| 00:16.01 | ManxPower | teknoprep: I think the answer is "it might, it might not, depending on your requirements" |
| 00:16.27 | teknoprep | just sip connections |
| 00:16.32 | teknoprep | to phones and a provider |
| 00:16.37 | ManxPower | no meetme or IAX2 trunking? |
| 00:16.41 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 00:16.41 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 00:17.40 | ManxPower | Then the answer is "it might, it might not" |
| 00:18.22 | jaytee | as a certified hardcore nutjob I'm just going to keep polishing my bare metal server :-) |
| 00:18.47 | teknoprep | lol i have a bare metal server |
| 00:18.53 | teknoprep | i also have my house running on vmware server |
| 00:19.00 | teknoprep | which has yet to have a problem ever |
| 00:19.24 | jaytee | your house runs on a vmware server? |
| 00:19.26 | teknoprep | using iSCSI backend storage with GFS and redhat cluster service |
| 00:19.29 | jaytee | virtual kitchen? |
| 00:19.33 | jaytee | virtual bathroom? |
| 00:19.39 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 00:19.41 | teknoprep | well the vmware server is actually at a clients office |
| 00:20.00 | teknoprep | i asked if i could use a portion to host a server and cut him a small break on my monthly billing rate for support |
| 00:20.35 | Vec | When using the AMI, is it possible to send multiple Action commands in parrellel and then tag them in some way to match them to responces, or can only 1 action be performed at a time ? |
| 00:26.17 | ManxPower | Vec: there was no information in manager.txt about this? |
| 00:26.50 | Vec | ManxPower : let me look, if there is my apologies never knew where to look. |
| 00:27.24 | jaytee | Vec, what about the book? |
| 00:28.04 | Vec | jaytee : nothing in the book. |
| 00:28.16 | Vec | jaytee : read everything on the AMI in the book |
| 00:29.07 | Vec | ManxPower : already read manager.txt, on http://www.asterisk.org/doxygen/1.4/ |
| 00:29.14 | Vec | nothing there on that |
| 00:32.49 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:34.35 | Vec | I read on voip-info something about taging actions but it only mentioned, there was no real indication on how to do it. |
| 00:36.23 | *** join/#asterisk RoyK (n=roy@ip-2-52-149-91.dialup.ice.no) |
| 00:36.57 | *** join/#asterisk nitam (n=eskali@173-205-231-201.fibertel.com.ar) |
| 00:37.02 | nitam | hi |
| 00:38.28 | nitam | does anybody know if there is a way to set a (SIP) registration limit ? |
| 00:40.21 | ManxPower | nitam: number, frequency, or simul? |
| 00:41.52 | nitam | ManxPower, actually, whatever that helps ... coz someone was triying to hack my asterisk using some dictionary-style tool |
| 00:42.16 | nitam | in my full log file i got over 100 tries in a short time frame |
| 00:42.23 | ManxPower | nitam: are you the one that posted to the mailing list over the past day or two? |
| 00:42.44 | ManxPower | nitam: nothing in sip.conf.sample jumped out at you? |
| 00:42.45 | nitam | no, this happend today ... |
| 00:43.11 | ManxPower | well, someone else posted to the mailing list with the same issue, only on his system they succeeded in hacking in |
| 00:44.10 | nitam | mm no, its wasn't me .. i just notice this this afternoon |
| 00:44.37 | ManxPower | sucks to be you. might want to check the mailinglist archives, and of course, sip.conf.sample |
| 00:44.38 | nitam | i was thinking about some iptables rules, with limit-burts, but i couldn't make it work over UDP protocol. |
| 00:44.46 | nitam | it just works over TCP ... i guess. |
| 00:45.00 | ManxPower | nitam: the mailing list had some suggested iptables rules |
| 00:45.21 | jaytee | where's the mailing list found? |
| 00:46.21 | ManxPower | in fact 367 messages since yesterdayt |
| 00:46.24 | ManxPower | ~mailinglist |
| 00:46.25 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 00:46.57 | ManxPower | "sip extension compromised" is part of the subject of the thread |
| 00:47.11 | nitam | thanks man |
| 00:47.30 | ManxPower | I should check my one internet accessable system (out of like 15) |
| 00:47.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c80943302ac8f2f0) |
| 00:48.23 | ManxPower | we only send calls out the internet when EVERYTHING (two offices, two carriers) are down |
| 00:49.16 | nitam | i see |
| 00:50.00 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:53.51 | nitam | crap .. i can't find that thread... did you jaytee? |
| 00:54.11 | jaytee | nope, I'm browsing but I'm not sure which section it's in |
| 00:55.59 | *** join/#asterisk lordmortis (n=lordmort@203-206-116-137.dyn.iinet.net.au) |
| 00:56.12 | jaytee | nitam, I think I found it. http://lists.digium.com/pipermail/asterisk-users/2008-June/214514.html |
| 00:57.20 | nitam | yeah, great jaytee |
| 01:00.40 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 01:01.25 | nitam | there is no help, just a few thoughs |
| 01:02.21 | *** join/#asterisk juanjoc (n=juanjoc@host223.190-225-200.telecom.net.ar) |
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| 01:12.29 | *** join/#asterisk jsolares (n=jsolares@67.230.4.74) |
| 01:12.44 | jsolares | does anyone know if 1.4.20.1 also had lockup problems? |
| 01:16.09 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 01:22.44 | *** part/#asterisk RoyK (n=roy@ip-2-52-149-91.dialup.ice.no) |
| 01:24.53 | jsolares | arghh, i have a dual quad core xeon machine with 2 tc400 cards and it still seems that it's not enough :S |
| 01:28.01 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176000517.dsl.bell.ca) |
| 01:29.24 | JT | not enough for that? |
| 01:29.38 | JT | s/that/what/ |
| 01:30.22 | jsolares | 4 e1's of sip g729 traffic |
| 01:30.37 | jsolares | i downgraded to 1.4.18.1 |
| 01:31.00 | jsolares | hopefully it wont die again |
| 01:32.29 | JT | that's disappointing |
| 01:33.55 | jsolares | hmm actually it's just one quad core, X5460@3.16ghz, but still it has the transcoder cards so it wont have to do g729 on the cpu, and 8gb of ram |
| 01:34.13 | jsolares | 1.4.20+ have just been a pain tho, so it might just be that |
| 01:35.11 | JT | what does it do? |
| 01:35.44 | jsolares | and yes it is dissapointing having those issues as i had a dual xeon 2.8ghz handling 90 calls just fine, we upgraded so it could handle 8e1's meh |
| 01:36.26 | jsolares | 1.4.20+? it just dies, show channels shows nothing, stop now does nothing, kill -9 <pidof asterisk> does nothing, and we have a <defunct> asterisk, gotta restart the machine |
| 01:38.21 | jsolares | 1.4.18.1 has survived longer than 1.4.20.1 or 1.4.21.1 with 100 calls |
| 01:38.46 | jsolares | and just as i type that it dies |
| 01:40.41 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 01:41.07 | jameswf-home | heh :) http://dontcallmyboss.blogspot.com/2008/07/next-big-battle-digium-vs-walmart.html |
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| 01:41.53 | imcdona | about to do a voip rollout. I was under the assumption MPLS is cheaper than point to point T's. correct? |
| 01:42.48 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 01:48.08 | jsolares | well since 1.4.18.1 also crashed i'm going to try removing the transcoder cards and using software g729 |
| 01:52.59 | jsolares | seems that the transcoder cards are just paper weights :S |
| 01:56.04 | jsolares | very expensive paper weights |
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| 01:58.49 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:00.19 | jsolares | seems exvito was right, tc400b == baaaaad |
| 02:05.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585204.dsl.bell.ca) |
| 02:06.04 | rift0r | anyone here use vicidial? |
| 02:06.08 | rift0r | or astguiclient |
| 02:07.06 | rift0r | ok i feel like a tard asking this, but what is the command to turn off sip debug in cli? i tried sip debug off, disable etc |
| 02:07.19 | rift0r | i cant figure it out |
| 02:07.20 | rift0r | heh |
| 02:07.29 | jaytee | sip set debug off |
| 02:07.44 | rift0r | sip set debug off |
| 02:07.44 | rift0r | No such command 'sip set' (type 'help' for help) |
| 02:07.51 | jsolares | sip no debug |
| 02:07.56 | jsolares | what version of asterisk do you have? |
| 02:08.02 | rift0r | that worked |
| 02:08.09 | jsolares | probably 1.2.x then |
| 02:08.10 | rift0r | 1.2 tree |
| 02:08.12 | rift0r | ya |
| 02:08.16 | rift0r | i thought i tried no debug |
| 02:08.18 | rift0r | weird |
| 02:08.22 | rift0r | maybe i fat fintered |
| 02:08.24 | rift0r | fingered |
| 02:09.16 | rift0r | thx |
| 02:15.02 | jsolares | heh software g729 had been working for 31mins, 28mins more than with the transcoder cards :S |
| 02:15.59 | jsolares | *has |
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| 02:16.44 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 02:33.02 | zackz | i have a TDM808P, seems like every hour or so, each Zap channel will open up sequentially signalling a call to the system so it puts all the "calls" into the incomming context |
| 02:33.11 | zackz | anyone have an idea on wtf is happening |
| 02:36.01 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 02:37.21 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 02:43.34 | russellb | zackz: sounds like you should call your telco |
| 02:43.40 | russellb | and ask "wtf?" |
| 02:43.44 | zackz | i did |
| 02:43.53 | zackz | up to the demarc, the lines work fine |
| 02:44.06 | zackz | but the guy said when he put a bridge clip on (dont know what that is) the system went crazy |
| 02:44.59 | ManxPower | if nothing else, try swapping one of the pairs. |
| 02:45.09 | ManxPower | i.e. reverse them |
| 02:46.01 | zackz | like on the 66 block? |
| 02:46.22 | zackz | i just have cables punched onto the phone block plugged into the tdm808p |
| 02:47.12 | nitam | guys, is there a way to set different authname than the extension ? ... so, if extension is 200, use another one in order to authenticate to the asterisk server ? (soz if its a lame question) |
| 02:47.15 | ManxPower | ~mailinglist |
| 02:47.15 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 02:47.41 | ManxPower | Also search the mailing list for people reporting similar problems, see if they found a solution. |
| 02:52.16 | *** join/#asterisk geek_cl (n=geek@pc-21-33-83-200.cm.vtr.net) |
| 02:53.40 | geek_cl | what about predictive dialer ? |
| 02:53.55 | JT | is that even a question? |
| 02:54.12 | rift0r | lol |
| 02:56.03 | riddlebox | geek_cl, you need a predictive dialer? |
| 02:56.17 | geek_cl | yes... vicidial is a option? |
| 02:56.17 | zackz | so do you not have to define signalling type in zapata.conf anymore? |
| 02:56.19 | rift0r | I would try vicidial |
| 02:56.28 | rift0r | works well for pred. dialing |
| 02:56.40 | riddlebox | geek_cl, I use astercrm |
| 02:56.51 | *** join/#asterisk lordmortis (n=lordmort@203.8.160.250) |
| 02:57.04 | geek_cl | oh ok |
| 02:57.08 | geek_cl | i will check |
| 02:57.46 | rift0r | hmmm you like astercrm riddlebox ? |
| 02:57.56 | rift0r | is it web gui for agents? |
| 02:58.13 | riddlebox | rift0r, I think it is easy enough to setup a predictive dialer in it, but never used it for agents |
| 02:58.19 | JT | geek_cl: try asking a PROPER QUESTION. |
| 02:58.21 | rift0r | what do you use it for? |
| 02:58.26 | riddlebox | I need to look at vicidial |
| 02:58.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 02:58.46 | geek_cl | ok, thanks |
| 02:58.52 | geek_cl | is for outbound calls |
| 02:59.10 | riddlebox | rift0r, do you like vicidial? |
| 02:59.15 | JT | waits for a question that isn't a pile of random blurts |
| 02:59.24 | riddlebox | lol |
| 02:59.35 | geek_cl | gnudialer is out? |
| 02:59.47 | rift0r | riddlebox it is powerful, but has some bugs i don't like |
| 02:59.50 | rift0r | and the interface is damn ugly |
| 03:00.04 | geek_cl | ok |
| 03:00.15 | riddlebox | rift0r, I didnt really like the interface, but I tried it along time ago |
| 03:00.16 | rift0r | seems like it is very quirky with certain procedures, and don't things out of order eff it up |
| 03:00.29 | rift0r | /s/don't/doing |
| 03:00.54 | riddlebox | astercrm worked well when I tested it, the only thing I didnt like was, it wont redial a campaign with the people that didnt answer |
| 03:01.21 | geek_cl | :O |
| 03:01.47 | rift0r | oh it doesn't do lead recycling? |
| 03:01.54 | rift0r | vici does that |
| 03:02.23 | rift0r | The requested URL /astercrm_documents/installation was not found on this server. |
| 03:02.27 | rift0r | lovely |
| 03:02.43 | riddlebox | rift0r, I think it was moved to sourceforge |
| 03:03.09 | rift0r | yeah i found it |
| 03:04.15 | riddlebox | wow I knew he was going to change the site, but its way different even though thats default drupal |
| 03:04.58 | riddlebox | he is pretty good about checking the forums too, if you ask questions he will respond, I was having issues with the dialer and fxo ports and he looked into it and we got it all working |
| 03:05.34 | rift0r | nice |
| 03:05.53 | rift0r | yeah the vicidial guy is good about that too, he answers shit quickly |
| 03:06.57 | riddlebox | I havent seen anything on vicidial about the predictive dialer part yet |
| 03:07.23 | rift0r | http://www.eflo.net/vicidial.php |
| 03:10.49 | riddlebox | yeah I am going through the demo |
| 03:11.56 | riddlebox | I dont see where you start a campaign and how you can have it call all leads that didnt answer |
| 03:12.03 | *** join/#asterisk joshopkins (n=j@69.88.71.98) |
| 03:12.17 | rift0r | you create a new campaign |
| 03:12.23 | rift0r | and set it to active |
| 03:12.35 | rift0r | then upload leads list |
| 03:12.49 | riddlebox | ahh I see |
| 03:13.02 | rift0r | then go to campaigns |
| 03:13.05 | rift0r | leads recycle |
| 03:13.20 | *** join/#asterisk MrNaz (n=naz@ppp121-44-233-18.lns2.mel4.internode.on.net) |
| 03:14.25 | riddlebox | so its about the same as astercrm |
| 03:14.59 | riddlebox | astercrm has the ability to schedule a campaign now |
| 03:15.29 | rift0r | astercrm def looks nicer |
| 03:15.33 | rift0r | and i like the css ability |
| 03:15.43 | riddlebox | I liked it better when I was looking at dialers |
| 03:15.50 | joshopkins | anyone had problems with te205p and trixbox ce 2.6.1 |
| 03:16.45 | joshopkins | i can not receive inbound calls or outbound calls |
| 03:17.04 | joshopkins | the pri status is good |
| 03:17.44 | riddlebox | ~trixbox |
| 03:17.45 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 03:23.06 | pputman | I have a pri where the telco is not recognizing the setup message, because of the display information element. From the pri debug I have: Display (len= 7) Charset: 31 [ device ]. They are rejecting the setup because of this, does anyone know of a way to change this field. i.e. a zapata.conf configuration option? |
| 03:23.08 | *** join/#asterisk tristanbob (n=tristanb@ubuntu/member/tristanbob) |
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| 03:47.26 | rift0r | ok riddlebox i installed astercrm |
| 03:47.54 | rift0r | now to play |
| 03:59.17 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 03:59.30 | riddlebox | rift0r, let me know how it goes |
| 03:59.43 | riddlebox | you can email me at james@nigmatech.com |
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| 04:41.49 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 04:41.54 | pcrane | hello |
| 04:42.01 | pcrane | I've got a question |
| 04:42.14 | pcrane | does anyone know how to disable text dialing on the snom phones? |
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| 04:52.48 | TJNII | Gaah! DAMN GENTOO! Why does ALSA have to break with EVERY FARKING UPDATE! |
| 04:52.58 | TJNII | needs to switch distros |
| 04:53.15 | drmessano | Gentoo, brute? |
| 04:56.39 | DigitalIrony | drmessano: thats funny |
| 04:57.20 | drmessano | You just earned 3 points on the drmessano karma scale |
| 04:57.35 | drmessano | Not because you thought it was funny, but because you at least appreciated it |
| 04:58.58 | DigitalIrony | drmessano: im a horrible speller if it weren't for spell check, but I know my Lit. |
| 04:59.51 | drmessano | Just like that horrible line - Ubuntu is an old african word meaning "I can't install debian" |
| 05:00.12 | drmessano | It's not "HAW!" funny, but just appreciating the fuckedupedness of it, is all thats needed |
| 05:01.06 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.34) |
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| 05:20.02 | x_or | Anyone know if voicepulse is down? I cannot reach the website. |
| 05:20.54 | pcrane | anyone know anything about polycom phones? |
| 05:20.57 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 05:40.19 | cvox | what about them? |
| 05:40.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:43.50 | *** part/#asterisk x_or (n=cdawson@sub19-181.member.dsl-only.net) |
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| 05:51.10 | gramulhaozin | Hey Guys |
| 05:52.19 | jblack | hi. |
| 05:54.11 | gramulhaozin | hey jblack |
| 05:54.16 | gramulhaozin | Have you tried any IP Phone ? |
| 05:54.26 | gramulhaozin | I have used the Cisco's and I like them very much. |
| 05:54.42 | gramulhaozin | Specially with the capability of processing XML so I can write anything on the screen. |
| 05:54.56 | gramulhaozin | Have you tried anything ? |
| 05:55.25 | JT | wow |
| 05:55.42 | JT | some people like those bastard childs it seems :) |
| 05:55.52 | jblack | I have been supporting polycoms. |
| 05:56.15 | jblack | I avoid cisco equipment when possible |
| 05:56.17 | pcrane | cvox: I was wanting to know how to configure the digit map via xml and tfpt, I managed to figure it out though |
| 05:56.37 | gramulhaozin | jblack: do the polycom process XML ? |
| 05:56.38 | imcdona | Is MPLS cheaper than point to point T-1's? |
| 05:56.39 | JCJC | cisco isnt a bad product though right? its just overpriced? |
| 05:56.49 | jblack | their configs are xml, yes. |
| 05:57.03 | gramulhaozin | jblack: config doesn't matter |
| 05:57.06 | JT | JCJC: cisco ip phones aren't that great |
| 05:57.11 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 05:57.17 | gramulhaozin | jblack: what I'm saying is showing XML to the user using the phone screen. |
| 05:57.25 | jblack | much too overpriced, and tend to be limited. A many people swear by cisco, though, from a "nobody got fired for using ...." standpoint. |
| 05:57.38 | jblack | That depends upon the phone you guys.I believe the 501s do http |
| 05:57.47 | gramulhaozin | believe ? |
| 05:58.04 | gramulhaozin | I have never tried something else than cisco's 7940 and 7960 |
| 05:58.08 | JT | i think all the polycom soundpoints have a minibrowser with new enough firmware |
| 05:58.12 | jblack | The phones I'm supporting right now are about 3,000 miles away. |
| 05:58.48 | gramulhaozin | and when I press the button services on the phone, I have features that I have developed in PHP + XML |
| 05:58.56 | gramulhaozin | JT believe ? |
| 05:58.59 | gramulhaozin | JT tested it ? |
| 05:59.15 | jblack | Yeah, you can do that with the higher end polycoms. |
| 05:59.23 | gramulhaozin | jblack: what model ? |
| 05:59.46 | JT | read the SIP Administrator's Manual |
| 05:59.46 | jblack | I don't have a list offhand. probably anything 5xx and above. |
| 06:00.02 | JT | phones lower than the 5xx do microbrowser too |
| 06:00.05 | JT | 430 does |
| 06:00.11 | JT | possibly 320 and 330 too |
| 06:00.48 | gramulhaozin | 430 is 186 dollars |
| 06:01.06 | gramulhaozin | and with that small screen I'm not going to browse anything |
| 06:01.24 | jblack | maybe what you're looking for is a pc with a headset and a softphone |
| 06:01.47 | gramulhaozin | jblack: now |
| 06:01.49 | gramulhaozin | no |
| 06:01.59 | gramulhaozin | I'm trying to find an option to the cisco phones |
| 06:02.18 | JT | 186 dollars, what big screen cisco can you find for less than that? |
| 06:02.20 | gramulhaozin | the polycom is $186 new the cisco 7940 is $195 new |
| 06:02.32 | gramulhaozin | I don't see the cisco overpriced as the screen is much bigger |
| 06:02.42 | jblack | If you're happy, go with it |
| 06:02.44 | frogonwheels | - :) I have a Intel Quad Duo processor which I've got headphones and use it to connect to asterisk box... |
| 06:02.50 | gramulhaozin | JT not less, just $9 dollars more and I get a cisco 7940 |
| 06:03.06 | frogonwheels | and I find my PAP2t connected to POTS doesn't give as much problems :) |
| 06:03.11 | imcdona | I am doing a voip rollout and need to know if point to point t's are the way to go, or if MPLS is a cheaper route. Anyone? |
| 06:03.17 | JT | frogonwheels: intel quad duo... uhh, what? |
| 06:03.26 | frogonwheels | oph whatever |
| 06:03.26 | gramulhaozin | PAP2T ? |
| 06:03.35 | frogonwheels | you intel duo quadcore |
| 06:03.37 | frogonwheels | swap it |
| 06:03.47 | JT | gramulhaozin: the polycom ip430 is $144 btw |
| 06:03.56 | frogonwheels | or whatever it's called - a fastish cpu |
| 06:04.06 | JT | ... |
| 06:04.14 | JT | perhaps an Intel Core 2 Quad Core |
| 06:04.16 | gramulhaozin | frogonwheels: PAP2T is an FXO version of the PAP2 ? |
| 06:05.06 | Strom_C | gramulhaozin: no |
| 06:05.29 | gramulhaozin | Is there any SIPURA FXO box ? |
| 06:05.36 | Strom_C | SPA-3102 |
| 06:05.47 | gramulhaozin | I need to put 3 fxo in an asterisk |
| 06:06.05 | gramulhaozin | frogonwheels just gave me the idea of using SIP boxes instead of cards |
| 06:06.59 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 06:07.00 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 06:07.12 | gramulhaozin | Wow |
| 06:07.25 | Strom_C | just go with an interface card |
| 06:07.30 | Strom_C | save yourself the hassle |
| 06:07.36 | gramulhaozin | So I can use 3 of those SPA-3102 instead of a TDM400P with 3 FXO modules. |
| 06:07.48 | Strom_C | TDM400P is discontinued |
| 06:07.54 | gramulhaozin | Strom_C have you experienced any problem ? |
| 06:08.02 | gramulhaozin | Strom_C: what are people using lately ? |
| 06:08.04 | Strom_C | you could go for a TDM800 with a quad FXO module and a quad FXS module too |
| 06:08.08 | Strom_C | or a TDM410 |
| 06:08.17 | JT | gramulhaozin: also, the polycom IP320 supports the minibrowser too and is USD$84 |
| 06:08.39 | gramulhaozin | JT $84 new ? where ? |
| 06:08.55 | gramulhaozin | JT is there any polycom with big screen ? |
| 06:09.09 | JT | www.telephonydepot.com |
| 06:09.19 | JT | sure, IP501, IP550, IP601, IP650 |
| 06:09.51 | gramulhaozin | You guys have had trouble using little SPA3000 as the gateway for PSTN ? |
| 06:10.16 | JT | and IP560 |
| 06:10.25 | JT | SPA3000 is discontinued |
| 06:10.29 | JT | it's now the SPA-3102 |
