| 00:05.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 00:08.50 | drfreeze | Ok, in sip.conf I added "call-limit=1" to the extension and that fixes the audio problem |
| 00:09.14 | drfreeze | But causes another problem in that the user can't see new calls coming in and field them |
| 00:09.23 | drfreeze | Kind of defeats the purpose of a switch board |
| 00:10.02 | WIMPy | sip.conf is not on your phone. |
| 00:10.58 | drfreeze | I would like to understand this problem |
| 00:11.09 | drfreeze | Not sure why it has cropped up. |
| 00:11.40 | WIMPy | Search at the phone, not at *. |
| 00:12.38 | WIMPy | For * it won't make a difference if a new call goes to a phone in use or to another. |
| 00:12.48 | WIMPy | Except fot that call-linit, off course. |
| 00:19.49 | root52 | Good Day All. Is there a good primer on NAT and Asterisk I could look at? I am having a problem where I am placing a call between two endpoints using sip. Each endpoint is registered to a seperate PBX. There is no Audio when the channels are bridged togeather. Both the ASterisk boxes are behind NAT devices with ports fowarded and SIP.conf has the network infromation for extern and localnet. |
| 00:20.26 | *** join/#asterisk blaines (~blaines@wsip-70-184-85-19.ph.ph.cox.net) |
| 00:21.55 | WIMPy | ~sipnat |
| 00:21.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:25.24 | root52 | Thank You infobot. Robots are usualy more helpful than humans ;-) |
| 00:26.55 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:27.48 | *** join/#asterisk mathslinux (~mathslinu@120.42.46.126) |
| 00:33.26 | drfreeze | WIMPy: ok, changing chord to silence for CALLWAITING AND CALLWAITINGLONG on the phone config fixed it |
| 00:33.36 | drfreeze | but, now there is no audible idicator |
| 00:33.43 | drfreeze | sighs |
| 00:43.44 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 00:44.28 | *** join/#asterisk VITAS (~VITAS@port-2739.pppoe.wtnet.de) |
| 00:45.04 | VITAS | hi anyone here can help me with an asterisk -> exchange 2010 UM Problem? |
| 00:46.36 | chuckf | can anyone here tell what VITAS' problem is without VITAS telling us what it is? |
| 00:47.10 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 00:48.37 | VITAS | lol |
| 00:49.13 | VITAS | i could make a /nick and explain it |
| 00:49.18 | VITAS | ;) |
| 00:49.56 | *** join/#asterisk mathslinux (~mathslinu@74.86.0.138) |
| 00:51.49 | VITAS | chuckf i want to access exchange Unified Messaging from an asterisk 1.6something |
| 00:52.26 | VITAS | ive got the peering up (sip.conf entry for the exchange) and added some rules into extensions.conf |
| 00:54.11 | VITAS | sip show peers says that the connection to exchange is working but if i try to call the mailbox on the excange i get a "forbidden" in asterisk and windows tells me that asterisk has stopped the handshake at the codec/media part |
| 00:54.51 | VITAS | i tried to change the settings in exchange to gsm,g711 without change |
| 00:55.33 | VITAS | any advice? |
| 00:55.51 | *** join/#asterisk Kyosh (whoa@pool-74-108-16-250.nycmny.fios.verizon.net) |
| 00:58.38 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 01:02.27 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 01:02.37 | pabelanger-lap | VITAS: enable SIP debugs |
| 01:13.00 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 01:13.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:14.52 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 01:19.11 | *** join/#asterisk arielb27 (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 01:19.22 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 01:24.27 | VITAS | did that allready |
| 01:24.37 | VITAS | chan_sip.c:17865 handle_response_invite: Received response: "Forbidden" from '"Exchange User" <sip:11001@10.254.0.16>;tag=as33692990' |
| 01:24.46 | VITAS | he hangs up |
| 01:27.15 | drfreeze | Where do I get info on the sip.ld files that come from polycom spip zip files? |
| 01:27.29 | drfreeze | It looks like version of sip.ld exist: 3111-40000-001.sip.ld |
| 01:27.57 | drfreeze | but I can't find info on the versions |
| 01:28.22 | WIMPy | strings? |
| 01:30.34 | pabelanger-lap | VITAS: pb your sip.conf |
| 01:30.48 | pabelanger-lap | VITAS: and debug log of call |
| 01:32.13 | drfreeze | WIMPy: seems to be no difference (via strings) between 3111-15600-001.sip.ld 3111-40000-001.sip.ld |
| 01:33.06 | WIMPy | Ohm I thought you were trying to find the version. |
| 01:36.03 | *** part/#asterisk encinoman (~matthew@pool-173-63-34-15.nwrknj.fios.verizon.net) |
| 01:36.33 | *** join/#asterisk hyperactivecrond (~chris@unaffiliated/hyperactivecrond) |
| 01:37.15 | hyperactivecrond | I know that it's very doable to take an incoming PRI line and use it for outgoing calls |
| 01:37.26 | hyperactivecrond | however, is it possible to do the reverse with Asterisk? |
| 01:37.56 | hyperactivecrond | i.e. take a SIP connection with X channels and offer it as a T-1 line to legacy hardware |
| 01:37.57 | WIMPy | ??? |
| 01:38.20 | WIMPy | Ah, Sure. |
| 01:38.23 | drfreeze | hrm, looks like a bug to me |
| 01:38.40 | drfreeze | changing the sip.ld file doesn't seem to fix it either |
| 01:39.08 | hyperactivecrond | actually, could it be ISDN BRI/ |
| 01:39.38 | *** join/#asterisk Jamuel (~jamuel@c-24-7-72-218.hsd1.ca.comcast.net) |
| 01:40.09 | VITAS | http://pastebin.com/znfthPnw <- sip.conf & debug |
| 01:41.53 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 01:42.23 | *** join/#asterisk pwell (~pwell@ool-435255fc.dyn.optonline.net) |
| 01:42.27 | jstapleton | what might cause - " == Spawn extension (devices, o, 0) exited non-zero on 'SIP/aethra-0000005a' |
| 01:42.55 | jstapleton | I am trying to setup a zero destination from voicemail. pretty sure that it was working in 1.4 but not in 1.6. |
| 01:48.33 | Jamuel | Hi--can anyone talk about DTMF on SIP channels? We have our ITSP peer configured for RFC2833 which works fine most of the time, occasionally the peer sends an INVITE with only PCMU/8000 offered (no telephone-event/8000). When this happens we end up with DMTF issues on our IVRs ostensbily because * is expecting DTMF via RFC2833 and not inband. |
| 01:49.10 | *** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-177-61.mia.bellsouth.net) |
| 01:49.27 | Jamuel | is there anything I can do to configure inband decoding of DTMF as a fallback? |
| 01:49.46 | Jamuel | I'm running * v1.6.2.9 |
| 01:51.07 | Jamuel | so would dtmfmode=auto solve that problem? |
| 01:55.19 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 01:59.53 | *** join/#asterisk gadams999 (gadams999@173-165-184-25-atlanta.hfc.comcastbusiness.net) |
| 02:00.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:00.39 | pabelanger-lap | VITAS: transport = udp is your problem |
| 02:01.15 | *** join/#asterisk mathslinux (~mathslinu@120.42.46.126) |
| 02:01.32 | pabelanger-lap | Jamuel: no |
| 02:01.34 | pabelanger-lap | Jamuel: https://issues.asterisk.org/view.php?id=17571 |
| 02:02.23 | WIMPy | Just out of interest: So it does speak udp, but won't accept calls unless you use tcp? |
| 02:06.19 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 02:08.21 | *** part/#asterisk root52 (~root52@ip24-252-251-246.cl.ri.cox.net) |
| 02:16.55 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 02:17.36 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:18.20 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 02:19.58 | *** join/#asterisk Junior (~Juni@unaffiliated/junior) |
| 02:28.42 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 02:28.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:34.32 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 02:35.02 | *** join/#asterisk sarah_p (~sarah_p@pool-71-117-252-127.ptldor.fios.verizon.net) |
| 02:45.12 | *** join/#asterisk [Outcast]_ (~anonymous@24-151-143-187.dhcp.oxfr.ma.charter.com) |
| 02:54.15 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 02:54.31 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 03:08.30 | *** join/#asterisk coppice (~chatzilla@202.64.176.19) |
| 03:22.12 | *** join/#asterisk itbuddha (~itbuddha@c-69-254-4-174.hsd1.ga.comcast.net) |
| 03:24.36 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:35.06 | *** join/#asterisk sourcode (~code@ppp-58-8-239-141.revip2.asianet.co.th) |
| 03:35.09 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 03:57.36 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:57.58 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 04:06.42 | *** join/#asterisk incognito53 (~Guest627@ip70-178-69-17.ks.ks.cox.net) |
| 04:07.13 | *** part/#asterisk itbuddha (~itbuddha@c-69-254-4-174.hsd1.ga.comcast.net) |
| 04:10.44 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.194) |
| 04:13.49 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 04:20.07 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 04:22.04 | boodu | ciao |
| 04:38.00 | *** join/#asterisk aidinb (~Aidin@166.134.78.128) |
| 04:43.32 | *** part/#asterisk Tobarja (~chatzilla@24.136.151.107) |
| 04:46.51 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ppdytfhclbxxvgic) |
| 04:52.49 | *** join/#asterisk eye-scuzzy (~light@sun28.ipfw.su) |
| 05:01.05 | ChannelZ | moof |
| 05:08.30 | *** join/#asterisk itbuddha (~itbuddha@c-69-254-4-174.hsd1.ga.comcast.net) |
| 05:21.39 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-66-25.home.otenet.gr) |
| 05:22.46 | *** join/#asterisk mathslinux (~mathslinu@120.42.46.126) |
| 05:26.20 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:38.34 | *** join/#asterisk BarthezZ (~bart@ipd50a21c9.speed.planet.nl) |
| 05:50.52 | *** join/#asterisk eye-scuzzy (~light@sun28.ipfw.su) |
| 05:54.01 | *** join/#asterisk [T]ank (~T]ank@c-71-195-199-101.hsd1.ut.comcast.net) |
| 05:55.40 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 05:56.07 | [T]ank | I am trying to install h323 on a asterisk server that is already been running for a while. Its version 1.4 |
| 05:56.07 | [T]ank | I have installed the opal packages from yum. This is a cent os 5 server. When i do a make distclean ./configure and make menuselect It still shows that dependancies have not been met for h323. says I still need openh323. But everywhere I see says that opal is the correct package. can anyone assist? |
| 05:59.06 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qipzbjjlyzremanq) |
| 05:59.09 | [T]ank | seems pretty quiet... everyone sleeping? |
| 05:59.16 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 05:59.39 | drmessano | No, you just said h323 |
| 05:59.49 | drmessano | The other guy using it died back in 2002 |
| 06:00.07 | drmessano | RIP Shleppy |
| 06:00.23 | [T]ank | hahaha.... connecting to an Avaya PBX, so its a neccessary evil |
| 06:00.56 | drmessano | Ok, so let me take a stab |
| 06:01.05 | drmessano | which packages did you install? |
| 06:01.09 | drmessano | specifically |
| 06:01.21 | [T]ank | k, let me get this exact... sec |
| 06:01.43 | carrar | yum install all |
| 06:01.47 | drmessano | LOL |
| 06:02.13 | drmessano | I see opal and opal-devel when I do a yum list |
| 06:02.16 | [T]ank | I did a yum install opal opal-devel |
| 06:02.21 | drmessano | ok |
| 06:03.03 | [T]ank | then i did a make distclean then ./configure then make menuselect in the asterisk install directory |
| 06:04.01 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 06:05.40 | [T]ank | but when i do the make menuselect the h323 option is still XXX'ed out. |
| 06:06.14 | drmessano | I see a patch for using opal that pertains to 1.6.2.x |
| 06:06.44 | drmessano | As in, using opal instead of... |
| 06:06.46 | drmessano | https://issues.asterisk.org/view.php?id=17226 |
| 06:07.07 | [T]ank | ...reading |
| 06:07.53 | [T]ank | hmmmm... ok... so I guess the other read I found was wrong.... I thought everything on the internet was accurate and true. |
| 06:08.00 | drmessano | haha |
| 06:08.25 | [T]ank | k, back to google. thanks |
| 06:11.45 | carrar | everything on the internet IS accurate and true, you just have the wrong perspective |
| 06:13.38 | drmessano | You went to the wrong internet |
| 06:13.42 | drmessano | Try Wikipedia |
| 06:13.45 | drmessano | IT is accurate |
| 06:13.58 | Pimmetje | drmessano: Depends |
| 06:14.06 | Pimmetje | I SHOULD BE |
| 06:14.26 | Pimmetje | but everybody can write EVERYthing |
| 06:14.46 | drmessano | In theory Wikipedia can never be accurate |
| 06:15.23 | drmessano | Accuracy requests a constant.. and there are no constants, other than that Windows sucks |
| 06:15.38 | drmessano | and my distro is better than yours, even if we are using the same distro |
| 06:15.43 | drmessano | MY ISO IS BETTER |
| 06:16.59 | coppice | pi is a constant. apple pi less so. |
| 06:18.36 | carrar | You need to get on the IPv6 network, eveything there is 100% SAFE |
| 06:18.57 | [TK]D-Fender | coppice: Yes, half-baked, and likely poisoned :0 |
| 06:20.29 | drmessano | Walmart so screwed up |
| 06:21.45 | drmessano | Walmart's flameless candles have these "try me" pushbuttons that you can access from the front of the box. They're wired to a CR2032 battery in a little plastic case that attaches to a little 2 pin connector on the bottom of the candle |
| 06:22.23 | drmessano | The primary power is 3 AAA batteries.. But I now have a pigtail to run the candle off a power supply lol |
| 06:23.24 | drmessano | and more importantly.. to switch on/off with an X10 universal module.. which I plan to control with the BIG A |
| 06:23.39 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 06:23.49 | drmessano | Asterisk Flameless Candle project FTW |
| 06:28.23 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
| 06:28.24 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-asiwbsjuvcgyhaon) |
| 06:41.53 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 06:43.04 | [T]ank | drmessano: so I have been trying to install openh323 manually.... when running the make opt command, I am getting the following error: In file included from ixjunix.cxx:491: |
| 06:43.04 | [T]ank | /usr/src/openh323_v1_18_0/include/ixjlid.h:277:28: error: linux/compiler.h: No such file or directory |
| 06:43.24 | [T]ank | Im not finding anything when searching for that error that leads me to understand what i may be doing wrong. |
| 06:45.13 | SiNGLer | maybe you need kernel headers? |
| 06:46.12 | [T]ank | shows they are installed. |
| 06:46.18 | [T]ank | this is a currently running asterisk installation.... |
| 06:46.24 | [T]ank | just adding h323 after the fact |
| 06:46.46 | [T]ank | the file that the make is looking for does not exist in the h323 install directory. |
| 06:51.05 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 06:56.06 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 07:02.57 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 07:04.42 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
| 07:04.42 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:04.42 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 07:04.42 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 07:04.42 | *** join/#asterisk Scorcerer (root@czlug.icis.pcz.pl) |
| 07:04.42 | *** join/#asterisk DND (~arabia@94.200.7.26) |
| 07:04.42 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 07:04.42 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 07:04.42 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 07:04.42 | *** join/#asterisk fenrus (fenrus@oklart.com) |
| 07:04.43 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 07:06.28 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 07:06.42 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 07:10.31 | *** join/#asterisk ming_zym (~ming_zym@121.0.29.237) |
| 07:19.29 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:31.32 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:38.49 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:39.13 | *** join/#asterisk cdahmedeh (~cdahmedeh@62.68.65.245) |
| 07:40.15 | cdahmedeh | hello, i have setup a sip server in some other country.. and in the current country.. i am able to register, and make the call.. but i get no audio... all the ports are correctly forwarded.. and the client is behind a dmz |
| 07:40.48 | cdahmedeh | though the dmz is only from the router... |
| 07:41.02 | cdahmedeh | isp here seems to be blocking alot of ports |
| 07:49.39 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 07:49.45 | ChannelZ | I love it when they answer their own questions |
| 07:57.57 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 07:58.11 | *** join/#asterisk lhz (~shrekz@c-dba672d5.021-158-73746f34.cust.bredbandsbolaget.se) |
| 07:58.31 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 07:59.07 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 08:07.11 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 08:15.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:19.56 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-tkejibnxtchiymhn) |
| 08:31.08 | *** join/#asterisk [T]ank (~T]ank@c-71-195-199-101.hsd1.ut.comcast.net) |
| 08:38.02 | *** join/#asterisk centoslinux (~centoslin@1x-193-157-204-216.uio.no) |
| 08:38.05 | *** join/#asterisk matagou_ (5ef36b93@gateway/web/freenode/ip.94.243.107.147) |
| 08:38.21 | matagou_ | hello, asterisk |
| 08:39.55 | matagou_ | please suggest how can i get the latest svn of asterisk-addons for asterisk 1.6 |
| 08:41.00 | puzzled | matagou_: http://www.asterisk.org/developer/resources/svn |
| 08:51.44 | *** join/#asterisk ruied (~ruied@89.180.240.144) |
| 08:53.08 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 09:10.03 | *** join/#asterisk ruied (~ruied@89.180.240.144) |
| 09:11.29 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:13.01 | *** join/#asterisk pejotr (~pejotr@abab21.neoplus.adsl.tpnet.pl) |
| 09:28.02 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 09:33.04 | *** join/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 09:39.02 | *** join/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 09:42.20 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 09:48.13 | *** join/#asterisk kinsey (~chatzilla@58.85-200-224.bkkb.no) |
| 09:49.30 | kinsey | I just ran yum update on a my AsteriskNow box, and all of a sudden asterisk refuses registrations from my IAX2 devices - any clues about how to find out why? |
| 09:51.48 | SiNGLer | does it produce any errors? |
| 09:51.56 | Tim_Toady | kinsey, udated from 1.4 to 1.6? updated to a newer version of 1.4 that requires call tokens? in any case read doc/IAX2-security.txt from asterisk source tree |
| 09:54.49 | kinsey | Tim: No, it was at 1.6 (or 1.7) when I started |
| 09:55.02 | kinsey | Call tokens was fixed on the last update :) |
| 09:55.31 | kinsey | but what is the best way to debug? Which logs to enable/read etc? |
| 09:55.55 | Tim_Toady | /var/log/asterisk/full |
| 09:56.05 | *** part/#asterisk Han (~han@unaffiliated/han) |
| 09:56.27 | Tim_Toady | it is enabled by deaful i think in asterisk now, check logger.conf |
| 09:57.52 | kinsey | ah, fixed |
| 09:58.16 | kinsey | for some reason I now needed to set the requirecalltoken to 'no' |
| 10:01.01 | *** join/#asterisk kinsey (~chatzilla@58.85-200-224.bkkb.no) |
| 10:15.13 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 10:20.16 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 10:23.03 | xheliox | yeah, it's a mystery.. if only that was documented in some sort of changelog or readme. :P |
| 10:23.07 | xheliox | oh.. wait. |
| 10:26.16 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:46.20 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:52.51 | pejotr | hi, i have a problem with SayUnixTime, when i set SayUnixTime(kMS) it says "twelve thirty three AND two SECONDS" why ? |
| 10:58.33 | SiNGLer | 12 hour 33 minutes and 2 seconds, what is the problem? |
| 10:59.04 | pejotr | why it says word 'seconds' and 'and' ? |
| 11:00.30 | SiNGLer | I assume because it is natural to say so :) |
| 11:05.22 | pejotr | i expected "twelve thirty three two". i'd like it to say "twelve thirty three two seconds" and i assume it is not possible ? |
| 11:05.54 | Gugge | it is possible, just not with sayunixtime :) |
| 11:06.53 | *** join/#asterisk sourcode (~code@ppp-61-90-12-241.revip.asianet.co.th) |
| 11:08.31 | pejotr | and that's what i want to hear, thanks |
| 11:09.41 | *** join/#asterisk gadams999 (gadams999@173-165-184-25-atlanta.hfc.comcastbusiness.net) |
| 11:10.59 | *** join/#asterisk daltio (~daltio@nccpl.pk) |
| 11:11.21 | daltio | guys i want to integrate asterisk with seimens highpath 3000 any idea |
| 11:11.44 | daltio | high path support sip |
| 11:15.41 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 11:28.37 | leifmadsen | pejotr: in American english, it is normal to say "and 2 seconds" |
| 11:29.12 | leifmadsen | if you don't want that, then feel free to modify app_sayunixtime.c (or whatever C file has that application) |
| 11:29.37 | leifmadsen | xheliox++ |
| 11:33.16 | pejotr | i just look into the source and found that language i use has hardcoded to say 'and' and 'second' :) thank for your help |
| 11:51.59 | *** join/#asterisk Belgarath (~Belgarath@banda.pl) |
| 12:01.34 | *** join/#asterisk sourcode (~code@ppp-61-90-41-181.revip.asianet.co.th) |
| 12:04.40 | *** join/#asterisk zorp75ck (~zorp75ck@141.158.199.197) |
| 12:05.39 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 12:07.29 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 12:15.17 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 12:17.20 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 12:17.32 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:27.30 | *** join/#asterisk traxx (~traxx@91-64-218-219-dynip.superkabel.de) |
| 12:31.56 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 12:38.37 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 12:42.28 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:51.19 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 12:52.33 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:55.55 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
| 12:56.02 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:05.08 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:06.03 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:10.23 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
| 13:10.39 | *** part/#asterisk TheDavidFactor (~chatzilla@nc-71-52-237-137.dhcp.embarqhsd.net) |
| 13:10.54 | *** join/#asterisk Naikrovek (~Steve@unaffiliated/naikrovek) |
| 13:13.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 13:17.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:21.08 | *** part/#asterisk [T]ank (~T]ank@c-71-195-199-101.hsd1.ut.comcast.net) |
| 13:22.56 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 13:24.42 | *** part/#asterisk CaT[t3] (~cat@nessie.weebeastie.net) |
| 13:26.46 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:26.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:34.53 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 13:37.10 | *** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com) |
| 13:38.36 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 13:45.12 | *** join/#asterisk zamba (marius@flage.org) |
| 13:45.21 | evilbit | hi, there are two subroutines(?) in the example extensions.conf called stdexten and stdPrivacyexten. Is it possible to specify options to the device dialed, like the ability for the device to transfer a call? |
| 13:49.17 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:53.08 | leifmadsen | evilbit: what kind of device? |
| 13:53.19 | evilbit | sip device |
| 13:53.35 | leifmadsen | evilbit: then the sip device is responsible for requesting the transfers -- you don't have to do anything with Asterisk to enable that |
| 13:53.39 | evilbit | so, the normal entry would be Dial(SIP/evilbit) |
| 13:53.51 | leifmadsen | just use the transfer buttons on the device |
| 13:53.52 | evilbit | er, Dial(SIP/hhoffman,,tT) |
| 13:54.01 | leifmadsen | that is not necessary unless you're doing DTMF based transfers |
| 13:54.07 | evilbit | ah, I didn't realize that... I thought it was necessary |
| 13:54.09 | evilbit | thanks :-) |
| 13:54.09 | leifmadsen | which is usually only necessary for things like analog devices |
| 13:57.39 | [TK]D-Fender | evilbit: those options tel * to listen for DTMF commands for things other phones get to take for granted. For crap phones where the maker didn't add the normal functionality in (Limited like X-Lite), analog phones on dumb ATA's, etc. |
| 13:58.04 | evilbit | gotcha, thanks guys :-) |
| 13:58.37 | *** join/#asterisk rare1980_ (~as@109.169.28.42) |
| 13:59.29 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:59.37 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:01.21 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 14:01.30 | rare1980_ | hi all when i try to do outbound calls i get this error msg on asterisk cli---- Using SIP VRTP TOS bit 136 and Using SIP RTP TOS bits 184.. please any help on this.. |
| 14:01.50 | rare1980_ | why this msg is coming up.. 1 day back it was working fine |
| 14:03.08 | Tim_Toady | check sip.conf for tos_sip tos_audio and tos_video |
| 14:04.04 | rare1980_ | tos_sip tos_audio....... wht info do they contain? |
| 14:04.53 | rare1980_ | i don't have tos_sip and tos_audio under asterisk folder |
| 14:04.59 | rare1980_ | where these files are located? |
| 14:05.07 | Tim_Toady | the value for the TOS bit at IP header |
| 14:05.30 | rare1980_ | where i can find these files? |
| 14:05.38 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:05.58 | Tim_Toady | the values are set in sip.conf |
| 14:06.04 | rare1980_ | humm |
| 14:07.02 | rare1980_ | let me c |
| 14:08.32 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:14.30 | *** join/#asterisk Martinblr (~Miranda@61.12.17.170) |
| 14:15.04 | Martinblr | by remote I am not able to register a softphone with password, without giving password I am able to register |
| 14:15.29 | pabelanger-lap | rare1980_: SIP debugs? |
| 14:20.15 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:20.20 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 14:20.37 | *** join/#asterisk p0seidon (~nnscript@89.181.51.12) |
| 14:24.02 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 14:25.10 | evilbit | should the natted-phone template in sip.conf include qualify=yes and canreinvite=no by default? |
| 14:27.48 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 14:27.55 | [TK]D-Fender | ~sipnat |
| 14:27.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:27.58 | [TK]D-Fender | ^^^^^^ |
| 14:28.11 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:30.04 | *** join/#asterisk BezNalogov (~BezNalogo@62.88.17.85) |
| 14:30.18 | BezNalogov | Hi people. I'm trying to put an iax account into my asterisk server. This account is meant for incoming calls. I've entered a register => line into iax.conf. When I check with iax2 show registry I can see the account is online. Also I made in iax2.conf an item for this account, where I have also set a context, then I've added into that context an extension S with just a dial command. Now when I dial in from my landline to the DID number, the server w |
| 14:30.18 | BezNalogov | ill say |
| 14:30.31 | BezNalogov | that it rejected the connection attempt for who was trying to reach 's@' |
| 14:30.36 | BezNalogov | What am I missing here? |
| 14:38.09 | *** join/#asterisk dailylinux (~fedora@s0021-0018.dsl.start.no) |
| 14:38.55 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:39.57 | *** join/#asterisk DiegoIT86 (~zdzxzx@host74-151-static.93-94-b.business.telecomitalia.it) |
| 14:40.32 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:40.35 | *** part/#asterisk DiegoIT86 (~zdzxzx@host74-151-static.93-94-b.business.telecomitalia.it) |
| 14:44.45 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 14:47.31 | *** join/#asterisk brycebaril (~bbaril@sea02-v612-nat.marchex.com) |
| 14:48.48 | *** join/#asterisk ming_zym (~ming_zym@124.160.123.83) |
| 14:52.31 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-79-112.home.otenet.gr) |
| 14:54.01 | *** join/#asterisk vrban (~chatzilla@p50997436.dip0.t-ipconnect.de) |
| 15:00.02 | *** join/#asterisk chuckp (~chuck.pea@c-76-106-198-76.hsd1.fl.comcast.net) |
| 15:01.45 | chuckp | Anyone using SendFAX from Fax For Asterisk? ReceieveFAX is working fine, however the SendFAX connects then drops with Auto fallthrough, channel 'SIP/*' status is 'UNKNOWN' marking the call as completed. t.38 negotiates just fine. Then the fax session ends with: FAX session '3' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
| 15:08.22 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 15:09.45 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:31.58 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:33.48 | *** join/#asterisk Ambiguity (~Ambiguity@adsl-179-117-51.gnv.bellsouth.net) |
| 15:39.26 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 15:39.28 | wcselby | o/ |
| 15:40.47 | *** join/#asterisk clintc (~clintc@n128-227-105-14.xlate.ufl.edu) |
| 15:45.26 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:47.06 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 15:49.35 | Naikrovek | wcselby: i had a blinking clock problem again |
| 15:49.38 | Naikrovek | guess what it was |
| 15:49.39 | Naikrovek | lol |
| 15:49.51 | wcselby | what? |
| 15:50.32 | chuckp | time server went down? |
| 15:51.14 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:55.13 | raden_work | I need a billing solution for asterisk anyone have any recommendations |
| 15:55.32 | wcselby | roll your own? |
| 15:55.35 | wcselby | a2billing? |
| 15:55.57 | wcselby | http://lmgtfy.com/?q=asterisk+billing |
| 15:56.00 | raden_work | rolling my own feels like a lot of work at the moment :( |
| 15:56.15 | wcselby | :) |
| 15:58.46 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 15:59.51 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 16:00.12 | Naikrovek | wcselby: ftp server didn't start on reboot |
| 16:00.17 | Naikrovek | that's it |
| 16:00.22 | wcselby | hahqaha |
| 16:00.29 | Naikrovek | no idea why that would affect the time but damn it took me a while to notice |
| 16:00.35 | Naikrovek | because all the phones came back up properly |
| 16:00.39 | wcselby | lol |
| 16:00.40 | Naikrovek | they were just using older configs |
| 16:00.46 | wcselby | that's funny |
| 16:00.47 | chuckp | Naikrovek: is it failing over to tftp? |
| 16:00.53 | Naikrovek | no not failing over just failing |
| 16:01.01 | Naikrovek | again, no idea why taht would result in a blinking clock |
| 16:01.03 | Naikrovek | but it did |
| 16:01.05 | chuckp | that is odd usually it persists the time |
| 16:01.05 | wcselby | polycom's don't have an option to failover, right? |
| 16:01.19 | Naikrovek | wcselby: no failover |
| 16:01.26 | chuckp | don't believe so only aastra that I am aware of |
| 16:01.29 | wcselby | yeah that's what I though |
| 16:01.30 | Naikrovek | chuckp: that's why it took me so long to notice the ftp server being down |
| 16:01.32 | wcselby | thought* |
| 16:05.13 | chuckp | that is pretty annoying, unless do you have a time server entry in your DHCP server? |
| 16:05.35 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 16:05.42 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:12.19 | *** join/#asterisk vAd0r (~vAd0r@unaffiliated/vad0r) |
| 16:12.30 | vAd0r | if the channel is 2010-06-23 18:08:00 SIP/82.188.183.1752-b6d69090 5007 "Anonymous" <5007> s ANSWERED 19 |
| 16:12.35 | vAd0r | what is the source ip? |
| 16:12.47 | vAd0r | 82.188.183.175 |
| 16:19.42 | *** join/#asterisk sarah_p (~sarah_p@71.117.223.226) |
| 16:20.04 | *** join/#asterisk itiliti (~itiliti@74.222.60.231) |
| 16:21.02 | itiliti | afternoon All |
| 16:22.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:34.01 | *** join/#asterisk hfb (~hfb@pool-98-112-239-44.lsanca.dsl-w.verizon.net) |
| 16:34.41 | *** join/#asterisk Yoda_1204 (~chatzilla@12.111.169.98) |
| 16:37.20 | Yoda_1204 | does anyone know where i could download the channel driver module "chan_woomera"? |
| 16:37.52 | *** join/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 16:38.11 | *** join/#asterisk imox1234 (~imox1234@p4FC5C517.dip0.t-ipconnect.de) |
| 16:38.16 | xheliox | what's for lunch? |
| 16:38.51 | Yoda_1204 | mmm "5 guys burgers and fries" :) |
| 16:39.04 | xheliox | that would be tasty, i'm not sure we have any of those around here |
| 16:39.12 | p0seidon | Any one with experience in x100p clone card? |
| 16:39.50 | p0seidon | i've bought a card and it wont even boot my server |
| 16:40.06 | p0seidon | i know...shity card and that.... |
| 16:40.27 | xheliox | p0seidon: What do you mean it won't "boot my server"? Does the system even post? |
| 16:40.49 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:40.57 | *** join/#asterisk DJDeFi (~DeFi@209-234-189-154.static.twtelecom.net) |
| 16:41.04 | p0seidon | xheliox yes system wont even post. and weird thing is power led stays on all the time but a faint light on it. |
| 16:41.22 | xheliox | Bad card and/or something is wrong with your mobo? Try another PCI slot? |
| 16:41.47 | xheliox | As crappy as those generic cards are, they don't cause a system not to post. :P |
| 16:41.56 | xheliox | They might sound like crap and drop calls, but other than that.. |
| 16:42.14 | p0seidon | xheliox: i've tried other mobo system posts but i get that weird faint power light. |
| 16:42.26 | xheliox | Power light on what? |
| 16:42.39 | p0seidon | sorry for my bad english |
| 16:42.57 | p0seidon | let me try to explain |
| 16:43.11 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:43.55 | p0seidon | when i install the card and plug in the power cord to the power supply , as soon as I plug in the power cord, i get the power led on, but with a faint light... and i cant power the system. |
| 16:44.20 | p0seidon | but that power led gets somehow a litle bit of power , lol! |
| 16:45.12 | xheliox | What card are you using? |
| 16:45.32 | p0seidon | motherboard : gigabyte |
| 16:45.34 | xheliox | Sounds to me like you're using some sort of FXS card.. not a x100p. |
| 16:45.35 | p0seidon | card: x100p |
| 16:45.50 | *** join/#asterisk Whtsup (~sssi@WimaxUser376-208.wateen.net) |
| 16:45.59 | xheliox | Someone correct me if I'm wrong, but I've never seen an FXS x100p. |
| 16:46.00 | *** join/#asterisk joe_k (~joe@99-25-240-140.lightspeed.ltrkar.sbcglobal.net) |
| 16:46.13 | p0seidon | xheliox : x100p its fxo |
| 16:46.19 | p0seidon | 1 fxo port |
| 16:46.21 | xheliox | And you have to plug power to it? |
| 16:46.22 | Whtsup | <PROTECTED> |
| 16:46.33 | Whtsup | guys need help |
| 16:46.40 | chuckp | Anyone here using Fax For Asterisk, SendFAX function? |
| 16:46.40 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
| 16:46.43 | p0seidon | xheliox power? Like plug in the pstn? |
| 16:46.56 | xheliox | Whtsup: Make sure you have dahdi_dummy loaded and configured or some sort of timing device like a Digium card. Are you running on a virtualized system? |
| 16:46.56 | Whtsup | wht shud i do |
| 16:47.22 | Whtsup | well i have connected 2 asterisk boxes |
| 16:47.24 | Whtsup | with iax |
| 16:47.24 | xheliox | You're plugging in the phone line, you mean? |
| 16:47.40 | xheliox | Whtsup: And not answering my question helps you none. |
| 16:47.46 | p0seidon | xheliox: no i did not pluged in the phone line on it. Do you think that's the problem? |
| 16:48.18 | xheliox | p0seidon: No.. |
| 16:48.18 | Whtsup | xheliox listen i have connected 2 asterisk boxes with iax |
| 16:48.27 | xheliox | Oh ffs.. |
| 16:48.34 | Whtsup | one side tdm400p is installed |
| 16:48.47 | Whtsup | there is no error on that side |
| 16:49.01 | xheliox | Yes and you're trying to trunk those two boxes together, you can't use IAX trunking without a timing source. I already know what you're trying to do. |
| 16:49.10 | xheliox | BECAUSE THATS A TIMING DEVICE. |
| 16:49.14 | xheliox | throws things. |
| 16:49.35 | Whtsup | <PROTECTED> |
| 16:49.41 | p0seidon | xheliox, hmm... I guess i'll have to wait for x100p support. :| thanks for your help. Do you have any other idea? |
| 16:50.02 | xheliox | p0seidon: I'm still confused what you're "plugging in" -- a x100p should not require additional power from the power supply. |
| 16:50.34 | Whtsup | for fxs it required aaddtional powe supply |
| 16:50.39 | Whtsup | not for fxo pot |
| 16:51.04 | p0seidon | xheliox, I was saying if you think that the post problem is because I did not pluged in the phone line on the card. |
| 16:51.12 | Chainsaw | Any echo cancellation hardware (if that card supports such extensions) might require power even with an FXO port though. |
| 16:52.26 | xheliox | Have you ever seen a generic x100p with EC? |
| 16:52.54 | p0seidon | sorry xheliox what do you want to say with "EC" ? |
| 16:53.23 | Chainsaw | Oh, the rebadged PCI modem? |
| 16:53.27 | xheliox | p0seidon: I was asking Chainsaw, but EC is echo cancelation. |
| 16:53.27 | Chainsaw | You deserve all you get. |
| 16:53.29 | Chainsaw | walks away |
| 16:54.40 | xheliox | I already covered that part. ;) |
| 16:54.59 | xheliox | But his system isn't posting, so the card is probably just f'ed. |
| 16:55.27 | *** join/#asterisk Jomu (~Jomu@188.124.200.2) |
| 16:55.31 | xheliox | Beyond being f'd in other ways too. |
| 16:55.53 | [TK]D-Fender | p0seidon: [12:52]<xheliox>Have you ever seen a generic x100p with EC? <- lol... there is no 1-port analog card with EC |
| 16:55.59 | joe_k | i would be using my x100p but there are no more drivers for freebsd :( |
| 16:56.22 | [TK]D-Fender | (HWEC) |
| 16:56.35 | p0seidon | damn card :P |
| 16:56.54 | xheliox | [TK]D-Fender: Yeah, I figured as much. :P |
| 16:57.10 | xheliox | Then what the hell is he trying to plug in? ;) |
| 16:57.20 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:57.26 | *** join/#asterisk resno (~bryan@cpe-075-177-155-247.nc.res.rr.com) |
| 16:58.08 | [TK]D-Fender | xA fried piece of shit |
| 16:58.21 | [TK]D-Fender | p0seidon: Your card is crispy. Get over it. |
| 16:58.39 | p0seidon | LOL i know |
| 16:58.49 | *** part/#asterisk resno (~bryan@cpe-075-177-155-247.nc.res.rr.com) |
| 16:58.56 | p0seidon | shity card... |
| 16:59.16 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 16:59.24 | xheliox | It's like no one is willing to believe those cards are crap until they actually go out and waste the $15 |
| 16:59.33 | Chainsaw | xheliox: *nod* Very strange. |
| 16:59.35 | p0seidon | thanks for your help anyway :) |
| 17:00.22 | [TK]D-Fender | xheliox: His problems have nothing to do with the X100P being what it is. His card is fried. That is all |
| 17:00.35 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 17:00.35 | xheliox | Chainsaw: Hell, I did years ago too. I'm not unwilling to let them see for themselves how bad they are, it's not like they're hard to configure if it's a functioning piece of hardware. |
| 17:01.16 | xheliox | [TK]D-Fender: Yeah, I wasn't really commenting on his specific situation, more or less a general observation. |
| 17:01.46 | p0seidon | yep my card probably iss fried i'll ask x100p for a refund and get a sangoma or something like that |
| 17:02.21 | p0seidon | i just wanted to learn something but guess i've just toke the wrong step in hardware |
| 17:03.36 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 17:04.55 | Whtsup | can i use jitterbuffer in single iax peer |
| 17:08.20 | drmessano | [12:55] <joe_k> i would be using my x100p but there are no more drivers for freebsd :( <-- Not a FreeBSD guy, but doesn't BSD pride itself on security and stability. Maybe that extends to the supported hardware too. Kudos to BSD! |
| 17:09.01 | [TK]D-Fender | p0seidon: "just wanting to learn" isn't a great reason to buy hardware like that |
| 17:09.03 | xheliox | snickers |
| 17:09.22 | [TK]D-Fender | p0seidon: Because what little it takes to use it says that after you've spent that 10 minutes.... whats the point? |
| 17:10.42 | *** join/#asterisk markfeatherston_ (~mark.feat@71.4.41.98.ptr.us.xo.net) |
| 17:10.59 | itiliti | afternoon. |
| 17:11.47 | itiliti | Is there any way I can not use the answer app to answer a call, but dial a call out, and when the dialed call is answered, answer the first call? |
| 17:12.31 | itiliti | I am using asterisk ro record calls from a dialer, but when I use Answer(), first, the dialer thinks the call was answered, so it screen pops to the user while still ringing... |
| 17:13.01 | *** part/#asterisk Jamuel (~jamuel@c-24-7-72-218.hsd1.ca.comcast.net) |
| 17:13.15 | [TK]D-Fender | itiliti: So stop calling Answer() <- |
| 17:15.10 | jstapleton | what might cause - " == Spawn extension (devices, o, 0) exited non-zero on 'SIP/aethra-0000005a' |
| 17:17.33 | itiliti | I tried to not use Answer, but then it tells me the call comonig from the ip of the source is not an authorized peer. |
| 17:23.22 | chuckp | Anyone here using Fax For Asterisk, SendFAX function? |
| 17:23.59 | itiliti | If I dont answer it, how can I get it to dial out, and when that call is connected, conenct the incoming call, and then record it? I have the recording piece figured out. It is just the signaling to the source I want happen last, after the call is answered on the trunk side... |
| 17:24.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:24.10 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:24.17 | *** join/#asterisk sarah_p_ (~sarah_p@108.1.169.2) |
| 17:29.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:29.30 | *** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 17:33.18 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.112.66.dsl.dyn.forthnet.gr) |
| 17:33.57 | [TK]D-Fender | itiliti: You are dialing OUT, so what is this nonsense of an INCOMING call? |
| 17:34.42 | [TK]D-Fender | [13:17]<itiliti>I tried to not use Answer, but then it tells me the call comonig from the ip of the source is not an authorized peer. <- this is completely wrong |
| 17:35.03 | [TK]D-Fender | itiliti: The dialplan has NO impact on your call being authed |
| 17:41.52 | itiliti | I have a dialer that is coming form a PRI to SIP gateway, then through asterisk, to another SIP to PRI gateway. |
| 17:42.29 | itiliti | I want to record all of the calls, but the dialer thinks that the call has already been answered, becasue of using the Answer(). |
| 17:42.45 | itiliti | Here is what my dialplan looks like now... |
| 17:43.27 | [TK]D-Fender | itiliti: Don't. Call. Answer() <- |
| 17:46.42 | BarthezZ | I want to make a small set of special extensions in my asterisk, in which a user can "login" so that extension is available on whatever phone he's sitting at, now i'm thinking of what would be the best way to achieve that. The internal astdb to make it dynamic or a queue-like way with just one agent? |
| 17:48.33 | *** join/#asterisk pieter30 (~pieter@source.andrewx.net) |
| 17:49.16 | pieter30 | Hi ! I would like to sell an older PRI card from Digium , can someone recommend any forum that has a sell/buy section ? |
| 17:49.41 | itiliti | here is the link to my dialplan right now.... pastebin.com/t6WamGHB |
| 17:49.42 | [TK]D-Fender | BarthezZ: chan_agent is kinda icky. I'd do it in more regular dialplan personally. |
| 17:50.11 | BarthezZ | hmm ok [TK]D-Fender, so am i on the right way like this: |
| 17:50.17 | [TK]D-Fender | itiliti: You seem to have a serious learning issue |
| 17:50.25 | [TK]D-Fender | itiliti: exten => _X.,1,Answer() <--------STOP ANSWERING THE CALL HERE! |
| 17:50.42 | BarthezZ | 123 to login, add fromchan (SIP/userid) to the db, create 190 till 199 and point them to the SIP/userid entry in the db? |
| 17:51.30 | BarthezZ | it feels so "hacky", but it's the only thing i can come up with :P |
| 17:52.01 | [TK]D-Fender | BarthezZ: Something like that. |
| 17:52.46 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:52.48 | [TK]D-Fender | BarthezZ: and yes, it is kinda "hacky" in that you are doing the processing code yourself, but not that big a deal. You'll also have to lookup a CID name & number to present for calls from the device, etc |
| 17:54.15 | BarthezZ | yeah, i was already looking at that |
| 17:54.31 | [TK]D-Fender | BarthezZ: not that bad really. |
| 17:54.41 | BarthezZ | found the app DumpChan(), but no variable in ther actually represents the information I need |
| 17:54.56 | BarthezZ | it has for example IAX2/bartlaptop-215, but still have to stripp off the -XXXX |
| 17:55.27 | [TK]D-Fender | BarthezZ: another idea is to script up something to mod your sip.conf to change the "mailbox=" line and change the box it monitors so that you can get MWI to follow you as well. |
| 17:56.24 | [TK]D-Fender | BarthezZ: For your CID lookup you'll need to lookup in another table (AsDB,, external, maybe some dialplan direct, etc) |
| 17:56.43 | [TK]D-Fender | BarthezZ: Depends where you you want to update things and how big a setup you are looking at |
| 17:57.12 | [TK]D-Fender | BarthezZ: For a small one I would actually just use pure dialplan and not even AstD for looking up the name & number |
| 17:57.14 | BarthezZ | uhmm, updates should take place at locally connected SIP hardphones (the "logging in and out") and maybe from the manager or something later on |
| 17:57.31 | itiliti | So, if I stop answering the call, then it will still dial out, and also copnnect the call after it is connected? |
| 17:57.39 | [TK]D-Fender | itiliti: YES. |
| 17:57.54 | [TK]D-Fender | itiliti: when the far side answers, the calling channel gets answered and they are bridged |
| 17:57.55 | BarthezZ | well it's, for now, probably just about 10 clients who need to be able to allways be reachable, but the setup has some grow-perspective |
| 17:58.13 | [TK]D-Fender | BarthezZ: Whats the realistic size it might grow? |
| 17:58.33 | BarthezZ | it now has abou 60 clients on 1 location, it could grow up to 400 |
| 17:58.56 | BarthezZ | slowly, but that's about the end point in i think 6 months |
| 17:58.59 | [TK]D-Fender | BarthezZ: Your call as to where to start. |
| 17:59.34 | BarthezZ | calls to the extension you mean? |
| 17:59.59 | BarthezZ | originating from internal users and possibly a "press your extension to directly contact person" from the outgoing line |
| 18:03.39 | [TK]D-Fender | meant "call" as "choice" |
| 18:03.52 | [TK]D-Fender | ok, off for a while. |
| 18:03.54 | [TK]D-Fender | BBL |
| 18:04.19 | BarthezZ | oh lol :) |
| 18:04.22 | BarthezZ | thanks [TK]D-Fender :p cya |
| 18:04.31 | itiliti | Gr8, thanks for the spanking :) |
| 18:04.32 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 18:04.42 | itiliti | dont mean to be learnign challenged... |
| 18:05.41 | markfeatherston_ | I'm helping a friend who recently had an intrusion on their asterisk system and started making calls trying to trick people out of credit card info. They had a default manager password, but are there any gotchas that this person could use to get back in? |
| 18:07.52 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 18:07.59 | *** part/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 18:09.27 | Chainsaw | markfeatherston_: An owned box can not be trusted. |
| 18:09.55 | Chainsaw | markfeatherston_: Take it offline, make backups of config files only (no kernels, no binaries, no exceptions) and reinstall from scratch. |
| 18:10.04 | Chainsaw | markfeatherston_: Anything less is irresponsible at best. |
| 18:13.01 | xheliox | And any config files you do copy, comb thru them very very very throughly. |
| 18:13.10 | Chainsaw | ^ I second that. |
| 18:13.51 | markfeatherston_ | I agree, but I'm only doing this as a favor. I'll be making that recommendation though |
| 18:13.51 | WIMPy | Or use a trusted backup or diff them to it. |
| 18:14.07 | markfeatherston_ | ... backups would have been nice |
| 18:14.17 | Chainsaw | markfeatherston_: Then it depends on whether you want the repeat business (consultancy basis) or whether you want it properly fixed (a friend). |
| 18:14.34 | markfeatherston_ | true, i'll have to talk to them about that |
| 18:14.44 | Chainsaw | markfeatherston_: With people who leave management passwords on default... that's a choice you'll have to make now. |
| 18:15.15 | markfeatherston_ | the guy who set it up originally doesn't work there anymore, so it's not a mistake that would be happening again |
| 18:15.41 | xheliox | And you might want to read... README-SERIOUSLY.bestpractices.txt in the Asterisk source directory. If they had a default manager password on an outside interface, chances are they're doing many other stupid things. |
| 18:16.47 | xheliox | And are you sure it was because of the manager password or were they doing a dialplan injection? |
| 18:17.19 | *** join/#asterisk Delido1983 (~Delido198@212.144.236.196) |
| 18:17.43 | Chainsaw | markfeatherston_: This sounds like a possible full-time job opportunity. Unmaintained & unloved software installations of any kind are. |
| 18:17.59 | Delido1983 | hey which dahdi complete version from svn can i download? i dont find a branch version only an tag download |
| 18:18.05 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:18.06 | *** join/#asterisk githogori (~githogori@168.61.10.135) |
| 18:18.46 | Delido1983 | what is "tags"? is this a final release? |
| 18:20.03 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 18:21.05 | Qwell | Delido1983: There are no branches of dahdi-linux-complete. It only exists as tags. If you want to use a branch, you'll need to install dahdi-tools and dahdi-linux separately. |
| 18:22.00 | Delido1983 | @Qwell: is tags an stable? |
| 18:22.28 | leifmadsen | tags == versions |
| 18:22.36 | leifmadsen | tagged versions do not change |
| 18:22.40 | leifmadsen | ~asterisk-versioning |
| 18:22.40 | infobot | asterisk-versioning is, like, http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 18:22.46 | leifmadsen | please read that |
| 18:23.25 | Delido1983 | @Qwell: @leifmadsen: thanks for the link |
| 18:23.45 | markfeatherston_ | xheliox: I'm not sure if it was a dialplan injection. I'm not sure what that is myself so I'll have to look into it more. They were calling out from several user extensions so I just assumed they just used manager to read the config files. |
| 18:25.50 | xheliox | Don't assume. |
| 18:25.57 | xheliox | Read that best practices txt, it's important. |
| 18:26.19 | xheliox | And if you don't understand it, make sure you do. Or have them hire someone with experience. I know a guy. :) |
| 18:27.42 | Chainsaw | Don't play hard to get xheliox, just give him your business card already :) |
| 18:27.52 | xheliox | Too transparent? :P |
| 18:28.26 | markfeatherston_ | xheliox: small company, they only have a consulting company for IT for the most part |
| 18:29.33 | xheliox | markfeatherston_: Then hire a consultant for the Asterisk box. |
| 18:29.45 | markfeatherston_ | xheliox: I agree, they should. Not my call though |
| 18:30.02 | markfeatherston_ | xheliox: I recommended it already : / |
| 18:30.08 | *** join/#asterisk x-demon (xdemon@2001:ba8:1f1:f0b8:216:5eff:fe00:135) |
| 18:30.13 | xheliox | Well when they get a giant phone bill or have the cops come by asking questions, they might see there was value in taking care of it properly the first time. |
| 18:30.36 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 18:30.45 | markfeatherston_ | xheliox: they went from 150 calls a day to >700 a day (started yesterday). |
| 18:31.42 | xheliox | Yeah, it's not uncommon. I see attempts a lot on most of my systems. |
| 18:32.01 | Chainsaw | *nod* They try at least once a week. |
| 18:32.34 | markfeatherston_ | yikes |
| 18:33.34 | Chainsaw | Believe it or not, the internet is not a friendly place. |
| 18:33.38 | markfeatherston_ | hah |
| 18:33.39 | Chainsaw | Do not hook up unprotected systems to it. |
| 18:35.41 | markfeatherston_ | I've seen some of the code by this developer... he doesn't concern himself with security much |
| 18:36.58 | xheliox | that makes xheliox a sad panda. |
| 18:38.20 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:39.27 | *** join/#asterisk [Outcast] (~james@64.202.62.5) |
| 18:41.19 | wcselby | I got a fail2ban email last night ---> The IP 67.212.176.82 has just been banned by Fail2Ban after 56040 attempts against ASTERISK. |
| 18:41.23 | wcselby | 56040! |
| 18:41.38 | wcselby | I'm pretty sure it monitors the file for changes, and usually catches stuff within a minute at the most |
| 18:41.49 | wcselby | I haven't had a chance to login to the server yet to reveiw the logs, but damn |
| 18:41.52 | xheliox | Yeah. |
| 18:42.02 | xheliox | They send multiple login attempts at once. |
| 18:42.04 | wcselby | usually it's in the range of 200-300 |
| 18:42.08 | xheliox | So fail2ban doesn't catch it instantly. |
| 18:42.14 | wcselby | yeah, I've seen it |
| 18:42.33 | xheliox | It's not ideal, but even if they get into a SIP account, in theory, they'll get banned within 60 seconds. |
| 18:42.34 | markfeatherston_ | wow... |
| 18:43.01 | *** join/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 18:43.04 | *** part/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 18:43.10 | xheliox | And I haven't seen what passwords they're using, but I suspect it's like 200/200 and not anything more complex like the passwords I'm using. |
| 18:44.01 | xheliox | And just for good measure, for any remote users, we have a dialplan pin they have to enter for LD calls, so even if you got the SIP account, you probably couldn't do much with it without also then guessing the 5 digit pin -- easily brute forced, but who is scripted for that? |
| 18:44.23 | xheliox | And of course, even more ideally, have them VPN and also use an IP ACL. |
| 18:45.14 | wcselby | Corydon76-dig - your email to the list about randomness in password generation just hurt my head. thought I'd share. :) |
| 18:45.38 | xheliox | Now I'm curious. :) |
| 18:45.59 | xheliox | the main Asterisk list? |
| 18:46.04 | wcselby | http://pastebin.com/V7Cd3yr5 |
| 18:46.14 | wcselby | there you go xheliox |
| 18:46.33 | xheliox | blinks |
| 18:46.37 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 18:46.41 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 18:46.57 | *** join/#asterisk k4tana (~roni@190.196.71.206) |
| 18:47.09 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 18:47.21 | k4tana | hi everybody .. a little question . how do i compile asterisk with sqlite support ? |
| 18:47.43 | wcselby | is that in asterisk-addons? |
| 18:48.04 | p3nguin | I think think you could install sqlite, then configure asterisk to use it. |
| 18:48.38 | wcselby | i think there's a module for it, same as there's one for MySQL |
| 18:49.15 | k4tana | is in asterisk , but was not compiled .. mysql is in the addons .. but i dont remember if i compile after install sqlite .. i will check that |
| 18:49.19 | p3nguin | Install sqlite before building asterisk. See if it is available in the menu. |
| 18:50.07 | xheliox | You need the development libs installed for sqlite, obviously. |
| 18:50.39 | xheliox | cdr sqlite is in 1.4 & 1.6 and res_config_sqlite is in 1.6 |
| 18:50.42 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 18:50.51 | xheliox | it's just mysql that's in the addons. |
| 18:52.26 | *** join/#asterisk Z_God (~julius@wlan226192.mobiel.utwente.nl) |
| 18:53.02 | markfeatherston_ | Has anyone here ran asterisk on ec2? |
| 18:53.48 | xheliox | Yeah. |
| 18:53.54 | Naikrovek | yeah i have also |
| 18:53.56 | Naikrovek | it works well |
| 18:54.00 | markfeatherston_ | Good to hear |
| 18:54.02 | xheliox | Eh. |
| 18:54.03 | Naikrovek | it did in my case, anyway |
| 18:54.11 | Naikrovek | people who haven't tried it will tell you it doesn't work |
| 18:54.17 | Naikrovek | but i tried it and it works well |
| 18:55.52 | markfeatherston_ | I did get a large instance running. getting the modules was a bit of a pain. Are there any problems you've had from ec2 doing this? |
| 18:56.39 | Naikrovek | no |
| 18:56.58 | Naikrovek | i tested by putting a few phones in a meetme and listened for any audio issues |
| 18:57.01 | Naikrovek | heard none |
| 18:57.34 | Naikrovek | then i did some long-term echo tests and recorded the audio. looked for things other than the sine wave i put in, found nothing |
| 18:57.47 | xheliox | It's not "audio" issues.. it's the timing of when the other person is heard. As the conversation goes longer, the length of time that the other callers hear the audio increases. |
| 18:57.48 | markfeatherston_ | awesome, thanks |
| 18:57.59 | Naikrovek | then i put about 40 phones in a single meetme and listened for any issues |
| 18:58.04 | Naikrovek | again, nothing. worked great |
| 18:58.35 | markfeatherston_ | xheliox: so the audio delay gets larger depending on how long the call is? |
| 18:58.46 | xheliox | markfeatherston_: that's my experience. |
| 18:58.51 | Naikrovek | you're thinking of latency, which is not what you described. it does not increase as the call lengthens |
| 18:59.18 | Naikrovek | at least i think you're thinking of latency |
| 18:59.35 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:00.05 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-70-171.home.otenet.gr) |
| 19:00.54 | xheliox | I'm telling you the consequences of having no timing device or a poor one and using MeetMe. It's more complicated than just hearing poor audio quality. I certain I'm not smart enough to explain why it does what it does, merely that I can tell you what I've experienced. |
| 19:01.09 | Naikrovek | everyone says that |
| 19:01.20 | Naikrovek | but linux 2.6 and xen (what ec2 uses) cooperate on timing |
| 19:01.30 | Naikrovek | you get good hardware timing from the cpu via dahdi_dummy |
| 19:01.58 | Naikrovek | just as i do on my local hyper-v and vmware esxi machines |
| 19:02.18 | xheliox | Not having using MeetMe on EC2, I can't tell you if it's any good or not, but I am advising what I have seen elsewhere with no timing device. |
| 19:02.30 | Naikrovek | EC2 HAS A TIMING DEVICE |
| 19:02.38 | Naikrovek | just like vmware or hyper-v |
| 19:02.39 | xheliox | And based on my experiences elsewhere, your test wasn't sufficient enough to determine if you had problems or not. |
| 19:02.50 | xheliox | And did I say at any point that it didn't? |
| 19:02.56 | Naikrovek | right. hours of testing isn't enough |
| 19:03.00 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 19:03.21 | xheliox | If those hours were spent merely on listening for audio quality problems, you were fishing with the wrong pole. |
| 19:03.52 | Naikrovek | friend of mine runs his entire phone system on ec2 |
| 19:03.54 | Naikrovek | using asterisk |
| 19:03.59 | Naikrovek | has for about 18 months |
| 19:04.02 | Naikrovek | NO PROBLEMS |
| 19:04.06 | markfeatherston_ | I'm looking to migrate my call center there now |
| 19:04.16 | markfeatherston_ | xheliox: is there anywhere I can read more about this issue? |
| 19:04.17 | Naikrovek | he has hundreds of endpoints |
| 19:05.04 | xheliox | Great and most of Asterisk's modules/apps do not require a solid timing device. So I completely BELIEVE YOU. But thanks for yelling. |
| 19:05.43 | wcselby | markfeatherston_ - go through a few of the astricon archieves for last year |
| 19:05.52 | wcselby | there were some talks on xenifiing asterisk |
| 19:06.04 | wcselby | also using asterisk in the cloud (amazon's cloud, specifically) |
| 19:06.16 | wcselby | apparently someone already has an ec2 template that has asterisk ready to go |
| 19:06.40 | wcselby | i forget the name of the person who gave that talk, but they were from rf.com I think (Ring Free, or something like that) |
| 19:07.19 | Naikrovek | verdict: EC2 is a proven good platform for Asterisk |
| 19:07.22 | *** join/#asterisk Mango (~iMango@d154-20-89-230.bchsia.telus.net) |
| 19:07.35 | Naikrovek | I'm so tired of people who've never done it saying otherwise |
| 19:07.47 | xheliox | scrolls back |
| 19:07.52 | xheliox | Where did anyone say that? |
| 19:07.58 | wcselby | Naikrovek - ec2 sucks for asterisk, plain and simple |
| 19:08.01 | wcselby | ...... |
| 19:08.02 | wcselby | :P |
| 19:08.07 | xheliox | :P dick. |
| 19:08.09 | wcselby | }=) |
| 19:08.19 | Naikrovek | not talking about you - and this has been talked about MANY times in the year or so I've hung out in here |
| 19:08.28 | Naikrovek | this is not the first time this conversation has come up at all |
| 19:08.56 | wcselby | it's not your faul xheliox - he's right, alot of people come in here with preconceived ideas and opinions on the matter, who haven't actually tried it. |
| 19:09.01 | Naikrovek | it usually consists of a guy joining, asking about EC2, getting negative responses, leaving, then me asking if anyone who responded has actually tried it, and they all answer "well, no" |
| 19:09.38 | Naikrovek | then they show me articles about it written in 2006 on 32-bit platforms without virtualization support as evidence |
| 19:09.44 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 19:10.10 | Mango | Does anyone here have a CallWithUs account and five minutes to help me test something? |
| 19:10.14 | *** join/#asterisk [T]ank1 (~ckwall@206.71.78.158) |
| 19:10.15 | pieter30 | Hi ! I would like to sell an older PRI card from Digium , can someone recommend any forum that has a sell/buy section ? |
| 19:10.19 | Naikrovek | on any MODERN virtualization platform (64-bit, hardware virtualization support) asterisk runs perfectly fine |
| 19:10.40 | [T]ank1 | pieter30: ebay |
| 19:10.56 | xheliox | wcselby: Sure, that's true, but I wasn't making any claim either way, just highlighting some potential concerns. And that I felt that Naikrovek's process of testing may have been flawed. |
| 19:11.09 | Naikrovek | maybe it is flawed |
| 19:11.18 | Naikrovek | but the result warranted a real-world test |
| 19:11.21 | Naikrovek | which was done |
| 19:11.26 | Naikrovek | that result was favorable |
| 19:11.37 | Naikrovek | now my friend has i think 250 phones connected to an EC2 asterisk server |
| 19:11.39 | pieter30 | [T]ank1: ebay doesn't have a forum ! |
| 19:11.45 | [T]ank1 | anyone here able to assist me in getting h323 added to my asterisk installation? I have installed libopenh323. using ubuntu 10 and asterisk 1.6.2.9 |
| 19:12.07 | [T]ank1 | pieter30: why do you want a forum? I have had great success selling digium hardware on ebay. |
| 19:12.15 | xheliox | So after all this, wouldn't have been easier to say you had real users on the system using MeetMe rather than saying you had 40 SIP test calls listening for audio issues? |
| 19:12.17 | [T]ank1 | larger community of people looking for hardware |
| 19:12.26 | pieter30 | [T]ank1: I didn't and I am not in the states |
| 19:12.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:12.44 | [T]ank1 | ebay is world wide. |
| 19:12.59 | [T]ank1 | so, to answer your direct question then... no, i dont know of a forum. |
| 19:13.28 | wcselby | ebay is a forum, of sorts. not necessarily an internet forum though |
| 19:13.39 | wcselby | sort of like saying IRC is a forum |
| 19:13.47 | wcselby | or astricon is a forum |
| 19:13.50 | wcselby | etc |
| 19:13.54 | Naikrovek | xheliox: perhaps, but I don't use that system day to day. I only know about it |
| 19:14.18 | [T]ank1 | back to my h323 question... so i have installed libopenh323 on ubuntu and asterisk 1.6.2.9. When i do my make menuselect it still shows that the chan_h323 option is missing the openh323 dependency. What could I be missing? |
| 19:14.22 | wcselby | and if I recall, ebay does have a forum, my wife used to chat on it |
| 19:14.23 | Naikrovek | the gurus in here usually say to listen for timing issues on meeme calls and echo tests, so I did. I listened for a long time |
| 19:14.37 | pieter30 | [T]ank1: on "which" ebay did you have success selling digium , there are multiple |
| 19:14.45 | [T]ank1 | ebay.com |
| 19:14.47 | Naikrovek | and didn't have a single skip in the audio or anything - it was good |
| 19:14.53 | wcselby | [T]ank1 - there have been troubles lately getting h323 to work |
| 19:15.06 | [T]ank1 | I also tried in 1.4 with no success |
| 19:15.07 | wcselby | i remember reading a thread about it on the mailing list, I think tzafrir_laptop was a part of it |
| 19:15.08 | pieter30 | that isn't an option for me, as shipping overseas would be too expensive |
| 19:15.38 | wcselby | pieter30 - where are you shipping from? |
| 19:15.41 | [T]ank1 | pieter30: the buyer pays it... I just sold over seas, it was not an issue for the buyer. |
| 19:16.41 | wcselby | and to whoever was asking about it earlier, the ec2 talk was I believe called "voip in a cloud", on Wednesday. But I just clicked the link and the presentation wasn't available on the astricon site |
| 19:16.49 | pieter30 | [T]ank1: as I am familiar with EBAY.COM they require to charge only for inland shipping, I am not sure if you can specify your own price |
| 19:16.55 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 19:17.09 | pieter30 | wcselby: Europe |
| 19:17.27 | [T]ank1 | pieter30: may want to look for another option then. Sorry I could not help |
| 19:17.47 | pieter30 | [T]ank1: no problem, at least you answered |
| 19:18.10 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 19:18.11 | wcselby | pieter30 - at least in the states you can state that the buyer pays actual shipping costs. I don't know about any other country's ebay. |
| 19:18.23 | WIMPy | pieter30: I just got one from ebay. |
| 19:18.38 | pieter30 | [T]ank1: did you put the auction starting at 1 usd ? |
| 19:18.44 | [T]ank1 | no |
| 19:18.51 | wcselby | pieter30 - try maybe the digium.com forums, the trixbox forums, or someplace like that |
| 19:18.55 | ruben23 | hi nay idea, when i call using SIP on my asterisk server using VOIp i got this error log----------------------------> http://pastebin.com/n9zTizUS |
| 19:19.32 | [T]ank1 | ruben23: share your configs... |
| 19:19.38 | [T]ank1 | looks to me like a config issue |
| 19:20.17 | ruben23 | [T]ank1: on what..? sip.conf or dial plan..? |
| 19:20.22 | [T]ank1 | yes |
| 19:20.37 | [T]ank1 | both |
| 19:20.44 | [T]ank1 | otherwise all we can do is guess |
| 19:23.27 | grinder13 | hello! i have the following problem: Two Asterisk servers. I am trying to connect them via IAX. A registered IAX user in the first server tries to call registered IAX extension in the second server, but keeps getting "No authority found". When the same IAX registered user calls an IAX extension in the same server, there is no problem. You can see my confs of the two servers here: http://pastebin.com/KrypXZcg , http://pastebin.com/SJqPXHmB |
| 19:23.34 | ruben23 | this is my config----> http://pastebin.com/iDj02yz2 |
| 19:23.35 | wcselby | wow, gaylord hotel costs 319. / night. I hope there's going to be an astricon convention discount rate |
| 19:23.52 | ruben23 | <PROTECTED> |
| 19:24.44 | Naikrovek | grinder13: are you registering each extension on the other server in iax.conf?? |
| 19:25.21 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 19:26.14 | *** part/#asterisk Delido1983 (~Delido198@212.144.236.196) |
| 19:26.15 | Mango | How does md5secret increase security? The only way I can think of is if I use the same password in multiple places, it will be difficult to find out what that is. Is there another advantage? |
| 19:26.32 | Naikrovek | md5 passes the hash of the password, not the password itself |
| 19:26.47 | [T]ank1 | ruben23: do you have the configuration set up on the peer you are sending the call to? |
| 19:27.12 | grinder13 | Naikrovek check my confs. you 'll see what I am doing. I am following the example from here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 19:27.36 | Naikrovek | grinder13: i've never done it that way... |
| 19:27.42 | grinder13 | user 3001 registers in server A and tries to call extension 4000 registered in server B |
| 19:27.44 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 19:28.05 | Naikrovek | your configs say trunk=no, example says trunk=yes |
| 19:28.11 | ruben23 | <PROTECTED> |
| 19:28.20 | [T]ank1 | can you share that config too? |
| 19:28.32 | [T]ank1 | need to see it all to know whats going on |
| 19:28.33 | grinder13 | Naikrovek: yes, but that's no a problem. even with trunk=yes the problem remains |
| 19:28.40 | Naikrovek | alright |
| 19:28.52 | ruben23 | <PROTECTED> |
| 19:28.56 | ruben23 | thats all |
| 19:29.03 | [T]ank1 | have you registered? |
| 19:29.26 | ruben23 | its IPbased sutherntication and amy asjerisk is behind NAT and firewall |
| 19:29.34 | ruben23 | all ports are opened |
| 19:29.52 | [T]ank1 | then why not just dial to the ip? |
| 19:29.53 | grinder13 | SIP example from here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html works perfectly, but with IAX there is the "no authority problem" |
| 19:30.44 | ruben23 | how do i set the string for that..? |
| 19:31.51 | wcselby | grinder13 - there were new iax authentication routines added sometime around 1.4.26 I think, you may need to read up on those |
| 19:33.06 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 19:33.36 | [T]ank1 | ruben23: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 19:34.47 | [T]ank1 | ruben23: are you behind nat? |
| 19:34.54 | ruben23 | yes |
| 19:35.07 | ruben23 | but all are port forwarded and ports are all opened |
| 19:35.23 | grinder13 | wcselby, can you point to any link about that? i 've already read the relevant info at voip-info.org, but I can't see what I am doing wrong |
| 19:35.29 | [T]ank1 | need to specify that somewhere. nat=yes |
| 19:36.26 | [T]ank1 | ruben23: i recommend looking at the example sip.conf in the install directory. there are quite a few examples there that i think would really answer you questions. |
| 19:36.27 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 19:37.21 | [T]ank1 | ruben23: who is your host? |
| 19:37.37 | [T]ank1 | US based? |
| 19:38.32 | markfeatherston_ | Can anyone point me in the right direction for sending out a sip notify when record-on-demand is used? |
| 19:39.16 | wcselby | grinder13 - the sample iax.conf file in source |
| 19:39.57 | ruben23 | yes |
| 19:40.00 | ruben23 | delridgeis |
| 19:40.19 | [T]ank1 | How are their rates? What do they charge per minute? |
| 19:41.01 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:41.34 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 19:45.19 | ruben23 | 0.011 |
| 19:50.39 | *** join/#asterisk brettnem (~brett@63.246.161.15) |
| 19:50.43 | brettnem | hello all! |
| 19:51.04 | [T]ank1 | ruben23:how much for DIDs |
| 19:51.07 | brettnem | hey, has anyone seen Poly IP330s randomly freeze? |
| 19:51.17 | ruben23 | [T]ank1: i think i see the problem, it rings when i used dialcode 1 and it will congest when no dialcode..but when i sued a dialcode like 12127775678 it says like this, it rings and stop ----> http://pastebin.com/g424UGsG |
| 19:51.45 | [T]ank1 | so your not sending what they are expecting it sounds like |
| 19:51.57 | ruben23 | but ti should not used dial 1 since the caller and the server itself dialing is in the US. |
| 19:52.05 | [T]ank1 | they must be expecting 11 digits. |
| 19:52.12 | [T]ank1 | I have seen carriers require it |
| 19:52.44 | ruben23 | but why, i tried 11 digits it ring once then stop---see this http://pastebin.com/g424UGsG |
| 19:53.14 | ruben23 | there some how some kind of bridging |
| 19:53.26 | brettnem | with these Polycom 330s, we'll just come up to one and see that the clock stopped updating a few hours ago.. buttons don't do anything.. totally locked up.. reboot and it's fine.. maybe a day.. maybe two |
| 19:54.40 | [T]ank1 | ruben23: have you set the nat=yes? also if they are doing a reinvite and such you will want to specify externhost=<ip> |
| 19:56.47 | ruben23 | ow i havent, ok ill try it now |
| 19:56.48 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 20:04.10 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:04.13 | *** join/#asterisk ldiamond (60167564@gateway/web/freenode/ip.96.22.117.100) |
| 20:05.19 | ldiamond | I need a client to test my setup (i.e. place calls to my sip address). I'm running linux. Any suggestions? (I tried ekiga but it didn't work) |
| 20:05.43 | chuckf | twinkle works for me |
| 20:06.56 | chuckf | brettnem: are the clocks flashing? |
| 20:09.10 | *** join/#asterisk sirdopes (~user@espion.houston.hostgator.com) |
| 20:09.57 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 20:10.33 | [T]ank1 | so, does the issue with h323 stem from asterisk? Or is it an issue with openh323? |
| 20:12.24 | ldiamond | I'm looking for a good and cheap SIP provider that supports SMS, ideally located in Canada, otherwise in US but with Canada DIDs available. |
| 20:12.53 | ldiamond | Anyone had good experience with certain providers? |
| 20:13.52 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 20:14.07 | boodu | hello |
| 20:18.39 | brettnem | chuckf: No the clocks are frozen.. buttons frozen.. clock displays time when it was locked up.. back to the future style |
| 20:19.09 | brettnem | chuckf: I have dozens of Polycom 330s at several sites on several asterisk boxes all freezing up. |
| 20:19.17 | Mango | ldiamond: If you find one that supports SMS, I would love to know about it. |
| 20:19.30 | brettnem | not all at the same time.. doesn't appear to be "caused" by doing anything on the phone..might happen on it's own |
| 20:19.40 | FutureWeb | I don`t know if anyone can help me, but I currently Need Incoming Numbers in Malta as in Maltese (356) DID Numbers, if anyone knows a website which accepts paypal and provides them, please let me know in a pm. thanks :) |
| 20:19.42 | ldiamond | Mango, I just found callcentric.com, however, I have not yet been able to test it properly. |
| 20:20.57 | Mango | Ah, I didn't realize they supported SMS. |
| 20:26.45 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:26.47 | [sr] | howdy |
| 20:26.50 | [sr] | hi WIMPy |
| 20:26.54 | [sr] | one question guys |
| 20:27.07 | [sr] | i'm tring to access my mailbox, using *97 or *98 |
| 20:27.09 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:27.13 | [sr] | when i enter the password |
| 20:27.15 | WIMPy | hi [sr] |
| 20:27.24 | [sr] | or the mailbox number when asked |
| 20:27.50 | [sr] | i get the numbers doubled and can't login, for example this |
| 20:27.54 | [sr] | [Jul 1 21:24:01] VERBOSE[32727] logger.c: -- Incorrect password '11' for user '69' (context = default) |
| 20:28.02 | [sr] | i just entered "1" |
| 20:28.31 | [sr] | this for sure is a softphone problem, using the phisical SIP phone, works just fine |
| 20:28.53 | Naikrovek | brettnem: i've heard that before, the freezing thing |
| 20:28.58 | Naikrovek | someone else was in here talking about it at some point |
| 20:29.03 | Naikrovek | latest firmware? |
| 20:29.26 | brettnem | it's not.. I'm in the process of tring to get the latest to give it a shot.. |
| 20:29.55 | FutureWeb | so no one has an idea ? :/ |
| 20:30.01 | brettnem | wanted to see in the meantime if there were any known problem.. weird thing is.. we have plenty of phones that do work too.. and the phones are not rebooting.. just freezing up |
| 20:31.04 | Naikrovek | yeah |
| 20:31.10 | Naikrovek | someone spoke of this a couple weeks back |
| 20:31.17 | Naikrovek | don't remember who it was (maybe it was you) |
| 20:31.33 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:32.08 | FutureWeb | Hello ? :P anyone read my question above please ? this is the last time I`m asking if no one replied i`ll it as no one knows :// |
| 20:33.09 | [sr] | WIMPy: did you ever saw something like this? |
| 20:33.24 | WIMPy | no |
| 20:33.26 | chuckp | I have a strange issue, we have a Polycom SoundStation2W which is a wireless conference phone. It plugs into an analog line that goes back to an Astribank. When the customer calls a conf. # they have and enter the conference ID, it missreads the DTMF. However if I call into a # to read back the DTMF it works fine, along with other conference #s. If the customer plugs an analog line into the same port the 2W is plugged into, the conf |
| 20:33.40 | Naikrovek | FutureWeb: sorry, don't know any maltese DID providers |
| 20:33.46 | [sr] | WIMPy: never mind..i'll report to the softphone owner... |
| 20:34.10 | brettnem | FutureWeb: I use callcentric too..and when I look for that area, it brings up ireland :/ well crap. :) |
| 20:34.11 | Naikrovek | chuckp: we didn't get all that |
| 20:34.16 | k4tana | <FutureWeb> : this is the last time I`m asking if no one replied i`ll it as no one knows .... I think that everybody is afraid .. |
| 20:34.33 | FutureWeb | lol I have been searching for ages :S |
| 20:34.33 | Naikrovek | IRC support isn't always fast... |
| 20:34.39 | FutureWeb | well sorry then :P |
| 20:34.50 | FutureWeb | but I just got ripped off by some fake site |
| 20:34.54 | Naikrovek | ew |
| 20:34.56 | Naikrovek | that sucks |
| 20:34.56 | FutureWeb | so I kinda gave up trying websites |
| 20:35.04 | FutureWeb | and thought i`ll ask here :/ |
| 20:35.14 | [sr] | WIMPy: ei, got a question for you!! imagine i have now a setup of 2 NT TE ports...with a group os 10 numbers [0-9], and i'm going to route each number to the desired SIP extension... all fine here, but if I want tell the SIP extension to use ALWAYS the number 234377560 as CID, is it possible, how? |
| 20:36.34 | WIMPy | [sr]: Unless you replace it the call will go out with the number configured on the phone or what you configure in sip.conf. |
| 20:37.12 | FutureWeb | anyway if anyone finds anything about Maltese DIDs let me know or somethin in a Private message thanks :) |
| 20:37.12 | wcselby | hates SQL queries |
| 20:37.47 | [sr] | WIMPy: whatever i set, even if it is invalid/inexistent, will go out as CID? |
| 20:37.53 | Naikrovek | ?!? |
| 20:37.57 | Naikrovek | sql is awesoem |
| 20:38.02 | Naikrovek | well not AWESOME |
| 20:38.04 | Naikrovek | but good |
| 20:38.05 | Naikrovek | i like it |
| 20:38.24 | FutureWeb | or to make the deal hotter :P I`ll give you a FREE shell for Asterisk in europe, for that person who gets me a Maltese DID :D |
| 20:38.34 | wcselby | Naikrovek - it's awesoem? |
| 20:38.35 | WIMPy | [sr]: Yes, but if it's not a number of that line, your telco will replace it with your default number. |
| 20:38.37 | wcselby | :p |
| 20:38.56 | wcselby | SQL is nice, you can do all kinds of shit with it. But I only use it once every few months, so I hate designing queries for it |
| 20:38.59 | WIMPy | [sr]: ... unless you've got CLIP no screening. |
| 20:39.13 | Naikrovek | 4th normal form and java are my friends |
| 20:39.17 | wcselby | well, I use it all the time, but I only have to write custom queries every few months, I guess I should say |
| 20:39.22 | Naikrovek | not the fastest combo in the world, proper ORM is slow |
| 20:39.24 | Naikrovek | but it's nice |
| 20:39.28 | [sr] | WIMPy: hum interesting...so since it's ISDN, i can use any of the 10 numbers... and anyone will be valid |
| 20:39.49 | WIMPy | [sr]: yes. |
| 20:39.51 | [sr] | WIMPy: still got to learn about CLIP.. don't know what is it yet |
| 20:40.13 | WIMPy | Calling Line Identification Presentation |
| 20:40.47 | [sr] | going to read :) |
| 20:40.54 | Naikrovek | oops. s/4th normal form/5th normal form/ |
| 20:41.09 | voxter | If i want asterisk to forward to a new host and not use Local/, which 3XX sip message should i use? |
| 20:41.21 | voxter | 302 Temp Moved seems to always use Local/ even if i have specified a new sip host/ip in the 302 |
| 20:41.22 | voxter | 301? |
| 20:42.09 | Naikrovek | why are you playing with response codes |
| 20:42.43 | Naikrovek | why doesn't exten => 123,1,Dial(SIP/trunk_to_new_host/${EXTEN}) work |
| 20:43.13 | voxter | Maybe i'll rephrase my question, i have a host i am dialing that is sending me a 302 redirect with a new host on it, but asterisk forces 302's to always try with the Local/ context, and ignores the new host/ip in the 302 from the far end. |
| 20:43.33 | voxter | either, how do i make asterisk not handle 302's that way, or what response code should i have the far end send instead of 301 to not have asterisk behave that way? |
| 20:43.48 | voxter | I WANT it to forward to a new ip. the "302" is expected. |
| 20:43.53 | Naikrovek | hmm. |
| 20:43.59 | Naikrovek | not sure. |
| 20:44.25 | *** join/#asterisk markfeatherston_ (~mark.feat@71.4.41.98.ptr.us.xo.net) |
| 20:48.35 | ldiamond | Anyone know good SIP providers that support SMS? ideally based in canada, or at least providing canadian dids and canadian servers. |
| 20:48.53 | FutureWeb | guys, is this >> Linksys RTP300 good for connecting an analogue telephone line to Asterisk ? :P |
| 20:49.21 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:52.16 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 20:52.23 | *** join/#asterisk zyphlar (~z@wsip-70-182-59-230.ph.ph.cox.net) |
| 20:52.27 | zyphlar | heya |
| 20:53.11 | zyphlar | what's the right way to change an extension number in asterisk? |
| 20:53.23 | zyphlar | i.e. same phone, voicemail, routes, etc-- just different internal number |
| 20:57.30 | brettnem | zyphlar: really depends on how it's done.. not something that can be answered generically |
| 20:58.36 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 20:58.59 | zyphlar | so i guess generally... what is the link between the exten => and the voicemail system? |
| 20:59.37 | zyphlar | and/or can i change the voicemail box's extension without clearing all its message |
| 20:59.48 | FutureWeb | is this good for asterisk LINKSYS PAP2 NA SIP VOIP Phone Adapter 2 Port ? |
| 20:59.49 | FutureWeb | lol |
| 21:00.24 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:03.00 | markfeatherston_ | FutureWeb: You're probably best off searching for reviews of the phone model. I'm told linksys works decently, but voice quality isn't that great. I use aastra phones and they work great. Polycom phones are supposed to be some of the best voip phones. Just look for what meets your price point at acceptable quality. I know of a call center that uses linksys phones, though I don't know the model. |
| 21:05.05 | FutureWeb | its not a phone :P its a router, I want to know if I can use it to connect my phone line to asterisk |
| 21:05.51 | *** join/#asterisk ACK-NAK (~Miranda@kfife.com) |
| 21:06.36 | *** join/#asterisk TheSov (~al@dsl081-136-068.chi1.dsl.speakeasy.net) |
| 21:06.47 | FutureWeb | http://www.voiplink.com/Linksys_PAP2_NA_p/linksys-pap2-na.htm |
| 21:07.07 | markfeatherston_ | FutureWeb: http://www.voip-info.org/wiki/view/Linksys < look for pap2 on that page |
| 21:08.53 | FutureWeb | what I dont get is, if it is good to connect a PHONE line to it, I dont want it to connect phones to it I want to connect a phone line so people can call ont he land lien number and the call goes to asterisk |
| 21:11.08 | Godfather_ | FutureWeb, you need FXO, not FXS |
| 21:11.25 | Naikrovek | yes |
| 21:11.30 | Godfather_ | FutureWeb, probably you are searching for a spa3102 |
| 21:11.42 | Godfather_ | it has 1 FXO and 1 FXS |
| 21:11.44 | Naikrovek | some devices are expected to provide dial tone (FSX) some are expected to receive dial tone (FXO) |
| 21:11.47 | *** join/#asterisk sarah_p (~sarah_p@71.117.223.226) |
| 21:11.55 | zyphlar | FXS |
| 21:11.58 | Naikrovek | s/FSX/FXS |
| 21:12.06 | Naikrovek | yes |
| 21:12.33 | Godfather_ | FutureWeb, but quality isnt good. |
| 21:12.34 | FutureWeb | Linksys RTP300-NA VoIP < is that FXO or FXS ? :P |
| 21:13.17 | Godfather_ | FutureWeb, you can see it in the description, do it yourself |
| 21:13.26 | zyphlar | FXS = "provides dialtone to a Station (a phone)", FXO = "gets dialtone from an Office (the phone company) |
| 21:14.38 | ACK-NAK | Think of it as fxO is a ph'O'ne and fxS is 'S'ervice |
| 21:14.52 | zyphlar | that works too |
| 21:15.09 | [sr] | WIMPy: how can the TELCO's send invalid CID's?, like alpha chars, and unexistent numbers? 'cause they are the TELCO and have control over that? normaly this invalid numbers and alpha CID's happem when i get a SMS on mobil |
| 21:15.12 | Qwell | O, O, O. not 0. |
| 21:15.56 | zyphlar | it's just confusing cuz the S stands for station and the O stands for Office... but FXS is not what is in a phone and FXO is not in the phone company's office |
| 21:16.12 | Qwell | zyphlar: it absolutely is. |
| 21:16.17 | zyphlar | FXS provides for stations... FXO connects back to the central office |
| 21:16.27 | Qwell | Remember that FXS devices provide FXO signalling. |
| 21:16.36 | Qwell | rather, use |
| 21:16.41 | zyphlar | haha even worse |
| 21:16.51 | zyphlar | lemme guess, FXO uses FXS? |
| 21:17.01 | Qwell | that would be the obvious conclusion |
| 21:17.06 | zyphlar | delicious |
| 21:17.06 | WIMPy | [sr]: Phone "numbers" are not restricted to digits. Neither in SIP not in ISDN. Non numeric ones are just not used for obvious reasons. |
| 21:17.34 | ACK-NAK | When a call comes on a TDM400 FXO port, The call is routed to 's' in [context] but I want to go to, 1234 in [context]. I can't find the way to do this. Can anyone direct me? |
| 21:17.40 | Qwell | a *phone* is actually an FXO. |
| 21:18.03 | ACK-NAK | a phOne |
| 21:18.13 | ACK-NAK | ph'O'ne |
| 21:18.28 | zyphlar | ACK-NAK: i'm not getting it... could you emphasize the O a bit more? |
| 21:18.36 | Qwell | You just have to think about it the correct way, and it makes perfect sense. |
| 21:18.39 | Naikrovek | nO kidding |
| 21:18.47 | Naikrovek | :P |
| 21:18.55 | Godfather_ | ACK-NAK, you can configure it in dahdi_channels.conf |
| 21:19.08 | Godfather_ | dahdi-channels.conf |
| 21:19.15 | ACK-NAK | Godfather_: of course, but which parameter? |
| 21:19.22 | WIMPy | [sr]: With SMS it's a slightly different story, as they are usually delivered via IP oder things like UCP or TAP, where little filtering is done. |
| 21:19.39 | ACK-NAK | zyphlar: lol |
| 21:20.08 | [sr] | WIMPy: so i almost can spoof an SMS message sent from asterisk... am i correct? |
| 21:20.56 | ACK-NAK | _ |
| 21:20.56 | ACK-NAK | ph\_/ne |
| 21:21.20 | WIMPy | [sr]: Depend on what way you send it an what your provider allows. But there are goood chances, you can send whatever you like, incl. names as OAdC. |
| 21:21.30 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:21.32 | ACK-NAK | Darn it. So much for my attempt at ascii art to over, over, OVER emphasize the O in phOne |
| 21:21.51 | [sr] | WIMPy: cool...i'll do that test!! |
| 21:22.07 | Godfather_ | ACK-NAK, i mistaked, sorry |
| 21:22.33 | zyphlar | ph*~,~*~,ooOOOooo~*~,~*~,ne |
| 21:22.42 | Godfather_ | ACK-NAK, maybe you can do Goto in extensions.conf, but no idea how to redirect it from dahdi-channels.conf to a specify extension |
| 21:22.49 | WIMPy | [sr]: But it might happen, that you kill your provider. Happened to me once. Bad luck I only did what they stated in their documentation. |
| 21:23.31 | [sr] | WIMPy: what do you mean by kill, mobil service stop working?:) |
| 21:24.17 | ACK-NAK | I could send the call to [othercontext], and GOTO the call to 1234 in [context] but that's a flaming pile of kludge. |
| 21:24.39 | ACK-NAK | zyphlar: You're more artistic than I |
| 21:25.49 | ACK-NAK | zyphlar: This is what I had tried, but the IRC parser swallowed it. http://pastebin.com/JYHvfZjn |
| 21:26.38 | WIMPy | [sr]: The messaging service of that provider went down, yes. |
| 21:26.50 | [sr] | WIMPy: that is very nice :P |
| 21:26.54 | *** part/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 21:27.03 | ACK-NAK | Anyone?? For 25 points: call comes on a DAHDI FXO port, The call is normally routed to 's' in [context] but we want to go to, 1234 in [context]. Is this Possible? It's too obvious NOT to be, but I can't find it. |
| 21:27.07 | [sr] | WIMPy: i'll try to send as CID stuff like \c'||33 |
| 21:27.12 | WIMPy | yes, they were not very happy about that. |
| 21:27.30 | [sr] | WIMPy: eheh... i imagine... |
| 21:28.00 | WIMPy | But it was all within the secifications they handed out. Just their software wan;t up to what was written there. |
| 21:28.42 | [sr] | WIMPy: when i try here, i'll let you know the results with the 3 mobil operators here |
| 21:28.46 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 21:28.54 | markfeatherston_ | Can someone help me with the StartMonitor/StopMonitor events? It's giving me a channel when it's triggered, but when I pass the channel back to getvar, it seems like it's an invalid channel. Does anyone know what might cause this, or how I can verify that it is a valid channel name? |
| 21:29.07 | Katty | hey guys |
| 21:29.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:29.16 | markfeatherston_ | Howdy |
| 21:29.22 | Katty | http://pastebin.com/xFjT3n9R <- anyone have some insight on this warning? |
| 21:29.52 | Katty | a remote user logs in, initates a call, but we don't have audio. i doublechecked the firewall on my side that it's allowing traffic from anyone on port 5060 and 10k-20k |
| 21:30.00 | Katty | not sure what else to check |
| 21:30.39 | markfeatherston_ | Katty: Using sonicwall by chance? |
| 21:30.42 | Katty | no |
| 21:31.01 | Katty | any chance you know exactly waht that warning message means? |
| 21:31.17 | Katty | i get it's trying to retransmit..something... |
| 21:31.23 | Katty | and it hit the maximum ammount of retries |
| 21:31.43 | markfeatherston_ | sip reinvite i'm guessing |
| 21:32.46 | *** join/#asterisk ruied (~ruied@bl6-129-85.dsl.telepac.pt) |
| 21:33.52 | markfeatherston_ | could the endpoint be behind anything that would interfere with the traffic? |
| 21:34.03 | Katty | possibly |
| 21:34.14 | Katty | but of course i don't have any good way to tell. i can't just drive over there |
| 21:35.15 | WIMPy | Katty: The peer seems to have disappeared. |
| 21:36.17 | Katty | disappeared |
| 21:36.29 | Katty | so it can't find the ip it's on? |
| 21:36.33 | Katty | correct? |
| 21:36.43 | WIMPy | Retransmissions occour when no response is received. |
| 21:36.49 | Katty | hrmmmmmmm |
| 21:36.51 | Katty | interesting |
| 21:36.51 | Katty | k |
| 21:36.55 | WIMPy | I can't tell you the reason. |
| 21:37.25 | Katty | well it works inside a vpn connection |
| 21:37.29 | Katty | so i'm convinced it's something to do with nat |
| 21:37.42 | Chainsaw | Katty: The only time I've seen that message so far is for e164.org verification calls. |
| 21:37.43 | Katty | i don't think it's on our side, simply because port 5060 and rtp ports are opened up |
| 21:37.52 | Chainsaw | Katty: (They purposely disconnect the SIP session very, very early) |
| 21:37.55 | Katty | would someone be willing to run a quick test with me? |
| 21:38.00 | Chainsaw | Katty: Sure, what do you need? |
| 21:38.58 | WIMPy | Well the call was accepted, So there must have been responses. But then they seem to stop. |
| 21:39.29 | WIMPy | Maybe be a reinvite issue? |
| 21:39.34 | WIMPy | Time for sip debug. |
| 21:46.59 | *** join/#asterisk guilhermebr (~Guilherme@189.63.65.56) |
| 21:50.40 | *** join/#asterisk mrchrisadams (~Adium@CPE-124-179-78-253.lns5.cht.bigpond.net.au) |
| 21:59.53 | *** join/#asterisk mrchrisadams (~Adium@CPE-124-179-78-253.lns5.cht.bigpond.net.au) |
| 22:03.00 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 22:16.28 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
| 22:17.22 | *** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 22:17.49 | nny | anyone know of a way to remove the -- from grep -A NUM when it finds concurrent matches? |
| 22:19.33 | Qwell | `grep -v "--"` ? |
| 22:20.08 | nny | invert, i'll try that thanks |
| 22:21.26 | ACK-NAK | Qwell: you're a guy who knows stuff. Is there a way to send an inbound call to a specific FXO port to a specific extension without giving it its own private context? |
| 22:21.54 | ACK-NAK | calls go to s. I want them to go to 1234 for example |
| 22:24.36 | xheliox | how about something like.. |
| 22:25.01 | xheliox | GotoIf($["${CHANNEL}" = "DAHDI/1-1"]?label1) |
| 22:25.40 | ACK-NAK | xheliox: that's very clever. |
| 22:25.40 | xheliox | s,n(label1),Dial(SIP/whatever) |
| 22:26.12 | xheliox | Asterisk variables are your friend. Use the variables, Luke. |
| 22:26.12 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 22:27.13 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 22:27.17 | ACK-NAK | It just seems messier than it shoudl be. Seems like there should be a context=1234@default |
| 22:27.55 | xheliox | If you come up with a better way, let me know, because that's all I'm coming up with off the top of my head. I'd be curious. |
| 22:28.32 | ACK-NAK | xheliox: Thanks. I'll do that! |
| 22:28.44 | xheliox | no problem. :) |
| 22:32.56 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:40.28 | viq | Kinda Off Topic. I want to get myself one of those PSTN/SIP DECT phones, and can't make up my mind between Siemens C470/C475 and S685, any advice? |
| 22:41.38 | WIMPy | viq I tried one of the Telekom branded Siemens things an it's pretty shitty. |
| 22:41.46 | *** join/#asterisk p0seidon (~nnscript@89.181.55.57) |
| 22:41.54 | p0seidon | HI |
| 22:42.23 | viq | Hm. |
| 22:42.26 | WIMPy | If one handset goes missing (e.g. low battery) it won't ring the others any more either until it's rebooted. |
| 22:42.36 | viq | o.O |
| 22:42.50 | viq | What model was that? |
| 22:42.51 | p0seidon | [TK]D-Fender: problem solved , it was my motherboard the x100p is working. Its detected , still cant get it to pick up the calls but dahdi detects it tomorrow i'll get it working with the phone line. |
| 22:43.18 | *** join/#asterisk kotp (~vgoff@96.2.187.67) |
| 22:43.18 | WIMPy | Not sure. The branded name is Sinus 501V. |
| 22:43.21 | viq | Any other advice on something that would have PSTN and SIP and wireless handsets? |
| 22:44.01 | p0seidon | viq i'm using siemens a580ip |
| 22:44.02 | WIMPy | I'd go for an ordinary dect phone behind Asterisk. |
| 22:44.26 | p0seidon | entry level and it works greate with Asterisk |
| 22:44.31 | p0seidon | great* |
| 22:44.34 | xheliox | p0seidon: what was the problem? |
| 22:44.40 | WIMPy | p0seidon: Can you access the webinterface without disabling ecn? |
| 22:44.45 | p0seidon | xheliox: hi |
| 22:45.40 | viq | hm, that sinus thing does look like C4xx |
| 22:46.07 | p0seidon | WIMPy, disabling what? |
| 22:46.16 | viq | p0seidon: I want to use it both with PSTN and a couple of external SIP accounts, and be able to easily change between those |
| 22:46.21 | WIMPy | TCP ECN |
| 22:46.53 | p0seidon | xheliox, the board just wont boot with this card...not compatible. in other board I have its working ok. |
| 22:47.36 | p0seidon | WIMPy, i'm sorry i dont know that protocol ecn |
| 22:47.43 | p0seidon | let me check |
| 22:48.01 | WIMPy | Explicit Congestion Notification |
| 22:49.24 | p0seidon | i can access webinterface of siemens without having to disable anything on my network |
| 22:53.40 | WIMPy | Looks like it is a S685 IP. |
| 22:54.10 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:54.11 | viq | yeah, just found same info |
| 22:54.48 | viq | grumble, can't say that made my choice easier :P |
| 22:54.49 | p0seidon | viq, i use a580 with several sip accounts |
| 22:55.08 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 22:55.12 | viq | p0seidon: how does it work, how do you like it? Any features you lack and would like to see? |
| 22:55.35 | p0seidon | i just use it for home pourpose and for that its really ok |
| 22:55.41 | p0seidon | good range, good sound quality |
| 22:55.52 | viq | one handset or more? |
| 22:55.57 | p0seidon | u can have 6 |
| 22:56.14 | p0seidon | just have 1 |
| 22:56.34 | viq | I am somewhat afraid of what WIMPy said |
| 22:56.52 | p0seidon | my terminal is a58h |
| 22:56.56 | viq | As I want to have 2 handsets |
| 22:57.08 | p0seidon | there are reviews if you google it |
| 22:57.26 | p0seidon | there is a kit with 2 handsets |
| 22:57.30 | WIMPy | I have two handsets on the same SIP account. |
| 22:57.34 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 22:57.37 | viq | hm, yeah, will have to look some more... |
| 22:57.37 | ManxPower | ~answers |
| 22:57.38 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 22:58.10 | p0seidon | you can have them on separeted sip accounts |
| 22:58.24 | p0seidon | like extension 100 and extension 200 |
| 22:58.30 | WIMPy | I have a 2nd sip account configured already that is intended for a third handset, but that's not connected, yet. |
| 23:01.50 | viq | OK, thank you guys for the answers and hints, now time to head to sleep, I'll look at it tomorrow. |
| 23:02.37 | *** join/#asterisk mrchrisadams (~Adium@CPE-143-238-67-118.lns6.cht.bigpond.net.au) |
| 23:03.06 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:09.23 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:16.17 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:22.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:27.45 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:28.49 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 23:37.16 | *** join/#asterisk guilhermebr (~Guilherme@189.63.65.56) |
| 23:37.46 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 23:42.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:55.40 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 23:57.24 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |