| 00:01.00 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 00:02.28 | blitzrage | rhombus: right -- pager doesn't attach the file |
| 00:03.08 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 00:03.31 | rhombus | blitzrage: thanks -- funny that isn't really described anywhere |
| 00:04.31 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 00:09.38 | mrdigital | anyone know how to install hamachi on linux? |
| 00:11.22 | blitzrage | what is a hamachi? |
| 00:11.47 | blitzrage | I don't think that is asterisk related |
| 00:12.42 | *** join/#asterisk StarSong (n=illusion@200.68.73.133) |
| 00:13.05 | mrdigital | hamachi is a VPN software |
| 00:13.13 | mrdigital | has anyone installed it on their asterisk box to remotely admin it |
| 00:13.19 | DrukenLPY | is there something with tftp where it only works locally? |
| 00:13.20 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 00:13.43 | StarSong | Hi, im new into this, what im i supposed to use to serve up like 50 lines from my computer ? |
| 00:13.58 | _VoiceMeUp_Com | Drunken check the bind address |
| 00:14.05 | _VoiceMeUp_Com | im out night |
| 00:14.12 | StarSong | night |
| 00:14.46 | DrukenLPY | _VoiceMeUp_Com: nah.. not that... if i have my phones on the local network, they use the tftp fine, if i try from abroad, no luck... but it DOES grab the file... |
| 00:14.53 | DrukenLPY | might be a nat problem.... i dunno |
| 00:15.26 | *** join/#asterisk grndslm (n=grndslm@host-69-59-127-198.nctv.com) |
| 00:16.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:18.10 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 00:20.56 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:21.03 | JT | mrdigital: i doubt many would bother with a vpn just to admin their asterisk boxes |
| 00:22.12 | mihinomenest | so, asterisk is all but deployed at work and management decided they have just one other thing they want. |
| 00:22.48 | mihinomenest | they'd like to have a "high-priority" line that gets handled differently from the rest of the calls. |
| 00:22.55 | mihinomenest | how hard is that to implement? |
| 00:23.16 | JT | depends |
| 00:23.25 | JT | you'll need to explain what that actually means |
| 00:23.47 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 00:23.48 | flenders | jt: morning |
| 00:23.49 | tessier_ | Hello all! |
| 00:24.24 | tessier_ | In my musiconhold.conf file I used to have mode=mp3 but I just changed it to mode=quietmp3 but the music still seems rather loud. |
| 00:24.31 | JT | flenders: hey :) |
| 00:24.54 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 00:25.04 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 00:25.07 | flenders | do you not have vpn access to your asterisk box at work? |
| 00:25.20 | BSD_Tech | I work for my self |
| 00:25.24 | BSD_Tech | right now |
| 00:25.36 | BSD_Tech | and my box is in the same room as my phone |
| 00:25.52 | JT | flenders: what on earth do i need a VPN to use SSH for? :) |
| 00:26.23 | flenders | so you do have ssh access? |
| 00:26.28 | JT | yes |
| 00:26.30 | flenders | but not through vpn |
| 00:26.35 | JT | right |
| 00:26.45 | flenders | ok, I have ssh access through vpn |
| 00:26.53 | JT | seems redundant |
| 00:27.03 | flenders | the thing is not just encryption |
| 00:28.23 | flenders | all the servers here can only be accessed through vpn |
| 00:28.23 | flenders | very few have http or some other port open |
| 00:28.23 | tessier_ | nat sucks. :( |
| 00:28.23 | BSD_Tech | thats why you iax trunk them |
| 00:28.23 | flenders | I can ssh to the firewalls, though |
| 00:28.23 | BSD_Tech | so you dont have nat issues |
| 00:28.23 | JT | flenders: ok |
| 00:28.27 | JT | meh |
| 00:28.34 | JT | it isn't actually that hard to get SIP working through NAT |
| 00:28.39 | JT | unless your NAT device is rubbish |
| 00:29.09 | JT | also, people don't usually need IAX Trunking, IAX will do for most |
| 00:29.23 | JT | well, IAX2 |
| 00:29.24 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
| 00:30.05 | mihinomenest | JT: well, a normal call comes in on the normal sip connections, gets sent to the menus, then into the queues or something. they want a call from this "high priority" line to come in and get sent straight to a queue that rings a different set of phones. |
| 00:30.21 | elriah | Hi all. I have 200 DIDs I need to forward to a specific context or individual sip peers. I also need to set an account code on the dids prior to sending them to a context or the sip peer. I know how to do it one at a time, is there a more effecient way to do a large qty of dids this way? |
| 00:30.29 | elriah | Oh, 1.2.16. |
| 00:30.53 | JT | mihinomenest: if someone is calling from a different DID or callerid, it should be quite doable |
| 00:31.10 | JT | elriah: realtime |
| 00:31.31 | elriah | I need to be able to use qualify in my sip.conf so realtime is out (sigh). |
| 00:32.10 | JT | i don't get it |
| 00:33.44 | pfn | I would just do _NPANXXXXXX => Dial(SIP/${EXTEN} ...) and maybe some preprocessing to nuke off the prefix |
| 00:34.00 | mihinomenest | I don't suppose I can do this on a specific sip account? |
| 00:34.03 | pfn | qualify... why use realtime, just use static config and sip reload |
| 00:34.22 | elriah | hrm... |
| 00:34.34 | JT | why won't qualify work with realtime? |
| 00:34.46 | pfn | rtcachefriends should let qualify work |
| 00:35.07 | JT | mihinomenest: why not? |
| 00:35.50 | mihinomenest | don't all incoming sip calls get sent to the context specified in "[default]" in sip.conf? |
| 00:36.30 | pfn | JT, http://www.voip-info.org/wiki-Asterisk+RealTime -- re realtime + qualify, etc. |
| 00:36.42 | JT | no, you set the context per sip.conf entry, mihinomenest |
| 00:36.46 | pfn | but yeah, you need to set rtcachefriends |
| 00:36.53 | mihinomenest | oh. |
| 00:36.55 | JT | default should not normally need to be used |
| 00:37.44 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 00:38.05 | *** part/#asterisk dasenjo (n=be185fc7@acuario.unicauca.edu.co) |
| 00:38.50 | mihinomenest | so all I really have to do is set it's context in sip.conf to a context in extensions.conf that does what I want. I wish I'd thought of that. |
| 00:38.58 | JT | heh |
| 00:42.27 | JT | pfn: a lot of poorly worded explanations there, but if i read and understand asterisk realtime correctly, if you use the STATIC realtime method with DATABASE config storage, qualify should work fine? |
| 00:43.12 | mihinomenest | I'll have to come up with a way to make it sound complex and boring. otherwise management will think that they can have other business' phone system on my wee box. |
| 00:43.21 | mihinomenest | whereas they're just wrong in that thought. |
| 00:43.45 | pfn | JT no, if you use pure realtime config then chan_sip never knows about any sip peer/user until a call is made |
| 00:43.50 | pfn | at which point it loads from the database |
| 00:44.05 | JT | pfn: can't you use realtime realtime and static realtime? |
| 00:44.13 | pfn | jt from what I've read, no |
| 00:44.23 | pfn | you can use one or the other, not both at the same time (for the same file) |
| 00:44.49 | pfn | jt and yeah, static realtime works fine |
| 00:45.12 | JT | pfn: the way i understand it is realtime realtime it reads the database when there's a connect attempt or what not, but static realtime you need to do a sip reload and it caches what's in the db |
| 00:45.19 | JT | right |
| 00:45.20 | JT | so problem solved |
| 00:45.25 | pfn | jt that's right |
| 00:45.31 | pfn | but static realtime is kind of a pain, imo |
| 00:45.40 | pfn | "realtime" realtime is what's cool |
| 00:46.10 | JT | elriah is just too impatient to hear the answer it would seem |
| 00:46.19 | JT | well clearly it's not implemented that well |
| 00:46.25 | pfn | indeed |
| 00:46.55 | JT | but a database is a much easier way to handle 200 extensions than flatfile |
| 00:47.08 | pfn | have you tried realtime static? it's practically a flatfile |
| 00:47.19 | pfn | except "ported" to a database |
| 00:47.21 | *** join/#asterisk angom (n=angom@red-corp-200.79.141.185.telnor.net) |
| 00:48.16 | *** join/#asterisk lokiloch (n=me@203.82.44.179) |
| 00:48.45 | JT | pfn: yes but it's easier to make interfaces to manipulate a db, than flat files |
| 00:48.52 | Corydon76-home | pfn: it's STILL res_config_odbc |
| 00:49.28 | pfn | Corydon76-home, just not called that anymore :p |
| 00:49.37 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
| 00:49.38 | Corydon76-home | pfn: yes, it is |
| 00:49.41 | pfn | is it, heh |
| 00:49.59 | Corydon76-home | There aren't any plans to change it, either |
| 00:53.07 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 00:59.16 | flenders | hey does anyone know how much does the enterprise asterisk cost? |
| 00:59.30 | flenders | or the asterisk for businesses? |
| 01:00.10 | flenders | nevermind, its on the website |
| 01:15.22 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 01:15.30 | anonymouz666 | <PROTECTED> |
| 01:16.35 | StarSong | flenders: How much how much! |
| 01:16.55 | StarSong | :o :O :o :O |
| 01:21.29 | flenders | almost 1K |
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| 01:31.50 | DrukenLPY | anyone have experince with aastra phones? |
| 01:32.33 | Dr-Linux|home | does Cepstral work with asterisk? |
| 01:32.44 | DrukenLPY | yes |
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| 01:46.26 | lee_is_me | Dr-Linux: Cepstral works very nicely for me on 1.2.14 |
| 01:56.59 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 01:57.31 | SwK | anyoen have a link to the vonage "no work around" on the patents story? |
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| 02:00.31 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:00.55 | tengulre | hi,all |
| 02:02.37 | tengulre | I have a jabber, how to setting the asterisk to connect it? |
| 02:03.14 | Corydon76-home | Anybody try to interface a Coral ISBX with Asterisk? |
| 02:04.55 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-51-85.dsl.irvnca.pacbell.net) |
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| 02:37.51 | tengulre | anybody active? |
| 02:38.55 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 02:39.16 | *** join/#asterisk i3inary (i=i3inary@ip68-8-91-87.sd.sd.cox.net) |
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| 02:43.23 | DocHolliday | hey swk :) |
| 02:46.51 | i3inary | wondering if someone could help me out with http://pastebin.ca/443929 |
| 02:47.43 | i3inary | i am basically looking for a solid efficient method of banning originating numbers which are in .call files |
| 02:48.56 | DocHolliday | anyone here do canada termination / origination or able to recommend me to someone who does? |
| 02:49.33 | JT | i3inary: what is the blocked call detection method? |
| 02:49.58 | i3inary | manual atm |
| 02:50.13 | JT | what would you like it to be |
| 02:50.19 | JT | i assume you have something in mind |
| 02:50.44 | i3inary | heheh...yeah it would be a customer that isnt generating any revenue for me after x period of time |
| 02:51.11 | i3inary | but for right now its a manual method..i dont have the algorithim for it right now |
| 02:51.16 | JT | yes but i asked about the detection method |
| 02:51.19 | JT | ip address |
| 02:51.23 | JT | phone number |
| 02:51.27 | i3inary | either or |
| 02:51.37 | JT | do they have an account? |
| 02:51.47 | i3inary | yes but its not required at the moment |
| 02:52.20 | i3inary | right now users can make calls for free without even logging in...the only advantage to logging in is enhanced features |
| 02:52.35 | JT | sounds like a recipe for a big phone bill |
| 02:52.37 | i3inary | which are few and far between atm |
| 02:52.45 | i3inary | heheh thats ok its just a prototype |
| 02:52.52 | i3inary | its just to get vc money |
| 02:53.55 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:54.02 | i3inary | basically i currently have couple hundred users...only a few of them talk on the phone to certain numbers for hours on end...and i want to ban them for now |
| 02:54.29 | i3inary | since i foot the bill and they arent making my life easy atm |
| 02:54.45 | JT | then do it |
| 02:54.48 | i3inary | couple long calls ...im ok with...but these are talkaholics |
| 02:56.07 | i3inary | so if im going to ban by phone number what would you suggest? |
| 02:56.36 | JT | that you check the phone number dialled before puting it through? |
| 02:56.55 | i3inary | in the php not the .conf right? |
| 02:57.07 | i3inary | do it before it gets to asterisk then right |
| 02:57.32 | JT | either way, up to you |
| 02:57.42 | JT | the dialplan can check too |
| 02:58.31 | i3inary | im assuming its better to not burden asterisk if possible right? |
| 02:58.32 | JT | i'm not sure how high the burden really is |
| 02:58.41 | JT | main thing is that it's secure from user interference |
| 02:58.51 | i3inary | right agreed |
| 03:07.10 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:11.45 | pigpen | is there any easy way in the dialplan to tell if an extension is an iax or sip peer? |
| 03:12.54 | pigpen | ie: if sip, then dial SIP/ or if iax then dial IAX2/ |
| 03:13.41 | DocHolliday | anyone used callcentric? |
| 03:13.45 | mrdigital | hey jt can i pm you |
| 03:15.26 | DocHolliday | guess not :( |
| 03:16.44 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 03:17.13 | jarrod | what is the best way to integrate a presence utility into our voip environment w/ asterisk/ser |
| 03:18.25 | Nugget | Hire an intern to constanly cycle through the office and keep notes on who is present and who is away. |
| 03:18.59 | pigpen | Personally, I have a blue light this is turned on and off when I am on the phone....you know..kinda like Kmart "blue light special" |
| 03:19.11 | jarrod | that is great :-D |
| 03:19.12 | jarrod | haha |
| 03:22.00 | *** join/#asterisk jmacz (n=jmacz@201.244.171.105) |
| 03:23.46 | pigpen | anyone know if you can do a #include "mine/*.conf" in the dialplan? |
| 03:23.55 | pigpen | if not, it would be kinda cool. |
| 03:24.07 | Nugget | neat idea |
| 03:24.28 | pigpen | yeah..it would probably suck resources...or something to screw my day. |
| 03:26.30 | Nugget | Why stop there. We need remote includes... #include http://macnugget.org/asterisk/generic.conf |
| 03:27.09 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:27.15 | wunderkin | http://macnugget.org/porn/good/gay/littleboysex.jpg |
| 03:27.17 | wunderkin | lol j/k |
| 03:27.22 | DocHolliday | are there any channels where voip providers tend to be present? |
| 03:27.28 | Nugget | this one. |
| 03:28.14 | pigpen | yeah... |
| 03:28.16 | DocHolliday | Nugget, i havent had very good luck.. still trying to find a Canada origination provider |
| 03:28.37 | mrdigital | DocHolliday: define orgination |
| 03:28.43 | Nugget | perhaps irc isn't the best approach if you have specific needs like that. |
| 03:28.50 | wunderkin | to originate |
| 03:28.52 | pigpen | DocHolliday, Vonage. |
| 03:28.53 | pigpen | :) |
| 03:29.03 | mrdigital | DocHolliday: VoiceMeup.com |
| 03:29.08 | mrdigital | is from Canada |
| 03:29.19 | DocHolliday | pigpen, wholesale :) |
| 03:29.41 | mrdigital | DocHolliday: VMU does Wholesale |
| 03:29.45 | DocHolliday | Nugget, right.. just trying to get general ideas.. with so many voip providers out there it would be nice to narrow the list |
| 03:29.49 | pigpen | I wasn't serious anyway. In fact, I am suprised I didn't get kicked from the channel for that one. |
| 03:30.18 | pigpen | shit, kick me out of the state for that one. |
| 03:30.19 | pigpen | :) |
| 03:30.23 | Nugget | You might try posting to the asterisk-biz mailing list. |
| 03:30.25 | pigpen | big state...big kick. |
| 03:30.28 | wunderkin | send him all the way back to new kids on the block |
| 03:30.34 | wunderkin | or wtf that was |
| 03:30.44 | DocHolliday | Nugget, good idea. |
| 03:31.30 | mrdigital | can anyone offer suggestions on features i should add to the IVR Menu |
| 03:31.39 | mrdigital | we're a online retail clothing store |
| 03:31.46 | mrdigital | i already added a order status lookup mod |
| 03:31.50 | wunderkin | now pigpen is a dirty whore voip dude |
| 03:32.14 | pigpen | wunderkin, yeah..I am a general slut....but I have standards now. |
| 03:32.21 | pigpen | its a start. |
| 03:32.24 | wunderkin | 'you must be this tall' |
| 03:32.35 | *** join/#asterisk mjun007 (n=mjun007@221.221.149.55) |
| 03:33.05 | wunderkin | 'no shirt or shoes required, preferrably not' |
| 03:33.22 | pigpen | hmm..sounds like my single days... |
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| 03:38.05 | Nugget | http://macnugget.org/legal/ |
| 03:38.28 | wunderkin | /barelylegal :P |
| 03:39.55 | Nugget | heh |
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| 03:56.26 | Defraz | Does anyone know of a simulator for the Polycom, Cisco, or Aastra phones. I want to try some XML stuff and I don't have the cash to buy the phones for play. I have clients that have them but just nothing to develop with? |
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| 04:02.12 | pfn | wcfxs: disagrees about version of symbol zt_receive |
| 04:02.13 | pfn | wcfxs: Unknown symbol zt_receive |
| 04:02.15 | pfn | hmm, wtf |
| 04:03.41 | pfn | nevermind, old zaptel 1.0 stuff |
| 04:05.10 | kuku5 | When I do an assisted transfer, is there a way to do a beep so everyone know that the call is transfered ? |
| 04:07.34 | pfn | why? it's an attended transfer |
| 04:07.49 | JunK-Y | my grand-ma is using 1.0 :) |
| 04:08.09 | pfn | heh, I was using 1.0 up until a few minutes ago |
| 04:08.30 | pfn | now I'm using nothing... until I get branch-1.4 configured |
| 04:08.50 | JunK-Y | i c. |
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| 04:13.01 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 04:24.57 | JoelSolanki | Hi all |
| 04:25.55 | JoelSolanki | i m back with my problem of early media. |
| 04:26.06 | JoelSolanki | someone told me to talk on irc |
| 04:26.16 | JoelSolanki | does asterisk support early media disabled ? |
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| 04:38.34 | illsci | hey its either sip or iax2 right... |
| 04:38.37 | illsci | you don't use both |
| 04:38.41 | illsci | correct? |
| 04:38.44 | FuriousGeorge | yeah |
| 04:38.46 | FuriousGeorge | correct |
| 04:38.49 | FuriousGeorge | well |
| 04:38.51 | FuriousGeorge | actually no |
| 04:38.53 | FuriousGeorge | incorrect |
| 04:39.04 | illsci | for incomming and outgoing calls... |
| 04:39.05 | FuriousGeorge | i can make a call from my sip phone via asterisk |
| 04:39.11 | FuriousGeorge | using an iax probider |
| 04:39.13 | FuriousGeorge | provider |
| 04:39.41 | FuriousGeorge | that is why asterisk is sometimes (never in here) referred to as a b2bua (back to back user agent) |
| 04:40.02 | JT | in sip terms it's a B2BUA |
| 04:40.13 | FuriousGeorge | it sits between two users, in my scenario your sip phone and your iax provider, and makes the connections |
| 04:40.29 | FuriousGeorge | JT: thats what i said sans caps |
| 04:40.41 | JT | and it acts as a user agent on each side, not a proxy |
| 04:40.51 | JT | well it's a sip specific term really |
| 04:40.56 | FuriousGeorge | i see what you mean |
| 04:41.02 | FuriousGeorge | in sip terms |
| 04:41.18 | FuriousGeorge | so i have this one server that like to deadlock |
| 04:41.24 | FuriousGeorge | *likes |
| 04:41.31 | ManxPower | It's not true if you don't have the caps. |
| 04:41.31 | FuriousGeorge | trying to figure out what it is |
| 04:41.59 | FuriousGeorge | my newest greatest theory is the overloaded UPS its attached to |
| 04:42.19 | FuriousGeorge | though the damn thing doesnt beep at me or anything |
| 04:42.28 | ManxPower | FuriousGeorge: I have found that random hard lockups with Linux are usually some failing or bad piece of hardware |
| 04:42.37 | FuriousGeorge | Power alarm on module 1, resetting! |
| 04:42.42 | FuriousGeorge | thats in my dmesg |
| 04:42.50 | ManxPower | Ugh. Analog cards. |
| 04:42.57 | FuriousGeorge | i think its referring to my zap card |
| 04:43.01 | ManxPower | FuriousGeorge: perhaps the power supply is too weak. |
| 04:43.09 | FuriousGeorge | its a 600 watt mushkin |
| 04:43.20 | ManxPower | FuriousGeorge: what did your google search for that term turn up? |
| 04:43.35 | FuriousGeorge | since i was here i tried the #asterisk search first :) |
| 04:44.13 | FuriousGeorge | b/c ironically i was telling Mercestes earlier that deadlocks always coincide with use of zaptel hardware |
| 04:45.43 | ManxPower | Well is it sharing interrupts? |
| 04:45.52 | FuriousGeorge | ManxPower: negative |
| 04:45.54 | FuriousGeorge | apic |
| 04:45.57 | FuriousGeorge | 64bit |
| 04:46.03 | FuriousGeorge | i do have two cards though |
| 04:46.15 | FuriousGeorge | one with three fxo and one with four fxs |
| 04:46.22 | ManxPower | I have seen APIC put multiple devices on the same IRQ |
| 04:46.35 | FuriousGeorge | u thinking of acpi? |
| 04:47.06 | FuriousGeorge | im wrong |
| 04:47.07 | FuriousGeorge | ur right |
| 04:47.12 | ManxPower | Hmm? I always use acpi unless I have few devices enabled and have plenty of XT-PIC IRQs available |
| 04:47.27 | FuriousGeorge | i do share interrupts, but the wctdms got there own |
| 04:47.37 | ManxPower | Cool. |
| 04:47.41 | FuriousGeorge | so apic does not necessarily assign irq exclusively |
| 04:48.12 | FuriousGeorge | really though, my discovering that message isnt what is confusing me. its electronics in general |
| 04:48.39 | FuriousGeorge | there is this brand-x UPS running my two servers and a gateway i installed for them |
| 04:49.21 | ManxPower | The backup mail server, located a 3 hr drive away keeps hard locking. I keep telling them that it is cleaper to replace it than to figure out what the problem is. We always have need of a server box. |
| 04:49.34 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 04:49.53 | FuriousGeorge | and this location is haunted by failing hw, so i google the UPS and its only 970 watts, running two 500W+ servers, a ~300W gateway |
| 04:50.09 | FuriousGeorge | i guess PSU dont constantly draw at their max output, do they? |
| 04:50.32 | JT | of course not |
| 04:50.42 | JT | that's quite a small UPs |
| 04:50.45 | JT | UPS |
| 04:51.08 | FuriousGeorge | JT: what you mean 'of course'? you were born this smart or something :P |
| 04:51.16 | ManxPower | UPSs fail, he larger ones are expensive. Put a dedicated UPS on each box. |
| 04:51.48 | FuriousGeorge | and a dedicated one for all the switches? |
| 04:51.54 | FuriousGeorge | and gizmos |
| 04:52.03 | FuriousGeorge | and juice blender |
| 04:52.08 | ManxPower | Whaever works. |
| 04:52.10 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 04:52.18 | flenders | get an expensive one then |
| 04:52.36 | ManxPower | I have a fan blowing into the two racks and that is plugged into a UPS as well. |
| 04:52.44 | JT | FuriousGeorge: do you have any idea how much heat would be generated if they pulled max power all the time? or how big your power bill would be? |
| 04:52.53 | FuriousGeorge | i have an idea |
| 04:53.19 | FuriousGeorge | 1.8 kw X .13 dollars X 24 # 31 |
| 04:53.29 | FuriousGeorge | s/#/X |
| 04:53.41 | JT | switchmode power supplies are highly efficient |
| 04:54.00 | groogs[h] | ManxPower: that seems like a waste of battery.. if the powerfails, you don't get that much time anyways, does it matter if the servers get a bit warm? |
| 04:54.19 | FuriousGeorge | i appreciate "youze guys's" input as always, but what are the odds that this is causing my deadlocks without the UPS complaining? |
| 04:54.55 | FuriousGeorge | i could see if the m/fer was beeping at me constantly, but it isnt |
| 04:55.31 | FuriousGeorge | english needs a 2nd persons plural |
| 04:55.38 | ManxPower | groogs[h]: It takes little power and the equipment gets hot. Also I am sometimes gone for 2 weeks at a time and if the power goes out the fan will NOT start back up. |
| 04:55.52 | JT | groogs[h]: air conditioning generally fails if power fails |
| 04:55.58 | Defraz | Does anyone know of a simulator for the Polycom, Cisco, or Aastra phones. I want to try some XML stuff and I don't have the cash to buy the phones for play. I have clients that have them but just nothing to develop with? |
| 04:56.03 | groogs[h] | ManxPower: oh, thats a good reason then |
| 04:56.20 | groogs[h] | JT: yeah, but if you only get, say, 30 mins runtime anyways.. |
| 04:56.50 | JT | groogs[h]: it's good to blow the heat away from the servers then |
| 04:57.49 | ManxPower | and with a fan you might only get 25 mins of run time. You can never afford to enough power backup for an extended outage and most short outages last less then 25 mins. |
| 04:57.52 | groogs[h] | i dunno, a fan is another 60-80W of load i would guess |
| 04:58.10 | groogs[h] | yeah. its probably a marginal difference |
| 04:58.43 | groogs[h] | guess it really comes down to, do you want extra 5 mins runtime, or do you want your servers to stay 5 degrees cooler.. |
| 04:59.10 | CunningPike | Defraz: Polycom has an excellent 501 emulator for only US$215 |
| 04:59.10 | ManxPower | I have a lot of stuff in my racks besides servers. |
| 04:59.29 | CunningPike | Defraz: It's just like the real thing :D |
| 04:59.32 | mosty | does anyone know how the g729 register program works? the latest version doesnt work on my box (cant determine host id) and the old version i have handy cant connect to the digium server (connection refused) |
| 05:00.16 | ManxPower | mosty: the host id is determined by all the ethernet addresses of the system. |
| 05:00.26 | illsci | hey do you know of an iax based soft phone for linux? |
| 05:00.27 | Qwell | mosty: What are the interface names? |
| 05:00.33 | illsci | something like http://www.laser.com/dante/diax/diax.html |
| 05:00.37 | Qwell | illsci: that one...heh, umm |
| 05:00.49 | Qwell | ugh, what's that one Zoa makes called? |
| 05:01.07 | [TK]D-Fender | qwell : idefisk |
| 05:01.11 | kaldemar | zoiper |
| 05:01.13 | Qwell | that's the one |
| 05:01.16 | mosty | qwell: they are renamed to stuff like uplink and lan (box has 4 net interfaces) |
| 05:01.24 | Qwell | mosty: yeah, don't do that |
| 05:01.30 | mosty | i have to |
| 05:01.36 | illsci | so idefisk? |
| 05:01.37 | Qwell | Then you can't use the register tool |
| 05:01.43 | mosty | the old register util worked, i think |
| 05:01.59 | mosty | it just cant connect to the license server |
| 05:02.01 | Qwell | On Linux, the interfaces *must* be named eth* |
| 05:02.12 | Qwell | you need at least one with that name |
| 05:02.29 | CunningPike | Has anyone used an Audiocodes FXS gateway with Asterisk and figured out how to get each port to register separately for initiating calls? Calls _to_ the ports work great, but call _from_ don't authenticate properly |
| 05:02.31 | JT | groogs[h]: i think you really are guessing, 80W is an industrial fan |
| 05:02.56 | mosty | qwell: is that just a requirement of the register util? nothing else in linux requires that ethX be used |
| 05:02.59 | Qwell | yes |
| 05:03.26 | Qwell | for the record, I've suggested changing that, but it would mess with existing registrations |
| 05:03.27 | mosty | qwell: is there a way i can create an aliased interface? i need to use these names |
| 05:03.32 | Qwell | no idea |
| 05:03.37 | Qwell | why "must" you use those names? |
| 05:04.18 | mosty | because when we boot different kernels the order of the interfaces changes, and my network config breaks |
| 05:04.47 | FuriousGeorge | i do use oej's metermaid patch with 1.2.X. these guys use parking a lot |
| 05:04.48 | mosty | i need to bind them to a permanent name |
| 05:05.05 | Qwell | mosty: udev rules |
| 05:05.09 | FuriousGeorge | anyone know if metermaid patch has been known to cause deadlocks? |
| 05:06.15 | mosty | qwell, i'll try that then i guess |
| 05:13.50 | mosty | Qwell, do you happen to know what host/port the register util contacts? |
| 05:14.20 | Qwell | no |
| 05:15.26 | mosty | would anyone be willing to run tethereal when running register and tell me which digium host/port it connects to? |
| 05:17.22 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:22.10 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 05:22.37 | pfn | duh... how do I use dialplan functions? |
| 05:23.07 | pfn | [Apr 16 22:22:08] WARNING[32422]: pbx.c:1783 pbx_extension_helper: No application '${CUT' for extension (macro-incoming, s, 1) |
| 05:25.42 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:27.28 | SwK | 207.210.100.226 <--- beware of this IP address it is currently trying to place calls on random SIP gateways to the ivory coast... |
| 05:27.53 | SwK | (shouldnt be a problem if ou have your stuff configured properly... but.... |
| 05:29.22 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:29.38 | FuriousGeorge | the official recommended limit of tdm400p cards per system is still 2, right? |
| 05:30.16 | SwK | i thought it was 0 |
| 05:30.28 | SwK | :P |
| 05:32.40 | FuriousGeorge | SwK: i think thats the ideal limit |
| 05:33.02 | FuriousGeorge | SwK: you use sangoma or something? |
| 05:33.10 | SwK | I use a T1 card and a channel bank |
| 05:33.17 | FuriousGeorge | lucky you |
| 05:33.28 | SwK | do the math its not that bad |
| 05:33.35 | FuriousGeorge | maybe where you are |
| 05:33.38 | *** join/#asterisk `p4r14h (n=j0sh@69.92.206.192) |
| 05:33.43 | FuriousGeorge | im here 10 min from nyc |
| 05:33.58 | FuriousGeorge | and a t1 is b/t 400-500/mo |
| 05:34.17 | SwK | no you miss the point |
| 05:34.24 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:34.26 | FuriousGeorge | i thought we were doing math |
| 05:34.29 | SwK | 2 x TDM400 w/ 4 ports each is like $800 |
| 05:34.38 | FuriousGeorge | ok |
| 05:34.59 | FuriousGeorge | i see where you are going |
| 05:35.10 | aptura | looksing at asterisk sounds dont see anything called record the message |
| 05:35.17 | SwK | 1 T1 card is $595 + $400 for a channel bank... and if you ever decide to upgrade... |
| 05:35.24 | *** join/#asterisk `p4r14h (n=j0sh@69.92.206.192) |
| 05:35.30 | aptura | Swk hello |
| 05:35.42 | SwK | hello |
| 05:35.51 | FuriousGeorge | ive never interfaced with a t1, but just from my experience with chan_zap, i'd love to try |
| 05:36.01 | FuriousGeorge | regardless its out of most of my clients' price range |
| 05:36.21 | JT | FuriousGeorge: a channel bank has nothing to do with what your local telco offers for T1s :) |
| 05:36.37 | FuriousGeorge | o right |
| 05:36.45 | FuriousGeorge | you can set up channels to be fxo as well |
| 05:36.49 | FuriousGeorge | on some of them |
| 05:37.20 | JT | yes, that is channel bank dependant |
| 05:37.25 | FuriousGeorge | like i said, never used a t1 or c.b. |
| 05:37.30 | SwK | FuriousGeorge: the whole point of the T1 + channel bank is by the time you load out 2 TDM400's a T1 card and a channelbank for FXO/FXS ports is only about 2 to 300 more and handles 3 times more ports |
| 05:38.27 | JT | another thing |
| 05:38.39 | FuriousGeorge | SwK: i see what you guys mean now. i was thinking for some reason the c.b. was fxs only |
| 05:38.42 | JT | interfacing 320832727024 analogue lines directly into your asterisk box is a pain in the arse |
| 05:39.06 | FuriousGeorge | yeah, i hate when the number of lined is a float |
| 05:39.10 | FuriousGeorge | *lines |
| 05:40.28 | FuriousGeorge | which brings me back to my original problem of the deadlocks. im researching how to debug them, and the wiki is all like: "step one: fire up your debugger; step two: wait for deadlock; step three: debug" |
| 05:40.37 | SwK | furiousgeorge: some cb are fxs only... most have cards you can swap out in groups or 4 to 6 |
| 05:40.39 | pfn | <PROTECTED> |
| 05:40.44 | FuriousGeorge | now i'm not a coder, this is all somewhat over my head |
| 05:40.47 | pfn | hmm, why do I get that? why doesn't it go to the t extension? |
| 05:41.16 | FuriousGeorge | pfn: i think you need a 'g' in there with the options in the dial command, or something |
| 05:41.24 | FuriousGeorge | maybe |
| 05:41.27 | mosty | Qwell, i managed to create a fake ethernet interface and register with that :) |
| 05:41.45 | pfn | FuriousGeorge, this is on an incoming call |
| 05:41.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:42.00 | JT | FuriousGeorge: how is 320832727024 a floating point number? |
| 05:43.24 | FuriousGeorge | JT: i was being funny, but i thought that would be because it is larger than 2^32 and my os is 32-bit |
| 05:44.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:44.01 | FuriousGeorge | JT: either way it makes my point as to why debugging deadlocks is just not for me |
| 05:44.27 | FuriousGeorge | best i can do is swap things basically at random and see if i start doing better than a deadlock every 100 hours |
| 05:44.42 | FuriousGeorge | or i can have asterisk restart daily and settle for a deadlock every 100 days |
| 05:45.12 | JT | FuriousGeorge: floats have decimal points, the number above is just an int, or long int, depending on lang/arch :) |
| 05:45.15 | *** part/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 05:45.31 | *** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net) |
| 05:45.37 | FuriousGeorge | i thought longer than a long int was expressed as a float |
| 05:45.38 | FuriousGeorge | my bad |
| 05:46.12 | FuriousGeorge | s/expressed/"sored or something" |
| 05:46.17 | FuriousGeorge | s/sored/stored |
| 05:46.24 | FuriousGeorge | anyhoo |
| 05:48.00 | *** join/#asterisk moranil (n=moranil@122.162.67.129) |
| 05:49.11 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:50.47 | *** join/#asterisk stridernzl (n=neville@125-237-110-53.jetstream.xtra.co.nz) |
| 05:51.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 05:53.54 | FuriousGeorge | you know, ive never had to pay anyone to fix something * related for me, but i think im going to have to this time |
| 05:54.05 | FuriousGeorge | im going to lose this client with another deadlock or two |
| 05:54.40 | JT | so the UPS... ruled it out? |
| 05:55.38 | FuriousGeorge | JT: no, but im just finding it hard to convince myself the ups is causing asterisk to deadlock without complaining at all. especially considering how noisy those things are when they are unhappy |
| 05:56.10 | JT | i suppose |
| 05:56.43 | FuriousGeorge | first thing i was gonna try was reinstalling OS and software from the almost identical happy server, then i'm going to swap tdm400p cards and modles |
| 05:56.45 | FuriousGeorge | modules |
| 05:56.59 | FuriousGeorge | and im gonna teach one of them to ssh in and reboot the server when needed |
| 05:57.08 | pfn | [Apr 16 22:56:50] WARNING[32255]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 |
| 05:57.11 | pfn | why do I get that... |
| 05:57.18 | aptura | FuriousGeorge might be a good idea. |
| 05:57.20 | JT | in the long term, a switch to a TDM800P or TDM2400P or something might be the go also |
| 05:57.50 | FuriousGeorge | never heard of the 800, was looking at the 2400... hmmmm 800.. is that what i think it is.... brb |
| 05:58.12 | JT | the 800 is a small 2400 |
| 05:58.16 | JT | it takes 2400 modules |
| 05:58.43 | FuriousGeorge | nuts |
| 05:59.01 | FuriousGeorge | was hoping it would take my 400p modules |
| 05:59.10 | JT | they won't fit :) |
| 05:59.19 | FuriousGeorge | until i get my hammer |
| 05:59.27 | FuriousGeorge | then we'll see |
| 06:00.23 | aptura | FuriousGeorge yea and the credibility with that one clinet. |
| 06:00.41 | JT | you could get a hardware watchdog card, FuriousGeorge |
| 06:00.50 | aptura | JT what is that? |
| 06:00.51 | FuriousGeorge | aptura: ive lost that already, im afraid |
| 06:01.01 | FuriousGeorge | JT that smdc stuff? |
| 06:01.18 | Strom_M | the 800 can take modules from the 400 |
| 06:01.24 | Strom_M | but it can only fit four of them |
| 06:01.25 | aptura | FuriousGeorge thats the pisser with phone technoligy needs to be up 99% of the time unlike other products. |
| 06:01.32 | FuriousGeorge | aptura: sure is |
| 06:01.35 | JT | a hardware watchdog client sits in a slot and receives heartbeats from a driver, if it fails to receive them after a period of time it reboots the pc |
| 06:02.41 | FuriousGeorge | JT: yeah, those would probably be a good idea anyway. of course, im only guessing that my tdm400p is the cause of the issue |
| 06:02.47 | osiris | i used them in a bitchy video survalence system |
| 06:03.07 | osiris | nice little pice |
| 06:03.17 | osiris | er oiece |
| 06:03.23 | osiris | er piece |
| 06:05.24 | osiris | ended those anyoing "it locked up again" service calls |
| 06:05.24 | JT | nice |
| 06:06.18 | FuriousGeorge | hate those |
| 06:06.36 | aptura | yea |
| 06:06.56 | FuriousGeorge | "try this: hit the 'power' button" |
| 06:07.23 | *** join/#asterisk ComaVN (n=blaargh@unaffiliated/comavn) |
| 06:07.26 | aptura | trying to get my ivr going in a way that I created a dial in to record my message then pass the info to the ivr. |
| 06:07.29 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:07.46 | osiris | put it in place till a REAL solution comes up |
| 06:08.05 | osiris | it doesnt have to be permantent. they arnt that much if i remember right |
| 06:08.17 | pfn | wtf is format 6... |
| 06:08.19 | osiris | mine might have been vendor specific |
| 06:10.18 | osiris | dumb question, but you ruled out things like temp file flooding and other system components, correct ? |
| 06:10.43 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 06:12.36 | *** join/#asterisk grEvenX (n=even@ti500720a080-9665.bb.online.no) |
| 06:13.55 | FuriousGeorge | osiris: is that to me? |
| 06:14.03 | osiris | yes |
| 06:14.04 | supjigatr | Any maxtnt folks here? |
| 06:14.21 | FuriousGeorge | havent looked into temp file flooding |
| 06:14.50 | FuriousGeorge | as to other system components, replaced mb and psu |
| 06:15.28 | FuriousGeorge | think im gonna round-robin swap tdm400p, but im stuck with my current modules |
| 06:16.19 | FuriousGeorge | was just about to ask if hooking up 4-pin molex to a tdm400p w/ only fxo was in any way beneficial |
| 06:17.04 | FuriousGeorge | osiris: and what temp files should i be looking at? |
| 06:17.58 | osiris | idk. im troubleshooting on more of a general system level. if you know it goes down around 100 days, take a look at the stats around the time it dies |
| 06:18.16 | osiris | a few days before |
| 06:18.48 | osiris | is the hd running out of space, is the ram gone. is a service forked |
| 06:20.18 | aptura | exten => s,n(restart) what does the restart part of this dialplan do |
| 06:20.18 | FuriousGeorge | i was just talking when i said 100 days. after i replaced mb and reinstalled OS it was up for only 100 hours before deadlock. i was joking that if i have asterisk restart daily this particular server has been shows to work for 100 days at a time |
| 06:20.18 | creativx | aptura: its just a name tag for that line |
| 06:20.18 | FuriousGeorge | but thats not a real solution |
| 06:20.25 | aptura | I see |
| 06:20.27 | FuriousGeorge | the smdc card on the other hand may make a good stopgap |
| 06:20.27 | creativx | you can use it with gotoif() to skip execution |
| 06:20.30 | creativx | its like a label |
| 06:20.41 | aptura | I dont need it |
| 06:20.49 | creativx | its very practical when doing logic |
| 06:20.57 | creativx | and you need to jump around in the dialplan |
| 06:21.00 | FuriousGeorge | aptura: no, but if you arent numbering your priorities you will eventuially have to |
| 06:21.11 | FuriousGeorge | what creativx said |
| 06:21.16 | aptura | I am giving it the n priority after 1 |
| 06:21.34 | creativx | eventually you will see that a label here and there is in place ;) |
| 06:21.43 | FuriousGeorge | aptura: so like he said, if you wanna use a goto, you will want a label |
| 06:22.03 | FuriousGeorge | since goto(s,n) wont help much |
| 06:23.13 | FuriousGeorge | JT: you dont think the smdc will go all willy nilly restarting my server when it isnt needed, do you? |
| 06:24.03 | aptura | well trying to get mine going again. |
| 06:24.08 | aptura | 'ivr that is. |
| 06:24.15 | JT | what is smdc? |
| 06:24.34 | aptura | also need to head off to bed. |
| 06:24.43 | tengulre | [Apr 17 14:19:21] ERROR[7615]: res_jabber.c:480 aji_act_hook: gnuTLS not installed. |
| 06:25.12 | tengulre | but. ls /usr/lib/libgnutls.* |
| 06:25.33 | tengulre | have /usr/lib/libgnutls.a /usr/lib/libgnutls.la /usr/lib/libgnutls.so /usr/lib/libgnutls.so.13 /usr/lib/libgnutls.so.13.7.0 |
| 06:25.44 | tengulre | anybody know why? |
| 06:25.46 | FuriousGeorge | isystem management daughter card |
| 06:26.06 | FuriousGeorge | JT: which i think, having never used one, is the same as what you called a watchdog card, which i agreed might work well till i find out WTF is going on |
| 06:26.58 | aptura | are all Background audio files stored in /var/lib/asterisk/sounds or /tmp |
| 06:27.11 | aptura | ones that i record of course. |
| 06:27.40 | aptura | I need to go |
| 06:27.40 | FuriousGeorge | aptura: the former |
| 06:27.50 | aptura | fomer what |
| 06:27.50 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 06:28.15 | FuriousGeorge | <PROTECTED> |
| 06:28.24 | aptura | so all files I record goto /var/lib/asterisk/sounds |
| 06:28.32 | aptura | latter ? |
| 06:28.34 | FuriousGeorge | oh, you mean using the dialplan record app |
| 06:28.37 | FuriousGeorge | heh |
| 06:28.38 | aptura | yes |
| 06:28.44 | aptura | I need to create my ivr |
| 06:29.20 | aptura | i cannot stay away night :) |
| 06:29.37 | FuriousGeorge | i think you tell it where to put the file with the first option |
| 06:29.53 | FuriousGeorge | not sure if you can specify a dir with filename, or what the default is |
| 06:30.42 | creativx | default goes to tmp |
| 06:30.54 | creativx | havent tried making it write to another dir, but why shouldnt it work :-) |
| 06:30.55 | *** join/#asterisk apardo (n=apardo@87.217.145.64) |
| 06:31.12 | FuriousGeorge | creativx: i ask myself that daily :) |
| 06:31.25 | FuriousGeorge | though maybe not for the same reasons always |
| 06:32.12 | creativx | hehe |
| 06:32.16 | FuriousGeorge | like the other day i drove my car through a puddle. everyone else was doing it. why shouldnt it work? got some water in my air intake, water doest compress (or combust), broke my engine. might be totalled |
| 06:32.27 | creativx | well |
| 06:32.34 | creativx | you ought to know where your intake is located before doing that |
| 06:32.40 | FuriousGeorge | i knew where it was |
| 06:32.49 | FuriousGeorge | and the water didnt go up that high |
| 06:33.01 | creativx | bad luck then |
| 06:33.02 | creativx | or bad car? |
| 06:33.04 | FuriousGeorge | until the wave from the dude in front of me sped up |
| 06:33.31 | FuriousGeorge | hit my bumper, splashed against my grille, the end |
| 06:33.35 | creativx | hehe |
| 06:33.47 | creativx | that sucks indeed |
| 06:34.09 | creativx | perhaps you bent a conrod on the compression stroke then |
| 06:34.24 | FuriousGeorge | thats what were thinking |
| 06:34.31 | FuriousGeorge | appraiser will be out there tomorrow |
| 06:34.53 | FuriousGeorge | hope its not totalled |
| 06:35.36 | creativx | what car is it then |
| 06:36.09 | FuriousGeorge | and as im sitting there, waiting for my buddy to get to me with galoshes so we can push out before the water gets up to my door im thinking "this could only get worse if that damn asterisk server deadlocks" :) |
| 06:36.15 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 06:36.18 | FuriousGeorge | a6 2002 2.7t |
| 06:36.51 | FuriousGeorge | hey oej |
| 06:37.06 | oej | morning, FuriousGeorge |
| 06:37.13 | oej | Greetings from SIPit in Antwerp |
| 06:37.20 | creativx | hehe, not the cheapest one to destroy either |
| 06:37.49 | FuriousGeorge | creativx: yeah. my insurance covered it. you think its a total if the camrod(s) is(are) bent |
| 06:38.00 | FuriousGeorge | s/covered/will cover |
| 06:38.43 | creativx | nah a conrod can bend without any other damage |
| 06:38.50 | creativx | it just depends where the excess energy went |
| 06:38.56 | creativx | if the conrod absorbed it all by bending it |
| 06:39.19 | creativx | but not that easy to say without splitting the engine and seeing |
| 06:39.44 | creativx | if you were really lucky the water went out of there on the exhaust stroke and into the turbine housing |
| 06:40.17 | JT | turbine? |
| 06:40.22 | JT | what sort of engine does he have? |
| 06:40.25 | pfn | damnit, why is my call that is incoming on a SIP channel and outgoing on IAX2 negotiating as g729 for the SIP leg? |
| 06:40.39 | FuriousGeorge | the guy said it makes a noise when it turns over coming from the pan |
| 06:40.44 | FuriousGeorge | said maybe bent a rod |
| 06:40.44 | pfn | asterisk keeps dropping the call because it doesn't support g729... even though I disallow=g729 in sip.conf |
| 06:40.55 | JT | FuriousGeorge: is it diesel? |
| 06:40.56 | creativx | JT: 2.7t where t stands for hair blower :) |
| 06:40.57 | FuriousGeorge | pfn: thats proprietary |
| 06:41.12 | FuriousGeorge | 2.7t where t stands for turbo, actually :) |
| 06:41.13 | creativx | 2.5 is tdi |
| 06:41.21 | creativx | 2.7 is gasoline |
| 06:41.24 | JT | what about 2.7 |
| 06:41.29 | creativx | an aint it biturbo |
| 06:41.32 | creativx | or was that only the s models |
| 06:41.34 | FuriousGeorge | it is |
| 06:41.38 | JT | FuriousGeorge: so it's not diesel? |
| 06:41.40 | pfn | FuriousGeorge, duh, how do I make it not support g729 in the SIP channel |
| 06:41.42 | FuriousGeorge | no |
| 06:41.45 | FuriousGeorge | not diesel |
| 06:41.45 | JT | ok |
| 06:41.46 | creativx | i only use audi s-parts for my vw engine.. |
| 06:41.49 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:41.57 | FuriousGeorge | nor does it run on flood water apparently |
| 06:42.01 | JT | diesel engines are much more likely to bend conrods with water |
| 06:42.05 | pfn | the audi s-models don't have turbo engines anymore... |
| 06:42.16 | creativx | i didnt specify what year |
| 06:42.17 | creativx | hehe |
| 06:42.20 | FuriousGeorge | no new audi's do afaik |
| 06:42.31 | FuriousGeorge | maybe the rs models |
| 06:42.35 | creativx | my engine is mainly s2/4/6 parts |
| 06:42.56 | osiris | so, is anyone here proficient in polycom phones ? |
| 06:42.59 | FuriousGeorge | JT: so what are gas engines more likely to do? |
| 06:43.12 | JT | FuriousGeorge: haven't heard of it happening, but it's possible |
| 06:43.35 | osiris | im trying to get together a polycom channel if anyone wants to help |
| 06:43.46 | JT | water could make the turbo angry and stress fracture pieces of compressor blade and therefore put metal into the cylinder |
| 06:44.27 | FuriousGeorge | sounds bad |
| 06:44.33 | FuriousGeorge | perhaps worse |
| 06:45.25 | JT | either way it's lucky you have insurance |
| 06:45.40 | *** join/#asterisk mkl1525 (n=mkl1525@pD95335B8.dip0.t-ipconnect.de) |
| 06:46.03 | FuriousGeorge | i even got insurance on my insurance |
| 06:46.09 | FuriousGeorge | (gap insurance) |
| 06:46.34 | JT | heh |
| 06:46.37 | FuriousGeorge | i just dont want it to be totalled. then i gotta come up with another d.p. |
| 06:46.47 | FuriousGeorge | and no one wants that |
| 06:46.48 | JT | dp? |
| 06:46.53 | FuriousGeorge | down payment |
| 06:47.00 | JT | ah ok |
| 06:47.01 | FuriousGeorge | im to lazy to be clear |
| 06:47.13 | FuriousGeorge | ~s/too/to |
| 06:47.26 | JT | well if the engine is stuffed, i don't think that is enough to write off most cars |
| 06:47.41 | FuriousGeorge | my blue book value is about 15k |
| 06:47.43 | JT | body repairs are what causes a car to get written off quick smart |
| 06:48.01 | pfn | replacing an engine is only $10kish |
| 06:48.45 | FuriousGeorge | perhaps if it is totalled i can take the computer out of the car and see if it runs asterisk better than my current server at that location |
| 06:49.02 | *** join/#asterisk ams1701 (n=ams1701@202.189.249.206) |
| 06:49.05 | FuriousGeorge | plus itll keep the mileage low |
| 06:49.30 | JT | FuriousGeorge: did the water enter the passenger area? |
| 06:49.42 | CunningPike | osiris: Sounds interesting - where would such a channel live? |
| 06:49.49 | FuriousGeorge | JT: no or it would be totally hosed, thank god |
| 06:49.50 | osiris | <PROTECTED> |
| 06:49.55 | osiris | here |
| 06:49.59 | CunningPike | osiris: On which network? |
| 06:49.59 | osiris | freenode |
| 06:50.12 | JT | FuriousGeorge: freshwater or saltwater? |
| 06:50.22 | CunningPike | osiris: The freenode guardians may have thoughts about that - Polycom is a commerical produt |
| 06:50.29 | CunningPike | s/produt/product/ |
| 06:50.36 | FuriousGeorge | fresh as any water on the streets of newark, nj can be |
| 06:50.50 | osiris | well, that as the case may be. |
| 06:50.55 | mkl1525 | Hi, (* 1.4) I'm trying to setup some kind of call forwarding: first normal number is called 444 if nobody picked up for 10 sec send it to a queue of phones for 10 seconds if even then nobody is answering send it to the users cell phone. this works but if the cell phone is called using isdn capi line I get the ringing on the cell phone but after short time the call is aborted with "nobody picked up in 10000ms" but afaik there's no explic |
| 06:50.56 | mkl1525 | it timeout set - any hints where this timeout could come from? |
| 06:51.09 | FuriousGeorge | didnt see much fresh sewage |
| 06:51.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:51.24 | CunningPike | osiris: Speak to [TK-Fender] - he's the resident Polycom guru |
| 06:51.47 | osiris | roger that |
| 06:52.16 | FuriousGeorge | windows is proprietary too |
| 06:52.25 | FuriousGeorge | no one busts that channel up |
| 06:52.49 | JT | to be technically correct it should be ##polycom not #polycom |
| 06:52.56 | JT | but i don't care too much for that policy |
| 06:53.18 | FuriousGeorge | policy schmolicy |
| 06:53.20 | osiris | i know it should be ## |
| 06:53.28 | osiris | but until someone claims it |
| 06:53.34 | osiris | =) |
| 06:53.47 | osiris | ill gladly step down if asked |
| 06:53.58 | JT | heh |
| 06:54.04 | FuriousGeorge | after a prolonged and bloody fight, of course |
| 06:54.15 | osiris | absolutley |
| 06:54.24 | FuriousGeorge | mostly scratching and biting |
| 06:54.41 | osiris | i refer to it as cussin and spittin |
| 06:54.55 | FuriousGeorge | heh |
| 06:54.59 | pfn | hmm, does Set(SIP_CODEC=ulaw) do anything? |
| 06:55.10 | pfn | my call is still being passed to iax2 as g729... |
| 06:55.34 | osiris | tell yer provider to give you g711 |
| 06:56.32 | pfn | well, I am my provider... heh |
| 06:56.41 | pfn | I guess I should login to the as5400 and tell it to use g711u instead |
| 06:56.55 | osiris | yes |
| 06:57.10 | pfn | 'cept I don't know the commands, heh |
| 06:57.30 | osiris | are you sure it will dole out g711 if asked ? |
| 06:57.40 | pfn | yes |
| 06:57.53 | osiris | by other devices i mean |
| 06:58.15 | pfn | huh? |
| 06:59.04 | osiris | idk. do you have other boxes/trunks os that same as5600 that teminate g711 |
| 06:59.25 | osiris | os=to |
| 06:59.32 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 06:59.43 | FuriousGeorge | brb |
| 07:01.07 | pfn | nevermind, I messed up, I had g729 enabled in my iax.conf (from an old config) |
| 07:04.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:05.40 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 07:06.40 | lesouvage | Can somebody plesae tells me what is wrong with this line: Set(GESPREKS_KOSTEN= $[ ${AANTAL+SEC} * ${SEC_PRIJS} ]) I got an error message ast_yyerror(): syntax error: syntax error, unexpected '*', expecting $end; Input: |
| 07:06.42 | lesouvage | <PROTECTED> |
| 07:06.43 | lesouvage | <PROTECTED> |
| 07:10.28 | pfn | ${AANTAL+SEC} ? |
| 07:10.51 | JT | the syntax there certainly doesn't look like it should work |
| 07:12.32 | tzafrir | lesouvage, is AANTAL+SEC a name of a variable? Did you mean ${AANTAL}+${SEC} ? |
| 07:13.21 | lesouvage | tzafrir: you are right, just a stupid typo. Thanks for the help. |
| 07:14.10 | creativx | gespreks |
| 07:14.27 | creativx | that sounds dutch |
| 07:15.01 | tzafrir | --> lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) has joined #asterisk |
| 07:15.45 | creativx | i was at the kitchen ;) |
| 07:16.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:16.42 | creativx | but thank you for pointing that out tzafrir |
| 07:17.01 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:18.34 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:18.57 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 07:20.10 | *** join/#asterisk [BAF64] (i=ferrisr@allegro/botmaster/baf) |
| 07:23.02 | *** join/#asterisk mkl1525 (n=mkl1525@pd953076a.dip0.t-ipconnect.de) |
| 07:23.10 | FuriousGeorge | tzafrir: does that mean "the eggs", literally |
| 07:23.41 | FuriousGeorge | ? |
| 07:24.23 | tzafrir | FuriousGeorge, why are you looking at me? |
| 07:24.49 | FuriousGeorge | u said it sounds dutch |
| 07:25.18 | FuriousGeorge | les oefs (i think) means eggs in french |
| 07:25.18 | creativx | that was me actually |
| 07:25.23 | FuriousGeorge | ovage ~ovum |
| 07:25.25 | FuriousGeorge | sorry |
| 07:25.40 | creativx | the only dutch I know is limited to hoe gaat het met jou mijn goede vriend |
| 07:26.19 | FuriousGeorge | i know dankya vel |
| 07:26.23 | FuriousGeorge | or something |
| 07:26.29 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 07:26.52 | FuriousGeorge | creativx: i wanna get me some rs6 turbos for my car (assuming it lives) |
| 07:28.10 | creativx | wouldnt that mean new manifolds and a lot more |
| 07:28.11 | creativx | hehe |
| 07:28.57 | FuriousGeorge | creativx: im no expert but i think you can just drop'em in |
| 07:29.14 | creativx | might be.. im not that familiar with newer audi engines |
| 07:30.13 | FuriousGeorge | what engine did your vw's parts come from. i assumed it was the same as mine more or less |
| 07:31.02 | creativx | ive got a 4cyl 16v vw engine |
| 07:31.07 | creativx | its the same as a 2.2t |
| 07:31.09 | creativx | except 1 less cyl |
| 07:31.23 | FuriousGeorge | i c now |
| 07:31.25 | creativx | so im using the s6 exhaust manifold, k24 turbo, s2 intake |
| 07:31.31 | creativx | chopped and welded |
| 07:31.39 | creativx | h-profile conrods and s2 pistons |
| 07:31.51 | FuriousGeorge | hey, thats all assembler to me |
| 07:31.54 | creativx | hehe |
| 07:31.55 | creativx | i see |
| 07:32.10 | creativx | well its gonna be a fun little engine |
| 07:32.44 | FuriousGeorge | i know a lot of gti have 2.7 which are parts compatible with my car and the s/rs 4/6 |
| 07:32.56 | creativx | yeah thats VAG for ya |
| 07:33.00 | creativx | things are sorta interchangeable |
| 07:33.21 | creativx | im using a 60-2 crank triggerwheel from an 2002 Audi A2 |
| 07:33.37 | creativx | magically it bolts right onto my 1989 KR block |
| 07:33.43 | FuriousGeorge | heh |
| 07:34.01 | FuriousGeorge | engines make this voip stuff seem easy |
| 07:34.16 | JT | lies |
| 07:34.38 | FuriousGeorge | no one ever said perception was reality |
| 07:35.13 | creativx | engines are easy until you start trying to control them |
| 07:35.19 | creativx | with wasted spark ignition and all that |
| 07:35.43 | creativx | and fuel/ignition maps.. thats when it gets up around the voip level |
| 07:36.09 | JT | coilpack per spark plug is a winner ;) |
| 07:37.31 | creativx | that requires more than one trigger signal.. and well yeah |
| 07:37.41 | creativx | i'll be satisfied with wasted spark :-) |
| 07:37.59 | creativx | the advantages with fully sequential vs wasted spark can be argued |
| 07:38.05 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 07:38.26 | JT | no spark leads is pretty cool |
| 07:40.13 | creativx | hehe |
| 07:40.13 | FuriousGeorge | i just thought of something ironic. when i get water in my engine, that's called a hydrolock, right? |
| 07:40.13 | creativx | that could be any fluid FuriousGeorge.. like fuel |
| 07:40.13 | FuriousGeorge | ok |
| 07:40.13 | creativx | but yeah the idea is correct |
| 07:40.13 | FuriousGeorge | and im in here because asterisk deadlocks |
| 07:40.16 | FuriousGeorge | coincidence |
| 07:40.17 | creativx | liquid mass that causes the engine to lock up |
| 07:40.17 | creativx | hehehe |
| 07:40.26 | FuriousGeorge | you think i got water in my server |
| 07:40.29 | creativx | perhaps if you try water in the server |
| 07:40.33 | FuriousGeorge | lol |
| 07:40.39 | creativx | it might reverse things |
| 07:40.41 | creativx | who knows :-) |
| 07:40.49 | FuriousGeorge | one of us must be right, using this logic |
| 07:41.08 | JT | FuriousGeorge: it's only a hydrolock if your engine ceases rotation due to fluid(s) within one or more cylinders :) |
| 07:41.25 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) [NETSPLIT VICTIM] |
| 07:41.26 | *** join/#asterisk Rutro (n=Rutro187@wsip-68-96-31-95.om.om.cox.net) [NETSPLIT VICTIM] |
| 07:41.44 | FuriousGeorge | JT: so youre saying i had a fan failure in my server |
| 07:42.15 | creativx | with a hydrolock it is possible to drain the engine to make it work again |
| 07:42.18 | creativx | if you fry a chip |
| 07:42.29 | creativx | well.. lost bits?? |
| 07:43.20 | FuriousGeorge | so is there any theoretical advantage to having your engine rebuilt? |
| 07:43.47 | FuriousGeorge | assuming the issue that caused it to need it to be rebuilt is 100% addressed in the process |
| 07:44.10 | creativx | depends on the engine wear and tear |
| 07:44.22 | FuriousGeorge | creativx: in the rebuilding process? |
| 07:44.40 | creativx | i think i misunderstood |
| 07:45.05 | creativx | but new gaskets etc is always good |
| 07:45.10 | creativx | prolongs the engine life |
| 07:45.46 | JT | rebuilt often involves boring of the cylinders |
| 07:45.54 | JT | maybe surfacing of head |
| 07:45.59 | JT | if there is significant wear |
| 07:46.03 | FuriousGeorge | im asking if my rebuilt engine may be better (in terms if a reliability, not performance) than it would have been had it never broke and needed rebuilding |
| 07:46.05 | creativx | not boring.. honing |
| 07:46.15 | FuriousGeorge | JT: that is a bad thing? |
| 07:46.17 | creativx | it depends on how bad it was before FuriousGeorge :) |
| 07:46.31 | creativx | e.g uneven compression, warped heads |
| 07:46.37 | JT | FuriousGeorge: depends if the rebuild is done well |
| 07:46.46 | JT | creativx: yes boring. |
| 07:47.18 | creativx | that would mean oval cyls or great wear |
| 07:47.25 | creativx | and new overdimensioned pistons |
| 07:47.29 | JT | yes |
| 07:47.37 | creativx | well it all depends |
| 07:47.46 | JT | "significant wear" |
| 07:48.09 | FuriousGeorge | JT: creativx: i guess im asking if they would get a chance to do any maintenance they otherwise would have been unable to do, given the chance to get in there and rebuild the thing |
| 07:48.21 | FuriousGeorge | it sounds like you guys are saying its a definite maybe |
| 07:48.48 | JT | FuriousGeorge: sure, they could clean it and replace the seals at a minimum |
| 07:49.05 | creativx | change anything that is accessible with the engine apart |
| 07:49.10 | FuriousGeorge | damn straight, my insurance company is paying for it |
| 07:49.17 | JT | and replace bearings |
| 07:49.22 | creativx | you might get a replacement engine |
| 07:49.30 | FuriousGeorge | and that troublesome flux capacitor |
| 07:49.37 | creativx | and coil supercharger |
| 07:49.49 | FuriousGeorge | which from time to time has prevented me from going back to the future |
| 07:50.16 | JT | what would be worse would be if it prevented you from coming back to the present |
| 07:50.41 | FuriousGeorge | that would require at the very minimum one sequel |
| 07:52.41 | FuriousGeorge | ok guys so take a stand: now that the water is drained, it makes a "clank-ety-clank" noise coming from "the pan" when they turn it over. insurance is going out later today |
| 07:52.45 | FuriousGeorge | is it totalled? |
| 07:52.57 | FuriousGeorge | car value ~15.5K |
| 07:54.46 | creativx | damn.. 15k |
| 07:55.02 | FuriousGeorge | i wont hold you to it, im just taking a poll |
| 07:55.10 | creativx | that less than 50% of what its value is here |
| 07:55.20 | creativx | well the clank noise sounds like new conrod and new bearings |
| 07:55.26 | FuriousGeorge | where are you? |
| 07:55.28 | pfn | FuriousGeorge, 15k with a 2.7t? totalled |
| 07:55.44 | FuriousGeorge | pfn: 16 if its mint |
| 07:55.48 | creativx | if the piston has touched the cylinder wall.. perhaps not worth it |
| 07:55.53 | creativx | FuriousGeorge: im in norway |
| 07:55.56 | FuriousGeorge | mine is just very good condition |
| 07:56.49 | FuriousGeorge | ok, so another definite maybe :) |
| 07:58.58 | pfn | FuriousGeorge, no other water damage? |
| 07:59.04 | pfn | how'd you manage to hydrolock? |
| 07:59.41 | FuriousGeorge | pfn: a big puddle wasnt up to my air intake till the wake from the car entering in front of me |
| 07:59.50 | FuriousGeorge | and by car i mean truck |
| 08:00.03 | pfn | FuriousGeorge, weird, you run some sorta funky intake or something? |
| 08:00.09 | FuriousGeorge | not me |
| 08:00.19 | FuriousGeorge | it just splashed up on my grille |
| 08:00.54 | FuriousGeorge | and my engine must have gotten a gulp |
| 08:01.09 | FuriousGeorge | all the other cars made it |
| 08:01.13 | *** join/#asterisk oej_ (n=olle@cust224-125.dsl.versadsl.be) |
| 08:01.16 | FuriousGeorge | cept for one other dude |
| 08:01.40 | FuriousGeorge | he got rescued by fire compnay |
| 08:02.11 | FuriousGeorge | "rescued" from 30 cm of water |
| 08:02.36 | FuriousGeorge | i pushed my car out |
| 08:10.29 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:16.16 | *** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net) |
| 08:17.15 | FuriousGeorge | JT: you havent personally used a watchdog card with *, have you? |
| 08:22.12 | JT | no |
| 08:24.03 | nemski | hey guys |
| 08:24.18 | nemski | got a phone that's behind an ADSL router |
| 08:24.45 | nemski | if I turn NAT on in the config, then it registers but no voice |
| 08:24.57 | nemski | if I turn NAT off and try port forwarding it doesn't register |
| 08:25.04 | *** join/#asterisk smurfix (n=smurf@debian/developer/smurf) |
| 08:25.58 | nemski | it appears that it's using the internal address not the internet address when it's talking to the asterisk server |
| 08:29.05 | *** join/#asterisk pressureman (n=pressure@210.48.105.162) |
| 08:30.42 | *** join/#asterisk oej (n=olle@cust224-125.dsl.versadsl.be) |
| 08:30.42 | pressureman | i've got a problem with asterisk not providing ringback tone to the caller when they get blind transferred to another extension |
| 08:30.54 | pressureman | it works with basically an identical config in openpbx - sip debug shows a 180 Ringing the first time around, and after the transfer, opbx generates the ringback inband. no matter what i do, i can't get asterisk to generate ringback tone in band |
| 08:31.20 | pressureman | ...unless i resort to fake ringing with 'r' in the Dial(), which is really ugly, because the remote end might be busy, not ringing |
| 08:31.53 | jeffgus | is it pretty common for carriers to only support 56kbit channels and not clear64 on a PRI??? |
| 08:32.00 | jeffgus | i've talked to 2 so far |
| 08:32.05 | jeffgus | XO Communications |
| 08:32.19 | jeffgus | and Telepacific and they say they only do 56kbit PRI channels |
| 08:32.54 | jeffgus | it seems to me that true PRI should always be 64kbit |
| 08:38.01 | kumbalae | hi |
| 08:38.30 | kumbalae | what is the maximum number of PRI Zap conference channels allowed in a pentinum computer ? |
| 08:39.22 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:41.37 | Supaplex | 2! |
| 08:41.54 | tzafrir | "allowed" is not the right word. You're not allowed to use it ;-) |
| 08:42.41 | tzafrir | It also depends which "pentium" exactly |
| 08:43.01 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 08:47.14 | Zeeek | ??? |
| 08:48.21 | Supaplex | Zeeek: no matches for ???. Please restate your query. |
| 08:49.04 | Zeeek | m.??day |
| 08:49.42 | Zeeek | i should be working |
| 08:54.44 | *** join/#asterisk felipex (n=dsfdsf@88-149-172-71.f5.ngi.it) |
| 08:55.35 | felipex | hi at all |
| 08:57.37 | felipex | i have to check if the time is between tue 9:30 - 13:00 or thu 14:30 17:30 |
| 08:57.55 | felipex | if yes go to exten if no another exten |
| 08:58.02 | felipex | can you help me? |
| 08:59.58 | *** join/#asterisk hermuli (n=Eladamri@cs185062.pp.htv.fi) |
| 09:00.50 | *** part/#asterisk pressureman (n=pressure@210.48.105.162) |
| 09:01.27 | mosty | felipex, use GotoIfTime |
| 09:04.36 | felipex | mosty ok but how can i check the 2 time ? |
| 09:05.29 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:07.08 | mosty | lookat the docs on the wiki |
| 09:15.45 | felipex | mosty thanks very much |
| 09:16.47 | mosty | no problem |
| 09:18.32 | mosty | how long i just registered some g729 licences and restarted asterisk, now it;s running at 99% cpu usage, and there are no calls. is this normal? |
| 09:18.52 | mosty | er, s/^how long// |
| 09:27.02 | e-ddie | not really |
| 09:28.55 | Ahrimanes | mosty, sounds.. well wrong |
| 09:28.57 | Ahrimanes | hey e-ddie |
| 09:29.13 | e-ddie | hi Ahrimanes |
| 09:31.27 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 09:37.25 | kumbalae | mosty: just contact digium for this problem, they will give you updated module |
| 09:42.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:43.37 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 09:57.13 | JT | jeffgus: yeah 56kbit/s is very old school |
| 09:57.53 | jeffgus | JT, why is it so common still? |
| 09:58.33 | jeffgus | the XO guy said that it would be expensive to upgrade the switches... duh... of course |
| 09:58.56 | jeffgus | but it's not true ISDN unless the switches due 64kbit |
| 10:01.26 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 10:01.26 | zeeesh | hi |
| 10:01.52 | mosty | kumbalae, i've been emailing digium every day for a week. no response at all |
| 10:03.21 | tzafrir | what consumes the CPU? (look at top) |
| 10:03.41 | pfn | heh, boo, this hacked g729 I was toying with doesn't work... |
| 10:03.46 | *** join/#asterisk Ast001 (n=uros@77-105-51-136.adsl-1.sezampro.yu) |
| 10:03.48 | mosty | tzafrir, "asterisk" |
| 10:03.49 | Ast001 | hello |
| 10:03.59 | Ast001 | any good nagios plugin for checking sip on asterisk ? |
| 10:04.18 | tzafrir | mosty, hmm... next thing: look which thread consumes CPU |
| 10:04.40 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 10:04.42 | tzafrir | Not sure which switches to pass to ps to get threads information, though |
| 10:07.36 | zeeesh | I hv 2 sip peer (peer100 and peer200) at asterisk server "A" which can communicate with each other by using this extensions " exten => 101,1,Dial(SIP/101) ,,, exten => 100,1,Dial(SIP/100) … same as I hv 2 sip peers (peer 200 and peer201) at asterisk server "B" .. how can server "A" peers will communicate with server "B" …. ????? |
| 10:08.37 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 10:09.30 | JT | jeffgus: i don't think it's common at all |
| 10:09.34 | JT | jeffgus: where on earth are you? |
| 10:11.30 | Ast001 | extension/context/ipaddr or something like that |
| 10:12.18 | jeffgus | JT, welp, XO Communications isn't small and they told me they can't do switched PRI data |
| 10:12.26 | jeffgus | only voice (56kbit) |
| 10:12.34 | JT | never heard of XO communications |
| 10:13.14 | pfn | XO ... used to be covad |
| 10:13.23 | jeffgus | http://en.wikipedia.org/wiki/XO_Communications |
| 10:14.06 | jeffgus | covad is still covad isn't it? |
| 10:14.30 | pfn | hmm, I must be confused |
| 10:14.35 | pfn | erm, or was it concentric |
| 10:14.37 | pfn | xo = concentric |
| 10:14.57 | JT | jeffgus: where are you? |
| 10:15.07 | pfn | ah, it's concentric, not covad, it was a "C-name" |
| 10:15.11 | jeffgus | i also talked to a smaller company called Telepacific (based in California) and they said their switches don't do 64kbit either |
| 10:15.25 | jeffgus | JT, Los Angeles, CA, US |
| 10:16.08 | jeffgus | the the AT&T people are slow to get back to me, but i would think they would do 64kbit since they provide BRI service |
| 10:16.16 | JT | yeah |
| 10:16.33 | JT | you should speak with Strom_C when he's next around |
| 10:18.37 | *** join/#asterisk Elfe (n=elfe@td9091a48.pool.terralink.de) |
| 10:21.46 | jeffgus | JT, ok... i'll keep an eye out |
| 10:23.18 | JT | he is in that area, and knows a LOT about telcomms :) |
| 10:24.38 | *** join/#asterisk hypn0tek (n=eleve@linagoraberri.pck.nerim.net) |
| 10:24.53 | hypn0tek | hello |
| 10:26.03 | Elfe | hi, does asterisk have an option to send an udp packet to prevent nat timeouts? (server side ttl) |
| 10:26.33 | pfn | qualify |
| 10:26.51 | hypn0tek | 2 phones installed : an xlite and a Thomson phone, when calling the Thomson phones from xlite it works |
| 10:27.25 | Elfe | thanks |
| 10:27.46 | hypn0tek | but when trying to call from the Thomson phone the microphone of Xlite seems to be working, but we can't hear nothing from the Thomson |
| 10:31.33 | *** join/#asterisk Inez (n=faceoff@devel4.net) |
| 10:31.36 | Inez | hi |
| 10:32.39 | *** join/#asterisk robin_sz (n=robin@212.243.40.130) |
| 10:32.44 | robin_sz | morning girls |
| 10:32.46 | robin_sz | so ... |
| 10:33.25 | robin_sz | imagine I want some sort of web page thing, to allow clients to listen to an IAX channel |
| 10:33.39 | robin_sz | "click here to call" or whatever .. right? |
| 10:34.27 | robin_sz | I already bought Balbir wossisnames active-X thing |
| 10:34.39 | robin_sz | which is fine, but now I need somehting that works in firefox |
| 10:34.48 | robin_sz | clues? |
| 10:34.51 | robin_sz | Cloos? |
| 10:34.57 | mosty | ahh, if you have multiple g729 codec files in the module dir, asterisk goes crazy |
| 10:35.46 | robin_sz | ideally, a GPL bit of code, especially if its the sort of GPL code where you get the source |
| 10:36.24 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 10:36.25 | pfn | so why hasn't asterisk switched to using UUID for its cdr uniqueids yet? |
| 10:37.52 | *** part/#asterisk Elfe (n=elfe@td9091a48.pool.terralink.de) |
| 10:38.00 | robin_sz | so no ideas then |
| 10:52.07 | Ast001 | any nagios-plugin for monitoring asterisk ? |
| 10:52.47 | tzafrir | Ast001, I saw a thread about it in asterisk-users |
| 10:53.03 | tzafrir | I don't remember if there was an actual link to such a plugin there |
| 10:53.47 | tzafrir | robin_sz, moziax? (a firefox extension) |
| 10:54.33 | robin_sz | tzafrir, coo, really? |
| 10:55.12 | tzafrir | http://moziax.mozdev.org/ . Haven't tested it |
| 10:55.50 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:55.55 | skirmisha | hello guys |
| 10:56.12 | skirmisha | can someone tell me is /etc/asterisk dir hardcoded in binary |
| 10:56.43 | nemski | I've forwarded the ports to my hard phone, I'm using NAT< yet I still can't get any voice |
| 10:56.50 | nemski | anyone got any ideas? |
| 10:57.13 | skirmisha | did u set ast to qualify endpoints |
| 10:57.19 | skirmisha | and also to use nat? |
| 10:57.45 | nemski | asterisk is set to use nat |
| 10:57.52 | nemski | unsure about qualify |
| 10:58.19 | skirmisha | also what codec do u use |
| 10:58.19 | nemski | is qualify set on a per-user basis? |
| 10:58.20 | *** join/#asterisk oej (n=olle@cust224-125.dsl.versadsl.be) |
| 10:58.31 | skirmisha | yes qualify is on per user basis |
| 10:58.38 | nemski | gsg729a |
| 10:58.54 | nemski | k, I'll make sure it's set |
| 10:59.01 | nemski | I dont' have access to the asterisk box at the moment |
| 10:59.26 | nemski | *g729a |
| 10:59.32 | skirmisha | is call ok with g711 |
| 10:59.38 | skirmisha | have u tried g711 first |
| 10:59.52 | *** join/#asterisk friedrich| (n=friedric@e177253068.adsl.alicedsl.de) |
| 11:00.22 | robin_sz | tzafrir, seems to be the right thing, if a bit undocumented |
| 11:00.28 | nemski | yep, it doesn't work either |
| 11:00.49 | *** join/#asterisk oej_ (n=olle@cust224-125.dsl.versadsl.be) |
| 11:01.41 | skirmisha | first see if user is registered with ast |
| 11:02.15 | tzafrir | robin_sz, if you ask about it in asterisk-users, the author will probalbly answer... |
| 11:02.16 | nemski | it is |
| 11:02.53 | nemski | 1 111 REGISTERED 3595 349 itnexus.homelinux.org:5060 |
| 11:02.54 | nemski | 2 111 REGISTERED 3595 106 itnexus.homelinux.org:5060 |
| 11:03.52 | skirmisha | then what happens when u call |
| 11:04.02 | nemski | no sound |
| 11:04.30 | nemski | incoming or outgoin |
| 11:06.22 | Ast001 | thanks tzafrir |
| 11:06.30 | robin_sz | ok |
| 11:06.37 | robin_sz | thanks tzafrir |
| 11:06.52 | Ast001 | i found one on voipinfo org monitor asterisk nagios |
| 11:07.01 | Ast001 | but I strugle to configure it |
| 11:07.06 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:07.07 | robin_sz | tzafrir, is that not an irc channel then? |
| 11:07.48 | Ast001 | yuppi I found out check_sip!sip:101/agent is answer on my |
| 11:08.03 | Ast001 | concerns |
| 11:08.04 | nemski | oh wait |
| 11:08.10 | nemski | outgoing sound, no inbound sound |
| 11:08.11 | tzafrir | robin_sz, I just remember the author answering questions about it in the asterisk-users mailing list. this is not to say that he is or is not active elsewhere |
| 11:09.36 | robin_sz | ah, ok asterisk-users is a mailing list .. right |
| 11:11.38 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 11:12.24 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 11:12.28 | nemski | it appears that asterisk and addressing my phone via the internal address |
| 11:12.34 | nemski | not the internet address |
| 11:12.34 | Ast001 | but unfortunately it works only for default context |
| 11:12.37 | Ast001 | grrrrr |
| 11:16.38 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 11:17.07 | stoffell | is there any comparison as to what is the big difference between TE110P and TE120P ? |
| 11:17.34 | cpm | 20 |
| 11:17.59 | stoffell | cpm, yeah, at first glance that seems to be the only difference.. :p |
| 11:18.20 | cpm | except that it's really only 10 |
| 11:18.36 | hi365 | how can i let the end user controll playback speed or volume of a podcast? |
| 11:18.40 | stoffell | i was thinking more like .. technically.. |
| 11:20.26 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 11:20.41 | creativx | technically its the double of 5 |
| 11:25.30 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:26.37 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:26.41 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:26.49 | stoffell | well, seems you guys know as much bout it as me then.. nothing :p |
| 11:26.53 | *** join/#asterisk CASP3R (n=YourMum@c58-107-236-202.sunsh7.vic.optusnet.com.au) |
| 11:27.38 | CASP3R | hello |
| 11:27.56 | CASP3R | need some help with Asterisk server bethind NAT and remote phone behind NAT |
| 11:28.58 | CASP3R | the phone register with its Real IP but will ring |
| 11:29.09 | CASP3R | but voice traffic is sent to its LAN address |
| 11:29.33 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 11:30.17 | nemski | skirmisha: this is the guy who setup the asterisk box i"m trying to access |
| 11:38.26 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 11:40.03 | zeeesh | <PROTECTED> |
| 11:42.52 | *** join/#asterisk michael-i (n=michael-@141.41.40.191) |
| 11:47.35 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 11:48.54 | JT | stoffell: yeah it's a new chipset, supposedly meant to have better interrupt handling |
| 11:51.21 | stoffell | thanks JT, will order the TE120P then, the newer the better I hope ;) |
| 11:51.40 | JT | what are your requirements? |
| 11:52.15 | michael-i | Does anyone have Asterisk logging CDRs to syslog? Did I miss this config somewhere or do I have to hack around a bit. Google only found me a commercial product. |
| 11:52.35 | stoffell | small office, 1x E1, approx. 24 phones, a small (but busy in the winter) call center to take incoming orders |
| 11:52.39 | JT | i'm not sure why you'd want to do that, michael-i |
| 11:54.09 | michael-i | JT, I'm working on an embedded Asterisk solution and need to circularly record cdrs to avoid having my ram disk getting eaten up. My syslogd is already circular and this would solve my problem. |
| 11:54.21 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 11:54.33 | JT | hrm it can use sql too |
| 11:56.25 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 11:59.31 | *** join/#asterisk ltd (n=z@202-161-2-210.dyn.iinet.net.au) |
| 12:02.09 | DrukenLPY | people are sleeping... |
| 12:02.18 | MrChimpy | use clever bits of dialplan to achieve what I need to do... and CDRs become utterly screwed |
| 12:02.44 | MrChimpy | there's a promising "forkcdr" call. documentation for it is one line. |
| 12:03.01 | MrChimpy | that'll learn me. never ever use dialplan. |
| 12:03.43 | DrukenLPY | huh? |
| 12:04.43 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 12:04.52 | MrChimpy | i should've just used AGI from the start |
| 12:05.13 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 12:13.05 | defswork | my zap trunk it setup for 10 channels and incoming calls work ok - but I can only get one outgoing call |
| 12:13.17 | *** join/#asterisk pressureman (n=pressure@60-234-213-71.bitstream.orcon.net.nz) |
| 12:13.21 | defswork | anyone know what that could be down to ? |
| 12:13.28 | michael-i | JT, sorry I was gone, I'm aware of the sql stuff but that's too expensive an option |
| 12:14.02 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 12:14.08 | pressureman | has anyone run into the problem of no ringback tone being generated for the caller after a blind transfer? |
| 12:14.13 | tzanger | Good morning |
| 12:14.16 | pressureman | (sip channels) |
| 12:14.49 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:14.57 | JT | michael-i: expensive? |
| 12:15.00 | tzanger | does anyone know if there is a decent (i.e. no magic invocations of *8 or other crazy codes) to allow one IP501 to pick up a call that's ringing another IP501 when they're both in the same pickupgroup? |
| 12:15.19 | pressureman | ringback tone is generated ok if caller enters an IVR, which then does a Dial() |
| 12:15.34 | Zeeek | hey guys - is there a way to NOT mix the channels after an automon recording? |
| 12:15.37 | pressureman | but not of the caller is blind-transferred by somebody |
| 12:15.46 | michael-i | JT, memory usage |
| 12:17.56 | creativx | tzanger: how about pickup() ? |
| 12:18.21 | tzanger | creativx: and invoking that from an IP501 involves magic DTMF invocations... I was hoping I could map a hardkey or something to other than a speed dial |
| 12:21.57 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 12:22.55 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 12:23.40 | creativx | what is an ip501 then? |
| 12:23.42 | *** join/#asterisk JoanaDoe (n=chatzill@193.174.25.23) |
| 12:23.44 | creativx | since it requires magic |
| 12:24.00 | DrukenLPY | defswork: are you dialing the group? or zap/1/ ? |
| 12:24.09 | defswork | Dr-Linux: zap/1 |
| 12:24.15 | defswork | DrukenLPY: zap/1 |
| 12:24.23 | DrukenLPY | well, there's your problem :) |
| 12:24.28 | *** join/#asterisk hal2k (n=am@2002:5470:9fd9:0:0:0:0:1) |
| 12:24.30 | defswork | oh |
| 12:24.38 | JT | creativx: it's a phone |
| 12:25.05 | creativx | does this other ip501 have an extension |
| 12:25.16 | creativx | that you would be willing to hard code into a speed dial button |
| 12:25.38 | defswork | DrukenLPY: so change it to zap/g1 ? |
| 12:26.06 | DrukenLPY | defswork: generally that is a good idea... providing you have your zap interfaces in group=1 |
| 12:26.07 | tzanger | creativx: you don't understand my question. I know I can use *8 to pick up a call that is ringing a phone in my pickupgroup. I don't want to tell my users to remember all these damn codes. *8, *72, etc. The Polycom phones have hard and soft buttons, and they're also SIP phones, so in theory picking up calls in my pickupgroup should be in the SIP spec; I shouldn't need to fall back to in-band DTMF codes to communicate to *. |
| 12:26.39 | creativx | yeah I understand that |
| 12:26.40 | tzanger | creativx: I am asking if anyone knows if there is a way to get a Polycom hard or soft button to map to "pickup" without making the button a speed dial that dials *8. |
| 12:26.52 | DrukenLPY | tzanger: sounds to me like your looking for a sip implimentation of the pickup.... |
| 12:26.59 | creativx | Then I'm afraid I cant help |
| 12:26.59 | tzanger | *8, *72, etc are all hacks in the SIP world |
| 12:27.06 | creativx | I implemented the pickups in a different manner |
| 12:27.23 | tzanger | DrukenLPY: precisely. |
| 12:27.33 | tzanger | hinting doesn't work since that is just monitoring |
| 12:27.41 | tzanger | i.e. the hint can indicate ringing, but I can't do shit with it |
| 12:28.10 | defswork | DrukenLPY: it works now - thanks - I was advised to change it to zap/1 when setting it up :/ |
| 12:31.44 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 12:32.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:33.43 | DrukenLPY | you were advised wrong.... |
| 12:34.24 | DrukenLPY | tzanger: did you check the papers on sip? does it even support a call pickup? |
| 12:34.34 | Ast001 | check_sip!sip:101@aa.bb.cc.dd is it right for nagios plugin ? |
| 12:34.57 | tzanger | DrukenLPY: no not yet, I was going to ask tk when he comes in since he's the resident polycom expert :-) |
| 12:35.41 | DrukenLPY | ahh, well it sounds like it's more of a sip thing... no? |
| 12:35.42 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:36.03 | DrukenLPY | cause you could always just program the stupid *8 into a speed dial on the phone.... |
| 12:36.35 | DrukenLPY | my aastra has a sip park, but do you think asterisk can use it?? NO...... |
| 12:37.54 | tzanger | heh |
| 12:40.02 | DrukenLPY | quite annoying |
| 12:41.17 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-178-65.buckeyecom.net) |
| 12:43.58 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 12:45.09 | DrukenLPY | tzanger: http://bugs.digium.com/view.php?id=5014 |
| 12:46.33 | tzanger | DrukenLPY: wow, ok, gotta go through this :-) |
| 12:46.46 | DrukenLPY | tee hee :) |
| 12:49.21 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 12:49.24 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:49.36 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 12:50.06 | hi365 | how can i let the end user controll playback speed or volume of a podcast? |
| 12:50.09 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 12:50.26 | *** part/#asterisk pressureman (n=pressure@60-234-213-71.bitstream.orcon.net.nz) |
| 12:51.13 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:51.19 | msetim | Hi guys |
| 12:51.57 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 12:52.01 | Uatec | Greetings |
| 12:52.11 | msetim | I'd like to have a asterisk cli with highlight. How I can make it? |
| 12:52.24 | tzanger | http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime |
| 12:52.26 | tzanger | hahahaha |
| 12:52.28 | tzanger | love the Easter one |
| 12:52.53 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:53.21 | *** join/#asterisk ourkid (n=none@host-80-76-192-193.bytel.net.uk) |
| 12:54.37 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 12:56.02 | DrukenLPY | kinda sad that someone actually put the time into figuring all those dates out.... |
| 12:56.33 | anonymouz666 | chan_bluetooth is nice |
| 12:56.40 | anonymouz666 | but does not seem to work with motorola v3 |
| 12:56.53 | JT | it's deprecated |
| 12:57.00 | JT | people have moved to chan_cellphone |
| 12:57.26 | anonymouz666 | chan_cellphone works only on 1.4 I think |
| 12:57.32 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:58.24 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 12:58.53 | *** join/#asterisk Rick999 (n=rpulido_@72.243.170.17) |
| 12:58.54 | Uatec | 1.4? |
| 12:58.54 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 12:58.57 | DrukenLPY | am i correct in assuming chan_cellphone allows you to send calls to your cellphone (within bluetooth range) ?? |
| 12:58.59 | Uatec | What version is the business Edition? |
| 12:59.11 | JT | DrukenLPY: yes |
| 12:59.27 | anonymouz666 | JT: do you know the homepage? |
| 12:59.29 | anonymouz666 | chan_cellphone |
| 12:59.49 | Vec | My SIP phone support both G729 and alaw, I want calls made to my IAX trunk to be G729 (straight from the SIP phone) and my calls made to my ZAP channels to be alaw, is there a way to do this, I have tried to allow alaw and g729 in sip.conf but it always chooses the first match ? |
| 13:01.47 | msetim | Someone know how can I enabled cli asterisk highlight? |
| 13:02.15 | LeddyHM | where are people |
| 13:02.22 | slmnhq | Hi... would you folks recommend getting a Voip connection from your local Telco or a regular POTS line and using a digium card to connect with the Asterisk server? Are there any pro/cons to consider? |
| 13:02.26 | LeddyHM | where are people's personal greetings stored byu default? |
| 13:02.34 | Vec | msetim : if you run asterisk -cvvv it does the highlighting if u connect to it asterisk -r, it does'nt, not sure how to change that |
| 13:02.43 | Zeeek | slmnhq it depends on your situation |
| 13:03.06 | Vec | slmnhq : first thing is what country do you live in ? |
| 13:03.10 | slmnhq | USA |
| 13:03.19 | Zeeek | oej does MONITOR_OPTIONS work in 1.2.16 ? |
| 13:03.25 | JT | slmnhq: digital PRI is superior to POTS and voip |
| 13:03.46 | slmnhq | and lets say that my application is a little beyond Small Office / Home Office |
| 13:03.58 | Vec | Vec : How many lines r u going to need, (external lines) not extensions ? |
| 13:04.02 | JT | slmnhq: digital PRI for sure |
| 13:04.14 | JT | slmnhq: T1 or fractional T1 with PRI ISDN signalling would be optimal |
| 13:04.16 | msetim | Vec: Thanks. Exist some themes for it? |
| 13:04.19 | LeddyHM | nm, found it |
| 13:04.59 | Vec | msetim : not that I know of |
| 13:05.42 | Vec | JT : any idea how to do what I asked a little earlier ? |
| 13:06.17 | msetim | Vec: Thanks Vec |
| 13:06.18 | JT | Vec: no idea what you asked "a little earlier" :) |
| 13:06.38 | Vec | My SIP phone support both G729 and alaw, I want calls made to my IAX trunk to be G729 (straight from the SIP phone) and my calls made to my ZAP channels to be alaw, is there a way to do this, I have tried to allow alaw and g729 in sip.conf but it always chooses the first match ? |
| 13:06.43 | Vec | << that |
| 13:06.58 | anonymouz666 | I can't move to version 1.4 because I don't know if chan_unicall works on that version |
| 13:07.06 | JT | not sure, that sounds like a pain in the arse :) |
| 13:07.07 | *** join/#asterisk viperdude (n=jon@195.74.96.113) |
| 13:07.15 | JT | Vec: sip phone is on your lan? |
| 13:07.34 | Vec | SIP phone (Codec A) > Asterisk > ZAP -- AND -- SIP phone (codec B) > asterisk > IAXtrunk |
| 13:07.51 | *** join/#asterisk nasls_lsa (n=chatzill@85.75.130.107) |
| 13:07.57 | viperdude | hi guys how do I get asterisk to use more than 1 enum lookup domain? |
| 13:07.59 | JT | simple solution (i think) |
| 13:08.00 | Vec | JT : yeh, but doing it that way will not require my asterisk box to do any transcoding. |
| 13:08.21 | JT | Vec: allow only alaw on sip phone connection |
| 13:08.32 | JT | allow only g.729 on iax entry |
| 13:08.35 | DrukenLPY | tzanger: did that park thing help you at all ? |
| 13:08.39 | JT | transcoding is probably unavoidable |
| 13:09.05 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 13:09.07 | Ast001 | can someone tell me what is full sip uri |
| 13:09.28 | Vec | JT : it seems silly, thats what I am trying to avoid, then I can use a slow PC. |
| 13:09.31 | Ast001 | for extension 101 context agent at server a.b.c.d |
| 13:10.00 | Ast001 | sip:101@a.b.c.d or sip:101/agent@a.b.c.d |
| 13:10.16 | JT | Vec: i suspect you'd have to modify the source code |
| 13:10.56 | Vec | JT : I think it is something that the developers need to look into |
| 13:11.08 | JT | maybe |
| 13:11.13 | JT | not sure how big the use case is |
| 13:12.15 | *** part/#asterisk moranil (n=moranil@122.162.67.129) |
| 13:12.19 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 13:13.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:13.29 | *** join/#asterisk jeanmiii (n=besnard@mic92-2-82-67-216-218.fbx.proxad.net) |
| 13:13.34 | jeanmiii | hello |
| 13:15.12 | jeanmiii | I have set an account in sip.conf (something very basic) and whenever I try to register am seeing "SIP/2.0 401 Unauthorized |
| 13:16.00 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 13:16.04 | jeanmiii | at least I get this when I set no realm |
| 13:16.11 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 13:16.46 | jeanmiii | and when I set a realm, then I am getting wrong password (though I have check a hundred times that my username + password on the sip phone were the same as the one I set in sip.conf) |
| 13:16.54 | *** join/#asterisk Zdrulio (n=sux_@82.119.72.130) |
| 13:16.57 | Zdrulio | hello all |
| 13:19.25 | jeanmiii | I am actually getting this "SIP/2.0 401 Unauthorized" also when I intentionally set the wrong password in my sip phone |
| 13:19.43 | jeanmiii | so I guess the problem lies a layer below the proper username/password authentication |
| 13:21.34 | zeeesh | app_dial.c:1024 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 13:24.51 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 13:25.01 | Zefk | Hi all. What packages should I install on CentOS 4.4 in order to be able to compile cdr_odbc in asterisk 1.4.2? I just installed unixODBC and unixODBC-devel. |
| 13:26.17 | *** join/#asterisk _mike3_ (n=mike3@dhcp-0-13-10-78-a2-54.cpe.mountaincable.net) |
| 13:26.29 | _mike3_ | yawn |
| 13:27.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:27.44 | _mike3_ | Hey guys i'm looking for some cool addons for asterisk. (EG: alarm) |
| 13:27.59 | _mike3_ | Anyone got a goot site I can check out with addons? |
| 13:28.20 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:30.12 | *** join/#asterisk AgarGuest (n=agargues@CPE001839ee0745-CM00111ade4822.cpe.net.cable.rogers.com) |
| 13:30.17 | AgarGuest | hi |
| 13:30.22 | AgarGuest | anyone around?? |
| 13:33.59 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 13:34.48 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:36.28 | *** part/#asterisk Ast001 (n=uros@77-105-51-136.adsl-1.sezampro.yu) |
| 13:36.45 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:36.55 | blitzrage | patience is a virture |
| 13:37.10 | blitzrage | s/virture/virtue/ |
| 13:39.10 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:39.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:39.19 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:40.42 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 13:42.48 | *** join/#asterisk caw (n=caw@mail.compucorp.net) |
| 13:42.52 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 13:43.50 | *** join/#asterisk intralanman (n=intralan@pool-162-83-125-163.norf.east.verizon.net) |
| 13:44.13 | Vec | Where would one get all the international call costs, for billing in asterisk, my telco does not seem like they will provide them in a digital form, they send me a fax, and I am not sure if that fax lists everywhere ? |
| 13:44.49 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 13:47.35 | Uatec | Vec, if the telco doesn't give it to you, then you're stuffed |
| 13:47.50 | Uatec | i would expect that they are legally required to provide you with all that information in SOME format |
| 13:47.58 | Uatec | if they wont do it digitally then... bummer |
| 13:50.15 | *** join/#asterisk codefreeze (i=steve_mu@nat/digium/x-029f31510a20dc0e) |
| 13:50.28 | *** join/#asterisk thinwires (n=thinwire@ny-amherst-C4-1-bg2a-1-245.bflony.adelphia.net) |
| 13:50.40 | thinwires | morning everyone |
| 13:51.33 | Vec | Uatec : They obviously want u to buy if u want it digitally, because the fax thing is crazy |
| 13:51.42 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 13:52.21 | Uatec | they can't charge you for service information... |
| 13:52.32 | robin_sz | sure they can |
| 13:52.39 | Uatec | that's like going in to a restaurant, and you can look at what's on offer, but if you want to know how much things are you have to pay for the price list |
| 13:52.46 | robin_sz | no |
| 13:52.56 | robin_sz | its like havingto pay for looking at the recipie |
| 13:53.01 | thinwires | welcome to evil corpoation 101, charge for everything, charge twice to make sure they paid the first time |
| 13:53.23 | robin_sz | s/evil corporation/real world/ |
| 13:56.12 | Uatec | robin_sz, it's not a recipe |
| 13:56.19 | Uatec | it's nothing like that |
| 13:56.21 | Uatec | the price list... |
| 13:56.34 | Uatec | that's... just like a price list |
| 13:56.39 | Uatec | lol, thinwires |
| 13:56.47 | Uatec | someone gave us £5k today by accident |
| 13:56.52 | Uatec | when we offered it back they said keep it |
| 14:00.16 | Mercestes | thinwires: I worked fo ra company like that. |
| 14:00.35 | Mercestes | and 5k euro is like, what, 20 bucks in US dollars? |
| 14:00.38 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:00.46 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 14:00.54 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:00.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:01.37 | thinwires | unfortunatley that was a GBP symbol, and they are worth almost 2X USD |
| 14:01.57 | Mercestes | impossible. |
| 14:02.18 | GiantPickle | Can anyone help me? My zaptel channels are not loading. safe_asterisk is running, and I've got one zip channel from and did provider. How do I get my zaptel to reload? or load for that matter? |
| 14:02.27 | GiantPickle | sry... zip=sip |
| 14:04.44 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:04.49 | e-ddie | GiantPickle: zip != sip |
| 14:04.51 | *** join/#asterisk lerat (n=dnormand@bas2-montreal19-1177750105.dsl.bell.ca) |
| 14:05.43 | lerat | hi everybody |
| 14:05.57 | *** join/#asterisk Lavmol (n=chatzill@SDBRON98-1168104984.sdsl.bell.ca) |
| 14:06.11 | Mercestes | GiantPickle, reload chan_zap.so |
| 14:06.12 | thinwires | 1 British pound = 1.9838 U.S. dollars |
| 14:06.16 | lerat | i m having a big issue here |
| 14:06.22 | GiantPickle | e-ddie: I mean that 'zip channel' was supposed to be 'sip channel' |
| 14:06.33 | GiantPickle | Mercestes: from the console? |
| 14:07.19 | Mercestes | but that won't fix your zip channels. |
| 14:07.19 | Mercestes | GiantPickle, Yes. |
| 14:07.19 | Mercestes | GiantPickle, But htat only fixes zap channels, not zip or sip channels |
| 14:07.19 | lerat | some of my remote phone logout randomly and i dont know why... any clue? |
| 14:07.29 | thinwires | are any of the devices over a nat? |
| 14:07.34 | Mercestes | lerat: By log out randomly, you mean they go "TOO LAGGED" and stop responding for about 30 seconds, and then come back online? |
| 14:07.39 | GiantPickle | Mercestes: I'm getting errors on that |
| 14:07.39 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:07.49 | lerat | not really |
| 14:07.51 | Mercestes | GiantPickle, I'm pretty sure those errors are the source of your problems then. |
| 14:08.04 | slmnhq | How does one go about setting their own phone call rates (on top of Telco charges) if one was providing a premium service via phone? |
| 14:08.08 | Mercestes | lerat, Then elaborate on "log out randomly." |
| 14:08.08 | GiantPickle | Mercestes: you are probably right.. =) |
| 14:08.34 | GiantPickle | Mercestes: [Apr 17 07:06:57] WARNING[4373]: chan_zap.c:11067 process_zap: Ignoring signalling |
| 14:08.53 | lerat | they simply go off line. they can call outside but cannot recieve any call. the call go imidiately in the voicemail |
| 14:08.54 | GiantPickle | [Apr 17 07:06:57] ERROR[4373]: chan_zap.c:10426 build_channels: Unable to reconfigure channel '1' |
| 14:08.55 | Mercestes | GiantPickle, That's a warning, not an error, and is normal. |
| 14:09.06 | GiantPickle | next line is an error |
| 14:09.20 | thinwires | www.pastebin.ca ? |
| 14:09.23 | slmnhq | eg: If someone calls my service number to query some restaurant information, how can I charge them a premium per minute? |
| 14:10.09 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:11.37 | thinwires | hey guys, when I make an outbound call from my poly it goes straight to busy signal, does that sound like a port forwarding issue? |
| 14:11.46 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:11.46 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:12.00 | Uatec | What's your opinion of the Aastra phone sets? |
| 14:12.13 | lerat | Mercestes ... any clue |
| 14:13.14 | lerat | by the way the peer is not consider UNREACHABLE |
| 14:13.49 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 14:14.02 | lerat | Sorry .. it is consider UNREACHABLE |
| 14:15.26 | *** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell) |
| 14:15.26 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 14:16.10 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:16.10 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:16.38 | *** join/#asterisk snowy_owl (i=0@200.218.196.2) |
| 14:16.51 | Lavmol | Hey all wondering if someone can give me a little advice here I have 3 phones outside the LAN that can make calls to the internal LAN but the internal phones call those phones and I get directed to VM? |
| 14:17.00 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:17.15 | lerat | Same problem here |
| 14:18.05 | Lavmol | lerat: You running poly's |
| 14:18.11 | lerat | Lavmol : is this problem come random?? |
| 14:18.41 | Lavmol | lerat: not sure just recently made the changes to the configuration! |
| 14:19.06 | lerat | Not me personnaly but there is one companie i know how does and they have that problem |
| 14:19.38 | lerat | the phones momentarely stop reponding for no reason |
| 14:19.39 | osiris | kinda sounds like a nat traversal error |
| 14:19.51 | JT | sounds very obviously like one actually :) |
| 14:20.06 | lerat | what type of nat problem? |
| 14:20.09 | JT | sounds like someone has canreinvite= set to yes |
| 14:20.12 | JT | instead of no |
| 14:20.15 | Lavmol | Ya that is what I am thinking... Maybe the router on the remote end |
| 14:20.24 | *** join/#asterisk Corydon76-home (i=beige@pdpc/supporter/sustaining/Corydon76-home) |
| 14:20.24 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:20.49 | osiris | Lavmol, sounds like more your end, if you cant go outbound |
| 14:20.59 | thinwires | lol I'm having the same problem, except my real problem is I can't get my boss to port forward the 10000-20000 range due to "security" issues :-( |
| 14:21.29 | JT | the problem is the integrity of the information in his brain has been compromised? |
| 14:21.48 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:22.06 | Lavmol | I have to agree it is a big range if you are not sure... |
| 14:22.20 | snowy_owl | i'm back.. I've a question for you: the version 1.4 (asterisk, course) has a different behavior of 1.2 when we talk about 'canreinvite'. The 1.4 tells: "In Asterisk 1.4 this setting also affect direct RTP at call setup (a new feature in 1.4 - setting up the call directly between the endpoints instead of sending a re-INVITE". The version 1.4.0 works fine. But 1.4.2 hasnt this behavior. |
| 14:22.28 | thinwires | he's a bit weird when it comes to that stuff... the other real question is do I go behind his back and just call the Data center and have them do it for me |
| 14:22.45 | JT | Lavmol: it's just a range |
| 14:23.24 | snowy_owl | I'm using the same conf files to them. 1.4.2 works like 1.2, sending the INVITE or UPDATE after 200 OK. |
| 14:23.34 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 14:23.47 | snowy_owl | Is it a bug? |
| 14:25.13 | JT | thinwires: you can also alter the range rtp uses |
| 14:25.50 | lerat | JT : the only thing i need to do to correct the logoff problem is to make the router NOT REINVITE |
| 14:25.54 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:26.06 | thinwires | hm, if I make it smaller it wont effect the quality? |
| 14:26.11 | etfonhomey | If a call comes in without CallerID information, it appears to on my sip stations as "asterisk", however, if CallerID info exists, it shows up on my sip stations correctly. Anyone know where I can change it from displaying "asterisk"? |
| 14:26.17 | JT | lerat: yes i already said how to fix that, and i think you mean asterisk not router |
| 14:26.17 | *** join/#asterisk znoG (n=gs@OL132-95.fibertel.com.ar) |
| 14:26.38 | JT | thinwires: it will affect how many simultaneous calls and calls per second |
| 14:26.49 | lerat | weel actually my asterisk is not set on reinvite on any ext. |
| 14:26.51 | JT | call setups/second or time |
| 14:26.53 | GiantPickle | Folks. I hate to do this. I really need help. I've spent about 2 hours on trying to fix my * box this morning. I'm pretty new to it, and I'm at a loss for how to fix it. Can anyone help me with some trouble shooting? |
| 14:27.13 | JT | GiantPickle: you haven't given us much to go off |
| 14:27.24 | thinwires | JT: oh, well I have 4-5 poly's, that shouldn't really need 10K ports then eh? |
| 14:27.41 | GiantPickle | JT: true... I did a bit ealier... I can't get my zaptel channels to load |
| 14:27.56 | etfonhomey | GiantPickle, what version of * and zaptel? |
| 14:27.58 | JT | thinwires: port each way, nat mappings take a while to timeout if that's an issue |
| 14:28.13 | lerat | JT : none of my phone are set with REINVITE and i still have a logoff problem |
| 14:28.48 | thinwires | JT: well the phone is recieving calls like a champ, when dialing out they go to busy and the asterisk console only shows info when in debug and the port numbers in the debug i know are firewalled |
| 14:29.56 | GiantPickle | etfonhomey: I believe zaptel is 1.4.0, and * is also 1.4.0 but that seems a bit odd |
| 14:30.20 | JT | thinwires: right |
| 14:30.21 | etfonhomey | GiantPickle, Why is that odd? |
| 14:30.45 | GiantPickle | etfonhomey: not sure... just didn't expect to be same version number |
| 14:31.24 | etfonhomey | GiantPickle, OK, it doesn't load, tell is more about the "not loading" part. |
| 14:31.30 | etfonhomey | is=us |
| 14:31.54 | JT | lerat: need more info to being to diagnose that |
| 14:32.06 | GiantPickle | when I do a reload chan_zap.so I get an error... [Apr 17 07:31:40] ERROR[4373]: chan_zap.c:10426 build_channels: Unable to reconfigure channel '1' |
| 14:32.07 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 14:32.11 | lerat | what do you need? |
| 14:32.38 | Mercestes | GiantPickle, Nice, pastebin your zapata.conf, zaptel.conf and lspci of whatever card you are using. |
| 14:32.40 | JT | lerat: quick summary of the setup, the problem, and any pertinant config or debugging info |
| 14:33.01 | JT | pastebin is at pastebin.ca, not this channel window, btw :) |
| 14:33.09 | GiantPickle | Mercestes: k |
| 14:33.13 | GiantPickle | jussec |
| 14:34.43 | *** join/#asterisk mkl1525 (n=mkl1525@pD953076A.dip0.t-ipconnect.de) |
| 14:34.46 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:35.26 | lerat | JT : the phone is an Aastra 480i it has a 2000 register , no reinvite, behind nat (linksys router-basic),the phone is obviously remote. The problem is that the phone comes offline with an UNREACHABLE message. The person can make call but cannot receive?? |
| 14:35.28 | mkl1525 | Hi, is the phone.conf used for all (sip) phones or some special ones? |
| 14:37.11 | JT | what do you mean 2000 register? |
| 14:37.27 | JT | mkl1525: it's not used. |
| 14:37.43 | GiantPickle | Mercestes: http://www.pastebin.ca/444773 |
| 14:38.03 | defswork | whats the cheapest way to get my home phone served by asterisk ? |
| 14:38.40 | lerat | JT : it s the QUALIFY time 2000 (ms) |
| 14:38.47 | mkl1525 | JT, thanks so for what is it used normally? |
| 14:39.02 | Mercestes | I could be wrong but, shouldn't you have fxo in one config file, and fxs in the other config file? |
| 14:39.09 | JT | lerat: that has nothing to do with registering |
| 14:39.13 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-0-0-cust301.bagu.broadband.ntl.com) |
| 14:39.19 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 14:39.20 | Mercestes | like fxo expects fxs signalling and fxs expects fxo signalling or something else anti-intutive like that? |
| 14:39.23 | JT | mkl1525: it's an old relic of the past i believe |
| 14:39.39 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:39.40 | lerat | JT : can i msg you |
| 14:39.44 | GiantPickle | Mercestes: this system was working pretty much perfect up until yesterday... fyi |
| 14:39.51 | mkl1525 | JT, ok thanks |
| 14:39.53 | JT | lerat: not unless absolutely necessary |
| 14:40.03 | Mercestes | GiantPickle, Ok, what happened yesterday? |
| 14:40.12 | lerat | Ok |
| 14:40.23 | etfonhomey | GiantPickle, I don't see anything glaringly wrong with those. |
| 14:40.28 | GiantPickle | I'm not sure... came in this morning and the box was unresponsive... and spewing stuff to the screen |
| 14:40.34 | GiantPickle | had to reboot |
| 14:40.34 | tzafrir | Mercestes, you have the counter-intuitive signalling in botrh zaptel.conf and zapata.conf |
| 14:40.45 | etfonhomey | GiantPickle, What was it spewing? |
| 14:40.53 | GiantPickle | etfonhomey: couldn't read it |
| 14:41.16 | Mercestes | tzafrir, Ah, so if it's an FXO card you have FXS in both zapata.conf and zaptel.conf? Whew, that's a relief. Atleast it's consecutive across allconfig files. :) |
| 14:41.53 | lerat | JT : just confirm this to me : The Qualify if the time the phone has to respond and Regitry is how long the phone is Login ? |
| 14:42.00 | JT | no |
| 14:42.07 | JT | it has NOTHING to do with registering |
| 14:42.08 | lerat | ok ... |
| 14:42.12 | Zand3r | Hi all... I am configuring an asterisk server and all is well so far however I would like the voicemail system to say the date/time as well as the caller id before playing the message. I have set "saycid = yes" and this worked. Am I correct that "envelope = yes" and "tz=Europe/London" is the correct configuration for playing the date/time? Am I using the tz= correctly and will this factor in daylight savings or do I need t |
| 14:42.12 | Zand3r | o do this manually when the clocks change? |
| 14:42.14 | tzafrir | Mercestes, what do you think about: http://bugs.digium.com/view.php?id=9496 |
| 14:42.32 | JT | qualifying just means asterisk sends a SIP OPTIONS packet every so often to see if it can still see the other SIP endpoint |
| 14:42.41 | Mercestes | GiantPickle, Ok, try this. stop now. Drop out of asterisk. Do a wanrouter restart. Then do your ztcfg -s and ztcfg -ccv or something like that, then do an asterisk -cvvvvvvvv and see if it comes up. Could be wanrouter isn't running. |
| 14:42.48 | GiantPickle | Last entry in my log is from 3:39 am... at what time it seemed to be working fine |
| 14:42.54 | tzafrir | Zand3r, the timezone factors in daylight savings, yes |
| 14:42.55 | JT | if it exceeds (in this case 2000 milliseconds) it will deem the device as UNREACHABLE |
| 14:43.00 | GiantPickle | Mercestes: k.. will try |
| 14:43.17 | lerat | JT : ok and the register |
| 14:43.28 | Mercestes | tzafrir, That would be helpful. :) |
| 14:43.31 | JT | lerat: registering is seperate |
| 14:43.49 | JT | lerat: regular registering or qualifying should both hold most NAT connections open |
| 14:43.50 | Mercestes | tzafrir, What would be even more helpful is some level of autodetection on Asterisk's part, but, that wouldnt' be very linux-like. |
| 14:43.56 | JT | then again some NAT routers are rubbish |
| 14:44.03 | tzafrir | Well, I need someone to test it on actual hardware. With our drivers it's there from day 1 |
| 14:44.15 | Mercestes | GiantPickle, This is a controlled system restart btw where you are implicitly starting each service in the ordre that it needs to be restarted. |
| 14:44.28 | Zand3r | tzafrir: Thanks for the confirmation. Is "tz=Europe/London" the correct format for that particular timezone (i.e. using capital letters, the forward slash, etc.)? |
| 14:44.28 | *** join/#asterisk Nugget (i=nugget@dazed.notslacker.com) |
| 14:45.01 | tzafrir | Mercestes, you mean something like: http://bugs.digium.com/view.php?id=7613 |
| 14:45.32 | ourkid | I am currently using a Cisco 2821 as a ISDN>SIP gateway, and i have my asterisk set up to trunk into this and it works fine for incoming and outgoing calls, i use 7940 handsets. However when i transfer an incoming DID call to another extension, they hear me behind asterisk, however i dont hear them. Any ideas? |
| 14:45.49 | tzafrir | Anything more would require better sysfs support. With decent sysfs support, you could do pretty cool udev hotplugging tricks |
| 14:45.51 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 14:46.07 | Mercestes | tzafrir: Yes. :) |
| 14:46.08 | JT | ourkid: any NAT? |
| 14:46.29 | GiantPickle | Mercestes: can you tell me a bit more about "ztcfg -ccv or something like that"? |
| 14:46.32 | lerat | JT : so wtih what i told you do you have any advice on why my phones are loging off |
| 14:46.44 | Mercestes | GiantPickle, Gah... |
| 14:46.47 | ourkid | well i am not using any devices outside the current subnet, the 79xx's and the cisco router are all on the same subnet, the phones have to be nat=no |
| 14:46.59 | GiantPickle | Mercestes: I'm sorry... I told you I was new |
| 14:47.21 | JT | lerat: is the CLI providing any info on whether the phones appear to be reachable or what not? |
| 14:47.28 | Mercestes | GiantPickle, Sure, let me cat my /etc/init.d/zaptel so I can see how it automaticallydoes it for me in one line with a "start" "stop" and "restart" interface since I use gentoo. :P |
| 14:47.43 | etfonhomey | GiantPickle, you should stop asterisk, rmmod zaptel (and related modules) and start from there. |
| 14:47.47 | Mercestes | GiantPickle, Just type ztcfg |
| 14:47.52 | JT | ourkid: sounds like one direction or the RTP stream is shooting off into cyberspace, so to speak |
| 14:47.57 | lerat | JT : i will check now |
| 14:48.01 | Mercestes | rmmod? I thought it was modprobe -r |
| 14:48.05 | etfonhomey | GiantPickle, modprobe each one and run dmesg and see if you get any errors. |
| 14:48.34 | etfonhomey | modprobe -r is probably cleaner |
| 14:48.51 | lerat | JT : right now it s UNREACHABLE |
| 14:48.58 | ourkid | JT: the irony is when i transfer between the local handsets internal calls, it works fine, however the issue seems to be only with the SIP trunk |
| 14:49.34 | Mercestes | lerat: Sounds like a nat/firewall/router/network issue. |
| 14:49.41 | etfonhomey | JT or Mercestes: Do you know the proper setting for echotraining on a Digium TDM400P? |
| 14:49.43 | Mercestes | Cheap switch, cheap router, nat, etc. |
| 14:49.57 | GiantPickle | Mercestes: doing what you said seems to have done the trick |
| 14:50.03 | kovger | Hi, i got some problem with fax detection, i can receive fax if i directly call the extension with rxfax in, but if i try to let the fax detect decide if it's a fax or voice, the fax extension never get called. I switch faxdetect=both in my zapata.conf. I'm useing te110p+* 1.2.17 with zaptel-1.2.16. Plz can somebody help? (and sorry english is not my native language) |
| 14:50.22 | Mercestes | etfonhomey, unfortunately no, I think it's one of those process of elimination discovery things where you just have to try values until it works. |
| 14:50.26 | lerat | Mercestes : by cheap switch you mean my asterisk |
| 14:50.28 | GiantPickle | Mercestes: thanks |
| 14:50.32 | GiantPickle | Ethon: thanks |
| 14:50.36 | GiantPickle | err... |
| 14:50.41 | GiantPickle | etfonhomey: thanks |
| 14:50.42 | Mercestes | GiantPickle, Cheers. |
| 14:50.44 | JT | ourkid: terminology thing, no such thing as a sip trunk, BTW.. care to attack it with network packet sniffers? |
| 14:50.45 | etfonhomey | Mercestes, should it atleast be yes? |
| 14:50.51 | GiantPickle | I'll save a log of this for furture reference |
| 14:50.56 | JT | etfonhomey: not sure |
| 14:51.05 | Mercestes | etfonhomey, Oh, I was thinking fxotune. Um, well, that depends. |
| 14:51.12 | etfonhomey | GiantPickle, np |
| 14:51.30 | mkl1525 | on some calls we lose the 1-2 seconds of the beginning of a call from isdn/e1 to our internal snom360 sip phones although the * server hasn't a high load. Any suggestions what the cause could be? |
| 14:51.37 | Mercestes | etfonhomey, If your running fax over it then it should be no, if not, and there is no echo on yoru line then it should still be no. If there is echo, it should be no and you should have the echo fixed through the rx and tx values and through your telco |
| 14:51.48 | Mercestes | etfonhomey, If you simply cannot fix the echo no matter what, or you are lazy, then it should be set to yes. |
| 14:52.01 | Mercestes | That's my personal feelings on it, |
| 14:52.04 | etfonhomey | Mercestes, then I'll set it to yes. LOL! |
| 14:52.08 | Mercestes | :D |
| 14:52.22 | etfonhomey | Mercestes, Actually, I only have echo for the first few seconds of the call. |
| 14:52.31 | ourkid | JT: can do, my meaning is traffic between the 2821<->Asterisk<->79xx ends up 1 way audio, calls 79xx<->Asterisk<->79xx ends up fine |
| 14:52.50 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 14:52.58 | Mercestes | etfonhomey, I've heard of that. Forget what causes it, I think the echo training on the remote end perhaps. |
| 14:53.09 | JT | ourkid: does the 2821 reinvite to the 79xx? |
| 14:53.31 | JT | Mercestes: it'd be asterisk echo training |
| 14:54.34 | Mercestes | JT: Yea, maybe echocancel=no would fxi the problem by making it echo continuously. |
| 14:54.42 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 14:54.48 | GiantPickle | Mercestes: so, if in fact it was that the wanrouter was not running... why would that be? any ideas? |
| 14:54.55 | ourkid | JT: I have canreinvite enabled for the handset's account |
| 14:55.07 | JT | ourkid: try with it disabled |
| 14:55.10 | Mercestes | GiantPickle, Because wanrouter is not part of yoru startup process. |
| 14:55.25 | Mercestes | GiantPickle, wanrouter is your driver for that card. Until wanrouter is running your box cannot see the card. |
| 14:55.45 | *** join/#asterisk docelm0 (n=vircuser@c-68-45-140-42.hsd1.de.comcast.net) |
| 14:55.48 | Mercestes | GiantPickle, I just programmed it into my /etc/init.d/zaptel startup/shutdown process but, that's distro specific. |
| 14:56.07 | *** join/#asterisk ManOfMilk (n=CpnPlnet@70-56-29-78.eugn.qwest.net) |
| 14:56.49 | lerat | Mercestes : you told me i have cheap devices ... can you tell me what kind of router i should use instead? |
| 14:57.15 | ourkid | JT: attempting now |
| 14:58.03 | Mercestes | lerat, I said I've seen cheap devices do that and listed it among the causes I've seen. Cisco routers tend to be pretty ok, as long as you don't use them with cisco switches and cisco phones. |
| 14:58.05 | JT | lerat: it might not be the device |
| 14:58.30 | JT | lerat: also make sure there is no firewall interfering, on the device or otherwise |
| 14:59.01 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 14:59.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:59.13 | DeeJayTwo | how can I be sure my rtp stream is P2P ? |
| 14:59.25 | DeeJayTwo | I got an asterisk server and 2 polycom phones. |
| 14:59.31 | lerat | Jt : but there is of course a firewall on my asterisk should i do something about it |
| 14:59.38 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 14:59.39 | DeeJayTwo | I'd like to be sure the rtp stream doesn't flow thru asterisk. |
| 14:59.44 | lerat | JT : or leave it as is |
| 14:59.46 | JT | canreinvite=yes and ensure you don't record calls or similar, DeeJayTwo |
| 14:59.59 | DeeJayTwo | JT: Is there any command giving a clue? |
| 15:00.17 | JT | lerat: err, there's a firewall on your asterisk box and you don't think it's important? |
| 15:00.22 | JT | lerat: try disabling it |
| 15:00.42 | JT | DeeJayTwo: i've already provided the relevant configuration line |
| 15:00.54 | thinwires | firewall on *server = very not yes |
| 15:01.02 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 15:01.23 | lerat | JT : i m not very familiar with firewall so what do you think might happen if i disable it? |
| 15:01.24 | Mercestes | thinwires, we used iptables firewalls on our asterisk boxes and they worked great... |
| 15:01.31 | DeeJayTwo | thinwires I have it..but can I see a clue of it in action? |
| 15:01.32 | JT | lerat: it might work |
| 15:01.41 | Mercestes | thinwires: of course, the guy responsible for them was a total freak. I don't even think he was from this planet. |
| 15:01.53 | thinwires | mercestes: haha |
| 15:01.57 | lerat | Thinwires: why do you say it s bad to have a firewall on my Asterisk? |
| 15:02.12 | JT | lerat: because it STOPS PACKETS |
| 15:02.16 | JT | if it's setup wrong |
| 15:02.17 | JT | that's bad |
| 15:02.26 | Mercestes | wel,l that's kinda what a firewall is supposed to do |
| 15:02.42 | thinwires | lerat: you need to have a bunch of ports opened up and the firewall stops all unknown traffic unless you forward it through |
| 15:03.35 | lerat | i m gonna go check the config of my firewall and come back to you ... this might be my problem from the beginning |
| 15:03.40 | etfonhomey | Mercestes and JT: How does rxgain/txgain interact with fxotune? |
| 15:03.43 | thinwires | honestly, the person responsibly for ports and firewalling pretty much fuels all of the IT field... the only reason we have jobs eh |
| 15:03.57 | JT | lerat: next step is to break out sip debug on the cli and network packet sniffers :) |
| 15:04.09 | JT | etfonhomey: i try to avoid analogue where possible :) |
| 15:04.29 | etfonhomey | JT: You do SIP or PRI's? |
| 15:04.34 | JT | etfonhomey: yes |
| 15:04.36 | JT | to both |
| 15:04.54 | Mercestes | etfonhomey, Well, rxgain/txgain are basically just volume controls. Fxotune is supposed to create those volume settings by testing different values and spitting out whatever eliminates echo I believe. |
| 15:05.04 | heison | anyone here has experience with USB FM tuner or soundcard line-in as MOH? |
| 15:05.45 | etfonhomey | Mercestes, so after I've run fxotune -i x and then fxotune -s /etc/fxotune.conf, will it affect my echo if I increase rxgain? |
| 15:06.05 | Mercestes | etfonhomey, Should. |
| 15:06.22 | *** join/#asterisk _mike3_ (n=mike3@dhcp-0-13-10-78-a2-54.cpe.mountaincable.net) |
| 15:06.52 | *** join/#asterisk wunderkin (n=kev@65.39.92.95) |
| 15:07.04 | lerat | thinwires: what does that means : Reject new packet without SYN? |
| 15:07.05 | [TK]D-Fender | DeeJayTwo: Pastbin the CLI output of a call at verbose 10, and then do "show channels concise" followed by show channel [channel]" for each leg of the call. |
| 15:07.26 | etfonhomey | Mercestes, how do I get the right balance then? My receiving volume on calls is very low while the transmit volume is correct. There is virtually no echo now. I'd like to increase the volume on the rx side, but not add echo. |
| 15:07.34 | JT | [TK]D-Fender: i'm going for the world record polycom call on hold :P |
| 15:07.53 | [TK]D-Fender | JT : what are you waiting FOR? |
| 15:07.56 | etfonhomey | JT: You're caling Polycom support? |
| 15:08.00 | JT | no |
| 15:08.15 | JT | i have a polycom here that has had some calls on hold for 530 hours now |
| 15:08.30 | _mike3_ | I'm looking for a good web source with third party asterisk addons. Eg: Alarm. Anyone have a site? |
| 15:08.55 | JT | it doesn't display time in "days" and "weeks" on the screen |
| 15:09.01 | JT | hours seems to be its biggest unit |
| 15:10.42 | groogs[h] | JT: the day it does display 'days' and 'weeks' on hold is the day i will be scared |
| 15:10.43 | JT | groogs[h]: well call duration in general |
| 15:10.43 | JT | it's just the call timer |
| 15:10.43 | Mercestes | etfonhomey, Try echotraining=400 echocancel=yes echochancelwhenbridged=yes then. |
| 15:10.43 | *** join/#asterisk rogerz (i=jon13@cpe-24-195-144-82.nycap.res.rr.com) |
| 15:10.43 | Mercestes | etfonhomey, As long as you rnot faxing over these lines. |
| 15:10.48 | JT | happy it hasn't crashed though |
| 15:11.37 | etfonhomey | Mercestes, not faxing, so I'll try it. |
| 15:12.17 | etfonhomey | Mercestes, do you need to run fxotune -s /etc/fxotune.conf after every time you load the wctdm and zaptel modules? |
| 15:12.44 | JT | anyone know somewhere other than voipsupply that sells the new A101D? |
| 15:12.57 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:12.59 | groogs[h] | yeah even so, >=24 hours on the phone is a bit excessive for my tastes. nice that it can be done though i guess. ;) |
| 15:13.13 | Mercestes | etfonhomey, no. Just once. |
| 15:13.30 | JT | groogs[h]: heh, you don't think i've actually been sitting at this phone for 530 hours do you? :P |
| 15:15.16 | EzWay | everything seem running fine on my system ; but in my log i am getting this : no zaptel transcoder support ; what that mean exactly ? |
| 15:15.55 | lerat | Jt : any idea what that means : it s in my firewall option : reject new packect without SYN |
| 15:16.38 | JT | lerat: that option relates only to TCP |
| 15:16.39 | Mercestes | JT: If yes, then it's shower time. |
| 15:16.46 | JT | UDP is stateless |
| 15:17.10 | lerat | JT : so it has nothing to do wtih the communication |
| 15:17.40 | _mike3_ | wher can I get a good monitoring gui app for asterisk? I want something so I see how is connected how many people on the phone etc, etc... |
| 15:17.45 | JT | asterisk SIP and IAX2 only uses UDP |
| 15:17.56 | tzafrir | etfonhomey, yes, |
| 15:17.56 | lerat | ok |
| 15:18.12 | tzafrir | make it mart of the zaptel init.d script and forget about it |
| 15:18.24 | lerat | i m gonna got check some more |
| 15:19.21 | JT | lerat: how about you put the output of the two commands following into pastebin.ca?: iptables -L |
| 15:19.26 | JT | iptables -t nat -L |
| 15:19.29 | Vec | To get features.conf working, I have unhased blind transfer and attendedtransfer reloaded asterisk set reinvite=no, but when I push #1 nothing happens ? |
| 15:20.00 | Mercestes | _mike3_, Try FOP |
| 15:20.04 | [TK]D-Fender | Vec: pastebin your attempt at verbose 10 |
| 15:20.33 | _mike3_ | FOP ok.. |
| 15:20.54 | etfonhomey | Mercestes or tzafrir, who's right on the fxotune -s? Both of you have given valuable info on here. |
| 15:21.40 | Mercestes | If someone contradicted me on any of my answers concerning fxotune then I vote for the other guy. I'm not an expert at fxotune by anymeans. |
| 15:21.51 | Mercestes | I'm just reciting what I remember reading from voip-info |
| 15:21.57 | Vec | [TK]D-Fender : there is no indication of a key being pressed on the log, just a normal call setup, the other part just hears beep beep ? |
| 15:22.17 | etfonhomey | Mercestes, OK. |
| 15:22.21 | [TK]D-Fender | Vec: *PASTEBIN* |
| 15:22.32 | Vec | [TK]D-Fender : ok ok |
| 15:22.58 | etfonhomey | tzafrir, can I get your opinion on increasing rxgain after using fxotune and it's affect on echo? |
| 15:23.22 | ourkid | JT: tried setting canreinvite to no on all the handsets aswell as the sip out settings but still no incoming audio |
| 15:23.24 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 15:23.32 | ourkid | JT : after transfer |
| 15:23.37 | tzafrir | etfonhomey, the thing is that you need to set the parameters to the channel after it has been created (and before it gets opened by asterisk) |
| 15:23.49 | tzafrir | It gets generated when you load wctdm |
| 15:24.07 | JT | ourkid: i blame it on the ciscos :) |
| 15:24.19 | _VoiceMeUp_Com | Apr 16 14:52:48 WARNING[15726]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8713fd8 (len 893) to 712.3.12.1.123:0 returned -1: Invalid argument |
| 15:24.26 | _VoiceMeUp_Com | again and then no more zap ... |
| 15:24.31 | _VoiceMeUp_Com | its since .17. |
| 15:24.41 | _VoiceMeUp_Com | or can a sangoma dis in less the n 3-4 months ? |
| 15:24.58 | etfonhomey | tzafrir, you mean having the settings you want to use in zapata.conf before starting Asterisk, right? That any edits to zapata.conf require an * restart? |
| 15:24.58 | ourkid | JT : the 7940's, would rulling it out by using softphones be a good measure |
| 15:25.05 | JT | [TK]D-Fender: heard of the A101D? |
| 15:25.05 | Mercestes | _VoiceMeUp_Com, how does a chan_sip.c warning shutdown your zap channels? |
| 15:25.09 | JT | ourkid: probably |
| 15:25.10 | DeeJayTwo | If the rtp connection is point to point...should I be able to stop asterisk and keep the conversion up? |
| 15:25.12 | _VoiceMeUp_Com | no idea |
| 15:25.17 | _VoiceMeUp_Com | but when i see that... |
| 15:25.22 | [TK]D-Fender | JT : Welcome to 2 WEKKS AGO :) |
| 15:25.24 | JT | ourkid: i suspect the isdn gateway might be a problem |
| 15:25.24 | _VoiceMeUp_Com | the zap cant be reused |
| 15:25.28 | [TK]D-Fender | WEEKS* |
| 15:25.37 | JT | [TK]D-Fender: heh, it was a quiet announcement |
| 15:25.45 | _VoiceMeUp_Com | all new calls say.. channel is blah all frozen .. even stop now |
| 15:25.48 | _VoiceMeUp_Com | needs a killall |
| 15:25.53 | JT | it's pointless at voipsupply's crap prices though |
| 15:26.01 | JT | mayaswell get 2 ports for the difference |
| 15:26.12 | lerat | JT : i m not very familiar with pastebin.ca...how is it suppose to work? |
| 15:26.24 | Mercestes | _VoiceMeUp_Com, Very strange. |
| 15:26.28 | _VoiceMeUp_Com | yes |
| 15:26.29 | JT | lerat: you paste stuff into it so you don't flood the channel |
| 15:26.34 | _VoiceMeUp_Com | its always call from pstn |
| 15:26.39 | etfonhomey | lerat, think of it as a scratch pad for notes that you want to pass to people here. |
| 15:26.40 | _VoiceMeUp_Com | to asterisk |
| 15:27.02 | etfonhomey | lerat, you paste stuff in there, hit submit, and then paste the URL you get into here. |
| 15:27.22 | Mercestes | _VoiceMeUp_Com, how does a call from a PSTN generate a chan_sip warnign? |
| 15:27.39 | _VoiceMeUp_Com | zap/1 to REMOTEast01 |
| 15:27.42 | lerat | ok thanks |
| 15:27.43 | _VoiceMeUp_Com | ;) |
| 15:27.43 | etfonhomey | lerat, you might want to stay away from pasting in real PSTN numbers and passwords... |
| 15:27.49 | _VoiceMeUp_Com | but not sure |
| 15:27.53 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:28.28 | _VoiceMeUp_Com | its like.. if iun between pri and asterisk.. the hangup was not trasnmitted |
| 15:28.49 | lerat | #444836 |
| 15:29.01 | _VoiceMeUp_Com | so pri hardware got not got hangup but aserisk did.. then on it wanting to reuse.. it says that channel is locked |
| 15:29.10 | *** join/#asterisk alexns (n=alex@static-acs-24-154-114-15.zoominternet.net) |
| 15:29.23 | *** join/#asterisk ars247 (n=no@64-142-43-180.dsl.static.sonic.net) |
| 15:29.29 | Vec | [TK]D-Fender : here is the pastebin http://pastebin.ca/444849 |
| 15:29.34 | alexns | need some quick help.... cant remove /sbin/ztcfg permission denied as root |
| 15:29.55 | _VoiceMeUp_Com | killall -9 ztcfg first |
| 15:30.00 | _VoiceMeUp_Com | maybe its running and locked ? |
| 15:30.02 | Qwell[laptop] | no |
| 15:30.03 | _VoiceMeUp_Com | no idea |
| 15:30.03 | Vec | I assume you can transfer a call even if you made the call ? |
| 15:30.04 | Qwell[laptop] | it's just a file |
| 15:30.29 | alexns | got no process killed |
| 15:30.30 | _VoiceMeUp_Com | true |
| 15:30.38 | Qwell[laptop] | _VoiceMeUp_Com: f that didn't work, you'd never be able to upgrade things like glibc, or your kernel, or rpm |
| 15:31.20 | Qwell[laptop] | unlike...*cough*windows*cough* |
| 15:31.29 | _mike3_ | I'm looking for a good web source with third party asterisk addons. Eg: Alarm. Anyone have a site? |
| 15:31.40 | alexns | tried lsattr ------dA----- ztcfg |
| 15:31.45 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-98-7.red.bezeqint.net) |
| 15:31.57 | *** join/#asterisk mavior (n=chatzill@81-174-45-129.f5.ngi.it) |
| 15:32.08 | alexns | need some ideas |
| 15:32.56 | Dovid | hi guys. i have an off topic question. my provider is doing authentication based on IP and i am trying new equpiment on a DSL connection that gets changed. I went to send the calls thru a linux box. anyone know of sofware that will accept data on a certain port and then just pass it along on the same por to a diffrent iP ? |
| 15:33.05 | [TK]D-Fender | Vec: Executing [s@macro-zapDial:1] Dial("SIP/fax2-9bf51600", "Zap/g1/0118845293") in new stack |
| 15:33.07 | Dovid | IP* |
| 15:33.34 | [TK]D-Fender | Vec: See anything missing in there concerning the ABILITY to TRANSFER? |
| 15:33.58 | Vec | [TK]D-Fender : yeh, I am trying to transfer the call I initiated, from the SIP/fax2 ? |
| 15:34.09 | Vec | its a phone not a fax, just called it fax |
| 15:34.27 | [TK]D-Fender | Vec: think about what OPTIONS you have to pass DIAL to allow the caller to TRANSFER calls... |
| 15:35.08 | Vec | [TK]D-Fender : oh yeh, dumb, sorry tT, errr |
| 15:35.15 | Vec | thanks |
| 15:35.20 | Vec | I should have seen that |
| 15:35.40 | mavior | hello everybody,i have some problems with my phones and my "r" buttons that seems to be related to that problem http://www.asteriskguru.com/archives/asterisk-users-flash-hook-hangup-problem-vt30039.html?highlight=flash+button , now i want to use the callwaiting feature, anybody can say how can i set a simple extension to flash my channel to achieve the same behaviour as I pressed my... |
| 15:35.42 | mavior | ...hook/flashbutton ? tnx |
| 15:37.17 | _VoiceMeUp_Com | so what does this error mean ? tehcnically |
| 15:37.28 | DeeJayTwo | I have canreinvite=yes in sip.conf.. when two sip phones get on a conversaion |
| 15:37.40 | DeeJayTwo | a rtp debug IP show the rtp packets... |
| 15:37.57 | mavior | ? |
| 15:38.55 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 15:40.07 | etfonhomey | Can anyone recommend a good ATA with at least 2 FXO ports? |
| 15:40.10 | [TK]D-Fender | DeeJayTwo: Pastbin the CLI output of a call at verbose 10, and then do "show channels concise" followed by show channel [channel]" for each leg of the call. |
| 15:40.20 | [TK]D-Fender | DeeJayTwo: ^^^^^ I asked you this a LONG time ago.... |
| 15:40.24 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-bc3cfac2b73264df) |
| 15:41.12 | _VoiceMeUp_Com | ok i found out |
| 15:41.38 | _VoiceMeUp_Com | Apr 16 14:52:48 WARNING[15726]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8713fd8 (len 893) to 12.3.12.1.123:0 returned -1: Invalid argument |
| 15:41.38 | mavior | anybody flashing here ? :P |
| 15:41.43 | Mercestes | _VoiceMeUp_Com, Ok, what does it mean? The suspense is killing me. |
| 15:41.44 | _VoiceMeUp_Com | this means.. the PORT is bad.. the user has a default ip addrss in case of not online.. |
| 15:41.59 | _VoiceMeUp_Com | but the asterisk is maping in memory has default :0 |
| 15:42.15 | Mercestes | _VoiceMeUp_Com, So 5060 does not exist on the specified peer? |
| 15:42.19 | _VoiceMeUp_Com | why choose port 0 if theres an ip in htere.. the default should be 5060 at least no ? |
| 15:42.26 | Mercestes | _VoiceMeUp_Com, or whatever port it's trying to send to |
| 15:42.27 | _VoiceMeUp_Com | it exists |
| 15:42.35 | _VoiceMeUp_Com | when he regusters its addr :1.2.3.4:5060 |
| 15:42.42 | _VoiceMeUp_Com | but default 1.2.3.4:0 |
| 15:42.58 | _VoiceMeUp_Com | sip show peer blah |
| 15:43.13 | Mercestes | _VoiceMeUp_Com, So your sending a call to an invalid peer? |
| 15:43.20 | _VoiceMeUp_Com | yes |
| 15:43.23 | _VoiceMeUp_Com | not me |
| 15:43.33 | Mercestes | _VoiceMeUp_Com, but I do that all the time and don't get that warning. |
| 15:43.45 | _VoiceMeUp_Com | .17 ? |
| 15:43.56 | _VoiceMeUp_Com | ADDr->IP : 1.2.3.4 port 5060 |
| 15:44.04 | _VoiceMeUp_Com | Defaddr-> : 1.2.3.4:0 |
| 15:44.18 | _VoiceMeUp_Com | you see that asterisk loads the default ip but forgets about port |
| 15:44.24 | _VoiceMeUp_Com | unless hte current peer is regged |
| 15:44.30 | _VoiceMeUp_Com | then both show same info |
| 15:44.42 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 15:44.55 | _VoiceMeUp_Com | meaning there is no default port or mainlay a default of 0 ( since it cant think of a default port to use ) wow.. how about 5060 |
| 15:45.05 | ctooley | Anyone know if there's a Call-ID field in Zap or IAX channels? Is that just a SIP thing? |
| 15:45.12 | lerat | is there a command to see the firewall setting in cli???? |
| 15:45.36 | Mercestes | lerat: which CLI? which firewall? |
| 15:45.38 | ctooley | lerat, which firewall? |
| 15:45.39 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:45.58 | Mercestes | _VoiceMeUp_Com, Here is my problem tho.... |
| 15:46.03 | lerat | i have a firewall set on my asterisk server |
| 15:46.10 | ctooley | if it's a linux iptables firewall "iptables -L -n && iptables -t nat -L -n" |
| 15:46.12 | Mercestes | _VoiceMeUp_Com, I send calls to offline peers all the time and I don't get this error. why are you getting this error? |
| 15:46.24 | Mercestes | _VoiceMeUp_Com, you can't hav ethe only offline sip peers in the asterisk community. |
| 15:46.27 | lerat | and i want to see it just like a SIP ext. in cli |
| 15:47.38 | _VoiceMeUp_Com | i have a default addy to contact in the mysql |
| 15:47.49 | _VoiceMeUp_Com | as a field for realtime.. but could be in sip if you want |
| 15:47.51 | _VoiceMeUp_Com | deffaddr |
| 15:48.02 | Mercestes | _VoiceMeUp_Com, I see. Maybe you should update that to be 5060 instead of 0 then. |
| 15:48.05 | _VoiceMeUp_Com | do you use defaults ? if not then thats it |
| 15:48.12 | _VoiceMeUp_Com | i cant .. unless i mode source |
| 15:48.17 | _VoiceMeUp_Com | im hunteing right now |
| 15:48.26 | Mercestes | you can't? |
| 15:48.33 | Mercestes | didn't you just say it was realtime? |
| 15:48.37 | _VoiceMeUp_Com | well ineed to mod source |
| 15:48.43 | _VoiceMeUp_Com | but cant find where that is defined |
| 15:48.46 | Mercestes | isn't it a mysql field? |
| 15:48.51 | _VoiceMeUp_Com | there is no default port field |
| 15:48.55 | _VoiceMeUp_Com | no |
| 15:49.04 | Mercestes | so where are you seeing defaddr? |
| 15:49.10 | _VoiceMeUp_Com | argh |
| 15:49.23 | *** join/#asterisk jmls (n=JBouncer@host86-135-47-194.range86-135.btcentralplus.com) |
| 15:49.26 | Mercestes | I've used realtime too and I stil didn't have this problem. |
| 15:49.33 | Mercestes | and no, it wasn't realtime on 1.2.17. |
| 15:50.19 | _VoiceMeUp_Com | i have ipadr field.. then port then defaultip |
| 15:50.23 | _VoiceMeUp_Com | no defaultport |
| 15:50.29 | _VoiceMeUp_Com | is that flag even alive ? |
| 15:50.35 | _VoiceMeUp_Com | i cant just invent them |
| 15:50.52 | _VoiceMeUp_Com | hmm |
| 15:51.06 | _VoiceMeUp_Com | line 328 chan_sip.c #define DEFAULT_SIP_PORT |
| 15:51.09 | _VoiceMeUp_Com | =5060 |
| 15:51.12 | _VoiceMeUp_Com | so no idea |
| 15:51.41 | Mercestes | I don't think I used a default Ip field, I'm pretty sure I left it null. |
| 15:51.56 | *** join/#asterisk Lavmol (n=chatzill@69.159.222.24) |
| 15:51.58 | _VoiceMeUp_Com | yeah i removed option |
| 15:51.59 | Mercestes | do a select distinct defaut ip from your sip table. does it return multiple values or just one? |
| 15:52.01 | _VoiceMeUp_Com | line 12548 |
| 15:52.05 | _VoiceMeUp_Com | is where it gets default ip |
| 15:52.11 | _VoiceMeUp_Com | there is no read for default port |
| 15:52.13 | _VoiceMeUp_Com | only port |
| 15:52.16 | Mercestes | or you could just modify the source code and recompile... |
| 15:52.18 | *** part/#asterisk jmls (n=JBouncer@host86-135-47-194.range86-135.btcentralplus.com) |
| 15:52.18 | _VoiceMeUp_Com | which is hsared with ipaddr |
| 15:52.26 | Mercestes | and then ask us for help when you have an error no one else has. |
| 15:52.33 | _VoiceMeUp_Com | so.. but disconnecting a peer FORCES the entries to 0.0.0.0. : 0 |
| 15:52.45 | _VoiceMeUp_Com | hence the defualt trying to use the port 0 |
| 15:53.27 | DeeJayTwo | [TK]D-Fender : http://www.pastebin.ca/444887 |
| 15:56.44 | _VoiceMeUp_Com | ah |
| 15:56.46 | _VoiceMeUp_Com | <PROTECTED> |
| 15:56.53 | _VoiceMeUp_Com | so its missing something |
| 15:56.54 | _VoiceMeUp_Com | found it |
| 15:57.02 | _VoiceMeUp_Com | line 12524 chan_sip.c of 1.2.17 |
| 15:57.22 | _VoiceMeUp_Com | <PROTECTED> |
| 15:57.26 | _VoiceMeUp_Com | that needs an ELSE |
| 15:57.50 | _VoiceMeUp_Com | right ? |
| 15:58.03 | _VoiceMeUp_Com | i dont want to start adding more problems then there is |
| 15:58.11 | _VoiceMeUp_Com | but merc.. if you think its safe |
| 15:58.12 | GiantPickle | Mercestes: the wanrouter is actually in my startup routines, it would seem. |
| 15:58.13 | [TK]D-Fender | DeeJayTwo: And where is the first part I asked for? |
| 15:58.33 | Mercestes | GiantPickle, Then my "fix" doesnt' make sense, does it? |
| 15:58.46 | DeeJayTwo | [TK]D-Fender sorry.. one moment... |
| 15:59.18 | Mercestes | _VoiceMeUp_Com, I'm no expert on C but, wouldn't peer->addr.sin_port = 0; } suggest that you are already assigning addr.sin_port=0; in the event that addr.sin_port is not otherwise assigned? |
| 16:00.16 | _VoiceMeUp_Com | it looks like no.. that sounds like if you have a port.. then backup current port to default port and make it 0 lol |
| 16:00.26 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.234) |
| 16:00.36 | Mercestes | _VoiceMeUp_Com, In fact, that appears to check to see if addr.sin_port has a value at all and assigning it to 0 if not. |
| 16:01.14 | DeeJayTwo | [TK]D-Fender : http://www.pastebin.ca/444902 |
| 16:01.15 | _VoiceMeUp_Com | <PROTECTED> |
| 16:01.17 | _VoiceMeUp_Com | they saty |
| 16:01.24 | _VoiceMeUp_Com | <PROTECTED> |
| 16:01.25 | Mercestes | I don't know. |
| 16:01.41 | *** join/#asterisk _trace (n=_trace@c-75-69-191-164.hsd1.vt.comcast.net) |
| 16:01.45 | _VoiceMeUp_Com | but .. is (1) { def=current; cur=0 } |
| 16:01.48 | _VoiceMeUp_Com | is a translation |
| 16:01.56 | _VoiceMeUp_Com | but .. is (1) { def=current; current=0 } |
| 16:02.00 | Mercestes | yea it's starting to look like that. |
| 16:02.01 | [TK]D-Fender | DeeJayTwo: see this? -- Executing [s@macro-dial:3] Dial("SIP/107-08277140", "SIP/102||wWtT") in new stack |
| 16:02.10 | [TK]D-Fender | DeeJayTwo: thats your dial line right? |
| 16:02.13 | _VoiceMeUp_Com | so it port ... then backup port to default and make port 0 lol |
| 16:02.15 | DeeJayTwo | yes |
| 16:02.16 | _VoiceMeUp_Com | that bad |
| 16:02.25 | Mercestes | _VoiceMeUp_Com, So then change it.... |
| 16:02.32 | Uatec | Hey, is there a way I could connection a Skype like client to Asterisk? maybe using SIP?, which could then be anywhere on the network without need of hardware ? |
| 16:02.36 | [TK]D-Fender | DeeJayTwo: Typically this means * is FORCED in the way of RTP |
| 16:02.38 | _VoiceMeUp_Com | well.. thing is why woudnt it cause probs to others |
| 16:02.53 | Zdrulio | hm asterisk GUI is a free software right ? |
| 16:02.53 | Mercestes | _VoiceMeUp_Com, That is a very good question. |
| 16:02.54 | DeeJayTwo | ok |
| 16:03.00 | [TK]D-Fender | DeeJayTwo: any DTMF feature removes the ability for re-invite |
| 16:03.01 | DeeJayTwo | how to set it properly? |
| 16:03.11 | Zdrulio | i read at digum website 1000$ ? |
| 16:03.12 | [TK]D-Fender | DeeJayTwo: looks like you CAn"T |
| 16:03.18 | Mercestes | _VoiceMeUp_Com, Maybe you should look and see whenthe call is made if it uses def_port first and then tries the abs_port or if it uses the abs_port first. |
| 16:03.37 | Mercestes | _VoiceMeUp_Com, It could just be a bad utilization of variables. |
| 16:04.01 | CunningPike | Has anyone experienced problems with a TE410P and zaptel 1.4.0? |
| 16:04.29 | _VoiceMeUp_Com | no |
| 16:05.18 | _VoiceMeUp_Com | i looked into it further.. its on the read oof otuboundproxy/... is that a new variable ? |
| 16:05.44 | _VoiceMeUp_Com | means if dynamic and not outboundproxy then .. if not in memory flush the port lol |
| 16:06.09 | DeeJayTwo | [TK]D-Fender: So how can I achieve phone 2 phone RTP stream? |
| 16:06.10 | DeeJayTwo | I mean... |
| 16:06.17 | DeeJayTwo | What's the proper way to do things to get it done? |
| 16:06.51 | tzanger | [TK]D-Fender: morning |
| 16:06.54 | tzanger | or rather afternoon now |
| 16:07.35 | rogerz | how many kbytes a second is a sip call? trying to figure what I'll need for outgoing bandwidth for about 20 phones |
| 16:07.44 | Qwell[laptop] | rogerz: it depends |
| 16:07.45 | *** join/#asterisk ucfMethod (n=ucfmetho@office.eyestreet.com) |
| 16:07.57 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-51-85.dsl.irvnca.pacbell.net) |
| 16:08.12 | rogerz | Qwell, how about max |
| 16:08.12 | BSD_Tech | ok where is the guy who does the friday podcast |
| 16:08.22 | Qwell[laptop] | rogerz: again, it depends |
| 16:08.23 | BSD_Tech | I need to catch up with him |
| 16:08.34 | Mercestes | _VoiceMeUp_Com, Aren't you glad you didnt' change it now? |
| 16:08.34 | Qwell[laptop] | For all we know, you could be doing h264 video |
| 16:09.20 | Qwell[laptop] | it depends almost entirely on the codec you use |
| 16:09.20 | _VoiceMeUp_Com | well.. still trying to figure this out.. makes no sense putting it to 0 |
| 16:09.20 | rogerz | ahh alright |
| 16:09.20 | _VoiceMeUp_Com | it should ratther be loader like defaultip |
| 16:09.20 | Qwell[laptop] | G.711 uses about 80k/s, something like G.279 uses about 13 |
| 16:09.20 | _VoiceMeUp_Com | as defaulport so ill lookinto it |
| 16:09.20 | Qwell[laptop] | I think |
| 16:09.25 | _VoiceMeUp_Com | then ill make a patch for my boxes.. since.18 will mess things up |
| 16:09.39 | Mercestes | _VoiceMeUp_Com, If I were a developer with the source memorized, capable of troubleshooting a unique error you have by lookin gat a single line of code....I'd be working for digium, don't you think? |
| 16:09.43 | ucfMethod | simple question, 'show channels verbose' displays channels and calls.. I want to gauge the number of PRIs we need, so the active channels would be channels on a PRI correct? |
| 16:10.21 | Mercestes | ucfMethod, not necessarily. Many people just call themselves. You'd want to look at the number of outbound calls leaving the PBX. But, yes ,that's a good place to start. |
| 16:11.31 | ucfMethod | Mercestes: thanks... |
| 16:12.47 | _VoiceMeUp_Com | hehe |
| 16:12.55 | _VoiceMeUp_Com | yeah that why i pasted line #'s |
| 16:12.56 | _VoiceMeUp_Com | ;) |
| 16:13.15 | *** join/#asterisk xkev (i=kevin@orbit.xmission.com) |
| 16:13.47 | xkev | new polycom 2.x firmware, is there a way to get it to stop showing @realm on calls? |
| 16:14.07 | Mercestes | which does me no good because I'm not running 1.2.17 |
| 16:14.27 | Mercestes | but I dont' randomly patch my source everything I see a warning message either. |
| 16:14.44 | Mercestes | so even if I was running 1.2.17 I'm not 100% sure it would do me any good |
| 16:15.14 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 16:16.01 | *** join/#asterisk mrdigital (n=mrdigita@207-172-229-15.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 16:16.11 | mrdigital | can someone tell me how to use a zoom 5801 with asterisk as a ATA |
| 16:16.18 | Mercestes | I applaud your initiative, of course but...I'm not really sure what yoru running now, ya know? and I seriously haven't seen that error, nro does it make sense. |
| 16:16.19 | mrdigital | i have it configured but it says not ready to make calls |
| 16:16.29 | Mercestes | so my concern is, why is that happening to you? |
| 16:24.48 | CunningPike | Has anyone seen this error in /var/log/messages relating to zaptel: kernel: TE4XXP: Version Synchronization Error! |
| 16:25.31 | *** join/#asterisk the_planarian (n=the_plan@bas4-ottawa23-1088826734.dsl.bell.ca) |
| 16:25.43 | CunningPike | We just attempted an upgrade, but had to back out because our PRIs were unstable, and /var/log/messages has lots of that error in it |
| 16:25.45 | the_planarian | hello! |
| 16:26.21 | CunningPike | xkev: Do you have URI dialing enabled? |
| 16:27.09 | *** join/#asterisk sooth (n=soothsay@bas5-montreal02-1167963639.dsl.bell.ca) |
| 16:28.08 | *** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
| 16:28.10 | ryguillian | Die |
| 16:28.11 | AndrewGearhart | does asterisk have the ability to do any sort of voice recognition? |
| 16:28.27 | *** join/#asterisk darylvoip (n=darylvoi@pool-72-78-162-79.phlapa.east.verizon.net) |
| 16:28.49 | darylvoip | Anyone have a good way to bind to low ports (port 30 specifically) when running asterisk as non-root with the init.d script? |
| 16:29.08 | CunningPike | AndrewGearhart: Many people have used Festival with success - there is another one, too - can't recall what it is |
| 16:29.27 | CunningPike | darylvoip: There's a good way to do that? |
| 16:29.38 | sooth | Festival is speech synthesis |
| 16:31.02 | darylvoip | Well, any way ;) |
| 16:31.03 | Mercestes | darylvoip, I think there is a kernel option to allow non-root users to bind to low ports somewhere in make menuconfig. I couldn't tell you specifically how to do that, but it would be a global kernel change allowing all services to do that. |
| 16:31.03 | CunningPike | sooth: Aye - what's the other one |
| 16:31.03 | darylvoip | Not my choice to be using those ports, but its brutal to get around the Qatar firewall. |
| 16:31.03 | robin_sz | darylvoip, you need to be root then drop privileges after assigning the low port |
| 16:31.03 | sooth | sooth: Other what? Sphinx is another speech synthesis system |
| 16:31.03 | darylvoip | Right....robin_sz |
| 16:31.03 | CunningPike | sooth: Right - thanks |
| 16:31.03 | darylvoip | Was wondering if there were a safe_asterisk or simial script that was already out there to do that |
| 16:31.03 | CunningPike | AndrewGearhart: Sphinx, according to sooth |
| 16:31.03 | AndrewGearhart | CunningPike: I'm looking actually to go the other way... from speech to entry... |
| 16:31.20 | *** join/#asterisk rrocha (n=bastard@201.47.29.34.adsl.gvt.net.br) |
| 16:31.21 | robin_sz | Mercestes, nah, almost all proceses eg apache are non-root yet run on low ports, you just open and then drop privs |
| 16:31.24 | AndrewGearhart | oooh |
| 16:31.49 | AndrewGearhart | thanks CunningPike and sooth |
| 16:31.51 | Mercestes | robin_sz, That would be the sane way to do it, yes. |
| 16:32.04 | darylvoip | Sane is good. Otherwise it just won't get done. |
| 16:32.09 | robin_sz | Mercestes, well, sane is good |
| 16:32.52 | mavior | hello ,anybody can say how can i set a simple extension to flash my channel to achieve the same behaviour as I pressed my hook/flashbutton(to use the callwaiting feature) ? tnx |
| 16:32.58 | *** join/#asterisk Shoeb (n=chatzill@64.34.69.9) |
| 16:32.59 | sooth | AndrewGearhart: http://cmusphinx.sourceforge.net/sphinx4/ |
| 16:33.12 | Shoeb | What does it mean when you can call extensions, but not outside numbers? |
| 16:33.18 | sooth | AndrewGearhart: I made a mistake. Sphinx is recognition, not synthesis |
| 16:33.34 | AndrewGearhart | sooth: actually... that's what I want.... is recognition. |
| 16:33.50 | AndrewGearhart | sooth: so, your mistake answered my real question. :) |
| 16:33.50 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-98-7.red.bezeqint.net) |
| 16:33.55 | sooth | AndrewGearhart: Yes. I was correcting myself. Above I said Sphinx is speech synthesis. |
| 16:34.04 | AndrewGearhart | ah |
| 16:34.06 | Dovid | how do i see what codec a specific call is being used ? |
| 16:34.12 | sooth | AndrewGearhart: http://www.voip-info.org/wiki-Sphinx |
| 16:34.17 | darylvoip | Shoeb - little more info. All of your extensions can't call out? This is a new problem or a new setup? What is your outbound (sip, zap channels, etc.). Any debugging info? |
| 16:34.17 | Dovid | i tried show channel X and it wont show the codec in use |
| 16:34.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:35.50 | mavior | i have written something like "exten => 42,1,Flash()" but dialing 42 when i'm on calling does not work (even cli does not shows up nothing) like if dtmf are not recognized by asterisk |
| 16:38.34 | Dovid | how do i see what codec a specific call is being used ? |
| 16:38.36 | Dovid | i tried show channel X and it wont show the codec in use |
| 16:38.44 | mavior | please guys...i'm gonna to be mad on this |
| 16:39.18 | AndrewGearhart | anybody here worked with LumenVox vs Sphinx? |
| 16:41.50 | the_planarian | hmm... too busy "Open Source-ing" for a friendly "hello" back, i see... ;) |
| 16:43.29 | xkev | cunningpike, I do have uri dial enabled |
| 16:43.36 | xkev | I shall try that knob |
| 16:43.48 | xkev | <2.x didn't matter on what it displayed |
| 16:43.52 | CunningPike | xkev: Try turning it off |
| 16:44.00 | CunningPike | We turned ours off for the same reason |
| 16:45.53 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 16:46.05 | the_planarian | looking for some help with a much more basic, introductory question...can anyone please help? |
| 16:46.56 | CunningPike | the_planarian: Only if you ask |
| 16:47.38 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-226-244.red.bezeqint.net) |
| 16:47.55 | *** join/#asterisk kizmet (n=kizmet@AeriaSolutionsPtyLtd.fe0-1.aes-brd-0.agl.cbr.as-ip.net.au) |
| 16:50.38 | *** join/#asterisk ZefK (n=Zefk@81.181.249.106) |
| 16:51.36 | xkev | cunningpike, thanks, worked great |
| 16:51.46 | CunningPike | xkev: Excellent |
| 16:53.00 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 16:53.54 | plasmid | what tool can I use to catch sip wireless traces? In other words my WIfi phone is not registering with my provider so I would like to troubleshoot and catch the packets(?) |
| 16:53.56 | [TK]D-Fender | the_planarian: Don't ask to ask.... |
| 16:54.30 | _VoiceMeUp_Com | ah |
| 16:54.37 | _VoiceMeUp_Com | ok after 2 gig of file checking |
| 16:54.47 | xkev | ngrep |
| 16:54.50 | _VoiceMeUp_Com | :15 DEBUG[12901] chan_sip.c: SIP message could not be handled, bad request: 0e35be991136bc1b765c372b74e12dc8@ |
| 16:54.55 | _VoiceMeUp_Com | bad request |
| 16:55.02 | xkev | ngrep -W byline -d eth0 <- for example |
| 16:55.05 | _VoiceMeUp_Com | hmm just before a crash down |
| 16:55.18 | rogerz | We have a dsl line coming into an asterisk box, all outgoing calls through sip account. We want to get a new line for web/email servers 7mb/1mb. If I hook this into the switch, change dhcp for default gateway to the 7mb/1mb line, all calls should still function properly throught he asterisk box going through the dsl correct? or will changing the default gw make them call through the new line |
| 16:55.23 | xkev | ..you need to have the traffic passing through eth0 from phone<->provider |
| 16:55.39 | xkev | (then probably put on a libpcap filter, like 'port 5060' on the end) |
| 16:55.59 | xkev | plasmid ^^ |
| 16:56.02 | *** join/#asterisk davidcsi (n=davidcsi@213.201.53.222) |
| 16:56.37 | xkev | if you want to catch wireless, you need a card that supports promiscuous, but much easier just to catch it when it hits the wire |
| 16:56.43 | davidcsi | guys question: is it possible to DISABLE sdp mess 180? I set it up to send 183, but i need * NOT to send the 180 before de 183 |
| 16:57.06 | *** join/#asterisk sharp (n=sharp@pool-71-242-110-119.phlapa.east.verizon.net) |
| 16:57.30 | xkev | rogerz, default gateway == gateway |
| 16:57.41 | xkev | if you want something specific to not use the gateway, set a more specific route |
| 16:58.02 | *** join/#asterisk AF-Slash (n=AF-Slash@71-210-59-29.hlna.qwest.net) |
| 16:58.14 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:58.21 | plasmid | xkev, err.. promiscuous? |
| 16:58.36 | xkev | e.g. ip route add 4.3.2.0/24 via 199.104.120.1 |
| 16:58.38 | rogerz | so even though the phone connects to the asterisk box, the calls will still be made through the default gateway? |
| 16:58.44 | xkev | (doh, gave away my subnet) |
| 16:59.03 | xkev | plasmid, there is not an easy way |
| 16:59.17 | Sweeper | omg I haxxors j00r compf |
| 16:59.29 | xkev | best to snag it as it passes through some linux pc (like using a bridge, which is what I do for sniffy in lab) |
| 16:59.52 | *** join/#asterisk icel (n=dan@65.200.26.80) |
| 17:00.35 | the_planarian | CunningPike: it's a stupid question... but one that I imagine has already been asked before.... 1.) why is there no version of asterisk for Windows and 2.) will there ever be one... so that transitioning users can have a "fighting chance" to make the switch much more smoothly from windows to linux? |
| 17:01.05 | the_planarian | lol |
| 17:01.08 | CunningPike | the_planarian: There is a version for Windows. It |
| 17:01.13 | the_planarian | err... |
| 17:01.14 | CunningPike | It's as good as Windows is |
| 17:01.27 | davidcsi | anyone¿ |
| 17:02.24 | the_planarian | re: the "About" page on www.asterisk.org -- "Asterisk® is a complete IP PBX in software. It runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris" |
| 17:02.32 | the_planarian | i see no mention of windoze. :P |
| 17:02.34 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-094-225.pools.arcor-ip.net) |
| 17:02.40 | [TK]D-Fender | the_planarian: Correct. |
| 17:02.50 | [TK]D-Fender | the_planarian: how "astute" of you. |
| 17:02.56 | the_planarian | err.. |
| 17:03.08 | Mercestes | google asterisk windows |
| 17:03.12 | Mercestes | there is a windows port for it |
| 17:03.19 | [TK]D-Fender | the_planarian: Have you seen a copy of Apple's Safari browser for windows available anywhere yet? |
| 17:03.20 | Qwell[laptop] | and it's only 3 years old |
| 17:03.36 | the_planarian | o_O |
| 17:03.40 | [TK]D-Fender | Qwell :Virtualized BS... its not a "windows" app/ |
| 17:03.44 | plasmid | xkev, ahh.. good I see a selection for "capture packets in promiscuous mode" |
| 17:04.15 | davidcsi | guys question: is it possible to DISABLE sdp mess 180? I set it up to send 183, but i need * NOT to send the 180 before de 183 |
| 17:04.35 | Mercestes | the_planarian, They're just jealous of the microsoftonian empire. They too will be assimiliated. |
| 17:04.44 | the_planarian | i don't blame them. ;) |
| 17:05.25 | the_planarian | unfortunately i have already been assimilated, now i'm looking to gain further independance from them, that's all. |
| 17:05.57 | the_planarian | :) |
| 17:07.05 | sooth | Can someone tell me what the minimum system requirements for no-frills Asterix (one line + voicemail stuff)? |
| 17:07.14 | blitzrage | A S T E R I S K |
| 17:07.29 | wunderkin | asstricks |
| 17:07.33 | blitzrage | or that |
| 17:07.39 | sooth | blitzrage: Sorry |
| 17:07.40 | Lavmol | Hey all is their a way to see if the is any call forwarding on extensions |
| 17:07.45 | sooth | asterisk |
| 17:07.47 | blitzrage | Asterix is a French comic |
| 17:07.58 | wunderkin | o rly? |
| 17:08.01 | blitzrage | yes |
| 17:08.09 | wunderkin | ic, thats why eh |
| 17:08.15 | blitzrage | what is why? |
| 17:08.19 | *** part/#asterisk Shoeb (n=chatzill@64.34.69.9) |
| 17:08.21 | blitzrage | they are not related |
| 17:08.27 | davidcsi | anyone on my sdp 180/183 message thing?? |
| 17:08.28 | wunderkin | why people say that sometimes :P... shrug |
| 17:08.33 | blitzrage | I guess |
| 17:08.36 | wunderkin | could be ha |
| 17:08.41 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:08.46 | blitzrage | they are lazy and don't know how to pronounce Aster-isk |
| 17:08.49 | CunningPike | the_planarian: Save yourself the agony and start on Linux |
| 17:08.50 | mrdigital | blitzrage: im trying to use Zoom 5801 as a adapter to hook up a analog phone to the * Box |
| 17:08.52 | icel | Does anyone have a digium card working with a voice T1? I have a TE405P that isn't working yet and I don't know if it is my zaptel configs or something with asterisk |
| 17:08.54 | sooth | blitzrage: Yes, I know the difference, believe me. I'm punished by Google every time |
| 17:08.55 | mrdigital | it says not ready to make calls |
| 17:08.59 | mrdigital | even though its configured |
| 17:08.59 | icel | configs are at http://pastebin.ca/444977 |
| 17:09.01 | mrdigital | any ideas? |
| 17:09.22 | blitzrage | sooth: to answer your question... Asterisk could do one-line and voicemail on like... a PII easily |
| 17:09.30 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:09.42 | sooth | blitzrage: 128 meg RAM is sufficient? |
| 17:09.51 | blitzrage | yep |
| 17:10.02 | blitzrage | RAM isn't really as important as CPU |
| 17:10.19 | _VoiceMeUp_Com | youll have spike on cpu on transcoding/recordging and the such |
| 17:10.22 | sooth | blitzrage: Thanks. One more question. Can a softmodem be used as a FXS? |
| 17:10.26 | blitzrage | once things are loaded up, that's about all the RAM Asterisk will use (until you get a LOT of calls) |
| 17:10.27 | _VoiceMeUp_Com | moh |
| 17:10.35 | reber | hi |
| 17:10.39 | blitzrage | sooth: certain ones can -- you don't want to use them -- they SUCK |
| 17:10.55 | reber | mpg123 is cpu hungry, isn't it ? |
| 17:11.07 | blitzrage | don't use mpg123 -- use native MoH |
| 17:11.32 | the_planarian | CunningPike: I am... unfortunately my first desktop-worthy experience has ended up being Knoppix.. :/ |
| 17:11.33 | blitzrage | mpg123 is ol' sk00l (and not the in the good way) |
| 17:11.41 | reber | blitzrage, how to ? |
| 17:11.59 | anonymouz666 | mpg123 sucks ballz |
| 17:12.22 | anonymouz666 | I got one mpg123 process running using 99% CPU |
| 17:12.36 | reber | anonymouz666, exactly the same prob here |
| 17:12.41 | _VoiceMeUp_Com | ok so a 3 year old fix neds to be refixxed |
| 17:12.42 | _VoiceMeUp_Com | http://bugs.digium.com/bug_view_page.php?bug_id=0000956 |
| 17:12.57 | reber | "native MoH" how that |
| 17:12.59 | sooth | blitzrage: I will only be using the softmodem temporarily (I need it working ASAP). |
| 17:13.02 | sooth | How can I tell if my device is compatible? Is there a list somewhere? Will Asterisk tell me? |
| 17:13.11 | the_planarian | oh, bugger me... look what i've found.... www.asterieriskwin32.com :P |
| 17:13.26 | Mercestes | _VoiceMeUp_Com, Of course. that explains why there is so much information on your warning. |
| 17:13.28 | the_planarian | serves me right for asking stupid questions first thing in the morning... |
| 17:13.32 | blitzrage | sooth: you're better off using an ITSP then -- those softmodems are a waste of time -- no CallerID or remote disconnect supervision |
| 17:13.45 | _VoiceMeUp_Com | http://lists.digium.com/pipermail/asterisk-gui/2007-March/000264.html |
| 17:13.55 | _VoiceMeUp_Com | this also.. seems it in march 30 buf |
| 17:13.58 | _VoiceMeUp_Com | for the gui |
| 17:14.12 | sooth | blitzrage: Erm, no CallerID is a deal breaker for me. |
| 17:14.29 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 17:14.35 | xkev | I never got the callerid to work w/ the cheap softmodem either |
| 17:14.45 | chefrs | I just got a fax and it was only like an inch tall. Is there any way to fix this? |
| 17:14.47 | xkev | it'd come through, but it wouldn't parse |
| 17:14.50 | blitzrage | sooth: exactly -- use an ITSP, not a crappy cheap modem. Or get a TDM400P |
| 17:14.53 | xkev | sound was horrid too |
| 17:14.58 | blitzrage | and echo |
| 17:15.03 | blitzrage | its a waste of $10 |
| 17:15.12 | blitzrage | better of taking the $10 and using a pre-paid ITSP account |
| 17:15.23 | lude | does anyone else have trouble getting cisco 7960 phones to register against 1.4.2, when they worked fine in 1.2.x |
| 17:15.24 | Mercestes | the_planarian, It's not a stupid question. Windows is a beast to program for so....very little support for it. |
| 17:15.49 | blitzrage | lude: nope -- has worked fine for the last 3 months on 1.4 for me |
| 17:16.03 | lude | i wonder if i'm just doing somethign stupid |
| 17:16.16 | blitzrage | possibly :) |
| 17:16.30 | icel | can anyone help me with a voice T1? |
| 17:16.40 | chefrs | icel: I just set mine up today. |
| 17:16.43 | Strom_M | icel: maaaaaybe |
| 17:16.49 | icel | configs are at http://pastebin.ca/444977 |
| 17:17.00 | icel | basically zaptel loads fine but doesnt seem to actually 'work' |
| 17:17.09 | Strom_M | icel: is it a voice T1, or is it ISDN PRI? |
| 17:17.13 | icel | it says it is configured but asterisk can't create a channel. PRI |
| 17:17.20 | chefrs | What hardware is it? |
| 17:17.26 | lude | blitzrage: you just used the same configs from 1.2 ? |
| 17:17.27 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 17:17.28 | icel | TDM405P |
| 17:17.34 | the_planarian | Mercestes: tell me about it... :P i just hope that one day i'll be knowlegable enough about linux to quite even possibly host "transitioner" services from windows to linux. |
| 17:17.50 | chefrs | Did you get any new drivers for it? |
| 17:17.53 | blitzrage | lude: yep |
| 17:17.56 | the_planarian | (and perhaps especially for stuff like Asterisk) |
| 17:17.57 | reber | "native MoH" what is that ? |
| 17:18.03 | Strom_M | icel: what about dialing ZAP/G1/13115552368 |
| 17:18.24 | icel | chefrs: I just used zaptel 1.4 |
| 17:18.26 | blitzrage | reber: what does google tell you? |
| 17:18.37 | chefrs | icel: I had to get a .c file when I put mine in. |
| 17:18.57 | lude | blitzrage: can you paste me a snippit from your sip.conf for your 7960 please? |
| 17:19.04 | Strom_M | icel: try dialing ZAP/G1 instead of ZAP/1 |
| 17:19.06 | lude | i maybe have some dumb option set that i shouldn't or something |
| 17:19.24 | icel | Strom_M: how do I dial from the console? |
| 17:19.29 | Strom_M | no no |
| 17:19.33 | Strom_M | from your dialplan |
| 17:19.44 | Strom_M | extensions.conf and all that :) |
| 17:20.27 | reber | blitzrage, http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 17:20.28 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 17:20.34 | icel | Strom_M: same error when I try to dial ZAP/G1 |
| 17:20.42 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:20.46 | icel | 'No channel type registered for Zap' |
| 17:20.55 | Strom_M | what happens when you type "PRI show span 1" at the console? |
| 17:20.55 | blitzrage | reber: yes, I know where the link is, thx |
| 17:21.11 | reber | blitzrage, np |
| 17:21.34 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 17:21.42 | icel | strom_m: no such command, PRI. I am using 1.4 |
| 17:21.59 | chefrs | Whuuuu? |
| 17:22.10 | anonymouz666 | how slow is using realtime extensions? |
| 17:22.27 | Strom_M | icel: does your zapatq.conf have a [channels] section above the settings? |
| 17:22.47 | [TK]D-Fender | Strom_M: Better prepare yourself for some serious pain & handholding... |
| 17:22.49 | icel | strom_m: no |
| 17:23.01 | Strom_M | icel: well there's your problem then |
| 17:23.10 | chefrs | Anyone know anything about receiving faxes? |
| 17:23.30 | Strom_M | chefrs: yeah, you make a high pitched whine for a while, then you gurgle a bit |
| 17:23.34 | Strom_M | and then the fax comes through |
| 17:23.51 | icel | strom_m: thx. I'll look for a doc on it I guess |
| 17:24.04 | chefrs | lol |
| 17:24.13 | Strom_M | icel: just add [channels] at the top and then restart asteisk |
| 17:24.15 | Strom_M | asterisk |
| 17:24.21 | wunderkin | asterix |
| 17:24.48 | Strom_M | assfucking |
| 17:25.10 | Corydon-w | Anybody have resources for DTMF tweaking in chan_zap beyond relaxdtmf=yes ? |
| 17:25.26 | chefrs | Strom_M: I can see the system accept the fax but the .pdf is only like, an inch tall. |
| 17:25.33 | Strom_M | Corydon-w: I gave up on it and had my client buy polycoms instead |
| 17:25.59 | Corydon-w | Strom_M: this is to interface to a legacy system via D4/AMI/E&M |
| 17:26.19 | Strom_M | yeah |
| 17:26.29 | Strom_M | what weirds me out is that it all worked perfectly for a year |
| 17:26.41 | Strom_M | and then my client started having weirdo DTMF issues |
| 17:26.53 | _VoiceMeUp_Com | you have a channel => 1 def in zapata.conf ? |
| 17:27.01 | Corydon-w | The issue I'm having is that Asterisk is not picking up all the DTMF |
| 17:27.22 | Corydon-w | The legacy system sends *92279 and the Asterisk system sees 97, for example. |
| 17:29.22 | icel | chefrs: what .c file did you have to get? |
| 17:30.05 | icel | strom_m: still doesn't have a registered channel. Any other ideas? |
| 17:30.07 | blitzrage | Corydon-w: really? that is really wierd...... thats like what I was seeing on chan_sip... |
| 17:30.30 | *** join/#asterisk grndslm (n=grndslm@host-69-59-102-128.nctv.com) |
| 17:31.02 | grndslm | what does this whole verizon vs vonage thing have to do with asterisk?? anything? |
| 17:31.37 | grndslm | are there still going to be DID providers and such? |
| 17:31.47 | illsci | hey is there a way to bind asterisk to a single ip |
| 17:32.01 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 17:32.02 | illsci | i have a box with a /27 dedicated to it... and i have one virtual ip |
| 17:32.05 | Strom_M | grndslm: give up now and go back to using two tin cans and a string |
| 17:32.08 | chefrs | icel: r1t1.c |
| 17:32.11 | illsci | i see registrations bouncing between the two ips i have |
| 17:32.25 | *** join/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 17:32.32 | illsci | is there a way to set it to just use a single ip address... |
| 17:32.41 | chefrs | illsci: Tell your phones to just use one IP? |
| 17:33.19 | grndslm | well, if Verizon does end up winning against Vonage, wouldn't that mean that all voip solutions are infringing on Verizon's patents? |
| 17:33.44 | the_planarian | well folks, thanks for all the help on this matter! wish me luck on days ahead! :) |
| 17:34.13 | Corydon-w | grndslm: Please consult a lawyer. We are unqualified to give you legal advice, and what you've asked for is legal advice. |
| 17:34.22 | *** join/#asterisk lwh (n=lwh192@rdsl-0230.tor.pathcom.com) |
| 17:34.33 | chefrs | Gah. VOIP isn't a new idea. That whole thing is silly. |
| 17:35.02 | Lavmol | Say can anyon tell me why extension 1 is rining when I dial extension 2??? |
| 17:35.17 | grndslm | no kidding...but if verizon wins this patent case...i'm just wondering wonder the future of asterisk is like |
| 17:35.18 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 17:35.34 | grndslm | if they could/would even catch asterisk users |
| 17:35.46 | Corydon-w | idle speculation |
| 17:35.55 | deeperror | anyone familiar with implementing qos on asterisk packets? |
| 17:36.29 | *** part/#asterisk the_planarian (n=the_plan@bas4-ottawa23-1088826734.dsl.bell.ca) |
| 17:36.30 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 17:37.32 | lee_is_me | hi all, is there a way to place a call a "holding pattern" and then access it later on? |
| 17:37.47 | chefrs | Park a call. |
| 17:38.02 | chefrs | http://www.voip-info.org/wiki-Asterisk+call+parking |
| 17:38.32 | AndrewGearhart | anybody have suggestions on ITSPs? |
| 17:38.38 | _VoiceMeUp_Com | hmm you can adda setvar for a peer how neat is that... |
| 17:38.42 | _VoiceMeUp_Com | in the sip.cofn definition |
| 17:38.47 | _VoiceMeUp_Com | that is soooooo neat |
| 17:38.56 | AndrewGearhart | I had vonage... loved it. No phone numbers in my area ... and they are looking like a sinking ship. |
| 17:39.09 | plasmid | AndrewGearhart, vitelity.com, teliax.com, les.net |
| 17:39.16 | AndrewGearhart | so... time to look at what I can do. |
| 17:39.19 | AndrewGearhart | ooh. thanks plasmid |
| 17:39.27 | aydiosmio | packet8 |
| 17:40.02 | deeperror | anyone familiar with qos and asterisk that could offer some assistance |
| 17:40.28 | aydiosmio | deeperror: just ask |
| 17:40.41 | plasmid | deeperror, i setup q0s and asterisk with my router. |
| 17:40.54 | deeperror | i've got ipcop setup with qos and i'm setting up rules now |
| 17:41.07 | deeperror | do i just put them on ports 5004 - 5082 and call that voip traffic? |
| 17:41.19 | deeperror | what about rtp traffic? 1024 - 64000? |
| 17:41.22 | aydiosmio | depends on what ports your UDP traffic are on |
| 17:41.29 | deeperror | thats the question |
| 17:41.32 | aydiosmio | you don't need to prioritize SIP traffic |
| 17:41.55 | aydiosmio | standard RTP ports are 10000-20000, which I belive are asterisk's default |
| 17:42.04 | *** join/#asterisk NoTurbo (n=cornholi@ip5457284a.direct-adsl.nl) |
| 17:42.25 | deeperror | ok so any rtp on those ports and any udp in the 5004-5082 give high priority |
| 17:42.35 | aydiosmio | why UDP on 5004-5082? |
| 17:42.46 | deeperror | so udp is just signaling packets |
| 17:42.47 | deeperror | ? |
| 17:42.55 | deeperror | were just concerned with the data rtp |
| 17:42.58 | aydiosmio | RTP is UDP |
| 17:42.59 | lee_is_me | chefrs: Thanks. My mistake...I want to access the channel from AGI script or outside process later on |
| 17:43.16 | aydiosmio | deeperror: SIP, the signalling protocol is TCP, which you don't need to prioritize |
| 17:43.34 | deeperror | ok that makes more sense |
| 17:43.53 | deeperror | now it says in voip-info.org that * changes the headers for a TOS response |
| 17:43.56 | blitzrage | SIP signalling is UDP, not TCP |
| 17:44.09 | *** join/#asterisk lilwookie (n=lilwooki@30-82-252-216-static.enter-net.com) |
| 17:44.10 | blitzrage | Asterisk doesn't do TCP |
| 17:44.16 | lilwookie | Hi folks ;) |
| 17:44.21 | lee_is_me | chefrs: Prolly AMI is a better choice for this |
| 17:44.29 | plasmid | blitzrage, so packets are sent via UDP on wireless? |
| 17:44.37 | blitzrage | yah |
| 17:44.42 | blitzrage | the medium doesn't mean anything |
| 17:44.49 | plasmid | blitzrage, that is if I want to capture them and analyze 'em |
| 17:44.59 | blitzrage | UDP is layer 4, IP is layer 3, ethernet is layer 2, wireless is layer 1 |
| 17:45.33 | blitzrage | whether it is wireless or not means nothing |
| 17:45.34 | lilwookie | so I have a asterisk box setup and is in use, works fine but everynow and then when picking up a call (sip phone) I get grishing noise and other party keeps ringing |
| 17:45.35 | deeperror | so if asterisk is changing the ip header to include a request for type of service....how could i read this on my router to id the packets as voip |
| 17:45.38 | blitzrage | you are still capturing at layer 2 |
| 17:46.13 | blitzrage | deeperror: the router has to understand the ToS bits in the IP packet header |
| 17:46.35 | deeperror | ipcop allows rules to be setup by ports or layer-7 |
| 17:46.55 | blitzrage | it routes at the application layer? interesting :) |
| 17:47.16 | blitzrage | you probably configure at the application layer -- filtering would still be done around layer 4 |
| 17:47.28 | blitzrage | ~osi |
| 17:47.39 | jbot | i heard osi is see OSIRM Application - 7, Presentation - 6, Transport - 5, Session - 4, Data Link - 3, Network - 2, Physical - 1.. Open Source Initiative |
| 17:47.39 | deeperror | layer-7 will have a cpu hit though and there will be a large volume of calls going over the router in full production |
| 17:49.21 | grndslm | is packet8 going to be sued by verizon if they win against vonage? would you guys think a vonage customer moving to packet8 would be a wise decision? |
| 17:50.14 | cpm | vonage will just have to pay verizon some settlement amount and enter into a license with them. I don't think they will go away |
| 17:51.00 | deeperror | blitzrage: should level7 rule be setup for SIP and that would work? Or should i push it further down than that? |
| 17:51.13 | *** join/#asterisk drega (n=drega@host217-36-49-65.in-addr.btopenworld.com) |
| 17:51.44 | grndslm | cpm: ...and other voip providers will need to do the same then? |
| 17:51.56 | cpm | probably |
| 17:51.59 | grndslm | just seems crazy that at&t can't sue verizon for connecting to their network |
| 17:52.01 | cpm | unless there is some good news |
| 17:52.40 | cpm | This is just another arrow in the thick hide of the bloated pest known as software patents |
| 17:53.02 | cpm | at some point, someone will loose enough money over this idiocy to start yelling |
| 17:53.11 | grndslm | i hope |
| 17:53.17 | HarryR | cpm, people do loose enough money over it, and are yelling :) |
| 17:55.57 | chefrs | Okay so when someone faxes me, I don't get all of the fax. Any ideas why? |
| 17:56.27 | deeperror | chefrs: bad connection |
| 17:56.46 | *** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell) |
| 17:56.46 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 17:56.52 | chefrs | It's a T1 PRI |
| 17:57.04 | chefrs | It was working fine when it was just FXO. |
| 17:57.18 | deeperror | using a codec? compression? packet loss? |
| 17:57.31 | chefrs | Haven't changed anything since the switch. |
| 17:57.37 | deeperror | except the switch |
| 17:57.45 | chefrs | Except the means of getting the faxes yes. |
| 17:58.08 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:03.06 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 18:03.06 | chefrs | deeperror: Any ideas? |
| 18:03.11 | icel | chefrs and strom_m: thx for help, got it up and running now |
| 18:03.12 | chefrs | icel: Good to hear. |
| 18:03.24 | icel | literally |
| 18:03.29 | chefrs | Hah |
| 18:03.44 | deeperror | i think it has something to do with running over the t1 you are compressing the data of the fax with some type of codec and this creates problems with faxing |
| 18:04.20 | chefrs | deeperror: I'm running some tests with some free online fax sites. I guess we'll see... |
| 18:05.06 | chefrs | It seems the first page gets cut off pretty badly, but subsequent pages seem alright. |
| 18:06.57 | chefrs | Crap. I'm all out of free fax testings |
| 18:08.38 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:13.11 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 18:16.03 | *** join/#asterisk ingenius (n=syntax@81-190-114-200.fibertel.com.ar) |
| 18:16.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-136-120.red.bezeqint.net) |
| 18:18.49 | *** join/#asterisk randalk (n=rk@theres.no.place.like.home.data102.com) |
| 18:19.00 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 18:19.13 | PupenoR | Hello. |
| 18:19.43 | randalk | heyya, anybody in here familiar with cisco 7940s? have a question, and google & newsgroups haven't gotten the job done. |
| 18:19.47 | PupenoR | When I try to make Asterisk play any audio, the demo for example, I get a debug message "Oooh, format changed to 2" and I can't hear anything on the phone. This setup was mostly unchanged, any ideas what might be wrong? |
| 18:20.07 | Hmmhesays | man that online turbo tax is a good deal |
| 18:22.32 | randalk | ... |
| 18:23.21 | randalk | anyways, my problem is that I only get 2 working softkeys on the 7940 w/ load 7.5. I'd like to at least get a CFwdAll key on there, but there seems to be no way to do it via SIPxx.cnf. Any suggestions would be helpful. |
| 18:25.01 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 18:25.10 | [[blah]asfd | anyone using the followme feature in 1.4? |
| 18:26.51 | *** join/#asterisk apardo (n=apardo@87.217.144.67) |
| 18:27.54 | lude | why on earth would this phone send an authenticate sip header to 1.2 and not to 1.4 |
| 18:28.39 | Mercestes | lude: The phone is not 1.4 compliant. contact the phone manufacturer and ask them to update their firmware. |
| 18:28.53 | lude | latest 7960 cisco firmware? |
| 18:28.54 | *** join/#asterisk mindCrime (n=chatzill@216.187.233.42) |
| 18:28.56 | aydiosmio | http://www.pastebin.ca/445077 |
| 18:28.57 | lude | should be.. |
| 18:29.01 | Mercestes | PupenoR, is the demo transcoded in a format your not supporting? |
| 18:29.20 | PupenoR | Mercestes: I am not sure, since I haven't really changed anything in the install. |
| 18:29.22 | aydiosmio | registered with sip.voipstream.co.za, but no response to INVITES |
| 18:29.27 | aydiosmio | looks like a network problem, now? |
| 18:29.29 | aydiosmio | no? |
| 18:29.29 | Mercestes | PupenoR, What codecs are you allowing. |
| 18:30.43 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-0e3afb0ed4733256) |
| 18:30.48 | PupenoR | Mercestes: let me check. |
| 18:31.13 | Mercestes | aydiosmio, Looks like it. It' snot getting respnoses back from 41.204.198.76 |
| 18:32.28 | aydiosmio | yeah... but if it registered, what the hell could the issue be? |
| 18:32.44 | aydiosmio | same ip:port |
| 18:33.08 | PupenoR | Mercestes: I've re-make samples, so I have the standard config for 1.2.17. |
| 18:35.09 | PupenoR | Afret adding "allow=all" to the sip account I get: Unable to find a codec translation path from g723 to ulaw. Now I am getting somewhere. |
| 18:36.21 | Mercestes | PupenoR, don't allow all. Disallow=all then allow=ulaw. |
| 18:36.28 | Mercestes | then try again |
| 18:37.09 | aydiosmio | * doesn't support g723, so if one of your enpoints requires it, you're SOL |
| 18:38.35 | PupenoR | Mercestes: still nothing. |
| 18:38.39 | Mercestes | aydiosmio, The ability to register does not indicate an ability to pass voice or any other packets on any other ports. |
| 18:38.55 | Mercestes | PupenoR, Then see aydiosmio's comment. Where are you getting g723 from? |
| 18:39.36 | PupenoR | The device is a Snom 360, everything worked yesterday, or the day before yesterday. |
| 18:40.10 | Mercestes | PupenoR, That changes everything. |
| 18:40.21 | PupenoR | Mercestes: what changes everything? |
| 18:40.59 | PupenoR | The codec preference of this Snom 360 for this account is G.711u, G.711a, etc. |
| 18:41.10 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 18:41.52 | Cybertoy | hi .. does anyone know exactly what patent Verizon claims to have? Is it really on SIP? |
| 18:42.24 | GreyFoxx | Where are you seeing that ? |
| 18:42.46 | Cybertoy | I heard that but can't really read it anywhere. |
| 18:42.51 | _VoiceMeUp_Com | anyway to simply DROP any subscribe packets ? |
| 18:43.07 | _VoiceMeUp_Com | before one does DOS with that |
| 18:43.18 | Cybertoy | but people ask if they're going after Sunrocket, Packet8 and Broadvoice next ... which would imply it's SIP they're talking about. |
| 18:43.27 | Cybertoy | but I really have no clue |
| 18:43.51 | _VoiceMeUp_Com | ahah |
| 18:43.53 | Hmmhesays | anyone else having trouble matching *72 in the dialplan? |
| 18:43.53 | _VoiceMeUp_Com | verizon no |
| 18:44.00 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 18:44.02 | _VoiceMeUp_Com | i think it was on soemthign else |
| 18:44.14 | _VoiceMeUp_Com | like ip based messaging.. so voice falls into it |
| 18:44.45 | _VoiceMeUp_Com | its like claiming the web.. coz you made http |
| 18:44.50 | _VoiceMeUp_Com | or sock for the matter |
| 18:45.12 | PupenoR | Now I get "chan_sip.c:1415 __sip_ack: Stopping retransmission on '3c4753a4d467-li9s5iv5swej@snom360-000413230550' of Response 2: Match Found". |
| 18:45.34 | _VoiceMeUp_Com | i think that laws exist in these cases.. its like.. if i patend H2O .. can i get a cut on all people making dist water ? |
| 18:45.43 | _VoiceMeUp_Com | since we can now patent human genes.. why not |
| 18:45.57 | blitzrage | patents are dumb |
| 18:46.02 | _VoiceMeUp_Com | better yet.. ill get a patent on fecondation |
| 18:46.03 | _VoiceMeUp_Com | ;) |
| 18:46.09 | _VoiceMeUp_Com | then ill get a piece of your ass |
| 18:46.13 | _VoiceMeUp_Com | literally |
| 18:46.15 | Cybertoy | how about a patent on breathing |
| 18:46.24 | Hmmhesays | is asterisk 1.2 able to handle extensions starting with *? |
| 18:46.25 | _VoiceMeUp_Com | well .not breathing that a function.. |
| 18:46.32 | _VoiceMeUp_Com | but tranfomraing oxygen into xx |
| 18:46.36 | PupenoR | Cybertoy: it probably falls into "business process". |
| 18:46.44 | aydiosmio | _VoiceMeUp_Com: because DNA are molecules that can be unique, water is what we'd call "diluted by use" |
| 18:46.48 | aydiosmio | nyuk nyuk |
| 18:46.54 | _VoiceMeUp_Com | still.. microsfot ocudnt get awya with it |
| 18:47.00 | _VoiceMeUp_Com | hmm |
| 18:47.03 | _VoiceMeUp_Com | aydiosmio, ;) |
| 18:47.22 | Hmmhesays | anyone else have this problem? |
| 18:47.30 | aydiosmio | patents are a method of protecting a process, not the end result |
| 18:48.14 | aydiosmio | yikes, this is a little beyond the scope of this channel. |
| 18:48.25 | Cybertoy | yeah ... I'm sorry |
| 18:48.26 | Hmmhesays | do I have to prefix something to the extension when that extension starts with *? |
| 18:49.41 | aydiosmio | Cybertoy: patent on breathing? see the many ventilator and SCUBA patents:) |
| 18:51.07 | PupenoR | Hmmhesays: I think not, just a * should be enough. But I am not sure. I'll try scaping it with \ |
| 18:51.29 | Hmmhesays | exten => \*72,1,NoOP(holy crap) |
| 18:51.57 | PupenoR | Does anybody know why I am sudenly not getting any audio? This: http://paste.lisp.org/display/39877 is why I get when trying to run the demo... any hints? |
| 18:53.39 | Hmmhesays | ugh this sucks, asterisk will not match a * |
| 18:53.59 | Strom_M | why are you prefixing the * with a \? |
| 18:54.00 | _VoiceMeUp_Com | yeah welcome to my world |
| 18:54.09 | StarSong | But its asterisk |
| 18:54.15 | _VoiceMeUp_Com | had to map all featuees to 4 digits |
| 18:54.29 | PupenoR | Strom_M: scroll up. |
| 18:54.34 | Strom_M | asterisk will quite definitely match * |
| 18:54.36 | Hmmhesays | Strom_M: i'm not, this is what I have *72,1,NoOP(Match Damnit) |
| 18:54.48 | Hmmhesays | Strom_M 1.2 is definately not |
| 18:54.54 | Lavmol | Anyone know what this means "dialparties.agi: dbset CALLTRACE/201 to 113" |
| 18:54.54 | *** part/#asterisk StarSong (n=illusion@200.68.73.133) |
| 18:55.01 | Strom_M | Hmmhesays: is it in the right context? |
| 18:55.06 | Strom_M | pastebin extensions.conf |
| 18:55.06 | PupenoR | Hmmhesays: no need for scaping. |
| 18:55.07 | Hmmhesays | yep |
| 18:55.22 | Strom_M | what kind of telephone set are you using? |
| 18:55.36 | Hmmhesays | when I remove the * and just have 72,1,NoOP(Match damnit) it matches |
| 18:55.44 | Hmmhesays | ekiga |
| 18:55.44 | Strom_M | what kind of telephone set are you using? |
| 18:55.50 | _VoiceMeUp_Com | maybe phne has that feature i its own options and not even sending to asterisk |
| 18:55.52 | aydiosmio | ek? |
| 18:55.56 | aydiosmio | eki? |
| 18:56.03 | Strom_M | well perhaps ekiga is intercepting *72 and doing its own thing with it |
| 18:56.14 | aydiosmio | most likely |
| 18:56.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:57.38 | Hmmhesays | it is not |
| 18:57.46 | Hmmhesays | the invite includes my *72 |
| 18:57.49 | Mercestes | rofl |
| 18:57.53 | Mercestes | asterisk cannot match an asterisk |
| 18:58.04 | Hmmhesays | it sure seems that way in 1.2.17 |
| 18:58.25 | Strom_M | i suspect operator error |
| 18:58.32 | aydiosmio | somethign fishy |
| 18:58.37 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 18:58.49 | Supaplex | [^0-9#] ? |
| 18:59.21 | aydiosmio | bah, I should be doing actual work. |
| 18:59.27 | Supaplex | slacker! |
| 18:59.38 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:59.59 | Hmmhesays | exten => *72,1,NoOP(Match damnit) is what I have in the dialplan |
| 19:00.12 | *** join/#asterisk angler_ (n=angler@12-218-74-162.client.mchsi.com) |
| 19:00.19 | PupenoR | I see RTP being send from the phone to the server, but none from the server to the phone. |
| 19:00.43 | Hmmhesays | put extension *72 in the dialplan and try it |
| 19:01.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:02.13 | _VoiceMeUp_Com | hmm says |
| 19:02.30 | _VoiceMeUp_Com | or map 792 to *72 |
| 19:02.34 | _VoiceMeUp_Com | or whatever |
| 19:03.43 | Hmmhesays | ok i'm an idiot today |
| 19:03.46 | Hmmhesays | must be the taxes |
| 19:04.07 | Supaplex | smart people finished those atleast a week ago |
| 19:04.49 | [TK]D-Fender | Hmmhesays: Fix your phone's dialplan |
| 19:05.16 | [TK]D-Fender | Hmmhesays: and ensure that you have "pedantic=yes" under [general] in sip.conf |
| 19:06.41 | PupenoR | Here I have even more output: http://paste.lisp.org/display/39877#1 , I can't even find an error there :( |
| 19:10.34 | lesouvage | Is there a way to end the other leg of the call in the h, extension when both legs in a meetme room. I tried exten => h,20,System(asterisk -rx meetme kick ${KLANT:1:9} all) and a hangup after the MeetMe(123134) line of the leg that didn't hang up but this doesn't seem to work. The channel that hasn't hang up keeps active. |
| 19:10.36 | PupenoR | Nobody has a clue about that? |
| 19:12.07 | lude | dangit |
| 19:12.15 | lude | why are people telling me their 7960's work fine in 1.4.2 |
| 19:12.24 | lude | i can't get it working for the life of me |
| 19:13.12 | Hmmhesays | GotoIf($[${LEN(${cfCheck})}!=0] ok is that valid |
| 19:14.39 | SplasPood | anyone ever had an issue where app_voicemail will activate PIN changes in memory, but not write them out to voicemail.conf ? I see nothing in my logs... |
| 19:16.49 | lude | i know mine won't write changes if the permissions on voicemail.conf are screwy |
| 19:16.56 | lude | dunno if that helps you |
| 19:18.21 | drega | I'm seeing some strange issues with phones in /var/log/asterisk/messages that i'm wondering if anyone has any advice on.. |
| 19:18.31 | drega | Apr 17 16:47:06 NOTICE[5448] chan_sip.c: Peer '4011' is now UNREACHABLE! Last qualify: 24 |
| 19:18.31 | drega | Apr 17 16:47:11 NOTICE[5448] chan_sip.c: Peer '4039' is now REACHABLE! (122ms / 2000ms) |
| 19:18.41 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 19:19.22 | drega | the extensions are split onto two POE switches directly connected to asterisk. No additional traffic is going over the voice network. |
| 19:19.32 | drega | it doesn't happen all the time but does happen randomly. |
| 19:19.41 | Mercestes | drega: seen that alot recently. mostly due to NAT/firewalls/cheap hardware/other networking issues. |
| 19:19.50 | AndrewGearhart | anybody interested in discussing the vonage/patents/voip issue (it was discussed here earlier) I encourage you to join #voip-future |
| 19:19.55 | lesouvage | drega:I need a new router and when I see that I restart my router. That normally fix the problem. |
| 19:20.21 | aydiosmio | drega: is your NAT connection timeout set equal or lower than the qualify period? |
| 19:20.34 | drega | Mercestes: & lesouvage hardware is all totally new. I'm using Cisco Catalyst switches that are brand new |
| 19:20.42 | drega | new polycoms. |
| 19:20.44 | aydiosmio | if there's even NAT involved... |
| 19:20.49 | drega | no nat. |
| 19:21.27 | aydiosmio | have you run a packet capture and exacmined the OPTIONs for the qualify during these periods? |
| 19:21.29 | *** part/#asterisk NoTurbo (n=cornholi@ip5457284a.direct-adsl.nl) |
| 19:21.45 | aydiosmio | perhaps they're being dropped randomly |
| 19:21.46 | drega | nope haven't gone that far. |
| 19:22.08 | _VoiceMeUp_Com | anyone have vitelity address ? |
| 19:22.15 | drega | I'm just befuddled as to why.. |
| 19:22.24 | Mercestes | drega: Are these cisco phones? |
| 19:22.24 | aydiosmio | _VoiceMeUp_Com: define vitelity address |
| 19:22.31 | drega | polycom 430's |
| 19:22.41 | drega | brand spanking new as well |
| 19:23.01 | drega | what issues could this cause for call quality.. |
| 19:23.19 | Mercestes | drega: Hrm. Watch the nat tables in the cisco router and see if the packets are incrementing nat ports at random |
| 19:23.34 | drega | atm I've got 40 extensions and quality varies but getting some complaints of delay and some echo. |
| 19:23.41 | drega | could this be related? |
| 19:23.55 | bkruse | use ulaw, not gsm |
| 19:23.56 | bkruse | :D |
| 19:23.57 | Mercestes | drega: LOL. you have a network loop. |
| 19:24.08 | Mercestes | drega, Someone plugged the switch back into itself. SIP doesn't echo |
| 19:24.14 | bkruse | Mercestes: true that. |
| 19:24.20 | bkruse | i can just picture it now...lol |
| 19:24.23 | _VoiceMeUp_Com | need to port # out |
| 19:24.29 | *** join/#asterisk wundaboy (n=look@adsl-68-122-41-10.dsl.pltn13.pacbell.net) |
| 19:24.32 | bkruse | does the LAN plug go into WAN? |
| 19:24.45 | lude | does anyone have a working example for cisco 7960 in 1.4.2? the exact same config from 1.2 doesn't work |
| 19:24.50 | wundaboy | i know this is not very "asterisk", but does anyone have experience with the polycom "digitmap"? |
| 19:24.59 | drega | Mercestes: there is no router involved. It's set out like server eth1 => switch1 linked to switch2 over to patchpanel over to phoens |
| 19:25.08 | bkruse | lol |
| 19:25.12 | bkruse | thats just as bad, no router? |
| 19:25.13 | bkruse | eww |
| 19:26.10 | Mercestes | drega, The fact that you believe that there is not a router involved puts your network knowledge in question for me. |
| 19:26.23 | Mercestes | wundaboy, Yes. |
| 19:26.30 | bkruse | Mercestes: well, you dont NEED a router, but your right, a router is upstream somewhere |
| 19:26.34 | bkruse | but the fact they dont have one.....herm |
| 19:26.42 | bkruse | switch -> patch cable -> switch -> computer |
| 19:26.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:26.49 | Mercestes | bkruse, doing kernel level routing with IP tables is using your asterisk box as a router. |
| 19:26.50 | wundaboy | Mercestes: can I pm you and ask you some questiosn? |
| 19:27.01 | Mercestes | wundaboy, Sure! Thanks for asking. |
| 19:27.07 | bkruse | Mercestes: no, i would not use my asterisk box as a router, but ya, youc ould |
| 19:27.09 | Mercestes | bkruse, *something* functinos as a router somewhere or the whole system would fail |
| 19:27.13 | bkruse | just buy something cheap and get it working |
| 19:27.15 | drega | ok I'm not claiming I'm the network god here. Thats why i'm questioning what's going on. |
| 19:27.17 | bkruse | Mercestes: lol, kinda |
| 19:27.18 | bkruse | its like multicast + dhcp but with actual networking packets |
| 19:27.49 | Dr-Linux | anybody is using dynamic conferencing? |
| 19:27.50 | drega | eth0 is connected to data side where there is a router and all other swithes eth1 is handing out dhcp to phones connected to switches |
| 19:28.08 | Dr-Linux | i wanna limit the PIN number digits |
| 19:28.37 | bkruse | now he says theres a router Mercestes |
| 19:28.38 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 19:28.50 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 19:28.51 | SplasPood | lude: that was my thought, so I made em 666 just to be sure... I'm wondering if it has something to do with the fact that i'm using #include |
| 19:28.53 | drega | bkruse: not on the voice side of things |
| 19:29.13 | bkruse | drega: why in the world not? |
| 19:29.22 | SplasPood | and if so that puts a major crimp in my plans.. |
| 19:29.51 | drega | eth1 => switch => patch panel => phones eth1 is handling dhcp ... |
| 19:30.00 | drega | was just trying to segment things out as much as possible. |
| 19:30.57 | lude | using an #include in the voicemail config? |
| 19:31.43 | bkruse | SplasPood: dialplan reload and see if you get errors |
| 19:32.00 | bkruse | router = segmentation, wouldnt you agree Mercestes? |
| 19:32.08 | bkruse | it routes, breaks up broadcast domains, its quite the bomb |
| 19:32.19 | lude | gaaah why doesn't this work |
| 19:32.23 | *** part/#asterisk icel (n=dan@65.200.26.80) |
| 19:32.24 | lude | what the hell changed |
| 19:33.15 | Mercestes | Can you have a polycom automatically prepend digits to a dialstring before it outputs it to Asterisk??? |
| 19:33.34 | anonymouz666 | Apr 17 16:33:05 WARNING[30726]: pbx.c:3677 ast_context_create: Tried to register context 'dynamic_states', already in use |
| 19:33.58 | Mercestes | drega: Let me state again, sip does *NOT* echo.... |
| 19:34.09 | [TK]D-Fender | Mercestes: in 2.1.X yes |
| 19:34.13 | Mercestes | drega, it's not possible, the only way to get "echo" on SIP is to broadcast the exact same packets twice. |
| 19:34.22 | lude | Mercestes: you can probably set a variable for the polycoms in the sip confiig, and then check for that variable in your dialplan |
| 19:34.23 | Mercestes | drega, what you have is a network loop. |
| 19:34.36 | Mercestes | [TK]D-Fender, Can you hint me on how to do it? |
| 19:34.43 | mrdigital | whats the feature code to put someone on hold |
| 19:34.44 | bkruse | Mercestes: well, asterisk does broadcast twice, because its UDP |
| 19:35.01 | drega | Mercestes: ok you think that is also making phones go Unreachable and back again ? |
| 19:35.14 | Mercestes | drega, yes. Absolutely. |
| 19:36.07 | drega | hurm.. great well that gives me something to look at. |
| 19:36.15 | lude | drega: check the load on your switches, how big is the vlan the phones are in? |
| 19:36.22 | drega | there is no vlan. |
| 19:36.31 | drega | voice for phones is totally stegmented |
| 19:36.51 | lude | it's one flat layer 2 network? |
| 19:36.53 | lude | how big is it? |
| 19:36.54 | drega | I didn't want to mess with QOS or vlans |
| 19:37.09 | lude | and how bad is the load on your switches? |
| 19:37.14 | drega | na just for the phones on the other side of things there is a data network. |
| 19:37.32 | [TK]D-Fender | Mercestes: www.polycom.com. There is an independant addendum to the admin guide concerning it. |
| 19:37.45 | [TK]D-Fender | Mercestes: I have not attempted it personally. |
| 19:37.48 | Mercestes | Sweet. Thanks. |
| 19:38.03 | drega | I've got two brand new cisco catalyst switches with 24 people on it using G.711 with giga uplink |
| 19:38.17 | drega | and that is the ONLY traffic on them |
| 19:38.33 | drega | correct me if I'm wrong but I don't think the switches are overloaded |
| 19:39.29 | lee_is_me | Faxing: I'm following a thread on the mail list concerning faxing and am not quite sure what the conclusion is. Can I hook up a standard fax machine to an FXO port without problems? |
| 19:40.33 | lude | drega: might not have cef enabled or who knows what |
| 19:40.35 | Mercestes | lee_is_me, *shakes magic 8 ball* All indications point to no. |
| 19:40.48 | lee_is_me | lol |
| 19:40.49 | lude | how long is the subnet the phones are in |
| 19:40.54 | bkruse | Mercestes: good answer |
| 19:41.13 | PupenoR | re-starting my computers solved the Asterisk problem. |
| 19:41.21 | lee_is_me | so, the best bet is to avoid bringing the fax line through asterisk at all? |
| 19:41.34 | PupenoR | my computer is where the Asterisk is running... |
| 19:41.52 | aydiosmio | I hope it's running on a computer |
| 19:42.32 | bkruse | aydiosmio: :X |
| 19:42.35 | PupenoR | aydiosmio: it could be running in a server... if restarting a workstation solves a problem in a server... then I would start turning the lights on-off every time there's a problem in any computer ;) |
| 19:43.05 | aydiosmio | a server is a computer. bah |
| 19:43.53 | PupenoR | bah! |
| 19:43.59 | aydiosmio | "When I use a word," Humpty Dumpty said in rather a scornful tone. "It means just what I choose it to mean - neither more or less." |
| 19:44.03 | Mercestes | lee_is_me, The best bet is yes. It will still work that way and I've had success, but never "problem free." |
| 19:44.47 | drega | lude: data network is using 10.1.80.* ip range and voice is on 192.168.10.* range again totally segmented. |
| 19:44.57 | lee_is_me | Mercestes: Thanks. Customer has no problem doing it either way. Just need to be able to fax. |
| 19:45.27 | lude | drega: maybe duplex mismatches? |
| 19:45.42 | lude | issue is kinda broad |
| 19:45.45 | drega | now thats a good idea |
| 19:46.52 | drega | mm. well I'm just wondering how the phones going unreachable and reachable again could be effecting call quality. |
| 19:47.11 | drega | in watching the logs and monitoring users some of the extensions do that while people are on a call |
| 19:47.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:47.29 | Mercestes | drega, I would say that the unreachable/reachable status is a symptom, not a cause. |
| 19:48.16 | drega | ok so I'm digging the duplexing idea and will defo check for a loop |
| 19:48.24 | Mercestes | Good luck |
| 19:49.05 | lude | argh |
| 19:49.08 | lude | this is so stressful |
| 19:49.24 | drega | but I've got this down to the simplest setup possible so i can't imagine there is a loop. I mean this could be the most holistic voice setup possible.. I'm concerned about the talk about routing though. |
| 19:49.45 | mrdigital | can someone point me to a page on faxing with asterisk |
| 19:49.52 | lude | drega: i don't think there's a network loop |
| 19:50.03 | lude | that's really only caused by routers or stale macs or whatever |
| 19:50.16 | lude | span the port on a phone, and test that way for duped packets |
| 19:50.18 | *** part/#asterisk deeperror (n=deeperro@mail.banctel.com) |
| 19:50.22 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 19:50.27 | lee_is_me | Mercestes: your 8 ball is needed again... |
| 19:51.07 | irule | is it possible to tal the ata to not rng 0.5 seconds every time it registers? |
| 19:51.29 | drega | ya thanks lude Mercestes bkruse I'll give it all a look over in the morning. |
| 19:51.43 | bkruse | kk, awesome |
| 19:51.45 | bkruse | keep us posted |
| 19:51.59 | *** join/#asterisk Waverly360 (n=irc@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 19:52.02 | drega | will do I'll check back in tomorrow evening and tell the tale |
| 19:52.19 | lude | anyone active that has a 7960 working with asterisk 1.4 ? |
| 19:55.10 | Mercestes | lee_is_me, alright... |
| 19:55.29 | aydiosmio | drega: it shouldn't affect in progress calls |
| 19:56.03 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 20:01.34 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 20:03.09 | onecentld | having problems with dtmf with ver 1.4.2 --- anyone have any ideas --- where are using G729 and rfc2833 |
| 20:03.39 | wundaboy | i need the sip 2.1.1 for polycom ip501. it is not on www.freedomephones.net and I cannot download it from polycom... |
| 20:03.46 | wundaboy | does anyone have access to it? |
| 20:04.06 | [TK]D-Fender | wundaboy: Go ask your reseller. They are supposed to provide it to you |
| 20:04.50 | wundaboy | [TK]D-Fender i bought them on ebay |
| 20:05.13 | Mercestes | wundaboy, Well, that represents a problem. |
| 20:05.54 | wundaboy | thanks |
| 20:10.14 | onecentld | having problems with dtmf with ver 1.4.2 --- loosing dtmf when dialing into the asterisk box --- anyone have any ideas --- where are using G729 and rfc2833 |
| 20:10.57 | *** join/#asterisk ixela (i=ixela@nat/digium/x-668b47c4076e2234) |
| 20:13.02 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 20:13.10 | linagee | does anyone know of a way to get memorable 800 numbers? |
| 20:13.28 | linagee | 800-4AS-TERISK, etc? |
| 20:14.23 | drega | ah linagee I've seen a website that would generate vanity stuff from numbers.. |
| 20:14.34 | drega | give google a go with something like that |
| 20:14.35 | linagee | phonepeople.com? |
| 20:15.12 | linagee | drega: also, i'm using voicepulse, why is toll free 5 cents/min? is that expensive? |
| 20:15.36 | drega | are you looking to purchase an 800 or just looking for a vanity number then go purchase that one. |
| 20:15.57 | linagee | hrm |
| 20:15.59 | drega | ah linagee I'm not sure what country your in but "toll free" means the call is free |
| 20:16.16 | linagee | drega: for the asterisk side. ;) |
| 20:16.29 | drega | so anything more than 0 is too expensive..with the caveat that again I don't know where your calling from |
| 20:16.37 | linagee | USA to USA |
| 20:18.24 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 20:19.18 | linagee | run! it's the thoughtpolice! |
| 20:19.39 | aydiosmio | a.k.a. witcops |
| 20:21.26 | Lavmol | Anyone know where I can find some documentation on the difference between the sip_additional.conf file and the extension configuration in freepbx??? |
| 20:21.33 | onecentld | having problems with dtmf with ver 1.4.2 --- loosing dtmf when dialing into the asterisk box --- anyone have any ideas --- where are using G729 and rfc2833 |
| 20:21.36 | *** join/#asterisk wyle-e-kyote (n=Doug@gw.hypercube-llc.com) |
| 20:22.06 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 20:24.58 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:25.30 | defsdoor | Anyone use aastra phones or know if you can make it autodial when a 3 digit number beginning with 2 is dialled |
| 20:27.58 | Waverly360 | Hey all, I'm looking at some voip gateways from Mediatrix, AudioCodes, and Multitech. Can any of you share your thoughts on any of these companies and their products? |
| 20:28.39 | *** join/#asterisk ncampion (i=chatzill@nat/ibm/x-eed9061c63a0a537) |
| 20:36.09 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 20:37.38 | Hmmhesays | i wish _X. matched a *72 |
| 20:47.00 | Mercestes | <PROTECTED> |
| 20:47.00 | BSD_Tech | ok the .configure script sees zaptel.h and it passes the requirement |
| 20:47.00 | BSD_Tech | and I build asterisk |
| 20:47.00 | BSD_Tech | but after a reboot zaptel loads and ztdummy loads |
| 20:47.00 | BSD_Tech | but when I start astersk and do a zap show channels |
| 20:47.01 | BSD_Tech | Connected to Asterisk SVN-branch-1.4-r61666M currently running on mythtv (pid = 661) |
| 20:47.01 | BSD_Tech | mythtv*CLI> zap show channels |
| 20:47.01 | BSD_Tech | No such command 'zap show' (type 'help' for help) |
| 20:47.01 | BSD_Tech | it seems not to see zaptel |
| 20:47.01 | BSD_Tech | but zaptel is loaded and ztdummy is loaded |
| 20:47.53 | onecentld | having problems with dtmf with ver 1.4.2 --- loosing dtmf when dialing into the asterisk box --- anyone have any ideas --- where are using G729 and rfc2833 |
| 20:48.08 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:51.43 | reber | is madplay fine ? |
| 20:51.45 | Mercestes | onecentld, answre first. |
| 20:51.51 | onecentld | help |
| 20:52.03 | Mercestes | they'll tell you to just use native. |
| 20:52.15 | Mercestes | onecentld, Call an Answer(). |
| 20:52.21 | Hmmhesays | apparently asterisk comparisons need a space between the operator and the expressiosn |
| 20:53.33 | Mercestes | Hmmhesays, for example??? |
| 20:56.10 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 20:56.58 | onecentld | having problems with dtmf with ver 1.4.2 --- loosing dtmf when dialing into the asterisk box - IVR picks up the calls - asks to entry the phone number to connect to - at this point it doesn't catch all the digits --- anyone have any ideas --- where are using G729 and rfc2833 |
| 20:57.23 | Hmmhesays | exten => *72,n,GotoIf($[${LEN(${cfCheck})} != 0]?${EXTEN},$[${PRIORITY} + 1]:s-ERROR,1) |
| 20:57.34 | Hmmhesays | that works where exten => *72,n,GotoIf($[${LEN(${cfCheck})}!=0]?${EXTEN},$[${PRIORITY} + 1]:s-ERROR,1) |
| 20:57.35 | Hmmhesays | doesn't |
| 20:58.21 | Mercestes | ... |
| 20:58.38 | lude | i'm gonna break this stupid phone |
| 20:58.52 | Hmmhesays | you see where the difference is Mercestes? |
| 20:58.53 | lude | i can't imagine i'm the only one seeing this weirdness |
| 20:58.58 | lude | but google is no help |
| 20:59.16 | Mercestes | Hmmhesays, Yea, the #!=# with spaces. |
| 20:59.28 | onecentld | Mercestes - i dont know |
| 20:59.29 | Hmmhesays | i didn't know the spaces were required |
| 20:59.40 | Mercestes | me neither |
| 20:59.51 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:01.31 | *** join/#asterisk sharp (n=sharp@pool-71-242-110-119.phlapa.east.verizon.net) |
| 21:02.05 | *** join/#asterisk NirS (n=Nir@84.94.210.27.cable.012.net.il) |
| 21:02.33 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 21:03.08 | onecentld | Mercestes - i am new to asterisk - i don't know where to look |
| 21:03.31 | froguz | russellb, is it true you're working on upload capability for asterisk web server? |
| 21:03.32 | onecentld | can you point me into the right direction please |
| 21:04.34 | Mercestes | Sure! |
| 21:04.36 | Mercestes | ~book |
| 21:04.37 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:04.43 | Mercestes | onecentld: There you go. |
| 21:06.28 | russellb | froguz: already done ... |
| 21:07.16 | russellb | froguz: it's in trunk |
| 21:07.27 | froguz | cool! |
| 21:07.57 | froguz | thanks a lot |
| 21:08.11 | russellb | no problem :) |
| 21:08.51 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 21:10.09 | wyle-e-kyote | anyone done any work integrating SBE's Channelized DS3 cards into Asterisk? (or any other DS3 level connectivity on the TDM side?) I've been googling and besides finding places where it seems to fade in and out of interest there doesn't appear to be anything solid. |
| 21:12.02 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 21:14.50 | zuez | curious, wanted to learn asterisk/voip at home .. besides having an IP Phone and an asterisk install, what would I need to access a PTNS so I can be reached externally, an account with vonage or something similar? |
| 21:15.41 | Strom_M | what the hell is "a PTNS"? |
| 21:15.53 | zuez | public telephone network switch |
| 21:15.59 | [TK]D-Fender | Strom_M, A tyop obviously! ;) |
| 21:16.03 | Strom_M | uh, that's not the acronym :) |
| 21:16.10 | Strom_M | it's "the PSTN" |
| 21:16.16 | zuez | errr PSTN, apologies |
| 21:16.16 | Strom_M | the public switched telephone network |
| 21:16.55 | errr | is there a way to test how my asterisk server will hold up with out using that app that is for windows? |
| 21:18.24 | Strom_M | zuez: you want either an account with an ITSP, or you want some sort of telephone line interface card |
| 21:19.08 | zuez | Strom_M: okay great, the whole purpose was so I can learn asterisk and all. I'd pay for an ITSP, they should be cheap. I can't imagine more than $10-15/mo. |
| 21:19.32 | zuez | Have an extra box at home I can pop asterisk onto, then I want to eventually put some office phones on VoIP and switch us away from legacy PBX |
| 21:19.40 | *** join/#asterisk tornad (n=Regis@84.6.23.244) |
| 21:20.36 | Strom_M | the ITSP shouldn't charge you monthly |
| 21:20.46 | Strom_M | get one that charges you per-minute |
| 21:21.21 | zuez | Strom_M: great, then I can just DHCP an IP to the phone from my ghetto lan switch at home and be on my way to breaking things. :-) |
| 21:21.50 | Strom_M | terrific |
| 21:22.02 | zuez | Thanks for the insight, it's appreciated. |
| 21:22.23 | *** join/#asterisk opioid (n=karl@207.191.91.242) |
| 21:22.58 | opioid | hello all! i've been having loads of fun with this stuff!! |
| 21:23.11 | opioid | anyone else used the polycom 501 phones? |
| 21:23.27 | Strom_M | opioid: no, you are the only person in the history of the universe to use that phone |
| 21:24.25 | [TK]D-Fender | opioid, ecept for the majority of us in here of course... |
| 21:24.57 | Strom_M | damnit, [TK]D-Fender, you're ruining my hyperbole-for-comic-effect :D |
| 21:25.30 | [TK]D-Fender | Strom_M, But I'm extending it to the "duh, AND we have you outnumbered!" level ;) |
| 21:25.49 | Strom_M | heh |
| 21:29.01 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:29.12 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-51-85.dsl.irvnca.pacbell.net) |
| 21:29.19 | irule | what is the propper method of reloading 1.4? just reload gives me an error message |
| 21:29.20 | BSD_Tech | back |
| 21:29.26 | BSD_Tech | any ideas |
| 21:29.58 | blitzrage | irule: 'reload' |
| 21:30.43 | blitzrage | errr: I think you want to use SIPp |
| 21:30.56 | blitzrage | [TK]D-Fender: I don't want to know your name! |
| 21:31.01 | blitzrage | I just want... |
| 21:31.04 | [TK]D-Fender | blitzrage, I just want.... |
| 21:31.09 | blitzrage | ! ! ! |
| 21:31.09 | [TK]D-Fender | blitzrage, ! ! ! |
| 21:31.16 | blitzrage | that never gets old |
| 21:31.21 | [TK]D-Fender | NEVER |
| 21:31.30 | *** join/#asterisk alexns (n=alex@static-acs-24-154-114-15.zoominternet.net) |
| 21:32.00 | alexns | need some help with polycom sidetone over tdm400 |
| 21:33.27 | errr | blitzrage: I tried that one and it brought my server to a grinding hault |
| 21:33.48 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 21:33.54 | blitzrage | errr: then you tried to setup too many calls |
| 21:34.19 | blitzrage | remember that any console output is going to cause issues at high call volume |
| 21:34.33 | blitzrage | so make sure you set verbose 0, set debug 0, and comment out any logging to the console |
| 21:34.52 | blitzrage | (in logger.conf if it's not obvious that's what I meant) |
| 21:35.01 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 21:37.12 | errr | ok thanks |
| 21:39.22 | BSD_Tech | I need zaptel |
| 21:39.32 | BSD_Tech | grrr |
| 21:39.42 | BSD_Tech | asterisk is punked |
| 21:39.55 | BSD_Tech | Asterisk SVN-branch-1.4-r61666M |
| 21:41.31 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 21:41.53 | TheCompWiz | can anyone tell me if rtp packets use the tos flags specified in sip.conf? or do I need to specify something else in rtp.conf? |
| 21:44.58 | drega | clear |
| 21:45.02 | drega | grrr. |
| 21:45.05 | wunderkin | *charge* |
| 21:45.10 | wunderkin | *splat* |
| 21:45.21 | Mercestes | He's dead, Jim |
| 21:45.24 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 21:46.13 | slmnhq | Are there any reports that talk about Asterisk performance given different number of channels and different levels of call volume? |
| 21:46.19 | slmnhq | Are there any reports that talk about Asterisk performance given different number of channels and different levels of call volume? |
| 21:46.39 | alexns | need some help with polycom sidetone over tdm400 |
| 21:46.39 | *** part/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 21:46.53 | Qwell[laptop] | way to wait 8 seconds |
| 21:47.09 | Mercestes | rofl |
| 21:47.25 | opioid | Strom_M: not very nice! whats up with the volume bullshit on this phone? will a firmware update fix that? the "forgetful" handset volume memory.. |
| 21:47.35 | Mercestes | Come on, this is the 21st century! Answers should be instantaneous. |
| 21:47.54 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 21:47.58 | Mercestes | opiod: Did you try volume.persist.headset=1. |
| 21:48.07 | Mercestes | opiod: Speaking of bullshit, did you bother reading the admin guide? |
| 21:48.22 | [TK]D-Fender | Mercestes, do you even have to ask? |
| 21:48.33 | Mercestes | [TK]D-Fender, ...sorta, yea :D it makes me feel better. |
| 21:48.39 | [TK]D-Fender | opioid, how hard did you look in the web interface for this setting? |
| 21:48.43 | Mercestes | it's childish, I know. semi-trollish. |
| 21:48.52 | Mercestes | web interface? |
| 21:48.55 | Mercestes | O.O |
| 21:52.14 | wunderkin | pointy clicky.. woohoo pretty buttons |
| 21:52.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:52.25 | Waverly360 | Any of you guys had any experience with AudioCodes products? |
| 21:53.11 | *** join/#asterisk codefreeze (i=steve_mu@nat/digium/x-fe26cf1a58ccc21e) |
| 21:53.29 | opioid | [TK]D-Fender: i thought i looked pretty hard. i also checked the manual, which says that the volume resets after each call to conform to standards.. |
| 21:53.45 | wunderkin | but.. |
| 21:55.00 | [TK]D-Fender | wunderkin, "Harker to 'Tooth Fairie', the troll has been baited!" |
| 21:55.12 | Mercestes | opiod: Did you check under the line settings in the web interface? |
| 21:55.16 | wunderkin | geronimo |
| 21:57.09 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:59.11 | [TK]D-Fender | wunderkin, I really outdid myself on that... so where's my Academy Award?! |
| 21:59.51 | wyle-e-kyote | its on its way, but the intertubs are full right now with all the pointy clicky stuff going on. |
| 22:01.36 | [TK]D-Fender | wunderkin, Thhaaas not a knife! |
| 22:12.30 | *** join/#asterisk greyarea (n=light@ip68-109-167-150.ph.ph.cox.net) |
| 22:12.37 | pfn | damn, I thought I had g729 working... |
| 22:20.11 | *** join/#asterisk mivck (n=mv@134.42.128.66.PPPoECali.dynamic.telesat.net.co) |
| 22:22.35 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 22:26.23 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-156.sd.sd.cox.net) |
| 22:26.46 | *** join/#asterisk neoalex (n=neoalex@user-0ccengj.cable.mindspring.com) |
| 22:26.52 | *** join/#asterisk Inverted (n=steven@cuervo.unwiredbuyer.com) |
| 22:27.12 | neoalex | hey guys, does anyone have the admin guide for the pap2t |
| 22:29.06 | Inverted | why does record_file timeout after 30 minutes if you've sent it a -1 as a timeout value? |
| 22:32.25 | JunK-Y | Inverted: dunno, never tried to record > 30 minutes :) |
| 22:32.35 | JunK-Y | Inverted: which * ? |
| 22:33.36 | BSD_Tech | this issues baffels me |
| 22:33.42 | *** join/#asterisk ruied (n=ruied@bl7-213-44.dsl.telepac.pt) |
| 22:36.18 | *** join/#asterisk Stridernzl (n=neville@125-239-173-41.jetstream.xtra.co.nz) |
| 22:40.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:44.02 | bkruse | and i agree |
| 22:44.58 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:46.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:46.25 | Inverted | 1.2.13 |
| 22:47.19 | Inverted | I've tested it out 4 times, and it record_file returns with 0 status +-1 second every time |
| 22:47.49 | BSD_Tech | is there a cli in the gui |
| 22:48.40 | bkruse | no |
| 22:49.30 | BSD_Tech | grrrr |
| 22:49.32 | BSD_Tech | ok |
| 22:49.48 | Mercestes | Inverted, I recorded a 2 hour meeting by not passing it a timeout. |
| 22:50.55 | Inverted | Mercestes: hmm alright, let me take out the -1 timeout value |
| 22:51.24 | Mercestes | Record(${timestamp}-9998:gsm) is all I have. |
| 22:51.29 | neoalex | hey guys, does anyone have the admin guide for the pap2t |
| 22:51.56 | *** join/#asterisk Vec (n=Vec@dsl-243-80-127.telkomadsl.co.za) |
| 22:52.29 | greyarea | I think if you google such terms as administration guide<insert model> pdf |
| 22:52.32 | greyarea | you should find it |
| 22:52.46 | greyarea | seriously though, not just saying that to be a dick. |
| 22:52.51 | Mercestes | greyarea, you really think they have it on the google? |
| 22:53.08 | greyarea | I find the spaxxx on there all the time |
| 22:54.00 | greyarea | the google is great for that type of infos ;) |
| 22:56.07 | Mercestes | and porn |
| 22:56.24 | intralanman | google is good for porn? |
| 22:56.29 | Mercestes | sure! |
| 22:56.32 | greyarea | yep |
| 22:56.40 | intralanman | hmmm, i'll have to try that sometime |
| 22:56.40 | greyarea | turn off protection in the settings |
| 22:56.42 | Mercestes | just image search goatscx |
| 22:56.43 | nemski | google is good for anything |
| 22:56.50 | Mercestes | or lemonparty |
| 22:56.51 | nemski | hehe |
| 22:56.57 | greyarea | heh |
| 22:56.59 | Mercestes | or "athens" |
| 22:57.00 | intralanman | or xtube |
| 22:57.01 | nemski | www.lemonparty.com!, best porn site |
| 22:57.05 | intralanman | :) |
| 22:57.06 | Mercestes | ROFL |
| 22:57.15 | Mercestes | bad, Nemski, Bad! |
| 22:57.20 | nemski | :( |
| 22:57.43 | Mercestes | don't click the link, btw...It will change your life....for the worst |
| 23:02.25 | *** join/#asterisk uhb (n=anon@CPE000d3a2bac7d-CM00159a6a31ee.cpe.net.cable.rogers.com) |
| 23:03.25 | Inverted | Mercestes: im using EAGI, you? |
| 23:03.40 | Mercestes | Inverted: strait record. |
| 23:03.48 | Mercestes | Inverted, phpAGI? |
| 23:07.14 | SplasPood | ahh... my voicemail pin changing issue seems to be that asterisk will not write out new pins in #include(d) files from voicemail.conf |
| 23:07.49 | BSD_Tech | svn is broken on asterisk |
| 23:07.51 | BSD_Tech | grrr |
| 23:08.13 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-113-108.cia.com) |
| 23:08.45 | Inverted | Mercestes: http://www.voip-info.org/wiki/view/record+file |
| 23:09.53 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:13.08 | Mercestes | ok |
| 23:13.18 | Mercestes | butI don't use agi. |
| 23:13.23 | Mercestes | I just call Record() |
| 23:16.13 | *** join/#asterisk augustz (n=11@unaffiliated/augustz) |
| 23:17.17 | augustz | i'm getting a 941. |
| 23:17.35 | augustz | it's going to be sitting behind a nat, I do think I can open up ports however to allow reinvites (with stun on 941) |
| 23:17.43 | augustz | am I way off base? |
| 23:18.50 | BSD_Tech | ok anyone using todays cvs |
| 23:20.24 | BSD_Tech | I mean svn |
| 23:23.52 | *** join/#asterisk Shoeb (n=chatzill@64.34.69.9) |
| 23:24.11 | Shoeb | What does it mean when both parties ring, caller and callee, but voices behind heard from both sides is not a possibility? |
| 23:24.23 | *** join/#asterisk logicwrath (n=some@c-68-60-121-112.hsd1.mi.comcast.net) |
| 23:24.26 | Shoeb | I checked the NAT settings, and I made sure both of them are DMZ'd. |
| 23:24.33 | Shoeb | And it still does the same. |
| 23:27.47 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 23:30.59 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 23:31.18 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:32.04 | BSD_Tech | this is bs |
| 23:32.04 | BSD_Tech | where are the devs when you need input |
| 23:32.04 | BSD_Tech | grrrrr |
| 23:36.52 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:39.06 | JT | BSD_Tech: this is not an official tech support venue |
| 23:43.51 | BSD_Tech | well its a asterisk support venue |
| 23:44.58 | BSD_Tech | and this is a major issue |
| 23:46.03 | Mercestes | did you try #asterisk-devs? And what is the issue?? |
| 23:46.21 | BSD_Tech | its been empty the last few times I joined |
| 23:46.42 | tornad | join #asterisk-devs |
| 23:46.59 | tornad | join #asterisk-dev <- with no S sorry |
| 23:47.14 | Mercestes | lol |
| 23:47.16 | Mercestes | might help. :D |
| 23:47.39 | tornad | sometime... :p |
| 23:47.40 | tornad | :) |
| 23:48.12 | Qwell | #asterisk-dev is also not a support venue ;) |
| 23:48.22 | Qwell | in fact, it's far, far less so than here |
| 23:49.00 | Mercestes | Qwell: asterisk is broken in SVN! zOMG! |
| 23:49.02 | Mercestes | havne't you heard? |
| 23:50.09 | *** join/#asterisk Fieldy (i=s58tPSoJ@gentoo/contributor/Fieldy) |
| 23:50.12 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:50.15 | *** join/#asterisk BB|AtWork (n=karl@38.99.18.98) |
| 23:50.18 | logicwrath | i heard they are coming out with a new distro of AsteriskNOW called AsteriskNOOB for me |
| 23:50.19 | JT | BSD_Tech: doesn't matter if it's a major issue, this is not an official digium support venue |
| 23:50.32 | Mercestes | ROFLMAO |
| 23:50.38 | Mercestes | #asteriskn00b. I love it |
| 23:50.57 | BB|AtWork | so i think i've got my t1 + card configured but i get a slow busy signal when i dial the number for it from the outside. Is there any way i can watch something to see if asterisk is even getting the call? |
| 23:51.29 | JT | BB|AtWork: is the span even up, can you make outgoing calls? |
| 23:51.41 | Mercestes | BB|AtWork, I suggest the CLI |
| 23:51.55 | BB|AtWork | JT, no idea if the span is up. i'm not sure if i have asterisk setup right |
| 23:52.11 | BB|AtWork | i'm using freepbx to configure the trunks/{in,out}bound routes |
| 23:52.14 | JT | BB|AtWork: does the console say anything about it? |
| 23:52.15 | Mercestes | ... |
| 23:52.15 | JT | arrgh |
| 23:52.18 | JT | ~freepbx |
| 23:52.34 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:52.34 | Mercestes | ~freepbx |
| 23:52.37 | jbot | freepbx is, like, unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:52.37 | Mercestes | bwhahahaha |
| 23:52.37 | Mercestes | die, jbot, die |
| 23:52.39 | BB|AtWork | well i really wanted to know if there was a way i could see an incomming call |
| 23:52.43 | JT | jbot: you are a slow pile of bits |
| 23:52.45 | jbot | JT: what are you talking about? |
| 23:52.46 | BB|AtWork | so |
| 23:52.47 | Qwell | jbot: You nub |
| 23:52.55 | BB|AtWork | i came to the right place for the question i asked |
| 23:53.13 | JT | BB|AtWork: do you even know how to access the asterisk cli? |
| 23:53.17 | BB|AtWork | yes |
| 23:53.32 | JT | freepbx is like a major change to asterisk |
| 23:53.33 | logicwrath | have you tried sip show peers? |
| 23:53.38 | BB|AtWork | i have verbosity at 7 and debug at 99. would it start spewing stuff it it was actually recieving it? |
| 23:53.45 | JT | logicwrath: wtf, he's talking about pri |
| 23:53.46 | Mercestes | Yes. |
| 23:53.57 | Mercestes | and no you did NOT come to the right place to ask your question :P |
| 23:54.06 | JT | BB|AtWork: try pri intense debug |
| 23:54.14 | BB|AtWork | Mercestes, yes i did. i asked nothing about freepbx |
| 23:54.21 | JT | and pri show span 1 |
| 23:54.38 | JT | BB|AtWork: if you keep arguing like that, few here will want to help you |
| 23:54.46 | Mercestes | If you have even tainted your install with the presence of freePBX.....then this is not the channel for you. |
| 23:54.50 | Shoeb | ~druid |
| 23:54.59 | Mercestes | so unless you wanted *DIRECTIONS* to #freepbx....then your in the wrong place for your question. |
| 23:55.13 | BB|AtWork | heh |
| 23:55.15 | JT | Mercestes: we can probably help a little |
| 23:55.25 | BB|AtWork | cmon i just wanted to know where to find debug information |
| 23:55.26 | JT | unless he keeps arguing about it being the right place |
| 23:55.43 | JT | BB|AtWork: have you read up on setting up pri interfaces in zaptel? |
| 23:55.45 | Mercestes | BB|AtWork, what does zap show channels show? |
| 23:56.07 | JT | BB|AtWork: so as i asked, what is the output of pri intense debug and pri show span 1? |
| 23:56.07 | Mercestes | that would be "zap show channels" in the asterisk cli under asterisk -r from your linux CLI that you get to via ssh. |
| 23:56.11 | BB|AtWork | JT, alot. i can't seem to get it to work. yeargh. the t1 isnt in the channels |
| 23:56.16 | JT | ~pb |
| 23:56.18 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:56.24 | JT | ~thebook |
| 23:56.26 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:56.38 | JT | BB|AtWork: read relevant sections of the book? |
| 23:57.03 | Mercestes | zap show status ??? |
| 23:57.20 | BB|AtWork | going to read that book |
| 23:57.33 | Mercestes | probably a good call. |
| 23:57.35 | BB|AtWork | Mercestes, zap show status isn't showing the right stuff. its not showing any of the t1 channels |
| 23:57.38 | Mercestes | oh..and get rid of that freepbox stuff. :P |
| 23:57.44 | Mercestes | BB|AtWork, what type of card is it? |
| 23:57.44 | BB|AtWork | heh |
| 23:57.49 | BB|AtWork | TE110P |
| 23:58.14 | Mercestes | Ah. |
| 23:58.15 | JT | BB|AtWork: pastebin zapata.conf and zaptel.conf |
| 23:58.21 | Mercestes | and modprobe -l |
| 23:58.31 | Mercestes | you never know. |
| 23:58.34 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 23:59.01 | BB|AtWork | /etc/zaptel.conf http://rafb.net/p/V9LfKR91.html |
| 23:59.01 | BB|AtWork | /etc/asterisk/zapata.conf http://rafb.net/p/bnYSst10.html |
| 23:59.04 | samy_b1 | can some on etell me how to get the DID of a SIP trunk when the provider doesn't send it ? |
| 23:59.23 | samy_b1 | so i can point it to my IVR ? |
| 23:59.35 | JT | samy_b1: magic |
| 23:59.39 | Mercestes | samy_b1: someone sold you a sip trunk? |
