| 00:06.35 | mrdigital | ManxPower: PM? |
| 00:08.05 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:12.49 | *** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net) |
| 00:13.08 | mrdigital | wheres the log for asterisk |
| 00:14.31 | *** join/#asterisk shodan (n=shodan@ip166.99-113-216.pppoe4.joliette.intermonde.net) |
| 00:14.54 | shodan | call from asterisk to a skipe user, possible, impossible, almost there ? |
| 00:15.00 | shodan | *skype |
| 00:16.07 | [TK]D-Fender | shodan, possible at cost, |
| 00:16.09 | Qwell | ... |
| 00:16.13 | Qwell | [TK]D-Fender: remember brian? |
| 00:16.31 | MikHell | [TK]D-Fender: I went through the Wiki for NAT and I do not see it addressing my case. :( |
| 00:16.48 | Qwell | ...the one in the ban list still, heh |
| 00:17.11 | shodan | [TK]D-Fender, is it a one time cost or $ per unit of time ? is there a free but unreliable way ? |
| 00:17.47 | Qwell | [TK]D-Fender: it was a firewall problem the entire time |
| 00:18.07 | [TK]D-Fender | qwell : And how did you come to this conclusion? |
| 00:18.27 | [TK]D-Fender | shodan, go look it up on the WIKI |
| 00:18.28 | [TK]D-Fender | ~wikis |
| 00:18.32 | jbot | [wikis] http://www.voip-info.org |
| 00:18.32 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:18.51 | [TK]D-Fender | qwell : You didn't exactly ban him in the most sweeping way... |
| 00:20.46 | shodan | k |
| 00:23.17 | Vec | Has anyone been able to get speex to compile with asterisk 1.4.2, I am having no end of trouble. |
| 00:24.36 | Qwell | [TK]D-Fender: he msg'd me and told me |
| 00:25.52 | [TK]D-Fender | Qwell : Brilliant little ^%$@-tard isn't he? Oh no... we can't possibly see anything in debug or CLI output... no... only his useless 3 line rephrased description will do and we are all jsut a bunch of losers.... |
| 00:26.56 | mrdigital | why when i call an exten it says all circuits are busy |
| 00:26.56 | [TK]D-Fender | qwell : I'm shopping for a new blade... maybe I can get a "volunteer" for a cutting test ;) |
| 00:26.59 | mrdigital | no one is using it |
| 00:27.30 | thekidrio | [TK]D-Fender, get a kyocera ceramic if ya want to cut human |
| 00:27.38 | [TK]D-Fender | mrdigital, Maybe you should look at what its DIALING. And if you want us to tell you what your own eyes should be telling you, perhaps you be so kind as to SHOW US. |
| 00:27.47 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 00:27.59 | [TK]D-Fender | thekidrio, http://aocomputing.net/bushi <- My current baby |
| 00:28.02 | mrdigital | this is the exten |
| 00:28.05 | mrdigital | dialplan |
| 00:28.10 | [TK]D-Fender | thekidrio, THAT is for cutting humans |
| 00:28.19 | ringhals | I would like to make new ringtones for my Cisco IP phones anyone have a sox command they would share for cutting the files to the right length |
| 00:28.25 | mrdigital | http://rafb.net/p/66tWSF54.html |
| 00:28.30 | Qwell | [TK]D-Fender: why doesn't it glint? |
| 00:28.31 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 00:28.36 | mrdigital | i changed the exten from 700 to 97 |
| 00:28.38 | Qwell | all cool swords glint |
| 00:28.45 | DocHolliday | sup [TK]D-Fender |
| 00:28.56 | mrdigital | unless 97 is also used |
| 00:29.20 | thekidrio | [TK]D-Fender, nice and yeah that could cut up some mean human steak |
| 00:29.43 | Qwell | that's the word of the day, BTW |
| 00:29.48 | thekidrio | needed to use macro lens on P1010018.JPG btw |
| 00:29.48 | ManxPower | mrdigital: That is not CLI output |
| 00:29.56 | thekidrio | glint? |
| 00:29.58 | Qwell | yes |
| 00:30.01 | thekidrio | good word |
| 00:30.03 | [TK]D-Fender | mrdigital, the why are you showing me something that isn't REAL. and as ManxPower said, its not CLI OUTPUT |
| 00:30.13 | mrdigital | i got it |
| 00:30.17 | Qwell | google doesn't have a definition of it.. only synonyms |
| 00:30.20 | DocHolliday | Qwell, where can i grab the nvfaxdetect source? |
| 00:30.22 | mrdigital | exten 97 is used |
| 00:30.24 | DocHolliday | newmantelecom is down |
| 00:30.27 | Qwell | DocHolliday: got me |
| 00:30.29 | thekidrio | really qwell? |
| 00:30.31 | thekidrio | thats funny |
| 00:30.55 | DocHolliday | Qwell its my only option for SIP fax right? |
| 00:30.59 | ManxPower | mrdigital: You are using some GUI for Asterisk rean't you? |
| 00:31.27 | mrdigital | im using the conf files |
| 00:31.28 | thekidrio | [TK]D-Fender, nice design in P1010022.JPG |
| 00:31.34 | ManxPower | DocHolliday: Uh, NVFaxDetect only DETECTS a fax tone then jumps to an extension named "fax" IT does nothing else. |
| 00:31.37 | Qwell | ManxPower: You should test out my zaptel branch |
| 00:31.43 | thekidrio | hehe ManxPower the gui is coming the gui is coming better watch ou |
| 00:31.44 | thekidrio | y |
| 00:31.53 | ManxPower | So is is basically for voice and fax on the name number |
| 00:32.00 | ManxPower | Qwell: Why? |
| 00:32.13 | Qwell | because I don't know if it actually works in a realworld scenario, heh |
| 00:32.17 | thekidrio | haha |
| 00:32.28 | DocHolliday | ManxPower, thats exactly what i need, i have a SIP ATA with a fax connected to it.. |
| 00:32.28 | Qwell | I only know it works with my uber-cool new "echo can" |
| 00:32.29 | ManxPower | Qwell: what is it supposed to do? |
| 00:32.38 | [TK]D-Fender | thekidrio, I am currently considering this : http://www.casiberia.com/product_details.asp?id=SH1018 |
| 00:32.44 | DocHolliday | thus asterisk --> SIP ATA --> Fax Machine |
| 00:32.46 | Qwell | ManxPower: give you a dump of the pre- and post- echocan audio |
| 00:33.02 | ManxPower | ah. |
| 00:33.05 | Qwell | which would eventually be useful for debugging |
| 00:33.06 | Vec | Trying to compile speex with asterisk and getting "symbol lookup error: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_decode_int" when I try load asterisk, any ideas ? |
| 00:33.12 | ManxPower | that's why kpflemming was talking about? |
| 00:33.16 | Qwell | yeah |
| 00:33.35 | Qwell | zaptel/team/qwell/echocan-debug/ I think was the name of it |
| 00:33.39 | [TK]D-Fender | thekidrio, it lacks a little cutting power for the close POB, and slightly lower weight. |
| 00:33.56 | [TK]D-Fender | thekidrio, But for me, agility = flying death. |
| 00:34.07 | ManxPower | Qwell: I'll try it on sunday. The highest usage of the system is on the weekends |
| 00:34.16 | [TK]D-Fender | thekidrio, And my newfound love of the extended tsuka |
| 00:34.17 | Qwell | sounds good |
| 00:34.24 | Juggie | i've answered so many questions around here, but i'm stuck myself now, who wants to help me with this (real non clone) x100p someone gave me. |
| 00:34.32 | Juggie | it appears to be stuck off hook. |
| 00:34.53 | Juggie | (i allways use digital hardware at work, sigh, analog!) |
| 00:34.54 | ManxPower | Juggie: I've not seen that problem before. What does zttool show you? |
| 00:35.06 | Juggie | ManxPower, just says OK. |
| 00:35.08 | Qwell | ManxPower: You should also try enabling the uber-cool JP1 echo can in that tree, and see if it sounds similar |
| 00:35.16 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 00:35.22 | Juggie | ManxPower, i might have it configured totally wrong. |
| 00:35.38 | Qwell | the JP1 echo can is the way of the future... |
| 00:35.47 | ManxPower | Juggie: it's pretty much exactly the same as a 1xFXO tdm400P except the driver is different |
| 00:35.56 | Juggie | ManxPower, i've never configured analog ever. |
| 00:36.02 | Qwell | the quality is about on par with lpc10, but the codec size is the same as slin |
| 00:36.11 | Juggie | ManxPower, only T1 :) |
| 00:36.28 | ManxPower | You city boys don't know nutthin about analog. |
| 00:36.33 | Juggie | ManxPower, do i need Loop/Ground or Kewl start? |
| 00:36.43 | [TK]D-Fender | Juggie, kewlstart |
| 00:36.49 | Qwell | nubstart |
| 00:37.13 | ManxPower | Juggie: fxs ks (kewlstart) Remember fxo ports use fxs signalling and fxs ports use fxo signalling |
| 00:38.00 | Juggie | yep, i did do that properly. |
| 00:39.12 | Juggie | ok, so thats all fine |
| 00:39.20 | Juggie | but its stuck offhook. |
| 00:39.28 | Juggie | Hookstate (FXS only): Offhook |
| 00:39.39 | Juggie | as soon as i plug the line in, it changes to offhook. |
| 00:39.41 | ManxPower | Juggie: uh, notice the FXS only |
| 00:40.04 | Juggie | er, touche |
| 00:40.07 | Juggie | its changing though. |
| 00:40.14 | Juggie | when i unplug the line it goes to onhook. |
| 00:40.25 | Juggie | but consequentially, calling my house is busy now |
| 00:40.28 | Juggie | so its off hook. |
| 00:40.55 | DocHolliday | Qwell, is there another project equivalent to nvfaxdetect? |
| 00:41.03 | Qwell | dunno |
| 00:41.05 | ManxPower | Juggie: /etc/zaptel.conf: "loadzone = us" and "defaultzone=us" and "fxsks=1" |
| 00:41.18 | Juggie | just those 3 lines? |
| 00:41.27 | ManxPower | I'm a minimalist |
| 00:41.36 | Juggie | i dont need channels=1? |
| 00:41.47 | ManxPower | not in /etc/zaptel.conf you don't. |
| 00:41.49 | _VoiceMeUp_Com | i think its channel=1 ( no s) |
| 00:42.04 | Juggie | nah its channels. |
| 00:42.11 | _VoiceMeUp_Com | not in zaptel he is right |
| 00:42.11 | Juggie | ok, well its in the example, but i'll try. |
| 00:42.25 | ManxPower | If you wait a min ya young whippersnapper I'll give you a /etc/asterisk/zapata.conf |
| 00:42.39 | _VoiceMeUp_Com | no its in zapata channel=>1 |
| 00:42.42 | _VoiceMeUp_Com | no S |
| 00:42.47 | _VoiceMeUp_Com | the context is channels |
| 00:42.57 | _VoiceMeUp_Com | channel => 1-21 |
| 00:43.09 | Juggie | _VoiceMeUp_Com, we are talking about /etc/zaptel.conf |
| 00:43.12 | Juggie | not zapata |
| 00:43.17 | _VoiceMeUp_Com | eah sorry |
| 00:43.21 | _VoiceMeUp_Com | he asked abotu channel=1 |
| 00:43.31 | _VoiceMeUp_Com | so i said its channel no (s) and its in zapata |
| 00:43.37 | DocHolliday | _VoiceMeUp_Com, heh docelmo never got back to me :( |
| 00:43.37 | _VoiceMeUp_Com | Juggie, ; ) |
| 00:44.10 | ManxPower | <PROTECTED> |
| 00:44.57 | thekidrio | DocHolliday, yeah i don't think he got back to me either |
| 00:45.12 | *** join/#asterisk CrazyYoss (n=luther@c-24-5-165-3.hsd1.ca.comcast.net) |
| 00:46.03 | DocHolliday | ack |
| 00:52.35 | _VoiceMeUp_Com | Found my eriupe TDM termination.. |
| 00:52.37 | _VoiceMeUp_Com | sounds great |
| 00:52.48 | _VoiceMeUp_Com | eriup is europe after an earthquake |
| 00:53.07 | _VoiceMeUp_Com | friday night huumor.. i guess im not a sand up comedian |
| 00:58.40 | DocHolliday | _VoiceMeUp_Com, tough crowd tonight |
| 01:00.41 | Juggie | ManxPower, still not happy. |
| 01:00.52 | Juggie | the x100p is really pissing off my ATA |
| 01:04.30 | Juggie | maybe its broken, because if i try an outbound call, i dont even see it dial. |
| 01:04.46 | Juggie | er, hear it. |
| 01:09.05 | shodan | what's the verdict on "uplink skype to sip adapter" ? I just tried it out, won't work because I'm running x64 :\ is it worth sacrificing another box just for it ? |
| 01:10.10 | tzafrir_laptop | Juggie, "cannot see it dial"? how? with a phone in parallel to it? |
| 01:10.37 | tzafrir_laptop | Juggie, or with ztmonitor? |
| 01:11.33 | Juggie | hmm good idea |
| 01:11.37 | Juggie | i should build ztmonitor |
| 01:11.49 | Juggie | i mean when i have it hooked up, and i pick up a second phone to listen, the card isnt dialing. |
| 01:12.12 | Juggie | also the card is constantally off hook, as soon as i plug it in, its offhook, i can see it off hook on my ATA. |
| 01:17.29 | *** join/#asterisk Opperior (n=chatzill@c-75-69-241-84.hsd1.nh.comcast.net) |
| 01:18.05 | ManxPower | Juggie: I assume you have only 1 line plugged into 1 port on the X100P |
| 01:18.06 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 01:18.12 | Juggie | correct. |
| 01:18.21 | ManxPower | I assume you have also checked /proc/interrupts to make sure there is no IRQ sharing |
| 01:18.27 | xpot | anyone know if T38 Fax is implemented in *1.4? |
| 01:18.52 | Juggie | ManxPower, its actually shared w/ usb. |
| 01:18.57 | Juggie | but there are no usb devices plugged in. |
| 01:19.13 | Juggie | i dont see how a shared irq would keep this card persistantally offhook. |
| 01:19.51 | Juggie | i should try it with the card powered off. |
| 01:22.38 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:22.55 | [TK]D-Fender | Juggie, Sure you're using the right port on the back? |
| 01:23.10 | [TK]D-Fender | Juggie, And have you tested the cable? |
| 01:23.42 | Juggie | [TK]D-Fender, i've tried dif cables |
| 01:24.04 | Juggie | and i've also tried both ports, though the correct one is clearly labeled 'line' |
| 01:24.13 | Juggie | even with the card in my hand, if i plug it in, i go offhook |
| 01:25.31 | Juggie | seems odd eh |
| 01:26.08 | Juggie | i supposed it could be the ATA its connected to |
| 01:26.12 | Juggie | b |
| 01:27.18 | Juggie | ahhh |
| 01:27.21 | Juggie | i see the problem |
| 01:28.58 | [TK]D-Fender | Juggie, What ATA? |
| 01:29.06 | ManxPower | You plugged the card into the FXO port of the ATA didn't you? |
| 01:29.12 | [TK]D-Fender | Juggie, Don't tell me you plugged it into an FXO or ethernet port :) |
| 01:29.21 | Juggie | no |
