| 00:14.11 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 00:15.20 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7bfd6b9fbd015550) |
| 00:23.55 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 00:28.18 | jsaunders | Anyone ever had an issue with their tdm2400 (or 400 for that matter) and having your fxo lines lose their ability to hear anything? Example. I call the ivr, Zap/13 picks up, I can hear the recording fine. dtmf does not work though because the channel cannot hear anything. ztmonitor 13 -vv shows tx changing variably which coincides with me being able to hear the recording. But rx side is totally dead. Keypresses or blowing in the mic, nothing. |
| 00:29.03 | jsaunders | If I restart the whole server (not just asterisk) it fixes it for a short period of time but it the problem eventually comes back. And this happens on all 8 of our fxo lines. |
| 00:30.57 | *** join/#asterisk anthm (n=anthm@mb10736d0.tmodns.net) |
| 00:30.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:33.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:35.23 | litage|w | jsaunders: can you try another TDM card? |
| 00:36.15 | jsaunders | Unfortunately no. :( Don't have one. The only odd thing I've found is chan 16 has the rx side at full bore, so there's major noise on the line. I'm guessing this must be the cause so I just pulled the line from the bix rail, restarted the box, and am now about to check the other lines for noise. |
| 00:37.26 | J4k3 | shit or get off the POTS. |
| 00:37.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:38.50 | tripps | maybe i'm doing something wrong, but i'm finding some discrepancies between the "book" and what my * box likes in the CLI. E.g., "core show functions" - what am I missing? |
| 00:38.52 | JT | jsaunders: how many POTS lines do you have? |
| 00:39.15 | Sweeper | oi, who's got a sip provider of decent price that will do LNPs? |
| 00:39.26 | Sweeper | low-volume |
| 00:40.00 | jsaunders | JT: 8 |
| 00:40.13 | jsaunders | I pulled chan 16 from the equation. Will monitor and see if issue comes back. |
| 00:40.30 | tripps | "no such command 'core'" is what I get if I try that command |
| 00:40.43 | TJNII | Bah. I don't have the tool I need to put the new intake manifold on my car. I guess I'll work on my * box. |
| 00:41.02 | J4k3 | TJNII: no torque wrench, aye? |
| 00:41.04 | JT | jsaunders: seems excessive |
| 00:41.19 | [hC] | anyone know of an issue sending alert info packets to polycom 2.2.x firmware? |
| 00:41.27 | TJNII | J4k3: Actually, a 9/16" socket with a build in universal joint. |
| 00:41.27 | J4k3 | ;) |
| 00:41.44 | JT | J4k3: not my fault your telcos are retarded ;) |
| 00:41.51 | J4k3 | TJNII: icky... time to chase the snap-on truck |
| 00:41.58 | TJNII | Yea.... |
| 00:42.09 | J4k3 | JT: eh, at least my ITSP isn't retarded ;) |
| 00:42.23 | J4k3 | my one POTS line left has been down since saturday.. |
| 00:42.24 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
| 00:42.40 | Sweeper | so nobody knows a decent provider that does LNPs in the US? :3 |
| 00:43.04 | J4k3 | Sweeper: vitelity ported in an SBC/ATT number in about 2 weeks |
| 00:43.06 | J4k3 | for me |
| 00:43.11 | J4k3 | but they charge more than they used to for porting, which sucks |
| 00:43.56 | *** join/#asterisk |Vulture| (n=Vulture@136.246.189.72.cfl.res.rr.com) |
| 00:44.32 | |Vulture| | Is ${CDR(duration)} accessible via the dialplan? Or is there any way to track a call duration via the dialplan? |
| 00:45.14 | jsaunders | Tnx fer the banter gentleman. Always appreciated. Later. |
| 00:45.22 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 00:45.37 | |Vulture| | wow there is a name I haven't seen in forever.. sup Juggie |
| 00:46.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:48.49 | TJNII | hmmmm.... Anyone know of a good FXO adapter (To connect a POTS line from the telco, in case I got my terminology wrong) that doesn't cost several hundred dollars? |
| 00:49.19 | litage|w | TJNII: get a Linksys ATA, or a Digium TDM card |
| 00:49.49 | TJNII | Digium TDMs are pricy, though. I don't have that kind of scratch. |
| 00:52.06 | J4k3 | POTS costs entirely too much in every way imaginable. |
| 00:52.18 | |Vulture| | PRI is where its at |
| 00:52.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:52.34 | J4k3 | PRI is where its at if you're terminating a few hundred extensions |
| 00:52.34 | |Vulture| | cept for the taxes they hurt |
| 00:52.46 | J4k3 | theres this huge in-between market thats totally missed except by the itsp. |
| 00:52.59 | J4k3 | but the i in itsp makes them suck mildly at best. |
| 00:53.32 | |Vulture| | yea but luckily there are a few semi reliable |
| 00:53.54 | |Vulture| | I am terminating a few toll frees from an itsp right now, until I can get them assigned by our PRI provider |
| 00:54.11 | J4k3 | yep |
| 00:54.19 | |Vulture| | its truly a case of you get what you pay for though |
| 00:54.37 | J4k3 | yep, and your internet connection needs to not suck in order to make it halfway reliably |
| 00:55.21 | J4k3 | personally vitel used to be about 400 landline-network-miles from me, and a handful of hops |
| 00:55.37 | J4k3 | now they're about 3500 network miles away, and 12-14 hops |
| 00:56.49 | |Vulture| | thats quite a change |
| 00:57.11 | |Vulture| | too bad they don't provide multiple proxies |
| 00:57.48 | J4k3 | yeah, all their addresses follow the same route |
| 01:00.42 | |Vulture| | still trying to find out how to access the call duration via the dialplan but it doesn't look like it is possible :( |
| 01:06.34 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:06.35 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:07.03 | [hC] | anyone using polycom sip firmware 2.2.0 notice anything wrong with sending "Ring Answer" in alert info? |
| 01:07.26 | blitzrage | Poll to the channel: When you think of the absolute must have features in a PBX (SoHo type environment), what are they? i.e. what set of features can you not do without? |
| 01:07.29 | peanut- | is there a way to forward in incomming call out of the system and have it preserve the CPN easily? |
| 01:07.45 | blitzrage | CPN? |
| 01:07.51 | peanut- | calling party number |
| 01:07.56 | blitzrage | CID? |
| 01:07.56 | peanut- | callerid(ani) |
| 01:07.59 | blitzrage | aha |
| 01:08.04 | blitzrage | PRI? |
| 01:08.12 | |Vulture| | CID yes, PRI no |
| 01:08.47 | [hC] | blitzrage: QoS in some form or another.. and a link that is stable enough (jitter less than 30ms) to carry voice are my top two |
| 01:08.48 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:09.13 | peanut- | when an external caller calls in now, and I do #NXXNXXXXXX to an external line, it forwards the caller as my CID, I want it to preserve the caller's |
| 01:09.30 | blitzrage | Use 'o' in Dial() |
| 01:10.33 | |Vulture| | [hC]: wow we are still using 1.6.5... unless you are talking bootrom I can check that |
| 01:11.06 | peanut- | ah, sweet. |
| 01:11.17 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 01:11.30 | |Vulture| | Is ${CDR(duration)} accessible via the dialplan? Or is there any way to track a call duration via the dialplan? |
| 01:12.23 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 01:12.24 | |Vulture| | trying to read the call duration to have it insert into a mysql db, seperate from our CDR database |
| 01:12.45 | [hC] | |Vulture|: nope... 2.2.0 sip firmware:) its pretty new.. but 1.6.5 is also pretty damn old. |
| 01:12.53 | grimsy | Does anyone know of a conference phone that does POE? |
| 01:13.06 | |Vulture| | [hC]: kinda going on the if its not broke approach |
| 01:14.35 | |Vulture| | any new features or is it all for the IP-501 series? |
| 01:14.39 | [hC] | |Vulture|: good approach :) |
| 01:15.00 | [hC] | theres tons of new features along the way... however, if you arent using them, and you dont know of them, you dont need them, so dont bother :) |
| 01:15.02 | |Vulture| | 1.6.5 was the first firmware for 501 |
| 01:15.08 | |Vulture| | I believe |
| 01:15.22 | [hC] | there are a bunch of microbrowser enhancements, blf, crashes, etc.. |
| 01:15.27 | [hC] | id you dont have any issues with how it is now, id stay there. |
| 01:15.54 | |Vulture| | well we do get presence issues but I don't think they actually cause a problem.. only a notice |
| 01:16.10 | *** part/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:16.30 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:18.20 | |Vulture| | trying to find a logger error on it |
| 01:18.32 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 01:18.36 | [hC] | |Vulture|: with the 500 server error? |
| 01:18.36 | |Vulture| | I do know before 1.6.5 we had some major issues with random reboots |
| 01:18.39 | |Vulture| | yea |
| 01:18.41 | |Vulture| | you got it |
| 01:19.05 | [hC] | |Vulture|: change these kind of occurances voIpProt.server.1.transport="DNSnaptr" |
| 01:19.15 | [hC] | change DNSnaptr to TCPpreferred |
| 01:19.19 | [hC] | thats the 'fix' |
| 01:19.24 | [hC] | but, it is just a warning |
| 01:19.35 | |Vulture| | [hC]: doing it now |
| 01:20.08 | |Vulture| | yup mine was "" |
| 01:21.07 | Juggie | hc |
| 01:21.08 | Juggie | wanna trace to my ip |
| 01:21.18 | Juggie | tell me what you see |
| 01:21.18 | Juggie | do a mtr |
| 01:22.09 | [hC] | sure. |
| 01:22.33 | [hC] | what do you want to know about it? |
| 01:22.47 | [hC] | im getting 57% loss to you |
| 01:22.48 | |Vulture| | Juggie: I am seeing loss from Jacksonville, FL |
| 01:23.06 | Juggie | i'm seeing like 50% packet loss on my 3rd hop out |
| 01:23.14 | [hC] | 17. gi-4-0-0.gw03.flfrd.phub.net.cable.rogers. 50.0% 43 85.2 83.5 72.0 90.5 4.5 |
| 01:23.15 | |Vulture| | looks like rogers |
| 01:23.21 | [hC] | its that hop for me |
| 01:23.43 | [hC] | now there's 100% loss to you at the previos hop |
| 01:23.46 | [hC] | previous* |
| 01:23.52 | sevard | Erbert and Gerbert subs are _amazing_ |
| 01:23.58 | [hC] | something in the last couple hops to you is screwed for sure jug |
| 01:24.34 | *** join/#asterisk Onyxyte (i=Onyxyte@r75-110-104-20.rmntcmtc01.rcmtnc.ab.dh.suddenlink.net) |
| 01:24.44 | Juggie | hc, what hop is generating all the loss |
| 01:24.45 | Juggie | and consequentially you see me also w/ los @ hop 18 right? |
| 01:24.45 | Juggie | *loss. |
| 01:24.45 | Juggie | assuming my router responds to icmp and it should |
| 01:25.52 | Juggie | ? |
| 01:26.03 | Juggie | is it on my hop (the last one) or the hop before me? |
| 01:26.48 | [hC] | theres routing changes occurring right now |
| 01:26.56 | [hC] | they seem to be trying to re route around the problem |
| 01:27.15 | [hC] | I keep seeing path changes every few seconds |
| 01:27.58 | [hC] | the router at "flfrd.phub.net.cable.rogers.com" is the one with the problem |
| 01:28.09 | [hC] | there are multiple ports/paths on that router that mtr keeps taking |
| 01:30.55 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 01:31.58 | Juggie | ya, still a problem in my area |
| 01:31.58 | Juggie | today it was totally down and now i'm losing about 50% packets |
| 01:32.12 | [hC] | damn kids with their small udp packets and their "voip" |
| 01:32.18 | [hC] | GET OFF MY LAWN!!! |
| 01:33.06 | sevard | hahhahaha |
| 01:33.28 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 01:33.50 | [hC] | ok, so if i let polycom discover what ntp server address to use (and ignore what i set in sip.cfg) from DHCP, what dhcp option does it poll? |
| 01:34.20 | fujin | I'm going to put money on the NTP SERVER DHCP OPTION |
| 01:34.50 | *** join/#asterisk s0lid (n=_freq@210.213.198.98) |
| 01:34.54 | fujin | that's option 42, IIRC. |
| 01:34.55 | [hC] | you bastard fujin :) |
| 01:35.02 | [hC] | Thats what i was looking for :) |
| 01:35.07 | [hC] | me and google arent speaking right now |
| 01:35.11 | [hC] | it let me down all day. |
| 01:35.17 | fujin | hehe |
| 01:35.22 | sevard | fujin: you just made him flip in his grave |
| 01:35.27 | fujin | Who? |
| 01:35.41 | sevard | anyone, ford |
| 01:36.02 | fujin | I'm sorry; You've lost me. |
| 01:36.21 | sevard | :), maybe you made an hhgttg reference w/out knowing it. |
| 01:36.42 | fujin | indeed |
| 01:37.01 | sevard | well, read hitch hikers guide to the galaxy |
| 01:37.14 | fujin | I'll work on it. |
| 01:37.37 | [hC] | I totally didnt get it until just now. |
| 01:37.43 | [hC] | How i didnt spot that earlier, i dont know |
| 01:41.13 | fujin | watches |
| 01:41.16 | fujin | shit I'm having a bad day. |
| 01:41.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:41.37 | [hC] | youve never heard of the reference to '42' being the answer to the live, universe, and everything? |
| 01:42.23 | [hC] | google even has a calculator for it. |
| 01:42.24 | [hC] | http://www.google.com/search?hl=en&c2coff=1&client=safari&rls=en&q=the+answer+to+life%2C+the+universe%2C+and+everything&btnG=Search |
| 01:42.25 | [hC] | :) |
| 01:47.19 | fujin | oh, yes indeed I have |
| 01:47.20 | fujin | I've seen the movie |
| 01:47.43 | fujin | yet wasn't aware that I made reference to it |
| 01:51.11 | [hC] | so.. im just getting my fingers into diigum's svn again.. is there a way to check out a 'trunk' version of 1.4? (or does that even exist) |
| 01:51.51 | [hC] | Im looking for the latest development changes in the 1.4 tree, but i dont want 1.6.. if i understand correctly trunk is just 'pre 1.6' - but current 1.4 releases are not built out of snapshots of trunk? |
| 01:52.21 | *** join/#asterisk e1mer (n=elmer@unaffiliated/e1mer) |
| 01:52.50 | fujin | http://svn.digium.com/view/asterisk/branches/1.4/ |
| 01:53.11 | [hC] | ah, so branches is the 'trunk' version of 1.4 then? |
| 01:53.29 | file | there is no trunk version of 1.4, trunk is a name presently given to the development tree |
| 01:53.38 | file | 1.4 is not a development tree, it receives only bug and security fixes |
| 01:53.51 | phix | Do I need an external program / module to create a conference call? |
| 01:53.56 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 01:53.59 | [hC] | oh i see. so once it becomes stable, no features go into it. |
| 01:54.16 | [hC] | That makes a lot more sense now. |
| 01:54.24 | file | that is the present way of things, it is in flux right now and will change |
| 01:54.32 | file | (for future versions) |
| 01:54.41 | [hC] | I guess i'll ask again soon? :) how do you think its going to change? |
| 01:54.52 | [hC] | by that i mean, what do you think it will change to.. |
| 01:54.54 | file | there was a document posted to the asterisk-dev mailing list detailing things |
| 01:55.00 | [hC] | ill check there. |
| 01:55.08 | [hC] | Thanks file |
| 01:58.30 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:58.49 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 01:59.32 | ZaVoid | anyone know a decent iax load balancer? |
| 02:04.07 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 02:05.22 | Ritzerisk | anyone familar with amanda |
| 02:05.49 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 02:06.33 | docelmo | hey Digium guys.. You might want to tell marketing tomorrow that they messed up the link on the Digium/Asterisk logo in the email they just sent for Digium Asterisk World |
| 02:06.35 | [hC] | so there appears to be an interesting issue with asterisk running as non-root and sending mail as the asterisk user on the system ,instead of the serveremail= address in voicemail.conf |
| 02:09.14 | docelmo | Wow.. Dead channel |
| 02:11.39 | Qwell | docelmo: eh? |
| 02:16.53 | [hC] | When running asterisk as non-root, is it asterisk's fault or the sendmail command's fault for not taking what was specified in serveremail in voicemail.conf? As root I had it set to no-reply@domain.com, and now it goes out as asterisk@myhostname.domain.com, ignoring serveremail |
| 02:17.11 | Juggie | [hC], sendmail permissions perhaps? |
| 02:17.47 | [hC] | Juggie: well, it should just be text that goes out in the from:, i dont know what permissions would need to be changed... theoretically you should be able to pass anything there, regardless of who you are. |
| 02:17.48 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-db56dc77376b8900) |
| 02:19.46 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:21.57 | Juggie | [hC], sendmail may be configured to reduce spam |
| 02:22.36 | docelmo | Qwell PM me your email and I will send you the email I got and you can see whats wrong with it. |
| 02:22.41 | phix | sooo |
| 02:23.02 | phix | is an external program / module required? or can it be done using stock asterisk? |
| 02:23.17 | phix | in the dialplan |
| 02:23.20 | Qwell | docelmo: qwell@ |
| 02:23.35 | docelmo | what? |
| 02:23.37 | phix | Qwell: ? |
| 02:23.45 | fujin | phix: ? |
| 02:24.11 | fujin | You haven't said anything in here for a while, you're going to have to repeat your initial question |
| 02:24.12 | phix | fujin: conference calls, joining at least 3 calls together |
| 02:24.23 | fujin | Use MeetMe? |
| 02:24.33 | phix | That is an external program / module? |
| 02:24.50 | phix | It is not in the core of Asterisk ? |
| 02:24.53 | fujin | No, It's included. |
| 02:24.54 | [hC] | Juggie: postfix... id have to check... just dont know what to search for |
| 02:25.18 | phix | fujin: ok |
| 02:25.21 | fujin | [hC]: postfix will send mails from the logged-in-user, i.e.; if asterisk is running as the user 'asterisk' they'll appear to come from that. |
| 02:25.23 | fujin | this is configurable, though |
| 02:26.15 | phix | fujin: how would I learn how to do it? I have googled it but can only find examples that use ZAP channels, I want to join at least three SIP channels together |
| 02:26.44 | fujin | http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 02:26.59 | phix | nice |
| 02:27.02 | phix | Thank you |
| 02:27.07 | fujin | fuckinggoogleit.com |
| 02:27.30 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:28.08 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 02:31.03 | phix | ok so a meetme conference is a static number yuo call, enter in a pin to join it. Are there other ways to do this? like call a number and specified the number you want to add to the conference? |
| 02:31.52 | fujin | uhghr? |
| 02:31.54 | fujin | probablty |
| 02:31.57 | fujin | use dialplan |
| 02:35.07 | phix | hmmm |
| 02:35.26 | fujin | what are you trying to do |
| 02:35.31 | phix | ok I would like to use the dialplan, I just don't seem to have that knowledge already. |
| 02:35.33 | fujin | make a kind of invite conference? |
| 02:35.55 | phix | yeah, just an easy way for a user to add in another person to the call |
| 02:36.14 | fujin | I'm not sure that'd be easy, but definitely possible |
| 02:36.17 | phix | without needing to be a meetme admin or something like that |
| 02:36.38 | phix | like dial *45 then number of person to add, something like that |
| 02:36.45 | fujin | I've given you the tools you require. |
| 02:36.47 | ZaVoid | can i strip part of a number in the sip.conf entries and not ina dialpan? |
| 02:36.52 | phix | fujin: ok |
| 02:37.01 | phix | fujin: I will RTFM and google |
| 02:37.29 | phix | I feel motivated today |
| 02:37.29 | fujin | heh |
| 02:37.30 | fujin | http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 02:37.41 | fujin | instead of pissing about in irc, you could have scrolled down on the original link I gave you |
| 02:37.41 | phix | yay |
| 02:38.13 | fujin | although that looks messy |
| 02:40.54 | phix | yep |
| 02:40.56 | phix | Messy indeed |
| 02:44.11 | *** join/#asterisk J_5 (n=j@cpe-71-72-210-44.cinci.res.rr.com) |
| 02:47.11 | *** join/#asterisk Op3r (n=Op3r@125.212.63.243) |
| 02:55.27 | *** join/#asterisk blq (n=Bl@dslb-088-067-029-086.pools.arcor-ip.net) |
| 03:01.25 | *** join/#asterisk jmacz (n=jmacz@201.244.173.79) |
| 03:03.38 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:10.31 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:17.40 | *** join/#asterisk PepOSX (n=pepOSX@190.72.154.149) |
| 03:20.37 | *** join/#asterisk juanmanuel (n=jmacz@190.24.97.187) |
| 03:22.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 03:25.07 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 03:25.10 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 03:27.39 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ff2674fcbb20071f) |
| 03:30.13 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:34.31 | phix | Oct 23 13:14:19 ERROR[1104]: chan_sip.c:11078 handle_request_subscribe: Got SUBSCRIBE for extension 105@default from 10.0.0.33, but there is no hint for that extension |
| 03:50.24 | *** join/#asterisk asdx (n=kde-deve@adsl-145-217.click.com.py) |
| 03:52.09 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 03:52.15 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 03:59.26 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128663708.dsl.bell.ca) |
| 04:11.02 | [hC] | fujin: any idea how to allow that to be overridden in postfix? |
| 04:11.31 | fujin | not off the top of my head sorry |
| 04:11.39 | [hC] | no worries, i'll make friends with google |
| 04:19.14 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 04:20.10 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 04:25.41 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com) |
| 04:27.41 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com) |
| 04:29.31 | phix | hmmm |
| 04:29.34 | phix | any ideas? |
| 04:29.40 | phix | It is just annoying seeing that message |
