| 00:00.07 | *** join/#asterisk serafie (~erin@75.76.38.159) |
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| 00:33.24 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 00:40.21 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zdnrieqznedgbtqy) |
| 00:47.48 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 00:49.12 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 00:49.13 | k1ng | How to configure freepbx for username based SIP authentication? I like to give softphone users a username such as "abc1234" and pw instead of their extension number. So, they can register their client to server. |
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| 01:48.15 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 01:48.28 | nickgaw | Hi, Does anyone here know of an apt repository for debian so I don't have to go threw the full hand install as asterisk is in the main debian repository but even with contrib and non-free no freepbx? |
| 02:04.02 | WIMPy | k1ng: Ask in #freepbx |
| 02:04.27 | WIMPy | nickgaw: Same |
| 02:07.27 | nickgaw | are there normal pc's that have been built for asterisk that I can buy? |
| 02:08.10 | WIMPy | That's a contradiction in itself, isn't it? |
| 02:08.26 | robl^ | just use a normal PC for a normal PC |
| 02:08.52 | WIMPy | Any PC should do. Or even anything else as long as it has the resources for what you plan to do. |
| 02:10.16 | robl^ | I've use Dell Power Edge server for high end. Intel Atom mobo/cpu barebones systems in generic cases, even a $35 Raspberry Pi. |
| 02:10.49 | nickgaw | what I meant is a computer that contains telephone cards preinstalled with both fso and fsx ports? |
| 02:11.27 | WIMPy | Several companies sell Asterisk based solutions. |
| 02:12.54 | robl^ | yes.. Rhino, Xorcom, Rockbochs (Phonebochs) are examples. Can't say anything about how goood they are or how well they work, as I've not personally used them |
| 02:14.12 | robl^ | there are many others as well.. just google a little ;-) |
| 02:14.31 | WIMPy | The software is probably questionable, the hardware should be pretty much standard everywhere. |
| 02:15.11 | nickgaw | are most asterisk distributions centos based are there no debian based ones or other versions of linux? |
| 02:16.29 | WIMPy | What do you want? If you wan t your favourite distro, go for it. If you don't want to do anything yourself, why do you care? |
| 02:16.39 | robl^ | asterisk doesn't really care. CentOS is common only due to being based on RedHat Enterprise Linux. You can run it on any Linux, run it on BSD even. |
| 02:17.42 | *** join/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 02:18.49 | nickgaw | what I am trying to find is a debian based either a .deb package or a updated distribution of linux either debian or centos with newest asterisk that supports google voice. I have looked into pbx in a flash but the kernel is old and lots of the components are outdated they even suggest not to upgrade your kernel or to the next centos release? |
| 02:19.59 | WIMPy | As we found out a while ago, you shouldn't trust RedHead kernels anyway. |
| 02:22.23 | nickgaw | true I am also blind and none of the centos kernels include any accessibility features like debian does. once something is installed I can always ssh into it do set it up but like the web interface provided all images have alt texts and freepbx 2.10 does not have good accessibility. I am setting up my own internet based phone system so I don't have to always have a land line in the house and can use my cell when needed to mak |
| 02:23.35 | WIMPy | As mentioned before, if you want FreePBX, ask in #freepbx. We don't (can't) support it here. |
| 02:25.54 | nickgaw | for just an asterisk version would using a netbook work provided I am just using sip phones? |
| 02:26.32 | WIMPy | It can even work with telephony interfaces. |
| 02:26.58 | nickgaw | how can that be done as you can't install any hardware into the netbook? |
| 02:27.22 | WIMPy | USB |
| 02:27.47 | SeRi | WIMPy: I am curious why we shouldnt trust RedHat's kernels? |
| 02:27.52 | nickgaw | so the normal telephone plugs into the adapter that is plugd into the usb port? |
| 02:28.48 | WIMPy | SeRi: Because we had someone here trying to use misdn for three day without any luck until he used a kernel from kernel.org. |
| 02:29.22 | WIMPy | Something I was used to way back when Suse used to be popular. I thought those days were long gone. Obviousely not :-( |
| 02:29.38 | SeRi | WIMPy: ah. Ok. Thanks for the info. |
| 02:29.51 | WIMPy | nickgaw: There are some USB interfaces available. |
| 02:30.17 | SeRi | I remember does days. I was a user of SuSe. I even bought it back than from compusa. Along with FreeBSD. |
| 02:30.34 | WIMPy | SeRi: And together with RH's history of shipping broken gcc versions, they are definitely on my to avoid list now. |
| 02:31.29 | SeRi | WIMPy: LOL. They put food on my table. But I agree with you. I work with it day in and day out and there is a reason why I dont use it at home. |
| 02:31.57 | *** part/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 02:32.46 | WIMPy | I have tried some other distros from time to time, but somehow I always ended up staying Slackware. No fancy shit that backfires. |
| 02:32.57 | SeRi | WIMPy: amen. |
| 02:33.04 | SeRi | What version you on? |
| 02:33.10 | WIMPy | current |
| 02:33.16 | SeRi | I am waiting for 14 so I can slap it on a SSD |
| 02:33.19 | SeRi | nice |
| 02:33.23 | SeRi | I stick with stable |
| 02:34.10 | SeRi | I am using 13.37 and when 14 comes out I will upgrade to an SSD and slap it there |
| 02:34.28 | WIMPy | On the way to 14 I ran into an pretty evil udev hiccup indeed. |
| 02:34.38 | SeRi | yes I am aware |
| 02:34.42 | SeRi | eject is one of them, |
| 02:34.54 | WIMPy | But I always upgrade my netbook first. |
| 02:35.00 | SeRi | but there is more. |
| 02:35.07 | SeRi | hehehe smart. |
| 02:35.27 | WIMPy | No, it failed to boot. So that was rather annoyiong. |
| 02:35.50 | SeRi | ouch. There is a long thread about issues like that at linuxquestions |
| 02:35.51 | WIMPy | But it was fixed the day after I tried it. |
| 02:36.03 | SeRi | cool |
| 02:36.14 | SeRi | I cant live on the edge that way. LOL |
| 02:36.20 | WIMPy | Yes, that's why I wouldn't normally install udev on a server. |
| 02:36.51 | WIMPy | Until I found out that you need it for Digium cards with HWEC. |
| 02:37.14 | SeRi | I see. |
| 02:37.39 | SeRi | well guys nice chatting. time for bed. good old days of staying up late are long gone. :( |
| 02:38.00 | jpsharp | Lightweight |
| 02:38.03 | WIMPy | It's already far too late. |
| 02:38.24 | SeRi | lol |
| 02:38.34 | SeRi | g/n guys. |
| 02:38.51 | WIMPy | zzZzZZ |
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| 05:53.10 | schmidts | good morning |
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| 06:02.51 | krotos | hi all |
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| 07:41.46 | dym | Morning fellows! Is it possible instead of having a MOH while beeing a queuememeber, to have just an orderly ringtone generated without having to record such a tone as MOH? |
| 07:43.02 | salz212 | hi all, is there a possibility of getting chopped off SIP X headers when passed to different proxies?. Or if they follow standard, it should pass X headers. |
| 07:43.04 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 07:44.16 | dym | you should check with your proxys maintainer |
| 07:44.21 | dym | its always possible |
| 07:46.14 | salz212 | lets say I do not know about the far end proxies.. then what should I do? |
| 07:47.57 | cusco | dym: Queue() has a option for that |
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| 07:58.24 | dym | cusco: thanks. |