| 01:29.23 | ManxPower | ~fxofxs |
| 01:29.32 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 01:29.34 | Juggie | looks like during the shipping one of the resistors got broken |
| 01:29.48 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 01:30.19 | ManxPower | ah |
| 01:30.52 | DocHolliday | how can i set the CID Name? |
| 01:31.26 | ManxPower | DocHolliday: For what? |
| 01:31.40 | DocHolliday | for a SIP DID |
| 01:31.49 | ManxPower | incoming or outgoing? |
| 01:31.55 | DocHolliday | outgoing |
| 01:32.00 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
| 01:32.00 | ManxPower | You can't |
| 01:32.11 | DocHolliday | oh? |
| 01:32.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:32.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:32.23 | ManxPower | Well, you CAN, but the telco will ignore it and look up the name associated with the sent CLID number. |
| 01:32.36 | ManxPower | not the originating telco, the terminating telco. |
| 01:32.49 | DocHolliday | how can i ensure that the name is resolved? |
| 01:32.59 | ManxPower | Huh? |
| 01:33.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:33.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:33.14 | ManxPower | Set it to a valid phone number. |
| 01:33.29 | DocHolliday | how do traditional telcos set CID Name then? |
| 01:34.23 | mrdigital | DocHolliday? pm |
| 01:34.31 | ManxPower | The carrier of the calling line puts it into some sort of shared database (called CNAME). The terminating telco then takes the calling phone number, looks up the name associated with that number and then hands that to the destination |
| 01:34.44 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:34.46 | [TK]D-Fender | ManxPower, I've got telcos here that let you set CID name... |
| 01:34.53 | ManxPower | VoIP carriers generally do not offer the option of putting a name in the CNAM database |
| 01:35.24 | ManxPower | [TK]D-Fender: my client's telco does to, but once the call leaved the telco's network the name goes awayt |
| 01:35.40 | Juggie | hmmmm |
| 01:35.45 | Juggie | how am i gonna fix this |
| 01:36.10 | ManxPower | I could set a callerid name of "God Himself" and the moment it leaves XFone's network the terminating telco does a CNAME dip. If the call stays on XFone's network I am still God |
| 01:36.19 | ManxPower | Juggie: you throw out the card and get another one |
| 01:36.30 | Juggie | ManxPower, send me one then :P |
| 01:37.00 | ManxPower | Juggie: I have two I think. Somewhere. Maybe. |
| 01:37.18 | ManxPower | At least one of them was destroyed in the flood |
| 01:37.22 | Juggie | it looks like the brown resistor got tore off |
| 01:37.33 | Juggie | Manx, dont you live on a mountain |
| 01:37.55 | ManxPower | I used to live 13 miles east of where the eye of Katrina made landfall. |
| 01:38.15 | ManxPower | And about 1.5 miles inland from the ocean. |
| 01:38.43 | ManxPower | I still have a TDM400P that is all corroded from the salt water. |
| 01:40.03 | ManxPower | Hmm.. It sounds like there is a dying water buffalo outside. They must be doing kareoke now. |
| 01:41.27 | Juggie | ManxPower, RMA it |
| 01:41.30 | Juggie | that would be funny :) |
| 01:41.48 | Juggie | i'm not 100% sure where this brown resistor connects |
| 01:42.04 | Juggie | i dunno if anyone has this same card kicking around |
| 01:45.41 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au) |
| 01:46.15 | ManxPower | jello shooters + kareoke machine = horror |
| 01:46.22 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:47.53 | DocHolliday | ManxPower, heh |
| 01:50.04 | [TK]D-Fender | ManxPower, the one I say worked cross network... can't explain it... |
| 01:51.05 | blitzrage | party time |
| 01:51.07 | blitzrage | excellent |
| 01:51.21 | [TK]D-Fender | blitzrage, Party on Wayne! |
| 01:51.28 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:51.29 | blitzrage | Party on Garth |
| 01:51.35 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:52.09 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 01:55.58 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:56.25 | dseeb_ | ~ seen voipy |
| 01:56.54 | jbot | voipy <n=pirch@a81-84-60-131.cpe.netcabo.pt> was last seen on IRC in channel #asterisk, 30d 4h 32s ago, saying: 'Does anyone use Chan_cellphone and knows how to solve the bluetooth pairing prob on bluez-utils 3.7-1?'. |
| 01:57.19 | Juggie | anyone have a REAL digium x100p handy |
| 01:58.05 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 02:01.53 | tzafrir_laptop | Juggie, I should have one here |
| 02:02.48 | tzafrir_laptop | sorry, not a real x100p. Just some "clone". If it really matters. |
| 02:07.10 | Juggie | tzafrir, i was just wondering about one of the resistors on the board |
| 02:07.26 | Juggie | just wanted to double check where it was connected. |
| 02:11.55 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:12.15 | kiwoneka | good evening to all |
| 02:13.56 | kiwoneka | my someone help/point me in the direction of a good resource to setup voicemail |
| 02:14.07 | kiwoneka | paging |
| 02:14.11 | kiwoneka | thanks |
| 02:15.52 | ManxPower | tzanger: glue a heatsink on it and claim it is a real X100P. Everyone else seems to. |
| 02:16.32 | [TK]D-Fender | kiwoneka, Look at the sample voicemail.conf |
| 02:16.49 | mrdigital | asterisk is fun |
| 02:20.15 | Juggie | heh |
| 02:20.28 | Juggie | damn i was so looking forward to setting this up so i could use my unlimited longdistance frmo anywhere |
| 02:23.14 | blitzrage | Juggie: yah... that sucks :( |
| 02:26.19 | *** join/#asterisk Fieldy (i=Exidp8Ei@gentoo/contributor/Fieldy) |
| 02:30.12 | Juggie | yah, it does. |
| 02:30.35 | [TK]D-Fender | Juggie, Whats your LD going through? |
| 02:30.49 | [TK]D-Fender | Juggie, ATA + locked ITSP? |
| 02:31.40 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:41.29 | aptura | :) |
| 02:42.25 | aptura | Yea good way to end the week being that is friday. |
| 02:43.13 | aptura | BTW who here is from Las Vagas? I will be there come this fall for a few days. |
| 02:43.55 | aptura | also, looking for bootrom.ld ver 1.5.3 |
| 02:43.59 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:45.16 | *** join/#asterisk VoIPMasta (n=fabio@201.139.138.21.cable.dyn.cableonline.com.mx) |
| 02:45.36 | mrdigital | tk: i have 3 incoming 800 #s how do i have it where if someone calls 1 of them |
| 02:45.44 | VoIPMasta | Hi there |
| 02:45.45 | mrdigital | the 1-800 # appears on all the extenstions |
| 02:45.53 | VoIPMasta | does anyone know how can I play a recording into an active call? |
| 02:47.13 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 02:47.15 | aptura | mr you want to know if you can see which number thay are calling in from? what phone are you using? |
| 02:48.12 | mrdigital | right now a softphone as im still developing the system |
| 02:48.12 | mrdigital | xlite |
| 02:48.24 | mrdigital | no |
| 02:48.31 | mrdigital | i dont want to see the CALLERs # |
| 02:48.40 | mrdigital | i want to see the 1-800 # their calling on the extenstons CID |
| 02:49.42 | aptura | k |
| 02:49.48 | mrdigital | you know how to do this? |
| 02:50.24 | VoIPMasta | mrdigital are you using AGI? |
| 02:50.48 | mrdigital | no |
| 02:51.04 | VoIPMasta | because it's the only way I can think of to achieve what you want to do |
| 02:51.46 | mrdigital | can i do distencive ringing for each # |
| 02:51.58 | VoIPMasta | another way would be to assing each DID/extension to a different line (sip account) in your X-lite and therefore you'll be able to see what number/extension the user dialed just by looking which line is active |
| 02:52.09 | mrdigital | well |
| 02:52.14 | mrdigital | we're going to be going to 1 line phones |
| 03:00.11 | DocHolliday | VoIPMasta, or just consider setting the CID Name? |
| 03:01.34 | VoIPMasta | DocHolliday: but the phone doesn't have a display |
| 03:02.07 | VoIPMasta | ohh you meant mrdigital's issue :) |
| 03:02.13 | VoIPMasta | I thought you were talking about mine |
| 03:02.24 | VoIPMasta | yes that could work for mrdigital |
| 03:03.22 | mrdigital | ??? |
| 03:04.05 | VoIPMasta | mrdigital you can set the CID name according to the dialed extension and it would show up in X-lite's display (DocHolliday's solution) |
| 03:04.15 | blitzrage | VoIPMasta: only thing I can think of (for playing music into a current channel) is to use whisper paging somehow... |
| 03:04.20 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.26 | mrdigital | where do i set it |
| 03:05.11 | *** join/#asterisk Journeyman (i=matt@72-161-187-112.dyn.centurytel.net) |
| 03:05.24 | Journeyman | do you have to have a voice modem for asterisk to work? |
| 03:05.40 | blitzrage | Journeyman: absolutely not |
| 03:05.48 | blitzrage | 2.6 can be the timing source |
| 03:05.52 | blitzrage | 2.6 kernel* |
| 03:06.12 | Journeyman | oh good |
| 03:06.20 | blitzrage | you basically want to use the ztdummy driver to pull timing from the kernel |
| 03:06.24 | blitzrage | what OS? |
| 03:06.43 | DocHolliday | VoIPMasta, good point.. didnt know that' |
| 03:07.06 | Journeyman | is it stupid to want to set up asterisk for my own personal voip system? |
| 03:07.11 | mrdigital | no |
| 03:07.12 | Journeyman | linux |
| 03:07.15 | Journeyman | Linux Ubuntu 2.6.17-11-generic #2 SMP Thu Feb 1 19:52:28 UTC 2007 i686 GNU/Linux |
| 03:07.31 | VoIPMasta | blitzrage: whisper paging? |
| 03:08.11 | Journeyman | I don't know much about this technology and I want to learn by setting up my own system and playing with it |
| 03:08.16 | [BAF64] | anyone here know how to access asterix gui after installing it? |
| 03:09.37 | blitzrage | Journeyman: no way |
| 03:09.42 | blitzrage | Journeyman: asterisk is awesome |
| 03:09.54 | blitzrage | Journeyman: perfect start |
| 03:10.18 | blitzrage | Journeyman: you should be good to use the ztdummy driver on that |
| 03:10.25 | blitzrage | see http://www.asteriskdocs.org |
| 03:10.28 | blitzrage | and |
| 03:10.29 | blitzrage | ~doc |
| 03:10.42 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 03:10.42 | blitzrage | ~docs |
| 03:10.44 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 03:10.56 | tzafrir_laptop | [BAF64], http://hostname:8088/ ? |
| 03:11.20 | tzafrir_laptop | ~asterix |
| 03:11.22 | jbot | i heard asterix is a fearless fighter of the Roman tyranny |
| 03:11.38 | [BAF64] | nope |
| 03:11.54 | [BAF64] | i didnt see anything about the GUI anyplace |
| 03:12.04 | [BAF64] | i got it setup and installed, at least according to the readme |
| 03:12.09 | [BAF64] | but nothign says how to actually access it |
| 03:12.10 | kiwoneka | check your directory permission |
| 03:12.29 | tzafrir_laptop | jbot, no, asterix is a fearless fighter of the Roman tyranny, who enjoys sueing Penguins: http://mobilix.org/ |
| 03:12.32 | jbot | okay, tzafrir_laptop |
| 03:12.42 | [BAF64] | kiwoneka, which directory |
| 03:13.17 | tzafrir_laptop | [BAF64], which system is it? asterisknow or self-installed? |
| 03:13.43 | [BAF64] | self installed |
| 03:13.56 | tzafrir_laptop | netstat -lntp | grep 8088 |
| 03:14.20 | [BAF64] | nothing |
| 03:14.24 | tzafrir_laptop | This should show you who, if at all, is listening on port 8088 |
| 03:14.44 | tzafrir_laptop | So I guess you need to edit /etc/asterisk/http.conf and reload |
| 03:14.51 | tzafrir_laptop | and/or, that is |
| 03:14.56 | [BAF64] | what should http.conf say |
| 03:15.03 | [BAF64] | i put enabled and enable-static in it like the README said |
| 03:15.09 | [BAF64] | tcp 0 0 0.0.0.0:5038 0.0.0.0:* LISTEN 21261/asterisk |
| 03:15.10 | [BAF64] | tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 21261/asterisk |
| 03:15.15 | [BAF64] | ^ those are the only 2 ports asterix is listening on |
| 03:16.03 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 03:17.24 | [BAF64] | i have http.conf, and i have restarted asterisk |
| 03:20.33 | [BAF64] | got it |
| 03:22.38 | [BAF64] | well, make checkconfig says its okay, but it still doesn't listen on 8088 |
| 03:23.15 | masked | you need to change the bindaddr |
| 03:23.33 | masked | it'll be set to something other than what the notes says is default |
| 03:23.37 | masked | iono why that is.. |
| 03:25.33 | [BAF64] | didn't change it |
| 03:25.41 | ManxPower | generally you can just remoe the bindaddr, |
| 03:26.45 | [BAF64] | I commented out the bindaddr and it still doesn't appear to be listenin |
| 03:26.47 | [BAF64] | listening |
| 03:26.59 | [BAF64] | netstat doesn't list it and I can't get to it to run the setup |
| 03:29.22 | [BAF64] | masked, any other ideas? |
| 03:30.11 | [BAF64] | I get Apr 13 23:28:32 WARNING[22850] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 03:30.14 | [BAF64] | in the message log |
| 03:30.18 | [BAF64] | is that anything relevant? |
| 03:33.59 | masked | [BAF64]: bindaddr = 0.0.0.0 |
| 03:34.11 | [BAF64] | in http or manager |
| 03:35.08 | [BAF64] | that doesn't appear to make a difference |
| 03:37.59 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:40.19 | [BAF64] | so, how are you supposed to install the GUI.. perhaps I did something wrong |
