| 00:01.17 | vee8 | Hi all. I cant make outgoing calls from sip phones...it continually rings from the sip side...but the called phone never rings...my zap phone calls out fine and incoming is fine. This is a new install using TDM11B / Asterisk 1.4.19.1 / FreePBX 2.4.0.0 / PBXinaFlash...odd thing is that outgoing used to work until I started customizing...any advice for troubleshooting? |
| 00:02.27 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 00:09.53 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:10.48 | Maliuta | are the outgoing calls to a sip line or a zap line? |
| 00:13.23 | vee8 | I just have a POTS line connected to the FXO on the TDM card for both in and outgoing |
| 00:14.44 | Maliuta | so you are trying to call sip->zap? |
| 00:15.02 | Maliuta | and how are you doing it in your dialplan? |
| 00:15.32 | vee8 | yea i'm trying to call my cell phone from a grandstream ip phone |
| 00:15.53 | notbright | A guy calls into an inbound DID number of mine, but if his callerid matches 2121221111, I want it to do different things.. like playback, etc.. what can I do for that? |
| 00:18.08 | trnzmeta | @#$#^%$# grandstream pieces of shit |
| 00:19.21 | vee8 | Maliuta: My zap trunk has an outgoing dial rule 617NXXXXXX |
| 00:19.43 | Maliuta | vee8: pastebin the config |
| 00:20.11 | Maliuta | notbright: write a dialplan that deals with it |
| 00:20.17 | Maliuta | notbright: read the book |
| 00:20.44 | notbright | mudak |
| 00:21.59 | Maliuta | RTFM fixes most things |
| 00:24.40 | hardwire | like a dictionary flung at your head in english class. |
| 00:25.03 | vee8 | Maliuta: There's so many config files which to paste? Sorry im a bit moobish |
| 00:26.09 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:27.43 | DrkShadow | I could use some help here... the asterisk console gets disconnected at "[May 15 18:25:08] VERBOSE[5307] logger.c: -- Executing [forwardNum@macro-dialexten:13] Gosub("SIP/1098-14a774a0", "<context>|<other-worldly phone no>|1") in new stack" -- and then asterisk crashes. Anything see anything particularly bad about that? My outbound dialing is included into <context>, and the pattern... |
| 00:27.44 | DrkShadow | ...matches the .... oh shit... that must be why... two patterns, each matching everything... |
| 00:27.50 | DrkShadow | hmm. |
| 00:27.52 | Maliuta | vee8: dialplan is extension.conf |
| 00:28.01 | DrkShadow | well no, one matches 3 digits, the other 10 digits. |
| 00:28.50 | DrkShadow | so yeah. Why does Asterisk crash at that line? |
| 00:29.08 | Maliuta | DrkShadow: pastebin your dialplan |
| 00:29.47 | DrkShadow | maliuta: thought you'd say that, and I can't. Aside from it being around 10-15 files, I don't feel I should give away that particular macro. |
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| 00:32.40 | Maliuta | DrkShadow: you can edit it to take out specific numbers etc |
| 00:33.01 | Maliuta | DrkShadow: and you use pastebin not put it into channel |
| 00:33.57 | Maliuta | DrkShadow: and you expect help, but aren't willing to give us enough information to let us diagnose? |
| 00:34.08 | DrkShadow | hold, I just broke my ivr.. |
| 00:35.33 | DrkShadow | maybe I didn't. But OMFG... two hours ago I changed my phone's DTMF from both to RTP because it was dialing numbers twice... and that fixed it. For a bit. Now, it's not dialing any goddamn numbers >:-( |
| 00:36.54 | km2 | i just ported a phone number to my PRI, but when i call it, i see this in the CLI: -- Accepting call from '4435068970' to '4154892874' on channel 0/1, span 1 |
| 00:37.45 | km2 | the 443 # is my cell phone, but the 415 # is NOT the number i dialed, and i don't recogize it. could this be aliases somewhere in the config? (i didn't set up this pbx) |
| 00:38.02 | vee8 | Maliuta: Here's extensions.conf http://pastebin.ca/1019641 |
| 00:38.05 | km2 | s/aliases/aliased/ |
| 00:39.05 | DrkShadow | and.. crash is fixed.. |
| 00:39.10 | DrkShadow | wtf |
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| 00:41.56 | DrkShadow | can I set an ANI with asterisk? does that have meaning? |
| 00:42.49 | DrkShadow | I can... ok. |
| 00:42.55 | JT | you can set CLI |
| 00:42.58 | JT | you cannot set ANI |
| 00:44.50 | DrkShadow | jt: when you mean I cannot set ani.. what do you mean by that, exactly? :-) |
| 00:45.03 | DrkShadow | as in, I'm not authorized, or something else? |
| 00:45.14 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:45.37 | JT | only the telco can set ANI |
| 00:45.57 | DrkShadow | what is a telco in VOIP land? what does vocalocity, vonage, etc do for ani? |
| 00:46.18 | JT | unless they speak SS7, they don't have real ANI |
| 00:47.14 | DrkShadow | so are you implying that e911 and such goes off standard callerid? |
| 00:47.45 | JT | no |
| 00:48.04 | DrkShadow | then they set ani.. or they don't use it by phone number. |
| 00:48.36 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 00:48.43 | JT | the telco generates ANI |
| 00:48.57 | JT | sometimes CLI is used as a basis for that |
| 00:49.01 | DrkShadow | so everyone from Vocalocity gets e911 with the same phone number, same ANI. |
| 00:49.03 | DrkShadow | cli? |
| 00:49.07 | DrkShadow | ah. |
| 00:49.15 | JT | calling line identification/cid |
| 00:49.29 | DrkShadow | which is different from callerid number? |
| 00:49.39 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
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| 00:50.01 | JT | that is callerid |
| 00:50.34 | DrkShadow | in the CALLERID function, I see ani.. but not CID. How do I set that? |
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| 00:51.48 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 00:52.11 | Vec | Hi, I need some help kinda desprately, I am running asterisk 1.4.2 with zaptel 1.4.1 and libpri 1.4.0, I tried to upgrade zaptel and libpri to the latest current stable releases, it did not work, I am now trying to revert back to the previous versions, everything compiles fine but when I do a modprobe zaptel after installing zaptel it says error "zaptel not found" any ideas ? |
| 00:52.31 | alancio | Vec: depmod -a |
| 00:53.06 | Vec | alancio : depmod -a, then modprobe zaptel, same error ? |
| 00:53.44 | JT | DrkShadow: i keep trying to tell you, you can't set it without telco gear |
| 00:53.49 | alancio | Vec: is the module really installed? Run this: find /lib/modules/`uname -r` -name zaptel.ko |
| 00:53.51 | JT | you need to speak SS7 |
| 00:54.12 | DrkShadow | jt: <JT>you can set CLI \n <JT>you cannot set ANI |
| 00:54.32 | JT | why are you pasting back what i said? |
| 00:54.42 | DrkShadow | because you just contradicted yourself. |
| 00:54.47 | JT | no i didn't |
| 00:54.51 | JT | that still applies |
| 00:55.11 | alancio | does anybody in here have experience with Grandstream's GXP2000? |
| 00:55.12 | JT | ANI is not CLI |
| 00:55.18 | Vec | alancio : yeh you right, thats odd everything compiles and installs |
| 00:55.19 | DrkShadow | I asked how I can set CID, you said I can't. First you said I can set CID. How's that not a contradiction? I believe it's an excellent example. |
| 00:55.31 | JT | read what i said again |
| 00:55.35 | JT | you can set CID |
| 00:55.38 | JT | you cannot set ANI |
| 00:55.44 | JT | how much simpler can i make it? |
| 00:55.49 | alancio | Vec: what distro are you using? are you installing from source or package? |
| 00:56.07 | DrkShadow | egh, term confusion -- cli = cid |
| 00:57.05 | Vec | alancio : running RHEL4 installing from source |
| 00:57.42 | alancio | Vec: you have tu run make, then make install |
| 00:57.56 | Vec | alancio : yeh I know |
| 00:58.09 | Vec | alancio : I think it may be an issue with the kernel sources I am using |
| 00:58.29 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 00:58.38 | Vec | alancio : I think it may be an issue with the kernel sources I am using |
| 00:59.22 | alancio | Vec: then fix it ;) |
| 01:00.36 | Vec | alancio : is /proc/version defintately the kernel version I am currently running, can I specify the kernel source folder in zaptel ./configure |
| 01:01.12 | alancio | Vec: yes, /proc/version is the current kernel version |
| 01:02.04 | alancio | Vec: why don't you just copy zaptel.ko somwhere inside /lib/modules/`uname -r`/extra, then run depmod -a |
| 01:02.31 | Vec | alancio : nice idea, I am not so familiar with kernel modules, will try that one |
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| 01:08.10 | Vec | alancio : it seems like I only have the kernel source for 2.6.9-42.0.10.EL-x86_64 but I am running 2.6.9-42.0.10.ELsmp-x86_64, any idea how I can get the kernel source again ? |
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| 01:10.54 | deltaray2 | What is a trunkgroup? Like I have two pots lines that I'm planning on hooking up to my 4 port FXO card. Are those two pots lines one trunk group? |
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| 01:35.25 | Vec | How does zaptel know which kernel source to use ? |
| 01:35.46 | Dossy | Does Asterisk have the ability to provide MWI? |
| 01:35.50 | Vec | its 2.35 pm and I really have to get this fixed in 2 hours or I am dead |
| 01:35.59 | CoffeeKid | my guess is it goes to /usr/src/linux |
| 01:36.11 | CoffeeKid | thats where the latest kernel is usually on most linux systems |
| 01:36.14 | JT | uname -a probably too |
| 01:36.35 | CoffeeKid | yeah could be /usr/src/`uname -a` |
| 01:37.05 | Vec | ok, for some reason it is not installing the module in the correct lib/modules and when I do a copy it manuelly and do a modprobe zaptel, I get invalid module format ? |
| 01:37.07 | Maliuta | or it could look at /lib/modules/uname -a/build |
| 01:37.28 | CoffeeKid | /usr/src/`uname -r` |
| 01:37.31 | CoffeeKid | thats the one |
| 01:37.32 | Maliuta | or source |
| 01:39.06 | Vec | Maliuta : it looks like my /lib/modules/uname -a/build does not exist, that kinda makes sence that that would cause an issue |
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| 01:39.50 | Vec | ? |
| 01:40.37 | JT | Vec: you probably don't have the kernel source |
| 01:41.30 | Vec | JT: pretty sure I do its in /usr/src/linux which links to /usr/src/kernels/2.6.9-42.0.10.EL-x86_64 |
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| 01:43.44 | Maliuta | Vec: you'd need to have the kernel source package installed |
| 01:43.56 | Maliuta | or at least the headers |
| 01:43.56 | dFence | d'arrr typical netgear: advertise a 24port PoE switch and conveniently mention in the small-print "oh well.. supports only 12 PoE devices"... BULLCRAP |
| 01:44.20 | JT | most cheap PoE switches are like that |
| 01:47.30 | drmessano | Good rule of thumb |
| 01:47.41 | drmessano | There is no such thing as a 1u 24 port PoE switch |
| 01:49.24 | dFence | 1u? |
| 01:49.43 | dFence | single-unit |
| 01:49.48 | drmessano | yah |
| 01:50.11 | JT | sure there is |
| 01:50.15 | JT | if you pay enough |
| 01:50.21 | drmessano | There is? |
| 01:50.29 | drmessano | I havent even seen a Cisco that's 1U |
| 01:50.30 | JT | i think cisco and hp make one |
| 01:51.43 | drmessano | Ciscos are usually 1U per power rail of 12 |
| 01:51.59 | drmessano | So a 2U has two power buses |
| 01:52.30 | JT | http://www.cisco.com/en/US/products/ps6571/index.html |
| 01:52.37 | JT | Cisco Catalyst Express 500-24PC Switch |
| 01:53.50 | JT | also 2960-24PC-L |
| 01:54.00 | dFence | what about linksys srw 2024? |
| 01:54.17 | drmessano | Those seem to be a bit underpriced for full 24 port PoE |
| 01:54.19 | dFence | doesn't specifically say _not_ supporting all 24 ports with poe ;D |
| 01:54.42 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:56.02 | *** join/#asterisk jsolis (n=jimmy@201.230.135.91) |
| 01:56.30 | dFence | and for the love of god... why are there almost only 8port switches and then right on with 24 ports!? |
| 01:56.40 | notbright | A call comes in, if the callerID on that incoming call matches, I want it to give busy tone, if not move on and do what you do. -> Shall it need to be done by If($CALLERID(num)=4194423422)?Playback(sounds/busy) or something like that? |
| 01:56.55 | dFence | i gotta wire 13 devices per floor + 2 uplinks *garrgarr* |
| 01:57.17 | JT | dFence: i think you mean the SRW2024P |
| 01:57.22 | JT | the SRW2024 has no PoE |
| 01:57.31 | JT | and wrt the SRW2024P |
| 01:57.33 | JT | Standards-based IEEE 802.3af Power over Ethernet (PoE) - Up To 12 ports at 15.4W for 180W of total PoE power |
| 01:57.56 | dFence | JT: datasheet says so |
| 01:58.27 | jsolis | why i cant call WARNING[2421]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 01:59.04 | JT | dFence: says what? |
| 01:59.08 | jsolis | zapata.conf // |
| 01:59.08 | jsolis | group=0 |
| 01:59.08 | jsolis | callerid = asreceived |
| 01:59.08 | jsolis | context=DID_trunk_1 |
| 01:59.08 | jsolis | channel => 1,2 |
| 01:59.15 | JT | ~pb |
| 01:59.16 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 01:59.18 | drmessano | JT: I need to get a better look at those switches.. Havent seen those before.. looks like they're 370W switches |
| 01:59.26 | jsolis | [prueba] |
| 01:59.26 | jsolis | exten => _XXXXXXX,1,Dial(Zap/g0/${EXTEN},60) |
| 01:59.29 | dFence | naming in the linksys-portfolie seems kinda random to me anyways.. once found a router that went by 3 different model-numbers depending on version and country |
| 01:59.39 | JT | drmessano: i bet they cost enough |
| 01:59.40 | dFence | JT: http://www-de.linksys.com/servlet/Satellite?blobcol=urldata&blobheadername1=Content-Type&blobheadername2=Content-Disposition&blobheadervalue1=application%2Fpdf&blobheadervalue2=inline%3B+filename%3DSRW2024v1.2_ds.pdf&blobkey=id&blobtable=MungoBlobs&blobwhere=1193772803987&ssbinary=true&lid=8978137406B02 |
| 01:59.52 | drmessano | JT: $1500 <-- which isnt bad |
| 02:01.20 | JT | dFence: i don't need a link, i need you to explain what you mean |
| 02:01.24 | JT | what does it say? |
| 02:01.45 | JT | drmessano: wow |
| 02:01.50 | dFence | JT: that it has PoE but it doesn't say how many ports it can serve with poe |
| 02:02.02 | JT | drmessano: what is your price on a cisco 24 port gigabit switch? |
| 02:02.17 | JT | dFence: that link says nothing about PoE that i can see |
| 02:02.22 | JT | that's the non PoE version |
| 02:02.35 | JT | and the one that is a PoE version says 12 ports, as i pasted before. |
| 02:03.18 | dFence | JT: in the "Standards"-Tab: 802.3af |
| 02:03.40 | drmessano | JT: I would need to check |
| 02:04.00 | dFence | what do you guys think of the snom-hardphones? |
| 02:04.26 | JT | dFence: an error in the datasheet. |
| 02:04.34 | JT | there SRW2024 has no PoE |
| 02:04.38 | JT | s/there/the/ |
| 02:05.09 | dFence | cute bot! JT: wouldn't surprise me.. linksys ^^ |
| 02:05.18 | notbright | Actually, better idea, a phone call comes into [incoming], first thing I want to do is check a text file denied.txt where there are phone numbers, and check if the callerid of the incoming call matches any in the text file, if yes, playback or whatever, if not, continue to whatever. How can I make something like that? |
| 02:05.30 | JT | drmessano: i've seen prices here equivalent to USD$4000 |
| 02:05.32 | JT | which is insane |
| 02:05.58 | JT | dFence: if it has PoE, they're would advertise it loudly, not hide it in the standards |
| 02:06.11 | JT | they advertise is loudly for the SRW2024P |
| 02:07.19 | notbright | I think what I wanna do is cool. Is it THAT cool that no one knows how to do it? |
| 02:08.29 | JT | notbright: heaps of people have setup blacklists before |
| 02:08.34 | JT | there's a few different ways to do it |
| 02:08.53 | notbright | JT: Would you be able to point me inthe right direction please? What can I search for? |
| 02:09.24 | JT | if you only want to check against a couple of numbers, you can just have a different dialplan match for them |
| 02:09.26 | km2 | anyone know why i would see this in the CLI: B-channel 0/1 successfully restarted on span 1 (one for each of our T1 lines?) |
| 02:09.34 | km2 | there's no problem, just haven't seen that before |
| 02:10.26 | JT | exten => <CALLED>/<CALLING,1,Blah... |
| 02:11.51 | notbright | exten=>2120001200/4192291112,1,Answer |
| 02:11.53 | notbright | Like that? |
| 02:12.05 | notbright | Where 212 is the caller number, and 419 is the calling guy? |
| 02:12.33 | JT | 212... is the callED number |
| 02:12.38 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 02:13.22 | notbright | Sorry, that's what I meant. |
| 02:13.34 | notbright | How can I do so if there is more than one number that I want to check for? |
| 02:14.18 | JT | more than one extension |
| 02:14.30 | JT | if it gets to a lot you may want to do something with a db |
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| 02:19.42 | notbright | Just a few numbers, like 5-10. |
| 02:19.48 | notbright | I wanna use it from a list. |
| 02:21.20 | notbright | Seems like everyone does the blacklist on trixbox, etc. No one shows how I can do it in the dialplan manually lol |
| 02:22.58 | JT | there's so many ways you can do it |
| 02:23.01 | JT | astdb |
| 02:23.05 | JT | variables |
| 02:23.10 | JT | sql db |
| 02:23.41 | JT | all you need to do is check against the list and perform a different action depending on a number's status in a list |
| 02:25.05 | notbright | Yeah, is there a way that doesn't require using dbs? I dont know much on using databases and not much proficient on using the dbput dbget functions. |
| 02:26.28 | JT | you can use agi or system |
| 02:27.23 | notbright | lol looks like its all the functions I've stayed away from! |
| 02:36.41 | dFence | hm... any recomendations for a wake-up-call feature for asterisk? or do i have to bastel one myself? |
| 02:39.25 | km2 | dFence, you could use call files in /var/spool/asterisk/outgoing. see http://www.the-asterisk-book.com/unstable/call-file.html |
| 02:39.44 | dFence | merci |
| 02:40.02 | JT | also see sample.call in the source |
| 02:40.13 | JT | you could tie it in with cron |
| 02:41.24 | km2 | i read somewhere that it's importamt to use mv, not cp, when putting call files in that directory, because of the way the file system works |
| 02:43.18 | jsolis | hey how to uninstall zaptel-1.4.10.1 |
| 02:43.32 | JT | yeah apparently |
| 02:43.52 | JT | so the best way is to copy to a temp file then move the temp |
| 02:44.08 | km2 | right |
| 02:44.51 | jsolis | hey guys how to uninstall zaptel-1.4.10.1 i cant make calls by group :( |
| 02:45.08 | JT | i have a nice little callback system hacked together with System() and a .call file generated by a shell script :) |
| 02:45.26 | JT | don't really need to use it much though |
| 02:45.56 | km2 | anyone recommend good asterisk books aside from the o'reilly one? |
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| 02:50.52 | drmessano | I just saw someone describe having two channels as "Hey, I got two line rollover" |
| 02:51.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:51.17 | drmessano | channels != rollver |
| 02:51.19 | drmessano | channels != rollover |
| 02:52.16 | outtolunc | ringing != barking? |
| 02:53.10 | drmessano | woof |
| 02:53.57 | JT | interesting term |
| 02:54.06 | JT | they normally call it line hunt with analogue lines here |
| 02:55.45 | drmessano | or a hunt group or just a hunt |
| 02:55.54 | drmessano | Rollover is slang |
| 02:56.00 | drmessano | Either way |
| 02:56.12 | drmessano | 2 SIP Channels aren't even a hunt |
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| 03:04.49 | notbright | How can I not load a skinny module? |
| 03:05.05 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 03:06.58 | notbright | noload => chan_skinny.so |
| 03:07.11 | notbright | seems to have worked, because chan_skinny doesn't show up when I do "show modules" anymo |
| 03:08.02 | notbright | [May 15 22:07:43] WARNING[24961]: app_lookupblacklist.c:106 lookupblacklist_exec: LookupBlacklist is deprecated. Please use ${BLACKLIST()} instead. |
| 03:08.04 | notbright | Great! |
| 03:09.51 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:11.34 | *** join/#asterisk gmoque (n=gmoque@189.180.125.106) |
| 03:12.55 | notbright | Does anyone know what the soundfile for a busy tone is? |
| 03:13.34 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 03:15.06 | notbright | Hmm, got it. |
| 03:15.14 | notbright | I'm just using Busy(5) |
| 03:15.21 | notbright | The problem is it's not taking 5 as a timeout. |
| 03:16.48 | notbright | How can I make the channel hangup after doing Busy for 5 seconds? |
| 03:16.56 | notbright | The 5 second timeout doesn't seem to be working.. |
| 03:18.19 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 03:19.03 | deltaray2 | Is there a way to display in the log what the callerid was of a number that called? Like what is the function to do this? |
| 03:20.36 | notbright | deltaray2: Try NoOp or Verbose (show application verbose) |
| 03:21.24 | deltaray2 | notbright: Ah thanks |
| 03:21.52 | notbright | Try it first, lol i'm still a newb |
| 03:22.12 | notbright | Why doesn't the timeout for Busy(5) work!!! |
| 03:24.56 | deltaray2 | Hmmm, it doesn't seem to be getting the caller id. |
| 03:25.46 | C4colo | drmessano: if "rollover" on sip lines was the only difference between VoIP and PSTN |
| 03:25.53 | C4colo | that is hard to explain |
| 03:26.20 | deltaray2 | I guess what I want is the number that is calling, not the caller id (if thats any different). |
| 03:27.47 | deltaray2 | Ahh, there we go, $CALLERID(num) |
| 03:28.07 | *** join/#asterisk adr3nalin3 (n=afink@ip68-13-89-102.om.om.cox.net) |
| 03:30.21 | drmessano | heh |
| 03:32.31 | Bhaal | how do I stop asterisk from sending the local extension number along with the callerid? |
| 03:34.01 | *** join/#asterisk rcy` (n=rcy@S010600207810532a.vc.shawcable.net) |
| 03:37.27 | dFence | eeeerrrrm.... i have a variable ${day}=10 how can i modify the output in-time so that i get 11 (day+1)? |
| 03:42.14 | adr3nalin3 | Hello, I am having some trouble when recording voice menus. The quality of the recording is very poor and hard to understand. It is very similar to the poor quality I had on voicemails when I had no codecs selected on each user. Is there anything you can guys can suggest I check? |
| 03:46.24 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 03:49.41 | *** join/#asterisk drbrown (n=drbrown@oh-76-1-215-18.dhcp.embarqhsd.net) |
| 03:53.44 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 04:03.25 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583729.dsl.bell.ca) |
| 04:10.05 | *** join/#asterisk thx2000 (n=bob@netblock-208-127-150-56.dslextreme.com) |
| 04:15.06 | mackes | So I have a question. |
| 04:15.23 | mackes | What Sip phone does most Asterisk users use> |
| 04:18.46 | JT | polycom |
| 04:19.23 | km2 | same |
| 04:19.32 | thx2000 | I like Aastra |
| 04:19.58 | adr3nalin3 | I like my snom 360s |
| 04:20.31 | alancio | have any of you seen the snom m3? |
| 04:20.38 | drmessano | Most asterisk users use Skype |
| 04:20.48 | drmessano | Ok, no |
| 04:20.49 | rob0 | I think I'm going to get a Polycom IP-320 for around US$100 |
| 04:21.04 | drmessano | I like my Linksys stuff |
| 04:21.05 | rob0 | looks like a nice choice |
| 04:21.13 | adr3nalin3 | alancio: yes look pretty cool |
| 04:21.43 | alancio | I wonder why they used DECT instead of wifi |
| 04:23.37 | adr3nalin3 | most likely to eliminate interference problems it uses less common freqs |
| 04:24.21 | adr3nalin3 | and Snom is from Germany so its probably more standard for them |
| 04:24.34 | alancio | I want to buy this phone |
| 04:25.05 | alancio | I'm going to germany in a month, I wonder if I can see it in person before buying it |
| 04:25.11 | JT | more likely because mobile wifi sucks big fat donkey balls for streaming media like voip |
| 04:25.18 | JT | dect performs way better |
| 04:25.59 | alancio | my other option would be to buy an ATA for my wireless panasonic phone |
| 04:26.05 | JT | mobile wifi is a hopeless cause unless you spend zillions of dollars on your wireless infrastructure |
| 04:26.57 | alancio | why does wifi suck so much for voip? |
| 04:27.04 | alancio | range? |
| 04:27.16 | alancio | battery life? |
| 04:27.45 | JT | well most of the phones out there suck for battery life, but that's not the reason |
| 04:27.54 | adr3nalin3 | commonly used freqs and EMI RFI interference. Try using one in a steel building |
| 04:27.58 | JT | mobile wifi has lots of jitter and packet loss issues |
| 04:28.52 | dFence | oi!? strptime isn't working |
