| 00:04.59 | *** join/#asterisk ariel_ (~abatista@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:18.21 | *** join/#asterisk shapr (~shapr@nat/digium/x-vjkclhptfvnnirnu) |
| 00:35.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:42.25 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.131) |
| 00:53.15 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 00:57.34 | *** join/#asterisk sahafeez (~sahafeez@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 01:13.48 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 01:24.59 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 01:27.20 | LemensTS | Is there not a STARTTIME and ENDTIME that can be used to know when a call starts/ends? |
| 01:28.14 | LemensTS | Im trying to get the STARTTIME of the call after the call is answered, not when it starts ringing |
| 01:36.28 | *** join/#asterisk jzknight (~jason@pool-96-242-37-152.nwrknj.fios.verizon.net) |
| 01:38.52 | *** join/#asterisk Kumbang (~Kumbang@180.245.137.5) |
| 01:42.28 | *** part/#asterisk Kumbang (~Kumbang@180.245.137.5) |
| 01:49.32 | *** part/#asterisk jzknight (~jason@pool-96-242-37-152.nwrknj.fios.verizon.net) |
| 01:50.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 02:00.56 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 02:03.33 | *** part/#asterisk newsham (~chat@udp216902uds.hawaiiantel.net) |
| 02:04.39 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176003582.dsl.bell.ca) |
| 02:10.04 | *** join/#asterisk pagec (~chatzilla@s1.corebizserv.com) |
| 02:10.22 | pagec | what is the usb timing source for dahdi (i.e. the ztdummy equivalent) |
| 02:14.29 | jaytee | dahdi_dummy |
| 02:22.29 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 02:29.17 | *** join/#asterisk coppice (~chatzilla@m121-202-95-152.smartone-vodafone.com) |
| 02:33.22 | *** join/#asterisk \DSAFEW\ (~dsafew@ip70-190-95-86.ph.ph.cox.net) |
| 02:33.59 | \DSAFEW\ | hello, I'm very new to asterisk, and I'm trying to make my new X100P work with the phone line basically. |
| 02:34.10 | \DSAFEW\ | how does one go about that with 1.6 on gentoo? |
| 02:40.51 | \DSAFEW\ | all the guides I read mention zaptel and zapata.conf etc... I have no such package installed nor do I see it anywhere |
| 02:42.14 | Freeaqingme | \DSAFEW\, if i recall correctly it's in asterisk-addons |
| 02:42.30 | \DSAFEW\ | ahh, ty I'll look there |
| 02:42.43 | X-Rob | \DSAFEW\, you know, you could try googling 'gentoo zaptel' |
| 02:43.20 | X-Rob | second link is this: http://www.voip-info.org/wiki/view/Asterisk+Linux+Gentoo |
| 02:45.45 | \DSAFEW\ | I read that yesterday, there's no zaptel package, use flag and the latest stable isn't asterisk-0.9.0 it's 1.6.2 |
| 02:45.57 | \DSAFEW\ | but I thank you for the addons advice, it's compiling now |
| 02:46.45 | Freeaqingme | \DSAFEW\, did you check if it's actually in there? |
| 02:47.11 | \DSAFEW\ | it's already installed, I'm checking now |
| 02:47.18 | Freeaqingme | kk |
| 02:48.56 | \DSAFEW\ | hah, no it's not in there, it's got a polish word counting module though |
| 02:48.58 | \DSAFEW\ | how strange |
| 02:52.32 | \DSAFEW\ | Zaptel is DAHDI now? |
| 02:52.55 | \DSAFEW\ | that makes things much easier to figure out now. |
| 02:53.03 | *** part/#asterisk \DSAFEW\ (~dsafew@ip70-190-95-86.ph.ph.cox.net) |
| 02:58.04 | *** join/#asterisk ectospasm (~trey@75.76.170.25) |
| 03:15.44 | *** join/#asterisk dlynes_laptop (~dlynes@bas6-hamilton14-1279410619.dsl.bell.ca) |
| 03:18.39 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 03:22.59 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 03:25.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:33.35 | *** join/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 03:35.25 | flux_control | I'm having some issues with using a SIP registrar. Currently I can make outgoing calls, but not receive incoming calls. I'm having problems with media on the outgoing calls, though I'm guessing that's a separate issue from the incoming calls. |
| 03:36.19 | flux_control | When I make outgoing calls, I can see signalling happen in the asterisk console. When I try to make an incoming call to my service via another phone, I don't see any activity in the asterisk console. |
| 03:36.39 | flux_control | Would anyone be able to help me with fixing my registration? |
| 03:37.13 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 03:38.14 | flux_control | I currently am using type=friend and I don't have a register line in the [general] section (tried that, but that resulted in neither incoming nor outgoing working, whereas not having it but setting type=friend at least gets me outgoing). |
| 03:40.14 | WIMPy | How do you exprect a non existant registration to work? |
| 03:40.33 | WIMPy | But the whole thing sonds like a firewall isssue. |
| 03:41.14 | flux_control | I don't expect it to work. |
| 03:41.34 | flux_control | As I said, I tried setting it up, but I couldn't get it to work. |
| 03:41.54 | flux_control | I've seen differing documentation on how the register line is supposed to look, so I'm sure I did something wrong. |
| 03:42.43 | flux_control | I'm unclear on how the register line is really supposed to look, so any help explaining that would be appreciated. |
| 03:42.46 | WIMPy | Always use teh sample configuration file that came with the version you're using as a template. |
| 03:42.56 | flux_control | I tried that |
| 03:43.07 | flux_control | The register line I tried was something like this: |
| 03:43.16 | shapr | flux_control: Certainly sounds like firewall issues in both cases. |
| 03:43.34 | shapr | At least, if you're using Network Address Translation from your asterisk server to you SIP registrar. |
| 03:43.59 | flux_control | register => number@server:password:authname@host/context |
| 03:44.18 | flux_control | It could be a firewall issue even when the outbound signalling is working? |
| 03:44.58 | shapr | Yes, very much so. |
| 03:44.59 | flux_control | I understand the media is a firewall issue, but I thought that having the SIP signalling working fine in the outbound direction would be the same for the inbound. |
| 03:45.08 | shapr | NAT is designed to allow outgoing connections and deny incoming connections. |
| 03:45.20 | flux_control | Hmm.. |
| 03:45.31 | flux_control | Unless it's ported inward. |
| 03:45.35 | flux_control | Which it is. |
| 03:45.51 | shapr | Do you have 5060 on your external IP forwarded to your asterisk server? |
| 03:45.54 | flux_control | The asterisk server is on a different host than I connect from with my own client. |
| 03:45.58 | flux_control | Yup |
| 03:46.01 | flux_control | My client can connect fine. |
| 03:46.07 | shapr | From the internet at large? |
| 03:46.13 | flux_control | And I can get media between my client and the asterisk server. |
| 03:46.14 | flux_control | yes |
| 03:47.08 | shapr | In that case, it sounds like your SIP registrar is not pointing to the correct IP address for your incoming number. |
| 03:47.29 | flux_control | Any way to fix that in asterisk? |
| 03:48.12 | flux_control | Is the register line strictly needed? |
| 03:48.59 | flux_control | I've also read that setting up a specific host section and having type=user or type=friend would enable it to work for incoming (and I have a registrar over IAX set up this way in iax.conf which works fine). |
| 03:50.55 | shapr | flux_control: I'd check your SIP registrar's settings about which IP and port to contact for calls going to you. |
| 03:51.45 | flux_control | I don't have access to that. |
| 03:52.20 | shapr | Can you ask them? |
| 03:53.00 | flux_control | I'll try, but if I'm understanding what you're suggesting I don't think they have such a thing necessarily. |
| 03:53.13 | flux_control | Or maybe they do, in which case that would definitely explain the problem... |
| 03:53.22 | flux_control | It's VoIP service bundled with my net. |
| 03:54.07 | flux_control | They provide all the SIP settings, so I hadn't considered that they might lock it to whatever my current assigned IP address is (which is dynamic), but I suppose they could be. |
| 03:54.48 | shapr | Well, from what you're saying, your configuration is such that they need to contact you back when you have an incoming call. |
| 03:56.11 | flux_control | To make sure I'm not misunderstanding you, what specifically do you mean by "contact me back"? |
| 03:56.55 | shapr | SIP providers can have several ways of doing outgoing and incoming calls. |
| 03:57.20 | flux_control | Yes |
| 03:57.54 | shapr | If it's the "two Asterisk SIP servers" style setup, then each server must be able to contact the other on the SIP port 5060, right? |
| 03:58.59 | flux_control | What specifically is the "two * SIP servers" setup? I don't see that as a single type of setup, since you can make 2 * boxes work any of the ways that calls can be provided, yes? |
