| 00:03.16 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:05.27 | Katty | :< |
| 00:05.31 | Katty | there has been a death in the family |
| 00:06.03 | vk2dgy | ok, so what am I doing wrong. Edited logger.conf, added in the [logfiles] section, debug => debug then logger reload; sip debug peer 213xxxxxxxx - debug info visible in console, but not in the debug log file... |
| 00:06.13 | adnc | huhu katty |
| 00:07.24 | Katty | sighs |
| 00:07.28 | Katty | i didn't even know he was sick |
| 00:09.22 | skrite | ManxPower-work, ok, did a fresh reboot, and then service dahdi start and i got this error http://pastie.org/836036 |
| 00:09.54 | *** join/#asterisk lost_soul (~noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 00:11.55 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:15.25 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 00:25.20 | Katty | sulks |
| 00:25.48 | voipmonk | logger reload vk2dgy |
| 00:26.08 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:26.11 | voipmonk | add console |
| 00:26.17 | voipmonk | then start asterisk manually |
| 00:27.17 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 00:32.52 | vk2dgy | well, it's logging stuff - but nothing that I think is going to be useful. Lots of "background noise" - other peers comming and going etc - but there's no activity on this peer when I try to place a call. Just a warning message (as previously posted). |
| 00:36.32 | skrite | ok all, how would i go about completely removing asterisk, anything of zaptel or dahdi and everything else to start fresh, i have learned a lot but may have scrambled things too bad to recover as it is now |
| 00:44.06 | ChannelZ | make uninstall |
| 00:47.12 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 00:51.36 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 00:52.37 | skrite | ChannelZ, thanks |
| 00:58.53 | adnc | for matching local calls like 123456 would an entry like _X. be enough or do i need to consider anything else? |
| 01:17.08 | ChannelZ | no that would work |
| 01:18.24 | adnc | ChannelZ, if i would have mailbox at number 99 and someone would call 994444 that wouldn't match would it? |
| 01:22.38 | *** join/#asterisk aandrade (~aandrade@189.34.121.136) |
| 01:23.14 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:23.38 | ChannelZ | correct |
| 01:23.58 | adnc | ChannelZ, thanks |
| 01:27.42 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:38.12 | *** join/#asterisk ELBunce (~erik@kde/developer/bunce) |
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| 02:43.15 | ManxPower-work | mailbox? |
| 02:43.38 | ManxPower-work | 99 would match _X. |
| 02:43.47 | adnc | yes |
| 02:44.48 | adnc | one matching rule is exten => 99... |
| 02:47.34 | *** part/#asterisk vk2dgy (~rossw@ali-syd-3.albury.net.au) |
| 02:48.40 | adnc | is it possible to negate a matching rule? |
| 02:49.25 | adnc | like match anything but 49 |
| 02:49.29 | florz | _[0-35-9][0-8]. |
| 02:59.47 | adnc | hi florz |
| 03:01.29 | florz | hi ;-) |
| 03:04.36 | Katty | hi |
| 03:04.46 | jaytee | hi |
| 03:05.26 | Katty | jaytee: dunno if you heard the bad news or not |
| 03:05.34 | Katty | jaytee: we've had a death in the family |
| 03:05.35 | jaytee | no, what? |
| 03:05.43 | Katty | jaytee: had to bury Shire tonight...he passed away |
| 03:06.06 | Katty | Sammy seems to be okay..his fur isn't falling out yet. |
| 03:06.14 | jaytee | oh, I'm sorry to hear that hon |
| 03:06.20 | Katty | thanks. |
| 03:06.37 | jaytee | they're cute and can "get under your skin" just like cats |
| 03:06.54 | Katty | i'm more concerned about Sammy right now |
| 03:07.02 | jaytee | being alone? |
| 03:07.10 | Katty | Shire was Sammy's brother |
| 03:07.22 | *** join/#asterisk nightrid3r (kvirc@41.214.134.113) |
| 03:07.24 | Katty | and when ferrets bond, and then lose another... |
| 03:07.37 | Katty | it can be very traumatic for them |
| 03:07.52 | jaytee | yeah |
| 03:07.58 | jaytee | like swans |
| 03:08.13 | *** join/#asterisk nightrid3r (~kvirc@41.214.134.113) |
| 03:08.26 | Katty | nods |
| 03:09.31 | Katty | i'm hoping that the others will provide enough comfort for him to cope |
| 03:14.32 | jaytee | and if you give him a little extra attention for a bit that'll get him through this |
| 03:14.40 | Katty | most definately. |
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| 03:58.29 | rift0r | my carrier is telling the following and i need to change my remote party ids |
| 03:58.36 | rift0r | This is the wrong format on the Invites we are receiving from your PBX: |
| 03:58.49 | rift0r | Remote-Party-ID:"310"<sip:310@myip>;privacy=off;screen=no |
| 03:59.00 | rift0r | they want it as |
| 03:59.19 | rift0r | Remote-Party-ID:"mydid#"<sip:mydid#@63.230.216.66>;privacy=off;screen=no |
| 03:59.35 | rift0r | 310 is the extension it is coming from a sip peer |
| 03:59.40 | rift0r | how do i do this? |
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| 04:01.18 | *** part/#asterisk albasheers (~basheer@89.148.43.51) |
| 04:03.59 | rift0r | anyone? |
| 04:08.59 | *** join/#asterisk joelsolanki (joelsolank@202.160.161.94) |
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| 04:28.17 | *** part/#asterisk dzup (dzup@support.team.at.shellium.org) |
| 04:41.31 | skrite | how do i get what channel types are registered? |
| 04:42.07 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 04:44.00 | voipmonk | what modules have you loaded? |
| 04:45.11 | skrite | voipmonk, me? wcwctdm24xxp |
| 04:46.53 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 04:49.10 | rift0r | can someone help me get my caller id working |
| 04:49.22 | rift0r | for some reason it is always showing my 1+my extension |
| 04:49.27 | rift0r | for whoever i call |
| 04:59.13 | p3nguin_ | 1+ your what? |
| 04:59.37 | p3nguin_ | 1+ your phone number? That's good. |
| 05:00.06 | rift0r | no my sip phone is showing up |
| 05:00.09 | rift0r | so i am extensions 300 |
| 05:00.12 | rift0r | it shows up as 1300 |
| 05:00.13 | rift0r | on caller id |
| 05:00.21 | p3nguin_ | Change it in sip.conf. |
| 05:00.21 | rift0r | so 800 numbers are blocking my calls |
| 05:00.44 | rift0r | isn't that fromuser=? |
| 05:00.47 | p3nguin_ | callerid= |
| 05:01.46 | p3nguin_ | You'll probably want to also set trustrpid=yes and sendrpid=yes for the service peer. |
| 05:09.57 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:14.13 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 05:16.24 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 05:20.16 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 05:20.24 | KingDavidNYC | <PROTECTED> |
| 05:20.40 | KingDavidNYC | 1.6.2 |
| 05:21.30 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 05:29.52 | joelsolanki | Hello all |
| 05:29.59 | joelsolanki | please check this dialplan http://pastebin.ca/1805434 |
| 05:30.20 | joelsolanki | gotoif says to go to test1 context. |
| 05:31.00 | joelsolanki | but i want to send it to 4th priority of same dialplan means this line exten => 222,4,Busy(5) |
| 05:31.03 | joelsolanki | how can i do this please? |
| 05:31.08 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:33.15 | joelsolanki | any suggestions |
| 05:34.22 | ManxPower-work | why are you numbering your priorities. That was deprecated in like 1.2 Asterisk |
| 05:35.40 | ManxPower-work | Also your question makes NO sense. |
| 05:36.43 | ManxPower-work | http://pastebin.ca/1805439 |
| 05:37.15 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 05:39.00 | joelsolanki | hmm ok |
| 05:39.01 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:39.09 | joelsolanki | oh why ? |
| 05:39.42 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 05:40.06 | joelsolanki | i see . |
| 05:40.12 | joelsolanki | exten => 222,n,GoToIfTime(11:00-14:00|thu|1|sep?busy) |
| 05:40.16 | joelsolanki | so this should work right ? |
| 05:40.24 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 05:40.25 | joelsolanki | exten => 222,n(busy),Busy(5) |
| 05:40.44 | joelsolanki | actully i dont want to create new context. |
| 05:40.57 | joelsolanki | i want to write dialplan in one single dialplan. |
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| 05:57.30 | jayesh | Hi All |
| 05:57.54 | carrar | ~book |
| 05:57.55 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:00.33 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-cgkrgrkpotvzunly) |
| 06:01.36 | *** join/#asterisk Chotaire (chotaire@to.to.to) |
| 06:04.04 | b14ck | What's up everyone? |
| 06:05.00 | p3nguin_ | joelsolanki: exten => 222,n,GoToIfTime(11:00-14:00|thu|1|sep?busy) says to go to the busy label on the same extension in the same context. exten => 222,n(busy),Busy(5) is where it goes. That is not a new context. |
| 06:09.36 | KingDavidNYC | do you guys know how I can get the value from a call on chan_spool.c? |
| 06:09.55 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
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| 06:11.50 | joelsolanki | p3nguin: yes correct go it :) |
| 06:11.56 | joelsolanki | i was not aware of label. |
| 06:12.06 | joelsolanki | but now i understand what label is |
| 06:12.22 | joelsolanki | so this is going to be interesting. and can do much good routing :):) |
| 06:12.24 | joelsolanki | thanks |
| 06:13.34 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:16.11 | jayesh | <PROTECTED> |
| 06:18.19 | rift0r | i am having an issue when trying to dial an extension while playback is going |
| 06:18.22 | rift0r | i get [Feb 21 23:14:09] WARNING[11642]: pbx.c:4332 __ast_pbx_run: Channel 'SIP/192.168.47.68-000129f8' sent into invalid extension '3' in context 'fromtrunk', but no invalid handler |
| 06:18.46 | rift0r | this is when i am calling my did from my cell phone |
| 06:18.56 | rift0r | i try to dial extension 300 while background(greeting) is playing |
| 06:19.01 | rift0r | and it gives me that and hangs up |
| 06:19.16 | rift0r | i never had an issue with this config on 1.4 |
| 06:19.19 | rift0r | this is on 1.6 |
| 06:19.33 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 06:23.58 | *** part/#asterisk albasheers (~basheer@89.148.43.51) |
| 06:35.00 | p3nguin_ | rift0r: Do you have exten 300 in the fromtrunk context? If not, do you include a context that does? |
| 06:35.25 | p3nguin_ | The invalid handler is the 'i' extension, but you didn't define any rule for it. |
| 06:46.38 | jayesh | I was trying to setup a dictation server using asterisk. I was able to do this with limited success. I am using the dictate app for this purpose. |
| 06:47.05 | jayesh | the dictate app comes pre built set of keypad mapping, how can I change that keypad mapping ? |
| 06:48.04 | *** join/#asterisk oej (~olle@ns.webway.se) |
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| 07:03.07 | rift0r | thx p3nguin_ that fixed it |
| 07:03.09 | rift0r | i added |
| 07:03.10 | rift0r | exten => _3[0-5]X,1,Dial(SIP/${EXTEN},15,to) |
| 07:03.16 | rift0r | to fromtrunk |
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| 08:53.33 | elliot98 | hiya all! |
| 08:53.35 | elliot98 | great to be back! |
| 08:53.39 | elliot98 | anyone here this time of day? |
| 08:53.53 | gr0mit | nods |
| 08:54.13 | *** join/#asterisk jayesh (~jay@122.172.124.160) |
| 08:54.32 | elliot98 | winks |
| 08:55.28 | elliot98 | for some odd reason asterisk after running for some time looks like it creates over 100 unix sockets |
| 08:56.04 | elliot98 | when I run netstat -anp, a whole bunch of "CONNECTED... /var/run/asterisk.ctl" show up |
| 08:56.20 | elliot98 | and from then on I cannot connect to asterisk with asterisk -r |
| 08:58.33 | *** join/#asterisk d-k-t (~D@112.202.232.46) |
| 08:58.52 | elliot98 | there are no zombie scripts accessing asterisk, can't figure out what they're coming from |
| 09:02.16 | jayesh | I am using app_dictate for a project of mine, but I wish to change the keypad being used by default by the module. How can I accomplish this ? |
| 09:02.36 | *** join/#asterisk Godfather_ (~Godfather@205.Red-88-0-3.dynamicIP.rima-tde.net) |
| 09:16.33 | Faustov | elliot98: try changing the ulimit maybe |
| 09:16.40 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 09:16.53 | elliot98 | what is ulimit? |
| 09:17.21 | elliot98 | but even more importantly, where are they coming from? |
| 09:18.16 | Faustov | man ulimit |
| 09:19.30 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:21.34 | elliot98 | Faustov: thanks...I'll look into that |
| 09:21.45 | elliot98 | however, it doesn't solve where the memory leak is coming from |
| 09:22.34 | Faustov | you just said many sockets are created... what does memory leak have to do with it? |
| 09:22.38 | Faustov | is confused |
| 09:23.01 | *** join/#asterisk fofware (~chatzilla@host147.190-229-137.telecom.net.ar) |
| 09:23.45 | elliot98 | Faustov: I am just assuming that since no scripts are creating these sockets, there could be some memory leak somewhere that is creating them |
| 09:24.24 | *** join/#asterisk klashniv (~klashniv@41.191.76.68) |
| 09:24.25 | Faustov | not closing open file descriptors has nothing to do with memory leaks |
| 09:25.06 | elliot98 | Faustov: true...how are they opening up to begin with? |
| 09:27.13 | Faustov | idk, you can monitor that in the first place - maybe they are legimitate |
| 09:27.30 | elliot98 | how do I monitor them? |
| 09:27.36 | elliot98 | they just pop up |
| 09:28.15 | Faustov | log in to the console, set high verbosity and watch for netstat changes corresponding to asterisk events |
| 09:28.53 | klashniv | hullo all, quick question: is there any way of clearing hang sip channels? Asterisk 1.4.27.1, sip show channels shows 748 active SIP channels but show channels has 386 active channels and 193 active calls |
| 09:32.24 | klashniv | lots of the channels seem to have hung on INVITE |
| 09:34.40 | klashniv | <peer ip> <called number> 64639ed263e 00102/00000 0x0 (nothing) No Init: INVITE |
| 09:36.24 | Faustov | I had that on 1.6.1.10 |
| 09:36.31 | Faustov | upgrade to 1.6.1.14 fixed the issue for me |
| 09:39.19 | klashniv | thanks, guess its time for an upgrade |
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| 10:13.23 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 10:19.47 | *** join/#asterisk Ceeriael (~ceeriael@93.167.108.90) |
| 10:21.40 | Ceeriael | Its strange... i have to internal extentions. Its takes me about 5 sec to call from one to another ? |
| 10:21.56 | Ceeriael | it used to be "instant". |
| 10:21.59 | *** join/#asterisk soman (~somnath@212.226.87.141) |
| 10:22.18 | Ceeriael | any idea what could be causing this delay ? (or how i can find out what is causing it) |
| 10:22.47 | *** part/#asterisk klashniv (~klashniv@41.191.76.68) |
| 10:25.28 | tzafrir | Ceeriael, what type of extensions? What devices? |
| 10:25.39 | quietdev | hey anyone here based in Japan? |
| 10:25.58 | Ceeriael | Sip devices (xlite phones) |
| 10:26.02 | Ceeriael | tzafrir ^ |
| 10:26.15 | tzafrir | Any chance you use overlap dialing? |
| 10:26.23 | Ceeriael | i dont know what that is ? |
| 10:26.29 | quietdev | just a totally off-topic question, but has fax loosen its grip on the Japanese consumer world? or are faxes still how grandpapas and grandmamas use to converse with their family? |
| 10:26.33 | tzafrir | dial digit by digit |
| 10:26.50 | Ceeriael | dont think so... i can press anything, and then i press dial. |
| 10:26.56 | quietdev | fax machine has never grown any sufficient balls/vulvas to deal with fax spam in north america |
| 10:27.22 | quietdev | so i can see why they just died, but people keep one around (probably just a fax modem) for shits and giggles |
| 10:27.28 | tzafrir | Ceeriael, next step: look at the dialplan trace |
| 10:27.31 | Ceeriael | i can also change what ive pressed (if i press wrong). so i dont think i use overlap dialing |
| 10:27.45 | tzafrir | (core set verbose 3 , and look at the CLI) |
| 10:27.49 | Ceeriael | ok |
| 10:27.53 | Ceeriael | ill do that now |
| 10:29.13 | Ceeriael | it seems that it hangs when checking for inbound recording..... |
| 10:29.47 | Ceeriael | is it supose to be that slow ? |
| 10:30.03 | Ceeriael | tzafrir |
| 10:30.32 | Ceeriael | it "hangs" to places... one at the Checking -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck |
| 10:31.41 | *** join/#asterisk helloritesh (~hellorite@121.243.172.188) |
| 10:31.56 | Ceeriael | and again at: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi |
| 10:32.02 | Ceeriael | this is where the time is spent..... |
| 10:32.14 | Ceeriael | tzafrir any idea on how i can make it faster ;) |
| 10:32.40 | helloritesh | [TD]D-Fender.. A quick question for you. You had helped earlier on the call recording using ChanSpy and it worked fine on 1.4. Now I upgraded to 1.6 and I am noticing someting strange. |
| 10:33.19 | helloritesh | Bascailly, I create a special channel using AMI, answer it start ChanSpy and dump it into a local channel that records. |
| 10:33.54 | helloritesh | I noticed that if the channel I call from AMI is a remote channel, all is well but if I switch to a local channel, the call never goes to the context where I record the call |
| 10:34.06 | helloritesh | is there something different in 1.6? Used to work fine in 1.4 |
| 10:34.41 | helloritesh | basically, usign AMI Originate, if the channel is local, then the context/Exten/Prio never answers the call |