| 04:40.24 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 04:41.36 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) |
| 05:04.48 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 05:14.12 | *** join/#asterisk saftsack (n=saftsack@pD9E041E0.dip.t-dialin.net) |
| 05:15.51 | [TK]D-Fender | phix, you have a phone thats trying to look for the status of ext 105 and you don't have any HINTS set up in your dialplan for presence support |
| 05:19.18 | *** join/#asterisk neax (n=newdle@203-114-176-86.dsl.sta.inspire.net.nz) |
| 05:22.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:25.16 | *** join/#asterisk s0lid (n=_freq@210.213.198.98) |
| 05:32.33 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 05:35.32 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 05:35.37 | ussrback | hi alll |
| 05:36.25 | ussrback | is there mysql-vm for * 1.4 ? |
| 05:37.21 | Raky-2 | mysql voicemail? |
| 05:37.27 | ussrback | yes |
| 05:37.34 | ussrback | for 1.4 version |
| 05:37.40 | Raky-2 | 1.4,.* or just 1.4 flat. |
| 05:37.46 | Raky-2 | i believe there is buddy, i've got it working. |
| 05:37.53 | ussrback | i downloaded addons |
| 05:37.56 | Raky-2 | give me a second, let me show you the link. |
| 05:38.10 | ussrback | but i cant find there |
| 05:38.15 | ussrback | ok |
| 05:38.25 | Raky-2 | http://www.voip-info.org/tiki-index.php?page=Asterisk+voicemail+database |
| 05:38.34 | Raky-2 | the addon you require is the req_mysql.so |
| 05:38.38 | kiscokid | what's the advantage of putting vm on mysql? |
| 05:38.38 | Raky-2 | i think that's what it is. |
| 05:38.56 | Raky-2 | well, it allows for ease of use with web applications. |
| 05:39.03 | Raky-2 | so as you can imagine, you'd be able to add users through PHP. |
| 05:39.13 | ussrback | asterisk-addons/mysql-vm-routines.h |
| 05:39.26 | ussrback | where can i find it |
| 05:39.52 | ussrback | it says that i have to edit makefile USE_MYSQL_VM_INTERFACE=1 |
| 05:39.54 | *** join/#asterisk epaulin (n=epaulin@219.136.238.247) |
| 05:39.55 | Raky-2 | sorry! |
| 05:39.58 | ussrback | and then make install |
| 05:40.01 | Raky-2 | i gave you the incorrect lik |
| 05:40.03 | Raky-2 | link |
| 05:40.06 | ussrback | sooooo... |
| 05:40.13 | kiscokid | raky: thanks |
| 05:40.42 | Raky-2 | http://www.voip-info.org/wiki/view/Asterisk+sip+mysql+peers |
| 05:41.01 | Raky-2 | you want to use this, and apply the same kind of theory for voicemail |
| 05:42.31 | ussrback | ok but what shoud i put in voicemailconf file? |
| 05:44.28 | Raky-2 | http://www.voip-info.org/tiki-index.php?page=Asterisk+voicemail+database |
| 05:44.30 | Raky-2 | look at section 3 |
| 05:44.35 | Raky-2 | that's what you put in the voicemail section |
| 05:47.15 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 05:47.28 | *** join/#asterisk bantu (n=Miranda@rz-du-mbx-136-213.rz.uni-karlsruhe.de) |
| 05:47.49 | ussrback | what are the mandatory columns for voicemail database table? |
| 05:51.36 | epaulin | I got my digium card Wed at last week, then I applied hpec-licensing from Digium website, now has been a week, still got no response from Digium, what should I do, is here anyone from digium can help me? |
| 05:52.35 | Sweeper | epaulin: call them on the phone |
| 05:52.46 | epaulin | I also wrote a mail to Digium customer service, no response too, I just don't know how hard it could be. |
| 05:53.01 | Sweeper | well, it IS 2 am in digiumland |
| 05:53.09 | epaulin | Sweeper: I don't know how to call a IAXTel, and I;m not in us. |
| 05:54.00 | Sweeper | epaulin: it's pretty easy..get an iax softphone, and dial :v |
| 05:54.23 | epaulin | Sweeper: OK, tnx, I'll do that. |
| 05:55.41 | ussrback | how can i get callerid variable and pass it ro perl AGI ? |
| 05:56.00 | Sweeper | ussrback: see the FastAGI docs |
| 05:56.04 | Sweeper | you can pass it in the url |
| 05:56.20 | ussrback | why fastagi ? |
| 05:56.26 | ussrback | and not AGI |
| 05:56.37 | Sweeper | oh, you're using plain agi? :v |
| 05:56.47 | Sweeper | well, plain agi is ok if performance isn't an issue |
| 05:56.59 | Sweeper | I dunno how you'd do it, but I suspect it is easy |
| 05:57.15 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:57.23 | ussrback | u mean that fast agi is better that AGI ? |
| 05:57.56 | Sweeper | depends on the application |
| 05:58.06 | Sweeper | fastagi needs its own server instance |
| 05:58.13 | Sweeper | and will thus take a bit more to configure |
| 05:58.30 | Sweeper | regular agi is slow, and spawns an interpreter for every call |
| 05:58.31 | ussrback | yes sure.... i kno fastagi is good for sounds and such applications |
| 05:58.48 | ussrback | i use agi for database interaction |
| 05:58.53 | Sweeper | hmm |
| 05:59.14 | Sweeper | I really recommend fastagi |
| 05:59.28 | *** join/#asterisk saftsack (n=saftsack@s0776.vpn.hrz.tu-darmstadt.de) |
| 06:00.12 | Sweeper | personally, I use adhearsion to serve my fastagi stuff, but I'm sure there are perl scripts that will do it |
| 06:00.28 | ussrback | ok. thanks for advice. i dont think that i shoud change my perl scripts if i jump to the fastAGI |
| 06:00.48 | Sweeper | it probably won't change much |
| 06:01.20 | Sweeper | they'll just be run in a permanently-loaded interpreter, so that will cut down on processing time |
| 06:01.49 | *** join/#asterisk ltd (n=z@202.55.146.65) |
| 06:02.53 | ussrback | thats good, but what about load, should it decrease load on * server |
| 06:03.31 | Sweeper | yea |
| 06:03.52 | Sweeper | with fastagi, you could even move the perl scripts to a different server |
| 06:04.01 | Sweeper | so it scales much more nicely |
| 06:04.35 | Sweeper | and even if you don't, you only end up with a single instance of the perl interpreter running at any one time, so it cuts down on memory usage |
| 06:04.56 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:05.37 | ussrback | ok greattt. is there more variables and commands for fastagi |
| 06:05.45 | ussrback | or they are the same as in AGI |
| 06:05.46 | ussrback | ? |
| 06:06.25 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 06:06.26 | Sweeper | same |
| 06:07.17 | ussrback | good |
| 06:07.20 | ussrback | thanks |
| 06:07.31 | ussrback | ill try my scripts with fastagi |
| 06:07.54 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:11.22 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 06:12.14 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 06:12.48 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:16.49 | *** join/#asterisk i3inary (i=i3inary@ip72-207-113-253.sd.sd.cox.net) |
| 06:35.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:37.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:50.52 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:54.46 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 06:55.34 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:56.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:00.21 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:05.31 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 07:18.44 | *** join/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 07:19.59 | *** join/#asterisk harpal (n=Harpal@124.125.254.227) |
| 07:24.18 | harpal | I am new to VOIP. from Where I start? |
| 07:26.16 | jql | ~book |
| 07:26.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 07:35.06 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 07:37.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:39.57 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ff2674fcbb20071f) |
| 07:39.59 | LukinoVoip | Hi all, i have troubles in connecting AST with a Philips PBX, i see in CLI duplicates Q931 messages...i would understand if is it possible that it depends of overlap dialing...Any ideas? |
| 07:46.43 | *** join/#asterisk Op3r (n=Op3r@125.212.63.243) |
| 07:46.56 | *** join/#asterisk ltd (i=foobar@202.55.149.246) |
| 07:48.59 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6b5d538d7afc56cc) |
| 07:52.39 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 08:00.31 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 08:01.46 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:03.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:12.16 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 08:16.45 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 08:16.59 | *** join/#asterisk el_ektro (n=golf_gti@hyperspule.fs.ei.tum.de) |
| 08:17.27 | el_ektro | hello, got a question about asterisk/asterisknow: |
| 08:18.09 | el_ektro | Is it possible to do some kind of CLIP routing, as in "I have a list of callers, who are forwarded to +491234567 when they call" |
| 08:18.11 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 08:18.27 | el_ektro | "all other callers and callers without CLIP are forwarded to ext 85" |
| 08:19.39 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 08:20.34 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:23.23 | Sweeper | el_ektro: yes |
| 08:23.57 | el_ektro | Sweeper: got a tip how to do that? |
| 08:24.51 | Sweeper | el_ektro: if your list is static, you can do it with plain old dialplan |
| 08:25.07 | *** part/#asterisk Raky-2 (n=John@220.157.75.246) |
| 08:25.12 | Sweeper | if you want to update the list fairly freuquently, use adhearsion+rails or some other AGI method |
| 08:25.13 | *** join/#asterisk flenders_ (n=fserto@unaffiliated/flenders) |
| 08:25.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 08:35.50 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:38.29 | el_ektro | ah ok, thanks... gonna check that... |
| 08:40.19 | el_ektro | and another thing, I didn't find too much information about using an AVM Fritz!Card USB, is that similar to using a AVM Fritz!Card PCI or is it totally different? |
| 08:47.59 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:48.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:55.36 | *** part/#asterisk el_ektro (n=golf_gti@hyperspule.fs.ei.tum.de) |
| 09:03.02 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 09:08.41 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 09:11.51 | *** part/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:2) |
| 09:14.05 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 09:15.53 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 09:17.59 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:20.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 09:21.53 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:22.24 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 09:26.55 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:30.02 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:30.59 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 09:31.02 | casix | hello |
| 09:34.08 | casix | I'm having a lots of <<chan_sip.c: = No match Their Call ID: 37a4fa0a6bbce490523854d33ecb3d44@212.36.71.106 Their Tag as196ef169 Our tag: as1bc08c19>> errors. Anybody knows what this error mean?? It is possible that asterisk is hanging up calls?? |
| 09:35.50 | jql | are those followed by a "Found Call ID" or whatnot? |
| 09:37.13 | *** join/#asterisk bantu (n=Miranda@rz-du-ubx-140-93.rz.uni-karlsruhe.de) |
| 09:37.26 | casix | no |
| 09:37.46 | jql | then you may indeed have a problem |
| 09:38.39 | jql | while that message alone is normal for a high debug/verbose level, receiving messages which don't "match" a known Call ID is often a problem |
| 09:38.46 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 09:39.04 | jql | let me rephrase: don't match *any* known Call ID |
| 09:39.45 | casix | which can of problems?? |
| 09:39.56 | casix | establishing a calls? |
| 09:40.16 | casix | with established calls?? is possible that asterisk hang up a call? |
| 09:40.43 | jql | sip messages are sent for every step in a call, from setup to teardown |
| 09:40.54 | jql | any point could fail |
| 09:41.12 | jql | so, that's a qualified "yes, I suppose" |
| 09:42.29 | casix | and do you know how can I see whats wrong?? have I to debug the sip messages? |
| 09:52.30 | casix | sometimes yes that I have a Found Their Call ID |
| 09:55.36 | *** join/#asterisk saftsack (n=saftsack@s0640.vpn.hrz.tu-darmstadt.de) |
| 10:04.31 | *** join/#asterisk Phuntom (n=Phuntom@80.233.159.254) |
| 10:04.35 | Phuntom | hi ya! |
| 10:05.35 | Phuntom | my mISDN module not loaded. it is a big trouble? |
| 10:11.11 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 10:13.36 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 10:14.53 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 10:19.09 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 10:27.46 | *** join/#asterisk LT (n=lt@unaffiliated/lt) |
| 10:29.33 | *** join/#asterisk Bobocop (n=whooopa@ip-89-174-40-121.multimo.gtsenergis.pl) |
| 10:32.21 | Bobocop | Hi all |
| 10:32.39 | Bobocop | Do you know, how to increase wait time for entering number to dial from analog phones? Is it zaptel's setting? |
| 10:33.36 | kaldemar | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout |
| 10:40.03 | *** join/#asterisk el_ektro (n=golf_gti@hyperspule.fs.ei.tum.de) |
| 10:40.06 | el_ektro | yo |
| 10:40.24 | Phuntom | oy |
| 10:40.25 | Bobocop | kaldemar THX! you're great - I couldn't find it anywhere :) Now I need to figure out how to modify trixbox scripts to change it.... Any clues? |
| 10:41.05 | el_ektro | anyone in here with a clue how to get an AVM fritz!card usb v2.1 to work on asterisk? the big great google didn't help me that much, and rtfm neither... |
| 10:42.21 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 10:44.50 | el_ektro | i am running trixbox CE 2.2.4 (asterisk 1.2.23 on CentOS 4) |
| 10:49.10 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-6b5d538d7afc56cc) |
| 10:51.23 | kaldemar | Bobocop: yes, go and ask in #trixbox. i'm pretty sure no one will help you here. |
| 10:52.20 | Phuntom | I got: set_address_from_contact: '' is not a valid sip contact in my log |
| 10:52.27 | Phuntom | what does it mean? |
| 10:52.44 | kaldemar | el_ektro: to you too, trixbox is not supported here. you'll probably have better luck in #trixbox. |
| 10:52.58 | Phuntom | ( missing SIP ). Trying to use anyway... |
| 10:55.04 | el_ektro | kaldemar: mh ok, then I'll ask more generically: anyone have a clue how to get the avm fritz!card usb driver compiled on linux? |
| 10:55.28 | roxlu | hi |
| 10:55.34 | roxlu | someone here who uses budgetphone? |
| 10:56.39 | kaldemar | Phuntom: what version of asterisk are you using? what were you doing when you got that? where did you get it? what is your setup like? |
| 10:57.34 | Phuntom | asterisk=1.4.9-0.1-1 |
| 10:57.45 | Phuntom | i use asterisknow |
| 10:57.56 | kaldemar | Phuntom: looking at chan_sip.c in 1.4.13, that line lets you know that a sip URI is missing the sip: part in the beginning. e.g. sip:123@domain vs 123@domain. |
| 10:58.50 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:59.22 | *** join/#asterisk dez71 (i=dez@216.83.0.172) |
| 10:59.42 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 10:59.50 | kaldemar | Phuntom: what were you doing when you got the message? it looks like you're trying to dial with SIP, but with empty contact info. |
| 11:00.02 | Phuntom | i did nothing |
| 11:00.18 | Phuntom | it looks like, i can see |
| 11:00.26 | kaldemar | where did you dig up the message then? |
| 11:00.46 | Phuntom | in console and in log file |
| 11:02.01 | kaldemar | there must have been some activity. |
| 11:02.28 | Phuntom | ya, i have my sip app launched |
| 11:02.40 | Phuntom | 2 sip apps |
| 11:02.59 | Phuntom | 1 for calling, 2 - for receiving calls |
| 11:04.18 | Bobocop | kaldemar: thx again for help, I have to modify it manually, at #trixbox nobody knows how to change it :) |
| 11:04.37 | Bobocop | bye |
| 11:05.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:06.01 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 11:07.30 | Phuntom | how can i log cli messages (while debug is on and verbose set to 5) ? |
| 11:08.21 | Phuntom | i`m trying to find out which packet make this shit |
| 11:08.32 | Phuntom | makes |
| 11:10.54 | phix | hmmm, ok so I dont have any HINTS setup in my dial plan, ok, I guess I will read up on that then |
| 11:11.21 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 11:11.29 | phix | ok another issue, I just signed up with a VoIP service, I can make outgoing calls but I cannot ring my asterisk box using the number allocated to me. |
| 11:12.03 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:13.06 | *** join/#asterisk myiagy (n=myiagy@201.56.109.2) |
| 11:18.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:18.58 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 11:19.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:20.30 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 11:24.04 | *** join/#asterisk Arc^^ (n=Arc_@a82-95-179-89.adsl.xs4all.nl) |
| 11:24.36 | Arc^^ | Hi, anyone know how to get incoming calls on ISDN to work? I'm getting: |
| 11:24.36 | Arc^^ | P[ 1] GOT IGNORE SETUP P[ 1] CC_RELEASE_COMPLETE|CONFIRM [TE] |
| 11:24.42 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:24.44 | Arc^^ | and then a busy tone/error tone on the dialing side |
| 11:25.08 | Arc^^ | I'm using MISDN 1.2, Asterisk 1.2 on a digium B410P |
| 11:25.27 | Arc^^ | outgoing calls work fine |
| 11:28.23 | phix | hmmmm |
| 11:31.42 | phix | SIP/2.0 407 Proxy Authentication Required |
| 11:32.01 | phix | I have a register => line though :/ |
| 11:32.05 | phix | what else am I missing? |
| 11:33.57 | phix | Do I need to specify a realm for the sip proxy I am trying to register as? |
| 11:36.51 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 11:39.23 | *** join/#asterisk Fl1p (n=fl1p@port-83-236-208-171.static.qsc.de) |
| 11:41.10 | Fl1p | hi all, is there any asterisk addon to let the sip clients change their extensions behavior ? Something like a Web Interface with Options DND, when away redirect to no. xxx ? |
| 11:42.04 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:50.25 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:55.24 | *** join/#asterisk saftsack (n=saftsack@s0662.vpn.hrz.tu-darmstadt.de) |
| 11:55.39 | Arc^^ | Ok simpler question: I'm getting an EXTCANTMATCH from misdn |
| 11:55.57 | Arc^^ | However it doesnt matter what direct DID i fill in for my extension i keep getting extcantmatch |
| 11:56.08 | Arc^^ | I tried all the numbers i'm getting on the dad: list |
| 12:05.00 | *** join/#asterisk toscaba (n=no@hs-bihamk.europronet.ba) |
| 12:06.58 | toscaba | need help on making updatecdr property in agent.conf start working |
| 12:07.02 | toscaba | any help appreciated |
| 12:07.37 | toscaba | thanks in advance |
| 12:10.34 | blitzrage | phix: register does not mean the call will authenticate -- it simply is a method used to tell the far end where you exist on the network |
| 12:11.00 | blitzrage | phix: you still need to setup the account and authorization in sip.conf -- the O'Reilly book tells you all this |
| 12:11.07 | blitzrage | FYI |
| 12:12.28 | Arc^^ | how can i debug the list asterisk is trying to match an incoming call to? |
| 12:14.10 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 12:14.56 | Arc^^ | i'm starting to think its better to config asterisk using the config files |
| 12:15.28 | phix | blitzrage: I have a context for my sip provider |
| 12:15.42 | blitzrage | phix: ok... |
| 12:15.54 | phix | blitzrage: it specifies the realm, fromdomain, fromuser, auth=md5, etc.. |
| 12:17.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:17.45 | blitzrage | 407 Proxy Auth. is normal after the INVITE comes in. That's because Asterisk sends some information the other end needs for authentication purposes (a nonce) -- then the other end should send another INVITE with the authentication information, and either Asterisk will accept the call, or possibly reject it again if you don't have your username/password right -- or -- your Asterisk will send a 404 Not Found if the requeste |
| 12:17.45 | blitzrage | d extension does not exist in the context defined for that peer |
| 12:18.01 | phix | blitzrage: [myVoipContext] type=friend host=myVoipProvider.com realm=myVoipProvider.com fromdomain=myVoipProvider.com username=myUsername secret=myPassword auth=md5 reinvite=yes canreinvite=yes qualify=no nat=yes |
| 12:18.08 | phix | blitzrage: ok |
| 12:18.14 | *** join/#asterisk el_ektro (n=golf_gti@hyperspule.fs.ei.tum.de) |
| 12:18.29 | blitzrage | phix: the 'sip debug' would be handier (and it should go into a pastebin -- not into this channel) |
| 12:18.31 | phix | blitzrage: I will tell you what the sip debug message tell me |
| 12:18.38 | phix | blitzrage: yep :) I wasdoing that |
| 12:18.38 | blitzrage | ~pb |
| 12:18.39 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:18.45 | phix | blitzrage: ok |
| 12:21.41 | Arc^^ | any way to debug asterisk extension matching? |
| 12:23.04 | agx | Arc^^, "show dialplan" perhaps? |
| 12:25.35 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:29.58 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 12:29.58 | phix | blitzrage: http://rafb.net/p/cPJkoH99.html |
| 12:29.58 | phix | blitzrage: I hope you help me :) |
| 12:30.42 | blitzrage | if that's all you got -- it looks like the other end didn't get the 407 back |
| 12:31.28 | phix | hmmmm, I am not blocking outgoing connections |
| 12:31.49 | blitzrage | not sure -- but the other end is not replying if thats all you got |
| 12:32.16 | blitzrage | is your asterisk box behind NAT? |