| 03:59.05 | shapr | From what you've said, you can contact your Asterisk server on port 5060 from outside of your firewall. |
| 03:59.21 | flux_control | Do you mean peering two asterisk servers? Rather than one being the user of another? |
| 04:00.31 | flux_control | This is the setup I'm trying to achieve: client --> asterisk --> provider (where --> means "user of") |
| 04:00.46 | flux_control | All three are different networks |
| 04:01.21 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:01.26 | flux_control | My client registers to my asterisk server just fine, and I can make and recieve calls through it (via means not involving the provider in question) |
| 04:02.40 | flux_control | I'd like my asterisk to register to the provider in question as a client, so I can use it to manage calls/media rather than using my client directly (e.g., for voicemail when I'm not connected to the net myself, as the asterisk server is on 24/7). |
| 04:14.25 | Beltechs | Hello I was upgrading from 1.6 to 1.8 and now * will not start Thanks in Advance. http://pastebin.com/TXNv9FJm |
| 04:14.52 | Beltechs | Segmaentation Fault |
| 04:15.46 | shapr | Beltechs: clean out /usr/lib/asterisk/modules/ |
| 04:15.56 | shapr | try ls -otr /usr/lib/asterisk/modules/ |
| 04:16.08 | shapr | er, you want ls -tr now that I think about it.. |
| 04:16.17 | shapr | get rid of the older modules from the 1.6 install, see if that fixes it |
| 04:16.35 | Beltechs | omit th o |
| 04:17.19 | shapr | Eh, I like long listings :-) |
| 04:18.12 | Beltechs | ok showed a list in Green Text |
| 04:18.57 | Beltechs | amortal restart - same error as original |
| 04:19.05 | Beltechs | amportal |
| 04:20.25 | shapr | Beltechs: Have you checked the logs for errors? |
| 04:22.54 | Beltechs | http://pastebin.com/aH8XEESL |
| 04:23.08 | Beltechs | I dont see anything that sticks out |
| 04:23.56 | shapr | Beltechs: Do you have codec_g729a.so or res_skypeforasterisk.so or one of the other binary-only modules installed? |
| 04:24.15 | Beltechs | i dont think so |
| 04:24.45 | Beltechs | why would that affect my upgrade? |
| 04:27.59 | shapr | Because the module API is different for 1.8 than for 1.6 and trying to load shared objects built for some other version of Asterisk can cause weird stuff to happen. |
| 04:28.24 | shapr | Beltechs: did you build this asterisk 1.8 on the system where you are running it? |
| 04:28.56 | Beltechs | give me a sec. |
| 04:30.00 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 04:31.00 | *** join/#asterisk tstorm (~tstorm@c-98-207-90-194.hsd1.ca.comcast.net) |
| 04:34.16 | Beltechs | Its an upgrade using a tutorial from one of the moderators |
| 04:34.37 | Beltechs | let me get the link |
| 04:38.54 | Beltechs | http://forums.asterisk.org/viewtopic.php?f=14&t=75822&p=149739#p149739 |
| 04:42.06 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 04:43.16 | shapr | Beltechs: I don't know... it certainly sounds like some of the binary modules for asterisk 1.6 are still installed. |
| 04:43.30 | shapr | And yum would clean out any that were installed by the packaging system |
| 04:43.45 | Beltechs | yum udate? |
| 04:43.48 | shapr | thus my question as to whether you've got modules from g729 or skype installed. |
| 04:43.50 | Beltechs | update? |
| 04:44.04 | Beltechs | ah |
| 04:44.09 | shapr | If you followed Qwell's post, I would expect that to work in general. |
| 04:44.45 | Beltechs | where would i find the g729 and skype modules to see if thery were installed |
| 04:44.46 | Beltechs | ? |
| 04:45.24 | shapr | They'd be in /usr/lib/asterisk/modules/ |
| 04:45.36 | shapr | and they'd have a different modified time than the other modules |
| 04:45.47 | Beltechs | let me see whats there |
| 04:45.54 | shapr | thus my original suggest of doing ls -tr to find the oldest .so files in /usr/lib/asterisk/modules/ |
| 04:47.20 | Beltechs | everything thats there was modified on the same date 10 days ago |
| 04:48.36 | Beltechs | 20 hours machine down .... time for ISO... |
| 04:54.33 | phix | hey, I want to wait for a DTMF button to be pressed, should I set autofallthrough to no, use a queue or is there another method to do this? |
| 04:54.57 | shapr | background or waitexten? |
| 04:55.11 | phix | Basically what I am trying to do is playback a message if someone calls me out of business times then allows them to press 1 to dial me any way or 2 or wait for timeout to leave a voice message |
| 04:55.46 | Beltechs | vmxl locater? |
| 04:56.07 | phix | Not sure what vmxl is |
| 04:57.00 | Beltechs | http://www.freepbx.org/news/2007-09-13/follow-me-or-vmx-locater-which-one-is-for-you |
| 04:58.59 | shapr | phix: You want the background application in dialplan |
| 05:01.18 | Beltechs | shapr shouuld I just reinstall from ISO? |
| 05:02.40 | phix | shapr: I am using both background and waitexten, testing now |
| 05:05.36 | *** join/#asterisk Justman (~just@78.108.73.46) |
| 05:05.52 | Justman | hi |
| 05:05.54 | shapr | Beltechs: Heck if I know... it certainly sounds like some sort of .so is leftover, but I'm not sure how to track it down |
| 05:06.27 | Beltechs | yea its time to whip out the ISO |
| 05:06.31 | Justman | it's possible to tune up astdb size more than 65 kbytes ? (sorry for bad english) |
| 05:07.43 | Justman | sometimes i see this warning in the CLI "Could not create persistent member string: out of space" |
| 05:07.58 | Justman | asterisk 1.6.2.13 |
| 05:14.13 | shapr | Justman: Do you have space free on the disk? |
| 05:14.24 | Justman | shapr, yes |
| 05:18.57 | phix | oh nice, it works awesomely! thnx! |
| 05:22.29 | shapr | Justman: I don't know... maybe someone else here will know? |
| 05:23.38 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 05:37.09 | *** join/#asterisk Howlader (howlader@soft.bdcom.com) |
| 05:49.50 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 05:51.02 | *** part/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 06:01.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-gupohzfefktxooel) |
| 06:03.56 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 06:11.05 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 06:15.52 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:23.03 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 06:23.15 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 06:31.06 | Balistic | Can one use mISDN v2 with asterisk 1.6? |
| 06:31.50 | shapr | You can, but I'd not recommend it. |
| 06:31.56 | shapr | Digium doesn't support mISDN, only DAHDI |
| 06:32.50 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 06:40.00 | Balistic | shapr: ive got a HFC-S ISDN BRI card but dahdi doesn't want to play well with it |
| 06:41.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:46.39 | shapr | Balistic: Might be your best choice then |
| 06:47.07 | *** join/#asterisk sourcode (~code@ppp-58-8-68-59.revip2.asianet.co.th) |
| 06:48.43 | Balistic | with both misdn and dahdi the kernel drivers detect the card properly. But im struggling to compile asterisk with misdn support. Configuration on asterisk fails with the flag '--with-misdn=/root/mISDNuser/' |
| 06:49.27 | shapr | Not sure about that. |
| 06:53.02 | Balistic | hmm, lemme try the dahdi route again. I got to the point where 'dahdi show channels' listed 2 channels as expected but 'pri show spans' showed the span is down |
| 06:59.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:04.59 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:05.05 | schmidts | good morning |
| 07:05.41 | *** join/#asterisk tstorm (~tstorm@c-98-207-90-194.hsd1.ca.comcast.net) |
| 07:07.27 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:10.02 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:14.49 | *** join/#asterisk tstorm (~tstorm@c-98-207-90-194.hsd1.ca.comcast.net) |
| 07:15.54 | kleszcz | good morning |
| 07:18.09 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9dd1:1b05:e9a6:3ea0) |
| 07:21.08 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:25.16 | *** join/#asterisk scruz (29d9cd32@gateway/web/freenode/ip.41.217.205.50) |
| 07:25.45 | scruz | hello. |
| 07:26.17 | scruz | could someone please recommend a primer on AEL? i couldn't seem to find any in the documentation on www.asteriskdocs.org |
| 07:29.37 | wdoekes2 | ~newbook |
| 07:29.38 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 07:30.14 | wdoekes2 | perhaps? I haven't looked though |
| 07:30.59 | wdoekes2 | oh.. I see "Once you learn this language, it should be fairly easy to transition to AEL or LUA, should you desire." |
| 07:31.06 | wdoekes2 | never mind then :) |
| 07:31.34 | shapr | Is AEL mentioned in that book? |
| 07:35.19 | wdoekes2 | mentioned, yes |
| 07:35.26 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 07:35.32 | wdoekes2 | "NamePurposePopularity/Status |
| 07:35.32 | wdoekes2 | pbx_aelAsterisk Extension Logic offers a dialplan scripting language that looks like a modern programming language.Usable[a] |
| 07:35.36 | wdoekes2 | pbx_configThis is the traditional, and most popular, dialplan language for Asterisk. Without this module, Asterisk cannot read extensions.conf.Useful |
| 07:35.39 | wdoekes2 | pbx_dundiPerform data lookup on remote Asterisk systems.Useful |
| 07:35.41 | wdoekes2 | pbx_loopbackPerforms something similar to a dialplan include, but in a deprecated manner.Insignificant[b] |
| 07:35.44 | wdoekes2 | pbx_luaAllows creation of dialplan using Lua scripting languageUseful |
| 07:35.47 | wdoekes2 | pbx_realtimeFunctionality related to Asterisk Realtime ArchitectureUseful |
| 07:35.49 | wdoekes2 | pbx_spoolOutgoing spool support relating to Asterisk call filesUseful |
| 07:35.52 | wdoekes2 | oops |
| 07:35.54 | wdoekes2 | sorry |
| 07:36.00 | wdoekes2 | I meant to say: "We have not found too many people using AEL. We suspect this is because most developers will tend to use AGI/AMI if they do not want to use traditional dialplan" |
| 07:37.31 | wdoekes2 | that applies to me as well.. I considered using lua, but went with traditional dialplan because I figured it'd have fewer bugs / better support |
| 07:38.34 | shapr | I'd rather use Python or Haskell for my Asterisk scripting. |
| 07:40.11 | wdoekes2 | don't underestimate the power of lua |
| 07:40.32 | scruz | thanks |
| 07:40.37 | shapr | I used to use the ion window manager lots, and I messed with WoW scripting a bit, so I've written some Lua. |
| 07:40.53 | shapr | But now there's xmonad, and it's scripted in Haskell and I broke the WoW addiction... |
| 07:41.50 | *** join/#asterisk DND (~ctcarino@94.200.7.26) |
| 07:42.00 | DND | hi. how can i do qos on sip packets |
| 07:42.27 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 07:42.42 | shapr | god morgon oej, hur mår du idag? |
| 07:44.41 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85de.bcn.adamo.es) |
| 07:56.27 | *** join/#asterisk iulhk (~iulhk@175.110.11.42) |
| 08:03.03 | *** join/#asterisk erol (~chatzilla@195.155.223.246) |
| 08:03.07 | erol | hi |
| 08:03.19 | shapr | hi erol |
| 08:03.37 | erol | with excuse can i ask a question about cisco call manager? |
| 08:03.46 | shapr | ~question |
| 08:03.46 | infobot | from memory, question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 08:05.17 | erol | i am trying to setup translations for DID numbers. They all start with 2123640xxx and because of this our 212 DN doesnt work |
| 08:05.28 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 08:05.30 | erol | how can i solve this? |
| 08:08.25 | erol | no ideas? |
| 08:10.55 | shapr | I've never used cisco products. |
| 08:12.03 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 08:24.44 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:25.48 | *** join/#asterisk Secret_Hamster (~pdiddly@host81-149-30-66.in-addr.btopenworld.com) |
| 08:26.26 | Secret_Hamster | Just wondering if anyone has come across an issue with a system ringing other extensions when an external call is dropped |
| 08:27.41 | Secret_Hamster | What appears to happen is the first group in a dial-plan puts the call down (and I'm seeing it as the other end is not on hook), and it then rings the next hunt group |
| 08:28.55 | Secret_Hamster | we've actually had it report that the other end of the call (the one on the other end of the external trunk) is also getting their phones ringing on the main huntgroup |
| 08:35.10 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:40.34 | X-Rob | Secret_Hamster, it's a bug in your dialplan. |
| 08:43.40 | Secret_Hamster | I'd kinda reckoned on that, just wondering if anyone possibly could guess what. I've inherited the system from the previous admin, and asterisk is not particularly an expertise of mine |
| 08:44.34 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:47.28 | *** join/#asterisk arkygeek (~arkygeek@host86-136-133-134.range86-136.btcentralplus.com) |
| 08:47.36 | arkygeek | hi everyone.  I am trying to connect to my asteriskNOW server (1.4 with freepbx) using a SIP trunk⦠ I am having difficulties with the settings⦠I am on a Mac using Telephone Version 0.14.3 (23) and my Asterisk server's IP is 192.168.1.105 |
| 08:47.46 | arkygeek | Telephone wants Description: Full Name: Â Domain: Â Username: Â and Password: |
| 08:48.02 | arkygeek | I followed these instructions: http://pebble-tree.co.uk/forum/default.aspx?g=posts&m=79 |
| 08:48.17 | arkygeek | What do I enter into Telephone to connect? |
| 08:48.50 | Justman | shapr, http://dpaste.com/526240/ |
| 08:49.30 | Justman | shapr, i think, it's hardcoded size of astdb (65 kb) |
| 08:49.39 | Justman | but can't find it in sources |
| 08:53.21 | *** join/#asterisk UQlev (~chatzilla@212.50.99.8) |
| 08:54.28 | UQlev | can anybody help to find out reason of ERROR[1172]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("proton.sallbay.com", "(null)", ...): hostname nor servname provided, or not known |
| 08:55.27 | UQlev | is there any nameserver should be mentioned in configs? does asterisk use resolv.conf? |
| 08:57.20 | shapr | yes |
| 08:58.11 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:03.53 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:08.32 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ndodywirlelixcde) |
| 09:10.23 | Justman | shapr, i found it, its PM_MAX_SIZE :) |
| 09:10.32 | shapr | cool! |
| 09:11.01 | Justman | yeah ;) |
| 09:11.46 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:12.58 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 09:23.12 | Secret_Hamster | anyone here used an Aastra 610D dect phone with Asterisk? |
| 09:25.11 | shapr | We're using a Panasonic KX-TPA50 here. |
| 09:26.57 | Secret_Hamster | the only reason I ask is that we are moving over to the Aastra's from Cisco (cheaper), and I'm hoping that the config will pretty much be the same |
| 09:27.39 | Secret_Hamster | however, I get the sense I have to buy some extra RFP equipment to make it work, which is horrifically expensive |
| 09:28.33 | Secret_Hamster | shapr, but for £59 it's going to be hard to beat that. What's it like? |
| 09:28.53 | *** join/#asterisk Howlader (~howlader@soft.bdcom.com) |
| 09:29.35 | shapr | The KX-TPA50 has surprising range, we set up the base in the center of our three-story building (well, you've seen Digium?) and it lets us roam around outside as well. |
| 09:29.53 | shapr | It also has a standard headset adapter option, and the battery lasts a long time. |
| 09:30.38 | shapr | All of us compared and contrasted... we tried to find an affordable SIP/wifi phone, but they don't seem to exist. |
| 09:31.08 | Secret_Hamster | I'm just looking for a decent dect phone, with sip capabilities |
| 09:31.36 | Secret_Hamster | i.e we have our asterisk server, we have a network port to plug in the connection |
| 09:32.04 | Secret_Hamster | I'm not wanting a WiFi version as, tbh, the quality suxs |
| 09:32.33 | Secret_Hamster | there is probably things I could do, but I'm just looking for the simplest option |
| 09:32.46 | Secret_Hamster | we had a digium IAX box, but it gone bang |
| 09:33.07 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 09:34.13 | shapr | Which box? |
| 09:34.21 | shapr | is idly curious |
| 09:36.39 | Secret_Hamster | I believe it's an IAXy 2 |
| 09:37.16 | Secret_Hamster | it's ancient |
| 09:37.31 | shapr | oh that is ancient! I've actually seen one of those! |
| 09:37.38 | shapr | I have never seen an X100P though. |
| 09:38.25 | Secret_Hamster | so in essence we had a standard dect phone (which still works, but again if friggin' ancient) and the adapter box |
| 09:38.35 | Secret_Hamster | I'm looking to replace it with an all in one |
| 09:39.10 | Secret_Hamster | We run Aastra's and Cisco, so ideally looking to replace it with an Aastra |
| 09:41.38 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 09:45.29 | longword | I was in last week with some Cisco 7962G troubles - 10 second delays making outbound calls |
| 09:45.39 | longword | Turns out downgrading the firmware was the answer. |
| 09:46.32 | shapr | suggests Linux |
| 09:46.59 | longword | The latest, 9.1 is epic fail, doesn't even register. 8.5(4) was mostly working, with the 10 second delay. 8.4(4) had the same problem. Now running 8.3(5) and it seems OK. |
| 10:07.46 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 10:14.52 | *** join/#asterisk Tuju (~tuju@84-50-207-175-dsl.est.estpak.ee) |
| 10:37.02 | *** join/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 10:38.29 | StaRetji | p3nguin: Hi there, not sure if you remember, but you were helping me last week with mobile and master asterisk issue. I'm still having problem if I try to reach mobile asterisk. I call several times before I was able to reach mobile user/peer |