| 04:29.52 | rob0 | Obese equidae testicle suction |
| 04:30.20 | rob0 | s/cle/cular/ |
| 04:31.58 | alancio | I currently have 3 grandstream gxp2000 phones |
| 04:32.14 | alancio | thay all work fine, despite all the negative comments I've found on every single forum |
| 04:32.37 | JT | have you ever used a better phone? |
| 04:32.41 | [TK]D-Fender | ~gs |
| 04:32.46 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 04:32.49 | alancio | good point! hehe |
| 04:32.49 | [TK]D-Fender | ~grandstream |
| 04:32.49 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 04:33.08 | JT | it's hard to know what you're missing out on if you haven't tried better :) |
| 04:33.23 | drmessano | oh god |
| 04:33.25 | alancio | thats true, I'm planning on buying 8 more phones |
| 04:33.31 | drmessano | alancio: Remember when I said I didnt hate you |
| 04:33.35 | alancio | I don't know if I should go with these again |
| 04:33.45 | drmessano | alancio: Grandstream? You crossed the line now |
| 04:33.47 | rob0 | Hmmm, I use X101P's and Grandstream exclusively. |
| 04:33.55 | drmessano | WOFFLE |
| 04:34.04 | JT | rob0: GENUINE X101Ps? :P |
| 04:34.05 | alancio | drmessano: don't hate me, I also have a Linksys |
| 04:34.19 | alancio | a Linksys SPA961 |
| 04:34.26 | drmessano | X101P and a BT102 = Two turntables and a microphone |
| 04:34.35 | drmessano | WHERE ITS AT |
| 04:34.45 | alancio | but I don't see whats wrong with grandstream |
| 04:34.47 | rob0 | Mine are Chinese. Bought 'em from a guy on a street corner in a trench coat. |
| 04:36.26 | dFence | what's wrong with turntables?! |
| 04:36.35 | alancio | I'm planning to convert two of these grandstreams into doorphones |
| 04:36.48 | JT | s/phones/stops/ |
| 04:36.49 | JT | :P |
| 04:36.52 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:36.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:37.21 | alancio | so, whats wrong with them? |
| 04:37.48 | JT | apart from very buggy firmware, awful design, cheap manufacture, and echo, not much i guess |
| 04:37.49 | alancio | they have PoE, big illuminated lcd screen, 7 speed-dials |
| 04:38.29 | drmessano | My Rotex watch has backlight display too.. $9.95 from a chinaman on a street corner |
| 04:38.48 | alancio | my Linksys SPA961 doesn't have backlight display |
| 04:38.54 | alancio | it doesn't have speed-dials either |
| 04:39.15 | JT | or bugware bingo |
| 04:39.20 | drmessano | Thats crap |
| 04:39.28 | alancio | maybe I just had good luck with my firmware |
| 04:39.36 | drmessano | My SPA941 has a damn directory |
| 04:39.44 | drmessano | How can your 961 not? |
| 04:39.52 | rob0 | load => chan_turntable.so |
| 04:39.59 | hardwire | turntable? |
| 04:40.11 | alancio | sorry, I have a 941, just the same as yours |
| 04:40.34 | rob0 | load => chan_8track.so |
| 04:40.37 | alancio | drmessano: I mean hardware buttons, which you can assign to speeddials |
| 04:40.44 | hardwire | rob0: geek |
| 04:41.09 | rob0 | OLD one. I actually HAD a car with an 8-track player. |
| 04:41.17 | drmessano | alancio: Thats a waste of space |
| 04:41.47 | drmessano | alancio: and who needs a backlit display in a lit office? |
| 04:42.03 | alancio | drmessano: I'm using them at home |
| 04:42.09 | drmessano | alancio: so> |
| 04:42.11 | drmessano | alancio: so? |
| 04:42.17 | alancio | I'd like to see the callerid without turning on the light at night |
| 04:42.29 | drmessano | alancio: Oh forget it.. enjoy your grandstreams |
| 04:42.59 | alancio | I bought the grandstreams new for 70$, the SPA961 used for the same price |
| 04:43.11 | drmessano | You just said you had a 941 |
| 04:43.12 | JT | $70 too much |
| 04:43.18 | alancio | errr 941! |
| 04:43.36 | alancio | 70$ too much for which? |
| 04:43.37 | hardwire | load => chan_irc.so |
| 04:43.50 | alancio | 70$ was a good deal for the SPA941 |
| 04:43.52 | JT | a grandstream |
| 04:43.54 | drmessano | Which you claim didnt have speed dials.. which makes me doubt you own anything but some cheap ass BT101s |
| 04:44.40 | alancio | drmessano: the gxp2000 have 7 buttons for speed-dials |
| 04:44.41 | drmessano | You got a used SPA-941 for $70? |
| 04:44.46 | drmessano | I got one new for $85 |
| 04:45.00 | alancio | really? which store? |
| 04:45.01 | drmessano | and you paid $70 for a grandstream? |
| 04:45.05 | hardwire | speaking of SPA stuff, I'd really love an open-source firmware for most of the sipura phones |
| 04:45.31 | alancio | yes, 70$ |
| 04:46.00 | alancio | BTW, the SPA941 can only use 2 lines, I think I need a license for the other two |
| 04:46.08 | hardwire | nyet |
| 04:46.10 | JT | not anymore |
| 04:46.11 | hardwire | upgrade the firmware |
| 04:46.23 | alancio | mmm ok |
| 04:46.28 | drmessano | Thats old ass new |
| 04:46.29 | drmessano | Thats old ass news |
| 04:46.42 | hardwire | drmessano: give us some new ass news. |
| 04:46.56 | drmessano | doesnt think alancio owns an SPA-941 |
| 04:46.58 | hardwire | you can put it there, next to yesterdays ass news. |
| 04:47.37 | drmessano | hardwire: I wasn't addressing you |
| 04:47.58 | alancio | drmessano: just checked the ebay message, its a SPA941, it cost 60$ |
| 04:47.59 | hardwire | drmessano: :| |
| 04:48.15 | alancio | drmessano: it has 4 buttons on the right, with leds |
| 04:48.29 | drmessano | alancio: I can look at google images too |
| 04:48.44 | hardwire | http://www.voip-info.org/wiki/view/SPA-941 |
| 04:48.48 | alancio | haha |
| 04:50.40 | alancio | http://www.mizrahi.com.ve/spa941.png |
| 04:51.25 | hardwire | woot |
| 04:51.27 | hardwire | I'm in a screen shot |
| 04:51.49 | hardwire | And you Won something on Ebay. |
| 04:51.51 | drmessano | 4.1.8 firmware? |
| 04:51.59 | drmessano | Is it 2002? |
| 04:52.00 | denon | and your clock is really hosed |
| 04:52.02 | alancio | yeah, its the phone, I was checking the price |
| 04:52.13 | alancio | my click? its 12:20 here |
| 04:52.19 | alancio | s/click/clock/ |
| 04:52.25 | hardwire | alancio: what's sexy or not? |
| 04:52.32 | denon | 12:20? |
| 04:52.34 | drmessano | and you spend too much time on "Sexy or not" |
| 04:52.34 | denon | what tz? |
| 04:52.37 | alancio | http://www.sexyono.com/ |
| 04:52.40 | hardwire | haha |
| 04:52.42 | alancio | GMT-4.5 |
| 04:52.44 | hardwire | this is funny |
| 04:52.54 | denon | huh, didnt know there were fractions of gmt |
| 04:52.56 | hardwire | 1008 unread messages, shameful. |
| 04:52.57 | denon | I guess I should have .. |
| 04:53.01 | drmessano | Thats really sad |
| 04:53.05 | hardwire | at least you use amarok |
| 04:53.16 | drmessano | So what is your "Sexy or not" score |
| 04:53.23 | hardwire | I'd like to know too |
| 04:53.23 | drmessano | This is SOO going on Digg... so keep talking |
| 04:53.33 | alancio | I'm looking at the pictures there |
| 04:53.38 | drmessano | Ohhh |
| 04:53.46 | denon | alancio: so is that like caracas? |
| 04:53.51 | alancio | yeah |
| 04:53.52 | drmessano | yeah.. I heard that one before.. back when yahoo mattered |
| 04:53.55 | drmessano | Anyway |
| 04:53.57 | drmessano | Score? |
| 04:53.58 | hardwire | woohoo.. 60 gigs of md5sums done |
| 04:53.59 | denon | or venezuala or somethin? |
| 04:54.11 | alancio | yes, caracas, venezuela |
| 04:54.23 | denon | coo |
| 04:54.28 | alancio | we used to be GMT-4, until our stupid president wanted a new headline |
| 04:55.11 | drmessano | alancio: Are you a 5.5? |
| 04:55.26 | alancio | 5.5? |
| 04:55.31 | hardwire | I bet I'd be a 6.3 |
| 04:55.54 | hardwire | how can IRC go on DIGG? |
| 04:55.59 | hardwire | online logs? |
| 04:56.01 | drmessano | That pic is |
| 04:56.04 | hardwire | oh :) |
| 04:56.10 | drmessano | Along with some nice captioning |
| 04:56.18 | hardwire | you should slap it up on tinypic.com or something real quick |
| 04:56.35 | alancio | why are you so amazed with the screenshot? |
| 04:56.44 | drmessano | As soon as I get done photoshopping it |
| 04:57.39 | hardwire | alancio: boredom.. |
| 04:57.42 | hardwire | why else does digg exist? |
| 04:58.03 | alancio | there is a blogger in my city |
| 04:58.32 | hardwire | just the one? |
| 04:58.37 | drmessano | There's probably lots of bloggers in your city |
| 04:58.44 | alancio | he goes into cybers and looks for files that users left on the desktop |
| 04:58.45 | drmessano | "Theres a bathroom in my town" |
| 04:58.54 | hardwire | is being the town blogger a bit like being the town idiot? |
| 04:59.04 | drmessano | Ask SexyOrNot |
| 04:59.11 | hardwire | alancio: don't you own a cafe? |
| 04:59.22 | alancio | no |
| 04:59.34 | drmessano | alancio: Please tell me you're wearing pants |
| 04:59.38 | hardwire | hmm.. some other caracas person was in here a year ago doings some telephony evil |
| 04:59.47 | alancio | http://archivosabandonados.blogspot.com/ |
| 05:00.10 | alancio | thats what I was talking about |
| 05:00.18 | alancio | all these pictures were left behind on public access computers in the city |
| 05:00.27 | alancio | he just makes stories about them |
| 05:00.35 | hardwire | NSF.. Who's at work? |
| 05:00.38 | ManxPower | I should stop by Digium sometime soon |
| 05:00.58 | alancio | hardwire: do you remember the nickname? |
| 05:01.01 | ManxPower | Well, apparently I was scrolled back several hours. nevermind. |
| 05:01.07 | drmessano | LOL |
| 05:01.14 | hardwire | alancio: I .. maybe? |
| 05:01.49 | drmessano | ManxPower: Just to fill you in, alancio is a 5.5 and may or may not be wearing any pants while he asterisks |
| 05:02.06 | hardwire | Jimmy Laguna |
| 05:02.19 | alancio | I'm only wearing soxs, but don't worry, I'm a 6.8 |
| 05:02.26 | alancio | s/soxs/socks/ |
| 05:02.27 | hardwire | He was in Caracas, but he goes by bolivian |
| 05:02.37 | hardwire | gmail ftw. |
| 05:03.55 | *** join/#asterisk gomel77 (n=gomel77@wn2nat45.beelinegprs.ru) |
| 05:04.07 | alancio | shouldn't have posted that screenshot |
| 05:04.26 | hardwire | alancio: I would have thought first. :) |
| 05:04.47 | hardwire | I've done my fair share of posting my passwords to IRC, as well as incredibly accidentally indecent screenshots in help channels. |
| 05:05.14 | hardwire | alancio: what are you doing with *, anyways? |
| 05:05.25 | drmessano | alancio: Before posting any screenshots online, you must ask yourself one thing |
| 05:05.36 | drmessano | alancio: Am I sexy, or not? |
| 05:05.48 | alancio | I just bought a digium TDM card, some phones, and I'm trying to get all working as I want to |
| 05:06.05 | alancio | drmessano: did you visit the website? |
| 05:06.18 | hardwire | alancio: what dist of linux? |
| 05:06.29 | alancio | CRUX, http://crux.nu/ |
| 05:06.37 | drmessano | alancio: No, im afraid to |
| 05:06.38 | Bhaal | I have a cordless phone connected to an ATA, the cordless phone has a single line display for CID ... How do I stop its extension number from being displayed instead of the actual CID number? |
| 05:07.18 | alancio | sexyornot.com is a porn website? |
| 05:07.21 | drmessano | That doesnt make any sense |
| 05:08.06 | drmessano | If the extension number of the cordless is being displayed as the CID when you receive a call, you have something painfully borked. |
| 05:08.06 | alancio | hardwire: I'm also trying to use the phones as doorbells |
| 05:08.49 | Bhaal | drmessano: Well no, the actual CID is being sent aswell, but as the extension number gets sent first, thats whats displayed on the phone... |
| 05:09.16 | Bhaal | If I have a phone with a 2 line display, it does display both the extension number and the CID |
| 05:09.22 | alancio | and I use it to talk with my family, they leave in other countries |
| 05:10.25 | Bhaal | drmessano: Just like when using trixbox, the queue the call has been sent to is displayed on the phone along with the CID |
| 05:10.51 | drmessano | Bhaal: you are making no sense.. You have a cordless with a single line display.. and then you're telling me about its CID being SENT |
| 05:11.03 | drmessano | What does its DISPLAY have to do with how it's CID is SENT? |
| 05:11.42 | Bhaal | *sigh* the ATA the phone is plugged into connects to asterisk ... I am talking about whats being sent to the phone when a call comes in... not out |
| 05:11.48 | Bhaal | :) |
| 05:12.22 | drmessano | Seems like you need support for your cordless phone |
| 05:12.25 | drmessano | not asterisk |
| 05:12.26 | Bhaal | Sorry, really annoying day here in .au ... Wife is sick, too sick to look after daughter, so work is shitty with me for not coming in, etc etc etc |
| 05:12.27 | drmessano | get a new phone |
| 05:12.52 | Bhaal | drmessano: It is just a regular cordless phone... |
| 05:12.55 | drmessano | If its not displaying CID thats working on other phones, it's a shit phone.. toss it |
| 05:12.59 | Bhaal | connected with an ATA |
| 05:13.02 | Bhaal | to asterisk |
| 05:13.08 | drmessano | Bhaal |
| 05:13.10 | drmessano | You said |
| 05:13.17 | drmessano | [01:09] <Bhaal> If I have a phone with a 2 line display, it does display both the extension number and the CID |
| 05:13.23 | drmessano | So Asterisk is working |
| 05:13.29 | Bhaal | I know this... |
| 05:13.32 | Bhaal | I know asterisk is work |
| 05:13.33 | Bhaal | ing |
| 05:13.45 | Bhaal | I know this is the default for how asterisk behaves... |
| 05:13.46 | drmessano | You have a phone with a one line display that displays the CID in a shitty way |
| 05:13.49 | drmessano | Toss the phone |
| 05:13.56 | alancio | Bhaal: what phone is it? |
| 05:13.57 | Bhaal | I am asking if there is a way to change this in the config files |
| 05:14.51 | Bhaal | alancio: Panasonic 1.8Ghz phone, unsure of actual model... |
| 05:15.28 | alancio | if asterisk sets the CID number and name, will the name be displayed in the phone? |
| 05:15.39 | Bhaal | No, tried... |
| 05:15.43 | alancio | or is the name looked up in the phone's directory, based on the number |
| 05:15.49 | Bhaal | yes |
| 05:16.45 | alancio | I still don't understand the problem |
| 05:17.14 | alancio | you said: "How do I stop its extension number from being displayed instead of the actual CID number?" |
| 05:17.23 | drmessano | The problem is that the crappy phone doesnt display all the CID info |
| 05:17.30 | drmessano | Which has nothing to do with * |
| 05:17.43 | Bhaal | When asterisk sends the call to the ATA, the callerID information it sends consists of the ATAs extension number, followed by the CID of the incoming call |
| 05:18.12 | alancio | Bhaal: maybe some configuration in the ATA |
| 05:18.29 | Bhaal | No... I see the same details from any asterisk server... |
| 05:18.45 | alancio | Bhaal: what happens with another phone and the same ATA and asterisk server? |
| 05:19.16 | Bhaal | alancio: I have 1 regular POTS fone which is capable of displaying all the information sent to it... |
| 05:19.20 | alancio | Bhaal: it must be the ATA, because the phone doesn't know its own extension number |
| 05:19.37 | Bhaal | It displays the extension number on the first line, followed by the CID on the second line |
| 05:20.04 | alancio | Bhaal: does the other phone display its extension too? |
| 05:20.15 | Bhaal | alancio: I know the fone doesnt know its own extension number, these a normal POTS fones ... The information is being sent from asterisk... |
| 05:20.31 | Bhaal | alancio: Yes, the other phone displays the extension number as well... |
| 05:20.39 | alancio | then its the ATA or asterisk |
| 05:21.00 | Bhaal | alancio: I know that the cordless does work... If I connect the ATA directly to my SIP provider, I get the caller ID correctly... |
| 05:21.03 | alancio | are you using set(CALLERID(num)=xxx)? |
| 05:21.29 | alancio | check your extensions.conf, see if you are setting the callerid to something wrong |
| 05:21.53 | Bhaal | Set(CallerIDString=${CALLERID(num) |
| 05:22.47 | Bhaal | }) is meantr to be on the end |
| 05:22.55 | Bhaal | didnt grab it with the cut'n'paste |
| 05:23.07 | alancio | thats not setting the caller id, thats just setting a var |
| 05:23.14 | alancio | use pastebin |
| 05:23.25 | Bhaal | sorry |
| 05:23.54 | Bhaal | What should the line be? |
| 05:24.52 | alancio | maybe it looks like this: Set(CALLERID(num)=${EXTEN} ${CallerIDString}) |
| 05:25.23 | Bhaal | Thats not it presently, but what should it be? |
| 05:25.41 | Bhaal | Set(CALLERID(num)=${CallerIDString}) |
| 05:25.43 | Bhaal | ? |
| 05:25.45 | alancio | you shouldn't need it |
| 05:26.15 | alancio | it is used only to put something else in the caller id |
| 05:26.33 | alancio | and since you are getting your own extension in the caller id, maybe you are using it wrongly |
| 05:26.39 | Bhaal | Okey, well I shall see how that goes now... |
| 05:27.05 | alancio | ok, I hope I helped |
| 05:28.09 | alancio | now I'll read my 1008 emails |
| 05:28.40 | Bhaal | :) |
| 05:28.47 | Bhaal | Thanks for the help |
| 05:28.56 | alancio | you are welcome |
| 05:37.06 | km2 | i just ported a phone number to my PRI, but when i call it, i see this in the CLI: "-- Accepting call from '4435068970' to '4154892874' on channel 0/1, span 1". that's odd because the -2874 number is NOT the number i dialed (the new port), though it is one of our DIDs. is this probably just a problem with the PRI provider? if not, where could i investigate on my end? |
| 05:43.29 | ManxPower | the problem is not on your end |
| 05:44.22 | trnzmeta | guys how to check if sip connection is up? |
| 05:44.42 | km2 | trnzmeta, asterisk -rx 'sip show peers' |
| 05:45.17 | km2 | ManxPower, thanks, i had hoped so |
| 05:46.20 | *** join/#asterisk erojasv (n=erojasv@190.40.54.80) |
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| 06:00.59 | trnzmeta | guys: show sip peers says my voip provider status is OK |
| 06:01.39 | trnzmeta | when I make outgoing phone calls |
| 06:01.43 | trnzmeta | SIP/engin-c182 is circuit-busy |
| 06:03.17 | hardwire | that happens |
| 06:03.32 | hardwire | sip devices can return busy if their side is busy or something similar |
| 06:03.51 | trnzmeta | no consistently |
| 06:03.58 | trnzmeta | configuring a replacement asterisk box |
| 06:04.07 | hardwire | I didn't say it was an inconsistent only error :) |
| 06:04.21 | hardwire | we have some sip peers that we haven't paid in a while - they return congested... |
| 06:04.30 | hardwire | or if the sip call reaches a congested PSTN, it returns that code |
| 06:04.46 | trnzmeta | well it's sitting side by side the working asterisk box |
| 06:05.00 | hardwire | did you pay the other box? |
| 06:05.09 | trnzmeta | using same account details but different ISP gateways |
| 06:05.42 | hardwire | whats the "congested" side complaining about? logging? |
| 06:05.49 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:07.12 | trnzmeta | Everyone is busy/congested at this time (1:0/1/0) |
| 06:07.25 | trnzmeta | any other ways to show more in depth logs |
| 06:09.24 | trnzmeta | ok sip debug mode on |
| 06:12.07 | jeffspeff2 | hi all, i did some system software updates earlier, one of them was the kernel... i have since noticed that my conference rooms aren't working... i haven't changed a thing in the dialplans or files... here is the error i get when i try to go to a meetme exten... "build_conf: Unable to open pseudo device" |
| 06:14.50 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:23.40 | jeffspeff2 | trnzmeta, any ideas on the above? |
| 06:25.57 | trnzmeta | sorry, I'm out :( |
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| 07:07.15 | dexpdx | ok |
| 07:07.29 | dexpdx | what is the recommended method for doing procedure call backs to a database |
| 07:07.31 | dexpdx | func_odbc |
| 07:07.33 | dexpdx | agi? |
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| 07:10.05 | aiksa[LV] | I would rely on AGI |
| 07:10.18 | aiksa[LV] | far more controll over the whole process |
| 07:10.33 | Adrellias | does anybody know of reporting software for asterisk that shows unanswerd calls per extension etc |
| 07:10.35 | aiksa[LV] | ability to add excpetion handlers, retry clauses etc. |
| 07:11.35 | aiksa[LV] | Adrellias: I am not sure of such software, but adding this funcionality if you save CDRs to database should be trivial |
| 07:12.12 | Adrellias | hrmm |
| 07:12.13 | Adrellias | :P |
| 07:12.16 | Adrellias | was just thinking |
| 07:12.27 | Adrellias | if someone has already done it why should i redo it |
| 07:12.32 | aiksa[LV] | I imagine (if php for web reporting is your preference) it shouldnt take more than 50 lines of code to have fairly robuist systems for reporting this |
| 07:12.33 | Adrellias | the only one i can find is quemetrics |
| 07:12.38 | Adrellias | but the last release is 2007 |
| 07:12.50 | Adrellias | aiksa[LV], yeah true :P |
| 07:12.56 | Adrellias | o well then i shall code it rather |
| 07:13.15 | aiksa[LV] | 2007 is not that old |
| 07:13.17 | aiksa[LV] | :) |
| 07:13.47 | aiksa[LV] | I suppose you can find opensource projects which has not been maintained for a couple of years |
| 07:15.24 | Adrellias | :P |
| 07:15.29 | Adrellias | hehe yeah true |
| 07:15.38 | Adrellias | but my installation is delicate at the most |
| 07:15.42 | Adrellias | and to use quemetrics |
| 07:15.47 | Adrellias | i need to recompile apache |
| 07:15.51 | Adrellias | with tomcat |
| 07:15.55 | Adrellias | or at least replace apache |
| 07:16.02 | Adrellias | and put apache with tomcat on |
| 07:16.04 | Adrellias | but then again |
| 07:16.12 | Adrellias | i could always just change the port it listens on :P |
| 07:16.18 | aiksa[LV] | Adrellias: then you will be better of - making it your own |
| 07:16.34 | aiksa[LV] | at least I suppose it would be faster |
| 07:16.55 | aiksa[LV] | in terms of time to deploy |
| 07:17.29 | aiksa[LV] | And! - Busineses will ALWAYS like to have a modification of the reporting systems. |
| 07:18.15 | Adrellias | hehe true |
| 07:18.24 | aiksa[LV] | In a month or two time, you will have at least one strange request (a standart deviation of non-answered calls during specific hours of the day for example |
| 07:18.51 | aiksa[LV] | then - you will be better of having your own backyard piece of code which you can easily and fastly modify |
| 07:19.18 | Adrellias | :P |
| 07:19.27 | Adrellias | hehe pluss im pretty good at php |
| 07:19.28 | Adrellias | :D |
| 07:20.09 | aiksa[LV] | well - there you go |
| 07:20.14 | aiksa[LV] | issue solved |
| 07:20.21 | Adrellias | :P |
| 07:20.27 | Adrellias | yeah i was being lazy but ok \ |
| 07:24.52 | *** join/#asterisk peterpan-z (n=stelio@dsl-240-68-168.telkomadsl.co.za) |
| 07:25.02 | peterpan-z | anyone around? |
| 07:26.37 | peterpan-z | where do i change the default ringtime before asterisk picks up and IVR kicks in? |
| 07:34.16 | aiksa[LV] | i suppose you want something like this: http://www.voip-info.org/wiki-Asterisk+cmd+Ringing ? |
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| 07:39.24 | aiksa[LV] | peterpan-z: did it work for you? |
| 07:42.11 | peterpan-z | im going to try it |
| 07:44.11 | peterpan-z | aiksa[LV], basically when a call comes in from my local telco i want my asterisk box to pickup immediately and then play my IVR |
| 07:45.00 | Adrellias | aiksa[LV], do you know if the cdr db has a document somwhere |
| 07:45.04 | Adrellias | which explains last app etc |
| 07:45.13 | Adrellias | like i dont want to count calls as missed if its a group ring |
| 07:45.15 | Adrellias | etc |
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| 08:16.55 | frogonwheels | anybody have an opinion on whether an Alix 2-2 (www.yawarra.com.au) c 500mhz CPU would handle Asterisk with maybe a couple of channels a a time only?/ 3-4 way conferencing? |
| 08:17.08 | frogonwheels | (strictly home use) |
| 08:18.54 | frogonwheels | I have an asus router (200mhz I think) that handles 1 channel transcoding from gsm -> ulaw or 2 channels with no transcoding (80%) |
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| 09:01.29 | DarnoQ | anyone knows the estimated processor power needed for recording one call on 1.4.1 ? |
| 09:02.02 | DarnoQ | what processor would i need for recording 20 simultainous calls ? |
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| 09:08.46 | rolek | Can anyone tell me if it is possible to expire host entries in sip.conf? |
| 09:09.39 | rolek | My DNS changes seem not to be noticed by * |
| 09:09.49 | rolek | sip.conf: http://www.pastebin.org/36318 |
| 09:11.59 | frogonwheels | presumably dig on that machine noticed the changes.. and that sip reload didn't do it. |
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| 09:19.06 | rolek | frogonwheels: well, sip reload forces the change, but I would like to be able to change my dns without doing a sip reload on 40+ machines.. |
| 09:19.41 | frogonwheels | ah - the missing piece of the puzzle. |
| 09:20.25 | rolek | frogonwheels: The registration moves from one server to the other when I change the host name, but the peer remains associated with the old IP address.. |
| 09:21.08 | rolek | frogonwheels: s/change the host name/change the IP address for the host name/ |
| 09:21.57 | frogonwheels | is this something that happens regularly / a one off? |
| 09:22.54 | frogonwheels | i don't know enought about * to know whether it can handle it itself.. but some shell scripts might be able to do it... |
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| 09:26.10 | rolek | frogonwheels: Yeah, I could tie it together with some shell scripts, but my guess is that it would be proper if * would handle this by itself.. |
| 09:26.29 | rolek | frogonwheels: I wondered if I am missing some setting that would do this. |
| 09:27.00 | frogonwheels | shrugs. |
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| 09:35.18 | *** join/#asterisk sysadmin-lb22 (n=asdf@mail.splendor.net) |
| 09:35.20 | sysadmin-lb22 | Hi I think I need /usr/bin/safe_asterisk, I ran ./configure make make clean and make install on the asterisk source but I do NOT have that file I.e. "/usr/bin/safe_asterisk" anything I missed here ? |
| 09:43.38 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:46.02 | rolek | sysadmin-lb22: if you just ran ./configure you've probably installed asterisk in /usr/local |
| 09:47.02 | rolek | sysadmin-lb22: try running "make distclean" and start with "./configure --prefix=/usr" |
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| 09:51.21 | sysadmin-lb22 | rolek, thanks let me try |
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| 09:57.27 | sysadmin-lb22 | rolek, is make distclean something similar to uninstall ? |
| 10:01.32 | rolek | sysadmin-lb22: no make distclean will make sure that your sources are clean again, e.g. no previous ./configure settings remain |
| 10:03.04 | sysadmin-lb22 | rolek, thanks good to know that |
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| 10:51.05 | rvhi | hi |
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| 11:11.25 | marc7 | hey |
| 11:11.32 | Zyna | hey@all |
| 11:11.33 | marc7 | is there any way i could "cascade" calls? |
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| 11:12.07 | ctoep | hi |
| 11:12.08 | marc7 | so for example, John gets an incoming call... it rings desk 1 three times, then desks 1, 2, and 3 three times... then goes to voicemail. |
| 11:12.59 | marc7 | i have two Dial() statements at this point, one is Dial(SIP/desk1,10,r) and the other is Dial(SIP/desk1&SIP/desk2&SIP/desk3,10,r) |
| 11:13.23 | marc7 | the problem is that desk1's phone shows "2 missed calls" instead of one |
| 11:14.21 | ctoep | short question: ist it possible to connect 2 or more sip-phones to one number and if one phone is in use, set the other phones to busy? |
| 11:16.37 | marl | hi, im using asterisk 1.4.14 at present, it has been workingwithout any problems for quite some time now, but after a power outage a couple of weeks ago, i have noticed that the followme option no longer appears to accept pressing any digits to accept the call from the phone, eg. someone calls my number, * uses followme to dial my mobile, i press '1' to accept the call on my mobile and * compleates the call, this has worked fine upto the power outage, now howev |
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| 11:24.35 | Uatec | i don't understand this. My asterisk server is registering with my SIP proxy fine. but when my asterisk server tries to send an invite to the sip proxy, the proxy asks for authentication and asterisk just assumes that it's had Access Denied |
| 11:24.44 | Uatec | it reports "Failed to authenticate on invite" |
| 11:25.11 | Uatec | well that's true, it has, but it's supposed to then the credentials then, isn't it? |
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| 11:35.43 | Zyna | how much b/s do alaw and ulaw transfer approx? |
| 11:36.00 | Uatec | isn't it like 50kb/s or something? |
| 11:36.23 | Uatec | and if you use one of the compressions you can get down to like 5kb/s |
| 11:36.26 | Uatec | me thinks... |
| 11:37.17 | Zyna | k |
| 11:37.18 | Zyna | thx |
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| 11:37.41 | stephbul | According to me, it is 64kb/s |
| 11:37.53 | Uatec | same kinda order... |
| 11:38.14 | Uatec | 64kb/s being the bandwidth of an ISDN channel... |
| 11:38.17 | stephbul | A and µ law are G711 |
| 11:38.20 | Uatec | just enougy for one call |
| 11:38.42 | Uatec | *enough |
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| 11:41.22 | stephbul | Does anybody know a tutorial on how generate deb package of asterisk? |
| 11:41.44 | stephbul | I would like to do it for the 1.6beta8 version. |
| 11:43.20 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:43.20 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:45.24 | rob0 | I've never made .deb's, but the only thing asterisk-specific about it would be the options given to ./configure script. And possibly not even those. |
| 11:46.08 | rob0 | IOW, you're asking in the wrong place, it's a #debian question. |
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| 11:50.12 | stephbul | thx rob0 |
| 11:50.34 | geoff_k | i've got another possibly slightly off topic question, im new to asterisk, jsut readin around at reviews on softphones, i've read about kiax, linphone, x-lite couple of others, only thing i have used so far is zoiper. Now im looking at a huge list on voip-info.org and overwhelmed with choice, does anyone recommend any i should look at for free to use works linux with both AIX and SIP support preferably |
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| 11:50.55 | lmadsen | geoff_k: I'm a big fan of zoiper |
| 11:51.00 | tzanger | zoiper? |
| 11:51.12 | lmadsen | geoff_k: runner up is x-lite. Other than that, I've never used anything as good as those 2. |
| 11:51.18 | geoff_k | lmadsen, its been easy to use for me but i unpraded to latest and now i can only have 2 accounts it says |
| 11:51.20 | tzanger | ah softphone |
| 11:51.29 | tzanger | hmm it is cross platform |
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| 11:51.37 | lmadsen | geoff_k: yep, that is true -- x-lite only allows 1 account |
| 11:51.38 | tzanger | I shall have to try it, I use twinkle right now (KDE) |
| 11:51.47 | geoff_k | im not bothered about cross platform i just want it for me on linux |
| 11:52.03 | lmadsen | tzanger: ya... it's the phone by zoa, is SIP/IAX, does T.38 faxing, and is cross-platform |
| 11:52.03 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 11:52.06 | geoff_k | twinkle was one i just been reading a review on |
| 11:52.13 | tzanger | lmadsen: re-he-healy |
| 11:52.22 | lmadsen | tzanger: yeppers -- great phone |
| 11:52.25 | tzanger | I like twinkle but it's got its warts too |
| 11:52.25 | geoff_k | was jsut wondering if there are any in the list on the website i may have missed out too many to try them all |
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| 11:52.39 | lmadsen | tzanger: I never use a single phone because most of the time I'm going testing between phones :) |
| 11:52.47 | lmadsen | ekiga maybe |
| 11:52.48 | lmadsen | that's about it |
| 11:52.53 | geoff_k | its seems the ones i been readin bout so far are hte main ones |
| 11:53.06 | lmadsen | I'd say the only phones worth looking at are: zoiper, x-lite, ekiga, and twinkle (since apparently someone uses it :)) |
| 11:53.09 | tzanger | ok how do I downlod it |
| 11:53.17 | tzanger | the linux download link is cyclic |
| 11:53.23 | lmadsen | tzanger: really? |
| 11:53.25 | geoff_k | was hoping for one with AIX and SIP other than zoiper really |
| 11:53.26 | tzanger | ahh ther it is |
| 11:53.27 | tzanger | nevermind |
| 11:53.29 | lmadsen | heh |
| 11:53.31 | lmadsen | it's early |
| 11:53.35 | lmadsen | you're forgiven |
| 11:53.49 | lmadsen | geoff_k: can't get everything you want for free |
| 11:54.07 | geoff_k | lmadsen, yeah im prepared for that to be the case and i not a problem |
| 11:54.17 | rob0 | A SIP client for AIX? |
| 11:54.27 | lmadsen | he never said that |
| 11:54.28 | geoff_k | no a softphone supports both i want |
| 11:54.28 | *** join/#asterisk stony (n=oloch@p57B39B44.dip0.t-ipconnect.de) |
| 11:54.29 | stony | hi |
| 11:54.36 | rob0 | 11:53 < geoff_k> was hoping for one with AIX and SIP other than zoiper really |
| 11:54.40 | tzanger | ahh, binary only |
| 11:54.45 | geoff_k | if there are any other than zoiper for linux |
| 11:54.45 | cpm | no want not one both always |
| 11:55.07 | lmadsen | rob0: you reworded it in such a way that you were more wrong than him -- only thing wrong with the sentence was s/AIX/IAX |
| 11:55.21 | rob0 | 'twas a joke |
| 11:55.23 | cpm | wants to run iax on aix |
| 11:55.25 | stony | can anyone see an error here ? the var is always empty and the resultvars are filled (doing a NoOp before the if) "exten => s,n,set(gdtries=${IF($[${doorexists_res} == yes]?${gdmaxtries}:${gdtries})})" |
| 11:55.30 | lmadsen | I'm in no mood for your humour!!!!!!!!!! |
| 11:55.35 | geoff_k | yeah sorry i have a habbit of getting that wrong IAX |
| 11:55.39 | rob0 | Do we have coffee sprinklers in here? |
| 11:55.48 | cpm | yeah, rob0, stop it. |
| 11:55.58 | lmadsen | wonders when rob0 will know he is just bustin' his chops :) |
| 11:56.14 | rob0 | triggers the smoke alarm to test |
| 11:56.21 | lmadsen | I need some tea... |
| 11:56.25 | cpm | whispers to lmadsen, I'll get behind him and kneel down, and then you, , , |
| 11:56.41 | frogonwheels | anybody have an opinion on whether an Alix 2-2 (www.yawarra.com.au) c 500mhz CPU would handle Asterisk with maybe a couple of channels a a time only?/ 3-4 way conferencing? |
| 11:56.47 | lmadsen | pushes cpm over tzanger |
| 11:57.08 | lmadsen | frogonwheels: 500MHz should be able to do like 6 channels without transcoding at least |
| 11:57.12 | lmadsen | maybe even 10-15 |
| 11:57.22 | lmadsen | depending on how the box is designed |
| 11:57.33 | frogonwheels | lmadsen: so should be able to support a few channels in a conference then? |
| 11:57.48 | *** join/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 11:57.59 | rob0 | feels guilty for eating frog legs yesterday ... now we have frogonwheels |
| 11:58.02 | *** part/#asterisk Shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 11:58.22 | frogonwheels | kicks rob0 |
| 11:58.33 | rob0 | but you HAVE NO LEGS |
| 11:58.45 | frogonwheels | wheels rob0 harder ;) |
| 11:59.03 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:59.04 | tzanger | so this is zoa's phone... looks good, hopefully it's able to hit dbus and stuff too |
| 11:59.26 | coppice | rob0: its wasteful to only eat their legs. we eat the whole frog |
| 11:59.37 | coppice | well, except for the bones |
| 12:00.12 | lmadsen | tzanger: do you know a good way I can tell asterisk to log PRI debug and SIP debug to a file? I have the full debug going on... but I think I want to see the connection stuff too |
| 12:00.24 | lmadsen | tzanger: getting random dropped calls, and trying to figure out how and why |
| 12:00.33 | lmadsen | hates random issues that can't be reproduced |
| 12:00.46 | yang | says hi to lmadsen |
| 12:00.58 | lmadsen | says hi back to yang |
| 12:01.17 | lmadsen | frogonwheels: I've never used it... but I would think so |
| 12:01.29 | tzanger | lmadsen: just look at the full debug log then, that should do it all. IIRC there was also a way to say "pri debug filename" to log just the PRI stuff |
| 12:01.34 | lmadsen | frogonwheels: your requirements sound pretty modest, so I think you should be good |
| 12:01.57 | lmadsen | tzanger: ya, I enabled full -- but I think that just gives you debug,verbose,notice,warning,error |
| 12:02.06 | lmadsen | I will see about the command you just gave me though |
| 12:02.24 | rob0 | lmadsen could have the * console in screen(1), log everything to a file, then grep or otherwise separate/parse the contents of the file. |
| 12:02.27 | lmadsen | I guess I was hoping for everything to be logged to a single file so I could go back through it when they call me to tell me a call was dropped |
| 12:02.41 | frogonwheels | lmadsen: ta. I've got it working on a err around 200mhz router at the moment. |
| 12:02.43 | lmadsen | ya... I guess I would just tee the output of the CLI to a file... |
| 12:02.49 | tzanger | lmadsen: yes, full debug to file, then issue pri debug ... |
| 12:02.51 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 12:03.00 | lmadsen | oh right... because that'll be in VERBOSE right? |
| 12:03.03 | tzanger | I don't have a * system with pri at the moment |
| 12:03.06 | tzanger | lmadsen: correct |
| 12:03.08 | frogonwheels | lmadsen: so I've got all the settings for low to no transcoding. |
| 12:03.09 | lmadsen | that makes sense |
| 12:03.28 | lmadsen | frogonwheels: ya -- you're going to want to make sure you're not doing transcoding because that'll eat a lot of CPU |
| 12:03.57 | frogonwheels | lmadsen: it handles one transcoding - but another channel kills it... |
| 12:04.14 | frogonwheels | lmadsen: and forget about a conference... |
| 12:04.31 | lmadsen | ya... conference should be ok with no transcoding for 3-4 channels |
| 12:04.35 | frogonwheels | - but I need to try that again now that I have my QoS set up. |
| 12:04.56 | frogonwheels | seems it was buming out on the outgoing packets. |
| 12:05.12 | frogonwheels | lmadsen: assume that multiplexing the channels together is what would kill it. |
| 12:05.48 | lmadsen | frogonwheels: ya.. that'll certainly be more intense than just bridging calls together... but a single conference with 3-4 participants should be ok... |
| 12:05.58 | lmadsen | tzanger: thanks -- that seems to be working |
| 12:06.21 | tzanger | lmadsen: always glad to help |
| 12:06.32 | lmadsen | merci garcon :) |
| 12:06.52 | frogonwheels | lmadsen: interesting. |
| 12:07.15 | frogonwheels | possibly there are bigger problems on the router than just cpu speed - like swap-space on a usb stick :) |
| 12:07.46 | frogonwheels | - andhandling wifi encoding at the sme time. |
| 12:08.38 | lmadsen | frogonwheels: oh ya... if you're not *just* doing asterisk on that box... you're certainly going to have diminished capacity |
| 12:09.11 | frogonwheels | lmadsen: yeah - it's my wireless router - so it's doing pretty well to be doing what it's doing. |
| 12:09.18 | frogonwheels | if that makes sense :) |
| 12:09.44 | masus | exit |
| 12:10.17 | rob0 | exit |
| 12:10.18 | frogonwheels | lmadsen: I might muck about with moving the conferencing onto another computer... |
| 12:10.24 | rob0 | it didn't work for me |
| 12:10.41 | mwalling | rob0fail |
| 12:10.55 | cpm | chuckles |
| 12:11.12 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:11.15 | frogonwheels | rob0: I'm sure somebody with Op priv. could arrange to make it LOOK like it worked. |
| 12:11.54 | lmadsen | chuckles |
| 12:14.12 | frogonwheels | hey is there somewhere I could upload a fun little script I've got that takes email via Xmail filter and unwraps/sox's and places it into asterisk mailbox? |
| 12:16.50 | [TK]D-Fender | frogonwheels: WIKI is a nice bet |
| 12:18.01 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:18.15 | *** join/#asterisk MaliutaLap (n=nikolai@kiev.lusan.id.au) |
| 12:22.35 | *** join/#asterisk nighty^ (n=nighty@y096122.dynamic.ppp.asahi-net.or.jp) |
| 12:27.02 | frogonwheels | [TK]D-Fender: which wiki would we want to write on? |
| 12:27.23 | frogonwheels | [TK]D-Fender: i mean - which one? |
| 12:27.34 | [TK]D-Fender | ~wikis |
| 12:27.37 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 12:28.03 | *** join/#asterisk asdx (n=diego@adsl-128-242.click.com.py) |
| 12:30.15 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 12:32.36 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 12:32.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:38.07 | *** part/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 12:38.50 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 12:41.01 | Uatec | so, i've got my asterisk box which is inviting to my SIP proxy, my proxy is returning 401 Unauthorised, asking for credentials, but asterisk isn't sending on the credentials. |
| 12:41.11 | Uatec | it's working using the same credentials to register, but not to invite |
| 12:41.15 | Uatec | i don't understand |
| 12:41.24 | Uatec | can anybody help? |
| 12:44.25 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:44.32 | jks | anyone else experiencing problems with app_queue in combination with realtime? |
| 12:45.30 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:49.11 | shido6 | whats wrong jks? |
| 12:49.45 | jks | shido6, the problem I'm experiencing is that app_queue somehow stop being able to ring a specific phone |
| 12:50.00 | jks | shido6, i.e. the phone is not in use and there's people waiting in the queue, but it's not ringing |
| 12:50.16 | shido6 | is asterisk still connected to your dsn? |
| 12:50.19 | jks | shido6, the system will work fine for days, and then suddenly this happens... |
| 12:50.31 | jks | shido6, sorry, dsn is what? |
| 12:50.49 | jks | shido6, everything not related to the queue works fine (i.e. I can Dial() the phone without problems) |
| 12:51.09 | jks | shido6, if I remove the queue from the database and add it again (the exact same configuration), things starts working again |
| 12:52.07 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:52.15 | jks | shido6, it's quite clear in the queue log... normaly when it's trying to ring a phone, it tries over and over to determine when the phone is no longer busy and is ready to receive a call |
| 12:52.56 | jks | shido6, but when this problem happens, it seems to not be doing this at all, or only be doing it every queue timeout (for me, I have set it to 120 seconds) |
| 12:55.03 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 12:55.42 | *** join/#asterisk Penol- (n=Penol@195.204.248.62) |
| 12:56.15 | Penol- | How to setup the asterisk to play music when its waiting for an ext? |
| 12:56.53 | jks | shido6, I don't know if this gives you any ideas? |
| 12:57.20 | jks | shido6, I've found some mentions of similar behaviour on the bugtracker, which indicated that added call limits would do the trick... I have added 9999 call limits, but still experiencing the problem |
| 12:57.39 | [TK]D-Fender | Penol-: waiting where? |
| 12:58.22 | Penol- | I want my asterisk to play music when its waiting for someone to ask |
| 12:58.55 | Penol- | Something like MusicOnHold but that dont call the SIP's |
| 12:59.02 | Penol- | when its waiting |
| 12:59.45 | *** part/#asterisk asdx (n=diego@adsl-128-242.click.com.py) |
| 13:00.46 | [TK]D-Fender | Penol-: Only way is to Background it. |
| 13:00.59 | Penol- | Ok |
| 13:01.45 | Uatec | when inviting to my SIP provider, they give me 407 proxy authentication required, asterisk replies with ACK then trys to invite again but with the credentials. |
| 13:02.09 | Penol- | But when i put backround on it the music dont start and the asterisk hungs up |
| 13:02.14 | Uatec | however when doing the same with my SIP proxy, asterisk gets 407 and replies ACK, but then doesn't try to invite again with the credentials |
| 13:02.34 | Uatec | as such, i can't make any calls to my SIP proxy becuase asterisk just doesn't try to authenticate |
| 13:02.46 | Uatec | does anybody know why this might be? |
| 13:03.10 | [TK]D-Fender | Penol-: PASTEBIN some backup to your problem so we cans ee why it failed. |
| 13:03.24 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 13:03.51 | Penol- | [TK]D-Fender: http://pastebin.no/5846 |
| 13:04.51 | [TK]D-Fender | Penol-: You can't dial in the middle of your exten like that |
| 13:05.05 | [TK]D-Fender | Penol-: * won't be waiting for input. |
| 13:05.24 | Penol- | how to fix? |
| 13:05.35 | [TK]D-Fender | Penol-: and please show the CLI output of your failed call at verbose 10. |
| 13:05.46 | Penol- | ok |
| 13:05.53 | [TK]D-Fender | Penol-: use Waitexten, set your timeouts, etc |
| 13:06.15 | [TK]D-Fender | Penol-: And you have 3 priority "3"'s in there. BAD. |
| 13:06.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.47 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583729.dsl.bell.ca) |
| 13:09.41 | marl | is it posible to install a second copy of * on a machine and change from one version to the other? (the machine has NO telephony hardware) am currently running 1.4.14 and want to try my configs on 1.4.20 or 1.6 before i commit to the new version? a pointer to a howto or something would be great :) |
| 13:10.34 | rob0 | change all relevant paths, or merely run it in a chroot |
| 13:12.11 | [TK]D-Fender | marl: What kind of telephony hardware? |
| 13:12.21 | marl | there is no hardware |
| 13:12.21 | [TK]D-Fender | nvm |
| 13:12.25 | [TK]D-Fender | blind thi moring |
| 13:12.39 | [TK]D-Fender | marl: Yeah, as rob0 said |
| 13:12.40 | marl | am using this purly for IAX |
| 13:13.00 | rob0 | might also have to bind a different port/IP address |
| 13:13.56 | marl | was going to shutdown the original * and then start the new one, so i can confirm all the ext/iax settings work |
| 13:14.03 | marl | so only running one at a time |
| 13:14.26 | rob0 | oh okay |
| 13:14.32 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:15.57 | Katty | good morninable asteriskables! |
| 13:16.20 | shido6 | err |
| 13:16.25 | shido6 | jks, you still there? |
| 13:16.34 | shido6 | instead of readding the info |
| 13:16.35 | jks | shido6, yep? |
| 13:16.39 | Katty | hugs _ShrikE |
| 13:16.39 | shido6 | what happens when u do a reload |
| 13:16.43 | Katty | hugs [TK]D-Fender |
| 13:16.46 | jks | shido6, module reload, you mean? |
| 13:17.00 | shido6 | reloading just the db pieces |
| 13:17.02 | Uatec | waves to Katty |
| 13:17.05 | _ShrikE | hugs Katty! |
| 13:17.08 | Katty | eyes Uatec |
| 13:17.08 | [TK]D-Fender | Katty: Mew. |
| 13:17.08 | shido6 | yes module reload |
| 13:17.12 | Katty | Uatec: allo person! |
| 13:17.14 | jks | shido6, nothing much |
| 13:17.21 | jks | shido6, it reloads ofcourse, but the problem persists |
| 13:17.23 | Katty | [TK]D-Fender: how are mew |
| 13:17.24 | shido6 | which modules do you reload? |
| 13:17.37 | jks | shido6, I've tried "module reload" (verbatim) |
| 13:17.41 | shido6 | whats your extconfig look like? |
| 13:17.50 | shido6 | which modules do you reload? |
| 13:18.00 | jks | shido6, doesn't it reload all the modules? |
| 13:18.38 | jks | shido6, what do you need to know about the extconfig file? - it only contains statements for sipusers, sippeers, voicemail, queues and queue_members |
| 13:19.00 | frogonwheels | Ok - added http://www.voip-info.org/wiki/view/Xmail+filter+to+Voicemail+script |
| 13:19.27 | [TK]D-Fender | Katty: I feel broken today. Arm is messed up, I have no energy today. Feel like *bleh* |
| 13:19.32 | shido6 | you said you were using realtime. This requires you also add either mysql or odbc on the line you're using such as your "queues" or "queue_members" |
| 13:19.38 | Katty | [TK]D-Fender: :< |
| 13:19.45 | jks | shido6, I am using realtime |
| 13:19.49 | Katty | [TK]D-Fender: what happened to your arm? |
| 13:19.54 | Katty | [TK]D-Fender: swordy related stuff? |
| 13:20.02 | jks | shido6, for example: "queues => mysql,mydatabase,asterisk_phone_queues" |
| 13:20.18 | shido6 | mysql |
| 13:20.20 | shido6 | or |
| 13:20.21 | [TK]D-Fender | Katty: Dunno. My left forearm is just locked rotationally. |
| 13:20.27 | jks | shido6, I'm using mysql |
| 13:20.30 | shido6 | are you using mysam or innodb in the mysql db ? |
| 13:20.50 | jks | shido6, I'll check... why is that important in this case? (I'm just wondering) |
| 13:21.00 | shido6 | Im wondering if a table is being locked |
| 13:21.06 | shido6 | instead of just a row |
| 13:21.07 | jks | shido6, it's not |
| 13:21.19 | jks | shido6, we can do changes to the table easily |
| 13:21.43 | jks | shido6, and all the other queues in the systems are working fine |
| 13:21.52 | shido6 | can you explain the readding of info process to me |
| 13:21.59 | jks | shido6, info process? |
| 13:22.03 | shido6 | when u remove it and readd this info it works flawlessly, right? |
| 13:22.21 | jks | shido6, I'm not 100% sure, but it has worked in one instance |
| 13:22.48 | jks | shido6, as this happens only once every few days, it takes a bit of time to get any real hard facts on what works and what doesn't |
| 13:23.00 | jks | shido6, it was an idea I got yesterday, and there it worked and fixed the problem |
| 13:23.06 | shido6 | make sure debug is turned on and bin log is on on the mysql server....... and do it again |
| 13:23.18 | shido6 | hrmmmm |
| 13:23.19 | jks | shido6, there's no problems with the mysql server |
| 13:23.23 | jks | shido6, the problem is in app_queue |
| 13:23.25 | *** join/#asterisk lupino3 (n=andrea@217-133-45-108.b2b.tiscali.it) |
| 13:23.37 | shido6 | do u have that crappy cache crap turned on in sip.conf.. |
| 13:23.42 | jks | shido6, the problem is with app_queue kind of deciding that the phone in question is in a different state than it really is |
| 13:24.00 | jks | shido6, rtcachefriends you mean? |
| 13:24.08 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-210-156.nycmny.east.verizon.net) |
| 13:24.08 | shido6 | oui |
| 13:24.15 | lupino3 | hello everybody; I have a strange problem with the Flash Operator Panel: some events are not recognized until I manually reload the page containing the flash .swf.. Anybody can help me? |
| 13:24.17 | jks | shido6, yep, it's on |
| 13:32.21 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:36.58 | *** join/#asterisk BrokenNoze (n=Simon_Kr@212.183.134.131) |
| 13:37.40 | BrokenNoze | Hi, does anyone have any experience using ringinuse? |
| 13:38.03 | *** join/#asterisk shinao1 (n=shinao1@41.219.208.136) |
| 13:40.16 | Dossy | Any reason why inbound SIP call won't recognize DTMF? Ugh. |
| 13:41.20 | Uatec | there's a configuration somewhere that specifies how long a tone has to be to be recognised |
| 13:41.27 | Uatec | i suspect you need to shorten that |
| 13:41.34 | Dossy | Uatec: ah, interesting. |
| 13:41.42 | Uatec | from sip phones, and ISDN phones and stuff DTMF was fine for me |
| 13:41.54 | Uatec | but callers originating on analogue phone could not use it |
| 13:42.14 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:42.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:42.19 | Dossy | that's exactly the problem - LOL - I'm actually dialing in to my VoIP DID from a cellphone |
| 13:42.33 | BrokenNoze | i've set ringinuse=yes but the sip extensions still only ever allow one call into them, why is that?? |
| 13:42.50 | Dossy | I have a different problem testing from my HandyTone-386 ATA - can send audio, but can't recv it |
| 13:42.53 | [TK]D-Fender | Dossy: go fix the DTMFMODE for your inbound peer in sip.conf |
| 13:43.03 | [TK]D-Fender | Dossy: for your 1-way audio : |
| 13:43.05 | [TK]D-Fender | ~sipnat |
| 13:43.06 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:43.07 | [TK]D-Fender | ^^^^^^^^^ |
| 13:43.16 | Dossy | [TK]D-Fender: help me "fix" it - I've tried "inband", I've tried "rfc2833", I've tried "info" and "auto" - none work. |
| 13:43.27 | Dossy | Is there another, undocumented, setting I should try? |
| 13:43.48 | [TK]D-Fender | Dossy: pastebin the incoming call at verbose 10, sip debug enabled, and include your sip.conf masking only passwords |
| 13:43.51 | Dossy | The SIP isn't behind NAT - at least, not with respect to the ATA and Asterisk. |
| 13:44.13 | Dossy | [TK]D-Fender: thanks. let me do that now |
| 13:44.14 | [TK]D-Fender | Dossy: include a failed call for the ATA as well and its config |
| 13:44.48 | Dossy | [TK]D-Fender: does "relaxdtmf" need to be yes or now? |
| 13:44.49 | Dossy | er, no. |
| 13:45.13 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 13:45.57 | [TK]D-Fender | Dossy: last I checked that option is only valid in zapata.conf |
| 13:46.20 | *** join/#asterisk b1shop (n=b1shop@c-76-16-229-8.hsd1.il.comcast.net) |
| 13:46.23 | Dossy | it's included in the example sip.conf - shrug |
| 13:47.14 | [TK]D-Fender | Dossy: could be valid... if you get SOME dtmf but not all from that entry then it matters. And this still only applies to inband |
| 13:47.45 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 13:49.38 | eric2 | I have a DID set to ring a sip phone and a cell phone at the same time, it does this but the caller only hears one ring until the called party picks up, any way to have the caller hear ringing as normal? |
| 13:51.10 | BrokenNoze | anyone used ringinuse? know why it might not work? |
| 13:52.32 | BrokenNoze | it doesn't want to work with dynamic agents? |
| 13:52.41 | plik | eric2: put an 'r' at the end of your dial string |
| 13:52.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:54.31 | eric2 | like this: exten => _4164223742,1,Dial(SIP/4164223742&SIP/4163265248@vpri_gw2,r) ?? |
| 13:55.24 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 13:55.29 | [TK]D-Fender | eric2: exten => 4164223742,1,Dial(SIP/4164223742&SIP/4163265248@vpri_gw2,,r) |
| 13:56.15 | plik | yeah, what [TK]D-Fender said |
| 13:56.41 | eric2 | oh that's the sh*t!! thanks |
| 13:57.19 | *** join/#asterisk ManxPower (n=manxpowe@57.sub-70-220-93.myvzw.com) |
| 13:58.05 | peterpan-z | if i do zap show channels.. how do these correspond to my trunks? |
| 14:00.53 | [TK]D-Fender | peterpan-z: what "trunks"? |
| 14:02.51 | peterpan-z | ZAP trunks |
| 14:02.56 | lmadsen | peterpan-z: depends how you have them configured |
| 14:03.09 | lmadsen | in N.A. usually 1 span = 23B+1D |
| 14:03.35 | coppice | and not a single A or C in sight |
| 14:03.36 | peterpan-z | i have channel 1 from zaptel |
| 14:03.40 | lmadsen | not at all |
| 14:03.50 | lmadsen | goes back to real work.. |
| 14:04.05 | [TK]D-Fender | peterpan-z: So what are you trying to do exactly? |
| 14:04.06 | ManxPower | peterpan-z: we don't usually call them trunks here. "zap show channels" shows your CONFIGURED zap channels. |
| 14:04.32 | peterpan-z | add a ZAP trunk |
| 14:04.57 | Dossy | Hey, is there an easy way to redirect the console to a file? |
| 14:05.06 | peterpan-z | for the Zap identifier, i have used the trunk name g1 |
| 14:05.12 | ManxPower | peterpan-z: You're one of those people with a fetish for the word "trunk", aren't you? |
| 14:05.12 | Dossy | [TK]D-Fender: trying to get you the verbose/sip debug log - but cut and paste from console is getting truncated :-( |
| 14:05.28 | [TK]D-Fender | Dossy: get a better client |
| 14:05.28 | ManxPower | peterpan-z: no, you use a GROUP. |
| 14:05.44 | lmadsen | Dossy: see logger.conf, or: asterisk -rvvv | tee /tmp/output.txt |
| 14:05.53 | [TK]D-Fender | NO LOGGER |
| 14:05.57 | [TK]D-Fender | bleh |
| 14:06.13 | Dossy | lmadsen: ah, tee? |
| 14:06.20 | lmadsen | messages => verbose,warning,error,notice |
| 14:06.24 | Dossy | [TK]D-Fender: yeah, i'm using asterisk -r |
| 14:06.30 | ManxPower | Dossy: tee is a standard unix tool. |
| 14:06.40 | Dossy | yes, I know about tee :) |
| 14:06.41 | [TK]D-Fender | Dossy: "thats nice". Now access that via a better client. |
| 14:06.41 | lmadsen | handy as all hell |
| 14:06.59 | Dossy | [TK]D-Fender: is there a better CLI non-GUI client you can recommend? |
| 14:07.08 | [TK]D-Fender | Dossy: PuTTY |
| 14:07.13 | ManxPower | Dossy: I use PuTTY |
| 14:07.22 | [TK]D-Fender | high-5's ManxPower |
| 14:07.23 | lmadsen | I use terminal |
| 14:07.30 | Dossy | I'm using SecureCRT - I guess I can increase scrollback lines. |
| 14:07.34 | glaz | I use xterm |
| 14:07.38 | glaz | Dossy: yes you can. |
| 14:07.40 | [TK]D-Fender | lmadsen: Never use a client that sounds like its going to die ;) |
| 14:07.47 | lmadsen | [TK]D-Fender: I live on the edge |
| 14:08.04 | glaz | I like nutty also, you can click the hyperlinks. |
| 14:11.32 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 14:12.12 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:12.34 | Dossy | OK, here's the verbose/SIP debug - http://pastebin.com/d68cce333 |
| 14:12.41 | Dossy | I'm pasting up the sip.conf now |
| 14:13.04 | shimi | Hi all. I'm having in issue. I am creating a call file to call me and throw me into DISA. When I throw the file into outgoing/ - asterisk initiates the call. When I pick up, I get dialtone. When I dial the numbers, I see in the full log that DTMF digits are pressed. However, after some seconds, I see in the log: app_disa.c: DISA extension entry timeout on chan Zap/1-1 and asterisk hangs up the line. What am I doing wrong? |
| 14:14.24 | phix | hi |
| 14:14.27 | Zeeek | hi |
| 14:14.41 | phix | sup? |
| 14:14.58 | phix | how's your PBX coming along/ |
| 14:15.40 | Zeeek | shimi you should put the various files in the pastebin |
| 14:16.32 | phix | yeah |
| 14:16.45 | Zeeek | yeah! |
| 14:16.57 | shimi | Zeeek, http://pastebin.com/d51c7c2c5 < that's the call file |
| 14:17.05 | BrokenNoze | Hi All, in a call queue, if I add a SIP/ channel multiple calls will ring on the channel, if however I add an AGENT/ channel it only delivers 1 call to the phone, why is this? anyone any ideas? |
| 14:17.21 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 14:17.37 | putnopvut | BrokenNoze: It has to do with the reported device state of the phone. |
| 14:17.54 | phix | BrokenNoze: sup |
| 14:17.57 | jduggan | hey guys, if i do restart now in the cli, it'll disconnect me but it doesnt seem to fully restart (you can asterisk -r again), i cant make calls untill i restart it using my init scripts.. is there anything i can look at as to why this happens? |
| 14:17.58 | putnopvut | BrokenNoze: With SIP devices, they will always report "Not in Use" unless you have specified a call-limit |
| 14:18.23 | putnopvut | BrokenNoze: If you specify a call limit for a SIP device, then when it makes or receives a call, it will change to "In use" |
| 14:18.57 | putnopvut | BrokenNoze: and in queues.conf, there is an option called "ringinuse." If you set this to "no" and put a call-limit on the SIP queue member, then that member will only be able to receive one call at a time. |
| 14:19.08 | BrokenNoze | putnopvut: OK well the SIP channels are actually doing what I want the AGENT channels to do, how do I set the agent channel to return Not In Use? |
| 14:19.19 | Dossy | here's the sip.conf - http://pastebin.com/d68bfb07f |
| 14:19.24 | putnopvut | BrokenNoze: oh, heh. I misunderstood. |
| 14:19.32 | BrokenNoze | no, your spot on |
| 14:19.35 | putnopvut | BrokenNoze: and that's a very good question too :) |
| 14:20.06 | BrokenNoze | putnopvut: so... no answer for me :-( damn it! |
| 14:20.18 | putnopvut | BrokenNoze: Yeah, sorry. :( |
| 14:20.40 | BrokenNoze | anyone? deliver multiple calls to a single agent at the same time? |
| 14:20.55 | putnopvut | BrokenNoze: I'm going to try a little experiment for that. Hold tight. |
| 14:21.24 | BrokenNoze | cheers! I'm dying here, stuck on a customer site with them breathing down my neck! |
| 14:21.24 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 14:21.45 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 14:22.59 | shimi | are you trying to implement queues on agents ? |
| 14:23.26 | putnopvut | BrokenNoze: Okay, so when an agent picks up a call, his state moves to "Busy" and there's no way to tell a queue to dial a busy member. |
| 14:23.43 | putnopvut | I think that short of source code modification, there would be no way to change that for an agent channel. |
| 14:23.46 | *** join/#asterisk aksyn (n=aksyn@nat76.mia.three.co.uk) |
| 14:24.13 | BrokenNoze | Ah, i don't mind the pickup, but the agent seems to be busy even if he's ringing? |
| 14:24.21 | BrokenNoze | so only one call is delivered |
| 14:24.32 | shimi | I think that if he wants all the agents to get called every time a call comes to the queue - just use a simple queue with a ringall strategy |
| 14:24.43 | shimi | and enable callwaiting on relevant extensions |
| 14:24.45 | shimi | and that's it |
| 14:25.01 | BrokenNoze | can i write a Macro and place it in my Dial( M(setagenttonotbusy)) state? |
| 14:25.50 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 14:25.53 | BrokenNoze | I need my agents to see all the calls so they can select the most important ones? |
| 14:26.22 | shimi | and the queue is a bad method because? |
| 14:27.03 | BrokenNoze | Well I could use groups, can i add members to groups dynamically? |
| 14:27.17 | *** join/#asterisk dFence (n=chatzill@p54980C1F.dip0.t-ipconnect.de) |
| 14:27.24 | shimi | anything can be done with enough programming :) |
| 14:27.25 | [TK]D-Fender | Dossy: You need to prevent reinvites on all of your devices |
| 14:28.22 | shimi | the important question is what are you implementing |
| 14:28.59 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 14:29.00 | shimi | for example, in the front desk in my company, i just make all the phones in a queue, and any of them can be picked, whether one of the secretaries is talking on the phone now or not |
| 14:29.11 | BrokenNoze | Shimi: I simply need a call queuing system where an agent can login to multiple queues but see on his handset calls coming in from all of those queues |
| 14:29.14 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 14:29.27 | masus | **UDP Flood to Host** does anyone know what this meaning ? |
| 14:29.28 | ManxPower | I hate queues. Fortunately, we don't need anything more than the functionality of chan_local and some clever dialplan stuff to fake queues. |
| 14:29.40 | shimi | and if they're not there, well, the phone will just ring (until the timeout I've set then it returns to main IVR) |
| 14:29.51 | masus | i cant hear the other client and have see in the modem log **UDP Flood to Host** |
| 14:29.59 | masus | can anyone help ? |
| 14:30.22 | Dossy | [TK]D-Fender: thanks - I'll try that |
| 14:30.27 | shimi | masus: It means that your modem is trying to be smart. VoIP sends a constant stream of small UDP packets. you need to disable this "protection" |
| 14:30.37 | Dossy | Is that a general "I should do this" or is this why I'm having the one-way audio problem? |
| 14:30.49 | ManxPower | [root@bourbon asterisk]# grep -ri "UDP Flood to Host" * |
| 14:30.49 | ManxPower | grep: asterisk-1.4.13/configs/.#extensions.conf.sample: No such file or directory |
| 14:30.49 | ManxPower | [root@bourbon asterisk]# |
| 14:31.03 | ManxPower | masus: Your problem is not an Asterisk issue. |
| 14:31.23 | s0ck | does anyone |
| 14:31.25 | s0ck | anywhere |
| 14:31.32 | s0ck | have an AEX800 card that actually works? |
| 14:31.57 | masus | ManxPower: but i cant hear the other sip user ... |
| 14:32.06 | s0ck | could it be topic'd by any chance... |
| 14:32.15 | masus | shimi: where to disable it ? |
| 14:32.16 | Zeeek | too much bourbon |
| 14:32.28 | s0ck | Digium ssh'd into my pbx and couldn't get it working either |
| 14:32.31 | s0ck | at a bit of a loss now. |
| 14:32.32 | ManxPower | masus: Once you fix the non-asterisk issue, you can start working on any remaining Asterisk issues. |
| 14:32.41 | ManxPower | s0ck: It's a digium card. Digium cards come with support. |
| 14:32.43 | shimi | masus, your modem web interface ? |
| 14:32.49 | s0ck | see above. |
| 14:33.37 | masus | ok |
| 14:33.37 | ManxPower | s0ck: then return the card |
| 14:33.37 | s0ck | i have |
| 14:33.37 | s0ck | the new card is sitting in the box |
| 14:33.37 | s0ck | same problem |
| 14:33.37 | ManxPower | and you contacted Digium again? |
| 14:33.37 | shimi | maybe the problem is in the box |
| 14:33.38 | s0ck | i just did |
| 14:33.38 | s0ck | thought i'd come here and whinge in the meantime |
| 14:33.38 | Zeeek | shimi :) |
| 14:33.38 | ManxPower | Perhaps the card is simply not compatible with the system. |
| 14:34.06 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 14:34.15 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 14:34.17 | s0ck | that would be a real pain as i have only lower spec boxes to move it onto |
| 14:34.20 | ManxPower | If you can't make it work, even with Digium's help, you should consider Sangoma |
| 14:34.20 | s0ck | nevertheless |
| 14:34.22 | s0ck | i think i best go try |
| 14:34.32 | phix | wooties |
| 14:34.44 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 14:35.10 | pingwin | hello, i am assisting a buddy who can't get online. He has an asterisk server using a PRI and needs to forward all of his calls to a new number outside of our area code. The problem is our provider requires a code to be entered when dialing long distance numbers ...any ideas on how to make the dial plan do this automatically? |
| 14:35.19 | pingwin | sorry for the long message ... and thank you :) |
| 14:35.40 | *** join/#asterisk atomspace (n=huh@adsl-220-196-117.bhm.bellsouth.net) |
| 14:35.49 | ManxPower | pingwin: Look at the D() option of Dial. That option requires the far end to send and Answer before sendin the DTMF. |
| 14:35.51 | b11d` | just enter the code before the number being dialed.. i'd think |
| 14:35.56 | b11d` | cool |
| 14:35.57 | fiddur | Hi. Does anyone know how to get app_jabber to work with asterisk 1.4? I get "Module 'app_jabber.so' did not register itself during load" after a "load app_jabber.so"... |
| 14:36.11 | pingwin | ManxPower: i'll look at that, thank you |
| 14:36.34 | atomspace | are there threads for certain ip phones? I have a grandstream gxp-2000 and am looking for some input |
| 14:36.41 | ManxPower | ~gs |
| 14:36.41 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:36.57 | b11d` | aye |
| 14:37.01 | atomspace | well, opinion. |
| 14:37.06 | [TK]D-Fender | atomspace: http://voxilla.com/ <- go read the forums |
| 14:37.07 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 14:37.07 | atomspace | I've had no problems with ours |
| 14:37.15 | atomspace | thanks fender |
| 14:37.42 | phix | 1~2 |
| 14:37.43 | shimi | Hi all. I'm having in issue. I am creating a call file to call me and throw me into DISA. When I throw the file into outgoing/ - asterisk initiates the call. When I pick up, I get dialtone. When I dial the numbers, I see in the full log that DTMF digits are pressed. However, after some seconds, I see in the log: app_disa.c: DISA extension entry timeout on chan Zap/1-1 and asterisk hangs up the line. What am I doing wrong? This is the call file: http:/ |
| 14:37.43 | shimi | /pastebin.com/d51c7c2c5 |
| 14:37.45 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 14:38.32 | ManxPower | shimi: you have STMF problems |
| 14:38.35 | ManxPower | and DTMF too |
| 14:38.42 | phix | DTMF is great |
| 14:38.53 | shimi | on two different phones? |
| 14:39.02 | rob0 | s0ck, some specifics about the system? CPU? Motherboard? Maybe it doesn't support a recent enough PCI spec. I had that problem with an el cheapo ECS board with onboard VIA C3 CPU. |
| 14:39.13 | phix | I like it how it doesn;t work in a compresed audio stream, it is great |
| 14:39.17 | rob0 | (was a TDM400 however) |
| 14:39.20 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 14:39.20 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:39.23 | ManxPower | shimi: the message you are getting indicates DISA did not hear any DTMF. You draw your own conclusions. |
| 14:39.23 | phix | I has a TDM400 |
| 14:39.24 | shimi | (if so, why does asterisk log received the number completely correct?) |
| 14:39.27 | phix | I need another one |
| 14:39.31 | phix | thy are great too |
| 14:40.11 | jm|home | anyone use sipgate? |
| 14:40.17 | shimi | maybe something needs to be done so DISA can "hear" the DTMF that asterisk DOES SAY that it received ? |
| 14:40.27 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:40.36 | [TK]D-Fender | shimi: pastebin that complete CLI output of your attempt with full debug. |
| 14:40.42 | ManxPower | shimi: nope, all DTMF detection is done by the core asterisk, not DISA |
| 14:40.58 | s0ck | rob0: it's a low end HP ML110 G4 |
| 14:40.59 | ManxPower | shimi: what codec, what dtmfmode are you using? |
| 14:41.01 | s0ck | a test rig basically |
| 14:41.05 | shimi | codec? |
| 14:41.06 | s0ck | but i can't even get it working on that |
| 14:41.11 | shimi | it's a PRI. |
| 14:41.16 | shimi | no voip here. |
| 14:41.49 | rob0 | ~cheap |
| 14:41.50 | jbot | well, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 14:41.53 | [TK]D-Fender | shimi: And you should not call an app direct like that but rather dump them to an exten that at least does an explicit Answer. |
| 14:42.02 | ManxPower | shimi: then my guess would be your audio gains are messed up. Did you have to modify rxgan/txgain to remove echo? |
| 14:42.17 | s0ck | people used to say that about x100p's |
| 14:42.20 | s0ck | they worked, regardless |
| 14:42.47 | rolek | Can anyone tell me if it is possible to expire host entries in sip.conf? |
| 14:42.47 | rolek | My DNS changes seem not to be noticed by * |
| 14:42.47 | rolek | sip.conf: http://www.pastebin.org/36318 |
| 14:42.49 | shimi | ManxPower, pardon my stupid question, but, in that case, wouldn't asterisk have NOT said: chan_zap.c: DTMF digit: 4 on Zap/1-1 per each number I am dialing ? |
| 14:42.52 | s0ck | i've also just realised, none of the other boxes i have have pcie slots |
| 14:43.03 | shimi | [TK]D-Fender, isn't Answer() for calls coming INTO asterisk ? |
| 14:43.25 | [TK]D-Fender | shimi: jsut do it. |
| 14:43.42 | s0ck | you work for Nike? :) |
| 14:43.59 | shimi | no, their slogan is "just do it" |
| 14:43.59 | shimi | :) |
| 14:44.05 | s0ck | LOL |
| 14:44.22 | rob0 | they had a Learjet, "N1KE", when I was in ATC. |
| 14:44.26 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:44.28 | Nugget | http://www.theonion.com/content/node/39171 <-- Nike |
| 14:44.38 | Zeeek | oh? |
| 14:44.39 | atomspace | does anyone have experience with power over ethernet? I assume most of the time, voip phones are powered that way. |
| 14:44.40 | shimi | so basically you're saying that the Application: option in a call file is something that doesn't work |
| 14:44.49 | Nugget | N1KE is a famous plane. it's the one that had landing gear trouble and had to circle over boston for hours in an emergency. |
| 14:44.52 | shimi | atomspace, i do PoE |
| 14:45.02 | atomspace | shimi: --- do you use an injector? |
| 14:45.14 | atomspace | or some sort of poe switch? |
| 14:45.25 | shimi | nope, 3com PoE switch. I have a WiFi AP though, that works with an injector |
| 14:45.26 | rob0 | Arriving into my area once, they were held up high, and I had to give them a restriction to cross the arrival fix at 10000. |
| 14:45.47 | atomspace | *looking into that 3com switch now |
| 14:45.53 | rob0 | They said it might be a problem to meet the restriction. "Just do it," I said. |
| 14:46.17 | rob0 | and it's a Lear ... they did it :) |
| 14:47.03 | Dossy | [TK]D-Fender: I added "canreinvite=no" to sip.conf - no difference. |
| 14:47.11 | Dossy | I changed "nat=yes" and no difference. |
| 14:47.22 | *** part/#asterisk notbright (n=chatzill@unaffiliated/yourname/x-837320) |
| 14:47.40 | atomspace | so shimi, how exactly do you use the injector with the wap? |
| 14:47.43 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 14:48.03 | atomspace | the wifi AP itself is powered over ethernet? |
| 14:48.14 | shimi | ethernet cable comes from switch into injector, ethernet cable goes from injector to WiFi AP, and a power supply is connected to the injector |
| 14:48.31 | [TK]D-Fender | Dossy: Make sure to add it to EVERY entry. |
| 14:48.39 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:48.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:48.55 | atomspace | shimi:...and why do you use an injector with that wap? why not go strait from poe switch, into wap? |
| 14:48.58 | [TK]D-Fender | atomspace: I run PoE for my phones & Cisco AP's |
| 14:49.11 | shimi | the PoE is for VoIP, different network segment. :> |
| 14:49.17 | atomspace | ah, gotcha |
| 14:49.32 | atomspace | Fender, what about you, do you use a poe switch, or an injector? |
| 14:49.38 | shimi | (if it only had VLAN support...) |
| 14:49.47 | atomspace | its kind of my understanding that they do the same thing, but but the switch gives you more ports, right? |
| 14:50.13 | Dossy | [TK]D-Fender: OK. Trying. |
| 14:50.13 | shimi | anyways it doesn't make any difference. a PoE switch is just a switch that has PoE injector built in |
| 14:50.13 | rob0 | Um, Major Kong, a message just in on the CRM 114 ... you won't believe this ... it says "Wing attack, Plan R". |
| 14:50.13 | atomspace | ah, ok |
| 14:50.32 | shimi | a 24 port one could cost you something around $200-300 I think |
| 14:51.01 | atomspace | really? |
| 14:51.08 | atomspace | I that 3com shows ~$1700 |
| 14:51.25 | atomspace | well, you're right thought, newegg has some for under $300 |
| 14:51.37 | Dossy | [TK]D-Fender: no change. still nothing. :-( |
