| 00:00.41 | jaytee | vegbox, you need an ATA with FXS ports |
| 00:01.01 | vegbox | either that or buy ip phones right? |
| 00:01.09 | jaytee | or a tdm card with fxs modules from a company like Digium or Sangoma |
| 00:02.04 | jaytee | yes, if you don't want to buy new phones you'll need either the ATA which connects via ethernet to * using SIP or a card in the server. |
| 00:02.35 | vegbox | well the * box is colocated, so i want something that doesnt require another computer running in the house |
| 00:02.50 | vegbox | i was tinking a linksys phone adapter? |
| 00:02.59 | jaytee | the Linksys Sipura SPA-2102 comes with 2 FXS ports so you could have two seperate extensions. |
| 00:03.27 | vegbox | yeah i like that |
| 00:03.55 | vegbox | and i just tell that thing to connect to * |
| 00:04.00 | vegbox | via an extension and secret code? |
| 00:05.19 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 00:10.26 | vegbox | whats a cheap joint to get a DID and unlimited incoming calls? |
| 00:21.22 | jaytee | vegbox, you need to read the part of the book about connecting to a SIP provider and probably read more if you're going to do any administration on the colo * box. |
| 00:21.27 | jaytee | ~book |
| 00:21.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 00:22.06 | jblack | . |
| 00:22.26 | vegbox | well i know how to use ip phones and soft phones in general |
| 00:22.30 | vegbox | but i dont want to spend 50 bucks online |
| 00:22.33 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 00:22.36 | jblack | About the lynksys'... I have an 8K, and I think it sucks energy |
| 00:22.40 | vegbox | to find out this linksys sipura doesnt do what i need to do |
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| 00:41.01 | pheller_ | anyone have any hints for how to deal with out-of-order RTP frames carrying rfc2833 dtmf data? |
| 00:41.16 | jblack | the spec doesn't say? |
| 00:42.09 | pheller_ | jblack: was that in response to me? |
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| 00:57.12 | jblack | pheller: yeah |
| 00:58.41 | pheller | jblack: well, the general idea would be to do some sort of out of order compensation with a buffer on the asterisk side. there seems to be a facility to do this (jitterbuffer), but it doesn't seem to apply to rtp type 101 |
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| 01:06.06 | jblack | I don't know what the spec calls for |
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| 02:00.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:00.55 | *** join/#asterisk trix123 (n=trix@60-242-131-154.static.tpgi.com.au) |
| 02:01.01 | trix123 | hey guys |
| 02:01.16 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 02:02.25 | trix123 | im using trixbox at the moment and im having a problem where when i ring phone1 there is a 15 second delay between dialing and phone2 ringing |
| 02:03.24 | trix123 | any ideas? |
| 02:05.19 | leifmadsen | trix123: see #trixbox |
| 02:05.38 | trix123 | im in there too :) |
| 02:06.18 | eppigy | that is the only place you should be |
| 02:06.33 | jaytee | unless you are dave |
| 02:06.40 | eppigy | yes |
| 02:06.55 | eppigy | then you do what you want |
| 02:07.29 | jaytee | dave is like the kwisatz haderach of IRC channels, he can be many places at once |
| 02:07.53 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:07.55 | eppigy | SMOKE SPICE BY THE POUND |
| 02:08.00 | jaytee | LOL |
| 02:10.29 | trix123 | yeh ok its quiet in there lol |
| 02:11.03 | trix123 | we also run a plain asterisk and it works fine |
| 02:11.22 | trix123 | just thought there might be some simple configuration we missed in this testing |
| 02:11.30 | trix123 | stupid trixbox |
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| 03:10.37 | toughmarketing | [TK]D-Fender: Last night you said something about using time within an extension but I still cannot find anything like that. Maybe I misunderstood? I am trying to put time based calls inside of my database for extensions. |
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| 03:30.23 | vegbox | Hello, anyone know of good cheap providers for outgoing calls? |
| 03:35.12 | Faiz | Hey everyone, any ideas where I can find a beginners guide on setting up Fax with Asterisk? |
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| 04:01.27 | Rob3Rt | sup |
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| 04:03.55 | joelsolanki | Hi all |
| 04:04.15 | joelsolanki | anybody recommend canada DID provider. need hunting group facility from them too. |
| 04:04.22 | joelsolanki | maybe someone is here ? |
| 04:04.34 | joelsolanki | Need DIDs for toronto |
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| 04:43.04 | mike345 | hi guys. after my asterisk came up from a power outage zttool shows all my T1s as RED. All the other non-Asterisk T1's look good, is this a ztcfg problem? |
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| 04:48.49 | s14ck | somebody can make calls with letters extension at polycom 330 ip phone? |
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| 04:56.06 | darkmadda | Anyone worked with asterisk-perl for interfacing programs with asterisk. |
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| 06:03.49 | k4mi1 | hi all |
| 06:03.50 | k4mi1 | :) |
| 06:04.00 | k4mi1 | I have a question about '+' character in number e.g. +49 71 123 456 |
| 06:04.07 | k4mi1 | do You think that this character is double-zero: 00 |
| 06:04.14 | k4mi1 | and after this my CO will receive number 0049 71 123 456 |
| 06:04.17 | k4mi1 | ? |
| 06:15.54 | WindowsUser | http://en.wikipedia.org/wiki/List_of_international_call_prefixes |
| 06:16.48 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 06:18.37 | WindowsUser | if you're asking asterisk for +49 71 etc you might have to convert it to 004971 etc in the dialplan |
| 06:22.00 | k4mi1 | ok |
| 06:22.01 | k4mi1 | thanks |
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| 06:51.03 | AlmightyOatmeal | has anyone had any dial lag with a spa1001 ata adapter? |
| 06:51.18 | AlmightyOatmeal | its like i dial the number and wait another 20 seconds before i hear ringing/call connects |
| 06:51.21 | AlmightyOatmeal | sometimes it doesn't at all |
| 07:04.57 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:07.21 | Dovid | when do you see the call in the asterisk CLI ? |
| 07:07.30 | Dovid | and on a sip debug when does the call come in ? |
| 07:09.02 | AlmightyOatmeal | next time i make a call (or when i get home), i'll time when i dial on the spa adapter to when i see it in the cli |
| 07:09.34 | AlmightyOatmeal | i just hope its not my dialplan or the CoS settings in my switch |
| 07:09.46 | AlmightyOatmeal | a softphone over vpn has no dial lag so i couldn't imagine it would be the dialplan |
| 07:11.21 | Dovid | prob. the device |
| 07:11.23 | Dovid | r u using IP or DNS ? |
| 07:12.27 | AlmightyOatmeal | ip |
| 07:12.50 | AlmightyOatmeal | its on the local network of course |
| 07:19.23 | WindowsUser | its probably a problem with the dialplan in the spa device |
| 07:19.48 | AlmightyOatmeal | in the spa device itself and not my asterisk dialplan? |
| 07:20.03 | WindowsUser | well yea |
| 07:20.18 | WindowsUser | when you pick up the phone and dial, its the spa handling that |
| 07:20.21 | AlmightyOatmeal | can't really configure much of the spa itself as far as dialplans, the web UI is rather limited in that way |
| 07:20.30 | WindowsUser | you dial some digits, and usually that goes to the asterisk |
| 07:20.38 | AlmightyOatmeal | yeah |
| 07:20.43 | WindowsUser | (*xx|[3469]11|0|00|5366S0|[2-9]xxxxxxxxxS0|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 07:20.51 | AlmightyOatmeal | mmm |
| 07:22.08 | AlmightyOatmeal | http://pastebin.ca/1501808 <-- thats my asterisk outbound dialplan |
| 07:22.13 | WindowsUser | the S0's mean that if it matches those options, it doesn't wait |
| 07:22.30 | AlmightyOatmeal | and thats a spa dialplan? |
| 07:23.06 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:23.27 | WindowsUser | yea |
| 07:25.50 | AlmightyOatmeal | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 07:25.54 | AlmightyOatmeal | thats what i have right now |
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| 07:26.14 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 07:26.24 | WindowsUser | so if you dial 16085551212 it should immediately go out |
| 07:26.47 | AlmightyOatmeal | ah |
| 07:27.10 | AlmightyOatmeal | but if i dial 5551212 it needs to check with the asterisk dialplan? |
| 07:27.34 | WindowsUser | well all calls go to asterisk in your case |
| 07:27.40 | AlmightyOatmeal | well yeah |
| 07:27.53 | AlmightyOatmeal | i'm just trying to figure if dialing the number without the +1608 has any affect |
| 07:28.28 | WindowsUser | well with the SPA dial plan you can modify what gets dialed if you want |
| 07:29.12 | WindowsUser | ie <:1608>xxxxxxx to add the 1608 in front of a 7 digit number |
| 07:29.38 | AlmightyOatmeal | i want people to be able to dial 16085551212, 6085551212 (and add the +1), or 5551212 (add the +1608) |
| 07:29.51 | AlmightyOatmeal | can that be done from one line with the spa dialplan? |
| 07:31.02 | WindowsUser | yea |
| 07:32.06 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 07:32.36 | AlmightyOatmeal | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx|1608xxxxxxx.) <-- like that? |
| 07:32.40 | *** join/#asterisk CrazyTux[m] (n=Brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 07:33.58 | WindowsUser | no, dont take the dot |
| 07:34.17 | AlmightyOatmeal | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx|1608xxxxxxx) <-- like that then? |
| 07:34.23 | AlmightyOatmeal | or |
| 07:34.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ed28ecc147be0e01) |
| 07:34.34 | AlmightyOatmeal | yeah |
| 07:34.37 | WindowsUser | also 1xxx[2-9]xxxxxxS0 matches any 1608xxxxxxx calls |
| 07:35.05 | AlmightyOatmeal | ah, i want an automatic addition of 1608 on dialing a 7 digit number so i would probably take that rule out |
| 07:35.30 | WindowsUser | (*xx|[3469]11|0|00|<:1608>[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) then |
| 07:36.38 | AlmightyOatmeal | 2-9? |
| 07:37.20 | WindowsUser | (*xx|[3469]11|0|00|<:1608>[2-9]xxxxxxS3|[2-9]xxxxxxxxxxS0|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) <-- I'm curious if this'd work |
| 07:37.50 | WindowsUser | AlmightyOatmeal: 2-9 aka not a 1 |
| 07:38.17 | AlmightyOatmeal | would that prevent a 1 or 0 from being dialed in the rest of the number? |
| 07:38.36 | AlmightyOatmeal | or am i not understanding that right |
| 07:38.43 | WindowsUser | its a regex |
| 07:39.04 | WindowsUser | so if you dial 1 first, it will skip that rule |
| 07:39.13 | AlmightyOatmeal | aah |
| 07:41.39 | WindowsUser | one problem with doing 7 digit dialing like that is 310xxxx calling wont work :) |
| 07:42.06 | AlmightyOatmeal | well i only want 7 digit dialing to prefix a 1608 to it, thats all |
| 07:42.18 | AlmightyOatmeal | otherwise 10 digit dialing would audomatically prefix a 1 |
| 07:42.27 | AlmightyOatmeal | and 11 digit dialing goes through like it should |
| 07:43.09 | WindowsUser | test the rule with S3 like dial 7 digits, then try 10 digits |
| 07:43.23 | AlmightyOatmeal | so like <:1608>xxxxxxxS3|<:1>xxxxxxxxxxS3|xxxxxxxxxxxS3 |
| 07:43.25 | AlmightyOatmeal | why S3? |
| 07:43.48 | WindowsUser | I had [2-9]xxxxxxxS0 (7digits) on mine and I wasn't able to dial 10 digit number |
| 07:44.27 | AlmightyOatmeal | or like like <:1608>[2-9]xxxxxxxS3|<:1>[2-9]xxxxxxxxxxS3|xxxxxxxxxxxS3 |
| 07:44.43 | AlmightyOatmeal | that looks like a winner to me, though I don't get the S0/S3 thing |
