| 00:00.50 | aces1up | paulc ok.. i thought fxs is what i needed.. ok so this might be dumb, but our REAL phone lines aren't going to be hooked up to the system for 2 days.. And i wanted to use an ata connected to a voip provider to test this system, so will the system know the difference when using an ata? just curious... also, can anyone suggest some good ata's to use? |
| 00:01.04 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:01.32 | Micc | Why is my polycom ip450 sorting buddies alphabetically instead of by spead dial number like I have seen it do in the past? |
| 00:01.44 | paulc | aces1up: What you want to do will totally work. As for ATAs.. hmm.. Linksys/Sipura are pretty good, I like 'em.. cheap as chips too.. |
| 00:03.05 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.12 | aces1up | paulc have any voip providers that work well with those ata's that i can use for testing? also, will it still be fxs ports on the atas if i connect the ata's to an asterisk box that connects to the voip gateway? sooo asterisk --> ATA/FXS --> Merlin Phone system |
| 00:03.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.173.128.dsl.dyn.forthnet.gr) |
| 00:05.47 | paulc | aces1up: Vitelity have a decent rep. People on here talk about voip.ms a lot too. VoipInfo wiki has a ton of providers. I quite like Link2Voip (use paulc as a referral if you're feeling generous!) |
| 00:06.02 | aces1up | cool |
| 00:06.15 | aces1up | paulc and how bout connecting the ata to asterisk and then to the merlin? |
| 00:06.50 | paulc | aces1up: yup, totally - VoIP Provider <--> Asterisk <--> ATA <--> Merlin |
| 00:06.59 | aces1up | ok nice. |
| 00:07.05 | paulc | some dialplan tweaking on the ATA, depending on how the Merlin's dialing out |
| 00:07.19 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 00:07.20 | paulc | some Asterisk dialplan work to route calls back and forth |
| 00:07.23 | aces1up | sooo it seems its hard to find an unlocked ata around here locally, might use this magic jack :) what you think of that idea? |
| 00:07.36 | paulc | aces1up: Magic Jack != ATA |
| 00:07.42 | aces1up | paulc no? |
| 00:07.44 | paulc | it would work for what you want to do, on a single line |
| 00:07.50 | paulc | but it won't talk to Asterisk |
| 00:07.53 | paulc | it's not really a SIP device |
| 00:08.00 | aces1up | paulc right no, i know it won't talk to asterisk |
| 00:08.09 | paulc | it'd be "good enough" to test your Merlin functionality, using a single trunk line though |
| 00:08.18 | aces1up | nice |
| 00:08.20 | aces1up | thats all i need. |
| 00:08.42 | aces1up | althought it would be great if i had more than one, only have 1 right now, so i can test multiple incoming lines :( |
| 00:09.04 | aces1up | know any places locally that sell unlocked ata's looks like there aren't any from what i'm finding. |
| 00:09.16 | aces1up | best buy, fry's, staples, no luck. |
| 00:09.31 | paulc | aces1up: Where is "locally" ? |
| 00:09.37 | aces1up | las vegas for me. |
| 00:11.02 | paulc | aces1up: http://lasvegas.craigslist.org/ele/1583656629.html might be worth a look |
| 00:11.17 | paulc | aces1up: Also http://lasvegas.craigslist.org/ele/1605047359.html but it's probably locked to Vonage |
| 00:12.56 | aces1up | clear voip is not locked? |
| 00:13.36 | paulc | I'm not sure |
| 00:13.42 | paulc | might be worth a try for $25 |
| 00:14.11 | paulc | the alternative is to buy one online.. and get it early next week.. but then you said your phone lines are coming in a couple of days anyway, no? |
| 00:14.14 | aces1up | hrmm cool, well thanks for your help i gotta run get to testing this system before monday rolls around. |
| 00:14.22 | paulc | good luck! |
| 00:14.26 | aces1up | heh thanks. |
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| 00:17.59 | idespinner | does dialplan cmd ResetCDR() actually remove the CDR entry? |
| 00:19.31 | Walsh86 | Has anyone ever come across an issue where IVR inluding voicemail and phonebook work fine from within the PBX's network but if you call from PSTN through a SIP trunk it doesn't respond to ANY tones. however it can't be the trunk not passing tones properly because if I forward incoming calls to a DISA extension it understands my password and all keys from that point on |
| 00:19.55 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 00:20.20 | idespinner | Walsh86, are you referring to DTMF tones? |
| 00:20.26 | Walsh86 | yes |
| 00:20.27 | Walsh86 | sorry |
| 00:20.34 | *** join/#asterisk V4mpire (~gary@82.118.111.252) |
| 00:20.56 | paulc | Walsh86: Is the call being "answered" in both cases? |
| 00:21.06 | idespinner | I would double check the DTMF signaling method is set the same for both ends. |
| 00:21.08 | Walsh86 | yes 100% sure of that |
| 00:21.27 | idespinner | dtmf=... inline... rfc2866 or whatever.. sip info... |
| 00:24.12 | Walsh86 | I've tried all modes available for dtmfmode= . I'm fairly never to asterisk, is dmtf= and dtmfmode= still used? i thought i read somewhere that dtmf= wasn't used anymore in the config but maybe i read wrong |
| 00:24.28 | Walsh86 | new* not never |
| 00:25.43 | paulc | Generally dtmfmode=rfc2833 works pretty swell |
| 00:25.54 | Walsh86 | thats waht im using right now |
| 00:26.17 | paulc | when you say "transfer to DISA" - is the call coming in to an extension, being answered, then "transferred"? or like forwarded? |
| 00:27.15 | *** join/#asterisk rickross (~rickross@supporter/active/rickross) |
| 00:27.16 | Walsh86 | the incoming route is set to forwarded to DISA, sorry if my terminology is a little off |
| 00:28.01 | idespinner | Walsh86, did you enable DTMF debugging in logger.conf |
| 00:28.24 | paulc | Walsh86: Does it make any difference if you play a prompt before going to DISA? |
| 00:30.19 | Walsh86 | idespinner: No i havn't thanks for the suggestion I'll look into that |
| 00:30.48 | *** join/#asterisk jpvoip (~c88ad599@gateway/web/freenode/x-agfxwjkirplratmu) |
| 00:31.05 | Walsh86 | paulc: does DISA asking for a password count as a prompt? |
| 00:31.58 | jpvoip | hi there |
| 00:32.14 | jpvoip | any suggestion of research resources for implementing Asterisk + XMPP? |
| 00:33.43 | Walsh86 | Anyways I think I i will try and look into this for another few hours and see if I can figure out a little more before I take up any more of your time. If i can't figure it out any further maybe ill have a better idea of what to ask when i come back |
| 00:33.43 | paulc | jpvoip: I read this just earlier today - http://leifmadsen.com/sites/default/files/Unusual_Uses-What_You_Didnt_Know_Asterisk_Could_Do.pdf |
| 00:33.52 | paulc | jpvoip: there's a section in there on XMPP |
| 00:33.59 | idespinner | i thought asterisk had a chan XMPP/jabber... |
| 00:34.09 | paulc | Walsh86: good luck! :) |
| 00:34.14 | Walsh86 | thanks |
| 00:34.29 | jpvoip | paulc: cool! ill read this! |
| 00:34.40 | jpvoip | idespinner: yes... jingle |
| 00:34.54 | jpvoip | idespinner: and gtalk.. to talk to... gtalk! |
| 00:38.51 | ruben23 | hi does Asterisk 1.4.21.2 supports jabber or gtalk integration..? |
| 00:39.24 | idespinner | pretty sure. Dont think they would make channel drivers for either if it didnt.. |
| 00:40.04 | *** join/#asterisk lgbr (~lgbr@laserbunny.net) |
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| 00:42.48 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 00:48.16 | jpvoip | ruben23: what u want to do whith xmpp? |
| 00:49.29 | *** part/#asterisk lgbr (~lgbr@laserbunny.net) |
| 00:53.06 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 00:53.16 | ruben23 | jpvoip: iwant to integrate gtalk to send me message when incoming calls are coming, displaying its number..even its missed call.. |
| 00:53.43 | ruben23 | any idea..? i have installed iksemel already and inital config, but im stock here. |
| 00:57.00 | jpvoip | google it, thereis a lot of tutorials |
| 00:57.12 | jpvoip | its easy |
| 00:59.51 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 01:00.12 | jpvoip | i asked because im searching unified communications topics using asterisk + xmpp to implement in my final course project |
| 01:02.54 | rickross | anyone have any reason to believe Time-Warner cable is using deep packet inspection (or some other QOS-type tool) to limit the max amount of upstream bandwidth used for voip? |
| 01:03.39 | rickross | we seem to max out and go into a huge rut at about 384 kbps of upstream, although for normal ftp and other upstream we can hit 1.6-1.7 mbps |
| 01:07.02 | *** join/#asterisk xpot (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 01:11.28 | jblack | rickross: See if it still happens with iax. I bet not |
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| 01:11.51 | rickross | interesting |
| 01:11.55 | rickross | I will check |
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| 01:34.48 | rickross | jblack, I don't know how to check that - we are seeing the problem manfest even when we use local extensions connected via sip to our remote asterisk - the polycom phones don't have any way to use IAX to speak to the asterisk server , afaik |
| 01:38.53 | jpvoip | try to change the port |
| 01:39.43 | *** join/#asterisk Zettatronic (~nick.croc@mail.maverickforms.biz) |
| 01:44.31 | ruben23 | hi, i i run jabber show connected-and type help for show jabber---command not found, is the asterisk broken or the jabber function is just not supported..? |
| 01:45.02 | jpvoip | whats the answer for jabber show connected? |
| 01:46.45 | ruben23 | No such command 'jabber show connected' (type 'help jabber show' for other possible commands) |
| 01:47.10 | jpvoip | hlep jabber show shows what? |
| 01:47.20 | ruben23 | help jabber show---> No such command 'jabber show'. |
| 01:47.46 | *** join/#asterisk ellisdee (~ellisdee@cosmic.sized.penisinyourface.com) |
| 01:47.50 | dlynes | ruben23, load chan_jabber.so |
| 01:48.22 | dlynes | ruben23, or chan_gtalk.so or whatever it's called |
| 01:48.58 | jpvoip | ruben23: are the libs loaded? |
| 01:49.53 | ruben23 | http://pastebin.com/QLpe8CK4 |
| 01:50.23 | ruben23 | what could be the problem is it with the asterisk compile..? i did not select jabber module.? |
| 01:50.58 | jpvoip | have u look for a tutorial? |
| 01:51.50 | ruben23 | actually i have existing asterisk installed, i re-compile because i want to add gtalk |
| 01:51.59 | *** join/#asterisk aces1up (aces1up3@wsip-24-234-80-107.lv.lv.cox.net) |
| 01:52.12 | ruben23 | so what i did is re compile then follow the step on how to install iksemel.. |
| 01:52.42 | ruben23 | configire jabber.conf and gtalk, but as i see now asterisk si not suing jabber..i think |
| 01:52.49 | jpvoip | but have u look for a step by step tutorial for using asterisk + gtalk |
| 01:53.00 | ruben23 | yes i have.. |
| 01:53.13 | aces1up | so... crazy question, can asterisk work with a modem's jacks to send voice? |
| 01:54.03 | ellisdee | no |
| 01:54.05 | ruben23 | http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 01:54.32 | ellisdee | aces1up need a fxs card |
| 01:54.59 | jpvoip | list yuor modules |
| 01:55.45 | jpvoip | try load res_jabber.so or chan_gtalk.so |
| 01:56.00 | ellisdee | ruben23 i see you are almost there :P |
| 01:56.12 | ruben23 | yeah this is the output--->http://pastebin.com/QLpe8CK4 |
| 01:56.24 | ruben23 | the jabbe module is not there.. |
| 01:56.43 | ruben23 | ellisdee:im having hardtime..but still psuhing hard |
| 01:57.12 | jpvoip | RES_jabber.so |
| 01:58.41 | ellisdee | ruben23 nice |
| 01:59.05 | ellisdee | did you successfully compile the module earlier? |
| 01:59.22 | ellisdee | may need to copy the module to a specific location so that you can load the module within asterisk command line |
| 02:00.17 | ruben23 | yes |
| 02:00.28 | ruben23 | what modules should i copy..? |
| 02:02.37 | hipitihop | I'm still chasing why my asterisk runs @ 100% cpu, some here pointed me at trying to debug/trace the process but so far no luck, so wondering if anyone else is experiencing similar problem with 1.6.2.4 |
| 02:04.25 | jpvoip | ruben23: u need to have res_jabber.so/chan_gtalk.so !!!! |
| 02:04.51 | jpvoip | ruben23: RES_jabber.so and not CHAN_jabber.so |
| 02:04.51 | ruben23 | i cant find it on my asterisk system...both res_jabber.so/chan_gtalk.so |
| 02:05.24 | jpvoip | are using 1.4? |
| 02:05.53 | ruben23 | Asterisk 1.4.21.2 |
| 02:08.27 | *** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net) |
| 02:08.37 | ddickenson | ran into an unforseen problem with an install that runs two different businesses. I have been able to separate everything except music on hold. Anyone know how to work contexts and such into moh? |
| 02:11.25 | ddickenson | anyone? |
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| 02:21.06 | Micc | Do polycom phones have a way to format numbers in the speed dial list? |
