| 00:00.10 | hmmhesays | Qby: set up your queue right? |
| 00:00.16 | hmmhesays | and the dialplan accordinging |
| 00:00.17 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:00.20 | hmmhesays | *accordingly |
| 00:00.25 | TripleFFFF | whats the vcdial crap name again ? |
| 00:00.39 | hmmhesays | vicidial |
| 00:00.47 | TripleFFFF | ho |
| 00:00.57 | TripleFFFF | lol |
| 00:00.59 | TripleFFFF | http://blog.tmcnet.com/blog/tom-keating/voip/dial-up-voip-claim.asp |
| 00:01.12 | TripleFFFF | New Patented Technology Will Allow Free Phone Calls Even Over Dial Up Connection. |
| 00:01.32 | hmmhesays | crazy |
| 00:01.42 | hmmhesays | what is this VoIp you speak of? |
| 00:01.43 | TripleFFFF | VoiP Technology Companies offer their services to broadband users only. Now using a new patented technology anyone with an Internet connection can use VoiP to make free phone calls. |
| 00:01.57 | TripleFFFF | ?????? |
| 00:01.58 | TripleFFFF | Thanks to a small technology company based out of Ireland the option of using VoiP soft phones over a dial up connection successfully has arrived |
| 00:02.11 | hmmhesays | um people have been doing that for a long time |
| 00:02.24 | TripleFFFF | http://www.superiorvoip.com/pr-release.htm |
| 00:02.29 | TripleFFFF | no wonder site doesnt exist |
| 00:02.38 | hmmhesays | gsm or ilbc over a 33.6k connection is fine as long as ping times are decent |
| 00:02.39 | TripleFFFF | lol man |
| 00:02.42 | hmmhesays | and there isn't much jitter |
| 00:02.47 | TripleFFFF | you need 8KB |
| 00:02.51 | TripleFFFF | on g729 |
| 00:03.08 | hmmhesays | yeah |
| 00:03.14 | hmmhesays | overhead about 12k |
| 00:04.24 | QbY | hmmhesays.. Do you have any suggestions, or suggested reading material for Queue and Dialplan configuration? |
| 00:04.40 | hmmhesays | are you using freepbx? |
| 00:04.59 | QbY | No regular * |
| 00:05.11 | hmmhesays | then you should know exactly how to set it up right, |
| 00:05.17 | hmmhesays | if you set the dialplan up in the first place |
| 00:05.39 | QbY | And I did. It works perfectly, except for my calls in queue.. |
| 00:08.12 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 00:11.16 | *** join/#asterisk mne (n=mne@chello080108001212.35.11.tuwien.teleweb.at) |
| 00:13.41 | mne | hi there! I'm just playing with asterisk. I know that modem/fax over a codec is the worst one can do, however it seesm that using G.711 or something similar allows slow (1200bps and similar) connections. So I installed iaxmodem, connected 2 iaxmodems to asterisk over IAX2 and tried to connect them. so far I had no success ;( did anyone have success with this ? |
| 00:16.55 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:16.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:17.23 | hmmhesays | send calls to a diaplan extension that doesn't have voicemail QbY |
| 00:19.15 | mne | there is no voicemail in the dialplan for the iaxmodems |
| 00:19.34 | QbY | hmmhesays.. I'm building a new context for the agents to loginto.. |
| 00:19.37 | QbY | log into* |
| 00:24.43 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 00:28.17 | *** join/#asterisk RF_MIA (n=Administ@68-235-157-35.miamfl.adelphia.net) |
| 00:30.58 | nick125_lappy | anyone here good with "customizing" meetme? i.e. putting custom greetings, etc, on there without changing all the sound files globally? |
| 00:31.43 | rikstah | nick125_lappy, can't just set the language in the dialplan |
| 00:32.09 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 00:32.31 | nick125_lappy | I only want a few meetmes to have the custom greeting.. |
| 00:32.57 | nick125_lappy | I wonder if it would just be easier to make my own prompts (for creating a conference, etc), then, call meetme when needed |
| 00:33.11 | nick125_lappy | but, I don't think theres a way to make a custom prompt in dialplan |
| 00:35.07 | nick125_lappy | for my little free conferencing service, I got a custom prompt for the intro, then, when they go to create a conference, it goes to the standard asterisk prompts...doesn't blend real well |
| 00:37.26 | *** join/#asterisk Givemelove (n=bozoo@208.57.229.162) |
| 00:37.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
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| 00:41.14 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 00:42.48 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:45.00 | *** join/#asterisk Givemelove (n=bozoo@208.57.229.162) |
| 00:48.31 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 00:50.59 | X-Rob | nick125_lappy, replace the .gsm files that it's using with whatever you want |
| 00:51.14 | X-Rob | oh, only a few |
| 00:51.28 | X-Rob | set them to a different language before going in then |
| 00:54.52 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:56.53 | *** join/#asterisk RF_MIA (n=Administ@68-235-157-35.miamfl.adelphia.net) |
| 00:57.12 | rikstah | all he needs to do is set the language in the dialplan and then put the prompt files in the new directory |
| 01:04.10 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:08.40 | *** join/#asterisk HaMYaI (i=CKGLOB@61.47.115.113) |
| 01:09.43 | HaMYaI | I have 1xTDM400 + 1xTE110P, which one should come first in zaptel.conf? |
| 01:11.11 | HaMYaI | when I put the TE110P first and the /proc/zaptel/1 shows info from TDM and when I put TDM400 first it just behaves the other way around |
| 01:11.18 | HaMYaI | very confusing |
| 01:11.38 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 01:13.03 | Strom_C | what order are you loading the drivers in? |
| 01:14.59 | HaMYaI | Strom_C, let me check |
| 01:15.05 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:16.44 | HaMYaI | Strom_C, wctdm first and then wct1xxp |
| 01:17.02 | HaMYaI | Strom_C, in /etc/modprobe.d/zaptel |
| 01:17.11 | HaMYaI | hope I look at the right place |
| 01:17.36 | Strom_C | i would load them the other way round |
| 01:17.48 | Strom_C | load your t1 first, then the tdm card |
| 01:18.30 | Strom_C | then configure the t1 as channels 1-24 |
| 01:19.03 | HaMYaI | Strom_C, more info http://pastebin.ca/163897 |
| 01:19.27 | HaMYaI | Strom_C, ok will give it a try |
| 01:19.37 | Strom_C | yeah, b-channels are not to be found on analog cards |
| 01:19.41 | Strom_C | hang up and try again please |
| 01:21.01 | nick125_lappy | X-Rob, rikstah: thanks |
| 01:21.11 | nick125_lappy | I'll give that a try once I'm done cleaning my room :) |
| 01:21.18 | rikstah | np |
| 01:21.32 | rikstah | nick125_lappy, look at voip-info.org |
| 01:24.59 | RF_MIA | Hi. I need to be able to force a BUSY on all Zap channels on a T1 to notify a downstream Legacy PBX that the trunk is not available......Any thoughts on how I might approach this? |
| 01:25.14 | HaMYaI | Strom_C, ohh, that works like a charm |
| 01:25.32 | Strom_C | RF_MIA: busy on ALL channels? |
| 01:25.35 | Strom_C | at the same time?? |
| 01:25.38 | HaMYaI | Strom_C, thanks dude |
| 01:25.40 | RF_MIA | Yes...at the same time |
| 01:25.44 | Strom_C | HaMYaI: you're welcome |
| 01:26.01 | RF_MIA | I have Asterisk setup as a T1-to-IAX gateway on a legacy Mitel pbx |
| 01:26.03 | Strom_C | RF_MIA: ......what the hell kind of bonkers stupid PBX requires /that/? |
| 01:26.11 | RF_MIA | See above :) |
| 01:26.42 | Strom_C | under what conditions would the trunk not be available? |
| 01:26.48 | Strom_C | and am I correct in assuming this is CAS? |
| 01:26.49 | RF_MIA | In order to allow for failover if my IAX trunk goes down or the Internet I need to notify the Mitel that the T1 tie-line between the Mitel and Asterisk is completely busy |
| 01:27.39 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 01:27.55 | RF_MIA | d4,ami |
| 01:28.02 | Strom_C | blech |
| 01:28.09 | Strom_C | hmm |
| 01:28.14 | RF_MIA | The Mitel is an oldie |
| 01:28.21 | Strom_C | yeah, I figured :) |
| 01:28.31 | orlock | Mitel.. Mitel Networks? |
| 01:28.34 | Dr-Linux | question, i have a sip trunk between two * boxes, so when the call being forwarded it rings , how can i skip this ring? |
| 01:28.46 | RF_MIA | Short of just doing a Dial(all freaking Zap channels)....I can't think of anything more elegant |
| 01:29.01 | RF_MIA | Mitel SX-200 i believe |
| 01:29.12 | Strom_C | what is the alternate route that the mitel should route out over? |
| 01:29.17 | Dr-Linux | anyclue in my question? |
| 01:29.22 | RF_MIA | It has it's own T1/PRI |
| 01:29.38 | RF_MIA | which would be route 2 if the IAX link goes down |
| 01:29.46 | Strom_C | RF_MIA: what about....using a dual-span card and letting asterisk handle the failover routing |
| 01:29.48 | RF_MIA | Kind of a least cost routing |
| 01:30.15 | RF_MIA | damn....I didn't think of that one |
| 01:30.22 | *** part/#asterisk HaMYaI (i=CKGLOB@61.47.115.113) |
| 01:30.24 | RF_MIA | but that would work.... |
| 01:30.36 | Strom_C | certainly there's less kludgery involved |
| 01:31.07 | Strom_C | dr-Linux, how are you handling the forwarding? |
| 01:31.15 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:31.29 | RF_MIA | I just have a 1 X t1 in that server now...so that is a limitation |
| 01:31.45 | RF_MIA | Plus I'd rather have the Mitel be the judge of where to route the calls..... |
| 01:32.39 | Strom_C | what happens if asterisk just returns congestion on that specific channel? |
| 01:32.41 | Dr-Linux | Strom_C: i have sip trunk, simply i have a parttern on serverA after dialing that users goes to serverB |
| 01:33.00 | Strom_C | dr-Linux, show me the dial statement in question |
| 01:33.04 | RF_MIA | If I return congestion or busy on a specific Zap then the Mitel tries the next available channel |
| 01:33.05 | Dr-Linux | Strom_C: and during this move user listen a ring |
| 01:33.28 | Dr-Linux | Strom_C: hold on |
| 01:33.48 | Strom_C | RF_MIA: hmmm |
| 01:34.00 | RF_MIA | A doozy eh?! |
| 01:34.08 | Strom_C | can you provision a PRI from the mitel to the asterisk box? |
| 01:34.51 | RF_MIA | Tried that one...The old clunker needs a license upgrade to do anything other then RBS |
| 01:35.08 | RF_MIA | so it's a T1 E&M |
| 01:35.47 | Strom_C | so how do you have a PRI coming into it now? is it a per-span license? |
| 01:35.51 | Dr-Linux | Strom_C: |
| 01:36.11 | Strom_C | dr-Linux, take that r off the end of the dial command |
| 01:36.25 | Strom_C | dr-Linux: you should never need to use the r flag |
| 01:36.36 | Dr-Linux | awww i see |
| 01:36.41 | RF_MIA | I don't manage the Mitel....but the Admin informs me that in fact it is a per-span license |
| 01:36.48 | Dr-Linux | Strom_C: this r is for ring? :S |
| 01:36.56 | Strom_C | dr-Linux, yes |
| 01:37.08 | Dr-Linux | great help |
| 01:37.36 | Dr-Linux | Strom_C: actually i have different servers are trunking with eacher other |
| 01:37.41 | Strom_C | RF_MIA: honestly, i think it would be easier to get a dual-span card for the asterisk box and let it make the iax/pri decision |
| 01:38.19 | Dr-Linux | Strom_C: so what you suggest, caller dial string should be on first server or the serverB whre caller is being forwarded? |
| 01:38.37 | Strom_C | dr-Linux: just try removing the r first and see if that works |
| 01:38.39 | RF_MIA | Strom_C, I agree with you. I would certainly feel more comfortable using that scenario...but alas the customer might not |
| 01:39.08 | RF_MIA | I'll keep digging around for some clever way of doing this...thanks for the help though Strom! |
| 01:39.17 | Dr-Linux | Strom_C: correct that i understand .. that question is cleared :) |
| 01:39.46 | Strom_C | I think it's time for a nap |
| 01:39.56 | Dr-Linux | :S |
| 01:40.39 | RF_MIA | I have another good one for any takers... |
| 01:41.04 | Strom_M | im not gone yet |
| 01:41.08 | RF_MIA | Anyone with experience trunking Asterisk E1--E1 Cisco using QSIG? |
| 01:41.08 | Strom_M | im just on the laptop now :) |
| 01:41.37 | Dr-Linux | Strom_C: do you have any clue on my 2nd question? |
| 01:41.37 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:41.40 | file | Strom_M: why leave? :( |
| 01:42.43 | Strom_M | ? |
| 01:43.22 | Dr-Linux | Strom_M: so what you suggest, caller dial string should be on first server or the serverB whre caller is being forwarded? |
| 01:43.40 | Strom_M | Dr-Linux, i honestly have no clue what you're going on about |
| 01:43.54 | Strom_M | also, for inter-asterisk box trunking, use iax2 :) |
| 01:45.15 | Dr-Linux | Strom_C: like on serverA if a caller dials 32XXXXXX and go to serverB |
| 01:45.16 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 01:45.19 | DasTech | hello |
| 01:45.30 | DasTech | and how do you reset aastdb |
| 01:45.38 | DasTech | it seems mine is not updating |
| 01:46.03 | Dr-Linux | the other way reached to serverB and dials 32XXXXXXX , what will be the good |
| 01:46.20 | DasTech | phones cant register |
| 01:46.32 | Dr-Linux | :( don't know how should i ask my question .. anyway ... |
| 01:46.42 | RF_MIA | There is a database command I believe in the CLI |
| 01:46.42 | DasTech | ? |
| 01:47.49 | DasTech | nothing about clearing it out and having it reset |
| 01:49.23 | RF_MIA | DasTech: short of reloading or restarting Asterisk...I'm not sure |
| 01:50.09 | DasTech | if I rm the astdb file will it rebuild it ? |
| 01:51.15 | RF_MIA | Not sure dastech |
| 01:51.25 | RF_MIA | backup just to be safe :) |
| 01:53.14 | RF_MIA | QSIG + Asterisk....any takers? |
| 01:55.18 | *** part/#asterisk RF_MIA (n=Administ@68-235-157-35.miamfl.adelphia.net) |
| 01:56.43 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:57.17 | *** join/#asterisk doolph (n=notengo@200.46.148.58) |
| 01:59.35 | hmmhesays | fun fun |
| 02:00.49 | doolph | hey anyone know a real good doc or tutorial about Asterisk behind NAT, sip clients is trying to connect from outside |
| 02:03.19 | *** part/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 02:03.23 | *** join/#asterisk _Vile (i=_Vile@198.175.14.242) |
| 02:05.11 | tengulre | who are come from china? |
| 02:05.27 | justinu|laptop | all your base are belong to us |
| 02:12.42 | hmmhesays | lol |
| 02:14.22 | *** join/#asterisk nitrus^ (n=asdf@ip70-187-148-2.oc.oc.cox.net) |
| 02:14.52 | nitrus^ | how do i get asterisk to work behind a nat talking to a soft phone behind a nat, is that even possible at this point? * - nat - net - nat - xlite |
| 02:19.55 | doolph | it is possible |
| 02:20.16 | nitrus^ | i setup all the nat stuff in asterisk for sip.conf and it still gives me no sound |
| 02:20.30 | nitrus^ | i can login just fine and the dial goes through and rings a phone but when i pickup there is no sound |
| 02:21.28 | doolph | its pretty hard, the best way is install vpn |
| 02:21.36 | doolph | best and easier |
| 02:22.01 | nitrus^ | hmm good point |
| 02:22.35 | nitrus^ | maybe ill try xtunnels |
| 02:23.35 | doolph | I have the same problem, but I am worst because I am trying to use a sip hardware |
| 02:24.15 | nitrus^ | ick |
| 02:24.37 | doolph | are you using linux router? |
| 02:24.39 | nitrus^ | youd have to put your entire network inside the VPN on the remote side |
| 02:24.40 | nitrus^ | yeah |
| 02:24.55 | doolph | so are you forwarding the ports |
| 02:25.00 | doolph | 5060 and 10000-20000 |
| 02:25.05 | nitrus^ | hmm |
| 02:25.07 | nitrus^ | just 5060 |
| 02:25.11 | nitrus^ | maybe that's why |
| 02:25.12 | doolph | that's why |
| 02:25.20 | nitrus^ | is that where the sound packets come in and out? |
| 02:25.23 | doolph | yes |
| 02:25.27 | nitrus^ | lol |
| 02:25.31 | nitrus^ | whoops |
| 02:25.33 | nitrus^ | ill try that |
| 02:25.38 | doolph | 5060 is just for login |
| 02:26.54 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 02:27.32 | doolph | nitrus^ but i think for you the best solution is vpn network to network |
| 02:27.48 | nitrus^ | i agree |
| 02:27.59 | nitrus^ | im just hoping the added encryption time etc doesnt lag the call |
| 02:28.51 | doolph | I don't think so, it could be even better |
| 02:29.23 | doolph | if you do compression |
| 02:29.25 | *** join/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 02:30.09 | doolph | by the way, can you paste your firewall script |
| 02:31.18 | nitrus^ | the entire thing? |
| 02:31.36 | nitrus^ | it's 400 lines |
| 02:31.50 | *** part/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 02:31.53 | nitrus^ | do you want nat relavent lines? |
| 02:32.15 | doolph | you can use pastebin.ca |
| 02:32.23 | doolph | yes only that port forward rules |
| 02:32.28 | nitrus^ | alright |
| 02:32.30 | nitrus^ | ill PM you |
| 02:32.32 | doolph | I don't know why mine isn't working |
| 02:34.11 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 02:34.14 | Strom_M | what do you mean "it could be even better if you do compression" |
| 02:46.27 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 02:48.16 | doolph | there's vpns with compressions |
| 02:49.18 | fafnir | theres vpns with werewolf plugins |
| 02:49.22 | doolph | anyone know why I cannot register when I am behind nat (the error is unauthorized) |
| 02:50.17 | Strom_C | doolph: you'll lose more in compression time than you will just spitting out the uncompressed voice data |
| 02:50.48 | Strom_C | compression only reduces transit time in situations where the data being transferred is not time sensitive |
| 02:51.53 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 02:52.09 | doolph | really |
| 02:52.14 | Strom_C | yes |
| 02:52.20 | Strom_C | i.e. transferring an iso file or somesuch |
| 02:52.26 | *** join/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 02:52.52 | Strom_C | but with voice, you know you are going to be transferring a specific number of bits in a given amount of time |
| 02:53.14 | Strom_C | transferring those bits faster doesn't help, because they can't be played faster into the other party's ear |
| 02:53.17 | anonymouz666 | memcpy() is producing a seg fault in my code |
| 02:53.26 | doolph | can you suggest me something good for qos |
| 02:53.54 | florz | anonymouz666: then you are probably using it wrongly =:-) |
| 02:54.06 | hmmhesays | bah bah bah, i need some fargo nd did's |
| 02:54.11 | doolph | i have adsl 786/256, 1 linux router (vpn), 1 asterisk box, a lan with 4 computer |
| 02:54.15 | hmmhesays | what companies are you guys using for inbound DID's? |
| 02:54.36 | doolph | hmmhesays US? |
| 02:55.05 | hmmhesays | yeah |
| 02:55.26 | doolph | free or paying? |
| 02:55.51 | hmmhesays | paying |
| 02:56.24 | doolph | paying there's lots |
| 02:56.37 | doolph | vonage, deltathree, etc |
| 02:56.40 | doolph | just gogle |
| 02:57.05 | doolph | all of them are good |
| 02:57.13 | hmmhesays | yeah something I can use with asterisk though |
| 02:57.45 | doolph | any that can supports SIP |
| 02:58.44 | hmmhesays | vonage won't let you, deltathree shy's upon it |
| 02:58.55 | Strom_C | and remember: sip == headache |
| 02:59.39 | hmmhesays | not really |
| 02:59.49 | anonymouz666 | why SIP == headache? |
| 03:00.27 | doolph | everyone is still using SIP |
| 03:00.34 | doolph | but it's headache really |
| 03:01.09 | Strom_C | ok...well, granted, SIP isn't as much of a headache as H.323, where the H actually /stands/ for "Headache" |
| 03:01.28 | Strom_C | but SIP was not designed with reality in mind |
| 03:01.40 | Strom_C | and so therefore the P stands for "Pain-in-the-ass: |
| 03:01.44 | Strom_C | s/:/"/ |
| 03:02.14 | FuriousGeorge | do any of the major bells here in the us have a service that one can interface w/ asterisk |
| 03:02.21 | Strom_C | yes |
| 03:02.32 | Strom_C | it's called ISDN Primary Rate Interface |
| 03:02.34 | doolph | what QOS solution do you suggest Strom_C? |
| 03:02.35 | FuriousGeorge | nj, usa Strom_C? |
| 03:02.41 | FuriousGeorge | oh |
| 03:02.44 | FuriousGeorge | not funny |
| 03:02.52 | Strom_C | how is that not funny? |
| 03:02.53 | Strom_C | it's a service |
| 03:02.57 | Strom_C | it interfaces with asterisk |
| 03:03.01 | Strom_C | it meets your criteria |
| 03:03.09 | FuriousGeorge | thats a physical line comming into your building |
| 03:03.22 | Strom_C | over which services are provided |
| 03:03.37 | FuriousGeorge | im just talking about termination over someone else's braodband connection type o' voip |
| 03:03.41 | file | you were not specific enough, so Strom_C is correct |
| 03:03.47 | Strom_C | technically, PRI /is/ the service; the physical line is called "T1" |
| 03:04.19 | file | you could have a point to point T1 and run VoIP over it ... |
| 03:04.42 | Strom_C | doolph: what kind of qos problem are you trying to solve |
| 03:05.28 | doolph | I don't know if it is internet problem or the asterisk CPU problem |
| 03:05.35 | doolph | or memory |
| 03:05.45 | Strom_C | well, explain it to me so I don't have to turn on my ESP |
| 03:06.03 | FuriousGeorge | my parents use a service from att that works with their router device. after a shaky first month or two its been reliable as a landline. ive tried a couple of different sip and iax services with asterisk, and there's always periods where users complain they sound "robotic to others" |
| 03:06.15 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 03:06.17 | FuriousGeorge | i dunno if its my server or my provider, but im leaning toward the later |
| 03:06.19 | Strom_C | FuriousGeorge: what codecs are you using? |
| 03:06.24 | FuriousGeorge | ulaw |
| 03:06.40 | doolph | ok I got an office with 4 machines, 2 hardphones, 2 softphones, 2 remote office (2 phones) |
| 03:06.43 | Strom_C | what does your pipe to the provider look like |
| 03:06.52 | FuriousGeorge | a piece of cat 5 :) |
| 03:07.00 | Strom_C | uh, no |
| 03:07.05 | FuriousGeorge | actually a coaxial cable |
| 03:07.19 | Strom_C | cable company? |
| 03:07.23 | FuriousGeorge | yeah |
| 03:07.35 | Strom_C | bandwidth / latency / variation in latency? |
| 03:07.47 | FuriousGeorge | 300kBytes up and like 2000 kil bytes down i ping google and other boxes at like 13 ms |
| 03:07.54 | FuriousGeorge | 13-20 |
| 03:08.10 | doolph | then I am getting bad quality sometimes |
| 03:08.21 | doolph | maybe someone is trying to download emails or browsing |
| 03:08.24 | Strom_C | doolph: what does your internet connection look like |
| 03:08.28 | DasTech | man sip/nat need to die |
| 03:08.39 | FuriousGeorge | Strom_C: ones and zeoros |
| 03:08.42 | DasTech | or nat needs to be replaced with something better |
| 03:08.43 | doolph | I have 786/256 |
| 03:09.00 | Strom_C | doolph: what codec are you using for calls? |
| 03:09.12 | doolph | remote office are using g729 |
| 03:09.16 | *** part/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 03:09.18 | doolph | local ulaw |
| 03:09.26 | Strom_C | and to the ITSP? |
| 03:09.35 | doolph | g729 |
| 03:09.40 | Strom_C | eeeewww |
| 03:10.08 | doolph | so i need to upgrade my internet eh |
| 03:10.09 | Strom_C | so yeah, sounds like your problem is that you're not prioritizing your voip traffic |
| 03:10.21 | doolph | yes I am not prioritizing nothing |
| 03:10.29 | Strom_C | you have two options: |
| 03:10.30 | orlock | double negative |
| 03:10.42 | Strom_C | 1. get a real router and do traffic shaping |
| 03:10.43 | orlock | if you are not prioritising nothing, you must be prioritising somehting |
| 03:11.05 | Strom_C | 2. get a second IP pipe for voice and segregate your voice and data networks |
| 03:11.19 | doolph | I have a real router |
| 03:11.26 | doolph | it is supposing to do the work |
| 03:11.27 | *** join/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 03:11.32 | Strom_C | and by "real" what do you mean |
| 03:11.37 | Strom_C | "netgear"? |
| 03:11.43 | doolph | a linux router |
| 03:11.53 | doolph | i setup it from scratch with debian |
| 03:11.55 | Strom_C | ah, iptables |
| 03:12.04 | orlock | iptables and ctb :) |
| 03:12.05 | Strom_C | so ok, traffic shaping is the first thing to try |
| 03:12.34 | DasTech | ok when putting 2 nics in 1 box how do you bind the the ips in asterisk so it uses the outside interface to connect to voip providers and the inside nic for the phones |
| 03:12.47 | DasTech | hmm k |
| 03:12.53 | FuriousGeorge | doolph: i find traffic shaping is very useful in preventing some guys ftp transfer from screwing up everyone's calls, for instance |
| 03:13.08 | FuriousGeorge | but it doesnt do squat to the traffic once it leaves your network |
| 03:13.14 | FuriousGeorge | or so i hear |
| 03:13.32 | doolph | I am trying to use cbq.init |
| 03:13.35 | file | can you hear the music? |
| 03:14.01 | doolph | but I think that I need something better |
| 03:14.17 | Strom_C | doolph: see option 2 above |
| 03:14.21 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:14.34 | Strom_C | and also, with option 2, you can stop using that icky-sounding g729 :) |
| 03:15.02 | doolph | what's better than g729? |
| 03:15.06 | Strom_C | ulaw |
| 03:15.12 | Strom_C | by leaps and bounds |
| 03:15.20 | doolph | but the provider is using g729 |
| 03:15.38 | Strom_C | so tell the provider to use ulaw or pound sand |
| 03:16.13 | doolph | my solution isn't get another internet |
| 03:16.20 | doolph | because I have users with softphones |
| 03:16.37 | Strom_C | ...how does that make any sense? |
| 03:16.46 | Strom_C | separate pipes for voice calls and data calls |
| 03:16.54 | Strom_C | you can route between two subnets, you know |
| 03:17.13 | doolph | umm |
| 03:17.46 | doolph | example I have a computer with ip 192.168.1.100 |
| 03:18.04 | doolph | i browse, download emails and I have xten |
| 03:18.12 | *** join/#asterisk adker (n=chatzill@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 03:18.24 | Strom_C | and you put your asterisk box on 192.168.0.100, and route accordingly |
| 03:19.27 | doolph | the problem is the people that is browsing and downloading emails |
| 03:19.45 | doolph | they eat all the bandwidth |
| 03:20.01 | Strom_C | sigh. you're not listening to me |
| 03:20.09 | Strom_C | you have two separate data pipes |
| 03:20.13 | Strom_C | one for data, one for voice |
| 03:20.28 | Strom_C | you route voice traffic out of one such pipe, and data traffic out of the other |
| 03:20.57 | doolph | how can I do that |
| 03:21.02 | Strom_C | you can do vlans and subnets and all sorts of neato IP routing stuff |
| 03:21.19 | doolph | a workstation is considered as data |
| 03:21.39 | Strom_C | ok, you should probably get a firm grasp of how IP routing works before administering an IP PBX |
| 03:22.03 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 03:22.11 | Strom_C | i suggest either "TCP/IP" from No Starch Press, which is far too large to carry around practically but sits magnificently on your coffee table |
| 03:22.14 | Strom_C | or |
| 03:22.17 | Strom_C | ~hafc |
| 03:22.18 | jbot | from memory, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 03:22.33 | justinu|laptop | TCP/IP illustrated by Stephens owns |
| 03:23.33 | anonymouz666 | stevens |
| 03:23.39 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.32) |
| 03:23.57 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.32) |
| 03:25.12 | doolph | umm |
| 03:27.58 | tengulre11 | how to connect remote H323 gateway, anybody can give me some tips? |
| 03:28.35 | doolph | get oh323 |
| 03:28.52 | tengulre11 | doolph, I installed h323. |
| 03:28.56 | Strom_C | I think it's time for fish tacos |
| 03:31.06 | doolph | what h323 do you have |
| 03:31.28 | tengulre11 | doolph, asterisk/channels/h323/ |
| 03:32.21 | doolph | well, I don't have that one |
| 03:32.39 | tengulre11 | anybody like hear yanni's 'one man's dream'? |
| 03:33.27 | Strom_C | tengulre11: i think that's possibly even more off topic than fish tacos |
| 03:33.48 | *** part/#asterisk DasTech (n=DasTech@d47-69-168-46.col.wideopenwest.com) |
| 03:35.13 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 03:35.14 | file | pizza is the goodness |
| 03:37.03 | file | remembering that I own headphones, also the goodness |
| 03:37.11 | *** join/#asterisk postel_ (n=jp@Wikimedia/Postel) |
| 03:38.10 | anonymouz666 | Jon Postel |
| 03:38.23 | Nugget | RIP. |
| 03:38.45 | Strom_C | catsex |
| 03:39.47 | tengulre11 | I want using asterisk to connect remote H.323 gateway, does it have to a static public ip in this asterisk?? |
| 03:40.33 | doolph | what do you mean |
| 03:42.36 | tengulre11 | doolph, asterisk in my office using 192.168.0.xxx, remote H.323 gateway using public ip. ( fuck english :P ) |
| 03:42.57 | Nugget | I think he's asking if h.323 can work through nat. |
| 03:43.09 | tengulre11 | Nugget: yes!!! |
| 03:43.27 | doolph | ah |
| 03:43.57 | doolph | um never had test that |
| 03:44.35 | tengulre11 | doolph, I see. is it not possible? |
| 03:45.13 | doolph | I really don't know I cannot assure nohitng |
| 03:46.31 | tengulre11 | doolph: thanks . |
| 03:47.03 | tengulre11 | It 's time to lunch |
| 03:47.06 | tengulre11 | bye! |
| 04:10.26 | knight_ | anyone run Asterisk on a WRAP here? |
| 04:19.20 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:28.04 | *** join/#asterisk hax (n=hax@httpcraft/hax) |
| 04:29.08 | hax | heya all |
| 04:30.15 | knight_ | hey |
| 04:30.22 | hax | this may be the wrong place to ask, but maybe someone friendly can point me in the right direction... i'm doing a startup, and i need to get a 800 line (preferably a vanity one)... i also guess i need a regular line for outbound (even though the ANI should show the 800 number)... anyone have any recommendations on where to actually get the lines from? |
| 04:30.35 | hax | i intend to run my own private asterisk on my debian server |
| 04:32.57 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 04:33.03 | knight_ | that should be no prob |
| 04:34.47 | hax | well, i don't really know who to use |
| 04:34.53 | hax | sellvoip.net looks like the cheapest, by a lot |
| 04:35.08 | hax | but they don't really explain to me how i can get 1-800-4-HAXHAX |
| 04:37.00 | hax | knight_: thoughts? |
| 04:40.31 | hax | x86: yo, help me :) |
| 04:42.34 | *** part/#asterisk adker (n=chatzill@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 04:43.25 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 04:47.28 | hax | oh c'mon now, i'm sure someone here can offer *some* clue |
| 04:50.34 | *** join/#asterisk tainted_ (n=point@adsl-69-230-201-74.dsl.irvnca.pacbell.net) |
| 04:51.22 | tainted_ | ~seen qwell |
| 04:51.38 | jbot | qwell <n=north@unaffiliated/qwell> was last seen on IRC in channel #asterisk, 12h 44m 1s ago, saying: 'umm, no'. |
| 04:51.38 | tainted_ | !seen qwell |
| 04:51.52 | MikeJ | classic |
| 04:54.50 | file | hello there hmmhesays |
| 04:56.43 | *** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-72b0197d399f302e) |
| 04:59.20 | Strom_C | wheeeeeeee |
| 04:59.51 | x86 | hax: what's up? |
| 05:00.01 | hax | x86: see above :) |
| 05:00.14 | hax | x86: well, you might be a little bias, running your own provider ;) |
| 05:00.31 | hax | x86: but i guess the big question is, where do i get a vanity 800 number from? |
| 05:00.41 | x86 | check /msg |
| 05:00.47 | x86 | ;-) |
| 05:01.02 | hax | kk |
| 05:07.53 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 05:08.18 | Jason99 | Is there a way that if I dial out on a SIP trunk and it timesout or fails to try on a second SIP trunk? |
| 05:09.55 | hads | Can anyone think of a reason I would be getting distorted audio when doing a pickup (*8) on an SIP phone (using trunk). |
| 05:09.56 | Strom_C | yes |
| 05:10.03 | Strom_C | maybe |
| 05:10.51 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:14.29 | hads | I do get a couple of messages saying "New channel is zombie" and "Old channel is zombie" |
| 05:14.56 | hads | I'm looking for ways to debug it if anyone has any suggestions. |
| 05:20.07 | linagee | Strom_C: jeopardy time. what is an asterisk number station? :-D |
| 05:20.16 | Strom_C | mein fraulein! |
| 05:20.30 | linagee | lol |
| 05:21.08 | linagee | Strom_C: is that what the fish on american dad always said? |
| 05:21.40 | Strom_C | beats me |
| 05:21.46 | Strom_C | i havent watched tv since 2002... |
| 05:21.49 | linagee | is it german? |
| 05:21.53 | linagee | Strom_C: heh |
| 05:22.55 | *** join/#asterisk dprevite (n=dprevite@c-67-162-110-89.hsd1.il.comcast.net) |
| 05:23.31 | linagee | lol |
| 05:23.36 | linagee | that was hilarious |
| 05:23.37 | Strom_C | hehehehe |
| 05:23.48 | Jenna | guys one quick not so smart question: I need to setup 100/150 phone pabx. Which project would get me going the quickest. thought of asteriskathome but it seemed not fit for 100/150 lines. currently Im inclined toward trixbox. Is that good idea. |
| 05:23.50 | Strom_C | goad you enjoyed it :D |
| 05:24.01 | Strom_C | Jenna: oh god no |
| 05:24.07 | linagee | Strom_C: did you guys ever consider festival? :p |
| 05:24.08 | linagee | lol |
| 05:24.12 | linagee | Strom_C: would have been easier. :) |
| 05:24.18 | Strom_C | linagee: but not as fun |
| 05:24.22 | linagee | lol |
| 05:24.26 | *** join/#asterisk dprevite (n=dprevite@c-67-162-110-89.hsd1.il.comcast.net) |
| 05:24.38 | Strom_C | Jenna: how much asterisk experience do you have and how much time are you willing to sink into the project before going lice |
| 05:24.40 | Strom_C | s/lice/live/ |
| 05:25.26 | linagee | wow. that could get annoying. |
| 05:25.35 | linagee | s/annoying/very annoying/ |
| 05:25.43 | Jenna | I did fidlle with asteriskathome once or twice. but yeah Im willing to go whole 9 yards |
| 05:26.00 | Strom_C | Jenna: how much time? |
| 05:26.14 | Jenna | 2-3 weeks |
| 05:26.28 | Strom_C | you only have three weeks before going live? |
| 05:26.34 | Jenna | I am good at rtfms |
| 05:26.43 | Strom_C | rtfm is not enough |
| 05:26.51 | Strom_C | how much traditional telephony experience do you have |
| 05:26.52 | linagee | Jenna: 2-3 weeks working 13 hour days, or just spending an hour each day? :-D |
| 05:27.13 | hads | Not enough either way :) |
| 05:27.39 | linagee | hads: 3 weeks of 13 hour days is not enough? what if on day zero you went to the bookstore and bought up every asterisk book? lol |
| 05:27.50 | Jenna | by traditional if u mean I have ever conversed over phone. yeah. I did. but my preference was to blow smokes |
| 05:28.20 | Jenna | okay make that 1.5 month |
| 05:28.25 | Strom_C | so basically you have no telephone system engineering experience |
| 05:28.30 | Jenna | is that any good |
| 05:28.55 | Jenna | would linux/unix sysadmin eperience count |
| 05:29.10 | linagee | Jenna: it's easy. just outsource the phone voice work, and all the asterisk work. :-D |
| 05:29.20 | Strom_C | do you understand proper numbering plan design, principles of telephony, codecs, TDM, and so on? |
| 05:29.40 | Jenna | I did take digital communications systems course back in the college. would that count ? |
| 05:29.58 | Strom_C | Jenna: i think at this point, the most reliable thing i can tell you is: |
| 05:30.01 | Strom_C | ~hafc |
| 05:30.02 | jbot | extra, extra, read all about it, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 05:30.15 | Jenna | and implemented few 3G datalink layer proctocols |
| 05:30.27 | Strom_C | otherwise you will likely shoot yourself in the foot at some point |
| 05:30.42 | Strom_C | every nub shoots him/herself in the foot at least twice on their first install |
| 05:30.51 | linagee | Strom_C: freaking consultant or phreaking consultant? lol |
| 05:30.58 | Strom_C | hjahahahaha |
| 05:31.13 | Strom_C | well if you're talking about me, then i'd be a phreaking consultant |
| 05:31.25 | JT | haq0r |
| 05:31.43 | Jenna | but working with asteriskathome was breeze |
| 05:31.59 | Strom_C | and asterisk@home/trixbox is completely NOT suited for any kind of production PBX install |
| 05:32.06 | Strom_C | especially not the size you're looking to install |
| 05:34.02 | Jenna | how much load would trixbox handle. I dont need complex/fancy features. just plain phone extensions would do |
| 05:34.15 | Strom_C | plus, if this is your first install, you are likely to make mistakes that will result in you having to completely rebuild the system later |
| 05:34.20 | Strom_C | its not a load issue |
| 05:34.20 | JT | it's less about load |
| 05:34.26 | JT | more about manageability |
| 05:34.30 | hads | A system should be configured by hand. |
| 05:35.19 | Jenna | I've plenty of machine for spare |
| 05:36.28 | Jenna | machines* |
| 05:36.31 | Strom_C | Jenna: well, give yourself a week wotk asterisk |
| 05:36.37 | Strom_C | s/wotk/with |
| 05:36.53 | Jenna | anyway what sort of work (level) could be achieved with trixbox |
| 05:37.06 | Strom_C | if, at the end of that week, you don't feel completely 100% confident using asterisk, hire a consultant |
| 05:37.15 | Strom_C | Jenna: DO NOT USE TRIXBOX |
| 05:37.18 | JT | Jenna: toy setups at home |
| 05:37.20 | Strom_C | not for a 150-station PBX |
| 05:37.25 | Strom_C | trixbox is a toy |
| 05:37.57 | Jenna | what would u recommend ? does asterisk have rpms or I have to deal with tgz |
| 05:38.06 | Strom_C | use the tarballs |
| 05:38.11 | Strom_C | or use subversion |
| 05:38.14 | Jenna | yeah. 1 weeks seems logical. sure |
| 05:39.16 | shodan | is it possible at all to use ADSI on a spa-2102 fxs ? |
| 05:39.38 | Jenna | btw would playing toys prepare me for ultimate thing better ? |
| 05:39.38 | hads | No |
| 05:39.56 | JT | oh come on, maybe a LITTLE |
| 05:40.03 | JT | but that bad habits it will teach you |
| 05:40.04 | Strom_C | playing with the real thig will prepare you for the real thing better |
| 05:40.07 | JT | will take time to unlearn |
| 05:41.20 | linagee | Strom_C: set up two xen guest OSes each with asterisk. have one as a backup that just does soemthing basic like redirects all lines to a voicemail. :-D |
| 05:41.22 | Jason99 | Anyone know if there is a way for asterisk to know that a Dial failed and it will attempt again on a second trunk ? |
| 05:41.33 | Strom_C | Jason99: yes |
| 05:41.33 | Jenna | hmm. would I be able to plug in the WebGUI in the real thing |
| 05:41.41 | linagee | Strom_C: keep the backup down unless you need to service the primary. :-D |
| 05:41.50 | Jason99 | Strom_C: Can you point me in the right direction so that I can read up on it ? |
| 05:41.51 | hads | A phone system doesn't need a web GUI. |
| 05:42.02 | Strom_C | Jenna: please, I can't tell you enough: no GUIs. No trixbox. Stop it. |
| 05:42.17 | Strom_C | Jason99: look at how the stdexten macro handles dial status |
| 05:42.23 | Jenna | anyway thanx Strom_C JT etc.. |
| 05:42.53 | Jenna | btw have u guys seen crash3m around lately |
| 05:43.26 | linagee | hads: the flash based asterisk gui is fun. click to dial. :-D reminds me of a child's toy. :-D |
| 05:43.41 | linagee | hads: "C is for Cookie Monster" |
| 05:44.07 | Jason99 | If the SIP trunk I'm sending to does not respond would that call CHANUNAVAIL ? |
| 05:44.40 | Strom_C | well, it will set DIALSTATUS to CHANUNAVAIL |
| 05:44.49 | Strom_C | you do your conditional branching from there |
| 05:45.17 | Jason99 | from what I can see I have to wait for the Dial to timeout (we'll say 20 seconds) before it moves to the second line, correct ? |
| 05:45.27 | Jenna | linagee, : I can understand the pro view-point when it comes to building thing. I hate webmin too for configuring system |
| 05:45.39 | Jenna | anyway love u guys for all the insight |
| 05:46.11 | Strom_C | Jason99: if the channel isn't available, it's not going to wait 20 seconds to time out |
| 05:46.22 | Jason99 | perfect.. that's what I need |
| 05:46.23 | Jason99 | thanks |
| 05:46.25 | Jason99 | :) |
| 05:46.31 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 05:47.34 | Jenna | night all |
| 05:47.39 | JT | bye |
| 05:48.46 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 05:50.19 | shodan | so... adsi on the spa-2101, yes/no ? |
| 05:50.38 | Strom_C | i'm going with "unlikely" |
| 05:50.59 | orlock | oh man |
| 05:51.09 | orlock | brockies dead :-( |
| 05:51.13 | JT | yep |
| 05:51.24 | Strom_C | whoo? |
| 05:51.29 | orlock | damn |
| 05:51.29 | JT | australian celebrity death week |
| 05:51.31 | JT | http://www.smh.com.au/news/national/peter-brock-killed-in-crash/2006/09/08/1157222310976.html |
| 05:51.42 | orlock | now i have no chance of getting the torana signed |
| 05:51.52 | JT | hrm |
| 05:51.53 | orlock | Strom_C: famous aussie racer |
| 05:51.57 | Strom_C | hmm, ok |
| 05:52.00 | X-Rob | NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO |
| 05:52.07 | X-Rob | NOT BROCKY! |
| 05:52.13 | orlock | X-Rob: yup :-( |
| 05:52.19 | Strom_C | i have two days left to put the dioxin in Paul Hogan's food , i guess |
| 05:52.40 | *** join/#asterisk daysmen3 (n=primus@host86-141-241-193.range86-141.btcentralplus.com) |
| 05:52.45 | orlock | hah |
| 05:52.51 | JT | You call THAT poison? Now THIS is poison? |
| 05:53.06 | X-Rob | Dude, plutonium is the most toxic substance |
| 05:53.19 | Strom_C | ok fine fine fine |
| 05:53.25 | Strom_C | i'll just spike his drink with ricin |
| 05:54.25 | orlock | http://www.ultimatecarpage.com/frame.php?file=car.php&carnum=2 |
| 05:54.45 | orlock | Only six examples of the Daytona Coupe were constructed, making it one of the most sought after vehicles in the world today. |
| 05:54.46 | hads | Wow, two Aussie celebs in a week. |
| 05:54.57 | benjk | ricin oil is probably more effective in the long term |
| 05:55.33 | JT | orlock: nasty |
| 05:56.10 | orlock | heh |
| 05:56.14 | orlock | guess its veen rarer now |
| 05:56.33 | JT | the wreck will be worth something |
| 05:56.35 | JT | read a lot |
| 05:56.47 | orlock | yeah |
| 05:57.17 | JT | mega rare car + racing champion died in it |
| 05:57.23 | benjk | these Aussies do all this unhealthy shit |
| 05:57.45 | JT | eh? |
| 05:57.50 | orlock | yeah, it distracts us from the binge drinking |
| 05:58.04 | JT | speak for yourself :) |
| 05:58.07 | hads | It's called living. |
| 05:58.19 | benjk | yeah, aussie rules football, binge drinking, corocodile teasing, .... :) |
| 05:58.49 | X-Rob | Trials bike riding |
| 05:58.49 | X-Rob | Ooh |
| 05:59.06 | JT | swimming with stingrays... |
| 05:59.12 | orlock | driving toranas... |
| 05:59.24 | JT | crashing toranas |
| 05:59.30 | X-Rob | How cool are these: http://osetbikes.com/ |
| 05:59.48 | Kumbang | how's rob bredl, i think this man more wild than steve irwin , right? |
| 05:59.49 | benjk | come on, he had a saaaaeeeve woorking distance from that animal at all times |
| 05:59.55 | orlock | hmm.. i wonder if my brother in laws walkinshaw is worth more now |
| 06:00.01 | X-Rob | orlock, no |
| 06:00.04 | orlock | probably not, cos that was made after the brock-holden split |
| 06:00.08 | X-Rob | walkinshaws were never worth much |
| 06:00.19 | X-Rob | you want the VL-pre-split with the polarisor |
| 06:00.21 | X-Rob | polariser? |
| 06:00.29 | X-Rob | those are the ones that are worth the megabucks |
| 06:00.33 | orlock | X-Rob: a bit, but nothing stupid like the bathurst specials |
| 06:00.39 | orlock | yeah |
| 06:00.57 | orlock | XU1's are 20-30k+ |
| 06:01.11 | orlock | A9X's are 50-80k |
| 06:04.50 | SplasPood | Has anyone used Cepstral with SVN trunk |
| 06:04.51 | SplasPood | ? |
| 06:07.39 | SplasPood | specifically via an application rather than an AGI |
| 06:16.33 | benjk | yes, I did, but not with Digium's code base |
| 06:17.04 | micky | hi, i'm getting -> Sep 7 18:16:09 WARNING[4834]: app_ices.c:176 ices_exec: Write failed to pipe: Broken pipe |
| 06:17.27 | benjk | SplasPood, you can use app_cepstral |
| 06:17.53 | benjk | if it doesnt match Digium's latest code base, it shouldn't be difficult to adapt |
| 06:21.40 | SplasPood | benjk: Yea its not compiling.. whats the quickest way to add an app to the new build setup? |
| 06:23.32 | benjk | if its not compiling, I'd say its probably due to some changes in the asterisk code base such as a different parameter list for some function, or changing the name of a function or something like that |
| 06:24.51 | benjk | like ast_foobar(int foo, int bar) became ast_foobar(ast_baz *baz, int foo, int bar) |
| 06:25.05 | benjk | they do shit like that all the time |
| 06:25.47 | SplasPood | well it seems to be an include issue at this point |
| 06:26.00 | SplasPood | /usr/include/asterisk/strings.h:280: error: conflicting types for 'strtoq'/usr/include/stdlib.h:197: error: previous declaration of 'strtoq' was here |
| 06:26.31 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:27.45 | benjk | yeah, that's precisely the sort of thing I was talking about |
| 06:27.49 | SplasPood | Yea, this would be easier for someone who A) knew C and B) knew asterisk's code base :P |
| 06:27.58 | benjk | they made their own strtoq function apparently |
| 06:28.31 | benjk | the thing is that changing things in the Digium code base is a waste of time |
| 06:28.48 | benjk | they don't want free software contributions |
| 06:29.32 | benjk | they only want code monkeys who work for them with the same kind of rights an employee has, but they don't want to pay for the work |
| 06:30.11 | benjk | and even if you have something you are willing to give them under their terms, the chance is high they reject it anyway |
| 06:30.30 | benjk | so I have little incentive to do anything with their code base |
| 06:30.47 | benjk | you can use OpenPBX though |
| 06:31.30 | benjk | app_cepstral works just fine there |
| 06:31.43 | SplasPood | yea it works just fine with 1.2 too :{ |
| 06:31.51 | benjk | then why upgrade ? |
| 06:31.56 | benjk | "upgrade" |
| 06:32.07 | benjk | has to be put in quotes |
| 06:32.49 | Strom_C | or maybe |
| 06:32.51 | benjk | if only they had a proper API with abstraction layers |
| 06:32.53 | Strom_C | just MAYBE |
| 06:32.57 | Strom_C | there's a BUG in the code!!! |
| 06:32.59 | Strom_C | (gasp) |
| 06:33.07 | benjk | then you could make changes to the implementation without breaking stuff |
| 06:35.10 | benjk | and I didn't need the plug "SplasPood: Yea, this would be easier for someone who A) knew C and knew asterisk's code base" |
| 06:35.26 | SplasPood | eh? |
| 06:35.27 | benjk | without that plug I wouldn't have mentioned it |
| 06:35.46 | SplasPood | Cause I don't know how to modify this myself? |
| 06:37.40 | benjk | maybe a misunderstanding, never mind |
| 06:38.39 | benjk | but anyway, if you want somebody to change things related to the asterisk code base for you, the better time to ask during US daytime |
| 06:38.56 | Strom_C | yeah, because at night, this place is troll city |
| 06:39.17 | benjk | depends on your viewpoint really |
| 06:40.02 | benjk | in any event, fanboys are worse than whatever you call trolls |
| 06:41.07 | benjk | in the US there are more folks who are happy to work under Digium's terms, that's fair enough |
| 06:41.32 | benjk | what is not fair enough is to expect that everybody else has to be willing to agree to those terms as well |
| 06:43.26 | hads | Yay, benjk again. |
| 06:47.00 | shodan | ~cpe |
| 06:47.02 | jbot | it has been said that cpe is Customer Premises Equipment. Telephone devices such as handsets and PBXs located at the customer.s site that interface with the public network. It includes equipment such as modems, terminals and routers supplied by the telephone company, installed at customer sites and connected to the telephone network. |
| 07:02.29 | *** join/#asterisk stagiaire (n=stagiair@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 07:04.03 | _Vile | blah |
| 07:06.53 | micky | any idea why i get error reading /dev/zap/pseudo ? and /dev/zap/pseudo says no device... but i've installed zaptel modules and are running in the kernel |
| 07:08.22 | GingerDog | micky: does the card show up through lspci (assuming it's a pci card) |
| 07:09.06 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 07:10.17 | micky | GingerDog it`s usb, it works with one kernel i have installed but i've recompiled because i just upgraded the memmory to 2GB and i needed highmem support active... and now it loads the module and sais error reading /dev/zap/pseudo |
| 07:11.11 | micky | do i need to configure something else ? |
| 07:12.01 | GingerDog | *shrug*; does 'dmesg' say anything useful? |
| 07:12.24 | micky | not really |
| 07:12.52 | GingerDog | does 'lsusb' show the device as being present? |
| 07:16.03 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:18.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:18.51 | *** join/#asterisk digtalp (n=steve@dns1.nyc.dns-roots.net) |
| 07:18.55 | digtalp | hello all |
| 07:20.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:20.23 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 07:20.57 | digitalp | can anyone tell me what would cause " exited non-zero on" |
| 07:21.38 | hads | digitalp: That's normal |
| 07:22.59 | digitalp | hads; i have this and after it plays reacord name and press # it hangs up the call |
| 07:23.00 | digitalp | exten => s,1,Wait(0.5) |
| 07:23.00 | digitalp | exten => s,2,Playback(vm-rec-name) |
| 07:23.00 | digitalp | exten => s,3,Setvar(SCREEN_FILE=/tmp/${CALLERIDNUM}) |
| 07:23.00 | digitalp | exten => s,4,Record(${SCREEN_FILE}.gsm) |
| 07:23.00 | digitalp | exten => s,5,Playback(screen-pls-wait) |
| 07:23.02 | digitalp | exten => s,6,Dial(SIP/113&bttx-sip/16095103665,30,mtM(screen^${SCREEN_FILE})) ;here you put dial string to your phone |
| 07:23.03 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:23.11 | hads | Don't paste in the channel |
| 07:23.24 | digitalp | sorry |
| 07:24.00 | hads | set verbose 5 and see if something interesting gets printed to the console |
| 07:25.35 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 07:27.25 | digitalp | nothing but it following the dialplan |
| 07:27.45 | digitalp | strange thing it goes right to h, after playing say your name. |
| 07:28.13 | micky | GingerDog so i'm getting the error: chan_zap.c:Unable to open '/dev/zap/pseudo': no such devide or address ; chan_zap.c unable to dup channel: no such device |
| 07:28.24 | micky | GingerDog could this be related to udev ? |
| 07:28.47 | hads | digitalp: Try adding debug to your console log and set debug 5 aswell and see if you can find something. |
| 07:30.28 | digitalp | hads: still nothing , i have debug at 5 and verbose at 5 starting asterisk with -vvvvgggr |
| 07:31.20 | micky | same here... |
| 07:32.22 | hads | digitalp: Did you add debug to the console line in logger.conf |
| 07:33.30 | hads | And what's with the ggg? |
| 07:34.11 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:34.30 | digitalp | yes its in logger.conf to send debug to console |
| 07:38.52 | GingerDog | micky: *Shrug* |
| 07:39.25 | GingerDog | micky: I'd try: 1) rebooting, 2) checking dmesg for errors on module insertion, 3) check lsmod output, 4) lsusb, 5) google |
| 07:40.46 | *** join/#asterisk dacleric (n=dacleric@p548232CD.dip0.t-ipconnect.de) |
| 07:44.29 | hads | digitalp: Seems odd that you are not getting any messages at all. What are you trying to do with that dialplan anyway? |
| 07:45.40 | hads | Looks like you are trying to do something similar to screening mode. |
| 07:46.26 | hads | Also, you should be using Set instead of SetVar and CALLERID(num) instead of CALLERIDNUM |
| 07:48.54 | *** part/#asterisk GingerDog (n=GingerDo@oak.palepurple.co.uk) |
| 07:54.13 | benjk | micky, yes, most likely cause is missing udev rules |
| 08:03.26 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:04.23 | CtRiX | janeway:~# ps -Af | grep asterisk | wc -l |
| 08:04.23 | CtRiX | 139 |
| 08:04.33 | CtRiX | this is the way it deadlocks............ |
| 08:04.36 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:06.39 | CtRiX | chan_sip really hangs on small load |
| 08:07.14 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:07.39 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:11.59 | Toadyus | anyone use 1-800#'s? |
| 08:12.49 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:14.05 | *** join/#asterisk ppyy (n=lala@222.188.139.146) |
| 08:15.38 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:17.08 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:23.46 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:26.25 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 08:27.27 | eject_ck | Hi all! Does anybody know how much traffic is used for intensive speach per hour ? |
| 08:29.10 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:32.52 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 08:49.40 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 08:49.49 | eject_ck | hey |
| 08:49.54 | eject_ck | please help |
| 08:58.29 | Jeffjohnson | howdy |
| 08:59.14 | Jeffjohnson | i need a pattern to match all numbers from 89100 0-298... Which is the correct pattern? _89100[0-298] don't work, _8199[0-2][0-9][0-8] also seem to be wrong |
| 09:00.25 | Jeffjohnson | _89100[0-2]X[0-8] don' match numbers with 89100X :O |
| 09:02.12 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:02.16 | Jeffjohnson | it must match variable length numbers... |
| 09:05.20 | eject_ck | I have servers with Asterisk under OpenBSD. Now I want connect to PSTN ... as I understand there is not possible do it with TDM400P. What another ways ? |
| 09:06.22 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 09:07.34 | *** join/#asterisk evilmnky (n=evilmnky@69-163-147-74.atlsfl.adelphia.net) |
| 09:12.49 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 09:13.03 | SeicherlBoB | can someone help me reading SIP-debug logs? |
| 09:13.19 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 09:13.29 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:13.43 | SeicherlBoB | i got some authentification issues with my external SIP accounts |
| 09:18.55 | stagiaire | SeicherlBOB, Can you describe your problem? |
| 09:19.46 | SeicherlBoB | i have 3 sip-accounts at the same provider |
| 09:20.15 | SeicherlBoB | all i do is catch every extension ( exten => _.,1,Answer() ) |
| 09:20.26 | SeicherlBoB | and then echo the dialed extension with a NoOp |
| 09:20.43 | SeicherlBoB | one account works perfectly fine, two others dont |
| 09:22.09 | benjk | jeffjohnson, _89100[0-1]XX, _891002[0-8]X, and _8910029[0-8] |
| 09:22.33 | Jeffjohnson | benjk: mmh ok, so i need 3 extensions |
| 09:22.43 | benjk | three matches yes |
| 09:23.02 | SeicherlBoB | now i got one log of the successful call and one of the broken one (broken means that * wont answer the call) |
| 09:23.54 | *** join/#asterisk Flusher (i=flusher@filer.euroserv.com) |
| 09:24.01 | benjk | eject_ck, there was some project trying to port zaptel drivers to BSD, not sure what the status is, but maybe you want to look for info on that |
| 09:24.03 | Flusher | hi |
| 09:24.21 | eject_ck | I have servers with Asterisk under OpenBSD. Now I want connect to PSTN ... as I understand there is not possible do it with TDM400P. What another ways ? I hear abt Sipura 3000. What difference between it features ? |
| 09:24.40 | benjk | I just told you |
| 09:24.52 | Flusher | Are FreePBX/Trixbox projects encouraged by Asterisk ? |
| 09:24.57 | benjk | and yes, Sipura-3000 can be used too |
| 09:25.09 | eject_ck | benjk, features ? |
| 09:25.37 | benjk | but if you need 4 ports, then perhaps a 4 port SIP FXO/FXS gateway is more suitable, because the Sipura3000 only has 1 FXO and 1 FXS port |
| 09:25.42 | eject_ck | functionallity is same ? (with 1 FXS/FXO) |
| 09:26.08 | benjk | in my experience the Sipura works better than analog zaptel |
| 09:26.13 | benjk | YMMV |
| 09:26.19 | eject_ck | As I understand sipura is as SIP station |
| 09:26.29 | benjk | its a SIP gateway |
| 09:26.36 | eject_ck | and I will use only SIP |
| 09:26.38 | benjk | with 1 FXO and 1 FXS port |
| 09:26.41 | eject_ck | ok |
| 09:26.49 | benjk | you use SIP to connect to the Sipura yes |
| 09:27.07 | benjk | POTS ----> sipura ----SIP----> asterisk |
| 09:28.56 | benjk | the only thing is that the Sipura is a beast to configure, it's got half a bazillion settings |
| 09:29.11 | benjk | most of which you don't need, but still |
| 09:29.12 | eject_ck | I have two offices and want to connect it with asterisk with main features - then i need buy 2 sipura and connect it throught FXO to my mini ATS |
| 09:29.37 | benjk | you mean you have one POTS line in each office? |
| 09:30.56 | benjk | POTS ---> [spa3k] --- [asterisk1] ===IAX trunk=== [asterisk2] --- [spa3k] <--- POTS |
| 09:31.26 | benjk | where asterisk1 is in office 1 and asterisk2 is in office 2 |
| 09:32.04 | eject_ck | yes |
| 09:32.17 | benjk | well, use that layout then |
| 09:32.24 | eject_ck | all right |
| 09:32.39 | benjk | which country is this for? |
| 09:32.40 | eject_ck | have u experience with sipura ? |
| 09:32.44 | eject_ck | Ukraine |
| 09:32.48 | eject_ck | and u ? |
| 09:32.52 | benjk | no ISDN in Ukraine? |
| 09:33.02 | eject_ck | yes in one office |
| 09:33.23 | benjk | you may want to use BRI PCI cards and BRIstuff |
| 09:33.28 | eject_ck | another is in other city |
| 09:33.34 | benjk | its WAAAAAAAAYYYYYYY better than POTS |
| 09:33.41 | eject_ck | I now |
| 09:33.50 | benjk | the cheapest BRI cards are less than $50 |
| 09:33.52 | eject_ck | not possible |
| 09:34.09 | benjk | passive single port BRI HFC cards |
| 09:34.17 | benjk | they work nicely with BRIstuff |
| 09:37.23 | Jeffjohnson | what pattern can i use to match 0-1 Characters? |
| 09:38.14 | *** join/#asterisk Chaz6 (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 09:40.11 | Jeffjohnson | benjk: it dont work, when i dial a number with a 3 character extension only 2 characters are recognized... 1 or 2 character extensions work |
| 09:41.06 | benjk | pastebin your exentions (the ones for the 89100... numbers you want to match) |
| 09:51.28 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 09:52.12 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 10:00.16 | *** join/#asterisk frawd (n=root@62-14-54-163.inversas.jazztel.es) |
| 10:06.48 | frawd | hi everyone, i have a small problem when i call via an analog zap interface, it seems that when i'm making a call, asterisk is bridging the call a bit late and i never hear the first word of the one i call ("allo" or "hi"). Anyone has experienced this problem? |
| 10:09.02 | frawd | I was thinking it could be related to echo cancelation (i have settings echocancel=64 and echotraining=600). Any idea about the issue? |
| 10:09.52 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f43870d3-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 10:10.20 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
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| 10:38.32 | S^P | how to implement caller Id blocking feature ? |
| 10:39.02 | benjk | S^P, use the blacklist |
| 10:39.16 | S^P | blacklist? |
| 10:39.19 | benjk | its got a lookup application |
| 10:39.31 | S^P | can you please give me some pointer on it? |
| 10:39.32 | benjk | if the caller id is in the list, it will block |
| 10:39.48 | benjk | search for Asterisk + blacklist on voip-info.org |
| 10:39.53 | S^P | ok |
| 10:42.42 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:43.06 | jmls | anyone been able to compile app_rxfax / app_txfax on the latest svn trunk ? |
| 10:45.58 | *** join/#asterisk jeffik (n=Jeff@CABLE-72-53-84-108.cia.com) |
| 10:47.18 | Sonderblade | when i make an outbound call, i sometimes get the messages "<dev> is proceeding passing it to <odev>" and "<dev> is making progress passing it to <odev>" and other times i don't get those messages. anyone know what they mean? |
| 10:50.01 | jeffik | All, anybody have experience with H.263 video phones? |
| 10:52.57 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:53.07 | benjk | hi matteo |
| 10:53.13 | benjk | long time no see |
| 10:53.54 | Sonderblade | jeffik: no experience, but i saw one demonstrated at an exhibition.. :) it looked amazing |
| 11:01.56 | jeffik | sounderblade: which one did you see? |
| 11:05.49 | Sonderblade | jeffik: i forgot, but it was some kind of wireless phone with a big lcd in the holder |
| 11:06.20 | *** join/#asterisk adorah (n=Administ@84.94.208.214.cable.012.net.il) |
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| 11:18.25 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.11.220.revip2.asianet.co.th) |
| 11:20.01 | HaMYaI | a question regarding the zaptel.conf |
| 11:20.07 | HaMYaI | do we put fxoks=1-4 and then span=1,1,0,ccs,hdb3,crc4 |
| 11:20.12 | HaMYaI | or fxoks=1-4 and then span=2,1,0,ccs,hdb3,crc4 |
| 11:20.38 | HaMYaI | for 1 tdm amd 1 pri |
| 11:22.03 | HaMYaI | s/tdm/tdm400 |
| 11:23.04 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:24.32 | *** join/#asterisk vlt (n=dm@p54B340E3.dip0.t-ipconnect.de) |
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| 11:36.05 | adamowitz | has anyone used pingplotter ( http://www.pingplotter.com/ ) to test a connection with a service provider? Is there a similar program that runs on linux or mac osx? |
| 11:36.49 | vlt | Hello. After having had more and more problems with the ubuntu packaged asterisk I appreciated compiling a fresh one myself (what I never did before). I downloaded the 1.2.11 tar from digium, extracted it to /usr/src/asterisk, did an `apt-get build-dep asterisk` and then tried `make` in the src dir. Compilation stops after 1m40s and 257 lines of output saying: "chan_zap.c:9025: error: 'pri_event_setup_ack' has no member named 'call'". The kernel |
| 11:36.49 | vlt | <PROTECTED> |
| 11:40.01 | adamowitz | vit: to compile asterisk from source, read and follow the instructions here: http://www.asterisk.org/download |
| 11:40.43 | adamowitz | ignore your packaging system altogether. |
| 11:40.52 | tzafrir_laptop | vlt, hmmm... incompatible libpri? |
| 11:41.04 | tzafrir_laptop | what version of libpri-dev do you have? |
| 11:41.54 | tzafrir_laptop | unlike zapel, libpri is rather close in nature to asterisk. consider building it from source if you can't easily get a more recent one packaged. |
| 11:43.33 | tzafrir_laptop | vlt, OTOH, don't event think about building openh323 yourelf. Get a decent deb, and than the Debian for saving you that mess ;-) |
| 11:44.30 | tzafrir_laptop | HaMYaI, something seems wrong. Please pastebin cat /proc/zaptel/* |
| 11:44.54 | *** join/#asterisk frenzy (n=frenzy@196.46.104.77) |
| 11:45.15 | frenzy | Hi |
| 11:45.34 | tzafrir_laptop | HaMYaI, also try the script xpp/genaptelconf |
| 11:46.05 | frenzy | all of a suden after reboot cant start asterisk.. keep getting ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 11:46.25 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:47.08 | HaMYaI | tzafrir_laptop, ok |
| 11:50.56 | *** join/#asterisk isede (n=qed@cpe-74-65-225-244.nyc.res.rr.com) |
| 11:52.29 | HaMYaI | tzafrir_laptop, http://pastebin.ca/164205 |
| 11:53.34 | HaMYaI | tzafrir_laptop, "wctdm" is loaded before "wcte11xp" |
| 11:53.41 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 11:54.13 | HaMYaI | tzafrir_laptop, so I assigned fxoks=1-4 for the tdm400 with quad fxs |
| 11:54.58 | HaMYaI | tzafrir_laptop, so wondering if it has to be span=2... |
| 11:56.04 | tzafrir_laptop | yes, it hould be span 2 |
| 11:56.36 | HaMYaI | tzafrir_laptop, do I have to reboot after changing it? |
| 11:56.49 | tzafrir_laptop | no. Just re-run ztcfg |
| 11:57.09 | tzafrir_laptop | Do you need to re-order modules? just rmnmod both and modprobe them in the right order |
| 11:57.37 | tzafrir_laptop | However you need to make sure that they will lod in that order after boot |
| 11:58.59 | HaMYaI | I have tried to change the order but how do I know if they are following what's placed in /etc/modprobe.d/zaptel? |
| 11:59.35 | *** join/#asterisk jeffik (n=Jeff@CABLE-72-53-84-108.cia.com) |
| 11:59.50 | HaMYaI | recompiling zaptel always changes my order |
| 12:00.37 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 12:02.38 | vlt | tzafrir_laptop: libpri-dev 1.2.2-3 is installed. Where can I get a newer one? (And do I need openh323?) |
| 12:03.33 | jeffik | All: again, anybody have experience with H.263 video phones/ |
| 12:04.26 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 12:06.37 | tzafrir_laptop | vlt: recent asterisk needs libpri 1.2.3 |
| 12:08.14 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 12:08.22 | tzafrir_laptop | there are libpri 1.2.3 debs, I believe . Edgy , as well as Debian Etch and Sid have 1.2.3 |
| 12:08.36 | tzafrir_laptop | You can probably rebuild those debs |
| 12:08.58 | tzafrir_laptop | you need openh323 if you want chan_h323 |
| 12:09.26 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 12:10.27 | jamincollins | anyone know of a way to find out why * is triggering a disconnect? I am using it for media translation (TDM to SIP) and in at least some cases it appears that * is generating a disconnect for both the TDM and SIP portion of the call passing through it |
| 12:12.59 | *** join/#asterisk af_ (n=af@ip-164-15.sn2.eutelia.it) |
| 12:18.00 | *** join/#asterisk Chaz6 (n=chaz@2001:630:1c0:2001:20c:29ff:fe4d:1bd5) |
| 12:19.29 | *** join/#asterisk jeffik (n=Jeff@CABLE-72-53-84-108.cia.com) |
| 12:19.42 | HaMYaI | tzafrir_laptop, I loaded tdm first and then wcte11xp but in zaptel.conf I put the config of wcte11xp first |
| 12:19.47 | HaMYaI | and it works |
| 12:19.51 | HaMYaI | any comment? |
| 12:21.22 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 12:21.22 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 12:23.08 | Assid | man.. waiting for december is gonna be difficult |
| 12:25.07 | *** join/#asterisk maxbel (n=maxi@r190-0-143-250.dialup.adsl.anteldata.net.uy) |
| 12:25.47 | *** join/#asterisk frawd (n=root@62-14-54-163.inversas.jazztel.es) |
| 12:29.04 | RoyK | http://karlsbakk.net/fun/IrishWeatherMachine.jpg |
| 12:29.28 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:29.43 | tzafrir_laptop | HaMYaI, the order of entries in zaptel.conf doesn't matter, as long as you get the number right |
| 12:30.21 | Qwell | what is this silly floating red line in xchat for? |
| 12:30.40 | Qwell | ahh, nm |
| 12:30.51 | Qwell | "marker line" |
| 12:36.54 | pablus | morning |
| 12:46.16 | vlt | tzafrir_laptop: Mmh, I can't find a src for a newer libpri. Do I need this when I only want to use misdn/BRI and SIP channels? |
| 12:50.12 | *** join/#asterisk speekac (n=alwin@60.51.217.61) |
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| 12:53.42 | adamowitz | vlt: o compile asterisk from source, read and follow the instructions here: http://www.asterisk.org/download |
| 12:53.43 | adamowitz | ignore your packaging system altogether. |
| 12:55.42 | *** join/#asterisk ppyy (n=lala@222.185.16.160) |
| 13:04.44 | vlt | adamowitz: Thank you. I downloaded libpri from digium and untared it to /usr/src/libpri. In the README file there's no description how to install it. Just `make` and `make install`? |
| 13:05.29 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:06.27 | jamincollins | anyone know a means of locating what within * triggered a disconnect? |
| 13:07.25 | adamowitz | vlt: have you visited the link I just gave you? You must not have because your question is answered there. |
| 13:08.06 | vlt | adamowitz: Yes, that's where I got the src tar from. (Am I damn blind???) |
| 13:08.31 | adamowitz | has anyone used pingplotter ( http://www.pingplotter.com/ ) to test a connection with a service provider? Is there a similar program that runs on linux or mac osx? |
| 13:09.08 | jamincollins | adamowitz, never used it, but the closest I've seen to what it seems to be is smokeping |
| 13:09.10 | vlt | adamowitz: Aah, ok, I am .... It's hidden in the SVN section :) |
| 13:11.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.32 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:14.11 | adamowitz | thanks jamincollins. |
| 13:14.39 | jamincollins | adamowitz: np, I've been using smokeping for a while never knew of pingplotter |
| 13:15.27 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 13:16.54 | jamincollins | now, if only I could locate the source of this seemingly random disconnect issue |
| 13:18.01 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:21.06 | jamincollins | anyone seen this before: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request |
| 13:21.40 | jamincollins | should I be concerned that the peerstate was "Connect Request" rather than "Active"? |
| 13:24.46 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:25.16 | puzzled | hi |
| 13:30.01 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
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| 13:40.47 | *** join/#asterisk rados___ (n=rados@c-68-62-71-76.hsd1.mi.comcast.net) |
| 13:41.27 | rados___ | I'm trying to interconnect asterisk with avaya definity |
| 13:41.48 | rados___ | I actually am able to make calls but have trouble connecting the audio |
| 13:42.03 | rados___ | it's using h.323 |
| 13:42.30 | rados___ | i know it seems to be a codec issue but I can't figure it out |
| 13:42.43 | rados___ | has anyone here played with this set up? |
| 13:45.56 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:48.31 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
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| 14:15.22 | *** join/#asterisk aixa (n=Miranda@www.crediweb.lv) |
| 14:15.25 | aixa | hi everyone |
| 14:15.50 | aixa | any idea where I could locate the functions which siptapi or asttapi exposes to the operating system? |
| 14:16.19 | aixa | tapi 2.X looks HUGE and I highly doubt that every feature there is supported |
| 14:16.40 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:17.09 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 14:18.05 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 14:18.40 | }btorch{ | what's up with this "warnning, flexible rate not heavily test" ? |
| 14:18.57 | Qwell | You fixed a typo and introduced another one... |
| 14:19.06 | }btorch{ | it only seems to happen during moh on meetme calls |
| 14:22.21 | *** join/#asterisk xming (n=xming@gentoo/user/xming) |
| 14:28.05 | alawguy | }btorch{: it's a message from mpg123 (when playing VBR MP3) |
| 14:32.34 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:33.48 | ghenry | wildfire and spark client with asterisk plugin rule |
| 14:35.01 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
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| 14:49.00 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 14:51.20 | paolob | Hi guys! My asterisk system works with a sipura spa-3000 and various linksys pap2. When I make a external call from an extension, I see a lot of net traffic on the server. Isn't there a way to get asterisk communicate directly the pap2 of the calling extension with the spa3000 without having net traffic on the asterisk server? thank you |
| 14:51.52 | MrChimpy | guys, i'm keeping my samples in GSM format but i'd like to avoid having to transcode. i'm getting calls in as normal audio on E1 lines. what format should I be using? I'm guessing 8KHz mono wav? |
| 14:52.21 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 14:54.00 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:54.15 | GerbilWrk | paolob, you will need to turn on reinvite in the sip.conf files |
| 14:54.47 | paolob | GerbilWrk, shouldn't I do nothing in the sipura and linksys configuration? |
| 14:54.50 | *** join/#asterisk chode (n=chode@p54B028ED.dip0.t-ipconnect.de) |
| 14:55.02 | GerbilWrk | you shouldn't need to modify them, just the sip.conf file |
| 14:58.09 | paolob | GerbilWrk, is it the canreinvite=yes command? |
| 14:58.18 | GerbilWrk | i believe so |
| 14:58.27 | GerbilWrk | if they are both behind a nat, it won't work though |
| 14:59.01 | paolob | GerbilWrk, the spa3000 and the pap2 are behind the same nat |
| 14:59.12 | GerbilWrk | then it might work |
| 14:59.18 | GerbilWrk | i know behind seperate nats it doesn't |
| 15:00.17 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 15:00.19 | Bert- | hello there |
| 15:00.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 15:01.23 | Bert- | a little pb with modules : in modules .conf, I have noload => cdr_addon_mysql.so, but module is loaded at startup |
| 15:01.31 | Bert- | I have load => cdr_csv.so |
| 15:01.31 | Bert- | load => cdr_manager.so |
| 15:01.31 | Bert- | load => cdr_custom.so |
| 15:01.41 | Bert- | but these are not automaticaly loaded |
| 15:01.50 | Bert- | then I don't understand |
| 15:02.08 | Bert- | I can load or unload modules in CLI it is okay |
| 15:02.27 | Bert- | but I would like to understand why they are not loaded |
| 15:03.38 | Bert- | I restart asterisk : [ Booting...Sep 8 17:02:55 NOTICE[7869]: cdr.c:1191 do_reload: CDR simple logging enabled. |
| 15:03.56 | Bert- | but no no cdr_csv.so loaded in show modules |
| 15:04.14 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 15:05.47 | Idle | I currently have a context setup and it works fine, but I want to have any call from one of my SIP lines to perform an action (notify to be exact) BEFORE it goes to any of those... I tried adding a new context, and changing that to the default, with it having 'exten => i,1,blah... exten => 1,2,Goto(oldcontext,${EXTEN},1)' |
| 15:05.52 | Idle | but that doesn't work |
| 15:07.22 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:07.53 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 15:08.00 | trevarthan | yo yo... what's up? |
| 15:08.06 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 15:08.32 | trevarthan | Hey, I'm curious about VoiceXML. Is there anything production ready that integrates voicexml and asterisk? |
| 15:09.12 | paolob | GerbilWrk, I set canreinvite=yes in sip.conf, but the traffic still passes throug the server. I saw that the docs say that if in the dial command there is a t,T,w,W, ecc. option the reinvite isn't issued. What's the reason for that? |
| 15:09.17 | Qb3rt | when i call one particular number asterisk tell me PROGRESS with cause code 28 received |
| 15:10.24 | GerbilWrk | paolob, those allow the call to be transfered by the user, or recorded by the user |
| 15:10.54 | paolob | GerbilWrk, but why doesn't asterisk obey the reinvite when they are set? |
| 15:11.18 | GerbilWrk | you've updated and reloaded the sip.conf files right? |
| 15:12.01 | GerbilWrk | for both sip clients? |
| 15:12.07 | trevarthan | does publicVoiceXML integrate with asterisk yet? |
| 15:12.19 | file | Qb3rt: Cause No. 28 - Incorrect number (invalid number format, address incomplete)/Special intercept announcement |
| 15:13.37 | Qb3rt | file: the number i dial is 5145743176 and if i dial 5145743174 i working good! |
| 15:13.37 | paolob | GerbilWrk, yes, both have reinvite set, I checked it with sip show peer <peer name> |
| 15:14.00 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 15:14.03 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:14.23 | GerbilWrk | well, check the devices them, see if they have a setting for reinviting |
| 15:14.30 | GerbilWrk | also, are both devices using the same codec? |
| 15:15.23 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:15.30 | bkw_ | both devices don't have to speak the same codec to do a reinvite.. they just have to have SOMETHING in common ... but then again asterisk does this wrong |
| 15:16.22 | benjk | bkw, you're trolling and you got it wrong, asterisk does it right and the devices do it wrong |
| 15:16.36 | benjk | all devices |
| 15:16.36 | GerbilWrk | voip-info says they have to speak the same codec |
| 15:17.10 | benjk | in order to establish a voice call between them they have to have at least one codec in common, yes |
| 15:17.54 | benjk | but in order for them to talk to each other to negotiate a codec it matters not what the initial / first preferred setting is |
| 15:18.25 | devel | somebody please tell me what this means: -- Channel 0/1, span 1 got hangup |
| 15:18.42 | benjk | it means what it says |
| 15:18.47 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 15:18.56 | devel | no, i want to hear it in english from somebody. |
| 15:19.01 | benjk | channel 0 received a hangup signal |
| 15:19.16 | devel | so the PRI (i.e the CO) sent a hangup for that channel |
| 15:19.23 | benjk | channel 0 of span 1 that is |
| 15:19.37 | stephane_ | re |
| 15:19.41 | benjk | the remote end hung up on you yes |
| 15:19.52 | devel | that's what i wanted to know. |
| 15:19.57 | devel | now, will somebody else say that. |
| 15:19.59 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 15:20.28 | benjk | you can always test the channel variable HANGUPCAUSE in your dialplan to get it more detailed |
| 15:20.43 | benjk | to find out what reason there was |
| 15:20.57 | benjk | NoOp(${HANGUPCAUSE}) |
| 15:21.02 | rados___ | <PROTECTED> |
| 15:21.03 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:21.15 | rados___ | asterisk that is and Avaya |
| 15:21.20 | benjk | and the hangup cause codes are listed in include/asterisk/causes.h |
| 15:21.31 | Idle | the s extension is supposed to catch ALL before they dial the actual extension, right? |
| 15:22.40 | tzanger | no |
| 15:22.43 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:22.50 | tzanger | 's' is the default extension hit when no extension is given. |
| 15:22.57 | *** join/#asterisk mne (n=mne@chello080108001212.35.11.tuwien.teleweb.at) |
| 15:23.01 | Idle | this is driving me nuts... I have no idea how to add something before a user dials... |
| 15:23.16 | Idle | this extensions config is so disgusting... everything includes everything else |
| 15:23.18 | tzanger | i.e. Dial(IAX2/user@peer/1234) will hit extension '1234' or a match for it, but will NOT fall back to 's', since an extension WAS given |
| 15:23.26 | Idle | ok |
| 15:23.33 | tzanger | Dial(IAX2/user@peer) will hit the 's' extension since none was given. |
| 15:24.11 | Idle | now, how would I set it so that with my SIP phone, it does 'Notify', then dials the extension from the office context? |
| 15:24.24 | mne | hi guys. how far is current asterisk support for T.38 ? I would like to set up an email to fax gateway using T.38 over a sip provider |
| 15:24.46 | jbalcomb | Anyone else read the asterisk-biz mailing? I just read through the G.729/G.723 threads. Some of those people are both crazy and retarded. |
| 15:25.01 | jbalcomb | file: You work for Digium right? |
| 15:25.09 | Idle | I tried another extension with i,1,Notify, i,2,Goto(office,${EXTEN},1) but that didn't work |
| 15:25.26 | jamincollins | benjk: that suggestion will output the q.931 cause code, right? |
| 15:25.30 | jamincollins | err right... |
| 15:25.46 | benjk | yes, but only if you are on a BRI or PRI channel |
| 15:25.56 | benjk | unfortunately |
| 15:26.24 | Qb3rt | file: what can cause this error if the number really exist and i am sure i am dialing it properly?? |
| 15:27.09 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:28.04 | jamincollins | benjk: hmmm, won't help in my situation then, I'm already seeing the cause code, but need to know more about why the disconnect happened.... cause code is 16 which is a normal call clearing, but this is anything but normal |
| 15:29.02 | benjk | you can always do a pri intense debug and look at the cleartext translations of the messages |
| 15:30.02 | jamincollins | benjk: under just pri debug, there doesn't seem to be anything kicking off the disconnect, other than asterisk |
| 15:30.06 | Qb3rt | problem ---> http://pastebin.ca/164361 |
| 15:30.16 | jamincollins | asterisk seemingly just decides to disconnect the call |
| 15:30.40 | benjk | did you do *intense* debug? |
| 15:31.01 | benjk | it shows you if its an incoming or outgoing message |
| 15:31.25 | jamincollins | pri debug shows that's it's an outgoing |
| 15:31.31 | jamincollins | > vs < |
| 15:31.57 | benjk | ok |
| 15:32.09 | benjk | whats the nature of the previous incoming message |
| 15:32.18 | benjk | before * sends the hangup |
| 15:32.28 | jamincollins | there's nothing for several seconds |
| 15:32.57 | jamincollins | pulling it from the log, one sec |
| 15:32.59 | benjk | sounds like a timeout of some sort |
| 15:33.12 | benjk | I'd still try *intense* debug though |
| 15:33.28 | jamincollins | it just happens in the middle of calls, somtimes 2 minutes in, sometimes 10, somtimes 40 |
| 15:33.51 | jamincollins | I thought timeout too, but I've seen it happen during both parties speaking |
| 15:34.03 | benjk | happy debug :) |
| 15:34.19 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:35.11 | jamincollins | previous pri message was an outbound CONNECT ACKNOWLEDGE, 5 minutes prior to asterisk initiating a DISCONNNECT |
| 15:35.26 | benjk | you probably want to look at placing some debug statements into libpri |
| 15:35.42 | benjk | recompile and then see if you can narrow it down |
| 15:36.42 | bkw_ | moving targets are fun |
| 15:40.47 | Bert- | asterisk 4 fun : call 2 unknown (or known) people then auto conference |
| 15:40.53 | Bert- | if you are bored at desk |
| 15:40.55 | Bert- | good game :) |
| 15:40.57 | Di[Lv] | hi may be someone can help width asterfax - we got the problem to send out default tiff and windows maked tiff asterisk crashing width out errors incoming faxes we handling ok - we runing asterisk-1.2.9.1 ghostscript-8.54 tiff-v3.6.0 spandsp-0.0.3pre22 |
| 15:42.17 | *** join/#asterisk frenzy (n=frenzy@196.46.104.77) |
| 15:42.56 | frenzy | hi.. I get alot of " SIP response 406 "Not Acceptable" back from ATA-IP " on incoming calls |
| 15:43.14 | frenzy | What causes this? |
| 15:43.22 | CtRiX|h | codecs, probably |
| 15:43.22 | frenzy | am using Sipura 9000 |
| 15:44.14 | *** part/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 15:44.15 | *** join/#asterisk Filar (n=none@sw.m5net.com) |
| 15:44.37 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:45.55 | Qb3rt | can you please help me with my problem ---> http://pastebin.ca/164361 |
| 15:47.01 | eKo1 | WTF?! For some reason, the sip CLI commands have disappeared... |
| 15:47.15 | devel | so benjk, in this intense pri debugging, is '<' to or from asterisk? |
| 15:47.42 | benjk | outbound I think |
| 15:48.21 | devel | that's what i thought too, but at the start of a call, with the callerid info and such, they were all '<' messages, so.... |
| 15:50.38 | frenzy | the codecs are all ulaw but still keeps getting SIP/2.0 406 Not Acceptable |
| 15:50.38 | jamincollins | a "<" is inbound to the * |
| 15:50.46 | jamincollins | a ">" is outbound from the * |
| 15:50.47 | benjk | its been a while that I did PRI debugging, so I might be wrog |
| 15:50.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:51.03 | devel | ok, that would make sense then jamincollins |
| 15:51.04 | jamincollins | I'm pretty sure I've got that bit down... |
| 15:51.18 | jamincollins | been staring at the PRI traces for the last two days |
| 15:51.25 | benjk | you can always look at the sources in libpri to verify |
| 15:51.50 | Di[Lv] | for sure noone using asterfax |
| 15:52.10 | jamincollins | Di[Lv]: I opted for iaxmodem instead of asterfax |
| 15:52.35 | devel | what we have is an issue where we get a call, the ATA says "redirect", we start rerouting the call, then it _looks_ like the PRI channel is hung up inbound (which isn't the case from us holding the phone dialing in our hand) |
| 15:53.34 | devel | so we're trying to look at the inbound PRI to verify... anybody have any ideas on that, or seen the likes before? |
| 15:55.28 | Di[Lv] | <jamincollins> we are using beronet card width misdn drivers -other side call ringing and try get fax but other side stops width mesage emty fax |
| 15:56.13 | _DAW | I need some help with a sip configuration issue. |
| 15:56.34 | jamincollins | Di[Lv]: nope, no beronet card, just a TE110P and the iaxmodem software along with hylafax |
| 15:56.34 | Di[Lv] | if we try to send out recived fax than everything works |
| 15:57.04 | _DAW | I need to know if it is possible to remove the a=silenceSupp:off attribute from SDP in invites. |
| 15:57.41 | Di[Lv] | <jamincollins> so U sugest to use hylafax not asterfax |
| 15:57.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:57.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:58.33 | *** part/#asterisk frenzy (n=frenzy@196.46.104.77) |
| 15:59.20 | jamincollins | Di[Lv]: /I/ had problems with asterfax and found that iaxmodem just worked |
| 15:59.48 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 16:01.11 | Di[Lv] | <jamincollins> tnx , will try |
| 16:01.51 | oej | ~seen ahrimanes |
| 16:01.58 | jbot | ahrimanes <n=michael@81.7.159.2> was last seen on IRC in channel #asterisk, 4d 5h 4m 8s ago, saying: 'yay!'. |
| 16:02.20 | benjk | Di[Lv] iaxmodem can but doesn't have to be used with hylafax |
| 16:02.41 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:04.25 | Di[Lv] | now ai get point - when I starting asterfax I got error ./nohupasterfax.sh: line 27: iaxmodem: command not found - so I will install iaxmodem |
| 16:06.28 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:11.03 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:11.56 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:15.10 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 16:15.50 | *** join/#asterisk Vec (n=Vector@dsl-146-113-250.telkomadsl.co.za) |
| 16:16.25 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:21.38 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 16:25.46 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 16:25.46 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 16:27.16 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:28.33 | stubert | any known issues with gxp2000 and DTMF? |
| 16:29.01 | CunningPike | stubert: Are you using inband or rfc2388? |
| 16:29.27 | stubert | rfc2388 |
| 16:29.38 | stubert | let me give you more... |
| 16:29.38 | CunningPike | Hmm - shouldn't be an issue then..... |
| 16:31.14 | stubert | gxp2000 gsm -> asterisk iax2/gsm -> asterisk pstn -> endpoint |
| 16:31.25 | stubert | DTMF does not reach endpoint |
| 16:31.33 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 16:31.48 | CunningPike | stubert: Does it reach the second asterisk? |
| 16:31.49 | devel | well there you go. i changed the inter-switch to use SIP rather than IAX2, and my problem of PRI hangup went away.... |
| 16:32.18 | stubert | CunningPike: my stupid is showing, how do I tell... |
| 16:32.45 | stubert | iax2 debug ??? |
| 16:32.54 | CunningPike | Try a test extension on your second asterisk that reads back your dtmf digits to you |
| 16:33.08 | stubert | will do... |
| 16:35.08 | *** join/#asterisk Filar (n=none@sw.m5net.com) |
| 16:37.39 | frawd | hi all! is there any way to do some sound amplification in asterisk (for crappy SIP phones with too low volume)? |
| 16:37.47 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 16:38.26 | [TK]D-Fender | frawd: You can adjust Zaptel gains, but thats about it. |
| 16:38.45 | frawd | damn :( |
| 16:38.51 | c4t3l | ever heard of VM password unable to be changed? what casues this? |
| 16:39.12 | *** join/#asterisk watchy2 (n=watchy@office2.gwhsi.com) |
| 16:39.48 | Nugget | might be permissions on the voicemail.conf file. |
| 16:40.17 | c4t3l | should they be 640 ? |
| 16:40.19 | frawd | i also have some problem with zap lines, when i make any outgoing call, i always loose the first word of the called person (first second of the call or so)... I think it's related to echo cancelling, but not sure. anyone has a clue? |
| 16:40.36 | c4t3l | user is root gourp is asterisk |
| 16:40.41 | c4t3l | group** |
| 16:41.03 | CunningPike | c4t3l: Are you running asterisk as root? |
| 16:41.43 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 16:43.17 | jamincollins | if not, that's a problem, as group only has read permissions |
| 16:45.09 | frawd | my settings are echocancel=64 and echotraining=600, so i was thinking maybe the echo training makes me loose the first 600ms of every call i make. Can this be it? |
| 16:47.09 | c4t3l | * is running as root |
| 16:47.15 | frawd | and in that case, is there a workaround (appart from disabling echo training of course) |
| 16:47.19 | frawd | ? |
| 16:47.20 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 16:47.46 | sx-wks | is there a howto run asterisk as non-root ? |
| 16:48.42 | watchy2 | can a 2.4ghz box handle 60 sip phones connected to it with maybe 10 outgoing calls at a time? |
| 16:49.02 | joe | are there any OSS soft phone clients for linux these days? |
| 16:49.09 | [TK]D-Fender | frawd: Thats exactly it. |
| 16:49.12 | jamincollins | sx-wks: I don't know of a specific howto, but debian's asterisk packages run as the asterisk user instead of root |
| 16:49.32 | [TK]D-Fender | watchy2: Without transcoding, no problem at all |
| 16:49.32 | joe | watchy2: it should |
| 16:49.53 | [TK]D-Fender | joe: Ekiga, kphone, linphone, etc |
| 16:49.57 | frawd | [TK]D-Fender: thanks, but do you know of any other solution (i badly need echo cancellation)? |
| 16:50.15 | [TK]D-Fender | frawd: Get a better card with onboard EC. |
| 16:50.35 | joe | [TK]D-Fender: I've tried linphone and kphone w/ very very shitty quality a while back :/ I'll try ekiga. thanks |
| 16:50.51 | pdt | has anybody that uses the presence stuff on polycom phones played with the 2.x sip software, it seems to break it... hints are still being updated, but the phone watch doesn't seem to actually watch |
| 16:50.59 | frawd | [TK]D-Fender: good idea, are you sure that these card have no "training" period? |
| 16:51.33 | joe | anyone have a polycom 4000 conference phone that goes in loops when trying to configure it? |
| 16:51.58 | [TK]D-Fender | frawd: I can tell you that there is no perceived delay or loss of any quality with the Sangoma EC cards I've used |
| 16:52.32 | frawd | [TK]D-Fender: thanks so much, i'll buy one of these right away to try! |
| 16:53.04 | tzanger | hmm |
| 16:53.48 | tzanger | T100P to channel bank to POTS. incoming calls have very little echo, but outgoing calls typically have bad echo. guessing that it has to do with the echo canceller training up before the circuit path is completely solidified on the telco side |
| 16:53.55 | tzanger | any ideas on how to overcome this? |
| 16:54.59 | CunningPike | ~non-root |
| 16:55.01 | jbot | non-root is, like, what you should irc as |
| 16:55.06 | [TK]D-Fender | tzanger: If there is an inherent delay in connecting the path you can maybe elongate your training period sso taht you still overlap enough to do the job.... |
| 16:55.15 | CunningPike | You're a mine of information, jbot |
| 16:55.32 | tzanger | [TK]D-Fender: good point, let me see if I can do that |
| 16:55.46 | CunningPike | sx-wks: http://www.voip-info.org/wiki-Asterisk+non-root |
| 16:55.57 | CunningPike | sx-wks: giyf |
| 16:56.01 | CunningPike | ~giyf |
| 16:56.02 | jbot | extra, extra, read all about it, giyf is Google Is Your Friend, or see also: STFW |
| 16:56.25 | [TK]D-Fender | ~stfw |
| 16:56.27 | jbot | stfw is probably Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
| 16:56.32 | [TK]D-Fender | what I though ;) |
| 16:56.34 | Strom_M | haha |
| 16:56.38 | CunningPike | :D |
| 16:59.19 | jmls | anyone managed to compile app_txfax and app_rxfax on svn trunk ? Any clues ? |
| 17:02.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@80.75.184.179) |
| 17:02.47 | Cresl1n | tzanger did you try turning echotraining off? |
| 17:03.01 | tzanger | Cresl1n: it was off |
| 17:03.02 | tzanger | I turned it on |
| 17:03.10 | tzanger | and failing that, I'll try bumping it out to 1500s |
| 17:03.12 | tzanger | er 1500ms |
| 17:03.54 | Cresl1n | echo training is probably not going to be the answer |
| 17:04.04 | Cresl1n | what signallng method are you using? |
| 17:04.55 | jmls | hey, I've just tried including a file in queues.conf and it worked. Is this supported behaviour ? Didn't know it woirked outside the dialplan |
| 17:05.03 | jmls | (worked) |
| 17:05.06 | stubert | CunningPike: Does not work using gsm, works 70% of the time using ulaw |
| 17:05.29 | Strom_C | jmls: yes, i believe that works in almost every config file except voicemail.conf |
| 17:05.39 | jmls | cool :) |
| 17:05.47 | jmls | is it documented ? |
| 17:06.07 | CunningPike | stubert: Hmmm - sounds like you're using inband somewhere in the path - let me check something |
| 17:08.19 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 17:08.57 | CunningPike | stubert: dtmf over iax is inband by the very nature of iax, methinks - unless I'm talking bollocks |
| 17:09.22 | CunningPike | stubert: What's the connection between the two asterisks, network-wise |
| 17:09.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:09.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:09.44 | CunningPike | stubert: Also, try the same experiment on the first asterisk server |
| 17:10.19 | Strom_C | DTMF over iax is inband in the sense that it travels over the same UDP stream, but out of band in the sense that it's sent as a signaling message and not a media frame |
| 17:10.29 | eKo1 | Does a voicemail mailbox have to be of the form number@context? Can I use letters instead of numbers? |
| 17:12.10 | stubert | CunningPike: two asterisk systems are on lan |
| 17:12.55 | Dr-Linux|work | i noticed ths warning on CLI few times: Warning, flexibel rate not heavily tested! |
| 17:13.01 | Dr-Linux|work | what does this mean? |
| 17:13.02 | stubert | CunningPike: It appears the first asterisk box is fine with dtmf |
| 17:13.53 | stubert | CunningPike: although iax2 does send it's tones inband, it errors big time if the devices are set to inband |
| 17:14.38 | file | that's because there is supposed to only be 1 way of transporting DTMF across IAX2 |
| 17:14.56 | file | devices are not supposed to send it inband... but you could |
| 17:15.08 | russellb | it shouldn't be possible to do that |
| 17:15.19 | file | russellb: other implementations do it |
| 17:15.26 | russellb | file: wtf? |
| 17:15.27 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 17:15.33 | file | russellb: yeah. |
| 17:15.50 | russellb | lame |
| 17:16.05 | russellb | well, it certinaly could not possibly work with asterisk |
| 17:16.10 | stubert | So, should I try sip between the asterisk boxes to see if DTFM works? |
| 17:17.36 | wunderkin | dtfm problems on astrix? o rly? |
| 17:18.36 | chode | Hello everyone |
| 17:18.55 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 17:19.24 | THX2000 | anyone using teliax on the west coast? |
| 17:19.30 | stubert | I am... |
| 17:19.42 | THX2000 | u noticed any artifacting in the calls recently? |
| 17:19.48 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 17:20.00 | stubert | yes... since about 12 noon yesterday |
| 17:20.03 | stubert | PST |
| 17:20.05 | stubert | PDT |
| 17:20.05 | THX2000 | yea, me too |
| 17:20.07 | stubert | whatever |
| 17:20.16 | THX2000 | well at least it isn't just me |
| 17:20.42 | THX2000 | Not sure if thats something to be relieved about or not |
| 17:20.50 | stubert | Only coming out of my system |
| 17:21.09 | THX2000 | your incoming is fine? |
| 17:21.19 | stubert | today... |
| 17:21.32 | stubert | It may be a backbone issue |
| 17:21.54 | chode | are there available freelancers on this channel ? i need asterisk related work to be done for money... anyone ? |
| 17:21.56 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 17:22.30 | benjk | you may want to tell us the nature of what you want to do |
| 17:22.58 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.247) |
| 17:23.11 | chode | anyone who's interested please look at http://www.voipscout.net/sip_rtp_cdr_agi.html |
| 17:23.16 | chode | those are the requirements |
| 17:23.53 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:25.10 | *** join/#asterisk adamowitz (n=adamowit@ip68-109-23-191.ri.ri.cox.net) |
| 17:26.07 | eKo1 | seems pretty simple |
| 17:27.10 | tzanger | chode: yep |
| 17:27.31 | bkw_ | chode, and you want to use asterisk? |
| 17:27.44 | eKo1 | Did you make that document? |
| 17:27.47 | CunningPike | stubert: You could try changing the codec....... |
| 17:28.05 | eKo1 | chode: as I said, I'm not interested. |
| 17:28.08 | chode | bkw_: i would use anything, but asterisk seems to be mostly available |
| 17:29.05 | chode | eKo1: yes, i wroye that doc |
| 17:29.05 | tzanger | that looks like fun. |
| 17:29.09 | tzanger | and by fun I mean tedios |
| 17:29.11 | tzanger | er tedious |
| 17:29.12 | tzanger | wow |
| 17:29.22 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:29.49 | chode | i am opened to suggestions or "bids" quotes, anyone who's got time and needs money, please respond... |
| 17:30.19 | russellb | i'll do it for 1 BILLION DOLLARZ ! |
| 17:30.51 | eKo1 | dollarz? Where do they use those? |
| 17:31.38 | THX2000 | I'll help russellb for a cut :P |
| 17:32.30 | chode | ppl, you got your deal, but i pay in dollarZ this amount.... and by fax ofcourse ;-) |
| 17:36.13 | *** join/#asterisk Rick_Hunter (n=rhunter@10-216.008.popsite.net) |
| 17:38.07 | *** part/#asterisk Rick_Hunter (n=rhunter@10-216.008.popsite.net) |
| 17:38.16 | Strom_C | chode: you realize loopstart /can/ detect call pickup if you have the telco provision the line such that it does polarity reversal for supervision, right? |
| 17:42.35 | chode | member:identifier:strom_c: Yes, but that's not the case this time... |
| 17:42.51 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:42.59 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 17:45.04 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 17:46.22 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:47.54 | *** join/#asterisk Qwell (i=north@unaffiliated/qwell) |
| 17:47.54 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:48.54 | *** join/#asterisk Adrian__ (n=foo@zux221-091-123.adsl.green.ch) |
| 17:49.02 | Adrian__ | good evening all |
| 17:52.00 | stoffell | evening Adrian__ |
| 17:52.32 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
| 17:52.56 | ctooley | When is the next Astricon |
| 17:53.17 | oej | Never, no more real Astricons ;-) |
| 17:54.09 | tessier__ | Anyone know why my agents get call waiting indications when there are calls waiting in the queue and they are already on a call? |
| 17:54.42 | jmls | are they agents or devices ? |
| 17:54.58 | jmls | member=>agent/1 or member=>SIP/712 |
| 17:55.02 | tessier__ | Devices |
| 17:55.13 | tessier__ | Actually we are not using member=> |
| 17:55.28 | jmls | are they multiple line phones ? |
| 17:55.31 | tessier__ | I'm not yet clear on the best way to set up queues |
| 17:55.33 | tessier__ | Yeah, Snom 220 |
| 17:56.16 | tessier__ | We used to put people in the queue with AgentCallbackLogin in the dialplan but we need agents to be able to login to multiple queues |
| 17:56.17 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 17:56.42 | tessier__ | We found that if we use the asterisk manager interface to put people into the queue with QueueAdd action we can script putting them into multiple queues |
| 17:56.46 | tessier__ | We give it SIP/extension |
| 17:58.27 | jmls | I think that if a device (SIPA) is busy on queue A and a call comes through on queue B then as far as the system is concerned the device is not busy, i.e agent status is not across queues, only within queues. I could be wrong. |
| 17:58.53 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 17:59.02 | jmls | we got round it by only allowing a single line on the phone. Then when queue B tries, the phone is busy. |
| 17:59.08 | jmls | That was 1.2 |
| 17:59.26 | jmls | in 1.4 we use jabber to check the presence of the agent before trying to call |
| 18:00.21 | jmls | in our app we set the agent presence to busy / not busy / wrapup etc |
| 18:00.40 | tessier__ | Disabling all but one of the lines on our expensive phones is not an option |
| 18:00.55 | jmls | yup. we have 7940/7960's |
| 18:00.56 | Strom_C | how about this then |
| 18:01.04 | Strom_C | give each line a separate appearance |
| 18:01.06 | Strom_C | er |
| 18:01.17 | Strom_C | give each appearance a separate numbered extension |
| 18:01.25 | Strom_C | allow only one extension for queueing |
| 18:01.32 | Strom_C | and leave the rest free for other nonsense |
| 18:02.07 | jmls | or use chan/local and check for some db key / chanisavail / etc *before* you dial |
| 18:06.46 | Di[Lv] | plz hellp - txfax crashing asterisk when sending out fax using latests spandsp spandsp-20060907.tar and tiff-v3.6.0 |
| 18:07.13 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:07.21 | tessier__ | We may have to just write our own queuing code |
| 18:07.29 | tessier__ | I am surprised the stuff that comes with * out of the box is not more functional |
| 18:07.43 | nextime | Di[Lv] : consider to try iaxmodem instead of rxfax and txfax. |
| 18:10.01 | Di[Lv] | nextime but I need to send fax to outside number from e-mail and rxfax works fine |
| 18:11.54 | nextime | Di[Lv] : sure, but with hylafax you can do mail to fax, fax to mail, and many other things with many systems and softwares, it is more stable, and is indipendent from asterisk itself, so you can upgrade, downgrade, recompile without need to patch anything |
| 18:11.58 | tessier__ | Why did they create yet another language with string handling functions instead of using perl or python or something? Ugh |
| 18:12.47 | justinu|laptop | good question |
| 18:13.35 | Di[Lv] | tnx nextime -I will try to install hylafax |
| 18:14.58 | nextime | Di[Lv] : hylafax is the "fax server software", to use it with asterisk and spandsp you need also iaxmodem, that emulate a "real hardware serial connected modem" creating a virtual tty device to attach to with hylafax |
| 18:17.23 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 18:17.35 | CtRiX | Di[Lv], consider trying openpbx which has T38 termination working perfectly |
| 18:17.46 | CtRiX | nextime, ciao |
| 18:17.54 | nextime | ciao CtRiX |
| 18:19.31 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 18:19.31 | CtRiX | Di[Lv], openpbx.org has T38 termination with modified code and support txfax 2 rxfax,that is rtp faxing out of the box. |
| 18:22.20 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:22.33 | redder86 | who said something about iaxmodem ? |
| 18:23.02 | CtRiX | <Di[Lv]> plz hellp - txfax crashing asterisk when sending out fax using latests spandsp spandsp-20060907.tar and tiff-v3.6.0 |
| 18:23.06 | CtRiX | <nextime> Di[Lv] : consider to try iaxmodem instead of rxfax and txfax. |
| 18:23.13 | CtRiX | here you are redder86 ! |
| 18:23.40 | redder86 | :-) |
| 18:26.34 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 18:26.54 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 18:26.57 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 18:28.02 | chode | CtRiX: are you interested in some work for money ? check these requirements out http://www.voipscout.net/sip_rtp_cdr_agi.html |
| 18:32.07 | *** join/#asterisk Avalone (n=Avalone_@dial-183.vl-cen-as1.avtlg.ru) |
| 18:32.59 | Avalone | hi all ... how (at 1.2) i'm able join 2-way talk to conference? |
| 18:33.04 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 18:34.09 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 18:41.23 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 18:42.09 | stephane_ | soir |
| 18:44.05 | tessier__ | voulez vous couche avec moi, c'est soir? |
| 18:45.02 | joe | tessier__: at least get it right, if you're going to say such silly things :P |
| 18:45.14 | tessier__ | yeah yeah |
| 18:46.38 | Ox0F0-0FF | salut stephane_ |
| 18:46.43 | tessier__ | asterisk dialplan syntax suuuuucks.... :( |
| 18:46.50 | stephane_ | soir Ox0F0-0FF |
| 18:46.54 | Ox0F0-0FF | :) |
| 18:47.10 | benjk | tessier, you're kidding right? |
| 18:47.36 | tessier__ | No, I'm not. |
| 18:47.38 | Ox0F0-0FF | stephane_, n=stephane@merlin.cabale.ne <=== quebec ? |
| 18:47.41 | benjk | :D |
| 18:47.45 | stubert | CunningPike: Just an update, it seems to work fine asterisk sip -> asterisk |
| 18:48.06 | benjk | tessier you are in danger of being promoted to troll here |
| 18:48.15 | tessier__ | How so? |
| 18:48.27 | benjk | tessier__: asterisk dialplan syntax suuuuucks.... |
| 18:48.44 | tessier__ | We are only allowed to say nice things about asterisk here? |
| 18:49.04 | benjk | reasonable comment of course, but |
| 18:49.08 | tessier__ | I've been here for 3 years. I think I can say something sucks without being accused of trolling. |
| 18:49.14 | benjk | hehe |
| 18:49.20 | benjk | I thought so too |
| 18:49.28 | CunningPike | stubert: OK - great |
| 18:49.48 | benjk | but I ppl tell me it wasn't so |
| 18:49.54 | stubert | CunningPike: can you think of a reason why though? |
| 18:49.56 | Ox0F0-0FF | I think it is good when someone says something sucks... It helps highlighting issues and help to have the software better.... just by criticism... |
| 18:49.58 | tdonahue-laptop | does anyone know if there is a variable i can use to access the voicemail box assigned to a sip account? |
| 18:50.04 | stephane_ | Ox0F0-0FF, non France :) |
| 18:50.19 | CunningPike | stubert: Not really :( |
| 18:50.23 | justinu|laptop | tessier: you're not the only one who feels that way |
| 18:50.27 | benjk | trolling day today here |
| 18:50.29 | benjk | I like it |
| 18:50.35 | benjk | feels like home again |
| 18:50.39 | justinu|laptop | hehe |
| 18:50.53 | stubert | It seems that every time I try to use IAX2 it just doesn't work... |
| 18:51.00 | justinu|laptop | troll!! |
| 18:51.03 | benjk | hehe |
| 18:51.44 | CunningPike | stubert: Pastebin your iax.conf |
| 18:51.46 | CunningPike | ~pb |
| 18:51.48 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 18:52.18 | benjk | justinu, note the absence of intervention from any fanboy, that's refreshing isn't it |
| 18:52.27 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:52.48 | justinu|laptop | they're probably just afk |
| 18:53.00 | benjk | probably |
| 18:53.04 | stubert | CunningPike: maybe I should refraise my statement... |
| 18:53.12 | benjk | otherwise we would have been whacked by now |
| 18:53.15 | tessier__ | I am having to implement my own queue stuff in the dialplan. Trying to get the second example from http://www.voip-info.org/wiki/view/Agents+without+agent+channel working |
| 18:53.26 | tessier__ | Trying to figure out how this works |
| 18:53.31 | CunningPike | stubert: ?? |
| 18:53.56 | benjk | the fanboys usually respond to that, that Voip-Info.org is hopelessly wrong or out of date |
| 18:54.03 | justinu|laptop | tessier: honestly, i would give you a hand but I haven't needed to look at agents yet |
| 18:54.27 | justinu|laptop | this channel is also a cesspool of rumor and ignorance |
| 18:54.32 | benjk | but, who am I to say its not so, or it is so, just passing on the wisdom of the channel |
| 18:54.32 | stubert | CunningPike: It seems that whenever I use IAX2 to connect to another asterisk box I get wierd problems... could it have anything to do with clocking? |
| 18:54.41 | [TK]D-Fender | justinu|laptop: Don't forget the slander! |
| 18:54.51 | justinu|laptop | actually, i don't believe that... but some others feel that way about #asterisk i guess |
| 18:55.10 | tessier__ | extension priorities...that's similar to order order of execution in imperative progamming. Jumping the prioritiy by 100 for decision making...that sounds like flow control. DBGet(foo="/key/tree") is a lot like foo = DBGet("/key/tree") in saner languages... |
| 18:55.14 | justinu|laptop | i know that a lot of ppl have gotten a lot of problems solved by a lot of regulars here |
| 18:55.23 | CunningPike | stubert: No - could be your config - pastebin it, man |
| 18:55.46 | tessier__ | justinu|laptop: Thanks anyhow, I appreciate it. We'll work something out. Just need time to go over this code. |
| 18:55.53 | justinu|laptop | slander!! [TK]D-Fender is a nub! |
| 18:56.50 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 18:56.57 | [TK]D-Fender | OMGZ! |
| 18:57.52 | benjk | tessier, don't even look at pbx.c (where this crap, er beauty is implemented), I guarantee you that you will throw up yesterday's meals all at once |
| 18:58.40 | stubert | [general] |
| 18:58.40 | stubert | delayreject=yes |
| 18:58.40 | stubert | language=en |
| 18:58.40 | stubert | disallow=all |
| 18:58.41 | stubert | allow=gsm |
| 18:58.43 | stubert | allow=ulaw |
| 18:58.45 | stubert | tos=0x10 |
| 18:58.48 | stubert | dtmfmode=rfc2833 |
| 18:58.50 | stubert | jitterbuffer=yes |
| 18:58.50 | justinu|laptop | ~pb |
| 18:58.52 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 18:58.53 | stubert | forcejitterbuffer=no |
| 18:58.56 | stubert | dropcount=2 |
| 18:58.58 | stubert | maxjitterbuffer=1000 |
| 18:59.00 | stubert | maxjitterinterps=10 |
| 18:59.03 | stubert | resyncthreshold=1000 |
| 18:59.04 | benjk | hehe |
| 18:59.05 | stubert | [actusa] |
| 18:59.08 | stubert | type=peer |
| 18:59.10 | stubert | auth=md5 |
| 18:59.13 | stubert | secret=<secret> |
| 18:59.15 | benjk | oh dear, how long is this? |
| 18:59.16 | stubert | host=voipserver |
| 18:59.18 | stubert | context=demo |
| 18:59.20 | stubert | accountcode=demo |
| 18:59.23 | stubert | disallow=all |
| 18:59.25 | stubert | allow=gsm |
| 18:59.34 | stubert | it's done |
| 18:59.50 | benjk | do you think that was cool? |
| 18:59.56 | Strom_M | NEVER DO THAT AGAIN |
| 19:00.02 | Strom_M | !!!!!!!!!!!!!!!!!!! |
| 19:00.12 | stubert | OK |
| 19:00.25 | justinu|laptop | hahaha |
| 19:00.33 | CunningPike | stubert: I told you _twice_ to pastebin - and even gave you the link |
| 19:00.39 | benjk | stubert, there are places where you can paste your stuff and then put the URL here |
| 19:00.52 | stubert | sorry... |
| 19:00.56 | stubert | my fault |
| 19:01.34 | shodan | hmm I'm using analog phone -> fxs(spa-2102) -> * -> fxo(x100p) to call another pbx , but when I dial the extension the other pbx isn't understanding my dmtf ?! I'm using ulaw btw |
| 19:01.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:02.27 | benjk | are you sure that the spa and asterisk are configured to use the same DTMF method? |
| 19:04.47 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 19:05.06 | stubert | http://pastebin.ca/164563 |
| 19:05.57 | CunningPike | stubert: Give me a while - on the phone |
| 19:06.07 | stubert | CunningPike: np |
| 19:06.12 | *** part/#asterisk Avalone (n=Avalone_@dial-183.vl-cen-as1.avtlg.ru) |
| 19:06.41 | benjk | shodan, benjk: are you sure that the spa and asterisk are configured to use the same DTMF method? |
| 19:08.10 | hmmhesays | anyone in here use broadvoice? |
| 19:08.20 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 19:08.54 | CunningPike | stubert: I don't think the dtmfmode is a valid entry for iax.conf - maybe remove it and see what happens |
| 19:09.48 | hmmhesays | i'm curious how many simultaneous calls they allow |
| 19:12.29 | Strom_M | hmmhesays, two or so, i think |
| 19:13.06 | hmmhesays | yeah I wish they listed it |
| 19:15.51 | stubert | nope... same issue |
| 19:15.56 | tessier__ | If the key requested by DBGet is not found does asterisk still jump to n+101? |
| 19:16.09 | stubert | CunningPike: nope same issue |
| 19:16.14 | hmmhesays | isn't DBGet deprecated? |
| 19:16.18 | tessier__ | Because mine seems to be going to n+1 whether it gets a value or not |
| 19:16.26 | tessier__ | Deprecated or broken? |
| 19:16.30 | tessier__ | There is a difference. |
| 19:16.55 | stubert | CunningPike: I seem to always be having problems consistantly with IAX2 asterisk -> asterisk |
| 19:17.27 | stubert | CunningPike: delays, DTMF, Short Frame Errors |
| 19:18.04 | CunningPike | stubert: Network issues, maybe? |
| 19:18.08 | [TK]D-Fender | tessier__: DBGet is deprecated. Use DBEXISTS and DB for that now. |
| 19:18.27 | CunningPike | stubert: Our IAX Just Works(tm)....... |
| 19:18.51 | CunningPike | stubert: Try without the jitterbuffer |
| 19:18.56 | tessier__ | [TK]D-Fender: I don't see dbexists on the asterisk dialplan commands page |
| 19:19.15 | [TK]D-Fender | tessier__: "lookup "asterisk functions" |
| 19:19.36 | stubert | CunningPike: I don't have issues with Teliax and IAX2... |
| 19:19.36 | [TK]D-Fender | tessier__: The WIKI isn't always up to date in every spot. |
| 19:19.48 | tessier__ | db_exists |
| 19:20.11 | CunningPike | stubert: Don't know what else to suggest then..... |
| 19:20.20 | *** join/#asterisk ToTo (n=ToTo@host149-109.pool8258.interbusiness.it) |
| 19:20.26 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 19:20.29 | CunningPike | stubert: Maybe pastebin the iax.conf from the other asterisk server |
| 19:21.12 | sevard | Does anyone know what extension _s is? |
| 19:21.18 | stubert | CunningPike: thanks man... I have some other things I can try, like swap out one side of the connect |
| 19:21.49 | CunningPike | sevard: What does your dialplan say it is? :) |
| 19:22.21 | sevard | In the standard vmexten macro listed on voip-info it includes a _s |
| 19:22.33 | CoffeeIV_ | I have a digium T1 card installed, but no T1 hooked up to it -- asterisk doesn't start, giving the error "Unable to specify channel 1: No such device or address" -- is it possible my * is configured correctly but just needs the T1 actually connected to work ? |
| 19:23.11 | [TK]D-Fender | CoffeeIV_: You have defined the channels, but likely do not have the module loaded... |
| 19:24.14 | FuriousGeorge | hey all. pci bays are backwards compatible right? the last time i stuck a tdm400p in this tyan tomcat mb it fried the thing |
| 19:24.19 | FuriousGeorge | so i rma'd it and got iot back |
| 19:24.22 | FuriousGeorge | and now im scared |
| 19:24.50 | *** join/#asterisk dprevite (n=dprevite@c-67-162-110-89.hsd1.il.comcast.net) |
| 19:25.13 | CoffeeIV_ | D-Fender: the kernel module wct1xxp and zaptel s loaded, is there another module I have to load also ? |
| 19:26.45 | [TK]D-Fender | CoffeeIV_: PB your zapata & zaptel. Before doing that try "ztcfg -vvvv" and if it doesn't bomb out on you try starting * again. |
| 19:31.07 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 19:33.01 | *** part/#asterisk steveaj (n=steve@62.55.147.53) |
| 19:33.01 | *** join/#asterisk bkunyiha (n=bgitonga@66-113-78-5.rev.ibsinc.com) |
| 19:33.44 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 19:33.47 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:37.30 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 19:39.08 | C6Vette | IM getting seg faults and no core dumps. I just see "stdout: Broken pipe" then "Ouch ... error while writing audio data: : Broken pipe" then "Segmentation fault" |
| 19:39.11 | C6Vette | any ideas? |
| 19:39.37 | C6Vette | running version: Asterisk SVN-branch-1.2-r40901M |
| 19:40.26 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 19:41.15 | CoffeeIV_ | I'm having trouble getting a Digium T1 card to work -- I have made a pastebin, any comments would be greatly appreciated: http://pastebin.ca/164576 |
| 19:41.57 | tessier__ | Anyone know if res_python is still alive? Nothing on the wiki about it. |
| 19:41.57 | tessier__ | res_perl has some docs. |
| 19:42.08 | tessier__ | I prefer python but I'll use whatever is most supported I guess. |
| 19:46.28 | CoffeeIV_ | I wish pastebin had a little hit tracker on it so I could tell if any of you guys looked at my problem |
| 19:47.41 | C6Vette | pastebin.ca does |
| 19:48.03 | C6Vette | Total Paste Views: 6 |
| 19:48.16 | CoffeeIV_ | I see it now -- cool |
| 19:48.27 | CoffeeIV_ | unfortunately like 3 or 4 of those were me |
| 19:48.50 | C6Vette | you did mobrobe for the device correct? |
| 19:49.05 | CoffeeIV_ | yes, and it is listed in lsmod (the module) |
| 19:49.06 | C6Vette | s/mobrobe/modprobe/s |
| 19:49.17 | C6Vette | ok |
| 19:50.50 | [TK]D-Fender | CoffeeIV_: Please PB "cat /proc/interrupts" |
| 19:50.58 | CoffeeIV_ | ok |
| 19:51.16 | *** join/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 19:51.32 | *** join/#asterisk aixa (n=aixa@195.2.112.177) |
| 19:51.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 19:52.31 | CoffeeIV_ | D-Fender: I added /proc/interrupts to the bottom of that pastebin |
| 19:52.45 | aixa | any idea if we can expect some fax sending utility from authors of mISDN, so that we could use inbuilt dsp functionality of beronet cards? |
| 19:53.02 | aixa | maybe such code exists somewhere as experiemntal test? |
| 19:55.04 | [TK]D-Fender | CoffeeIV_: new link please |
| 19:55.24 | file | [TK]D-Fender: eep! |
| 19:56.32 | aixa | [TK]D-Fender: hi, that forwarding we talked about the other day - works perfectly |
| 19:56.33 | FuriousGeorge | last time i put a tdm in this tyan tomcat mb it fried. pci-bays are backwards compatible, right? |
| 19:56.50 | aixa | no SEGFAULTS yet |
| 19:56.58 | [TK]D-Fender | aixa: You're welcome..... what was it about again? :) |
| 19:57.25 | [TK]D-Fender | aixa: Oh yes, nesting Local dials... |
| 19:57.33 | aixa | [TK]D-Fender: yup the same |
| 19:57.35 | *** join/#asterisk DarKnesS_WolF (n=wolf@80.75.184.20) |
| 19:57.39 | aixa | not only local |
| 19:57.51 | aixa | but yes in Local |
| 19:59.12 | CoffeeIV_ | D-Fender -- sorry -- here it is: http://pastebin.ca/164587 |
| 19:59.14 | hmmhesays | go around a time or two just to waste my time with you |
| 19:59.31 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 20:01.05 | CunningPike | CoffeeIV_: Dude, where's your card? |
| 20:01.41 | CoffeeIV_ | it's listed in lspci |
| 20:01.57 | CunningPike | CoffeeIV_: I don't see it in /proc/interrupts, though...... |
| 20:02.23 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 20:02.40 | CoffeeIV_ | do I need an int=<something> argument to the modprobe ? What does a /proc/interrupts with a working card look like ? |
| 20:04.14 | CunningPike | CoffeeIV_: It should list wctNxxp somewhere |
| 20:04.28 | chode | Everyone, i'm looking to pay money to asterisk coder, details here: http://www.voipscout.net/sip_rtp_cdr_agi.html |
| 20:04.34 | CunningPike | CoffeeIV_: Are you running a 2.6 kernel? |
| 20:04.51 | CunningPike | CoffeeIV_: And if so, did you make the necessary changes to udev? |
| 20:06.13 | CoffeeIV_ | CunningPike: yes to both, I can pastebin uname -a and hte udev changes if you like |
| 20:06.55 | CunningPike | CoffeeIV_: And you rebooted after making the udev changes....? |
| 20:07.04 | CoffeeIV_ | several times |
| 20:09.00 | CunningPike | CoffeeIV_: Well, I would contact Digium then....... |
| 20:11.47 | CoffeeIV_ | I added the udev info and the uname -a: http://pastebin.ca/164610 |
| 20:13.34 | CoffeeIV_ | I remember back in the bad old days when I used asterisk@home it had a genzaptelconf command that would do all this for me . . . is that utility available outside of asterisk@home ? |
| 20:14.02 | hmmhesays | the last episode of star trek is on |
| 20:14.45 | file | CoffeeIV_: is the driver loaded? |
| 20:15.22 | CoffeeIV_ | yes -- I can zaptel and wct1xxp in lsmod |
| 20:15.41 | file | and dmesg shows it detected the card fine? |
| 20:16.54 | CoffeeIV_ | I think so -- the lines "Zapata Telephony Interface Registered on major 196" and "Zaptel Version: 1.2.8 Echo Canceller: KB1" appear in there |
| 20:17.20 | file | ummm |
| 20:17.39 | file | load wcte11xp |
| 20:19.28 | CoffeeIV_ | that gives an error -- "line 223: Cannot get number of tones for channel 1" and "line 223: Cannot init tones for channel 1" -- I tried it while the other module was loaded and also when it was unloaded |
| 20:19.51 | file | call Digium technical support |
| 20:20.26 | CoffeeIV_ | ok |
| 20:21.22 | file | installation support for the win! |
| 20:21.54 | C6Vette | IM getting seg faults and no core dumps. I just see "stdout: Broken pipe" then "Ouch ... error while writing audio data: : Broken pipe" then "Segmentation fault" |
| 20:21.54 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-9be55e089249b49c) |
| 20:22.00 | C6Vette | running version: Asterisk SVN-branch-1.2-r40901M |
| 20:22.04 | C6Vette | any ideas? |
| 20:23.51 | Zodiacal | anyone know how outlook email clients send to a contacts fax number? i.e. using asterfax, etc.. in outlook's new mail message, when i select the contact's fax item it puts the contacts name in the to: field. but what acctualy does outlook put in the to: field? cuz asterfax requires the # to just be digits and have a domain appended... i.e. 2390432@fax.local. is there a way to get outlook's contacts to append that @fax.local to the address? |
| 20:24.02 | Zodiacal | or is outlook expecting some kind of direct fax machine |
| 20:24.18 | Zodiacal | maybe i guess i have to create a second email address for the strange fax email address |
| 20:24.43 | teknoprep | i use a global folder |
| 20:24.51 | teknoprep | then give the global folder an email addy |
| 20:24.54 | teknoprep | in exchagne |
| 20:25.19 | Zodiacal | i guess i was hoping that the users woudn't have to type the fax # twice.. |
| 20:25.22 | Zodiacal | when seting up contacts |
| 20:29.39 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 20:30.29 | Jason99 | Does call-limit actually work? if I set call-limit=1 on a SIP peer would that prevent the customer from placing more then 1 call at a time? |
| 20:31.37 | *** join/#asterisk Vec (n=Vector@dsl-146-113-250.telkomadsl.co.za) |
| 20:39.23 | *** join/#asterisk ltd (n=z@202-161-28-106.dyn.iinet.net.au) |
| 20:40.40 | teknoprep | how long does digium usually take to send me those g729 codec keys? |
| 20:49.08 | *** join/#asterisk _Guhit (n=amistry@cpe-24-210-75-119.columbus.res.rr.com) |
| 20:54.47 | *** join/#asterisk adorah (n=Administ@84.94.122.156.cable.012.net.il) |
| 20:55.25 | _Guhit | I've got a weird problem. When I dial out from my asterisk box on the Zap/1 channel I still hear a ringing sound even if the other person answers the phone. This happens regardless of the 'r' option to Dial |
| 20:55.35 | jamincollins | anyone here familiar with the T203 and T200 counters? |
| 20:55.44 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 20:55.44 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:56.35 | jamincollins | specifically wondering if a T203 expiry would cause * to disconnect a PRI channel |
| 21:07.36 | teknoprep | whats the best way to have failover ... 2 identical asterisk boxen ... one dies.. box 2 picks up and has all voicemail and other settings as box 1 |
| 21:11.03 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-0e420fe7b9563827) |
| 21:11.03 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 21:11.47 | *** part/#asterisk mogorman (i=mogorman@nat/digium/x-0e420fe7b9563827) |
| 21:13.03 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:15.43 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 21:16.20 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 21:18.23 | CunningPike | Jason99: Try it!! :) |
| 21:18.32 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:18.55 | Jason99 | CunningPike: thanks for answering.. I did try it and it didnt seem to work |
| 21:19.19 | Jason99 | there was about 20 calls through the sip account, and I put a limit for 12 and calls could still come in |
| 21:20.16 | CunningPike | Jason99: Wait - hold the phone - calllimit in peers is to limit inbound calls to that peer, no? |
| 21:20.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 21:22.14 | Jason99 | i tested inbound.. I didnt test outbound |
| 21:22.18 | Jason99 | let me do some tests |
| 21:22.35 | CunningPike | Jason99: OK |
| 21:23.39 | *** join/#asterisk hax (n=hax@httpcraft/hax) |
| 21:23.41 | hax | hey guys |
| 21:23.53 | *** join/#asterisk trumee (n=trumee@cpc4-cmbg1-0-0-cust330.cmbg.cable.ntl.com) |
| 21:24.04 | hax | i have a question... is it okay (legal?) to record business conversations with VoIP? |
| 21:24.29 | hax | i think there's rules about that if you're using real phone lines, but i don't know if that applies to things like asterisk |
| 21:25.33 | trumee | guys, i cannot hear any sound on a sip call (using voiptalk.org). Although iax works. Can you guys give any suggestions? |
| 21:25.38 | teknoprep | well if they are calling you from an analoug line .. its pretty much the same thing |
| 21:26.35 | C6Vette | hax: in Arizona you can record any conversation IF one of the party is aware you are recording. |
| 21:27.12 | hax | C6Vette: well, from a business perspective... i could just encode every voip conversation to like 56kbps, and i'd never lose anything |
| 21:27.24 | hax | it'd take no space and probably save me all kinds of problems |
| 21:27.34 | hax | but i don't know if that's okay to do |
| 21:27.37 | hax | C6Vette: does asterisk make it easy to do? |
| 21:27.41 | trumee | On making a sip call i get in asterisk. Executing Dial("SIP/1234-ad5a", "SIP/voiptalk/00448003769036") in new stack; Called voiptalk/00448003769036; SIP/voiptalk-9ae7 is making progress passing it to SIP/1234-ad5a; SIP/voiptalk-9ae7 answered SIP/1234-ad5a; Attempting native bridge of SIP/1234-ad5a and SIP/voiptalk-9ae7 |
| 21:28.00 | trumee | Although, the connection is made. i dont hear any sound :( |
| 21:30.07 | trumee | my sip connection to voipstunt although works fine. I am wondering if this is a problem with voiptalk rather than asterisk setup. |
| 21:30.24 | trumee | does asterisk depend on alsa/oss? |
| 21:31.27 | CunningPike | hax: You need to check the laws in your jurisdiction - they usually don't discriminate between VOIP and POTS calls |
| 21:31.46 | hax | CunningPike: okay |
| 21:32.34 | hax | CunningPike: apparently my state requires one-party notification |
| 21:32.47 | hax | and i'm certainly one of the parties |
| 21:33.33 | CunningPike | hax: I would always notify the other party - most call centers do, if you think about it (your call is being recorded so we can can the ass of the agent if they mess it up) |
| 21:33.42 | hax | CunningPike: yeah |
| 21:34.46 | CunningPike | hax: But the intent of most of the laws is to allow people to record their own calls for their own records - they don't have to notify the other party. It's just courtesy to do so |
| 21:35.50 | hax | CunningPike: yeah, i see |
| 21:36.07 | hax | CunningPike: it's just for internal use anyway, i just want attach any phone conversations to my tickets |
| 21:36.14 | hax | CunningPike: so when shit asplodes, i'll know what i promised to do |
| 21:36.14 | hax | heh |
| 21:36.17 | CunningPike | hax: Yup |
| 21:37.15 | shodan | hunterjedispirit.ytmnd.com |
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| 21:50.29 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-145-124.dyn.embarqhsd.net) |
| 21:50.56 | syzygyBSD | how can i check what audio codec a current sip call is using? |
| 21:51.24 | C6Vette | sip show channels |
| 21:51.44 | De_Mon | I need to setup SIP over TCP/IP and currently debating between the TCP/IP patch or setting up openSER |
| 21:51.59 | syzygyBSD | thanks, now i feel stupid |
| 21:52.02 | De_Mon | any advice? |
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| 22:02.03 | *** join/#asterisk dannyman (n=djh@67.120.109.211) |
| 22:02.12 | dannyman | hello :) |
| 22:05.16 | tessier__ | Anyone know how one call tell from dialplan logic if there is already a call active on a particular device or extension? |
| 22:05.21 | tessier__ | We are trying to fix up the asterisk queueing stuff |
| 22:06.22 | teknoprep | anyone here have any problems trying to register g729 digium with asterisk? |
| 22:08.50 | mog | teknoprep, what seems to be the problem |
| 22:09.19 | bkruse | teknoprep: you there? |
| 22:10.31 | sx-wks | tessier__: chanisavail ? |
| 22:11.25 | sx-wks | tessier__: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail |
| 22:12.39 | teknoprep | nvm i was using the wrong versin |
| 22:12.59 | teknoprep | i thought trixbox would have used version 1.4 of asterisk.. it uses 1.2 |
| 22:13.45 | tessier__ | sx-wks: That won't do what I need. It just tells you if the phone can accept another call. |
| 22:14.05 | CunningPike | De_Mon: My inclination (although I have no direct experience, mind) would be OpenSER |
| 22:14.07 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 22:14.23 | CunningPike | De_Mon: Better than patching asterisk, imho |
| 22:14.25 | sx-wks | tessier__: hmmm... guess you were not clear |
| 22:14.26 | TripleFFFF | if io pass "" <> as callerid ? what will that do |
| 22:14.32 | tessier__ | sx-wks: My Snom 320 phones can accept a ton of calls. We are setting up a queue for a call center. It makes no sense that asterisks queueing system calls the operator who is already handling another call and puts a call waiting tone into their headset. |
| 22:14.44 | TripleFFFF | i mean is that bad |
| 22:14.51 | tessier__ | sx-wks: I am trying to figure out how this queue implementation works for anyone at all. |
| 22:15.08 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 22:15.11 | tessier__ | chanisavail checks to see if the phone has the capacity to handle more calls, not if it already has an active call |
| 22:15.11 | sx-wks | tessier__: hmmm |
| 22:15.12 | TripleFFFF | i mean will asterisk freak ? |
| 22:15.45 | sx-wks | tessier__: then you should have read that wiki page all the way to the end |
| 22:18.52 | tessier__ | sx-wks: Are you referring to: According to bug 4506 Chanisavail is not intended to detect if a phone is in use or not at all, it's only intended to check if asterisk could send the call there. |
| 22:19.25 | sx-wks | no... For telling if Sip peers are online or not, when you are using qualify, then you may wish to just use the SipPeer('name':status) function, and jump based on that. ChanIsAvail doesn't seem to tell you the difference between a Sip peer that's online, and one that's offline. |
| 22:21.33 | tessier__ | sx-wks: We tried SipPeer('name':status) and did not get the results we needed but we are trying it again now. |
| 22:21.55 | sx-wks | hah |
| 22:23.13 | De_Mon | CunningPike thats kinda where I was leaning too, one less patch to maintain |
| 22:23.53 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
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| 22:24.28 | CunningPike | De_Mon: Exactly - and openSER may yield more benefits in other areas (acting as a SIP proxy for your Asterisk PBX) |
| 22:24.51 | tessier__ | sx-wks: SipPeer('name':status) returns "OK (123 ms)" |
| 22:25.01 | tessier__ | sx-wks: That does not tell us if there is already a call on that line. |
| 22:25.06 | sx-wks | hmm |
| 22:25.11 | CunningPike | De_Mon: When our server begins to approach registration capacity , we may consider openSER instead of a second asterisk box |
| 22:25.16 | tessier__ | What is it supposed to return? |
| 22:25.36 | tessier__ | It would be nice if that wiki page specified whether it should return a boolean or a string or an enumerated list of possible values etc. |
| 22:25.55 | CunningPike | tessier__: Can you set callsperlinekey (or its equivalent) on Grandstream phonies? |
| 22:27.28 | tessier__ | CunningPike: I think callsperlinekey is a polycom thing isn't it? Not sure what it does. But we currently only get one call per line appearance. The problem is this phone has a lot of line appearances and the next one rings while the operator is handling a call from the queue. |
| 22:27.42 | sx-wks | tessier__: another idea would be to store busy status in the database ? |
| 22:28.46 | tessier__ | sx-wks: Possibly. We are looking into that as well. |
| 22:28.51 | CunningPike | tessier__: I see (and yes, it is a Polycom thing) - you must have more than one key per registration, then? |
| 22:29.24 | tessier__ | CunningPike: Not sure how polycom phones work but on our Snom 320's any available key can ring for any registration. |
| 22:29.50 | CunningPike | tessier__: Hmm - that's a bit inconvenient...... |
| 22:30.05 | TripleFFFF | _X.,1,DIAL(SIP/ |
| 22:30.08 | tessier__ | I think it is a reasonable way for the phone to behave. |
| 22:30.10 | TripleFFFF | shoudl match anything right ? |
| 22:30.23 | tessier__ | What is inconvenient is that the queue calls phones which are already handling a call |
| 22:30.33 | CunningPike | TripleFFFF: _X.,1,DIAL(SIP/S{EXTEN}) |
| 22:30.37 | TripleFFFF | yeah |
| 22:30.41 | TripleFFFF | i know |
| 22:30.43 | TripleFFFF | duh |
| 22:30.45 | CunningPike | ;) |
| 22:30.49 | TripleFFFF | the keyword was match |
| 22:30.56 | TripleFFFF | hence the part intereting awas the _x./ |
| 22:30.59 | tessier__ | TripleFFFF: Isn't it just _. ? |
| 22:30.59 | TripleFFFF | ;) |
| 22:31.17 | tessier__ | I think _X. might require two digis |
| 22:31.19 | tessier__ | digits |
| 22:31.25 | tessier__ | Maybe that is what you want... |
| 22:31.26 | CunningPike | tessier__: Not a good idea - it literally matches _anything_ - including t, i, etc |
| 22:31.40 | tessier__ | CunningPike: He asked to match _anything_...but maybe anything isn't what he really wanted. |
| 22:31.45 | CunningPike | _X. is one digit match |
| 22:31.53 | CunningPike | tessier__: Pedant ;) |
| 22:33.30 | wunderkin | tessier__: so you have a person on a phone taking a call, is it from the queue? and the same person is getting a call from the queue again? |
| 22:33.38 | tessier__ | wunderkin: Yes |
| 22:33.53 | tessier__ | And the call waiting tone goes off and another line appearance blinks and it is really annoying |
| 22:34.11 | wunderkin | tessier__: using chan_agent? |
| 22:34.23 | tessier__ | wunderkin: Yes. |
| 22:35.04 | wunderkin | tessier__: ringall strategy? |
| 22:35.22 | tessier__ | wunderkin: We are using roundrobin. |
| 22:36.08 | tessier__ | wunderkin: Head of CS says they will kill us if we ring all phones. |
| 22:37.09 | wunderkin | yes ringall sucks, i just don't see yet why you would be getting a call from the queue if the agent is already on a queue call and you are using chan_agent |
| 22:38.50 | wunderkin | the queue member in queues.conf is Agent/blah? |
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| 22:39.35 | wunderkin | it would be possible to get an incoming direct call but not another queue call if you are using chan_agent... |
| 22:40.20 | tessier__ | We are using AddQueueMember in the dialplan to put agents into the queue. |
| 22:40.30 | *** join/#asterisk vlt (n=daniel@dslb-088-073-198-145.pools.arcor-ip.net) |
| 22:40.32 | wunderkin | exactly |
| 22:40.34 | tessier__ | We are definitely getting multiple queue calls |
| 22:40.57 | wunderkin | AddQueueMember(blah|SIP/blah) right |
| 22:41.20 | tessier__ | exten => login,7,AddQueueMember(${j},Local/${agent}@agent_call); |
| 22:41.35 | wunderkin | oh, right, the new way |
| 22:41.44 | tessier__ | That's another problem |
| 22:41.54 | tessier__ | There are at least 3 different ways to put people into the queue |
| 22:42.03 | tessier__ | And none of them are clearly marked as to which are deprecated etc |
| 22:42.08 | tessier__ | So we have spent a lot of time trying bogus stuff |
| 22:42.09 | wunderkin | and in ${agent}@agent_call it is dialing a sip phone or something like that, which is basically the same thing |
| 22:42.22 | tessier__ | There is AddQueueMember, AgentCallbackLogin, and a way to do it through the manager API |
| 22:42.33 | wunderkin | hehe, agentcallbacklogin is going |
| 22:42.46 | wunderkin | you want people to be called back, not be logged in all of the time right |
| 22:42.54 | file | you're not using groups, so it doesn't reject the call |
| 22:42.58 | wunderkin | right |
| 22:43.11 | wunderkin | it is not marked because it is not being marked deprecated until 1.4 and gone on 1.6 |
| 22:43.17 | wunderkin | and 1.4 is not out yet :) |
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| 22:43.33 | tessier__ | file: We've looked into groups and can't figure out what the heck they are supposed to do. |
| 22:43.43 | wunderkin | tessier__: keep count of the number of calls |
| 22:43.48 | vlt | Hello. I successfully compiled asterisk for the first time (1.2.11) and now it's running but I can't connect with `asterisk -r`: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 22:43.49 | vlt | srwxr-xr-x 1 root root 0 2006-09-09 00:34 /var/run/asterisk.ctl |
| 22:44.33 | sevard | vlt: confirm that asterisk is in fact running (ps awuux | grep asterisk ) |
| 22:45.12 | tessier__ | wunderkin: I got this code from somewhere else, the wiki I think: |
| 22:45.15 | tessier__ | [queue-to-agent] |
| 22:45.15 | tessier__ | exten => _XXX,1,Set(GROUP()=${EXTEN}) |
| 22:45.15 | tessier__ | exten => _XXX,2,NoOP(Group count is ${GROUP_COUNT()}, group is ${GROUP()}, exten is ${EXTEN}) |
| 22:45.23 | vlt | root 13772 0.0 0.1 3708 840 pts/12 S 00:35 0:00 /bin/sh /usr/sbin/safe_asterisk -p -U asterisk |
| 22:45.23 | vlt | asterisk 13778 0.0 1.1 16924 7880 pts/12 Sl 00:35 0:00 /usr/sbin/asterisk -p -U asterisk -vvvg -c |
| 22:45.24 | tessier__ | So we call 123 at queue-to-agent |
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| 22:46.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:46.37 | vlt | I can watch its activity in /var/log/asterisk/messages |
| 22:47.14 | tessier__ | wunderkin: Someone else had the same problem. http://www.voip-info.org/wiki/view/Asterisk+Queue+with+limited+calls+per+IAX+agent |
| 22:47.27 | tessier__ | Exact same problem we have except he used IAX soft phones and he claims to have fixed it with groups |
| 22:48.54 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
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| 22:50.39 | tessier__ | wunderkin: -- Executing Set("Local/124@queue-to-agent-95af,2", "GROUP()=124") in new stack |
| 22:50.39 | tessier__ | <PROTECTED> |
| 22:50.58 | tessier__ | No matter how many calls we send into the queue GROUP_COUNT always stays 1 |
| 22:51.20 | tessier__ | And it keeps ringing phones where the operator is already handling a call |
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| 22:51.34 | wunderkin | hold on |
| 22:51.50 | file | that's because you're not putting /n at the end, so the Local channel (where the GROUP is actually set) is not there anymore because it gets optimized out |
| 22:52.09 | tessier__ | file: At the end of what? |
| 22:52.18 | file | so you have two options |
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| 22:54.12 | dannyman | heyas. |
| 22:54.29 | file | use /n at the end of the Local dial line that you give to AddQueueMember, or figure out OUTBOUND_GROUP that can be used with app_dial |
| 22:54.54 | dannyman | assuming you're a fairly clever SysAdmin, who has done the basics with telephony stuff, including per-set PBX admin, how much of a red pill is it to enter hte asterisk world? |
| 22:55.27 | dannyman | I'm thinking it would be neat if I could bring in a T1 from the telco and provide dialtone plus whatever other added services to my local handsets. |
| 22:55.49 | dannyman | But I'm not clear on how hard it ias to master asterisk, nor quite what hardware I'd need. |
| 22:56.25 | wunderkin | dannyman, you can learn easily enough if you play around at home |
| 22:57.10 | teknoprep | in the CLI whats the codec view thing that shows the - - - for passthrough and the numbers for codecs you can transcode |
| 22:57.16 | dannyman | So, I expense a one-port card to stick in some old hardware? :) |
| 22:57.45 | tessier__ | dannyman: It's a fair bit of a learning curve but in a few days you can set up something simple. It will take quite a while to become proficient enough to set up a company pbx though. |
| 22:57.54 | tessier__ | dannyman: I've been using asterisk for 2 years and still run into stupid problems. |
| 22:58.03 | tessier__ | Almost 3 actually |
| 22:58.05 | dannyman | how hard is it to get asterisk talking to the T1? |
| 22:58.11 | teknoprep | ahh .. show translation |
| 22:58.31 | tessier__ | dannyman: Harder than it should be. I recommend getting a Cisco box to do PRI and have a Cisco guy config it up for you and have * talk SIP to it. |
| 22:58.47 | tessier__ | dannyman: Avoid putting any PCI cards into your * box if you can help it. |
| 22:59.01 | tessier__ | They are all more complicated than they need to be and none of them have hardware dsp or any thing really. |
| 22:59.13 | teknoprep | T1 telephone or T1 inet? |
| 22:59.24 | tessier__ | I was assuming telephone. |
| 22:59.27 | dannyman | The old handbook draft I have found shows the * box connecting to a channel bank ... |
| 22:59.28 | teknoprep | i am just asking |
| 22:59.37 | dannyman | T1 telephone. |
| 22:59.49 | tessier__ | dannyman: You can connect to a channel bank with T1 if you need a bunch of FXO or FXS, sure. |
| 23:00.02 | dannyman | Or can we just purchase SIP through our data upstream these days? |
| 23:00.51 | tessier__ | file: Dial() takes /n as an argument? |
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| 23:01.06 | Qwell | tessier__: chan_local does |
| 23:01.07 | *** join/#asterisk dannyman (n=djh@67.120.109.211) |
| 23:01.15 | wunderkin | Local/blah/n |
| 23:01.57 | dannyman | oooh or it looks like i could recycle the existing analog lines instead of breaking out a T1, if I wanted to. |
| 23:03.52 | tessier__ | I am not currently using local/ anywhere. |
| 23:04.10 | tessier__ | I guess I need to route the call through that to get it into my queue or something? |
| 23:04.22 | wunderkin | [15:41] <tessier__> exten => login,7,AddQueueMember(${j},Local/${agent}@agent_call); |
| 23:04.31 | *** join/#asterisk Kylun (i=StarHawk@adsl-068-157-090-228.sip.bct.bellsouth.net) |
| 23:04.37 | tessier__ | Ah. We copied that from somewhere else. Ok. |
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| 23:05.02 | Kylun | Hola. Has anyone here ever setup asterisk w/ XO Communications XOptions Flex package? |
| 23:05.33 | wunderkin | well i mean i think that would go in the queues.conf? member => Local/blah/n i think |
| 23:05.46 | wunderkin | not on the exten |
| 23:06.13 | wunderkin | Kylun, no.. will they break out into a pri yet or is it still analog? |
| 23:06.27 | Kylun | wunderkin: thats what I was trying to figure out. |
| 23:06.29 | Kylun | :) |
| 23:06.47 | wunderkin | Kylun, has been a few months, i was told that they were going to offer pri soon... ask them :D |
| 23:06.49 | Kylun | the tech told me I could get a PRI card for the router they installed, but... |
| 23:07.28 | Kylun | I was actually wondering how hard it would be to just figure out what voip settings they were using, and get a T1 card for my asterisk box, and just recreate what they do.. |
| 23:07.37 | Kylun | bypassing the router entirely, basically. |
| 23:08.26 | wunderkin | shrug, they probably wouldn't be very happy about that, i dunno |
| 23:08.35 | Kylun | i doubt they would at all. :-/ |
| 23:08.43 | Kylun | cheaper though, wouldn't it be? :) |
| 23:09.20 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 23:09.28 | Dr-Linux | hi guys |
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| 23:10.07 | Kylun | wunderkin: what would i need to make PRI work for that then, just a T1 card in my * box? |
| 23:10.33 | Kylun | well, that and the pri card for the cisco.. |
| 23:10.51 | Dr-Linux | Kylun: Cisco pri card? :S |
| 23:11.01 | Kylun | the tech mentioned needing an expansion card. |
| 23:11.01 | wunderkin | yeah i guess so |
| 23:11.43 | marl | hi, can some one point me in the write direction for this problem? ive got asterisk dialing a mobile number and not bridging the call unless # is pressed, using dial(Zap/2c/MOBILE_NO,20,r) but im looking for some way to play a sound file to the called person? (like call from CALLERID_NO) ? |
| 23:11.49 | Dr-Linux | topic :P |
| 23:12.05 | Dr-Linux | looks like people are flooding file :) |
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| 23:20.29 | tessier__ | wunderkin: -- Called Local/124@drjays/n |
| 23:20.35 | tessier__ | wunderkin: It still rings the phone again. |
| 23:21.13 | file | are you doing group count checking? |
| 23:21.26 | tessier__ | file: We need \n AND group count checking? |
| 23:21.28 | file | is there logic present that looks at a group and says, "hey there's a call in progress - don't call this person" |
| 23:21.30 | file | yes |
| 23:22.12 | tessier__ | Ok, let me put my group count logic back in there |
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| 23:27.46 | tessier__ | <PROTECTED> |
| 23:27.50 | tessier__ | SWEET BABY JEBUS! |
| 23:27.53 | wunderkin | yey |
| 23:28.38 | tessier__ | It only two employees 18 man-hours to figure that out! Sweet! |
| 23:29.06 | wunderkin | meep |
| 23:29.54 | tessier__ | I need to ponder exactly what this local channel stuff is good for because I still don't grok it. But at least it works. |
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| 23:33.23 | SplasPood | Anyone bored and wanna get app_cepstral working with svn trunk? :) |
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| 23:45.28 | marl | can anyone point me to docs for the 'c' option in the dial command? eg. dial(Zap/1c/number) ? as i cant see anything on the viop-info/cmd+dial page |
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