| 07:44.46 | WindowsUser | just S3 on 7 digits |
| 07:45.00 | *** join/#asterisk caraconan (n=caracona@194.179.83.89) |
| 07:45.14 | AlmightyOatmeal | like <:1608>[2-9]xxxxxxxS3|<:1>[2-9]xxxxxxxxxxS0|xxxxxxxxxxxS0 ? |
| 07:45.22 | WindowsUser | (*xx|[3469]11|0|00|<:1608>[2-9]xxxxxxS3|<:1>[2-9]xxxxxxxxxxS0|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 07:45.25 | WindowsUser | yea |
| 07:45.42 | AlmightyOatmeal | what does S3 do? |
| 07:45.45 | AlmightyOatmeal | vs S0 |
| 07:45.55 | WindowsUser | S0 is wait for zero seconds after a match |
| 07:46.05 | AlmightyOatmeal | aah |
| 07:46.08 | WindowsUser | so im hoping that S3 is a 3 second time out |
| 07:46.17 | AlmightyOatmeal | why would i want that? |
| 07:46.33 | WindowsUser | I had [2-9]xxxxxxxS0 (7digits) on mine and I wasn't able to dial 10 digit number <-- this is why |
| 07:46.52 | AlmightyOatmeal | oh heh |
| 07:46.57 | WindowsUser | 6042809000 and it'd dial 6042809 |
| 07:47.13 | AlmightyOatmeal | aah lol makes sense |
| 07:47.25 | AlmightyOatmeal | now i can't wait to go home and try out that dialplan hehe |
| 07:47.51 | AlmightyOatmeal | hopefully that takes care of that lag oddity |
| 07:47.53 | caraconan | Hi. I would do a hylafax question: I have '/etc/hylafax/config.ttyS0' with 'LogFileMode: 0666', but '/varspool/hylafax/xferfaxlog' are 0600. Some help? |
| 07:48.37 | WindowsUser | AlmightyOatmeal: just the S3 and S0 parts should take care of it, the <:1> stuff just rewrites what goes out |
| 07:48.57 | AlmightyOatmeal | WindowsUser: good to know, i feel more productive this morning hehe :) |
| 07:49.02 | WindowsUser | all your calls will match the _1NXXNXXXXXX rule btw |
| 07:49.25 | CrazyTux[m] | AlmightyOatmeal: thats because you ate your oats :) |
| 07:49.29 | AlmightyOatmeal | yeah, but i have the other outbound rules just incase i'm using a softphone |
| 07:49.31 | AlmightyOatmeal | CrazyTux[m]: lol |
| 07:49.50 | WindowsUser | ah |
| 07:50.06 | WindowsUser | some of the rules are @ivr and @sip.broadvoice.com not sure if they're = |
| 07:50.35 | AlmightyOatmeal | ivr is my dialplan context and sip.broadvoice.com is my sip provider context |
| 07:51.10 | WindowsUser | oh, thats an odd way of doing it |
| 07:51.27 | AlmightyOatmeal | it takes the number to be re-written and re-passes it to the ivr context to pass to the first rule to be passed to my sip provider |
| 07:51.36 | AlmightyOatmeal | at least that was the theory, and odd is how i do things hehe |
| 07:51.44 | WindowsUser | but it works? |
| 07:51.54 | AlmightyOatmeal | strangely enough yes |
| 07:52.24 | AlmightyOatmeal | i probably don't need to set caller id so many times though, plus my sip provider doesn't do ss7 or subscribe to any outbound cname databases so i could probably just drop those alltogether |
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| 07:55.41 | AlmightyOatmeal | time for me to earn my $7/hr and do the night audit |
| 07:57.02 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
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| 08:12.43 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-82-2.hag.east.verizon.net) |
| 08:13.04 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 08:13.42 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d420:f99e:15a5:c4a9) |
| 08:14.33 | *** part/#asterisk oilinki (n=oil@ip-208-109-22-97.ip.secureserver.net) |
| 08:15.41 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 08:16.53 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 08:21.12 | *** join/#asterisk proute (n=AnthonyC@LMontsouris-152-63-16-150.w217-128.abo.wanadoo.fr) |
| 08:21.22 | *** join/#asterisk mileslai (n=miles@220.130.175.184) |
| 08:21.29 | proute | hello |
| 08:22.58 | proute | I have troubles with dtmf on sip. For example on my * 1.4.25, I have a DID number to an ivr. if this DID number is received by patton on BRI connection, it's work. the caller push 1 or 2 and the IVR work. |
| 08:23.17 | proute | If I receive a call via SIP to ivr, the dtmf don't work... |
| 08:23.26 | proute | my dtmfmode is rfc2833 |
| 08:23.43 | proute | I tried with inband and info, but nothing change |
| 08:23.54 | proute | any idea about this? |
| 08:24.32 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:30.42 | mileslai | Do you have enabled videosupport? |
| 08:33.45 | pif | anyone else is having trouble with 1.6.1.x and iaxmodem faxing? |
| 08:36.48 | proute | no |
| 08:39.22 | pif | what fax options do I have besides iaxmodem with asterisk 1.6 ? |
| 08:43.20 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 08:45.14 | *** join/#asterisk Dovid[Laptop] (n=chatzill@213.8.121.90) |
| 08:45.56 | mileslai | try dtmfmode to auto, maybe the SIP client only support inband |
| 08:46.36 | proute | mileslai: I just try .... but the problem is the same :( |
| 08:46.55 | proute | I tried auto, info, inband, rfc2833 |
| 08:47.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 08:47.14 | proute | and When a call come from SIP via Internet.. dtmf not work |
| 08:47.52 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 08:54.03 | mileslai | asterisk -rx "sip show peer xxxxx" , xxxxx is your sip client , and show the DTMFmode information |
| 08:59.09 | proute | mileslai: For my trunk I have : DTMFmode : rfc2833 |
| 09:00.52 | proute | mileslai: For my Patton, I have no information about dtmf |
| 09:01.11 | proute | oups sorry |
| 09:01.14 | proute | I have DTMFmode : auto |
| 09:06.13 | *** join/#asterisk cjk_ (n=cjk@vodsl-9494.vo.lu) |
| 09:06.58 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:10.56 | cjk_ | hi, has anyone used externalIVR? i am not able to get it working, maybe someone has a mini example |
| 09:16.09 | Dovid | what issue r u having ? |
| 09:16.13 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 09:16.44 | Dovid | in logger.conf under full put in dtmf |
| 09:16.46 | Dovid | then have a look at the log. see if your dtmf is coming in |
| 09:17.33 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 09:20.30 | proute | Dovid: I put dtmf under full in logger.conf. |
| 09:21.08 | Dovid | ok now: asterisk -rx 'logger reload' ; tail -f /var/log/asterisk/full |
| 09:21.21 | Dovid | and see if when you press some keys if you see it there |
| 09:21.38 | proute | ok thanks, I try now |
| 09:21.45 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:23.27 | proute | Dovid: Nothing about DTMF. After, I try call via Patton and I saw dtmf... |
| 09:24.22 | Dovid | proute: so if you have carreir->Asterisk no dtmf and if you have carrier->patton->Asterisk then it does work ? |
| 09:25.18 | proute | Dovid: call from Internet (SIP) -> Asterisk dtmf not work | Call from Patton -> Asterisk, dtmf is ok |
| 09:25.31 | Dovid | then its prob. ur carrier ;) |
| 09:25.41 | Dovid | try dtmfmode=info |
| 09:25.44 | Dovid | see if that works |
| 09:25.53 | proute | Dovid What is carrier? |
| 09:25.54 | Dovid | if not set the codec to ulaw and then set dtmfmode=inband |
| 09:26.11 | Dovid | this part: call from Internet (SIP) -> Asterisk |
| 09:26.14 | Dovid | how are you testing that ? |
| 09:26.24 | proute | I try inband, info, auto, rfc2833... with alaw for inband... |
| 09:28.26 | proute | Dovid, the thing is strange . So When I call IVR via SIP dtmf work | Phone -> Asterisk -> SIP (Internet) dtmf is ok |
| 09:29.11 | Dovid | how are you trying from the internet ? you have an ITSP that you are using? |
| 09:30.15 | proute | So, On my asterisk, I put DID number to IVR. I call this DID number via my mobile phone.... |
| 09:30.36 | proute | (With Patton I put a DID number to the same IVR= |
| 09:30.57 | proute | and yes |
| 09:31.02 | proute | I have an ITSP |
| 09:41.33 | proute | But, with my Patton, there is a "sip connection" like my ITSP... and my dtmf work... |
| 09:41.49 | proute | Dovid: Do you think that my problem come from my ITSP? |
| 09:42.00 | Dovid | probably |
| 09:51.03 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 09:52.27 | proute | Dovid: I try witj |
| 09:52.30 | proute | arf |
| 09:52.53 | proute | I try with other ITSP (dtmf rfc2833) and it works... |
| 09:53.14 | Dovid | so it is your ITSP. you need to speak to him |
| 09:53.17 | Dovid | them* |
| 09:53.27 | proute | yes, thanks for your help :) |
| 09:55.00 | *** join/#asterisk mileslai (n=mileslai@220.130.175.184) |
| 10:05.36 | *** join/#asterisk drmay (n=drmay@221.Red-80-38-95.staticIP.rima-tde.net) |
| 10:06.20 | *** join/#asterisk DarthPointer (n=no@82.218.68.216.DED-DSL.fuse.net) |
| 10:07.51 | drmay | I want my asterisk to ring my mobile telephone, anyopne know a good n cheap VoIP company? |
| 10:09.44 | dwery | it's normal that "queue show xxx" always shows the members as "Not in use" ? |
| 10:21.06 | *** join/#asterisk troubled_ (n=troubled@unaffiliated/troubled) |
| 10:25.52 | *** join/#asterisk justfahdi (n=fahad@116.71.16.229) |
| 10:26.01 | justfahdi | Hi |
| 10:26.07 | justfahdi | I'm very newbie in asterisk |
| 10:26.37 | justfahdi | My teacher assigned me a task to create pc 2 phone service. from asterisk |
| 10:27.03 | Chainsaw | justfahdi: So you have joined today because you would like us to do your homework. |
| 10:27.09 | justfahdi | please tell me what I need to do after installing asterisk |
| 10:27.19 | Chainsaw | justfahdi: You will need to configure it. |
| 10:27.31 | justfahdi | from? |
| 10:28.08 | Chainsaw | From any text editor you like. |
| 10:28.14 | Chainsaw | The configuration files are in /etc/asterisk on most systems. |
| 10:28.46 | justfahdi | ok which files I need to edit? |
| 10:29.07 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 10:29.24 | Chainsaw | justfahdi: I am sure that part of doing homework involves showing some initiative. |
| 10:29.44 | Chainsaw | justfahdi: Open a few configuration files, look at the comments inside and check whether it seems relevant to what you are doing. If not, move on to the next one. |
| 10:31.11 | justfahdi | Chainsaw: yeah u are right but after wasting my 2 days. My friend suggested me to join IRC channel and get help from there |
| 10:31.45 | Chainsaw | justfahdi: You spent 48 hours on this, but the notion of editing configuration files in /etc/asterisk was new to you? |
| 10:33.18 | justfahdi | no it wasn't. I was trying to do it with A2billing. So I made conf. files and dialled plan from there. But didn't get any luck. So now I decided to leave a2billing and do it simple way |
| 10:33.50 | Chainsaw | justfahdi: If you want a "simple way" that does not involve configuration files, plain Asterisk is not what you want. |
| 10:36.28 | justfahdi | ohh really. :S that's why I'm here. I just want to know what steps I should follow. like first I've to edit conf. file. then should I need to add any carrier to make outgoing calls for USA? definitely yes. But how I don't know :( I'm absolute beginner. Please just give me the outline |
| 10:36.57 | Chainsaw | justfahdi: What you still haven't done is ask a clear question. |
| 10:37.31 | Chainsaw | justfahdi: "My name is justfahdi, and today I want to do X with Asterisk, it needs to be able to do Y and I'm comfortable using Z to configure it" |
| 10:38.24 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 10:40.08 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 10:41.10 | justfahdi | ok. My name is Fahdi, and today I want to do PC2Phone service with Asterisk, I don't know how I can do it. I tried to do it with A2Billing. Am I going on the right way. Can I create a PC2Phone service with A2billing? |
| 10:41.39 | Chainsaw | justfahdi: "PC2Phone" doesn't mean anything to me. Tell me what you want to do. |
| 10:42.37 | justfahdi | I want to call at a USA number using Asterisk |