| 02:21.24 | Micc | or can I format them in the xml file? |
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| 02:38.15 | rickross | is there any convenient way to add encryption between my Polycom phones and my asterisk server? |
| 02:43.28 | aces1up | i know this isn't an asterisk question, but has anyone here have experience with merlin magix systems? |
| 02:46.49 | carrar | put a VPN between them |
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| 02:57.20 | rickross | carrar - no easy way to do it |
| 02:58.22 | rickross | well, maybe I could force all the phones onto a vlan |
| 02:58.22 | rickross | and somehow get that onto a vpn link |
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| 03:01.09 | adnc | Katty, huhu |
| 03:01.52 | hardwire | nice.. anybody running 1.6.2 have voicemail crash when you tell it to skip instructions? |
| 03:01.56 | hardwire | well.. asterisk crash :) |
| 03:02.37 | carrar | rickross, why isn't a vpn easy? |
| 03:03.08 | rickross | the phones are all on a lan with a ton of other computers |
| 03:03.26 | carrar | well you can do a vpn destination based |
| 03:03.40 | carrar | for packets heading to the asterisk box send to the vpn tunnel |
| 03:04.03 | rickross | hmm, it'd completely hide the voip from the cable-modem ISP, wouldn't it? |
| 03:04.19 | carrar | if it's encrypted it would |
| 03:04.28 | carrar | openvpn is free |
| 03:04.33 | carrar | and very nice |
| 03:04.54 | rickross | we think Time-Warner is somehow limiting our total voip upload bandwidth to a fraction of our real bandwidth |
| 03:05.05 | carrar | and depending on the load of your asterisk box might even be able to run it on the same box |
| 03:05.22 | rickross | so, I would open a vpn tunnel between my lan and the * server, with openvpn |
| 03:05.31 | carrar | yup |
| 03:05.31 | rickross | and direct all the phones to use that? |
| 03:05.49 | carrar | yup, destination based rou ting |
| 03:06.45 | rickross | is there something like TLS support in Asterisk? (or is it not completed?) |
| 03:07.01 | carrar | TLS would not encrypt the RTP |
| 03:07.18 | rickross | I see, and that's the main rub, isn't it? |
| 03:07.24 | carrar | err |
| 03:07.40 | rickross | how is TW detecting that the packets are voip rtp packets? |
| 03:08.03 | carrar | possibly packet timing |
| 03:08.06 | rickross | is there any trivial way to camouflage the packets? |
| 03:08.15 | carrar | You are sending packets out every 20ms |
| 03:08.50 | carrar | or they may be looking deeper |
| 03:08.56 | carrar | who knows |
| 03:09.09 | rickross | well, it is fine for up to 3 outbound calls at 80 kbps per, but the 4th one pushes it over some kind of threshold |
| 03:09.10 | carrar | have you called them? |
| 03:09.23 | rickross | not yet, we just discovered this tonight |
| 03:09.33 | carrar | Whats your circuit type and speed? |
| 03:09.38 | rickross | after their business people go home |
| 03:09.50 | rickross | we have 10 down, 2 up |
| 03:09.57 | rickross | on a docsis 2 cable modem |
| 03:10.17 | rickross | it'll happily push 1.7 mbps up for ftp and other types of packets |
| 03:10.19 | carrar | sure there isn't a issue on the far end? |
| 03:10.26 | rickross | but it craps out on the voip |
| 03:10.54 | rickross | not positive, but we have been going to significant lengths to remove local causes |
| 03:11.07 | rickross | hmm, you mean on the * server? |
| 03:11.26 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 03:11.30 | carrar | yeah |
| 03:11.37 | rickross | we can push at 1.7 mbps to it for file xfers, but not for voice |
| 03:11.42 | carrar | that side anyways |
| 03:11.57 | rickross | we've never seen the full, actual 2 mbps, always a bit less |
| 03:12.14 | carrar | have you had more then 4 calls on your * server? |
| 03:12.18 | carrar | from the outside |
| 03:12.20 | rickross | yes |
| 03:12.42 | rickross | it has a hige pipe and a reasonably hefty processor |
| 03:12.51 | rickross | and is mostly free to service asterisk |
| 03:13.20 | carrar | maybe you are having local network issues |
| 03:13.33 | carrar | with lots of small packets |
| 03:13.54 | rickross | we thought so |
| 03:13.55 | carrar | but 4 calls is not a lot |
| 03:14.14 | rickross | but there's no sign of significant load on the local router |
| 03:14.29 | rickross | and we completely removed that one and put in a different type of device to test |
| 03:14.38 | rickross | it crapped out at exactly the same point |
| 03:15.09 | rickross | these are approx. 80 kbps calls using Polycom HDVoice |
| 03:15.48 | carrar | g711 is 88k per call |
| 03:15.50 | rickross | in fact, if there are any cals going at all, our upstream bandwidth appears to get limited |
| 03:16.34 | rickross | yeah, I don't know the exact amount Polycom uses, but it is right around 80-per, a little higher |
| 03:16.56 | rickross | I don't even know if it is actually G.722 or some variant they are pushing |
| 03:17.05 | rickross | but it sounds pretty good |
| 03:17.21 | carrar | first try testing something |
| 03:17.30 | carrar | use a different port other then 5060 |
| 03:17.34 | carrar | for sip |
| 03:17.39 | rickross | for the sip signalling? |
| 03:17.51 | *** join/#asterisk sacitec (~anre@189.190.92.32) |
| 03:18.06 | carrar | or change your rtp range maybe |
| 03:18.21 | carrar | those are much simpler changes I think then a vpn |
| 03:18.36 | *** part/#asterisk sacitec (~anre@189.190.92.32) |
| 03:18.47 | carrar | might just call them |
| 03:18.49 | carrar | and ask |
| 03:19.02 | rickross | good ideas, thx |
| 03:19.38 | rickross | would srtp obscure this traffic? |
| 03:20.04 | carrar | hard to say |
| 03:24.51 | carrar | TLS should encrypt the audio |
| 03:25.10 | carrar | don't know asterish status as to tls though |
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| 05:10.21 | titter | Is there a general location to place requirecalltoken=no ... obviously tried it in the general section of iax.conf |
| 05:10.47 | hipitihop | after having proplems with 1.6.2.4 I have installed a ubuntu testing repo of 1.6.2.0 however it seems it has not installed any init.d ot upstart .. what is the correct way to manually run the daemon from the command line ? |
| 05:12.08 | idespinner | hipitihop, Running Asterisk is actually rather straight forward. Asterisk, if run with no arguments, is launched as a daemon process: |
| 05:12.21 | idespinner | from: http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping |
| 05:13.14 | hipitihop | idespinner, reading..thanks for link |
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| 05:38.21 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 05:38.33 | ManxPower | ~answers |
| 05:38.34 | infobot | extra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 05:44.06 | hipitihop | idespinner, running successfully thanks... now I have something to report regarding 1.6.2.4 compiled form source using 100% cpu whereas now installed 1.6.2.0~rc2-0ubuntu1.2 is fine, where is best to report such stuff |
| 05:44.12 | skor | what's the rationale behind providers doing free termination to toll free ? |
| 05:45.07 | idespinner | hipitihop, i suppose https://issues.asterisk.org/main_page.php |
| 05:45.16 | idespinner | but no direct experience with it here... |
| 05:46.16 | idespinner | hipitihop, i think i would suggest unloading modules 1 by 1 first though, find out which is causing the cpu usage |
| 05:46.27 | idespinner | or it could be asterisk itself that does it |
| 05:47.30 | idespinner | hipitihop, unload them in /etc/asterisk/modules.conf |
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| 05:51.47 | hipitihop | idespinner I see, but I have now installed previous |
| 05:52.08 | hipitihop | idespinner, is there a command to list loaded modules ? |
| 05:52.15 | idespinner | yea |
| 05:52.22 | idespinner | i think its like core module show |
| 05:52.26 | idespinner | or core show module |
| 05:52.32 | idespinner | maybe its module show |
| 05:52.54 | ManxPower | hipitihop, you can't report a bug like that. |
| 05:53.06 | ManxPower | you have to test between two compiled from source versions |
| 05:54.42 | hipitihop | manxPower, sounds reasonable, is it posisble to have two versions installed from source ? |
| 05:55.04 | hipitihop | soerry if these are newbie questions, just a little new territory for me |
| 05:56.35 | ManxPower | just delete everything in /usr/lib/asterisk/modules You can't easily have two versions installed at the same time |
| 05:57.57 | hipitihop | ManxPower, hmm, sound slike I will need to set asside some time to do that.... well at least now I have proved I can get it to work and so my sip.conf and extensions.conf are transportable between versions |
| 05:58.53 | ManxPower | hipitihop, if you switch between major versions like 1.4, 1.6.0, 1.6.1, 1.6.2, etc then you should read the UPGRADE*.txt. Upgrading between minor versions should not require config file changes |
| 06:00.51 | hipitihop | ManxPower, indeed, although I doubt I will drop bellow 1.6 series and unlikely bellow 1.6.2 anyway |
| 06:06.20 | Micc | hipitihop, what features in 1.6.2 do you require? |
| 06:06.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:07.47 | Micc | skor, if you find out why some providers provide free toll free termination, let me know. I'm guessing if they have a high priced per minute rate they fiture they'll make up for it in usage. |
| 06:08.26 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:09.03 | Micc | fiture/figure |
| 06:09.42 | Micc | i'm practically blind right now. I lost my glasses in a snowboarding accident last weekend. |
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| 06:39.20 | hipitihop | Micc, I'm a first time * user so I'm not really aware of what the baseline capabilities are never mind diff between versions at this point |
| 06:40.50 | Micc | hipitihop, I would suggest staying with 1.6.1 tree for now until 1.6.2 becomes more stable. |
| 06:40.58 | hipitihop | Micc, also I'm only setting it up at home with only a voip provider and my existing ATA with handset .. so I doub't I'm going to be stretching it's abilities :-) |
| 06:42.29 | hipitihop | Micc, so realy only an experiment although I do want it to be reasonably reliable considering the voip line is my only thing available at home... any specific area of instability or just general |
| 06:43.11 | Micc | hipitihop, just in general. 1.6.2 is the newest branch. |
| 06:43.44 | Micc | and I doubt you will need any of the new features in 1.6.2 |
| 06:47.05 | hipitihop | Micc, I still have a very simple setup afaik only the insecure=port,invite is relatively new, so I guess it would be pretty straight forward to drop back |
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| 06:48.18 | hipitihop | one thing that confuses me, I'm running on a ubuntu, and here someone suggested to install from source, which I did but then had issues with 100% cpu...in ubuntu channel they suggested to stick to ubuntu packages |
| 06:48.45 | ManxPower-work | hipitihop, packages are not supported here so you would be on your own if you used a package |
| 06:50.09 | ManxPower-work | Personally I'd stick to 1.6.1.x |
| 06:51.28 | hipitihop | ManxPower-work, I think I remember you suggesting that a few days back ;-) ... go on, you know you want to say it |
| 06:52.30 | hipitihop | and I sure do appreciate the help from this channel, so would much rather stay on from source .. especially since I'm not really committed to either way at the moment. |
| 06:52.53 | ManxPower-work | hipitihop, Asterisk is almost the only thing I ever install from source |
| 06:53.57 | hipitihop | ManxPower-work, it certainyl is the only thing I remember having to go to source...I lie, one other Simon listens too. |
| 06:54.43 | hipitihop | ManxPower-work, so I should remove the PPAand package. ... anything I need to do to remove the previous 1.6.2.4 build from source ? |
| 06:54.58 | ManxPower-work | Asterisk is a repidly evolving and updated project, packages are ALWAYS out of date. |
| 06:55.17 | hipitihop | makes perfect sense. |
| 06:55.27 | hipitihop | and I love the bleeding edge |
| 06:55.40 | ManxPower-work | ClamAV is the only other package I can think of that I'd install from source. |
| 06:56.01 | hipitihop | that's virus stuff isn't it |
| 06:56.52 | ManxPower-work | yes -- also a rapidly evolving and updated project. |
| 07:02.46 | benngard | hipitihop: if u love "bleeding edge" try "trunk version" ;) |
| 07:05.50 | hipitihop | benngard, heh .. I'm going with ManxPower-work recommendation this time, I doubt I need the new features and stability for my home voip will be more important then something to play with ... the mythtv running along side is bleeding edge enough for now |
| 07:06.55 | benngard | hipitihop: for a home solution, any * version will do the trick for u |
