| 00:03.21 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 00:06.04 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 00:14.38 | *** join/#asterisk guilhermebr (~Guilherme@189.63.91.131) |
| 00:16.12 | citywok | Yup, i found the bug in realtime_update_peer, it uses ast_config_AST_SYSTEM_NAME which is set to null regardless of the asterisk.conf setting for configuring the systems hostname. |
| 00:35.24 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 00:49.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:17.26 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 01:19.33 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 01:21.02 | cmendes0101 | I'm building asterisk from SVN for chan_rtmp support. When I run "make" on asterisk I get these errors http://pastebin.com/KivfjxcW |
| 01:21.46 | cmendes0101 | I can't seem to find how to give CFLAGS the -fPIC flag like the error suggests |
| 01:37.48 | *** join/#asterisk dbtid (utauanvf9r@unaffiliated/dbtid) |
| 01:38.14 | dbtid | howdy... can someone point me to some docs that describe the asterisk/dahdi framework? how audio data flows through the system? |
| 01:41.38 | russ | I can point you to the fact that asterisk is just plain awesome |
| 01:42.11 | russ | What information are you looking for specifically? |
| 01:42.15 | *** join/#asterisk n0tk (~n0tk@66-193-170-248.static.twtelecom.net) |
| 01:42.43 | dbtid | russ: data flow from hardware through dahdi_ec_span, dahdi_receive and dahdi_transmit |
| 01:42.58 | dbtid | an explanation of spans, channels, etc |
| 01:49.33 | *** join/#asterisk coppice (~chatzilla@m121-202-1-215.smartone-vodafone.com) |
| 01:51.25 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
| 02:13.21 | *** join/#asterisk OrNix (~ornix@l49-48-15.cn.ru) |
| 02:19.55 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:23.25 | *** join/#asterisk Alagar (~Administr@122.164.24.117) |
| 02:27.14 | *** join/#asterisk moy (~moy@74.12.96.53) |
| 02:50.56 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177778310.dsl.bell.ca) |
| 03:09.11 | *** join/#asterisk guilhermebr (~Guilherme@187.113.37.229) |
| 03:21.08 | *** join/#asterisk moy (~moy@74.12.96.53) |
| 03:29.29 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:29.36 | joako | how is 1-way audio when there is no NAT possible? |
| 03:36.22 | *** join/#asterisk whtsup (~tun@119.63.136.51) |
| 03:36.30 | whtsup | hello |
| 03:41.26 | Maliuta | joako: mute? |
| 03:41.48 | whtsup | can i check the log of hungup call |
| 03:41.54 | whtsup | reason of hungup calls |
| 03:43.19 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 03:43.21 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:43.46 | Maliuta | whtsup: depends on if you have debugging on and if you're logging those messages |
| 03:44.09 | joako | Maliuta, if it was only that simple. No everything is setup right one IP 192.168.3.28 and the other 192.168.3.188 but 100% I get 1 way audio\ |
| 03:47.33 | joako | Now I have all these odd messages pop up: Sep 1 23:47:15 WARNING[17537]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0x9313e78 (len 467) to 192.168.3.28:5060 returned -1: Bad file descriptor |
| 03:52.06 | *** join/#asterisk radic (~radic@dslb-094-216-232-212.pools.arcor-ip.net) |
| 03:52.56 | joako | Is it normal to see in my calls Nativeformats: ulaw, WriteFormat: slin, ReadFormat: slin? |
| 03:56.05 | *** join/#asterisk moy (~moy@74.12.96.53) |
| 04:01.44 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 04:18.50 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 04:19.49 | joako | Between the same hosts I changed from SIP to IAX and still have 1-way audio. how is this possible? |
| 04:20.42 | joako | There is no nat, router, firewall between the two they are plugged into the same switch on the same subnet each machine only have 1 network card |
| 04:21.49 | *** join/#asterisk mpe (~mpe@65.161.206.209) |
| 04:26.17 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 04:27.05 | kaldemar | joako: a broken phone? |
| 04:28.16 | joako | kaldemar, No the phone works. The problem is I am having 1-way audio on calls between asterisk 1 and asterisk 2 which are on the same network |
| 04:29.39 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 04:32.14 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 04:32.39 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:37.21 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 04:39.04 | *** join/#asterisk sourcode (~code@ppp-58-8-115-232.revip2.asianet.co.th) |
| 04:54.27 | *** join/#asterisk sourcode_ (~code@ppp-58-8-115-232.revip2.asianet.co.th) |
| 04:58.43 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 05:02.15 | whtsup | IF I want to connect 2 asterisk boxes sip is better or iax .. |
| 05:02.47 | uqlev | whtsup, iax2 is dedicated for it |
| 05:05.23 | joako | Well I tried IAX and I have the exact same 1-way audio problem. I am stumped since both machines are on the same LAN |
| 05:06.26 | whtsup | uqlev |
| 05:06.34 | whtsup | i have connected 2 asterisk boxes |
| 05:06.38 | whtsup | client and server |
| 05:06.49 | whtsup | on client side tdm400 card is installed |
| 05:06.50 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 05:07.14 | whtsup | when call comes from server to client |
| 05:07.20 | whtsup | mostly calls r hungup |
| 05:07.28 | whtsup | just showing hungup on server side |
| 05:07.33 | whtsup | some calls r answered |
| 05:07.44 | whtsup | i dont know wht is the reason |
| 05:14.25 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 05:21.20 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 05:24.37 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 05:28.23 | ChannelZ | wrong signalling method? |
| 05:29.04 | [TK]D-Fender | Firewalls |
| 05:29.10 | [TK]D-Fender | (probably) |
| 05:29.17 | [TK]D-Fender | Shouldn't affect IAX2 normally mind you |
| 05:30.43 | whtsup | hmmmmm |
| 05:30.50 | whtsup | signalling method on which side ? |
| 05:31.15 | whtsup | well i m doing wholesale termination |
| 05:31.31 | whtsup | not all calls hungup |
| 05:31.34 | whtsup | mostly calls |
| 05:32.11 | [TK]D-Fender | SIP & IAX is irrelevant normally. |
| 05:32.23 | whtsup | <PROTECTED> |
| 05:32.24 | whtsup | ? |
| 05:32.33 | whtsup | any suggestion plz |
| 05:32.51 | [TK]D-Fender | [01:31]<whtsup>not all calls hungup <- there is no "detection. All VoIP protocols have setup and teardown, progress etc. Ther is no "detection". It is a fact. |
| 05:33.10 | [TK]D-Fender | whtsup: I recommend SIP wherever you can and IAX only where you have to |
| 05:33.43 | whtsup | soo my setup is |
| 05:33.54 | whtsup | like my client is using some sort of voipswitch |
| 05:34.06 | whtsup | i have registered him as a sip |
| 05:34.23 | whtsup | 0 |
| 05:34.42 | whtsup | and my both boxes connected on iax trunk |
| 05:35.13 | *** join/#asterisk coppice (~chatzilla@m121-202-1-215.smartone-vodafone.com) |
| 05:35.21 | whtsup | can i connect 2 asterisk boxes over sip using g723 codec |
| 05:35.22 | *** join/#asterisk Nephfl (~Nephfl@rrcs-67-78-149-118.se.biz.rr.com) |
| 05:35.24 | whtsup | ? |
| 05:36.23 | Nephfl | hello |
| 05:38.23 | [TK]D-Fender | whtsup: SIP is just a call protocol, and G.723 is jsut a codec. |
| 05:38.48 | [TK]D-Fender | whtsup: So YES. Except that * cannot TRANSOCDE that codec normally unless you have the TC400 transcoder card |
| 05:39.09 | [TK]D-Fender | why* can pass the call off device to device however. Just no transcoding. |
| 05:40.08 | ChannelZ | I thought he said he had a TDM400 |
| 05:40.38 | ChannelZ | (on the whole hangup thing which I'm not sure I even understood what he was after) |
| 05:42.12 | [TK]D-Fender | Hangup problem has nothing to do with SIP/IAX |
| 05:49.52 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 05:57.06 | [TK]D-Fender | checkout time, later all |
| 06:02.59 | ChannelZ | yeah I know that |
| 06:04.35 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:08.22 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 06:14.45 | *** join/#asterisk hmmhesays (~hmmhesays@24-116-107-203.cpe.cableone.net) |
| 06:14.53 | Nephfl | so, why do systems music on hold always sound static-y? |
| 06:15.35 | sawgood | Did you encode PCM as a .wav or as a .MP3? |
| 06:16.20 | Nephfl | this one is .MP3, but seems most peoples systems are that way... I imagine I may need to transcode better maybe some sort of compressor |
| 06:16.26 | *** join/#asterisk Justman (~just@78.108.73.46) |
| 06:16.40 | sawgood | probably ... I use GoldWave neat program |
| 06:18.43 | ChannelZ | you mean 'other people's in general'? |
| 06:21.31 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 06:22.10 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 06:27.34 | sawgood | ChannelZ: I downloaded the README from Digium on how to add/setup G.729A licenses ... have you done this much? |
| 06:30.48 | *** join/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 06:32.50 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 06:35.16 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 06:35.33 | drclue | Having a lot of fun with Asterisk AsterClick. So far the test application (A Asterisk HUD) is doing meetme,call queues,xfer,parking , chat , spy functions, monitoring functions , and responding to every single Asterisk event and command. |
| 06:35.56 | AliRezaTaleghani | can someone tell me more about this log? |
| 06:36.00 | AliRezaTaleghani | Auto fallthrough, channel 'SIP/192.168.11.136-2ab94000' status is 'NOANSWER' |
| 06:36.27 | AliRezaTaleghani | 11.136 is a ciscoCallmanager which try to send call to asterisk.. |
| 06:41.11 | kaldemar | AliRezaTaleghani: the other end didn't answer within a timeout and your dialplan had no more lines for asterisk to execute. |
| 06:41.13 | drclue | Sounds like the recipient of the call did not answer before the attempt expired |
| 06:42.35 | drclue | Yes and as Kaidemar expressed , your dialplan had no entry to account for the outcome of "NO ANSWER" and fell through to a default behavior |
| 06:46.01 | AliRezaTaleghani | kaldemar: drclue: tnx |
| 06:46.28 | drclue | So what would folks really prefer as the second WBEA application that makes HTML5 work without a browser? The current WBEA makes AsterClick web based applications run as executables on Windows and Linux (just a little more to finish_ but should the next one be for Droid or iPhone? |
| 06:50.27 | drclue | Basically Asterclick is an HTML5 context with a matching server side AMI bridge that provides non-polling event driven interactions with asterisk. The WBEA makes those web applications run as desktop executables in windows and Linux without a browser. The same tool is being made available to the smart phone context, so which phone should get the WBEA first? |
| 06:55.00 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ayjevjadkmlkgwes) |
| 06:58.49 | drclue | So it looks like aside from having to flip a coin over which next context will be able to run AsterClick applications , probably nobody is interested in asterisk configuration editing that is so live hot that it can be applied to even a call in progress if desired |
| 07:00.00 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 07:00.01 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:01.56 | drclue | Must be that it is just Thursday. |
| 07:03.07 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:10.36 | *** join/#asterisk BANSAL (~bansal@117.199.124.134) |
| 07:13.01 | *** join/#asterisk odenkos_mobile (~odenkos@stip-static-181.213-81-187.telecom.sk) |
| 07:23.41 | *** join/#asterisk MeisterM (~MeisterM@dslb-084-059-029-154.pools.arcor-ip.net) |
| 07:23.59 | MeisterM | hello, is this channel english only? |
| 07:24.22 | MeisterM | i have problems with SIP Peer Registration with Arcor/Vodafone (German) |
| 07:24.26 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 07:24.46 | MeisterM | there is a registration Timeout every 20 Seconds and no connection is established |
| 07:25.27 | joako | I have 1-way audio without any NAT or firewall and I don't understand how that is possible. |
| 07:25.48 | MeisterM | This is my Register line: register => 06109xxxxx:xxxxx:06109xxxxx@06109.sip.arcor.de/06109xxxxx |
| 07:27.33 | joako | MeisterM, registration timeout means asterisk did not get a response to its registration request |
| 07:28.50 | odenkos_mobile | MeisterM i would be willing to help you in german when i get home |
| 07:29.05 | ChannelZ | joako: is this a softphone or..?" |
| 07:29.10 | ChannelZ | and which way works |
| 07:31.05 | MeisterM | odenkos_mobile Danke für dein Angebot |
| 07:31.26 | MeisterM | joako I have now forwarded Port 5060 from my Router to my Asterisk Server but still the same problem |
| 07:31.41 | MeisterM | do i need several additional ports? |
| 07:31.45 | joako | MeisterM, That shouldn't really matter. |
| 07:32.19 | MeisterM | okay |
| 07:32.50 | MeisterM | is there a possibility to debug the problem. when i use a softphone on my PC i can connect and register with the SIP Host without any problem |
| 07:33.00 | MeisterM | so i think there is a mistake in my sip config |
| 07:33.10 | *** join/#asterisk Diffen2 (~diffen2@81-234-228-6-no32.tbcn.telia.com) |
| 07:34.17 | kaldemar | ~sipnat |
| 07:34.17 | infobot | i guess sipnat is Quick guide on configuring Asterisk + 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 |
| 07:34.21 | kaldemar | MeisterM: ^^ |
| 07:37.45 | MeisterM | but what is the problem with NAT? my Softphone works great without additional NAT configuration |
| 07:37.53 | joako | MeisterM, now if you turn off your softphone and restart your router will it connect (could be broken nat)? also did you set all the nat options in sip.conf? |
| 07:38.32 | joako | ChannelZ, it is between 2 asterisk! But I just resolved it with "service iptables stop" |
| 07:38.39 | joako | firewalls driving me crazy this week! |
| 07:38.56 | kaldemar | MeisterM: your asterisk puts a private address in the registration message unless you tell it do not to. |