| 10:42.55 | justfahdi | now is it clear question? |
| 10:43.19 | Chainsaw | justfahdi: It's getting better. |
| 10:43.35 | Chainsaw | justfahdi: Will this PC be running Asterisk? What software will you be using the make the actual call? |
| 10:46.24 | Chainsaw | justfahdi: You can see Asterisk like a phone exchange in software. It can connect an incoming call to an external number (and that can be a USA number). |
| 10:47.21 | Chainsaw | justfahdi: If you want to make a phone call with the microphone and speakers of a PC, you'll need "soft phone" software. |
| 10:48.18 | Chainsaw | justfahdi: You can connect the soft-phone to Asterisk and then connect to a USA number (using a VoIP telephony provider or a telephony card that you connect to an ISDN line or regular phone line). |
| 10:48.39 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 10:48.39 | Chainsaw | justfahdi: I can't see your assignment from here, but a soft-phone might be all you need. You may not need Asterisk at all. |
| 10:49.26 | Chainsaw | justfahdi: Unless of course you have multiple PCs with multiple soft-phones and your Asterisk is not running on those PCs, but on a central server somewhere. |
| 10:51.52 | justfahdi | I've a seperate central server. I want to make calls from this server to any usa number usin softphone on my pc. I've already asterisk there. Already registered the extensions. Now what I need to make calls |
| 10:52.25 | Chainsaw | justfahdi: Either a telephony card and a phone line, or a VoIP telephony provider. |
| 10:52.32 | Chainsaw | justfahdi: Both will cost money. |
| 10:53.22 | *** join/#asterisk pepe (i=kvirc@130.161.41.252) |
| 10:53.45 | justfahdi | Ok. How Can I do it. Money doesn't metter. But I want to do calls from my server. Don't want to use their server. |
| 10:54.16 | Chainsaw | justfahdi: Do you have a phone line near your server that you can use? |
| 10:54.51 | Chainsaw | justfahdi: Are you planning to offer multiple simultaneous calls? |
| 10:54.51 | justfahdi | No i dont have |
| 10:54.51 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 10:55.02 | Chainsaw | justfahdi: Okay, so adding a telephony adapter to the server is not an option, because you have nothing to connect it to. |
| 10:55.16 | Chainsaw | justfahdi: So you'll need to open an account with a VoIP telephony provider. You'll likely talk to it over SIP. |
| 10:55.33 | justfahdi | hmm then... |
| 10:55.49 | Chainsaw | sprays water on justfahdi |
| 10:55.52 | Chainsaw | The power of google compels you! |
| 10:56.05 | justfahdi | :) ok. |
| 10:59.11 | pepe | Hello, I'm using Asterisk 1.6.2.0-beta3 with asterisk-addons-1.6.2.0-rc1 to write cdr into mysql. I followed http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk, and it is mostly working however CLID is not written to mysql, although it appears in Master.csv. What do I possibly missing? TIA |
| 11:00.01 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 11:02.53 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 11:03.36 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 11:04.21 | pepe | are there somewhere archives of the asterisk irc channel ? |
| 11:05.10 | Zhad | Doubtfull |
| 11:06.23 | pepe | :| ok |
| 11:11.56 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 11:13.33 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 11:24.59 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 11:25.39 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:28.00 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:35.13 | afink | Zhad: a few http://ircarchive.info/asterisk/2007/5/20/1.html |
| 11:41.24 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 11:42.27 | Orbixx | I'm creating a user extension for a SIP softphone, what details do I use in order to register with the pbx? |
| 11:43.05 | *** join/#asterisk justfahdi (n=fahad@116.71.6.236) |
| 11:45.41 | *** join/#asterisk Router222 (n=CK@212.98.141.199) |
| 11:49.48 | tzafrir_laptop | ~logs |
| 11:49.49 | infobot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 11:49.58 | tzafrir_laptop | pepe, ==^ |
| 11:50.31 | tzafrir_laptop | <PROTECTED> |
| 11:50.52 | tzafrir_laptop | <PROTECTED> |
| 11:51.36 | *** join/#asterisk jgoo (n=r3rman@athedsl-4549667.home.otenet.gr) |
| 11:52.05 | jgoo | I was asking about autoconferencing ring groups - is this possible? I want to dial 10 numbers, and as they pickup, to be added to a conference |
| 11:52.30 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 11:52.57 | *** join/#asterisk bhodder (n=blake@142.166.111.209) |
| 11:53.01 | jgoo | someone got as far as asking me my asterisk version, which is possible the most annoying thing ever, I can tell you my gimp, ff, thunderbird, pidgin, xchat or almost any other version, but there is something inherently idiotic about the asterisk version numbers that make them forgettable |
| 11:53.38 | jgoo | 1.4.22-2 |
| 11:53.46 | bminish | With Asterisk 1.4 and up one can set DTMF lenght value (via RFC2833), how do I do this ? |
| 11:55.20 | pepe | tzafrir_laptop: thanks |
| 11:55.25 | bhodder | infobot |
| 12:01.29 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 12:03.10 | mbrevda | .ç ùãà ÷øïãì-ïê |
| 12:05.09 | tzafrir_laptop | well, at least it was UTF-8 |
| 12:05.34 | mbrevda | ops |
| 12:05.36 | mbrevda | it was?? that weird |
| 12:06.33 | tzafrir_laptop | I think it was |
| 12:06.51 | mbrevda | if it was, its soem sort of windows bug :) |
| 12:06.54 | tzafrir_laptop | I can't read it anyway. |
| 12:07.49 | tzafrir_laptop | actually, no. I guess my client shows random non-UTF-8 bytes as ISO-8859-1 characters |
| 12:08.01 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:15.41 | jgoo | I am grepping the grandstream 201 user guide - I can't find a way of setting a dial plan... wtf |
| 12:16.12 | jgoo | I know they support early dialing, but do they also support dialplans? I've seen people showing examples, but the user guide and the interface don't show how to set it... |
| 12:17.04 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:18.02 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 12:23.50 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:24.23 | Dovid | i forgot which sip is rining |
| 12:24.24 | ariel_ | Morning |
| 12:24.27 | Dovid | 183 or 180 ? |
| 12:25.17 | Dovid | which one tells me to generate one and which one which one tells me to pass audio ? |
| 12:27.27 | mbrevda | 180 is ringing |
| 12:29.01 | Dovid | and 183 says i will pass the early media ? |
| 12:32.17 | *** join/#asterisk justfahdi (n=fahad@116.71.1.241) |
| 12:33.29 | mbrevda | something like that |
| 12:34.13 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 12:34.39 | Dovid | or is it the opposite ? |
| 12:35.25 | mbrevda | 180 Ringing |
| 12:35.29 | mbrevda | 183 Session Progress |
| 12:35.52 | Dovid | where did u see that ? |
| 12:36.03 | mbrevda | Google, where else? |
| 12:36.03 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d420:f99e:15a5:c4a9) |
| 12:36.11 | mbrevda | http://en.wikipedia.org/wiki/List_of_SIP_response_codes |
| 12:38.04 | Dovid | i suck at googling |
| 12:38.19 | mbrevda | suck at googling=suck |
| 12:39.08 | aurax | sup Dovid |
| 12:39.17 | Dovid | nm |
| 12:39.17 | dwery | jgoo: on my bt 100 there's no dialplan that I can see |
| 12:43.46 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 12:44.01 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:45.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 12:54.50 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:54.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:55.27 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 12:57.41 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
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| 13:01.03 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:03.08 | Anth8708 | morning guys. I'm looking to see if there is a dialplan variable that will give hint status. Anyone know? I've found variables to give listing of devices with hints, but nothing to get the actual hint status |
| 13:03.15 | *** join/#asterisk Von_Lorenz (n=lorenzo@ip-89-162.sn1.eutelia.it) |
| 13:03.23 | Von_Lorenz | Hi to all! |
| 13:03.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:05.09 | Von_Lorenz | I have a little question, how can i set default language for all channels end all users in asterisk 1.6.0 ? |
| 13:06.34 | Anth8708 | quit |
| 13:06.54 | Chainsaw | Anth8708: The door's over there *points* |
| 13:07.08 | Anth8708 | heh. oops. sorry, typing without looking after alt-tabbing |
| 13:08.12 | *** join/#asterisk awk_r (n=awk@nat/digium/x-4541873b2870d7bf) |
| 13:08.39 | [TK]D-Fender | Von_Lorenz: You don't, its per channel driver |
| 13:09.05 | dwery | mm asterisk 1.6.0. was using 70% of the CPU.. I had to restart it.. there's no indication on the cause in the logs. I'll have to program an extension to restart it so the user can do on its own :( |
| 13:10.06 | Von_Lorenz | I must use language = xx for all kind of config files? True? |
| 13:10.16 | *** join/#asterisk pepe (n=pepesz@aran.et.tudelft.nl) |
| 13:11.20 | *** join/#asterisk yahh (n=root@122.169.93.71) |
| 13:12.38 | Von_Lorenz | If i want to change default language entries in pbx core settings, i need to edit asterisk.conf, its that true? |
| 13:13.58 | [TK]D-Fender | Von_Lorenz: No such thing there |
| 13:16.29 | Von_Lorenz | thanks D-fender |
| 13:19.26 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 13:19.29 | Katty | mew. |
| 13:21.04 | [TK]D-Fender | Katty: Mew. |
| 13:25.16 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 13:26.16 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.85) |
| 13:28.06 | Katty | i wish it was christmas time. |
| 13:29.13 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:29.54 | Faustov | hi |
| 13:30.32 | Katty | hi |
| 13:30.35 | Faustov | I'm having a problem dialing from a lan via asterisk to fring on a mobile - does anyone have a working example? |
| 13:30.56 | Katty | i don't |
| 13:30.56 | Faustov | i've set it up similarly to a lan phone, but with host=dynamic |
| 13:31.12 | Faustov | WARNING[17363]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 13:31.15 | Faustov | this is what im getting |
| 13:31.27 | Faustov | cause could be more descriptive ;) |
| 13:31.46 | *** join/#asterisk eliyahu (n=eliyahu@77.126.64.188) |
| 13:32.12 | Katty | makes a note to find some construction paper fall leaves cutouts to hang around her office. |
| 13:33.24 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 13:37.47 | eppigy | Katty: D: |
| 13:37.47 | Katty | :> |
| 13:37.47 | Katty | eppigy: let's go out for breakfast and get pumpkin bagels. |
| 13:37.51 | eppigy | oh dude I am down |
| 13:38.18 | Katty | i'm starving for real :< brb |
| 13:38.30 | eppigy | yeah dude Im going to hit mcdonalds |
| 13:38.37 | eppigy | in about 20 minutes |
| 13:40.04 | stope | app_fax.so won't load, error message --> Error loading module 'app_fax.so': /usr/lib64/asterisk/modules/app_fax.so: undefined symbol: t30_set_header_info |
| 13:40.14 | stope | is there a quick fix for it? |
| 13:40.21 | stope | asterisk 1.4.25 |
| 13:41.39 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 13:45.10 | Katty | eppigy: thats bad for you |
| 13:45.54 | tzafrir_laptop | stope, that symbol should come from spandsp |
| 13:46.19 | tzafrir_laptop | what do you get from: ldd /usr/lib64/asterisk/modules/app_fax.so |
| 13:46.53 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:47.21 | stope | http://pastebin.ca/1502033 |
| 13:47.27 | Katty | eppigy: of course i got a soda this morning too from the vending machine :< and that's bad for me. |
| 13:49.39 | stope | maybe I'll go to an older version of spandsp |