| 12:32.21 | phix | ok it is working now, I need to add in insecure=very |
| 12:32.34 | blitzrage | that should be insecure=invite |
| 12:32.51 | blitzrage | insecure=very is old syntax, meaning "invite,port" |
| 12:32.59 | phix | what does that mean? I don't like the word insecure |
| 12:33.03 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 12:33.53 | phix | no my asterisk box has direct Internet access |
| 12:33.59 | *** join/#asterisk michael-i (n=michael-@Wb6bc.w.pppool.de) |
| 12:35.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:35.55 | blitzrage | phix: check the docs -- it'll explain what that option means |
| 12:36.55 | mvanbaak | heya all |
| 12:37.09 | mvanbaak | does anyone know if the polycom phones have extension keypads ? |
| 12:37.32 | mvanbaak | ~phones |
| 12:37.33 | jbot | extra, extra, read all about it, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream ... |
| 12:37.57 | phix | blitzrage: is it good to have that option set? |
| 12:38.00 | phix | blitzrage: ok |
| 12:38.36 | Arc^^ | hm is freepbx adding stuff that doesn't really exist in asterisk? such as 'routes' |
| 12:38.43 | phix | blitzrage: is it better to log CDR to a RDBMS or a csv file? |
| 12:38.51 | blitzrage | phix: when you know what the option means, then you'll be able to come to your own conclusion as to whether that is good or bad :) |
| 12:39.05 | blitzrage | Arc^^: not sure -- this isn't #freepbx |
| 12:39.19 | blitzrage | phix: define: "better" |
| 12:39.26 | phix | blitzrage: hehe ok ok I get the hint. |
| 12:40.06 | phix | blitzrage: "better" as in if I do something else instead of having insecure=invite I won't get raped |
| 12:40.25 | blitzrage | phix: the name is misleading -- it's not really that insecure |
| 12:40.33 | phix | ok :) |
| 12:40.46 | blitzrage | you mean "better" as in the CDRs are off the box? then I guess so |
| 12:41.18 | phix | blitzrage: oh, CDR question right, better as in more efficient? |
| 12:41.19 | blitzrage | many people think that putting something into a DB is "better" just because it's in the DB and don't really understand why they want it in a DB (that's why I was asking you to define "better") |
| 12:41.44 | phix | blitzrage: Efficient to store and to search I mean |
| 12:41.46 | [TK]D-Fender | mvanbaak: Extensions keypads? |
| 12:42.06 | blitzrage | phix: it'll have overhead because it's not writing to the filesystem, but yes, it can make it easier to search, etc... |
| 12:42.23 | phix | blitzrage: in other words, would I be better off using perl and regex or SQL statements and indexes in my RDMBS? |
| 12:42.28 | phix | RDBMS even |
| 12:42.38 | blitzrage | I don't know -- it depends what you're doing |
| 12:42.44 | blitzrage | both answers are equally correct |
| 12:42.51 | mvanbaak | [TK]D-Fender: like for receptionists |
| 12:43.16 | [TK]D-Fender | mvanbaak: You mean speedial/presence panel? |
| 12:43.21 | mvanbaak | yeah |
| 12:43.23 | [TK]D-Fender | mvanbaak: If so then yes |
| 12:43.34 | [TK]D-Fender | mvanbaak: for the IP 601/650 |
| 12:43.36 | phix | Well I guess I would like to create a statistics page with charts and stuff |
| 12:43.58 | *** join/#asterisk nexilus (n=nexilus@gate.compodium.se) |
| 12:44.18 | nexilus | how would i manually reset all channels of Zap ? |
| 12:44.33 | phix | Either using perl, PHP or java servlet / jsp / some type of webapp framework |
| 12:44.42 | phix | nexilus: zap reload ? |
| 12:44.50 | nexilus | cause i have 31 channels, and 27 of them are in PRI-state "resetting"... |
| 12:44.53 | blitzrage | phix: depending how you would like to implement it, then both could be correct -- but it almost sounds like that application would work better with SQL |
| 12:45.05 | *** join/#asterisk mildk (n=mil@duke.code3.dk) |
| 12:45.09 | phix | blitzrage: yeah |
| 12:45.14 | phix | aawww |
| 12:45.23 | phix | but you wer every helpful! |
| 12:45.25 | nexilus | phix: is no command named zap reload |
| 12:45.41 | phix | asterisk -rx "zap reload" |
| 12:45.48 | phix | sudo asterisk -rx "zap reload" |
| 12:45.50 | mildk | does anyone know how to hide callerid in a pri line? hidecallerid=yes in zapata.conf does not do the trick |
| 12:46.08 | nexilus | phix: im in the asterisk cli, it says "no such command" |
| 12:46.12 | phix | oh |
| 12:46.16 | phix | ummmm help zap? :) |
| 12:46.29 | phix | *shrugs* I am knew :) |
| 12:46.33 | phix | knew = new |
| 12:47.09 | [TK]D-Fender | nexilus: "module reload chan_zap.so" |
| 12:47.16 | nexilus | ive tried doing a full reload of zaptel, aswell as restarting asterisk, but the channels dont stop being in mode "resetting" unless i reboot so far |
| 12:47.21 | nexilus | ill try that |
| 12:48.19 | phix | ok, [TK]D-Fender knows all :) |
| 12:48.21 | nexilus | [TK]D-Fender: where exactly should i do that? tried in asterisk cli and in the shell, none had that command :s |
| 12:48.31 | [TK]D-Fender | nexilus: what ver of *? |
| 12:48.35 | phix | [TK]D-Fender: hey, how do I get rid of that HINT message? |
| 12:48.49 | nexilus | Asterisk 1.2.21.1 |
| 12:48.56 | [TK]D-Fender | phix: Alrady told you, its your PHONE thats requesting the usage info from *. Stop your PHONE. |
| 12:49.02 | nexilus | should i get a newer one? |
| 12:49.09 | [TK]D-Fender | nexilus: OLD... "reload chan_zap.so" |
| 12:49.50 | phix | [TK]D-Fender: ok, but can / should I allow my phone to get that information? or isn't that supported in Asterisk ? |
| 12:50.11 | [TK]D-Fender | phix: And like I told you last night, sure, but you didn't set up the hint for it to do so. |
| 12:50.26 | phix | oh ok :) I probably dosed off |
| 12:50.40 | phix | ok, so I need to setup a hint ay, what would that be under? :) |
| 12:51.02 | [TK]D-Fender | phix: extensions.conf. Fo look up "presence" on the WIKI |
| 12:51.03 | [TK]D-Fender | ~wikis |
| 12:51.04 | jbot | methinks wikis is http://www.voip-info.org |
| 12:51.28 | phix | thnx |
| 12:51.54 | phix | now I know what I am looking for it will be alot easier then search randomly through wiki / forums |
| 12:54.17 | phix | nice found it |
| 12:54.20 | phix | thank you [TK]D-Fender! |
| 12:56.07 | docelmo | phix there are a couple ways to setup presence in asterisk. If you dont set the directive for presence context then it will use the context= direcrective. So if your peer is setup to use default in extensions.conf you would put exten => peer,hint,technology/peer or however you choose under that same [default] context |
| 12:57.30 | [TK]D-Fender | docelmo: "exten => peer,hint,technology/peer " <- ummm, almost :) |
| 12:57.59 | docelmo | what did I miss? its exten => 102,hint,sip/102 right? or along those lines |
| 12:58.14 | [TK]D-Fender | docelmo: "exten => extension,hint,technology/peer " <-more like it... |
| 12:58.24 | [TK]D-Fender | docelmo: What happens when my peer is named FRED <---- |
| 12:58.40 | docelmo | put fred there? :) |
| 12:58.52 | docelmo | I dont do named peers they are too much of a pain in the ass |
| 12:59.04 | *** join/#asterisk ToTo (n=ToTo@209.8.41.65) |
| 12:59.15 | [TK]D-Fender | docelmo: Extens are typically numerical (except by softphone addict psycho's who think analog phones can DTMF alpha-numerically :p) |
| 12:59.29 | ToTo | i all |
| 12:59.33 | docelmo | haha |
| 12:59.40 | [TK]D-Fender | docelmo: Well that would jsut be you..... a huge number of people do name them :) |
| 13:00.25 | docelmo | eh.. I find it simpler to do a exten => _1XXX,1,Dial(SIP/${EXTEN}) than having to put EVERY single one in there |
| 13:00.29 | ToTo | can i force reinvite? i wold rtp streeming bypass asterisk... |
| 13:00.41 | docelmo | toto yes canreinvite=no |
| 13:00.50 | [TK]D-Fender | docelmo: UGLY... unless you're guaranteed to be using all 1000 possibilities :) |
| 13:01.02 | ToTo | docelmo, tnx |
| 13:01.16 | mvanbaak | [TK]D-Fender: what do you recommend for a basic office setup with like 12 phones |
| 13:01.17 | [TK]D-Fender | docelmo: An ugly practice... |
| 13:01.21 | mvanbaak | the 601 or the 650 ? |
| 13:01.31 | docelmo | err bypass.. canreinvite=yes and make sure your codecs match and rtp shouldnt go thru asterisk unless you are recording or something |
| 13:01.43 | mvanbaak | 1 or 2 of them need the speeddial/blf thingies |
| 13:01.46 | [TK]D-Fender | mvanbaak: IP 650 for receptionist, IP320's for the rest |
| 13:01.53 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:02.00 | mvanbaak | ok |
| 13:02.02 | docelmo | I like my 601 w/ sidecar |
| 13:02.14 | mvanbaak | gonna see if we can get the polycom stuff here in .nl |
| 13:02.43 | nexilus | hm.. |
| 13:02.55 | docelmo | [TK]D-Fender yes true.. no I havent actually setup 1000 extensions HOWEVER I have setup 100+ and its a bitch to have to write each and every one |
| 13:02.57 | [TK]D-Fender | mvanbaak: You can, but the price is really high on import. Thats why I usually suggest Linksys there.... |
| 13:03.18 | [TK]D-Fender | docelmo: copy/paste FTW :) |
| 13:03.25 | docelmo | ewww.. |
| 13:03.35 | docelmo | still have to change parts of the lines.. :) |
| 13:03.47 | nexilus | hm |
| 13:03.56 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.182.176) |
| 13:03.59 | nexilus | fender, didnt see your reply after i said my version, care to repeat? |
| 13:04.05 | docelmo | I think I would if I did name them write a script or something to parse the number to a name or use the astdb functions |
| 13:04.14 | phearless | [TK]D-Fender : you were right, about the fact of using Dial instead of queues.conf |
| 13:04.36 | phearless | @everybody : listen to the great [TK]D-Fender he KNOWS |
| 13:04.44 | docelmo | hehe |
| 13:04.56 | docelmo | there are quite a few of us in here who KNOW.. :) |
| 13:05.02 | phix | docelmo: ok nice, yeah I have a context for my local sip user extensions, I put the hints in there. |
| 13:05.08 | docelmo | so TK when do you plan to make it to a astricon? |
| 13:06.09 | [TK]D-Fender | nexilus: Look 2 lines BELOW it. |
| 13:06.19 | [TK]D-Fender | docelmo: 2010! |
| 13:06.24 | docelmo | haha |
| 13:06.29 | nexilus | ...i cant, my service providers switches hickuped and i got booted |
| 13:06.36 | nexilus | just came back |
| 13:06.59 | [TK]D-Fender | nexilus: OLD... "reload chan_zap.so" <------- |
| 13:07.04 | nexilus | aah aight |
| 13:07.07 | nexilus | thanx |
| 13:07.18 | ToTo | docelmo, i use canreinvite= no but asterisk is always in the mediapath... |
| 13:07.19 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
| 13:07.24 | [TK]D-Fender | nexilus: But if you're saying that restarting * didn't do it, I doubt this would... |
| 13:07.34 | phearless | is it possible to dial (with the cmd Dial) the number of an autoresponder, and automatically press the button [1] ? |
| 13:07.42 | docelmo | toto yes |
| 13:07.50 | nexilus | [TK]D-Fender: yeah, didnt do the trick :( |
| 13:08.16 | loca|host | hello all |
| 13:08.16 | ToTo | docelmo, i woldn't it in mediapath.. |
| 13:08.21 | nexilus | only "trick" ive found so far that works is restarting the whole dam machine... |
| 13:08.26 | loca|host | i cant download cvs from cvs.digium.com |
| 13:08.29 | nexilus | and that seems like quite a bit of overkill.. |
| 13:08.30 | phearless | anybody understand my question, or I do have to reformulate ? |
| 13:09.17 | loca|host | where can i checkout for asterisk, zaptel and libpri ? |
| 13:09.23 | file | loca|host: we haven't used CVS in ... years |
| 13:09.25 | [TK]D-Fender | phearless: Yes... "show application dial" |
| 13:09.34 | loca|host | lol |
| 13:09.42 | phearless | thanks [TK]D-Fender |
| 13:09.54 | [TK]D-Fender | loca|host: www.asterisk.org for more current instructions... |
| 13:09.57 | loca|host | file, am trying to install asterisk at home with my new TDM01B |
| 13:10.08 | loca|host | and got this guide |
| 13:10.10 | loca|host | http://www.automated.it/guidetoasterisk.htm#_Toc49248766 |
| 13:10.33 | nexilus | Anyone have any idea if Zap usage overall has improved notably from version 1.2 to 1.4 in asterisk? |
| 13:10.42 | [TK]D-Fender | loca|host: BAD guide |
| 13:11.00 | loca|host | is there any GOOD guide ? |
| 13:11.09 | [TK]D-Fender | loca|host: Anything using RH 8.0 as a base tells you that you arre more than a few years off target |
| 13:11.21 | [TK]D-Fender | loca|host: Go downlaod... THE BOOK |
| 13:11.23 | [TK]D-Fender | ~book |
| 13:11.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:12.01 | [TK]D-Fender | loca|host: And here is a very basic sample you can start with : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 13:12.34 | [TK]D-Fender | nexilus: depends what you mean by "usage" |
| 13:14.24 | loca|host | [TK]D-Fender, :( same thing: |
| 13:14.25 | loca|host | ftp.digium.com |
| 13:14.33 | loca|host | host unreachable |
| 13:15.38 | file | the FTP server no longer exists |
| 13:15.44 | loca|host | i'll get from here http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.13.tar.gz |
| 13:15.50 | file | http://downloads.digium.com/ |
| 13:16.30 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:16.43 | [TK]D-Fender | loca|host: tHE GUIDE WAS MORE FOR CONFIGURING, NOT INSTALLING. |
| 13:17.10 | [TK]D-Fender | \o/ capslock |
| 13:17.16 | loca|host | [TK]D-Fender, i see |
| 13:17.27 | loca|host | but it started by "Download the latest 1.4 version of Asterisk from ftp.digium.com." |
| 13:17.44 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:17.46 | loca|host | and i reads docs from the begin :) |
| 13:17.48 | [TK]D-Fender | loca|host: [09:09]<[TK]D-Fender>loca|host: www.asterisk.org for more current instructions... <----- |
| 13:17.56 | [TK]D-Fender | loca|host: Like I said first.. |
| 13:18.01 | loca|host | ok dude :) |
| 13:18.25 | loca|host | it's my very first hour with asterisk, be easy man :) |
| 13:20.12 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:20.42 | loca|host | [TK]D-Fender, will the 1.1 handle unobstrosive JS by default ? and adding an abstraction layer over JS frameworks and not to force to a given fwk ? |
| 13:20.46 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:20.57 | loca|host | because proto+scriptaculous is a shit |
| 13:21.07 | [TK]D-Fender | loca|host: ..... huh? |
| 13:21.09 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:21.24 | loca|host | when having 70kb to download for these two framworks when just calling Javascript helper |
| 13:21.36 | [TK]D-Fender | loca|host: What does * have to do with JS? |
| 13:21.43 | loca|host | lol |
| 13:21.45 | loca|host | sorry |
| 13:21.52 | loca|host | was wrong on the channel |
| 13:21.55 | loca|host | sorry |
| 13:21.57 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:23.38 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:23.46 | phearless | [TK]D-Fender: does not work, it send the [1] too early |
| 13:23.51 | phearless | [TK]D-Fender: -- Sending DTMF '1' to the called party. |
| 13:24.04 | *** join/#asterisk mavior (n=mavior@host-84-221-233-242.cust-adsl.tiscali.it) |
| 13:24.17 | [TK]D-Fender | phearless: when DOES it send it? |
| 13:24.29 | [TK]D-Fender | phearless: Because that paste doesn't say anything useful. |
| 13:25.17 | [TK]D-Fender | phearless: And what techa re you using to place your call? |
| 13:25.56 | phearless | [TK]D-Fender: http://pastebin.ca/746745 |
| 13:27.04 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 13:27.26 | phearless | sorry I dot disconnected |
| 13:28.26 | mavior | hello...i'm going to buy an OpenVox A400 instead of a TDM4oop...any experience with that card? is there so many differences with the digium one in terms of quality? |
| 13:30.30 | phearless | [TK]D-Fender: did you get my CLI log ? |
| 13:30.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:30.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:31.52 | [TK]D-Fender | phearless: Yes... what card? And whene xactly did it send the dtmf? |
| 13:32.08 | [TK]D-Fender | mavior: ... |
| 13:32.10 | [TK]D-Fender | ~wglwat |
| 13:32.10 | jbot | wglwat is, like, well, good luck with all that |
| 13:32.23 | [TK]D-Fender | mavior: Few people here would touch it |
| 13:32.41 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 13:32.48 | [TK]D-Fender | mavior: if you're lucky is exactly the same... which give the TDM400P ... isn't a GOOD thing. |
| 13:33.01 | phearless | [TK]D-Fender: I use a UK PRI to dial numbers, and I used : exten => 496,1,Dial(Zap/g1/0015166xxxxxx,20,D(1)) |
| 13:33.24 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:33.26 | phearless | [TK]D-Fender: the card is a Sangoma |
| 13:33.38 | mavior | <[TK]D-Fender> : dunno understand...what is not a good idea? |
| 13:33.41 | [TK]D-Fender | phearless: and when does it send? |
| 13:33.48 | [TK]D-Fender | mavior: That card. |
| 13:33.57 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:34.09 | wwalker | When is ztdummy actually needed? I know meetme uses it, but if you are not using meetme, when do you need ztdummy? |
| 13:34.16 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:34.20 | wwalker | (in a pure SIP RTP environment) |
| 13:35.10 | [TK]D-Fender | wwalker: MeetMe & IAX2 Trunking |
| 13:35.52 | mavior | <[TK]D-Fender> i know..but is cheaper than the digium one (i own already a digium 400p) and i am doing a 'budget' consideration..... |
| 13:36.05 | phearless | [TK]D-Fender: http://pastebin.ca/746751 here we go ! |
| 13:36.33 | keith4 | if I bought a digium card on ebay, any chance of unlocking the hw echo cancelling? |
| 13:36.48 | [TK]D-Fender | phearless: So that's pretty instant from the time of answer... |
| 13:37.01 | [TK]D-Fender | keith4: There is no "locking" |
| 13:37.01 | phearless | [TK]D-Fender: yes... :( |
| 13:37.10 | phearless | [TK]D-Fender: i'd like to delay this during 1 second |
| 13:37.12 | [TK]D-Fender | keith4: LOL.. this isn't like a cell phone you know.. |
| 13:37.26 | [TK]D-Fender | phearless: Dunno, see if there is something in the isntructions for how to delay. |
| 13:37.28 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 13:37.30 | keith4 | right, but don't you need some key from digium? |
| 13:37.47 | [TK]D-Fender | keith4: for HPEC LICENSES, yes. |
| 13:37.55 | keith4 | yah, that! |
| 13:37.56 | phearless | [TK]D-Fender: ok .... I think that i'm screwed |
| 13:38.21 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 13:38.28 | lsodi | jambo! |
| 13:38.45 | phearless | [TK]D-Fender: the documentation say : D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) |
| 13:38.56 | [TK]D-Fender | keith4: it'd have to be under warranty, and then again, that'd be for SOFTWARE EC, not HWEC like you were jsut asking about. |
| 13:38.58 | phearless | [TK]D-Fender: where can I add this w ? i'm confused |
| 13:39.10 | [TK]D-Fender | keith4: HARDWARE EC doesn't need licensing. |
| 13:39.21 | keith4 | [TK]D-Fender: oh. so why would i want software EC? |
| 13:39.22 | phearless | [TK]D-Fender: I will try wwwwD(1) |
| 13:39.28 | [TK]D-Fender | phearless: D(wwwwwww12345) |
| 13:39.36 | phearless | ok [TK]D-Fender |
| 13:39.40 | [TK]D-Fender | keith4: You don't. |
| 13:40.18 | keith4 | wait... so for the TDM400, why do they license a software EC, when it has hardware EC? |
| 13:40.53 | JT | it doesn't have hardware EC |
| 13:40.54 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 13:40.57 | JT | not an option |
| 13:41.08 | [TK]D-Fender | keith4: it DOESN'T have HWEC. |
| 13:41.18 | keith4 | ah |
| 13:41.22 | keith4 | good reason! |
| 13:41.23 | [TK]D-Fender | keith4: You are clearly completely lost about the products your are researching... |
| 13:41.30 | keith4 | yep |
| 13:42.01 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 13:42.56 | keith4 | so... digium cards on ebay, probably not under warranty. no way to use HPEC? |
| 13:43.17 | [TK]D-Fender | keith4: Sure, you can pay 10$ or so a channel on their site.... |
| 13:43.39 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 13:43.42 | [TK]D-Fender | keith4: and I highly advise you to stop cheaping-out on your hardware purchases. |
| 13:43.47 | [TK]D-Fender | ~ygwypf |
| 13:43.48 | jbot | well, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 13:43.49 | [TK]D-Fender | ~cheap |
| 13:43.49 | jbot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 13:44.03 | [TK]D-Fender | keith4: the TDM400P is a LOW target. |
| 13:44.39 | keith4 | what's the next step up? |
| 13:46.40 | [TK]D-Fender | keith4: What exactly are you setting up? |
| 13:46.56 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 13:47.04 | keith4 | home office system |
| 13:47.07 | keith4 | 2 analog trunks |
| 13:48.52 | lsodi | Cisco cme4.2 and Asterisk sip Trunk... has any one worked on that kind of solution? |
| 13:49.51 | [TK]D-Fender | keith4: I guess if you're willing to take your chances with it maybe not so bad. |
| 13:50.12 | keith4 | i like to live on the edge |
| 13:50.18 | [TK]D-Fender | keith4: But go new so its warranteed |
| 13:50.35 | [TK]D-Fender | keith4: Be prepared to get cut |
| 13:50.53 | keith4 | where is a good place to buy them new? (US) |
| 13:51.39 | blitzrage | TDM800P uses a different chipset than the TDM400P |
| 13:52.04 | blitzrage | which all Digium hardware has switched to (other than the TDM400P) |
| 13:53.07 | *** join/#asterisk bantu (n=Miranda@p54A3296F.dip0.t-ipconnect.de) |
| 13:53.46 | [TK]D-Fender | keith4: www.telephonydepot.com |
| 13:53.57 | keith4 | [TK]D-Fender: thanks |
| 13:54.52 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 13:56.28 | phix | How to figure out how to transfer calls |
| 13:56.43 | phix | I just need to enable the featuremap or I need to define an exten as well? |