| 07:39.12 | MeisterM | ok can you tell me the parameter i have to set? |
| 07:39.34 | MeisterM | is it nat=yes ? |
| 07:40.02 | joako | MeisterM, and externip and localnet |
| 07:40.39 | sawgood | If I want a SIP (IP to IP call) to leave an Asterisk box (to call anon. to another Asterisk box) ... (and not use the ITSP) ... how would I write that in the dialplan |
| 07:40.51 | sawgood | SIP/exen@IP address |
| 07:40.58 | sawgood | did not work (it tried to use the ITSP trunk) |
| 07:41.03 | kaldemar | MeisterM: it's all in the guide that infobot just posted. |
| 07:41.21 | joako | sawgood, that part is fine the problem is your extension matching |
| 07:41.57 | sawgood | joako: Thank you ... can you offer a tip/solution |
| 07:42.34 | kaldemar | sawgood: how did it not work? is the ITSP accepting anonymous calls btw.? |
| 07:43.17 | sawgood | Well, with SIP set debug on ... I see a call trying to leave out to the SIP trunk provider (ITSP) and not to the IP of the other Asterisk box (which does have anon. sip on) |
| 07:43.55 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 07:44.08 | sawgood | I kind of want this call to not use the ITSP at all (and make a direct IP to IP call) (as if the ITSP account was not even an option to use) |
| 07:44.21 | drclue | So it appears that of all the new features coming to AsterClick , the most excitement is to be had in the merging of FastAGI with AsterClick, allowing dialplan changes to be reflected in even an ongoing call without any dial plan reload. The things that excite folks in this vast feature set will always amaze me |
| 07:44.45 | sawgood | I did not want to make a trunk/peer between the two Asterisk boxes |
| 07:45.04 | MeisterM | now i set externhost to my dynamicDNS Address and externrefresh to 120 and nat to yes but without success |
| 07:45.24 | sawgood | did you do a SIP reload? |
| 07:46.04 | MeisterM | i did a full reload |
| 07:47.09 | drclue | I guess the reload thing does occupy a lot of folks thinking so having the config hot is going to be a big deal |
| 07:47.51 | ChannelZ | sawgood: SIP/x.x.x.x/exten |
| 07:48.11 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 07:48.27 | sawgood | ChannelZ: thank you (I'll try) |
| 07:48.38 | sawgood | actually, I think that was my first try ... but I'll try again |
| 07:48.38 | joako | sawgood, post the dialplan show output for itsp and other call |
| 07:48.58 | ChannelZ | if it didn't work, it's for some other reason |
| 07:49.42 | drclue | Hot config has been available in Asterisk for multiple versions , so I guess if we are all still talking about reload , this is going to be awesome |
| 07:50.03 | sawgood | So, the call did not work (it did not leave the 1st box) (I have consoles open on the source and destination boxes) |
| 07:50.54 | ChannelZ | and it said... |
| 07:51.37 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:51.58 | sawgood | -- SIP/220.109.21.88-0000000c is circuit-busy |
| 07:52.16 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 07:52.34 | MeisterM | is there a possibility to see the sip messages? |
| 07:53.51 | kaldemar | MeisterM: core set verbose 10, sip set debug on <--- CLI commands |
| 07:55.09 | MeisterM | thanks but its to fast how can i log my console in a file? |
| 07:55.19 | ChannelZ | hmm that IP doesn't respond for me so it's saying that for a reason |
| 07:55.39 | ChannelZ | IE a better reason, like because the remote box rejected it, etc |
| 07:58.15 | ChannelZ | MeisterM: logger.conf |
| 08:01.01 | ChannelZ | turn on 'verbose' for whatever logfile (like messages) |
| 08:01.15 | ChannelZ | or full |
| 08:01.46 | MeisterM | ok i guess my own IP Address in the SIP Message is correct |
| 08:02.26 | MeisterM | From: <sip:06109xxxxx@06109.sip.arcor.de>;tag=as5b517593 |
| 08:02.26 | MeisterM | To: <sip:06109xxxxx@06109.sip.arcor.de> |
| 08:02.31 | MeisterM | is this the right one? |
| 08:02.46 | MeisterM | there is a field "Contact" with my IP |
| 08:03.38 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:04.52 | ChannelZ | I missed the beginning of your problem... you can't register? |
| 08:04.53 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:05.01 | MeisterM | yes i get a timeout |
| 08:05.16 | odenkos_ntbk | MeisterM: do you still need anything? |
| 08:05.28 | ChannelZ | so either something is blocking outgoing port 5060, or the return traffic is being blocked |
| 08:05.32 | MeisterM | Asterisk is sending the register sip message to my sip provider 6 times then i get a timeout |
| 08:06.00 | MeisterM | i forwarded incoming port 5060 udp to my asterisk server |
| 08:06.01 | ChannelZ | (by outgoing 5060 I mean traffic going TO 5060 out on the net) |
| 08:06.17 | MeisterM | outgoing traffic isn't blockes |
| 08:06.18 | MeisterM | blocked |
| 08:06.45 | ChannelZ | Then it's blocking the reply |
| 08:06.52 | ChannelZ | What kind of firewall is this? |
| 08:07.46 | *** join/#asterisk Secret_Hamster (~pdiddly@host81-149-30-66.in-addr.btopenworld.com) |
| 08:07.47 | MeisterM | its a Netgear ProSafe Router |
| 08:08.02 | sawgood | I think in my case, when I dial 4999 (the 1st Asterisk box is trying to 'answer' the call locally) (I do not think it is trying to leave on the Internet to deliver the call) ... do I need to put 4999 in my outbound route? |
| 08:08.04 | ChannelZ | hmm |
| 08:08.40 | ChannelZ | sawgood: that has no meaning. What exactly are you trying to do? |
| 08:08.48 | MeisterM | the local linux firewall isnt running and i can register my phones on my asterisk server without any problems |
| 08:09.03 | sawgood | MeisterM: did you turn off SIP ALB on the Netgear router? |
| 08:09.06 | sawgood | very important |
| 08:09.12 | sawgood | SIP ALG (sorry) |
| 08:09.31 | ChannelZ | MeisterM: Well I don't know why but your Netgear is either not allowing your SIP packets out, or it's not letting the replies come back in. |
| 08:10.17 | sawgood | Asterisk box 1 wants to make an IP to IP call (using the open Internet) and not its ITSP SIP trunk (Asterisk box 2 is waiting for the incoming anonymous SIP call) |
| 08:10.17 | MeisterM | but my softphone is working |
| 08:10.24 | MeisterM | on the other machine |
| 08:10.46 | sawgood | I have a customer extension on * box 1 set to do this: SIP/IP address of * 2/extension |
| 08:10.46 | ChannelZ | to the same external host through the same firewall? |
| 08:10.53 | MeisterM | yes |
| 08:11.51 | ChannelZ | well again all I can tell you is that if you're getting timeouts, that means your * box isn't getting a reply. Why that is specifically for your network, I don't know. |
| 08:12.22 | ChannelZ | Maybe the remote end is ignoring you even, it's difficult to say. Or double-check you're trying to register to the right address even |
| 08:13.10 | MeisterM | asterisk is getting the right ip address by dns lookup and write it in the outgoing sip package |
| 08:13.35 | MeisterM | Reliably Transmitting (NAT) to 212.144.112.76:5060: |
| 08:13.38 | ChannelZ | sawgood: and the result of that is 'circuit busy' on the calling side... which more than likely means the remote side rejected it |
| 08:13.47 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-gpzcithvjbxcfnkw) |
| 08:15.08 | sawgood | you are right ... when I turn SIP set debug ON (Asterisk box 2) I see the SIP messages arriving (I was simply looking at the console) with debug/verbose set at 5 (and nothing was coming in) |
| 08:16.31 | ChannelZ | hmm it should be saying _something_ |
| 08:16.42 | MeisterM | so asterisk send Call-ID: 0eda60054235b48172807fdc611f378c@127.0.0.2 is this correct with this end ip? |
| 08:17.09 | sawgood | SIP/2.0 401 Unauthorized |
| 08:17.10 | MeisterM | otherwise the sip package from asterisk and my softphone looks the same |
| 08:17.26 | ChannelZ | so then you don't have 'anonymous' calls setup right on *2 |
| 08:17.35 | sawgood | I'll double check (BRB) |
| 08:17.41 | *** join/#asterisk tiny_D (~tiny_D@mail.snom.com) |
| 08:17.59 | ChannelZ | either that or you have a SIP peer defined in *2 that is matching *1's IP |
| 08:18.23 | ChannelZ | or its name, if it's a friend and not a peer |
| 08:18.39 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:19.39 | Alton35 | Any other good ideas for outbound providers? I'm using teliax and voipms now |
| 08:19.58 | ChannelZ | what is the problem with either of them? |
| 08:20.08 | Alton35 | nothing, only 10 channels each though |
| 08:20.19 | *** join/#asterisk schmidts (d5ebd4c1@gateway/web/freenode/ip.213.235.212.193) |
| 08:20.22 | schmidts | hello all |
| 08:20.40 | ChannelZ | ~itsplist-us |
| 08:20.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 08:21.06 | Alton35 | thanks, I need to copy this :-) |
| 08:21.47 | Alton35 | got it, thanks |
| 08:22.43 | ChannelZ | MeisterM: the Call-ID is kind of arbitrary |
| 08:23.06 | sawgood | ChannelZ: So on * #2: even with anon. set to yes (will I still need an incoming route to send the anon. calls to x4999 voicemail)? |
| 08:23.49 | *** join/#asterisk jetlag (jetlag@pool-173-61-217-74.cmdnnj.east.verizon.net) |
| 08:24.00 | ChannelZ | by anonymous I assume you mean 'allowguest' and that you have a default context defined |
| 08:24.21 | sawgood | yes, and let me check the context ... (BRB) |
| 08:24.24 | ChannelZ | But as I said, you might be getting the 401 reply because it's matching a peer in sip.conf |
| 08:26.40 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:30.48 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 08:36.13 | ChannelZ | bed time |
| 08:37.37 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 08:40.31 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 08:40.54 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 08:40.57 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:41.09 | AliRezaTaleghani | exten => _!,1,Dial(SIP/7088485${EXTEN}@192.168.11.136) |
| 08:41.20 | AliRezaTaleghani | what is the meaning of _! ? |
| 08:41.28 | AliRezaTaleghani | what will be match with this pattern? |
| 08:42.01 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:44.18 | kaldemar | AliRezaTaleghani: _ makes it a pattern. without it, ! would be a literal exclamation mark. |
| 08:44.47 | kaldemar | AliRezaTaleghani: so, _ doesn't match anything, it just denotes a pattern. |
| 08:45.40 | kaldemar | AliRezaTaleghani: what _! matches, you'll find here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 08:45.58 | *** join/#asterisk darkskiez_ (~dz@195-11-205-216.suip.mezzonet.net) |
| 08:46.05 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 08:47.01 | *** join/#asterisk Tim_Toady (~moi@77.49.107.168.dsl.dyn.forthnet.gr) |
| 08:50.09 | sawgood | ChannelZ: got it working (thanks for your tips and advice) |
| 08:54.08 | *** join/#asterisk BANSAL (~bansal@117.199.125.194) |
| 08:55.13 | sawgood | cool ... I can have an open Internet (IP to IP) call in 'becon' |
| 08:56.37 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-168-208.clienti.tiscali.it) |
| 08:59.45 | EmleyMoor | has asked his usual PC parts supplier if they can get AEX cards... |
| 09:00.14 | EmleyMoor | The initial response was "Let me find out." |
| 09:04.31 | sawgood | By default, is there a 'time limit' on long a VM could be left? |
| 09:04.45 | sawgood | I meant 'how long' a VM can be left (in time) |
| 09:08.21 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 09:13.05 | *** join/#asterisk jetlag (jetlag@pool-173-61-237-92.cmdnnj.east.verizon.net) |
| 09:21.03 | *** join/#asterisk mpe (~mpe@65.161.206.209) |
| 09:27.39 | J4zen | Im a bit confused on what hardware to pick as an E1 card from Digium, how does the number of ports on the card relate to the number of channels you can use? |
| 09:27.42 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 09:28.25 | J4zen | for example, i'll need to traffic about 30 simultaneous calls. What sort of E1 card would i need? |
| 09:29.42 | sawgood | Use a SIP trunk and get unlimted concurrent simultaneous calling |
| 09:30.18 | J4zen | We don't want SIP-trunks, we have our own interconnect |
| 09:30.48 | kaldemar | J4zen: one E1 port is 30 channels, one T1 port is 24. |
| 09:31.14 | J4zen | Thanks Kaldemar, that's static info for all cards? |
| 09:32.06 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 09:32.52 | J4zen | Let me rephrase, it's 30 channels per E1 for all cards (forced standard) or does it differ per manufacturer? |
| 09:33.08 | Diffen2 | Hello, Is it possible to show the DID number when using P-Asserted-Identity? For example some variable that shows the DID of the incoming call. |
| 09:37.18 | kaldemar | J4zen: yes, defined by standards. |
| 09:41.12 | *** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 09:43.04 | J4zen | thank you Kaldemar, apreciate it. |
| 09:51.10 | tzafrir_laptop | 30 channels: single E1 port |
| 09:51.16 | tzafrir_laptop | J4zen, ==^ |
| 09:52.39 | J4zen | i found some more info on one of the cards i'm looking at, it states 1 E1 port supports 32 channles. I assume these extra two channels are actually used for signalling and therefor cannot be used for actual calls? |
| 09:53.17 | AliRezaTaleghani | can someone help me to debug this problem? |
| 09:53.20 | AliRezaTaleghani | http://paste.ubuntu.com/487156/ |
| 09:53.42 | AliRezaTaleghani | we have a Cisco which call asterisk ... |
| 09:54.00 | AliRezaTaleghani | when the calls route from asterisk to cisco, all thing are correct |
| 09:54.05 | J4zen | quoting for a two port E1 PCI card from digium -> **The new module enables users to eliminate echo tails up to 128ms or 1024 taps across all 64 channels in E1 mode or 48 channels in T1/J1 modes. |
| 09:54.08 | AliRezaTaleghani | we can talk on buth side |
| 09:54.22 | AliRezaTaleghani | but when the call route from cisco to asterisk |
| 09:54.46 | AliRezaTaleghani | just we can hear the asterisk side, |
| 09:55.00 | AliRezaTaleghani | and the out voice call carry to asterisk |
| 09:55.04 | AliRezaTaleghani | :-/ |