| 14:51.39 | pingwin | ManxPower: thanks for that info earlier, took care of it like a charm :D |
| 14:51.44 | shimi | [TK]D-Fender, OK I am trying to use a callfile with an extension that I added to extensions_custom. However asterisk says I need at least one app or extension... |
| 14:51.46 | *** part/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 14:51.54 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 14:52.11 | shimi | atomspace, take a "baseline" 3com, not high end switches ;) |
| 14:52.14 | Dossy | Hmm, could be firewalling rules. Let me sniff. |
| 14:52.33 | shimi | [TK]D-Fender, whoops, not extension. I meant a context. |
| 14:53.42 | Dossy | aha! that was it - RTP inbound to Asterisk was being filtered (ugh!) - need to specify a range so I can open the fw rules |
| 14:55.03 | shimi | [TK]D-Fender, ok I just used a bogus '0'. That seems to work. I don't understand what's the difference, though... |
| 14:55.56 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:56.02 | [TK]D-Fender | shimi: Merry Christmas. Always Answer your calls. |
| 14:56.18 | shimi | had it been an incoming call, I WOULD AGREE with you |
| 14:56.20 | shimi | however it is not. |
| 14:56.28 | [TK]D-Fender | atomspace: D-Link Des-1228P |
| 14:56.30 | shimi | now I removed the Answer from the custom DISA context |
| 14:56.32 | shimi | STILL WORKS. |
| 14:56.55 | [TK]D-Fender | shiFinal lesson, don't call "application" for call-files :) |
| 14:56.55 | shimi | I guess it's a bug in * :\ |
| 14:57.21 | shimi | well, if the feature exists, it should work; if it doesn't, zap it out (as if it was a channel ;)) |
| 14:58.07 | shimi | now phase 2: create an ext that calls my external program to do my callbacks ;) |
| 14:59.03 | Dossy | OK, opening up the RTP ports seems to have fixed things! yay. |
| 14:59.52 | Dossy | thanks, [TK]D-Fender |
| 15:00.01 | *** join/#asterisk murdock_ut (n=chatzill@70.99.184.194) |
| 15:00.14 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 15:00.16 | [TK]D-Fender | Dossy: np |
| 15:00.24 | deeperror | is there any way to run a queue that will ring sip extensions and pstn extensions? |
| 15:00.41 | [TK]D-Fender | deeperror: PSTN = serious trouble. |
| 15:00.47 | b11d` | Asterisk is hard coded to use an RTP packet size of .020 correct? |
| 15:01.03 | deeperror | thats what i figured |
| 15:02.11 | Dossy | now, if I can just figure out why playback of pre-recorded sound files gets truncated |
| 15:02.13 | Dossy | :) |
| 15:03.03 | b11d` | sigh.. i guess i'll buy a g729 licence and give it a shot.. |
| 15:03.43 | [TK]D-Fender | Dossy: All, or jsut the start of each? |
| 15:04.03 | Dossy | the start |
| 15:04.26 | km2 | does the telecom acronym TN = telephone number? |
| 15:04.28 | Dossy | like, if 4 or 5 sounds are played ... sometimes I only hear the 4th and 5th |
| 15:05.29 | [TK]D-Fender | Dossy: I'd suggest explicitly Answering the call, then doing something like Wait(2) |
| 15:05.33 | *** join/#asterisk xenonex (n=xenonex@82.200.211.120) |
| 15:10.11 | rolek | Any clues on dns expiry of si ppeer entries? |
| 15:10.39 | lmadsen | I usually do something like Playback(silence/1&file-I-want-to-play) |
| 15:11.21 | lmadsen | since Playback() will answer the channel if not already answered -- the silence/1 is almost always enough to prevent truncated audio |
| 15:11.35 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 15:12.30 | Zeeek | lmadsen but if this happens in the forest with no one nearby, will it make a noise? |
| 15:12.41 | lmadsen | of course it will |
| 15:12.48 | lmadsen | I'm omnipresent |
| 15:13.12 | Zeeek | If you believe that, then go here http://voipusersconference.org and phone in to the bridge today |
| 15:13.36 | jduggan | hey guys, if i do restart now in the cli, it'll disconnect me but it doesnt seem to fully restart (you can asterisk -r again), i cant make calls untill i restart it using my init scripts.. is there anything i can look at as to why this happens? |
| 15:13.50 | *** join/#asterisk Darrenmv (n=darren@adsl-155-80-80.btr.bellsouth.net) |
| 15:13.54 | Zeeek | just intgested a huge amount of garlic in my wife's home made humus |
| 15:14.07 | lmadsen | jduggan: you get disconnected, but then you can reconnect because it restarted........ |
| 15:14.15 | lmadsen | that is perfectly expected behaviour... |
| 15:14.30 | Zeeek | lmadsen like sawing off the branch you're sitting on, basically |
| 15:14.43 | lmadsen | the other reason it might not work / be delayed would be because something is blocking |
| 15:16.02 | tzanger | coppice: found something interesting |
| 15:16.04 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:16.20 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:16.23 | coppice | an endless supply of cash? |
| 15:16.38 | tzanger | I've hacked your sliptest program to bits to emit known data |
| 15:16.58 | tzanger | since it's sending in slinear, I have a "desired ulaw pattern to slinear converter" function I picked up online |
| 15:17.06 | tzanger | it works reasonably well |
| 15:17.17 | tzanger | but doesn't handle a few patterns correctly |
| 15:17.19 | *** join/#asterisk TrentCreek (n=trent@cpe-70-117-198-98.rgv.res.rr.com) |
| 15:17.23 | Zeeek | now posesses the greatest cordless phone ever invented http://x2z.eu/h for asterisk |
| 15:17.42 | tzanger | the pattern I am emitting is just 0x00 -> 0xff repeating |
| 15:18.13 | tzanger | what gets emitted is very close... 00 is converted to 0x02, and 0xff is converted to 0x7f |
| 15:18.39 | tzanger | all other patterns convert between ulaw and slinear correctly |
| 15:18.50 | tzanger | I guess the zaptel slin->ulaw isn't quite the same as my ulaw->slin |
| 15:19.16 | coppice | the zaptel ulaw and alaw were both broken the last time I looked |
| 15:19.25 | tzanger | orly |
| 15:20.31 | Zeeek | CDG |
| 15:21.00 | *** join/#asterisk railsmunky (n=nick@82-70-72-101.dsl.in-addr.zen.co.uk) |
| 15:21.22 | railsmunky | Howdy people. Can anyone help me with my extensions. I have this |
| 15:21.23 | railsmunky | [trunk-test] |
| 15:21.23 | railsmunky | exten => _9!,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) |
| 15:21.33 | coppice | actually, most people's alaw converters are brokn in the same way. someone (openser I think) discovered that recently, and suggested I fix spandsp. If they'd looked they would have found spandsp is one of the few things to do it right :-) |
| 15:21.45 | railsmunky | to make outgoing calls but i can't get asterisk to match on that |
| 15:21.56 | coppice | however, zaptel had unique breakage of ulaw the last time I looked |
| 15:21.59 | railsmunky | just keeps telling me that it's an invalid extension |
| 15:22.57 | tzanger | coppice: hmm |
| 15:23.18 | tzanger | I'm not using your ulaw->slinear conversion routine, in fact I dind't know spandsp had it |
| 15:23.26 | tzanger | I just found some code online :-) |
| 15:23.49 | tzanger | the zaptel code isn't off by much... the code emits 0x7f instead of 0xff, and 0x02 instead of 0x00 |
| 15:24.11 | coppice | if it only gets 2 values wrong, you have no EC or gain pads or other forms of distortion in the path. the errors in zaptel are probably still there |
| 15:24.54 | tzanger | coppice: yes, this is only tdmoe and no analog or any bit-manglers |
| 15:25.11 | Zeeek | please, this is a family channel |
| 15:25.25 | *** part/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 15:25.40 | tzanger | Zeeek: interesting phone |
| 15:25.49 | Zeeek | I LOVE THIS PHONE - seriously |
| 15:26.09 | Zeeek | I will make everyone sick of it by talking about it all the time. Until I get enough wine |
| 15:26.11 | railsmunky | anyone :) |
| 15:26.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:26.42 | Zeeek | Seems it isn't available in the US for the moment: S675IP |
| 15:27.08 | ManxPower | railsmunky: show us the Dial line AFTER it's evaluated (you can see that in the CLI) |
| 15:27.30 | railsmunky | [May 16 16:18:37] NOTICE[4230]: chan_sip.c:13952 handle_request_invite: Call from '828' to extension '901274641911' rejected because extension not found. |
| 15:27.41 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 15:27.48 | railsmunky | you mean that? |
| 15:28.23 | ManxPower | railsmunky: that is not the DIAL line. |
| 15:28.47 | ManxPower | That means 901274641911 is not found in the expected context |
| 15:29.07 | ManxPower | normally the context is also mentioned in that error message. I don't know why it wasn't,. |
| 15:29.12 | railsmunky | ah ok. |
| 15:29.23 | ManxPower | The Dial line in the CLI would have the word Dial in it. |
| 15:29.26 | [TK]D-Fender | ManxPower: His dialplan isn't even being called because he didn't point his device to the right context. |
| 15:29.39 | [TK]D-Fender | railsmunky: SIP debug will tell you what context its looking for. |
| 15:29.41 | ManxPower | [TK]D-Fender: so you'll help him then? |
| 15:29.45 | railsmunky | So do i need to get out of the current context for the extension i'm on? |
| 15:29.52 | [TK]D-Fender | ManxPower: You've just seen the extent of it :) |
| 15:30.08 | [TK]D-Fender | railsmunky: You aren't ON an "extensions". |
| 15:30.18 | ManxPower | railsmunky: we need to teach you to crawl before we teach you to walk. |
| 15:30.27 | [TK]D-Fender | railsmunky: your SIP device POIS to a context. That context clearly does not link to that exten you pasted for us |
| 15:30.39 | [TK]D-Fender | POINTS* |
| 15:30.49 | ManxPower | now, put a copy of your extensions.conf and sip.conf, masking only the passwords, and the CLI output of a failed call (I don't want to see sip debug at this point). |
| 15:31.00 | ManxPower | put them on pastebin.ca |
| 15:31.00 | railsmunky | Yeah i get that - so obvisouly if i put the _9! in the context my extension is in - it works |
| 15:31.26 | ManxPower | railsmunky: well you can't call an extension if you dno't have it in your context |
| 15:31.37 | ManxPower | that's the whole POINT of contexts -- access control. |
| 15:31.42 | [TK]D-Fender | railsmunky: Well tahts like saying "why can't I find to book I left on shelf 1?" when you actually put it on shelf *2* |
| 15:31.51 | railsmunky | yeah i get that completly |
| 15:32.05 | ManxPower | railsmunky: cool, that was an easy fix |
| 15:32.09 | Dossy | [TK]D-Fender: the audio "drop out" also was happening in the middle of voicemail app ... hmm |
| 15:32.13 | railsmunky | i just don't get moving from my current context into the context for outgiong |
| 15:32.15 | [TK]D-Fender | railsmunky: Then if you want 1 context to be able to use extens of another you need to " include =>" it |
| 15:32.30 | [TK]D-Fender | Dossy: maybe bandwidth related. |
| 15:32.42 | railsmunky | _9! is still considered an extension? |
| 15:33.01 | ManxPower | railsmunky: contexts are both one of the most important things to learn with Asterisk -- it is also the most difficult things to learn. |
| 15:33.14 | ManxPower | railsmunky: does it have an exten => before the _9! ? |
| 15:33.14 | [TK]D-Fender | railsmunky: its a pattern. its an extension if you have a "exten => _9!,1,Blah.... |
| 15:33.59 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 15:34.03 | railsmunky | yeah which i get - so if i include trunk-test in my context for extensions that will put calls which match _9! in the context of trunk-test? |
| 15:34.06 | railsmunky | or extensions? |
| 15:34.26 | [TK]D-Fender | railsmunky: Yes |
| 15:34.39 | railsmunky | ManxPower: no this is my outgoing lines |
| 15:34.50 | Dossy | [TK]D-Fender: maybe - although I have 30Mbit down/5Mbit up. :( |
| 15:34.55 | Dossy | Latency? |
| 15:34.56 | Zeeek | It's 12 PM EDT: in about 26 min |
| 15:35.00 | railsmunky | ok cool. thanks guys. I know how to tinker - i just want to make sure it's right |
| 15:35.11 | tzanger | coppice: yeah I guess you're right... I am not testing all slinear values, only 256 of them |
| 15:35.24 | Dossy | aha - http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone - "Some (older?) GS phones have a setting for "Voice Frames per TX" with a default value of 10." - let me try setting it to 2 |
| 15:35.46 | [TK]D-Fender | Dossy: INDEED 20MS IS WHAT YOU SHOULD BE USING. |
| 15:35.55 | ManxPower | railsmunky: perhaps it is time for you to spend some time reading The Good Book |
| 15:36.24 | Dossy | hmm, it was set to 2. that rules that out. |
| 15:36.46 | Dossy | and, I'm using ulaw codec. mah |
| 15:40.11 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-124-226.se.biz.rr.com) |
| 15:40.37 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:40.52 | ManxPower | Dossy: I suspect 2 means 2ms, not 20ms |
| 15:42.15 | *** join/#asterisk asdx (n=diego@adsl-128-242.click.com.py) |
| 15:42.17 | asdx | ~book |
| 15:42.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:42.18 | Dossy | ManxPower: no, 2 means 20ms. 10 means 100ms. |
| 15:42.39 | Zeeek | http://VoipUsersConference.com in 20 minutes for Click 2 Call apps from TringMe and 1ezphone. This stuff works and may be of interest to your customers |
| 15:42.50 | Zeeek | ^Zeeek |
| 15:42.55 | Zeeek | ~Zeek |
| 15:42.59 | Zeeek | ~Zeeek |
| 15:42.59 | jbot | extra, extra, read all about it, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 15:43.13 | Zeeek | I never get tired of that |
| 15:43.40 | ManxPower | Dossy: What moron set it to 100ms? |
| 15:43.55 | Dossy | ManxPower: old firmware defaulted to 100ms. newer firmware (which I'm using) defaults to 20ms. |
| 15:44.13 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:44.13 | Dossy | but, obviously, old wiki pages still point out that if it's set to 100ms, to drop it to 20ms. mine already was. |
| 15:45.36 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 15:45.53 | variable_office | is there a way i can make a catchall sip peer in sip.conf? |
| 15:46.24 | ManxPower | variable_office: you mean a guest? |
| 15:46.51 | ManxPower | generally whatever is in [general] will be used for unauthenticated calls. Asterisk accepts unauthenticated calls BY DEFAULT |
| 15:47.22 | variable_office | Well, i want asterisk to stop saying "Postgresql RealTime: Could not find any rows in table sip_users." |
| 15:47.41 | ManxPower | variable_office: I have no idea how you would do that when using REALTIME. |
| 15:47.56 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:48.14 | variable_office | well if that is a catchall, i can just enter it into sip.conf and asterisk won't even both checking realtime hopefully |
| 15:49.12 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 15:49.25 | variable_office | humm, that doesnt seem to work |
| 15:50.36 | *** join/#asterisk sp00k3y (n=chatzill@74.202.4.2) |
| 15:55.02 | eric2 | so I have this linksys 2102 gateway and when playing xbox online, the connection cuts out constantly, anyone know what needs to be adjusted on the voip router to prevent this? |
| 15:58.30 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 15:59.23 | variable_office | i am just going to recompile a new version is there anything special that needs to be done from 1.4.16.2 to 1.4.19.2 |
| 15:59.35 | jblack | eric2: It's either your router or your xbox that need adjusting. |
| 15:59.57 | jblack | the xbox is hogging all the bandwidth, so either get the xbox to be more polite, or start googling for traffic shaping on your router. |
| 16:00.37 | eric2 | haha... good old traffic shaping |
| 16:00.59 | eric2 | I made a change to the QOS setting.. phone is fine, xbox takes a dump |
| 16:01.35 | jblack | then get more bandwidth |
| 16:01.57 | shido6 | hrmm |
| 16:02.16 | *** part/#asterisk shido6 (n=shido6@209.114.208.192) |
| 16:02.36 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 16:04.42 | [TK]D-Fender | variable_office: No. |
| 16:07.14 | variable_office | [TK]D-Fender, my asterisk box is taking between 2-3 invites to respond to the other side, (SIP) 2 invities works, but on the 3rd invite bandwidth.com (provider) sends the invite from a different host and then my box responds to the 1st host and the whole call gets messed up and dropped. The only thing I can figure is that the 5 or so "Postgresql RealTime: Could not find any rows in table sip_users." messages are wasting time in call proc |
| 16:07.14 | variable_office | essing; but i have added the first "VIA:" as a peer and it still happens; any ideas? |
| 16:08.31 | b11d` | hrm.. i've got my g729 codec, registration succeeded, yet "show g729" isnt available as a command.. |
| 16:08.39 | b11d` | show modules lists codec_g729a as a loaded module.. |
| 16:10.02 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 16:10.16 | watchy | anyone here ever do distinetive ringing with polycoms? |
| 16:10.31 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
| 16:10.53 | mackes | Watchy: Yep |
| 16:10.54 | jblack | variable_office: Try show codecs, perhaps? |
| 16:11.05 | jblack | then from there, perhaps show codec |
| 16:11.19 | watchy | mackes: got any howtos on doing it> |
| 16:12.24 | mackes | You define them in your polycom config files and then, call them with the Alert command in your dial plan |
| 16:12.41 | variable_office | jblack, i am just using g711 |
| 16:13.18 | jblack | variable_office: I was talking to b11d, actually. |
| 16:13.40 | b11d` | show codecs lists it.. but "show codecs" says its for info only, doesnt show config.. |
| 16:13.50 | mackes | Google This: Polycom distinctive ring asterisk |
| 16:14.02 | jblack | b11d`: Now try show codec g729 (notice: no s) |
| 16:14.14 | b11d` | tried that.. it expects a numeric value.. not a name.. |
| 16:14.35 | b11d` | show codec 256 shows g729, but again, its just for info.. |
| 16:14.37 | jblack | get the name from the show codecds list. |
| 16:14.39 | b11d` | doesnt mean anything.. |
| 16:15.00 | jblack | Ok. I have no idea then |
| 16:15.30 | b11d` | thanks :) |
| 16:16.18 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 16:16.27 | *** join/#asterisk Defraz (n=T0tal@69.92.19.83) |
| 16:17.51 | maqr | before i subscribe to an ITSP, what would you guys recommend i use to test if voip will even be viable for me and my connection? |
| 16:18.14 | [TK]D-Fender | maqr: what do you have? |
| 16:18.25 | maqr | [TK]D-Fender: a fresh asterisk 1.6 beta install with no configuration :) |
| 16:18.36 | [TK]D-Fender | maqr: no, your CONNECTION <------ |
| 16:19.17 | maqr | [TK]D-Fender: oh, i'm on a comcast business account (2.5MB/s down, 250KB/s up), but my pbx is going to sit on a linode VPS, to which i ping 35ms |
| 16:19.28 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:19.29 | maqr | (xen vps) |
| 16:19.35 | [TK]D-Fender | maqr: your upstream is bad. |
| 16:19.48 | maqr | [TK]D-Fender: that's really good upstream... not kilobits, kilobytes |
| 16:19.55 | s0ck | especially if that is Kb/s and not KB/s |
| 16:20.10 | maqr | 250 kilobytes per second |
| 16:20.17 | s0ck | nice |
| 16:20.47 | maqr | i've got enough bandwidth in all directions, i think |
| 16:20.53 | maqr | i'm more concerned about latency |
| 16:20.56 | [TK]D-Fender | maqr: ok, then you're fine. |
| 16:20.57 | dandre | hello |
| 16:21.04 | maqr | i'm thinking FWD might be a good place to test? but maybe that's not the right way to do it |
| 16:21.16 | maqr | [TK]D-Fender: maybe there's some kind of 'jitter testing tool' or something, like iperf, but for voip? |
| 16:21.17 | dandre | I have difficulties to build astmanproxy |
| 16:21.22 | [TK]D-Fender | maqr: SIP call to something more or less as far. |
| 16:21.46 | maqr | [TK]D-Fender: are there any way to get benchmark numbers? or is it just whatever 'sounds okay'? |
| 16:21.57 | [TK]D-Fender | maqr: Latter |
| 16:22.35 | maqr | [TK]D-Fender: and if i don't have an ITSP yet, how would i go about testing? should i use FWD or some other service? is there some test-number that i should be calling? |
| 16:24.07 | [TK]D-Fender | maqr: FWD is a decent test. |
| 16:24.15 | [TK]D-Fender | maqr: test call to Digium, etc. |
| 16:24.33 | maqr | [TK]D-Fender: digium has a number to call? |
| 16:24.35 | *** part/#asterisk ology (n=gene@67-210-207-232.generic.webformix.com) |
| 16:24.41 | maqr | got a link or soemthing about that? i'm new :) |
| 16:25.01 | [TK]D-Fender | maqr: Look at the sample extensions.conf |
| 16:25.08 | maqr | oh, alright |
| 16:25.11 | maqr | thanks :) |
| 16:26.04 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 16:26.18 | dandre | tzafrir ? |
| 16:26.18 | *** join/#asterisk mintee (n=mintee@72-165-177-94.dia.static.qwest.net) |
| 16:26.47 | mintee | anyone know of a good VOIP USB Phone that works well with *? |
| 16:27.01 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 16:28.31 | *** join/#asterisk [T]an2 (n=ckwall@206.71.78.158) |
| 16:28.49 | [TK]D-Fender | mintee: * doesn't work with USB phones. |
| 16:29.02 | *** part/#asterisk [T]an2 (n=ckwall@206.71.78.158) |
| 16:29.35 | maqr | [TK]D-Fender: oh, one more question, for asterisk 1.6.0-beta9, which ztdummy driver should i get? |
| 16:29.46 | maqr | zaptel-1.4.10.1 maybe? |
| 16:30.09 | atomspace | is there a way to make one phone (in my case, an analog phone connected via ATA) not have to dial 9 first? Basically, any number dialed by this one phone would get put out to the trunk? |
| 16:30.11 | [TK]D-Fender | maqr: 1.6 is in BETA and not recommended for production, and you should use the latest versiosn listed in the channep topic. |
| 16:30.37 | [TK]D-Fender | atomspace: The only reason you have to dial 9 first is because YOU set your dialplan up taht way. |
| 16:30.52 | maqr | [TK]D-Fender: oh, then i'll install that :) |
| 16:30.57 | atomspace | oh, so I can just give that 1 extension a different dialplan? |
| 16:31.08 | atomspace | I didn't put that together, thanks a lot Fender |
| 16:31.19 | [TK]D-Fender | atomspace: And stop calling your "phones" as "EXTENSIONS" |
| 16:31.33 | [TK]D-Fender | atomspace: Chenge the patterns you match against. |
| 16:31.51 | [TK]D-Fender | atomspace: And don't forget your ATA has rules of its own about what to allow to apss on to *. |
| 16:32.02 | atomspace | apss ? |
| 16:32.09 | atomspace | oh, gotcha |
| 16:32.11 | maqr | notes downloads.digium.com is slow today |
| 16:34.07 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 16:34.31 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.143.86) |
| 16:34.38 | *** join/#asterisk tobias (n=tobias@cpe-069-134-226-227.nc.res.rr.com) |
| 16:36.29 | s0ck | any of you chaps in the uk? |
| 16:36.46 | s0ck | trying to work out what happens, in terms of voltage, when a call is picked up on a pstn line |
| 16:36.48 | s0ck | bt/uk specific |
| 16:37.13 | Nasra | have a question: the ATA --> SPA3102-[NA] <----Does that mean anything ...do they all have the 1 -fxs + 1-fxo nothing to worry about and are they all unlocked? |
| 16:37.42 | ManxPower | Nasra: NA means "NOT LOCKED" |
| 16:37.50 | b11d` | whats up with the codecs found at http://asterisk.hosting.lv/ ? |
| 16:37.51 | Nasra | okay |
| 16:37.53 | b11d` | legit? |
| 16:37.55 | ManxPower | But for all the other information, look on SIPura's web site. |
| 16:38.20 | ManxPower | b11d`: the only legit codecs are either free or licensed from Digium |
| 16:38.24 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:38.49 | b11d` | i figured.. |
| 16:38.50 | ManxPower | b11d`: references to those pirated codecs are the only thing I have ever seen Digium remove from the mailinglist archives. |
| 16:38.56 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:39.01 | b11d` | i just bought g729 codec from digium.. |
| 16:39.08 | b11d` | cant get it to work :) |
| 16:39.18 | b11d` | Manx.. funny.. I found that link on the mailing list archives.. |
| 16:39.21 | [TK]D-Fender | b11d`: CALL THEM |
| 16:39.21 | ManxPower | b11d`: Intel licenses THEIR code and libraries free for non-commercial use, but you still have to get a license for the actual codec from the patent holders. |
| 16:39.47 | ManxPower | and there is no "non-commercial use" exception for the codec patents |
| 16:39.53 | Nasra | ManxPower: another one, so If I have spa3102-NA ...so don't need card TDM400P ( I am a newbie...you can tell).. |
| 16:39.59 | b11d` | interesting.. |
| 16:40.02 | ManxPower | b11d`: then report a link to the message |
| 16:40.10 | b11d` | will do.. |
| 16:40.19 | ManxPower | Nasra: what does the SIPura web page for the SPA3102 say? |
| 16:40.48 | [TK]D-Fender | Nasra: the SPA does what it says. If that what you wanted the TDM400P for, then you have your own answer |
| 16:40.51 | ManxPower | b11d`: Digium risks losing their G729 license if they don't do at least minimal things to prevent unlicensed usage of G729 |
| 16:41.05 | Nasra | ManxPower: good answer ...have to read...thanks |
| 16:41.17 | b11d` | sure.. that makes sense.. |
| 16:41.20 | b11d` | i'll fire them an email.. |
| 16:42.00 | ManxPower | b11d`: I think I'll report that site to the patent holders as well |
| 16:42.01 | jblack | changing queue configuration seems awkard. There seems to be no "queue reload", and "reload" doesn't seem to fully reconfigure queues, thus requiring a full restart of asterisk. Does this sound accurate? |
| 16:42.17 | [TK]D-Fender | ManxPower: IIRC, Digium paid up front for the licenses they distribute, its a done deal. If I wrote an infringing codec for X-Lite, are they liable if they don't make their product incompatible with my "hack"? |
| 16:42.23 | putnopvut | jblack: module reload app_queue.so |
| 16:42.30 | jblack | of course |
| 16:42.33 | jblack | Thanks |
| 16:42.53 | ManxPower | [TK]D-Fender: no, but they could be considered liable if they let you post links to your codec on digium run systems |
| 16:42.58 | ManxPower | licenses can be revoked |
| 16:44.06 | [TK]D-Fender | ManxPower: the most Digium could do is try to remove the post, at which poitn every mirror would still have it. This is such a thin veil its not funny. |
| 16:44.29 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-210-156.nycmny.east.verizon.net) |
| 16:44.49 | b11d` | Manx.. since you're gonna email them .. the post can be found at: http://lists.digium.com/pipermail/asterisk-bsd/2008-January/003127.html |
| 16:44.55 | b11d` | whoops.. |
| 16:44.56 | b11d` | meant http://lists.digium.com/pipermail/asterisk-bsd/2008-January/003121.html |
| 16:45.10 | ManxPower | There, sent off |
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| 16:45.26 | ManxPower | b11d`: I e-mailed the patent holders, not Digium |
| 16:45.29 | b11d` | oh ok |
| 16:45.39 | b11d` | *sniff* |
| 16:45.44 | b11d` | i did my part ;) |
| 16:45.51 | ManxPower | you can notify Digium abut the mailing list archoves |
| 16:45.53 | [TK]D-Fender | ManxPower: ".lv" what country is that? I'm betting one that doesn't support the concept of "software patents" |
| 16:45.58 | b11d` | latvia isnt it? |
| 16:46.09 | ManxPower | [TK]D-Fender: not my problem -- it's the patent holder's problem |
| 16:46.11 | b11d` | TK.. undoubteld.y |
| 16:46.37 | ManxPower | the G729 google group isn't hosted there at least. |
| 16:47.31 | ManxPower | I just did them the courtesy of notifying them of a violation |
| 16:47.47 | [TK]D-Fender | ManxPower: From my personal book of alternative definitions : salmonella ; (state) When you're sick and tired of constantly swimming up stream. |
| 16:49.47 | blablabla_123543 | [TK]D-Fender: I fixed my prob from yesterday, i dont know if you remember |
| 16:50.16 | blablabla_123543 | it was indeed the caller id |
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| 16:56.02 | Kobaz | where can i find some info on doing a hook flash transfer |
| 16:56.44 | ManxPower | Kobaz: press the FLASH or RECALL key on your phone. |
| 16:56.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:56.52 | Kobaz | i mean in asterisk |
| 16:57.19 | ManxPower | Kobaz: set transfer and threeway calling in zapata.conf for your ZAP ports. |
| 16:57.26 | Kobaz | ah |
| 16:57.27 | [TK]D-Fender | Kobaz: From what device to what other device? |
| 16:57.40 | Kobaz | fxo on the asterisk box |
| 16:57.49 | Kobaz | from another pbx |
| 16:57.50 | ManxPower | Kobaz: you mean FXS, right? |
| 16:57.53 | [TK]D-Fender | Kobaz: details dammit. |
| 16:58.08 | Kobaz | working on it |
| 16:58.08 | [TK]D-Fender | Kobaz: what HARDWARE exactly. |
| 16:58.29 | Kobaz | sangoma fxo, hooked up to an fxs on a notel pbx |
| 16:58.39 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 16:58.43 | Kobaz | get incomming call, do a hook flash to transfer to another number |
| 16:58.54 | ManxPower | Kobaz: now you are starting to give us useful information. "show application flash" |
| 16:59.02 | Kobaz | k |
| 16:59.08 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 16:59.42 | ManxPower | Kobaz: YOU do the hookflash, or you want Asterisk to do the hookflash? Asterisk does not support passthru of hookflash |
| 16:59.42 | Kobaz | asterisk does it |
| 17:00.08 | ManxPower | call comes in, your dialplan runs Flash, dial the correct number, hangup. |
| 17:00.19 | Kobaz | something like that |
| 17:00.31 | Kobaz | call comes in, goes into a queue |
| 17:00.57 | Kobaz | the agent answers a call in the queue, agent monkeys with our web console and types in a transfer to xyc |
| 17:01.00 | ManxPower | Kobaz: I'm sorry I cannot help you further unless you step back, figure out what you need to ask, then we can try again. |
| 17:01.30 | ManxPower | Ah, sorry, I can't help with web consoles |
| 17:01.33 | Kobaz | and then ast would need to do a hook flash to transfer the call away |
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| 17:01.38 | Kobaz | oh, i dont need help with that |
| 17:01.42 | Kobaz | just the asterisk part |
| 17:01.51 | ManxPower | Kobaz: you now have all the information you need for the Asterisk part |
| 17:01.56 | Kobaz | yeap |
| 17:02.36 | ManxPower | the Flash application (you might want to look at the Transfer app too, but I don't think it's correct for your rather complicated requirements) |
| 17:02.36 | Kobaz | k |
| 17:08.44 | mintee | [TK]D-Fender, i meant a SIP phone, that just happens to be USB... |
| 17:09.12 | [TK]D-Fender | mintee: SIP has nothing to do with USB. |
| 17:09.36 | mintee | i didn't say it did |
| 17:09.58 | mintee | i meerly asked... if anyone knew of a USB SIP phone that worked with * |
| 17:10.10 | [TK]D-Fender | mintee: No such thing as a SIP USB phone. |
| 17:10.39 | mintee | ORLY |
| 17:10.52 | [TK]D-Fender | mintee: Sure, go link me to one. |
| 17:11.04 | mintee | http://www.google.com/products?q=usb%20sip&ie=UTF-8&oe=utf-8&rls=org.mozilla:en-US:official&client=firefox-a&um=1&sa=N&tab=wf |
| 17:11.05 | RypPn | the usb phone is really a usb dsp for the audio into a softphone |
| 17:11.26 | RypPn | so sip is the softphone |
| 17:11.31 | [TK]D-Fender | mintlink me directly to the exact product please. |
| 17:11.46 | *** join/#asterisk redback (n=noname@mail.datadream.co.uk) |
| 17:11.47 | mintee | the first link |
| 17:12.01 | redback | has anyone had success in getting MYSQL commands to work within an ael file - I keep getting app_addon_sql_mysql.c:268 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '\ password\ from\ agents' at line 1 |
| 17:12.08 | [TK]D-Fender | mintee: 1st link is a WiFi SIP phone. |
| 17:12.13 | mintee | it's WiFI but charges via USB |
| 17:12.32 | redback | but I am sure the command is correct, show dialplan shows the statement as MYSQL(Query RES_ID ${CON_ID} SELECT\ password\ from\ agents) |
| 17:12.39 | [TK]D-Fender | mintee: whoopde doo, it doesn't TALK via USB |
| 17:12.53 | mintee | you assumed that's what I was asking |
| 17:12.59 | mintee | I wasn't |
| 17:12.59 | [TK]D-Fender | mintee: thats like saying I have a 110v AC that talks SIP to *. |
| 17:13.12 | mintee | whatever, don't be so nieve |
| 17:13.19 | mintee | your gonna blow another vessle |
| 17:13.20 | [TK]D-Fender | mintee: USB on that phone has nothing to do with anything other than it charging its battery |
| 17:13.54 | mintee | so, basically, your saying there are no actual SIP phones that talk via USB.... |
| 17:14.00 | mintee | hum |
| 17:14.06 | mintee | that's a bunch of suck |
| 17:14.19 | [TK]D-Fender | mintee: SIP is a a protocol over TCP/IP <----- |
| 17:14.28 | [TK]D-Fender | mintee: that has nothing to do with USB. |
| 17:14.31 | mintee | USB network device... |
| 17:14.37 | mintee | just saying |
| 17:15.01 | mintee | i figured someone made something like that |
| 17:15.13 | mintee | *sigh* |
| 17:15.32 | [TK]D-Fender | mintee: Technically possible, but if its counted as a network device then thats an OS driver issue. And from there once functional it'd still jsut be a NETWORK device and doesn't talk directly to * over USB, but rather through its NETWORK interface. |
| 17:15.59 | mintee | yeah, correct. that's kinda what I was hoping |
| 17:16.16 | [TK]D-Fender | mintee: Yeah I have USB network devices.... like http://www.tigerdirect.ca/applications/SearchTools/item-details.asp?EdpNo=1354224&CatId=200 |
| 17:16.29 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 17:16.30 | [TK]D-Fender | mintee: but who the hell would do that and build a PHONE into it? |
| 17:16.39 | mintee | for a phone I could take to a hotel and just plug into my laptop and reconnect with * via SIP |
| 17:16.49 | mintee | lol |
| 17:17.04 | [TK]D-Fender | mintee: You just never seem to get it and missed the comment above... |
| 17:17.31 | [TK]D-Fender | mintee: use a SOFTPHONE on your PC, and a USB **handset** on your laptop |
| 17:17.53 | [TK]D-Fender | mintee: the handset is jsut a dumb audio device that is used by the softphone. |
| 17:18.10 | TrentCreek | softphones suck |
| 17:18.16 | mintee | that's what I've been doing, but thought it'd be cool to actually have a hardware phone.. but another idea of my fails. |
| 17:19.50 | [TK]D-Fender | mintee: Nearly useless idea. Next yourlaptop is going to know that it needs to act like a ROUTER for that device too? Double NAT problems? EEK. boat-loads of stupid there.. |
| 17:20.28 | [TK]D-Fender | mintee: there is no vaguely profitable market for such a thing. |
| 17:20.43 | mintee | yeah, i guess.... I was kinda depending on a VPN to solve that problem |
| 17:21.14 | redback | this MYSQL error is driving me nuts. I am escaping the spaces, i tested the command through mysql client on the same machine with same user and it worked |
| 17:21.29 | redback | but whatever I put in my .ael file it fails |
| 17:21.50 | mintee | wonders how many people beat up [TK]D-Fender during his childhood to make him say things like "boat-loads of stupid there.." |
| 17:21.51 | mintee | ;) |
| 17:23.00 | *** join/#asterisk MrWork (n=Mr@206.245.153.216) |
| 17:23.17 | redback | man I have even made the statement as simple as possible and still nothing |
| 17:23.25 | [TK]D-Fender | mintee: None. You should rather wonder how many people come in here with seriously lame ideas and no clue. Then take that number and add yourself to that count :p |
| 17:23.27 | redback | is loosing the will to live |
| 17:23.42 | mintee | hahaha... touche |
| 17:24.18 | [TK]D-Fender | redAnd you are not showing us anything so that we can help you. PASTEBIN is your friend. |
| 17:24.20 | [TK]D-Fender | ~pb |
| 17:24.20 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:24.38 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:26.35 | redback | http://www.pastebin.ca/1020260 - sorry I did paste the line that errors out but here is the context |
| 17:27.35 | Katty | oooh my |
| 17:27.37 | Katty | i'm stuffed |
| 17:27.42 | Katty | sprawls |
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| 17:31.57 | TrentCreek | ~destroy <[TK]D-Fender> |
| 17:31.58 | jbot | no |
| 17:32.01 | TrentCreek | ;-) |
| 17:32.35 | TrentCreek | ~burn <[TK]D-Fender> |
| 17:32.36 | jbot | ACTION pours gasoline all over <[TK]D-Fender>, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 17:32.59 | rob0 | Yuck! Petrol fumes spoil good marshmallows. |
| 17:33.08 | redback | heh |
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| 17:38.57 | xacatecas | hi guys, i've got a problem when asterisk bridges between a SIP and ZAP trunk. It's as if when I speak the other end gets cut out, so whilst i speak i can't hear them, and then i get scratchy noises when i start/stop speaking. |
| 17:39.08 | xacatecas | has anybody else observed that ... and is there a known fix? |
| 17:40.37 | *** join/#asterisk asteriskmonkey (n=philip@137.122.72.219) |
| 17:41.33 | [TK]D-Fender | xacatecas: What SIP device and Zap device? What networking between your SIP device & *? |
| 17:41.43 | [TK]D-Fender | TrentCreek: Careful playing with fire there... |
| 17:42.01 | TrentCreek | buwahahaha |
| 17:42.15 | ManxPower | xacatecas: sounds like the classic symptom if AGGRESSIVE ECHO CANCEL being enabled |
| 17:42.28 | xacatecas | [TK]D-Fender, Grandstream BT 200 phones, same LAN, no issues when breaking out over VoIP (IAX2) trunk. digium TDM844P zaptel side. |
| 17:42.54 | [TK]D-Fender | xacatecas: use a softphone and see if the problem repeats |
| 17:42.59 | xacatecas | ManxPower, how can I check that? |
| 17:43.11 | ManxPower | That zaptel option really should be named "half duplex voice" |
| 17:43.15 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 17:43.20 | *** part/#asterisk MrWork (n=Mr@206.245.153.216) |
| 17:43.23 | xacatecas | [TK]D-Fender, I've had it with an IPT1000H and one of the snom phones as well. |
| 17:43.27 | ManxPower | xacatecas: I think it's in ztconfig.h in the zaptel config |
| 17:43.36 | ManxPower | ..er.. in the zaptel source dir |
| 17:43.41 | xacatecas | ManxPower, on a pre-packaged binary? |
| 17:43.52 | ManxPower | xacatecas: you can't check it on that. |
| 17:43.55 | [TK]D-Fender | xacatecas: Ok, What Zaptel versions? Which EC compiled? |
| 17:44.08 | ManxPower | compile from source, or talk to the package maintainer for support. |
| 17:44.15 | xacatecas | bugger. |
| 17:44.38 | ManxPower | xacatecas: why did you think we hate packages around here? |
| 17:44.49 | ManxPower | Your issue is a classic example of why. |
| 17:45.03 | Damin | <PROTECTED> |
| 17:45.06 | xacatecas | same issue why i hate them. they never seem to get it right. |
| 17:45.08 | Damin | Oops.. |
| 17:45.20 | ManxPower | Exactly. |
| 17:45.31 | xacatecas | which is why i'm more and more inclined to use source-based distro's. |
| 17:45.59 | xacatecas | ManxPower, if i set echocancel=no it should disable it right? |
| 17:46.00 | anonymouz666 | [TK]D-Fender |
| 17:46.06 | ManxPower | xacatecas: One of the VERY few applications I compile from source is Asterisk (ClamAV, Spamassassin, are the other two) |
| 17:46.06 | anonymouz666 | what a good example of FXO ATA? |
| 17:46.09 | xacatecas | that should at least confirm your theory? |
| 17:46.14 | ManxPower | xacatecas: It should |
| 17:46.27 | ManxPower | You'll get echo, but you should not get half duplex audio |
| 17:46.34 | *** join/#asterisk srand06 (n=srand@77.206.204.33) |
| 17:46.38 | srand06 | hi guys |
| 17:46.44 | xacatecas | rigs a phone over this 384/128kbps adsl line ... and prays that it'll give reasonable performance. |
| 17:46.51 | srand06 | Hangup issue :-( |
| 17:47.05 | [TK]D-Fender | anonymouz666: SPA-3102 |
| 17:47.30 | ManxPower | srand06: turn off callprogress= and busydetect= (they default to no. If you have them set to yes, then set them to no or remove them) |
| 17:47.41 | [TK]D-Fender | anonymouz666: a bit of a misnomer. ATA = analog TERMINAL adapter and implies that you'll attach a phone. But close enough.... |
| 17:47.45 | ManxPower | I can't help you further without a GOOD description of your problem. |
| 17:47.47 | srand06 | ManxPower : let me try |
| 17:48.09 | srand06 | ManxPower : general options I suppose ? |
| 17:48.11 | ManxPower | srand06: those two options are commonly called "randomlydisconnectmycalls=yes" around here. |
| 17:48.19 | [TK]D-Fender | xacatecas: ouch. |
| 17:48.41 | [TK]D-Fender | xacatecas: Better be using G.729 or at least GSM .... |
| 17:48.43 | ManxPower | srand06: remove those options anywhere you find them in zapata.conf |
| 17:48.44 | anonymouz666 | [TK]D-Fender: it's a fxs/fxo ATA that I am looking for. you can register a SIP account to FXO and dials to it. I am using Grandstream shit, but they sucky |
| 17:48.58 | *** join/#asterisk SQLDarkly (n=admin@199-117-163-66.dia.static.qwest.net) |
| 17:49.02 | xacatecas | GSM atm ... but considering switching the g729 around to face to the phone temporarily. |
| 17:49.13 | [TK]D-Fender | anonymouz666: SPA-3102 is generally pretty decent |
| 17:49.29 | srand06 | ManxPower : but I have no zapata device |
| 17:49.39 | srand06 | ManxPower : I'm using a pstn gateway |
| 17:49.48 | ManxPower | srand06: As I said, I cannot help you further without a good description of your problem. |
| 17:49.49 | *** join/#asterisk saftsack (n=oliver@p54A71493.dip0.t-ipconnect.de) |
| 17:50.06 | SQLDarkly | Ok I have set up an SIP extension. I have setup port=5060 in this extension its defaulting to port 46444 when I do SIP Show Peers in asterisk CLI |
| 17:50.13 | ManxPower | This is NOT a good description of your problem "srand06: Hangup issue :-(" |
| 17:50.21 | srand06 | ManxPower : you're right |
| 17:50.24 | SQLDarkly | I mean its functional and working but why is it refusing to use the specified default port |
| 17:50.34 | *** join/#asterisk sbrobou (n=sbrobou@201.48.5.105) |
| 17:50.41 | ManxPower | SQLDarkly: you are setting the destination port, but are looking at the source port. The source port can be any random port number |
| 17:50.45 | xacatecas | ok, great, g729, now i can hear the scratching. |
| 17:50.46 | sbrobou | Sbrobous has come |
| 17:50.59 | srand06 | ManxPower : I use Hangup in my extensions.conf and it does not work properly |
| 17:51.04 | SQLDarkly | Should I not be concerned about this then? |
| 17:51.13 | SQLDarkly | Is it actually using 5060 ? |
| 17:51.16 | ManxPower | SQLDarkly: don't manually set the ports anwhere. |
| 17:51.16 | xacatecas | echocancel=off ... same issue. |
| 17:51.34 | ManxPower | SQLDarkly: you understand that every IP connection has two ports associated with it, right? |
| 17:51.38 | srand06 | ManxPower : So I tried the following : (play a digit and Hangup ) |
| 17:51.40 | SQLDarkly | Yes |
| 17:52.08 | ManxPower | SQLDarkly: what command are you using that you see the 46444 port number? |
| 17:52.08 | srand06 | ManxPower : It plays the digit but after 2 second it says call is unknown |
| 17:52.33 | ManxPower | srand06: what HAPPENS when hangup is issued. do you see the hangup running in the CLI? |
| 17:52.35 | SQLDarkly | sip show peers in asterisk cli (asterisk -r ) |
| 17:52.39 | xacatecas | could a slightly damanged FXO port cause it? although, I've seen this on three different boxes now. |
| 17:52.41 | srand06 | ManxPower : yes |
| 17:52.44 | ManxPower | SQLDarkly: and that shows you the REMOTE SOURCE PORT. |
| 17:52.55 | ManxPower | you don't really care about the remote port. |
| 17:53.02 | SQLDarkly | AH |
| 17:53.14 | sbrobou | Sarava. Friends... I'm trying to set ringing tone up in zaptel. I already set the zapata.conf's "language" and "progzone" and the indications.conf, but I hear the same ringing tone |
| 17:53.19 | SQLDarkly | Ok i was not aware of that Max |
| 17:53.38 | ManxPower | SQLDarkly: I've asked the devs to make that output clearer about the port number |
| 17:53.50 | ManxPower | I don't know if they will or not. |
| 17:54.00 | sbrobou | how can I change? I also tried the "dial(.......${exten}r1), but I hear a continuous tone this way |
| 17:54.02 | SQLDarkly | Yes it should be defined clearer |
| 17:54.10 | SQLDarkly | but I do thank you for that tidbit |
| 17:55.03 | sbrobou | is there a page explaining how to do that? |
| 17:55.03 | ManxPower | sbrobou: r1 specifies the destination phone ringing cadence, not the ringback (what the caller hears) |
| 17:55.56 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 17:56.37 | ManxPower | SQLDarkly: you really need to fully understand TCP/UDP, NAT, networking, routing to be any good at Asterisk |
| 17:56.45 | ManxPower | sbrobou: I have no idea. |
| 17:56.59 | xacatecas | hmm, another question - i've got a hardware echo canceller on the TDM844 ... I shouldn't even need software echo cancel at all - or could this double echo cancelling be exactly what's causing my problem? |
| 17:57.07 | sbrobou | I tought that defining a cadence, it would work |
| 17:57.16 | srand06 | ManxPower : it says the following http://www.pastebin.sk/6601/ |
| 17:57.23 | sbrobou | uou... |
| 17:59.18 | sbrobou | so, let's see, my people. Question one: does it is possible? Is possible to change the ringback from zaptel? Yes or No? :) |
| 17:59.57 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 18:00.15 | sbrobou | im going to lose this fight... Round two... fight! |
| 18:00.25 | *** join/#asterisk bannanabrain (n=twisted@64.129.235.200) |
| 18:00.47 | srand06 | ManxPower : Any idea ? |
| 18:00.47 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 18:01.30 | unpaidbill | has anyone here set up a foundry switch with data/voice vlans and voip phones with the port for a computer on them |
| 18:02.00 | sbrobou | your situation is worst than mine |
| 18:07.17 | *** join/#asterisk killmel8tr (n=IceChat7@ip-198-22-65-131.quickconnectusa.com) |
| 18:08.39 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:10.16 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 18:10.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:11.04 | *** join/#asterisk s0lid (n=s0lid@210.213.198.203) |
| 18:12.27 | jeffspeff2 | hi all, i did some system software updates earlier, one of them was the kernel... i have since noticed that my conference rooms aren't working... i haven't changed a thing in the dialplans or files... here is the error i get when i try to go to a meetme exten... "build_conf: Unable to open pseudo device" |
| 18:12.41 | xacatecas | stuff this ... i'm building asterisk onto my gentoo box now. i've had enough of this binary crap. |
| 18:12.45 | mvanbaak | jeffspeff2: recompile zaptel |
| 18:13.16 | jeffspeff2 | mvanbaak, is it because of the new kernel? |
| 18:13.34 | mvanbaak | jeffspeff2: yes, your zaptel kernel modules were build for the older kernel |
| 18:13.41 | mvanbaak | so they wont load with this new kernel |
| 18:13.54 | mvanbaak | you have to recompile them so they get installed for this new kernel |
| 18:14.15 | jeffspeff2 | ahhh... that's what i was thinking, but wanted to make sure... thanks.... also, do i have to recompile asterisk or anything else? |
| 18:14.40 | [TK]D-Fender | jeffspeff2: yes, every time you recompile zaptel, you should do * again as well. |
| 18:14.41 | mvanbaak | nah, just recompile zaptel |
| 18:14.46 | mvanbaak | why ? |
| 18:14.59 | mvanbaak | I never had to do that |
| 18:15.02 | [TK]D-Fender | safeties sake |
| 18:15.12 | [TK]D-Fender | can't go wrong. |
| 18:15.42 | jeffspeff2 | [TK]D-Fender, if i recompile * is it going to overwrite my dialplans? |
| 18:15.48 | mvanbaak | no |
| 18:15.56 | jeffspeff2 | ok, just checking |
| 18:16.45 | lmadsen | as long as you don't run 'make samples' |
| 18:16.46 | killmel8tr | just make sure you do a 'make samples' ;) |
| 18:16.49 | lmadsen | do NOT run it again :) |
| 18:17.02 | lmadsen | kills killmel8tr right now |
| 18:17.05 | *** join/#asterisk azasterisk (i=azasteri@wsip-70-166-101-146.ph.ph.cox.net) |
| 18:18.40 | azasterisk | Hello, is there any way to see what a clients SIP password is? I have my asterisk box set up and one of my users is tring to connect with the wrong password, I would like to use the password he has chosen as there is no other phone service there to contact him. |
| 18:19.25 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:19.50 | delparnel | Hi.. is there any way accomplish this... I call a DID on my asterisk box, which puts me to a DISA... I enter a number, the server hangs up on me, calls me back, and connects the call? |
| 18:19.54 | [TK]D-Fender | azasterisk: e-mail him. |
| 18:20.19 | [TK]D-Fender | delparnel: Yes you can do this. you'd use an AMI originate, or call-file, not Disa |
| 18:20.49 | *** join/#asterisk gomel77 (n=gomel77@wn2nat35.beelinegprs.ru) |
| 18:20.58 | delparnel | Which would be better? |
| 18:21.23 | azasterisk | [TK]D-Fender....I could do that....but no one is there right now....is the SIP password encrypted? I have tcpdump up and running but have heard that the SIP may be sent using MD5? |
| 18:21.25 | *** join/#asterisk steliosk (n=Stelios@athedsl-284816.home.otenet.gr) |
| 18:21.31 | delparnel | My objective is to be able to use Callback with my address book on my blackberry |
| 18:21.55 | [TK]D-Fender | delparnel: either would do the job. |
| 18:22.07 | [TK]D-Fender | delparnel: Call-files is the simpler approach. |
| 18:22.26 | delparnel | Ok, I'll look into it. Thanks |
| 18:23.19 | *** part/#asterisk gomel77 (n=gomel77@wn2nat35.beelinegprs.ru) |
| 18:23.41 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:24.29 | xacatecas | is it possible to set up a custom context that does Echo() but with a one second delay? |
| 18:25.08 | Strom_M | xacatecas: why do you want to induce echo into a circuit? |