| 13:49.43 | stope | down to 4.x |
| 13:49.46 | stope | from 5.x |
| 13:50.19 | coppice | why does going backwards sound like a good idea? whatever happened to progress? |
| 13:51.04 | Katty | sometimes going backwards is a good idea. |
| 13:51.09 | Katty | maybe we'd all be healthier. |
| 13:51.18 | Katty | maybe our lives would be more simple. |
| 13:51.33 | coppice | not very often though, especially when you are already starting out with something ancient |
| 13:51.46 | Katty | i wasn't talking about asterisk there. |
| 13:53.50 | eppigy | :> |
| 13:54.08 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 13:54.36 | leifmadsen | Katty: I've quit drinking pop |
| 13:55.30 | Faustov | looks like the id for frings needs to be ID@IP_of_my_asterisk to dial it |
| 13:55.47 | Faustov | could that be the casE? |
| 13:55.51 | tzafrir_laptop | <stope> app_fax.so won't load, error message --> Error loading module 'app_fax.so': /usr/lib64/asterisk/modules/app_fax.so: undefined symbol: t30_set_header_info |
| 13:56.30 | tzafrir_laptop | stope, how is spandsp installed? from source? from a binary package? |
| 13:56.37 | stope | from an rpm |
| 13:56.45 | stope | but I've got the src and will compile |
| 13:56.51 | stope | then try to reload the little bugger |
| 13:56.52 | tzafrir_laptop | rpm -qa | grep spandsp |
| 13:57.14 | stope | lib64spandsp1-0.0.5-0.pre4.1mdv2009.0 |
| 13:57.16 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 13:57.28 | tzafrir_laptop | stope, no -devel package? |
| 13:57.33 | stope | n |
| 13:57.35 | Katty | leifmadsen: yeah so did i, for the most part. it's been several weeks. now that i have the soda, i took a couple sips and don't really want it. |
| 13:57.47 | coppice | interesting they put 2009 into such an old version's name :-\ |
| 13:57.47 | stope | I'll put the devel pkg in the system |
| 13:57.48 | leifmadsen | Katty: ya, I never get that feeling though :) |
| 13:57.49 | tzafrir_laptop | How about asterisk? |
| 13:58.06 | stope | oh, I can't admit that I have the rpm for * installed as well |
| 13:58.16 | tzafrir_laptop | hmm... is app_fax.c a backport from 1.6.0? your backport? |
| 13:58.22 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 13:58.40 | stope | its from the rpm for my distro |
| 13:59.20 | dwery | stope: you might want to try iaxmodem with hylafax. it's easy to setup and can be built with all the libs static |
| 13:59.46 | stope | ok, I'll try it.... :) |
| 14:00.00 | tzafrir_laptop | hmm... so that package was built vs. spandsp.so.1 (spandsp 0.0.5 -something) |
| 14:00.15 | stope | yes |
| 14:00.36 | stope | I went and got spandsp 4 pre 16 |
| 14:00.50 | stope | as it apparently had no rxfax txfax issues |
| 14:00.57 | coppice | bad direction. 0.0.6pre12 is the right thing to use these days |
| 14:01.05 | tzafrir_laptop | stope, that is wrong |
| 14:01.21 | stope | hmm, ok, I'll use 6 pre12 |
| 14:01.33 | stope | i was just reading the docs on the spandsp site |
| 14:02.49 | tzafrir_laptop | coppice, btw: spandsp 0.0.6 also resolves http://bugs.debian.org/537529 :-) |
| 14:03.01 | dwery | coppice: I've been told by the iaxmodem author that the are no particular improvements on the fax side of libspandsp in the latest version wrt 0.5 |
| 14:03.20 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:03.47 | coppice | no particular improvements in the T.31 FAX modem part. massive improvements elsewhere, though |
| 14:04.21 | stope | ok, I'll try it later on tonight... tx for the suggestions! |
| 14:04.52 | dwery | T.38 and T.30 I've been told |
| 14:06.10 | ariel_ | is having a very hard time configuring a audioCodes MP-112. There are so many settings for h323 that just don't make Sense.... argh why are people out there still using h323? Why? |
| 14:06.18 | coppice | tzafrir_laptop: people actually build this for the AVR32? :-\ |
| 14:06.37 | mbrevda | ariel_: why are you? |
| 14:07.09 | tzafrir_laptop | coppice, they build the whole Debian archive |
| 14:07.27 | tzafrir_laptop | Now whether or not the code gets run is a different story |
| 14:07.56 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:09.44 | *** join/#asterisk nemster (n=nemster@193.8.60.254) |
| 14:10.14 | coppice | tzafrir_laptop: the AVR32 doesn't have an FPU so it must be seriously slow running this stuff |
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| 14:15.17 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 14:16.36 | nemster | how can i call a sip extension and then playback a file if that extension answers? |
| 14:17.15 | *** join/#asterisk ketema (n=ketema@71.43.207.50) |
| 14:17.19 | [TK]D-Fender | nemster: "core show application dial" <- A() |
| 14:17.37 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 14:18.14 | ickmund | If I have the following in the dialplan, "exten => _1XX,1,ChanSpy(${EXTEN:1}|q)", and later wants to do a ChanSpy(...|qw) if * is pressed, how could I let asterisk know which extension it's on? |
| 14:21.30 | nemster | [TK]D-Fender: completely overseen A() thanks |
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| 14:26.02 | jaytee | I hate the fact that if I want to use tail on a file on a Windows machine I have to install the Win2K3 Server Resource Kit tools to get tail for Windows. Gaaah!!!! |
| 14:27.09 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 14:27.11 | nemster | jaytee: afair there is tail32 |
| 14:27.38 | Katty | jaytee: we need a vacation. let's go camping. |
| 14:28.03 | *** join/#asterisk mphill (n=mphill@174.37.19.92) |
| 14:28.04 | Katty | jaytee: a cooler full of alcamahols sounds exciting |
| 14:29.00 | mphill | does anyone know off hand if aastra config files support an include syntax so you can include other external config files? |
| 14:29.07 | *** join/#asterisk Joel (n=jjshoe@CPE-70-92-78-170.new.res.rr.com) |
| 14:29.21 | coppice | stope: what did you see on the spandsp site that lead you to try ancient versions? |
| 14:29.26 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 14:29.32 | jaytee | Katty, sounds like fun |
| 14:29.56 | Katty | jaytee: yesh |
| 14:30.01 | jaytee | gotta go test a POS credit card transaction, bbiab |
| 14:30.03 | afink | is this the correct syntax? exten => _3100,1,SetVar(cidnumber=${CALLERID(num)) |
| 14:30.07 | Katty | k |
| 14:30.31 | Katty | i think you're missing a } |
| 14:30.33 | nemster | mphill: yes |
| 14:30.50 | mphill | nemster: thank you, do you know what it is? |
| 14:31.08 | afink | oh yeah duh, do I need " " around the ${CALLERID(num)} to evaluate it? |
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| 14:31.38 | nemster | mphill: err, i might be wrong. we just have the global file which then includes the mac-specific files |
| 14:32.38 | mphill | nemster: i think you are right. i searched the admin documentation. What a bummer. |
| 14:33.25 | nemster | so you might need to generate the files if you want to include which would be simple toughy |
| 14:34.17 | [TK]D-Fender | afink: No, that is still 1.0 crap |
| 14:34.31 | [TK]D-Fender | afink: Mixed 1.0 / 1.2 |
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| 14:36.02 | leifmadsen | afink: you don't need " " to evaluate |
| 14:36.22 | leifmadsen | afink: exten => 3100,1,Set(cidnumber=${CALLERID(number)}) |
| 14:36.22 | stope | coppice, I just went with the rpm that was readily available.... but I'll upgrade to the latest |
| 14:36.26 | leifmadsen | (num or number both work) |
| 14:36.40 | leifmadsen | afink: additionally -- you only need to prefix your extension with _ when you are doing a pattern match |
| 14:37.29 | coppice | stope:ah, OK. I just checked through the site, and found a typo that says 0.0.6pre2 rather than 12 is current, but I can't see any references to 0.0.5 or 0.0.4 |
| 14:38.21 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:38.28 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 14:38.50 | [TK]D-Fender | afink: And what are you going to do with that var that you couldn't with the function istelf? |
| 14:39.24 | afink | [TK]D-Fender: passing it to a macro |
| 14:39.52 | stope | http://www.soft-switch.org/downloads/spandsp/ |
| 14:39.57 | [TK]D-Fender | afink: You could pass it directly in the call |
| 14:40.16 | FreakGuard | what's a good introduction into the whole VOIP stuff? |
| 14:40.59 | afink | [TK]D-Fender: how so? |
| 14:42.56 | *** join/#asterisk Orbixx (i=Orbixx@office.exoware.net) |
| 14:44.10 | [TK]D-Fender | afink: Pass it jsut like anything else... |
| 14:44.27 | [TK]D-Fender | FreakGuard: "the whole VOIP stuff"? Huh? |
| 14:46.11 | FreakGuard | [TK]D-Fender: hm. sorry, I'm trying to configure an asterisk server using adhearsion, but I doubt I'll get far by just trying things out, I'm looking for some founding knowledge |
| 14:47.50 | [TK]D-Fender | FreakGuard: GUI's aren't supported here, and "founding knowledge... for that your question is entirely too vague |
| 14:48.13 | spackle | FreakGuard: go check out the book (or download) Asterisk: the future of IP Telephony |
| 14:48.32 | Orbixx | The book is seriously comprehensive. |
| 14:48.35 | Orbixx | Pack some patience. |
| 14:49.41 | *** join/#asterisk Joel (n=jjshoe@CPE-70-92-78-170.new.res.rr.com) |
| 14:50.10 | spackle | Anybody have experience registering an Adtran 712/706 phone to Asterisk? |
| 14:50.15 | FreakGuard | Orbixx: that's not problem there |
| 14:50.25 | [TK]D-Fender | FreakGuard: What is your reason for choosing Adhearsion? |
| 14:51.07 | *** join/#asterisk Joel (n=jjshoe@CPE-70-92-78-170.new.res.rr.com) |
| 14:51.09 | FreakGuard | [TK]D-Fender: I was looking for something to cleanly configure asterisk |
| 14:51.12 | Orbixx | Can somebody explain how I can pipe incoming calls from an IAX * pbx to an extension? |
| 14:51.26 | Orbixx | (and outgoing) |
| 14:51.28 | FreakGuard | well, sure, you can use the default asterisk configs, but they're a bit... unreadable |
| 14:51.41 | [TK]D-Fender | FreakGuard: Poor choice |
| 14:51.51 | [TK]D-Fender | FreakGuard: All it is is another layer like AEL |
| 14:52.25 | [TK]D-Fender | FreakGuard: And putting one shaky surface on another won't make your long-term experience too stable |
| 14:52.45 | [TK]D-Fender | Orbixx: "core show application dial" |
| 14:53.39 | FreakGuard | [TK]D-Fender: Well, let's see what the other guys are doing |
| 14:55.01 | afink | Would making a macro for the first 4 lines of this:http://pastebin.com/m1a8d03c7 be the best way to do this? I have these 4 lines repeated lots of places in my dialplan |
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| 14:59.39 | [TK]D-Fender | afink: Sure. first, make it a GOSUB that sets a variable for whether or not it is bad and not a MACRO, and call that. |
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| 15:02.41 | afink | [TK]D-Fender: ok makes sense thanks |
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| 15:05.23 | Von_Lorenz | see you |
| 15:05.40 | spackle | Anybody have experience registering an Adtran 712/706 phone to Asterisk? |
| 15:06.44 | [TK]D-Fender | spackle: About as many as 15 minutes ago |
| 15:07.22 | spackle | D-Fender, indeed, but people come and people go. |
| 15:07.54 | *** join/#asterisk pepe (n=pepesz@aran.et.tudelft.nl) |
| 15:08.01 | [TK]D-Fender | spackle: Oh look, another! |
| 15:09.18 | [TK]D-Fender | spackle: And I've only heard someone even ask about those products once, not even owning one. |
| 15:10.58 | spackle | D-Fender, I've registered dozens of freaky phones from Grandstream, Hitachi, Uniden, Sipura and Polycom and have not had the problems with these. It may be time to call Adtran. |
| 15:13.52 | [TK]D-Fender | spackle: Quite possibly. |
