| 00:00.05 | gusto | ok |
| 00:00.18 | gusto | asking differently, what is the minimal ram requirement for asterisk? |
| 00:00.38 | [TK]D-Fender | gusto: Yes. |
| 00:00.45 | WIMPy | That depends on the modules you want to load/use. |
| 00:01.04 | gusto | yes, minimal, sip, conf needed, iax maybe |
| 00:01.29 | gusto | i see that on my openwrt router it takes about 8 MB of ram, what is really low |
| 00:01.39 | gusto | on my desktop i did not manage to get it under 14 MB |
| 00:02.11 | catphish | i imagine there are several ways to make it smaller |
| 00:02.18 | gusto | well |
| 00:02.31 | gusto | i am curious |
| 00:02.31 | catphish | 32 vs 64 bit, debug symbols, compiler optimizations |
| 00:02.37 | gusto | ah |
| 00:02.43 | WIMPy | Like compiling with -Os |
| 00:02.44 | catphish | obviously modules and their options |
| 00:03.26 | gusto | that's maybe the case why i did not manage it to make it use up less, because the desktops are x86_64 nowadays and MIPS is still 32 but |
| 00:03.28 | gusto | bit |
| 00:05.05 | gusto | well with modules you can not do much, because you need some to have a functional setup |
| 00:05.08 | catphish | why is cdr(exten) read-only? |
| 00:05.38 | catphish | guess i can strip things in my views later |
| 00:05.59 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 00:06.42 | WIMPy | Sounds like a plan. |
| 00:07.09 | catphish | i already parse local to international format for billing |
| 00:07.23 | catphish | so stripping 141 isnt a great hardship |
| 00:07.40 | WIMPy | That's what I use the loopback switches for. |
| 00:08.06 | WIMPy | Otherwise I'd have to match the same numbers in different formats. |
| 00:08.11 | *** join/#asterisk DarthExpeditor (~IceChat9@c-24-11-175-127.hsd1.mi.comcast.net) |
| 00:09.02 | WIMPy | That's especially an issue if you want to use dundi or enum while it lasted. |
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| 00:31.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:40.43 | *** join/#asterisk powerunits (b6b1a8d2@gateway/web/freenode/ip.182.177.168.210) |
| 00:40.48 | powerunits | hello all |
| 00:40.57 | powerunits | how is every one. |
| 00:41.12 | powerunits | dear team i m getting small error message on asterisk 1.6 |
| 00:41.22 | powerunits | [Jun 25 12:25:14] ERROR[15018]: utils.c:1177 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 00:41.43 | *** join/#asterisk rjb_77777 (~rjb_77777@mta-76-176-43-24.san.rr.com) |
| 00:41.44 | powerunits | i m trying to reload asterisk from AMI using PHP |
| 00:42.14 | powerunits | plz help |
| 00:43.13 | powerunits | ?? |
| 00:44.42 | pabelanger | powerunits, fix your AMI script |
| 00:45.17 | pabelanger | you are likely closing the socket before asterisk has a chance to response |
| 00:45.49 | powerunits | http://pastebin.com/U9LXa0i1 |
| 00:45.56 | powerunits | i think script is correct |
| 00:45.58 | powerunits | please check |
| 00:46.12 | pabelanger | please use an existing library |
| 00:46.32 | powerunits | which library? |
| 00:46.47 | powerunits | please can you give me some hint |
| 00:47.26 | pabelanger | I'd just google it, so best to start there |
| 00:47.31 | powerunits | is that script is correct |
| 00:47.37 | powerunits | ? |
| 00:47.38 | pabelanger | lots of existing libraries |
| 00:47.39 | pabelanger | no |
| 00:47.44 | pabelanger | you are not login off the manager |
| 00:47.52 | pabelanger | you are just dropping the socket |
| 00:48.24 | powerunits | let me google it.. |
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| 01:06.52 | powerunits | @pabelanger: you there? |
| 01:07.00 | powerunits | thanks that issue is fixed |
| 01:07.35 | powerunits | now one more small question regarding on gtalk with asterisk. |
| 01:07.44 | powerunits | i have configured gtalk with asterisk |
| 01:07.49 | powerunits | but when i make a calls |
| 01:08.00 | powerunits | i get these messages on asterisk CLI |
| 01:08.01 | powerunits | http://pastebin.com/104vea3W |
| 01:08.07 | KNERD | powerunits: give up |
| 01:08.15 | KNERD | it dont work anymore |
| 01:08.24 | powerunits | and no voice on both sides |
| 01:08.49 | powerunits | ?? |
| 01:08.51 | powerunits | why |
| 01:08.59 | powerunits | :( |
| 01:09.05 | KNERD | not even the asterisk people know |
| 01:09.24 | pabelanger | well, google likely changes something again |
| 01:09.31 | pabelanger | Asterisk 11 will have a new channel driver |
| 01:09.36 | powerunits | on this its not working fine |
| 01:09.38 | pabelanger | should be more reliable |
| 01:09.40 | powerunits | :S |
| 01:09.41 | powerunits | but on one of my system gtalk is working 100% fine.. but this is my new system |
| 01:10.02 | powerunits | hum |
| 01:10.11 | KNERD | when is 11 due out? |
| 01:10.18 | powerunits | soo when asteisk11 will be on the market? |
| 01:10.29 | powerunits | yes |
| 01:10.34 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 01:10.55 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:10.58 | powerunits | ? |
| 01:11.07 | mjordan | powerunits: read to the bottom of the page. |
| 01:11.08 | pabelanger | well you could do what it stays |
| 01:11.10 | pabelanger | says* |
| 01:11.20 | pabelanger | Found a loopback IP on the system, check your network configuration or set the bindaddr attribute. |
| 01:12.20 | KNERD | according to that, * 11 should be out now |
| 01:13.02 | powerunits | @pabelanger: let me check it out |
| 01:13.03 | pabelanger | no |
| 01:13.04 | p3nguin | knerd: Huh? What doesn't work? I use Google Voice with Asterisk every day. |
| 01:13.12 | pabelanger | likely Oct. 28this |
| 01:13.16 | KNERD | yeah..suuuuuree.. |
| 01:13.23 | KNERD | I have not had it functioning in over a year |
| 01:13.31 | mjordan | KNERD: what do you mean it should be out now? |
| 01:13.38 | p3nguin | Crazy. It works fine for me. |
| 01:13.49 | KNERD | maybe an older version |
| 01:13.55 | p3nguin | Older version of what? |
| 01:13.57 | KNERD | one day it just stopped functioning |
| 01:13.59 | KNERD | asterisk |
| 01:14.06 | p3nguin | How old are you talking about? |
| 01:14.10 | mjordan | KNERD: feature freeze is 3rd Wednesday of July, first release is 3rd Wednesday of October |
| 01:14.15 | KNERD | well at leats more than a year |
| 01:14.39 | p3nguin | Asterisk 1.8.8.0 built by rob @ cpe-e650 on an i686 running Linux on 2011-12-20 23:27:42 UTC |
| 01:14.46 | KNERD | the chart shows asterisk 11 out now |
| 01:15.13 | p3nguin | Not much over six months on that build, but I don't know when it was released. |
| 01:15.15 | mjordan | KNERD: uhm. Maybe my screen is more stretchy? |
| 01:15.25 | KNERD | possibley |
| 01:15.50 | KNERD | p3nguin: oh? one day it stopped working..worked work a few months then stopped workign again all of a sudden |
| 01:15.59 | mjordan | suffice to say, the dates are what is on the bottom of the screen. Feature freeze is endish of July, first release is endish of October |
| 01:16.21 | p3nguin | I make and receive calls with Google Voice every single day. |
| 01:16.39 | mjordan | KNERD: what version of Asterisk are you running? |
| 01:16.51 | KNERD | i have not been able to successfully make a call in over a year, and a LOT of others are in the same boat |
| 01:17.02 | KNERD | I juts put 1.8.13.0 on today |
| 01:17.06 | p3nguin | I do remember some time ago there was a patch to fix something that they changed, then they changed it back and we had to revert the patch... but then I guess they quit messing with it. |
| 01:17.33 | mjordan | KNERD: https://wiki.asterisk.org/wiki/display/AST/Help+Maintain+Google+Talk+and+Voice. There is an open source developer who is attempting to maintain the current chan_gtalk in the face of google's changes |
| 01:17.46 | p3nguin | I tend to not be current when it comes to software. |
| 01:18.12 | KNERD | "Page Not Found " |
| 01:18.37 | p3nguin | Loads fine for me. |
| 01:18.48 | p3nguin | Overview |
| 01:18.50 | p3nguin | Asterisk has, since 1.8.0, supported inbound and outbound calling through Google Voice ... |
| 01:18.56 | mjordan | k, I'm looking at the page now. So either you don't have an account on the Asterisk wiki, or - wherever you are - you have bigger issues then just gtalk :-D |
| 01:19.32 | KNERD | yes it has, bt as you mention there was changes back and forth, and it just does not work anymore for me and a lot of thers... |