| 03:41.01 | [BAF64] | I did a svn co on it, configure, make, make install, make samples, made a http.conf and edited manager.conf as per the readme from the svn co, make checkconfig (reported it was ok, etc), and it doesnt work |
| 03:41.22 | [TK]D-Fender | [BAF64], Looks like its trying to start an ODBC connection and unixODBC isn't stup right |
| 03:41.40 | [BAF64] | i don't want ODBC |
| 03:41.42 | [BAF64] | :P |
| 03:41.54 | [BAF64] | is that preventing the gui or whatever from starting up? |
| 03:42.04 | tzafrir_laptop | if the bind address was incorrect, it would still show in netstat on port 8088 |
| 03:42.28 | [BAF64] | its not on netstat :\ |
| 03:43.10 | [TK]D-Fender | [BAF64], perhaps you should PASTBIN your startup attempt for us to SEE... |
| 03:43.10 | [TK]D-Fender | ~pb |
| 03:43.20 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:43.20 | tzafrir_laptop | [BAF64], could you please pastebin your http.conf and manager.conf (with passwords modified...) |
| 03:43.38 | [BAF64] | one sec |
| 03:44.58 | [BAF64] | http://pastebin.bafserv.com/951 |
| 03:45.31 | [TK]D-Fender | [BAF64], and for what I asked for? |
| 03:45.41 | [BAF64] | it starts up fine |
| 03:45.49 | [BAF64] | what verbosity level do you want? |
| 03:46.10 | [TK]D-Fender | [BAF64], Oh, so * starts fine, its just the HHTP server for GUI that doesn't? |
| 03:46.16 | [BAF64] | yeah |
| 03:46.18 | [TK]D-Fender | ah |
| 03:48.00 | [TK]D-Fender | [BAF64], look like you're missing bindport |
| 03:48.50 | [BAF64] | bindport = 8088 under httpd.conf? |
| 03:49.08 | [TK]D-Fender | [BAF64], yes, or similar |
| 03:49.10 | [BAF64] | i mean i didnt see anything on what these confs should look like in the docs |
| 03:49.23 | [BAF64] | bindport = 8088 did nothing |
| 03:49.27 | [TK]D-Fender | [BAF64], you don't need doncs.. just look at the SAMPLE |
| 03:49.47 | [BAF64] | what sample |
| 03:50.06 | [BAF64] | the sample in README doesnt have anything about port |
| 03:50.23 | [BAF64] | just enabled and enablestatic for http, and enabled and webenabled for manager |
| 03:50.43 | [TK]D-Fender | I have it from when I did "make samples" just fine, which means it in the source samples folder . |
| 03:51.35 | [BAF64] | i ran make samples and it didnt appear to do much |
| 03:51.50 | [TK]D-Fender | configs/http.conf.sample |
| 03:52.02 | [BAF64] | the copy i have only has providers.conf.sample |
| 03:52.05 | [TK]D-Fender | [BAF64], then you've clearly done something wrong along the way. |
| 03:52.10 | [BAF64] | revision 658 |
| 03:52.13 | [TK]D-Fender | go check that out in your * source folder |
| 03:52.58 | *** join/#asterisk bmg505 (n=leon@196.209.176.132) |
| 03:53.08 | [BAF64] | I installed it via portage on Gentoo |
| 03:53.12 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:53.39 | [TK]D-Fender | [BAF64], Oh... then that can potential void who knows how much otherwise useful advice... |
| 03:53.56 | [BAF64] | hmm |
| 03:54.05 | [BAF64] | where is http.conf.sample supposed to come from? |
| 03:54.08 | [BAF64] | * or *-gui |
| 03:55.28 | [TK]D-Fender | * |
| 03:55.29 | *** join/#asterisk antlers (n=antlers@ip70-173-90-39.lv.lv.cox.net) |
| 03:55.29 | [BAF64] | hmm |
| 03:55.29 | [TK]D-Fender | [BAF64], FYI : http://www.pastebin.ca/439283 |
| 03:55.49 | [BAF64] | hmmm |
| 03:56.02 | [TK]D-Fender | Packaged * = ASS. Anyone depending on packages for this ... YMMV |
| 03:56.10 | [TK]D-Fender | and... |
| 03:56.15 | [TK]D-Fender | ~wglwat |
| 03:56.23 | jbot | i heard wglwat is well, good luck with all that |
| 03:57.02 | [BAF64] | heh |
| 03:57.26 | [BAF64] | how nice |
| 03:57.45 | [BAF64] | more broken packages in portage |
| 03:58.26 | *** join/#asterisk bawb2 (n=bawb2@ip48200.estcmp.ku.edu) |
| 03:58.58 | [TK]D-Fender | [BAF64], Unheard of! |
| 03:59.06 | [TK]D-Fender | [BAF64], Use the source, Luke! |
| 03:59.16 | [BAF64] | BarfTheDog configs # ls *http* |
| 03:59.16 | [BAF64] | ls: *http*: No such file or directory |
| 03:59.21 | DocHolliday | s/force/source |
| 03:59.21 | shodan | anyone knows why "uplink skipe2sip" doesn't hangup the skype call when the asterisk side hangs up ? is there a fix for that ? |
| 03:59.57 | [TK]D-Fender | [BAF64], And don't go telling me "But its Gentoo... it IS source....." straight from the freezer with modified corn-starch and BHT till the cows come home... |
| 04:00.19 | [BAF64] | yeah i know |
| 04:00.24 | [BAF64] | gentoo likes to break stuff :P |
| 04:00.25 | [TK]D-Fender | shodan, Nope, and Gl finding people here at this hour using it..... |
| 04:00.34 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.50 | [BAF64] | the funny thign is there isn't anything on http |
| 04:00.58 | [BAF64] | 1.2.14 is the version it's using |
| 04:01.05 | [BAF64] | is that what it should be using? |
| 04:01.17 | [TK]D-Fender | OMG... the GUI only works on 1.4! |
| 04:01.24 | [BAF64] | srsly? |
| 04:01.32 | [TK]D-Fender | [BAF64], Dear God... you missed the BIG FRIGGEN PRINT |
| 04:01.35 | [BAF64] | lmao |
| 04:01.42 | [BAF64] | how n00bish of me |
| 04:01.43 | [TK]D-Fender | ~osmosis |
| 04:01.47 | jbot | well, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 04:02.03 | [TK]D-Fender | uNF! |
| 04:02.04 | shodan | [TK]D-Fender, oh well, at least it works great when it does work, the audio is smooth on my SPA-3XXX |
| 04:02.13 | [BAF64] | haha |
| 04:02.20 | [BAF64] | i've been up too many hours when i miss stuff like this |
| 04:02.56 | [BAF64] | wait |
| 04:02.59 | [BAF64] | are you serious, [TK]D-Fender? |
| 04:03.04 | [TK]D-Fender | [BAF64], YES |
| 04:03.09 | [BAF64] | i see no mention of 1.4 in the readme or #asterisk-gui chan |
| 04:03.38 | shodan | why would asterisk need a gui !?, iptables doesn't have one !! |
| 04:04.13 | *** part/#asterisk Fieldy (i=Exidp8Ei@gentoo/contributor/Fieldy) |
| 04:04.34 | [BAF64] | alright |
| 04:04.37 | [BAF64] | i'll deal with this tomorrow |
| 04:04.40 | [BAF64] | thanks for the help guys |
| 04:04.44 | [BAF64] | later |
| 04:04.53 | [TK]D-Fender | [BAF64], I'm going to go collect my hair now... |
| 04:04.54 | [TK]D-Fender | :) |
| 04:04.58 | [TK]D-Fender | [BAF64], later |
| 04:04.58 | [BAF64] | hehe |
| 04:05.27 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
| 04:05.28 | [BAF64] | you can have some of my hair |
| 04:05.33 | [BAF64] | i pulled a bunch out too :] |
| 04:06.54 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 04:16.17 | clyrrad | Hey anyone here from the UK? |
| 04:16.31 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 04:20.54 | clyrrad | anyone even awake? |
| 04:24.02 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 04:25.45 | coppice | I'm from the UK. I escaped, though |
| 04:25.57 | clyrrad | coppice: are you there now? |
| 04:26.05 | clyrrad | coppice: or local to it? |
| 04:26.17 | coppice | I just said. I escaped |
| 04:26.27 | clyrrad | do you by chance have a UK DID? |
| 04:26.42 | coppice | I'm local to the UK in galactic terms. i'm in asia |
| 04:26.55 | clyrrad | heh - I mean in calling terms |
| 04:26.57 | VoIPMasta | you can get UK 0870 DID for free almost everywhere |
| 04:27.04 | clyrrad | I have a UK toll free DID and I have no way to test it if it works |
| 04:27.08 | clyrrad | Im in Canada |
| 04:27.36 | VoIPMasta | clyrrad: do you want me to call you? I can dial UK toll free DIDs |
| 04:27.38 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 04:27.54 | clyrrad | VoIPMasta: yes - you wont get me - but a PBX (if it works) yes that would be great |
| 04:28.01 | clyrrad | VoIPMasta: may I PM? |
| 04:28.06 | VoIPMasta | ok go ahead |
| 04:28.12 | clyrrad | great thanks |
| 04:38.01 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 04:40.37 | blitzrage | Canada kicks ass |
| 04:40.37 | blitzrage | going to bed though --- night! |
| 04:50.11 | notoriousrab1982 | anyone know how to start festival working in asterisk 1.4.0 - using the book, but cannot find festival.scm locaed in /etc/ or /usr/share/festival |
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| 05:10.35 | CessnaFlyer | hello, all! |
| 05:10.55 | notoriousrab1982 | hi cessnaflyer, think everyone is asleep |
| 05:11.16 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 05:12.21 | CessnaFlyer | hrm... too bad... im having a spot of trouble with asterisk... was hoping someone could help me troubleshoot the demo greeting stuttering very badly |
| 05:12.44 | notoriousrab1982 | im no expert, but what is it doing |
| 05:13.02 | notoriousrab1982 | its not a QoS problem with bandwidth is it? |
| 05:13.32 | CessnaFlyer | i just installed asterisk on a VMware linux install, using the demo scripts, and when i connect using x-lite (dialing "2" for the demo greeting), the sound stutters very badly |
| 05:13.57 | CessnaFlyer | i hope its not a bandwidth issue, since its traveling across a virtual network on the same machine |
| 05:14.27 | notoriousrab1982 | i had similar problem, i recorded a greeting in wav format, converted it to gsm and put it onto the dialplan, its probably best to record a greeting from a handset and try that |
| 05:16.11 | notoriousrab1982 | cessna: try something like that and see if it still stutters, record your own greeting - http://pastebin.ca/439331 |
| 05:22.54 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:26.19 | CessnaFlyer | hrm... i must have done something wrong, since now im getting a 404 |
| 05:26.24 | *** join/#asterisk CunningPike_ (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:26.52 | notoriousrab1982 | what do you mean by a 404: isn't that when a phone cannot register |
| 05:27.51 | CessnaFlyer | i think so, yes |
| 05:28.04 | CessnaFlyer | let me fire up the 'ol wireshark... |
| 05:28.43 | notoriousrab1982 | the code i put on pastebin was for extension.conf, create an extension which points into the [record] context, i think the phone not registering is a separate issue |
| 05:30.44 | *** join/#asterisk moranil (n=moranil@122.162.67.129) |
| 05:34.40 | CessnaFlyer | ah, that was my fault... it wasnt jumping from the default context to the record context, and i had commented out the demo context |
| 05:35.09 | CessnaFlyer | but now its showing an error in the log: Unable to open xiptel_welcomemessagerecordingfacility (format 0x4 (ulaw)): No such file |
| 05:35.09 | CessnaFlyer | or directory |
| 05:35.52 | notoriousrab1982 | sorry, that is a file I had on my own asterisk box, the pastebin was to give you a guide on how to record from the dialplan using the Record() application |
| 05:35.56 | CessnaFlyer | along with several other similar errors |
| 05:36.01 | CessnaFlyer | ah |
| 05:36.46 | notoriousrab1982 | so any lines which say playback, you could comment out to get a welcome message stored on your asterisk box |
| 05:37.51 | notoriousrab1982 | the pastebin will store the gsm file in /var/lib/asterisk/sounds/custom/ and will be given a unique code from the call detail record, from there you can do what you want with it |
| 05:39.00 | notoriousrab1982 | ie rename it and put it in your dialplan so when you ring your extension, you hear that message you recorded before, that will rule out if it is a sound-recording issue as it has been recorded directly from a handset in your phone system |
| 05:49.29 | CessnaFlyer | ok, finally got enough bugs worked out that i was able to record and playback, and the playback has the same stutter |
| 05:50.24 | CessnaFlyer | one could also describe it as a really, really bad echo |
| 05:52.42 | CessnaFlyer | is there a different softphone i could use to test? |
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| 07:19.40 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:34.53 | *** join/#asterisk sahafeez (n=sahafeez@m0c0e36d0.tmodns.net) |
| 07:42.12 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:11.16 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 08:13.05 | *** part/#asterisk moranil (n=moranil@122.162.67.129) |
| 08:18.17 | *** join/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 08:19.03 | kev009 | I've got a TDM400P with an FXO on an AsteriskNOW box, and as far as I know it is configured but when I try to place a call I get a busy signal |
| 08:19.40 | jnc | kev009: are you placing a call from the pstn or from a SIP phone |
| 08:20.30 | kev009 | sip phones |
| 08:21.17 | jnc | this might be #asterisk-gui material, but I had to set up advanced SIP settings in the GUI and make explicit the codecs I wanted to allow, and also set the domain name configuration items too |
| 08:21.55 | jnc | the asterisk-gui section for codecs has been inconsistent for me on a fresh install, might just be my web browser (stale version of firefox) |
| 08:22.38 | kev009 | what sections did you have to edit? |
| 08:23.16 | jnc | I hit up Options -> [Advanced] |
| 08:23.21 | jnc | then SIP -> ... |
| 08:24.23 | jnc | Domain, {Codecs} => Allowed Codecs (adpcm, gsm |
| 08:24.44 | jnc | and well I also had to patch Ekiga since I was using Ekiga as a softphone and it has a bug |
| 08:24.51 | jnc | hope that helps |
| 08:25.09 | JT | well this is #asterisk |
| 08:25.14 | JT | we don't do guis here |
| 08:25.25 | jnc | JT: thanks, I'd mentioned that |
| 08:25.48 | jnc | sorry for the spam |
| 08:27.07 | Juggie | i'm sure we can support asterisk's own gui |
| 08:27.15 | Juggie | though, i dont think many people have expirence with it |
| 08:27.21 | Juggie | 3rd party are definitally not however |
| 08:29.22 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 08:31.35 | *** join/#asterisk saftsack (n=saftsack@pD9E06820.dip.t-dialin.net) |
| 08:35.55 | *** join/#asterisk Star568 (i=Star@24-205-92-96.dhcp.psdn.ca.charter.com) |
| 08:36.02 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 08:36.52 | *** join/#asterisk mcmx2 (n=dan@5ac0690d.bb.sky.com) |
| 08:37.01 | JT | Juggie: there's a seperate channel for the gui |
| 08:37.19 | Star568 | i am using * 1.2.12.1. Dial out command with HmL(x:y:z) option. the music on hold is so CHOPPY |
| 08:37.32 | Star568 | any body knows how to fix it? |
| 08:38.36 | Star568 | i tried gsm, mp3, wav all, but still the same thing |
| 08:40.06 | Star568 | is it a codec issue or clock issue? |
| 08:41.33 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 08:41.44 | ManxPower | do you have a slow system? |
| 08:54.10 | *** join/#asterisk mcmx3 (n=dan@5ac51e2a.bb.sky.com) |
| 08:54.19 | mcmx3 | hey anyone around? |
| 08:56.58 | mcmx3 | i got major probs with my setup when i put outside numbers into ring groups and a call comes in the outside number rings and when answered it is silent untill the incoming phone rings off then all hangs up |
| 09:00.14 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:00.21 | Strom_M | mcmx3: I have absolutely no idea what the hell you're trying to accomplish |
| 09:01.07 | mcmx3 | lol its hard to explain... |
| 09:01.19 | Strom_M | first off |
| 09:01.25 | Strom_M | are you using freepbx / trixbox? |
| 09:01.41 | mcmx3 | that is how i initially setup the system |
| 09:02.02 | mcmx3 | this is not a problem with the GUI tools though |
| 09:02.23 | Strom_M | yeah, thats what you think |
| 09:02.38 | Strom_M | have you looked at the disaster of dialplan code that thing uses? |
| 09:02.56 | Strom_M | ~trixbox |
| 09:03.07 | jbot | Trixbox is a full linux distro that includes *, FreePBX, and other 3rd party add-ons. It is these things on top of * which make it seriously painful to support and hence you will find little help here for it. Try asking in #freepbx , or their forums at http://www.trixbox.org/modules/newbb/ |
| 09:03.40 | mcmx3 | i see |
| 09:04.03 | mcmx3 | the config files did look a nightmare lol |
| 09:04.34 | Strom_M | yeah, i'd recommend that you not run any kind of production system on that platform |
| 09:05.01 | mcmx3 | why you say that? |
| 09:05.30 | mcmx3 | just because its difficuilt to bug find? |
| 09:05.56 | Strom_M | trust me - ive tried it |
| 09:06.00 | Strom_M | it's difficult to debug and it's prone to developing really weird problems as time goes on |
| 09:06.15 | mcmx3 | i see |
| 09:06.42 | mcmx3 | it seems like a lot of work programming it by hand though |
| 09:11.52 | jnc | mcmx3: don't do it by hand with a trixbox system as the base |
| 09:12.36 | jnc | start with a debian etch install, asterisk-1.2.x stable release |
| 09:12.53 | jnc | takes about 2-4 days to get a grasp of the extensions.conf syntax |
| 09:15.28 | mcmx3 | i see |
| 09:16.28 | jnc | mcmx3: I found it helpful myself to leverage the comments from extensions.ael.sample of asterisk 1.4 branch |
| 09:16.49 | *** join/#asterisk friedrich| (n=friedric@e177247098.adsl.alicedsl.de) |
| 09:16.49 | jnc | that's for AEL, which is flattened into extensions.conf syntax |
| 09:17.02 | Strom_M | no, not as much as you'd think |
| 09:17.03 | Strom_M | doing it by hand also allows you much greater control and flexibility |
| 09:17.22 | jnc | the comments are very helpful to understand what contexts are |
| 09:18.24 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:18.47 | *** join/#asterisk MaartenB (n=Maarten@213-73-178-251.cable.quicknet.nl) |
| 09:18.52 | MaartenB | hey everyone |
| 09:18.59 | mcmx3 | yeh |
| 09:19.16 | MaartenB | I am having problems with incomming calls, I have no idea why, but they are handled by a timeout |
| 09:19.38 | MaartenB | it says "Timeout on IAX2/speakup01-1" where speakup01 is my voip provider |
| 09:21.17 | MaartenB | and I think it is giving this timeout on "Set("IAX2/speakup01-1", "CALLERID(name)=Private") in new stack" |
| 09:21.24 | MaartenB | which is akward |
| 09:31.53 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:36.23 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:38.28 | Vec | Has anyone been able to compile speex with Asterisk 1.4.2 on a x64 system ? |
| 09:49.15 | kev009 | I wish there was a better compact flash asterisk distro |
| 09:49.27 | kev009 | something like m0n0wall for * |
| 09:55.39 | *** join/#asterisk nemski (n=nemesis@65.111.176.146) |
| 09:58.26 | Vec | kev009 : I also wish. |
| 10:00.19 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
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| 10:48.28 | *** part/#asterisk kev009 (n=kev009@ip70-162-43-70.ph.ph.cox.net) |
| 11:02.57 | *** join/#asterisk sashion (n=djbdsf@dsl-241-227-89.telkomadsl.co.za) |
| 11:03.44 | sashion | whats the chances of asterisk going threaded with calls. that way if asterisk crashes, your calls in progress remain active, instead of being cut off |
| 11:05.08 | tzafrir_laptop | Vec, http://packages.debian.org/speex shows it on amd64 as well. |
| 11:06.01 | tzafrir_laptop | asterisk has a thread for each channel (which is basically one "leg" of the call) |
| 11:07.10 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 11:07.33 | sashion | tzafrir_laptop: true, however, if asterisk crashes, those calls get cut... |
| 11:08.23 | tzafrir_laptop | who exactly should maintain them alive (keep pumping audio frames, and such) |
| 11:08.58 | sashion | tzafrir_laptop: well I'm not programmer, but say like a helper application that just knows the channel data |
| 11:09.19 | tzafrir_laptop | helper application run where? |
| 11:09.29 | tzafrir_laptop | on the same server? on another server? |
| 11:09.55 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 11:11.21 | sashion | same server... |
| 11:12.53 | tzafrir_laptop | sashion, and if the server crashes? |
| 11:13.49 | sashion | tzafrir_laptop: yeah ok that we can't help unless you run a form of dundi (but you'll still loose calls that are on that system) |
| 11:14.11 | sashion | what I'm thinking of is incase the asterisk process crashes (which is does from time to time)... |
| 11:15.08 | tzafrir_laptop | sashion, asterisk shouldn't crash. If it does crash, don't count on it to become instantly available. |
| 11:15.31 | tzafrir_laptop | (in other words: solve the problem anddon't try patching it with a buggy "safe_asterisk") |
| 11:16.57 | sashion | tzafrir_laptop: I hear what you're saying... but in the event of preventing call drops when asterisk crashes, whats the possibility of having a "helper" app that is spawned on an icnoming call outside of asterisk core process |
| 11:17.10 | sashion | that way if asterisk does crash, atleast your calls stay up... |
| 11:17.25 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 11:18.35 | tzafrir_laptop | copying all that information as the tim of a crash (when you can't really count on it to be non-curropted) is probably not the best idea |
| 11:19.11 | sashion | true... I mean it would be a great concept... |
| 11:19.31 | sashion | but I mean, what is the best way to go about building a stable system ? |
| 11:28.43 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 11:29.00 | *** join/#asterisk threat (n=phix@60-240-43-214.static.tpgi.com.au) |
| 11:32.15 | DrukenLPY | asterisk it's self shouldn't crash... course... i do have one machine that it does sometimes all of a sudden drop... |
| 11:33.26 | DrukenLPY | but i also haven't done any investigating as to why... since it's only my home server, and only my calls are affected |
| 11:34.57 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:35.45 | threat | I have a Wildcard TDM400P, are there any known problems with some analog POTS phones when connected to this card? |
| 11:35.56 | threat | or issues with asterisks itself? |
| 11:36.22 | threat | I keep getting cut out when I use a panasonic cordless phone |
| 11:36.36 | threat | the cabled phone works fine |
| 11:40.33 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 11:41.37 | tzafrir_laptop | threat, what problems do you notice? |
| 11:42.16 | threat | tzafrir_laptop, read up :) |
| 11:42.23 | tzafrir_laptop | sashion, for that your starting point may be having two servers |
| 11:42.26 | threat | cut outs |
| 11:42.39 | threat | phone rings, I pick up, it cuts out in a few secs |
| 11:43.22 | tzafrir_laptop | threat, just the audio, or also the call itself? |
| 11:43.31 | threat | tzafrir_laptop, also another annoyance I have found is if a person rings then hangs up, my phone still rings for a few more seconds |
| 11:43.36 | threat | the call |
| 11:43.43 | threat | well I assume the call |
| 11:43.54 | sashion | tzafrir_laptop: all good having 2 servers, however, running 2 servers as a cluster proves to be difficult when using an inbound call center |
| 11:44.47 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 11:45.25 | tzafrir_laptop | don't assume. set verbose 3 |
| 11:45.41 | tzafrir_laptop | Or run 'show channels' occasionally |
| 11:46.48 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 11:47.41 | [TK]D-Fender | threat, You "continued ringing in reference to an analog line on zaptel? |
| 11:49.24 | threat | ok |
| 11:49.28 | threat | [TK]D-Fender, what? |
| 11:49.46 | threat | [TK]D-Fender, yes |
| 11:49.53 | [TK]D-Fender | threat, for the call you mentioned cuts out. Was this call coming in from an analog phone line? |
| 11:49.58 | threat | the analog phone is connected to the asterisk box |
| 11:50.16 | threat | [TK]D-Fender, ummm probably |
| 11:50.40 | threat | [TK]D-Fender, incomming calls can come from landline or IAX VoIP provider |
| 11:51.12 | [TK]D-Fender | threat, Well the problem with analog is that you get a ring indication from telco and * will keep ringing until after it is sure a ring has been SKIPPED. It can't know instantly that the line will be dead or cut off in a fraction of a second |
| 11:51.18 | threat | [TK]D-Fender, tzafrir_laptop ok so I should use the asterisk console, set debug higher and get some more useful information? |
| 11:51.42 | threat | [TK]D-Fender, it can't? but the ring tone changes |
| 11:51.46 | [TK]D-Fender | landline is understandable, but IAX carrier should be on PRI on their side and be near-instant. |
| 11:51.57 | tzafrir_laptop | by default debug information is not sent to the console. And it would generally overwhelm the console |
| 11:52.11 | threat | hmm |
| 11:52.14 | [TK]D-Fender | threat, to know that its stopped it has to skip an ENTIRE ring + a little bit extra |
| 11:52.22 | tzafrir_laptop | threat, so look at logger.conf on ways of sending this to a log file or the console. |
| 11:52.32 | [TK]D-Fender | analog = ass |
| 11:53.10 | threat | [TK]D-Fender, ok, so there is no way to reduce the extra bit of time it waits? |
| 11:53.26 | threat | tzafrir_laptop, ok |
| 11:53.27 | [TK]D-Fender | threat, in indications.conf if you're lucky, don't know off-hand |
| 11:53.36 | threat | [TK]D-Fender, ok thank you :) |
| 11:54.01 | threat | [TK]D-Fender, yes I am still learning asterisk :) I am trying to debug problems on an already configured system |
| 11:54.19 | tzafrir_laptop | indications.conf? what does it have to do with this? |
| 11:54.32 | threat | tzafrir_laptop, I asked two questions |
| 11:54.44 | sashion | does chan_agent by default mix the two legs of a recording (*-in.wav and *-out.wav) after the call is complete? |
| 11:54.51 | sashion | if so, how can you turn that feature off ? |
| 11:54.51 | threat | tzafrir_laptop, 1) drop outs 2) phone keeps ringing even though the caller hang up |
| 11:54.53 | [TK]D-Fender | tzafrirfor his incoming FXO, for the ring timeout. No chance to tweak zaptels treatment? |
| 11:55.13 | tzafrir_laptop | zapata.conf ? |
| 11:55.20 | [TK]D-Fender | tzafrir_laptop, as far as ring time-out is concerned |
| 11:55.41 | tzafrir_laptop | It also depends on the disconnection supervision |
| 11:55.43 | [TK]D-Fender | tzafrirnormally your cadences, etc are in indications.conf IIRC |
| 11:56.02 | tzafrir_laptop | If he must resort to busydetect, there's no real escape here |
| 11:56.16 | [TK]D-Fender | tzafrir_laptop, Should KS also do a reversal/cut for end-of-ringing? |
| 11:56.46 | tzafrir_laptop | basically, yes. But what do you mean by "reversal"? |
| 11:56.59 | [TK]D-Fender | tzafrir_laptop, No, not busydetect, just though the INBOUND ring cadence (not TONE), was in there as well. I could be entirely mistaken :) |
| 11:57.33 | [TK]D-Fender | tzafrir, CPD is normally either a polarity reversal, or a cold-cut depending |
| 11:58.02 | threat | the diconnection supervision ay |
| 11:58.46 | threat | PRI is some type of ISDN / digital line? |
| 11:59.10 | [TK]D-Fender | threat, PRI is a signalling type of T1/E1/J1 |
| 11:59.24 | threat | [TK]D-Fender, ok |
| 11:59.28 | [TK]D-Fender | threat, Includes full call progress indications |
| 11:59.34 | threat | [TK]D-Fender, expensive |
| 12:00.15 | [TK]D-Fender | threat, SIP supports pretty much the same informations set so when used with a SIP phone it'll know the moment calls get answered, ring on inbound, etc. |
| 12:00.20 | mcmx3 | hi is there an alternative to callprogress=no if i have it no, my ring groups to outside numbers work if it is yes then it loses the ability to hang up detect (Im in UK btw) |
| 12:00.25 | [TK]D-Fender | threat, expensive depends on your needs. |
| 12:00.32 | [TK]D-Fender | threat, for HOME... uhh.. yeah ;) |
| 12:00.37 | threat | [TK]D-Fender, one line SOHO configuration |
| 12:01.01 | Vec | Off topic: Looking for a premium rated number in the UK, that I can recieve over IAX/SIP, anyone know where I can look ? |
| 12:01.11 | [TK]D-Fender | mcmx3, callprogress=yes actually means "randomly disconnect my calls". |
| 12:01.26 | threat | [TK]D-Fender, why would you want that for? |
| 12:01.32 | [TK]D-Fender | Vec, Thats not really OT so much..... |
| 12:01.41 | [TK]D-Fender | threat, PRI? |
| 12:01.45 | threat | [channels] |
| 12:01.45 | threat | busydetect => yes |
| 12:01.45 | threat | ;callprogress => yes |
| 12:01.51 | Vec | [TK]D-Fender : yeh, well some people are picky |
| 12:01.53 | [TK]D-Fender | threat, EW |
| 12:02.01 | threat | [TK]D-Fender, :) |
| 12:02.04 | [TK]D-Fender | threat, ditch busydetect as well. |
| 12:02.11 | threat | [TK]D-Fender, why? |
| 12:02.23 | threat | [TK]D-Fender, problems? |
| 12:02.35 | [TK]D-Fender | Vec, You miss our major diversions to politics, movies, music, martial-arts, etc :) |
| 12:02.58 | threat | [TK]D-Fender, how about I paste my entire configuration files on some paste site for you to look over :P |
| 12:03.42 | Vec | [TK]D-Fender : I guess I don't spend that much time on #asterisk to notice :), too busy trying to get things to work, like SPEEX :( |
| 12:04.07 | Vec | ~pastebin |
| 12:04.19 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 12:04.19 | *** join/#asterisk saftsack (n=saftsack@pd9e06820.dip.t-dialin.net) |
| 12:04.43 | [TK]D-Fender | threat, Sure... I'll pass you paypal address and I'll let you know what the "renovation" fee will be like :) |
| 12:06.04 | Vec | ~help |
| 12:06.16 | threat | [TK]D-Fender, ok :) how much? |
| 12:07.10 | sashion | whats the possibility of asterisk chowing CPU if you are using chan_agent with recording and 118 calls come in at once... |
| 12:07.30 | sashion | since I see in the code, joinfiles just runs soxmix |
| 12:07.36 | Vec | cnn is at http://www.cnn.com |
| 12:07.42 | Vec | Where is cnn ? |
| 12:08.27 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:09.29 | Vec | Has anyone got speex 1.2 working with asterisk, or should I stick to 1.1 ? |
| 12:10.15 | tzafrir_laptop | Vec, About a year ago I rebuilt the Debian speex package with 1.2 and rebuilt asterisk using it |
| 12:13.36 | Vec | tzafrir_laptop : the problem I am having is firstly when using speex 1.0.4, which is availible on RHN, and I try to run asterisk I get "undefined symbol speex_decode_int", I wanted to try 1.2 but it does not have a RPM-devel for x64, so I am now trying to compile it from source, but I sometimes get dependince failures when I do ./configure --with-speex=/spfol ? |
| 12:14.02 | Vec | I am running RHEL 4 |
| 12:14.12 | Vec | with asterisk 1.4.2 |
| 12:23.42 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:24.35 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:33.07 | Dr-Linux | Vec: same :) |
| 12:33.19 | Dr-Linux | i'm going to upgrade my servers to 1.2.17 |
| 12:33.27 | mcmx3 | how do i get proper hangup detection on a TDM400 in the UK? |
| 12:34.13 | Vec | Dr-Linux : upgrade ? what u running at the moment ? |
| 12:34.44 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 12:35.09 | Dr-Linux | Vec: 1.2.0 |
| 12:35.27 | Dr-Linux | Vec: also zaptel etc |
| 12:35.40 | Dr-Linux | last week i'm upgraded my other servers as well |
| 12:35.59 | Vec | Dr-Linux : I am running 1.4.2 zaptel 1.4.1 so I don't think upgrading will help. |
| 12:36.13 | Dr-Linux | ohh |
| 12:36.17 | Dr-Linux | Vec: what's your problem? |
| 12:37.00 | Vec | read a few lines up get, "undefined symbol speex_decode_int" when asterisk tries to load the speex module. |
| 12:37.19 | *** join/#asterisk Owlet (n=gufo@ip-245-22-dyn.adsl.intratec.it) |
| 12:37.51 | Owlet | Hi all |
| 12:37.52 | Vec | that is with speex 1.0.4 though, I am going to try the other versions just now and I'll tell u what happens |
| 12:38.04 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 12:38.26 | Owlet | maybe someone can help me... |
| 12:38.52 | Owlet | i'm using asterisk 1.4.2, and I have problem with the ring tone of outbound calls with a grandstream 486 ata |
| 12:39.20 | Owlet | with the same server softphones are ok |
| 12:40.02 | Owlet | but the ata, only for outgoing calls, either doesn't ring or rings even if the other party isn't already called by my proxy server |
| 12:40.14 | Owlet | depending on the dial() function specified |
| 12:40.45 | Owlet | if it's exten => _X.,1,Dial(SIP/${EXTEN:0}@provide,60,) |
| 12:40.52 | Owlet | no ring |
| 12:41.07 | Owlet | if exten => _X.,1,Dial(SIP/${EXTEN:0}@provider,60,r) it rings even if the call isn't established |
| 12:41.54 | Owlet | is there a known workaround or my ata configuration/asterisk configuration is buggy? |
| 12:54.32 | Dr-Linux | ehh |
| 12:55.02 | Dr-Linux | i didn't delete /usr/lib/asterisk/modules/.... and i upgraded :S |
| 12:55.26 | Dr-Linux | will new module override new ones? :S |
| 12:55.34 | Dr-Linux | or it's wrong .. i should do again |
| 13:03.01 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 13:04.04 | EmleyMoor | I am getting very poor performance on a local echo test on phones on this computer - what are good things to check? |
| 13:08.43 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 13:09.27 | *** join/#asterisk moranil (n=moranil@122.162.67.129) |
| 13:20.56 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D4F3.access.telenet.be) |
| 13:25.21 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 13:33.18 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-182-254.buckeyecom.net) |
| 13:44.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:00.30 | *** join/#asterisk `p4r14h`work (n=josh@72.22.238.36) |
| 14:01.30 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 14:02.14 | Falle | Hi all! Is there some variable that i can use in extensions.conf to get the clients IP-address? |
| 14:03.45 | sashion | Falle: Just use an agi script to get the IP address and then make a variable |
| 14:04.37 | Falle | okey. i just hoped there was a quicker solution than that :) |
| 14:06.11 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 14:07.09 | GreyFoxx | SIPPEER() has an ip value |
| 14:07.47 | GreyFoxx | ${SIPPEER(${CALLERIDNUM}:ip)} |
| 14:08.03 | GreyFoxx | At least it works in 1.2.x |
| 14:08.15 | Falle | thanks :) |
| 14:17.53 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
| 14:18.21 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 14:32.08 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 14:32.29 | *** join/#asterisk sooli (n=sooli@AMontpellier-156-1-180-55.w90-36.abo.wanadoo.fr) |
| 14:32.33 | sooli | Hi |
| 14:33.11 | sooli | I'm a newbie and i want to setup my wengo account with asterisk ... but i have always pbx.c:1700 pbx_extension_helper: No application 'Playback' for extension |
| 14:33.39 | sooli | I have erros for each functions... Dial, Playback, etc ... |
| 14:36.23 | *** part/#asterisk moranil (n=moranil@122.162.67.129) |
| 14:36.55 | sooli | in my log i didn't see Dial and Playback as function registered ! |
| 14:36.55 | mcmx3 | hi i cant seem to get asterisk to hang up after a call im in the UK any ideas? |
| 14:37.06 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.17 | bulle | mcmx3: sip ? |
| 14:41.33 | mcmx3 | no zap - tdm400p |
| 14:42.46 | *** join/#asterisk _Roman (n=roman@87.254.76.159) |
| 14:43.16 | kumbalae | mcmx3: did you the zone to uk ? |
| 14:43.55 | mcmx3 | yeah every thing says uk |
| 14:45.21 | sooli | how can i access to Dial, Background or Mp3player ? |
| 14:46.21 | tzafrir_laptop | mcmx3, "no zap"? could you pastebin your zapata.conf ? |
| 14:47.09 | mcmx3 | yp.. |
| 14:49.14 | sooli | anyone ? |
| 14:50.25 | mcmx3 | http://www.pastebin.ca/439764 |
| 14:54.23 | _Roman | Can anyone please recommend a good book on Asterisk 1.4 |
| 14:54.42 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 14:55.42 | mcmx3 | any solutions on that? |
| 14:57.27 | bulle | _Roman: im not sure there is any 1.4 specific books out yet 1.4 is pretty new |
| 14:57.42 | bulle | _Roman: the book, should still be a good read, if you run 1.4 |
| 14:58.19 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 14:59.46 | _Roman | bulle: ok thanks, I have been reading the book. It is pretty good, just didn't want to confuse myself with stuff that may have changed. |
| 15:00.24 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 15:00.50 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 15:02.10 | jm|laptop | chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission OGE3YjM1YzkwNjExOWEwMGUxMTQyMjRlMGQxYjczMDc. for seqno 1 (Critical Response) |
| 15:02.13 | jm|laptop | what's all that then?! |
| 15:02.24 | _Roman | The main problem that I am having is getting festival to work, I am using the instructions in the book. I keep getting the following error: WARNING[23434]: utils.c:725 tvfix: warning too large timestamp -1211197404.136543853. I see that there was a bug reported (http://bugs.digium.com/view.php?id=8754) which I presume that this fix has not made it into the main 1.4 relese yet? |
| 15:04.46 | *** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net) |
| 15:05.13 | knarfly | anyone else using the X101P clone? |
| 15:05.59 | knarfly | I'm having some troubles...sometimes * picksup and starts the welcome message when I'm on the other extension |
| 15:15.42 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 15:18.39 | Qwell | knarfly: yeah, it does that |
| 15:20.30 | knarfly | Qwell: Any workaround known for it...or why does it do it? |
| 15:20.40 | Qwell | because it's a crappy card |
| 15:20.46 | knarfly | hehe |
| 15:20.50 | Qwell | ~ygwypf |
| 15:21.00 | jbot | hmm... ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 15:21.11 | knarfly | my words exactly....some nappy card eh! |
| 15:21.30 | jm|laptop | is anyone using a binat solution for their asterisk box? |
| 15:21.36 | jm|laptop | it seems to be causing me problems :/ |
| 15:22.06 | knarfly | jbot: that's not entirely the case...you can spend $30 to test before you spend $400 and find out it's not what you wanted. |
| 15:22.37 | knarfly | what is I spent $400 on a digium card and it too answered during another call....! |
| 15:22.54 | knarfly | if |
| 15:23.20 | Qwell | knarfly: then you'd call support |
| 15:26.05 | knarfly | yes but doesn't that particular problem sound like one of electrical frequency... |
| 15:26.40 | knarfly | I first thought it was the call waiting signal causing the problem but then I found out it was just the unstable nature of the card. |
| 15:28.30 | *** join/#asterisk Mnabil (n=Mnabil@82.201.214.245) |
| 15:29.11 | Mnabil | hello, how can i make video one to one with asterisk ? |
| 15:29.14 | *** join/#asterisk clinthome (n=clinthom@c-71-63-5-40.hsd1.va.comcast.net) |
| 15:33.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:34.24 | knarfly | \q |
| 15:39.07 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:42.59 | *** join/#asterisk khronos (n=khronos@c-76-110-134-230.hsd1.fl.comcast.net) |
| 15:45.17 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:45.44 | khronos | Hi guys. |
| 15:47.11 | khronos | Anyone had trouble building Zaptel 1.2.16 on Centos 5? |
| 15:52.00 | e-ddie | not really |
| 15:52.09 | e-ddie | i dont use systems named by their price |
| 15:52.41 | *** join/#asterisk etfonhomey (n=etfonhom@74-140-213-69.dhcp.insightbb.com) |
| 15:53.26 | *** join/#asterisk dweazle (i=dweazle@s5591373c.adsl.wanadoo.nl) |
| 15:54.11 | dweazle | hi all, i've got asterisk sip to work and i can call from my pc to my phone (nokia n80), but there's no sound.. do i need to configure codecs or something? |
| 15:54.50 | dweazle | i'm using ekiga on the pc site of things |
| 15:54.53 | dweazle | side* |
| 15:55.33 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:55.51 | etfonhomey | dweazle, sounds like NAT issues. |
| 15:56.20 | dweazle | i don't NAT, this is all internal |
| 15:56.34 | dweazle | maybe it's my firewall though |
| 15:56.51 | dweazle | i've opened up 5060 and 5061 UDP, should be enough, right? |
| 15:57.01 | etfonhomey | dweazle, NO |
| 15:57.05 | dweazle | oh |
| 15:57.07 | dweazle | :> |
| 15:57.23 | etfonhomey | dweazle, those are only for SIP, not the actually media stream |
| 15:57.29 | etfonhomey | *actual |
| 15:57.33 | *** join/#asterisk dasenjo (n=be185974@acuario.unicauca.edu.co) |
| 15:57.34 | dweazle | ok |
| 15:57.52 | dweazle | so .. then what ports do i need to open? |
| 15:58.51 | etfonhomey | dweazle, Let me get this straight, you're going across a firewall, but not NAT'ing? |
| 15:59.13 | dweazle | yup, i'm briding from my wlan to my lan |
| 15:59.16 | dweazle | bridging |
| 15:59.21 | dasenjo | Hi! I'm having SIP one way audio, I can't hear any message from my asterisk server. I used wireshark to debug .. .there is _one_ RTP packet from asterisk to the peer and a lot from the peer to asterisk, any idea? |
| 15:59.54 | etfonhomey | dweazle, if you're on the same network, why are you traversing a firewall? |
| 16:00.18 | dasenjo | I have used a gs bt100, ekiga and xlite to test with the same results |
| 16:00.24 | dweazle | lan = 10.0.0.x , wlan = 10.0.1.x .. asterisk is running on the server, which also bridges between lan and wlan (and internet), pc is on lan, nokia is on wlan |
| 16:01.56 | dweazle | (i've got a wifi card in my server in master mode) |
| 16:02.39 | etfonhomey | dweazle, ok. |
| 16:02.55 | etfonhomey | dweazle, why dont you put your wlan and lan on the same network? |
| 16:03.08 | dweazle | ehm, because i don't trust wlan traffic :) |
| 16:03.12 | dasenjo | what could be a reason for asterisk stops sending RTP packets? |
| 16:04.10 | etfonhomey | well, your * server has an IP address in the wlan network, right? |
| 16:04.25 | dweazle | yup |
| 16:04.35 | etfonhomey | dweazle, and it has an IP address in your lan network, right? |
| 16:04.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 16:04.40 | dweazle | yup |
| 16:05.05 | etfonhomey | And you're trying to send data from a device in one network to a device in the other network, right? |
| 16:05.16 | dweazle | but the nokia can connect directly to the lan, i've opened up the port |
| 16:05.19 | dweazle | it's bridged |
| 16:05.38 | etfonhomey | So you're running some routing process on your * server? |
| 16:05.39 | dweazle | i can ping the phone from my lan and everything |
| 16:05.42 | *** join/#asterisk hal2k (n=am@2002:5470:9fd9:0:0:0:0:1) |
| 16:05.51 | etfonhomey | To route between networks. |
| 16:06.10 | dweazle | . . ye .. something like /proc/net/ip_forward = 1 |
| 16:06.53 | etfonhomey | In your sip.conf entries for the two devices, do you have canreinvite=no? That will force the media to be bridged by * as well. |
| 16:07.24 | dweazle | yes i have that |
| 16:07.52 | dweazle | maybe i should just tcpdump and figure out which ports it uses |
| 16:07.55 | etfonhomey | dweazle, why don't you start by opening up everything and see if it works, then close it down. |
| 16:07.58 | dweazle | but it's prolly a dynamic range or something |
| 16:08.08 | etfonhomey | It is dynamic |
| 16:08.19 | dweazle | well now that i know there's a seperate media stream i might just try that :) |
| 16:08.22 | dweazle | thanks |
| 16:08.32 | etfonhomey | That's why SIP doesn't work through NAT |
| 16:08.41 | etfonhomey | And you're basically doing NAT. |
| 16:08.52 | etfonhomey | From 10.0.1.x to 10.0.0.x |
| 16:08.54 | dweazle | it's not translating anything , it's just a firewall sitting in between |
| 16:09.38 | ManxPower | dweazle: Port 5060/UDP and 10000-20000/UDP |
| 16:09.48 | *** join/#asterisk ToyMan (n=Stuart@ool-45784fde.dyn.optonline.net) |
| 16:09.49 | dweazle | ManxPower: thanks i'll try that |
| 16:10.02 | ManxPower | Plus whatever ports the other side decides to use. |
| 16:10.08 | etfonhomey | :) |
| 16:10.17 | etfonhomey | Can someone define NAT for me please? |
| 16:10.23 | Qwell | ~nat |
| 16:10.34 | jbot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:10.34 | ManxPower | Network Address Translation |
| 16:11.27 | etfonhomey | No, is dweazle not NAT'ing |
| 16:12.06 | etfonhomey | No, he's not. |
| 16:12.08 | etfonhomey | Nevermind. |
| 16:12.11 | dweazle | lol :) |
| 16:12.26 | etfonhomey | He's just restricting what can get routed between the 2 nets, right? |
| 16:13.01 | dweazle | indeed |
| 16:13.06 | dweazle | i'm paranoid |
| 16:13.06 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 16:13.26 | ManxPower | most of the time on this channel when someone says "firewall", they mean "nat". |
| 16:14.33 | etfonhomey | ManxPower, are the TDM400P cards crap? |
| 16:15.00 | blitzrage | "Paranoia, paranoia, everyone's coming to get me. Just say you never met me. I'm going underground with the moles, hear the voices in my head. I swear to God it sounds like they're snoring, but if you're bored then you're boring" |
| 16:16.29 | etfonhomey | dweazle, read up on SIP. In the first part of a SIP session, the 2 ends negotiate what port to send the media stream over. |
| 16:16.34 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:16.48 | dweazle | ok |
| 16:18.01 | \BAF64\ | [Apr 14 12:17:38] NOTICE[3741] chan_local.c: No such extension/context executecommand@asterisk_guitools creating local channel |
| 16:18.01 | \BAF64\ | [Apr 14 12:17:38] NOTICE[3741] channel.c: Unable to request channel Local/executecommand@asterisk_guitools |
| 16:18.14 | \BAF64\ | Any idea why I get tons of that when I try to go to the http asterisk gui setup page? |
| 16:19.06 | etfonhomey | dweazle, I think in * you can say what port you want to start at when negotiating. I could be mixing that up with a config on my Polycom phones. |
| 16:19.41 | etfonhomey | ManxPower, do you remember last night when I was talking about presence problems with a buddy on the line key of my 301? |
| 16:20.48 | etfonhomey | dweazle, do you have the Asterisk: The Future of Telephony PDF? |
| 16:20.56 | dweazle | etfonhomey: yes that's what i used to configure * |
| 16:20.59 | *** join/#asterisk saftsack (n=saftsack@pD9E06820.dip.t-dialin.net) |
| 16:21.02 | dweazle | i'm a total noob on this topic :) |
| 16:21.25 | Hmmhesays | fun |
| 16:21.30 | etfonhomey | dweazle, I'm only at the nooB + 1 level. |
| 16:21.36 | Hmmhesays | fun fun |
| 16:22.03 | dweazle | cool it works :D |
| 16:22.08 | dweazle | i just opened up UDP on all ports |
| 16:22.12 | dweazle | haha |
| 16:22.14 | dweazle | nice! |
| 16:22.30 | Hmmhesays | 10000-20000 is what the sample configs use |
| 16:22.31 | etfonhomey | dweazle, did it work? |
| 16:22.34 | Hmmhesays | you can change that |
| 16:22.36 | dweazle | etfonhomey: yes :) |
| 16:22.41 | Hmmhesays | rtp.conf I believe |
| 16:22.48 | etfonhomey | dweazle, yes, it's rtp.conf |
| 16:22.51 | dweazle | ah ok :) |
| 16:23.05 | dweazle | rtpstart=10000 |
| 16:23.05 | dweazle | rtpend=20000 |
| 16:23.09 | dweazle | good enough |
| 16:23.09 | etfonhomey | dweazle, rtpstart and rtpend |
| 16:23.26 | etfonhomey | dweazle, what phone are you using again? |
| 16:23.46 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 16:24.02 | *** part/#asterisk Owlet (n=gufo@ip-245-22-dyn.adsl.intratec.it) |
| 16:24.59 | dweazle | etfonhomey: nokia n80 |
| 16:26.00 | etfonhomey | Anyone hear with a lot of experience with the TDM400P card? |
| 16:26.00 | dweazle | mm .. now let's add sip srv records to my dns |
| 16:26.25 | etfonhomey | Qwell or ManxPower, I need professional help... ...with this stupid TDM400P card. |
| 16:26.26 | toot | what ever happened to vocal? :) i used to do dev on it then left and was just wondering |
| 16:26.47 | etfonhomey | dweazle, what SRV records are you adding? |
| 16:27.01 | etfonhomey | dweazle, never messed with SRV records... |
| 16:27.47 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 16:27.50 | Hmmhesays | they're fun |
| 16:28.10 | Hmmhesays | you can do half assed load sharing |
| 16:28.22 | etfonhomey | What is their purpose? |
| 16:28.35 | *** join/#asterisk clinthome (n=clinthom@c-71-63-5-40.hsd1.va.comcast.net) |
| 16:28.52 | Hmmhesays | service records |
| 16:28.56 | Hmmhesays | in dns |
| 16:29.06 | Hmmhesays | they identify services like... sip |
| 16:29.32 | dweazle | yeah so people can call me over the internet :) |
| 16:29.33 | etfonhomey | Tell you where the asterisk server is or something? |
| 16:29.36 | dweazle | yup |
| 16:29.38 | Hmmhesays | yes |
| 16:30.09 | etfonhomey | OK. I've only seen them used by MS Exchange servers and Active Directory domain integrated DNS servers... |
| 16:30.30 | dweazle | they're also used for things like kerberos |
| 16:30.30 | Hmmhesays | i use them |
| 16:30.38 | dweazle | which is also what MS-AD uses indeed |
| 16:30.39 | Hmmhesays | I use them for load balancing |
| 16:30.51 | Hmmhesays | round robin dns |
| 16:31.04 | dweazle | you don't need srv records to do that, do you? |
| 16:31.17 | Hmmhesays | no, but for the endpoints i'm using it works better |
| 16:31.26 | etfonhomey | So, you need SRV records for someone to call you directly across the internet using SIP? |
| 16:31.35 | dweazle | i think so |
| 16:31.39 | Hmmhesays | um no |
| 16:31.47 | Hmmhesays | don't be silly |
| 16:31.52 | IgorG | Also it's can be used for fallback |
| 16:32.01 | dweazle | no? mm .. i thought it was something like a MX record for mail, that you needed it to be able to make sip calls to a domain |
| 16:32.25 | Hmmhesays | you can use dns to make calls across the net but they aren't required |
| 16:32.26 | dweazle | but then again, i'm still a newbie :) |
| 16:32.32 | etfonhomey | I know you can register with FWD or something similar and make calls that way. |
| 16:32.32 | Hmmhesays | you can use an ip address |
| 16:32.45 | dweazle | ah ok |
| 16:32.59 | Hmmhesays | no different than any network communication |
| 16:33.13 | etfonhomey | What's the typical use with SRV records with SIP, then. |
| 16:33.26 | Hmmhesays | to allow you to register with a server |
| 16:33.32 | Hmmhesays | by resolving a hostname |
| 16:33.47 | dweazle | mm can i also use names instead of extension numbers to call an asterisk client? |
| 16:33.51 | dweazle | if so: how? |
| 16:34.11 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 16:34.29 | etfonhomey | So, when I register with FWD, I query their DNS server at fwd.pulver.com for it's SRV records, right? |
| 16:36.24 | ManxPower | As I understand it, SRV records allow you to query a domain or hostname and find out the actual name/IP of the server. |
| 16:36.44 | dweazle | yes |
| 16:37.01 | ManxPower | For example the SIP address eric@fnordsl.org might hand back a hostname of bourbon..fnords.org as the actual SIP server |
| 16:37.42 | ManxPower | dweazle: exten => satan,1,Noop(Now I have your soul, my pretty) |
| 16:37.57 | dweazle | hah :D |
| 16:38.20 | etfonhomey | ManxPower, did you see my previous message to you? |
| 16:38.56 | ManxPower | (11:26:09) etfonhomey: Qwell or ManxPower, I need professional help... |
| 16:38.59 | ManxPower | I suggest a therapist. |
| 16:39.14 | etfonhomey | That response was NOT predictable... |
| 16:39.38 | etfonhomey | I was talking about the even earlier one about the Polycom presence issue I brought up last night. |
| 16:39.55 | ManxPower | etfonhomey: 1.4 or 1.2 |
| 16:39.57 | Hmmhesays | heh |
| 16:40.00 | ManxPower | ? |
| 16:40.39 | etfonhomey | ManxPower 1.4, [TK]D-Fender had the exact same issue with a 301 he had. |
| 16:41.13 | ManxPower | Can't help you with 1.4 other than to say that 1.4.0 was basically broken when it comes to presence. |
| 16:41.35 | *** join/#asterisk deltaray (n=deltaray@static1-66-244-85-183.stfd.smithvilledsl.net) |
| 16:41.42 | etfonhomey | Well, the exact same config (only changing MAC addresses on the configs) works on a 501. |
| 16:42.03 | ManxPower | I also said that I don't do presence on anything except for 601s |
| 16:42.09 | etfonhomey | Anywho, wanna help a poor soul with a TDM400P? |
| 16:42.27 | deltaray | Would there generally be and problems with using asterisk on a server that acts as an ecommerce webserver? Mainly, I'm looking for any limitations |
| 16:42.37 | etfonhomey | I'm trying to figure out if the problem is the telco or the card. |
| 16:42.47 | mrdigital | deltaray: pm |
| 16:43.06 | ManxPower | deltaray: VoIP is very sensitive to spikes in CPU usage. |
| 16:43.24 | ManxPower | Generally it is recommended that you run Asterisk on a dedicated server. |
| 16:44.14 | etfonhomey | ManxPower, do you not do anything with the analog cards? |
| 16:44.21 | deltaray | ManxPower: Thanks |
| 16:45.04 | ManxPower | etfonhomey: you have not described your problem |
| 16:46.03 | etfonhomey | ManxPower, I'm getting a random fast busy on an incoming call. |
| 16:46.21 | etfonhomey | ManxPower, I've got 2 FXO pors on the card. |
| 16:46.31 | ManxPower | turn off busydetect and callprogress |
| 16:47.06 | ManxPower | call from cell or POTS -> Asterisk gives the caller a fasy busy? |
| 16:47.34 | etfonhomey | That's correct. What are the defauls for busydetect and callprogess? |
| 16:47.47 | etfonhomey | (if you leave them out of the config?) |
| 16:47.51 | ManxPower | off |
| 16:48.21 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 16:48.43 | ManxPower | What do you see on the CLI? |
| 16:49.01 | etfonhomey | Here's the current zapata.conf: http://www.pastebin.ca/439898 |
| 16:50.35 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-170.dsl1.mdl.ny.frontiernet.net) |
| 16:51.09 | etfonhomey | http://www.pastebin.ca/439901 |
| 16:51.21 | etfonhomey | It just happened. The CLI results are at that pastebin. |
| 16:52.18 | etfonhomey | FYI, this box is * 1.2.17 and zaptel 1.2.16. |
| 16:53.03 | \BAF64\ | what's the easiest way to set up a dialplan |
| 16:53.49 | \BAF64\ | i just want the asterisk-gui stuff to work |
| 16:53.53 | \BAF64\ | like record a voice menu |
| 16:53.57 | Corydon76-home | Easiest way is to pay someone else to do it |
| 16:54.24 | \BAF64\ | ... |
| 16:54.33 | \BAF64\ | there isnt just a basic dialplan that will work |
| 16:54.39 | etfonhomey | ManxPower, this pastebin shows it when it works and goes to voicemail: http://www.pastebin.ca/439910 |
| 16:56.37 | etfonhomey | ManxPower, another piece of info that may be useful: The fast busy only happens when someone calls in the Zap/1-1 line. Calls coming into Zap/3-1 always work. I'm wondering if it's not the FXO module. |
| 16:57.20 | *** part/#asterisk deltaray (n=deltaray@static1-66-244-85-183.stfd.smithvilledsl.net) |
| 16:57.31 | ManxPower | etfonhomey: I have no idea how to fix this. Perhaps it is the universe punishing you for using Analog |
| 16:58.02 | etfonhomey | ManxPower, I believe you're right. |
| 16:58.32 | etfonhomey | ManxPower, it's a small Dr.'s office with 2 incoming analog lines, and 3 Polycom phones. |
| 16:58.54 | etfonhomey | ManxPower, Ithey have DSL, I'm not confident enough to run SIP over their DSL connection. |
| 16:59.07 | etfonhomey | They* |
| 17:02.46 | etfonhomey | ManxPower, that's BellSouth DSL... |
| 17:03.05 | ManxPower | etfonhomey: I have no idea how to fix this. |
| 17:03.12 | *** join/#asterisk Fieldy (i=QMnTir44@gentoo/contributor/Fieldy) |
| 17:03.57 | etfonhomey | ManxPower, would you run an office of that size with SIP over a DSL Intenet connection? |
| 17:05.32 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 17:05.46 | toot | we run ours, but we have a dialplan to allow us to swap to pstn when the lads are downloading. um. stuff |
| 17:06.21 | toot | 7 users using VoIP on a 8MB dsl line in crappy ireland |
| 17:06.34 | ManxPower | etfonhomey: I would never run ANY office over an internet connetion |
| 17:06.42 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 17:06.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:07.05 | etfonhomey | ManxPower, you use dedicated connections for VoIP? |
| 17:08.01 | etfonhomey | ManxPower, i.e a partial T1 for VoIP and another T1 for Internet access? |
| 17:08.16 | ManxPower | Um no, I use PRIs |
| 17:08.44 | etfonhomey | So, you never use SIP trunks? |
| 17:09.29 | ManxPower | I have a client with clinics and an administrative offices. I use fractional T-1s between offices with a PRI at the main office, then use QoS on the dedicated inter-office links, the asteirsk server at the main office and all polycom phones. Each office has a dedicated POTS line. |
| 17:10.02 | etfonhomey | Who do you get your PRI from, Bellsouth? |
| 17:10.11 | ManxPower | etfonhomey: The only time I would consider "sip trunks" is in a situation where I have enough PRI channels to handle normal traffic volume and then roll over to a SIP provider in the event of more calls than normal. |
| 17:10.24 | ManxPower | etfonhomey: From XFone a CLEC |
| 17:10.46 | etfonhomey | I wonder if XFone is a CLEC for Louisville. |
| 17:11.22 | ManxPower | I don't like pain and customers with their main connection to the PSTN running over a DSL connection is pretty much the same as saying "Users please beat me up!" |
| 17:14.14 | ManxPower | At me primary client pretty much all the department heads are constantly trying to take over the IT department because the IT department has the largest budget. If there are ANY phone problems they start circling like rabid wolves. The phones MUST work and they MUST work ALL THE TIME. |
| 17:15.04 | etfonhomey | ManxPower, do you use the POTS connection at the each office as a failover and/or 911 access? |
| 17:15.51 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:16.12 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:16.32 | ManxPower | We have a couple of red phones in each office directly connected to the fax line for emergency and 911 |
| 17:16.54 | ManxPower | The T-1s go down like once per year or less so we don't worry about it too much |
| 17:17.00 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 17:17.35 | etfonhomey | ManxPower, is it XFone.com? |
| 17:17.47 | ManxPower | I assume so. |
| 17:18.42 | ManxPower | The sales rep I have is one of the founders of I-55 (a company that was bought by XFone) and so I basically have the cell phone of a tech that is a VP and has access to their switch. |
| 17:18.56 | etfonhomey | ManxPower, I'd love to see your setup some time. |
| 17:19.13 | ManxPower | multiple setups. |
| 17:19.16 | ManxPower | I do consulting |
| 17:20.49 | etfonhomey | ManxPower, I knew that, I meant the setup you were describing. |
| 17:23.18 | dweazle | etfonhomey: thanks for the input |
| 17:23.19 | dweazle | cya |
| 17:23.21 | *** part/#asterisk dweazle (i=dweazle@s5591373c.adsl.wanadoo.nl) |
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| 17:34.41 | mrdigital | whats the best music for on hold |
| 17:34.51 | Strom_M | disco records |
| 17:35.01 | mrdigital | disco records? |
| 17:35.06 | Strom_M | disco records |
| 17:35.36 | mrdigital | ok |
| 17:35.40 | etfonhomey | You know, 70's porn music. |
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| 17:39.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:39.40 | etfonhomey | toot, have you had good luck with a SIP trunk over a DSL connection? |
| 17:40.10 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:41.09 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 17:41.15 | etfonhomey | Any others out there use a SIP trunk over a DSL connection? |
| 17:41.32 | gambolputty | Bananaphone by Raffi is the best music on hold. |
| 17:43.25 | joat | wasn't there a site that offered MP3's of porn theme music? |
| 17:46.56 | ManxPower | joat: just download pretty much any of the crappy electronica 8-) |
| 17:48.40 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 17:51.16 | bulle | gambolputty: i have bananaphone as my ring signal on my cellphone |
| 17:52.06 | ManxPower | joy. storms coming |
| 17:54.53 | gambolputty | I have Bananaphone as a ringtone too for one of my IP phones, sped up 40% of course. |
| 17:55.24 | ManxPower | I don't even know (or really care) what a "Banannaphone" ringtone is. |
| 17:57.00 | bulle | ManxPower: its the ringtone to have, it improves call quality by 25%! |
| 17:57.00 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 17:57.55 | ManxPower | I find that non-standard ringtones increase your chances of being killed by an angry mob by 25% |
| 17:58.53 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 18:09.27 | sooli | Hi |
| 18:09.44 | sooli | I have this problem on a fresh asterisk install : pbx.c:1700 pbx_extension_helper: No application 'Dial' for extension |
| 18:09.47 | sooli | what's wrong ? |
| 18:10.05 | sooli | Backroung, Hangup and Answer are ok |
| 18:12.23 | JunK-Y | sooli: type: load module app_dial.so |
| 18:13.01 | JunK-Y | Backgroung, Hangup and Answer are all core apps. |
| 18:13.27 | sooli | JunK-Y: in my asterisk console ? |
| 18:14.10 | sooli | and what about mp3player ? |
| 18:14.31 | JunK-Y | yes, in ur CLI |
| 18:14.36 | JunK-Y | same thing. |
| 18:14.42 | JunK-Y | type show modules |
| 18:14.51 | sooli | ok thanks |
| 18:14.56 | sooli | Dial work fin now :) |
| 18:17.49 | sooli | ok now my softphone ring ... but when i open conversation, connection is closed :/ |
| 18:22.08 | sooli | here my log : http://pastie.caboo.se/53854 |
| 18:24.49 | *** join/#asterisk fbffff (n=fbffff@cpe-24-175-234-33.stx.res.rr.com) |
| 18:27.02 | *** join/#asterisk dawizard (n=dawizard@mimas.xios.be) |
| 18:27.17 | JunK-Y | take a look at voip-info.org theres so much infos regarding that issue. |
| 18:30.59 | *** join/#asterisk colepc (n=colepc@24-176-126-45.static.jcsn.tn.charter.com) |
| 18:31.22 | fbffff | <PROTECTED> |
| 18:31.23 | fbffff | oops |
| 18:31.41 | colepc | hello. I'm very new to linux and to asterisk and need some advice on a couple of items... |
| 18:32.08 | fbffff | and what would those items be/ |
| 18:32.23 | colepc | I believe I've got Asterisk 1.2.17 installed ok, but don't have a clue how to instantiate it? |
| 18:33.54 | fbffff | you type asterisk with the flags you want most likely -cv or something |
| 18:34.06 | colepc | let me try that... |
| 18:35.09 | fbffff | i saw a wild gecko last night |
| 18:35.50 | colepc | I've got a terminal session on the box, sitting in the ~/asterisk-1.2.17 dir. I typed 'asterisk -cv' and got a result of "bash: asterisk: command not found" |
| 18:36.45 | fbffff | s i keep getting my channels mixed up |
| 18:37.44 | fbffff | then you need to install asterisk some somewhere in your path, or figure out where you installed it and adjust your path |
| 18:37.47 | fbffff | or you didn't install it at all |
| 18:38.21 | colepc | following the very short example at asterisk.org, I installed with the command 'make clean; make install'. |
| 18:38.53 | colepc | it failed several times as certain packages were not installed in the os, but after installing those, it appeared to finally install successfully |
| 18:39.26 | colepc | the dir I'm sitting in has a file labeled as 'asterisk' (no quotes) |
| 18:39.33 | *** join/#asterisk alexjcg (i=alexjcg@201.22.150.201.adsl.gvt.net.br) |
| 18:39.37 | *** part/#asterisk alexjcg (i=alexjcg@201.22.150.201.adsl.gvt.net.br) |
| 18:39.44 | fbffff | you have to type man |
| 18:39.45 | fbffff | make |
| 18:39.48 | fbffff | then make install |
| 18:39.56 | fbffff | or else your not really installing anything |
| 18:40.10 | colepc | let me try that again...the install took about 3 minutes to execute completely the last time... |
| 18:40.16 | *** join/#asterisk tmjb (n=tmjb@212.200.240.90) |
| 18:40.28 | colepc | you're suggesting that I type 'make' <enter> and then 'make install' <enter> ? |
| 18:40.42 | fbffff | yes, you may also want to buy a book or two |
| 18:40.47 | colepc | :) |
| 18:40.52 | fbffff | :) |
| 18:40.52 | tmjb | hello i got problem with misdn this it error any ideas chan_misdn_log: Extension can never match, so disconnecting |
| 18:41.01 | colepc | no doubt... |
| 18:41.17 | fbffff | or find a decent source of ebooks |
| 18:41.28 | fbffff | as most useful books cost way to much money as it is |
| 18:42.13 | ManxPower | fbffff: Did the Gecko say "G'day, Mate!" |
| 18:42.16 | colepc | agreed. I'm trying to wing my way thru this. has been a good teacher in the past. I'm hot on M$, but just learning about things 'UX |
| 18:42.47 | *** join/#asterisk kurtisb1 (n=pa@67.105.142.34.ptr.us.xo.net) |
| 18:42.58 | kurtisb1 | Good morning! |
| 18:43.01 | fbffff | it didn't have time, as soon as i saw it i stopped smoking and ran inside, i was afraid it was going to try and sell me a some sort of insurance policy |
| 18:43.32 | kurtisb1 | Anybody have any luck sending callerID from when intgrating with a legacy PBX? |
| 18:43.32 | colepc | typing 'make' <enter> resulted in this line... "*** [pbx_dundi.o] Error 1" |
| 18:43.47 | fbffff | humm that's not right :) |
| 18:44.06 | kurtisb1 | ..... let me clarify.... sending callerid Name to a legacy PBX (that supports it)? |
| 18:44.14 | colepc | (lots of other stuff scrolled by too!) |
| 18:44.16 | fbffff | that means that the compiler encountered a error when trying to build the module pbx_dundi |
| 18:44.43 | colepc | is pbx_dundi a package (or related to a package) that I may not have installed in the OS yet? |
| 18:45.44 | fbffff | it's a module asterisk loads at runtime |
| 18:45.45 | fbffff | asterisk is comprised of the core asterisk binary, and a verity of modules it loads at runtime |
| 18:46.03 | colepc | in YaST > Software management, I searched for 'pbx' and it produced 2 items unchecked (assmuning, not installed): "asterisk" and "asterisk-debuginfo" |
| 18:46.32 | fbffff | well, what error was make giving you |
| 18:47.50 | colepc | when it started at the 'pbx_dundi.c' lines during make, I'll post a few of them; don't know what is significant... |
| 18:47.58 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 18:48.05 | fbffff | ah post them to pastebin.ca and paste the url here |
| 18:48.25 | fbffff | sadly, im going to go take a shower then i have to go to the gulf coast for something |
| 18:48.31 | colepc | one sake |
| 18:48.34 | fbffff | so i'll be pretty much gone for a few hours :( |
| 18:48.34 | colepc | one sec... |
| 18:50.11 | colepc | http://pastebin.ca/440034 |
| 18:50.27 | colepc | that's cool....didn't know about that site! |
| 18:51.25 | colepc | gulf coast...what state are you in? |
| 18:51.33 | colepc | I've got client in Biloxi |
| 18:52.32 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 18:55.52 | *** part/#asterisk tehmaze (i=wijnand@unaffiliated/tehmaze) |
| 18:56.38 | kurtisb1 | Anybody successfully send CallerID NAME to a Norstar when integrating with asterisk? |
| 18:57.19 | fbffff | right im now im in really southern texas |
| 18:57.56 | fbffff | ah that's a crazy zaptel error |
| 18:58.12 | fbffff | by any chance did you install zaptel? |
| 18:58.20 | colepc | I dont' think so |
| 18:58.23 | fbffff | humm |
| 18:59.06 | fbffff | oh wait |
| 18:59.12 | fbffff | nevermind, you just have to install zlib |
| 18:59.23 | fbffff | it has nothing to do with zaptel at all |
| 18:59.26 | colepc | one sec... |
| 18:59.29 | colepc | installing |
| 18:59.43 | colepc | zaptel is hardware specific, right? |
| 18:59.49 | fbffff | just grab the zlib source then run configure make and make install, or if your useing yast make sure you install the -dev package |
| 18:59.57 | fbffff | nah but ignore the fact i said zaptel at all |
| 19:00.04 | colepc | ok |
| 19:00.09 | fbffff | zlib is what you need |
| 19:00.20 | fbffff | http://www.zlib.net/ |
| 19:00.35 | fbffff | build and install that and you should be one step closer to making phone calls :) |
| 19:00.51 | colepc | thanks...you gone for a while? |
| 19:01.08 | fbffff | well, i'm going to go smoke and get some coffee then i have to get moving |
| 19:01.10 | fbffff | ya prob. |
| 19:01.14 | fbffff | infact brb |
| 19:01.16 | colepc | thanks for your help. |
| 19:05.22 | colepc | ***much better result***... |
| 19:06.18 | colepc | is there a graphical interface to asterisk? |
| 19:06.25 | colepc | (ver 1.2.17) |
| 19:10.09 | colepc | . |
| 19:10.12 | fbffff | nope |
| 19:11.02 | colepc | books, huh? |
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| 19:24.03 | mrdigital | colepc: pm? |
| 19:26.32 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-216-188.socal.res.rr.com) |
| 19:30.26 | colepc | say again mrdigital? |
| 19:31.00 | colepc | stupid question...where's the sip.conf file? |
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| 19:47.49 | jm|laptop | Apr 14 20:45:04 WARNING[12272]: pbx_spool.c:347 scan_service: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting |
| 19:47.50 | jm|laptop | :( |
| 19:47.58 | jm|laptop | it is 0777 asterisk:asterisk ! |
| 19:48.56 | kratzers | anyone know why channel variables aren't showing up when AgentCalled event is generated? |
| 19:49.14 | kratzers | variables set as Set(__varname=value) |
| 19:49.56 | kratzers | in a macro just before calling the Queue app |
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| 19:51.17 | *** join/#asterisk colepc (n=colepc@24-176-126-45.static.jcsn.tn.charter.com) |
| 19:51.39 | Feral_Kid | Is there a way from the command line to send a command to asterisk.... What I would like to do, is write a script that will check my dynamic IP address every 10 minutes, and if my IP address as changes, cause asterisk to do a "sip reload" to update the changes... |
| 19:52.19 | Feral_Kid | I meant to say if my IP address has changed, then send a "sip reload" to asterisk... |
| 19:53.24 | kratzers | http://pastebin.ca/440103 |
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| 19:58.19 | jm|laptop | [TK]D-Fender: wake up :) |
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| 20:11.45 | CessnaFlyer | hello, all |
| 20:12.32 | CessnaFlyer | im having a spot of trouble with asterisk... was hoping someone could help me troubleshoot the demo greeting stuttering very badly |
| 20:13.42 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 20:16.16 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:17.27 | Zefk | does anyone knows a good ip hard phone for use in call center ? |
| 20:18.21 | SwK | zefk: Polycom or Cisco |
| 20:20.42 | Zefk | SwK: I bought 3 ip phones in order to test, but I need also a second opinion. I bought: GS BudgeTone 200, AAstra 9133i and a Polycom 301 |
| 20:20.59 | gambolputty | The Snom 300 is only $100. |
| 20:24.01 | Zefk | After a smoke test with BudgeTone 200 I realize that this phone is only for doorman. |
| 20:25.38 | JT | s/doorman/doorstop/ |
| 20:26.22 | SwK | i dont care for the snom's some people swear by them |
| 20:26.28 | SwK | personally I swear by the polycoms |
| 20:26.47 | *** join/#asterisk ZefK (n=Zefk@81.181.249.106) |
| 20:27.18 | JT | s/doorman/doorstop/ |
| 20:27.29 | JT | yeah i don't see what's so good about snoms |
| 20:27.35 | ZefK | Aastra and Polycom are good quality just the price is a litle high |
| 20:27.41 | JT | they definitely don't look very good :) |
| 20:27.56 | gambolputty | The Snom features are great. |
| 20:28.23 | ZefK | gambolputty: is it good the audio quality ? |
| 20:29.50 | gambolputty | I am thinking of its SIP features |
| 20:29.59 | gambolputty | its audio is good too |
| 20:30.20 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 20:30.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:30.34 | ZefK | I intend to buy another 2-3 phones to have more testing. I am thinking to buy a GXP 2000, maybe a snom and a linksys |
| 20:30.40 | *** join/#asterisk Mavvie (n=edwin@ppp24-120.lns1.syd6.internode.on.net) |
| 20:30.53 | JT | ~gs |
| 20:31.00 | jbot | gs is, like, South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 20:32.10 | JT | don't waste money on more grandstream phones |
| 20:32.16 | JT | ZefK: what country are you in? |
| 20:32.19 | ZefK | jbot: I live in Europe, I don;t know how is in South Georgia but I get your point :) |
| 20:32.21 | jbot | You live in Europe, I don;t know how is in South Georgia but I get your point :)? |
| 20:32.49 | JT | ZefK: the south georgia bit is a totally different factoid to grandstream :) |
| 20:33.00 | ZefK | I'm living in Romania but I buy from Italy |
| 20:33.45 | JT | hrm there's a few romanian foss telephony users :) |
| 20:34.15 | mrdigital | jt: is there a decent phone priced the same as a Grandstream BT |
| 20:34.50 | *** join/#asterisk kurtisb1 (n=pa@67.105.142.34.ptr.us.xo.net) |
| 20:35.12 | kurtisb1 | Anybody know if Digium cards support sending q.932 ? |
| 20:35.15 | JT | mrdigital: no |
| 20:35.23 | JT | decent phones aren't that cheap |
| 20:36.15 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:36.28 | ZefK | We have in our call center a solution with Avaya S8700, perfect quality and features but we are payng a lot for all these. I'm trying to implement a resonable solution with asterisk and I don't want to buy phones with more than 120EUR |
| 20:37.18 | hal2k | ZefK: what about using a softphone? |
| 20:38.01 | ZefK | I tested 5-6 softphones, but the quality is not good at all |
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| 20:40.18 | ZefK | Even with gs phones the quality is better than with a softphone |
| 20:40.25 | SwK | zefk I wasnt impressed with the NAT handlng on the asstra never could get them to work when the phone was behind nat and asterisk was on a public IP.... have used polycoms in carriers where they do things like IP-PBXs and they hold up good |
| 20:41.47 | [TK]D-Fender | Polycom > All |
| 20:42.05 | ZefK | I did not tested yet my Polycom 301 because I received the phone with US power source. Monday I will buy an adapter |
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| 20:44.35 | ZefK | Anyway, my problem is if I should buy phone sin the range of 110-120 EUR or shoud I move to phones in the range of 160-170EUR ..!? |
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| 20:47.47 | ZefK | polycom and Aastra belong to second price interval |
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| 20:52.26 | Star568 | manager API originat calls, how can i get music on hold on local leg while waitting for outside leg connecting? |
| 20:52.34 | [TK]D-Fender | ZefK, What are your expectations, needs and description of call volume/usage? |
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| 20:56.19 | ZefK | I intend to move a group of 50-70 operators from Avaya to Asterisk. This project does not requires Avaya features so the movement should be easy. It is just an outbound project. It is very important for me to integrate in our web application a dialer in order to have the operator dial from a web page. |
| 20:57.41 | ZefK | Another thing is I dont want to have complanins about voice quality. |
| 20:57.59 | gambolputty | ZefK: see my private message |
| 20:58.01 | JT | then it'd be a bad idea to go backwards :) |
| 21:01.43 | ZefK | I'm thinking also to use manager interface to place calls in automatic but if I found a phone with some "click and dial" facilities will be great |
| 21:03.31 | ZefK | Anyway, the manager interface provides a general solution that is "phone independent" |
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| 21:07.51 | ZefK | Another concern of mine is if asterisk is ready for professional solutions in call centers? I am not an asterisk expert, I just realized some interconnections world wide via IAX, and a few scripts |
| 21:08.21 | JT | it can be if you use decent phones |
| 21:08.32 | JT | but i guess you need to test if your setup is stable |
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| 21:10.38 | ZefK | On the other hand I heard about some call centers that runs only on asterisk ... like Acer in Germany. |
| 21:13.07 | Vec | What processes in asterisk require a timer like ztdummy or a digium card, i.e. when would I encounter problems without one ? |
| 21:13.28 | JT | zap hardware, iax trunking, meetme conferences |
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| 21:17.58 | Vec | JT : how good is ztdummy, because I am going to have to do some IAX trunking ? |
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| 21:19.14 | JT | results can be pretty haphazard |
| 21:19.27 | JT | do you really need to do iax trunking? |
| 21:19.41 | Vec | JT : well I could do SIP style trunking |
| 21:20.10 | JT | sip doesn't do trunking |
| 21:20.25 | JT | trunking is optional in iax2 |
| 21:20.30 | Vec | JT: well just route the calls individually through using SIP |
| 21:20.43 | JT | eh? |
| 21:20.51 | Vec | just with trunking I get the performance benefit of no protocol overhead |
| 21:21.07 | JT | there is overhead |
| 21:21.13 | Vec | I am going to link to a SIP gateway |
| 21:21.21 | Vec | or IAX gateway |
| 21:21.37 | Vec | yeh but the overhead decreases with the increase in simultanious calls with IAX trunking |
| 21:22.08 | JT | do you think it really makes that much difference? |
| 21:23.50 | Vec | JT : not sure, have not played around with it that much, just checked out http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 |
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| 21:24.22 | JT | there's a lot of hype around iax2 trunking, i just dunno if it's really that great |
| 21:24.39 | JT | also it chokes if you have lots of simultaneous calls on the one trunk |
| 21:24.52 | Vec | JT : well thats sh1t |
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| 21:36.37 | MaartenB | I am having problems with incomming calls, I have no idea why, but they are handled by a timeout |
| 21:36.47 | MaartenB | it says "Timeout on IAX2/speakup01-1" where speakup01 is my voip provider |
| 21:37.04 | MaartenB | and I think it is giving this timeout on "Set("IAX2/speakup01-1", "CALLERID(name)=Private") in new stack" |
| 21:37.06 | MaartenB | which is akward |
| 21:48.37 | drfreeze | MaartenB: I thought the Set syntax was Set(name1=value1|name2=value2|...[options]) |
| 21:49.54 | MaartenB | drfreeze, you are right, it is set like that in my extensions.conf, the text above is what is displayed in the log |
| 21:49.59 | drfreeze | JT: I have heard a lot of problem reports from people using Iax2. Never used it myself. |
| 21:50.12 | drfreeze | JT: SIP seems to be the way to go |
| 21:50.27 | drfreeze | MaartenB: oh |
| 21:51.45 | drfreeze | MaartenB: has this ever worked? |
| 21:52.50 | MaartenB | yes, actually it still works |
| 21:53.03 | MaartenB | but it fails in 1 out of 5 cases |
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| 22:34.08 | yenno | hi.. where does that "you have successfully installed asterisk.." message coming from and how do i get rid of it? |
| 22:36.01 | blitzrage | yenno: from the dialplan (extensions.conf) in the [demo] context |
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| 22:36.22 | blitzrage | if you're asking this -- you have a bunch of reading to do |
| 22:36.23 | blitzrage | ~docs |
| 22:36.33 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
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| 22:39.31 | blitzrage | JunK-Y: nice commit :) |
| 22:41.52 | JunK-Y | blitzrage: wait my next one, ya will like it. |
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| 22:46.12 | yenno | thanks blitzrage, well normally i'm only using cisco callmanager, but my sip implementation needs to work with asterisk, too :-/ |
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| 22:53.53 | colepc | can someone advise me on a VOIP service provider I can test my shiny new trixbox with? |
| 22:59.53 | etfonhomey | FWD |
| 23:09.25 | colepc | thanks! |
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