| 10:39.52 | StaRetji | so, qualify = 1000 didn't help. Internet on both sides is perfect. Any other ideas? |
| 10:59.29 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 11:01.55 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:09.43 | *** join/#asterisk wolf_be (~wolf_be@212.123.9.210) |
| 11:09.52 | wolf_be | . |
| 11:17.49 | *** part/#asterisk wolf_be (~wolf_be@212.123.9.210) |
| 11:28.04 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:31.56 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 11:34.19 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 11:51.52 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:53.52 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 11:58.53 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 12:05.15 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 12:15.33 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:17.46 | Balistic | What does this mean: "PRI span 1/0: Provisioned, Down, Active" from "pri show spans" |
| 12:21.24 | WIMPy | There's nothing connected? |
| 12:21.50 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:25.25 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 12:27.20 | Balistic | WIMPy: meaning? no current connections or no cable? |
| 12:27.43 | WIMPy | no cable |
| 12:28.02 | WIMPy | Try dahdi show status as well. |
| 12:28.25 | Balistic | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 12:28.29 | Balistic | HFC-S PCI A ISDN card 0 [TE] OK 0 0 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 12:28.32 | Balistic | DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 12:29.12 | WIMPy | Ok, so there's something connected but not configured or down. |
| 12:30.09 | Balistic | how do you see that? |
| 12:30.20 | WIMPy | Yellow alarm |
| 12:30.35 | Balistic | The options column? |
| 12:31.07 | WIMPy | Wait. Is that the right column? |
| 12:31.43 | WIMPy | No. Sorry. Misread that because of the wline warpping around. |
| 12:32.17 | WIMPy | "OK" doesn't look too bad :-) |
| 12:33.23 | Balistic | heh, but im still missing something. |
| 12:33.40 | Balistic | This card+cable works in another box running asterisk 1.4+misdn |
| 12:33.42 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 12:33.42 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 12:33.47 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:34.04 | Balistic | Currently trying to get it working on a new box with asterisk 1.6+dahdi |
| 12:35.12 | _Corey_ | Balistic: What kind of card is that? |
| 12:35.36 | WIMPy | I'm not sure, wehn it puts an extra down there. But IIRC will only say OK, when L1 is up. |
| 12:35.51 | Balistic | _Corey_: HFC-S ISDN BRI |
| 12:35.53 | WIMPy | read HFC-S :-) |
| 12:36.30 | Balistic | Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] |
| 12:36.59 | _Corey_ | I haven't had my coffee yet, but you're trying to bring up a PRI with a BRI card? |
| 12:37.40 | tzafrir | Balistic, what version of DAHDI is it that you use dahdi_dummy? |
| 12:38.10 | tzafrir | What's the output of lsdahdi ? |
| 12:38.19 | Balistic | DAHDI Version: 2.2.1 Echo Canceller: MG2 |
| 12:39.00 | WIMPy | thinks it's usually a good idea to use up-to-date versions for BRI stuff. |
| 12:39.05 | Balistic | lsdahdi -> http://pastebin.com/HkxRtZvf |
| 12:39.27 | Balistic | only dahdi 2.1/2.2 has drivers for HFC-S chips |
| 12:39.44 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 12:39.57 | tzafrir | huh? dahdi-2.4.1 (now in Debian testing) has zaphfc |
| 12:40.29 | Balistic | _Corey_: my actuall problem is that asterisk doesn't allow outbound and shows nothing in debug for inbound |
| 12:40.30 | tzafrir | anyway, let's first see what we have |
| 12:41.01 | tzafrir | asterisk -rx 'pri show span 1' |
| 12:41.45 | Balistic | http://pastebin.com/uDhpG1Qb |
| 12:43.41 | ujjain | query p3nguin hi. |
| 12:43.46 | ujjain | hmm, :p |
| 12:43.47 | LemensTS | is there a "pickuptime" variable to find out when the call is answered, rather than when it starts to ring? |
| 12:44.44 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 12:44.53 | Balistic | tzafrir: dahdi 2.4 from http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/ does not have the zaphfc driver |
| 12:45.12 | tzafrir | Balistic, DAHDI never includes zaphfc |
| 12:45.53 | WIMPy | You always need a patched or 3rd party enhanced dahdi. Or you use misdn or LCR. |
| 12:46.04 | tzafrir | Balistic, did you set signalling=bri_cpe_ptmp ;? |
| 12:46.09 | Balistic | yes |
| 12:46.26 | Balistic | i have also tried ptp, but im old config ran ptmp |
| 12:46.41 | tzafrir | What happens when a call comes in? |
| 12:46.57 | tzafrir | asterisk -r |
| 12:46.59 | Balistic | nothing show on astrisk console with verbose/debug > 10 |
| 12:47.10 | tzafrir | pri set debug 1 span 1 |
| 12:47.45 | Balistic | nothing |
| 12:48.34 | Balistic | tzafrir: this is a ubuntu box so it should be easy to get the debian dahdi going. ill give that a try |
| 12:48.47 | Balistic | is the dahdi_dummy module required for timing? |
| 12:49.27 | tzafrir | Balistic, as of 2.3.0, dahdi_dummy is no longer required. The main dahdi code provides timing, if required |
| 12:49.53 | Balistic | i noticed if dahdi_dummy is loaded first the dahdi_test functions but not the other way around |
| 12:51.25 | tzafrir | Balistic, the patch from the package that you actually need is http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra |
| 12:53.06 | zkn | how do I cut the last 19 caracters off the dialed exten? |
| 12:53.53 | Balistic | tzafrir: that patch is for dahdi-linux-complete-2.4.1+2.4.1.tar.gz from asterisk |
| 12:54.19 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 12:54.21 | tzafrir | Balistic, for the linux subdirectory |
| 12:54.21 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 12:54.40 | tzafrir | It should apply with 'patch -p1' there |
| 12:54.55 | Balistic | k, download is a bit slow so ill check back in 15min or so |
| 12:56.09 | *** join/#asterisk Howlader (Howlader@soft.bdcom.com) |
| 12:57.02 | *** join/#asterisk Howlader (Howlader@soft.bdcom.com) |
| 12:58.39 | saxa | anybody in here knows what should I set in Brasil / Bahia in cidsignalling to get the calling number appear ? |
| 13:00.07 | saxa | if I use a small box I bought for cid, and I connect it to the phone line, the number apears on the small box, this means that my telco is sending the number to me |
| 13:00.23 | saxa | but it does not apear on the asterisk console |
| 13:01.01 | saxa | any ideas on how should i set cidsignalling ? |
| 13:01.37 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:07.20 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 13:11.33 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 13:14.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:19.21 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:19.21 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:21.02 | *** join/#asterisk Shnootz (~Hanan@bzq-218-154-21.cablep.bezeqint.net) |
| 13:25.41 | *** join/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 13:25.46 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 13:25.52 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:25.55 | datacompboy | Hello! Anyone knows sip<=>h323 proxy with video support? |
| 13:26.52 | datacompboy | I can't even chan_ooh323 make working :( in code is support for H263, but on call initiate video codec lost... |
| 13:29.25 | *** part/#asterisk tasca (~tasca@189.34.27.64) |
| 13:30.27 | Balistic | tzafrir: dahdi 2.4.1 is running now without dummy module and dahdi_test functions. original issue remains. debug on pri now gives "Sending TEI management message 1, TEI=127" every now and then |
| 13:32.52 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 13:34.26 | *** part/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 13:36.18 | datacompboy | or, maybe anyone knows working MCU for sip? i have tried openvcs, and get mixed video OK, but video have about 1 second lag (that is ok for me), but sound lag 5 second is completely not accepted :( |
| 13:36.29 | Cadey | Hi guys, with MixMonitor you can amplify the sound, can you do the same with Monitor - I dont see the option on teh wiki only flags |
| 13:37.07 | datacompboy | Cadey: you can run any external software after monitor finished |
| 13:37.27 | Cadey | Im running sox and lame to murge and convert the files |
| 13:37.38 | Cadey | do any of those have such a feature? |
| 13:37.44 | datacompboy | Cadey: dox can do amplify/normalize sound |
| 13:37.58 | Cadey | kk |
| 13:40.38 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 13:45.44 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 13:48.30 | Cadey | hum |
| 13:48.38 | Cadey | does anyone use sox to amplify? |
| 13:49.19 | Cadey | I just read a little about it and apparenty gain -n -3 should help normalise and amplify the file |
| 13:49.28 | Cadey | only the file is still really really quiet |
| 13:50.09 | datacompboy | Cadey: hmmm... i have used normalization |
| 13:50.38 | Cadey | is putting gain -n not correct? |
| 13:51.07 | datacompboy | Cadey: how exactly your commandline looks? |
| 13:51.22 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 13:51.32 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 13:52.23 | Cadey | "/usr/local/bin/sox -M {$RAMLeftChannel} {$RAMRightChannel} {$RAMMixFile} gain -n -3" |
| 13:52.54 | datacompboy | you should put gain -n -3 before {$RAMMixFile} |
| 13:53.03 | Cadey | kk ill try :) |
| 13:53.04 | datacompboy | since it should be applied before save |
| 13:53.45 | Cadey | I didnt think the order of options would matter as all params get loaded at initalise but ill try |
| 13:54.06 | datacompboy | Cadey: all options supplied affects on file next after them |
| 13:54.22 | Cadey | oh right |
| 13:54.37 | datacompboy | at least, format options. may be effects not |
| 13:54.41 | datacompboy | let me read man... |
| 13:54.50 | Cadey | putting in there has faile the script :) |
| 13:55.23 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:55.24 | datacompboy | try: "/usr/local/bin/sox --norm -M {$RAMLeftChannel} {$RAMRightChannel} {$RAMMixFile}" |
| 13:55.42 | Cadey | tryinh |
| 13:56.52 | Cadey | might be a bit louder |
| 13:56.58 | Cadey | but its still really quiet :( |
| 13:57.23 | Cadey | I have to turn the volum up on the speakers to half and if I dont remember to turn them back down and play some music it about blows them :( |
| 13:58.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:58.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:58.47 | datacompboy | hmmm... show me please output of "sox yourfile.wav -e stat -v" |
| 13:59.05 | Cadey | kk |
| 13:59.55 | Cadey | sox FAIL sox: --encoding: `stat' is not one of: signed-integer, unsigned-integer, floating-point, ms-adpcm, ima-adpcm, oki-adpcm, gsm-full-rate, u-law, mu-law, a-law. |
| 13:59.57 | Cadey | humm |
| 14:00.05 | Cadey | im using wav in the monitor app |
| 14:00.56 | Cadey | what was stat supposed to be because its thinking stat is the encoding I want? |
| 14:01.06 | datacompboy | what version of sox you have? |
| 14:01.28 | Cadey | 14.3.2 |
| 14:01.47 | Cadey | should be the latest I belive |
| 14:02.11 | Cadey | yeah it is |
| 14:02.26 | datacompboy | ok, then "sox yourfile.wav /dev/null stats" |
| 14:03.44 | Cadey | now it saying it cant open input file, arggg |
| 14:03.51 | Cadey | sorry |
| 14:03.53 | Cadey | my fail |
| 14:04.16 | Cadey | sox FAIL formats: can't determine type of `/dev/null' |
| 14:04.33 | datacompboy | pffr... |
| 14:04.58 | Cadey | what are you trying to find out dude, the type of file im putting into sox? |
| 14:05.20 | datacompboy | :) i'm really stupid today. "sox yourfile.wav -n stats" |
| 14:05.33 | Cadey | arrr |
| 14:05.34 | datacompboy | most intresting is "Pk lev dB" |
| 14:05.36 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:05.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:05.45 | Cadey | Pk lev dB -0.17 |
| 14:06.20 | datacompboy | hm... it looks like already normalized. |
| 14:06.38 | Cadey | it was the in file from a monitor output |
| 14:06.44 | datacompboy | what RMS lev ? |
| 14:06.53 | Cadey | RMS lev dB -20.76 |
| 14:07.07 | datacompboy | ahha... there was in recording some peak |
| 14:07.22 | datacompboy | this why file is not normalized -- it get normalized across peak, not RMS |
| 14:07.37 | Cadey | im new to this :) ? |
| 14:08.18 | datacompboy | then, you need to use "/usr/local/bin/sox -M {$RAMLeftChannel} {$RAMRightChannel} {$RAMMixFile} gain -B -3" |
| 14:08.22 | datacompboy | please try this way |
| 14:08.31 | Cadey | kk |
| 14:09.53 | Cadey | still really quiet :( |
| 14:10.17 | datacompboy | hmm... what now stats said on that output file? |
| 14:10.18 | Cadey | i took out --norm like your above line |
| 14:11.05 | datacompboy | ooops, sorry, i forgot -n |
| 14:11.12 | datacompboy | "/usr/local/bin/sox -M {$RAMLeftChannel} {$RAMRightChannel} {$RAMMixFile} gain -Bn -3" |
| 14:14.03 | Cadey | :( |
| 14:14.29 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:14.56 | datacompboy | Cadey: could you send me sample sound file? |
| 14:15.16 | Cadey | the wav or mp4 after lame has had it way? |
| 14:15.29 | datacompboy | Cadey: wav |
| 14:15.32 | Cadey | kk |
| 14:16.22 | *** join/#asterisk m4xmr (~m4xmr@93-36-129-84.ip60.fastwebnet.it) |
| 14:17.35 | Cadey | they have been sent dude |
| 14:18.15 | Cadey | brb making a coffee - only be 2 mins |
| 14:19.48 | m4xmr | hi, i'm trying to configure my pbx to have the busy signal on the extensions "on the phone"..., i used the call-limit=1 but now the problem is that, if an extension put the channel on hold an try to call someone it couldn't :( Do you have some idea, please? |
| 14:20.36 | Cadey | well the phone needs to line to make the other call when the first line is on hold |
| 14:20.53 | m4xmr | yes |
| 14:20.55 | Cadey | is there a way on your phone to only have 1 line accept incomming calls? |
| 14:21.39 | m4xmr | this could be a good idea, i have to verify..., i talking about sipura 942 |
| 14:22.00 | m4xmr | isn't there a way passing thru asterisk? |
| 14:22.02 | m4xmr | some IF clause.., maybe? |
| 14:22.19 | m4xmr | i'm looking for a way to handle this problem in the extensions.conf |
| 14:23.23 | Cadey | um yeah |
| 14:23.33 | Cadey | you could use the asterisk db for that i guess |
| 14:23.46 | Cadey | on a new call put in a variable like Extension and on call |
| 14:23.50 | Cadey | so like, 3000, 1 |
| 14:24.08 | Cadey | and before calling 3000 again check to see if the db variable is a 1, if it is they are busy |
| 14:24.46 | *** join/#asterisk orn (~orn@178.19.49.1) |
| 14:25.10 | Cadey | for example |
| 14:25.32 | Cadey | if (DB(3000) != 1) |
| 14:25.49 | m4xmr | this could be a good solution, if i used a DB ;) |
| 14:25.51 | Cadey | and set it with Set(DB(3000)=1) |
| 14:26.03 | Cadey | asterisk has a built in database m4xmr |
| 14:26.06 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 14:26.17 | m4xmr | Cadey, sorry missunderstanding |
| 14:26.43 | m4xmr | nice, this is a good idea Cadey, thanks |
| 14:26.49 | Cadey | http://www.voip-info.org/wiki/view/Asterisk+func+db |
| 14:27.10 | Aut0ExeC | hi guys... does Congestion() generate something like a disconnect tone? |
| 14:27.11 | Cadey | just remember to clear the value aftr the dial so when the call ends it clears the 1 |
| 14:27.47 | Aut0ExeC | I cant get the correct disconnect tone for my ATA so I have ot try and find another way... my calls are not hanging up when disconnected |
| 14:27.52 | _Corey_ | Aut0Exec: Yeah, it's a fast-busy tone |
| 14:28.05 | m4xmr | Cadey, this could be a problem if occurs a non standard hangup |
| 14:28.06 | Aut0ExeC | _Corey_: do you think that will work instead of using my ATA's? |
| 14:28.22 | _Corey_ | you may want to take a look at the Hangup() command though for codes |
| 14:28.33 | Aut0ExeC | ok thanks |
| 14:28.46 | _Corey_ | Running hangup with the right argument will cause other equipment to recognize the hangup |
| 14:28.53 | _Corey_ | otherwise you're just playing a fast-busy noise |
| 14:28.58 | Cadey | m4xmr : this is true but can you use the t extension to catch that? |
| 14:29.08 | Cadey | is it the t extension... I forget |
| 14:29.38 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-vxukeyguuwuwmajv) |
| 14:29.54 | m4xmr | Cadey, I have to try this, thanking for help |
| 14:30.23 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 14:30.46 | Aut0ExeC | _Corey_: i'm using a cisco spa3102 and it fails miserably at detecting disconnects |
| 14:30.55 | E-bola | If i have 2 SIP provider accounts i can dial out with, what options do i have to let the users select which one to use? |
| 14:31.05 | Aut0ExeC | _Corey_: i figure I might be able to simulate on the asterisk side |
| 14:31.07 | _Corey_ | is that acting as FXO or FXS for you? |
| 14:31.25 | Aut0ExeC | yes |
| 14:31.31 | E-bola | I culd setup up 2 accounts on each telephone which would map to two different outgoing nubmers, but i guess i could also let them press 1 or 2 infront of the number they are dialing? |
| 14:31.33 | Aut0ExeC | FXO |
| 14:31.51 | _Corey_ | so you're connecting POTS lines into this thing and providing service to Asterisk? |
| 14:35.16 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 14:35.32 | Aut0ExeC | _Corey_: yes |
| 14:35.56 | _Corey_ | Well, if your problem is the ATA not detecting far-end hangup properly Asterisk won't help |
| 14:36.05 | Aut0ExeC | ok |
| 14:36.14 | _Corey_ | if you're having trouble detecting when someone on Asterisk hangs up, then the Hangup code will help |
| 14:36.38 | Aut0ExeC | k |
| 14:36.48 | Aut0ExeC | yeah I'm using SPA3102 |
| 14:36.57 | Aut0ExeC | apparently this is a common problem |
