :P |
| 17:51.12 | fullstop | pico > nano... :-P |
| 17:51.48 | [TK]D-Fender | more>less |
| 17:51.51 | bmoraca_work | my knowledge of vim is ":wq" and "i" |
| 17:52.58 | carrar | What else do you need to know |
| 17:53.02 | p3nguin | Hmm. I haven't used "i" since the advent of the Insert key. |
| 17:53.04 | fullstop | bacon>pie>cake |
| 17:53.14 | Qwell | baconcake>pie |
| 17:53.16 | fullstop | I still use i. It's closer than insert |
| 17:53.24 | bmoraca_work | p3nguin: the Insert key doesn't work when connected to a system via serial console |
| 17:53.35 | carrar | Are you suggesting there are other editing commands? :) |
| 17:53.40 | p3nguin | I'll keep that in mind. |
| 17:53.49 | fullstop | I'll have to think about the baconcake |
| 17:53.54 | bmoraca_work | it's important |
| 17:54.12 | p3nguin | Have you seen the candied bacon cookies? |
| 17:54.53 | bmoraca_work | the little supermicro boxes i use for my asterisk servers are really neat...i can redirect all BIOS and POST output to serial, and then with linux, I can use a serial TTY. makes it that much more appliance-like |
| 17:55.57 | fullstop | I think I will actually make bacon ice cream. |
| 17:56.21 | bmoraca_work | i made watermellon sorbet once...it was pretty good |
| 17:57.21 | bmoraca_work | although i was so busy this summer that i never got around to getting the ice cream maker out |
| 17:57.46 | *** join/#asterisk toothkit (~betch2k@66.87.4.124) |
| 17:57.54 | p3nguin | I was just saying yesterday that I would like to make a batch of blackberry ice cream. |
| 17:58.01 | bmoraca_work | ooo |
| 17:58.03 | Qwell | blackberry isn't bacon. |
| 17:58.03 | bmoraca_work | that sounds good |
| 17:58.10 | bmoraca_work | bacon sorbet? |
| 17:58.31 | fullstop | I got 3 watermelons this year. I would have had 4, but my wife picked one when I was out of town... and it was white on the inside. |
| 17:58.32 | bmoraca_work | i wonder how fine my food processor could grind the bacon up...or maybe someone makes bacon powder...hmmmmm |
| 17:58.47 | fullstop | a blender would probably do a better job |
| 17:59.08 | fullstop | the food processor would make nice bacon bits, though. |
| 17:59.20 | leifmadsen | BACON! |
| 17:59.38 | fullstop | get a mortar and pestle and go to town. |
| 17:59.46 | bmoraca_work | lol |
| 18:00.07 | bmoraca_work | hmm |
| 18:00.26 | tzanger | http://erkie.github.com/ |
| 18:00.28 | tzanger | SFW, fun waste of time (on a real browser) |
| 18:00.30 | bmoraca_work | i'm getting off early today...maybe i'll dig out the ice cream machine and make something fancy for my sister's birthday tomorrow |
| 18:00.39 | leifmadsen | tzanger: that thing is fun! |
| 18:00.48 | tzanger | so's your mom |
| 18:00.51 | leifmadsen | I'm putting the rest of my BBQ together tonight and making burgers |
| 18:00.58 | leifmadsen | tzanger: my moms dead |
| 18:01.15 | p3nguin | Crack open a cold one. |
| 18:01.22 | tzanger | yeah, she doesn't make a lot of noise |
| 18:02.21 | leifmadsen | tzanger: O.O |
| 18:02.48 | bmoraca_work | necrophelia jokes...yay! |
| 18:03.51 | tzanger | wwhat kind of bbq did you get? |
| 18:04.04 | tzanger | is craving some pulled pork now |
| 18:08.05 | leifmadsen | tzanger: I have 10 lbs of pork butt in the freezer for that :) |
| 18:08.23 | tzanger | woot, party at leif's |
| 18:08.43 | tzanger | I'm gonna go get me a jimmy john's sub for lunch though |
| 18:08.51 | tzanger | ultimate porker, #17 with cheese |
| 18:08.56 | tzanger | total awesome |
| 18:10.47 | leifmadsen | tzanger: yummy -- I go this BBQ: http://www.charbroil.com/ProductInfo/67-430-1834/Commercial-Series-500-Threeburner-Infrared.aspx |
| 18:12.27 | tzanger | .. infrared? |
| 18:12.56 | Qwell | yeah I don't think they understand what infrared means.. |
| 18:13.11 | tzanger | but it's QUANTUM INFRARED |
| 18:13.19 | Qwell | they also have *GAS* infrared grills. |
| 18:13.39 | tzanger | on the drive down yesterday I swear I had some infrared gas |
| 18:13.49 | tzanger | I'm very happy it was just me in the car |
| 18:13.56 | tzanger | although I am positive I have rotted out the foam in the seat |
| 18:15.02 | bmoraca_work | electric grill? eww :P |
| 18:15.29 | bmoraca_work | oh, wait...that's a gas infrared gril...who the heck does that work? |
| 18:15.42 | Qwell | <Qwell> yeah I don't think they understand what infrared means.. |
| 18:15.56 | bmoraca_work | indeed |
| 18:16.02 | fullstop | tzanger: I always thought that a funny idea for a movie would be to show years of people sitting in airplane seats, eating beans and expelling large quantities of gas. Then, the plane makes an unexpected water landing and someone uses that seat as a flotation device, allowing years of flatulence to be released and causing them to pass out. |
| 18:16.25 | Qwell | umm |
| 18:16.34 | Qwell | really? you think that'd be a movie? a funny one, at that? |
| 18:16.47 | fullstop | oh no, part of a movie. |
| 18:16.53 | tzanger | it wouldn't be a bad youtube clip |
| 18:17.13 | fullstop | Have you seen piranha 3d? |
| 18:17.20 | fullstop | You don't need much these days. |
| 18:17.54 | bmoraca_work | 3D boobs...makes it ALMOST worth it |
| 18:18.03 | bmoraca_work | bad moveis are still bad, though |
| 18:18.23 | fullstop | Shark Attack 3: Megalodon is one of my favorite bad movies. |
| 18:18.34 | fullstop | It's absolutely awful. |
| 18:18.41 | fullstop | ...but so fun |
| 18:19.13 | bmoraca_work | i watched Megashark vs. Giant Octopus on Netflix...that was a terrible movie, lol |
| 18:19.18 | fullstop | I still don't know how they roped John Barrowman into that one. |
| 18:19.35 | fullstop | I've not seen that one. Someday soon. |
| 18:19.53 | bmoraca_work | i haven't gotten around to watching Killer Clowns from Outer Space or Ghost in a Teeny Bikini yet |
| 18:20.04 | bmoraca_work | so many aweful B movies on Netflix and so little itme |
| 18:20.17 | Naikrovek | i watched Xanadu once |
| 18:20.28 | Naikrovek | one of the worst movies ever made |
| 18:20.39 | fullstop | They actually made a sequel to shark attack 3, but it was titled Shark in Venice. It starred a baldwin brother (not the successful one) and Scarlett Johansson's older sister. |
| 18:21.10 | bmoraca_work | Scarlett Johanssen isn't bad...but there are better |
| 18:21.17 | fullstop | It's her sister. |
| 18:21.33 | fullstop | the one you never hear about that stars in sub D-grade movies. |
| 18:22.06 | [TK]D-Fender | Sub D-Cup? |