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| 15:32.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.42 | bmoraca | Mitel makes a really cool VoIP phone that has a SunRocket thin client built in to it. too bad it's a proprietary protocol |
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| 15:42.27 | Katty | so. |
| 15:42.38 | Katty | i'm thinking about getting a camcorder. to take videos of the pup. |
| 15:42.45 | Katty | what should i get. |
| 15:43.29 | Katty | thinking something with a harddrive in it. |
| 15:44.01 | Katty | http://www.newegg.com/Product/Product.aspx?Item=N82E16830120227 <- like that. |
| 15:44.45 | bmoraca | With an SPA3102, I can have the FXO port associated with a SIP peer on my PBX, correct? Basically, what needs to happen is when a call comes in on that particular analog line, I need it to ring a certain number of phones. I shouldn't have an issue doing that, correct? |
| 15:46.12 | [TK]D-Fender | bmoraca: Correct, and no. |
| 15:46.23 | bmoraca | good to hear. |
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| 15:50.50 | bmoraca | completely unfamiliar with those devices...i know the concept, but the manual is rather unclear. |
| 15:54.00 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
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| 16:05.19 | synapseattack | Anyone ever try setting very low RTP ports? We are trying to set rtpstart=100 and rtpstop=1000 but it doesn't seem to be following this as the ports are still high after restarting the asterisk process. I know this isn't exactly the best way to go about this but we are trying to get past some blocking. I didn't see anything in issues.asterisk.org related to this. |
| 16:07.30 | afink | with gui: http://pastebin.com/m6f346f3 without gui: http://pastebin.com/m567a653b plus this one has the users in it! what a difference |
| 16:08.01 | leifmadsen | synapseattack: you can't use ports below 1024 |
| 16:08.18 | leifmadsen | synapseattack: it's a linux thing -- they are restricted ports |
| 16:08.25 | leifmadsen | (not an asterisk problem) |
| 16:08.32 | synapseattack | thats what i figured |
| 16:08.33 | leifmadsen | synapseattack: is asterisk running as root? |
| 16:08.52 | *** join/#asterisk eliyahu (n=eliyahu@77.126.64.188) |
| 16:09.08 | synapseattack | @leifmadsen: yes |
| 16:09.31 | leifmadsen | synapseattack: ya, I think you're outta luck from using those ports then |
| 16:09.36 | synapseattack | @leifmadsen: it's cool we will find a different way to go about it. |
| 16:09.43 | synapseattack | @leifmadsen: thanks |
| 16:09.46 | dwery | tho those who like to play with the Siemens C470IP I just found that it can be auto provisioned. The problem is that it can be easily crippled as I did with my own :D you will find several clues using "strings" on the firmware or on the binary modules distributed with the source code. there's an hidden page to send a dump of your configuration to Siemens and another one to probably st a remote source for the phonebook. |
| 16:10.35 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:10.58 | spackle | Anybody have experience registering an Adtran 712/706 phone to Asterisk? |
| 16:11.47 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:11.53 | bmoraca | spackle: maybe if you detailed the issue you're having (sip debugs, etc), someone might be able to steer you in the right direction...SIP is SIP is SIP, so it's likely someone's had the same problem with a different phone before. |
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| 16:12.39 | *** join/#asterisk pawpro (n=pawpro@213.166.12.73) |
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| 16:13.54 | spackle | bmoraca: I forgot about the debug commands. But I'm not sure the phone is sending anything. IT supports several lines and has options for registrars and proxies. I have a call in to Adtran, but it seemed some more phone-savvy people were about just now. |
| 16:13.56 | *** join/#asterisk Joel (n=jjshoe@CPE-70-92-78-170.new.res.rr.com) |
| 16:14.15 | spackle | bmoraca: thanks for your suggestion, I will give it a try. |
| 16:14.28 | pawpro | Hello everybody. Is there a way to access a peer/friend name or defaultuser in dialplan (extensions.conf)? The peers are set in realtime. Thank you. |
| 16:14.32 | bmoraca | spackle: if you haven't done a sip debug, how can you know whether or not the phone is attempting to register? |
| 16:15.07 | leifmadsen | pawpro: eh? what do you mean? |
| 16:15.35 | pawpro | leifmadsen: the sipphone is making a call and it's put into certain context. can i get the it in there? |
| 16:15.56 | leifmadsen | pawpro: I don't understand what you're trying to do... |
| 16:16.07 | leifmadsen | the question doesn't make sense to me |
| 16:16.22 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 16:16.36 | pawpro | leifmadsen: I need to pass distinct reference to this peers account to another asterisk using custom sip header... |
| 16:16.43 | bmoraca | i think he's looking for a dialplan variable that will give him the name of the sip peer |
| 16:16.55 | pawpro | bmoraca: exactly! |
| 16:17.11 | leifmadsen | pawpro: ${CHANNEL} |
| 16:17.41 | pawpro | leifmadsen: ill this not give me a bit more that i need? :) |
| 16:17.46 | leifmadsen | exten => start,n,Set(peername=${CUT(CUT(CHANNEL,-,1),/,2)}) |
| 16:18.02 | spackle | bmoraca: good, call, I'm missing a config option somewhere on the phone. It does nt seem to be poking at Asterisk at all. hmmm. |
| 16:18.02 | leifmadsen | pawpro: your questions confuse me greatly... |
| 16:18.24 | leifmadsen | pawpro: just strip off the parts you don't need |
| 16:19.24 | pawpro | leifmadsen: peer is authenticated on sip gateway (asterisk) and then call goes to siptrunk app (asterisk) via openser. I need the siptrunk app to know the peers name so i'll use custom header to pass it to him. App is written in PHP AGI. |
| 16:19.59 | pawpro | leifmadsen: This will be sufficient. Thank you leifmadsen! |
| 16:20.00 | Qwell | <3 TCPA |
| 16:20.06 | leifmadsen | pawpro: ok, so go ahead ;) Set(SIP_HEADER(X-origpeer)=${CUT.........}) |
| 16:20.16 | Qwell | at this rate, Chase is going to be paying off my car for me. |
| 16:20.47 | bmoraca | tcpa is the devil |
| 16:21.02 | Qwell | bmoraca: only if you violate it :p |
| 16:21.38 | jameswf | Qwell: your being hurassed by a blogger? |
| 16:21.42 | bmoraca | pawpro: this page might be useful to you in the future: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List |
| 16:21.49 | Qwell | jameswf: is @chase just a blogger? well, whatever |
| 16:21.51 | pawpro | Thank you! |
| 16:21.57 | jameswf | lol |
| 16:22.00 | Qwell | people get the idea :P |
| 16:22.09 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 16:22.21 | jameswf | @chase will be like wait wtf did I do |
| 16:22.29 | Qwell | jameswf: he's used to it I'm sure |
| 16:22.54 | jameswf | that is what he gets for name squatting on twitter |
| 16:22.58 | Qwell | mmhmm |
| 16:24.04 | Qwell | better? |
| 16:26.36 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
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| 16:32.49 | ocnarf | Does anyone know what this error means, ERROR[27295] astobj2.c: refcount -1 on object 0xb2b47408? |
| 16:33.18 | ocnarf | When asterisk crash, i always see that on the /log/asterisk/messages |
| 16:33.30 | Qwell | ocnarf: it's a "Bad Thing". what version of Asterisk? |
| 16:33.52 | ocnarf | Asterisk 1.4.25.1 |
| 16:33.59 | ocnarf | Do i need to change version? |
| 16:34.25 | ocnarf | ASterisk crashes almost 2-3x a week |
| 16:34.30 | Qwell | you could try the latest 1.4.26 RC (or wait a bit and 1.4.26 will be released) |
| 16:35.01 | ocnarf | any suggestion? |
| 16:35.09 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 16:35.15 | ocnarf | any other version i could try? |
| 16:35.23 | *** part/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 16:35.23 | Qwell | I just said... |
| 16:36.36 | ocnarf | hmmm.. is it logical to downgrade? |
| 16:36.58 | Qwell | unlikely |
| 16:37.20 | Qwell | you may end up running into different (fixed..) bugs instead |
| 16:39.12 | *** join/#asterisk eliyahu (n=eliyahu@77.126.64.188) |
| 16:40.03 | jaytee | "Forward!!! Into the past!!! Tally ho, men!" |
| 16:40.30 | ocnarf | thanks Qwell |
| 16:40.41 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:41.41 | pawpro | leifmadsen: What am I missing "/usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_get_encoded_str"? |
| 16:42.40 | leifmadsen | pawpro: I have no idea -- please just ask the room generally |
| 16:45.18 | pawpro | Hello everybody. When i run "exten => _X.,1,SIPAddHeader(X-SIPTrunkID: ${CUT(CUT(CHANNEL,-,1),/,2)})" asterisk crashes with "chan_sip.so: undefined symbol: ast_get_encoded_str". Can anyboduy help please? |
| 16:46.32 | Qwell | pawpro: what version of Asterisk? |
| 16:46.39 | pawpro | btw. i have autoload=yes. Asterisk 1.6.0.5 compiled on FC8 2.6.23.1-42 |
| 16:48.32 | Qwell | sounds like you're using a module that wasn't compiled against/with that version of Asterisk |
| 16:48.54 | pawpro | Qwell: i'll compile 1.6.1 do you agree? |
| 16:49.37 | Qwell | pawpro: yes |
| 16:49.45 | Qwell | 1.6.1.1 anyways |
| 16:49.48 | [TK]D-Fender | pawpro: Don't run a nested CUT |
| 16:50.04 | Qwell | [TK]D-Fender: as written, that actually should work |
| 16:50.14 | pawpro | [TK]D-Fender: on the console it looks like its doing the right thing |
| 16:50.28 | Qwell | You'll note he didn't use ${CUT()} for the inner one - which is correct |
| 16:51.00 | pawpro | -- Executing [078675768@from-sip:1] SIPAddHeader("SIP/officetrixbox-09a8b730", "X-SIPTrunkID: officetrixbox") in new stack |
| 16:51.11 | Qwell | trixbox. there's your problem |
| 16:51.24 | pawpro | Qwell trisbox is just peer |
| 16:51.59 | [TK]D-Fender | Qwell: I noticed. |
| 16:52.15 | [TK]D-Fender | pawpro: remove the nesting |
| 16:52.43 | pawpro | [TK]D-Fender: Could you assist me with extracting peername from CHANNEL then? Please |
| 16:53.04 | [TK]D-Fender | pawpro: I just said don't STACK THEM. Split it into 2 lines. |
| 16:53.17 | pawpro | [TK]D-Fender: ok thank you |
| 17:00.40 | Katty | has cheddar tomato dumplings cooking. |
| 17:01.11 | Katty | REF: http://www.tasteofhome.com/Recipes/Cheddar-Tomato-Dumplings/Print |
| 17:01.15 | *** join/#asterisk errotan (n=errotan@62.201.123.54) |
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| 17:04.21 | [TK]D-Fender | pawpro: and? |
| 17:06.17 | Orbixx | Can anybody either explain or direct me to a guide that explains how to set up IVR, queues and other stuff a callcenter would be expected to have? |
| 17:06.25 | *** join/#asterisk path (n=~tarro@pc-95-122-161-190.cm.vtr.net) |
| 17:07.03 | [TK]D-Fender | Orbixx: "core show function TIMEOUT", "core show application waitexten" |
| 17:09.43 | Orbixx | [TK]D-Fender: So instead of directing an incoming call to an extension, how do I get waitexten to execute immediately when somebody dials? |
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| 17:12.36 | [TK]D-Fender | Orbixx: .... call comes in. It lands on an extension. Jump to "s" in a new context and set up your timeouts, Background() some prompts and await input |
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| 17:14.10 | Katty | omnomnomnomnom |
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| 17:26.05 | path | how can I forward all my incoming calls to an analog or cellphone phone number ? |
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| 17:26.29 | Katty | path: call your telco |
| 17:26.31 | Orbixx | How do I invoke an audio message to play back? |
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| 17:26.36 | Katty | path: and tell them to put a temporary forward on your pri/lines |