| 01:19.37 | p3nguin | You don't even need an account to view pages on the wiki. |
| 01:19.54 | *** join/#asterisk newmember (~chatzilla@static-66-11-81-77.ptr.terago.net) |
| 01:20.06 | KNERD | it was the period at the end |
| 01:20.19 | mjordan | KNERD: have you filed issues in JIRA, so that the developers who have volunteered to maintain chan_gtalk can fix the issues? |
| 01:20.20 | p3nguin | Periods always ruin things. |
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| 01:21.12 | KNERD | i just gave up on it..seems everyday there was a patch...oops...remove patch...etc |
| 01:22.23 | newmember | Where do I turn off call forwarding for an extension in cfg files for *? |
| 01:25.00 | p3nguin | newmember: I'd guess you turn it off in the same file where you turned it on. |
| 01:25.24 | newmember | p3nguin: Only I was not here when it got turned on. but thanks |
| 01:25.30 | p3nguin | And, since extensions are configured in extensions.conf, I would suspect extensions.conf is the file where you turned it on. |
| 01:26.46 | p3nguin | There is also a possibility that the phone itself is responsible for the call forward that you are encountering. |
| 01:27.23 | powerunits | ooooh great |
| 01:27.29 | powerunits | gtalk is also working fine |
| 01:27.48 | powerunits | i will only bind the IP address of asterisk in gtalk.conf |
| 01:27.59 | powerunits | and started working fine :D |
| 01:28.05 | KNERD | well I guess I gotta try again |
| 01:28.27 | newmember | p3nguin: good thought, we have polycom phones, so I am thinking that the cfg is coming from the server. I am thinking yes if I had a softphone maybe I couls set that in the client |
| 01:29.13 | p3nguin | Polycom phones, like most other phones, have the ability to set call forwarding on itself. |
| 01:29.30 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 01:29.31 | p3nguin | But config files for the phones also can control that setting. |
| 01:30.22 | dijib | anybody got any i3 or i5 laptops they want to part with? |
| 01:30.32 | dijib | how is everyone also |
| 01:31.30 | newmember | p3nguin: hmmm i see the forward all option |
| 01:31.46 | p3nguin | Yep. |
| 01:31.55 | newmember | So what is happening is that all calls to an ext in the asterisk are call forwarding to the guys cell phone |
| 01:32.19 | newmember | So that is why I was going with the * conf files |
| 01:32.22 | p3nguin | It can only be controlled in the dial plan or in the phone. |
| 01:32.25 | p3nguin | There is no other place. |
| 01:34.51 | p3nguin | I would first look on the phone, since that is what users have access to. |
| 01:35.26 | p3nguin | It could also be determined by watching the CLI during a call to the extension that the phone uses. |
| 01:35.58 | newmember | k |
| 01:36.15 | p3nguin | If the phone is doing it, you would see some type of diversion in the SIP debug. |
| 01:37.56 | newmember | :Extension = 5654 3|Dial|Local/912xxxxxxxxx@internal |
| 01:38.07 | newmember | thats what is in the logs |
| 01:38.11 | dijib | p3nguin: did you find anything yet to trade for this sangoma a200? |
| 01:38.21 | p3nguin | Nope. |
| 01:38.31 | dijib | well then i better send it to u anyways |
| 01:38.40 | dijib | how is the us these days? |
| 01:39.05 | dijib | do i have any use for it whatsoever... ? it the forseeable future |
| 01:39.05 | p3nguin | I've been wishing I had stuff to trade. People are always having things to get rid of that I could potentially use. |
| 01:39.26 | dijib | you would be more likely to deploy it than me |
| 01:39.41 | dijib | althought it did install automatically in the last build |
| 01:39.50 | p3nguin | Send it over if you're tired of seeing it. |
| 01:40.00 | dijib | my server i think is running well being a nuub but i think i need to harden it now |
| 01:40.14 | p3nguin | Encase it in cement. |
| 01:40.32 | dijib | i kindof like seeing it with the red lights in the chassis, but thats all the effect i enact with it |
| 01:41.15 | p3nguin | If you wanted to encase it in cement but still see the LEDs, you could extend the wires and put the LEDs on the outside of the concrete case. |
| 01:41.37 | dijib | im broke as hell these days, although i have aquired 2x 53" proj lcd |
| 01:41.47 | dijib | one working one sony with lamp out |
| 01:41.51 | dijib | need lamp |
| 01:42.23 | dijib | no corp phone systems in the bins yet |
| 01:44.05 | newmember | I see the error in the log files: : Extension = 0: DB_NOTFOUND: No matching key/data pair found |
| 01:45.22 | dijib | p3nguin: email me some delivery info and i will see about shipping this usps.canadapost |
| 01:47.04 | dijib | dijib at hot male. com |
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| 01:49.20 | dijib | k back to my game, later lads |
| 01:50.36 | powerunits | team one more small issue.. my asteriskcdr was just working fine. i don't what i did but now its not showing any result in cdr table... when i call on extension. asterisk does update cdr table but it does not enter any record into it. |
| 01:51.19 | powerunits | i can only see 0 0 into cdr table |
| 01:51.27 | powerunits | any idea why is that? |
| 01:52.45 | powerunits | http://pastebin.com/vmdTunWP |
| 01:52.52 | powerunits | here is my cdr_mysql.conf |
| 01:55.59 | powerunits | where as i can see complete call logs in tail -f /var/log/asterisk/cdr-csv/Master.csv |
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| 03:41.05 | vassilux | hi alls, I have some troubles with dahdi 2.6.1 and wct4xxp driver. my carte is TE205P 5th Gen. I got "No interrupts" in dmesg. First span stay green all time. I can provide any log cause I'm out of the office but may be someone can give me an idea :-). I tried some different version of dahdi with the same result. |
| 03:42.06 | WIMPy | Well, no interrupts looks like a hardware issue. |
| 03:42.32 | vassilux | I changed the cart slot with the same result. I can't desactivate usb in the BIOS cause I must install r-series with usb plug. |
| 03:43.49 | vassilux | Yes but I found in the goggle the someone speaking about alarmdebounce=2500 |
| 03:44.06 | WIMPy | That doesn;t change interrupts. |
| 03:45.10 | vassilux | in the bios ? |
| 03:45.35 | WIMPy | Wherever, the issue may be. |
| 03:46.02 | WIMPy | BIOS, board, the card or just a bad connection. |
| 03:49.26 | vassilux | I'm not sure if it is a bad connection cause I used in my dev box one other TE205P and for test just make crossover connection beetwen two ports. |
| 03:50.05 | WIMPy | Between the card and the board. |
| 03:50.40 | WIMPy | If the card doesn't work, it doesn't matter what you connect to the ports. |
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| 04:43.55 | ectospasm | vassilux: call Digium technical support +1.256.428.6161, they will help you as long as your TE205 is registered. |
| 04:44.23 | ectospasm | ...they have technicians available from 7pm Sunday through 7am Saturday, US CDT (UTC-5) |
| 04:44.50 | ectospasm | (disclaimer: I work for Digium Technical Support) |
| 04:45.06 | pabelanger | LIES |
| 04:45.41 | pabelanger | (disclaimer: I used to work for Digium) |
| 04:52.28 | KNERD | on a weekend? Buiahaha! |
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| 05:28.07 | MosquitoCR | Anyone knows a good nagios plugin for Asterisk 1.6 and 1.8 ? |
| 05:48.10 | KNERD | nope |
| 05:48.13 | KNERD | never heard of it |
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| 05:53.37 | din3sh | mrning all |
| 05:56.14 | KNERD | evenin' guvnor |
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| 06:10.02 | schmidts | good morning |
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| 06:27.51 | Celorio10 | Good morning from Spain guys! |
| 06:32.38 | KNERD | hola crayola cero |
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| 06:36.48 | Samuel_CTX | MosquitoCR: we have some shell scripts that query asterisk (rasterisk -x) and give the results to Nagios |
| 06:43.13 | Celorio10 | Anyone can help me with asterisk problem? |
| 06:43.35 | drmessano | Maybe |
| 06:43.55 | Celorio10 | Do you know GAI-TRONICS Phones? |
| 06:44.15 | MosquitoCR | Samuel_CTX: can you send them ? |
| 06:44.41 | drmessano | Celorio10: Looking at their website.. |
| 06:45.10 | drmessano | Celorio10: What is the problem? Still waiting for it |
