00:02.09 | afrosheen | nice and quiet here today.... |
00:03.03 | asterisk99 | anyone familiar with zaptel compiles on ubuntu? |
00:03.21 | afrosheen | not really but what is it doing |
00:04.03 | asterisk99 | I am getting a 'Module wctdm missing' when I modprobe wcfxs |
00:04.22 | afrosheen | what kernel, and are you using udev |
00:04.44 | asterisk99 | afrosheen: 2.6.12 |
00:04.54 | afrosheen | so you did the make linux26 for the make command right |
00:04.55 | ManxPower | asterisk99, what zaptel version are you using? |
00:05.05 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
00:05.08 | Ariel_ | hello folks |
00:05.14 | asterisk99 | afrosheen: cd .. |
00:05.16 | afrosheen | sup Ariel_ |
00:05.31 | asterisk99 | afrosheen: 1.2 |
00:05.41 | Ariel_ | just stopping by to see how everyone is doing. |
00:06.27 | ManxPower | wcfxs and wctdm are the same module |
00:06.37 | afrosheen | yep, has been for awhile |
00:06.46 | afrosheen | hence it being missing |
00:07.02 | afrosheen | asterisk99: did you do a 'make' or a 'make linux26' when you built it |
00:07.32 | asterisk99 | afrosheen: I did a make linux26 |
00:08.27 | afrosheen | asterisk99: did the make install throw out any errors at the end? |
00:08.47 | asterisk99 | afrosheen: NOt that I noticed |
00:09.15 | afrosheen | ManxPower: how will I know if fxotune worked or not |
00:10.05 | afrosheen | asterisk99: what happens when you do a 'modprobe wctdm' |
00:10.34 | ManxPower | afrosheen, should be in dmesg |
00:10.37 | ManxPower | -- Setting echo registers: |
00:10.37 | ManxPower | -- Set echo registers successfully |
00:10.40 | asterisk99 | I am getting a 'Module wctdm not found' |
00:11.26 | ManxPower | asterisk99, that would mean that that module was not installed in the right place. |
00:11.39 | ManxPower | I suspect you have a version mismatch between your kernel source and your running kernel |
00:11.51 | afrosheen | asterisk99: have a look here http://blog.kirjava.net.nz/posts/76 |
00:11.53 | asterisk99 | afrosheen: The make linux26 ends with 'leaving directory usr/src/linux-headers-2.6.12-386' |
00:12.06 | afrosheen | that should be right |
00:12.53 | afrosheen | ManxPower: sweet, it worked on reboot :) |
00:13.42 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
00:15.24 | afrosheen | gah this zaptel dtmf issue is driving me up the wall |
00:15.35 | afrosheen | half the time it doesn't match the digits on the ivr |
00:15.47 | asterisk99 | afrosheen: Am digesting link u sen |
00:16.21 | afrosheen | asterisk99: ok |
00:17.00 | ManxPower | afrosheen, what interface? |
00:17.20 | ManxPower | afrosheen, in analog ports messed up gains can screw up DTMF. |
00:17.30 | afrosheen | how so |
00:17.51 | afrosheen | coz these are fxo ports with incoming business lines jacked in |
00:18.07 | ManxPower | well if the gain is too high, then the audio will be distorted, if it's too low, the system may not detect all DTMF digits. |
00:18.22 | ManxPower | also relaxdtmf=yes can mess it up too. |
00:18.38 | afrosheen | well the strange thing is, if you call the ivr from a cellphone, the digits are always perfect |
00:18.49 | afrosheen | if you dial it from some landlines, dtmf tones get lost |
00:19.06 | afrosheen | so I guess ma bell isn't transmitting as loudly as t-mobile |
00:19.17 | ManxPower | afrosheen, perhaps your cell company transmits DTMF at a higher or lower gain and analog phones. |
00:19.40 | afrosheen | how do I balance that without introducing nasty echo, just bump the rx gain a little? I think it's already at +3 |
00:19.45 | ManxPower | increse your rxgain by 4 then try, if that does not work try decreasing it by 4 |
00:20.01 | ManxPower | so if rxgain-0 then try rxgain=4 and rxgain=-4 |
00:20.36 | ManxPower | rxgain won't mess with echo, only txgain |
00:20.40 | afrosheen | whew |
00:20.45 | afrosheen | that's what I'm worried about |
00:21.09 | afrosheen | after I make changes should I only reboot * |
00:21.30 | [av]bani | gaaaaaaaaaaaah |
00:21.36 | [av]bani | FuriousGeorge mangled my edits |
00:21.47 | [av]bani | FuriousGeorge: "preview" is your friend |
00:21.53 | ManxPower | either of these two ways: 1) stop and start asterisk or 2) unload chan_zap.so then load chan_zap.so |
00:22.12 | ManxPower | reload chan_zap.so might do it, but I don't know. |
00:23.14 | Ariel_ | ManxPower, don't you have to also restart the zaptel service? |
00:24.22 | ManxPower | Ariel_, why? gains are in /etc/asterisk/zapata.conf not /etc/zaptel.conf |
00:25.16 | Ariel_ | great I have always stoped and restarted the service as well. I guess I was doing more then what is needed. Good to know. |
00:25.28 | *** join/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net) |
00:25.55 | themikester60 | has anyone here used a wrt54g with a third party firmware installed in their network setup for an asterisk server? |
00:26.04 | ManxPower | any changes to /etc/zaptel.conf needs a ztcfg to apply the changes (which will kill all active calls) |
00:26.56 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167033017.pppoe-dynamic.nb.aliant.net) |
00:28.04 | Abydos313 | themikester60 someone in here said they have it running on openwrt |
00:28.18 | ManxPower | themikester60, I suspect most people here just buy compatable hardware |
00:28.34 | ManxPower | rather then spending 20 hours making something work that you can just buy for under $500 |
00:28.56 | themikester60 | ManxPower, I imagine they would, I just have a slight issue for anyone who uses a wrt54g, I've been asked to try to get the QoS features of it to work as a switch rather than a router but have had no luck so far using the alchemy firmware |
00:29.05 | Abydos313 | maybe asterisk will be included in future firmware like seavofts |
00:29.10 | MGSsancho | themikester60 i am |
00:29.20 | afrosheen | themikester60: yeah I had trouble with dd-wrt firmware's QoS also |
00:29.25 | MGSsancho | wrt54gs |
00:29.31 | ManxPower | Ahrimanes, Wireless QoS. |
00:29.35 | Ariel_ | I have uses openwrt and ti work |
00:29.41 | ManxPower | Next they will be asking you to turn lead into gold. |
00:29.42 | Abydos313 | i have freemans' basic if you want to try it |
00:29.43 | themikester60 | MGSsancho, have you found a way to get the router to act as a switch while still utilizing the QoS featuers? |
00:29.44 | orlok | wireless QoS is an oxymoron :) |
00:29.58 | MGSsancho | themikester60 no :( |
00:30.20 | ManxPower | What most people don't realize is that their network doesn't need QoS |
00:30.21 | afrosheen | I'm not looking for wireless QoS, just QoS for the polycom plugged into it |
00:30.27 | themikester60 | I'm not using any wireless with this router, it is simply being used a switch, QoS works as a router but I'm trying to find a firmware that can make the QoS work as a switch |
00:30.41 | themikester60 | ManxPower, what makes you say this? |
00:30.41 | ManxPower | or should I say, that their LAN does not need QoS |
00:30.47 | MGSsancho | dont use the wan port then <_< |
00:31.01 | orlok | hmm |
00:31.09 | orlok | these grandstreams dont wanna register with * |
00:31.21 | ManxPower | themikester60, have you measured the data thruput of your LAN? |
00:31.24 | Abydos313 | freemans basic will let you assign port priority :) |
00:31.25 | orlok | has anybody used a GXP-2000 with asterisk? |
00:31.29 | themikester60 | MGSsancho, the wan port isnt used, I have 3 ports in use, 1 and 2 should have higher priority than 3 and I have configured it so within the QoS page, but no QoS is occuring at all |
00:31.32 | MGSsancho | that will make it a switch. then disable dchp |
00:31.38 | themikester60 | ManxPower, yes I have |
00:31.58 | ManxPower | Even using ulaw, a call will use something like .008 of a 100Mbps LAN |
00:32.06 | themikester60 | Abydos313, so does alchemy, but it doesn't seem to be having any effect, have you tested freemans basic's QoS when operating as a switch instead of a router? |
00:32.07 | MGSsancho | only need QoS over wan when you wanna make calls and your dling "linux distros" on bit torrent |
00:32.31 | themikester60 | MGSsancho, I have disabled dhcp, its working as a switch like I want it to but I'm not noticing any QoS occurring |
00:33.10 | MGSsancho | i havent been able to get working like that unless you do static routing to get multiple routers working |
00:33.12 | themikester60 | I agree with all of you on the part that QoS isnt really necessary, but the networks we install these asterisk servers on tend to have high bandwidth usage so for these instances it is needed or else the calls begin to have breakup |
00:33.16 | ManxPower | themikester60, there's a good chance the hardware does not support QoS between the LAN ports. |
00:33.19 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:33.38 | MGSsancho | thats possible |
00:33.41 | themikester60 | ManxPower, I had assumed so but was hoping I might find out otherwise, do you think it is because of the hardware though or is it possible a different firmware could control it? |
00:33.52 | ManxPower | themikester60, *nod* If they have high bandwidth usage on the local lan, then perhaps QoS would help. |
00:34.05 | ManxPower | On the other hand, it might be easier to just put each phone on it's own Ethernet port. |
00:34.07 | MGSsancho | useally it means QoS over wan since thats the slowest port |
00:34.21 | afrosheen | QoS is critical for this stuff...if the packets don't make it out of the wan to our trunking provider, people cry |
00:34.27 | ManxPower | assuming the phones and the asterisk server are on the same switch. |
00:34.40 | themikester60 | MGSsancho, I had assumed so, though within the options for the QoS you can specify QoS as either for Wan or Wan&LAN so I assumed that it might work |
00:34.42 | Ariel_ | for qos to work on a local lan you need to have a good layer 2 or 3 switch in the mix |
00:34.50 | ManxPower | afrosheen, QoS on WAN is important |
00:34.53 | afrosheen | and it's necessary for all our remote users with phones at their homes |
00:35.02 | MGSsancho | if you have a lot of users, then you might want to get a more powerfull switch to handle the load |
00:35.24 | themikester60 | I was considering a more powerful switch, but the modified firmware makes these routers handle QoS beautifully |
00:35.31 | themikester60 | if only they had QoS as a switch it would be perfect |
00:35.39 | MGSsancho | werd |
00:35.41 | afrosheen | I'd like to be able to roll out a router package that guarantees our home users don't have issues with calls regardless of what they're doing on their computers |
00:35.56 | MGSsancho | hmm |
00:36.04 | themikester60 | afrosheen, thats actually pretty similar to what I'm trying to do |
00:36.05 | MGSsancho | i see what you mean |
00:36.22 | afrosheen | I thought the wrt54g would be a good candidate so I bought one and played with different firmwares, could never get it to act right |
00:36.28 | ManxPower | If the Asterisk server is a dedicated machine, and the phones are not sharing their port with a PC, and the phones and the server are on the same switch, I can't imagine QoS helping any. |
00:36.41 | afrosheen | ManxPower: that's probably true |
00:37.05 | MGSsancho | i agree |
00:37.19 | MGSsancho | have the phones on a powered QoS switch |
00:37.22 | ManxPower | since if the switch is doing it's job, phones<->server won't have to contend for bandwith |
00:37.26 | themikester60 | ManxPower, I have that setup, but the PCs are peers to the asterisk server, so traffic theyre generating through the switch is killing calls |
00:37.29 | MGSsancho | and use a another one for PCs |
00:37.47 | ManxPower | themikester60, define "peers" |
00:38.41 | ManxPower | MGSsancho, why a QoS switch |
00:38.52 | MGSsancho | good point |
00:39.04 | MGSsancho | put phones on a POE switch |
00:39.05 | themikester60 | ManxPower, the two networks converge on one point which is the switch (wrt54g) which I'm trying to get to utilize QoS, each network has its own router with its own static IP, so one dsl line with two static IPs. When traffic gets too demanding on the PC network, it effects the quality of the cals |
00:39.15 | ManxPower | Now, if the phones and the PC are sharing the same port, etc. |
00:39.28 | themikester60 | MGSsancho, forgive me for sounding stupid, but what exactly is a POE switch? Power over ethernet? |
00:39.33 | MGSsancho | yes |
00:39.34 | ManxPower | themikester60, of lan calls or wan calls? |
00:39.40 | themikester60 | wan calls |
00:40.03 | ManxPower | Well, yes, for WAN you would want to do some QoS |
00:40.27 | ManxPower | I thought these phones and the server were all in the same location, since you talked about QoS between the LAN ports |
00:40.36 | themikester60 | ManxPower, right, and thats why I was trying to get the wrt54g to use QoS when acting as a switch, but so far the QoS only seems to work when its operating as a router |
00:40.41 | MGSsancho | may also have to keep them separate. have the phones on their own dsl line. and pcs on another |
00:40.47 | themikester60 | oh no, sorry about that I shouldve clarified |
00:41.12 | MGSsancho | since each call is like 10K, and in the US, we only get like 30K up :( |
00:41.14 | *** join/#asterisk jpablo (n=jpablo@201.139.54.46) |
00:41.22 | themikester60 | MGSsancho, that would make my life much easier, but my boss seems to want to conserve and only use one DSL line, because the voice really doesnt consume much bandwidth, but it does get affected if too much is being used |
00:41.25 | ManxPower | Unless you control the other end of your internet access line, QoS will still be somewhat limited. |
00:41.33 | jpablo | hey people, any idea what' current chan_bluetooth cvs ? |
00:41.38 | ManxPower | Since you can only apply QoS to TRANSMITTED traffic. |
00:42.06 | themikester60 | ManxPower, well I have control over it to some degree, enough that when I had QoS working as a router and I had the bandwidth being used to the maximum the calls suffered no loss |
00:42.08 | MGSsancho | true |
00:42.20 | ManxPower | themikester60, you are just starting to realize why I don't put calls on a WAN |
00:42.39 | afrosheen | someone needs sdsl |
00:42.58 | MGSsancho | tell ur boss that dsl is like $14 a month. and if you want to expand as a buisness, the current dsl can only handle so much traffic |
00:43.10 | ManxPower | And why I turn down cheap customers that want me to do a half-assed job. |
00:43.18 | MGSsancho | mmmm sdsl |
00:43.24 | MGSsancho | or get a T1 |
00:43.41 | afrosheen | if he's too cheap for sdsl or an upgrade, he sure as hell won't get a t1 |
00:43.49 | afrosheen | how many people work there, 4 or 5? |
00:43.51 | ManxPower | MGSsancho, if they won't spend money on a real switch, they are not going to spend money on a T-1/E-1 |
00:44.03 | MGSsancho | i know trying to be funny :P |
00:45.13 | Nugget | it's not as effective as outbound, obviously, but it's much more effective than you'd expect. |
00:45.43 | Nugget | for sites where tcp is the bulk of the traffic, it's quite feasible to have useful inbound queuing |
00:46.22 | Nugget | that said, I think it's downright foolish to want to do business voip over the internet. |
00:47.41 | ManxPower | CTRL-A DEL There, all caught up on asterisk-users. And people wonder why I don't want web based "forums" |
00:48.03 | Nugget | web forums are the cesspools of the internet. |
00:48.30 | jpablo | grr, where's people getting channel bluetooth |
00:49.11 | MGSsancho | this chan is about VOIP systems |
00:49.17 | MGSsancho | not bluetooth sorry |
00:49.27 | Nugget | chan_bluetooth is an asterisk module, you doof. |
00:49.29 | jpablo | humm, isn't this channel about asterisk ? |
00:49.33 | Nugget | that couldn't be more on-topic |
00:50.03 | afrosheen | argggh this dtmf stuff is unfixable |
00:50.23 | jpablo | afrosheen, what's the problem? |
00:50.28 | themikester60 | sorry I walked away for a bit, the more expensive connection resolution doesnt really work here because my boss is trying to implement QoS as more of a "Save Our Ass" feature so that the customers never have to complain about call quality issues |
00:50.35 | ManxPower | Perhaps by reading this page: http://www.asterisk.org/developers |
00:50.43 | afrosheen | my zaptel interfaces (fxo) don't register dtmf tones correctly most of the time |
00:50.48 | themikester60 | thats why I'm trying to get a QoS switch, we've tried more expensive ones in the past but none have ever quite worked well enoug |
00:51.00 | afrosheen | I've tried everything from rxgain = -4 to rxgain = 10 and nothing changes |
00:51.02 | themikester60 | the QoS on the wrt54g worked well enough, but it appears so far to work only when its functioning as a router |
00:51.13 | ManxPower | Um, if you don't want customers to complain about call quality, don't send the calls over the Internet |
00:51.13 | themikester60 | but one of our installs needs it to work as a switch, thats where I find myself up river without a paddle |
00:51.16 | jpablo | afrosheen, analog or {e,t}1? |
00:51.27 | afrosheen | jpablo: analog, tdm400 card |
00:51.34 | themikester60 | ManxPower, yea but the whole focus for us is VOIP |
00:51.47 | themikester60 | ManxPower, thats how we supply the majority of channels for the users |
00:51.51 | jpablo | afrosheen, works fine for me ... |
00:52.03 | ManxPower | themikester60, What are you going to do if the box simply cannot do LAN QoS |
00:52.10 | afrosheen | jpablo: lol thanks buddy |
00:52.20 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
00:52.31 | *** join/#asterisk welles (n=welles@61.150.43.113) |
00:52.44 | themikester60 | ManxPower, most likely find a different switch solution, something is bound to be able to work as a QoS switch, I'd just rather find a solution with this switch and move on to more important issues |
00:52.50 | welles | hi al |
00:53.08 | afrosheen | arg this is so retarded, it gets the digits from my cellphone perfect every time, but not from an inbound landline call |
00:53.16 | jpablo | afrosheen, doesn't zaptel has something liek relaxdtmf ? |
00:53.24 | afrosheen | relaxdtmf didn't help either :( |
00:53.48 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
00:53.51 | jpablo | afrosheen, how many diferent phones have you tested ? maybe the landline phone is b0rked ... |
00:53.54 | afrosheen | I need to just get our business lines ported to commpartners and take a big hammer to this tdm400 card |
00:54.10 | ManxPower | Sorry, from here: http://www.asterisk.org/download |
00:54.11 | afrosheen | jpablo: about 10 different landlines, all sbfc |
00:54.18 | ManxPower | relaxdtmf selcom helps |
00:54.21 | afrosheen | <PROTECTED> |
00:54.26 | welles | why my asterisk can not receive packets from another asterisk using iax2 protocol? |
00:55.11 | jpablo | welles, that's network question not really a * question. |
00:55.48 | robin_z | here, you'll be needing this |
00:55.53 | jpablo | afrosheen, does zaptel has a jitter buffer ? maybe disabling it will help |
00:56.21 | ManxPower | zaptel has something called a jitter buffer, but it's not one, in the sense of VoIP |
00:56.42 | *** join/#asterisk wellng (n=welles@61.150.11.54) |
00:56.47 | ManxPower | afrosheen, can you fax over the lines? |
00:57.06 | ManxPower | afrosheen, are you talking about RECEIVING DTMF or SENDING DTMF |
00:57.11 | jpablo | afrosheen, is your card missing interrupts somehow ? |
00:57.31 | afrosheen | receiving dtmf |
00:57.44 | glm2k | afrosheen: does a verbose setting show the dtmf digits being pressed? |
00:57.46 | afrosheen | and it's only having trouble receiving from landlines, we don't use any faxing |
00:58.07 | ManxPower | because if you are trying to send DTMF to remote IVRs then that's the classic DTMF length time problem, see digits.h in the zaptel source, change the DTMF length, recompile |
00:58.11 | afrosheen | glm2k: it shows a few, i.e. I dial 235 and it sees 255 or 25 or 355 |
00:58.45 | ManxPower | afrosheen, classic rxgain probleml. |
00:59.01 | ManxPower | I had to bring my rxgain up to 8 before my asterisk would see DTMF from the PSTN |
00:59.08 | glm2k | ouch |
00:59.37 | afrosheen | ManxPower: is this related at all to the settings fxotune writes out |
00:59.43 | ManxPower | glm2k, I believe the audio level coming from the telco is low. |
00:59.50 | afrosheen | mine wrote 8,0,0- for each module |
00:59.59 | ManxPower | afrosheen, no. |
01:01.49 | *** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net) |
01:03.17 | *** part/#asterisk themikester60 (n=mikey@209-83-240-53-static.dsl.oplink.net) |
01:04.17 | afrosheen | lol I turned the rxgain up to 50 and now it won't detect anything |
01:06.24 | glm2k | time to complain to sbc? |
01:06.31 | afrosheen | yeah I suppose |
01:06.44 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
01:06.44 | afrosheen | anyway I need to get these numbers ported away from sbc |
01:06.55 | afrosheen | I fully trust commpartners now |
01:07.21 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
01:07.34 | *** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
01:08.28 | fjean | Hello guys, I need some help regarding DeadAGI, apparently when I dial directly an IAX2 route from extensions.conf I hear the ring tone, but the same route from a DeadAGI would not give me the ring tone...anybody has a hint ? :- ) |
01:09.25 | fjean | (I use 1.0.9) |
01:10.19 | websae | is it possible to have sip and iax connections at the same time with an asterisk box? |
01:12.00 | jpablo | websae, yes |
01:12.20 | jpablo | that's the whole fun of asterisk |
01:13.36 | websae | i need to setup an client asterisk box to connect to mine via sip |
01:13.41 | websae | what's the best way to do this? |
01:14.00 | *** join/#asterisk welles (n=welles@61.150.60.46) |
01:15.36 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
01:15.48 | jpablo | create friends in both ends, and voila |
01:16.29 | websae | on my end...have a user setup with type=friend |
01:16.44 | websae | and then on his end, setup the same exact friend |
01:16.46 | websae | with same settings |
01:16.48 | websae | correct? |
01:17.01 | jpablo | not same settings, the ip will change |
01:17.06 | websae | host= my ip on my end |
01:17.12 | websae | host=his ip on his end |
01:17.19 | jpablo | that should work |
01:17.23 | websae | do i set his ip on my box |
01:17.26 | jpablo | then just dial(SIP/peername/bla) |
01:17.40 | jpablo | just like you typed |
01:18.03 | websae | i am just trying to understand how it knows to peer to my server |
01:18.20 | websae | i think i would have to type is ip as host on my server |
01:18.22 | jpablo | you host = bla in his side ? |
01:18.31 | websae | and he'd have to type my ip for host on his |
01:18.42 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
01:18.50 | websae | ohh---- host=hisip on my* server |
01:19.05 | websae | and host=my ip on his*server |
01:19.13 | jpablo | yes |
01:19.29 | websae | okay |
01:19.37 | websae | don't need a register command or anything? |
01:20.11 | welles | hi all. now i want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while |
01:20.11 | welles | iaxclient can not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated! |
01:21.34 | websae | nat |
01:21.34 | jpablo | websae, you only need to register if you are in a dynamic ip |
01:21.46 | websae | he has a dynamic ip address |
01:22.13 | jpablo | then you configure in your vox host = dynamic and he must put a register command |
01:22.28 | jpablo | s/vox/box |
01:22.33 | websae | what would hist register command be? |
01:22.40 | websae | register => |
01:22.43 | websae | what the |
01:22.46 | fjean | i am not sure if i got it, but I had this experience too where if I use an * box and softfone out of the same public IP to connect to another public IP I would not be able to call back the * box that has the same public IP as my softfone, it would teel me "in a call" |
01:22.48 | websae | *then |
01:23.25 | jpablo | don't remember search the wiki or see the examples that are already in sip.conf |
01:23.45 | jpablo | is basically will have a user name (the peer name in your box) a secret, and your ip |
01:25.02 | websae | jpablo: any familiarity with iax? |
01:25.07 | jpablo | yes |
01:25.14 | websae | is that hard to setup? |
01:25.21 | jpablo | almost the same as sip |
01:25.33 | websae | what's different |
01:25.43 | jpablo | humm, it's called iax :P |
01:26.00 | jpablo | so you use dial(iax/peer) :P |
01:26.33 | jpablo | really the configurations are basically the same, the dial command gets more options as it understand more options on the remote box like contexts and stuff |
01:26.49 | jpablo | but at a configuration level they are basically the same |
01:27.19 | welles | jpablo, will u help me? |
01:27.28 | websae | iax more stable then sip ? |
01:27.47 | jpablo | maybe with especific questions im having my own fun |
01:28.03 | a1fa | hey |
01:28.07 | jpablo | humm, same stability, it uses just one port so it doesn't have all the rtp nat nightmares |
01:28.10 | a1fa | what is that voice mail menu? |
01:28.14 | a1fa | VoiceMailMenu()? |
01:28.47 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
01:29.07 | jpablo | voicemailmain |
01:29.17 | a1fa | uff.. thats what i ment.. |
01:29.30 | a1fa | it wont move me into menu.. it wants an extension |
01:29.44 | a1fa | i want it to be a menu where you put your extension |
01:29.46 | jpablo | you need to configure voicemail.conf |
01:29.57 | jpablo | what do you mean ? |
01:30.04 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-20-52.buckeyecom.net) |
01:30.11 | a1fa | voicemail.conf is configured |
01:30.12 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
01:30.46 | jpablo | a1fa, i didn't understand you, what do you want to do |
01:31.27 | a1fa | Feb 20 01:31:16 WARNING[11776]: pbx.c:1688 pbx_extension_helper: No application 'VoicemailMain ' for extension (myphones, 28346, 1) |
01:31.32 | a1fa | this is the error i get now |
01:31.54 | jpablo | humm |
01:32.18 | jpablo | do you have /usr/lib/asterisk/modules/app_voicemail.so ? |
01:32.19 | file | you have a space... in your extensions.conf entry that calls VoicemailMain |
01:32.20 | a1fa | weerd |
01:32.23 | jpablo | ah, wait |
01:32.27 | jpablo | there a space, corrent |
01:32.35 | a1fa | its there |
01:32.40 | jpablo | delete it |
01:32.43 | a1fa | but for some reason it is pwned by root |
01:32.52 | a1fa | delete it why? |
01:33.01 | file | he meant delete the space |
01:33.06 | a1fa | ok |
01:33.13 | file | you have something like: |
01:33.18 | a1fa | right i see it now |
01:33.19 | a1fa | :( |
01:33.23 | file | yeah... fix it |
01:35.05 | a1fa | interesting |
01:35.08 | a1fa | it is not authenticating |
01:35.36 | a1fa | uff it is |
01:35.48 | a1fa | really interesting |
01:36.01 | *** join/#asterisk in-side (n=lowgitek@es-217-129-30-48.netvisao.pt) |
01:36.10 | in-side | Hi anybody here works with ser or openser? |
01:36.39 | in-side | I was wondering if anybody knows how to concat two avp strings to only one? |
01:36.41 | FuriousGeorge | hmmmmm, so how hard would it be to monitor parking spots from the manager and set dev states based on that? |
01:36.45 | a1fa | are you guys using md5secret or clear secret? |
01:37.01 | FuriousGeorge | cuz im not gonna be able to get it working in the dialplan just right |
01:37.09 | FuriousGeorge | (with meetmes) |
01:37.39 | in-side | nobody here use ser? or openser ? |
01:39.57 | fjean | hi, anybody uses AGI->Exec("Dial... with the ring option ('r') in his script ? |
01:42.04 | in-side | why ? |
01:42.20 | in-side | I use it |
01:42.25 | in-side | wazz up? |
01:42.34 | [av]bani | FuriousGeorge: "preview" is your friend |
01:43.18 | fjean | in-side, it seems the 'r' option doesn work, we can hear nothing until the person answers... |
01:43.53 | *** join/#asterisk wellng (n=welles@219.144.238.194) |
01:44.17 | fjean | in-side, this how i do it: $dialstr = "IAX2/$res->{path}/$phone|60|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)|r"; |
01:44.42 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
01:44.56 | in-side | sorry i don't work with iax but let me check it.. |
01:45.08 | fjean | cool |
01:45.15 | in-side | why ":60000:30000) |
01:45.16 | file | fjean: remove the | in between ) and the r |
01:45.34 | fjean | ok..let me test |
01:46.05 | in-side | no man the | is need |
01:46.20 | FuriousGeorge | [av]bani: preview? preview what... btw, i decided the only way its gonna work right is with an API, so I may as well start reading |
01:46.59 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
01:47.16 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
01:47.29 | fjean | ok |
01:47.47 | file | fjean: did it work? |
01:47.55 | in-side | ya... |
01:47.59 | fjean | i am switching user, 2 secs |
01:48.00 | in-side | the | is not need sorry |
01:48.03 | in-side | it is a option |
01:48.09 | in-side | as r |
01:48.15 | in-side | so should be together |
01:49.10 | in-side | well nobody works here with ser sip proxy server? |
01:49.31 | in-side | or openser ? and knows how to contact two avps to only one? |
01:49.32 | fjean | in-side, nope, both ways don't work... |
01:49.43 | fjean | this is so trange |
01:49.58 | file | I'd ask why it's strange, but it'll just end up with me getting irritated |
01:50.24 | fjean | hehe |
01:50.33 | in-side | fjean: try to only send r |
01:50.34 | file | so I'll do it anyway |
01:50.37 | file | why is it strange? |
01:50.39 | in-side | to see if it works |
01:50.52 | fjean | right..let me check |
01:51.03 | file | I could give you a speech about what the difference is... |
01:51.53 | in-side | or try to put r first |
01:52.39 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
01:52.55 | fjean | this guy did not ring: $dialstr = "IAX2/$res->{path}/$phone|60|r"; |
01:53.15 | fjean | I am using a SIP user, could that be translation problem ? |
01:53.25 | in-side | err.. anyway r option sux |
01:53.34 | in-side | have no ideia just use sip on it |
01:53.38 | in-side | maybe codec |
01:53.41 | fjean | i know... |
01:54.06 | fjean | well, maybe the codec, i ll try a couple |
01:54.08 | file | in-side: you're an interesting individual... but anyway |
01:54.26 | file | fjean: what exactly is the situation, what's happening that should/shouldn't be happening |
01:54.54 | in-side | file: thanks for your opnion about myself but I didn't asked it... |
01:55.39 | in-side | fjean: why are you forcing the it ring anyway? |
01:55.42 | [av]bani | FuriousGeorge: voip-info. your edit borked my feature request |
01:55.58 | [av]bani | snom-360 |
01:56.32 | fjean | in-side, because when i dial from the script, the caller cannot hear ringing...but, the funny thing is that if i dial directly from extensions.conf it gives it |
01:57.33 | in-side | hmmm... sorry can't help you more, all my sip signals handling are done in openser |
01:57.44 | fjean | cool, no prob |
01:59.02 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
01:59.14 | Katty | hihi |
01:59.23 | Ariel_ | Katty, how are you? |
01:59.33 | Katty | getting better. |
01:59.36 | Katty | i think. |
01:59.47 | Katty | swelling should peak tonight |
02:00.15 | orlok | chan_sip.c: Registration from '<sip:1234@192.168.1.243;user=phone>' failed for '192.168.1.211' |
02:00.21 | in-side | fjean: progressinband ? |
02:00.32 | orlok | Any ideas? grandstream gxp-2000 to asterisk |
02:00.43 | in-side | is true for sip client? |
02:01.09 | fjean | in-side: mmmmm, good question, i guess its the default value, letme check |
02:01.29 | in-side | default is no |
02:01.31 | Ariel_ | swelling??? what did you do? Katty |
02:01.32 | in-side | so checkit out |
02:01.51 | in-side | take care that inband must be also set in sip client |
02:01.58 | fjean | ok |
02:02.05 | in-side | and not all codecs supports it properlly |
02:02.11 | Katty | Ariel_: surgeons cut out my wisdom teeth friday |
02:02.25 | justinu | file: what are these two talking about? |
02:02.26 | Ariel_ | argh sorry... I understand |
02:02.31 | justinu | blind leading blind? |
02:02.38 | Katty | and file won't come take care of me. |
02:02.40 | Katty | sniffle. |
02:02.56 | file | justinu: I like my ignore list |
02:02.59 | justinu | hah |