| 17:26.42 | WindowsUser | Orbixx: Playback()? |
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| 17:27.11 | Katty | path: we do that here in case of extended power outages and so forth |
| 17:27.23 | path | so far I've got this http://codepad.org/HnLILLCA |
| 17:27.43 | *** join/#asterisk hmmhesays (n=hmmhesay@97-112-201-74.frgo.qwest.net) |
| 17:27.55 | hmmhesays | whats going on guys? |
| 17:28.06 | Katty | hi matt |
| 17:28.19 | path | Katty: telco == my SIP provider ? |
| 17:28.20 | Katty | path: you do not get a gold star for not listening. |
| 17:28.32 | path | :( |
| 17:28.32 | hmmhesays | hola |
| 17:28.53 | *** join/#asterisk holos (n=cosmond@209.167.131.35) |
| 17:28.57 | hmmhesays | you're stuck with a mr yuck sticker |
| 17:29.01 | *** join/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 17:29.42 | Katty | *hee* |
| 17:29.46 | martha | anyone currently in canada here who could do me a very quick favor? I need to see if a toll-free number reaches my asterisk server or not.... |
| 17:30.01 | path | Katty: so that means I need permission from my local company to forward calls ? |
| 17:30.02 | WindowsUser | whats the number |
| 17:30.13 | Katty | facepalm |
| 17:30.14 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 17:30.49 | WindowsUser | path: if you want ALL calls forwarded the suggestion was to do it before calls get to asterisk, be it at telus or your sip provider |
| 17:31.16 | path | ok I get it now |
| 17:31.27 | [TK]D-Fender | Orbixx: Blackground() <--- |
| 17:31.38 | martha | 1-866-490-1409 |
| 17:31.41 | hmmhesays | thats not marthag is it? |
| 17:31.51 | path | WindowsUser: just for curiosity, why should it be before ? |
| 17:31.59 | path | performance issues? |
| 17:32.07 | WindowsUser | why bother looping? |
| 17:32.26 | Orbixx | [TK]D-Fender: What format does that accept and where does it look for it? |
| 17:32.26 | path | thanks :-) |
| 17:33.54 | p4tr0p1 | hi, I have a queue with MusicOnHold, I need to put another audio when the call is not answered yet informing that soon somebody will answer the call, anybody knows how? |
| 17:34.12 | Katty | p4tr0p1: play them the audio file, then put them on hold. |
| 17:34.13 | WindowsUser | Orbixx: it'll take files that asterisk will play, like .wav (8000hz) ulaw g729 etc etc, and you can give it a full path |
| 17:34.23 | Katty | p4tr0p1: and setup an extension that records it, then dial the extension and record what you want. |
| 17:35.23 | [TK]D-Fender | Orbixx: it will play any of the formats that * can read. "core show modules like format" |
| 17:35.33 | Orbixx | WindowsUser: If I'm receiving a call from PSTN which uses alaw codec (UK) and I have different codecs to that, what happens? |
| 17:35.46 | Orbixx | I don't seem able to play any messages back. |
| 17:35.52 | [TK]D-Fender | Orbixx: And without an absolute path it looks in 'sounds" in the Lib folder listed in asterisk.conf |
| 17:36.00 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:36.11 | [TK]D-Fender | Orbixx: * will transcode as needed / able |
| 17:36.53 | Orbixx | Whatever I'm playing (default sounds) seems to come out as a garble. |
| 17:37.10 | p4tr0p1 | Katty, I guess I'll mix the files =) |
| 17:38.37 | *** part/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 17:39.19 | [TK]D-Fender | *b00m* |
| 17:39.21 | WindowsUser | default sounds as in stuff included with asterisk? |
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| 17:39.26 | Orbixx | hmm |
| 17:39.26 | Orbixx | it seems to be playing back but it's incredibly stuttery |
| 17:39.27 | Katty | p4tr0p1: oh you mean like you want it randomized in there? |
| 17:39.27 | Orbixx | WindowsUser: Yes. |
| 17:39.27 | Katty | does the musiconhold() have a time on it? |
| 17:39.47 | [TK]D-Fender | Orbixx: Bandwidth, jitter, CPU (VM) may all contribute to this as well as : |
| 17:39.47 | [TK]D-Fender | ~gsmbug |
| 17:39.48 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 17:39.48 | Katty | 1,1,playback(KEEPYOURPANTSON) 1,2,musiconhold(default) <- with a time of 60 seconds, 1,3,Playback(KEEPYOURPANTSON) |
| 17:39.48 | [TK]D-Fender | ^^^^^^^^^^^ |
| 17:39.48 | WindowsUser | ewww I only get stutter when my router is crashing |
| 17:39.50 | [TK]D-Fender | darn, infobot is out. |
| 17:40.01 | [TK]D-Fender | ^^^^^^^^^^^ |
| 17:40.01 | [TK]D-Fender | there |
| 17:40.15 | Orbixx | [root@r26209 en]# gcc --version |
| 17:40.15 | Orbixx | gcc (GCC) 4.1.2 20080704 (Red Hat 4.1.2-44) |
| 17:40.55 | [TK]D-Fender | Orbixx: Describe the rest of your scenario. |
| 17:41.17 | Orbixx | Well at the moment, I'm just messing around, it's just literally playing back a file when somebody dials in and that's it. |
| 17:41.29 | Orbixx | exten => 03333408856,1,Playback(vm-tooshort) |
| 17:41.45 | Orbixx | Plus does that bug exist in 1.6.1.1? |
| 17:42.12 | p4tr0p1 | Katty, yes. I want to play something with a 20 seconds interval while the music on hold it playing |
| 17:42.33 | *** join/#asterisk Strale (n=sk@79.126.252.16) |
| 17:42.42 | Katty | p4tr0p1: check out the music on hold application and see if that's an option. i think it is. |
| 17:42.53 | Katty | p4tr0p1: or you can do what we do. |
| 17:43.15 | Katty | p4tr0p1: and record averts, with music on hold, and make the moh go in order rather than randomized. |
| 17:43.34 | Katty | p4tr0p1: then every 20 seconds insert the appropriate file. 01 02 03 04 etc |
| 17:44.08 | *** join/#asterisk Strale (n=sk@79.126.252.16) |
| 17:44.10 | Strale | How many channels are occupied by this dialplan : exten -> answer() | exten -> Set(something)=$channel | exten -> dial(12345) |
| 17:44.38 | p4tr0p1 | Katty, hummm, i'll read about that! Tks =) |
| 17:44.54 | Orbixx | [TK]D-Fender: Happening with alaw too, not just gsm. |
| 17:44.59 | Katty | p4tr0p1: 2 |
| 17:45.11 | Katty | p4tr0p1: incoming line is held open when you dial OUT |
| 17:45.18 | Katty | p4tr0p1: unless that's an internal sip call. |
| 17:45.23 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:45.24 | [TK]D-Fender | Orbixx: Really... what format is your SOURCE recording in? |
| 17:45.27 | Katty | p4tr0p1: in which case just your incoming line/channel is held open |
| 17:45.43 | Katty | p4tr0p1: if you do a blind transfer out the building, it will hold open both channels until the cellphone hangs up |
| 17:45.53 | [TK]D-Fender | Strale: 2 |
| 17:45.57 | Orbixx | [TK]D-Fender: Sorry, I don't quite understand. |
| 17:46.08 | WindowsUser | Orbixx: what are you playing back |
| 17:46.10 | *** join/#asterisk apeiron_ (i=apeiron@isuckatdomains.net) |
| 17:46.13 | Katty | oops. s / p4tr0p1 / Strale :P |
| 17:46.19 | WindowsUser | .gsm? .alaw? |
| 17:46.25 | WindowsUser | .su? ;) |
| 17:46.29 | Orbixx | .alaw |
| 17:46.30 | Katty | ulaw, he she or itlaw |
| 17:46.36 | Strale | [TK]D-Fender : how to set channel variable of dialed number, I like to hangup from asterisk manager interface |
| 17:46.43 | [TK]D-Fender | Orbixx: Then its one of the other reasons I told you |
| 17:46.46 | *** join/#asterisk hfb (n=hfb@pool-96-251-62-168.lsanca.dsl-w.verizon.net) |
| 17:46.59 | Orbixx | [TK]D-Fender: I just tested it over IP only, it works fine. |
| 17:47.05 | Orbixx | It just screws up when calling from a PSTN. |
| 17:47.23 | [TK]D-Fender | Orbixx: that is a really hollow and meaningless answer.. |
| 17:47.31 | [TK]D-Fender | Orbixx: DETAILS |
| 17:47.41 | Orbixx | I'm not sure what to tell you. |
| 17:47.42 | Strale | [TK]D-Fender : or maybe it's ok to hangup the channel that I have alredy setup like that dialplan ?? |
| 17:47.51 | Orbixx | Sorry. |
| 17:48.05 | [TK]D-Fender | Orbixx: Gee, i dunno.. how the hell do you GET to the PSTN? |
| 17:48.20 | Anth8708 | hey guys, quick question on comedian mail. can you "skip" the message envelope (not have to listen to the intro who/when info) with a key? |
| 17:48.32 | Orbixx | [TK]D-Fender: IAX to the PBX of a provider who has PSTN access. |
| 17:48.33 | [TK]D-Fender | Anth8708: "#" |
| 17:48.42 | Anth8708 | [TK]D-Fender thanks man |
| 17:48.55 | [TK]D-Fender | Orbixx: Then they (or your connection to them) has issues |
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| 17:49.29 | Orbixx | [TK]D-Fender: But I'm making incoming and outgoing calls using alaw codec through them and they're fine. |
| 17:49.52 | *** join/#asterisk ReD-MaN (i=rox-ur-s@209.183.147.106) |
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| 17:50.52 | Strale | [TK]D-Fender if I hangup something will the second channel be automatically hangup ?? : exten -> answer() | exten -> Set(something)=$channel | exten -> dial(12345) |
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| 17:51.26 | [TK]D-Fender | Orbixx: you still didn't specify codecs, etc... Where did your recording come from.... |
| 17:51.38 | Anth8708 | [TK]D-Fender, can that key be customized in voicemail.conf? I see a few customizations there? Trying to mimic callpilot as best I can. |
| 17:51.48 | [TK]D-Fender | Anth8708: No |
| 17:51.50 | Orbixx | The recordings came from digium [TK]D-Fender, I've tried gsm and alaw playback. |
| 17:51.59 | Anth8708 | [TK]D-Fender: rgr. no biggie then. |
| 17:53.58 | Anth8708 | [TK]D-Fender: ahh . I didn't state clearly. When logged into a voicemail box and listening to messages, can the listener skip the introductory who/when of the voicemail with a key? |
| 17:54.38 | [TK]D-Fender | Anth8708: Not afaik. Stop playing them all the time and let them coose to play it |
| 17:55.14 | Anth8708 | [TK]D-Fender: Roger. Will ask the user their preference as we setup voicemail boxes. Sounds like a winner. Thanks. |
| 17:55.43 | *** join/#asterisk asasas (n=asasasas@c-68-61-183-129.hsd1.mi.comcast.net) |
| 17:56.08 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.26 (2009/07/21), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.2.0.1 (2009/06/30), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asteris |
| 17:56.36 | leifmadsen | Asterisk 1.4.26 has been released! For more information see the release announcement at http://www.asterisk.org/node/48610. Thank you for your continued support of Asterisk! |
| 17:59.15 | Orbixx | [TK]D-Fender: A friend suspects an echo cancellation problem. |
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| 18:03.46 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:04.36 | raden_work | I have my DSL modem bridged to my router is there a way to create a route in my router via the wan to access the modem control panel @ 192.168.0.1 |
| 18:04.45 | raden_work | or is this not possible because of the bridge |
| 18:05.52 | [TK]D-Fender | raden_work: Bounce it off a *NIX box |
| 18:09.11 | *** join/#asterisk raden (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:09.16 | [intra]lanman | anyone gonne be here except jtodd? http://www.digium.com/en/events/viewevent/102 |
| 18:09.21 | raden | [TK]D-Fender, how so ? |
| 18:09.51 | [TK]D-Fender | raden : SSH tunnel, etc |
| 18:10.13 | Qwell | [intra]lanman: Please don't advertise here. |
| 18:10.27 | [intra]lanman | Qwell: are you kidding me? |
| 18:10.31 | [intra]lanman | it's on digium's site |
| 18:10.41 | Qwell | What's your point? |
| 18:10.56 | raden | [TK]D-Fender, modem connected to Wan on Netgear router how would i SSH to modem if i cant get a route to it ? |
| 18:11.05 | [intra]lanman | Qwell: well... it's not me that's advertising |
| 18:11.22 | [TK]D-Fender | [intra]lanman: No, it'll only be jtodd. He'll be so lonely... also it'll be hard to explain to his boss about the cost effectiveness of sitting in an empty room having wasted airfare, lodging, etc.... |
| 18:11.34 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) |