| 06:48.21 | Celorio10 | i have 8 IP phones (Auteldac 4 VoIP without buttons) and i need, when i off hook this phones calls to a extension, i have configured the sip.conf and the extensions.conf but doesn't works :( |
| 06:48.45 | KNERD | nope..nobody here canhelp you |
| 06:48.49 | KNERD | what is asterisk? |
| 06:48.50 | EmleyMoor | Anyone know a good RJ11-431A adapter with ring cap? |
| 06:48.57 | KNERD | nope..nobody |
| 06:49.23 | EmleyMoor | KNERD: Read the topic and stop being a fool |
| 06:49.26 | Celorio10 | in the phones i put in memory 1 "sip:103@192.168.1.100" to call (the same that i use with 3CX PBX) |
| 06:49.50 | KNERD | then why did you ask a fool question "anyone know..." |
| 06:50.01 | KNERD | i responded correctly |
| 06:51.02 | *** join/#asterisk af_ (~getsmart@88.149.212.175) |
| 06:52.32 | Celorio10 | 103 is the extension and 192.168.1.100 is the IP of the server who has Asterisk |
| 06:53.02 | drmessano | Celorio10: Do you see the call attempt in the CLI? |
| 06:53.23 | Celorio10 | No |
| 06:53.35 | Celorio10 | then i think that the problem is in the call from the phone |
| 06:54.16 | RiceCracker | is your sip extension registering? "sip show peers" |
| 06:54.26 | Celorio10 | yes its all ok |
| 06:55.33 | KNERD | *43 work fine? |
| 06:55.43 | Celorio10 | The problem is that i dont have buttons in the phone and i have tu put the extensión in the memory |
| 06:55.49 | Celorio10 | *to |
| 06:56.37 | Celorio10 | and i try directly with 103, SIP/103,103:pass@192.168.1.100 ...........and nothing :( |
| 06:56.49 | drmessano | Celorio10: If the call isn't hitting asterisk, then it's the phone |
| 06:57.03 | RiceCracker | can you make it work using a softphone like xlite? |
| 06:57.29 | Celorio10 | yes, i supose it :( nobody here works with GAI-TRONICS phones? |
| 06:58.03 | drmessano | Celorio10: I doubt you're going to find someone at random who has used something so specific. Might want to contact the vendor |
| 06:58.24 | drmessano | Celorio10: Sounds like you have a dialplan issue with the device. Not sure what else to tell you |
| 06:58.35 | Celorio10 | RiceCracker: today i will try to use x-lite |
| 06:59.23 | Celorio10 | drmessano i'm trying to contact gai-tronics i hope have an answer today....thank you so much! |
| 06:59.55 | RiceCracker | agreed with drmessano, some phones do require you to input the dialplan on the phone. if you can get it to work with x-lite, then for sure its a setting on your phone. |
| 07:00.36 | Celorio10 | Then this morning i will try to works with x-lite! Thanks RiceCracker!! |
| 07:00.46 | RiceCracker | np |
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| 07:51.18 | *** join/#asterisk jkroon (~jkroon@dsl-244-22-173.telkomadsl.co.za) |
| 07:51.25 | jkroon | hi guys, with dahdi-2.6.1 and asterisk 1.8.12.1 we're seeing analog (FXS signalling) going into strange states from time to time, and presumably dropping calls. |
| 07:51.33 | jkroon | has anyone else seen this? known resolutions? |
| 07:51.41 | jkroon | a restart of asterisk usually fixes it for a few hours... |
| 07:52.22 | jkroon | [Jun 26 09:42:40] WARNING[10208] sig_analog.c: Ring/Off-hook in strange state 6 on channel 1 |
| 07:52.33 | jkroon | after that we're dead in the water until restart |
| 07:55.23 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
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| 08:04.26 | *** join/#asterisk applemango (~ahmad.2.h@182.185.232.172) |
| 08:04.40 | applemango | hello every one |
| 08:04.49 | applemango | i am new with asterisk |
| 08:04.51 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:04.55 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 08:05.11 | applemango | i am facing a problem |
| 08:05.14 | *** join/#asterisk ChannelZ (channelz@173.160.35.173) |
| 08:05.42 | applemango | <PROTECTED> |
| 08:06.06 | applemango | there does'nt raised AgentCalled event |
| 08:06.39 | applemango | could any one help me |
| 08:06.44 | applemango | ? |
| 08:08.29 | applemango | helllllo |
| 08:08.36 | applemango | anybody there] |
| 08:08.39 | applemango | ? |
| 08:09.16 | aberrios | applemango: I believe that's because the queue call has already been answered. |
| 08:09.42 | jkroon | https://issues.asterisk.org/view.php?id=811 - this seems to be related to my problem, however, I can't imagine that this is still open. |
| 08:10.22 | jkroon | well, the fix there is to explicitly "Answer" and "Ringing" an analog channel, which IMHO is wrong, why can't we simply answer the incoming analog channel once the answering SIP channel answers? |
| 08:11.33 | applemango | actully i am using manager API |
| 08:11.58 | applemango | and i want to triger events when agent are called |
| 08:12.14 | ChannelZ | jkroon: does the console actually show proper call progress? |
| 08:12.23 | applemango | when first time call enters in queue its work fine] |
| 08:12.29 | applemango | yes |
| 08:12.46 | applemango | but when agent transfer call to other agent |
| 08:12.54 | applemango | there is no event |
| 08:12.58 | aberrios | applemango: attended or blind? |
| 08:13.07 | applemango | attended |
| 08:13.43 | jkroon | ChannelZ, call comes in, enters the dialplan, goes through everything it does normally, SIP/ channel starts ringing, SIP/ channel answers and then it's claimed that the call drops, so yea, I think it does. |
| 08:14.02 | ChannelZ | That isn't really what I asked |
| 08:14.04 | aberrios | applemango: you know what, I'm not entirely sure what the order of events is here but I would reckon since the Queue Call gets answered you would never see another AgentCalled event for that queue call again |
| 08:14.14 | ChannelZ | Turn on some verbose and pastebin the whole process |
| 08:14.22 | ChannelZ | Let |
| 08:14.36 | ChannelZ | I'd like to see what it thinks is happening to the channels |
| 08:15.28 | applemango | channelZ : i am facing call tranfer issue |
| 08:16.00 | applemango | mean when agent transfer a call to other agent how could i get that event |
| 08:16.01 | applemango | ? |
| 08:16.16 | ChannelZ | sorry I'm talking to jkroon not you |
| 08:16.51 | ChannelZ | I've no idea on your problem, I don't have any extensive experience with queues sorry |
| 08:17.03 | applemango | its okay |
| 08:21.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:22.39 | jkroon | ChannelZ, I'll get stephen to trap one in the act of "failing", verbosity at level 10? |
| 08:23.16 | ChannelZ | 3 or 4 is fine (I forget, * doesn't actually log anything above 4 I think) |
| 08:24.56 | jkroon | I've reconfigured the CLI stuff now as well, which apparently from internet reports seems to mostly be what triggers it ... |
| 08:25.14 | ChannelZ | eh? |
| 08:27.30 | *** join/#asterisk shadebob (~shadebob@41.141.164.56) |
| 08:28.04 | applemango | could som1 tell me how to get CallTranfer Event |
| 08:28.11 | applemango | manager api |
| 08:28.12 | applemango | ? |
| 08:28.26 | *** join/#asterisk savenger (~savenger@p5496EF91.dip.t-dialin.net) |
| 08:35.19 | kaldemar | applemango: where did you come up with such an event? |
| 08:38.47 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:40.52 | ChannelZ | wanders off to bed |
| 08:41.44 | savenger | Hi again, I still cannot make asterisk open the tls port. Can anyone of you show me sip.conf where tls is working for you? I followed https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial but it did not work. |
| 08:42.07 | kaldemar | applemango: there is one called Transfer though. |
| 08:42.19 | savenger | asterisk is not giving any error messages about tls |
| 08:50.21 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 08:54.34 | jkroon | ChannelZ, i googled a bit and found two or three reports that claims to have tampered with cid detections settings that ended up resolving their problems. |
| 08:58.12 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
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| 09:09.06 | savenger | Guys! When I start asterisk via "asterisk -cvvvvvvvvvv" it opens the tls port! If I start it with "/etc/init.d/asterisk start" it doesn't!!!! Do you know why? |
| 09:10.06 | kaldemar | see what your init script does. |
| 09:18.15 | *** join/#asterisk jol (jol@ph4.se) |
| 09:23.00 | *** join/#asterisk dandate2 (~dan@180.190.218.211) |
| 09:23.33 | dandate2 | in the pap2 ATA device does Silence Threshold set to "High" mean long wait for suppression or short? |