02:03.47 | fjean | in-side, isee it goes in general settings, not in individual blocks |
02:04.14 | in-side | Katty: tas a ver a ideia nao tas... tao prontos.. Vai te lixar o meu caralho! |
02:04.32 | *** join/#asterisk websae_ (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
02:04.47 | Katty | in-side: i speak english. |
02:05.02 | fjean | in-side, que palavrao feio ;-) |
02:05.10 | file | I speak I/O talk |
02:05.20 | in-side | Katty: tao o problema é teu! err.. not my problem sucker |
02:05.28 | in-side | fjean: anda aqui po pvt |
02:06.00 | *** join/#asterisk nozey (n=nozey@20150042085.user.veloxzone.com.br) |
02:06.35 | websae_ | i don't get why my phone sometimes has great sound quality---and then voice quality is gargley--gets lound and quiet, loud and quiet----any ideas anyone? |
02:06.48 | nozey | hi ... im trying to run asterisk here, i can dial my friend, but theres no sound ... i already opened 5060 udp(and tcp) at my router ... what can be wrong? |
02:06.49 | justinu | write(fd,"english, muthafucka, do you speak it?",53); |
02:06.56 | nozey | sorry for my english ... im brazilian #) |
02:07.13 | file | justinu: yes |
02:07.15 | file | :P |
02:07.16 | nozey | im using sip |
02:07.18 | justinu | cool |
02:07.50 | in-side | nozey: nao ligues pa estes gajos sao todos eliteee... err.. |
02:07.56 | Ariel_ | wow google could not translate that |
02:08.04 | Katty | in-side: err, k |
02:08.05 | in-side | file: ha um bug nisso pa |
02:08.32 | NewSole | on a sip call how can I pass it from asterisk server to outside server |
02:08.48 | justinu | quite simply |
02:09.24 | *** join/#asterisk mzo (i=user@ool-435193b3.dyn.optonline.net) |
02:09.25 | in-side | http://bugs.digium.com/view.php?id=3695 |
02:09.27 | nozey | heh in-side voce é espanhol? |
02:09.39 | in-side | nozey: nao mane sou brasileiro |
02:09.46 | Katty | i don't have the patience for this |
02:09.48 | mzo | someone give me a sledge-hammer |
02:09.54 | mzo | i want to smash this 9800 all in-wonder pro to bits |
02:09.57 | in-side | Katty: really cvall you mum |
02:10.01 | Katty | especially not after having my widom teeth out. |
02:10.06 | nozey | i can hear the ring, but theres no voice sound |
02:10.06 | Katty | in-side: she's right here, hunny. |
02:10.08 | justinu | Katty: i don't blame you... it's tiring even without the mouth pain |
02:10.32 | Katty | justinu: yeah, the stitches feel insanely tight. |
02:10.34 | in-side | Katty: whatever... |
02:10.53 | mzo | but insanely tight is sometimes a good thing |
02:11.03 | Katty | mzo: yeah, but not with stitches and swelling |
02:11.06 | justinu | hah |
02:11.19 | Katty | well, maybe swelling |
02:11.25 | Katty | but definately /not/ stitches |
02:11.44 | websae_ | any suggestions why my phone sometimes works great..no problems, but then in the middle of a call on and off it gets loud and soft, and voice becomes gargly????any suggestions, anyone? :) greatly appreciated |
02:11.50 | I-MOD | !mp3 |
02:11.56 | in-side | file: are you there??? |
02:11.59 | Ariel_ | websae_, network issues |
02:12.03 | I-MOD | sry |
02:12.04 | Ariel_ | crappy phone? |
02:12.04 | I-MOD | nm |
02:12.16 | websae_ | Ariel: have a grandstream |
02:12.23 | Ariel_ | then the 2nd |
02:12.25 | websae_ | i use to have a sipura---it was even worst! |
02:12.41 | Ariel_ | sounds like then it's your network what's it like |
02:12.42 | [av]bani | sipura 841 :D |
02:12.59 | websae_ | any suggestions |
02:13.04 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
02:13.10 | Ariel_ | websae_, more info needed |
02:13.19 | websae_ | what info? |
02:13.27 | websae_ | i connect with g729 |
02:13.33 | websae_ | to a server in LA---im in wisconsin |
02:13.41 | websae_ | the server is on a gig-e connection out there |
02:13.43 | Ariel_ | are you on cable |
02:13.49 | websae_ | static ip dsl |
02:14.03 | Ariel_ | what is your ping times |
02:14.10 | websae_ | 80ms |
02:14.14 | Ariel_ | also what firewall/nat router are you using? |
02:14.27 | websae_ | dlink |
02:14.31 | Ariel_ | try ulaw and see how that one sounds.... |
02:14.32 | websae_ | but the phone registers fine |
02:14.49 | Ariel_ | sound goes out on rtp ports between 10,000 to 20,000 |
02:15.06 | websae_ | ulaw---what's the speed on that |
02:15.23 | NovceGuru | 80kbps I think |
02:15.24 | Ariel_ | it's 80k but it's sound is better no compression needed |
02:15.26 | NovceGuru | with overhead |
02:15.45 | Ariel_ | if your on a static IP with dsl you should be able to do ulaw |
02:15.45 | nozey | i have no audio using sip ... can someone help? |
02:15.54 | nozey | im behind a router |
02:16.01 | Ariel_ | nozey, sure have you looked up on the wiki about nat |
02:16.04 | Ariel_ | ~doc's |
02:16.19 | nozey | yep |
02:16.19 | websae_ | iax---the answer to everything |
02:16.29 | websae_ | no worries about rtp and nat |
02:16.29 | Ariel_ | websae_, no it's not |
02:16.34 | nozey | well ... first i need to learn sip |
02:16.40 | nozey | then im going to try iax |
02:16.49 | Ariel_ | nozey, sip registers on port 5060 or 5061 |
02:17.03 | Ariel_ | they are needed to be open for registation |
02:17.16 | websae_ | yep |
02:17.17 | nozey | i opened 5060 both udp and tcp |
02:17.23 | Ariel_ | are the ports forwarded to the asterisk box |
02:17.26 | nozey | i can dial |
02:17.27 | websae_ | okay---then something with your voice packets |
02:17.27 | NewSole | on a sip call how can I pass it from asterisk server to outside server |
02:17.32 | nozey | yes Ariel_ |
02:17.40 | websae_ | <PROTECTED> |
02:17.42 | websae_ | with your provider |
02:17.43 | nozey | i can dial ... just can hear the other people speaking |
02:17.44 | in-side | nozey: check pvt |
02:17.50 | Ariel_ | how about rtp.conf what ports are there? have you forwarded them? |
02:18.27 | Ariel_ | again sound is via rtp from the range of 10,000 to 20,000 depending on device |
02:18.41 | nozey | well |
02:18.46 | in-side | the default starts at 8000 i think |
02:18.52 | Ariel_ | this can be set by editing the rtp.conf file located in the /etc/asterisk directory |
02:18.59 | Ariel_ | in-side, really |
02:19.01 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
02:19.04 | nozey | can i open only the 10.000 port? |
02:19.10 | in-side | nao |
02:19.11 | in-side | nopes |
02:19.13 | nozey | or should i open the range? |
02:19.21 | in-side | you need a range.. same srange |
02:19.25 | in-side | that is in rtp.conf |
02:19.30 | in-side | as Ariel_ told you |
02:19.30 | Ariel_ | 10,000 to 11,000 set it up on the rtp.conf for asterisk then forward the ports |
02:19.36 | nozey | ok |
02:19.37 | nozey | let me try |
02:19.39 | nozey | one minute |
02:19.42 | Ariel_ | you don't really need 10,000 ports open |
02:19.53 | in-side | ya.. |
02:20.58 | Ariel_ | wow just checked out the first server I setup with asterisk it was back in april 2002.... almost 4 years ago.... |
02:21.05 | nozey | udp, tcp or both? |
02:21.10 | Ariel_ | udp |
02:22.24 | Ariel_ | you might have to check your device like phone sipura likes in the range of 16,380 and cisco like 8000 range. (Seems like they don't like getting a standard setup). |
02:22.45 | websae_ | peering two asterisk servers...what does one have to do |
02:23.29 | Ariel_ | peering is fairly easy. if you only have a few channels needed use iax2 if not then more then 8 use sip |
02:23.45 | nozey | didint worked |
02:23.57 | gaupe | websae_: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
02:23.59 | Ariel_ | nozey, what devices |
02:24.35 | websae_ | on server a, make entry for server b and on server b make entry for server a? |
02:24.49 | Ariel_ | websae_, pretty much |
02:25.00 | Ariel_ | nozey, which version of asterisk do you have? |
02:25.23 | websae_ | server b is dynamic----so i have to do a register on server b with the credentials from the entry on server a? |
02:25.23 | nozey | 1.2.4 |
02:25.45 | nozey | i have 5060 and 10000-11000 opened and forwarded to my pc |
02:25.57 | Ariel_ | your pc |
02:25.58 | nozey | my friend is behind a router too |
02:26.06 | nozey | yes ... im running asterisk here |
02:26.10 | nozey | only for learn |
02:26.24 | Ariel_ | then you might have a problem with your friends router |
02:26.41 | Ariel_ | nozey, setup a free fwd account and do some testing from that |
02:26.51 | nozey | let me try |
02:26.54 | nozey | tkz again |
02:27.00 | Ariel_ | double nat is a bear to fix |
02:27.11 | nozey | i can see that |
02:27.15 | nozey | :( |
02:27.17 | Ariel_ | it can be fixed but takes a bit |
02:27.48 | Ariel_ | are you running it on vmware? |
02:28.24 | nozey | nop |
02:28.28 | nozey | slack 10.2 |
02:28.39 | *** join/#asterisk wellng (n=welles@61.150.43.113) |
02:28.44 | nozey | on local machine, not vmware |
02:28.55 | nozey | sorry for me english ... i know it sucks :( |
02:29.38 | Ariel_ | well it's better then my brazillian |
02:30.17 | nozey | heh :P |
02:31.02 | nozey | like what? |
02:31.05 | websae_ | dropped calls...what's the best way to debug that |
02:31.10 | websae_ | with 'sip debug' |
02:31.14 | websae_ | and just screen that |
02:31.19 | *** join/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com) |
02:31.20 | Ariel_ | ethereal |
02:31.36 | websae_ | what would i do with ethereal? |
02:31.42 | websae_ | i think i have that on my box |
02:31.46 | kc5cqm | quick question: I'm ztcfg isn't configuring a channel, but zttool is finding the unconfigured x101p car |
02:31.57 | kc5cqm | debian 3.1 |
02:32.04 | nozey | is ethereal a gui for tcpdump? |
02:32.22 | Ariel_ | ethereal gui hehehe |
02:32.31 | WasPhantom | well, ethereal uses the same pcap library to capture the packets |
02:33.03 | Ariel_ | or ngrep could help as well |
02:33.06 | mzo | i'm being stupidly brave and configuing FWD for asterisk :P |
02:33.12 | mzo | someone call me and see if something good happens? |
02:33.17 | websae_ | what command do i run with ethereal |
02:33.20 | websae_ | to do this debuggin? |
02:33.25 | Ariel_ | number |
02:33.33 | mzo | 749414 |
02:33.43 | mzo | maybe it'll blow up :P |
02:34.04 | Ariel_ | no just busy |
02:34.13 | mzo | hmm, is that their end or mine? |
02:34.18 | websae_ | do i just type ethereal at the command line? |
02:34.25 | websae_ | or do i need to specify something with ethereal? |
02:34.31 | Ariel_ | webmind, google how to use ethereal |
02:34.33 | mzo | what'd you get? a busy signal or an asterisk prompt? |
02:34.40 | Ariel_ | busy |
02:34.40 | websae_ | webmind? |
02:34.42 | websae_ | whose that? |
02:35.53 | nozey | well |
02:36.03 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
02:36.08 | mzo | im sure i fucked something up :P back soon with the 2x4 |
02:36.09 | nozey | i just tested with a friend that is not behind a router, and he can listen to me :-) |
02:36.34 | Ariel_ | nozey, ok so he can hear you. But you do not hear him |
02:36.49 | Ariel_ | which means your rtp is not coming back to you outbound is ok |
02:37.02 | nozey | yep |
02:37.06 | nozey | router problem |
02:37.23 | Ariel_ | ngrep and ethereal are programs you run in the command line |
02:37.41 | justinu | tethereal is cli |
02:37.45 | justinu | ethereal is gui |
02:38.54 | Ariel_ | http://www.ethereal.com/docs/man-pages/tethereal.1.html |
02:39.44 | Ariel_ | nozey, most routers will open and leave the ports open when it goes out |
02:42.57 | mzo | hmm, iax status: rejected = bad? :P |
02:43.29 | Ariel_ | mzo, well yes |
02:44.11 | mzo | haha, i must have made a typo :P |
02:45.22 | mzo | i wonder if there is a lag |
02:46.56 | *** join/#asterisk dmaust (n=dmaust@adsl-67-120-175-165.dsl.lsan03.pacbell.net) |
02:46.59 | dmaust | hi |
02:48.37 | mzo | anyone got an idea? :P |
02:48.37 | dmaust | I just set up Asterisk a couple days ago and was trying to connect to FWD. When I set it to verbose output, it said "IAX2/192.246.69.186:4569/7 is circuit-busy". |
02:48.37 | mzo | http://pastebin.com/563801 |
02:48.42 | dmaust | What does that mean? |
02:49.10 | mzo | no idea, i get rejected :P |
02:49.13 | mzo | oh wait |
02:49.14 | mzo | duh |
02:50.01 | dmaust | after that message, I get Registration of '749404' rejected: Registration Refused |
02:51.16 | mzo | i checked my email i thought maybe i had to confirm, but nothing, same error :P |
02:51.18 | dmaust | mzo, what were you trying to do? |
02:51.32 | mzo | oh just connect using the instructions on the site :P |
02:51.51 | dmaust | I see |
02:51.58 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
02:51.59 | dmaust | exactly the same as what I was trying to do |
02:52.44 | mzo | oh well |
02:53.29 | dmaust | I just know 55555 is tech support :-). |
02:53.39 | mzo | i'll mess with it tommorow |
02:53.40 | dmaust | unfortunately, it's rejecting my login |
02:53.44 | dmaust | one idea... |
02:54.10 | dmaust | they do have an option to activate IAX in their Extra Features section |
02:54.15 | mzo | i'llmess with it later |
02:54.44 | dmaust | okay, good luck. |
02:54.49 | mzo | i just turned that on |
02:54.51 | mzo | let's see |
02:55.09 | dmaust | weird... |
02:55.12 | dmaust | I just turned it off |
02:55.13 | dmaust | and now it works |
02:56.06 | dmaust | well.. not quite, but I did get Registered 'dmaust' (AUTHENTICATED) at 192.168.5.128:4569 |
02:56.08 | mzo | i dunno. |
02:56.10 | mzo | haha. |
02:56.12 | mzo | lucky you |
02:56.41 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
02:57.22 | dmaust | still doesn't work |
02:57.33 | dmaust | That message just got my hopes up for a second. |
03:01.39 | dmaust | oh. That was a totally unrelated message. Since I turned the verbosity up, now it notifies me when my other IAX2 client re-registers. |
03:02.56 | Abydos313 | hey guys, i have asterisk already installed and want to start using cvs.. do i need to remove before i setup cvs |
03:05.47 | *** join/#asterisk CrashHD (i=user@c-67-187-241-56.hsd1.ca.comcast.net) |
03:05.48 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300771.sympatico.ca) |
03:07.06 | asteriskmonkey | i need some help with an echo issue.. the issue is the caller from the pstn side gets echo .. caller => pstn => pri => asterisk box => sip phone |
03:07.50 | wellng | hi all. i meet problems. just like this url describle http://lists.digium.com/pipermail/asterisk-users/2003-December/030787.html. anyone would like to help me? |
03:08.04 | *** join/#asterisk TuckerAdel (n=Tucker@58.160.196.17) |
03:09.17 | TuckerAdel | <PROTECTED> |
03:11.48 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
03:12.04 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-128-96.gdrpmi.dsl-w.verizon.net) |
03:12.06 | wellng | anyone will help me:my iaxclient receive too many vnak frames. |
03:14.41 | *** join/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2) |
03:15.28 | nozey | sip + nat is a pain |
03:15.35 | *** part/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2) |
03:15.45 | *** join/#asterisk Aum-Aum (i=Aum-Aum@66.226.248.2) |
03:16.02 | TuckerAdel | <PROTECTED> |
03:17.05 | Aum-Aum | I get the following error starting Asterisk |
03:17.09 | Aum-Aum | <PROTECTED> |
03:17.22 | orlok | My grandstream is failing to register - ANy hints? |
03:17.25 | Aum-Aum | Modules load and ztcfg finds ports |
03:17.51 | asteriskmonkey | anyone know of any toronto telco 1004hz test phone numbers? |
03:19.18 | TuckerAdel | <PROTECTED> |
03:19.58 | TuckerAdel | <PROTECTED> |
03:22.00 | nozey | Ariel_, tkz for the help ... i will try agian later |
03:22.03 | nozey | see ya |
03:26.26 | *** part/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com) |
03:26.39 | tengulre11 | Hi,all |
03:26.58 | tengulre11 | I want to building a asterisk system without hardware, anybody can give me some tips? |
03:27.24 | TuckerAdel | <PROTECTED> |
03:30.58 | Aum-Aum | tengulre11 - you mean no Zaptel hardware |
03:31.13 | Abydos313 | ztdummy needs to be loaded for timing issues |
03:31.26 | Abydos313 | so you have to install zaptel anyways according to book |
03:31.47 | rt | anyone here running asterisk on openwrt? |
03:32.08 | rt | just looking for a thumbs up or thumbs down on it before I go to the effort of trying it out. :-) |
03:34.12 | shido6 | Aum-Aum 2.6 + ? |
03:35.00 | Aum-Aum | Yes |
03:35.05 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
03:35.41 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
03:35.46 | *** part/#asterisk TuckerAdel (n=Tucker@58.160.196.17) |
03:35.50 | Aum-Aum | Shido6 - Yes 2.6.14-1 |
03:36.12 | shido6 | README.udev is your friend |
03:36.15 | Aum-Aum | Shido6 - Software versions |
03:36.15 | Aum-Aum | <PROTECTED> |
03:36.15 | Aum-Aum | <PROTECTED> |
03:36.15 | Aum-Aum | <PROTECTED> |
03:36.15 | Aum-Aum | <PROTECTED> |
03:36.15 | Aum-Aum | <PROTECTED> |
03:36.17 | Aum-Aum | <PROTECTED> |
03:37.07 | shido6 | in teh zaptel dir you can make config after you make install and run /etc/init.d/zaptel restart to get better results... after you've editted /etc/zaptel.conf of course. |
03:46.27 | Aum-Aum | Shido6 - udev all ready set |
03:46.40 | shido6 | kewl |
03:46.50 | shido6 | zaptel all compiled? |
03:46.58 | Aum-Aum | yes |
03:47.02 | shido6 | make install |
03:47.04 | shido6 | done ? |
03:47.11 | shido6 | make config done, too ? |
03:47.32 | Aum-Aum | make clean, make linux26, make install, make, make config all done |
03:48.00 | shido6 | what kind of zaptel gear do you have installed? |
03:48.06 | orlok | wtf |
03:48.08 | Aum-Aum | Using init.d/zaptel gives same result |
03:48.14 | Aum-Aum | TDM2413E |
03:48.23 | shido6 | oooh, fun! |
03:48.54 | *** join/#asterisk bmg505 (n=leon@dsl-165-129-29.telkomadsl.co.za) |
03:49.03 | shido6 | how many FXS modules and how many FXO modules are REALLy on the board? |
03:49.04 | orlok | icmp dest. unreach can be sent as a response to a udp packet, correct? |
03:49.39 | orlok | hmm, but theres no udp packets in this dump.. |
03:49.59 | Aum-Aum | one green and three red (1 FXS and 3 FXO) FXS in slot one |
03:50.25 | Aum-Aum | Slot one being closest to bracket |
03:54.54 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
03:56.07 | *** join/#asterisk litage (n=nick@203.220.55.70) |
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04:06.37 | *** join/#asterisk nozey (n=nozey@20150042085.user.veloxzone.com.br) |
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04:12.51 | nozey | zael, man ... this is a pain |
04:22.16 | *** join/#asterisk websae_ (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
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04:26.16 | Mavvie | aha. now that makes sense.... |
04:26.53 | Mavvie | the fact that ^W deleted until the beginning of the line instead of to the next non-whitespace place is an emacs thing |
04:27.20 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
04:32.08 | *** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
04:32.25 | jeebusroxors | haha badass - dual output is amazing |
04:33.33 | websae_ | how does one setup a peer? |
04:33.38 | [hC] | dual output? |
04:34.05 | websae_ | have the same entry in each server? |
04:34.16 | websae_ | and then host=the opposite server? |
04:34.23 | websae_ | type=friend? |
04:35.22 | [hC] | voip-info.org has an example labelled 'dual servers' |
04:35.23 | [hC] | i think |
04:35.34 | websae_ | hrm |
04:35.53 | websae_ | very confusing |
04:39.36 | *** join/#asterisk pengyong (n=lala@222.185.5.141) |
04:47.28 | websae_ | how's everybody doing? |
04:49.16 | Mavvie | looks like the removal of ChanIsAvail and just pushing everything through zap/g2 has resolved my congestion problems. |
04:49.33 | Mavvie | question of course is, like always: why? |
04:53.51 | *** join/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
04:55.55 | Uberbot | Hi all. |
04:56.26 | *** join/#asterisk ctmsydney (n=ctm@obrien6.lnk.telstra.net) |
04:58.06 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
04:58.29 | Uberbot | Some time ago, I heard a rumour that they were going to re-write the voicemail system in *. Is this true? |
05:00.31 | rajiv | Uberbot: i heard the same thing. there is a wiki page about it |
05:01.06 | Uberbot | I'll look. If not, I'm thinking of doing something different as an agi... Any thoughts? |
05:01.48 | Corydon76-home | Some time ago, the voicemail system WAS rewritten. We're on version 2. |
05:15.00 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
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05:23.20 | *** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
05:23.23 | *** join/#asterisk Ubeyguy (n=chatzill@204.9.119.130) |
05:23.40 | Ubeyguy | hello can someone help me with this: Xlite - 21:19:41.9 Proxy slot #1 () - Failed to register! error-code: 403, msg: 'Forbidden' ? |
05:25.36 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
05:30.12 | *** join/#asterisk Maxx4life (n=Maxx4lif@71-35-210-12.slkc.qwest.net) |
05:35.44 | Mavvie | hmm... new syntax of CUT is euhm... very unusual |
05:38.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:52.07 | Ubeyguy | Mavvie: ?? |
05:56.45 | Mavvie | well, if you're used to "Cut(foo=bar,,2)",then "Set(foo=${CUT(bar,,2)})" is something you need to look at once or twice before you understand it. |
05:58.13 | Ubeyguy | i did |
05:58.19 | Ubeyguy | i cant figure out whats wrong |
05:59.29 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
05:59.45 | wellng | my iaxclient send Excessive VNAK frame .anybody know why? |
06:04.44 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
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06:19.32 | [av]bani | ... |
06:21.53 | *** join/#asterisk Z-Knight (i=user@cpe-67-10-25-30.houston.res.rr.com) |
06:23.24 | Z-Knight | stupid question - What are some motherboards you'd suggest for an * setup (possibly with TDM2400P card) ?? please? |
06:24.51 | MGSsancho | any? |
06:25.17 | Z-Knight | anyone here? |
06:26.24 | Z-Knight | don't all of you respond at once! ;) |
06:27.24 | MGSsancho | any mobo will do |
06:27.31 | MGSsancho | all you need is a PCI slot |
06:27.36 | Z-Knight | not really...it needs the right pci slot |
06:27.47 | MGSsancho | and enough room in your case for that long as board |
06:27.50 | Z-Knight | but tthe TDM2400P has a particular slot...am I wrong? |
06:28.05 | Z-Knight | I'm looking at the pictures and it looks like it won't fit most boards |
06:28.12 | Z-Knight | that's what has me confused |
06:28.33 | MGSsancho | its pci 2.2 |
06:28.38 | MGSsancho | any new comp will have it |
06:28.52 | MGSsancho | just need to make sure you have clearence |
06:28.52 | Z-Knight | ok...here is a stupid observation then |
06:28.58 | Z-Knight | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM2400P |
06:29.01 | Z-Knight | look at that picture |
06:29.10 | MGSsancho | i am |
06:29.12 | Z-Knight | the pci keying is "small....large....small" |
06:29.15 | Z-Knight | see the bottom of it |
06:29.21 | Z-Knight | http://www.digium.com/index.php?menu=whatpcislot |
06:29.24 | MGSsancho | thats fine |
06:29.25 | Z-Knight | now look at that picture |
06:29.29 | Z-Knight | the slots 3 4 5 |
06:29.35 | MGSsancho | oh |
06:29.35 | Z-Knight | don't have that little front notch |
06:29.40 | MGSsancho | dont matter |
06:29.42 | Z-Knight | you need slot 2 |
06:29.44 | MGSsancho | 345 will do |
06:29.48 | Z-Knight | ahh |
06:29.49 | Z-Knight | ok |
06:29.49 | MGSsancho | 12 are PCIX |
06:29.59 | Z-Knight | yeah I misspoke |
06:30.06 | Z-Knight | so you think those are fine then? |
06:30.08 | Z-Knight | the 3 4 5 |
06:30.12 | MGSsancho | no 12 are pci 64bit |
06:30.15 | MGSsancho | yeah |
06:30.20 | Z-Knight | yes, ok |
06:30.23 | Z-Knight | thank you |
06:30.30 | Z-Knight | have you ever used the TDM2400P ? |
06:30.34 | MGSsancho | i would personaly hold it down with electrical tape or something |
06:30.39 | MGSsancho | nope :( |
06:30.42 | Z-Knight | ahh |
06:30.42 | Z-Knight | k |
06:30.44 | MGSsancho | <-- poor |
06:30.48 | Z-Knight | :) |
06:30.55 | Z-Knight | yeah that thing is expensive from what I see |
06:31.15 | Z-Knight | I was just wondering how really long it is...I suspect even smaller micro-atx cases should support it |
06:31.36 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
06:31.50 | MGSsancho | not sure about it |
06:32.10 | Z-Knight | what hardware have you used? any? or mostly SIP? |
06:32.18 | MGSsancho | better of getting it. then once you have it in ur hands, goto Frys or something and look to see what mobos and cases will house it |
06:32.28 | Z-Knight | true |
06:32.31 | MGSsancho | use me router |
06:32.37 | MGSsancho | WRT54GS |
06:32.41 | MGSsancho | <_< |
06:32.44 | Z-Knight | you embedded it in there? |
06:32.48 | MGSsancho | no |
06:32.50 | Z-Knight | * in wrt54gs? |
06:32.53 | MGSsancho | yes |
06:33.00 | MGSsancho | flash the firmware |
06:33.02 | Z-Knight | yup |
06:33.07 | Z-Knight | was that easy? |
06:33.12 | MGSsancho | reletivly |
06:33.21 | MGSsancho | all you have to do is update it :) |
06:33.22 | Z-Knight | I'm guessing there is no way to get voicemail |
06:33.28 | MGSsancho | then enable the asterisk module |
06:33.35 | MGSsancho | then use sip |
06:33.38 | Z-Knight | k |
06:33.40 | MGSsancho | you can |
06:33.42 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
06:33.47 | MGSsancho | the router can |
06:33.49 | Z-Knight | really? how much space does the flash have? |
06:33.56 | Z-Knight | I mean the router? |
06:33.57 | MGSsancho | but my router has like 8mb flash so its limited |
06:34.08 | MGSsancho | 4mb ram 8mb ram |
06:34.12 | MGSsancho | something like that |
06:34.35 | MGSsancho | i wish routers had usb ports >_< |
06:34.36 | Z-Knight | ahh...you'd have to do a hardware mod to add more then....beyond my capability right now |
06:34.48 | MGSsancho | yeah |
06:34.49 | Z-Knight | some do...one port |
06:34.55 | Z-Knight | wait |
06:34.56 | Z-Knight | never mind |
06:34.57 | MGSsancho | or the WRT54GL has 32mb flash |
06:35.00 | Z-Knight | I'm thinking cable modem |
06:35.30 | Z-Knight | so with the router you have support for at least 4 direct SIP lines |
06:35.49 | Z-Knight | i'm guessing with a hub, etc you can then get more |
06:36.16 | MGSsancho | yup |
06:36.31 | MGSsancho | but limited to the bandwith your ISP will give you |
06:36.36 | Z-Knight | yeah |
06:36.44 | MGSsancho | and the dinky 200mhz processor in my router |
06:36.55 | Z-Knight | that is really cool....I want to setup a * server that can support an office of 12 people ...that is my goal |
06:37.13 | MGSsancho | then yeah lol you will need a real pc |
06:37.16 | Z-Knight | yup |
06:37.25 | Z-Knight | that's why the question of motherboards |
06:37.30 | MGSsancho | and sdl or a good cable connection |
06:37.31 | Z-Knight | really hard to settle on one |
06:37.34 | MGSsancho | *sdsl |
06:37.35 | Z-Knight | yes |
06:37.52 | Z-Knight | it will likely be a mix of SIP and regular lines |
06:37.59 | MGSsancho | yeah |
06:38.14 | MGSsancho | get a mobo that has like 4 or more pci slots |
06:38.17 | MGSsancho | <_< >_> |
06:38.30 | Z-Knight | why so manY? |
06:38.44 | MGSsancho | add more lines |
06:38.46 | MGSsancho | :) |
06:38.48 | Z-Knight | ahh |
06:38.57 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
06:38.59 | MGSsancho | but 12 isnt that bad |
06:39.11 | MGSsancho | you can just do the entire phone system over ehternet |
06:39.13 | Mavvie | 124 lines per PCI slot... |
06:39.18 | MGSsancho | IP phones |
06:39.40 | MGSsancho | and VOIP service from a provider. then just have your PBX do its job |
06:39.48 | MGSsancho | 124 <_< damn |
06:40.11 | Z-Knight | say I have 4 regular incoming land lines then I just need one of the DIgium cards with the 4 inputs |
06:40.34 | Z-Knight | I'd only need the TDM2400 if I want to connect to regular phones (not IP phones) |
06:40.47 | Z-Knight | else I'd need a router for the IP phones |
06:40.49 | Z-Knight | right? |
06:40.49 | glm2k | Mavvie: wouldn't that be dependent on the codec? 124 using ulaw? |
06:41.12 | MGSsancho | the output ones are for umm |
06:41.15 | Mavvie | glm2k: no, it's depending on the hardware. A QuadPRI card is 124 lines. |
06:41.16 | MGSsancho | making calls |
06:41.19 | Mavvie | :-) |
06:41.23 | MGSsancho | like you connect a regular phone to it |
06:41.54 | glm2k | Mavvie: egad. |
06:42.14 | glm2k | all you really need is 1 slot in most cases |
06:42.44 | MGSsancho | https://shop.eikonex.net//catalog/product_info.php?products_id=250 |
06:42.48 | MGSsancho | that? |
06:42.51 | Z-Knight | my goal is to upgrade an office using regular phones now and move them to VOIP....so actually probably have 2 hard lines coming in and then upgrade most phones to IP phones while keeping a couple that for regular connection |
06:43.13 | MGSsancho | you can keep the 4 lines |
06:43.33 | MGSsancho | cuz you can only use those 4 lines at most (internal is independant |
06:43.35 | Z-Knight | yeah, I could, but it would be better to migrate slowly away from land lines |
06:43.51 | MGSsancho | yeah good thinking |
06:43.51 | Z-Knight | I'd replace the 4 lines with VOIP eventually |
06:43.55 | MGSsancho | :) |
06:44.05 | Z-Knight | is that about the right price for the TE411P? $2400? |
06:44.19 | Z-Knight | I thought those would be cheaper, but that's about the price I'm seeing |
06:44.25 | Z-Knight | seems expensive as hell for only 4 lines |
06:45.04 | Z-Knight | or am I missing something about what this card actually does? |
06:45.21 | X-Rob | it's 120 lines, not 4. |
06:45.30 | X-Rob | 4 ports. 1 port == 30 lines (E1) |
06:45.34 | Z-Knight | ahhh |
06:45.42 | Z-Knight | I am missing something then....I need to read more |
06:45.57 | MGSsancho | so you need internal phones? |
06:46.00 | MGSsancho | 12? |
06:46.15 | X-Rob | Normal lines use a TDM card, not a TE |
06:46.18 | Z-Knight | well, the office has 12 internal phones now (all regular phones, not IP phones) |
06:46.19 | X-Rob | TE is ISDN |
06:46.22 | Z-Knight | yes |
06:46.25 | Z-Knight | k |
06:46.32 | Z-Knight | so the TDM2400P |
06:46.37 | Z-Knight | for the current phones |
06:47.09 | Z-Knight | I guess I'm confused then how you use this TE411P....what does one of the 4 connections go to? a router? |
06:47.10 | X-Rob | I stronly suggest you use SIP phones, not analog. |
06:47.21 | X-Rob | Many more features. |
06:47.24 | Z-Knight | yes |
06:47.26 | Z-Knight | I agree |
06:47.28 | X-Rob | Z-Knight, It's an ISDN connection, it comes from your telco. |
06:47.46 | Z-Knight | so the ISDN connects to the those ports? |
06:47.50 | Z-Knight | those are all incoming ports? |
06:47.57 | Z-Knight | support for 4 ISDN lines? |
06:48.02 | X-Rob | Support for 4 PRI's |
06:48.05 | X-Rob | Primary Rate Interfaces |
06:48.06 | Z-Knight | AHH |
06:48.08 | Z-Knight | yes |
06:48.09 | Z-Knight | thanks |
06:48.21 | X-Rob | or T1's for the yanks |
06:48.25 | Z-Knight | :) |
06:48.35 | MGSsancho | mmm T1 |
06:49.00 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
06:49.56 | Z-Knight | so your PBX would have the PRI lines coming in to this TE411P, then the PBX connects to a router/switch through a regular ethernet card, and then all other IP phones connect to the router/switch as well. |
06:50.01 | Z-Knight | that is the basics right? |
06:50.19 | X-Rob | Yeah |
06:50.23 | Z-Knight | sweet |
06:50.32 | X-Rob | doing the cablign is the easy bit though |
06:50.37 | X-Rob | it's the configuration that's the difficult one. |
06:50.42 | Z-Knight | I can see that |
06:50.46 | Z-Knight | there is so much in * |
06:50.53 | Z-Knight | freaking steep learning curve |
06:51.05 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
06:51.09 | Z-Knight | I want to write my own interface to it using...make a control panel using Java |
06:51.36 | X-Rob | There's a flash operator's panel |
06:51.41 | Z-Knight | I know |
06:51.47 | X-Rob | that might give you some hints. |
06:52.00 | Z-Knight | yes, but that is sort of like an Admin phone receptionist panel |
06:52.12 | Z-Knight | I want AMP I think |
06:52.19 | Z-Knight | trying to remember the name...the PHP tool |
06:52.26 | Z-Knight | PHP interface to the PBX |
06:52.32 | Z-Knight | for configuration of it |
06:52.35 | X-Rob | AMP |
06:52.37 | X-Rob | I'm a developer |
06:52.50 | Z-Knight | I guess I was mistaken by saying control panel...I mean config panel |
06:52.52 | Z-Knight | that's cool |
06:52.58 | Z-Knight | why PHP by the way? |
06:53.02 | X-Rob | *shrug* |
06:53.16 | Z-Knight | any thought to doing JAVA? Do you know anyone doing a JAva interface? |
06:53.59 | Z-Knight | basically I'd like to do what Fonality has |
06:54.11 | X-Rob | Why java? |
06:54.12 | Z-Knight | and what AMP is |
06:54.16 | X-Rob | all the config has to be done on the server |
06:54.19 | Z-Knight | *shrug* |
06:54.22 | Z-Knight | :) |
06:54.24 | X-Rob | so you'll have a java gui that talks to .. what? |
06:54.28 | Z-Knight | I like java |
06:54.30 | Z-Knight | no |
06:54.35 | Z-Knight | java webserver |
06:54.38 | Z-Knight | like PHP server |
06:54.41 | Z-Knight | no different |
06:54.45 | Z-Knight | not java gui |
06:54.55 | Z-Knight | just using Java servlets, JSPs |
06:55.00 | Z-Knight | sorry should have clarified |
06:55.08 | Z-Knight | mysql interface, etc |
06:57.16 | Z-Knight | I'm still not decided, but I felt that maybe I can write a Java interface so I can learn it from scratch and really understand the ins/outs of controlling * |
06:57.39 | *** join/#asterisk camonz (n=camonz@200.8.20.210) |
06:57.42 | camonz | hi |
06:57.48 | Z-Knight | hey |
06:59.57 | Z-Knight | X-Rob....how hard is the AMP code to grasp? I know that is a nebulous question but I'm curious what your feels are on it? |
07:00.18 | X-Rob | It's not all that difficult really - espeically with the v2 stuff (currently in CVS) as it's all modular. |
07:00.29 | Z-Knight | that's good |
07:01.25 | Z-Knight | I'll need to go into then, I've not learned PHP yet but can't be any more difficult than ASP,JSP,etc ....plus if the base is there it might be easier to use PHP |
07:01.32 | Z-Knight | is a PHP server hard to config? |
07:03.14 | X-Rob | no, it's usually already enabled in most linux distributions |
07:03.39 | Z-Knight | that's better than Java...geez, I'd have to setup a Tomcat server |
07:04.19 | Z-Knight | for your stuff do you use Asterisk@Home or do you just use your own flavor of Linux and install * on it? |
07:04.33 | Z-Knight | or what would you suggest? |
07:05.25 | X-Rob | If you're starting out, grab A@H |
07:05.42 | X-Rob | It's a nice beginners piece, but it's quite complex under the hood |
07:05.57 | X-Rob | so you can get the feel of it, and figure out how stuff works by looking at the generated configs. |
07:06.17 | Z-Knight | I am starting out, though I do have programming/computer experience and I'd like to get a system that I could sell |
07:07.22 | Z-Knight | I'm trying to figure out if for a production server (or whatever you'd call it) I'm not sure I should go with Asterisk@Home, though I really don't have any reason that I shouldn't |
07:08.19 | X-Rob | *shrug* give it a shot. |
07:08.25 | Z-Knight | hehe |
07:08.35 | X-Rob | It's a good distro (CentOS 4.2) which I'd recommend anyway, and it's got all the required bits you need |
07:08.45 | Z-Knight | yes |
07:08.49 | X-Rob | so even if you don't end up using AMP to manage *, it's still a good starting point. |
07:08.58 | Z-Knight | I really don't know why I wouldn't go with *@home |
07:09.19 | Z-Knight | thanks for all your help/info |
07:12.49 | wellng | hi Z-Knight |
07:12.52 | Z-Knight | hey |
07:12.59 | camonz | if my * box is behind a NAT device i should set the externip value to my pub ip address right? |
07:13.18 | Z-Knight | trying to think... |
07:13.21 | Z-Knight | one sec to confirm |
07:13.36 | camonz | i've been running some tests with some friends and we're experiencing 1 way audio only |
07:13.55 | Z-Knight | that's what I did |
07:14.01 | Z-Knight | do you have nat=yes? |
07:14.14 | Z-Knight | are they using xlite phones by chance? |
07:14.17 | camonz | normally the client can listen to me but i'm not getting their rtp packets |
07:14.19 | camonz | yep |
07:14.28 | camonz | but nat=yes is when the client is behind a nat |
07:14.42 | camonz | i have both cases, the client and the * server are behind different nats |
07:15.08 | Z-Knight | do you have reinvite=no and canreinvite=no on the sip.conf for them |
07:15.15 | camonz | nope |
07:15.21 | Z-Knight | I think you may need that |
07:15.36 | Z-Knight | and also when I had this issue I had to make sure all the portforwarding was set |
07:15.45 | Z-Knight | you not only need the 5060 port |
07:16.03 | Z-Knight | but * uses ports 10000 to 20000 (I think) as well for the SIP (I think again) |
07:16.15 | camonz | 10k to 20k for rtp audio |
07:16.18 | Z-Knight | the port forwarding was one thing that got me |
07:16.42 | Z-Knight | wait...maybe nat=no ( I don't recall what is should be ) |
07:17.04 | camonz | what i really can't understand as to why i'm not getting the audio packets from the client is that the * server is on my router dmz |
07:17.22 | camonz | so i should be getting all outside packets |
07:17.37 | Z-Knight | yeah, but you sure your friend is not blocking it |
07:17.49 | Z-Knight | can you get him to remove his router out of the equation |
07:18.15 | camonz | nope, but i ran tests this afternoon with his gf wich is on a public ip |
07:18.19 | camonz | one way audio as well |
07:18.19 | Z-Knight | the last time I did this config was about 3 weeks ago so I managed to forget my solutions |
07:18.25 | camonz | she can hear me but i cannot hear her |
07:18.27 | Z-Knight | ok |
07:20.08 | Z-Knight | for my config that worked I had this: |
07:20.11 | Z-Knight | type=friend |
07:20.13 | Z-Knight | nat=no |
07:20.17 | Z-Knight | username=... |
07:20.25 | Z-Knight | secret=.. |
07:20.29 | Z-Knight | host=dynamic |
07:20.36 | Z-Knight | context=testcontext |
07:20.40 | Z-Knight | reinvite=no |
07:20.44 | Z-Knight | canreinvite=no |
07:20.58 | camonz | ok |
07:20.59 | Z-Knight | also in [general] I had |
07:21.07 | Z-Knight | externip = my external ip (hehe) |
07:21.19 | Z-Knight | localnet = 192.168.0.1/255.255.255.0 |
07:21.26 | Z-Knight | not sure if those are needed |
07:21.46 | Z-Knight | I also accidentally had a nat=yes in the [general] but I'm not sure it even belongs there |
07:22.12 | camonz | i'm wondering right now if adding externip value in [general] is going to screw up the set up for my house network |
07:22.17 | Z-Knight | then I totally opened up my router to allow my friend to connect and the PBX to be totally accessible |
07:22.42 | camonz | i have 4 interfaces for my lan, as well as 2 extensions outside of it |
07:22.44 | Z-Knight | I can't help you with that because I don't quite know |
07:23.05 | camonz | no prob :->, gonna change the value and reload to see what happens |
07:23.21 | Z-Knight | good luck.... |
07:24.03 | Z-Knight | I had my box succeeding 3 weeks back and then I had to take it apart so I need to do that same test soon again....I'm surprised a detailed description of this is not available |
07:24.29 | Z-Knight | by the way...do their XLITE phones register successfully? (does it say logged in on the xlite?) |
07:24.58 | camonz | yep, |
07:25.17 | camonz | for all extensions, the in house ones and the outside the nat ones |
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07:25.23 | Z-Knight | I'd try the reinvite,canreinvite and see what happens |
07:25.27 | camonz | the prob is with the audio |
07:25.32 | Z-Knight | yes I know what you mean |
07:25.42 | Z-Knight | I had the exact same thing and then I started tinkering |
07:26.04 | Z-Knight | unfortunately between the time it didn't work and the time it did work I changed like 3 things so I can't pinpoint which was the solution |
07:27.27 | Z-Knight | I'm really thinking that having this (canreinvite=no) for each user will make it work |
07:27.37 | Z-Knight | oh and nat=no |
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07:30.25 | puzzled | morning |
07:30.44 | Z-Knight | good morning |
07:31.01 | camonz | morning |
07:31.09 | Z-Knight | any look, camonz? |
07:31.14 | camonz | yep |
07:31.18 | Z-Knight | any luck* |
07:31.24 | camonz | for my lan it still works |
07:31.26 | Z-Knight | geez...I can't type |
07:31.28 | camonz | aparently |
07:31.35 | camonz | wanna run a test |
07:31.42 | Z-Knight | sure |
07:31.55 | Z-Knight | let me get xlite |
07:31.58 | Z-Knight | and mic |
07:31.59 | X-Rob | wtf is with this '*' shit? My son does it when he's msn'ing me. |
07:32.09 | camonz | ok |
07:38.49 | Qwell | X-Rob: ? |
07:40.58 | X-Rob | <Z-Knight> any look, camonz? |
07:41.01 | X-Rob | <Z-Knight> any luck* |
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07:41.28 | X-Rob | if you use regexps, jbot even dixes it for you. |
07:41.31 | X-Rob | s/dixes/fixes/ |
07:42.14 | Qwell | X-Rob: kids don't know regex :p |
07:42.21 | X-Rob | they damn well shoud! |
07:42.30 | Qwell | teach your kid :p |
07:42.33 | X-Rob | Should be taught in primary school along with clicking and double clicking! |
07:42.36 | Qwell | maybe he'll teach others |
07:42.43 | Qwell | and it'll spread through the internet |
07:42.49 | Qwell | and eventually...we'll never see it again. :D |
07:42.54 | X-Rob | teh interwebs! |
07:43.00 | Qwell | intarweb |
07:43.05 | X-Rob | thazzit. |
07:44.01 | Qwell | don't msg me |
07:44.08 | X-Rob | I didn't. |
07:44.10 | Qwell | not you |
07:44.13 | X-Rob | I know |
07:44.15 | X-Rob | coz I didn't! |
07:44.15 | Qwell | :p |
07:44.29 | X-Rob | who do we get to mock? |
07:44.31 | Qwell | wellng: |
07:44.58 | MGSsancho | my mom!! |
07:45.02 | X-Rob | wellng takes it up the arse, doo dah, doo dah. |
07:50.08 | *** join/#asterisk [hC] (n=hardcore@S0106000fb56d814d.vc.shawcable.net) |
07:52.31 | wasim | <PROTECTED> |
07:56.12 | Mavvie | Zap/99-1 s@SJH-AAPT1:1 Rsrvd (None) |
07:56.18 | Mavvie | heh... wonder who to get rid of these beautifies |
07:56.19 | *** join/#asterisk apardo (n=apardo@87.218.45.71) |
07:56.22 | Mavvie | beauties |
07:57.05 | X-Rob | I used to work for AAPT |
07:57.08 | X-Rob | many moons ago |
07:57.14 | Qwell | aapt? |
07:57.25 | X-Rob | Qwell, small telco in .au |
07:57.35 | *** join/#asterisk lorinc (n=ang@caracas-0632.adsl.interware.hu) |
07:57.44 | X-Rob | Mavvie, I got better! |
07:57.50 | X-Rob | (They turned me into a newt!) |
07:58.00 | Mavvie | :-P |
08:01.20 | `Sauron | Mavs |
08:01.30 | Mavvie | now I have a PRI full of Rsrvd channels, and trunkavail() automatically jumped to the second PRI! |
08:01.31 | Mavvie | yay |
08:01.34 | Mavvie | hi `Sauron |
08:01.38 | `Sauron | you're all bark no bite, buddy :) |
08:01.53 | Mavvie | `Sauron: all hissing, no clawing. |
08:03.13 | X-Rob | Bad kitty! |
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08:03.26 | X-Rob | *squirt squirt* |
08:03.26 | Mavvie | oh oh |
08:03.40 | Mavvie | double oh oh |
08:03.47 | Mavvie | it's raining outside again. |
08:03.52 | Mavvie | that's two loads of washing getting wet. |
08:04.01 | X-Rob | Bugger. |
08:04.13 | Mavvie | is only hanging there now for three days. |
08:04.17 | X-Rob | Gladstone's good for that. We don't get rain here. |
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08:05.26 | robin_z | Washing? Outside? ... coo. How very 20th century ;) |
08:05.51 | Mavvie | robin_z: much cheaper than using a dryer. |
08:06.15 | robin_z | errr ... well, apart from the fact that its dripping wet, yes. |
08:06.34 | Mavvie | it's always dry in the afternoon. |
08:09.55 | robin_z | hmmm . this "group" thing is weird ... it seems to hava count 1 higher than I expect |
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08:11.31 | robin_z | shrug .. but at least it works, unlike call_limit |
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08:27.07 | jeebusroxors | no dice |
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08:44.49 | welles | my * send TXREJ and LAGRQ very fast, and my iaxclient all response VNAKs .then the call hangup .what's wrong? |
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09:03.30 | [hC] | hmm. i have mixmonitor recording calls, but it seems like quite a lot of the time, even though the cli seems to report proper mixmonitor start/stop, the recordings come out truncated, like half way thru a call |
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09:09.29 | pycsusz | Hi Everybody! |
09:09.35 | GBLRA | hi |
09:09.42 | X-Rob | Hi doctor nick! |
09:10.00 | pycsusz | Somedody cab help me, how can I provide sip with asterisk? |
09:10.08 | pycsusz | How can I configure? |
09:10.12 | X-Rob | oh fuck me. |
09:10.24 | X-Rob | pycsusz, you've got a pile to learn. start with http://www.voip-info.org |
09:10.53 | pycsusz | X-rob thanx, but there where? |
09:11.03 | [hC] | and maybe the oreilly asterisk book, but voip-info is a good right-away source. |
09:11.08 | X-Rob | search for sip.conf |
09:11.36 | X-Rob | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
09:11.58 | pycsusz | X-Rob I use Asterisk for one year, but still I didn't set sip trunk |
09:12.35 | pycsusz | X-Rob I would lik to create a SIP trunk |
09:12.40 | X-Rob | well look at /usr/src/asterisk/configs/sip.conf.sample |
09:13.08 | pycsusz | X-Rob Thanx, I will try it! |
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09:13.51 | saftsack | hi |
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09:16.14 | webmind | good morning (or your favorite part of the day), I've got a Fritz! isdn card, and trying to get asterisk to work with chan_capi, but it keeps bugging about "cc_init_capi: CAPI not installed, CAPI disabled!" and the fritz driver install page on voip-info seems to be for 2.4 (I'm running 2.6.15) |
09:16.23 | webmind | anyone like to help me out ? |
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09:19.21 | Kernel_core | hi all |
09:19.40 | Kernel_core | anybody compiled PWLIB 1.9.0 on Debian ?! |
09:19.57 | pycsusz | Kernel_core yes |
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09:20.24 | Kernel_core | pycsusz: which version of GCC do you use ? |
09:20.57 | pycsusz | Kernel_core 3.4 |
09:21.11 | pycsusz | Kernel_core it don't compile with 4 |
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09:46.08 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
09:46.36 | Winkie | hey guys, anyone got any experience with grandstream gxp-2000s? I need to know if it's possible to muck about with their LEDs |
09:47.34 | [av]bani | yep |
09:47.48 | [av]bani | blf is simple |
09:47.49 | Winkie | ah excellent, i need to have some sort of agent logged on indication |
09:47.58 | Winkie | blf? |
09:48.04 | X-Rob | Aaah. You need devstate |
09:48.05 | [av]bani | busy lamp field |
09:48.09 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
09:48.10 | X-Rob | which is part of bristuff |
09:48.25 | [av]bani | if you want agent tracking, thats something else... devstate may let you do that |
09:48.51 | Winkie | not tracking particularly, just want a simple agent logon procedure that will light up an LED on the phone |
09:49.03 | Winkie | X-Rob: is there any documentation on it? |
09:49.48 | [av]bani | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate |
09:50.09 | [av]bani | ~devstate |
09:50.10 | jbot | from memory, devstate is http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate |
09:50.59 | Winkie | i should learn to use ~ more often |
09:52.27 | Winkie | heh, bristuff is useless for this because we have a PRI it would seem |
09:52.31 | Winkie | still BLF might be enough |
09:53.03 | [av]bani | blf wont really do agent logon stuff like that |
09:54.38 | Winkie | hmm |
09:54.43 | Winkie | it's quite worrying that |
09:55.35 | [av]bani | blf without devstate is only good for tracking extension status |
09:55.51 | Winkie | yeah, from the look of bristuff i can install it but use just devstate without affecting the PRI? |
09:58.40 | pycsusz | X-Rob Can you help me sill a bit? |
10:00.10 | pycsusz | X-Rob I have got 2 subnetwork, and in both I have got telephones, but just that can log in whitch is on same subnetwork as asterisk. Why? |
10:00.32 | X-Rob | pycsusz, firewall or bad routing. DCHP wrong? Plenty of reasons. |
10:02.03 | pycsusz | X-Rob thanx |
10:03.14 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
10:05.28 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
10:10.58 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
10:19.55 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
10:21.56 | Dabba | any know how to replace this line with a line that will handle a varying number of rows so i dont have to repeat the fetch fifty times |
10:22.01 | Dabba | exten => s,3,MYSQL(Fetch fetchid ${resultid} myteam) |
10:22.30 | Dabba | exten => s,4,MYSQL(Fetch fetchid ${resultid} myteam1) |
10:22.31 | Dabba | etc |
10:22.33 | Dabba | etc |
10:24.48 | Dabba | i.e interpolate the variable name with a 1,2,3,4 etc |
10:25.13 | *** join/#asterisk sherif_ (n=sherif@212.103.170.135) |
10:26.13 | sherif_ | i have a question when a phone rings and no one is there to answer i want to ba able to grap the call to my set. |
10:26.59 | glm2k | sherif_: you're probably looking for pickupgroup and callgroup |
10:27.01 | Dabba | enable call pickup |
10:27.06 | Dabba | :-) |
10:27.07 | Mavvie | hmmm... which oart of zt_pvt lets me know if I'm in a call or not. |
10:27.24 | glm2k | Dabba: jinx! :) |
10:28.40 | glm2k | Dabba: i think you can use a loop for that |
10:29.08 | glm2k | but i'm not so sure about calling mysql in such a way |
10:29.44 | Dabba | i use it regular it works well, but is a pain with a row count that varies |
10:30.00 | Dabba | or at least its a pain at the moment |
10:30.07 | glm2k | hmm, would you be able to capture the row count ahead of time? |
10:31.42 | Dabba | oh i think i see where your going |
10:31.45 | glm2k | before you reach that part of the dialplan, do you already know how many rows to fetch? |
10:32.32 | FLeiXiuS | Asterisk is not allowing me to dial without receiving a busy tone...anyone have any ideas? Line's are up and configured. I'm running SCCP. |
10:32.36 | Dabba | no i dont i wonder if a GoSUb in a Gosub will kill it all |
10:33.07 | FLeiXiuS | I receive a dial tone but I'm unable to push more than 1 button without receiving the busy tone. |
10:33.14 | Dabba | > FLeiXius a console output ?? |
10:33.37 | FLeiXiuS | Dabba: I don't see anything... |
10:33.52 | Dabba | verboseity on ? |
10:34.45 | FLeiXiuS | Yep, nothing relevant. |
10:35.47 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
10:36.20 | [av]bani | fulgas: what model cisco? |
10:36.28 | FLeiXiuS | dabba, let me check out the debug, I was wondering about it. |
10:38.54 | FLeiXiuS | I still don't see any output which would restrict my dialing to only 1 number before it receives a busy tone. |
10:39.23 | Winkie | anyone got a clue whether pgsql support for databases of sip peers etc is in svn or planned at any poing? |
10:39.24 | Winkie | point* |
10:40.40 | FLeiXiuS | Dabba: Care to see the output when a button is pushed? |
10:40.54 | Dabba | sure pastebin it |
10:41.22 | FLeiXiuS | Dabba: http://pastebin.com/564164 |
10:42.10 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-121-114.telkomadsl.co.za) |
10:42.20 | X-Gen | hey freaks |
10:46.37 | Dabba | FLeiXiuS as you said looks ok, what are you doing with 9 in the dialplan, i havent ever used sccp so im blind really, my 7960 is running sip firmware |
10:47.16 | FLeiXiuS | Dabba: 9 does absolutely nothing...I'm just giving an example. After pushing one number on the keypad it instantly sends the busy tone. |
10:47.41 | RoyK | I wonder why they change function and variable names within the 1.2 "stable" track |
10:47.49 | Mavvie | eeks... app_trunkisavail.so is loaded before chan_zap.so is loaded, and thus the loader complains about it. |
10:48.26 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
10:48.49 | RoyK | zoa: hi |
10:48.52 | *** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt) |
10:48.59 | zoa | hey ho |
10:49.06 | _foxfire_ | hi |
10:49.11 | RoyK | zoa: tried the jb patch on 1.2.svn? they have changed function- and variablenames again :( |
10:49.26 | zoa | nopez, didnt try it yet |
10:49.30 | zoa | oh no, not again |
10:49.32 | zoa | :( |
10:49.39 | RoyK | fscking idiots |
10:49.49 | RoyK | it's supposed to be frozen, isn't it? |
10:49.58 | zoa | dont think so, no |
10:50.04 | RoyK | 1.2 is |
10:50.12 | zoa | ah yes |
10:50.16 | RoyK | they won't allow a single line of new code into it |
10:50.27 | RoyK | but changing names is ok, it seems |
10:51.49 | Mavvie | also wonder who killed my agi-bin directory! |
10:52.33 | puzzled | RoyK: where can I find that patch again? |
10:52.44 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
10:53.13 | RoyK | mantis |
10:53.16 | RoyK | i'll find the bug |
10:53.25 | RoyK | and i may as well upload a new 1.2.trunktoday patch |
10:53.43 | RoyK | #3854 |
10:54.54 | puzzled | RoyK: thanks. in case you are interested, I made 1.2.4 compatible patches for #2863 and #5374 |
10:55.41 | *** join/#asterisk Bambr (n=Bambr@213-35-237-173-dsl.end.estpak.ee) |
10:56.03 | _foxfire_ | i am upgrading an asterisk 1.0.9 to an asterisk 1.2.4 , all seemed well but i have one strange behavior, when i dial a user using "dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r)" and some channels do not exist like the IAX2 channel even if the user ANSWERS the cdr will always show "FAILED" , this didn't happen in 1.0 any idea why ? |
10:56.22 | puzzled | RoyK: which one in #3854 should I try on 1.2 svn? ast_jb-1.2.0.patch3? |
10:57.28 | RoyK | puzzled: the one i just uploaded :P |
10:57.40 | RoyK | 2006-02-20-svn-1.2-rev10558.patch |
10:58.17 | puzzled | got it. thanks |
10:58.33 | RoyK | puzzled: be a good friend and test it and test it and test it :) |
10:58.57 | puzzled | RoyK: will include it in my rpm and see how it works |
10:59.21 | puzzled | RoyK: any particular conf setting I should avoid/should use? |
10:59.38 | RoyK | just see the ones in sip.conf-sample |
10:59.44 | puzzled | ok thanks |
11:00.02 | RoyK | i don't know if adaptive jitterbuffer is as well tested as the fixed one, but the fixed one works very well |
11:00.05 | *** join/#asterisk WasPhantom (n=neil@203.86.197.11) |
11:00.23 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.133) |
11:00.27 | Kernel_core | hi all |
11:00.32 | RoyK | we tested on a 704/128kbps adsl link with some 20 concurrent downloads with g.711a and we didn't have a single hichup |
11:00.36 | Kernel_core | anybody compiled PWLIB 1.9.0 on Debian ?! |
11:00.40 | RoyK | latency from the depth of hell, though |
11:00.56 | zoa | royk, depends how much ms buffer you take :) |
11:00.56 | puzzled | RoyK: hehe. will try the static forst then |
11:01.04 | zoa | normally you have enough with 40ms or so |
11:01.15 | RoyK | the PLC really works well |
11:01.28 | Kernel_core | RoyK: I hate it too ... because after 2days playing hard with it , I couldn't comiple it on DEBIAN ! |
11:01.38 | puzzled | zoa: is that the jb-max-size option? |
11:01.42 | RoyK | we had the ATAs tuned up with larger JB and PLC turned on as well |
11:01.43 | zoa | yes |
11:01.47 | puzzled | ok |
11:01.53 | zoa | im off |
11:01.54 | zoa | food |
11:01.58 | puzzled | enjoy |
11:02.27 | welles | hi all,i want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while iaxc |
11:02.27 | welles | lient can not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated! |
11:02.35 | *** join/#asterisk GBLRA (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
11:03.12 | Mavvie | is there a way to force the export of a function in a channel-module? |
11:05.05 | welles | hi zoa ,what's your opinion about my issue? |
11:05.23 | ChrisUK | Anyone know when the new firmware for the Grandstream GXP2000 the v1.2 is getting released officially.? |
11:07.55 | Kernel_core | anybody compiled PWLIB 1.9.0 on Debian ?! :(( |
11:08.04 | GBLRA | have they actually fixed the bugs yet? |
11:08.29 | Kernel_core | or is there any Package for debian ? |
11:10.47 | _foxfire_ | i am upgrading an asterisk 1.0.9 to an asterisk 1.2.4 , all seemed well but i have one strange behavior, when i dial a user using "dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r)" and some channels do not exist like the IAX2 channel even if the user ANSWERS the cdr will always show "FAILED" , this didn't happen in 1.0 any idea why ? |
11:22.12 | kmilitzer | Hello everyone. Is it possible that there is a Bug in handling the hangupcause in ast_softhangup_nolock ? |
11:23.12 | Mavvie | of course there is. |
11:23.20 | *** join/#asterisk _foxfire_ (n=_foxfire@cica-adm.fe.up.pt) |
11:23.23 | Mavvie | (the possibility that) |
11:24.31 | kmilitzer | Mavvie: OK, i guess I put it the wrong way. I see strange behavior in passing the hangupcause back to the dial application i am doing some debugging right now to verify that |
11:25.09 | kmilitzer | I only have to figure out, if the problem lies within asterisk itself, or if it is a bug of chan_ss7 that i use |
11:25.31 | RoyK | cypromis: ping |
11:26.38 | *** join/#asterisk mac7 (n=karsten@c184067.adsl.hansenet.de) |
11:28.13 | remiss | any suggestions for fun things to do with the dialplan? |
11:28.55 | remiss | except rm'ing it... |
11:29.14 | iDunno | that's not a fun thing to do in the dialplan. |
11:29.44 | remiss | uh? |
11:30.54 | sherif_ | hum the defualt value for picking up the call is *8 ;-) |
11:31.25 | mac7 | I'm currently unable to register with iax.fwdnet.net! any other have this problem? |
11:32.26 | mac7 | connection to iax.fwdOUT.net is OK |
11:34.30 | *** join/#asterisk Mw3 (n=mw3@daisy.chains.ch) |
11:35.05 | sherif_ | what dose call parking means ? |
11:35.26 | iDunno | means that you can park the call |
11:35.31 | iDunno | think 'hold' but different. |
11:35.47 | iDunno | basically, when you park a call you should get given an extension number that it was parked on... |
11:36.00 | iDunno | if you then hang up and dial that extension, you'll pick the call back up |
11:36.09 | iDunno | (does that make sense?) |
11:36.48 | remiss | any way to transfer it instead of parking it? |
11:37.29 | remiss | blindxfer? |
11:38.03 | iDunno | push the transfer button and transfer to an extension that isn't call parking? ;) |
11:38.13 | remiss | yes, yes, but... |
11:44.09 | znoG | iDunno: but.. in which cases is it useful to do call parking? |
11:44.21 | remiss | #1 is transfer |
11:44.26 | remiss | no.. just sharp |
11:44.29 | znoG | apart from parking it then going to another extension and dialing it back up, i can't see much use for it |
11:45.33 | iDunno | znoG: when you're not sure who needs to take the call next? ;) |
11:45.45 | remiss | no, no, no |
11:45.49 | remiss | first you park the call |
11:46.19 | remiss | then you send mail to everyone with the extension and tell someone to pick it up |
11:46.19 | sherif_ | remiss: yes i can transfare calles |
11:46.19 | iDunno | so you phone round and find out who wants it then you tell them the extension that $person is on ;) |
11:46.20 | iDunno | remiss: *grin* - that's evil ;) |
11:46.20 | remiss | hehe |
11:46.40 | sherif_ | remiss: but the problem is i'm new in this PBX things but i know how to transfare it :) u click hold then ext number them transfer or read ur phone manuals ;-) |
11:46.49 | sherif_ | now i'll play with music on hold :D |
11:47.37 | FLeiXiuS | i wish I could play moh without a soundcard. |
11:47.42 | remiss | asterisk is kind of cool :) |
11:47.44 | FLeiXiuS | with mpg321 |
11:47.54 | remiss | FLeiXiuS: use something else? |
11:48.05 | FLeiXiuS | Ideas? |
11:48.08 | remiss | mplayer? |
11:48.24 | FLeiXiuS | will mplayer play .pls'? |
11:48.49 | remiss | .pls.. |
11:48.57 | remiss | what is that? |
11:49.04 | FLeiXiuS | shoutcast stream.. |
11:49.10 | remiss | oh |
11:49.13 | puzzled | I use madplay |
11:49.17 | FLeiXiuS | I'll check it out |
11:49.17 | znoG | iDunno: in which sort of asterisk setup would that be ... useful? |
11:49.22 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
11:49.23 | remiss | FLeiXiuS: i don't think so.. |
11:49.48 | iDunno | znoG: it's only 3 lines or so of dialplan and enabling it in features, it's probably useful to have anyway. |
11:50.06 | znoG | iDunno: yeah i know it's easy to setup, i'm just wondering what sort of business could make use of it |
11:50.10 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
11:50.31 | iDunno | large places that aren't sure where the call needs to go to, long term hold, etc etc. |
11:50.44 | znoG | i see the use when going from one office to another, for example |
11:51.28 | znoG | would be neat if the call park would ring certain extensions when they're free and connect them to the parked ext |
11:51.31 | znoG | but i guess thats what queues are for |
11:51.33 | iDunno | or if the person that you're going to transfer to is engaged, might be easier to park the call first ;) |
11:55.19 | trixter | Microsoft To Offer Free Wireless VoIP http://it.slashdot.org/it/06/02/20/0227200.shtml |
11:55.34 | kmilitzer | I found a bug in ast_softhangup_nolock |
11:56.41 | RoyK | trixter: they prolly just want your soul, but that'll be considered 'free' anyway, won't it? |
11:57.27 | trixter | they prolly are just integrating something into office, and its really going to be a wifi->office thing and not pstn |
11:58.10 | trixter | but I havent read the article so I am just guessing |
11:58.14 | kmilitzer | The Hangupcause is not set there, so it is not correctly transmitted to the other channel |
11:58.25 | kmilitzer | I inserted a chan->hangupcause = cause; and now it works ... |
11:58.49 | kmilitzer | What do I have to do to file this bug? Never did something like that before ... |
12:04.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
12:04.32 | kmilitzer | Helloooo anyone there that could answer my questions? |
12:05.25 | remiss | http://bugs.digium.com/main_page.php <-- probably |
12:07.39 | puzzled | kmilitzer: yes post the diff to kmilitzer or send an email to the asterisk-deve mailinglist or mention it in #asterisk-dev |
12:07.57 | puzzled | kmilitzer: post the diff to bugs.digium.com off course :) |
12:10.53 | mutilator | can you turn on/off threeway calling on a zap chan |
12:10.56 | mutilator | per channel |
12:11.03 | mutilator | or is that a global setting |
12:11.45 | puzzled | If not possible in a config file I guess you could limit it via Set(GROUP=... |
12:12.01 | puzzled | and check for the number of active calls and limit to one |
12:12.30 | mutilator | ya |
12:13.46 | remiss | anyone got a nice link to configuring musiconhold? |
12:14.09 | _foxfire_ | i am migrating from 1.0 to 1.2 , everyting worked ok with the exception of one anoying detail. if i dial Dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r) an for example IAX2/foxfire channel does not exist and SIP/foxfire accepts the call the cdr record will return "FAILED" instead of the expected "ANSWERED" , is this a bug or has some option changed ? |
12:15.01 | _foxfire_ | this does not happen with version 1.0 |
12:15.57 | *** part/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
12:20.07 | sherif_ | what is the bitrate for the hold on music ? |
12:20.22 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
12:22.25 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
12:24.51 | GBLRA | 8bit |
12:25.02 | kmilitzer | puzzled: Thanks, went to asterisk-dev now |
12:25.22 | RoyK | GBLRA: 8bit is not a bitrate :) |
12:25.27 | puzzled | kmilitzer: yup, seen it |
12:25.32 | RoyK | sherif_: 8khz |
12:26.26 | *** join/#asterisk Mw3 (n=mw3@national.t-error.hu) |
12:26.32 | *** join/#asterisk Bambr (n=Bambr@213-35-238-17-dsl.end.estpak.ee) |
12:26.38 | GBLRA | sorry |
12:26.47 | GBLRA | not thinking |
12:29.46 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
12:30.07 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
12:31.10 | l-fy | does anyone knows a library that analizes voice delays and jitters and things like that? |
12:31.44 | fjean | Hello everybody, anybody would have a suggestion on where to start looking for SIP phones that don't hear ringing tone on an IAX2 channel that is in progress ? |
12:31.45 | zoa | there are some |
12:31.49 | zoa | but they work on the network level |
12:31.55 | l-fy | hey zoa |
12:32.01 | zoa | meaning they look at the timestamps only |
12:32.08 | l-fy | oooo |
12:32.22 | l-fy | i need something that does convolution and corelation of the voice stream |
12:32.27 | l-fy | something not very fancy |
12:33.50 | fjean | it happens for SIP only, not for IAX2 devices... |
12:34.01 | l-fy | i want to send a voice to some voip equipment do some echo there and than get the voice back and compare |
12:36.12 | fjean | does anyone knows if IAX2 send ringing tone inband or outband ? |
12:37.20 | zoa | outband |
12:37.26 | fjean | ok |
12:37.30 | zoa | l-fy not that i know of |
12:37.36 | l-fy | ok zoa thank you |
12:37.44 | l-fy | zoa where are you living this days? |
12:37.52 | zoa | bulgaria still |
12:38.09 | l-fy | wow |
12:38.18 | l-fy | you haven't been in .nl? |
12:39.35 | zoa | im from .be |
12:39.40 | zoa | im going there tomorrow |
12:40.27 | *** join/#asterisk coppice (n=chatzill@19.206.17.210.dyn.pacific.net.hk) |
12:40.30 | fjean | zoa, do you think that my ATAs are listening inband for ringing tone and that would be why they hear nothing ? what are the chances ? :- ) |
12:40.48 | zoa | hmm, that doesnt matter |
12:40.51 | l-fy | sorry :( |
12:40.54 | zoa | as your ata will be sip |
12:40.55 | l-fy | i'm soooo stupid |
12:41.00 | l-fy | i've forgot |
12:45.38 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
12:47.15 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
12:48.04 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
12:56.02 | fjean | sip.conf: does progressinband=yes really work ?? |
12:56.23 | GBLRA | i found setting it to no stopped the external and internal ringing on my Aastras |
12:57.03 | fjean | ah, ok |
12:57.30 | fjean | you know any other settings that affect ringing for sip devices ? I can't manage to get one |
12:57.58 | GBLRA | what a ring? |
12:58.56 | fjean | mm? |
13:00.28 | fjean | yes, a ring |
13:00.38 | fjean | sorry.. |
13:00.41 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
13:01.53 | welles | zoa ,there? |
13:01.57 | zoa | yes |
13:02.03 | welles | see my question? |
13:02.09 | zoa | where ? |
13:02.28 | welles | ok i post again |
13:02.38 | welles | i want my iaxclient call an extension ,this extension locate asterisk server A. when server A receives call ,server A will transfer the call to an extension of asterisk server B. according to the http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers, this can be done easily.now i can call server B through server A. but the call will hangup several seconds. i grap the packets and found that server B can receive iaxclient 's packets while iaxclient c |
13:02.38 | welles | an not receive B's packets when my iaxclient using a private ip. if my iaxclient using a public ip ,they can work fine. what's wrong? any reply will be greate appreciated! |
13:03.04 | zoa | ah dunno |
13:03.09 | zoa | that seens to work fine for me |
13:03.31 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
13:03.37 | welles | it works fine when using public ip |
13:03.51 | welles | but when use private ip it can not work |
13:04.35 | welles | and myiaxclient receive too many TXREJ and send too many VNAK |
13:04.37 | fjean | you have one server on the same public IP as your softfone ? |
13:05.00 | welles | no |
13:05.50 | welles | softphone use private ip and two aterisk use different public ip |
13:06.48 | welles | if softphone use public ip. it works fine |
13:08.17 | mexuar-tim | welles, what is the 'cause' in the TXREJ ? |
13:09.01 | welles | if asterisk can not transfer successfully .why not use relay mode. the asterisk seem to try too many times to transfer |
13:09.17 | welles | mexuar-tim, let me have a check |
13:11.29 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
13:14.43 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
13:15.16 | _Paulo_ | Hi all... |
13:15.35 | welles | mexuar-tim, i dunno why it transfer fail.maybe my softphone behind a nat which aterisk can not access |
13:15.36 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
13:16.25 | welles | the debug info only show 'unable to transfre ...'' |
13:17.39 | mexuar-tim | Who ever sends the TXREJ must include a 'cause' field in the TXREJ packet. |
13:17.57 | welles | ok |
13:18.00 | mexuar-tim | I was hoping that might give a clue. |
13:18.04 | _Paulo_ | I got a TDM400P. I want to connect an external modem for faxing. |
13:18.29 | _Paulo_ | I heard that TDM400P is not reliable for fax. |
13:18.33 | _Paulo_ | Is that true? |
13:20.04 | _Paulo_ | I want to use the external modem because none of the "pure software" solutions (app_txfax and iaxmodem) are working. |
13:20.42 | *** part/#asterisk exonic (n=exonic@209.172.11.54) |
13:21.27 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
13:22.39 | welles | the cause code is 27 |
13:23.17 | holmeh | How would you guys go by shaping the two last bytes of a string? |
13:23.47 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.229) |
13:23.53 | Kernel_core | hi all |
13:23.58 | Kernel_core | anybody compiled PWLIB 1.9.0 on Debian ?! :(( |
13:24.24 | holmeh | err |
13:24.35 | mexuar-tim | welles, 27 means "Destination out of order ". Sorry that isn't much of a clue. |
13:24.36 | holmeh | I am in the wrong channel trolling =) |
13:26.38 | coppice | _Paulo_ if iaxmodem and spandsp don't work the external modem won't work either |
13:32.22 | _Paulo_ | coppice, Im receiving well with spandsp. Sending is the problem. |
13:33.10 | coppice | a few people complain about that, but nobody follows up well enough to find why it doesn't work for everyone |
13:34.40 | _Paulo_ | Im not sure the problem lies in spandsp, so I decided to try an oldfashioned modem. |
13:34.54 | welles | mexuar-tim, sorry. i check the packet carefully. the it show the data field of TXREJ is all zero. |
13:35.02 | coppice | that's just asking for more trouble |
13:35.07 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
13:35.51 | fugitivo | coppice: do you have any record of r2 working with asterisk connected to another pbx? |
13:36.27 | fugitivo | i mean asterisk --- R2 --- some pbx |
13:37.03 | coppice | fugitivo: lots of people do that. However, because there are lots of variants, and many implementations are buggy there is no guarantee any particular setup will work |
13:37.48 | welles | mexuar-tim, TXREJ 's subclass is 0x1c but ethreal shows that it is ox1b. what's wrong? |
13:37.53 | fjean | we do have it working here in brazil... |
13:38.04 | fugitivo | fjean: what pbx are you using? |
13:38.13 | fugitivo | i need to do that with a siemens hipath 3550 |
13:38.44 | fjean | sorry, i would have to ask the guys at the other office, but i know its there and working :- ) |
13:39.00 | coppice | fugitivo: it will depend on configuration and software versions, as well as the model of PBX |
13:39.40 | fugitivo | coppice: software versions? |
13:39.44 | *** join/#asterisk pengyong (n=lala@218.19.188.13) |
13:40.29 | coppice | of the software in the PBX |
13:40.41 | _Paulo_ | coppice, Why the analog modem is worst than the soft ones? |
13:40.55 | fugitivo | coppice: ugh, i can't control that |
13:41.25 | coppice | _Paulo_ because the existing analogue port options, like the TDM400, are mostly flaky |
13:42.11 | welles | any ideas about my issueïĵ |
13:42.16 | _Paulo_ | coppice, I heard something in that line... :-( |
13:43.08 | _Paulo_ | coppice, I has hoping they have somewath improved... |
13:43.09 | fugitivo | coppice: does the pbx need any special config or it's just luck with the software version? |
13:43.28 | *** join/#asterisk alerios (n=alerios@201.244.246.58) |
13:43.29 | _Paulo_ | fugitivo, you will have to try |
13:43.45 | mexuar-tim | i would have to see the ethereal packet dump to understand that. |
13:43.49 | _Paulo_ | fugitivo, I think its a model by model issue. |
13:45.23 | _Paulo_ | fugitivo, you will offer solutions for people who already have a PBX? |
13:45.28 | fugitivo | yes |
13:45.53 | _Paulo_ | fugitivo, If so, I think you canot claim to support every PBX out there. |
13:46.15 | fugitivo | i don't |
13:46.35 | fugitivo | i told the customer that it could work or not |
13:46.47 | _Paulo_ | fugitivo, After some time, you will have a few PBX validated. |
13:48.13 | fugitivo | do you have any? |
13:52.15 | _Paulo_ | I have none, you should ask the maillist |
13:54.13 | _Paulo_ | coppice, How can i help diagnosing the issue with app_txfax? |
13:54.52 | *** join/#asterisk nozey (n=nozey@20150042008.user.veloxzone.com.br) |
13:55.29 | nozey | hi ... im having problems with sip and nat. can someone help? |
13:58.47 | docelm0 | nozey, try stun |
13:59.03 | fugitivo | nozey: describe your scenario |
13:59.22 | _foxfire_ | i am migrating from 1.0 to 1.2 , everyting worked ok with the exception of one anoying detail. if i dial Dial(SIP/foxfire&SIP/3001&IAX2/foxfire,30,r) an for example IAX2/foxfire channel does not exist and SIP/foxfire accepts the call the cdr record will return "FAILED" instead of the expected "ANSWERED" , is this a bug or has some option changed ? |
13:59.26 | *** join/#asterisk zael (n=zael@20150034147.user.veloxzone.com.br) |
13:59.59 | nozey | i instaled asterisk here on my machine(thats behind a router) ... its running ... i can dial to my friends. If my friend is not behind a router(like me) he can hear me .... but i cant hear him |
14:00.13 | nozey | if hes behind a router too ... none of us can hear each other |
14:00.24 | nozey | sorry for my english ... im brazilian #) |
14:00.40 | fugitivo | nozey: try stun like docelm0 said |
14:00.47 | docelm0 | Its natting.. You need to put your asterisk box on the outside of your router. If its on the inside it makes life VERY hard |
14:00.59 | RoyK | hm. may sip/udp packet ever fragment? |
14:01.12 | nozey | well hes inside |
14:01.12 | RoyK | or will all consist of a single udp packet? |
14:01.16 | *** join/#asterisk Tuttle_ (n=Tuttle@kelinat210.keli.cz) |
14:01.31 | nozey | and unfortunally i need to make it work inside a nat |
14:01.36 | Tuttle_ | What is the favorite SW phone for Linux? |
14:01.41 | nozey | but let me try stun ... thanks for the help |
14:01.43 | docelm0 | RoyK, if there is an inconsistancy of a udp packet it will never frag.. it will just be dropped |
14:01.43 | zoa | idefisk! :p |
14:01.44 | _foxfire_ | nozey : are you using IAX2 or SIP ? |
14:01.48 | nozey | sip |
14:01.56 | RoyK | good... |
14:01.56 | zoa | (its mine so im not objective) |
14:02.15 | nozey | i need to use sip :( ... but i learned that iax is much simplier to tranverse nats |
14:02.18 | docelm0 | RoyK, one of the drawbacks to UDP.. They are never rebuilt and sent again like TCP |
14:02.21 | _foxfire_ | sip is the best but when nat comes in the middle .... ikes |
14:02.34 | nozey | i can see that |
14:02.37 | zael | hey guys, i'm with nozey in this problem... we need to set an asterisk server using sip through a nat |
14:02.52 | _foxfire_ | did u forward any ports in your router ? |
14:02.57 | nozey | 5060 |
14:03.03 | nozey | the rtp ports too |
14:03.05 | docelm0 | zael does your router have the option for DMZ and also is your IP static? |
14:03.32 | _foxfire_ | did u use the ports explained in the asterisk site ? |
14:03.40 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
14:03.47 | zael | i dunno, mine is a d-link 502g... i cannot access the router page through firefox |
14:03.58 | zael | is there any way to config this using telnet? |
14:04.05 | docelm0 | ya.. they suck.. You have to use IE |
14:04.14 | nozey | no way |
14:04.17 | zael | but how? :) |
14:04.23 | zael | i'm using slack |
14:04.26 | zael | lol |
14:04.27 | nozey | i can use dmz at my router |
14:04.42 | nozey | but i dont thinks thtas the problem, cause i already forwarded all the ports that sip uses |
14:04.49 | Tagor | I've the following problem with asterisk; I am using a sip provider for my calls. When I make an external call everything works fine. But when I try to call the asterisk server using a normal phone, then it just says; 'trying to connect'. When I try 'sip debug' on the CLI, I see nothing. Does anyone know how to find out what's going wrong? |
14:04.52 | fugitivo | last firmware of dlink routers works with ie only :) |
14:05.00 | nozey | zael, is the one that im dialing to |
14:05.16 | docelm0 | But you can set a DMZ.. What I did at my house was to static assign my internal IP to something like 200.. Then DMZ 200 and make asterisk forward on the public IP as its own.. |
14:05.31 | docelm0 | but it has to be static.. dynamic will work for the most part till it changes |
14:05.42 | zael | my internal IP is static |
14:05.50 | docelm0 | nozey you must not understand how SIP works |
14:06.01 | zael | but i can't change these DMZ settings in the router |
14:06.04 | docelm0 | zael, what bout external? |
14:06.10 | zael | external is dynamic |
14:06.14 | docelm0 | Good luck |
14:06.16 | nozey | mine too |
14:06.16 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:06.21 | nozey | docelm0, what do u mean? |
14:06.26 | _foxfire_ | nozey try http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules for more help |
14:06.38 | Tagor | Nobody? :( |
14:06.49 | docelm0 | Tagor, wait your turn or leave.. |
14:07.13 | nozey | i made this already _foxfire_ |
14:07.17 | nozey | im logging that port too |
14:07.49 | _foxfire_ | tagor : probable you don't have an extension assosiated with th eincoming number |
14:07.59 | nozey | i can see the connections ... but no audio ... i think audio uses the rtp ports described in rtp.conf, or am i worng? |
14:08.05 | docelm0 | nozey, zael, When a sip packet comes in it works cause of the invite being setup on the originating location. When you send out from the nat it broadcasts the private IP block of information which is unroutable. This is bad.. You need to tell it to use the public IP over the private |
14:08.25 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
14:08.47 | zael | yeah, we know this |
14:08.48 | Tagor | _foxfire_ >> The extension in extensions.conf is the same as the extension name in sip.conf and the context in sip.conf |
14:08.53 | zael | but we dont know HOW to do this |
14:08.54 | nozey | thats the problem |
14:09.17 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
14:09.18 | _foxfire_ | nnozey youi can limit the range of the rtp ports on asterisk and on all major software and hardware clients |
14:09.19 | SplasPood | it'd be nice if all the sipbroker sip providers would list their DIDs in the ENUM db |
14:09.41 | docelm0 | So when your RTP tries to sync up it tries to use the private.. Either A.. Use stun which asterisk to my knowledge doesnt fully support.. OR manually push your public IP from the asterisk box |
14:10.12 | _foxfire_ | tagor maybe i didn't understand your problem right can you be a bit more detailed |
14:10.16 | nozey | let me try stun |
14:10.20 | docelm0 | _foxfire_, thats all fine and dandy.. but how are they gonna change the information in the SIP Header Packet? |
14:10.28 | SplasPood | Or if I could otherwise get a list of all their assigned DIDs somewhere |
14:10.35 | nozey | thanks for the help guys ... we are going to try it ... |
14:10.39 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
14:11.29 | _foxfire_ | docelm : iup thats a prob ... , another one is if he has an dynamic ip |
14:11.47 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
14:12.01 | docelm0 | Good lord what have I been saying |
14:12.08 | docelm0 | They both have dynamic.. |
14:12.09 | docelm0 | geesh |
14:12.19 | Tagor | _foxfire_ >> I use a sip provider to handle outgoing and incomming calls. When I call internally everything works fine. Also I can make calls to external phone numbers. But when an external phone tries to call the asterisk server then nothing happens. The phone tries to connect but after a few times it says that the network is busy. When I try 'sip debug' to see if there is any signal from the voip provider, I see nothing |
14:13.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:13.44 | _foxfire_ | Tagor hmm did you tried using tcpdump to see if any connections are conecting to the server ? |
14:13.59 | docelm0 | Tagor, are you natted also or public ip? |
14:14.03 | _foxfire_ | Tagor also did you "register" on your remote server ? |
14:14.45 | synthetiq | why would message fowarding work only half the time |
14:15.36 | Tagor | _foxfire_ >> No, let me try that. Yes, it registers in sip.conf. Though I am not sure if it is possible to put the register line in front of the extension |
14:15.46 | *** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com) |
14:16.03 | Tagor | docelm0 >> natted, but all routers have static ip's and use DMZ. If I try x-lite it's no problem to get incoming calls |
14:17.14 | Tagor | _foxfire_ >> Is there a specific port orso which I can monitor with tcpdump? Else I get too much output |
14:18.43 | _foxfire_ | hmm choose only udp trafic |
14:19.01 | _foxfire_ | yoiu can also give the src host if you know its IP |
14:19.27 | docelm0 | Where does Xlite sit in reference to your * box? |
14:19.34 | docelm0 | same side of router or opposite sides |
14:20.03 | docelm0 | Xlite also supports Stun |
14:20.11 | Tagor | _foxfire_ >> If I try 'tcpdump udp' and call asterisk then it does nothing |
14:20.34 | Tagor | docelm0 >> I 'replaced' the server with the xlite computer |
14:21.00 | docelm0 | You have a config issue on your ast box. |
14:21.20 | _foxfire_ | hmm hold on tagor use the src host then |
14:21.27 | docelm0 | thats all it can be then but as I said.. Xten does use STUN as asterisk doesnt |
14:21.44 | *** join/#asterisk _deg_ (n=deg@200.150.147.29) |
14:22.47 | _foxfire_ | tagor tcpdump sees it as IP |
14:23.40 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
14:23.59 | Tagor | _foxfire_ >> I tried 'tcpdump src host ip.address.of.sip-provider' and this also doesn't output anything |
14:24.29 | Tagor | docelm0 >> I understand that there is some setting wrong. But I can't find out which one since 'sip debug' doesn't show anything |
14:25.11 | Tagor | docelm0 >> Note it has worked before with the current router settings. So I don't think it's a nat problem |
14:25.23 | docelm0 | I dont know where to even tell you begin.. do a sip show settings and make sure its bound to all the correct IP's |
14:25.25 | *** join/#asterisk macaco (n=tiago@joltid-gw.joltid.org) |
14:25.30 | macaco | hey |
14:25.37 | fjean | hello, anybody uses a grandstream 286 here ? |
14:25.42 | *** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net) |
14:25.47 | docelm0 | Good lord no |
14:25.53 | fjean | :- ) |
14:26.00 | macaco | has anyone been using the svn asterisk? there's a huge memory leak when establishing zap/sip channels |
14:26.07 | Tagor | docelm0 >> As far as I can see the ip's are correct. Else I also wouldn't be able to make calls |
14:26.15 | macaco | 132 bytes to be precise, corresponding to the pool string allocation |
14:26.18 | macaco | anyone noticed this? |
14:26.23 | _foxfire_ | fjean i use an grandstream gxp2000 and likes it |
14:26.26 | docelm0 | macaco, check the bugs |
14:26.38 | *** join/#asterisk coppice (n=chatzill@221.162.17.210.dyn.pacific.net.hk) |
14:26.47 | docelm0 | _foxfire_, I have 75 GXP2000's deployed in 3 countries |
14:26.59 | fjean | foxfire: you had no ring tone with it at some point while you were using it ? |
14:27.37 | docelm0 | Tagor, thats right.. forgot your not using your linux box as a route |
14:27.39 | docelm0 | router |
14:27.40 | _foxfire_ | nope i tftpd a few ring tones of my own , but never had problems with it |
14:27.46 | docelm0 | which is my setup |
14:28.11 | sherif_ | where can i get hold on music or asterisk ? |
14:29.26 | macaco | docelmo: there's no bug about it |
14:29.34 | docelm0 | macaco, submit one |
14:29.36 | xachen | Freeplaymsic.com sherif_ |
14:29.42 | docelm0 | sherif_, any MP3 works |
14:29.52 | macaco | ok... i'd just like to know if someone had already noticed that :) |
14:29.56 | xachen | yeah but Freeplaymusic.com is royalty free :) |
14:30.26 | docelm0 | macaco not me.. then again I use ZAP/SIP @ my office only |
14:30.44 | sherif_ | docelm0: no not all is working :-s i put some MP3 which is not working.; |
14:30.47 | docelm0 | macaco, what did you use to find the leak? |
14:30.50 | sherif_ | i need like 8khz music |
14:31.01 | xachen | it'll transcode it |
14:31.03 | xachen | to 8khz |
14:31.15 | docelm0 | Im using full blown MP3's and they work fine |
14:31.17 | xachen | I usually take mp3s and convert them to raw format though. saves lots of cpu |
14:31.24 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
14:31.49 | xachen | docelm0: yeah, that works but its way easier to just transcode it to raw first and watch your CPU loads drop so they can be used for other important tasks |
14:32.15 | macaco | docelm0: i just connect a sip client and terminate asterisk |
14:32.44 | Tagor | _foxfire_ >> Any other suggestion? |
14:32.52 | docelm0 | macaco, I feel ya.. |
14:32.58 | docelm0 | Tagor, Call Digum |
14:33.03 | docelm0 | err Digium |
14:33.19 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:33.19 | *** mode/#asterisk [+o anthm] by ChanServ |
14:33.20 | macaco | docelm0: my guess is that the pool isn't freed when the channel is freed... |
14:33.38 | _foxfire_ | tagor what does and "sip show peers" show by the way ? |
14:33.39 | docelm0 | macaco what are you using to find this leak? |
14:34.11 | macaco | docelm0: ah sorry :) I'm using valgrind |
14:34.21 | sherif_ | xachen: can i see ur onholdmusic.conf ? |
14:34.30 | xachen | sure :p |
14:34.56 | xachen | [default] |
14:34.56 | xachen | mode=files |
14:34.56 | xachen | directory=/home/telecom/root/var/lib/asterisk/moh |
14:34.58 | xachen | thats it |
14:35.48 | Tagor | _foxfire_ >> It shows: 12connect/**MY-PHONE-NUMBER** **IP-OF-THE-PROVIDER** N 5060 Unmonitored |
14:36.24 | Tagor | docelm0 >> Why should I call them? |
14:36.30 | sherif_ | xachen: where the command to run it :D? |
14:36.30 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
14:37.04 | xachen | erm |
14:37.05 | xachen | there is none |
14:37.08 | xachen | its * 1.2 |
14:37.12 | xachen | it has its own builtin mp3 player |
14:37.39 | _foxfire_ | tagor and "sip show registry" ? |
14:37.58 | docelm0 | Tagor, your asked for more help |
14:37.59 | Tagor | _foxfire_ >> All registered |
14:38.09 | fjean | --- hi anyone would know how to force a ring tone on * for SIP devices ? |
14:38.41 | sherif_ | xachen: mine is Asterisk CVS-v1-0-03/29/05-08:24:40 built by root@localhost on a i686 running Linux |
14:38.47 | xachen | erm |
14:38.52 | xachen | Update :) |
14:38.58 | xachen | thats real old |
14:39.05 | sherif_ | xachen: seems too ;-) i don't why why i have this one already !! |
14:39.05 | xachen | your going to have nothing but problems with that version |
14:39.20 | sherif_ | i'm not the guy who installed it. |
14:40.03 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
14:40.25 | GBLRA | dont you have to pay freeplay.com for usage? |
14:40.40 | _foxfire_ | Tagor it's not easy to debug like this , are you shure you put the correct local extension ate the end of the register line |
14:41.16 | Sebb | hi.. is it possible that dial gives a busy, if any of the called channels ("ZAP/g1/11&ZAP/g1/12&SIP/13") is busy? that would make sense if there is only one person and some phones.. ;) |
14:41.32 | *** join/#asterisk lthnnpwr (n=sdf9sfo3@195.166.60.12) |
14:41.53 | Tagor | _foxfire_ >> The register line is just like this: register => **MY-PHONE-NUMBER**@**IP-OF-THE-SIP-PROVIDER** |
14:42.10 | lthnnpwr | hi, may I ask you a question guys? you might know the answer, hopefully |
14:42.26 | _foxfire_ | hmm my sip register lines are a bit more comples |
14:42.38 | Tagor | _foxfire_ >> Or should I use something like this: register => **MY-PHONE-NUMBER**@**IP-OF-THE-SIP-PROVIDER**/contextname |
14:42.47 | Tagor | _foxfire_ >> Can you give me an example? |
14:43.03 | _foxfire_ | ;register => xxxxx:yyyyy@proxy01.sipphone.com/1001 |
14:43.14 | _foxfire_ | where 1001 is my local extension |
14:43.26 | _foxfire_ | at least thios worked for me |
14:44.03 | _foxfire_ | Sebb i am having a similar problem |
14:44.54 | Sebb | _foxfire_: perhaps that is possible with chanisavail, but i'm not sure if that would work with zap-channels (zaphfc), too |
14:45.05 | _foxfire_ | Sebb i did an upgrade from asterisk 1.0 to 1.2.4 |
14:46.21 | lthnnpwr | Hi. I am trying to install pwlib152 - the updated one, and the compilation fails with theses errors: http://pastebin.com/564375 |
14:46.39 | _foxfire_ | when i try to call SIP/foxfire&SIP/3001&IAX2/foxfire and one channel does exist the CDR always gives me FAILED even if it is answered . |
14:47.08 | _foxfire_ | Sebb does this happen to you too ? |
14:47.23 | Tagor | _foxfire_ >> Just tried that too, but I don't get any incoming call :( |
14:47.26 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:47.27 | fugitivo | _foxfire_: why don't you create an extension and call that extension using Local/xxx |
14:48.08 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
14:48.25 | Sebb | _foxfire_: no. i don't have problems with unexpected behaviour, i just want to extend functionality.. so, if i already use one phone, other people who try to call my get a busy, because i can't use two phones at the same time |
14:48.48 | _foxfire_ | fugitivo , hmm i wanted to ring several phones simultaneosly |
14:49.14 | fjean | can we setup call progress indication with ztdummy ? |
14:49.24 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.201) |
14:49.25 | fugitivo | _foxfire_: create the extension with multiple devices, then call that extension using Local, don't call the extension directly |
14:49.29 | Kernel_core | hi there ... |
14:49.51 | Kernel_core | is ASterisk 1.2.4 Debian Package availabe? |
14:50.13 | _foxfire_ | Sebb my useres have 2 logins ext and login and i want to ring both at the same time |
14:50.13 | razu | is it normal that RDNIS info doesn't go forward in IAX trunk ? |
14:50.47 | _foxfire_ | Sebb and it works but the CDR record is incorect |
14:50.51 | lthnnpwr | has anyone else hot any problems compiling pwlib152? |
14:50.59 | Sebb | _foxfire_: sorry, i don't know.. |
14:51.16 | fugitivo | _foxfire_: why don't you try what i told you? |
14:51.40 | _foxfire_ | fugiutivo i will can you give me an example |
14:52.38 | _foxfire_ | Tagor sorry i am running out of ideas |
14:52.53 | _foxfire_ | did you try this with an other provider |
14:53.07 | _foxfire_ | like free world diallup for example |
14:53.22 | ketanp | Tagor: what provider are you using? |
14:53.25 | _foxfire_ | maybe it's not you problem |
14:53.26 | Tagor | No haven't tried that yet, _foxfire_. Thanks for your help |
14:53.35 | Tagor | ketanp >> 12connect (astate) |
14:53.49 | iCEBrkr | yo yo yo |
14:54.33 | ketanp | Tagor: oh ok, sorry can't help then... try it with a free provider and see what happens... otherwise, go get some coffee and try again later, it may become obvious then |
14:58.36 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
14:59.02 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
15:01.32 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
15:01.34 | ramtha | peace |
15:01.58 | ramtha | hmm asterisk overlapdial=yes is not recognized by asterisk 1.2.4 |
15:02.05 | ramtha | why is that? |
15:02.06 | ramtha | any hint |
15:02.08 | ramtha | ? |
15:03.05 | tzafrir | ramtha, no idea here, but it would help if you mentioned where it has last worked for you |
15:03.25 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
15:04.04 | mutilator | or hell even the imail imapd chokes on that much |
15:04.06 | MRH2 | hi anyone forsee a problem using both monitor and mixmonitor in the dialplan to have 2 recording files |
15:04.12 | mutilator | wtf |
15:04.13 | mutilator | anyone know if linksys wrt54g's can do WARP? |
15:05.35 | ramtha | hm no one uses overlapdial funktion? |
15:05.37 | MRH2 | like: 4,MixMonitor(/tmp/MM/${CALLFILE}.g729|bv(1)V(1)) ... 5,Monitor(gsm,${CALLFILE},bm) |
15:06.54 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
15:07.30 | nextime | ramtha : i use overlapdial on a PRI |
15:07.58 | lthnnpwr | any idea why the pwlib152 update fails to compile? anyone encountered any problems with that? |
15:08.09 | ramtha | nextime: ok, mei pstn switch dials only the firs 4 digits and pushes the call |
15:08.27 | ramtha | reload from asterisk says me: setup_zap: Ignoring overlapdial |
15:08.42 | ramtha | does this option go away? |
15:08.58 | ramtha | i have a wildcard in the mashine |
15:10.15 | *** join/#asterisk pnviking (n=pnviking@c83-248-7-150.bredband.comhem.se) |
15:13.32 | lthnnpwr | aw. has ANYone encountered any problems while upgrading to pwlib152? |
15:13.34 | *** join/#asterisk pycsusz (n=infocare@pluto.euronetrt.hu) |
15:13.41 | pycsusz | hi Everybody! |
15:13.54 | pycsusz | I have got a question |
15:14.14 | pycsusz | Feb 20 16:11:37 NOTICE[2115]: chan_sip.c:10915 handle_request_register: Registration from '5004<sip:5004@80.95.69.180>' failed for '80.95.69.180' - Not a local SIP domain |
15:14.30 | pycsusz | somebody can tell me something for this? |
15:19.29 | *** join/#asterisk lodeon (n=not4u@h119n5c1o1023.bredband.skanova.com) |
15:21.42 | *** join/#asterisk peanuter (n=saasdf@216.176.177.138) |
15:23.31 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
15:25.01 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
15:28.43 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
15:32.49 | *** join/#asterisk coppice (n=chatzill@189.201.17.210.dyn.pacific.net.hk) |
15:34.33 | *** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com) |
15:34.57 | Seyr | Anyone know how to set the time on Polycom phones? The servers time is NOT what is displayed on it |
15:35.57 | ChrisUK | need a NTP service running on your server |
15:37.04 | Seyr | besides that :-) |
15:37.10 | Seyr | I dont *have* to have one |
15:38.13 | Seyr | I was looking more for: "Seyr: The Polycom phones get their time setting from DHCP, so make sure your DHCP server is dishing out the correct NTP server and time offset" |
15:38.55 | _Sam-- | if you know so much, what is the point of asking a question |
15:39.24 | Seyr | because I just stumbled across the web page with that info, thank you very much |
15:41.15 | [TK]D-Fender | Seyr : What is providing DHCP for your network? |
15:43.19 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
15:43.40 | *** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net) |
15:43.46 | paulhuynh | please help me asap |
15:43.55 | paulhuynh | i have major problem this morning |
15:44.21 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc04v.dialup.mindspring.com) |
15:44.32 | paulhuynh | i did a asterisk upgrade overnight and this morning non of the device is registering to the server |
15:44.47 | paulhuynh | server seem to be working |
15:44.48 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc04v.dialup.mindspring.com) |
15:45.00 | Hmmhesays | [TK]D-Fender: how goes it? |
15:46.23 | paulhuynh | please help me |
15:46.23 | paulhuynh | anyone here |
15:46.23 | [TK]D-Fender | Hmmhesays : Getting by. recovering from a cold, but my * consulting projects are really picking up and I'm getting a job offer because of it from a digital based co going VoIP |
15:46.40 | Hmmhesays | cool, cool |
15:46.44 | [TK]D-Fender | definately.. |
15:46.51 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
15:46.52 | Hmmhesays | I tanked the intro to "my own worst enemy" in front of 300 people last night |
15:46.58 | Hmmhesays | :D |
15:47.01 | file | every time you need me you know I will be there |
15:47.04 | file | you know I really care |
15:47.07 | file | say you will, say you dare |
15:47.17 | file | hot'n'sexy Hmmhesays!!! |
15:47.21 | Hmmhesays | hey file |
15:47.24 | file | hiya |
15:47.26 | [TK]D-Fender | GF situation is flakey but I'm attracting some attention and who knows what may develop so life is "improving" slowly. Gotta go Appt hunting soon though... |
15:47.36 | paulhuynh | please can anyone help me |
15:47.48 | paulhuynh | my offic is completely dead in the water |
15:47.51 | *** join/#asterisk danzig (n=chatzill@130.226.173.92) |
15:47.51 | paulhuynh | no phone service |
15:47.53 | [TK]D-Fender | Hmmhesays : "Tanked" =good? |
15:48.07 | danzig | EHLO * gurus :-) |
15:48.21 | Seyr | [TK]D-Fender: Not sure. It is at a customers site in another state and they have a seperate network team that works on their network |
15:48.23 | Hmmhesays | i went up and played a telecaster, never done it in my life, i completely bombed the 1st bar |
15:48.27 | file | paulhuynh: that's why you keep backups... and revert if it fails |
15:48.31 | Hmmhesays | as in screwed it up bad |
15:48.36 | nozey | can someone help me with stun? |
15:48.38 | [TK]D-Fender | Seyr : are you provisioning the phones? |
15:48.39 | Seyr | [TK]D-Fender: I am trying to figure out if I can set the NTP servers via a config file, so I can do it remote |
15:48.41 | file | Hmmhesays: did they notice? |
15:48.53 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
15:48.55 | NovceGuru | paulhuynh, state your problem |
15:48.57 | Hmmhesays | hahah hell yeah they did i stopped smiled and said "lets try that again" |
15:49.00 | Seyr | [TK]D-Fender: All I have found so far is setting the ntp via dhcp |
15:49.01 | [TK]D-Fender | Hmmhesays : OH... |
15:49.07 | paulhuynh | nothing is register to the * |
15:49.08 | file | Hmmhesays: ah yes |
15:49.27 | Hmmhesays | then I got more people on the dance floor than any of the other jammers that night |
15:49.28 | file | paulhuynh: are you using trunk? if so, don't. it can be broken at any given time |
15:49.30 | Hmmhesays | with pictures |
15:49.31 | [TK]D-Fender | Seyr : You can have the phones pick it up through DHCP or with SIP 1.6.3 (maybe .4 or .5) you can have it in the provisioning files. |
15:49.51 | Seyr | [TK]D-Fender: You know a link that shows all the config file params? |
15:50.16 | paulhuynh | no i use trunk just for sip acrrier and ext for the user |
15:50.16 | [TK]D-Fender | Seyr : its in the changelog. |
15:50.29 | paulhuynh | my * was from asterisk@home |
15:50.47 | paulhuynh | and i did a * to 1.2.4 from 1.2.1 |
15:50.52 | Seyr | [TK]D-Fender: kk, thanks :-) thats all I needed |
15:51.14 | NovceGuru | My first experience was with asterisk@home, it wasn't a good one |
15:51.17 | paulhuynh | system status is showing everything is working |
15:51.24 | Hmmhesays | ahh the "if its not broke, don't fix it" theory comes into play |
15:51.32 | paulhuynh | but no deviced or network can register to my * |
15:51.45 | [TK]D-Fender | file: !!! |
15:51.50 | file | [TK]D-Fender: !!! |
15:51.55 | NovceGuru | pastebin your sip.conf (remove passwds) |
15:52.00 | Seyr | Seyr: !!! |
15:52.16 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@71-36-209-237.dlth.qwest.net) |
15:52.21 | Seyr | hey file, invoice is HTML, can I view it online? |
15:52.33 | file | Seyr: can you? dunno, but I can |
15:52.34 | [TK]D-Fender | * has been kind to me. Last friday's complete overhaul was virtually perfect, and I'm getting consulting jobs coming out all over the place now... |
15:52.48 | Seyr | file: if you can view it, go ahead and pay for it as well :-) |
15:52.53 | [TK]D-Fender | And now a job offer... |
15:52.54 | file | Seyr: I can have it resent as well |
15:53.11 | paulhuynh | tk d-fender |
15:53.19 | paulhuynh | can you please help me |
15:53.23 | Seyr | file: naw, it auto-debits, i was just curious how much this new client ran |
15:53.29 | file | Seyr: ah I can go look |
15:53.33 | file | Seyr: which account? |
15:53.38 | Hmmhesays | I could use some more asterisk jobs, mine are running a bit dry |
15:53.40 | Seyr | hrm.. no clue |
15:53.57 | Seyr | file: i'll check and get back with you in #asterlink later |
15:54.02 | file | Seyr: great |
15:54.04 | danzig | If I download asterisk 1.2 and build from source, will conferencing work straight off? Last time I looked one needed to install zaptel and make a fake loopback adaptor... Is this still the case? |
15:54.15 | generalhan | hey guys can i get a little help ?? My sip phones are unregistering ALL THE TIME. sometimes they will stay registered, once i restart the phone, for an hour, sometimes an entire day. this all happened after i switched over from a VoIP provider to a PRI, and upgraded to 1.2.1 ... this is my sip.conf and a show peers if anyone could take a look at it and give me some suggestions :: http://generalhan.pastebin.ca/42213 |
15:54.21 | Seyr | danzig: yeh |
15:54.48 | danzig | seyr: yeh it will work or yeh I need zaptel? :-) |
15:54.52 | Seyr | danzig: just read the README in zaptel dir and make sure you do all the needed stuff for ztdummy |
15:55.09 | danzig | ok, thanx. |
15:55.21 | Seyr | danzig: its flawless if you follow the instructions :-) |
15:55.28 | file | zaptel pseudo channels are cool... |
15:55.29 | shido6 | got any zaptel interfaces or usb interfaces , generalhan ? |
15:55.50 | generalhan | <PROTECTED> |
15:56.18 | paulhuynh | can someone please tell me how can i restore it from backup? |
15:56.30 | generalhan | all my phones are sip phones directly into the network that the asterisk server is on. the PRI line comes into that server and then the nic connects it to the rest of the phones on the network |
15:57.32 | generalhan | when the phone registers i get this " -- Registered SIP '7106' at 192.168.0.106 port 5060 expires 120" is that 120 minutes ? or what? maybe i have to set that way higher some how ?? |
15:57.39 | file | 120 seconds |
15:57.42 | *** join/#asterisk buutymalapico (n=jorgito@82.113.32.241) |
15:57.43 | buutymalapico | hi |
15:57.44 | generalhan | haha |
15:57.54 | generalhan | so that would explain it huh !! |
15:58.07 | generalhan | what conf file can i set that to something else ? |
15:58.28 | file | it's on 'da phone |
15:58.33 | generalhan | hmm |
15:58.33 | buutymalapico | have problem with festival, i can hear just half of sentence .. |
15:59.19 | paulhuynh | anyone here can help please |
15:59.40 | Seyr | [TK]D-Fender: congrats on the job offer! |
15:59.41 | paulhuynh | if $$ is involed please PM me |
16:01.11 | paulhuynh | HELP ME |
16:02.38 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
16:02.39 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
16:03.32 | wunderkin | lol, paulhuynh, did you make any config file changes after the upgrade? were you running a release? just reinstall the old version if you dont have a backup and deal with it later |
16:03.32 | Hmmhesays | she was on google talk earlier |
16:06.44 | *** join/#asterisk lalito (n=erg@201.137.152.125) |
16:09.17 | Tagor | Which port does asterisk by default use for incoming calls? |
16:10.03 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
16:10.33 | lunaphyte | hello |
16:12.17 | docelm0 | Depends on technology |
16:12.31 | fugitivo | Tagor: port for what? |
16:12.37 | fugitivo | Tagor: sip? |
16:12.46 | Tagor | Yeah |
16:13.02 | fugitivo | 5060 udp, 10k-20k udp |
16:13.04 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
16:13.26 | buutymalapico | somebody using festival here ? |
16:13.38 | *** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net) |
16:13.43 | nozey | hey fubster i changed the rtp ports to 10000-10010 ... is this a problem? |
16:14.16 | nozey | fugitivo ... not fubster ... sorry #) |
16:14.21 | iCEBrkr | ha |
16:14.55 | *** join/#asterisk Crontibs (n=frank@office-nat.choopa.net) |
16:14.59 | fugitivo | nozey: if you have a lot of sip connections it will, 10k-20k are for rtp packets |
16:16.19 | lunaphyte | is it true that i need a certain firmware image file to get an older 12sp+ to work with asterisk? |
16:19.04 | buutymalapico | somebody using festival here ? |
16:19.36 | puzzled | no I like my ears |
16:19.48 | fugitivo | buutymalapico: www.cepstral.com |
16:20.15 | *** join/#asterisk robb_ (n=robb@kapow.vm.bytemark.co.uk) |
16:20.33 | buutymalapico | fugitivo, well i need help about festival.. |
16:21.04 | paulhuynh | so no taker on my problem |
16:22.24 | puzzled | let's see if I can find a taker for my mysterious problem... |
16:22.27 | puzzled | HELP |
16:22.38 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F41EA.dip0.t-ipconnect.de) |
16:22.41 | buutymalapico | puzzled, what s the problem _ |
16:22.54 | puzzled | buutymalapico: that was not supposed to work :) |
16:22.58 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
16:23.03 | puzzled | and I don't have a problem either actually |
16:23.23 | zoa | you rang my lord ? |
16:23.29 | _foxfire_ | fugitivo : i tried what you suguested and dial a local extension , i receive a valid CDR now , but also the invalid CDR so i end up with 2 . |
16:23.37 | puzzled | zoa: hehe |
16:28.35 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
16:29.22 | fugitivo | _foxfire_: hmmm |
16:29.37 | fugitivo | _foxfire_: isn't that better than only only invalid record? :) |
16:29.57 | danzig | Can anyone explain the logic of when CDR started/ended/written? Im messing with ForkCDR and ResetCDR(w), and niether do exaclty what I would expect... I need to do things like user comes in, gets dialtone, dials new number [start new CDR] |
16:33.09 | danzig | I'm getting error "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line 723" - but my sip.conf has 100 included files. Any clever tricks to figure out which file the error is in? |
16:33.47 | fugitivo | 100 included files??? |
16:33.54 | fugitivo | god |
16:33.58 | fugitivo | why did you do that? |
16:34.04 | peanuter | i live in ny looking to play with iax and asterisk. any suggestions as to where to get cheap service? |
16:34.20 | danzig | 100 users :-) They can only muck up their own config... |
16:34.40 | fugitivo | there're better ways to do that |
16:34.46 | fugitivo | 100 include files is impossible to admin |
16:34.53 | danzig | fug: how? :-) |
16:35.04 | fugitivo | danzig: a web page and a database? |
16:36.10 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
16:36.11 | danzig | They do it themselves with a web interface. Works fine. Until now, there must be an error in our web interface that let someone wirte something wrong... Could put it in a DB, but would then have to chuck it into files anyway... |
16:36.24 | iCEBrkr | 100 includes?!?! |
16:36.26 | iCEBrkr | WTF |
16:36.40 | fugitivo | danzig: into a single file |
16:36.54 | fugitivo | danzig: you don't need one include file for each user |
16:37.28 | _foxfire_ | fugitivo : yeah , but it messes up my scripts, ;-) i will stay with 1.0.9 for now . |
16:38.03 | danzig | well, actually you just do #include "sip-conf.d/trunk-*.conf" |
16:39.39 | Crontibs | wow |
16:39.45 | Crontibs | thats pretty intense dan |
16:40.27 | danzig | it solves a concurrency problems - 1 user=1file=only 1 person trying to alter file at a time. which is the only thing a database has to offer in this case Works great. Have been in production 8 months. |
16:41.14 | *** join/#asterisk littlejohn (n=little@host130-254.pool8263.interbusiness.it) |
16:43.52 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
16:43.57 | mikefoo | iCEBrkr: sup sup |
16:45.55 | iCEBrkr | yo |
16:46.02 | iCEBrkr | Mon Feb 20 11:45:57 EST 2006 |
16:46.03 | *** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com) |
16:49.13 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.159) |
16:50.40 | peanuter | anyone else having compiling problems with zaptel on fbsd 5.4? |
16:51.16 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
16:51.28 | mikefoo | peanuter: you have zaptel card or emulating it? |
16:51.36 | peanuter | emulating |
16:51.45 | tmccrary | Is there any functionality in asterisk to "barge into" a converstaton between two people? Like forcing a conference call? |
16:52.26 | trixter | tmccrary: you can always transfer people into a meetme |
16:52.46 | trixter | depending on system speed it might be anything from noticable to barely perceptable |
16:53.07 | trixter | I think thgere is a barge app though not sure |
16:53.21 | tmccrary | oh ok, thanks |
16:53.40 | mikefoo | trixter: hey sup.. |
16:53.55 | trixter | not much |
16:54.02 | *** join/#asterisk trelane_ (n=trelane@209.43.90.13) |
16:54.24 | mikefoo | any other decdent books on asterisk besides the voip-info one? |
16:54.32 | mikefoo | need to get some reading material at b&n today. |
16:54.37 | trixter | ~docs |
16:54.38 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
16:54.53 | [TK]D-Fender | mikefoo : the only one I could suggest would be TFOT |
16:54.58 | [TK]D-Fender | ~thebook |
16:55.00 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
16:55.05 | trixter | oh that way, no not really ,there is another asterisk related voip book, but there are many non asterisk voip books that are good to read |
16:55.14 | tmccrary | also, is there a way to "barge in" but only have one side of the conversation hear you? Say for training so you can talk to YOUR rep but not have the "customer" hear what you're saying (for like coaching lets say) |
16:55.16 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
16:55.26 | trixter | they tend to cover different aspects of voip in general, some in great detail about the technology itself |
16:56.24 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
16:57.49 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
16:58.32 | devel | hey all. anybody make a decent wall mount sip phone? |
16:59.03 | [TK]D-Fender | devel : What does it need to do? |
16:59.44 | devel | nothing special, just hang on a wall. everything i have here (sipura, grandstream, polycom. snom) won't mount on a wall... |
17:00.11 | *** join/#asterisk websae (i=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
17:00.31 | mutilator | use an ata? |
17:00.39 | mutilator | and a wall mount phone.. |
17:00.54 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
17:01.00 | [TK]D-Fender | devel : Well.... its not a great phone but I bought Uniden UIP-200's for wall mounting in public places (less $ risk if vandalized, and they support PoE |
17:01.07 | _Paulo_ | devel, try connecting a standarrd wall phone to an ATA, |
17:01.11 | [TK]D-Fender | Though I wouldn't suggest them for anything else.... |
17:01.26 | [TK]D-Fender | devel : Actually yeah... ATA would be the best way.... |
17:01.32 | [TK]D-Fender | cheapest too. |
17:01.52 | devel | yeah, they're for use in a "shop area", and i didn't really want to go the ATA way. but from a cheapness standpoint, i guess i can consider it. |
17:01.59 | tmccrary | is there a way to "barge in" but only have one side of the conversation hear you? Say for training so you can talk to YOUR rep but not have the "customer" hear what you're saying (for like coaching lets say) |
17:02.09 | devel | thanks, all. |
17:02.10 | mutilator | tmccrary: no |
17:02.16 | mutilator | so stop asking |
17:02.55 | mutilator | if they're for use in a 'shop' i'de say do it anyway |
17:03.02 | mutilator | don't want a $100 phone hangin on the wall |
17:03.06 | mutilator | rather a $10 |
17:03.15 | mutilator | u can put the ata in a closet somewhere |
17:04.09 | devel | yeah, i guess that does make more sense, just put the ATA in the wiring cabinet. |
17:05.52 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
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17:12.48 | mutilator | =\ |
17:13.01 | mutilator | i just noticed, the lil bumps on my f and j keys are gone |
17:13.28 | *** join/#asterisk GoRK (n=GoRK@amarillo.energynet.com) |
17:14.34 | [TK]D-Fender | devel : Whats wrong with ATA? It mean you can put a phone that will get abused there... |
17:15.02 | mutilator | i think we already knocked sense into him |
17:15.04 | mutilator | o_O |
17:15.06 | mikefoo | [TK]D-Fender: thanks for the book advice, I will download the pdf today along with the other docs :) |
17:15.28 | [TK]D-Fender | mikefoo : I duplex'd it here a few times for backup for when the WIKI goes down :) |
17:15.54 | mikefoo | :) |
17:17.31 | mikefoo | could someone point me in the right direction if I wanted to run a incoming line with ani ability, what I would need to do |
17:19.50 | t0ke | anyone know if I can to have enable musiconhold if I havent soundcard on server? |
17:19.55 | *** join/#asterisk T42X (n=T42X@193.219.62.88) |
17:20.30 | [TK]D-Fender | t0ke : Yes |
17:20.44 | GoRK | t0ke: mpg123 does not require a sound card, it decodes to stdout where asterisk reads the stream. Native moh will also work |
17:20.57 | [TK]D-Fender | mikefoo : ANI? As in "CALLED" # (like DID)? |
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17:21.18 | mikefoo | [TK]D-Fender: yup |
17:21.19 | t0ke | ok, thnks..then will be error mine in extensions.conf |
17:21.33 | mikefoo | http://en.wikipedia.org/wiki/Automatic_Number_Identification |
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17:22.11 | [TK]D-Fender | mikefoo : Your choices typically look like PRI/T1 or a VoIP provider that does the same. |
17:22.16 | [TK]D-Fender | What scale of solution? |
17:22.25 | GoRK | mikefoo: as a customer it is very difficult to get ANI. typically you can only get it on things like 900 numbers but sometimes your provider will give it to you on 800 numbers.. do you have a line with ANI already? |
17:22.48 | danzig | Ha - found error. Was not actually from the web interface that edits the include files, was manual edit by the other sysadmin :-( |
17:23.02 | GoRK | mikefoo: in any case you can only read ANI over ISDN/PRI D channels |
17:23.09 | danzig | Can anyone explain the logic of when CDR started/ended/written? Im messing with ForkCDR and ResetCDR(w), and niether do exaclty what I would expect... I need to do things like user comes in, gets dialtone, dials new number [start new CDR] continue with new CDR |
17:23.23 | mikefoo | GoRK: no line with ANI as of now. |
17:23.37 | mikefoo | so I need to find a voip provider that is offering ani, you are saying? |
17:23.54 | GoRK | mikefoo: why do you need ANI instead of just caller id? |
17:24.30 | mikefoo | my clients don't want to see "blocked call" on records. |
17:24.59 | [TK]D-Fender | mikefoo : I take it you wnt a number of differnt DID's from them? |
17:24.59 | mikefoo | its for lead generation so they can call back who ever calls them. |
17:25.17 | mikefoo | [TK]D-Fender: what do you mean? |
17:25.20 | GoRK | mikefoo: well i will tell you now that telling that to anyone capable of providing you ANI service will not be a sufficient reason to sell it to you |
17:25.52 | devel | yeah, [TK]D-Fender, i don't really know what i was thinking there... kind of a "all ethernet" mindset. i have indeed had sense knocked in to me :) thanks |
17:25.53 | [TK]D-Fender | mikefoo : as in you want a # of differnt phone #'s targeting your PBX and for your outbound calls to identify as coming from the specific # that will lead to the agent that called them? |
17:25.59 | GoRK | mikefoo: you can sometimes get ANI on 800# service though so that's probably your best bet |
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17:26.26 | [TK]D-Fender | devel : I run ATA's in my switching room, and plug the el-cheapo phones into the RJ45 jack direct. works great... |
17:27.10 | mikefoo | GoRK: but I was reading and came across: The Federal Communications Commission allows anyone who is paying for a phone call to know who is calling them. |
17:27.39 | GoRK | mikefoo: that is why you can sometimes get it on 800 number service and why you can on 900/976 service |
17:27.44 | Hmmhesays | freaking autofall through |
17:28.09 | mikefoo | Ahh ok.. |
17:28.29 | mutilator | anyone wanna buy a benchmade billabong butterfly knife? $150 |
17:28.55 | mutilator | http://www.icegn.net/knife/ |
17:29.13 | mikefoo | [TK]D-Fender: oh, well no its other way around, I have numbers pointing to PBX, client calls into PBX with *67 for instance, my client is charged for the call, but doesn't know who called, so thats why they are bitching. |
17:29.26 | GoRK | mikefoo: but if your caller is not calling into your 800/900/976 number, then no ANI.. also the only way to actually receive the ANI is via ISDN/PRI -- so you'd need an ISDN/PRI or a T1/PRI at the very least.. or buy an 800# from a voip provider that will relay the ANI to you instead of the normal callerid |
17:29.32 | paulhuynh | does anyone here work on asterisk@home |
17:29.35 | paulhuynh | can help me |
17:29.39 | paulhuynh | I broke my |
17:30.12 | paulhuynh | asterisk will not register any device anymore after upgrade centos + asterisl1.2.1 to 1.2.4 |
17:30.15 | mikefoo | GoRK: yah I was thinking of getting 800# from voiup provider that can possibly relay ANI information to me. |
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17:30.54 | GoRK | mikefoo: it would be a specialized sort of deal.. afaik no protocols have provisions for both callerid and ANI at the same time, so they'd have to replace callerid with the ANI on calls coming to you |
17:31.51 | GoRK | mikefoo: but i dont see any reason they couldn't do it from a technical perspective.. they would of course have to be able to receive and capture the ANI with whatever equipment they have |
17:32.06 | mikefoo | oh yah? ahh ok then maybe I need to not charge clients for blocked calls I guess, I will see how that goes. |
17:32.29 | mexuar-tim | GoRK: actually IAX has an info element for ANI. I don't know if any one supports it. |
17:32.46 | GoRK | mexuar-tim: ah didnt know that/ guess i should read more code :) |
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17:34.11 | mexuar-tim | GoRK: I'm deep into debugging some IAX code, so I have the spec open all the time :-) |
17:34.55 | GoRK | mikefoo: anyway normally the ANI is simply used behind the scenes by whoever is providing the 800# service for billing purposes.. so it's not strictly necessary to pass it to the customer WITH the call.. so you will have to negotiate with a provider to do this for you.. even if you are dealing with a CLEC or something, they may (and usually are) buing 800# service from someone else so they may not even be getting ANI themselves |
17:34.59 | mikefoo | well I need to buy some dids anyway.. anyone have experince with virtualphoneline? |
17:36.49 | *** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com) |
17:37.38 | file | we do get ANI at Asterlink, but we don't pass it to the customer... like GoRK said we use it for rare billing instances |
17:37.48 | file | ie: US cellphone customer roaming in Canada, gets billed at Canada inbound rate |
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17:39.52 | mocker | Does asterisk have a drop folder where I can make it call by placing a file in that directory? |
17:40.05 | wasim | mocker: /var/spool/asterisk/outgoing |
17:40.23 | mocker | wasim: Awesome, thanks. |
17:41.41 | GoRK | mocker: it's very greedy about .call files though; make sure you create them somewhere else on the FS then use mv to put them in the spool directory.. it doesnt do any kind of checking to make sure it's got a complete file not open by other processes before it reads it in.. if you use cp or write directly to the dir sometimes asterisk will get at the file before you have finished creating it |
17:42.47 | GoRK | mocker: also be wary about too many files in the spool dir at once.. try to serialize them a bit if possible with a slight dealy between -- 500ms to 1s should do |
17:43.33 | file | if god had a name what would it be and would you call it to his face if you were faced with him in all his glory |
17:43.42 | file | what would you ask if you had just one question? |
17:44.26 | Nivex | file: are you feeling alright? (that was directed at you, not God) |
17:44.34 | file | always |
17:45.18 | GoRK | i'd ask him who wants to buy a benchmade butterfly knife? WTF is it with all the offtopic crap? we might as well all start with the Chuck Norris stuff |
17:46.14 | file | GoRK: it's very very... calm today |
17:46.51 | GoRK | Chuck Norris while having a calm day at home was watching television when he became furious and began walking down the street punching every kid he saw and screamed "Trix are for Chuck Norris" |
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17:51.09 | file | dang nabbit my printer is broken |
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17:51.57 | GerbilWrk | Can you have hardcoded agents in a queue, plus agents that can call in and add themselves to it? |
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17:56.02 | [TK]D-Fender | GerbilWrk : yup |
17:56.45 | robb_ | i'm having a problem getting pstn->voip to work. i have an asterisk box running on fedora kitted out with an x100p. when i ring the line from the pstn the recorded greeting sounds fine, but when i try and dial an extension to a sip/iax client there's a lot of hiss. does anyone know why that might be the case? |
18:00.53 | mikefoo | file: so I am not passing it to customer, I am just including it in billing statements. so my clients don't see a bill for a blocked number. |
18:01.33 | *** join/#asterisk Cresl1n (n=matt@146.229.177.231) |
18:04.10 | [TK]D-Fender | robb_ : Is your X100 on its own IRQ? Running any other nasty processes like X? |
18:04.16 | mikefoo | file: I also am thinking about going with asterlink, I heard some good things. |
18:04.47 | robb_ | no, i've killed X and other nasty things like that. it's on its own irq |
18:04.52 | robb_ | irq 10 |
18:05.02 | robb_ | getting 1000 interrupts a seconds |
18:05.16 | file | mikefoo: well you're more then welcome |
18:05.19 | [TK]D-Fender | hmm... set up an IVR option for an echo test and see if it sounds clean while it remains within *. |
18:05.44 | robb_ | ok. good idea |
18:06.02 | *** join/#asterisk Mother (n=m@53.Red-217-126-93.staticIP.rima-tde.net) |
18:06.11 | *** join/#asterisk ncjp (n=switch@61.206.115.5.user.ad.il24.net) |
18:06.14 | Mother | greetings |
18:06.30 | Mother | is there a way to get asterisk to answer SDP payloads? |
18:06.59 | Mother | or if not, any pointers as to where in code I could start prodding? |
18:07.15 | zoa | robb_: thats normal |
18:07.20 | austinnichols101 | ~seen opsys |
18:07.25 | jbot | opsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 7d 12h 26m 23s ago, saying: 'betaboi" true'. |
18:07.25 | zoa | 1000 interrupts per second are ok |
18:08.18 | Mother | I have a bunch of a= records coming from a client, but asterisk only sends back a bare 200 reply |
18:09.11 | robb_ | zoa: yeah i meant that its not an irq issue because it's getting the 1000 interrupts a second |
18:09.41 | GerbilWrk | anyone have experience making snom 360's not show a missed call if the missed call goes to a queue? |
18:13.04 | [TK]D-Fender | GerbilWrk : Sorry... if a SIP phone misses a call it'll have no idea what the server will do with it after in order to make a decision.... |
18:13.04 | robb_ | [TK]D-Fender: during the echo test there was a low frequency noise even when there was no audio being input |
18:13.37 | [TK]D-Fender | robb_ : Plug a real phone into the jack you use for the X100. Does it sound crappy as well? |
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18:14.08 | robb_ | [TK]D-Fender: tried that already. the line sounds ok |
18:14.41 | [TK]D-Fender | robb_ : Hmmm.... played with the gain/loadzone any? |
18:15.00 | [TK]D-Fender | is the noise a stable constant hum or variable? |
18:15.20 | robb_ | constant hum. played a bit with the gain a while ago |
18:15.39 | robb_ | hum becomes distortion when i speak into the line |
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18:17.37 | Mother | so, no takers on the SDP issue? |
18:18.52 | [TK]D-Fender | robb_ : Hmmm... maybe an impedance thing or other electrical interference.... |
18:20.15 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp32-adsl-178.ath.forthnet.gr) |
18:20.35 | robb_ | [TK]D-Fender: sounds plausible alright. |
18:20.51 | robb_ | [TK]D-Fender: might try moving it. cheers |
18:22.49 | *** join/#asterisk rene- (n=rene-@201.127.101.127) |
18:22.54 | [TK]D-Fender | robb_ : Glad if it helps... |
18:22.59 | rene- | hello |
18:23.27 | rene- | is the tdm400 5v or 3.3v? |
18:24.06 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
18:24.19 | [TK]D-Fender | rene- : dual IIRC |
18:25.15 | rene- | [TK]D-Fender: well it seems that it would fit better in the pci-x slots of the server than in the 5v 32bit rated ones |
18:25.44 | rene- | the tdm400 has an extra rib not found in the pci socket but well it did went all the way so |
18:26.04 | [av]bani | \o/ |
18:26.25 | iCEBrkr | rene-: I believe it's auto-sensing. I can only make this assumption from the picture. |
18:26.47 | iCEBrkr | rene-: It's slotted for both |
18:28.26 | rene- | i see, well i need to turn the server on and see if it works, |
18:28.55 | rene- | thanks, otoh is there anyone from signate in here? i have some questions regarding their product |
18:29.07 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
18:29.34 | *** part/#asterisk mac7 (n=karsten@c184067.adsl.hansenet.de) |
18:31.59 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
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18:36.24 | QbY | does anyone know how a MOS score can be calculated automatically for each call? |
18:37.05 | *** join/#asterisk |omni| (i=rob@net98.limelyte.net) |
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18:42.55 | trixter | you can automatically calculate it |
18:43.08 | trixter | there are 2 basic ways of doing MOS scores |
18:43.23 | trixter | one is you have a group of people rate the call 1-5 average em together there is your score |
18:43.27 | trixter | the 2nd is twice as good |
18:43.30 | trixter | they rate it 1-10 |
18:43.44 | *** join/#asterisk ZipKid (n=ZipKid@d51A48D49.access.telenet.be) |
18:44.55 | Abydos313 | good morning everyone |
18:44.59 | trixter | I have asked digium about a year ago if they had any MOS scores and under what environment that was done, they told me (at that time) they hadnt done any such testing. and in reality the system, network, endpoints all add into that so you may end up with a lower mos score just because you were using a worse phone or had network congestion |
18:45.26 | Qwell[] | or can't figure out how to fix echo properly |
18:45.31 | Abydos313 | i'm having an issue with AMP. all works except the 'extensions' link |
18:45.35 | mutilator | or doing it over 9 mile wifi links |
18:45.36 | Qwell[] | ~amp |
18:45.37 | jbot | methinks amp is NOT supported here! people using it should join #amportal |
18:45.48 | Qwell[] | mutilator: or 11 in ManxPower's case |
18:45.55 | mog_work | hey qwell |
18:45.56 | trixter | you can try to guess what the score might be if you look for high jitter, dropped packets, and run sphinx or something looking for the word 'what' |
18:46.00 | Qwell[] | mog_work: y0 |
18:46.01 | mutilator | yea, we have 1 20 mile link |
18:46.08 | Qwell[] | mog_work: Why isn't today a holiday? |
18:46.14 | mog_work | you have been working on chan_skinny a lot lately |
18:46.18 | Qwell[] | or, rather...why am _I_ at work on this holiday? :( |
18:46.18 | mog_work | we dont work for the gov |
18:46.21 | mutilator | try not to do more than 10 tho |
18:46.30 | Qwell[] | mog_work: I work for a bank...95% of the people here are off. :( |
18:46.38 | Qwell[] | it's so lonely here, heh |
18:46.39 | *** part/#asterisk rene- (n=rene-@201.127.101.127) |
18:46.46 | trixter | what holiday? |
18:46.52 | Qwell[] | presidents day or something |
18:46.57 | mog_work | lol |
18:47.03 | Qwell[] | mog_work: never touched chan_skinny...only chan_sccp |
18:47.09 | mog_work | thats what i meant |
18:47.17 | mog_work | i need some help writing rtp handlers |
18:47.21 | mog_work | for the channel im writing |
18:47.33 | Qwell[] | I've not gotten that far into it... |
18:47.37 | mog_work | ahh okies |
18:48.02 | Qwell[] | chan_skinny is probably unbloated in that regard though |
18:48.11 | Qwell[] | might be a good place to look for advice |
18:48.26 | Qwell[] | (the real chan_skinny...) |
18:48.48 | mog_work | ahh |
18:48.54 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
18:48.56 | mog_work | reading chan_sip gave me a headache |
18:50.01 | ZipKid | hi, simple question, is it possible to use Asterisk with a simple external serial modem for the incoming telephone line? |
18:50.25 | Qwell[] | ZipKid: and do what with the modem? |
18:50.33 | Qwell[] | use it as a modem, over voip? |
18:50.37 | cypromis | d beep beep beep |
18:51.00 | ZipKid | recieve incoming calls and sent them on to an internal softphone for instance |
18:51.06 | Qwell[] | no |
18:51.08 | ZipKid | no, pots |
18:51.11 | Qwell[] | you need an fxo |
18:51.18 | Qwell[] | modem != fxo |
18:51.43 | trixter | or someone to press the bare phone wires on their tounge and is really good with a morse code keyer so they can put the bits on the ethernet wire in realtime |
18:51.45 | mog_work | well it would if someone wrote a driver for it, and it was a capable modem |
18:51.46 | ZipKid | ah, ok.... |
18:51.49 | trixter | but watch for jitter on that setup |
18:51.55 | Qwell[] | mog_work: well, sure |
18:52.09 | Qwell[] | but, in that case...you could use anything, really |
18:52.19 | ZipKid | Qwell[]: thanks, that is not clearly defined anywhere i can find.. |
18:52.25 | mog_work | ZipKid, you can get an x100p off of ebay for 10 bucks or a digium tdm01b for like a hundred |
18:52.27 | Qwell[] | rj11 over serial port, rj11 over ethernet (hey, if there are drivers, it could work, right?) |
18:52.35 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
18:52.53 | ZipKid | mog_work: i have seen thos things but i wanna play now ! :-) |
18:52.55 | Qwell[] | actually, rj11 over serial would be kinda cool, heh |
18:53.16 | Qwell[] | just get an el-cheapo adapter for like $1.50 |
18:53.33 | mog_work | well get one overnighted |
18:53.35 | Qwell[] | you don't need a dsp, do you? |
18:53.52 | mog_work | do it in software qwell |
18:53.58 | Qwell[] | yeah, that's what I'm thinking |
18:54.07 | Qwell[] | there are already thinsg that do that, right? |
18:54.10 | Qwell[] | in asterisk, that is |
18:54.14 | mog_work | thats what i was telling marko |
18:54.19 | mog_work | we should stop selling t1 cards |
18:54.22 | Qwell[] | heh |
18:54.26 | mog_work | and just do ethernet |
18:54.27 | Qwell[] | just do it over ethernet? |
18:54.30 | Qwell[] | yeah, totally |
18:54.32 | ZipKid | TDM01B : 1 FXO port 124.22EUR |
18:54.36 | mog_work | and do all framing signalling garbage in software |
18:54.42 | ZipKid | no shipping included |
18:54.52 | Qwell[] | mog_work: good idea, except the whole paying Digium's bills thing :p |
18:54.55 | mog_work | thats a bit of a markup we sell it state side for around 100 i thought |
18:55.13 | mog_work | well and it would be immensally difficult to do in real life |
18:55.29 | Qwell[] | Kevin could knock it out in 4 hours! |
18:55.30 | Qwell[] | heh |
18:55.40 | Abydos313 | g4 kevin? |
18:55.46 | mog_work | getting kevin for 4 hours is incredibly difficult though |
18:55.54 | Qwell[] | mog_work: indeed, heh |
18:56.07 | Qwell[] | the other day, Kevin said "anything in asterisk can be done in 4 hours" |
18:56.16 | Qwell[] | "or...that's what the sale guys tell everybody" |
18:56.20 | Qwell[] | ;] |
18:56.30 | mog_work | sounds like sales |
18:56.40 | denon | thats what the tech guys always say too |
18:56.50 | ZipKid | Qwell[]: but one of those crappy winmodem cards would do the trick? Or have i again terribly misread things...? |
18:56.59 | mog_work | lol |
18:57.02 | Qwell[] | ZipKid: You've terribly misread things... |
18:57.07 | Qwell[] | ~fxofxs |
18:57.18 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
18:57.22 | Qwell[] | thank you |
18:57.34 | denon | ~fwap Qwell |
18:57.47 | Qwell[] | ZipKid: If you wrote zap drivers for the modem, you could use it |
18:57.51 | Qwell[] | but...good luck |
18:57.58 | ZipKid | hum.. no thanks |
18:58.18 | denon | not to mention, not all modems would duplex well, as I understand it .. |
18:58.20 | Qwell[] | denon: My standard answer is "a few days" |
18:58.34 | denon | er, FD |
18:58.46 | Qwell[] | "Oh, you need a typo fixed? That'll take me a few days" |
18:59.31 | denon | "But ... but .. it's just on the wiki!" |
18:59.52 | Qwell[] | "Yes, but there are change control processes you're forcing me to follow. So...it'll take me a few days." |
19:01.16 | *** join/#asterisk teleniek0 (n=marc@167.Red-80-35-144.staticIP.rima-tde.net) |
19:01.44 | *** join/#asterisk DanielArndt (n=DanielAr@reverse-82-141-61-29.dialin.kamp-dsl.de) |
19:01.56 | teleniek0 | hi ppl. would you give me some clues on getting a TE110P working on Spain ? when I plug the card to the PRI box a nice "ALM" led blights on it ;( |
19:02.09 | *** join/#asterisk clint_ (n=clint@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
19:02.12 | Qwell[] | teleniek0: got the right kind of cable? |
19:02.22 | Qwell[] | I know zero about T1, so...yeah |
19:02.50 | teleniek0 | Qwell[] the cable is a straight cat5? |
19:02.58 | Qwell[] | got me |
19:03.18 | teleniek0 | wasn't sure but hope I guessed hehe |
19:03.31 | teleniek0 | it's E1 not T1 anyway :)) |
19:03.41 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
19:03.42 | *** join/#asterisk PaulHuynh (n=paulhuyn@c-68-37-18-82.hsd1.de.comcast.net) |
19:03.43 | Qwell[] | should be the same |
19:03.49 | PaulHuynh | hello everyone |
19:03.59 | PaulHuynh | i'm rebuilt the asterisk@home |
19:04.07 | teleniek0 | maybe, anyway it's red red red all the time ughh x"D :) |
19:04.10 | Qwell[] | teleniek0: tried calling Digium? You get free installation support for buying a card |
19:04.14 | PaulHuynh | i'm going to recreate all the ext + config again |
19:04.21 | Qwell[] | They're there to help...and they're really good I hear |
19:04.24 | teleniek0 | Qwell[] didn't know! hehehe thanks :) |
19:04.29 | PaulHuynh | but i wonder how can i keep my old CDR |
19:04.54 | MRH2 | hi anyone forsee a problem using both monitor and mixmonitor in the same extension to create 2 recording files |
19:05.16 | Qwell[] | MRH2: besides that it'd be pointless? |
19:05.29 | MRH2 | yeah besides that |
19:05.36 | Qwell[] | probably. try it |
19:05.41 | MRH2 | something like 4,MixMonitor(/tmp/MM/${CALLFILE}.g729|bv(1)V(1)) ... 5,Monitor(gsm,${CALLFILE},bm) |
19:05.52 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
19:07.03 | Qwell[] | seems kinda silly to me, to write 3 streams to disk at a time when one will do |
19:07.26 | MRH2 | mix monitor let me down earlier this year so i am trying to ease into it |
19:08.00 | *** join/#asterisk trelane_ (n=trelane@209.43.90.13) |
19:08.05 | *** join/#asterisk Cresl1n (n=matt@146.229.177.231) |
19:09.20 | jontow | woo, fixed another issue with the VM machine.. longstanding one too :) |
19:09.37 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
19:09.41 | Qwell[] | VM machine? |
19:09.50 | Qwell[] | that kinda like an ATM machine? |
19:10.19 | jontow | voicemail for our CLEC ;) |
19:10.36 | Qwell[] | your CLEC carrier? |
19:10.39 | jontow | it sits behind a coppercom softswitch, and gets calls thrown at it via forward-on-busy-or-unavailable |
19:10.44 | jontow | we are the CLEC |
19:10.44 | jontow | :) |
19:10.53 | Qwell[] | You don't get it yet, do you? :p |
19:10.57 | jontow | of course not! |
19:10.58 | Qwell[] | too subtle, I guess |
19:11.09 | Qwell[] | VM machine is redundant :P |
19:11.24 | jontow | how so? |
19:11.30 | Qwell[] | M = machine |
19:11.35 | jontow | M = mail |
19:11.35 | Qwell[] | oh, haha...voicemail |
19:11.40 | jontow | ;) |
19:11.44 | trixter | but its not EM |
19:11.46 | trixter | wonder why |
19:11.50 | Qwell[] | EM? |
19:11.53 | trixter | email |
19:11.53 | Qwell[] | email? |
19:11.56 | Qwell[] | hmm |
19:11.57 | jontow | trixter: because the E is the only important letter! |
19:11.58 | trixter | as in electronic mail |
19:12.01 | Qwell[] | yeah |
19:12.08 | jontow | mail has happened forever.. but *E*mail, now thats something new and exciting! |
19:12.09 | Qwell[] | though, some people do refer to it as vMail |
19:12.11 | trixter | and its not SM for snail mail |
19:12.18 | Qwell[] | no, but SM + squirrelmail |
19:12.23 | trixter | although content filters may have a problem with people tossing letters like S&M around |
19:12.36 | Qwell[] | s/+/=/ |
19:13.07 | *** join/#asterisk bjohnson_ (n=bjohnson@66.11.164.106) |
19:13.15 | jontow | but, aside the semantics of the issue.. we had a longstanding problem because we had to swap the fields that were being shoved into the voicemail system, so it would ring into * as a DID, though none of the numbers were actually destined for it.. creates a small problem when you also have a generic "dial X number from outside our network to check your voicemail from anywhere!" |
19:13.39 | jontow | so i had to play some magic with a perl script named 'ismbox.pl' and a SQL query or two.. |
19:13.57 | jontow | felt like an ass once i'd done the proof-of-concept deployment on my dev. box and realized it was super-easy to write |
19:14.01 | Mother | yo |
19:14.01 | jontow | :P |
19:14.12 | Mother | trixter....as in the tscm-l trixter? |
19:14.13 | Qwell[] | brb |
19:14.21 | jontow | took 25mins to write it.. mostly because i don't know perl |
19:14.23 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
19:14.25 | trixter | yes |
19:14.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
19:14.30 | Mother | well greetings |
19:14.38 | trixter | hi |
19:14.44 | trixter | which list? there are so many |
19:14.53 | Mother | JMA's one |
19:15.00 | trixter | oh the sucky one |
19:15.02 | Mother | *the* TSCM list |
19:15.04 | Mother | LOL! |
19:15.23 | Mother | the Uhrigh-bashing one |
19:15.30 | trixter | its all lame jokes and random spam from him, since moving I noticed he is no longer using it to harass someone (against yahoos rules and most likely why they canceled the group) |
19:15.50 | Mother | yeah, was the group actually cancelled by yahoo? |
19:16.03 | Mother | I was rather busy at that time and have some 200 backlogged messages |
19:16.15 | *** join/#asterisk MikeJ__ (n=vircuser@71-36-209-237.dlth.qwest.net) |
19:16.17 | trixter | yeah after 6 months of posting personal information and using it to harass someone... he claims its becuase he posted information that is publicly available on any ham radio site |
19:16.37 | trixter | 200? I get that many in a day from all the lists especially asterisk-users |
19:16.43 | Mother | wtf... |
19:16.54 | Mother | oh yeah, the asterisk one is quite an exercise :D |
19:17.11 | Mother | at one stage I had 20k unread on that one |
19:17.14 | trixter | I think asterisk-users is averaging about 150/day now |
19:17.22 | PaulHuynh | anyone? |
19:17.36 | Mother | jeez |
19:17.46 | PaulHuynh | i need to tranfer some cdr info over from my dead asterisk to my working one |
19:17.53 | Mother | any talk of PoC? I'm trying to get it working with the Nokia client |
19:17.56 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
19:18.16 | Mother | but there's a bunch of crap on the SDP portion of the register message, to which * doesn't answer |
19:19.07 | Mother | I thought of writing a proxy that took care of that and translated it asterisk-friendly, but I'd rather have it do it natively |
19:19.53 | *** part/#asterisk godfree2 (n=user@hlfxns0148w-142167238144.ns.aliant.net) |
19:19.56 | Abydos313 | would be nice if asterisk only used one or two static ports for everything. would help with fireall issues |
19:19.57 | file | a REGISTER shouldn't/needn't contain SDP |
19:20.24 | file | and last I checked the Nokia client used a codec that Asterisk didn't/couldn't support due to licensing |
19:20.51 | Mother | Abydos313: it's called IAX |
19:21.15 | remiss | eeks |
19:21.24 | Mother | file: yes, it's AMR, but I'm not sure on the licensin issue - you can download C source from the 3GPP site |
19:21.29 | Mother | *licensing |
19:22.00 | [av]bani | bleh |
19:22.03 | Abydos313 | Mother do you have a preference for softphones for windows that use iax? or hardware adapter |
19:22.04 | Mother | file: besides, I don't want it doing translation right now, all it needs to do is act as marshall and gateway, so-to-speak |
19:22.17 | remiss | Abydos313: idefisk works... |
19:22.45 | Abydos313 | i have that one but haven't tried it yet. i really like xlite. wish it were iax |
19:22.47 | file | Mother: Asterisk acts as B2BUA, ie: two separate call legs... not as a proxy |
19:23.40 | Mother | file: agreed, but it can just simply forward the RTP right? there is no need for it to support the codec, as with G729, where it has to be licensed if you want to translate between it and other codecs |
19:24.00 | file | it needs to know about the codec |
19:24.09 | file | as it doesn't simply forward the RTP to start with |
19:24.43 | Mother | hmmmm define *know* in regards of G729 when it's not licensed on a particular asterisk server |
19:24.56 | Qwell[] | Mother: the codec has to be defined in * |
19:24.58 | file | internally in the core Asterisk still knows about G729 |
19:25.01 | Qwell[] | at the very least |
19:25.11 | file | it knows it exists, it knows it's SDP information |
19:25.16 | Mother | Qwell[]: agreed |
19:25.53 | Mother | OK, that's a starting point - could I then define the extra headers that the Nokia client sends, associated to AMR/8000 for example? |
19:26.03 | file | ummm hold |
19:26.13 | file | http://quark.file-radio.com/asterisk/amr_passthru.diff |
19:26.30 | file | that's for 16000, but you can uh... use it as a base |
19:26.45 | Mother | great stuff, thanks! |
19:27.00 | Mother | now I need to implement floor control and all the other crap :) |
19:29.09 | Mother | oh...and...."convergence"....seemed to be *the* buzzword at 3GSM |
19:29.33 | *** join/#asterisk sch19 (n=sch19@adsl-9-107-161.mia.bellsouth.net) |
19:29.35 | file | Asterisk - Now with SpeedyConvergence(tm) technology! |
19:29.51 | Mother | lol you could make millions with that |
19:30.28 | _Sam-- | <PROTECTED> |
19:30.35 | _Sam-- | <PROTECTED> |
19:30.37 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
19:30.43 | Mother | there was a stand that all it had was a platform, and half-naked women, and they sold something 'convergent' |
19:30.58 | _Sam-- | when you were gone on friday, switch-04 decided to not work |
19:31.03 | _Sam-- | *sigh* |
19:31.12 | file | _Sam--: bkw and MikeJ also work for Asterlink |
19:31.14 | file | so you can bug them too |
19:31.25 | _Sam-- | im done bugging people i just want stuff to work! |
19:31.39 | _Sam-- | i have 10.4% loss at the last hop before switch-* |
19:33.20 | file | bkw_: poke |
19:33.39 | trixter | Mother file: yes, it's AMR, but I'm not sure on the licensin issue - you can download C source from the 3GPP site |
19:33.44 | trixter | AMR is patented and not free |
19:33.54 | trixter | anything that uses it (like GSM/AMR) requires a license |
19:34.34 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-111-54.38-151.net24.it) |
19:35.38 | file | _Sam--: AIM. |
19:36.02 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
19:38.37 | websae | i am curious--is it possible to setup such a network where you have two WAN connections, internally you have an asterisk server and sip phones as well as machines surfing the internet,---is it possible to hook up one of WAN connections to the asterisk box (with 2 lan cards) so that all sip trafic goes through one connection |
19:38.52 | Qwell[] | websae: Sure |
19:39.05 | Qwell[] | asterisk can listen on a single interface |
19:39.12 | Qwell[] | or, rather, on an IP |
19:39.44 | trixter | _Sam--: do you have network problems to other sites? maybe try www.yahoo.com and www.google.com |
19:39.51 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
19:39.52 | trixter | it may be a general network problem and not any one specifically |
19:40.01 | trixter | that should at least tell you if its closer to you or not |
19:40.40 | *** join/#asterisk backblue (n=moo@87-196-12-97.net.novis.pt) |
19:41.26 | Mother | trixter: you surely mean in the US.... |
19:42.19 | Mother | we haven't invented software patents yet |
19:42.38 | *** join/#asterisk lazzarello (n=lee@dsl254-077-209.nyc1.dsl.speakeasy.net) |
19:43.27 | trixter | just becuase the implementation is in software doesnt mean the patent is only software |
19:43.44 | websae | what type of server hardware would one suggest for an asterisk box supporting 30 end users |
19:43.45 | websae | ? |
19:43.57 | trixter | remember you cant really patent an implementation as such, that is what copyrights are for, you can patent the process indifferent to the implementation |
19:44.00 | Mother | er...well...if there was a *chip* doing AMR then yes, but this is hardly the case |
19:44.29 | *** join/#asterisk breakdecks (n=breakdec@adsl-065-006-209-021.sip.mem.bellsouth.net) |
19:44.32 | trixter | but it may not matter depending on things.. if the patent is on a process and the process is implemented in software that doesnt mean its a software patent per se |
19:44.38 | backblue | anyone knows any way to get TEI fixed with zaphfc? |
19:44.43 | Mother | agreed |
19:44.47 | trixter | it might, it depends on a variety of factors |
19:45.42 | Mother | but in the case of codecs, I don't think that applies, at least if it was so no doubt there'd be a flurry of lawsuits in EU in a matter of minutes |
19:45.42 | trixter | AMR has to do with voice coding and odds are they wrote it in such a way to cover both hardware as well as software implementations sine its likely to be both |
19:45.42 | breakdecks | I am having an issue registering with FWD |
19:45.42 | trixter | AMR isnt a codec per se |
19:45.52 | [av]bani | ilbc 4-evar |
19:45.56 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
19:46.07 | trixter | its a way of doing parts that can be used in a codec. such as GSM/AMR is the GSM codec but with the AME stuff.. as opposed to the GSM/FE which is free |
19:46.25 | _Sam-- | is above.net any better than cogent? |
19:46.26 | trixter | doesnt ilbc use as much cpu as g7239 but more bandwidth? |
19:46.30 | websae | any suggestions for what requirements i would need to setup a asterisk machine supporting 30 end users...anyone? |
19:46.38 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
19:46.45 | trixter | webmind: more information would be a start |
19:46.51 | breakdecks | I put in my correct register => Number:Password@iax2.fwdnet.net in iax.conf, but my console says it can't log in |
19:46.57 | Mother | http://www.dsprelated.com/showmessage/17150/1.php |
19:47.04 | Mother | that's some bit of info |
19:47.05 | trixter | what are those 30 people going to be doing? how many concurrent calls? transcoding? hardware? |
19:47.06 | trixter | etc |
19:47.33 | [av]bani | according to show translations, ilbc takes slightly less cpu than g729 |
19:47.35 | [av]bani | on x86_64 |
19:47.35 | austinnichols101 | sam: you lookin for a provider? |
19:47.42 | *** join/#asterisk Seldon1975 (n=someone@toronto-HSE-ppp4239697.sympatico.ca) |
19:47.46 | websae | i am wondering what hardware i would need.....it would only transcode for voicemail....and prolly 10-15 concurrent calls |
19:47.55 | _Sam-- | austinnichols101 : no , just looking for a better route to some. |
19:47.57 | *** join/#asterisk Seldon1975 (n=someone@CPE0013105d0913-CM0014e8b6162c.cpe.net.cable.rogers.com) |
19:48.05 | Qwell[] | websae: you could get away with about 1ghz |
19:48.11 | _Sam-- | wondering if above.net is any better than cogent |
19:48.15 | Qwell[] | so...any recent machine would do the job nicely, for only 15 calls |
19:48.27 | [av]bani | g729 uses 8kbps, ilbc uses 13 |
19:48.34 | iCEBrkr | _Sam--: IT RUNS! IT RUNS! |
19:48.43 | _Sam-- | [av]bani: above or cogent? |
19:48.47 | austinnichols101 | sam: in terms of reliability or performance? |
19:48.47 | websae | cogent |
19:48.51 | [av]bani | _Sam--: yes |
19:48.54 | _Sam-- | lol |
19:49.01 | websae | cogent---reliability and performance |
19:49.04 | [av]bani | _Sam--: um, neither? |
19:49.12 | [av]bani | websae: reliability? lollerskates. |
19:49.14 | denon | avoid cogent .. |
19:49.18 | denon | at all costs :) |
19:49.21 | _Sam-- | if you could get data to a host...and that host was on cogent or above...which way would you go |
19:49.22 | Qwell[] | cogent = pissy ex-girlfriend |
19:49.56 | [av]bani | both abovenet and cogent are poo |
19:49.56 | denon | I know they're offering you 20Gbps for $9.95/mo .. |
19:49.56 | denon | but its not worth it |
19:49.56 | [av]bani | they both are very spam-friendly and get blacklisted often |
19:49.56 | [av]bani | and cogent especially is subject to lots of ddos |
19:50.09 | [av]bani | because any script kiddie can buy hosting on there and setup irc server and warez and bots... |
19:50.22 | _Sam-- | seems whichever way i set up my remote gateway, its an every other day affair...one day its fine, the next it isnt. |
19:50.36 | _Sam-- | and i am doing nothing but chasing my own tail day after day |
19:50.42 | Mother | trixter: it looks like there are companies developing AMR codecs for licensing, not just one, so I guess the copyright is on the implementation, and not the purpose, so-to-speak |
19:51.01 | Mother | IMHO you could develop your own version/implementation, and be free from IP issues |
19:51.26 | austinnichols101 | IMO, interNap is the network choice |
19:51.30 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@71-36-209-237.dlth.qwest.net) |
19:51.51 | [av]bani | austinnichols101: internap is also very spammer-friendly ("bullet proof hosting") |
19:51.57 | AndyCap | Mother: copyright and patents are not the same. |
19:52.19 | austinnichols101 | [av]bani: sounds like you're thinking of Interland. |
19:52.30 | *** join/#asterisk jtodd (n=jtodd@mccpool-2.ci.monterey.ca.us) |
19:52.33 | trixter | you c ant patent the implementation only the process, the implementation would be copyrighted, and in effect you can end up oweing to two people one who holds the copyright and the other the patent |
19:52.35 | austinnichols101 | [av]bani: InterNap does't put up with spam at all |
19:52.54 | Mother | AndyCap: I'm aware of that |
19:52.59 | [av]bani | austinnichols101: no. pnap.net. definitely. |
19:53.06 | [av]bani | austinnichols101: they DO NOT respond to complaints. |
19:53.28 | _Sam-- | [av]bani: what backbone does your VOIP traffic take to your remote gw for terminating calls? |
19:53.33 | denon | austinnichols101: what's your experience with internap these days? Ive talked to a few carriers who have dropped them in the couple years because they were "slipping" |
19:53.46 | AndyCap | Mother: so how do you see people developing an AMR implementation free from patent issues? |
19:54.02 | austinnichols101 | [av]bani: I've had a completely different experience. I get excellent response from them. |
19:54.05 | sevard | denon: were you the one who made the weather script using Alison Smith? |
19:54.23 | Mother | AndyCap: who holds the patent to start with? I don't think there is a patent on AMR - Ericsson claimed to have one but it was rejected |
19:54.30 | [av]bani | austinnichols101: are you a spammer =) |
19:54.42 | austinnichols101 | [av]bani: but I've never worked with them as a non-customer... |
19:54.51 | Mother | and now 3GPP has standarised it as the mandatory codec for 3G, so I can hardly see *everyone* having to pay royalties to 3GPP for it |
19:54.57 | _Sam-- | [av]bani : how sure are you that BLF works fine with GXP2000 / * 1.2.*? |
19:55.01 | [av]bani | austinnichols101: pnap were hosting vibe direct media for ages. (NET-64-94-51-128-1). unrelenting spammers. |
19:55.09 | _Sam-- | because ive done some more tests and can break it pretty much at will |
19:55.09 | [av]bani | austinnichols101: and pnap took like 8 months before they wree booted |
19:55.11 | AndyCap | Mother: I guess voiceage claims to hold the licensing rights to the "patent pool" |
19:55.37 | [av]bani | _Sam--: i havent used it heavily, but it goes blinky blinky in my tests |
19:55.50 | austinnichols101 | denon: I get a 100% SLA from internap as they're really just running the seven largest us carriers across their flow-control-platform hardware. |
19:55.52 | _Sam-- | the extensions you are hinting, are gxps also? |
19:55.52 | iCEBrkr | Blinky lights are the essense of technology |
19:55.56 | [av]bani | _Sam--: yes |
19:55.59 | AndyCap | Mother: that may of course be FUD or they have a legitimate patent. :P |
19:56.00 | _Sam-- | how many? |
19:56.04 | *** join/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com) |
19:56.06 | Mother | AndyCap: well, possibly, but I don't think they could get away with it if challenged - people using their codec will pay because they've done the implementation work |
19:56.09 | *** join/#asterisk konfuzed (n=KonfuzeD@H135.C72.B0.tor.eicat.ca) |
19:56.17 | austinnichols101 | denon: and I've never had a second of downtime in a little over two years as a customer |
19:56.32 | DagMoller | hi... is possible to utilize 2 TDM04B in one machine? |
19:56.39 | austinnichols101 | [av]bani: no, but I get a few hundred a day... |
19:56.43 | [av]bani | _Sam--: 5 so far |
19:56.46 | denon | austinnichols101: but with regards to their latency and peering |
19:56.46 | AndyCap | Mother: and what's the 3gpp's policy on patents in the standards? some allow it and others don't |
19:57.03 | _Sam-- | there is a note in older firmware about it not working right |
19:57.09 | _Sam-- | and apparently its still not fixed |
19:57.16 | [av]bani | austinnichols101: its possible > 5 makes it break. shrug.. |
19:57.24 | [av]bani | s/austinnichols101/_Sam--/ |
19:57.46 | _Sam-- | no, i got it to break with 3 or 4 |
19:57.54 | _Sam-- | and the hint extensions werent even gxps |
19:58.02 | _Sam-- | this is part of it: |
19:58.04 | _Sam-- | "NOTE: (Nov29/05) When receiving the "''Incoming call: Got SIP response 415 "Unacceptable Content-Type" error, the BLF doesn't seem to work - looks like the phones are refusing the BLF messages all together. - conexim |
19:58.05 | _Sam-- | " |
19:58.36 | _Sam-- | i got those all over the console before things really start to break |
19:58.40 | austinnichols101 | denon: check out interpulse.net and set focus 'from internap / to internap' |
19:58.49 | Mother | AndyCap: I see your point, maybe it's true that they have nominated VoiceAge, Nokia and Ericsson as the "administrators" of the patent rights |
19:58.50 | AndyCap | Mother: of course they're not saying outright what patents you would be paying for. :-P http://www.voiceage.com/amr_faqs.php scum |
19:59.05 | denon | austinnichols101: well, that doesnt really tell us jitter over time |
19:59.09 | Mother | hehe indeed |
19:59.24 | austinnichols101 | denon: I'm getting that now... |
19:59.25 | AndyCap | Mother: and it seems you only need to license it for "non-wireless" use |
19:59.36 | [av]bani | austinnichols101: http://www.spamhaus.org/sbl/sbl.lasso?query=SBL14726 |
19:59.48 | [av]bani | austinnichols101: http://www.spamhaus.org/sbl/listings.lasso?isp=internap.com 23 active listings |
19:59.56 | [av]bani | austinnichols101: including some still active from 2004 ... |
19:59.59 | DagMoller | hi... is possible to utilize 2 TDM04B in one machine? |
20:00.30 | [TK]D-Fender | DagMoller : yup |
20:00.43 | DagMoller | [TK]D-Fender, ??? |
20:00.54 | austinnichols101 | [av]bani: can't speak to the spam issue other than they've been very responsive when we've complained to them |
20:00.57 | AndyCap | Mother: nah, misinterpreted that. If you want wireless use you have to beg the competition for a license (both ericsson and nokia) |
20:01.16 | [av]bani | austinnichols101: if internap was indeed intolerant of spam, that list would be empty. not stuffed with active listings up to 2 years old |
20:01.18 | austinnichols101 | [av]bani: but thanks for the link - hadn't seen that one before |
20:01.41 | Mother | AndyCap: yep...however a curious point is that it says you need a license if you will use it one product only...how about many products...it's a bit unclear |
20:01.45 | [av]bani | austinnichols101: you might want to ask them why they host and protect so many spammers. they'll pribably ignore you though |
20:01.53 | Mother | sorry, not product, application |
20:02.05 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
20:02.15 | DagMoller | [TK]D-Fender, yes or no??? |
20:02.58 | AndyCap | Mother: so in essence they sell you a pig in a poke. |
20:03.01 | AndyCap | :P |
20:03.23 | trixter | better a pig in a poke than to poke a pig |
20:03.27 | trixter | cause no one likes a pig poker |
20:03.35 | [TK]D-Fender | DagMoller : I said yes already.... |
20:03.43 | Seldon1975 | i do |
20:03.52 | DagMoller | [TK]D-Fender, no, you say 'yup' |
20:04.08 | [av]bani | austinnichols101: do let me know their response. i'll be interested to know it. |
20:04.08 | Mother | AndyCap: LOL yes |
20:04.14 | DagMoller | [TK]D-Fender, thanks... |
20:04.24 | Mother | well, gotta get going, cheers all |
20:04.39 | *** part/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com) |
20:04.46 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
20:05.06 | AndyCap | I can't believe that the actual patents should be so hard to find |
20:06.05 | PakiPenguin_ | evening |
20:06.18 | iCEBrkr | PakiPenguin_: G'afternoon |
20:06.21 | iCEBrkr | :) |
20:07.33 | austinnichols101 | denon: Jan 2006 Stats: N. America Latency 37.3143ms Packet Loss 0.0359% Network Availability 100% Jitter 0.01164ms |
20:08.25 | austinnichols101 | denon: N. America-Europe 70.3305ms 0.0256% 100% 0.01206ms for the same time period |
20:08.35 | Hmmhesays | i'm sorry i'm bad, i'm sorry i'm blue, i'm sorry about all the things I said to you |
20:08.45 | lazzarello | I recently heard rumors about ISPs in the states introducing jitter into consumer broadband connections. anyone else hear the same? |
20:09.20 | Hmmhesays | that would be a good way to piss a lot of people off |
20:09.28 | lazzarello | sure would |
20:09.38 | _Sam-- | it would only piss them off if they knew about it. |
20:09.43 | PakiPenguin_ | hey iCEBrkr |
20:09.46 | Hmmhesays | jitter while i'm playing quake would piss me off |
20:10.03 | lazzarello | or convince them to "use cablevision's digital phone service" because it's "better". |
20:10.06 | Abydos313 | why? is regular phones being discarded for voip too quickly ? |
20:10.17 | _Sam-- | lazzarello: i heard the same rumors about comcast.net |
20:10.29 | _Sam-- | and that they messed with voip traffic...i think its just a rumor personally |
20:10.37 | iCEBrkr | Mangle QoS settings mid-stream.. Man, that'd piss me off |
20:11.13 | _Sam-- | i wouldnt be terribly surprised if they did things that did interfere with voip traffic to make their own services seem better |
20:11.15 | austinnichols101 | [av]bani: just sent support an inquiry regarding the spam listing. Let see what they say... |
20:11.18 | lazzarello | verizon's FTTP service blocks incoming TCP traffic on port 80. they don't advertise /that/ anywhere. |
20:11.38 | _Sam-- | lazzarello : yes they do, they specifically tell you you cant run servers |
20:11.44 | _Sam-- | usually, anythig that runs on port 80 is a server |
20:12.03 | iCEBrkr | Brighthouse/RoadRunner doesn't block any inbound ports. |
20:12.04 | _Sam-- | i just read the FIOS terms of service |
20:12.06 | _Sam-- | it says right there |
20:12.32 | *** join/#asterisk lalito (n=erg@201.154.202.110) |
20:13.10 | lazzarello | ooooh, watch out for the "servers"! |
20:13.13 | _Sam-- | erisk ("..quit..") |
20:13.17 | _Sam-- | Verizon FiOS Internet Service consumer packages include 10 MB of personal Web space. The consumer offers do not permit customers to host any type of server, personal or commercial. |
20:13.25 | Seldon1975 | if the voicemail.conf entry format is: 1234 =>{password},Some User,email@address.com how can I let users change their own passowrd? |
20:13.28 | *** join/#asterisk saftsack (n=saftsack@p54A7E210.dip.t-dialin.net) |
20:13.47 | _Sam-- | you are the one saying they dont tell you that... |
20:13.49 | lazzarello | sweet. so they give you 1.5 megs inbound and block inbound traffic! |
20:13.52 | _Sam-- | but in fact they do, that is my only point |
20:13.57 | iCEBrkr | Seldon1975: phpsuexec + PHP and a voicemail.conf class :P |
20:13.59 | _Sam-- | you have to pick your argument |
20:14.05 | _Sam-- | if you are arguing they dont tell, they do. |
20:14.11 | _Sam-- | if you are complaining that it sucks, that is different |
20:14.20 | lazzarello | I didn't read the fine print. |
20:14.27 | Seldon1975 | iCEBrkr: no, really |
20:14.31 | PaulHuynh | hello |
20:14.35 | austinnichols101 | [av]bani: the first link you sent me (rokso / datran media) is in the same CoLo facility as we are (http://www.napoftheamericas.net) |
20:14.43 | iCEBrkr | Seldon1975: Really. |
20:14.49 | PaulHuynh | where is the custom voice promt located on the asterisk server? |
20:14.59 | iCEBrkr | Seldon1975: Actually, doesn't Asterisk allow users to dial in and change their passwords? |
20:15.12 | Seldon1975 | iCEBrkr: I hope so |
20:15.13 | iCEBrkr | PakiPenguin_: Custom voice prompt?? |
20:15.15 | [av]bani | austinnichols101: nice |
20:15.15 | Seldon1975 | iCEBrkr: but how? |
20:15.24 | iCEBrkr | Seldon1975: VoiceMailMain()? |
20:15.33 | PakiPenguin_ | no |
20:15.37 | Seldon1975 | users have said they've tried that and the password didnt change |
20:15.44 | Seldon1975 | iCEBrkr: users have said they've tried that and the password didnt change |
20:15.52 | GerbilWrk | anyone know what could cause this to start coming up, Feb 20 14:15:14 NOTICE[1567]: chan_zap.c:8203 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
20:16.01 | iCEBrkr | Seldon1975: File permissions are different from user:group Asterisk is running as? |
20:16.02 | austinnichols101 | [av]bani: yeah - I wonder where their cabinet is. I might be able to solve this problem myself :-) |
20:16.13 | iCEBrkr | PaulHuynh: Huh? Voice prompt? |
20:17.10 | Seldon1975 | iCEBrkr: hmm, ill have a look' |
20:18.03 | Seldon1975 | iCEBrkr: thx |
20:19.28 | *** join/#asterisk breakdecks (n=breakdec@adsl-065-006-209-021.sip.mem.bellsouth.net) |
20:19.30 | [av]bani | austinnichols101: i'd be interested in hearing internap's response. $50 says it's a non-response |
20:19.44 | [av]bani | austinnichols101: eg a response with no content, just doublespeak |
20:20.33 | breakdecks | I have a DVG-1120S VoIP gateway, but when I dial digits in a menu, I get the error 'codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?' |
20:20.36 | breakdecks | what do I do? |
20:21.16 | |omni| | I think I'm going to write a windows based asterisk manager interface |
20:21.27 | iCEBrkr | |omni|: Yea, we all say that |
20:21.43 | austinnichols101 | [av]bani: $50 not necessary - I'll do it just out of pure curiosity. And I'll definitely share the results, doublespeak or no. |
20:21.53 | |omni| | was just looking at what's available and I've got some things I need handled |
20:22.03 | iCEBrkr | |omni|: me too |
20:22.14 | iCEBrkr | |omni|: Personally all the desktop call managers out there suck ass |
20:22.17 | |omni| | just did a system for a real-estate office and could really do some cool things with the manager interface |
20:22.20 | |omni| | yea they do, so far |
20:22.40 | |omni| | did you write the callerid popup? |
20:22.40 | iCEBrkr | |omni|: One of the many. |
20:22.46 | |omni| | ya, just looked at yours a bit ago |
20:22.58 | iCEBrkr | Very basic.. Kind of a trial I guess.. |
20:23.00 | rayvd | I am not a midget! |
20:23.12 | iCEBrkr | |omni|: I have bigger/better plans. |
20:23.20 | |omni| | cool |
20:23.27 | [av]bani | breakdecks: dvg-1120s doesnt support ilbc |
20:23.47 | iCEBrkr | My callerID thing is pretty shitty itself :) But like I said, it was just a start and of course I wanted to see what kind of interest was out there |
20:23.53 | |omni| | I was just messing with quick and dirty commands via a little Delphi app...can do some neat things with the manager interface |
20:24.02 | |omni| | ya definitely |
20:24.03 | iCEBrkr | |omni|: yup |
20:24.34 | iCEBrkr | |omni|: I'm not sure how true it is today (v1.2.x), but there were some scalability issues with the manager port/interface |
20:24.35 | Nivex | I'm trying to figure out the best way to have astersk query an HTTP server for a call URL and then Dial it. Any pointers on where to start? |
20:24.52 | iCEBrkr | Nivex: CURL()? |
20:24.56 | |omni| | scalability issues using a proxy or no? or either? |
20:25.10 | iCEBrkr | |omni|: Without a proxy and having all the clients connect directly to Asterisk |
20:25.13 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
20:25.35 | |omni| | hmm..should play with that on a test box |
20:25.41 | |omni| | would be interesting to see how much better it does with a proxy |
20:25.46 | |omni| | if at all |
20:25.50 | |omni| | still has to pass out a lot of info |
20:26.04 | iCEBrkr | |omni|: I think it'd do better for the machine in general as there's less socket management |
20:26.06 | austinnichols101 | [av]bani: sending you a copy of the message in a separate window |
20:26.14 | |omni| | could be |
20:26.19 | mocker | Is there a variable for the dialed number? |
20:26.19 | Tagor | Hmm, I give it another try; I have a problem with incoming calls. Outgoing calls work fine. But for some reason I don't get any signal from my voip provider. When I try tcpdump or sip debug, I just don't see anything when I try to call my number. Anyone an idea how to get more information about why it isn't working? |
20:26.32 | |omni| | mocker ${EXTEN} |
20:26.44 | iCEBrkr | |omni|: The real key factor in the proxy is mangling the data to/from the client. So you can do some extra custom stuff. |
20:26.45 | Nivex | ~iCEBrkr++ |
20:26.56 | mocker | |omni|: Thanks, I'll try that. |
20:27.04 | iCEBrkr | Nivex: Think that'll work for ya? |
20:27.13 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
20:27.24 | iCEBrkr | mocker: ${EXTEN} isn't always the dialout number |
20:27.33 | |omni| | yea that might be cool... actually with a proxy you could use TCP between proxy <-> asterisk and then make it a bit lighter from the proxy -> client using UDP messages..almost like an IM app |
20:27.44 | |omni| | at least for some things. |
20:28.03 | Nivex | iCEBrkr: should be just what the doctor ordered. |
20:28.04 | iCEBrkr | |omni|: Yea, use the proxy to actually process the data, and then spew UDP to the client |
20:28.10 | iCEBrkr | Nivex: cool stuff |
20:28.27 | mocker | |omni|: Hrm, that just says 's'. What I'm doing is dialing out w/ the drop folder, and I'm using a system command to send an email of what number was dialed. |
20:28.41 | iCEBrkr | mocker: You're using a call file? |
20:28.48 | Nivex | basically I'm going to write a script on the server that will return either a call URL or UNAVAIL and then the Asterisk dialplan can react accordingly |
20:28.51 | iCEBrkr | mocker: Then most likely you won't be able to get the dialed number. |
20:29.15 | |omni| | are you using an AGI to create the call file ? |
20:29.31 | |omni| | you could pass the ${EXTEN} to your AGI |
20:29.43 | mocker | iCEBrkr: Yeah, I'm using a call file. |
20:29.47 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
20:29.49 | iCEBrkr | |omni|: ${EXTEN} doesn't mean the dialed number |
20:29.57 | harryvv | moker, what are using the call file for |
20:30.10 | Nivex | iCEBrkr: I'm reading the func_curl page on the wiki, but I don't see anything about handling error conditions. |
20:30.12 | |omni| | within your extensions it will return the dialed number, no? |
20:30.23 | mocker | harryvv: Basically call someone, when they press 2 to acknowledge, send an email saying they acknowledged. |
20:30.24 | iCEBrkr | ...and since ${DIALEDPEERNUMBER} is broken along with ${DNIS} |
20:30.42 | mocker | But trying to get who it called into the email is my roadblock. ;) |
20:30.45 | harryvv | sending a email from whom? the called party? |
20:30.48 | iCEBrkr | mocker: Your best bet is to: SetVar: _dialednumber='123-123-1234' in your callfile.. |
20:31.02 | mocker | iCEBrkr: Oh, well that would solve everything. |
20:31.06 | iCEBrkr | |omni|: It returns the dialed 'extension' which isn't always the number you've called. |
20:31.10 | mocker | I didn't know I could set variables like that. :) |
20:31.18 | |omni| | okay well..fair enough |
20:31.29 | |omni| | I guess it returns in the dialed number in my implementation |
20:32.14 | iCEBrkr | |omni|: if NXXNXXXXXXX is your extension, sure.. |
20:32.30 | |omni| | it will be for certain contexts |
20:32.34 | iCEBrkr | |omni|: if you land in 's' then ${EXTEN} is going to equal 's' |
20:32.35 | |omni| | and I can grab and stuff into a var at that point |
20:32.54 | |omni| | I see what you're saying |
20:32.57 | iCEBrkr | mocker: I'm having the same problems now, but I'm too far along in the project to start making changes. |
20:33.05 | *** join/#asterisk Skid (i=cm@unaffiliated/skid) |
20:33.11 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
20:34.03 | fjean | hey guys, how are you.. |
20:34.06 | |omni| | in most contexts I match the incoming (or outgoing) exten with the full number just so I know what's going on..at least the first time it hits before jumping to another context |
20:34.31 | iCEBrkr | |omni|: then ${EXTEN} is gonna change throughout the dialplan :) |
20:34.53 | harryvv | mocker, how many calls do u want your astrisk box to make at one time/ |
20:34.54 | harryvv | ? |
20:35.00 | |omni| | right, but it will be the dialed number the first time it's grabbed in any matching context |
20:35.08 | fjean | -- can someone help me with an AGI script that dials a string ? |
20:35.15 | iCEBrkr | |omni|: sure, depending on your pattern matching and what-not |
20:35.22 | |omni| | guess it really depends on how you organize everything.. mine's pretty simple. |
20:35.37 | mocker | iCEBrkr: That worked like a charm. |
20:35.51 | mocker | iCEBrkr: Any way to get it to not have the ' ' around the variable output? |
20:36.01 | iCEBrkr | mocker: Then you just NoOp(${dialednumber})? |
20:36.04 | mocker | Oh, that's in what I'm setting. |
20:36.29 | iCEBrkr | Yeah, it's habit to quote values.. |
20:36.45 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
20:38.14 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
20:38.42 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
20:38.43 | fjean | i have this in an AGI script: $dialstr = "IAX2/okscm/559132316200|60"; which does not give me ring back, but this line will give it to me: exten => _0XX32316200,1,Dial(IAX2/okscm/559132316200,60); |
20:39.02 | fjean | -- anybody knows why i would get it in the AGI ? |
20:39.04 | asterisk99 | anyone here have Asterisk running on UBUNTU? |
20:40.12 | asterisk99 | iefbr14 |
20:40.35 | lazzarello | asterisk99, what's your question? |
20:41.17 | asterisk99 | lazzarello: Did you have any problems with the zaptel modules on ubuntu? |
20:41.30 | PaulHuynh | help |
20:41.34 | PaulHuynh | i got this error |
20:41.35 | PaulHuynh | No such file or directory in /var/www/html/admin/ivr.php on line 112 |
20:41.53 | lazzarello | I don't run ubuntu but that's should make much difference. did you compile yourself? |
20:42.03 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
20:42.11 | asterisk99 | lazzarello: I did the make clean; make linux26; make install .... but when I try modprobing 'em, I get errors |
20:42.18 | *** join/#asterisk Inkubot (n=inkubot@200.74.186.41) |
20:42.22 | Inkubot | hi |
20:42.28 | lazzarello | asterisk99, what kind of errors? |
20:42.50 | Inkubot | i've got a Asterisk Server behind NAT, working with sip.. i can't register from the internet.. i've got something like this |
20:43.02 | Inkubot | Feb 20 16:01:55 DEBUG[2090] chan_sip.c: Target address 200.74.186.41 is not local, substituting externip |
20:43.03 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
20:43.19 | Inkubot | i set externip and localnet y sip.conf |
20:43.22 | Inkubot | also nat=yes |
20:43.49 | Inkubot | any idea on what could be the problem ? |
20:43.52 | Hmmhesays | yeah good luck with that |
20:43.56 | Seldon1975 | when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist? |
20:44.07 | lazzarello | the fact that SIP behind NAT is /HELL/ could be the problem. |
20:44.20 | Inkubot | haha |
20:44.23 | Inkubot | damn |
20:44.35 | lazzarello | Inkubot, install ngrep and start capturing headers for a ingle faild session. read the headers and figure out your problem. |
20:45.22 | dudes | asterisk99 - did you modprobe zaptel first? if not, try that, otherwise, create "misc" in /lib/modules/(your kernel)/ |
20:45.31 | austinnichols101 | lnkubot: what do you have set on the remote phone? |
20:45.38 | dudes | then goto zaptel ... make clean && make linux26 && make install |
20:45.40 | asterisk99 | lazzarello: modprove wctdm gets me FATAL ERROR: module wctdm not found |
20:45.59 | Inkubot | the ip address of the Router->Asterisk Server |
20:46.02 | lazzarello | he he. I guess the module isn't found. what's modinfo wctdm output? |
20:46.04 | Inkubot | public ip address |
20:46.26 | dudes | Debian/Ubuntu do this something (something to do with Udev) |
20:46.27 | austinnichols101 | lnkubot: is the remote side behind nat too? |
20:46.41 | asterisk99 | lazzarello: modprobe zaptel ... same thing |
20:46.43 | lazzarello | dudes, Debian and ubuntu are not the same thing. |
20:46.54 | Inkubot | austinnichols101 yes.. i forget that |
20:47.00 | *** part/#asterisk PaulHuynh (n=paulhuyn@c-68-37-18-82.hsd1.de.comcast.net) |
20:47.06 | asterisk99 | lazzarello: isn't ubunu a derivative of debian? |
20:47.10 | lazzarello | asterisk99, I'm running Debian stable. when I run 'modinfo wctdm' it outputs this: |
20:47.10 | Inkubot | let me see if NAT is activated in the client configuration |
20:47.11 | Seldon1975 | when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist? |
20:47.12 | lazzarello | lummox:~# modinfo wctdm |
20:47.12 | lazzarello | filename: /lib/modules/2.6.8-2-386/extra/wctdm.ko |
20:47.26 | lazzarello | asterisk99, what does your system say? |
20:47.28 | asterisk99 | lazzarello: 1 sec |
20:47.31 | austinnichols101 | lnkubot: can you move the remote phone into the DMZ on the remote router for testing? |
20:47.37 | dudes | lazzarello - DUDE, I use Debian and Ubuntu and get this issue every once in awhile |
20:47.39 | lazzarello | asterisk99, don't paste the whole thing. |
20:47.50 | *** join/#asterisk bronze (n=clark@c-24-91-157-212.hsd1.ma.comcast.net) |
20:47.58 | asterisk99 | lazzarello: not found |
20:48.01 | lazzarello | dudes, NOT THE SAME! you run testing or unstable then. |
20:48.02 | justinu | d00d |
20:48.17 | lazzarello | asterisk99, then you didn't install the modules for your running kernel. what's in /lib/modules? |
20:48.17 | Inkubot | austinnichols101 i can't.. i'm in another place.. |
20:48.21 | *** join/#asterisk areski (n=areski@162.Red-83-34-10.dynamicIP.rima-tde.net) |
20:48.39 | dudes | lazzarello - I never said they were, they same ... They still are similiar though. |
20:48.39 | austinnichols101 | lnkubot: what type of device are you trying to register? |
20:48.47 | lazzarello | asterisk99, does ubuntu have a package for all the asterisk 1.2 stuff. |
20:48.48 | asterisk99 | lazzarello: (perosnally, I prefer gentoo, but it doesn't like my NIC) |
20:48.49 | Inkubot | yeaaaaaaaa |
20:48.53 | Inkubot | austinnichols101 |
20:48.59 | Inkubot | i forgot nat=yes in the clients :D |
20:49.03 | dudes | asterisk99 - I told you how to fix it |
20:49.05 | asterisk99 | lazzarello: no package |
20:49.14 | lazzarello | is there a backport of some kind? |
20:49.19 | lazzarello | cause debian stable has a package for 1.2.1 |
20:49.39 | dudes | It is Rocket Science to fix ... which I have already mentioned |
20:49.46 | lazzarello | in fact, I just finished building the zaptel-modules for the 386 and 686 kernel in sarge. |
20:50.08 | austinnichols101 | lnkubot: now let the clients idle for a couple of minutes and check 'sip show peers' on the server and verify that they maintain the registration (I'm assuming that you have qualify = yes on the server) |
20:50.11 | asterisk99 | lazzarello: I wonder if I should go with debian |
20:50.16 | lazzarello | I wonder... |
20:50.26 | asterisk99 | lazzarello: I ain;t married to ubuntu |
20:50.26 | lazzarello | oh yeah, and the base system is only 140 megs |
20:50.33 | lazzarello | that's pretty cool too |
20:50.36 | dudes | I've had this issue on Stable, testing, Unstable, ubuntu, and kubuntu ... |
20:50.41 | lazzarello | ubuntu is like, 1.5 gigs, right? |
20:50.51 | asterisk99 | dudes: I will look in messages above for your fix |
20:51.08 | lazzarello | asterisk99, you didn't tell me what's in /lib/modules |
20:51.10 | asterisk99 | lazzarello: You can install a 'server' version of ubuntu |
20:51.29 | asterisk99 | lazzarello: Hang on :) I have 1 monitor on 2 machines |
20:51.36 | asterisk99 | lazzarello: Cheeeeeep! |
20:52.20 | asterisk99 | lazzarello: /lib/modules has 2.6.12-10-386 directory |
20:52.27 | asterisk99 | lazzarello: plus a few more |
20:52.32 | lazzarello | okay! what's the few more? |
20:52.47 | Seldon1975 | when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist? |
20:52.51 | *** join/#asterisk Inkubot (n=inkubot@200.74.183.55) |
20:52.55 | Inkubot | i kick the adsl.. |
20:52.56 | Inkubot | :D |
20:53.01 | Inkubot | austinnichols101 everything works now |
20:53.18 | lunaphyte | why might asterisk segfault when a skinny phone answers a call from a sip phone? |
20:53.29 | asterisk99 | lazzarello: 2.6.12 , 2.6.12-10-386 , 2.6.12-10-686 , and 2.6.12-9-386 |
20:53.31 | austinnichols101 | lnkubot: party on! |
20:53.51 | lazzarello | asterisk99, now what does the output of 'uname -a' say? |
20:55.10 | asterisk99 | lazzarello: Linux mymachine 2.6.12-10-386 # date time ... |
20:55.28 | *** join/#asterisk pengyong (n=lala@218.19.188.13) |
20:55.48 | asterisk99 | lazzarello: mymachine is my name of the kernel |
20:55.51 | dudes | mkdir /lib/modules/`uname -r`/misc && cd /usr/src/zaptel && make clean && make linux26 && make install |
20:55.58 | lazzarello | asterisk99, now... 'find /lib/modules -name 'wctdm.ko'' |
20:56.21 | lazzarello | get rid of the first set of quotes. |
20:56.29 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
20:56.38 | hensema | hi, is "spawn extension (...) exited non-zero on 'channel name'" a problem, or do all dialplans end that way? |
20:57.17 | dudes | once complete ... cp *.ko /lib/modules/`uname -r`/misc && make clean && make linux26 && make install && modprobe zaptel && modprobe wctdm |
20:57.34 | lazzarello | dudes, that's what the makefile is for |
20:57.39 | fjean | is there a difference between a Dial from extensions.conf and a Dial from an AGI, should we do answer() before ? |
20:57.44 | asterisk99 | lazzarello: /lib/modules/2.6.12/misc/wctdm.ko |
20:58.01 | Qwell[] | asterisk99: debian? |
20:58.08 | lazzarello | asterisk99, there you go. that's your problem. your running kernel is different then where the modules were installed. |
20:58.10 | dudes | lazzarello - I think I know how to resolve this issue, but since you're the smart guy |
20:58.15 | lazzarello | reboot into the 2.6.12 kernel. |
20:58.17 | asterisk99 | lazzarello: aye carumba |
20:58.21 | Qwell[] | no, no, no |
20:58.34 | Qwell[] | You just need to tell the Makefile where the modules should go |
20:58.39 | asterisk99 | lazzarello: I didn't compile the kernel |
20:58.56 | lazzarello | dudes, it's not resolved if you have to type fancy shell scripts. the installer package should do that automatically or it's a bug. |
20:59.13 | lazzarello | asterisk99, then why do you have a module directory for 2.6.12? |
20:59.21 | dudes | lazzarello - it's once they haven't fixed in a very long time |
20:59.28 | Qwell[] | KVERS=`uname -r` make install |
20:59.32 | Qwell[] | simple as that.. |
20:59.32 | asterisk99 | lazzarello: I ast-install'ed the kernel-source |
20:59.47 | lazzarello | then reboot into that kernel and your problem's solved. |
20:59.49 | dudes | asterisk99 - do as I said and see the magic |
20:59.51 | Qwell[] | Debian defs KVERS |
20:59.57 | Qwell[] | lazzarello: That kernel doesn't exist |
21:00.14 | Qwell[] | My solution will work, or your money back - guaranteed. ;] |
21:00.19 | asterisk99 | lazzarello: I think it's because the kernel comes precompiled |
21:00.29 | asterisk99 | lazzarello: So the trick is to get the right source |
21:00.34 | Qwell[] | ho hum |
21:00.34 | lazzarello | I don't compile kernels either. something really broken with ubuntu and zaptel... |
21:00.39 | Qwell[] | I talk, but nobody listens... |
21:00.50 | Qwell[] | ifndef KVERS |
21:00.50 | Qwell[] | KVERS:=$(shell uname -r) |
21:00.50 | Qwell[] | endif |
21:00.56 | asterisk99 | Qwell[]: sorry ... Too many messages ... let me catch up |
21:00.57 | Qwell[] | Like I said...heh |
21:01.03 | Qwell[] | Debian does funkiness with KVERS |
21:01.08 | lazzarello | if there's a bug somewhere it should be fixed, not just hacked around with some shell scripting via IRC. |
21:01.13 | dudes | Qwell[] - I said my solution ... But since this guy doesn't seem to want to listen to me |
21:01.24 | Qwell[] | lazzarello: Go talk to the debian guys about it. It's their problem |
21:01.35 | lazzarello | Qwell, I just installed zaptel on debian stable and didn't have that problem. |
21:01.37 | *** join/#asterisk cosa (n=tmalkut@fw.orasoft.net.pl) |
21:01.44 | lazzarello | so maybe not? |
21:01.46 | asterisk99 | Qwell[]: do I put KVERS:=$(shell uname -r) in Make? |
21:01.51 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
21:01.51 | asterisk99 | Qwell[]: do I put KVERS:=$(shell uname -r) in Makefile? |
21:01.52 | Qwell[] | asterisk99: no |
21:01.58 | Qwell[] | KVERS=`uname -r` make install |
21:02.03 | asterisk99 | Qwell[]: that |
21:02.09 | asterisk99 | that's a command??? |
21:02.13 | dudes | if you ran the two commands above it'd resolve it too |
21:02.13 | Qwell[] | debian won't def KVERS if it's already def'd - just as zaptel won't |
21:02.23 | Qwell[] | so, if you explicitly def it...neither will chang eit |
21:02.40 | shmaltz | anybody else having trouble with the list? or is it G again? |
21:03.05 | Qwell[] | dudes: Your way will work, but it isn't the "proper" way. The stuff would be compiled against 2.6.x, and not 2.6.x-blah |
21:03.22 | asterisk99 | Qwell[]: bash hates KVERS |
21:03.31 | Qwell[] | asterisk99: type it exactly as I did |
21:03.32 | asterisk99 | Qwell[]: command not found |
21:03.33 | lazzarello | Qwell[], or you can download zaptel-source and use module-assistant. |
21:03.35 | Qwell[] | KVERS=`uname -r` make install |
21:03.41 | lazzarello | but I don't know if ubuntu has modass |
21:03.45 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
21:03.56 | asterisk99 | Qwell[]: aha!!! no spaces!!!! |
21:04.05 | dudes | Qwell - Debian makes a symlink from the sources to the dir anyway, so it really doesn't matter |
21:04.15 | Qwell[] | dudes: yeah, but the modules |
21:04.21 | Qwell[] | it's just...funky how they do it |
21:04.38 | Qwell[] | hensema: All the time |
21:04.42 | Seldon1975 | when users change their VM password using Comedian Mail's menu option it does change it successfully but when Asterisk is restarted their password reverts to the value in voicemail.conf - how can I allow these changes to persist? |
21:04.48 | hensema | Qwell[]: is it a problem? |
21:04.55 | Qwell[] | Seldon1975: allow writes to voicemail.conf |
21:05.02 | Qwell[] | I believe there is an option at the top |
21:05.14 | Qwell[] | hensema: don't think so |
21:05.16 | asterisk99 | Qwell[]: after KVERS= shouldn't I do a make clean; make linux26; make install ??? |
21:05.18 | *** join/#asterisk Speeder (n=root@a81-84-241-120.cpe.netcabo.pt) |
21:05.25 | Qwell[] | asterisk99: no need |
21:05.38 | Qwell[] | you can do a make clean first, if you'd like though |
21:05.39 | asterisk99 | Qwell[]: just make install ? |
21:05.42 | Qwell[] | linux26 is detected |
21:05.50 | Qwell[] | no...type it EXACTLY as I did...one line |
21:05.56 | lazzarello | asterisk99, what does apt-cache search module-assistant output? |
21:06.28 | Speeder | Hi. I'm having a problem with asterisk & zaphfc. For outgoing call's it appears can't create channel zap, for incomming call's i'm getting busy tone |
21:06.32 | asterisk99 | lazzarello: 1 min please |
21:07.40 | asterisk99 | Qwell[]: still modules not found |
21:07.57 | Qwell[] | pastebin the output |
21:08.02 | Qwell[] | the full output - including the command you typed |
21:08.21 | Qwell[] | you probably have to mkdir the misc dir. |
21:09.01 | asterisk99 | lazzarello: apt-cache search module-assistant no output |
21:09.04 | Seldon1975 | Qwell: thx |
21:09.21 | *** join/#asterisk NotFreak (n=extmail@cp12193-e.tilbu1.nb.home.nl) |
21:09.26 | asterisk99 | Qwell[]: paste from what to what? |
21:09.36 | asterisk99 | Qwell[]: the whole make output? |
21:09.42 | Qwell[] | yes |
21:09.48 | lazzarello | asterisk99, switch to sarge. all this stuff is apready in packages and installable through apt. |
21:10.02 | Qwell[] | I'll actually save you the trouble though |
21:10.13 | Qwell[] | mkdir /lib/modules/`uname -r`/misc |
21:10.13 | asterisk99 | Qwell[]: sarge is better? |
21:10.20 | [av]bani | ahahaha. just got pictures of a customer computer |
21:10.26 | [av]bani | they're a cigar smoker. |
21:10.28 | Qwell[] | do that, then do the make install again (with the KVERS) |
21:10.32 | [av]bani | it is N A S T Y |
21:10.52 | Qwell[] | there is more funkiness with install -D that I've seen |
21:12.17 | NotFreak | is there some one around here that has some knowledge about setting and reading registers on the X100P card? |
21:13.03 | Seldon1975 | Qwell: it doesn't look as if there is a setting in Voicemail.conf to allow writes - do you know where you saw such a thing? |
21:13.22 | [av]bani | ew |
21:13.25 | asterisk99 | Qwell[]: is that mkdir work? it gives me an error .... it thinks uname -r is part of the name |
21:13.31 | [av]bani | _Sam--: OMG ILLEGAL |
21:13.33 | [av]bani | _Sam--: :) |
21:13.40 | _Sam-- | lol |
21:13.43 | Qwell[] | no, I don't recall...I'm probably thinking extensions.conf |
21:13.48 | Qwell[] | asterisk99: with the backticks? |
21:13.48 | Qwell[] | ` |
21:13.50 | _Sam-- | living close to the canadia border must have its benefits |
21:13.56 | _Sam-- | s/canadia/canadian/ |
21:13.57 | iCEBrkr | _Sam--: Man, did you hear me?! It runs! It Runs! |
21:14.00 | iCEBrkr | :) |
21:14.05 | Seldon1975 | Qwell: I think so |
21:14.06 | [av]bani | _Sam--: there's probably an entire fidel castro inside this PC |
21:14.12 | _Sam-- | lol! |
21:14.18 | _Sam-- | iCEBrkr : did you ride the thing? |
21:14.20 | Qwell[] | Seldon1975: It should "jsut work". Do you get any errors/warnings? |
21:14.24 | iCEBrkr | _Sam--: Yea, Saturday night |
21:14.55 | asterisk99 | Qwell[]: no have backtick on tis ^%$^% keyboard |
21:15.01 | Qwell[] | umm |
21:15.05 | Qwell[] | okay, do this |
21:15.06 | _Sam-- | iCEBrkr : so what are you buying next for it? :) |
21:15.15 | Qwell[] | mkdir /lib/modules/$(uname -r)/misc |
21:15.23 | asterisk99 | Qwell[]: how about I just do uname -r and paste result in the dir name |
21:15.27 | iCEBrkr | _Sam--: Clutch does this weird thumping thing when it's in neutral and I have it all the way out.. I think it just needs some adjusting. |
21:15.30 | Qwell[] | that would work too, yeah |
21:15.35 | Qwell[] | the second command should also work though |
21:15.42 | iCEBrkr | _Sam--: Light smoked windscreen :P |
21:15.59 | file | how personal |
21:16.20 | [av]bani | _Sam--: http://bani.anime.net/cigar_smoker/ |
21:16.21 | Qwell[] | too much rockstar...heart is racing...ugh |
21:16.54 | _Sam-- | what, ALL your pc's dont look like that? |
21:17.07 | asterisk99 | Qwell[]: Still didn;t work |
21:17.14 | shmaltz | Interesting: |
21:17.16 | shmaltz | http://news.yahoo.com/s/cmp/20060218/tc_cmp/180203910;_ylt=AgRQNsJD3E9GRhi7Elz.3pCor7oF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
21:17.21 | Qwell[] | asterisk99: if backtick doesn't work on that keyboard...how are you setting KVERS? |
21:17.33 | Qwell[] | that also needs backticks |
21:17.38 | |omni| | heh..that's awesome... I've been without a bike forever now |
21:17.38 | _Sam-- | [av]bani bow old is that pc? |
21:18.14 | Seldon1975 | Qwell: no errors or warnings; as I say the password change does work,.. but only until Asterisk is restarted |
21:18.17 | [av]bani | _Sam--: dunno, a year maybe |
21:18.29 | [av]bani | its on a table across the room and i can smell it from here |
21:18.31 | Qwell[] | Seldon1975: Yeah, it should be saving... Does * run as root? |
21:18.41 | _Sam-- | i had one that looked like that after about 5 years of being in an office with 6 cigarrette smokers |
21:18.53 | asterisk99 | Qwell[]: nopers |
21:19.04 | [av]bani | if thats what ends up in a pc, think of what goes in your lungs |
21:19.05 | Seldon1975 | Qwell: yyes |
21:19.08 | Qwell[] | asterisk99: see above |
21:19.29 | _Sam-- | my lungs have other things to worry about :) |
21:19.32 | asterisk99 | Qwell[]: I used a regular single apostophie ' |
21:19.37 | Qwell[] | backticks :P |
21:19.44 | *** part/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
21:19.45 | Qwell[] | I said *exactly* as I typed it, heh |
21:20.05 | asterisk99 | Qwell[]: OK - so I can;t find backtick on this keyboard |
21:20.10 | Qwell[] | replace the backticks with the output of uname -r, or $(uname -r) |
21:20.52 | iCEBrkr | asterisk99: backtick is to the left of the number 1 key. below ESC and above TAB.. Well, typically, anyhow |
21:21.06 | Qwell[] | iCEBrkr: Silly foreign keyboards. ;) |
21:21.08 | iCEBrkr | asterisk99: :D |
21:21.16 | iCEBrkr | oh well, there IS that problem :) |
21:21.57 | asterisk99 | Qwell[]: I FOUND it !!!!!! - thanks iCEBrkr |
21:22.21 | dpolitech | ```! |
21:22.23 | *** join/#asterisk fjean (n=fjean@201.29.122.10) |
21:22.26 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
21:22.32 | Qwell[] | asterisk99: So, it's working then? |
21:23.24 | Inkubot | i need an outbound proxy |
21:23.26 | Inkubot | free.. |
21:23.46 | Inkubot | maybe you know some ? |
21:24.13 | Qwell[] | Inkubot: usually, the outbound proxy is the box you want to call through. Is that not the case? |
21:24.14 | asterisk99 | Qwell[]: somthing differed ... error running install prgram for wctdm |
21:24.23 | Qwell[] | asterisk99: there you go |
21:24.28 | Qwell[] | now you just need to fix your configs |
21:24.35 | asterisk99 | Qwell[]: wctdm module not found |
21:24.41 | *** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net) |
21:24.50 | Qwell[] | or not |
21:25.05 | Qwell[] | asterisk99: ls -l /lib/modules/`uname -r`/misc |
21:25.10 | Qwell[] | Do you see the zap modules in there now? |
21:25.18 | iCEBrkr | :) |
21:25.21 | FLeiXiuS | I'm having a problem with dialing my extensions, It's not correctly dialing my extensions at all. For example, this is the line I have in my dial plane: "exten => _100x,1,Dial(SCCP/100x,15,tT)" |
21:25.39 | Qwell[] | FLeiXiuS: X. I believe case matters |
21:25.44 | iCEBrkr | FLeiXiuS: That's because.. what Qwell[] said. |
21:25.47 | Mavvie | FLeiXiuS: use ${EXTENSION} instead of 100x in teh Dial string |
21:25.47 | Qwell[] | sorry, not X. just X |
21:26.07 | Qwell[] | Mavvie: ${EXTEN} |
21:26.08 | puzzled | FLeiXiuS: exten => _100x,1,Dial(SCCP/${EXTEN},15,tT) |
21:26.22 | iCEBrkr | and there is the who SCCP/100x part that's the problem.. puzzled gave you the correct formatting |
21:26.37 | puzzled | except I missed the lowercase "x" |
21:26.43 | puzzled | FLeiXiuS: exten => _100X,1,Dial(SCCP/${EXTEN},15,tT) |
21:26.43 | asterisk99 | Qwell[]: nothing in that directory |
21:26.52 | Mavvie | Qwell[]: true. was too enthousiast :-) |
21:26.57 | Qwell[] | asterisk99: shame. pastebin the output of te make install |
21:27.02 | Qwell[] | including the command you typed |
21:27.24 | asterisk99 | ok |
21:27.36 | FLeiXiuS | Ahh, I was wondering what the variable was, thankyou puzzled Qwell[] iCEBrkr |
21:27.40 | asterisk99 | Qwell[]: ok - now I have to figure oute pastebin |
21:28.19 | puzzled | FLeiXiuS: read the document README.variables that's included in the source. Explains all the variables |
21:28.28 | Qwell[] | pastebin.com |
21:28.34 | Qwell[] | asterisk99: paste the stuff there, then give us the link |
21:28.42 | FLeiXiuS | puzzled: Will do, I'm also looking on voip-info for a wiki page, I'm sure it's on there some where! |
21:29.15 | puzzled | FLeiXiuS: yeah sometimes stuff is a bit difficult to find. Just use the search function |
21:30.44 | NotFreak | hey guys, i'm doing some hardware hacking on X100P so it can support other CallerID standards but i need some help on reading and setting the registers of the X100P chip that is used, tiger320/md3200 |
21:31.00 | iCEBrkr | NotFreak: Wrong place :P |
21:31.02 | Qwell[] | NotFreak: That's a little out of scope of this channel |
21:31.11 | iCEBrkr | NotFreak: But I wish you luck |
21:31.24 | NotFreak | ok |
21:31.28 | NotFreak | thanks for the info |
21:32.01 | NotFreak | this is more about the general configuration of asterisk ? |
21:32.03 | asterisk99 | Qwell[]: http://pastebin.ca/42329 |
21:32.16 | Qwell[] | NotFreak: Pretty much |
21:32.19 | NotFreak | ok |