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| 07:08.00 | benngard | btw, we have like 0.5 m of snow in gothenburg/sweden and it's raining like hell |
| 07:08.20 | benngard | gonna be fun today, NOT :( |
| 07:09.11 | hipitihop | benngard, well that's what I thought, but my intial attempt was 1.6.2.0 rc2 package from standard ubuntu repos, that had some strange bug which treated my incoming as an outgoing...dropped that, went to 1.6.2.4 from source, and that has a 100% cpu usage, now I'm on 1.6.2.0 testing which works but I'm being recommended to go back to source |
| 07:09.58 | hipitihop | so theory doesn't always always work out as easy as practice :-) |
| 07:10.11 | drmessano | 1.6.1.x |
| 07:10.30 | benngard | i had about the same experience with * and debian so i started to use source |
| 07:10.52 | drmessano | Source is the ONLY way to go with Asterisk |
| 07:11.18 | hipitihop | yup just removed the package .. about to remove the ppa, and then will insall 1.6.1.16 from source |
| 07:11.21 | drmessano | I can handle someone elses semi-optimized binaries for non-realtime applications, but for Asterisk, no way |
| 07:13.23 | benngard | btw, talking about source, i have a heavy patched ooh323 channel on my *, alexandr nees to commit those pathes, must remind him to do that |
| 07:13.48 | hipitihop | drmessano, sinse I already went from source for 1.6.2.4, do I need to do anything to get rid of that before I try the 1.6.1.x ? |
| 07:15.04 | *** join/#asterisk Rajmohan (~raj@122.164.69.15) |
| 07:15.17 | drmessano | As was explained earlier, clear out your modules directory at the very least |
| 07:15.44 | drmessano | [00:56] <ManxPower> just delete everything in /usr/lib/asterisk/modules You can't easily have two versions installed at the same time |
| 07:16.04 | Rajmohan | hi, i want to have a voip usa phone number configured to a analog phone, should i go for a linksys pap2 or linksys spa |
| 07:17.19 | benngard | remove /var/lib/asterisk/ to, just to be safe |
| 07:17.24 | hipitihop | drmessano, indeed my bad.. sometimes forgot what I have already asked |
| 07:17.48 | Rajmohan | i have asterisk setup done... how do i get a voip number configured to it... can any one guide me and also should i go for linksys pap2 or linksys spa to connect to analog phones |
| 07:18.24 | drmessano | There's a couple dozen "Linksys spa" devices.. Which one?? |
| 07:18.25 | benngard | spa2102 works fine for me, but i am in sweden, guess it will work in usa to |
| 07:18.44 | Rajmohan | Linksys SPA3102 |
| 07:18.57 | drmessano | You dont need the 3102 |
| 07:19.04 | Rajmohan | which is phone adapter + pstn gateway |
| 07:19.05 | drmessano | PAP2 is fine |
| 07:19.14 | benngard | thats the on with both fxs and fxo, correct? |
| 07:19.16 | drmessano | I know what it is, kthx |
| 07:19.21 | drmessano | yeah |
| 07:20.05 | Rajmohan | ok fine, when i have pap2, and voip number configured to it, i can make outgoing call as well from analog phones with caller id? |
| 07:20.19 | drmessano | yes |
| 07:20.45 | benngard | ofc, why shouldnt u not be able to do that? |
| 07:20.46 | Rajmohan | go great, thanks for guiding, iam going with pap2 |
| 07:22.16 | Rajmohan | one more question, i have asterisk setup done.. since iam newbie, my question might make nosense. can i get a voip number configure in asterisk or directly configure it to linksys pap2 |
| 07:22.19 | *** join/#asterisk neurosys (~neurosys@166.192.64.227) |
| 07:22.32 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 07:22.37 | drmessano | Yes |
| 07:22.59 | drmessano | ~itsp |
| 07:23.00 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 07:23.25 | drmessano | ~answers |
| 07:23.26 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 07:23.46 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 07:23.54 | drmessano | ~happyclownpbx |
| 07:23.54 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 07:24.14 | drmessano | Hmm |
| 07:24.34 | drmessano | The 200LP-A12 is out now |
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| 07:42.30 | hipitihop | drmessano, no /usr/lib/asterisk at all never mind /usr/lib/asterisk/modules ... is it likely elsewhere depending on repo ? I'm on ubuntu |
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| 07:43.42 | drmessano | Well, maybe its not |
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| 07:45.50 | p3nguin_ | infobot: forget happyclownpbx |
| 07:45.50 | infobot | p3nguin_: i forgot happyclownpbx |
| 07:46.25 | p3nguin_ | infobot: happyclownpbx is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 07:46.26 | infobot | okay, p3nguin_ |
| 07:46.47 | p3nguin_ | infobot: happyclownpbx |
| 07:46.48 | infobot | extra, extra, read all about it, happyclownpbx is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 07:47.13 | p3nguin_ | manxpower must have added that originally. |
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| 07:48.35 | ManxPower-work | hipitihop, now you start to see why we hate packages -- stuff isn't always where you expect it to be. |
| 07:49.31 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 07:50.23 | hipitihop | ManxPower-work, I removed the package that's fine.. I'm trying to cleanup from the previous install from source of 1.6.2.4 before I build 1.6.1.16 |
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| 07:51.17 | hipitihop | ManxPower-work, and I certainly see lots in /var/lib/asterisk/* |
| 07:51.37 | hipitihop | but can't see the modules that was recommended to clean |
| 07:51.56 | ManxPower-work | /var/lib/asterisk != /usr/lib/asterisk |
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| 07:55.12 | drmessano | p3nguin_, why are you messing with my happyclownpbx? |
| 07:55.33 | p3nguin_ | Had to correct a word that didn't make sense. |
| 07:55.38 | p3nguin_ | ~happyclownpbx |
| 07:55.39 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 07:55.47 | p3nguin_ | It's fixed up, now. |
| 07:55.56 | drmessano | I see |
| 07:56.34 | drmessano | Fine, then..it's looks better now |
| 07:56.50 | p3nguin_ | Say what? |
| 07:58.01 | drmessano | I said, its much better now thanks to you're corrections, which we're accurate |
| 07:59.13 | *** join/#asterisk RichardLynch (~RichardLy@c-98-193-36-91.hsd1.il.comcast.net) |
| 07:59.13 | p3nguin_ | Ah, your write. Thank's for clearing that up. |
| 07:59.23 | drmessano | OMFG |
| 07:59.30 | p3nguin_ | That hurt to type. |
| 07:59.36 | drmessano | Thank's <-- One of my favs |
| 08:00.32 | p3nguin_ | Yes, the apostrophe means "here comes an S." |
| 08:00.46 | RichardLynch | I just compiled and installed asterisk 1.6.2.2 on top of an old 1.4 box. format_mp3.so is causing segfaults, so I've left it out. What am I missing with that, and how do I re-compile just that?... My boss is the asterisk geek, but I don't want to call him at this hour... |
| 08:01.25 | *** join/#asterisk SAngeli (~SAngeli@host242-77-static.82-94-b.business.telecomitalia.it) |
| 08:01.25 | RichardLynch | (It's 2am here) |
| 08:01.33 | p3nguin_ | For you, too? |
| 08:01.47 | p3nguin_ | I guess it's 2 everywhere in Illinois. |
| 08:02.19 | RichardLynch | I think there is some podunk town in eastern IL that sticks with Indiana time, actually... |
| 08:02.40 | p3nguin_ | I think most people rename/move/delete the modules directory prior to compiling and installing any newer version of Asterisk. |
| 08:03.01 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:03.34 | RichardLynch | Yes, well, it's a DEV box, and it was easier to blow it up and then move out all the suspect ones from make install listing and then put them back one by one until the culprit was found... At least for this naive user. |
| 08:03.41 | p3nguin_ | I don't know if that has any relation to the problem you've described, though. |
| 08:04.18 | RichardLynch | format_mp3.so was in the modules dir, and now it isn't. If its' there, it segfault. If it's not it doesn't. At least not immediately. :-) |
| 08:04.53 | nix8n82 | did you run the make menuconfig and make sure it was set to compile as a module? |
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| 08:07.42 | RichardLynch | I think I need to go find the 1.6.2.2 version of asterisk-addons and compile that, since that's where I'm locate-ing format_mp3.* files, and I didn't, and it does not compile from the source I have, from 1.4 days... |
| 08:07.49 | kaldemar | RichardLynch: format_mp3 is from asterisk-addons. if you didn't compile the addons package, just delete it. |
| 08:08.29 | RichardLynch | We probably need the mp3 format though. I'll leave it out for now. |
| 08:08.45 | kaldemar | RichardLynch: it's a good practise to delete all modules before installing a new version. makefiles can do that for you with the uninstall target. |
| 08:10.42 | RichardLynch | I'm compiling the 1.6 version of asterisk-addons to replace all those. Will have to ask the boss where all the other modules came from tomorrow... |
| 08:12.25 | RichardLynch | Well, it doesn't segfault at launch anymore, so that's progress. :-) |
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| 08:21.09 | hipitihop | is there no reverse steps for make, make install ? |
| 08:21.47 | p3nguin_ | If you took my advice of using checkinstall, it wouldn't matter. |
| 08:22.00 | drmessano | make uninstall |
| 08:22.34 | skor | ~itsplist-us |
| 08:22.35 | infobot | itsplist-us is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 08:22.59 | p3nguin_ | drmessano: Your sure their's an uninstall target? |
| 08:23.16 | p3nguin_ | I feel so dirty. |
| 08:23.21 | hipitihop | p3nguin_, is that aimed at me ? did I miss something you said ? |
| 08:23.59 | p3nguin_ | hipitihop: Yes, if you use checkinstall in lieu of make install, you wouldn't have to worry if there is an uninstall target. |
| 08:24.46 | hipitihop | p3nguin_, I'm sorry I did not see that advice, first time I heardof it ... Im still in the middle of make .. is it too late now ? |
| 08:24.48 | p3nguin_ | hipitihop: Since you should ALWAYS use your package manager, and most people TYPICALLY want to install Asterisk from source, checkinstall is the way to do both. |
| 08:24.55 | p3nguin_ | It is not too late. |
| 08:25.09 | p3nguin_ | Which distro are you using? |
| 08:25.18 | hipitihop | ubuntu karmic |
| 08:25.59 | p3nguin_ | See if checkinstall is already installed. If not, "sudo aptitude install checkinstall" to add it. |
| 08:26.31 | p3nguin_ | Instead of running sudo make install, you run sudo checkinstall and answer any questions it asks you. |
| 08:27.11 | p3nguin_ | Ideally, you would have changed the --prefix= setting when you ran configure, but it's not a super-huge deal that you didn't. |
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| 08:29.25 | p3nguin_ | I always set --prefix=/usr --sysconfdir=/etc when I run configure, that way the paths are proper for a package. |
| 08:30.13 | hipitihop | p3nguin_, I see, I'll make note of that for next time |
| 08:30.37 | hipitihop | Ia ssume those are arguments to configure |
| 08:31.36 | p3nguin_ | Yes, ./configure --prefix=/usr --sysconfdir=/etc |
| 08:31.50 | hipitihop | p3nguin_, and once the make completes, can I just go back to the ./configure step and add those parms and then runmake again ? |
| 08:32.08 | p3nguin_ | Yes. |
| 08:32.29 | hipitihop | infact I can probably just interupt the make |
| 08:32.31 | p3nguin_ | You could even interrupt it now if you plan to change it and run it again. |
| 08:32.36 | p3nguin_ | SIGINT |
| 08:32.45 | hipitihop | <ctrl-c> |
| 08:32.49 | p3nguin_ | exactly |
| 08:34.25 | hipitihop | way too many moons ago I used to develop using C although not on linux, but a lot of this build stuff should be familiar you;d have thought :-) |
| 08:35.06 | hipitihop | ok reran config with those args, assuming make stays the same |
| 08:36.10 | hipitihop | no parameters on checkinstall ? |
| 08:36.29 | hipitihop | and does it need sudo rights ? |
| 08:37.21 | p3nguin_ | You can pass some options to it if you want to avoid answering the questions. |
| 08:38.20 | hipitihop | ahh does need sudo. |
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| 08:39.25 | p3nguin_ | I thought I mentioned that earlier. |
| 08:40.48 | p3nguin_ | I'm obsessive about software being installed on my systems. I no longer install anything into global locations unless from a package -- even if I make the package myself using tools like checkinstall. |
| 08:43.02 | p3nguin_ | Of course I run ArchLinux, so I use yaourt or just write a pkgbuild if one doesn't exist and then run makepkg... but it's basically the same. |
| 08:44.39 | hipitihop | p3nguin_, well you're a jedi master compared to my skills at the moment.. it's all baby steps for me |
| 08:45.47 | hipitihop | and so, once checkinstall finishes, to reverse a given build hat does one do |
| 08:45.58 | hipitihop | s/hat/what |
| 08:46.21 | hipitihop | eek, failed to isntall Debian paclage |
| 08:47.40 | hipitihop | hmm conflict with libspeex1 , I guess I'll just remove it, can't think of anything that needs that |
| 08:49.45 | hipitihop | p3nguin_, ok seems there are a few things that depend on it, including some mythtv elements, bugger. |
| 08:51.06 | p3nguin_ | On RH-like systems, checkinstall doesn't actually install... you still have to use rpm or yum to install the package. I assume you will still have to sudo dpkg -i whateverthepackagenameis to install it. |
| 08:51.39 | hipitihop | oh I see.. so ignore the error ? |
| 08:51.48 | p3nguin_ | uh no |
| 08:52.06 | hipitihop | I am on ubuntu, so where does it create the package ? |
| 08:52.19 | p3nguin_ | I mean, if you don't want it to work, ignore all errors. |
| 08:52.55 | hipitihop | where does checkinstall create the log ? |
| 08:52.58 | p3nguin_ | No clue, I won't touch Debian or anything that looks like Debian. I can only assume checkinstall will tell you, since it does on RH-like systems. |
| 08:54.14 | hipitihop | it only asked me to if I wanted to see the log but no idea where it was |
| 08:55.51 | tzafrir_laptop | hipitihop, what version of ubuntu? |
| 08:56.02 | tzafrir_laptop | what exactly do you miss? |
| 08:56.53 | tzafrir_laptop | speex is already packaged. No need to bother with it when you can use the package manager |
| 08:57.04 | hipitihop | tzafrir_laptop, karmic, it complained about libspeex1 which is already there but I'm guessing I have a later version and it seems my mythtv stuff wants it too |
| 08:57.21 | tzafrir_laptop | you need the -dev package to build with it |
| 08:57.38 | tzafrir_laptop | install libspeex-dev . Probably also libspeexdsp-dev as well |
| 08:57.49 | hipitihop | any idea where checkinstall would have put the log ? |
| 08:58.11 | tzafrir_laptop | Don't use checkinstall for things you already have proper packages for. |
| 08:58.20 | tzafrir_laptop | checkinstall generates rather broken packages |
| 08:58.57 | p3nguin_ | How so? It copies the output of make install. |
| 08:59.06 | hipitihop | tzafrir_laptop, I'm using checkinstall to build asterisk |
| 08:59.31 | p3nguin_ | You're using checkinstall to make a package out of an asterisk build. |
| 08:59.40 | tzafrir_laptop | This means you'll probably miss the obvious dependencies for libraries |
| 08:59.40 | p3nguin_ | make is what builds it. |
| 09:00.07 | tzafrir_laptop | hipitihop, why not modify the existing deb packaging of asterisk? |
| 09:00.24 | tzafrir_laptop | apt-get source asterisk |
| 09:00.57 | tzafrir_laptop | Or even better: if you have devscripts: debcheckout asterisk # will probably work as well |
| 09:01.06 | tzafrir_laptop | Never tried it on Ubuntu |
| 09:01.42 | hipitihop | tzafrir_laptop, because I don't know what I'm doing ... I installed the PPA testing of asterisk, then was told to go to source, now I'm trying that.. it is difficult for me to guage what is correct advice |
| 09:02.18 | tzafrir_laptop | Why would you need to install from source (specifically) ? |
| 09:02.28 | hipitihop | I then followed normal make, make install etc and was recommended to use checkinstall |
| 09:03.18 | hipitihop | tzafrir_laptop, because I was told that support in here is best for those that build from source |
| 09:03.28 | p3nguin_ | Many people frown on using the prepackaged asterisk from the repos. |
| 09:03.48 | p3nguin_ | Personally, if it has the features you want, I don't see why it matters. |
| 09:03.48 | tzafrir_laptop | p3nguin_, and that is because? |
| 09:03.53 | p3nguin_ | no idea, dude. |
| 09:04.04 | p3nguin_ | If it has what you need, I say use the package. |
| 09:04.19 | p3nguin_ | Others get bitchy over it. |
| 09:04.33 | hipitihop | so, not meaning to ppitch you guys against each other in opinion, but how does someone unqualified judge what is best advice |
| 09:04.36 | p3nguin_ | So the solution is to make your own package. |
| 09:06.07 | hipitihop | so since, checkinstall is complaining about things, then how do I install a specific version of a package e.g. I want asterisk 1.6.1.16 |
| 09:07.02 | hipitihop | tzafrir_laptop, because 1.6.2.x is not considered as stable and that's the normal repo version for karmic or the testing PPA |
| 09:07.30 | p3nguin_ | Why can't you solve the dependencies on your own? |
| 09:07.56 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 09:07.59 | p3nguin_ | I guess I kinda made the assumption that if you could compile asterisk you would also know how to solve deps. |
| 09:08.05 | tzafrir_laptop | rather: 1.6.2 is rather stable. However the version packaged with Karmic is a bit dated and not as stable |
| 09:08.26 | tzafrir_laptop | what I meant is that you use the existing packaging and package a newer version |
| 09:08.39 | hipitihop | happy to, but I don't even know where checkinstall puts is logs to see exactly what it was whining about and then if it is agenuine conflict then I have little experience... I am very used to the package system just doing the right thing |
| 09:08.51 | tzafrir_laptop | which is basically, once you're in the packaging directory: |
| 09:09.15 | tzafrir_laptop | dch -i # to add a new changelog entry - set the version you want there |
| 09:09.23 | tzafrir_laptop | debian/rules get-orig-source |
| 09:09.43 | tzafrir_laptop | and rebuild it (with e.g. debuild) |
| 09:11.33 | hipitihop | tzafrir_laptop, sorry you'll have to point me more specifically all of that is more new stuff |
| 09:12.21 | hipitihop | so I have to install devscripts to even get dch |
| 09:12.23 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 09:12.44 | tzafrir_laptop | hipitihop, (or just edit debian/changelog yourself) |
| 09:12.48 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 09:14.07 | hipitihop | ok devscripts installed .. now 'dch -i #' I assume # is version number ? is that refering to the version I want ? |
| 09:15.22 | hipitihop | sorry this is obviously second nature to some people, so I hate to be wasting people's time |
| 09:15.37 | *** join/#asterisk DarKnesS_WolF (~wolf@unaffiliated/sherif) |
| 09:15.53 | tzafrir_laptop | http://pkg-voip.alioth.debian.org/ ; http://www.debian.org/doc/manuals/maint-guide/ch-build.en.html |
| 09:16.15 | tzafrir_laptop | hipitihop, no. '#' is a comment in the shell |
| 09:16.31 | tzafrir_laptop | It was intended to make the rest of the line harmless if you paste it as-is |
| 09:16.40 | DarKnesS_WolF | tzafrir_laptop: ! hey mate been a long time :) |
| 09:16.54 | tzafrir_laptop | DarKnesS_WolF, hi, what's up? |
| 09:17.16 | DarKnesS_WolF | everything is okay , |
| 09:17.17 | DarKnesS_WolF | and there? |
| 09:17.24 | tzafrir_laptop | all's well generally |
| 09:18.13 | *** join/#asterisk alex702 (aleeeeex72@ip98-160-155-80.lv.lv.cox.net) |
| 09:18.19 | OrNix | help me please with AEL. I do: goto ${DIALSTATUS}. then NOANSWER:... CONGESTION:... e.t.c. What pattern should i use to catch all else values of ${DIALSTATUS}? '*', '_*','_.' don't work. |
| 09:18.27 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 09:19.18 | *** part/#asterisk alex702 (aleeeeex72@ip98-160-155-80.lv.lv.cox.net) |
| 09:19.49 | hipitihop | tzafrir_laptop, if you have time can you step me through ... this is taking me hours, just to get basic asterisk install only because it is suggested to go from source ... I'd rather be spending these hours learning actual asterisk functionality |
| 09:20.47 | hipitihop | I might just go back to the PPA testing version of 1.6.2.0 which actually worked and set aside to read all these links a learn how to build it myself another time |
| 09:21.04 | tzafrir_laptop | (it's not a PPA) |
| 09:21.41 | tzafrir_laptop | hipitihop, can you try, as a user: apt-get source asterisk |
| 09:21.56 | tzafrir_laptop | this should download asterisk and extract it in a subdirectory |
| 09:22.15 | hipitihop | as a user meaning not sudo ? |
| 09:22.56 | hipitihop | unable to find sourc epackage |
| 09:23.22 | tzafrir_laptop | right, not requiring sudo |
| 09:23.35 | tzafrir_laptop | (package building does not need to run as root) |
| 09:24.00 | hipitihop | ok, "E: Unable to find a source package for asterisk" |
| 09:26.15 | tzafrir_laptop | in /etc/apt/sources.list , make sure you also have deb-src lines for universe |
| 09:26.24 | tzafrir_laptop | not just 'deb' lines |
| 09:26.37 | tzafrir_laptop | I suppose those are commented out or something |
| 09:26.56 | hipitihop | checking |
| 09:28.48 | p3nguin_ | ornix: If it isn't one of the known statuses, then it must be unknown... match all known, then all unknowns can do something else. if, else, fi |
| 09:29.36 | Rajmohan | hi is there any tutorials for beginners in asterisk |
| 09:29.36 | OrNix | p3nguin_, there are few ${DIALSTATUS}'es, but a lot of ${HANGUPCAUSE} |
| 09:30.15 | OrNix | maybe digital ${HANGUPCAUSE} will match '_.' |
| 09:30.40 | hipitihop | Rajmohan, there is an online book, see asterisk.org |
| 09:30.56 | hipitihop | Rajmohan, "the future of ..." |
| 09:31.42 | Rajmohan | ok thanks, does it cover the latest version, or it is fine to refer older version as asterisk as well |
| 09:31.43 | Rajmohan | ? |
| 09:32.29 | *** join/#asterisk neurosys (~neurosys@32.160.97.201) |
| 09:33.35 | hipitihop | Rajmohan, probably only up to 1.6.1.x afaik |
| 09:34.00 | hipitihop | tzafrir_laptop, ok fetching source now |
| 09:34.18 | Rajmohan | ok |
| 09:35.24 | hipitihop | tzafrir_laptop, btw got a notice "'asterisk' packaging is maintained in the 'Bzr' version control system at:...." and it suggests to use bzr get.... |
| 09:36.56 | hipitihop | tzafrir_laptop, anyway it is in the process of getting 1:1.6.2.0~rc2-0ubuntu1 |
| 09:37.47 | hipitihop | which looks suspiciously like the packaged version in the repos which didn't work for me and had some bug with inbound calls .but not that I know anything |
| 09:38.36 | hipitihop | tzafrir_laptop, ok, got the source, what now ? |
| 09:38.50 | tzafrir_laptop | do you have devscripts installed? |
| 09:38.56 | hipitihop | yes |
| 09:39.14 | p3nguin_ | ~book |
| 09:39.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 09:39.17 | tzafrir_laptop | start with the obvious: find missing build dependencies |
| 09:39.26 | tzafrir_laptop | just run: debuild |
| 09:39.30 | p3nguin_ | rajmohan: book ^^^ |
| 09:39.45 | tzafrir_laptop | it will probably fail quite fast and tell you what packages you're missing |
| 09:39.52 | Rajmohan | yes pl |
| 09:40.32 | hipitihop | tzafrir_laptop, I need to change into the new subdir first I assume |
| 09:40.52 | tzafrir_laptop | right. sorry, I forgot that |
| 09:41.58 | hipitihop | tzafrir_laptop, ok a fair list of unmet dependencies and aborted |
| 09:42.36 | hipitihop | tzafrir_laptop, so does that mean I just install those packages using apt-get install ? |
| 09:42.46 | tzafrir_laptop | basically: yes |
| 09:43.01 | tzafrir_laptop | (remove versions and such) |
| 09:43.08 | hipitihop | things like quilt, debhelper etc |
| 09:43.32 | hipitihop | and I assume packages not source |
| 09:44.47 | hipitihop | yep, tried first one and debuild again and now not listed in missing.. standby will do the rest |
| 09:50.28 | *** join/#asterisk Diogo (~info@a89-152-10-8.cpe.netcabo.pt) |
| 10:03.00 | skor | reading the support forums of some of these providers is entertaining |
| 10:03.21 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 10:04.40 | skor | ~itsplist-us |
| 10:04.41 | infobot | methinks itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 10:04.45 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 10:04.48 | cusco | hi folks |
| 10:05.01 | cusco | We just installed some new digium cards |
| 10:05.10 | cusco | recompiled dahdi etc |
| 10:05.32 | cusco | pri show spans - shows them all Provisioned, In Alarm, Down Active |
| 10:05.40 | cusco | any ideas? |
| 10:06.12 | tzafrir_laptop | cusco, what alarm? |
| 10:07.30 | cusco | yellow |
| 10:08.03 | cusco | I think it was.. hold.. I just restarted it for something else... |
| 10:09.30 | cusco | actually |
| 10:09.31 | cusco | red |
| 10:13.04 | cusco | asterisk full log states: Detected alarm on channel 1: Red Alarm // Registered channel 1: ISDN PRI Signalling |
| 10:13.17 | cusco | and doe that until channel 158 |
| 10:16.10 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:16.54 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:33.28 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 10:35.42 | hipitihop | tzafrir_laptop, nearly there with installing all dependencies |
| 10:36.51 | hipitihop | tzafrir_laptop, ok, all satisifed debuild is away |
| 10:37.38 | hipitihop | tzafrir_laptop, so when it's done, does that procude a deb file which I then have to install or does it do more |
| 10:38.17 | tzafrir_laptop | that shows you can build asterisk. next step, is get a more up-to-date one |
| 10:39.28 | hipitihop | tzafrir_laptop, oh, so when this is done what's next |
| 10:40.06 | hipitihop | tzafrir_laptop, so this is simply building what is already in the repo ? |
| 10:41.23 | tzafrir_laptop | yes |
| 10:41.31 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 10:41.35 | tzafrir_laptop | now we get the newer version |
| 10:41.47 | hipitihop | well taht will explain why the version looks familiar, so how do I get latest now ? |
| 10:41.53 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 10:41.58 | cusco | hi |
| 10:42.05 | cusco | Im havving all spans as red alarm |
| 10:42.06 | cusco | :( |
| 10:42.10 | cusco | can't figure out why |
| 10:58.19 | hipitihop | tzafrir_laptop, ok seems to have failed on last bit, "could not find signing program pgp or gpg |
| 11:04.41 | hipitihop | tzafrir_laptop, you have gone ? |
| 11:05.26 | cusco | please help!! |
| 11:05.47 | cusco | just installed new digium cards, recompiled dahdi and all channels are on RED ALARM!! |
| 11:05.50 | cusco | any ideias?? |
| 11:06.11 | tzafrir_laptop | cusco, you're not getting through to the other side. Is the cable actually connected? |
| 11:07.00 | tzafrir_laptop | hipitihop, basically: add to the debuild command: -uc -us |
| 11:07.19 | tzafrir_laptop | but we can now tell you have the build deps and such |
| 11:07.29 | tzafrir_laptop | next: dch -i |
| 11:08.05 | tzafrir_laptop | and in the new entry, change the version number from 1.6.2.0~rc2 to 1.6.2.5 |
| 11:08.15 | tzafrir_laptop | (in the first line only) |
| 11:09.02 | cusco | tzafrir_laptop: yes... could there be another reason non-hardware? |
| 11:09.07 | cusco | like siugnaling, perhaps? |
| 11:09.21 | cusco | what config could result in red alarm? |
| 11:09.36 | cusco | dahdi_scan states channel is red |
| 11:09.40 | drmessano | Nothing else changed other than the cards and the recompile? |
| 11:10.00 | cusco | correct, only ran dahdi_genconf |
| 11:10.07 | drmessano | .... |
| 11:10.15 | cusco | is there any miss-configuration in the files that would resuylt in rRED alarm? |
| 11:10.22 | tzafrir_laptop | cusco: is it E1 or T1? |
| 11:10.23 | cusco | or must it be hardw? |
| 11:10.32 | cusco | E1... i think |
| 11:10.47 | cusco | yrp e1 |
| 11:11.17 | cusco | u think it must be cabling? |
| 11:11.27 | cusco | no chance of wrong configuration |
| 11:11.45 | tzafrir_laptop | Is the card configured as E1 or as T1 ? |
| 11:12.22 | cusco | E1 |
| 11:12.54 | cusco | how do I double-check? |
| 11:13.02 | cusco | the thing is.. we have 2 cards, both not working |
| 11:13.04 | hipitihop | tzafrir_laptop, leave -0ubuntu2 I assume |
| 11:13.08 | cusco | differente PRIs too |
| 11:13.26 | hipitihop | tzafrir_laptop, curretn first line is asterisk (1:1.6.2.0~rc2-0ubuntu2) karmic; urgency=low |
| 11:13.27 | drmessano | New circuits? |
| 11:13.33 | tzafrir_laptop | hipitihop, doesn't really matter. |
| 11:14.01 | drmessano | Cusco: NEW CIRCUITS as well? |
| 11:14.33 | cusco | no... what circuits? |
| 11:14.56 | hipitihop | tzafrir_laptop, I'm changing to "asterisk (1:1.6.2.5~-0ubuntu2) karmic; urgency=low" ok ? |
| 11:15.01 | drmessano | The thing your Digium card plugs into |
| 11:15.19 | cusco | the thing is... how do I debug a RED alarm configuration-wise? |
| 11:15.28 | cusco | same circuits |
| 11:15.30 | cusco | same cables |
| 11:15.39 | cusco | new machine, new cards |
| 11:15.54 | drmessano | Wheres the old config? |
| 11:16.13 | cusco | we have it, but it's no good because we had a different card |
| 11:16.20 | cusco | but we've tried putting it too, same problem |
| 11:16.20 | drmessano | So?? |
| 11:16.30 | drmessano | The parameters are the same |
| 11:16.51 | drmessano | It should be obvious what's specific to the card |
| 11:16.56 | hipitihop | tzafrir_laptop, althouhg I see it is mentioned also on line 7 |
| 11:17.44 | hipitihop | tzafrir_laptop, aah seems like comments form previous builders |
| 11:18.39 | cusco | erm.. its not quite the same. where before I would state: group=1 |
| 11:18.39 | cusco | channel => 1-15,17-31 |
| 11:18.39 | cusco | now it states: |
| 11:18.39 | cusco | group=1 // channel => 1-23 |
| 11:18.57 | drmessano | Yeah |
| 11:18.59 | cusco | dahdi_scan shows different dchan on this card |
| 11:19.09 | cusco | anyway it seems all good |
| 11:19.16 | drmessano | So make it right |
| 11:19.16 | cusco | can I debug red alarm anyway |
| 11:19.30 | cusco | drmessano: dchan is 24 |
| 11:19.32 | cusco | so it is right |
| 11:19.41 | drmessano | You have an E1? |
| 11:19.44 | cusco | yes |
| 11:19.56 | hipitihop | tzafrir_laptop, anyway, I cahnged line one, saved and exited... dch warned "no orig tarball found for the new version." |
| 11:20.58 | tzafrir_laptop | ./debian/rules get-orig-source |
| 11:22.06 | drmessano | [06:18] <cusco> channel => 1-15,17-31 |
| 11:22.10 | drmessano | dchan is 16 |
| 11:22.19 | hipitihop | tzafrir_laptop, I'm still in 1.6.2.0 directory, I asume back out a level ? |
| 11:22.39 | drmessano | Like I said, you see what needs to be fixed |
| 11:23.54 | hipitihop | tzafrir_laptop, if ./debian/rules get-orig-source was meant for me then "No such file or directory" |
| 11:24.32 | drmessano | cusco, fix the config and restart dahdi |
| 11:25.11 | cusco | drmessano: it is fixed. believe you mee I tried many typoes of config |
| 11:25.21 | drmessano | Did you fix the system.conf? |
| 11:25.26 | drmessano | Did you restart dahdi? |
| 11:25.38 | drmessano | No, i dont mean asterisk or reload asterisk, I mean DAHDI |
| 11:25.47 | cusco | YES |
| 11:25.50 | cusco | !!!!!!!!!!!!!!!!! |
| 11:25.55 | drmessano | Pastebin the configs |
| 11:26.31 | drmessano | [06:19] <cusco> drmessano: dchan is 24 |
| 11:26.31 | drmessano | [06:19] <cusco> so it is right <---- this is WRONG, which is why everything is now in total question |
| 11:31.32 | drmessano | Duh |
| 11:32.37 | hipitihop | tzafrir_laptop, appreciate your patience so far ... how do we get around this problem |
| 11:37.20 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 11:37.26 | cusco | I'm back |
| 11:37.45 | cusco | we have now seen that we have type=digital-T1 in one card and digital E1 in the other |
| 11:38.00 | cusco | tha's not right, i think |
| 11:38.16 | cusco | where do I configure this? |
| 11:38.49 | *** join/#asterisk P424D0X (~DL7RAY@2001:470:1f0b:449:215:f2ff:fef4:a244) |
| 11:40.53 | P424D0X | I get the message "Registration from 'sip' failed for 'ip' - no matching peer found. What can i do?! |
| 11:49.45 | Gugge | does the error really show 'sip' and 'ip' ? |
| 11:54.09 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 11:56.33 | P424D0X | Gugge: No.. SIP="sip:1234567@192.168.1.1" and IP="192.168.1.2" |
| 11:59.15 | hipitihop | hmm, looks like I have lost tzafrir_laptop |
| 12:00.12 | *** join/#asterisk lesouvage (~lesouvage@82.73.69.76) |
| 12:05.05 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:07.53 | *** join/#asterisk kalib (~lkhlui@osiris.aspec.com.br) |
| 12:08.56 | kalib | Hi there. Im my company the asterisk is configured to receive calls only on week days. Today we do have to work and the guy who was supposed to care about asterisk is not here. So the incoming calls are not going to the ULA. In which file should I check this? |
| 12:09.45 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 12:10.54 | kalib | ops.. *URA |
| 12:15.36 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 12:15.53 | *** join/#asterisk pawz (~pawz@ppp118-208-8-80.lns20.bne1.internode.on.net) |
| 12:34.31 | micols | - Called local/2230185 [Feb 27 13:04:16] WARNING[16921]: chan_sip.c:15902 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '10fb22546fa0013b281eaead604da94a@87.72.215.77'. Giving up. -- SIP/local-082eb708 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) |
| 12:34.42 | micols | any idea what this means? non-existing call leg on other UA ? |
| 12:34.51 | micols | i just get a busy tone |
| 12:34.54 | micols | and cannot call out |
| 12:35.01 | micols | no in/out |
| 12:35.42 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 12:36.10 | *** join/#asterisk V4mpire (~gary@82.118.111.252) |
| 12:39.02 | tzafrir_laptop | hipitihop, here? |
| 12:39.26 | tzafrir_laptop | ./debian/rules does not exists, or reports a problem? |
| 12:48.00 | dlynes | tzafrir_laptop, fwiw, he's probably running ubuntu, too |
| 12:48.18 | tzafrir_laptop | he does |
| 12:48.24 | tzafrir_laptop | (I'm not) |
| 12:48.59 | dlynes | kalib, ura? ula? |
| 12:52.40 | *** join/#asterisk mangefoo (~mangefoo@c83-250-205-114.bredband.comhem.se) |
| 13:07.29 | *** join/#asterisk superc (~superc@port-87-234-216-82.static.qsc.de) |
| 13:09.15 | superc | Hi. I try to send a fax from within a phpagi script. after the SENDFAX command ends I see the message 'WARNING[23030]: app_fax.c:764 transmit: Transmission error' |
| 13:09.36 | superc | This is probably not the problem. But then the php script is aborted. Why? |
| 13:10.01 | superc | I see 'AGI Tx >> 200 result=-1' after that warning |
| 13:11.08 | superc | AGI Script fax.php completed, returning -1 |
| 13:12.47 | superc | isn't there a way to stop asterisk killing the php script? |
| 13:17.40 | *** join/#asterisk puzzled (~patrick@535335AA.cable.casema.nl) |
| 13:18.19 | puzzled | hi |
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| 15:18.15 | tcpip4000 | Hi all, I'm having some problems receiving a fax using fax for asterisk, the system show the error: retrans_pkt: Maximum retries exceeded on transmission |
| 15:18.44 | tcpip4000 | using asterisk 1.4.25.1 |
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| 15:34.04 | birnenkind | hoi |
| 15:34.17 | birnenkind | my asterisk service does not start, how can i trace the problem? |
| 15:34.20 | smooth_penguin | ahoi |
| 15:35.35 | birnenkind | ./asterisk restart gives no error |
| 15:36.05 | smooth_penguin | add set -x to the init.d script |
| 15:36.09 | smooth_penguin | and take a peek |
| 15:37.58 | kaldemar | birnenkind: start asterisk directly with the binary, "asterisk -vvvvvc" |
| 15:38.26 | kaldemar | birnenkind: you'll see output for each loaded module and where it fails if it doesn't start. |
| 15:40.14 | *** part/#asterisk tcpip4000 (~fulano@200.26.136.114) |
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| 16:12.30 | jkroon | hi guys, having a problem with a link that keeps dropping and recovering. the remote device is a 30 channel PRI SIM-farm unit (fct mux), E1, with crc4. |
| 16:12.37 | jkroon | how do I go about trouble shooting it? |
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| 17:00.18 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 17:14.02 | zonyl | Hello. Whenever my ISP goes offline, my asterisk box will fail to re-register. I have read a lot about UDP timeouts, however, I have seen some conflicting information. My NAT router is set to timeout UDP unreplied at 10 secs. It appears asterisk is retrying every 20secs. Do I need to increase or decrease my UDP timeout or something else? |
| 17:14.52 | jkroon | connection tracking usually. |
| 17:15.04 | jkroon | switch your asterisk off for 60 seconds, then restart, if that works, buy a better router. |
| 17:15.32 | zonyl | jkroon: Yes, that works. I have a WRT with Tomato on it (I thought that was pretty good). |
| 17:15.53 | jkroon | ok, if you run tcpdump on the box, does the REGISTER requests continue going out? |
| 17:16.10 | jkroon | or is your renew time perhaps too long? in other words, your asterisk thinks it's still registered? |
| 17:16.52 | zonyl | asterisk shows it is unregistered and it is currently attempting (getting a console message every time it does) |
| 17:17.36 | zonyl | Actually it shows it as unreachable |
| 17:18.06 | zonyl | going to find the tcpdump binary |
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| 17:31.18 | ruben23 | hi, does jabber module included with asterisk 1.4 by default during compile..? oe i need to select it..on the menu select..? |
| 17:31.23 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 17:31.28 | ruben23 | to support gtalk |
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| 17:38.38 | p3nguin_ | ruben23: module show like jabber |
| 17:39.46 | ruben23 | <PROTECTED> |
| 17:40.27 | p3nguin_ | ruben23: I clearly meant: On Asterisk's CLI, run "module show like jabber" and see what shows. |
| 17:41.06 | ruben23 | <PROTECTED> |
| 17:42.20 | ruben23 | sorry i got this----> 0 modules loaded |
| 17:42.38 | p3nguin_ | res_jabber.so AJI - Asterisk Jabber Interface 0 |
| 17:42.51 | p3nguin_ | So that means you didn't install the module, but I did. |
| 17:43.29 | ruben23 | <PROTECTED> |
| 17:43.50 | p3nguin_ | Do you REALLY think that will work? |
| 17:44.31 | ruben23 | <PROTECTED> |
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| 17:50.17 | idespinner | ruben23, I would suggest recompiling with make menuselect |
| 17:53.43 | *** join/#asterisk smooth_penguin (~smoove@115.240.36.97) |
| 17:54.33 | ruben23 | idespinner:ok thanks i will do that.. |
| 18:01.10 | p3nguin_ | ruben23: What do you plan to do with res_jabber? |
| 18:01.58 | p3nguin_ | I presume you want to connect to Gtalk, but what's your goal? |
| 18:03.00 | ruben23 | <PROTECTED> |
| 18:03.22 | p3nguin_ | ruben23: Your phone doesn't have "missed calls" on it? |
| 18:04.12 | ruben23 | the problem is the incoming call are all routed to agent, but when all agent are busy attending others..i cant track the missed call |
| 18:04.21 | *** join/#asterisk smooth_penguin (~smoove@115.240.114.186) |
| 18:04.24 | p3nguin_ | I see. |
| 18:04.31 | ruben23 | might gtalk helps |
| 18:04.37 | p3nguin_ | possible |
| 18:05.03 | ruben23 | 1st problem to activate it..and the jabber module.. |
| 18:05.28 | ruben23 | 2nd, is the dial plan setup with our existing dial plan for production how i integrate it.. |
| 18:05.40 | ruben23 | to add the function. |
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| 18:15.59 | Micc | ruben23, we do exactly that with a 24 hour emergency call center. It works great. We also tell them when the user selects spanish so they can answer in spanish |
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| 18:22.23 | ChannelZ | Last time I tried to play around with gtalk in asterisk I discovered the Google talk client doesn't seem to have a dialpad, so you can't use IVRs. |
| 18:22.24 | *** join/#asterisk Rapier1024 (~Rapier102@d-69-161-89-227.cpe.metrocast.net) |
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| 18:23.09 | Rapier1024 | hi folks, can anyone answer a question for me about the permit and deny fields in sip.conf? |
| 18:23.21 | ChannelZ | If you ask one |
| 18:24.14 | Rapier1024 | I'm curious if there is a way to get asterisk to resolve a dynamic domain name for the permit field |
| 18:24.49 | Rapier1024 | I've got a phone outside the lan, and that would help keep unauthorized users out of the system. |
| 18:26.39 | Rapier1024 | I've tried entering: permit=phone123.dyndns.org / 255.255.255.255 but it denies the login |
| 18:27.34 | ChannelZ | I'm pretty sure ip/mask are only allowed, not hostnames |
| 18:27.46 | ruben23 | Micc: very nice.. |
| 18:28.12 | ruben23 | this is inbound calls only right..? |
| 18:28.40 | Rapier1024 | this is a type friend extention. It gets dropped into the local context, so no. |
| 18:28.55 | drmessano | I guess cusco never figured out whether he had an E1 or T1, or what exactly they were |
| 18:29.26 | Rapier1024 | but when it goes to register, it is an inbound, so yes, I guess that would be accurate ruben23 |
| 18:31.12 | Micc | yes, inbound calls only |
| 18:31.29 | ruben23 | <PROTECTED> |
| 18:31.31 | Micc | It could send a gtalk message on outbound calls too, but they don't care about that. |
| 18:32.02 | Rapier1024 | sorry, wrong conversation, :) |
| 18:32.31 | Micc | ruben23, sure, what do you need to know? I just followed the instructions. |
| 18:32.35 | Rapier1024 | will asterisk take a variable in place of a hard ip/mask? |
| 18:33.02 | Micc | Rapier1024, in what? Dialplan? Conf file? |
| 18:33.10 | Rapier1024 | sip.conf |
| 18:33.24 | Rapier1024 | as part of the permit= line |
| 18:33.35 | ruben23 | Micc: you have guides for it or documentation. |
| 18:33.37 | Micc | Rapier1024, that I'm not sure. I've never needed to do anything like that. |
| 18:34.26 | ruben23 | my purpose is only integrate to inbound calls so i can get the numbers of missed calls and do a follow up call to them.. |
| 18:34.26 | Micc | ruben23, do you use voip-info.org? |
| 18:34.40 | *** join/#asterisk CheBuzz_Home (~CheBuzz_H@88-202-61-65.ip.skylogicnet.com) |
| 18:34.42 | *** join/#asterisk theBruno (~ChrisBrun@32.128.35.85) |
| 18:34.44 | Rapier1024 | hmmm |
| 18:34.55 | ruben23 | Micc: yes i followed it, but there is not dial plan integration for inbound... |
| 18:35.14 | *** part/#asterisk theBruno (~ChrisBrun@32.128.35.85) |
| 18:35.30 | Micc | http://www.voip-info.org/wiki/view/Asterisk+Jabber |
| 18:35.41 | Micc | It has the dialplan stuff in there. |
| 18:36.02 | ruben23 | ow ok..the script..? |
| 18:36.19 | Micc | jabbersend is the dialplan command. |
| 18:37.06 | CheBuzz_Home | I have a question. I have my asterisk server behind NAT on a 192.168.3.0/24 network. I can register just fine. But when I go to make a call, the INVITE seems to go just fine to the public IP address, but then when I get the "Packet2Packet bridging" message on the console, for some reason my SIP phone begins sending data to the servers local IP (192.168.3.*) instead of it's public IP. Any ideas why this is? |
| 18:37.24 | drmessano | ~sipnat |
| 18:37.25 | infobot | [~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 |
| 18:37.36 | drmessano | Follow that |
| 18:38.48 | p3nguin_ | chebuzz_home: "its" public IP |
| 18:39.13 | CheBuzz_Home | Huh, a grammar nazi in #asterisk :) |
| 18:39.32 | CheBuzz_Home | Maybe I meant it is and just suck at teh English |
| 18:39.48 | p3nguin_ | Like I've never heard that term before. |
| 18:39.53 | ruben23 | Micc:thanks, so much |
| 18:39.56 | drmessano | p3nguin_, What are you're problem; I are spake the penultimate vocabularyism |
| 18:40.16 | drmessano | Hold on |
| 18:40.22 | drmessano | Let me count my score |
| 18:41.08 | drmessano | 48 + 50 for using penultimate incorrectly.. |
| 18:41.11 | p3nguin_ | It's always the bratty ones that try to hurt my feelings with the "grammar nazi" retaliation. |
| 18:41.36 | hardwire | there trying to point something out.. not hurt you.. |
| 18:41.53 | drmessano | p3nguin_, Stop correcting my poor english please.. |
| 18:42.01 | hardwire | heh |
| 18:42.21 | p3nguin_ | falls off the chair |
| 18:42.29 | drmessano | p3nguin_, I don't need your help not sounding like an illiterate 4th grade dropout. I can not not do that on my own. |
| 18:42.45 | hardwire | so anyways |
| 18:42.49 | hardwire | I'm awesome.. that is all. |
| 18:42.55 | hardwire | leaves for works. |
| 18:43.28 | drmessano | I spent 20 mins arguing over a bundt cake pan |
| 18:43.39 | hardwire | over or over? |
| 18:43.40 | drmessano | I can handle a "Your, you're, you are" |
| 18:43.47 | p3nguin_ | What was the other person trying to call it? |
| 18:43.51 | drmessano | lol |
| 18:43.59 | p3nguin_ | bunt? butt? |
| 18:44.06 | hardwire | bunted. |
| 18:44.29 | drmessano | I spent 20 minutes engaged in an argument concerning a bundt cake pan |
| 18:45.39 | hardwire | the pan was conerned? |
| 18:45.40 | p3nguin_ | drmessano: As hard as it may be to believe, I will occasionally run into a person that actually thanks me for letting them know about the incorrect possessive pronouns they are using. |
| 18:45.47 | hardwire | concerned. |
| 18:46.04 | drmessano | p3nguin_, It really doesn't bother me.. I just ramble |
| 18:46.16 | drmessano | and it's "thank's" |
| 18:46.22 | hardwire | p3nguin_: I think it's odd that you're concerned with how ineffective peoples insults are.. when in reality most people are insulted to be corrected in the first place. |
| 18:46.24 | hardwire | it's odd. |
| 18:46.37 | hardwire | it doesn't really bother me either.. until I think about it enough to let it. |
| 18:47.03 | drmessano | Take this bundt cake pan, for example |
| 18:47.19 | drmessano | I haven't made anything "bundt" in almost two years |
| 18:48.20 | drmessano | But, it's mine |
| 18:48.44 | drmessano | I can not not use it all I want |
| 18:49.03 | CheBuzz_Home | p3nguin_: I was trying to hurt your feelings. |
| 18:49.08 | drmessano | lol |
| 18:49.37 | drmessano | Its like I learned from Ghostbusters |
| 18:49.39 | hardwire | I think 'Grammar Nazi' is shorthand for a few pages of flames. |
| 18:49.58 | p3nguin_ | There is very little that a person can say on here to actually affect me. |
| 18:50.25 | drmessano | In order to fight a little evil, sometimes you need to strap the unlicensed nuclear accelerator to your back and switch it on |
| 18:50.35 | hardwire | p3nguin_: your shoes are untied.. |
| 18:50.40 | drmessano | So what if it's untested |
| 18:50.43 | hardwire | crap.. I can't physically do anything to you. |
| 18:50.48 | [TK]D-Fender | p3nguin_: I can think of at least 3 things :) |
| 18:50.57 | CheBuzz_Home | So here's another question: is there anyway to set core verbosity in the config files so I don't have to set in on the console every time I need to restart *? |
| 18:51.10 | drmessano | yes |
| 18:51.12 | hardwire | asterisk.conf |
| 18:51.21 | drmessano | He didn't ask where |
| 18:51.24 | drmessano | "yes" |
| 18:51.27 | drmessano | Next question |
| 18:51.39 | hardwire | drmessano: FINE! |
| 18:51.42 | CheBuzz_Home | Haha, where? |
| 18:51.50 | drmessano | asterisk.conf |
| 18:51.54 | hardwire | Does anybody here like nachos? |
| 18:51.56 | CheBuzz_Home | Ah, thanks. |
| 18:52.06 | drmessano | "nacho's" |
| 18:52.14 | hardwire | Natcho's. |
| 18:52.29 | hardwire | Hey maaaan.. that's naacchoooo cheeese. |
| 18:52.36 | drmessano | We have this mexican/soul food place here called "Nacho Mama's" |
| 18:52.47 | ManxPower-work | Int3rg10^e |
| 18:52.52 | hardwire | This Nacho Mama's Cookin? |
| 18:52.56 | drmessano | Yep |
| 18:53.10 | hardwire | drmessano: I wanna start a place called F'n A's |
| 18:53.16 | hardwire | food so good you curse when you taste it. |
| 18:53.22 | drmessano | Just like the james brown joke with the Halo Statues |
| 18:53.57 | hardwire | -> breakfast.. my fast must be broken. |
| 18:54.46 | CheBuzz_Home | Well, you guys are crazy. But thanks for the help. That did solve my problem. |
| 18:54.48 | p3nguin_ | [tk]d-fender: I should have been more specific in saying that comments with negative connotation don't bother me. On the other hand, if you give me some beneficial feedback such as telling me that I have incorrectly used a word, like saying except when I should have said accept, that actually does affect me in that it can improve myself and try to never make the mistake again. |
| 18:55.49 | drmessano | Where he asks one of his hired hands to get him, and you have to sound it out, "halo statues".. So they get him all these angel statues with big halo's on them.. and he says "no no, I want the thing on the wall where they ring the doorbell and I can push the button and say "halo, statue?" |
| 18:56.15 | p3nguin_ | halo's, huh? |
| 18:56.22 | Micc | ruben23, did you get that jabber thing figured out? it should be pretty easy. |
| 18:56.25 | drmessano | Hello, s'that you |
| 18:57.03 | ruben23 | Micc:im recompileing my asterisk and others now to have the jabber module loaded.. |
| 18:57.03 | drmessano | One of those elephant shoes sort of jokes |
| 18:57.33 | *** part/#asterisk CheBuzz_Home (~CheBuzz_H@88-202-61-65.ip.skylogicnet.com) |
| 18:58.24 | drmessano | Surely setting up postfix to mail relay to google has to be easier than this crap |
| 19:00.56 | p3nguin_ | What's the issue? |
| 19:00.58 | raden_work | http://pastebin.com/b2g39w5D |
| 19:01.08 | raden_work | when i dial from the office number comes up unavailable |
| 19:02.03 | drmessano | I'm not entirely certain.. but I have followed what i know of the examples online and it fails authentication, I think because of the CNAME's they use and the certificate name for the TLS session |
| 19:02.33 | drmessano | I only vaguley remember the error now from last week.. But the more examples I see of how this should be set up, they all vary so widely |
| 19:03.19 | p3nguin_ | Are you simply trying to configure google to be your relayhost? |
| 19:03.30 | drmessano | yes |
| 19:04.52 | p3nguin_ | What hostname are you setting for postfix's relayhost? |
| 19:05.44 | drmessano | relayhost = [smtp.gmail.com]:587 |
| 19:06.05 | raden_work | p3nguin_, I cant seem to get my gotoif callerid thing working |
| 19:06.07 | raden_work | http://pastebin.com/b2g39w5D |
| 19:06.54 | p3nguin_ | drmessano: What about using 465 rather than the submission port? |
| 19:07.53 | raden_work | how can i output the callerid variable to console ? |
| 19:07.56 | drmessano | Asking me to think back a ways.. but there was a reason.. All the examples use 587 for that reason |
| 19:08.05 | drmessano | Whatever it may be |
| 19:09.05 | *** join/#asterisk puzzled (~patrick@535335AA.cable.casema.nl) |
| 19:09.25 | drmessano | Postfix does not deliver mail via TCP port 465 (the obsolete "wrappermode" protocol). See TLS_README for a solution that uses the "stunnel" command. |
| 19:10.03 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 19:10.18 | Micc | raden_work, have you tried NoOp(${CALLERID(number)}) ? |
| 19:10.42 | raden_work | Misc just did that |
| 19:10.55 | raden_work | what I need is if its a 3 digit number to set it as well |
| 19:10.56 | p3nguin_ | I think Verbose() would be a better command, since NoOp() is only going to show up with verbosity increased. |
| 19:11.01 | raden_work | phones are passing there extension number |
| 19:12.22 | raden_work | how would I do if caller id = xxx ???? |
| 19:12.27 | raden_work | is there a way todo that ????? |
| 19:12.49 | Micc | only 3 digits, no more? |
| 19:13.35 | p3nguin_ | drmessano: Could the problem be within your sasl_passwd file? |
| 19:13.59 | Maliuta | raden_work: so if ${CALLERID} == pr0n; do doggy(); ??? ;) |
| 19:14.22 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:14.22 | p3nguin_ | drmessano: It appears that you set your gmail credentials within that file. |
| 19:14.44 | Micc | Anyone know a good AEL tutorial? |
| 19:14.47 | drmessano | I've done that |
| 19:15.08 | raden_work | Micc, correct |
| 19:15.14 | raden_work | all our extensions are 3 digits |
| 19:15.37 | raden_work | <PROTECTED> |
| 19:15.50 | raden_work | thats what callerid is getting set to... |
| 19:16.10 | raden_work | exten => _NXXNXXXXXX,n,GotoIf($["${CALLERID(number)}" = ""]?setcid:pass) <<< is not working cause the field not empty |
| 19:16.34 | raden_work | so i need if empty or less than 3 digits or less than 10 actually |
| 19:16.41 | p3nguin_ | If it isn't empty, then it will jump to the "pass" label. |
| 19:17.05 | raden_work | p3nguin_, and thats my problem cause the CID being set as the extension in asterisk |
| 19:17.08 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.16 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 19:17.26 | p3nguin_ | raden_work: It will be set to whatever you have configured it to be set as. |
| 19:17.30 | Maliuta | Raden: check for if digits and < 999999999 ??? |
| 19:17.35 | raden_work | p3nguin_, so I need it to go to setcid if it less than 10 digits basically |
| 19:17.57 | raden_work | Maliuta, hmmm |
| 19:18.01 | p3nguin_ | raden_work: If it is the same as the extension, fine. If it is set to the phone's account number, fine. If it is something completely different, that's okay too. |
| 19:18.03 | Maliuta | raden_work: that solution is a major hack |
| 19:18.04 | Micc | anyone have experience with zycoo ip phones? are they any good? I can't find any pricing anywhere. |
| 19:18.41 | raden_work | cant i just have a line that goto if callid 3 digits like XXX ? |
| 19:18.57 | raden_work | it has to be this way for many complex reasons :( |
| 19:19.08 | Micc | raden_work, http://www.voip-info.org/wiki/view/Asterisk+func+len |
| 19:19.17 | Micc | will that help? |
| 19:20.07 | raden_work | I think so |
| 19:20.27 | raden_work | now how do i do a goto if that variable less than 9 ? |
| 19:20.46 | Maliuta | raden_work: have you though about passing that section of code out to an agi? |
| 19:21.04 | p3nguin_ | Do all outbound calls need to have the same CALLERID(num) on them? |
| 19:21.13 | raden_work | p3nguin_, no thats the issue :( |
| 19:21.13 | Micc | GotoIf($[Len(${CALLERID(number)}) = 3]?setcid:pass) |
| 19:21.40 | p3nguin_ | raden_work: Does every peer have its own outbound CID number? |
| 19:21.50 | raden_work | p3nguin_, no all 1 phone number 20 channels |
| 19:21.57 | raden_work | p3nguin_, I cannot set in phones context |
| 19:22.12 | p3nguin_ | Your information isn't adding up. |
| 19:22.21 | raden_work | cause if there is a redirect in the database then its just the callerid num in the context and they want to know whos calling them :( |
| 19:22.48 | raden_work | so if extension 101 has a redirect to his cell he dont want to see the shop number he wants to see number calling him etc... |
| 19:23.13 | p3nguin_ | Are you using followme? |
| 19:23.26 | raden_work | if i set cid in sip.conf under [101] it will use the shop callerid all the time |
| 19:23.46 | raden_work | p3nguin_, exten => _*72.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:3}) |
| 19:24.08 | raden_work | p3nguin_, and then also the callerid of all the phones being the same messes that up as well :( |
| 19:24.17 | raden_work | so it has to be done in outbound :( |
| 19:24.21 | p3nguin_ | I have one regular outbound context where all calls get the regular phone number, and then I also have a followme_outbound context which does NOT force any CID on it, allowing the already-attached CID to pass along. |
| 19:24.42 | raden_work | I just need a quick solution :( |
| 19:24.49 | raden_work | Micc, working with yours right now :) |
| 19:24.54 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 19:25.43 | Micc | raden_work, I need to do something similar with my dialplans. I have a dialout macro that sets the callerid, but I have some internal numbers that I also need to set the callerid to something other than the 3 digit internal number. |
| 19:25.46 | raden_work | Micc, is there a way to set a range in that field like 1-9 ? |
| 19:26.32 | Micc | raden_work, yeah you would have to use an or |
| 19:26.39 | Micc | < 1 or > 9 |
| 19:26.55 | Micc | not sure the exact syntax. I'd have to look it up. |
| 19:27.00 | raden_work | [Feb 27 13:26:38] ERROR[30816]: ast_expr2.y:1065 op_func: Error! 'Len' is not possibly a function name! |
| 19:27.19 | Micc | raden_work, core show function len |
| 19:27.31 | p3nguin_ | drmessano: Now you've got me wanting to set up gmail relaying just to see if I can get it working. |
| 19:27.36 | Micc | Oh, sorry, its LEN |
| 19:27.38 | Micc | all caps |
| 19:27.52 | raden_work | Just noticed that |
| 19:28.03 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.173.128.dsl.dyn.forthnet.gr) |
| 19:28.09 | Micc | p3nguin_, Setting up a relay for gmail is on my todo list too. |
| 19:28.32 | p3nguin_ | I'm going to get started in 1.7 minutes. |
| 19:28.32 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 19:28.49 | raden_work | NoOp("SIP/103-08264aa0", "103") in new stack |
| 19:28.49 | raden_work | [Feb 27 13:28:27] NOTICE[30829]: ast_expr2.y:757 compose_func_args: argbuf allocated 4 bytes; |
| 19:28.49 | raden_work | [Feb 27 13:28:27] NOTICE[30829]: ast_expr2.y:776 compose_func_args: argbuf uses 3 bytes; |
| 19:28.53 | raden_work | it worked |
| 19:29.10 | *** part/#asterisk Rapier1024 (~Rapier102@d-69-161-89-227.cpe.metrocast.net) |
| 19:29.18 | raden_work | ok that must have had nothing todo with it |
| 19:29.38 | raden_work | Micc, so now how do i make it if it 0-9 digits ???? im trying to figure something out but coming up blank :( |
| 19:30.10 | [TK]D-Fender | raden_work: how about showing us what you've got so far... |
| 19:30.34 | raden_work | exten => _1NXXNXXXXXX,n,GotoIf($[LEN(${CALLERID(number)}) = 3]?setcid:pass) |
| 19:30.34 | raden_work | exten => _1NXXNXXXXXX,1,GotoIf($["${CALLERID(number)}" = ""]?setcid:pass) |
| 19:31.00 | raden_work | works for 3 digit extensions to set CID and passing through redirects from db nicely |
| 19:31.09 | afink | hello all, I am trying to use g729 in pass through mode. I think I have something wrong in somewhere. When making a call to my * system I immediately get this error: [Feb 27 13:28:04] NOTICE[3046]: chan_sip.c:5634 process_sdp: No compatible codecs, not accepting this offer! Is there a place besides sip.conf where I need to tell * to use g729 |
| 19:31.11 | afink | ? |
| 19:32.29 | raden_work | [TK]D-Fender, Id like to have exten => _1NXXNXXXXXX,n,GotoIf($[LEN(${CALLERID(number)}) = >9 ?setcid:pass) but that dont work |
| 19:32.46 | raden_work | so if the digit count was 9 or less it would setcid |
| 19:33.14 | Micc | raden_work, check out the logical operators http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 19:34.13 | Micc | raden_work, oh you need a comparison operator. |
| 19:34.22 | Micc | raden_work, >= |
| 19:34.47 | raden_work | yeah i got it :) |
| 19:34.57 | raden_work | i had = > 9 illogical :( |
| 19:34.58 | [TK]D-Fender | raden_work: you didn't reference LEN as a FUNCTION. |
| 19:35.11 | Micc | raden_work, I thought it needed to be greater than 0 though. so you would need to do < 1 | > 9 |
| 19:35.30 | raden_work | 0-9 I want it to setcid 10 pass |
| 19:35.43 | raden_work | [TK]D-Fender, refrence howso ? |
| 19:36.15 | [TK]D-Fender | raden_work: ($[ <---- you don't start a function refernce with a square brace... |
| 19:36.39 | [TK]D-Fender | raden_work: You appear to be attempting to abbreviate having a function reference inside of an EXPRESSION |
| 19:36.42 | Micc | TKD-Fender, thats to evaluate |
| 19:37.01 | Micc | TKD-Fender, isn't the whole thing an expression? |
| 19:37.16 | raden_work | so it should be (${ ???? |
| 19:37.39 | [TK]D-Fender | Micc: You need to format the expression. You need to make the complete function reference. you can't just smash the syntax into one broken hybrid |
| 19:37.58 | raden_work | Your loosing me with your big words LOL |
| 19:38.10 | [TK]D-Fender | raden_work: You need the $[ for the expression. You ALSO need the ${ to start the function reference |
| 19:38.12 | Micc | raden_work, if yoiur going to use $[ you have to close it ] |
| 19:38.27 | [TK]D-Fender | Micc: ARGH |
| 19:38.39 | [TK]D-Fender | Holy shit you people can't count braces... |
| 19:38.57 | raden_work | I got it :) |
| 19:38.59 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 19:39.11 | [TK]D-Fender | exten => _1NXXNXXXXXX,n,GotoIf($[${LEN(${CALLERID(number)})} >9]?setcid:pass) |
| 19:39.18 | raden_work | hands [TK]D-Fender beer |
| 19:39.35 | raden_work | [TK]D-Fender, thanks for point that out :) |
| 19:39.50 | [TK]D-Fender | $[] <- outer wrapper of expression |
| 19:40.07 | [TK]D-Fender | ${LEN()} <-- LEN still needs the wrapper for it to be referenced |
| 19:40.42 | raden_work | exten => _1NXXNXXXXXX,1,GotoIf(${LEN(${CALLERID(number)}) > 9 }?setcid:pass) |
| 19:40.55 | raden_work | working |
| 19:40.59 | [TK]D-Fender | raden_work: I just GAVE you the completed answer.... |
| 19:41.08 | [TK]D-Fender | reaches for his ClueBat (tm) |
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| 19:43.38 | raden_work | [TK]D-Fender, so why does $[ ] have to be around it, Thats my only question at the moment |
| 19:45.33 | Micc | I'm not good for giving exact syntax. I can bearly see without my glasses. looking up that kind of stuff would take me a while. |
| 19:45.47 | raden_work | LOL :) |
| 19:46.05 | raden_work | I get where TK going doing some reading has todo with precedence |
| 19:46.14 | [TK]D-Fender | raden_work: $[] <- this means the contents are an EXPRESSION. |
| 19:46.28 | raden_work | [TK]D-Fender, Thanks ;) |
| 19:46.55 | [TK]D-Fender | raden_work: If you don't have it, then operators like > don't evaluate and that leave just a bunch of crap characters before the ? to be judged on. |
| 19:47.17 | [TK]D-Fender | [14:40]<raden_work>exten => _1NXXNXXXXXX,1,GotoIf(${LEN(${CALLERID(number)}) > 9 }?setcid:pass) <--- this would evaluate as ..... |
| 19:47.53 | [TK]D-Fender | [14:40]<raden_work>exten => _1NXXNXXXXXX,1,GotoIf(12 > 9?setcid:pass) <- assuming you didn't screw up the } right before the ? |
| 19:48.46 | [TK]D-Fender | raden_work: Gotoif checks the contents (as parsed) before the ? are ZERO or NOT. Anything other than a ZERO = true. that includes being "FRED" |
| 19:49.26 | [TK]D-Fender | raden_work: Expressions are supposed to return 0 or 1 based on a true/false result. |
| 19:49.52 | [TK]D-Fender | raden_work: If you even have 1 junk space charater in there you have just screwed yourself |
| 19:49.59 | Micc | I want to learn how to do AEL dialplans. Regular dial plans are so hard to read. |
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| 19:51.21 | raden_work | NOT WORKING :( |
| 19:51.30 | ManxPower-work | raden_work, then you did something wrong. |
| 19:51.32 | raden_work | exten => _NXXNXXXXXX,n,GotoIf($[${LEN(${CALLERID(number)})} >9]?setcid:pass) |
| 19:52.21 | raden_work | what am I missing ? |
| 19:52.43 | ManxPower-work | pastebin the cli output, that's the way you debug this stuff |
| 19:54.40 | raden_work | http://pastebin.com/ASfjgABu |
| 19:55.13 | [TK]D-Fender | exten => _NXXNXXXXXX,n,GotoIf($[${LEN(${CALLERID(number)} >9]?setcid:pass) |
| 19:55.34 | [TK]D-Fender | ummm.. |
| 19:56.21 | ruben23 | hi can eyebeam softphones handles DTMF...? like hardphones...to be used for asterisk.. |
| 19:56.58 | [TK]D-Fender | ruben23: Of course |
| 19:57.07 | [TK]D-Fender | raden_work: where do we see the CID output? |
| 19:57.19 | ruben23 | [TK]D-Fender: with voip trunk using softphones..? |
| 19:57.24 | *** part/#asterisk jkroon (~jkroon@196.212.81.236) |
| 19:57.27 | [TK]D-Fender | ruben23: huh? |
| 19:57.47 | ruben23 | i mean im using voip for my softphones eyebeam.. |
| 19:57.52 | raden_work | [TK]D-Fender, ypou want me todo a noop before dial ? |
| 19:57.57 | [TK]D-Fender | raden_work: -- Executing [7152815035@outbound:1] NoOp("SIP/103-0824c160", "103") in new stack <-- THIS MEANS THE cid IS 3 CHARS LONG, RIGHT? HAVING 103 AS THE VALUE, RIGHT? |
| 19:58.10 | raden_work | [TK]D-Fender, correct |
| 19:58.14 | raden_work | and its not setting it |
| 19:58.16 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 19:58.23 | raden_work | lemee paste Extensions.conf |
| 19:58.43 | [TK]D-Fender | raden_work: well, 3 is NOT greater than 9, thus it returns FALSE. |
| 19:59.08 | raden_work | I need less than 9 duh :( |
| 19:59.10 | [TK]D-Fender | <PROTECTED> |
| 19:59.16 | raden_work | using to many examples and not my brain |
| 19:59.38 | [TK]D-Fender | reaches for his ClueBat (tm) again and just starts swinging |
| 19:59.42 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 20:03.17 | ManxPower-work | raden_work, sounds like fatigue to me |
| 20:03.42 | raden_work | being rushed to much |
| 20:05.19 | p3nguin_ | Hmm, gmail accepted my relayed mail with a 250 2.0.0 OK, but I still haven't received the mail in my other inbox. |
| 20:05.40 | jblack | did you check the spam filter? |
| 20:05.48 | p3nguin_ | Oh, it's greylisted. So that means gmail is relaying. |
| 20:06.14 | p3nguin_ | The server received it, but greylisted it... that's why it didn't show up in the inbox yet. |
| 20:06.24 | p3nguin_ | So gmail relay works fine. |
| 20:06.27 | p3nguin_ | simple setup. |
| 20:07.11 | Micc | raden_work, again, I was just pointing you in the right direction with > 9. |
| 20:07.26 | p3nguin_ | I'm not a big fan of this message, though: certificate verification failed for smtp.gmail.com[74.125.95.109]:587: untrusted issuer |
| 20:08.16 | raden_work | Micc, appreciate all the help I was just not thinking like i said |
| 20:09.25 | Gugge | p3nguin_: you just need to add the Thawte ca cert to your system |
| 20:09.40 | p3nguin_ | I thought I had it already. I'll double check. |
| 20:10.39 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:10.52 | Gugge | p3nguin_, maybe its Equifax, i cant really remember :) |
| 20:11.07 | *** join/#asterisk Tim_Toady (~moi@193.92.197.188.dsl.dyn.forthnet.gr) |
| 20:11.19 | p3nguin_ | /C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Premium Server CA/emailAddress=premium-server@thawte.com |
| 20:11.24 | p3nguin_ | You were right the first time. |
| 20:11.39 | p3nguin_ | I have /etc/ssl/certs/Thawte_Premium_Server_CA.pem |
| 20:12.27 | p3nguin_ | It is actually a symlink to /usr/share/ca-certificates/mozilla/Thawte_Premium_Server_CA.crt |
| 20:12.52 | Gugge | put the content from that file in the cacert.pem your postfix uses ... /etc/postfix/cacert.pem or whereever you have it :) |
| 20:13.48 | p3nguin_ | Eww, the relayed email arrived with my gmail account name as the sender rather than the actual sender. |
| 20:14.26 | Gugge | if i remember correctly you can add what mail adresses you are allowed to send from |
| 20:14.37 | Gugge | somewhere in gmails settings |
| 20:14.48 | p3nguin_ | I won't be relaying through gmail anymore, so I'm not going to bother with it. I just wanted to see if I could get postfix to do it without too much trouble. |
| 20:15.14 | p3nguin_ | It's good to know in case I want to do something with it later, though. |
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| 20:41.33 | hardwire | anybody seen an odbc connector for clouddb? |
| 20:41.41 | hardwire | err.. couchdb |
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| 21:11.41 | fuxu2 | anyone here have some suggestions with regards to blade servers? |
| 21:11.50 | raden_work | <PROTECTED> |
| 21:11.59 | jql | define "suggestion" |
| 21:12.34 | fuxu2 | preferences.. models to look at.. reasons for or against certain models.. etc |
| 21:13.10 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:13.14 | raden_work | why the F would you need a blade server for asterisk ? |
| 21:13.45 | fuxu2 | it's not for asterisk itself actually.. but it will be used for testing asterisk for awhile.. |
| 21:13.58 | raden_work | get whatever then |
| 21:14.09 | jql | not only do you have the fun a choosing between hp/ibm/dell, there are also atca-form systems to choose from |
| 21:14.11 | raden_work | your not going to bog it down |
| 21:14.26 | jql | lots of choices these days -- really depends on what additional needs you have |
| 21:14.40 | *** join/#asterisk brezular (~brezular@adsl-dyn61.78-99-24.t-com.sk) |
| 21:14.55 | jql | dell is a decent value, IMO. |
| 21:14.56 | *** join/#asterisk ketin (~asdfgas@cpe-24-242-47-201.hot.res.rr.com) |
| 21:15.11 | fuxu2 | jql: biggest thing right now is keeping it under 13k for the chassis and a blade |
| 21:15.24 | fuxu2 | ever tried the supermicro blades? |
| 21:15.24 | raden_work | what are u using for ? |
| 21:15.28 | ketin | Anyone willing to help a little with an asterisk/festival issue? |
| 21:15.38 | fuxu2 | raden: I do security testing |
| 21:15.46 | jql | I'm a bit gunshy about supermicro, but they have volume |
| 21:15.57 | fuxu2 | why are you gunshy about them? |
| 21:16.32 | fuxu2 | in short, I'm going to be setting up vm's to run for security testing. |
| 21:16.40 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 21:16.43 | ketin | Festival logs show a connection then disconnection immediatley, console output in ast looks fine and cache file for festival is generated but no audio. |
| 21:16.49 | dddh | hi |
| 21:16.53 | dddh | guyz! |
| 21:16.54 | jql | failure rate of their 1U servers have been hit-or-miss in my experience |
| 21:16.55 | dddh | ^_^ |
| 21:17.06 | fuxu2 | jql: that's good to know |
| 21:17.56 | dddh | I need help with my voip device, I wanna test if incoming calls are working or not |
| 21:17.57 | dddh | ;( |
| 21:18.35 | fuxu2 | dddh: what protocol? |
| 21:18.52 | dddh | fuxu2, sip |
| 21:19.14 | [TK]D-Fender | dddh: Want to test? TRY IT. |
| 21:19.46 | eppigy | 8[] |
| 21:19.47 | dddh | [TK]D-Fender, I do not have second voip-phone at home to call myself |
| 21:20.09 | jblack | I wonder how penguin is doing |
| 21:20.31 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 21:20.37 | dddh | I'll ask questions about setting asterisk later, right now I just want to test if my device works or not |
| 21:21.06 | ChannelZ | Softphone |
| 21:21.07 | jblack | Never mind. they went centos/redhat only. :( |
| 21:21.46 | [TK]D-Fender | dddh: What device? |
| 21:22.11 | [TK]D-Fender | dddh: What have you done on it to accept calls since you apparently haven't even set up * yet? What will it prove even if you try? |
| 21:22.16 | dddh | ChannelZ, some voip phone based on broadcom chipset, doesn't really matter |
| 21:22.53 | dddh | [TK]D-Fender, I have already setup an account at iptel, I can call some service numbers, but didn't call myself yet |
| 21:23.05 | dddh | ChannelZ, sorry, last message was for [TK]D-Fender |
| 21:23.50 | [TK]D-Fender | dddh: IPTEL is a full ITSP, or just a SIP peering provider? |
| 21:25.35 | jblack | has anyone bought a digitial camera in the last 12 months or so? I'm wondering whether or not shutter delay and light sensitivity has improved |
| 21:26.56 | dddh | [TK]D-Fender, iptel is iptel.org |
| 21:27.24 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 21:29.01 | [TK]D-Fender | dddh: Seems they are just a SIP proxy |
| 21:30.19 | dddh | [TK]D-Fender, actually I want to setup asterisk to point _sip._udp and _sip._tcp to my own server and use several providers for usa, uk and so on, but now I just have a sip voip phone |
| 21:30.50 | ketin | Eh, anyone know much about festival configuration? |
| 21:31.10 | [TK]D-Fender | dddh: Umm... that doesn't really say much.... |
| 21:31.55 | ketin | ¬ 03:16:43pm| ketin: Festival logs show a connection then disconnection immediatley, console output in ast looks fine and cache file for festival is generated but no audio. |
| 21:33.47 | dddh | [TK]D-Fender, my isp offered me pstn number, but I did not sign papers yet, that is why I am just playing with sip now, just need some method to call my sip: uri to test if I'm doing everything right or not |
| 21:37.19 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:41.30 | [TK]D-Fender | dddh: Provide a URI to call and someone might help out |
| 21:42.04 | dddh | ok |
| 21:43.47 | dddh | I planned to pm.. |
| 21:43.54 | dddh | sip:123450@iptel.org |
| 21:44.13 | ketin | [TK]D-Fender: Any advice? |
| 21:44.37 | ketin | dddh: Calling. |
| 21:45.19 | ketin | Request timed out... strange. |
| 21:45.27 | dddh | ketin, strange |
| 21:46.10 | *** join/#asterisk fofware (~chatzilla@190.224.76.151) |
| 21:56.42 | dddh | ketin, one more time please |
| 22:02.07 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 22:02.12 | *** join/#asterisk PhunTelTek (~root@cpe-76-188-232-92.neo.res.rr.com) |
| 22:07.05 | *** join/#asterisk catojo (~catojo@189.24.45.223) |
| 22:29.00 | dddh | are there any howtos on howto setup asterisk for home use? |
| 22:29.14 | [TK]D-Fender | ~book |
| 22:29.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:29.16 | [TK]D-Fender | ^^^^^^ |
| 22:29.45 | [TK]D-Fender | dddh: And any of the dozens of blog entries that are easily Google-able going into further specific and simple detail |
| 22:30.18 | dddh | [TK]D-Fender, ok, thank you, I'll try to find something |
| 22:37.18 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 22:37.56 | *** join/#asterisk catojo (~catojo@189.24.45.223) |
| 22:41.48 | p3nguin_ | dddh: AsteriskNOW might be the all-in-one solution for you. |
| 22:42.51 | dddh | p3nguin_, why do you think precompiled is better? |
| 22:43.28 | p3nguin_ | dddh: It's apparent that you are pretty new to the scene, so you can have your system up and running inside of 30 minutes with AsteriskNOW. |
| 22:44.17 | p3nguin_ | For home use, as you requested, AsteriskNOW would be pretty well-suited. |
| 22:44.44 | dddh | I see there is some "wizard-driven web interface" |
| 22:45.12 | p3nguin_ | I don't know about "wizard-driven," but it certainly does provide the FreePBX web interface. |
| 22:45.27 | hipitihop | tzafrir_laptop, you about ? |
| 22:45.36 | tzafrir_laptop | hipitihop, yes |
| 22:47.19 | hipitihop | thanks for your help yesterday .. do you have time to continue ? you asked me to "./debian/rules get-orig-source" however that complains of no such directory |
| 22:49.51 | hipitihop | tzafrir_laptop, we proved we could debuild yesterday but we now need to get the latest source |
| 22:50.20 | tzafrir_laptop | right. what about ./debian/rules get-orig-source |
| 22:50.25 | tzafrir_laptop | what does it do? |
| 22:50.49 | hipitihop | no such file or directory |
| 22:50.55 | dddh | p3nguin_, I will try to read documentation first to understand if asterisk is what I need and it will support my providers |
| 22:51.24 | p3nguin_ | dddh: If they offer SIP or IAX2, it supports your providers. |
| 22:52.09 | hipitihop | tzafrir_laptop, is there something else we need to install ? since ./debian/rules doesn't exist |
| 22:52.48 | tzafrir_laptop | in which directory are you? |
| 22:53.07 | dddh | p3nguin_, currently they offer SIP, but I need to understand how asterisk works to use it;) |
| 22:53.12 | tzafrir_laptop | there should be a subdirectory debian/ under the asterisk-1.* directory |
| 22:53.29 | p3nguin_ | dddh: Not really. AsteriskNOW lets you type in the information and press Apply. |
| 22:54.46 | hipitihop | tzafrir_laptop, yup used locate .. now get 404 when downloading http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.5~.tar.gz |
| 22:55.59 | dddh | p3nguin_, but my phone itself supports several providers, I need to know features it supports to use them |
| 22:57.56 | tzafrir_laptop | svn co svn://svn.debian.org/pkg-voip/asterisk/trunk asterisk-svn |
| 23:00.03 | dddh | s/it supports/asterisk supports |
| 23:02.16 | p3nguin_ | I don't even know what you're talking about. |
| 23:17.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:23.30 | ManxPower-work | dddh, What is your specific question? |
| 23:24.04 | ManxPower-work | dddh, Oh, I see you are using a GUI. Nevermind. I wish you the BEST of luck. |
| 23:25.18 | hardwire | if a queue member is Local, is there any way to figure out which queue was called? |
| 23:26.38 | hardwire | other than throwing that into the extension |
| 23:26.47 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:28.42 | hardwire | I was hoping app_queue would pass RDNIS |
| 23:46.36 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:55.04 | Amorsen | hardwire: Set a variable before entering the queue? |
| 23:55.20 | hardwire | it doesn't get passed to queue members |
| 23:55.26 | hardwire | unless I'm mistaken |
| 23:55.33 | Amorsen | It does if you underscore it them |
| 23:55.35 | Amorsen | then |
| 23:57.05 | hardwire | _IMAHO=yes? |
| 23:57.10 | Amorsen | Yes |
| 23:57.15 | hardwire | you brat. |
| 23:57.56 | Amorsen | If you want it to survive infinite levels you need two underscores |
| 23:58.04 | Amorsen | Otherwise it'll only survive once |
| 23:58.12 | afink | anyone around that could help me with registering to an ITSP for sip service? |
| 23:59.20 | afink | I keep getting Registration timed out, trying again |