| 00:00.13 | LemensTS | If I have 'T' option in my dial cmd, I cannot pass DTMF to outside IVR's (sip call) |
| 00:00.21 | LemensTS | Any idea why? |
| 00:08.29 | Zelic | One last question, if you may. Where would I be able to get the services for this request and about how much would it cost? |
| 00:08.43 | Zelic | It sounds like it may be a little more complicated than what I know |
| 00:13.47 | *** join/#asterisk engrxyz (~hhghgfhgf@host81-150-217-173.in-addr.btopenworld.com) |
| 00:15.39 | Freeaqingme | Zelic, that depends. think there's plenty of people in here who provide these services. |
| 00:16.01 | Freeaqingme | otherwise you could contact digium they probably do so too |
| 00:16.14 | Freeaqingme | (I would do it for you right now, but I dont know the voicemail module at all) |
| 00:16.15 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 00:17.18 | Zelic | ah I see. Thank you Freeaqingme. I will have to bear it out for a few more days. I appreciate your support and services. |
| 00:17.43 | Zelic | I'll probably drop by tomorrow or the next day if there will be someone who can perform these services. |
| 00:17.45 | Zelic | Thanks! |
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| 01:09.39 | joobie | hey guys.. can anyone recommend a good / cheap ATA device to do 1 phone? |
| 01:09.57 | Thedr | the spa2102 isn't bad |
| 01:10.30 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 01:11.47 | joobie | Thanks |
| 01:11.54 | joobie | any idea what a good voip cordless phone is ?:P |
| 01:13.02 | mateu | Can one reload asterisk too many times or does asterisk manage well a bunch of reload requests that occur around the same time? |
| 01:13.46 | Thedr | Joobie, there aren't that many cordless VoIP phones out there, try looking at the Cisco Iphone |
| 01:13.51 | *** part/#asterisk LedZeplin (jbearer@74.46.1343.static.theplanet.com) |
| 01:13.53 | Thedr | not the apple one ;) |
| 01:14.55 | p3nguin | joobie: I think the most popular ATA is the Linksys PAP2T. |
| 01:15.53 | mateu | is wondering if he needs to queue up reload requests. |
| 01:16.19 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:17.52 | WiretapWork | mateu, why would you queue up kill-my-calls requests? |
| 01:18.17 | mateu | reload = kill-my-calls? I didn't know. |
| 01:19.40 | Thedr | A reload won't kill calls |
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| 01:34.36 | SaiSoma|AtHome | hey guys, connecting * to a cisco via PRI. straight through cable or crossover? |
| 01:40.26 | WiretapWork | SaiSoma|AtHome, that entirely depends on the configuration of your PRI card |
| 01:41.25 | Thedr | Hey guys does anyone know of a web interface to log an agent in and out of a queue? |
| 01:41.29 | SaiSoma|AtHome | rgr. completely vanilla configuration generated by dahdi. Just trying to get a green light. The card is a Sigium TE220 |
| 01:43.29 | WiretapWork | green light on what? |
| 01:44.14 | Freeaqingme | I'm doing a: RECORD /home/kandtranet/domains/domain.nl/var/pbx/voicemails/kndlr/29:wav, but now asterisk complains: No such format 'nl/var/pbx/voicemails/kndlr/29:wav' |
| 01:44.20 | Freeaqingme | any way to escape that dot? |
| 01:44.38 | WiretapWork | you're aware if you need to use a crossover you shouldn't use an ethernet crossover I assume |
| 01:59.38 | atan | joobie, stay away from PAP2 adapters on eBay |
| 01:59.52 | atan | Actually, stay away from all old Vonage equipment. I have had more bad luck than like... errrrg! |
| 02:09.05 | WiretapWork | atan, the pap2t-na on ebay are genuine boards in counterfeit cases in my experience |
| 02:24.40 | ferdna | can i have 5 phones use the same extension? |
| 02:24.59 | Thedr | why would you need that |
| 02:25.28 | WiretapWork | ferdna, use a ring group or queue |
| 02:28.16 | Thedr | you will get a lot of issues if you try to use the same extension on multiple phones |
| 02:29.00 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 02:29.43 | knarfly | how do I clear /var/log/messages? it's filling up my /var folder |
| 02:30.09 | Thedr | you can just delete it |
| 02:30.18 | Thedr | or back it up to another location then delete |
| 02:30.40 | knarfly | should I stop * first and then restart after delettion |
| 02:30.54 | Thedr | shouldn't need to |
| 02:32.06 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 02:40.01 | ferdna | WiretapWork, ok thanks =) |
| 02:47.40 | knarfly | can anyone explain this message: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. I'm running FreeBSD, not linux |
| 02:47.52 | Freeaqingme | checked syslog? |
| 02:48.23 | knarfly | what would I look for in syslog? |
| 02:48.28 | Freeaqingme | last few lines |
| 02:51.13 | *** join/#asterisk brianc (~brian@99-61-68-214.lightspeed.austtx.sbcglobal.net) |
| 02:52.19 | knarfly | looks like it may have been due to the /var folder being full |
| 02:52.43 | *** part/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 02:52.55 | Freeaqingme | he could have hung around a tad longer... |
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| 03:22.43 | tacloban | quit |
| 03:28.42 | SaiSoma|AtHome | Wiretap: thanks for responding. I actually did NOT know that and found the proper pinout and am up and running, thanks! gnite all |
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| 03:46.29 | WiretapWork | the mind does boggle at how often people look at an RJ45 jack and simply say 'it must be ethernet' |
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| 04:13.24 | titter | Quickly, any reason for callerid to show the IP address? Incoming calls are showing the full sip URI |
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| 05:17.22 | Dovid | .j #asterisk-il |
| 05:17.39 | Dovid | titter: That depends on the phone |
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| 05:48.13 | Thedr | Does anyone know a way to use a web page script to call a number in the PBX? I want to be able to open a website or run a batch file that will log a user in and out of a queue |
| 05:48.30 | Wiretap | Thedr, should be possible via AMI |
| 05:48.58 | celthunder | Thedr: gv ? |
| 05:49.47 | kaldemar | Thedr: originate is the keyword. |
| 05:50.03 | Wiretap | that said |
| 05:50.11 | Wiretap | you should be able to log a user in and out without using originate |
| 05:50.47 | Thedr | I just need it to pass something like dial 7777* from user 100 |
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| 06:19.32 | zkn | Hello, question about Queue app... inbound call comes in over IAX2 and is sent to the Queue, but how to play MOH to callers when they are indeed waiting in the queue (agent busy) and hear ringing when the agent is actually ringing ? |
| 06:20.12 | Thedr | thanks guys, I figured something out with php and AMI |
| 06:20.51 | zkn | "R ? stops moh and rings once an agent is ringing (Asterisk Trunk)" |
| 06:20.58 | zkn | when exactly does that apply? |
| 06:22.12 | zkn | seems to not work in my above scenario.. |
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| 06:28.58 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:29.00 | schmidts | good morning |
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| 06:35.13 | Thedr | afternoon |
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| 06:55.16 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 06:55.20 | jacc0 | hi all!! |
| 06:55.22 | jacc0 | good mrning |
| 06:55.44 | Thedr | afternoon |
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| 06:58.14 | jacc0 | How are things @ digium? everything back to normal again? is the dev. svn up already? |
| 06:58.31 | Wiretap | jacc0, why do you ask this twice each day? |
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| 07:04.44 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 07:04.45 | kaushal | hi |
| 07:10.49 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 07:14.19 | kaushal | Please guide me about http://pastebin.ubuntu.com/605598/ |
| 07:16.09 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 07:18.31 | kaldemar | kaushal: how did you install wanpipe? and what card do you have? |
| 07:18.48 | jacc0 | sangoma? |
| 07:18.52 | kaushal | jacc0: yes |
| 07:18.56 | kaushal | kaldemar: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT%28Debian%2FUbuntu%29 |
| 07:18.57 | jacc0 | then you have bad luck |
| 07:19.16 | jacc0 | there is an issue with sangoma and new ubuntu/debian kernal |
| 07:19.20 | jacc0 | they dont compile |
| 07:19.24 | kaushal | ok |
| 07:19.56 | jacc0 | I have mail about this |
| 07:20.00 | jacc0 | I'll share it with you |
| 07:20.04 | jacc0 | 1 momoent |
| 07:20.05 | kaushal | jacc0: sure |
| 07:20.40 | jacc0 | This is an update on our Asterisk 1.8 policy for Sangoma hardware/software using Chan_woomera. |
| 07:20.40 | jacc0 | Sangoma had previously decided to move our Chan_woomera related software to maintenance only mode but with the large number of changes in Asterisk 1.8 we have decided to completely end of life that project. |
| 07:20.40 | jacc0 | All our hardware (PRI, BRI, and analog) is now fully supported in SMGV3. For our BRI line of cards, SMGv3 will also be our only conduit for Asterisk. |
| 07:21.11 | kaldemar | kaushal: wanpipe is not a part of asterisk. and what card do you have? |
| 07:21.24 | jacc0 | and on top of that: SMGv3 doesn't compile with the new ubuntu/debian kernel |
| 07:21.38 | jacc0 | a500 I guess |
| 07:21.42 | jacc0 | from sangoma |