| 10:01.41 | tzafrir_laptop | J4zen, E1 is theoreticall 32 channels. But two of those are used for administrative usage |
| 10:02.01 | tzafrir_laptop | (chan 0: timing and such, chan 16: D channel) |
| 10:02.16 | tzafrir_laptop | So you're left with only 30 channels of actual calls |
| 10:02.30 | tzafrir_laptop | That's per port. So total of 60 channels |
| 10:02.47 | tzafrir_laptop | AliRezaTaleghani, on the same LAN? |
| 10:03.37 | AliRezaTaleghani | tzafrir_laptop: it has been solve |
| 10:03.48 | AliRezaTaleghani | so turn off the caller-id facility on cisco |
| 10:03.58 | J4zen | Thanks tzafrir_laptop |
| 10:04.07 | AliRezaTaleghani | i don't no what was the reall reason... but it's solved right now |
| 10:07.03 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:21.38 | *** join/#asterisk MeisterM (~MeisterM@dslb-084-059-029-154.pools.arcor-ip.net) |
| 10:21.41 | MeisterM | hi |
| 10:21.59 | MeisterM | i got asterisk to work. now i can receive incoming calls over my SIP Line to sip provider |
| 10:22.19 | MeisterM | but when i try to call out i get the error Forbidden |
| 10:22.34 | MeisterM | in the sip message i get: Warning: 399 arcor.de " Authentication: control MD5 response failed" |
| 10:23.00 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 10:25.12 | tzafrir_laptop | MeisterM, and you're sure you got the username/password right? |
| 10:26.29 | MeisterM | oh shit... in one field i added the @ postfix.... |
| 10:26.34 | odenkos | hi all, a quick question concerning dialplan patterns.. if I use for ex. [1237-9], can't I use [12379]? wouldn't it have the same effect? |
| 10:26.40 | MeisterM | i think i have done enough today... ;-) thanks# |
| 10:27.09 | tzafrir_laptop | [1237-9] means [123789] |
| 10:27.34 | tzafrir_laptop | Which is also the same as [1-37-9] |
| 10:28.34 | odenkos | aha, now I get it, thanks tzafrir_laptop.. I can't read apparently :/ |
| 10:29.49 | tzafrir_laptop | is impresed. People actually manage to help themselved today |
| 10:33.41 | fors1 | anyone got a init.d script for wanpipe that supports debian squeeze run-time dependencies ? |
| 10:34.25 | tzafrir | fors1, I don't really have it. But it shouldn't be difficult to add |
| 10:34.39 | tzafrir | what should it run after? |
| 10:34.48 | tzafrir | Should any other service run after it? |
| 10:35.45 | tzafrir | (if so, the Should-Start fields of those services should be updated) |
| 10:36.06 | fors1 | boot order is not important, really. I'm no expert in init.d scripting. I know how to alter a init.d script to support LSB, but I can't find any init.d script for wanpipe, only rc.d, which I can't get working |
| 10:36.36 | tzafrir | fors1, actually the order is very important |
| 10:37.00 | tzafrir | The dependencies are all about the startup order |
| 10:37.37 | fors1 | hm. yeah, I expressed my self poorly. after dahdi, before asterisk |
| 10:37.50 | *** join/#asterisk koudaddy (~gouki@67.91.6.130) |
| 10:38.26 | fors1 | uhm. I must be blind. just found wanrouter script in init.d :) sorry for bothering you. will try to alter it to support dependencie booting |
| 10:42.03 | tzafrir | are wanpipe and wanrouter different services? |
| 10:42.55 | fors1 | nah. wanpipe is the package name, wanrouter is the tool to activate the drivers |
| 10:45.56 | *** join/#asterisk l0st-soul (~lost@scylla.sysif.net) |
| 10:49.31 | fors1 | tzafrir: I guess I should add "wanrouter" to the "Should-Start:" in the asterisk init.d script? |
| 10:55.58 | fors1 | tzafrir: got it working by adding the proper init info, and altering the asterisk init info. will talk to moy, maybe he can add a debian specific init script in the next package of wanpipe |
| 10:56.40 | *** join/#asterisk darkskiez_ (~dz@62-50-207-46.client.stsn.net) |
| 11:05.40 | *** join/#asterisk m0t3jl (~petr.mote@77.78.93.40) |
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| 11:07.55 | *** join/#asterisk dbtid (nfol5porss@unaffiliated/dbtid) |
| 11:08.02 | dbtid | hrm got bumped off and didn't see a reply |
| 11:08.07 | dbtid | howdy... can someone point me to some docs that describe the asterisk/dahdi framework? how audio data flows through the system? |
| 11:08.15 | dbtid | russ: data flow from hardware through dahdi_ec_span, dahdi_receive and dahdi_transmit |
| 11:08.21 | dbtid | an explanation of spans, channels, etc |
| 11:08.28 | dbtid | thanks. |
| 11:09.08 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 11:11.10 | *** join/#asterisk schmidts (d5ebd4c1@gateway/web/freenode/ip.213.235.212.193) |
| 11:11.13 | schmidts | re |
| 11:11.34 | schmidts | could someone point me to a good irc client for mac osx ? |
| 11:14.03 | joobie | x-chat aqua |
| 11:14.53 | schmidts | thx |
| 11:15.37 | kaldemar | colloquy or irssi are fine also. |
| 11:16.05 | schmidts | for irc i want some colourful clicky interface ;) |
| 11:16.57 | joobie | ive tried them all |
| 11:17.01 | joobie | and for free |
| 11:17.04 | joobie | x-chat aqua |
| 11:17.13 | joobie | unless u really like ANSI type IRC consoles |
| 11:17.14 | joobie | that suck ass |
| 11:17.19 | schmidts | downloading right now and will try this ;) |
| 11:17.27 | joobie | you need to change the colours tho |
| 11:17.31 | joobie | like black background |
| 11:17.34 | joobie | stuff like that |
| 11:17.44 | joobie | it looks pretty crap by default |
| 11:17.54 | joobie | .. so u just switched to mac? |
| 11:18.20 | schmidts | nope using it since 4 years, but not using irc for a long time ;) |
| 11:18.30 | joobie | ahh.. shame on you |
| 11:18.31 | joobie | ;P |
| 11:18.41 | schmidts | :D |
| 11:18.41 | joobie | IRC is where it's happening |
| 11:18.49 | joobie | you are sitting in the social hub right now |
| 11:18.55 | joobie | veyr talkative |
| 11:19.09 | schmidts | i just read bash.org, does this count too ? ;) |
| 11:19.21 | joobie | never heard of that site |
| 11:19.24 | joobie | checks it out |
| 11:19.45 | schmidts | :D |
| 11:20.02 | joobie | http://bash.org/?5300 |
| 11:20.04 | joobie | teehee |
| 11:20.34 | schmidts | i was sure this would happening ;) |
| 11:20.36 | joobie | http://bash.org/?835030 |
| 11:20.37 | joobie | ahahha |
| 11:20.40 | joobie | that one is class |
| 11:20.45 | joobie | check it out |
| 11:21.16 | schmidts | damn you, i´ve sitting in a meeting |
| 11:21.28 | schmidts | my face looks like i try to supress a fart :D |
| 11:22.15 | joobie | haahha |
| 11:22.34 | joobie | interrupt the meeting |
| 11:22.38 | joobie | to show them that post |
| 11:22.45 | schmidts | ;) |
| 11:23.14 | joobie | so wat mac u got |
| 11:36.29 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-eweqbxjwqucnrfyh) |
| 11:40.59 | Faustov | hello, what controls the timeout to dial a number when i get a "free signal"? |
| 11:41.55 | kaldemar | Faustov: response timeout, see core show function TIMEOUT |
| 11:42.04 | schmidts | i have a normal macbook 13 " with 2,2 ghz intel core duo processor |
| 11:42.29 | Faustov | kaldemar: thanks |
| 11:42.36 | kaldemar | in dialplan though, don't know about phones that are just lifted off-hook without entering the dialplan. |
| 11:42.41 | schmidts | i had a old Powerpc powerbook before but this was real slow ;) |
| 11:44.22 | Faustov | kaldemar: what would be a good way of setting that globally for all dialplans? |
| 11:44.26 | Justman | we have two queues with similar weight, and two callers, first caller in first queue has priority=10 (for example) and second caller has priority=0, who will precessed first ? |
| 11:44.35 | Faustov | in [general]? |
| 11:44.41 | Justman | second caller in second queue |
| 11:45.03 | Justman | and one member (sip/1234) registered in both queues |
| 11:46.13 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:46.42 | kaldemar | Faustov: do you mean for all phones in whatever dialplan? |
| 11:46.47 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 11:47.00 | Faustov | kaldemar: yeah |
| 11:47.20 | Faustov | kaldemar: because I'd rather avoid having this specific per dialplan rule |
| 11:47.37 | kaldemar | Faustov: maybe setvar=TIMEOUT(response)=... in device definitions in sip.conf. |
| 11:48.47 | Faustov | kaldemar: hmm, I've only got sip accounts there, but let me try setting that for the test account |
| 11:48.51 | kaldemar | Faustov: afaik setvar won't work under [general]. |
| 11:49.25 | Faustov | k |
| 11:49.44 | Faustov | sip reload did not spit an error about setvar in the account thought |
| 11:52.08 | *** join/#asterisk fazendeiro (~chatzilla@200.249.86.178) |
| 11:53.32 | Faustov | kaldemar: didn't work there, though |
| 11:54.09 | kaldemar | Faustov: where is it supposed to work? what does your dialplan look like? |
| 11:54.30 | Faustov | w8, lemme test it in the dialplan first |
| 11:54.34 | Faustov | see if it does the job actually |
| 11:54.37 | Faustov | well lol |
| 11:54.44 | Faustov | what dialplan if I'm not even dialing a number? |
| 11:55.37 | kaldemar | that's a phone feature if you're talking about the timeout before even dialing. asterisk has no control over what happens before you dial. |
| 11:57.30 | Faustov | I see |
| 11:57.36 | Faustov | I thought it might be waiting for input |
| 11:58.27 | Faustov | with a time limit |
| 11:58.41 | kaldemar | no, a SIP phone takes no contact to asterisk before you dial a number. some phones do have a configurable timeout after which they dial a preconfigured number. |
| 12:10.36 | Faustov | think I got it |
| 12:10.49 | Faustov | spa922 has such option |
| 12:12.00 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
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| 12:45.46 | schmidts | hello |
| 12:46.08 | *** join/#asterisk robert83a2 (~robert83a@gandalf.topolauniverzal.com) |
| 12:46.48 | robert83a2 | hello, I've changed the Default Manager password in asterisk and restarted the server, now my clients can call outside but local calls between locales not possible, do I need to change some code on the phones themselves? |
| 12:47.14 | robert83a2 | amportal.conf , manager.conf |
| 12:47.23 | robert83a2 | amportal stop then amportal start |
| 12:47.43 | upb | there is no such thing as amportal in asterisk |
| 12:47.49 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:48.02 | robert83a2 | oh |
| 12:48.03 | robert83a2 | trixbox |
| 12:48.10 | robert83a2 | I'm in wrong room then? :) |
| 12:48.16 | kaldemar | ask in #trixbox or #freepbx |
| 12:48.24 | robert83a2 | thank you |
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| 13:04.10 | odenkos | why is it that when I add the options h and H into the Dial application, asterisk transmits no audio.. it is not the fault of incompatible codecs, because when I remove them from the dial application, everything works just fine.. |
| 13:04.19 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 13:12.39 | *** join/#asterisk jojo^ (jol@ph4.se) |
| 13:13.13 | jojo^ | What's the prefered way of building an asterisk 3rd party app/module? Drop the app_h324m.c into apps and run make from top directory? |
| 13:13.29 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.114.173) |
| 13:16.11 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 13:16.25 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 13:19.22 | [TK]D-Fender | odenkos: perhaps you should look at the actual call and the configs for whatever is on each end of that call. |
| 13:21.55 | dbtid | howdy... can someone point me to some docs that describe the asterisk/dahdi framework? how audio data flows through the system? |
| 13:22.03 | odenkos_ntbk | [TK]D-Fender: and what should I look for? |
| 13:22.08 | dbtid | i'm specifically looking for info about spans and channels |
| 13:22.19 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:22.23 | [TK]D-Fender | odenkosWhere are packets going, and told to go... |
| 13:23.51 | odenkos_ntbk | [TK]D-Fender: but the calls work if I remove the options.. so the packets are going to the destination they should... |
| 13:25.43 | [TK]D-Fender | odenkosWho says the call flow is the SAME? |
| 13:26.10 | odenkos_ntbk | [TK]D-Fender: I say it should be, according to all logic |
| 13:26.56 | odenkos_ntbk | but I'll check that.. |
| 13:27.39 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
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| 13:43.08 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 13:47.19 | odenkos | [TK]D-Fender, I'm clueless.. should I paste the SIP debug for you? |
| 13:47.44 | [TK]D-Fender | [09:19]<[TK]D-Fender>odenkos: perhaps you should look at the actual call and the configs for whatever is on each end of that call. |
| 13:48.48 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 13:50.21 | odenkos | [TK]D-Fender, the configs on either side of the call are identical, with the exception of ip address and name... and what do you exactly mean by "look at the call"? |
| 13:50.46 | [TK]D-Fender | odeComplete call CLI output, with SIP DEBUG cleafrly |
| 13:51.21 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-sufplimsrinzrmld) |
| 13:53.34 | hrhrhr | odenkos: i forget what does h / H do? |
| 13:54.06 | odenkos | hrhrhr, allow caller/callee to hangup using * |
| 13:54.28 | odenkos | [TK]D-Fender, looked at the CLI output, and with hH the call doesn't bridge |
| 13:54.29 | hrhrhr | cool |
| 13:55.30 | [TK]D-Fender | odenkos: go and try Tt instead of Hh and I'll bet it also doesn't work.... |
| 13:55.36 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:55.41 | odenkos | [TK]D-Fender, on to it |
| 13:56.29 | odenkos | [TK]D-Fender, I don't have transfers configured though... I'll try it anyway |
| 13:58.06 | odenkos | [TK]D-Fender, you were right, it doesn't bridge either.. what does that mean? |
| 13:59.27 | [TK]D-Fender | odenkos: Perhaps you should tell us what you're working with... |
| 14:00.08 | odenkos | [TK]D-Fender, endpoint clients? |
| 14:00.26 | [TK]D-Fender | odenkos: What phones, where are they, what networking sits between, etc... |