| 13:57.34 | peanut- | that aussie sounds like a durka. |
| 13:57.43 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 13:58.09 | [TK]D-Fender | phix: depends on your phone. |
| 13:58.31 | Dabba | anyone know of a wiki entry to get the cfwd button to work on a cisco or linksys |
| 13:59.13 | phix | peanut-: wtf? |
| 13:59.22 | [TK]D-Fender | Dabba: should "just work". What's it doing? |
| 13:59.24 | phix | [TK]D-Fender: softphone |
| 13:59.37 | [TK]D-Fender | phix: Which? |
| 13:59.38 | phix | x-lite |
| 13:59.40 | [TK]D-Fender | ~softphone |
| 13:59.40 | jbot | something that should be drug out into the street and shot |
| 13:59.52 | JT | dragged out? |
| 14:00.01 | [TK]D-Fender | phix: "show application dial" |
| 14:00.02 | JT | peanut-: what do you mean? |
| 14:00.20 | [TK]D-Fender | phix: And go read up on features.conf |
| 14:00.27 | Dabba | D-Fender if you puch it and punch in a normal pstn number you get |
| 14:01.19 | [TK]D-Fender | Dabba: PASTEBIN a failed attempt at verbose 10, SIP DEBUG enabled. |
| 14:01.21 | [TK]D-Fender | ~pb |
| 14:01.22 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:01.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 14:01.50 | JT | peanut-: what is a durka? |
| 14:02.04 | *** join/#asterisk HarryR`Work (n=harryr@77.240.56.17) |
| 14:02.21 | *** join/#asterisk saftsack (n=saftsack@pD9E041E0.dip.t-dialin.net) |
| 14:02.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:03.00 | [TK]D-Fender | JT : http://www.urbandictionary.com/define.php?term=durka |
| 14:03.03 | Dabba | D-FENDER |
| 14:03.07 | Dabba | http://pastebin.ca/746779 |
| 14:03.26 | Dabba | where 11111 is the pstn number |
| 14:03.41 | Dabba | its dumping the call into default :-( |
| 14:03.47 | Dabba | which of course has no pstn access |
| 14:04.38 | [TK]D-Fender | Dabba: it transfers to the context that [1004] uses. |
| 14:05.01 | [TK]D-Fender | Dabba: Which is apparently [default] and doesnt' have an exten to match. |
| 14:05.36 | Dabba | it isnt default |
| 14:05.45 | Dabba | its defined as officeusers |
| 14:05.49 | *** join/#asterisk Derky (n=derky@193.141.36.251) |
| 14:06.11 | [TK]D-Fender | Dabba: Show me you phone's entry and do the next pastbin with the WHOLE call, and with sip debug enabled. |
| 14:08.58 | *** join/#asterisk littleball (n=littleba@bb220-255-76-180.singnet.com.sg) |
| 14:12.03 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:22.10 | Dabba | [TK]D-Fender: http://pastebin.ca/746802 |
| 14:22.53 | agx | Anyone connected * to an Innovaphone? i'm trying to register * to Innovaphone but he reply "404 NOT FOUND" |
| 14:23.07 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 14:23.13 | [TK]D-Fender | Dabba: So far I don't see a valid exten for that forward to work... |
| 14:24.00 | Dabba | the ip6netusers context contains pstn access |
| 14:24.13 | badcfe | is it somehow possible to use another dtmf digit than '*' for the H option of dial? (like a work around?) |
| 14:24.15 | [TK]D-Fender | Dabba: please show me something COMPLETE and useful... |
| 14:24.29 | [TK]D-Fender | badcfe: You have the sourcecode.... get busy |
| 14:24.40 | badcfe | [TK]D-Fender: hehe |
| 14:25.00 | *** join/#asterisk lirakis (n=eric@69.24.142.1) |
| 14:25.10 | lirakis | hey everyone |
| 14:25.28 | lirakis | .. my queues dont seem to be taking the strategy that I am setting... |
| 14:26.20 | Dabba | the point is why is it doing No such extension/context 01737822860@default |
| 14:27.14 | putnopvut | lirakis: what strategy are you setting, and what's happening wrong? |
| 14:27.48 | lirakis | putnopvut: i set "strategy=rrmemory" in queues.conf. When i do show queues from cli .. it says strategy is ringall |
| 14:28.13 | lirakis | <PROTECTED> |
| 14:28.23 | putnopvut | That would be the problem. |
| 14:28.30 | putnopvut | You have to set the strategy inside the queue. |
| 14:28.31 | lirakis | putnopvut: has to be per queue? |
| 14:28.33 | lirakis | okay |
| 14:28.34 | putnopvut | Yes. |
| 14:28.38 | lirakis | putnopvut: thanks |
| 14:28.41 | putnopvut | np |
| 14:28.53 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 14:29.15 | Dabba | [TK]D-Fender: http://pastebin.ca/746808 |
| 14:29.32 | Dabba | thats everything ! |
| 14:29.34 | lirakis | .. now .. for some reason.. i went into cli "asterisk -vvvvvr" .. and i issued a reload... nothing shows on the screen... so .. i wait still nothing .. i try again .. and it says "the previous reload has not finished" ... hrmm... this system has live calls.. so i cant really restart asterisk |
| 14:31.08 | [TK]D-Fender | Dabba: please PB 1006 as well |
| 14:31.53 | lirakis | the cli is also not responding to some commands ... like "show queues" .. just sits there... but "show channels" and "sip show peers" ..seem to work fine ... whats going on? |
| 14:32.23 | Dabba | tk |
| 14:32.32 | Dabba | [TK]D-Fender: http://pastebin.ca/746811 |
| 14:32.39 | Dabba | no default in there either |
| 14:33.02 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:33.05 | Dabba | lirakis: reboot needed |
| 14:35.11 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:35.14 | lirakis | Dabba: reboot? .. or restart of asterisk |
| 14:35.21 | lirakis | Dabba: .. and what would cause this? |
| 14:36.35 | Dabba | i have no idea, i know ours does that sometimes and i have to do restart now |
| 14:37.17 | [TK]D-Fender | Dabba: :/ |
| 14:37.24 | Dabba | ya |
| 14:37.50 | Dabba | quit |
| 14:38.01 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:38.02 | Dabba | doh wrong windy |
| 14:43.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:44.52 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:45.38 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 14:47.40 | *** join/#asterisk socken23 (n=socken@123-117.77-83.cust.bluewin.ch) |
| 14:48.11 | socken23 | Hi all! I'm new to asterisk and trying to setup incoming fwdnet according to http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+6#61FreeWorldDialupFWDbspan |
| 14:48.37 | socken23 | But I always get a Registration refused .. any idea why!? |
| 14:48.51 | socken23 | (user and password are correct, I checked that twice ;-) ) |
| 14:49.20 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 14:53.37 | [TK]D-Fender | socken23: apparently not |
| 14:54.09 | socken23 | I tried to configure it directly in iax.conf and extensions.conf and via the freePBX webinterface |
| 14:54.11 | socken23 | same effect |
| 14:55.24 | [TK]D-Fender | ~freepbx |
| 14:55.25 | jbot | methinks freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:55.34 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 14:55.41 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-221.pools.arcor-ip.net) |
| 14:56.13 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 14:56.40 | socken23 | Is there any way to increase verbosity so I can see what is actually failing?? |
| 14:58.20 | Dabba | [TK]D-Fender: so any ideas |
| 14:58.20 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:58.44 | [TK]D-Fender | Dabba: not offhand... |
| 14:58.58 | [TK]D-Fender | socken23: iax2debug |
| 14:59.04 | [TK]D-Fender | socken23: "iax2 debug" |
| 14:59.20 | [TK]D-Fender | socken23: http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 |
| 14:59.53 | socken23 | thanx! |
| 15:00.21 | socken23 | ah, that's the configuration I tried by hand... |
| 15:00.24 | socken23 | didn't work |
| 15:00.41 | Dabba | another unsolved asterisk caveat |
| 15:03.18 | *** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell) |
| 15:03.18 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 15:03.21 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 15:03.44 | [TK]D-Fender | socken23: Guess you missed something. |
| 15:03.56 | [TK]D-Fender | socken23: go prove your account is right with a softphone login. |
| 15:04.22 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 15:05.08 | socken23 | OK, I'll try that |
| 15:06.13 | socken23 | using FWD:Communicator works !? Strange.. |
| 15:06.19 | socken23 | Maybe there's a NAT problem.. |
| 15:07.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:08.57 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
| 15:09.21 | loca|host | re |
| 15:09.39 | loca|host | am allways getting this error when trying to subscribe a sipphone: Registration from '<sip:fourat.zouari@shark.tux>' failed for '10.10.1.196' - No matching peer found |
| 15:10.23 | loca|host | and i have in sip.conf username=fourat.zouari ... |
| 15:10.27 | loca|host | that seems correct |
| 15:10.35 | *** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt) |
| 15:11.09 | _foxfire_ | can any help with asterisk licensing |
| 15:11.33 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 15:12.20 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 15:13.27 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:15.24 | Ritzerisk | haha do you live here Fender ? |
| 15:16.39 | naitram | using AMI send Hangup for sip channel, reply is no such channel. does Hangup work for sip channels? |
| 15:17.41 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 15:19.34 | creativx | yes naitram |
| 15:20.31 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 15:20.59 | naitram | creativx: do you know if it works in v 1.2? |
| 15:22.37 | naitram | I call the hangup using the same string for the channel that I use to originate a call so the channel name is right, but it comes back with no such channel |
| 15:23.57 | creativx | you need to use the channel name that asterisk gives it |
| 15:23.58 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 15:24.07 | *** join/#asterisk Op3r (n=edwin@125.212.63.243) |
| 15:24.34 | Ritzerisk | asterisk Licensing ?? isnt it all open source ? |
| 15:24.50 | naitram | creativx: gives me when? |
| 15:25.15 | [TK]D-Fender | _foxfire_: Typicall we all run the OOS common release fo * which is well... GPL.... there IS no licensing.. |
| 15:25.26 | [TK]D-Fender | Ritzerisk: Mostly, yes |
| 15:25.57 | [TK]D-Fender | naitram: the channel you dial is not your CALL'S channel. |
| 15:26.14 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:26.50 | c0rnflake | hey guys, i have a question about attended transfers |
| 15:27.26 | c0rnflake | is there a way to let the recipient know that a call was successfully transferred to them, like a beep or something? |
| 15:27.44 | [TK]D-Fender | c0rnflake: Nope. |
| 15:27.45 | naitram | [TK]D-Fender: ok, so I need to do some more reading. Thanks All |
| 15:29.09 | loca|host | anyine ? |
| 15:29.11 | loca|host | anyone ? |
| 15:29.14 | loca|host | am allways getting this error when trying to subscribe a sipphone: Registration from '<sip:fourat.zouari@shark.tux>' failed for '10.10.1.196' - No matching peer found |
| 15:29.18 | loca|host | and i have in sip.conf username=fourat.zouari ... |
| 15:31.53 | file | loca|host: username is for when you are placing a call to a peer and they request authentication, it tells chan_sip what username to use for that authentication - it is NOT the username of devices registering to your Asterisk |
| 15:32.55 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:33.09 | naitram | local|host: Do you have the asterisk book? |
| 15:33.16 | loca|host | file, so how to fix the error: No matching peer found |
| 15:35.20 | loca|host | no |
| 15:38.01 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 15:38.23 | *** join/#asterisk blq (n=Bl@dslb-088-064-143-231.pools.arcor-ip.net) |
| 15:38.27 | naitram | local|host: go to Oreilly.com or to asterisk.org and get the book Asterisk The future of telephony. It has a complete example of how to set up a sip softphone. without doing a lot of reading, you will not be able to set this up very easy. IMHO |
| 15:38.44 | naitram | oh, its a free download by the way |
| 15:40.13 | *** join/#asterisk nybble (n=jhurley@about/apple/performa/nybble) |
| 15:40.54 | nybble | hey all, anyone have a method for turning down manager interface verbosity on the console? |
| 15:44.51 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 15:45.27 | *** join/#asterisk spaghetty (n=spaghett@lugbari/people/spaghetty) |
| 15:46.34 | spaghetty | hi .. i just set up asterisk 1.2.16 on ubuntu box ... i get some message about "unsupported scheme" from my app |
| 15:46.53 | spaghetty | seems that asterisk loose the "sip" tag before the uri |
| 15:46.59 | spaghetty | in message 100 |
| 15:47.38 | spaghetty | i know this behaveaur was patched but i need to applay the smallest change set on server |
| 15:47.53 | spaghetty | someone can suggest me where to found the selective patch for this ? |
| 15:48.22 | *** join/#asterisk SM0TVI (i=teodor@c80-217-63-18.bredband.comhem.se) |
| 15:52.00 | Ritzerisk | hmm strange .. it says username mismatch but i can still dial |
| 15:53.00 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:57.13 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 15:57.29 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:57.33 | agx | need to replace 2 Grandstream 8xFXS gateway: any idea wich other product is nice? |
| 15:59.44 | phix | ummm a few TDM cards in a core 2 duo |
| 15:59.52 | phix | That would go down nice |
| 16:00.02 | phix | or perhaps a dual or quad xeon |
| 16:02.17 | agx | phix, no way i'm using TDM cards, need to use SIP/FXS gateway, they are placed in a different building then the PBX |
| 16:03.07 | *** join/#asterisk punkgode (n=punkgde@rev-200-40-119-222.netgate.com.uy) |
| 16:03.08 | [TK]D-Fender | agx: SPA-8000 |
| 16:05.25 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:05.35 | punkgode | wwalker, did call forward work? |
| 16:06.20 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.247) |
| 16:07.35 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 16:07.55 | jstew | Hey guys, I come seeking recommendations. |
| 16:08.19 | jstew | Is there a desktop app that I can use to manage the switchboard other than hudlite? |
| 16:08.48 | nestAr | anyone got a second to look at a Set(GROUP) config and tell me why it doesn't work? |
| 16:09.17 | Alan_Hicks | Is there an rc script for zaptel on Slackware, or should I just modify zaptel.init to suit my purposes? |
| 16:09.49 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 16:10.27 | jstew | You could roll your own and put it in rc.local or something Alan_Hicks. I haven't used slackware in years though :) |
| 16:11.12 | *** part/#asterisk harpal (n=Harpal@124.125.254.227) |
| 16:12.45 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 16:12.52 | HarryR`Work | Alan_Hicks, just modify the zaptel.init one |
| 16:13.04 | HarryR`Work | it shouldn't be dependant on anything distribution specific iirc, perhaps just LSB |
| 16:13.07 | nestAr | anyone got a second to look at a Set(GROUP) example that doesn't seem to work for me? |
| 16:13.31 | Alan_Hicks | HarryR`Work: Thanks. I figured that would be the case. Google didn't turn up anything too promising. |
| 16:13.35 | agx | funny :) Oct 23 17:57:23 192.168.1.99 GXW4008: [00:0B:82:0E:EA:AA] SIGSEGV raised |
| 16:14.29 | HarryR`Work | slackware's init system is pretty basic, so any shell script'll do :\ |
| 16:14.45 | *** join/#asterisk ManxPower (n=manxpowe@42.sub-70-223-100.myvzw.com) |
| 16:14.52 | Alan_Hicks | Yeah. That's one of the things I like about it. |
| 16:14.59 | jsaunders | Anyone ever had fxo chans on a zapata tdm card lose their ability to "hear" anything after awhile? ie, doing a ztmonitor and rx is completely dead. Was working and then suddenly stopped, on all 8 chans. tx works fine however. |
| 16:15.39 | *** join/#asterisk Blueneon (n=blue@dsl-146-29-195.telkomadsl.co.za) |
| 16:17.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:17.38 | Blueneon | hi im trying to figure out why after upgrading to the latest version of asterisk when i hit the (R) button on my phone the caller doesnt get the on-hold music and is instead left with a silence, but if I do a transfer (R) Ext, then they will get the on-hold music... before hand all I had to do to place my callers "on hold" to hear music while i did something was to simply use the (R) button. |
| 16:17.50 | ussrback | hi all |
| 16:17.50 | Blueneon | I'm using TDM400 and standard analog handsets |
| 16:17.59 | ussrback | how can i call chanspy using agi |
| 16:18.01 | ussrback | $AGI->ChanSpy("$channel"); ? |
| 16:18.17 | ussrback | $AGI->ChanSpy("$channel",wW); ? |
| 16:18.27 | ussrback | i use perl |
| 16:19.52 | ManxPower | ussrback: I didn't know that asterisk-perl supported AGI->ChanSpy. You will have to use AGI->exec |
| 16:20.57 | *** part/#asterisk spaghetty (n=spaghett@lugbari/people/spaghetty) |
| 16:21.38 | ussrback | but how can i pass perl variable in chanppy command if ill use exec |
| 16:21.39 | ussrback | ? |
| 16:21.51 | ManxPower | you do not use perl exec |
| 16:21.52 | TJNII | Is there a queue reload command or do I need to do a whole restart? |
| 16:21.54 | ManxPower | you use AGI exec |
| 16:23.08 | ussrback | ok ... the same with voicemail ? |
| 16:23.33 | ManxPower | $AGI->exec($app, $options) |
| 16:23.42 | ManxPower | read the manpage, dude. |
| 16:23.54 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:24.49 | ussrback | gime link |
| 16:24.52 | ussrback | ;) |
| 16:25.57 | ManxPower | A link? The man pages are installed when you install asterisk-perl |
| 16:26.01 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 16:26.28 | ManxPower | but if you can't figure out how to read the manpage then you can use the Place of Last Resort -- the Wiki -- http://www.voip-info.org/wiki/view/Asterisk+perl+agi |
| 16:27.12 | *** join/#asterisk nm2588 (i=user@165.138.2.140) |
| 16:27.43 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 16:28.30 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:29.13 | *** join/#asterisk corpcomp (n=corpcomp@125-236-174-30.broadband-telecom.global-gateway.net.nz) |
| 16:31.50 | corpcomp | I have just installed a test server as per "http://www.freepbx.org/support/documentation/installation/install-procedure-for-centos-4-3" when I goto http://myserver/admin and get mysql://asteriskuser:amp109@localhost/asterisk. Yes I used defaults but this is just to test asterisk. Any comments would be helpful |
| 16:35.28 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 16:35.52 | ussrback | anyone uses here perl for AGI ? |
| 16:36.00 | ussrback | anyone uses perl for AGI ? |
| 16:36.36 | outtolunc | what now? (meaning what in that agi i showed you didn't you understand) |
| 16:36.50 | *** part/#asterisk nm2588 (i=user@165.138.2.140) |
| 16:37.58 | ussrback | http://pastebin.ca/index.php |
| 16:38.28 | outtolunc | how about the link to that actual paste |
| 16:38.42 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:38.54 | outtolunc | click 'submit' it will give you a link |
| 16:39.17 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:40.02 | ussrback | sorry they forged it |
| 16:40.03 | ussrback | http://sial.org/pbot/28188. |
| 16:41.14 | [TK]D-Fender | corpcomp: ..... |
| 16:41.16 | [TK]D-Fender | ~freepbx |
| 16:41.17 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:42.04 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 16:42.35 | outtolunc | Voicamail hmm |
| 16:42.52 | xheliox | Friggin Teliax, agian. |
| 16:43.09 | *** join/#asterisk stunsch (n=stunsch@104.Red-83-63-195.staticIP.rima-tde.net) |
| 16:43.52 | stunsch | I'm trying to make an outbound call on a billion bri card |
| 16:43.57 | ussrback | outtolunc: i changed it on voicemail .... but problem is, how to pass variables there |
| 16:44.22 | stunsch | It gives a cause 66 channel not implemented error |
| 16:44.27 | outtolunc | the same way you do to other exec's |
| 16:44.50 | outtolunc | which there are a shitload of examples in that agiIVR.agi i showed you |
| 16:46.00 | outtolunc | meaning exec("appname","appparams") |
| 16:46.15 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 16:46.24 | outtolunc | whereas you.. are still using it as a function with ()'s and all |
| 16:46.25 | l0verb0y | hey hows it going? |
| 16:46.38 | l0verb0y | does anyone know how i can have my ring group calls recorded? |
| 16:46.49 | Blueneon | I cannot figure out why my callers dont hear the onhold music when i press the (R) button, but when i transfer a call they get the music, any ideas? |
| 16:47.07 | wwalker | punkgode: haven't heard from him today. |
| 16:47.09 | [TK]D-Fender | l0verb0y: "show application monitor" , " show application mixmonitor" |
| 16:47.22 | punkgode | wwalker, oh ok |
| 16:47.42 | l0verb0y | thanks |
| 16:48.55 | ussrback | outtolunc: $AGI->exec("ChanSpy", "$channel","wW"); |
| 16:48.58 | ussrback | lie this? |
| 16:49.42 | ManxPower | well, the example shows TWO options, but you are giving it THREE options. |
| 16:49.46 | ManxPower | you do the math |
| 16:50.03 | ManxPower | You've never programmed in perl before have you? |
| 16:50.19 | outtolunc | you need to concat the params and feed it 2 like manx said |
| 16:53.20 | *** join/#asterisk asdx (n=kde-deve@adsl-145-217.click.com.py) |
| 16:53.55 | `Sauron | ~openpbx |
| 16:53.55 | jbot | well, openpbx is something that started off as an asterisk fork, but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in asterisk. see http://openpbx.org/ for more info, or join #openpbx |
| 16:54.09 | `Sauron | hehe |
| 16:55.18 | tzafrir | ~callweaver |
| 16:56.38 | stunsch | I'm trying to make an outbound call on a billion bri card but get a "cause 66 channel not implemented error". Any help? |