| 07:21.58 | kaushal | jacc0: so any workaround ? |
| 07:22.22 | kaushal | kaldemar: 02:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 07:22.50 | kaushal | SMGV3 ? |
| 07:23.46 | jacc0 | work around is to install centOS and run ubutu within a virtual machine if you realy need ubuntu |
| 07:24.02 | kaushal | ok |
| 07:24.25 | jacc0 | go bug mark and the others from sangoma support |
| 07:24.27 | kaushal | jacc0: is there a description about the issue ? |
| 07:24.30 | jacc0 | at #sangoma |
| 07:24.40 | kaushal | that it doesnot work with the new kernel |
| 07:24.45 | kaushal | of ubuntu |
| 07:26.01 | jacc0 | and debian |
| 07:26.04 | jacc0 | Marc Celsie (marc@sangoma.com) |
| 07:26.12 | jacc0 | go bug him about it |
| 07:26.25 | jacc0 | they will be online on this irc in about 6 houres |
| 07:26.31 | jacc0 | #sangoma |
| 07:26.44 | jacc0 | also support_leo is on this issue |
| 07:26.53 | jacc0 | Sorry for the late reply. Marc informs me that he has been in contact with you this morning on IRC chat. I had emailed the developer yesterday and just received notification from him and he indicated that he will have something available tomorrow. I know you have heard that one before too. I have the most senior engineer here putting pressure on the release as well now |
| 07:27.01 | *** join/#asterisk iulhk (iulhk@117.102.32.140) |
| 07:27.06 | jacc0 | that message was from 1,5 month ago |
| 07:27.25 | jacc0 | still there is no solution |
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| 07:32.08 | *** join/#asterisk e7e5 (~rudenko@77.239.239.27) |
| 07:32.43 | jacc0 | a message months earlier: |
| 07:32.45 | jacc0 | I looked into your ticket again and the issue here is you have asterisk 1.8 installed on your system and our current driver has issues with this. A bug report has been filed and You will get notification once the issue is resolved. Now if you don't hear anything with in 4-5 business days just message me on here and we will go fromthere. |
| 07:33.24 | jacc0 | so don't count on a quick-fix :P |
| 07:33.52 | Wiretap | thats a pain |
| 07:34.01 | Wiretap | since I'm plannign to build a new PBX for $work |
| 07:34.10 | Wiretap | and it would be reliant on a Sangoma A500 |
| 07:34.15 | Wiretap | and running debian |
| 07:34.28 | jacc0 | ;) don't use the squeeze versiuo |
| 07:34.32 | jacc0 | *version |
| 07:34.39 | jacc0 | or don't use 1.8 |
| 07:34.52 | jacc0 | or wait |
| 07:34.58 | Wiretap | I plan to deploy Cisco Unified phones, so 1.8 is kinda important, and yes |
| 07:35.00 | Wiretap | I will be waiting |
| 07:35.08 | Wiretap | its not in the budget for a minimum of 6mo |
| 07:35.12 | jacc0 | why don't we all start complaining to marc:(marc@sangoma.com) |
| 07:35.43 | jacc0 | it will make them give this a higher prio. |
| 07:36.03 | jacc0 | I'll mail him to update me about the issue |
| 07:39.16 | jacc0 | I'm going to wrap my a200 and a500 in a box and send it back to them |
| 07:39.48 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 07:40.09 | Wiretap | these things aren't cheap either |
| 07:40.39 | Wiretap | I priced up what it would cost me to replace the A500 I have here (two line cards and an echo canceller), NZ$1250 |
| 07:42.26 | jacc0 | @wiretap: you would realy help me if you put some pressure on sangoma support |
| 07:42.57 | jacc0 | marc@sangoma.com |
| 07:43.01 | Wiretap | jacc0, I will rustle up one of my spare boxes in the coming days once I have my own * install working properly |
| 07:43.10 | Wiretap | and fire off a complaint when I can't get it to compile :P |
| 07:43.22 | jacc0 | ty ;) |
| 07:43.34 | jacc0 | that would realy help me |
| 07:43.57 | jacc0 | installing asterisk is simple: apt-get install asterisk |
| 07:44.05 | Wiretap | haha |
| 07:44.12 | Wiretap | until you want the cisco unified patch |
| 07:44.42 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:4ddc:dc75:641e:385) |
| 07:44.50 | jacc0 | ;) |
| 07:45.53 | jacc0 | you can also add: |
| 07:45.55 | jacc0 | deb http://packages.asterisk.org/deb squeeze main |
| 07:45.55 | jacc0 | deb-src http://packages.asterisk.org/deb squeeze main |
| 07:46.12 | jacc0 | to /etc/apt/sources.list |
| 07:46.33 | Wiretap | jacc0, I actually have a functioning * PBX right now |
| 07:46.34 | jacc0 | after that apt-get install asterisk will install 1.8 |
| 07:46.41 | jacc0 | okay |
| 07:46.57 | Wiretap | but it doesn't have cisco unified phone support |
| 07:47.07 | Wiretap | so I'm building a custom asterisk that includes it |
| 07:47.14 | Wiretap | keep ending up with voicemail not working though |
| 07:47.19 | Wiretap | trying to work out where its going wrong |
| 07:48.21 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
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| 07:50.03 | Wiretap | right, so it was DEFINITELY one of my config options causing the problem |
| 07:53.49 | ChannelZ | it usually is :P |
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| 08:04.55 | kaushal | Can someone please explain me about the difference of features available in 1.8 and 1.6 ? |
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| 08:15.27 | kaldemar | kaushal: http://svn.digium.com/svn/asterisk/tags/1.8.4/UPGRADE.txt |
| 08:16.10 | kaldemar | kaushal: http://www.asterisk.org/node/51444 |
| 08:19.18 | Dovid | I keep mixing up types. If I have incoming and outgoing with 1 company. Asterisk seems to send a 401 to my ITSP. How do i break them up in sip.conf? i know one needs to be peer and the other ? |
| 08:19.50 | *** part/#asterisk zapotek6 (~zap@mail.comelit.it) |
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| 08:30.33 | kaldemar | Dovid: if your ITSP does not authenticate to you, use insecure=port,invite for the mathcing peer. |
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| 08:57.51 | b14ck | Yo, I've got a funky problem, was wondering if any dialplan gurus are around. |
| 08:58.02 | b14ck | I'm doing: AGI(/some/script,arg1,arg2) |
| 08:58.04 | b14ck | the issue is: |
| 08:58.33 | b14ck | arg is a string that looks like: '{"a": "1", "b": "2"}' |
| 08:58.52 | b14ck | And what's happening is that the comma there is making Asterisk send 3 command line arguments to my agi script instead of 2 |
| 08:58.58 | b14ck | so saterisk is running: |
| 08:59.25 | b14ck | <PROTECTED> |
| 08:59.31 | b14ck | which breaks my stuff |
| 08:59.41 | b14ck | Is there a way to escape the comma character inside of the AGI command? |
| 08:59.42 | kaldemar | escape the comma |
| 08:59.47 | b14ck | How? |
| 08:59.52 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 08:59.54 | b14ck | I've tried \ |
| 09:00.03 | kaldemar | try more \'s. |
| 09:00.12 | b14ck | seriously? :o |
| 09:00.15 | kaldemar | yes. |
| 09:00.18 | b14ck | ok |
| 09:00.21 | b14ck | let me see |
| 09:00.46 | b14ck | trying with 2 \\ |
| 09:00.59 | kaldemar | i don't remember which func it was, but one needed four (yes, 4) \'s to escape a comma. those things may happen. |
| 09:02.51 | b14ck | Oh wow. |
| 09:03.28 | b14ck | well, 2 & 3 dont work |
| 09:03.31 | b14ck | hopefully 4 does :) |
| 09:03.52 | Wiretap | lol 4! |
| 09:04.17 | b14ck | nope :( |
| 09:04.30 | b14ck | Executing [s@partyline_ban_check:1] AGI("SIP/PA-CO-CG01-0000155c", "/nfs/asterisk/main/programs/partyline/tools/api_proxy.py,'/caller/is_banned_partyline/','{"partyline": "610XXXXXXX"\\\\, "caller": "818XXXXXXX"}'") in new stack |
| 09:04.41 | b14ck | still no good, lol =/ |
| 09:04.51 | b14ck | is there any other way to escape a comma? hrm |
| 09:05.41 | kaldemar | is the 3 arguments vs 2 a problem? you can always combine them in the AGI script if the escaping is a problem. |
| 09:05.54 | b14ck | ya, that's the problem right now |
| 09:06.05 | b14ck | I'm not sure how to combine them is the problem. |
| 09:06.11 | b14ck | My string MUST have that comma in it |
| 09:06.16 | b14ck | Well.. |
| 09:06.19 | b14ck | I guess I could do that. |
| 09:06.24 | b14ck | what the fuck, let me give it a go |
| 09:06.28 | b14ck | why not! |
| 09:09.22 | jacc0 | what about base64 encoding it? |
| 09:09.28 | jacc0 | en decoding it later on |
| 09:09.53 | jacc0 | ${BASE64_ENCODE(Some Interesting Text to encode)} |
| 09:11.46 | b14ck | that's a pretty good idea |
| 09:11.53 | b14ck | tries |
| 09:11.57 | jacc0 | is full of good ideas |
| 09:12.23 | jacc0 | it takes care of all other strange chars to |
| 09:12.52 | jacc0 | like ) maybe? |
| 09:13.30 | jacc0 | or ",: and } |
| 09:15.04 | b14ck | im trying that now |
| 09:15.07 | b14ck | i think that's an awesome way to do it |
| 09:15.10 | b14ck | hopefully it works ok |
| 09:15.10 | b14ck | :o |
| 09:16.04 | *** join/#asterisk wonderworld (~ww@port-92-201-95-80.dynamic.qsc.de) |
| 09:17.05 | jacc0 | be aware tho |
| 09:17.27 | jacc0 | the length of the base64 encoded string is not allowed to be longer the 4096 bytes or asterisk will fail |
| 09:17.31 | jacc0 | i guess |
| 09:17.36 | b14ck | :o |
| 09:17.42 | b14ck | its pretty short |
| 09:17.45 | jacc0 | it is |
| 09:21.05 | jacc0 | dialplan rows are not allowed to be longer then 4096 en asterisk will past the base64 encoded sting in place |
| 09:21.33 | jacc0 | that's what I believe is causing it |
| 09:22.47 | jacc0 | I updated a bug report: https://issues.asterisk.org/view.php?id=19203 |
| 09:23.12 | b14ck | hey jacc0 that works GREAT |
| 09:23.17 | b14ck | thanks for the suggestion |
| 09:23.29 | b14ck | this has been a crazy headache |