| 14:00.48 | [TK]D-Fender | odenkos: You don't seem very thorough in trying to debug things involved... |
| 14:01.13 | [TK]D-Fender | odenkos: Youve provided no details about your problem except "hH doesn't work" |
| 14:01.23 | anonymouz666 | anyone know what is "dahdi faxbuffer transmit policy"? |
| 14:01.24 | *** join/#asterisk coppice (~chatzilla@210.17.198.88) |
| 14:02.40 | *** join/#asterisk Diffen2 (~diffen2@81-234-228-6-no32.tbcn.telia.com) |
| 14:03.34 | odenkos | [TK]D-Fender, I'm kinda fairly non-experienced with asterisk... these phones, http://www.joyce.cz/produkt-310A521-well-3195if-ip-tel-2xsip-iax-2x16-znak-5x-prog-tl-router.html , all on a lan, no nat between nothing.. |
| 14:10.16 | nitram | anyone have a SEP file including a 7975 and a 7916 handy? |
| 14:12.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:13.06 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 14:15.39 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:20.15 | odenkos_ntbk | is going away, be back in around 4 hours |
| 14:23.43 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 14:24.36 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 14:32.19 | path | is this right? http://pastebin.com/cP5vFYy3 |
| 14:32.48 | path | Im worried about $FAXFILE |
| 14:38.00 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 14:38.49 | [TK]D-Fender | path: Don't worry ... $FAXFILE doesn't exist anywhere there. Nothng to fear from it |
| 14:39.47 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 14:41.22 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177778310.dsl.bell.ca) |
| 14:43.26 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 14:43.35 | *** join/#asterisk jim55 (~jshillida@216.158.22.254) |
| 14:45.54 | *** join/#asterisk coppice (~chatzilla@210.17.198.88) |
| 14:46.33 | Dovid | [TK]D-Fender: when did you start hanging out in #free-pbx ? |
| 14:47.12 | [TK]D-Fender | Dovid: Years ago |
| 14:50.13 | Dovid | i live and learn ;) |
| 14:50.35 | Dovid | so this means when people ask me about freepbx and have work I can send them to you ? |
| 14:51.01 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:51.12 | Qwell | Dovid: Only if you want to cure them of it. |
| 14:51.58 | Dovid | haha. |
| 14:52.30 | Dovid | I have lots of people that set it up, have no idea what's going on and when they get hacked they come crying |
| 14:58.43 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 14:59.28 | golikwid|mac | is it hard to upgrade from asterisk 1.4 ro 1.6 |
| 14:59.53 | bougyman | it's easier than robbing a bank. |
| 15:00.21 | bougyman | harder than tic-tac-toe |
| 15:00.28 | golikwid|mac | hm |
| 15:00.35 | golikwid|mac | which bank? |
| 15:00.59 | golikwid|mac | i suppose its harder than yum update |
| 15:01.00 | golikwid|mac | ;) |
| 15:01.05 | golikwid|mac | or upgrade rather |
| 15:01.42 | [TK]D-Fender | [10:52]<Dovid>I have lots of people that set it up, have no idea what's going on and when they get hacked they come crying <- not a FreePBX problem... a general idiot administrator problem... |
| 15:01.58 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:02.11 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 15:03.42 | doolittlework | hi there i am trying out something about the CDR stuff, is there a way to get the billsec cdr into a variable after the call has been hanged up? |
| 15:04.46 | Dovid | [TK]D-Fender: More like an ID-10-T error. they think it's simple and try to make a quick buck till they have an issue |
| 15:05.38 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:08.04 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:08.12 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:08.54 | path | [TK]D-Fender: that means its right ? |
| 15:09.48 | [TK]D-Fender | patThat means what you referred to does not exist in that pastebin. And you do have code errors in there... |
| 15:10.07 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:12.20 | doolittlework | does the cdr that generated in the Master.csv file only get genereated when the call has ended? |
| 15:12.54 | *** join/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 15:13.57 | doolittlework | another question if i may, can one determin the end of a call in agi and then the variable of bill sec and then pass it on to exteral applications? |
| 15:14.55 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 15:20.00 | dmz | anyone know if this is in the current released *??: https://issues.asterisk.org/view.php?id=6119 |
| 15:21.21 | Katty | throws things. |
| 15:21.26 | thehar | catches them |
| 15:21.50 | malcolmd | dmz: that's a negative |
| 15:21.58 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:22.51 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:23.10 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 15:24.34 | *** join/#asterisk mducharme (~nothing@S0106000e0cc03cff.wp.shawcable.net) |
| 15:27.27 | dmz | so if i want to integrate agent signin/signout from phone i'll have to patch and build myself? |
| 15:28.20 | *** join/#asterisk Diffen2 (~diffen2@81-234-228-6-no32.tbcn.telia.com) |
| 15:28.39 | [TK]D-Fender | dmz: via THAT method, yes |
| 15:29.23 | dmz | any hints on other methods :) |
| 15:29.34 | dmz | it was just the 1st hit on google search :) |
| 15:30.32 | [TK]D-Fender | dzmYou know that logging in & out is an ASTERISK thing right? With like... dialplan apps & stuff? And things you can call via otehr triggers you can set up many phones to send... |
| 15:32.27 | *** join/#asterisk garymc (~chatzilla@host81-148-79-26.in-addr.btopenworld.com) |
| 15:34.29 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 15:37.14 | dmz | but i want a soft button i can hit that shows "logged in/logged out" :) |
| 15:38.01 | dmz | it's a computer w/microbrowser, why not take advantage of it; plus my brother & other non-techs in the co don't like to remember dialing details; they want to hit a button :) |
| 15:38.28 | Katty | stitches the Staple's Easy Button to dmz's jacket |
| 15:38.44 | dmz | if it was only that easy |
| 15:39.07 | Katty | it can be. |
| 15:39.20 | Katty | hire it done |
| 15:39.41 | Katty | has anyone tried crest whitening strips? |
| 15:39.46 | Katty | that stuff actually work? |
| 15:40.01 | dmz | heh |
| 15:40.14 | dmz | but why would i hire someone if i can figure it out |
| 15:40.14 | Katty | surely someone knows. |
| 15:40.35 | dmz | that's why i ask here; usually someone here has constructive criticism (or even sometimes creative insights!) |
| 15:40.50 | *** join/#asterisk CunningPike (~CunningPi@204.239.12.183) |
| 15:40.56 | Katty | support local economy. |
| 15:41.06 | Katty | CunningPike: have you used crest whitening strips? |
| 15:41.24 | [TK]D-Fender | [11:37]<dmz>but i want a soft button i can hit that shows "logged in/logged out" :) <-- go setup a line-key speed-dial + presence |
| 15:41.47 | *** join/#asterisk flea (~flea@unaffiliated/flea) |
| 15:41.48 | dmz | thx |
| 15:42.00 | dmz | hmm flea..i know a flea elsewhere but that's on efnet |
| 15:42.05 | [TK]D-Fender | dmz: As for the microbrowser I have live queue stats for 2 queues, 2 VM boxes, and 4 agents login/pause/call status on mine |
| 15:42.24 | flea | not the same, i stayed away from efnet and been here since opn |
| 15:42.39 | dmz | D-Fender fun! |
| 15:43.41 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:45.28 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 15:46.49 | doolittlework | quick dum question sorry not close to a system to test( if i wan tto cut the first 3 digits off the number 0123456789 is ${EXTEN:0:3} coorent should give me 012 ???// |
| 15:46.57 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 15:48.40 | [TK]D-Fender | doolittlework: If the number you want is the EXTEN |
| 15:50.16 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:50.45 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 15:54.57 | *** join/#asterisk rocknfra (~rocknfra@host63-229-dynamic.17-87-r.retail.telecomitalia.it) |
| 15:59.41 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 16:00.00 | rocknfra | I'm going mad.. Configured nat in sip_nat.conf, configured nat=yes e canreinvite=no on device options of each extension... no audio on calls... rtp debug shows packets being sent to the local ip of the destination and not to its public ip address.. |
| 16:00.02 | Micc_ | How can I find out who the underlying carrier is that owns one of my DIDs? I'm tired of getting the run around from voip innovations. |
| 16:01.12 | *** join/#asterisk korcan (~johnynum5@67.216.31.34) |
| 16:02.43 | Micc_ | I guess I'll have to find out later. gotta go |
| 16:03.17 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:4c:62d9:2ac0:762d) |
| 16:03.20 | cusco | hi... |
| 16:04.25 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 16:06.31 | drmessano | rocknfra: Pastebin your sip_nat.conf |
| 16:07.21 | *** join/#asterisk oryxtec (oryxtec@119.152.221.163) |
| 16:07.55 | cusco | I can pass variables with SipAddHeader from one asterisk to another.... |
| 16:08.08 | cusco | but how can the second asterisk pass back a new value to the first one? |
| 16:09.31 | *** join/#asterisk ariel_ (~chatzilla@93-62-188-210.ip23.fastwebnet.it) |
| 16:10.02 | oryxtec | hello every one.. |
| 16:10.20 | oryxtec | plz tell me how is this CPU to run asterisk Intel Xeon DP E5405, 1x Quadcore |
| 16:10.27 | oryxtec | with 4 GB Ram |
| 16:11.00 | cusco | fine, I run it with less resources |
| 16:11.02 | cusco | :) |
| 16:11.17 | rocknfra | drmessano: http://pastebin.com/dqZZPcTn |
| 16:11.22 | oryxtec | humm |
| 16:11.32 | cusco | that depends on your needs |
| 16:11.33 | bougyman | oryxtec: what kind of traffix are you doing? |
| 16:11.36 | bougyman | also, are you recording? |
| 16:11.45 | cusco | there is a page adressing those hardware requirements on the wiki oryxtec |
| 16:11.50 | oryxtec | i m thinking to run virtualization on this server and then asterisk |
| 16:12.03 | bougyman | virtualization is a whole other problem. |
| 16:12.09 | cusco | ouch |
| 16:12.11 | bougyman | don't do it. |
| 16:12.18 | oryxtec | how is KVM then? |
| 16:12.27 | bougyman | it's also virtualization. |
| 16:12.34 | bougyman | what's wrong with dedicated hardware? |
| 16:12.54 | drmessano | rocknfra: Your hostname is going to resolve to the local IP when you perform a DNS lookup.. Pick another external hostname that resolves to the external IP or use externip |
| 16:12.59 | cusco | he's got a very powerfull machine, but wants to take more out of it |
| 16:13.01 | bougyman | there's a reason avaya and cisco and the other big boys haven't released a virtualized phone switch, y'know.... |
| 16:13.06 | oryxtec | i want to ve more thn one server on one dedicated machine for my clients |
| 16:13.11 | cusco | should have gotten two half-good machines then :P |
| 16:13.20 | bougyman | only Mitel has that solution, and it sucks ass (Mitel communicator in VMWare) |
| 16:13.36 | drmessano | rocknfra: Think about it.. your machine name is the same as the host you're trying to resolve external. When you PING it, what address do you think Asterisk will return? |
| 16:13.40 | bougyman | the software is free, which frees up room to pay for dedicated, good hardware. |
| 16:13.44 | bougyman | that's my view on it. |
| 16:13.58 | rocknfra | mmm.. you are right |
| 16:14.41 | rocknfra | but.. I even tried nslookup |
| 16:15.03 | rocknfra | but I think that the server doesn't even query the dns server because he knows its own ip... |
| 16:15.06 | Qwell | ~sox |
| 16:15.06 | infobot | well, sox is Sound Processing Tool. URL: http://sox.sourceforge.net/ |
| 16:15.10 | Qwell | ~wav |
| 16:15.10 | infobot | wavplay will play .wav, apt-get install wavtools. xmms also supports .wav files |
| 16:15.13 | oryxtec | Mitel communicator in VMWare .. is it linux based? |
| 16:15.25 | Qwell | quick, somebody give me a sox command to make an asterisk-compat wav file from something |
| 16:15.27 | oryxtec | wht is your feedback on promox |
| 16:15.28 | bougyman | oryxtec: as if mitel would say so. |
| 16:15.28 | oryxtec | ? |
| 16:15.29 | rocknfra | in fact the nslookup resolve to te public address but a ping is done to 127.0.0.1 |
| 16:15.31 | drmessano | rocknfra: correct.. use another externhost or use externip instead |
| 16:15.35 | bougyman | Qwell: 2 secs... |
| 16:15.55 | *** join/#asterisk imox1234 (~imox1234@p4FC5C531.dip0.t-ipconnect.de) |
| 16:15.58 | Qwell | bougyman: was hoping the bot would know. fail. |
| 16:16.01 | rocknfra | I put the dyndns host that actually is the host pointed from the server host |
| 16:16.05 | cusco | Qwell: convert what? to wav or from wav? |
| 16:16.05 | rocknfra | gonna try right now |
| 16:16.09 | Qwell | to wav |
| 16:16.13 | oryxtec | let me see Mitel communicator in VMWare on google |
| 16:16.17 | oryxtec | i will b back |
| 16:16.36 | cusco | Qwell: asterisk -vrx "file convert /bla/file.ogg /bla/file.wav" |
| 16:16.38 | cusco | :p |
| 16:16.55 | Qwell | cusco: yeah, I'm adding a new format. Using Asterisk to convert to said new format kinda defeats the purpose. ;p |
| 16:17.04 | cusco | ow |
| 16:17.14 | cusco | so whats wrong? |
| 16:17.28 | Qwell | need to do it with something else, so I can verify that Asterisk is reading/writing it correctly. |
| 16:17.28 | cusco | sox -c 1 -r 8000 file.bla file.wav |
| 16:17.34 | Qwell | that's it? |
| 16:17.40 | bougyman | we just do sox -c1 -r8000 infile outfile.wav |
| 16:17.42 | bougyman | yeah, same thing |
| 16:18.05 | drmessano | Qwell: Adding WMA? Thank goodness |
| 16:18.08 | cusco | only one channel, there is no stereo in voip yet :p |
| 16:18.11 | Qwell | drmessano: yep, exactly |
| 16:18.13 | cusco | or there is.. just not used |
| 16:18.14 | bougyman | oryxtec: it's a 7 digit pricetag on mitel communicator. |
| 16:18.35 | Qwell | stupid sox doesn't know about sln16 |
| 16:18.40 | cusco | hahaha |
| 16:18.42 | rocknfra | drmessano, same problem : ( http://pastebin.com/MWmrMmqB |