| 10:35.11 | helloritesh | if I switch the channel to remote channel usign AMI Originate, then the context/Exten/Prio answers fine... |
| 10:35.17 | helloritesh | that's the only difference |
| 10:36.10 | helloritesh | Anyone else noticed this? |
| 10:36.57 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-kwbihstuaxnncmzf) |
| 10:41.04 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 10:44.17 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 10:44.42 | hluesea | hello i need ring tone in dialplan for asterisk svn |
| 10:44.57 | hluesea | default ringing command is not allowed in svn :( |
| 10:45.50 | helloritesh | here is the code http://pastebin.ca/1805594 |
| 10:46.30 | tzafrir | Ceeriael, how does it check? Could you pastebin the trace? |
| 10:49.27 | Ceeriael | tzafrir found the error ... |
| 10:50.22 | helloritesh | Folks, anyone out there? |
| 10:50.22 | Ceeriael | it was my DNS service that asterisk wasent happy abount. |
| 10:50.22 | Ceeriael | tzafrir https://www.trixbox.org/forums/trixbox-forums/help/suddenly-everything-slow |
| 10:50.22 | Ceeriael | tzafrir thx for you help anyway ! it took me on the correct path! |
| 10:50.29 | hluesea | thank you <helloritesh |
| 10:50.43 | helloritesh | Thank you? |
| 10:50.55 | tzafrir | helloritesh, I suppose hluesea and Ceeriael are here |
| 10:51.04 | helloritesh | I see |
| 10:51.05 | Ceeriael | yep =) |
| 10:51.11 | tzafrir | pretends not to be here |
| 10:51.15 | benngard | is there a way to se what sound-files asterisk is playing when u for eaxmple dial into voice-mail? |
| 10:51.27 | helloritesh | cli? |
| 10:51.31 | helloritesh | that should help |
| 10:51.37 | tzafrir | benngard, a high anough verbosity level |
| 10:51.40 | helloritesh | asterisk -r |
| 10:51.45 | benngard | was looking for that but cant figure out how |
| 10:51.55 | benngard | did try set core verbose 10 |
| 10:51.57 | helloritesh | start asterisk as asterisk -vvvvvvvvvvvvgc |
| 10:52.10 | helloritesh | or login to cli and "set verbose 10" |
| 10:52.14 | tzafrir | I think 2 should be enough, but 3 will surely do |
| 10:52.25 | benngard | sec let me try |
| 10:52.38 | helloritesh | core set verbose 10 |
| 10:52.56 | helloritesh | anything from 1-10 |
| 10:52.58 | helloritesh | you pick |
| 10:53.01 | helloritesh | 3 is enough |
| 10:53.20 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 10:53.32 | *** join/#asterisk Tako-san (~Tako-san@p4022-ipad69osakakita.osaka.ocn.ne.jp) |
| 10:53.45 | benngard | 3 did the trick |
| 10:53.47 | benngard | thx |
| 10:54.15 | benngard | strange i am (almost) sure i did it with 10 and didnt se any result |
| 10:54.15 | spenguin[work] | hey, if I have dtmfmode=auto in sip.conf |
| 10:54.27 | spenguin[work] | is there a way to figure what mode is being actually used |
| 10:54.39 | spenguin[work] | or on what basis is the mode automagically selected? |
| 10:55.01 | helloritesh | spenguin[work] you can debug the channel |
| 10:55.38 | helloritesh | core set debug channel |
| 10:55.48 | helloritesh | and give the channel name or core set debug channel all |
| 10:55.52 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.163.204.dsl.dyn.forthnet.gr) |
| 10:55.53 | helloritesh | at CLI |
| 10:56.11 | spenguin[work] | ok |
| 10:56.21 | helloritesh | or the entire trunk by core set debug ip <trunk IP> |
| 10:56.28 | spenguin[work] | ok |
| 10:59.49 | *** join/#asterisk Madkiss (~madkiss@freenode/staff-emeritus/madkiss) |
| 10:59.52 | Madkiss | hi folks. |
| 10:59.59 | Madkiss | does anyone know about a C# interface for asterisk? |
| 11:05.27 | *** join/#asterisk areay (~areay@188-220-19-194.zone11.bethere.co.uk) |
| 11:06.50 | *** join/#asterisk Rajmohan (~raj@122.164.73.254) |
| 11:12.03 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 11:13.50 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 11:15.40 | areay | hi all... i was wondering if it's possible to have 2 register lines in sip.conf, both using the same provider... i seem to remember it not being possible previously, but i haven't tried since 1.6.x has been out (and i don't want to break my production system) |
| 11:16.11 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 11:16.16 | Pimmetje | Madkiss: http://www.voip-info.org/wiki/view/Asterisk+manager+Examples |
| 11:23.29 | ruyo | Did anyome manage to get hfcpci (1 port BRI driver) on kernel 2.6.26? |
| 11:25.17 | tzafrir | ruyo, for which version of mISDN? |
| 11:25.49 | ruyo | 1.1.8/9 |
| 11:25.53 | ruyo | Tested everyone. |
| 11:26.11 | ruyo | Well, it kind of works. |
| 11:26.22 | ruyo | I just can't unload the module after loading. |
| 11:29.17 | ruyo | The problem is simmilar to http://www.mail-archive.com/misdn-asterisk@lists.beronet.com/msg00711.html |
| 11:29.51 | ruyo | I've also posted on the ML, but no one was able to help me either. |
| 11:34.04 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 11:35.13 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:35.30 | areay | ok i've decided to start testing on my production system (1.6.1.13 i686) as nobody replied... luckily (i think because i added the new register line below my old one) my inbound calls are still working for the first peer... but i'm getting "chan_sip.c check_auth username mismatch" errors for the second peer; in other words, because i have 2 sip peers that need to register with one host, for some reason asterisk can't tell the differenc |
| 11:35.30 | areay | e. i am however able to add a second sip peer from *another* host (which doesn't solve my problem btw) |
| 11:36.23 | areay | am i doing something wrong? surely i can't be the only person to want to register 2 sip peers with the same host... |
| 11:36.54 | benngard | whats the best way to save voice-prompts, wanna have the best possible qulity, .wav? |
| 11:37.06 | areay | i use wav |
| 11:37.20 | areay | but then i don't care about space |
| 11:37.32 | areay | if you care about space use gsm or something |
| 11:37.44 | benngard | i dont care a shit about space |
| 11:37.50 | areay | go for wav then ;) |
| 11:37.53 | benngard | thx |
| 11:38.08 | areay | np |
| 11:39.16 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 11:39.39 | benngard | just 1 strange thing (or my not enough competent) did record se/vm-and.wav when i play it and watch in cli: Playing 'vm-and.slin' (language 'se'), is .slin the same as .wav? |
| 11:40.15 | benngard | i dont have any se/vm-and.slin |
| 11:40.31 | areay | i think .slin is what asterisk uses internally... |
| 11:40.32 | ruyo | I think slin is u/alaw |
| 11:40.50 | benngard | oki, thats explain, i am using alaw |
| 11:41.13 | benngard | so what i see is the "converted" format? |
| 11:42.57 | ruyo | You can tell what format you want your record in if you add the extension, if I'm not mistaken. |
| 11:43.15 | benngard | yesm playing around with it atm |
| 11:43.55 | helloritesh | Folks, I am still strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 |
| 11:44.04 | benngard | notice if i record a voice-prompt as alaw, Playing 'vm-password.alaw' (language 'se') |
| 11:44.16 | helloritesh | Not sure why the local channel won't send to context while the remote does |
| 11:44.47 | benngard | maybe it is better to record in alaw? i am always using alaw... |
| 11:44.48 | *** join/#asterisk nightrid3r (kvirc@41.214.154.95) |
| 11:45.16 | *** join/#asterisk nightrid3r (kvirc@41.214.154.95) |
| 11:45.24 | ruyo | If the voice channel is in alaw then it's the best format, since wav won't add more quality. |
| 11:49.10 | benngard | did notice that, did record se/vm-and.alaw and vm-and.wav, asterisk picked .alaw, Playing 'vm-and.alaw' (language 'se') |
| 11:58.41 | benngard | can any recommend a .gsm and .alaw player for windows? |
| 12:00.18 | *** join/#asterisk Crusnik01 (~mh@nat-01.core.rbnet.no) |
| 12:00.29 | *** join/#asterisk matata (~matata@Wikipedia/Bassem-JARKAS) |
| 12:01.22 | Crusnik01 | Hi! Anyone have any good referance to where one might find information on building desktop applications that interact with an Asterisk server? Like, when a call comes in, it performs some action? |
| 12:02.57 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 12:03.05 | *** join/#asterisk albasheers (~basheer@89.148.43.28) |
| 12:03.08 | *** part/#asterisk albasheers (~basheer@89.148.43.28) |
| 12:03.22 | ruyo | Crusnik01, have a look at http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 12:03.48 | ruyo | And http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 12:06.23 | Crusnik01 | thanks ruyo |
| 12:12.14 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 12:13.36 | *** join/#asterisk Godfather_ (~Godfather@193.Red-88-8-12.dynamicIP.rima-tde.net) |
| 12:14.20 | tzafrir | benngard, a SIP soft-phone, connected through Asterisk :-) |
| 12:16.21 | benngard | tzafrir: :) |
| 12:16.47 | benngard | quicktime can play .gsm but not .alaw :( |
| 12:17.15 | tzafrir | alaw is RAW PCM |
| 12:17.26 | tzafrir | convert it to wav with sox |
| 12:17.48 | benngard | could do that ofc |
| 12:17.55 | tzafrir | (well, G.711a-encoded, but just the same) |
| 12:18.03 | matata | Hi! Is there any document describes the way that allow/disallow extension (or group of extensions) using outbound routes ? |
| 12:18.37 | tzafrir | matata, yes, it's called the dialplan. .e.g extensions.conf |
| 12:19.05 | tzafrir | matata, do you ask with respect to some specific dialplan, such as the one used in FreePBX? |
| 12:19.35 | areay | i know somebody must know wtf i'm talkin bout... i want asterisk to handle multiple sip registrations from the same host/provider... right now it's getting confused |
| 12:20.38 | matata | tzafrir, yes, actually it's Elastix machine which is based on FreePBX |
| 12:21.49 | tzafrir | areay, is your problem with outgoing calls or with incoming calls? |
| 12:23.33 | tzafrir | matata, baiscally freepbx uses a single context for all the extensions. Thus you can't use the standard solution used by Asterisk for that (different contenxts) |
| 12:24.53 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:25.04 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 12:26.14 | areay | tzafrir, incoming... basically our company merged with another one and we're hosting the inbound calls for both... right now we're using 2 separate providers, one of which is expensive and they limit the number of inbound concurrent calls... we want to change so we use the same provider for both; i'm just testing it now, but asterisk is getting confused... if i have 2 register lines, and 2 sip peer sections in sip.conf i get a username |
| 12:26.15 | areay | <PROTECTED> |
| 12:26.31 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 12:27.15 | tzafrir | areay, how are incoming calls expected to be identified? |
| 12:27.47 | areay | by the sip account (auth name/username) |
| 12:27.48 | tzafrir | Do they actually authenticate on incoming calls? |
| 12:28.41 | areay | tzafrir, no idea... all i know is it works for one or the other, but not for both... i've tried separate contexts, and using the same context with different extensions |
| 12:29.32 | areay | there are posts on google with people having the same problem, and referring to it as a bug in asterisk... it seems this "bug" has been around for a long time... and it's starting to look like less of a bug and more of an antifeature |
| 12:30.19 | areay | it seems like someone made an initiative to rewrite chan_sip.c, but that was a long time ago and the links are all dead |
| 12:30.38 | areay | one would assume the changes made their way into recent versions of asterisk |
| 12:31.49 | areay | in fact, it seems like the digium-hosted links have been deliberately broken in some sort of strange attempt to influence people to buy digium's premium products/support |
| 12:34.56 | areay | anyone? |
| 12:37.55 | *** join/#asterisk cnu (~cnu@the.ultimate.lamer.la) |
| 12:38.29 | areay | forget everything else and somebody (for god's sake) please answer this one simple question: is it or is it not possible for asterisk handle multiple sip registrations to the same host/provider? |
| 12:39.33 | *** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net) |
| 12:44.38 | *** join/#asterisk cgc (~cgc@2001:0:53aa:64c:24ce:ac4:a13e:9c7f) |
| 12:44.42 | cgc | hi everyone |
| 12:44.49 | cgc | does anyone know how to get 'Cologne Chip Designs GmbH ISDN network controller [HFC-PCI]' working with asterisknow? |
| 12:45.05 | cgc | can I do this with dahdi? |
| 12:45.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:45.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:46.36 | kaldemar | areay: yes it is. remember that register lines have little to do with incoming calls. you need properly matching peer definitions and a dialplan. it depends on the provider how you must do it. |
| 12:49.04 | tzafrir | cgc, for dahdi you need the driver called zaphfc |
| 12:49.24 | tzafrir | e.g. from the dahdi-extra repo |
| 12:49.44 | cgc | tzafrir: ahh, cool, thanks :) |
| 12:57.52 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 12:59.54 | angryuser_ | ~fxs |
| 12:59.55 | infobot | [fxs] foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 13:07.01 | areay | kaldemar, i've set it up according to my provider's instructions... if i only have one register line, one peer definition, and one dialplan context for my provider it works (for one or the other), but if i list both i get a conflict |
| 13:08.11 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 13:12.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:18.57 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:19.27 | helloritesh | [TD]D-Fender... need your quick help. |
| 13:19.41 | helloritesh | I am still strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 |
| 13:19.41 | helloritesh | Not sure why the local channel won't send to context while the remote does |
| 13:21.04 | [TK]D-Fender | helloritesh: Link is no good |
| 13:21.14 | helloritesh | let me do that again |
| 13:21.35 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 13:21.50 | helloritesh | http://pastebin.ca/1805692 |
| 13:21.51 | helloritesh | Try this |
| 13:21.54 | helloritesh | please |
| 13:22.32 | romb | temporarily atsutstvuet |
| 13:23.18 | helloritesh | Basically, [record-call] context is not triggering if I used local channel. Triggers fine when I use remote channel |
| 13:23.32 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 13:23.49 | [TK]D-Fender | helloritesh: Show me the attempt |
| 13:24.02 | helloritesh | Attempt? |
| 13:24.13 | helloritesh | the AMI output? |
| 13:24.34 | [TK]D-Fender | helloritesh: AMI & CLI |
| 13:24.43 | helloritesh | sure |
| 13:26.06 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:26.13 | helloritesh | Here is the AMI output http://pastebin.ca/1805698 |
| 13:27.17 | *** join/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 13:27.19 | *** part/#asterisk Crusnik01 (~mh@nat-01.core.rbnet.no) |
| 13:27.32 | helloritesh | CLI doesn't have much to post...Only walks throught the call-record-leg |
| 13:27.42 | helloritesh | OMG.. Lost power |
| 13:28.00 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 13:28.24 | helloritesh | back |
| 13:28.28 | asteriskmonkey | sip (box a) -> sipbox(boxb) -> sip box(c) .... box a to b is good audio yet when call is passed to 3rd box audio is mangled.. anyone else experience this? |
| 13:29.10 | [TK]D-Fender | asteriskmonkey: What about B>C direct? |
| 13:29.18 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:29.21 | helloritesh | on CLI [record-call] never happens |
| 13:29.22 | asteriskmonkey | b->c is good direct |
| 13:29.41 | asteriskmonkey | funny thing is too when i do a->b(sip) then pass to c using iax its good |
| 13:29.58 | asteriskmonkey | of sip fro b-> c its good just not sip on multi box pass |
| 13:30.16 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 13:30.24 | asteriskmonkey | any ideas... ? |
| 13:31.13 | [TK]D-Fender | asteriskmonkey: what codecs involved? |
| 13:31.22 | asteriskmonkey | ulaw all the way |
| 13:32.13 | [TK]D-Fender | asteriskmonkey: Have you confirmed that reinvites have been disabled across the board? |
| 13:32.34 | asteriskmonkey | yep |
| 13:32.57 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:33.32 | asteriskmonkey | see why im stumped :) ? |
| 13:36.18 | [TK]D-Fender | asteriskmonkey: Yup... is this effect 100% consistent? where S>SIP, vs SIP>IAX always reacts as described? |
| 13:36.34 | asteriskmonkey | yes |
| 13:37.11 | [TK]D-Fender | asteriskmonkey: Idea : have A>B = IAX, and B>C = SIP and test |
| 13:37.14 | asteriskmonkey | sip>iax 100% works and sip>sip 100% works (sip only when b->c) |
| 13:37.19 | asteriskmonkey | cant |
| 13:37.26 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 13:37.28 | asteriskmonkey | A>B = metaswitch to asterisk |
| 13:37.29 | [TK]D-Fender | asteriskmonkey: Why not? |
| 13:37.35 | [TK]D-Fender | :/ |
| 13:37.57 | asteriskmonkey | damn class 5 switches and there non conformity to 3rd party protocol |
| 13:38.13 | [TK]D-Fender | asteriskmonkey: Only thing I can think of is if B>C = hostile path to SIP by mean ISP |
| 13:38.20 | asteriskmonkey | no |
| 13:38.24 | asteriskmonkey | canreinvite = no |
| 13:38.34 | asteriskmonkey | so the media path is being mangled somehow by asterisk |