| 18:11.35 | leifmadsen | won't be there. |
| 18:12.26 | [TK]D-Fender | raden : go read your router's manual. |
| 18:13.03 | raden | been doing that guess ill keep doing that |
| 18:13.38 | [intra]lanman | [TK]D-Fender: haha, that's grade A humour... so i'm guessing you won't be attending, huh? |
| 18:13.47 | [intra]lanman | s/humour/sarcasm/ |
| 18:14.15 | [TK]D-Fender | [intra]lanman: Nope. I can chat with jtodd any time... far cheaper too :p |
| 18:15.05 | *** part/#asterisk Strale (n=sk@79.126.252.16) |
| 18:15.06 | [intra]lanman | yeah, that's how i feel about most conferences... most are more of a commercial to buy stuff than informative |
| 18:18.58 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
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| 18:22.33 | raden | is it even possible to create a route to a bridged modem since the router has an external IP from the ISP ? \ |
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| 18:23.07 | [TK]D-Fender | raden: how does the router talk to the modem? |
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| 18:27.12 | mercutioviz | has anybody been working on the S-prize? |
| 18:29.47 | [TK]D-Fender | mercutioviz: You mean the X-Prize wasn't enough? What would you give for an "H"? |
| 18:30.10 | [TK]D-Fender | inflates the market |
| 18:30.22 | *** join/#asterisk voxter (n=voxter@190.241.15.56) |
| 18:30.32 | mercutioviz | http://blogs.digium.com/2009/02/18/s-prize/ |
| 18:30.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 18:33.29 | bmoraca | raden: no. |
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| 18:34.16 | raden | [TK]D-Fender, the modem is set as bridged, the modem ethernet port is connected to wan the router does all the authentication with the ISP |
| 18:34.19 | *** join/#asterisk davevg-btwtech (n=davevg-b@75.97.64.33) |
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| 18:34.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 18:34.51 | *** mode/#asterisk [+o angler] by ChanServ |
| 18:34.55 | bmoraca | raden: by virtue of being a bridge, that modem is completely invisible to your router. |
| 18:35.07 | raden | i figured that |
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| 18:37.14 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:37.30 | hmmhesays | that blog link broken for anyone else? |
| 18:37.32 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:38.10 | mercutioviz | hmm, strange. it isn't responding for me even though I just copied it... |
| 18:38.32 | Qwell | we're apparently having internet difficulties atm... |
| 18:38.46 | mercutioviz | quick! blame n. korea |
| 18:38.57 | hmmhesays | apache must be avoiding deadlock |
| 18:39.01 | Qwell | though...I'm still here. wtf |
| 18:39.34 | [intra]lanman | hmmhesays: :-D |
| 18:39.57 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 18:41.22 | *** join/#asterisk MmixX (n=mix@61.14.191.137) |
| 18:41.51 | hmmhesays | [intra]lanman, thats quite the smile [intra]lanman |
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| 18:42.14 | [intra]lanman | hmmhesays: sorry... forgot to brush... :-| |
| 18:42.46 | hmmhesays | yeah that happens |
| 18:43.24 | raden | brb |
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| 18:47.51 | *** mode/#asterisk [+o angler] by ChanServ |
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| 18:48.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 18:53.19 | mercutioviz | okay, seems to be back: http://blogs.digium.com/2009/02/18/s-prize/ |
| 18:53.38 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 18:53.39 | mercutioviz | anyway, I was just curious if anybody was working for that steak dinner. :P |
| 18:54.54 | [TK]D-Fender | mercutioviz: Yeah... I'm seeing all sorts of "curious" folk around here today... |
| 18:55.23 | mercutioviz | is curious |
| 18:57.04 | Olobola | I can't get DTMF tones to pass to Verizon Wireless' phone maze. DTMF works elsewhere. Where should I start? |
| 18:58.22 | *** join/#asterisk Sincan2 (n=sincan@203.89.24.71) |
| 18:58.30 | Sincan2 | hi hello |
| 18:58.43 | Sincan2 | can you help me about install asterisk |
| 18:58.45 | pepe | Hello, I'm using Asterisk 1.6.2.0-beta3 with asterisk-addons-1.6.2.0-rc1 to write cdr into mysql. I followed http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk, and it is mostly working however CLID is not written to mysql, although it appears in Master.csv. What do I possibly missing? TIA |
| 19:00.06 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:00.40 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 19:00.48 | [TK]D-Fender | Sincan2: Instructions are in the tarball |
| 19:00.57 | [TK]D-Fender | Siwith the giant title "README" |
| 19:01.03 | [TK]D-Fender | Sincan2: with the giant title "README" |
| 19:01.09 | Sincan2 | emm |
| 19:01.20 | Sincan2 | <PROTECTED> |
| 19:01.21 | Sincan2 | Connected to Asterisk 1.4.24 currently running on localhost (pid = 2537) |
| 19:01.21 | Sincan2 | Verbosity was 3 and is now 7 |
| 19:01.21 | Sincan2 | <PROTECTED> |
| 19:01.21 | Sincan2 | <PROTECTED> |
| 19:01.21 | Sincan2 | <PROTECTED> |
| 19:01.23 | Sincan2 | <PROTECTED> |
| 19:01.25 | Sincan2 | <PROTECTED> |
| 19:01.27 | Sincan2 | <PROTECTED> |
| 19:01.29 | Sincan2 | <PROTECTED> |
| 19:01.31 | Sincan2 | <PROTECTED> |
| 19:01.35 | Sincan2 | this found |
| 19:01.35 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 19:01.37 | Sincan2 | why ? |
| 19:01.46 | [TK]D-Fender | Sincan2: do NOT spam in here again. Use a PASTEBIN |
| 19:01.48 | [TK]D-Fender | ~pb |
| 19:01.49 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 19:01.50 | Sargun | o.O |
| 19:01.50 | [TK]D-Fender | ^^^^^^^^ |
| 19:01.59 | *** join/#asterisk Defraz (n=T0tal@24-117-156-215.cpe.cableone.net) |
| 19:02.00 | Sincan2 | ok so sory |
| 19:02.02 | Sincan2 | sory |
| 19:02.16 | [TK]D-Fender | SargunAlso GUI's are not supported in this channel. Please refer to #freepbx |
| 19:02.17 | Sargun | it's two "R"s |
| 19:02.20 | [TK]D-Fender | #freepbx |
| 19:02.27 | Sargun | [TK]D-Fender, eh? |
| 19:02.31 | [TK]D-Fender | SargunBad aim |
| 19:02.40 | *** join/#asterisk seeker2921 (n=sumone@cpe-70-123-169-127.hot.res.rr.com) |
| 19:02.46 | [TK]D-Fender | Sincan2: Also GUI's are not supported in this channel. Please refer to #freepbx |
| 19:02.52 | [TK]D-Fender | ~freepbx |
| 19:02.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:03.13 | Sincan2 | woek thanks about this |
| 19:03.42 | Sincan2 | woke thanks |
| 19:04.27 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 19:07.09 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 19:09.59 | seeker2921 | Hello, I'm looking for assistance in configuaring a recent asterisk install. |
| 19:10.13 | [TK]D-Fender | ~ask |
| 19:10.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:10.16 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-230-234.telecom.net.ar) |
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| 19:11.07 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:13.20 | seeker2921 | I have two sip trunks setup and registered. I have the inboud and outboud routes configured and I have a softphone installted and registered to the system. Problem is when placing a call to one of the sip numbers the system kicks it back and the voice mail from the provider picks up. also when dialing a number using the softphone I get a recording saying there was a error and to try again. |
| 19:14.10 | [TK]D-Fender | seeker2921: Go look at SIP debug for the inbound call attempt and see what's going on |
| 19:15.42 | *** join/#asterisk blazon (n=blazon@unaffiliated/blazon) |
| 19:16.07 | *** join/#asterisk raden (n=jon@69-179-99-17.stat.centurytel.net) |
| 19:16.36 | raden | which version of asterisk would be considered stable ? |
| 19:17.23 | [TK]D-Fender | raden: Best choice for now is 1.6.0.latest full release |
| 19:17.41 | [TK]D-Fender | raden: 1.6.1 branch is still a little too new. |
| 19:18.15 | [TK]D-Fender | raden: I'd give it another release or two to get a better impression that its stable 9although it might be now) |
| 19:19.20 | seeker2921 | In CLI I enabled sip debug but I'm not sure where the logs would be to view them? |
| 19:21.45 | [TK]D-Fender | seeker2921: No logs, just local CLI |
| 19:22.45 | seeker2921 | I'm getting alot displaying there. heres one of it |
| 19:22.46 | seeker2921 | Scheduling destruction of SIP dialog '2b8b46ff2d76173433896d636ac5f6bb@127.0.0.1' in 32000 ms (Method: REGISTER) |
| 19:22.47 | seeker2921 | localhost*CLI> |
| 19:22.47 | seeker2921 | <--- SIP read from 216.115.20.41:5061 ---> |
| 19:22.47 | seeker2921 | SIP/2.0 200 OK |
| 19:22.47 | seeker2921 | Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK5b913109;rport |
| 19:22.48 | seeker2921 | From: <sip:15412050779@sphone.vopr.vonage.net>;tag=as7784146d |
| 19:22.50 | seeker2921 | To: <sip:15412050779@sphone.vopr.vonage.net> |
| 19:22.52 | seeker2921 | Call-ID: 62ce33ce2e36fad8757e041016eb514f@127.0.0.1 |
| 19:22.54 | seeker2921 | CSeq: 123 REGISTER |
| 19:22.56 | seeker2921 | Contact: <sip:s@192.168.0.117:5060>;expires=20 |
| 19:22.58 | seeker2921 | Content-Length: 0 |
| 19:23.01 | Orbixx | ... |
| 19:23.06 | *** kick/#asterisk [seeker2921!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 19:23.35 | *** join/#asterisk seeker2921 (n=sumone@cpe-70-123-169-127.hot.res.rr.com) |
| 19:23.42 | [TK]D-Fender | seeker2921: do NOT spam in here : |
| 19:23.43 | [TK]D-Fender | ~pb |
| 19:23.44 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 19:23.45 | [TK]D-Fender | ^^^^^^^^ |
| 19:24.07 | mercutioviz | flooding the channel is a no-no |
| 19:24.25 | jaytee | almost as big a no-no as flooding New Orleans |
| 19:24.27 | seeker2921 | Sorry, didn't realise I was flooding. |
| 19:24.44 | Orbixx | That generally happens when you paste lots of stuff into IRC. |
| 19:24.58 | [TK]D-Fender | jaytee: And I don't wanna swim! |
| 19:25.03 | mercutioviz | rule of thumb: don't paste more than 2 lines into the channel |
| 19:25.22 | seeker2921 | Will keep that in mind, Again sorry. |
| 19:26.12 | beek | Don't worry -- if you forget [TK]D-Fender will remind you... ;-) |
| 19:26.18 | mercutioviz | it's all good. |
| 19:26.44 | seeker2921 | :) |
| 19:26.49 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 19:27.41 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 19:27.46 | seeker2921 | So as for my problem, The main thing I'm seeing is "distruction of SIP dialog" |
| 19:28.55 | raden | is there a online howto on isntall and setup of asterisk , my first time |
| 19:29.20 | Qwell | ~book |
| 19:29.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:29.24 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:29.27 | [TK]D-Fender | seeker2921: Place an incoming call. If yousee nothing then they either aren't sending the call to you or you have a networking problem |
| 19:29.35 | raden | Qwell, i have that so i guess im good |
| 19:31.04 | raden | just wish it was more step by step seems like the book skips around a lil maybe it is just me |
| 19:31.41 | pepe | raden: www.das-asterisk-buch.de/2.1/ - if you don't mind reading german |
| 19:31.42 | raden | i downloaded Asterisk 1.6.0.10 do i need to download zaptel from somewhere ? |
| 19:31.46 | jaytee | raden, what distro are you running? |
| 19:31.52 | raden | opensuse 11 |
| 19:32.03 | jaytee | raden, 1.6 does not use zaptel, it uses dahdi |
| 19:32.09 | jaytee | which replaced zaptel |
| 19:32.41 | raden | jackal, thank you thats why i was asking for a howto |
| 19:32.56 | raden | so i can skip the zaptel chapter |
| 19:34.33 | jaytee | there have been issues with running * on Suse 10, not sure if they were fixed in 11. Not much in the way of how-to's for Suse. mostly Debian and CentOS (RedHat without the branding) |