| 16:56.54 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:57.15 | ManxPower | stunsch: paste just the actual Dial line from the CLI output. |
| 16:57.18 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 16:57.23 | tzafrir | ~callweaver |
| 16:57.23 | jbot | methinks callweaver is something that started off as a fork of Asterisk (b the name of openpbx), but is more of a rewrite of the internals and all good old GPL instead of the split licence stuff in Asterisk. see http://callweaver.org/ for more info, or join #callweaver |
| 16:58.41 | stunsch | ManxPower: http://pastebin.com/d228d6fbd |
| 17:00.22 | tzafrir | ~openpbx |
| 17:00.23 | jbot | i heard openpbx is a free software PBX written in PERL. Written by Voicetronix. Maybe you meant callweaver, which was once caller openpbx. |
| 17:00.58 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 17:03.59 | n00dle | Question: Anyone using russell's func_devstate? |
| 17:05.09 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@adsl-99-162-117-1.dsl.austtx.sbcglobal.net) |
| 17:06.31 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:07.49 | CoffeeIV_ | I am passing a call from one asterisk to another using Dial() and IAX2. I want the caller id information to accompany the call -- how can I do that ? |
| 17:10.05 | n00dle | CoffeeIV_, use the CALLERID() function to set it before sending the call. (see "core show function CALLERID" on CLI for help) |
| 17:11.10 | CoffeeIV_ | n00dle: thanks, I was thinking of doing that -- I thought that there was some argument to Dial() that passed it on ? maybe not |
| 17:11.47 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) |
| 17:11.55 | n00dle | Nope, set it before the dial and it will be passed. |
| 17:12.00 | tripps | ~iax |
| 17:12.01 | jbot | somebody said iax was port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 17:12.06 | steve | hi all |
| 17:12.18 | n00dle | Hi steve |
| 17:12.28 | steve | can a standard voice modem be used to transfer PSTN calls to voip? |
| 17:12.31 | Blueneon | why am i getting no music when starting a threewaycall ? |
| 17:12.34 | steve | handled by asterisk |
| 17:12.36 | tripps | jbot: thanks! |
| 17:12.36 | jbot | my pleasure, tripps |
| 17:12.41 | tripps | heh |
| 17:13.25 | n00dle | steve: it's not supported unless its a digium card, but some people have claimed to have made it work. |
| 17:13.40 | steve | n00dle: so what kind of hardware do I need? |
| 17:13.41 | flujan | hi guys... |
| 17:13.51 | flujan | the busy-limit option for sip.conf is working on 1.4.13? |
| 17:14.34 | n00dle | steve, digium/sangoma/rhino or similar analog interface card. check out digium.com and/or voipsupply.com. |
| 17:14.58 | CoffeeIV_ | steve: get one of those cheap-ass Wildcard x100p clones off of ebay (low quality), or an ATA module like the cisco one (better, more expensive) |
| 17:15.12 | steve | thanks |
| 17:15.27 | *** join/#asterisk Egonis (n=roman@tfi1meg.1meg.golden.net) |
| 17:15.27 | steve | for the better quality what kind of prices do you think? |
| 17:16.18 | Egonis | I have a tor2 T1 card, and a WCTDM2400P, and ran genzaptelconf, ztcfg -vvv, and ran asterisk -- but no channels appear. I noticed that there is a file called zapata-channels.conf in /etc/asterisk, but how do I make the channels appear when typing 'zap show channels'? |
| 17:17.16 | n00dle | steve: I'm using a rhino 8-port card we paid about $500 for. Single port clone should probably run about $35 for a decent one. |
| 17:17.24 | steve | nice |
| 17:17.39 | steve | and excuse my ignorance, but will a US version work in the UK? |
| 17:18.20 | n00dle | steve, check the support site for each card to see the set of countries supported: most are configurable. |
| 17:20.29 | tzafrir | Egonis, echo '#include zapata-channels.conf' >>/etc/asterisk/zapata.conf |
| 17:20.31 | sevard | Anyone remember how dialplan strings in a sipura work? |
| 17:20.34 | ManxPower | stunsch: I doubt "isdn/g:isdn/629411470||r" is a valid Dial line. |
| 17:21.06 | ManxPower | stunsch: now paste just the 1 Dial line from extensions.conf |
| 17:22.12 | *** join/#asterisk qs- (i=qs@pi.nxs.se) |
| 17:25.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:26.22 | roxlu | hi |
| 17:26.41 | roxlu | can someone help me with this message: http://paste-it.net/4113 |
| 17:26.52 | roxlu | As far as I can see this is correct |
| 17:27.59 | ManxPower | roxlu: you forgot the leading _ in your pattern match |
| 17:28.01 | *** join/#asterisk Quintana (n=sylvain@213.215.63.10) |
| 17:28.01 | *** part/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 17:28.07 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 17:28.12 | putnopvut | You also misspelled "incomingcalls" in your goto |
| 17:28.29 | ManxPower | ALSO you CANNOT Goto() a pattern match. |
| 17:28.34 | ManxPower | you have to goto a real number |
| 17:28.43 | roxlu | pff sorry... you are right :$ gonna fix it |
| 17:29.14 | ManxPower | exten => _3171XXXXXX,1,Goto(incomincalls,${EXTEN},1) shold do it. |
| 17:29.30 | ManxPower | of course you have to spell "incomming" correctly |
| 17:29.41 | ManxPower | roxlu: are you drunk? |
| 17:29.42 | roxlu | tyes |
| 17:29.43 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 17:29.50 | roxlu | Thanks a lot it is working now :D |
| 17:30.07 | roxlu | (next step is outgoing calls :-) |
| 17:30.07 | EmleyMoor | Have FWD withdrawn free access to US 800 numbers? |
| 17:30.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:30.20 | ManxPower | *grumple* I have 3 T-1s down |
| 17:30.21 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 17:30.41 | WilliamK | ManxPower, that bytes |
| 17:30.45 | roxlu | ManxPower: for incoming calls, is it necessary to have a default extension ? |
| 17:31.00 | ManxPower | roxlu: that depends on how the calls come into the system |
| 17:31.33 | roxlu | Okay. At the end of the register I have /[number] (which is used on line 7) |
| 17:31.38 | ManxPower | if the call is coming in on a FXO port then Asterisk does NOT know the numbered dialed. When Asterisk does not know the number dialed, it sets the number to "s" and exten => s,1,whatever will be matched. |
| 17:32.00 | ManxPower | if asterisk DOES know the dialed number then an exten => line matching the dialed number will be matched and NOT the "s" extension. |
| 17:32.02 | *** join/#asterisk mugawuki (n=admin_ae@extranet.lehighgas.com) |
| 17:32.15 | roxlu | ok great (I think I understand a bit more how Asterisk works now :) |
| 17:32.22 | *** join/#asterisk deadkode (n=pshively@extranet.lehighgas.com) |
| 17:32.47 | ManxPower | the /number at the end of the register => line tells Asterisk to REQUEST the remote server send calls to that extension on Asterisk when a call comes in. The provider is not required to accept the request. |
| 17:32.50 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 17:33.08 | ManxPower | (I think most providers to accept it) |
| 17:33.25 | roxlu | it's working now, so it looks like my provider accepts it |
| 17:33.36 | dlynes_laptop | ParkAndAnnounce should allow you to come back to a specified context, priority and extension, right? |
| 17:34.04 | ManxPower | dlynes_laptop: it is documented that way, but it has never worked as documented for me. |
| 17:34.35 | dlynes_laptop | ManxPower: ah, ok...I've been shaking my head at it for the better part of 2 weeks now, and can't seem to get anywhere with it |
| 17:34.38 | dlynes_laptop | ManxPower: thanks |
| 17:34.40 | *** join/#asterisk deadkode (n=deadkode@extranet.lehighgas.com) |
| 17:37.06 | ManxPower | watch the console, it will tell you where it is trying to timeout to |
| 17:37.26 | ManxPower | then create an exten and context that is referenced in the error message. |
| 17:38.53 | EmleyMoor | Has anyone got a toll-free number in the Netherlands, Norway or Germany I could try? |
| 17:39.02 | roxlu | ManxPower: when I try an outbound call asterisk goed to my incoming channel: http://paste-it.net/k50ed95 |
| 17:39.02 | *** join/#asterisk jsaunders (n=nevermin@70.70.0.33) |
| 17:39.38 | ManxPower | roxlu: then you have a dialplan design problem. |
| 17:40.05 | ManxPower | your incoming calls from untrusted sources should land in one context, your actual phones should be in a different context. |
| 17:41.36 | roxlu | Okay |
| 17:41.42 | ManxPower | On MY systems, I have calls from untrusted sources (like the PSTN or VoIP provider) land in the context [incoming] then route the call in the dialplan from there. |
| 17:42.03 | ManxPower | roxlu: if you wait, I'll see if I can find an example for you |
| 17:42.26 | roxlu | ManxPower: so using Goto to route your incoming (from the default) to [incoming] ? |
| 17:43.34 | ManxPower | roxlu: no, calls land in [incoming] automatically because I have context=incoming on zapata.conf or sip.conf for those channels/providers |
| 17:43.53 | roxlu | ah like that okay |
| 17:43.59 | ManxPower | roxlu: give me about 15 mins to write up an example I can post. |
| 17:44.05 | roxlu | ManxPower: strangely that didn't work for me |
| 17:44.18 | roxlu | I had to put the /##### at the end of the register, and route it in default |
| 17:44.19 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:45.08 | anonymouz666 | _432423[0-4][0-9],1,someapp_here - pattern should work or not? |
| 17:46.10 | anonymouz666 | hehe |
| 17:48.08 | ManxPower | roxlu: http://www.fnords.org/~eric/dialplan-example.txt |
| 17:48.30 | EmleyMoor | Has anyone got an interesting US/CA tollfree number other than 1-800-555-TELL? |
| 17:48.43 | keith4 | sure |
| 17:48.50 | roxlu | ManxPower: thanks |
| 17:48.56 | keith4 | EmleyMoor: 1 800 938 8487 |
| 17:49.07 | ManxPower | anonymouz666: yes, "432423", followed by a single digit between 0 and 4, followed by a single digit 0-9 (which is "X", BTW) |
| 17:49.29 | ManxPower | so that could be _432423[0-4]X,1,Whatever |
| 17:49.36 | EmleyMoor | keith4: Getting a reject on that |
| 17:49.37 | anonymouz666 | yeap |
| 17:49.41 | roxlu | ManxPower: The line for toll-access, that's for outgoing right? |
| 17:49.41 | anonymouz666 | that way works |
| 17:49.49 | anonymouz666 | using [0-9] doesn't. |
| 17:49.52 | ManxPower | roxlu: we may have to customize it for your system |
| 17:50.06 | keith4 | EmleyMoor: booooo, they took it down |
| 17:50.15 | ManxPower | roxlu: toll-access is the context that allows dialing out, that is why the sip.conf sets the SIP devices to be in that context. |
| 17:50.31 | ManxPower | then I use include => to give access to the exten lines for dialing the actual device. |
| 17:50.42 | ManxPower | this allows us to segment what is permitted my who/what |
| 17:50.45 | roxlu | yes, so can I use anything for the Dial(Zap/G1), like: Dial(SIP/1002/${EXTEN:1}) okay |
| 17:51.05 | ManxPower | roxlu: you mean in the [toll-access] context> |
| 17:51.08 | ManxPower | ? |
| 17:51.13 | roxlu | yes |
| 17:51.23 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 17:51.43 | ManxPower | the only exten lines you want in toll-access are exten lines that dial outside the PBX to the PSTN |
| 17:51.59 | keith4 | EmleyMoor: 1 800 938 2548 |
| 17:52.06 | *** join/#asterisk ciphercast (n=cipherca@pool-151-204-79-229.delv.east.verizon.net) |
| 17:52.12 | roxlu | but how do I know which ones those are? (I've got a SIP account at budgetphone.nl) |
| 17:52.21 | clyrrad | <PROTECTED> |
| 17:52.26 | sevard | Does anyone use FWD for 1-800 termination? |
| 17:52.31 | EmleyMoor | Unintelligible |
| 17:52.43 | sevard | I'm trying to program a good dialplan for my sipura 2100 |
| 17:52.52 | EmleyMoor | sevard: I do |
| 17:53.15 | ManxPower | roxlu: paste a Dial line that you use to dial to the PSTN via your provider. |
| 17:53.23 | sevard | EmleyMoor: Can you paste me your dial plan in your line? |
| 17:53.38 | roxlu | ManxPower: I don't know how that line must look like |
| 17:53.54 | ManxPower | roxlu: nobody knows except you and your provider. |
| 17:54.06 | ManxPower | roxlu: you've never been able to send calls to your provider before? |
| 17:54.18 | roxlu | ah, well... my provider doesn't give me the line (we don't offer support for asterisk) |
| 17:54.33 | roxlu | ManxPower: not using asterisk, with only x-lite it works |
| 17:54.49 | ManxPower | Those bastards. put a copy of your sip.conf on pastebin.ca, change ONLY the passwords. |
| 17:55.20 | roxlu | Okay |
| 17:55.26 | EmleyMoor | sevard: What exactly do you want to know about it? I use a one-liner to do it |
| 17:55.38 | [TK]D-Fender | sevard: (x.T|#.T|*.T) |
| 17:55.45 | sevard | EmleyMoor: i'm trying to write a one-liner. My 1800 PSTN termination dosn't work |
| 17:55.57 | ManxPower | roxlu: Remember Asterisk is not really a PBX, it is a PBX toolkit that lets you design your PBX. There are a million ways to do most things in Asterisk. I am teaching you the way *I* do it and I've been using Asterisk for a long time. |
| 17:56.01 | flujan | guys, it is possible to use pickup command with asterisk? |
| 17:56.09 | flujan | where can i specify the groups to use with it? |
| 17:56.09 | roxlu | yes |
| 17:56.10 | sevard | i tried writing them myselves, but maybe their 1800 termination is just broken |
| 17:56.19 | roxlu | I'm really glad you are helping me out |
| 17:56.29 | *** part/#asterisk myiagy (n=myiagy@201.56.109.2) |
| 17:56.38 | Shaun222 | anybody have any experience with other providers (using voicepulse connect) that support in/out callerID name support? also looking for one that supports more than 4 concurrent calls. |
| 17:57.05 | EmleyMoor | sevard: OK - well, you need to catch 1800 etc. ahead of where you catch 1anythingelse... I catch it in my default context as it's free to call... |
| 17:57.25 | ManxPower | roxlu: you don't make me wait very long for answers to my questions, you do not argue with me, you know enough linux to make changes quickly. All that helps me help you. |
| 17:57.33 | sevard | EmleyMoor: I'm used to doing this in asterisk aswell |
| 17:57.41 | sevard | EmleyMoor: but I don't have an * box available |
| 17:57.46 | EmleyMoor | I only know how to do it in asterisk |
| 17:57.51 | ManxPower | roxlu: what is your native language? |
| 17:58.23 | sevard | [TK]D-Fender: that's not working ;\ |
| 17:58.39 | *** join/#asterisk SirWhit (n=sirjames@blk-11-12-158.eastlink.ca) |
| 17:58.54 | roxlu | ManxPower: one sec... have a phonecall :-) |
| 17:59.02 | SirWhit | anyone familiar with the TC400B here? |
| 17:59.20 | ManxPower | roxlu: OK |
| 17:59.31 | sevard | [TK]D-Fender: I plug in that dialplan, try dialing 18004664411 (goog 411), i prepend it with *, or **, or nothing, and I get a busy signal every time |
| 17:59.34 | Qwell | SirWhit: what would you like to know? |
| 18:00.23 | SirWhit | Quell: Just wondering if its possible to put more then one card into a computer.. so instead of 96 G729 lines.. it can handle 184 |
| 18:00.55 | ManxPower | SirWhit: I suspect that is a question for Digium sales. |
| 18:01.01 | flujan | according to the documentation, asterisk can pick up extensions, not channels... |
| 18:01.12 | flujan | it is possible to pickup a SIP extension? |
| 18:01.16 | Qwell | SirWhit: it should be fine, yes, but please do ask Sales for the "official" answer. |
| 18:01.24 | SirWhit | will do .. thanks though... |
| 18:02.30 | ManxPower | flujan: what documentation? |
| 18:02.36 | [TK]D-Fender | sevard: (x.T|#x.T|*x.T) |
| 18:03.20 | flujan | voip-info.org |
| 18:03.33 | flujan | ManxPower: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 18:03.34 | sevard | [TK]D-Fender: that's exactly what I put in |
| 18:03.35 | ManxPower | flujan: that is NOT documentation!!!!!!!!! |
| 18:03.46 | [TK]D-Fender | sevard: I just added "x"'s |
| 18:03.47 | ManxPower | "show application pickup" in the Asterisk CLI. |
| 18:03.50 | sevard | [TK]D-Fender: is this ATA munched if it doesn't understand it? |
| 18:03.58 | flujan | ManxPower: already did it... |
| 18:03.59 | ManxPower | you do not have ANY idea what version of Pickup the wiki is talking about. |
| 18:04.11 | [TK]D-Fender | sevard: Possibly.. I THINK it was right.. one of the 2 should work... |
| 18:04.43 | flujan | ManxPower: it is also saing extension... not the specific sipchannel... |
| 18:05.02 | ManxPower | correct, it says extension so it means extension, not channe |
| 18:05.05 | ManxPower | l |
| 18:05.59 | sevard | [TK]D-Fender: that's not working for me either. |
| 18:06.07 | ManxPower | so you can do an exten => _*XXX,1,Pickup(${EXTEN:1}) but you cannot do Pickup(SIP/1234) |
| 18:06.17 | Shaun222 | [TK]D-Fender: whats the .T for in the dialplan on the polycoms |
| 18:06.53 | [TK]D-Fender | Shaun222: any # of extra digits + timeout |
| 18:07.19 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 18:07.27 | Shaun222 | ah, that would explain why i have that on 9,011xxx.T |
| 18:07.34 | [TK]D-Fender | ManxPower: Strom_M would have a complete shit-fit if he saw that... |
| 18:09.26 | ManxPower | [TK]D-Fender: why is that? |
| 18:09.40 | [TK]D-Fender | ManxPower: You know he's the VSC-Nazi here... |
| 18:09.46 | ManxPower | VSC? |
| 18:09.49 | [TK]D-Fender | ~vsc |
| 18:09.50 | jbot | [vsc] Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 18:10.08 | ManxPower | [TK]D-Fender: Ah! There isn't a VSC for pickup. |
| 18:10.22 | ManxPower | but you and he are right. |
| 18:10.26 | [TK]D-Fender | ManxPower: No, but your "sample" violates all others :) |
| 18:10.39 | ManxPower | IIRC *9X is reserved for local stuff |
| 18:10.43 | [TK]D-Fender | ManxPower: Oh no.. just HIM... I think of them as "suggestions" personally :) |
| 18:11.05 | ManxPower | on production boxes I try very hard to stick to using good VSCs |
| 18:11.31 | ManxPower | We do things like paging using a leading # |
| 18:11.42 | ManxPower | we never require a trailing # for anything, BTW. |
| 18:12.32 | Shaun222 | whats with this one.. [0-1][2-9]xxxxxxxxx\ |
| 18:12.45 | roxlu | ManxPower: sorry, I'm back |
| 18:12.53 | roxlu | I'll read up .. |
| 18:13.20 | roxlu | ManxPower: I'm dutch |
| 18:13.42 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-7a51d5484c32834e) |
| 18:14.00 | ManxPower | roxlu: where were we? |
| 18:14.04 | steve | will FXO products branded as "trixbox compatible" still work on asterisk? |
| 18:14.13 | ManxPower | oh, your sip.conf sans passwords |
| 18:14.17 | roxlu | hmm I was reading your dialplan at: http://www.fnords.org/~eric/dialplan-example.txt |
| 18:14.24 | ManxPower | steve: should be. |
| 18:14.42 | [TK]D-Fender | steve: ... |
| 18:14.45 | [TK]D-Fender | ~trixbox |
| 18:14.45 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 18:14.53 | roxlu | I'm not sure if I need nat=yes for my extension in sip.conf, i've got a install like this: [test-pc]--->[router/asterisk]--->inet |
| 18:15.02 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 18:15.05 | [TK]D-Fender | steve: And the way you're shopping and comparing is sounding cheap & dangerous. |
| 18:15.05 | Shaun2222 | errr |
| 18:15.16 | ManxPower | roxlu: you do not need it at this point. |
| 18:15.25 | ManxPower | we may have to add it latter if we have issues. |
| 18:15.27 | Shaun2222 | got kicked off.. |
| 18:15.32 | Shaun2222 | whats with this one.. [0-1][2-9]xxxxxxxxx |
| 18:15.36 | Shaun2222 | i guess that makes it work if they use 0 or 1 infront of a area+number? |
| 18:16.04 | roxlu | ManxPower: what is that qualify=no flag |
| 18:16.07 | el_critter | hi |
| 18:16.39 | ManxPower | roxlu: just trust me on that. |
| 18:16.49 | roxlu | I'll :D |
| 18:17.32 | ManxPower | now put your sip.conf on pastebin.ca |
| 18:17.46 | roxlu | okay |
| 18:18.39 | ManxPower | without passwords. |
| 18:19.11 | alrs | anyone have any tips for manually editing astdb outside of Asterisk? |
| 18:19.27 | roxlu | http://paste-it.net/i1246da1 <--- there |
| 18:19.48 | ManxPower | this paste is either expired or it never existed at all! |
| 18:19.59 | roxlu | http://paste-it.net/i1246da |
| 18:20.21 | ManxPower | alrs: I believe it is just a Berkely DB1 file. |
| 18:21.26 | alrs | ManxPower: I've never had to edit a Berkeley DB1 file. Is there a commandline utility to do that? |
| 18:21.55 | ManxPower | roxlu: you would want to try something like exten => _X.,1,Dial(SIP/${EXTEN}@31717113433) to call outside the PBX. |
| 18:22.06 | ManxPower | that would be in the [incoming] context |
| 18:22.16 | roxlu | in the incoming? |
| 18:22.20 | ManxPower | alrs: Google is your friend. |
| 18:22.27 | roxlu | ManxPower: not in outgoing_calls ? |
| 18:22.32 | alrs | ManxPower: not really, it takes me to a shitbox oracle page |
| 18:22.32 | ManxPower | roxlu: VERY sorry, I meant the toll-access context. |
| 18:22.54 | roxlu | yes so for that is in one of the [phones] context? |
| 18:23.06 | roxlu | ManxPower: line 39 |
| 18:23.50 | ManxPower | that context is terribly named. |
| 18:24.01 | alrs | ManxPower: http://www.google.com/search?q=edit+berkeley+db&ie=utf-8&oe=utf-8&aq=t&rls=org.debian:en-US:unofficial&client=iceweasel-a |
| 18:24.01 | ManxPower | but yes, in your setup it would be in the phones context |
| 18:24.08 | roxlu | I know :$ (though didn't make it up myself :-) ) |
| 18:24.13 | ManxPower | I still recommend you use my naming of contexts. |
| 18:24.21 | roxlu | okay I'll |
| 18:24.22 | roxlu | one second |
| 18:24.49 | ManxPower | roxlu: just make a backup copy of all your .conf files for asterisk |
| 18:28.26 | dlynes_laptop | Is there a way, whereby I can automatically transfer an incoming call into the parking lot say using ValetParkCall? |