| 13:38.39 | [TK]D-Fender | asteriskmonkey: I mean f C's ISp is screwing with you on purpose |
| 13:38.47 | asteriskmonkey | no |
| 13:38.50 | asteriskmonkey | its all local |
| 13:38.57 | [TK]D-Fender | asteriskmonkey: Crap. |
| 13:39.19 | asteriskmonkey | yeah my thoughts exacrtly |
| 13:39.34 | [TK]D-Fender | asteriskmonkey: Only thing left is checking jitter-buffer status. |
| 13:39.50 | asteriskmonkey | but there should be not jitterbuffer required |
| 13:39.55 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 13:40.45 | [TK]D-Fender | asteriskmonkey: One would think... sometimes bad switches + PL = ick |
| 13:40.53 | asteriskmonkey | jb is on by default, but ill try packing it with some more ms.. i think though this might be some f'd up rtp timing issue in 1.4.x ... will have to roll up a lab setup of 1.6.x if i cant get an answer here |
| 13:41.13 | asteriskmonkey | i have 80k worth of switching here :/ not my switches... |
| 13:41.17 | trontonic | hi, I'm a beginner when it comes to Asterisk. Currently, I'm trying to configure an Asterisk server (with FreePBX) for SIP. Outbound calls work fine, but not incoming. What are the usual steps to be taken to start debugging and finding the problem? I've tried "asterisk -vvvvr", but I find it hard to figure out why incoming calls aren't routed like I want them to. Any suggestion would be great. Thanks |
| 13:41.39 | asteriskmonkey | trontonic: piss off this is not freepbx support |
| 13:42.00 | trontonic | asteriskmonkey: Asterisk support is sufficient |
| 13:42.17 | asteriskmonkey | k, you did your own build, setup your own sip... |
| 13:42.31 | asteriskmonkey | your own extensions.conf? or you use that gui wizard |
| 13:42.36 | trontonic | asteriskmonkey: no, someone else did and I'm the guy that has been given the task to fix it |
| 13:43.10 | [TK]D-Fender | asteriskmonkey: Umm... cool it a bit... |
| 13:43.12 | asteriskmonkey | ah, join #freepbx, we can help you from a command line persepective but freepbx generates code like mad |
| 13:43.30 | [TK]D-Fender | ~freepbx |
| 13:43.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:43.49 | trontonic | I don't neccesarily need freepbx support, Asterisk support would be just fine |
| 13:43.53 | [TK]D-Fender | trontonic: Indeed FreePBX is nto supported here. They can help you 2 doors to the left. |
| 13:44.06 | trontonic | Well, forget I mentioned FreePBX, I need Asterisk support |
| 13:44.21 | [TK]D-Fender | trontonic: in what way are calls not routed like tyou expect? |
| 13:44.49 | helloritesh | [TD]D-Fender: Anything funny you spot in the AMI response? http://pastebin.ca/1805698. Just wondering why the call is not dumping into [record-call |
| 13:44.53 | helloritesh | context |
| 13:45.26 | trontonic | [TK]D-Fender: when I call in with my cellphone, I get a busy signal. I would expect it to be routed to one of the phones, but I don't know where to start when it comes to debugging, except "asterisk -vvvvr" |
| 13:45.34 | [TK]D-Fender | helloritesh: I never see priority4 executing <- |
| 13:45.47 | [TK]D-Fender | trontonic: And do you see anything there? |
| 13:46.27 | trontonic | [TK]D-Fender: yes, a lot of output. The "sip" command also gives me useful output. I see that I'm correctly registered with the external SIP-vendor (if that's the correct term) and that the right IP-phones are online and happy. |
| 13:46.47 | asteriskmonkey | what context do you have your sip provider gong to? |
| 13:46.57 | trontonic | [TK]D-Fender: When I call inn, my phone number shows up, and it gets 200 ACK, but I'm having a hard time interpreting all the output |
| 13:47.03 | helloritesh | When the the actual transfer slated? I mean, I thought the moment the Channel answers, the call is sent to the assigned Context/Exten/Priority, no? |
| 13:47.22 | helloritesh | I might not have complete capture. I will make some edits and capture as much data |
| 13:47.42 | asteriskmonkey | helloritiesh, if your using 1.6.x you have to add originate to your ami permissions |
| 13:47.56 | trontonic | asteriskmonkey: in several configuration files, lines start with "context=". Which configuration file will tell me which context I have my sip provider going to? |
| 13:47.56 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:48.09 | trontonic | asteriskmonkey: sip.conf? |
| 13:48.13 | asteriskmonkey | you need to find in the sip.conf |
| 13:48.27 | asteriskmonkey | the provider name in [provider-a] and look for the line context=.... |
| 13:48.37 | asteriskmonkey | the ... will be where the calls go to in the extensions.conf |
| 13:48.42 | trontonic | it's going to context=default |
| 13:48.51 | [TK]D-Fender | trontonic: PASTEBIN the failed call attempt with SIP DEBUG enabled |
| 13:48.53 | [TK]D-Fender | ~!pb |
| 13:48.54 | [TK]D-Fender | ~pb |
| 13:48.55 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:48.57 | [TK]D-Fender | ^^^^^^66 |
| 13:48.59 | asteriskmonkey | do in default in extensions.conf look at the rules |
| 13:49.06 | Ceeriael | I have a strange problem ;) I have en external SIP Xlite phone... i can Call it with sound, but when i call from it. i get no sound (call gets though) ... no sound in both directions.... |
| 13:49.11 | trontonic | [TK]D-Fender: ok, thank you |
| 13:49.27 | Ceeriael | So calling it internal -> External works.,... but External -> internal gives me no sound. |
| 13:49.28 | asteriskmonkey | Ceeriael: sounds like firewall issue |
| 13:49.33 | *** join/#asterisk SirThomas (~tomc@mail.kendeco.com) |
| 13:49.40 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
| 13:49.49 | Ceeriael | but wouldnt it also fuck up internal -> external ? |
| 13:50.07 | asteriskmonkey | no, some firewalls create openings based on outgoing origination |
| 13:50.07 | Ceeriael | also, this was working till some days ago.... then it "just stoped" working.... :S |
| 13:50.24 | asteriskmonkey | use ziopper and iax much better for firewalls |
| 13:50.36 | Ceeriael | do you know an IAX client ? |
| 13:50.44 | asteriskmonkey | i just said one "zoipper" |
| 13:50.50 | Ceeriael | ahh =) |
| 13:50.54 | [TK]D-Fender | ~zoiper |
| 13:50.55 | infobot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 13:50.57 | Ceeriael | tells a bit about my knolage ;) |
| 13:50.58 | asteriskmonkey | ist free :) |
| 13:50.59 | *** join/#asterisk cgc (~cgc@2001:0:53aa:64c:389f:2faf:a13e:9c7f) |
| 13:51.09 | Ceeriael | free is gooooood ;) |
| 13:51.32 | [TK]D-Fender | OSS = better |
| 13:51.37 | cgc | to install zaphfc do I need to remove dahdi and install zaptel? |
| 13:51.51 | Ceeriael | well it cant all be gold. |
| 13:53.03 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
| 13:53.57 | asteriskmonkey | cgc: why are you going backwards in technology? dahdi replaced zap |
| 13:54.09 | asteriskmonkey | your also have to roll back your asterisk... mmm alot :) |
| 13:54.37 | [TK]D-Fender | asteriskmonkey: Says who |
| 13:54.53 | asteriskmonkey | your using zaptel drivers with 1.6.2? |
| 13:54.54 | cgc | asteriskmonkey: so I can get zaphfc to work with dahdi?, I need to get the following card to work: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] |
| 13:54.57 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 13:55.09 | [TK]D-Fender | asteriskmonkey: Who says he's on 1.6.2? |
| 13:55.25 | cgc | im on 1.4.24 |
| 13:55.35 | [TK]D-Fender | ^^^ |
| 13:55.36 | asteriskmonkey | hes asking if hes having to uninstall dahdi which indicates hes at the cutover period from zap |
| 13:55.44 | [TK]D-Fender | asteriskmonkey: incorrect |
| 13:55.56 | asteriskmonkey | enlighten, im all eyes :) |
| 13:56.03 | [TK]D-Fender | asteriskmonkey: 1.4 supports either through current, and is LTS |
| 13:56.27 | asteriskmonkey | thanks |
| 13:56.40 | asteriskmonkey | do you know if mfcr2 is support nativly in 1.6.x and dhadi? |
| 13:56.46 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:57.08 | [TK]D-Fender | asteriskmonkey: I've heard Newest versions of things are going to have proper r2 across the board... not sure if that's 1.6.2 or 1/8 |
| 13:57.11 | [TK]D-Fender | 1.8 |
| 13:57.18 | tzafrir | asteriskmonkey, mfcr2 is included in chan_dahdi as of 1.6.2 . |
| 13:57.38 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-spbiujxzbqxbjimw) |
| 13:57.53 | asteriskmonkey | thanks, now i can finally through away my last asterisk 1.2.x box in brasil and upgrade without worry :) |
| 13:58.17 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:02.42 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 14:03.55 | *** join/#asterisk Grombels (~info@pd95b5894.dip0.t-ipconnect.de) |
| 14:05.00 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 14:05.39 | Grombels | hello someone here who speaks german ? |
| 14:05.46 | *** join/#asterisk sasargen (~chatzilla@70.42.35.159) |
| 14:06.03 | Chotaire | nein |
| 14:06.34 | tzafrir | Isn't there a #asterisk-de channel? |
| 14:07.05 | ruyo | Has anyone tried Debian's asterisk package? |
| 14:07.26 | Grombels | ok i see i found the channel thks for support |
| 14:07.52 | kaldemar | the german channel is #asterisk.de |
| 14:09.13 | asteriskmonkey | ruyo: yes the debian packages work nice (im using ubuntu variant |
| 14:09.29 | Grombels | at least i have installed the packages from debian lenny its version 1.4.21 |
| 14:10.17 | ruyo | asteriskmonkey, do you have any analog and/or BRI card on it? |
| 14:10.42 | asteriskmonkey | ruyo: no just sip, using gateways on that one |
| 14:10.55 | ruyo | Ok, thanks anyway. |
| 14:11.03 | ruyo | I'll try it out. |
| 14:11.04 | Katty | good morning |
| 14:11.04 | asteriskmonkey | ruyo: if you have analog cards see if dahdi is in your repository |
| 14:12.07 | [TK]D-Fender | 1.4.21= Zaptel. 1.4.22 = start of DAHDI |
| 14:12.08 | *** join/#asterisk captiancrash (~jonathan@12.71.218.232) |
| 14:12.08 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:12.36 | tzafrir | ruyo, /me has |
| 14:13.03 | Katty | Qwell: soooo |
| 14:13.06 | Katty | Qwell: you get your package yet |
| 14:13.07 | ruyo | Can I compile dahdi if I need analog drivers and asterisk will detect it? |
| 14:13.09 | *** join/#asterisk Skeeter- (~Skeeter@190-141.cgocable.ca) |
| 14:13.18 | ruyo | dahdi and mISDN? |
| 14:13.28 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 14:13.32 | beek | Morning [TK]D-Fender |
| 14:13.35 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 14:13.35 | beek | hugs Katty |
| 14:13.35 | tzafrir | ruyo, what version of asterisk? What card? |
| 14:14.07 | ruyo | My problem is with B100P (Openvox) because of hfcpci on Debian Lenny. |
| 14:14.09 | areay | im trying to get asterisk to register to two separate accounts with the same sip provider. i've tried just adding register lines, peer definitions, and dialplan rules for each account, and they work independently, but not together... if this is a limitation of asterisk, would using opensips as a sip proxy provide a solution? |
| 14:14.11 | ruyo | Kernel 2.6.28 |
| 14:14.26 | Katty | hugs beek |
| 14:14.26 | ruyo | Asterisk 1.4.2x |
| 14:14.33 | Katty | beek: i lost a ferret the other night :< |
| 14:14.52 | ruyo | *kernel 2.6.26, sorry |
| 14:14.57 | beek | Katty: As in it died or it got away? |
| 14:15.02 | Katty | beek: he died. |
| 14:15.07 | michael-i | I'm seeing "chan_dahdi.c:8146 in handle_init_event: Unable to play dialtone on channel 2, do you have defaultzone and loadzone defined?" in my latest builds (Dahdi 2.2.0 + Asterisk 1.6.1.14). My settings are the same as those that worked before. Any pointers? |
| 14:15.16 | beek | I'm sorry to hear that. What happened to the little guy? |
| 14:15.17 | tzafrir | ruyo, IIRC that driver indeed won't build with kernel >= 2.6.27 (and I can provide you patches, if you really want) |
| 14:15.19 | Katty | beek: but he had a long happy life, i think |
| 14:15.22 | Katty | beek: not sure :/ |
| 14:15.25 | beek | Just old age? |
| 14:15.27 | Katty | beek: didn't even know he was ill. |
| 14:15.32 | Katty | beek: not sure of that either. |
| 14:15.40 | Katty | beek: when i adopted him, they didn't know how old he was |
| 14:15.41 | ruyo | tzafrir, sorry, I meant 2.6.26 |
| 14:15.49 | beek | Ahh... did you get him from a shelter? |
| 14:15.51 | Katty | beek: i guessed he was 2 or 3 when i adopted him |
| 14:16.05 | Katty | beek: not exactly, i adopted him from a couple that was going through a divorce |
| 14:16.05 | areay | [TK]D-Fender, i know you can help me with this one |
| 14:16.17 | tzafrir | ruyo, standard packages should build. One of the maintaiers of the package has a HFC-S card |
| 14:16.22 | beek | How old was he (including your guess of 3) |
| 14:16.29 | Katty | about 5 |
| 14:16.38 | beek | Is that the normal lifespan of those critters? |
| 14:16.46 | Katty | 5-7 yeah |
| 14:16.51 | tzafrir | michael-i, can you post your /etc/dahdi/system.conf ? |
| 14:17.02 | beek | He had a good life then... I'm sure he loved your household. |
| 14:17.05 | *** part/#asterisk sasargen (~chatzilla@70.42.35.159) |
| 14:17.07 | [TK]D-Fender | areay: I have no reason to trust you did it right and you haven't shown anything yet. |
| 14:17.15 | Katty | beek: i sure hope so :< |
| 14:17.35 | ruyo | tzafrir, I'm going to try the package, although my problem is with mISDN's hfcpci. |
| 14:17.48 | areay | [TK]D-Fender, gettin info now brb |
| 14:18.03 | TheDavidFactor | Good morning all! Does anyone know if you can run the 32bit version of Fax For Asterisk on a 64bit version of CentOS? |
| 14:18.06 | tzafrir | ruyo, I'm not familiar with mISDN. |
| 14:18.35 | tzafrir | (it's been kept out of Debian due to inherent stability issues) |
| 14:18.52 | captiancrash | when i dail out on our PRI using the dial() command, local and long distance calls result in a tone for about 20 seconds then the message " You have no sent enough digits, please try again". I'm sending a 10 or 11 digit numbers, which have worked fine up until now. |
| 14:19.00 | michael-i | tzafrir: will do, waiting to setup the test system again here in my lab (2 mins.) |
| 14:19.20 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:19.53 | captiancrash | toll-free numbers work as expected, meaning the call completes. ... there have been no configuration changes to asterisk, should i suspect the providor, or is there somewhere else i need to look? i'm out of ideas. |
| 14:19.57 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:20.18 | ruyo | tzafrir, yes, I know. I need it for the BRI card though. I'll have it a go with the package anyway. :> |
| 14:20.40 | coppice | The David Factor: if you run a complete 32 bit Asterisk it should. otherwise you can run 64 bit Asterisk with app_fax |
| 14:21.24 | [TK]D-Fender | captiancrash: Where si the complete call attempt with PRI debug for us to examine? |
| 14:21.25 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:21.26 | [TK]D-Fender | ~pb |
| 14:21.27 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:21.28 | [TK]D-Fender | ^^^^^^^^ |
| 14:24.41 | captiancrash | [TK]D-Fender, successful call to 18002472020 http://pastebin.ca/1805738 |
| 14:24.54 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-457f22ad.dyn.optonline.net) |
| 14:24.57 | FuriousGeorge | hey all |
| 14:25.10 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:25.14 | captiancrash | [TK]D-Fender, failed call to 12703652024 <http://pastebin.ca/1805737> |
| 14:25.18 | FuriousGeorge | is multiparking / metermaid finally in the newest *? |
| 14:25.44 | FuriousGeorge | or is it still a patch that only works with version 1.X.Y.Z.A.B.C |
| 14:26.29 | Kobaz | metermaid? |
| 14:26.52 | FuriousGeorge | Kobaz: multiple "parking lots" |
| 14:26.54 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:27.06 | michael-i | tzafrir: my system.conf (zone loading problem...) http://pastebin.ca/1805740 |
| 14:27.18 | FuriousGeorge | so that usera can park on 70X-70Y and userb can park on 80X-80Y, etc |
| 14:27.39 | michael-i | tzafrir: I've had this problem with these custom drivers and a report from the Askozia community of the same error message on a trusted digium analog card |
| 14:27.43 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:28.22 | FuriousGeorge | Kobaz: there used to be a patch called metermaid, which i never tried, but im a bit behind on the latest asterisk releases... kinda moved on after 1.4.21.something |
| 14:28.25 | coppice | ooh, trusted analogue computing :-) |
| 14:28.35 | captiancrash | [TK]D-Fender, is that what is needed? |
| 14:28.53 | michael-i | s/trusted/previouslyworking ;) |
| 14:29.46 | *** join/#asterisk seanbright (~chatzilla@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:29.53 | tzafrir | michael-i, is this freebsd? |
| 14:30.21 | FuriousGeorge | http://www.voip-info.org/wiki/view/Asterisk+call+parking <--- leads me to believe that somehow multiparking is still not in asterisk after all, but perhaps the docs are just old, can anyone verify? |
| 14:30.26 | michael-i | tzafrir: no, this is my own Linux distro now. |
| 14:31.02 | FuriousGeorge | (for those that don't know, I'm talking about the functionality provided by the metermaid patch) |
| 14:31.27 | tzafrir | michael-i, so what do you see on cat /proc/dahdi/* #? |
| 14:32.09 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 14:33.23 | michael-i | tzafrir: cat + dahdi_scan output http://pastebin.ca/1805748 |