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| 11:18.19 | AdvoWork | Hi there, i'm using trixbox ontop of asterisk, our calls go out via an ISDN line. Initially they went out over SIP. After changing from SIP > ISDN the caller id has started showing as anonymous. Any idea how to debug this at all? |
| 11:20.29 | WIMPy | Are you familiar with the Asterisk CLI? |
| 11:21.46 | WIMPy | If not, then |
| 11:21.50 | WIMPy | ~trixbox |
| 11:21.51 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 11:26.07 | *** join/#asterisk af_ (~getsmart@88.149.212.175) |
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| 11:42.09 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 12:02.35 | savenger | still another problem with tls: "Problem setting up ssl connection: error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure". Debug-Output: http://pastebin.com/uNYhuUHN . Any clue what I have to do to fix this? |
| 12:05.22 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:06.26 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 12:47.41 | slav3_kitten | stupid question time |
| 12:47.48 | slav3_kitten | can asterisk do SMS? |
| 12:48.51 | schmidts | slav3_kitten yes ;) |
| 12:49.10 | slav3_kitten | really needs time to read the book |
| 12:49.48 | slav3_kitten | been trying to figure out A) where to get the latest firmware for my 7921, and B) how to configure the 7911 and 7921 phones he has |
| 12:50.16 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:58.17 | [TK]D-Fender | * doesn't really do SMS. |
| 12:59.50 | *** join/#asterisk n3hxs (~ed@50-76-3-4-static.hfc.comcastbusiness.net) |
| 13:01.08 | slav3_kitten | [TK]D-Fender, now i'm confused |
| 13:01.53 | [TK]D-Fender | slav3_kitten, All docs seem to show is support of SENDING SMS over E1. Nothing else |
| 13:02.43 | slav3_kitten | hrmm |
| 13:03.51 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-qkxczknmazoblnmy) |
| 13:08.32 | *** join/#asterisk francisvgarcia (~francisvg@190.80.239.124) |
| 13:09.28 | francisvgarcia | Hi everyone |
| 13:09.57 | francisvgarcia | I got a question for u guys |
| 13:10.46 | francisvgarcia | do you have a patch for the DTMF issues on chan_mobile when using disa? |
| 13:12.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-uifvojiaeznadiiu) |
| 13:12.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:15.05 | [TK]D-Fender | DISA has not special implication with DTMF |
| 13:15.10 | *** join/#asterisk oxnard (~oxnard@ws25-ilnk.elkhart.net) |
| 13:17.27 | francisvgarcia | [TK]D-Fender what happens is that I got asterisk working ok and I am using chan_mobile for connecting asterisk to the GSM network as everyone knows |
| 13:18.00 | [TK]D-Fender | francisvgarcia, I don't, but that's fine. DTMF either works or it doesn't. DISA is not special. |
| 13:18.02 | francisvgarcia | the issue is when I make an incoming call to the asterisk server it responds but no DTMF tones are procesed by asterisk |
| 13:19.07 | francisvgarcia | only the * and the 0 are reconigzed by asterisk |
| 13:19.11 | *** join/#asterisk rossand (~aross@CPE0023695c1457-CM00222ddf42dd.cpe.net.cable.rogers.com) |
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| 13:31.26 | aberrios | slav3_kitten: I can confirm I had * sending and receiving SMS over E1. |
| 13:31.43 | slav3_kitten | cool |
| 13:31.50 | aberrios | twas 1.6 tho, havent tried with 1.8 |
| 13:32.08 | WIMPy | aberrios: You sent a whole service? |
| 13:32.20 | aberrios | expensive for sending here though, obviously free to receive |
| 13:32.25 | aberrios | SMSs'eses's |
| 13:32.51 | aberrios | messagesss's'ses |
| 13:44.01 | *** join/#asterisk slashp (~corte@cpe-071-068-040-205.carolina.res.rr.com) |
| 13:46.17 | slav3_kitten | i am mostly curious sending sms alerts of voicemail |
| 13:46.21 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:46.21 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:47.43 | _Corey_ | slav3_kitten: You could look into using an externotify application with a provider like Twilio or Voxeo who offer an SMS api |
| 13:47.50 | [TK]D-Fender | slav3_kitten, Get some other service to do it then. that's what the "pager" option in voicemail.conf is for |
| 13:48.04 | _Corey_ | (or just use an e-mail gateway if you want to manage that) |
| 13:48.14 | [TK]D-Fender | slav3_kitten, Most cell co's offer an e-mail>SMS option |
| 13:48.46 | slav3_kitten | [TK]D-Fender, my cell is really bare bones basic |
| 13:49.08 | [TK]D-Fender | slav3_kitten, it isn't your PHONE that needs to offer anything |
| 13:49.25 | [TK]D-Fender | slYour CELL COMPANY probably offers an e-mail address to send you SMS's |
| 13:49.26 | slav3_kitten | well i should say more verizon charges for everything |
| 13:49.41 | slav3_kitten | i'll check into i |
| 13:49.43 | slav3_kitten | t |
| 13:49.59 | slav3_kitten | < misspoke the first time as he recently woke up again and has the flu |
| 13:51.12 | Katty | hello my asterisk does not work at all how to fix plz |
| 13:51.37 | aberrios | You're not winning this time Katty .... |
| 13:51.51 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:52.04 | slashp | man i'm going crazy with a 481 Call leg/transaction does not exist response...would someone mind reading over a post and giving any input they could? i would really appreciate it...the post is located here: http://fonality.com/trixbox/forums/community-edition/help/lync-2010-kamailio-trixbox-2623-asterisk-14 |
| 13:52.29 | [TK]D-Fender | sla1.4 is EOL and no longer supported |
| 13:52.37 | [TK]D-Fender | slashp, 1.4 is EOL and no longer supported |
| 13:52.51 | Katty | aberrios: i'm always winning. |
| 13:52.55 | Katty | aberrios: I"M A WINNER |
| 13:53.09 | beek | waves to [TK]D-Fender and hugs katty |
| 13:53.10 | [TK]D-Fender | And Trixbox is dead. Just completely and utterly dead |
| 13:53.18 | Katty | hugs beek |
| 13:53.22 | Katty | beek: how're you dear? |
| 13:53.27 | beek | Doing fine. Yourself? |
| 13:53.28 | [TK]D-Fender | beek, 'lo |
| 13:53.30 | [TK]D-Fender | Katty, Mew. |
| 13:53.38 | Katty | [TK]D-Fender: howdy. |
| 13:54.12 | leifmadsen | omg my asterisk is down and we're loosing<sic> money by the second! I need an adult! |
| 13:54.35 | Katty | oh noes!!! |
| 13:54.39 | WIMPy | Asterisk adultery? |
| 13:54.44 | slashp | [TK]D-Fender, i understand that (if i had a choice I wouldn't be on this version, i'd be using PBX in a flash or something)..it's just the way our production environment is currently set up, but if no one will help i'll pass the message to my boss :) |
| 13:55.10 | [TK]D-Fender | PIAF = different flavour of crap... but OK.... |
| 13:55.14 | slashp | haha is it? |
| 13:55.17 | slashp | what would you recommend? |
| 13:55.26 | [TK]D-Fender | Actually.. not that bad. ELASTIX was the other bad one |
| 13:55.32 | slashp | yeah i don't like ELASTIX that much |
| 13:55.55 | [TK]D-Fender | slashp, Currently AsteriskNOW, or the FreePBX ISO if you're looking at the FreePBX angle... |
| 13:56.06 | slashp | yeah i've heard good things about AsteriskNOW |
| 13:57.46 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 13:58.37 | beek | slashp: astlinux |
| 13:59.19 | slashp | haven't seen this one before :) |
| 14:01.09 | *** join/#asterisk mbrit (~magnux@186.120.97.195) |
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| 14:03.46 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:03.46 | francisvgarcia | any help about chan_mobile and DTMF on incoming calls? |
| 14:08.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:08.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.24 | *** join/#asterisk polysics (~polysics@host41-65-dynamic.54-79-r.retail.telecomitalia.it) |
| 14:09.26 | polysics | hello |
| 14:09.45 | polysics | does an AMI command exist to check if a context or extension exists? |
| 14:09.49 | polysics | before I add it |
| 14:10.22 | polysics | or a cli command of sorts |
| 14:10.31 | polysics | I use dialplan add extension |
| 14:10.37 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 14:10.51 | WIMPy | dialplan show? |
| 14:12.45 | polysics | yeah, I think I will have to parse that |
| 14:13.56 | WIMPy | You did notice the optional parameters? |
| 14:18.22 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 14:18.29 | polysics | such as? |