| 18:28.27 | roxlu | ManxPower: is the [extensions] block a 'special' predefined one? |
| 18:28.53 | *** join/#asterisk circas (n=Dominic@atelka.info) |
| 18:28.55 | ManxPower | roxlu: no, but I still recommend it, because all my examples assume that is the context name. |
| 18:29.14 | ManxPower | roxlu: once you get it working with my names, feel free to adapt them to whatever you want them to be. |
| 18:29.17 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-22-211.dsl.chcgil.ameritech.net) |
| 18:29.21 | roxlu | yes |
| 18:29.24 | ManxPower | but at least you will have a working system to start with. |
| 18:29.26 | roxlu | 'm rename now |
| 18:29.33 | jordanb | Is there a way to make asterisk accept collect calls? |
| 18:29.42 | ManxPower | jordanb: it will do so by default. |
| 18:29.57 | jordanb | Hrmm. |
| 18:30.00 | ManxPower | well, it will at least accept the call, a human would have to accept the charges, of course. |
| 18:30.13 | Shaun2222 | other than saving numbers with a 9 infront of them is there a simple way to have my directory get around the pressing 9 before dialing a outside number? |
| 18:30.15 | jordanb | That's what I mean. |
| 18:30.24 | jordanb | Especially through some sort of authentication. |
| 18:30.39 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 18:30.39 | ManxPower | play an audio file saying that you accept the charges. |
| 18:30.42 | roxlu | ManxPower: I'll paste them again |
| 18:30.43 | jordanb | Like when she asks to state your name, if you could type in a code, and then have that validate with asterisk. |
| 18:31.05 | steve | will FXO products branded as "trixbox compatible" still work on asterisk? |
| 18:31.36 | ManxPower | steve: they should since trixbox is just Asterisk with the most bizarre config files you will EVER see. |
| 18:32.15 | roxlu | ManxPower: this is what I have now |
| 18:32.18 | roxlu | http://paste-it.net/v63ffd6 |
| 18:33.06 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:33.07 | ManxPower | roxlu: any context=whatever line in sip.conf MUST match a [whatever] section of extensions.conf |
| 18:33.32 | ManxPower | you have context=incomingcalls in sip.conf with no [incomingcalls] context in extensions.conf |
| 18:33.35 | roxlu | ah.. so the incoming ..... |
| 18:33.47 | ManxPower | THAT is what links the two files and that is the only thing that links the two files. |
| 18:34.00 | roxlu | so like [from-budgetphone] ? |
| 18:34.41 | *** join/#asterisk Boones (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 18:34.46 | ManxPower | roxlu: why limit yourself by calling the context budgetphone? Just a nice generic name so when you get sick of that provider you can change with out everything being very confusing. |
| 18:34.52 | *** join/#asterisk dmangot (n=dmangot@pnapgw.terracottatech.com) |
| 18:34.53 | roxlu | true |
| 18:34.57 | ManxPower | just use my context names and dialplan design. We are wasting time. |
| 18:35.06 | roxlu | so like [from-outside] ? |
| 18:35.12 | ManxPower | I am trying to get your configs into a state that we can try making some calls. |
| 18:35.31 | ManxPower | as I said, use my names now you can change them when I am done helping you. |
| 18:35.40 | roxlu | you didnt' add a sip account for your provider |
| 18:35.56 | roxlu | you only had 100 and 101? |
| 18:35.59 | ManxPower | roxlu: no, but you can use the one you have |
| 18:36.05 | roxlu | like 101 ? |
| 18:36.16 | roxlu | or incoming? |
| 18:36.52 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 18:36.52 | ManxPower | I used generic stuff in my example config. |
| 18:36.52 | roxlu | okay I'm using incoming as context for my provider now |
| 18:36.56 | ManxPower | good, you have the exten => line I gave you on the channel for dialig out? |
| 18:37.04 | ManxPower | dialing out, that is. |
| 18:37.24 | roxlu | I'll replac eit now |
| 18:37.41 | ManxPower | the design of the context relationships is CRITICAL. |
| 18:37.50 | roxlu | ManxPower: this one: exten => _NXXNXXXXXX,1,Dial(Zap/G1/${EXTEN}) ? |
| 18:38.11 | roxlu | I've put it exactly like yours |
| 18:38.29 | roxlu | but that wan't work of course as I don't have a Zap/G1 |
| 18:38.39 | ManxPower | ManxPower: roxlu: you would want to try something like exten => _X.,1,Dial(SIP/${EXTEN}@31717113433) to call outside the PBX. |
| 18:39.13 | roxlu | ah sorry |
| 18:39.35 | roxlu | Okay |
| 18:39.38 | roxlu | I've used that one |
| 18:39.42 | roxlu | but still the same erorr |
| 18:39.56 | roxlu | I'll paste my confs |
| 18:40.05 | clyrrad | <PROTECTED> |
| 18:40.16 | ManxPower | yes, paste your confs and the CLI output of the failed call. |
| 18:40.22 | roxlu | yes |
| 18:42.07 | roxlu | ManxPower: http://paste-it.net/4120 |
| 18:42.10 | clyrrad | anyone have any tips may steer me in the right direction? |
| 18:42.55 | dmangot | clyrrad, is it choppy with the asterisk supplied prompts, or are these custom? |
| 18:43.02 | ManxPower | roxlu: I really cannot edit your configs on paste-it.net |
| 18:43.09 | clyrrad | dmangot: both |
| 18:43.16 | roxlu | ah |
| 18:43.21 | roxlu | do you know another one? |
| 18:43.28 | ManxPower | pastebin.ca |
| 18:43.31 | clyrrad | dmangot: but not with music on hold |
| 18:43.34 | ManxPower | ~pb |
| 18:43.35 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:43.35 | dmangot | clyrrad: the load on the server is pretty low? |
| 18:43.46 | clyrrad | dmangot: yes 4 active channels |
| 18:43.58 | roxlu | ManxPower: there you go: http://pastebin.ca/747056 |
| 18:44.00 | dmangot | clyrrad: what is your cpu? |
| 18:44.04 | _x86_ | are there any decent CDR reporting tools besides areski / asterisk-stat? |
| 18:44.18 | clyrrad | dmangot: its a Xeon, and its stilling at about 3 percent used |
| 18:44.33 | dmangot | mmm |
| 18:44.44 | clyrrad | ? |
| 18:45.03 | dmangot | clyrrad: should be plenty, nothing on the * console? |
| 18:45.04 | [TK]D-Fender | _x86_: Notepad |
| 18:45.16 | clyrrad | dmangot: negative, console looks good |
| 18:45.44 | roxlu | ManxPower: I'll paste my log |
| 18:45.50 | Shaun2222 | [TK]D-Fender: come on now.. notepad... vim :) |
| 18:45.57 | clyrrad | dmangot: its even sound files recording using Asterisks Record() function have the same choppy sound to them |
| 18:46.57 | roxlu | ManxPower: this is one with the CLI log: http://pastebin.ca/747062 |
| 18:46.58 | clyrrad | dmangot: the best adivse I found on Google was to check if the NIC is in Full Duplex which I have verified it is |
| 18:47.11 | dmangot | clyrrad: well it's probably not the network if the MoH files are ok, and the CPU doesn't seem bound, I've only had bad sound when my MoH files weren't in the perfect encoding format, which wouldn't be the case here with system sounds... |
| 18:47.42 | clyrrad | dmangot: right, this was my thought process so far too, I am at a loss |
| 18:47.43 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:47.45 | ManxPower | roxlu: MANY changes: http://pastebin.ca/747064 |
| 18:47.54 | dmangot | dmangot clyrrad: I doubt it's the client having trouble if MoH sounds ok |
| 18:48.08 | ManxPower | roxlu: the order of general and globals is important on some versions of asterisk |
| 18:48.24 | roxlu | ahhhh |
| 18:48.25 | clyrrad | dmangot: yes, you can call in from the PSTN and hear the choppyness |
| 18:48.26 | ManxPower | and you should not generally allow=alaw AND allow=ulaw |
| 18:48.59 | ManxPower | clyrrad: PSTN calls are Zap or SIP or IAX? |
| 18:49.00 | roxlu | okay |
| 18:49.15 | roxlu | I'll replace your paste |
| 18:49.16 | dmangot | clyrrad; do you head the choppiness on the local network over SIP or IAX? |
| 18:49.24 | dmangot | err, hear |
| 18:49.25 | ManxPower | once you make the changes see if you can dial a number and pastebin.ca the CLI output |
| 18:49.30 | clyrrad | ManxPower: PSTN calls are from a regular Land Line to IAX DID |
| 18:49.53 | ManxPower | clyrrad: that eliminates the possibility of a zaptel IRQ issue |
| 18:50.11 | clyrrad | dmangot: no there are no sound issues when your call is going, and extension to extension talking is fine. Its just when it plays back the IVR promps |
| 18:50.17 | roxlu | ManxPower: do you route an incoming phonecall to sip/1002? |
| 18:50.24 | clyrrad | ManxPower: yup....... im confused what could be the issue |
| 18:50.24 | roxlu | ah.. of course |
| 18:50.50 | *** part/#asterisk circas (n=Dominic@atelka.info) |
| 18:51.17 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 18:51.43 | dmangot | I am having the same problem as this guy: http://forums.digium.com/viewtopic.php?t=18560&start=0&postdays=0&postorder=asc&highlight=timestamp+voicemail But of course, the registration on asterisk.org is broken so I can't post on the forums about it |
| 18:52.20 | Qwell | dmangot: how is it broken? |
| 18:52.33 | dmangot | I register but I never get the email with my password |
| 18:52.43 | roxlu | ManxPower: where is the 1002 extension? in extensions.conf (you use Goto(1002) there? |
| 18:53.02 | Qwell | dmangot: interesting - on both counts |
| 18:53.06 | dmangot | Qwell: I've waited over 12 hours but still no show |
| 18:53.37 | ManxPower | roxlu: you can Goto 1002 because we include => extensions |
| 18:53.46 | ManxPower | HOWEVER, I did make a mistake in the example goto. |
| 18:53.48 | roxlu | ahhh |
| 18:53.59 | dmangot | Quell: yeah sux. I upgraded to 1.4.13 from 1.2 and the problem showed up, same configuration. I tried setting TZ in the init.d script but no love |
| 18:54.13 | Qwell | it worked with 1.2? |
| 18:54.25 | ManxPower | roxlu: http://pastebin.ca/747067 notice the ,1 added to the Goto. you MUST always have a priority when using a goto. |
| 18:54.30 | dmangot | Qwell: yeah, worked no problem in 1.2 |
| 18:55.15 | *** join/#asterisk LeRat (n=danoshky@70.55.91.45) |
| 18:55.20 | dlynes_laptop | can you do presence detection on valetpark? |
| 18:55.27 | dmangot | Qwell: looks like from the forum post that it was fine in 1.4.12! |
| 18:55.34 | ManxPower | roxlu: fully understanding contexts is required to be able to understand Asterisk. And contexts are one of the most difficult things to learn, |
| 18:55.43 | roxlu | yes |
| 18:56.03 | *** join/#asterisk hsoj (n=josh@209.223.48.71) |
| 18:56.13 | LeRat | Hi ALL ... anyone knows a good low rates CAN & US trunk provider ?????????????? |
| 18:56.15 | roxlu | you didn't change much for the [1000] extension and [1002] in sip.conf right? |
| 18:56.18 | ManxPower | roxlu: contexts are also the key to Asterisk security. Did you notice the context=INVALID in [general] section of sip.conf? |
| 18:56.20 | hsoj | can anybody provide a worthy term/orig that is tier2? |
| 18:56.31 | hsoj | company that is |
| 18:56.36 | Qwell | dmangot: that's odd, it doesn't look like asterisk adds the time header |
| 18:56.38 | ManxPower | roxlu: only removed allow=ulaw, for those devices, I think. |
| 18:56.40 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 18:57.02 | ManxPower | roxlu: the only time you might want to allow ulaw is if you are using a provider in USA/Canada. |
| 18:57.17 | Qwell | or, maybe it does |
| 18:57.19 | roxlu | ManxPower: I didn't see context=INVALID in sip.conf?? |
| 18:57.23 | ManxPower | not calling numbers in usa/canada, only if you are using a provider located there. |
| 18:57.27 | Egonis | I am trying to dial out via a T1 PRI configured via genzaptelconf, and it automatically creates 96 channels as a result -- when I try to dial out via 'Dial (Zap/1/#####) I get 'Unable to create channel of type: Zap' |
| 18:57.45 | Egonis | obviously I'm doing something wrong, but no clue as to what |
| 18:57.46 | ManxPower | roxlu: sorry, I did not include that. don't worry about it for now. |
| 18:57.47 | roxlu | ManxPower: pff.. still the same error here :( |
| 18:57.55 | ManxPower | roxlu: but we ARE making progress |
| 18:57.55 | dmangot | Qwell: the funny thing is in the body of the message, it has the right time for the incoming VM. But all the timestamps in my postfix logs are correct |
| 18:57.59 | dlynes_laptop | LeRat: try www.calltermination.com |
| 18:58.00 | ManxPower | now pastebin the CLI output. |
| 18:58.29 | LeRat | dlynes_laptop : thanks ! |
| 18:58.35 | ManxPower | Egonis: you did not have zaptel installed when you built asterisk, therefore asterisk dis not build support for Zap channels. |
| 18:58.40 | Qwell | Corydon76-dig: any idea about why the TZ would be off? |
| 18:58.55 | ManxPower | don't ask me how to fix that. it has something to do with rerunning configure or removing some file. |
| 18:58.56 | Egonis | ManxPower: but zap show channels shows all 96 channels, and their contexts |
| 18:59.22 | ManxPower | Egonis: then channel 1 is in use. |
| 18:59.26 | roxlu | ManxPower: Corydon76-dig: any idea about why the TZ would be off? |
| 18:59.30 | roxlu | oh |
| 18:59.35 | ManxPower | or your telco refused the call |
| 18:59.37 | roxlu | http://pastebin.ca/747071 |
| 18:59.57 | Egonis | ManxPower: It's a brand new T1, is it possible that it's not connected to the correct port? |
| 19:00.56 | objective | does srvlookup=yes seriously break anything? |
| 19:00.59 | Egonis | ManxPower: i.e. I have 4 ports on my tor2, and if I'm trying to dial Zap/1 when it should be Zap/72, would this cause the issue? |
| 19:01.09 | ManxPower | roxlu: that is strange. change _X. to _0XXXX |
| 19:01.14 | ManxPower | then issue a RELOAD in the CLI and try again |
| 19:01.17 | ManxPower | .e.r.. |
| 19:01.23 | ManxPower | _0XXXX. |
| 19:01.27 | ManxPower | notice the . at the end |
| 19:01.28 | roxlu | ManxPower: like: diaplan reload |
| 19:01.31 | roxlu | and sip reload? |
| 19:01.35 | ManxPower | Egonis: I really can't help you. |
| 19:01.44 | ManxPower | roxlu: for now just do a "reload" |
| 19:01.52 | Egonis | ManxPower: Okay, thank you anyway for trying |
| 19:02.06 | ManxPower | roxlu: once you have things working then you can issue a reload for just the part you have changed. |
| 19:02.27 | [TK]D-Fender | Egonis: Pastebin your failed calls CLI output. |
| 19:02.30 | roxlu | did it, but again ..... |
| 19:02.36 | ManxPower | roxlu: if it still fails, put the new extensions.conf and the CLI output of the failed call on pastebin.ca |
| 19:02.45 | roxlu | okay |
| 19:03.18 | ManxPower | roxlu: I *think* you may have a problem with your provider, but I need to test more before being sure. |
| 19:03.28 | _x86_ | [TK]D-Fender: you crazy bastard... notepad is not a viable solution to ~120,000 CDR records in a database ;-) |
| 19:03.31 | Qwell | Damin: ping? |
| 19:03.44 | [TK]D-Fender | _x86_: You're right... its too big... Wordpad then :p |
| 19:03.58 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 19:04.35 | ManxPower | roxlu: building a PBX with Asterisk is a very complex task, do not worry. |
| 19:05.00 | roxlu | ManxPower: http://pastebin.ca/747075 |
| 19:05.12 | roxlu | yes, though, incoming is already working :-) |
| 19:06.11 | roxlu | I didn't remove the bindaddr in sip.conf |
| 19:06.19 | *** join/#asterisk BBHoss (n=hoss@proxy-srv.uah.edu) |
| 19:06.21 | roxlu | but I don't think that could cause this? |
| 19:06.42 | ManxPower | roxlu: it could. |
| 19:07.00 | ManxPower | I am going to try something you may think is strange. standby for a pastebin |
| 19:07.33 | roxlu | :D |
| 19:07.47 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 19:09.05 | dmangot | Qwell: could it be related to this? 2007-09-17 20:16 +0000 [r82594-82676] Russell Bryant <russell@digium.com> |
| 19:09.06 | dmangot | <PROTECTED> |
| 19:09.06 | dmangot | <PROTECTED> |
| 19:09.06 | dmangot | <PROTECTED> |
| 19:09.06 | dmangot | <PROTECTED> |
| 19:10.23 | russellb | hmmm? |
| 19:10.38 | *** part/#asterisk hsoj (n=josh@209.223.48.71) |
| 19:12.52 | ManxPower | roxlu: changes to sip.conf. I split the budgetphone entry into a type=peer and type=user http://pastebin.ca/747084 |
| 19:13.06 | ManxPower | also extensions.conf dial line change http://pastebin.ca/747079 |
| 19:13.28 | ManxPower | It is normal practice to split provider entries into type=peer and type=user |
| 19:13.46 | roxlu | okay |
| 19:14.01 | ManxPower | roxlu: for some reason the call IS going to the provider and the provider is sending the call BACK to you. |
| 19:14.21 | roxlu | where do you see that? (i mean where in hte CLI output) |
| 19:15.09 | ManxPower | roxlu: I don't have it on my screen, but the got nnn back from..... is the line |
| 19:15.15 | ManxPower | the 1st or 2nd line after the Dial |
| 19:15.19 | roxlu | ah okay |
| 19:15.24 | roxlu | i'm now changing my config |
| 19:15.39 | Egonis | I am suddenly receiving a device or resource busy message when doing a 'module load chan_zap.so' -- how do I find out what is hooking it? |
| 19:15.49 | ManxPower | roxlu: we should test outgoing first, once we get that working, we can fix any new issues with incoming calls |
| 19:15.59 | ManxPower | Egonis: "show channels" |
| 19:16.41 | roxlu | ManxPower: okay, did you change something with the [1000] or [1002] ? |
| 19:16.47 | ManxPower | roxlu: no. |
| 19:17.44 | dmangot | russellb: I am having the same problem after upgrading to 1.4.13 as this guy: http://forums.digium.com/viewtopic.php?t=18560&start=0&postdays=0&postorder=asc&highlight=timestamp+voicemail |
| 19:17.50 | dlynes_laptop | Is there a way to blf parked calls that have been parked using valetparkcall? |
| 19:18.06 | roxlu | Still the same :( |
| 19:18.22 | ManxPower | show me the cli output on a pastebin. |
| 19:18.23 | roxlu | Thoug I see: Back from 17.19.3.1 ? |
| 19:18.33 | ManxPower | roxlu: yes. |
| 19:19.08 | ManxPower | I think I may see the issue. |
| 19:19.15 | ManxPower | I'll look at the CLI output |
| 19:19.23 | roxlu | There: http://pastebin.ca/747090 |
| 19:21.26 | Egonis | How do I dial out through a PRI? e.g. I have 96 channels over 4 spans, and have one T1 cable plugged in -- what do I do to test it? (yes, slap me for being an idiot) |
| 19:22.11 | [TK]D-Fender | Egonis: Have you considering trying to place a call with it/to it? |
| 19:22.12 | alrs | [TK]D-Fender: that's a bit over-the-top |
| 19:22.18 | ManxPower | roxlu: in sip.conf change svrloookup=yes to svrloookup=no and host=budgetphone.nl to host=sip.budgetphone.nl |
| 19:22.34 | Egonis | [TK]D-Fender: I want to call through it, not into it from the outside world. It's intended to be outbound only |
| 19:22.38 | *** join/#asterisk krondorl (n=chatzill@tfi1meg.1meg.golden.net) |
| 19:22.47 | roxlu | okay |
| 19:22.50 | krondorl | nice bitch slap [TK]D-Fender |
| 19:22.51 | tripps | what is the general meaning of cause "no authority found" cause code: 50 in terms of iax2 debug messages? |
| 19:22.53 | [TK]D-Fender | Egonis: Then by all means dial :) |
| 19:23.07 | Egonis | [TK]D-Fender: I am trying Dial(Zap/G2/thenumber) -- G2 is channels 1-23 |
| 19:23.21 | roxlu | ManxPower: ahh a bit more now.... |
| 19:23.22 | [TK]D-Fender | Egonis: Sounds like a decent start.... |
| 19:23.32 | Egonis | [TK]D-Fender: The resulting message is: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 19:23.32 | ManxPower | roxlu: pastebin the CLI output. |
| 19:23.35 | roxlu | yes |
| 19:23.38 | [TK]D-Fender | Egonis: so <drphil> Hows that working out for you? |
| 19:23.48 | alrs | Egonis: what does pri debug span 1 say? |
| 19:23.53 | Egonis | [TK]D-Fender: However, zap show channels shows 96 channels |
| 19:24.00 | Egonis | alrs: how do I do that? this is new to me |
| 19:24.04 | alrs | Egonis: up, active, etc? |
| 19:24.06 | [TK]D-Fender | Egonis: pastebin your zaptel & zapata |
| 19:24.20 | alrs | Egonis: better yet, just pri show span 1 |
| 19:24.21 | [TK]D-Fender | Egonis: And the full CLI oputput of the failed call. |
| 19:24.46 | roxlu | ManxPower: http://pastebin.ca/747096 |
| 19:24.56 | Egonis | [TK]D-Fender: Doing so now, and alrs, I have ran pri debug span 1, and it is repeating the messages 'Sending Set Asynchronous Balanced Mode Extended' |
| 19:25.52 | *** join/#asterisk clive- (n=pirch@dsl-242-170-00.telkomadsl.co.za) |
| 19:25.58 | ManxPower | roxlu: that looks like a password/secret problem |
| 19:26.17 | Egonis | alrs: what should the usual output be from a pri debug span 1? |
| 19:26.34 | ManxPower | if your secret= line is correct in sip.conf, then put a full copy of your sip.conf on pastebin to make sure you have all the changes I gave you |
| 19:27.10 | Egonis | alrs: pri show spans results in 'PRI span 1/0: Provisioned, In Alarm, Down, Active -- for all four |
| 19:27.34 | ManxPower | Egonis: in alarm means "line not working" |
| 19:28.01 | Egonis | ManxPower: so the fact that all four are 'In Alarm' means that there's an issue with the carrier? |
| 19:28.23 | ManxPower | Egonis: or a cable issue |
| 19:28.27 | alrs | Egonis: If it's in Alarm it should show as RED or YEL in zttool |
| 19:29.22 | Egonis | alrs: How do I compile zttool? When I make menuconfig zaptel-1.4.5 zttool has 'XXX' beside it |
| 19:30.05 | ManxPower | Egonis: do "cat /proc/zaptel/1" at the command prompt (not asterisk CLI) |
| 19:30.10 | alrs | Egonis: I've never had any need to compile Zaptel. |
| 19:30.14 | ManxPower | that will give you the info zttool would give you. |
| 19:30.28 | ManxPower | Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF |
| 19:30.33 | ManxPower | that would show red or yellow, I think |
| 19:31.59 | roxlu | ManxPower: when I change the host from sip.bu... to only budgetphone.nl I get the loop error again. |
| 19:32.06 | roxlu | ManxPower: the passwords are correct |
