| 00:00.03 | rhombus | JT: Is that the Mexican? I thought you were referring to one who is still alive |
| 00:00.03 | JT | rhombus: right |
| 00:00.19 | rhombus | JT: the real answer is sorta |
| 00:00.25 | JT | mmkay |
| 00:00.36 | antimoof | viva zapata! |
| 00:00.56 | rhombus | JT: meaning only that it will meet the needs of anybody who needs zapata |
| 00:01.04 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 00:01.06 | MrTelephone | hi |
| 00:01.10 | rhombus | JT: and the result will be much better |
| 00:01.33 | Mad|Cow | Is there any way to register the same extension on the PBX several times? I have two phones in different offices, I want them both to ring (on the same extension) but I also want them to register with Asterisk as the same username. I played around with SER which can do this. The only work around I have found with Asterisk is to have my dial plan ring different extensions when the caller calls the primary extension. Any ideas? |
| 00:01.46 | JT | rhombus: i like that idea |
| 00:01.58 | JT | rhombus: it's good not being tied to zaptel |
| 00:01.58 | MrTelephone | madcow, u do it with your dialplan |
| 00:02.03 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 00:02.04 | rhombus | JT: It's a great idea. I think this card is going to kick major ass. I have to run -- but I will revisit this topic later |
| 00:02.06 | JT | but it's also good to have as an option |
| 00:02.07 | ManxPower | Mad|Cow: You do NOT want your SIP USER ID to be the same as your EXTENSION. They are two totally different things. |
| 00:02.15 | rhombus | see you boys later. |
| 00:02.17 | *** part/#asterisk rhombus (i=user239@74.12.124.179) |
| 00:02.28 | Mad|Cow | MrTelephone: yeah... thats what i thought :-( |
| 00:02.43 | [hC] | ManxPower: I tried setting it to both unavailable and prohib, it didnt seem to help. |
| 00:02.45 | MrTelephone | madcow exten => 222,1,Dial(SIP/phone1&SIP/phone2) |
| 00:02.47 | ManxPower | You do not "ring extensions" in Asterisk. You ring devices. |
| 00:03.06 | ManxPower | [hC]: you have to work with your carrier. |
| 00:03.07 | MrTelephone | will ring both phones, first one who picks up, wins |
| 00:03.15 | Mad|Cow | ManxPower: sorry... incorrect use of terminology. |
| 00:03.30 | ManxPower | We set the SIP userid to be the MAC address of the device. |
| 00:03.39 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 00:03.48 | ManxPower | well the MAC with a -a -b -c etc on it for each line appearance. |
| 00:04.31 | MrTelephone | why manxpower? |
| 00:04.35 | MrTelephone | why havemore than one line appearance |
| 00:05.02 | ManxPower | MrTelephone: because then *I* control how calls roll over to different line appearances |
| 00:05.18 | MrTelephone | right |
| 00:05.30 | ManxPower | On many phones, the first line appearance is the person's DID, the 2nd and 3rd line appearances are the main DID for the office. |
| 00:05.35 | MrTelephone | so if appearance 3 rings on different criteria |
| 00:05.49 | MrTelephone | i see what your saying |
| 00:06.18 | ManxPower | MrTelephone: I have all the gory details of this wrapped up in a macro. You just Set() a couple of variables and then run the macro. |
| 00:06.36 | MrTelephone | how do your clients like that? |
| 00:06.45 | ManxPower | How do you mean? |
| 00:07.02 | MrTelephone | having all thephones ring if someone doesn't pick an extension |
| 00:07.24 | ManxPower | MrTelephone: Oh, I think it is totally STUPID, but if the client wants it, the client gets it. |
| 00:07.35 | ManxPower | Also there are some clients that have 1 phone shared between 2 people. |
| 00:07.42 | MrTelephone | i really really wish there was a way to group sip devices into a group that has a call limit |
| 00:07.43 | ManxPower | so each person gets their own line apperarance |
| 00:08.11 | MrTelephone | manxpower, i know its hard to get people used to the no line appearance |
| 00:08.41 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 00:08.50 | ManxPower | MrTelephone: I force them |
| 00:09.08 | MrTelephone | did u see my really really wish sentence? you think thats possible to do in the dialplan |
| 00:09.09 | ManxPower | But there are MANY reasons to be able to control each line appearance seperatly |
| 00:09.30 | ManxPower | MrTelephone: Are you famialiar with the GROUP_ variables? |
| 00:10.00 | MrTelephone | when i started using asterisk i didn't remember seeing them |
| 00:10.04 | MrTelephone | should I look that up then? |
| 00:10.12 | ManxPower | look in README.variables |
| 00:10.26 | ManxPower | We never use call limits. |
| 00:10.43 | ManxPower | We don't limit outgoing calls and for incoming calls to phones, we just turn off call waiting on the phone |
| 00:11.27 | ManxPower | MrTelephone: Also look on the Wiki and the mailing list archives. The whole GROUP thing is massivly confusing. |
| 00:11.48 | ManxPower | MrTelephone: Do you know about setvar=examplevariable=examplecontents in sip.conf? |
| 00:12.04 | MrTelephone | its hard when I do the billing-- I want to charge as if it was one FXS |
| 00:12.08 | MrTelephone | i mean FXO |
| 00:18.14 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:18.14 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- Flame suits required... |
| 00:18.25 | MrTelephone | i got this dhcp server right now and it doesn't have the option in the bios to startup after power failure |
| 00:18.33 | MrTelephone | and its a single point of failure |
| 00:18.35 | MrTelephone | totally sucks |
| 00:18.41 | ManxPower | get a different server |
| 00:19.10 | MrTelephone | thats like job #1,353,405 |
| 00:19.30 | kiscokid | option tftp-server-name and option tftp-server are syntax errors |
| 00:19.35 | ManxPower | MrTelephone: only until it fails and you are out of town and nobody is answering their cell phone |
| 00:19.38 | kiscokid | next-server doesn't work |
| 00:19.43 | MrTelephone | one sec |
| 00:19.58 | ManxPower | kiscokid: what phone are you using? |
| 00:20.07 | kiscokid | Aastra 480i |
| 00:21.09 | MrTelephone | next-server 69.71.79.164; |
| 00:21.28 | kiscokid | that seems to be ignored by the phone |
| 00:21.41 | ManxPower | kiscokid: perhaps that is for a different DHCP option |
| 00:21.59 | ManxPower | I know that it works in the default Polycom setup |
| 00:22.04 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 00:22.33 | Mad|Cow | Other than using VMWare (or a simular virtulization software) has anyone had any success hosting several PBX's from the same physical hardware? |
| 00:22.35 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-67216c1ecb112992) |
| 00:23.02 | ManxPower | Mad|Cow: I do it everyday. |
| 00:23.13 | ManxPower | It is called "contexts" in Asterisk |
| 00:23.24 | ManxPower | They allow you to build multi-client PBXS. |
| 00:23.27 | Mad|Cow | ManxPower: ahhh.. thats what i though |
| 00:23.39 | ManxPower | the only significant issue that I know of is for call parking, which does not support contexts |
| 00:23.58 | JT | PBXes :D |
| 00:24.06 | JT | Mad|Cow: vmware is too slow |
| 00:24.16 | JT | Mad|Cow: something like Xen of openvz may work |
| 00:24.21 | Mad|Cow | ManxPower: what if you have two clients with the same extensions? how do you dial them? |
| 00:24.22 | JT | not as easy to setup though |
| 00:24.36 | tzafrir_laptop | Xen and openvz are quite different |
| 00:24.39 | shido6 | if the two clients are in different contexts it doesnt matter :) |
| 00:24.39 | ManxPower | Mad|Cow: that works just find, they just won't be able to call each other |
| 00:25.01 | JT | tzafrir_laptop: yes, they are quite different. |
| 00:25.03 | Mad|Cow | JT: never used Xen or openvz. Are they any good? |
| 00:25.07 | tzafrir_laptop | and openvz would be much nicer for Asterisk |
| 00:25.21 | tzafrir_laptop | or linux-vserver |
| 00:25.24 | JT | Mad|Cow: yes, if you want to do virtualisation with any performance |
| 00:25.30 | JT | Mad|Cow: KVM is another one, too |
| 00:26.22 | tzafrir_laptop | it's still basically in the same class of Xen and qemu: you partition all of the resources (except, maybe, CPU time) between the different hosts |
| 00:28.32 | Mad|Cow | ManxPower: So lets say your hosting two PBX's on the same box. Company A has an extension 1234 and Company B has an extension 1234. When the user dials 1234 on CompanyA's system, I assume the device you ring is 1234CompanyA? |
| 00:29.24 | ManxPower | Mad|Cow: Is the user in Company A or Company B? |
| 00:29.44 | Mad|Cow | The user is in Company A |
| 00:30.00 | ManxPower | Then the call will match exten => 1234 in Company A's context |
| 00:30.16 | *** join/#asterisk bkruse_home (i=kruz@nat/digium/x-09c8d35bfd1bc9ab) |
| 00:30.31 | ManxPower | Mad|Cow: What you do NOT want to do is set the SIP userid in sip.conf and on the device to be the same as the extension. |
| 00:30.39 | ManxPower | We set them to be the MAC of the device |
| 00:31.22 | ManxPower | anyway I think I'm going to go to the bar. |
| 00:31.37 | Mad|Cow | ManxPower: hehehe... nice. thanks for the info |
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| 00:33.56 | ServerGod | Hi all, i have an HA cluster of two 1.4 * boxes, i can get to the shared ip and the one that is the master, but the standby i cant get to the web interface. If i ssh to the box, it lets me log in. Sync stat shows all is good (consistant). I go to the primary box and i can go to the asterisk CLI. the slave, not so much :Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?). and amportal wont start it either. ha.d file is fin |
| 00:34.37 | tzafrir_laptop | are you using safe_asterisk with a HA setup? |
| 00:35.06 | blitzrage | I don't think HA starts those services until it detects a failure on the primary |
| 00:35.32 | JT | ServerGod: sorry, doesn't sound like an asterisk question |
| 00:35.35 | JT | ~freepbx |
| 00:35.36 | jbot | freepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 00:35.54 | tzafrir_laptop | safe_asterisk may interact quite badly with HA setups |
| 00:35.56 | ServerGod | no, i saw an article about that though. |
| 00:36.17 | Mad|Cow | Anyone know why ManxPower was saying he uses the MAC address of the device for the SIP userid? |
| 00:36.22 | JT | ServerGod: "web interface"? |
| 00:36.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 00:36.46 | JT | Mad|Cow: because having the sip peer being the same name as the extension is bad practice |
| 00:37.21 | ServerGod | there are three ip's all have astgui lets say .106 .107 and .108 |
| 00:37.44 | ServerGod | 108 is also 10.0.0.2 and 107 is 10.0.0.1 106 is the virtual ip |
| 00:37.52 | Mad|Cow | JT: gotcha |
| 00:38.06 | ServerGod | i was able to get to the web gui on all three |
| 00:38.18 | ServerGod | now the .108 is toast |
| 00:38.25 | tzafrir_laptop | does that web gui require asterisk to be up? |
| 00:38.53 | ServerGod | no, there is also hylafax and a2zbilling and unified messaging there also |
| 00:39.14 | ServerGod | just a jumpstart page |
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| 00:43.42 | tzafrir_laptop | ServerGod, we can help you with Asterisk. Not with some unnamed GUI |
| 00:47.17 | ServerGod | slightly modded asterisk 1.4.2 gui |
| 00:48.03 | ServerGod | i stared off with the adminsparadise bundle and did a little work to it. |
| 00:48.16 | ServerGod | *started |
| 00:52.17 | JT | ServerGod: asterisk doesn't come with a gui |
| 00:52.32 | *** join/#asterisk theacolyte_ (n=theacoly@unaffiliated/theacolyte) |
| 00:53.09 | theacolyte_ | How do most people handle multiple offices? VPN links? |
| 00:58.47 | x86 | theacolyte_: point to point or frame relay traditionally... nowadays MPLS is the hip thing |
| 00:59.07 | x86 | theacolyte_: VPN over the internet is highly unreliable, in my experiences |
| 00:59.16 | theacolyte_ | Hmmm |
| 00:59.17 | theacolyte_ | Got it |
| 00:59.22 | theacolyte_ | So 1 server for the whole org |
| 00:59.39 | theacolyte_ | New company I'm going to does some outsourced VOIP stuff, but may want to switch to * soon |
| 00:59.49 | theacolyte_ | Interoffice calling is the only thing I worry about |
| 01:00.10 | theacolyte_ | they're gonna wanna use their existing T1's though |
| 01:00.15 | x86 | wait, no one said anything about a single asterisk server.. |
| 01:00.23 | theacolyte_ | oh |
| 01:00.25 | x86 | that's just silly ;) |
| 01:00.28 | theacolyte_ | So 1 per site? |
| 01:00.35 | x86 | you need at least 2 for redundancy |
| 01:00.37 | theacolyte_ | Yeah - also, all calls will be recorded |
| 01:00.42 | x86 | 1 per site is _ideal_ |
| 01:00.55 | theacolyte_ | Oh, I see what your'e saying |
| 01:01.00 | x86 | you doing all outbound / inbound calling over dedicated voice T1's? |
| 01:01.03 | x86 | or PRI's? |
| 01:01.06 | theacolyte_ | PRI |
| 01:01.16 | x86 | PRI at each site? |
| 01:01.18 | theacolyte_ | Yeah |
| 01:01.24 | theacolyte_ | Most likely |
| 01:01.34 | x86 | then how would you get away with a single asterisk server anyway? :) |
| 01:01.40 | theacolyte_ | Very true |
| 01:01.46 | theacolyte_ | I'm about 45 minutes into considering this stuff hehe |
| 01:01.52 | x86 | ah |
| 01:02.24 | x86 | what I do, is have an asterisk server at each branch, with a CAS T1 running into it (we're an outbound call center, no need for PRI) |
| 01:02.30 | theacolyte_ | I suggested going CCM to be honest... but they don't have 500k sitting around |
| 01:03.01 | x86 | then, if the T1 gets saturated, I failover to another branch, or to the main office (with 3 T1's) until the call is finally routed out |
| 01:03.11 | theacolyte_ | Got it... I'd do something similar |
| 01:03.18 | theacolyte_ | How many stations at each site? |
| 01:03.23 | x86 | so it's redundant in the fact that it will always find a path out, no matter if it's local or via another branch |
| 01:03.41 | x86 | it varies... I would say about 30 on average |
| 01:03.48 | x86 | we have one office that has 50 |
| 01:04.07 | theacolyte_ | I'm looking at about 80-60-60-30 |
| 01:04.12 | x86 | which was the main reason for setting the failover up, because the 50 user office always has a pegged T1 |
| 01:04.16 | theacolyte_ | Right |
| 01:04.21 | JT | pegged? |
| 01:04.26 | sevard | x86! what's up man |
| 01:04.29 | x86 | 24 concurrent calls ;) |
| 01:04.33 | x86 | sevard: hey man, ltns ;) |
| 01:04.46 | sevard | same bro, how you been |
| 01:04.48 | JT | full? right |
| 01:04.58 | theacolyte_ | Yeah, pegged = saturated = 100% capacity |
| 01:05.08 | x86 | theacolyte_: drop 3 T1's into the 80 office, 2 T1's into the 60 offices, and 1 T1 into the 30... you'll be fine |
| 01:05.22 | x86 | JT: 24 channels is the maximum on a CAS T1, yes |
| 01:05.23 | JT | weird slang for saturated |
| 01:05.33 | x86 | no, it's rather common here |
| 01:05.34 | theacolyte_ | x86: are you also counting in regular data traffic? |
| 01:05.37 | JT | x86: i realise, i had no idea what you meant by pegged |
| 01:05.44 | JT | x86: yes, there, hence weird slang |
| 01:05.45 | theacolyte_ | pegged is a very common word |
| 01:05.46 | x86 | JT: he did ;) |
| 01:05.52 | sevard | yeah dude, get with the lingo. |
| 01:05.54 | JT | theacolyte_: maybe where you live |
| 01:05.56 | x86 | hehe |
| 01:06.01 | theacolyte_ | He didn't get the memo |
| 01:06.11 | sevard | i bet he isn't even filing his TPS reports |
| 01:06.17 | x86 | JT: i was talking to him, so it doesnt really matter if someone else knows what I'm talking about, as long as he does... |
| 01:06.23 | x86 | sevard: haha |
| 01:06.23 | JT | i am, just with the old cover sheet |
| 01:06.30 | theacolyte_ | x86: 3 T1's with 80 users + data traffic? |
| 01:06.42 | x86 | theacolyte_: no man, 3 T1's of voice.... |
| 01:06.45 | theacolyte_ | I mean... I guess I could set up QoS |
| 01:06.49 | JT | x86: actually it does, because there's PMs if you want to have a conversation that doesn't involve channel participants :) |
| 01:06.51 | x86 | theacolyte_: 3 PRI's ;) |
| 01:06.53 | theacolyte_ | Oh |
| 01:06.58 | theacolyte_ | I see what you're saying |
| 01:07.05 | theacolyte_ | I'm talking about just the VOIP part for interoffice |
| 01:07.12 | theacolyte_ | But I guess that's not too much traffic anyway |
| 01:07.19 | x86 | JT: no, it doesnt matter to me if you know what i mean when i'm talking directly to someone else... sorry ;) |
| 01:07.36 | x86 | theacolyte_: are you a call center? |
| 01:07.38 | theacolyte_ | Yeah, in this case I'll have to get pretty close to 1:1 with PRI/Stations |
| 01:07.52 | theacolyte_ | x86: basically yes, but lots of inbound too |
| 01:08.08 | JT | x86: well that's pretty antisocial to be honest |
| 01:08.15 | x86 | i would say oversaturate the biggest office, so in a worst case scenario, your other branches can fail over to it |
| 01:08.20 | theacolyte_ | right |
| 01:09.12 | theacolyte_ | Pretty neat stuff all around |
| 01:09.28 | theacolyte_ | Current place uses Altigen, new place currently outsources to Covad |
| 01:09.47 | theacolyte_ | Been waiting to do a * deployment |
| 01:10.13 | Hmmhesays | so who wants to help me with a regex |
| 01:10.40 | x86 | Hmmhesays: with perl? |
| 01:10.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 01:10.47 | Hmmhesays | pcre yeah |
| 01:10.53 | Hmmhesays | I need to pull the value out of an html tag |
| 01:11.02 | x86 | theacolyte_: you can always drop VoIP T1's in |
| 01:11.20 | JT | x86: as i said, there are private messages that can be used if you don't want anyone else to see/comment on a message, heh |
| 01:11.23 | Hmmhesays | <input type="text" value="this_is_variable_string_I_need_to_get"> |
| 01:11.29 | x86 | theacolyte_: there are providers here doing data T1's dedicated to SIP, promising 60 concurrent calls on a single T1 |
| 01:11.46 | theacolyte_ | x86: Yeah but I'm if I understand them correctly, they don't want to oursource anything like that -- meaning just a straightforward super reliable PRI |
| 01:11.54 | theacolyte_ | current issue == reliability |
| 01:11.55 | x86 | JT: you're still on that? i dropped that conversation a good 5 minutes ago ;) |
| 01:12.19 | x86 | theacolyte_: i hear ya... they're the same way where I am |
| 01:12.21 | theacolyte_ | x86: lol I was just about to say that myself |
| 01:12.27 | Hmmhesays | anyone anyone? ;) |
| 01:12.59 | theacolyte_ | What's nice is they already own all the phones (Cisco 7960's) |
| 01:13.02 | x86 | theacolyte_: i wanted to replace all the point to point links with MPLS, and drop a DS3 right in the middle for voice, and another DS3 for Internet... they wouldn't budge |
| 01:13.05 | JT | eww, ciscos |
| 01:13.20 | JT | and 60 sip calls on a T1, would sound pretty awful |
| 01:13.32 | x86 | JT: g729 nub ;) |
| 01:13.36 | Hmmhesays | depending on the voice codec used |
| 01:13.41 | JT | yes G.729 sounds awful. |
| 01:13.44 | theacolyte_ | x86: $$ no doubt. I looked at a frac DS3 (20/mbit) and it was $5k/mo |
| 01:13.45 | JT | compared to G.711 |
| 01:13.49 | theacolyte_ | and that's out here in the bay area |
| 01:13.49 | JT | it's not terrible |
| 01:13.53 | JT | but it's not that great |
| 01:13.57 | x86 | JT: yeah but for the cost per channel you can't beat it |
| 01:14.02 | Mad|Cow | JT: I've been playing around with my SIP username since you and Max said I should be using the MAC address. How do I get my phone to register as the MAC address but yet still work with my dialplan (so when I dial the extension, it rings the phone)? They only way I can figure out how to set this up, I would have to dial the MAC in my dial plan. |
| 01:14.13 | [hC] | Imho, the difference in quality is not that great. |
| 01:14.14 | x86 | theacolyte_: the major costs with a DS3 is the loop |
| 01:14.18 | theacolyte_ | yup |
| 01:14.30 | theacolyte_ | data was like.... 1/6 of the cost |
| 01:14.37 | x86 | theacolyte_: the advantage of doing MPLS and dropping a DS3 in the middle is that you only pay a cross connect at the CO, no loop charge :) |
| 01:14.43 | JT | x86: whenevr i call a a callcentre and hear G.729 encoded audio i think "this company are stingey idiots" |
| 01:14.43 | [hC] | i mean yeah g729 sounds compressed, of course, but its not that bad compared to g711. unless you're REALLY listening for it, most people dont notice. |
| 01:14.57 | theacolyte_ | interesting.... |
| 01:15.04 | x86 | [hC]: exactly |
| 01:15.14 | x86 | g723 is also nice |
| 01:15.32 | JT | [hC]: it's quite good for the bandwidth usage, but the audio is still poor, especially with VAD |
| 01:15.34 | x86 | speex and gsm aren't terrible either, but more bandwidth |
| 01:15.44 | theacolyte_ | to me g729 always sounds robotic... I run g711 on my current system |
| 01:15.55 | JT | g.723 has poor patent licensing policies |
| 01:16.01 | x86 | JT: i agree with you there... VAD is horrible |
| 01:16.02 | [hC] | speex to me sounds horrible |
| 01:16.07 | [hC] | gsm usually sounds horrible too |
| 01:16.10 | theacolyte_ | tried g729 though... have a few locations that only have a single t1 |
| 01:16.13 | theacolyte_ | speex is OK |
| 01:16.53 | JT | most people over 30 who don't work in telecomms can't tell G.711 and G.723 apart |
| 01:16.58 | JT | G.729 |
| 01:17.00 | JT | i meant |
| 01:17.13 | [hC] | judging that most 'shitty audio' experiences come from the quality of the phone the other person is using, g729 doesnt have a big enough noticable difference to me to bother wasting so much more bandwidth over. |
| 01:17.24 | *** join/#asterisk ukris (n=ukris@aa20060807547d355914.userreverse.dion.ne.jp) |
| 01:17.26 | JT | but i'm sure most people could tell them apart if they listened for it |
| 01:17.35 | JT | but these days people are accustomed to shitty audio |
| 01:17.38 | JT | with bad POTS |
| 01:17.41 | JT | and cellphones |
| 01:17.41 | [hC] | well you can tell a lot of things if you're actively trying to compare.... anything.. |
| 01:17.56 | [hC] | yep. |
| 01:18.16 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 01:18.34 | JT | i'll not use G.729 as it is clearly noticable when you are connecting to the PSTN to a business phone system on BRI/PRI |
| 01:18.44 | JT | and business clients can be fusy |
| 01:20.10 | JT | rightly so imho |
| 01:20.22 | [hC] | i missed the conversation earlier on wether or not anyone had tried SLA in asterisk 1.4 and if it was worth giving a shot |
| 01:20.26 | JT | "landline" calls should sound like they are landlines ;) |
| 01:20.52 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 01:21.04 | [hC] | it makes it kinda hard to offer a price break when you use ulaw, since people have to buy internet connections capable of handling the number of calls they need in ulaw, and pay more from an ITSP cause it costs them more per channel now. |
| 01:21.04 | theacolyte_ | x86: well, thanks for the pointers, one last question for ya: do you know of any good * best practices guides? |
| 01:21.39 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 01:21.56 | JT | [hC]: i think quality is worth a bit extra |
| 01:22.07 | JT | i won't use an ITSP that disallows G.711 |
| 01:24.37 | JT | [hC]: i think in a lot of markets using VoIPoI over a telco is not economical unless there's added value of some sort |
| 01:26.04 | bkw_ | JT then don't use your cellphone |
| 01:26.30 | bkw_ | GSM/AMR vs ERVC |
| 01:26.32 | JT | bkw_: i don't if i am at a location with a desk phone |
| 01:26.44 | bkw_ | honestly AMR is awesome |
| 01:27.00 | JT | it's not bad |
| 01:27.02 | bkw_ | ERVC is too but its a bit heavy on cpu |
| 01:27.16 | JT | amr is much nicer than gsm |
| 01:27.16 | x86 | theacolyte_: the book ;) |
| 01:27.19 | x86 | ~thebook |
| 01:27.20 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:27.28 | x86 | the best guide there is |
| 01:27.32 | theacolyte_ | excellent |
| 01:27.33 | theacolyte_ | thanks again |
| 01:27.44 | JT | but not all of the things in the book are best practice |
| 01:27.47 | theacolyte_ | oh, even better... it's free |
| 01:27.48 | JT | some things are out of date |
| 01:27.51 | JT | keep that in mind |
| 01:28.01 | theacolyte_ | k |
| 01:28.01 | JT | yes but if you like it, you should consider buying it |
| 01:28.11 | *** join/#asterisk MdeP (n=mdep@204-87-22-190.adsl.tie.cl) |
| 01:28.12 | JT | second edition is meant to be released "any month now" |
| 01:28.54 | `Sean | i wish devel's would work on better fax support |
| 01:28.55 | `Sean | :( |
| 01:29.04 | `Sean | they seem soo lazy on that module specificly |
| 01:29.08 | JT | doesn't seem to be a priority for them |
| 01:30.45 | blitzrage | JT: actually it is -- there is 2-3 months of production required where we don't have much of a say of what is going on. Plus, we spent about 2-3 extra months writing more stuff into this edition than was originally intended (I wrote a new chapter on func_odbc that was not originally planned). We are already past QC1 (the last time the authors are allowed to make changes), and we've just approved the index |
| 01:31.06 | tzanger | I bet that was a fun job |
| 01:31.09 | tzanger | going through the index |
| 01:31.22 | blitzrage | Plus we used docbook which O'Reilly doesn't use very often, so it took them longer to get it into Frame Maker |
| 01:31.32 | JT | blitzrage: wasn't it originally slated for release in june? |
| 01:31.33 | blitzrage | tzanger: ya, it was a blast |
| 01:31.47 | blitzrage | JT: not sure... I don't think we ever said when we would release it |
| 01:31.53 | blitzrage | I've been saying Aug/Sept |
| 01:32.03 | JT | oh, well that's what amazon said on it earlier this year iirc |
| 01:32.21 | blitzrage | and like I said, we were just supposed to update it for 1.4, and we wrote a lot more stuff and did a lot of work on the appendices |
| 01:32.25 | JT | maybe o'reilly gave some promises |
| 01:33.22 | blitzrage | JT: don't listen to that... it's just a random date that they "think" it could be released. The Cookbook was supposed to be release Aug. 1st according Amazon too. It's just marketing and it's hard to control when the authors actually release things. Plus, we certainly don't get rich doing this, and we all have real day jobs. |
| 01:33.52 | JT | blitzrage: so that's probably o'reilly's doing? |
| 01:34.19 | blitzrage | it's the marketing department at O'Reilly just picking a 6-8 month date from the time of signing the contract |
| 01:34.25 | Juggie | blitzrage, is the cookbook even close to done? |
| 01:34.33 | JT | nice |
| 01:34.42 | blitzrage | Juggie: www.asteriskcookbook.com <-- you tell me |
| 01:34.59 | JT | what's the cookbook meant to be? |
| 01:35.23 | blitzrage | whatever the community wants (although I have an outline of how I want to build it) |
| 01:35.24 | Juggie | blitzrage, no? :) |
| 01:35.28 | blitzrage | Juggie: :) |
| 01:35.34 | bkw_ | blitzrage, you going to be at Astricon? |
| 01:35.39 | blitzrage | bkw_: I hope so |
| 01:35.44 | bkw_ | I'll see you there then |
| 01:35.49 | blitzrage | nice nice |
| 01:36.07 | bkw_ | anyone here going to speechtek? |
| 01:36.46 | blitzrage | JT: hopefully next time you'll realize it is a priority for us ;) |
| 01:36.56 | blitzrage | some things are out of our hands though |
| 01:37.12 | JT | blitzrage: realise what is a priority? |
| 01:37.47 | blitzrage | <JT> second edition is meant to be released "any month now" |
| 01:37.47 | blitzrage | <JT> doesn't seem to be a priority for them |
| 01:38.16 | blitzrage | anyways... it should be released "any week now!" :) |
| 01:38.50 | JT | blitzrage: |
| 01:39.04 | JT | < `Sean> i wish devel's would work on better fax support |
| 01:39.05 | JT | < JT> doesn't seem to be a priority for them |
| 01:39.15 | blitzrage | ahhhhhhhhhhhhhhhhhhhhhh |
| 01:39.16 | blitzrage | stupid IRC |
| 01:39.17 | JT | confusion ;) |
| 01:39.23 | blitzrage | that makes sense then lol |
| 01:39.28 | blitzrage | my apologies |
| 01:39.41 | JT | it certainly sounds like writing TFOT 1 IS a priority |
| 01:39.44 | JT | 2 |
| 01:39.45 | JT | even |
| 01:39.47 | JT | :P |
| 01:39.58 | `Sean | blitzrage may i ask when t38 or other fax components will be a priorty for asterisk? |
| 01:40.24 | [TK]D-Fender | `Sean, Go ask Satan about that snow-blower ;) |
| 01:40.27 | blitzrage | `Sean: as soon as someone with the skills to wrote those components comes forward to write them probably |
| 01:40.40 | blitzrage | or someone sponsors such an effort |
| 01:40.54 | bkw_ | `Sean, why not find someone to sponsor it? |
| 01:40.57 | blitzrage | writing that stuff is quite expensive since the skills to write that kind of code well is fairly rare (thus expensive) |
| 01:41.06 | *** join/#asterisk djPepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 01:41.11 | `Sean | are you saying the current asterisk dev's do not have the skills neccesary to fix the fax problems with asterisk :P? |
| 01:41.21 | bkw_ | `Sean, what do you plan on using for a t.38 stack? |
| 01:41.23 | JT | blitzrage: there is someone who writes all the sort of stuff, maybe digium could pay him? ;) |
| 01:41.26 | djPepse | hi guys, anyone use a cisco 7910 phone that can clue me in as to how it works? |
| 01:41.36 | `Sean | bkw_ elaborate please? |
| 01:41.44 | blitzrage | NEXT!!! |
| 01:41.51 | bkw_ | Well you can't use spandsp |
| 01:42.11 | DrukenLPY | what will asterisk call management enable do for me? |
| 01:42.13 | djPepse | In the 7910's settings I can make it "unlock", but it still won't let you change any settings :) |
| 01:42.21 | blitzrage | `Sean: I'm going to assume the 380+ bugs in the tracker and other projects |
| 01:42.30 | bkw_ | damn 380 bugs |
| 01:42.43 | blitzrage | that includes the GUI too though |
| 01:43.03 | file | and feature patches, and AsteriskNOW |
| 01:43.08 | bkw_ | true |
| 01:43.17 | bkw_ | I still don't get why i'm banned in #asterisk-dev |
| 01:43.18 | blitzrage | actually, only 379 :) |
| 01:43.20 | file | as usage goes up... bugs go up |
| 01:43.36 | bkw_ | guess my many patches and years of work doesn't gain me anything these days |
| 01:43.44 | `Sean | nopes |
| 01:43.50 | `Sean | isn't that the mark spencer way lol |
| 01:44.03 | file | bkw_: unbanned |
| 01:44.13 | `Sean | file pm |
| 01:44.22 | bkw_ | well I am very critical asterisk and some of its short comings |
| 01:44.33 | bkw_ | but then again it wasn't the tool I needed for the job |
| 01:44.44 | bkw_ | and trying to fit that square peg in that round hole didn't quite pan out |
| 01:44.49 | file | `Sean: hrm? I don't accept pms from people unless you are on my friend list |
| 01:44.50 | `Sean | lol |
| 01:45.04 | `Sean | wow, a guy on IRC with 'freinds list' |
| 01:45.07 | `Sean | heh |
| 01:45.14 | bkw_ | anyway.. I have round peg and round hole.. and square peg and square hole... nice fit |
| 01:45.33 | JT | `Sean: most people have friends lists, but they're mental lists |
| 01:45.35 | DrukenLPY | what about my big peg and a small hole? :) |
| 01:45.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:45.41 | `Sean | JT indeed |
| 01:45.49 | JT | DrukenLPY: careful stretching required |
| 01:45.57 | `Sean | ;p; |
| 01:45.58 | bkw_ | DrukenLPY, haha well what I have now can scale from small peg to large peg |
| 01:45.58 | `Sean | lol |
| 01:46.06 | bkw_ | and everything inbetween |
| 01:46.18 | DrukenLPY | :P |
| 01:46.19 | JT | bkw_: but can it do T.38? :P |
| 01:46.29 | JT | (and everything inbetween) |
| 01:46.31 | blitzrage | fax is dead. move along. |
| 01:46.36 | JT | lies |
| 01:46.44 | bkw_ | JT we'll have t.38 passthru and t.38 termination once we get UDPTL in there |
| 01:46.45 | `Sean | fax is still very much indeed alive ... |
| 01:46.53 | JT | bkw_: so "no" :) |
| 01:46.56 | bkw_ | but I think their is a bug in the passthru |
| 01:46.59 | bkw_ | but we are working on it |
| 01:47.08 | bkw_ | well we can do regular faxing with spandsp over our socket interface |
| 01:47.12 | bkw_ | ie T1 in |
| 01:47.28 | `Sean | bkw get support for t38 termination, and i'll love you :D |
| 01:47.37 | bkw_ | `Sean, Its on my list! ;) |
| 01:47.42 | `Sean | cool :) |
| 01:47.43 | file | `Sean: Callweaver has T38 termination using spandsp, have you tried it? |
| 01:47.48 | bkw_ | I want it as much as you do |
| 01:47.57 | JT | also, modem calls need to work in open source telephony software |
| 01:48.02 | `Sean | bah, i'm too lazy to make the switch, to callweaver i will have to redo almost every single thing :( |
| 01:48.14 | bkw_ | `Sean, use them together |
| 01:48.18 | bkw_ | asterisk does t.38 passthru |
| 01:48.23 | `Sean | but if bkw gets it working with callweaver in a stable point i'll switch to callweaver :) |
| 01:48.24 | JT | like you have a PRI, and an FXS for a modem, it'd be great if that worked well |
| 01:48.26 | bkw_ | then you can pass them off to callweaver to terminate them |
| 01:48.35 | JT | as many people have POS terminals etc |
| 01:48.41 | bkw_ | JT I think thats a zaptel issue isn't it? |
| 01:48.50 | JT | probably |
| 01:48.53 | file | I would also suggest if you are using callweaver either keep it isolated from the outside, or go through the Asterisk security advisories and patch it if applicable |
| 01:49.11 | `Sean | heh ive got a sangoma A400 in my closet never used i'll just use it if it indeed is a zaptel issue |
| 01:49.15 | `Sean | zaptel is quite buggy aswell tho |
| 01:49.23 | bkw_ | `Sean, you can use that with OpenZAP and fs now |
| 01:49.26 | `Sean | but cant complain, Opensource stuff :) |
| 01:49.33 | djPepse | any tips on how to resolve "callerid.c:607 callerid_feed: Caller*ID failed checksum" |
| 01:49.34 | djPepse | ? |
| 01:49.43 | bkw_ | all the analog hardware from both sangoma and digium works great via OpenZAP |
| 01:49.54 | bkw_ | and soon pika will work if we get time :) |
| 01:50.21 | bkw_ | it also has basic PRI support... for making and taking calls but nothing fancy |
| 01:50.41 | Juggie | bkw_, your working w/ pika? |
| 01:50.49 | `Sean | hrmp so its a replacement for zaptel then correct? |
| 01:50.58 | bkw_ | you still use zaptel kernel part |
| 01:51.04 | JT | oh, and bri support, that's another bitch i have with open source telephony |
| 01:51.10 | bkw_ | JT coming soon |
| 01:51.14 | bkw_ | gotta get a line and hardware |
| 01:51.15 | file | bkw_: if you guys haven't looked yet take a look at the latest Lumenvox email spam thingy, they are adding new configuration options and making some old ones deprecated (if you use them) |
| 01:51.21 | JT | bkw_: really good BRI and PRI support, that is |
| 01:51.24 | Juggie | bkw_, does pika support their T1/E1 cards yet? |
| 01:51.25 | JT | bkw_: in anything |
| 01:51.26 | Hmmhesays | so buy a bri sip gateway |
| 01:51.31 | bkw_ | JT yes |
| 01:51.35 | JT | no project has good bri support |
| 01:51.41 | JT | bkw_: i meant to say TE and NT mode |
| 01:51.49 | JT | bkw_: you can get euroisdn? |
| 01:51.53 | bkw_ | we do TE |
| 01:51.55 | Juggie | bkw, has their kernel support gotten any better? |
| 01:51.59 | bkw_ | NT coming and euro is being tested |
| 01:52.05 | JT | bkw_: yes, everyone does TE |
| 01:52.07 | bkw_ | the PRI stack we have came from a nice guy in the UK |
| 01:52.14 | Hmmhesays | I really like channelredirect in 1.4 whoever wrote that, props |
| 01:52.24 | bkw_ | we just have to write the state machine NT |
| 01:52.38 | bkw_ | s/NT/for NT/ |
| 01:53.07 | bkw_ | Hmmhesays, is that the 2bct? |
| 01:53.13 | JT | like malicious call trace, etc |
| 01:53.18 | bkw_ | JT our goal is to support all features possible |
| 01:53.26 | bkw_ | AOC, TBCT |
| 01:53.29 | Juggie | bkw_, i refuse to look @ pika until they compile montecarlo for 64bit/smp |
| 01:53.33 | Juggie | i've told them that a million times. |
| 01:53.36 | Hmmhesays | what? |
| 01:53.38 | JT | tbct? is that 2bct |
| 01:53.38 | bkw_ | Juggie, we'll see |
| 01:53.44 | bkw_ | tbct is the same |
| 01:53.47 | bkw_ | 2 or two |
| 01:53.50 | bkw_ | duh :P |
| 01:53.52 | JT | ECT |
| 01:53.54 | Juggie | bkw_, they were kissing my ass trying to get me to trial their stuff |
| 01:54.08 | Juggie | since we are they feds and they know we use asterisk |
| 01:54.11 | bkw_ | Juggie, well we are going to be working with them on a few things... I can't say much more than that |
| 01:54.41 | bkw_ | but rest assured what ever comes out of it will be open source for the world to benefit from |
| 01:54.51 | Juggie | bkw_, well hopefully they pull their head out of their ass, i dont know many people will want to run high density asteirsk servers non 64bit or non smp |
| 01:54.54 | *** join/#asterisk ManxPower (n=manxpowe@adsl-222-26-172.msy.bellsouth.net) |
| 01:54.58 | bkw_ | file, I think Lumenvox will be irrlevant soon |
| 01:55.26 | file | oh? |
| 01:55.47 | bkw_ | but who knows |
| 01:55.55 | Hmmhesays | so don't buy a million channels worth of their speech engine? |
| 01:55.56 | bkw_ | we will be interfacing with them with MRCP |
| 01:56.05 | bkw_ | use MRCP |
| 01:56.08 | bkw_ | or DIE |
| 01:56.33 | JT | oh and if any project gets a proper skype channel driver, PHBs everywhere will love you |
| 01:56.45 | bkw_ | skype will die if they keep this up |
| 01:56.59 | Hmmhesays | um sure? |
| 01:57.01 | JT | it seems to still be growing amongst nubs :( |
| 01:57.01 | bkw_ | the best bet for skype is to offer a business account with a sip interconnect |
| 01:57.11 | `Sean | meh iif asterisk business edition had fax id buy it |
| 01:57.17 | bkw_ | `Sean, it can't |
| 01:57.22 | bkw_ | because SpanDSP is GPL |
| 01:57.24 | `Sean | but its gayness dont got it either,, i guess its time i started reading, about Callweaver |
| 01:57.27 | `Sean | and did the switch |
| 01:57.28 | bkw_ | and steve will not give an exception for that |
| 01:57.39 | `Sean | Oh |
| 01:57.40 | JT | bkw_: what if digium pay him? :) |
| 01:57.42 | `Sean | thats why |
| 01:57.46 | bkw_ | he will not budge |
| 01:57.50 | Juggie | bkw_, someone told me lumenbox was doing some work on mrcp |
| 01:57.52 | bkw_ | people have tried to pay him |
| 01:57.55 | Juggie | but i didnt got the details |
| 01:57.57 | `Sean | heh |
| 01:57.59 | `Sean | they must have :) |
| 01:58.04 | bkw_ | Juggie, yes takea guess who's MRCP stack they'll use |
| 01:58.07 | `Sean | but cant you write a SpanDSP similar thing?? |
| 01:58.09 | Hmmhesays | hard to buy someone out who is comfortable financially |
| 01:58.11 | `Sean | and licence it :D? |
| 01:58.19 | Juggie | bkw_, no idea, i'm kinda distant from that. |
| 01:58.23 | bkw_ | Juggie, ours |
| 01:58.30 | bkw_ | the one Cepstral and Our guys did |
| 01:58.31 | JT | `Sean: it's very difficult |
| 01:58.33 | bkw_ | http://fisheye.freeswitch.org/browse/OpenMRCP |
| 01:58.35 | Juggie | i had no idea you wrote one, i know we talked about it before |
| 01:58.43 | Juggie | awesome, congrats ;) |
| 01:58.44 | JT | `Sean: it implements a full software modem in dsp, for starters |
| 01:58.44 | bkw_ | its a V1 and V2 stack |
| 01:58.48 | JT | err |
| 01:58.50 | JT | not in dsp |
| 01:58.52 | JT | in host cpu |
| 01:59.03 | bkw_ | JT you would be supprised how easy a software modem is once you understand it fully |
| 01:59.15 | `Sean | where is spandsp homepage? |
| 01:59.20 | `Sean | *is the |
| 01:59.21 | JT | bkw_: then write one ;) and all the other stuff that spandsp has |
| 01:59.27 | Hmmhesays | soft-switch.org ? |
| 01:59.29 | bkw_ | http://fisheye.freeswitch.org/browse/~raw,r=260/OpenZAP/trunk/src/zap_callerid.c |
| 01:59.42 | bkw_ | we had to do it for callerid in openzap |
| 01:59.52 | JT | callerid is a lot more of a simple case |
| 02:00.09 | bkw_ | exactly |
| 02:00.15 | bkw_ | but its very similar |
| 02:00.20 | Juggie | bkw_, mrcp is definitally the way to go, one unified interface. |
| 02:00.25 | *** join/#asterisk bbryant_ (n=Greenbox@user-24-214-124-177.knology.net) |
| 02:00.32 | bkw_ | Juggie, thats the reason I pushed for getting the stack written |
| 02:00.48 | bkw_ | no good open source stacks for both client and server existed till we started on it |
| 02:00.51 | JT | i personally hope skype goes bankrupt |
| 02:00.55 | JT | but someone will buy it :/ |
| 02:00.56 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:01.03 | bkw_ | just don't use Skype |
| 02:01.07 | JT | i don't |
| 02:01.08 | bkw_ | its aweful software |
| 02:01.10 | Qwell | somebody already bought skype :p |
| 02:01.12 | JT | crazy humans do |
| 02:01.17 | `Sean | oh your talking about coppice |
| 02:01.19 | Juggie | bkw_, i'm fairly certain we discussed this in the past, am i correct? |
| 02:01.21 | JT | it's indeed terrible |
| 02:01.45 | bkw_ | http://fisheye.freeswitch.org/browse/~raw,r=355/OpenMRCP/trunk/docs/OpenMRCP%20Design |
| 02:01.52 | Hmmhesays | wow the new buy.com commercial is just awful |
| 02:01.53 | bkw_ | Juggie, I think we did |
| 02:02.06 | Juggie | i believe so, we talked about the lack of a mrcp stack. |
| 02:02.37 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:02.39 | bkw_ | Juggie, that is no longer the case |
| 02:02.40 | bkw_ | ;) |
| 02:03.08 | Juggie | hopefully someone writes the speech mrcp hook for asterisk :) |
| 02:03.16 | *** join/#asterisk ManxPower (n=manxpowe@015-847-806.area5.spcsdns.net) |
| 02:04.02 | bkw_ | Juggie, Asterisk will have to provide an exception for OpenMRCP to be used in Asterisk |
| 02:04.15 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 02:04.52 | ManxPower | Hiya bkw_ |
| 02:04.56 | bkw_ | hey ManxPower |
| 02:04.57 | BugKhaM | Juggie: hi |
| 02:05.11 | Juggie | bkw_, why, whats the license? |
| 02:05.36 | bkw_ | MPL |
| 02:05.46 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:05.49 | ManxPower | Juggie: Digium does not include GPL software in Asterisk because of Asterisk Business Edition -- they don't want two codebases. |
| 02:06.15 | ManxPower | one would assume most open source licenses would have similar issues. |
| 02:06.17 | bkw_ | well OpenMRCP is MPL so commercial companies can take advantage of it without having to pay feels thus lowering the entry point for people to speak MRCP |
| 02:06.21 | ManxPower | BSD and LGPL are OK |
| 02:06.34 | BugKhaM | Juggie: I figured out yesterday after you left that my agi problem was from EXEC DIAL |
| 02:06.44 | bkw_ | never exec dial in an AGI |
| 02:06.48 | bkw_ | thats a bad thing to do in the first place |
| 02:06.57 | Juggie | bkw_, apparentally a2billing does it. |
| 02:07.04 | BugKhaM | Juggie: It exits as soon as the call hangs up |
| 02:07.19 | bkw_ | its a bad idea |
| 02:07.28 | Juggie | BugKhaM, dialing with a g wont really help either because that will only half fix it |
| 02:07.35 | bkw_ | doesn't DeadAGI fix that? |
| 02:07.47 | Juggie | bkw_, deadagi was never intended to run on a live channel |
| 02:07.59 | Juggie | this script was taking advantage of that bug, and running a agi on a live channel in dead mode |
| 02:08.22 | Juggie | there was also a bug in exec which didnt return the proper value from the exec |
| 02:08.26 | bkw_ | oh thats fun |
| 02:09.08 | BugKhaM | Juggie: yeah, it didn't seem that it works well with the 'g' either |
| 02:09.19 | Juggie | its fixed in trunk now. |
| 02:09.36 | Juggie | but since its more of a feature then a bug, its kinda hard to justify backporting it. |
| 02:09.41 | BugKhaM | Juggie: the call just didn't want to hangup untill timeout in the 'L' param |
| 02:10.08 | Juggie | BugKhaM, i dont know i've never tested dial within an agi. |
| 02:10.13 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:10.35 | Juggie | bkw_, since i havnt read the MPL license, what about it doesnt agree w/ the asterisk license. |
| 02:10.59 | Juggie | in theory though, i assume it could be a candidate for asterisk-addons |
| 02:11.23 | bkw_ | it has additional restrictions that say if you modify the code you must give those changes back even if you don't distribute |
| 02:11.24 | BugKhaM | Juggie: by placing another DeadAGI in the 'h' extension probably |
| 02:11.43 | bkw_ | the MPL is basically what's mine is mine and what's your's is your's |
| 02:11.46 | Juggie | bkw_, back to pika though, their linux support used to be horrible |
| 02:12.01 | bkw_ | Juggie, I think they are going to try to rectify that |
| 02:12.13 | Juggie | when i first tested their stuff for them 2-3 years ago, the only way to configure the boards was a gui app in xwindows |
| 02:12.24 | Juggie | i shit on them for that, and alot of other things. |
| 02:12.46 | bkw_ | yah that blows |
| 02:13.01 | Juggie | but yeah if they would rectify that, it would be nice |
| 02:13.13 | bkw_ | well if we have anythign to say about it.. they'll have a better interface |
| 02:13.26 | Juggie | lack of 64bit kernel support is just bad |
| 02:13.35 | bkw_ | but what we have now allows all the Sangoma (Native API) and Digium (Zaptel API) to work with FreeSWITCH in both 32bit and 64bit systems |
| 02:13.36 | Juggie | lumenvox i believe also has that problem |
| 02:13.49 | bkw_ | the sangoma stuff is going to work on FreeBSD and many other BSD's soon |
| 02:13.54 | bkw_ | and if I have my way.. Mac OS X |
| 02:13.58 | Juggie | we actually tested some Sangoma boards a few weeks ago |
| 02:14.07 | Juggie | we pushed 500,000 calls over a weekend with no problems |
| 02:14.27 | bkw_ | not bad |
| 02:14.52 | bkw_ | I have to get a few things into freeswitch before the first offical release but that list is a small list |
| 02:15.01 | Juggie | bkw_, pika keeps having an intreast in us because they know we are huge business, we run HUGE ivr farms, and also their HQ is like 20 minutes from ours |
| 02:15.50 | Juggie | i would guess we have probally no less then a thousand jct-120-ls cards in production |
| 02:16.06 | bkw_ | oh that reminds me |
| 02:16.31 | bkw_ | I have to sign this NDA with a company so I can get access to their code and start writing a module :P |
| 02:16.34 | Juggie | good luck if they start you with opendialogic |
| 02:16.43 | Juggie | i spent like a whole day trying to get those drivers to work |
| 02:16.44 | bkw_ | actually we'll have dialogic soon too |
| 02:16.47 | Juggie | they dont seem complete too me. |
| 02:16.49 | Juggie | *to |
| 02:16.58 | bkw_ | they were at Cluecon |
| 02:17.00 | Juggie | and i dont have a recent copy of the version 6 drivers |
| 02:17.01 | bkw_ | so was Pika |
| 02:17.33 | Juggie | i have an old copy of the linux drivers but not with the newer kernel support |
| 02:17.42 | Juggie | and i dont really want to run rhel3 :) |
| 02:17.47 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.211.11.revip2.asianet.co.th) |
| 02:18.43 | Juggie | i considered working on a dialogic driver alot |
| 02:18.57 | Juggie | considering i spent alot of time working with dialogic in a past life |
| 02:19.00 | HaMYaI | Juggie: you know how the bug get fixed, I will probably see if there's a work around for the time being |
| 02:19.03 | bkw_ | the idea is to plug Dialogic into OpenZAP |
| 02:19.10 | bkw_ | then the API from the top looks the same |
| 02:19.20 | Juggie | but i've never worked with the api required to stream audio to the board |
| 02:19.43 | bkw_ | so mod_openzap.c in freeswitch will be clueless on what hardware its driving |
| 02:19.56 | Juggie | then i coudnt get my hands on the latest driver, so i moved onto something else |
| 02:19.57 | bkw_ | infact we have a box here driving a sangoma T1 card and a tor2 at the same time.. |
| 02:20.09 | bkw_ | its quite neat to see |
| 02:21.16 | Juggie | it would be nice if zap be it zaptel or openzap was better at addressing cards in the system |
| 02:21.29 | bkw_ | OpenZap addresses this issue |
| 02:21.30 | Juggie | i mean its ok if it takes all the boards and addresses the channels 1 through whatever. |
| 02:21.46 | Juggie | but i'd like to be able to do Zap/board/span/channel |
| 02:21.51 | bkw_ | you get to control what is what.. and we translate it in the API |
| 02:22.08 | bkw_ | its totally generic from the ap layer |
| 02:22.11 | bkw_ | er app |
| 02:22.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:23.10 | bkw_ | well OpenZAP is trying to carry Jim Dixon's vision forward |
| 02:23.21 | bkw_ | a completly hardware agnostic abstraction layer |
| 02:23.28 | bkw_ | vendor neutrual |
| 02:23.35 | bkw_ | damn can't spell tonight |
| 02:23.43 | Juggie | HaMYaI, are you running 1.2 or 1.4 |
| 02:23.48 | bkw_ | vendor neutral |
| 02:24.09 | bkw_ | If you ever get a chance to meet Jim Dixon.. I recommend it |
| 02:24.19 | bkw_ | I have sadly only talked to him on the phone about 100 times.. funny man |
| 02:24.52 | Juggie | i've heard the name but i dont think i have unless he's been @ astricon |
| 02:24.58 | bkw_ | zaptel creator |
| 02:25.03 | bkw_ | the man that started it all |
| 02:25.36 | Juggie | can you say which company your going into an NDA with? |
| 02:25.48 | bkw_ | not right now |
| 02:25.50 | Juggie | HaMYaI, you havnt answered me. |
| 02:26.16 | bkw_ | if you hang in #freeswitch you might know :P |
| 02:27.21 | Juggie | bkw_, this is whats causing his problem likely, http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065&r2=71656 |
| 02:27.34 | Juggie | although he wont answer for me to tell him that |
| 02:28.00 | HaMYaI | Juggie: ohh sorry it's 1.2 |
| 02:28.07 | *** join/#asterisk jmacz (n=jmacz@190.25.34.209) |
| 02:28.07 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:29.02 | HaMYaI | Juggie: I just saw from the change log -> res/res_agi.c: Issue 10035 - handle_exec returns a result inconsistent with all of the other AGI commands |
| 02:29.05 | Juggie | HaMYaI, http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065&r2=71656 revert that change in your source and compile see if it works. |
| 02:29.42 | HaMYaI | Juggie: okie, will try that |
| 02:33.10 | Juggie | heres the thing, previously a hangup during a dial was not returning a RESULT_FAILURE to agi, so the agi continued, but now that a hangup does, its causing asterisk to hangup immeditally. |
| 02:33.33 | Juggie | hmmmm |
| 02:34.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:35.25 | bkw_ | Juggie, not sure |
| 02:36.04 | Juggie | HaMYaI, ok, as i told you PLEASE PLEASE submit a bug to the tracker with your expirence |
| 02:36.13 | bkw_ | our system will eventually turn into a hybrid Asterisk + FreeSWITCH |
| 02:36.16 | Juggie | i will fill in the details once you do this |
| 02:36.27 | bkw_ | if I get my way i'll have 2 DS3 cards per box |
| 02:36.38 | bakermd | Should OOB signaling negotiate as magic number 97 or magic number 101 |
| 02:36.38 | Juggie | bkw_, someones gotta make one first |
| 02:36.50 | bkw_ | Juggie, someone does.. just not ready ;) |
| 02:37.04 | Juggie | i'd like to know what ever happened to digiums |
| 02:37.07 | HaMYaI | Juggie: okie |
| 02:37.12 | bkw_ | you can't channelize a DS3 with Zaptel |
| 02:37.19 | bkw_ | its too inefficent |
| 02:37.36 | Juggie | its only software, its fixalble |
| 02:37.50 | bkw_ | you might think that but to fix it you might as well start over |
| 02:38.06 | bkw_ | you can't work on the idea of 1000ms interrupts |
| 02:38.22 | bkw_ | you needto work with larger chunks of data |
| 02:38.26 | bkw_ | on a DS3 |
| 02:44.08 | HaMYaI | Juggie: tested, and that works |
| 02:46.41 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 02:47.23 | *** join/#asterisk djPepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
| 02:47.29 | djPepse | woah |
| 02:47.59 | djPepse | so i got my 7910 to connect by aliasing a 169.254 address to the * machine.. when i press the HOLD button on the phone (while the handset is hung up), Asterisk restarts :D |
| 02:48.22 | Juggie | HaMYaI, still post the bug, its a change in behaviour |
| 02:48.30 | Juggie | and those arnt accepted in a release branch |
| 02:48.51 | Juggie | bkw_, hows openzap gonna fix it |
| 02:50.59 | sweeper | what's voltage tolerance like on polycom ip430's? |
| 02:52.12 | JT | wide |
| 02:53.14 | sweeper | will 19v do for it? |
| 02:53.35 | sweeper | oh, and it's 330, my bad |
| 02:53.37 | Qwell | djPepse: chan_skinny? |
| 02:53.53 | djPepse | yeah |
| 02:53.56 | Qwell | if so, I fixed that like...yesterday |
| 02:54.07 | djPepse | nice |
| 02:54.24 | djPepse | do i configure the buttons in the config of the phone or somewhere in *? |
| 02:54.37 | sweeper | phone config |
| 02:54.39 | JT | sweeper: the ip430 comes with a 24v supply |
| 02:54.47 | sweeper | JT: my 330 did not :3 |
| 02:54.52 | JT | right |
| 02:54.56 | JT | they're PoE phones |
| 02:54.58 | sweeper | yea |
| 02:55.04 | sweeper | and I need to rig an adaptor |
| 02:55.17 | *** join/#asterisk ZX81 (n=matt@202.20.97.200) |
| 02:55.24 | sweeper | I have no poe stuff handy, and I need to do some dev work with the xml stuff |
| 02:55.29 | djPepse | do i need a special cable for PoE adapter -> phone? |
| 02:55.43 | Qwell | for the 7910? |
| 02:55.45 | djPepse | yeah |
| 02:55.54 | Qwell | not sure if those do PoE |
| 02:56.01 | djPepse | oh. if they did, would I? |
| 02:56.04 | Qwell | I doubt they do 802.3af |
| 02:56.22 | *** join/#asterisk ZX81_ (n=matt@202.20.97.200) |
| 02:56.26 | snuff-work | usually only the 79x1 do proper 802.3af |
| 02:56.37 | Qwell | 7970 |
| 02:56.38 | ZX81_ | ok, here's a weird question |
| 02:56.39 | Qwell | but yeah |
| 02:56.42 | djPepse | if they don't, and I don't need a special cable, then I know why my PoE adapter wasn't powering it on :) |
| 02:56.56 | ZX81_ | if I am doing an install for 3000 homes, how many lines do you reckon I'd need? |
| 02:57.05 | sweeper | ZX81_: a lot |
| 02:57.07 | Qwell | ZX81: 300ish? |
| 02:57.14 | ZX81_ | ok cool |
| 02:57.16 | snuff-work | bout 10:1 ratio |
| 02:57.16 | djPepse | i'm also wondering why i can't dial the 7910's extension from another extension, or dial any extension from the 7910 |
| 02:57.21 | djPepse | but i can use the dialplan just fine |
| 02:57.37 | *** join/#asterisk Stridernzl (n=neville@125-239-179-163.jetstream.xtra.co.nz) |
| 02:57.41 | ZX81_ | for residential? |
| 02:58.04 | ZX81_ | holiday homes |
| 02:58.24 | sweeper | 10:1 is still pretty much what you want |
| 02:58.28 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-107-175-93.ny325.east.verizon.net) |
| 02:58.32 | ZX81_ | ok' |
| 02:58.38 | sweeper | you're talking total, or just inbound? |
| 02:58.46 | ZX81_ | total |
| 02:58.50 | sweeper | oh, ok |
| 02:58.53 | sweeper | yea, 10:1 |
| 02:58.53 | djPepse | Qwell: since you seem to know aboot chan_skinny, what's "find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on '501'" mean? |
| 02:58.54 | ZX81_ | we're going to supply ddi numbers |
| 02:59.06 | djPepse | (warning) |
| 02:59.07 | Qwell | djPepse: skinny set debug off ;) |
| 02:59.08 | ZX81_ | and outbound via our own demarc |
| 02:59.27 | ZX81_ | +doing fiber + tv |
| 03:00.02 | sweeper | JT: so will the 330 be happy with 19v? |
| 03:00.11 | djPepse | heh, it won't let me set it off |
| 03:00.22 | JT | sweeper: yes |
| 03:00.25 | djPepse | oh skinny nodebug |
| 03:00.26 | djPepse | sorry |
| 03:00.27 | sweeper | aight |
| 03:01.04 | JT | main screen turn on |
| 03:01.31 | *** join/#asterisk juanmanuel (n=jmacz@190.24.102.126) |
| 03:02.36 | djPepse | hm that's weird. |
| 03:02.46 | djPepse | i can't dial any extensions directly |
| 03:04.32 | djPepse | oh i think i might know why :/ |
| 03:06.12 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:09.41 | Qwell | djPepse: If you find out that it does PoE, could you let me know? |
| 03:11.46 | djPepse | Qwell: From what I've read, it should. but it didn't work with a regular patch cable |
| 03:11.55 | djPepse | Qwell: voip-info wiki says I might need a special cable |
| 03:14.28 | Qwell | I haven't tried it, but supposedly you can make one to "fool" cdp |
| 03:28.37 | *** join/#asterisk Corydon76-home (i=gold@pdpc/supporter/sustaining/Corydon76-home) |
| 03:28.37 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 03:28.39 | *** join/#asterisk jmacz (n=jmacz@190.24.100.200) |
| 03:30.41 | sweeper | JT: well, it's not turning on, but I don't smell the blue smoke... |
| 03:33.10 | JT | sweeper: wrong polarity? |
| 03:33.31 | sweeper | eh, pinout for poe says it's right |
| 03:33.49 | sweeper | or is it expecting some sort of negotiation? |
| 03:34.13 | JT | :o |
| 03:34.17 | JT | doesn't it have a socket |
| 03:34.45 | JT | 802.3af has negotiation, not sure if you need it or not |
| 03:34.55 | sweeper | it does, but wiring up a poe adaptor was easier than cuttin a plug off of something and attatching it to the power supply |
| 03:38.59 | *** join/#asterisk ilovephp (n=ilovephp@CPE-121-223-215-117.static.nsw.bigpond.net.au) |
| 03:39.02 | JT | i'd advise you to use the socket |
| 03:39.21 | *** join/#asterisk xanderp (n=xanderp@74-132-35-142.dhcp.insightbb.com) |
| 03:40.34 | sweeper | w00t |
| 03:40.35 | sweeper | worked |
| 03:40.36 | sweeper | thanks~ |
| 03:40.43 | JT | :) |
| 03:40.47 | ilovephp | hello all |
| 03:40.56 | xanderp | help please, asterisk noob trying to get inter-tel 8662 to register with trixbox and I get Registration from "1000" <sip:1000@10.1.1.212:5060> failed for 10.1.1.107 ACL error (permit/deny) I can't figure out how to fix the ACL error. Is there a conf file that I need to edit to tell it 'legal' IP's that phones can have? |
| 03:41.27 | sweeper | ilovephp: gtfo :v |
| 03:41.42 | xanderp | i couldn't find any acl.conf or anything to tell it what subnets I want to allow. |
| 03:42.19 | sweeper | nothing personal, just on principle :D |
| 03:42.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:42.39 | ilovephp | ? |
| 03:42.49 | sweeper | your nick :P |
| 03:42.57 | ilovephp | ooooo lol, i had to pick something |
| 03:43.19 | *** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au) |
| 03:43.24 | [TK]D-Fender | xanderp, .... |
| 03:43.27 | [TK]D-Fender | ~trixbox |
| 03:43.28 | jbot | somebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 03:44.23 | ilovephp | i take it your all long time users of asterisk? |
| 03:44.40 | [TK]D-Fender | ilovephp, and all this time I though I was in #muffins ! |
| 03:44.42 | sweeper | some more than others :) |
| 03:44.47 | sweeper | mmm, muffins |
| 03:45.05 | ilovephp | how do you find it, easy to use? |
| 03:45.16 | [TK]D-Fender | ilovephp, sure |
| 03:45.16 | sweeper | arg. I put the electrical tape away without taping up this splice *rummage* |
| 03:45.46 | sweeper | ilovephp: if you like dbs, be prepared to struggle a bit, but it's manageable if you can afford enough of a performance hit to use FastAGI |
| 03:46.45 | ilovephp | i c |
| 03:46.49 | sweeper | on the whole tho, it's a lot easier off the bat than the up-and-comers, eg freeswitch |
| 03:46.59 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:47.16 | ilovephp | how is it with integration with php and mysql application? easy? |
| 03:47.22 | [TK]D-Fender | sweeper, way to tailor the question to a * newb :) |
| 03:47.28 | *** join/#asterisk Blackthorn (n=Elive_us@w-l4.smyth.net) |
| 03:47.48 | sweeper | ilovephp: if you can afford the hardware to handle FastAGI overhead, yea |
| 03:47.48 | [TK]D-Fender | ilovephp, be specific about your goals and we'll be specific with our answers :) |
| 03:48.02 | sweeper | if you need to process massive amounts of calls, it'll be a lot harder |
| 03:48.05 | [TK]D-Fender | sweeper, Again, when did he mention AGI? |
| 03:48.10 | sweeper | php! |
| 03:48.19 | [TK]D-Fender | sweeper, its a friggen NICK. |
| 03:48.21 | sweeper | he wants his web stuff to talk to his phone stuff |
| 03:48.32 | ilovephp | pretty much |
| 03:48.36 | sweeper | < ilovephp> how is it with integration with php <-- |
| 03:48.38 | sweeper | see |
| 03:48.44 | Stridernzl | [TK]D-Fender: can i grab you for some mins to knock this over? |
| 03:48.59 | [TK]D-Fender | Stridernzl, I'm only up for a few more.... |
| 03:49.18 | [TK]D-Fender | Stridernzl, e-mail me the IP & dyndns and I'll have it in place in about 12H |
| 03:49.21 | sweeper | ilovephp: if you've got some time on your hands, you might want to take a look at adhearsion and ruby on rails. if you've got an existing php app tho, just stick with whatever FastAGI libraries exist for php |
| 03:49.31 | ilovephp | for example, the caller enters in a 'orderid' onto the keypad, and the telephone software needs to grab lets say the 'shipping status' of that order |
| 03:49.43 | [TK]D-Fender | sweeper, sure, yougo and assume what that means to HIM.... |
| 03:49.59 | Stridernzl | [TK]D-Fender: yeah sorry .. me just lost calls now for 4/5 days as remote extn just to suspect :( |
| 03:50.12 | sweeper | ilovephp: FastAGI is the fast way to use external scripting languages with asterisk |
| 03:50.33 | sweeper | however, if your needs are REALLY basic, you could get away with raw odbc queries in the dialplan |
| 03:51.13 | [TK]D-Fender | ilovephp, how many simultaneous callers in this lookup system? |
| 03:51.42 | bkw_ | php sucks |
| 03:51.52 | bkw_ | ok had to get that off my chest :P NEXT!!! |
| 03:51.57 | sweeper | bkw_: well that's a given :P |
| 03:52.11 | ilovephp | well i'm just trailing it now, but it needs to be scalable to handle up to 25 calls at any given time |
| 03:52.25 | [TK]D-Fender | bkw_, For 5 bucks more it'll swall, and for 25 it'll even lie and said it liked it ;) |
| 03:52.38 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 03:52.39 | [TK]D-Fender | swallow* |
| 03:53.00 | Qwell | for $50 more, it will like it |
| 03:53.14 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 03:53.18 | ilovephp | what other scripting languages are there besides php? theres asp.net and thats about it |
| 03:53.29 | sweeper | ruby |
| 03:53.31 | sweeper | is <3 |
| 03:53.35 | Qwell | perl, sh, ruby, python, awk |
| 03:53.47 | Juggie | Qwell |
| 03:53.54 | Juggie | M10035 |
| 03:53.55 | Qwell | that isn't a scripting language |
| 03:53.58 | sweeper | Erlang! |
| 03:54.05 | Juggie | that bug fix caused a behaviour change |
| 03:54.05 | sweeper | that's the most apropos for * :D |
| 03:54.09 | ilovephp | none as popular as php |
| 03:54.14 | sweeper | php sucks :v |
| 03:54.17 | Qwell | ilovephp: umm... |
| 03:54.18 | Qwell | yeah |
| 03:54.33 | Stridernzl | [TK]D-Fender: emailed |
| 03:54.34 | ilovephp | it does the job so i'm not complaining |
| 03:54.34 | sweeper | that's like saying myspace is cool because it's popular |
| 03:54.36 | JT | popularity doesn't make a language good |
| 03:54.37 | Juggie | just wondering on your thoughts, previously AGI could continue after a failed dial, but now it doesnt, i've seen some reports of this breaking scripts, specifically a2billing. |
| 03:54.51 | JT | php is one of the worst web scripting languages currently in use |
| 03:54.57 | *** join/#asterisk jicksta (n=jicksta@S0106001839e7062a.vc.shawcable.net) |
| 03:55.17 | [TK]D-Fender | Stridernzl, Got it, will have ready tomorrow |
| 03:55.58 | Stridernzl | [TK]D-Fender: you also never answered re x100P cards .. which i gladely post across the globe just for you :) |
| 03:56.02 | sweeper | ilovephp: http://www.oreillynet.com/lpt/a/7067 |
| 03:56.07 | sweeper | take a look at that :D |
| 03:56.16 | *** join/#asterisk juanmanuel (n=jmacz@190.24.103.182) |
| 03:56.25 | ilovephp | sweeper: i think the dailplan will be the best way to go |
| 03:56.34 | [TK]D-Fender | Stridernzl, it'd cost more in shipping than they're worth :) |
| 03:56.51 | sweeper | ilovephp: fi that's ALL you need to do, sure |
| 03:56.54 | Qwell | sending it USPS standard deliver would cost more than they're worth |
| 03:57.00 | sweeper | just compile the extension and bam |
| 03:57.01 | Qwell | (33 cent stamp) |
| 03:57.03 | Stridernzl | <[TK]D-Fender>: just think recycle .. i don't care about post i have an acocunt with them :) |
| 03:57.14 | ilovephp | yeah, easy way to get started |
| 03:57.33 | [TK]D-Fender | Qwell, .... he's in NEW ZEALAND <-------- |
| 03:57.58 | Stridernzl | [TK]D-Fender: they came from your way I am happy to send back or they will just end up in the rubbish ! |
| 03:58.04 | sweeper | man, I just basically recreated half of trixbox with 400 loc.... |
| 03:58.05 | sweeper | <3 |
| 03:58.12 | ilovephp | but i'll definately have a read through ahearsion, i'll need to work out how far i want the telephone system to go |
| 03:58.19 | sweeper | :D |
| 03:58.29 | sweeper | OVER NINE THOUSAAAAAND imo |
| 03:58.30 | JT | sweeper: what is adhesion/ |
| 03:58.39 | sweeper | JT: ruby FastAGI api |
| 03:58.42 | JT | ah ok |
| 03:58.52 | sweeper | well |
| 03:58.55 | Qwell | adhearsion* or something |
| 03:59.07 | sweeper | it does AMI too, supposedly >.> |
| 03:59.21 | Qwell | it's a bit more than just an AGI API |
| 03:59.27 | sweeper | maybe when that 0.8 version comes out it'll ahve that missing comma <.< |
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| 05:15.41 | errr | hmm having issues building 1.2.23 |
| 05:17.11 | Strom_M | what kind of issues? |
| 05:17.58 | errr | codec_zap.c:613: error: dereferencing pointer to incomplete type |
| 05:19.29 | errr | any idea what would cause that? |
| 05:19.59 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 05:20.33 | errr | I built & install 1.2.19 zaptel then went to build and install 1.2.23 asterisk but it fails there |
| 05:23.45 | errr | hmm I see in the list someone else had this but they had an old version of zap |
| 05:24.35 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 05:25.27 | errr | ah crap I fat fingered my way intot he old zap folder I had and reinstalled my old version of zap |
| 05:25.30 | errr | lol yay me! |
| 05:25.47 | MrTelephone | not very good :-/ |
| 05:25.57 | *** join/#asterisk saftsack (n=oliver@p54A7E7BB.dip.t-dialin.net) |
| 05:26.02 | errr | oh well now its building fine |
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| 05:38.04 | bakermd | Anyone know much about Cisco AS5300 gateways with E-1s? |
| 05:40.35 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 05:42.13 | jarod14 | hi guys |
| 05:42.44 | tengulre | hi,all |
| 05:42.50 | tengulre | anybody alive? |
| 05:47.25 | Strom_M | ALL DEAD HERE |
| 05:48.40 | MrTelephone | shot and killed |
| 05:53.13 | *** join/#asterisk saftsack (n=oliver@p54A7D268.dip.t-dialin.net) |
| 05:56.22 | tengulre | haha... |
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| 05:57.25 | tengulre | where have cheap compatible asterisk card ? |
| 05:58.14 | bakermd | Man, ISDN is a biotch |
| 05:58.23 | bakermd | Cisco makes it soo difficult sometimes |
| 05:58.27 | bakermd | Invalid information element contents |
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| 06:00.31 | tengulre | what's different between voice gateway and voice card for dailing users? |
| 06:09.21 | Strom_M | tengulre: you're not making a lick of sense |
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| 06:11.45 | MrTelephone | heh |
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| 06:46.06 | *** join/#asterisk RSAMan (n=aa@196.210.154.4) |
| 06:46.16 | RSAMan | hi |
| 06:46.19 | RSAMan | quick question |
| 06:46.27 | RSAMan | i am setting up a sip server |
| 06:46.28 | Strom_M | the answer is "cheese" |
| 06:46.31 | RSAMan | thanks |
| 06:46.34 | RSAMan | and.. |
| 06:47.08 | RSAMan | can i use variables "exten => 2203,1,Dial(2203) |
| 06:47.19 | RSAMan | variables instead of 2203 |
| 06:47.32 | RSAMan | so exten => var,1,Dial(var) |
| 06:47.36 | RSAMan | is this possible |
| 06:47.55 | RSAMan | so that everyone who called for anyone will be transferred to them via sip |
| 06:48.01 | RSAMan | providied that person exists |
| 06:48.40 | RSAMan | ? |
| 06:48.41 | creativx | no |
| 06:48.41 | jarod14 | you can use a pattern and use ${EXTEN} variable like exten => _22XX,1,Dial(SIP/${EXTEN}) |
| 06:48.48 | creativx | do like jarod14 said |
| 06:48.51 | creativx | :) |
| 06:49.03 | jarod14 | thx creativx |
| 06:49.08 | creativx | np gg |
| 06:49.18 | jarod14 | I love to be approved ;o) |
| 06:50.00 | RSAMan | kk |
| 06:50.03 | RSAMan | i c |
| 06:50.14 | RSAMan | because waht is someone phones a peter |
| 06:50.22 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:50.43 | RSAMan | then i want everyone who is calling a peter to call peter on sip |
| 06:50.48 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 06:50.57 | RSAMan | am i making sense,,, |
| 06:50.59 | *** join/#asterisk Penggu (i=foobar@220-245-200-87.static.tpgi.com.au) |
| 06:51.04 | Strom_M | RSAMan: not really |
| 06:51.04 | RSAMan | like i want to cover all names in sip.conf |
| 06:51.08 | creativx | you are thinking of using named extensions |
| 06:51.20 | creativx | like exten => peter,1,dial(sip/peter) |
| 06:51.21 | RSAMan | i will look that up |
| 06:51.25 | Penggu | hi all. we have snom phones adn asterisk on the local net. what should i use in sip.conf for nat=yes/no/never/etc ? |
| 06:51.26 | RSAMan | yes |
| 06:51.31 | RSAMan | like that creativx |
| 06:51.50 | creativx | are you dialing the number "peter" ? |
| 06:51.51 | RSAMan | creativx> : but isnt there a better way to do that ? |
| 06:51.55 | RSAMan | name |
| 06:51.59 | RSAMan | creativx : name peter |
| 06:52.02 | Penggu | also, i have a vpn connection using windoze client/server for one of the sip soft-phones, but it ends up on the same network anyway, so what about that one? nat=?? |
| 06:52.11 | RSAMan | creativx : what if i had 1000 names |
| 06:52.26 | RSAMan | creativx : would i have to set the extension for all of them ? |
| 06:52.28 | creativx | RSAMan: and nobody is to have numeric extensions? only named ? |
| 06:52.56 | creativx | Penggu: nat !? lan |
| 06:53.04 | RSAMan | creativx : numeric would be a better idea |
| 06:53.17 | JT | ~sipnat |
| 06:53.17 | 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 |
| 06:53.37 | RSAMan | creativx : basically, do i have to route each and every line in the rules ? |
| 06:53.43 | RSAMan | creativx : basically, do i have to route each and every line in the extensions |
| 06:53.46 | RSAMan | creativx : basically, do i have to route each and every line in the extension.conf |
| 06:54.11 | JT | i think you need to read the book |
| 06:54.13 | JT | ~thebook |
| 06:54.14 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:54.26 | RSAMan | kk |
| 06:54.33 | RSAMan | been reading |
| 06:54.39 | RSAMan | at about p 110 |
| 06:54.49 | RSAMan | set up the sip server |
| 06:54.55 | RSAMan | who can phone itself |
| 06:54.59 | Penggu | if i type: asterisk*CLI> restart when convenient i can't type any more commands unless i exit (which works) re-enter asterisk -r |
| 06:55.23 | Penggu | creativx: dunno.. the default settings in sip.conf... didnt touch'em.. wondering which one to choose |
| 06:56.22 | Penggu | i changed all to =no |
| 06:56.29 | Penggu | may be i should put it in the general section |
| 06:58.00 | Penggu | would type=friend work under [general] ? |
| 06:58.07 | Penggu | all my sip people are my friends |
| 07:04.57 | creativx | Penggu: you should read up on sip.conf |
| 07:05.26 | creativx | RSAMan: the idea is the map a numeric extension to a named sip user |
| 07:05.31 | creativx | is to |
| 07:05.33 | Penggu | ive got the page open on voip-info.. it doesn't say which settings will apply from [general] or not |
| 07:06.14 | Penggu | ok hang on |
| 07:06.19 | Penggu | i missed that line |
| 07:06.36 | Penggu | sorry, thanks |
| 07:06.38 | creativx | also did you check out the sample sip.conf |
| 07:06.39 | creativx | :) |
| 07:07.41 | Penggu | im heavily editing my ex-sample to reduce repeats |
| 07:09.20 | RSAMan | creativx : so i should manually input each numeric extension to a person using the sip |
| 07:12.40 | RSAMan | http://mirror.internode.on.net/pub/fedora/linux/releases/7/Fedora/i386/iso/F-7-i386-DVD.iso |
| 07:12.42 | RSAMan | oops |
| 07:12.54 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 07:12.54 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:13.30 | mvanbaak | mornin all |
| 07:14.30 | Penggu | ahh... here goes nothing... "reload" |
| 07:14.38 | Penggu | i should set up svn for asterisk config files |
| 07:16.31 | Penggu | cleaning up sip.conf by putting the common stuff under general reduced the file size from 28880 to 9276 bytes |
| 07:16.32 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 07:16.46 | Penggu | (not worried about size.. but will make it easier to maintain) |
| 07:16.51 | mvanbaak | yeah |
| 07:16.58 | mvanbaak | configs in subversion is nice |
| 07:17.02 | mvanbaak | we have that as well |
| 07:17.18 | mvanbaak | heckt, I put everything in subversion cept my music and movies collection |
| 07:17.19 | Penggu | mvanbaak: good for stuffing around |
| 07:17.39 | mvanbaak | my whole homedir is in svn |
| 07:17.50 | Penggu | i did that in windoze |
| 07:17.52 | Penggu | things got messy |
| 07:17.56 | Penggu | REALLY messy |
| 07:18.02 | mvanbaak | and all my boxen have this default /etc checkout with a host specific branch |
| 07:18.57 | Penggu | was ael the alternative congif format? |
| 07:19.06 | Penggu | i dont see much mention /example of it on voip-info |
| 07:19.15 | Penggu | althogh tfot touched on it |
| 07:22.02 | Penggu | nyway, im out |
| 07:22.04 | Penggu | cyas |
| 07:23.57 | creativx | hmmm |
| 07:23.59 | creativx | that was a good idea |
| 07:24.04 | creativx | why havent i svn'ed my configs :-) |
| 07:26.02 | mvanbaak | no idea |
| 07:26.09 | mvanbaak | lack of time to do so ? |
| 07:26.33 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 07:26.35 | hank | hi there |
| 07:26.56 | creativx | mvanbaak: it just hadnt occured to me |
| 07:27.53 | mvanbaak | it really is nice |
| 07:28.19 | mvanbaak | if a customer calls: "yesterday it worked great, today it's acting weird" |
| 07:28.35 | mvanbaak | you can do svn log && svn diff && svn up -r<somerev> |
| 07:28.59 | creativx | yeah |
| 07:29.10 | creativx | or just right click it with tortoisesvn ;) |
| 07:29.12 | creativx | for us in the win world |
| 07:32.33 | mvanbaak | yup |
| 07:35.09 | flart | morning everyone |
| 07:36.08 | tzafrir_laptop | Autoreply: morning everyone |
| 07:36.25 | mvanbaak | ;) |
| 07:36.27 | flart | any ideas how i could verify that asterisk is "seeing" my isdn-ntba? "capi show channels" tells me asterisk sees two isdn-channels, but i can't place any successful call |
| 07:36.31 | flart | ;) |
| 07:36.31 | tzafrir_laptop | Autoreply: Autoreply: morning everyone |
| 07:36.38 | tzafrir_laptop | that's it |
| 07:36.44 | creativx | hehe |
| 07:37.24 | flart | oh, and i mean calls from the outside to the asterisk server |
| 07:37.38 | mvanbaak | flart: any logs when you try to dial out ? |
| 07:38.13 | flart | i just tried to dial in |
| 07:38.43 | flart | and i don't see anything during that (verbosity on 5) |
| 07:38.46 | penguinFunk | anyone here got isdn30 in the UK with international calls working? |
| 07:39.35 | mvanbaak | flart: can you dialout ? |
| 07:41.07 | flart | i don't have a network-connection at the moment to the server ;) i'm connected via serial-console at the moment |
| 07:42.14 | flart | is there a way to dial out from the asterisk console? |
| 07:45.04 | creativx | dial |
| 07:46.07 | RSAMan | call the functions ?> |
| 07:46.14 | RSAMan | for asterisk |
| 07:46.15 | RSAMan | ? |
| 07:46.22 | RSAMan | wouldnt that work? |
| 07:46.34 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 07:46.46 | Chris-NB | hi |
| 07:47.15 | Chris-NB | how do I check if there is a @ in the ${EXTEN} varialbe |
| 07:48.00 | Chris-NB | I tried this, but wasn't successful: $["${EXTEN}" =~ "@"] |
| 07:48.30 | Chris-NB | is this correct? do I have to escape the @ ? |
| 07:49.38 | sweeper | if the polycom says "error saving application", does this mean it's trying to get the sip.ld and not finding it? |
| 07:49.54 | sweeper | and how do I tell it in the config file not to grab the config file at all? |
| 07:53.46 | JT | flart: capi is old school, what card do you have? |
| 07:54.06 | flart | fritzcard pci |
| 07:54.11 | flart | crappy thing i think |
| 07:54.49 | sweeper | err |
| 07:54.56 | sweeper | s/config file/application/ |
| 07:55.12 | flart | but it's just for testing purposes |
| 07:58.49 | JT | is there only drivers for capi? |
| 07:59.38 | flart | the capi-setup was the only one that seems to work |
| 08:01.37 | JT | tried bristuff? |
| 08:01.48 | JT | i don't remember if it has drivers for that or not |
| 08:01.57 | JT | is the fritz a hfc-s based card? |
| 08:02.00 | mvanbaak | no |
| 08:02.02 | JT | if so there's zaphfc |
| 08:02.05 | mvanbaak | it's a capi based thing |
| 08:02.13 | JT | yuck |