| 14:19.13 | WIMPy | [[extension@]context] |
| 14:22.22 | polysics | oh, i can use the context |
| 14:22.39 | polysics | it's not the same thing as an explicit "question" but still works, thanks |
| 14:26.32 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:26.40 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:28.10 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:32.09 | EmleyMoor | Managed to get temporary inside lines wired up now - but my router is in a mess so can't actually restore my phones until the new one arrives |
| 14:33.23 | EmleyMoor | I've told people "Taking the number? I'm taking the whole exchange with me!" |
| 14:36.24 | *** join/#asterisk mbrit (~magnux@186.120.97.195) |
| 14:37.53 | cusco | hello |
| 14:38.28 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:38.33 | cusco | when a caller is joining a queue, is there some how that I can allow him to press 8, only after 50 seconds in queue? |
| 14:41.11 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:42.36 | [TK]D-Fender | cusco, Nope. |
| 14:43.24 | slav3_kitten | what's pressing 8 do in your system cusco? |
| 14:44.53 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:45.19 | autofsckk | good morning everybody |
| 14:45.45 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 14:46.26 | ThinkGNU- | Is there a good way to allow users to input the numbers that will be dialed in FollowMe()? |
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| 14:47.45 | *** join/#asterisk MosquitoCR (~mosquitoc@186.176.157.48) |
| 14:50.08 | [TK]D-Fender | slav3_kitten, Whatever he wants it to be |
| 14:50.26 | [TK]D-Fender | ThinkGNU-, Considered Read() yet? |
| 14:50.27 | slav3_kitten | [TK]D-Fender, obviously but i was curious what he was doing with it |
| 14:50.48 | [TK]D-Fender | slav3_kitten, Good odds on "leave a message", or "emergency" |
| 14:51.18 | slav3_kitten | *nods* what i was thinking. or operator |
| 14:54.09 | Greenlight | Can asterisk "translate" DTMF modes between trunks, for example I have a trunk with an incomming call dtmfmode info and then that dials out on another trunk with dtmfmode rfc2833 will Asterisk "translate" from one to the other? |
| 14:54.17 | cusco | slav3_kitten: asks for the contact number to be contacted later by us |
| 14:54.23 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 14:54.34 | slav3_kitten | ah interesting :) |
| 14:54.38 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v026-202.mobile.uci.edu) |
| 14:55.51 | WIMPy | Greenlight: Should work. But I remember there was an issue with one of the translations. IIRC the length was lost or something. |
| 14:58.49 | *** join/#asterisk Defraz (~Defraz@67-60-210-130.cpe.cableone.net) |
| 14:59.32 | Katty | why am i hungry |
| 14:59.34 | Katty | it's only 10 |
| 15:00.31 | newtonr | Katty: it's always lunchtimes somewhere |
| 15:00.54 | Qwell | Katty: come make me lunch! |
| 15:01.02 | Qwell | excuse to leave early for lunch |
| 15:01.05 | coppice | Katty: isn't 10 maximum on the hunger scale? |
| 15:01.25 | coppice | 10 is the right time for lunch in Pheonix |
| 15:02.03 | [TK]D-Fender | ThinkGNU-, No. |
| 15:02.27 | [TK]D-Fender | ThinkGNU-, That conf & app are virtually worthless. Nothing you can't do in dialplan yourself |
| 15:03.40 | [TK]D-Fender | ThinkGNU-, If you need this functionality.. apparently |
| 15:03.52 | [TK]D-Fender | ThinkGNU-, and stop using /notice for talking to poeple. |
| 15:05.34 | cusco | another question.. in AEL using goto context,exten,priority, it states: no label main|6 exists in the context Lusomundo-novo or its inclusions |
| 15:05.42 | cusco | there is extension main |
| 15:05.57 | cusco | dialplan show context, shows 6 |
| 15:06.53 | [TK]D-Fender | cusco, Stop using "|" as a delimiter |
| 15:07.14 | cusco | I'm not |
| 15:07.20 | cusco | asterisk is outputing it that way |
| 15:07.28 | [TK]D-Fender | cusco, Show us the precise code & output |
| 15:07.39 | cusco | I'm using: goto Lusomundo-novo,main,6 |
| 15:07.39 | cusco | ok |
| 15:07.48 | cusco | [Jun 26 16:03:57] ERROR[25806]: ael/pval.c:1316 check_goto: Error: file /etc/asterisk/extensions_partner/extensions_Lusomundo-novo.ael, line 124-124: goto: no label main|6 exists in the context Lusomundo-novo or its inclusions! |
| 15:08.03 | cusco | <PROTECTED> |
| 15:08.05 | *** join/#asterisk DarthExpeditor (~IceChat9@adsl-76-235-175-148.dsl.klmzmi.sbcglobal.net) |
| 15:08.07 | cusco | ... |
| 15:08.08 | [TK]D-Fender | cusco, actual ful dialplan, dump of "dialplan show" and athe FULL call to Goto, not just the ERROR |
| 15:08.12 | [TK]D-Fender | ~pb |
| 15:08.12 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:08.13 | [TK]D-Fender | ^^^ |
| 15:08.22 | cusco | ok |
| 15:10.16 | cusco | [TK]D-Fender: http://paste.debian.net/176423/ here is dialplan show, and the ael calling goto, (wich is not yet in dialplan show, as asterisk errored out on 'ael reload) |
| 15:10.32 | *** join/#asterisk fulcan (~brads@li186-148.members.linode.com) |
| 15:11.11 | fulcan | can someone recommend an asterisk friendly sip/sms carrier |
| 15:11.13 | fulcan | ? |
| 15:11.42 | cusco | I would not use sms in asterisk, rather integrate it externally |
| 15:12.34 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:12.36 | fulcan | cusco smsq looks like it will do exactly what I want it to do. why is this? |
| 15:20.49 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:20.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:23.44 | WIMPy | tries a little OT, although it's related to phones. Does someone have experience with LCDs with a glued flex band between the controller and the glass? The glue seems to fail with time leading to |
| 15:23.50 | WIMPy | dead rows or collumns. |
| 15:24.03 | WIMPy | Is it possible to fix those? |
| 15:25.05 | coppice | get Superman to use his laser beam eyes |
| 15:27.58 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 15:32.18 | cusco | [TK]D-Fender: ok I found out I can set a label/alias for a priority |
| 15:32.22 | cusco | it works |
| 15:33.43 | cusco | another question... can I make a member leave a queue, playback something, and make him join the queue and keep the same position? so he does not go to end of line? |
| 15:33.48 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 15:40.52 | slav3_kitten | so i might need some help help with sccp... |
| 15:41.07 | slav3_kitten | anyone here ever use deploy that chan? |
| 15:42.22 | WIMPy | Yes, but I probably won't be able to help, not understnading how it works. |
| 15:44.49 | slav3_kitten | *nods* |
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| 15:51.50 | cusco | math works like NoOp($[3-2]); |
| 15:51.52 | cusco | right? |
| 15:52.06 | leifmadsen | use the MATH() function |
| 15:52.11 | leifmadsen | but ya that shold work too |
| 15:52.22 | cusco | !nan |
| 15:52.24 | leifmadsen | but you'll likely get more consistent results if you use the function |
| 15:56.27 | *** join/#asterisk rjb_77777 (~rjb_77777@rrcs-74-62-26-26.west.biz.rr.com) |
| 15:56.57 | cusco | its ok, calculating ints |
| 15:57.16 | cusco | set epoch, then checking setted var with current eopch |
| 15:57.29 | cusco | the var had a typo, thus not working heh.. |
| 15:58.16 | rjb_77777 | Hey all. I have a routing issue I am hoping you can help with. |
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| 16:00.07 | rjb_77777 | We have an Asterisk system that was configured by a third party vendor. We have Polycom 300 series we give to users who work remotely. |
| 16:00.36 | rjb_77777 | The remote phones work great unless they have a Vonage or Verizon on the same network. |
| 16:01.10 | rjb_77777 | About 2/3 of the remote users can't use the remote phones. |
| 16:01.40 | rjb_77777 | Is there a way to set up a VPN device at the remote location and terminate the phones behind the Asterisk? |
| 16:02.08 | WIMPy | Yes, but that's something for #networking. |
| 16:02.25 | WIMPy | Unless you get phones that will do it for you. |
| 16:03.42 | rjb_77777 | Thanks wimpy. We have everything set up properly and connectivity both directions, but the Asterisk won't communicate with the phones. They don't register. They will pull NTP, DNS, and FTP down, but the Asterisk won't communicate with it on the LAN, only out the WAN. |
| 16:04.24 | rjb_77777 | What phones can do VPN. What would be the endpoints? |
| 16:04.38 | leifmadsen | I'm not sure any do that |
| 16:04.55 | leifmadsen | unless you have a laptop on the vpn using a softphone |
| 16:05.03 | leifmadsen | or you place the device behind a VPN hardware device |
| 16:05.05 | WIMPy | Not on the LAN? |
| 16:05.22 | rjb_77777 | Is there a config that tells the Asterisk which interface the phones are on? |
| 16:05.41 | WIMPy | Snom 370 does OpenVPN. The 8xx series probably as well, but I haven;t looked in to them. I think they are just over the top. |
| 16:06.09 | rjb_77777 | The issue is that the Asterisk will not route to the phones if the next hop is on the LAN. I have set permanent routes on eth1 pointing to the VPN firewall. |
| 16:06.13 | WIMPy | It will find out when the phone tries to register. |
| 16:06.51 | WIMPy | That sounds like a general networking issue. |
| 16:09.23 | *** join/#asterisk DarthExpeditor (~IceChat9@c-24-11-175-127.hsd1.mi.comcast.net) |
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| 16:10.27 | rjb_77777 | We have a hardware VPN device at the remote location connecting to a VPN firewall at the Asterisk. The tunnel terminates behind the Asterisk. We have connectivity in both directions, but the Asterisk will not communicate with the phones unless they have a LAN address. It won't route the connections. |
| 16:11.10 | rjb_77777 | Where can I look to configure the Asterisk to also register phones that are on subnets other than the DHCP scope? |
| 16:12.13 | WIMPy | We need more detailed ionformation. So far I don't see an Asterik Issue. |
| 16:12.17 | [TK]D-Fender | * isn't restricted to any specific scope |
| 16:12.35 | [TK]D-Fender | It doesn't care where your phone is unless YOU restricted it in your configs |
| 16:12.55 | rjb_77777 | Where would that restriction be configured? |
| 16:12.58 | WIMPy | Or somewhere on the network. |
| 16:13.53 | WIMPy | Does Asterisk receive the registration requests at all? |
| 16:14.24 | rjb_77777 | Yes, the requests are hitting the *. We can get good DNS, NTP, and FTP. |
| 16:15.27 | WIMPy | And what about the resonse? What does it respond? Where does it go? |
| 16:16.18 | *** join/#asterisk navaismo (~navaismo@189.144.194.18) |
| 16:16.40 | rjb_77777 | Response goes out the WAN. |
| 16:17.01 | rjb_77777 | I can ping the phone private IP from the * itself. Routing is configured properly. |
| 16:17.48 | slav3_kitten | grumble grumble grumbles about cisco firmware |
| 16:18.05 | WIMPy | I don't want to know about pings. I want to know about the response. |
| 16:18.20 | rjb_77777 | Response goes out the WAN. |
| 16:20.37 | WIMPy | Show us a SIP debug. |
| 16:20.51 | rjb_77777 | New territory. How do I pull that? |
| 16:21.02 | *** join/#asterisk viro (~viro101@cpe-71-72-174-83.woh.res.rr.com) |
| 16:21.08 | WIMPy | sip set debug on |
| 16:21.29 | WIMPy | You can restrict it to an ip if you want. |
| 16:22.07 | rjb_77777 | OK. Will have to wait until tomorrow. The remote phone is at home right now. |
| 16:23.32 | rjb_77777 | Thanks Wimpy. |
| 16:24.17 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 16:26.07 | *** join/#asterisk senator (lebbeous@nox.esilibrary.com) |
| 16:35.51 | *** join/#asterisk rx4change (~joshelson@216.241.45.82) |
| 16:36.16 | rx4change | Does anyone know if there is a solution for Page to use confbridge instead of meetme in Asterisk 1.8? |
| 16:36.34 | leifmadsen | there is not |
| 16:36.58 | WIMPy | Not in 1.8. |
| 16:37.00 | rx4change | well that settles that.... is there a solution on 10? |
| 16:37.00 | leifmadsen | updating app_page to not use app_meetme hooks would require some developmen effort. I think that may have been changed in trunk for Asterisk 11 |
| 16:37.22 | leifmadsen | rx4change: check the CHANGES and UPGRADE.txt files in trunk to determine that kind of change |
| 16:37.34 | WIMPy | Not on 10, either? |
| 16:37.57 | leifmadsen | I think that change was post Asterisk 10 |
| 16:38.12 | leifmadsen | as part of the clean up to get everything switched over to app_confbridge usage |
| 16:38.19 | leifmadsen | it might be in 10, but I can't confirm that without effort |
| 16:38.31 | leifmadsen | mjordan might know for sure |
| 16:40.00 | rx4change | In my representative 10 install, I don't even have Page installed. I assume would be a dahdi dependency issue. |
| 16:40.53 | mjordan | WIMPy: rx4change: that is an item that was addressed for Asterisk 11 |
| 16:41.10 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Projects#Asterisk11Projects-apppagerefactor |
| 16:43.01 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:43.07 | mjordan | and yes, that should already be in trunk - it was committed on March 10th |
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| 17:10.11 | senator | hi all. if i have a pri span correctly configured with dahdi and asterisk 1.6.2.18, shouldn't i have commands available in the asterisk console that start with 'pri', like 'pri show spans', etc |
| 17:10.15 | senator | ? |
| 17:11.25 | WIMPy | Yes. |
| 17:11.39 | WIMPy | But you want to upgrade anyway. |
| 17:11.56 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 17:13.24 | senator | WIMPy: thanks. (and yes i do) so /proc/dahdi/1 shows 23 channels clear/in-use and the 24th as HDLCFCS (this is a pri of 'national' type with b8zs/esf coding, and i do see channels in asterisk console with 'dahdi show channels' and the like, |
| 17:14.10 | senator | but outbound calls fail with unknown status and the telco says 'the d-channel is down, nothing we can do' |
| 17:14.14 | senator | any hints as to what i'm missing? |
| 17:14.23 | senator | i suspect it's the same thing stopping me from having any pri* commands |
| 17:14.27 | WIMPy | libpri probably |
| 17:14.31 | senator | doh |
| 17:14.36 | senator | that would be too obvious |
| 17:14.37 | senator | cheks |
| 17:14.39 | senator | *checks |
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| 17:45.23 | volga629 | Hello All |
| 17:47.18 | volga629 | I created new test account on mail server and I tried leave voice message and that what I see http://fpaste.org/NU45/ |
| 17:47.46 | volga629 | Is that actual delivery of email ? |
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| 19:01.23 | wonko | ok, so for inbound google voice calls do i *really* need to Wait() for 8 seconds? |
| 19:07.31 | file | 2-3 is enough, but you do need to send a DTMF digit of 1 |
| 19:07.45 | file | (even the google talk plug-in in gmail does it) |
| 19:08.19 | wonko | yeah, i've got the sendDTMF(1) i nthere |
| 19:08.27 | wonko | just wasn't sure if i really needed to wait so long |
| 19:09.11 | wonko | awesome, thanks! |
| 19:09.19 | wonko | now to clean up that awful callerid string. :) |
| 19:13.22 | *** join/#asterisk Phican (~Phican@c-75-71-178-102.hsd1.co.comcast.net) |
| 19:13.41 | Phican | Anybody mind dabbling with a sip URI issue? |
| 19:14.12 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 19:16.23 | Phican | Point and laugh, but Im using pbx in a flash, purple edition with asterisk 1.8. Incoming sip URI works fine, trying to dial out with sip uri to known good addresses errors out with "Congested/busy" Rather hoping to find a way to get more verbosity then I have, or find a way to troubleshoot the issue. |
| 19:18.01 | drmessano | The Asterisk CLI is where you want to be.. enable sip debug and look at the call |
| 19:18.26 | Phican | Alright, Ill give that a shot, Ive been defaulting with asterisk -rvvv |
| 19:18.47 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:19.07 | newtonr | Phican: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
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| 19:19.58 | Phican | also, anything wrong with this? |
| 19:20.16 | Phican | Dial(SIP/<ext>@Domain.com) ? |
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| 19:27.05 | [TK]D-Fender | Phican, I highly recommend you look at the ACTUAL ATTEMPT |
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| 19:39.49 | Phican | I plan to, however I wanted to make sure I had the dial string correct, several examples use a matching format, Im using an extension to dial by sip |
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| 19:51.49 | jero | hi, I have a strange behavior in asterisk 1.8, can't get call waiting on a dahdi channel with callwaiting=yes in chan_dahdi.conf; dunno if related but "dahdi show channel 1" shows CallWait: <None> |