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| 08:07.46 | n1x0n | Hello, I know it's not asterisk specific question - but anyone using it on debian sqeeze and upgraded to latest vesion from repo today ? Mine was fine for ~4 years and just broke with chan_sip not loading (unresolved symbols etc..) , was wondering if it's just mine machine then :) |
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| 08:21.17 | n1x0n | ignore ^ http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=688053 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=688062 |
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| 08:23.00 | schmidts | n1x0n have you recompiled asterisk after updates? or do you use a debian package for it? |
| 08:24.01 | schmidts | looks like there are some not recognized depandicies in the debian package |
| 08:28.02 | *** join/#asterisk LgK (~chatzilla@bureau.nirwana.fr) |
| 08:28.07 | LgK | hi |
| 08:29.09 | LgK | I have a little problem after upgrading my asterisk, chan_sip.soi do not load anymore |
| 08:29.45 | LgK | I have upgraded to the 1.6.2.9-2+squeeze7 on debian squeeze |
| 08:30.00 | LgK | Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: sip_pvt_lock_full <== someone have an idea to fix this ? |
| 08:30.38 | schmidts | LgK look here http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=688062 :D |
| 08:30.54 | schmidts | did you made a debian update lately? |
| 08:32.11 | LgK | I have exactly same problem as describe on this link |
| 08:32.34 | LgK | hmmm no upgrade, i have just apply the security fix on asterisk |
| 08:33.06 | schmidts | it looks like an unresolved debian dependecy after the upgrade, so you can try to redo your debian upgrade, or rebuild asterisk using your own sources |
| 08:34.17 | LgK | i have try to remove en readd asterisk but now debian always install the last version with this f****** fix :( |
| 08:35.12 | bulkorok | I know why I have a test-system :-) |
| 08:35.17 | salz212 | any one know.... which RPC asterisk use for user agent field? in sip? |
| 08:35.54 | bulkorok | what is RPC !? |
| 08:41.04 | dym | I just read a security announcement on the debian sec list saying that 1:1.6.2.9-2+squeeze7 fixed some DOS vulnerabilities on the AMI |
| 08:41.14 | dym | could have something to do with that |
| 08:41.25 | dym | LgK: Whats your version? |
| 08:41.26 | LgK | problem fixed temporaly |
| 08:41.28 | dym | The one you upgraded to? |
| 08:41.38 | LgK | i have downgraded to the previous version |
| 08:41.45 | dym | what was the one you upgraded to? |
| 08:42.12 | LgK | <PROTECTED> |
| 08:42.29 | fling | hello |
| 08:42.35 | LgK | if you install this version on debian squeeze, sip module don't load and asterisk don't work |
| 08:43.14 | LgK | need to uninstall asterisk and asterisk-config and reinstall the previous manually asterisk_1.6.2.9-2+squeeze6 and asterisk-config_1.6.2.9-2+squeeze6 |
| 08:43.43 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:46.47 | dym | Has been reported to list. |
| 08:46.56 | dym | I assume a fix will be built quite quickly. |
| 08:50.16 | fling | how to join this together? |
| 08:50.18 | fling | exten => _XXXXXX,1,Dial(SIP/${EXTEN}@metro74) |
| 08:50.23 | fling | exten => _8800.,1,Dial(SIP/${EXTEN}@metro74) |
| 08:50.39 | fling | I want to simplify my dialplan |
| 08:50.59 | zamba | anyone experienced with polycom sip phones and provisioning/upgrading software? |
| 08:51.17 | fling | kaldemar: hey :p |
| 08:51.38 | dym | fling: Check out http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 08:51.54 | fling | is checking |
| 08:52.38 | fling | is not # mean sharp? |
| 08:52.53 | dym | hash, sharp, whatever |
| 08:53.01 | dym | germans have 10 words for it :D |
| 08:53.13 | dym | (slightly exaggerated) |
| 08:53.32 | dym | fling: sharp is more in a musical sense :) |
| 08:53.43 | fling | Rautenzeichen |
| 08:54.01 | dym | Americans say pound-key |
| 08:54.03 | fling | dym: ok, I will call it Doppelkreuz |
| 08:54.22 | dym | Or Lattenzaun |
| 08:54.50 | fling | I will not ask about '*' |
| 08:55.05 | dym | People quite willingly accepted "star" :) |
| 08:55.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 08:56.09 | fling | dym: I have read the link and idk how to join '_XXXXXX' with '_8800.' |
| 09:00.33 | fling | dym: can I join it or not? |
| 09:02.22 | fling | ChannelZ: hello |
| 09:08.09 | *** join/#asterisk nismapk (nisam@223.226.0.64) |
| 09:08.56 | nismapk | hi |
| 09:09.39 | nismapk | hi experts |
| 09:10.03 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:10.15 | nismapk | i require one help |
| 09:12.50 | dym | fling: It depends if you have more numbers to be dialed. |
| 09:13.20 | dym | fling: you could wildcard with _X. that would match for both. (but then agian all numbers) |
| 09:13.58 | dym | _8800. means 8800 + anything dialed afterwards |
| 09:14.30 | dym | _XXXXXX matches any 6 digit number from values 0-9 |
| 09:17.22 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 09:30.36 | *** join/#asterisk nisam (nisam@106.204.4.81) |
| 09:31.14 | nisam | hi friends |
| 09:31.49 | cusco | hi friend |
| 09:32.06 | cusco | if you need help, ask |
| 09:32.16 | cusco | don't aks for help... ask your real question |
| 09:32.17 | nisam | can i ask one question |
| 09:32.30 | nisam | thanks |
| 09:32.39 | cusco | and next time, don't ask to ask, just ask |
| 09:34.32 | wdoekes | LgK: try: apt-get install 'asterisk=1.6.2.9-2+squeeze6' |
| 09:39.34 | *** join/#asterisk nisam (nisam@106.202.10.207) |
| 09:40.06 | *** join/#asterisk pietro (~pietro@81.174.3.44) |
| 09:40.09 | pietro | hello |
| 09:40.16 | pietro | is issues.asterisk.org down ? |
| 09:40.26 | wdoekes | no |
| 09:40.47 | nisam | my problem: whan i am installing asterisk, chan_mobile is not getting enabled.wat may be the pblm.? |
| 09:42.23 | nisam | any body there..? |
| 09:43.19 | nisam | hi friends |
| 09:44.08 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 09:44.19 | mcolombo | hi all |
| 09:46.57 | bulkorok | nisam: can you load the module in asterisk cli? |
| 09:47.32 | nisam | no..it is showing error |
| 09:48.27 | bulkorok | so... what does it tell you?! |
| 09:49.23 | nisam | bulkorok:Error loading module 'chan_mobile': /usr/lib/asterisk/modules/chan_mobile.so: cannot open shared object file: No such file or directory |
| 09:50.50 | nisam | i have given this command "module load chan_mobile" |
| 09:51.12 | fling | dym: so it is not possible? |
| 09:51.24 | bulkorok | type: module load [TAB] and check if chan_mobile is listed |
| 09:51.49 | mcolombo | anyone say how can i enable this line "Reason: SIP;cause=xxx;text="xxx"" in Asterisk's Cancel? |
| 09:52.42 | mcolombo | i don't find how can enable it in SIP cancel method.. |
| 09:53.12 | bulkorok | nisam: or check at first that chan_mobile.so is in the modules folder of asterisk (check asterisk.conf) and if the asterisk-process is allowed to read... |
| 09:53.23 | nisam | bulkorok: it is not there |
| 09:53.28 | nisam | how can i make it |
| 09:54.00 | bulkorok | nisam: do you compile asterisk from source? |
| 09:54.20 | nisam | yes..i installed from source |
| 09:54.47 | dym | fling: Have you read what i wrote? |
| 09:54.52 | pietro | is there no news about this issue: https://issues.asterisk.org/jira/browse/ASTERISK-19003 ? |
| 09:55.08 | bulkorok | then check via 'make menuselect' what dependencies are needed to compile the module |
| 09:55.26 | fling | dym: yes |
| 09:55.41 | dym | Doubtful |
| 09:56.33 | nisam | when i am doing make menuselect, chan_mobile showing XXX .and not showing any dependency details |
| 09:57.04 | bulkorok | go to the entrsy with the cursor... on the bottom you will see bluetooth |
| 09:57.31 | nisam | yes |
| 09:58.31 | nisam | i think i already installed bluetooth..because i am able to send file to my fone |
| 09:58.35 | bulkorok | so get bluetooth (maybe just the libs) and ./configure and make menuselect again |
| 09:58.48 | bulkorok | get the libs / devs |
| 09:58.55 | bulkorok | you need the sources... |
| 09:59.11 | nisam | wat is the command for that |
| 09:59.26 | bulkorok | what's your distro? |
| 09:59.36 | nisam | CentOS 5.5 |
| 09:59.41 | bulkorok | ew,,, |
| 09:59.49 | bulkorok | never used |
| 10:00.06 | bulkorok | check your packet-manager |
| 10:00.12 | nisam | oh god |
| 10:00.30 | bulkorok | it's yum ?! |
| 10:00.35 | bulkorok | I think |
| 10:00.38 | nisam | can you tell me pacage name |
| 10:00.45 | nisam | exactly..it is yum |
| 10:01.11 | bulkorok | google!? |
| 10:01.55 | bulkorok | I have a debian with aptitude and it suggests libbluetooth-dev ... |
| 10:02.46 | bulkorok | but in fact... get the bluetooth-devs and ./configure and make menuselect the asterisk-source... |
| 10:04.44 | nisam | k..i will try now |
| 10:04.49 | *** join/#asterisk wonderworld (~ww@dsdf-4db5eaf7.pool.mediaWays.net) |
| 10:04.57 | nisam | thanks for suggession |
| 10:05.07 | nisam | will come back to you |
| 10:05.20 | bulkorok | if I can help :-) |
| 10:16.06 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 10:18.59 | *** join/#asterisk nisam (3b5ab491@gateway/web/freenode/ip.59.90.180.145) |
| 10:21.33 | *** join/#asterisk disposable (disposable@shell.websupport.sk) |
| 10:24.04 | salz212 | SIP RFC question: what is the meaning of "H14.43" I could not find this clause any where .. context: "20.41 User-Agent: The semantics of this header field are defined in [H14.43]." |
| 10:29.30 | wdoekes | salz212: http rfc ( http://www.ietf.org/rfc/rfc2616.txt ) |
| 10:29.41 | wdoekes | (and no, that wasn't immediately clear to me neither) |
| 10:30.13 | wdoekes | Rather than repeating the syntax and semantics here, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [8]) |
| 10:33.21 | salz212 | thanks a lot. |
| 10:35.35 | salz212 | Actually I am stuck in User-Agent grammar thing... I want to customize the User-Agent from my UAClient like an Andriode phone.. I am setting a delimiter separated 8-10 fields. But I am not sure if this packet pass through other proxies and will it remain the same or chopped off.. also what is the max length of a User-Agent header..? |
| 10:42.31 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 10:44.10 | salz212 | what is the max length of a User-Agent allowed by asterisk..? and SIP.. I have gone through RFCs but could not find the exact value, even in chan_sip has no length specified. |
| 10:45.01 | *** join/#asterisk WebSprocket (~WebSprock@static-159-253-76-34.karoo.kcom.com) |
| 10:45.37 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 10:46.05 | Greenlight | Howdy folks. Does Asterisk handle SRV DNS records correctly to allow failover? |
| 10:47.27 | WebSprocket | Hey Guys, I have an issue and wondering if anyone can shine a light on it. We have an E1 (ISDN) with different DDI for our clients to dial out with) however it doesn't include the area code on the number, I want to setup a trunk with 2 voip lines but when i send the call to them trunks i need the prefix adding onto the callerid number is there anyway to do this? |
| 10:53.59 | Greenlight | According to voipinfo.org asterisk only reads the first SRV record, and ignores the rest - is this still the case? |
| 11:00.00 | leifmadsen | Greenlight: yes |
| 11:00.43 | leifmadsen | WebSprocket: just: Set(CALLERID(num)=123${CALLERID(num)}) |
| 11:00.46 | wdoekes | salz212: normally you'll want to keep the entire sip packet within 1300 bytes (including the sdp).. so that'll be a limiting factor. sure you're allowed to go bigger, but then you "should" move to tcp or suffer packet fragmentation issues |
| 11:10.35 | salz212 | My User-Agent length won't be more than 80-100 chars |
| 11:16.48 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 11:29.59 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 11:31.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:43.59 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
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| 11:54.54 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 11:58.53 | Greenlight | How can I view the audio/RTP IP addreses in use for a SIP call? |
| 11:59.24 | Greenlight | I do "sip show channel <ChanelID>" but it only shows it's own IP as the Audio IP, I want to know the remote side |
| 12:03.12 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 12:04.38 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:10.01 | leifmadsen | Greenlight: might have to look at core show channels |
| 12:10.08 | leifmadsen | (try verbose on the end too) |
| 12:10.28 | leifmadsen | when looking at the sip channels, they are probably independent and don't show bridging information |
| 12:11.32 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:13.12 | bulkorok | Greenlight: do you need the info in realtime or for later use like billing? |
| 12:21.00 | *** join/#asterisk nisam (3b5ab491@gateway/web/freenode/ip.59.90.180.145) |
| 12:21.16 | nisam | hi |
| 12:21.56 | nisam | hi friends |
| 12:23.38 | bulkorok | nisam... |
| 12:23.58 | bulkorok | what is chan_mobile doing!? |
| 12:26.42 | nisam | i really want to connect my mobile to asterisk using bluetooth..can you help me |
| 12:27.23 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 12:27.56 | leifmadsen | note that chan_mobile is not well maintained, and is possible it won't even work well in recent versions of asterisk |
| 12:28.16 | leifmadsen | but what you're looking for is chan_mobile for bluetooth interaction in asterisk |
| 12:34.16 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 12:36.49 | *** join/#asterisk nisam (3b5ab491@gateway/web/freenode/ip.59.90.180.145) |
| 12:37.54 | nisam | hi |
| 12:39.36 | bulkorok | as mentioned earlier,you need the header-files for bluetooth... |
| 12:41.25 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-pdesmkkarktdwpwy) |
| 12:43.41 | *** join/#asterisk nisam_ (3b5ab491@gateway/web/freenode/ip.59.90.180.145) |
| 12:44.15 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 12:44.38 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:44.51 | nisam_ | bulkrok: i have these much package. gnome-bluetooth-0.7.0-10.2.el5 |
| 12:45.01 | nisam_ | bluez-utils-3.7-2.2.el5.centos |
| 12:45.12 | nisam_ | nautilus-sendto-bluetooth-1.0.1-6.el5.centos |
| 12:45.25 | nisam_ | bluez-hcidump-1.32-1 |
| 12:45.33 | nisam_ | bluez-gnome-0.5-5.fc6 |
| 12:45.43 | nisam_ | gnome-bluetooth-libs-0.7.0-10.2.el5 |
| 12:45.52 | nisam_ | bluez-libs-3.7-1.1 |
| 12:45.59 | nisam_ | gnome-bluetooth-devel-0.7.0-10.2.el5 |
| 12:46.15 | nisam_ | anything more required..? |
| 12:46.15 | Greenlight | bulkorok: Need it in realtime - I'm trying to check that the RTP stream is being correctly remotely bridged |
| 12:47.47 | Greenlight | Hmm, does directmedia=yes overrule directrtpsetup=yes? |
| 12:47.51 | *** join/#asterisk jetlag (~jetlag@pool-71-168-253-208.cmdnnj.east.verizon.net) |
| 12:49.16 | Greenlight | Seeing 20mbs+ traffic on a server that should not be in the RTP path and trying to diagnose why |
| 12:49.46 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-ayglakqohkqxqbih) |
| 12:50.30 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 12:50.31 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 12:51.19 | bulkorok | Greenlight: maybe just tcpdump to file and check with wireshark?! |
| 12:51.52 | bulkorok | the Telephony menu is very useful... |
| 12:51.59 | Greenlight | Yea got a capture running now |
| 12:52.07 | Greenlight | Will soon see :) |
| 12:52.14 | bulkorok | :) |
| 12:52.16 | Greenlight | It may well just be calls "ringing" |
| 12:52.19 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 12:52.31 | Greenlight | As iirc the remote bridge happens upon connection, but i'm not 100% sure |
| 12:52.36 | bulkorok | kind of early media or sth!? |
| 12:52.49 | Greenlight | early media yea |
| 12:52.54 | *** join/#asterisk serafie (~erin@nat/digium/x-xhhcixdrvxzsgcar) |
| 12:53.07 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:53.17 | Greenlight | But it's making my ASA get twitchy with the number of packets and load |
| 12:54.02 | *** join/#asterisk Sprocks (~Sprocks@bas1-brampton37-1176135265.dsl.bell.ca) |
| 12:56.07 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 12:57.33 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 12:58.02 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:4:0:414:11) |
| 12:59.05 | jeffspeff | Greenlight, how many users do you have? |
| 12:59.14 | jeffspeff | also call volume? |
| 12:59.41 | Greenlight | 500+. 1mil/day |
| 13:00.04 | jeffspeff | what asa do you use? 5510? |
| 13:00.27 | Greenlight | 5505 iirc |
| 13:00.41 | Greenlight | It's running a little hot tho |
| 13:01.25 | jeffspeff | I didn't think a 5505 could handle that kind of volume |
| 13:01.38 | Greenlight | It's just the SIP, not RTP |
| 13:01.42 | *** join/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 13:01.49 | Greenlight | RTP gets handed off |
| 13:02.25 | Greenlight | Well, it's supposed to... |
| 13:03.07 | Greenlight | Mind we're planning on putting another box behind it that will be handling RTP streams, for approx 250 channels |
| 13:03.12 | Greenlight | So think we'll have to upgrade |
| 13:04.11 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gokjueebuvuczjsl) |
| 13:04.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:05.17 | *** join/#asterisk deo__ (~dnepangue@58.71.19.178) |
| 13:08.09 | Greenlight | Ah ha - had a server set to use ulaw, so that's why remote bridging wasnt happening |
| 13:08.23 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 13:10.37 | *** part/#asterisk deo (~dnepangue@58.71.19.178) |
| 13:10.40 | *** join/#asterisk aboven (~aboven@94.139.41.14) |
| 13:12.23 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 13:12.24 | darkdrgn2k | hi |
| 13:12.56 | darkdrgn2k | wheni call is transfered to an outside number over sip, does asterisk hold on to the call or is there a way to have it released from the sip trunk but still transfered? |
| 13:13.03 | *** part/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 13:23.38 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust1000.sgyl.cable.virginmedia.com) |
| 13:28.53 | *** join/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 13:38.44 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 13:41.10 | [TK]D-Fender | darkdrgn2k, You are unclear on precisely what is on the other end |
| 13:49.21 | darkdrgn2k | i have a sip trunk comming in |
| 13:49.49 | darkdrgn2k | when a call comes in on that trunk. i want to forward it (from the phone) to another pstn number, the transfer will be outbound from the same trunk |
| 13:50.08 | darkdrgn2k | will asterisk keep both trunk channels open durring the length of the transfered call? (IN and OUT) |
| 13:50.34 | bulkorok | depends on directmedia option I would say... |
| 13:50.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:50.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:51.08 | darkdrgn2k | directmedia? |
| 13:51.15 | darkdrgn2k | canreinvite? |
| 13:52.54 | bulkorok | the same in different versions |
| 13:53.09 | darkdrgn2k | so if the trunk provider allows for canreinvite im good? |
| 13:53.25 | bulkorok | you must check on your asterisk |
| 13:53.44 | darkdrgn2k | check for what? canreinvite? |
| 13:54.07 | bulkorok | i would say so... sip.conf |
| 13:54.20 | darkdrgn2k | so if canreinvite=yes it wont hold the call other wise it will? |
| 13:56.31 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 13:58.42 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 13:58.54 | *** join/#asterisk youjelly (~bwahahaha@39.47.245.77) |
| 14:00.40 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-81-77-81.hsv.bellsouth.net) |
| 14:03.38 | *** join/#asterisk aross42 (~aross@216.113.92.2) |
| 14:04.20 | *** join/#asterisk bchia (~Adium@nat/digium/x-pkafygljiuxxtgfm) |
| 14:07.00 | [TK]D-Fender | darkdrgn2k, tons of things can force * to stay in the path : transcoding, recording, forced audio ringing, DTMF feature codes, etc. If not, and re-invites are allowed and the provider SUPPORTS it then * would hand off the RTP |
| 14:07.40 | Greenlight | What's forced audio rining? |
| 14:07.43 | Greenlight | *ringing |
| 14:08.16 | darkdrgn2k | fake ring tone on the lien |
| 14:08.19 | darkdrgn2k | : |
| 14:08.22 | darkdrgn2k | ? |
| 14:08.39 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:08.47 | [TK]D-Fender | "r" |
| 14:09.02 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:09.19 | Greenlight | Ahh |
| 14:09.49 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 14:11.34 | Greenlight | I think the issue i'm getting is that event with directrtpsetup=yes and directmedia=yes, asterisk is remaining in RTP path until call connects, so it's handling RTP for early media |
| 14:11.42 | Greenlight | Which is a lot of traffic |
| 14:12.29 | Greenlight | IS this by design? Should this be the case? |
| 14:13.10 | *** join/#asterisk sekil (~sekil@217.52.33.166) |
| 14:14.23 | [TK]D-Fender | thre is nothing to bridge until the other side answers |
| 14:14.33 | [TK]D-Fender | * is not a proxy |
| 14:14.46 | [TK]D-Fender | or a SIP router |
| 14:14.50 | Greenlight | Hmm gotcha |
| 14:15.08 | Greenlight | That's a pain |
| 14:15.28 | Greenlight | To the tune of 20mbs load ;/ |
| 14:15.59 | bulkorok | Greenlight: with that "much" calls I would suggest some SIPproxy or loadbalancing... |
| 14:16.36 | Greenlight | I'd looked at OpenSIPs, but went with Asterisk because I knew how to get it working how I wanted |
| 14:16.51 | Greenlight | Was happy setting it up and getting it working |
| 14:17.28 | *** join/#asterisk wonderworld (~ww@dsdf-4db5eaf7.pool.mediaWays.net) |
| 14:18.12 | Greenlight | This system autherises each call against a MSSQL database and also pumps CDR's back for billing purposes. What would be an alternative, opensips? |
| 14:18.57 | bulkorok | opensips/kamailio.... freeswitch is a good one for big installations too |
| 14:19.29 | bulkorok | freeswitch has failover capability without dropping calls... |
| 14:19.37 | Greenlight | And they can be set to call database procs to autherise calls and the like? |
| 14:19.40 | *** join/#asterisk cbdev (~cbdev@gateway/web/irccloud.com/x-xgsceohejtvjtjkq) |
| 14:19.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:20.07 | bulkorok | dont know exactly but I'm pretty sure... |
| 14:20.16 | Greenlight | Ok cool, thanks, it's much appriciated |
| 14:21.28 | bulkorok | kamailio has some AUTH and DB modules... |
| 14:22.23 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ohxnwhtgthjideoo) |
| 14:25.21 | *** join/#asterisk synju (~cocopops@196-215-75-30.dynamic.isadsl.co.za) |
| 14:29.32 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:29.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:29.40 | *** join/#asterisk sekil (~sekil@217.52.33.166) |
| 14:29.55 | synju | I'm hoping someone here can guide me a little bit, I am looking at using asterisk to allow clients to go through to 1 of 2 departments.. sales & support.... Now my confusion comes in where the client initially calls 1 number... my problem is now that the line becomes 'engaged' .. Do I need to contact my telecom SP and ask them for additional line usage or something? eg more than 1 call on the same line at a time.... |
| 14:30.45 | pabelanger | synju: depends |
| 14:30.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:30.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:30.55 | pabelanger | how many channels do you have from your telco?? |
| 14:31.02 | pabelanger | if only 1, then you can only do 1 call |
| 14:31.14 | synju | so its called 'channels' ? |
| 14:31.23 | pabelanger | yes |
| 14:31.26 | synju | okay |
| 14:31.33 | synju | I'm in South Africa |
| 14:31.40 | synju | my telco is Telkom... |
| 14:31.41 | synju | .. |
| 14:31.48 | pabelanger | so, how many calls do you want to accept at 1 time? |
| 14:32.01 | pabelanger | that is how many channels you will need |
| 14:32.41 | synju | Well I'm looking at offering PBX System Setups and Maintenance as a Service and Product... meaning We will set it up according to what the company (any in particular) needs. |
| 14:33.30 | synju | I'm not too knowledgable about it yet but I'm investigating it and could possible hire people who are knowledgable enough to handle it. |
| 14:33.50 | synju | So I need to see how it would 'basicly' work. |
| 14:34.31 | synju | And what I imagine is 1 or more clients would call a 'number' ... they would select an option (pre-scripted) and then go through to the desired location... |
| 14:34.50 | synju | thanks for answering my question about the 'Channels' part |
| 14:40.53 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:41.11 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.166) |
| 14:41.17 | StaRetji | Howdy folks |
| 14:41.27 | Greenlight | hiya |
| 14:41.37 | StaRetji | can someone help me out, how to blacklist clid and dnid? |
| 14:42.03 | StaRetji | I have list of clid and list of dnid which I would like to block, but don't know from where to start |
| 14:42.24 | Greenlight | exten => _CLI/.,1,Hangup() |
| 14:42.51 | StaRetji | can it read from some .txt file? |
| 14:42.56 | StaRetji | I have tons :/ |
| 14:43.08 | Greenlight | YOu could write ascript to create the dialplan |
| 14:43.40 | bulkorok | putting the numbers in a databse and use func_odbc |
| 14:43.48 | Greenlight | Or that too |
| 14:44.01 | bulkorok | I really like func_odbc ;-) |
| 14:44.07 | StaRetji | thx both, bulkorok idea is out of my knowledge |
| 14:44.08 | StaRetji | lol |
| 14:44.22 | bulkorok | learning! |
| 14:44.25 | StaRetji | and writing scripts can maybe be done |
| 14:44.32 | StaRetji | yeah, hehe |
| 14:44.38 | bulkorok | :) |
| 14:44.45 | StaRetji | if I can learn in 30 minutes, that would be cool |
| 14:44.46 | StaRetji | hahahha |
| 14:44.51 | Greenlight | You could even use excel to write the script, if push came to shove and that's not your forté |
| 14:45.07 | StaRetji | I have client and his client is killing us :/ |
| 14:45.17 | Greenlight | Ahh |
| 14:45.20 | Greenlight | In that case |
| 14:45.35 | Greenlight | exten => _CLI/.,1,Playback(tt-monkeys) |
| 14:45.40 | Greenlight | exten => _CLI/.,n,Hangup() |
| 14:45.47 | Greenlight | ;) |
| 14:46.55 | StaRetji | lol |
| 14:46.58 | StaRetji | nice one ;) |
| 14:48.29 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 14:49.23 | StaRetji | Greenlight: can you give me example for DNID also (destination number) |
| 14:49.24 | StaRetji | ? |
| 14:49.26 | StaRetji | thx mate |
| 14:49.45 | StaRetji | hm |
| 14:49.58 | StaRetji | I did that, he still makes calls |
| 14:51.04 | StaRetji | exten => _0000/.,1,Hangup() |
| 14:51.05 | StaRetji | exten => _X.,n,NoOp(A2Billing Start) |
| 14:51.05 | StaRetji | exten => _X.,n,AGI(a2billing.php,1) |
| 14:51.05 | StaRetji | exten => _X.,n,Hangup |
| 14:51.18 | StaRetji | he is still able to make calls |
| 14:51.25 | StaRetji | oh, wait, dialplan reload? |
| 14:51.26 | StaRetji | lol |
| 14:51.56 | [TK]D-Fender | That is BROKEN |
| 14:51.59 | Greenlight | Was just about to ask that |
| 14:52.05 | [TK]D-Fender | _X has no FIRST priority |
| 14:52.10 | Greenlight | Indeed |
| 14:52.19 | [TK]D-Fender | and the concept of where the CID match goes is backwards |
| 14:52.40 | StaRetji | ehm |
| 14:52.43 | Greenlight | Ahh yea you're right - always get them mixed up |
| 14:52.46 | StaRetji | how to block clid 0000 |
| 14:52.56 | Greenlight | exten => ./_CLI,1,Hangup() |
| 14:52.56 | StaRetji | now you lost me :) |
| 14:53.26 | [TK]D-Fender | ..... |
| 14:53.44 | Greenlight | exten => ./0000,1,Hangup() |
| 14:54.02 | [TK]D-Fender | Greenlight, So far none of those is functional..... |
| 14:54.11 | StaRetji | lol |
| 14:54.23 | StaRetji | and I applied 1 by 1 on my live server |
| 14:54.25 | [TK]D-Fender | Greenlight, You may want to start reading THE BOOK again.... |
| 14:54.28 | StaRetji | but no problem |
| 14:54.30 | [TK]D-Fender | ~book |
| 14:54.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:54.34 | StaRetji | I appreciate you are trying to help |
| 14:54.54 | Greenlight | exten => _./0000,1,Hanguo() |
| 14:54.59 | Greenlight | exten => _./0000,1,Hangup() |
| 14:55.00 | StaRetji | please folks, help me :) |
| 14:55.02 | Greenlight | Seesh |
| 14:55.34 | Greenlight | I just got the CLI and EXTEN wrong way around and missed an _ |
| 14:55.42 | StaRetji | which is now, hehe |
| 14:55.44 | StaRetji | last one? |
| 14:56.38 | Greenlight | Should be, but I'll let [TK]D-Fender judge incase he's cruel again |
| 14:57.22 | StaRetji | [TK]D-Fender: what say you? :) |
| 14:58.33 | [TK]D-Fender | Greenlight, Cruel? Far from. Putting out "fair warning" when I see 4 distinct mistakes in the very first field in an Exten line ... that's a public service. |
| 14:58.56 | [TK]D-Fender | starWhere you put things also matters and you your mini-flood in itself isn't all of it.... |
| 14:59.04 | StaRetji | and thx for that, appreciated |
| 14:59.21 | Greenlight | What were the 4 mistakes? |
| 14:59.33 | StaRetji | it seems I blocked all my traffic |
| 14:59.57 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 15:00.13 | Greenlight | Remove the line we added, and do a reload, that'll get your traffic back |
| 15:00.20 | StaRetji | with this one exten => _./0000,1,Hangup() |
| 15:00.27 | Greenlight | Yea |
| 15:00.29 | StaRetji | yes, but in both cases I'm doomed |
| 15:00.33 | StaRetji | lol |
| 15:00.41 | StaRetji | need to make it work, or I'm done |
| 15:01.00 | Greenlight | ANd apologies for the incorrect syntax, was trying to help, but it's been a few busy months since I last did any dialplan editing and I appear to be rusty - sorry! |
| 15:01.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:01.22 | StaRetji | no no, you very nice, thx for trying to help out this noob :/ |
| 15:01.29 | StaRetji | you're |
| 15:01.34 | Greenlight | To block calls from CLI 0000 (is that correct CLI?) then use the last line I posted, but as [TK]D-Fender pointed out it needs to be in the correct place |
| 15:01.45 | zamba | i need a pattern matching where i want to replace two or more 0 at the beginning of the number with '+'.. how? |
| 15:02.14 | zamba | of the number dialed, that is |
| 15:02.50 | StaRetji | you mean, it should work, but not in front of a2billing stuff? |
| 15:03.41 | Greenlight | Well you'll need to put it in the correct place, without seeing your dialplan I can't say where that is. |
| 15:04.22 | [TK]D-Fender | zamba, ${somevar:2} = that var with 2 digits chopped off |
| 15:04.59 | zamba | [TK]D-Fender: but i don't know how many 0s there is.. it could be 3 |
| 15:05.09 | zamba | or even 4 |
| 15:05.28 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:05.30 | zamba | the theory is that more than one 0 at the beginning of a number should be replaced with + |
| 15:05.52 | zamba | or maybe not even replaced, but just stripped off.. because this will indicate an attempt of dialing international |
| 15:07.01 | Greenlight | Why not use a loop, and keep looping if the left most character is 0, and each iteration chop a character off |
| 15:07.17 | zamba | Greenlight: sure.. but i'm not that fluent with asterisk dialplan ;) |
| 15:07.24 | Greenlight | Me neither, apparently ;)( |
| 15:07.28 | zamba | Greenlight: hehe |
| 15:07.30 | StaRetji | Greenlight: it seems it is working mate :) hehehe, can you give me example for DNID and will test it ;) THX! |
| 15:07.52 | zamba | the pattern matching looks to be '_00.X.', at least |
| 15:07.56 | Greenlight | StaRetji: As in, when your user dials a specific number? |
| 15:07.58 | zamba | if i've understood this correctly |
| 15:08.34 | Greenlight | _00.X. will match anything starting with 00 then anything then a number then anything again |
| 15:08.40 | Greenlight | So, basically anything starting 00 |
| 15:08.52 | zamba | which is fine and what i want |
| 15:09.05 | zamba | eh.. maybe not 00 then anything |
| 15:09.13 | StaRetji | Greenlight: exten => _./0000,1,Hangup() in front of a2billing |
| 15:09.20 | zamba | it should be 00<something> or 000000<something> |
| 15:09.41 | Greenlight | Sorry - I gotta leave the office now, catch yas later |
| 15:09.42 | StaRetji | Greenlight: yes, when use dial specific number |
| 15:09.53 | StaRetji | okay, cheers mate |
| 15:10.08 | [TK]D-Fender | zamba, tehn you'll have to run a loop to count how many are at the front. THEn replace them. |
| 15:10.36 | zamba | '_00.X.' seems to match what i want, at least |
| 15:11.48 | zamba | [TK]D-Fender: got an example of that? |
| 15:11.58 | zamba | gotoif? |
| 15:12.14 | [TK]D-Fender | zamba, You can't use "." in the middle of a pattern. |
| 15:12.27 | zamba | [TK]D-Fender: seems to be working here.. |
| 15:12.37 | zamba | exten => _00.X.,1,Noop(yo) |
| 15:12.47 | zamba | Executing [004741921679@standard_sip:1] NoOp("SIP/marius-00000041", "yo") in new stack |
| 15:12.52 | [TK]D-Fender | zamba, .... because EVERYTHING after the "." is ignored. |
| 15:12.54 | zamba | Executing [0000004741921679@standard_sip:1] NoOp("SIP/marius-00000042", "yo") in new stack |
| 15:13.00 | zamba | oh |
| 15:13.00 | [TK]D-Fender | zamba, X is meaningless there |
| 15:13.04 | zamba | ok |
| 15:13.09 | zamba | so how should i write the pattern instead? |
| 15:13.23 | zamba | '_00.'? |
| 15:13.57 | [TK]D-Fender | zamba, that would match 00 + 1 or more chars |
| 15:14.31 | zamba | can you please tell me what i should use instead? |
| 15:14.55 | zamba | i need to match 2 or more 0s |
| 15:20.35 | *** join/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 15:20.35 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:23.26 | [TK]D-Fender | zamba, there is no pattern for variable number of specific digits. You need 2 or MORE. that will match it. Getting the exact count is something you'll do IN that basic pattern |
| 15:26.52 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:34.05 | zamba | ok, so then '_00.' is a good pattern |
| 15:35.40 | *** join/#asterisk Tagor (~Tagor@s55978a13.adsl.wanadoo.nl) |
| 15:36.50 | Tagor | I upgraded Asterisk to stable on my debian pc, but now 'sip show registry' doesn't work anymore and Asterisk does not register my phonelines. Anyone know what happened? |
| 15:38.16 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:39.53 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 15:40.08 | Tagor | I get this: No such command 'sip show registry' |
| 15:40.25 | Tagor | I'm on 1.6.2.9-2 |
| 15:44.16 | Tagor | Anybody? Looks like all sip commands are gone. core show help doesn't show any sip command |
| 15:47.46 | [TK]D-Fender | Tagor, if it doesn't work then chan_sip didn't even load |
| 15:48.04 | [TK]D-Fender | Tagor, You've either broken your configs, or your overall * install |
| 15:48.24 | [TK]D-Fender | Tagor, "module load chan_sip.so" |
| 15:48.41 | [TK]D-Fender | Tagor, see what happens when you try to load it manually |
| 15:49.03 | zamba | [TK]D-Fender: should i use while? |
| 15:51.04 | *** join/#asterisk navaismo (~navaismo@189.144.58.34) |
| 15:52.02 | mcolombo | Hi a |
| 15:52.04 | mcolombo | hi all |
| 15:53.04 | mcolombo | anybody, say how to enable the line "Reason: SIP " in SIP Cancel messages? |
| 15:53.32 | mcolombo | example : Reason: SIP;cause=16;text="Normal Call Clearing" |
| 15:53.39 | mcolombo | thanks |
| 15:59.34 | Tagor | [TK]D-Fender: thanks! i found out there's a problem with the sip module in 1.6.2.9-2+squeeze7, rolled back to 1.6.2.9-2+squeeze5 (http://comments.gmane.org/gmane.linux.debian.packages.voip.devel/20193) |
| 16:02.41 | *** join/#asterisk Guest39614 (yang@freenode/sponsor/cacert.assurer.yang) |
| 16:03.39 | Guest39614 | Is it possible to have an encrypted VoIP connection to some voip uplink provider ? |
| 16:03.54 | Guest39614 | which one sells these services ? |
| 16:05.46 | pabelanger | yes |
| 16:05.53 | pabelanger | however, you need to share keys |
| 16:07.52 | zamba | [TK]D-Fender: got it working! :) |
| 16:08.28 | zamba | http://pastie.org/4754418 |
| 16:08.31 | zamba | can you approve that? |
| 16:10.29 | Guest39614 | pabelanger: do you know such provider ? |
| 16:10.46 | pabelanger | no |
| 16:10.55 | pabelanger | but you _really_ need to trust them to do it |
| 16:11.04 | pabelanger | since they will have the keys and can monitor the traffic |
| 16:11.18 | pabelanger | if you want it secure, you need to build it yourself |
| 16:14.27 | lorsungcu | wat do |
| 16:16.48 | *** join/#asterisk DropsOfSerenity (~justin@cpe-76-169-90-202.socal.res.rr.com) |
| 16:19.53 | chris_n | is there any reason not to use rsync to keep a warm standby * server config up-to-date? |
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| 16:28.44 | *** part/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 16:29.35 | *** join/#asterisk danfromuk (~IceChat77@2.27.28.56) |
| 16:30.13 | danfromuk | Is anyone having problem with Voxbone today? |
| 16:40.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 16:41.40 | [TK]D-Fender | zamba, Link won't open |
| 16:46.14 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 16:54.32 | zamba | oh.. what tool do you generally use for pasting? |
| 16:55.14 | unicron | anybody use voip.ms? |
| 16:56.25 | [TK]D-Fender | unicron, Lots of people |
| 17:03.36 | unicron | i'm trying to use my asterisk server to forward calls to a toll free UK DID and i'm not sure why it's not working |
| 17:03.53 | unicron | err, to forward inbound calls from the DID to a US cell phone |
| 17:06.09 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
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| 17:17.07 | unicron | i figured it out... apparently i don't want to be sending 011 |
| 17:21.50 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 17:25.30 | zamba | [TK]D-Fender: but i have a strange problem.. i added a pattern match for numbers starting with '+', but this doesn't work with polycom phones |
| 17:26.24 | zamba | [TK]D-Fender: in the missed calls section of the polycom the numbers are listed with +47<number>.. and when i then try to call these numbers back, i get the error that the extension isn't found in the context |
| 17:26.43 | zamba | [TK]D-Fender: whereas, if i dial +47<number> from my softphone, it works perfect with the pattern matching i've done |
| 17:26.46 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 17:27.11 | [TK]D-Fender | zamba, I should be seeing a PASTEBIN for these calls.... |
| 17:28.14 | zamba | [TK]D-Fender: i'll provide that tomorrow.. i'm off work now :) |
| 17:28.32 | zamba | http://pastebin.com/GubtWt6Y |
| 17:28.40 | zamba | here's that paste from earlier, if you're able to open pastebin.com links |
| 17:29.38 | WIMPy | How many 0s do you want to be able to dial without effect? |
| 17:30.00 | zamba | WIMPy: what do you mean? |
| 17:30.16 | zamba | WIMPy: 2 or more 0 in the beginning of the number should be replaced with '+' |
| 17:30.19 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 17:30.29 | WIMPy | You have a loof there to remove any number of leading 0s. |
| 17:30.43 | zamba | WIMPy: or, rather, i add that '+' when i dial, but that loop should remove all 0s if there's 2 or more, yeah |
| 17:30.44 | WIMPy | "or more"? |
| 17:31.01 | zamba | yeah? |
| 17:31.03 | zamba | or 3, or 4 |
| 17:31.07 | WIMPy | Intergalactic calls? |
| 17:31.14 | zamba | well.. just to be sure :) |
| 17:32.00 | zamba | because 00 should be international calls, and since we earlier had to prefix -all- external calls with 0 anyway, and some users are used to dialing 000, i want to include that as well |
| 17:32.21 | *** join/#asterisk cmendes0101 (~cmendes01@216.104.164.5) |
| 17:32.23 | zamba | so it's basically just two cases, 00 and 000, but might as well make it as general as possible :) |
| 17:32.45 | zamba | (we're switching pbx at our offices) |
| 17:33.17 | WIMPy | And how do you know if someone dialling 00... wants a nationla or an international call then? |
| 17:33.54 | zamba | because 00 is the international prefix |
| 17:34.12 | zamba | 2 or more 0s means international |
| 17:35.14 | zamba | of course, we have some other odd cases, which i guess is what you're getting at.. because we have some national numbers like for instance 05000.. and if i apply this pattern here and we have a user dialing 005000 like he used to, we have a problem |
| 17:35.31 | zamba | so i guess i need to add a new pattern for that |
| 17:35.40 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 17:38.44 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:39.13 | bmoraca_work | Hey, does Asterisk still have issues with multiple NICs? i recall in version 1.4, it didn't work all that well. |
| 17:40.04 | zamba | bmoraca_work: it works for me |
| 17:40.17 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:40.22 | bmoraca_work | zamba: do you have multiple routes out each interface? |
| 17:40.39 | zamba | nope |
| 17:40.54 | bmoraca_work | i.e., i'll have a default route out one interface and a few more specific network routes out the other interface |
| 17:41.01 | zamba | ah, that should work |
| 17:41.57 | bmoraca_work | i'd hope so |
| 17:42.15 | bmoraca_work | i just remember a few years ago that sometimes Asterisk would send responses back out the wrong interface |
| 17:42.26 | zamba | strange |
| 17:42.37 | zamba | one would think it'd send it to an ip and then just let the kernel handle the rest |
| 17:43.09 | bmoraca_work | something about putting the correct source IP in the SIP messages or something |
| 17:43.55 | bmoraca_work | i.e. one interface will be public network, the other would be private network. the private network will not have any routing out to the public internet at all, so no way to get to the public address...aside from the other negatives of the asymmetric routing that would ensue |
| 17:44.03 | bmoraca_work | guess i'll just have to try it and see how it goes |
| 17:44.37 | zamba | right now i have two nics in my asterisk box.. one has the default route out to the world, whereas the other nic only communicates with a sip trunk |
| 17:44.46 | zamba | and that works |
| 17:44.55 | bmoraca_work | well, that's a good sign |
| 17:46.12 | bmoraca_work | i'm going to be moving some of my hosted pbx customers into tunneled private networks to remove NAT...so we'll see what happens |
| 17:49.31 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 17:49.37 | *** join/#asterisk sustav (~vpp@nat/digium/x-krrrlcdgoahgbanq) |
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| 18:19.37 | *** join/#asterisk silvano_ (d8054fce@gateway/web/freenode/ip.216.5.79.206) |
| 18:20.06 | silvano_ | hi. does anyone know an open source PBX project with voicemail blasting features? |
| 18:20.08 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
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| 18:25.26 | ChannelZ | Yes. turn the volume on your phone up as loud as it can possibly go |
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| 18:38.02 | volga629 | process_sdp: ignoring 'video' media offer because port number is zero Is this error can cause by codec ? |
| 18:39.25 | [TK]D-Fender | silvano_, Yes there is this one called "Asterisk". You should take a look at it... |
| 18:40.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 18:44.38 | silvano_ | [TK]D-Fender: lol, I meant something with a grafical UI, i.e. FreePBX |
| 18:44.44 | silvano_ | graphical* |
| 18:46.49 | specing | silvano_: I'd recommend Asterisk too |
| 18:49.32 | wdoekes | if you mean freepbx, you should probably look in #~ |
| 18:50.37 | silvano_ | I am trying to that wdoekes... I've been waiting for one hour already :) |
| 18:50.42 | cresl1n | or in ## |
| 18:51.19 | silvano_ | so I guess nobody here uses web interfaces for their asterisk installations. That's fine then. |
| 18:56.42 | specing | silvano_: nope, we are all hardcore here |
| 18:57.00 | WIMPy | And S&M. |
| 18:57.07 | specing | exactly :D |
| 18:58.40 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:00.48 | silvano_ | good for you :) I'd rather give people easy tools they can use to make changes themselves so I don't have to waste my time writing configurations by hand all the time. |
| 19:01.00 | *** join/#asterisk sustav_ (~vpp@nat/digium/x-nixfrkiiibszxcrg) |
| 19:01.23 | specing | "give"? |
| 19:01.31 | specing | Hmmmm... |
| 19:03.26 | silvano_ | give, offer, hand them a solution, allow access to... I think you can get the point. |
| 19:03.55 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 19:34.09 | danfromuk | Is anyone having problem with Voxbone today? |
| 19:35.40 | silvano_ | working fine here danfromuk |
| 19:36.29 | *** join/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:36.44 | oquidave | hello , why am i getting 0 modules loaded when i type module show yet when asterisk starts, it shows NOTICE[8477]: loader.c:1133 load_modules: 147 modules will be loaded. |
| 19:36.47 | danfromuk | silvano_: do you have any UK did's with them? We got a few that haven't been working for 48 hours now. They keep saying they've got infrastructure problems |
| 19:37.12 | silvano_ | yes, let me check those |
| 19:38.45 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:39.54 | silvano_ | I see calls coming in... |
| 19:40.19 | danfromuk | Strange. |
| 19:40.23 | danfromuk | Thanks for checking |
| 19:40.53 | silvano_ | np |
| 19:41.56 | oquidave | here's my modules.conf file http://pastebin.com/HAQXbkVt |
| 19:42.17 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:44.53 | oquidave | when i also check /usr/lib/asterisk/modules , there're over 70 modules there...why then is my asterisk saying 0 modules loaded? |
| 19:50.43 | danfromuk | oquidave: this is my modules.conf and it loads fine |
| 19:50.45 | danfromuk | http://pastebin.com/k0ZY4CAn |
| 19:52.37 | oquidave | danfromuk okay, you dont have alot of noload lines |
| 19:52.49 | oquidave | so perhaps i need to remove them too |
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| 19:56.58 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:56.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:59.23 | oquidave | am trying to restart asterisk and it says No such command 'restart now' (type 'core show help restart' for other possible commands) !!! |
| 19:59.43 | jpsharp | "core restart now" |
| 20:00.25 | oquidave | that too doesn't work...No such command 'core restart now' (type 'core show help core' for other possible commands) |
| 20:00.51 | leifmadsen | did you type the command? |
| 20:01.03 | leifmadsen | or did you change the cli_permissions.conf file? |
| 20:01.08 | leifmadsen | what version? |
| 20:01.22 | oquidave | Asterisk 1.8.16. |
| 20:02.48 | leifmadsen | core restart now -- is definitely a command :) |
| 20:02.58 | leifmadsen | only thing I can think of is that cli_permissions.conf is disallow you to use it |
| 20:03.21 | jpsharp | Is it in a module he didn't load? |
| 20:03.24 | oquidave | leifmadsen where's that file and what can i do with it |
| 20:03.31 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:03.41 | leifmadsen | oquidave: all conf files are in your /etc/asterisk/ dir |
| 20:03.50 | leifmadsen | jpsharp: core commands are part of the core system |
| 20:03.54 | leifmadsen | no modules are required |
| 20:04.08 | jpsharp | Built into the /usr/sbin/asterisk binary? |
| 20:04.41 | leifmadsen | into the core system of Asterisk |
| 20:04.53 | WIMPy | jpsharp: That's why those commands have been moved to 'core ...'. |
| 20:05.11 | jpsharp | I've seen weirder in systems. :) |
| 20:05.38 | oquidave | meanwhile my asterisk has 0 modules loaded...perhaps it's the cause of the problem. am trying to restart the server |
| 20:07.09 | jpsharp | might just have to nuke it from orbit with kill. |
| 20:07.16 | *** join/#asterisk af_ (~getsmart@78-134-106-25.v4.ngi.it) |
| 20:07.55 | oquidave | but it says load_modules: 182 modules will be loaded. when starting!!! |
| 20:08.16 | oquidave | let me post my module.conf file |
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| 20:08.19 | leifmadsen | sounds like problems with your conf files |
| 20:08.31 | leifmadsen | just reset the modules.conf file and use autoload=yes |
| 20:08.38 | WIMPy | Or some module has some problem starting. |
| 20:08.43 | WIMPy | (the first) |
| 20:09.17 | oquidave | here http://pastebin.com/3sg9yBHa |
| 20:09.43 | oquidave | leifmadsen> i am using autoload=yes |
| 20:16.09 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:16.49 | oquidave | could it be a file permission issue? i created a user called asteriskpbx and a group asteriskpbx and changed the file permission of the asterisk file system to be owned by asteriskpbx:asterisk as advised in the asterisk book e.g...-rwxr-xr-x 1 asteriskpbx asteriskpbx 4687189 Sep 17 15:58 chan_sip.so |
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| 20:21.38 | oquidave | shouldn't i just re-install asterisk all togather? |
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| 20:28.37 | retentiveboy | Anyone know how to adjust the digit map on a Digium D40 once it's configured for use with Switchvox? It's not handling 10-digit dialing as expected. |
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| 20:39.52 | pabelanger | ~digiumphones |
| 20:40.03 | pabelanger | infobot: ! |
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| 21:00.14 | retentiveboy | pabelanger: my irc foo is lacking... was that directed at me? |
| 21:00.47 | pabelanger | infobot: snack |
| 21:00.47 | infobot | ACTION belches loudly. Oup, 'begpardon. |
| 21:00.56 | pabelanger | ~digiumphones |
| 21:01.00 | pabelanger | infobot: ! |
| 21:01.03 | pabelanger | heh |
| 21:01.16 | pabelanger | retentiveboy: yes, contact Digium for support |
| 21:01.46 | retentiveboy | pabelanger: will but they take *forever*. |
| 21:01.52 | retentiveboy | thx |
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| 21:08.03 | cusco | ~test |
| 21:08.03 | infobot | If you say tset I will not be happy |
| 21:08.15 | cusco | ~listvalues *digium*phone* |
| 21:08.25 | cusco | ~digiumphones |
| 21:08.39 | cusco | :/ |
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| 21:32.54 | oquidave | i still don't get why my modules aren't loading....i've set the right file permission e.g drwxr-xr-x 2 asteriskpbx asteriskpbx 12288 Sep 17 15:58 modules, for the module.conf file -rw-r--r-- 1 asteriskpbx asteriskpbx 1286 Sep 19 22:58 modules.conf and asterisk is running as asteriskpbx 1001 9611 99.8 0.2 16768 6348 pts/1 Rl+ 00:11 23:28 /usr/sbin/asterisk -gvvvc |
| 21:33.05 | oquidave | so what is really wrong?? please help |
| 21:33.57 | oquidave | am running asterisk as a non-root user |
| 21:34.10 | WIMPy | Doesn't it give any clue when you start it with -cvvvdddd |
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| 21:34.36 | mzb | asterisk is a member of the asteriskpbx group? |
| 21:35.19 | oquidave | mzb yes it is |
| 21:35.41 | mzb | change your group permissions on the file and see if it works |
| 21:36.13 | WIMPy | Who needs the group if it's the right user? |
| 21:36.46 | mzb | because 'asterisk' is the user, but the file is owned by 'asteriskpbx' user |
| 21:36.57 | specing | su -s /bin/bash asterisk and see if it works |
| 21:37.10 | specing | if the file is readable/writeable/whatever, that is |
| 21:38.16 | oquidave | specing what does su -s /bin/bash asterisk do? |
| 21:38.41 | specing | oquidave: makes your machines halt and catch fire |
| 21:38.48 | mzb | agreed |
| 21:40.05 | oquidave | specing your kidding...anyway i've done that and it has droped me to the asteriskpbx user |
| 21:42.20 | *** part/#asterisk cresl1n (~Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 21:47.38 | mzb | unless you have *some* diagnostic skills, running as root achieves nothing and is more likely to make your permissions issue worse |
| 21:56.58 | oquidave | mzb so what do you suppose i do? |
| 21:57.23 | mzb | what I said before you were distracted |
| 21:58.28 | oquidave | mzb changing my group permissions? to what? |
| 21:58.52 | mzb | to anything that works |
| 21:59.13 | mzb | if it still doesn't work then you're probably barking up the wrong tree |
| 21:59.26 | mzb | ie: got other issues that need fixing first |
| 22:00.12 | oquidave | like what? i thought you change file permissions, not group permissions...anyway can u give an example |
| 22:00.30 | mzb | chmod g+rwx |
| 22:00.37 | mzb | sorry, but you're going to have to do all the heavy lifting for yourself as I've got to start work now |
| 22:02.10 | oquidave | mzb thanks though |
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| 22:50.13 | navaismo | With the help of asterisk, usb sound card & Raspberry Pi you can emulate an IP DoorBell cheaper than Helios or ITS Pantel http://raspimods.blogspot.mx/2012/09/portero-ip-con-asterisk-y-tarjeta-de.html |
| 22:53.34 | jpsharp | And if they had added sound input on the RasPI, you wouldn't even need the USB card. |
| 22:55.43 | navaismo | only have output |
| 22:56.00 | navaismo | dont know if in rev 2 they added Input |
| 22:56.10 | volga629 | ccss.c:936 ast_set_ccbs_available_timer: 0 is an invalid value for ccbs_available_timer. Retaining value as 4800 what is mean this message ? |
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| 23:08.17 | volga629 | 2012-09-19 19:08:00] WARNING[23500]: res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short ? |
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| 23:13.14 | doug | anyone know of a good web front-end for conference calls (on asterisk, acourse)? |
| 23:18.31 | *** part/#asterisk doug (doug@breakout.horph.com) |
| 23:31.41 | navaismo | volga629, in the mailist say this http://lists.digium.com/pipermail/asterisk-biz/2007-August/023111.html and http://lists.digium.com/pipermail/asterisk-users/2007-July/192717.html |
| 23:33.13 | volga629 | Orrrrrrrrrrr this look like duplex or bad cable |
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| 23:38.37 | volga629 | ignoring 'video' media offer because port number is zero |
| 23:42.25 | volga629 | navaismo thank you |
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