| 18:25.23 | plik | delparnel: why even have * answer your call - you could have it drop your incoming call when it recognises the callerid, then call you straight back and ask for the number you want to connect you to |
| 18:25.43 | xacatecas | i'm trying to troubleshoot a voice issue, and having to disturb somebody on the other end the whole time when they cannot hear me is annoying the client. |
| 18:25.54 | xacatecas | i can hear them, they can't hear me. |
| 18:26.11 | mvanbaak | xacatecas: are they behind nat or is asterisk behind nat ? |
| 18:26.32 | xacatecas | asterisk on my side is behind nat, them not. going through an intermediate switch. |
| 18:26.34 | delparnel | plik: Because then you can't use the address book to dial |
| 18:26.44 | mvanbaak | ~sipnat |
| 18:26.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:26.57 | plik | ah ok... just a thought |
| 18:27.45 | xacatecas | mvanbaak, i do have DNAT set up on my side to port forward port 4569 to my asterisk server. |
| 18:28.09 | mvanbaak | that's IAX |
| 18:28.34 | xacatecas | jip. |
| 18:29.41 | jeffspeff2 | in * make menuselect... what are the "Module Embedding" options for? |
| 18:29.58 | *** part/#asterisk azasterisk (i=azasteri@wsip-70-166-101-146.ph.ph.cox.net) |
| 18:32.18 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 18:32.31 | xacatecas | ok well, it seems to work reasonably well now, i just forgot to actually Answer() before running Echo() |
| 18:32.38 | delparnel | plik: Works great if you have photographic memory. |
| 18:34.27 | plik | people used to be good at remembering numbers - it's only in the last 20 years or so of mobiles being really popular that once again the precieved technological advantage has eroded our natural faculties and continues to turn us in to morons :/ |
| 18:34.41 | *** part/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 18:35.13 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 18:36.09 | xacatecas | has anybody here gotten the codec renogotiation patch at http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch working yet? |
| 18:37.08 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 18:37.30 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 18:41.13 | TrentCreek | not in Malatecas |
| 18:42.30 | xacatecas | plik, 20 years? heck, i only got my first cell phone about 8 years ago. and i could memorise phone numbers by simply hearing them once. |
| 18:43.18 | *** join/#asterisk jelly-bean (i=user@63-76-119-176.directcom.com) |
| 18:45.20 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:46.34 | plik | xacatecas: can you still memorise phone numbers by simply hearing them once? |
| 18:46.44 | mackes | http://paste.lisp.org/ doesnt seem to have access to #asterisk ? |
| 18:46.54 | xacatecas | sometimes. depends on how badly i want to remember it. |
| 18:47.56 | *** join/#asterisk s0lid (n=s0lid@210.213.198.203) |
| 18:48.29 | plik | heh, a good point... |
| 18:48.30 | bkruse | mackes: #asterisk requires registration, I believe that is anonymous |
| 18:48.58 | TrentCreek | how can we get * to run across two LANs? |
| 18:50.04 | mackes | I run Asterisk across several Lans |
| 18:50.11 | mackes | What is the issue you are having- |
| 18:50.34 | sbrobou | your are the RainMan |
| 18:50.40 | sbrobou | you are.. |
| 18:51.10 | TrentCreek | one lan is connected by wifi. it supplies my connection. The other is my own local LAN. The * box is suppling the Wifi |
| 18:51.11 | mackes | How are these LAN connected? Routed/Bridged or Nat'd |
| 18:51.44 | TrentCreek | My own lan is connected to a routers connecting all the boxes together |
| 18:52.03 | TrentCreek | the ATA is on internal LAN |
| 18:52.53 | xacatecas | plik, to give you an idea ... i can at the very least recognise numbers i've seen before, and I can remember probably around 15 of my client's numbers off the top of my head. |
| 18:52.54 | TrentCreek | the ATA is fixed IP as well as the LAN card on * box in internal lan |
| 18:52.57 | *** part/#asterisk ghantoos (n=ghantoos@kimlus.ghantoos.org) |
| 18:53.07 | TrentCreek | but it is not communicating with * |
| 18:53.17 | TrentCreek | though the dial plan is correct |
| 18:54.06 | TrentCreek | I am in the middle of changing ISP, and connected to friend's LAN, getting access through his service |
| 18:54.21 | plik | yeah, it's really a case of #use it, or lose it' ... if you stop using the no's in your head, recall generally gets harder |
| 18:54.22 | mackes | Please provide more information as to your network design- What device sits between these two LAN's- Who makes it, and what does it do? |
| 18:54.37 | mackes | oh. |
| 18:54.40 | mackes | I See. |
| 18:54.45 | mackes | You are NATing |
| 18:54.55 | mackes | SIP and NAT does not work well. |
| 18:55.33 | mackes | Are you attempting to use your friends Internet Access via WIFI to supply your Asterisk server Internet? |
| 18:55.41 | TrentCreek | It was working before...that is till I put the wireless card in and used it for internet access |
| 18:55.45 | TrentCreek | yes |
| 18:56.03 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.06 | TrentCreek | It is receiving calls fine, but not forwarding them to the ATA |
| 18:56.14 | TrentCreek | nor cal I call out |
| 18:56.21 | TrentCreek | *can |
| 18:56.25 | redback | hurray I worked out what was wrong (3 hours later) the manual said to escape the query string with backslahes - but this must be out of date as it works if I do not escape them, I did however need to escape commas |
| 18:57.21 | mackes | Ok- You have alot going on there to trouble shoot- First, I would use a Wireless Bridge, like a Linksys with DDWRT to connect to your Friends WiFI.. and plug the Ethernet of your sever to that bridge |
| 18:58.04 | mackes | It sounds as if the Network Setup on your server is out of wack- not so much Asterisk- |
| 18:58.14 | mackes | Make sure you can ping everything |
| 18:58.24 | TrentCreek | I have a US Robotics 5461 that has bridge mode, but not so sure how to set that up |
| 18:58.29 | mackes | Make sure all points can ping back |
| 18:59.01 | mackes | Its all in the networking- I dont think its Asterisk |
| 18:59.32 | xacatecas | does anybody have any feedback re the nat_sip module in the kernel? |
| 18:59.40 | TrentCreek | maybe so..since both routers outside and in, are of same brand..Linksys |
| 19:00.04 | TrentCreek | maybe because the LAN addresses are the same |
| 19:00.19 | mackes | Na, that shouldnt matter- can you ping the internet from your Asterisk box? |
| 19:00.25 | mackes | Ohhhhhh. |
| 19:00.57 | TrentCreek | so both I think have 192.168.1.xxx |
| 19:01.02 | mackes | All of your devices can be on the same subnet but cannot have the same addresses |
| 19:01.18 | TrentCreek | I mean the routers |
| 19:01.32 | TrentCreek | 192.168.1.1 I think |
| 19:02.37 | jelly-bean | hey i am setting up new IT infrastructure for a local startup. they just leased the building today so we're making decisions like what type of internet to bring in and what type of phone system to use... |
| 19:02.59 | jelly-bean | since we only have a small call center of 7 agents receiving inbound only calls, we're looking at a remote/hosted VoIP solution. that way, ideally, the only equipment we'd need is a couple desktops, usb softphones |
| 19:02.59 | xacatecas | TrentCreek, yea, that would kinda kill it. |
| 19:02.59 | jelly-bean | we need it up within the next week. i know asterisk has potential. can anyone advise? |
| 19:03.38 | xacatecas | jelly-bean, i say go for it. |
| 19:03.45 | TrentCreek | Maybe the bridge mode would be a better idea. Got a general idea of how that is suppose to work? |
| 19:04.02 | jelly-bean | can anyone recommend a good VoIP host? |
| 19:04.10 | jelly-bean | ringcentral.com looks good but i'm not sure its asterisk |
| 19:04.23 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-131-41.w90-30.abo.wanadoo.fr) |
| 19:04.28 | jelly-bean | i was also considering TrixBox but i would rather have it hosted externally so if our internet goes down people still get voicemail |
| 19:04.32 | xacatecas | doesn't matter, as long as they do SIP, but I'd prefer a host that provides IAX2 if I were you. |
| 19:04.40 | bougie | hello =) |
| 19:05.10 | xacatecas | jelly-bean, trixbox in my experience is a pain. it works, but all too often i want something that is not catered for. |
| 19:05.29 | xacatecas | it's going to be more effort, but it may be worth it to just install a raw asterisk and build the config by hand. |
| 19:05.33 | xacatecas | use any distro you like. |
| 19:05.58 | xacatecas | as long as you work within it's framework - it works well though. |
| 19:06.19 | jelly-bean | xacatecas: if i did that, i'd rather pay someone to set up the hardware |
| 19:06.32 | xacatecas | the hardware is the easy part. |
| 19:06.32 | jelly-bean | it seems more risky to manage the hardware and software setup ourselves |
| 19:06.46 | jelly-bean | our business depends on our inbound sales calls |
| 19:06.54 | cpm | doesn't everyones? |
| 19:06.58 | xacatecas | ok, i might just be suicidal :) |
| 19:07.03 | jelly-bean | cpm: no. :) |
| 19:07.03 | Strom_C | jelly-bean: where are you located? |
| 19:07.06 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-140-147.client.mchsi.com) |
| 19:07.06 | xacatecas | also consider response times. |
| 19:07.14 | jelly-bean | Strom_C: Utah |
| 19:07.36 | xacatecas | you're on site the whole time probably if you're part of the business ... if stuff goes wrong - response time == immediate. if it's manged externally it can be up to a few hours. |
| 19:07.42 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 19:07.42 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:07.44 | jelly-bean | xacatecas: how to response times vary between custom build and hosted? |
| 19:07.53 | jelly-bean | ah you're right |
| 19:08.03 | jelly-bean | if it went down remotely we're at their mercy completely |
| 19:08.26 | jelly-bean | does anyone here offer an asterisk setup service? :D |
| 19:08.34 | xacatecas | jip, but remember that since you're probably going to be using an "upstream" trunk, you're going to be on that trunks mercy in any case. |
| 19:08.34 | jelly-bean | if i built the box and got you ssh... |
| 19:08.48 | Strom_C | jelly-bean: I do :) |
| 19:11.52 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:13.00 | thepacmanfan | gah, this is driving me up a wall |
| 19:13.26 | thepacmanfan | zaptel can't find my kernel sources in centos 5.1 |
| 19:13.52 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 19:14.08 | xacatecas | thepacmanfan, install kernel-devel or something. |
| 19:14.34 | thepacmanfan | i have kernel-devel, and a link from /usr/src/linux to /usr/src/kernels/2.6.18.xxxxxx |
| 19:16.45 | xacatecas | and from /var/lib/modules/$(uname -r)/build ? |
| 19:17.25 | xacatecas | which EC should I prefer when compiling from source? |
| 19:18.15 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-210-156.nycmny.east.verizon.net) |
| 19:18.24 | thepacmanfan | xacatecas: no i don't, how do i do that one? |
| 19:18.43 | xacatecas | ls -ld /var/lib/$(uname -r)/build |
| 19:19.10 | thepacmanfan | do that while in /usr/src? |
| 19:19.56 | xacatecas | from anywhere |
| 19:20.08 | *** join/#asterisk ikevin (n=kevin@2001:5c0:99b9:0:20f:b0ff:fe4a:4589) |
| 19:20.35 | thepacmanfan | hmm... there is no uname -r in /var/lib |
| 19:21.06 | Strom_C | sigh |
| 19:21.15 | Strom_C | <PROTECTED> |
| 19:21.19 | thepacmanfan | 2.6.18-53.el5 is my kernel, btw |
| 19:21.37 | thepacmanfan | but there's no folder for currentver in /var/lib |
| 19:25.11 | xacatecas | thepacmanfan, ls -ld /var/lib/modules/`uname -r`/build ? |
| 19:26.36 | thepacmanfan | ls: /var/lib/2.6.18-53.el5/build: No such file or directory |
| 19:27.24 | thepacmanfan | when i ls /var/lib, there's nothing with a version number in it. |
| 19:27.30 | xacatecas | you've got a missing _modules_ folder between lib and 1.6.18... |
| 19:28.35 | thepacmanfan | huh? |
| 19:28.56 | thepacmanfan | oh |
| 19:29.32 | thepacmanfan | one problem... there is no modules folder in lib |
| 19:30.26 | thepacmanfan | somehow my sources must not have properly installed... is kernel-devel the same thing as kernel headers? |
| 19:30.46 | thepacmanfan | i just did yum install kernel-devel |
| 19:30.48 | *** join/#asterisk kingsob (n=kingsob@HMTNON14-1242538569.sdsl.bell.ca) |
| 19:31.30 | *** join/#asterisk machpo (n=mcp@ketone.jawfish.com) |
| 19:33.42 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:33.42 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:34.59 | xacatecas | no, not the same. |
| 19:37.51 | *** join/#asterisk TrentCreek (n=trent@cpe-70-117-198-98.rgv.res.rr.com) |
| 19:38.30 | thepacmanfan | errr, are you sure? several guides tell me otherwise. |
| 19:39.33 | xacatecas | very, very different. kernel-headers is used by userspace apps, not stuff that links against the kernel. |
| 19:39.54 | xacatecas | whereas kernel-devel is the stuff required to actually build kernel modules. |
| 19:40.01 | thepacmanfan | yum install kernel-headers tells me "nothing to do" |
| 19:40.15 | *** join/#asterisk dFence (n=chatzill@p54980C1F.dip0.t-ipconnect.de) |
| 19:40.17 | xacatecas | yea, it's one of those core packages that all systems must have type thing. |
| 19:40.21 | *** join/#asterisk mike-ekim (n=digiport@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 19:40.24 | thepacmanfan | ok, so i have it... |
| 19:40.45 | thepacmanfan | wonder why the heck i don't have /var/lib/modules |
| 19:40.57 | errr | thepacmanfan: yes I am sure |
| 19:41.14 | mike-ekim | if I have something like: exten => exten,1,Set(DN=${SIP_HEADER(TO)}), etcetc... and change it to exten => exten,1,Macro(macroname) |
| 19:41.19 | errr | really has no idea what he is sure of though |
| 19:41.20 | [TK]D-Fender | thepacmanfan: "yum install kernel* |
| 19:41.26 | thepacmanfan | errr: very very nice... i bow to thee. |
| 19:41.28 | TrentCreek | It's working now, but I had to play with the connections after I changed the IP address on the outside router. Seems the box wants to use internet via the first network card that is initialized |
| 19:41.29 | mike-ekim | and have SET(DN=${SIP_HEADER(TO)}) in a macro |
| 19:41.34 | mike-ekim | would it still recognize the values? |
| 19:41.50 | mike-ekim | or would i need to pass it as arg |
| 19:43.22 | thepacmanfan | D-Fender: all that nets me is kernel-PAE, kernel-debug, kernel-doc, kernel-xen, and devel versions for most of those. i don't need any of those for zaptel, do i? |
| 19:45.27 | thepacmanfan | yum list installed shows kernel, kernel-devel, and kernel-headers, all of the same version. |
| 19:45.29 | *** join/#asterisk dmcmorris_esi (n=dmcmorri@dmcmorris.net) |
| 19:45.59 | [TK]D-Fender | thepacmanfan: just do it |
| 19:47.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:47.53 | *** join/#asterisk broheem (n=broheem@72.16.138.81) |
| 19:47.53 | *** join/#asterisk watchy2 (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 19:48.04 | watchy2 | anyone setup paging on polycoms and does it work ok? |
| 19:48.48 | [TK]D-Fender | watchy2: yes, and fine |
| 19:48.52 | dmcmorris_esi | there's a page on voip-info.org about paging, with polycoms specifically mentioned |
| 19:49.06 | watchy2 | whats it do if the caller is on the phone? |
| 19:50.52 | *** part/#asterisk broheem (n=broheem@72.16.138.81) |
| 19:51.14 | dmcmorris_esi | watchy2: voip-info.org has a little explanation, including a way to check to see if their channel is busy before paging them |
| 19:51.27 | [TK]D-Fender | watchy2: it will simply ring (cw) |
| 19:51.38 | watchy2 | ah ok |
| 19:51.56 | [TK]D-Fender | watchy2: and you might want to check the status first |
| 19:52.03 | watchy2 | yea i agree |
| 19:54.43 | *** part/#asterisk mike-ekim (n=digiport@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 19:55.14 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 19:56.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.48 | thepacmanfan | ok, i finally found my modules folder! |
| 19:57.58 | thepacmanfan | here's the results: lrwxrwxrwx 1 root root 43 May 15 04:01 /lib/modules/2.6.18-53.el5/build -> ../../../usr/src/kernels/2.6.18-53.el5-i686 |
| 19:58.35 | thepacmanfan | that seems to point to the kernel source correctly. |
| 19:58.40 | *** join/#asterisk xenonex (n=xenonex@82.200.211.120) |
| 19:59.22 | Kobaz | anyone know if the rhino rcbfxx is supposed to support hook flash |
| 19:59.37 | Kobaz | <PROTECTED> |
| 19:59.37 | Kobaz | [May 16 15:56:10] WARNING[12168]: app_flash.c:97 flash_exec: Unable to flash channel Zap/1-1: Device or resource busy |
| 20:01.32 | *** join/#asterisk uluatu (n=deg@mail.ondacorp.com.br) |
| 20:02.57 | *** join/#asterisk mltlnx (n=mltlnx@pool-96-232-210-156.nycmny.east.verizon.net) |
| 20:05.35 | Kobaz | never mind |
| 20:05.50 | TrentCreek | Darn! The multiple lan setup is not working |
| 20:06.09 | TrentCreek | Phone rings, but Cannot hear voice |
| 20:06.37 | Kobaz | it works if i wait |
| 20:06.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.11 | [TK]D-Fender | TrentCreek: You still NAT'd to him? |
| 20:08.18 | TrentCreek | yes |
| 20:08.28 | [TK]D-Fender | TrentCreek: And he's NAT'd to the internet? |
| 20:08.34 | TrentCreek | yeps |
| 20:08.54 | TrentCreek | i can call out also, but seems other end does not pick up |
| 20:09.29 | [TK]D-Fender | TrentCreek: Does he need VoIP out his connection? |
| 20:09.35 | TrentCreek | no |
| 20:09.48 | thepacmanfan | my bad... i forgot to reboot after installing kernel-devel |
| 20:10.03 | thepacmanfan | zaptel is compiling with no issues at this point. :) |
| 20:11.58 | TrentCreek | [TK]D-Fender: Maybe do port forwarding to my machine? |
| 20:13.41 | [TK]D-Fender | TrentCreek: set your peer to nat=yes, don't forward anything, "qualify=yes" for that peer |
| 20:13.55 | [TK]D-Fender | TrentCreek: and if you have multiple phones on your side you're probably screwed |
| 20:14.11 | watchy2 | exten => 4203,1,SIPAddHeader(Alert-Info: Ring Answer) |
| 20:14.15 | watchy2 | is that correct? |
| 20:14.31 | TrentCreek | just one phone. Do I set that on his routers? |
| 20:15.57 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 20:17.14 | *** join/#asterisk gardo (n=gardo@121.97.192.59) |
| 20:20.18 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 20:20.23 | talntid | cows say moo |
| 20:22.08 | [TK]D-Fender | TrentCreek: you don't touch his |
| 20:22.56 | TrentCreek | [TK]D-Fender: Okay..trying now to find my network config files |
| 20:23.10 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
| 20:23.54 | [TK]D-Fender | TrentCreek: what do you mean network config files? |
| 20:24.22 | *** join/#asterisk marc7 (n=marc@s207-6-221-31.bc.hsia.telus.net) |
| 20:25.04 | TrentCreek | [TK]D-Fender: to do your instructions. I don't know any other way to set "qualify=yes" |
| 20:25.15 | [TK]D-Fender | TrentCreek: sip.conf <- |
| 20:25.34 | [TK]D-Fender | TrentCreek: thats not something you'd term "network config files" |
| 20:25.58 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 20:26.39 | [TK]D-Fender | ok, checkout time, BBIAB |
| 20:27.06 | [TK]D-Fender | watchy2: Could be, depends on your sip.cfg and the rest of your dialplan. |
| 20:29.31 | *** join/#asterisk ragb (n=ricardo@92.0.217.14) |
| 20:37.10 | TrentCreek | qualify=yes |
| 20:37.11 | TrentCreek | nat=yes |
| 20:37.11 | TrentCreek | <PROTECTED> |
| 20:37.54 | talntid | Nobody has an obligation to answer this, but I had a quick question that is most likely very simple for you guys... my Asterisk guru is on vacation, but I had a date wrong for something, and I need to make it so all incoming calls go directly to another phone #.... can someone assist me with that? |
| 20:38.34 | talntid | I know to edit extensions.conf, and reload it, but.. not sure what the proper syntax is to make it forward properly... |
| 20:38.50 | TrentCreek | ~book |
| 20:38.50 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:39.01 | talntid | excellent |
| 20:39.23 | TrentCreek | All about * in there |
| 20:41.04 | talntid | chapter 5 :) |
| 20:43.05 | TrentCreek | groovy |
| 20:45.05 | talntid | Hm. I hope I don't break it! ;) |
| 20:47.07 | *** join/#asterisk uluatu (n=deg@mail.ondacorp.com.br) |
| 20:48.47 | Kobaz | errr |
| 20:49.02 | Kobaz | Flash() seems broken |
| 20:49.03 | errr | Kobaz |
| 20:49.07 | Kobaz | haha |
| 20:49.10 | errr | :) |
| 20:49.58 | Kobaz | Flash() very much doesn't seem to be doing a flash, it's going on hook |
| 20:50.15 | Kobaz | and it's sending a dtmf of "1" for some odd reason |
| 20:51.15 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 20:51.40 | Kobaz | anyone familiar with Flash() ? |
| 20:52.17 | Strom_C | what kind of circuit are you trying to flash? |
| 20:52.42 | Kobaz | i have an fxo on my box |
| 20:52.47 | Kobaz | and an fxs on a simulator |
| 20:53.18 | *** join/#asterisk watchy2 (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 20:53.27 | *** part/#asterisk dmcmorris_esi (n=dmcmorri@dmcmorris.net) |
| 20:53.29 | watchy2 | using meetme for paging isnt a good idea is it |
| 20:54.39 | _ShrikE | watchy2: that is what app_page does, and it works OK |
| 20:54.55 | _ShrikE | for smaller page groups at least |
| 20:55.15 | TrentCreek | [TK]D-Fender: "qualify=yes nat=yes" did not seem to work |
| 20:56.10 | *** join/#asterisk bakermd (n=none@38.104.0.102) |
| 20:56.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:56.29 | bakermd | I am not getting MWI on my PolyCom 501 - where should I look first to troubleshoot? |
| 20:57.22 | Strom_C | do you have the mailbox= entry set correctly in the relevant sip.conf entry? |
| 20:57.31 | bakermd | Will double check |
| 20:57.57 | Kobaz | Strom_C: http://www.pastebin.ca/1020521 |
| 20:58.24 | Kobaz | Strom_C: everything is working fine up until 33:26.23 |
| 20:58.24 | bakermd | Strom_C: mailbox=100@device |
| 20:58.52 | Kobaz | Strom_C: i do a Flash() on asterisk, and my simulator gets a "1" and asterisk goes on hook |
| 20:58.52 | Strom_C | is the mailbox in the "device" context in voicemail.conf? |
| 20:59.28 | bakermd | Nope - default - Thanks!! |
| 21:00.08 | Kobaz | Strom_C: would you think this is asterisk related or line card related? |
| 21:00.20 | Strom_C | what kind of line card are you using? |
| 21:00.29 | Kobaz | i have a sangoma in the mail due on monday, i'm using a rhino rcbfxx card right now |
| 21:00.48 | Kobaz | i wouldn't be surprised if it's rhino's fault |
| 21:00.56 | Kobaz | their t1 drivers are still broken |
| 21:01.20 | Strom_C | yeah, see if the new card fixes the problem |
| 21:01.36 | Kobaz | what i dont get, is why i'm getting a dtmf of "1" |
| 21:01.41 | Strom_C | who knows |
| 21:01.43 | Kobaz | heh |
| 21:01.57 | Strom_C | what happens if you monitor the circuit with a buttset? |
| 21:02.23 | Kobaz | the asterisk side goes on hook |
| 21:02.48 | Kobaz | circuit goes byebye |
| 21:03.15 | Strom_C | i would wait for the new card and see what happens |
| 21:03.30 | Kobaz | yeah |
| 21:04.10 | redback | ${CHANNEL} gives me the channel name which is mainly made up of the handset name but is the a variable that can give me the handset name i.e. the one in iax.conf / sip.conf |
| 21:04.31 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:06.29 | Strom_C | redback: how aboutn CALLERID(num)? |
| 21:06.35 | Strom_C | s/utn/ut/ |
| 21:07.37 | *** join/#asterisk rvhi (n=chatzill@udp197017uds.hawaiiantel.net) |
| 21:07.39 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:08.29 | rvhi | hi, got issues with a few seconds of silence during a call |
| 21:08.34 | redback | Strom_C: coming up blank |
| 21:08.36 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 21:08.41 | rvhi | then it came back itself |
| 21:08.47 | rvhi | anyone had this before? |
| 21:09.03 | Strom_C | redback: well, have you set caller ID for each of your stations in sip.conf? |
| 21:09.53 | [TK]D-Fender | TrentCreek, you'd better pastebin you complete setup for us then |
| 21:10.03 | redback | the name but not number yet but I need the full string i.e. IAX2/ext1000 instead of just 1000 |
| 21:10.17 | Strom_C | ${CHANNEL} is what you want then, probably |
| 21:10.47 | redback | is there a way to remove the dash and number from the channel - any sting manipulation commands? |
| 21:10.49 | Strom_C | and you probably want to do something about all those run-on sentences too |
| 21:11.03 | [TK]D-Fender | redback, "core show function CUT" |
| 21:11.23 | redback | Strom_C: heh, yeah - tired. Sorry. |
| 21:11.30 | Strom_C | always some lame excuse |
| 21:11.44 | redback | [TK]D-Fender: cheers |
| 21:12.10 | Strom_C | if you're so tired that you can't even construct sentences properly, lord knows what horrors you're doing to your configuration files :) |
| 21:13.09 | redback | Strom_C: nobody is perfect! The sentence was contrusted properly, it just lacked puntuation. I was being lazy. This is IRC, so informal. But as I said. Sorry. |
| 21:17.22 | TrentCreek | okay |
| 21:19.02 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:19.15 | iratik | Is there a way to say "reset as soon as there are no calls active?" |
| 21:19.32 | Strom_C | restart when convenient |
| 21:21.03 | iratik | yeah... thats how to say it |
| 21:21.09 | iratik | but how do i tell asterisk that? |
| 21:21.35 | iratik | i guess i could create an ssh script that invokes a php script that checks the number of open channels with ami, the script runs again and again.. when the script returns 0 .. then restart |
| 21:21.40 | iratik | but that.. is alot of work |
| 21:21.53 | TrentCreek | [TK]D-Fender: Here is sip.conf http://pastebin.com/mb4416bf |
| 21:21.56 | Strom_C | iratik: that's the command you issue at the asterisk CLI |
| 21:22.00 | Strom_C | "restart when convenient" |
| 21:22.09 | iratik | no _s? |
| 21:22.14 | iratik | thats awesome |
| 21:22.43 | [TK]D-Fender | TrentCreek, I thought you said he didn't use his internet connection for VoIP.... |
| 21:23.39 | TrentCreek | <[TK]D-Fender>: Correct//I was thinking you were talking about setting up the router. been months since I touched a * config |
| 21:24.43 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 21:25.03 | [TK]D-Fender | TrentCreek, ok, not sure what to advise here... other than of course stop double NAT-ing and jsut bridge your networks. |
| 21:25.37 | TrentCreek | [TK]D-Fender: Got any advice on how to bridge? |
| 21:25.38 | *** join/#asterisk acren (n=pmowry91@12.166.196.9) |
| 21:25.57 | [TK]D-Fender | TrentCreek, check your routers config. |
| 21:26.16 | [TK]D-Fender | TrentCreek, if it can't replace it. |
| 21:26.22 | ManxPower | TrentCreek: what router do you have? |
| 21:27.00 | acren | Sorry to just jump right in with a question, but can anyone give me a hand with t.38 passthrough? |
| 21:27.29 | TrentCreek | [TK]D-Fender: I have a few different brands. 1 One Linksys.. Another is 1 USR 5461 that has a "bridge mode" as a option, ut not sure ow that works |
| 21:27.30 | ManxPower | acren: not many people use it. If you don't get an answer here, you might want to ask on the mailinglists |
| 21:27.52 | ManxPower | TrentCreek: on a Cisco router I'd just set up a GRE tunnel |
| 21:28.03 | ManxPower | but with those boxes......well...best of luck. |
| 21:28.34 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 21:28.45 | TrentCreek | hey..Linksys is suppose to be Cisco ;-) |
| 21:29.26 | [TK]D-Fender | TrentCreek, You have optiosn. Get off your ass and TRY THEM. |
| 21:29.39 | ManxPower | TrentCreek: they are not real cisco routers |
| 21:30.02 | TrentCreek | yes I am, but I am not sure how to bridge two routers |
| 21:30.03 | iratik | Refressh my memory.. i'm trying .. reading the guides. .. exten=> _*555X.,1, ChanSpy(SIP/${EXTEN:4}) .... if i dialed *5555121 ... it would execute ChanSpy(SIP/5121) right? |
| 21:30.25 | ManxPower | bicycle is to motorcycle as Linksys is to Cisco Routers (running IOS) |
| 21:30.45 | ManxPower | TrentCreek: there is no direct connection between the two networks? |
| 21:31.09 | TrentCreek | no |
| 21:31.32 | ManxPower | TrentCreek: then you cannot bridge them. With the hardware you have you also cannot set up a tunnel between them |
| 21:31.43 | ManxPower | iratik: yes |
| 21:32.19 | acren | ok, how bout a g729 question, can you * route a call between 2 SIP gateways if it does not have the g729 codec itself? I have a low bandwidth connection, that's why I'm looking at g729 and t.38. |
| 21:32.24 | iratik | ManxPower: sure? , awesome! i might just be getting the hang of this ... is there a way to reload extensions.conf without dropping all calls on the system |
| 21:32.28 | TrentCreek | ManxPower: so you mean a wire going between them? |
| 21:32.35 | ManxPower | I don't know if "ChanSpy(SIP/5121)" is valid or not (I don't use chanspy), but the exten:4 is will give you 5121 |
| 21:32.39 | Kobaz | iratik: dialplan reload |
| 21:32.49 | ManxPower | iratik: I think it is time to read The Good Book |
| 21:32.52 | *** join/#asterisk adr3nalin3 (n=afink@ip68-13-89-102.om.om.cox.net) |
| 21:33.10 | ManxPower | "extensions reload" will do it in 1.2, in 1.4 you may have to add "core" to the beginning of the command |
| 21:33.32 | talntid | How do I merge 2 wav files that were recorded? :\ |
| 21:33.36 | ManxPower | TrentCreek: "direct connection" could be a wire, a VPN tunnel, a data T-1 |
| 21:33.42 | Kobaz | 1in 1.4 extensions reload is depricated in favor of dialplan reload |
| 21:33.47 | ManxPower | talntid: try #sox |
| 21:33.59 | talntid | ah, sox. that's the name. i couldn't remember it for the life of me |
| 21:35.29 | iratik | hmm |
| 21:35.39 | iratik | extensions reload worked |
| 21:36.05 | TrentCreek | ManxPower: the * box is connected to a another LAN where it gets its connected to the internet. Inside the house here is another router that once was connected to the internet. ATA is connected inside to the router along with the * box also. |
| 21:36.07 | plik | don't sound so surprised |
| 21:36.36 | iratik | lol |
| 21:37.20 | Strom_C | iratik: just be aware that *555x. conflicts with NANP vertical service codes |
| 21:37.21 | ManxPower | TrentCreek: diagram it like SIP Phone -> Local LAN -> Asterisk -> NAT Router -> Internet |
| 21:37.24 | ManxPower | or however it is |
| 21:38.35 | TrentCreek | ManxPower: no... SIP Phone and Asterisk are on same Local LAN. |
| 21:39.00 | [TK]D-Fender | TrentCreek, No they're not, you're NAT'd across *2* class-C subnets |
| 21:39.04 | TrentCreek | And Asterisk Connected to another LAN via wireless connection getting internet |
| 21:39.27 | ManxPower | TrentCreek: DIAGRAM it like I did. |
| 21:39.41 | TrentCreek | ManxPower: no |
| 21:39.45 | [TK]D-Fender | ManxPower, SIP phones (subnet A) > NAT router > Asterisk (subnet B) > Internet |
| 21:39.48 | ManxPower | Then I can't help ypi/ |
| 21:39.50 | ManxPower | you. |
| 21:40.08 | [TK]D-Fender | ManxPower, SIP phones (subnet A) > NAT router > Asterisk (subnet B) > NAT router > Internet |
| 21:40.11 | [TK]D-Fender | ^^ rather |
| 21:40.11 | ManxPower | [TK]D-Fender: sounds like you can remove the nat router and renumber the local lan |
| 21:40.25 | TrentCreek | It is: SIP Phone -> Local LAN -> Asterisk -> NAT Router |
| 21:40.41 | TrentCreek | Asterisk -> NAT Router -> Internet |
| 21:40.49 | ManxPower | TrentCreek: and what is the IP address of the SIP phone and the Asterisk server |
| 21:40.59 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 21:41.04 | [TK]D-Fender | TrentCreek, You finished tell me over again that your phone was behind a router relative to * |
| 21:41.09 | ManxPower | This is such basic TCP/IP |
| 21:41.42 | ManxPower | TrentCreek: I assume you meant SIP Phone -> Local LAN -> Asterisk -> NAT Router -> Internet |
| 21:41.42 | TrentCreek | SIP: 192.168.1.101, Asterisk #1: 192.168.1.106, Asterisk #2 192.168.2.101 |
| 21:42.07 | ManxPower | TrentCreek: I did not see numbers on the Asterisk server in your diagram |
| 21:42.37 | talntid | Hm, Nobody in #Sox, and my attempt merged the files end-to-end. oops. :) |
| 21:42.38 | ManxPower | TrentCreek: Dude, you are trying to do one of the most difficult things in VoIP. Start being more accurate in your questions, answers, and information or you'll never get it working, |
| 21:42.48 | [TK]D-Fender | TrentCreek, and where is * #2 magically appearing from? |
| 21:43.05 | ManxPower | As it is as far as I can tell you keep telling us different things |
| 21:43.05 | TrentCreek | Not magical |
| 21:43.08 | TrentCreek | No |
| 21:43.23 | ManxPower | So diagram it accurately this time. |
| 21:43.29 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 21:43.53 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 21:43.53 | TrentCreek | A bit hard with single line text |
| 21:44.14 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:44.17 | ManxPower | Best of luck. I'm going back to paying work. |
| 21:44.23 | TrentCreek | SIP: Local LAN |
| 21:44.29 | generalhan | hey all ! |
| 21:44.32 | TrentCreek | Asterisk: Local LAN |
| 21:44.40 | acren | talntid: it's not sox -m wav1 wav2 out.wav ? |
| 21:44.46 | TrentCreek | Asterisk -> NAT -> Internet |
| 21:44.58 | TrentCreek | The * box has two cards |
| 21:45.13 | generalhan | anyone in here ever used the Asterisk-IM XMPP plugin ??? |
| 21:45.42 | talntid | do'h! typo'd. |
| 21:45.45 | talntid | thanks acren |
| 21:45.47 | TrentCreek | the 192.168.2.101 is connected to another NAT router getting ISP |
| 21:46.55 | *** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com) |
| 21:48.44 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.247) |
| 21:49.09 | *** join/#asterisk comprookie2000 (n=comprook@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 21:52.08 | dexpdx | Anyone got any suggestions of an "IVR engine" that would work well with asterisks |
| 21:52.13 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:52.32 | dexpdx | I don't want to be writing 10 dialplans a week for customers who want to change their voice prompts |
| 21:53.19 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 21:54.16 | ManxPower | dexpdx: I wrote a recording interface. |
| 21:54.28 | ManxPower | Customers can't change their dialplan, but they can record all their prompts |
| 21:55.07 | dexpdx | I just want something for me, to steamroll IVR's without having to clutter my dialplan |
| 21:55.32 | dexpdx | I suppose I could write something using DeadAGI + EAGI but I would think someone would have wrote something better already |
| 21:55.57 | ManxPower | dexpdx: uh, IVRs are so simple in Asterisk people here can write them in their sleep. |
| 21:56.03 | ManxPower | no need for silly AGIs, etc |
| 21:56.17 | ManxPower | There IS an external IVR application for Asterisk. |
| 21:56.38 | ManxPower | I'm sure you know that already as you did a "core show applications like IVR" in Asterisk CLI, right? |
| 21:56.40 | dexpdx | IVR's have always been busy work though |
| 21:57.27 | dexpdx | hmm ExternaIVR |
| 21:57.30 | dexpdx | first I've seen it |
| 21:57.50 | dexpdx | feels stupid |
| 21:58.28 | ManxPower | dexpdx: every major upgrade of Asterisk "core show applications", read the UPGRADE.txt for every release of Asterisk you can find. |
| 21:59.23 | dexpdx | In a manor of speaking I'm still really new to elements of asterisk |
| 21:59.46 | dexpdx | I got contracted about a year and a half ago to replace a crappy agi application with a good agi application |
| 22:00.01 | dexpdx | long story short I ended becoming this switches admin |
| 22:00.08 | dexpdx | s/switches/pbxs |
| 22:00.27 | dexpdx | and reelmented the entire application in func_odbc and AEL |
| 22:01.05 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 22:01.10 | acren | hehe, that's seems to be the way it goes. Make one little change in a sub aplication, then you own the whole thing ;) |
| 22:01.27 | dexpdx | Ok god this thing was shitty |
| 22:01.37 | ManxPower | 741/*/*-*+ |
| 22:01.38 | ManxPower | * |
| 22:01.41 | dexpdx | it would take channel control through DeadAGI |
| 22:01.46 | dexpdx | but then block()! |
| 22:01.47 | *** join/#asterisk RoyK (n=roy@ti0034a380-3704.bb.online.no) |
| 22:02.00 | dexpdx | + not multithreaded |
| 22:02.02 | dexpdx | nightmare |
| 22:02.30 | dexpdx | I ended rebuilding it from the ground up in Perl using Asterisk::AGI and then building my own FastAGI server because the one on CPAN didn't work |
| 22:03.05 | dexpdx | which reminds me I intended on starting a new branch in the Asterisk::* namespace and perl and reimplementing AGI as an api rather than just a parser |
| 22:03.42 | dexpdx | It would be nice to build applications that handled Manager and AGI api events like a HTTP MVC framework |
| 22:06.58 | Dossy | Hey, is there anything "special" I need to do with Asterisk to be able to receive faxes through it? Set up is VoIP DID to Asterisk to ATA that speaks T.38 connected to analog fax modem. |
| 22:07.44 | ManxPower | Dossy: Does the VOIP DID support T.38? |
| 22:07.58 | ManxPower | Dossy: also what version of Asterisk? |
| 22:08.08 | Dossy | Not sure. I was able to successfully send a fax _outbound_ through the VoIP provider ... |
| 22:08.11 | Dossy | Asterisk 1.4.19.2 |
| 22:08.39 | dexpdx | I was actually just reading about fax today |
| 22:08.43 | ManxPower | Dossy: if the provider does not support T.38 then you are not using T.38 |
| 22:08.50 | dexpdx | although I have zap channels |
| 22:09.40 | ManxPower | T.38 is a protocol for converting Fax into data at the PSTN/VOIP and VOIP/FAX Machine interfaces. |
| 22:09.50 | Dossy | ManxPower: okay, so assuming the provider _doesn't_ - would there be a reason why I can successfully _send_ faxes out, but not receive? |
| 22:10.05 | ManxPower | You could, in theory try using fax without T.38, but only if you don't have much jotter on your network and are using the ulaw or ala codecs. |
| 22:10.10 | dexpdx | fax detection |
| 22:10.18 | ManxPower | Dossy: without T.38 faxing will be hit and miss |
| 22:10.24 | Dossy | ManxPower: aha, ok. |
| 22:11.52 | ManxPower | Fortunatly I've found a way to make faxing work just as reliable as using POTS and the PSTN and a fax machine. |
| 22:12.16 | acren | Your ata may be switching to G711 when detecting the fax tones. At the moment I'm having trouble with an ata (audiocodes MP-112) set for T.38. The inbound call to the ATA starts off as G729, The ata sends a reinvite for T.38, but * is not passing it along to the other SIP gateway. |
| 22:12.37 | Dossy | oh, I have canreinvite=no ... heh |
| 22:12.40 | acren | manxPower: scan it an email ;) |
| 22:12.52 | ManxPower | acren: technically Asterisk does not pass ANYTHING between the two channels. |
| 22:12.57 | Dossy | Should I not do that? I also have t38pt_udptl=yes. |
| 22:13.20 | ManxPower | acren: a PSTN POTS line and an analog fax machine. No voip, no Asterisk, etc. |
| 22:13.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:13.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:13.37 | ManxPower | Dossy: T.38 is so new almost nobody knows anything about it. |
| 22:13.59 | Dossy | ManxPower: yeah, I have a PSTN POTS line now |
| 22:14.08 | Dossy | but, the line costs me $16/mo. |
| 22:14.13 | Dossy | I can get a VoIP DID for $2/mo :) |
| 22:14.20 | ManxPower | Dossy: I wonder why you did not find anything useful when you searched the mailing list archives |
| 22:14.33 | ManxPower | Dossy: the $12 difference will mean nothing if your fax does not work well |
| 22:14.44 | ManxPower | sorrt, $14 difference |
| 22:15.06 | Dossy | maybe I should turn off T.38 in the ATA, then. |
| 22:15.47 | ManxPower | Dossy: maybe you should do more reading about T.38 and Asterisk |
| 22:16.23 | Dossy | yeah. wiki isn't of much use |
| 22:16.33 | Dossy | wiki assumes fxo card it seems |
| 22:16.49 | plik | Dossy: feel free to bring it up to date with your research ;) |
| 22:16.53 | ManxPower | ~mailinglist |
| 22:16.54 | jbot | [mailinglist] Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 22:17.00 | Dossy | plik: oh, if I get this working, I definitely will |
| 22:17.11 | *** join/#asterisk watchy2 (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 22:17.15 | TrentCreek | ah yes...good ol T38 Air Force training fighter from the 70s ;-) |
| 22:17.22 | watchy2 | using this polycom alertinfo stuff |
| 22:17.29 | watchy2 | is it going to make everyphone show busy |
| 22:18.09 | ManxPower | watchy2: maybe because they ARE busy. |
| 22:18.26 | ManxPower | But that should not be an issue, as you should be able to make the calls rollover to the 2nd line appearance on the phone. |
| 22:19.35 | *** part/#asterisk geoff_k (n=geoff_k_@209.116.gr5.adsl.brightview.com) |
| 22:20.37 | watchy2 | yea |
| 22:20.47 | watchy2 | but wouldnt it make a 601's sidecar go insane? |
| 22:22.28 | ManxPower | watchy2: never has on any of my systems |
| 22:22.41 | ManxPower | We did have a problem with MeetMe. |
| 22:22.50 | watchy2 | but i mean |
| 22:23.00 | ManxPower | we shrunk the size of the paging group and that fixed it |
| 22:23.05 | watchy2 | when using the intercom feature of a polycom |
| 22:24.02 | ManxPower | watchy2: well the auto answer feature of the polycom, the polycom doesn't have an actual page/intercom feature -- that's all handled on the server |
| 22:24.38 | watchy2 | yea i know |
| 22:25.00 | watchy2 | but if i intercom to like 20 phones |
| 22:26.13 | watchy2 | its going to freak out like 20 extensions on the call thing |
| 22:26.33 | watchy2 | but im sure they will live |
| 22:26.50 | ManxPower | this is a copy from a live system: |
| 22:26.57 | ManxPower | PAGE_GROUP_1=Local/3920@corporate&Local/3921@corporate&Local/3922@corporate&Local/3923@corporate&Local/3924@corporate&Local/3925@corporate&Local/3926@corporate&&Local/3927@corporate&Local/3928@corporate&Local/3930@corporate&Local/3933@corporate&Local/3937@corporate&Local/3938@corporate&Local/3940@corporate&Local/3943@corporate&Local/3949@corporate&Local/3950@corporate |
| 22:27.02 | _ShrikE | I have seen polycoms crap badly when page groups get very large |
| 22:27.30 | ManxPower | You would have to patch Asterisk in order to get a line that long in the dialplan, BTW |
| 22:28.25 | ManxPower | _ShrikE: I've herd of reports of problems, but not experienced them myself |
| 22:28.30 | ManxPower | I think we ue SIP 2.1.1 |
| 22:28.38 | watchy2 | im using 2.2.0 i think |
| 22:28.53 | ManxPower | Might want to upgrade from the .0 release |
| 22:29.24 | watchy2 | im using 2.2.2 actually |
| 22:29.33 | watchy2 | you like the 330s manx? |
| 22:29.47 | watchy2 | small screen but they seem ok to me |
| 22:30.38 | ManxPower | watchy2: I've not used them. I think there are some deployed, but I deal more with the Asterisk side of it, there are people on site to deal with the physical stuff like beating users senseless and dumping them in the canal |
| 22:30.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:30.50 | watchy2 | haha |
| 22:31.08 | ManxPower | I do most of my work remote |
| 22:31.10 | watchy2 | i have to deal with it all, i work for a small company and im the only voip/phone guy |
| 22:31.28 | watchy2 | but i also got them by the nuts because in south arkansas who else you gonna find to do * |
| 22:32.06 | ManxPower | We found that paging was used much less once everyone started using DIDs |
| 22:32.06 | watchy2 | i deployed a phone system that cisco quoted $200k on for like $40k phones were the main cost |
| 22:32.23 | ManxPower | watchy2: anywhere near Texarkana? |
| 22:32.30 | watchy2 | 1:30 mins |
| 22:32.51 | watchy2 | i actually maybe putting in a huge voip system in texarkana |
| 22:32.55 | watchy2 | a insane install |
| 22:32.57 | ManxPower | After Katrina I lived in Atlanta Texas (near Texarkana) for a couple of months. |
| 22:33.08 | watchy2 | at an old army base |
| 22:33.54 | watchy2 | you contract out? |
| 22:33.54 | ManxPower | The local feed and farm supply store sold computers (and when I was in there the guy was trying to make Cisco phones work with Asterisk) |
| 22:33.54 | ManxPower | I grabbed toto and ran. |
| 22:33.55 | ManxPower | watchy2: /msg jbot manxpower |
| 22:34.22 | watchy2 | nice, this system is beyond my capability i think to a point |
| 22:34.55 | watchy2 | i would probably want a backup there, i do */polycom only but i need someone to guide me through a large install |
| 22:35.20 | watchy2 | cisco quoted $220k |
| 22:35.21 | *** join/#asterisk RoyK (n=roy@ip-122-19-149-91.dialup.ice.no) |
| 22:35.32 | ManxPower | My largest install has at least 90 polycoms, PRI, tellabs echocancel |
| 22:35.48 | watchy2 | this is going to be 60 polys i think |
| 22:35.52 | watchy2 | 24 lines if i remember |
| 22:36.02 | watchy2 | if i can get the quote for cisco mind if i send it to your email |
| 22:36.05 | ManxPower | 23 lines if you are lucky |
| 22:36.35 | watchy2 | what would you charge for a 60 phone pri system? |
| 22:36.40 | watchy2 | just a ball park |
| 22:37.03 | ManxPower | watchy2: As much as I thought I could. |
| 22:37.11 | watchy2 | yea i agree |
| 22:37.17 | watchy2 | i think i can get $100k+ |
| 22:37.25 | ManxPower | And I can't give ANY kind of estimate until we decide who will be doing what. |
| 22:37.48 | *** join/#asterisk marc7 (n=marc@s207-6-221-31.bc.hsia.telus.net) |
| 22:37.54 | watchy2 | well if i email you the quote dont steal my customer |
| 22:37.55 | watchy2 | :( |
| 22:37.57 | Strom_M | the classic mistake of charging by the job ad not by the hour |
| 22:38.00 | Strom_M | s/ad/and/ |
| 22:38.05 | ManxPower | Installing Asterisk and polycoms are so simple I could do it in my sleep. Figuring out what the customer wants when the customer doesn't even know what the want -- THAT is what takes the time. |
| 22:38.21 | jjshoe | I've done a 150 phone install :) |
| 22:38.33 | jjshoe | not counting the 100 or so softphones |
| 22:38.34 | watchy2 | i agree, im still changing things for a customer who still isnt %100 sure what they want a year later |
| 22:38.37 | ManxPower | My rates is $120/hr base rate. Discounts of up to 20% are available. |
| 22:39.59 | watchy2 | well id be willing to profit share on the sell for sure |
| 22:40.33 | ManxPower | watchy2: e-mail me when you decide/figure out what you would want a consultant to do in this project. |
| 22:40.46 | watchy2 | yea will do |
| 22:40.54 | watchy2 | i gotta find out if the company i work for even wants to do it |
| 22:41.12 | watchy2 | if not im going to turn it over to someone i know |
| 22:41.28 | watchy2 | probably someone here, i dont know anyone in real life with * exp i trust |
| 22:41.54 | watchy2 | and i know the folks that idle here know whats up |
| 22:41.58 | watchy2 | well some |
| 22:41.59 | ManxPower | Some people think I'm an Asterisk Guru, others think I'm just an asshole, but most seem to think both things at once. |
| 22:42.11 | ManxPower | I'm very good at the areas that I do work in. |
| 22:42.16 | jjshoe | irc, realtime slashdot (tm) |
| 22:42.43 | watchy2 | im trying to learn everything i can, lately ive learned quite a bit about polycoms |
| 22:43.29 | ManxPower | I've been doing Asterisk installs since before 1.0 |
| 22:43.33 | ManxPower | whenever that was. |
| 22:43.41 | russellb | 1.0 was Fall of 2004 |
| 22:43.42 | watchy2 | wow |
| 22:43.49 | ManxPower | russellb: what about 0.65? |
| 22:43.53 | russellb | no idea |
| 22:44.10 | ManxPower | 0.65 is when we started our first production prototypes |
| 22:44.55 | *** join/#asterisk km2 (n=x@c-24-23-252-175.hsd1.ca.comcast.net) |
| 22:45.43 | russellb | 0.7.0 appears to have been January 2004 ... |
| 22:46.25 | russellb | 0.5.0 was September 2003 |
| 22:49.25 | seanbright | warranty company decided to reimburse me for my transmission! |
| 22:49.34 | seanbright | is paying approximately *squat* |
| 22:49.36 | seanbright | woo |
| 22:49.54 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 22:49.58 | seanbright | sorry, OT :) |
| 22:49.59 | Nasra | ManxPower: why you write a book? |
| 22:50.43 | Nasra | should say write a book |
| 22:58.09 | riddlebox | I have a fxo port with a line in it, and a modem connected to a fxs port, if I use that modem I connect at 12k per sec but if I go straight to the line I get 32k per second? |
| 22:58.44 | Strom_M | riddlebox: analog-digital conversions |
| 22:59.37 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:02.40 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
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| 23:07.39 | *** part/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 23:07.53 | ManxPower | Nasra: Other people, better at many of the things I'm good at, some experts in their own areas of Asterisk already wrote a book. |
| 23:13.15 | clickonce | feels sad |
| 23:18.05 | riddlebox | Strom_M, the connections are all on a TDM card in the server |
| 23:21.20 | Strom_C | riddlebox: yes, I know |
| 23:21.25 | Strom_C | the ports are still analog. |
| 23:21.30 | Strom_C | the switching is digital/ |
| 23:24.59 | drmessano | I am gonna write an asterisk book |
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