| 09:23.45 | jacc0 | ;) |
| 09:31.10 | *** join/#asterisk BarthezZ (~bart@ip51cd391a.speed.planet.nl) |
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| 09:42.22 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:42.40 | eject_ck | How can I wipe stats what I see during (fax show stats) |
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| 10:30.03 | *** join/#asterisk elzid (~IceChat7@du-012-0026.freeuk.com) |
| 10:30.37 | arunas | Hi, i have a problem with chan_323. instalation vent succesfully |
| 10:30.56 | arunas | everything seems fine when i get inbound calls |
| 10:31.11 | arunas | but when trying to do outbound |
| 10:31.24 | elzid | hi - any idea why this wouldnt work: exten => s,n,GotoIf($[ $[ ${a} > 0 ] & $[ ${a} <= 80 ] ]?:TWO) |
| 10:31.48 | arunas | after dial cmd, looks like asterisk is thinking for a minute(!) |
| 10:32.03 | arunas | and only then sends something to h323 trunk |
| 10:32.08 | arunas | any ideas? |
| 10:32.51 | arunas | shoulnd it be && insted of &? |
| 10:33.20 | elzid | I think not - asterisk uses | and & for logical OR and AND |
| 10:33.26 | jacc0 | @elzid: if ${a} is empty the syntax will fail |
| 10:33.46 | jacc0 | try: |
| 10:33.52 | jacc0 | exten => s,n,GotoIf($[ $[ 0${a} > 0 ] & $[ 0${a} <= 80 ] ]?:TWO) |
| 10:33.55 | elzid | but i@jacc0 - ${a} is never empty |
| 10:34.27 | elzid | @jacc0 - what is 0 prefix for? |
| 10:34.48 | jacc0 | if ${a} is empty syntax will fail |
| 10:34.58 | jacc0 | if you put a 0 in front it doesn't |
| 10:35.15 | elzid | i see - i will try it now - thanks |
| 10:35.46 | elzid | currently it always evaluates true - even if ${a} is 100 |
| 10:36.17 | jacc0 | do you have loging of that? |
| 10:36.45 | jacc0 | asterisk CLI output? |
| 10:37.07 | jacc0 | and can you add noop(${a}) right above the line that fails |
| 10:37.26 | elzid | ok |
| 10:37.28 | elzid | will do |
| 10:37.30 | elzid | brb |
| 10:38.20 | jacc0 | I'm going for a cigaret, brb |
| 10:41.28 | kaldemar | elzid: it does work. what do you expect it to do? |
| 10:41.52 | jacc0 | back |
| 10:49.23 | elzid | playback an audio file |
| 10:50.37 | elzid | @jacc0 - really weirdly my syntax is working now! ${a} is certainly being populated - that's probably why - many thanks for your help :) |
| 10:50.52 | kaldemar | that only goes to n(TWO) if a is not empty is not greater than 0 and less than or equal to 80. |
| 10:52.03 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 10:53.49 | elzid | @kaldemar - yes - thanks - for some reason, the logic was not working before, and it may have been because ${a} was not being set, so it always equated to true |
| 10:54.42 | Dovid | anyone here that worked on Asterisk Fax T.38 ? |
| 10:56.07 | WiretapMac | jacc0: mzb reckons freedoh might get you around your Sangoma issues :P |
| 10:56.58 | mzb | let me double check |
| 10:58.12 | mzb | which version are you trying to install? |
| 10:58.58 | mzb | and what flags (if any) are you using to install with? |
| 10:59.20 | mzb | I'm almost certain freedoh will work for you |
| 10:59.44 | mzb | not certain enough to make a bet, but I reckon the odds are in freedoh's favour ;) |
| 10:59.59 | mzb | only "latest" works iirc |
| 11:00.10 | mzb | "stable" won't work |
| 11:11.01 | jacc0 | I've been trying latest a couple of months ago |
| 11:11.26 | jacc0 | marc and leo would update me if there would be a new version that addresses the issue |
| 11:11.38 | jacc0 | I'm bugging them evey day |
| 11:12.16 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 11:12.22 | hetii | hello :) |
| 11:12.31 | jacc0 | hi hetii |
| 11:13.42 | hetii | I had a trouble with one of my trunk, when i call to my did he is passed by inbound route to my sip extenision. The problem is because i got this: |
| 11:13.44 | hetii | <PROTECTED> |
| 11:13.44 | hetii | <PROTECTED> |
| 11:13.55 | jacc0 | @mzb: have you tested it with debian squeeze icw asterisk 1.8? |
| 11:14.35 | mzb | well ... freedoh installs latest asterisk (and in fact now uses asterisk 1.8 svn branch) |
| 11:14.56 | mzb | but the issue for me is that I don't actually have sangoma hardware |
| 11:15.07 | WiretapMac | jacc0: build yourself a dev box and give it a shot, it can't hurt |
| 11:15.13 | mzb | so I've had to rely on ad-hoc connections to ppl with problems |
| 11:15.34 | jacc0 | okay, the issue is with the sangoma software: they nolonger support chan_woomera |
| 11:15.38 | mzb | once it's fixed I don't really get the opportunity to go back and play ;) |
| 11:15.55 | jacc0 | you should use SMGv3 en register it to asterisk as a normal sip client |
| 11:15.57 | mzb | well, let me put it this way |
| 11:15.58 | *** join/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 11:16.08 | WiretapMac | jacc0: trust me on this one bro... stick the card in a spare box and have a go |
| 11:16.18 | WiretapMac | if mzb says it will work, he's probably right |
| 11:16.28 | epart | wer can find a downloadable PDF of Asteriskâ¢: The Definitive Guide 3rd Ed. \ |
| 11:16.30 | mzb | if you try freedoh out, and find that it doesn't work, I will fix it if you allow me access to the box |
| 11:16.52 | mzb | s/do my best to fix it |
| 11:16.53 | WiretapMac | ~thebook |
| 11:16.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 11:16.53 | jacc0 | @epart: you should read theBook |
| 11:17.09 | jacc0 | ;) |
| 11:17.11 | mzb | jacc0: sound fair? |
| 11:17.16 | WiretapMac | right, sleeptime |
| 11:17.43 | jacc0 | very fair, but I don't have access to the box @ this moment |
| 11:17.51 | jacc0 | it's at my home with no power connected |
| 11:18.25 | mzb | ok, pm me if/when you need me ... more than happy to help out (and fix something in freedoh!) |
| 11:18.40 | epart | jacc0: yes I will, is there anylink were I can download a PDF file.. of it.. I would like to read it on my computer if im offline |
| 11:18.54 | mzb | nice to hear the "home" reference .. pls read the license on freedoh ;) |
| 11:19.19 | mzb | you can get the links anytime (they redirect to the real sites) in #freepbx by typing |
| 11:19.26 | mzb | ? freedoh |
| 11:19.27 | mzb | and |
| 11:19.31 | mzb | ? googlevoice |
| 11:19.37 | epart | #vicidial |
| 11:19.42 | mzb | but bottomline they are: |
| 11:19.54 | mzb | http://freedoh.net |
| 11:20.03 | mzb | webpages are HORRIBLY basic |
| 11:20.20 | mzb | freedoh page is comprehensive (ish) |
| 11:20.29 | mzb | but none are "pretty" ;) |
| 11:20.31 | epart | anyone here knows were to download free the Asteriskâ¢: The Definitive Guide 3rd ed. |
| 11:20.43 | epart | PDF file |
| 11:21.02 | jacc0 | ask leifmadsen; I believe he wrote the bigger part of it |
| 11:21.06 | mzb | have a play. More than happy to add/fix/play to improve freedoh's abilities |
| 11:21.48 | mzb | ps: freedoh is a "closed source" *cough* bash script ... I hope you appreciate the humor ;) |
| 11:22.28 | epart | pls help free dowlnload of PDF Asteriskâ¢: The Definitive Guide |
| 11:23.00 | mzb | jfgi ? |
| 11:23.13 | mzb | or is it purchase only? |
| 11:23.59 | mzb | hehehehe |
| 11:24.09 | mzb | mzb-> ~google PDF Asterisk: The Definitive Guide |
| 11:24.09 | mzb | <mzb-> [Google] http://astbook.asteriskdocs.org/ [Asterisk: The Future of Telephony] |
| 11:24.38 | epart | cant find any free download link in google.. |
| 11:24.40 | jacc0 | @mzb: Sangoma told me : we have decided to completely end of life that project |
| 11:24.48 | jacc0 | are you going to finish there work? |
| 11:24.58 | mzb | err |
| 11:25.01 | mzb | gawd |
| 11:25.04 | jacc0 | :p |
| 11:25.18 | mzb | surely their driver still supports the card? |
| 11:25.32 | jacc0 | jep, but des not support 1.8 |
| 11:25.36 | mzb | *or* if you know a version of the driver ... |
| 11:25.37 | mzb | oh |
| 11:25.42 | mzb | well |
| 11:25.58 | mzb | here's the only consolation I can offer |
| 11:26.08 | jacc0 | :) |
| 11:26.10 | mzb | there are three modes of installation in freedoh |
| 11:26.23 | mzb | EASY, INTERMEDIATE and ADVANCED |
| 11:26.50 | epart | Hellow |
| 11:27.10 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:27.17 | mzb | although I've pretty much concentrated on EASY of late, it's possible to specify any version of DAHDI/SANGOMA, Asterisk and FreePBX |
| 11:27.19 | mzb | BUT |
| 11:27.31 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 11:27.31 | mzb | I just haven't supported Asterisk 1.6 yet |
| 11:27.34 | epart | pls have a kind heart to give me a free link on the new Asteriskâ¢: The Definitive Guide 3rd Ed.... free download PDF |
| 11:27.48 | mzb | epart: ^^ pls check the link I pasted |
| 11:27.59 | mzb | <mzb> <mzb-> [Google] http://astbook.asteriskdocs.org/ [Asterisk: The Future of Telephony] |
| 11:28.11 | kaldemar | epart: it's not free. you get the PDF when you buy the paper book or the e-book. |
| 11:28.17 | mzb | oh ... sorry it's only 2nd edition |
| 11:28.46 | mzb | re-reads and pulls his head in |
| 11:29.15 | jacc0 | @mzb: does it support asterisk 1.8? |
| 11:29.51 | mzb | freedoh supports 1.8, yes |
| 11:30.03 | mzb | after a couple of minor changes ;) |
| 11:30.05 | epart | Kaldemar: really? :( |
| 11:30.46 | jacc0 | @sangoma told me I needed to instal SMGv3 because chan_woomera is not support in 1.8 end will never be |
| 11:31.12 | jacc0 | SMGv3 doesn't compile on the latest debian/ubuntu kernel |
| 11:31.39 | jacc0 | how does freedoh fix that? |
| 11:31.56 | mzb | well, as I say I've got no sangoma hardware but I'm pretty sure if their driver supports your card that freedoh will build, install + configure it |
