| 00:00.39 | SeRi | jpsharp: lol |
| 00:01.00 | SeRi | jpsharp: my internet is fixed. yay! :) |
| 00:01.31 | jpsharp | Found comcast and applied a 2x4 directly to their head? |
| 00:02.59 | SeRi | rofl |
| 00:03.01 | SeRi | almost |
| 00:06.33 | dijib | why cant you connect? |
| 00:06.48 | SeRi | dijib: your openfire server |
| 00:07.01 | dijib | you cant connect to it? |
| 00:07.07 | dijib | with a client ? |
| 00:07.08 | *** join/#asterisk cisco (~cisco@201.20.110.36) |
| 00:07.15 | dijib | im connected to it right now |
| 00:07.16 | SeRi | send me the options you have enable in your client |
| 00:07.24 | dijib | defaults |
| 00:07.29 | dijib | 5222 is the port |
| 00:07.38 | SeRi | send me the options you have enabled in your cleint |
| 00:08.36 | SeRi | hey I cant talk |
| 00:08.41 | SeRi | it seems to ber blocking me |
| 00:08.48 | SeRi | thats why I am pressing keys man |
| 00:09.01 | SeRi | dijib: ^^ |
| 00:09.57 | dijib | try #1 |
| 00:10.15 | SeRi | ok |
| 00:31.26 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:31.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:32.12 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 00:37.53 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:39.25 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 01:17.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 01:19.47 | *** join/#asterisk deo (~deo@112.198.79.19) |
| 01:22.18 | *** join/#asterisk cisco (~cisco@201.20.110.36) |
| 01:33.11 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 01:39.29 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 01:52.34 | *** join/#asterisk hebber (~hebber@125.25.206.20) |
| 01:54.34 | hebber | Hi, I need some assist on how to include queue.conf as static realtime into a mysql database. Most conf files are easy to move, but the queue definition doesn't really have a [var_name] and a [val_name] syntax |
| 01:57.41 | hebber | I have found some old instructions to create a queue table with a similar structure as sippeers - is that the correct instructions? |
| 01:57.53 | *** part/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 01:57.55 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 01:57.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:11.50 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 02:12.20 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 02:17.04 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 02:21.57 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-byoxiibruexjyvbh) |
| 02:21.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:32.06 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 02:32.15 | saint_ | hi all |
| 02:41.06 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 02:42.37 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 02:54.17 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.107) |
| 02:57.30 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 02:57.30 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:05.31 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:15.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-spdvixugdnbizvgh) |
| 03:15.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:21.48 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 03:29.37 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 03:35.57 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 03:36.04 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 03:36.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:44.38 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 03:44.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:45.26 | *** join/#asterisk jsjc (~Adium@2.136.89.82) |
| 03:55.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 04:07.02 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 04:36.05 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 05:10.19 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-qgawbuyvjhoytckq) |
| 05:15.30 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 05:27.28 | *** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158) |
| 05:35.02 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 05:47.13 | *** join/#asterisk cyborg-one (~cyborg-on@188.115.131.197) |
| 05:52.15 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 05:52.20 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 06:12.32 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 06:14.02 | Russ | hmmm...If I connect to my did via IAX, is there a good way to make some calls use a different codec? |
| 06:32.07 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.107) |
| 06:51.00 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 06:57.15 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 07:01.29 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:04.28 | ectospasm | Russ: what do you mean? |
| 07:05.14 | ectospasm | ...if the peer is set in Asterisk to use a list of codecs, the only way to enforce a particular codec on any given call is to have the peer only request the codec it wants. |
| 07:05.32 | ectospasm | there's no way to switch codecs per CID, for instance. |
| 07:07.43 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 07:41.06 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:42.33 | bulkorok | hi |
| 07:45.26 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:50.14 | ectospasm | Russ: another way of saying that is there's no way to change the codec with dialplan, that I know of anyway. |
| 07:51.30 | Russ | I want to try to give faxes a fighting chance |
| 07:51.52 | Russ | so the codec would be set for specific lines |
| 07:51.57 | Russ | (outgoing only) |
| 07:52.08 | ectospasm | what do you mean "lines?" |
| 07:52.24 | ectospasm | there ain't no such thing as "lines" in VoIP. |
| 07:53.02 | ectospasm | you could have two different peers pointing at the same place, one with the codecs you want |
| 07:53.22 | ectospasm | and besides, if you're using T.38 the fax stack should ignore whatever you've set for codec |
| 07:55.29 | v0lZy | could u configure two accounts on 1 phone |
| 07:55.30 | kaldemar | chan_sip lets you set codecs with channel variables, but chan_iax does not have means to do that in dialplan afaik. |
| 07:56.07 | v0lZy | then switch between them at will... you could maybe do somethign like that for faxes. |
| 08:00.16 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:01.00 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:08.51 | *** join/#asterisk elico (~Thunderbi@bzq-109-64-187-2.red.bezeqint.net) |
| 08:19.55 | *** join/#asterisk areski (~areski@80.174.255.87.dyn.user.ono.com) |
| 08:20.24 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:f816:1f36:fa84:8d72) |
| 08:22.25 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 08:27.11 | *** join/#asterisk wonderworld (~w@dsdf-4db5ee2c.pool.mediaWays.net) |
| 08:29.38 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:32.37 | *** join/#asterisk keith_hanson (~keith@c-50-130-109-232.hsd1.la.comcast.net) |
| 08:33.34 | keith_hanson | Hi all :) Does anyone have any tips on port forwarding? I have asterisk running locally successfully, ports forwarded via my router (udp 5060, udp 10000 - 20000), but can't seem to use my softphone to connect |
| 08:35.33 | keith_hanson | I also set my externip and localnet in sip.conf |
| 08:35.51 | bulkorok | what's youor CLI telling you?! |
| 08:36.02 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:36.07 | bombev | hi all |
| 08:36.34 | keith_hanson | bulkorok: nothing, unfortunately. Doesn't seem to be punching through my router unfortunately. |
| 08:37.12 | keith_hanson | bulkorok: but I do see successful registrations via localhost |
| 08:37.18 | bulkorok | core set verbose 5 |
| 08:37.22 | bulkorok | sip set debug on |
| 08:37.24 | v0lZy | keith_hanson: whats the problem u are having? u cant register the phone to the pbx? |
| 08:37.42 | v0lZy | or you cant dial out/in ? |
| 08:37.47 | keith_hanson | bulkorok: will try that. |
| 08:37.59 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 08:38.04 | bulkorok | sip debug shows you all sip messages that come to your asterisk |
| 08:38.21 | keith_hanson | v0lZy: can't register. I'm using SFLphone in Linux, and it can't seem to go from my machine, to external IP, and back |
| 08:38.23 | bulkorok | if nothing is to see, you have to check your FW |
| 08:39.52 | v0lZy | keith_hanson: If i understand correctly, your phone client is separated from the server via a router? |
| 08:40.41 | keith_hanson | v0lZy: well, I'm running asterisk locally on my computer, phone client is local as well. I'm point my phone client to my public IP, hoping to get through my router and back to my local server |
| 08:41.13 | bulkorok | ew... dubble NAT... |
| 08:41.20 | v0lZy | thats ugly |
| 08:41.37 | keith_hanson | v0lZy: haha, well, I'd love to not have to setup an external asterisk server while I'm building my app |
| 08:42.01 | keith_hanson | v0lZy: also, pretty new to asterisk in general. Is there some standard practice for testing this sort of thing? |
| 08:42.34 | v0lZy | why not use a virtual machine... |
| 08:42.38 | keith_hanson | bulkorok: yeah. So no sip messages coming through while the client is trying to connect |
| 08:42.56 | bulkorok | then it's your routing/NETing/firewalling... |
| 08:43.06 | bulkorok | s/NETing/NATing |
| 08:43.20 | keith_hanson | v0lZy: well, I definitely need it accessible from the net. I'm trying to integrate Phono.js (http://phono.com/), which connects from their servers over sip |
| 08:43.25 | keith_hanson | bulkorok: seems that way, yeah. |
| 08:43.39 | kaldemar | keith_hanson: pointing your local client to your public address won't necessarily work. try the client from outside of your network to really test it. |
| 08:44.09 | bulkorok | nat out and nat in with port forwarding seems to be very confused... |
| 08:44.26 | keith_hanson | kaldemar: ah, good to know. I'll give it a shot. worst case I'll spin up an EC2 box or something. |
| 08:44.47 | keith_hanson | thanks for the help all :) |
| 08:44.49 | kaldemar | and that's not a real world usage case anyway. a local client would be connecting locally, not via your public ip. |
| 08:44.56 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.147) |
| 08:45.07 | keith_hanson | kaldemar: true true. I just wanted to duck tape it together for testing purposes. |
| 08:46.01 | v0lZy | id just run a virtual machine for asterisk keith_hanson |
| 08:46.53 | keith_hanson | v0lZy: or that :D |
| 08:47.36 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:47.53 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.158) |
| 08:48.02 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:48.06 | v0lZy | u could also just use the phone in the virtual machine |
| 08:48.13 | v0lZy | whichever u find easier :D |
| 08:48.27 | v0lZy | id suggest asterisk in vm though, less hassle with sound and all that stuff :D |
| 08:49.07 | keith_hanson | cool. I'm going to throw together a simple demo with Phono.js and see if it can connect to the local box for testing. If that fails, then I'll give a VM a shot, and if *that* fails, I'll just startup an ec2 |
| 08:51.42 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 08:52.41 | kaldemar | less hassle with sound in a virtual machine? there are many opinions that counter that statement. |
| 08:53.27 | kaldemar | putting the asterisk in a local virtual machine just makes the setup more complex. |
| 08:56.54 | mcolombo | hi guys |
| 08:58.41 | mcolombo | when my asterisk receive an UPDATE Messages, he answers with a 501 NOT IMPLEMENTED |
| 08:58.56 | mcolombo | it's possible to answer with a 200 OK Messages? |
| 08:59.19 | *** join/#asterisk Nickinator (~Nickinato@220-244-173-9.static.tpgi.com.au) |
| 09:01.50 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 09:02.17 | *** part/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 09:04.58 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 09:06.26 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 09:10.25 | kaldemar | mcolombo: when is the update sent and which version of asterisk are you using? |
| 09:12.56 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 09:16.25 | mcolombo | Kaldemar : thanks for reply, the update are sent during the call , i'm using asterisk 1.8 with oej patch for prack |
| 09:16.42 | *** join/#asterisk pbxman (c335d91f@gateway/web/freenode/ip.195.53.217.31) |
| 09:18.44 | mcolombo | here the sip debug : http://pastebin.com/tVE1NDtF |
| 09:22.54 | weinerk | Hi. Please help figure out why I see a [h@macro-dialout-trunk:1] Macro "hangupcall|" |
| 09:23.08 | weinerk | ast1.4 - voip call-in -> AGI1 makes a dial#1 to -> local/ext1 AGI2 -> makes a dial#2 to -> Local/18882223333 which dials over sip trunk -> callee answers. |
| 09:23.22 | weinerk | Shortly after all legs are answered - I see: |
| 09:23.22 | weinerk | -- Executing [h@macro-dialout-trunk:1] Macro("Local/18882223333@from-internal-3ad0,2", "hangupcall|") in new stack |
| 09:23.34 | weinerk | The call continues as normal until hangup by either side. But why is there a macro-hangupcall ??? |
| 09:24.16 | kaldemar | weinerk: the local channel is dropped from the path because you don't have /n configured for it. |
| 09:28.45 | weinerk | kaldemar: 1) thanks for helping. 2) please explain /n configured? do you mean dial() option ? |
| 09:30.39 | kaldemar | when you dial a local channel, you have something along Dial(Local/exten@context). if you use Dial(Local/exten@context/n), the local channel remains in the path when the channels are bridged. without the /n, the local channel is hung up and the "real" channels are bridged directly. |
| 09:30.44 | jacekowski | i've got digium D40 phones and when i press 9 on them when handset is up i get Failed straight away |
| 09:33.21 | weinerk | kaldemar: 1) Thanks!!!! 2) I will research and try it thanks!!!!!!!! |
| 09:35.34 | kaldemar | weinerk: note that hanging up the local channel is not a problem unless your dialplan expects to do something with it. probably no need to be concerned about that. |
| 09:45.32 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 09:45.38 | weinerk | kaldemar: my problem is that - in the middle of running a macro on a channel - this hangup macro is called and clobers variables. |
| 09:48.52 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 09:49.22 | *** join/#asterisk Sconk2 (~krh@srv.jtw.dk) |
| 09:49.42 | mcolombo | kaldemar : but normally, asterisk reply with 200 ok? |
| 09:50.47 | Sconk2 | Hi I'm trying to make a php agi script to add and remove members from a que budt I'm unable to get the add and remove to work.. are there someone that can point me the right direction ? |
| 09:53.04 | kaldemar | mcolombo: asterisk handles UPDATE methods differently in different situations. updating session properties during a call is not supported, hence the 501. |
| 10:00.54 | mcolombo | kaldemar: ok, thanks |
| 10:01.51 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat27.beelinegprs.ru) |
| 10:03.57 | *** part/#asterisk deo (~deo@112.198.79.19) |
| 10:05.41 | *** join/#asterisk alsuren (~dlaban@80.169.133.251) |
| 10:09.39 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 10:10.14 | jacekowski | is there a way to update callerid on attended transfer? |
| 10:16.22 | con3x | jacelowski: There is a CallerId application you can use in your dialplan |
| 10:16.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:17.34 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-213-033.pools.arcor-ip.net) |
| 10:20.49 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.9.246) |
| 10:23.14 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 10:23.33 | jacekowski | con3x: but that's when calling |
| 10:23.42 | jacekowski | on transfer i get call from extension |
| 10:23.55 | jacekowski | and then if transfer is finished |
| 10:24.37 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 10:30.53 | *** join/#asterisk janmate (~janmate@188-112-68-156.3pp.slovanet.sk) |
| 10:31.18 | janmate | hi everybody, please is there any way to use SIP templates (from sip.conf) in LDAP? … the goal is to create minimal LDAP peer accounts and set the global shared settings in template (in sip.conf) |
| 10:32.13 | weinerk | kaldemar: thanks so much for helping! Can I send you a little thankyou like paypal or bitcoin etc? |
| 10:36.36 | kaldemar | weinerk: you're welcome. |
| 10:39.49 | v0lZy | kaldemar, I can include another file in extensions.conf with #include right? |
| 10:39.58 | kaldemar | v0lZy: yes. |
| 10:40.15 | *** join/#asterisk chris-NB (~chris@fw.commpany.at) |
| 10:40.31 | *** join/#asterisk schmidts (~schmidts@fw.commpany.at) |
| 10:40.33 | schmidts | Hi Folks |
| 10:42.27 | v0lZy | thanks |
| 10:42.46 | Ice_Strike | How to uninstall Asterisk? |
| 10:47.08 | kaldemar | Ice_Strike: depends on how you installed. |
| 10:49.00 | Ice_Strike | kaldemar compiled from source |
| 10:50.21 | kaldemar | the makefile has an uninstall target, so "make uninstall". that will not remove your configuration files. |
| 10:50.39 | v0lZy | kaldemar: So using include, I can do something like this now: Keep a separate file with a list of internal extensions and hosts's of my asterisk boxes. I can then in each box configure an extension 0XX to point to the context under which that file is included. I hookup the asterisk boxes together via VPN and maintain a single file I rsync over to each box and run through a script to remove all lines which contain the servers hostname |
| 10:52.04 | v0lZy | everytime i update the file, i rsync it and reload the extensions.conf on all boxes, everyone should be up to date and my different locations can call eachothers internal phone numbers without using a sip provider |
| 10:52.51 | v0lZy | neat :D |
| 10:55.05 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:57.19 | kaldemar | v0lZy: or you could just make extensions that send the calls to the appropriate boxes. or use DUNDi, which is a protocol to do number queries for you. |
| 10:58.02 | v0lZy | yeah but id then have to do that on each box. |
| 10:58.57 | bulkorok | v0lZy: what about realtime dialplan with odbc-database!? |
| 10:59.16 | bulkorok | *just a hint... gone for some minutes...* |
| 10:59.35 | v0lZy | I guess that would work with a lot of extensions |
| 11:00.31 | v0lZy | but it just seems like yet another technology to maintain... |
| 11:00.43 | v0lZy | too big for my setup |
| 11:14.13 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 11:14.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 11:14.39 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:15.43 | weinerk | Hi. Please help with AGI + dial() + G option |
| 11:15.47 | weinerk | ast1.4 - when I try this: $agi->exec('Dial', "SIP/extension||G(mycontext^2222^111)"); |
| 11:15.47 | weinerk | Dial returns right away and calling side progresses to hangup. |
| 11:15.56 | weinerk | Without G option call is bridged just fine between caller and callee. |
| 11:25.04 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.158) |
| 11:37.45 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 11:47.00 | *** join/#asterisk slash213 (slash213@176.57.74.111) |
| 11:47.15 | slash213 | hi guys |
| 11:47.28 | slash213 | quick question: what are the differences between te420 and te420p? |
| 11:47.34 | slash213 | what does "p" stand for? |
| 11:48.02 | slash213 | my colleagues bought two 420p and one 420 card, and i want to know what we're dealing with |
| 11:50.05 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.246) |
| 11:55.15 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 11:56.40 | *** join/#asterisk italorossi (~italoross@189.124.196.68) |
| 12:05.08 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:08.13 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 12:08.49 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 12:16.31 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 12:17.39 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:17.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:19.16 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 12:19.44 | *** join/#asterisk TimeRider (~steve@81.136.216.215) |
| 12:31.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:34.09 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:38.43 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 12:38.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:38.47 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 12:50.02 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:51.08 | nunne | Running 11.0.1 is it possible to get the old dialplan behaviour of _X. triggering on 1 digit extensions? Not it has to be at least two digits.. So i'm forced to have both _X and _X. |
| 12:56.26 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 13:00.18 | kaldemar | nunne: X. has never matched 1 digit. |
| 13:00.29 | nunne | kaldemar: it does it 1.4 |
| 13:00.42 | nunne | I think :P |
| 13:00.48 | kaldemar | try again. |
| 13:01.28 | nunne | kaldemar: you know WHY they don't recommend using _.? |
| 13:01.49 | nunne | i actually want EVERYTHING to get "hit" :) |
| 13:02.57 | kaldemar | yes, because it will match all standard extensions like s, h, i and t. |
| 13:04.13 | nunne | Ah, i see. well. that will not be a problem in my case :) thanks |
| 13:09.29 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.171.31) |
| 13:10.02 | *** join/#asterisk felimwhiteley (~quassel@92.251.255.5.threembb.ie) |
| 13:13.26 | wdoekes | mintos: _X! |
| 13:13.33 | wdoekes | s/mintos/nunne/ |
| 13:14.35 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:14.58 | nunne | wdoekes: thanks alot! :) |
| 13:16.14 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:18.11 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 13:24.29 | Sconk2 | I still have problems using Add/Remove QueueMember from a php agi script, can someone point me to some info how that can be done ? |
| 13:26.32 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 13:29.30 | bulkorok | Sconk2: you can use fsockopen to intiate an AMI session and issue the manager commands |
| 13:30.34 | Sconk2 | okay |
| 13:30.50 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 13:30.52 | [TK]D-Fender | bulkorok, that doesn't answer the "problems" part of his request. You just redescribed what he's trying to do. |
| 13:31.24 | [TK]D-Fender | Sconk2, And if you're in AGI... that's all you need. They are DIALPLAN APPS |
| 13:31.39 | mcolombo | hi guys |
| 13:31.46 | mcolombo | <PROTECTED> |
| 13:31.59 | mcolombo | <PROTECTED> |
| 13:32.07 | [TK]D-Fender | Sconk2, If you'd like our assistance you should show us your attempt along with the related code so we can actually possibly point out where the error is |
| 13:32.07 | mcolombo | sorry for duoble post |
| 13:32.17 | mcolombo | it's possible to answer with a 200 OK Messages or asterisk does not support this answer? Thanks |
| 13:33.20 | Sconk2 | [TK]D-Fender, i have no errors right now, just trying to figure out how to do it.. as from what i see the *queuemember function don't working from agi.. |
| 13:33.41 | [TK]D-Fender | Sconk2, Show us where you see this. |
| 13:34.13 | [TK]D-Fender | Sconk2, It is a dialplan app just like any other. It works. If it didn't, then AGI wouldn't work at all for anybody. |
| 13:34.52 | [TK]D-Fender | <mcolombo> it's possible to answer with a 200 OK Messages or asterisk does not support this answer? Thanks <---- 501 NOT IMPLEMENTED |
| 13:40.08 | kaldemar | mcolombo: was my earlier answer not good enough? |
| 13:41.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:44.02 | Sconk2 | [TK]D-Fender, my script is here http://pastebin.com/U928b4Dv if it gives any meaning for you. |
| 13:46.41 | mcolombo | kaldemar: yes your answer is very good, I just seems a little strange that Asterisk does not support this thing |
| 13:48.24 | kaldemar | yes, it's a SIP feature alright, but not supported by asterisk. |
| 13:50.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:55.54 | *** join/#asterisk serafie (~erin@nat/digium/x-wpehhimwzisflgtz) |
| 13:56.59 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-tyfuhnebaikijweg) |
| 13:56.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:01.17 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:05.25 | *** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl) |
| 14:05.41 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:09.26 | [TK]D-Fender | Sconk2, execute_agi('AddQueueMember("110","SIP/'. $agi_callerid .'")'); |
| 14:09.40 | [TK]D-Fender | Sconk2, You do not use brackets with AGI app calls. |
| 14:09.47 | [TK]D-Fender | Sconk2, Or put quotes around everything |
| 14:10.47 | [TK]D-Fender | Sconk2, Also, you need to use EXEC to call a dialplan app. You don't just call them as a direct command |
| 14:10.50 | [TK]D-Fender | ~book |
| 14:10.51 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:10.52 | [TK]D-Fender | ^^^ |
| 14:14.21 | jmetro | that and the wiki |
| 14:19.06 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 14:20.30 | Sconk2 | thanks |
| 14:23.09 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 14:24.00 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:26.24 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.246) |
| 14:28.09 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:29.22 | *** join/#asterisk Ta^3 (~tacvbo@187.191.13.43) |
| 14:33.02 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 14:34.50 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:34.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:35.57 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 14:36.09 | *** join/#asterisk Ta^3 (~tacvbo@187.191.13.43) |
| 14:37.38 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 14:37.38 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 14:38.14 | as001 | Hi I have asterisk with one 1 fxo and 1 fxs port. When someone calls me I can see incomming callerid on asterisk CLI but later in dialplan when I Dial FXS device (DAHDI/1) I can just see "Incomming" on display instead of cellerid. When I plug fxs device directly in wall I can see callerid. Why I can't pass callerid from FXO to FXS device ? (Asterisk is 1.4.41) |
| 14:38.56 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:40.12 | [TK]D-Fender | as001, Show us the call coming in and displaying CID, then calling out. Also include your DAHDI configs. |
| 14:40.13 | [TK]D-Fender | ~pb |
| 14:40.14 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:40.16 | [TK]D-Fender | ^^^ |
| 14:45.39 | *** join/#asterisk gerhard7 (~gerhard7@82.168.115.40) |
| 14:45.50 | con3x | Anybody in the UK using a BT ISDN 30e line? |
| 14:46.10 | WIMPy | ~polls |
| 14:46.10 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 14:47.17 | con3x | ok, do BT lines do anything strange or can I just use a standard PRI card like the TE110P |
| 14:47.38 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 14:47.41 | WIMPy | That should be bog standard DSS1 PRI. |
| 14:48.34 | WIMPy | And E1 is E1 anyway. |
| 14:48.49 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.6.77) |
| 14:49.28 | con3x | Something you think would come up in a google search. I was confused when they said you can only connect 30 lines (was pretty sure E1 is 32 data lines) |
| 14:50.19 | WIMPy | 30 B0channel, 1 D-channel and one unused/for synchronization. |
| 14:50.35 | WIMPy | s/0c/-c/ |
| 14:51.11 | con3x | Ahh that clears things up :) |
| 14:51.43 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 14:53.23 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 14:54.48 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:57.46 | *** join/#asterisk Matthias (~Matthias@2001:15c0:670f:ffff::2) |
| 14:59.40 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.246) |
| 14:59.50 | [TK]D-Fender | ~E1 |
| 14:59.50 | infobot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 15:00.16 | *** join/#asterisk Ta^3 (~tacvbo@187.191.13.43) |
| 15:00.25 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 15:02.25 | *** join/#asterisk DanYWalker (c87b087b@gateway/web/freenode/ip.200.123.8.123) |
| 15:02.32 | *** part/#asterisk DanYWalker (c87b087b@gateway/web/freenode/ip.200.123.8.123) |
| 15:02.42 | *** join/#asterisk Ta^3 (~tacvbo@187.191.13.43) |
| 15:03.22 | *** join/#asterisk minotaur01 (~minotaur@24.215.3.50) |
| 15:09.27 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-lsueykljwtgupczf) |
| 15:09.50 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 15:13.39 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 15:14.09 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 15:16.30 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 15:24.00 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.246) |
| 15:24.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:24.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:24.39 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 15:32.56 | *** join/#asterisk StaRetji (~LittleAll@178.79.10.247) |
| 15:34.12 | StaRetji | hi there folks, is it normal that I have 144 active channels, 837 actve calls 1570 calls processed? something is strange here, I have a 253 active sip dialogs ( I have traffic on the server) |
| 15:34.18 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:34.37 | StaRetji | and number of active calls is raising constantly until asterisk simply hangs |
| 15:34.37 | *** join/#asterisk tuxfarva (hidden-use@chtest46.gta.com) |
| 15:35.22 | StaRetji | I use a2billing, so I doubt if agi scripts do not hang the line properly or something. Can anyone help me out here with some tips where to look/read, thx |
| 15:41.09 | *** join/#asterisk JoTo (~JoTo@p5DC16BBD.dip.t-dialin.net) |
| 15:41.25 | JoTo | hi all, sorry, i'm puzzled |
| 15:41.29 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:41.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:41.36 | JoTo | i try to setup a queue with german language |
| 15:41.47 | JoTo | all works fine, but the position is announced in english |
| 15:41.55 | JoTo | there are german digit audio files though |
| 15:42.01 | JoTo | i checked permissions and format |
| 15:42.25 | JoTo | the cli says playing digits/2.alaw (Language: de) |
| 15:42.26 | bulkorok | did you checked the files self!? |
| 15:42.35 | JoTo | hi bulkorok |
| 15:42.39 | bulkorok | in fact listen to one?! |
| 15:42.58 | bulkorok | maybe they are really english ones... |
| 15:42.59 | JoTo | i tried to convert one of them with sox again (only the 2.gsm because i test only with 2 phones right now) |
| 15:43.02 | JoTo | but no success |
| 15:43.08 | JoTo | no, that i checked |
| 15:43.12 | bulkorok | k |
| 15:43.20 | JoTo | if i playback them in a media player it speaks german :) |
| 15:43.28 | bulkorok | asterisk is converting if needed... you don't have to |
| 15:43.56 | JoTo | well, i have to e.g. when i come from a windows wav file...asterisk dont like windows wav files :) |
| 15:44.09 | JoTo | the strange thing is |
| 15:44.24 | JoTo | it plays 2.alaw...but in de directory there are only .gsm files |
| 15:44.31 | JoTo | it dont even try to play the gsm file |
| 15:44.48 | JoTo | but it plays the other queue announcement in gsm format from the de directory very well |
| 15:44.52 | bulkorok | myabe gsm is not allowed on the sip trunk!? |
| 15:44.55 | JoTo | e.g. queue-callswaiting.gsm |
| 15:45.00 | bulkorok | ah... |
| 15:45.29 | JoTo | is there any debuging to see why he dislikes the digit files? |
| 15:45.40 | JoTo | when i just have you on my hook *grin* |
| 15:45.47 | bulkorok | verbose |
| 15:45.54 | JoTo | do you know a way to filter the output of the cli a bit? |
| 15:46.18 | JoTo | when i want to follow whats going on its very confusing because its a productive system with calls here and there |
| 15:46.31 | JoTo | yeah i know asterisk -vvvvvr or that |
| 15:46.38 | bulkorok | you can asterisk -rvvvvv | tee /tmp/cli-ouput.txt |
| 15:47.14 | JoTo | but there i can only see that it plays digits/2.alaw (Language: de) :( |
| 15:47.30 | JoTo | tee? never heard that command...thank you, will try that |
| 15:48.17 | bulkorok | maybe core set debug 10 helos too... |
| 15:48.19 | SuperNull | Anyone run G729 much ? enough to provide me some insight on its ability to fubar outward/inward DTMF (over audio of course) for PBXes |
| 15:48.23 | bulkorok | helps |
| 15:48.28 | SuperNull | for analog pbxes i should say. |
| 15:48.48 | JoTo | can i go up to 10 in verbose? |
| 15:48.51 | JoTo | wow |
| 15:48.55 | JoTo | i thought 5 is max |
| 15:49.01 | bulkorok | I suppose 99 is possible :-) |
| 15:49.09 | JoTo | i read tee manpage...thanks might come handy |
| 15:49.35 | JoTo | then at least i can search for the call id to follow the call in the gazillion of lines :) |
| 15:49.49 | bulkorok | sure... |
| 15:50.59 | JoTo | thanks bulkorok |
| 15:51.08 | bulkorok | :) |
| 15:58.24 | JoTo | ok...gotta run...cu soon |
| 15:58.27 | *** part/#asterisk JoTo (~JoTo@p5DC16BBD.dip.t-dialin.net) |
| 15:58.38 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 16:13.32 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 16:25.50 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 16:28.07 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:30.51 | parasitodelsur | p3nguin: does flowroute allow you to pass your own cid? |
| 16:42.11 | *** part/#asterisk StaRetji (~LittleAll@178.79.10.247) |
| 16:44.23 | *** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26) |
| 16:44.26 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 16:45.36 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 16:53.56 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 16:54.58 | JasonL | When I execute a "reload" I get "The previous reload command didn't finish yet". First time I see that.. Anyone else run into this? Using 1.8.8.0 |
| 16:56.37 | fredericve | I've seen that on asterisk 1.2 before |
| 16:57.07 | fredericve | on asterisk 1.8... try core show locks see if there's a lock being held anywhere |
| 16:58.41 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 17:12.12 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.158) |
| 17:38.03 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.156) |
| 17:38.50 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.156) |
| 17:40.53 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:48.49 | *** join/#asterisk fukuda2 (~IceChat9@183.181.70.86.rev.sfr.net) |
| 17:52.54 | fukuda2 | hi |
| 17:53.16 | fukuda2 | I'm few problems with hylafax, asterisk and iaxmodem |
| 17:56.28 | Qwell | JasonL: What happened when you tried a version of Asterisk from this year? |
| 17:56.51 | fukuda2 | when send a fax with the command : senfax -d "number fax" "files" i'm a message : no carrier detected (on faxstat -s) |
| 17:58.46 | fukuda2 | or a warning message : no local dialtone |
| 17:59.58 | [TK]D-Fender | fukuda2, Why isn't all of your debug in a pastebin to examine? |
| 18:00.00 | [TK]D-Fender | ~pb |
| 18:00.00 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:00.01 | [TK]D-Fender | ^^^ |
| 18:02.14 | fukuda2 | i'm sorry, i'm not access server for copy and paste the debug :( |
| 18:02.58 | fukuda2 | i'm a few screenshot of configuration |
| 18:05.09 | fukuda2 | the iaxmodem is running and idle according to faxstat -s |
| 18:05.16 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:08.12 | p3nguin | parasitodelsur: Yes. |
| 18:08.50 | *** join/#asterisk fritz09 (~Adium@pop1-765.catv.wtnet.de) |
| 18:10.30 | fukuda2 | help please :/ It's a project of school. the project is developped a faxtomail server |
| 18:10.34 | Russ | ectospasm, the fax machine has a dedicated line to it from my VoIP box (DAHDI) |
| 18:10.40 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 18:12.41 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 18:12.46 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 18:17.28 | [TK]D-Fender | fukuda2, You have no access to your server. Do you ask your mechanic what's wrong with your car without giving it to him to look at? |
| 18:18.34 | fukuda2 | I have access to the server but not at home |
| 18:19.00 | [TK]D-Fender | fukuda2, If you don't have it now, then we can't help you now. |
| 18:20.58 | fukuda2 | But I have some configuration screens. |
| 18:21.02 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 18:21.56 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:21.56 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:23.25 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:26.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:27.38 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:27.59 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 18:29.00 | p3nguin | parasitodelsur: It would be somewhat useless if they did not allow that. Having DIDs on one provider and using flowroute to terminate, it would be bothersome to not be able to set the callerid number to that of your DID number. |
| 18:35.06 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 18:35.23 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.156) |
| 18:40.02 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 18:46.14 | *** join/#asterisk areski (~areski@80.174.255.87.dyn.user.ono.com) |
| 18:47.47 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 18:48.38 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 18:50.08 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 18:52.13 | *** join/#asterisk josefig (~josefig@unaffiliated/josefig) |
| 18:56.26 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 19:03.03 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 19:05.08 | josefig | hello, someone can help me when I dial I get a 503 Service Unavailable this is my log http://paste2.org/p/2561632 |
| 19:06.23 | jpsharp | I see a 404, not a 503. |
| 19:07.16 | jpsharp | And really, I don't even see a call attempt there. |
| 19:09.03 | josefig | let me try again |
| 19:22.51 | *** join/#asterisk twodogs (~twodogs@telok.vitiate.me) |
| 19:24.09 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:26.58 | SuperNull | damn you start asking questions about problems and "Quit: jrose_atDigium" haha |
| 19:30.25 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 19:37.28 | *** join/#asterisk jsjc (~Adium@55.228.222.87.dynamic.jazztel.es) |
| 19:38.11 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 19:41.26 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:43.18 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 19:48.25 | jacekowski | i've got a question about emergency services |
| 19:48.52 | jacekowski | how do you prove that it's working without calling the number? |
| 19:49.26 | leifmadsen | jacekowski: you still have to call the number -- you just need to follow the rules for your area. You can typically call a non-emergency number and schedule a time to test it. |
| 19:52.07 | jpsharp | Yep. I've done that to test E911 location data. Just called the local county sherriff's office regular number and asked about it. |
| 19:52.51 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.156) |
| 19:53.11 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 19:53.17 | jpsharp | They told me to call during a certain time and first thing was to say "This is a non-emergency call testing my 911 system, do you have time to assist?" |
| 19:54.02 | *** join/#asterisk mokmeister (~mokmeiste@109.77.113.219) |
| 19:54.06 | jpsharp | And if no, to hangup immediately. |
| 20:10.04 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 20:11.17 | wasabi1 | Has anybody spent much time trying to get Asterisk integrated with Lync and Unified Messaging? |
| 20:25.02 | *** join/#asterisk ircNewbz (~ircnewbz@unaffiliated/ircnewbz) |
| 20:28.59 | *** join/#asterisk cklimos (~Claude@209.5.121.232) |
| 20:29.23 | ircNewbz | Excuse me, but could someone please point me in the direction of a step-by-step walkthrough that a call goes through, explaining what each detail in the /var/log/asterisk/messages file indicates for a call? If such a thing exists, that is. |
| 20:30.15 | *** join/#asterisk radic (~radic@dslb-094-216-239-109.pools.arcor-ip.net) |
| 20:30.22 | navaismo | ~book |
| 20:30.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:30.34 | navaismo | or hire a consultant |
| 20:30.41 | navaismo | ircNewbz, ^^ |
| 20:34.31 | ircNewbz | Thanks! |
| 20:35.52 | weinerk | Hi. Please help with AGI + dial() + G option |
| 20:35.53 | weinerk | Dial returns right away and calling side progresses to hangup. |
| 20:35.53 | weinerk | How can I make dial(G()) work in AGI? |
| 20:39.39 | *** join/#asterisk mogra (477b818a@gateway/web/freenode/ip.71.123.129.138) |
| 20:43.10 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 20:46.56 | *** part/#asterisk ircNewbz (~ircnewbz@unaffiliated/ircnewbz) |
| 20:52.38 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:58.35 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:00.13 | mbrit | hello everyone, i've got this problem, i make a video call and i receive video normally on the video port i specified on the SDP, but when i start sending my video from another different port, asterisk starts sending it to this second one. what could be happening there? |
| 21:01.00 | mbrit | i tried to draw it here http://pastebin.com/tq17FPk7 |
| 21:02.49 | WIMPy | You probably enabled NAT. |
| 21:05.45 | *** join/#asterisk areski (~areski@80.174.255.87.dyn.user.ono.com) |
| 21:05.46 | mbrit | oh ok, it automatically changes the destination port that way then? i thought you would only change that parameter if the peer expressed it on an SDP |
| 21:06.42 | WIMPy | Nope. That's exactely what it does. |
| 21:13.25 | mbrit | thanks WIMPy that was very helpful |
| 21:16.23 | *** join/#asterisk fritz09 (~Adium@2002:542e:5301:e472:85d0:853c:5b0c:85e6) |
| 21:16.51 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:24.21 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.156) |
| 21:49.17 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:51.54 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 21:53.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:00.57 | *** join/#asterisk wonderworld (~w@dsdf-4d0a0d39.pool.mediaWays.net) |
| 22:06.03 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 22:17.53 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 22:31.16 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 22:33.08 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 22:34.54 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:35.04 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 22:35.41 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 22:43.58 | *** join/#asterisk P-NuT (~P-NuT_@5ad3493e.bb.sky.com) |
| 22:45.58 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:01.33 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:06.42 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 23:08.53 | n8ideas | Anyone have any workarounds for some limitations in application FollowMe? Specifically, dealing with offline phones or voicemail detection, or even something as simple as ringing instead of MoH ? |
| 23:09.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 23:10.18 | *** join/#asterisk Alex25 (~kvirc@bzq-79-183-209-222.red.bezeqint.net) |
| 23:10.24 | p3nguin | Are you having trouble with getting the follow me prompt on the cell phone voicemail? |
| 23:12.12 | n8ideas | That happens as well |
| 23:12.42 | Alex25 | Can any DID number redirectable to SIP addresses be used as a fax number? |
| 23:13.14 | wasabi1 | Any of you used repro to route stuff? |
| 23:13.30 | p3nguin | n8ideas: If that problem is secondary, what was your primary issue? |
| 23:14.18 | n8ideas | I would really like to be able to run AMD on answered calls to figure out if we had a real connection or not, for the case where confirm = no |
| 23:14.49 | p3nguin | Doesn't the "press 1 to accept this call" pretty much take care of it being real person? |
| 23:14.54 | n8ideas | in the case where confirm = yes, the prompt on the voicemail is annoying... |
| 23:15.10 | n8ideas | Yes... if I am OK with leaving voicemails on people's voicemail most of the time |
| 23:15.31 | n8ideas | I don't necessarily want to have to confirm |
| 23:15.42 | p3nguin | The easiest way that I have found is to have longer outgoing voice mail messages. |
| 23:15.50 | n8ideas | in which case you basically can't guarantee the forwarding is actually legitimate |
| 23:16.09 | p3nguin | If the ring time on the followme is 30 seconds, make sure the outgoing message is at least 30 seconds long. |
| 23:16.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:16.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:16.57 | p3nguin | That should also take care of instances where the call goes immediately to voice mail. |
| 23:17.15 | p3nguin | dual purpose |
| 23:17.42 | n8ideas | Yeah... I suppose, but that's an awful thing to have to do |
| 23:17.43 | p3nguin | Maybe ring time wasn't the correct term there... |
| 23:18.12 | n8ideas | Also, it appears you can't even change between Music on Hold and ringing in the follow me state |
| 23:18.13 | p3nguin | I'm thinking of the setting that determines how long the call can last before being accepted/denied before it just gives up. |
| 23:18.36 | n8ideas | Oh.... is there a setting on FollowMe for that? |
| 23:18.38 | p3nguin | For that, do not use the FollowMe app. Replicate the behavior with a macro. |
| 23:18.41 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 23:19.31 | n8ideas | I wish I could do that... it was my original approach |
| 23:19.49 | n8ideas | but the Macro(confirm) on Dial actually kills all ringing channels before the call confirm is actually accepted |
| 23:19.55 | n8ideas | I couldn't figure out a way for that not to happen |
| 23:20.06 | Alex25 | Can any DID number redirectable to SIP addresses be used as a fax number? |
| 23:20.45 | p3nguin | I don't know what you mean by "kills all ringing channels." |
| 23:21.42 | p3nguin | Dial() once to the desk phone. When the timeout on that expires, move on to another Dial(M()) for the cell phone. |
| 23:21.45 | n8ideas | When I have a Dial(SIP/1&SIP/2,60,M(confirm)) and SIP/1 answers and hits the confirm macro, SIP/2 immediately stops ringing |
| 23:21.56 | n8ideas | whether or not the call is ever confirmed |
| 23:22.32 | *** join/#asterisk deo (~deo@112.198.79.92) |
| 23:22.44 | p3nguin | In the case where you use FollowMe, you first have a Dial(), then FollowMe(). I.e. two apps. |
| 23:22.55 | n8ideas | Correct... in some cases |
| 23:23.01 | p3nguin | Just replace the FollowMe() with another Dial using the macro. |
| 23:23.02 | n8ideas | depends on how the user configured the forwarding settings |
| 23:23.23 | n8ideas | Right... that would work, expect if the Dial step has multiple phones in it, the behavior above occurs |
| 23:23.46 | n8ideas | Hitting M(confirm) immediately kills all other ringing channels, which isn't the behavior in FollowMe |
| 23:23.46 | p3nguin | You want to follow TWO phones? |
| 23:23.54 | n8ideas | Right... |
| 23:24.04 | p3nguin | Hmm, that might change things. |
| 23:24.11 | n8ideas | so I might want to have two phones rung after 15 initial seconds of ringing the main extension |
| 23:24.37 | p3nguin | Try putting a local channel in between the dial plan and the cell phones. |
| 23:24.40 | SeRi | p3nguin: how is the earpiece working out? |
| 23:24.43 | p3nguin | No clue. |
| 23:25.04 | SeRi | p3nguin: Havent tryed it? |
| 23:25.15 | p3nguin | Correct. I haven't even collected it from the mail. |
| 23:25.47 | SeRi | o. ok. Interesting. Last I saw it was "delivered" |
| 23:25.59 | SeRi | maybe it was a different pkg |
| 23:26.19 | p3nguin | It probably is delivered, but I haven't collected it. |
| 23:26.52 | SeRi | ok cool. |
| 23:26.56 | n8ideas | I did try the local channel as well |
| 23:27.08 | n8ideas | it doesn't make any difference in the behavior of the dial application |
| 23:27.39 | p3nguin | That's too bad. That was the only other thing I could think of right off hand. |
| 23:29.09 | SeRi | p3nguin: so if I wanted to use flowroute as out bound I can set my did to my voip.ms and just use voip.ms for incoming. |
| 23:29.38 | p3nguin | Your DIDs are already on voip.ms, aren't they? |
| 23:29.52 | SeRi | yes. |
| 23:29.57 | SeRi | ops |
| 23:29.59 | SeRi | cid |
| 23:30.03 | SeRi | I meant cid |
| 23:30.04 | SeRi | lol |
| 23:30.06 | n8ideas | Yeah.. seems to be a painful limitation on the Dial macro handler itself |
| 23:30.21 | p3nguin | Just change your outbound extensions to use your flowroute peer. Nothing else needs to be done. |
| 23:30.24 | n8ideas | I would happily write this directly in dialplan, but I can't have the Dial macro disconnecting all channels when the answer macro is hit |
| 23:30.40 | p3nguin | Your callerid number should be set on the outbound extensions anyway, so no other changes need to be made. |
| 23:30.44 | SeRi | got it p3nguin. Thanks |
| 23:31.04 | wasabi1 | If anybody has experience with repro. Any idea why it might be ignoring a route, and instead trying to route by DNS (which resolves to itself, and thus causes a loop?) |
| 23:31.32 | p3nguin | What is repro? |
| 23:33.17 | wasabi1 | A SIP proxy. |
| 23:33.21 | *** join/#asterisk radic (~radic@dslb-094-216-251-170.pools.arcor-ip.net) |
| 23:34.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tyfuhnebaikijweg) |
| 23:34.49 | SeRi | wasabi1: I am not sure you will ahve luck with that here |
| 23:35.08 | SeRi | s/ahve/have/ |
| 23:35.15 | wasabi1 | Well, there's not much else of a place for asking where there's a chance of it. And I've seen lots of people using it to proxy to/from asterisk to other systems. |
| 23:35.32 | p3nguin | It isn't something people here have mentioned in my presence. |
| 23:36.20 | p3nguin | Check back occasionally and maybe someone who has used it will pop up. |
| 23:46.25 | *** join/#asterisk minotaur01 (~minotaur@S01060018e7f9c7df.hm.shawcable.net) |