| 06:10.35 | adorah | Atcom in China makes generic TDM400 - not too bad and cheap but avoid the fxs modules.. |
| 06:10.53 | JT | not much more trouble than any other analogue interface |
| 06:10.55 | JT | analogue sucks |
| 06:11.33 | gramulhaozin | adorah: generic TDM400 ? ? |
| 06:11.40 | JT | clone, fake |
| 06:11.45 | Strom_C | gramulhaozin: avoid the clone crap |
| 06:11.45 | gramulhaozin | :P |
| 06:11.58 | adorah | <gramulhaozin>right and not fake no copyrights on the design |
| 06:11.59 | gramulhaozin | :P that's what I though |
| 06:12.12 | JT | there are only two brands worth considering for cards, digium and sangoma |
| 06:13.06 | gramulhaozin | adorah: no copyrights ? you are talking about a zapata card ? |
| 06:13.46 | gramulhaozin | hehe |
| 06:13.51 | adorah | <gramulhaozin>zaptel card yeah..it shows exactly as tdm400 when u setup |
| 06:13.55 | gramulhaozin | I can see Polycom doesn't like big screens :P |
| 06:14.04 | gramulhaozin | adorah: even the PCI ID ? |
| 06:14.12 | adorah | <gramulhaozin>right |
| 06:14.24 | gramulhaozin | if it clones the PCI ID it means it's a clone. |
| 06:14.37 | gramulhaozin | PCI ID has the manufacturer code + hardware code doesn't it ? |
| 06:15.07 | jblack | correct. |
| 06:15.31 | adorah | # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) |
| 06:15.48 | gramulhaozin | adorah: I would not appreciate if someone clone my hardware, even if I didn't copyrighted the design. It doesn't sound good. |
| 06:15.56 | gramulhaozin | adorah: how much are those clones ? |
| 06:16.09 | JT | gramulhaozin: how big do you want them to be? if that's not big enough, get a damn computer |
| 06:16.19 | adorah | LOL now you're talking biz..less than half the price FOB |
| 06:16.23 | gramulhaozin | JT as big as the cisco :P |
| 06:16.36 | JT | gramulhaozin: useless pointless penis extension crap |
| 06:16.48 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:16.50 | adorah | 50$ main board 35$ each fxo module |
| 06:16.50 | JT | adorah: and less than half the quality |
| 06:16.54 | JT | adorah: what a waste of time |
| 06:16.55 | jblack | let's all play runescape on your phone system! |
| 06:17.03 | gramulhaozin | :P |
| 06:17.04 | JT | misers have no place in telecomms |
| 06:17.04 | adorah | <JT>I disagree |
| 06:17.15 | JT | adorah: please don't make up quotes of me |
| 06:17.16 | jblack | JT:I heartily disagree. |
| 06:17.37 | adorah | And that is the only way I can integrate small pbx under 700US$ for 2 FXO |
| 06:17.47 | jblack | Pennywise, pound foolish, I'd agree on. |
| 06:18.06 | gramulhaozin | I sold an IP PBX with 4 refurbished 7940 for 3k |
| 06:18.12 | JT | misers, as in people with a low cost only tunnel vision |
| 06:18.23 | JT | i'm not saying go and get ripped off and buy everything cisco :) |
| 06:18.34 | gramulhaozin | I would not risk the chances of disapointing the customer because the TDM card didn't work because I've got a counterfit board. |
| 06:18.41 | jblack | I was drawn to * because for $300, I cut my monthly phone costs by about 75% |
| 06:19.13 | JT | but it's a different matter when deploying it to a business, you still don't want it to suck |
| 06:19.22 | JT | even if saving money |
| 06:19.22 | gramulhaozin | How much are you guys selling a IP PBX Setup ? |
| 06:19.38 | gramulhaozin | JT you are right |
| 06:19.58 | gramulhaozin | right, the business want the features, BUT THEY NEED THEM TO WORK |
| 06:20.02 | adorah | Well it is not a question of saving money..small customers won't buy otherwise |
| 06:20.14 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 06:20.20 | gramulhaozin | adorah: do you sell those cards or use them ? |
| 06:20.23 | adorah | And it works just that the fxs module sucks |
| 06:20.38 | JT | adorah: then don't go for that end of the market |
| 06:20.43 | JT | adorah: they're not worth it |
| 06:20.45 | gramulhaozin | adorah: FXO works ? |
| 06:20.53 | JT | not all customers are worth having |
| 06:20.59 | gramulhaozin | JT true JT |
| 06:21.01 | gramulhaozin | JT is the man. |
| 06:21.04 | gramulhaozin | That's true |
| 06:21.06 | adorah | If you want to open up the market than you can't be too picky |
| 06:21.13 | gramulhaozin | some customers are not even worth taking. |
| 06:21.39 | JT | the customers who "nickel and dime" over everything are also the biggest hassle and have the highest support costs |
| 06:21.51 | adorah | He whoe is able to hand pick his customers is a lucky devil.I'm not |
| 06:21.57 | JT | you end up losing money on them |
| 06:21.59 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:22.25 | jblack | adorah: I understand where you're coming from. |
| 06:22.27 | gramulhaozin | adorah: do you sell those cards or use them ? |
| 06:22.29 | adorah | I know only two type of customers: Those that pay and those that are trouble.. |
| 06:22.36 | adorah | I do sell them |
| 06:22.48 | gramulhaozin | that's the point. |
| 06:22.49 | gramulhaozin | :P |
| 06:22.52 | adorah | MArket here is tough |
| 06:22.56 | gramulhaozin | And you are probably on a small market. |
| 06:23.07 | jblack | There's an old saying I can't quite place my finger on... |
| 06:23.10 | gramulhaozin | for those in a small market I will just be clear with my customer: you have two options |
| 06:23.14 | adorah | PRI Ibuy mostly Digium\ |
| 06:23.19 | gramulhaozin | by an analog pbx for $1000 |
| 06:23.25 | gramulhaozin | or get a virtual pbx from your phone provider. |
| 06:23.26 | adorah | <PROTECTED> |
| 06:23.44 | gramulhaozin | I'm honest with my customers |
| 06:23.47 | jblack | Anyone can sound good on a steinway, but it takes a virtuoso to make a cheap keyboard sound good. |
| 06:23.58 | gramulhaozin | I will not deploy a asterisk installation for less than $2500 |
| 06:24.17 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 06:24.28 | gramulhaozin | trnzmeta: Linus Torvalds ? |
| 06:24.35 | trnzmeta | yeah I wish |
| 06:24.37 | adorah | I sell it for small ones that need only 2 fxo or where i use FIxed GSM gateways..the rest I use Digium/Sangoma |
| 06:25.05 | gramulhaozin | TWO fxo ? |
| 06:25.11 | gramulhaozin | TWO lines they don't need a PBX |
| 06:25.16 | adorah | But it works fine including fax to mail |
| 06:25.20 | gramulhaozin | they need just phones. |
| 06:25.32 | trnzmeta | guys: what keywords should I google when I'm after |
| 06:25.33 | gramulhaozin | RCA makes good phones for up to 16 lines. |
| 06:25.34 | gramulhaozin | ops |
| 06:25.34 | JT | actually, 2 fxo can have multiple sip connections to an ITSP also |
| 06:25.41 | gramulhaozin | up to 4 FXO and 16 extensions |
| 06:25.43 | adorah | gramulhaozin>Sorry you really don't know what u r talking about |
| 06:26.02 | gramulhaozin | adorah: you are talking about cheap customers |
| 06:26.04 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 06:26.06 | gramulhaozin | look |
| 06:26.10 | trnzmeta | being able to iniate phone calls via web interface (computer) to sip phone |
| 06:26.20 | adorah | The system with GSM gateways save them a fortune on overseas and from overseas calls |
| 06:26.26 | trnzmeta | kinda like what skype does for phone numbers on webpages |
| 06:26.36 | gramulhaozin | My business is deploying good and reliable stuff to my customers. |
| 06:26.44 | gramulhaozin | I work with Security Cameras also |
| 06:26.58 | gramulhaozin | And I have seen many people selling camera systems for $2500 dollars |
| 06:27.03 | gramulhaozin | but my system is $6500 |
| 06:27.04 | trnzmeta | oh who else works with cctv and cameras? |
| 06:27.11 | gramulhaozin | WHY ? |
| 06:27.13 | trnzmeta | I work in that industry |
| 06:27.16 | gramulhaozin | Because my system has quality |
| 06:27.17 | JT | gramulhaozin: but i can buy a cctv system for $500 on ebay! |
| 06:27.30 | gramulhaozin | JT go ahead and deal with the crap |
| 06:27.34 | JT | :) |
| 06:27.37 | gramulhaozin | :P |
| 06:27.45 | gramulhaozin | JT there is a big difference in quality |
| 06:28.05 | adorah | I have to compete with all the smart arses that download Trixbox for free..than they get into trouble and crying for help..with those I rip them off for service hehe |
| 06:28.10 | pputman | Anyone have a clue to this? I have a pri where the telco is not recognizing the setup message, because of the display information element. From the pri debug I have: Display (len= 7) Charset: 31 [ device ]. They are rejecting the setup because of this, does anyone know of a way to change this field. i.e. a zapata.conf configuration option? |
| 06:28.12 | gramulhaozin | My cameras are 540 lines, competition 420 or 380 |
| 06:28.26 | gramulhaozin | My DVR is H.264, competition is MJPEG |
| 06:28.36 | JT | adorah: rip them off.. i think you mean "charge appropriately for the difficulty level involved" ;) |
| 06:28.41 | gramulhaozin | My stuff works and you can call me and I will be happy to assist you |
| 06:28.50 | gramulhaozin | The competition will sell you the stuff and run away |
| 06:29.17 | adorah | RHey don't run they simply helpless..hehe |
| 06:29.57 | adorah | Every kid here with some knowhow of Linux thinks he can download and deploy for free in 20 minutes.. |
| 06:30.02 | gramulhaozin | pputman: did someone told you that or you found that ? did you check your configs? |
| 06:30.18 | gramulhaozin | adorah: and they do and it works man. |
| 06:30.29 | gramulhaozin | adorah: trixbox is there and with the right hardware it works. |
| 06:30.36 | adorah | gramulhaozin>right until they get into the first trouble.. |
| 06:30.43 | gramulhaozin | what trouble ? |
| 06:30.46 | gramulhaozin | I don't see any trouble man |
| 06:30.55 | pputman | gramulhaozin, That's what the telco is complaining of, and I checked the switchtype. My pri debug shows me sending that out, and then I receive a status messaging declaring an invalid information element. So obviously I called the telco, and they're complaining of that field. |
| 06:31.20 | gramulhaozin | JT didn't wanted to take personally about the camera stuff, just wanted to demonstrate my point because the FAKE cards are similar to cheap cameras, got it ? |
| 06:31.28 | adorah | updating..getting under the bonnet..working with endpoints, echo problem, jitter in wirless networks and so forth |
| 06:31.33 | JT | gramulhaozin: i was joking |
| 06:31.41 | gramulhaozin | trnzmeta: what area you are ? |
| 06:31.46 | JT | of course a $500 complete cctv setup is a piece of crap |
| 06:32.14 | gramulhaozin | JT is your business deploying PBX's ? |
| 06:32.26 | gramulhaozin | I mean, do you sell telephony equipment / services ? |
| 06:32.35 | JT | pputman: what switchtype is your telco advising of/ |
| 06:32.45 | JT | gramulhaozin: somewhat |
| 06:32.56 | JT | more datacentre related right now |
| 06:33.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:33.08 | gramulhaozin | JT USA ? |
| 06:33.13 | JT | australia |
| 06:33.18 | gramulhaozin | I'm in US |
| 06:33.46 | pputman | JT, national 2. |
| 06:33.59 | JT | pputman: what switchtype is in zapata.conf? |
| 06:34.03 | gramulhaozin | adorah: I charge the same amount for all my customers, doesn't matter if the have tried something or not. |
| 06:34.04 | pputman | JT, national |
| 06:34.39 | pputman | JT, I was suspecting the same thing, I'm going to get them to verify that it really is a national 2 line. |
| 06:34.53 | pputman | unless you have any more suggestions. |
| 06:34.57 | JT | pputman: that should be right |
| 06:35.04 | JT | pputman: you could always randomly change it :P |
| 06:35.04 | gramulhaozin | adorah: I know what you mean about people trying to be cheap and bypass you, but I love to get smart customers and I love talking to them and showing them how it works and collecting my hourly bill. |
| 06:35.28 | JT | like try ni1, 5ess and dms100 in the usa |
| 06:35.37 | adorah | gramulhaozin>right same here hehe |
| 06:35.52 | gramulhaozin | adorah: how much you charge per hour ? |
| 06:35.56 | pputman | JT, I'll try, thanks. |
| 06:36.04 | gramulhaozin | adorah: in US dollars please. |
| 06:36.14 | adorah | 60$ off-site 100$ on-site.. |
| 06:36.42 | adorah | But the $ is weak so I do it in a historic rate hehe |
| 06:37.18 | adorah | about 30% more in nominal terms.. |
| 06:37.32 | gramulhaozin | adorah: are you in Israel or USA ? |
| 06:37.40 | adorah | gramulhaozin>Israel |
| 06:37.52 | gramulhaozin | I am in USA and I charge $75 an hour |
| 06:38.12 | gramulhaozin | and I always work on-site. |
| 06:38.17 | adorah | on-site or off site? |
| 06:38.28 | gramulhaozin | I always visit the customer |
| 06:38.49 | gramulhaozin | I rarely ever get the customer equipment into my office. |
| 06:38.50 | adorah | gramulhaozin>you r cheap come to work over here hehe |
| 06:39.04 | adorah | I support remotely |
| 06:39.09 | adorah | mpst of the time |
| 06:39.17 | adorah | =most.. |
| 06:39.37 | pputman | JT, are the different switchtypes ITU specifications? |
| 06:39.50 | gramulhaozin | adorah: your customers aren't frugal , they are cheap. |
| 06:40.01 | *** join/#asterisk tparcina (n=tparcina@cisco16.fesb.hr) |
| 06:40.09 | tparcina | hi channel! |
| 06:40.12 | pputman | frugal is when you get quality stuff out of someone elses trash and fix it =) |
| 06:40.13 | gramulhaozin | adorah: jewish people are cheap by nature |
| 06:40.18 | JT | pputman: i don't think most of them are in ITU docs, but i could be wrong, they may have been added |
| 06:40.30 | adorah | gramulhaozin>Indeed they r cheap that is why they pay more and don't be that racist.. |
| 06:40.48 | tparcina | is "sudo apt-get install zaptel" enough to install zaptel drivers on Ubuntu? |
| 06:40.53 | gramulhaozin | adorah: not being racist are all, it's natural |
| 06:40.57 | JT | uhuh |
| 06:41.02 | JT | which is why they are paying more? |
| 06:41.03 | adorah | gramulhaozin>that is why they r smart ..sometimes too smart for their own good hehe |
| 06:41.03 | JT | right |
| 06:41.20 | gramulhaozin | adorah: they don't pay more they are cheap and that's it. But if you explain to them that they are getting counterfit hardware, they will pay the extra to get the real stuff. |
| 06:42.04 | Strom_C | gramulhaozin: wow...you're just digging your own hole deeper and deeper |
| 06:42.35 | gramulhaozin | Strom_C: I have lots of jewish customers, and usually they are good customers. |
| 06:42.48 | adorah | I don't think the TDM400 card that is very crude design and make and made in China with Chinese chipset is such a "masterpiece".. |
| 06:42.48 | gramulhaozin | always get paid, always clear. |
| 06:42.49 | JT | gramulhaozin: how long have you lived in israel? |
| 06:43.04 | gramulhaozin | I live in US |
| 06:43.19 | JT | so how do you know what customers are like in israel? |
| 06:43.29 | gramulhaozin | I actually can say that most of my best customers are jewish |
| 06:43.45 | gramulhaozin | JT I have Israeli customers here in USA |
| 06:43.51 | JT | that's not the same thing as people living in israel |
| 06:43.54 | tparcina | I have executed "sudo apt-get install zaptel" but when I try to call on zaptel channel I get this message - Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 06:44.01 | gramulhaozin | JT probably not. |
| 06:44.34 | gramulhaozin | JT but Israel, brazil or saudi arabia, good business would not opt to get the counterfit hardware |
| 06:45.04 | adorah | This is not a counterfeit it is generic and not worse make than that of DIgium |
| 06:45.12 | gramulhaozin | :P |
| 06:45.21 | gramulhaozin | I will have to try that before deploying |
| 06:45.43 | *** join/#asterisk ker2x (n=chatzill@AToulouse-257-1-6-167.w86-221.abo.wanadoo.fr) |
| 06:45.50 | adorah | U probably don't buy Taiwanese PC or Dell PC coz they r all IBM "counterfeit: :P |
| 06:45.56 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:46.24 | gramulhaozin | no comments on that |
| 06:46.30 | adorah | LOL |
| 06:46.33 | Strom_C | adorah: he can't even spell "counterfeit" |
| 06:46.57 | gramulhaozin | Strom_C: so you like the "counterfeit" hardware too ? |
| 06:47.10 | JT | it's a clone of english |
| 06:47.17 | JT | a copy, with slight changes |
| 06:47.35 | JT | or counterfeit english, depending on who you ask :P |
| 06:47.37 | adorah | For bigger deployment i use IBM server not because they r "real" but because they suitable..and I like their local service.. |
| 06:48.01 | gramulhaozin | Have you guys been to China or Taiwan ? |
| 06:48.10 | adorah | SUre |
| 06:48.37 | gramulhaozin | I have never seen a counterfeit do 100% or have the same quality of an original. |
| 06:48.38 | adorah | I lived in east Asia for years |
| 06:49.01 | gramulhaozin | Cell phones, clothes, tv's , stereo's |
| 06:49.25 | JT | extraneous apostrophes |
| 06:49.25 | ker2x | mmmm, hi. i just can't find the API reference on asterisk.org, any link please ? :) |
| 06:49.26 | adorah | <gramulhaozin>I think u simply don't understand the meaning of "counterfeit" and the difference with generic |
| 06:49.27 | JT | :D |
| 06:49.47 | gramulhaozin | never seen ONE to say that it worked and performed equally |
| 06:50.09 | gramulhaozin | adorah: if you clone the PCI ID and the Manufacturer ID you are not making something GENERIC |
| 06:50.15 | gramulhaozin | adorah: you are making fake hardware |
| 06:50.48 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 06:50.54 | ker2x | (and i can't find the licence too) |
| 06:50.54 | creativx | ker2x: API for what? the manager interface? |
| 06:50.58 | adorah | well I'm not sure the cloned the ID but * recognizes it as the original TDM400 and that is good enough for me |
| 06:51.14 | gramulhaozin | adorah: don't come with IBM / Clones, that's nothing to do with fake hardware, IBM PC model is on the market for many years and have nothing to do with fake / counterfeit equipment. |
| 06:51.21 | ker2x | creativx: the api to develop an app using asterisk |
| 06:51.45 | gramulhaozin | adorah: the kernel recognizes it as they cloned even the PCI ID |
| 06:51.58 | creativx | ker2x: are you familiar with how asterisk works? |
| 06:52.10 | gramulhaozin | adorah: I can picture you here man. |
| 06:52.13 | adorah | well chaps was nice talking to u but I have to start my day work..good nite to the Americans here |
| 06:52.32 | gramulhaozin | adorah: ROLEX WATCH, ARMANI SHIRT, ARMANI JEANS, PRADA SHOES |
| 06:52.37 | gramulhaozin | hauhauahau |
| 06:52.43 | ker2x | creativx: not at all, and that's why i'd like to take a look at the api. :D |
| 06:52.51 | gramulhaozin | all generic. |
| 06:52.53 | gramulhaozin | right :P |
| 06:53.01 | adorah | <gramulhaozin>I can tell u a story about fake jeans but another time..hehe |
| 06:53.05 | gramulhaozin | you wear those and tell people that you bought generic stuff :P |
| 06:53.29 | creativx | ker2x: i would recommend reading the book first, to grasp the concepts |
| 06:53.34 | creativx | ~tfot |
| 06:53.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 06:53.39 | gramulhaozin | I rather wear my cheap levi's than wear fake diesel. |
| 06:54.13 | ker2x | gramulhaozin: Generic<is, good>; :o) |
| 06:54.17 | gramulhaozin | adorah: and don't forget to buy that LV generic bag for your girlfriend, she is going to love it. |
| 06:54.42 | ker2x | creativx: ok, but, do that mean, there is no API reference available online ? :D |
| 06:54.56 | creativx | ker2x: not in the api sense you are thinking |
| 06:55.03 | ker2x | sigh |
| 06:55.05 | ker2x | ok, thx |
| 06:55.06 | creativx | ker2x: there is a manager interface, called AMI |
| 06:55.06 | gramulhaozin | ohh the guy left :P |
| 06:55.11 | gramulhaozin | JT there ? |
| 06:55.20 | creativx | ker2x: which you can do wonders with in controlling/listening to asterisk/ |
| 06:55.22 | JT | hi |
| 06:55.33 | creativx | ker2x: and most of it is documented at voip-info.org |
| 06:55.40 | gramulhaozin | JT I can't believe people accept this GENERIC thing man. |
| 06:55.54 | ker2x | creativx: ok thank you |
| 06:56.10 | JT | yeah the prices of digium and sangoma have come down enough |
| 06:56.15 | creativx | ker2x: also what language are you fluent in |
| 06:56.26 | JT | anyone not willing to pay those prices deserve what they get |
| 06:56.27 | ker2x | Perl and C#, mostly |
| 06:56.35 | creativx | ker2x: theres a lot of opensource apps that utilize the AMI, theres also a java and/or c# class library |
| 06:56.38 | gramulhaozin | JT :P |
| 06:56.47 | gramulhaozin | JT true man |
| 06:57.12 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 06:57.18 | gramulhaozin | I picture that guy "adorah" wearing PRADA, ARMANI, DIESEL, all counterfeit and I see no point. |
| 06:57.35 | gramulhaozin | having a Rolex watch and Armani clothes is not a matter of wearing it. |
| 06:57.39 | creativx | ker2x: http://www.voip-info.org/wiki/view/Asterisk+.NET |
| 06:57.53 | ker2x | thx |
| 06:57.54 | gramulhaozin | It's a lifestyle. you have to pay for that lifestyle. |
| 06:58.04 | jblack | gramulhaozin: You have all sorts of opinions. |
| 06:58.13 | jblack | Which is better? strawberry ice cream, or chocolate |
| 06:58.21 | gramulhaozin | napolitano |
| 06:58.27 | gramulhaozin | jblack: order napolitano |
| 06:58.36 | gramulhaozin | jblack: all three of them. :P |
| 06:58.42 | jblack | shame you're not so bright, though. |
| 06:59.36 | gramulhaozin | jblack: ??? |
| 07:00.38 | ker2x | creativx: that's what i was looking for. i just switched on my brain and found that it doesn't make sense to ask for the server API when i was looking for a client library that can connect to/use the asterisk server :) |
| 07:00.57 | ker2x | need more coffee |
| 07:01.15 | gramulhaozin | spa3102 has echo cancelation ? |
| 07:01.22 | JT | no |
| 07:01.26 | JT | afaik |
| 07:01.26 | creativx | ker2x: exactly ;) |
| 07:01.56 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 07:02.06 | gramulhaozin | :( |
| 07:02.45 | gramulhaozin | JT what do you use for FXO ? |
| 07:03.04 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 07:03.13 | JT | i avoid FXO like the plague |
| 07:03.20 | JT | but i just use a pci card or a SPA-3102 |
| 07:03.28 | JT | if i have to touch evil FXO |
| 07:03.29 | JT | :) |
| 07:03.34 | jblack | gramulhaozin: A common triage check for mild retardation involves asking children to chose among two similiar, but distinct, choices. You failed a basic check to choose between two options. |
| 07:04.41 | gramulhaozin | jblack: similar distinct choices ? I personally like "napolitano". By the way, how is mommy and daddy doing ? |
| 07:06.20 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:06.43 | jblack | My mother passed in 1994. My father is doing ok for his age. |
| 07:08.14 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 07:08.14 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:08.19 | gramulhaozin | jblack: don't come with personal stuff or it may turn back to you. |
| 07:08.22 | creativx | i must be a fucking genious, cause I dont need a test to spot the two children in the channel! |
| 07:08.49 | gramulhaozin | :P |
| 07:08.53 | gramulhaozin | LOL |
| 07:08.57 | jblack | Pardon? You asked a question. I gave you an honest answer. |
| 07:09.34 | gramulhaozin | SPA400-NA wow |
| 07:09.41 | gramulhaozin | found a 4 ports gateway from LINKSYS |
| 07:09.54 | gramulhaozin | that sounds better than a TDM card |
| 07:10.59 | *** join/#asterisk ArchSSM (n=noosjent@193.160.28.100) |
| 07:13.51 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:17.29 | ArchSSM | What "new and revolutionary" features will be available in Asterisk 1.6 as opposed to the 1.4 series? |
| 07:18.08 | creativx | asterisk will be self aware |
| 07:19.17 | ArchSSM | What do you mean by that? |
| 07:19.46 | pputman | fancy new lisp AI |
| 07:20.24 | ArchSSM | hehe. But seriously? |
| 07:21.17 | creativx | i have no idea to be honest and serious. |
| 07:21.27 | ArchSSM | I got the distributed presence, multiple parking lots (a very welcome feature) and TLS support |
| 07:21.30 | ArchSSM | ah.. ok :) |
| 07:25.36 | phpboy | hey all, what would the following two conditions mean? |
| 07:25.37 | phpboy | exten => 3000,4,GotoIf($[${DB(PBX/NS1)}=1]?afterhours-cartrack,${EXTEN},1) |
| 07:25.37 | phpboy | <PROTECTED> |
| 07:29.31 | gramulhaozin | biggest screen and least expansive Polycom is IP550 with 320x160 |
| 07:29.36 | gramulhaozin | Cisco 7940 is 320x222 |
| 07:29.44 | phpboy | hello |
| 07:29.53 | phpboy | I see what the end result it |
| 07:29.55 | phpboy | *is |
| 07:30.04 | phpboy | I just need to know what the conditions are |
| 07:31.31 | creativx | if the db valua PBX/NS1 equals 1 or 0 |
| 07:31.33 | creativx | value* |
| 07:34.41 | phpboy | which DB would that be? |
| 07:34.47 | phpboy | the MySQL db? |
| 07:36.02 | phpboy | I'm just trying to pin point this as to better learn the config of this PBX |
| 07:36.52 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 07:36.53 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:41.12 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:45.45 | creativx | phpboy: the astdb |
| 07:46.47 | pputman | http://www.voip-info.org/wiki/view/Asterisk+database |
| 07:47.43 | creativx | thankas |
| 07:58.17 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 08:01.18 | phpboy | ok, I've dumped the db and I see the values |
| 08:01.27 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 08:01.27 | phpboy | my question is, what would NS2 be for? |
| 08:04.27 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 08:09.43 | creativx | that depends who sets the value ns2.. |
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| 08:12.17 | *** join/#asterisk Datax (n=john@smirnoff.nurvnet.org) |
| 08:12.32 | Datax | Hi all |
| 08:13.00 | Datax | anyone know why when I am logged on the CLI I don't have the console command ? |
| 08:15.55 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 08:17.47 | Datax | bacardi*CLI> console |
| 08:17.47 | Datax | No such command 'console' (type 'help' for help) |
| 08:17.50 | Datax | :( |
| 08:18.01 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 08:25.06 | drmessano | What are you trying to do? |
| 08:25.39 | jblack | sound like he's trying to get in his house from the inside. |
| 08:25.49 | drmessano | Yes |
| 08:26.11 | drmessano | c:\> cmd |
| 08:27.20 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 08:27.34 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 08:27.52 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:28.31 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:31.38 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:32.58 | Datax | drmessano: I'd like to be able to use the dial command but the console command isn't recognized by the CLI |
| 08:33.18 | *** join/#asterisk defswork (n=andy@mx1.3gcomms.co.uk) |
| 08:33.54 | mandh | when i play an file , and through that if hangup occur , the phone line still busy , like as it still play the file? |
| 08:33.54 | drmessano | uh |
| 08:34.25 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 08:34.33 | drmessano | What about DIAL? |
| 08:34.52 | Datax | drmessano: no go |
| 08:34.57 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 08:34.58 | kaldemar | Datax: you need either chan_alsa.so or chan_oss.so loaded to get it. |
| 08:35.02 | Datax | ok |
| 08:35.12 | Datax | I don't think I have those loaded |
| 08:35.18 | Datax | I'll check, thanks for the tip |
| 08:35.22 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 08:36.06 | Datax | mhhh, no sound card in the server so the module didn't load :p |
| 08:37.10 | kaldemar | well there you go, forget about it. if you're just trying to originate calls, there's always the originate command. |
| 08:37.15 | Datax | I'll try setting up alsa with a dummy sound card |
| 08:39.37 | phpboy | creativx: I can't see it being 'set' anywhere in any of the config files |
| 08:39.37 | phpboy | creativx: I'm guessing it's not a 'standard' var |
| 08:40.23 | creativx | does not look like it belongs to asterisk no, who has made the dialplan logic? |
| 08:41.30 | *** join/#asterisk BeeBuu (n=beebuu@125.95.198.42) |
| 08:41.37 | BeeBuu | hello,all |
| 08:41.39 | kaldemar | smells like freepbx or some other gui. |
| 08:42.07 | BeeBuu | is this correct? "One group of Puritans, called the âPilgrims,â crossed the Atlantic in the ship Mayflower and settled at Plymouth, Massachusetts in 1620." |
| 08:42.25 | kaldemar | did they use asterisk? |
| 08:42.29 | BeeBuu | is mayflower? |
| 08:42.38 | drmessano | WTF |
| 08:42.55 | drmessano | BeeBuu, it was the SunFlower |
| 08:42.58 | drmessano | Not MayFlower |
| 08:42.59 | gr0mit | BeeBuu, this is not the History Channel |
| 08:43.00 | kaldemar | would that be a NAT issue? |
| 08:43.09 | pputman | kaldemar, navigation issue. |
| 08:43.15 | BeeBuu | gr0mit: :-P |
| 08:43.34 | drmessano | The SunFlower was also known as the Santa Maria |
| 08:43.40 | drmessano | So either will work |
| 08:43.42 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 08:44.01 | drmessano | Later followed by the Nina and the Pinto |
| 08:44.12 | ArchSSM | As late as 1620? |
| 08:44.18 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 08:44.32 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 08:45.16 | drmessano | the Pinto mission to american is chronicled in the movie "U571", BeeBuu |
| 08:45.17 | drmessano | It |
| 08:45.21 | drmessano | it is a must see |
| 08:46.20 | DigitalIrony | drmessano: you mean ninja and pento |
| 08:46.36 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 08:47.49 | *** join/#asterisk dauhuber (n=dauhuber@213.189.154.108) |
| 08:48.06 | drmessano | Ninja is actually a misnomer.. it was called the NienJah at it's time of commissioning, by King Ralph III of Saint-Elsewear, France |
| 08:49.32 | DigitalIrony | Oh, thought it was a different boat from King henry VIII of the Tudors |
| 08:49.41 | DigitalIrony | sors |
| 08:49.42 | drmessano | It was to be piloted by Captain Howiemandel, and it's cargo heated by the sun, which was developed by the great Edward Begley II |
| 08:50.04 | DigitalIrony | drmessano: yeah see I was thinking of the one with captain nemo |
| 08:51.15 | drmessano | Yes, much like the confusion between Benjamin Bratton and General Denzel Washington when discussing Revolutionary War law |
| 08:51.25 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 08:51.26 | DigitalIrony | yep |
| 08:51.34 | DigitalIrony | I always got those guys confused in school |
| 08:52.01 | drmessano | Kumbang.. I swear I have that movie on Betamax |
| 08:52.30 | DigitalIrony | wow. I wasn't even alive when they had betamax |
| 08:52.57 | drmessano | Kumbang in Bangkok II: Electric Bangaloo |
| 08:53.39 | DigitalIrony | There was some really good kungfu movie on spike today. It was really funny |
| 08:53.55 | phpboy | creativx: A 3rd part company that originally installed this system that my company has asked me to take over |
| 08:54.18 | phpboy | creativx: can I paste 4 lines that show's why this is a confusing ruleset? |
| 08:54.31 | Kumbang | kungfu panda? |
| 08:54.44 | creativx | phpboy: pastebin |
| 08:54.46 | creativx | ~pb |
| 08:54.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 08:54.49 | phpboy | ok |
| 08:56.52 | phpboy | creativx: http://paste.debian.net/9105/ |
| 08:56.57 | phpboy | tell me what you think |
| 08:57.26 | DigitalIrony | kumbang: nope, it was an old good one |
| 09:06.49 | phpboy | creativx: what do you think? |
| 09:06.55 | creativx | let me see |
| 09:08.43 | creativx | does not make sense |
| 09:08.46 | *** join/#asterisk rcy (n=rcy@66.183.58.28) |
| 09:08.50 | creativx | what version of asterisk |
| 09:10.47 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 09:13.47 | masus | hi all, after upgrade to "Asterisk 1.4.21.1" from "Asterisk 1.4 SVN" i get this warning "[Jul 2 11:09:53] WARNING[3286]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info." but everything works fine,does anyone know why i get this warning ? Thanks |
| 09:17.36 | mvanbaak | are you using mysql realtime ? |
| 09:19.03 | masus | yes |
| 09:19.51 | masus | mvanbaak: i use RealTime and everything works fine extensions and sipusers are RealTime , but every 5 seconds i get this warning. |
| 09:20.11 | phpboy | creativx: 1.2.27 |
| 09:25.10 | mvanbaak | did you check the debug info like it suggests ? |
| 09:26.38 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 09:30.36 | masus | mvanbaak: i have check /var/log/asterisk/messages , but there is the same info |
| 09:31.08 | masus | mvanbaak: where can i find the debug info ? |
| 09:38.53 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 09:47.53 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 09:48.14 | mvanbaak | look in /etc/asterisk/logger.conf |
| 09:54.03 | phpboy | creativx: Still not too sure, perhaps there's something else I can check on the system? |
| 09:57.16 | masus | mvanbaak: thank u |
| 10:11.18 | *** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it) |
| 10:15.48 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 10:29.13 | snailrails | hi. there a recommended SIP phone for OSX ? |
| 10:31.40 | *** part/#asterisk snailrails (n=snailrai@dsl-241-13-146.telkomadsl.co.za) |
| 10:32.25 | *** join/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-439d3e78915a2516) |
| 10:32.36 | XnOSX | hello everybody |
| 10:33.06 | XnOSX | friends i need to know what kind of mobile terminal is recommended to work with asterisk? |
| 10:40.03 | *** join/#asterisk badcfe (i=christia@peter.mindslice.net) |
| 10:40.06 | badcfe | on a sip channel i use SIP_HEADER(Call-ID) as identifier of a call. What do i use for a Zap channel? |
| 10:40.22 | mort_gib | Hi, I need some input... I have two sip extensions, I want to direct the "reception" extension to voicemail when they are both unavailable, but to four other extensions if they are in a call. Any idea as to ho to do that?? |
| 10:41.28 | mort_gib | Dialstatus retuen unavailable if they are in a call.... |
| 10:52.25 | *** join/#asterisk Gwayne (n=Gwayne@203.118.32.53) |
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| 11:20.31 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 12:12.50 | agx | i've a problem with CDR: does using Dial after answering a call will make a 2nd CDR line to be written? |
| 12:13.28 | mort_gib | How do I know if a SIP handset/Channel is in use?? |
| 12:13.52 | mort_gib | I get Unavailable if the user is in a call, or if the users has DND on! |
| 12:14.25 | phpboy | mort_gib: sip show channels |
| 12:14.36 | phpboy | mort_gib: or show channels |
| 12:14.46 | phpboy | depending on what you want, you'll of course type this on the console |
| 12:14.47 | mort_gib | But how do I use sip show channels in the dial plan?? |
| 12:14.53 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 12:14.59 | *** join/#asterisk mogie (i=opera@ppp121-44-216-213.lns1.hba1.internode.on.net) |
| 12:15.08 | patrick-- | is there a variable to generate a random number? |
| 12:15.10 | phpboy | that I'm not too sure of |
| 12:15.19 | phpboy | mort_gib: sip ? |
| 12:15.23 | phpboy | should sort you out |
| 12:15.37 | mogie | anyone know if with 1800 you can have a location based feature like the 13 numbers |
| 12:17.16 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.25 | mort_gib | phpboy: I tried ChanIsAvail following TK's advice, but I never managed to get it quite right |
| 12:19.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:19.37 | [TK]D-Fender | mort_gib: 15th times the charm! |
| 12:19.52 | mort_gib | TK -Sorry?? |
| 12:21.54 | patrick-- | [TK]D-Fender: when using the Monitor command to record a conversation to the file, i first hear all i said then hear all the guy on the other end said |
| 12:22.07 | mort_gib | ChanIsAvail gives 2 regardless |
| 12:22.19 | [TK]D-Fender | mort_gib: pastebin is your friend... |
| 12:23.11 | mort_gib | Yes, quite, but what do you want to see?? Apart from everything including /var/log/messages |
| 12:26.28 | [TK]D-Fender | mort_gib: log files are meaningless. CLI channel dumps, hint dumps, your dialplan, CLI for a call where you try to detect in-use, channel configs, etc. |
| 12:26.30 | codefreeze-lap | agx: yes, usually a dial command will result in a CDR. |
| 12:28.59 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 12:30.16 | *** join/#asterisk Gwayne (n=Gwayne@203.118.32.53) |
| 12:36.00 | *** join/#asterisk trafim (n=reallyma@212.200.84.70) |
| 12:36.57 | mort_gib | http://www.pastebin.org/47879 |
| 12:37.36 | tompaw | Hello. Imagine an Asterisk PBX running as a mobile gateway. It receives outgoing calls via SIP and then routes them to appropriate provider, depending on the network. |
| 12:37.58 | tompaw | Now, the network is determined by the number used, which works nicely with Asterisk's calling rules. Right? |
| 12:38.09 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 12:38.40 | [TK]D-Fender | tompaw: ok, fine, sure. |
| 12:38.44 | tompaw | However, with the Number Portability it is not that obvious. It may happen that a number that SEEMS to belong to one network has been ported to other one. |
| 12:39.34 | tompaw | And our whole scenario fails. Now, there are many possible ways to perform a Number Lookup and get its actual network code. The question is - how to integrate that with Asterisk, so that the routing decision is not being made basing on the number, but on some external service? |
| 12:39.52 | [TK]D-Fender | mort_gib: I've said it a hundred times, though not sure to you, but DND does NOT report back chanisavail status <- |
| 12:40.39 | [TK]D-Fender | tompaw: take your call in, call a script, process the "lookup", do whatever you want. |
| 12:41.22 | mort_gib | Then the next logical question HOW do I find out. I can use ChanIsAvail to see if the user is in a call, and maybe ${DIALSTATUS} to detect Unavail... |
| 12:42.25 | tompaw | [TK]D-Fender: I can handle that. The question is - how to use that information for an actual routing? How to explain it to Asterisk? |
| 12:42.40 | [TK]D-Fender | mort_gib: You don't get to find out. I fyou want them to be able to DND, then make a dialplan based DND for them. Those buttons are dead ends. |
| 12:42.54 | [TK]D-Fender | tompaw: Its your dialplan, it does whatever you tell it to. |
| 12:43.16 | mort_gib | How the !"$!"£$ do I set DND in the dial plan?? And will it reflect on the phone?? |
| 12:43.28 | [TK]D-Fender | tompaw: Go to this exten/priority, dial that. Set variable X to Y, yadda yadda... |
| 12:43.28 | tompaw | [TK]D-Fender: I already have the web-based script, as simple as http://myserver/nl?+48123123123, which returns the network ID. I can even name the routes the same as those IDs. But how do I PUT it into the dialplan? |
| 12:44.07 | *** join/#asterisk oilinki (n=oil@ppp-124-120-3-63.revip2.asianet.co.th) |
| 12:44.10 | [TK]D-Fender | mort_gib: make an exten to set a variable of AstDB value to indicate they are DND, and look for it everywhere you might dial that phone. |
| 12:44.12 | tompaw | [TK]D-Fender: sorry if my question sounds dumb, but is that solution possible with AsteriskNOW, or should I install Asterisk from the scratch? |
| 12:44.21 | [TK]D-Fender | tompaw: AGI <- |
| 12:44.48 | [TK]D-Fender | tompaw: You should probably be doing it from scratch as you're doing very custom stuff. |
| 12:44.48 | tompaw | [TK]D-Fender: that's it, thank you. |
| 12:45.22 | [TK]D-Fender | mort_gib: And for "visibility", use DeviceState patch to set a presence indicator on the phone. |
| 12:46.11 | *** join/#asterisk ManxPower (n=manxpowe@218.sub-75-202-16.myvzw.com) |
| 12:46.38 | mort_gib | Hmm, I wanted to avoid using astDB... Like keep it simple, but alas if that is the only way! |
| 12:47.00 | mort_gib | exity |
| 12:47.09 | *** join/#asterisk dauhuber (n=dauhuber@213.189.154.108) |
| 12:48.12 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 12:50.34 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:53.03 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 12:53.20 | *** part/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 12:56.31 | tompaw | [TK]D-Fender: I just called my friend and he suggested usting just a dialplan for that. First running a simple script and then performing a database lookup. |
| 12:58.50 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-176.usadatanet.com) |
| 13:00.42 | *** join/#asterisk Ubluzok (n=ubluzok@62.141.89.219) |
| 13:00.50 | [TK]D-Fender | tompaw: That'd do. |
| 13:02.03 | *** join/#asterisk d1mas (n=chatzill@host-99.dataart.net) |
| 13:02.28 | d1mas | Hello ppl. Someone can help with E1? :) |
| 13:02.40 | JT | can has questions please |
| 13:03.03 | [TK]D-Fender | JT: that's "haz" and "pleeze" |
| 13:03.20 | [TK]D-Fender | d1mas: Feel free to get specific now.. |
| 13:03.27 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:03.53 | d1mas | sure :) Ok, when remote end disconnects, I see DISCONNECT message (with pri debug span 1) but Asterisk does not disconnect the call - a person on a SIP phone hears busy tone. |
| 13:04.14 | d1mas | after some time, I see RELEASE message on PRI and the call actually gets disconnected |
| 13:04.59 | d1mas | so the question is: does it make any sense to keep the call beyond DISCONNECT and if there is no big sense - how to make Adterisk react on the first event ? |
| 13:05.22 | tzafrir_laptop | in hsort: yes |
| 13:05.26 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 13:05.32 | *** join/#asterisk Ubluzok (n=ubluzok@62.141.89.219) |
| 13:06.03 | d1mas | tzafrir: I'm actually switching from analog lines to E1 to keep these busy-tone callers from the conference :) |
| 13:06.13 | tompaw | [TK]D-Fender: Thanks for the advice. |
| 13:06.23 | d1mas | I mean to prevent others from hearing these beep-beep-beep |
| 13:07.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.55 | tzafrir_laptop | hmmm.... actually, yeah, you can disconnect the call from your side |
| 13:08.20 | tzafrir_laptop | it can be used for things such as a "the number you dialed is not valid" message |
| 13:09.35 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-162-129.red.bezeqint.net) |
| 13:09.44 | Dovid | anyone here from Germany ? |
| 13:11.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:11.26 | d1mas | tzafrir: you saying that if I dial wrong number, there is a DISCONNECT message first, then someones voice says "the number is invalid" and then goes the RELEASE ? |
| 13:17.09 | ManxPower | on a PRI if you dial a number not in service, the telco will send back a code to the PBX (HANGUP CAUSE) then Asterisk must play the correct message. you set this in extensions.comnf |
| 13:18.05 | d1mas | How to tell Asterisk I want to hangup the call on PRI DISCONNECT event, not a PRI RELEASE ? |
| 13:18.45 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 13:19.21 | ManxPower | d1mas: remove the priindication =inband from your zapata.conf |
| 13:19.27 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 13:19.31 | d1mas | oh my god |
| 13:19.39 | d1mas | I seen it is default and forgot it |
| 13:19.45 | ManxPower | there is a reason the sample says you should not use it. |
| 13:19.49 | d1mas | ManxPower: thanks a lot! will try |
| 13:20.02 | d1mas | yes, yes, I have seen it :) |
| 13:21.01 | *** join/#asterisk eric_at_footstep (n=eric@80.101.125.5) |
| 13:22.28 | eric_at_footstep | Hello, I'm trying to configure asterisk to work with a Cisco/Linksys SPA400. Could anyone help me out here? |
| 13:22.33 | tompaw | if using AsteriskNOW, are the calling rules (dialplans) also kept in extensions.conf, or somewhere else? |
| 13:22.53 | [TK]D-Fender | tompaw: Some of it there, some of it is generated based on users.conf. |
| 13:23.00 | [TK]D-Fender | tompaw: its craptastic. |
| 13:23.24 | ManxPower | tompaw: we are not 2nd level support for AsteriskNOW, ask on the #AsteriskNOW channel |
| 13:23.27 | d1mas | ManxPower: however... asterisk still disconnects only after receiving RELEASE not DISCONNECT |
| 13:23.29 | [TK]D-Fender | eric_at_footstep: Go follow the guides on the WIKI & voxilla.com ' s forums. |
| 13:24.00 | eric_at_footstep | [TK]D-Fender: Done all that. We're using the newest firmware on the SPA400. It's broken now. |
| 13:24.18 | [TK]D-Fender | eric_at_footstep: clarify "broken". |
| 13:24.25 | ManxPower | eric_at_footstep: You need to tell us what is not working |
| 13:25.01 | eric_at_footstep | [TK]D-Fender: Gives a 403 Forbidden |
| 13:25.06 | *** join/#asterisk proppy (n=proppy@rosiers.mekensleep.com) |
| 13:25.13 | ManxPower | eric_at_footstep: That tells you it IS working. |
| 13:25.18 | [TK]D-Fender | eric_at_footstep: thats not "broken", thats "configured wrong". |
| 13:25.43 | eric_at_footstep | I wish that would be it. The SPA400 doesnt take a password |
| 13:26.33 | eric_at_footstep | While it might be something buggie in the SPA400 (Not accepting connections from the new server ip opposing to that of the SPA9000 PBX) |
| 13:26.42 | ManxPower | eric_at_footstep: It would be the first Linksys/SIPura box that does not support passwords |
| 13:26.42 | *** join/#asterisk cabel (n=abel408@64.128.120.92) |
| 13:27.04 | eric_at_footstep | There are a few differences between the Asterisk and SPA9000 register headers |
| 13:27.22 | eric_at_footstep | I was wondering if that might be causing it |
| 13:28.05 | [TK]D-Fender | eric_at_footstep: pastebin is your friend. |
| 13:28.06 | eric_at_footstep | ManxPower: I assure you, it doesnt accept a password |
| 13:28.07 | [TK]D-Fender | ~pb |
| 13:28.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 13:28.09 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:28.46 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 13:28.53 | eric_at_footstep | Well the differences are quite small, dont really think I need to post the entire headers |
| 13:29.46 | eric_at_footstep | SPA9000 sends a "SIP Display info" header. And adds a name to a the contact headers: From: "NAME" <...> |
| 13:29.57 | ManxPower | According to Cory Andrews of VOIP Supply.com the SPA-400 will not work with Asterisk |
| 13:29.58 | [TK]D-Fender | eric_at_footstep: http://forum.voxilla.com/linksys-spa9000-spa400-support-forum/asterisk-cannot-register-spa400-22405.html |
| 13:30.08 | ManxPower | http://lists.digium.com/pipermail/asterisk-users/2006-September/166892.html |
| 13:30.10 | [TK]D-Fender | ManxPower: plenty of people using it. |
| 13:30.21 | eric_at_footstep | checking out url... brb |
| 13:30.30 | cabel | Hello, after updating zaptel and asterisk something broke. I cannot make internal calls anymore (and most likely external). Nothing is in the full log and this shows up in the messages log: "NOTICE[2677] chan_sip.c: Registration from '<sip:1024@192.168.16.6>' failed for '192.168.16.27' - No matching peer found". I can get into asterisk -r. After updating asterisk I had to change /etc/asterisk/asterisk.conf because the astrundir was wrong. |
| 13:30.31 | ManxPower | I guess Cory doesn't know what he's talking about. |
| 13:30.50 | ManxPower | cabel: you forgot to tell us the versions |
| 13:31.05 | [TK]D-Fender | eric_at_footstep: http://forum.voxilla.com/linksys-spa9000-spa400-support-forum/ |
| 13:32.21 | cabel | asterisk version 1.4.19.2 and zaptel version 1.4.10.1. I first started with zaptel version 1.4.11 but downgraded to see if it would help. and I started with asterisk version 1.4.20 but also downgraded |
| 13:32.42 | [TK]D-Fender | cabel: pastebin your sip.conf , and CLI output with SIP debug for a failed call |
| 13:32.44 | [TK]D-Fender | ~pb |
| 13:32.45 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 13:32.46 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:32.55 | [TK]D-Fender | cabel: masking only passwords |
| 13:33.03 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 13:33.41 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 13:34.02 | ManxPower | cabel: What version worked for you? |
| 13:34.03 | eric_at_footstep | Well, Links weren't much of a help. In the past the SPA400 could be used with asterisk. My guess is linksys changed something to avoid this. (They really want to sell their crappy SPA9000's) |
| 13:34.13 | cabel | ok, hold on |
| 13:34.39 | ManxPower | eric_at_footstep: You sure give up easily. |
| 13:34.50 | [TK]D-Fender | eric_at_footstep: I currently have no reason to believe taht to be the case. |
| 13:35.16 | [TK]D-Fender | eric_at_footstep: And since you're not showing us anything I don't see your situation changing any time soon. |
| 13:35.39 | eric_at_footstep | [TK]D-Fender: hahaha, i'm not giving up |
| 13:35.50 | eric_at_footstep | im trying to find out what linksys might have changed |
| 13:36.04 | [TK]D-Fender | eric_at_footstep: And you're not giving us anything to work with. Guess we won't be able to help you any further. |
| 13:36.07 | ManxPower | eric_at_footstep: perhaps a pastebin of the CLI output of a failed call? |
| 13:36.18 | eric_at_footstep | since firmware 1.0.1.4 people are reporting this authentication problem |
| 13:36.52 | eric_at_footstep | <ManxPower: there is no failed call, the SPA400 gives a Forbidden upon register |
| 13:37.05 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 13:37.39 | ManxPower | eric_at_footstep: You don't have asterisk register to the device, you have the device register to Asterisk. Perhaps the CLI output of a SIP debug. |
| 13:37.45 | d1mas | ManxPower: have you seen my message? Asterisk still keeps the call up until RELEASE PRI message |
| 13:37.53 | ManxPower | You're going to give us a pastebin one way or another. |
| 13:38.14 | [TK]D-Fender | d1mas: after making the change did you completely restart Asterisk? |
| 13:38.22 | d1mas | sure |
| 13:38.24 | ManxPower | d1mas: pastebin the CLI output of a failed call |
| 13:38.33 | d1mas | I did "restart now" |
| 13:38.40 | [TK]D-Fender | ManxPower: And yes, you do register to the SPA-400 |
| 13:38.45 | eric_at_footstep | ManxPower: thats not true, SPA400 functions as a peer (proxy perhaps?). |
| 13:38.48 | d1mas | 5 min |
| 13:38.49 | [TK]D-Fender | d1mas: include your configs |
| 13:38.49 | eric_at_footstep | thanks Fender |
| 13:39.18 | eric_at_footstep | Going to reset the SPA400 |
| 13:39.29 | eric_at_footstep | Will be back later, thanks for you suggestions so far |
| 13:39.36 | ManxPower | eric_at_footstep: Unless you start doind what you are told to do nobody will help you |
| 13:39.44 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:39.54 | eric_at_footstep | ManxPower: What would you like me to do then? |
| 13:39.56 | ManxPower | we are not paid to help you. |
| 13:40.21 | ManxPower | how about the CLI output with SIP debug enabled showing the problem |
| 13:40.30 | ManxPower | use pastenbin.ca |
| 13:41.00 | eric_at_footstep | ManxPower: k, hold on |
| 13:42.22 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 13:43.12 | ManxPower | eric_at_footstep: You should not need to register, assuming neither Asterisk nor the SPA are on dynamic IPs |
| 13:43.21 | eric_at_footstep | ManxPower: http://pastebin.ca/1060488 |
| 13:43.35 | eric_at_footstep | ManxPower: You do register to an SPA400 |
| 13:43.35 | cabel | Ok Here is my sip.conf http://pastebin.com/d522e56be |
| 13:43.53 | cabel | and my sip debug http://pastebin.com/d5eb0e7e0 |
| 13:44.01 | eric_at_footstep | ManxPower: The SPA9000 does it, so i'm sure i should do it too then |
| 13:44.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 13:44.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:44.46 | [TK]D-Fender | cabel: please permanently remove all commented lines from your sip.conf and re-pastebin it. |
| 13:44.59 | ManxPower | eric_at_footstep: paste the register => line sans password |
| 13:45.00 | cabel | ok |
| 13:45.21 | [TK]D-Fender | cabel: and you have no [1017] in three for your register to match against |
| 13:45.22 | ManxPower | cabel: I'm not going to wade thru 300 lines of comments |
| 13:45.26 | [TK]D-Fender | there* |
| 13:45.30 | d1mas | ManxPower: http://pastebin.com/m58d26df8 |
| 13:46.06 | eric_at_footstep | ManxPower: It doesnt take passwords: register =>SPA9000@192.168.199.3/SPA9000 |
| 13:46.06 | ManxPower | d1mas: now repaste WITHOUT the pri debug |
| 13:46.10 | ManxPower | eric_at_footstep: remove the /spa9000 |
| 13:47.05 | ManxPower | eric_at_footstep: it looks like the SIP debug is about SPA400 |
| 13:47.06 | eric_at_footstep | ManxPower: Will do, hold on |
| 13:47.21 | ManxPower | [Jul 2 15:21:46] WARNING[3738]: chan_sip.c:12526 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'spa400' to '192.168.199.3' |
| 13:47.23 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:47.32 | eric_at_footstep | ManxPower: yup, thats true |
| 13:47.45 | eric_at_footstep | i've been trying to mimic the SPA9000 situation |
| 13:47.55 | eric_at_footstep | the debug i pasted is a little bit older |
| 13:48.06 | eric_at_footstep | from when i exactly followed the guide |
| 13:48.08 | ManxPower | eric_at_footstep: Best of luck. I cannot help you further. |
| 13:48.11 | cabel | this is the only thing in my sip.conf http://pastebin.com/d6c679abe haha is that bad? |
| 13:48.29 | [TK]D-Fender | cabel: Where is your phone's config? |
| 13:48.48 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-168-144.dsl.teksavvy.com) |
| 13:48.56 | ManxPower | cabel: you have no devices set up. |
| 13:49.06 | eric_at_footstep | ManxPower: Np, thanks for your time and suggestions |
| 13:49.11 | [TK]D-Fender | ManxPower: Perhaps, and I think we both know where ti resides |
| 13:49.26 | cabel | well where di they go? |
| 13:50.09 | ManxPower | [TK]D-Fender: in a file named sip_addional.conf, right? |
| 13:50.19 | [TK]D-Fender | ManxPower: nope. |
| 13:50.34 | [TK]D-Fender | cabel: This is your config. Where they hell did you configure your phone? |
| 13:50.35 | ManxPower | [TK]D-Fender: users.conf? |
| 13:51.10 | cabel | well all my lines are in sip_addition.conf |
| 13:51.18 | cabel | I only have 2 lines right now |
| 13:51.29 | d1mas | ManxPower: ??? there is nothing to look at really - the call established fine, there is no warnings etc. The dialplan consist of single line - Dual(IAX2/1001).... I probably was not clear enough - the call eventually disconnects. the problem is only that sequence is: Asterisk receives PRI DISCONNECT and my user hears congestion tone for 30 seconds, then Asterisk receives RELEASE and actually... |
| 13:51.31 | d1mas | ...releases the line. What i want is to disconnect the call on the first vent (DIScONNECT) so user won't hear the congestion tone... |
| 13:51.51 | [TK]D-Fender | cabel: Nothing calls that file. Its contents are irrelevent |
| 13:51.55 | cabel | sip_additional.conf* |
| 13:52.06 | ManxPower | cabel: we CANNOT help you with GUI Asterisks |
| 13:52.11 | ManxPower | ~trisbox |
| 13:52.15 | ManxPower | ~trixbox |
| 13:52.16 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 13:52.29 | [TK]D-Fender | ManxPower: its not even FreePBX. Leftovers at best... |
| 13:52.37 | ManxPower | d1mas: ok then, pastebin the extensions.conf |
| 13:52.39 | [TK]D-Fender | cabel: And why are you using that file? |
| 13:53.14 | cabel | I thought it was an asterisk problem because trixbox doesn't display any errors |
| 13:53.35 | ManxPower | cabel: What part of "we can't support trixbox here" did you not understand |
| 13:53.56 | cabel | haha, alright thanks anyway |
| 13:54.02 | [TK]D-Fender | cabel: there is no such thing as an "Asterisk problem". There is only "I have no idea what I'm doing or how * works" poroblems |
| 13:54.07 | ManxPower | cabel: don't bother with the extensions.conf pastebin. I would not be able to read it anyway |
| 13:54.22 | ManxPower | We are NOT 2nd level support for TrixBox |
| 13:54.32 | d1mas | ManxPower: common :) it is one-line... but if you insist: http://pastebin.com/m2875b608 |
| 13:54.58 | ManxPower | d1mas: And THAT is your problem. |
| 13:55.31 | cabel | alright. I'm sincerely sorry guys for wasteing your time. You really don't need to bash me. We were all new at one point and I've learned a lot about asterisk in just a week even though I'm using trixbox. |
| 13:55.36 | ManxPower | d1mas: after the dial you must check the value of HANGUPCAUSE, then play a message based on that return code, or play a busy, or hangup, or something else. Asterisk does not magvally handle this, you must do it in your dialplan |
| 13:56.23 | ManxPower | d1mas: add a priority after the Dial with Noop(HANGUPCAUSE is ${HANGUPCAUSE}) then you can see the disconnect cause without all the PRI debug crap |
| 13:56.38 | [TK]D-Fender | cabel: Trixbox will teach you practically nothing about *. It adds so much crap that you will waste tons of time just to work your way around it. It isn't there to be learned from. |
| 13:57.13 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:57.17 | d1mas | ManxPower: look, the next priority (after the Dial) does not get executed until Dial finishes. And it does NOT finish immediately when the far end hangs up |
| 13:57.19 | ManxPower | cabel: then please leave. We do this stuff for free, we are not paid. We have real paying jobs that we could have been doing instead of spending half an hour with no chance ever of being able to solve your problem. |
| 13:57.39 | ManxPower | d1mas: Best of luck solving your problems. I cannot help you futher. |
| 13:57.50 | cabel | Yea well considering I've only worked with linux for a year I figured a gui interface would be best. Starting to think if I should just use asterisk |
| 13:57.58 | d1mas | ManxPower: thanks anyway |
| 13:58.08 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:58.24 | beek | cabel: absolutely. Just use Asterisk. And read this book: |
| 13:58.26 | beek | ~book |
| 13:58.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:58.37 | jaytee | enters and bows repeating "I'm not worthy!" over and over. |
| 13:58.45 | cabel | ManxPower: It was 10 min of your time and I had already appologized |
| 13:59.20 | kamanashisroy | hi, is it possible to bridge two channels from AGI script ? |
| 13:59.21 | [TK]D-Fender | cabel: Ok, jsut let go. If you want support for what you've got, you know where to go. When you're ready to move on, we'll still be here. |
| 13:59.35 | cabel | Thank you |
| 13:59.48 | beek | cabel: Don't take it personally. Both ManxPower and [TK]D-Fender can be extremely brutal... and the offer THE BEST tech support for *. There are none better. |
| 13:59.58 | jaytee | some of us will still be here. odds are I won't be if I don't lay off the cheese fries |
| 14:00.11 | tompaw | how can I test the DB function output from the CLI? |
| 14:00.26 | [TK]D-Fender | beek: pshaw! I'm only moderately brutal with a side order of fries & sarcasm! |
| 14:00.39 | ManxPower | beek: I'm doing better. Look at d1mas. All he did was tell me why he should do what I told him to do. Instead of ripping his head off I just stopped helping him. |
| 14:01.05 | tompaw | I am trying to follow the example from http://www.asteriskguru.com/tutorials/dbget_function.html |
| 14:01.09 | ManxPower | all he did was tell me why he should NOT do what I told him to do, that is. |
| 14:01.17 | beek | I've lurked for a few months and have watched you two in action. Brutal is an understatement. But jeez do you get to the heart of the problem and get it solved! |
| 14:01.17 | kamanashisroy | everytime I watch this channel and drupal channel .. I find people are fighting .. :D |
| 14:01.24 | tompaw | however, my DB(test/data) function doesn't seem to return anything! |
| 14:01.37 | tompaw | I've checked my db with 'database show' and the data is ther |
| 14:01.41 | tompaw | there. |
| 14:01.59 | kamanashisroy | tompaw: which db are you using ? |
| 14:02.02 | jaytee | kamanashisroy, hey! ya wanna piece o' me? huh? c'mon, put em up! :-) |
| 14:02.07 | ManxPower | tompaw: perhaps ${DB(test/data)} |
| 14:02.12 | *** join/#asterisk ursom (n=relas@port-92-195-88-113.dynamic.qsc.de) |
| 14:02.37 | tompaw | oh god |
| 14:02.42 | tompaw | ManxPower: how did you know? |
| 14:02.48 | [TK]D-Fender | beek: well by that definition sure. |
| 14:02.50 | ManxPower | tompaw: maybe not, but that is the most common error with FUNCTIONS |
| 14:02.59 | ursom | hello! I installed Asterisk 1.4-latest. But I think there is no SIP-support yet? |
| 14:03.06 | kamanashisroy | tompaw: :) |
| 14:03.24 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:03.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.25 | ManxPower | ursom: every version since 0.05 (8 years ago) has supported SIP |
| 14:03.48 | ursom | If I type sip in the cli |
| 14:03.48 | jaytee | ursom, * without SIP is like Ford forgetting the engine in an F-250 |
| 14:04.04 | kamanashisroy | ursom: did you try "core show channeltypes" ?? |
| 14:04.05 | [TK]D-Fender | ursom: * has had SIP since the start. |
| 14:04.07 | tompaw | ManxPower: it's my first time with Asterisk's functions, thanks :> |
| 14:04.26 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 14:04.29 | ursom | sip show peers |
| 14:04.30 | ursom | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 14:04.49 | ManxPower | ursom: did you install from source, or install from a package? |
| 14:04.53 | ursom | source |
| 14:04.58 | [TK]D-Fender | ursom: Quick guess : running a soft-phone or other SIP service on the same box? |
| 14:05.06 | ursom | no |
| 14:05.10 | ManxPower | ursom: waiting for the response to what kamanashisroy said |
| 14:05.28 | [TK]D-Fender | ursom: do "module load chan_sip.so" |
| 14:05.46 | kamanashisroy | ursom: I think you did noload chan_sip.so :) |
| 14:05.54 | ManxPower | ursom: is the first time Asterisk has been installed on the system? |
| 14:05.58 | kamanashisroy | ursom: or something like autoload=no |
| 14:06.10 | ursom | ManxPower: yes |
| 14:06.21 | ManxPower | ursom: what version of Asterisk. |
| 14:06.28 | ursom | okay works fine with module load chan_sip.so |
| 14:06.44 | [TK]D-Fender | ursom: Go look at your modules.conf |
| 14:06.53 | kamanashisroy | can I bridge channels from AGI script ? |
| 14:07.02 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:07.11 | [TK]D-Fender | kamanashisroy: huh? |
| 14:07.20 | ManxPower | kamanashisroy: the only way I can think of us to use meetme, but there might be another way. |
| 14:07.28 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 14:07.41 | kamanashisroy | ManxPower: I am mad with one of my application .. |
| 14:07.56 | ursom | okay works fine :) thx. |
| 14:07.57 | ManxPower | kamanashisroy: I have no idea what you just said. |
| 14:07.58 | kamanashisroy | it was calling the other end |
| 14:08.19 | kamanashisroy | and calling agi script in background thread |
| 14:08.41 | [TK]D-Fender | kamanashisroy: Your explanation is not getting any clearer... |
| 14:08.48 | kamanashisroy | and it is not supported anymore I think , I mean it is blocking two threads to run different applications .. |
| 14:08.54 | kamanashisroy | [TK]D-Fender: sorry |
| 14:09.09 | ursom | I've got another question: I've got a hfc-s card in te-mode. How to hide the callerid? Changing the callerid works, but not hiding. |
| 14:09.56 | ursom | I tried SetCALLERPRES(prohib_not_screened), but without any effect. |
| 14:10.01 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:10.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:10.11 | jaytee | lmadsen, hi |
| 14:11.17 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 14:11.49 | lmadsen | jaytee: hi hi |
| 14:12.32 | kamanashisroy | I wrote a small application earlier .. the application dials the destination channel .. and then after the call is established it feeds the caller channel, but it does not feed the called one .. And when the user press "*" in this stage it starts an agi script in the background and calls bridge function(which is blocking) ..... It was working earlier .. but not now .. -> This is the application specification ...... |
| 14:13.12 | kamanashisroy | I know that asterisk manager is the best choice to do this |
| 14:13.23 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 14:13.53 | kamanashisroy | There is also a schedular to do some scheduled tasks .. But it needs a callback where I cannot run the agi :( .... |
| 14:14.46 | kamanashisroy | Now I am trying a little different .. I want to exit the application when user press "*" .. and then I like to run an agi script that will bridge two channels .. |
| 14:14.55 | kamanashisroy | please help :D |
| 14:17.38 | kamanashisroy | ^^ the total idea is "listen before you accept a call" |
| 14:18.14 | [TK]D-Fender | kamanashisroy: Then this entire pile of scripting is a waste. Look at the "M()" options for Dial. |
| 14:18.23 | kamanashisroy | the agi script is required because it queries the database and shows some data in the ipphone screen while the call is established .. |
| 14:19.18 | kamanashisroy | I have seen that a lot of times .. review now in this position again :-P |
| 14:19.48 | kamanashisroy | s/review/reviewing .. |
| 14:20.23 | *** join/#asterisk intralanman (n=lanman@216.40.253.210) |
| 14:20.59 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 14:22.43 | kamanashisroy | [TK]D-Fender: one kind of joke there is that when the macro ends the called party is hanged up .. it seems good for other tasks .. but not in this case .. please fix me if I am wrong .. |
| 14:23.53 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 14:23.58 | *** join/#asterisk mond0 (n=lvsyijnc@galactic.umflint.edu) |
| 14:24.58 | [TK]D-Fender | kamanashisroy: No, the entire point of that option is for "privacy mode" implementation. |
| 14:25.14 | [TK]D-Fender | kamanashisroy: Keep reading the instructions and samples till your eyes bleed. |
| 14:25.43 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 14:25.52 | mond0 | What the trick to get NetMeeting working? The PBX works because I have two phones that can call each other, but with NetMeeting, I can dial a phone and establish a call, but there's no audio. I've tried combinations of codecs and even installed the GSM codec for NetMeeting. At times I've seen errors in the logs (while trying different codec combinations) but right now there's no errors at all. Should I turn on debugging? |
| 14:26.50 | kamanashisroy | sure |
| 14:27.14 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 14:28.42 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 14:29.16 | [TK]D-Fender | mond0: Which 2 "phones"? |
| 14:29.21 | tompaw | <PROTECTED> |
| 14:29.44 | kamanashisroy | tompaw: see the AGI application .. |
| 14:30.00 | [TK]D-Fender | tompaw: "System" or "AGI" |
| 14:30.13 | tompaw | thx guys. |
| 14:31.18 | tompaw | Now, correct me please if I made some mistakes in my plan. Upon a call, I DB()-put a '0' value under the index of EXTEN, and I launch external application passing the EXTEN parameter to it. |
| 14:31.53 | tompaw | Then, I make a loop that waits for DB()-get for that EXTEN-index to return something different than 0. If it does, I route the call with the provider pointed by that value. |
| 14:32.05 | tompaw | Does that sound logically? |
| 14:34.50 | [TK]D-Fender | tompaw: ... huh? just go TRY some stuff, and see what happens |
| 14:34.51 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 14:35.01 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 14:35.09 | tompaw | :) |
| 14:35.36 | tompaw | OK, so the last piece of the puzzle would be - how to set an Asterisk's database value outside the Asterisk? Do I need to use that AGI thingie for that? |
| 14:36.46 | [TK]D-Fender | tompaw: probably a good way. |
| 14:37.28 | [TK]D-Fender | tompaw: You can do it in AGI, or through calling "asterisk -rx ..." from linux CLI, via AMI, etc. |
| 14:40.06 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 14:40.09 | tompaw | righto! |
| 14:40.19 | *** join/#asterisk aaawrekng (n=aaawrekn@c-76-121-222-172.hsd1.wa.comcast.net) |
| 14:41.06 | mogie | who owns grabup? |
| 14:41.18 | mogie | woops. wrong chat |
| 14:43.48 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 14:45.51 | kamanashisroy | [TK]D-Fender: no dial is no good .. :( |
| 14:47.47 | [TK]D-Fender | kamanashisroy: Yes it is. You dial, run your privacy macro to see if the other side accepts, if so, bridge. |
| 14:48.00 | [TK]D-Fender | kamanashisroy: Keep reading till your eyes bleed. |
| 14:48.49 | kamanashisroy | [TK]D-Fender: no .. you know I need to do stuff, I prefer AGI for that, after it is bridged .. |
| 14:49.48 | kamanashisroy | [TK]D-Fender: I have seen the dial application code .. you are right that it will continue dialing if the macro does not set the variable .. |
| 14:49.49 | [TK]D-Fender | kamanashisroy: Dial is the right tool for this job, and unless you show us what you're doing, I'll simply believe that you are doing it wrong. |
| 14:50.22 | *** join/#asterisk huey23 (n=yea@64.192.209.34) |
| 14:51.14 | kamanashisroy | [TK]D-Fender: I like the way you think .. but as I told you it is a small agi script that queries the database and updates an ipphone screen when the call is bridged .. let me know if you need anything more .. |
| 14:51.59 | [TK]D-Fender | kamanashisroy: You haven't shown me an ATTEMPT or your code for trying it the way I suggested. |
| 14:52.29 | proppy | Hi, is there a way to test a connection to an asterisk server from the command line ? |
| 14:52.50 | kamanashisroy | proppy: what type of connection ? |
| 14:52.51 | [TK]D-Fender | proppy: what kind of "connection"? |
| 14:52.53 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca) |
| 14:53.03 | huey23 | [TK]D-Fender: i love you <3 |
| 14:53.17 | proppy | like a softphone connection but with comprehensive log and arguments |
| 14:53.47 | [TK]D-Fender | proppy: "like a softphone?" Yuo want something "like a softphone... the go RUN ONE. |
| 14:54.30 | proppy | [TK]D-Fender: want something like ekiga or twinkle but from commandline and with log I gues |
| 14:54.45 | proppy | I'm trying to implement http://www.voip-info.org/wiki/view/RolePlayingTwoPhonesTwoRooms |
| 14:55.06 | [TK]D-Fender | proppy: Just do what you're trying to do and debug that. |
| 14:55.19 | proppy | I've set the asterisk server, but I lack a way of testing it, since I've no voip phone with my right now |
| 14:55.55 | [TK]D-Fender | proppy: Fine. Go set up a softphone on another PC to test it |
| 14:55.55 | proppy | [TK]D-Fender: with which client are you suggestiong that I connect to the asterisk server ? |
| 14:56.11 | [TK]D-Fender | proppy: take your pick. Hardly matters which |
| 14:56.11 | proppy | sorry If my question sound stupid I'm very new to this area |
| 14:56.39 | proppy | [TK]D-Fender: do you know a softphone that produce connection/error logs ? |
| 14:56.41 | [TK]D-Fender | ~softphone |
| 14:56.42 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 14:56.58 | proppy | [TK]D-Fender: thanks |
| 14:57.02 | kamanashisroy | ~pastebin |
| 14:57.03 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 14:57.07 | [TK]D-Fender | proppy: You don't need the softphone to make any kind of logs. Look at ASTERISK's output to see what problems arise |
| 14:57.25 | proppy | [TK]D-Fender: good idea |
| 14:57.28 | [TK]D-Fender | proppy: and since you are sounding like you haven't even installed or tested one yet, you don't HAVE a problem to debug yet |
| 14:59.00 | proppy | [TK]D-Fender: thanks to AsteriskNOW and the configuration snippet of the howto it wasn't that hard to set |
| 14:59.16 | proppy | now I'm trying to connect to it, but ekiga hangs with no log |
| 14:59.49 | [TK]D-Fender | proppy: enable SIP DEBUG on * CLI and watch the traffic. |
| 15:01.11 | kamanashisroy | [TK]D-Fender: here is the application with description http://paste.uni.cc/19051 |
| 15:01.23 | kamanashisroy | sending you the agi it runs |
| 15:01.50 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:02.26 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 15:02.29 | Kobaz | http://www.pastebin.ca/1060547 |
| 15:02.44 | Kobaz | voicemail just completely dies if it has problems playing a voicemail emssage |
| 15:02.47 | Kobaz | is there any way to prevent that? |
| 15:02.53 | [TK]D-Fender | kamanashisroy: Thanks.. thats of no use. |
| 15:03.45 | [TK]D-Fender | Kobaz: COMPLETE CLI output for the call, max debug (dore & verbose) + channel (IAX2) |
| 15:03.46 | kamanashisroy | Kobaz: dies means ? asterisk goes down ? |
| 15:04.16 | kamanashisroy | [TK]D-Fender: do you like to see the script ? |
| 15:04.38 | [TK]D-Fender | kamanashisroy: You have not shown me your attempt to use the "M()" feature with Dial. |
| 15:04.50 | kamanashisroy | [TK]D-Fender: thanks |
| 15:04.52 | [TK]D-Fender | kamanashisroy: You instead decided to show me some junk in "C" |
| 15:05.16 | kamanashisroy | [TK]D-Fender: LOL .. it is not possible in M .. ! |
| 15:05.25 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:05.43 | kamanashisroy | [TK]D-Fender: thanks for talk |
| 15:05.55 | [TK]D-Fender | kamanashisroy: You said, call 1 guy. Then call out to another and let them press something to accept the call. Damn right this is possible with "M()". Now get off your ass and TRY IT |
| 15:06.00 | Kobaz | kamanashisroy: the channel hangs up |
| 15:06.05 | Kobaz | [TK]D-Fender: lemme get it |
| 15:06.26 | kamanashisroy | [TK]D-Fender: that is not the end |
| 15:06.46 | mond0 | [TK]D-Fender, the two phones I've configured that can call each other are a Polycom SoundPoint IP300 SIP and a Grandstream GXV-3000. They can both call each other. NetMeeting can call either and when I pick up, NetMeeting shows the call established. When I hang up, NetMeeting plays a tone immediately. Just no audio... |
| 15:06.47 | kamanashisroy | [TK]D-Fender: not near ! |
| 15:06.52 | kamanashisroy | [TK]D-Fender: thanks |
| 15:07.56 | [TK]D-Fender | mond0: pastebin your configs masking only passwords, and CLI output with SIP/H.323 debug enabled |
| 15:08.16 | mond0 | Furthermore, I have tested the NetMeeting client machine that its microphone works by recording an audio file with sndrec32.exe and the speakers work because I can hear stuff. |
| 15:08.43 | mond0 | [TK]D-Fender, okay. |
| 15:09.08 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:09.14 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:09.25 | [TK]D-Fender | mond0: And don't forget that * sucks at H.323 and should be avoided. Also you have not specified codecs. |
| 15:10.21 | Kobaz | [TK]D-Fender: http://www.pastebin.ca/1060549 |
| 15:10.25 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 15:10.27 | *** join/#asterisk spokra (n=spokra@host093-179-132.sea0.speakeasy.net) |
| 15:10.44 | tompaw | am I missing something in here: [admin@asterix ~]$ sudo asterisk -rx database show |
| 15:10.47 | tompaw | No such command 'database' |
| 15:10.47 | proppy | ALT-F9 to switch to *CLI |
| 15:10.47 | proppy | sip set debug :) |
| 15:10.50 | tompaw | ? |
| 15:10.56 | proppy | is there a way to attach the CLI*> from a standart login connection ? |
| 15:11.03 | Kobaz | [TK]D-Fender: i hit voicemail, hit 1 to listen to old messages.... and it goes to play the first message (which is a wav file with just a header), and it hangs up the channel |
| 15:11.23 | [TK]D-Fender | Kobaz: What ver of *, what device? |
| 15:11.32 | Kobaz | iax, and 1.4.14 |
| 15:11.49 | [TK]D-Fender | tompaw: do "asterisk -r" and run it there. |
| 15:12.01 | [TK]D-Fender | tompaw: and you need to put "database show" in quotes. |
| 15:12.15 | [TK]D-Fender | Kobaz: I do recommend you upgrade. Thats somewhat dated now. |
| 15:12.18 | tompaw | thats's it. thx. |
| 15:12.31 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:12.35 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 15:12.42 | [TK]D-Fender | Kobaz: you didnt' provide the complete call, no IAX debug, etc... |
| 15:12.47 | Kobaz | [TK]D-Fender: yeah, but this is a 100k calls a month box, we haven't finished doing testing on the newest asterisk |
| 15:13.02 | Kobaz | [TK]D-Fender: oh iax debug too, hmm |
| 15:13.45 | [TK]D-Fender | Kobaz: Figure I'd see more on core debug 10... |
| 15:14.00 | Kobaz | [TK]D-Fender: it is the complete call as far as set verbose 3 is concerned |
| 15:14.06 | Kobaz | let me try and duplicate the problem locally |
| 15:14.20 | [TK]D-Fender | Kobaz: core 10, verbose 10 <- |
| 15:14.22 | Kobaz | i cleaned up the non-working voicemail messages for that box |
| 15:14.35 | Kobaz | well i saved a copy |
| 15:15.35 | proppy | found out, asterisk -vvvvvr to attach the CLI*> |
| 15:15.50 | *** join/#asterisk adr3nalin3 (n=info@72-164-59-242.dia.static.qwest.net) |
| 15:18.04 | *** join/#asterisk NovceGuru (n=NovceGur@oh-65-40-70-180.sta.embarqhsd.net) |
| 15:19.38 | macros73 | Having an issue with NAT. My server incorrectly sees internal peers as being behind NAT, and external peers (my trunks, actually) as not being behind NAT. Thus, I can receive incoming calls but not make calls to the trunk. |
| 15:20.37 | Kobaz | [TK]D-Fender: http://www.pastebin.ca/1060560 |
| 15:20.43 | Kobaz | there it be |
| 15:20.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:21.04 | *** part/#asterisk mogie (i=opera@ppp121-44-216-213.lns1.hba1.internode.on.net) |
| 15:21.09 | Kobaz | [TK]D-Fender: all kinds of verbosity |
| 15:22.08 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:22.36 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:22.44 | macros73 | Nevermind, easy fix, added 'nat=yes' to the trunk's peer details |
| 15:27.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:28.07 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-92.revip2.asianet.co.th) |
| 15:29.00 | [TK]D-Fender | Kobaz: What codec is the call in? formats = 0x4 <- not sure what that is. And if you listen to a different message instead, does it work? (is the message itself corrupted perhaps) |
| 15:31.36 | *** join/#asterisk MrNaz (n=naz@ppp121-44-233-18.lns2.mel4.internode.on.net) |
| 15:32.21 | jaytee | could someone tell me if this is a serious issue? It only started showing up on my * console when I started using a SIP TAPI plugin for Win XP to do click2dial from Outlook. http://pastebin.ca/1060568 |
| 15:32.41 | jaytee | everything works great except I get that warning message. |
| 15:33.16 | tompaw | yeah!! |
| 15:33.23 | tompaw | check this out > http://pastebin.com/m240e730a |
| 15:33.28 | tompaw | Now I only need the last line :-) |
| 15:34.17 | tompaw | AND, it seems to be working as of now ;-) |
| 15:35.05 | [TK]D-Fender | tompaw: .... um... looks like an infinite loop.;... what does that script do? |
| 15:35.55 | markid | tompaw: ;P |
| 15:35.55 | tompaw | [TK]D-Fender: it checks for the number in the Numbers Portability database, then it writes the target route ID back to asterisk's database |
| 15:36.14 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-1-97.revip2.asianet.co.th) |
| 15:36.17 | tompaw | it's a simple 'wget + asterisk -rx' bash thingie. |
| 15:36.27 | [TK]D-Fender | tompaw: Ok... STILL seems to just spin in circles. Does it run as a daemon? |
| 15:36.36 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 15:36.51 | tompaw | [TK]D-Fender: actually, no. that 'nl' script will change this value and end the loop |
| 15:38.08 | [TK]D-Fender | tompaw: Always? |
| 15:38.09 | tompaw | I just need to add some protection there, like [break the loop if waited more than 5 seconds] |
| 15:38.36 | tompaw | [TK]D-Fender: yes. it will return a physical termination route on success, or if the network is not supported by the gateway.... |
| 15:38.49 | tompaw | ...it will return a route that just plays a message "sorry, blah blah blah" |
| 15:38.49 | [TK]D-Fender | tompaw: unless its running as a daemon, you don't need * to loop for anything.. your script won't continue in the dialplan till its finished |
| 15:38.52 | tompaw | (that's the plan) |
| 15:39.27 | tompaw | [TK]D-Fender: well, it's an asteriskNOW installation, i think it runs as a daemon, and the System() call is asynchronous. |
| 15:39.39 | [TK]D-Fender | tompaw: and if that script ever returns without changing that DB value it WILL loop indefinitely |
| 15:39.55 | [TK]D-Fender | tompaw: I'm talking about your SCRIPT running as a daemon. |
| 15:39.59 | tompaw | [TK]D-Fender: I agree, that anti-infinity protection must be added. |
| 15:40.04 | defswork | Jul 2 15:35:45 WARNING[18411] chan_zap.c: Detected alarm on channel 4: Red Alarm < any ideas what this might be ? |
| 15:40.20 | [TK]D-Fender | tompaw: right now there seems to be little poitn to a check, OR returning a DB value like that. |
| 15:40.49 | tompaw | [TK]D-Fender: I don't understand. what do you mean? |
| 15:41.31 | [TK]D-Fender | tompaw: Is your script starting some background process or is it just linar? |
| 15:41.37 | [TK]D-Fender | linear* |
| 15:41.50 | tompaw | [TK]D-Fender: I assumed that the System() call is asynchronous |
| 15:41.56 | tompaw | (Goes to background) |
| 15:41.59 | [TK]D-Fender | tompaw: if its just linear then there is no need to loop anything. |
| 15:42.04 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:42.06 | [TK]D-Fender | tompaw: NO. |
| 15:42.25 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:42.28 | tompaw | [TK]D-Fender: really? 100% sure? |
| 15:42.29 | [TK]D-Fender | tompaw: System does not execute anything in the background. * waits till it is done before continuing your call. |
| 15:42.33 | [TK]D-Fender | tompaw: yes I'm sure. |
| 15:42.44 | tompaw | OK then, I get rid of the loop |
| 15:42.47 | mond0 | [TK]D-Fender, for my NetMeeting issues, which configs might be pertinent? h323.conf and... |
| 15:44.06 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 15:44.47 | [TK]D-Fender | mond0: Whatever you set up. |
| 15:44.53 | tompaw | [TK]D-Fender: the last thing for me is to choose a provider basing on that value returned by the 'nl' script. I assume I should add those providers to asterisk first, then check their trunk ids, and then just use that 'trunkdial' macro, right |
| 15:44.57 | tompaw | ? |
| 15:45.10 | [TK]D-Fender | tompaw: Go try stuff. |
| 15:45.38 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:46.15 | tompaw | :) |
| 15:46.54 | proppy | strange asterisk seems to receive SIP packet with non-nated IP |
| 15:47.24 | [TK]D-Fender | proppy: PASTEBIN is your friend.... |
| 15:48.41 | proppy | http://pastebin.com/md5c72b8 |
| 15:48.43 | proppy | sorry :) |
| 15:51.01 | *** join/#asterisk billybigpotatoes (n=billybig@83-244-133-169.cust-83.exponential-e.net) |
| 15:51.28 | billybigpotatoes | hi - anyone using ices? |
| 15:51.36 | billybigpotatoes | with ezstream? |
| 15:51.43 | [TK]D-Fender | proppy: You've clearly not set your system up properly to ahndle NAT. Go read the guides : |
| 15:51.45 | [TK]D-Fender | ~sipnat |
| 15:51.46 | jbot | [~sipnat] Quick guide on configuring * + 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 |
| 15:52.05 | ManxPower | [TK]D-Fender: one of these days I'm going to write a good guide for NAT stuff. |
| 15:52.08 | [TK]D-Fender | proppy: And while you're at it, your credentials are wrong : SIP/2.0 401 Unauthorized |
| 15:52.19 | [TK]D-Fender | ManxPower: the first link there covers it. |
| 15:52.19 | jaytee | could someone tell me if this is a serious issue? It only started showing up on my * console when I started using a SIP TAPI plugin for Win XP to do click2dial from Outlook. http://pastebin.ca/1060568 |
| 15:52.21 | ManxPower | <PROTECTED> |
| 15:52.51 | [TK]D-Fender | jaytee: Single lines like that is meaningless junk |
| 15:52.58 | [TK]D-Fender | jaytee: FULL DEBUG DAMMIT |
| 15:53.35 | ManxPower | [TK]D-Fender: the first one is better than I expected. |
| 15:53.59 | [TK]D-Fender | ManxPower: deserves a minor tweak when i can bother to get around to it. |
| 15:54.13 | ManxPower | jaytee: that message usually means "client went away" |
| 15:54.40 | [TK]D-Fender | jaytee: or could mean "I have networking issues" |
| 15:54.50 | ManxPower | that's just a subset of "client went away" |
| 15:55.08 | [TK]D-Fender | ManxPower: or "client? What client?!" ;) |
| 15:55.10 | proppy | [TK]D-Fender: thanks for the link |
| 15:55.39 | [TK]D-Fender | proppy: and next time include the FULL output, not just from the middle of the call. |
| 15:55.56 | proppy | [TK]D-Fender: sorry :) |
| 15:56.11 | jaytee | [TK]D-Fender, ManxPower thanks. I only see it for a very brief period and not on every click2dial. If it appears more frequently I'll grab sip debug info on it and do a pastebin at some future point. |
| 15:56.35 | proppy | seems that I'm in the 4. case Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk |
| 15:56.58 | [TK]D-Fender | proppy: go read the FIRST link |
| 15:56.58 | tompaw | guys I need help with the dialplan grammar. Look at this macro call: exten=_4X!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid}) - let's assume I have this trunk's id (1) in separate variable. How should I write this line then? |
| 15:57.15 | tompaw | can I use... ${trunk_${TID}} ? |
| 15:57.31 | proppy | [TK]D-Fender: oups |
| 15:59.46 | WildPikachu | [TK]D-Fender, only 1 problem today with a transfer tone not being picked up that i'll investigate, but the callprogress=yes seems to of fixed all the problems :( |
| 16:00.09 | billybigpotatoes | anyone having success with app_ices with lame and ezstream - as per http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES - i am seeing broken pipe on asterisk v1.4.18 |
| 16:00.35 | *** join/#asterisk L-info (n=L-info@g0962184.demon.co.uk) |
| 16:01.02 | cabel | Just so you guys know I did end up fixing my problem. The problem was that files sip.conf, features.conf, extensions.conf, iax.conf already existed and freepbx wouldn't overwrite them. I deleted them and freepbx was able to create them. I think when I compiled and did make samples it screwed it up some how. So yes you were correct that it was in fact the guis fault that it was not writing to those files.... If you guys are interested |
| 16:01.37 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 16:02.13 | [TK]D-Fender | cabel: \o/ Back to FreePBX. Move along now :p |
| 16:02.49 | proppy | When following http://www.aocomputing.net/?p=3, I get http://pastebin.com/m4015a64c |
| 16:03.30 | cabel | Yea, well the problem is I'm leaving here after the summer so I can't install asterisk because the next user will not know what to do. So I have to stick with a gui |
| 16:03.42 | [TK]D-Fender | proppy: that isn't a NAT issue. Thats an "I didn't set up my auth right' issue |
| 16:03.56 | [TK]D-Fender | cabel: "That's nice" |
| 16:04.21 | [TK]D-Fender | cabel: Hope they like it and can deal with it all |
| 16:05.02 | proppy | [TK]D-Fender: maybe I miss some deny=0.0.0.0/0.0.0.0 permit=0.0.0.0/0.0.0.0 |
| 16:05.06 | proppy | to allow connection from all peers |
| 16:05.22 | [TK]D-Fender | proppy: remove BOTH |
| 16:05.59 | cabel | [TK]D-Fender: Yea well you'll be happy to know that I am planning on building an asterisk server for home use and not use any guis cause it's too many head aches and problems. So hopefully you'll see me around int the future :) |
| 16:06.04 | proppy | [TK]D-Fender: ok |
| 16:06.06 | mond0 | I'm having trouble using NetMeeting, but am very close. I'm using a stock install of AsteriskNOW which I haven't altered much. #asterisknow is kind of quiet, so I |
| 16:06.26 | [TK]D-Fender | mond0: Again you are providing NO backup. |
| 16:06.39 | jaytee | #asterisknow is usually quiet, #freepbx is usually like a morgue |
| 16:06.40 | mond0 | 'm here. Here's my h323.conf and a log of calls from NetMeeting to 2 phones. They connect, just no audio. http://pastebin.com/d2cdaa770 |
| 16:07.10 | proppy | [TK]D-Fender: thanks now its getting better, :) http://pastebin.com/m61f7e5f1 |
| 16:07.21 | proppy | I should just not try to register but only make a call I guess |
| 16:07.57 | [TK]D-Fender | mond0: H323 over NAT? Good luck with that... You'd be wise to ditch Netmeeting |
| 16:08.33 | tompaw | [TK]D-Fender: you were right. The System() call is synchronous. |
| 16:09.06 | mond0 | NAT? |
| 16:09.32 | mond0 | I know what NAT is, and we're not using it. |
| 16:09.57 | [TK]D-Fender | mond0: and Found peer capability G.723.1 <5>, Asterisk code is 1, frame size (in ms) is 120 <-- onse side says ULAW/GSM, and this is the OTHER. * can't transcode G.723. |
| 16:10.04 | [TK]D-Fender | mond0: Fix your codec selection. |
| 16:10.42 | tompaw | woot woot! it's working!! http://pastebin.com/m6be00ce0 |
| 16:11.07 | tompaw | 3 lines of code in 3 hours. not bad :> |
| 16:11.37 | [TK]D-Fender | tompaw: No, not bad at all for a start like this. |
| 16:11.45 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-181-18.dsl.tul2ok.sbcglobal.net) |
| 16:11.55 | [TK]D-Fender | mond0: Again, why are you using NetMeeting? |
| 16:12.21 | mond0 | It was requested. |
| 16:13.25 | proppy | adding host=dynamic fixed the registration issue |
| 16:13.29 | mond0 | We have a bunch of classrooms with "smart boards" but no phones. The smart boards are running XP and people apparently got tipped off that Asterisk can supposedly handle H.323, so they got to thinking why not just use the board to make calls, instead of installing phones... |
| 16:13.32 | proppy | now I get this http://pastebin.com/m51a9b260 |
| 16:14.07 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:14.08 | jaytee | what the hell is a "smart board"? |
| 16:14.15 | tompaw | [TK]D-Fender: thanks, I'll focus on that portability lookup now :) |
| 16:14.25 | proppy | wonder which extension is it talking about |
| 16:14.42 | mond0 | It's like a big touch screen thing. I have't even used it, I just know they have NetMeeting installed. |
| 16:14.53 | [TK]D-Fender | mond0: install a SIP softphone instead. |
| 16:15.01 | [TK]D-Fender | mond0: * works like SHIT with H.323 |
| 16:15.16 | mond0 | ha. okay. |
| 16:15.33 | mond0 | Recommendations for a softphone? |
| 16:16.01 | jaytee | some people like X-lite but your mileage may vary |
| 16:16.02 | *** part/#asterisk d1mas (n=chatzill@host-99.dataart.net) |
| 16:17.02 | [TK]D-Fender | mond0: X-Like / Zoiper |
| 16:17.05 | [TK]D-Fender | X-Lite* |
| 16:17.12 | *** join/#asterisk zwsegal (n=zwsegal@209.208.68.200) |
| 16:17.32 | zwsegal | hello, have a question about PRI behavior, any takers? |
| 16:17.51 | jaytee | my PRIs are very well behaved. |
| 16:17.55 | sysreq | ~ask |
| 16:17.56 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:18.21 | zwsegal | danke |
| 16:18.44 | proppy | seems that I forgot to properly set context= to a valid extension |
| 16:19.04 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:19.05 | zwsegal | my PRI has been reporting FCS errors and aborts randomly, esp during thunderstorms, is that typical? |
| 16:19.29 | zwsegal | how many FCS errors and aborts and d-channel drops should i consider tolerable? |
| 16:19.35 | [TK]D-Fender | proppy: And "context=" is not set to an "extension" |
| 16:19.47 | [TK]D-Fender | zwsegal: 0 :) |
| 16:20.10 | coppice | zwsegal: ask Baron von Frankenstein to disconnect it from the lightning rods |
| 16:20.17 | zwsegal | [TK]D-Fender ha, so i was right to assume paetec is just a crappy provider |
| 16:20.33 | zwsegal | coppice then hed want to connect them to me :P |
| 16:20.51 | zwsegal | so lightning and whatnot shouldnt be affecting the line as much as it is? |
| 16:20.55 | coppice | nothing's perfect |
| 16:21.08 | tompaw | <PROTECTED> |
| 16:21.18 | tompaw | Google finds only 3 answers, none in english. |
| 16:21.19 | proppy | [TK]D-Fender: context should correspond so something valid in extensions.conf doesn't it ? |
| 16:21.31 | proppy | I mean context= in sip.conf |
| 16:21.35 | [TK]D-Fender | tompaw: Don't jsut look at the error, look at what CAUSED it. |
| 16:21.50 | jaytee | zwsegal, we have frequent lightening and thunderstorms here in the midwest and my PRIs don't drop D channel or have FCS errors due to it but they come in via fiber not copper. |
| 16:21.56 | [TK]D-Fender | proppy: Yes, it should point to a valid context in extensions.conf... |
| 16:22.38 | zwsegal | jaytee ah i expect that would make a difference as copper would be more susceptible to electrical interference |
| 16:22.41 | tompaw | Right. that would be the lack of mpeg123, which moves the question to #asteriskNOW |
| 16:22.51 | jaytee | certainly would |
| 16:23.02 | zwsegal | anyone else use PAETEC? |
| 16:23.03 | proppy | [TK]D-Fender: understood that was my mistake |
| 16:23.31 | jaytee | even over copper if the lines are properly shielded you shouldn't have problems like that unless your telco sucks ass |
| 16:23.34 | [TK]D-Fender | proppy: A mistake, but not one responsible for the error you showed us |
| 16:23.59 | [TK]D-Fender | tompaw: What are you actually trying to accomplish? |
| 16:24.41 | zwsegal | jaytee i was operating under the same assumption. i have a feeling it might be the telco :P |
| 16:24.46 | *** join/#asterisk cabbiepete_ (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:27.20 | tompaw | [TK]D-Fender: play mp3. it requires mpg123. I want to install it using that fancy 'conary' package manager not to mess with the deps and stuff. |
| 16:28.11 | tompaw | but there's not much I can do, since: <ProtocolError for https://rmirror.digium.com/conary/: 500 Internal Server Error> |
| 16:28.14 | *** join/#asterisk gpowers (n=glenn@adsl-67-38-31-125.dsl.sfldmi.ameritech.net) |
| 16:28.17 | [TK]D-Fender | tompaw: No. To just play an mp3 file you should install "asterisk-addons", which will give you "format_mp3.so". |
| 16:29.01 | fogo | I'm getting all outbound calls on one T1 being dropped - the telco doesn't like that I'm sending "Info=1device" in the setup message. Any idea what option may remove that? |
| 16:29.37 | tompaw | [TK]D-Fender: sorry mate, I relied on the MP3Play() documentation, which says it requires mpg123. |
| 16:29.49 | [TK]D-Fender | tompaw: Then don't |
| 16:30.02 | tompaw | [TK]D-Fender: anyway, the whole conary repo from Digium seems to be down :/ |
| 16:30.16 | [TK]D-Fender | tompaw: rpath = meh |
| 16:31.31 | proppy | [TK]D-Fender: Yes you're right it seems unrelated to my error, since I get the same pb |
| 16:31.32 | proppy | http://pastebin.com/m78173b3 |
| 16:32.10 | proppy | when pointing to the correct context/extension |
| 16:33.35 | [TK]D-Fender | proppy: To: <sip:A@88.191.77.88> Looking for A in roleplaying (domain 88.191.77.88) SIP/2.0 404 Not Found |
| 16:33.53 | [TK]D-Fender | proppy: yeah... the error is pretty much standing in plain sight. Now go fix it ;) |
| 16:34.45 | proppy | yep |
| 16:35.07 | proppy | just figured out that my Dial line when bad in extensions.conf |
| 16:35.07 | proppy | lines |
| 16:35.14 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:35.49 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 16:36.08 | Kobaz | [TK]D-Fender: regarding my voicemail issue... the codec is ulaw |
| 16:36.17 | tompaw | Strange. It seems like *now has everything that's required to BUILD asterisk-addons :) |
| 16:36.32 | Kobaz | [TK]D-Fender: the message is definitly corrupted... there is just a wav header with no audio contents |
| 16:36.44 | [TK]D-Fender | Kobaz: That distro is nothing but trouble. |
| 16:36.56 | [TK]D-Fender | Kobaz: Better to build it for yourself on something more solid |
| 16:37.19 | Kobaz | [TK]D-Fender: which distro? |
| 16:37.28 | Kobaz | [TK]D-Fender: the built in voicemail app? |
| 16:37.53 | [TK]D-Fender | Kobaz: Sorry, wasn't meant to be directed at you. |
| 16:37.57 | Kobaz | heh |
| 16:38.09 | billybigpotatoes | anyone want to talk about app_ices? |
| 16:38.17 | Kobaz | [TK]D-Fender: the voicemail app shouldn't hang up the channel if the voicemail wav is bad |
| 16:38.23 | [TK]D-Fender | Kobaz: have you tested if its just that one singular message, or does it bomb on every and any message? |
| 16:38.39 | Kobaz | [TK]D-Fender: it works fine if i delete the corrupted wav and it plays a good one |
| 16:38.54 | [TK]D-Fender | Kobaz: Does * still create corrupted wavs? |
| 16:39.10 | Kobaz | [TK]D-Fender: not sure if this wav from asterisk 1.2 or from 1.4.14 |
| 16:39.24 | Kobaz | [TK]D-Fender: we upgraded from 1.2 several months ago |
| 16:39.47 | [TK]D-Fender | Kobaz: Ok, if it looks like a 1-off, I'd write it off. |
| 16:40.05 | Kobaz | [TK]D-Fender: but this has come up more than once... |
| 16:40.09 | [TK]D-Fender | BRB, rebooting. |
| 16:40.13 | proppy | [TK]D-Fender: aaaah, just found it |
| 16:40.23 | proppy | extension is not the same that an username |
| 16:40.38 | proppy | username is used for registerting |
| 16:40.42 | proppy | extension for calling |
| 16:41.47 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 16:43.13 | tompaw | I assume I have to "enable" format_mp3.so somewhere in the conf files, right? |
| 16:43.30 | ManxPower | No. The only extension is in extensions.conf. sip.conf contains devices |
| 16:43.39 | ManxPower | tompaw: download it first |
| 16:44.26 | tompaw | how about modules.conf? |
| 16:44.31 | tompaw | ManxPower: I just built it. |
| 16:44.59 | ManxPower | tompaw: there was not a "make install" to install it. |
| 16:45.07 | proppy | yep and the dialplan (extensions.conf) is the extra mapping between extension and sip action ? |
| 16:45.08 | ManxPower | ? |
| 16:45.18 | tompaw | ManxPower: there was. |
| 16:45.22 | ManxPower | proppy: for the 2nd time. They are sip devices. |
| 16:45.26 | ManxPower | tompaw: and did you run it? |
| 16:45.34 | tompaw | ManxPower: but that just put the module in appropriate directory. It didn't mess with the .conf files at all |
| 16:45.52 | tompaw | I'll try adding "load=format_mp3.so" to modules.conf |
| 16:46.06 | ManxPower | tompaw: you don't need to. just restart asterisk, it should then show up in "show formats" or show modules |
| 16:46.26 | ManxPower | you understand that format_mp3 is not app |
| 16:46.32 | ManxPower | app_mp3, right? |
| 16:46.38 | proppy | ManxPower: and extension are the number you dial ? |
| 16:46.52 | ManxPower | proppy: correct. The extension is the number you dial. |
| 16:47.07 | proppy | ManxPower: thanks for clearing that up |
| 16:47.14 | ManxPower | We, for example, make the SIP username be the same as the MAC address of the device. |
| 16:47.21 | tompaw | asterix*CLI> module show like mp3 > shows only app_mp3.so |
| 16:47.28 | tompaw | it doesn't list format_mp3.so :/ |
| 16:48.05 | tompaw | I did 'load module...' |
| 16:48.06 | ManxPower | tompaw: Does g729 and g723.1 show up in show modules? |
| 16:48.17 | ManxPower | tompaw: fine. "core load format_mp3.so" |
| 16:48.35 | ManxPower | But I will not help you further unless you do what I ask. |
| 16:48.39 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 16:48.43 | Mike8861 | hello all |
| 16:48.43 | *** join/#asterisk neurosys0 (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:48.50 | ManxPower | I asked if it showed in "show formats" and you totally ignored me. |
| 16:49.04 | tompaw | No such command 'show formats' (type 'help' for help) |
| 16:49.21 | ManxPower | try adding "core" in front of the command like almost all other 1.4 commands |
| 16:49.31 | tompaw | sorry, I thought you meant to use show formats OR show modules |
| 16:49.42 | tompaw | No such command 'core show' (type 'help' for help) |
| 16:50.08 | ManxPower | Geeeze. I'm going to have to find a 1.4 box a min. |
| 16:50.30 | Mike8861 | ManxPower: what command are u trying |
| 16:50.36 | Mike8861 | ManxPower: maybe i can help ? |
| 16:51.37 | tompaw | Mike8861: I am trying to enable mp3 playback. I manually loaded format_mp3.so with 'module load format_mp3.so' |
| 16:51.43 | ManxPower | Mike8861: he is trying to use format_mp3 |
| 16:51.45 | tompaw | Now it appears in the modules list, yet... |
| 16:51.57 | Mike8861 | ManxPower: i dunno about asterisk, in trixbox(asterisk 1.4) we used to use "core show file formats" for this |
| 16:52.07 | ManxPower | format_mp3 won't let you play mp3 files. |
| 16:52.08 | tompaw | -- Executing [551@numberplan-custom-3:2] MP3Player("SIP/6969-b590df78", "/home/admin/heart.mp3") in new stack |
| 16:52.10 | Mike8861 | ManxPower: for MP3 Playback, mplayer is easy way, or |
| 16:52.16 | tompaw | NOTICE[8730]: app_mp3.c:118 timed_read: Poll timed out/errored out with 0 |
| 16:52.19 | ManxPower | Mike8861: I don't want to hear about Trixbox |
| 16:52.22 | tompaw | that's the error I keep receiveing. |
| 16:52.29 | Mike8861 | ManxPower: show module |
| 16:52.50 | ManxPower | Mike8861: thank you for volunteering to help tompaw |
| 16:52.54 | ManxPower | I can get back to paying work now. |
| 16:52.56 | Mike8861 | ManxPower: try that out, its preety easy, and let me know the result |
| 16:53.12 | tompaw | Mike8861: try what? |
| 16:53.31 | tompaw | D-Fender said that "To just play an mp3 file you should install "asterisk-addons", which will give you "format_mp3.so"." |
| 16:53.52 | Mike8861 | tompaw: core show file fomats |
| 16:53.59 | Mike8861 | tompaw: show modules |
| 16:54.26 | Mike8861 | tompaw: installing mp3, will just result play back in monotone 8Khz g711u |
| 16:54.51 | Mike8861 | tompaw: asuuming u are using g711u as your codec, so i see no point for installing mp3 support |
| 16:55.01 | tompaw | Mike8861: ok, let's do it your way, to get the best quality possible. "core show file formats" returns: |
| 16:55.37 | tompaw | http://pastebin.com/m6f1d4737 |
| 16:56.22 | tompaw | 'show modules' returns more than 160 modules, however http://pastebin.com/m617c3225 |
| 16:56.36 | hardwire | hmmm.. I have a pool of iaxmodem's .. now how to dial them as a group.. |
| 16:57.07 | Mike8861 | tompaw: i will leave that for ManxPower |
| 16:57.16 | hardwire | It seems like that would be a bad idea.. a bunch of modems trying to answer at once then having to reset themselves when they didn't win. |
| 16:58.05 | Mike8861 | tompaw: unless u have technical issue with transcoding, it is recommand easy way to play back music |
| 16:58.42 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 16:58.50 | Mike8861 | tompaw: if u need a transcoder, i can provide u, which i am currently using(its legal) |
| 16:58.56 | tompaw | Mike8861: what do you mean by 'it'? |
| 17:00.26 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:03.18 | Mike8861 | tompaw: download transcoder software, this software can transcode (almost any, including mp3) to WAV |
| 17:03.23 | Mike8861 | tompaw: http://holas.myweb.hinet.net/MediaConvert.7z |
| 17:03.45 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:04.06 | Mike8861 | tompaw: choose desire codec, config bitrate etc, select file to transcode, click transcode(right most button) |
| 17:04.48 | Mike8861 | any one would like to help with dialplan problem ?? |
| 17:05.42 | Mike8861 | i am trying to dial 9000(int-ext), when connected to 9000, it will dialout to a external IVR, and send DTMF to ext ivr |
| 17:05.47 | Mike8861 | http://www.pastebin.sk/en/7242 please help |
| 17:08.17 | kaldemar | Mike8861: core show application dial => option D() |
| 17:09.07 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 17:09.16 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 17:09.21 | DarnoQ | hi! what could be that cause of situation that i can't login with softphone to any user ? i also don't see the information that softphone is trying to connect in CLI |
| 17:10.43 | Mike8861 | kaldemar: i have read document before asking, having trouble to understand. |
| 17:11.17 | Mike8861 | kaldemar: technology should be SIP, no doubt, no idea whats identifier and where goes the destination phone number..... |
| 17:12.05 | kaldemar | Mike8861: Dial(SIP/trunk1${EXTEN},,D(<dtmf_sequence>)) |
| 17:13.38 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:14.29 | Mike8861 | kaldemar: the result will be , exten => _91878200,n, Dial(SIP/trunk1${EXTEN},,D(<dtmf_sequence>)) ??? |
| 17:15.01 | Mike8861 | i want the user to dial 9000, and it forward the call to 1878200 and send some DTMF |
| 17:15.06 | kaldemar | if that exten line matches, yes |
| 17:15.28 | Mike8861 | you are a genius!!! i will try it out now |
| 17:15.33 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 17:17.25 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 17:18.28 | trafim | Hi guys. I have a little prob, mb you know the solution. I need to implement night-schedules for support team, so every night different operator (with different number) will answer. The point is that it must be roundrobin. Like, five guys, and it should every evening switch to another number, not depending on day of week. |
| 17:19.55 | trafim | Now i implemented night schedule through pack of GotoIfTime, but it's connected to day. And it must not be. |
| 17:21.59 | [TK]D-Fender | trafim: Then set a DB value to the counter position number for what it should be doing that day. You can automate the incremening via cron, etc, or have someone increment it manually. |
| 17:22.32 | Mike8861 | kaldemar: i have updated the dialplan, however its not working as expected |
| 17:22.38 | Mike8861 | heres the updated: http://www.pastebin.sk/en/7265/ |
| 17:23.05 | [TK]D-Fender | Mike8861: exten => exten => _91878200,n, Dial(SIP/trunk1${EXTEN},,D(12)) c<- PRDON? |
| 17:23.26 | [TK]D-Fender | Mike8861: you mashed 2 lines together and are referencing EXTEN as part of your PEER. |
| 17:24.13 | Mike8861 | [TK]D-Fender: my bad!!!!!i deserve this |
| 17:24.42 | trafim | [TK]D-Fender: i already thought about some script, but wanted to do it via dialplan.. well then, tnx. |
| 17:25.12 | [TK]D-Fender | trafim: You can do it via dialplan logic entirely with a little trickery. |
| 17:25.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:26.10 | trafim | [TK]D-Fender: er.. that would be good. can you give some hints? |
| 17:26.41 | Mike8861 | [TK]D-Fender: i have remove the duplicate exten=> however its still not working |
| 17:27.01 | [TK]D-Fender | Mike8861: and you didn't show me what you thought "fixed" should look like either. |
| 17:27.25 | Mike8861 | [TK]D-Fender: http://www.pastebin.sk/en/7267/ |
| 17:27.33 | [TK]D-Fender | trafim: "core show function DB" |
| 17:28.07 | Mike8861 | [TK]D-Fender: in addition, I have confirm that my dialplan will actually runsi have tried with playback(you-sound-cute) ! |
| 17:28.20 | [TK]D-Fender | Mike8861: exten => _91878200,n, Dial(SIP/trunk1${EXTEN},,D(12)) <- you can't jsut shove an "n" priority for a DIFFERENT exten in the middle of a bunch of "s" exten priorities like that. |
| 17:28.48 | [TK]D-Fender | Mike8861>[TK]D-Fender: in addition, I have confirm that my dialplan will actually runsi have tried with playback(you-sound-cute) ! <- thanks... this means NOTHING <- |
| 17:29.40 | trafim | [TK]D-Fender: uh. okay, tnx anyway. |
| 17:29.53 | Mike8861 | [TK]D-Fender: should i change it to all N or all S ?? |
| 17:30.39 | [TK]D-Fender | Mike8861: work on ONE exten at a time. "s" will haev NOTHING to do with "_91878200" |
| 17:30.55 | proppy | [TK]D-Fender: ManxPower: thanks a lot |
| 17:32.31 | Mike8861 | [TK]D-Fender: by EXTEN, you mean the first parameter right ? |
| 17:32.44 | [TK]D-Fender | trafim: Make a DB entry for the "entry person" to cal and another for the date last changedl. Every time you try to call that person, check the date to see if it is the same. If not, increment (and loop around if necessary. Then once you've determined the person (in sequence to call) call them |
| 17:33.18 | Mike8861 | [TK]D-Fender: therefore I replace all 3 lines from exten => s, .... to exten => _91878200, ... ??? |
| 17:33.46 | trafim | [TK]D-Fender: okay i'll try to implement this. thanks. |
| 17:36.12 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 17:39.35 | *** join/#asterisk quentusrex (n=quentusr@c-71-197-244-228.hsd1.or.comcast.net) |
| 17:40.01 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:40.11 | quentusrex | can someone help me figure out why my sip phone is reporting 404 for registration. |
| 17:40.50 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 17:47.46 | huey23 | quentusrex: it might not be able to find the SIP server, what program are you running? |
| 17:48.14 | quentusrex | I'm using the x-lite softphone on one laptop, and a grandstream phone |
| 17:48.33 | huey23 | what SIP server? |
| 17:49.05 | quentusrex | pbxinaflash |
| 17:49.13 | jaytee | roflcopter |
| 17:49.36 | Qwell | ~freepbx |
| 17:49.37 | jbot | [~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 |
| 17:49.40 | huey23 | did you apply changes after you added extensions? |
| 17:50.26 | quentusrex | well, then if that's how you all feel about freepbx. What is a good replacement? |
| 17:50.32 | huey23 | :) |
| 17:50.37 | quentusrex | huey23, yes I did. |
| 17:50.54 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 17:51.07 | jaytee | quentusrex, what's the name of this channel? |
| 17:51.23 | huey23 | install debian, download and compile asterisk, start reading...that's what [TK] made me do :P |
| 17:52.56 | huey23 | you don't need all of that extra crap that comes with those bundled software programs, if you look at a couple of dialplans, you'll figure it out real quick |
| 17:54.45 | quentusrex | how do I debug a sip connection? |
| 17:55.09 | quentusrex | my grandstream is registered, but it can't dial anything |
| 17:55.21 | huey23 | no one knows, i played around with the PIAF and FreePBX for about 1 hour and junked both of them |
| 17:56.07 | quentusrex | I use to use asterisk@home |
| 17:56.16 | huey23 | well, quit it |
| 17:56.19 | quentusrex | then it became trixbox, then I dumped it... |
| 17:57.03 | _ShrikE | ~centos52bug |
| 17:57.04 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 17:57.31 | *** join/#asterisk hi365_m (n=hi365@213.151.56.96) |
| 17:57.38 | jaytee | what's a good cordless VOIP phone? |
| 17:58.29 | huey23 | cisco 7920, different protocol but good quality |
| 17:59.06 | jaytee | thanks, but I want SIP. Don't want to mess with skinny or sccp |
| 17:59.26 | huey23 | ahh, you did not specify :) |
| 17:59.26 | Qwell | jaytee: ata + cordless phone |
| 17:59.41 | *** join/#asterisk cool_judge (n=topper_h@207.230.238.94) |
| 18:00.18 | jaytee | Qwell, thanks. I was thinking about that but the only ATA I've tried is the Grungestream Handytone 286 with a Panasonic cordless. Lotta hum and noise. |
| 18:00.52 | jaytee | Qwell, what's a good quality ATA? Linksys/Sipura? |
| 18:02.58 | huey23 | IVR, is there any addons or scripts that use this to ask the caller where they want to be transfered to and they can respond by voice? |
| 18:04.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:06.17 | [TK]D-Fender | jaytee: Linksys ATAs are decent enough |
| 18:06.35 | [TK]D-Fender | jaytee: I've got cordless phones & my Polycom SoundStation 2W running off of them quite well. |
| 18:07.12 | [TK]D-Fender | huey23: to respond by voice means Voice Recognition. So either Lumenvox or Sphinx |
| 18:07.35 | cool_judge | Hello everyone. I have a question about astagidir in astarisk.conf. This variable is supposed to hold the location of the agi folder. However, when I try to change it, the new value does NOT take effect after restart - Asterisk still looks in /usr/share/asterisk/agi-bin. Any ideas??? |
| 18:07.44 | huey23 | [TK]D-Fender: i'll check them out, thanks |
| 18:07.51 | jaytee | [TK]D-Fender, I hate the Handytone audio quality. I'd like to go with the Polycom Spectralink 8002 but my boss cringes at the price and is looking at an Aastra 480i |
| 18:08.10 | [TK]D-Fender | quentusrex: Feel free to use whatever Web GUI frontend you want, just know that 99% of your questions here will get a "It's not supported here" answer. |
| 18:08.50 | quentusrex | alright |
| 18:09.06 | [TK]D-Fender | jaytee: Aastra has decent battery, very good range and clarity. BUT... its terminally tied the regsitrations of the base. it is indeed to be considered a fixed "extension" o fthe base and not separate. |
| 18:09.11 | quentusrex | then my question is where are the asterisk logs. and how do I debug a sip connection? |
| 18:09.34 | jaytee | [TK]D-Fender, yeah and a bit pricey just to get cordless option |
| 18:09.35 | [TK]D-Fender | quentusrex: "sip debug" at * CLI. This is the place to go for debugging SIP issues. |
| 18:10.02 | jaytee | plus I'd rather stick with ATA's with analong phones or Polycom only for SIP phones. |
| 18:10.05 | [TK]D-Fender | jaytee: Yeah, shitty to ahve a fixed phone as a receiver. Seimens SIP DECT stuff is said to be pretty decent if you can find it affordably. |
| 18:10.10 | rwaite | i like watching the iax2 debug stuff scroll by. it soothes the mind. |
| 18:10.10 | huey23 | [TK]D-Fender: voip-info says that the accoustic model for sphinx is 8khz and not available in * |
| 18:10.37 | [TK]D-Fender | huey23: Plenty of guides showing how to integrate Sphinx & *. Go loko. |
| 18:10.41 | [TK]D-Fender | look* |
| 18:10.42 | jaytee | I was wondering about the IAXy ATA from Digium but I think I'm going to recommend getting a Linksys to test. |
| 18:11.05 | huey23 | kk |
| 18:12.20 | [TK]D-Fender | jaytee: ... you don't want that kind of pain :) |
| 18:13.05 | jaytee | I'm not really using IAX now. I was but that was just temporary and it wasn't much fun. |
| 18:14.42 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:19.00 | *** join/#asterisk angler_ (n=angler@216.207.245.1) |
| 18:21.00 | jaytee | [TK]D-Fender, any particular Linksys ATA model you'd recommend? I don't need FXO so I'm looking at the SPA-2102 or the PAPT2-NA |
| 18:21.17 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 18:23.29 | [TK]D-Fender | jaytee: If you think there's any chance of sending the ATA somewhere remote, get the SPA-2102 as it has a built in router, etc. Might be worth the extra 10$ no matter what. |
| 18:33.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:34.18 | jaytee | ok, my boss just ordered an Astra. the model that replaces the 480i. I really love being micromanaged :-( |
| 18:34.38 | jaytee | time to update the resume and start looking again, hate doing that at my age. |
| 18:35.35 | [TK]D-Fender | jaytee: Ycuk. I have an 57i CT myself.... |
| 18:35.50 | [TK]D-Fender | jaytee: Means well, but many physical "bleh" factors |
| 18:36.27 | jaytee | if only the Spectralink from Polycom was a little cheaper in price :-( |
| 18:37.21 | jaytee | I had a panic attack because he called me into his office and started showing me printouts from websites and the phones on them were Snom. |
| 18:37.36 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:37.50 | jaytee | the man has no frigging clue about supporting large telecom installs. |
| 18:39.06 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:39.17 | [TK]D-Fender | jaytee: this is where its good to have a CTO with power to match the title. |
| 18:39.41 | [TK]D-Fender | jaytee: So that general managers don't screw your company over with tech mistakes. |
| 18:40.33 | jaytee | [TK]D-Fender, the problem is that this guy is the IT Director |
| 18:40.38 | rwaite | chief tautology officer? |
| 18:40.56 | [TK]D-Fender | rwaite: Of course he's always right! |
| 18:41.04 | [TK]D-Fender | jaytee: Stock up on K-Y... |
| 18:41.13 | jaytee | tautology? that's for truth tables, when has an executive ever relied on the truth :-) |
| 18:41.28 | rwaite | lol |
| 18:41.31 | rwaite | falseoloy |
| 18:41.56 | jaytee | [TK]D-Fender, he actually asked me if I'd run Dr Watson to debug a problem with an application that would terminate unexpectedly. It was running on Windows XP. |
| 18:43.20 | jaytee | and he insisted we buy IP phones with an extra port for the computer so we didn't have to have more cat5 runs. I mentioned that if it became an issue and we wanted to VLAN the traffic that might be a hindrance. He insists we'll never need to VLAN voip traffic. |
| 18:44.13 | jaytee | so essentially yeah....... I need a railcar tanker full of K-Y |
| 18:45.40 | *** join/#asterisk Netgeeks (n=chris@204-16-157-174-static.ipnetworksinc.net) |
| 18:45.42 | [TK]D-Fender | jaytee: I prefer saving $ on the phones and investing in infrastructure personally. |
| 18:46.03 | [TK]D-Fender | jaytee: prevents you from getting tied down to any one solution. |
| 18:46.28 | [TK]D-Fender | jaytee: And minimizes blead-over risks as well as performance loss for PC's |
| 18:46.49 | patrick-- | meeb [TK]D-Fender :) |
| 18:46.52 | patrick-- | how is life these days? |
| 18:47.00 | [TK]D-Fender | patrick--: Getting by. Slowly... |
| 18:47.12 | patrick-- | thats not much, but sth. |
| 18:50.06 | jaytee | anyone running CentOS with * ? should I disable SELINUX or run it permissive? |
| 18:51.33 | huey23 | nope, deb |
| 18:53.20 | huey23 | and it works great :) |
| 18:54.06 | [TK]D-Fender | jaytee: Disable |
| 18:57.45 | rwaite | love debian |
| 18:58.11 | *** join/#asterisk naitram (n=naitram@216.77.58.40) |
| 18:59.11 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 19:00.54 | beek | jaytee: I use CentOS 5 -- love it. Do yourself a favor and disable SELINUX. I use SELINUX everywhere except on my * boxes. |
| 19:02.04 | huey23 | [TK]D-Fender: ok, sphinx it is |
| 19:02.26 | jaytee | beek, thanks. I've set it to disabled. That's what I'd originally done when I setup my first Red Hat server for * back in Nov 07. |
| 19:02.32 | [TK]D-Fender | huey23: Lumenvox is relatively inexpensive and far better.... |
| 19:02.47 | jaytee | and [TK]D-Fender thanks again |
| 19:03.23 | huey23 | [TK]D-Fender: i just looked at lumenvox and it seems that they went a little to far on the "sales pitch" |
| 19:03.26 | jaytee | Lumenvox only runs on 32 bit though so be forewarned. I'm running 64 bit RHEL 5 so I'm setting up a 32 bit box with * as my IVR |
| 19:03.34 | [TK]D-Fender | huey23: Your call. |
| 19:04.25 | jaytee | huey23, there's always Nuance! (roflcopter) |
| 19:04.37 | huey23 | [TK]D-Fender: i am researching lumwnvox a little more, just for you |
| 19:07.00 | [TK]D-Fender | huey23: I've never tried either personally, just recounting common opinion. |
| 19:12.19 | *** join/#asterisk skyggen (n=edward@rrcs-67-52-199-30.west.biz.rr.com) |
| 19:12.27 | skyggen | hello, hello. |
| 19:12.52 | skyggen | God AT&T are poky |
| 19:13.34 | skyggen | got a question about zt_pri_error unknown 500. Anyone had this happen? |
| 19:13.58 | huey23 | your hardware just took a dump :P |
| 19:14.08 | skyggen | serios |
| 19:14.21 | skyggen | so switching out the card would fix this? |
| 19:16.44 | [TK]D-Fender | skyggen: What card? What server? What signalling? What version of * & Zaptel? |
| 19:22.14 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:25.53 | huey23 | skyggen: ? |
| 19:28.21 | skyggen | TE205P zaptel-1.4.11 libpri-1.4.4 pri_cpe asterisk-1.4.11 |
| 19:30.35 | russellb | $ ./changes_since asterisk 1.4.11 |
| 19:30.35 | russellb | Changes since asterisk Version 1.4.11/ - svn revision 80193 |
| 19:30.36 | russellb | 1222 |
| 19:30.51 | russellb | Please update Asterisk to the latest version. If you still have trouble, please contact Digium technical support. |
| 19:30.53 | nny_1 | i am running a box with ztdummy as the timing source. When calling the Timeout function i get Function Timeout not registered. i am currently compiling each from it's own set of directories. (src/zaptel/zaptl-(version)/ and src/asterisk/asterisk-(version). I am recompiling * now, but could that directory structure be casuing issues with asterisk compiling for zap? |
| 19:31.25 | russellb | calling what timeout function? |
| 19:32.23 | nny_1 | actually misworded on my part, the line of code in the dialplan is exten => s,5,Set(TIMEOUT(digit)=2) |
| 19:32.48 | nny_1 | this is code from before i took over |
| 19:33.16 | nny_1 | i guess the app would be set in this case |
| 19:34.17 | huey23 | russellb: where are the GUI screenshots? |
| 19:34.50 | [TK]D-Fender | skyggen: I don't believe that you should be using that version of * and zaptel together... |
| 19:35.28 | [TK]D-Fender | skyggen: Zaptel is far newer than your * version. Bring them more into sync. |
| 19:35.31 | skyggen | what about libpri version? |
| 19:35.52 | [TK]D-Fender | skyggen: I'd suggest going to the latest "release" versions as per the topic |
| 19:36.09 | russellb | huey23: eh? which one? |
| 19:36.33 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 19:36.35 | nny_1 | hmm i am gonna look this over more. I am probably not grasping the entire concept of how this is implemented |
| 19:36.45 | huey23 | i thought that was your blog on the website |
| 19:36.52 | nny_1 | although asterisk is most definitely not aware of zap |
| 19:37.01 | russellb | it might be. the links might be broken now, if so, nothing i can do |
| 19:37.01 | nny_1 | recompiling |
| 19:37.37 | huey23 | the only one thats broken is the one to the screenshots |
| 19:38.03 | nny_1 | i guess the question is does zap and asterisk need to be compiled in the same directory? |
| 19:38.24 | russellb | huey23: there might be some on digium.com or asterisknow.org somewhere |
| 19:43.46 | nny_1 | nm lol figured it out |
| 19:45.52 | naitram | I get all 'busy here message' from a sip client within asterisk cli when called device is busy. How do I capture that within the dial plan? |
| 19:46.03 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 19:46.18 | [TK]D-Fender | naitram: ${DIALSTATUS} |
| 19:46.28 | nny_1 | if i have a box that is hardware less (no card) how do i know asterisk sees* zap for ztdummy? |
| 19:46.48 | *** join/#asterisk doolph (i=doolph@190.141.70.138) |
| 19:46.52 | doolph | hello |
| 19:47.37 | doolph | has anyone tried install asterisk on a HP ML110 |
| 19:48.06 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 19:48.14 | [TK]D-Fender | nny_1: "zap show status" |
| 19:48.39 | nny_1 | [TK]D-Fender: hmm zap application not being loaded |
| 19:48.45 | [TK]D-Fender | doolph: What problems have you encountered, and what other hardware involved? |
| 19:48.54 | [TK]D-Fender | nny_1: huh? |
| 19:48.54 | nny_1 | but i have compiled one before the other.. may be the subdirs each is compiled in |
| 19:49.01 | jeev | sup Fender |
| 19:49.04 | tompaw | [TK]D-Fender: hi mate, would you mind helping me a little more? that mp3... I built asterisk-addons and manually loaded format_mp3.so, but the error's still there |
| 19:49.12 | nny_1 | No such command 'zap' (type 'help zap' for other possible commands) |
| 19:49.25 | [TK]D-Fender | nny_1: * has to be recompiled after zaptel is installed for zaptel support to be compiled in. |
| 19:49.31 | naitram | [TK]D-Fender: thx |
| 19:49.38 | [TK]D-Fender | tompaw: don't use "mp3player", tec. use |
| 19:49.39 | doolph | well there's no problem yet, I have ordered the HP ML110 already, now where I purchase the Digium T1 card they told me that might not be compatible |
| 19:49.42 | [TK]D-Fender | "Playback" |
| 19:50.03 | tompaw | with that format_mp3.so module loaded, right? |
| 19:50.09 | [TK]D-Fender | doolph: Some HP's have had issue with Digium cards. Then again, they should be able to RMA it if you have issues |
| 19:50.15 | [TK]D-Fender | tompaw: Clearly. |
| 19:50.20 | nny_1 | [TK]D-Fender: yeah i just did that, but each is being compiled into it's own subdir of /src/ which i think is the issue |
| 19:50.48 | [TK]D-Fender | nny_1: No. Compile * install zaptel. INITIALIZE IT. Then compile * install *. |
| 19:51.01 | nny_1 | will try again thanks |
| 19:51.01 | [TK]D-Fender | nny_1: Do this off freshly extracted tarballs. |
| 19:51.11 | nny_1 | yeah it was |
| 19:51.13 | [TK]D-Fender | jeev: Getting by. |
| 19:51.18 | *** join/#asterisk bmg505 (n=leon@196-209-8-58-ndn-esr-2.dynamic.isadsl.co.za) |
| 19:52.54 | tompaw | [TK]D-Fender: http://pastebin.com/m2191db52 |
| 19:53.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:53.54 | [TK]D-Fender | tompaw: Quick lesson : you never provide the EXTENSION of the audio file to play back. * will match it automatically |
| 19:54.04 | tompaw | [TK]D-Fender: quick thx. |
| 19:55.05 | tompaw | [TK]D-Fender: workin fine! |
| 19:55.11 | [TK]D-Fender | tompaw: Glad to hear. |
| 19:57.06 | tompaw | [TK]D-Fender: and the quality is amazingly good, too! |
| 19:57.48 | [TK]D-Fender | tompaw: keep in mind that MP3 is a lot better than the lowest common telephony codec (G.711) at 8khz mono |
| 19:58.09 | [TK]D-Fender | tompaw: So you're definitely not losing anything, but it DOES put a transcoding load on your CPU |
| 19:58.29 | tompaw | [TK]D-Fender: Sure. |
| 19:58.42 | [TK]D-Fender | tompaw: You really shouldn't use MP3 unless you can't avoid it. best to convert them to more native formats (since that's what * will do to them live anyways) |
| 19:59.41 | tompaw | That's what I wanted to do anyway. Right now I'm just playing with it. The functionality of * is amazing, as today shown. I was almost sure that using some kind of API (AGI) will be a must for that number lookup. |
| 19:59.57 | tompaw | And it was all down to 3 lines of "code"! |
| 20:01.23 | [TK]D-Fender | tompaw: You seem to be doing well. I suspect that you will grow quickly with this. |
| 20:01.32 | tompaw | (-: |
| 20:01.41 | macros73 | Has anyone here used Asterisk as a SIP proxy? I have a situation where I would like to have an Avaya IP Office route certain calls to an Asterisk server via SIP. Asterisk would then handle routing the call out from there. |
| 20:02.08 | [TK]D-Fender | macros73: No. Asterisk is NOT a "SIP Proxy" |
| 20:02.50 | tompaw | But can be used as one. Yet it would be much faster to use something dumb like VoipSwitch for that thing in my opinion. |
| 20:03.10 | [TK]D-Fender | macros73: * is a B2BUA which should be able to serve as an intermediary server to bridge your Avaya to whatever external resource you were looking for however. |
| 20:03.38 | [TK]D-Fender | tompaw: No, it is not a proxy. At all. * does not "pass on" anything from one leg to another. |
| 20:03.50 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:04.08 | [TK]D-Fender | tompaw: A talks to *. * talks to B. |
| 20:04.38 | macros73 | Which would work for this admittedly obtuse path we might follow. |
| 20:05.50 | macros73 | Avaya IP Office supports find-me follow-me (but called mobile twinning.) IPO tries to send the calling number CLID so that the mobile phone can see the originating number, and not the BTN. Our LEC blocks sending CLID that aren't specifically on our account. |
| 20:06.15 | macros73 | Our ITSP does not, so long as it is not abused. |
| 20:06.56 | nny_1 | [TK]D-Fender: ok compiled zap from /root/packages-2008-06-27/src/zaptel/zaptel-1.4.11 and then loaded it. I got a message stating ztdummy was loaded, as this box has no card, and then ztcfg failed for the same reason. (although /etc/zaptel.conf exists.) I then compiled asterisk from /root/packages-2008-06-27/src/asterisk/asterisk-1.4.21 and started it. I would understand my mistake if zap show anything in CLI is supposed to be missing if only ztdummy is use |
| 20:06.58 | macros73 | So the thought is: route mobile twinning calls from IPO --> *, and * then bridges the call via the ITSP trunk |
| 20:07.05 | *** join/#asterisk Infested (i=Infested@oracle.riverstreet.intellitechsolutions.com) |
| 20:07.31 | tompaw | [TK]D-Fender: right. But it can pretend to be one, can't it? |
| 20:07.36 | macros73 | IPO passes the original CLID to *, and * passes it on to the phone. Problem solved, until we get around to a more permanent fix. |
| 20:07.37 | [TK]D-Fender | tompaw: No. |
| 20:07.47 | [TK]D-Fender | macros73: Sure, why not. |
| 20:08.26 | *** part/#asterisk naitram (n=naitram@216.77.58.40) |
| 20:09.22 | macros73 | [TK]D-Fender: That's what I'm here to ask, if anyone sees red flags (besides the unnecessary layer of complexity) with this. |
| 20:09.36 | [TK]D-Fender | macros73: No, thats fine |
| 20:10.00 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-201-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:10.46 | huey23 | [TK]D-Fender: do you have a personality or do you talk just like you type? |
| 20:11.20 | huey23 | [TK]D-Fender: i'm just wondering |
| 20:11.53 | macros73 | huey23: Did your mom drop you alot as a toddler? I'm just wondering. :D |
| 20:12.43 | ManxPower | nny_1: 1.4.21 has major bugs, use 1.4.21.1 |
| 20:12.51 | huey23 | [TK]D-Fender: probably, i won't dispute your insinuations |
| 20:13.01 | nny_1 | ManxPower: good to know, will update that today |
| 20:13.04 | huey23 | sorry, macros |
| 20:14.04 | [TK]D-Fender | huey23: Nice backhanded insult there :) I'm not even going to qualify that with anything more. |
| 20:14.20 | macros73 | Don't worry, I already qualified it. |
| 20:14.40 | huey23 | [TK]D-Fender: macros73 tried :P |
| 20:15.40 | macros73 | Codec question: Informally, a blind survey of our users shows they rate ulaw calls as a 5, gsm 2-3, and g729a as 4-4.5. (Ekiga soft phone, ulaw to *, * transcoded to target codec and sent via our ITSP). Do those ratings jive with what I should expect? |
| 20:15.58 | [TK]D-Fender | huey23: Just sparing you the kind of wrath I'm fully capable of :) |
| 20:16.22 | *** join/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 20:16.22 | huey23 | [TK]D-Fender: i'm well aware...please don't eat me |
| 20:16.43 | [TK]D-Fender | huey23: I wouldn't want another case of food-poisoning anyways |
| 20:17.07 | [TK]D-Fender | macros73: Its that a "scale of perceived audio quality"? |
| 20:17.08 | huey23 | [TK]D-Fender: yea, you don't want that kind of wrath i am capable of :) |
| 20:17.19 | [TK]D-Fender | values his intestinal tract |
| 20:17.31 | macros73 | [TK]D-Fender: Yeah, I just asked them to rate the voice quality from 1-5, 5 highest. |
| 20:18.06 | [TK]D-Fender | macros73: You should always use like codec on the highest you can afford. G.711 is really a good idea... |
| 20:18.16 | huey23 | [TK]D-Fender: i am not sure how to take the previous comment :0 |
| 20:18.45 | [TK]D-Fender | huey23: trace our commentary, I'm sure you'll see it shortly... |
| 20:19.19 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:19.29 | macros73 | [TK]D-Fender: G.711 is my preference, but constraints may force us to use a tighter codec. |
| 20:19.29 | huey23 | [TK]D-Fender: i was there...i was wondering if you were refering to your colon or your actual intestinal tract |
| 20:20.00 | [TK]D-Fender | huey23: Everything that fall victim to food-poisoning :) |
| 20:20.32 | [TK]D-Fender | huey23: Guess I could include any other system that would back-fire due to that as well, but hey... start at the source! |
| 20:20.51 | huey23 | [TK]D-Fender: yea yea, point your finger |
| 20:21.07 | macros73 | Let's not discuss food poisoning, I ate at Wendy's today and I'm expecting the blood to flow anytime now. |
| 20:21.40 | huey23 | macros73: i ate authentic mexican food yesterday, i think some just dribbled out |
| 20:21.58 | macros73 | Okay, no more use of the word "dribble" for huey23. |
| 20:22.04 | huey23 | :) |
| 20:22.34 | huey23 | i used that yesterday and i think everyone thought it was a little much |
| 20:23.27 | huey23 | i knew someone would appreciate it here |
| 20:23.29 | macros73 | Yeah, there are words that don't flow well together. "Anal", "dripple" is one such tuple. |
| 20:24.25 | *** join/#asterisk masus (n=ethemc@78.162.24.94) |
| 20:24.39 | huey23 | macros73: i am glad you like it, you can use it and form it to your liking, it is GNU/GPL |
| 20:24.45 | nny_1 | lol |
| 20:24.59 | [TK]D-Fender | macros73: Actaully... is all about the "flow" ;) |
| 20:25.06 | nny_1 | registers analdripple.com.. what?! taken? |
| 20:25.12 | huey23 | flow is good |
| 20:25.18 | macros73 | lol. |
| 20:25.24 | masus | hi all, have the same problem like this one -> http://bugs.digium.com/view.php?id=12269&nbn=7 does anyone know that it's fixed ? |
| 20:25.40 | huey23 | nny_1: check the whois, registered to huey23 :P |
| 20:25.43 | nny_1 | hahaha |
| 20:25.46 | masus | my asterisk-V Asterisk 1.4.21.1 |
| 20:26.22 | nny_1 | so i have a script that wgets asterisk-current. is there one considered "asterisk-stable" or is the 1.4.21 issue uncommon? |
| 20:26.37 | nny_1 | or is 1.2 stable in digium's eyes? |
| 20:26.38 | masus | Or does anyone know how to fix it? Thanks all. |
| 20:26.43 | *** join/#asterisk [Outcast] (n=bill@203-114-166-26.dsl.sta.inspire.net.nz) |
| 20:27.02 | nny_1 | (i remember they havce astbusiness on 1.2 for that reason, or did as of the beginning of this year or so) |
| 20:27.19 | [TK]D-Fender | nny_1: Shouldn't auto-dl any particular version. |
| 20:27.38 | quentusrex | ok, now I'm using asterisk. |
| 20:27.46 | [TK]D-Fender | nny_1: If you want to automate system deployment, you should manage your own repo for it. |
| 20:27.49 | quentusrex | How do I troubleshoot no inbound audio? |
| 20:28.18 | [TK]D-Fender | quentusrex: Typical problem is not being configured to operate behind NAT. Does this sound like your case? |
| 20:28.19 | huey23 | quentusrex: what phone? |
| 20:28.24 | nny_1 | [TK]D-Fender: yeah i am 1/2 in 1/2 out on that idea. Need to start doing that. May be the next step I take, although my experiece with building, creating and setting up packages for yum/rpms is limited |
| 20:28.46 | quentusrex | it could be nat, it's a grandstream phone |
| 20:29.00 | [TK]D-Fender | quentusrex: Read up : |
| 20:29.02 | [TK]D-Fender | ~sipnat |
| 20:29.02 | jbot | [~sipnat] Quick guide on configuring * + 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 |
| 20:29.04 | [TK]D-Fender | ^^^^^^ |
| 20:29.06 | huey23 | not familiar with granddribble phones |
| 20:29.38 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 20:29.47 | huey23 | i love voip-info, we had sex 2 days ago |
| 20:30.31 | macros73 | Hmm. I have allow=gsm&ulaw for this trunk, but it's defaulting to ulaw. How do I make it default to g729, but use ulaw if all g729 licenses are in use? |
| 20:30.45 | quentusrex | which ports need to be forwarded |
| 20:30.47 | outtolunc | allow=gsm,ulaw |
| 20:30.51 | outtolunc | not & |
| 20:30.56 | macros73 | Oh. Duh, thanks. |
| 20:32.33 | macros73 | Actually, that didn't appear to work. I have allow=g729,ulaw but it defauled to ulaw again. I did reload first. |
| 20:32.53 | WildPikachu | kicks his atcom 530 phone, darn thing is saying its my gateway IP when trying to register |
| 20:33.54 | macros73 | ...and NOW it works |
| 20:34.06 | *** join/#asterisk qdk (n=qdk@87.48.132.28) |
| 20:34.47 | [TK]D-Fender | quentusrex: read the guide, its all in there,. |
| 20:35.00 | quentusrex | <PROTECTED> |
| 20:35.10 | [TK]D-Fender | macros73: there is no way to do codec selection overflow based on available licenses. |
| 20:35.20 | quentusrex | how do I fix the anonymous |
| 20:35.37 | [TK]D-Fender | quentusrex: What should it be? |
| 20:35.49 | quentusrex | I don't know... that just doesn't seem right |
| 20:36.21 | [TK]D-Fender | quentusrex: If you have no idea what it should be, either don't worry about it, or think harder. |
| 20:36.27 | quentusrex | :) |
| 20:36.56 | masus | ManxPower: are u there ? |
| 20:37.44 | huey23 | alright, i'm out |
| 20:37.47 | huey23 | have a good one |
| 20:37.50 | [TK]D-Fender | huey23: later |
| 20:37.53 | huey23 | i love you |
| 20:37.56 | huey23 | :) |
| 20:38.22 | *** part/#asterisk huey23 (n=yea@64.192.209.34) |
| 20:40.44 | [TK]D-Fender | ok, heading home. Later all |
| 20:47.38 | quentusrex | what does this mean: -- Executing [202@from-sip-external:1] NoOp("SIP/anonymous.invalid-0a001278", "Received incoming SIP connection from unknown peer to 202") in new stack |
| 20:48.08 | quentusrex | I'm calling from one internal extension 201 to another internal extension 202 |
| 20:53.58 | *** join/#asterisk mrweasel (n=4stfed@core.spokanecomputing.com) |
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| 20:57.54 | nny_1 | hrrm |
| 20:57.54 | nny_1 | ast_func_write: Function Timeout not registered |
| 20:58.30 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 20:58.32 | nny_1 | from exten => s,4,Set(maintimeout=0) exten => s,5,Set(TIMEOUT(digit)=2) exten => s,6,Set(TIMEOUT(response)=10) |
| 20:58.39 | nny_1 | worked before this version iirc |
| 21:00.12 | *** join/#asterisk x3rus (n=x3rus@201.220.136.117) |
| 21:00.36 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 21:01.25 | masus | an_agent.c line 880 (agent_hangup): Error releasing mutex: Operation not permitted,does anyone have experience with this error ? |
| 21:02.30 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:02.48 | nny_1 | core show function TIMEOUT lists it's use though |
| 21:04.19 | nny_1 | lol i do it again for DISA later and it works >< |
| 21:04.37 | masus | http://rafb.net/p/wnKQdx52.html please ;( |
| 21:06.47 | *** join/#asterisk anthm (n=anthm@mbd0736d0.tmodns.net) |
| 21:07.14 | WilliamK | patch for http://bugs.digium.com/view.php?id=12954 works - fixed the issue entirely |
| 21:07.24 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:08.49 | masus | WilliamK do u say to me ? |
| 21:08.58 | WilliamK | ? |
| 21:09.10 | WilliamK | was having the same issue with it crashing |
| 21:09.14 | nny_1 | meh Request to schedule in the past?!?! no cpu load, not using mpg etc any advice? |
| 21:12.18 | WildPikachu | this stupid atcom phone just doesn't want to register, asterisk keeps saying no matching peer found, i put in the pbx's ip, username and password ... what else could it be ... i wonder |
| 21:15.13 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:18.21 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:19.22 | cool_judge | Regarding my previous question, the problem was that asterisk.conf had a section label [global], instead of [directories]. Hope this info is useful to any of you. Bye for todat. |
| 21:19.23 | masus | http://rafb.net/p/wnKQdx52.html please ;( |
| 21:19.57 | *** part/#asterisk cool_judge (n=topper_h@207.230.238.94) |
| 21:20.11 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:22.07 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:22.50 | [TK]D-Fender | masus: What ver of *? |
| 21:25.25 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:25.30 | ManxPower | WildPikachu: |
| 21:25.35 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:25.47 | ManxPower | WildPikachu: I'm sure it says other stuff. Can you guess what we need now? That's right! A pastebin of the failure! |
| 21:25.50 | WildPikachu | working now ManxPower .... darn thing has "Phone Number" which is actually username |
| 21:26.07 | WildPikachu | throws it at ManxPower ..... kick it ! |
| 21:26.14 | ManxPower | WildPikachu: Welcome to the world of phones nobody but you use. |
| 21:26.22 | Yourname`` | Is there a way to disconnect a connected manager user from the CLI? |
| 21:26.28 | WildPikachu | its a stupid phone i'm testing for a dumb cheap client :) |
| 21:26.29 | ManxPower | Buy a decent phone that others use and someone might be able to help you in the future. |
| 21:26.40 | ManxPower | WildPikachu: cheap clients are expensive. |
| 21:26.46 | WildPikachu | they bought a snom today |
| 21:26.58 | ManxPower | That's a start, I guess. |
| 21:27.01 | ManxPower | ~phones |
| 21:27.02 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 21:27.09 | WildPikachu | i just wanted to see why this one is not working as it should ... well i found that one out ... cheap dumb phone |
| 21:28.00 | WildPikachu | btw, why not consider grandstream? |
| 21:28.12 | ManxPower | really for customers on a small budget I suggest the SPA-9XX phones, but for only a little more you can get a Polycom |
| 21:28.40 | WilliamK | if customer can't afford a poly 320, someones got issues |
| 21:28.48 | WilliamK | 320/330 |
| 21:28.56 | WilliamK | those were only 80.00 a while back |
| 21:29.01 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:29.01 | ManxPower | WildPikachu: because their developers are apparently meth heads, as they have never released firmware that is stable for most situations. With a grandstream you just keep trying different firmwares until you find on that more-or-less works for you. |
| 21:29.05 | *** join/#asterisk qdk (n=qdk@87.48.132.28) |
| 21:29.16 | WildPikachu | ah, i will avoid them |
| 21:29.30 | ManxPower | Their hardware is not great either, but at least it generally works |
| 21:29.58 | WilliamK | anyone found a good qual video phone that's reasonable in pricing? |
| 21:30.50 | masus | [TK]D-Fender: Asterisk 1.4.21.1 |
| 21:31.20 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:31.27 | [TK]D-Fender | masus: ok, no idea then |
| 21:31.59 | masus | [TK]D-Fender: Ok Thank U. |
| 21:32.41 | masus | have the same problem like this one -> http://bugs.digium.com/view.php?id=12269&nbn=7 |
| 21:32.43 | Yourname`` | Hello, Is there a way to disconnect a connected manager user from the CLI? |
| 21:32.56 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:33.56 | [TK]D-Fender | Yourname``: not from * |
| 21:34.24 | Yourname`` | [TK]D-Fender: Ok, cool. thanks.. I thought there might be some elusive CLI command. |
| 21:34.54 | [TK]D-Fender | Yourname``: "help" <- try typing that into * CLI and pay attention. |
| 21:35.30 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:35.33 | Yourname`` | That's what I meant when I said elusive, unless the command is escaping me when I typed help |
| 21:43.12 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 21:44.57 | jameswf-home | got a mac and thinks the world is about to end.... |
| 21:46.23 | mchou | drmessano: you around? |
| 21:50.47 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 21:52.26 | WildPikachu | best to send dtmf via sip_info or rfc? |
| 21:53.21 | [TK]D-Fender | WildPikachu: generally either. rfc if possible |
| 21:53.34 | WildPikachu | rfc is via rtp, right? |
| 21:54.19 | drwelby | Anyone run into static issues on Polycom phones that are only the caller's voice within their own handset? And does it totally randomly? |
| 21:54.35 | drwelby | I'm evaluating the phone, so I can't switch it out at the moment. |
| 21:54.44 | [TK]D-Fender | drwelby: Whats on the other end of the call? |
| 21:55.09 | drwelby | [TK]D-Fender: POTS |
| 21:55.32 | [TK]D-Fender | drwelby: generally... blame the card/line |
| 21:55.46 | [TK]D-Fender | drwelby: And next time "POTS" is not a complete answer |
| 21:55.58 | drwelby | Caller on other end doesn't hear it though. It's only in the headset. |
| 21:56.30 | [TK]D-Fender | drjust because its heard on the handset doesn't mean it STARTS form the handset |
| 21:56.31 | drwelby | And for the sake of completedness, there's an AA50 in the middle |
| 21:58.21 | drwelby | It's the only phone that exhibits this problem, and it's one of two Polycoms we're testing |
| 21:58.45 | bkruse | mvanbaak: you there? |
| 21:58.50 | drwelby | I don't suspect the handset, since it's occurs on maybe 10% of calls |
| 21:59.06 | [TK]D-Fender | drwelby: Do you ever get it phone-phone? |
| 21:59.44 | [TK]D-Fender | drwelby: What happens when you transfer this call to another phone? Does the problem follow? |
| 21:59.57 | drwelby | No, and phone-phone calls are strictly on our LAN. It's only on calls outside on POTS |
| 22:00.18 | drwelby | [TK]D-Fender: thank you, that's a good suggestion |
| 22:00.19 | [TK]D-Fender | drwelby: not quite what I was asking. |
| 22:00.32 | [TK]D-Fender | drwelby: So phone-phone calls are always fine? |
| 22:01.21 | drwelby | Phone-phone calls are always fine |
| 22:01.39 | mvanbaak | bkruse: gheh, ur lucky |
| 22:01.54 | mvanbaak | just booted my new osx 10.5 install ;) |
| 22:01.57 | *** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net) |
| 22:02.11 | [TK]D-Fender | drwelby: And call comes in from analog port on AA50 to Polycom. Polycom encounters "static". They then transfer that call to another handset. Do THEY get static? |
| 22:02.31 | mvanbaak | it got thru faster then I hoped. |
| 22:02.48 | mvanbaak | I thought I had to keep it running while going to bed, but it is all done already |
| 22:02.48 | drwelby | Ah, that has not been tested yet |
| 22:02.54 | mvanbaak | so yeah, I'm here ;) |
| 22:02.58 | drwelby | That's why I thanked you for that suggestion |
| 22:03.12 | drwelby | I will try that the next time the call goes sour |
| 22:03.16 | [TK]D-Fender | drwelby: because the likelyhood that I'd pin this on a Polycom phone is remarkably low. |
| 22:03.23 | [TK]D-Fender | drwelby: Do be thorough |
| 22:03.40 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 22:03.41 | drwelby | Yes, that's why I don't want to blame the phone |
| 22:05.50 | drwelby | All my other phones are Ciscos, so if the static travels then I suppose I need to look at the AA50 |
| 22:06.06 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 22:06.45 | masus | [TK]D-Fender: Have disabled 2 lines of code in chan_agent.c now it works, on line 880 and 1022. |
| 22:07.41 | masus | :) |
| 22:08.03 | masus | <PROTECTED> |
| 22:08.18 | [TK]D-Fender | masus: Do share you experience on the bug tracker. |
| 22:08.44 | *** join/#asterisk Segnale007 (n=Segnale0@host163-249-dynamic.23-79-r.retail.telecomitalia.it) |
| 22:08.49 | masus | i will but my english is not enough they will dont understand me i think |
| 22:09.56 | [TK]D-Fender | masus: I think you'd do fine. |
| 22:10.25 | masus | :) |
| 22:11.20 | Fuzix | Hello guys, could anyone point me in the direction of the answer to my next question? |
| 22:11.29 | Fuzix | Why is my Asterisk install saying: Warning! Asterisk is not thread safe. |
| 22:11.32 | Fuzix | and how do I fix it? |
| 22:14.20 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 22:14.54 | [TK]D-Fender | Fuzix: What version of *, and what OS? |
| 22:16.23 | Fuzix | Asterisk 1.4.21.1 on Debian 2.6.24-etchnhalf.1-686 |
| 22:17.08 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 22:19.03 | bkruse | jicksta!~ |
| 22:19.38 | *** join/#asterisk jicksta (n=jicksta@adsl-76-204-20-193.dsl.pltn13.sbcglobal.net) |
| 22:20.44 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.66.155) |
| 22:22.32 | jpcansa | hello, i´m having troubles transfering CID from out calls, when phone A gets an out call and then it trasnfer the call to phone B, phone B will get phone A´s CID |
| 22:22.45 | jpcansa | any idea on how to fix this? |
| 22:23.11 | [TK]D-Fender | jpcansa: Srtop doing attended transfers, and start doing blind transfers |
| 22:23.17 | [TK]D-Fender | stop* |
| 22:25.55 | jpcansa | Fender, but i need to talk to person in phone B before I transfer the call |
| 22:26.20 | ManxPower | jpcansa: The idea is that when you pick up the phone you will be talking to the person that shows on the callerid |
| 22:26.27 | mvanbaak | I'm going to get some sleep |
| 22:26.29 | mvanbaak | latero all |
| 22:26.41 | [TK]D-Fender | jpcansa: then call them first, terminate that call, then blind transfer them after |
| 22:27.08 | ManxPower | If that is a blind transfer then the CID should be the original CID, as that is who you are talking to.; If it is an attended transfer then you are talking to the person doing the transfer and so their callerid should show up |
| 22:27.33 | *** join/#asterisk philipp64 (n=chatzill@nat/microsoft/x-82a09e07e834694f) |
| 22:28.20 | jpcansa | so there is no way to transfer the CID of a call i got on attended transfer?? |
| 22:28.48 | ManxPower | jpcansa: Oh, there might be, but none of us care enough to figure it out for you. |
| 22:28.54 | implicit | jpcansa: yeah you can |
| 22:28.54 | [TK]D-Fender | jpcansa: Sure... you've got the source clode just like the rest of us |
| 22:28.59 | ManxPower | this is a VOLUNTEER channel, not paid support channel |
| 22:29.11 | implicit | ManxPower: chill he didn't ask you to code it up |
| 22:29.11 | [TK]D-Fender | code* |
| 22:29.45 | jpcansa | thx implicit |
| 22:29.57 | implicit | jpcansa, it depends on a few things, including the protocol you're using and whether the UAs support updating of the callerid |
| 22:30.03 | *** join/#asterisk RoyK (n=roy@ip-2-52-149-91.dialup.ice.no) |
| 22:30.19 | *** part/#asterisk RoyK (n=roy@ip-2-52-149-91.dialup.ice.no) |
| 22:30.37 | implicit | as well, in asterisk's case using sip, since it's a b2bua, how the sip channel driver handles attended transfers |
| 22:30.57 | implicit | so it's a bit of a complicated question |
| 22:31.42 | implicit | jpcansa: what kind of phones are you using? |
| 22:32.05 | jpcansa | spa942 and 962 |
| 22:32.07 | jpcansa | linksys |
| 22:32.30 | jpcansa | why depends on protocols? |
| 22:32.47 | implicit | well you're using sip, so don't worry about that |
| 22:32.58 | jpcansa | yeah |
| 22:33.22 | implicit | in SIP you can do it for sure but you will have to append some header fields during the re-invite |
| 22:33.29 | implicit | honestly it will be tough to do w/ asterisk |
| 22:33.47 | jpcansa | hmm I see |
| 22:33.51 | implicit | what do you do on your platform exactly? what is the call flow? |
| 22:34.08 | implicit | maybe something can be figured out |
| 22:35.00 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 22:35.52 | jpcansa | my setup: Digium TDM2400P with 10 pstn lines, all calls from the pstn lines are forwarded to an Operator(SPA962) from default context |
| 22:37.10 | implicit | to do what you want to do the easiest solu tion i can think of is to put an openSER in front of asterisk, use it for registration and on-net calls, use asterisk as your media server and gateway (handling REFER's and all that for transfers) |
| 22:37.35 | implicit | in openser you can write a config to append the proper P-Asserted-Identity, etc |
| 22:37.40 | implicit | to the messages you need |
| 22:38.14 | implicit | and to select the one you want from the DB, whether it be the transferer or original caller |
| 22:38.48 | implicit | still it's not simple, so if it's not really important to you it is probably not worth it |
| 22:38.57 | masus | by all |
| 22:39.27 | jpcansa | yeah, u´re right |
| 22:39.44 | implicit | especially since it seems you're running a pretty small operation |
| 22:39.48 | implicit | only 10 channels |
| 22:41.16 | jpcansa | yeah |
| 22:41.22 | JT | 10 FXO lines :( |
| 22:42.07 | jpcansa | i´m getting 6 more next week ;) |
| 22:42.19 | JT | why don't you get a T1? |
| 22:42.26 | implicit | he's in costa rica |
| 22:42.28 | implicit | they don't have T1s :) |
| 22:42.31 | jpcansa | right!!!! |
| 22:42.34 | JT | that sucks |
| 22:42.37 | implicit | E1s |
| 22:42.39 | implicit | hehe and thats 30 chans |
| 22:42.48 | JT | you can't get partial E1s there? |
| 22:42.59 | implicit | latin america is tough man |
| 22:43.03 | implicit | takes forever to do stuff |
| 22:43.06 | implicit | and people rip you off |
| 22:43.09 | jpcansa | no, we just have one telco here |
| 22:43.11 | implicit | i've had a lot of experience there |
| 22:43.32 | jpcansa | they dont offer much solutions |
| 22:43.35 | implicit | not like it is back here in the states |
| 22:43.50 | JT | i'm not in the states |
| 22:43.53 | jpcansa | even if they have the technology |
| 22:43.58 | implicit | i'm not saying |
| 22:43.59 | implicit | u are |
| 22:44.01 | implicit | i'm in the states :) |
| 22:44.15 | implicit | JT, you're an openser guy too, nice |
| 22:44.47 | JT | just a tip when you're a tourist overseas though, don't say "back in the states" as it annoys the shit out of everyone :D |
| 22:44.49 | implicit | I hadn't met you before |
| 22:45.01 | implicit | JT, i'm just saying it cause i'm here right now hahaha |
| 22:45.09 | implicit | overseas it doesn't make sense |
| 22:45.21 | jpcansa | LOL |
| 22:45.28 | implicit | anyway, lets get back to our discussion |
| 22:45.38 | JT | i've bumped into a few american tourists who were all "back in the states... blah blah blah" |
| 22:45.57 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 22:46.54 | implicit | lol |
| 22:47.00 | implicit | i know a lot of ignorant as hell americans |
| 22:47.21 | implicit | but i would criticize them for a lot more stpid comments than that |
| 22:47.48 | implicit | i had a friend who had never been out of the US and I went on a trip with him to south america, i wish i'd gone alone |
| 22:48.29 | implicit | jpcansa: send me a message if you want to talk about your issue more |
| 22:48.41 | drwelby | Loud American Syndrome? |
| 22:48.42 | implicit | jpcansa: maybe i can help you hack something quick together |
| 22:48.55 | jpcansa | thnks |
| 22:49.44 | implicit | i think the US is a great country, which others may or may not agree with. but still i think americans who say america's the best country but have never been out of the country are really stupid, how the hell do they know |
| 22:58.52 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:02.09 | [TK]D-Fender | America is a great country, the only problem is too many Americans ;) |
| 23:03.43 | Qwell | [TK]D-Fender: "At least it isn't Canada." |
| 23:03.59 | [TK]D-Fender | :) |
| 23:04.21 | Strom_M | well, you know what they say about Quebec |
| 23:04.26 | Qwell | nothing |
| 23:04.26 | Strom_M | it's like the Canada of Canada |
| 23:04.31 | Qwell | lol |
| 23:06.11 | Strom_M | "SI VOUS LICHEZ MON DICK JE PROMETTRAI D'ETRE MOINS FRANCAIS" said the Quebecois |
| 23:07.28 | Strom_M | "JOUET POUR CHATS!?" responded the tourist from Ottawa |
| 23:08.10 | Strom_M | "GARDEZ-VOUS AU REFRIGERADEUR" interjected the milk |
| 23:09.50 | Strom_M | Tune in next week for an all-new episode of "Strom's Terrible Stories En Francais" |
| 23:11.56 | [TK]D-Fender | "Il parles bilingue pour sauver du temps mon ostie!" |
| 23:13.23 | Strom_M | i don't know what "ostie" is |
| 23:13.40 | [TK]D-Fender | Strom_M: Semi-generic swear-word. |
| 23:13.46 | Strom_M | ah |
| 23:14.03 | Strom_M | what would be the rough equivalent in English? |
| 23:14.47 | [TK]D-Fender | Strom_M: "He speaks bilingually (simultaneously) to save time, shit!" |
| 23:14.52 | Strom_M | ah |
| 23:14.58 | [TK]D-Fender | Franglais <- |
| 23:15.15 | [TK]D-Fender | Strom_M: Classic "Elvis Gratton" (movie) |
| 23:16.21 | *** join/#asterisk eclark (n=eclark@75-164-243-153.ptld.qwest.net) |
| 23:17.03 | eclark | I'm having some problems making outbound SIP calls - specifically, audio doesn't seem to go through (but the call connects). Is this the place to ask? |
| 23:17.19 | Strom_M | eclark: SEE |
| 23:17.23 | Strom_M | ~sipnat |
| 23:17.24 | jbot | [~sipnat] Quick guide on configuring * + 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 |
| 23:17.26 | [TK]D-Fender | eclark: So far. Quick guess. Your *, or a remote phone behind NAT? |
| 23:17.54 | eclark | yeah, the asterisk box is behind nat. I have it DMZ'ed though |
| 23:17.55 | [TK]D-Fender | ccesario: If so follow the first guide linked above |
| 23:18.08 | [TK]D-Fender | eclark: Not enough. Follow the guide |
| 23:18.10 | eclark | checks link |
| 23:19.39 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:20.09 | JT | don't DMZ |
| 23:21.09 | Strom_M | gardez-moi dans le refrigerateur |
| 23:27.34 | *** part/#asterisk jpcansa (n=jpbenavi@201.201.66.155) |
| 23:28.34 | [TK]D-Fender | Strom_M: You'd love it... a week ago I hit this restaurant that must be the king of all paces to buy poutine from. They had an entire PAGE of the menu dedicated to something like 15 different kinds :) |
| 23:29.13 | Strom_M | omg |
| 23:29.17 | Strom_M | :( |
| 23:29.50 | JT | there's almost no restaurants here with poutine :( |
| 23:30.16 | [TK]D-Fender | JT: Where are you located again? |
| 23:30.36 | JT | sydney, australia |
| 23:31.11 | [TK]D-Fender | JT: Oh yeah.. sorry, for a split second I mixed you up with JayTee. Yeah I knoew where you are :) |
| 23:31.24 | [TK]D-Fender | JT: And yeah.. the commute for it would suck :p |
| 23:31.26 | JT | there is a chain of fast food outlets springing up in Melbourne that do poutine, but i doubt it's very authentic |
| 23:31.35 | JT | they call themselves Lord of The Fries |
| 23:31.42 | [TK]D-Fender | JT: But if you're even up this way, I'll buy you a beer :) |
| 23:32.03 | JT | one day that would be nice |
| 23:32.03 | JT | :) |
| 23:34.43 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:34.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:37.07 | Robba | Hey guys |
| 23:37.35 | Robba | i'm looking at sliming down my extensions.conf file |
| 23:37.56 | Robba | any suggestions would be great. |
| 23:38.11 | LiNeTuX|Home | cat " " > /etc/asterisk/extensions.conf |
| 23:38.51 | bkruse | Robba: use more macros |
| 23:38.56 | bkruse | LiNeTuX|Home: be nice... |
| 23:39.41 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 23:40.03 | Maliuta | LiNeTuX: yeah, cat is too much overhead, recommend echo ;) |
| 23:40.18 | Robba | http://rafb.net/p/U3mZtH71.html |
| 23:40.19 | LiNeTuX|Home | oops ;0 |
| 23:40.23 | Maliuta | or rm |
| 23:44.04 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
| 23:44.23 | doolph | Hello I need some help |
| 23:44.57 | doolph | My analog card always get stuck and I have to restart asterisk |
| 23:45.00 | doolph | why? |
| 23:45.13 | doolph | this is my show channels http://pastebin.ca/1060902 |
| 23:46.46 | [netman] | what analog card is? |
| 23:46.48 | JT | doolph: the stuck channels, what are they doing, are they in an IVR, conference, inbound call or what? |
| 23:50.14 | doolph | JT no, just they are transfering calls |
| 23:50.21 | doolph | I think |
| 23:50.37 | doolph | I think its the threewaycalling problem |
| 23:50.41 | doolph | and transfering calls |
| 23:51.05 | doolph | all calls is going to 1 SIP phone that can handle 4 lines, then the person will transfer them to fxs lines |
| 23:52.13 | JT | then what happens? |
| 23:53.10 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 23:53.22 | doolph | then the person with fxs line answer the call, maybe it transfer it again to another fxs |
| 23:53.39 | doolph | I really cannot recreate the problem, just sometimes it get stuck all lines |
| 23:53.51 | doolph | then I have no more channels to call or receive calls |
| 23:53.52 | JT | all lines get stuck at once? |
| 23:53.59 | doolph | no at once |
| 23:54.02 | JT | or they build up? |
| 23:54.04 | doolph | yeah |
| 23:54.07 | doolph | build up |
| 23:54.23 | JT | my normal advice would be to get rid of analogue |
| 23:54.30 | JT | though it is also possible there is a bug |
| 23:55.19 | doolph | yes I know |
| 23:55.25 | doolph | but what can I do now |
| 23:55.35 | doolph | its not mine, and they wont want to invest more money |
| 23:56.18 | unpaidbill | sweet. click to dial firefox extension nearly complete! |
| 23:56.32 | unpaidbill | i must say, the first extension is quite a whorebag to get working correctly |
| 23:58.28 | doolph | :( |
| 23:59.18 | JT | unpaidbill: open source? |
| 23:59.28 | unpaidbill | yeah i'll send it to you |
| 23:59.38 | unpaidbill | i used some other dudes code for my base |
| 23:59.43 | unpaidbill | some cisco dialer |
| 23:59.58 | unpaidbill | it requires a cgi running on the asterisk server though |