| 19:32.29 | tripps | i have 2 * servers connected and registered to each other via IAX2. Let's call them office and remote. I have SIP phones registered with office. I wish to have dialplan such that _X555 from SIP phones call SIP phones registered with remote all over IAX2 trunk. can someone tell me what i'm doing wrong? ultimately I want to actually dial _X. through the IAX2 trunk and out the other side |
| 19:32.35 | ManxPower | roxlu: they have something weird in their setup. that is why I specified sip.budgetphone.nl to try to work around that issue |
| 19:32.52 | roxlu | yes |
| 19:33.02 | roxlu | maybe I need to add budgetphone in my /etc/hosts |
| 19:33.18 | ManxPower | roxlu: I doubt that will fix it. |
| 19:33.24 | roxlu | ok |
| 19:33.29 | ManxPower | roxlu: paste the output of "sip show peers" |
| 19:33.51 | ManxPower | and the output of "sip show registry" |
| 19:34.07 | ManxPower | roxlu: now we are moving beyond configuration into real troubleshooting. |
| 19:34.11 | tripps | i have a dialplan for office so exten -> _5XXX,n,Dial( |
| 19:34.28 | tripps | i have a dialplan for office so exten -> _5XXX,n,Dial(IAX2/remote/${EXTEN}) |
| 19:34.35 | tripps | => i mean |
| 19:34.52 | tripps | do I need to set up a dialplan on the remote site? |
| 19:35.04 | *** join/#asterisk guillote_GNU (n=bancaria@host127.190-30-104.telecom.net.ar) |
| 19:35.14 | ManxPower | tripps: yes. |
| 19:37.10 | krondorl | [TK]D-Fender and alrs: the pastebin for egonis is at: http://pastebin.com/d8d4dddc |
| 19:37.33 | tripps | ManxPower: ok great - what would that dialplan look like? |
| 19:38.04 | ManxPower | tripps: I'm already helping someone, you are on your own for complicated stuff like that. |
| 19:38.40 | roxlu | :P |
| 19:39.07 | roxlu | ManxPower: i'm pasting it now |
| 19:39.11 | tripps | ManxPower: ok - tell me if it's an easy Dial() type deal or if I have to dig deeper to bridge the IAX2 trunk with the SIP outbound |
| 19:40.05 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 19:40.05 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) [NETSPLIT VICTIM] |
| 19:40.09 | roxlu | ManxPower: http://paste-it.net/xc1291f |
| 19:40.16 | ManxPower | tripps: you need to set the remote side to accept the call (sip.conf) then handle the call (extensions.conf) |
| 19:40.29 | tripps | ManxPower: k thanks! |
| 19:40.34 | [TK]D-Fender | Egonis: pastebin "zap show channels" , " pri show span1" , and confirm that you are indeed using NI2 (national) signalling for your PRI with your provider. I also highly doubt you need any LBO settings in your zaptel.conf |
| 19:41.16 | ManxPower | roxlu: that looks good "sip.budgetphone.nl:5060 31717111111@ 105 Registered Tue, 23 Oct 2007 21:35:16" |
| 19:41.34 | roxlu | yes |
| 19:42.00 | ManxPower | roxlu: OK, in the CLI do a "sip debug peer budgetphone" and try a call and paste the CLI output. there will be much CLI output. |
| 19:42.12 | roxlu | okay |
| 19:42.31 | roxlu | oh it's deprecated |
| 19:42.34 | CoffeeIV_ | my asterisk is not setting the callerid correctly on outbound calls. If I print out CALLERID(num) right before the Dial command, it is correct, but when the call is received the number shows up incorrectly. The call is coming in from another asterisk via IAX2, and it is leaving on a PRI line. Any ideas of how I can make the caller id go through correctly ? |
| 19:43.50 | *** join/#asterisk blq (n=Bl@dslb-088-064-143-231.pools.arcor-ip.net) |
| 19:44.01 | roxlu | ManxPower: chan_sip.c:12298 handle_response_register: Outbound Registration: Expiry for sip.budgetphone.nl is 120 sec (Scheduling reregistration in 105 s) |
| 19:46.25 | roxlu | ManxPower: here is a piece of the CLI: http://paste-it.net/u51a787 |
| 19:49.32 | ManxPower | roxlu: add "fromdomain=budgetphone.nl" back into the [budgetphone] section of sip.conf. and reload and retry the call. |
| 19:50.06 | roxlu | okay |
| 19:51.21 | nestAr | anyone got a second to look at a Set(GROUP) example that doesn't seem to work for me? |
| 19:51.30 | Qwell | Strom_M: ping |
| 19:51.44 | Strom_M | Qwell: pong |
| 19:52.00 | roxlu | ManxPower: YESSSSSSSSSSSSSSSSSS!!!!!! |
| 19:52.04 | roxlu | ManxPower++ |
| 19:52.10 | Qwell | Strom_M: on jabber? |
| 19:52.21 | Strom_M | Qwell: not at the moment |
| 19:52.27 | ManxPower | roxlu: now test incoming calls |
| 19:52.49 | roxlu | ManxPower: yep!!! |
| 19:52.59 | ManxPower | roxlu: Goot! |
| 19:53.00 | roxlu | ManxPower: you rule!! you made me and a lot of other people happy! |
| 19:53.12 | roxlu | The company couldn't get it to work either :$ |
| 19:53.15 | roxlu | budgetphone.nl |
| 19:53.16 | ManxPower | roxlu: now MAKE A BACKUP COPY OF extensions.conf, sip.conf |
| 19:53.25 | roxlu | I WILL definitely!! |
| 19:53.45 | ManxPower | you WILL break it and it will be good to have working examples to refer to as you make changes to Asterisk |
| 19:53.55 | roxlu | I'll gonna make an howto for all other budgetphone.nl useres tomorrow, are you still there tomorrow? |
| 19:54.14 | *** join/#asterisk blq (n=Bl@dslb-088-064-143-231.pools.arcor-ip.net) |
| 19:54.27 | ManxPower | roxlu: You are welcome to send a donation to me via paypal. I should be online tomorrow, but I cannot guarantee. |
| 19:54.45 | roxlu | :-) |
| 19:55.01 | ManxPower | roxlu: There are many ways we could have setup your sip.conf, other users may be confused by the way I did it. But it IS working. 8-) |
| 19:55.10 | *** join/#asterisk clive-- (n=pirch@dsl-242-181-70.telkomadsl.co.za) |
| 19:55.12 | roxlu | yes |
| 19:55.30 | roxlu | though the hard part was the [budgetphone] separation |
| 19:55.39 | roxlu | wny was that btw? |
| 19:55.45 | clive-- | ~seen waverly360 |
| 19:56.12 | jbot | waverly360 <n=waverly@ns2.dalcon.com> was last seen on IRC in channel #asterisk, 36d 5h 47m 50s ago, saying: 'JerJer: Hmm...I don't have one...might have something else..will look around.'. |
| 19:56.13 | ManxPower | Just remember that in sip.conf your phones should be in a context that ONLY has exten lines for dialing outside the PBX and an include => the context the exten lines for the phones are in. |
| 19:56.33 | roxlu | okay |
| 19:56.43 | ManxPower | roxlu: I'm sure we could have used type=friend, but I don't like doing that for gateways -- it always seems to cause me problems eventually. |
| 19:57.01 | *** join/#asterisk Doodluv (n=brad@24.214.206.158) |
| 19:57.01 | roxlu | okay |
| 19:57.14 | ManxPower | roxlu: none of the providers I have used required a fromdomain= set |
| 19:57.20 | roxlu | ManxPower: ... and one thing which you probably know ... how can I directly start incoming/outgoing calls? |
| 19:57.36 | roxlu | ManxPower: I heard budgetphone is very difficult to configure |
| 19:57.40 | ManxPower | what do you mean by "directly start" |
| 19:58.05 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 19:58.07 | roxlu | well, I want to record my phonecalls |
| 19:59.11 | ManxPower | roxlu: I don't record phone calls so I cannot give you much help with it. Check the wiki for examples, but remember the Wiki has much incorrect information |
| 19:59.29 | ManxPower | the extensions.conf application is "monitor" or "mixmonitor". |
| 19:59.40 | roxlu | okay |
| 19:59.45 | roxlu | but really again, thanks a lot! |
| 19:59.53 | ManxPower | I would suggest you fully set up your pbx before you try something like recording. |
| 19:59.53 | roxlu | I've been working/testing for the last 5days |
| 19:59.55 | ManxPower | you are welcome |
| 20:02.45 | tripps | ManxPower: when you say that the remote server needs entries in sip.conf and extensions.conf, note I have those entries as it relates to the existing sip endpoints on the remote end which work just fine. would those suffice or do i need to add extra entries to handle the incoming iax2 --> sip hand off other than whats already there and the new stuff in iax2.conf files? |
| 20:04.16 | roxlu | ManxPower: when I want to make a friend of my part of the 'phone network' (or how you call it), is that possible? (he is not on the same network though) |
| 20:04.20 | Alan_Hicks | Whoa! "make samples" installs a LOT of configuration files. |
| 20:04.46 | ManxPower | roxlu: if you see me tomorrow we can talk about it. I must go do errands now. |
| 20:04.54 | roxlu | okay |
| 20:05.03 | roxlu | cu later than! |
| 20:06.13 | clive-- | Hi, can anyone give me some pointers regarding fastagi ? |
| 20:06.23 | Doodluv | We have a fully functional 80+ phone * system in use at our healthcare facility. One complaint I have been receiving (mainly from grouchy docs who hate learning new things like dialing phone numbers) is that when we press 9 to get an outside line you hear no dailtone. Is there a way to make * give you a dial tone when you press 9? If so, could somebody direct me to a tutorial or documentation that may hint as to how this is done? |
| 20:07.40 | Shaun2222 | any of you logo'd up your polycom phones? |
| 20:08.08 | [TK]D-Fender | Doodluv: Not cleanly. You could always remove that silly prefix from your dilaplan entirely and save yourself the trouble |
| 20:08.45 | Doodluv | hmm ok...I was afraid that may be the answer. |
| 20:08.59 | [TK]D-Fender | Doodluv: One factor is the phones you're using... what are they? |
| 20:09.20 | tripps | Doodluv: ignorepat => 9 if you're not using voip phones |
| 20:09.21 | Doodluv | polycom phones |
| 20:09.47 | clive-- | seems like fastAGI is not well used .... oh well |
| 20:09.49 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 20:09.54 | tripps | polycoms allow you do have the secondary tone in the internal dialplan I think but not sure |
| 20:09.56 | [TK]D-Fender | Doodluv: Ok, then we CAN fix that.... go download SIP 2.2.0 and upgrade all your phones. After this you'll be able to mod the Polycom dialplan to continue dialtone internally. |
| 20:10.08 | Doodluv | ahhhhhhh sweet |
| 20:10.42 | Doodluv | [TK]D-Fender great, thanks I will give that a shot. |
| 20:12.59 | *** join/#asterisk irule (n=irule@201.151.52.150) |
| 20:13.16 | irule | hi, where may I get the second edition book? |
| 20:13.40 | styelz | amazon |
| 20:14.12 | irule | I thought it was a creative commons license |
| 20:14.40 | *** join/#asterisk Shmattie (n=Shmattie@cpe-75-179-191-147.woh.res.rr.com) |
| 20:15.14 | styelz | there is a free pdf version |
| 20:15.27 | styelz | books cost money though |
| 20:15.34 | [TK]D-Fender | ~book |
| 20:15.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:16.02 | *** join/#asterisk whist (n=whistler@71-81-91-121.dhcp.stls.mo.charter.com) |
| 20:16.20 | Doodluv | arrrgh gotta be a reseller to download the sip 2.2.0! |
| 20:16.32 | peanut- | to downwhatnow? |
| 20:17.07 | Doodluv | well....loooking at the polycom site....since I have polycom phones.... |
| 20:17.08 | roxlu | Does someone know if there is a way to keep a central phonebook? |
| 20:17.14 | Doodluv | maybe going about this the wrong way. |
| 20:17.17 | *** part/#asterisk Strom_M (n=strom@208.127.172.112) |
| 20:18.39 | [TK]D-Fender | Doodluv: Correct, you should be contacting YOUR reseller |
| 20:18.45 | Doodluv | ok. |
| 20:21.05 | Shmattie | Anyone know how to restrict a max of only 1 inbound call to a sip phone? |
| 20:22.09 | nestAr | Shmattie: you doing call queues? |
| 20:23.11 | Shmattie | nestAr: Not in this application. |
| 20:23.51 | nestAr | Oh, well, I am trying to basically do the same thing in a call queue, but it's not working for me either.. |
| 20:23.58 | Shmattie | NestAr: Perhaps I should rephase my goal. If the person is on the phone, I don't want another call to ring at there phone. They should be able to make outbound calls or transfers |
| 20:24.03 | etfonhomey | I'm a reseller. |
| 20:24.24 | *** join/#asterisk guillote_GNU (n=bancaria@host127.190-30-104.telecom.net.ar) |
| 20:25.00 | jcanfield | umm...getting the firmware is quite easy if you modify the d/l link. Shhh. |
| 20:25.53 | nestAr | Shmattie: basically, i want to do the same thing, i used an example found on voip-info, but it didn't seem to work the way i thought it should. |
| 20:26.09 | *** join/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 20:26.19 | roxlu | is it possible (maybe a bit strange) to just ring a phone only once and than hangup ? |
| 20:26.21 | Shmattie | NestAr: Did you try it with Group_count? |
| 20:26.22 | nny | anyone here good with aastra hardware? |
| 20:26.26 | nny | in particular the 480i CT |
| 20:26.43 | nestAr | Shmattie: yes, i will pastebin it, hold on. |
| 20:26.54 | Shmattie | NestAr: I have used that. |
| 20:27.10 | nny | trying to see if there is a way to have the base station ring on a 480i CT when the handset is in use |
| 20:27.12 | dlynes_laptop | I've got a locked mutex, and I'm wondering how to clear it, short of 'killall -9 asterisk'? I've got a pastebin at http://pastebin.ca/747182 |
| 20:27.43 | Shmattie | NestAr: I don't really like that solution because later if I use queues or other applications, I don't think it will respect the group_count. |
| 20:28.07 | roxlu | mvanbaak: are you there? |
| 20:28.40 | nestAr | Shmattie: well, the example was for queues, according to the page.. |
| 20:28.42 | nestAr | http://www.pastebin.ca/747185 |
| 20:28.51 | nestAr | http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent |
| 20:32.00 | Shmattie | NestAr: That is similar to how I have tried it. I also tried it using the new dev_state function but it seems to only work with IAX phones. |
| 20:32.06 | CoffeeIV_ | I'm updating some old dialplans to work on newer asterisk, and the newer stuff doesn't have SetGroup -- what replaced that application ? |
| 20:32.27 | nestAr | Shmattie: that sucks. |
| 20:32.34 | CoffeeIV_ | sorry, it's actually CheckGroup I need -- I figured out Set(GROUP= ) |
| 20:32.37 | nestAr | I need to figure out something to work. |
| 20:33.12 | Shmattie | nestAr: Why doesn't the normal operation for queues work for you? |
| 20:33.22 | dlynes_laptop | CoffeeIV_: Not in 1.2.x: The CheckGroup application has been deprecated, please use a combination of the GotoIf application and the GROUP_COUNT() function, example: |
| 20:33.26 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 20:33.32 | dlynes_laptop | CoffeeIV_: that's straight from the wiki |
| 20:33.48 | CoffeeIV_ | yeah, I found it |
| 20:33.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:35.21 | nestAr | Shmattie: I get call waiting in the ear |
| 20:36.12 | dlynes_laptop | nny: I'm somewhat good with Aastra...but never used the 480i's or the 480iCT's |
| 20:36.27 | dlynes_laptop | nny: only the 9112i, 9133i, and the 57i/560M |
| 20:36.40 | Shmattie | nestAr: You get call waiting for an agent that is currently on the phone with a customer and then an internal employee calls that agent? |
| 20:37.01 | dlynes_laptop | nny: but the handsets and the 480i CT's all act like the same phone? |
| 20:37.44 | nestAr | well, i'm doing AddQueueMember, if the agent takes one call, and then another caller comes in, it beeps callwaiting, so i want to limit the calls only for queue calls. |
| 20:40.13 | Shmattie | nestAr: I didn't realize it would beep for callwaiting in that situation. I don't use queues so I didn't know that is the standard behavior. |
| 20:41.03 | Shaun2222 | any of you logo'd up your polycom phones? |
| 20:41.38 | mvanbaak | roxlu: I'm here now |
| 20:42.02 | nestAr | Shmattie: it's a real pain in the as |
| 20:42.04 | nestAr | ass |
| 20:43.04 | Shmattie | NestAr: What type of phones are you using and what version of Asterisk? |
| 20:43.34 | nestAr | Polycom Ip550 and 1.4.x |
| 20:43.48 | nestAr | Asterisk 1.4.13 built by root @ mecca on a x86_64 running Linux on 2007-10-19 11:11:33 UTC |
| 20:44.11 | Alan_Hicks | Anyone know of any good MOH tunes, preferably Creative Commons or similar licensing? |
| 20:44.43 | dlynes_laptop | I've got a locked mutex, and I'm wondering how to clear it, short of 'killall -9 asterisk safe_asterisk'? I've got a pastebin at http://pastebin.ca/747182 |
| 20:45.01 | dlynes_laptop | It happened when I had no calls, and I issued the 'stop when convenient' command |
| 20:45.26 | dlynes_laptop | It's basically preventing asterisk from exiting to a command prompt |
| 20:45.29 | Shaun2222 | nestAr: i've going tohave that same problem here soon, i figured i would probably have to have a seperate context for queued calls which checked the channel to see if it was in use before sending the call |
| 20:46.03 | dlynes_laptop | Or maybe some way of preventing it from happening in the future? |
| 20:46.04 | nestAr | Shaun2222: i have that, but i'm too stupid to make it work. |
| 20:46.12 | nestAr | i must be missing something. |
| 20:46.45 | Shaun2222 | sound be fairly simple with ChanIsAvail() and check checking the status variable that gets set i would think... |
| 20:46.48 | Shaun2222 | i havnt tackled it yet |
| 20:47.31 | Shaun2222 | right now i've moved onto more important things... like trying to load a bitmap on my display on the 550 phones... figured it would be fun and easy... what was i thinking... |
| 20:47.35 | Shaun2222 | :) |
| 20:49.13 | nestAr | lol |
| 20:49.45 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:50.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net) |
| 20:50.55 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:51.36 | Shmattie | NestAr: I agree with Shaun2222, that ChanIsAvail() should do the trick. I will write up some code now and try to test it. |
| 20:51.57 | nestAr | i will be happy to test it out. |
| 20:55.37 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:56.28 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:57.58 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-170-112.red.bezeqint.net) |
| 20:58.19 | Dovid | Hi room. i have a client that complained that he is not getting the number dialed in the from header in the sip invite. |
| 20:58.36 | Dovid | is this the solution:http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header ? |
| 20:58.44 | *** join/#asterisk mugawuki (n=mugawuki@extranet.lehighgas.com) |
| 21:02.32 | Shmattie | NestAr: Here is some info from the voip-info.org wiki |
| 21:02.41 | Shmattie | Nestar: "So: If you want to use ChanIsAvail to determine whether the SIP peer is known and registered, it will work fine. If you want to use it for limiting simultaneous calls to the peer, it will not work reliably for you. " |
| 21:03.43 | Shmattie | nestAr: I am still going to test it out more to see how reliable it is or isn't |
| 21:05.46 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:07.45 | Shmattie | nestAr: Using an Aastra 55i, call I get is a status of 0 which means unknown. I get this when the phone is in use and when it is not in use. |
| 21:07.56 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 21:10.25 | roxlu | where can I define my default language and check if there are any sounds for that? |
| 21:10.29 | Shaun2222 | i think if you specify the s option it chanisavail will always show the status as unavailible if the phone is in use |
| 21:10.47 | *** join/#asterisk Speedy2 (n=Javier_6@cpe-66-75-4-134.san.res.rr.com) |
| 21:11.21 | Speedy2 | Has anyone used Asterisk + PC sound card in full-duplex as an "FXS"? |
| 21:11.30 | marc7 | has anybody here had any success flashing a 7970 to SIP firmware? |
| 21:12.19 | nestAr | Shmattie: that sucks |
| 21:12.53 | Shaun2222 | marc7: cisco phone? think i used to use 7960's |
| 21:12.58 | Shaun2222 | while ago... though |
| 21:13.33 | marc7 | Shaun2222: yeah, the 7970's are drastically different... java-based platform with an XML configuration file... and I've been having trouble getting it to recognize the new firmware... |
| 21:13.50 | marc7 | the voip-info.org wiki article is a bunch of garbage |
| 21:14.30 | [TK]D-Fender | marc7, Poor choice of phone. |
| 21:14.48 | tzafrir | Speedy2, you can. It's just not so convinient an interface |
| 21:15.06 | [TK]D-Fender | Speedy2, no, FX involves powering a PHONE. What you're describing sounds more like a SOFTPHONE. |
| 21:15.12 | [TK]D-Fender | FXS* |
| 21:16.00 | marc7 | [TK]D-Fender: come on now... you trashed our choice of getting the linksys SPA-962's so we halted our order and sent them all back... this is one I can't send back, and we need a good color phone that works. telling me i bought a shitty phone isn't super helpful |
| 21:16.00 | Speedy2 | [TK]D-Fender: You're correct, I should have said "soft phone" |
| 21:16.42 | Shaun2222 | marc7: java based... no wonder :) |
| 21:17.01 | [TK]D-Fender | marc7, I never said you could be helped, and the SPA would have been a better choice in all likelyhood. |
| 21:17.03 | Speedy2 | tzafrir: Is it possible to get Asterisk to do DTMF recognition on the incoming sound? I guess normally a Zaptel device does the DTMF encoding/decoding, etc. |
| 21:17.05 | [TK]D-Fender | marc7, Oh well. |
| 21:17.21 | [TK]D-Fender | Speedy2, Huh? |
| 21:17.25 | marc7 | the SPA was a pretty lousy phone, we did manage to try them out |
| 21:18.06 | [TK]D-Fender | marc7, lousy in what way, and why this race for colour? |
| 21:18.07 | tzafrir | Speedy2, how exactly do you expect to connect to the PSTN? |
| 21:18.13 | fujin | Haven't had any issue with the spa962/942's here.. |
| 21:18.14 | jordanb | I'm wanting to get a SIP phone. |
| 21:18.19 | Speedy2 | tzafrir: No PSTN, just need to hook to a real telephone |
| 21:18.20 | fujin | Probably better than the Cisco ones |
| 21:18.25 | jordanb | Well, I'm thinking a PDA with a SIP client. |
| 21:18.27 | jordanb | Or something. |