21:32.31 | Qwell[] | grr :p |
21:32.33 | NotFreak | so i better should try the mailing list i presume? |
21:32.34 | Qwell[] | asterisk99: ONE LINE |
21:32.36 | alephcom | Does anybody see a problem with this as a timeout? Invalid timeout specified: 'L(100000:60000:30000)' |
21:32.41 | Qwell[] | KVERS=`uname -r` make install |
21:32.44 | Qwell[] | EXACTLY like that |
21:32.57 | asterisk99 | Qwell[]: 1 min |
21:33.04 | Qwell[] | If it was supposed to be two lines, I would have typed it on two lines. :) |
21:33.41 | iCEBrkr | ha |
21:35.01 | iCEBrkr | alephcom: That's at the end of your Dial() statement? |
21:36.05 | asterisk99 | Qwell[]: http://pastebin.ca/42234 |
21:36.24 | Qwell[] | eh? |
21:37.05 | Qwell[] | You don't need to (re)compile your kernel or anything...just type what I said |
21:37.34 | asterisk99 | Qwell[]: I think I did that ... one one line |
21:37.39 | Qwell[] | no, you did two lines |
21:37.45 | Qwell[] | according to your pastebin |
21:37.51 | asterisk99 | Qwell[]: I'll do it again |
21:39.17 | *** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net) |
21:39.43 | asterisk99 | Qwell[]: http://pastebin.ca/42239 |
21:39.56 | alephcom | iCEBrkr: yes, that's at the end of it. I can post the line if you want. |
21:39.58 | Qwell[] | umm |
21:40.13 | iCEBrkr | alephcom: Post just that line.. |
21:40.15 | Qwell[] | I have absolutely no clue what that says |
21:40.51 | Seldon1975 | Qwell: I have set the file permission on Voicemail.conf to -rw-rw-rw but it still doesn't get edited by comedian mail |
21:41.06 | alephcom | <PROTECTED> |
21:41.13 | Qwell[] | asterisk99: Was that the right link? |
21:41.37 | Qwell[] | there we go |
21:41.46 | asterisk99 | Qwell[]: http://pastebin.ca/42339 (sorry :( ) |
21:41.57 | Qwell[] | much better |
21:42.22 | iCEBrkr | alephcom: I meant the line from the dialplan, not console :P |
21:42.26 | asterisk99 | Qwell[]: I should get IRC running on ubunto so I don;t have to keep switching |
21:43.00 | Abydos313 | apt-get install xchat2 |
21:43.06 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
21:43.11 | Qwell[] | asterisk99: What does uname -r say? |
21:43.18 | Qwell[] | 2.6.12-10-386? |
21:43.29 | alephcom | Sorry, I'll try again. |
21:43.53 | asterisk99 | Qwell[]: yes |
21:44.04 | Qwell[] | and what does it say when you modprobe wctdm? |
21:44.19 | Qwell[] | and/or ls -l /lib/modules/`uname -r`/misc/ |
21:44.49 | asterisk99 | Qwell[]: modprobe says module wctdm not found |
21:45.03 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
21:45.36 | asterisk99 | Qwell[]: I do the modprobe in /usr/src/zaptel-1.2 right? |
21:45.45 | alephcom | iCEBrkr: Here it is: exten => _X.,1,Dial(SIP/17808473033@atlasvoice|L(100000:60000:30000)) |
21:45.50 | Qwell[] | anywhere. it doesn't matter |
21:45.58 | Qwell[] | you're just typing "modprobe wctdm" ? |
21:46.04 | asterisk99 | Qwell[]: yes |
21:46.09 | Qwell[] | no .ko or anything? |
21:46.25 | asterisk99 | Qwell[]: nope --- look at bootom of pastebin |
21:46.51 | Qwell[] | you're typing "modprobe wcfxs" |
21:47.09 | Qwell[] | That file was removed |
21:47.14 | file | alephcom: you're putting options into the place where you put in the call timeout |
21:47.50 | iCEBrkr | alephcom: Yea, you're missing a parameter |
21:48.23 | Qwell[] | asterisk99: Honestly, if you aren't going to type EXACTLY what I say...please don't ask me for help... |
21:48.26 | alephcom | Please tell me which one??? :-) This used to work and I'm almost banging my head against a wall. :-( |
21:48.33 | file | add another | |
21:48.39 | asterisk99 | Qwell[]: what did I do wrong??? |
21:48.41 | file | beside the existing | |
21:48.45 | Qwell[] | wcfxs != wctdm |
21:48.50 | alephcom | I see, I see, tks. |
21:48.53 | Qwell[] | I asked if you were modprobe'ing wctdm, and you said you were. You aren't |
21:49.31 | alephcom | you just make my day!!! Tks |
21:49.37 | alephcom | or maybe saved my sanity |
21:49.47 | asterisk99 | Qwell[]: modprobe wcfxs gives me module wctdm not found |
21:49.47 | alephcom | If I had any to start with |
21:49.47 | Qwell[] | alephcom: Sanity? Can I borrow some? |
21:49.56 | Qwell[] | asterisk99: Then don't type "modprobe wcfxs" |
21:50.00 | Qwell[] | type what I said |
21:50.09 | asterisk99 | Qwell[]: gotcha |
21:50.20 | file | I need a weapon... |
21:50.23 | file | something sharp |
21:50.55 | dpolitech | broken broom stick handle? |
21:50.55 | *** join/#asterisk gambolputty (n=root@64.74.225.135) |
21:51.06 | generalhan | Can anyone tell me where i define my offset on my ntpd server? all my phones think we are EST, when we are PST right now. |
21:51.15 | gambolputty | hi |
21:51.17 | generalhan | then i have to figure out how to turn it off, so that DST doesnt mess us up |
21:51.35 | lazzarello | asterisk99, install debian stable and use the zaptel-source package. you think less, you make less mistakes, the hard part is taken care of. |
21:51.57 | gambolputty | I have an analog card that won't recognize caller ID. I wait 2 seconds, then answer, and no caller ID result. Any ideas? |
21:52.00 | Qwell[] | lazzarello: It still won't work, if he doesn't modprobe the right driver |
21:52.24 | lazzarello | Qwell[], a reboot loads the driver automatically. |
21:54.01 | asterisk99 | Qwell[]: I'm going to switch IRC on the * machine... that way I can cut and paste |
21:54.01 | Qwell[] | same thing if you compile from source and use the init script |
21:54.15 | _Sam-- | Qwell: when did you become so patient |
21:54.29 | Qwell[] | _Sam--: I had a large energy drink this morning |
21:54.34 | asterisk99 | I'll be bacaaack |
21:54.58 | _Sam-- | hmm maybe i need to try drinking them at my work |
21:55.05 | _Sam-- | patience is not a virtue i currently possess |
21:55.43 | Qwell[] | yeah...he's never gonna get it |
21:56.16 | _Sam-- | you could just have him download tzafrir's debian zaptel packages |
21:56.18 | _Sam-- | they work fine |
21:56.29 | _Sam-- | i dont know what kernel that guy was on |
21:56.37 | _Sam-- | <didnt pay that much attn> |
21:57.43 | Qwell[] | brb |
21:57.51 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
21:57.56 | Qwell[] | or not |
21:57.58 | _Sam-- | lo |
21:58.14 | asterisk99 | Quell[]: I'm back now on 1 machine |
21:58.16 | _Sam-- | this place has tzafrir's modules: |
21:58.16 | _Sam-- | http://rapid.dotsrc.org/unstable/ |
21:58.31 | Qwell[] | asterisk99: good, now type modprobe wctdm |
21:58.33 | Qwell[] | and nothing else |
21:58.33 | _Sam-- | simple as downloading then 'dpkg -i somethingorother-zap.deb |
21:59.32 | _Sam-- | plus if it doesnt work, then you can make tzafrir do the support ;) |
21:59.38 | asterisk99 | Quell[]: root@konnex:/usr/src/zaptel-1.2# modprobe wctdm |
21:59.38 | asterisk99 | FATAL: Module wctdm not found. |
21:59.38 | asterisk99 | <PROTECTED> |
21:59.48 | Qwell[] | better |
21:59.57 | asterisk99 | Quell[]: :) |
22:00.05 | Qwell[] | now ls -l /lib/modules/`uname -r`/ |
22:00.17 | Qwell[] | if stuff isn't there...something is broken. I saw the make install put them there |
22:00.30 | lazzarello | sheesh, there's even packages on backports.org for a 1.2.1 zaptel stuff. |
22:01.06 | Qwell[] | erm |
22:01.09 | Qwell[] | now ls -l /lib/modules/`uname -r`/misc/ |
22:01.12 | Qwell[] | that one |
22:01.51 | asterisk99 | Quell: It's big --- u want me to pastebin that? |
22:01.58 | Qwell[] | are there files there? |
22:02.04 | Qwell[] | wctdm.ko in particular |
22:02.26 | Qwell[] | hurry up, I need to pee :p |
22:02.35 | asterisk99 | Quell[]: root@konnex:/usr/src/zaptel-1.2# ls -l /lib/modules/`uname -r`/ |
22:02.35 | asterisk99 | total 1268 |
22:02.35 | asterisk99 | lrwxrwxrwx 1 root root 36 2006-02-18 13:32 build -> /usr/src/linux-headers-2.6.12-10-386 |
22:02.35 | asterisk99 | drwxr-xr-x 2 root root 4096 2006-02-17 12:27 initrd |
22:02.35 | asterisk99 | drwxr-xr-x 11 root root 4096 2006-02-17 12:26 kernel |
22:02.36 | asterisk99 | drwxr-xr-x 2 root root 4096 2006-02-17 12:27 madwifi |
22:02.38 | asterisk99 | drwxr-xr-x 2 root root 4096 2006-02-20 16:15 misc |
22:02.40 | asterisk99 | -rw-r--r-- 1 root root 244258 2006-02-20 16:38 modules.alias |
22:02.43 | FLeiXiuS | GRRR |
22:02.44 | asterisk99 | -rw-r--r-- 1 root root 69 2006-02-20 16:38 modules.ccwmap |
22:02.45 | FLeiXiuS | pastebin. |
22:02.46 | asterisk99 | -rw-r--r-- 1 root root 298510 2006-02-20 16:38 modules.dep |
22:02.48 | asterisk99 | -rw-r--r-- 1 root root 813 2006-02-20 16:38 modules.ieee1394map |
22:02.50 | asterisk99 | -rw-r--r-- 1 root root 1141 2006-02-20 16:38 modules.inputmap |
22:02.51 | Qwell[] | meh |
22:02.52 | FLeiXiuS | WOW.. |
22:02.52 | asterisk99 | -rw-r--r-- 1 root root 21256 2006-02-20 16:38 modules.isapnpmap |
22:02.53 | FLeiXiuS | lmao |
22:02.54 | asterisk99 | -rw-r--r-- 1 root root 226143 2006-02-20 16:38 modules.pcimap |
22:02.56 | asterisk99 | -rw-r--r-- 1 root root 1135 2006-02-20 16:38 modules.seriomap |
22:02.58 | asterisk99 | -rw-r--r-- 1 root root 123227 2006-02-20 16:38 modules.symbols |
22:03.00 | asterisk99 | -rw-r--r-- 1 root root 315491 2006-02-20 16:38 modules.usbmap |
22:03.02 | asterisk99 | drwxr-xr-x 2 root root 360 2006-02-18 15:56 volatile |
22:03.06 | asterisk99 | Sorry... he needs to pee |
22:03.11 | mocker | http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message |
22:03.14 | Qwell[] | I said the misc dir :P |
22:03.20 | dpolitech | uhh |
22:03.20 | Qwell[] | nevermind, I'm done |
22:03.32 | mocker | In this example is says that it would be possible to reschedule the call, can someone give me a rough idea of how to do that? |
22:03.38 | Qwell[] | _Sam--: He's all yours |
22:03.39 | puzzled | asterisk99: use pastebin.ca if you want to paste more than 4 lines and post the link here |
22:03.46 | asterisk99 | Quell[]: 0 files in .misc |
22:03.51 | Qwell[] | .misc? |
22:03.56 | puzzled | maybe they live in extra |
22:04.04 | Qwell[] | puzzled: no, I saw the make install put them in misc |
22:04.25 | puzzled | maybe he's been haz0red then :) |
22:04.26 | Qwell[] | asterisk99: I'll be back in like 10 minutes. If you give me ssh access to your machine, I'll have it fixed in about 2 |
22:04.33 | _Sam-- | how about "updatedb" then "find wctdm.ko" ? |
22:04.39 | _Sam-- | er locate wctdm.ko |
22:04.46 | Qwell[] | _Sam--: was gonna say... |
22:04.49 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
22:05.03 | Qwell[] | asterisk99: okay? |
22:05.05 | asterisk99 | Quell[]: I could give u ssh if needed |
22:05.12 | Qwell[] | alright, brb then...get it ready for me |
22:05.16 | *** join/#asterisk darby_t (n=tom@abcf71.neoplus.adsl.tpnet.pl) |
22:05.17 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
22:05.46 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
22:05.49 | _Sam-- | asterisk99 : that is a nice offer. hope you buy that guy a case of good beer or something. |
22:06.42 | puzzled | the real stuff like Grolsch :) |
22:06.53 | dpolitech | which reminds me |
22:06.58 | dpolitech | looks like beer-thirty |
22:07.02 | dpolitech | quittin time! |
22:07.07 | _Sam-- | lol |
22:07.08 | dpolitech | later |
22:07.22 | _Sam-- | drink a few back for us |
22:07.31 | dpolitech | will do |
22:10.15 | asterisk99 | _Sam--: I'll buy a round for everyone if this works!!! |
22:10.40 | lazzarello | beer o'clock! |
22:10.43 | _Sam-- | be careful what you say...we will take you up on that, and it could be an expensive round :) |
22:10.46 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
22:10.48 | asterisk99 | _Sam--: But I don;t think PayPal accepts alcohol deposits |
22:11.13 | _Sam-- | we have some creative souls here...im sure if beer and/or other alcohol is involved for free, we will find some way |
22:11.14 | Qwell[] | okay, back |
22:11.15 | austinnichols101 | asterisk99: not true - just select the 'other' type of payment from the listing |
22:11.45 | asterisk99 | _Sam--: And I'm talking good 'ol 10%-by-volume Quebec Beer - better than that Canadian pee |
22:12.05 | Qwell[] | awesome, my proxy server isn't letting me connect to my other server |
22:12.16 | Qwell[] | there we go D: |
22:12.23 | Qwell[] | asterisk99: msg me the ip/password |
22:12.38 | *** join/#asterisk rob- (n=robbie@haylott.plus.com) |
22:13.19 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-5.dynamic.qsc.de) |
22:14.33 | lunaphyte | why might asterisk segfault when a skinny phone answers a call from a sip phone? |
22:14.52 | jontow | skinny is an ugly, ugly, horrible thing. |
22:15.24 | _Sam-- | someone else today was talking about chan_skinny, probably qwell / mog |
22:16.23 | lunaphyte | i gather.. :) |
22:16.49 | lunaphyte | the phone (12sp) rings, but when i pick it up, asterisk seg faults. |
22:19.23 | lunaphyte | and when i try to call x-lite from the 12sp, asterisk doesn't complete the call, seeming to be waiting for more digits. |
22:19.51 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
22:20.14 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
22:25.51 | cypromis | w 8 |
22:27.54 | [TK]D-Fender | asterisk99 : Thats right... American beer is fit only for children and the elderly :D |
22:28.46 | *** join/#asterisk rene- (n=rene-@201.127.101.127) |
22:29.02 | jontow | lunaphyte: i had "ok" luck with a 30VIP/12SP+ |
22:29.39 | puzzled | lunaphyte: the stuff from chan-sccp.berlios.de seems to work ok |
22:29.57 | lunaphyte | actually, that 12sp i mentioned is a 12sp+ |
22:30.28 | lunaphyte | puzzled: thanks, i'll give it a shot |
22:31.56 | rene- | im using a centos based asterisk distro from signate, zaptel isnt working, signate is telling me to upgrade to centos 4.2, i have tried compiling asterisk from source and it aint working, im getting an invalid module format error when i try to modprobe zaptel modules, would compiling kernel from source be an option for me? |
22:32.11 | puzzled | lunaphyte: and if you have an issue with it, subscribe to the list with your problerm. the author is quite responsive |
22:32.37 | *** part/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
22:33.29 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
22:35.19 | cypromis | rene-: no |
22:35.28 | cypromis | but we have a shitload of servers with centos |
22:35.35 | cypromis | and no issues with them |
22:36.38 | rayvd | shitload = 17 :( |
22:37.12 | |omni| | I don't like CentOS |
22:37.18 | |omni| | not really a fan of RedHat though |
22:38.02 | freezer | i love debian |
22:38.08 | rene- | cypromis: what are you choosing when you first install them? server, custom? and, is udev an issue? |
22:38.35 | FLeiXiuS | Is there a function which will send an incoming call to all extensions? |
22:38.50 | rene- | i love debian too, i only like centos because it knows about scsi controllers and its free unlike redhat |
22:39.09 | FLeiXiuS | rene-: redhats free version is now known as fedora |
22:39.14 | |omni| | I'm a slackware dude... on the linux side anyway... FreeBSD is my fav but asterisk likes slack better |
22:39.50 | freezer | whats better at freebsd? |
22:40.07 | rene- | i have had success with fedora 4 and scsi controllers, but the only thing i could load on a dell 2650 was white box enterprise |
22:40.11 | |omni| | better? ..just preference |
22:40.31 | |omni| | I've been using FreeBSD longer and all of our webservers and mapservers run BSD |
22:40.51 | |omni| | but early on asterisk had issues with BSD and my partner is a slack guy so I started messing with that for the phone stuff |
22:41.49 | cypromis | rene: minimal :) |
22:41.52 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
22:41.55 | Tagor | I have a DSL connection which I want to use for several computers. I also want to connect my Asterisk server to the internet, to make external calls. Is there a router which doesn't have NAT problems? Or is this impossible? |
22:41.55 | cypromis | and than yum it to what I want it to be |
22:42.15 | X-Rob_ | <PROTECTED> |
22:42.24 | rene- | cypromis: i dont think it has minimal, this signate centos distro |
22:42.48 | remiss | rene-: it has, but choose something else or you'll be spending your days resolving dependencies |
22:42.54 | Tagor | X-Rob_ >> Hmm, the problem when I do that, is that the connection gets very slow |
22:43.07 | Tagor | X-Rob >> Or are there decent programs that can handle that? |
22:43.53 | remiss | rene-: you probably want workstation or server... :) |
22:44.46 | rene- | hehe |
22:44.49 | Seldon1975 | when i upgrade from * 1.2.1 top 1.2.4 do I have to recompile the wctdm module? |
22:45.33 | jarrod | is there a way to set caller-id with ser when matching source account |
22:45.34 | jarrod | ? |
22:49.48 | mzo | bleh i broke my asterisk trying to do FWD. Every number i dial is 404 now. YAY :) |
22:50.32 | jarrod | sounds like all your numbers are matching a blackhole inbound ext |
22:50.34 | Hmmhesays | cool |
22:50.46 | Hmmhesays | i'll fix it for a cool $40 bucks |
22:51.02 | file | Hmmhesays: does that involve putting the cash in the freezer? |
22:51.08 | Hmmhesays | file: yes |
22:51.12 | Hmmhesays | wet cash |
22:51.15 | file | excellent |
22:51.24 | Hmmhesays | ugh, in 7 hours I turn 24 |
22:51.30 | remiss | don't brake it... |
22:51.57 | file | Hmmhesays: you're O-L-D |
22:52.03 | Hmmhesays | I know! |
22:52.32 | Hmmhesays | i can feel the aching in my joints |
22:52.36 | X-Rob_ | Tagor, most people use a linux box as their router, they just don't know it. |
22:52.42 | Abydos313 | Happy Bday Hmmhesays!!! |
22:52.47 | Hmmhesays | thanks |
22:52.53 | remiss | Hmmhesays: i do that as well and i'm only 20... |
22:52.58 | Hmmhesays | i can't believe i've been hanging out in this channel for 2 years |
22:53.02 | Hmmhesays | sweet geebus |
22:53.17 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
22:53.18 | file | I've been here longer then... that's more scary |
22:53.22 | Abydos313 | i bet you've learned alot |
22:53.36 | Hmmhesays | I come in here to BS a lot |
22:53.42 | Hmmhesays | help out every once in awhile |
22:54.00 | remiss | can you help me? i need to fall asleep... |
22:54.00 | Hmmhesays | as a pleasant distraction |
22:54.03 | Tagor | X-Rob_ >> But can this be handled by a 500mhz 128mb ram computer? Can this computer run as router (for 4 pc's), Asterisk server and CUPS server? |
22:54.17 | Qwell[] | I don't know how to save a screen session |
22:54.18 | *** join/#asterisk jtodd (n=jtodd@mccpool-2.ci.monterey.ca.us) |
22:54.25 | Qwell[] | erm |
22:54.31 | Hmmhesays | 128mb of ram would be cutting it a little short |
22:54.31 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
22:54.38 | X-Rob_ | Tagor, yes. Most ADSL routers are tiny little 50mhz MIPS machines that easily handle full ADSL speeds |
22:54.41 | file | Hmmhesays: hot'n'sexy Hmmhesays |
22:55.00 | Hmmhesays | yes file? |
22:55.02 | Tagor | Ok, thanks a lot X-Rob_! |
22:55.04 | file | Hmmhesays: how goes |
22:55.12 | Hmmhesays | still living |
22:55.16 | remiss | Qwell[]: save a screen session? |
22:55.21 | Qwell[] | nothing, ignore that |
22:55.23 | file | my ADSL modem is actually a 48MHz Motorola chip |
22:55.23 | Hmmhesays | got up and rocked the house last night at jam night |
22:55.29 | file | with 4MB of RAM and 1MB of flash |
22:56.25 | Hmmhesays | my nintendo was a 4mhz chip |
22:56.44 | b0ef | any reason I cannot transfer a call to voicemail? - I can transfer all other calls |
22:56.44 | Mavvie | Hmmhesays: don't think you can do cups on that :-) |
22:56.53 | Hmmhesays | Mavvie You never know |
22:57.01 | Mavvie | (that's not strictly true, there were print-servers on XTs in the past :-) |
22:57.09 | file | Hmmhesays: damn, you are O-L-D |
22:57.26 | Hmmhesays | I know it |
22:57.37 | file | Hmmhesays: so what'cha gonna do to celebrate? |
22:58.13 | b0ef | is it just me or can anybody else transfer calls to voicemail? |
22:58.24 | file | b0ef: see, there's a problem with that question |
22:58.45 | file | b0ef: you write how the dialplan works, so while other people may write their dialplan so it works fine and dandy, you may not have |
22:59.53 | file | you tell Asterisk what to do, so if you don't tell Asterisk a method of transferring calls directly to voicemail well then it won't do it |
23:07.36 | Qwell[] | ugh |
23:09.42 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167041112.pppoe-dynamic.nb.aliant.net) |
23:09.43 | rene- | im getting while modprobing zaptel. zaptel: version magic '2.6.9-22.0.1.ELsmp SMP 686 REGPARM 4KSTACKS gcc-3.4' should be '2.6.9-22.0.1.ELsmp SMP 586 REGPARM 4KSTACKS gcc-3.4' fatal: error inserting zaptel.... invalid module format, how can i fix it... |
23:10.22 | Juggie | stop using pre compiled versions of asterisk |
23:10.25 | Juggie | compile it yourself |
23:12.22 | Netgeeks | why oh why do people write AGI programs to do stuff asterisk does inherently? |
23:12.29 | SplasPood | well that may not be his/her issue |
23:13.29 | mzo | jarrod, probably, you mean my problem all my numbers matching a blackhole inbound extension? :P |
23:13.34 | rene- | Juggie: iam compiling zaptel from source |
23:13.36 | SplasPood | Netgeeks: I've been tempted to before.. simply cause the dialplan syntax (even with AEL) sorta sucks |
23:13.56 | SplasPood | rene: you're just not compiling it using the same compiler settings as your kernel was originally compiled for |
23:14.17 | SplasPood | rene: note the 686 vs 586 |
23:14.52 | mzo | which is 686, pentium III or better? |
23:15.17 | SplasPood | yes, I do believe |
23:15.21 | *** join/#asterisk paxr0 (n=paxr0@216-155-79-99.bk1-dsl.surnet.cl) |
23:15.30 | Netgeeks | I was just trying to figure out why A@H has an AGI for the directory function... especially when the AGI doesn't work |
23:15.39 | mzo | i think i was drunk when i recompiled 1.2.3 so I don't know when that happened. |
23:15.44 | SplasPood | Netgeeks: heh.. |
23:15.54 | mzo | Netgeeks, i broke my AAH install quite nicely. Someday it'll actually be working ;) |
23:16.13 | *** join/#asterisk Shido (n=shido6@d221-68-216.commercial.cgocable.net) |
23:16.22 | *** join/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk) |
23:16.23 | SplasPood | hey, can someone DCC me a stock A@H dialplan (extensions.conf) |
23:16.45 | Netgeeks | I refuse to use A@H myself |
23:17.08 | rene- | SplasPood: yes i notice that, would compiling a new kernel for this machine would be of any help? |
23:17.09 | Netgeeks | however I get alot of 'please help me fix this I can't get it to work' calls |
23:17.13 | mzo | what's wrong with it? :P |
23:17.21 | mzo | i mean i break it at least once a week. ;) |
23:17.44 | SplasPood | rene: chances are if you compile a new kernel it'd compile using the same settings zaptel does, however you may cause yourself other problems if you don't do it right.. |
23:18.32 | rene- | I once compiled a kernel for vserver usage, so i am no expert but i have done it, the only tricky part i have seen is getting the support for the scsi card and the netcard right |
23:18.49 | mzo | actually somewhere along the way |
23:18.50 | rene- | and the SMP part |
23:18.54 | mzo | AAH added support for SMP kernals |
23:18.59 | mzo | it picked up my SMP out of the box |
23:19.22 | paxr0 | anybody know softphone (for palm OS) ? |
23:19.40 | Tamarisk | Can anyone recommend a lowend SIP phone I could look to use on a home network to learn *. I want to remove my slow computer issues when trying with softfones. TIA |
23:19.44 | rene- | i guess the only way to know is trying |
23:19.45 | mzo | so who wants to laugh at me a lot and figure out why i broke FWD :) |
23:20.54 | *** join/#asterisk kremin (n=kremin@dslb-084-063-106-035.pools.arcor-ip.net) |
23:21.06 | rene- | i am only interested in the first part but im quite busy right now, so i would have to pass on your offer |
23:21.22 | rene- | jk |
23:21.25 | mzo | i'm going to fix asterisk by -rm -rf :P |
23:21.52 | De_Mon | whats -rm -rf do? |
23:22.04 | Qwell[] | De_Mon: Gives a command not found error |
23:22.12 | kremin | is this the right place for questions about asterisk configuration? |
23:22.15 | mzo | don't try it :P |
23:22.20 | De_Mon | mzo I don't think that's gona fix asterisk |
23:22.21 | Qwell[] | kremin: yes |
23:22.33 | mzo | De_Mon i know it won't :P i broke it bad |
23:22.51 | De_Mon | mzo try rm -rf /etc/asterisk and rebuild the examples |
23:22.51 | kremin | ok ... i guess most of you guys are from the us? |
23:22.56 | kremin | US= |
23:22.59 | kremin | US? |
23:23.02 | De_Mon | I'm not |
23:23.03 | rene- | mx here |
23:23.16 | mzo | De_Mon, it works, i broke it when i added FWD, I can jsut remove that and try again |
23:24.03 | kremin | anyway .. i need to know if i must use a crossover cable to connect two HFC-S cards to two german "Anlagenanschlüsse" .. that are sprecial ISDN connections. |
23:24.08 | De_Mon | mzo naaa you broke it worse than that, I suggest going with the rm -rf |
23:24.17 | kremin | ahh ... it was to long ... agaiN: |
23:24.23 | Netgeeks | but then again I have AGI with a red circle / slash tatood on my forhead |
23:24.42 | luckyduck | kremin: depends on the cards you use |
23:24.56 | kremin | i have two HFC-S cards |
23:24.59 | luckyduck | kremin: in the case of quadbri cards, you dont have to |
23:25.01 | luckyduck | single port? |
23:25.08 | kremin | yes single port |
23:25.15 | kremin | two of them |
23:25.23 | luckyduck | in that case, you need crossover isdn cables for the nt modus |
23:25.44 | luckyduck | if you want to connect a tka to the cards |
23:25.56 | kremin | whats a "tka" ? |
23:26.06 | luckyduck | if you only want to connect to the ntba, you need the te modus |
23:26.07 | b0ef | file: well, that's not really my problem: when I call my voicemail to read messages, I want to transfer it to a conference room so that other people can listen whil I go through my voicemail messages |
23:26.08 | mzo | De_Mon you think? lemme try the rm -rf :P |
23:26.11 | luckyduck | and no crossover |
23:26.19 | luckyduck | kremin: something like a eumax from the t-com |
23:26.22 | luckyduck | and so on |
23:26.25 | mzo | ~wiki |
23:26.33 | luckyduck | or eumex |
23:26.36 | puzzled | anyone know how I fix this error compiling asterisk-1.2.4? cli.c:49:30: error: asterisk/version.h: No such file or directory |
23:26.39 | kremin | i want to connect to the NTBA |
23:26.44 | luckyduck | a hardware based, quite cheap pbx |
23:26.52 | luckyduck | then, you dont need a crossover |
23:27.08 | kremin | i want to get rid of the pbx and exchange it with asterisk |
23:27.17 | luckyduck | it's only needed to the nt modus, i.e. in the case where you have to provide an nt port |
23:27.20 | De_Mon | puzzled figure out why version.h wasn't in your source |
23:27.34 | luckyduck | kremin: =) |
23:27.53 | puzzled | De_Mon: problem only occurs on ppc. on x86 it compiles fine |
23:28.36 | kremin | so i wired the two HFC-S to two NTBAs ... but crossover. in zapata.conf i said "signalling=bri_net" |
23:28.41 | kremin | that ok isnt it? |
23:29.04 | De_Mon | ppc? there, i can't help. |
23:29.35 | luckyduck | kremin: sorry, never played around with the zapata.conf so far. |
23:29.49 | luckyduck | however, have to go to bed. already late |
23:29.52 | luckyduck | night |
23:29.55 | kremin | night |
23:30.42 | kremin | anybidy else using HFC-S cards connected to NTBAs? |
23:33.28 | Tamarisk | Can anyone let me know what they recommend as a basic SIP phone on their asterisk server? |
23:33.58 | austinnichols101 | Tamarisk: price range? |
23:34.38 | Tamarisk | I do not know at moment, I guess less then £50 each this is for home/hobby use so 2nd from ebay would do |
23:34.41 | |omni| | handset or ata type? |
23:35.01 | Tamarisk | looking for more of a telephone style |
23:35.25 | Tamarisk | I want to stop using softphones |
23:35.30 | |omni| | not a big fan of the cheap handsets...cause they feel cheap...but apparently some of the lower end Snom's are pretty good |
23:35.42 | |omni| | but..you could always just get a Sipura ATA (like a 2000) |
23:35.46 | |omni| | and use any regular phone you want |
23:35.49 | austinnichols101 | I definitely didn't like the sipura 841 |
23:36.07 | Tamarisk | OK I am writing these down so I can search for them later |
23:36.13 | austinnichols101 | but the low-end aastra (9112, I think) was good |
23:36.22 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
23:36.24 | mzo | stupid question what's your default [your-main-context-whatever-it-may-be] if i'm using aah? |
23:36.53 | exonic | I have a problem with caller id not being set on outgoing calls. It only shows one number (that's not even in my configuration!? ) |
23:37.29 | Tamarisk | my asterisk is purely IP based PC so no cards fitted |
23:37.33 | |omni| | even when you use SetCallerID() ? |
23:37.50 | |omni| | Tamarisk: : http://www.sipura.com/products/spa2000.htm |
23:38.08 | |omni| | 2 lines, cheap, no cards needed (SIP) and you can use any standard POTS phone you want |
23:38.10 | Tamarisk | Cheers |
23:38.29 | Tamarisk | Is that one of the adapters |
23:38.36 | |omni| | ATA, ya |
23:39.02 | Tamarisk | OK catching on slowly |
23:39.22 | FLeiXiuS | Any idea why "exten => 500,2,VoiceMailMain(${CALLERID(num)})" is not picking up the CID number. |
23:39.32 | |omni| | on the desktop I really like the Cisco phones.. like 7940 or 7960s...you can find them on ebay for a couple hundred bucks |
23:39.59 | FLeiXiuS | |omni|: Plus, their SCCP support is quite nice ;-) |
23:40.12 | austinnichols101 | tamarisk: I would have to agree with omni - ATA is better than any of the phones that are below USD $100. |
23:40.22 | |omni| | ya actually I just switched my 7960 to the SIP image |
23:40.24 | rene- | how much time does a kernel compile should take on a p4 3.4? |
23:40.43 | austinnichols101 | FleiXiuS: why do you like the sccp? |
23:40.43 | |omni| | I was using chan_sccp until a couple days ago |
23:41.01 | rene- | id like to see sccp realtime |
23:41.14 | |omni| | but the sip image has more usable options (CfwdAll, Conferencing on the phone, etc.) |
23:41.15 | Qwell[] | rene-: chan_sccp? |
23:41.19 | |omni| | rene-: , apparently they're working on it |
23:41.27 | Qwell[] | working on - and finished |
23:41.34 | |omni| | oh really? |
23:41.37 | Qwell[] | now...go yell at Sergio to implement my patch |
23:42.29 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
23:42.29 | rene- | great news, i was told that the feature set on sccp was superior to sip on cisco phones |
23:42.33 | mzo | should i be using IAX or SIP for FWD? |
23:43.20 | Tamarisk | OK thanks for the tips all, at least I can go search ebay UK or worldwide. My softphones work sometimes sometimes not so need to remove one type of issue |
23:43.22 | rene- | i need to get realtime, well im still waiting on the firmware for my phones so no rush |
23:44.50 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.130) |
23:46.53 | *** part/#asterisk fjean (n=fjean@201.29.122.10) |
23:47.25 | Tamarisk | Sorry another question on ATA's what do they mean by "Unlocked"? |
23:47.45 | Qwell[] | Tamarisk: not locked to a provider - like Vonage |
23:48.43 | Tamarisk | AHHH Ok just found an sipora Grandstream and wondered. |
23:48.50 | *** join/#asterisk DarthClue (n=DarthClu@adsl-69-152-233-136.dsl.snantx.swbell.net) |
23:49.04 | Tamarisk | OK I will leave you all at the hard work thanks for assistance night |
23:49.06 | Qwell[] | DarthClue: Eww...it's...you |
23:49.16 | _Sam-- | [av]bani : you are Dan on the * users mailing list? |
23:49.48 | [av]bani | _Sam--: ? |
23:50.00 | *** part/#asterisk Tamarisk (n=adrian@user-4200.lns4-c10.dsl.pol.co.uk) |
23:50.12 | _Sam-- | i see some guy active in the gxp discussion..with domain anime.net... |
23:50.18 | _Sam-- | i thought it was you |
23:50.26 | _Sam-- | asterisk-users |
23:50.57 | _Sam-- | <_Sam--> asterisk-users |
23:50.58 | _Sam-- | er |
23:54.58 | _Sam-- | [av]bani : your isp is visp? |
23:55.41 | [av]bani | whatever gives you that idea? |
23:55.48 | _Sam-- | just speculation |
23:55.51 | [av]bani | nosy :) |
23:57.18 | _Sam-- | this guy sounds crafty: "Patent Pending Permissions Technology" |
23:57.26 | _Sam-- | and even authored the patent |
23:58.09 | rene- | but its flaky on os x |
23:58.21 | rene- | sp: flacky |
23:58.41 | _Sam-- | WasPhantom: flakey |
23:58.43 | _Sam-- | er |
23:58.55 | rene- | thanks |
23:58.57 | _Sam-- | flakey like a nice fresh croissant |