| 14:33.33 | Grombels | iam searching for an person who could teach me configure and build an asterisk server |
| 14:34.16 | beek | ~book |
| 14:34.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:34.21 | beek | Grombels: ^^^^^^^^^^^^^^^^^^^^^ |
| 14:34.32 | [TK]D-Fender | captiancrash: * appears to be the side that decided to hangup |
| 14:34.41 | Grombels | already read this books |
| 14:35.41 | Grombels | but its more sucessfull to have an helping hand |
| 14:35.51 | ManxPower-work | ~ask |
| 14:35.51 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:36.13 | [TK]D-Fender | captiancrash: 38 Progress from telco. 50 * Decides to stop. |
| 14:37.04 | helloritesh | [TK]D-Fender: Should the context be called as soon as the local channel answers the call? |
| 14:37.16 | *** join/#asterisk x86 (~x86@p3m/member/x86) |
| 14:37.21 | helloritesh | specially, this has been working fine in 1.4. I might downgrade now |
| 14:37.21 | [TK]D-Fender | helloritesh: I see no answer. |
| 14:37.28 | areay | [TK]D-Fender, http://pastebin.com/m449cfedf here's all the relevent info... i didn't do a full sip debug tho so if you need that just say |
| 14:37.38 | helloritesh | It happens |
| 14:37.44 | helloritesh | i am trying to capture more log |
| 14:37.46 | areay | there are error messages from asterisk tho |
| 14:37.55 | helloritesh | even the record and other things happnes |
| 14:38.01 | helloritesh | the exact code works for 1.4 |
| 14:38.19 | captiancrash | [TK]D-Fender, at about line 45 on the paste, I hang up the call. after I hear the "you have no dialed enough digits". |
| 14:38.21 | [TK]D-Fender | helloritesh: Whatever. I don't see as much as i'm suppoed to for this. |
| 14:38.28 | *** part/#asterisk Grombels (~info@pd95b5894.dip0.t-ipconnect.de) |
| 14:38.28 | tzafrir | michael-i, I noticed that channels 5 and 6 are not in use by Asterisk. Is that intentionally, or part of the problem? |
| 14:38.30 | [TK]D-Fender | areay: type=peer |
| 14:38.32 | helloritesh | sure.. i am posting in a couple of minutes |
| 14:38.53 | areay | [TK]D-Fender, lemme try that |
| 14:39.15 | [TK]D-Fender | areay: Barring that, I'd check type=user. You'd then need to make 2 peers for them. |
| 14:39.42 | michael-i | tzafrir: that's another module which has nothing attached to it at the moment. I could remove it and regenerate the .conf files for cleaner output if you'd like. |
| 14:40.35 | areay | [TK]D-Fender, sorry i don't understand what you mean there |
| 14:40.52 | [TK]D-Fender | areay: try peer first, then user |
| 14:41.03 | Kobaz | do de do |
| 14:41.07 | Kobaz | i gots a qrestion |
| 14:41.16 | areay | but for user i need to define separate peers? |
| 14:41.39 | Kobaz | i have a sangoma t1 hooked up to an avaya, and i'm getting static from the asterisk t1 to the avaya analog lines |
| 14:41.52 | Kobaz | any ideas what would be the cause of the static? |
| 14:42.15 | Kobaz | it comes and goes... it usually happens if there's been no activity for a while... so i'll call in, and the beginning of a track will be all staticy... and then it will be good for a while |
| 14:42.32 | areay | exit |
| 14:42.34 | Kobaz | i have this same problem on another system hooked up to an avaya as well... except the other avaya system, is all digital stations |
| 14:42.35 | areay | oops |
| 14:43.06 | Kobaz | someone told me something about VAD? |
| 14:44.32 | areay | [TK]D-Fender, both peer and user give the same result... |
| 14:44.53 | Kobaz | i've turned off echo cancellation on the t1 card, and i still get static |
| 14:45.24 | areay | i've also tried using the same context for both peers, and then defining the extensions inside of the context... but i get the same result |
| 14:45.29 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.22.80) |
| 14:46.01 | *** join/#asterisk evangelion (~manzy_zet@212.183.163.203) |
| 14:46.10 | [TK]D-Fender | areay: insecure=port,invite <- set to "no" and use "user" |
| 14:47.31 | areay | [TK]D-Fender, do i have to set both to user, or just the affected one? only reason i ask is because i think i may have temporarily fubar'd the outbound by changing it lol |
| 14:47.31 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 14:47.42 | areay | the one that's working is the one we use for outbound btw |
| 14:48.00 | [TK]D-Fender | areay: to both, And as i say you wil have to make matching peers to dial out |
| 14:48.08 | Kobaz | i've never seen static on t1 lines before this... all systems are completely under my control so it should be able to be tracked down |
| 14:48.30 | areay | [TK]D-Fender, with all the same info, except for the type right? it doesn't matter if they're named the same, right? |
| 14:48.55 | *** join/#asterisk wikki (~wikki@doraemon.wikki.com) |
| 14:49.08 | [TK]D-Fender | areay: the [name] should be different. ti matches on username for your user, so that one should be named a little different |
| 14:49.42 | areay | ah so the name doesn't really matter... that's interesting to know |
| 14:50.31 | areay | as long as it's unique obviously |
| 14:50.31 | wikki | anyone know where I can find a good tutorial on how to setup a sip to sip gateway with asterisk? |
| 14:50.39 | ManxPower-work | Wow, many n00bs here today that think Asterisk is as simple as installing vi. |
| 14:51.06 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 14:51.07 | seanbright | it is. make ; make install ; make config ; make samples |
| 14:51.14 | seanbright | bing bang boom - you're done |
| 14:51.22 | [TK]D-Fender | wikki: ... |
| 14:51.24 | [TK]D-Fender | ~book |
| 14:51.25 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:51.25 | helloritesh | [TK]D-Fender. Here the CLI and the entire AMI response http://pastebin.ca/1805763 |
| 14:51.26 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 14:51.31 | wikki | apt-get install asterisk |
| 14:51.49 | ManxPower-work | wikki: That will surely not work the way you want |
| 14:52.18 | wikki | thanks for the links |
| 14:52.26 | wikki | why do you say that manxpower-work? |
| 14:52.50 | ManxPower-work | wikki: 1) because you have no idea what the packager did 2) we don't support packaged Asterisk's here. |
| 14:53.07 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:18a7:546d:a6d0:dcfd) |
| 14:53.13 | [TK]D-Fender | helloritesh: Answer first, playback silence/2 then proceed. pastebin the new attempt |
| 14:53.17 | wikki | oh the apt-get |
| 14:53.27 | helloritesh | Sure |
| 14:53.28 | helloritesh | thx |
| 14:55.19 | areay | [TK]D-Fender, did that, outbound still works this time, but i'm getting the same error as before :( |
| 14:55.45 | [TK]D-Fender | areay: pastebin new configs |
| 14:58.20 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:04.22 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:05.49 | areay | [TK]D-Fender, http://pastebin.com/m688a4eea |
| 15:07.18 | [TK]D-Fender | areay: Ok, not sure at this point... |
| 15:08.59 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:18a7:546d:a6d0:dcfd) |
| 15:10.21 | areay | [TK]D-Fender, yea it's kinda weird... would using opensips as a sip proxy fix my problem? |
| 15:10.46 | [TK]D-Fender | areay: no idea |
| 15:10.59 | [TK]D-Fender | areay: I'm sure * can do this fine... jsut not sure on what I've missed... |
| 15:11.00 | Chainsaw | What is this slin format that Asterisk keeps whining about? [2010-02-22 15:09:02] NOTICE[6841]: channel.c:2960 __ast_read: Dropping incompatible voice frame on SIP/729-00000000 of format slin since our native format has changed to 0x4 (ulaw) |
| 15:11.08 | Chainsaw | I haven't *allowed* slin. Just ulaw & alaw. |
| 15:11.21 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:11.29 | p3nguin_ | slin is asterisk's native format, I believe. |
| 15:11.32 | ManxPower-work | sln is the intermediate format when transcoding. |
| 15:11.56 | Chainsaw | Alright, so playing a message first and then switching to fax is not possible yet? |
| 15:12.14 | areay | [TK]D-Fender, i hope it can, because using opensips would require me to reconfigure everything |
| 15:12.19 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 15:12.34 | Chainsaw | That seems to be what's upsetting it, the actual application did what I wanted: [2010-02-22 15:09:02] NOTICE[6841]: app_nv_backgrounddetect.c:208 nv_background_detect_exec: Redirecting SIP/729-00000000 to fax extension |
| 15:13.04 | Chainsaw | It's just that ReceiveFax is now trying to transmit: [2010-02-22 15:09:15] WARNING[6841]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=48: Disconnected after permitted retries. |
| 15:14.10 | ManxPower-work | That's a totally different thing. |
| 15:14.27 | ManxPower-work | Sounds like you are trying to do FaxOverVoiceOverIP |
| 15:17.50 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 15:19.05 | helloritesh | [TK]D-Fender: Here is the entire thing again http://pastebin.ca/1805792 |
| 15:19.16 | helloritesh | This time with Playback like you suggested |
| 15:19.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:19.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:21.20 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:21.57 | Chainsaw | ManxPower-work: Yes. I'm trying to deal with a bodged divert. |
| 15:22.18 | Chainsaw | ManxPower-work: All my DDIs are coming in on a single number (for now) and I'd like to take the fax calls away and deal with them. |
| 15:22.24 | ManxPower-work | 1) Huh? 2) don't expect fax over voice over IP to work over SIP. |
| 15:23.02 | [TK]D-Fender | helloritesh: I don't see it triggering on answer.. this is just screwed up somehow... |
| 15:23.14 | Chainsaw | ManxPower-work: The fax machine is on an analog gateway which is T38-capable. |
| 15:23.15 | helloritesh | yeah |
| 15:23.21 | helloritesh | that's what I was suggesting |
| 15:23.26 | helloritesh | I think there is a bug in 1.6.1 |
| 15:23.34 | helloritesh | can not think of anything else |
| 15:24.05 | Chainsaw | ManxPower-work: I have a range of numbers being diverted. Instead of xx29 ending up on xx29, everything ends up on xx00. |
| 15:24.48 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:24.53 | ManxPower-work | Chainsaw: that's all well and good, except you are not using T.38 |
| 15:25.13 | Chainsaw | ManxPower-work: I noticed that, yes. |
| 15:25.43 | Chainsaw | ManxPower-work: Presumably this "I am ReceiveFAX and I am *sending* stuff" has something to do with it. |
| 15:25.50 | ManxPower-work | I wish you the BEST of luck, but you are crazy to expect FaxOverVoiceOverIP to work. |
| 15:26.20 | p3nguin_ | It works pretty good for me. I don't do a ton of faxing, though. |
| 15:26.34 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 15:26.35 | Chainsaw | p3nguin_: In ManxPower's world, a lot of things don't work though. |
| 15:26.43 | p3nguin_ | I know. |
| 15:27.29 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
| 15:27.54 | sbrath | Not sure what I have wrong, but with both my Cisco 7960 and Grandstream phones calling into a Credit Card company and using DTMF dosen't quite work. |
| 15:28.23 | ManxPower-work | sbrath: What Digium card are you using? |
| 15:28.30 | sbrath | On the Cisco when you press, say 9, it dosen't send what the callee is expecting, and takes you to a menu. |
| 15:28.36 | sbrath | TE110P |
| 15:28.41 | sbrath | with a PRI |
| 15:28.55 | ManxPower-work | set your tonelength to 500 |
| 15:29.13 | sbrath | ok, I'll try that. |
| 15:29.28 | p3nguin_ | I've also seen results change when using different dtmf modes in sip.conf. |
| 15:29.41 | ManxPower-work | p3nguin_: that helps less often than people think |
| 15:30.16 | sbrath | Will that also fix that on the Grandstream it seems that when I go thru the remotes "menu", I select a menu option by pressing 1, then the remote party starts speaking but I never hear what they said... |
| 15:30.28 | ManxPower-work | sbrath: one thing at a time |
| 15:30.33 | sbrath | ok |
| 15:30.45 | sbrath | let me try the tonelength first. |
| 15:33.12 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 15:33.40 | sbrath | tonelength != toneduration |
| 15:33.41 | sbrath | ? |
| 15:33.50 | ManxPower-work | it might be toneduration |
| 15:34.11 | *** join/#asterisk jdnWEST (~jdn@mx1.westparkcom.net) |
| 15:34.39 | *** join/#asterisk matteo` (~matteo@openwrt/developer/matteo) |
| 15:34.39 | *** join/#asterisk jdnWEST (~jdn@mx1.westparkcom.net) |
| 15:35.56 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:36.28 | sbrath | ManxPower-work: I'll have to do a complete reboot to change that it seems. |
| 15:37.08 | ManxPower-work | sbrath: no, you don't. |
| 15:37.09 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:37.30 | ManxPower-work | module unload chan_dahdi.so / module load chan_dahdi.so |
| 15:39.09 | sbrath | ok, I still have to unload the module so I'll take the whole switch down for a minute while I do that... Maybe over lunch I can sneak it in.. |
| 15:39.42 | *** join/#asterisk luca`gervasi (~ashura@95.239.172.120) |
| 15:39.44 | luca`gervasi | Hallo |
| 15:40.09 | ManxPower-work | sbrath: It will take down ZAP CALLS, not all calls. |
| 15:42.38 | luca`gervasi | hallo. i have some throuble configuring res_config_mysql (for realtime loading of my extensions from mysql database). is anybody using it? |
| 15:42.42 | vader-- | adtran is sending me a demo unit |
| 15:42.48 | vader-- | of that total access 924e |
| 15:44.54 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:44.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:46.19 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-457f22ad.dyn.optonline.net) |
| 15:50.08 | *** join/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
| 15:50.56 | *** join/#asterisk Greek-Boy (~Monching@41.188.154.137) |
| 15:51.34 | cgc | hi |
| 15:52.02 | cgc | i've been trying to get zaphfc working but I cannot start asterisk now |
| 15:52.06 | cgc | http://pastebin.ca/1805811 |
| 15:52.19 | cgc | have I configured dahdi wrong or something? |
| 15:54.38 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 15:54.47 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 15:55.11 | *** join/#asterisk chazzm (~chazz@173-24-238-25.client.mchsi.com) |
| 15:55.21 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 15:55.30 | *** join/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
| 15:55.53 | imcdona | what is the command I can use to check what digum hardware is in a machine? |
| 15:56.19 | chilicuil | hi there, I'm new to asterisk and I've set up my first asterisk installation and everything is working great, but still I've a little issue, I cannot deal numbers bigger than 10 digits, what's the parameter I'm missing? |
| 15:56.40 | elred_ | imcdona, dahdi_cfg -vv ? |
| 15:56.50 | [TK]D-Fender | chilicuil: What are you dialing on? What error do you get? |
| 15:56.52 | tzafrir | imcdona, dahdi_hardware |
| 15:56.59 | imcdona | ok ty |
| 15:57.34 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 15:57.55 | *** join/#asterisk edwin_quijada (~macaruchi@200.26.172.50) |
| 15:57.58 | edwin_quijada | Ji! |
| 16:00.11 | chilicuil | I'm getting this error, [TK]D-Fender "-- Got SIP response 603 "Declined" back from 200.76.112.13", but it's because that number doesnt exist, I wanna deal this number: 044 55 35037481 but at the logs I see asterisk deal this one: "SIP/5512048885/0445535037,50,Ttm") |
| 16:00.18 | edwin_quijada | i AM have a link with an Avaya PBX.H323, for this link it sends me calls to go into an IVR to capture data, my question is this link has 23 channels in Avaya side defined , how can I receive all channels into my IVR ? |
| 16:00.44 | chilicuil | as you can see it split the number at 0445535037, [TK]D-Fender |
| 16:01.20 | [TK]D-Fender | chilicuil: I don't see your actual dialplan, so i don't know what # you started with. PASTEBIN your dialplan and the complete call attempt with SIP DEBUG enabeld. |
| 16:01.21 | [TK]D-Fender | ~pb |
| 16:01.22 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:01.23 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:02.49 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:06.47 | Skeeter- | is it possible to have 2 reg server with the same extensions? |
| 16:07.11 | Skeeter- | [TK]D-Fender, Polycom guru, can you have more then 1 phone on the same extension? |
| 16:07.11 | *** join/#asterisk matteo (~matteo@openwrt/developer/matteo) |
| 16:07.26 | sbrath | It's possible, but both phones have to support BLA |
| 16:07.35 | sbrath | Bridged Line Appearance |
| 16:07.50 | sbrath | sometimes also called Shared Call Appearance. |
| 16:07.55 | chilicuil | ok, [TK]D-Fender, my dealplan to manage outgoing calls is here, http://www.pastebin.com/m5949af43 and the debuging output is here http://www.pastebin.com/m29610093, thx for your time |
| 16:08.43 | sbrath | Skeeter-: I've usually just registered both phones as a XXXX-a and XXXX-b and re did the dialplan to DIAL(SIP/XXXX-a&SIP/XXX-b) etc. |
| 16:08.50 | *** join/#asterisk jpcansa (~jpbenavid@201.196.95.200) |
| 16:10.09 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:10.20 | [TK]D-Fender | chilicuil: -- Executing [10445535037@numberplan-custom-1:1] Dial("SIP/592-b71ad2d0", "SIP/5512048885/0445535037,50,Ttm") in new stack <-- this says [numberplan-custom-1] not [outgoing_calls] |
| 16:10.59 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
