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00:14.33 | Jasnejac | anybody tried 1.8.1.2 yet? I have two installations and neither will make sip calls although they will receive them |
00:15.05 | Chainsaw | Damn, is it out? |
00:15.07 | Chainsaw | Did I miss it? |
00:15.11 | Jasnejac | yes |
00:15.26 | Chainsaw | I blame leifmadsen for not updating the topic! |
00:15.28 | Jasnejac | Leif announced it on VUC earlier |
00:15.29 | Chainsaw | updates CVS |
00:15.46 | Jasnejac | but I cant get it to make SIP calls :D |
00:16.03 | Chainsaw | So some ancillary features don't work. But the core is very stable ;D |
00:16.13 | Jasnejac | lol |
00:16.15 | citywok | Jasnejac: did it work in 1.8.1.1? |
00:16.31 | Jasnejac | yes |
00:16.55 | citywok | k. compiling on dev box |
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00:17.47 | citywok | what's the error? |
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00:18.48 | Jasnejac | http://pastebin.com/AYDE3Lr4 |
00:19.49 | citywok | Jasnejac: is that supposed to be dial (SIP/Pete,30) and not .30? |
00:19.54 | Jasnejac | that's for a host not there of course but the effect is the same |
00:20.16 | citywok | or do you actually have a peer named "pete.30" |
00:20.24 | Chainsaw | Jasnejac: You can't rely on ChanIsAvail or Dial not connecting to non-existent hosts in 1.8; use SIPPEER instead. |
00:20.37 | Jasnejac | lol - you are kidding me. doh.... I've spent the last two hours trying to make this work! |
00:20.55 | citywok | your first clue should have been unable to resolve pete.30 |
00:21.22 | drmessano | PETE 3.0 is much cooler than 2.5 |
00:21.33 | citywok | yea but he's unstable. too new. |
00:21.35 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
00:21.41 | carrar | but PETE still needs a upgade |
00:21.43 | carrar | upgrade |
00:21.44 | citywok | i'd wait until pete 3.0.1 |
00:22.02 | carrar | obvously PETE 1.8.1.2 was a fail |
00:22.21 | drmessano | lol |
00:22.21 | Chainsaw | Jasnejac: I tend to check whether context != "" for said SIPPEER before I really connect. |
00:22.26 | Jasnejac | nah - that wasn't it |
00:22.27 | Chainsaw | Jasnejac: That works on my system. |
00:22.40 | drmessano | I did an SVN update this AM.. haven't tested my calls yet |
00:22.46 | drmessano | At home anyway |
00:23.01 | carrar | Auto upgrade productionsboxes? |
00:23.03 | Jasnejac | still can't get it to work - new pastebin http://pastebin.com/T2Wam4Kp |
00:23.11 | drmessano | Working fine here |
00:23.44 | citywok | = Everyone is busy/congested at this time (1:0/0/1) |
00:23.46 | citywok | sip show peers |
00:24.10 | Chainsaw | shines a bright light on the SIPPEER command he wrote about before, then walks off |
00:27.24 | CoYo-T | Jasnejac, looks like some invalid string in your peer |
00:28.10 | Jasnejac | this is strange - I've not chaged anything and now it works. I'll look a little further, all typos asside :D |
00:29.50 | CoYo-T | i have some similar issue, and think pthread is bad allocating memory spaces.. but its just an idea... not low level debug yet done to asume this |
00:30.31 | CoYo-T | im using 1.8.1.1 and have lot of crashes with strange "memory chunks" |
00:36.26 | Jasnejac | there was a patch for 1.6.2.15 (I think) for choppy IAX audio. Is this sorted in 1.8.1.2? |
00:36.40 | citywok | check CHANGES |
00:36.55 | Jasnejac | I have but they aren't clear |
00:37.09 | Jasnejac | not to me anyway! |
00:37.32 | citywok | it has 3 changes listed under IAX2. what isn't clear? |
00:38.50 | Jasnejac | those are added facilites not bug fixes |
00:39.23 | Jasnejac | this was a patch, noted on the users list. I couldn't find it included in the 1.8.2 rc1 list |
00:40.40 | Jasnejac | I trunked to a 1.4.19 box (live) and it gave choppy audio - ended up sending it over dahdi to get dtmf to work |
00:42.08 | CoYo-T | i ask if the new sip status labels are intentional or a misstype... cause this is messing some control panels... and no answer yet |
00:42.25 | CoYo-T | any url where find info about this? i can't find anything around |
00:43.19 | CoYo-T | maybe Jasnejac can find info there too |
00:47.39 | citywok | check the ChangeLog files |
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02:07.04 | v1s | is there any conversion wen going from sip to iax ? |
02:08.00 | v1s | like if my did provider can do sip and iax and I am doing didprov <sip> serv1 <iax2> serv2 |
02:08.11 | v1s | should I just try to keep it all iax2 or does it not mater |
02:14.26 | Cresl1n | It usually doesn't hurt to keep it all the same :-) |
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02:20.45 | Lantizia | Hey has anyone written anything that uses the official Digium "Fax for Asterisk" that does T.38 to E-mail and E-mail to T.38? |