| 19:53.19 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 19:54.57 | [TK]D-Fender | jero, And what do you have plugged in? |
| 19:59.30 | jero | [TK]D-Fender: i think it's some kind of custom device :> |
| 19:59.48 | *** join/#asterisk darrenlooby (~darrenloo@cpc1-shep3-0-0-cust1012.8-3.cable.virginmedia.com) |
| 20:00.48 | darrenlooby | I'm trying to originate a call using a call file which auto answers... Most of the stuff I can find talks about AMI, and I'm getting errors when I try to do similar with a call file. Anyone with experience of doing this? |
| 20:01.48 | [TK]D-Fender | jero, What card? Channel? |
| 20:02.12 | [TK]D-Fender | darrenlooby, Yes. No show us the actual call file, failed attempt, etc |
| 20:02.15 | [TK]D-Fender | Now* |
| 20:03.23 | [TK]D-Fender | ~pb |
| 20:03.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:03.25 | [TK]D-Fender | ^^^^ |
| 20:04.41 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 20:06.20 | justdave | idly wonders if Digium will ever provide a RHEL6/CentOS6 repo for asterisk/dahdi, since the packages in EPEL don't seem to be kept up-to-date (don't really expect an answer, just sharing ;) |
| 20:06.45 | justdave | seems strange to be able to get newer Asterisk on rhel5 than I can on rhel6 (without resorting to building it myself) |
| 20:07.31 | pabelanger | justdave, update the RPM and submit it back to digium |
| 20:07.38 | pabelanger | it is open source after all |
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| 20:08.01 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:08.15 | justdave | pabelanger: I did that once, the bugs in the tracker never got touched. despite having patches on them. |
| 20:09.35 | justdave | but there is no rhel6 repo right now. every time I asked if they were going to add one i was told "you can get it from EPEL" (EPEL didn't have it for EL5) |
| 20:09.55 | darrenlooby | [TK]D-Fender, http://pastebin.com/P6yjCcEB |
| 20:10.45 | darrenlooby | [TK]D-Fender, Attempting to play a file in this one |
| 20:10.46 | pabelanger | justdave, then ask for commit access to commit the patches |
| 20:10.48 | Qwell | It's not a matter of not having the spec file updated. It's a matter of getting a system built, building them, maintaining them, etc. |
| 20:10.50 | pabelanger | post them on reviewboard |
| 20:10.50 | [TK]D-Fender | darrenlooby, and where did you invent that option from? "call-info". That isn't a valid parm.... |
| 20:10.57 | pabelanger | get more active in asterisk-dev |
| 20:10.59 | Qwell | updating the specs is trivial |
| 20:11.11 | pabelanger | people will be more then happy to give you access if you work on it |
| 20:11.31 | darrenlooby | Found it here: http://www.voip-info.org/wiki/view/Asterisk+cmd+Page |
| 20:12.09 | [TK]D-Fender | darrthat isn't just something you can shove in a call file |
| 20:12.25 | [TK]D-Fender | darrenlooby, that is ablatant call to a separate dialplan app. |
| 20:12.39 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 20:12.50 | [TK]D-Fender | darrenlooby, If you plan on doing more, you'll have to dial a LOCAL CHANNEL and have dialplan do your dirt-work for you |
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| 20:14.06 | darrenlooby | [TK]D-Fender, the main thing I need to do - is add a call file to trigger a auto answering call, to play a file. I can't touch the dialplan files - because they get overwritten by other processes |
| 20:14.21 | [TK]D-Fender | darrenlooby, You have no choice. |
| 20:14.27 | darrenlooby | [TK]D-Fender, do you think I would be better approaching this using AMI? |
| 20:14.34 | [TK]D-Fender | darrenlooby, No, it's the same thing |
| 20:14.45 | [TK]D-Fender | darrenlooby, just a different trigger |
| 20:14.50 | darrenlooby | Okay, cheers [TK]D-Fender |
| 20:15.14 | [TK]D-Fender | darrenlooby, what "ther processes"? |
| 20:15.18 | [TK]D-Fender | other* |
| 20:18.03 | darrenlooby | [TK]D-Fender, we're using someone else's asterisk built - with GUI. When they add new features etc, it runs and update and overwrites most of the config files |
| 20:18.17 | darrenlooby | [TK]D-Fender, as a result, anything that we put it will get lost |
| 20:18.30 | darrenlooby | [TK]D-Fender, will have to just convince them to put it in |
| 20:19.31 | darrenlooby | [TK]D-Fender, cheers for your help :) |
| 20:23.43 | Katty | hello. |
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| 20:27.15 | [TK]D-Fender | checkout time, BBIAB |
| 20:29.57 | *** join/#asterisk kontinuity (~Adium@122.178.199.29) |
| 20:31.50 | kontinuity | hi guys |
| 20:32.33 | kontinuity | I am trying to simulate a call by placing a call file in the spool, and it does call but whenever the other guy starts speaking the call is dropped automatically |
| 20:34.34 | *** part/#asterisk kontinuity (~Adium@122.178.199.29) |
| 20:34.40 | *** join/#asterisk kontinuity (~Adium@122.178.199.29) |
| 20:35.38 | kontinuity | and I get this log in the console pawn extension (incoming, 2000, 2) exited non-zero on 'SIP/192.168.3.103-00000067' |
| 20:38.34 | beek | kontinuity: Pastebin your call file and the dialplan in the context you're dropping the call into. |
| 20:41.52 | kontinuity | beek: http://pastebin.com/in4sVrCF |
| 20:42.05 | kontinuity | beek: I am using a 3 asterisk setup to simulate my server setup |
| 20:42.18 | Phican | In my senario, sip debug shows the call being set up correctly, however, remote peer appears to come back with congested/busy. when someone has a chance, would anyone mind taking a peak at my debug output to see if Im missing anything? |
| 20:42.43 | kontinuity | beek: we have two sip peers, 1 that gives us inbound calls and another that we forward the calls to |
| 20:44.57 | beek | looks |
| 20:47.20 | beek | kontinuity: Without spending a huge amount of time trying to figure out exactly what you're doing (you didn't put in the call file you used) I'd suggest adding an 'Answer()' as the first line of the context... before the dial. |
| 20:47.34 | kontinuity | Channel: SIP/2000@outbound |
| 20:47.35 | kontinuity | Extension: s |
| 20:47.40 | kontinuity | beek: thats the call file |
| 20:47.53 | kontinuity | beek: thanks, let me give it a shot |
| 20:48.12 | kontinuity | beek: although if I make a call from a softphone the whole setup works |
| 20:48.12 | beek | You have no 'outbound' context in your pastebin. |
| 20:50.40 | beek | Once again, I'm just shooting in the dark here. |
| 20:50.47 | kontinuity | beek: thats the outbound context used in all 3 asterisk servers |
| 20:50.47 | kontinuity | http://pastebin.com/t7U3UZsD |
| 20:51.40 | kontinuity | beek: does call file calls have any limitations as compared to softphone calls? I am guessing I've missed something critical here. Maybe call files can't listen to whatever is being played back |
| 20:51.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.13 | volga629 | message and that what I see http://fpaste.org/NU45/ |
| 20:52.15 | volga629 | <volga629> Is that actual delivery of email ? |
| 20:52.48 | volga629 | http://fpaste.org/NU45/ |
| 20:53.07 | [TK]D-Fender | volga629: Do you SEE the message? |
| 20:53.26 | volga629 | I created test new account and that what I see on mail server |
| 20:53.36 | volga629 | Is that actual delivery of email ? |
| 20:54.01 | volga629 | for Imap |
| 20:54.36 | [TK]D-Fender | mailop - Adding Message: <--- what does this look like to you? |
| 20:55.02 | Phican | http://pastebin.com/1YnEE4rA |
| 20:55.29 | beek | kontinuity: Where is the "s" extension in your dialplan? |
| 20:55.42 | volga629 | Look like delivery, but I am not sure if after should be something else, because it mail folder I don't see anything. |
| 20:56.08 | [TK]D-Fender | PhiCalled SIP/31337@did.voip.les.net |
| 20:56.20 | kontinuity | beek: I am always calling the same _2XXX extension, 2000 in my call file case |
| 20:56.30 | volga629 | that why I ask to see which side need look for the problem |
| 20:56.33 | [TK]D-Fender | Phican: <--- SIP read from UDP:64.34.181.47:5060 ---> SIP/2.0 404 Not Found |
| 20:56.44 | [TK]D-Fender | PhiThey clearly don't like what you dialed |
| 20:57.21 | Phican | I wonder if its not passing correctly, seems to be the same error on multiple uris |
| 20:58.15 | [TK]D-Fender | Phican: Where do you see that 31337 is a valid number to dial there? |
| 20:58.35 | Phican | http://les.net/products/product_ipwholesale.php |
| 20:58.37 | beek | kontinuity: Have you cranked up verbosity and watched the console? |
| 20:58.50 | kontinuity | yes the call comes to the destination asterisk |
| 20:59.02 | Phican | Ive called it before, do you know of some quick URIs I can use? |
| 20:59.04 | beek | And then what? |
| 20:59.07 | Phican | rather, for testing? |
| 20:59.13 | kontinuity | and ends abruptly (probably) with Spawn extension (incoming, 2000, 2) exited non-zero on 'SIP/192.168.3.103-00000067' |
| 20:59.41 | kontinuity | beek: it kind of skips the playback part when the call is from a call file |
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| 21:00.23 | volga629 | http://fpaste.org/dZ3E/ Look on another it add message and delete message ????? |
| 21:02.00 | [TK]D-Fender | Phican: Well so far it looks like they killed that off... |
| 21:02.05 | [TK]D-Fender | Or just broken |
| 21:02.52 | kontinuity | beek: this is interesting, http://pastebin.com/tLEZmmKf |
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| 21:03.28 | kontinuity | beek: the log from the destination asterisk, first one is with call file, second one a softphone, you can see the playback being skipped in the first one |
| 21:03.37 | stimoceiver | so is there a good list of community-recommended SIP trunk providers somewhere? |
| 21:03.53 | stimoceiver | preferrably sorted by POP location |
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| 21:05.38 | beek | kontinuity: I'm looking something up. |
| 21:06.00 | stimoceiver | or maybe one of you asterisk gear heads can relate, your criterion for choosing SIP providers |
| 21:06.25 | stimoceiver | ive noticed some problems when the SIP provider doesnt have a local presence in my metro area - which happens to be Chicago |
| 21:07.21 | Phican | TK - Doesnt dial for you either? |
| 21:10.34 | beek | kontinuity: As I thought. The "person" in a call file is emulated by a context. |
| 21:11.01 | volga629 | I can't believe problem solved it is delete voice mail option on asterisk, I removed delete and right I see message in VoiceMail folder |
| 21:11.12 | kontinuity | beek: so I need to add the context to the call file |
| 21:11.21 | kontinuity | beek: ? and what if its not set? |
| 21:11.40 | beek | kontinuity: Yes, and you need to program the dialplan to act as the other person. At the very least, Answer(). |
| 21:12.05 | beek | Then you get the result that you have seen. The call is placed but the calling side simply drops the call as there is nothing more for it to do. |
| 21:13.19 | kontinuity | beek: bear with me please, so you mean to say that the outgoing context has to answer? How is that done? |
| 21:13.57 | kontinuity | beek: a sample dialplan/context would be much appreciated |
| 21:14.05 | beek | Working on it... hang on. |
| 21:17.24 | [TK]D-Fender | Phican: No, but what you dialed matches what they are asking for, and looks to be formatted right... |
| 21:18.23 | beek | kontinuity: http://pastebin.com/E2eWDNgG |
| 21:18.53 | beek | kontinuity: This is the Perl code which connects via AMI, but same concept. |
| 21:19.32 | beek | Call is placed and the calling side then drops into the 'page-by-phone' context for processing. It just continues to repeat a message until the person being "paged" hangs up. |
| 21:20.25 | [TK]D-Fender | Just picture the VOICEMAIL that would leave behind.... |
| 21:20.43 | kontinuity | beek: so in essence you are waiting endlessly until the other side hangs up |
| 21:21.15 | beek | Yes. The call is placed by AMI, then that open channel is dropped in to page-by-phone, priority 's'. |
| 21:21.36 | beek | The background being played says "Grab a pencil and paper and press 1 to hear the details". |
| 21:22.10 | beek | Once the person does that we jump to exten 1, priority 1, and loop until they hang up. |
| 21:22.22 | beek | I meant to say "page-by-phone, extension 's', priority 1" |
| 21:23.00 | kontinuity | beek: can I just use wait instead? |
| 21:23.07 | kontinuity | why use background |
| 21:23.16 | beek | Do whatever makes you feel good. Really! |
| 21:23.19 | volga629 | What can cause sent both Imap and regular mail as voice mail notification ? |
| 21:23.26 | beek | It is just a dialplan to do whatever you want, so wait would be fine. |
| 21:23.46 | beek | You just need to write a context that acts like the human who is using your softphone. |
| 21:24.07 | beek | And if that person does nothing more than listen, then wait(20) would be sufficient. |
| 21:24.37 | kontinuity | beek: awesome! let me try this out, thanks! |
| 21:24.49 | beek | Bottom line... the call file simply places the call, then expects to drop into dialplan somewhere. |
| 21:24.56 | mjordan | volga629: I don't understand your question. Do you mean why did you receive an e-mail notification that you were left a voicemail, and a message through IMAP that was the actual voicemail left? |
| 21:26.33 | Phican | TK, thank you for your input, I tried a uri path from carrierx and the call went through, seems alot of the URIs Ive been using are not longer in service |
| 21:26.48 | volga629 | yes, I got both to VoiceMail folder and Inbox |
| 21:27.51 | mjordan | volga629: do you have an e-mail address specified for the voicemail user, in addition to their IMAP credentials? |
| 21:29.55 | volga629 | 101 => 1455,SnomDesk,,,vmcontext=realtime-ldap| no |
| 21:30.12 | volga629 | passwd temporary this test machine |
| 21:30.46 | kontinuity | beek: brilliant! it worked, thank you so much |
| 21:30.54 | beek | kontinuity: You're welcome. |
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| 22:02.21 | navaismo | sorry for sound like a bot but: |
| 22:02.21 | navaismo | Hi im trying to add a simple web page to edit the *_customs files so far it works but i cant enable the auth using the freepbx system, any hint please? |
| 22:02.41 | drmessano | You're in the wrong channel |
| 22:04.00 | navaismo | LOL you are right |
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| 22:41.34 | KNERD | jpsharp: http://hiousi.tumblr.com/post/2745684940/a-freepbx-ari-module-to-send-fax-from-the-user-portal |
| 22:41.37 | KNERD | That has the same settings as you showed me...still not functioning |
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| 22:46.24 | vastina | in order to make a test PRI line as CO all I need to do is edit the /etc/asterisk/chan_dahdi.conf and modify signalling to pri_net and have a T1 crossover cable, no? |
| 22:47.04 | newtonr | vastina: pretty much |
| 22:47.31 | newtonr | assuming you have coding, framing, etc setup how you need |
| 22:47.59 | vastina | right, ESF framing, loadzone, etc... |
| 22:48.25 | vastina | bipolar with 8 zero subbing |
| 22:48.28 | vastina | I think I've got it |
| 22:48.36 | newtonr | cool |
| 22:48.51 | vastina | and I can also emulate an E1 european style with 32 channels if I wanted I can see |
| 22:48.54 | vastina | sweet stuff |
| 22:49.19 | newtonr | yup, wide range of signalling and protocol settings |
| 22:49.48 | vastina | any suggestions for load testing dahdi cabinets? sipp can't be what I need for benchmarking would it? |
| 22:50.54 | newtonr | i've never done any load testing with TDM stuff. are you wanting to do calls from SIP -> TDM ? |
| 22:51.23 | newtonr | if you originate from the SIP side, then sure I guess you could do that with sipp |
| 22:51.33 | vastina | yeah pretty much, it's a SIP appliance working with TDM cabinets |
| 22:51.37 | vastina | alright |
| 22:51.55 | vastina | I work for a manufacturer and that's one of my tasks, load testing the older TDM stuff |
| 22:52.20 | newtonr | you would just setup the basic dialplan to get the SIP call passed through out the DAHDI channels, then do all your magic with sipp |
| 22:52.21 | vastina | with an asterisk "appliance" |
| 22:52.26 | newtonr | neat |
| 22:52.43 | vastina | ah, alright, thank you |
| 22:53.16 | vastina | end of my day here, but I'll be asking more tomorrow if needed |
| 22:53.22 | vastina | much appreciated |
| 22:53.29 | newtonr | anytime! later |
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| 23:54.55 | WIMPy | vastina: Don't forget to change to provice timing on that interface in dahdi/system.conf. |