| 16:11.00 | [TK]D-Fender | chilicuil: and i don't don't see the debug of the call coming in FROM SIP/592 |
| 16:11.11 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:11.30 | [TK]D-Fender | chilicuil: And you never told me what that device is. |
| 16:11.30 | jpcansa | is there any way to open a linksys phone web interface from the linux console? |
| 16:11.40 | *** join/#asterisk wpbrown (~wpbrown@wh-gtw-0001.woolfharris.com) |
| 16:11.58 | [TK]D-Fender | jpcansa: "man links" |
| 16:12.04 | idespinner | jpcansa, you could use lynks or lynx or links(a text based web browswer) but i doubt that will get you too far |
| 16:12.30 | wpbrown | I have a quick question: What inside of Asterisk calculates hold time. The PBX tells users their call time is estimated 30 mins. |
| 16:12.53 | jpcansa | idespinner, cool, i just need to change an ext number, may be that could be possible |
| 16:13.04 | wpbrown | Any quick tip on how to purge the information it apparently is sticking with? |
| 16:15.18 | ManxPower-work | sbrath: BLF and SLA are not the same thing. |
| 16:15.45 | idespinner | jpcansa, depends. if the browser can handle the config page. Formatting may be off a bit... |
| 16:16.32 | ManxPower-work | Linksys uses extensive Javascript in any of their pages I've worked with. |
| 16:16.44 | sbrath | BLF as busy light != SLA you're right. I've seen some places call SLA a different accronym... But you're right. |
| 16:16.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:17.25 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.48 | *** join/#asterisk prometheanfire (~mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net) |
| 16:17.50 | chilicuil | ops, yep, [TK]D-Fender, but the numberplan-custom-1 only includes [outgoing_calls] http://pastebin.com/m3684103e , I attach another debug log, http://pastebin.com/m139d9ac1 , I'm not sure what do you mean by device |
| 16:19.26 | [TK]D-Fender | chilicuil: INVITE sip:10445535037@10.9.8.13 SIP/2.0 <-- this is the number originally passed from your SPA-2102. |
| 16:19.27 | *** join/#asterisk E-bola (~bola@smtp.techbiz.dk) |
| 16:19.29 | E-bola | Hello |
| 16:19.37 | chilicuil | yep, [TK]D-Fender |
| 16:19.49 | [TK]D-Fender | chilicuil: the SPA has its OWN dialplan that controls how many digits it accepts before sending the call onwards. fix THAT |
| 16:19.51 | chilicuil | no |
| 16:20.06 | E-bola | I have 2 different IAX providers setup on my 1.6.x system. I would like to use them equally so they each handle 50% of the outgoing calls, and if one is down the other one handles 100% of the calls |
| 16:20.15 | E-bola | do anybody know where i can find an example of such a config? |
| 16:20.20 | chilicuil | owowow, [TK]D-Fender but it happens with every phone, even softphones |
| 16:20.21 | E-bola | google isnt giving me much |
| 16:20.38 | [TK]D-Fender | chilicuil: That call shows the number the PHONE sends. * has nothing to do with that |
| 16:21.03 | ManxPower-work | E-bola: app_random/func_random |
| 16:21.13 | chilicuil | ok, then I'm gonna check the local conf (taking about the phones), thx a lot [TK]D-Fender |
| 16:21.13 | ManxPower-work | Or you can use GROUP* functions |
| 16:21.21 | bmoraca_work | E-bola: i suspect you're not going to find a canned solution to this |
| 16:21.36 | E-bola | ManxPower-work: ya i already got a macro using random |
| 16:21.40 | bmoraca_work | like Manx says, I'd use GROUP functions and ChanIsAvail to determine if it's down |
| 16:21.41 | [TK]D-Fender | E-bola: this is just dialplan. Set a flag to say "A". If A, then dial out A. Flip the flag to say B. Wash. Rinse. Repeat |
| 16:21.43 | E-bola | i was just hoping for something already done |
| 16:21.46 | idespinner | seems to me app_random may or may not do 50/50.... |
| 16:22.04 | ManxPower-work | idespinner: it should be close to 50/50 when time=infinity |
| 16:22.05 | [TK]D-Fender | E-bola: This is 2-3 lines of dialplan. Who would BOTH with this? |
| 16:22.06 | E-bola | [TK]D-Fender: Always nice to steal the code from somewhere instaid of re-inventing the wheel if i can help it |
| 16:22.17 | idespinner | yea... you got me on a technicality though... |
| 16:22.36 | [TK]D-Fender | E-bola: Searching for 2-3 lines of dialplan to steal is just sad... and expecting them to be perfectly formatted to your needs... well... |
| 16:22.37 | ManxPower-work | E-bola: you're not reinventing the wheel. You're reinventing a piece of charcol |
| 16:22.55 | E-bola | lazy ftw! |
| 16:22.57 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:23.18 | E-bola | i got the macro to select one of the server balanced |
| 16:23.28 | E-bola | but i wasnt completelæy sure how to check if a call fails |
| 16:23.37 | bmoraca_work | i'd rather build it myself so that I know it works properly within the context of my own dialplan, rather than ripping something off from someone else |
| 16:23.45 | E-bola | i guess i gotta check some sort of return var and if its = failed then dial out via the other line? |
| 16:24.11 | E-bola | bmoraca_work: i always prefer to build off something already working and then adjust to my needs |
| 16:24.35 | bmoraca_work | E-bola: that's almost always more work than just sitting down and doing it yourself |
| 16:24.50 | jaytee | most of my best ideas I stole from jared, leif and [TK]D-Fender :-) |
| 16:25.08 | E-bola | bmoraca_work: sometimes it is, sometimes it isnt. In any case its always my prefered way to do it |
| 16:25.10 | *** part/#asterisk prometheanfire (~mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net) |
| 16:25.37 | beek | mornin' jaytee |
| 16:25.38 | idespinner | E-bola, I found a post that assigns the outbound trunks as call queue agents to load balance... |
| 16:26.18 | E-bola | idespinner: hehe ya i think i stumbled upon that, seems a bit weird/overkill though |
| 16:26.30 | jaytee | mornin' beek |
| 16:27.14 | luca`gervasi | hallo. i have some throuble configuring res_config_mysql (for realtime loading of my extensions from mysql database). is anybody using it? the mappings seems ok, but the new extensions doesn't get listed in "sip show peers"... |
| 16:28.25 | E-bola | I'm not even sure how i would get asterisk to failover to a second provider |
| 16:28.35 | *** join/#asterisk mikealeonetti (~mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 16:28.44 | E-bola | since i would have to interpret what the #1 provider returns in case he somehow fails |
| 16:29.21 | mikealeonetti | what module would I have to reload to change the call-limit variable in sip.conf? I tried reloading chan_sip.so but the call-limit is still there. |
| 16:29.56 | E-bola | like if the party u are calling is busy |
| 16:30.07 | E-bola | asterisk shouldnt try the number one more time with the other provider |
| 16:32.13 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 16:32.43 | E-bola | [TK]D-Fender: I'd actualy like to see you pull it off in 2-3 lines of dialplan, i very much doubt so thats possible |
| 16:33.34 | leifmadsen | E-bola: use ${DIALSTATUS} |
| 16:34.13 | [TK]D-Fender | [11:32]<E-bola>[TK]D-Fender: I'd actualy like to see you pull it off in 2-3 lines of dialplan, i very much doubt so thats possible <- translation. Attempt to prove me wrong, and in so doing, provide me with the very code I'm looking to rip off rather than code myself. I understand perfectly |
| 16:34.18 | E-bola | leifmadsen: But wouldnt it be possible that some providers mighr return BUSY as an error as well as if u call a person thats busy? |
| 16:34.22 | *** join/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 16:34.23 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:34.32 | leifmadsen | E-bola: it's possible a provider could suck, yes :) |
| 16:34.51 | E-bola | Mmmm so i guess i shouldnt include BUSY as an error state |
| 16:35.01 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 16:36.55 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 16:37.01 | idespinner | any of you familiar with CDR issues and asterisk AMI originate? |
| 16:37.17 | idespinner | more specificaly where the total called time is always <2 seconds.... |
| 16:37.27 | idespinner | err >2 seconds... |
| 16:37.47 | leifmadsen | E-bola: something like this (depending on provider): http://pastebin.ca/1805850 |
| 16:37.52 | ManxPower-work | idespinner: Are you using analog lines? |
| 16:38.03 | idespinner | no but it is Dahdi and ZAPTEL |
| 16:38.06 | idespinner | pri |
| 16:38.26 | idespinner | this is with a recent 1.4 asterisk and an older ABE C rev... |
| 16:39.02 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 16:39.05 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 16:39.47 | leifmadsen | E-bola: or slightly modified to handle BUSY ${DIALSTATUS}. Again, depends on the provider and how good they are at sending the right status: http://pastebin.ca/1805854 |
| 16:40.06 | E-bola | leifmadsen: ya looks valid, but it doesnt pick a new provider in case of fail right? |
| 16:40.29 | E-bola | ohh nm |
| 16:40.34 | ManxPower-work | E-bola: Keep trying. I'm sure you'll manage to fool someone into writing it for you. |
| 16:40.44 | E-bola | it does missunderstood the last goto |
| 16:41.13 | E-bola | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS actualy have a useable example |
| 16:41.48 | leifmadsen | every time someone quotes voip-info.org, God kills a kitten. |
| 16:42.13 | E-bola | only place i know to get asterisk info... |
| 16:42.14 | idespinner | leifmadsen, whats wrong with voipinfo? its the best source of documentation ive seen for asterisk |
| 16:42.28 | leifmadsen | out of date almost exclusively |
| 16:42.32 | *** join/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-azqibsakotmirkmx) |
| 16:42.32 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:42.35 | ManxPower-work | E-bola: the official docs are included in the Asterisk source. |
| 16:42.38 | E-bola | Its a shame there isnt any better but thats open source for ya |
| 16:42.44 | ManxPower-work | leifmadsen: not to mention just plain wrong in some cases. |
| 16:43.15 | ManxPower-work | E-bola: *nod* We try to keep the official documentation (and there's quite a bit of it) hidden so users will have a hard time with Asterisk. |
| 16:43.16 | xuser | leifmadsen: when are you going to write a new edition of the book? ;-) |
| 16:43.20 | idespinner | to be honest, i'm not really sure how one would browse the source to find documentation... |
| 16:43.27 | leifmadsen | xuser: working on it now |
| 16:43.35 | ManxPower-work | idespinner: cd /path/to/src/asterisk/doc/ |
| 16:43.36 | xuser | nice |
| 16:43.36 | tzafrir | E-bola, if you care about it, help maintaning it. It's a wiki |
| 16:43.54 | idespinner | wasnt there a recent version of TFOT about a year ago? |
| 16:44.07 | E-bola | tzafrir: I cant help correct all the wrongs int he world, no matter how nice it would be... |
| 16:44.33 | tzafrir | ManxPower-work, I normally give links to files under http://svn.asterisk.org/svn/asterisk/<branch-name>/ |
| 16:44.50 | ManxPower-work | tzafrir: Yes, I know. Sad. |
| 16:44.51 | tzafrir | E-bola, so fix just one thing. Not everything |
| 16:44.59 | *** join/#asterisk slashtom (~tom@k-rad.co.uk) |
| 16:45.05 | ManxPower-work | Like an end user would know their "branch name" |
| 16:45.38 | tzafrir | ManxPower-work, I know. http://svn.asterisk.org/svn/asterisk/branches/1.6.1/doc/ |
| 16:46.13 | ManxPower-work | tzafrir: I guess that site requires javascript to be enabled. |
| 16:46.14 | E-bola | http://svn.asterisk.org/svn/asterisk/branches/1.6.1/doc/queue.txt |
| 16:46.16 | E-bola | riiiiight |
| 16:46.23 | E-bola | a plethora of info.... |
| 16:46.23 | tzafrir | ManxPower-work, you have NoScript? |
| 16:46.26 | slashtom | i'm building asterisk-1.2.29 and i have selected chan_skinny, but our previous build used chan_sccs - which one should i be using, as i understand that one replaces the other |
| 16:46.29 | ManxPower-work | E-bola: Dude, that EXACT file is ALREADY on your system. |
| 16:46.32 | ManxPower-work | tzafrir: yes. |
| 16:46.32 | slashtom | 1.4.29* |
| 16:46.45 | E-bola | ManxPower-work: My point is its even less informative than voip-info |
| 16:46.46 | tzafrir | It's not javascript. But it's a sort of XML or something |
| 16:46.49 | E-bola | did u actualy look in it? |
| 16:47.04 | ManxPower-work | E-bola: That is the official documention. |
| 16:47.15 | ManxPower-work | tzafrir: it is a blank screen to me. |
| 16:47.23 | E-bola | ManxPower-work: Thats my point |
| 16:47.37 | E-bola | So thank god for voip-info.com and #asterisk@freenode :) |
| 16:47.39 | tzafrir | ManxPower-work, Right. Sadly to see it you have to enable asterisk.org with noscript. |
| 16:49.20 | ManxPower-work | voip-info.com is the biggest source of misinformation about Asterisk on the internet. You'll learn that for yourself eventually. |
| 16:50.12 | E-bola | Doesnt matter as long as there isnt an alternative |
| 16:50.49 | E-bola | How come the official docs arent online somewhere in an html'ish format? |
| 16:51.10 | etfonhomey | E-bola, do you listen to your GPS when it tells you to turn right into a lake like on "The Office"? |
| 16:51.11 | sbrath | ManxPower-work: After changeing the DTMF duration is still does not work from connected SIP phones, but a old Key System phone that is connected via another TE110P card, and it can do the DTMF to the remove IVR correctly. |
| 16:51.15 | leifmadsen | E-bola: why do you say that? |
| 16:51.23 | ManxPower-work | E-bola: Oh, I'm sure they are somewhere. It's just so much easier to read the local info. |
| 16:51.26 | leifmadsen | E-bola: http://www.asterisk.org/docs |
| 16:51.39 | E-bola | http://www.asterisk.org/support |
| 16:51.44 | TheDavidFactor | Does anyone know if Digium's Fax For Asterisk is supported on Asterisk 1.6.2? |
| 16:51.44 | leifmadsen | wents to Asterisk.org, clicked "Documentation" |
| 16:51.47 | E-bola | weird its not listed there |
| 16:52.18 | leifmadsen | probably because there is a fully "DOCUMENTATION" link at the top directly beside the "SUPPORT" link |
| 16:52.28 | E-bola | Must have missed it :) |
| 16:52.31 | elred_ | Hello. I am experiencing a recurrent deadlock problem on A* 1.4.25. Here is a pastebin of the backtrace (from a coredump) : http://www.pastebin.com/m34f5413c Any suggestion on how to solve this would be greatly appreciated |
| 16:52.57 | ManxPower-work | elred_: everyone will tell you to upgrade |
| 16:52.59 | E-bola | still the support page has a "online documentation" section |
| 16:53.13 | E-bola | which does not list http://www.asterisk.org/docs..... thats a bit odd |
| 16:53.31 | carrar | man |
| 16:53.35 | carrar | how did anyone learn asterisk |
| 16:53.38 | elred_ | ManxPower-work, yea but maybe some develloper could tell me if this is a known bug and a fixed one, so that i don't upgrade for nothing. |
| 16:53.48 | bmoraca_work | sauce code! |
| 16:53.49 | elred_ | Carrar: reading the book |
| 16:53.53 | carrar | heh |
| 16:54.16 | elred_ | (and voip-info.org ;o ) |
| 16:54.19 | etfonhomey | leifmadsen, what's new in the next edition of ATFOT? |
| 16:54.39 | E-bola | should get his boss to order a copy of the book |
| 16:54.49 | E-bola | I asume some part of the profit goes tot he author? |
| 16:55.19 | carrar | leifmadsen volunteers |
| 16:55.56 | leifmadsen | etfonhomey: http://leifmadsen.wordpress.com/tag/tfot/ |
| 16:56.01 | ManxPower-work | E-bola: I doubt the authors get more than 5 cents per copy |
| 16:56.23 | leifmadsen | I don't know... I think we end up with something like $300 every quarter |
| 16:56.34 | leifmadsen | I'm not getting rich, that's for sure :) |
| 16:56.38 | [TK]D-Fender | E-bola: dialplan apps are fairly well documented. Same with functions. tarball has docs on the vars, and config files. The boko describes a few concepts. what you actually need is all withing the core tarball. |
| 16:56.48 | carrar | Thats a night out drinking BEER |
| 16:56.50 | leifmadsen | I learned Asterisk when documentation didn't even exist for it |
| 16:56.59 | leifmadsen | it is certainly doable |
| 16:57.42 | E-bola | Sure, it always is with opensource as long as you can read even half decent code |
| 16:57.53 | E-bola | But nice docs make everything ohh so much easier :) |
| 16:58.33 | E-bola | mmm didnt mean that the asterisk code is half decent, just that u only need a basic code understanding to read it hehe |
| 16:58.46 | slashtom | um... is there an issue with chan_sccp for asterisk-1.4.29? |
| 16:59.31 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 16:59.32 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:59.54 | Dovid | hi. is there any option in cmd dial like r but that it will only play a ringing sound till it gets a 183 ? |
| 17:00.03 | *** join/#asterisk aandrade (~aandrade@187.5.33.100) |
| 17:02.23 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 17:03.26 | *** join/#asterisk wpbrown (~wpbrown@wh-gtw-0001.woolfharris.com) |
| 17:04.17 | etfonhomey | leifmadsen, thanks for putting in the time to write The Book. |
| 17:05.34 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 17:08.22 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:13.19 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:14.24 | ariel_ | Hi folks |
| 17:20.37 | chilicuil | [TK]D-Fender: hi dude, it worked!, the SPA-2102 device configuration was wrong, thank you |