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02:27.08 | v1s | i had a fax to email script let me see if I can find it not sure if I still have it |
02:31.53 | v1s | sorry cant find the script I used. But I found this in my bookmarks hopefuly it can help you it has some script for that |
02:31.55 | v1s | http://www.teamforrest.com/blog/156/integrating-fax-for-asterisk/#more-156 |
02:34.43 | CoYo-T | try avantfax |
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03:07.42 | Lantizia | CoYo-T, does this avantfax register to asterisk as a peer then to send/receive T.38 faxes? |
03:08.00 | CoYo-T | via iaxmodem |
03:08.38 | CoYo-T | google for some info.. i have it working and goes great |
03:09.10 | Lantizia | no it's not t.38 - not want I want sorry |
03:11.27 | CoYo-T | nop.. a hint .. http://www.freelancer.com/projects/Script-Installation-System-Admin/Install-Hylafax-Avantfax-with-Softmodem.html |
03:11.35 | CoYo-T | works w t38 too |
03:12.47 | CoYo-T | let me see if i can find some more info 4 u.. but it works w/wout t.38 protocol.. and is a nice frontend to manage faxes via web |
03:12.50 | Lantizia | not natively though |
03:13.28 | Lantizia | and I specifically want to use the "Fax for Asterisk" module which already does T.38 |
03:13.44 | Lantizia | just need to find a nice web interface for faxes using it _IN Both Directions_ |
03:13.50 | Lantizia | and/or e-mail to fax / fax to e-mail |
03:14.08 | Lantizia | but I don't care about faxing over normal sip as it never works or faxes over tdm - just t.38 |
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03:18.48 | CoYo-T | i didn't try. but maybe avantfax can be pluggued to asterisk directly |
03:19.29 | CoYo-T | cause it manage modems... you can try using iax2 between asterisk and avantfax (as a modem) to manage all the fax stuff |
03:19.46 | CoYo-T | not usin iaxmodem ofcourse |
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03:21.25 | Lantizia | I've never even seen T.38 over IAX |
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03:22.58 | CoYo-T | iaxmodem is made for it (handles t38 too, if your sip trunk supports it) |
03:23.26 | CoYo-T | now asterisk can do that part of the job... so... for sure.. iaxmodem can be ommited |
03:24.08 | CoYo-T | look for info of how avantfax can be pluggued into asterisk.. |
03:25.00 | CoYo-T | cause even you can only use the interface... |
03:25.21 | CoYo-T | asterisk grab the faxes.. avanfax can read it from the storage dir... |
03:30.01 | Lantizia | CoYo-T, Do Not Want |
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05:26.45 | Kobaz | hmm |
05:27.00 | Kobaz | there's an IFTIME, but no IFDAY |
05:31.57 | v1s | what u trying to do |
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05:32.37 | raden | [Jan 14 23:32:23] WARNING[18208]: chan_sip.c:2803 retrans_pkt: Maximum retries exceeded on transmission 3cd17d6e-71d5b102@192.168.1.201 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. |
05:33.05 | Kobaz | day of the week |
05:33.23 | v1s | Â GotoIfTime(<time range>|<days of week>|<days of month>|<months>?[[context|]extension|]pri) |
05:34.04 | Kobaz | yeah but i want a function |
05:34.33 | v1s | it is the same I belive |
05:34.37 | v1s | u use that format |
05:34.56 | v1s | IFTIME(*|mon|8-15|nov?goodtime:badtime) |
05:35.04 | v1s | see this page |
05:35.05 | v1s | http://www.voip-info.org/wiki/view/Asterisk+func+iftime |
05:35.22 | v1s | timespec is on this page |
05:35.23 | v1s | http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime |
05:36.53 | Kobaz | which asterisk version |
05:36.55 | Kobaz | IFTIME(<timespec>?[<true>][:<false>]) |
05:37.00 | Kobaz | i'm on 1.6.0 still |
05:37.07 | v1s | it says 1.6 |
05:37.16 | Kobaz | hmm |
05:37.19 | v1s | Asterisk 1.6 |
05:37.19 | v1s | Â GotoIfTime(<time range>,<days of week>,<days of month>,<months>?[[context,]extension,]pri) |
05:37.22 | Kobaz | i guess i can see if it works |
05:37.29 | Kobaz | yeah i know GotoIfTime supports all that |
05:37.39 | Kobaz | in 1.6 |
05:50.45 | Kobaz | the example is wrong |
05:51.08 | Kobaz | i should look at the source code to see how it parses timespec |
05:51.21 | Kobaz | but just from some playing around, this works: Set(daymode=${IFTIME(*|mon|*|*|0:00-4:00|?1:0)}); |
05:51.24 | Kobaz | well, seems to work |
05:58.08 | Kobaz | ast_build_timing() |
05:58.25 | Kobaz | mm, that's not the right spot for the time range |
06:00.19 | Kobaz | probably should be added to the documentation, but it's time|weekday|monthday|month |
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06:21.38 | reseller | hi everybody |
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06:37.41 | jasonh | hello |