| 21:18.33 | nny | dlynes_laptop: sorry was on phone |
| 21:18.34 | tzafrir | you can dial to the sound card channel directly (with the "dial" command) |
| 21:18.36 | Speedy2 | tzafrir: I can build the hybrid to go to/from soundcard to the phone |
| 21:18.36 | jordanb | I don't want cellphone service |
| 21:18.45 | jordanb | But I want to be able to SSH from it as well as use SIP. |
| 21:18.45 | [TK]D-Fender | Speedy2, Go but an SPA-2102 then |
| 21:19.07 | fujin | jordanb: My TyTN has a sip client, and there's a cab available for pocketputty |
| 21:19.12 | fujin | which has been working super |
| 21:19.12 | Speedy2 | [TK]D-Fender: Since Linksys bought Sipura, the support has been terrible. I have friends with SPA-1001 and latest firmware breaks things, Linksys hasn't been responsive. |
| 21:19.21 | nny | dlynes_laptop: yeah the handset and basestation are essentially the same phone, just you can use line 1 on the handset and line 2 on the phone |
| 21:19.23 | fujin | although sip over UMTS probably isn't such a great idea; |
| 21:19.38 | jordanb | fujin, I'd like to avoid putty if possible. I've been looking at the Nokia E61 as it has SIP and Putty. |
| 21:19.42 | [TK]D-Fender | Speedy2, downgrade. |
| 21:19.49 | Shaun2222 | fujin: does the sip client work that well over the cell phones shitty internet? |
| 21:19.50 | jordanb | I might wait for the openmoko but it doesn't have a keyboard. |
| 21:19.53 | [TK]D-Fender | Speedy2, I've never seen a need to upgrade personally. |
| 21:20.07 | fujin | Shaun2222: in HSDPA areas, yes, it works fine |
| 21:20.08 | Shaun2222 | i have the cingular 8125 and seen sip clients for it but i cant imagine it would work all that good |
| 21:20.09 | Speedy2 | [TK]D-Fender: Which Sipura do you personal use? |
| 21:20.15 | Speedy2 | personally. |
| 21:20.30 | jordanb | Cingular doesn't make phones. |
| 21:20.33 | [TK]D-Fender | Speedy2, I've have 2000,3000, 3201, 1001 |
| 21:20.46 | Shaun2222 | hell ssh sucks over it when it comes to it being responsive.. |
| 21:20.49 | jordanb | I have a Sipura 3102 that I couldn't get the FXO port to work on. |
| 21:20.50 | fujin | jordanb: I'm not sure what SIP/SSH software the E61 has, although you wouldn't catch me dead running a Symbian fun. |
| 21:20.54 | jordanb | Peice of shit. |
| 21:20.58 | fujin | s/fun/phone/ |
| 21:21.00 | fujin | rhgm |
| 21:21.01 | jordanb | I eventually bought a TDM400P. |
| 21:21.02 | Speedy2 | [TK]D-Fender: Do you have a prefernece on which device works best? |
| 21:21.24 | jordanb | jbot, Yeah that's the problem. |
| 21:21.30 | [TK]D-Fender | Speedy2, All seemed the same to me. |
| 21:21.38 | jordanb | I've thought about a Zaurus 6000. |
| 21:21.53 | jordanb | But I don't know if its sound setup would be any good for using it as a phone. |
| 21:22.00 | jordanb | Also it's still pretty expensive. |
| 21:22.02 | Shaun2222 | man this polycom takes a year to boot with the 2.2.0.0047 |
| 21:22.14 | roxlu | What is the difference with Record() and "One touch recording" ? |
| 21:22.39 | [TK]D-Fender | Shaun2222, Then you're doing something very wrong. |
| 21:22.59 | [TK]D-Fender | roxlu, go READ. Its night & day between them. Completely different purposes |
| 21:23.03 | Shaun2222 | [TK]D-Fender: how long should it take? |
| 21:23.21 | roxlu | oh okay |
| 21:23.39 | [TK]D-Fender | Shaun2222, 1:45 on an IP 501 |
| 21:24.14 | Shaun2222 | 550 took about 247 seconds from the looks of the app log |
| 21:24.29 | Shaun2222 | actaully wait |
| 21:27.00 | Shaun2222 | some of this might have to do with spanning tree... |
| 21:27.18 | Shaun2222 | should only be another 30 seconds though. |
| 21:27.22 | mcab | Shaun2222: change your "DHCP Menu\DHCP <mumble>" setting from "Option 66/Custom" to Static |
| 21:27.45 | Shaun2222 | why? |
| 21:28.47 | *** join/#asterisk Egonis (n=roman@tfi1meg.1meg.golden.net) |
| 21:28.52 | *** join/#asterisk jozu (n=torrent@84.120.220.97.dyn.user.ono.com) |
| 21:28.55 | jozu | hi to all |
| 21:29.03 | mcab | Shaun2222: BR 4.0/SIP 2.2.0 added support for DHCP INFORM. If the phone doesn't get a Boot server from DHCP, it will try DHCP INFORM, then fall back on the information you programmed in by hand |
| 21:29.31 | Egonis | I just found out that my customer really ordered a T1 DAL, and not a T1 PRI -- what's the difference, is this a simple configuration change? |
| 21:29.32 | Shaun2222 | i want it to use dhcp to pull network info... |
| 21:29.33 | mcab | Shaun2222: unfortunately DHCP INFORM takes a sod of a long time to timout :-p |
| 21:29.38 | Shaun2222 | i dont use bootp, use ftp |
| 21:30.06 | Shaun2222 | would use sftp is the polycom's wernt broken |
| 21:30.25 | mcab | Shaun2222: you can use DHCP, but either provide the boot server address in the OFFER, or explicitly tell the phone to use the static BootServer host |
| 21:30.56 | Shaun2222 | i want somthing with some authentication... not tftp. |
| 21:30.57 | alrs | Egonis: I've never heard of a T1 DAL. If it means something like "direct analog lines" have them switch to a PRI |
| 21:31.34 | mcab | Shaun2222: the setting I'm talking about doesn't disable DHCP entirely, just tells the phone to not expect to get a bootserver from the DHCP offer (sorry, that probably wasn't clear, and If I had a Polycom handy I'd've checked what the actual name was :-) ) |
| 21:31.49 | Shaun2222 | i see what your saying. |
| 21:32.58 | *** join/#asterisk callguy (n=callguy@pool-72-70-79-233.bstnma.east.verizon.net) |
| 21:34.34 | Shaun2222 | guess i'm not understanding the app log's timestamp.. |
| 21:34.56 | *** join/#asterisk J4k3 (n=jsuter@pimpin.aint.easy.in.grapeland.us) |
| 21:35.09 | roxlu | [TK]D-Fender: I can't find anything about recording calls in the Asterisk book? I searched for record |
| 21:35.56 | [TK]D-Fender | roxlu, Go read the WIKI |
| 21:36.02 | roxlu | ah |
| 21:36.26 | mcab | Shaun2222: what's wrong? |
| 21:36.39 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:36.39 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:36.58 | Shaun2222 | was just trying to figure out why sip.ld was taking so long to load... not really that big of a issue... |
| 21:37.07 | Shaun2222 | i was trying to figure out the idle logo on this though |
| 21:37.10 | Shaun2222 | havnt had much luck. |
| 21:38.07 | *** join/#asterisk devonmeyers (n=devonmey@adsl-76-255-251-185.dsl.snfc21.sbcglobal.net) |
| 21:38.19 | devonmeyers | hello |
| 21:39.05 | J4k3 | roxlu: welcome to the fun of asterisk :) |
| 21:39.24 | roxlu | haha indeed |
| 21:39.29 | roxlu | it's really great! |
| 21:40.05 | devonmeyers | oh joy, i actually have a few Q's if there is anyone in here with experience with abe and the new appliance |
| 21:40.33 | peanut- | ~ask |
| 21:40.33 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:41.46 | devonmeyers | ok then, well I am using ABE on the new asterisk appliance, and I need to be able to edit the config file that is provisioned to the phone handsets |
| 21:42.41 | Shaun2222 | mcab: looks like that fixed it |
| 21:42.53 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:42.59 | Shaun2222 | shit only took about 30 seconds at the most to load sip.ld |
| 21:43.31 | Shaun2222 | mcab: you have any experience with loading idle bitmaps now :) |
| 21:44.13 | mcab | mcab: hah, glad it worked, and nope none at all. Sorry :-) |
| 21:44.42 | devonmeyers | jbot any thoughts? |
| 21:45.04 | [TK]D-Fender | lol |
| 21:45.20 | [TK]D-Fender | devonmeyers, may as well ask a wall :) |
| 21:45.22 | [TK]D-Fender | ~jbot |
| 21:45.23 | jbot | hmm... jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 21:46.14 | devonmeyers | does anyone else have any idea how I can get to editing those files? are there any digium moderators in here? |
| 21:46.43 | peanut- | ~[TK]D-Fender |
| 21:46.43 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 21:46.52 | peanut- | nice. |
| 21:46.56 | [TK]D-Fender | devonmeyers, Call support... witht he price I'm sure you paid for it, you deserve it. |
| 21:47.13 | devonmeyers | only email support offered, and I asked them already |
| 21:47.23 | [TK]D-Fender | devonmeyers, And what'd they have to say? |
| 21:47.45 | devonmeyers | they havent replied back yet |
| 21:48.19 | [TK]D-Fender | devonmeyers, welcome to toaster-ville. |
| 21:48.48 | HarryR | :) |
| 21:48.56 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585336.dsl.bell.ca) |
| 21:49.11 | devonmeyers | ok |
| 21:49.26 | devonmeyers | so thats a no then? does anyone know the root where those files are kept at least? |
| 21:49.30 | devonmeyers | the folder? |
| 21:49.34 | devonmeyers | anything? |
| 21:49.49 | HarryR | the config those phones are provisioned with? |
| 21:50.04 | [TK]D-Fender | devonmeyers, Realize for a moment how few people have bought that appliance or run ABE here... |
| 21:50.17 | dmangot | Shaun2222: I've got a logo going on my Polycom phones |
| 21:50.24 | Shaun2222 | fuckin a... |
| 21:50.31 | devonmeyers | i dont, thats why i was asking if anyone has any experience with ABE and noone replied |
| 21:50.32 | Shaun2222 | 550 doesnt use the 500 configuration.... |
| 21:50.36 | Shaun2222 | it uses the 600 config... |
| 21:50.39 | Shaun2222 | go polycom... |
| 21:50.41 | Shaun2222 | jeez |
| 21:50.55 | devonmeyers | but im getting the impression that right now there is noone that knows ABE, so i will probably go back to searching the web |
| 21:50.55 | [TK]D-Fender | Shaun2222, You'd think it might be in the admin guide ;) |
| 21:50.59 | Shaun2222 | figured 500 would be general for all 5xx devices since there wasnt anything seperate for 550 |
| 21:51.04 | Shaun2222 | haha ya... |
| 21:51.14 | [TK]D-Fender | Shaun2222, considering Oh I don't know.. maybe the fact the 550 = 600 scree-res ;) |
| 21:51.17 | Shaun2222 | i had to find it on some wiki |
| 21:51.37 | [TK]D-Fender | Shaun2222, So um... kinda opbvious that 500 specs don't match. |
| 21:52.05 | Shaun2222 | well, i just assumed 500 would be for 500 series |
| 21:52.11 | mcab | Shaun2222: heh, one would think... However, the 550 is a 650 that's missing a few line keys and EM support. I agree that it's not terribly obvious though... |
| 21:52.30 | [TK]D-Fender | Shaun2222, .... |
| 21:52.31 | [TK]D-Fender | ~assume |
| 21:52.32 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 21:52.40 | Shaun2222 | well it's working.. |
| 21:52.41 | Shaun2222 | looks like shit |
| 21:53.08 | [TK]D-Fender | Shaun2222, fix your colour & res |
| 21:53.27 | devonmeyers | ugh |
| 21:54.03 | Shaun2222 | ya.. need to, the logo is all faded looking |
| 21:54.08 | roxlu | I want to create create a voicemail message, but I want to make the unavailable messages dynamic to the time.. like: Goodmorning, Goodeveneing.. is that possible? |
| 21:54.46 | [TK]D-Fender | roxlu, its already generic if you don't HAVE a recording made. |
| 21:55.03 | roxlu | what do you mean generic? |
| 21:55.58 | [TK]D-Fender | roxlu, "the person at box XXX is (busy / on the phone)," etc |
| 21:57.22 | roxlu | yes generic like "busy" or "not available" but is it possible to play a message when someone calls between 9-12 in the morning and record the message to the voicemail? and when someone calls between 12-18 another sound/message |
| 21:57.35 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:57.48 | [TK]D-Fender | roxlu, that'd be much more complex |
| 21:57.56 | Nugget | the answer to just about any "is it possible to..." question is invariably "yes" |
| 21:58.04 | Nugget | the only question is how much effort you're willing to put in |
| 21:58.14 | TrentCreek | Fender geeeetar is always on here |
| 21:58.22 | TrentCreek | Sure this is not you? |
| 21:58.23 | TrentCreek | http://publications.mediapost.com/index.cfm?fuseaction=Articles.showArticleHomePage&art_aid=29415 |
| 21:58.24 | [TK]D-Fender | roxlu, You've have to check in the dialplan if they HAD a recording and if not do a time check, then play back the appropriate recording OUTSIDE of VM and then dump them in with NO recording playback |
| 21:58.37 | objective | is it possible to run asterisk on the iphone? |
| 21:58.48 | roxlu | okay |
| 21:58.51 | [TK]D-Fender | TrentCreek, death can't stop me! |
| 21:58.52 | objective | i'd like to make even more money off of AAPL |
| 21:59.06 | dmangot | Qwell or russellb, any more thoughts on why my Voicemail emails have the wrong timestamp? It's not even like it's sending them in GMT, it's like it's doing GMT -7 -7 (aka -14) and we are in PST |
| 21:59.07 | Dovid | objective: start hacking...... |
| 21:59.54 | TrentCreek | Typing your fingers to the bone literally |
| 22:00.10 | *** part/#asterisk nny (n=Scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:01.15 | TrentCreek | No hacking..iPhone should be a cut down version of OS X ;-) |
| 22:02.08 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-58-243.pskn.east.verizon.net) |
| 22:03.19 | dmangot | okay, since no one knows about my timestamp problem, anyone know when the registration will be fixed on Asterisk.Org? It never sends me the email |
| 22:04.32 | TrentCreek | check system time and OS time? |
| 22:06.58 | dmangot | all the log files have the correct time |
| 22:07.07 | dmangot | date returns the correct time |
| 22:07.21 | *** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net) |
| 22:07.34 | TrentCreek | isn;t there a confile file that indicates the time zone you are in? |
| 22:07.36 | dexpdx | Anyone seen this error before: |
| 22:07.40 | dexpdx | Anyone seen this error before |
| 22:07.43 | dexpdx | wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8c4895b |
| 22:07.51 | dexpdx | ? |
| 22:07.59 | dmangot | TrentCreek: which one? I've even set TZ in the init.d script |
| 22:08.09 | dexpdx | I'm assuming it has something todo with timing source for zaptel? |
| 22:08.10 | TrentCreek | the asterisk ones |
| 22:08.20 | TrentCreek | just speclating |
| 22:08.32 | dmangot | I'm having the same problem as this guy: http://forums.digium.com/viewtopic.php?t=18560&start=0&postdays=0&postorder=asc&highlight=timestamp+voicemail |
| 22:09.19 | dmangot | it worked perfectly on 1.2, I just upgraded to 1.4.13 when the problem appeared |
| 22:09.50 | dmangot | I would be surprised if I had to include the timezone in my config files just from upgrading, it's not mentioned anywhere |
| 22:10.06 | *** part/#asterisk Egonis (n=roman@tfi1meg.1meg.golden.net) |
| 22:10.38 | TrentCreek | what about callerID? Is the correct date/time being sent? |
| 22:11.58 | dmangot | callerid date/time? |
| 22:12.15 | dmangot | the email message itself has everything correct, the callerid even the time! |
| 22:12.26 | dmangot | it's the headers that have the wrong timestamp |
| 22:13.14 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:13.21 | TrentCreek | that's a tough one there. Sounds like it is coming from the code itself |
| 22:13.29 | *** join/#asterisk saftsack (n=saftsack@pD9E041E0.dip.t-dialin.net) |
| 22:13.40 | dmangot | if I type: echo blah | mail myemailaddress it shows up with the correct time, so it's obviously being munged within asterisk |
| 22:14.17 | TrentCreek | go through the source and find the bug ;-) |
| 22:14.36 | dmangot | yeah, I guess I need to find the time to do that :( |
| 22:15.10 | TrentCreek | its in C or C++? |
| 22:15.39 | dmangot | C |
| 22:16.44 | TrentCreek | well that's not too bad then. It would be nightmaree in C+ tryng to find your way throught that maze of objects and NAMES |
| 22:17.15 | russellb | C can be just as much of a nightmare :) |
| 22:17.36 | dexpdx | asterisk is screwing the timestamp? |
| 22:17.37 | russellb | app_voicemail is large, but not hard to follow, IMO ... |
| 22:17.39 | dexpdx | where? |
| 22:18.12 | dmangot | dexpdx: in the email headers |
| 22:18.19 | TrentCreek | yeah but it's just a bunch of functions. Try going through different classes and inheritence |
| 22:18.47 | dmangot | it appears to be doing the time offset twice (haven't even looked at the code yet) |
| 22:21.37 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 22:25.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:29.08 | nestAr | blarg. |
| 22:29.53 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 22:31.18 | thansen|laptop | can I dynamically get the mailbox number on a given sip extension? |
| 22:34.42 | TrentCreek | dial the box or exten number? |
| 22:37.09 | thansen|laptop | TrentCreek: I want to know what the mailbox= value is for the given sip extension |
| 22:37.14 | tripps | anyone know if there is an errata out for the "book"? there are serious typos and errors in there . . . |
| 22:37.28 | tripps | just spent an entire afternoon wrestling with that. . . ;) |
| 22:42.37 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:42.45 | roxlu | I'm trying to fix my voicemail. internally it's working |
| 22:43.07 | roxlu | but when I could from outside, my internal phone rings, but it never gets to the voicemail... what could be wrong? |
| 22:43.29 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 22:44.07 | TrentCreek | isn;t there a different dial plan for calls coming in? |
| 22:44.21 | roxlu | yes but I route it to my extensions. |
| 22:44.59 | roxlu | welll.. tommorow another day :-) |
| 22:45.06 | TrentCreek | have you looked at examples of see how others are doing it? |
| 22:45.15 | roxlu | yes thanks |
| 22:45.33 | roxlu | bye bye! |
| 22:49.24 | lirakis | later all |
| 22:49.26 | *** part/#asterisk lirakis (n=eric@69.24.142.1) |
| 23:09.54 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 23:10.27 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 23:10.55 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 23:13.29 | *** join/#asterisk irule (n=irule@201.151.52.150) |
| 23:13.56 | irule | ~book |
| 23:13.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 23:15.18 | TrentCreek | ~delete book |
| 23:15.19 | jbot | ACTION glares at book and then takes every step necessary to completely delete book and destroy any and all evidence that book ever existed |
| 23:16.19 | *** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net) |
| 23:16.53 | _ShrikE | ~itsp |
| 23:16.53 | jbot | i guess itsp is an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 23:17.04 | _ShrikE | ~ITSP |
| 23:17.05 | jbot | i heard itsp is an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 23:21.54 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:32.33 | *** join/#asterisk bkw__ (n=brian@ppp-70-128-126-184.dsl.tulsok.swbell.net) |
| 23:32.56 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 23:32.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:34.41 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 23:38.38 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:40.31 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 23:40.49 | ManxPower | Yay! The T-1s are back up! Only took 7 damn hours. |
| 23:41.53 | HarryR | the janitor pull the cable again by accident? |
| 23:41.58 | *** join/#asterisk cygar (n=cygar@200.26.191.3) |
| 23:42.06 | cygar | hello |
| 23:42.17 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 23:42.20 | ManxPower | HarryR: VERY bad storms in New Orleans. |
| 23:42.26 | HarryR | at the office they close the door to the makeshift server room |
| 23:42.36 | ManxPower | There was isolated severe flooding. |
| 23:42.39 | HarryR | without realizing that 50 servers in a room with no aircon and a closed door = bad |
| 23:42.43 | HarryR | ooh :\ that sucks |
| 23:42.58 | ManxPower | HarryR: I REMOVE the actual door. |
| 23:43.01 | HarryR | lol |
| 23:43.05 | HarryR | or get aircon |
| 23:43.10 | [T]ank | i am issuing a exten => XXXXXXX,n,Busy(). Asterisk calls it, but I cannot hear it. I have restarted asterisk. Did not fix it. I can make it work on other servers. Any ideas? |
| 23:43.19 | ManxPower | Either the room has AC or it has no door. The customer's option. |
| 23:43.31 | cygar | I got a problem, i am working on an old "trixbox" where extensions used to be created using FreePBX, now I am have changed it to realtime and I am facing the following problem when trying to dial to the extension I create manually in the db: recordingcheck|20071023-203957|1193182797.4428: No AMPUSER db entry for 211. Not recording |
| 23:43.46 | cygar | the extension is well registered and working properly to make outbound calls |
| 23:43.48 | ManxPower | cygar: make sure you have /etc/asterisk/indications.conf (the default one is fine). |
| 23:44.31 | cygar | ManxPower: I am calling from SIP to SIP, what does indications.conf has to be with this? |
| 23:44.52 | cygar | I am willing there's a DB [ where it tries to find this AMPUSER ] somewhere, but I can not find it... |
| 23:45.08 | cygar | since this happens to me when I create the users "manually" and not through freepbx |
| 23:46.44 | cygar | ManxPower: as far as I know indications.conf is where you configure local stuff for different signalling/tones, etc used in differents countries ( related to ZAP and not IP like here) |
| 23:47.54 | Trevor_b | Anyone use polycom 501's and 320/330 series in the same office? |
| 23:49.18 | Trevor_b | AND/OR anyone have 301's displays go blank after loading up once the sip application has loaded? |
| 23:53.24 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 23:54.12 | grandpapadot | Hi all. |
| 23:56.43 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:56.56 | [T]ank | CLI> Shows: -- Executing Playback("SIP/3594-09cdc348", "busy") in new stack. But I hear nothing. Other files I hear just fine. |