| 17:20.49 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:21.04 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:21.19 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 17:21.46 | *** join/#asterisk nickaugust_ (~anonymous@216-160-159-80.hlrn.qwest.net) |
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| 17:30.47 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:33.43 | p3nguin_ | slashtom: What "issue" are you wanting to know about? |
| 17:34.09 | *** join/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 17:34.24 | flux_control | Anyone know of an asterisk user group in Japan? |
| 17:35.07 | flux_control | Ah, nevermind, I asked too soon. Just found one. :) |
| 17:35.11 | *** part/#asterisk flux_control (~flux@sourcemage/wizard/flux) |
| 17:35.36 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 17:35.56 | *** join/#asterisk Tim_Toady (~moi@77.49.236.7.dsl.dyn.forthnet.gr) |
| 17:38.59 | ManxPower-work | Sometimes if you just tell someone something enough times eventually they will believe you. |
| 17:39.27 | *** join/#asterisk nickaugust_ (~anonymous@216-160-169-17.hlrn.qwest.net) |
| 17:41.37 | spenguin[work] | Katty !! |
| 17:41.58 | *** join/#asterisk nickaugust__ (~anonymous@71-33-201-207.hlrn.qwest.net) |
| 17:42.34 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 17:42.52 | QbY | Anyone know of a DID Provider in Japan? Specifically Okinawa? |
| 17:43.01 | QbY | bb |
| 17:43.04 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 17:43.16 | b14ck | sup everyone |
| 17:43.41 | coppice | ManxPower-work: that sounds more like a repetitive strain injury than a teaching exercise |
| 17:44.31 | *** join/#asterisk nickaugust__ (~anonymous@71-33-193-63.hlrn.qwest.net) |
| 17:46.27 | *** join/#asterisk nickaugust_ (~anonymous@71-33-196-132.hlrn.qwest.net) |
| 17:46.35 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 17:46.58 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
| 17:47.08 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
| 17:50.27 | ManxPower-work | coppice: *nod* chilicuil just would not believe it was not an Asterisk problem |
| 17:52.57 | *** join/#asterisk skrite (~skrite@69.55.17.185) |
| 17:53.15 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 17:53.52 | skrite | ManxPower-work, p3nguin_ Hey, big thanks for your help over the last couple of days, i managed to get everyting installed and working. Appreciate it a lot. |
| 17:54.32 | Geminizer | Hello all. Is there any way to "boost" the volume level on a given channel (invoked via Dial(..))? |
| 17:58.49 | voipmonk | are you handing off to a sip itsp or your own dahdi tdm cards, Geminizer ? |
| 17:59.08 | leifmadsen | Geminizer: VOLUME() function (in later versions of Asterisk) |
| 17:59.40 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:00.52 | Kobaz | [Feb 22 13:00:37] ERROR[10496]: chan_dahdi.c:10594 dahdi_pri_error: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX |
| 18:00.55 | Kobaz | hmm |
| 18:00.57 | Kobaz | that's odd |
| 18:04.04 | Geminizer | voipmonk: sip itsp |
| 18:04.30 | *** join/#asterisk hfb (~hfb@pool-98-112-226-53.lsanca.dsl-w.verizon.net) |
| 18:06.37 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 18:07.07 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 18:09.32 | voipmonk | what version of asterisk, Geminizer ? |
| 18:10.18 | p3nguin_ | skrite: That's great to hear! Glad everything came together for you. |
| 18:10.20 | Geminizer | 1.4.24 |
| 18:10.39 | [TK]D-Fender | Geminizer: No ral options |
| 18:10.41 | [TK]D-Fender | real* |
| 18:10.50 | voipmonk | does the callee on the far end barely hear you or can you hardly hear the person speaking to you? |
| 18:11.11 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 18:11.12 | Geminizer | the receiver / callee is having trouble hearing the caller |
| 18:11.24 | voipmonk | is it the same caller with this problem? |
| 18:11.37 | ruben23 | hi |
| 18:12.22 | Katty | i don't wanna go to the dr :< |
| 18:12.24 | Katty | pouts |
| 18:12.34 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 18:12.48 | voipmonk | hot date last night? |
| 18:12.56 | Geminizer | Katty, I wouldn't want to go to the DR either, being that it is next to Haiti |
| 18:13.07 | Katty | voipmonk: yearly torture. |
| 18:13.13 | voipmonk | :( |
| 18:13.37 | Geminizer | voipmonk: It's multiple callers that are experiencing the problem |
| 18:13.38 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 18:13.44 | Corydon76-lap | Is it torture only because you have girly parts? |
| 18:13.52 | voipmonk | same provider, Geminizer ? |
| 18:13.57 | Geminizer | correct |
| 18:14.03 | luca`gervasi | hallo. i have some throuble configuring res_config_mysql (for realtime loading of my extensions from mysql database). is anybody using it? the mappings seems ok, but the new extensions doesn't get listed in "sip show peers"... |
| 18:14.14 | Katty | Corydon76-lap: errr |
| 18:14.16 | fifer | Anyone know if there is a diference between the regular Aastra 6755i and the 8x8 model of the 6755i? |
| 18:14.18 | voipmonk | what happens when you try a different provider, Geminizer ? |
| 18:14.48 | voipmonk | one comes with different shipping options |
| 18:14.53 | Corydon76-lap | luca`gervasi: and they won't, until they're loaded dynamically |
| 18:15.05 | voipmonk | ( and a sandwich ) |
| 18:15.08 | luca`gervasi | do i need to totally erase my .conf giles? |
| 18:15.21 | Corydon76-lap | luca`gervasi: such as at sip device registration |
| 18:15.33 | Corydon76-lap | or 'sip show peer foo load' |
| 18:15.42 | fifer | I was assuming that the only diference might be a diferent firmware, though I would expect that I can flash with the standard sip fw |
| 18:15.44 | luca`gervasi | Corydon76-dig, what? wait a sec... |
| 18:15.45 | ruben23 | hi any have idea, suddenly when i set command sip show peers on my asterisk CLI i get this---> No such command 'sip show peers' (type 'help sip show' for other possible commands) but it did work before i restart the asterisk server. |
| 18:15.46 | Geminizer | hmm.. I will have to try that |
| 18:15.54 | fifer | might not even be that much diference though |
| 18:16.06 | ruben23 | did my asterisk crashed..? |
| 18:16.11 | ManxPower-work | ruben23: you screwed up your config files again |
| 18:16.16 | fifer | Hm, what kind of sandwich? |
| 18:16.26 | ManxPower-work | ruben23: "module load chan_sip.so" |
| 18:16.29 | luca`gervasi | Corydon76-lap, is it in sip.conf? |
| 18:16.58 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.14 | Corydon76-lap | luca`gervasi: I have no idea what you're talking about |
| 18:17.23 | ruben23 | ManxPower-work: http://www.pastebin.com/m2fb3faad |
| 18:17.35 | BCS-Satori | Would it be possible to have a phone dial its own extension to check its voice-mail. I.E.: if extension 6030 dials 6030 they are able to retrieve their voice-mail? |
| 18:17.43 | Corydon76-lap | luca`gervasi: I gave you a command to run on the CLI |
| 18:18.16 | luca`gervasi | Corydon76-dig, i have no such command... 1.6.x |
| 18:18.28 | Corydon76-lap | 'sip show peer foo load' means show the config for peer foo, and load it from realtime, if possible |
| 18:18.45 | ManxPower-work | ruben23: that confirms you screwed up your sip.conf |
| 18:18.54 | ManxPower-work | chances are it is line 1 |
| 18:19.01 | luca`gervasi | firegate*CLI> sip show peer 999 load - > Peer 999 not found. |
| 18:19.16 | luca`gervasi | Corydon76-dig, 999 is actually defined in my mysql schema |
| 18:19.20 | ManxPower-work | BCS-Satori: you mean like If CALLERID=EXTEN then Voicemail? |
| 18:19.29 | Corydon76-lap | luca`gervasi: then you don't have realtime configured correctly |
| 18:19.36 | BCS-Satori | ManxPower-work: yep |
| 18:19.49 | ManxPower-work | BCS-Satori: that is standard dialplan stuff. |
| 18:19.54 | ManxPower-work | ~answers |
| 18:19.55 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 18:20.12 | [TK]D-Fender | BCS-Satori: "core show application gotoif" |
| 18:20.20 | [TK]D-Fender | BCS-Satori: "core show function CALLERID" |
| 18:20.22 | luca`gervasi | Corydon76-dig, can you lead me to the right way? i followed some docs on voipinfo, but i was unsuccesfully |
| 18:20.46 | *** join/#asterisk nix8n82 (~AndChat@63.162.27.14) |
| 18:21.09 | Corydon76-lap | luca`gervasi: 'core show config mappings'... what does it show for sippeers? |
| 18:21.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:22.07 | luca`gervasi | pasting on nopaste |
| 18:22.22 | ruben23 | ManxPower-work: thanks it ok now. |
| 18:23.02 | luca`gervasi | Corydon76-dig, http://nopaste.info/99d452b5fb.html |
| 18:23.42 | Corydon76-lap | luca`gervasi: and do you have a section called [asterisk] in res_mysql.conf ? |
| 18:24.25 | luca`gervasi | Corydon76-dig, argh... i tough that "asterisk" was the database name, not an actual section... Trying to correct now! (please wait) |
| 18:25.49 | luca`gervasi | Corydon76-dig, added the section, but the peer still doesn't show |
| 18:26.03 | Corydon76-lap | luca`gervasi: did you restart Asterisk? |
| 18:26.11 | luca`gervasi | yes, with core restart now |
| 18:26.33 | luca`gervasi | the mappings are still in place |
| 18:26.44 | Corydon76-lap | Check for typos |
| 18:27.04 | Corydon76-lap | Verify that type=peer in the db and not type=user |
| 18:29.08 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 18:29.26 | luca`gervasi | Corydon76-dig, done... no change... |
| 18:30.09 | Corydon76-lap | Dunno then |
| 18:30.18 | luca`gervasi | do you have it working? |
| 18:30.25 | Corydon76-lap | I use ODBC |
| 18:30.48 | luca`gervasi | i can give it a try, if you land me your (censored) config :$ |
| 18:31.08 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:31.18 | Corydon76-lap | Sorry, no, it's not here, and it's quite a bit of work to get it |
| 18:31.46 | Corydon76-lap | and I have other work I need to get done |
| 18:32.14 | luca`gervasi | Corydon76-dig, ok. thanks for you help anyway ^_^... where did you get the infos to get it working? |
| 18:33.07 | Corydon76-lap | I read the sample configuration files |
| 18:33.37 | luca`gervasi | ok thanks again :D |
| 18:35.34 | ruben23 | hi any idea about sometimes, when calling with asterisk the other party cannot hear you but some times they can hear you, just a random problem... |
| 18:35.42 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 18:36.07 | Katty | ihad that issuew ith some polycoms before |
| 18:36.11 | Katty | turned out to be a firmware issue |
| 18:36.38 | ruben23 | <PROTECTED> |
| 18:36.44 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 18:37.07 | Katty | ruben23: ah, well it probably isn't that then :P |
| 18:37.50 | ruben23 | :-D |
| 18:38.01 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 18:38.47 | Corydon76-lap | causes random problems for Katty |
| 18:39.34 | atis_work | whew, 4 programmers are able to push jeep into garage.. :p |
| 18:41.35 | ChannelZ | 4? I think 2 could probably push one over :) |
| 18:42.00 | [TK]D-Fender | Only takes 1 to crash it <- |
| 18:42.26 | ChannelZ | heh |
| 18:43.08 | ChannelZ | true dat |
| 18:43.12 | Katty | Corydon76-lap: :<<<< |
| 18:44.34 | Corydon76-lap | Katty: maybe you need a female doctor |
| 18:44.44 | Katty | i have one |
| 18:44.58 | Corydon76-lap | and she still makes you uncomfortable? |
| 18:45.11 | Katty | her specifically? no. |
| 18:45.13 | Katty | the process? yes. |
| 18:45.44 | Corydon76-lap | Pap smear? |
| 18:46.20 | Katty | bagel with smear. |
| 18:46.37 | p3nguin_ | cream cheese? |
| 18:46.47 | Katty | mhmm |
| 18:46.47 | p3nguin_ | I bet the doctor doesn't like that too much. :/ |
| 18:47.02 | Katty | we're not talking about |
| 18:47.07 | Katty | that is my clever attempt to change the subject. |
| 18:47.13 | p3nguin_ | OH! |
| 18:47.20 | p3nguin_ | I didn't catch on. |
| 18:47.49 | Corydon76-lap | Daffy... penguin |
| 18:54.22 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 19:00.14 | *** part/#asterisk BadHorsie (~nile@201.198.239.167) |
| 19:02.51 | *** join/#asterisk Alagar (~Administr@122.164.33.221) |
| 19:03.09 | Kobaz | is there a way to ignore specific events in the AMI |
| 19:03.24 | atis_work | well, push yes, but if you have to push it through the snow.. it has dead engine :p |
| 19:03.53 | Kobaz | like, i have some agi scripts... and if there's lots of callers calling in it'll be generateing upwards of several hundre agiexec events per second |
| 19:03.58 | seanbright | Kobaz: you can use the Events: header to specify what category of event you want to receive, but not to exclude specific onces |
| 19:04.06 | Kobaz | yeah i know about Events: |
| 19:04.09 | seanbright | (the Events header of the Login action) |
| 19:04.12 | Kobaz | yeah |
| 19:04.20 | seanbright | then no |
| 19:04.20 | Kobaz | i was thinking of adding an EventsFilter option to login |
| 19:04.27 | seanbright | use a proxy |
| 19:04.31 | seanbright | or add it yourself |
| 19:04.33 | Kobaz | yeap |
| 19:04.36 | Kobaz | i'm adding i'm adding |
| 19:11.16 | ManxPower-work | astmanproxy has a good reputation, but it did not work for me. |
| 19:13.24 | fifer | Anyone have any very old aastra 480i firmware files laying around? |
| 19:15.31 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 19:17.05 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.44 | *** join/#asterisk smooth_penguin (~smoove@59.95.14.193) |
| 19:18.49 | p3nguin_ | Hmm... decisions... |
| 19:18.56 | p3nguin_ | Pillowtop or EuroTop? |
| 19:19.53 | carrar | wood board |
| 19:20.01 | carrar | or futon |
| 19:20.09 | p3nguin_ | Too stiff, not good on the back. |
| 19:20.30 | carrar | Memory Foam |
| 19:20.36 | ManxPower-work | EuroTops are hot. |
| 19:20.53 | p3nguin_ | I don't have an extra $1500 lying around, or I would have a TempurPedic. |
| 19:21.21 | carrar | EuroTop come with a slutty swedish chic? |
| 19:21.43 | p3nguin_ | I think that's extra. |
| 19:22.20 | Kobaz | p3nguin_: when you do happen to have an extra 1500 lying around... you can give it to me |
| 19:22.44 | p3nguin_ | I'll put you on the list right under my TempurPedic mattress. |
| 19:23.49 | carrar | hell add me too |
| 19:23.51 | etfonhomey | What is the correct sip.conf setting to set it so that I re-register with my ITSP every 60 seconds? |
| 19:23.56 | p3nguin_ | I passed up an awesome deal about two years ago. Someone had a queen TempurPedic, brand new (so they claimed) for sale for $450. |
| 19:24.24 | carrar | complete with bed bugs! |
| 19:24.44 | p3nguin_ | I didn't realize I needed it at the time, but now I'm kicking myself for not at least going to take a look at it to see if it was, in fact, brand new. |
| 19:25.28 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:26.02 | p3nguin_ | etfonhomey: defaultexpiry= maybe |
| 19:26.33 | etfonhomey | I always get the maxexpirey and defaultexpirey confused. |
| 19:27.16 | p3nguin_ | maxexpiry is for incoming registrations, where defaultexpiry should control both ways. |
| 19:27.23 | etfonhomey | I have maxexpirey=180 and defaultexpirey=160, but when I do a sip show registry it says 45 under the "Refresh" column. |
| 19:27.48 | p3nguin_ | That might be the time that your ITSP has told your client to use. |
| 19:28.02 | etfonhomey | And looking at SIP debug, it is re-registering ever 45 seconds. |
| 19:28.14 | p3nguin_ | "Hey, here I am." "Okay, now register again in 45 seconds." |
| 19:28.17 | ManxPower-work | etfonhomey: this is a good thing with NAT |
| 19:28.22 | etfonhomey | My ITSP says to set it at 60. |
| 19:28.46 | p3nguin_ | Is it causing a problem the way it is? |
| 19:29.31 | bmoraca_work | there's nothing wrong with short sip registration times...it's not like registering is incredibly resource intensive...no more so than qualify, anyway |
| 19:29.52 | etfonhomey | I thought it was the max/defaultexpirey commands. The frequency of failed incoming calls has increased lately and I just want to conform to what the ITSP expects so we can look elsewhere. |
| 19:30.14 | p3nguin_ | What type of failures? |
| 19:30.30 | bmoraca_work | etfonhomey: that says to me that your registration timeout is longer than your NAT timeout |
| 19:30.46 | etfonhomey | bmoraca_work, that is what I suspect as well. |
| 19:32.37 | bmoraca_work | WSFTP is the best FTP client evar |
| 19:33.27 | etfonhomey | bmoraca_work, My NAT/firewall device is a Cisco 871W router. So, i think it is either the dynamic NAT mapping timing out. Or, the application inspection that opens the firewall for the return traffic timing out. |
| 19:33.33 | p3nguin_ | No results for wsftp... so I'm going to say that probably isn't the case. |
| 19:33.46 | p3nguin_ | If it were truly the best, it would be available in my repos. |
| 19:33.47 | etfonhomey | ws_ftp, maybe? |
| 19:34.07 | etfonhomey | I think it's only Windows based, though. |
| 19:34.24 | p3nguin_ | Ah, that further supports the fact that it isn't the best. |
| 19:34.34 | etfonhomey | :) |
| 19:35.18 | ManxPower-work | etfonhomey: pretty simple to find out. |
| 19:36.02 | ManxPower-work | ip nat translation port-timeout udp 5060 3600 |
| 19:36.27 | bmoraca_work | that should do her |
| 19:37.09 | ManxPower-work | no ip nat service sip udp |
| 19:38.24 | ManxPower-work | "no ip nat service sip udp port 5060" might be more correct |
| 19:38.50 | etfonhomey | Also did this: ip inspect udp idle-time 3600 (wouldn't let me get port specific) |
| 19:39.18 | ManxPower-work | etfonhomey: you must not be running the same IOS I am running. |
| 19:39.30 | ManxPower-work | inspect is usually for the stateful firewall stuff, not basic nat |
| 19:39.47 | ManxPower-work | etfonhomey: did you try the two commands I gave you? |
| 19:40.24 | etfonhomey | ManxPower-work, I'm using CBAC on this 871w as well. |
| 19:40.51 | etfonhomey | ManxPower-work, I set both the inspection timeout and the nat timeout. |
| 19:40.55 | etfonhomey | (just now) |
| 19:40.57 | ManxPower-work | etfonhomey:Sorry to waste your time. Nothing I said applies to you. |
| 19:45.27 | etfonhomey | ManxPower-work: http://www.pastebin.ca/1805957 |
| 19:45.57 | ManxPower-work | etfonhomey: I know nothing about CBAC |
| 19:48.58 | etfonhomey | ManxPower-work, oh. The way I have configured it doesn't do any inspection past the src/dst port in the tcp/udp header. Just opens up the FW for return traffic. So, it may have been timing out in addition to the entries in the NAT table. |
| 19:50.00 | etfonhomey | ManxPower-work, do you know what the default timeout is for entries in the NAT table? |
| 19:50.45 | ManxPower-work | etfonhomey: no. |
| 19:51.18 | ManxPower-work | Most routers are 60 seconds |
| 19:53.27 | carrar | etfonhomey, why don't you set a timeout |
| 19:54.01 | etfonhomey | carrar, I did. |
| 19:59.26 | *** join/#asterisk alex_voip (~alex@201.161.45.81) |
| 20:01.59 | alex_voip | hello, * 1.4.28 with Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) recording stops on attended transfer. Any ideas? |
| 20:02.44 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:03.37 | p3nguin_ | Mine does that, too. 1.4.29 |
| 20:08.07 | *** join/#asterisk Reemo (~Reemo@p5B3EEB79.dip.t-dialin.net) |
| 20:10.23 | Reemo | Hey guys, I'm using Asterisk 1.6.2.0-1 (Debian) with CAPI and SIP. Asterisk works along with my analog telephone and just routes incoming calls from capi to my sip phones. Problem is: If theres no SIP Phone online, it instantly jumps to voicemail. How can I let Asterisk wait a total of 30 seconds before handing over to voicemail? Will pastebin the dialplan in a sec. |
| 20:11.44 | Reemo | http://asterisk.pastebin.com/d15dcb4b1 |
| 20:11.47 | bmoraca_work | Reemo: use ChanIsAvail or devstate to determine whether or not your SIP phones are available. if they aren't, Wait(30) or play ringing, and then transfer. |
| 20:14.26 | hardwire | bmoraca_work: probably easier than using groups. however groups are great if you want to know more about the devstate for a specific function. |
| 20:14.55 | hardwire | I use it to check to see if the phone is currently handling an inbound customer and avoid it if so, but other functions still ring through. |
| 20:15.56 | bmoraca_work | i've never used groups for anything other than determining whether a channel is congested or not, but i suspect there are lots of other uses for them |
| 20:15.59 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 20:16.29 | bmoraca_work | or, rather, i should say "whether a channel SHOULD BE congested or not" |
| 20:18.04 | p3nguin_ | reemo: http://asterisk.pastebin.com/dcc1b9fb |
| 20:18.32 | hardwire | bmoraca_work: they are very useful |
| 20:18.45 | hardwire | I like being able to add channels to a named group and check their existence |
| 20:18.58 | *** join/#asterisk donatas (~donatas@ton.czv.lt) |
| 20:19.27 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 20:19.54 | donatas | hey. if I use Background(file) I can hear it via phone, but if I use SetMusicOnHold(default), I can't hear.. |
| 20:20.01 | donatas | how to fix it ? |
| 20:20.17 | Reemo | Thanks for the hints, p3nguin_ will try this out. But just for my understanding: why doesnt the dowait solution i pasted work? |
| 20:20.23 | p3nguin_ | donatas: If you use SetMusicOnHold(default), you can't here what? |
| 20:20.27 | p3nguin_ | hear |
| 20:20.51 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 20:20.54 | p3nguin_ | reemo: I didn't see any dialplan code, so I have no idea what you tried. |
| 20:21.05 | Reemo | http://asterisk.pastebin.com/d15dcb4b1 |
| 20:21.06 | ManxPower-work | donatas: try MusicOnHold |
| 20:21.25 | ManxPower-work | You are only setting the music for when a call is placed on hold. Are you placing the call on hold? |
| 20:22.41 | donatas | p3nguin_: i hear nothing.. |
| 20:23.02 | p3nguin_ | donatas: Yeah... why would setting the MoH class cause you to hear anything? |
| 20:23.05 | donatas | wait, i will paste some |
| 20:23.32 | p3nguin_ | You would have to actually do something to play MoH to hear MoH. |
| 20:23.37 | Reemo | ahhh p3nguin_ how can you know my sip alias if you haven't seen my dial plan? ;) |
| 20:23.47 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 20:23.55 | p3nguin_ | reemo: I saw what you pasted, but it was not dialplan code. |
| 20:24.15 | ManxPower-work | Reemo: "sip alias"?? |
| 20:24.24 | p3nguin_ | reemo: What I pasted for you was dialplan. |
| 20:24.48 | p3nguin_ | reemo: And if you use what I provided, the result should be exactly what you asked for. |
| 20:24.55 | Reemo | well it is "dialplan show". http://asterisk.pastebin.com/d2088c5d8 |
| 20:24.55 | ManxPower-work | maybe you mean sip peer/friend/user |
| 20:25.08 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:25.55 | donatas | here: http://pastebin.com/m14e6ceba |
| 20:26.07 | p3nguin_ | reemo: No idea. I'm not likely to ever try to get that to work. I would write it exactly the way I did for you. |
| 20:26.08 | Reemo | p3nguin_, i'm just curious. I'm new to asterisk and I usually learn most from my mistakes |
| 20:26.33 | p3nguin_ | reemo: I try not to kludge things together if I know a good way to handle it. |
| 20:26.58 | ManxPower-work | ~answers |
| 20:26.59 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 20:28.10 | ManxPower-work | donatas: in the CLI do "core show application dial" pay special attention to the "m" option. Also NEVER ADD SPACES like you did in the Dial line. |
| 20:28.59 | Reemo | p3nguin_, i don't see the point why it shouldn't work, but appearently it doesn't. But thanks for your code :) |
| 20:29.01 | ManxPower-work | Obviously you did not provide with your actual extensions.conf |
| 20:30.53 | p3nguin_ | reemo: Give it a try and see if you like it. Make sure you put in your real mailbox number if what I used is wrong. |
| 20:31.25 | p3nguin_ | Yep, I did. I put in Voicemail(910778@default,u) when it should have been Voicemail(200@default,u) |
| 20:31.38 | p3nguin_ | err, Voicemail(2000@default,u) |
| 20:32.05 | Kobaz | anyone have the following problem with the new polycom firmware (3.2)..... ring a single phone for 10 seconds on, 2 seconds off, and after about 20 cycles, the phone returns CANCEL, but the phone doesn't actually stop the call ringing the line... so now you have a phantom call |
| 20:32.05 | donatas | ManxPower-work: do you mean this: Dial(SIP/${ZILVINAS},m(vadyba),15); ? |
| 20:32.28 | p3nguin_ | Here's an example of kludge: http://thereifixedit.com/2010/02/21/epic-kludge-photo-hope-it-doesnt-have-a-tilt-feature/ |
| 20:32.59 | ManxPower-work | donatas: read more carefully, notice where the timeout and the options go. |
| 20:33.18 | donatas | ManxPower-work: yea, i mixed order |
| 20:33.32 | benngard | yay, did move all our faxes from an audiocodes/avaya soloution to an spa2102/asterisk solution,loooks like 1000% better :) |
| 20:33.42 | donatas | I got this now: http://pastebin.com/m7f73a2b9 |
| 20:34.45 | ManxPower-work | donatas: great! |
| 20:35.08 | donatas | but no music |
| 20:35.10 | donatas | :( |
| 20:35.48 | ManxPower-work | donatas: fix your audio file and turn off comfort noise on your client |
| 20:36.08 | ManxPower-work | donatas: why don't you try getting it working with the STOCK MOH before you try doing custom stuff? |
| 20:36.27 | donatas | ah, one more thing. I use g729.. |
| 20:36.40 | ManxPower-work | donatas: how many G729 licenses do you have? |
| 20:37.00 | donatas | m, no licenses. asterisk.hosting.lv |
| 20:37.18 | ManxPower-work | donatas: then you are not going to be able to transcode from ulaw/wav to G729 |
| 20:37.31 | donatas | :( |
| 20:38.49 | carrar | h4X0r |
| 20:39.17 | carrar | donatas, get some licenses |
| 20:39.23 | carrar | they aren't too expensive |
| 20:39.33 | p3nguin_ | $8 per channel? |
| 20:39.38 | ManxPower-work | $10/channel |
| 20:39.52 | donatas | a year? |
| 20:39.54 | donatas | a month? |
| 20:39.58 | ManxPower-work | donatas: no. |
| 20:40.12 | carrar | 4 LIFE |
| 20:40.17 | ManxPower-work | $10/simultaneous channel, one time fee. |
| 20:40.19 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 20:40.35 | donatas | where can i buy them? |
| 20:40.45 | ManxPower-work | I guess Digium could give them away for free, but I don't think they'd be able to recover the tens of thousands of dollars they spent to license the codec. |
| 20:40.47 | carrar | They will work after life too |
| 20:40.49 | ManxPower-work | donatas: digium.com |
| 20:41.58 | benngard | i must have missed something, why are people using codecs like g729, i know they save bandwith, but what the heck, bandwith is almost fre nowadays |
| 20:42.17 | Reemo | p3nguin_, what is the code DIALSTATUS has when one SIP phone is busy? |
| 20:42.27 | donatas | ManxPower-work: thanks. |
| 20:42.44 | p3nguin_ | reemo: I don't know. Try it and see. |
| 20:43.28 | ManxPower-work | Reemo: 17 |
| 20:43.30 | p3nguin_ | reemo: Add an ,n,Verbose(${DIALSTATUS}) in there. |
| 20:43.47 | ManxPower-work | oh, dialstatus would be BUSY. hangupcause would be 17 for busy. |
| 20:44.04 | ManxPower-work | Reemo: you should have seen that info in "core show application dial" |
| 20:44.33 | luca`gervasi | is anybody using res_config_mysql to achieve asterisk realtime peering ? I'm unable to load any peer from a mysql database. Please help me :( |
| 20:45.15 | *** join/#asterisk smooth_penguin (~smoove@59.95.14.193) |
| 20:45.20 | QubeZ | anyone had issues with Asterisk where in the middle of the call, there was 3-5 secs of complete silence? I've checked my switchports for udp errors and asterisk as well, but cannot find any reason why this is happenning. There is plenty of bandwidth available during those calls. We have 40+ mb of bandwidth of which only 12% is being utilized. |
| 20:45.51 | ManxPower-work | QubeZ: Have you tried using SIP instead of IAX2? |
| 20:46.02 | QubeZ | ManxPower-work we are using SIP |
| 20:46.06 | QubeZ | these are internal phones on our network |
| 20:46.23 | ManxPower-work | QubeZ: perhaps you should reword your question with some useful information? |
| 20:46.29 | QubeZ | Polycom IP-301's. The issue is very random and we cannot recreate it. |
| 20:46.39 | ManxPower-work | QubeZ: turn off CDP on the phones. |
| 20:46.54 | QubeZ | ok |
| 20:53.11 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 20:54.09 | Reemo | p3nguin_, thanks for your help, it works. But wouldn't my solution be somewhat... cleaner? The idea behind my pasted dialplan was to "simulate" an extension to dial that just waits N seconds. I still don't get why that doesn't work |
| 20:55.34 | p3nguin_ | reemo: I think mine is cleaner, since it wasn't a kludge to fake a device's existence. |
| 20:55.51 | p3nguin_ | reemo: Mine used normal dialplan logic. |
| 20:56.19 | Reemo | But yours needs a GotoIf for every status code DIALPLAN can have except the one that occurs when one SIP Phone actually answers the call |
| 20:56.51 | Reemo | I'm just curious |
| 20:57.26 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 20:57.52 | p3nguin_ | You were adding a completely different context, even. |
| 20:58.05 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 20:58.07 | Mhaddog | epbx |
| 20:58.14 | p3nguin_ | Creating a local channel, running an extension in another context... |
| 20:58.28 | p3nguin_ | I don't see how that is better than my checking of DIALSTATUS. |
| 20:58.33 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 20:58.44 | *** part/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 20:59.06 | Reemo | the context is just an util context, exclusive just for doing the Wait stuff, and it is reusable |
| 20:59.38 | Reemo | it doesn't matter anyway. Yours works, mine doesn't. But I'm very interested why |
| 20:59.38 | p3nguin_ | And it also runs every time you place a call to that extension. |
| 20:59.57 | p3nguin_ | I don't know why -- you're asking the wrong person. |
| 21:00.32 | donatas | where to download moh sound files? |
| 21:00.58 | Reemo | okay, thanks anyway p3nguin_ |
| 21:01.55 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 21:02.34 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 21:02.35 | ujjain | Can I fax via SIP account, without having Asterisk? |
| 21:02.58 | p3nguin_ | maybe, but that wouldn't be a question for #asterisk. |
| 21:03.06 | bmoraca_work | that sounds like the absolute perfect question to ask in #asterisk |
| 21:03.14 | ujjain | :) I agree. |
| 21:03.19 | Deeewayne | Katty, ping |
| 21:04.08 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:04.15 | p3nguin_ | If you want to know if you can fax with SIP and Asterisk... yes. |
| 21:04.33 | ujjain | Hmm,, :( It requires .tif input |
| 21:04.45 | p3nguin_ | Not if you have a fax machine. |
| 21:05.53 | ujjain | Interesting. I forgot I do not have one lol. |
| 21:06.02 | ujjain | I do not have one. |
| 21:11.27 | *** join/#asterisk jayrod422 (~jayrod422@pool-71-99-94-9.tampfl.dsl-w.verizon.net) |
| 21:11.56 | jayrod422 | anyone ever see the issue where asterisk hangs up calls after a sip 200 becuase the remote endpoint doesnt ack back? |
| 21:13.00 | *** join/#asterisk justsomedood (~justsomed@mail.serverplus.com) |
| 21:13.09 | *** part/#asterisk justsomedood (~justsomed@mail.serverplus.com) |
| 21:14.05 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 21:15.08 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 21:15.58 | SuPrSluG | is there a new way asterisk handles sip traces? i'm getting [Feb 22 16:13:57] VERBOSE[4013] chan_sip.c: --- (12 headers 11 lines) --- |
| 21:16.10 | SuPrSluG | instead of the full sip dialog |
| 21:16.44 | SuPrSluG | asterisk version is 1.6.1.12 |
| 21:20.13 | voipmonk | still have to enable sip debug |
| 21:20.24 | voipmonk | sip set debug on or something |
| 21:22.36 | QubeZ | ManxPower-work disabling CDP is not an option for us because the phones get their correct VLAN and other network specific parameters via DHCP. |
| 21:23.11 | SuPrSluG | no, my problem is it's not showing the sip dialog just -> chan_sip.c: --- (13 headers 0 lines) --- |
| 21:24.04 | SuPrSluG | it's trimming the output. I can see it's there if i use the tail command. when I use cat it trims it. |
| 21:24.54 | ManxPower-work | QubeZ: One classic Polycom issue is having problems when they are configured to use CDP and CDP is not correctly working on the network. |
| 21:25.55 | etfonhomey | ManxPower-work, i ran in to a good one with my Polycom phones this weekend. |
| 21:26.09 | jayrod422 | is there something in a sip packet that would force a phone not to ACK a 200 OK when call is hungup? |
| 21:26.23 | jayrod422 | er answered not hungup |
| 21:26.46 | etfonhomey | ManxPower-work, I had callerid="Unknown" <Unknown> in [genera] of sip.conf thinking its format should mirror the format of callerid= for the peer entries. |
| 21:27.36 | QubeZ | ManxPower-work so what should my next step be? I disabled CDP and the phone works but is not getting the proper IP address. |
| 21:27.45 | ManxPower-work | etfonhomey: did sip.conf.sample, which lists where each option can be used, not give you that information? |
| 21:28.03 | ManxPower-work | QubeZ: investigate a possible problem with CDP on your network. |
| 21:28.13 | QubeZ | thanks |
| 21:28.20 | ManxPower-work | QubeZ: what SIP firmware are you using? |
| 21:28.28 | QubeZ | ManxPower-work 3.0.3 |
| 21:28.30 | Reemo | p3nguin_, got mine working. Ringing() before Wait did the job. |
| 21:28.40 | ManxPower-work | QubeZ: you might want to upgrade |
| 21:28.41 | etfonhomey | ManxPower-work, callerid can be used in [general], I just didn't think it's format would be different. I wanted incoming calls with no caller ID to say Unknown rather than asterisk |
| 21:28.44 | p3nguin_ | *shrug* |
| 21:28.58 | ManxPower-work | etfonhomey: do that in your dialplan |
| 21:28.58 | p3nguin_ | its format |
| 21:29.38 | ManxPower-work | One of the first thing I do on incoming calls is run a callerid fixup script. |
| 21:30.21 | etfonhomey | ManxPower-work, I learned that the hard way. Check out what it did to the sip conversation with my Polycom phone. Here's the Polycom's app log with sip debugging turned up (lines 16/17 in particular): http://pastebin.ca/1805805 |
| 21:30.22 | Katty | sigh |
| 21:30.23 | Katty | i miss shy |
| 21:32.15 | etfonhomey | ManxPower-work, because of that, my phones (all on a LAN) would go to LAGGED status randomly. |
| 21:32.45 | p3nguin_ | Misconfiguration giving unexpected behaviors. Imagine that. |
| 21:32.57 | *** join/#asterisk dlewis (~dlewis@about/security/staff/dlewis) |
| 21:33.02 | etfonhomey | p3nguin_, yeah, yeah |
| 21:33.09 | *** part/#asterisk dlewis (~dlewis@about/security/staff/dlewis) |