| 16:18.43 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 16:19.00 | cusco | yea, my sox don't know about ordinary sln |
| 16:19.16 | rocknfra | I even rebooted the machine |
| 16:19.18 | bougyman | hrm, mine does. |
| 16:19.29 | Qwell | bougyman: as a filetype? |
| 16:19.36 | cusco | -t, --type FILE-TYPE |
| 16:19.42 | bougyman | sox input.foo -r 16000 -c 1 -t sw output.sln16 |
| 16:19.46 | bougyman | yeah. |
| 16:19.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:19.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:20.04 | cusco | See soxformat(7) for a list of supported file types. |
| 16:20.16 | cusco | sw? |
| 16:20.38 | drmessano | rocknfra: Where is your canreinvite=no ? |
| 16:20.59 | rocknfra | in the device options of each extensions |
| 16:21.08 | Qwell | cusco: s16 |
| 16:21.12 | rocknfra | and I put it also in the PEER DETAILS of the trunk even if I think it's useless there |
| 16:21.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 16:21.34 | bougyman | ah, but i installed libsox-fmt-all (debuntu package) |
| 16:21.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 16:21.45 | cusco | yes |
| 16:21.47 | cusco | Deprecated aliases |
| 16:21.50 | cusco | its s16 now |
| 16:21.54 | drmessano | rocknfra: Needs to be in that same file as well, so it's in the [general] section |
| 16:21.56 | cusco | didn't know about that |
| 16:22.05 | cusco | but asterisk reads .sln files |
| 16:22.30 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 16:22.34 | rocknfra | drmessano, sorry what you mean for same file ? |
| 16:22.59 | rocknfra | drmessano, sip.conf ? |
| 16:23.06 | cusco | bougyman: did install libsox-fmt-all too, tho proud to say Debian package :P |
| 16:24.00 | drmessano | rocknfra: SIP_NAT.CONF |
| 16:24.10 | drmessano | canreinvite=no |
| 16:24.20 | drmessano | ~sipnat |
| 16:24.20 | infobot | extra, extra, read all about it, sipnat is Quick guide on configuring Asterisk + 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 |
| 16:24.24 | bougyman | man soxformat shows s16 supported here. |
| 16:26.40 | rocknfra | drmessano, added canreinvite=no and restarted asterisk, same thing.. |
| 16:29.04 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 16:29.05 | bmoraca_work | fail2ban is frickin awesome |
| 16:29.10 | drmessano | rocknfra: Paste your debug |
| 16:29.18 | rocknfra | ok |
| 16:29.21 | rocknfra | just a while |
| 16:29.26 | b14ck | sup guys |
| 16:33.31 | *** join/#asterisk tiny_D (~tiny_D@e178233242.adsl.alicedsl.de) |
| 16:36.56 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 16:38.29 | rocknfra | I've got the log in the server machine but I don't know how to transfer it.. |
| 16:40.43 | p3nguin | pastebinit |
| 16:41.18 | p3nguin | Or you can use any of several file transfer protocols to get it to your local machine. |
| 16:41.20 | *** join/#asterisk radic (~radic@dslb-094-216-232-212.pools.arcor-ip.net) |
| 16:41.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:42.04 | rocknfra | ok istalled smb-client |
| 16:42.42 | p3nguin | That wasn't really what I had in mind, but if it works for you, I guess that's what matters. |
| 16:44.43 | p3nguin | scp/sftp is usually the fastest way to copy a file from one computer to another computer and does not require configuration of an ftpd. |
| 16:45.18 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
| 16:45.26 | *** join/#asterisk golikwid|mac_ (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 16:45.49 | rocknfra | drmessano, p3nguin http://pastebin.com/tY3Xy1b0 |
| 16:45.51 | [TK]D-Fender | Most default installations of ProFTPD don't need any configuration to be usable |
| 16:45.54 | rocknfra | sorry for the time.. |
| 16:45.57 | [TK]D-Fender | (out o fthe box) |
| 16:46.32 | p3nguin | It does require installation and enabling, though. Not everyone HAS it by default. Everyone has ssh/sshd and the associated tools. |
| 16:47.30 | doolittlework | could the guru please have a look at this and give some advice why its not working or what i can do to make it better, thx |
| 16:47.30 | rocknfra | Yes, I usually use ssh to transfer files but on this pc didn't have any tool... but got a network drive so found quicker to install samba-client |
| 16:47.36 | doolittlework | could the guru please have a look at this and give some advice why its not working or what i can do to make it better, thx http://pastebin.org/799349 |
| 16:49.41 | p3nguin | rocknfra: That's ONE HUGE pastebin. What exactly am I supposed to be looking for? |
| 16:50.18 | chuckf | p3nguin: his problem |
| 16:50.30 | p3nguin | chuckf: And that would be? |
| 16:50.52 | rocknfra | I don't know which part of the log drmessano wanted to take a look at.. |
| 16:50.54 | chuckf | that's the game. He shows you the output, you guess the problem and solution |
| 16:51.01 | p3nguin | /: |
| 16:51.03 | rocknfra | I explain you the problem |
| 16:52.04 | rocknfra | here you can see my sip_nat.conf http://pastebin.com/3RFzQ5z9 |
| 16:52.24 | rocknfra | When I make calls no sounds passes... |
| 16:52.38 | rocknfra | firewall and forwarding is ok |
| 16:52.47 | rocknfra | the sip_nat.conf seems ok |
| 16:53.04 | rocknfra | on every device options I've set nat=yes and canreinvite=no |
| 16:53.10 | doolittlework | sorry to waste your time im just a dumbass |
| 16:53.32 | doolittlework | solved uploaded the wrong file to my server,,lol,,long day |
| 16:53.57 | rocknfra | but rtp packets goes to the local address of the remote person, not it's public address, as you can see from the first lines of the first pastebin |
| 16:54.35 | p3nguin | I experienced a similar problem when I used a Cisco router. The RTP packets contained the private IP address rather than the public one needed to be able to communicate. No one here was able to help solve the problem. I had to use a different router. |
| 16:55.00 | rocknfra | I've got a wrt54gl |
| 16:55.04 | rocknfra | tried with ddwrt and openwrt |
| 16:55.27 | rocknfra | also tried different versions of openwrt |
| 16:55.52 | rocknfra | so I guess it is like I've tried with two different routers.. |
| 16:56.04 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 16:56.17 | kaldemar | rocknfra: is the peer behind a NAT? |
| 16:56.26 | rocknfra | yes |
| 16:56.36 | p3nguin | Since no one here knows how to overcome the problem, consider another BRAND of router. |
| 16:56.59 | kaldemar | rocknfra: did you configure it as a device behind a NAT in sip.conf? |
| 16:57.23 | p3nguin | It's an RTP issue. |
| 16:57.38 | rocknfra | if you're asking if I put nat=yes and canreinvite=no on the device details of the peer, yes, I've done |
| 16:58.11 | Naikrovek | Katty: glad to hear it\ |
| 16:59.08 | rocknfra | it sounds to me so strange... |
| 16:59.30 | kaldemar | wrt54gl shouldn't be a problem. |
| 16:59.35 | rocknfra | why a router shouldn't nat correctly rtp packets rather than any other packet |
| 16:59.46 | rocknfra | why any other service works ? |
| 16:59.54 | kaldemar | feel free to pastebin the relevant configurations. |
| 16:59.58 | [TK]D-Fender | Why aren't you REALLY LOOKING at the call? |
| 17:00.06 | p3nguin | I wouldn't have expected a Cisco 831 to be a problem either, but getting rid of it solved my RTP problem. |
| 17:00.22 | kaldemar | and disable rtp debug, enable sip debug and pastebin the result of a call. |
| 17:00.28 | rocknfra | sorry, [TK]D-Fender what you mean ? |
| 17:00.41 | p3nguin | The "call" probably works fine. Mine did. |
| 17:00.54 | p3nguin | The sip debug shows that everything is fine in the signaling. |
| 17:01.00 | [TK]D-Fender | rocknfra: where is the PASTEBIN with complete SIP DEBUG for the failed call? |
| 17:01.13 | p3nguin | It's the RTP that doesn't flow because the RTP packets contain the wrong address in them. |
| 17:01.15 | kaldemar | p3nguin: we haven't seen any sip debug for his call yet. |
| 17:01.20 | rocknfra | [TK]D-Fender, kaldemar ok coming |
| 17:02.49 | *** join/#asterisk tiny_D (~tiny_D@e178233242.adsl.alicedsl.de) |
| 17:08.41 | Katty | omnomnomnomlunchnomnom |
| 17:08.46 | Kobaz | cross post from #-dev... in case any of you guys know |
| 17:08.55 | Kobaz | i have an asterisk thread that sucking up a bunch of cpu, how do i find out what the actual thread is? |
| 17:09.07 | Kobaz | it's chewing 47.9cpu and it's been going for 8ish hours, so i'm wondering how to map the thread id in core show threads, so some sort of thread info i can get from ps |
| 17:09.18 | Katty | hi Kobaz |
| 17:09.28 | rocknfra | sorry for the time |
| 17:09.33 | rocknfra | it's a 1/2mb file.. |
| 17:09.42 | Kobaz | hey Katty |
| 17:09.42 | rocknfra | which part should I paste ? |
| 17:10.35 | rocknfra | also... until now I was making call to a friend that now is not at home, so I took my mobile phone, disable wireless connection and activated only 3g |
| 17:10.52 | rocknfra | registered to my server with 3cx and used it to make the call for the debug |
| 17:11.06 | rocknfra | and I were able to hear the audio from both side... |
| 17:11.26 | [TK]D-Fender | [13:00]<[TK]D-Fender>rocknfra: where is the PASTEBIN with complete SIP DEBUG for the failed call? <------------------ |
| 17:11.46 | rocknfra | its 512 kb file... |
| 17:11.54 | rocknfra | should I past it all ? |
| 17:12.16 | p3nguin | That's a huge call. |
| 17:12.23 | Katty | sounds like a rough day in here |
| 17:12.28 | rocknfra | is there a logfile with only sip debug ? |
| 17:12.32 | rocknfra | I downloaded the full log fikle |
| 17:12.34 | rocknfra | file* |
| 17:12.48 | [TK]D-Fender | * CLI w/ SIP DEBUG. |
| 17:12.52 | [TK]D-Fender | PASTEBIN |
| 17:12.54 | rocknfra | ok |
| 17:12.54 | p3nguin | Yes, it's called sip set debug. |
| 17:13.03 | rocknfra | yes I enabled it |
| 17:13.05 | rocknfra | rasterisk |
| 17:13.10 | rocknfra | sip set debug on |
| 17:13.20 | rocknfra | but I thought that that log were into a file... |
| 17:13.25 | p3nguin | If you're using PuTTY to ssh to your server, make sure you increase the scroll buffer before you begin. |
| 17:13.38 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 17:14.16 | kaldemar | rocknfra: it logs the call into the CLI, not a file. |
| 17:15.55 | rocknfra | http://pastebin.com/vfq4qCiZ |
| 17:16.41 | [TK]D-Fender | rocknfra: COMPLETE CALL. |
| 17:16.54 | [TK]D-Fender | rocknfra: The ENTIRE thing from the very first packet |
| 17:17.29 | oryxtec | guys which CPU is best Intel Xeon DP E5405 Quadcore or 2.8GHz Dual Core?? |
| 17:17.50 | Qwell | oryxtec: Which one is more expensive? |
| 17:18.00 | oryxtec | both are same |
| 17:18.08 | oryxtec | price at hosting company |
| 17:18.28 | p3nguin | The hosting company is giving you a price to USE the hardware. |
| 17:18.47 | p3nguin | Find the retail price or the cost of the hardware. |
| 17:18.48 | rocknfra | did rasterisk > /tmp/siplog this time |
| 17:18.50 | rocknfra | http://pastebin.com/sA142RNq |
| 17:19.22 | oryxtec | yes |
| 17:19.32 | oryxtec | correct ... i ve to pay rent |
| 17:20.28 | heffer | tzafrir: are you working on porting your dahdi-extras to 2.4.0? otherwise i'd give a shot at trying to find out what changed and look what i can come up with without much c skills:D |
| 17:20.59 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn216.78-98-74.t-com.sk) |
| 17:21.03 | CunningPike | Katty: You can tell from there!!!??!! Wow - my teeth must be bright ;-) |
| 17:21.45 | tzafrir | I should do it today |
| 17:21.55 | heffer | tzafrir: wow that's great news! thanks a lot |
| 17:24.28 | *** join/#asterisk oryxtec (oryxtec@119.152.221.163) |
| 17:25.05 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:27.08 | Katty | who what huh? |
| 17:27.15 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 17:27.57 | oryxtec | guys how we can check to hosted company network quality. If i will run asterisk on any hosted company there will not be any voice issues? |
| 17:28.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.82) |
| 17:30.39 | anonymouz666 | oryxtec: testing |
| 17:30.47 | oryxtec | how? |
| 17:30.48 | bougyman | oryxtec: SLA |
| 17:30.53 | oryxtec | i can test it |
| 17:30.56 | oryxtec | SLA? |
| 17:30.59 | bougyman | yes. |
| 17:31.03 | bougyman | service-level agreement. |
| 17:31.13 | bougyman | ours has exact specs on latency and packet delivery and such. |
| 17:31.29 | bougyman | if 10 packets don't get delivered, they owe us X amount of dollars... |
| 17:31.40 | bougyman | and X amount of dollars for anything over 10ms pings, etc. |
| 17:31.52 | bougyman | the good companies will have a rock-solid SLA |
| 17:32.38 | anonymouz666 | lots of people use 'ping' to test the network quality... |
| 17:32.52 | Kobaz | ping is worthless |
| 17:33.21 | bougyman | they _could_ theoretically still suck with a good SLA, but 1: you get paid when they do and 2: they allow an opt-out after N amount of SLA underperformance. |
| 17:34.06 | bougyman | if you ask for their SLA and they say "What SLA?" keep seeking. |
| 17:34.28 | oryxtec | humm |
| 17:34.35 | oryxtec | let me googld it i will be back |
| 17:34.41 | oryxtec | SKA |
| 17:34.42 | oryxtec | SLA |
| 17:39.24 | p3nguin | Heh, if the company does not know what an SLA is, they're in the wrong business. |
| 17:39.35 | rocknfra | nothing about my log ? |
| 17:39.41 | oryxtec | Kobaz: i wana quick test one IP ... how can i test it through SLA plz can you guide me in a easy way thanks |
| 17:39.46 | oryxtec | :( |
| 17:39.59 | bougyman | SLA is just a contract. |
| 17:40.03 | p3nguin | sheesh |
| 17:40.44 | Kobaz | oryxtec: run mta over a period of a few weeks |
| 17:40.51 | Kobaz | er... mtr rather |
| 17:41.08 | [TK]D-Fender | ~sla |
| 17:41.09 | infobot | sla is probably service level agreement, or shared line appearances |
| 17:41.17 | Kobaz | ~mtr |
| 17:41.18 | infobot | Network diagnostic tool. URL: http://www.bitwizard.nl/mtr/ |
| 17:41.35 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 17:41.44 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 17:41.45 | bougyman | http://www.rackspace.com/managed_hosting/support/servicelevels/intensivesla.php |
| 17:41.52 | bougyman | something that covers stuff like that. |
| 17:42.03 | bougyman | of course you usually need to sign an NDA to get the actual SLA |
| 17:42.58 | [TK]D-Fender | bougyman: Which probably says that if you ditch them for the fact they suck you won't be allowed to tell anyone ;) |
| 17:43.22 | bougyman | [TK]D-Fender: i've never seen that in one. |
| 17:43.50 | p3nguin | That's what NDA means. |
| 17:44.08 | bougyman | non-disclosure just means you can't disclose their proprietary information/ |
| 17:44.20 | bougyman | "They sucked so I fired them" is not proprietary information. |
| 17:44.29 | KavanS | bougyman, uhm, NDA can mean anything the clause means... |
| 17:44.33 | KavanS | NDA is just the general term |
| 17:44.48 | KavanS | in the NDA the "scope" of what information can vary |
| 17:44.53 | bougyman | sure, an NDA can cover anything the two parties agree upon. |
| 17:45.01 | cmendes0101 | I asked this yesterday but hoping someone is online that might know about this. I'm building asterisk from SVN for chan_rtmp support. When I run "make" on asterisk I get these errors http://pastebin.com/KivfjxcW |
| 17:45.10 | bougyman | but the general one you see convers proprietary information from one or both parties. |
| 17:45.12 | [TK]D-Fender | YOU WILL NOT DISCLOSE HOW MUCH WE SUCK WHEN YOU CURB US! |
| 17:45.16 | KavanS | I'm sure personal assistants of celebrities have to sign something like that ;) |
| 17:45.29 | KavanS | i.e. you can never tell anyone the horrifying shit you see here |
| 17:46.03 | tzanger | hello from .aw! |
| 17:46.19 | p3nguin | Hello, A&W! |
| 17:47.53 | tzanger | heh not quite :-) |
| 17:48.19 | Qwell | tzanger: Aruba? |
| 17:49.20 | *** join/#asterisk bcrisp (~bcrisp@70.102.242.138) |
| 17:50.44 | tzanger | Qwell: yep |
| 17:51.20 | bcrisp | anyone have helpful tips for troubleshooting garbled audio that occurs intermittently using softphone in asterisk? I know this could be due to a large variety of potential issues / configurations, but thought I'd ask anyway. Internet connectivity does not appear to be a problem, and we're using ulaw codec |
| 17:51.21 | Qwell | I appear to have misplaced my ticket. |
| 17:52.09 | bougyman | bcrisp: can you try something for me (i've had this issue creep up a lot)? |
| 17:52.19 | bougyman | bcrisp: next time you get one, have them transfer the call to themselves. |
| 17:52.26 | bougyman | when this happens to us that corrects the issue. |
| 17:52.38 | bcrisp | what codec(s) do you support? |
| 17:53.03 | bougyman | turning canreinvite=no on the sip.conf helped, but didn't completely alleviate it. |
| 17:53.03 | bougyman | bcrisp: only ulaw |
| 17:53.07 | bougyman | when we transfer the call to ourselves, though, the garble goes away |
| 17:53.11 | bougyman | it's the strangest thing. |
| 17:53.16 | bougyman | though we've only had it with aastras |
| 17:53.33 | bougyman | and it's not just on asterisk... same with yate, fs, asterisk, openser + rtpproxy |
| 17:53.39 | bcrisp | its only happening when we bridge call between softphone client and outside number (inbound or outbound has this issue) |
| 17:54.23 | bcrisp | whats interesting is that i was on a call that i received from a phone in canada into my softphone in the office - i could hear the other party perfectly but they were complaining about digitized sound |
| 17:54.26 | bcrisp | coming from my end |
| 17:54.43 | bcrisp | and cutting out etc |
| 17:57.03 | bcrisp | we have a separate in bound and out bound SIP provider .. maybe i need to start traceroutes from the asterisk box to the provider's servers |
| 17:57.50 | anonymouz666 | deep purple 1974 - mistreated (california jam) |
| 17:57.51 | anonymouz666 | very nice song |
| 17:58.05 | bougyman | bcrisp: yes, this sounds like the exact smae issue. |
| 17:58.08 | anonymouz666 | I'll put as my MOH |
| 17:58.11 | bougyman | provider didn't matter, even happened on PRI |
| 17:58.17 | anonymouz666 | (just to be ontopic heh) |
| 18:03.12 | *** join/#asterisk mpe (~mpe@65.161.206.209) |
| 18:03.20 | bcrisp | bougyman: what did you see in pri? |
| 18:03.36 | bcrisp | i mean... did you see anything odd in the debug? |
| 18:03.37 | bougyman | nothing, we couldn't capture the 'garble' |
| 18:03.38 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:03.50 | bcrisp | im having strange CDR errors that recently started showing up |
| 18:04.06 | bcrisp | errors thrown here could possibly interrupt the speed of the system i wonder |
| 18:04.18 | bougyman | Do You Want To Install Proprietary blah blah. |
| 18:04.22 | bougyman | just say yes. |
| 18:04.48 | bcrisp | can debug / verbosity settings have a significant impact on the system? |
| 18:08.43 | bougyman | yes |
| 18:08.45 | bougyman | always |
| 18:08.51 | bougyman | when you measure something, you affect it. |
| 18:09.00 | bougyman | that's Heisenberg, can't be gotten around. |
| 18:09.10 | *** join/#asterisk mducharme-laptop (cebc7904@gateway/web/freenode/ip.206.188.121.4) |
| 18:09.18 | mducharme-laptop | hey leifmadsen |
| 18:09.31 | p3nguin | Heisenberg... you just reminded me of how long I have to wait for another season of Breaking Bad. |
| 18:09.32 | mducharme-laptop | I got the snmp going |
| 18:09.34 | mducharme-laptop | thanks for the document |
| 18:09.46 | mducharme-laptop | I do have some feedback |
| 18:10.05 | bougyman | never seen breaking bad. |
| 18:10.15 | bougyman | they have something about the Uncertainty Principle in it? |
| 18:10.40 | p3nguin | The main character takes on a "business name" alias of Heisenberg. |
| 18:11.19 | p3nguin | For the obvious reasons, of course. |
| 18:11.49 | doolittlework | dumb question time: how many queries can asterisk send to mysql before making an impact on performance? |
| 18:12.05 | Naikrovek | a lot, if you have decent hardware |
| 18:12.13 | p3nguin | http://blogs.amctv.com/breaking-bad/talk/2009/03/heisenberg.php |
| 18:12.38 | doolittlework | Naikrovek:how many is alot say on a core 2 quad? |
| 18:13.10 | Naikrovek | doolittlework: i don't know, but I used to do 100 inserts per second on a P3 and load was never an issue at all. |
| 18:13.26 | Naikrovek | never got above .3 as i recall, but it was a while ago |
| 18:13.29 | doolittlework | Naikrovek: thanks that puts me at ease |
| 18:13.53 | Naikrovek | doolittlework: remember, your db engine matters, your table structure matters, and all of that |
| 18:13.55 | Naikrovek | but i think you'll be fine |
| 18:14.01 | Naikrovek | mysql is plenty fast |
| 18:14.08 | doolittlework | ta |
| 18:14.15 | Naikrovek | haven't heard 'ta' in a while |
| 18:14.15 | bcrisp | hmm - I set up cdr for mysql a while ago - noticed some errors relating to a missing field, 'lastms' - where can i get an updated list of required fields necessary for recording cdr info? |
| 18:19.52 | *** join/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 18:20.37 | *** join/#asterisk mpe (~mpe@65.161.206.209) |
| 18:20.46 | leifmadsen | mducharme-laptop: awesome! any feedback is appreciated |
| 18:20.49 | leifmadsen | glad you got it working |
| 18:21.01 | *** join/#asterisk AlHafoudh_ (~AlHafoudh@adsl-dyn110.78-98-89.t-com.sk) |
| 18:21.50 | mducharme-laptop | I have some feedback on both the document, but mainly the actual way snmp is implemented is less than ideal |
| 18:22.01 | mducharme-laptop | and limits the monitoring options available |
| 18:22.58 | mducharme-laptop | I've gone through the snmp schema for asterisk and it's not as useful as it could be |
| 18:25.59 | doolittlework | here i go again, sorry my connection droped erlier, so i dont know if anyone replied,, i have to get the duration data from the call after it ends, without hanging up the call, can this be done with standaard dialplan or must i involve uncle agi? |
| 18:26.01 | dbtid | howdy... can someone point me to some docs that describe the asterisk/dahdi framework? how audio data flows through the system? |
| 18:26.49 | leifmadsen | mducharme-laptop: gotcha -- there probably won't be much I can do about the actual SNMP schema unless you wish to provide a patch :) |
| 18:27.44 | mducharme-laptop | leifmadsen heh |
| 18:28.05 | mducharme-laptop | leifmadsen well there are two main things.. there are not enough performance counters for things people would care about |
| 18:28.13 | mducharme-laptop | you can graph the number of active channels but that's about it |
| 18:28.19 | leifmadsen | mducharme-laptop: agreed |
| 18:28.48 | mducharme-laptop | and the other thing is it's bad design to have an "active channels table", it's nearly useless |
| 18:28.52 | mducharme-laptop | b/c it will constantly change |
| 18:29.35 | mducharme-laptop | if you try to graph a channel from it you will end up always graphing some different random channel and sometimes graphing nothing |
| 18:29.47 | mducharme-laptop | because the oids keep changing on the fly as channels become active and deactive |
| 18:30.16 | mducharme-laptop | it makes no sense |
| 18:31.05 | mducharme-laptop | there are also lots of strings in there, like you can call up the language settings for things |
| 18:31.18 | mducharme-laptop | I have no idea why you would want to pull that to begin with |
| 18:31.53 | mducharme-laptop | thus most of the stuff available in snmp is useless trivial data that you would not care about monitoring to begin with |
| 18:32.02 | mducharme-laptop | no actual important data is there |
| 18:32.08 | leifmadsen | mducharme-laptop: what you are talking about it sounds like issues with Asterisk itself, and not so much the documentation we wrote -- we can only write about how things are implemented in Asterisk |
| 18:32.27 | mducharme-laptop | right, I was just talking about that first |
| 18:33.06 | mducharme-laptop | for me I had to restart the entire server to get it all working |
| 18:33.18 | mducharme-laptop | loading the module, restarting the snmp module etc did not get it going, restarting asterisk did not get it going |
| 18:33.26 | mducharme-laptop | I rebooted the server and it magically started working |
| 18:34.23 | leifmadsen | weird... |
| 18:34.34 | leifmadsen | maybe it had to do with the SNMP service? |
| 18:34.38 | leifmadsen | snmpd |
| 18:36.09 | mducharme-laptop | I had restarted snmpd, I know how to configure snmpd.. I use it on many servers |
| 18:36.20 | mducharme-laptop | I've used net-snmp for a long time |
| 18:36.55 | mducharme-laptop | it magically started working on reboot even though I had restarted all modules etc.. I cannot explain it |
| 18:37.24 | mducharme-laptop | I had loaded the module and told asterisk reload when convenient.. and it reloaded.. and the module was loaded, but it wasn't working |
| 18:37.35 | mducharme-laptop | perhaps /etc/init.d/asterisk restart might do something I do not know |
| 18:37.50 | leifmadsen | odd |
| 18:37.50 | mducharme-laptop | the other thing is I don't know whether I would suggest redirecting the sysobjectid |
| 18:38.10 | leifmadsen | you're probably better off sending all these comments in an email since I can't update the documentation right now :) |
| 18:38.26 | leifmadsen | or you could wait until we get OFPS running in a couple weeks and comment directly ons-ite |
| 18:38.29 | leifmadsen | on-site |
| 18:38.38 | bcrisp | anyone know the documentation filename for cdr? looking in src/doc |
| 18:38.59 | mducharme-laptop | ok that's fine, I can wait until then, might be easier |
| 18:40.38 | leifmadsen | mducharme-laptop: ya likely is -- I'll ping you when we get that up and running |
| 18:53.08 | mducharme-laptop | leifmadsen ok sounds good.. you can by email or I will also likely be online but may not be in here, I am in #zenoss a lot |
| 18:53.13 | *** part/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 18:53.21 | mducharme-laptop | because we use the zenoss open source nms instead of opennms |
| 18:53.21 | leifmadsen | mducharme-laptop: *thumbs up* |
| 18:53.26 | leifmadsen | gotcha |
| 18:53.30 | mducharme-laptop | I find it a lot better than opennms actually heh |
| 18:54.38 | bcrisp | would love to hear anyone share their experience with using jitterbuffers in * - trying to zero in on some garbled audio issues and am not sure if the jitterbuffer features would help / hurt the situation |
| 18:54.56 | bcrisp | (or have no effects) |
| 19:01.06 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 19:01.31 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 19:08.20 | *** join/#asterisk ^scott^ (~scott@stthom.org) |
| 19:09.02 | ^scott^ | Hi, I'm using cdr_mysql and I'm noticing that with some phone calls, the cdr.src column is '', even though I know it was an inbound phone call. Does anyone know under what circumstances that column is supposed to be blank like that. |
| 19:09.45 | iscario | hello. I have a question about iax client : i noticed that when two clients have done a call, Asterisk server thinks that they are still busy. So that if A hangup after a call to B, if C is trying to call A, it won't work, and C will be in touch with A's answering machine. So where does the problem comes from ? i notived that fact with 2 different client. Thx |
| 19:22.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:22.41 | dbtid | hello? |
| 19:22.44 | iscario | no suggestion ? |
| 19:23.24 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 19:23.30 | Chainsaw | Hi? |
| 19:24.57 | carrar | hi!! |
| 19:26.16 | bmoraca_work | hrm...can I have a phone subscribe to hints on a server to which that phone is not registered? |
| 19:27.06 | [TK]D-Fender | bmoraca_work: Go check the manual for that phone |
| 19:27.25 | [TK]D-Fender | bmoraca_work: and Registration != Subscription |
| 19:27.59 | bmoraca_work | [TK]D-Fender: I know that the phone can subscribe to anything, my questions was more "does Asterisk have a problem letting unassociated devices subscribe to hints?" |
| 19:28.32 | bmoraca_work | are there any potential security issues or permissions issues I might run in to? |
| 19:28.43 | [TK]D-Fender | bmoraca_work: un-associated? Don't know in un-authed calls will point to [general]'s context for that... Maybe... |
| 19:29.24 | bmoraca_work | good point, I'll need to make sure the hints context is available for "guest" peers |
| 19:29.57 | bmoraca_work | server is not accessible from outside the network, so i'm not worried about that |
| 19:34.21 | *** join/#asterisk citrus2 (~citrus2@72.215.183.28) |
| 19:34.52 | dbtid | howdy... can someone point me to some docs that describe the asterisk/dahdi framework? how audio data flows through the system? |
| 19:35.37 | citrus2 | hey all, i am trying to execute "sip show registry" with php so i can get the result in a web browser |
| 19:36.13 | [TK]D-Fender | bmoraca_work: What are you working with that wouldn't be reg's? 3rd aprt stack? |
| 19:36.14 | citrus2 | anyone know how i should go about that? php's shell_exec uses a different user so asterisk complains i was thinking AGI but i haven't got into that yet |
| 19:36.16 | [TK]D-Fender | party* |
| 19:36.31 | Katty | hoo-rah! |
| 19:36.42 | Naikrovek | wonder if asterisk would run on this: http://www.adafruit.com/index.php?main_page=product_info&cPath=46&products_id=278 |
| 19:36.50 | [TK]D-Fender | citrus2: AMI + Command |
| 19:37.14 | ^scott^ | Uhhh, I've got debug enabled in logger.conf for Console, but cdr_mysql isn't spitting out its debug information. Is there an additional option for cdr_mysql or cdr? |
| 19:37.27 | bmoraca_work | [TK]D-Fender: no, i have a customer that has multiple sites with multiple PBXes, and they want BLF of users at all sites...so a phone at Site A that would not be registered to the PBX at Site B needs to subscribe to the hints for phones at Site B |
| 19:37.33 | citrus2 | [TK]D-Fender: is there a prebuilt way i can execute a command via AMI with php? |
| 19:38.09 | [TK]D-Fender | bmoraca_work: Any reason not to reg there? |
| 19:38.27 | [TK]D-Fender | citrus2: PHP_AGI <- Google it |
| 19:38.34 | bmoraca_work | [TK]D-Fender: no reason to register there...the phones are not going to have extensions at the other sites (nor do they want it) |
| 19:38.34 | Naikrovek | bmoraca_work: i think with 1.6.2 you can do that. cross-server hints |
| 19:38.49 | Naikrovek | i forget which version added that |
| 19:38.49 | [TK]D-Fender | bmoraca_work: No, this monitoring phone... |
| 19:39.07 | citrus2 | [TK]D-Fender: i have been but i haven't found any good examples for excuting something via a web call i have found plenty of stuff for handeling incoming calls.. but thats not what i want to do |
| 19:39.25 | [TK]D-Fender | bmoraca_work: http://snapvoip.blogspot.com/2008/06/distributed-presence-for-asterisk-in.html |
| 19:39.28 | bmoraca_work | citrus2: you want to use AMI...it's a simple telnet interface. |
| 19:39.33 | *** join/#asterisk apocn (~apo@unaffiliated/apocn) |
| 19:39.46 | citrus2 | bmoraca_work: yes i know that |
| 19:39.49 | [TK]D-Fender | citrus2: COMMAND <- |
| 19:39.51 | citrus2 | but i want to use it via a php call |
| 19:40.01 | bmoraca_work | citrus2: right...so look up how to use sockets in PHP |
| 19:40.10 | apocn | Hello all, Im having a problem with a H323 Trunk. inbound calls are working great, but not outbound calls. I get this when I try: http://pastebin.com/raw.php?i=jyaugJ4T |
| 19:40.11 | citrus2 | i am just looking for a php example to connect to the AMI |
| 19:40.36 | apocn | its a local network (no nat), etc. Asterisk <--> SBC (Acme) |
| 19:40.38 | [TK]D-Fender | citrus2: I just handed you a LIBRARY to let you do that |
| 19:40.41 | *** join/#asterisk mpe_ (~mpe@65.161.206.209) |
| 19:40.42 | ^scott^ | citrus2: May not help, but I noticed that there is a CPAN module for AMI in Perl. |
| 19:40.51 | bmoraca_work | citrus2: http://www.geckotribe.com/php-telnet/ maybe? |
| 19:41.21 | bmoraca_work | citrus2: this might be more useful: http://php.net/manual/en/function.fsockopen.php |
| 19:41.39 | bmoraca_work | google is excellent |
| 19:41.40 | apocn | the provider told me I needed to pass: '8' (pause) 1 (pause) 0 (pause) 9 (pause) phone_number_to_call |
| 19:42.07 | bmoraca_work | apocn: use the 'w' in your dial string |
| 19:42.24 | apocn | bmoraca_work: you mean /w ? |
| 19:42.30 | bmoraca_work | no |
| 19:42.31 | citrus2 | i have been googling for the last few hours.. i just figured there had to be a better way to issue a simple command then learning how to use fsockopen and establishing a socket to the AMI port |
| 19:42.48 | [TK]D-Fender | citrus2: [15:38]<[TK]D-Fender>citrus2: PHP_AGI <- Google it |
| 19:42.49 | bmoraca_work | "w" when in your dialstring will cause a pause |
| 19:43.05 | apocn | ok |
| 19:43.27 | bmoraca_work | citrus2: fsockopen is like 7 lines of code to execute an AMI command...3 of the 7 lines are identical |
| 19:43.35 | *** join/#asterisk jdoe_ (jdoe@falseprophet.ca) |
| 19:44.42 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 19:44.51 | *** join/#asterisk rafael-ec (~rafael@200.110.234.162) |
| 19:44.52 | citrus2 | [TK]D-Fender: Thanks, but i think anyone that made it into this channel knows how to use google.. i was hoping for a little experiance not just PHP_AGI GOOGLE IT.. like haven't done that. |
| 19:45.12 | citrus2 | it looks like there is two ways.. |
| 19:45.26 | [TK]D-Fender | s/two/tons/ |
| 19:45.27 | citrus2 | either using fsocket or using the php_AGI prebuilt thing. |
| 19:45.27 | *** join/#asterisk beardy_ (~beardy@unaffiliated/beardy) |
| 19:45.28 | bmoraca_work | apocn: so, for instance, Dial(zap/1/wwww5551234) will open the channel, wait 2 seconds, and then dial it. |
| 19:45.45 | [TK]D-Fender | citrus2: There are a fwe other frameworks as well |
| 19:46.02 | apocn | ok |
| 19:46.16 | apocn | so Instead of: OOH323/8093727262@172.19.11.50:1720|300| I will execute: OOH323/8w1w0w9w8093727262@172.19.11.50:1720|300| |
| 19:46.16 | bmoraca_work | apocn: however, h323 sucks. :P |
| 19:46.19 | [TK]D-Fender | bmoraca_work: Except he mentioned using an SBC <- Doesn't sound like analog to me |
| 19:46.20 | apocn | it does! |
| 19:46.23 | apocn | I hate it :S |
| 19:47.11 | bmoraca_work | maybe |
| 19:47.13 | bmoraca_work | no idea |
| 19:47.28 | bmoraca_work | why are you using h323 anyway? |
| 19:48.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 19:49.34 | apocn | bmoraca_work: just trying to help a friend (they use it at he's company) |
| 19:49.35 | ^scott^ | I'm trying to get debug working with CDR MySQL so that I can figure out why the src information isn't getting to the DB |
| 19:50.05 | apocn | where I work we use sip all the way (for interconnection between all providers) |
| 19:50.29 | ^scott^ | I've tried core set debug 10, setting console to debug on logger.conf, and reload'ng the cdr_mysql_addon.so file, but it's still not logging the SQL commands to the console. Is there another debug command I need to run? |
| 19:50.34 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 19:51.53 | *** join/#asterisk plundra (404@article.se) |
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| 19:52.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:53.44 | BCS-Satori | How does one go about change the outbound calling name of the phone system when the phone system calls an extension internally. i.e.: if i do a console dial to an extension it says from "asterisk". I would like to change that to something else. |
| 19:54.58 | citywok | BCS-Satori: in the dialplan set the callerid to whatever you want |
| 19:55.23 | BCS-Satori | citywok: gotcha didn't know if there was a global variable someone hiding |
| 19:59.06 | leifmadsen | BCS-Satori: CALLERID() function |
| 20:00.49 | ^scott^ | Okay, forget about MySQL, even the cdr-csv has a blank src |
| 20:01.01 | BCS-Satori | actually I think I found what I am looking for in /etc/asterisk/asterisk.conf with "systemname = pbxname" or better yet "autosystemname= true" |
| 20:01.23 | apocn | bmoraca_work: didn't work :( same error |
| 20:01.38 | ^scott^ | CDR, in general, isn't logging where incoming phone calls are coming from, it just has an empty string, has anyone else seen this or have any troubleshooting ideas? |
| 20:01.55 | bmoraca_work | apocn: not sure what to tell you then...talk to whoever controls the SBC |
| 20:03.32 | apocn | ok, thanks man |
| 20:05.35 | p3nguin | Is there a way to set the codec on a per-call basis? |
| 20:20.53 | *** join/#asterisk seanjohn (~admin@173.50.101.10) |
| 20:26.23 | Katty | ATTENTION p3nguin |
| 20:26.30 | Katty | incoming hug! PREPARE YOURSELF. |
| 20:26.39 | Katty | hugs p3nguin |
| 20:27.40 | iscario | I just copy once my question from few time ago, expecting an answer :D. I have a question about iax client : i noticed that when two clients have done a call, Asterisk server thinks that they are still busy. So that if A hangup after a call to B, if C is trying to call A, it won't work, and C will be in touch with A's answering machine. So where does the problem comes from ? i notived that fact with 2 different IAX client. Thx |
| 20:27.50 | p3nguin | :) Wow. Even though I had a warning, I still didn't see it coming. |
| 20:28.10 | *** join/#asterisk geneg1 (~gene@bas3-toronto01-1177778310.dsl.bell.ca) |
| 20:28.24 | Qwell | Katty: I see how it is |
| 20:28.45 | beek | is jealous of p3nquin. |
| 20:28.54 | Chainsaw | Hi Katty :) |
| 20:29.09 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 20:30.39 | Katty | hugs on beek |
| 20:30.42 | Katty | hugs on Chainsaw |
| 20:30.46 | Katty | Qwell: s'ok, i still love you. |
| 20:30.51 | Katty | hugs on Qwell |
| 20:30.52 | Chainsaw | Katty: *hug* |
| 20:31.20 | beek | Katty: *hug* |
| 20:33.21 | Diffen2 | Hello, does anyone of you guys know of some device that you can connect to RJ45 and I guess that it registers into a extension. When there are a call on that extension the device sounds loud like hell or start flashing. Good for factories and loud enviroment. I know there are analog devices on the market. |
| 20:36.29 | Chainsaw | Diffen2: Something like this? http://www.algosolutions.com/products/Audible-and-Visual-Alerting/8180-sip-audio-alerter.html |
| 20:37.16 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 20:39.21 | p3nguin | Should Set(SIP_CODEC_OUTBOUND=xxxx) work in 1.4.35? |
| 20:42.29 | Diffen2 | Chainsaw exactly! have you tried that one out? |
| 20:42.48 | Chainsaw | Diffen2: Nope, I was trying some random google queries and that came up. |
| 20:43.00 | Diffen2 | ok :) |
| 20:44.16 | Diffen2 | I have called the blind and the deaf people organisation here in sweden and all the companies that sell electronic junk but no one had heard of this device. Thanks again Chainsaw! |
| 20:44.28 | Chainsaw | Diffen2: You're welcome :) |
| 20:45.12 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
| 20:47.08 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:49.10 | Qwell | Diffen2: Why not just connect a "normal" device that they would sell to an ATA? |
| 20:51.32 | *** join/#asterisk bsaxon (~bsaxon@75.216.124.255) |
| 20:52.13 | *** join/#asterisk ahowlader (howlader@180.234.37.227) |
| 20:53.14 | Katty | Qwell: i just fell in love. |
| 20:53.23 | Katty | Qwell: black 2005 corvette |
| 20:56.17 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 21:05.41 | Diffen2 | Qwell we dont have good experience of ATA boxes. |
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| 21:19.37 | seanjohn | I was able to get grub to install and boot but it tries to boot from /vmz, i guess i need to edit menu.1st? |
| 21:19.49 | *** part/#asterisk apocn (~apo@unaffiliated/apocn) |
| 21:19.51 | seanjohn | of course, everything is in /boot/vmz |
| 21:19.53 | Qwell | seanjohn: wrong channel |
| 21:22.09 | seanjohn | thanks qwell, sorry |
| 21:22.52 | chuckp | I am trying to make an IAX2 connections between two * servers to test, and I keep getting "No Authority" error http://pastebin.com/wJiKCwEX can anyone tell me what in my conf is wrong? |
| 21:22.53 | p3nguin | and it is menu.lst, not menu.1st |
| 21:23.32 | chuckp | Ultimately I am trying to get it so one server can call into the other server, it only needs to be one way |
| 21:23.41 | chuckp | just been so long since I did an IAX2 conf |
| 21:24.55 | p3nguin | The host settings are probably reversed, for one thing. |
| 21:25.13 | p3nguin | The usernames are not really needed. |
| 21:26.09 | chuckp | Ill give that a shot |
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| 21:35.00 | p3nguin | chuckp: http://pastebin.com/m2jLpMcX |
| 21:36.48 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:39.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:39.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:39.40 | chuckp | Thank you p3nguin I actually just got it, but my conf is much more simplistic, I am going to add in the items and compare with what you pasted |
| 21:39.56 | chuckp | is auth=plaintext safe? |
| 21:40.39 | p3nguin | I suppose it depends on your definition of safe. |
| 21:41.38 | chuckp | It seems to be working without plaintext but it looks like the passcode is just a simple hash anyways so I guess it probably doesn't mattery anyways |
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| 22:00.45 | nitram | p3nguin: weren't you the one with a 7925? |
| 22:03.43 | p3nguin | I think we talked about it because I use 7900 series phones with SCCP images, but I don't specifically have a 7925. |
| 22:06.30 | nitram | ok, then it was someone else |
| 22:06.57 | nitram | i just set up a 7925 today and leaving the wireless range and reentering did crash the asterisk server |
| 22:10.32 | p3nguin | That's pretty crappy. What version of Asterisk and chan_sccp are you using? |
| 22:18.41 | *** join/#asterisk [hC] (~hardcore@173-9-132-153-Miami.FL.hfc.comcastbusiness.net) |
| 22:18.49 | [hC] | anyone here used a cisco 7920/7921? |
| 22:19.52 | Chainsaw | Just 7960, sorry. (And hated it!) |
| 22:21.25 | *** join/#asterisk seanjohn (~admin@173.50.101.14) |
| 22:22.05 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:22.44 | p3nguin | You don't like your 7960? |
| 22:22.49 | [hC] | Chainsaw: heh, yeah, well past that ;) |
| 22:23.16 | Chainsaw | p3nguin: No. Waiting for sign-off on my POs for Polycom. |
| 22:24.14 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 22:24.33 | p3nguin | I'll give you $20 for the old 7960. |
| 22:24.51 | [hC] | Heh, I hate me some polycom lately too. |
| 22:25.48 | exothermc | When the carrier says they are providing their T1 with loop start, do I set my configs to FXS or FXO loopstart? |
| 22:26.20 | [hC] | T1 is not FXO or FXS. Its PRI |
| 22:26.54 | exothermc | [hC]: Well actually a T1 does not have to be an ISDN pri, that is a pretty limited understanding of the tech. |
| 22:27.35 | *** join/#asterisk mode14 (~mode14@209.189.234.189) |
| 22:27.58 | *** part/#asterisk mode14 (~mode14@209.189.234.189) |
| 22:28.24 | [hC] | exothermc: touche. My assumption not ever having spoken to you was that you were one of the hundreds of people that come in here every day not understanding the technology. my bad. |
| 22:29.15 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 22:30.13 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:30.42 | exothermc | [hC]: Well in this case this is just a voice T1 with CAS (channel associated signaling) also known as Stollen Bit Signaling, but I'm not experienced/smart enough to know if I should be using FXS or FXO for the signaling. |
| 22:31.03 | [hC] | exothermc: ive never used that particular configuration either, sorry.. |
| 22:31.25 | exothermc | The only thing I'm told is "Loop start" which still has some ambiguity. |
| 22:32.08 | nitram | p3nguin: asterisk 1.6 from debian squeeze and chan-sccp-b trunk |
| 22:32.23 | nitram | [hC]: yes, i hav a 7920 |
| 22:33.14 | nitram | [hC]: too bad the 7920 is still 802.11b only |
| 22:33.24 | nitram | at least it got WPA/TKIP |
| 22:34.04 | [hC] | nitram: sweet. yeah im trying to firmware upgrade it but it only ever asks for a CTLSEP file and a random firmwar.e I just tried a factory reset to get it to ask for its xmldefault.cnf.xml again |
| 22:34.25 | nitram | eeks |
| 22:35.12 | nitram | didn't that require the OS7920.TXT? |
| 22:35.18 | nitram | in that i have cmterm_7920.4.0-03-02 |
| 22:35.37 | [hC] | mine doesnt even ask for OS7920.TXT or XMLDefault.cnf.xml |
| 22:35.42 | nitram | ouch |
| 22:35.45 | [hC] | it just asks for CTLSEP<mac>.tlv |
| 22:35.54 | [hC] | and then CMTERM_7920.4.0-03-02.LOADS |
| 22:36.00 | [hC] | but i have newer firmware than that |
| 22:36.06 | nitram | you have? |
| 22:36.08 | [hC] | that id have to tell it about - somehow- |
| 22:36.21 | nitram | i did not know there was a newer one available |
| 22:36.21 | [hC] | I have uhh. |
| 22:36.30 | [hC] | CP7921G-1.3.4SR1.LOADS |
| 22:36.38 | nitram | see |
| 22:36.41 | nemith | so you haven't tried factory reset yet? |
| 22:36.43 | nitram | that is 7921 |
| 22:37.02 | [hC] | Just factory reset it |
| 22:37.07 | [hC] | now trying to get it to join a wireless network |
| 22:37.26 | [hC] | it doesnt like joining networks ive created on my laptop (macosx) using internet sharing |
| 22:37.27 | nitram | i just upgraded the 7925 today, worked like a charm |
| 22:37.34 | [hC] | and the existing wireless network is WPA (this firmware doesnt have it) |
| 22:37.57 | nitram | [hC]: what device do you have now? |
| 22:38.08 | nitram | 7920 or 7921? |
| 22:38.22 | [hC] | 7921 |
| 22:38.41 | nitram | and it asks for 7920 firmware? |
| 22:39.18 | nitram | isn't the CTLSEP<mac>.tlv for certificate loading and stuff? |
| 22:39.29 | nitram | did you set some security foo? |
| 22:40.31 | [hC] | ok here we go.. factory reset, wifi connected. tftp set. lets see what it asks for |
| 22:40.56 | [hC] | repeatedly asking for CTLSEP0021553DBA98.tlv now, which is fine. doesnt exist. thats normal for all these 79xx devices, they usually move on |
| 22:41.31 | nitram | yup |
| 22:41.56 | nitram | don't you get the tftp from dhcp? |
| 22:42.00 | [hC] | weird, this time it didnt ask for anything else at all, and now says "Not in service" |
| 22:42.09 | [hC] | nitram: dhcp is off in this particular case, i set the tftp server on the phone itself. |
| 22:42.38 | nitram | try to wireshark |
| 22:42.51 | [hC] | I see tftp requests coming in for CTLSEP0021553DBA98.tlv to the tftp server. |
| 22:43.04 | [hC] | Now it says "Opening <tftp server ip>" wiith no more tftp requests coming in. |
| 22:43.11 | [hC] | Like its trying to connect to that ip with sccp. |
| 22:43.22 | [hC] | maybe ill set up sccp and feed it a new load id that way |
| 22:43.36 | nitram | yup |
| 22:43.52 | nitram | i'd wireshark first, see what's happening |
| 22:44.03 | nitram | you only monitor the tftp requests |
| 22:45.12 | *** join/#asterisk Greek-Boy (~Greek-B0y@41.188.154.137) |
| 22:47.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:51.03 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 23:09.44 | *** part/#asterisk thehar (thehar@thehar.xmission.com) |
| 23:10.27 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 23:10.38 | boodu | hello |
| 23:10.42 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 23:12.40 | boodu | what is the difference between a fxo cards and ISDN BRI Cards ? |
| 23:13.19 | bougyman | yes |
| 23:14.01 | [TK]D-Fender | bougyman: Digital signalling vs analog signalling. |
| 23:14.16 | [TK]D-Fender | bougyman: Having DID's, the ability to send CID, etc |
| 23:14.16 | bougyman | oh, _what_ is the difference. |
| 23:14.22 | bougyman | i thought he asked if there was a difference. |
| 23:14.26 | bougyman | learns to read |
| 23:17.53 | boodu | bougyman, i don't know what i must buy :S |
| 23:18.08 | boodu | fxo or bri cards :s |
| 23:18.09 | bougyman | boodu: what kind of line are you getting? |
| 23:18.16 | boodu | ISDN |
| 23:18.59 | boodu | with 4 number of phone (and line) |
| 23:19.23 | bougyman | well i think you just answered your question, then. |
| 23:19.28 | bougyman | you need digital for ISDN |
| 23:19.32 | boodu | i must buy a 2 BRI S/T interface cards? |
| 23:20.03 | WIMPy | Has anyone here ever experienced MOH playing at different volumes on occation? |
| 23:20.16 | ChannelZ | the same song? |
| 23:20.39 | WIMPy | The same file, yes. |
| 23:20.41 | ChannelZ | or file anyway.. maybe your MOH is rap, not qualifying as music/song |
| 23:20.58 | boodu | bougyman, ok fxo and fxs is for rtc |
| 23:20.59 | boodu | thx |
| 23:21.06 | ChannelZ | Haven't noticed; * doesn't do any sort of processing (besides possible codec conversions) |
| 23:21.25 | boodu | and with 2 BRI S/T interface, I will have 4 lines ? |
| 23:21.26 | [TK]D-Fender | boodu: What you are looking as is ISDN-BRI |
| 23:21.49 | WIMPy | That's what I thought. But I have had complaints about the music being far too lound when it was perfectely ok for be. |
| 23:22.22 | WIMPy | But when handling multiple calls I had a short snipplet of MOH myself that I actually found far too loud as well. |
| 23:22.37 | WIMPy | So I think there's somethign fishy going on somewhere. |
| 23:22.50 | ChannelZ | Hmm. What kind of lines? I guess if you're on analog you might have different gain settings for some channels |
| 23:22.57 | [TK]D-Fender | WIMPy: Well you haven't described your call or your MoH an any useful detail |
| 23:23.49 | WIMPy | Unfortunaletly I can't recall the exact circumstances. :-( I was handling multiple callers at that time and was concentrating on other things. |
| 23:24.10 | [TK]D-Fender | WIMPy: You aren't even describing how these calls come IN. |
| 23:24.10 | WIMPy | No analogue here. Only SIP and ISDN. |
| 23:24.19 | [TK]D-Fender | WIMPy: Seriously.. you've got NOTHING for us? |
| 23:24.54 | WIMPy | Unfortunaletly not. Otherwise I wouldn't have aske if anyone ever observed something similar. |
| 23:25.26 | [TK]D-Fender | WIMPy: WIMPy You don't even know about your calls. We have nothing to base any kind of assessment on. We're only psychic on TUESDAYS |
| 23:25.58 | ChannelZ | plays The Price Is Right fail sound |
| 23:26.18 | ChannelZ | http://www.youtube.com/watch?v=1ytCEuuW2_A |
| 23:26.30 | WIMPy | I didn't ask why it happens. Just if that ever happened to someone else. |
| 23:26.58 | exothermc | [TK]D-Fender: You ever done a T1 with Loop Start? I have a Qwest CAS T1 circuit that they say is Loop Start, but I'm not sure if I should set to FXS or FXO on my end. |
| 23:28.08 | sawgood | exothermc: if they tell you it is 'loop start' then MOST likely they have taken a digital channel or two off the T1 and made it into a 'loop start' analog FXO line |
| 23:28.19 | sawgood | usually, mounted on a 66 block on the wall near the IAD T1 device |
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| 23:29.04 | exothermc | sawgood: nope this is coming out of antran IAD and they say it is a full 24 channels. |
| 23:29.53 | sawgood | Well, then there is a big confusion on 'terms'. Loop start lines would have to come from an analog base (or maybe the entire T1 is ready for analog use) |
| 23:30.08 | boodu | [TK]D-Fender, I search to make a IPBX for 4 lines ISDN |
| 23:30.24 | sawgood | what is your goal and/or point? |
| 23:30.39 | WIMPy | boodu: 4 lines or 4 channels? |
| 23:30.42 | exothermc | sawgood: Well it isn't an ISDN circuit, it is Channel associated signaling. |
| 23:30.54 | sawgood | It is a T1/PRI probably, right? |
| 23:31.30 | sawgood | CAS is digital T1 .. the Adtran simply converts from telco signal to what you and I would call "Ethernet" |
| 23:31.34 | exothermc | sawgood: it is a T1, but not a PRI (PRI is ISDN) |
| 23:31.47 | sawgood | PRI can be ISDN ... not always ... |
| 23:32.36 | sawgood | What is the goal of the T1 circuit you have (was this just installed, is it an existing piece of equipment, where does it plug into now)? |
| 23:32.40 | exothermc | sawgood: odd so they say it is Loop Start for their DSS service. |
| 23:32.46 | boodu | WIMPy, 4 channels yes |
| 23:32.49 | exothermc | sawgood: new install. |
| 23:32.55 | [TK]D-Fender | [19:31]<sawgood>PRI can be ISDN ... not always ... <-- O RLY? And when ISN'T it? |
| 23:33.35 | sawgood | Well, here in California XO has a flex PRI (which is data and analog) |
| 23:33.42 | sawgood | that is an example (I guess) |
| 23:34.28 | sawgood | What do you want to do with the T1 CAS circuit you have? |
| 23:34.33 | exothermc | [TK]D-Fender: So my side would be FXO Loop Start then right? |
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| 23:34.47 | exothermc | sawgood: make outbound calls with it. |
| 23:35.16 | sawgood | exothermc: so, the only piece of equipment they 'gave' you was an Adtran IAD device ... which goes to what piece of equipment? |
| 23:35.38 | exothermc | sawgood: Sangoma A104DE |
| 23:35.42 | [TK]D-Fender | exothermc: LS is a CAS signalling not PRI |
| 23:35.53 | sawgood | Is there a 'smart jack' on the WALL near the IAD? |
| 23:36.02 | exothermc | [TK]D-Fender: Ya there is no indication this is PRI. |
| 23:36.05 | [TK]D-Fender | And the jack shape won't determine this <- |
| 23:36.16 | [TK]D-Fender | Its all about ENCODING |
| 23:36.44 | exothermc | [TK]D-Fender: They claim AMI coding. |
| 23:37.07 | exothermc | sawgood: The smart jack is built into the IAD. |
| 23:37.26 | seanjohn | where does dahdi look for the kernel sources /usr/src/kernels or /usr/src/linux? |
| 23:37.30 | sawgood | ok .. so you are not sure if this is a DATA or voice circiut then? |
| 23:37.37 | [TK]D-Fender | exothermc: So they seem to pass the calls off as one would to an old T1 PBX. Retards |
| 23:37.40 | sawgood | Why not call your carrier (using your circuit ID_ |
| 23:37.57 | exothermc | sawgood: I'm very sure this is a digital voice T1 unless they royally screwed up the build. |
| 23:38.50 | exothermc | [TK]D-Fender: Well ya it is an old DMS10 they are working with. I know the DMS10 can do PRIs, but they don't have the right gear in it. So only the old crappy voice T1s out of it. |
| 23:39.35 | sawgood | So, from the Adtran, you were planning on running a cable to your IP PBX T1 card? |
| 23:40.13 | exothermc | sawgood: Yup which is what we have done for other T1s, (although they were wink start.) |
| 23:40.46 | sawgood | So, to me ... it seems you need your 'cut sheet' from the provider which will tell you the specs (which in turn you would use to configure your T1 card) ... |
| 23:40.56 | rocknfra | [TK]D-Fender, about my problem I've find out that if the other person is natted |
| 23:41.45 | [TK]D-Fender | rocknfra: Shouldn't matter |
| 23:42.08 | rocknfra | I've make some tests with a friend of mine |
| 23:42.11 | sawgood | What model is the Adtran? |
| 23:42.25 | rocknfra | each one at his own house so with different public ip addresses |
| 23:42.28 | exothermc | sawgood: Well they have given me that except for two ambiguous pieces. They say Coding is AMI, and Loop Start. Now the odd part is the Tech flopped back and forth on D4 or ESF for the framing. Also he didn't know if it was FXS or FXO on the Loop Start. |
| 23:42.38 | rocknfra | with the very same software |
| 23:42.50 | exothermc | sawgood: ya not at the location right now so I can't say. |
| 23:42.56 | rocknfra | if he connects trought his router via dsl the problems appear |
| 23:43.23 | rocknfra | if he connects via mobile network (directly without any nat device) the call is perfect |
| 23:44.30 | seanjohn | where does dahdi look for the kernel sources /usr/src/kernels or /usr/src/linux? The headers are installed for the correct version but dahdi complains "You do not appear to have the 2.6.18-194.3.1.el5.centos.plus header files installed" |
| 23:49.51 | exothermc | hmm looks like I have Loss Of Frame alarm, and Alarm Indication Signal both on. |
| 23:55.33 | exothermc | [TK]D-Fender: you have any ideas where I should go from here? |
| 23:56.18 | fenrus | if you have LOF from the provider, somthing's b0rked :) |
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