| 18:22.20 | Naikrovek | vanessa johansson if you wanna google her |
| 18:22.25 | fullstop | imdb doesn't have a photo |
| 18:22.30 | Naikrovek | google does |
| 18:22.38 | Naikrovek | http://www.google.com/imgres?imgurl=http://www.cinemahorror.it/images_bank/news/2008/settembre/Vanessa-Johansson-01.jpg&imgrefurl=http://www.cinemahorror.it/notizie/notizia.asp%3Fid%3D1555&h=345&w=230&sz=19&tbnid=gk8bVnBYLwHHQM:&tbnh=275&tbnw=183&prev=/images%3Fq%3DVanessa%2BJohansson&zoom=1&q=Vanessa+Johansson&usg=__r3tfSCgxLO7pjobuOlC4nlKbLD4=&sa=X&ei=yoOjTO7tDJKenge1mdGIBA&ved=0CBYQ9QEwBg |
| 18:22.42 | Naikrovek | oh man what a link that is |
| 18:22.43 | Naikrovek | sorry |
| 18:23.29 | Naikrovek | cripes she looks awful as a blonde, too |
| 18:23.35 | Naikrovek | stay natural vanessa |
| 18:23.37 | fullstop | So, regarding airplane seat cushion flatulence.. there has been much worse. |
| 18:23.58 | fullstop | and, yes, I would find a clip of that scene amusing. |
| 18:24.01 | Naikrovek | i sat in a theater to watch transformers once and wound up sitting in a freshly peed-on seat |
| 18:24.12 | Naikrovek | got my money back and then some |
| 18:24.15 | fullstop | it wasn't that scary. |
| 18:24.18 | fullstop | :-P |
| 18:24.28 | fullstop | no need to wet yourself |
| 18:24.33 | Naikrovek | heh |
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| 19:01.48 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 19:01.58 | ManxPower | I just want to say Adtran support is *awesome* |
| 19:03.10 | bmoraca_work | i've never had a problem dealing with them |
| 19:04.25 | ManxPower | bmoraca_work, it is refreshing to have a tech call you back less than 15 mins after submitting the ticket, he knew what he was doing, and solved both issues I was having. |
| 19:04.57 | carrar | "Did you plug the power cord in" |
| 19:04.59 | bmoraca_work | yep, i've never talked to one who couldn't help me |
| 19:05.10 | bmoraca_work | do you mind my asking what the issues you were having were? |
| 19:05.20 | bmoraca_work | (curious) |
| 19:05.20 | ManxPower | I toured their HQ and labs last year. very impressive. |
| 19:05.44 | ManxPower | bmoraca_work, My permit/deny in the VPN selector ACL did not match the permit/deny on the VPN server. |
| 19:05.50 | bmoraca_work | ahhh |
| 19:05.58 | ManxPower | I was lazy and specified a much larger range on my side. |
| 19:06.02 | bmoraca_work | yeah, Adtrans are more picky than Ciscos as far as that |
| 19:06.39 | bmoraca_work | i've experienced that before when connecting an Adtran NetVanta to a Cisco IOS router using IPSec VPNs |
| 19:06.49 | ManxPower | bmoraca_work, you know about dhcp excluded ranges, right? |
| 19:06.54 | bmoraca_work | yeah |
| 19:07.11 | bmoraca_work | or is there some bug with Adtran and htem? |
| 19:07.13 | ManxPower | It seems Cisco and Adtran treat them the opposite of how ISC dhcpd treat them. |
| 19:07.23 | fullstop | So.. when recording voicemail as g729, when my stream is g729 passthrough.. why does it try to go to slin as an intermediate format? |
| 19:07.35 | fullstop | Can't it just write the 729 stream? |
| 19:08.03 | bmoraca_work | Cisco IOS and Adtran have their DHCP server setups very similar from what I recall...both specify the network to the DHCP server and then refine it by specifying excluded addresses |
| 19:08.03 | ManxPower | fullstop, what formats do you allow in your voicemail.conf? |
| 19:08.08 | fullstop | g729 |
| 19:08.16 | ManxPower | fullstop, NO other formats? |
| 19:08.31 | fullstop | no other formats |
| 19:08.41 | fullstop | http://pastebin.com/JSeiqBX9 |
| 19:08.55 | ManxPower | fullstop, and ALL your VM prompts are in g729 format and you are not answering the line first and don't have any weird dial lines involved? |
| 19:09.14 | fullstop | correct. all audio files are 729 encoded |
| 19:09.19 | fullstop | just a single line |
| 19:09.37 | ManxPower | bmoraca_work, on ISC if you don't exclude the addresses of your statically defined hosts it gets upset. In adtran you must NOT include those IPs in your statically defined gosts. |
| 19:10.51 | bmoraca_work | ahhh |
| 19:11.30 | bmoraca_work | are you speaking about static reservations or statically assigned IPs on the device? |
| 19:11.56 | ManxPower | static reservations (i.e. hardware-address THEMAC for a "pool" of 1 ip |
| 19:12.05 | bmoraca_work | gotcha |
| 19:12.12 | bmoraca_work | well, i can see the merits of either |
| 19:12.40 | ManxPower | at home virtually all my hosts are static reservations |
| 19:12.50 | bmoraca_work | with Cisco, you have to create a whole separate DHCP pool to do reservations. haven't done them with Adtran because I usually have a server onsite that does DHCP |
| 19:13.02 | ManxPower | I got tired to connecting to the router to find the IP of my TiVo, for example. |
| 19:13.22 | ManxPower | bmoraca_work, they do it exactly the same way as Cisco. |
| 19:13.28 | bmoraca_work | ahh |
| 19:13.33 | ManxPower | but different from the ISC DHCP I use on linux almost every day. |
| 19:13.35 | bmoraca_work | i always thought that was clunky as hell |
| 19:14.06 | bmoraca_work | i use Windows DHCP server most times...never really let me down, though it can have problems with multiple scopes and helper addresses |
| 19:14.24 | bmoraca_work | i usually use the Cisco layer 3 switch to provide DHCP if I'm doing voice vlans |
| 19:14.27 | ManxPower | fullstop, See this? Executing [s@macro-stdexten:5] Dial("SIP/2612-00000022", "SIP/2611,17,tTr") in new stack |
| 19:14.33 | bmoraca_work | for the voice vlan, that is |
| 19:14.37 | ManxPower | you can't do that and expect passthru to work |
| 19:15.24 | fullstop | ManxPower: the phone didn't answer, and it goes to VM. |
| 19:15.34 | fullstop | Also, that phone has a 729 license |
| 19:16.06 | ChannelZ | I use post-it notes with IPs scribbled on them |
| 19:16.30 | *** join/#asterisk Faithful (~Faithful@180.194.2.32) |
| 19:16.30 | ManxPower | fullstop, do a "sip show channels" while the VM app is playing the prompts. |
| 19:16.48 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:17.05 | myster | I use post-it notes with different colors. Shapes if it's a printer. |
| 19:18.36 | ManxPower | bmoraca_work, my boss bought me a netvanta 3120 to replace my Cisco 1721 |
| 19:19.00 | fullstop | ManxPower: http://pastebin.com/ayVKbXq9 |
| 19:19.23 | fullstop | I found a solution. |
| 19:19.31 | fullstop | http://forums.digium.com/viewtopic.php?f=1&t=3429 |
| 19:19.48 | bmoraca_work | ManxPower: I like the 3120s...they're cute. they're way weak in the VPN department, though. don't try to put more than 2 on it or you'll have lots of problems |
| 19:20.38 | Qwell | fullstop: that Qwell guy seems pretty awesome. |
| 19:20.44 | fullstop | He sure is. |
| 19:20.45 | ManxPower | bmoraca_work, two is the most I would ever use, also most of my traffic is non-vpn. |
| 19:20.54 | fullstop | Even if he turns into the incredible hulk at times. :-P |
| 19:21.01 | Qwell | indeed! |
| 19:21.57 | bmoraca_work | fullstop: if you specify "t" or "T" as dial options (or "w" or "W"), asterisk has to be in the media path, which means you need to be able to transcode g729 |
| 19:22.14 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 19:22.40 | tzanger | hmm, I really need to get a system in play that after 10 invalid sip registrations from an IP it auto-drops them on the iptables level |
| 19:23.25 | ManxPower | tzafrir_laptop, fail2ban |
| 19:23.45 | fullstop | bmoraca_work: the server that I am connecting with over IAX does have a 729 license. |
| 19:24.01 | carrar | HOT++ |
| 19:24.02 | p3nguin | Takes like five minutes to set up that with fail2ban. |
| 19:24.15 | fullstop | However, the one I am connecting these phones to is not intel, and I can't get 729 licenses for it. |
| 19:24.17 | ManxPower | tzafrir_laptop, surprisingly easy to set up too. |
| 19:24.21 | bmoraca_work | fullstop: the server that is issuing the Dial() with the tT needs to have the g729 licenses |
| 19:24.34 | tzanger | p3nguin: true |
| 19:24.35 | ManxPower | fullstop, um, phones use chips that have g729 in them. |
| 19:24.47 | bmoraca_work | fail2ban is awesome |
| 19:24.51 | fullstop | We are talking about 2 different things. |
| 19:24.51 | ManxPower | I'm not aware of any IP phone that does not support G729, I'm sure there are some |
| 19:25.02 | fullstop | yes, the phone has 729 |
| 19:25.11 | fullstop | the server which the phone connects to does not. |
| 19:25.23 | ManxPower | fullstop, Sounds like poor planning to me. 8-) |
| 19:25.28 | fullstop | the server which that asterisk server talks to, the one which interfaces with dahdi, does have 729 licenses. |
| 19:25.48 | ManxPower | Is your non-Intel box a PIKA or some other PoS? |
| 19:25.56 | bmoraca_work | fullstop: if you're trying to record a voicemail in g729 on a server that doesn't have g729 licenses, you cannot have asterisk in the media path (which means no DTMF transfer (tT) or call recording (wW)) |
| 19:26.52 | fullstop | not pika, but it is arm based. |
| 19:27.29 | bmoraca_work | fullstop: why not use a Supermicro 5015A? tiny 1U box that's easily wall-mountable and is Intel-based |
| 19:27.39 | ManxPower | I have little interest in running a PBX that is so underpowered they need to use busybox. |
| 19:28.38 | fullstop | This box is $80, and we are using it to put phones at remote locations over a vpn tunnel. |
| 19:29.01 | fullstop | Not a lot of phones, but a few here and there. It is powerful enough for this purpose. |
| 19:29.09 | bmoraca_work | why not just put the phone there and have it connect straight back? |
| 19:29.14 | fullstop | no vpn |
| 19:29.35 | p3nguin | Supermicro 5015A ... Atom based? |
| 19:29.42 | bmoraca_work | p3nguin: yep |
| 19:30.01 | fullstop | you are concerned about invalid sip registrations and setting up fail2ban.. but I only have to be concerned with a single port. |
| 19:30.16 | bmoraca_work | fullstop: so you've got your little $80 box doing a VPN and asterisk? |
| 19:30.21 | fullstop | yes |
| 19:30.39 | fullstop | and if it goes tits-up, it can be replaced easily. |
| 19:31.03 | carrar | We're gonna need to vote on that |
| 19:31.06 | bmoraca_work | fullstop: well, your only real option, then, is to use g711 as the codec. or maybe iLBC and have the box transcode to ulaw to the phones (unless your phones support ilbc) |
| 19:31.13 | bougyman | fullstop: we're using atom netbooks for that. |
| 19:31.17 | Katty | pants |
| 19:31.23 | carrar | shirts |
| 19:31.29 | bougyman | a bit pricier, but not a whole lot, and you get a built-in console and wifi router. |
| 19:31.37 | bougyman | plus ups |
| 19:31.48 | fullstop | bmoraca_work: I don't think you understand... I have voicemail working... |
| 19:31.49 | fullstop | and.. |
| 19:32.03 | tzanger | lacy thong and bra...er... right. <whistles> |
| 19:32.04 | fullstop | I don't need call recording for in-office calls. |
| 19:32.36 | fullstop | I need call recording on calls going into a queue, all coming from the server with dahdi cards. |
| 19:32.58 | jdoe | tzanger: it's okay, I like to feel pretty too. |
| 19:33.18 | fullstop | For this, tT and wW works -- the dtmf is ignored by the * server on the ARM and the parent server takes care of this. |
| 19:33.32 | bmoraca_work | fullstop: your issue is that the voicemails are recording in slin and not g729, right? well, I've given you the answer to that. if you don't care about that, then there's no need to ask a question. however, it's likely that all of your calls are being changed down to g711 because you're attempting to do something which can't be done (transcode calls on a box which can't transcode g729) |
| 19:33.42 | fullstop | bmoraca_work: read up a bit |
| 19:33.57 | tzanger | jdoe: :-) |
| 19:34.00 | fullstop | bmoraca_work: setting an option (no silence detection) allowed my VM to work as 729. |
| 19:34.20 | fullstop | There is no need for asterisk to transcode the VM, other than to watch for silence. |
| 19:35.10 | fullstop | bmoraca_work: http://forums.digium.com/viewtopic.php?f=1&t=3429 |
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| 19:40.40 | Katty | carrar: ha |
| 19:40.55 | Katty | carrar: i worked out. now i am hungry. feed me :< |
| 19:42.05 | Kobaz | what did you do? |
| 19:43.09 | fullstop | guesses zumba |
| 19:43.44 | carrar | feeds katty cheesecake |
| 19:44.02 | thehar | NOM |
| 19:46.41 | p3nguin | bmoraca_work: There isn't room inside that chassis for four hard drives, is there? I see the specs say 4x SATA. |
| 19:46.50 | bmoraca_work | no |
| 19:47.07 | bmoraca_work | you can fit two 2.5" SATA drives if you buy an optional caddy |
| 19:47.21 | bmoraca_work | i only ever put 1 3.5" drive |
| 19:47.22 | p3nguin | They expect you to hang the disks off the back, or what? |
| 19:47.40 | fullstop | How comparable is the supermicro atom to a core i3? |
| 19:47.43 | bmoraca_work | i'd never put that many drives in one of those |
| 19:48.02 | bmoraca_work | fullstop: not possible to compare. vastly different architecture (Atom is in-order, for instance) |
| 19:48.10 | bmoraca_work | so, it works great for some things, not so good for others |
| 19:48.22 | bmoraca_work | as a pbx, though, it works pretty well |
| 19:48.27 | fullstop | I'm speaking in terms of asterisk.. :-) |
| 19:48.32 | p3nguin | I'd imagine they're good for a PBX. |
| 19:48.34 | fullstop | overall performance |
| 19:48.59 | Kobaz | axeterisk |
| 19:49.03 | p3nguin | I run a PIII Coppermine, so I'm sure the Atom can do just fine. |
| 19:49.30 | fullstop | my $80 box is 400MHz MIPS.. |
| 19:49.39 | p3nguin | Even at 50 calls, Asterisk doesn't use over 5% CPU. |
| 19:49.46 | fullstop | which might beat the coppermine. ;-) |
| 19:50.22 | Kobaz | as long as you dont do any transcoding, you can support bunches of calls |
| 19:50.26 | Kobaz | bunches of oats |
| 19:50.37 | tzanger | that reminds me I better get over to jimmy john's |
| 19:50.52 | fullstop | tzanger: they had $1 subs in philadelphia the other day |
| 19:51.08 | tzanger | axeterisk, that's when you run it on heavy metal. |
| 19:51.13 | tzanger | fullstop: nice |
| 19:51.20 | Kobaz | i got pizza in philly last week |
| 19:51.23 | Kobaz | oh man that was good |
| 19:51.27 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:51.33 | fullstop | You are in Michegan on contract, right? |
| 19:51.39 | p3nguin | Even with all 50 calls in a MeetMe, it barely goes above idle. |
| 19:51.42 | fullstop | wow, where did my brain go |
| 19:51.45 | Kobaz | fullstop: who? |
| 19:51.58 | fullstop | Kobaz: tzanger, in Michigan |
| 19:52.02 | Kobaz | oh |
| 19:52.04 | Kobaz | i dunno |
| 19:52.34 | tzanger | fullstop: yes |
| 19:53.05 | fullstop | we had the conversation about JJ's before. |
| 19:53.21 | fullstop | I was surprised that they had JJ's in canada.. but you're not in the great white north right now. |
| 19:53.54 | tzanger | um |
| 19:53.59 | tzanger | michigan is definitely great white north :-) |
| 19:54.14 | *** join/#asterisk fracBlend (~fracBlend@unaffiliated/fracblend) |
| 19:54.28 | fracBlend | Whee, figured out 9.0.3 firmware |
| 19:54.41 | fracBlend | anyone got any tips on getting mwi to work? |
| 19:54.54 | fullstop | no no no. The line is clearly drawn at the border. |
| 19:55.13 | fullstop | Everything south of there is rainbows, green grass and sunshine. |
| 19:56.02 | tzanger | fullstop: haha |
| 19:56.37 | wcselby | hey Naikrovek - did your kindle ever come in? |
| 19:57.02 | Katty | carrar: <3 |
| 19:57.18 | Katty | carrar: i'm actually making cheesecake tonight or tomorrow night |
| 19:57.40 | p3nguin | I like cheesecake. |
| 19:58.02 | p3nguin | Probably a little too much. |
| 19:59.17 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 20:02.58 | leifmadsen | p3nguin: cheesecake <3 |
| 20:03.12 | Kobaz | beefcake! |
| 20:03.22 | thehar | yes leif is |
| 20:03.50 | leifmadsen | flexes |
| 20:04.04 | thehar | diggity |
| 20:04.32 | Katty | p3nguin: i'm making oreo cheesecake. |
| 20:04.41 | leifmadsen | pounces on Katty |
| 20:04.43 | Katty | p3nguin: the kind with oreo crust, and little oreos stuck into the cheesecake |
| 20:04.55 | Katty | hugs leifmadsen |
| 20:05.02 | leifmadsen | Katty: I think I will go finish building my BBQ now so I can make burgers... |
| 20:05.09 | leifmadsen | OH! I need to call the wife to pick up propane |
| 20:05.18 | Katty | that sounds very manly. |
| 20:05.18 | thehar | waits for invite and flight |
| 20:05.20 | Katty | i approve. |
| 20:05.28 | leifmadsen | Katty: I am a man! |
| 20:05.40 | Katty | most excellent. |
| 20:05.44 | leifmadsen | thehar: you are invited -- I do not provide travel :) |
| 20:05.51 | thehar | haha |
| 20:05.53 | thehar | shakes fist |
| 20:05.59 | thehar | i'll bring good american beers! |
| 20:06.04 | wcselby | lol |
| 20:06.19 | Katty | do those two words actually go in the same sentance? |
| 20:06.23 | Katty | erm 3 |
| 20:06.23 | thehar | yes |
| 20:06.29 | *** join/#asterisk vader-- (vader@c-71-225-201-226.hsd1.nj.comcast.net) |
| 20:06.32 | thehar | fightin' words |
| 20:06.41 | Katty | i hate beer. |
| 20:06.44 | Katty | it's disgusting. |
| 20:06.47 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 20:06.50 | Qwell | Katty: boo |
| 20:06.56 | thehar | Qwell: exactly |
| 20:07.05 | Katty | Qwell: more for you. |
| 20:07.31 | LemensTS | Hey All. Anyone know if the Polycom boot methods/cfgs are the same across 231, 501, and 550's? |
| 20:07.58 | LemensTS | Sorry not cfg's, formats |
| 20:09.42 | *** part/#asterisk fracBlend (~fracBlend@unaffiliated/fracblend) |
| 20:10.05 | fullstop | sorry, beer is awesome.. and there are plenty of good american brews. |
| 20:10.11 | leifmadsen | I love beer |
| 20:10.13 | leifmadsen | I also love wine |
| 20:10.22 | fullstop | leifmadsen: where are you located? |
| 20:10.28 | leifmadsen | Toronto, ON, Canada |
| 20:10.47 | fullstop | you should have a good selection there. |
| 20:11.10 | leifmadsen | fullstop: yep, plenty of beers at The Beer Store |
| 20:11.44 | fullstop | I don't know if Souther Tier ships up there, but if you like wheat beer their Hop Sun is tasty. |
| 20:12.00 | LemensTS | Get some Spotted Cow from Minnesotta if you ever are close to there, mmmm |
| 20:12.07 | leifmadsen | fullstop: ya haven't heard of that one |
| 20:12.21 | LemensTS | I mean Wisconsin not Minessotta |
| 20:12.24 | fullstop | http://beeradvocate.com/beer/profile/3818/17497/?ba=bros |
| 20:13.04 | Katty | how about mojitos instead of beer. |
| 20:13.35 | fullstop | whiskey on the rocks? |
| 20:13.47 | fullstop | surely we must agree on something. :) |
| 20:14.24 | Katty | no :< |
| 20:14.29 | Katty | whisky is not good. |
| 20:14.33 | Katty | neither is tequila. |
| 20:14.37 | fullstop | =/ |
| 20:14.42 | Katty | vodka and rum is okay tho. |
| 20:14.46 | Katty | when mixed with something. |
| 20:14.49 | fullstop | rum & coke? |
| 20:14.57 | Katty | i got drunk on rum and coke the first time. |
| 20:15.04 | Katty | i tend to avoid it now |
| 20:15.13 | thehar | mmm beer |
| 20:15.27 | wcselby | rum and dr. pepper is good |
| 20:15.31 | Katty | malibu and cherry dr pepper is good. |
| 20:15.32 | wcselby | jack and coke |
| 20:15.38 | wcselby | crown and coke |
| 20:15.41 | leifmadsen | I hate jack |
| 20:15.43 | Katty | those are whiskys |
| 20:15.47 | wcselby | crown and coke tastes like vanilla coke |
| 20:15.50 | fullstop | bourbon |
| 20:16.00 | leifmadsen | weisers deluxe please |
| 20:16.01 | wcselby | jack is good when it's in something else |
| 20:16.05 | fullstop | and there is a difference between whiskey and whisky |
| 20:16.09 | leifmadsen | jack is good when used to clean guns |
| 20:16.12 | wcselby | i don't really drink much of anything straight up anymore |
| 20:16.20 | leifmadsen | jack is not good for drinking |
| 20:16.21 | Katty | DO YOU LIKE PINA COLADAS |
| 20:16.27 | wcselby | YES KATTY |
| 20:16.44 | wcselby | margaritas can be good, too. get a lot of those down here in houston |
| 20:18.03 | fullstop | I don't drink much straight up these days. I will have scotch on the rocks, though. |
| 20:18.45 | fullstop | The only thing I've had straight up recently was sambuca. |
| 20:18.54 | leifmadsen | ceasars++ |
| 20:19.02 | leifmadsen | mamosa++ |
| 20:19.21 | beardy | Sambuca is nice. I've gotten some taste for whiskey. |
| 20:19.28 | beardy | I rarely drink at all though. |
| 20:19.54 | carrar | Katty, AND BEING CAUGHT IN THE RAIN |
| 20:20.02 | beardy | Otherwise, I like White Russians. |
| 20:25.08 | Katty | carrar: :>>> |
| 20:25.22 | Katty | carrar: also rain is wet. and often cold :< |
| 20:26.09 | leifmadsen | I like Bailey's on the rocks |
| 20:26.14 | WIMPy | Sometimes it's so cold, that it isn't wet any more. |
| 20:27.30 | Katty | WIMPy: :> |
| 20:28.16 | *** join/#asterisk the_weard (~arthur@41-132-111-196.dsl.mweb.co.za) |
| 20:36.03 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 20:36.12 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 20:38.42 | *** join/#asterisk imcdona (imcdona@173.160.189.68) |
| 20:40.03 | *** part/#asterisk imcdona (imcdona@173.160.189.68) |
| 20:49.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:50.50 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 20:59.57 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 21:01.20 | *** part/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 21:02.07 | *** join/#asterisk syadnom (~syadnom@216.187.176.106) |
| 21:02.10 | *** join/#asterisk imcdona (~imcdona@2001:470:e8f1:1:f581:38be:f71c:ab15) |
| 21:02.25 | syadnom | hi. I am seeing something strange and having a strange experience. as follows |
| 21:02.43 | carrar | mmm drugs |
| 21:03.12 | syadnom | my 'full' log shows my chan_zap as detecting each dialed digit multiple times. so I see 'Dialing 'wwww6720129' and then I get lines like |
| 21:03.20 | carrar | psychedelic |
| 21:03.33 | syadnom | Detected digit '6' 3 or 4 times, then Detected digit '7' 3 or 4 times. |
| 21:04.08 | syadnom | when dialing, sometimes I can getting a missing digit, sometimes the wrong digit because of the strange dialing. |
| 21:04.26 | syadnom | I added the wwww to put a delay in because I thought I was dialing before the dialtone. |
| 21:04.28 | syadnom | Any ideas? |
| 21:05.33 | *** join/#asterisk rossand (~aross@173.243.47.194) |
| 21:06.09 | *** part/#asterisk rossand (~aross@173.243.47.194) |
| 21:07.09 | Katty | lay off the brownies. |
| 21:09.14 | *** join/#asterisk mducharme-laptop (cebc7904@gateway/web/freenode/ip.206.188.121.4) |
| 21:09.18 | mducharme-laptop | afternoon |
| 21:09.29 | mducharme-laptop | I have a routing question |
| 21:09.49 | [TK]D-Fender | ~ask |
| 21:09.49 | 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. |
| 21:09.56 | mducharme-laptop | I am forwarding incoming calls on a PRI on one asterisk server to another by having a route the calls 7777 on the other asterisk server |
| 21:10.23 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 21:10.32 | mducharme-laptop | it is working but it's not checking the DIDs, so the calls are all going to the default route |
| 21:11.12 | mducharme-laptop | I think it's because of how it's dialing the 7777, so the did is not getting passed to server #2 |
| 21:11.32 | mducharme-laptop | is there a different way of redirecting all calls from one server to another that will preserve DIDs? |
| 21:11.33 | *** part/#asterisk Brack10 (~tbrackett@66.35.2.50) |
| 21:12.24 | [TK]D-Fender | mducharme-laptop: PARDON? All this speak of "DID's" is irrelevant. The call comes in on the exten it lands on with the CallerID that it has. How you pass that to your 2nd server is up to you including what number you dial over to it. |
| 21:12.29 | *** join/#asterisk [T]ank (~ckwall@c-71-195-199-101.hsd1.ut.comcast.net) |
| 21:12.59 | mducharme-laptop | tkd-fender let me explain |
| 21:13.00 | [TK]D-Fender | mducharme-laptop: and "dialing the 7777" tells us nothing. that is a number. There is no implied functionality associated with it |
| 21:13.19 | [T]ank | im getting brute forced by someone trying to log into my sip peers. can someone quickly help me with the commands i need to add only specific ips to my iptables? googling is taking a while and i want to hurry and get this shut down |
| 21:13.26 | mducharme-laptop | I have one server that is fully set up as the phone server, it has all the extensions and people on it |
| 21:13.26 | [TK]D-Fender | mducharme-laptop: PASTEBIN is your friend. |
| 21:14.00 | mducharme-laptop | I have another server running asterisk, with no extensions, nothing, only the PRI coming in and a trunk to the asterisk server with all the phones |
| 21:14.11 | [TK]D-Fender | [T]ank: iptables -a INPUT -s 1.2.3.4 -d DROP |
| 21:14.25 | [T]ank | that drops the offending IP? |
| 21:14.33 | [TK]D-Fender | [T]ank: yes |
| 21:14.36 | [T]ank | thank you |
| 21:14.53 | mducharme-laptop | what I want is for any incoming call through the PRI to be directed through the IAX trunk |
| 21:15.10 | mducharme-laptop | so that it is as though the PRI was plugged into the server with all the extensions |
| 21:15.22 | Qwell | just use an IAX2 switch. |
| 21:15.37 | Qwell | that's like... ridiculously easy. |
| 21:15.43 | [TK]D-Fender | Qwell: Not happening.... |
| 21:15.45 | Qwell | leifmadsen: Why don't people use switches anymore? :( |
| 21:15.53 | [TK]D-Fender | Qwell: Check your channels... |
| 21:15.55 | Letoric | Anybody know a reliable method for telling Asterisk to roll logs after x size, or x days? |
| 21:16.04 | Qwell | fail |
| 21:16.08 | leifmadsen | Qwell: because they are not well documented and are old school? :) |
| 21:16.11 | Qwell | Letoric: logrotate |
| 21:16.11 | [TK]D-Fender | Letoric: logrotate <- JFGI :) |
| 21:16.14 | Qwell | check the contrib/ dir |
| 21:16.22 | Letoric | separate product? |
| 21:16.22 | p3nguin | letoric: You don't have Asterisk do it, you have logrotate do it. |
| 21:16.26 | Letoric | ahh ok |
| 21:16.28 | Qwell | leifmadsen: but they are awesome! |
| 21:16.30 | bougyman | i hate logrotate. |
| 21:16.32 | mducharme-laptop | how do I do that for incoming calls? |
| 21:16.32 | leifmadsen | Qwell: I'm awesome! |
| 21:16.34 | p3nguin | It's used in almost EVERY distro by default. |
| 21:16.34 | Qwell | and your face is oldschool |
| 21:16.47 | bougyman | so is SysV, hate that, too. |
| 21:16.49 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:16.51 | Letoric | I'll check it out. Thank you p3nguin and [TK]D-Fender |
| 21:16.56 | Qwell | leifmadsen: You should put yourself in the acknowledgements in your book. |
| 21:17.03 | Qwell | "Myself: For being awesome." |
| 21:17.05 | boodu | salut |
| 21:17.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:17.37 | Qwell | (I'm totally kidding BTW.) |
| 21:17.40 | boodu | lo |
| 21:18.09 | p3nguin | letoric: http://pastebin.com/QzHgZHmx |
| 21:18.11 | leifmadsen | Qwell: I've already done it |
| 21:18.17 | Qwell | ^_^ |
| 21:18.24 | mducharme-laptop | qwell how do you se up an iax2 switch? |
| 21:18.34 | Qwell | I should totally stop joking about that. You actually take me seriously when I say things. |
| 21:18.44 | mducharme-laptop | like I know where to put the switch command |
| 21:18.49 | mducharme-laptop | but I don't know what section heading to use |
| 21:18.52 | mducharme-laptop | or understand what it does |
| 21:19.05 | Qwell | You're using FreePBX. You can't. |
| 21:19.14 | mducharme-laptop | oh |
| 21:19.27 | mducharme-laptop | I can edit the config file manually can't I? |
| 21:20.06 | mducharme-laptop | the one server receives calls in the from-pstn context |
| 21:20.17 | mducharme-laptop | so if I set up a from-pstn-custom |
| 21:20.31 | mducharme-laptop | as that would be included in from-pstn.. and put the switch in there, would that do the trick? |
| 21:21.13 | mducharme-laptop | I'm sorry for being so newbie here |
| 21:21.23 | mducharme-laptop | but I can't get in touch with my pbx vendor |
| 21:21.28 | mducharme-laptop | and it's a bit of an emergency |
| 21:22.16 | mducharme-laptop | b/c if it doesn't happen in the next little while people may die.. the system is used for an emergency medivac call centre |
| 21:23.17 | p3nguin | Don't blame your usage of FreePBX on us. |
| 21:23.46 | Letoric | p3nguin: That pastebin you showed, it uses a wildcard and the info I read guided against that - anything I need to worry about? |
| 21:24.53 | Letoric | p3nguin: nm, I am guessing compress will give them a new extension so the wildcard won't catch them |
| 21:24.57 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 21:24.59 | Letoric | thanks again |
| 21:26.05 | [TK]D-Fender | mducharme-laptop: Anything that sensitive to outage should have planned a proper administrator |
| 21:26.27 | [TK]D-Fender | mducharme-laptop: And pumping voip calls over IAX between multiple server is far from "reliable" by their normal standards |
| 21:26.33 | [TK]D-Fender | mducharme-laptop: is that over the internet? |
| 21:27.53 | bougyman | this reminds me of the hospital I got sent to who had a switchvox and hadn't paid support. |
| 21:28.23 | mducharme-laptop | tkdfender yes it's over the internet, unfortunately this was the result of a temporary set up |
| 21:28.38 | mducharme-laptop | we had to start setting up a server to ge tthe phones going because the vendor was going to be late with the phone appliance |
| 21:28.48 | mducharme-laptop | they got it to us late, and it was plugged in just two hours before the PRI went live |
| 21:29.03 | mducharme-laptop | so we have this temporary server all set up and running with everybody, but the PRI in a different box |
| 21:30.28 | [TK]D-Fender | mducharme-laptop: If lives depend on it the this looks like to idea of a moron. |
| 21:31.00 | Letoric | p3nguin: Not very nice ;P |
| 21:31.18 | p3nguin | letoric: What's the problem? |
| 21:31.50 | p3nguin | letoric: Asterisk logs don't have "extensions." |
| 21:32.03 | p3nguin | letoric: It's just a regular old file name without a dot in it. |
| 21:32.13 | Letoric | it doesn't back up the files, it just makes a new messages file that is empty |
| 21:32.14 | [TK]D-Fender | mducharme-laptop: Either way all you need is a peer on each end to pass of the calls. On your PRI side point your PRI channels to a custom context that will Dial() the other server via that peer. On the receiving end, point the context to from-trunk. And set the DID's up there. The End |
| 21:32.36 | Letoric | p3nguin: I'm working with it now, it's on the lab system, so no real harm |
| 21:32.39 | p3nguin | # ls -l /var/log/asterisk/messages* |
| 21:32.40 | p3nguin | -rw-r----- 1 asterisk asterisk 20733 Sep 29 16:30 /var/log/asterisk/messages |
| 21:32.40 | p3nguin | -rw-r----- 1 asterisk asterisk 19951 Sep 26 06:37 /var/log/asterisk/messages.1.gz |
| 21:32.43 | p3nguin | -rw-r----- 1 asterisk asterisk 23010 Sep 19 04:03 /var/log/asterisk/messages.2.gz |
| 21:32.46 | p3nguin | Looks fine to me. |
| 21:33.09 | Letoric | p3nguin: hrm, I figured you were pranking me since you told me last week not to paste y'alls code on a live system without checking it hehe |
| 21:33.20 | Letoric | p3nguin: I'll see if I made a typo or something |
| 21:34.19 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:34.30 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 21:34.41 | kerx | hi all, my polycom phone is trying to make an outbound call, but I get this error: |
| 21:34.52 | kerx | NOTICE[24331]: chan_sip.c:14422 handle_request_invite: Failed to authenticate user "pedram" <sip:pedram@192.168.1.1>;tag=73DCE29D-B67620D0 |
| 21:35.37 | p3nguin | letoric: logrotate rotates logs. It's what it does. If you tell it to compress, it will compress the files after rotation. If not, the logs will still be rotated, but will not be compressed. |
| 21:36.39 | p3nguin | messages.1.gz as compared to messages.1 |
| 21:37.11 | Letoric | p3nguin: I understand that, but for some reason, it isn't creating the .1.gz - that's the part I'm looking into |
| 21:37.52 | p3nguin | Copy my file contents, pasted it into a new file with an appropriate name in the appropriate directory. |
| 21:38.28 | p3nguin | Mine is asterisk under /etc/logrotate.d/ as you can see. |
| 21:39.19 | Letoric | yeah, that's what I did |
| 21:39.30 | Letoric | now I'm running logrotate asterisk -f |
| 21:39.58 | Letoric | it sends the remote commands ok, I see those pass in the console |
| 21:40.02 | p3nguin | logrotate -f asterisk would be more sensible. |
| 21:40.17 | p3nguin | (even though you don't need to force it) |
| 21:40.23 | Letoric | I'll see if that makes a difference |
| 21:40.24 | ManxPower | kerframil, you do not have a [pedram] section of sip.con |
| 21:41.17 | Letoric | p3nguin: same deal |
| 21:41.27 | kerx | on internal phones in the call center, should the type=peer or type=friend ? |
| 21:41.30 | Letoric | it creates the new file, but doesn't rotate |
| 21:41.36 | Letoric | I'm using CentOS 5.4 |
| 21:41.41 | Letoric | err, 5.5 actually I think |
| 21:41.45 | p3nguin | How does it create a new file if it isn't rotating? |
| 21:42.02 | p3nguin | If it isn't rotating, then the file already exists, and cannot be created again. |
| 21:42.19 | Letoric | well, that's where the -f was probably changing the reaction (just a guess) |
| 21:42.51 | adelas | is it possible to use a Dual T1 card (TE220B) with 2 different T1 PRI provideres? |
| 21:43.01 | bougyman | should be |
| 21:43.16 | Letoric | yeah, -f forces it, but it's still not rotating, only clearing the log |
| 21:43.18 | p3nguin | logrotate essentially does this: mv messages messages.1 && touch messages && gzip messages.1 |
| 21:43.41 | bougyman | i've never tried it with a digium, but i've got other cards with multiple-providers per card. |
| 21:44.06 | adelas | damn i hope so |
| 21:44.17 | adelas | did you use zaptel? |
| 21:44.22 | adelas | or the newer drivers? |
| 21:45.00 | Letoric | well, could it be placing the files somewhere else? I'm operating as root, that's the only difference between your script and mine |
| 21:45.20 | p3nguin | If you aren't getting the results you expect, you could have a permissions issue, a file name issue, or a mis-copy of my asterisk logrotate config. |
| 21:45.45 | p3nguin | If you're running asterisk as root, that's your first mistake. |
| 21:45.49 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 21:46.10 | Tim_Toady | trying to setup a sangoma card in asterisknow, i got the wanpipe rpm from this page: http://wiki.sangoma.com/wanpipe-linux-asterisk-asterisknow but i have a prob with chan_woomera its not loading: http://pastebin.com/nrsZTmNR |
| 21:46.26 | p3nguin | Not that it'll prevent logrotate from working, but a mistake nonetheless. |
| 21:46.29 | Letoric | agreed, regarding running it as root, and that's in the lab to be worked through, it's how it ran on Solaris when the boss had it, so I started small ;p |
| 21:46.42 | Letoric | I'm running logrotate in verbose to see if I can see what's up. Thanks again for your help |
| 21:46.45 | Tim_Toady | any ideas where i can get the source of chan_woomera sangome uses and compile it myself? |
| 21:47.06 | Tim_Toady | s/sangome/sangoma |
| 21:47.55 | Letoric | ok, this is messed up. It says it's doing it properly when I run it verbosely, but it doesn't show the files lol |
| 21:48.05 | adelas | bougyman, did you have faxes running through the cards? |
| 21:48.14 | bougyman | adelas: sure. |
| 21:48.14 | p3nguin | s/s\/sangome\/sangoma/s\/sangome\/sangoma\/ |
| 21:48.36 | adelas | were you running 1.6? |
| 21:48.39 | adelas | or 1.4? |
| 21:48.41 | p3nguin | Shit, I messed it up too. |
| 21:48.45 | bougyman | i did not use zaptel, used openzap |
| 21:48.47 | Letoric | lol |
| 21:48.48 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 21:48.51 | bougyman | now freetdm |
| 21:48.57 | Letoric | it's cleaning up the archive file right after it makes it |
| 21:49.52 | Letoric | p3nguin: I added rotate 5, and now it is working |
| 21:50.06 | Letoric | p3nguin: maybe a difference between the packages we have? |
| 21:50.06 | p3nguin | I wonder if there is any issue with empty files. |
| 21:50.19 | ManxPower | Tim_Toady, you might get a better if you asked your question on a channel where it was on-topic. |
| 21:50.26 | p3nguin | logrotate 3.7.8 - Copyright (C) 1995-2001 Red Hat, Inc. |
| 21:50.37 | Letoric | I'm on 3.7.4 |
| 21:50.44 | ManxPower | logrotate has an option to not rotate if the file is empty |
| 21:50.53 | Letoric | yeah, I was using force, so it was definitely rotating |
| 21:50.55 | ManxPower | see /etc/logrotate.conf or whatever it is called for your distro. |
| 21:51.03 | Letoric | but it would delete the .gz right after |
| 21:51.39 | p3nguin | I suspent you didn't copy my rotate file exactly, or your file names are different from mine. |
| 21:51.40 | Letoric | logrotate.conf has defaults at weekly/4 |
| 21:51.48 | p3nguin | s/suspent/suspect/ |
| 21:52.04 | Letoric | I did p3nguin I'll even pastebin if ya want! |
| 21:52.07 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:52.13 | Letoric | either way, adding rotate fixed the issue |
| 21:52.17 | p3nguin | I also use weekly and rotate 4. |
| 21:52.27 | Letoric | maybe logrotate isn't processing it's config file properly? |
| 21:52.39 | Get_The_Fish | can someone shed some light on the EXTENSION_STATE function? What is meant by "hinted extension" in the function notes? |
| 21:53.32 | ManxPower | Get_The_Fish, voip-info.org was not helpful? |
| 21:53.56 | ManxPower | Get_The_Fish, A hinted extension is an extension with a "pseudo" priority of "hint" |
| 21:53.57 | Get_The_Fish | not enough for me to wrap my head around... I just need a little push |
| 21:54.17 | ManxPower | exten => 1234,hint,SIP/9876 |
| 21:54.44 | Get_The_Fish | and then you can get the state of 1234, right? |
| 21:55.00 | Get_The_Fish | instead of SIP/9876 (is it really that simple?) |
| 21:55.08 | ManxPower | the state of extension 1234 would be based on the device SIP/9876 |
| 21:55.28 | Get_The_Fish | ah I see I see... I told you I just needed a little push |
| 21:55.54 | ManxPower | n00bs frequently set their SIP user IDs to be the same as the user's extension, but that is a bad idea. |
| 21:56.32 | Get_The_Fish | I seem to remember that there was an issue with using realtime extensions with DEVICE_STATE, do you think that the same would apply to EXTENSION_STATE the same, or is that no longer an issue? |
| 21:56.47 | p3nguin | SIP/<random letters and numbers> |
| 21:56.56 | ManxPower | I cannot comment on Realtime in any way. |
| 21:57.23 | ManxPower | p3nguin, I use the MAC of the device with -a -b -c, etc appended to register each call appearance independently. |
| 21:57.55 | Get_The_Fish | yeah, I am using a rather complicated hotdesking schema, doing these "extension to user mappings" (for lack of a better term) addresses some issues. Thanks ManxPower. |
| 22:00.27 | p3nguin | letoric: Are all the other log files rotating as prescribed by their respective config files? |
| 22:03.31 | Letoric | p3nguin: TBH, Asterisk is the only thing running on the box outside of the built-in system stuff. I don't know how often the system logs *should* rotate, but /var/log/messages is definitely rotating without compression |
| 22:08.04 | Letoric | it's working now with create and compress commented out (I would rather have them easily viewable, disk space isn't an issue) |
| 22:08.18 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 22:08.21 | Letoric | I know when I move asterisk to run as asterisk, I'll have to put that back in though |
| 22:09.29 | Letoric | off to play pool, see yas tomorrow, and thanks again for the help |
| 22:13.36 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 22:13.55 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 22:14.50 | BenC[UK] | evevneing guys, quick question on agi... I am sending a call directly into an agi script, which prompts the caller for some information, if they enter valid information, I want to put them into a queue... I've got everything working up to sending them to the queue - just not sure how to do that |
| 22:15.06 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 22:16.17 | ManxPower | BenC[UK], in your AGI set a dialplan variable, when you exit the AGI check for that variable and send the call to a queue using the dialplan if that variable is set. |
| 22:16.51 | ManxPower | You do not want to execute long running applications like Dial or Queue from in an AGI |
| 22:17.07 | BenC[UK] | ah |
| 22:17.42 | ManxPower | Your AGI will suspend until the application exits. |
| 22:18.05 | BenC[UK] | that makes sense.. just not sure how to edit the dialplan! |
| 22:18.44 | ManxPower | sounds like you are using a GUI |
| 22:19.06 | BenC[UK] | well, I have been using FreePBX, but doing some things from the command line too |
| 22:19.31 | BenC[UK] | I've started with asterisk late last night |
| 22:20.03 | russellb | your AGI application will only suspend if you write it that way. |
| 22:20.26 | russellb | if you don't want to suspend, don't block waiting on the response :-) |
| 22:21.33 | russellb | the dialplan _will_ block, in AGI you at least have the option not to block if you'd like. |
| 22:21.40 | russellb | just my 3.14159 cents. |
| 22:21.51 | BenC[UK] | russellb: how do I set it not to block? |
| 22:22.18 | *** join/#asterisk oDesk (~chatzilla@188.48.112.166) |
| 22:22.21 | BenC[UK] | i'm using phpagi |
| 22:22.47 | russellb | ah, if you're using a library, you probably can't. they're usually written to block. |
| 22:23.08 | russellb | but in any case, you're just in the same boat as the dialplan is. |
| 22:23.33 | *** join/#asterisk Tim_Toady (~moi@178.128.56.127.dsl.dyn.forthnet.gr) |
| 22:23.47 | BenC[UK] | I dont mind blocking.. we're talking about 10 users at a time max.. shouldnt really cause that many problems ... hopefully, if it does, I can revisit |
| 22:23.55 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:24.01 | oDesk | does wcfxo accept fwringdetect argument ? |
| 22:28.10 | ManxPower | those cards have not been manufactured in close to 10 years |
| 22:31.30 | *** join/#asterisk [canniballllera] (~cannibale@201-3-228-210.fnsce703.dsl.brasiltelecom.net.br) |
| 22:31.40 | *** part/#asterisk [canniballllera] (~cannibale@201-3-228-210.fnsce703.dsl.brasiltelecom.net.br) |
| 22:32.22 | *** join/#asterisk cornbread7733 (~IceChat7@unaffiliated/cornbread7733) |
| 22:34.02 | cornbread7733 | Anyone here brilliant with Aastra dial plans? I have a few phones cannot dial LD. Can dial local. Can dial local and LD with xLite and iSIP via iPhone |
| 22:35.34 | cornbread7733 | My Aastra models are 53i, 55i, 57iCT |
| 22:39.38 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-150-41.cinci.res.rr.com) |
| 22:50.33 | carrar | cornbread7733: might start back at the default and workit out how you want it; sip dial plan: "x+#|xx+*" |
| 22:51.07 | carrar | instructions are in the firmware admin guide |
| 22:52.32 | carrar | ftp://216.94.98.106/Downloads/Admin_Guides/IP%20Phone%20Admin%20Guide_2.6.0_41-001160-05_REV00_IPP_AG_1005.pdf |
| 22:52.40 | carrar | page 5-56 |
| 22:55.21 | BenC[UK] | ok, I give up for a minute with the Queue.. |
| 22:55.35 | BenC[UK] | next question, I keep getting "all circuits are busy" |
| 22:57.20 | BenC[UK] | <PROTECTED> |
| 22:57.31 | cornbread7733 | carrar: I will check the guide out, but I have pretty much started with default already, and added dial plans to attempt to correct. |
| 22:57.53 | carrar | then go back to were it was working |
| 22:57.59 | carrar | and figure out what you did wrong |
| 23:00.03 | cornbread7733 | It was never working... that is the problem. |
| 23:00.22 | cornbread7733 | ... it is working from xlite and iSIP |
| 23:00.35 | cornbread7733 | ...but not Aastra 53i, 55i, 57iCT |
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| 23:04.50 | *** part/#asterisk [canniballllera] (~cannibale@201-3-228-210.fnsce703.dsl.brasiltelecom.net.br) |
| 23:10.23 | fifer | I moved from * 1.6.0 to 1.6.2.13 on Friday and * is now no longer logging cdr to MySQL. Nothing has changed in the config. |
| 23:12.21 | *** join/#asterisk jeffik (~chatzilla@69-196-165-181.dsl.teksavvy.com) |
| 23:15.03 | fifer | if my * config was setup properly to log cdr to MySQL should "cdr show status" indicate something under "Registered Backends" |
| 23:22.53 | *** part/#asterisk cornbread7733 (~IceChat7@unaffiliated/cornbread7733) |
| 23:26.17 | *** join/#asterisk Faithful (~Faithful@180.194.2.32) |
| 23:29.47 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 23:32.27 | fifer | can you restart cdr with out an * restart? |
| 23:40.46 | *** join/#asterisk cusco (~trilili@174.196.108.93.rev.vodafone.pt) |
| 23:40.50 | cusco | hello folks |
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| 23:50.50 | jsidhu | so ive got asterisk running on openwrt box, and am using a sdcard to record voicemails etc. If I call that extension direct and leave a voicemail, it comes out fine. When I forward calls from my main PBX via IAX2 trunk and leave a voicemail, the beginning is allright, but the rest of the message appears to be garbled (sort of like its playing at 3x-4x the normal speed). anyone have any tips to troubleshoot this one? |