| 11:32.14 | Dovid | hi all. when does AsteriskFax decide to send the re-invite for T.38 ? |
| 11:32.16 | jacc0 | okay, will check it later on |
| 11:32.17 | mzb | *if* you can show me a *bug* in freedoh I'll do whatever I can to fix it |
| 11:32.31 | jacc0 | :) |
| 11:32.43 | jacc0 | I'll will surely get back to you |
| 11:32.45 | jacc0 | ;) |
| 11:33.04 | jacc0 | and so will sangoma support and dev. team |
| 11:33.06 | jacc0 | :p |
| 11:33.13 | mzb | if you have a look at freedoh you'll probably realise that I've got a bit of imagination ... so there's a chance that I might win ;)) |
| 11:33.29 | mzb | we'll see |
| 11:33.36 | mzb | let me know if you need me |
| 11:33.41 | Dovid | anyone here know T.38 ? |
| 11:34.42 | kaldemar | epart: really. |
| 11:34.53 | mzb | time to watch a TV show with $WOMAN, thanks for your interest jacc0, and thanks for you recommendation Wiretap ... talk to you both later |
| 11:38.46 | jacc0 | bye |
| 11:38.49 | jacc0 | ttyl |
| 11:39.44 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 11:41.03 | jacc0 | @epart: go to download.com and search for html2pdf |
| 11:41.09 | jacc0 | @epart:good luck |
| 11:41.22 | *** join/#asterisk gjb83 (6de083cf@gateway/web/freenode/ip.109.224.131.207) |
| 11:42.39 | jacc0 | "Generate PDF from any webpage instantly with a single single click" |
| 11:45.33 | gjb83 | Hi all. I've got 3 VoIP phones behind a NAT gateway, each with unique local SIP ports and RTP ranges, which are port forwarded. Two of the phones are connecting to Asterisk with the same (conflicting) port number of 55970; the third phone is using a different port number of 55967. Any pointers on how I can identify the cause of this conflict please? :) |
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| 12:04.08 | m4xx | jacc0 around? |
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| 12:37.17 | jacc0 | yeah |
| 12:37.31 | jacc0 | but I'm going for a cigaret first, brb |
| 12:37.43 | m4xx | k |
| 12:38.22 | arunas | i think resolved my problem if someone is insterested, i should have set UserByAlias=yes, because i dont have dns |
| 12:41.36 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 12:42.30 | jacc0 | back |
| 12:43.23 | jacc0 | @arunas: what was your problem? asterisk querying DNS every once in a while? |
| 12:43.45 | jacc0 | @arunas:where did you set UserByAlias=yes? |
| 12:46.36 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:46.50 | gjb83 | Hi all. I've got 3 VoIP phones behind a NAT gateway, each with unique local SIP ports and RTP ranges, which are port forwarded. Two of the phones are connecting to Asterisk with the same (conflicting) port number of 55970; the third phone is using a different port number of 55967. Any pointers on how I can identify the cause of this conflict please? :) |
| 12:47.49 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 12:49.17 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:49.18 | jacc0 | @gjb83: did you try to do it some other way; like just registering and no portforwarding and turning on qualify=30 to keep open the route? |
| 12:51.03 | *** join/#asterisk marlowe_ (~marlowe@72.44.190.250) |
| 12:53.29 | m4xx | I'm still trying to figure out why my call is being terminated locally when i SendDTMF(*) from my dialplan. I've reset the binding in features.conf and in agents.conf i've set to endcall to no |
| 12:54.29 | jacc0 | 1.8.3.3 it was? |
| 12:54.35 | m4xx | not sure if i've missed anything but doesn't the feature need to be enabled with the h/H flags when using dial? |
| 12:54.36 | m4xx | yes |
| 12:54.42 | jacc0 | did you try 1.8.4-RC3 |
| 12:54.43 | jacc0 | ? |
| 12:55.23 | m4xx | no sir, i'm installing from ports on freebsd, i had troubles compiling from csv |
| 12:55.31 | m4xx | *cvs |
| 12:56.05 | m4xx | i will attempt it again |
| 12:56.14 | gjb83 | jacc0: unfortunately I need port forwarding as there is only one public IP available. Is it the NAT router or Asterisk that negotiates the port to use? |
| 12:58.36 | jacc0 | @m4xx: when turing on debug can you see what line of code in what file is causing the hangup? then comment the line in source and recomiple? |
| 12:59.27 | m4xx | "sip set debug on" ? |
| 12:59.53 | jacc0 | @gjb83: I've 3 phones behind not inn another subnet here in the office; no potforwarding needed |
| 13:00.01 | m4xx | nope |
| 13:00.02 | jacc0 | *behind nat |
| 13:00.15 | jacc0 | core set debug 9 |
| 13:00.20 | jacc0 | + sip set debug on |
| 13:00.28 | gjb83 | jacc0: are you using a proxy or something? |
| 13:00.34 | jacc0 | nope |
| 13:01.04 | gjb83 | jacc0: is your server behind the same NAT? |
| 13:01.05 | jacc0 | just let them register en turn up the qualify ferq. to keep open the routers |
| 13:01.23 | jacc0 | my server is not behind nat; my phone are |
| 13:01.28 | jacc0 | *phones |
| 13:02.10 | gjb83 | my understanding was that phones either had to use a SIP proxy, or be configured with different local SIP/RTP ports |
| 13:02.53 | jacc0 | _Support_Marc_ is online @ #sangoma :) |
| 13:03.34 | jacc0 | kaushal and wiretap ; jump on him :p |
| 13:07.49 | m4xx | jacc0, i dont see any line numbers or function or line numbers |
| 13:08.04 | m4xx | * line numbers or function names |
| 13:10.50 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:13.49 | jacc0 | "support dialing of the featuremap disconnect code while performing an attended tranfer " |
| 13:13.57 | jacc0 | i found that in /main/features.c |
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| 13:16.28 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:17.20 | jacc0 | you could add some verbose info to the source to check if that is the code that is bugging you |
| 13:20.29 | Faustov | is there a dialstatus specific for missing extension only? I got chanunavail catching it in my phone macro right now, I was wondering if there's something more precise perhaps |
| 13:21.26 | kaldemar | ~sipnat |
| 13:21.26 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 13:22.47 | kaldemar | gjb83: no portforwarding needed. just tell asterisk that the phones are behind a NAT with nat=yes and use qualify parameter for NAT keepalive. for forwarding should not be needed in the NAT of the phone end. |
| 13:22.57 | jacc0 | @m4xx: try adding some debug info in source to see what like of code is causing the hangup: ast_verb(4, "we are now at point X"); |
| 13:23.15 | jacc0 | *line of code |
| 13:23.37 | jacc0 | I would start with /main/features.c |
| 13:23.43 | kaldemar | Faustov: extension as in a dialplan extension? |
| 13:24.48 | m4xx | i'm trying |
| 13:24.52 | Faustov | kaldemar: yeah, like someone dials something that is not specified anywhere - with the default macro-phone, he falls into CHANUNAVAIL dialstatus, but he'd also get that status if the extension was there but there was a network connectivity issue as well |
| 13:24.57 | jacc0 | :) |
| 13:25.10 | Faustov | so I was wondering if perhaps there wasn't something more precise, like EXTEN_NOT_FOUND :P |
| 13:25.16 | m4xx | i just set disconnect_code = '****************'; //builtin_features[x].exten |
| 13:25.22 | m4xx | to see if that will make a difference |
| 13:25.23 | jacc0 | @mx44: search for "disconnect" |
| 13:26.01 | gjb83 | kaldemar: many thanks, I'll have a read through those links :) |
| 13:28.37 | kaldemar | Faustov: i don't know what your macro-phone is, but function DIALPLAN_EXISTS can be used to test whether an extension exists in the dialplan. |
| 13:29.03 | kaldemar | Faustov: other than that, cause codes are helpful. |
| 13:31.15 | *** part/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 13:31.32 | jacc0 | @m4xx: I guess that will fix your issue :P |
| 13:31.43 | m4xx | building now |
| 13:32.37 | m4xx | if this works i'm sending a hooker to your house |
| 13:33.35 | jacc0 | :) lucky you that prostetution is leagal in my country |
| 13:33.44 | m4xx | hahaha |
| 13:34.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:34.21 | m4xx | hmm |
| 13:34.29 | m4xx | features show shows "*" as still being the default |
| 13:35.25 | m4xx | ok so that didn't work =[ |
| 13:36.34 | m4xx | i see this ast_verb(4, "User hit '%s' to disconnect call.\n", code); |
| 13:36.38 | m4xx | but i never see that in logs |
| 13:39.14 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:39.17 | m4xx | im going to try commenting out { AST_FEATURE_DISCONNECT, "Disconnect Call", "disconnect", "*", "*", builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF, "" }, |
| 13:41.42 | jacc0 | :P |
| 13:41.54 | jacc0 | don't you love open source |
| 13:42.23 | m4xx | would be a whole lot easier if i used linux =[ |
| 13:42.30 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 13:43.15 | kaii | hm .. whats the english term for "switching between a held and a connected party"? in german we have the term "makeln" ("broker"). is there an english term for that use case? experts, please help |
| 13:44.23 | kaii | i think "broker" is a financial term. |
| 13:44.27 | leifmadsen | I'm not sure there really is a term... I just say "caller" |
| 13:44.50 | leifmadsen | ya Broker is a more financial term for someone who sells on behalf of someone else |
| 13:46.08 | kaii | hm.. was just wondering, if there really is no term for that use case |
| 13:48.44 | m4xx | same result |
| 13:50.49 | jacc0 | :S |
| 13:51.10 | m4xx | im about to jump off a bridge =[ |
| 13:52.49 | jacc0 | damn, no hooker |
| 13:52.55 | m4xx | lol |
| 13:53.18 | m4xx | that successfully removed that feature |
| 13:53.23 | jacc0 | I guess I'll have to do with my girlfriend |
| 13:53.29 | m4xx | it's no longer shown in "features show" |
| 13:53.52 | m4xx | unless that's just the part that rints em out |
| 13:53.59 | m4xx | *prints |
| 13:54.11 | jacc0 | could be :P |
| 13:54.45 | m4xx | but if it was a features issue |
| 13:54.49 | m4xx | shouldnt i see this ast_verb(4, "User hit '%s' to disconnect call.\n", code); |
| 13:54.51 | m4xx | somewhere? |
| 13:55.33 | jacc0 | I don't realy know but I exspect it to showup when : core set debug 4 |
| 13:55.37 | jacc0 | or higher |
| 13:55.48 | *** join/#asterisk Janos (~Janos@186.32.53.52) |
| 13:56.49 | m4xx | i've got debug level set to 5 |
| 13:56.57 | m4xx | and haven't received that message |
| 13:59.01 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:59.43 | arunas | jacc0: thanks, yeah, asterisk was queueing dns every time i tried to call. anyways changing UserByAlias to yes helped. |
| 14:00.31 | *** join/#asterisk cxk287 (~zorp75ck@client-75-102-91-32.mobility-up.psu.edu) |
| 14:02.33 | Janos | hey there, i have the following macro http://pastebin.com/t2VGRFQA, the idea with this macro is to return the call to the operator "2000" if it's not answered in 20 seconds, it used to work, but at some point it stopped and i have no idea what changed |
| 14:03.58 | arunas | i was wondering has anyone tried using lua insted of extensions or ael in production? has anyone succeed doing this? |
| 14:04.28 | Janos | this is my relevant verbose output in the asterisk console http://pastebin.com/xPirH5Y6, the only thing i find interesting here is this <ZIMBIE> which i have no idea what it means, now question is, is the dial application suppose to stop the macro execution after the timeout, or should it simply continue with the next line ? |
| 14:04.31 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
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| 14:06.48 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 14:07.06 | asilva | Does anyone know how to set the video call resolution? |
| 14:07.34 | *** join/#asterisk cherzberg (~chatzilla@91-66-40-188-dynip.superkabel.de) |
| 14:08.16 | cherzberg | Hi folks |
| 14:09.03 | m4xx | jacc0, does any of this mean anything to you? http://paste2.org/p/1408530 |
| 14:09.47 | *** join/#asterisk sereal (~jjrh@2001:4830:16ca:1:21a:6bff:fe6a:3cd2) |
| 14:11.41 | sereal | Hello, I'm trying to setup outbound calling and i'm getting a busy signal when I try to call - right after a reload I can make a call, but after less than a minute I get the busy signal. This provider uses a outboundproxy. |
| 14:11.50 | sereal | Any idea what the issue might be, is it on my end or theirs? |
| 14:11.53 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 14:11.59 | sereal | I have outbound calls working with other providers. |
| 14:12.01 | m4xx | do you have hangup() at the end of your dial plan? |
| 14:12.23 | sereal | That shouldn't be a issue with dialing out. |
| 14:14.10 | asterisk-learner | Hello,I asked this question here before but just wanted to have another opinion, does WaitForSilence() wait for total silence or continuous silence ? |
| 14:14.19 | Janos | so, any idea what this ZOMBIE means and if dial should continue with next line after timeout ? |
| 14:15.06 | jaytee | a ZOMBIE is either a dead channel that was part of a call setup or a person waiting to eat your brains |
| 14:15.39 | jacc0 | @sereal: i was reading this in sourcecode: |
| 14:15.41 | jacc0 | * For chan_dahdi, the channel is crippled until the hangup is |
| 14:15.41 | jacc0 | <PROTECTED> |
| 14:15.41 | jacc0 | <PROTECTED> |
| 14:15.41 | jacc0 | <PROTECTED> |
| 14:16.05 | cherzberg | I have an asterisk 1.6.2.18 running in ubuntu 10.10 LTS. Some times asterisk stop working. The core is running but without any connection to sip phones or trunks. After a restart asterisk works again. Any ideas? |
| 14:16.46 | sereal | jacc0 for sake of ruling out this issue I added a Hangup after, and it didn't do anything different. |
| 14:16.55 | jacc0 | @cherzberg: is it a realtime configuration using mySQL? |
| 14:17.05 | sereal | And for the record I have my other SIP out's with out a Hangup |
| 14:17.06 | cherzberg | yes it is |
| 14:17.16 | jacc0 | than I might have a clue |
| 14:17.39 | cherzberg | sounds good to me |
| 14:18.05 | jacc0 | one moment; I'll look it up |
| 14:19.17 | sereal | Requiring Hangup would indicate that I should have to try and dial out once, which isn't the case, if I'm not very fast trying to dial out it just doesn't work. |
| 14:19.28 | jacc0 | I'm not sure it is an issue in 1.67 but read this: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg252971.html |
| 14:19.50 | jacc0 | *1.6 |
| 14:20.36 | m4xx | hmm, setting dtmf mode to inband and the call no longer dies |
| 14:20.37 | jacc0 | in my.cnf I added this under [mysqld]: |
| 14:20.43 | jacc0 | wait_timeout=864000 |
| 14:20.43 | jacc0 | interactive_timeout=864000 |
| 14:22.37 | cherzberg | Sounds like our problem. With 1.8.3.3 asterisk is crashing every 10 minutes. With 1.6 just 2 to 4 times a day. |
| 14:22.48 | *** join/#asterisk thegoat (~thegoat@c-71-224-170-221.hsd1.pa.comcast.net) |
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| 14:24.34 | nite613 | Hi guys. I've got an easy question today. By default is there a maximum number of channels on a SIP trunk? |
| 14:24.45 | leifmadsen | no |
| 14:24.46 | cherzberg | our installation is with 1 E1 line and 14 Agents in 3 queus. Asterisk has to handle 2000 to 2500 calls a day |
| 14:24.59 | nite613 | leifmadsen: that was to me? |
| 14:25.02 | leifmadsen | nite613: asterisk does not limit you to the number of channels |
| 14:25.13 | nite613 | Thanks, as always! |
| 14:25.18 | leifmadsen | only CPU and memory limits will .... limit you |
| 14:25.32 | Janos | jaytee, do you think this channel dying could be causing my macro to terminate ? |
| 14:25.59 | Janos | jaytee, this is my macro by the way, http://pastebin.com/t2VGRFQA |
| 14:27.37 | jacc0 | @nite613: if using sip you also have a limitation of ip ports |
| 14:27.40 | leifmadsen | Deeewayne: 1.8.4 is available for you now in the drop down |
| 14:27.59 | jacc0 | @nite613: also check you configured file handles |
| 14:29.01 | Deeewayne | leifmadsen, thanks |
| 14:29.21 | cherzberg | @jacc0 I inserted the two lines. Hope it will solv our problem. Thanks anyway |
| 14:29.59 | leifmadsen | this isn't twitter, you don't need @<name> :) |
| 14:30.13 | nite613 | jacc0: Thanks for the tips! |
| 14:30.31 | cherzberg | :-) |
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| 14:33.47 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4 (2011/05/10), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 14:34.16 | leifmadsen | Asterisk 1.8.4 is now available. See the release announcement at http://www.asterisk.org/node/51631 for more information. |
| 14:35.42 | jaytee | "You'll get my Asterisk 1.2 when you pry it from my cold, bony, arthritic dead fingers!" |
| 14:35.52 | pabelanger | leifmadsen: \o/ |
| 14:37.22 | *** join/#asterisk MikeH_ (~mike@2001:470:1f09:14ed:3058:a330:1cac:4612) |
| 14:38.05 | MikeH_ | What reason would there be for a piece of hardware to be recognised by dahdi, but not asterisk? |
| 14:38.48 | MikeH_ | I've followed the installation instructions to the T, yet the command it asks me to run under asterisk is not there (gsm show spans), so I'm guessing there is an issue between dahdi and asterisk somewhere? |
| 14:38.51 | MikeH_ | or something I've missed? |
| 14:39.09 | sereal | @name was long before twitter |
| 14:39.10 | jacc0 | woei!! going to compile 1.8.4 right away |
| 14:39.31 | cherzberg | is q.8.4 stable? |
| 14:39.41 | jacc0 | 1.8 is not in my opinion |
| 14:39.41 | cherzberg | I mean 1.8.4 :-) |
| 14:39.57 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:39.57 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:40.09 | leifmadsen | cherzberg: we don't know -- do your local stability tests pass on Asterisk 1.8.4? |
| 14:40.12 | leifmadsen | they pass on mine |
| 14:40.53 | jacc0 | but going to test 1.8.4 right away to see if it makes a difference |
| 14:40.57 | jacc0 | :) |
| 14:42.01 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:42.05 | MikeH_ | anyone? |
| 14:42.33 | MikeH_ | Is there perhaps something missing here: http://www.chan-extra.org/index.php/OpenVox_G400P_User_Manual_for_dahdi#3.3.29_load_driver_and_run_asterisk ? |
| 14:42.53 | jacc0 | No changes to 1.8.4-rc3 :S |
| 14:43.34 | leifmadsen | jacc0: of course not, the release is always a snapshot of the last release candidate for that version |
| 14:43.43 | leifmadsen | we don't do release candidates then change things at release |
| 14:43.47 | jacc0 | okay |
| 14:44.09 | leifmadsen | release candidates beyond RC1 are always snapshots of RC1 + specific changes |
| 14:44.13 | leifmadsen | and not snapshots of the branch |
| 14:44.27 | leifmadsen | which means there are many more fixes in the branch that are not in the latest RCs |
| 14:44.30 | *** join/#asterisk Freeaqingme| (~dolf@dsl-083-247-011-232.solcon.nl) |
| 14:44.39 | cherzberg | a ok. We do just testing with 1.8.3.3 and have had a lot of trouble with crashing asterisk |
| 14:45.17 | leifmadsen | cherzberg: did you file issues with backtraces and all the information required to debug and resolve the issue? |
| 14:46.49 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
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| 14:47.37 | Janos | problem seems to be when i transfer calls to my other asterisk server, macro does return the call, when a local sip peer is used |
| 14:47.55 | MikeH_ | Hrm, can no one even point me in the right direction for my issue? |
| 14:50.53 | *** join/#asterisk QuantumSchema (~QuantumSc@rrcs-24-227-92-3.se.biz.rr.com) |
| 14:52.18 | jacc0 | what hardware are you using MikeH_ ? |
| 14:52.31 | MikeH_ | jacc0, Openvox G400P |
| 14:52.52 | QuantumSchema | Good day all! |
| 14:53.34 | jacc0 | I don't have a clue; did you ask your suplier? |
| 14:53.52 | leifmadsen | ya contact Openvox support for help[ |
| 14:53.53 | cherzberg | leifmadsen: We search all log files and found nothing. asterisk just stop working but the console is just working fine. Only connections will no handled anymore. Same with 1.6.2.18 but with less crashes. |
| 14:53.55 | *** join/#asterisk robertj (~Rob@dumbledore.athenshousing.org) |
| 14:54.05 | leifmadsen | cherzberg: sounds like a deadlock then |
| 14:54.11 | leifmadsen | so you should debug it as such |
| 14:54.23 | leifmadsen | ~collectdebug |
| 14:54.23 | infobot | from memory, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:54.28 | MikeH_ | I've found Openvox support to be both slow and difficult unless you speak Chinese. |
| 14:54.48 | MikeH_ | Which is why I came here looking for possible reasons why something would be loaded in dahdi but not Asterisk |
| 14:54.55 | leifmadsen | then you should purchase hardware from someone who speaks your native language |
| 14:54.55 | MikeH_ | as a generic question rather than hardware specific |
| 14:55.58 | cherzberg | I did, but they are out of ideas with the problem |
| 14:56.24 | MikeH_ | leifmadsen, the only options that weren't ridiculously expensive were Atcom and OpenVOX. I originally bought the Atcom, had problems and purchased an OpenVOX card as a repalcement based on the advice I was given here :/ |
| 14:56.46 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:56.46 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:57.10 | MikeH_ | leifmadsen, So that's as far as the support here goes then? It's the suppliers issue :/ |
| 14:57.16 | jacc0 | @cherzberg: use nmap to check if asterisk is still responsive; if not the kill and restart |
| 14:57.44 | jacc0 | *then kill |
| 14:58.22 | cherzberg | jacc0: asterisk is not responding. after restart it works fine. I did the changes to the my.conf you told me. No I can just send a pray and wait. :-) |
| 14:58.25 | leifmadsen | MikeH_: other people are welcome to answer your questions, but there is nothing I can do to help you |
| 14:58.38 | jaytee | Sometimes cheap isn't always better. That's why I prefer Digium hardware |
| 14:58.41 | leifmadsen | I am unfamiliar with Openvox hardware |
| 14:58.49 | jacc0 | right now I'm using the following tools to keep asterisk running: safe_asterisk, astcanary, monit and flip1405(to activate a failover server) |
| 14:59.00 | jacc0 | I can not call that stable |
| 14:59.09 | MikeH_ | jaytee, I purchased a Digium analogue card, but there is limited options with GSM stuff. |
| 14:59.23 | jaytee | ah, GSM |
| 14:59.39 | MikeH_ | leifmadsen, Surely my question is a generic software question being the issue is clearly with dahdi -> asterisk and not the hardware? |
| 14:59.51 | MikeH_ | as far as I understood it, Dadhi handles drivers and interfaces with asterisk? |
| 15:00.16 | robertj | MikeH_, you doing your own cell stuff? |
| 15:00.26 | *** join/#asterisk moy (~moy@CPE003048b11058-CM00222d6b4d65.cpe.net.cable.rogers.com) |
| 15:00.47 | MikeH_ | robertj, It's being used to forward calls to my mobile for free |
| 15:01.01 | robertj | Cool, what piece of hardware generates the radio signal? |
| 15:01.10 | QuantumSchema | I had a question about __MONITOR_EXEC not actually kicking off... |
| 15:01.33 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-163-82.dsl.stlsmo.sbcglobal.net) |
| 15:01.43 | QuantumSchema | I have exten => 6226,n,Set(__MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) |
| 15:01.44 | QuantumSchema | exten => 6226,n,Set(__MONITOR_EXEC=/usr/local/parselog/update_mix_mixmonitor.pl ${UNIQUEID} q${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.wav) |
| 15:01.50 | QuantumSchema | Woop |
| 15:02.07 | MikeH_ | robertj, OpenVox G400P |
| 15:02.14 | QuantumSchema | I have the following in my dialplan.... |
| 15:02.16 | QuantumSchema | exten => 6226,n,Set(__MONITOR_FILENAME=/var/spool/asterisk/monitor/q${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}) |
| 15:02.20 | QuantumSchema | exten => 6226,n,Set(__MONITOR_EXEC=/usr/local/parselog/update_mix_mixmonitor.pl ${UNIQUEID} q${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID}.wav) |
| 15:02.40 | leifmadsen | ~pb |
| 15:02.40 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:02.46 | leifmadsen | QuantumSchema: ^^^ |
| 15:02.49 | robertj | MikeH_, do you get data as well? |
| 15:02.59 | MikeH_ | I'm not sure |
| 15:03.12 | QuantumSchema | Thanks Leif |
| 15:05.35 | QuantumSchema | Alrighty.... |
| 15:05.37 | robertj | needs a packet radio that looks like a cell phone and runs android ;P |
| 15:05.48 | QuantumSchema | I've posted the dialplan and output here: http://pastebin.com/xnsaSAQb |
| 15:06.02 | robertj | we've got some 153 & 155 spectrum that is just lying fallow around here |
| 15:06.16 | QuantumSchema | The it looks like the monitor_exec is supposedly being executed but it doesn't acutally do anything. |
| 15:06.16 | sezuan | shouln'd the ~pb message itself point to a pastebin link? it's more than three lines ;) |
| 15:06.28 | QuantumSchema | LoL sezuan |
| 15:07.20 | *** join/#asterisk circut (~circut@c-24-1-32-67.hsd1.il.comcast.net) |
| 15:07.25 | circut | howdy all |
| 15:07.26 | jacc0 | :P |
| 15:07.36 | QuantumSchema | If I run the command manually with all the variables in place (copy and paste from the log), it executes without a problem. |
| 15:07.48 | circut | i was just wondering if there is a way (in asterisk 1.4) to determine if the call stack will be empty before calling Return() |
| 15:08.03 | QuantumSchema | I've ensured that the owner for any of the directories in question is asterisk too... |
| 15:13.43 | sezuan | What's the purpose of the autoregister option in jabber.conf? |
| 15:13.45 | QuantumSchema | Well... the update_mix_monitor.pl script is owned by root, could that be it? (I'm not thinking so but that's all I got left).... |
| 15:13.54 | Faustov | kaldemar: thanks, looks good, I can hook a condition into the chanunavail dialstatus with this function (glad I upgraded to 1.6 :D) |
| 15:17.41 | QuantumSchema | Is there anyway to see what the returned output is (0 or 1) from what MONITOR_EXEC actually does? |
| 15:21.18 | jacc0 | new SMGv3.3 from sangoma is just released (and should now support A200 & A500 on the latsted ubuntu and debian(squeeze) kernel) find it here: ftp://ftp.sangoma.com/linux/smg/3.2/ |
| 15:21.34 | jacc0 | bye all |
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| 15:59.15 | Preytell | I am having an issue with sip registration and * 1.8.3. Phones will register just fine when the system boots, but slowly the phones will fall off one by one, I pulled traces and see that the phones are attempting to register, but * does not respond. |
| 15:59.41 | Preytell | Phones can still make calls, but of course cannot get internal, or transferred calls. |
| 16:01.41 | leifmadsen | Preytell: sounds like a dead lock |
| 16:01.53 | leifmadsen | you should file an issue with 'core show locks' info |
| 16:01.56 | leifmadsen | ~collectdebug |
| 16:01.56 | infobot | well, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:02.22 | leifmadsen | there is a link on the left hand side for Backtraces which contains info about how to get a backtrace and 'core show locks' output |
| 16:02.31 | Preytell | k |
| 16:02.38 | leifmadsen | after you've gathered the info per those instructions, please do file an issue on https://issues.asterisk.org |
| 16:08.28 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 16:08.43 | cusco | hey folks |
| 16:09.03 | cusco | when I queue() with a macro option |
| 16:09.17 | cusco | will the macro run when joining the queue or when leaving |
| 16:09.31 | cusco | Im getting macro results when I leave queue, but I would like to get them on joining |
| 16:09.36 | cusco | is there something I can do about it? |
| 16:09.58 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 16:11.00 | leifmadsen | cusco: the macro option executes the macro just prior to bridging the caller and the agent |
| 16:11.10 | leifmadsen | that's how it works |
| 16:11.22 | leifmadsen | there is no "macro on entering the queue" option |
| 16:12.21 | Preytell | I do see a lot of this: chan_sip.c: sip_xmit of 0x8fac528 (len 388) to 216.115.69.144:5060 returned -1: Operation not permitted, it's on my sip trunk that as far as I know is working fine. |
| 16:16.28 | *** join/#asterisk tuxx- (~tuxx@nightshade.nl) |
| 16:17.55 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 16:20.01 | cusco | leifmadsen: is there a way to know the estimated queue-hold-time prior to queuing the caller?? |
| 16:20.03 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 16:20.58 | leifmadsen | cusco: core show function QUEUE_VARIABLE |
| 16:21.03 | leifmadsen | cusco: core show function QUEUE_VARIABLES |
| 16:21.52 | cusco | leifmadsen: prior to queue() the caller ? |
| 16:22.11 | leifmadsen | yes |
| 16:22.17 | leifmadsen | look at the content and try it |
| 16:22.40 | cusco | I know about QUEUEHOLDTIME |
| 16:22.50 | cusco | but I thought it would onlybe available after Queue() |
| 16:22.58 | leifmadsen | I don't know, you'll have to try it |
| 16:23.03 | cusco | how do I specify WHICH queue ?= |
| 16:23.14 | leifmadsen | [Syntax] |
| 16:23.14 | leifmadsen | QUEUE_VARIABLES(queuename) |
| 16:23.25 | cusco | sorry yes |
| 16:23.32 | cusco | my asterisk says nothing of that lol |
| 16:23.45 | leifmadsen | I'm running trunk, but it certainly would be in 1.8, and probably 1.6.2 |
| 16:23.49 | leifmadsen | I haven't used 1.4 in years |
| 16:23.55 | cusco | http://paste.debian.net/116554/ |
| 16:24.01 | cusco | I'm runniong 1.6 |
| 16:24.02 | cusco | I'm runniong 1.62 |
| 16:24.09 | leifmadsen | shrugs |
| 16:24.34 | cusco | let me try then! |
| 16:25.33 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 16:27.08 | cusco | on a non-related subject |
| 16:27.20 | cusco | May 10 17:16:11] NOTICE[16159] chan_sip.c: Sending fake auth rejection for device "265 - Bruno Castanheira" <sip:265@10.100.100.5>;tag=4fc9dae9 |
| 16:28.01 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 16:28.02 | Lantizia | anyone else think the skype trunk digium helped make will go down the pan now MS owns skype? |
| 16:28.22 | leifmadsen | I don't |
| 16:28.46 | Lantizia | asterisk does directly compete after all |
| 16:28.56 | leifmadsen | I'm sure there is some sort agreement in place that still needs to be dealt with |
| 16:29.16 | leifmadsen | the agreement would have been with Skype, and that wouldn't be negated just because someone else bought the company |
| 16:29.25 | leifmadsen | however after the agreement then all things are game I suppose |
| 16:29.28 | leifmadsen | IANAL |
| 16:29.54 | Lantizia | theyd be silly to loose the linux/android/symbian/mac users though since they get money from skype out - worth maintaining the clients for that alone |
| 16:30.34 | Lantizia | but i reckon MS will try and entice the business/enterprise customers off asterisk by withdrawing skype for asterisk eventially |
| 16:30.47 | leifmadsen | s/loose/lose/ |
| 16:31.04 | Lantizia | keyb is on low poer - keep missing letters :S |
| 16:31.08 | Lantizia | *power |
| 16:31.12 | leifmadsen | you hit too many letter s:) |
| 16:31.41 | Lantizia | same issue :P |
| 16:32.24 | *** join/#asterisk bradland (~bradland@c-69-138-99-118.hsd1.fl.comcast.net) |
| 16:32.40 | *** part/#asterisk bradland (~bradland@c-69-138-99-118.hsd1.fl.comcast.net) |
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| 16:39.27 | *** part/#asterisk mpe (~mpe@office.ipvision.dk) |
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| 16:43.03 | *** join/#asterisk unixSnob (~unixSnob@212.117.169.230) |
| 16:43.05 | raden | Katty, :D |
| 16:45.10 | QuantumSchema | I was wondering if anyone had any suggestions about my MONITOR_EXEC not actually executing? |
| 16:51.59 | cj | anyone have an fxs and two fxo modules they want to sell? |
| 16:53.55 | *** join/#asterisk Preytell (~Preytell@64-199-153-59.ip.mcleodusa.net) |
| 16:55.00 | cj | sorry, got that backwards. two fxs modules and one fxo module :) |
| 16:56.00 | *** join/#asterisk sled-dog (~luser@65-124-95-55.dia.static.qwest.net) |
| 16:56.30 | sled-dog | where was astricon USA last year? in Carolina? |
| 16:57.33 | leifmadsen | sled-dog: http://www.astricon.net/2010/default.aspx |
| 16:57.46 | sled-dog | thanks. |
| 17:02.02 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 17:18.35 | sereal | how do I create a variable for a extension (I have a extension number that is fairly long and would rather not type it out every time) |
| 17:19.17 | *** join/#asterisk jkroon (~jkroon@197.173.250.40) |
| 17:22.16 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.70) |
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| 17:31.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 17:42.59 | leifmadsen | sereal: use 'same => ' |
| 17:43.09 | leifmadsen | sereal: it depends what version of Asterisk you're using |
| 17:43.14 | *** join/#asterisk ph8 (ph8@unaffiliated/ph8) |
| 17:43.52 | *** join/#asterisk golam (4531720a@gateway/web/freenode/ip.69.49.114.10) |
| 17:44.46 | leifmadsen | sereal: if you're using 1.6.2 or later, then you can do something like: |
| 17:44.53 | leifmadsen | exten => _NXXNXXXXXX,1,NoOp() |
| 17:44.59 | leifmadsen | same => n,Verbose(2,Hello there) |
| 17:45.10 | leifmadsen | replace the pattern match with your variable if you wish |
| 17:45.57 | golam | yesterday I have modifed this file /etc/dahdi/system.conf and today it was reverted back to yesterdays version.....we do restart asterisk every night.....any idea?? |
| 17:47.25 | leifmadsen | not sure, unless dahdi_genconf got run again |
| 17:51.14 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 17:51.34 | eXcAliBuR | how can i register => to my sip provider and have it go as TCP and not UDP ? |
| 17:52.40 | leifmadsen | eXcAliBuR: per the sample configuration file: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 17:52.47 | leifmadsen | use transport:// |
| 17:52.48 | cj | http://www.asterisk.org/doxygen/trunk/sip_tcp_tls.html maybe? |
| 17:52.56 | cj | ah, nice. |
| 17:52.57 | leifmadsen | documentation ftw! |
| 17:53.11 | cj | leifmadsen: o/ |
| 17:53.14 | leifmadsen | cj: ohai :) |
| 17:53.32 | leifmadsen | ok afk for a bit as I need to get a walk in |
| 17:55.00 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:56.12 | *** join/#asterisk MarKsaitis (~MarKsaiti@82.144.242.158) |
| 17:58.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-048.mobile.uci.edu) |
| 18:01.44 | eXcAliBuR | i don't know how to follow the syntax :{ |
| 18:02.32 | eXcAliBuR | i need an example i did register => transport:tcp username:secret@host:5060/200 |
| 18:02.41 | cj | register => tcp://user@sipprovider.com:mypassword:myauthuser@providerhostname:5060/200 |
| 18:02.51 | eXcAliBuR | oh |
| 18:02.52 | eXcAliBuR | hahaha |
| 18:03.43 | eXcAliBuR | this is so exciting |
| 18:03.57 | cj | so maybe register => tcp://username@sipprovider.com:secret:username@host:5060/200 |
| 18:03.59 | eXcAliBuR | i went and got the new book the future of telephony is now |
| 18:04.12 | cj | yeah, i got that too. you know leifmadsen wrote it, right? :) |
| 18:04.34 | eXcAliBuR | hahaha his name is on it |
| 18:04.42 | cj | it's true :) |
| 18:04.43 | eXcAliBuR | i didn't realize |
| 18:04.48 | eXcAliBuR | how kool is that |
| 18:04.56 | cj | go go gadget open source. |
| 18:05.08 | cj | I've seen at least one of the other authors here, too :) |
| 18:07.12 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.8.6) |
| 18:07.14 | DelphiWorld | hey :P |
| 18:07.23 | DelphiWorld | good bye chan_skype:) |
| 18:07.26 | DelphiWorld | leifmadsen: :D |
| 18:07.33 | cj | DelphiWorld: you think? |
| 18:07.50 | DelphiWorld | cj, was a joke |
| 18:07.51 | cj | I was thinking I'd get a chan_skype license... |
| 18:08.21 | *** join/#asterisk tomaw_ (tom@freenode/staff/tomaw) |
| 18:08.34 | DelphiWorld | cj, cause Microsoft buy skype |
| 18:08.47 | cj | but why would they get rid of the chan_skype product? |
| 18:08.55 | cj | seems like a revenue stream... |
| 18:09.13 | DelphiWorld | cj, was a joke :D |
| 18:09.32 | cj | okay. my humerator must be broken. |
| 18:09.43 | jaytee | May 10, 2011 Microsoft buys Skype, June 12, 2011 first reported BSOD on an Android smartphone running Skype. |
| 18:11.15 | neurosys | exited non-zero on disconnect. What exactly does this mean? |
| 18:11.36 | bdfoster | neurosys, usually that's a good thing |
| 18:11.47 | neurosys | Ok :) |
| 18:12.07 | DelphiWorld | inquire and sue neurosys |
| 18:12.43 | neurosys | :D |
| 18:13.56 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.8.6) |
| 18:14.45 | pabelanger | If anybody using the Asterisk packages Digium is providing for Debian or Ubuntu, 1.8.4 has been release. I'd be interested in hearing feedback from upgrading to a previous version. |
| 18:15.32 | eXcAliBuR | my asterisk is working :} |
| 18:15.34 | eXcAliBuR | many thanks |
| 18:15.43 | eXcAliBuR | i'm connecting to the outside world |
| 18:15.52 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 18:24.38 | carrar | People still use skype?! :) |
| 18:29.45 | leifmadsen | cj: it's ok -- delphiworld isn't very funny |
| 18:32.06 | *** join/#asterisk twanny796 (~root@46.11.88.154) |
| 18:32.19 | twanny796 | how do I call the console? |
| 18:32.25 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 18:35.23 | ChannelZ | yell "hey, console!" |
| 18:35.39 | leifmadsen | twanny796: call the console, or call form the console? |
| 18:35.41 | leifmadsen | from* |
| 18:35.53 | leifmadsen | calling the console I think is just: Dial(Console/dsp) |
| 18:37.53 | twanny796 | leifmadsen; calling to the console |
| 18:38.26 | twanny796 | 1236 ;) |
| 18:38.38 | neurosys | skype = muckrosofty now |
| 18:38.49 | leifmadsen | neurosys: welcome to several hours ago :) |
| 18:38.49 | m4xx | deal went through today? |
| 18:39.02 | neurosys | leifmadsen: heh I be a lil slowww |
| 18:39.06 | *** join/#asterisk angryuser (53a79c5a@gateway/web/freenode/ip.83.167.156.90) |
| 18:55.01 | eXcAliBuR | leifmadsen are you back ? |
| 18:56.30 | leifmadsen | eXcAliBuR: define "back" |
| 18:56.44 | *** join/#asterisk mpe (~mpe@91.100.62.248.generic-hostname.arrownet.dk) |
| 18:56.53 | eXcAliBuR | yay |
| 18:57.06 | eXcAliBuR | i must apologize... I didn't realize you were who you are |
| 18:57.12 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |
| 18:57.14 | leifmadsen | I don't know who I am either :) |
| 18:57.18 | eXcAliBuR | cj pointed it out to me |
| 18:57.23 | eXcAliBuR | I have your book :} |
| 18:57.27 | eXcAliBuR | i feel special |
| 18:57.44 | leifmadsen | :) |
| 18:57.47 | leifmadsen | hope you enjoy it |
| 18:59.40 | cj | leifmadsen: I also often don't know who I am. It's good for one to have folks remind one who one is... |
| 18:59.53 | leifmadsen | aye :) |
| 19:00.52 | cj | what are the chances that I burned out my FXS module by plugging it in to the telco link and sending ringing voltage through it? |
| 19:01.10 | leifmadsen | high |
| 19:01.12 | cj | 'cuz the ring test isn't causing the station to ring |
| 19:01.14 | cj | yay. |
| 19:01.28 | leifmadsen | voltage into voltage producing circuits is not good |
| 19:01.34 | cj | inorite |
| 19:01.47 | cj | okay, well, I'll order a couple new ones, then. |
| 19:03.24 | *** join/#asterisk kdekorte (~kdekorte@63-163-137-113.iprev.kci.net) |
| 19:10.12 | *** part/#asterisk mpe (~mpe@91.100.62.248.generic-hostname.arrownet.dk) |
| 19:13.53 | Preytell | in the log I see this: Call-ID: 1856943b72739c047a03d0423bb59436@10.0.111.8 where does the system get that ip address? It's not an ip on the server. |
| 19:16.14 | Preytell | nm, found it, errant entry in the hosts file. |
| 19:18.31 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 19:23.27 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:38.19 | *** join/#asterisk knarfly (~vlad@c-98-242-233-74.hsd1.fl.comcast.net) |
| 19:39.31 | knarfly | I keep seeing this message on my console doing dnsmgr_lookup for 'sip.my_sip_provider.com' is there a way to stop this? |
| 19:40.07 | leifmadsen | that's normal |
| 19:40.14 | leifmadsen | if you have dnsmgr enabled |
| 19:40.30 | leifmadsen | you'd have to remove the line of code that is creating that message or disable dnsmgr entirely |
| 19:40.47 | knarfly | I guess I do...I see it's not harming anything but it's filling up the console screen |
| 19:41.00 | leifmadsen | options have been provided |
| 19:47.26 | ChannelZ | you can lower the refresh. |
| 19:47.40 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
| 19:49.08 | sezuan | any linphone users here? When I try to subscribe an extension I get the following error: SUBSCRIBE failure: no Accept header: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'LocalSets', subscribeuri: '' |
| 20:03.50 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 20:13.39 | *** join/#asterisk Oliver1 (~oliver@manz-590f361c.pool.mediaWays.net) |
| 20:21.02 | Oliver1 | Ich experimentiere im Auftrag meine Familie mit eine Asterisk rum. Wir würden gerne SIP-DDI Accounts als Trunk einbinden. Dabei ist die Frage aufgekommen, ob eine Asterisk in der Lage ist einen QoS darzustellen? |
| 20:23.00 | _Corey_ | auf English bitte |
| 20:24.12 | Oliver1 | sorry, thougt this channel is for german language. |
| 20:25.18 | _Corey_ | There may be some german speakers here, though you'll get help faster if you ask your question in english |
| 20:26.29 | leifmadsen | heh... |
| 20:28.04 | Oliver1 | okl, I will try it. These are my first steps with Asterisk. And I' m a little bit scare to do this in english... |
| 20:28.16 | Oliver1 | anyway |
| 20:29.18 | _Corey_ | if you spend more time in here you will see at least 10% of the questions are obviously written in chinese and pasted into Babelfish or something, so fear not |
| 20:29.50 | Oliver1 | so, I want to use Astersik with some SIP-DDI Accouts. The first thing I want to know, is, if Asterisk is able to handle QoS? This is the Distri I favour http://www.iptam.com/ |
| 20:30.48 | _Corey_ | Asterisk can set TOS tags on your packets, but your network equipment needs to handle the QoS |
| 20:31.20 | Oliver1 | Please, what "TOS"? |
| 20:32.22 | Oliver1 | _Corey_: thanks, I will try it ;-) |
| 20:32.25 | leifmadsen | Type Of Service |
| 20:32.29 | Oliver1 | ahhh |
| 20:32.30 | Oliver1 | ok |
| 20:32.33 | Oliver1 | sorry |
| 20:34.22 | Oliver1 | so, this is the thing I also want to know; we have some Softphones on XP-machines. Between the machines and the Asterisk we use actually are a non intelligent switch. What is a good way to implement QoS? |
| 20:34.49 | _Corey_ | probably best not to think about QoS unless you're working with remote users |
| 20:35.06 | Oliver1 | Btw; I' m just a salesmen, not a IT-guy ;-) |
| 20:35.37 | Oliver1 | no, all the users are in one local network |
| 20:35.48 | Oliver1 | and this will not change! |
| 20:36.23 | _Corey_ | it is very unusual to need QoS on a LAN |
| 20:37.49 | Oliver1 | the thing is, that sometimes we have echo' s and time delays |
| 20:38.31 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:38.34 | _Corey_ | do you have a hard phone or just XP soft phones? |
| 20:38.37 | Oliver1 | in out LAN we have VoIP, and outside on the Asterisk-Machine there is a normal connection to the PSTN |
| 20:38.57 | Oliver1 | XP Softphones from Global-IP |
| 20:39.08 | *** join/#asterisk draeath (~draeath@unaffiliated/draeath) |
| 20:39.11 | Oliver1 | *in our |
| 20:39.54 | Oliver1 | on the LAN we handle also our normal Data-Usage like Exchange |
| 20:40.12 | Oliver1 | Internet, some cloud-services |
| 20:40.53 | _Corey_ | well, it is more likely an issue with the PC than bandwidth |
| 20:41.15 | Oliver1 | with all PC's? |
| 20:42.45 | _Corey_ | not sure, but rule things out one at a time... if you're only using softphones try using a hardphone to see if you have issues there too |
| 20:43.01 | Oliver1 | that sounds goog! |
| 20:43.06 | Oliver1 | *good |
| 20:43.08 | _Corey_ | other factors could be load or problems on the asterisk server itself |
| 20:43.32 | _Corey_ | LAN bandwidth is less likely than most other things |
| 20:43.40 | Oliver1 | ok |
| 20:44.17 | Oliver1 | VoIP like a own science ;-) |
| 20:45.14 | Oliver1 | does it makes sense to upgrade all our inhouse-cable up to CAT 6 or 7? |
| 20:47.28 | _Corey_ | no |
| 20:47.47 | _Corey_ | (not to solve this problem anyway) |
| 20:47.59 | Oliver1 | ok |
| 20:49.10 | *** join/#asterisk foxb (~chatzilla@216.191.245.130) |
| 20:49.21 | Oliver1 | what I want to know; in companies who use VoIP with Asterisk, did they route there VoIP-traffic together with there normal Data over the same cable in case of using CTI? |
| 20:50.24 | foxb | Hi I have problem installing 1.8.4 in debian - It does not enable res_crypt, but openssl is installed ... any ideas |
| 20:52.41 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:57.23 | russellb | foxb: you need openssel-dev |
| 20:57.26 | russellb | er, openssl-dev |
| 20:57.29 | russellb | or not |
| 21:02.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:03.45 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 21:05.47 | Jasnejac | Hi all. Anybody have a connection with Global Crossing? I have an issue where all calls fail after 15 minutes. According to GC, and I have no traces for this, it is effectively because asterisk is ignoring the UAC response in the initial invite. Has anyone else seen this please? |
| 21:08.18 | _Corey_ | Jasnejac: We have Global... been using their SIP for ~4 yrs, very solid |
| 21:09.05 | Jasnejac | I only have this problem with them. I'm frankly confused! |
| 21:09.54 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 21:11.24 | Jasnejac | d_Corey_: do you have a session-refresher=uas setting in your sip.conf by any chance? |
| 21:18.55 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
| 21:21.14 | _Corey_ | nope |
| 21:22.04 | *** join/#asterisk Pathin (~root@gladsheim.nullbytestudios.net) |
| 21:23.09 | _Corey_ | other than dtmfmode=inband and progressinband=yes, it's pretty much all defaults |
| 21:24.05 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
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| 23:42.50 | pdtpatrick_ | On asterisk after changing voicemail.conf .. is there something I need to do for the user's extension to start going to VM ? |
| 23:46.24 | leifmadsen | you need to direct it to voicemail in the dialplan |
| 23:46.31 | leifmadsen | it doesn't happen automatically |
| 23:48.37 | pdtpatrick_ | @leifmadsen can you please elaborate. Dailplan = ? |
| 23:48.47 | pdtpatrick_ | where do i need to go or look |
| 23:49.00 | p3nguin | extensions |
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| 23:49.06 | p3nguin | extensions.conf |
| 23:49.09 | Preytell | now that's a question. |
| 23:49.36 | p3nguin | If you don't know what extensions are, you're in the wrong business. |
| 23:50.24 | pdtpatrick_ | I know what extensions are im just not familiar with asterisk.. im usually a programmer/linux guy. Trying to learn asterisk :) |
| 23:51.17 | p3nguin | It was clear to me that you don't know what extensions are. |
| 23:51.27 | p3nguin | You think phones are extensions. They are not. |
| 23:52.34 | leifmadsen | ~thebook |
| 23:52.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 23:52.45 | leifmadsen | reading ftw! |
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