| 08:02.29 | mvanbaak | misdn should work as well |
| 08:02.55 | mvanbaak | it's not a chan_capi issue I think. because flart can do 'capi show info' and that shows them they have 2 B channels |
| 08:03.04 | mvanbaak | that means it "should" work in theory |
| 08:04.10 | JT | misdn, working, no way ;) |
| 08:04.38 | flart | yeah, it "should" ;) |
| 08:04.40 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:04.52 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 08:07.22 | flart | hm, i think i try asking our phone-guys if they have something to test the ntba and the isdn-cable |
| 08:08.56 | mvanbaak | I was about to ask if you were sure if the lines were working |
| 08:10.36 | flart | they should be working. but i gonna verify this |
| 08:10.54 | flart | ...i really begin to hate this whole phone stuff |
| 08:10.59 | flart | nothing you can ping |
| 08:11.10 | mvanbaak | that's why we went to a pure voip setup |
| 08:11.13 | mvanbaak | indeed |
| 08:11.18 | mvanbaak | no mtr, no ping |
| 08:13.07 | penguinFunk | woah |
| 08:13.13 | penguinFunk | been banging my head about for 2 days now |
| 08:13.18 | penguinFunk | and i have found the answer |
| 08:13.33 | penguinFunk | pridialplan=unknown should be set |
| 08:13.35 | mvanbaak | and it was 42 ? |
| 08:13.39 | penguinFunk | lol |
| 08:13.45 | penguinFunk | the default is national only |
| 08:14.44 | penguinFunk | [16:56:11] <penguinFunk> i have noticed that in order to make calls you have to leave out the preceeding 0 |
| 08:14.44 | penguinFunk | [16:56:43] <penguinFunk> so to dial 01554 723 345 you need to dial 1554 723 345 |
| 08:14.44 | penguinFunk | [16:56:51] <penguinFunk> because of the way isdn30 works |
| 08:14.44 | penguinFunk | [16:56:56] <penguinFunk> but what about international calls? |
| 08:14.49 | penguinFunk | [16:58:57] <penguinFunk> i have tried leaving out one 0, both 0's leaving the number completely intact, nothing |
| 08:15.01 | flart | mvanbaak: the sip-only test worked perfectly :) |
| 08:15.20 | penguinFunk | phoned our provider, they didn't know. phoned BT they didn't know |
| 08:15.25 | penguinFunk | bleh |
| 08:18.49 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
| 08:24.16 | mvanbaak | flart :) |
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| 09:01.29 | E-bola | Do anybody have experience with the snom 370? |
| 09:02.00 | sweeper | mmmthis is awsome |
| 09:02.16 | sweeper | I pick up the handset, and the phone reboots |
| 09:03.04 | mvanbaak | sweeper: nice |
| 09:04.36 | enioreh | sweeper: strange, this isn't supposed to happen. i work with snom300/320/360 and we never had such behavior |
| 09:04.41 | *** join/#asterisk Grapsus (n=grapsus@rai21-2-82-228-65-50.fbx.proxad.net) |
| 09:04.57 | Grapsus | Hi ! |
| 09:05.42 | sweeper | polycom 330 :P |
| 09:05.59 | enioreh | sweeper: sorry , misread ;) |
| 09:06.19 | sweeper | and I'm running 2.1.1.0037.... |
| 09:06.24 | sweeper | is there a later rev? :/ |
| 09:06.34 | enioreh | E-bola: what for ? |
| 09:07.39 | Grapsus | I'm trying to replace my sip.conf with a mysql table, it works well for all my users, but how do you entrer a line like "register => login:pwd@host" in the database ?? |
| 09:07.52 | E-bola | enioreh: i have a client who got grandstream 2000 phones |
| 09:08.02 | E-bola | and want to offer him some phones with better audio quality |
| 09:08.27 | E-bola | I was planning on giving him to 360's to play with |
| 09:08.42 | E-bola | but now i see there is a 370 oin the market too, which priced considerably higher |
| 09:09.27 | enioreh | E-bola: to me, the 370 won't offer a better sound quality but simply an improved screen |
| 09:09.40 | E-bola | enioreh: have u tried it? |
| 09:10.01 | enioreh | no |
| 09:10.02 | sweeper | oh ffs |
| 09:10.19 | sweeper | of COURSE this phone only works with 2.2.... |
| 09:10.34 | sweeper | and now it's been overwritten.... D: D: D: |
| 09:10.41 | E-bola | enioreh: How can u say it wont have better audio then? |
| 09:11.50 | Grapsus | I've read about MYSQL_FRIENDS option but it doesn't seem to exist anymore |
| 09:12.02 | enioreh | E-bola: wait a minute |
| 09:12.29 | sweeper | Grapsus: use realtime |
| 09:12.45 | E-bola | Im wondering if a snom 360/370 has better, worse, or similar audio quality compared to a polycom 501 |
| 09:12.58 | Grapsus | sweeper: I do, but how to enter a line like "register => foo:bar@baz" in the table ? |
| 09:13.09 | mvanbaak | E-bola: the 360 is great |
| 09:15.00 | Grapsus | sweeper: I use this structure http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip , it works for my users, but how to "register =>" ? |
| 09:15.13 | sweeper | Grapsus: not getting you |
| 09:16.26 | sweeper | how does one get in touch with polycom tech support? |
| 09:16.28 | enioreh | E-bola: we had to open our phones here, and snom320/360 were using the same hardware. I quickly took a look at the datasheets to search for differences about audio. There's noting about a better sound quality. |
| 09:16.59 | enioreh | E-bola: the main difference between the 320 and the 360 is the application running on it which manage the screen. |
| 09:17.04 | E-bola | well i think its known that the 230 and 360 has same audio |
| 09:17.10 | E-bola | but the 370 is being marketed as having improoved audio |
| 09:17.21 | E-bola | 230=320 |
| 09:17.24 | Grapsus | sweeper: in my sip.conf I have a line like "register => user:password@provider" I wanna move it to the database, it's not complicated ! |
| 09:18.00 | enioreh | E-bola: the point about 370 is that it has more memory and that memory has some influence on audio quality |
| 09:18.50 | E-bola | But you dont think it will make a noticeable difference? |
| 09:19.18 | enioreh | E-bola: i don't think so, but as we never get any 370 here, i can't affirm it. |
| 09:20.28 | *** part/#asterisk Axet (n=john@smirnoff.nurvnet.org) |
| 09:23.35 | sweeper | http://www.x2n.net/polycom/ <-- latest polycom firmware. get it while you can still get an index listing :D |
| 09:25.12 | enioreh | E-bola: i am reading throught the web site and some part of the 370 presentation sounds like commercial crap to me |
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| 09:27.20 | Grapsus | so nobody could help me ? I just want an example how to put a "register =>" line in sip.conf to a database |
| 09:27.45 | E-bola | enioreh: do you know the polycom phones? |
| 09:27.52 | E-bola | im wondering if they have better audio than the snom's |
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| 09:28.23 | enioreh | E-bola: know it only by name, we had one conferencing device from them but i didn't test it much and cannot help you about sound quality |
| 09:28.55 | enioreh | All i can say from my experience is that i really don't think the 370 has a better sound quality that the 360 or even the 320 |
| 09:29.09 | sweeper | oh WTF |
| 09:29.17 | enioreh | they present 370 features on the website that the 320 and the 360 already had |
| 09:29.22 | sweeper | new firmware, phone still dies :/ |
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| 09:36.31 | foo | I stuck in a new NIC card... do I have to tell asterisk to use eth1 instead of eth0 now? Or no? |
| 09:36.58 | Paul-T | Hello all, is this an appropriate place to ask about ASTCC? |
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| 09:53.28 | Grapsus | there's another strange thing with realtime, I moved my users to the database, they are connected, can call, but "sip show peers" or users displays nothing |
| 09:55.12 | enioreh | Grapsus: i remember having read somewhere that this is the behavior of realtime, but i am not sure about this |
| 09:55.52 | enioreh | and sip peers are updated in the database when connected |
| 09:56.29 | Grapsus | although I can see them with "database show" |
| 09:58.06 | Grapsus | and still my simple question, in [general] context of sip.conf I have a line "register =>xxx:yyy@zzz" how to delete it from sip.conf and put it in my mysql table sip_buddies ? |
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| 10:01.10 | Grapsus | is it possible ? |
| 10:02.07 | enioreh | no idea about it |
| 10:03.18 | Grapsus | cause I'm writing a webbased GUI, and I want all the configuration data in the database |
| 10:06.04 | enioreh | Grapsus: you may eventually some include=web-generated-file in your sip.conf |
| 10:06.15 | enioreh | i know this doesn't answer but it may be a workaround |
| 10:07.09 | Grapsus | Yes I tought abot it, there a script retrieve_sip_conf_from_mysql.pl |
| 10:07.22 | Grapsus | but maybe is there a cleanner way to do it :/ |
| 10:09.39 | enioreh | Grapsus: http://forums.digium.com/viewtopic.php?p=45131&sid=ade9eeb7acc25ea5a1fd31221016c396 |
| 10:09.43 | enioreh | i found this |
| 10:09.59 | enioreh | seems to be part of the answer you are looking for |
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| 10:11.41 | Grapsus | enioreh: intereseting, thanks |
| 10:13.58 | *** join/#asterisk morex (n=m@91.84.56.12) |
| 10:14.03 | morex | Hi all |
| 10:14.13 | morex | Don't know if the Asterisk-users list maintainer is here |
| 10:14.29 | morex | But somebody's autoresponder is causing it to spiral out of control... |
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| 10:34.40 | MadUni | hi all.. does anybody know smth about multiply calls in asterisk via flash button? |
| 10:36.31 | MadUni | I know 2 conference applications.. meetme & app_conference.. but I was said, that there is simplier solution |
| 10:37.37 | MadUni | ????? ?????! |
| 10:38.47 | enioreh | Typing lines full of ? won't make ppl answer ;) |
| 10:39.10 | pj_ | Yeah, it's not enough, nobody will care below 3 or 4 |
| 10:39.20 | pj_ | (amateurs) *sigh* |
| 10:42.20 | creativx | MadUni: you mean "multiple" ? |
| 10:42.41 | creativx | or multiple participants |
| 10:43.06 | MadUni | multiple ) |
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| 11:12.00 | Paul-T | Is anyone aware of any known issues with ASTCC detecting hangups in the latest svn? |
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| 11:27.07 | skitfish | Hello all, I have a quick, short question: Can asterisk detect when a call it has originated has been redirected (to an answering service, for example)? |
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| 11:28.52 | Fl1p | hi, i've got problems with installing zaptel on gentoo kernel 2.6.19, compiling works (there is no make linux26 anymore isn't it ?) but the modprobe gives me an error inserting zaptel invalid module format..... any clue ? |
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| 11:38.59 | skitfish | so can asterisk detect when a call it's originated has been transferred? |
| 11:42.16 | hank | It can detect answering machines afair |
| 11:42.46 | skitfish | cool |
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| 11:44.02 | skitfish | I've searched for configuration options using the terms 'redirect' and 'transfer' but I haven't found anything |
| 11:44.31 | Aurs | is the info from "sip show peer <peername>" available for realtime sip peers? |
| 11:44.40 | Aurs | (1.4.9) |
| 11:44.41 | skitfish | hank: do you remember roughly how I might make asterisk respond to said transfers? |
| 11:45.22 | tsurko | hi, is it necessary the value of accountcode field in sip.conf to be in " "? |
| 11:45.36 | enioreh | Aurs: as i know, not, you should use database show |
| 11:46.17 | Aurs | enioreh: that shows me registry, but not other useful info that you get from sip show peer, such as useragent, qualify status etc.. |
| 11:46.26 | enioreh | oh .. |
| 11:46.32 | enioreh | sorry, cannot help :) |
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| 11:47.50 | hank | skitfish: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 11:48.04 | skitfish | thanks hank |
| 11:48.19 | Aurs | tsurko: I don't use "" on the accountcode field in sip.conf |
| 11:51.29 | tsurko | Aurs, whithout "" this - test=${CDR(accountcode)} - shows this in CLI: [s@macro-int_call:4] Set("SIP/tsurko-0822bb58", "test=0"). Am I doing something wrong? |
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| 11:55.33 | Aurs | enioreh: found my answer: rtcachefriends=yes in general section of sip.conf :) |
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| 11:59.59 | sweeper | when I specify a mailbox in sip.conf, does that get dialed trhough the dialplan, or does it go straight to voicemail? |
| 12:00.33 | enioreh | sweeper: you have to specify it in the dialplan |
| 12:01.31 | Aurs | tsurko: i don't know... is the result different if you use ""? |
| 12:01.45 | Aurs | have anyone else seen this errormsg: "Function CD$R not registered"? |
| 12:02.29 | lirakis | morning |
| 12:04.26 | sweeper | morning~ |
| 12:05.04 | tsurko | Aurs, yes, but I think I figured out the problem. When there is dash in the accountcode everything messes up. It works now. |
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| 12:06.13 | Aurs | oh... i had a typo in my cdr_custom file.. oops |
| 12:06.25 | Aurs | tsurko: ok |
| 12:12.53 | johndo | mysql backend or flatfile? |
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| 12:19.05 | waKKu | morning ;) |
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| 12:20.43 | creativx | top of the middle of the day to you sir |
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| 12:26.36 | wothinn | I switched from 1.2.23 to 1.4.9 and all of a sudden 'exten => *99,1,Voicemailmain(${CALLERIDNUM})' in my extensions.conf is causing app_voicemail to ask for what mailbox I want to talk to when I dial *99. |
| 12:26.37 | zeeesh | hi |
| 12:26.44 | wothinn | Any ideas on that one? |
| 12:27.03 | zeeesh | how to uninstall asterisk newer versino .. |
| 12:27.06 | [TK]D-Fender | wothinn: Stop using deprecated variables |
| 12:27.17 | [TK]D-Fender | wothinn: And read the upgrade.txt |
| 12:27.24 | Aurs | wothinn: try ${CALLERID(num)} |
| 12:27.30 | wothinn | Thanks. I'll go read upgrade.txt. |
| 12:27.33 | [TK]D-Fender | wothinn: And atricles that tell you about al the changes out there. |
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| 12:54.11 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
| 12:54.14 | [TK]D-Fender | b00m |
| 12:55.13 | blitzrage | goes the dynamite |
| 12:57.18 | [TK]D-Fender | blitzrage: I DON'T WANT THE NET TO SPLIT |
| 12:57.24 | blitzrage | I JUST WANT... |
| 12:57.26 | [TK]D-Fender | blitzrage: ! ! ! |
| 12:57.51 | blitzrage | I could totally go for some ! ! ! |
| 12:58.30 | CM3_1_2_632 | is a Wildcard X100P FXO from ebay for $24.99 any good? |
| 12:59.25 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 12:59.30 | tzafrir_laptop | CM3_1_2_632, you can probably get it for less. 24$ includes S&H? It's really nice for experimentation and such |
| 12:59.45 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 13:00.22 | CM3_1_2_632 | tzafrir_laptop: same seller has old X100P with older components on board for $14.99 |
| 13:00.43 | CM3_1_2_632 | tzafrir_laptop: and yes all i need is 1 FXO so i can learn this puppy... |
| 13:00.46 | tzafrir_laptop | X100Ps are actualy not manufactured anymore |
| 13:01.56 | Optic | a crappy FXO can really taint your voip experience |
| 13:01.58 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 13:02.25 | CM3_1_2_632 | yeah but i don't want to spend a few hundred dollars buying from diguim |
| 13:03.05 | tzafrir_laptop | Optic, but it will work |
| 13:03.10 | Optic | if you're not using it for anything real the cheap cards will do something for you |
| 13:03.37 | Optic | the X100P style cards are basically a voice modem chipset that happens to work with Asterisk |
| 13:04.08 | CM3_1_2_632 | it was a motorola now an intel from what i've read |
| 13:04.37 | tzafrir_laptop | Right. I wish people could just use any such modem and get the same experince they get with the X100P |
| 13:05.34 | Strom_M | tzafrir_laptop: write the drivers then and make it possible :) |
| 13:06.03 | tzafrir_laptop | Strom_M, any specs availble for another such modem? |
| 13:06.05 | Optic | I think the analog side of most X100P style cards is poor |
| 13:06.22 | Strom_M | tzafrir_laptop: ? |
| 13:06.24 | Optic | at least the ones I had |
| 13:06.36 | tzafrir_laptop | Optic, it's poor. But if you're in the US, chances are it is good enough |
| 13:06.51 | tzafrir_laptop | I also used it for my first PBX |
| 13:07.00 | Optic | good enough to learn on :) |
| 13:07.47 | coppice | modems are more demanding than voice. if the analogue part were poor it would never function as a modem |
| 13:07.47 | CM3_1_2_632 | which is all i'm asking for..... |
| 13:08.20 | coppice | source is available for a simple driver for most modem cards in the Linux packages to use them as a modem. |
| 13:08.43 | CM3_1_2_632 | quality is not much of my concern....but for $24.99 it gets me a FXO / FXS card i think is worth it.... |
| 13:08.50 | tzafrir_laptop | coppice, most of them include an ugly binary blob in the middle, AFAIR |
| 13:08.59 | Strom_M | CM3_1_2_632: it's not FXS |
| 13:09.02 | Strom_M | FXO only |
| 13:09.11 | CM3_1_2_632 | right...sorry |
| 13:09.12 | CM3_1_2_632 | analog bypass it is.... |
| 13:09.24 | coppice | nope. most of them have source for a kernel driver, and the DSP is binary and sits in userland |
| 13:10.58 | tzafrir_laptop | I remember differently. Looking at the linmodems mailing list, most of them seem to need the Ant properitary drivers. Which is why I never bothered for modems on my system |
| 13:10.58 | CM3_1_2_632 | i'm building asterisk on CentOS 5.....trying to get the ebay seller to reply to my shipping inquiry coz i'm far |
| 13:11.06 | *** join/#asterisk af_ (n=getsmart@81-174-46-138.dynamic.ngi.it) |
| 13:11.32 | coppice | 24.99 is very expensive for one of those cards. 4.99 is more reasonable |
| 13:11.55 | CM3_1_2_632 | lol.....point me to the right direction please.... |
| 13:12.27 | coppice | you'll find some for that price. use them with the oslec EC module, and the results can be pretty good |
| 13:13.03 | CM3_1_2_632 | these final releases are surface mount components....better than those older releases with caps and shits on..... |
| 13:13.23 | *** join/#asterisk MaartenB (n=Maarten@213-73-177-32.cable.quicknet.nl) |
| 13:13.30 | MaartenB | hi all |
| 13:13.33 | coppice | I've never seen one that isn't surface mount |
| 13:14.08 | MaartenB | I have a problem with reconnecting to Asterisk, when I do asterisk -r, it does say "Unable to connect to remote asterisk", but asterisk is running |
| 13:14.32 | tzafrir_laptop | MaartenB, how can you tell asterisk is running? |
| 13:14.51 | CM3_1_2_632 | this is a X100P(C)....the X100P(A2) and X100A(B2) are not..... |
| 13:15.16 | tzafrir_laptop | "Unable to connect" - through /var/run/asterisk.ctl or /var/run/asterisk/asterisk.ctl |
| 13:16.56 | tzafrir_home | CM3_1_2_632, if you want to use us as better bulshit detectors, I suggest you feed us with proper data |
| 13:17.11 | tzafrir_home | links to the relevant ebay pages |
| 13:17.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:17.56 | MaartenB | tzafrir_laptop, it is working as normal, I can call everyone :) |
| 13:18.33 | CM3_1_2_632 | http://cgi.ebay.com/Digium-Wildcard-X100P-FXO-PCI-For-Asterisk-IP-PBX_W0QQitemZ150143708636QQihZ005QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 13:18.49 | CM3_1_2_632 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=150144750737&ssPageName=MERC_VI_RCRX_Pr3_PcY_BIN_Stores_IT&refitem=150143708636&itemcount=3&refwidgetloc=active_view_item&usedrule1=CrossSell_LogicX&refwidgettype=cross_promot_widget |
| 13:18.55 | *** join/#asterisk jmacz (n=jmacz@190.24.102.135) |
| 13:19.01 | MaartenB | tzafrir_laptop, it says "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)" |
| 13:19.11 | CM3_1_2_632 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=150144196217&ssPageName=MERC_VI_RCRX_Pr3_PcY_BIN_Stores_IT&refitem=150144750737&itemcount=3&refwidgetloc=active_view_item&usedrule1=CrossSell_LogicX&refwidgettype=cross_promot_widget |
| 13:19.39 | CM3_1_2_632 | bulshit detectors.....lofl |
| 13:21.15 | tzafrir_laptop | A lot of bullshit on that page |
| 13:21.56 | CM3_1_2_632 | for $24.99 they can bullshit all they want....all i care is 1 FXO port.... |
| 13:22.04 | CM3_1_2_632 | working....of course |
| 13:22.52 | tzafrir_laptop | First-off, it's not a Digium X100P. And them most they can say is that it is compatible with the discontinued junk card X100P of Digium |
| 13:22.59 | CM3_1_2_632 | looks like a descent card to me.... |
| 13:23.17 | coppice | choose your price - 14.99, 19.99 or 24.99 according to taste :-) |
| 13:23.24 | tzafrir_laptop | (as opposed to the TDMxxx cards of digium that are not junk) |
| 13:23.28 | CM3_1_2_632 | coppice: absolutely |
| 13:23.51 | CM3_1_2_632 | tzafrir_laptop: again, $24.99 mister.... |
| 13:23.59 | coppice | come on. the TDM400P isn't too great |
| 13:24.21 | tzafrir_laptop | well, but it's not like the X100P |
| 13:24.46 | *** join/#asterisk claudiotainen (n=claudiot@ppp-157-180.33-151.iol.it) |
| 13:24.51 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 13:25.08 | claudiotainen | I have a problem with voicemail configuring |
| 13:25.08 | CM3_1_2_632 | it doesn't have to be LIKE or identical.....it can look like CRAP but gives me a working FXO port they'll have my money for it.... |
| 13:25.19 | claudiotainen | can anyone help me ? |
| 13:25.43 | waKKu | claudiotainen whats problem ? |
| 13:25.47 | waKKu | i did it yesterday |
| 13:25.51 | tzafrir_laptop | well, the S&H that they charge is a fair price for that card :-) |
| 13:26.00 | tzafrir_laptop | claudiotainen, ask your question... |
| 13:26.42 | CM3_1_2_632 | tzafrir_laptop: only if you're in the states....i'm in Hong Kong |
| 13:27.29 | coppice | snap |
| 13:27.38 | CM3_1_2_632 | ouch |
| 13:27.52 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 13:28.15 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 13:28.34 | claudiotainen | oh sry |
| 13:28.36 | claudiotainen | :D |
| 13:28.46 | claudiotainen | I can't access my mailbox |
| 13:29.25 | waKKu | nice explanatory ... |
| 13:29.38 | claudiotainen | extensions.conf is configured so as to let the user access its mailbox when 9999 is dialed |
| 13:30.18 | claudiotainen | but then when i select the mailbox and type the password I always geta "login incorrect" message |
| 13:30.28 | waKKu | claudiotainen (i'm lazy today) http://www.the-asterisk-book.com/unstable/voicemail-einleitung.html |
| 13:31.09 | claudiotainen | thank you ;) |
| 13:31.19 | CM3_1_2_632 | waKKu: rofl..... |
| 13:33.31 | waKKu | :) |
| 13:33.55 | CM3_1_2_632 | TGIF everyone....let's go!!! =) |
| 13:34.22 | coppice | to bed? sure, in an hour or so :-) |
| 13:35.27 | *** join/#asterisk cVsup (n=cVsup@janeway2.mav.com.br) |
| 13:35.33 | *** part/#asterisk cVsup (n=cVsup@janeway2.mav.com.br) |
| 13:35.38 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:35.41 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 13:39.21 | Optic | hi sibbers |
| 13:42.31 | *** join/#asterisk friedrich| (n=friedric@e177240084.adsl.alicedsl.de) |
| 13:42.51 | tzafrir_home | claudiotainen, from which phone is it? |
| 13:43.24 | claudiotainen | i'm using ekiga |
| 13:43.38 | coppice | CM3_1_2_632: if you are using Wharf, aren't you using VoIP all the time? |
| 13:43.45 | tzafrir_home | maybe no DTMFs got to the voicemail app in the first place |
| 13:44.19 | claudiotainen | well I think they do |
| 13:44.33 | claudiotainen | because I can see debug messages in asterisk console |
| 13:44.57 | claudiotainen | which say "Incorrect password 1234 for user blablabla |
| 13:45.10 | claudiotainen | and 1234 is the password I dial |
| 13:45.37 | tzafrir_home | claudiotainen, so start with 'show voicemail users' to see if you have user 'blablabla' |
| 13:45.53 | claudiotainen | :) |
| 13:46.18 | enioreh | claudiotainen: is your voicemail context correct ? |
| 13:46.40 | claudiotainen | well that's a thing I'm not sure about |
| 13:47.01 | claudiotainen | the parameter "user" |
| 13:47.15 | claudiotainen | in voicemail extensions |
| 13:47.16 | enioreh | user@context |
| 13:47.18 | enioreh | from memory |
| 13:47.36 | claudiotainen | mmm |
| 13:47.42 | Strom_M | no |
| 13:47.43 | claudiotainen | hang on, I'll make a try |
| 13:47.46 | Strom_M | it's mailbox@context |
| 13:48.09 | Strom_M | in reference to the mailbox and associated context in voicemail.conf |
| 13:48.56 | wothinn | Has anyone had any luck with the message waiting indicator on the Polycom IP500? I've followed the suggestions at http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk but no dice and I'm not even sure how to begin debugging whether it's a * problem or a phone configuration problem. |
| 13:49.40 | Strom_M | wothinn: by default, it should work correctly with the included configs that come with the latest firmware |
| 13:49.53 | Strom_M | just set mailbox= in your sip.conf |
| 13:50.54 | wothinn | Hmm... unfortunately, I didn't find that to be the case. I'm running 2.1.2 and BootRom 3.2.3. |
| 13:51.12 | [TK]D-Fender | wothinn: PASTEBIN is your friend..... |
| 13:51.16 | [TK]D-Fender | claudiotainen: you too.... |
| 13:51.18 | [TK]D-Fender | ~pb |
| 13:51.18 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:51.26 | Strom_M | and you have the mailbox= setting set correctly in the sip.conf entry associated with the phone, right? |
| 13:51.57 | wothinn | http://pastebin.ca/635914 There we be. |
| 13:52.37 | skitfish | guys, I am trying to install app_amd.c in asterisk 1.4.8 |
| 13:52.51 | wothinn | I believe I do. I've tried the mailbox= with and without the @default. My extension number is the same as the mailbox number defined in voicemail.conf (voicemail *does* work flawlessly.) |
| 13:53.13 | skitfish | how can I do this? The script astxs seems to only be bundled with older versions of asterisk |
| 13:53.54 | claudiotainen | Fender: shouldn't I ask here? |
| 13:53.56 | file | skitfish: uh, app_amd is included with 1.4 |
| 13:54.12 | [TK]D-Fender | wothinn: Stop setting multiple entries up for the same phone! |
| 13:54.21 | skitfish | file: I'm following http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD |
| 13:54.28 | [TK]D-Fender | claudiotainen: you should be SHOWING us the problem. ALL of it. |
| 13:54.29 | wothinn | One's a user and one's a peer. Don't I need that for both inbound and outbound? |
| 13:54.55 | Strom_M | wothinn: for a telephone set, try setting a single "type=friend" entry instead |
| 13:54.58 | claudiotainen | ok |
| 13:55.04 | file | skitfish: well those instructions are for 1.2... you don't have to do anything with 1.4 as it is included |
| 13:55.10 | [TK]D-Fender | wothinn: here : http://pastebin.ca/635916 |
| 13:55.12 | wothinn | OK... I'll futz with that and see if it helps. Thanks for the suggestion. |
| 13:55.15 | skitfish | file: ok thanks |
| 13:55.30 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 13:55.44 | [TK]D-Fender | Strom_M: Friend in 1.4 is brolen for presence, and they've blurred the line of "type" badly... |
| 13:55.58 | wothinn | [TK]D-Fender: That just leaves me with a peer? Will that still allow both calls in and calls out? |
| 13:56.10 | [TK]D-Fender | file: Seriously... do you know why they mucked up "friend" and "peer" so bad? It auths PEERS on incoming calls! |
| 13:56.17 | [TK]D-Fender | wothinn: You on 1.4? |
| 13:56.20 | *** join/#asterisk SuurMyy (n=guacamol@195.238.211.98) |
| 13:56.22 | wothinn | Yep. 1.4.9 |
| 13:56.27 | [TK]D-Fender | wothinn: then it'll work |
| 13:56.31 | wothinn | Neat! Thanks. |
| 13:56.45 | file | [TK]D-Fender: the world of SIP doesn't fit into a user/peer universe |
| 13:56.56 | [TK]D-Fender | file: It seemed so clean before... |
| 13:57.19 | coppice | Still Incomplete Protocol |
| 13:57.20 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:57.58 | *** join/#asterisk karrotx (n=karrotx@ip67-91-24-2.z24-91-67.customer.algx.net) |
| 13:58.12 | [TK]D-Fender | coppice: Slightly Irritating Problem |
| 13:58.58 | Optic | ha ha |
| 13:59.02 | Optic | nice ones :) |
| 13:59.18 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:59.18 | Strom_M | Innovation Subscribers Don't Need |
| 13:59.24 | Strom_M | oh wait, wrong acronym |
| 13:59.41 | Optic | ISDN made things eaiser |
| 13:59.48 | Strom_M | it was a joke, Optic |
| 13:59.53 | Optic | hehe |
| 14:00.09 | coppice | Strom_M: It wasn't a joke. it was dead on |
| 14:00.16 | Strom_M | perhaps I should have said "Guaranteed Trouble Everytime" instead |
| 14:00.54 | *** join/#asterisk ManxPower (n=manxpowe@015-847-806.area5.spcsdns.net) |
| 14:01.26 | *** join/#asterisk jmacz (n=jmacz@201.244.175.40) |
| 14:02.23 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 14:03.40 | creativx | Strom_M: NT the |
| 14:03.41 | creativx | n |
| 14:03.45 | creativx | No Trouble? |
| 14:05.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:06.39 | *** join/#asterisk Grnd-Wire (n=groundwi@71-35-110-216.tukw.qwest.net) |
| 14:06.44 | Grnd-Wire | Good morning everyone! |
| 14:06.50 | Grnd-Wire | Has anyone ever worked with IAXmodem ? |
| 14:07.21 | coppice | nope. nobody ever did that |
| 14:07.33 | Grnd-Wire | uh huh |
| 14:07.39 | Optic | hi hi hi |
| 14:13.53 | blitzrage | It Still Doesn't Work |
| 14:14.08 | Strom_M | Get Telephone Elsewhere |
| 14:14.30 | blitzrage | all your base are belong to us |
| 14:15.18 | j-goddess | hehe |
| 14:15.21 | j-goddess | blitz |
| 14:15.29 | blitzrage | <-- blitzrage |
| 14:15.30 | [TK]D-Fender | take off every zig |
| 14:15.42 | [TK]D-Fender | you have no chance to survive! |
| 14:15.46 | [TK]D-Fender | make your time! |
| 14:15.51 | pepse | oops |
| 14:15.55 | pepse | for great justice! |
| 14:15.57 | pepse | damnit :) |
| 14:16.02 | [TK]D-Fender | <- knows too much, and the ninjas have already been dispatched to kill me |
| 14:16.17 | coppice | Hurray! |
| 14:16.18 | Mercestes | what's taking them so damn long? |
| 14:16.39 | blitzrage | [TK]D-Fender: ninja force 1; en route |
| 14:16.47 | pepse | my favorite of those pics was the one with Hitler surrounded by his men. He's pointing at a map with the caption "So you see, General.. If we set them up the bomb here, all their base will belong to us." |
| 14:17.11 | [TK]D-Fender | Strom_M: Took a look yesterday for the camera, and settled on the Panasonic DMC-FZ8KK I was looking at (got 14 days to check her out before returning if needed) |
| 14:17.20 | enioreh | pepse: if you have a link, i would enjoy getting it :) |
| 14:17.34 | [TK]D-Fender | Strom_M: Took a few shots with it, VERY intuitive |
| 14:17.42 | pepse | enioreh: it's been a long while, maybe i can find it again |
| 14:17.59 | j-goddess | o.O |
| 14:18.01 | pepse | [TK]D-Fender: I've noticed that Panasonic cameras take -amazing- pictures. Even ones on cellphones. |
| 14:18.40 | [TK]D-Fender | pepse: Only cell-phone I want is about a few months away :) |
| 14:18.52 | claudiotainen | I've "pastebined" my problem ;) |
| 14:18.55 | claudiotainen | http://pastebin.com/m4a98faf3 |
| 14:19.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:19.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:19.10 | pepse | yeh but the point is, Panasonic makes bad ass cameras. I bought my mom a cheap Lumix, and it's -awesome- |
| 14:19.13 | coppice | cameras don't take amazing pictures. people do |
| 14:19.32 | blitzrage | Guns don't kill people. I kill people. |
| 14:20.03 | coppice | Guns don't kill people. Hard work does |
| 14:20.07 | Grnd-Wire | [TK]D-Fender: So I got in my polycom 320 last night.. |
| 14:20.17 | [TK]D-Fender | claudiotainen: Using 1.4? |
| 14:20.27 | Grnd-Wire | [TK]D-Fender: No freakin' power brick! I'm so spoiled from Aastra.. and I don't have a PoE switch in my lab yet.. UGH! |
| 14:20.48 | pepse | does poe require a special cable? |
| 14:20.57 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-221-126.dsl.irvnca.pacbell.net) |
| 14:21.20 | [TK]D-Fender | claudiotainen: claudiotainen: exten => 9999,1,VoiceMailMain(${CALLERIDNUM}) <- this variable is DEPRECATED. Second, you are NOT using [default] as your context in voicemail.conf so you MUST specify it in VoicemailMain |
| 14:21.45 | [TK]D-Fender | claudiotainen: exten => 9999,1,VoiceMailMain(${CALLERID(num)}@sip_calls_vm) |
| 14:22.09 | claudiotainen | oh |
| 14:22.09 | blitzrage | pepse: nope |
| 14:22.16 | claudiotainen | thank you :) |
| 14:22.19 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:22.25 | mercestes | yay, I'm back |
| 14:22.27 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:22.27 | [TK]D-Fender | pepse: Not for that phone or any other standard 802.3af native phone |
| 14:22.28 | blitzrage | boooo |
| 14:22.56 | pepse | What about the Cisco 7910? :) |
| 14:23.14 | Grnd-Wire | [TK]D-Fender: Even though Polycom appears to have a "PoE cable" that allows you to plug the brick into the network cable, so that there is still one cable coming up on your desk. |
| 14:23.46 | *** join/#asterisk wunderkin (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 14:23.51 | [TK]D-Fender | Grnd-Wire: thats on the 30X & 50X |
| 14:24.07 | ghenry | Polycom ip501 a safe bet? |
| 14:24.11 | Grnd-Wire | [TK]D-Fender: ok, so it's only half of their product line then.. :D |
| 14:24.30 | [TK]D-Fender | pepse: 7910 is a BAD choice, and uses CDP Cisco PoE |
| 14:24.31 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:24.44 | [TK]D-Fender | Grnd-Wire: 301 is being phased out, 501 following. |
| 14:24.57 | pepse | [TK]D-Fender: yeah, i happened to get one for free. and I have a PoE adapter, but a regular cable doesn't work with it |
| 14:25.09 | pepse | [TK]D-Fender: what does CDP mean? |
| 14:25.11 | [TK]D-Fender | Grnd-Wire: And its 1/4 of their line ;) |
| 14:25.19 | JT | googleit |
| 14:25.20 | [TK]D-Fender | pepse: Cisco Discovery Protocol. |
| 14:25.22 | Zeeek | hello men |
| 14:25.35 | bakermd | I have 3 Cisco 7940's that were given to me - and I have tried 6 different FW versions, but cannot get these phones to work with Asterisk. I hate Cisco. My Polycoms work flawlessly (60+ of them now) |
| 14:26.16 | Zeeek | hello women and children |
| 14:26.19 | pepse | bakermd: i got my 7910 up in no time, i don't even have a tftp server or config for the phone :) |
| 14:26.39 | bakermd | This one started with firmware from 2002... |
| 14:26.53 | pepse | the phone had some 169.254 address set as its call manager, so I aliased that ip on * |
| 14:27.15 | pepse | (and uncommented some stuff in skinny.conf) |
| 14:27.34 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:27.48 | pepse | pack of sunflower seeds for me |
| 14:27.53 | Grnd-Wire | heh - There's nothing like changing the configuration of your network to match a device that is improperly configured.. :D |
| 14:28.19 | Zeeek | pizza? Licorice? Chocolate? TV guide? a hooker? |
| 14:28.21 | pepse | Grnd-Wire: there's nothing like slackin :D |
| 14:28.31 | russellb | beer for me, as well. |
| 14:28.35 | Zeeek | noted. |
| 14:28.59 | Zeeek | In this country we can choose between beer at 4,5 or 12%. WHich do you want? |
| 14:29.07 | russellb | definitely 12 |
| 14:29.17 | Zeeek | 1/2 litre or one full liter |
| 14:29.23 | russellb | full! |
| 14:29.33 | coppice | one half empty litre |
| 14:29.35 | Zeeek | russell has left the building |
| 14:29.48 | pepse | [TK]D-Fender: hm, so does cdp mean that even if I do make a proper cable for my generic PoE adapter, that it still may not work with the 7910? |
| 14:30.09 | Zeeek | coppice, an order of porc au caramel for you? |
| 14:30.17 | blitzrage | Zeeek: Canada has that too! :) |
| 14:30.34 | Zeeek | yes, only the US has 3.2% beer. Thats sooooo wimpy |
| 14:30.57 | Nugget | The US has the same range of beers that everyone else has. |
| 14:31.00 | Zeeek | You have to drink about eleventy gallons to get a buzz |
| 14:31.02 | [TK]D-Fender | pepse: It means that A) there is no "special cable, and B) a CDP phone will not work on 802.3af |
| 14:31.10 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:31.12 | [TK]D-Fender | b ) |
| 14:31.17 | Nugget | there's just a silly law that you have to call it 'malt liquor' if the alcohol percentage is above a certain level |
| 14:31.38 | blitzrage | anyone know if you're using sippeers and sipusers from realtime, if call-limit (inuse) will update correctly when you send a call (and answer) to a queue member? |
| 14:31.39 | Zeeek | and it can't be sold after 12AM in some states or at all on SUndays in others |
| 14:31.45 | pepse | [TK]D-Fender: this article suggests some phones work with this special cable.. http://www.voip-info.org/wiki-Cisco+POE |
| 14:31.51 | [TK]D-Fender | Zeeek: 12% beer? |
| 14:32.00 | Zeeek | yeah guaranteed headache |
| 14:32.01 | blitzrage | since I can't use 'show show peer <foo>' or 'sip show inuse' because the peers are not cached |
| 14:32.13 | Zeeek | >SIP SHOW BEERS |
| 14:32.15 | blitzrage | no headache if you don't go nuts |
| 14:32.23 | _DAW | Zeek.. Louisiana has something like 5.4%.. and drinking on sundays is manadatory :) |
| 14:32.28 | pepse | but later says "the phones issue a Cisco Discovery Protocol (CDP) during boot up to try to ascertain whether POE (or more correctly inline power) is available" |
| 14:32.29 | Zeeek | heh |
| 14:32.29 | blitzrage | 1 litre bottle of Labatt Ice Max is pretty much all you need :) |
| 14:32.54 | blitzrage | Labatt Maximum Ice* |
| 14:33.02 | Zeeek | very nice |
| 14:33.19 | j-goddess | now if only we could incorporate Beer.conf |
| 14:33.31 | j-goddess | show beer inuse |
| 14:33.33 | Zeeek | #include beer.conf |
| 14:33.40 | j-goddess | hehe |
| 14:33.48 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.135.109) |
| 14:33.53 | Zeeek | reload glasses |
| 14:34.03 | pepse | beer refill now |
| 14:34.09 | j-goddess | beer show netstats |
| 14:34.11 | [TK]D-Fender | pepse: ... |
| 14:34.12 | Zeeek | stop and drink gracefully |
| 14:34.14 | [TK]D-Fender | ~wglwat |
| 14:34.14 | jbot | hmm... wglwat is well, good luck with all that |
| 14:34.46 | j-goddess | boip |
| 14:34.46 | *** join/#asterisk RSAMan (n=a@dsl-242-44-247.telkomadsl.co.za) |
| 14:34.52 | RSAMan | hiyas guys |
| 14:34.53 | Zeeek | nice |
| 14:34.58 | pepse | are you saying that's bs or you just haven't seen such things? |
| 14:34.59 | pepse | :) |
| 14:35.29 | Grnd-Wire | [TK]D-Fender: hey - Is there a way to set a Zap channel to "administratively down" (busy/unavailable_, so that all of the dialplan elements that use it will skip it for outgoing routes? |
| 14:35.39 | [TK]D-Fender | pepse: I'm saying that this seems hit/miss with contradictory results. YMMV and "good luck" |
| 14:35.53 | pepse | nod. |
| 14:35.56 | [TK]D-Fender | Grnd-Wire: Not that I'm aware of |
| 14:36.04 | pepse | i'm not expecting it to work, but it's worth a try cause i don't have a power supply of my own |
| 14:36.12 | [TK]D-Fender | pepse: Another great reason to not buy Cisco |
| 14:36.26 | pepse | good thing it was free :) |
| 14:36.32 | Zeeek | has anyone ever found a way to have asterisk CLI continue to work when the network is down? Is there any way this could happen? |
| 14:36.34 | anonymouz666 | pepse: sip show beers |
| 14:36.45 | Grnd-Wire | pepse: You could probably ebay the phone dude.. :P |
| 14:37.33 | pepse | probably. but i've got it in my hands |
| 14:37.42 | pepse | if i ebay it then i'd have to look for a different phone |
| 14:37.53 | Zeeek | coppice a lot of people would like to talk about faxing in asterisk on the conference. I donb't suppose you'd consider.... |
| 14:38.02 | Grnd-Wire | heh.. You can buy an Aastra for half as much as you would get for selling it.. or ahem, a Polycom.. :P |
| 14:40.09 | pepse | hehe looking at ebay, i take it aastra makes a lot of non-voip phones |
| 14:40.27 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 14:40.40 | Grnd-Wire | yes.. |
| 14:40.52 | BSD_Tech | who what where when how why who with for what reason |
| 14:40.56 | coppice | Zeeek: people like talking. doing is another matter :-) |
| 14:41.17 | Zeeek | coppice at least a few questions could be answered. I wish it could happen |
| 14:41.31 | Zeeek | but I'm sure it's either too late or you're too busy |
| 14:41.49 | Zeeek | I haven't tried to update for a while now |
| 14:41.55 | pepse | 7910s seem to only be going for around 30 bux |
| 14:42.01 | Zeeek | I'm trying to opush our clients into the 21st century |
| 14:43.13 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:43.35 | pepse | heh, love this listing. "Aastra IP Phone".. yet.. "Compatible with Centrex, PBX, or standard telephone service" |
| 14:43.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:45.41 | *** join/#asterisk saftsack (n=saftsack@pD9E072E0.dip.t-dialin.net) |
| 14:47.20 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:51.12 | mercestes | define "compatible." |
| 14:51.26 | mercestes | because given that ad, my tin can and string is compatible with standard phone service. |
| 14:52.25 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 14:52.48 | *** join/#asterisk juanmanuel (n=jmacz@190.24.102.180) |
| 14:52.48 | mvanbaak | mercestes: wow, you still have a working set of those ? |
| 14:55.49 | Capps- | how do you change your ringer on asterisk? it the ring sounds the same as when you're calling someone so i never know if my phone is ringing or if the person next to me is calling someone on speakerphone [without me having to look at the phone.] |
| 14:55.54 | RSAMan | questio : whats is this asterisk gui... i am a new user, not lazy .. on page 130 for thebook |
| 14:56.09 | RSAMan | n |
| 14:56.14 | RSAMan | just curious |
| 14:56.23 | RSAMan | is the asterisk gui even worth looking at |
| 14:56.25 | RSAMan | ? |
| 14:56.30 | Capps- | yes. |
| 14:56.48 | RSAMan | how do i open it |
| 14:56.48 | RSAMan | ? |
| 14:56.48 | [TK]D-Fender | Capps-: Asterisk doesn't have a "ringer", your PHONE does. |
| 14:57.05 | [TK]D-Fender | RSAMan: Best forgotten |
| 14:57.11 | *** join/#asterisk sashion (n=sdgsdg@dsl-244-217-50.telkomadsl.co.za) |
| 14:57.25 | [TK]D-Fender | RSAMan: Its not part of the core setup, its an add-on. |
| 14:57.32 | Capps- | [TK]D-Fender: right. but it says it gets them from the asterisk server. |
| 14:57.33 | RSAMan | [TK]D-Fender: Understood, trying to learn the right way |
| 14:57.35 | sashion | what is the equivalent of GROUP_COUNT in asterisk 1.4 ? |
| 14:58.17 | [TK]D-Fender | Capps-: what says "get them" from "asterisk server"?! |
| 14:58.37 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 14:59.03 | Capps- | [TK]D-Fender: i fail at life. :[ |
| 14:59.19 | mvanbaak | http://xkcd.com/281/ <--- gheh, that could be me |
| 14:59.26 | mvanbaak | I'm waiting for my thinkpad to arrive |
| 15:01.06 | Nivex | that was me a couple weeks ago. woot.com sent me something via slowboat |
| 15:01.49 | mvanbaak | my thinkpad is in .nl already |
| 15:01.56 | mvanbaak | like 50 miles from here |
| 15:02.03 | mvanbaak | but it arrived there today |
| 15:02.13 | mvanbaak | so prolly wont be at my place before monday |
| 15:02.15 | mvanbaak | UGH |
| 15:02.27 | mvanbaak | I phoned them, and I cannot pickup the thing myself |
| 15:02.38 | mvanbaak | "it's all automagically stuff" |
| 15:02.40 | mvanbaak | dammit |
| 15:04.57 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 15:09.17 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
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| 15:14.14 | claudiotainen | a quick question about asterisk console |
| 15:14.15 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:14.48 | claudiotainen | is there a command to see which RTP ports are in use ? |
| 15:16.57 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 15:19.33 | russellb | claudiotainen: don't think there is a command for that |
| 15:20.27 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 15:21.04 | waKKu | u can check/define that range of ports to rtp on rtp.conf - if it helps |
| 15:21.08 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 15:21.42 | Qwell[] | russellb: are you back? |
| 15:22.19 | *** join/#asterisk jmacz (n=jmacz@190.25.32.86) |
| 15:22.21 | *** join/#asterisk rushowr (n=rushowr@cpe-65-24-149-191.columbus.res.rr.com) |
| 15:22.25 | *** part/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 15:22.32 | russellb | Qwell[]: no sir |
| 15:22.33 | russellb | just bored. |
| 15:22.36 | Qwell[] | heh |
| 15:23.32 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:24.55 | BSD_Tech | thiings/thinks |
| 15:25.03 | [TK]D-Fender | claudiotainen: While not "direct" you could dump a list of channels, and then cycle though each probably. |
| 15:25.13 | [TK]D-Fender | BSD_Tech: Good idea |
| 15:25.27 | russellb | BSD_Tech: see DUNDi :) |
| 15:25.29 | [TK]D-Fender | BSD_Tech: I've been working on a private #asterisk MeetMe for us to chat with. |
| 15:25.47 | [TK]D-Fender | russellb: Its an organization question, not a technological one :) |
| 15:26.17 | russellb | there are 2 global DUNDi networks ... dundi-test (easy to join) ... e164 (more difficult to join, especially for smaller installs) |
| 15:26.59 | russellb | there was just never a really good process for people to find peers for e164 |
| 15:27.25 | russellb | but people terminate millions of minutes over those networks ... |
| 15:28.49 | mocker | Anyone know much about http://www.voipsupply.com/product_info.php?products_id=2419&searchid=341962 ? |
| 15:31.08 | [TK]D-Fender | mocker: I'd go for an A200d instead in PCI..... |
| 15:31.29 | [TK]D-Fender | mocker: card is HUGE |
| 15:31.38 | mocker | [TK]D-Fender: That's my concern. :( |
| 15:31.46 | mocker | I don't know if it will *fit* in 2950 |
| 15:31.47 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 15:32.24 | BSD_Tech | brb |
| 15:32.25 | [TK]D-Fender | mocker: http://www.voipsupply.com/product_info.php?products_id=1341 |
| 15:33.02 | mocker | Yeah, that's much smaller. |
| 15:33.07 | mocker | + echo cancellation. |
| 15:33.15 | [TK]D-Fender | mocker: Cheaper here : http://www.telephonydepot.com/ProductDetails.asp?ProductCode=105%2D052%2DA200BRME |
| 15:33.16 | sevard | mercestes: he just ignores my love :( |
| 15:33.23 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 15:33.46 | mocker | [TK]D-Fender: Uh, by about *half* |
| 15:33.53 | mocker | wtfbbqvoipsupplymarkup |
| 15:33.55 | mocker | :) |
| 15:33.59 | mercestes | sevard: You desrve better. |
| 15:34.00 | [TK]D-Fender | mocker: No, you have to pick your modules. |
| 15:34.06 | mocker | Ahh. |
| 15:34.10 | [TK]D-Fender | mocker: and update the price. |
| 15:34.14 | sevard | mercestes: *wipes tears away* i'm strong, i'm strong. |
| 15:34.20 | [TK]D-Fender | mocker: But still a good bit cheaper |
| 15:34.29 | [TK]D-Fender | sevard: WUSS :p |
| 15:34.40 | [TK]D-Fender | load chan_needy.so |
| 15:34.57 | Zeeek | WHo ordered the turkey on rye? Here's the beer and the toffee, the decaf espresso and the condoms |
| 15:34.58 | mocker | [TK]D-Fender: That should work in a 2950, agree? |
| 15:35.40 | [TK]D-Fender | mocker: should work jsut fine |
| 15:35.56 | [TK]D-Fender | mocker: as long as you've got the 2 blackplane spaces for it |
| 15:36.01 | sevard | Zeeek: DECAF ESPRESSO |
| 15:36.12 | Zeeek | [TK]D-Fender your Polycom was too heavy to carry, it'll be delivered |
| 15:36.23 | sevard | wtf. |
| 15:36.30 | mercestes | sevard: Wouldn't that be an empty cup of air? |
| 15:36.31 | [TK]D-Fender | Zeeek: ? |
| 15:37.04 | sashion | i have a snome 320 which is mutli-lined. In a queue enviroment, this phone keeps accepting calls even when you are on the line. is there a way in 1.4 to limit amount of calls to specific peers ? |
| 15:37.24 | sevard | like that, ladies? |
| 15:37.32 | sashion | in asterisk@home, they used GROUP_COUNT < 1... is there a similar feature in 1.4 ? |
| 15:38.23 | fetcher | sashion: call-limit=1 in sip.conf? That might also restrict simultaneous outgoing calls, though |
| 15:39.24 | sashion | fetcher: its an option.. will enable and see how things go :P thanks |
| 15:41.27 | *** join/#asterisk jmacz (n=jmacz@190.25.32.86) |
| 15:42.18 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:42.27 | mercestes | POlycom custom ringtones are just a .wav file, aye? |
| 15:42.33 | polerin | meh |
| 15:42.33 | mercestes | and then some sip.cfg magick. |
| 15:42.38 | polerin | MERCESTEDESESDESDFDS |
| 15:42.39 | mercestes | Polerin!!!! |
| 15:42.45 | polerin | YESESDLKFJDSEEK |
| 15:42.50 | polerin | look at something for me? |
| 15:42.58 | mercestes | hrm? |
| 15:42.59 | polerin | ack |
| 15:43.05 | mercestes | what am I looking at? |
| 15:43.08 | polerin | msg. |
| 15:43.46 | mercestes | K, I put on my wizard hat. |
| 15:44.28 | *** part/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 15:46.45 | claudiotainen | just another [stupid] qquestion about ekiga and sip |
| 15:47.31 | claudiotainen | RTP packets are sent directly betwwen 2 user agents, the don't pass through asterisk, right ? |
| 15:47.43 | claudiotainen | I mean they're not "filtered" by the server |
| 15:47.48 | enioreh | claudiotainen: it depends on the option of your dial |
| 15:47.56 | [TK]D-Fender | claudiotainen: If they are going DIRECT, then they can't be going through ASTERISK. |
| 15:48.12 | [TK]D-Fender | claudiotainen: What colour was Napoleon's white horse? :) |
| 15:48.20 | claudiotainen | ehm ... |
| 15:48.24 | claudiotainen | don't know |
| 15:48.30 | claudiotainen | ;) |
| 15:48.37 | [TK]D-Fender | SMRT |
| 15:48.42 | *** part/#asterisk ManxPower (n=manxpowe@015-847-806.area5.spcsdns.net) |
| 15:48.51 | claudiotainen | the thing is how can I set to DIRECT ? |
| 15:49.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:49.28 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 15:50.00 | *** join/#asterisk VoicePulse_ (n=contact@unaffiliated/voicepulse) |
| 15:51.07 | lirakis | i just started having problems with my asterisk setup... when i make or recieve calls .. the call flows fine.. and eventually it says "attempting native bridge of SIP/4321-099c42d8 and SIP/1234-099d9110" .. and it never completes the bridge.. so both ends of the call just sit there |
| 15:51.15 | [TK]D-Fender | claudiotainen: "canreinvite=yes" for each of the people yuo want to be capable of doing this, an you must NOT be using any of "tTwW" or other dynamic features in your Dial command |
| 15:51.26 | enioreh | canreinvite=yes |
| 15:51.37 | lirakis | enioreh: .. to me? |
| 15:51.40 | [TK]D-Fender | lirakis: PASTEBIN the call, with sip debug |
| 15:51.58 | Zeeek | what the? I leave for az few minutes, and when I come back, you guys are actually talking about asterisk. What's up? |
| 15:52.03 | enioreh | lirakis: no, to claudiotainen |
| 15:52.28 | lirakis | k |
| 15:52.58 | enioreh | lirakis: both user agent are in the lan ? |
| 15:54.06 | lirakis | enioreh: no .. i havent tried ext->ext .. this is happeneing with every call i get/make to pstn though through my sip provider |
| 15:54.57 | enioreh | lirakis: this perhaps may be related to some nat issue |
| 15:55.16 | lirakis | enioreh: my pbx is on public ip |
| 15:55.50 | lirakis | enioreh: and it just started happenning.. things have been working for months before this |
| 15:56.02 | lirakis | .. ill get together a call log with sip debug |
| 15:56.22 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 15:57.02 | lirakis | arg.. its being bizarre.. |
| 15:57.08 | lirakis | it takes a long time to reload |
| 15:57.09 | lirakis | .. etc. |
| 15:57.20 | lirakis | .. its very strange whats started happening |
| 15:57.59 | enioreh | lirakis: what have you change last before it started failing ? |
| 15:58.08 | mocker | [TK]D-Fender: You've been blogged! http://www.mocker.org/sangoma-a200-vs-a400-cards/ :) |
| 15:58.41 | lirakis | enioreh: .. i cant think of anything honestly.. i dont poke at it b/c its my day to day phone system.. not a hobby/toy |
| 15:59.04 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.52.210) |
| 15:59.04 | [TK]D-Fender | mocker: I've got my own :) |
| 15:59.27 | enioreh | lirakis: can't help you much :/ |
| 15:59.28 | lirakis | enioreh: im wondering if im experiecing problems due to a pre-hardware failure... .. aka a disk or some thing is going bad |
| 15:59.47 | enioreh | lirakis: dmesg is your friend to know about hard disk failure |
| 15:59.57 | mocker | [TK]D-Fender: You have a blog now? |
| 16:00.03 | [TK]D-Fender | ~sipnat |
| 16:00.04 | 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 |
| 16:00.09 | [TK]D-Fender | above |
| 16:00.20 | [TK]D-Fender | mocker: I started it for things like that |
| 16:00.31 | mocker | [TK]D-Fender: What's the url? |
| 16:00.37 | mocker | I'll add you to my blogroll. |
| 16:00.40 | enioreh | time to go home , hooray \o/ |
| 16:02.31 | *** join/#asterisk awk (n=awk@kia.inet-corp.com) |
| 16:02.36 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:03.48 | Zeeek | Consider joining us at #asterisk-users-conference if only to heckle us! |
| 16:04.02 | *** part/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:04.05 | lirakis | [TK]D-Fender: http://pastebin.ca/636052 |
| 16:04.05 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 16:04.05 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:04.08 | Zeeek | Women are admitted FREE TODAY ONLY |
| 16:04.37 | Zeeek | Yes, it's ladies day at #asterisk-users-conference |
| 16:04.38 | lirakis | thats a copy paste of the cli with sip debug .. during my call... i called in to my did... then dialed extension 5000 |
| 16:07.51 | *** join/#asterisk jmacz (n=jmacz@190.24.102.76) |
| 16:08.52 | *** join/#asterisk friedrich| (n=friedric@e177240084.adsl.alicedsl.de) |
| 16:09.51 | lirakis | [TK]D-Fender: i just copied my old extensions.conf to a backup file.. and put together a really bare bones one that does ext->ext dialing, recieves 1 did and rings it strait to an ext, and allows dialing out to m sip provider. |
| 16:09.56 | lirakis | .. it seems to be working now |
| 16:10.34 | lirakis | ... i had been asking yesterday.. about whether using Answer() in the wrong places can make a call do wierd things.. any insight on that |
| 16:15.52 | *** join/#asterisk jcaceres (n=jcaceres@190.41.82.1) |
| 16:16.54 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 16:17.27 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 16:21.39 | *** join/#asterisk GMouse (n=michael@cpe-24-208-180-140.columbus.res.rr.com) |
| 16:21.53 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 16:21.57 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 16:22.24 | andresmujica | hi.. how can i calculate the use percentage for an ISDN channel?? |
| 16:25.00 | Hmmhesays | troubleshooting voice quality issues realy sucks |
| 16:25.02 | Hmmhesays | *really |
| 16:26.09 | irule | directory app spells names, how can I make it say the name instead with festival? |
| 16:27.32 | *** join/#asterisk taqua2008 (n=perdue@66.118.69.58) |
| 16:28.06 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 16:28.32 | jcaceres | hello, i am wringt from peru, i want to know if there is any troble in using a TE120P with an SMP kernel? |
| 16:29.00 | BSD_Tech | not if you compiled the driver on smp |
| 16:30.47 | Zeeek | Asterisk Users Conference right now |
| 16:31.06 | jcaceres | tnks BSD_Tech |
| 16:31.07 | Zeeek | join #asterisk-users-conference |
| 16:38.06 | irule | how can I search for ${EXTEN:3} and other options regarding the removal of digits in a number? I want to remove the first 3 digits, not the ones on the end, and learn a little more on the way, thankis |
| 16:39.08 | Strom_M | irule: search for "asterisk substrings" |
| 16:39.25 | irule | thanks |
| 16:39.44 | *** join/#asterisk MdeP (n=mdep@200.124.36.28) |
| 16:40.32 | jcaceres | hello i am trying to configure and TE120 and a TDM400 with one fxo and one fxs module |
| 16:40.48 | jcaceres | i have compiled zaptel correctly |
| 16:41.13 | jcaceres | when i do ztcfg i get all the channel configured |
| 16:42.13 | jcaceres | and i have configured in my zapata.conf the fxo and fx channel |
| 16:42.14 | jcaceres | <PROTECTED> |
| 16:42.18 | *** join/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 16:42.36 | n00dle | OT: anyone here use bitpim on linux? |
| 16:42.43 | jcaceres | <PROTECTED> |
| 16:42.49 | [TK]D-Fender | irule: ${EXTEN:3 }**DOES** remove the first 3.... |
| 16:43.08 | sashion | jcaceres: module load chan_zap.so |
| 16:43.09 | jcaceres | and whe i try to load the module chan_zap.so |
| 16:43.11 | irule | or the other way arround |
| 16:43.15 | jcaceres | i get this error |
| 16:43.18 | jcaceres | Unable to open channel 35: Device or resource busy |
| 16:43.20 | [TK]D-Fender | jcaceres: because Asterisk has to be compiled AFTER Zaptel. |
| 16:43.31 | Strom_M | [TK]D-Fender: WRONG |
| 16:43.39 | Strom_M | it's a configuration issue |
| 16:43.46 | [TK]D-Fender | Strom_M: News to me after so much I've read and done... |
| 16:43.47 | Strom_M | jcaceres: in what order did you load the drivers? |
| 16:44.05 | Strom_M | [TK]D-Fender: he has chan_zap, but it's not loading correctly |
| 16:44.11 | Strom_M | therefore his compile order is correct |
| 16:44.11 | jcaceres | the driver are loaded automaticale by zaptel |
| 16:44.37 | jcaceres | the modules are loaded automaticale by zaptel |
| 16:44.38 | Strom_M | jcaceres: you have 1-31 on the TE120 ad 32-25 on the TDM400? |
| 16:44.58 | Strom_M | s/ad/and/ |
| 16:46.04 | jcaceres | 1-31 te1 and 34 adn 35 fxo and fxs respecctibly |
| 16:46.25 | Strom_M | ok...did you make sure to connect the molex connector on the TDM400? |
| 16:46.58 | jcaceres | yes, the card is powerd |
| 16:47.01 | Strom_M | ok |
| 16:47.02 | jcaceres | powered |
| 16:47.14 | jcaceres | i also get this warning |
| 16:47.18 | jcaceres | already have an application 'ZapSendKeypadFacility |
| 16:47.30 | Strom_M | stop all asterisk processes |
| 16:47.45 | Strom_M | unload the zaptel kernel modules |
| 16:47.54 | Strom_M | and then load zaptel, wcte12xp, and wctdm |
| 16:48.05 | Strom_M | start asterisk again and see if you get the same error |
| 16:48.16 | Strom_M | if you do, pastebin your configuration files |
| 16:48.31 | sashion | jcaceres: what do you get when you run ztcfg -vvv |
| 16:49.31 | jcaceres | i get all channels configured, but the that resuls has been changing from time to time |
| 16:50.04 | Strom_M | jcaceres: please follow my directions |
| 16:50.21 | jcaceres | i also get an error about rtc:lost imterupts at 1024hz |
| 16:50.44 | jcaceres | ok Strom_M i'll doit |
| 16:51.47 | Strom_M | a poem for Quebec: |
| 16:51.58 | jcaceres | Strom_M, when u say unload zaptel kernel u men /etc/init.d/zaptel stop |
| 16:52.06 | Strom_M | rmmod |
| 16:52.12 | Strom_M | and modprobe |
| 16:52.14 | jcaceres | <PROTECTED> |
| 16:52.20 | jcaceres | oks |
| 16:52.26 | Strom_M | for this, do it one module at a time |
| 16:52.38 | Strom_M | a poem for Quebec: |
| 16:52.46 | Strom_M | Bonjour a votre lait homo |
| 16:52.53 | Strom_M | Je compose mauvais numero |
| 16:53.03 | Strom_M | Je me souviens sexe du chat |
| 16:53.05 | Strom_M | fin. |
| 16:55.01 | [TK]D-Fender | Merci.. salut la visite! |
| 16:55.24 | jcaceres | when i did modprobe wctd12xp i got : ZT_CHANCONFIG failed on channel 34: No such device or address (6) |
| 16:55.26 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 16:55.27 | b1shop | [TK]D-Fender: how it going TK? |
| 16:55.39 | Strom_M | jcaceres: that's fine; keep going on to wctdm |
| 16:55.43 | [TK]D-Fender | b1shop: Still breathing.... 3.5 hours to go! |
| 16:55.48 | b1shop | heh |
| 16:56.28 | jcaceres | and why is that the order in the chanes varies, some times te120 is first then is tdm400 |
| 16:56.28 | irule | what is the channel to the talkshow thing? |
| 16:56.35 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 16:56.49 | b1shop | is it possible to allow someone's vm greeting to allow them to xfer to another extension? |
| 16:57.16 | Strom_M | jcaceres: it depends on the driver load order |
| 16:57.33 | Strom_M | jcaceres: The first card you load the driver for will get channels starting with 1 |
| 16:58.07 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 16:58.24 | tzafrir_laptop | jcaceres, remove the install line for wctd12xp in /etc/modprobe.conf and run ztcfg explicitly after you loaded all modules |
| 16:58.33 | tzafrir_laptop | (or just run /etc/init.d/zaptel start) |
| 17:01.21 | [TK]D-Fender | b1shop: "a" or "o" |
| 17:01.40 | *** join/#asterisk Cybertoy (n=cybertoy@swillux.swill.org) |
| 17:02.18 | jcaceres | Strom_M, i get the same result |
| 17:02.26 | jcaceres | i'll paste bin my conf |
| 17:02.29 | Strom_M | cool |
| 17:02.47 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 17:03.36 | GMouse | I have an ordinary home telephone setup. Is it possible to reroute our phones through asterisk, rather than off to the phone provider? |
| 17:04.01 | Strom_M | what do you mean "reroute through asterisk"? |
| 17:04.13 | Strom_M | connect your phone line from the telco to your asterisk box? |
| 17:04.38 | GMouse | The other way around, I think. |
| 17:04.43 | GMouse | That is, bypass the telco. |
| 17:04.52 | jcaceres | http://pastebin.com/d51dfab17 |
| 17:05.00 | Strom_M | GMouse: that's doable too |
| 17:05.03 | jcaceres | Strom_M, http://pastebin.com/d51dfab17 |
| 17:05.43 | GMouse | I'm trying to evaluate costs, etc. to see if it would be worthwhile to do this, but I kinda need to know if it's even possible before I spend any money. ;) So, how exactly would that be done? |
| 17:05.47 | Strom_M | jcaceres: the fxs module should use FXO signaling |
| 17:05.51 | Strom_M | and vice versa |
| 17:06.04 | Strom_M | you have the green module in slot 3 and the red one in slot 4? |
| 17:06.28 | GMouse | I'm assuming that I'd need an fxs card of some sort, but I'm not sure where in our phone system the connection should be made. |
| 17:07.17 | b1shop | [TK]D-Fender: a or o? |
| 17:07.21 | [TK]D-Fender | GMouse: unplug your home from where the line comes in. PLug your home onto an ATA. Connect ATA to * and * to a VoIP provider. |
| 17:07.28 | [TK]D-Fender | b1shop: Standard Extensions. |
| 17:07.45 | [TK]D-Fender | GMouse: Technically you don't even NEED * |
| 17:07.59 | [TK]D-Fender | GMouse: But its good for a bunch of things. |
| 17:09.06 | GMouse | Can you link me to such an ATA? |
| 17:09.15 | jcaceres | Strom_M, i have the red one(fxo) in 3 and green one (fxs) in 4 |
| 17:09.22 | [TK]D-Fender | Google up "Linksys SPA-2102 |
| 17:10.09 | Strom_M | jcaceres: i don't see anything wrong with your config; i'd call digium at this point. you have free install support with the hardware. |
| 17:10.50 | jcaceres | is it a good idea to use genzaptelconf command? |
| 17:11.07 | GMouse | [TK]D-Fender: Thanks |
| 17:11.17 | Strom_M | i personally never use it, but i dont see anything wrong with what you've got |
| 17:11.37 | GMouse | It has a LAN port and a WAN port...? |
| 17:13.11 | [TK]D-Fender | GMouse: Yup, can act as a router as well if you want it to. |
| 17:13.26 | [TK]D-Fender | GMouse: Great little ATA's |
| 17:13.39 | jcaceres | is this the correct order,1) load zaptel and needed modules, 2) configure zaptel and zapata.conf, 3) run ztcfg --vv then start asterisk? |
| 17:13.57 | Strom_M | swap 2 and 1 |
| 17:14.26 | GMouse | [TK]D-Fender: huh, interesting |
| 17:14.57 | jcaceres | ok, i see know, this is because genzaptelconf does not function until you have started zaptel |
| 17:15.11 | Strom_M | jcaceres: call digium. |
| 17:17.16 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:17.16 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- Flame suits required... |
| 17:18.39 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:18.47 | shmaltz | anyone from nufone here? |
| 17:19.28 | [TK]D-Fender | mercestes: Convert to ULAW. The end. |
| 17:19.49 | sashion | jcaceres: start from scratch here please. confirm all zaptel drivers are unloaded (ie: lsmod shows no zaptel or wct400 or wct1xxp) |
| 17:20.09 | jcaceres | i am doiing it againg |
| 17:20.11 | mercestes | really? |
| 17:20.18 | mercestes | what's the sox command for that? |
| 17:20.48 | Strom_M | sashion: I already did that. |
| 17:20.59 | sashion | jcaceres: lol also confirm you do not have install wct1xxp && /... in your /etc/modules.conf or /etc/modprobe.conf |
| 17:21.36 | sashion | Strom_M: ok does his setup work now ? |
| 17:21.47 | Strom_M | sashion: no. I advised him to call digium. |
| 17:21.49 | [TK]D-Fender | mercestes: Don't recall offhand, its on the WIKI |
| 17:21.54 | Strom_M | he has free install support |
| 17:22.36 | neverblue | does using a new codec, switching from ulaw to G.729, have to implemented on both ends, at the provider and on my local * box ? |
| 17:22.40 | sashion | Strom_M: hmmm true |
| 17:22.58 | neverblue | s/to/to be/ |
| 17:23.04 | jcaceres | but i am in peru, long distance call, is not an option right now |
| 17:23.17 | Strom_M | jcaceres: there is voip alsp |
| 17:23.38 | Strom_M | install an IAX2 softphone on your workstation and call IAX2/guest@misery.digium.com/s |
| 17:23.55 | Strom_M | and it's FREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE |
| 17:24.06 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 17:24.21 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:24.29 | shmaltz | anyone from nufone here? |
| 17:25.30 | jcaceres | ok |
| 17:25.46 | neverblue | or can you just change codecs on the fly, to test different ones? |
| 17:26.05 | awk | anyone know how to clear all channels, I know that asterisk shows in the cli it clearing channels, but is there way to clear the spans manually? |
| 17:26.12 | Strom_M | neverblue: if the provider has them all turned on, you can just restrict codec choice |
| 17:26.13 | awk | re-loading the device doesn't work |
| 17:26.26 | mercestes | awk: stop now |
| 17:26.32 | mercestes | awk: What is broken? |
| 17:26.47 | neverblue | Strom_M, so if I had ulaw running locally, and the provider does not allow it, will I receive an error (hopefully) ? |
| 17:26.59 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 17:27.07 | Strom_M | neverblue: the call won't set up if both sides can't agree on a codec |
| 17:27.15 | neverblue | ok, thanks |
| 17:27.20 | neverblue | thats what I needed to hear |
| 17:27.32 | *** join/#asterisk `Sean (n=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 17:27.39 | awk | mercestes nothing is broken. just sometimes when you havea red light.. forcing the clearance of the channels |
| 17:27.48 | awk | mercestes just a round about understanding thats all im after |
| 17:27.54 | *** join/#asterisk Kirrus (n=Kirrus@squizzey.plus.com) |
| 17:28.00 | awk | just to clear the alarm.. |
| 17:28.41 | mercestes | what is the alarm for? |
| 17:29.00 | Kirrus | Hello.. how can I reset a sip peer's registration on Asterisk? the sip peer has gone away, but asterisk is holding a registration open for it, with an expiry time of 4000 seconds (and counting down) |
| 17:29.59 | awk | mercestes on the pri to indicate a problem, i know sometimes resetting the pri fixes it, but i know sometimes if the alarm is on. and in the cli asterisk says clearning spans or something to that affect, and after it does that the light goes green |
| 17:30.16 | creativx | murr. does anyone have a good hint as how to solve the following.. i want a message played back upon a caller joining a queue, and only played back once.. as far as i can see theres no "welcome to the queue" option in queues.conf.. |
| 17:30.16 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:30.55 | mercestes | you could try a reload chan_zap.so but....I would figure otu what is causing the alarm personally. |
| 17:31.04 | shmaltz | anyone here willing to do test call with me? |
| 17:32.04 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 17:32.22 | jcaceres | sashion, i have modified the i load modules for zaptel |
| 17:32.25 | jcaceres | http://pastebin.com/d6e7e65ae |
| 17:32.38 | jcaceres | tell me if its correct now |
| 17:34.11 | awk | mercestes: and what is the best troubleshooting techniques.. eg: a check this b check that, etc |
| 17:34.46 | mercestes | zap show status. see if the CLi gives you any errors or alarms, etc. |
| 17:34.52 | mercestes | zttool |
| 17:35.14 | sashion | jcaceres: hash out wct1xxp and wct2xxp and wctdm in /etc/modprobe.d/zaptel |
| 17:35.25 | sashion | so they don't get automatically installed on boot up |
| 17:35.36 | sashion | then load zaptel |
| 17:35.39 | sashion | modprobe zaptel |
| 17:35.52 | sashion | wait about 10 seconds. confirm its loaded by doing a ls /dev/zap |
| 17:36.41 | sashion | once loaded, load your PRI card: modprobe wct1xxp |
| 17:36.46 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 17:36.57 | Kirrus | oh well... guess its home time anyway. |
| 17:38.08 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 17:39.01 | Sci_05 | is there any way to have asterisk run a command after it get the hangup? |
| 17:40.32 | jcaceres | sashion, which module do i need wcte12xp or wct1xxp? |
| 17:41.02 | sashion | exten => h,1,System(/path/to/command arg1 arg2) |
| 17:41.13 | sashion | jcaceres: wct1xxp |
| 17:41.23 | sashion | jcaceres: i assume you have a 1 port PRI card ? |
| 17:41.38 | jcaceres | TE120P |
| 17:41.46 | Sci_05 | ok I will give it a shot, thanks sashion |
| 17:41.51 | jcaceres | yes one port |
| 17:42.12 | Strom_M | wcte12xp |
| 17:42.17 | Strom_M | not wct1xxp |
| 17:42.34 | sashion | ah jcaceres: wcte12xp |
| 17:42.39 | sashion | thanks Strom_M :) |
| 17:42.41 | Strom_M | that's what I just said |
| 17:42.44 | Strom_M | you're welcome |
| 17:42.44 | jcaceres | yes ia was wondering? |
| 17:43.32 | jcaceres | so finaly which modules should i have in /etc/modprobe/zaptel? |
| 17:43.35 | sashion | jcaceres: can you pastebin the output of cat /proc/zaptel/* |
| 17:43.38 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 17:43.58 | jcaceres | sashion, oks |
| 17:44.03 | sashion | jcaceres: don't really need anythere, rather load them with zaptel init.d script |
| 17:45.05 | jcaceres | sashion, i do no have proc zaptel, btw i do not have zaptel loaded |
| 17:45.15 | jcaceres | yes |
| 17:45.19 | jcaceres | yet |
| 17:45.22 | sashion | jcaceres: thought you had zaptel loaded :P |
| 17:45.24 | *** part/#asterisk n00dle (n=ccraft@204.10.248.123) |
| 17:46.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:47.02 | jcaceres | and which modules i must not load? |
| 17:48.01 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:48.50 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:48.50 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:49.09 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 17:49.14 | sashion | jcaceres: everything but wcte120p and wctdm |
| 17:49.31 | sashion | in redhat/fedora zaptel has a config file in /etc/sysconfig/zaptel |
| 17:49.39 | sashion | not sure if it exists in debain |
| 17:49.46 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:49.51 | sashion | but you can mod that file to only load the modules you need |
| 17:50.09 | jcaceres | i think it it /etc/modprobe/zaptel, ... look at this |
| 17:50.11 | jcaceres | http://pastebin.com/m3446012b |
| 17:50.35 | pepse | Any of you guys know anything about iCall? Is it possible to register with it? |
| 17:50.51 | sashion | jcaceres: looks great |
| 17:50.55 | sashion | ok start asterisk now |
| 17:51.09 | pepse | When I sniff iCall client's traffic, i see it's registering to a machine on port 4569, looks much like a regular iax2 registry |
| 17:51.28 | pepse | but when i compare it to a log of idefisk registering to my asterisk machine, it's off by just a couple bytes |
| 17:51.46 | *** part/#asterisk galeras (n=galeras@200.31.204.42) |
| 17:51.52 | jcaceres | http://pastebin.com/m53395ffd |
| 17:52.17 | jcaceres | thanks Strom_M , sashion |
| 17:52.30 | jcaceres | finaly i was a mater of loading modules |
| 17:52.46 | sashion | jcaceres: you still need to ensure on boot up the right modules load |
| 17:52.59 | *** join/#asterisk joaop (n=taken@201.22.13.157.adsl.gvt.net.br) |
| 17:53.05 | sashion | best thing to do is remove zaptel from startup (init.d) and then add in /etc/rc.d/rc.local |
| 17:53.09 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 17:53.24 | sashion | your modules to load |
| 17:53.45 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 17:54.01 | sashion | remember, with zaptel, you need to wait at least 10 seconds or so before loading your telephony drivers, else you'll get "Address does not exist" errors |
| 17:54.30 | jcaceres | well, in the configuration file i only left the wcte12xp and wctdm |
| 17:54.55 | sashion | jcaceres: ok great.. that will work too... do a reboot to ensure all comes up properly |
| 17:55.10 | jcaceres | this is my firt try wit card, i i was suffering a lot, i prommes i'll document all this |
| 17:56.09 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:56.32 | shmaltz | anybody here usning nufone? |
| 17:56.34 | shmaltz | this what I get: |
| 17:56.35 | shmaltz | http://private.dnsstuff.com/tools/lookup.ch?name=www.nufone.net&type=A |
| 17:56.38 | sashion | jcaceres: not a problem :P Generally these things work and you barely need to do any real configs here and there but oh well |
| 17:57.16 | jcaceres | it worked after the reboot, thnk |
| 17:57.25 | jcaceres | i'll go for luch |
| 17:57.30 | jcaceres | thnks again |
| 17:57.32 | jcaceres | bye |
| 18:03.30 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 18:05.26 | brodiem | In migrating to Ast14, I noticed AgentCallbackLogin is being depreciated. How are agent logins supposed to happen with this (and without using AgentLogin where a call is open the entire session)?? I know I can use dialplan and call Add/RemoveQueueMember, but that doesn't actually change the "status" of an agent (i.e. still logged off). |
| 18:07.38 | *** join/#asterisk Blumpkin (n=major@216.120.167.82) |
| 18:08.25 | Blumpkin | I have a quick question. I've just migrated to a new asterisk server and it looks like incoming calls are coming in on SIP, which they didn't do. I only want to use IAX. Is there a simple way to do this? |
| 18:09.58 | mercestes | Blumpkin, umm.. I guess your carrier would have to send you your calls on IAX instead of on SIP. |
| 18:10.44 | mercestes | So you call your carrier and scream at them, "No sip calls! IAX only! Ok? Ok! Bye bye!" and hang up really quicklky. |
| 18:12.47 | Blumpkin | well it looks like calls randomely come in on either IAX or SIP. I've never seen SIP calls come in before. |
| 18:13.14 | Blumpkin | I didn't know if there was a config setting to connect to my provider using only IAX. |
| 18:14.18 | jaiger | take the provider's authentication info out of your sip.conf |
| 18:14.37 | Blumpkin | now that these SIP calls come in, some of them don't hear the greeting when calling. |
| 18:14.51 | jaiger | I think that means take out the "register" lines in the sip.conf |
| 18:17.03 | [TK]D-Fender | Blumpkin: Itsw your config... go look at what you're doing.... |
| 18:18.18 | Blumpkin | what's even more strange is the configs are almost exact copies from the old server. |
| 18:18.48 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-227-163.dsl.irvnca.pacbell.net) |
| 18:18.49 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:19.27 | Blumpkin | so is the only solution really to comment out my connection strings in sip.conf to my provider? |
| 18:20.22 | sashion | brodiem: use your database :P |
| 18:20.51 | Optic | moo moo |
| 18:21.45 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:23.30 | Blumpkin | this is frustratin |
| 18:24.12 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 18:24.34 | *** join/#asterisk ewaldo (n=chatzill@206.80.91.195) |
| 18:24.48 | brodiem | sashion, You mean use astdb to keep track of agent logins/logoffs myself? |
| 18:25.14 | ewaldo | Hello. Is there any way to make a phone initiate a call using the Asterisk API? |
| 18:25.34 | waKKu | maybe is better use a .call file |
| 18:25.42 | ewaldo | er, Asterisk Manager API |
| 18:25.51 | phillipk | ewaldo: ACTION: Originate |
| 18:26.04 | ewaldo | Thank you very much phillipk |
| 18:28.07 | brodiem | sashion ? |
| 18:28.42 | sashion | brodiem: yup |
| 18:28.48 | sashion | its what I do on my systems |
| 18:29.19 | sashion | then use a customized func_devstate.c to display login indication on the phones (light on means logged in) |
| 18:29.28 | waKKu | folks... is this normal when doing a transfer ? |
| 18:29.28 | waKKu | [Jul 27 15:29:04] NOTICE[11205]: res_features.c:1241 ast_feature_request_and_dial: Don't know what to do about control frame: -1 |
| 18:29.29 | waKKu | [Jul 27 15:29:04] WARNING[11205]: cdr.c:830 ast_cdr_init: CDR already initialized on 'Local/882@ddi-2a71,1' |
| 18:29.53 | brodiem | sashion do you use chan_agent at all? |
| 18:30.06 | sashion | nope |
| 18:30.13 | sashion | Local channel for the call back |
| 18:30.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:30.34 | brodiem | hmm |
| 18:30.36 | sashion | so AddQueueMember(Local/${EXTEN}@agent-handler/n) |
| 18:31.14 | sashion | Queue calls Local -> goes into agent-handler context where I setup recording and CDR(userfield) and then calls the tech type of that extension |
| 18:31.24 | *** join/#asterisk pc500 (n=pc500@71.216.58.142) |
| 18:31.24 | pc500 | Anyone updated the firmware on a cisco 7960 ip phone? |
| 18:32.23 | Blumpkin | anyone know what would cause Asterisk to pick up, show that greetings are playing, but not hearing anything on the other end? |
| 18:32.37 | Blumpkin | ah it's gotta be my firewall |
| 18:32.38 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 18:32.48 | brodiem | sashion I like it :) |
| 18:33.02 | sashion | astdb is your FRIEND :P |
| 18:34.03 | brodiem | I can't think of any problems in being compatible with the current dial logic.. other then generating some events to the queue_log for queuemetrics |
| 18:34.21 | brodiem | and I love the idea of using func_devstate to show an indication on the phones |
| 18:34.33 | brodiem | been looking for a reason to start using that :) |
| 18:34.44 | *** join/#asterisk rhombus (i=user239@74.12.125.50) |
| 18:34.52 | Blumpkin | does anyone not use SIP for incoming/outgoing calls? |
| 18:35.10 | Mrtaz | i use analog |
| 18:35.10 | rhombus | Blumpkin: I use IAX for incoming |
| 18:35.14 | Mrtaz | ZAP |
| 18:35.18 | sashion | definately... you can hackup app_queue to create an QueueLog() function if you need to manually add data to queuelog.. other than that, you'll get Local/xxxx@agent-handler/n in your queuelog file... write a little perl script to clean that up :P |
| 18:35.20 | ewaldo | cool, got ACTION: Originate working but it doesn't ring my extension until the call is picked up by the outside party. Is there a variable I can set to make it immediately dial the extension while it is dialing the Channel? |
| 18:35.21 | rhombus | Blumpkin: or Zap |
| 18:36.20 | sashion | brodiem: i also use func_devstate for breaks... our agents have a "break" key that just simply puts the member into Pausemode (if not already paused) and then lights up the break key as well.. so supervisors can spot what agents are not working :P |
| 18:36.36 | brodiem | sashion I guess I re-define the agents in queuemetrics as Local/ instead of Agent/, and add the login/logoff events to queue_log |
| 18:36.41 | Blumpkin | I use voicepulse connect. They support IAX and SIP. If I comment out my register lines in my sip.conf, I should be using just IAX, right? |
| 18:36.52 | sashion | brodiem: Bingo! |
| 18:37.00 | brodiem | sashion nice |
| 18:38.04 | brodiem | sashion hmm maybe it is time to rid the old *78/*79 DND functions :) |
| 18:38.13 | sashion | brodiem: also what you can do is use astdb to keep a track of your "devstates" and then when asterisk reloads, you can automatically generate the states |
| 18:38.24 | sashion | specialling if you're using persistent members |
| 18:38.27 | sashion | LOL |
| 18:38.56 | brodiem | yeah true |
| 18:39.16 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 18:39.32 | mercestes | quit hacking me. :( |
| 18:40.22 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 18:40.32 | brodiem | time to grab devstate from svn |
| 18:41.12 | sashion | lol |
| 18:41.21 | brodiem | <PROTECTED> |
| 18:41.33 | sashion | nice one :P |
| 18:42.13 | Mrtaz | gah is there anyway to filter out the low level static on a Digium board (besides replacing it) |
| 18:42.30 | ewaldo | Does anybody know how to make the local extension ring before the channel has picked up when using ACTION: Originate? |
| 18:43.15 | Blumpkin | anyone know what would cause Asterisk to pick up, show that greetings are playing, but not hearing anything on the other end? |
| 18:43.19 | Blumpkin | I checked my firewall |
| 18:43.55 | sashion | Mrtaz: check your earthing ? |
| 18:44.16 | Mrtaz | hmm...box isnt explictly grounded |
| 18:44.28 | Mrtaz | maybe I gotta get the big ole ground cable out |
| 18:44.36 | sashion | Blumpkin: Does asterisk Answer() |
| 18:44.37 | ewaldo | Blumpkin: could be a codec licensing issue. Are you using SIP or ZAP? |
| 18:44.40 | sashion | Mrtaz: might help |
| 18:44.51 | Mrtaz | k ill give it ago thanks man |
| 18:45.37 | Blumpkin | ewaldo: iax |
| 18:45.46 | ewaldo | k, what codecs do you have enabled? |
| 18:45.58 | Blumpkin | sashion: Yes. It answers and I can see in console it plays the greeting. |
| 18:46.07 | Blumpkin | Whats odd is sometimes you hear it, sometimes you don't |
| 18:46.16 | Blumpkin | it's about 50/50 when repeatedly dialing |
| 18:46.24 | sashion | Blumpkin: lag perhaps ? |
| 18:46.34 | sashion | what is your qualify on the peer ? |
| 18:48.15 | Blumpkin | qualify? |
| 18:48.25 | sashion | Blumpkin: iax2 show peers |
| 18:48.30 | Blumpkin | It looks like asterisk requests ulaw every time - and gets it. |
| 18:48.51 | sashion | what does the status look like ? |
| 18:48.52 | Blumpkin | and the incoming call comes from the same ip every time too. |
| 18:49.14 | ewaldo | Blumpkin: Have you checked your and the other person's ears for earwax? :P |
| 18:49.28 | Blumpkin | ewaldo: I'm the other person calling. :) |
| 18:49.37 | Blumpkin | I'm calling while watching the console |
| 18:49.38 | sashion | ewaldo: LOL!!!! |
| 18:49.58 | Blumpkin | sashion: Sorry I'm sort of naive. How do I check the status? |
| 18:50.15 | sashion | Blumpkin: iax2 show peers |
| 18:50.16 | sashion | in cli |
| 18:51.18 | Blumpkin | shows 3 servers with 64 ms response or less |
| 18:51.23 | sashion | you'll see a Status Colum.. what are the timeouts (ie: xx ms) |
| 18:51.33 | sashion | hmmm thats good |
| 18:51.36 | Blumpkin | 59 61 and 64 |
| 18:51.49 | sashion | ok how are you calling ? |
| 18:51.56 | Blumpkin | using my cell phone |
| 18:52.08 | sashion | ok.. |
| 18:52.11 | Blumpkin | I'm migrating to a new server. Old server never had this issue. |
| 18:52.31 | sashion | Cellphone -> ISDN PRI (i assume) -> * -> IAX -> * number 2 ? |
| 18:52.46 | sashion | * = asterisk server ofcourse |
| 18:52.56 | Blumpkin | yeah not sure about the ISDN PRI |
| 18:53.09 | sashion | ok what telephony service have you got ? |
| 18:53.22 | sashion | how do you connect to your CO (telecoms provider) |
| 18:53.27 | Blumpkin | voicepulse |
| 18:53.36 | sashion | oooo ok |
| 18:53.55 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 18:53.55 | Blumpkin | I might hammer the old server and see if I get any audioless connections |
| 18:53.56 | sashion | so Cellphone -> VoicePulse -> IAX -> * right ? |
| 18:53.59 | Blumpkin | yeah |
| 18:54.40 | sashion | hmmm should work |
| 18:54.50 | sashion | what codec is your greeting done in? |
| 18:55.02 | sashion | file /path/to/voicegreeting.gsm |
| 18:55.06 | sashion | what does that return ? |
| 18:55.16 | Blumpkin | in console? |
| 18:55.21 | sashion | in linux :) |
| 18:55.25 | sashion | not asterisk console |
| 18:55.54 | *** join/#asterisk markgreene (n=markgree@209.12.142.2) |
| 18:56.34 | Blumpkin | well in bash, 'file' doesn't seem to be found |
| 18:56.48 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 18:56.51 | markgreene | Hello - Is there someone in the room that can tell me what to expect, and the best way of going about it, when upgrading from a 1.2 installation to a 1.4? |
| 18:56.55 | mercestes | whereis file |
| 18:57.04 | mercestes | markgreene, start over. |
| 18:57.04 | sashion | lol thanks mercestes |
| 18:57.06 | mocker | So are Sangoma cards not as finicky about IRQ ticks as Digium cards? |
| 18:57.18 | mercestes | sashion: np. ;) |
| 18:57.27 | Blumpkin | my guess is I don't have 'file' installed? |
| 18:57.37 | markgreene | mercestes: SO there is no chance of just compiling 1.4 ontop of 1.2? |
| 18:57.38 | Blumpkin | file: |
| 18:57.52 | sashion | Blumpkin: if you running CentOs.. you can do a yum -y install file |
| 18:57.55 | mercestes | markgreene, Well, there is some chance but....it's work either way. |
| 18:58.01 | sashion | or get the file rpm from www.rpmfind.net |
| 18:58.12 | mercestes | or format your HDD and install gentoo. |
| 18:58.25 | [TK]D-Fender | mocker: Correct |
| 18:58.53 | mocker | [TK]D-Fender: Is it a driver issue, or just the way the cards are manufactured? |
| 18:58.56 | markgreene | mercestes: when you say, "start over", are you refering to all of the config files as well? Extensions and DID mappings, etc? |
| 18:59.07 | [TK]D-Fender | mocker: Somewhere in between. |
| 18:59.18 | mercestes | Yea. *most* of the stuff will be ok, but the parts taht aren't are annoying. |
| 18:59.18 | mocker | Because the 'Disable USB, Disable on-board ethernet' game gets old. :( |
| 18:59.32 | mercestes | markgreene, But, you can upgrade, but there will be breakage... |
| 18:59.58 | sashion | markgreene: depends on which 1.2 version you are upgrading from :) |
| 19:00.04 | Blumpkin | sashion: File returns iqc-thankyou.gsm: data |
| 19:00.13 | [TK]D-Fender | mocker: You can forget all that nonsense. |
| 19:00.30 | Blumpkin | I copied my audio files from asterisk 1.2 to my new 1.4 box |
| 19:00.35 | markgreene | mercestes: I must say that it seems odd to me that the asterisk team would make it such an unknown when upgrading to the next version up |
| 19:00.42 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
| 19:00.57 | mercestes | markgreene, They changed alot of syntax and deprecated alot of stuff. |
| 19:01.17 | sashion | Blumpkin: ok so its GSM... should work.. perhaps in your iax.conf file |
| 19:01.29 | sashion | disallow=all |
| 19:01.33 | sashion | allow=gsm |
| 19:01.38 | Blumpkin | sashion: I just verified I'm getting the same issue on the old box. |
| 19:01.57 | mercestes | markgreene, And some of the variables changed. Extensions reload will tell you waht's deprecated. |
| 19:02.21 | Blumpkin | no idea why this is happening. What's odd is although I don't hear the greeting, I can enter DTMP codes and it picks them up. |
| 19:02.46 | sashion | Blumpkin: might be a codec issue.. ulaw -> gsm.... or perhaps your greeting is at the wrong hz setting... should be 8000 |
| 19:02.51 | neverblue | how much of a quality difference between using ulaw and G.729, in everyone's opinion? |
| 19:03.10 | sashion | Blumpkin: do you have the original recording for your greeting ? |
| 19:03.29 | mocker | [TK]D-Fender: I lost every 15th fax or so, I'm thinking it's because my zttest is 99.975 avg.. |
| 19:03.39 | Blumpkin | sashion: That's what I'm using. What I did is created an extension that would just record your voice to file and used those GSM files for greetings. |
| 19:03.48 | mocker | [TK]D-Fender: You don't think Sangoma would have that problem? |
| 19:03.52 | markgreene | mercestes: Thanks for the insight. I will set aside a saturday and a backup job for the task |
| 19:04.09 | mercestes | Good deal. |
| 19:04.15 | sashion | Blumpkin: ah then it will be at the right levels... can only assume its a codec transcoding issue.. |
| 19:04.25 | Blumpkin | and what's funny is the "welcome" schpeel is several audio files. Every one of them plays no audio if you don't hear audio when you call. |
| 19:04.25 | sashion | edit your iax.conf file |
| 19:04.31 | sashion | and under general set |
| 19:04.33 | [TK]D-Fender | mocker: No, it probably WOULD. |
| 19:04.35 | sashion | disallow=all |
| 19:04.37 | sashion | allow=gsm |
| 19:04.45 | mocker | [TK]D-Fender: Gotcha. |
| 19:04.46 | [TK]D-Fender | mocker: Fax on * period isn't so hot. |
| 19:04.47 | Blumpkin | and comment out all others? |
| 19:04.47 | sashion | and then do a module reload chan_iax2.so |
| 19:04.52 | sashion | yes |
| 19:04.56 | mocker | [TK]D-Fender: I know. :( |
| 19:05.02 | [TK]D-Fender | mocker: If your business relies on them, leave * the hell away |
| 19:05.06 | nDuff | [TK]D-Fender: It works for me. |
| 19:05.18 | [TK]D-Fender | nDuff: What reliability %? |
| 19:05.30 | Blumpkin | sashion: is gsm bandwidth intense? |
| 19:05.47 | sashion | Blumpkin: nope... 13.x odd kbits |
| 19:05.48 | [TK]D-Fender | mocker: RxFax or IAXModem+Hylafax? |
| 19:05.58 | sashion | compared to u/alaw which is about 64 odd kbits |
| 19:06.03 | mocker | nDuff: Looking at your /var/spool/hylafax/faxrcvd or whatever, do you ever get things like T.30 timeouts? |
| 19:06.07 | sashion | gsm is however CPU intensive :P |
| 19:06.11 | mocker | [TK]D-Fender: IAXModem+Hylafax. |
| 19:06.20 | mocker | [TK]D-Fender: rxfax shouldn't even be mentioned as a solution. ;) |
| 19:06.30 | rene- | can somebody help me to pinpoint one way audio issue on a lan, no nat with sip ? http://www.pastebin.ca/636248 |
| 19:06.31 | Blumpkin | sashion: Not an issue. Should I remove bandwidth=low as well? |
| 19:06.31 | nDuff | [TK]D-Fender: Failed faxes cause emails to our it-staff mailing list. They're very rare, and almost always trace back to something on the other end. |
| 19:06.58 | nDuff | [TK]D-Fender: that said, I don't have a percentage for you immediately... know offhand how to pull one? |
| 19:07.11 | sashion | Blumpkin: yeah leave that off for now.. also confirm you have trunk=yes |
| 19:07.18 | Blumpkin | sashion: I have bandwidth=low and jitterbuffer=on under general |
| 19:07.31 | Blumpkin | sashion: By "off" do you mean comment it out? |
| 19:07.52 | sashion | Blumpkin: yep |
| 19:07.58 | nDuff | mocker: no. Last time I saw those kinds of failures regularly was when we were using USRs (which are *very* lousy faxmodems) behind SPA-2100s. |
| 19:08.13 | Blumpkin | ok I put trunk=yes in as well |
| 19:08.22 | mocker | nDuff: PRI coming in? |
| 19:08.26 | nDuff | mocker: yes. |
| 19:08.35 | mocker | nDuff: To Sangoma or Digium? |
| 19:08.39 | sashion | Blumpkin: yep |
| 19:08.39 | nDuff | mocker: Sangoma. |
| 19:08.45 | mocker | nDuff: What's your zttest look like? |
| 19:09.05 | nDuff | mocker: 100.000000% |
| 19:09.37 | mocker | I think that's where my problem has to be. :( |
| 19:10.01 | mocker | [TK]D-Fender: Why don't you think that would be the problem? |
| 19:10.38 | [TK]D-Fender | mocker: Might be better, was thinking about RxFAX |
| 19:10.44 | rene- | i think i am going to set those anoying wip 330 's to connect to the small tank asterisk box so they can behave |
| 19:11.27 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 19:11.38 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 19:16.17 | DrukenLPY | afternoon everyone |
| 19:16.22 | mocker | nDuff: Hate to ask, but can you pastebin.ca your /var/spool/hylafax/etc/config.ttyIAX |
| 19:17.06 | nDuff | mocker: sure, I'll pastebin one of them. (I've got 6, one for each modem, but they're only trivially different). |
| 19:17.52 | mocker | nDuff: Thanks. |
| 19:18.15 | mocker | nDuff: The thing that stinks, is every time I try a fax it works flawlessly. |
| 19:18.27 | mocker | I need to buy a $5 fax from a garage sale! |
| 19:18.42 | nDuff | mocker: http://pastebin.ca/636263 |
| 19:19.24 | *** join/#asterisk minkus (n=minkus@pool-72-84-49-162.clrkwv.east.verizon.net) |
| 19:19.39 | mocker | nDuff: Thanks. |
| 19:21.22 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:21.34 | Katty | afternoon lovables. |
| 19:22.33 | file | Kattttttttty |
| 19:22.36 | [TK]D-Fender | Katty: Mew. |
| 19:24.14 | Katty | file: fiiiiillllllleee!! |
| 19:24.19 | Katty | [TK]D-Fender: mew. |
| 19:24.39 | Katty | i'm having a horrible day :< |
| 19:24.47 | waKKu | hmm... |
| 19:24.48 | waKKu | ¬¬ |
| 19:25.17 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 19:25.20 | Katty | Didn't take any caffeine with me to work. So i went to the kitchen to break a 5, here at the office, since they often keep sodas in there. Well, there was no change to break it... After seeing 3 of my co-workers, one of them finally had change. |
| 19:25.28 | Katty | So, back to the fridge i went!! but, alas, nothing diet with caffeine. |
| 19:25.41 | Katty | THEN, i went upstairs to the vending machine... and it ate my dollar. |
| 19:26.02 | waKKu | rofl |
| 19:26.12 | waKKu | kick it |
| 19:26.27 | DrukenLPY | Katty: are you diabetic? |
| 19:26.27 | Katty | so then they had to call the vending machine people.. who came down to fix it. |
| 19:26.39 | Katty | DrukenLPY: borderline, but not really. |
| 19:26.52 | Katty | ripped my dollar in half getting it out, so then i got paranoid and put in quarters. |
| 19:27.01 | Katty | i asked for diet mountain dew, it gave me... |
| 19:27.03 | Katty | dr. pepper |
| 19:27.18 | waKKu | lol.. nice movie |
| 19:27.22 | Katty | luckily someone was coming up for dr. pepper. |
| 19:27.27 | DrukenLPY | sounds like a bad day... i say you take the rest of the day off :) |
| 19:27.33 | Katty | so i picked dr. pepper and it gave me diet mt. dew ^_- |
| 19:27.56 | Katty | DrukenLPY: and do what? :P |
| 19:28.00 | sashion | Katty: And the moral of the story is ? |
| 19:28.10 | Katty | sashion: ummm. |
| 19:28.23 | Qwell[] | Katty: You have better luck with your vending machines that we do. |
| 19:28.30 | Katty | Qwell[]: :< |
| 19:28.42 | sashion | Qwell: :P |
| 19:28.43 | Qwell[] | We can choose Dr. Pepper, and get a Snicker's |
| 19:28.48 | Strom_M | hahaha |
| 19:28.50 | waKKu | lol |
| 19:28.51 | Katty | i'm going to have to put a mini fridge in my office, packed full of red bull, vault, and... *ahem* chocolate liquors. |
| 19:28.53 | sashion | LOL |
| 19:29.11 | Katty | Qwell[]: :< |
| 19:29.23 | Katty | Qwell[]: gee thanks, now i want chocolate. |
| 19:29.34 | nDuff | (the down side of keeping the coffee equipment there is that it's easy to make a mess in what should be a clean environment... the plug side is that it keeps other people out of my coffee, and building regs require alcohol to be in a locked area... server room := locked area.) |
| 19:29.52 | Katty | nDuff: teehee. |
| 19:30.00 | Katty | nDuff: our server room is so cold, no one wants to be in there more than a few seconds. |
| 19:30.17 | Katty | nDuff: i have a hoodie in the office, just for server room visits :> |
| 19:30.20 | nDuff | heh. |
| 19:30.59 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 19:30.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:31.29 | nDuff | our server room is halfway decent now, but the building tried to cut serious corners when putting it together for us. Contract specified only a "raised floor" => owner had a buddy of his build a wooden deck inside. |
| 19:31.55 | Sci_05 | lol |
| 19:32.08 | nDuff | ...had that ripped out, of course... but when we showed up on the first day, it turned out that all the non-AC 110V outlets were on the same circuit and the 220V outlets were actually wired for 110. |
| 19:32.10 | rhombus | nDuff: you have got to be kidding me |
| 19:32.16 | nDuff | rhombus: not kidding. |
| 19:32.31 | rhombus | was this a low bid? |
| 19:32.37 | nDuff | rhombus: yup. |
| 19:32.44 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 19:32.56 | rhombus | ...and the moral of the story is... |
| 19:32.58 | rhombus | :P |
| 19:33.39 | Sci_05 | rhombus: when you get a server room built, do it yourself? |
| 19:33.52 | rhombus | that's one -- there are many |
| 19:34.01 | rhombus | you're creative people, I'll let you guys come up with a few |
| 19:34.11 | nDuff | rhombus: ...don't let the cheapass CEO decide on a new building without letting the staff with real estate and contract drafting experience get involved? |
| 19:35.02 | rhombus | nDuff: sounds like a great one |
| 19:35.33 | rhombus | my favourite would be: don't be penny-wise and pound foolish |
| 19:35.48 | nDuff | *nod*. |
| 19:35.56 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:36.05 | rhombus | the deck image is scary. |
| 19:36.33 | DrukenLPY | Katty: and do what? buy the mini fridge for your office... what else? |
| 19:36.48 | nDuff | rhombus: yup. there were more than a few people expressing disbelief when we found out about that one. |
| 19:37.53 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 19:38.28 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:38.53 | markgreene | Can someone tell me why Mixmonitor would cause asterisk to crash once it's recieved the "stop" command |
| 19:38.54 | markgreene | ? |
| 19:39.07 | markgreene | I am running 1.2 by the way |
| 19:39.10 | rhombus | nDuff: is it okay to ask where this was? no specifics needed, just a region or province or state |
| 19:39.26 | nDuff | rhombus: we're in Austin, TX. |
| 19:39.47 | rhombus | nDuff: Hmn... Austin's a high-tech centre, you'd think that... |
| 19:39.54 | |Rain| | so... has anyone ever had a problem where a TDMoE interface was providing exactly half timing? (zttest says 8192 samples in 16384 sample intervals 0.000000%) |
| 19:40.16 | rhombus | did they at least put some deck chairs on it? |
| 19:41.36 | DrukenLPY | muskoka chairs, and perhaps a swim up wetbar |
| 19:41.55 | nDuff | heh. |
| 19:41.58 | rhombus | DrukenLPY: lol |
| 19:42.25 | DrukenLPY | god knows everyone "surfs" in a datacenter :) |
| 19:42.56 | DrukenLPY | hehe ok, bad joke |
| 19:43.20 | Innatech | hmm. Vitelity's not answering their phones. |
| 19:43.24 | brodiem | sashion you there? |
| 19:43.40 | brodiem | sashion never mind :) |
| 19:44.10 | sashion | brodiem: yes |
| 19:44.12 | sashion | ? |
| 19:44.33 | brodiem | sashion I was going to ask something about devstate/BLF but the answer came to me :) |
| 19:44.48 | sashion | brodiem: nice.. I wish I had that luck... |
| 19:45.27 | brodiem | sashion well it was a dumb thought I had to begin with lol |
| 19:46.04 | sashion | brodiem: ah one of those.. ok you're excused then :P |
| 19:46.07 | DrukenLPY | Innatech: which vitelity? |
| 19:46.36 | *** join/#asterisk jovu (n=bert@213.165.249.193) |
| 19:48.07 | Innatech | DrukenLPY: There's more than one? |
| 19:48.14 | Innatech | DrukenLPY: Vitelity.com |
| 19:48.23 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 19:49.07 | Innatech | 1.888.89.VITEL ---> queue ----> "we're too busy, leave a msg, kthxbye." |
| 19:49.14 | gardo | i'm having this problem w/ an x100p card |
| 19:49.27 | gardo | running modprobe wcfxo gives me this: |
| 19:49.35 | gardo | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 19:49.47 | nDuff | I've got DYNAMIC_FEATURES=automon in [globals], automon=>*1 in features.conf's [featuremap], but dialing *1 on any of my SIP phones isn't doing anything. (the features.conf values for call transfers and like features do work, however). I have TWK flags set in the outgoing Dial command. |
| 19:50.14 | nDuff | Any ideas as to something else I could be missing? |
| 19:50.31 | jovu | im trying to get faxdetect to work on a tdm400 and an analog line, it works if i manually call rxfax, but i want it to answer() and rxfax if a fax call is detected, or call voicemail otherwise |
| 19:50.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:51.23 | jovu | sofar, fax detection seems not to work at all, it always goes to voicemail |
| 19:52.10 | nDuff | jovu: as an aside, I don't recommend rxfax. |
| 19:53.12 | jovu | nduff, yeah i heard it's not great, but it seems functional... if i get fax detection working i can bounce to a hardware faxmodem on the serial port if necessary |
| 19:53.29 | nDuff | jovu: as for fax detection, I ended up using nvBackgroundTest while my voice menu is playing. |
| 19:54.00 | jovu | the nv* stuff doesnt work with asterisk 1.4.x, and the website for it is down in any case... |
| 19:54.10 | nDuff | jovu: I ported it to 1.4.x |
| 19:54.22 | jovu | yes? is there somewhere i can download the updated version? |
| 19:54.23 | nDuff | jovu: ...think that port should be available somewhere. |
| 19:54.24 | |Rain| | I did the same, 'cause I need detection on non-zap channels :/ |
| 19:54.44 | nDuff | jovu: dunno; it was a while ago. I can look and see if I still have the source sitting around somewhere. (I'd better, since otherwise I'll be doing it again next upgrade) |
| 19:55.16 | jovu | i found a patch pasted into a wiki, but i couldnt download the original source to patch |
| 19:55.16 | nDuff | jovu: iaxmodem+hylafax actually works *very* well for me. it's not SpanDSP but app_rxfax specifically that I don't trust. |
| 19:55.25 | *** join/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
| 19:55.58 | jovu | i'l try hylafax, first thing is to get detection working on it tho |
| 19:56.28 | nDuff | jovu: okay, found it. give me a minute and I'll pastebin the source. |
| 19:56.29 | |Rain| | http://themuffin.net/app_nv_backgrounddetect_ast1.4.c |
| 19:56.33 | |Rain| | jovu: ^ |
| 19:57.26 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 19:57.42 | jovu | aha |
| 19:57.46 | jovu | let me try |
| 19:58.18 | nDuff | http://pastebin.ca/636315 |
| 19:58.27 | fluffyfluffy | I have a single FXO PCI card from x100p.com and I'm pretty sure it's dead. All sorts of "initialize DAA" errors. Anyone have any experience with this card? Or could someone recommend a single FXO device that I should use? |
| 19:58.57 | Strom_M | fluffyfluffy: TDM01B |
| 19:59.01 | nDuff | and http://pastebin.ca/636316 |
| 19:59.08 | Strom_M | the x100p clone cards are crap, as you've just discovered |
| 19:59.30 | Nugget | that ought to be in the topic. |
| 19:59.39 | fluffyfluffy | strom_m: yep. |
| 19:59.42 | nDuff | Nugget: howdy. |
| 19:59.45 | Nugget | moo |
| 19:59.58 | fluffyfluffy | at least it was only $30 worth of crap. |
| 20:02.33 | Blumpkin | is there a codec that isn't too bandwidth intense that might help correct "choppyness" I'm experiencing? |
| 20:03.47 | Mercestes | g729 |
| 20:04.32 | Katty | i can't think any more! |
| 20:04.35 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:04.41 | Katty | i need more lemon juice! |
| 20:04.47 | *** join/#asterisk erisson (n=erisson@p54ACBC72.dip0.t-ipconnect.de) |
| 20:04.49 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:04.58 | Strom_M | ok zaphod |
| 20:05.01 | MrTelephone | hi |
| 20:05.14 | Mercestes | Katty: You need lemons |
| 20:05.24 | MrTelephone | 1.2.23, how come asterisk is climbing so fast |
| 20:05.27 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 20:07.06 | rhombus | MrTelephone: because the code is a disaster |
| 20:07.20 | [hC] | anyone know if when using "files" for music on hold, is there a way to have new files show up automatically in asterisk when dropped in the moh folder, without having to 'moh reload' ? |
| 20:07.38 | rhombus | Oh, and MrTelephone: Digium wants you to go to 1.4 :) |
| 20:07.58 | MrTelephone | why? |
| 20:08.22 | MrTelephone | im trying to think of a good failover solution for these systems, its hard |
| 20:08.33 | Mercestes | MrTelephone, CCM |
| 20:08.39 | Mercestes | >.> |
| 20:08.48 | MrTelephone | thats a swear word to me |
| 20:08.55 | Mercestes | uh huh |
| 20:09.13 | MrTelephone | cisco ata186s are neat though |
| 20:09.24 | MrTelephone | i just read the other day that if it can't contact the tftp server it uses the last known config |
| 20:09.34 | MrTelephone | so I don't have to worry about tftp ghoing down for them |
| 20:09.36 | Mercestes | nice |
| 20:10.07 | MrTelephone | if you point a tftp server to a host with multiple ip addresses it should auto check the next one you would think |
| 20:10.34 | MrTelephone | and if you use openser for load balancing what if openser goes down :( |
| 20:10.53 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 20:11.02 | MrTelephone | 2 openser servers with sip devices configured to fallback to the second one I guess |
| 20:11.03 | sashion | MrTelephone, have 2 openser boxes running heartbeat in active-active mode |
| 20:11.36 | sashion | if one ser goes down, heartbeat picks up, loads a virtual IP address (your global access) and you're up in roughly 3 seconds |
| 20:11.36 | MrTelephone | you know in the polycom 501 where you can configure 2 sip servers for each Line? |
| 20:11.45 | MrTelephone | 3 seconds? thats really good |
| 20:12.12 | sashion | ypu.. if you configure heartbeat correctly |
| 20:12.39 | rene- | jeezus |
| 20:12.43 | MrTelephone | what rene? |
| 20:13.02 | sashion | so what you do is have a global IP, say 10.0.0.1 which is an alias... when heartbeat is active, one system controls your 10.0.0.1 address.. when it fails, the other system loads the 10.0.0.1 address (heartbeat does this for you) |
| 20:13.12 | MrTelephone | heartbeat is probably the best thing to use then |
| 20:13.21 | jovu | nduff, your patched version fails to compile, AST_MODULE is not defined |
| 20:13.41 | MrTelephone | sashion are you using it? |
| 20:13.52 | sashion | yes |
| 20:13.52 | MindTheGap | why do one have 3 files for "unavaliable" unavail.gsm unavail.WAV unavail.wav . I can understand 2, one GSM for native gsm and a wav for conversion, but what about the third .WAV? its got about the same size of the gsm one... |
| 20:14.00 | sashion | check out www.ultramonkey.org |
| 20:14.03 | MrTelephone | ok |
| 20:14.11 | sashion | they have some examples on how to config heartbeat |
| 20:14.43 | MrTelephone | its nice to be able to take down a server and have everything still running |
| 20:14.57 | nDuff | jovu: no clue, then. I pulled it straight out of my asterisk-1.4.1 source tree. |
| 20:15.18 | jovu | ahh, i have 1.4.9 |
| 20:15.28 | MrTelephone | 2 open ser boxes, 1 t1 gateway box, 2 asterisk sip mgcp boxes |
| 20:15.43 | sashion | quite a setup there :P |
| 20:15.59 | sashion | don't forget UPS's, else that setup goes bye bye when power fails :P |
| 20:16.00 | MrTelephone | well that seems like the best thing to do right now.. im just pondering |
| 20:16.26 | MrTelephone | 1 seperate voicemail box |
| 20:16.40 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 20:16.45 | sashion | how big are you wanting this system ? |
| 20:17.00 | MrTelephone | not too big but redundancy is key here :-/ |
| 20:17.32 | MrTelephone | maybe just 2 asterisk boxes then on heartbeat, 1 t1 gateway asterisk box, one voicemail box |
| 20:17.51 | sashion | then don't go openser, will waste your time and resources |
| 20:18.02 | MrTelephone | but sometimes the box is fine but asterisk fails.. so are you using rmon to check 5060 status? |
| 20:18.39 | sashion | nah what we do is use safe_asterisk... |
| 20:18.50 | sashion | so if asterisk dies, it gets restarted... |
| 20:19.03 | MrTelephone | i find the t1 drivers I use, if something happens to the t1 card asterisk chokes |
| 20:19.30 | Katty | Mercestes: i /heard/ that lemon juice from toasted lemons was the best. |
| 20:19.30 | tzafrir_home | if asterisk dies, what makes you think it won't die again, immedietly? |
| 20:19.35 | *** part/#asterisk rhombus (i=user239@74.12.125.50) |
| 20:19.37 | sashion | if it doesn't come up again, heartbeat will automatically switches systems... also have a ISDNGuard to switch links from 1 system to the other |
| 20:19.53 | MrTelephone | it doesn't happen often but when it does, it sucks to hear from the customers and be like.. uhhh geez i thought everything was working |
| 20:20.06 | MrTelephone | ISDNGUARD? sounds cool |
| 20:20.09 | sashion | tzafrir_home: have a timeout... if res_watchdog doesn't come up system automatically fails over |
| 20:20.15 | tzafrir_home | safe_asterisk only makes HA more difficult because you don't really know tha Asterisk is down |
| 20:20.34 | MrTelephone | you need some kind of rmon script |
| 20:20.41 | tzafrir_home | sashion, so why do you need safe_asterisk in the first place? |
| 20:20.44 | MrTelephone | opernser is nice because it will detect if there is a response from asterisk on port 5060 |
| 20:21.06 | MrTelephone | but im hoping the polycom 501s will failover by adding second proxy information |
| 20:21.38 | *** join/#asterisk Ch0Hag (n=mking@knight.monnsta.net) |
| 20:21.48 | sashion | tzafrir_home: cause sometimes asterisk just dies but starts right up again perfectly (like last time it somehow got a # from a channel that was down, and sent asterisk for a loop) |
| 20:21.55 | Ch0Hag | How can I interrupt a (zap->zap) call, play a sound and hang up the line? |
| 20:22.06 | sashion | no point failling over if asterisk died cause of an unhandled event |
| 20:22.22 | MrTelephone | i never had asterisk die.. it always freezes and if I type stop now it could take 2 minutes to shut down |
| 20:22.26 | MrTelephone | but it doesn't happen too often |
| 20:22.27 | Ch0Hag | I need a catastrophic failure to get boring people off the phone. |
| 20:22.39 | MrTelephone | ch0hag, barge? |
| 20:22.48 | MrTelephone | lol |
| 20:22.51 | MrTelephone | thats hilarious |
| 20:23.12 | MrTelephone | record a 911 call and play it back on your boring conversation :-/ |
| 20:23.13 | sashion | MrTelephone: in that case, res_watchdog would stop responding and isdnguard would failover lines to other system |
| 20:23.24 | rene- | or a hotline call |
| 20:23.44 | rene- | i used to monitor calls in a hotline callcenter (for tech purposes ofcourse) |
| 20:23.52 | MrTelephone | how does that piece of junk work sashion, is it a good unit? |
| 20:24.20 | sashion | works great if your system fails and you don't wanna unplug isdn cables :P |
| 20:24.29 | MrTelephone | if you only do sip to zap calls on that gateway it shouldn't ever crash :-/ |
| 20:24.40 | MrTelephone | are you controlling it via snmp? |
| 20:25.15 | MrTelephone | imagine a system ran from memory, no harddrive, with a t1 card and basic asterisk config.. how sweet would that be |
| 20:25.21 | MrTelephone | it would never fail |
| 20:25.22 | MrTelephone | :-/ |
| 20:25.24 | sashion | nah isdnguard has a RS232 cable |
| 20:25.33 | Ch0Hag | Oh yeah, on a more useful note, has anyone ever seen a TDM400 get into a state where it doesn't get any details back from placed analogue calls until a cold boot? |
| 20:25.35 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:26.04 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 20:26.20 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 20:26.23 | MrTelephone | shashion, have you had a machine go down yet? |
| 20:26.44 | sashion | MrTelephone: not yet... about 80 days uptime on system... |
| 20:26.46 | rene- | my asterisk does crash, tho it is getting a bit more reliable |
| 20:26.53 | MrTelephone | what if hte machine that the rs232 cable runs to dies? |
| 20:26.54 | sashion | asterisk... probably a max of 3 days uptime or so... |
| 20:26.56 | rene- | i get 5-6 days uptime |
| 20:26.59 | rene- | of asterisk |
| 20:27.11 | rene- | on one box ii have about a month of asterisk uptime |
| 20:27.12 | MrTelephone | every crash gives you a chance to improve your configuration |
| 20:27.17 | rene- | true |
| 20:27.26 | MrTelephone | i noticed that sip only installs work forever |
| 20:27.31 | sashion | then heartbeat on the other system picks up the system has died (heartbeat is a udp broadcast) |
| 20:27.40 | MrTelephone | as soon as you deal with hardware like t1 cards the failure rate increases |
| 20:27.59 | sashion | rene-: so far, I have asterisk running as an SS7 gateway to a Nokia switch.. uptime on that is 26 days so far... |
| 20:28.09 | MrTelephone | you never had to restart? |
| 20:28.21 | sashion | nope |
| 20:28.21 | rene- | 4 weeks 5 hours |
| 20:28.25 | MrTelephone | I have to restart once a week right now because my t1 circuit starts getting corrupt |
| 20:28.28 | sashion | *taps wood) |
| 20:28.48 | rene- | it is good to restart the card |
| 20:28.52 | sashion | rene-: impressive.. what version of asterisk you runnig ? |
| 20:28.53 | [hC] | is there a known way to notice new files in the asterisk music on hold directory without issuing moh reload? |
| 20:28.53 | rene- | like restarting the driver |
| 20:29.04 | rene- | it is asterisk 1.2.18 |
| 20:29.07 | rene- | it would be like twice |
| 20:29.16 | rene- | but we didnt have a proper ups in place |
| 20:29.17 | MrTelephone | i have this channel bank and one of the channels will always begin to get distorted.. and the funny thing is the bad channel changes once in a while |
| 20:29.40 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:29.42 | rene- | it is impressive but it does not uses queues, local channels, recording or chan_spying, |
| 20:29.50 | MrTelephone | renen, i was using 1.2.12 and it was pretty buggy with my sangoma card |
| 20:30.14 | MrTelephone | i upgraded to 1.2.21 and besides a glitch in the dialplan code i've been running with no t1 crashes |
| 20:30.29 | sashion | rene-: ah.. yeah you start adding those features and asterisk tends to get a little more unstable.. |
| 20:31.04 | sashion | but I admit, I've had a lot more success with Sangoma cards than digium.. specially in installs that have flappy PRI lines :P |
| 20:31.22 | MrTelephone | im getting come carrier errors that I havn't investigated yet |
| 20:31.29 | MrTelephone | I have no carrier errors to the channel bank |
| 20:31.41 | MrTelephone | but to my telco I get about 2 a day |
| 20:31.52 | MrTelephone | not sure what the deal is there |
| 20:31.55 | Ch0Hag | Is there no way to do things in the console to running calls? |
| 20:32.04 | MrTelephone | running calls? |
| 20:32.08 | MrTelephone | i don't think so |
| 20:32.17 | MrTelephone | console only controls the oss channel |
| 20:32.19 | sashion | yeah |
| 20:32.22 | *** part/#asterisk Optic (n=dfraser@miso.capybara.org) |
| 20:32.26 | sashion | soft hangup TECH/xxx |
| 20:32.27 | sashion | :P |
| 20:32.35 | MrTelephone | stop now :-/ |
| 20:32.42 | sashion | lol |
| 20:32.42 | MrTelephone | that will shutdown your boring calls for sure |
| 20:32.53 | rene- | heh |
| 20:32.56 | MrTelephone | i grind my teeth when I do that hoping I didn't cut anyone off |
| 20:33.22 | MrTelephone | or if you don't have a backup system and you have to take your system down to reboot.. your heartrate goes up because you know 40 people are without service |
| 20:33.25 | MrTelephone | heh |
| 20:33.32 | MrTelephone | and its hard to work under pressure |
| 20:33.41 | sashion | i wonder when Asterisk will become threaded in the sence that each call gets handled by a new process.. that way.. if asterisk dies.. your calls aren't affected... could be cool |
| 20:33.46 | *** part/#asterisk fluffyfluffy (n=fluffyfl@h69-130-215-2.69-130.unk.tds.net) |
| 20:33.54 | MrTelephone | yeah that would be sweet |
| 20:34.03 | Corydon76-work | Uh, that's forked, not threaded |
| 20:34.12 | sashion | sorry forked.. my bad |
| 20:34.19 | MrTelephone | imagine that :-/ |
| 20:34.24 | Corydon76-work | and it's unlikely Asterisk will ever be run in a forked environment |
| 20:34.34 | Strom_M | that's forked up. |
| 20:34.38 | sashion | lol |
| 20:34.47 | Corydon76-work | It's too critical to be able to see into a common memory space |
| 20:35.01 | sashion | ah i see.... |
| 20:35.09 | sashion | was just saying.. it would be really cool :P |
| 20:35.19 | MrTelephone | well maybe handle rtp on a different level |
| 20:35.29 | MrTelephone | handle rtp on a sub process |
| 20:35.35 | Corydon76-work | No, it'd be slow and awful. |
| 20:35.35 | sashion | what good is that when your ZAP chan dies ? |
| 20:35.35 | MrTelephone | so if it dies it still gets forwarded |
| 20:36.09 | MrTelephone | does anyone here actually use sip reinvite? |
| 20:36.10 | MrTelephone | honestly |
| 20:36.43 | MrTelephone | ccm brags it can do a shitload of calls too but how many if the rtp traffic had to go through ccm |
| 20:36.46 | Strom_M | i don't |
| 20:37.17 | MrTelephone | before i started using voip technology I assumed the ata186's had tunneling support built in |
| 20:37.31 | MrTelephone | so you didn't have to worry about nat |
| 20:37.37 | MrTelephone | or ip addresses |
| 20:37.47 | MrTelephone | maybe thats bonkers to think that.. |
| 20:37.55 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 20:39.21 | MrTelephone | i signed up for voicemeup.com because they are 30ms from me and the calls sound good.. I asked them if they forwarded my callerid/callernum.. they said yes but the callerid(name) doesn't show up |
| 20:39.27 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-182-96.hsd1.wa.comcast.net) |
| 20:39.30 | MrTelephone | is that a cisco problem you think? |
| 20:39.39 | MrTelephone | asterisk sends callerid(name) fine |
| 20:42.15 | MrTelephone | well im gonna go build a fence around the yard |
| 20:42.23 | MrTelephone | god damn neighbors yard looks like shit |
| 20:42.56 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:42.57 | MrTelephone | digium should design phone kiosks |
| 20:43.10 | MrTelephone | get rid of those stinky quarter eaters |
| 20:43.16 | MrTelephone | haha |
| 20:43.20 | MrTelephone | alright have a good weekend guys |
| 20:43.49 | MrTelephone | struct piece_of_poop *MrTelephone |
| 20:43.51 | MrTelephone | :P |
| 20:43.56 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:45.18 | *** join/#asterisk galeras (n=galeras@201.244.199.31) |
| 20:45.52 | galeras | one minute of call recording = ?KB |
| 20:46.33 | Katty | why don't you call and find out (+ |
| 20:46.36 | Katty | (= |
| 20:46.43 | anonymouz666 | 100k |
| 20:46.44 | anonymouz666 | gsm |
| 20:46.48 | anonymouz666 | I think |
| 20:46.59 | galeras | i'm sizing a box to buy it! |
| 20:47.19 | galeras | anonymouz666: thanks |
| 20:47.20 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:47.41 | anonymouz666 | hi Katty |
| 20:47.59 | anonymouz666 | [TK]D-Fender was out of this channel? I can't believe |
| 20:48.11 | Katty | anonymouz666: hewwo. |
| 20:48.17 | Katty | [TK]D-Fender: GASP |
| 20:48.39 | [TK]D-Fender | :D |
| 20:51.07 | anonymouz666 | he smiles... |
| 20:51.13 | anonymouz666 | haha |
| 20:52.46 | Blumpkin | thanks for the help folks |
| 20:52.50 | Blumpkin | much appreciated |
| 20:54.32 | irule | can someone explain what is the new way of doing things in asterisk vs the old way? this is in regards to users.conf, it was mentioned in todays conference but I am interested in more technical details |
| 20:56.11 | *** join/#asterisk walter_rodrigues (n=walter@201-048-147-003.static.ctbctelecom.com.br) |
| 20:57.52 | [TK]D-Fender | irule, rm /etc/asterisk/users.conf |
| 20:58.20 | irule | [TK]D-Fender what is that? |
| 20:58.37 | walter_rodrigues | I am having problems with basic recognition of the TDM24xxP on kernel 2.6.22.1-27.fc7 ...when I modprobe wctdm24xxP I get this: Error inserting wctdm24xxp (/lib/modules/2.6.22.1-27.fc7/misc/wctdm24xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg) ...and DMESG returns this: wctdm24xxp: Unknown symbol pci_module_init |
| 20:58.43 | [TK]D-Fender | irule, Your IQ drops 20 points the moment you use it. |
| 20:58.47 | walter_rodrigues | Could anybody PLEASE hint me? thanks. |
| 21:00.07 | irule | [TK]D-Fender you still havent explained to me why |
| 21:00.23 | rene- | D-Fender: hehe |
| 21:01.24 | rene- | i think some politicians probably have used /etc/asterisk/users.conf more than once or twice |
| 21:01.48 | rene- | irule: users is an abstraction from the way asterisk works |
| 21:01.53 | [TK]D-Fender | irule, it crams too much shit together in one place assuming that VM box have to be related to "phone devices", etc. |
| 21:02.12 | rene- | irule: it is an easier way for new people to get into asterisk |
| 21:02.38 | rene- | probably it can be a faster but less flexible way for somebody who know what is doing |
| 21:02.51 | rene- | as in a bit faster and a lot less flexible |
| 21:04.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 21:05.52 | [TK]D-Fender | rene-, tiny big, though too much shit crammed together |
| 21:06.04 | [TK]D-Fender | bit* |
| 21:06.48 | rene- | it could be |
| 21:07.13 | rene- | a way for somebody that came from freepbx |
| 21:07.25 | rene- | to understand a bit more about asterisk |
| 21:07.51 | rene- | tho i have never used so wtf do i know |
| 21:07.52 | rene- | heh |
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| 21:13.29 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
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| 21:20.57 | [TK]D-Fender | Katty, They're down the street from me, want me to go loose a fire-hose on them directly? ;) |
| 21:22.19 | Katty | [TK]D-Fender: yes'r |
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| 21:23.44 | mercestes | Katty!!! |
| 21:27.10 | *** join/#asterisk mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 21:27.21 | nDuff | Under what circumstances is the "record" tone (from indications.conf) used? |
| 21:28.49 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:30.35 | mercestes | nDuff: If you call a monitor and tell it to indicate it is recording with a tone, I believe. |
| 21:31.16 | Innatech | seriously, what's the deal with Vitelity.....they don't answer their phones, at all. Don't respond to requests via web-form. Are they in some kind of trouble? |
| 21:31.55 | nDuff | mercestes: I don't see such a flag documented for the Monitor app. |
| 21:31.58 | *** join/#asterisk gardo (n=gardo@203.82.42.106) |
| 21:32.26 | mercestes | http://forum.voxilla.com/provider-rants-raves/beware-sixtel-net-vitelity-net-iax-cc-exgn-net-17614.html |
| 21:34.25 | *** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 21:35.45 | Innatech | Actually, I've read that thread. According to the last post, the OP ended up being very happy with them. |
| 21:35.59 | mercestes | http://troykelly.com/category/geek/ |
| 21:36.51 | Innatech | But now, if you call either their sales or customer service, you get but into a queue that times out, it takes a message and then disconnects you. No one ever calls back. That'd be pretty infuriating if your service stopped working. |
| 21:37.28 | Innatech | ah, well *that* is interesting. |
| 21:39.50 | *** join/#asterisk sysreq (n=sysreq@197.219-ppp.3menatwork.com) |
| 21:42.56 | mercestes | Innatech, most ppl seem to be happy with them but..."They dont' respond to me" tends to be a very common theme I'm seeing a google = vitality |
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| 21:46.08 | Innatech | yeah....kinda hard to figure out. |
| 21:46.40 | Innatech | I can't even get pricing out of them. |
| 21:47.06 | Katty | mercestes: ! |
| 21:47.08 | Katty | mercestes: :> |
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| 21:50.02 | Mad|Cow | Can someone help me understand the bandwidth requirements to sustain one SIP call. I have a T1 and trying to figure out how many simultanious calls I can support over SIP (using G711). |
| 21:50.31 | mercestes | http://www.voip-info.org/wiki/view/iax.cc |
| 21:50.48 | *** part/#asterisk galeras (n=galeras@201.244.199.31) |
| 21:51.04 | mercestes | just a common theme |
| 21:51.32 | mercestes | Lots of "service is great" posts but anytime I see bad service ,I see bad customer service. |
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| 22:02.14 | CunningPike | Mad|Cow: Reckon on roughly 80kbps per call |
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| 22:03.16 | *** join/#asterisk kkn088 (n=kikoun@84.4.50.39) |
| 22:05.52 | CunningPike | Mad|Cow: http://www.computerweekly.com/Articles/2007/06/28/225024/voip-bandwidth-fundamentals.htm |
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| 22:16.06 | *** join/#asterisk MrBIOS (n=aperez@dsl093-167-230.sfo1.dsl.speakeasy.net) |
| 22:16.21 | MrBIOS | Hi there, where can I get a reasonably new version of pwlib? the version linked from the openh323 site is ancient |
| 22:16.24 | MrBIOS | and incompatibl |
| 22:16.29 | MrBIOS | e+e |
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| 22:23.01 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:30.36 | tzafrir_home | MrBIOS, apt-get install libopenh323-dev #if you have the proper distro. |
| 22:30.38 | tzafrir_home | Otherwise: |
| 22:31.00 | MrBIOS | I don't but got it taken care of |
| 22:31.08 | tzafrir_home | http://www.voxgratia.org/ (or the sourceforge site) |
| 22:31.10 | Innatech | I'm reluctant to throw $35 at Vitelity just to try to get their attention. |
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| 23:03.46 | pc500 | 7960 |
| 23:03.57 | *** part/#asterisk pc500 (n=pc500@71.216.58.142) |
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| 23:15.43 | Nugget | I'll take "What phone is a pain in the ass to obtain firmware for" for $200, Alex. |
| 23:17.15 | ManxPower | Which firmware? |
| 23:17.37 | ManxPower | since I'm the network admin, that is not an issue for me 8-) |
| 23:17.46 | Nugget | the 7960 firmware. |
| 23:18.03 | Nugget | I was just taking pc500's nonsequiter as an opportunity to make a joke |
| 23:18.57 | Qwell[] | It's easier to pirate than it is to buy it from Cisco. And they do a good job of not letting people pirate it. |
| 23:19.06 | Nugget | heh, yup. exactly. |
| 23:19.46 | [hC] | i bought one smartnet contract for $9 and downloaded the entire firmware directory |
| 23:19.52 | [hC] | which contained firmware for every cisco phone, sip and sccp |
| 23:19.56 | [hC] | and called it a day. |
| 23:20.18 | Nugget | It's hard to find a company willing to do the two hours of paperwork involved in selling a smartnet contract for their cut of $9. |
| 23:20.43 | [hC] | yes, that is quite true. |
| 23:20.47 | Nugget | less hard if you're buying a phone at the same time, certainly, but even that's no guarantee |
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| 23:22.31 | ManxPower | Nugget: Well what do you expect when you buy Cisco phone? |
| 23:32.17 | Qwell[] | ManxPower: I expect Cisco to punch a baby and/or kill a kitten every time somebody buys a phone. |
| 23:32.36 | Qwell[] | would definitely explain all the cost and hassle |
| 23:34.27 | [hC] | Qwell: are you still working on skinny? |
| 23:38.53 | riddlebox | does anyone have a howto, on setting up a way to email me my voicemail messages when I receive one? |
| 23:40.59 | levi | riddlebox: I'm pretty sure there's one in the wiki |
| 23:41.08 | riddlebox | ohh ok |
| 23:41.21 | levi | Probably the example voicemail.conf has that set up, too. |
| 23:44.53 | MrBIOS | is there a sample asterisk init script? |
| 23:45.54 | riddlebox | levi, I have it setup to email the voicemail, but I need to know how to configure, sendmail or exim4 or something so that it will use an account to relay out of my mailserver |
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