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| 00:29.25 | Mahmoud | life sucks with my I$P |
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| 00:32.17 | jayk- | i just upgraded to asterisk 1.4.1 and I get No channel type registered for 'Zap'. i was using Zap/g1 to make calls.. |
| 00:32.32 | jayk- | anybody got any ideas how I can get that to work again? |
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| 00:36.26 | shido6 | why does it suck Mahmoud? |
| 00:36.37 | shido6 | change the ports your sip box uses |
| 00:36.40 | Mahmoud | my isp blocks SIP not by port, but at the application layer |
| 00:36.55 | shido6 | is that what they told you? |
| 00:37.02 | Mahmoud | nope |
| 00:37.12 | Mahmoud | i tried SIP on differnet UDP ports and the result is the same |
| 00:37.13 | shido6 | what ports are they blocking? |
| 00:37.19 | shido6 | what udp ports did you try? |
| 00:37.28 | Mahmoud | 5060, 6060, 53, 56 |
| 00:37.34 | shido6 | err |
| 00:37.39 | shido6 | use common ports |
| 00:37.43 | shido6 | like FTP ports |
| 00:37.48 | Mahmoud | heh... |
| 00:37.51 | Mahmoud | FTP is TCP |
| 00:37.56 | Mahmoud | i'll use TFTP port |
| 00:38.04 | shido6 | now ur thinking |
| 00:38.19 | Mahmoud | can you test my account? |
| 00:38.30 | Mahmoud | i'll give you uname,pword, domain name |
| 00:38.48 | Mahmoud | and i'll see if it connects or not |
| 00:39.29 | shido6 | test what account? |
| 00:39.34 | Mahmoud | my pbx |
| 00:39.40 | Mahmoud | test if you can connect to it or not |
| 00:39.57 | Mahmoud | then dial 611 to hear the automated voice menu and tell me its quality |
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| 01:11.28 | Mahmoud | back |
| 01:11.38 | distatica | I'm going to install asterisk here and help a buddy who needs to learn it for work, how much can actually be done without having a cheap card like the X100? Once testing is complete it would be setup on a machine that had cards. |
| 01:12.42 | mafkees | distatica: you dont need a zaptel card to use asterisk |
| 01:13.09 | mafkees | without a zaptel card it will work fine. only the conferencing and IAX trunking wont work |
| 01:13.39 | blitzrage | distatica: you don't need any hardware at all -- just use a pair of softphones |
| 01:13.46 | blitzrage | mafkees: ztdummy |
| 01:13.49 | mafkees | a plain IAX connection will work |
| 01:13.56 | mafkees | blitzrage: only on linux |
| 01:14.03 | mafkees | blitzrage: I'm using OpenBSD |
| 01:14.05 | distatica | i'm on linux |
| 01:14.11 | blitzrage | mafkees: sane people use linux |
| 01:14.28 | mafkees | blitzrage: s/sane/some/ |
| 01:14.44 | blitzrage | distatica: get zaptel and compile in ztdummy for timing support |
| 01:15.09 | blitzrage | distatica: x100p has been discontinued for a while -- it's ok for a hobby system, but isn't necessary for timing issues, and isn't a great card for FXO |
| 01:15.25 | mafkees | the x100p is evil |
| 01:15.51 | mafkees | I had nothing but trouble with it |
| 01:16.00 | blitzrage | because it's pretty much a useless card |
| 01:16.05 | mafkees | indeed |
| 01:16.12 | blitzrage | hence why Digium doesn't support it anymore |
| 01:16.15 | mafkees | call quality is very bad |
| 01:16.19 | blitzrage | x100p == support nightmare |
| 01:16.34 | mafkees | no CID support here |
| 01:16.38 | blitzrage | that too |
| 01:17.43 | mafkees | I trashed it |
| 01:17.51 | mafkees | IAX2 FTW |
| 01:18.49 | mafkees | every install under 10 lines is done wih an IAX2 ITSP |
| 01:19.17 | mafkees | if ppl really want to use POTS, I give them ISDN10 (partly E1) |
| 01:19.39 | JT | too bad if they have issues witht heir Internet link or ITSP |
| 01:19.42 | mafkees | the sangoma A102 with echo cancel is very good there |
| 01:20.25 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 01:20.32 | mafkees | JT: I can deliver DSL 1/1 mbit with qos for 45 euro/month |
| 01:20.48 | mafkees | I can run more channels there then dual ISDN2 |
| 01:21.01 | mafkees | the dual ISDN2 is the same price |
| 01:21.15 | mafkees | single ISDN2 is 22.45 euro/month here |
| 01:21.35 | JT | is that a dedicated dsl tail? |
| 01:21.40 | mafkees | yeah |
| 01:21.46 | mafkees | of course |
| 01:21.46 | JT | with a virtual path to itsp? |
| 01:21.48 | JT | heh |
| 01:22.00 | JT | better than using the Internet :) |
| 01:22.11 | JT | do they go over lines with a dialtone? |
| 01:22.25 | mafkees | ITSP gives IAX2 dialtone |
| 01:22.27 | ping2921 | hello |
| 01:22.40 | JT | no, does the pair have a dialtone? |
| 01:22.48 | mafkees | yes |
| 01:22.57 | JT | ok |
| 01:23.05 | mafkees | we use that as failsafe |
| 01:23.11 | mafkees | red-painted old telephone |
| 01:23.14 | ping2921 | how can I set the callerid before calling using dial() -- basically I have calls come into *, and then I do a dial() cmd/ |
| 01:23.30 | mafkees | 'if all else fails, use this phone to dial 911' |
| 01:23.38 | JT | good idea, i've heard here people with dsl over no dialtone pairs often get them disconnected by incompetant telco staff who think it's a spare pair they can use when they connect their buttinskis and hear nothing |
| 01:23.55 | *** join/#asterisk N9URK (n=icechat5@cpe-075-178-088-168.ec.res.rr.com) |
| 01:23.56 | ping2921 | when I do dial dial() to my cell phone.... I get callerid unknown. |
| 01:24.05 | mafkees | JT: yeah. we have that here as well |
| 01:24.13 | N9URK | hi, would someone please make a call to 1@n9urk.ath.cx? |
| 01:24.14 | mafkees | I always connect them using 2 pairs |
| 01:24.32 | mafkees | 1 pair with dialtone, second (linked) pair for DSL |
| 01:25.21 | mafkees | always get a 2pair line |
| 01:25.25 | JT | what do you mean? |
| 01:25.27 | blitzrage | ping2921: most ITSPs are not going to let you set your own CID to the PSTN |
| 01:25.28 | JT | 2 pairs, why? |
| 01:25.57 | mafkees | JT: because that way you can transfer the main telephone number from the first pair without loosing connectivity |
| 01:26.25 | mafkees | it means: telephone stuff is using first pair, internet is on the second pair |
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| 01:26.55 | JT | so there is no dialtone on the dsl pair? |
| 01:27.32 | mafkees | there is |
| 01:27.38 | mafkees | but no phonenr |
| 01:27.57 | mafkees | there actually is a phoneline on it |
| 01:28.06 | mafkees | but you dont get the number from the telco |
| 01:28.16 | JT | and it doesn't call anything? |
| 01:28.17 | mafkees | .nl telco is lame |
| 01:28.25 | mafkees | it does |
| 01:28.33 | mafkees | I tested several times |
| 01:28.41 | JT | does send cid? |
| 01:28.50 | mafkees | just connect the second pair to a default analog phone |
| 01:29.06 | mafkees | the telephone nr is (line1 telephonenr)+1 |
| 01:29.23 | mafkees | but no inbound calls |
| 01:29.30 | mafkees | I guess telco is blocking it |
| 01:29.31 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:29.39 | mafkees | outbound calln go fine |
| 01:30.18 | mafkees | soon all this shit will be history |
| 01:30.40 | mafkees | there's this plan to 'fiberize' every home |
| 01:30.48 | mafkees | and every location |
| 01:30.53 | N9URK | hi, would someone please make a call to 1@n9urk.ath.cx? |
| 01:30.58 | JT | nice |
| 01:31.00 | mafkees | no more dsl |
| 01:31.04 | mafkees | uhhuh |
| 01:31.16 | mafkees | more and more home stuff is fiber |
| 01:31.19 | JT | fibre is swet |
| 01:31.22 | JT | sweet |
| 01:31.24 | mafkees | or ethernet |
| 01:31.51 | mafkees | every new house has ethernet or fiber |
| 01:32.07 | mafkees | and they are switching DSL connections to ethernet |
| 01:32.27 | mafkees | fiber to block equipment |
| 01:32.36 | mafkees | ethernet to your livingroom |
| 01:32.56 | mafkees | it's becoming more and more common here in .nl |
| 01:32.58 | JT | yeah we have nothing like that here |
| 01:33.30 | mafkees | new houses get 10/10mbit ethernet for like 45 euro/month |
| 01:34.05 | mafkees | 100/100 mbit is 75/month |
| 01:34.05 | JT | unlimited? |
| 01:34.09 | mafkees | uhhuh |
| 01:34.11 | mafkees | no cap |
| 01:34.16 | JT | static ip? |
| 01:34.21 | JT | can runs servers? |
| 01:34.27 | mafkees | yup, yup |
| 01:34.35 | *** join/#asterisk SkinTex (n=SkinTex@ip503c3583.speed.planet.nl) |
| 01:34.40 | mafkees | with option to get a /27 for 5 euro/month |
| 01:34.51 | JT | that's crazy |
| 01:34.55 | mafkees | no |
| 01:34.57 | mafkees | it rox |
| 01:35.02 | JT | almost no incentive to go into a datacentre |
| 01:35.05 | JT | well there's a bit |
| 01:35.34 | mafkees | 999% of datacenters here in .nl have dual uplink and dual power |
| 01:35.43 | mafkees | not easy to get that at home ;) |
| 01:35.50 | JT | heh, only dual uplink, lame :P |
| 01:35.50 | mafkees | 99% |
| 01:36.06 | mafkees | stupid ibook registering double keys.... |
| 01:36.28 | JT | looking at the best carrier neutral datacentre here, it's connected to about 15 different telco exchanges from 10 telcos |
| 01:36.33 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 01:36.39 | mafkees | well, I've seen cut uplink cables too much to trust 1 supplier |
| 01:36.40 | JT | dual power, water, N+1 generator/cooling/UPS |
| 01:37.14 | mafkees | uhhuh |
| 01:37.31 | mafkees | that's why I keep all important stuff in datacentre |
| 01:37.38 | JT | yeah, dual uplink just doesn't sound like that much |
| 01:37.40 | JT | indeed |
| 01:37.42 | mafkees | netapp disk backup |
| 01:37.57 | mafkees | carrier neutral uplinks |
| 01:38.10 | mafkees | 40 days diesel generator power |
| 01:38.19 | mafkees | security stuff |
| 01:38.25 | JT | they have 40 days of fuel on site? |
| 01:38.31 | mafkees | yeah |
| 01:38.37 | JT | that's crazy |
| 01:38.42 | JT | is the datacentre big? |
| 01:39.06 | mafkees | uhhuh |
| 01:39.24 | mafkees | 10.000 racks |
| 01:39.30 | JT | they must have millions of litres of fuel |
| 01:39.46 | mafkees | they have 1 location for servers |
| 01:39.58 | mafkees | and 3 locations for their power stuff |
| 01:40.03 | mafkees | including diesel |
| 01:40.06 | JT | location? |
| 01:40.11 | mafkees | .nl |
| 01:40.30 | JT | yeah i don't know what you mean 1 location for servers, 3 for power |
| 01:40.40 | mafkees | and deal with gasstation around the corner to supply stuff at hourly basis |
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| 01:41.06 | mafkees | JT: I dont know the sizes, but it's like this |
| 01:41.25 | mafkees | 1square mile for servers, 3 square miles for power |
| 01:41.30 | mafkees | they do it like that |
| 01:41.39 | distatica | sorry, I had a phone call. So this ztdummy, it says it's a 'timer' what does that mean? |
| 01:41.47 | JT | it isn't actually tha big is it? |
| 01:41.55 | mafkees | every square mile of servers has 3 square miles of power |
| 01:42.12 | mafkees | they can hold 10.000 servers at total |
| 01:42.19 | mafkees | based on 1U servers |
| 01:42.26 | JT | oh |
| 01:42.29 | mafkees | 42U in one rack |
| 01:42.30 | JT | only 10000 servers |
| 01:42.39 | JT | i thought you meant 10000 racks |
| 01:42.43 | mafkees | no |
| 01:42.49 | mafkees | 10000/42 |
| 01:42.57 | mafkees | oh wait |
| 01:42.59 | mafkees | 38 |
| 01:43.06 | JT | well the datacentre i'm talking about has the capacity for 3000 racks i think |
| 01:43.18 | mafkees | because you get ups and network controlled power regulator |
| 01:43.22 | mafkees | and a switch |
| 01:43.34 | JT | they have 22MVA of generators |
| 01:43.35 | mafkees | so thats 39U per rack server space |
| 01:43.40 | JT | eventually will have 44MVA |
| 01:43.59 | N9URK | did you all see the eclipse? |
| 01:44.05 | mafkees | no |
| 01:44.09 | N9URK | (if it has already happened in your timezone) |
| 01:44.19 | orkid | no |
| 01:44.20 | mafkees | JT: we host at a small datacentre |
| 01:44.41 | JT | ok |
| 01:44.54 | mafkees | but we like what they offer |
| 01:45.05 | mafkees | and 10000 servers is not that bad |
| 01:45.06 | JT | fair enough |
| 01:45.15 | JT | i've never heard of 40 days fuel |
| 01:45.22 | JT | does the place get snowed in or something? |
| 01:45.28 | mafkees | nope |
| 01:45.39 | mafkees | they want to offer something unique |
| 01:45.53 | JT | do they sell whole racks, or just per RU? |
| 01:46.02 | mafkees | both |
| 01:46.07 | mafkees | we hire a whole rack |
| 01:46.19 | mafkees | but for 160euro/month you get 4U |
| 01:46.27 | mafkees | including 2mbit link |
| 01:46.31 | JT | not bad |
| 01:46.32 | JT | hrm |
| 01:46.39 | mafkees | nope |
| 01:46.40 | orkid | something unique, lol |
| 01:46.45 | JT | orkid: ? |
| 01:46.56 | mafkees | we hire a whole 42U rack |
| 01:47.05 | orkid | <JT> does the place get snowed in or something? |
| 01:47.05 | orkid | <mafkees> nope |
| 01:47.05 | orkid | <mafkees> they want to offer something unique |
| 01:47.07 | mafkees | it's stuffed rigght now |
| 01:47.14 | JT | orkid: so, what's funny? |
| 01:47.21 | mafkees | but we are moving from 4U machines to 1U sun machines |
| 01:47.22 | JT | mafkees: stuffed? |
| 01:47.26 | JT | full |
| 01:47.35 | mafkees | so soon we will have lots of space left |
| 01:47.43 | orkid | offering uniquencess for its own sake is not very productive |
| 01:47.45 | JT | stuffed usually means broken or fucked ;) |
| 01:47.50 | distatica | still confused, is a timer to actual driver? Or what does ztdummy do? How does it relate to a softphone? I'm googling but quite lost, heh. |
| 01:47.55 | mafkees | JT: I meat full |
| 01:48.09 | JT | orkid: if you've been following, you'd see there is usefulness |
| 01:48.44 | distatica | .. is a timer the* actual driver..." |
| 01:49.20 | orkid | it sounds funny though.. of course having duel for you gens is useful, but doing it to be unique is kindof weird. doing it to be better than others, and having some othe rationale for it might be more desirable |
| 01:49.26 | orkid | whatever |
| 01:49.42 | mafkees | well |
| 01:49.46 | JT | orkid: 40 days of fuel is unique and VERY USEFUL |
| 01:50.02 | mafkees | 2 weeks ago AMS-IX went down because of power failure |
| 01:50.11 | mafkees | and our setup was still online |
| 01:50.13 | JT | lol |
| 01:50.22 | mafkees | because all their links are redundant |
| 01:50.30 | JT | power failure, did a distribution board blow up? |
| 01:50.41 | JT | or did they actually not have enough backup? |
| 01:50.44 | mafkees | they have peering with AMS-IX, DE-IX and BE-IX |
| 01:51.13 | mafkees | JT: power in amsterdam went down, and generator was too loaded to keep everything up |
| 01:51.24 | mafkees | so AMS-IX core switches went offline |
| 01:51.27 | mafkees | was funny to see |
| 01:51.34 | JT | dodgy |
| 01:51.39 | JT | sounds like a bad datacentre |
| 01:51.49 | mafkees | yeah |
| 01:51.57 | JT | how long was amsterdam's power down? |
| 01:52.02 | mafkees | .nl is bad at 'new' technoligy |
| 01:52.06 | mafkees | 45 minutes |
| 01:52.10 | JT | uhh |
| 01:52.14 | JT | that's really bad! |
| 01:52.19 | mafkees | uhhuh |
| 01:52.21 | JT | that the dc didn't survive |
| 01:52.28 | type0 | mafkees.. what are you using in the 1U sun machines? |
| 01:52.28 | mafkees | yeah |
| 01:52.36 | mafkees | type0: X2100 |
| 01:52.47 | type0 | i was looking at those |
| 01:52.51 | mafkees | X2100 M2 |
| 01:53.02 | mafkees | the ones with VT |
| 01:53.11 | type0 | I have a Ultra 20 M2 on the way right now |
| 01:53.16 | type0 | hows asterisk do on solaris? |
| 01:53.31 | mafkees | I run OpenBSD on them |
| 01:53.37 | JT | mafkees: this datacentre i'm looking at here has our government signals intelligence agency approval, i don't know if that's good or bad ;) |
| 01:53.46 | mafkees | OpenBSD and debian linux |
| 01:54.05 | type0 | I read something that Solaris 10 can do something like 1400 calls |
| 01:54.12 | mafkees | JT: means the power and uplink is good, but everything is tapped ;) |
| 01:54.13 | type0 | google asterisk solaris |
| 01:54.23 | JT | mafkees: heh |
| 01:54.44 | mafkees | type0: noone in our company knows solaris, that's why we went with OpenBSD and Debian |
| 01:54.54 | mmlj4 | type0: depends on lots of things... what model of sun box? how much RAM? what else is on the box? how dumb is the admin? etc. |
| 01:55.02 | type0 | google it |
| 01:55.05 | type0 | they give the specs |
| 01:55.11 | type0 | with the mtmalloc on.. its like 1400 calls |
| 01:55.27 | type0 | too bad the 1U sun machines are so fucking expensive |
| 01:55.29 | mafkees | we run 800 calls on a p4 |
| 01:55.34 | mmlj4 | per day? i'd believe that |
| 01:55.48 | mafkees | type0: no way. the X2100 is like 1200 euro |
| 01:55.59 | type0 | http://www.thrallingpenguin.com/articles/asterisk-solaris.htm |
| 01:56.20 | type0 | One Celeron 2.4 GHz with 512 MB RAM |
| 01:56.20 | type0 | One Sun Fire x2100 with Opteron 175 and 2 GB RAM |
| 01:56.42 | type0 | solaris was doing 1400 calls per second |
| 01:56.45 | mafkees | yeah, the RAM is expensive for them sun machines |
| 01:56.49 | type0 | at 28% cpu systemn |
| 01:56.56 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 01:57.12 | mafkees | I dont care about calls/second |
| 01:57.26 | mafkees | all i care about is: how many simultanious calls |
| 01:57.33 | mafkees | in a real life setup |
| 01:57.39 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:57.47 | DocHolliday | can anyone help me with my tftp server? |
| 01:58.13 | mafkees | I have no customer that has calls of 1 second |
| 01:58.28 | mafkees | they setup a call, and keep talking for at least 20 seconds |
| 01:58.50 | JT | mafkees: yeah, different metrics for different situation |
| 01:58.51 | JT | s |
| 01:58.56 | mafkees | indeed |
| 01:59.01 | JT | calls/sec is more relevant if you don't handle media |
| 01:59.09 | mafkees | JT: indeed |
| 01:59.25 | DocHolliday | anyone here either used atftp or tftp-server? |
| 01:59.42 | mafkees | DocHolliday: used both with success |
| 01:59.54 | DocHolliday | mafkees, and you are my hero (think this is the second time) |
| 02:00.15 | DocHolliday | can you help me setup atftp for my cisco phones? (firmware is already loaded.. just for reoccurring resets) |
| 02:00.19 | mafkees | and counting... |
| 02:00.28 | DocHolliday | hah |
| 02:00.49 | DocHolliday | i loaded the RPM on to my machine already.. just need help configuring it :P |
| 02:01.04 | mafkees | there is no configuration (in debian) |
| 02:01.13 | mafkees | it simply uses /tftboot as root dir |
| 02:01.30 | mafkees | put all you files there, chmod 755 them |
| 02:01.35 | mafkees | and your there |
| 02:01.39 | DocHolliday | how does it know where the tftp root is? |
| 02:01.39 | DocHolliday | oh really? |
| 02:01.59 | mafkees | DocHolliday: check your /etc/inetd.conf |
| 02:02.10 | mafkees | there it tells you the root dir |
| 02:02.24 | mafkees | sometimes it's /opt/tftroot |
| 02:02.46 | mafkees | every distro and every version of a distro can do that differently |
| 02:03.16 | DocHolliday | i dont have an inetd.conf :P |
| 02:03.18 | sweeper | locate ftw |
| 02:03.25 | mafkees | lol sweeper |
| 02:03.30 | mafkees | man ftw! |
| 02:04.04 | mafkees | actually I'm wearing this t-shirt: black with white print on the back 'RTFM' |
| 02:04.22 | mafkees | helps me a lot at work |
| 02:04.31 | DocHolliday | hah |
| 02:04.42 | DocHolliday | mafkees, okay well when i reset the phone it didn't pickup any files at all |
| 02:04.46 | mafkees | just turn your back at them and they will know what to do |
| 02:05.00 | mafkees | DocHolliday: did you tcpdump it ? |
| 02:05.11 | mafkees | that will tell you the location it's requesting |
| 02:05.39 | mafkees | wireshark can help as well |
| 02:06.07 | DocHolliday | mafkees, can i set the tftproot in atftp? |
| 02:06.11 | DocHolliday | (when i execute it?)\ |
| 02:06.19 | mafkees | yeah you can |
| 02:07.35 | mafkees | man atftpd |
| 02:07.35 | mafkees | it's been a long time since I used atftpd |
| 02:07.35 | mafkees | using OpenBSD now all the time for tftp/dhcp/firewall |
| 02:08.09 | mafkees | ok, I'm off to have some fun with MrsMafkees ;) |
| 02:08.32 | mafkees | bye all |
| 02:09.17 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 02:11.13 | DocHolliday | mafkees, quick question? |
| 02:12.50 | DocHolliday | anyone here used atftp? |
| 02:22.16 | *** join/#asterisk etfonhome (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 02:23.06 | etfonhome | Anyone tested SLA in 1.4.1? |
| 02:31.56 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 02:32.22 | rhombus | Is there a way to make it so that a caller can break out of a queue by pressing a key? |
| 02:35.19 | rhombus | hello? |
| 02:35.35 | orkid | hello :) |
| 02:35.44 | orkid | relax a bit eh? |
| 02:35.49 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 02:36.54 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 02:37.04 | Mahmoud | any free online asterisk servers that we can join? |
| 02:37.19 | Mahmoud | i remember once i saw links in asterisk.org or in its wiki page but unsure where is it now |
| 02:38.02 | DocHolliday | anyone used tftpserver? |
| 02:38.22 | Mahmoud | what about it? |
| 02:38.43 | DocHolliday | Mahmoud, trying to configure it to work with my cisco 7941g's |
| 02:39.01 | DocHolliday | firmware is already on the phones.. just having trouble configuring the tftpserver |
| 02:39.04 | Mahmoud | how is this related to aterisk? |
| 02:39.24 | DocHolliday | very simple.. people often use tftpserver in conjunction with asterisk |
| 02:39.47 | DocHolliday | thus if people use one they will usually know how to use the other :P |
| 02:39.51 | rhombus | I'm relaxed -- things were so quiet I thought maybe my IRC client was broken. |
| 02:40.35 | rhombus | the queue docs on the wiki don't mention how a caller might break out of a queue and leave a message instead... forgive me if it's obvious |
| 02:42.14 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 02:43.31 | rhombus | never mind, got it -- "context" parameter in queues.conf |
| 02:43.35 | rhombus | thanks |
| 03:00.40 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 03:01.28 | tzafrir | which tftp server exactly? |
| 03:01.46 | tzafrir | tftpd/inetd? tftp-hpa? atftpd? |
| 03:05.51 | *** join/#asterisk x86__ (n=x86@p3m/member/x86) |
| 03:11.48 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:11.56 | *** join/#asterisk step_quasar (n=step_qua@190.48.196.34) |
| 03:12.20 | *** join/#asterisk TheMahmoud (n=fake@unaffiliated/mahmoud) |
| 03:12.38 | TheMahmoud | 100% sure that my ISP blocks SIP based on application layer (patterm match) |
| 03:13.04 | ez` | this is bad ... |
| 03:13.14 | Mahmoud | i tested the network by NetCat and all went fine |
| 03:13.30 | Mahmoud | I used port 5060 by netcat tests and every thing is blazing fast |
| 03:13.41 | Mahmoud | but when it comes to exchanging SIP data, it gets dropped |
| 03:14.24 | Mahmoud | poor sip.. |
| 03:15.04 | *** join/#asterisk step_quasar (n=step_qua@190.48.196.34) |
| 03:15.11 | Mahmoud | iax is fine, but all softphones supporting iax2 suck |
| 03:15.24 | ManxPower | all softphones suck |
| 03:15.29 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:15.32 | Mahmoud | eyebeam is cool |
| 03:15.51 | Mahmoud | at leats sip phones have video support |
| 03:16.01 | Mahmoud | while iax has no video support |
| 03:16.26 | JT | Mahmoud: you mean rtp gets dropped? |
| 03:16.30 | step_quasar | |
| 03:16.30 | step_quasar | somebody that speaks Spanish ? |
| 03:16.39 | Mahmoud | JT, yeah |
| 03:16.56 | Mahmoud | ÇáÍãÇÑ ÇáÓÑíÚ |
| 03:17.04 | JT | that's totally differen't to sip getting dropped |
| 03:17.18 | Mahmoud | sorry i don't mean RTP heh |
| 03:17.27 | Mahmoud | i mean, SIP messages exchanged over UDP 5060 |
| 03:18.09 | Mahmoud | where are the good old days, where things were blocked based on port number |
| 03:18.17 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 03:18.31 | iceyp | hey guys, anyone know what i can use instead of mpg123 as it keeps running at 99% CPU |
| 03:20.40 | N9URK | hi, would someone please make a call to 1@n9urk.ath.cx? |
| 03:21.53 | *** join/#asterisk coppice (n=chatzill@249.193.17.210.dyn.pacific.net.hk) |
| 03:22.19 | Mahmoud | i'm 100% sure that they didn't block SIP for b/w issues. it's all about forcing people to use POTS network |
| 03:22.52 | ManxPower | iceyp: in 1.2 and later you don't have to use mpg123 |
| 03:34.33 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 03:40.41 | *** join/#asterisk shinux__ (n=shinux@195.166.241.139) |
| 03:41.41 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 03:42.15 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 03:50.26 | hohum | hey someone? |
| 03:50.29 | hohum | I have a SIP question |
| 03:51.14 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 03:51.15 | hohum | regarding a server transaction |
| 03:51.22 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 03:51.37 | hohum | am I supposed to fire timers on the server side of the transaction? |
| 04:05.03 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:07.43 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 04:09.45 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 04:10.34 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 04:10.53 | iceyp | ManxPower sorry was upgrading firmware on my adsl router.... how do i use something other than mpg123? |
| 04:10.55 | iceyp | I using 1.2 |
| 04:13.19 | *** join/#asterisk bmg505 (n=leon@196.209.250.158) |
| 04:13.38 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 04:14.01 | *** join/#asterisk tecolote99 (n=ruben@netblock-68-183-84-199.dslextreme.com) |
| 04:17.08 | bkruse_home | anyone recommend a SIP book? or has anyone even bought one, ever? :P |
| 04:18.32 | shido6 | Mamoud u still having sip issues? tried iax? |
| 04:18.41 | shido6 | there's an "h" in there somewhere |
| 04:25.32 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 04:29.05 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 04:36.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 04:40.03 | hohum | bkruse_home: who needs a book? |
| 04:40.05 | hohum | rfc3261 |
| 04:41.12 | bkruse_home | hohum: i figured someone would sa that |
| 04:41.14 | bkruse_home | sa that |
| 04:41.17 | bkruse_home | say* |
| 04:41.19 | ping2921 | how do i set the callerid number before calling out using dial() cmd? |
| 04:41.26 | bkruse_home | i want something thats more, where SIP is going, has been, etc etc |
| 04:41.33 | bkruse_home | not just protocol specific, see what i mean? |
| 04:41.48 | JunK-Y | i dont imagine all employees in a company reading the same rfc. |
| 04:41.56 | bkruse_home | no |
| 04:42.13 | bkruse_home | but possible, and i might have already :] |
| 04:42.25 | JunK-Y | ping2921: Set(CALLERID(num)=5145551234) |
| 04:42.51 | bkruse_home | hohum: make sense? |
| 04:43.05 | hohum | sure |
| 04:43.09 | hohum | I can summarize it for you |
| 04:43.22 | hohum | SIP wasn't much, now it is, and later it'll be more |
| 04:43.52 | bkruse_home | hohum: alright! thanks! |
| 04:43.53 | bkruse_home | ....... |
| 04:43.58 | hohum | I mean really, protocols don't make much for a biographical page turner |
| 04:44.11 | bkruse_home | hohum: then you would be surprised. |
| 04:44.18 | bkruse_home | that was just an example, btw |
| 04:44.25 | hohum | SIP is used in too many things to fit neatly into a book |
| 04:45.03 | bkruse_home | hohum: 1,300 pages? |
| 04:45.10 | coppice | I guess a SIP book should start "Long ago and far away lived a man who always smoked the very best stuff." |
| 04:45.12 | bkruse_home | and yes, its just a session initiated protocol :P |
| 04:45.22 | bkruse_home | coppice: ha, i like! |
| 04:45.31 | Qwell | coppice: Why would a SIP book start off by explaining h323? |
| 04:46.09 | bkruse_home | ouch |
| 04:46.16 | bkruse_home | but true |
| 04:46.17 | coppice | SIP was only developed to make H.323 look good. MGCP was only developed to make both H.323 and SIP look good. |
| 04:46.30 | hohum | qwell: If you wanted to explain h323, the book should start off with "A bunch of sweeds got together and said 'how can we turn our garbage loose on the IP world" |
| 04:46.30 | bkruse_home | coppice: haha, so true |
| 04:47.00 | ping2921 | is it normal that when queue makes a holdtime announcement, the moh starts from zero when the announcement is over? |
| 04:47.46 | ping2921 | I would expect the moh to resume from the location before holdtime announcement. |
| 04:47.47 | hohum | the ITU-T should stay out of the IP world and stick to TDM where they belong |
| 04:48.14 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:49.33 | coppice | the IETF has actually done a worse job |
| 04:50.38 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 04:57.01 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 05:12.45 | *** part/#asterisk defend (i=defend@38.113.5.165) |
| 05:12.47 | *** part/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 05:25.47 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:49.17 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:50.35 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 05:50.41 | ZX81 | Hi all, whats the security vulnerability in the recently released asterisks? |
| 05:50.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 05:54.45 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.in.comcast.net) |
| 05:55.23 | *** join/#asterisk peanutb (n=paulb@71.197.218.111) |
| 05:56.22 | peanutb | I am having some trouble figuring out exactly what asterisk is, does anyone know of something to introduce me to the basics? |
| 06:05.27 | tzafrir_laptop | duh, just ran something like: cat chan_zap.c | le #should have been 'less' |
| 06:06.25 | tzafrir_laptop | It turns out that le is another text editor that I happened to have installed. It reads files from stdin |
| 06:06.34 | tzafrir_laptop | But s l o w l y |
| 06:06.52 | tzafrir_laptop | ~line per second |
| 06:08.21 | ZX81 | peanutb: www.asterisk.org |
| 06:08.48 | ZX81 | http://www.sineapps.com/news.php?rssid=1695 |
| 06:10.23 | JunK-Y | you could add that trixbox, packaged asterisk and openpbx (do we care) are affected too. |
| 06:12.13 | *** join/#asterisk sifusam (n=sifusam@nat-vlan0200.sat4.rackspace.com) |
| 06:12.15 | ZX81 | :) |
| 06:12.16 | ZX81 | cool |
| 06:16.45 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 06:17.36 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 06:27.31 | *** join/#asterisk jjshoe (i=jjshoe@39.sub-75-214-48.myvzw.com) |
| 06:27.34 | jjshoe | re |
| 06:29.53 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 06:30.17 | *** join/#asterisk techie (n=gus@voipops.net) |
| 06:30.18 | orkid | does anyone have experience with openVOX cards? |
| 06:38.52 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 06:40.00 | *** join/#asterisk kgx (n=karuna@port-60-234-196-160.jet.net.nz) |
| 06:41.21 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 06:45.13 | *** part/#asterisk [shodan] (n=shodan@ip176.96-113-216.pppoe1.joliette.intermonde.net) |
| 06:52.15 | jjshoe | what's the proper way to create a call from c? should I be hooking the asterisk manager? making a call file? or calling some c function? |
| 06:57.58 | ZX81 | use the originate command via the manager is easiest |
| 06:58.23 | jjshoe | actually I would argue that writing a file from c is easier then telneting into the manager :) |
| 06:58.24 | Nugget | telnet is eeeeeeevil! |
| 07:00.47 | *** part/#asterisk sifusam (n=sifusam@nat-vlan0200.sat4.rackspace.com) |
| 07:11.36 | ZX81 | try it :) |
| 07:21.11 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:21.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:26.44 | *** join/#asterisk linlin (n=will@c-67-184-229-49.hsd1.il.comcast.net) |
| 07:27.09 | linlin | what are the main directoris and files i need to remove in order to completly uninstall asterisk |
| 07:27.26 | linlin | i botched my install awhile back and i want to have a fresh start |
| 07:28.26 | *** join/#asterisk VTD (n=Timothy@ppp-70-255-106-4.dsl.hstntx.swbell.net) |
| 07:32.13 | tzafrir | jjshoe, writing a file from a shell script with netcat is easier that writing a file in C |
| 07:32.51 | tzafrir | jjshoe, or a little shell scirpt |
| 07:33.16 | tzafrir | shell scripts are generally better for things that involve many file operations |
| 07:35.36 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 07:37.36 | *** join/#asterisk h3x (n=hex@64.192.116.17) |
| 07:37.58 | h3x | whe |
| 07:38.22 | h3x | it looks a lot more mature now |
| 07:39.33 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 07:52.20 | *** join/#asterisk Witwolf (n=carel@c2-242-2.eno.dial.mweb.co.za) |
| 07:53.33 | Witwolf | Hi, in everyone's opinion, what is the best hardware phone to get for Asterisk? |
| 07:56.43 | *** join/#asterisk Witwolf (n=carel@c2-242-2.eno.dial.mweb.co.za) |
| 07:57.30 | Witwolf | Hi, sorry, my ethernet connection slipped out of my laptop. In everyone's opinion, what is the best hardware phone to get for asterisk? |
| 07:58.18 | JT | polycom |
| 08:00.10 | Witwolf | What about snom? Our local Digium distributer recommended Snom. |
| 08:00.30 | JT | apparently they're pretty flaky and the audion isn't good |
| 08:00.35 | JT | ~phones |
| 08:00.40 | jbot | [phones] http://bani.anime.net/phones/. SIP Hardphones in order of quality/suggestability : Polycom (any), Aastra 480i, Cisco 7940(+), Linksys SPA-9XX |
| 08:00.40 | JT | audio |
| 08:02.54 | Witwolf | I must say, I do not like their look. I really like the linksys phones, but how functional are they? |
| 08:03.17 | JT | look of what? |
| 08:03.29 | JT | linksys are ok, nothing spectacular |
| 08:03.33 | Witwolf | They look cool. |
| 08:04.28 | JT | polycom, aastra and cisco all look better than linksys |
| 08:05.07 | Witwolf | OK |
| 08:05.27 | Witwolf | I have quite limited options here in South Africa |
| 08:05.44 | JT | you can't import stuff? |
| 08:05.46 | Witwolf | My distributer does not have any of those makes |
| 08:06.13 | Witwolf | It just such a mission to import stuff! |
| 08:06.19 | JT | who cares what your distributor has, order what you want directly |
| 08:06.31 | Witwolf | You end up pay 30 - 40 % more |
| 08:06.46 | JT | i end up paying less by importing a lot of stuff |
| 08:06.51 | Witwolf | Shipping is not cheap! |
| 08:07.19 | JT | your distributors prices probably aren't either |
| 08:07.38 | Witwolf | Yes I know. |
| 08:07.51 | JT | it might be cheaper to import better products |
| 08:07.58 | Witwolf | But would it stil be cheaper if I only buy one or two phones? |
| 08:08.02 | *** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/sustaining/Corydon76-home) |
| 08:08.02 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 08:08.20 | JT | i don't know, you have to do the sums for your situation |
| 08:08.49 | Witwolf | YEAH, the polycom site is down! |
| 08:08.59 | Witwolf | Just my luck! |
| 08:09.27 | JT | weird, there are plenty of online stores with polycom prices though |
| 08:11.22 | Witwolf | Does not help being on a 56K dialup modem as well! LOL |
| 08:15.34 | linlin | forgive me, its been awhile, is it still required with asterisk 1.4 to download the asterisk-addons and asterisk-sounds packages or are they built in? |
| 08:17.09 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 08:39.44 | Witwolf | JT, I looked at all the phones and I like the Cisco 7941g the most. They say that the phone is a bit different than the other ones, do you know if it still works well? |
| 08:41.36 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 08:41.41 | JT | ciscos are a pain |
| 08:41.51 | JT | you need to get sip firmware |
| 08:42.13 | JT | and cisco aren't friendly unless you're on a support contract |
| 08:47.09 | *** join/#asterisk lorinc (n=ang@pool-3981.adsl.interware.hu) |
| 08:47.46 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 08:53.18 | linlin | whats the default location for asterisk-core-sounds? |
| 08:53.41 | h3x | look at asterisk.conf |
| 08:53.58 | h3x | some platforms move it |
| 08:55.06 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 08:59.56 | Witwolf | Where would I find information for redoing all the asterisk sounds with a South Africa Accent, I would at a Sound Studios, so it would be very easy to get a Voice over artist and record all the stuff again. |
| 09:07.44 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:07.54 | *** join/#asterisk lorinc (n=ang@pool-3981.adsl.interware.hu) |
| 09:11.05 | *** join/#asterisk lorinc (n=ang@pool-3981.adsl.interware.hu) |
| 09:14.03 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 09:17.02 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 09:48.40 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 10:03.26 | *** join/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 10:11.06 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 10:19.36 | *** join/#asterisk metacolo (n=ryan@ofb.net) |
| 10:19.52 | metacolo | hi. i upgraded 1.4.0 to 1.4.1 and now my nokia e61 sip phone won't register |
| 10:20.33 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 10:24.17 | orkid | that sucks metacolo |
| 10:25.59 | metacolo | has anyone done the upgrade? |
| 10:27.10 | metacolo | sip debug shows what look like legitimate messages (MD5, right realm) |
| 10:38.21 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 10:38.37 | metacolo | is there any list of the changes with 1.4.1? |
| 10:46.13 | *** join/#asterisk friedrich| (n=friedric@e177250021.adsl.alicedsl.de) |
| 10:51.25 | *** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com) |
| 10:52.05 | EmleyMoor | When I do a database show, the leading / is missing from one of the entries - why would that be? It's not the entry, nor is it the one above, at fault |
| 10:54.25 | EmleyMoor | If I show only the family in which it occurs, all entries look OK |
| 11:03.00 | tsurko | Hello, is there a way to get the phone number dialed on a specific zap channel? |
| 11:07.57 | tzafrir | tsurko, the name of the channel? |
| 11:08.20 | tzafrir | or the extension? |
| 11:08.33 | tzafrir | which type of zaptel: analog or digital? |
| 11:08.46 | tsurko | tzafrir, the extension |
| 11:08.59 | tsurko | digital I suppose - it's a PRI interface |
| 11:09.04 | tzafrir | $(EXTEN) ? |
| 11:12.42 | tsurko | I'm trying to give a higher priority to the SOS calls. I want to drop a call if there are no free channels, but I don't want to drop a SOS call. So I'm trying ot check the dialed extension on every channel before I free it. Can $EXTEN help me in this case? |
| 11:17.38 | RoyK | tsurko: just Dial and if it doesn't get through, check DIALSTATUS and take actions from there on |
| 11:20.39 | tsurko | RoyK, this is nice, but how to understand wheather to drop the call or not? |
| 11:29.54 | *** join/#asterisk coppice (n=chatzill@249.193.17.210.dyn.pacific.net.hk) |
| 11:32.05 | *** join/#asterisk shodan (n=shodan@ip176.96-113-216.pppoe1.joliette.intermonde.net) |
| 11:32.18 | RoyK | tsurko: dunno... perhaps GROUP is easier? |
| 11:32.57 | JT | tsurko: i have thought about this before, but i gave up because it was hard :P |
| 11:33.10 | JT | you need to play with global vars or ast db |
| 11:33.51 | tsurko | yes, I have something like that now, but I want to make it simlier |
| 11:33.59 | JT | does it work? |
| 11:34.50 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 11:37.13 | *** join/#asterisk NirS (n=Nir@87.68.42.164.cable.012.net.il) |
| 11:37.17 | NirS | hello all |
| 11:37.19 | tsurko | I guess so - someone else have done it. |
| 11:37.22 | NirS | anybody home ? |
| 11:37.47 | tsurko | Unfortunately, I'm not at home:( |
| 11:38.04 | NirS | anyone with experience with Asterisk Static RealTime and zapata.conf ? |
| 11:39.48 | NirS | hello? |
| 11:39.50 | NirS | anybody here ? |
| 11:39.55 | NirS | looks like everybody is asleep |
| 11:40.41 | tsurko | NirS, ask your question and somebody may help ypu |
| 11:40.56 | NirS | well, here's a questin |
| 11:40.58 | NirS | quesiton |
| 11:41.00 | NirS | question |
| 11:41.19 | NirS | in Asterisk Static RealTime you are able to define various configuration files that you want to manage |
| 11:41.41 | NirS | my question is, lets say I have a zapata.conf file which has at the end the following include lines: |
| 11:41.48 | NirS | #include zapata_fxs.conf |
| 11:41.53 | NirS | #include zapata_fxo.conf |
| 11:41.59 | NirS | #zapata_pri.conf |
| 11:42.27 | NirS | then, I have static RealTime control the configuration of zapata_fxo.conf, zapata_fxs.conf, zapata_pri.conf |
| 11:42.31 | NirS | will that work ? |
| 11:43.01 | NirS | if I understand correctly, as long as I don't store the actual zapata.conf file in the database, that will be handled by the static file |
| 11:43.08 | NirS | while the rest is managed by the database |
| 11:44.32 | NirS | hmmm... either everybody is asleep, or no one has a clue |
| 11:48.28 | JT | it's also possible you are very demanding and impatient |
| 11:50.53 | NirS | that may also be an answer ;-) |
| 11:50.53 | NirS | I'm simply on the other side of the world, that is all |
| 11:50.53 | JT | yeah, you might not get it in a whole 3 minutes |
| 11:50.53 | NirS | it's 13:50 over here |
| 11:50.53 | NirS | ;-) |
| 11:51.00 | JT | it's 2250 here |
| 11:51.18 | JT | people just don't like it when they're told they don't have a clue |
| 11:53.51 | coppice | dunno. when you tell them that, they usually respond with more information to emphasise just how clueless they really are :-) |
| 11:54.31 | JT | nice and cynical ;) |
| 11:58.18 | coppice | not cynical at all. there's almost a standard template for the response. "I've been working in XXXX for 10/20/30 years" is always an element, like you'll be more impressed if you know just how long its taken them to learn nothing. |
| 11:58.26 | *** join/#asterisk shodan (n=shodan@ip176.96-113-216.pppoe1.joliette.intermonde.net) |
| 11:58.59 | NirS | good one |
| 11:59.15 | NirS | well, I guess that my israeli tounge and manner still make a little rude from time to time |
| 11:59.32 | NirS | but, on the other hands, I tried it out, and it appears that it actually works |
| 11:59.39 | NirS | which is preaty cool |
| 11:59.40 | *** join/#asterisk hellop (n=hellop@udp112969uds.hawaiiantel.net) |
| 12:00.32 | hellop | Hello |
| 12:00.55 | *** join/#asterisk shodan (n=shodan@ip176.96-113-216.pppoe1.joliette.intermonde.net) |
| 12:01.19 | hellop | Is there a way to send an emil from extensions.conf? |
| 12:01.34 | JT | coppice: nice, i can definately agree with that |
| 12:02.19 | NirS | sending an e-mail from extensions.conf ? |
| 12:02.28 | NirS | you'll have to be a little bit more specific |
| 12:03.13 | hellop | NirS, I'd like to append a string var in extensions.conf, and then email the contents of that var |
| 12:05.25 | NirS | well, then simply do a Set(VAR=something) |
| 12:05.35 | NirS | then issue a system call to mail or something like that |
| 12:05.42 | hellop | is it even possible to SMTP messages these days? |
| 12:05.49 | hellop | from localhost |
| 12:06.10 | JT | mail? |
| 12:08.46 | *** join/#asterisk infi (n=infi@about/linux/staff/infi) |
| 12:09.03 | infi | greetings. what is the correct terminology for the concept of an "outdial" ? |
| 12:09.17 | infi | I am having trouble searching for outdial providers :| |
| 12:09.38 | hellop | JT, I try mail, but it just hangs after asking for subject... |
| 12:10.04 | *** join/#asterisk NirS_ (n=Nir@87.68.42.164.cable.012.net.il) |
| 12:10.32 | mzb | hellop: man mail? |
| 12:11.16 | hellop | mzb, barely understandable |
| 12:12.01 | mzb | you may need to investigate sudo if you have permissions+path problems |
| 12:12.11 | mzb | $ man mail |
| 12:12.36 | hellop | for instance, what is [...]? the message body? |
| 12:12.37 | infi | what I mean is a remote provider, who is willing (for a fee or free) to provide a VOIP->PSTN connection in a remote location. |
| 12:13.04 | mzb | mail [-eIinv] [-a header] [-b bcc-addr] [-c cc-addr] [-s subject] to-addr [...] |
| 12:13.30 | hellop | [-- sendmail-options [...]] |
| 12:14.09 | hellop | also that... You might guess that [...] is the message body, but then... there's 2 of them |
| 12:14.51 | hellop | At least one example of sending an email in the man page would be a nice feature. |
| 12:15.58 | *** part/#asterisk infi (n=infi@about/linux/staff/infi) |
| 12:16.12 | mzb | I see what you mean |
| 12:21.39 | hellop | If we could figure out a way to send an email from the Linux command line, we'd be millionaires. |
| 12:22.20 | hellop | I've also been working on an automatic punch card sorter for the past 50 years. |
| 12:22.35 | mzb | $ echo "This is a test mail |
| 12:22.49 | mzb | with two lines" > testmail |
| 12:23.09 | mzb | $ mail me@localhost -s "Test mail" < testmail |
| 12:24.17 | mzb | I'm guessing you should be able to use here strings as well with "<<" ... but YMMV |
| 12:24.47 | mzb | hmmm... although I can't prove it works atm ;) |
| 12:24.55 | hellop | me either |
| 12:27.36 | mzb | that last line might need an extra return |
| 12:30.09 | mzb | or pipe it into mail? ... sorry just guessing ... but interested ;) |
| 12:31.46 | hellop | mzb, Don't mail hosting companys these days block SMTP from an ip with no DNS/MX records? |
| 12:32.08 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:33.06 | mzb | I guess that would depend on the hosting coy? |
| 12:33.52 | mzb | I can confirm that this works: |
| 12:34.05 | mzb | $ echo "This is a test mail |
| 12:34.06 | mzb | > with two lines" | mail me@localhost -s "Test mail" |
| 12:34.32 | mzb | but external mails are prob another story ;) |
| 12:34.35 | mzb | I'll try |
| 12:39.14 | mzb | hmm ... may not have waited long enough ... perhaps "sendmail options" need to be employed? |
| 12:39.19 | mzb | bbl |
| 12:40.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:42.21 | *** join/#asterisk sandorp (n=sandor@dhcp-242.phx3.llnw.com) |
| 12:44.22 | hellop | mzb return then ctrl-D to send message... |
| 12:45.43 | hellop | mzb, so I'll probably have to use a legit SMTP server... |
| 12:46.33 | JT | all you need to do is have your local mta setup properly to use isp smtp server or whatever |
| 12:46.38 | JT | then sendmail command will work fine |
| 12:48.14 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 12:49.13 | hellop | JT, think I can use another SMTP server then my ISP's? |
| 12:50.01 | JT | well i assume you're actually allowed to send with your isp's server |
| 12:52.36 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 12:52.52 | sandorp | hellop: any properly configured SMTP server will refuse to forward email for any of your ISP's servers; if you can't use your ISP's SMTP server, you may have to set up your own; you should contact your ISP before doing that |
| 12:55.05 | *** part/#asterisk bulle (n=bulle@c-db2971d5.015-48-626c671.cust.bredbandsbolaget.se) |
| 13:00.16 | hellop | JT, well, not until Monday. Maybe, I can ssh into webserver and send a message from there. |
| 13:04.04 | hellop | So, mail command works from my commercial webserver... |
| 13:08.42 | hellop | sandorp, since my ISP blocks port 25, I can probably use a different port, and then port translation on SMTP server. |
| 13:12.13 | sandorp | hellop: did your ISP provide you with an IMAP or POP email account? if so, they should have given you access to their SMTP server as well; if not, then you probably need to use your own; how do you typically send regular email? |
| 13:13.20 | *** join/#asterisk FastFeet (n=FastFeet@CPE0013109fd25b-CM000f9fa60d7a.cpe.net.cable.rogers.com) |
| 13:15.49 | FastFeet | Question: Do I have to Backup my /etc/asterisk/ config(s) then uninstall and then re compile and reinstall asterisk just to upgrade from version 1.4.0 to 1.4.1? |
| 13:16.08 | FastFeet | Is there an easier way? |
| 13:17.30 | FastFeet | Ahh never mind, I found the UPGRADE.txt file... |
| 13:17.31 | FastFeet | Thanks |
| 13:17.39 | FastFeet | <--- Slaps Head |
| 13:17.56 | *** join/#asterisk sandorp (n=sandor@dhcp-242.phx3.llnw.com) |
| 13:18.13 | hellop | sandorp, no access to ISP SMTP.. |
| 13:18.33 | hellop | but my webhost already has smtp listening on a non-standard port |
| 13:19.46 | sandorp | hellop: sounds like you will have to use your own smtp server or push emails to the web server and resend |
| 13:22.52 | hellop | thanks for the help sandorp |
| 13:34.25 | *** part/#asterisk kezza491 (n=opera@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 13:36.33 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 13:39.04 | *** join/#asterisk malwcal (n=malcolm@ppp67-89.lns3.adl2.internode.on.net) |
| 13:40.32 | malwcal | Hello. I have a (hopefully) quick question i have been unable to answer via googling... |
| 13:43.38 | malwcal | I have a SIP phone (cisco 7912) which has a button labeled "Transfer to vm". This forwards an incoming call to the phone number defined for messages. The problem is, in extensions.conf I have this extension defined for VoiceMailMain, which is not what is needed for this situation. |
| 13:46.00 | malwcal | Is there an easy way to say "If this is a forwarded call then run VoiceMail, otherwise run VoiceMailMain"? |
| 13:54.22 | FastFeet | Question: Do I have to Backup my /etc/asterisk/ config(s) then uninstall and then re compile and reinstall asterisk just to upgrade from version 1.4.0 to 1.4.1? |
| 13:54.30 | FastFeet | Is there an easier way? |
| 13:54.50 | *** join/#asterisk cinthia (n=ccccc@host177-114.pool8250.interbusiness.it) |
| 13:54.53 | FastFeet | UPGRADE.txt does tell me anything about upgrading |
| 13:55.02 | FastFeet | doesn't* |
| 13:55.49 | FastFeet | Don't see anything on the Wiki how do so, nor does the handbook |
| 13:55.57 | *** part/#asterisk cinthia (n=ccccc@host177-114.pool8250.interbusiness.it) |
| 13:56.38 | FastFeet | Do I just recompile and install over top of my exsisting install of 1.4.0? |
| 13:59.35 | mafkees | malwcal: why not specify a seperate extension for this button ? |
| 13:59.38 | malwcal | mafkees: There is one extension defined for voicemail, and that is used when you press the :"messages" button to retrieve voicemail, and also used when transferring. |
| 14:00.17 | mafkees | hhmm |
| 14:00.23 | mafkees | maybe you can do: |
| 14:00.38 | mafkees | exten => vmexten/cid_of_phone,1,VoicemailMain() |
| 14:00.46 | mafkees | exten => vmexten,1,Voicemail() |
| 14:00.53 | mafkees | </pseudo code> |
| 14:01.52 | malwcal | OK, I will try. |
| 14:02.00 | *** join/#asterisk Ebola (n=Ebola@host86-143-156-147.range86-143.btcentralplus.com) |
| 14:04.40 | chrisknight | Is there a way I can trick my cisco 7960 into connecting to a forwarded tftp server? Not sure if I can explain this right... |
| 14:05.59 | chrisknight | I have a 7960 that I cant change the config on because its password protected... Im trying to send it new firmware from my tftp server... |
| 14:07.54 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:09.53 | reber | hi |
| 14:10.58 | malwcal | chrisknight: might be possible, although the easiest way is just reset it to factory defaults... |
| 14:13.22 | sandorp | chrisknight: or assign the IP of the "other" tftp server to your tftp server; most systems support multiple IP's per NIC; just make sure the phone and server are physically attached to the name network (i.e. eliminate the need for a gateway) |
| 14:13.46 | sandorp | oops: s/name/same/ |
| 14:17.42 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 14:17.54 | Mahmoud | i wonder why my IPFW is not blocking SIP's RTP packets although I set a deny any any at the end |
| 14:19.34 | Mahmoud | i permited udp 5060 and a deny any any statement at the end |
| 14:19.50 | Mahmoud | i can't find why RTP packets (using ports other than 5060 are still passing through |
| 14:20.02 | Mahmoud | it's nice to see it working, but i want to find the answer "why" it's working |
| 14:21.35 | mafkees | you have a permit rule for the other side ? |
| 14:22.23 | Mahmoud | it's actually in my LAN |
| 14:22.31 | Mahmoud | a single broadcast doimain (home network) |
| 14:23.37 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 14:23.59 | FastFeet | Do I just run ./configure, make menuselect, make install, over top of my already existing Asterisk 1.4.0 to upgrade to 1.4.1 ? |
| 14:24.51 | mafkees | Mahmoud: check if you have any permit rules for your local lan |
| 14:24.56 | mafkees | FastFeet: should work indeed |
| 14:25.00 | FastFeet | thanks |
| 14:25.08 | mafkees | make backups first ;) |
| 14:25.24 | FastFeet | of course, surprisingly how little information there is about this... |
| 14:25.26 | Mahmoud | oh yeah, i have a permit rule heh |
| 14:25.35 | FastFeet | thanks again |
| 14:25.58 | Mahmoud | i used tp permit any to any for my lan pcs, this includes the server it self |
| 14:26.06 | mafkees | ;) |
| 14:32.23 | malwcal | mafkees: Thanks that worked (once I fixed some stupid typos... ) |
| 14:32.43 | mafkees | malwcal: ur welcome |
| 14:33.20 | malwcal | Now the only issue with my 7912 phone is poor audio quality... |
| 14:37.49 | reber | x-lite (softphone) doesn't connect to asterisk. Any ideas if this could be the problem : Mar 4 15:24:42 WARNING[11713]: cdr_addon_mysql.c:295 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf |
| 14:38.16 | FastFeet | mafkees: Thanks for your help.... It worked just fine!! |
| 14:39.15 | FastFeet | :reber the error you posted has nothing to do with x-lite not connecting. |
| 14:39.59 | FastFeet | :reber the error has to do with keeping Call records with MySQL. My guess you do not have the MySQL client installed. |
| 14:40.24 | FastFeet | :reber I use x-lite a lot, so lets get this figured out.. |
| 14:41.13 | FastFeet | reber: You have your SIP.conf and users.conf configured? |
| 14:42.01 | reber | FastFeet, i have configured sip.conf and extensions.conf |
| 14:42.45 | *** join/#asterisk marc\cba (n=marc@cpc1-whit2-0-0-cust972.cdif.cable.ntl.com) |
| 14:42.50 | FastFeet | reber: Ok then, this should help you.... : http://www.asteriskguru.com/tutorials/xlite_softphone.html |
| 14:43.00 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 14:43.30 | marc\cba | folks im confused, my x-lite client is behind a home NAT gateway, my asterisk box is on a public ip elsewhere. Im registered. When i make a call the first packet from x-lite has my LAN ip address in it |
| 14:43.50 | marc\cba | whats puzzling is that asterisk starts trying to stream the audio then, x-lites private lan ip |
| 14:43.57 | marc\cba | where could i have gone wrong? |
| 14:44.00 | reber | FastFeet, i know this link and just already read it. Here is my config : http://pastebin.ca/380974 |
| 14:45.02 | FastFeet | Ok then |
| 14:45.04 | FastFeet | I see your problem |
| 14:45.17 | FastFeet | you need to add to your SIP.conf |
| 14:45.48 | FastFeet | just above secret=password add this: |
| 14:46.04 | FastFeet | user=2006 |
| 14:46.09 | FastFeet | and for your other account |
| 14:46.14 | FastFeet | add user=2007 |
| 14:46.19 | FastFeet | try it. |
| 14:46.33 | FastFeet | reber: get that? |
| 14:46.36 | *** part/#asterisk sandorp (n=sandor@dhcp-242.phx3.llnw.com) |
| 14:46.40 | reber | ok, i try |
| 14:48.20 | FastFeet | reber: That should work, if not show me your users.conf file |
| 14:50.19 | marc\cba | FastFeet any suggestions for me? |
| 14:50.27 | FastFeet | reber: That should work, if not show me your users.conf file |
| 14:50.29 | reber | FastFeet, it doesn't work, i still get the configuration window after x-lite tries to connect. http://pastebin.ca/380985 for the logs |
| 14:50.42 | reber | i'm pasting you users.conf |
| 14:51.44 | reber | FastFeet, i don't have any users.conf. Is this file mandatory ? |
| 14:51.58 | FastFeet | what version of Asterisk? |
| 14:52.20 | reber | 1.2.14 |
| 14:52.47 | FastFeet | That maybe why....Although I am using 1.4.0 and 1.4.1 |
| 14:52.54 | FastFeet | maybe different |
| 14:53.13 | JT | users.conf is optional |
| 14:53.17 | JT | and not really recommended |
| 14:53.44 | *** join/#asterisk onesandzeros (n=chris@softbank220041252002.bbtec.net) |
| 14:53.51 | *** part/#asterisk onesandzeros (n=chris@softbank220041252002.bbtec.net) |
| 14:54.01 | reber | mmm ... Any other ideas ? |
| 14:55.35 | FastFeet | not really, your SIP.conf looks fine now taht you added user= |
| 14:56.29 | reber | isn't it "username" instead of "user" ? |
| 14:56.51 | FastFeet | sec.... |
| 15:00.23 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 15:00.24 | FastFeet | here is a copy of my old but working sip.conf |
| 15:00.26 | FastFeet | http://pastebin.ca/380995 |
| 15:01.00 | reber | ok, can you paste your extensions.conf too ? |
| 15:01.09 | FastFeet | and yes your right |
| 15:01.13 | FastFeet | it is username= |
| 15:01.19 | FastFeet | my bad |
| 15:01.28 | FastFeet | try that |
| 15:03.09 | FastFeet | <--- Slaps Head |
| 15:03.52 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 15:04.05 | chrisknight | malwcal: sandorp: I cant reset the phone. It wont respond to **# or holding # during bootup. The tftp address password protected in the phone is 207.90.66.xx. I need it to access 172.16.16.201/24 |
| 15:04.29 | FastFeet | reber: Probley worked now that you fixed user= to username= |
| 15:05.40 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:06.25 | reber | FastFeet, it doesnt |
| 15:06.31 | malwcal | chrisknight: does the phone have a static IP address? |
| 15:06.35 | FastFeet | lol |
| 15:06.37 | FastFeet | weird |
| 15:07.04 | FastFeet | you did a reload from the CLI ? |
| 15:07.18 | chrisknight | It gets an address via my dhcp server. |
| 15:07.33 | reber | FastFeet, i restarted asterisk _and_ x-lite |
| 15:07.58 | FastFeet | reber: I'll see if can find a copy of my extension.conf |
| 15:08.15 | *** join/#asterisk Hypnotek (n=sdfjg@196.203.77.237) |
| 15:09.05 | malwcal | chrisknight: if you can control the dhcp server you might be able to force another tftp server via dhcp options. If this fails, try to run a dhcp server on server with the phone plugged in via a crossover cable. |
| 15:09.15 | reber | are you shure that this *error* problem about mysql doesn't block asterisk to work properly ? |
| 15:09.15 | chrisknight | I was going to use http://www.boutell.com/rinetd/ to forward the packets but that only works for TCP. |
| 15:10.04 | FastFeet | absolutly |
| 15:11.25 | FastFeet | reber: http://pastebin.ca/381007 |
| 15:11.32 | chrisknight | I have full control over the network/servers. This is at my house, I got a phone from work but its password protected. & the pass is not "cisco". I was trying to add the tftp option to my linux dhcpd config.. maybe im doing it wrong, I cant get it to work. |
| 15:12.33 | malwcal | chrisknight: what dhcp server are you running? |
| 15:12.49 | FastFeet | reber: I know the CDR is not your problem, because I have the same warning.. It is because my MySQL is not setup yet. |
| 15:12.51 | chrisknight | dhcpd. linux/smoothwall |
| 15:13.26 | *** join/#asterisk lenne_dk (n=lenne_dk@83.72.129.7.ip.tele2adsl.dk) |
| 15:13.31 | chrisknight | I can ssh into it and add options to /etc/dhcpd.conf... I must be doing it wrong... |
| 15:13.42 | FastFeet | reber: Got to run, good luck with it, I am sure it is something really simple |
| 15:13.52 | chrisknight | I tried: next-server 172.16.16.201; |
| 15:14.19 | lenne_dk | In 1.2 I can read/write the db in the console. |
| 15:14.26 | chrisknight | The phone boots, gets a dhcp lease but the tftp listed in the phones status is still the old one. |
| 15:14.30 | malwcal | hmm. When i did that at work I had to add "option option-150 code 150 = ip-address;" to the top of the config file, and then use that in the phone definition. |
| 15:14.38 | lenne_dk | It seems not possiblle in 1,4? |
| 15:15.17 | reber | FastFeet, ok, i try again |
| 15:15.20 | malwcal | chrisknight: the phones don't seem to use next-server, but rather this option 150 thing instead. |
| 15:15.37 | *** join/#asterisk botemia (n=false@196.205.124.73) |
| 15:15.45 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 15:16.18 | chrisknight | malwcal: ya lost me. option-150 code 150 = ip-address; never heard of it, then I have to do something with the "phone def"? hmmm.. |
| 15:16.42 | malwcal | Up the top of the /etc/dhcpd.conf add the line option-150 code 150... |
| 15:17.34 | malwcal | chrisknight: then in the same place you tried to use next-server instead say option-150 172.16.16.201 |
| 15:17.46 | chrisknight | option option-150 code 150 = ip-address; |
| 15:17.47 | chrisknight | or |
| 15:17.54 | chrisknight | option-150 code 150 = ip-address; |
| 15:18.05 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
| 15:18.28 | malwcal | Up the top: |
| 15:18.29 | malwcal | option option-150 code 150 = ip-address; |
| 15:18.34 | malwcal | then |
| 15:19.08 | malwcal | <PROTECTED> |
| 15:19.08 | malwcal | in your lease definition |
| 15:19.18 | Mahmoud | how does my SIP client know which RTP UDP port my Astetrisk is listening on? |
| 15:19.37 | malwcal | (changing the ip work's one to yours of course...) |
| 15:19.51 | chrisknight | ok ill try that.. Ill be disconnected because I have to reboot the smoothwall for the changes to take effect. I'll be back |
| 15:19.56 | mafkees | Mahmoud: it's negotiated on INVITE |
| 15:20.05 | Mahmoud | mafkees, I see |
| 15:20.45 | chrisknight | wait... least definition?? Im really green with voip... started playing with asterisks less than a week ago |
| 15:20.54 | Mahmoud | mafkees, can I use the same UDP 5060 port which is used by SIP? |
| 15:21.38 | mafkees | no |
| 15:22.26 | Mahmoud | can I just use one single UDP port for all RTP packets? |
| 15:22.50 | Mahmoud | "sockstat | grep asterisk" on freebsd shows that asterisk is using 4 RTP ports! |
| 15:23.39 | malwcal | chrisknight: sorry, lease definitions. |
| 15:24.48 | mafkees | Mahmoud: check rtp.conf |
| 15:25.01 | malwcal | chrisknight: you should have something that looks like subnet w.x.y.z netmask a.b.c.d { |
| 15:25.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:25.44 | chrisknight | yes |
| 15:25.59 | malwcal | chrisknight: that is where the second line i gave you goes. |
| 15:26.29 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 15:26.31 | malwcal | chrisknight: The first goes at the top, not within any { } sections |
| 15:27.00 | chrisknight | wait... im sorry for posting this huge post but... |
| 15:27.04 | chrisknight | subnet 172.16.16.0 netmask 255.255.255.0 |
| 15:27.04 | chrisknight | { |
| 15:27.04 | chrisknight | <PROTECTED> |
| 15:27.04 | chrisknight | <PROTECTED> |
| 15:27.04 | chrisknight | <PROTECTED> |
| 15:27.05 | chrisknight | <PROTECTED> |
| 15:27.07 | chrisknight | <PROTECTED> |
| 15:27.09 | chrisknight | <PROTECTED> |
| 15:27.11 | chrisknight | <PROTECTED> |
| 15:27.13 | chrisknight | <PROTECTED> |
| 15:27.15 | chrisknight | <PROTECTED> |
| 15:27.17 | chrisknight | <PROTECTED> |
| 15:27.19 | chrisknight | <PROTECTED> |
| 15:27.21 | chrisknight | <PROTECTED> |
| 15:27.23 | chrisknight | <PROTECTED> |
| 15:27.23 | *** mode/#asterisk [+b %chrisknight!*@*] by Corydon76-home |
| 15:27.29 | Mahmoud | lol |
| 15:27.45 | Mahmoud | malwcal, this means if i was behind a firewall, i must statically set RTP ports + SIP ports, right? |
| 15:27.49 | Corydon76-home | Don't paste to the channel |
| 15:27.54 | Corydon76-home | ~pb |
| 15:27.55 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:28.03 | malwcal | chrisknight: yes, you put it there, with the other options) |
| 15:28.06 | Mahmoud | chrisknight, pastebin.ca |
| 15:28.38 | *** mode/#asterisk [-b %chrisknight!*@*] by Corydon76-home |
| 15:28.48 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 15:28.54 | malwcal | Mahmoud: I am not at all following your conversation, perhaps you were thinking of someone else? |
| 15:29.09 | Mahmoud | oh sorry, tab completion |
| 15:29.19 | Mahmoud | mafkees, this means if i was behind a firewall, i must statically set RTP ports + SIP ports, right? |
| 15:29.52 | malwcal | chrisknight: the first line I gave you goes before that subnet definition, somewhere up the top. |
| 15:30.22 | chrisknight | malwcal: your saying I need it in the {} and the same command outside the {}... |
| 15:30.32 | *** join/#asterisk Jared_Leto (n=Lostprop@80-89-104-241.DSL.ycn.com) |
| 15:30.45 | lenne_dk | How do I access the db from command line? |
| 15:31.02 | lenne_dk | in 1.4 |
| 15:31.26 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 15:31.50 | malwcal | chrisknight: good night. |
| 15:31.53 | lenne_dk | in 1.2 I could use "db set ..." |
| 15:32.07 | mafkees | lenne_dk: database |
| 15:32.41 | chrisknight | later... thanks |
| 15:33.07 | lenne_dk | Tnx, maf |
| 15:35.00 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:44.26 | *** join/#asterisk TheMahmoud (n=fake@unaffiliated/mahmoud) |
| 15:44.47 | Mahmoud | SIP is insane. it takes lots of UDP ports |
| 15:44.53 | Mahmoud | the more phone calls are made, the more it takes |
| 15:45.03 | Mahmoud | SIP is just dirty |
| 15:47.16 | marc\cba | folks im confused, my x-lite client is behind a home NAT gateway, my asterisk box is on a public ip elsewhere. Im registered. When i make a call the first packet from x-lite has my LAN ip address in it |
| 15:47.17 | marc\cba | whats puzzling is that asterisk starts trying to stream the audio then, x-lites private lan ip |
| 15:47.20 | marc\cba | :o |
| 15:48.01 | *** join/#asterisk angryuser (n=Miranda@df01t2-212-195-193-150.d4.club-internet.fr) |
| 15:48.22 | marc\cba | my x-lite sends an INVITE Via: SIP/2.0/UDP 10.0.0.50 |
| 15:48.26 | marc\cba | to the asterisk server |
| 15:48.48 | marc\cba | despite the connection ovbiously originating from my public ip address to the * box |
| 15:49.02 | marc\cba | * still opts to stream audio to my private ip |
| 15:49.23 | marc\cba | instead of my public ip ready to be natted back to me |
| 15:49.57 | marc\cba | obviously, the asterisk box is on a different inet connection and 10.0.0.50 doens't exist from its point of view |
| 15:50.12 | file | marc\cba: you did set nat=yes in sip.conf for the x-lite client, right? |
| 15:50.26 | marc\cba | file, i've got my asterisk server running in a vm box |
| 15:50.38 | marc\cba | i had it on a different network last week |
| 15:50.43 | marc\cba | and all was working well |
| 15:50.48 | file | that was a yes/no question... |
| 15:51.00 | marc\cba | sec.. |
| 15:52.18 | marc\cba | i'll check now, my connection to the * box dropped, bear w/ me |
| 15:53.14 | marc\cba | nat=yes |
| 15:53.38 | Mahmoud | hmm |
| 15:53.42 | file | and your Asterisk machine has a public IP? |
| 15:53.45 | marc\cba | correct |
| 15:53.49 | marc\cba | and only a public ip |
| 15:53.50 | Mahmoud | any way to reload the "rtp.conf" configuration without restarting the server? |
| 15:53.51 | marc\cba | no private ip |
| 15:53.58 | file | sip debug, rtp debug - pastebin |
| 15:54.05 | marc\cba | iptables not enabled atm. |
| 15:54.28 | marc\cba | eth0 = 80.x.x.x/252 |
| 15:54.46 | file | oh, and pastebin sip.conf |
| 15:54.57 | Mahmoud | in etc/aterisk/rtp.conf file, it assignes RTP port ranges from 10000 to 20000, does this limit my number of phone calls? |
| 16:00.52 | marc\cba | file; http://pastebin.ca/381054 |
| 16:00.53 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 16:03.22 | file | why do you have externip and such set if you're not behind NAT? |
| 16:03.33 | marc\cba | ah, the box used to be natted |
| 16:03.39 | marc\cba | i'll comment it out, well spotted |
| 16:03.47 | file | and what is the username of the x-lite? |
| 16:03.53 | marc\cba | 301 |
| 16:04.30 | file | okay, comment out externip/externhost/localnet - restart Asterisk - try again and pastebin a sip debug and an rtp debug of the attempt |
| 16:04.37 | zoa | Fiiiiiiiiile |
| 16:04.43 | zoa | you little faggot |
| 16:04.46 | marc\cba | will do |
| 16:04.49 | zoa | come here |
| 16:05.00 | zoa | howsy ? |
| 16:05.05 | zoa | howdy dowdy ? |
| 16:05.09 | zoa | HELOOOOOOOO |
| 16:05.10 | zoa | :p |
| 16:05.15 | zoa | me not can wait long |
| 16:05.21 | zoa | mi little loverboy |
| 16:05.24 | file | uh oh |
| 16:05.25 | file | it's zoa |
| 16:08.59 | Corydon76-home | zoa: drunk much? |
| 16:09.08 | jjshoe | tzafrir are you on crack? that's so not true. then you have the overhead of spawning an external process. |
| 16:09.59 | zoa | not yet today |
| 16:10.18 | zoa | Corydon-w: you are just jaleous of the voices in my head! |
| 16:10.32 | Corydon76-home | I think you'd have to be drunk to call file a faggot and not get kicked |
| 16:10.58 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 16:11.18 | marc\cba | ah |
| 16:11.39 | zoa | hehe |
| 16:11.46 | zoa | nah, he knows im just joking |
| 16:11.51 | zoa | or am i |
| 16:11.53 | marc\cba | file: genious. many thanks sometimes you need someone else to help you see the wood through the trees |
| 16:12.04 | marc\cba | many thanks |
| 16:12.34 | zoa | how is corydon doing ? |
| 16:12.48 | zoa | and when did you turn 76 ? |
| 16:12.57 | Corydon76-home | in 1976 |
| 16:13.36 | Corydon76-home | That would make me 107, right? |
| 16:13.51 | zoa | jups |
| 16:13.56 | zoa | congratulations! |
| 16:14.22 | zoa | is it just me or is this channel quite quiet lately ? |
| 16:14.29 | zoa | a million people but nobody talking |
| 16:14.32 | zoa | or saying NEXT! |
| 16:14.36 | Corydon76-home | It's Sunday morning in America |
| 16:14.45 | zoa | yeah but i was here last night too |
| 16:14.53 | Corydon76-home | NEXT-boy is no longer a contributor |
| 16:14.59 | zoa | yeah i know |
| 16:15.12 | zoa | he is still here though |
| 16:16.29 | Corydon76-home | zoa: I'm about ready for a shower, actually |
| 16:16.45 | zoa | have fun! |
| 16:19.10 | *** join/#asterisk scarfy (n=scarfy@84-73-85-22.dclient.hispeed.ch) |
| 16:19.22 | zoa | im off too |
| 16:20.32 | zoa | cheers! |
| 16:20.35 | scarfy | Hi, can anyone tell me what this error message means? i can't find anything usefull when i google it. |
| 16:20.37 | scarfy | Mar 4 17:12:05 NOTICE[22117]: chan_sip.c:7305 handle_request: Failed to authenticate user <sip:12345678910@proxy01.sipphone.com>;tag=f087dc74 |
| 16:22.03 | jjshoe | http://www.voip-info.org/wiki-SIP+response+codes |
| 16:22.26 | jjshoe | you'll need to sip debug ip <theip> |
| 16:23.17 | scarfy | hmm ok thanks :) |
| 16:23.57 | tzafrir | jjshoe, what are you talking about? |
| 16:24.33 | jjshoe | tzafrir I scrolled back and responded to your silly response. |
| 16:24.52 | tzafrir | regarding? |
| 16:24.58 | jjshoe | happy scrolling. |
| 16:25.18 | tzafrir | out of my scroll buffer. nm |
| 16:25.29 | jjshoe | it's not worth it anyways, it was extremely stupid response. |
| 16:36.47 | jjshoe | "At least one of app or extension must be specified, along with channel and destination |
| 16:36.51 | jjshoe | that's from voip-info |
| 16:37.03 | jjshoe | do I need to speficy an app or extension if I specify a context in a call file? |
| 16:38.20 | ManxPower | jjshoe: The Wiki is often wrong. See the sample.call file in the Asterisk source tree. |
| 16:39.17 | ManxPower | If you specify an extension, then you need to specify the context |
| 16:39.46 | ManxPower | Remember each call has two "legs". |
| 16:39.46 | jjshoe | I forgot all about s |
| 16:39.49 | jjshoe | right |
| 16:40.39 | ManxPower | My scripts will frequently have 1 leg of the call be a PSTN call, and the other leg a local extension. |
| 16:40.45 | jjshoe | right |
| 16:41.04 | jjshoe | one of my leg's is going to be an outbound call, and one leg needs to drop into a local context |
| 16:42.15 | marc\cba | how many legs you got jjshoe? :p |
| 16:42.17 | *** join/#asterisk chrisknight (n=explodin@cpe-71-79-81-174.columbus.res.rr.com) |
| 16:43.35 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 16:43.36 | jjshoe | marc\cba how many legs could I have if I could have multilple legs? |
| 16:43.37 | jjshoe | :D |
| 16:44.10 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 16:48.03 | marc\cba | touché |
| 16:48.08 | marc\cba | and one-nil ;p |
| 16:49.13 | *** join/#asterisk juva (n=juva@gre92-3-81-56-50-45.fbx.proxad.net) |
| 16:55.14 | *** join/#asterisk dual-man (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 16:55.51 | dual-man | i would like to alert my guys if they've been on the phone for ten minutes by playing a beep, what is the best way to accomplish this? |
| 16:57.42 | angryuser | dual-man: core show application dial |
| 16:58.12 | angryuser | dual-man: you have a timeout there and nyway this can be done in many ways |
| 16:58.28 | *** join/#asterisk KnowWhat (n=KnowWhat@74-132-66-76.dhcp.insightbb.com) |
| 16:58.30 | KnowWhat | Hello |
| 16:58.42 | KnowWhat | i am connecting my asterisk remotely via sip |
| 16:59.06 | KnowWhat | i can dial through it, but i can not hear the other persons voice, what could be the problem, but iax protocol is running fine |
| 16:59.30 | angryuser | KnowWhat: port forwrding udtp 10000-20000 |
| 16:59.37 | angryuser | *udp |
| 17:01.24 | angryuser | KnowWhat: you call out with sip the iax client? |
| 17:03.14 | KnowWhat | hmm |
| 17:03.20 | KnowWhat | 10000 to 20000 |
| 17:03.23 | KnowWhat | are you sure |
| 17:03.37 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 17:03.43 | SoftIce | hi can somebody help me with FOP? |
| 17:03.45 | KnowWhat | no i call out sip to a number |
| 17:03.56 | SoftIce | I can only see parking lot, no extensions and no trunks? |
| 17:04.07 | KnowWhat | i connect to asterisk server at my office, though xlite at my home here |
| 17:04.25 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:04.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:04.49 | jjshoe | need to open rtp ports. |
| 17:05.08 | *** join/#asterisk inspired (n=mikael@62.141.128.222) |
| 17:05.10 | KnowWhat | rtp ports hmm |
| 17:05.22 | angryuser | ~ports |
| 17:05.32 | jbot | from memory, ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
| 17:05.32 | KnowWhat | well my firewall have udp and tcp ports option |
| 17:05.49 | angryuser | http://www.pastebin.ca/381113 here read this |
| 17:05.57 | KnowWhat | angler: i am using smoothwall |
| 17:07.13 | angryuser | open rtp ports KnowWhat: |
| 17:07.31 | SoftIce | how do you regenerate variables.txt in FOP ? |
| 17:08.36 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 17:08.36 | KnowWhat | ok i dont know how to do that but let me search on it how i can do that, is that in asterisk or in firewall? |
| 17:08.41 | Mahmoud | can SIP usrs call IAX users? |
| 17:08.52 | ManxPower | Mahmoud: of course they can. |
| 17:08.58 | jjshoe | Mahmoud with a gateway |
| 17:09.06 | jjshoe | like, asterisk, in the middle. |
| 17:09.14 | Mahmoud | yea with a gatewya |
| 17:09.18 | ManxPower | Asterisk is a multiprotocol PBX. IT would be pretty pointless if asterisk did not support that |
| 17:09.18 | Mahmoud | i have sip and iax users |
| 17:09.25 | Mahmoud | but sip users can not call sip IAX |
| 17:09.40 | ManxPower | Mahmoud: Then you have some other problem. |
| 17:09.43 | Mahmoud | it says "Unable to create channel of type 'IAX' (cause 66 - Channel not implemented)" |
| 17:09.55 | ManxPower | Mahmoud: IAX2 not IAX. |
| 17:10.09 | Mahmoud | oh |
| 17:10.14 | Mahmoud | so i must say IAX2/username ? |
| 17:10.19 | jjshoe | Mahmoud yes |
| 17:10.20 | Mahmoud | dial(IAX2/username) |
| 17:10.21 | Mahmoud | amaaazing |
| 17:10.22 | ManxPower | Mahmoud: try reading extensions.conf.sample |
| 17:10.25 | Mahmoud | THANKIEZ may god bless you |
| 17:12.19 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:24.33 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 17:35.21 | jjshoe | how do you pass a variable to a context using a call file? |
| 17:35.32 | jjshoe | set? |
| 17:35.59 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 17:36.47 | *** join/#asterisk djflux (n=djflux@mm.creditonefs.com) |
| 17:38.04 | *** join/#asterisk foxxtrot (n=craig@c-67-185-241-244.hsd1.wa.comcast.net) |
| 17:39.25 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 17:44.34 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 17:53.07 | angryuser | KnowWhat: it depends on your lan structure, if you asterisk ox is behind nat you need to route corresponding ports, also open them in firewall on asterisk machine |
| 17:56.33 | Mahmoud | any way to rn idefisk softphone over a non-standard port? |
| 17:56.37 | Mahmoud | s/rn/run |
| 17:59.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.216.2) |
| 18:06.21 | Mahmoud | there's no good voip protocol |
| 18:06.37 | Mahmoud | iax looks fine, but no conferencing |
| 18:06.44 | [TK]D-Fender | Mahmoud: Guess it would depend on what you consider "bad". |
| 18:07.05 | Mahmoud | sip has many features, but uses lots of port |
| 18:07.13 | Mahmoud | the more calls you make, the more udp ports it opens |
| 18:09.27 | [TK]D-Fender | Mahmoud: Soa few more ports.. big deal. Means simplifying the conent of each and being able to reroute individual channels all ofver the place easier. |
| 18:10.11 | Mahmoud | means, you should increase your port range if you have more clients |
| 18:10.37 | Mahmoud | and it gets tough with white list firewalls |
| 18:11.41 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 18:12.46 | Mahmoud | anyway, i have no way but iax (sip blocked) |
| 18:13.03 | Mahmoud | is there any way to change the destination port that's being used ty IAX2 clients? |
| 18:13.14 | Mahmoud | i'm using the one made by asteriskguru, and it doesn't have a way to change it |
| 18:13.23 | Mahmoud | if you know any better iax2 softphone please tell |
| 18:13.51 | wunderkin | you can't do host:port? |
| 18:13.56 | Mahmoud | hmm lemme try |
| 18:14.08 | BSDTech | is there a scriptto convert extensions.conf to extensions.ael ? |
| 18:14.41 | Mahmoud | wunderkin, amazing |
| 18:14.51 | Mahmoud | wunderkin, you are from heaven, god sent you to help me =P |
| 18:14.53 | ManxPower | Mahmoud: Of course IAX2 supports conferencing |
| 18:15.10 | BSDTech | btw we have zaptel/libpri/asterisk 1.4 on bsd now |
| 18:15.15 | Mahmoud | ManxPower, but IDEFISK (made by asteriskguru.org) doesn't have such feature? |
| 18:15.16 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 18:15.23 | BSDTech | and the add-ons should be done patching today |
| 18:15.31 | Mahmoud | ManxPower, know any better IAX2 softphone? |
| 18:15.32 | ManxPower | Mahmoud: correct. The specfic softphone you are using does not have conferencing |
| 18:15.49 | ManxPower | Mahmoud: All softphones are terrible. I would not waste my time on a softphone |
| 18:16.07 | Mahmoud | they are cheaper than hard phones |
| 18:16.31 | ManxPower | Mahmoud: if you have no money then you should not be in telecom. Telecom costs money. |
| 18:16.46 | Mahmoud | it's for home use |
| 18:16.50 | CrazyTux | ManxPower, maybe thats why he wants to get in, to make money? :) |
| 18:16.59 | ManxPower | Mahmoud: it does not matter how cheap something is if it does not meet your needs. |
| 18:17.06 | Mahmoud | okay forgive me.. sigh |
| 18:17.53 | Mahmoud | so, IAX2 support conferencing, video.. hmm nothing more needed.. it's the perfect solution |
| 18:18.04 | Mahmoud | ManxPower, i'll look for some cheap h/w phones |
| 18:18.18 | Mahmoud | better to be mobile, working with wireless networks |
| 18:19.22 | *** join/#asterisk Merlin (n=visi@bitcondom.bytesex.com) |
| 18:19.38 | Merlin | what are people using as a good Linux QoS reference? |
| 18:19.48 | Merlin | everything i found online seems to be outdated |
| 18:19.58 | ManxPower | Merlin: We use Cisco routers for that sitff. |
| 18:21.05 | [TK]D-Fender | BSDTech: No, but there is one for the reverse :) |
| 18:21.24 | BSDTech | to convert ael to .conf |
| 18:21.32 | BSDTech | hmm ok |
| 18:21.41 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 18:21.41 | BSDTech | did ael not take off ? |
| 18:21.42 | [TK]D-Fender | BSDTech: Correct... thats exactly what happens on load :) |
| 18:21.54 | [TK]D-Fender | BSDTech: Ask luke-jr_ .. he's about the only person using it :) |
| 18:22.01 | BSDTech | ok |
| 18:22.24 | BSDTech | well I about have asterisk-gui working now on bsd |
| 18:22.33 | [TK]D-Fender | BSDTech: Quick story : Doesn't do anything normal can't (because it gets parsed back to normal anyways on load). |
| 18:22.56 | BSDTech | ok |
| 18:23.22 | BSDTech | well 1.4.1 seems to work well |
| 18:23.27 | Merlin | ManxPower: not everyone does |
| 18:23.27 | luke-jr_ | O.o |
| 18:23.34 | Merlin | ManxPower: they are very expense :) |
| 18:23.38 | Merlin | expensive |
| 18:23.38 | BSDTech | have had it up and running with no issues |
| 18:23.49 | BSDTech | for 24 hours |
| 18:23.52 | BSDTech | but give it time we will see |
| 18:24.04 | BSDTech | I hate that meetme uses zaptel |
| 18:24.13 | BSDTech | I am going to find a fix for it |
| 18:24.25 | luke-jr_ | BSDTech: it's easier to convert complex (AEL) to simple (conf) :p |
| 18:24.33 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 18:24.39 | jjshoe | BSDTech it's called app_conference :) |
| 18:24.58 | BSDTech | where is app-confrence is there a url |
| 18:24.59 | luke-jr_ | BSDTech: I have some nice AEL functions tho |
| 18:25.07 | BSDTech | ahh ok |
| 18:25.19 | jjshoe | BSDTech yes, it's google.com |
| 18:25.25 | luke-jr_ | like RFC-compliant E164, strReplace, etc |
| 18:25.43 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 18:26.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:27.12 | BSDTech | it looks to be part of vicidial |
| 18:29.09 | BSDTech | and the svn of it I fine is over 16 months old |
| 18:29.15 | wunderkin | no, he made a mod of it |
| 18:29.53 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:29.57 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 18:31.28 | BSDTech | ok I am not getting a uptodate like with google |
| 18:31.36 | BSDTech | like/link |
| 18:33.05 | wunderkin | i don't know if it is in active development anymore, was it part of iaxclient or something? |
| 18:36.49 | DocHolliday | BSDTech = BSDaemon? |
| 18:37.21 | BSDTech | I found a sf svn |
| 18:37.24 | BSDTech | but its 1.0 |
| 18:37.40 | BSDTech | not I am not a Daemon yet but working on it |
| 18:42.34 | BSDTech | ok the one in vicidial has been updated so I will yank it out of there |
| 18:58.34 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 19:11.26 | *** join/#asterisk topping (n=topping@204.152.96.238) |
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| 19:20.40 | BSDTech | zaptel 1.4 seems to still have issues with smp |
| 19:20.44 | BSDTech | grr |
| 19:21.00 | bkruse_home | BSDTech: ohrly? |
| 19:21.02 | bkruse_home | kernel panic!? |
| 19:22.06 | BSDTech | its says it loads but when I start asterisk and do zap show channels is says zap show unknown command |
| 19:24.16 | file | did the asterisk configure script pick up zaptel? did you confirm chan_zap would be built via make menuselect? |
| 19:25.19 | BSDTech | it finds the zaptel.h |
| 19:25.32 | BSDTech | but I get xxx where the zaptel is |
| 19:25.38 | BSDTech | in the menu config |
| 19:26.23 | file | dependency wasn't met |
| 19:26.50 | file | the configure script also checks to make sure the installed zaptel has what it needs (vldtmf for example) |
| 19:26.54 | bkruse_home | ls /usr/lib/asterisk/modules | grep zap |
| 19:27.11 | bkruse_home | build zaptel then asterisk, right? |
| 19:27.14 | file | shown at almost the end of the configure run |
| 19:27.31 | file | example: checking for ZT_TONE_DTMF_BASE in zaptel/zaptel.h... yes |
| 19:27.36 | bkruse_home | yep |
| 19:29.52 | BSDTech | reruning brb |
| 19:32.56 | BSDTech | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 19:33.08 | BSDTech | so why is the config notfinding it |
| 19:33.18 | file | maybe it's not there? |
| 19:34.32 | BSDTech | it si I installed it |
| 19:34.52 | *** join/#asterisk [shodan] (n=shodan@ip154.96-113-216.pppoe1.joliette.intermonde.net) |
| 19:34.53 | BSDTech | I will have to look into it further |
| 19:41.33 | *** join/#asterisk linlin (n=will@c-67-184-229-49.hsd1.il.comcast.net) |
| 19:43.08 | BSDTech | I am sorry I forgot to say this is on bsd. and it seems the script does not look in /usr/local/include/zaptel |
| 19:43.16 | BSDTech | where we put zaptel.h |
| 19:43.33 | BSDTech | so its the 1.4.1 configure we have to patch now |
| 19:44.08 | file | you specified the zaptel path to configure? |
| 19:44.35 | BSDTech | the configure only seems to look in zaptel/ |
| 19:44.51 | BSDTech | and I have zaptel 1.4.0 installed |
| 19:45.03 | BSDTech | we got it installing lastnight |
| 19:45.22 | BSDTech | libpri 1.4.0 works ut of the box |
| 19:46.31 | BSDTech | but the configure script is not finding the zaptel.h wich means it needs to be fixed to look in the right places |
| 19:47.58 | file | try ./configure --with-zaptel=/usr/local/include |
| 19:49.46 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 19:50.11 | BSDTech | I had to add the like to the port make file |
| 19:50.15 | BSDTech | it was not there |
| 19:51.41 | *** join/#asterisk Vec (n=Vec@dsl-244-219-12.telkomadsl.co.za) |
| 19:51.48 | BSDTech | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 19:51.49 | BSDTech | configure: *** |
| 19:51.49 | BSDTech | configure: *** The Zaptel installation on this system appears to be broken. |
| 19:51.49 | BSDTech | configure: *** Either correct the installation, or run configure |
| 19:51.49 | BSDTech | configure: *** including --without-zaptel. |
| 19:51.49 | BSDTech | ===> Script "configure" failed unexpectedly. |
| 19:51.53 | BSDTech | hmmm |
| 19:52.12 | BSDTech | but I am using the bsd zaptel1.4 svn |
| 19:52.15 | BSDTech | hmmm |
| 19:52.24 | file | maybe it's not up to date enough to have that? |
| 19:52.25 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 19:52.43 | elriah | Alright! 1.4.1, woohoo! |
| 19:54.49 | BSDTech | ZT_DIAL_OP_CANCEL |
| 19:54.55 | BSDTech | is in the zaptel.h |
| 19:55.07 | BSDTech | I just checked |
| 19:55.26 | BSDTech | #define ZT_DIAL_OP_APPEND 1 |
| 19:55.26 | BSDTech | #define ZT_DIAL_OP_REPLACE 2 |
| 19:55.26 | BSDTech | #define ZT_DIAL_OP_CANCEL 3 |
| 19:57.33 | *** part/#asterisk Merlin (n=visi@bitcondom.bytesex.com) |
| 19:58.06 | BSDTech | it fails at the same eplace |
| 19:58.18 | BSDTech | btu its in the zaptel.h |
| 20:02.22 | BSDTech | even including the path seems not to work |
| 20:02.26 | BSDTech | this is a pisser |
| 20:02.40 | BSDTech | I want my sangoma card working but it needs zaptel |
| 20:03.58 | BSDTech | CONFIGURE_ARGS+= --with-zaptel=${LOCALBASE}/include --mandir=${PREFIX}/man |
| 20:07.03 | BSDTech | we dont use /usr/local/include/zaptel/zaptel.h |
| 20:07.15 | BSDTech | we just put it in /usr/local/include |
| 20:07.28 | BSDTech | so its looking for a dir called zaptel |
| 20:08.50 | *** join/#asterisk fndone (n=fndone@82-69-78-118.dsl.in-addr.zen.co.uk) |
| 20:10.28 | BSDTech | the config srcipt sucks that they force it to zatel/zaptel.h |
| 20:10.33 | BSDTech | thats stupid |
| 20:10.37 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 20:10.57 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 20:13.53 | *** join/#asterisk Nest0r (n=Nest0R@201.230.188.226) |
| 20:14.07 | Nest0r | hi |
| 20:14.36 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:16.20 | *** join/#asterisk HarryB (n=HarryB@bas12-toronto63-1088802200.dsl.bell.ca) |
| 20:16.27 | HarryB | hello and g'afternoon everyone |
| 20:16.30 | BSDTech | ok I just made a dir zaptel and cp the files and it makes |
| 20:17.13 | HarryB | I have two A200 sangoma cards installed |
| 20:17.27 | HarryB | 2 POTS lines are coming in |
| 20:17.32 | HarryB | 6 more to be added later on |
| 20:17.41 | HarryB | however, I am encountring a problem now |
| 20:17.47 | HarryB | When I dial in |
| 20:17.53 | HarryB | it keeps ringing for ever |
| 20:18.03 | HarryB | but asterisk -rvvvvvv shows calls answered |
| 20:18.09 | HarryB | and goes to Voice Mail |
| 20:18.33 | BSDTech | it has to do woth the ver of asterisk and the current sangoma driver |
| 20:18.50 | BSDTech | you have to get the updated sangoma driver |
| 20:18.51 | HarryB | and finall Hungsup and Spawn extension 0macro-vm, exit success, 2) exited non-zero on 'Zap/8-1' in macro 'vm' |
| 20:19.06 | BSDTech | I have had the same issue in the past |
| 20:19.16 | BSDTech | what ver of asterisk and zaptel ? |
| 20:19.26 | HarryB | one sec i will tell u |
| 20:19.35 | HarryB | what is the command to check that? |
| 20:19.46 | *** part/#asterisk Nest0r (n=Nest0R@201.230.188.226) |
| 20:20.02 | FastFeet | core show version |
| 20:20.29 | HarryB | core command does not exit |
| 20:20.57 | FastFeet | guess your not using 1.4 |
| 20:21.04 | FastFeet | show version |
| 20:21.12 | HarryB | wanpiupe is 2.3.4-2 |
| 20:21.13 | HarryB | i think |
| 20:21.24 | HarryB | it's trixbox i am using |
| 20:21.31 | FastFeet | :D |
| 20:21.32 | *** join/#asterisk Stridernzl (n=neville@222-152-248-128.jetstream.xtra.co.nz) |
| 20:21.43 | FastFeet | Never used it |
| 20:21.50 | FastFeet | Only Asterisk |
| 20:21.54 | HarryB | so how do i update to the newer one? |
| 20:22.35 | FastFeet | <PROTECTED> |
| 20:23.17 | HarryB | i can download ftp://ftp.sangoma.com/linux/current_wanpipe/wanpipe-2.3.4-7.tgz. |
| 20:23.25 | FastFeet | ? |
| 20:23.27 | HarryB | does it matter which directory i downalod it too? |
| 20:23.50 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 20:25.38 | reber | i have an asterisk behind a NAT, what do i have to do to access it from the internet with a softphone ? Hints are here : http://www.voip-info.org/tiki-index.php?page=Asterisk+How+to+connect+to+FWD |
| 20:26.33 | BSDTech | I have zap now |
| 20:27.32 | *** join/#asterisk kgx (n=kgx@60.234.20.178) |
| 20:29.08 | HarryB | reber do you use freepbx? |
| 20:29.18 | HarryB | if you do then make a sip extension and apply |
| 20:29.19 | HarryB | it |
| 20:29.36 | HarryB | then go into the extension again and put 'yes' for 'careinvite' field |
| 20:29.46 | HarryB | once does that you will be able to connect to it |
| 20:35.47 | *** join/#asterisk Barmal (i=Zilas_@c-24-99-8-2.hsd1.ga.comcast.net) |
| 20:35.53 | HarryB | can anyone tell me how to update wanpipe and it's utili ? |
| 20:35.58 | HarryB | for A200 card |
| 20:36.00 | BSDTech | I am basicly good to go till we get asterisk-addons ported |
| 20:36.02 | BSDTech | yes |
| 20:36.14 | BSDTech | and the a200 is working on bsd yes |
| 20:36.19 | BSDTech | cool this rocks |
| 20:36.26 | BSDTech | 1.4.1 I have arrived |
| 20:36.32 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:36.39 | Barmal | I am little lost here... pbx_dundi.c:1309 update_key: No such key 'dundi' for creating RSA encrypted shared key for |
| 20:36.54 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 20:39.00 | HarryB | wanpipe-util-2.3.4-7.i686.rpm |
| 20:39.09 | HarryB | [root@asterisk1 tmp]# rpm -i wanpipe-modules-2.6.9-34.0.2.ELsmp-2.3.4-7.i686.rpmWanpipe Modules located in /lib/modules/2.6.9-34.0.2.ELsmp |
| 20:39.23 | reber | HarryB, no, i use asterisk |
| 20:39.24 | HarryB | here ^^^^ is what i get when i try to install newer version |
| 20:39.45 | HarryB | what could be the prob? |
| 20:40.58 | HarryB | ??? |
| 20:41.12 | BSDTech | now the asteriskgui dont want to work |
| 20:41.14 | BSDTech | grrr |
| 20:41.16 | *** join/#asterisk Vec (n=Vec@c1-173-8.rrba.isadsl.co.za) |
| 20:41.41 | *** join/#asterisk budmang (n=budman@12-210-54-193.client.mchsi.com) |
| 20:41.51 | Barmal | I am lost with those keys... Where I can get the key for dundi please? |
| 20:42.28 | HarryB | is there any other asterisk help chnanel? |
| 20:42.49 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 20:42.58 | rhombus | Greetings! |
| 20:42.58 | *** join/#asterisk luisjose (n=ljd@unaffiliated/luisjose) |
| 20:43.25 | rhombus | Can anyone suggest a good Linux XML editor for the purpose of editing Polycom configuration files? |
| 20:45.14 | ManxPower | rhombus: JEdit. It's not really an XML editor, but it does do XML syntax highlighting |
| 20:45.20 | reber | i have an asterisk behind a NAT, what do i have to do to access it from the internet with a softphone ? |
| 20:45.23 | ManxPower | I use it for editing polycom config files |
| 20:45.40 | fetcher | rhombus: I think vim has an xml mode also |
| 20:45.47 | ManxPower | reber: we answer that question hundreds of times in a month. look in the mailinglist archive |
| 20:46.10 | ManxPower | rhombus: JEdit also has an SSH/SCP plugin so you can edit the files directly from a remote machine |
| 20:46.13 | rhombus | Yeah, I looked into vim's xml mode, but it looks complicated to set up |
| 20:46.23 | rhombus | also, it assumes you're using standard doctypes |
| 20:46.34 | rhombus | ManxPower: Nice. I'll have a look at it |
| 20:46.37 | ManxPower | JEdit is written in Java so you can run it on pretty much any OS that supports Java |
| 20:47.35 | rhombus | fetcher: My phones reboot in under 60 seconds! Isn't that great? |
| 20:48.08 | fetcher | rhombus: Polycom? which model is that? |
| 20:48.14 | rhombus | ManxPower: I've tried kxmleditor, and it breaks the file if you add a tag and then save it |
| 20:48.24 | rhombus | fetcher: The IP 650 |
| 20:48.46 | rhombus | ManxPower: I've also tried XMLmind, and have yet to figure out how to create a tag :) |
| 20:48.49 | fetcher | our slow ones here are IP 501's, running SIP 2.0 images. Probably have slower CPUs |
| 20:49.12 | fetcher | but it's the "checking software image", and "loading application..." stages that take the most time |
| 20:51.28 | fetcher | can Asterisk disable VAD / silence-suppression (inserting empty speech frames) when one endpoint is using it? |
| 20:52.53 | fetcher | VAD normally works well, and saves a lot of bandwidth, but one particular SIP peer can't deal with it well. On the phones it's an always-or-never setting |
| 20:55.45 | *** join/#asterisk dockmazter (n=dockmazt@gw-ham.iphh.net) |
| 20:56.11 | dockmazter | g'devening |
| 20:56.17 | dockmazter | (at least over here) |
| 20:58.31 | dockmazter | Maybe someone can help me out with my problem of the day. This is not my first Cisco 79xx registering at an * box but today (first 79xx on THIS * box) I am experiencing the problem with the phone's high source port. Phone sends REGISTER from e.g. 51242 and * replies to 51242 which is rejected by the phone b/c it is listening on 5060. I thought this was related to the NAT settings in sip.conf and tried nat=no and nat=never but I can not get to reply |
| 20:58.37 | *** join/#asterisk ClydeGoffe (n=ClydeGof@base/student/clydegoffe) |
| 21:01.50 | *** join/#asterisk harrybjb (n=HarryB@bas12-toronto63-1088802200.dsl.bell.ca) |
| 21:02.02 | harrybjb | does anyone know how to update Wanpipe? |
| 21:03.28 | *** part/#asterisk harrybjb (n=HarryB@bas12-toronto63-1088802200.dsl.bell.ca) |
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| 21:13.35 | *** join/#asterisk djs_2_6 (n=djstillm@cpe-071-077-052-156.nc.res.rr.com) |
| 21:14.36 | rhombus | fetcher: Which phone is having problems with VAD? |
| 21:15.29 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 21:18.57 | *** join/#asterisk LeddyHM (n=NONE@polar.artica.net) |
| 21:20.52 | imediax | is enumlookup depreciated in 1.4.1? I'm getting No application 'ENUMLOOKUP' |
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| 21:29.18 | *** join/#asterisk fab5freddy (n=vmware@bas1-montreal19-1177815388.dsl.bell.ca) |
| 21:29.51 | fab5freddy | Hi, I am following this tutorial http://www.ubuntuforums.org/showthread.php?t=136785 |
| 21:30.08 | fab5freddy | but this line doesn't work, svn checkout http://svn.digium.com/svn/asterisk-sounds/branches/1.0 asterisk-sounds-1.0 |
| 21:30.10 | *** join/#asterisk thoughtpolice (n=austin@ip68-98-250-69.lu.dl.cox.net) |
| 21:30.31 | fab5freddy | can anyody shed some light? thanks |
| 21:31.04 | *** join/#asterisk PMantis (n=pmantis@cpe-69-207-130-14.rochester.res.rr.com) |
| 21:31.53 | PMantis | Does anyone know how to determine (in dialplan) what channel a spscified channel is connected to? |
| 21:32.06 | ManxPower | PMantis: looks in README.variables |
| 21:32.16 | ManxPower | fab5freddy: what error message do you get? |
| 21:32.30 | PMantis | IOW, supply SIP/1032 as an argument, adn get back Zap/11 |
| 21:32.39 | *** join/#asterisk saftsack (n=oliver@pD9E07946.dip.t-dialin.net) |
| 21:32.44 | ManxPower | fab5freddy: also 1.0 is very, very old. 1.4.1 is the current version |
| 21:33.18 | ManxPower | PMantis: before the Dial the source channel is the only channel that exists |
| 21:34.01 | PMantis | ManxPower, I know... from a 3rd channel, I want to determine what channel a *different* phone is currently talking to. |
| 21:34.11 | PMantis | ...not my own. :) |
| 21:34.15 | fab5freddy | ManxPower: svn PROPFIND request failed |
| 21:34.42 | *** join/#asterisk abooker (n=tb@c-24-19-226-67.hsd1.mn.comcast.net) |
| 21:34.42 | fab5freddy | ManxPower: svn: Couldn't not open the requested SVN filesystem |
| 21:34.47 | PMantis | I expect it might be in the DB somewhere.. |
| 21:34.48 | ManxPower | fab5freddy: try svn checkout http://svn.digium.com:8080/svn/asterisk-sounds/branches/1.0 |
| 21:34.57 | ManxPower | IF that works then you have a crappy http proxy somewhere. |
| 21:35.19 | ManxPower | PMantis: what specifically are you looking for? |
| 21:36.19 | fab5freddy | ManxPower: same error messages, should i try changing 1.0 for 1.4.1? |
| 21:36.43 | PMantis | ManxPower, If a SIP phone dials outbound, and connects via ZAP channel, I want to be able to dial an exten and have the ZAP channel read to me. |
| 21:37.09 | PMantis | ManxPower, Of course, while the other call is still linked |
| 21:37.25 | ManxPower | so SIP -> Asterisk -> PSTN -> PSTN -> Asterisk |
| 21:37.37 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 21:37.47 | ManxPower | that is the only way you are going to make it work |
| 21:38.04 | PMantis | You're not following me yet. |
| 21:38.20 | PMantis | SIP -> Asterisk -> ZAP-PSTN |
| 21:38.32 | ManxPower | PMantis: you cannot do that |
| 21:38.45 | ManxPower | the ZAP-PSTN must do to the telco. |
| 21:38.50 | ManxPower | go -- go |
| 21:39.01 | ManxPower | if you plug your FXS into your FXO port then you can bypass the PSTN portion. |
| 21:39.02 | PMantis | Yes, of course... hangon, don't jump the gun. |
| 21:39.37 | PMantis | I only layed out a typlical scenario where someone can pickup a phone and call outbound... nothing more there. |
| 21:39.41 | PMantis | Then..... |
| 21:39.46 | fab5freddy | ManxPower: Can I complete the install without the sounds? |
| 21:40.16 | PMantis | Then, on a different SIP phone, pickup and dial an exten, and have ReadDigits(11) run, assuming that the previous call (on a different phone) is using ZAP/11 |
| 21:40.22 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 21:40.49 | PMantis | I just want to *lookup* what ZAP channel a SIP phone si currently using on the other leg of the call... |
| 21:41.53 | ManxPower | PMantis: then you already know about ${CHANNEL}, right? |
| 21:41.57 | abooker | n00b asking for a beating: I have any iaxy, I have a voip provider, I have a machine to run asterisk on, and I have a reasonable knowledge of networking. Where do I find docs to allow me to hook it all together in say 6hrs? Thx for your time. |
| 21:42.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:43.20 | ManxPower | PMantis: the SIP phone is not ON a zap channel until after Dial(Zap/whatever happens. |
| 21:43.32 | PMantis | OMG |
| 21:43.44 | ManxPower | and once Dial happen you can't magically start running dialplan stuff |
| 21:44.21 | PMantis | ManxPower, Somehow I led you to a *bad* conclusion... and I haven't been able to get you to "let go" of that. |
| 21:44.25 | PMantis | Let me start over. |
| 21:44.45 | ManxPower | PMantis: no, I think you don't want to accept that you can't do what you want to do. 8-=) |
| 21:44.57 | PMantis | ManxPower, wrong |
| 21:45.58 | PMantis | I know it's possible, and I can do it with extra dialan or DBPUTS, but I want to know if there's a way built in. |
| 21:45.58 | ManxPower | PMantis: describe how it would work from the point of view of the person using the SIP phone. |
| 21:45.59 | PMantis | From the top... |
| 21:46.18 | PMantis | My coworker picks up a SIP phone, dials 91234567 (or whatever), the call is shot over Zap/11 on a PRI and the other end answers... |
| 21:46.24 | JacksLivr | hello everyone. annyone have experience with the polycom soundpoint ip 500cs? |
| 21:47.05 | PMantis | While they are talking, *I* on a different SIP phone, dial an extension that is preprogramed to lookup SIP/1234, and report back to me what ZAP channel my coworker's phone is linked to. |
| 21:47.17 | ManxPower | PMantis: OK. 91234567 is a telephone number somewhere else? i.e. not a number on the PRI? |
| 21:47.22 | *** join/#asterisk jserve (n=mail@p54BCFA1D.dip.t-dialin.net) |
| 21:47.55 | ManxPower | PMantis: 1) The zap channel is not picked until Dial happens. Once the dial happens you can't put stuff into the database. |
| 21:48.02 | PMantis | ManxPower, Right, but that's insignificat to what I want to do. Either way there's a phone call already linked. |
| 21:48.28 | ManxPower | PMantis: best of luck with this. |
| 21:48.43 | ManxPower | I cannot help you futher. I still say you can't do what you want to do. |
| 21:48.49 | PMantis | Oh, man... |
| 21:48.57 | ManxPower | JacksLivr: 500CS? |
| 21:49.05 | PMantis | Then how does ChanSpy work? |
| 21:49.26 | ManxPower | PMantis: of course you could it if you wrote an asterisk application. |
| 21:49.35 | ManxPower | but I assume you do not want to write an asterisk application |
| 21:49.53 | PMantis | If there's no need, of course not. :) |
| 21:50.18 | JacksLivr | ManxPower: 500CS is the mgcp one that you can turn into sip |
| 21:50.30 | JacksLivr | but im apparently not smart enough to do it |
| 21:50.42 | ManxPower | JacksLivr: I doubt anyone here can turn it into SIP for you. |
| 21:50.46 | file | chanspy iterates through the channel list and is able to get the channel that the current one is bridged to |
| 21:50.53 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 21:51.15 | PMantis | But an example, you can't run ChanSpy on the same channel where the Dial is running... same thing you're saying to me. |
| 21:51.34 | kuku5 | When I have the digium t1 card ( no cable is plugged in ) and the light is blinking red, does that mean it found everything ? ( I cant find anythin int he docs regarding this LED blinking) |
| 21:52.09 | file | chanspy is running on the channel that is looking for stuff to listen on, it then attaches a spy (internal API) on a channel to get audio back |
| 21:52.12 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 21:52.19 | fab5freddy | Can somebody direct me to a walkthrough thats works to install Asterisk on Ubuntu Server 6.06 AMD64? |
| 21:52.34 | JacksLivr | ManxPower; maybe so. I sure cant. it can apparently be done though |
| 21:53.06 | J4k3 | anyone ever had to do a return through voipsupply? |
| 21:53.15 | JacksLivr | i have a 7910 that I can use. maybe i outta start trying to figure out how to use it. |
| 21:53.23 | Nugget | fab5freddy: the README file in the asterisk tarball. |
| 21:54.03 | fab5freddy | Nugget: Secondly, can you point me in the direction of a VOIP which is the easiest to get started with |
| 21:54.20 | Nugget | I don't know what you mean when you say "a VOIP" |
| 21:54.22 | file | PMantis: you would have to write a custom app to iterate through the channel list looking for a matching channel and get the channel it is bridged to... but even then the channels only get bridged once the other side picks up |
| 21:54.22 | fab5freddy | Nugget: VOIP phone, sorry! |
| 21:54.34 | Nugget | dunno, sorry. |
| 21:54.52 | fab5freddy | Nugget: What device do you use? |
| 21:54.58 | file | or you could do some channel variable stuff, have it be inherited down to the Zap channel... then you could iterate through the channel list looking for a matching Zap channel with the variable set |
| 21:54.58 | Nugget | I use cisco phones |
| 21:55.08 | wunderkin | J4k3, yeah... they don't do exchanges... they refund to your account and you have to repurchase... nice to know now huh? |
| 21:55.20 | fab5freddy | Nugget: are you an IT manager? |
| 21:56.34 | Nugget | no |
| 21:56.39 | dockmazter | did anyone read my question about the destination port for SIP udp-packets about 45 minutes ago? hmm. i should repost it |
| 21:57.55 | ManxPower | fab5freddy: telecom is a difficult and complicated thing. VoIP is at least 10 times as complicated. |
| 21:58.50 | ManxPower | You need to know about telecom, IP, linux, networking to start out, then you need to know SIP, the dialplan, NAT, routing. Then if you want to get fancy you need to learn AGI and Perl or C or PHP |
| 21:59.11 | file | requires problem solving skills and common sense |
| 21:59.20 | ManxPower | yes, that too 8-) |
| 22:02.14 | ManxPower | dockmazter: what was your question? |
| 22:02.49 | dockmazter | Maybe someone can help me out with my problem of the day. This is not my first Cisco 79xx registering at an * box but today (first 79xx on THIS * box) I am experiencing the problem with the phone's high source port. Phone sends REGISTER from e.g. 51242 and * replies to 51242 which is rejected by the phone b/c it is listening on 5060. I thought this was related to the NAT settings in sip.conf and tried nat=no and nat=never but I can not get to reply |
| 22:02.50 | abooker | I could say similar things about mail. There are lots of things to learn. But the fact of the matter is that one can begin exchaning mail with the world without knowing everything. |
| 22:02.56 | fab5freddy | I know a little about all those things, but clearly need to know more |
| 22:04.11 | ManxPower | dockmazter: Is the PHONE really sending from 51242 or is the phone sending from 5060 and the NAT router is doing port translation to src port of 51242? |
| 22:05.07 | ManxPower | phone src5060/dst5060 NAT src51242/dst5060 Asterisk |
| 22:05.42 | ManxPower | dockmazter: at least you have an interesting problem. 8-) |
| 22:05.43 | dockmazter | there is no NAT in between! they are even connected to the same switch |
| 22:05.51 | ManxPower | dockmazter: Oh! |
| 22:06.29 | ManxPower | dockmazter: now it gets more interesting. are you provisioning via an xml file on a tftp server? |
| 22:06.37 | kuku5 | When I have the digium t1 card ( no cable is plugged in ) and the light is blinking red, does that mean it found everything ? ( I cant find anythin int he docs regarding this LED blinking)? |
| 22:06.51 | ManxPower | dockmazter: Asterisk will reply to the port it got the request from. |
| 22:07.07 | ManxPower | kuku5: that question was answered on the mailing list last week. |
| 22:07.11 | dockmazter | manxpower: its this kind of plain file not xml..like SIPmacaddress.cnf |
| 22:07.41 | JT | kuku5: isn't the cable not being plugged in a bit of a giveaway |
| 22:08.09 | kuku5 | but does it mean the system found the card and all ? |
| 22:08.19 | dockmazter | manx: all NAT things in the SIPDefault.cnf and SIPmac.cnf are switched off |
| 22:08.23 | JT | means zaptel and card driver is loaded |
| 22:08.34 | kuku5 | :) - Thank you! |
| 22:08.52 | JT | doesn't guarantee it's configured right |
| 22:09.04 | kuku5 | Yes. |
| 22:09.12 | kuku5 | Just wanted make sure the other stuff is good. |
| 22:09.36 | rhombus | ManxPower: When I'm creating custom Polycom configs, do I need all the attributes in a tag, or only the ones I want to change? |
| 22:09.36 | ManxPower | dockmazter: what sip firmware version is on the phone? |
| 22:09.41 | dockmazter | manxpower: there is an article talking about this,... just a second |
| 22:09.59 | ManxPower | rhombus: in phone-macaddr.cfg? |
| 22:10.00 | dockmazter | manx: |
| 22:10.12 | dockmazter | huh.. manx: P0S3-08-5-00 |
| 22:10.31 | rhombus | Manx: yeah -- in that file you can specify overlapping configuration files, so that when you update sip.cfg, it doesn't break your phones |
| 22:11.47 | kuku5 | 08-6 is out |
| 22:12.59 | ManxPower | rhombus: see http://www.fnords.org/~eric/polycom-config-examples |
| 22:14.12 | ManxPower | dockmazter: I assume iptables/ipchains are NOT running? (lsmod to see) |
| 22:14.22 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 22:14.31 | rhombus | ManxPower: So that OVERRIDES tag is the only one I need, and then I can put attributes from any tag in there? |
| 22:14.45 | rhombus | For some reason I thought that file was uploaded by the phone :\ |
| 22:15.24 | ManxPower | rhombus: those are my ACTUALL production config files, except for the hostname was changed in the configs. |
| 22:15.48 | ManxPower | rhombus: the phone will overwrite that file if you let it |
| 22:15.49 | dockmazter | manx: would you go to http://www.voip-info.org/wiki/view/Standalone+Cisco+7941%252F7961+without+a+local+PBX and search for "unreachable" ? It describes my problem exactly |
| 22:16.22 | dockmazter | manx: (beside the fact that my phone is a 7960 and not a 7961) |
| 22:16.37 | linlin | can someone give me an example extension for an IAX2 extension ? |
| 22:17.26 | ManxPower | dockmazter: I assume you remove any nat= lines from sip.conf for that device? |
| 22:17.42 | ManxPower | linlin: exten => 666,1,Dial(IAX2/devil) |
| 22:17.52 | dockmazter | well, i have set "nat=no" or "nat=never" ... i should try to remove it completely..hmm |
| 22:18.10 | ManxPower | dockmazter: is there a nat=yes in ANY part of sip.conf? |
| 22:18.22 | ManxPower | dockmazter: nat=no should work, but you could just remove it |
| 22:18.36 | dockmazter | nat=no is in the global context |
| 22:18.52 | rhombus | ManxPower: in your macaddr-phone.cfg file -- does that PHONE_CONFIG tag matter, or can I name the tag anything? |
| 22:19.10 | ManxPower | rhombus: I would assume it is required. |
| 22:19.27 | ManxPower | rhombus: all I did was let the phone save it's config file to the FTP server, then modified it as required. |
| 22:19.43 | rhombus | ManxPower: Okay -- makes sense. Thanks. |
| 22:19.54 | ManxPower | my policy is to change a little as possible |
| 22:21.15 | ManxPower | dockmazter: you followed this? http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP |
| 22:21.30 | ManxPower | you should be able to switch to .xml config files with the same firmware. |
| 22:22.54 | dockmazter | manx: but only on the 79x1 i suppose |
| 22:23.50 | *** join/#asterisk BSDTech (n=RNeese@pool-71-108-9-139.lsanca.dsl-w.verizon.net) |
| 22:23.51 | ManxPower | dockmazter: I believe any SIP firmware above 6 supports .xml formatted config files |
| 22:24.24 | ManxPower | dockmazter: move the .cnf file somewhere, then reboot the phone, watch the tftp server logs to see all the files it tries to request |
| 22:24.48 | Vec | If cdr_mysql.conf and the cdr_mysql module is loaded will asterisk automatically save cdr records to the database ? |
| 22:24.55 | mafkees | yeah |
| 22:25.54 | Vec | What is res_mysql for ? (Working with mysql from extentions.conf ?) |
| 22:26.08 | Vec | i.e. not required for cdr_mysql |
| 22:26.17 | BSDTech | its for connecting to a sql database |
| 22:26.30 | Vec | is it required when using cdr_mysql ? |
| 22:26.45 | BSDTech | dontknow dont use mysql right now |
| 22:27.29 | dockmazter | manx: hmm... the same firmware works on a different asterisk box (with a different phone but same model) .. downgraded do 7-3 ... still not working... i would rather not play with xml config-files when the same setup works fine on another box.. this other * box I have with another phone also received FROM a high-port but replies to 5060 .. this works fine |
| 22:28.36 | ManxPower | dockmazter: do you have localnet= specified in sip.conf [general]? |
| 22:28.57 | *** join/#asterisk backblue (n=moo@87-196-6-60.net.novis.pt) |
| 22:30.21 | sweeper | ....wow |
| 22:30.24 | sweeper | unfortunately, I had to tell the guy I had mislaid my flux capacitor |
| 22:30.29 | sweeper | he didn't get it :( |
| 22:30.42 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 22:30.46 | J4k3 | haha |
| 22:31.03 | J4k3 | "heres your ticket, and since you're now late for court I'm going to have to arrest you" |
| 22:31.47 | J4k3 | last time I got pulled over I had to poke fun of the cop for having an entirely hard time catching up with a Hyundai SUV :P |
| 22:31.59 | dockmazter | manx: mm no I dont have! let me give it a try |
| 22:32.04 | sweeper | well, nah, he was like "oh, well that's a bit off!" and left :P |
| 22:32.11 | ManxPower | dockmazter: don't use it |
| 22:33.13 | dockmazter | manx: too late :-] but it doesn change anything |
| 22:35.32 | ManxPower | dockmazter: I can see how having that option might cause issues. |
| 22:36.25 | dockmazter | manx: well, its commented out again and still no change ... *sigh |
| 22:38.29 | *** join/#asterisk Bobocop (n=whooopa@ekc231.neoplus.adsl.tpnet.pl) |
| 22:42.31 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:43.16 | rhombus | ManxPower: Do you have a suggestion for how I get the phone to write an overrides file to the boot server? It's not even trying right now puzzled |
| 22:49.01 | dockmazter | manx: will go on debugging tomorrow.. should get some sleep. thanks for your help anyway! |
| 22:52.55 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 22:53.38 | ManxPower | rhombus: using FTP or TFTP? |
| 22:54.16 | rhombus | ManxPower: Using TFTP |
| 22:54.26 | *** part/#asterisk backblue (n=moo@87-196-6-60.net.novis.pt) |
| 22:56.04 | rhombus | ManxPower: Reading the Admin guide, one could get the impression it's possible to set the Server Menu settings from a config file (the phones default to FTP) but that seems like a "chicken or the egg" problem |
| 22:58.10 | *** join/#asterisk JacksLivr (n=JacksLiv@jules.dougstuff.com) |
| 22:58.47 | rhombus | ManxPower: In particular, the part at the beginning of the admin guide that talks about the "device.set" parameters -- it seems to let you set the server type :) |
| 23:00.21 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 23:01.53 | *** join/#asterisk dseeb_ (n=dcb@CPE-58-169-73-237.vic.bigpond.net.au) |
| 23:03.09 | rhombus | ManxPower: I figured it the overrides -- it doesn't write the file unless you change a setting through the user interface or web interface |
| 23:07.25 | *** join/#asterisk chema (n=chema@211.Red-88-7-4.staticIP.rima-tde.net) |
| 23:11.13 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:12.14 | stubert | Does 1.4.1 have any new dependancies for codec_zap? Do I even need codec_zap with a tdm400p? |
| 23:13.24 | Qwell | stubert: yes, but the configure script will fail to find it, and continue on without building it - and no, you do not |
| 23:14.37 | stubert | Qwell: so, what is it for? |
| 23:14.46 | Qwell | transcoder card |
| 23:14.53 | stubert | aww... |
| 23:14.58 | stubert | thanks |
| 23:16.25 | *** join/#asterisk heh_v_water (n=heh_v_wa@71-210-51-58.hlna.qwest.net) |
| 23:20.12 | JT | is there a way to set the volume of tones generated in asterisk? |
| 23:20.52 | *** part/#asterisk chema (n=chema@211.Red-88-7-4.staticIP.rima-tde.net) |
| 23:21.15 | ManxPower | JT: not just the tones |
| 23:21.36 | ManxPower | of course in SIP, etc the tones are generated by the device, not asterisk |
| 23:22.09 | JacksLivr | hey guys. after at least 12 hours of fighing with this polycom, i am, i think, almost there. can you tell me what i can check for this? I have looked in the groups and have made changes based on what i have see suggested. http://pastebin.ca/381597 |
| 23:22.44 | JacksLivr | i taken out and added the username and have reloaded the sip and asterisk |
| 23:23.03 | JT | ManxPower: yeah, i meant either indications.conf or zonedata.c generated tones |
| 23:23.50 | stubert | JacksLivr: Set host to dynamic, or don't have the policom register... |
| 23:24.09 | stubert | I had the same errors when we first rolled out our policoms |
| 23:24.27 | JacksLivr | ok, it was dyanmic before too. it still gives the same error |
| 23:25.16 | JacksLivr | i changed it back |
| 23:25.24 | JacksLivr | same error |
| 23:25.29 | stubert | Just out of curiousity, does the [name] and the username=name the same in your config? |
| 23:25.49 | JacksLivr | yes |
| 23:27.24 | stubert | then it should work... the only thing I can sugest is maybe adding an fromuser= entry as well... But that usually is only an issue when you have a different auth name then the name in the [] |
| 23:28.21 | ManxPower | stubert: No, but you almost never want them to be different. |
| 23:28.27 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 23:28.37 | JacksLivr | when i try to dial the extension, it gives: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:28.59 | JacksLivr | thanks |
| 23:29.06 | ManxPower | JacksLivr: that usually means the far side is not registered or the host=ip.add.ress is wrong |
| 23:29.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:29.29 | JacksLivr | this is all internal |
| 23:29.38 | Thazza | JacksLivr: You know the password you have is different in the sip and the poly config. |
| 23:29.51 | ManxPower | JacksLivr: my comment still applies |
| 23:29.53 | JacksLivr | oh $%^& |
| 23:30.16 | ManxPower | so the phone is not registered. |
| 23:30.46 | JacksLivr | rebooting |
| 23:32.56 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:34.02 | JacksLivr | ok, i fixed the papssword and I am still getting the same error |
| 23:34.31 | *** join/#asterisk Microgate (n=email@P7a24.p.pppool.de) |
| 23:34.32 | *** join/#asterisk MoutaPT (n=Blink@195-23-28-38.net.novis.pt) |
| 23:35.24 | Microgate | hello, can anyone help me? i want to install one AVM Fritz Card under TrixBox but it don't works |
| 23:35.39 | MoutaPT | What could make my static Agent from a queue receiving incoming calls while on call with a customer? Is there any recently know issue with queues or something else? I'm using TE110P too... |
| 23:35.47 | [TK]D-Fender | Microgate: ... |
| 23:35.49 | [TK]D-Fender | ~trixbox |
| 23:35.55 | jbot | i heard trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 23:37.10 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 23:38.21 | stubert | JacksLivr: I'm showing my reg.1.address= being the same as my reg.1.auth.userId= |
| 23:38.41 | stubert | in your case: polycom |
| 23:39.32 | JacksLivr | ok, i can try that |
| 23:39.55 | rhombus | ~trixbox |
| 23:39.56 | jbot | i heard trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/ |
| 23:40.01 | rhombus | LOL |
| 23:40.07 | MoutaPT | What could make my static Agent from a queue receiving incoming calls while on call with a customer? Is there any recently know issue with queues or something else? I'm using TE110P too.. |
| 23:40.10 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 23:41.16 | Bobocop | is any1 here successfull with Asterfax on Asterisk 1.4.1? Asterfax wants to replace app_txfax.c (from spandsp) with his version, but that one won't even compile... :( |
| 23:42.13 | *** part/#asterisk MoutaPT (n=Blink@195-23-28-38.net.novis.pt) |
| 23:43.35 | ltdwk | Is there any field in ast_channel or any related structure that just has the channel number in integer format, as opposed to name which has Zap/X-Y? |
| 23:43.36 | ManxPower | JacksLivr: http://www.fnords.org/~eric/polycom-config-examples/ |
| 23:43.58 | ManxPower | ltdwk: X is the channel number |
| 23:44.15 | ltdwk | manXPower: read the question again |
| 23:44.25 | ManxPower | Y would be the call number on the channel 1 for the first call, 2 for a 2nd call (call waiting) |
| 23:44.54 | ManxPower | ltdwk: and I am saying that it should be trivial to extract the integer channel number from the name |
| 23:44.57 | ltdwk | i know X is the channel number, but I was asking if it was in integer format accessable anywhere else |
| 23:45.16 | ltdwk | It's trivial to extract, but incorrect and hacky if it's available somewhere else |
| 23:45.17 | ManxPower | ltdwk: you might want to try #asterisk-dev as well. |
| 23:45.26 | *** part/#asterisk Microgate (n=email@P7a24.p.pppool.de) |
| 23:46.03 | ltdwk | manxpower: thanks, wasn't sure if such a channel existed |
| 23:46.36 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:50.20 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.146.9) |
| 23:54.05 | *** join/#asterisk googledude (n=googledu@h4609f8b0.area4.spcsdns.net) |
| 23:59.40 | Daejeo1 | guys plz have a look http://www.signalogic.com/index.pl?page=asterisk_ip_pbx |