| 14:36.58 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 14:37.03 | Aut0ExeC | finding the exact code for ur region |
| 14:38.18 | Aut0ExeC | _Corey_: are digium cards better at this? |
| 14:40.00 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 14:42.37 | _Corey_ | In my experience, the Digium cards work fine |
| 14:44.56 | Aut0ExeC | I just found out that I have to request that feature to be turned on at my telco ... lol sux |
| 14:46.53 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 14:47.36 | *** part/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 14:49.36 | *** join/#asterisk Sipster (~Sipster@modemcable115.15-200-24.mc.videotron.ca) |
| 14:52.29 | Secret_Hamster | http://pastebin.com/0xsKntgM |
| 14:52.55 | Secret_Hamster | as explained much earlier I'm having issues with the dialpland and AGI script in the pastebin |
| 14:53.02 | *** join/#asterisk wikki (wikki@68.64.241.250) |
| 14:53.35 | Secret_Hamster | it appears that if the far end keeps open the line, the system carries on down the dialplan |
| 14:54.05 | *** join/#asterisk moy (~moy@CPE003048b11058-CM00222d6b4d65.cpe.net.cable.rogers.com) |
| 14:54.14 | Secret_Hamster | as far as I know this did not used to happen, I'm unsure of when this started to happen, but sometime ago now that we do not know what the change was |
| 14:55.20 | Secret_Hamster | we have other dialplans that follow a similar method but do not appear to have the same issue |
| 14:56.05 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 14:59.07 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-67-213.zg3.cable.xnet.hr) |
| 15:00.53 | pecenipicek | Good day folks, got a question regarding CDR generation in the Asterisk 1.6 line. Is it possible to catch transfers and similar properly and get it to show up in CDR's, or is 1.8's CEL a better bet overall for that? |
| 15:01.28 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:01.44 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 15:02.31 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:05.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:06.29 | *** join/#asterisk m4xmr (~m4xmr@93-36-129-84.ip60.fastwebnet.it) |
| 15:07.37 | m4xmr | _Corey_: Your advice is really good but probably I'm to tired do achieve the result... |
| 15:08.16 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-iwpmkiukabpesusz) |
| 15:08.18 | m4xmr | the state of an extension should be modified when it starts a call and when it recieves a forward |
| 15:08.38 | zkn | Hi, so I have the following obstacle: I need to catch SIP calls which enter the system in format peer@systemname, by using CUT function I can cut the @systemname part and send the preceding part to Dial app, the question is how to best catch these calls? |
| 15:08.43 | m4xmr | this seems to be a problem, if i want to manage with that |
| 15:08.46 | *** join/#asterisk slim_ (c4dbe06b@gateway/web/freenode/ip.196.219.224.107) |
| 15:09.10 | slim_ | hello all, |
| 15:10.51 | slim_ | can i use asterisk as a sip server for chatting ( audio/video/text ) only ? |
| 15:11.40 | JerJer | zkn: unless i am missing something doesn't ${EXTEN} contain what you need already ? |
| 15:11.51 | Aut0ExeC | slim_: yeah thas what I wanted to know as well... |
| 15:12.29 | zkn | JerJer: nope, exten will contain peer@systemname, but I need to be able to dial SIP/peer |
| 15:12.38 | zkn | i was trying with exten => _.[@]systemname,1,Set(SIPEXTEN=${EXTEN}) but this is bad |
| 15:12.49 | Qwell | . must be at the end |
| 15:13.11 | JerJer | slim_: i wouuld look at kamailio for simple |
| 15:14.32 | slim_ | Aut0ExeC: JerJer thanks for reply, ... |
| 15:15.11 | slim_ | i search for a sip server that can integrate with MS lync server |
| 15:15.16 | zkn | exten => _[a-z][@]systemname,1,Set(SIPEXTEN=${EXTEN}) would be better but does not work |
| 15:15.32 | zkn | maybe there is smth wrong in that picture, too |
| 15:15.41 | Qwell | several things, yes |
| 15:16.41 | JerJer | zkn: you are trying do to blah@foo@asterisk.ip.address ???? |
| 15:17.11 | JerJer | as the sip uri ? |
| 15:17.13 | zkn | no.. just user@asteriskhostname |
| 15:19.31 | zkn | sipuri is empty |
| 15:20.24 | JerJer | so ${EXTEN} should simply be 'user' |
| 15:20.39 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:21.14 | zkn | it needs to become that, yes, and by using CUT i can get that but i would say the biggest problem is patternmatching with _. |
| 15:21.43 | JerJer | it is 'user' |
| 15:21.59 | JerJer | already |
| 15:22.38 | zkn | yes but i don't have user@systemname extensions created |
| 15:22.59 | JerJer | if you dial sip:bob@1.2.3.4 asterisk will set ${EXTEN} to 'bob' |
| 15:23.55 | zkn | nope, does not happen for me |
| 15:24.18 | JerJer | which version? |
| 15:24.21 | zkn | 1.8.3 |
| 15:24.39 | zkn | i don't think i have had that with any of the previous versons either |
| 15:24.57 | JerJer | of asterisk |
| 15:25.12 | zkn | yes |
| 15:26.00 | JerJer | then this behavior has changed!? I am looking at a 1.6 system right here and it uses alpha based extens an there is no @systemname |
| 15:27.55 | JerJer | are you setting systemname is sip.conf ? i'm checking out a 1.8 tree now |
| 15:29.08 | zkn | i am receving incoming calls through AMI, i have two fields set in an external CRM system which contain SIP address (user@asteriskserver) and extension (user) when I dial the extension, then as it also exists in my dialplan, the user is rung, I want to get the SIP address also to ring the user, so for that I either create seaparate set of extensions which contain also the asteriskserver/systemname part of the convention or I CUT the extra part off so I will |
| 15:29.08 | zkn | ring the user |
| 15:29.23 | zkn | systemname I set in asterisk.conf |
| 15:30.07 | *** join/#asterisk garymc (~chatzilla@host81-139-133-67.in-addr.btopenworld.com) |
| 15:31.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:31.38 | JerJer | ok - then the behavior has changed. I'm mostly stuck in 1.4 in old systems and have only starting deploying 1.6 on new systems in the last few months |
| 15:32.29 | JerJer | (or AMI is adding it) |
| 15:32.54 | zkn | well, by patternamtching with _. i catch too much how would I get a-z to patternmatch? e.g. _a-z@asteriskserver |
| 15:33.22 | zkn | a-z representing any alphabeticical username |
| 15:33.53 | JerJer | hmm brackets ? _[aA-zZ] |
| 15:34.12 | JerJer | <PROTECTED> |
| 15:35.13 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 15:35.17 | zkn | exten => _[aA-zZ][@]asteriskserver,1,Set(SIPEXTEN=${EXTEN}) |
| 15:36.10 | ujjain | Hi, I get this error in Asterisk ¨Unknown RTP codec 126 received from¨. Should I allow this codec in Asterisk or change X-lite audio codec settings? |
| 15:36.50 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 15:39.57 | zkn | _[aA-zZ] does not patternmatch usernames for me :( |
| 15:40.22 | zkn | ignores completely |
| 15:40.34 | zkn | 1 failed so falling back to exten 's' |
| 15:41.52 | JerJer | not sure then :( |
| 15:43.42 | *** join/#asterisk sahafeez (~sahafeez@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 15:44.50 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
| 15:46.42 | zkn | ffs, it does not even react to [0-9] |
| 15:48.28 | zkn | but X outside the brackets works |
| 15:49.10 | orn | does _[aA-zZ]. work? |
| 15:49.18 | orn | Didn't you leave out the period? |
| 15:49.24 | orn | or is the username just a single character? |
| 15:49.53 | zkn | no, not a single character, length veries |
| 15:50.03 | orn | then the period is missing |
| 15:50.10 | zkn | ok, let's try |
| 15:50.12 | orn | _[aA-zZ]. |
| 15:50.34 | _Corey_ | orn: If it could match a single character, use !, . assumes one more |
| 15:50.52 | orn | good point |
| 15:54.34 | zkn | _[aA-zZ]. did not work |
| 15:55.38 | zkn | ok, gotta run now, will try again later |
| 15:55.42 | zkn | thanks for the input! |
| 15:57.48 | *** join/#asterisk tstorm (~tstorm@173-164-230-21-SFBA.hfc.comcastbusiness.net) |
| 16:02.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:04.42 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 16:10.20 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:17.24 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:17.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:47.57 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 16:50.02 | *** join/#asterisk voila (IceChat77@117.212.71.104) |
| 16:50.11 | voila | hello guys . i m new to asterisk .. |
| 16:50.32 | voila | can i validate my dialplan thorugh soft phones ?? |
| 16:50.36 | voila | through* |
| 16:50.38 | Qwell | I wish I knew how to play the voila. |
| 16:50.52 | voila | lol |
| 17:01.12 | voila | anybody here |
| 17:03.44 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 17:04.39 | paulc | viola: Yes, and yes - you can test your dialplan using a soft phone - it just needs to be pointing to the right context(s) in sip.conf |
| 17:06.39 | voila | paulc: see i configure 3001 user account in my sosftphone , then as documented i dial 1000 to go to main ivrs menu it works . But now i want to make my ownd dialpan instead of using default one .. |
| 17:06.50 | voila | how can i do it ??? |
| 17:07.35 | paulc | voila: Make a new context, put the dialplan logic you want in it (in extensions.conf), then set your 3001 sip peer to use that context (in sip.conf) |
| 17:08.21 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 17:08.40 | voila | ok thanks , just one weird thing more .. i delelte everything from "extension.conf" , then i dial 1000 , still i m getting response from ivrs , how it is possible ??? its strange |
| 17:09.02 | Qwell | because you didn't reload? |
| 17:09.16 | voila | no i didn't reload |
| 17:09.20 | voila | oh ok .. |
| 17:09.27 | voila | let me try |
| 17:09.35 | voila | i have backup of extension.conf |
| 17:10.08 | voila | Qwell: yaa right .. i ddin't get anything |
| 17:10.20 | voila | thanks both qwell and paulc |
| 17:14.39 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 17:31.41 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:32.48 | Tuju | hi - I'm having problems to upgrade firmware to cisco 7975 phone - it fails with "Upgrading..... Auth Fail." <reset!> |
| 17:33.09 | longword | You're sure you want to upgrade? |
| 17:33.33 | Tuju | yes - especially now when this phone doesn't have a firmware anymore. |
| 17:33.47 | longword | Ah, I can see how that could be a problem. |
| 17:34.12 | Tuju | it has some kind of bootstrap that tries to load new firmware. |
| 17:34.15 | longword | I don't have any special insight there, but check the files on the TFTP server |
| 17:34.33 | Tuju | yes, i've been doing it couple days already :) |
| 17:35.07 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 17:35.15 | longword | One of them didn't accidentally get truncated or overwritten? |
| 17:35.22 | Tuju | i've couple theories - it could be caused by 802.1x or then by server MTU being 1500 which caused some issues with skinny sccp firmware releases. |
| 17:35.44 | Tuju | nope, i've been trying many of them |
| 17:35.49 | Tuju | and i'm not alone with his |
| 17:35.51 | Tuju | and i'm not alone with this |
| 17:35.52 | longword | Bump up the logging on the TFTP server, maybe try packet dumping |
| 17:36.01 | longword | Packet sniffing even. |
| 17:36.16 | Tuju | the problem is in phone eth interface |
| 17:36.31 | Tuju | i even implemented 802.1x authentication and got it working |
| 17:40.08 | *** join/#asterisk Pegasus_RPG (~chatzilla@rrcs-24-39-148-52.nyc.biz.rr.com) |
| 17:40.13 | Tuju | which is actually pretty kinky system. i recommend. |
| 17:43.38 | Pegasus_RPG | hello there. I've been thrown headfirst into the fun world of * behind NAT with an endpoint (phone) behind another NAT across the Internet. Long story short, do both routers need to support Symmetric NAT in order for it to work? |
| 17:45.56 | voila | i m trying to play mp3 file . like this ..... " exten => s, n , Background(c:\users\abc\amit.mp3) " ... its is not working , why ?? |
| 17:49.24 | serafie | voila: did you build Asterisk with MP3 support? |
| 17:49.44 | serafie | also, remove the file extension? |
| 17:49.49 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 17:50.07 | voila | serafie : i m suing asteriskwin32 , so i install it by default .. |
| 17:50.10 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:50.15 | lordvadr | Hello, I have two asterisk servers in a test enviroment, both running 1.6.2.17 and using iax2. One of them has 1 g729 license, the other has 4 licenses. If I set allow=g729,ulaw on each, the first call sets up g729 properly. The second call tries and fails. Is there a way to get it to fall back to another codec? |
| 17:54.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-175.mobile.uci.edu) |
| 17:55.57 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 17:59.52 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 18:00.01 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:859b:7080:580d:8fe3) |
| 18:03.41 | KavanS | any tips/concerns with asterisk presence and polycoms on asterisk 1.4? |
| 18:03.53 | KavanS | interested to setup a receptionist to monitor a couple "buddy" lines - have never done this |
| 18:03.59 | KavanS | is this hard to do? and/or a headache? |
| 18:04.49 | voila | http://pastebin.com/zudJnnca .. why i m getting error even in a very simple plan .?? |
| 18:10.01 | lordvadr | voila: it's your phone. Nothing to worry about |
| 18:10.28 | lordvadr | the phone is telling asterisk that it supports video, and asterisk doesn't like it |
| 18:11.44 | voila | lordvadr : but point is .. why i get "destination not found" (softphone display this) , even background is also not played |
| 18:14.14 | lordvadr | what are you dialing |
| 18:14.16 | lordvadr | ? |
| 18:15.39 | voila | lordvadr : point is i m learning , how to create dialplan , so in extension.conf , i see [demo] , context , i copy the part of this context under [default1] context , create new user in sip.conf , which is using this context . |
| 18:15.45 | voila | i m dialing 1000 |
| 18:16.10 | lordvadr | there isn't an extension for 1000 |
| 18:16.27 | *** part/#asterisk Pegasus_RPG (~chatzilla@rrcs-24-39-148-52.nyc.biz.rr.com) |
| 18:16.39 | lordvadr | put this in there anywhere... |
| 18:17.02 | lordvadr | exten => 1000,1,Goto(s,1) |
| 18:18.37 | voila | ohh ok |
| 18:19.06 | voila | let me try .. well thanks |
| 18:22.05 | lordvadr | voila: 's' is a predefined extension, means 'start'. It's the extension that a context uses if it doesn't have anything else to go on. |
| 18:22.08 | lordvadr | http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 18:22.09 | lordvadr | read that |
| 18:27.05 | voila | lordvadr : thanks |
| 18:30.56 | _Raptor_ | can i do this in extensions conf:http://pbot.icip.de/c6a1e2cfa7b02ae69ba3c31100947251 |
| 18:33.32 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 18:39.16 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 18:39.53 | serafie | _Raptor_: why not use a priority label? |
| 18:47.05 | serafie | _Raptor_: if you implement that and then run "dialplan show context", you will see that not only does Asterisk skip adding your explicit "2" priority (5th line), but it also skips the line after it. You can mix numerals and "n" for priorities, but all your numerals have to come first in order. Otherwise, you're going to see unexpected behaviors. |
| 18:47.30 | _Raptor_ | serafie: i see, thank you |
| 18:47.42 | _Raptor_ | serafie: i'll use numbers at all lines, thx |
| 18:48.40 | serafie | _Raptor_: you can use all "n"s (except 1 of course) and use a label like "ext,n(mylabel),App()" and then go there by doing GoTo(mylabel) |
| 18:49.16 | _Raptor_ | oh cool, i didn't know that |
| 18:51.05 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 18:51.29 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 18:54.23 | voila | Guys . i m done with simple dialplan .. now how can i record all calls using asterisk ?? Thnaks lordvadr |
| 18:54.27 | voila | thanks* |
| 18:55.14 | serafie | voila: https://wiki.asterisk.org/wiki/display/AST/Application_Monitor |
| 18:55.22 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:56.04 | serafie | wiki.asterisk.org is your friend. |
| 18:57.09 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:00.43 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 19:02.34 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 19:03.59 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 19:05.19 | voila | serafie: thanks .. but when i use Monitor , i m getting this eror "Could not create file /asterisk/spool/monitor/amit -n " .. why is it so ?? any idea ? |
| 19:05.41 | serafie | permissions? |
| 19:05.55 | serafie | Or check the file locations in asterisk.conf |
| 19:05.59 | serafie | Aren't you on Windows? |
| 19:06.28 | voila | serafie: i m using windows |
| 19:06.40 | voila | serafie : let me check conf file |
| 19:07.20 | *** join/#asterisk Sipster_ (~Sipster@modemcable115.15-200-24.mc.videotron.ca) |
| 19:07.42 | voila | serafie: in asterisk.conf .... astspooldir = /etc/spool is given |
| 19:07.57 | serafie | Looks like a Linux path to me. |
| 19:08.05 | serafie | Change it to something Asterisk has permissions to write to. |
| 19:08.11 | serafie | and then reload |
| 19:08.38 | voila | serafie : ok .. and i think i should give the absolute path .. like c:\users\voila\..... .. should i ??? |
| 19:09.01 | serafie | I would assume so, but I have absolutely no experience with Asterisk on Windows. |
| 19:09.14 | voila | serafie : ok .. let me hit and try |
| 19:09.17 | voila | :) |
| 19:11.46 | *** join/#asterisk Trapa (~Trapa@S01060017c596c06f.vc.shawcable.net) |
| 19:12.28 | Trapa | Hi everyone, most of the time I come here for asterisk CLI support, but i'm wondering Is this the place or is there a better place for support for a switchvox box and it's web GUI? |
| 19:13.38 | Trapa | If this is the place, Here's my Issue. I am on-site at a clients site, All call QUEUES, are getting the ringing, but whenever someone answers the phone the phone immidately says "Call ended" |
| 19:14.40 | Trapa | Some background, with the 'reputable' business person who provided the system for these guys, it would not surprise me if they stuck in a sabbatoge into the phone system to prevent calls from getting through (All incoming calls are routed to a reception queue.) |
| 19:16.56 | _Corey_ | Trapa: Pretty hard to sabotage Switchvox... it's not really open |
| 19:17.01 | voila | serafie: i m able to create file with monitor, if i donot rename it and change its format .. Monitor(wav , "amit") ... is this right way to change the format and name of the file ???? |
| 19:17.33 | _Corey_ | Trapa: Call Switchvox support though, you're likely to get help |
| 19:19.24 | Trapa | Corey: I'm thinking the guy just remoted in and set a variable or permission somewhere denying members of the queue permission from doing it. if i could access the asterisk cli i could watch what is happening. |
| 19:21.22 | _Corey_ | Trapa: There is a diagnostics screen where you can turn on a "capture" for X minutes and then download and view it |
| 19:21.29 | _Corey_ | You can't watch the CLI interactively |
| 19:21.39 | serafie | voila: syntax should be on that wiki page I linked to |
| 19:21.50 | _Corey_ | best bet is to do that for a couple minutes with all the debugging and AGI debugging turned on, then place your test calls |
| 19:21.55 | Trapa | Now thisis interesting, I'm looking at the switchvox call log. It's a little different than standard CLI but here's what it's saying. Incoming call, Rang Reception Queue, Call was hung up by "Joe Smith"<xxxxx> ... |
| 19:21.56 | _Corey_ | You should see a lot of info |
| 19:22.08 | Trapa | So the system is telling the log that the incoming caller hang up the call. |
| 19:22.09 | *** join/#asterisk lesouvage (~lesouvage@524947AA.cm-4-2b.dynamic.ziggo.nl) |
| 19:22.34 | Trapa | (even if i'm dialing the queue from another internal extension) (While internal extension dialing is working just fine) |
| 19:22.54 | _Corey_ | Check the debugging, maybe you'll see what it is |
| 19:23.57 | Trapa | Corey, .. sounds good .. oh interesting .. error log "Disk is full" a call recording has been aborted due to insuffiecent disk space. |
| 19:24.43 | Trapa | Now the question becomes, how in the hell do i delete old recordings. |
| 19:25.06 | tvc123 | Trapa: cron+find? |
| 19:25.14 | _Corey_ | The recordings screen lets you set thresholds for that stuff so your disk can't fill up |
| 19:25.18 | _Corey_ | try there |
| 19:25.31 | Trapa | Cool thanks, Looking now. |
| 19:25.55 | Trapa | I much prefer the cli .. but for what switchvox is doing their gui is pretty darn nice. |
| 19:26.28 | _Corey_ | seriously... makes FreePBX look like junk |
| 19:26.30 | _Corey_ | ;) |
| 19:29.40 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:29.42 | *** join/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 19:33.17 | *** part/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 19:36.52 | lesouvage | . |
| 19:38.35 | *** join/#asterisk Sipster (~Sipster@modemcable115.15-200-24.mc.videotron.ca) |
| 19:44.13 | *** join/#asterisk becks` (54e319fe@gateway/web/freenode/ip.84.227.25.254) |
| 19:44.31 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 19:44.43 | Aut0ExeC | google talk is availabel from 1.4 right? |
| 19:44.54 | becks` | hi, how can I stop my snom to automatically send out a NOTIFY whenever I pick up the phone? |
| 19:48.18 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 19:49.29 | serafie | Aut0ExeC: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 19:51.55 | Aut0ExeC | serafie: ok thanks... so can I set this up in 1.4? |
| 19:52.43 | Aut0ExeC | i dont understand why it said it needs 1.8 when in 1.4 all the config files are there |
| 20:00.26 | becks` | nobody knows how to disable all those NOTIFY the snom sends when using a line? :S |
| 20:02.33 | *** join/#asterisk voila (IceChat77@117.212.71.104) |
| 20:02.36 | voila | hi |
| 20:03.13 | voila | which command in asterisk enable me to send sms to softphones ??? |
| 20:11.20 | WIMPy | Does TIMEOUT(absolute) use te dialedtime or the answeredtime? |
| 20:22.40 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:23.22 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 20:31.01 | *** join/#asterisk sahafeez (~sahafeez@4.53.128.213) |
| 20:34.36 | justdave | what's the easiest way to figure out what my average and peak concurrent calls is? |
| 20:34.54 | justdave | are there well-known scripts anywhere to parse that kind of data out of the cdr? |
| 20:35.12 | justdave | (I imagine you'd have to do some crunching of the data to figure that out) |
| 20:39.01 | atan | Hmm, anyone have sipdroid setup to support video with Asterisk? |
| 20:40.11 | bmoraca_work | justdave: asterisk-stat-v2 is what i use |
| 20:40.26 | justdave | bmoraca_work: thanks, got a url? |
| 20:40.34 | bmoraca_work | no, but google does |
| 20:40.41 | justdave | ok, will look around |
| 20:41.17 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 20:41.22 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 20:41.54 | justdave | found it, that looks like it'll do the trick. thanks! |
| 20:41.57 | *** join/#asterisk Tecy (~Tagor@s55978a13.adsl.wanadoo.nl) |
| 20:42.38 | Tecy | Does anyone know a free flash or java script which visitors of my website can use to call me? |
| 20:44.04 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:56.14 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:07.54 | Freeaqingme | Tecy, I dont know such thing, but it would probably be a java applet |
| 21:15.57 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 21:19.12 | serafie | Tecy: you could do it by manipulating call files, but that functionality in general seems like a great way to introduce security issues. |
| 21:19.38 | serafie | or just annoying behavior in general. |
| 21:20.53 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 21:29.27 | *** join/#asterisk ccesario_ (~ccesario@189.111.7.213) |
| 21:31.14 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 21:37.05 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.191.92.49) |
| 21:40.23 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:48.05 | *** join/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 21:48.23 | *** part/#asterisk sjobeck (~sjobeck@206.83.218.66) |
| 21:59.26 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 22:02.56 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 22:11.32 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 22:30.52 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:42.43 | *** join/#asterisk digilink (~digilink@70.90.228.197) |
| 22:58.24 | *** join/#asterisk giesen (giesen@204.14.17.135) |
| 22:58.57 | giesen | Anyone have any experience getting res_snmp working on trixbox? I can get the module to load, and snmpd is configured with agent support, not able to snmpwalk the mib |
| 22:59.32 | giesen | bah never |
| 22:59.37 | giesen | freaking thing is working now =) |
| 23:07.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:07.51 | Greek-Boy | giesen: what are you using for monitoring? |
| 23:08.49 | giesen | Solarwinds Orion NPM |
| 23:08.51 | giesen | <groan> |
| 23:11.45 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 23:15.22 | Greek-Boy | isee |
| 23:20.31 | *** join/#asterisk StaRetji (~BigAll@91.143.222.166) |
| 23:22.19 | *** join/#asterisk m4xmr (~m4xmr@93-34-195-94.ip51.fastwebnet.it) |
| 23:25.46 | m4xmr | hi, I'm lookin for a solution to handle the busy signal for all those extensions, that are already "on phone"..., i need to implement something as the BLF but for the audio, any idea? |
| 23:37.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:37.58 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.131) |
| 23:42.02 | m4xmr | don't you have any idea..., how can i handle the busy signal when an extension is already involved in a conversation "a channel" ...? |
| 23:43.06 | *** join/#asterisk candrews (~candrews@fsf/member/candrews) |
| 23:46.08 | candrews | I'm using CURL to get caller info like this: Set(CALLERID(name)=${CURL(http://192.168.0.101/callerid/callerid.php?format=basic&num=${CALLERID(num)})}) |
| 23:46.24 | candrews | I'm wondering - how long can that CURL request take before it screws things up? |
| 23:46.46 | candrews | Or is asterisk okay with it taking a long time, and just won't proceed to the next instruction (the Dial cmd in my case)? |
| 23:54.25 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |