| 00:01.19 | SpaceBass | whats the difference between using yes or a numeric value for echotraining? |
| 00:02.55 | lesouvage | srt: it's not on my system and apt-get install db_dump doesn't work. I sit part of a package with another name? |
| 00:04.31 | srt | running debian? |
| 00:04.49 | lesouvage | Synapes: what is the cli output when you set verbose to 20. (do asterisk -vvvvvvvvvvvvvvvvvvvvr on the linux prompt) |
| 00:05.12 | lesouvage | srt: yes, xorcom rapid. |
| 00:06.00 | srt | hm on debian stable its called db4.2_dump on sid db4.3_dump |
| 00:06.14 | srt | and its in db4.2-util or db4.3-util. |
| 00:19.13 | Synapes | lesouvage:Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 3192) |
| 00:19.14 | Synapes | Verbosity was 3 and is now 20 |
| 00:19.18 | Triple1243 | yoyoyo |
| 00:19.18 | *** part/#asterisk Triple1243 (n=Triple12@modemcable171.79-70-69.mc.videotron.ca) |
| 00:25.06 | Ariel_ | Hello everyone |
| 00:25.55 | jhiver | bye everybody |
| 00:25.59 | *** join/#asterisk mog_home (n=cherry@user-24-236-84-48.knology.net) |
| 00:26.08 | jhiver | bed is looking too good :) |
| 00:40.07 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 00:43.02 | *** part/#asterisk kilobit2001 (n=locid@206-248-159-174.dsl.teksavvy.com) |
| 00:49.05 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 00:52.52 | De_Mon | I've got asterisk setup with an incoming-outgoing broadvoice line. when sip.conf has supportvideo=yes enabled broadvoice freaks out. |
| 00:53.21 | De_Mon | freaks out as in, woln't let me call out |
| 00:53.49 | marcus2_ | so set supportvideo=no |
| 00:54.31 | De_Mon | I need video support for internal users still |
| 00:54.50 | marcus2_ | so turn it off for just the broadvoice block |
| 00:55.40 | De_Mon | it is |
| 00:55.50 | marcus2_ | and it still flips? |
| 00:57.08 | De_Mon | yup, thats why im here |
| 00:57.18 | marcus2_ | sucky |
| 00:57.20 | De_Mon | afaik supportvideo only works in [general] |
| 00:57.21 | marcus2_ | find a new provider ;) |
| 00:57.42 | De_Mon | fix asterisk~ |
| 00:58.15 | marcus2_ | it sounds to me like broadvoice is the broken thing here |
| 00:58.40 | marcus2_ | i mean, asterisk could be enhanced to support supportvoice= on a per-block basis, but thta would just be to work around broken providers like broadvoice ;) |
| 00:59.38 | De_Mon | all the phones I've used support video, so I can't be sure but if this problem exists with all non-video phones it's not broadvoices fault |
| 01:00.14 | De_Mon | with *other* non-video phones |
| 01:12.04 | De_Mon | http://lists.digium.com/pipermail/asterisk-users/2005-March/092324.html |
| 01:13.22 | [av]bani | marcus2_: if asterisk cant do settings per-block, then imo asterisk needs to be fixed |
| 01:16.10 | justinu | [av]bani: agreed |
| 01:17.57 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 01:18.44 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 01:18.48 | justinu | any of you guys know a good tech at covad? |
| 01:18.57 | justinu | i have a screwy dsl problem |
| 01:19.13 | marcus2_ | i dont disupute that its a good idea for asterisk to support those settings per-block, but that doesnt mean this particular issue is asterisk' fault |
| 01:19.16 | [av]bani | justinu: #covad |
| 01:19.23 | justinu | [av]bani: lol i checked |
| 01:19.34 | [av]bani | it just seems to be the default answer here |
| 01:19.35 | justinu | no #covad on freenode |
| 01:19.46 | [av]bani | anything non-* is "try #bla" |
| 01:20.04 | justinu | i've helped out a lot here, i have the right to veer offtopic now and then |
| 01:20.12 | [av]bani | no exceptions |
| 01:20.18 | [av]bani | nobody is above the law |
| 01:20.36 | *** join/#asterisk craigb_ (n=craig@69.64.3.1) |
| 01:21.31 | justinu | did the sbc/at&t behemoth swallow covad too? |
| 01:21.32 | craigb_ | wehn I forward my extension to my cell phone and don't answer, how can i set it to let the cell provider voicemail answer instead of Asterisk? |
| 01:22.18 | dudes | have it answer the channel then dial? |
| 01:22.24 | *** part/#asterisk sivana (n=sivana@mixdown.ca) |
| 01:22.35 | craigb_ | dudes, thats all i need? |
| 01:22.44 | dudes | Try it |
| 01:23.33 | craigb_ | well, i was hoping to use the forward button on my Poly 501, nnot sure what i'd do to accomplish that in the dialplan |
| 01:24.17 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 01:27.27 | [av]bani | forward it to another extension which does dialing then falls to vm? |
| 01:28.50 | [av]bani | why would you want to use cell provider vm though :) |
| 01:29.35 | craigb_ | i don't, customer does |
| 01:30.51 | [av]bani | yeah, answer then dial seems the obvious way |
| 01:32.18 | craigb_ | this system was built with Amp, not sure where exactly to do that in the dialplan |
| 01:33.18 | [av]bani | i think you'd setup a extension "CUSTOM" with a dial string |
| 01:33.37 | craigb_ | hmm, sounds reasonable |
| 01:33.39 | craigb_ | thanks |
| 01:34.12 | craigb_ | but not sure where to move the order of answer so that it only affects particular extensions |
| 01:34.59 | [av]bani | AMP isnt really good for aynthing beyond basic stuff :() |
| 01:35.35 | craigb_ | i know, but they insist on having it so they don't have to pay someone to make trivial changes |
| 01:36.03 | justinu | you should be charging triple time to fix it then |
| 01:36.14 | [av]bani | set em up with sipX :) |
| 01:36.52 | justinu | can sipX act as a media gateway? |
| 01:38.47 | cypromis | no |
| 01:38.51 | kink0 | there anyway to connect to Asterisk and send a ANSWER just for one know channel ? I do actually with external application, that connect to Asterisk and send just "answer", but where ALL concurrent calls are answered instead just the one I want to be answered. |
| 01:38.52 | cypromis | sipx is a pure proxy |
| 01:39.14 | cypromis | although the package contains a auto attendand media server and a voicemail media server |
| 01:39.31 | *** join/#asterisk chalco (n=chatzill@pdpc/supporter/active/chalco) |
| 01:39.40 | justinu | is there anything besides * for a tdm gateway? (pri specifically) |
| 01:39.44 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 01:40.46 | ManxPowe | kink0, If you issue an Answer, only that one channel will be answered. |
| 01:40.57 | kink0 | I need something like asterisk -r -x 'soft hangup $channel' but for answer |
| 01:41.03 | ManxPowe | justinu, Ascend/Lucent MAX TNT |
| 01:41.21 | ManxPowe | kink0, you can't do that. |
| 01:41.25 | ManxPowe | answer it in your dialplan |
| 01:41.28 | kink0 | ManxPowe, but I do that ussing asterisk -r -x , and then asterisk doesn't know what channel must answered |
| 01:42.10 | kink0 | ManxPowe, yes, putting in the dialplan works fine, but I need to still ringing until an external event, then send the answer |
| 01:42.23 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 01:42.23 | ManxPowe | If you need to do that outside of Asterisk then use the Manager Interface |
| 01:42.26 | kink0 | in the same manner I do soft hangup or so |
| 01:42.51 | justinu | i guess i meant anything open source |
| 01:42.52 | ManxPowe | kink0, softhangup is only there to manually shutdown a channel incase something goes wrong. |
| 01:43.10 | ManxPowe | justinu, There's Bayonne and YATE |
| 01:43.27 | kink0 | ManxPowe, yes, but runs fine also when I do asterisk -r-x for a softhangup a channel |
| 01:43.50 | ManxPowe | kink0, answer using -r -x is not supported. If you don't like that then write a patch. |
| 01:43.59 | ManxPowe | or use the Manager Interface. |
| 01:44.01 | kink0 | the problem is how to answer a defined call only |
| 01:44.15 | ManxPowe | kink0, you can do that using the manager interface. |
| 01:44.54 | kink0 | AGI ? well that is another problem, because I run the external app on background , starting when a call arrives and ending when a call send the BYE |
| 01:45.15 | ManxPowe | Noreaga, not AGI. AGI is something different. The Manager Interface is different. |
| 01:45.25 | kink0 | I also tryed EAGI, but not way to send it to the background and still runing while this channel is in use. |
| 01:45.27 | ManxPowe | The manager interface is designed to control asterisk from outside Asterisk. |
| 01:45.49 | ManxPowe | look on the Wiki |
| 01:45.51 | ManxPowe | ~docs |
| 01:45.56 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:45.56 | kink0 | ManxPowe, ahh ok, then I am going to re-read docs about Manage interface |
| 01:45.58 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 01:46.10 | justinu | ManxPowe: thx... apparently freeswitch is also working with PRI now too |
| 01:46.10 | kink0 | yess, I have always at hand :) |
| 01:47.33 | *** join/#asterisk pengyong (n=lala@222.185.17.83) |
| 01:48.17 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 01:48.30 | Darwin35 | ok 1.2.2 where is voicemail stored ? |
| 01:48.45 | Darwin35 | is it /var/spoool/asterisk |
| 01:49.13 | Darwin35 | or /usr/local/share/asterisk/sounds |
| 01:49.37 | file | ummm, /var/spool/asterisk/voicemail |
| 01:49.53 | rob0 | I couldn't afford that many o's in "spool", I had to skimp. :) |
| 01:49.55 | cypromis | ommmmmmmm |
| 01:50.00 | ManxPowe | in fact unless you REALLY screw up the build process nothing will be /usr/local |
| 01:50.44 | *** join/#asterisk kilobit2001 (n=locid@206-248-159-174.dsl.teksavvy.com) |
| 01:50.51 | kink0 | ManxPowe, as I understand, Manager Interface allows the same as console commands, and not something like answer(channel) |
| 01:51.04 | ManxPowe | kink0, You do not understand than. |
| 01:51.10 | Darwin35 | there is when you build it on bsd |
| 01:51.27 | dudes | kink0 - what are you trying todo? Hangup a channel? |
| 01:51.27 | kilobit2001 | in cdr_mysql, is it normal to have every single action in extensions recorded? |
| 01:51.44 | kink0 | dudes answer a defined channel only. |
| 01:51.44 | kilobit2001 | i get one cdr row, for each row in extensions. |
| 01:52.13 | dudes | kink0 - answer it in the dialplan and use redirect/bridge to connect it to a channel |
| 01:52.20 | ManxPowe | kink0, the manager interface allows you to transfer calls, answer calls, redirect calls, change caller id. |
| 01:53.06 | ManxPowe | kink0, read manager.txt in the asterisk/docs directory |
| 01:53.12 | kink0 | ManxPowe, i did a show manager commands and as long as I saw, appear to be more restricted. |
| 01:53.21 | kink0 | ok, going ... |
| 01:53.35 | dudes | doQueueRedirect((*it)["channel"],tempagent,(*it)["campaign"],(*it)["leadid"]); |
| 01:53.56 | ManxPowe | <PROTECTED> |
| 01:54.05 | ManxPowe | there is the manager command you need. |
| 01:54.17 | ManxPowe | the Manager command Command allows you to run any command you want |
| 01:54.18 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:54.28 | ManxPowe | dudes, he wants to do it the hard way. |
| 01:54.39 | dudes | ah, hehe |
| 01:55.05 | kink0 | dudes yes, because I need to do it based on external event |
| 01:55.17 | JunK-Y | hey dudes! |
| 01:55.42 | Darwin35 | what manager interface |
| 01:55.52 | ManxPowe | kink0, The manager interface was DESIGNED to allow external events to control Asterisk |
| 01:55.57 | dudes | kink0 - then have a open listener socket to * and parse it |
| 01:56.00 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 01:56.16 | JunK-Y | Darwin35: AMI ? |
| 01:56.29 | ManxPowe | search for "FOP" or "Flash Operator Panel". It allows you to transfer calls from a Flash application running in a web browser. |
| 01:56.33 | kink0 | dudes: yes that was the way I was doing, but ussing asterisk -r -x instead AMI |
| 01:56.53 | dudes | parsing results from a asterisk -rx |
| 01:57.32 | kink0 | dudes: yes, but not problem to open a sock to the AMI port instead to use asterisk -rx |
| 01:57.45 | ManxPowe | kink0, use the Redirect manager command to send the call to an extension that runs Answer when your event happens. |
| 01:58.19 | chalco | rob0, mind a pm? |
| 01:58.37 | kink0 | ManxPowe, just that I was going to ask you, if use Redirect to an specific answer extension, since there not any "answer" command |
| 01:58.48 | ManxPowe | kink0, try it. |
| 01:59.01 | De_Mon | theres not any answer command? |
| 01:59.07 | kink0 | ok, I will try so. |
| 01:59.10 | ManxPowe | De_Mon, not on the CLI or the AMI |
| 01:59.28 | De_Mon | oh, i thought you were talking manager interface |
| 01:59.31 | kink0 | De_Mon, I need to answer a defined channel where more concurrents call are |
| 02:00.37 | ManxPowe | De_Mon, If you can find a Manager command to answer a call..... |
| 02:00.58 | rob0 | chalco: sure, go ahead |
| 02:01.07 | *** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar) |
| 02:01.33 | Math[laptop] | De_Mon: you could work something out with a little coding, look at app_pickup and implement a manager interface command |
| 02:02.19 | De_Mon | I added a phrase-recorder to my dialplan, am I better off leaving it in extensions.conf or #invluding a separate config? |
| 02:02.30 | De_Mon | #including |
| 02:07.01 | Tozaz2 | Hi, is it thrue that if I want tu set up asterisk for more than 20 users, I need a dual CPU and a lot off RAM ? thks |
| 02:07.58 | rob0 | file: I forgot the name of the company you work for :( |
| 02:10.56 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 02:12.13 | ManxPowe | But I'm on IRC chatting while I rip video from a DVD to upload to a TiVo so I obviously don't have a life either. |
| 02:12.14 | drumkilla | ManxPowe: sounds like a good idea to me |
| 02:14.15 | justinu | porn, no less |
| 02:19.42 | De_Mon | O_o |
| 02:23.24 | konfuzed | Tozaz2: no its not true |
| 02:24.27 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 02:24.42 | kink0 | ManxPowe, I got Message: Permission denied when I send Redirect ussing AMI |
| 02:25.23 | kink0 | I try it ussing a telnet to the port , to try before to add the code to my C external application |
| 02:26.30 | konfuzed | hey any one here in Toronto |
| 02:27.18 | konfuzed | may be on monday for $150 bucks ? |
| 02:28.19 | chalco | if I get to Toronto, can I have $150? US or Canadian? |
| 02:28.25 | Qwell | cdn |
| 02:29.40 | Darwin35 | manx make rooom i am moving in |
| 02:29.49 | Darwin35 | we can get a t1 |
| 02:30.06 | chalco | tell you what, we'll skip the meeting entirely, just wire me the $150 |
| 02:30.29 | dudes | paypal would be cheaper |
| 02:30.43 | Qwell | a check would be cheaper than that |
| 02:30.44 | chalco | even better |
| 02:31.30 | dudes | Yea I suppose PayPal does have their gay little fees |
| 02:32.12 | dudes | Amount: $2,078.40 USD Fee: -60.57 USD |
| 02:32.28 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 02:32.28 | Qwell | wtf did you get a paypal for $2k? :p |
| 02:32.40 | dudes | heh |
| 02:32.44 | Qwell | at that high, I'd ask for another form of payment, heh |
| 02:33.05 | dudes | a wire would have been cheaper |
| 02:33.16 | dudes | but only by 10.00 |
| 02:33.24 | dudes | maybe 15 |
| 02:33.51 | chalco | I don't care really - it's free money! or, maybe they wanted something in return.... that'd be inconvenient |
| 02:34.00 | dudes | nice |
| 02:34.08 | konfuzed | chalco: not quite free money |
| 02:34.21 | chalco | darn |
| 02:34.21 | konfuzed | 8:30 am to 9:30 pm |
| 02:34.27 | konfuzed | and in Canadian dollars |
| 02:34.46 | dudes | you want 13hr's for 150 CA ? |
| 02:35.02 | justinu | lol |
| 02:35.09 | konfuzed | well I think it may add up to $200 for sitting in a chair for 13 hrs |
| 02:35.21 | justinu | not worth it |
| 02:35.31 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 02:36.18 | konfuzed | there is a risk of falling asleep on the job so you have to bring your own caffiene |
| 02:36.30 | dudes | Or pills? |
| 02:36.32 | dudes | hehe |
| 02:36.48 | konfuzed | im bringing 5 ounds of chocolate |
| 02:37.15 | chalco | don't keep us in suspense... what's the job? |
| 02:37.36 | konfuzed | well its really easy money and id rather it went to someone i new or at least knows something technical |
| 02:37.54 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 02:38.08 | konfuzed | the job is to scratch names off the voter list and watch as I hand people their ballot and then at the end watch as I add up the ballots |
| 02:38.09 | SpaceBass | anyone having audio problems with BroadVoice tonight? |
| 02:38.31 | konfuzed | i know a little off topic |
| 02:38.44 | konfuzed | but easy cash is almost never off topic ;^) |
| 02:38.45 | *** join/#asterisk Cheetah (n=Akia@62.217.48.108) |
| 02:39.03 | konfuzed | Cheetah: are you from montreal or toronto? |
| 02:39.19 | Cheetah | neither |
| 02:39.33 | konfuzed | oh thought I recall you mention montreal |
| 02:39.48 | konfuzed | nvr mind then |
| 02:39.55 | dudes | is 150 CA like 110ish USD? |
| 02:40.06 | Qwell | $1.10USD |
| 02:40.10 | Cheetah | asterisk rocks :D |
| 02:40.58 | chalco | too bad I have to work Monday and I'm 9.5 hrs from Toronto |
| 02:41.03 | eieiyo | anybody have experience with 2 stage dialing plans with asterisk? |
| 02:41.32 | dudes | 2 stage? |
| 02:41.44 | dudes | Maybe I'm not up on the * lingo |
| 02:41.54 | konfuzed | dudes: you also have to be a canadian citizen but I understand that its not to hard to fake it. |
| 02:42.02 | SpaceBass | what is the difference b/t a numeric value and "yes" when using echo training ? |
| 02:42.05 | Qwell | just say "eh" a lot |
| 02:42.06 | justinu | eh? |
| 02:42.12 | chalco | lol |
| 02:42.15 | dudes | heh |
| 02:42.15 | konfuzed | just say Eh a lot eh! |
| 02:42.19 | eieiyo | i want to call in on free world dialup and then give myself another dialtone to be able to access pstn |
| 02:42.28 | kink0 | why from AMI action: listcommands only lists those commands that priv = none ? |
| 02:42.30 | konfuzed | justinu 's got it |
| 02:42.31 | eieiyo | so its sort of 2 stage i guess that is the appropriate terminology |
| 02:42.32 | dudes | I just don't get what this is all aboot. |
| 02:42.32 | chalco | I watch the Red Green Show |
| 02:42.56 | dudes | eieiyo - that's easy |
| 02:43.05 | SpaceBass | eieiyo: look up DISA |
| 02:43.06 | kink0 | I have for my user read: system, call ... write: systema, call in manager.conf |
| 02:43.13 | konfuzed | I watch the Corbert Report its very canadian |
| 02:43.16 | eieiyo | can you point me in the right direction? |
| 02:43.30 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 02:43.34 | SpaceBass | eieiyo: google :) sorry... |
| 02:43.53 | eieiyo | direct inward system access? |
| 02:44.00 | SpaceBass | yeah, thats it |
| 02:44.01 | dudes | just make a transfer context ... then dial |
| 02:44.06 | eieiyo | k, thanks guys |
| 02:44.28 | eieiyo | oh cool :) should not be too bad |
| 02:44.33 | dudes | nope |
| 02:45.07 | eieiyo | disa can give me some password security right? sort of like a password login so only authorized people could dial out? |
| 02:45.16 | Qwell | eieiyo: yeah |
| 02:45.18 | SpaceBass | yeah |
| 02:45.19 | *** join/#asterisk FastJack (i=fastjack@p5091E26A.dip.t-dialin.net) |
| 02:45.25 | eieiyo | ok thanks everybody... |
| 02:45.26 | Qwell | show application disa |
| 02:45.28 | konfuzed | anyway i just hope to avoid spending the day with Pat from saturday night live or something |
| 02:45.49 | SpaceBass | so, I upgraded from a PII 300mhz with 256mb ram to a nice 1.25ghz with 512mb and my jitter is out of control |
| 02:45.52 | SpaceBass | or echo... |
| 02:46.02 | Qwell | SpaceBass: asterisk requires a slow machine |
| 02:46.04 | SpaceBass | when someone calls into my zaptel lines I hear myself and its REALLY bad |
| 02:46.07 | justinu | lol |
| 02:46.09 | SpaceBass | lo |
| 02:46.10 | SpaceBass | lol |
| 02:46.21 | SpaceBass | seriously... it worked better on that old clunker |
| 02:46.21 | dudes | play with your rx/txgain |
| 02:46.29 | justinu | SpaceBass: irq conflict? |
| 02:46.40 | *** join/#asterisk anonymouz666 (n=lynx@ns2.redetaho.com.br) |
| 02:46.44 | SpaceBass | that was my next question... why would the gain effect echo... seems like that would just amplify one end or the other |
| 02:46.50 | SpaceBass | not doubting, just curious |
| 02:46.59 | Qwell | SpaceBass: which is exactly what causes echo sometimes |
| 02:47.07 | dudes | I know if I have it too high it echos like a mofo |
| 02:47.11 | anonymouz666 | Does anyone in here know a good IAX client? |
| 02:47.13 | justinu | dudes: true dat |
| 02:47.15 | SpaceBass | Iaxcomm |
| 02:47.22 | Qwell | idefisk |
| 02:47.26 | anonymouz666 | Free? |
| 02:47.30 | SpaceBass | if I'm hearing myself, would that be RX or TX? or is it hit or miss and I have to play |
| 02:47.31 | Qwell | free |
| 02:47.38 | justinu | SpaceBass: if the gain is too high, you're throwing out too much energy into the network and some of it reflects back |
| 02:47.45 | SpaceBass | anonymouz666: Iaxcomm |
| 02:47.52 | SpaceBass | justinu: ahhh gotcha |
| 02:47.53 | Qwell | idefisk! |
| 02:47.55 | justinu | spacebase: if you're hearing yourself, the echo is coming from the equipment on the PSTN side |
| 02:48.32 | SpaceBass | thats my first problem... the echo... the next is that BroadVoice is not having audio issues...not sure if its them or me, but when the call connects neither party can hear the other one |
| 02:48.45 | *** join/#asterisk a1fa||64 (n=a1fa@24.144.51.70) |
| 02:48.46 | a1fa||64 | sup |
| 02:48.49 | a1fa||64 | g00d people :P |
| 02:48.52 | dudes | it's f'ing Saturday and the only parties in this god for saken town is either a weenie roast or a teeny party. |
| 02:48.55 | justinu | SpaceBass: reinvite problems? |
| 02:48.58 | SpaceBass | justinu: so would 'turning down' my TX cause it to send less and thus echo less? |
| 02:49.09 | justinu | dudes: teeny party, definitely |
| 02:49.12 | a1fa||64 | dudes: nothing wrong with teen parties ;) ... as long as they are 18 |
| 02:49.16 | justinu | SpaceBass: yes, probably. |
| 02:49.18 | SpaceBass | justinu: thought that as well... so I changed canreinvite from NO to YES... no difference |
| 02:49.23 | Qwell | a1fa||64: or look 18 |
| 02:49.32 | dudes | the only 18+ party is a sausage fest |
| 02:49.33 | anonymouz666 | idefisk does not have the source code |
| 02:49.44 | Qwell | anonymouz666: so? Do you plan on making changes? |
| 02:50.03 | anonymouz666 | I want a IAX web client |
| 02:50.09 | SpaceBass | web client? |
| 02:50.11 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 02:50.13 | a1fa||64 | Qwell: officer, i swear.. she told me she was 18 |
| 02:50.15 | SpaceBass | interesting idea... dont know of any |
| 02:50.15 | a1fa||64 | :P |
| 02:50.20 | Qwell | a1fa||64: exactly |
| 02:50.22 | a1fa||64 | i even id her |
| 02:50.24 | anonymouz666 | web client |
| 02:50.30 | a1fa||64 | she must of had a fake |
| 02:50.34 | a1fa||64 | i dont have a id reader |
| 02:50.34 | a1fa||64 | :P |
| 02:50.50 | a1fa||64 | how many people updated to 1.2.2? |
| 02:50.52 | SpaceBass | justinu: I assume I can set my TX to a negitive number |
| 02:50.56 | Qwell | 27 people |
| 02:50.58 | SpaceBass | is that correct? |
| 02:51.01 | justinu | SpaceBass: yes |
| 02:51.05 | a1fa||64 | Qwell: lol 27 Only? |
| 02:51.10 | SpaceBass | Qwell: thought it was now 26...one regressed |
| 02:51.11 | justinu | now you're adjusting attenuation, not gain |
| 02:51.13 | a1fa||64 | Qwell: the questions is.. did you update? |
| 02:51.22 | Qwell | a1fa||64: no, I use trunk |
| 02:51.29 | a1fa||64 | lol |
| 02:51.31 | a1fa||64 | trunk what? |
| 02:51.35 | Qwell | but, 1.2.2 is worth it |
| 02:51.37 | Qwell | svn trunk |
| 02:51.40 | a1fa||64 | svn trunk |
| 02:51.40 | a1fa||64 | ok |
| 02:51.44 | justinu | all the cool kids run trunk |
| 02:51.52 | Qwell | s/cool/crazy/ |
| 02:51.54 | a1fa||64 | did u try that voice changer? |
| 02:52.11 | a1fa||64 | i want to patch it w/ voice changer |
| 02:52.15 | Qwell | a1fa||64: sure |
| 02:52.21 | a1fa||64 | Qwell: liar ;P |
| 02:52.26 | kilobit2001 | whats a good voip wifi phone? |
| 02:52.27 | Qwell | I've used it |
| 02:52.37 | a1fa||64 | will it compile on 1.2.2? |
| 02:52.42 | Qwell | should |
| 02:52.47 | SpaceBass | kilobit2001: I would have said the hitachi but I'm having problems altely |
| 02:52.50 | SpaceBass | lately |
| 02:52.58 | anonymouz666 | I wonder if someone use chan_ss7 in here :) |
| 02:53.04 | SpaceBass | speaking of which, anyone have a hitachi ip5000 and have the latest firmware? |
| 02:53.16 | Qwell | SpaceBass: no, but if you sent one this way, I'd give it a try |
| 02:53.21 | SpaceBass | lol |
| 02:53.32 | a1fa||64 | Qwell: so u used it? |
| 02:53.33 | Qwell | what? |
| 02:53.34 | kilobit2001 | how about UTStarcom F1000 |
| 02:53.36 | Qwell | a1fa||64: yes |
| 02:53.56 | SpaceBass | I do have 2 ipicassa phones that won't fully boot.... I'd gladly send one to someone to play with in exchange for a working config if you figure it out |
| 02:54.13 | SpaceBass | kilobit2001: I think thats the same as the prestige or what ever... I have one and it sucks |
| 02:54.14 | Qwell | ipicassa? |
| 02:54.29 | justinu | yeah, the prestige sucks |
| 02:54.37 | a1fa||64 | Qwell: how do you update svn.. svn update ? |
| 02:54.41 | Qwell | yes |
| 02:54.55 | Qwell | You don't want to run trunk in production...that's just dumb |
| 02:54.56 | SpaceBass | http://www.iridia.com/ipicasso.html |
| 02:54.58 | kilobit2001 | spacebass: whats exactly is the problem with it? |
| 02:55.12 | SpaceBass | kilobit2001: leme look at the starcom and see if its the same |
| 02:55.45 | Qwell | usb? wtf does the phone have usb? |
| 02:55.48 | kink0 | there any way to set a variable from CLI ? |
| 02:56.06 | Qwell | kink0: no, you need to associate it with a channel |
| 02:56.07 | a1fa||64 | vn update |
| 02:56.08 | a1fa||64 | At revision 8423. |
| 02:56.08 | SpaceBass | Qwell: not sure that page is accurate... there is no USB port as far as I can tell |
| 02:56.12 | a1fa||64 | is this correct? |
| 02:56.27 | SpaceBass | kilobit2001: that is newer than mine... my problem was two fold, the network support and no call waiting |
| 02:56.39 | kink0 | Qwell, ussing queue then ? |
| 02:56.56 | SpaceBass | kilobit2001: as far as I know no Wifi phone on the market supports WPA, so I ended up with a 2nd wifi subnet here at home just for my 2 wifi phones |
| 02:57.21 | Qwell | is this SIP or what? It doesn't say |
| 02:57.24 | SpaceBass | I wasn't willing to compermise my security down to WEP just for the phones.... |
| 02:57.35 | a1fa||64 | Qwell: svn update ?? what? |
| 02:57.42 | Qwell | a1fa||64: sure |
| 02:57.57 | kilobit2001 | spacebass: network support? |
| 02:58.05 | justinu | SpaceBass: lol, that sucks |
| 02:58.06 | SpaceBass | Qwell: the company that made it went under... google had a cache of their site for a while and it suggested that it has sip support... i got them on e-bay and they boot and say something like "waiting for softare...." |
| 02:58.16 | Qwell | oh... |
| 02:58.32 | a1fa||64 | Qwell: svn update -r 8423 |
| 02:58.34 | a1fa||64 | ? |
| 02:58.37 | Qwell | I might be able to get one working |
| 02:58.39 | Qwell | a1fa||64: just svn up |
| 02:58.52 | Qwell | if you don't know what you're doing...don't use trunk |
| 02:58.56 | SpaceBass | kilobit2001: i heard a rumor about some new wifi phone... might have been uniden or something that had G and WPA... looked slick as snot, but I'm not sure its on the market |
| 02:59.06 | *** join/#asterisk _blop (i=blop@openbeer.be) |
| 02:59.23 | a1fa||64 | WPA is evil :P |
| 02:59.49 | justinu | there's a rumored linksys wifi phone with all sots of features, including a color screen |
| 02:59.57 | SpaceBass | yeah, i prefer just hiding the SSID and maybe mac filtering... that is safe enough :) |
| 03:00.02 | a1fa||64 | justinu: can you change the background :p |
| 03:00.15 | justinu | probably |
| 03:00.22 | a1fa||64 | w00t |
| 03:00.23 | a1fa||64 | :( |
| 03:00.24 | justinu | cornflower blue |
| 03:00.35 | Qwell | ssid's can be sniffed, mac's can be spoofed |
| 03:00.43 | a1fa||64 | anybody running vwmare and asterisk willing to test |
| 03:00.45 | SpaceBass | I need a sip client (or iax) for my pocketpc |
| 03:00.50 | SpaceBass | been lazy on researching it |
| 03:00.54 | Qwell | sjphone |
| 03:01.01 | SpaceBass | a1fa||64: I ran AAH in vemware for a while |
| 03:01.07 | SpaceBass | sjphone for the PPC? |
| 03:01.21 | a1fa||64 | i need to test that voice changer with 1.2.2 |
| 03:01.23 | a1fa||64 | :P |
| 03:01.29 | a1fa||64 | i dont want to jump the code ;P |
| 03:01.30 | *** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net) |
| 03:01.35 | SpaceBass | b/c I did try stinky x-lite ppc and it sucked ass |
| 03:02.07 | SpaceBass | so, whats the best sub $50 sip phone? |
| 03:02.23 | justinu | lol |
| 03:02.43 | SpaceBass | i know... i know... |
| 03:03.01 | a1fa||64 | err |
| 03:03.19 | erickj_az | I ned som help with CDR. When an inbound call with no caller id comes in Asterisk records the last valid caller id in the record, even if the call is not from the same source. Any ideas? |
| 03:03.20 | SpaceBass | remodeled (and by that I mean built) a master bath and need a phone for it |
| 03:03.26 | SpaceBass | dont want to spend more than $50 |
| 03:03.37 | SpaceBass | kilobit2001: I hear good things about the cisco wifi phone |
| 03:03.38 | *** join/#asterisk niZon (n=ilt@S0106deadbeefbeef.wp.shawcable.net) |
| 03:03.58 | kilobit2001 | $$ |
| 03:04.04 | Qwell | only like $500 |
| 03:04.06 | a1fa||64 | SpaceBass: i bought those BudgetPhones |
| 03:04.09 | a1fa||64 | they are $50 |
| 03:04.11 | a1fa||64 | but they are crap |
| 03:04.12 | a1fa||64 | :P |
| 03:04.22 | SpaceBass | kilobit2001: ebay :) |
| 03:04.25 | a1fa||64 | <PROTECTED> |
| 03:04.27 | a1fa||64 | omfg |
| 03:04.28 | a1fa||64 | <PROTECTED> |
| 03:04.38 | a1fa||64 | not compatible with 1.2.2.. why wasnt it compiled |
| 03:04.39 | a1fa||64 | grr |
| 03:04.40 | SpaceBass | a1fa||64: crap how? I just want something that A) rings and B) makes calls |
| 03:04.46 | a1fa||64 | Budgetphone 101 |
| 03:04.49 | a1fa||64 | its $50 |
| 03:04.54 | SpaceBass | this is for a bathroom shitter... i mean, I don't need an LCD or anything |
| 03:05.03 | justinu | iaxy? |
| 03:05.09 | a1fa||64 | asterisk -V |
| 03:05.09 | a1fa||64 | Asterisk SVN-tag-1.2.1-r7367 |
| 03:05.14 | a1fa||64 | interesting |
| 03:05.19 | a1fa||64 | i just updated |
| 03:05.28 | *** join/#asterisk jef_ (i=fischer@p548457D8.dip.t-dialin.net) |
| 03:05.28 | Qwell | you need to switch tags |
| 03:05.49 | a1fa||64 | how? |
| 03:05.54 | a1fa||64 | manually? |
| 03:05.54 | Qwell | svn sw |
| 03:06.08 | SpaceBass | kilobit2001: this is the one i heard about: http://www.voipsupply.com/product_info.php?products_id=1067 |
| 03:06.20 | a1fa||64 | http://svn.digium.com/svn/asterisk/tags/1.2.2 |
| 03:07.30 | a1fa||64 | Qwell: should i try to compile the voice changer ;P |
| 03:07.44 | Qwell | sure, if you want it |
| 03:07.52 | pauldy | hrm voice mods in asterisk |
| 03:07.58 | a1fa||64 | it was designed for 1.2.0 |
| 03:08.00 | pauldy | I'm might just ream my shorts |
| 03:08.01 | a1fa||64 | i wonder if it will patch 1.2.2 |
| 03:08.14 | kilobit2001 | spaceboss: far too expensive |
| 03:08.27 | Qwell | a1fa||64: try it |
| 03:09.40 | *** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
| 03:09.47 | a1fa||64 | cp asterisk asterisk2 :P |
| 03:09.54 | a1fa||64 | just in case |
| 03:11.11 | a1fa||64 | its patching :P |
| 03:11.25 | SpaceBass | justinu: when I inquired bout my broadvoice problem (no audio) you suggested a reinvite issue... i tried yes and no... are there any other options? |
| 03:11.31 | SpaceBass | i didn;t see any in the docs |
| 03:12.54 | a1fa||64 | it worked :P |
| 03:13.22 | justinu | you looked at the SDP yet? |
| 03:13.42 | a1fa||64 | me? |
| 03:14.05 | justinu | no, spacebass |
| 03:14.28 | SpaceBass | thinking its a broadvoice issue at thsi point |
| 03:14.56 | a1fa||64 | broadvoice sucks |
| 03:15.00 | a1fa||64 | even though i use them |
| 03:15.06 | a1fa||64 | i get so much problems with them daily |
| 03:15.08 | SpaceBass | I love em |
| 03:15.09 | a1fa||64 | i gave up on them |
| 03:15.12 | a1fa||64 | they are cheap |
| 03:15.15 | SpaceBass | but the occassional problme is a pisser |
| 03:15.15 | a1fa||64 | great control pannel |
| 03:15.25 | a1fa||64 | but the voice quality and bandwidth are terrible |
| 03:15.28 | SpaceBass | in the process of porting my POTS number to them |
| 03:15.43 | a1fa||64 | wow |
| 03:15.45 | a1fa||64 | good luck |
| 03:15.45 | a1fa||64 | :P |
| 03:15.48 | justinu | heh |
| 03:15.53 | SpaceBass | I've got a 30mbs/10mbs connection... latency sucks but bandwidth isn't an issue |
| 03:16.03 | justinu | how's the latency? |
| 03:16.04 | SpaceBass | a1fa||64: yeah... a little worried now |
| 03:16.15 | a1fa||64 | i get 30ms with them |
| 03:16.30 | a1fa||64 | <PROTECTED> |
| 03:16.34 | a1fa||64 | <PROTECTED> |
| 03:16.40 | justinu | 30ms is decent |
| 03:16.43 | a1fa||64 | grrr. wtf.. did you guys get the same error? |
| 03:16.50 | *** join/#asterisk Cleyverson (n=cleyvers@201.29.182.20) |
| 03:16.55 | a1fa||64 | <PROTECTED> |
| 03:16.58 | a1fa||64 | contains old modules |
| 03:17.03 | a1fa||64 | i wonder if i should wipe them out manually |
| 03:17.29 | SpaceBass | i just averaged 15ms |
| 03:17.53 | SpaceBass | nope... 30ms too... |
| 03:17.57 | justinu | 15ms is awesome |
| 03:18.01 | justinu | 30ms is very good |
| 03:18.28 | tronix | what about 60ms? |
| 03:18.40 | tronix | (that's the latency for my upcoming dsl or cable modem) |
| 03:18.50 | justinu | 60 is good |
| 03:19.21 | SpaceBass | my FIOS connection is supposidly high latency |
| 03:19.35 | SpaceBass | I notice it when surfing but not for UDP or direct connections |
| 03:19.45 | justinu | maybe that's a shitty DNS |
| 03:20.26 | justinu | is it cacheing? |
| 03:20.47 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 03:20.49 | a1fa||64 | this is not working right |
| 03:20.50 | a1fa||64 | ;( |
| 03:20.56 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 03:20.58 | SpaceBass | was cacheing with my firewall for a while but not now |
| 03:21.06 | kll | tronix: what do you have to just the other end of that dsl connection? |
| 03:21.09 | a1fa||64 | <PROTECTED> |
| 03:21.10 | a1fa||64 | lol |
| 03:21.11 | a1fa||64 | its there |
| 03:21.38 | SpaceBass | re: my echo issue.... I changed txgain to =-1.0 and I still have it... call starts normally but echo gets wrose as call progresses |
| 03:21.47 | tronix | kll: not sure I understand your question -- can you rephrase? |
| 03:21.52 | a1fa||64 | Qwell: how can i check if its compiled right? |
| 03:22.24 | Qwell | by loading it |
| 03:22.28 | alexhopper | You spellec "voicechanger" wrong |
| 03:22.36 | kll | tronix: you were talking about 60 ms latency and I wondered if this was end-to-end or just from your dsl cpe to the other end of the dsl connection |
| 03:22.39 | alexhopper | *Spelled |
| 03:22.46 | tronix | kll: ahh-ha, gotcha. end-to-end |
| 03:23.00 | tronix | kll: time warner does some silly exit routing around here. |
| 03:23.04 | Cleyverson | hello everbody |
| 03:23.23 | Cleyverson | how does it mean? => Jan 22 01:20:40 WARNING[6746]: chan_iax2.c:6671 build_user: Unable to support trunking on user 'AsteriskB' without zaptel timing |
| 03:23.23 | Cleyverson | ?? |
| 03:23.24 | SpaceBass | justinu: does it make sense that if my TX was too high that the echo would worsen as the call progressed? |
| 03:23.36 | Qwell | Cleyverson: It means what it says |
| 03:23.41 | justinu | SpaceBass: not really |
| 03:23.41 | a1fa||64 | Qwell: no.. how do i list functions in asterisk console? |
| 03:23.42 | *** join/#asterisk FastJack_ (i=fastjack@p5091F9BA.dip.t-dialin.net) |
| 03:23.42 | Qwell | You can't use trunking without zaptel timing |
| 03:23.43 | tronix | kll: so traffic basically takes 40-60ms, a nice scenic drive, to get most anywhere. but between CPE and them is 2ms? |
| 03:23.49 | Cleyverson | ok but how to fix it? |
| 03:24.01 | Qwell | get a timer |
| 03:24.03 | justinu | SpaceBass: there's two attributes to echo, tail and erl |
| 03:24.07 | *** join/#asterisk Administrator (i=arv@j-chaos.net) |
| 03:24.09 | Qwell | hardware, or ztdummy |
| 03:24.11 | kll | tronix: kk |
| 03:24.23 | SpaceBass | justinu: I guess i need to research some |
| 03:24.35 | SpaceBass | dont know about tail and eri |
| 03:24.39 | justinu | erl = echo return loss (basically, the aplitude of the echoed signal) |
| 03:24.55 | tronix | how accurate/good is ztdummy with Linux kernel 2.6's RTC clock support? |
| 03:24.58 | justinu | tail = the lag the echoed signal comes in with |
| 03:24.59 | a1fa||64 | damn it |
| 03:25.00 | Cleyverson | where may i get a timer? |
| 03:25.04 | Qwell | hardware, or ztdummy |
| 03:25.24 | Cleyverson | ztdummy is software? |
| 03:25.26 | SpaceBass | justinu: so those are measurements or something, not parameters that need to be set/monitored |
| 03:25.37 | justinu | SpaceBass: yeah, measurements |
| 03:25.39 | tronix | Cleyverso: yes, it is |
| 03:25.52 | SpaceBass | justinu: how can I measure them and then mitigate them? |
| 03:26.03 | a1fa||64 | this sucks |
| 03:26.06 | justinu | but i'm saying, if your latency increased during the call... it could make it sound worse |
| 03:26.15 | tronix | Cleyverso: if you use ztdummy and Linux, strongly recommend enabling kernel's RTC clock option |
| 03:26.41 | tronix | (not sure if setting Hz to 100/250/1000 makes a particular difference for ztdummy.) |
| 03:26.43 | SpaceBass | justinu: this is going out over a zaptel line so not sure how network latency plays in.. i guess from the sip device to the server |
| 03:26.47 | Cleyverson | i would like to conect 2 * servers...should I use it? |
| 03:26.54 | Qwell | sure |
| 03:26.54 | a1fa||64 | i get "the number you dialed is not in service" when i try to dial-out with that patch |
| 03:27.05 | justinu | spacebass: the EC code in the zaptel drivers monitor those things and try and compensate |
| 03:27.19 | kll | ztdummy can use kernel timing if you set it to 1000Hz. but by default it now uses rtc afaik |
| 03:27.30 | tronix | kll: ahh, makes sense. thanks! |
| 03:27.36 | justinu | SpaceBass: thats part of the latency |
| 03:27.43 | a1fa||64 | Qwell: ( |
| 03:27.45 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 03:27.47 | Cleyverson | so i have just to download and install..an I right? |
| 03:27.51 | *** join/#asterisk ManxPowe (n=ewieling@dpc6745150107.direcpc.com) |
| 03:28.09 | justinu | spacebass: also the PSTN does have latency... |
| 03:28.15 | SpaceBass | justinu: is there a way to dynamically compensatate for LAN latency |
| 03:28.22 | tronix | Cleyverso: it's part of the zaptel source code build |
| 03:28.40 | justinu | spacebass: no, latency is just there... no way to compensate for it |
| 03:28.51 | SpaceBass | justinu: what about the echo as a result |
| 03:28.53 | Cleyverson | i already installed zaptel package |
| 03:29.01 | tronix | Cleyversion: what distro do you use? |
| 03:29.06 | justinu | spacebass: yes, that is what echo cancellers do |
| 03:29.18 | Cleyverson | it should works...I'm using Suse 9.3 |
| 03:29.26 | Qwell | 2.4 kernel? |
| 03:29.29 | tronix | Cleyverso: in Gentoo, it comes as part of the zaptel package |
| 03:29.48 | Cleyverson | 2.6 |
| 03:29.56 | Qwell | Then just modprobe ztdummy |
| 03:29.58 | a1fa||64 | Qwell: help man :P |
| 03:30.02 | tronix | Cleyverso: got km_zaptel installed? |
| 03:30.05 | SpaceBass | justinu: anything I can do besides echocancle=yes |
| 03:30.08 | tronix | km = kernel module, I believe |
| 03:30.11 | Qwell | a1fa||64: if it says the number isn't in service, then it isn't in service |
| 03:30.47 | a1fa||64 | the number is in service.. if i dial with 1 (which is Dial) |
| 03:30.52 | a1fa||64 | then it connects me right through |
| 03:30.57 | tronix | Qwell: had same problem a short time ago... solution was to deregister/re-register my SIP softphone |
| 03:30.59 | a1fa||64 | if i dial with 8 (which is VoiceChanger |
| 03:31.02 | justinu | SpaceBass: i'm not up to speed on how EC works on the zaptel cards |
| 03:31.06 | justinu | sorry :\ |
| 03:31.33 | Qwell | tronix: huh? |
| 03:31.36 | tronix | Qwell: but probably not same issue since this is in a different context. sorry :) |
| 03:31.47 | wunderkin | justinu ain't a zaptel boy |
| 03:32.14 | anonymouz666 | how it's possible for another context != default to see variables like ${EXTEN} |
| 03:32.14 | anonymouz666 | ? |
| 03:32.29 | Qwell | anonymouz666: huh? |
| 03:32.32 | a1fa||64 | exten => _9NXXNXXXXXX,1,VoiceChangeDial(SIP/${EXTEN}@sip.broadvoice.com|-6.0|20|hr) |
| 03:32.46 | a1fa||64 | same thing for 1, but 1 is Dial( |
| 03:32.50 | Qwell | a1fa||64: well, yeah, why are you sending the 9 to broadvoice? |
| 03:32.53 | Qwell | and why no 1? |
| 03:32.55 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 03:33.01 | a1fa||64 | 1 is normal |
| 03:33.05 | a1fa||64 | 9 is voicechanger |
| 03:33.06 | Qwell | yes, I know that |
| 03:33.06 | justinu | wunderkin: actually, i know a lot more about zaptel now... |
| 03:33.11 | Qwell | Why are you sending them the 9? |
| 03:33.16 | wunderkin | a1fa||64: SIP/1${EXTEN:1} |
| 03:33.18 | a1fa||64 | :P |
| 03:33.19 | wunderkin | justinu, oh ya? :D |
| 03:33.22 | a1fa||64 | ouch |
| 03:33.30 | a1fa||64 | <-- lame |
| 03:33.34 | Qwell | very |
| 03:34.21 | wunderkin | justinu, did i tell ya or did you see that you can correctly loop 2 and 4 port t1 cards now |
| 03:34.24 | *** part/#asterisk techie (i=gus@antibala.com) |
| 03:34.35 | anonymouz666 | Qwell: when you call your exten => will match in [default] I want something like this exten => _2400XXXX,1,Goto(new,s,1) and in [new] Dial(SIP/${EXTEN} ... |
| 03:34.36 | Qwell | wunderkin: You weren't able to before? |
| 03:34.57 | Qwell | anonymouz666: store ${EXTEN} in another var |
| 03:34.58 | wunderkin | nope, that was fixed a week or 2 or somethin ago |
| 03:35.00 | justinu | wunderkin: no... what fixed that? |
| 03:35.04 | kll | I'm getting US indication tones in my phone. changing the default in indications.conf has proven useless. is this solely up to the phone or is it configurable on the asterisk side? I am using g729 so I expect tones not to be sent inband.. |
| 03:35.12 | wunderkin | it was like a 2 liner patch |
| 03:35.20 | a1fa||64 | haha |
| 03:35.24 | a1fa||64 | voice changer works like a charm |
| 03:35.27 | justinu | kll: is it a SIp phone? |
| 03:35.29 | anonymouz666 | Qwell: do I need a global var for that? |
| 03:35.31 | Qwell | no |
| 03:35.50 | justinu | kll: if it's a sip phone, usually the tones are generated by the phone itself |
| 03:36.00 | Qwell | Why are you using the default context anyhow? That's just dumb |
| 03:36.33 | anonymouz666 | the sip call arrive there |
| 03:36.38 | Qwell | well, fix it |
| 03:36.38 | anonymouz666 | default context |
| 03:36.42 | wunderkin | http://lists.digium.com/pipermail/svn-commits/2006-January/009727.html thats it there |
| 03:36.53 | Qwell | otherwise, people can AND WILL call through your guest account, and cost you money |
| 03:37.34 | kll | justinu: it's an ATA box, so yes I suppose it's a sip phone |
| 03:38.19 | kll | behaves as one at least ;) |
| 03:38.38 | a1fa||64 | anybody wanna test this voice changer? |
| 03:39.43 | *** join/#asterisk st3v (n=st3v@netblock-66-245-204-218.dslextreme.com) |
| 03:40.20 | st3v | We have a dual span digium card, and we are using one of the ports for the pbx, but can I use the other one to connect a T1 internet connection? |
| 03:40.28 | st3v | so I can use the linux box as a router |
| 03:41.33 | wunderkin | the installation instructions on digium.com should cover hdlc setup |
| 03:43.51 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 03:43.54 | wunderkin | looking at maybe a spa-941 .. hmm..a snom 360 or polycom 601 would be nice.. maybe later.. |
| 03:44.02 | SpaceBass | damn powerbook died on me! |
| 03:44.11 | SpaceBass | was in the middle of learning about echo |
| 03:44.25 | SpaceBass | justinu: you were talking about tail and eri... |
| 03:45.13 | SpaceBass | I've lowered my TX to -2.0 and its still getting progressivly worse and I';m trying to figure out how I can mitigate either the local latency or the PSTN echo...which ever is causing the issue |
| 03:45.16 | benjk | do you mean the battery ran out or the machine actually broke down? |
| 03:45.42 | SpaceBass | battery? HA! the battery stopped charing months ago... only runs on AC now |
| 03:46.03 | SpaceBass | this powerbook looks like it was run over by a simi truck |
| 03:46.09 | benjk | oh dear, how old is that baby? |
| 03:46.13 | Qwell | a truck from simi valley? |
| 03:46.17 | SpaceBass | like 3 years or less |
| 03:46.46 | benjk | mine is 2 and a half years |
| 03:46.57 | SpaceBass | mine is dented, bent, broken... the screen bezel has split wide open... but hey, it ruuns |
| 03:46.58 | benjk | I had to send it in a year ago, the hard disk failed |
| 03:47.00 | SpaceBass | runs |
| 03:47.24 | benjk | I am getting a MacBook soon |
| 03:47.24 | Qwell | what, no warranty? |
| 03:47.28 | SpaceBass | my disk may be on the out and out... not sure... i have ultra slow periods and then it runs normally |
| 03:47.39 | benjk | Qwell, five years warranty |
| 03:47.44 | SpaceBass | Qwell: broke my own rule: always buy warrenty on laptop |
| 03:47.47 | Qwell | So...go get it fixed |
| 03:47.52 | Qwell | meh |
| 03:48.01 | st3v | we have one problem with the asterisk pbx system. when someone is on a call, and someone else picks up another line, there are pops and clicks. is there a way to fix that? |
| 03:48.21 | Qwell | st3v: what codec? |
| 03:48.22 | *** join/#asterisk tainted- (n=identd@ppp-71-134-157-119.dsl.irvnca.pacbell.net) |
| 03:48.23 | benjk | I always get extended warranty unless I buy a machine under 100 USD for some lab stuff |
| 03:48.44 | wunderkin | SpaceBass, did you try ECHO_CAN_MG2 in zaptel/zconfig.h? |
| 03:49.20 | *** join/#asterisk bmg505 (n=leon@dsl-146-46-215.telkomadsl.co.za) |
| 03:49.27 | SpaceBass | wunderkin: didn't compile myself... using AAH ... so short answer is no, I didnt change anything |
| 03:49.36 | st3v | I'm not sure what codec, I am used the guide from that o'reilly book |
| 03:49.53 | st3v | we are using zap channels on a digium te210p and a channel bank |
| 03:50.12 | Qwell | st3v: any voip, or just over the t1? |
| 03:50.13 | SpaceBass | I dont tend to buy warrenty on desktops... althought I should on all apple stuff just b/c its propritary |
| 03:50.18 | st3v | no voip yet |
| 03:54.04 | st3v | we are also using a TDM04B for the 4 phone lines |
| 03:55.07 | benjk | I never buy any desktops unless they are no more than 100 USD |
| 03:55.38 | SpaceBass | used to be able to ebay a desktop for around $100 now it seems to be like $150-200 |
| 03:56.29 | benjk | I bought truckloads of IBM slim-minitower systems for about 80 -90 USD |
| 03:56.29 | SpaceBass | i need a $40 box for ipcop or m0n0wall... need to search tonight |
| 03:56.31 | benjk | like NetVista |
| 03:56.37 | SpaceBass | benjk: WOW... where? |
| 03:56.41 | benjk | in Tokyo |
| 03:56.46 | rob0 | SpaceBass: if old junk is ok, featuremarketing.com (ships from Phoenix) |
| 03:56.54 | benjk | we have a boom in second hand stuff here now |
| 03:56.57 | rob0 | oh yes, ipcop fine on stuff like that |
| 03:57.08 | benjk | they even give you warranty on the second hand stuff |
| 03:57.12 | SpaceBass | I just bought a 8" touch screen for my bathroom...mostly to use as a TV but am trying to think of a thin-type client use for it... and need something like a netvista |
| 03:57.21 | benjk | there are shops that specialise on second hand |
| 03:57.51 | benjk | some even have their own packaging and the buy experience is just like buying a new system |
| 03:58.00 | benjk | buying experience |
| 03:58.20 | *** join/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
| 03:59.26 | *** join/#asterisk coppice (n=chatzill@204.206.17.210.dyn.pacific.net.hk) |
| 03:59.31 | SpaceBass | got a 404 on featuremarketing.com |
| 03:59.42 | SpaceBass | I need a box for ipcop and i need a box for that touch screen |
| 04:00.08 | rob0 | www.featuremarketing.com ? hmmm, their DNS died I guess. |
| 04:00.15 | benjk | I often use 30-40$ mini systems for firewall/VPN servers |
| 04:00.29 | rob0 | no, their server is down |
| 04:00.50 | SpaceBass | benjk: exactly... as long as the box runs and doesnt crash that is perfect |
| 04:00.51 | coppice | benjk: where do you get a system for $40? |
| 04:01.09 | SpaceBass | I looked into building a embedded box for m0n0 but it was like $150... |
| 04:01.12 | benjk | IBM Aptiva 143 are nice for firewall/VPN servers |
| 04:01.33 | rob0 | ( I got their email yesterday, so try again later ) |
| 04:01.36 | benjk | remove the hard disk and replace it with an IDE/CompactFlash adapter and a CF card |
| 04:02.01 | benjk | coppice: here in Tokyo |
| 04:02.15 | SpaceBass | I love my IPcop box... old ass dell that runs great... blows new dells out of the water in terms of relibality (there I just jinxed it) |
| 04:02.58 | benjk | the beauty of using very old stuff is that it's unlikely to break down ever |
| 04:03.04 | SpaceBass | but its a little too slow to do traffic shaping for my connection and it doesnt have a captive portal feature |
| 04:03.12 | benjk | the only things that are vulnerable are hard disks and fans |
| 04:03.45 | SpaceBass | i tend to replace a lot of HDs lately |
| 04:03.47 | benjk | since those boxes have survived that long, they are the survivors, will probably last another 5-10 years |
| 04:03.52 | coppice | then the power supply caps. then the caps on the boards |
| 04:03.58 | SpaceBass | and IDE controllers.. failing left and right in 4 year old dell |
| 04:04.21 | benjk | coppice: less of a problem with low power low clock rate systems |
| 04:04.47 | benjk | as for hard disks, I replace them with CompactFlash cards |
| 04:05.00 | benjk | and the fans are cheap |
| 04:05.14 | rob0 | FWIW, the Friday email from featuremarketing.com had numerous systems in the sub-$50 range. |
| 04:05.16 | benjk | the second hand shops usually put new fans in those boxes before they sell the machines |
| 04:05.20 | coppice | benjk: you must be thinking of the recent cap problems. however, all electrolytics dry out over time. the warmer, the faster |
| 04:05.34 | rob0 | <== disclaimer: no affiliation nor personal buying experience with them |
| 04:06.16 | coppice | benjk: when you see an electrolytic rated for +85C, that is its 1000 hour dry out time - 1 month of contnuous operation |
| 04:06.16 | SpaceBass | this echo is driving me nuts |
| 04:06.24 | coppice | this echo is driving me nuts |
| 04:06.32 | rob0 | I was gonna say that :( |
| 04:06.56 | SpaceBass | like going to rip out my * box, torch my cisco phones, throw a temper tantrum and call verizon and ask for the most expensive POTS line they have with the least features |
| 04:07.39 | benjk | coppice: I am aware of it, but I am running those boxes constantly for years now and never had a single capacitor fail. I once had a cpacitor fail in a WiFi base station and an onboard VGA card |
| 04:07.52 | benjk | other than that, it's hard disks and fans which fail |
| 04:08.10 | coppice | SpaceBass: waste of time. however much you try you'll never outdo a 2 year old for tantrums like that :-) |
| 04:08.31 | SpaceBass | I bet I could |
| 04:08.41 | tronix | has the cisco ata186 bug with REGISTER (req'ing tftp/reboot workaround) been fixed for any of the v3 firmware? |
| 04:09.10 | SpaceBass | I'm about ready to take this wifi phone and piss on it and jump up and down and yell and throw a full fledge fit! |
| 04:09.28 | benjk | which WiFi phone is that? |
| 04:09.34 | coppice | SpaceBass: if my sone was still 2, he would probably rise to a challenge like that |
| 04:09.51 | SpaceBass | hitachi ip5000 |
| 04:09.59 | benjk | Oh, I have two of those |
| 04:10.06 | benjk | what's the problem with it? |
| 04:10.23 | SpaceBass | benjk: worked pretty well before i upgraded my * box |
| 04:10.35 | benjk | upgraded to what? |
| 04:10.40 | *** join/#asterisk NK123 (n=NK123@ip68-227-192-219.dc.dc.cox.net) |
| 04:10.54 | SpaceBass | benjk: two fold... one its decided to only vibrate and not ring, even though vibrate is turned off and ring is on... but mostly, its the echo |
| 04:11.09 | coppice | "worked prettty well" but you just couldn't leave well alone, could you. :-) |
| 04:11.10 | SpaceBass | benjk: went from a 300mhz 256mb box to a 1.25ghz 1gb box |
| 04:11.39 | benjk | are you using the same WiFi base station? |
| 04:11.44 | SpaceBass | coppice: it was all a bad idea... decided to port my POTS number to broadvoice... so i wanted a more reliable box... well i should have stopped at the porting |
| 04:11.49 | NK123 | can anyone suggest me where i can find uninstall procedure for asterisk on f3 |
| 04:12.08 | SpaceBass | benjk: no, changed the AP too... but my wired lines (different subnet) have the echo too |
| 04:12.21 | benjk | NK123 I have a fancy uninstall_asterisk bash script |
| 04:12.36 | benjk | it's not specific to F3 |
| 04:12.44 | NK123 | ok |
| 04:12.44 | benjk | but should do the trick |
| 04:12.45 | wunderkin | i think there is a make uninstall now |
| 04:12.45 | SpaceBass | benjk: since the IP5000 doesnt support WPA, I had to create a WEP base station on its own subnet |
| 04:12.53 | tronix | well, guess we'll find out if the ata186 still has that bug with firmware rev 3.1.0. :-) |
| 04:13.00 | tronix | (if so, i'll report back) |
| 04:13.09 | wunderkin | drumkilla was cool and tested it on a production box :P |
| 04:13.36 | NK123 | can i get it benjk |
| 04:13.39 | benjk | SpaceBass: I have seen the WIP5000 refuse to work depending on the base station used, even with everyting turned off |
| 04:14.04 | SpaceBass | benjk: its using a wrt54g now... non hacked... |
| 04:14.25 | benjk | NK123 download it from http://www.sunrise-tel.com/asterisk-on-macosx.html |
| 04:14.44 | benjk | scroll down to where it says "Uninstalling ..." and "Asterisk Shell Scripts" |
| 04:14.53 | coppice | i've seen cisco cards refuse to work with cisco base stations, and centrinos hardly ever seem to work with cisco base stations. 802.11 sucks |
| 04:15.26 | SpaceBass | call me old school... but wired works... |
| 04:15.43 | SpaceBass | fiber b/t floors... cat 5 from port to switch |
| 04:15.46 | NK123 | ok thanks benjk |
| 04:15.53 | benjk | I only use WiFi for the WiFi phones |
| 04:15.53 | SpaceBass | I have a small tech problem... so I've been told |
| 04:16.05 | coppice | SpaceBass: If you were a radio engineer you would definitely agree with that :-) |
| 04:16.10 | SpaceBass | benjk: have a preference over the WIP5000? |
| 04:16.38 | benjk | I was about to get a UTStarcom |
| 04:16.46 | benjk | UTStarcom are cool |
| 04:16.57 | SpaceBass | benjk: really? i might need to look at one |
| 04:17.01 | benjk | I know them, trust them, they make solid stuff |
| 04:17.03 | coppice | people seem to laugh at the UT StarCom phones |
| 04:17.27 | SpaceBass | I have the UT pocketpc phone... its hoopty compaired to my old blackberry |
| 04:17.34 | benjk | UTStarcom have more experience in wireless phones than all the other WiFi phone folks put together |
| 04:18.11 | coppice | yeah, but I think they are rushing things out, cos the collapse of their PHS business caught them by surprise |
| 04:18.19 | SpaceBass | phs? |
| 04:18.28 | SpaceBass | pathways healthcare scheduling? |
| 04:18.28 | benjk | I wouldn't know about the PocketPC thing because that's likely not UTStarcom but just an OEM thing |
| 04:18.40 | benjk | Personal Handyphone System |
| 04:18.51 | SpaceBass | ahhhh |
| 04:19.10 | benjk | exatly 42 bazillion times better than WiFi |
| 04:19.19 | benjk | for telephony anyway |
| 04:19.30 | coppice | UT Starcom was build to a huge size on PHS. now its dying, they are diversifying quickly |
| 04:19.48 | SpaceBass | yeah... telephony seems to rely on almost zero latency and voip (by nature of IP ) isn't quite there |
| 04:20.29 | benjk | they launched a GSM/WiFi dual-mode phone this month |
| 04:20.39 | SpaceBass | I'm seeing more and more hospitals put in cisco phones... assume that is a good market for cisco |
| 04:21.09 | SpaceBass | my ppc has wifi and is thus basically a cdma/wifi phone |
| 04:21.27 | SpaceBass | but I still dont have a working sip client for it |
| 04:21.34 | *** join/#asterisk Mike (n=mike@201.145.88.41) |
| 04:21.38 | benjk | then its not a WiFi phone |
| 04:22.24 | SpaceBass | oui c'est vrai |
| 04:22.43 | benjk | how can they put a hospital inside a Cisco phone? |
| 04:22.48 | SpaceBass | anyhow... i gotta hit the sack... i'll deal with this echo and other prblems tomorrow |
| 04:22.55 | SpaceBass | benjk: very tiny doctors |
| 04:22.59 | benjk | haha |
| 04:24.27 | SpaceBass | its funny how much wireless is in hospitals, but they insist you turn your cell off |
| 04:24.43 | SpaceBass | woudln't want it interfearing with the equipments and all |
| 04:24.50 | SpaceBass | damn i cannot spell and type tonight |
| 04:25.34 | coppice | if they put a hospital inside a phone, won't the phone be full of dangerous chemicals, and radioactive materials? sounds dangerous |
| 04:25.59 | Administrator | xchat for windows is like.. gay |
| 04:26.10 | SpaceBass | windows is like... gay :) |
| 04:26.59 | SpaceBass | i hate to say that mIRC has it covered...cannot find anything for OS X or Linux that is as god |
| 04:28.35 | SwK | anyone know of a CVS LD (NANPA/Int'l) rate sheet I can grab off the intarweb quickly? |
| 04:29.52 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 04:30.32 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 04:32.51 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 04:33.39 | benjk | IRC clients generally suck ballz |
| 04:33.45 | benjk | unless they are command line |
| 04:34.42 | Cerlyn | Does anyone know if Optimum Voice (Cablevision) has any readily accessible SIP access? |
| 04:35.16 | *** join/#asterisk Johnnie (n=jdlewis@24.154.53.16) |
| 04:40.00 | *** join/#asterisk Corydon76-hm (i=red@pcp09181629pcs.nash01.tn.comcast.net) |
| 04:42.51 | *** join/#asterisk brookshire[home] (n=matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 04:43.35 | *** join/#asterisk techie (i=gus@antibala.com) |
| 04:45.29 | brookshire[home] | yay |
| 04:45.48 | Qwell | brookshire[home]: good thing you left! |
| 04:46.04 | brookshire[home] | i know right ;) |
| 04:46.47 | file | bye bye Digium |
| 04:46.58 | DarkFlibble | ib |
| 04:47.53 | Krill | does the 1.2.2 tarball have the latest version of ooh323 |
| 04:50.26 | drumkilla | Krill: no, that is in the asterisk-addons package |
| 04:50.28 | drumkilla | file: !!!!!! |
| 04:50.32 | file | drumkilla: !!!!!! |
| 04:51.34 | Krill | yeah i just reliased that when i dug thru the tarball ;) |
| 04:51.40 | brookshire[home] | drumkilla! |
| 04:51.46 | drumkilla | brookshire[home]: !!!!!!!!! |
| 04:51.51 | brookshire[home] | z0mg! |
| 04:51.53 | brookshire[home] | <3 |
| 04:55.33 | brookshire[home] | can you do variable variables in c? |
| 04:55.40 | drumkilla | ... what? |
| 04:55.40 | *** join/#asterisk budmang (i=budman@12-216-205-236.client.mchsi.com) |
| 04:55.43 | budmang | :-) |
| 04:55.45 | Qwell | like ${${EXTEN}} ? |
| 04:55.56 | budmang | Anyone get this working on Freebsd 5.4? |
| 04:56.05 | Qwell | budmang: get what working? |
| 04:56.19 | coppice | brookshire: make the stack space too small, and it makes lots of variables variable :-) |
| 04:56.27 | brookshire[home] | hehe |
| 04:56.52 | brookshire[home] | Qwell: i think so |
| 04:57.05 | brookshire[home] | hmm.. probably have to use an array |
| 04:58.16 | budmang | Just basic asterisk running |
| 04:58.22 | budmang | On my dedicated server. |
| 04:58.34 | budmang | ports is erroring out for me. |
| 04:58.35 | Qwell | budmang: sure |
| 04:58.38 | Qwell | don't use ports |
| 04:58.40 | budmang | Wondering if anyone could help |
| 04:58.41 | budmang | ok. |
| 04:58.45 | Qwell | just get the normal source, and compile it yourself |
| 04:58.48 | budmang | I have the source tar too |
| 04:58.54 | budmang | and make is erroring |
| 04:58.55 | Qwell | missing any libs? |
| 04:59.02 | Qwell | there is a list on the wiki |
| 04:59.04 | budmang | "Makefile", line 28: Missing dependency operator |
| 04:59.05 | Qwell | ~wikis |
| 04:59.07 | jbot | methinks wikis is http://www.voip-info.org |
| 04:59.15 | budmang | Im on that site. |
| 04:59.22 | budmang | Thats how I found this place. |
| 04:59.24 | Qwell | search for asterisk dependencies |
| 04:59.32 | brookshire[home] | just install linux :) |
| 04:59.36 | Qwell | or that :p |
| 05:00.21 | brookshire[home] | or buy a mac |
| 05:01.37 | rob0 | file: ping |
| 05:02.19 | budmang | Site is saying 1.2.1 for freebsd should I try that version? |
| 05:02.30 | Qwell | no, 1.2.2 should work just as well |
| 05:03.03 | brookshire[home] | is it erroring on zaptel? |
| 05:03.12 | budmang | "Makefile", line 28: Missing dependency operator |
| 05:03.12 | budmang | "Makefile", line 31: Need an operator |
| 05:03.19 | budmang | exact error on make on the 1.2.2 source |
| 05:03.22 | brookshire[home] | or asterisk |
| 05:03.31 | budmang | asterisk |
| 05:03.34 | brookshire[home] | ok |
| 05:04.09 | drumkilla | this is why I'm going to set up a FreeBSD box |
| 05:04.27 | drumkilla | so I can try to build everything on it, at least |
| 05:04.50 | file | rob0: hi |
| 05:04.55 | budmang | do I need zaptel? |
| 05:05.03 | rob0 | hey, I forgot the name of your company :( |
| 05:05.17 | Qwell | budmang: only if you have digium hardware, or need a timer |
| 05:05.22 | file | Asterlink |
| 05:05.28 | budmang | I will pay someone for help :-) |
| 05:05.32 | rob0 | ah thank you |
| 05:05.46 | Qwell | budmang: help with what? |
| 05:05.56 | dmz | budmang how much ya got :) |
| 05:06.01 | budmang | 20 |
| 05:06.03 | budmang | 30 |
| 05:06.04 | dmz | heh |
| 05:06.07 | budmang | 30$ for 1 hour |
| 05:06.09 | Qwell | dmz: he's all yours |
| 05:06.14 | budmang | i just want to use softphone |
| 05:06.16 | budmang | on 2 pcs |
| 05:06.22 | budmang | through my dedicated server |
| 05:06.42 | dmz | i'm still trying to get fwd working, i wish i could see the errors on the other side :| |
| 05:06.44 | brookshire[home] | drumkilla: i kevin has one i thought |
| 05:08.58 | *** join/#asterisk earthsound (n=another1@24.179.15.197) |
| 05:12.43 | *** join/#asterisk bch (n=bch@CPE-70-92-133-175.mn.res.rr.com) |
| 05:14.56 | bch | how do you pass a value back to * using PHP? I can't seem to get my script to pass anything back but 0 |
| 05:15.20 | Qwell | set a channel var |
| 05:16.49 | bch | so I can next statements? |
| 05:17.00 | Qwell | huh? |
| 05:17.05 | bch | sorry |
| 05:17.07 | bch | nest |
| 05:17.20 | Qwell | in php? sure, why not? |
| 05:17.45 | bch | so exten => 1,1,SetVar(test=AGI(test.php))? |
| 05:18.02 | Qwell | no, AGI isn't a function |
| 05:18.15 | Qwell | Have the script set a channel var |
| 05:18.39 | bch | ahh, good thinking |
| 05:18.41 | bch | thanks |
| 05:20.54 | hardwire | blah |
| 05:24.50 | brookshire[home] | bl4h |
| 05:25.07 | hardwire | blah |
| 05:25.15 | Qwell | plugh |
| 05:25.20 | brookshire[home] | blech |
| 05:27.02 | Corydon76-home | Heh, an AGI function |
| 05:27.11 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 05:27.11 | Corydon76-home | I suppose that might could be done |
| 05:27.21 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 05:27.37 | Qwell | AGI would need to be able to return a var to be useful, but... |
| 05:28.25 | Corydon76-home | Qwell: take a look at 5940... such an obvious solution... |
| 05:28.41 | Qwell | indeed |
| 05:28.41 | Math[laptop] | you mean AGI can't set a channel var right now? |
| 05:28.51 | Corydon76-home | Sure it can |
| 05:28.53 | Qwell | Math[laptop]: it can set channel vars, sure |
| 05:28.56 | Qwell | but it doesn't return a value |
| 05:29.08 | Corydon76-home | AGI can already even set functions |
| 05:29.10 | Qwell | Corydon76-home: saw that yesterday |
| 05:29.43 | Corydon76-home | Qwell: I thought that was just too obvious... I wonder why I didn't think of it before |
| 05:31.14 | Qwell | Corydon76-home: you rock for committing the queue stuff |
| 05:32.18 | *** join/#asterisk kshumard (n=kshumard@gateway.digium.com) |
| 05:32.18 | *** join/#asterisk reni (n=nubb@gateway.digium.com) |
| 05:32.22 | *** join/#asterisk angler (n=angler@gateway.digium.com) |
| 05:32.25 | Qwell | uh oh, they're back |
| 05:32.50 | brookshire[home] | yeah.. |
| 05:32.53 | coppice | anyone using fedora with a Via C3 machine? |
| 05:33.09 | rob0 | <== has one with Slackware 10 |
| 05:33.20 | brookshire[home] | gentoo with a c3 :) |
| 05:33.31 | Qwell | eeps |
| 05:33.34 | Qwell | sounds slow |
| 05:33.38 | Qwell | but efficient |
| 05:33.48 | rob0 | brookshire[home]: you are a masochist :) |
| 05:33.52 | Qwell | I almost ran gentoo on my 110mhz sparc... |
| 05:34.04 | brookshire[home] | rob0: it's a small server ;) |
| 05:34.07 | Qwell | That was...umm...interesting |
| 05:34.09 | brookshire[home] | i don't use that for my desktop |
| 05:34.10 | brookshire[home] | lol |
| 05:34.27 | brookshire[home] | it only took 3 days to install |
| 05:34.34 | Qwell | ugh |
| 05:34.41 | coppice | I have FC3 on a C3 machine. when I try to put FC4 on it the install dies, apparently somewhere in the SELinux part, even when i tell it not to install SELinux. I think something is built for a 686 |
| 05:34.44 | Qwell | it took 3 days to install Aurora (which is binary) on my sparc |
| 05:36.04 | coppice | It look three days to do my last FC4 install, though only an hour or so after I actually bothered to get off my lazy ass and answer the last of the questions :-) |
| 05:36.18 | Corydon76-home | Heh |
| 05:36.23 | rob0 | oh you definitely can't get away with i686 on a C3, but I guess you know that. |
| 05:36.30 | [av]bani | coppice: linux selinux=0 for the installer. |
| 05:36.30 | Qwell | well, for 3 days, my room was full of HD spinups |
| 05:36.38 | Qwell | spinup noises that is |
| 05:36.47 | rob0 | <== learned that the hard way |
| 05:37.16 | brookshire[home] | just don't install fedora on anything :) |
| 05:37.29 | Qwell | it's all about gentoo. |
| 05:37.36 | coppice | [av]bani: there was actually a question about selinux during the install, and I said no. do you mean that doesn't really work and the command line option is essential? |
| 05:37.43 | Qwell | USE="-kde -qt"...oh yeah, it's good |
| 05:37.45 | brookshire[home] | or debian or unbuntu |
| 05:38.35 | [av]bani | coppice: that's for enabling/disabling selinux _after_ the install is finished. selinux is enabled during the install unless you explicitly turn it off. you have to do this for eg reiserfs and xfs, otherwise the installer explodes. |
| 05:39.06 | [av]bani | to really disable selinux you have to selinux=0 while booting the installer, _and_ answer "disabled" during install |
| 05:39.09 | coppice | [av]bani: thanks. they woulds like it might be the trick I need |
| 05:39.33 | [av]bani | fedora has a raging hardon for selinux, tehy want to make it as difficult as possible to turn it off. |
| 05:39.47 | [av]bani | and like to deny that there is anyone actually having problems with it |
| 05:39.52 | eieiyo | has anybody written any cool agi scripts... and if so, what kind of things have you done before? |
| 05:40.17 | brookshire[home] | there is a cool one for podcasts out there somewhere |
| 05:40.49 | Qwell | there's an agi jukebox on the bug tracker...that's pretty cool |
| 05:40.57 | eieiyo | there is one about the weather in the asterisk book published by oreilly... i just wondered what some other functions could be made to integrate into asterisk |
| 05:40.58 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 05:40.58 | Qwell | seems to work well |
| 05:41.04 | eieiyo | oh cool |
| 05:41.16 | Qwell | uses festival to read you the songs and such |
| 05:41.26 | eieiyo | that's pretty cool! |
| 05:41.40 | coppice | [av]bani: there was a screwup with FC2 and Via C3s, cos the installer's kernel was built for 686. They were very quick and thorough about sorting that out. info on getting FC4 onto a C3 machine is less accessible. |
| 05:42.38 | *** part/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 05:42.41 | [av]bani | it's pretty much the assumption that nobody has a non-686 machine. they keep forgetting about C3s and stuff. |
| 05:42.52 | eieiyo | is festival speech capability pretty good? |
| 05:43.04 | brookshire[home] | yes :) |
| 05:43.13 | brookshire[home] | sometimes |
| 05:43.17 | [av]bani | its often easier to install FC on another pc, then compile the kernel you need, and then move the disk to your actual target hardware |
| 05:43.19 | Qwell | when it works |
| 05:43.40 | eieiyo | i think it would be cool to have festival read you some of the first five rss feeds from your favorite site or something |
| 05:43.48 | Qwell | it could |
| 05:44.01 | eieiyo | cool... i might just do something like that |
| 05:44.35 | [av]bani | the most annoying thing about FC is that redhat employees like to come up with just about any BS excuse to close bugs, oftentimes their responses show how completely out of touch they are with end users |
| 05:44.43 | [av]bani | some of them are pretty unbelievable |
| 05:44.55 | eieiyo | be kind of cool if you were on the road and unable to get internet you could get some news headlines |
| 05:45.08 | Qwell | traffic reports |
| 05:45.09 | DarkFlibble | [av]bani, a lot of developers are like that... |
| 05:45.14 | eieiyo | yep |
| 05:45.27 | Qwell | DarkFlibble: What are you trying to say? |
| 05:45.33 | eieiyo | that would just be cool.. stock quotes... i guess the possibilities are endless |
| 05:45.35 | [av]bani | DarkFlibble: in my experience, redhat is the worst. most you can argue with, but redhat closes them and cuts off discussion. |
| 05:45.53 | DarkFlibble | Qwell, many developers are monitored by how many bugs they fix... that is the problem |
| 05:45.56 | [av]bani | the only project worse than that would probably be mplayer |
| 05:46.26 | [av]bani | redhat are just annoying, mplayer are totally antisocial |
| 05:46.32 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 05:47.16 | [av]bani | DarkFlibble: thats a sucky metric to use... one i would expect of microsoft |
| 05:47.45 | DarkFlibble | its fairly common on projects that are established and has managers that don't code |
| 05:48.56 | *** join/#asterisk dominix (n=dominix@CA03F6B0.adsl.mana.pf) |
| 05:51.06 | coppice | DarkFlibble: you can tell those projects by the nature of bugs they have. all the trivial bugs get cleaned up, and only the deep ones remain |
| 05:51.45 | DarkFlibble | yeah.. |
| 05:51.53 | dominix | is asterisk@home upgraded to 1.2.2 yet ? |
| 05:52.00 | DarkFlibble | anywayz....I'm gonna go back bed and get a lie in |
| 05:52.02 | Qwell | dominix: ask them |
| 05:52.04 | *** join/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 05:52.09 | Qwell | DarkFlibble: get a lie in? |
| 05:52.22 | DarkFlibble | yeah... lounge in bed longer than normal |
| 05:52.23 | Qwell | as in, "No honey, I wasn't up all night. I just went to the bathroom." ? |
| 05:52.51 | coppice | "*what* were you doing all alone in the bathroom all night?" |
| 05:52.54 | DarkFlibble | a lock in would be similar... in a pub |
| 05:53.15 | [av]bani | coppice: sounds like asterisk :) |
| 05:53.30 | DarkFlibble | errr...brushing my teeth? |
| 05:53.49 | Qwell | DarkFlibble: What were you doing that required a 3 hour brushing? |
| 05:53.51 | DarkFlibble | anyway... bbl |
| 05:53.58 | DarkFlibble | yeah... |
| 05:53.59 | DarkFlibble | :P |
| 05:55.13 | dmz | doh, i hate when things don't work because of a simple typo |
| 05:55.42 | benjk | dmz: welcome to the club |
| 05:55.58 | dmz | had a 36 when it was suppose to be 63 |
| 05:56.20 | benjk | almost all of my software problems are related to silly typos |
| 05:56.44 | dmz | at least as i was seeing pages & pages of debug stuff going by i was able to go..hmm that doesn't grok |
| 05:56.47 | coppice | I used to live in Tai Po :-) |
| 05:56.59 | benjk | haha |
| 05:57.01 | Qwell | coppice: seriously? |
| 05:57.06 | benjk | I know Tai Po |
| 05:57.16 | coppice | yep. its a town in Hong Kong |
| 05:58.34 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 05:59.43 | *** join/#asterisk tengulre (n=tengulre@219.145.1.161) |
| 06:00.15 | Grubs | Is anyone here able to reliably recieve a fax from the PSTN via a TDM400P? RxFax simply doesnt want to work for me. |
| 06:01.24 | tengulre | Grubs: which fax modem are u using? |
| 06:01.41 | coppice | Grubs: for a lot of people the TDM400P will not work for any form of modem. They get data slips. Some people have no trouble at all. You definitely want the current zaptel drivers, though. There was a period were it would not work for anyone |
| 06:02.33 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 06:02.46 | Grubs | I can successfully divert an incoming call to a SIP channel then to a fax machine OK. Just cant get the TDM400P to receive the fax directly using RxFax with SpanDSP. |
| 06:03.08 | *** join/#asterisk [hC] (n=hardcore@S0106000e9b96114f.vf.shawcable.net) |
| 06:03.11 | Grubs | is fax to TIFF file on the asterisk box |
| 06:03.17 | Grubs | (ie) |
| 06:04.02 | coppice | Grubs: SIP to a fax machine is the opposite direction from fax machine to rxfax |
| 06:05.47 | Grubs | I mean I can accept an incoming fax call in asterisk, detect its a fax, and then dial up an ATA that has a fax machine plugged into it and use the fax machine to receive the call. However if I try to accept the fax "natively" in asterisk and asve to file using RxFax then I get nowhere. |
| 06:06.13 | Grubs | asve = save |
| 06:07.42 | Grubs | I tried the "|debug" switch for RxFax but I cant find any additional logging either on screen or in /var/log/asterisk/messsages |
| 06:08.31 | coppice | Grubs: well, what you have working is something that works by luck, rather than design. receiving to rxfax is normally rock solid, as long as there are no data slips |
| 06:09.00 | coppice | |debug should produce considerable debug output |
| 06:09.02 | Grubs | thats nice to know anyhow. |
| 06:09.46 | Grubs | *where* is the debug output produced? on the CLI or ina log file? |
| 06:10.11 | coppice | you should see stuff on the screen |
| 06:10.50 | coppice | what do you see after the message that rxfax is being started? |
| 06:11.41 | Grubs | thx. I see nothing here at all. Next line after RxFax is normally something like "Zap/1 hungup" after a 4-5 second delay. |
| 06:13.10 | Grubs | I know some people have success with A@home... so I think I'll swap out the hard drive and try a fresh isntall of A@H and see if it behaves different;y. My current system is Debian Sarge with Asterisk 1.2.2 |
| 06:13.59 | Grubs | At least it might give me some confirmation one way or the other. |
| 06:15.03 | *** join/#asterisk Drew (n=foo@zux221-073-098.adsl.green.ch) |
| 06:15.47 | *** join/#asterisk dalfry (n=vaibhav@66.250.170.114) |
| 06:15.47 | Grubs | thanks for the help. I have a better idea of what to expect even if its not working right now. :) |
| 06:25.06 | Drew | hi - i'm new to asterisk - and i seem to have a problem with my extensions.conf |
| 06:26.11 | Drew | i have a sip account at a local provider and a couple of softphones in my network - so i put the phones into the sip.conf and made extensions for them |
| 06:27.18 | Drew | now i'm trying to make a extension for incoming calls - its supposed to answer the call and ask for the extension number, then connect the caller to that extension |
| 06:27.52 | Drew | i know this is probably a very easy thing to do, but ive read the docs and i cant find my mistake |
| 06:28.23 | Drew | it allways terminates the call when the caller enters the extension... |
| 06:28.28 | Drew | anybody here to help? |
| 06:29.48 | *** part/#asterisk Cerlyn (i=ALEIN@pdpc/supporter/sustaining/Cerlyn) |
| 06:34.17 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 06:38.46 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 06:53.28 | Corydon76-home | Drew: post your dialplan to http://pastebin.ca |
| 06:53.56 | dalfry | shehjar: /window close |
| 06:54.01 | *** part/#asterisk dalfry (n=vaibhav@66.250.170.114) |
| 07:03.52 | *** join/#asterisk smoove (n=arcraig@c-67-187-184-149.hsd1.ca.comcast.net) |
| 07:20.34 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 07:25.00 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 07:28.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:29.55 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa24w-142167051174.pppoe-dynamic.nb.aliant.net) |
| 07:31.34 | dmz | hey y'all, anyone have any suggestions/opinions on pstn iax/sip providers? at the prices I see my 2nd house line is still cheaper ($14/mo for pac bell 2nd line) |
| 07:35.00 | *** join/#asterisk ighost (n=ighost@82.99.201.67) |
| 07:35.28 | *** join/#asterisk lithi (n=interp3@HSE-Montreal-ppp134788.qc.sympatico.ca) |
| 07:39.22 | Mark_Halverson | need some dial plan help - how do i set the dial number to $X so that later i can DIAL(SIP/${X}@provder) |
| 07:39.46 | Math` | prefix your extension with _ |
| 07:39.51 | Math` | and.... X matches a *single* digit |
| 07:39.57 | Qwell | use ${EXTEN} |
| 07:39.58 | Math` | and, you should use ${EXTEN} in your Dial command |
| 07:40.03 | Mark_Halverson | ok hold a sec |
| 07:40.30 | Mark_Halverson | hold a second let me post a couple lines and you'll see the prob |
| 07:40.49 | Mark_Halverson | exten => s,1,Random(12:CA,s,1) |
| 07:41.03 | Mark_Halverson | [CA] |
| 07:41.03 | Mark_Halverson | exten => s,1,Random(2:s,100) |
| 07:41.37 | Mark_Halverson | exten => s,100,Dial(SIP/${EXTEN}@provider) |
| 07:41.56 | Mark_Halverson | i have this over 1k times, to randomize my callerid |
| 07:42.08 | Mark_Halverson | oops....100 times |
| 07:42.09 | Mark_Halverson | lol |
| 07:42.20 | Math` | right |
| 07:42.30 | Mark_Halverson | s,100 is a mistake...1st it sets the callerid then 101 dials |
| 07:42.41 | Math` | use s,2 |
| 07:42.42 | Mark_Halverson | so will it hold, the exten then xfer to s |
| 07:42.43 | Mark_Halverson | ? |
| 07:43.23 | Mark_Halverson | start with: exten => _1NXXNXXXXXX,1,Goto(s,1) |
| 07:44.27 | Mark_Halverson | nope it tries to dial s@provider |
| 07:46.38 | Mark_Halverson | how do i set the dialed number to ${ARG1} ????? |
| 07:49.46 | DarkFlibble | why randomise the callerid? why not just set it to nothing or a fixed value that wont annoy people |
| 07:50.00 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 07:51.10 | Mark_Halverson | because i have 100 DIDs and I want to randomize what is sent |
| 07:51.42 | Mark_Halverson | and blank is not an option |
| 07:51.42 | tronix | darkflibb: guess he doesn't want people fixating on a particular DID for inbound calls |
| 07:52.20 | Mark_Halverson | that is most of it |
| 07:52.46 | Mark_Halverson | all outbound calls orginate from a single * box, but inbound is to multiple boxes |
| 07:52.55 | coppice | deep feelings of utter insignificance make him want to appear to be 100 different people :-) |
| 07:53.18 | DarkFlibble | oh... not just random 10 digit us number... random in a group |
| 07:53.23 | DarkFlibble | got ya |
| 07:53.32 | Mark_Halverson | yeah |
| 07:53.59 | Mark_Halverson | i have a list of 100 DIDs...NOT spoofing random numbers |
| 07:54.04 | [av]bani | how do i show debug for failed extensions? |
| 07:54.13 | [av]bani | it gives congestion by default, instead of logging it |
| 07:57.42 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-92.claranet.co.uk) |
| 07:58.21 | *** part/#asterisk ighost (n=ighost@82.99.201.67) |
| 08:03.32 | [hC] | Hows everyone tonight.. |
| 08:03.48 | DarkFlibble | I'm fine...but its 8am |
| 08:03.57 | justinu | midnight here |
| 08:04.07 | DarkFlibble | midwest? |
| 08:04.21 | justinu | pacific coast |
| 08:04.24 | DarkFlibble | ahh... |
| 08:04.57 | justinu | i was just out driving with the top down, was nice |
| 08:04.59 | justinu | :) |
| 08:05.07 | DarkFlibble | hard to guess timezones when you live 1/3rd of the way round the planet |
| 08:05.17 | justinu | uk? |
| 08:05.23 | DarkFlibble | yeah |
| 08:05.26 | justinu | cheers |
| 08:05.51 | DarkFlibble | uk is easy... its one of the only countries in this timezone besides a few places in africa |
| 08:06.04 | DarkFlibble | rest of europe is 1 hour forward |
| 08:06.14 | DarkFlibble | (except eastern europe) |
| 08:06.24 | [hC] | midnight here too |
| 08:06.26 | [hC] | <- vancouver |
| 08:06.37 | justinu | i wish i lived in vancouver |
| 08:06.49 | [hC] | Where do you live? |
| 08:07.01 | DarkFlibble | Leicester, UK here |
| 08:07.16 | justinu | los angeles |
| 08:07.25 | marcus2_ | hrm |
| 08:07.42 | marcus2_ | i wonder if theres any way to use the speaker on our polycom phones for paging |
| 08:07.48 | marcus2_ | or for background music |
| 08:07.48 | DarkFlibble | oh well... I suppose everyone has to live somewhere... :P |
| 08:07.59 | justinu | it sucks here |
| 08:08.10 | justinu | nice weather about the only bonus |
| 08:08.10 | [hC] | marcus2_: you can check out app_page, new to 1.2 |
| 08:08.23 | [hC] | Im curious about background music, havent seen a phone that does that yet. |
| 08:08.44 | *** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com) |
| 08:09.03 | marcus2_ | i dont really see how app_page would be useful for this |
| 08:09.41 | marcus2_ | i'm trying to convince my father to replace the 12+ yr old meridian system at his office, but he's hung up on paging and background music |
| 08:10.24 | [hC] | well, app_page does paging, heh |
| 08:10.33 | [hC] | background music, ive never seen accomplished |
| 08:10.37 | justinu | marcus2: what's background music? using the phone like a radio? |
| 08:10.41 | marcus2_ | ahh, and i see here's a way to get the polycom to auto-answer |
| 08:10.49 | DarkFlibble | worst case you can get a multiline phone and use a seperate line for mp3streaming... |
| 08:11.04 | [hC] | yeah. I had implemented paging before using auto answer and meetme, but that way sucked. I have intercom right now on auto answer lines |
| 08:11.12 | marcus2_ | justin; yeah, the meridian switch has an audio input, and any phone on the system can channel that to its speaker when the phone isnt in use |
| 08:11.25 | justinu | i see |
| 08:12.11 | coppice | marcus2_ sounds like you need a kind of app_annoyance :-) |
| 08:12.20 | justinu | lol |
| 08:12.30 | NDT | hey justinu...you do much php/mysql? |
| 08:12.46 | DarkFlibble | NDT, what do you need? |
| 08:12.51 | NDT | http://pastebin.com/517192 |
| 08:12.53 | justinu | no, i'm like a java/postgres dude :P |
| 08:12.58 | NDT | dunno why I am gettin that heh |
| 08:13.57 | DarkFlibble | are you selecting the database with error handling? |
| 08:14.30 | NDT | yeah |
| 08:15.05 | NDT | mysql_select_db("foo") or die(mysql_error()); Well just to spit out error |
| 08:15.26 | DarkFlibble | its hard to know without more info... |
| 08:15.34 | Qwell | [hC]: back in Canada already? |
| 08:15.43 | NDT | damn thing looks right to me heh |
| 08:16.11 | Math[laptop] | probably there's no db called "foo" |
| 08:16.15 | DarkFlibble | it looks okay to be from that fragment |
| 08:16.30 | DarkFlibble | Math[laptop], surely that would generate an error... |
| 08:16.33 | NDT | lol... |
| 08:16.39 | *** join/#asterisk cacrus (n=atif@213.132.40.2) |
| 08:16.46 | Math[laptop] | did you mysql_connect() first? |
| 08:16.53 | DarkFlibble | ')") |
| 08:16.54 | NDT | yeah |
| 08:16.57 | DarkFlibble | hmmm... |
| 08:17.12 | Math[laptop] | oh well, sounds like you need to read the docs ;) |
| 08:17.16 | Qwell | surely one of your vars have a ' in them |
| 08:17.40 | Qwell | or something else that breaks it...That is why you don't use PHP...just look at phpBb2 |
| 08:17.48 | DarkFlibble | I normally assign the query to a variable... and then echo the variable... |
| 08:17.50 | Qwell | VERY easy to break |
| 08:17.58 | DarkFlibble | you get to see how its expanded with variables |
| 08:18.11 | NDT | yeah will try that |
| 08:18.17 | Math[laptop] | or just use mysql_format_string() to any var you pass in an sql query |
| 08:18.27 | Qwell | indeed |
| 08:18.28 | cacrus | I am looking for a way to connect two offices , in different countries , connect using asterisk servers , and i could make calls between these offices as local call , on internet |
| 08:18.39 | cacrus | Ia m a newbie to asterisk and vopi |
| 08:18.46 | Math[laptop] | cacrus: only 2 offices or do you plan to add more? |
| 08:18.52 | cacrus | plan more maybe |
| 08:18.53 | DarkFlibble | cacrus, look at IAX and trucking... |
| 08:19.04 | DarkFlibble | cacrus, also read the wiki |
| 08:19.13 | DarkFlibble | trunking even |
| 08:19.26 | Math[laptop] | I'd just use DUNDi |
| 08:19.28 | cacrus | unfortunately my offices are in middleast , i am not sure what hardware will be required |
| 08:19.48 | Math[laptop] | cacrus: look at http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+IAX |
| 08:19.53 | cacrus | there is no voip provider here , i am thinking it might not vbe required as both ends will have asteriks |
| 08:19.57 | DarkFlibble | its harder for *us* to know what hardware will be required... |
| 08:20.18 | DarkFlibble | we don't know your offices |
| 08:20.30 | Math[laptop] | cacrus: if you want to call between offices, you don't need *any* provider except for internet connectivity |
| 08:20.34 | benjk | cacrus: which countries in the Middle East? |
| 08:20.40 | cacrus | Yes i fugured that too , |
| 08:20.57 | cacrus | between UAE and Kuwait |
| 08:21.09 | benjk | ok, you're lucky then |
| 08:21.09 | marcus2_ | hm whats this about the polycom 601 supporting a custom static xhtml idle screen |
| 08:21.17 | marcus2_ | has anyone else done anything with this? |
| 08:21.37 | Math[laptop] | uhm I know some xhtml menus for Polycoms are on the Wiki(tm) |
| 08:21.53 | marcus2_ | yeah i've played a little bit with the xhtml browser |
| 08:22.03 | benjk | most other countries made VoIP illegal |
| 08:22.03 | *** join/#asterisk [Baby20] (i=Connecti@213.186.175.113) |
| 08:22.12 | Math[laptop] | benjk: voip illegal? why? |
| 08:22.15 | cacrus | you guys were really helpful , thanks |
| 08:22.36 | DarkFlibble | state maintained telco monpolies |
| 08:22.36 | benjk | because the national telcos have a monopoly on international calls |
| 08:22.36 | cacrus | yes vopi is illegal in most of the middleeastern countries as well |
| 08:23.02 | benjk | they don't want VoIP to spoil the party for them |
| 08:23.03 | Math[laptop] | benjk: thats ridiculous |
| 08:23.13 | justinu | it's reality |
| 08:23.18 | benjk | but UAE and Kuwait is ok, they also have good infrastructure |
| 08:23.22 | DarkFlibble | most people just do it anyway tho... |
| 08:23.35 | NDT | duh...lol was one of the variables not coming thtough was getting just a '' LOL |
| 08:23.35 | benjk | Math: depends how you look at it |
| 08:23.39 | Math[laptop] | what other part of the world made voip illegal for a such reason? |
| 08:23.54 | cacrus | I read the asterisk handbook and pdf focument , still cant figure out the hardware requirement for two offices connectivity , is there any document which discusses this network |
| 08:23.55 | DarkFlibble | Math[laptop], Far east has a few countries like that |
| 08:23.56 | benjk | many countries outlaw VoIP |
| 08:24.09 | DarkFlibble | probably africa as well |
| 08:24.10 | [av]bani | hm, teliax dun like it when i set outgoing callerid :() |
| 08:24.18 | benjk | consider it a kind of tax |
| 08:24.18 | Math[laptop] | benjk: I look at it like "Oh no! Competition! Let's prevent that" |
| 08:24.21 | coppice | especially when the telco is still a government body |
| 08:24.27 | DarkFlibble | cacrus, okay to pm you? |
| 08:25.02 | coppice | or the private telcos have been given a franchise that lays down the level of competition they will face |
| 08:25.05 | benjk | Math, are you also in support of paying third world countries a fair price for Bananas, Cocoa and Coffee? |
| 08:25.07 | Qwell | cacrus: How many phones at each office? |
| 08:25.59 | Math[laptop] | benjk: I'd be, but it'd break the idea of capitalism, which is solely based on welth difference |
| 08:26.12 | Math[laptop] | so if everyone is equal, everyone's poor |
| 08:26.17 | benjk | because if we did pay them a fair price for their goods and not strongarm them into selling at any price we dictate to them, then they would have some cash there to support themselves and speed up liberalisation of their national monopolies |
| 08:26.58 | cacrus | we have more than 10 phones in each offices |
| 08:27.09 | DarkFlibble | <biblebasher> but in the eyes of <deity> everyone is created equal</biblebasher> |
| 08:27.21 | cacrus | what does pm you means ? |
| 08:27.28 | DarkFlibble | private msg |
| 08:27.41 | [av]bani | DarkFlibble: $deity to me results in null pointer |
| 08:27.59 | Math[laptop] | DarkFlibble: in the eyes of communism too, in theory |
| 08:28.26 | Math[laptop] | [av]bani: thats no problem as long as you don't try to (de)reference it |
| 08:28.34 | DarkFlibble | [av]bani, pretty much the same here... but I am gradually stating to understand why people follow doctrines... |
| 08:28.41 | coppice | look at all those south american countries using my R2 software for free. its bloody disgusting, I tell you :-) |
| 08:28.55 | justinu | coppice: lol! |
| 08:28.56 | Qwell | coppice: damn pirates |
| 08:28.59 | Math[laptop] | lol |
| 08:29.19 | benjk | well, coppice, at least you deserve the low banana prices in your local supermarket then |
| 08:29.27 | justinu | lol |
| 08:29.32 | benjk | the rest of us certainly don't deserve them |
| 08:30.09 | benjk | everytime we buy bananas from South America we should paypal some money to coppice |
| 08:30.15 | Math[laptop] | lol |
| 08:30.29 | benjk | in payment for his R2 software being used in South America |
| 08:30.31 | Qwell | what? he gets cheap bananas AND money? :P |
| 08:30.38 | Qwell | That's an outrage! |
| 08:31.36 | Math[laptop] | agreed |
| 08:31.41 | coppice | the peanut farmers pay me peanuts. the banana farmers pay me bananas. notice how all these fruity names for something very cheap is a produce from an oppressed nation? |
| 08:32.02 | benjk | indeed |
| 08:32.30 | benjk | and when they tried to protect their stuff, we send agents to steal the crops |
| 08:32.44 | Math[laptop] | indeed |
| 08:33.05 | coppice | or the CIA sends ITT to overthrow their government |
| 08:33.31 | justinu | itt? |
| 08:33.36 | benjk | and we forced China to allow us to sell opium there |
| 08:33.42 | Qwell | I'm okay with that, as long as I pay $.04 less per banana |
| 08:34.21 | coppice | benjk: not quite. in the china deal everyone was in the wrong |
| 08:34.27 | benjk | so, with our track record, it is understandable that some countries close certain industries to prevent us from dumping their prices and destroy income they need |
| 08:34.50 | benjk | coppice: yeah, it was one of those things that spiraled out of control |
| 08:35.07 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:35.13 | benjk | but still it wasn't exactly confidence building |
| 08:35.47 | DarkFlibble | benjk, without limiting forces everything will spiral out of control... |
| 08:35.55 | coppice | we sold them only the best quality stuff, at reasonable prices. they overcharged for their tea. who was really in the wrong? :-) |
| 08:36.00 | cacrus | DarkFlibble: are you getting my pm reply ? |
| 08:36.05 | DarkFlibble | cacrus, nope |
| 08:36.08 | cacrus | Oh |
| 08:36.16 | DarkFlibble | you might need to register first |
| 08:36.19 | cacrus | where do i register from ? |
| 08:36.29 | Qwell | /msg nickserv help |
| 08:36.31 | DarkFlibble | <PROTECTED> |
| 08:37.09 | DarkFlibble | you'll need to identify when you join in future with the same nick |
| 08:37.13 | coppice | benjk: what is really surprising about the opium/tea thing is how long the British took to be able to produce good tea within its empire |
| 08:42.10 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 08:43.00 | *** join/#asterisk [B]egum (n=anna_25f@213.186.175.113) |
| 08:53.24 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 08:54.33 | benjk | well, that's the thing isnt it, every business that looks easy at first turns out to involve some hard work after all |
| 08:56.47 | DarkFlibble | I have a nice MLM snow selling scheme for any innuit here... |
| 09:07.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:10.24 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 09:13.07 | *** join/#asterisk Mike (n=mike@201.135.48.190) |
| 09:15.44 | *** join/#asterisk Husk (n=Husk__@202.55.153.169) |
| 09:22.19 | tronix | hmm. i've got a pulse dialling phone... 'zap show channel 4' says: 'Pulse phone: no' |
| 09:22.29 | tronix | even though I've got 'pulsedial: yes' set in zapata.conf |
| 09:22.37 | tronix | am I missing something really obvious? |
| 09:25.44 | *** join/#asterisk yiddoX (n=yiddoX@host-84-9-43-72.bulldogdsl.com) |
| 09:34.59 | *** join/#asterisk lorinc (n=ang@caracas-3204.adsl.interware.hu) |
| 09:37.12 | *** join/#asterisk MYing (n=Ming@mying.enta.net) |
| 09:38.07 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 09:42.58 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 09:54.58 | *** join/#asterisk knobo (n=knobo@liberalitas.freecode.no) |
| 09:57.35 | knobo | I'm going to make a dial plan for a supportnumber on a helpdesk. There some external cell-phone-numbers shall be automaticly subscribed to a queue, based on a time-schedule made from the admin. What is the best way to do this? |
| 09:57.41 | knobo | with AGI? |
| 09:57.54 | knobo | maybe? |
| 10:00.02 | knobo | or with asterisk manager API? |
| 10:01.31 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 10:04.50 | knobo | Or maybe Asterisk RealTime Queue |
| 10:09.44 | knobo | where queue_member_table is a sql-view, that does a select baesd on the time... |
| 10:10.08 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 10:10.15 | robin_sz | doodz! |
| 10:10.56 | robin_sz | OK, I have a problem ... |
| 10:12.29 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:13.50 | *** join/#asterisk Johnnie (n=jdlewis@24.154.53.16) |
| 10:14.22 | robin_sz | using voipgate.com ... calls made outbound to number in say .ch, or mobiles work fine .. but some landlines are one-way audio only, is this possibly to do with coding standards? all calls are ilbc at the moment |
| 10:14.42 | tzafrir_laptop | knobo, why not use the caller ID number variable in your dialplan? |
| 10:15.09 | tzafrir_laptop | knobo, also check GotoIfTime |
| 10:15.16 | robin_sz | eg .ch landlines work fine, as do the USA, UK is one-way audio only .. or is this just a voipgate problem and theres nothing I can do in * to affect it? |
| 10:20.28 | *** join/#asterisk NuC|EaR (n=nuc@Toronto-HSE-ppp3751932.sympatico.ca) |
| 10:26.11 | *** join/#asterisk ToTo (n=ToTo@host197-231.pool870.interbusiness.it) |
| 10:32.39 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 10:51.38 | *** join/#asterisk coppice (n=chatzill@248.162.17.210.dyn.pacific.net.hk) |
| 10:53.43 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 11:08.55 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:15.06 | *** join/#asterisk svenna_ (n=svenna@p548D2AD1.dip0.t-ipconnect.de) |
| 11:20.32 | *** join/#asterisk RoyK (n=roy@87.80-202-9.nextgentel.com) |
| 11:20.57 | *** join/#asterisk Cleyverson (n=cleyvers@201.29.64.159) |
| 11:21.19 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 11:22.15 | axscode | just want to ask.. if a VoIP Gatekeeper can communicate to Asterisk? |
| 11:22.45 | kink0 | axscode, why no ? |
| 11:22.54 | axscode | why not? |
| 11:22.56 | axscode | or what? |
| 11:23.22 | kink0 | yes , can comunicate |
| 11:23.33 | axscode | ahh ok.. nice... |
| 11:23.38 | kink0 | anyone know how to retrieve a variable value from the CLI ? |
| 11:24.12 | axscode | coz i have a device that only supports VoIP GateKeeper. and dont support SIP... |
| 11:24.30 | axscode | or can i make the Asterisk as Gatekeeper instead? |
| 11:24.37 | *** join/#asterisk sunil (n=sunil@202.54.37.185) |
| 11:24.40 | kink0 | h323 ? |
| 11:24.47 | robin_sz | RGHH ,,, FSCKING zyxel crap .. this wifi phone is AWFUL |
| 11:24.55 | axscode | h323.. i guess.. |
| 11:25.02 | kink0 | well really you can no use any gatekeeper |
| 11:25.22 | axscode | so.. i dont need a gatekeeper.. and use the asterisk for that? |
| 11:25.42 | kink0 | axscode, yes, but depending what you need to do |
| 11:25.42 | axscode | do you happen to know what will i need in the asterisk config? |
| 11:25.56 | robin_sz | why did I even think this wifi phone would be ok huh? |
| 11:26.17 | sunil | anybody can help me on speech synthesis |
| 11:26.30 | robin_sz | sunil: festival. |
| 11:27.00 | kink0 | axscode, do you use gk for endpoints registering ? well , you can also register peers or users, or as both ( like does the gnugk ) ussing Asterisk |
| 11:27.31 | sunil | robin_sz: i installed it but i have some problem generating the utterance file |
| 11:28.00 | robin_sz | and you read this? http://www.voip-info.org/wiki-Asterisk+Festival+installation |
| 11:28.31 | Synapes | after creating extension and creating a digital receptionist for it, when trying to call from another extension i get into an error: "486 Busy here" and the digital recpctionst won't answer, any ideas? |
| 11:31.11 | robin_sz | sigh .. this poxy poxy poxy Zyxel phone doesnt work with the version of firmware with astersik .. you cant even upgrade the firmware because the poxy ting wont work well enough to even display the web configuration/upgrade page ..and they want 75p/minute for telephone support for their defective product .. |
| 11:31.35 | sunil | robin_sz: i had followed the method 2 |
| 11:32.54 | robin_sz | sunil: and you applied the patches? |
| 11:33.01 | sunil | yep |
| 11:34.31 | robin_sz | sunil: and you are on Debian Sarge right? |
| 11:35.37 | *** join/#asterisk fizgig (n=fizgig@fizgig.xs4all.nl) |
| 11:36.08 | fizgig | howdy all |
| 11:36.50 | sunil | robin_sz: no i am on Fedora Core 2 |
| 11:37.17 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-131.buckeyecom.net) |
| 11:37.35 | robin_sz | sunil: and no rpms for FC2 are available? |
| 11:37.42 | *** join/#asterisk RoyK (n=roy@87.80-202-9.nextgentel.com) |
| 11:37.52 | robin_sz | anyway .. what is the actual problem? |
| 11:38.54 | fizgig | does anyone have experience with mISDN on asterisk? or any pointers where i could read up it. i seem to be rather stuck on this one |
| 11:41.22 | sunil | i am tryinh to generate the utterance file using the command festival -b /usr/local/festvox/build_ldom.scm '(build_utts "etc/time.data")' and its giving the following error Can't find voice scm files they are not in |
| 11:41.22 | sunil | <PROTECTED> |
| 11:41.22 | sunil | <PROTECTED> |
| 11:41.22 | sunil | <PROTECTED> |
| 11:41.23 | sunil | SIOD ERROR: nil |
| 11:41.24 | sunil | closing a file left open: /usr/local/festvox/cmu_time_awb_ldom.scm |
| 11:41.26 | sunil | closing a file left open: /usr/local/festvox/build_ldom.scm |
| 11:44.27 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 11:50.58 | coppice | http://www.samefacts.com/archives/technology_and_society_/2006/01/technology_alert.php |
| 11:53.48 | niZon | lol |
| 11:55.04 | coppice | if you see a graph of borth rate and a graph of film sales volume over the last century for any developed country, they follow each other extremely well |
| 12:02.17 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 12:04.30 | *** join/#asterisk Assid (n=assid@203.115.64.10) |
| 12:06.17 | *** join/#asterisk zock (n=zock@p54B19988.dip0.t-ipconnect.de) |
| 12:06.21 | zock | Hi. |
| 12:07.05 | zock | Quick question about asterisk... what could that mean (bristuff, zaphfc, hfc-s as TE) : |
| 12:07.09 | zock | Jan 22 07:06:19 WARNING[25598]: chan_zap.c:7586 zt_pri_error: PRI: !! Got S-frame while link down |
| 12:09.34 | zock | No idea? |
| 12:13.58 | zock | Everybody asleep? |
| 12:14.45 | kll | zock: it means what it says ;) |
| 12:15.19 | zock | kll: Hi. Is it a serious warning, or can i ignore this one? |
| 12:16.15 | kll | I've seen it on my PRI links one some occasions like when I've just disconnected the cable and put it back in |
| 12:16.47 | kll | so ofcrouse it happens when something is not really right |
| 12:16.56 | kll | but it's 'just' a warning |
| 12:17.25 | kll | it hasn't cost me any trouble as far as I can remember |
| 12:17.32 | zock | kll: Hm. I actually building up my pbx and isdn<->asterisk communication is not working... so i am just searching a reason :-) |
| 12:18.17 | kll | ah |
| 12:18.28 | zock | NT <-> HFC-S (TE) <-> Asterisk <-> Softphone is the actual config. It justs does not dial out (okay, it dails, but without any success). |
| 12:18.43 | zock | I thought this message could have something to do with that. |
| 12:20.33 | kll | I'm not certain. do you get link on your zap interface? |
| 12:20.46 | kll | do you get anything in either direction? |
| 12:20.57 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 12:21.41 | zock | kll: i am not certain... just reviewing the zaphfc-configuration... |
| 12:23.27 | *** join/#asterisk coppice (n=chatzill@248.162.17.210.dyn.pacific.net.hk) |
| 12:24.21 | zock | kll: /proc/zaptel/1 looks okay. |
| 12:24.32 | coppice | anyone here using asterisk with a flash disk system? |
| 12:25.06 | zock | kll: show modules tells "chan_zap.so Zapata Telephony w/PRI 0" and many more |
| 12:25.28 | *** join/#asterisk Mother_zzZ (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 12:25.34 | zock | kll: "Use Count 0".... hmm |
| 12:26.14 | zock | kll: zap show channels lists 3 channels "pseudo" "1" and "2". |
| 12:27.31 | zock | kll: when dialing it say: |
| 12:27.34 | zock | Executing Dial("SIP/92-7590", "Zap/g1/601807/b") in new stack |
| 12:27.41 | zock | Requested transfer capability: 0x00 - SPEECH |
| 12:27.56 | zock | Called g1/601807/b (own phone no.) |
| 12:28.01 | zock | Channel 0/1, span 1 got hangup |
| 12:28.07 | zock | Zap/1-1 is circuit-busy |
| 12:28.12 | zock | Hungup 'Zap/1-1' |
| 12:28.49 | zock | kll: So it seems asterisk tries to call out, but does not succed. |
| 12:29.19 | zock | kll: and about every 10 seconds i get an warning about this s-frame thing. |
| 12:29.47 | kll | what happens if you dial from the pbx? do you get anything on the asterisk side? |
| 12:31.23 | zock | hm...from pbx? You mean from pstn? |
| 12:31.34 | kll | ah, yes |
| 12:33.00 | zock | oh damn... my phone is dead *g*...perhaps i should use another cable? |
| 12:33.28 | zock | kll: Or is PSTN <-> NT <-> PABX _and_ Asterisk not a valid config? |
| 12:33.51 | *** join/#asterisk bofh42 (n=bofh42@p54828EB1.dip0.t-ipconnect.de) |
| 12:40.03 | zock | hm...seemed not to be the best idea to hook up the hfc-s and the pbx at the same time :-) |
| 12:42.37 | zigman | te mode ? |
| 12:43.54 | zock | zigman: hfc-s in te-mode |
| 12:44.23 | Mother_zzZ | anyone here that has chan_bluetooth working ok? |
| 12:44.46 | zigman | ptmp ? or ptp ? |
| 12:45.15 | zock | zigman: PtMP (the pbx also uses PtMP). |
| 12:45.21 | zigman | should work |
| 12:45.48 | zock | zigman: As soon as i connect the cable between hfc-s and nt i can not dial out with my pbx. |
| 12:46.00 | zock | zigman: As soon as i disconnect i can dial out again. |
| 12:46.09 | zigman | what cable do you use ? |
| 12:46.31 | zock | zigman: Its a little home pbx (4 analog channels, 1 internal S0 channel). |
| 12:46.50 | zock | zigman: straight rj45 isdn cable. |
| 12:46.57 | zigman | odd |
| 12:48.04 | zock | zigman: yeah. |
| 12:48.43 | zigman | can you dial out with your pbx ? |
| 12:48.45 | zock | zigman: signalling = bri_net_ptmp |
| 12:48.55 | zigman | bri_cpue_ptmp |
| 12:49.00 | zigman | bri_cpe_ptmp |
| 12:49.04 | zigman | its TE .. remember |
| 12:49.16 | zigman | net is NT |
| 12:49.21 | zock | hm...cpe...hm |
| 12:50.07 | zock | k..changed, started... and now i will go downstairs to hook it up again :-) |
| 12:50.20 | zigman | go for it ;) |
| 12:50.26 | kink0 | I do ussing AMI Setvar variable=1 but folows a Getvar and then return the value 0 , any idea ? |
| 12:50.45 | zigman | kink0 paste full row |
| 12:52.36 | zock | zigman: hm... dial out on pbx is working now... |
| 12:53.07 | kink0 | zigman, ahh ok... appears was some spaces in variable name, as I did cut and paste the commands ... |
| 12:53.22 | zock | zigman: dial out on asterisk does not work...hm |
| 12:53.23 | zigman | zock ;) |
| 12:53.28 | kink0 | Setting global variable 'mivariable ' to '1 ' |
| 12:53.28 | kink0 | <PROTECTED> |
| 12:53.29 | zigman | not? |
| 12:53.47 | kink0 | that appears to be the failure, when I paste, there some extra spaces |
| 12:54.25 | zock | zigman: calls seem to come in... even if the get not handled correctly. |
| 12:54.27 | zock | zigman: Extension 's' in context 'default' from '6053601807' does not exist. Rejecting call on channel 0/2, span 1 |
| 12:54.37 | zock | zigman: But communication seems to be working now. |
| 12:55.01 | zock | zigman: Thanks... that error came from an example configuration i typed of from a website :-P |
| 12:55.29 | zock | zigman: So it's cpe if the channel is for te-mode and net if the channel is in te-mode, right? |
| 12:55.54 | zigman | yes |
| 12:56.03 | zock | zigman: Called g1/601807/b |
| 12:56.05 | zigman | ehh last te is nt |
| 12:56.11 | zock | zigman: Channel 0/1, span 1 got hangup |
| 12:56.23 | zock | zigman: typo...meant nt. |
| 12:56.29 | zigman | cat /proc/zaptel/* |
| 12:56.51 | zock | Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" AMI/CCS |
| 12:56.51 | zock | <PROTECTED> |
| 12:56.51 | zock | <PROTECTED> |
| 12:56.51 | zock | <PROTECTED> |
| 12:57.10 | zock | Just one card at the moment. |
| 12:57.16 | zigman | kk |
| 12:58.18 | *** join/#asterisk pengyong (n=lala@222.185.17.83) |
| 12:58.53 | *** join/#asterisk miguel (n=xc@217.116.243.18) |
| 12:59.00 | zigman | not sure |
| 13:00.16 | zock | zigman: Hm...do i need to provide a msn for outgoing calls? Perhaps asterisk sends the internal extension as msn and the pstn does not allow this? |
| 13:01.00 | zigman | depends on your telco |
| 13:01.09 | zigman | where are you from and what telco do you have ? |
| 13:01.18 | zock | zigman: germany "Deutsche Telekom". |
| 13:01.25 | zigman | dann machts nix ;) |
| 13:01.34 | zock | zigman: hmpf |
| 13:03.34 | zock | zigman: if i dial in i also get the voicemailbox :-) so communication pstn -> asterisk is working. also dtmf. |
| 13:03.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 13:04.21 | zigman | its only a config stuff then |
| 13:04.27 | zigman | thats you turn ;) |
| 13:04.30 | zock | zigman: hm... |
| 13:04.30 | zigman | your |
| 13:04.55 | zock | zigman: exten => _0.,1,Dial(Zap/g1/${EXTEN:1}/b) ... the only line that handles outgoing calls. |
| 13:05.03 | zigman | whats the /b ? |
| 13:06.38 | zock | zigman: hm...should be "early b3 connect" |
| 13:06.57 | *** join/#asterisk ToTo (n=ToTo@host197-231.pool870.interbusiness.it) |
| 13:07.11 | zigman | ahh a bristuff thing |
| 13:07.32 | zock | zigman: dunno... its state like that in a magazine. |
| 13:07.53 | zigman | pfff |
| 13:08.00 | zock | zigman: ARGH...without /b its working :-P |
| 13:08.09 | RoyK | - --- .-. -. .. -. --. |
| 13:08.22 | zock | ARGH |
| 13:08.29 | RoyK | wtf is /b? |
| 13:08.30 | zock | *rip rip rip*...damn article. |
| 13:08.36 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 13:08.46 | zigman | RoyK ? |
| 13:08.53 | zigman | you are not banned anymore ? ;09 |
| 13:09.06 | RoyK | me? |
| 13:09.52 | zigman | yeah |
| 13:09.55 | zigman | YOU |
| 13:09.59 | Jammy | heh |
| 13:10.00 | zock | then now it's just a matter of configuring the extensions. |
| 13:10.10 | zigman | zock have fun ;) |
| 13:10.38 | miguel | anybody have info about status of the Digium cards with OpenBSD? |
| 13:11.39 | RoyK | banned? damned? from where? |
| 13:12.26 | zigman | RoyK never mind... i guess its old stuff |
| 13:12.45 | RoyK | quite so |
| 13:14.01 | zock | zigman: Any suggestion on a url for explaining how to configure handling on a single specific incoming msn? "s" seems to handle ALL incoming calls. |
| 13:18.07 | Jammy | hmmm it should only pick up on what u tell it to, unless u have channels grouped |
| 13:18.38 | *** join/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 13:19.00 | xphreak | hi I'm an * newcomer |
| 13:19.26 | xphreak | is it possible to initiate a call from within asterisk without having a channel number |
| 13:19.46 | xphreak | thus make asterisk call a phone number |
| 13:20.38 | Jammy | yup i guess... |
| 13:20.51 | xphreak | do you know how ? |
| 13:21.17 | xphreak | I have looked at originate that does not suite me cause it requires a channel ID |
| 13:27.49 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 13:29.29 | ariel_ | xphreak, look on the wiki for call file you can drop a call file in and asterisk will do the calling for you. |
| 13:29.32 | ariel_ | ~docs |
| 13:29.38 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 13:29.45 | ariel_ | morning everyone |
| 13:30.20 | xphreak | ariel_: thank you will do that |
| 13:30.26 | xphreak | OK a simple one now |
| 13:30.36 | xphreak | how can I make a call using the * CLI ? |
| 13:30.40 | tzafrir_laptop | xphreak, call to where? |
| 13:30.43 | *** part/#asterisk chalco (n=chatzill@pdpc/supporter/active/chalco) |
| 13:30.43 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 13:30.50 | tzafrir_laptop | xphreak, you always have the channel Local/ |
| 13:31.04 | xphreak | just call an extension defined in the dilalplan |
| 13:31.16 | xphreak | I'm using asterisk 1.2.0 |
| 13:31.22 | *** join/#asterisk dimas (n=ds@84.53.210.46) |
| 13:32.03 | robin_sz | sigh ... OK, using receivefax() in * .. theres an AGI script called 'fax.php' refferred to in the docs .. but no clue as to what it does or where to get it from .. clues? |
| 13:33.29 | robin_sz | and ... HTF do you test an AGI script? |
| 13:33.38 | kll | xphreak: dial <extension>@<context> |
| 13:34.41 | kll | that will connect /dev/dsp to the call. and since most asterisk servers don't have a dsp device (ie, sound card) you won't hear anything. but after the call connects you can do a: transfer <extension>@<context> to transfer the call |
| 13:35.04 | zock | bbl |
| 13:35.33 | xphreak | here is what I've got on initiating the command dial |
| 13:35.34 | xphreak | *CLI> dial 2000@xphreak |
| 13:35.34 | xphreak | No such command 'dial' (type 'help' for help) |
| 13:35.46 | kll | xphreak: you don't have the correct module loaded |
| 13:35.53 | kll | it's pbx_something |
| 13:36.16 | kll | pbx_functions.so |
| 13:36.19 | kll | that one I beleive |
| 13:36.23 | xphreak | should I recompile asterisk or just load the module manually ? |
| 13:36.36 | Flauto | good morning guys |
| 13:36.41 | robin_sz | morning |
| 13:36.42 | Flauto | i have question |
| 13:36.47 | Flauto | hi robin |
| 13:36.51 | Flauto | good morning |
| 13:37.10 | robin_sz | 42 |
| 13:37.15 | ariel_ | the dial command is loaded from the start. |
| 13:37.22 | Flauto | i wanted to setup 911 to go through a zap connection in my extensions.conf |
| 13:37.33 | Flauto | but after i set it up |
| 13:37.40 | Flauto | it does not even dial |
| 13:37.46 | ariel_ | how did you set it up |
| 13:38.17 | Flauto | exten => 911,1,Dial(${DIALOUTANALOG}/${EXTEN}) |
| 13:39.04 | *** join/#asterisk razu (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 13:39.25 | ariel_ | Flauto, is your analog channel zap/1? |
| 13:39.32 | Flauto | yes |
| 13:40.09 | ariel_ | exten => _911,1,Dial(Zap/1/${EXTEN},20) |
| 13:40.23 | Flauto | okay |
| 13:40.25 | Flauto | let me try |
| 13:40.51 | ariel_ | this should be ahead of all your other rules. |
| 13:41.00 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 13:41.38 | kll | you could even use: exten => _911!,........ |
| 13:41.45 | tzafrir_laptop | dial is for interactive operation |
| 13:42.14 | tzafrir_laptop | If you want to initiate a call, use a Local/ channel and initiate it from a call file or from the manager interface |
| 13:45.48 | Flauto | ariel, samething. still does not work |
| 13:46.00 | Flauto | it gave me nothing for a while and then, busy tone |
| 13:46.14 | Flauto | on my asterisk, it does not even show that i was trying to call |
| 13:46.41 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 13:47.28 | Flauto | i have DIALOUTANALOG=Zap/1 in global so it should be the same way that they way you dial and my old way |
| 13:48.07 | Flauto | the only thing that i would think is that i have a dial partern setup with a prefix _9., later for another service |
| 13:48.43 | Flauto | but the thing is that other service is using a patern dial |
| 13:50.31 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 13:50.57 | xphreak | OK about the call file |
| 13:51.01 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 13:51.04 | xphreak | the call file needs channel |
| 13:51.21 | xphreak | I don't have a channel number since I have to make a call from the * server |
| 13:52.30 | xphreak | I'm writing an application that on a receipt of a message should call an appropriate number wait for that number to pick up and then call another number wait to pick up and then connect the two numbers |
| 13:52.37 | Flauto | ariel, are you there |
| 13:52.53 | ariel_ | Flauto, what do you get on the cli when you make the call |
| 13:53.13 | Flauto | ariel, nothing at all on cli |
| 13:53.31 | ariel_ | nothing?? there got to be something |
| 13:53.49 | Flauto | ariel, that is the strange part |
| 13:53.52 | Flauto | i got nothing |
| 13:54.00 | Flauto | absolutely nothing |
| 13:54.11 | Flauto | i tried 411 too |
| 13:54.14 | Flauto | i got the samething |
| 13:54.27 | ariel_ | set verbose 9 |
| 13:55.04 | Flauto | it says now verbose is 9 |
| 13:56.07 | ariel_ | try the call again |
| 13:56.17 | Flauto | samething |
| 13:56.18 | ariel_ | there has to be something on the cli |
| 13:56.24 | Flauto | nothing at all |
| 13:56.41 | ariel_ | then your device is not accessing the correct context |
| 13:56.43 | Flauto | i don't see anything at all |
| 13:56.52 | Flauto | but everythign else is working |
| 13:56.59 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 13:57.00 | xphreak | I'm using the Manager API, so when I get a JMS message from the application server I have to initiate a call to an agent |
| 13:57.00 | xphreak | how can I do that |
| 13:57.00 | xphreak | ? |
| 13:58.13 | xphreak | can anyone help me with this one ? |
| 13:58.27 | Flauto | ariel, what should i do then |
| 13:58.51 | Flauto | i even rebuilt asterisk a few days ago |
| 13:58.51 | Flauto | but the same problem is still there |
| 13:59.01 | *** join/#asterisk Drew (n=foo@zux221-158-032.adsl.green.ch) |
| 13:59.17 | Flauto | what i hear from the phone is just quiet for a while and then, busy tone |
| 13:59.49 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 14:01.04 | ariel_ | Flauto, but you said you see nothing on the cli |
| 14:01.19 | ariel_ | if you remove the rule does it try to send the call out your other trunk? |
| 14:02.18 | Flauto | let me try |
| 14:03.03 | *** join/#asterisk zeedo (n=zeedo@80-192-53-14.stb.ubr04.glen.blueyonder.co.uk) |
| 14:03.27 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:04.37 | Flauto | ariel, it does not make any difference |
| 14:04.43 | Flauto | i really don't know what to do |
| 14:06.33 | Flauto | let me exclude that _9 patern and to see what will happen |
| 14:07.49 | Flauto | still nothing at all |
| 14:08.47 | Flauto | ariel, any idea? |
| 14:10.33 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 14:11.42 | Z_God | hi all |
| 14:12.00 | ariel_ | Flauto, I would need to see your extensions.conf. But if your context are correct and your includes are correct the cli should display at least a dial command. |
| 14:12.29 | Z_God | I've got a problem getting asterisk to work with my ISDN card |
| 14:12.36 | tzafrir_laptop | xphreak, call from what to an agent? |
| 14:12.50 | Z_God | when I try to call asterisk, it doesn't seem to 'pick up' |
| 14:13.03 | tzafrir_laptop | Who is on the othe side of the call? |
| 14:13.14 | Z_God | any idea how I can check if asterisk uses my card? |
| 14:13.18 | ariel_ | I have to go out for a while. I should be back in a few hours. It's sunday and I have a list of honey doo's |
| 14:13.33 | tzafrir_laptop | Z_God, what type of asterisk channel do you use? what version of asterisk? |
| 14:13.48 | xphreak | tzafrir_laptop: I have no dial command I have just checked, I'm using asterisk 1.2.0, why is that ? |
| 14:14.07 | xphreak | on my laptop I use asterisk as well and have the dial command |
| 14:14.23 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:14.36 | Z_God | version from debian 3.1, sarge (I believe 1.0.7) with the modem channel |
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| 14:15.05 | *** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net) |
| 14:15.27 | Z_God | Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k |
| 14:16.13 | zishanov | My Music On Hold is distorted, need help. I've tried everything. I have a SIP phone line. SIP to SIP it is good but when call comes in from PSTN, it is distorted. |
| 14:17.02 | Z_God | my ISDN card uses the HiSax driver, (Eicon Diva 2.01 ISA) |
| 14:17.06 | Flauto | ariel, http://pastebin.ca/38004 |
| 14:17.31 | Z_God | I had everything working, but I had to reinstall due to a hd-crash |
| 14:17.38 | zishanov | Is there any detailed help on the Internet about Asterisk MoH? |
| 14:17.41 | Flauto | hey, ariel, thank you very much |
| 14:18.25 | xphreak | can anyone tell me why I cannot use the dial command from the CLI ?? |
| 14:18.59 | zishanov | xphreak, you can't use the dial command when your Linux hasn't installed sound card drivers |
| 14:19.20 | Flauto | xphreak, try to dial the number only without anything else |
| 14:19.33 | xphreak | I have the sound card drivers installed |
| 14:19.36 | xphreak | and they work |
| 14:19.39 | xphreak | OK |
| 14:19.58 | zishanov | dial works like 'Dial 201@internal' |
| 14:20.05 | zishanov | or whatever context it has |
| 14:20.28 | xphreak | *CLI> Dial 2000@xphreak |
| 14:20.29 | xphreak | No such command 'Dial' (type 'help' for help) |
| 14:20.37 | xphreak | here is the output for that command |
| 14:20.47 | xphreak | I have tried dial 2000 |
| 14:20.53 | xphreak | and it worked on the laptop |
| 14:20.59 | xphreak | but not on my desktop computer |
| 14:21.07 | xphreak | I get the same error |
| 14:21.13 | zishanov | xphreak, if you have drivers installed, Asterist doesn't know about them |
| 14:21.16 | xphreak | No such command 'dial' (type 'help' for help) |
| 14:21.23 | xphreak | hmm ... |
| 14:21.25 | zishanov | I have two linux systems, on one I use dial and on other I can't |
| 14:21.46 | zishanov | I know why these messages appear |
| 14:22.11 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-86.claranet.co.uk) |
| 14:22.14 | zishanov | try to modify modules.conf, I'll write in a but how |
| 14:22.32 | xphreak | what to modify |
| 14:22.59 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:24.29 | zishanov | in modules.conf, change ;noload => chan_oss.so to load chan_oss.so |
| 14:24.36 | xphreak | I have solved the problem |
| 14:24.45 | zishanov | what did you do? |
| 14:25.01 | xphreak | I have loaded chan_oss.so but I needed chan_alsa.so |
| 14:25.09 | xphreak | thanks for the tip zishanov |
| 14:25.11 | zishanov | thats right |
| 14:25.54 | zishanov | now can anybody help me on my MoH issue. I haven't slept all night trying to fix it. Now I want to sleep but can't, until it'll be fixed |
| 14:26.17 | zishanov | Music is distorted when call comes in from cell phone or any PSTN line, why is that |
| 14:29.35 | *** part/#asterisk gushi (i=danm@prime.gushi.org) |
| 14:29.41 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
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| 14:34.53 | nassy | zigman: i dont know. im new to asterisk. my guess: volume too loud |
| 14:35.58 | *** join/#asterisk wizard545 (n=wizard@tor/session/x-615631185c71cf22) |
| 14:36.43 | zishanov | volume is ok, because from other SIP lines it is clear |
| 14:37.11 | nassy | oh ok, maybe something to do with the codec or codec translation |
| 14:37.59 | Err | from a cell phone it could very easily be re-compression (since the cell phone system re-compresses the audio) |
| 14:38.19 | Err | landline phones don't necessarily digitally compress the audio, but they might - which could cause the audio to sound like crap as well |
| 14:38.55 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 14:38.58 | zishanov | what are the solutions? Is it good to use native MoH. If yes, then how to do I use it |
| 14:39.05 | *** part/#asterisk xphreak (n=zsolti@ns1.zrlocal.net) |
| 14:40.00 | Err | the only solution that I can think of is to use less compression on your links, so that the quality is degraded less on your end |
| 14:40.46 | *** join/#asterisk telmich (i=telmich@creme.schottelius.org) |
| 14:40.47 | telmich | hello |
| 14:40.51 | nassy | hello |
| 14:40.57 | zishanov | What's the difference between native MoH and regular MoH? |
| 14:41.11 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:41.19 | telmich | I've the problem that some people can't reach other people. I got that in my logs: Jan 22 15:42:14 NOTICE[18436]: chan_iax2.c:6775 socket_read: Rejected connect attempt from 212.59.62.244, who was trying to reach '1001@' |
| 14:41.27 | telmich | can you explain that to me |
| 14:42.01 | nassy | i cant, im not familar enough with asterisk. id guess your dial plan blocked them though |
| 14:42.42 | Drew | im having some problems with recieving incoming sip calls - the asterisk box is behind a nat firewall - do i have to forward the rtp port range aswell as 5060-5061 two the asterisk? and if so what is a sensible portrange to use? im not going to forward 10k ports... |
| 14:42.43 | telmich | exten => 1001,1,Dial(IAX2/nico,,rm) ; ring without time limit |
| 14:42.52 | telmich | that's the only entry I've for 1001 |
| 14:44.40 | *** part/#asterisk dimas (n=ds@84.53.210.46) |
| 14:45.28 | nassy | Drew: i dont know. if you dont get help here try the voip wiki they may have some suggestions in the asterisk section |
| 14:45.52 | nassy | telmich: i dont know, sorry. try turning on debugging on the cli |
| 14:46.14 | kink0 | Drew, try sip debug and see if messages arrives and also are sent without retrys |
| 14:46.57 | telmich | nassy: I've that message from the cli |
| 14:47.00 | kink0 | Drew or use tcpdump to see, I am also behind nat, but I did not need to touch my router config to pass ports, even most ports was closed. |
| 14:47.04 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 14:47.47 | markit | hi, the sound "inithelp.gsm" is listed in the sounds.txt, but I can't find it in the installed set... is it a bug of the sounds.txt or of the installation set? (svn 1.2.x stable) |
| 14:49.02 | Drew | kink - its not about incoming from the sip provider - that works (ie . caller uses POTS phone to call the number at my sip/voip provider) but incoming sip calls from ppl using my ip address (ie user@myname.dyndns.org) dont work |
| 14:50.03 | *** join/#asterisk bweschke (i=bweschke@dhcp64-134-228-208.hrp66.orl.wayport.net) |
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| 14:53.49 | kink0 | Drew what you got when a call to your IP arrives ? nothing ? even nothing debuging SIP ? |
| 14:54.51 | robin_sz | OK, so I have outgoing via voip gate going OK, incoming via ISDN 8 channels on an Eicon, 30 extesnions, 2 receptionists, mailboxes and forwarding all working ... |
| 14:55.08 | robin_sz | just the sodding FAX now ... |
| 14:55.44 | robin_sz | actually, no, just the fax and the emergency phone in the lift |
| 14:56.13 | Drew | gives me some kind of timeout error on the console and the caller gets an unavailable message |
| 14:56.19 | robin_sz | oh, and the lack of dancing girls |
| 15:00.27 | Drew | lol - who needs the emergancy phone to work?? ^^ |
| 15:00.51 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
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| 15:43.58 | neon_kl | hello everybody |
| 15:47.27 | tzafrir_laptop | hi |
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| 16:08.52 | Z_God | does anyone here know what command I can use in asterisk to see if it finds my modem (ISDN) channel correctly? |
| 16:09.17 | Z_God | asterisk doesn't seem to 'pick up' when I try to call it |
| 16:10.01 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:10.32 | bkw_ | give up |
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| 16:17.44 | sandra78 | hi does anybody knows if can be used 2 TDM04b in the same PC? |
| 16:18.01 | blop | -- Executing ChanIsAvail("Zap/1-1", "Zap/3&Zap/2") in new stack |
| 16:18.01 | blop | -- Hungup 'Zap/3-1' |
| 16:18.01 | blop | -- Executing Dial("Zap/1-1", "Zap/3-1|180|r") in new stack |
| 16:18.01 | blop | Jan 22 11:58:04 WARNING[18797]: chan_zap.c:7608 zt_request: Unknown option '-' in '3-1' |
| 16:18.09 | blop | => how can i fix this warning ? |
| 16:18.27 | *** join/#asterisk uneuronh (n=uneuronh@202.142.93.79) |
| 16:19.30 | *** join/#asterisk coppice_ (n=chatzill@96.157.17.210.dyn.pacific.net.hk) |
| 16:20.47 | inv_Arp | blop: errr, remve the unknow option? |
| 16:20.58 | robin_sz | blop the asnwer is in the message |
| 16:21.37 | blop | yeah but |
| 16:22.00 | blop | the dial is using the result of ChanIsAvail(), its ChanIsAvail which add a - in it |
| 16:22.28 | blop | i got |
| 16:22.28 | blop | exten => s,14,ChanIsAvail(Zap/3&Zap/2) |
| 16:22.28 | blop | exten => s,15,Dial(${AVAILCHAN},180,r) |
| 16:22.33 | *** join/#asterisk ToTo (n=ToTo@host197-231.pool870.interbusiness.it) |
| 16:22.42 | uneuronh | well is there any asterisk billing software done in PHP ? |
| 16:22.59 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 16:26.41 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 16:27.09 | *** join/#asterisk mallum (n=mallum@host86-136-17-6.range86-136.btcentralplus.com) |
| 16:27.19 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:27.42 | mallum | anyone have an example config of using asterisk as a sip<->iax gateway ? |
| 16:27.49 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 16:29.48 | *** join/#asterisk areski (n=areski@85.Red-83-34-12.dynamicIP.rima-tde.net) |
| 16:35.29 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:37.44 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 16:41.27 | RoyK | -.-. --.- |
| 16:41.37 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 16:42.14 | Druken | morning everyone |
| 16:43.52 | *** join/#asterisk hiha-ZzZz[20f] (n=NaSIr@85.108.149.247) |
| 16:44.13 | [TK]D-Fender | RoyK : CQ? |
| 16:45.07 | *** part/#asterisk uneuronh (n=uneuronh@202.142.93.79) |
| 16:45.12 | wasim | Cows Quarterly |
| 16:45.12 | RoyK | old calling in morse language |
| 16:45.22 | RoyK | seek you |
| 16:45.25 | RoyK | ~lart wasim |
| 16:45.35 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:45.50 | [TK]D-Fender | ah |
| 16:46.01 | RoyK | wasim: hehe. the .no ambassedor from .pk beat up a pakistani visitor today.... |
| 16:46.08 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 16:46.15 | wasim | RoyK: good start to the visit, eh? |
| 16:46.49 | RoyK | heh |
| 16:47.28 | Druken | cq, cq, this is wx409 calling tr576, cq, cq.... |
| 16:48.44 | Druken | ya know what i hate about web design? |
| 16:48.49 | Druken | the design part..... |
| 16:50.38 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 16:51.12 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 16:54.17 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 16:56.44 | robin_sz | Druken: the web part sucks too |
| 16:56.57 | Druken | not nearly as much as the design.... |
| 16:57.03 | robin_sz | shrug |
| 16:57.07 | sandra78 | hi does anybody knows if can be used 2 TDM04b in the same PC? |
| 16:57.08 | Druken | but i have no artistic value... so.. :) |
| 16:57.09 | robin_sz | these days, use Joomla |
| 16:59.29 | ManxPowe | sandra78, you should be able to. |
| 16:59.36 | ManxPowe | I would not recommend more than 2 |
| 17:05.01 | *** join/#asterisk SERGEUS (n=s@195.112.98.13) |
| 17:05.05 | *** join/#asterisk zock (n=zock@p54B197CD.dip0.t-ipconnect.de) |
| 17:05.13 | zock | re. |
| 17:07.10 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:09.23 | zock | kll: asterisk is working now as supposed. Next Task... using a second card for nt-mode ... but thats a task for next week ;-) |
| 17:09.33 | *** join/#asterisk ToTo (n=ToTo@host197-231.pool870.interbusiness.it) |
| 17:09.50 | kll | :) |
| 17:10.26 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:10.49 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 17:14.22 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:14.55 | *** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br) |
| 17:16.55 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 17:17.12 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 17:17.44 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:18.32 | *** part/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 17:19.13 | tekati | Can * detect distinctive ringing on a SIP line? The reason I ask is I have broadvoice with an add on line. It does not send any type of DNIS info so I can not determine which number is ringing. Broadvoice can however send distinctive ring tones so if I could intercept that I could tell which line was ringing. Or does anyone have any other ideas as well? |
| 17:20.19 | *** join/#asterisk mrdigital (n=mrdigita@pool-68-236-41-109.phil.east.verizon.net) |
| 17:20.40 | mrdigital | nice! i just hired to setup a asterisk server with 10 phones and 2 door buzzers my pay = $2,000 |
| 17:20.56 | mrdigital | *got |
| 17:21.39 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-223-184-33.mia.bellsouth.net) |
| 17:21.48 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:21.49 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-223-184-33.mia.bellsouth.net) |
| 17:24.15 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 17:24.38 | *** join/#asterisk fa_back (i=faceoff@217.153.189.28) |
| 17:24.46 | QbY | Is there something that would cause my PAP2 to not receive calls after a long period of idleness. But if I pick up the phone I can make a call right out? |
| 17:25.05 | nassy | where are you located, mrdigital |
| 17:25.53 | mrdigital | NY |
| 17:26.01 | mrdigital | this system is for a Doctors Office |
| 17:26.11 | *** join/#asterisk roulduke_ (i=o174vgg9@p508D1FDB.dip0.t-ipconnect.de) |
| 17:27.05 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:27.10 | nassy | ah ok, do you do migrations. i am in nyc and may need your services sometime in the future |
| 17:27.33 | nassy | i plan to propose migrating from our current toshiba strata ctx system to asterisk |
| 17:27.48 | mrdigital | pm me nassy |
| 17:28.16 | nassy | are you going to be here for a bit. i have to finish up an email |
| 17:28.21 | mrdigital | yea |
| 17:29.53 | [TK]D-Fender | QbY : Is your PAP2 local to your *? |
| 17:29.54 | redder86 | In the development of 1.2 from 1.0, Asterisk applications SetVar changed to Set, SetCID.. changed to Set(CALLERID(...)). The file UPGRADE.txt does not indicate, however, if these changes affected the Asterisk Call Manager syntax. Does anyone know? |
| 17:31.33 | QbY | TK. NO.. |
| 17:31.33 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc08e.dialup.mindspring.com) |
| 17:31.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:31.56 | [TK]D-Fender | QbY : Then you need to set the NAT keep-alive stuff in sip.conf and on the PAP2. |
| 17:32.07 | QbY | I have NAT=yes |
| 17:32.11 | QbY | what is the other keep alive? |
| 17:32.19 | [TK]D-Fender | QbY : qualify=yes |
| 17:32.30 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:32.36 | [TK]D-Fender | and on the PAP2, you might need to tell it you're behind NAT |
| 17:32.45 | QbY | ok, let me see |
| 17:33.09 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc08e.dialup.mindspring.com) |
| 17:34.09 | SERGEUS | i want to hide my asterisk from SIP provider :) is there any way to change asterisk's SIPUSERAGENT from dialplan, before calling DIAL(...) ? |
| 17:34.30 | redder86 | sorry, I meant AGI |
| 17:34.43 | redder86 | AGI syntax, not manager API syntax |
| 17:35.10 | *** join/#asterisk jcwunder (n=chris@ppp-82-135-2-145.mnet-online.de) |
| 17:36.05 | QbY | Is there a key sequence for hold? on an Analog phone. |
| 17:36.29 | jcwunder | おはよう |
| 17:36.45 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 17:36.55 | jcwunder | だらもわからないの？ |
| 17:37.09 | *** join/#asterisk Lurr_ (n=pr0ph3t@adsl-223-184-90.mia.bellsouth.net) |
| 17:37.28 | Qwell | wow, that question mark is excessively large... |
| 17:37.39 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 17:37.43 | *** part/#asterisk Lurr_ (n=pr0ph3t@adsl-223-184-90.mia.bellsouth.net) |
| 17:37.46 | PoWeRKiLL | hi |
| 17:38.01 | PoWeRKiLL | someone know why to correct this warning Jan 22 18:38:33 WARNING[25837]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x837b228', 10 retries! ? |
| 17:39.53 | [TK]D-Fender | QbY : The PAP2 has its own means of signalling SIP functionality like that through use of hook-flash & * codes. Look in your config as to how its set up. |
| 17:41.48 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:42.02 | QbY | i don't see one for Hold.. Would it be called something else in SIP Terms |
| 17:44.00 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 17:44.59 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 17:45.14 | frenzy | hey ya all ! |
| 17:47.28 | *** join/#asterisk aconda (n=anaconda@p5496F097.dip.t-dialin.net) |
| 17:48.36 | frenzy | what does this mean?? Jan 22 17:48:22 WARNING[12327]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/7777777-c9cb' |
| 17:51.58 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 17:53.02 | frenzy | ? |
| 17:54.00 | tekati | Can * detect distinctive ringing on a SIP line? The reason I ask is I have broadvoice with an add on line. It does not send any type of DNIS info so I can not determine which number is ringing. Broadvoice can however send distinctive ring tones so if I could intercept that I could tell which line was ringing. Or does anyone have any other ideas as well? |
| 17:55.05 | Math[laptop] | tekati: it should call the number which is ringing... |
| 17:56.09 | Katty | beep! |
| 17:56.23 | frenzy | bepp bepp |
| 17:57.59 | tzanger | katty |
| 17:58.00 | tzanger | katty |
| 17:58.05 | tzanger | katty |
| 17:58.13 | tzanger | :-) |
| 17:58.24 | frenzy | whhhhooooh |
| 17:58.55 | frenzy | tzanger: *blush* |
| 17:59.59 | tekati | Math: When Broadvoice sends the call it uses the primary number regardless of which number you call. I am guessing this is because it is the primary number that the sip registers too. |
| 18:00.14 | ManxPowe | there is no such thing as "distinctive ring" in SIP |
| 18:00.27 | ManxPowe | tekati, get multiple accounts. |
| 18:00.35 | Qwell | there are ALERT_INFO headers, but that's about it |
| 18:00.42 | Qwell | I doubt they send those |
| 18:00.58 | tekati | There must be or Broadvoice is a freak as that is setup option. |
| 18:00.58 | ManxPowe | tekati, remove the /number from your register line |
| 18:01.12 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:01.32 | tekati | Alternate Numbers Allow up to two additional phone numbers, with each number having a distinctive ringing pattern. |
| 18:01.55 | tekati | That comes from the configuration screen on their site. |
| 18:02.01 | tekati | Mine is turned on. |
| 18:02.04 | ManxPowe | tekati, that is only for use with an ATA. |
| 18:02.11 | Math[laptop] | do a SIP debug trace and check for ALERT_INFO |
| 18:02.15 | tekati | Ah now see that could be. |
| 18:02.22 | tekati | Math sounds reasonable stand by. |
| 18:02.30 | *** join/#asterisk coppice_ (n=chatzill@135.201.17.210.dyn.pacific.net.hk) |
| 18:02.37 | ManxPowe | tekati, if you remove the /number from your register line BV might do what you want it to. |
| 18:06.01 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 18:06.34 | tekati | INVITE sip:6615552121@66.75.215.63 SIP/2.0 |
| 18:06.34 | tekati | Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qkn8l300gk1v9gub301.1sr |
| 18:06.34 | tekati | From: "6615551212"<sip:6615551212@147.135.0.129;user=phone>;tag=SD2g75c01-575806738-1137952974682 |
| 18:06.34 | tekati | To: "Not Me"<sip:6615552121@sip.broadvoice.com;user=phone> |
| 18:06.34 | tekati | Call-ID: SD2g75c01-6ffb0fefa116f99a70e1356f321bd46a-js11002 |
| 18:06.35 | tekati | CSeq: 967062446 INVITE |
| 18:06.37 | tekati | Contact: <sip:6615551212@147.135.0.128:5060;ep=147.135.0.129;transport=udp> |
| 18:06.39 | tekati | Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY |
| 18:06.41 | tekati | Supported: 100rel |
| 18:06.43 | tekati | Accept: application/sdp,application/dtmf |
| 18:06.45 | tekati | Max-Forwards: 69 |
| 18:06.47 | tekati | Content-Type: application/sdp |
| 18:06.49 | tekati | Content-Length: 275 |
| 18:06.50 | JunK-Y | tekati: use pastebin |
| 18:06.51 | JunK-Y | ~pb |
| 18:06.54 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:06.54 | ManxPowe | DO NOT FLOOD THE CHANNEL! |
| 18:07.23 | ManxPowe | tekati, is that with /number on register => or without it? |
| 18:08.00 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:10.49 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 18:11.18 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 18:14.38 | *** join/#asterisk mne (n=mne@chello080108001212.35.11.tuwien.teleweb.at) |
| 18:17.55 | tekati | ManxPower: register => 6615551212@sip.broadvoice.com:eas7yjqre6:6615551212@sip.broadvoice.com/6615551212 |
| 18:18.00 | *** join/#asterisk pb__ (n=pb@cpc1-cmbg6-5-0-cust20.cmbg.cable.ntl.com) |
| 18:18.23 | tekati | That is the line if I try to remove the last part of the number it tells me when I reload SIP that the line is incorrect and I get a fast busy when trying to call the number. |
| 18:19.04 | QbY | tekati.. what version of asterisk are your urnning? |
| 18:19.13 | QbY | running |
| 18:19.13 | tekati | 1.2.1 I believe. |
| 18:19.49 | QbY | the registration syntax changed.. register=>6153491013:password@sip.broadvoice.com/6153491013 |
| 18:20.07 | tekati | I can try that. |
| 18:20.24 | Qwell | but do it without the /615blah at the end |
| 18:20.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:21.08 | ManxPowe | tekati, the /6615551212 tells BroadVoice "Please send all calls for this account to my exten => 6615551212 line/. |
| 18:22.15 | tekati | So should it just be: register => 6615551212:password@sip.broadvoice.com |
| 18:22.52 | QbY | depends on the rest of your dial plan |
| 18:23.15 | riddlebox | did anyone else have problems with broadvoice not registering this weekend? |
| 18:23.32 | Qwell | riddlebox: Did the day end in a 'y'? |
| 18:23.33 | QbY | i did when i upgraded to 1.2 |
| 18:23.36 | Qwell | if so, yes |
| 18:23.39 | QbY | that's how i knew of the new registration syntax |
| 18:24.36 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 18:25.03 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 18:25.34 | tekati | Okay I did that and setup the extension information correctly and it still comes through as the main number regardless of what number I called. |
| 18:25.55 | *** join/#asterisk joonysam (n=a@brom-245-243.flexabit.net) |
| 18:26.54 | joonysam | can anyone tell me which ports to map for a asterisk box behind a windows ICS NAT? |
| 18:27.19 | joonysam | I have 5060, 16384, 16394, and 8000, all UDP mapped out to the internal IP |
| 18:28.48 | [TK]D-Fender | joonysam : you should have from 10000-20000 mapped |
| 18:29.08 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:29.34 | trixter | default rtp.conf is 10000-20000 but that is easily changed |
| 18:29.38 | joonysam | hmm yeah, that's going to be a bit tricky, you can only map one port at a time in ICS :( |
| 18:29.57 | joonysam | is 10000-20000 for the audio channel? |
| 18:30.08 | trixter | and that also assumes that your internal devices dont directly bridge out, which is controlled through sip.conf |
| 18:30.17 | joonysam | a PC with x-lite on a public IP without a firewall can call, but not transmit audio, but hears the other side |
| 18:30.22 | trixter | joonysam: yes the rtp layer which does audio and other stuff |
| 18:30.54 | trixter | and it may be that depending on how you do your firewall you may need to allow other stuff ... |
| 18:31.05 | trixter | *most* people set up a firewall to restrict nothying out but limit what can go in |
| 18:31.12 | joonysam | I see |
| 18:31.29 | tekati | It does appear that the distinctive ring comes in the form of: Alert-Info: <http://127.0.0.1/Bellcore-dr3> |
| 18:31.49 | trixter | in that instance you can let udp into 5060 (default in sip.conf) and 10000-20000 (default rtp.conf) and let everything out go, and it should work |
| 18:32.15 | tekati | It does appear that the distinctive ring does come in the form of: Alert-Info: <http://127.0.0.1/Bellcore-dr3> with Broadvoice. |
| 18:32.21 | trixter | you also need to configure externip and localnet in sip.conf and turn nat on for your internal devices if you are natted |
| 18:32.42 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:32.55 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 18:33.02 | joonysam | so the devices in the internal network need to be natted even though it's in the same network with the * server? |
| 18:33.23 | tekati | IS there a way to use the Aleart-Info: within the extensions? |
| 18:34.18 | joonysam | because the devices inside the network can talk to each other |
| 18:34.31 | joonysam | which is why I didn't NAT them |
| 18:35.30 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 18:35.34 | nesys | hi folks |
| 18:35.41 | nesys | I've a lot of that: |
| 18:35.42 | nesys | Jan 22 19:34:00 WARNING[3112]: rtp.c:927 ast_rtp_settos: Unable to set TOS to 184 |
| 18:35.46 | nesys | what could I do? |
| 18:36.09 | trixter | change your tos or ignore it |
| 18:36.19 | trixter | I use 0x18 (high throughput low delay) |
| 18:36.37 | nesys | well, I've tos=0xb8 only on sip.conf |
| 18:36.40 | nesys | that's the problem? |
| 18:36.46 | Err | I would assume that that's a permission problem (that you don't have permission to set your TOS to that setting) |
| 18:36.56 | ManxPowe | I use 0xb8 as well |
| 18:37.09 | Err | 0xb8 == 184 decimal |
| 18:37.18 | ManxPowe | nesys, if you run Asterisk as non-root you will have that problem. |
| 18:37.33 | ManxPowe | There was a post on the mailing list a week or three ago that talked about how to work around that. |
| 18:37.35 | robin_sz | meep? |
| 18:37.53 | trixter | http://www.faqs.org/docs/linux_network/x-087-2-firewall.tos.manipulation.html |
| 18:38.12 | nesys | root 3091 0.0 0.0 2748 1324 pts/0 S 19:30 0:00 /bin/sh /usr/sbin/safe_asterisk -p -U asterisk |
| 18:38.12 | nesys | asterisk 3112 0.0 0.3 49852 7444 pts/0 S 19:30 0:00 asterisk -p -U asterisk -vvvg -c |
| 18:38.33 | nesys | yes, asterisk is running as asterisk user ManxPowe |
| 18:39.08 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:39.09 | ManxPowe | trixter, there's a way to allow specific users to change the TOS under linux. |
| 18:39.21 | fulgas | iptables |
| 18:39.37 | nesys | ManxPowe I use debian ... I check the ml now :) tnaks |
| 18:39.40 | nesys | thanks |
| 18:39.44 | trixter | I was giving that because it shows what the individual bits mean |
| 18:39.50 | Err | there are allowable classes that users can set - but I don't know that you can set all possible TOS values as a random user |
| 18:39.52 | ManxPowe | trixter, Ah. |
| 18:40.02 | *** join/#asterisk nroej (n=joern@heaven.cyphertext.de) |
| 18:40.04 | nroej | hi all! |
| 18:40.15 | ManxPowe | You can also use a router that allows you to change the TOS on the fly (like Ciscos) |
| 18:40.49 | Err | the problem is, I think, that you're setting the *precedence* bits by setting the high 3 bits - use 0x18 and not 0xb8 |
| 18:41.07 | trixter | I think so too |
| 18:41.14 | Err | ...not that it really matters, because every internet link on earth ignores precedence bits inbound |
| 18:41.17 | trixter | I didnt think tos was a full 8 bits, I thought it was 5 or something |
| 18:41.39 | Err | it is indeed - see RFC791 p. 12 |
| 18:41.57 | trixter | for what is meaningful anyway, and the bit mask table shows 5 bits.. |
| 18:42.05 | trixter | but meh |
| 18:42.12 | trixter | I have mine at 0x18 and that is good enough for me :) |
| 18:42.13 | Err | you're setting "Critical" precedence, which is pretentious at best :-P |
| 18:42.27 | trixter | why its ignored |
| 18:42.35 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 18:42.42 | trixter | becuase everyone sets those and unless your networking gear supports it most others wont or everyone would do that |
| 18:42.52 | Err | right |
| 18:43.05 | trixter | like setting your priority on AMPS mobile phones so you can get a call even if that means bumping someone else off due to congestion |
| 18:43.07 | Err | if you pay enough, your ISP might actually let you set precedence - but in general they ignore the TOS bits, too |
| 18:43.14 | Err | indeed |
| 18:43.20 | trixter | eventually mobile operators stopped using it, but it was great in the late 80s and very early 90s |
| 18:43.33 | Err | chances are good that the TOS bits are actually 0x00 by the time they get to the other end anyway :-) |
| 18:43.39 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 18:43.50 | trixter | congestion? not a problem some other schmuck drops their call so you can order a pizza :P |
| 18:44.14 | trixter | err: that would be improper for a company to do, odds are they ignore them rather than change them |
| 18:44.53 | Err | they're welcome to do whatever they want, as long as they update the checksum |
| 18:45.44 | *** join/#asterisk panic (i=chris@125.216.121.70.cfl.res.rr.com) |
| 18:45.45 | trixter | well yes/no, they cna introduce other issues which can cause more breakage and higher tech support queries.. while that specific change isnt likely to cause any real problems in terms of data not being received appropriately other similar things could |
| 18:45.57 | Err | "If the actual use of these precedence designations is of concern to a particular network, it is the responsibility of that network to control the access to, and use of, those precedence designations." <-- from RFC 791 |
| 18:46.05 | nesys | Err with 0x18 I've solved tnx :) |
| 18:47.03 | Err | I think these days most packets are prefixed with a header that designates QoS information, in a network's core, but updating bits in the header has been used in the past (since there are already fields for this sort of thing) |
| 18:47.17 | *** part/#asterisk panic (i=chris@125.216.121.70.cfl.res.rr.com) |
| 18:47.49 | nesys | someone has found problem to connect * (version after 1.0.9) with an ISP that uses sip port different of standard 5060? |
| 18:48.16 | trixter | running asterisk as root and giving people access to the cli (note the cli program doesnt have to have root, it just has to be able to talk to the stream pipe) you can exec commands as root - something that may not be allowed at a given installation (ie non-home users) and can allow someone to exceed their authorized access (ie a rogue employee about to quit) the manager interface has other issues. running asterisk as root to fix other problems is ge |
| 18:48.16 | trixter | nerally a bad idea for that reason. unless its just a home system or a very small company but not an enterprise type thing where permissions are segregated |
| 18:48.25 | nesys | * 1.0.9 works like a charm, 1.2.x has problems here with registration (request sent but not registered) |
| 18:48.33 | trixter | nesys: I dunno did someone ? |
| 18:48.42 | trixter | if the isp has DNS SRV entries it shouldnt be that big of a deal |
| 18:49.10 | nesys | trixter I've found that problem with messagenet (italian ISP) that uses 5061 istead of 5061 |
| 18:49.12 | nesys | 5061 |
| 18:49.17 | nesys | 5060 sorry :) |
| 18:49.56 | trixter | does that provider use a SRV entry in their DNS? they can map sip to 5061 easily that way and afaik asterisk does properly support that entry |
| 18:50.02 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:50.14 | trixter | there are also afaik ways to set the port it connects to in asterisk, something I have never had to look at so I dont really know how well that works |
| 18:50.15 | ManxPowe | nesys, are you SURE that's the destination port, not the SOURCE port? |
| 18:50.59 | nesys | ManxPowe I'm sure, cos same config on 1.0.9 works correctly :) |
| 18:51.08 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 18:51.20 | nesys | ManxPowe but I'll troubleshoot that later |
| 18:51.39 | trixter | ManxPowe: I have seen a lot of people pop ser on 5060 and asterisk on 5061 -- in that case however you should connect to 5060 most likely, but some people set up unauthenticated servers on 5061 hoping everyone will use ser on 5060 and not realize they can directly connect to the media gateway or whatever |
| 18:52.19 | nesys | trixter that's it |
| 18:52.20 | ManxPowe | trixter, until they get a 5 billion dollar bill from their provider. |
| 18:53.08 | *** join/#asterisk |omni| (i=rob@net98.limelyte.net) |
| 18:53.33 | nesys | well, as you say, isn't an asterisk problem .. right? |
| 18:55.08 | *** join/#asterisk scn (i=lkarsten@hyse.org) |
| 18:57.40 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 18:58.45 | *** part/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
| 18:59.30 | *** part/#asterisk scn (i=lkarsten@hyse.org) |
| 18:59.58 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:00.10 | *** join/#asterisk trym_ (n=trym@194.63.254.6) |
| 19:02.51 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 19:08.18 | *** part/#asterisk aconda (n=anaconda@p5496F097.dip.t-dialin.net) |
| 19:08.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:09.53 | *** join/#asterisk zishanov (n=mail@d57-249-149.home.cgocable.net) |
| 19:10.19 | zishanov | how to convert mp3 to gsm or ulaw for native MoH |
| 19:10.34 | Qwell | zishanov: sox |
| 19:10.42 | Qwell | just make sure to make it 8khz mono |
| 19:11.23 | riddlebox | geez broadvoice is making me mad right now |
| 19:11.40 | zishanov | Is it real time or I have to convert them all into gsm and then put them in moh-native folder? |
| 19:13.10 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:14.24 | austinnichols101 | Getting a voice cut-out during a call from a remote 7960G. Can hear first ring when dialing extension, then 3 rings worth of silence, then name, then silence for the rest of the call. Calling other things like *411 work fine. Any ideas on where to start troubleshooting? |
| 19:15.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:15.31 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 19:15.40 | zishanov | Is sox installed with Asterisk 1.2.2 |
| 19:15.48 | zishanov | Or I have to download is separately |
| 19:15.59 | dpryo | It's a separate application |
| 19:16.02 | DarkFlibble | zishanov, its seperate... |
| 19:16.29 | DarkFlibble | but I personally prefer to save prompts in ulaw... results in better quality with various codecs... |
| 19:17.06 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:17.15 | austinnichols101 | zishanov: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 19:17.23 | *** join/#asterisk rick283673 (n=rick1231@67-43-148-18.loudpacket.net) |
| 19:17.37 | zishanov | DarkFlibble, I am trying to fix the distorted MoH problem. Default MoH are in MP3 format and I want them to be played in native format |
| 19:18.12 | DarkFlibble | native format isn't gsm... |
| 19:18.18 | zishanov | austinnichols101, thanks |
| 19:18.21 | zishanov | yes DarkFlibble |
| 19:18.23 | DarkFlibble | gsm is just a codec... |
| 19:18.44 | DarkFlibble | Asterisk can play anything it has a format and codec for |
| 19:18.44 | austinnichols101 | zishanov: np |
| 19:19.19 | zishanov | I want Asterisk to use GSM when playing MP3 files. It is also a file format I think because all of the default sound prompts of Asterisk are in .gsm firmat |
| 19:20.13 | austinnichols101 | zishanov: slinear (.sln) starting in 1.2 |
| 19:20.20 | zishanov | I am trying to do something with sox now. I am really lost. What I am thinking now is that there should be a real time conversion of MP3s |
| 19:20.31 | zishanov | what is slinear? |
| 19:21.03 | austinnichols101 | check the link I sent |
| 19:21.11 | zishanov | ok. |
| 19:23.22 | zishanov | why asterisk doesn't come with proper MoH files? Or SOX for that matter? |
| 19:23.50 | DarkFlibble | it comes with 3 or 4 MoH mp3z which work fine for most people... |
| 19:24.03 | DarkFlibble | and Sox is developed by other people... |
| 19:24.09 | tronix | hmm maybe sound suppression (VAD) is enabled somewhere? |
| 19:24.16 | tronix | (re: 7960G cutout) |
| 19:24.23 | tronix | re: austinnic |
| 19:24.59 | DarkFlibble | if you turn on sip debugging there is a message about dropping frames or samples if vad is active iirc |
| 19:25.56 | Err | sox as in the sound utility? |
| 19:26.05 | DarkFlibble | Err, yes |
| 19:26.19 | *** join/#asterisk linlin2 (n=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 19:26.35 | Err | it's 450 years old - why should asterisk come with it |
| 19:26.36 | Err | ? |
| 19:26.42 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:27.10 | DarkFlibble | also Sox is used by 100s of utilities... not just asterisk... and asterisk doesn't *require* it |
| 19:27.32 | austinnichols101 | tronix: tks (googling to find more info...) |
| 19:28.06 | *** join/#asterisk svenna_ (n=svenna@p548D367E.dip0.t-ipconnect.de) |
| 19:28.11 | tronix | austinnic: vad basically transmits nothing during periods of silence and plays minor havoc with first second or two of next voice traffic frames |
| 19:28.26 | shmaltz | tzafrir,ping |
| 19:28.27 | austinnichols101 | tronix: sounds like what I'm seeing |
| 19:28.32 | shmaltz | ~seen tzafrir |
| 19:28.42 | jbot | tzafrir <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 105d 2h 12m 26s ago, saying: 'quasi2k, try #asterisk-de (is there such a channel?)'. |
| 19:28.44 | austinnichols101 | tronix: only seems to happen during relatively 'quiet' periods |
| 19:28.56 | tronix | austinnic: usually prefer to disable VAD to make it easier on everybody |
| 19:29.02 | shmaltz | ~seen tzafrir_lapto |
| 19:29.04 | jbot | i haven't seen 'tzafrir_lapto', shmaltz |
| 19:29.10 | shmaltz | ~seen tzafrir_laptop |
| 19:29.12 | jbot | tzafrir_laptop <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 3h 41m 45s ago, saying: 'hi'. |
| 19:31.22 | *** join/#asterisk elephantMan (n=elephant@7.205.103-84.rev.gaoland.net) |
| 19:31.24 | zishanov | how to untar the sox.tar file. It says gzip: stdin: not in gzip format |
| 19:31.38 | DarkFlibble | run file sox.tar |
| 19:31.44 | DarkFlibble | it will tell you what it is |
| 19:35.32 | zishanov | its ok now, untar successfully. Now I am trying to install it. But make install doesn't seem to be working. |
| 19:35.53 | DarkFlibble | did you read the readme? |
| 19:36.00 | DarkFlibble | and the install files? |
| 19:36.09 | zishanov | thats what I am doind right npw |
| 19:37.24 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:39.03 | austinnichols101 | tronix: doesn't appear to be enabled (checked config files on the 7960s). Is there an easy way to verify? |
| 19:40.11 | tronix | might also be referred to as rfc... 3389? |
| 19:40.40 | tronix | maybe not applicable to 7960G. not sure. |
| 19:40.59 | *** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:41.32 | austinnichols101 | the 7960G has it in the SIP<mac>.cfg file: # Enable VAD (0-disable (default), 1-enable) |
| 19:41.32 | austinnichols101 | enable_vad: "0" |
| 19:42.26 | austinnichols101 | but I think you're on to what the problem is - sounds too close to what is happening to be wrong |
| 19:43.56 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 19:45.52 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 19:47.13 | shmaltz | when is superbowl sunday? |
| 19:47.41 | iDunno | on a sunday. |
| 19:49.03 | shmaltz | iDunno, realy? |
| 19:49.46 | iDunno | apparently. |
| 19:50.02 | *** join/#asterisk annonimous (n=annonimo@dsl-200-78-52-201.prod-infinitum.com.mx) |
| 19:50.11 | annonimous | hello |
| 19:50.17 | shmaltz | iDunno, where you from? |
| 19:50.29 | shmaltz | nah, from UK |
| 19:50.44 | shmaltz | Brett, where in the UK you from? |
| 19:50.56 | *** join/#asterisk kram (n=mark@gateway.digium.com) |
| 19:50.57 | shmaltz | annonimous, hi |
| 19:51.13 | annonimous | hi |
| 19:51.17 | shmaltz | ~seen tzafrir_laptop |
| 19:51.25 | jbot | tzafrir_laptop <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 4h 3m 58s ago, saying: 'hi'. |
| 19:51.25 | zishanov | I installed sox, it said it needed libmad to read MP3 files. I installed libmad. Now do I need to reinstall SOX? |
| 19:51.27 | shmaltz | iDunno, where in the UK you from? |
| 19:51.34 | iDunno | currently in Brighton |
| 19:51.39 | iDunno | previously Norwich |
| 19:51.45 | iDunno | and before that Sudbury |
| 19:51.56 | DarkFlibble | Leicester here... |
| 19:52.13 | shmaltz | iDunno, I hate that country, the only good thing from there is my wife :P |
| 19:52.19 | DarkFlibble | used to work in Horsham... fairly close to brighton |
| 19:52.22 | tronix | shmaltz: I think it's two weeks from now? (American Football's Super Bowl) |
| 19:52.35 | shmaltz | tronix, it's on 2/5 |
| 19:52.39 | tronix | ah, nice. I had a great (working) trip to Basingstoke/London couple months ago |
| 19:52.40 | shmaltz | thanks |
| 19:52.59 | shmaltz | adn yo had what to eat? |
| 19:53.01 | iDunno | shmaltz: it's a lovely country, apart from the occasional bit of miserable weather :) |
| 19:53.07 | shmaltz | lol |
| 19:53.22 | shmaltz | tronix, you werent starving hungary while you were there? |
| 19:53.26 | tronix | oh no, not at all! |
| 19:53.29 | tronix | very well fed. |
| 19:53.42 | tronix | along with sit-down service that we take for granted here. ;) |
| 19:53.50 | tronix | (err, that we don't get enough of) |
| 19:53.51 | annonimous | is there anyway that my spa3000 can get "normal" trunk line and send it for my asterisk (cause i dont have the trunk cards and im wandering if thatsolutions will work =/) |
| 19:53.52 | shmaltz | hmmm, when I go to England I make sure I"m overweight, so I can tell everyone i'm on diet |
| 19:54.10 | shmaltz | annonimous, define normal |
| 19:54.55 | annonimous | shmaltz, normal i mean like companys wqho porovide analog accesss to pstn like humm att (mexico) telmex(mexico) not Voip companies |
| 19:55.25 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 19:55.32 | shmaltz | annonimous, of course, thats wha the FXO port on the SPA3000 is mean for |
| 19:56.19 | annonimous | shmaltz, oh ok but when i put my pstn to the fxo port i cant get line or maybe i need to configure more the spa3000? |
| 19:56.29 | annonimous | (sorry but im new with asterisk =/) |
| 20:00.02 | Flauto | annonimous, there is not much you can do to config from your asterisk end for your spa 3000 |
| 20:00.14 | Flauto | basically, it is depending on what you want |
| 20:00.15 | *** join/#asterisk razu (n=razu@217-159-187-162-dsl.prn.estpak.ee) |
| 20:00.19 | annonimous | Flauto, ah i see |
| 20:00.30 | shmaltz | annonimous, if you don't configure the spa then it wont work :( |
| 20:01.10 | annonimous | shmaltz, i have configured it but in the part of sipphone.com for sipping, but im looking for a link for how to configure fxo port =/ |
| 20:02.19 | shmaltz | annonimous, you have to configure the following on the SPA: |
| 20:02.20 | shmaltz | 1. That it registers with asterisk 2. That it has a DP for asterisk 3. and that DP is what any incoming call will use |
| 20:02.20 | Flauto | whta do you want to do with spa 3000 |
| 20:02.22 | shmaltz | In asterisk you configure: |
| 20:02.23 | shmaltz | 1. The sip.conf entry with a context. 2. an extension that matches the DP configured in step 2 above |
| 20:03.13 | annonimous | Flauto, trying to use it as line trunk instead of buy the fxo pci card cause i cant afford it |
| 20:03.55 | annonimous | shmaltz, ah i see so the spa needs to be configured and then i can use it? |
| 20:05.09 | Flauto | buy a x100p clone, it cost less then 20 dollars |
| 20:05.22 | Flauto | go to ebay |
| 20:05.22 | annonimous | x110p? |
| 20:05.26 | annonimous | ok lets see |
| 20:05.52 | Flauto | never played with x110p |
| 20:06.06 | annonimous | x11p |
| 20:06.09 | annonimous | x100p |
| 20:06.10 | shmaltz | the follwing DP for the PSTN line will make sure that any incoming call will go to extension s in asterisk: |
| 20:06.11 | shmaltz | (S0<:s>) |
| 20:07.34 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 20:07.35 | annonimous | shmaltz, and then any extension will ring? |
| 20:07.40 | annonimous | lol |
| 20:08.01 | annonimous | Flauto, digium wildcard oem? |
| 20:08.03 | shmaltz | annonimous, it depends on what you have setup in asterisk for the s extension |
| 20:08.29 | annonimous | shmaltz, ok |
| 20:09.06 | *** join/#asterisk jahani (n=l@adsl196-206-241-217-196.adsl196-16.iam.net.ma) |
| 20:09.14 | annonimous | shmaltz, any good book or how to that u can reccomend me? (cause sometimes im very shy to ask cause i dont feel that are "interesting questions =/) |
| 20:09.54 | shmaltz | annonimous, yes, Google, which can be found at: http://www.google.com/ |
| 20:09.59 | SERGEUS | are there any "sipdiscount" users? |
| 20:10.26 | annonimous | shmaltz, oh ok thanks =D! |
| 20:10.35 | SERGEUS | i have a problem last 2 days, maybe someone have the same? |
| 20:10.50 | SERGEUS | call breaks after 30-40 seconds from start... |
| 20:11.41 | *** join/#asterisk rob0 (i=1002@72.9.234.112) |
| 20:11.50 | Flauto | sergeus, i think you call for free only for under a minute if you use sipdiscount without paying |
| 20:12.01 | *** join/#asterisk NirS (n=NirS@62.90.49.95) |
| 20:12.33 | Flauto | or, use voipstunt |
| 20:12.42 | SERGEUS | Flauto, after call dropped, sipdiscount calling again to called party |
| 20:13.16 | Flauto | oh, that is strange |
| 20:13.23 | Flauto | sorry, never had that problem |
| 20:13.32 | SERGEUS | besides, such strange behaviour started 2 days ago, and i'm using sipdiscount for 2 weeks... |
| 20:13.58 | Flauto | sergeus, look at voipstunt |
| 20:14.03 | Flauto | they are the same company |
| 20:14.11 | Flauto | and using pretty much the same network |
| 20:14.17 | SERGEUS | Flauto, thanks for advice :) |
| 20:14.19 | Flauto | and it is free to many countries |
| 20:14.36 | Flauto | i have been using them for about two weeks now |
| 20:15.04 | Flauto | works great other than i can not set my callerid to be shown on ther paries phone |
| 20:19.31 | X-Files | Ppls, why message not work ? http://pastebin.ca/38065 |
| 20:20.01 | X-Files | SIP/2.0 415 Unsupported Media Type |
| 20:20.39 | rajiv | is there another web interface to voicemail other than vmail.cgi ? |
| 20:21.53 | *** join/#asterisk gkaca (n=kvirc@cpe-68-201-234-251.houston.res.rr.com) |
| 20:23.15 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 20:26.47 | *** join/#asterisk fuzzyr (n=fuzzy@204.238.218.130) |
| 20:27.45 | tainted- | would NAT issues give me 401 Unauthorized messages? |
| 20:27.56 | riddlebox | what would cause agi scripts to not run at all? I had a script that was working, now today, it doesnt |
| 20:28.12 | tainted- | riddlebox permissions? |
| 20:28.17 | tainted- | riddlebox what is it written in |
| 20:28.52 | riddlebox | tainted-, it is written in python, also I have one in c, and neither work now |
| 20:30.08 | tainted- | on the same machine? |
| 20:30.09 | tainted- | or fastagi |
| 20:30.51 | austinnichols101 | Getting a voice cut-out during a call from a remote 7960G. Can hear first ring when dialing extension, then 3 rings worth of silence, then name, then silence for the rest of the call. Calling other things like *411 work fine. Reviewed VAD but appeared to be disabled. Any ideas on where to start troubleshooting? |
| 20:31.11 | riddlebox | tainted-,on the same machine |
| 20:31.18 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:31.29 | tainted- | can the script run standalone |
| 20:31.32 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 20:31.49 | riddlebox | you mean, can I run it in python without executing it in asterisk? |
| 20:31.50 | tainted- | do u see it executing in CLI? |
| 20:31.53 | tainted- | yes |
| 20:32.22 | riddlebox | I see it in the asterisk cli, but it is supposed to use GET DATA, and it doesnt even play my recording |
| 20:33.20 | *** join/#asterisk iq|tablet (n=iq@71-38-74-41.omah.qwest.net) |
| 20:33.29 | tainted- | hm |
| 20:35.17 | *** join/#asterisk darby_t (i=darby_t@dkl13.neoplus.adsl.tpnet.pl) |
| 20:37.10 | riddlebox | do you want to see the code? |
| 20:37.22 | tainted- | yea |
| 20:37.33 | tainted- | i don't know python but i could take a look |
| 20:38.43 | riddlebox | http://pastebin.com/517966 |
| 20:39.08 | zock | cu |
| 20:39.13 | *** part/#asterisk zock (n=zock@p54B197CD.dip0.t-ipconnect.de) |
| 20:39.52 | tainted- | does it get to the "Received %s\n" line? |
| 20:40.05 | riddlebox | nope |
| 20:40.54 | *** join/#asterisk clive- (n=pirch@dsl-165-117-178.telkomadsl.co.za) |
| 20:46.39 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
| 20:47.10 | tainted- | not sure |
| 20:47.21 | tainted- | i would try running some of the perl agi's that come with asterisk |
| 20:47.53 | tainted- | see if those work |
| 20:48.00 | clive- | does nayone use mondo to do backups? |
| 20:48.07 | tainted- | is mondo around anymore? |
| 20:48.23 | tainted- | i never got it to work |
| 20:48.40 | clive- | tainted, I am looking for a simple way to backup/mirror disks |
| 20:48.48 | tainted- | same |
| 20:49.12 | justinu | ghost? |
| 20:49.35 | clive- | justin I hear ghost has issues with the gurb |
| 20:49.38 | Err | for mirroring, use rsync - for backup, use amanda :-) |
| 20:49.39 | clive- | oops...grub |
| 20:49.51 | clive- | amanda? |
| 20:49.57 | justinu | ic... can't say I've ever tried |
| 20:49.58 | Err | amanda's for tape backup |
| 20:50.00 | tainted- | she's his secretary |
| 20:50.05 | Err | (or to a hard drive, or any other filestore) |
| 20:50.25 | clive- | lol |
| 20:51.05 | tainted- | you can ghost linux? |
| 20:51.14 | Qwell | tainted-: sure |
| 20:51.20 | tainted- | wow.. didn't know that |
| 20:51.22 | Qwell | but, might as well just dd |
| 20:51.30 | clive- | tainted, so I have read, but there seems to be issues with grub |
| 20:51.42 | tainted- | what kind of issues |
| 20:52.12 | tainted- | yea if you're going to image the drive ... |
| 20:52.16 | clive- | dd seems to be the best way so far, but itsslow and not easy...one slip and bye bye hard drive |
| 20:52.24 | Err | any time you use a byte-for-byte copy of a disk partition, everything *has* to have the exact same geometry coming back out or booting won't work |
| 20:52.24 | Qwell | dd is quite easy |
| 20:52.44 | tainted- | are there incremental backup apps? |
| 20:52.46 | Qwell | Err: I disagree with that |
| 20:52.49 | Qwell | tainted-: rsync? |
| 20:52.55 | Err | well, actualy, grub understands filesystems |
| 20:53.02 | Err | then the only problem is partition tables :-) |
| 20:54.14 | clive- | the trouble with dd is that you need saem size drives |
| 20:54.23 | Qwell | X-Files: Please don't message me |
| 20:54.27 | Qwell | clive-: not true |
| 20:54.41 | *** join/#asterisk areski_ (n=areski@221.Red-88-5-213.staticIP.rima-tde.net) |
| 20:54.54 | Qwell | You can dd to a large drive very easily, and dd'ing to a smaller drive is also possible |
| 20:55.00 | riddlebox | tainted-, the perl example works |
| 20:55.03 | clive- | qwell,,,,I think you may be able to go from a small to a big driove, but not vice versa |
| 20:55.04 | Qwell | drive/partition even |
| 20:55.08 | Qwell | clive-: sure you can |
| 20:55.14 | clive- | oh really? |
| 20:55.16 | X-Files | Qwell: SIP/2.0 415 Unsupported Media Type <--- what u can say about ? |
| 20:55.17 | Qwell | absolutely |
| 20:55.23 | clive- | qwell, I need to do some more reading up":) |
| 20:55.31 | clive- | thanks for the info |
| 20:55.38 | Qwell | X-Files: means you're using a media type that isn't supported |
| 20:56.07 | Qwell | clive-: You need to be careful, but it's very possible |
| 20:56.57 | *** join/#asterisk lalo (n=erg@201.137.152.226) |
| 20:57.11 | X-Files | Qwell: what do I need for UP media type ? |
| 20:57.19 | Qwell | UP? |
| 20:57.39 | X-Files | to get up :) |
| 20:57.47 | Qwell | What is UP? |
| 20:58.14 | *** join/#asterisk masonc (n=lists@207.42.133.208) |
| 20:58.21 | X-Files | i don't know how to tell it ... xmm .. to make media type working well ... ? |
| 20:58.43 | masonc | can anyone help me understand ChanIsAvail? |
| 20:59.46 | Qwell | masonc: it checks if the channel is available |
| 20:59.49 | *** join/#asterisk SAnnis (n=a@adsl-68-89-0-10.dsl.spfdmo.swbell.net) |
| 20:59.55 | masonc | yes, got that |
| 21:00.04 | masonc | but it returns only the channel |
| 21:00.05 | Qwell | be more specific...what don't you understand? |
| 21:00.18 | masonc | if you are using IAX2 and you have to authenticate, how do you do that |
| 21:00.27 | Qwell | same as in dial |
| 21:00.30 | X-Files | Qwell what can I do with this media type? how can I fix it? |
| 21:00.30 | Qwell | I'd imagine |
| 21:00.49 | Qwell | bbl |
| 21:00.55 | masonc | example: ChanISAvail(IAX2/masonc@teliax&IAX2/fred@voxee) |
| 21:01.07 | Qwell | masonc: pretty much |
| 21:01.10 | masonc | only returns teliax |
| 21:01.16 | masonc | not masonc@teliaxc |
| 21:01.17 | Qwell | yes, it won't return both |
| 21:01.19 | SAnnis | I think I have what is a simple question if someone has time to address it.. |
| 21:01.20 | Qwell | oh, I see |
| 21:01.34 | Qwell | dunno, food |
| 21:01.35 | masonc | so what good is it in this situation |
| 21:01.38 | freq | anyone know of a voip service provider that lets you make free tollfree calls in the US , and supports IAX2 |
| 21:01.54 | *** join/#asterisk mlalkaka (n=mlalkaka@d205-250-96-41.bchsia.telus.net) |
| 21:02.29 | masonc | I am trying to make a failover dialplan |
| 21:02.34 | SAnnis | Is it possible to make asterix work with Packet8 so that I can put an extension in each room of my house.. |
| 21:02.37 | masonc | but I don't see hwo to do it with ChanISAvail |
| 21:02.44 | SAnnis | asterisk (sp0 |
| 21:03.35 | *** join/#asterisk Defraz (n=t0tal@72.165.56.43) |
| 21:03.40 | masonc | The only way I can get it to work is to do a test on a single channel |
| 21:03.41 | X-Files | Qwell what can I do with this media type? how can I fix it? |
| 21:03.43 | masonc | yes or no answer |
| 21:03.47 | masonc | and jump |
| 21:03.57 | masonc | but that gets old quick |
| 21:04.44 | tainted- | is it possible to hangup stuck SIP channels from CLI |
| 21:05.05 | clive- | tainted you can try the "soft hangup" command |
| 21:05.06 | justinu | yeah... soft hangup |
| 21:05.17 | justinu | also, try turning on the rtp timeout to prevent that from happening |
| 21:05.22 | tainted- | soft hangup <channel name> ? |
| 21:05.32 | tainted- | where do i set rtp timeout? |
| 21:05.34 | tainted- | rtp.conf? |
| 21:05.37 | justinu | sip.conf |
| 21:06.05 | tainted- | what should it be? |
| 21:06.16 | justinu | i dunno... maybe 30 seconds? |
| 21:08.37 | nroej | win 2 |
| 21:08.45 | tainted- | rtptimeout=30 in general |
| 21:08.47 | tainted- | ? |
| 21:09.01 | mlalkaka | How do I determine whether Asterisk supports my WinModem/LinModem? I have a Lucent modem? |
| 21:09.52 | mlalkaka | Does Asterisk only work with the hardware listed at http://www.asterisk.org/hardware ? |
| 21:11.02 | tainted- | if audio cuts out on my end, is it my issue or the other end |
| 21:12.03 | justinu | tainted: re: rtptimeout, yes |
| 21:12.23 | Drew | mlalkaka - i think you need a fullduplex modem - you probably have a halfduplex one.... but i dont know - im new to this stuff |
| 21:12.53 | mlalkaka | Drew, do you know how I can find out? |
| 21:13.10 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 21:13.32 | Darwin35 | ook is there any plan to fix the mem leak in mysql and asterisk |
| 21:13.33 | Darwin35 | <PROTECTED> |
| 21:14.24 | Darwin35 | it seems in realtime there is a mem leak |
| 21:14.28 | Drew | mlalkaka - http://www.voip-info.org/wiki/view/Asterisk+hardware - is a place to start |
| 21:14.44 | mlalkaka | Drew, thanks |
| 21:14.59 | Drew | you want to connect a hardline to your PBX? |
| 21:15.00 | Math[laptop] | Darwin35: if there's a bug report its probably gonna get fixed |
| 21:15.15 | Drew | or what is it you want to do with a modem? |
| 21:16.57 | tainted- | Jan 22 13:16:34 WARNING[27461]: cdr_custom.c:96 load_config: Failed to reload configuration file. |
| 21:17.02 | tainted- | what configuration file? |
| 21:18.36 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 21:19.15 | Mark_Halverson | is there a channel for dundi? |
| 21:19.58 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:22.28 | Darwin35 | ok how do you show a mem leak |
| 21:22.35 | Darwin35 | in the bug tracker |
| 21:23.22 | Darwin35 | but ast03 is working fine |
| 21:23.27 | Darwin35 | sorry |
| 21:26.58 | tainted- | i can't hangup my SIP channels |
| 21:26.58 | tainted- | I see it when i type 'sip show channels' |
| 21:27.23 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 21:29.30 | justinu | tainted: use tab completion for the soft hangup command |
| 21:32.40 | *** part/#asterisk mlalkaka (n=mlalkaka@d205-250-96-41.bchsia.telus.net) |
| 21:32.42 | *** join/#asterisk InetNomad (n=lenny@icus-gw.icus.com) |
| 21:34.07 | tainted- | stange thing is the SIP channels aren't in the autocomplete |
| 21:34.34 | justinu | oh... use "show channels" to get the name |
| 21:34.37 | justinu | not sip show channels |
| 21:36.40 | InetNomad | in the audio interview http://www.ronaldlewis.com/coffee/ Mark Spencer mentioned a solution for schools to call parents when they were absence, does anyone have a reference on that? |
| 21:37.24 | Qwell | InetNomad: probably a simple homebrew app |
| 21:37.54 | InetNomad | I'm sure, but it was made to sound like a product of some sort... |
| 21:38.17 | InetNomad | I can envision "how" I'd do it... but ... if it was done, I'd like to see their solution. |
| 21:38.53 | tainted- | justinu the two sip channels show up in 'sip show channels', but not in 'show channels' |
| 21:39.29 | InetNomad | Has anyone done Asterisk to Shoretel 6 interoperability (using new SIP in Shoretel)? |
| 21:39.41 | justinu | tainted: then those aren't real calls ;) |
| 21:39.47 | tainted- | lol |
| 21:39.49 | tainted- | but they stay there |
| 21:40.01 | tainted- | and sometimes after a day's use there are 100 of them |
| 21:40.09 | justinu | yeah, it's how the sip channel works... it creates a private struct even for registrations and crap |
| 21:40.13 | justinu | and options too, i think |
| 21:40.18 | tainted- | hmm |
| 21:40.20 | *** join/#asterisk elephantMan (n=elephant@180.205.103-84.rev.gaoland.net) |
| 21:40.29 | justinu | codec says "unkwn" right? |
| 21:40.42 | tainted- | yea |
| 21:40.51 | justinu | if they're actual calls, they'll have a real codec there |
| 21:40.51 | tainted- | and Last Message is (d) rx: BYE |
| 21:41.24 | justinu | they should eventually go away |
| 21:42.51 | tainted- | how do i force them to go away |
| 21:44.35 | justinu | you can't |
| 21:44.42 | justinu | they get auto cleaned up by the "garbage collector" |
| 21:46.33 | tainted- | justinu what about audio cut outs... sometimes i hear no audio for a few seconds |
| 21:46.48 | justinu | probably a network problem |
| 21:47.01 | justinu | lost packets |
| 21:47.35 | justinu | where do you get your calls from? |
| 21:49.17 | tainted- | US |
| 21:49.41 | justinu | hah, i mean what ip.... ping the ip with a command like this: ping -i0.2 -s180 <ip> |
| 21:50.36 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 21:51.52 | tainted- | yea crap |
| 21:51.58 | tainted- | network issues for sure |
| 21:52.13 | justinu | ping other stuff too... find out if its your feed, or theirs |
| 22:00.31 | *** join/#asterisk kilobit2001 (n=locid@206-248-153-239.dsl.teksavvy.com) |
| 22:03.00 | *** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 22:07.16 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 22:10.46 | *** join/#asterisk chiardon (n=chiardon@206.106.255.197) |
| 22:11.15 | *** part/#asterisk chiardon (n=chiardon@206.106.255.197) |
| 22:11.56 | *** join/#asterisk simon__ (n=simon@mindtrip.entered.net) |
| 22:12.10 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 22:15.09 | simon__ | Hi there, I'm having terrible time setting up outgoing ISDN with ISDN phones via asterisk, the TIMEOUT(digit) just doesn't work, is there anyone willing to help please? |
| 22:27.33 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 22:28.11 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 22:29.50 | *** join/#asterisk cjk (n=cjk@11.121.9.213.dsl.getacom.de) |
| 22:30.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:34.39 | *** part/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 22:35.36 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 22:35.40 | SpaceBass | heyhey folks |
| 22:35.48 | SpaceBass | oops s/heyhey/hey |
| 22:36.04 | Ariel_ | SpaceBass, how are you doing? it's been a few weeks |
| 22:36.10 | *** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca) |
| 22:36.14 | SpaceBass | Ariel_: yeah it has! I'm good... you? |
| 22:36.21 | Ariel_ | doing fine thanks |
| 22:36.33 | SpaceBass | Been laying tile, pouring cement and running cat 5 all day... nice to be sitting down |
| 22:37.08 | SpaceBass | also trying to iron out some bad echo on my zaptel connections and audio problems with my broadvoice |
| 22:37.12 | Ariel_ | wow hope it was for your own home or biz |
| 22:37.31 | SpaceBass | home... remodeling |
| 22:37.36 | Ariel_ | ahh I see |
| 22:38.02 | Ariel_ | I have not upgraded to aah 2.2 I am still on there 1.5 with amp 1.10.009 so far. |
| 22:38.19 | Ariel_ | it's working so don't have much need to upgrade for my home use. |
| 22:38.23 | SpaceBass | I'm on 2.2... loveing amp 1.10 thought! |
| 22:38.49 | *** join/#asterisk SoT (n=Owner@68-232-129-162.chvlva.adelphia.net) |
| 22:38.54 | SpaceBass | Well, that was my problem... mine was working so not sure why I upgraded... I did want to switch boxes for more horse power, so it seemed like a good time |
| 22:39.14 | *** join/#asterisk Dr-Linux (n=nah@202.59.75.58) |
| 22:39.20 | Ariel_ | ahh could it be the box it self |
| 22:39.44 | SpaceBass | I went from a 300mhz with 256mb to a 1.25ghz with 512mb |
| 22:39.50 | SpaceBass | would think that would help if anything |
| 22:39.58 | SpaceBass | but the zaptel echo (I hear myself) is driving us nets |
| 22:40.20 | Ariel_ | have your redone the kconfig.h to the newer mg2 |
| 22:40.39 | SpaceBass | I thought I was sending out too much power so I dialed my TXgain down to -2.5... the only result is that the other party couldn't hear me |
| 22:41.05 | SpaceBass | kconfig.h? no, not even sure what that does :) |
| 22:41.10 | Dr-Linux | is asterisk AGI works good with Java script? |
| 22:41.45 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 22:42.07 | Ariel_ | SpaceBass, in the zaptel directory you need to vi the file and change the echo routing from kb to the mg. It's fairly simple to do |
| 22:42.26 | SpaceBass | thanks... researching now |
| 22:43.54 | SpaceBass | the other problem just started yesterday... i have no audio on either end |
| 22:44.07 | pb__ | Is there a convenient expression syntax for returning a string with the last character removed? It doesn't look like ${STRING:0:-1} will do what I want. |
| 22:44.07 | SpaceBass | ive changed canreinvite from no to yes and it didn't make a change |
| 22:45.15 | Dr-Linux | anybody exprience with AGI? |
| 22:47.57 | SpaceBass | anyone else have broadvoice and having problems? |
| 22:48.30 | *** join/#asterisk backblue (n=moo@87-196-4-74.net.novis.pt) |
| 22:49.09 | Ariel_ | canreinvite should be no |
| 22:49.17 | Ariel_ | I have bv and it's working |
| 22:49.19 | wunderkin | ~striplastdigit |
| 22:49.23 | jbot | hmm... striplastdigit is ${EXTEN:0:$[${LEN(${EXTEN})} - 1]} , will remove the last digit from EXTEN, making 5551212 become 555121. Change the "1" to remove more digits. |
| 22:49.30 | backblue | hi, does anyone have problems with zaptel on gentoo? i frezze my OS. |
| 22:49.35 | Ariel_ | SpaceBass, are you sure there is no firewall issue? |
| 22:50.07 | *** join/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com) |
| 22:50.09 | SpaceBass | Ariel_: nothing has changed... it was working then stopped |
| 22:50.13 | pb__ | wunderkin: heh, thanks |
| 22:50.47 | *** join/#asterisk DaCat` (i=Dacat@68-190-18-68.dhcp.mtgm.al.charter.com) |
| 22:51.05 | Dr-Linux | ~agi |
| 22:51.07 | jbot | hmm... agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 22:54.03 | *** join/#asterisk Rowter (n=Rowter@201.145.5.26) |
| 22:54.12 | *** join/#asterisk Jammy (i=jammy@CPE0008740429bc-CM001404df6f46.cpe.net.cable.rogers.com) |
| 22:54.18 | SpaceBass | is there a reinvite setting at the trunk level or is it only at each device level? |
| 22:56.05 | *** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz) |
| 22:56.12 | Rowter | could I get dftm keys on a bridged call and react to them? on asterisk 1.0.10? or just with features on 1.2.x could be done? |
| 22:56.39 | DaCat` | ~rpt |
| 22:57.17 | Rowter | ahh http://bugs.digium.com/view.php?id=3764 >) |
| 23:00.28 | SpaceBass | damn latency is bad on my FiOS today |
| 23:01.34 | *** join/#asterisk prh (n=paul@wacka.mjr.org) |
| 23:05.34 | kuku5 | Anyone know how to reboot a cisco router via the command line ? |
| 23:06.52 | EriSanZzZ | kuku5, restart |
| 23:07.58 | Dr-Linux | anybody exprience with AGI? |
| 23:09.02 | SpaceBass | I have canreinvite=no on my device yet when I make a call over my broadvoice line the CLI reports that its attempting to bridge |
| 23:09.19 | kuku5 | EriSanZzZ: nope |
| 23:09.22 | JMcA | kuku5: reload |
| 23:09.36 | kuku5 | nope |
| 23:09.39 | dmz | hmm it's hard to get broadband w/out phone & not pay a fortune for it! |
| 23:09.40 | JMcA | yup |
| 23:09.42 | kuku5 | I'm talking about cisco. |
| 23:09.44 | JMcA | yup |
| 23:09.54 | JMcA | are you enabled? |
| 23:10.10 | kuku5 | admin>reload |
| 23:10.10 | kuku5 | Translating "reload"...domain server |
| 23:10.20 | JMcA | you're not enabled |
| 23:10.28 | kuku5 | ahh :) |
| 23:10.33 | kuku5 | thanks a bunch :) |
| 23:10.58 | SpaceBass | is there anywhere else I need to set canreinvite=no to prevent a bridge during a sip call? |
| 23:12.23 | kuku5 | JMcA: its not coming back up... |
| 23:12.28 | kuku5 | ah ..there it is |
| 23:13.03 | JMcA | IOS can take *forever* to reboot |
| 23:13.21 | JMcA | s/reboot/boot/ |
| 23:13.44 | kuku5 | you have experience with the vpn setup ? |
| 23:13.50 | JMcA | not in IOS |
| 23:13.53 | kuku5 | ah |
| 23:13.55 | kuku5 | k |
| 23:14.00 | JMcA | we do our VPNs on our Pix |
| 23:14.15 | kuku5 | its setup |
| 23:14.15 | kink0 | kuku5 you need be enabled to do a reload at cisco router |
| 23:14.33 | kuku5 | but for some reason NAT plays an important role with the vpn client |
| 23:14.33 | Ariel_ | Well Pit's are off to the SuperBowl. |
| 23:14.39 | kuku5 | kink0: thanks |
| 23:14.49 | JMcA | Ariel_: yup, been watching |
| 23:14.58 | Dr-Linux | anybody familiar with AGI ? |
| 23:15.13 | kink0 | Dr-Linux, some, what is your doubt ? |
| 23:15.13 | Ariel_ | SpaceBass, don't know why it's happening. But after dinner I might be able to help out. |
| 23:15.19 | DaCat` | Steelers vs. Seahawks! |
| 23:15.43 | eieiyo | go steelers |
| 23:16.31 | drumkilla | Seahawks have no chance! |
| 23:16.35 | drumkilla | Go Panthers!!! |
| 23:16.36 | drumkilla | :-p |
| 23:17.01 | tzanger | sivana: around?? |
| 23:17.22 | netsurfer | i've been messing with astcc for some time, made some db changes today and now the astcc config wont work just says "database creation failed" - any suggestions on how to debug this? |
| 23:17.59 | Ariel_ | don't know the next game should start in about 15 or 20 minutes |
| 23:19.17 | *** join/#asterisk SpaceBas1 (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 23:20.39 | DaCat` | Would anyone be available to speak to me in msg for help concerning the app_rpt project? |
| 23:21.09 | SpaceBas1 | anyone know what else I can do to not hear myself on zaptel calls? I've turned RX and TX gains way down, I've got echocancle on... |
| 23:22.51 | *** part/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 23:23.47 | *** join/#asterisk Johnnie (n=jdlewis@24.154.53.16) |
| 23:24.26 | *** join/#asterisk Mag1KaL (n=Mag1KaL@S010600112f0d62ac.wp.shawcable.net) |
| 23:24.36 | netsurfer | SpaceBas1 - using a x100p ? |
| 23:24.46 | SpaceBas1 | netsurfer: yep |
| 23:24.50 | Mag1KaL | Is there an good Asterisk port for Windows yet? Or does it still suck? |
| 23:24.54 | netsurfer | heh.. how'd I guess |
| 23:25.00 | Qwell | Mag1KaL: will always suck |
| 23:25.02 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 23:25.03 | gaupe | Mag1KaL: yes, windows still suck |
| 23:25.13 | SpaceBas1 | netsurfer: they worked fine for a year |
| 23:25.31 | netsurfer | SpaceBas1 - i've got one here, never got it to work right :( |
| 23:25.46 | *** join/#asterisk Husk_ (i=Husk@202.55.153.41) |
| 23:25.57 | SpaceBas1 | so I was porting my number to broadvoice... which now has gone dead and I get no audio at all on either end... i'm pretty screwed when it comes to telephoney right now |
| 23:27.58 | Husk_ | anyone know where I can find a howto on getting asterisk to find a free trunk in my trunklist for outgoing calls. at the moment it is giving up on the first one when its busy |
| 23:28.31 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:28.42 | *** join/#asterisk rene- (n=rene-@201.137.86.78) |
| 23:28.46 | rene- | hello |
| 23:28.59 | rene- | are there any clipcomm users online? |
| 23:29.03 | Ariel_ | Husk_, if your using 1.2 use dialing rules with n, if not then you can use n+101 |
| 23:29.07 | Ariel_ | ~docs |
| 23:29.08 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 23:29.20 | *** join/#asterisk _cleric_ (n=dacleric@p54828A46.dip0.t-ipconnect.de) |
| 23:29.40 | eieiyo | ~app_rpt |
| 23:30.53 | SpaceBas1 | would it make sense to turn rxgain positive to cancle echo |
| 23:31.34 | Ariel_ | actually I do use rxgain at 7.0 and txgain at 2.0 |
| 23:32.12 | SpaceBas1 | hummm |
| 23:32.29 | SpaceBas1 | I'll have to play around... my mother is tired of me calling her to test echo though :) |
| 23:32.57 | Ariel_ | you can use ztmonitor |
| 23:33.02 | Ariel_ | to see the gains |
| 23:33.10 | SpaceBas1 | looked for it... I have zttool but not monitor |
| 23:33.23 | SpaceBas1 | unless its not in the path somewhere... not sure where the zaptel stuff is in AAH |
| 23:33.28 | Ariel_ | in the /usr/src/zaptel do ./ztmonitor 1 |
| 23:33.56 | SpaceBas1 | ohhhh |
| 23:34.39 | SpaceBas1 | says it cannot show buffering |
| 23:34.57 | Ariel_ | now about bv have you tried ethereal to see if your send udp 10,000 - what ever for the rtp sound? |
| 23:35.20 | Ariel_ | you have to be on a call |
| 23:35.21 | SpaceBass | Ariel_: actually, yeah and it looks like its making it |
| 23:35.26 | SpaceBass | not sure its not a BV issue |
| 23:35.40 | eieiyo | help |
| 23:35.47 | Ariel_ | pm me your number I will give it a call |
| 23:36.01 | eieiyo | ~disa |
| 23:36.03 | jbot | extra, extra, read all about it, disa is direct inward system access. show application disa |
| 23:36.19 | Ariel_ | disa works it's fairly easy to setup |
| 23:38.15 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 23:39.02 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 23:39.06 | *** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net) |
| 23:39.14 | SpaceBass | back... connection |
| 23:39.32 | SpaceBass | Ariel_: rxgain=7.0 fixed my echo! |
| 23:39.44 | Qwell | 7.0? That's awful high |
| 23:40.09 | SpaceBass | Qwell: going to try dialing it back until ... DAMN ECHO IS BACK |
| 23:40.14 | SpaceBass | it gets wrose as the call goes on |
| 23:40.35 | Qwell | were you changing it by 1.0 or 0.5? |
| 23:40.40 | SpaceBass | 1.0 |
| 23:40.46 | SpaceBass | started at .5 until I got to 2 |
| 23:40.47 | Qwell | do .5 |
| 23:40.49 | shmaltz | ~seen tzafrir_laptop |
| 23:41.01 | jbot | tzafrir_laptop <n=tzafrir@local.xorcom.com> was last seen on IRC in channel #asterisk, 7h 53m 34s ago, saying: 'hi'. |
| 23:41.32 | SpaceBass | Ariel_: back... lost my connection... 202-318-1614 ... see if you get my IVR |
| 23:41.57 | SpaceBass | as broadcast that to the entire room |
| 23:42.12 | SpaceBass | I'm on a roll tonight |
| 23:42.53 | rene- | i have a voip analog gateway that makes calls to the pstn, it needs to be restarted each 5-10mins or so since all the sudden it gets blocked (cant be pinged, no web interfase, ongoing conversations are dropped) it is a clipcomm 410, do i have a bad unit? i tried with the only firmwarre update availabla at clipcomm web site without luck.. has anyone seen something like it before? |
| 23:43.29 | Ariel_ | SpaceBass, it's ok not much going on there right now |
| 23:43.40 | Ariel_ | but I got your vm and I head the sound just fine. |
| 23:43.50 | SpaceBass | thats interesting... |
| 23:43.58 | SpaceBass | wonder if its this device in particular |
| 23:46.37 | shmaltz | rene, I would exchange it for a new one |
| 23:47.59 | Ariel_ | rene-, is the unit very hot? |
| 23:48.20 | SpaceBass | anyone have firmware for the hitachi IP5000 ? |
| 23:48.38 | rene- | Ariel: it isnt hot at all |
| 23:48.40 | Ariel_ | which one |
| 23:48.50 | Ariel_ | then your going to need an rma |
| 23:49.06 | SpaceBass | Ariel_: it appears to be device specific.. just my wifi sip phone...which is on different subnet than my * box |
| 23:49.30 | Ariel_ | SpaceBass, good at least it's not the main trunk |
| 23:49.45 | SpaceBass | Ariel_: I'm thinking its attempting to make the bridge (despite my canreinvite=no) and since the wifi phone doesnt have access to the itnernet, it cannot complete it |
| 23:50.00 | Ariel_ | I have 2.03 |
| 23:51.32 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 23:52.20 | Ariel_ | I just ordered a F1000 from UTStarcom |
| 23:59.09 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 23:59.20 | TUplink | is there somthing like canreinvite for IAX? |
| 23:59.59 | TUplink | i call Asterisk from a PSTN to IAX gateway and then dial an extension but cant hear one end |