| 00:00.14 | ChannelZ-Wk | Hmm. Mine ended and showed a slide saying to look in the title bar of the little window to see if you passed, but I didn't see anything else interesting. And then I clicked back to the browser and it went away. Oh well. |
| 00:00.44 | ChannelZ-Wk | oh wait here we go |
| 00:01.47 | ChannelZ-Wk | Clicking the blue thumbnail of it under 'prior enrollments' took me to a new page. 100%! |
| 00:07.48 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 00:14.10 | Freeaqingme | if someone tries to call me through skype, is there any way of routing it to asterisk? |
| 00:14.17 | Freeaqingme | (vice versa is not required) |
| 00:18.37 | [TK]D-Fender | make sure yours is a business accoutn and you can use their SIP connector service |
| 00:20.00 | Freeaqingme | [TK]D-Fender, thanks. 'business account' = 'premium account'? |
| 00:20.18 | [TK]D-Fender | I believe so. Ther terms are all there |
| 00:20.28 | Freeaqingme | k, tnx |
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| 00:32.24 | volga629 | Hello Everyone, Is through IAX2 trunk video should work ? |
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| 01:22.15 | mnathani | how do I use call files to get asterisk to dial 2 numbers and conference them together? |
| 01:24.01 | WIMPy | Use application Dial and whatever you want to dial. |
| 01:24.36 | WIMPy | Or just use the extension if it's something callable from your dialplan. |
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| 01:28.10 | mnathani | WIMPy: ok thanks |
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| 02:15.43 | hjf | is the SPA3000 a decent gateway to connect to asterisk? for the FXO port |
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| 02:27.13 | mnathani | What do I need to do to secure my Asterisk install ? I see this in my logs currently: http://pastebin.com/FmM8ixUd |
| 02:35.45 | igcewieling | mnathani: fail2ban, iptables, allowguest=no alwaysauthreject=yes \ |
| 02:35.55 | igcewieling | or I could google it for you |
| 02:36.21 | mnathani | igcewieling: Thanks for getting me started |
| 02:37.17 | igcewieling | also leave context=default in [general] and set a specific context for each peer, which is not default. i.e. don't use a context named default |
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| 02:40.19 | WIMPy | Don't use default, not even for guests. |
| 02:42.55 | carrar | open root policy is best |
| 03:11.40 | ChannelZ | Why, is "default" some sort of magical context? |
| 03:12.05 | WIMPy | Yes |
| 03:17.46 | ChannelZ | Which is what exactly |
| 03:22.11 | WIMPy | I can't remember where it caught me. But I've read that it's used in multiple places. |
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| 05:35.56 | j4jackj | How much fun can one have with a codec? |
| 05:36.23 | hjf | what's the proper way to "time out" an unanswered call? like: exten=>1,1,Dial(SIP/hjf,20) exten=>1,n,Hangup() |
| 05:36.50 | hjf | (i want to disable voicemail) |
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| 05:53.43 | [TK]D-Fender | <PROTECTED> |
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| 07:18.31 | giucio | Hi, I'm trying to implement a system where an operator clicks on a web page and a call is automatically established between him/her and a phone number. I'd like this to be as quick as possible. The solution I implemented so far is a webservice doing an Originate call through AMI. The downside is that it establishes two different calls. Once the agent picks up the phone, it then dials the target number. Is there a way to automatically "connect" the operator |
| 07:19.34 | ChannelZ | What device is the operator using? |
| 07:19.41 | wdoekes | giucio: pass Auto-Answer headers |
| 07:20.50 | giucio | wdoekes: I'm doing it, but it looks like very few softphones implement it, which is pretty much why I was wondering if I'm doing it the right way. |
| 07:21.31 | giucio | I actually only got it working with jitsi |
| 07:21.46 | giucio | I'm trying to avoid non free-software solutions if possible |
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| 07:25.23 | pietro | hi |
| 07:25.54 | pietro | someone knows an RFC6035 capable voice quality reports collector ? |
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| 10:02.02 | eeezkil | Recently I was asked to start an asterisk server and I'm concerned a bit about security because I feel that it would be "a nice target for hackers" |
| 10:02.02 | eeezkil | <PROTECTED> |
| 10:02.02 | eeezkil | <PROTECTED> |
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| 10:04.46 | bacobart | http://kb.smartvox.co.uk/asterisk/secure-asterisk-pbx-part-1/ |
| 10:04.56 | bacobart | follow these instructions and you should be fine |
| 10:05.43 | bacobart | most important are disabling guest access and using secure passwords |
| 10:05.53 | eeezkil | bacobart, thank you.. I'll check it out |
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| 10:08.09 | kaldemar | eeezkil: http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 10:08.27 | eeezkil | kaldemar: thank you |
| 10:09.14 | kaldemar | you won't see any attacker ip addresses for sip in /var/log/messages, use the security event framework for those (security log in logger.conf). |
| 10:09.50 | kaldemar | many people use fail2ban too. |
| 10:10.05 | eeezkil | is that some script? |
| 10:10.09 | eeezkil | in python perl? |
| 10:10.30 | kaldemar | http://www.fail2ban.org |
| 10:11.01 | eeezkil | if the malicious users are using various proxies? |
| 10:11.32 | bacobart | if you get a distribution like asterisknow or freepbx those include fail2ban. although using those distributions is not popular here as ppl here tend to prefer editing text files / dislike the dialplan configurations used by those distros. |
| 10:12.24 | eeezkil | I'm comfortable with anything that gets the job done the best way possible |
| 10:12.38 | eeezkil | if I have to I'll write something myself |
| 10:13.25 | bacobart | asterisknow/freepbx include a webgui for configuring. this is helpful for people new to the system. ofcourse using asterisk and doing everything yourself will make you more of a guru later on:P |
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| 10:15.58 | eeezkil | I would like to get into asterisk fast.. but security is important (and maybe understanding asterisk on a really low level would help) |
| 10:16.00 | giucio | Hi, when using Originate through AMI to establish a call, two separate CDR records are written. Is it possible to avoid the CDR recording two different entries? |
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| 10:19.21 | bacobart | i'm using originate through ami and get just 1 cdr record per call |
| 10:20.08 | giucio | bacobart: I'm using the Local channel |
| 10:20.23 | giucio | it looks like I'm hitting this: https://issues.asterisk.org/jira/browse/ASTERISK-19996 |
| 10:20.31 | giucio | bacobart: are you using Local too? |
| 10:21.14 | bacobart | no |
| 10:24.24 | giucio | bacobart: do you think I could have a look at your originate call? |
| 10:25.20 | eeezkil | what are the differences between "Asterisk RealTime PostgreSQL" and normal Asterisk setups |
| 10:25.44 | eeezkil | maybe you can administer Аsterisk directly from postgre? |
| 10:26.14 | kaldemar | eeezkil: realtime is as normal as using "normal" config files. realtime is a way of using a database for configurations. |
| 10:26.17 | kaldemar | ~book |
| 10:26.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:26.33 | kaldemar | reading some of that will help you plenty. |
| 10:26.51 | eeezkil | kaldemar: thank you I have it |
| 10:27.05 | eeezkil | sorry for the annoying questions.. |
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| 10:37.47 | fsjdfh | can you send SMS with asterisk |
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| 10:51.49 | *** join/#asterisk eject_ck (~Eugene@95.67.72.22) |
| 10:52.59 | eject_ck | Hi, in dialplan I need set callerid with when I'm calling via external VoIP provider, but only for one internal extension, how can I check calling extention id? in dialplan? |
| 10:56.00 | giucio | bacobart: fixed it, thanks anyways |
| 10:56.09 | *** join/#asterisk eject_ck1 (~Eugene@95.67.72.22) |
| 10:57.02 | fsjdfh | Hello there, is it possible to send/recieve SMS though Asterisk |
| 10:57.22 | fsjdfh | *through |
| 11:00.43 | eject_ck1 | exten => _X.,n,Set(CALLERID(num)=(foo=${IF($[ ${EXTEN} = _7XXX.]?777:666)})) |
| 11:01.21 | eject_ck1 | how to use IF construction to match ${EXTEN} ? |
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| 11:12.01 | *** join/#asterisk vbrinza (~vbrinza@188.138.253.238) |
| 11:13.07 | vbrinza | hello all. is there any solution for asterisk behind nat and remote client behind an other nat than using STUN server? |
| 11:13.17 | vbrinza | and port forwarding on client side |
| 11:15.57 | kaldemar | vbrinza: asterisk is a fine solution for that. |
| 11:16.00 | kaldemar | ~sipnat |
| 11:16.00 | 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, directmedia, externhost or externaddr, and localnet. |
| 11:16.35 | kaldemar | eject_ck1: what are you trying to test? |
| 11:18.14 | kaldemar | eject_ck1: is this what you're really after: $["${EXTEN::1}" = "7"] ? |
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| 11:33.24 | OS_Florent2 | hi |
| 11:34.28 | OS_Florent2 | It is possible to have a small help on a behavior of the bridge application ? |
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| 11:40.08 | OS_Florent2 | After use Bridge application if 1 leg hangup it is possible to not hangup the other leg ? |
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| 12:01.46 | cusco | hi |
| 12:02.38 | OS_Florent2 | hi |
| 12:02.41 | WIMPy | OS_Florent2: IIRC they continue where they came from. |
| 12:03.07 | OS_Florent2 | in fact i dont use bridge application, but manager bridge command |
| 12:03.24 | OS_Florent2 | we cant pass option with the manager command? |
| 12:03.37 | OS_Florent2 | like they are in bridge application? |
| 12:03.55 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 12:05.07 | cusco | I'm wondering about using cdr_tds vs cdr_odbc |
| 12:05.11 | cusco | to store cdr in ms-sql |
| 12:05.12 | WIMPy | I don't think so. But you could try to redirect the call leg before doing the bridge. |
| 12:05.27 | OS_Florent2 | i just what i was thinking |
| 12:05.34 | OS_Florent2 | i will try tnak for help |
| 12:05.35 | cusco | states that unixodbc will use freetds, so I could make asterisk use it directly |
| 12:05.51 | cusco | but seems that odbc is always recomended |
| 12:05.55 | cusco | so I'm wondering. . . . |
| 12:07.10 | *** join/#asterisk andrewyager (~andrewyag@103.7.193.44) |
| 12:09.36 | cusco | also the difference between cdr_odbc and cdr_adaptative_odbc ? |
| 12:11.43 | igcewieling | cusco: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#Monitoring_id263137 |
| 12:15.25 | cusco | ok I got the adaptative part |
| 12:15.51 | cusco | how about freetds vs unixodbc |
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| 12:38.59 | jmetro | Anyone know how to increase visible characters on the Aastra screens <.< |
| 12:39.18 | jmetro | 8 characters is nothing, i've got space for atleast 15 >.> |
| 12:40.08 | tuxx- | which type of aastra? :P |
| 12:40.12 | tuxx- | 31i? |
| 12:40.14 | jmetro | 6757i |
| 12:40.30 | tuxx- | you got multiple lines, and you can scroll afaik |
| 12:40.51 | tuxx- | but you cant increase the charcount |
| 12:40.53 | jmetro | for softkeys? |
| 12:40.55 | tuxx- | oh right |
| 12:40.57 | tuxx- | yeah no |
| 12:41.02 | jmetro | bawrrgh |
| 12:41.04 | tuxx- | thats impossibru |
| 12:41.35 | jmetro | the 3 dots its displaying to cut off my text is all i need >.< |
| 12:42.31 | tuxx- | >_> |
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| 13:41.02 | hjf | what's the proper way to "time out" an unanswered call? like: exten=>1,1,Dial(SIP/hjf,20) exten=>1,n,Hangup() |
| 13:41.16 | hjf | i was reading that you can send a status code with Hangup |
| 13:41.24 | hjf | I used 19 (no answer from user) |
| 13:41.50 | hjf | but i'd rather do what the telco here does.. if it rings for more than 1 minute, it gives you a busy tone |
| 13:42.16 | kaldemar | are you sure it's a busy tone? |
| 13:42.25 | [TK]D-Fender | hjf: "core show application congestion" |
| 13:43.11 | [TK]D-Fender | that'd be a "reorder" tone |
| 13:43.14 | hjf | kaldemar: well, here we don't have the "fast busy" tone which means an error |
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| 13:43.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 13:47.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:48.20 | hjf | hmmm |
| 13:49.02 | hjf | congestion and hangup act the same on the SPA. they give a busy tone... the stupid CSipSimple doesn't recognize it, it just gives me a 480 error |
| 13:49.29 | eject_ck1 | can I enable enable music on hold during call via CLI ? |
| 13:53.11 | hjf | so my asterisk is working pretty nicely now |
| 13:53.16 | hjf | how can i break it again? |
| 13:53.19 | hjf | what can i try? :P |
| 13:53.37 | *** join/#asterisk Alex_h (~AlexHold@178.78.119.76) |
| 13:53.50 | hjf | i was thinking: is it possible to have "roaming" clients? that is, clients that can be inside and outside the LAN |
| 13:54.11 | hjf | or will everything just break because of NAT? |
| 14:00.43 | [TK]D-Fender | hjf: make 2 different profiles, one with the LAN, the other with the WAN IP |
| 14:01.01 | [TK]D-Fender | hjf: Or always VPN when remote |
| 14:01.24 | *** join/#asterisk apb1963 (~apb1963@174.134.98.138) |
| 14:01.34 | [TK]D-Fender | eject_ck1: You don't just 'enable" it. |
| 14:02.06 | [TK]D-Fender | eject_ck1: And you should be clear about the precise circumstances you want to create. |
| 14:02.15 | *** join/#asterisk eject_ck (~Eugene@95.67.72.22) |
| 14:10.32 | hjf | [TK]D-Fender: how do i deal with the RTP ports and the NAT? should i just forward the SIP ports or also a set of UDP ports for RTP? |
| 14:12.10 | kuku | I'm using Application: Bridge and in that diaplan I use Playback(), but the sound plays only on the side of the asterisk extension, not on the call recepient... |
| 14:12.14 | [TK]D-Fender | You always have to forward SIP & RTP |
| 14:12.35 | *** part/#asterisk eject_ck (~Eugene@95.67.72.22) |
| 14:23.45 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 14:24.03 | hrolf | Hi #asterisk |
| 14:24.29 | hrolf | Is it possible to receive call on SIP extensions? These SIP extensions are registered in IP office? |
| 14:25.11 | Greenlight | WHere does Asterisk fit into this ? |
| 14:25.55 | *** join/#asterisk wm_domino (~William@24-107-186-9.dhcp.stls.mo.charter.com) |
| 14:26.37 | [TK]D-Fender | hrolf: Please clarify what it is you're asking if it is possible to do. Your terminology is too vague |
| 14:27.51 | hrolf | [TK]D-Fender: I mean, can we use Asterisk as a SIP user agent? |
| 14:28.03 | [TK]D-Fender | hrolf: That is all Asterisk is |
| 14:28.06 | [TK]D-Fender | ~b2bua |
| 14:28.06 | infobot | methinks b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 14:28.09 | [TK]D-Fender | ^ |
| 14:28.59 | hrolf | [TK]D-Fender: Let me clarify a bit. The SIP extensions are created in Avaya IP Office. Now we usually connect on it in 3CX phone. |
| 14:29.24 | hrolf | [TK]D-Fender: Can we use Asterisk, to register on those extensions, and receive the calls from Avaya IP office? |