06:40.47 | v1s | hi |
07:03.17 | jasonh | could anyone offer me any advice? whenever I reboot my asterisk box my rhino cards come up unconfigured when I run dahdi_tool |
07:10.57 | carrar | jasonh, modules probably aren't getting loaded |
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07:12.01 | carrar | I don't use rhino cards |
07:12.36 | jasonh | let me check. i believe i checked that. in the /etc/dahdi/modules file? or a different one? |
07:12.51 | carrar | or you need to create a init.d script to load in the config |
07:14.44 | carrar | <PROTECTED> |
07:15.08 | carrar | but you didn't mention if the module was getting loaded or not |
07:15.20 | carrar | just that the card config wasn't there |
07:16.27 | carrar | Though I assume when you load the module it laods in the config too |
07:16.33 | carrar | don't know |
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07:18.55 | jasonh | modprobe -l shows them as being loaded.. lemme check the init script to see if it's loading the config.. thanks for the pointers! |
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07:45.38 | jasonh | awesome! got it working! :D |
07:46.23 | carrar | What was it |
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07:51.27 | jasonh | honestly, i'm not totally sure. but i think it was because I needed to run dahdi_genconf before dahdi_cfg. does that make sense? |
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11:18.03 | Alagar | hi |
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11:39.00 | bn-7bc | ,203, |
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12:15.12 | delx | Hey! |
12:18.11 | delx | I have an AGI script. The script dials a number. Is it possible to get the timestamp when the call receiver hearer has been picked ? |
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12:29.16 | HaMF | Hi! I've connected a SIP-Phone to my asterisk box and everything works fine as long as I specify the peer in sip.conf but as soon as I try to use postgresql realtime asterisk crashes after a the phone registeres (not neccesairly immediately). Could someone help me out there. I would happily provide any information you need. |
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13:15.35 | shtoom | delx: Its certainly possible through AMI there would be an event indicating channel state change (Ringing , answered etc etc ) |
13:17.59 | delx | shtoom, thanks for information! :) |
13:18.30 | shtoom | delx: :) |
13:19.17 | WIMPy | delx: You will also find it in your CDRs if you don;t need it during the calll |
13:19.33 | delx | WIMPy, CDRs? |
13:20.05 | WIMPy | Call Detail Records |
13:20.33 | delx | ah, ok :) |
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13:48.06 | mechbangirc | hi can any one recommend me some good SBC for asterisk |
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17:26.35 | shamelessn00b | anyone used VoiceRD? |
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18:45.45 | puzzled | hi |
18:47.34 | carrar | hi |
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19:44.18 | jaytee | does anyone know if there's an option in configuration of Polycom phones to give a tone indication for attended transfer on the phone the call is transferred to when the transfer has completed? |
19:44.47 | jaytee | I've been digging through the SIP Admin guide for several hours and googling. |
19:48.04 | jaytee | actually I'm probably meaning Consultative transfer because I'm not using the Conference softkey |
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20:02.44 | DoYouKnow | how do I set up asterisk so I can make a call file using google voice with jabber? I also want to play music with MP3Player |
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20:03.02 | DoYouKnow | (into the call) |
20:06.12 | DoYouKnow | I've seen some documents on configuring google voice, but none seem to describe how to set it up 100% fully, that I can tell anyway |
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20:13.52 | whippiii | hi there, I'm about to start the development of a call center supervising web application. I'm new in Asterisk's world and any sourrounding piece of code about it... |
20:14.07 | whippiii | Would you give me some advice about what language or languages should I consider to make this kind of job? |
20:14.27 | whippiii | please, sorry by my bad English, I speak Spanish |
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23:35.30 | gift | whats a good free voip provider to use for asterisk? ive googled but i think i have bad terminiology for pbx's |
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23:59.52 | DoYouKnow | gift: theoretically, you can use google voice callback with ipkall for inbound |