| 21:34.59 | jaytee | Katty, i still miss George and it's been almost 5 months now |
| 21:35.11 | *** join/#asterisk cidu (~PissInMyA@whthyt253-70.northwestel.net) |
| 21:35.14 | jaytee | but I expect I'll miss him till I'm dead |
| 21:35.45 | Katty | nods |
| 21:35.56 | Katty | jaytee: i found his previous owner on facebook. she's devistated too :/ |
| 21:35.58 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 21:36.08 | cidu | can anybody point me to a refrence for passing asterisk variables other than the 3 it does automatically to the externnotify command in voicemail.conf ? |
| 21:37.25 | jaytee | well, Shire is probably havin a wonderful time running around playing in the Summerlands and eventually the Goddess will send him back to the world for another turn. |
| 21:38.12 | Katty | i'm not sure i believe in all that |
| 21:38.26 | Katty | but if there was any suffering in the end, it's certainly over now |
| 21:38.35 | jaytee | true |
| 21:39.06 | Katty | tries to cheer up |
| 21:39.28 | jaytee | I like the Wiccan belief in the afterlife better than the Christian one cuz even our beloved pets and animals that serve us like horses and cattle can go there too |
| 21:39.35 | etfonhomey | ManxPower-work, Can I have a peak at your CallerID fixup script? |
| 21:41.38 | ManxPower-work | etfonhomey: http://pastebin.ca/1806077 |
| 21:42.03 | ManxPower-work | We normalize the callerid to have a leading 1 before it is sent to that script. |
| 21:42.42 | ManxPower-work | jaytee: I think belief in the afterlife is about as logical as believing in the easter bunny. |
| 21:43.19 | jaytee | ManxPower-work, HOW DARE YOU MOCK THE EASTER BUNNY!!!!! :-) |
| 21:44.03 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:44.25 | cidu | what im doing is ive got kannel set up and bound directly to our smsc, i want it to pass the callerid of voicemail messages in the mwi notification, but asterisk only passes context mailbox and number of messages to externnotify, specifying the variables directly after the command does not seem to work |
| 21:44.47 | jaytee | next he'll be telling me there's no Tooth Fairy. |
| 21:45.25 | p3nguin_ | WHAT?! There's no Tooth Fairy?! |
| 21:45.55 | cidu | now now, nobody is claiming that |
| 21:46.15 | cidu | how do ya think west virginians suppliment their welfare? |
| 21:49.36 | etfonhomey | ManxPower-work, thanks for the script! |
| 21:49.46 | etfonhomey | I'm going to modify it to handle length=0. |
| 21:51.00 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 21:51.09 | *** part/#asterisk mikealeonetti (~mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 21:53.17 | jaytee | cidu, LOL |
| 21:53.53 | QubeZ | ManxPower-work im not entirely sure how even if a network with CDP issues, that would just make the polycom not get the correct VLAN ID right? Does CDP do something else throughout the phone's existence (online) on the network? |
| 21:54.20 | jaytee | quittin time, bbl |
| 21:54.55 | cidu | so, no idea how to pass extra data to the externnotify command in voicemail.conf? |
| 21:55.13 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 21:55.30 | Geminizer | is there a more forceful way to drop a channel than issuing "soft hangup" ? |
| 21:56.39 | etfonhomey | QubeZ, do you use PoE with your phones or power bricks? |
| 21:56.47 | QubeZ | etfonhomey PoE |
| 21:57.10 | ManxPower-work | QubeZ: I don't know. All I know that over the years maybe 20 people have solved random, in the middle of the call audio drop outs by disabling CDP on the phones and they did not have CDP setup on their network. |
| 21:57.11 | etfonhomey | CDP can be used to negotiate power requirements as well. |
| 21:57.14 | QubeZ | sometimes when we run out of PoE cables, we use the bricks but 98% of the office is on PoE. |
| 21:57.46 | QubeZ | etfonhomey ok, im troubleshooting an issue where we have one group of agents complain about 3-5 sec audio delays then they are able to hear teh customer again |
| 21:58.30 | ManxPower-work | QubeZ: you said earlier that this happened on internal calls. |
| 21:58.36 | etfonhomey | QubeZ, I would go with ManxPower-work's suggestion. He has more experience than I. |
| 21:59.25 | QubeZ | ManxPower-work the phones are using SIP to register, but we are getting calls from outside |
| 21:59.28 | ManxPower-work | If it is happening on outside calls as well as inside calls then there is a million things that could cause that. If it's only happening on the local lan then there's only a few things that could cause that problem |
| 22:00.20 | QubeZ | ManxPower-work ya man, thats my issue here. It happens randomly, only one group of agents (our largest inbound group) is complaining of the issue. Also, happens intermittenly on inbound->inbound calls. |
| 22:00.22 | QubeZ | very puzzling |
| 22:01.56 | QubeZ | first ill upgrade a few of the phones over to new firmware and see what happens then take steps after that |
| 22:02.16 | QubeZ | not sure how cdp could behave strangely, seems simple enough of a protocol |
| 22:02.18 | jayrod422 | ok im banging my head on this one.. has anyone seen "Maximum retries exceeded on transmission issues" popping up more often recently? |
| 22:02.53 | ManxPower-work | jayrod422: that caller hung up |
| 22:02.57 | jayrod422 | no |
| 22:03.00 | jayrod422 | no one hung up |
| 22:03.02 | jayrod422 | just asterisk |
| 22:03.12 | ManxPower-work | It's either that or packets are being dropped |
| 22:03.20 | jayrod422 | seems like softphones are not relplying the ack back to 200 OKs |
| 22:04.28 | jayrod422 | ive done a million traces, tried a variety of softphone (zoiper/xlite) and keep seeing the 200ok going to the phones but they wont ack so asterisk drops the call after 5 tries |
| 22:04.55 | ManxPower-work | Soundds NATy to me. |
| 22:05.03 | jayrod422 | natty is right |
| 22:05.12 | jayrod422 | but i havent drank that in years |
| 22:05.18 | *** join/#asterisk ddickenson (~Drew@67-198-0-5.static.grandenetworks.net) |
| 22:05.33 | jayrod422 | did something change lately in the way asterisk handles nat |
| 22:05.52 | cidu | you using variable length class c subnets? i had quite a bit of call dropping in that situation |
| 22:05.53 | jayrod422 | i see the 200 going out of the asterisk box and on the x-lite box i see it coming in |
| 22:06.02 | ddickenson | any favorite sip/iax trunk providers for a small business? |
| 22:06.10 | cidu | gets the wrong mask somewhere and nothing comes back cause its acking to the wrong network |
| 22:06.17 | ManxPower-work | jayrod422: from the correct IP address and ports in the packet header and the sip data? |
| 22:06.28 | Katty | bmoraca_work: ping |
| 22:07.00 | jayrod422 | manx: yes from and to stuff in the headers look correct |
| 22:07.12 | etfonhomey | Does/Can the registrar tell the registering device what the re-registration period is? When i turn on sip debug for my ITSP, I see this a lot: NOTICE[30793] chan_sip.c: Outbound Registration: Expiry for 64.2.142.31 is 60 sec (Scheduling reregistration in 45 s) |
| 22:07.49 | cidu | etfonhomey, isnt that the qualify setting? |
| 22:07.59 | ManxPower-work | cidu: no, it is not |
| 22:08.25 | jayrod422 | i put a trace if anyone can see what the hell im going wrong: http://www.pastebin.com/d42791ce8 |
| 22:08.31 | etfonhomey | This line is the 200 OK response from my ITSP: Contact: <sip:s@MYIPADDRESS>;expires=60 |
| 22:09.15 | etfonhomey | cidu, qualify deals with OPTIONS, doesn't it? |
| 22:09.31 | ManxPower-work | etfonhomey: Are you expecting it to wait until the registration has expired before renewing? |
| 22:10.39 | etfonhomey | ManxPower-work, no, still trying to figure out sip.conf variable to set re-registration interval. |
| 22:12.23 | edwin_quijada | Hi! |
| 22:12.37 | etfonhomey | ManxPower-work, think I find it in the docs: register => [transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] <---------HERE |
| 22:12.39 | edwin_quijada | there is any changes to perl AGI |
| 22:13.08 | edwin_quijada | I am an AGI from my asterisk and this begin to exceute but it doesnt anything |
| 22:14.46 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 22:16.29 | etfonhomey | Actually, that was for 1.6.1, here's the same line in 1.4 docs: register => user[:secret[:authuser]]@host[:port][/extension] |
| 22:16.44 | p3nguin_ | etfonhomey: I'm not sure ... |
| 22:17.07 | p3nguin_ | etfonhomey: I was going to say I'm not sure if that first line was valid for all versions, but I think you complemented it with the second line. |
| 22:17.28 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:3ce4:10b0:a077:acde) |
| 22:17.29 | cusco | hi |
| 22:17.46 | cusco | what is the most common cause for this warning: "chan_sip.c:23861 build_peer: Invalid peer port configuration at line 0 : 0" ? |
| 22:18.01 | cusco | peer port configuration |
| 22:18.03 | cusco | ? |
| 22:18.14 | cusco | it is a realtime peer |
| 22:18.16 | cusco | in mysql db |
| 22:24.43 | Geminizer | does anyone know how to drop a channel w/o restarting asterisk ? |
| 22:25.17 | p3nguin_ | soft hangup sometimes works. |
| 22:26.12 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 22:28.14 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 22:28.25 | *** part/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 22:30.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:35.02 | edwin_quijada | I have a weird error from my AGI |
| 22:35.15 | edwin_quijada | any agi doesnt exceute with asterisk |
| 22:35.41 | ManxPower-work | edwin_quijada: stop asterisk, start it as "asterisk -cvvv" and then issue "agi debug" and try it agaihn |
| 22:35.54 | edwin_quijada | I can see in debug-cli of * this script running |
| 22:36.15 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 22:38.21 | edwin_quijada | ManxPower-work: i did but I dont understand -cvvvv options |
| 22:38.42 | edwin_quijada | how can I use "agi debug" |
| 22:39.06 | ManxPower-work | edwin_quijada: -c means "start asterisk in the forground, not the background. This means that the errors from your script will be seen on the console instead of being discarded. |
| 22:39.15 | ManxPower-work | edwin_quijada: type "agi debug on" that's how you use it. |
| 22:40.27 | edwin_quijada | ok |
| 22:44.40 | bmoraca_work | Katty: you have an awefully bad habbit of hollaring at me when i'm not here, heh |
| 22:44.47 | *** part/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 22:47.42 | edwin_quijada | ManxPower-work: Thks!! |
| 22:48.19 | edwin_quijada | I solved the problem, the module perl for asterisk it wasnt installed |
| 22:49.44 | cusco | hi... |
| 22:50.00 | cusco | ${EXTEN:2} of XXX gives me the last digit |
| 22:50.04 | cusco | how do I get the first? |
| 22:51.02 | cusco | like 621, I want to pick out the 6 only |
| 22:51.02 | cusco | please...? |
| 22:51.21 | bmoraca_work | ${EXTEN:1:1}, or 0:1...i can't remember if strings in asterisk are 0 or 1 based in position |
| 22:51.52 | bmoraca_work | it's 0-based |
| 22:51.59 | bmoraca_work | so ${EXTEN:0:1} |
| 22:53.48 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:56.27 | ManxPower-work | cusco: that is fully documented in asterisk.pdf and channelvariables.tex |
| 22:56.31 | ManxPower-work | As well as the Wiki |
| 22:59.52 | edwin_quijada | there is different TTS of cepstral, loquendo, festival and Verbio that can I use with asterisk? |
| 23:01.59 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 23:05.20 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 23:06.55 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 23:08.24 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 23:11.28 | edwin_quijada | I have a link with avaya PBX using H323 and this link has 23 channels but I dont know how can I receive all theses channels in my asterisk. I ahve an IVR in asterisk to repsond to all channels. I need to create a queue to do that? |
| 23:11.32 | edwin_quijada | Any cluue? |
| 23:14.26 | cusco | bmoraca_work: ok thanks |
| 23:14.30 | cusco | ManxPower-work: noted |
| 23:14.36 | cusco | another question |
| 23:15.13 | cusco | does iax consume less bandwidth than SIP ? |
| 23:16.34 | voipmonk | depends on the amount of channels and codec you use, cusco - what are you trying to do? |
| 23:17.27 | *** join/#asterisk Walsh86 (~Walsh86@CPE00c09fb5611e-CM0014e88f53ec.cpe.net.cable.rogers.com) |
| 23:18.02 | cusco | ok I was using SIP with ulaw |
| 23:18.25 | cusco | now we set up a local asterisk taht cnnects with main asterisk trough IAX2 ulaw as well |
| 23:18.33 | cusco | bandwidth seems to be the same tho |
| 23:18.55 | Kobaz | it depends on the codec |
| 23:18.58 | cusco | however there seems to be a slight change in the audio quality |
| 23:19.03 | Kobaz | iax and sip are just signalling |
| 23:19.08 | cusco | oh... |
| 23:19.13 | cusco | :( |
| 23:19.19 | *** join/#asterisk LemensTS (~customgt@70.238.161.207) |
| 23:19.37 | Kobaz | iax embeds the audio in the same stream as the signalling so it results in being nicer with nat |
| 23:19.55 | LemensTS | if i DIAL(SIP/1234&SIP/1235) and sip 1234 answers it, how can I know that sip 1234 answered it for use as a variable? |
| 23:20.10 | Kobaz | use a pickup macro |
| 23:20.12 | cusco | we have no nat (vpn) |
| 23:20.14 | Kobaz | look at the M option |
| 23:20.40 | Walsh86 | Hi, I have a TDM400P with two FXS's. Should the FXS's be putting out a dialtone regardless of configuration once DAHDI is installed? phones ARE getting power from card but no dial tone. Thanks |
| 23:21.00 | cusco | so the BW will always depend on the codec... |
| 23:21.22 | cusco | gsm is not good enough.. we are using ulaw, but strugling with BW |
| 23:21.24 | cusco | :-( |
| 23:21.39 | LemensTS | Kobaz: thanks |
| 23:22.23 | Kobaz | LemensTS: there's also U for using gosub, if you're using ast 1.6 |
| 23:22.31 | edwin_quijada | How can I receive a fwe channels from another pbx to my asterisk? |
| 23:22.40 | Kobaz | fwe? |
| 23:22.45 | edwin_quijada | few |
| 23:22.57 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:23.00 | Kobaz | you need a technology that links them, and the configuration set up |
| 23:23.01 | edwin_quijada | I have an integration with avaya using H323 so now |
| 23:23.16 | Kobaz | technology would be, analog,tdm,sip,iax,mgcp,etc |
| 23:23.39 | Kobaz | asterisk has h323 via some extra modules |
| 23:23.45 | edwin_quijada | the avaya send me 23 channels but I dont know how receive theses channels |
| 23:23.47 | Kobaz | i haven't had much luck getting h323 to work |
| 23:23.51 | Kobaz | edwin_quijada: t1? |
| 23:24.01 | edwin_quijada | the link works fine |
| 23:24.09 | Kobaz | what TYPE of link is it |
| 23:24.16 | edwin_quijada | H323 |
| 23:24.18 | Kobaz | okay |
| 23:24.37 | Kobaz | you'll need to poke someone who knows the h323 modules... i don;t |
| 23:24.45 | edwin_quijada | but i dont understand all calls from this channel |
| 23:24.55 | edwin_quijada | where going to |
| 23:29.36 | bmoraca_work | cusco: with IAX, the more channels you have, the less bandwidth it takes relative to SIP because the signalling is handled differently. so, as your number of active channels increases, the difference in total bandwidth also increases. however, SIP is generally recommended anyway |
| 23:31.19 | Walsh86 | hey sorry to repost my question again but dont know if it was seen or not... I have a TDM400P with two FXS modules. Should the FXS's be putting out a dialtone regardless of configuration once DAHDI is installed? phones ARE getting power from card but no dial tone. don't know if its a config or hardware issue. Thanks |
| 23:32.28 | cusco | bmoraca_work: I just noticed that |
| 23:32.31 | cusco | it is true |
| 23:32.32 | cusco | :) |
| 23:32.56 | cusco | takes less BW |
| 23:33.01 | cusco | why is SIP recommended? |
| 23:35.03 | Kobaz | cusco: much more control over the signalling |
| 23:35.16 | Kobaz | cusco: iax is also quite slow to tear down hung up channels |
| 23:35.39 | Kobaz | Walsh86: ports will not get dialtone if they are not configured properly in chan_dahdi.conf |
| 23:36.00 | Kobaz | Walsh86: and asterisk must be running |
| 23:36.45 | *** join/#asterisk TecR0c (tecr0c@never.met.a.leet-hacker.net) |
| 23:36.54 | TecR0c | hey |
| 23:37.08 | Walsh86 | ok thankyou, thats all i wanted to know. Just didnt want to get into it too far if there was a hardware problem |
| 23:37.15 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:38.24 | cusco | ok thanks bye |
| 23:41.13 | TecR0c | i have a trixbox asterisk box. All of sudden my snom voip phones and linksys phones are displaying initalzing network. zaptel zt_defaultzone failed invalid argument 22 and connect attempt from 127.0.0.1 unable to auth which is displayed in asterisk -r |
| 23:52.30 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:54.52 | jaskew | TecR0c: just guessing here. Could it be either a) a DHCP issue, or b) a DNS issue [or both...]? |
| 23:57.29 | dmast | Has anyone gotten MWI to work from Exchange 2010 to * using the unsolicited_mailbox option? |
| 23:57.40 | TecR0c | i think it was my ISP |
| 23:57.50 | TecR0c | just did a powercycle on the router |
| 23:58.08 | TecR0c | workiing now however the call is dsiorted |
| 23:58.58 | TecR0c | might take some time to calm down |
| 23:59.30 | tkrn | anybody configure a 3com 3101 sip phone in here? |