| 14:29.31 | [TK]D-Fender | hrolf: Yes |
| 14:29.39 | hrolf | [TK]D-Fender: How? |
| 14:29.44 | hrolf | in sip.conf? |
| 14:29.50 | [TK]D-Fender | hrolf: Clearly |
| 14:29.54 | hrolf | type=peer or friend? |
| 14:30.00 | [TK]D-Fender | hrolf: typically peer. |
| 14:30.16 | [TK]D-Fender | hrolf: This is a borking entry like jsut about any other |
| 14:30.20 | [TK]D-Fender | boring* |
| 14:30.24 | hrolf | [TK]D-Fender: Okay, and for username and password where do we specify that? |
| 14:30.33 | [TK]D-Fender | in the sip peer. |
| 14:30.38 | [TK]D-Fender | it's all the same |
| 14:30.51 | hrolf | [TK]D-Fender: Like I'm given 6001 extension, 123456 password and 192.168.0.231 the IP of IP Office |
| 14:30.59 | hrolf | [TK]D-Fender: I see. Thanks. Let me try it |
| 14:31.03 | [TK]D-Fender | hrolf: So go make a peer entry to match that |
| 14:34.19 | *** join/#asterisk navaismo (~navaismo@189.191.251.129) |
| 14:38.18 | hrolf | [TK]D-Fender: Do I have to make SIP as same as in IP office? Like if it is 6001 in IP office, then do I have to make [6001] in sip.conf? |
| 14:38.46 | hrolf | [TK]D-Fender: or can I name it anything, and space in username=? |
| 14:38.55 | [TK]D-Fender | hrolf: that's the default... or you can specify "username" and "defaultuser" |
| 14:40.39 | hrolf | [TK]D-Fender: Okay, for password, I use secret=? or remotesecret=? |
| 14:41.14 | [TK]D-Fender | secret |
| 14:41.17 | [TK]D-Fender | ~book |
| 14:41.17 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:41.19 | [TK]D-Fender | ^^^ |
| 14:41.26 | [TK]D-Fender | You should read up on your basics |
| 14:42.32 | hrolf | [TK]D-Fender: Yes, thanks. I'll look into it. (was in emergency, but yes I should do my homework.) |
| 14:47.10 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gwewlugcqtuouygm) |
| 14:47.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:55.47 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:02.57 | *** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru) |
| 15:02.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:05.12 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:05.21 | cusco | I was testing sipvicious last night, and noticed failed attempts on asterisk cli do not show remote ip, only local |
| 15:05.22 | *** join/#asterisk vlad_sta_ (~vlad_star@nat.canmos.ru) |
| 15:06.16 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:06.16 | hjf | i was playing with csipsimple and i noticed it has an option to send text messages |
| 15:06.29 | *** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru) |
| 15:06.31 | hjf | how are these messages delivered? is there some sort of SMS for voip?? |
| 15:06.56 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 15:07.30 | [TK]D-Fender | hjf: via SIP obviously |
| 15:08.17 | cusco | talking about that, the peer needs to be in a call to recieve the message? |
| 15:08.51 | hjf | [TK]D-Fender: yes lol. |
| 15:09.04 | [TK]D-Fender | cusco: No |
| 15:09.04 | hjf | oh |
| 15:09.06 | hjf | MESSAGE sip:6@10.42.42.35 SIP/2.0 |
| 15:09.12 | coppice | its SIMPLE really |
| 15:09.13 | hjf | SIP/2.0 405 Method Not Allowed |
| 15:09.52 | [TK]D-Fender | hjf: and not every version of * supports it |
| 15:10.11 | hjf | [TK]D-Fender: yeah i'm trying to find out. seems 1.4 needs a module for it |
| 15:10.45 | [TK]D-Fender | hjf: 1.4 is decrepit crap |
| 15:10.53 | hjf | yes |
| 15:10.56 | [TK]D-Fender | hjf: SIP Message support was added in * 10 |
| 15:10.56 | hjf | i'm using 1.8 |
| 15:11.10 | [TK]D-Fender | hjf: Upgrade time |
| 15:13.01 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 15:14.05 | hjf | [TK]D-Fender: yeah i used freebsd's ports asterisk18 thinking it was newer than asterisk10 |
| 15:14.35 | [TK]D-Fender | 10 is dead anyway. 11 is current |
| 15:14.46 | hjf | yes, i'm installing port asterisk11 |
| 15:14.53 | hjf | 11.5.0 |
| 15:14.55 | hrolf | I tried this register => 6001:123456@192.168.0.231 |
| 15:14.58 | hrolf | but it doesn't register |
| 15:15.07 | hrolf | and when I dial 6001 to receive call on asterisk it is busy |
| 15:15.25 | hrolf | but I can receive on 3cx or x-lite, which means asterisk is not registering, any ideas? |
| 15:15.28 | [TK]D-Fender | hrolf: "sip set debug on" <- look at what is actually happening. |
| 15:16.04 | [TK]D-Fender | hrolf: Idea : LOOK at the actual comms. You could have put lines in wrong places or messed up other settings. So go look at what * is actually doing |
| 15:16.06 | [TK]D-Fender | ~pb |
| 15:16.06 | 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. |
| 15:16.08 | [TK]D-Fender | ^^^^ |
| 15:16.19 | [TK]D-Fender | And PASTEBIN the results. |
| 15:16.33 | hrolf | [TK]D-Fender: Okay, but how do I make it attempt a register, in order to see what is happening. i.e. sip set debug on |
| 15:17.22 | [TK]D-Fender | [11:14]hrolfI tried this register => 6001:123456@192.168.0.231 <- |
| 15:18.51 | Katty | looks in |
| 15:18.58 | Katty | hi kids. |
| 15:19.05 | cusco | hi grown up |
| 15:20.06 | hrolf | [TK]D-Fender: http://pastebin.com/fpzbkxJz |
| 15:20.13 | hrolf | [TK]D-Fender: I have set sip set debug on |
| 15:20.20 | hrolf | [TK]D-Fender: There is no output, nothing. |
| 15:21.27 | [TK]D-Fender | hrolf: register => 6001:123456@192.168.0.231:/9000 <- this line has to be AFTER [general] and BEFORE any other section entry. Also the last ":" in there is bad |
| 15:21.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:22.27 | hrolf | [TK]D-Fender: Okay, sorry. Actually, what I'm trying to achieve by the :/9000 is that, it lands on extension '9000' in context 'access'. |
| 15:22.48 | [TK]D-Fender | hrthe last ":" is not valid |
| 15:29.35 | *** join/#asterisk vlad_sta_ (~vlad_star@nat.canmos.ru) |
| 15:34.26 | hrolf | [TK]D-Fender: Okay, it registered. But now when I call, it says Unauthorized. |
| 15:34.31 | hrolf | [TK]D-Fender: http://pastebin.com/AXKXPtYD |
| 15:35.00 | hrolf | [TK]D-Fender: Line 167. |
| 15:36.05 | [TK]D-Fender | hrolf: remove "remotesecret" which I told you you didn't need, and add "insecure=port,invite" |
| 15:37.38 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:40.14 | hrolf | [TK]D-Fender: Thanks. I received the call now. |
| 15:40.23 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:43.25 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:46.07 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:48.49 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:48.54 | *** join/#asterisk Alex_h (~AlexHold@178.78.119.76) |
| 15:49.25 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:49.25 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:49.52 | Alex_h | is there any way of showing the channels that have been up the longest on the asterisk cli? I know that core show channel will show an elapsed time for a specific channel, but is there any way of ordering this or showing the channel that has been up the longest? Im dealing with local channels, not calls specifically |
| 15:50.56 | hrolf | I get this error when playing a .wav file channel.c: Unable to find a codec translation path from 0x1 (g723) to 0x4 (ulaw |
| 15:51.10 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:51.18 | hrolf | file.c: Unable to open /var/lib/asterisk/sounds/access/Welcome.wav (format 0x1 (g723)): No such file or directory |
| 15:51.29 | [TK]D-Fender | Alex_h: No. You have to look one by one |
| 15:51.44 | [TK]D-Fender | hrolf: because * cannot transcode to G.723. |
| 15:51.59 | [TK]D-Fender | hrolf: Only via an expensive transcoder hardware card |
| 15:52.07 | [TK]D-Fender | hrolf: Pick your codecs better |
| 15:52.20 | hrolf | [TK]D-Fender: At which side? IP Office or Asterisk? |
| 15:52.20 | coppice | or wait until 2014 |
| 15:52.27 | [TK]D-Fender | hrolf: Asterisk |
| 15:52.39 | [TK]D-Fender | hrolf: You didn't set what * is allowed to agree with for your peer |
| 15:53.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:54.52 | hrolf | [TK]D-Fender: I have set allow=all |
| 15:55.00 | SuperNull | everyone having a good friday? |
| 15:55.10 | igcewieling | hrolf: then you almost guarantee to make it not work, |
| 15:55.35 | [TK]D-Fender | [11:54]hrolf[TK]D-Fender: I have set allow=all L- which is BAD |
| 15:55.54 | [TK]D-Fender | hrolf: because you are allowing * to agree to anything the other side offer.. including things it CAN'T support |
| 15:56.13 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:56.17 | [TK]D-Fender | hrolf: Go specify a STANDARD non-licensed codec that the other end can support |
| 15:56.28 | igcewieling | Asterisk needs an option which means "allow all codecs this system can transcode" |
| 15:56.33 | [TK]D-Fender | hrolf: You'll see what they offer in the SIP DEBUG |
| 15:57.05 | hrolf | igcewieling: [TK]D-Fender: Thanks, I'll check SIP DEBUG, it is in INVITE right? |
| 15:57.13 | [TK]D-Fender | hrolf: yes |
| 15:58.36 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe70:fce6) |
| 15:59.35 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:59.52 | hrolf | [TK]D-Fender: Thanks. |
| 15:59.56 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 16:00.07 | Alex_h | [TK]D-Fender: thanks again for your help |
| 16:04.17 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
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| 16:43.46 | *** join/#asterisk nny (~Scott@cpe-024-088-116-144.sc.res.rr.com) |
| 16:45.24 | nny | Quick question, I have a system with 4 FXO ports (Sangoma AFT B600) and the client wants to add four more. Can you put two PCI cards in a system or is it better to just have them buy a new 8 port card? |
| 16:46.01 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:46.05 | [TK]D-Fender | nny: 2 will be fine |
| 16:46.44 | nny | [TK]D-Fender: thank you |
| 16:46.45 | *** join/#asterisk ncrollo (~nolhay@d233-64-78-227.dim.wideopenwest.com) |
| 16:46.50 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 16:48.19 | igcewieling | nny: you can't add modules to that card? |
| 16:48.20 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 16:48.43 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:48.55 | [TK]D-Fender | igcewieling: Nope, 4fxo, 1 fxs fixed |
| 16:48.58 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 16:51.47 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:52.43 | igcewieling | ' |
| 16:53.02 | igcewieling | Has anyone used the Sangoma GSM cards? If so, how did they work out for you? |
| 16:53.50 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 16:53.59 | hrolf | Is there any way to pass named paramters to AGI() command? |
| 16:55.05 | [TK]D-Fender | hrolf: AGi has full access to asterisk channel variables as it is... there's very little point to "passing" anything |
| 16:55.28 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:55.59 | igcewieling | AGI(/path/to/hacknsa.php,mode=agressive,level=global) |
| 16:56.43 | igcewieling | [TK]D-Fender: best reason I can think of is so you can have one script for CLI and AGI without the hassle of using different options passing methods |
| 16:57.57 | igcewieling | BRB, someone's at the door. |
| 16:57.58 | igcewieling | 8-) |
| 16:58.08 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:00.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:04.39 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:05.28 | Penguin | This is ridiculous. |
| 17:05.59 | Penguin | file, malcolmd, mjordan, pabelanger, putnopvut, qwell, sruffell |
| 17:06.34 | malcolmd | ? |
| 17:06.54 | sruffell | ditto |
| 17:07.02 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:07.06 | Penguin | /b italorossi |
| 17:07.12 | Greenlight | +1 |
| 17:07.18 | Greenlight | It's every day |
| 17:07.32 | Penguin | Constantly in and out all day and night, never says a word. |
| 17:07.55 | sruffell | Ahh…I'm configured to not show the leave/join messages. |
| 17:08.16 | igcewieling | same here, no join/part messages |
| 17:09.29 | *** mode/#asterisk [+b italrossi!*@*] by malcolmd |
| 17:09.37 | sruffell | bam! |
| 17:09.39 | malcolmd | d'oh.. |
| 17:09.49 | Greenlight | Ty :) |
| 17:09.55 | *** mode/#asterisk [+b italorossi!*@*] by malcolmd |
| 17:09.58 | Penguin | Thanks! |
| 17:10.08 | *** mode/#asterisk [-b italrossi!*@*] by malcolmd |
| 17:10.16 | malcolmd | removing the wrong one... |
| 17:11.12 | Penguin | Here's a sample for those who had to luxury of missing it: http://pastebin.com/FGLKT89S |
| 17:11.51 | Penguin | the luxury, rather. |
| 17:12.18 | malcolmd | viewing that link will void any perception of luxury that you had before viewing the link. ;) |
| 17:12.34 | igcewieling | Penguin: Here's a nickle, kid. Go get yourself a real irc client. 8-) |
| 17:13.00 | Penguin | I'm not sure how to respond to that. |
| 17:13.09 | sruffell | "Gee Whiz! Thanks Mister!" I think is the only response. |
| 17:13.15 | igcewieling | that works! |
| 17:13.27 | igcewieling | admits to not using a real irc client either. |
| 17:14.30 | Penguin | Can you give a few examples which fit your definition of real? |
| 17:15.07 | _Corey_ | Unless you're using telnet to connect to IRC and interacting with it raw, I'd wager you're using some kind of "real" client. :) |
| 17:15.08 | igcewieling | Penguin: something which lets you not see join/parts and has persistent /ignore's |
| 17:15.08 | Nugget | telnet is eeeeeeevil! |
| 17:16.22 | *** join/#asterisk luce (~luce@NYUFWA-NYUSHANET-01.NATPOOL.NYU.EDU) |
| 17:18.01 | luce | hey all - I'm getting this error and not sure where to change the sample rate - chan_alsa.c:209 alsa_card_init: Rate not correct, requested 8000, got 16000 - any suggestions? |
| 17:19.07 | Penguin | I once used ii to get on IRC. That was pretty interesting. |
| 17:20.35 | Penguin | When I used telnet for IRC, I got tired of having to send PONGs all the time. |
| 17:21.33 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 17:24.39 | navaismo | igcewieling, I use XChat and I prefer Xcaht over Pidgin |
| 17:25.10 | Penguin | I've used irssi for years and will continue to use it for many more. |
| 17:25.31 | igcewieling | navaismo I may go back to Xchat |
| 17:26.34 | ChannelZ-Wk | luce: presumably in your alsa config. Although I thought you could add something to it to resample everything to a certain rate |
| 17:27.36 | luce | the alas config is the asound.conf file…i've adjusted those settings tremendously but still the same error |
| 17:30.33 | ChannelZ-Wk | defaults.pcm.rate_converter ? |
| 17:31.20 | hjf | hehe nice i updated to asterisk 11 and i have sms now |
| 17:31.31 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 17:32.19 | [TK]D-Fender | hjf: "SIP Messaging", not "SMS" |
| 17:33.12 | hjf | yes that |
| 17:37.57 | igcewieling | [TK]D-Fender: I wonder if he has sip trunks too |
| 17:42.46 | ncrollo | this is going to sound dumb but can you do TLS with anything other than a self signed certificate? |
| 17:43.03 | ncrollo | I'm assuming the answer is yes but I'm having a really difficult time |
| 17:43.45 | ChannelZ-Wk | as in you have a properly signed cert that isn't working? |
| 17:43.58 | ncrollo | yeah...its a godaddy wildcard |
| 17:48.28 | jeev | anyone here have two locations connected to eachother using trunks and also passing callerid's name ? i'm looking at the set up and it doesn't seem to be passing name |
| 17:48.48 | Penguin | Of course we do. |
| 17:48.55 | Penguin | How did you set the name? |
| 17:50.03 | jeev | i call into office A, office A answers the call, they see the caller id prefix, they see the phone number and they see the caller id name information. when they transfer a call from office A to me in office B, it sends the caller ID over but does not include the name. |
| 17:50.25 | jeev | unfortunately i know you guys dont support freepbx but i'm looking at it and its passing it through as a trunk |
| 17:50.30 | igcewieling | jeev: what DOES the callerid include? |
| 17:50.36 | Penguin | And the fact that it is an IAX2 trunk isn't really important. |
| 17:50.56 | jeev | Executing [s@macro-dialout-trunk:20] ExecIf("DAHDI/i1/2139251133-af9", "0?Set(CONNECTEDLINE(name,i)=CID:2139251133)") in new stack |
| 17:51.00 | Penguin | Trunked or not, the callerid data goes the same way. |
| 17:51.38 | jeev | this is from office A |
| 17:51.40 | jeev | http://pastebin.ca/2444533 |
| 17:51.50 | igcewieling | if you are using SIP you can use PAID/RPID to pass the callerid, then you can auth by username/passwotd |
| 17:51.53 | Penguin | (1248.55) <Penguin> How did you set the name? |
| 17:51.59 | Penguin | Still waiting on the answer to this. |
| 17:52.05 | hjf | is the SPA3000 a decent FXO gateway for my asterisk? |
| 17:52.46 | Penguin | You want to plug some phone lines into it so you can use asterisk on the PSTN? |
| 17:52.52 | [TK]D-Fender | hjf: Wouldn't use for business purposes but it's OK |
| 17:52.53 | hjf | yes |
| 17:52.56 | igcewieling | jeev: you see the 0? "] ExecIf("DAHDI/i1/2139251133-af9", "0?Set(CONNECTEDLINE(name,i)=" that means THE LINE DID NOT EXECUTE |
| 17:52.59 | hjf | no, for my house |
| 17:53.20 | hjf | i'll still run the DECT phones alongside asterisk |
| 17:53.36 | hjf | that is, in parallel (that's why i was asking yesterday if it will grab the line, i don't want it to) |
| 17:53.50 | Penguin | jeev: You're setting the CALLERID(all) value to only the number. That should remove the name. |
| 17:54.39 | jeev | this is how the call comes into B: Executing [s@macro-user-callerid:4] ExecIf("IAX2/officeA-13664", "1?Set(REALCALLERIDNUM=213...)") in new stack |
| 17:54.43 | jeev | ok Penguin, hm. |
| 17:54.58 | ChannelZ-Wk | ncrollo: do you have your cert and the key for it in the same file? |
| 17:55.02 | Penguin | CALLERID(all) should be name and number. But you don't even need to REset the callerid info anyway. Leave it alone and it should be fine. |
| 17:55.53 | igcewieling | freepbx and iax2 too kinky for me, you are on your own |
| 17:58.27 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:03.48 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 18:04.24 | jeev | yea, im looking at it and i can't see where its grabbing the name, all just means ${REALCALLERIDNUM}) i guess |
| 18:04.47 | Penguin | all means NAME and NUMBER. |
| 18:05.18 | Penguin | e.g., CALLERID(all)=Your Name <5551212> |
| 18:05.19 | *** join/#asterisk dongola7 (~dongola7@unaffiliated/blair/x-0911782) |
| 18:05.32 | jeev | i understand but in extensions_additional, it's being set by something else. |
| 18:05.45 | Penguin | There's num, which is just the number. There's name, which is just the name. There's all, which is name and number together. |
| 18:06.40 | jeev | exten => s,n(usercid),ExecIf($[${LEN(${USEROUTCID})} != 0]?Set(CALLERID(all)=${USEROUTCID})) |
| 18:06.52 | Penguin | If you set CALLERID(all) to just a number, expecting the name to be deleted makes sense to me. |
| 18:07.02 | jeev | that's what it looks like |
| 18:07.15 | jeev | is that happening because it considers it a trunk? |
| 18:07.30 | Penguin | No. Trunking isn't pertinent. |
| 18:07.42 | Penguin | Unless... |
| 18:08.17 | Penguin | If the rest of your dial plan is configured to explicitly leave off the name because you are sending it over DAHDI, then that would also make sense. |
| 18:08.41 | Penguin | You can't pass the name over the PSTN, so it doesn't need to be sent at all. |
| 18:09.10 | Penguin | But I don't debug freepbx dial plans. |
| 18:09.10 | jeev | so could this be fixed with a context? |
| 18:09.16 | jeev | a different context |
| 18:09.27 | Penguin | Context doesn't matter either. |
| 18:09.35 | jeev | oh |
| 18:09.55 | Penguin | You'll have to check the entire dial plan to see why the name is missing. |
| 18:10.10 | Penguin | At some point, the name is being left out. |
| 18:11.44 | *** join/#asterisk Linkforsoad (~Linkforso@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 18:21.02 | igcewieling | jeev: now you understand this.... |
| 18:21.04 | igcewieling | ~freepbx |
| 18:21.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:21.19 | jeev | yea i know |
| 18:24.55 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:27.34 | boom^time | So I'm trying to play a message after an answering machine by waiting for a certain amount of silence. But when I do 3 seconds it's too little most of the time and 4 seconds makes for a really long awkward pause. Does anyone have a more clever solution? |
| 18:27.48 | *** join/#asterisk maelaian (~alexis@38.96.32.242) |
| 18:28.13 | _Corey_ | boom^time: Check out the WaitForSilence command |
| 18:28.21 | boom^time | That's what I'm using |
| 18:28.37 | maelaian | I am looking for information on the name and specification of the protocol used in the norstar pbx M8X24-DS ( phones it uses are M7310 / NT8B20 ). I heard asterisk has some kind of compatibility with the system. |
| 18:28.43 | boom^time | WaitForSilence(3000,1,30) is a little too short some times |
| 18:28.49 | Penguin | I wait for 2000 ms if AMD() thinks it's a machine. |
| 18:28.53 | boom^time | and anything more than that makes for a really awkward pause before the message starts |
| 18:29.52 | _Corey_ | boom^time: It takes a lot of adjustment |
| 18:30.00 | Penguin | 500 ms if AMD() detects a human, 2000 ms if machine. |
| 18:30.22 | Penguin | Maybe your AMD needs tweaking. |
| 18:30.25 | boom^time | I'm testing against my cell and with t-mobile there is a very long delay between the end of your greeting and the beep |
| 18:30.32 | boom^time | Penguin, has nothing to do with AMD |
| 18:30.40 | Penguin | hmm |
| 18:30.49 | boom^time | it detects the machine properly, but the wait for silence portion is the issue. |
| 18:31.19 | Penguin | Why would 2 full seconds not be plenty of time to wait? |
| 18:31.24 | boom^time | it's like <your greeting><3 seconds of silence><beep> |
| 18:31.35 | boom^time | because it is triggered before the beep. |
| 18:31.36 | Penguin | Well that's just weird. |
| 18:31.39 | boom^time | agreed |
| 18:31.47 | Penguin | That part isn't YOUR problem... |
| 18:31.58 | Penguin | People shouldn't wait so long before ending their outgoing message. |
| 18:32.04 | *** join/#asterisk shailender123 (73f90d52@gateway/web/freenode/ip.115.249.13.82) |
| 18:32.23 | boom^time | That's the humorous part, this is MY outgoing message on my phone. I've tried ending the recording the instant I'm done talking |
| 18:32.39 | Penguin | T-mobile adds that much time, eh? |
| 18:33.13 | shailender123 | I have just installed asterisk on my CentOS system and unable to search good/usable softphone, please help me in which softphone i can directly install with the rpm instead of compilation |
| 18:33.14 | boom^time | just timed it, exactly 3 secs |
| 18:33.22 | Penguin | That's crazy stuff. |
| 18:34.00 | Penguin | shailender123: If you just put asterisk on your CentOS system, you don't really need a softphone on your CentOS system. |
| 18:34.43 | shailender123 | Penguin, how i will test dial plan, i will test the conversation |
| 18:34.47 | Penguin | That's like installing a browser on your web server and using your server for surfing your own site. |
| 18:35.56 | shailender123 | i don't have other machine so i want to test all softphone after assigning different virtual ip to same machine |
| 18:35.59 | *** join/#asterisk RZero (~androirc@host31-54-11-186.range31-54.btcentralplus.com) |
| 18:36.11 | Penguin | If you must install a soft phone on the asterisk box, use something that allows you to easily change the client ports. You won't be able to run your phone's listening port on 5060 since asterisk is already listening on that port. |
| 18:36.23 | Penguin | ~softphones |
| 18:36.48 | Penguin | *shrug* I don't remember what keyword infobot knows for the list of phones. |
| 18:37.13 | Penguin | twinkle, blink, zoiper... to name three |
| 18:37.24 | shailender123 | i already tried qutecom(ffmpeg error,unable to find concerned dependecny), xlite,ziper,takisrc, but none of them are usable for me |
| 18:37.44 | maelaian | So meridian norstar M8X24... No details on its wire communication protocol or anything? |
| 18:39.08 | [TK]D-Fender | maelaian: It's documented. and is a 2-wire ISDN derivative |
| 18:39.26 | Penguin | If every soft phone that you try doesn't work with your asterisk, did you consider that your asterisk may be the problem and not the phones being the problem? |
| 18:39.29 | maelaian | Does it have a name so I can find more info? |
| 18:39.56 | igcewieling | maelaian: what version of Asterisk are you using? |
| 18:40.08 | maelaian | igcewieling, None yet. |
| 18:40.09 | shailender123 | currently unable to install softphone after installation i can test the settings, asterisk installation looks good as i am able to see sip show peers etc |
| 18:40.21 | boom^time | Penguin, Is there any nifty applications that will listen for a break in silence during a background or playback and I can restart the playback if it happens? |
| 18:40.31 | Penguin | Yes. |
| 18:40.47 | boom^time | Neat! |
| 18:41.03 | Penguin | BackgroundDetect() |
| 18:41.43 | *** join/#asterisk dimitry7 (~antonello@187.174.147.162) |
| 18:41.45 | Penguin | There's also SpeechBackground(), but I haven't used it to know how it works. |
| 18:41.47 | maelaian | Trying to get into asterisk. It has a formidable learning curve. All the phone PBX systems do. |
| 18:42.11 | Penguin | ~book |
| 18:42.12 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:43.07 | [TK]D-Fender | maelaian: I've seen some particularly sharp people getin a system up and running with a few softphones, an ITSP and basic voicemail within a day. It will depend on your insight, ability to read documentation, and ability to communicate in channels like this |
| 18:43.20 | shailender123 | checking for ccgnu2-config... no checking for commoncpp2 version >= 1.6.0... not found *** The ccgnu2-config script installed by commoncpp2 0.99 *** or later could not be found. *** You need to install GNU Common C++ 2, whose later releases are *** available from http://www.gnu.org/software/commoncpp/ and any *** GNU mirror. |
| 18:43.28 | shailender123 | i got this error during compilation of twinkle |
| 18:43.36 | jeev | penguin, i got it to work. hm |
| 18:44.05 | maelaian | [TK]D-Fender, Sure, but they probably didn't understand as much as I would want to about the system, and its capabilities. I have no itch to scratch today, Im more interested in the possibilities. |
| 18:44.18 | igcewieling | shailender123: this is not a softphone support channel. |
| 18:44.34 | [TK]D-Fender | maelaian: possibilities are pretty huge. |
| 18:44.56 | shailender123 | yes, but i need some softphone to test my newly configured asterisk system |
| 18:45.13 | maelaian | [TK]D-Fender, I know. I wanted to figure out for now what my existing system speaks under the hood, see what I can do with that, and then work on an asterisk interface/enhancement to it. |
| 18:45.31 | igcewieling | shailender123: no, you don't. you could use a hardphone or install a softphone on a windows box or signup for an itsp -- and this is not a channel to support your softphone |
| 18:45.35 | Penguin | Use yum to solve dependencies rather than installing one rpm after another. |
| 18:46.15 | shailender123 | i don't have connectivity to internet in my vm, Ok i will try to configure internet access within my vm |
| 18:46.21 | [TK]D-Fender | maelaian: Nortel propriety gear is practically worthless. |
| 18:46.21 | [TK]D-Fender | maelaian: All we've got are a few gateway devices that will let you use Norstar sets as SIP devices. |
| 18:46.22 | *** join/#asterisk adam820 (~t3hrealad@pool-108-39-247-43.pitbpa.fios.verizon.net) |
| 18:46.57 | [TK]D-Fender | maelaian: They won't be as nice as dedicated actual SIP phones for instance however and the price/port ratio means you would probably only do this where you're restricted from changing wiring, etc |
| 18:47.02 | igcewieling | if the time to set up an Asterisk PBX is X, then the time to build a FrankenPBX with Asteirsk and your Nortel is X^2 (at least) |
| 18:47.12 | maelaian | [TK]D-Fender, It's not worthless :). Maybe from an asterisk perspective. |
| 18:47.45 | [TK]D-Fender | maelaian: practically worthless in almost all senses :) |
| 18:48.00 | [TK]D-Fender | maelaian: Norstar is a very dated and dead duck |
| 18:48.13 | maelaian | [TK]D-Fender, Our business runs on such a system, and has for 15 years. |
| 18:48.20 | igcewieling | maelaian: *nod* Looks to be worth $100 according to ebay |
| 18:48.33 | [TK]D-Fender | maelaian: Sorry, CARBON-dated, my bad... |
| 18:49.36 | maelaian | Well, i'm not exactly looking for a plug and play solution/answer. I can write software, and though I wouldnt consider myself an EE, I can manufacture and design my own pcbs. |
| 18:49.54 | maelaian | So if the possibility of an interface is even there, I am interested in exploring that. |
| 18:50.10 | igcewieling | maelaian: All that is trivial compared to getting Nortel to document their protocol for you |
| 18:50.34 | maelaian | igcewieling, I can figure out the protocol myself. Getting a jump start on its framing or basics wouldnt be bad. |
| 18:50.48 | [TK]D-Fender | maelaian: Well you said you don't currently even have an itch to scratch.... so this seems like the worst direction to put efforts towards... |
| 18:50.50 | Penguin | If the PBX has FXO ports on it, I guess that could be used as an interface to asterisk. |
| 18:51.23 | maelaian | Especially seeing as how CITEL probably has it figured out, and this system is very popular.. I would imagine there is some documentation atleast. |
| 18:51.25 | Penguin | I wouldn't want to do it, but you could put asterisk between the PBX and the PSTN. |
| 18:51.40 | maelaian | [TK]D-Fender, well, it elleviates my boredom. |
| 18:51.49 | [TK]D-Fender | maelaian: Whatever floats your boat I guess.... |
| 18:51.51 | maelaian | Penguin, Yea I was reading about such a setup with an identical pbx unit. |
| 18:51.54 | RZero | Just a quick question. Is anyone aware of any issues between kamailio 3.3 and ast 1.8 ? |
| 18:52.13 | maelaian | [TK]D-Fender, displacement floats my boat. |
| 18:52.59 | igcewieling | maelaian: you'll get little help from here. |
| 18:53.26 | maelaian | kk, figured if you interfaced with other pbxs may be known. But the ISDN tip is helpful. |
| 18:53.27 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:54.19 | maelaian | I can hookup my logic analyzer to the line and see what the deal is. |
| 18:55.33 | igcewieling | people generally don't interface Asterisk with other PBXs. |
| 18:55.57 | Penguin | Generally, that's true, but they do do it. |
| 18:56.12 | maelaian | I saw quite a few mailing list topics on it, my exact unit, so I figured id ask the question. |
| 18:56.30 | igcewieling | Penguin: I call those people "the crazies" 8-) |
| 18:56.35 | Penguin | hehe |
| 18:57.15 | igcewieling | The last Nortel I tried that with would have required $4,000 worth of license keys just to allow ISDN PRI |
| 18:57.45 | igcewieling | sorry ISDN PRI as a tandem style connection rather than a PSTN type connection |
| 18:58.29 | igcewieling | If you go with analog you'll often get "stuck" channels/calls |
| 18:58.54 | shailender123 | WARNING[1949]: chan_skinny.c:6888 get_input: Skinny Client sent less data than expected. Expected 4 but got 0 |
| 18:59.21 | shailender123 | Penguin - now i am trying to connect from my windows softphone but connection is failing |
| 18:59.32 | Penguin | why? |
| 18:59.56 | shailender123 | tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 1808/asterisk |
| 19:00.23 | shailender123 | i am connecting on 2000 and in which file i can bind asterisk port to 5060 with which parameter |
| 19:01.16 | shailender123 | do i need to use SIP domain/realm ( I am using Qutecom) |
| 19:01.31 | igcewieling | shailender123: Unless you are using some weird Cisco softphone you are NOT using the Skinny protocol. |
| 19:02.20 | *** part/#asterisk nny (~Scott@cpe-024-088-116-144.sc.res.rr.com) |
| 19:02.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.114) |
| 19:03.04 | igcewieling | shailender123: perhaps you should start by reading the Asterisk Book. |
| 19:03.06 | igcewieling | ~book |
| 19:03.06 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:14.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.114) |
| 19:15.11 | boom^time | Found a small bug the BackgroundDetect app doesn't support chaining audio files, ie en/silence/2&imacowboy/yeehaw/howdy_partner |
| 19:17.50 | Penguin | That's interesting. |
| 19:19.22 | boom^time | It seems to be really lagged as well. |
| 19:19.50 | boom^time | Meaning, if I say something to a backgrounddetect playback it will be like 2-3 seconds before it grabs the talk extension |
| 19:21.42 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 19:23.44 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 19:24.22 | Penguin | That's pretty lousy. |
| 19:25.24 | boom^time | No I think that last part is my fault. |
| 19:25.37 | boom^time | misunderstand what the sil argument was doing |
| 19:26.23 | Penguin | It's a little confusing at first. |
| 19:27.40 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 19:28.16 | pensmit | Anyone seen this /res/pjproject/version.mak: No such file or directory when doing a make install after menuselect? |
| 19:35.19 | ChannelZ-Wk | Something probably failed in your configure that you didn't notice, I'd guess. |
| 19:36.08 | ChannelZ-Wk | Do you have uuid-dev installed? I don't remember what the failure looked like when that's not on, besides chan_sip not being selectable to build |
| 19:42.16 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 19:42.20 | igcewieling | ChannelZ-Wk I think file fixed that issue. |
| 19:42.47 | *** join/#asterisk CeBe (~CeBe@port-92-206-44-68.dynamic.qsc.de) |
| 19:43.23 | pensmit | I redownloaded asterisk and went back through the compilation and it seems to be working now. |
| 19:43.27 | pensmit | Not sure what happened. |
| 19:45.34 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 19:51.09 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 20:01.36 | igcewieling | Every time I see this in the Polycom phone firmware release notes I laugh "84975/84908 In the locked state, the phone no longer initiates the call to the emergency 911 upon off-hooking the handset twice in quick succession, i.e., 2 seconds" |
| 20:16.29 | navaismo | i never get attached to polycom phone's they boot like a 150 old year man run. I hate them a lot, but the quality of voice was nice. |
| 20:17.45 | Penguin | How often do you really need to reboot phones, though? |
| 20:18.51 | Penguin | If you only have to boot them up once every 3-8 months, does it really matter if it takes four minutes to boot up? |
| 20:19.41 | Penguin | Same for computers. How often do you reboot a computer? I vote for never, or at least as infrequently as possible. |
| 20:20.01 | Penguin | Doesn't matter if it takes five minutes to start up once every couple years. |
| 20:20.21 | igcewieling | navaismo: modern firmwares boot much faster and do not require a reboot for most config changes |
| 20:24.21 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.16 | *** join/#asterisk Ice_Strike (~Ice_Black@84.92.51.164) |
| 20:28.29 | luce | if I use the dial(console/alsa) command, how can I make sure the dial plan continues after the call is ended? |
| 20:28.38 | luce | right now it just hangs out in the console |
| 20:33.27 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 20:33.31 | navaismo | Penguin, true about that |
| 20:33.53 | navaismo | igcewieling, since that time i never go back with polycom |
| 20:34.13 | j4jackj | Riddiddle. |
| 20:34.13 | navaismo | opted for aastra and others brands |
| 20:34.41 | boom^time | I have a cisco 7940 that has a horrible power cord. If I bump it the wrong way it'l reboot. |
| 20:35.06 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 20:35.17 | boom^time | It takes like 5 mins to boot. It can be irritating when you really needed to answer that call and now you assume they are trying to call back but can't. |
| 20:35.32 | boom^time | So POE to the rescue. |
| 20:35.41 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 20:37.32 | igcewieling | or swap the phone with someone in the company authorized to spend the money to replace the phone |
| 20:37.36 | j4jackj | :D:D:D:D:D:D:DDDD::D::D:D::D::D::D::D::D:D:::D:D |
| 20:37.44 | boom^time | It's my phone :\ |
| 20:38.04 | navaismo | boom^time, i hate cisco 79XX phones |
| 20:38.31 | navaismo | in fact i hate everyone |
| 20:38.48 | boom^time | navaismo, phones aren't people. |
| 20:39.34 | navaismo | ok, i hate all in this world(except my wife & childs) |
| 20:39.46 | Penguin | boom^time: Are you using SIP on the 7940? |
| 20:39.50 | boom^time | Yes |
| 20:39.51 | navaismo | and the ice cream everybody loves ice cream |
| 20:40.21 | Penguin | If you don't have a tftpd to offer up the configs, it'll take a whole lot longer to start up. |
| 20:40.48 | igcewieling | navaismo except those who are lactose intollerant |
| 20:40.58 | boom^time | Didn't realize that was the reason. I set one up for a provision before but I turned it down. |
| 20:41.03 | boom^time | After the upgrade |
| 20:41.07 | navaismo | igcewieling, yogurt ice cream |
| 20:41.37 | Penguin | It'll take about three minutes without a tftpd or about 15 seconds with one. |
| 20:41.53 | boom^time | Good to know. |
| 20:41.59 | Penguin | Without tftpd, it has to wait for the timeout before it falls back to stored configs. |
| 20:42.26 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 20:43.19 | navaismo | going for some ice cream |
| 20:47.15 | hjf | is there a way to set a default caller ID per peer? i'm sending SIP messages and they show up as "Unknown" sender |
| 20:47.38 | hjf | if i go into CSipSimple and set up a caller id for me, then i see the name i set |
| 20:47.52 | navaismo | set(CALLERID(num||name)) |
| 20:48.02 | navaismo | or in the peer settings |
| 20:48.37 | hjf | navaismo: what's the peer setting for this? |
| 20:48.51 | hjf | i'd like to have a default caller ID for the peer and let the user override it in his client if he wants |
| 20:48.52 | igcewieling | hjf: try reading sip.conf.sample sometime. |
| 20:50.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:50.11 | igcewieling | specifically like 1381 (at least in my sip.conf.sample) |
| 20:50.17 | igcewieling | s/like/line/ |
| 20:52.03 | navaismo | never sorted the sip messaging using sipml5 & asterisk |
| 20:52.27 | hjf | http://highsecurity.blogspot.com.ar/2012/03/asterisk-10-110-sms-messaging-or-sip.html |
| 20:53.35 | hjf | ${MESSAGE(from)} |
| 20:54.02 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 20:54.05 | navaismo | is that for me or a question? |
| 20:54.34 | hjf | 1 sec |
| 20:55.09 | igcewieling | navaismo: yes! |
| 20:55.11 | igcewieling | 8-| |
| 20:55.33 | hjf | ah.. it's the stupid csipsimple setting the "Unknown" name |
| 20:55.36 | navaismo | hahahahaha |
| 20:55.37 | hjf | From: "tablet" <sip:tablet@10.42.42.35> |
| 20:55.47 | navaismo | igcewieling, +100 |
| 20:55.48 | hjf | From: "Unknown" <sip:hjf@10.42.42.35> |
| 20:55.51 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:57.35 | navaismo | i can't get this--->http://xkcd.com/1216/ |
| 20:58.10 | *** join/#asterisk wm_domino (~William@24-107-186-9.dhcp.stls.mo.charter.com) |
| 20:58.19 | hjf | no wait, thats from the asterisk... f: <sip:hjf@10.42.42.35> vs From: "tablet" <sip:tablet@10.42.42.35> |
| 20:59.54 | hjf | peer tablet has a "caller id" setting, set to "tablet" |
| 20:59.58 | hjf | while hjf's is blank |
| 21:00.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:00.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:00.31 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-159-94.user.veloxzone.com.br) |
| 21:01.20 | hjf | i'm trying to understand where will asterisk try to pick up the name or default to Unkown |
| 21:01.20 | anonymouz666 | anyone already switched from 1.8 to ast 11? too much trouble? I know there is a file that shows what changes from version to version |
| 21:01.46 | hjf | anonymouz666: i just did today. then started asterisk -f and it spits out all warnings and errors to the screen |
| 21:02.19 | anonymouz666 | dialplan errors? |
| 21:02.25 | anonymouz666 | syntax I mean |
| 21:02.38 | navaismo | anonymouz666, many people did that already, you need to read the changelog. And about the "new" nat settings for sip |
| 21:02.55 | anonymouz666 | nat settings is OK |
| 21:03.16 | anonymouz666 | changes already read it, just wanted to know in the practise from who already did the upgrade |
| 21:04.51 | anonymouz666 | from 1.8 to 11 it seems is easier than it was from 1.4 to 1.8 |
| 21:05.47 | anonymouz666 | hjf: what kind of errors did you have? |
| 21:06.21 | navaismo | then go update & try (and possible come back and cry) |
| 21:08.22 | anonymouz666 | lol |
| 21:10.30 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 21:11.22 | anonymouz666 | navaismo: learning from others experience is also a good option. that's why history is important. |
| 21:12.08 | [TK]D-Fender | [17:01]anonymouz666anyone already switched from 1.8 to ast 11? too much trouble? I know there is a file that shows what changes from version to version <- lots of people. No. Correct. |
| 21:14.39 | navaismo | anonymouz666, history which history? human kind, your life or country? Because clearly history really sucks in all topic, wars, no food, fallen in love, asterisk libuuid-dev, force_rport comedia etc etc |
| 21:14.54 | navaismo | so better use your experience or open a ticket |
| 21:17.24 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gwewlugcqtuouygm) |
| 21:17.40 | anonymouz666 | never had any problems with libuuid-dev, rport or comedia. |
| 21:17.53 | navaismo | see |
| 21:18.31 | navaismo | and if you review the history of this IRC, you will find a lot of people bitching about libuuid-devel(including me) |
| 21:18.55 | file | which is why in the next 11 release it'll be an optional dependency... |
| 21:20.48 | navaismo | but break the ICE or something right? |
| 21:21.11 | file | as it is a requirement for ICE/TURN support that functionality would not be available |
| 21:21.46 | SuperNull | hey guys on "sip show peer <peer>" |
| 21:21.48 | SuperNull | woops. |
| 21:22.33 | SuperNull | on sip show peer.. i see Expire which seems to be validly counting down but.. sess-expires .. shows a number 3 times larger than the expire time ? is that normal ? |
| 21:22.48 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 21:24.40 | anonymouz666 | expire has to do with registers. session timers is another configuration in SIP |
| 21:25.14 | anonymouz666 | the idea is good, but the UAC has to support it. |
| 21:25.42 | anonymouz666 | and you can't count that every client will support so is useless for me, but useful for other people. |
| 21:26.57 | SuperNull | gotcha. im working on a realtime issue where fullcontact disappears but the ATA believes its still within the registration period.. will probably look more into it on monday. |
| 21:27.29 | Freeaqingme | What's the difference between asterisk 10 - standard and asterisk 10 -digiumphones ? |
| 21:27.31 | anonymouz666 | why monday? you still have today and tomorrow |
| 21:28.41 | SuperNull | 5:30 on a friday. im already in over time.. |
| 21:28.51 | SuperNull | if you wanna help i might stick around ;) |
| 21:28.55 | SuperNull | its 'random' .. |
| 21:29.15 | SuperNull | as in .. i poll the database every 2 seconds to check if fullcontact is cleared and it has no correlation with timing.. |
| 21:30.35 | *** join/#asterisk luce (~luce@cpe-24-90-234-19.nyc.res.rr.com) |
| 21:37.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:38.11 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 21:41.09 | *** join/#asterisk bipolar (~bipolar@204.186.46.94) |
| 21:59.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:03.41 | kuku | I'm using Application: Bridge and in that diaplan I use Playback(), but the sound plays only on the side of the asterisk extension, not on the call recepient... |
| 22:10.37 | [TK]D-Fender | kuku: Your description is vague. Show us the actual code & call. |
| 22:10.54 | WIMPy | 1.5 calls |
| 22:11.15 | ChannelZ-Wk | yeah after the fact or when are you Playbacking? |
| 22:11.27 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:12.04 | pensmit | Anyone know why you can't do a sip show channelstats for digium phones? |
| 22:13.16 | WIMPy | Because they don't do rtcp? |
| 22:16.17 | anonymouz666 | channelstats uses rtcp information? |
| 22:18.35 | WIMPy | What else could it use? |
| 22:23.52 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-jpcttcrgquidfexp) |
| 22:26.18 | *** join/#asterisk nam3l3zz (~quassel@86-46-203-155-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 22:30.56 | anonymouz666 | WIMPy: its own rtp stack? |
| 22:32.52 | WIMPy | That will obviousely only know onw half of the stats. |
| 22:32.54 | igcewieling | anonymouz666: also you can't do direct media if you want rtcp stats |
| 22:34.54 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 22:35.03 | anonymouz666 | anyone out there is using direct media? :P |
| 22:36.31 | anonymouz666 | WIMPy: make sense |
| 22:38.41 | anonymouz666 | channelstats is handy but I would use tshark, i think it does a better job in rtp stats |
| 22:39.37 | anonymouz666 | gotta go |
| 22:48.39 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 22:55.50 | navaismo | how can i know the IP of a peer via dialplan? |
| 22:56.31 | igcewieling | navaismo: "core show function CHANNEL" |
| 22:57.33 | navaismo | let me see |
| 23:02.21 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 23:08.52 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 23:09.41 | igcewieling | same => n,Noop(CHANNEL(peerip)='${CHANNEL(peerip)}' CHANNEL(recvip)='${CHANNEL(recvip)}' CHANNEL(from)='${CHANNEL(from)}') |
| 23:10.40 | navaismo | [Sep 6 18:09:44] WARNING[17760][C-00000001]: func_channel.c:482 func_channel_read: Unknown or unavailable item requested: 'peerip' |
| 23:10.54 | navaismo | let me see my compiler flags |
| 23:12.30 | navaismo | hmm |