| 19:35.11 | raden | i knew i should have installed debian |
| 19:35.32 | seeker2921 | [TK]D-Fender: I placed the call and it displayed alot of information regarding the call but doesnt list why it was droped. What should I be looking for in the output? |
| 19:36.35 | jaytee | raden, try googling for asterisk suse and install as keywords. there's not much for Suse on the wiki at voip-info.org but you might find another site where someone's posted a walkthrough or howto |
| 19:36.47 | raden | yeah been trying :( |
| 19:37.09 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 19:37.15 | jaytee | myself, I prefer CentOS as it seems to have attracted the largest base of followers. |
| 19:37.54 | jaytee | but there are many debian afficionados that run * and the wiki has a few good howtos for that distro also. |
| 19:39.01 | *** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com) |
| 19:39.04 | jaytee | word of advice, avoid precompiled packages for Asterisk as if they were the Black Death. |
| 19:39.22 | Orbixx | What's the reason for that? |
| 19:39.24 | beek | ... and AIDs combined. |
| 19:39.32 | Talirk81 | in 1.6 is it possible to merge 2 ConfBridge's? |
| 19:40.17 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:40.17 | jaytee | Orbixx, you talkin to me? (best NY mafia wiseguy accent) |
| 19:40.28 | jaytee | afternoon, beek |
| 19:40.48 | beek | Hi jaytee |
| 19:41.04 | Orbixx | jaytee: What if I am? |
| 19:41.08 | Orbixx | hurr, yes, I am |
| 19:41.13 | jaytee | just wondering dude |
| 19:41.22 | Orbixx | (I was playing along :>) |
| 19:41.47 | Orbixx | Yes, why should one avoid packaged Asterisk? |
| 19:41.49 | WindowsUser | precompiled is probably fine for sip only |
| 19:42.29 | jaytee | but if you were askin why packages are a bad idea then it's because with asterisk there is just too many factors to take into consideration and too much potential of missing dependencies. |
| 19:43.07 | Orbixx | I suppose it's not so bad anyway, because Asterisk is the sort of thing you reserve an entire box for, really. |
| 19:43.11 | jaytee | WindowsUser, agreed and usually only if your hardware is close to what the package builder was using to build the package on. |
| 19:43.16 | raden | linux-lms2:/usr/src/asterisk-1.6.0.10 # make clean |
| 19:43.16 | raden | make: -F.: Command not found |
| 19:43.18 | raden | ??? |
| 19:44.22 | raden | configure: error: *** termcap support not found |
| 19:44.31 | *** part/#asterisk seeker2921 (n=sumone@cpe-70-123-169-127.hot.res.rr.com) |
| 19:44.42 | jaytee | yum install libtermcaps libtermcaps-devel |
| 19:45.24 | [TK]D-Fender | Ah if only he waited a minute more.. |
| 19:45.27 | jaytee | you're just starting to get an idea about how much preparation it takes |
| 19:45.43 | jaytee | with Suse |
| 19:46.14 | howie | does anyone know if sprint still offers the unlimited cell to home calling? |
| 19:46.34 | WindowsUser | most companys offer a top 5 friends package |
| 19:46.48 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:46.48 | raden | jaytee, you for real ? |
| 19:46.52 | raden | [TK]D-Fender, ? |
| 19:47.24 | [TK]D-Fender | raden: He might be... |
| 19:47.34 | raden | lmao |
| 19:48.02 | [TK]D-Fender | raden: I'd pinch him to be sure, but that'd mean he's real, and I'm possibly gay. Can't have that... |
| 19:48.54 | raden | is a week enough to get asterisk working ? |
| 19:49.23 | [TK]D-Fender | raden: In my world, about 30 mins. |
| 19:49.31 | Alfio | raden its depend i know a guy who has two years working on it |
| 19:49.35 | [TK]D-Fender | raden: and "working" depends on specific goals |
| 19:49.41 | Alfio | :) |
| 19:49.45 | [TK]D-Fender | AlI know even more twits.. |
| 19:49.55 | raden | configure: error: *** termcap support not found |
| 19:50.03 | raden | i installed termcap package :( |
| 19:50.17 | Corydon76-dig | raden: and the -dev package |
| 19:50.27 | raden | not showing up in yast :( |
| 19:50.31 | Corydon76-dig | raden: really, ncurses-devel |
| 19:50.44 | Corydon76-dig | or libncurses5-devel |
| 19:51.01 | raden | have ncurses ill install that |
| 19:51.52 | raden | instead of novell partnering with ms they should be putting efforts to more practical things like making it work with other software |
| 19:51.58 | raden | things were so simple when i used debian |
| 19:53.14 | beek | raden -- for the time it takes to do it, why not shitcan your SuSE installation and put Debian (or CentOS5) on instead? |
| 19:53.46 | mercutioviz | is your pbx going to be on a dedicated machine? (correct answer is yes, btw) |
| 19:53.57 | Orbixx | [TK]D-Fender: It may interest you that using the same config files, my stuttering audio problem was solved by installing the latest SVN of Asterisk. |
| 19:54.00 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 19:54.05 | mercutioviz | if so, pick a distro that lends itself to a pbx |
| 19:54.07 | raden | mercutioviz, unfortunately not |
| 19:54.14 | mercutioviz | bummer |
| 19:54.52 | raden | PBX using .711 w/ 5 users / file server / sftp for remote sales reps / web server for company customer management |
| 19:55.21 | raden | i hope 3 ghz dual core w/ 4 gb ram enough |
| 19:55.38 | *** join/#asterisk kmem (n=email@cpe-66-25-166-104.austin.res.rr.com) |
| 19:56.02 | raden | oh and groupwise and e-directory |
| 19:56.03 | raden | i hate budgets |
| 19:56.12 | mercutioviz | hehe, good luck with all that |
| 19:56.13 | [TK]D-Fender | Orbixx: ok/fine/sure :) |
| 19:56.41 | Orbixx | No idea what the problem was, but that fixed it. |
| 19:57.11 | raden | I don't believe it will hold up to the load its going to experiance |
| 19:57.14 | raden | I could be wrong |
| 19:57.29 | raden | how long does make usually take > |
| 19:57.55 | raden | that didnt take long |
| 19:58.32 | raden | how would i go about uninstalling asterisk if i have to dow the road |
| 19:59.30 | raden | ok asterisk is installed |
| 20:01.40 | [TK]D-Fender | Orbixx: My ability to worry about the origin of a problem is inversely proportionate to how simple the "fix" seems to be |
| 20:02.05 | [TK]D-Fender | raden: "there is no exit strategy" - G.W.B. |
| 20:02.06 | Orbixx | Makes sense I suppose. |
| 20:02.29 | raden | lol |
| 20:03.01 | *** part/#asterisk awk_r (n=awk@nat/digium/x-4541873b2870d7bf) |
| 20:04.07 | *** join/#asterisk jcape (n=jcape@205.201.247.214) |
| 20:05.56 | raden | is it normal for /etc/asterisk to be empty ? |
| 20:06.09 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 20:06.18 | Alfio | did you make samples after install? |
| 20:06.33 | raden | no i did not |
| 20:07.33 | Alfio | that command will create samples files |
| 20:07.49 | raden | thanks |
| 20:08.38 | Alfio | you have to type it in your asterisk files folder, |
| 20:09.13 | Alfio | the one you uncompresed (if you downloaded a compresed files) |
| 20:10.01 | raden | yeah i understand |
| 20:12.50 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 20:15.17 | *** join/#asterisk trumee (i=rs4@cpc1-nmkt1-0-0-cust91.cmbg.cable.ntl.com) |
| 20:15.39 | trumee | guys, anybody from the UK? |
| 20:15.51 | trumee | <PROTECTED> |
| 20:16.08 | trumee | i have setup callback in freepbx and today tried it from a BT booth. I did get a ring back with a dial tone but the booth didnot accept any input. I was wondering if dtmf is intentionally disabled |
| 20:24.29 | jamesh1 | how do I turn off fax? |
| 20:24.34 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 20:26.07 | *** join/#asterisk ntbourey (n=ntbourey@c-75-74-236-16.hsd1.fl.comcast.net) |
| 20:26.20 | ntbourey | Hello everyone |
| 20:27.24 | jamesh1 | -- Executing [h@ext-fax:1] System("DAHDI/1-1", "/var/lib/asterisk/bin/fax-process.pl --to fax@mydomain.com --from freepbx@gmail.com --dest "2147201442" --subject "Fax from 9722989370 hostglobe9722989370" --attachment fax_9722989370.pdf --type application/pdf --file |
| 20:27.30 | jamesh1 | how do I turn this off? |
| 20:28.24 | [TK]D-Fender | jamesh1: You ask in #freepbx because it isn't supported here <- |
| 20:28.34 | jamesh1 | oh sorry wrong window |
| 20:28.34 | jamesh1 | lol |
| 20:29.02 | ntbourey | Has anyone ever run into an issues where an Perl AGI script gives you back incorrect digits when executing wait for digit? |
| 20:29.13 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:31.16 | [TK]D-Fender | checkout time, BBIAB |
| 20:31.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:35.59 | raden | bbl |
| 20:36.05 | Sargun | Would anyone be interested in my Erlang AGI/AMI handlers? |
| 20:36.20 | Qwell | checks the date |
| 20:46.32 | *** join/#asterisk lcuevas (n=lcuevas@fw1-ext-core.net-uno.net) |
| 20:47.04 | jamesh1 | is there a legend of reason calls get dropped or hungup? |
| 20:47.07 | jamesh1 | like cause 16 or cause 32 |
| 20:47.10 | jamesh1 | or cause 31 |
| 20:48.56 | lcuevas | google ISDN Disconnection Cause Codes or ICCs |
| 20:49.09 | jaytee | new vulnerability was just announced in dd-wrt firmware |
| 20:49.29 | leifmadsen | nice |
| 20:49.32 | jaytee | http://www.theregister.co.uk/2009/07/21/critical_ddwrt_router_vuln/ |
| 20:49.36 | leifmadsen | have they done a release since last December? |
| 20:49.53 | jaytee | haven't been to their site in ages. |
| 20:50.26 | jaytee | time to go home and cook tamales :-) be back later |
| 20:50.27 | Orbixx | Does waitexten accept 3 digit extensions? |
| 20:50.34 | Orbixx | Because it only seems to be catching 1 or 2 digits. |
| 20:53.08 | *** part/#asterisk CrazyTux[m] (n=Brandon@99-53-97-235.lightspeed.cyprtx.sbcglobal.net) |
| 20:55.34 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 20:59.34 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:59.48 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 21:02.06 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:07.23 | bmoraca | jamesh1: ISDN Cause Codes: http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 21:10.39 | *** join/#asterisk idc-dutch (n=idc-dutc@idc-vls.xs4all.nl) |
| 21:11.27 | telnettech | TK: do you know anything about SIPX? |
| 21:11.38 | idc-dutch | Hi, is video (h264) working on asterisk between SIP and IAX clients? |
| 21:13.53 | idc-dutch | I want to make this videosupport work: SipPhone -> Asterisk ---IAX-Trunk--->Asterisk----> SipPhone. Anyone can tell me if this can work on asterisk? |
| 21:20.41 | carrar | would work just like you show |
| 21:21.02 | carrar | and yes, it can work |
| 21:23.01 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 21:23.54 | [TK]D-Fender | telnettech: Yeah, its its ow SIP stack of services |
| 21:23.55 | Nugget | telnet is eeeeeeevil! |
| 21:24.35 | telnettech | what is the difference between it and asterisk. And is it all based on the SIP protocol |
| 21:25.01 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 21:25.03 | [TK]D-Fender | SPIX = only SIP. * = everything + kitchen sink. |
| 21:25.23 | [TK]D-Fender | telnettech: Go look at the actual project for yourself. |
| 21:25.31 | telnettech | so it is like calling IP to IP without * as a gateway |
| 21:26.13 | [TK]D-Fender | telnettech: it IS a gateway. And app server. And registrar, etc. its modular last I checked |
| 21:26.26 | carrar | which one is it |
| 21:26.32 | telnettech | ok thanks....i am going to check out the wiki |
| 21:26.50 | [TK]D-Fender | carrYES |
| 21:26.54 | carrar | heh |
| 21:27.00 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 21:27.29 | buttons840 | my asterisk server is behind a router, what ports do i need to forward so i could log into the asterisk server over the internet? |
| 21:27.48 | carrar | log in via SSH? |
| 21:28.10 | carrar | TCP:22 |
| 21:29.10 | buttons840 | 5060 is port for sip i guess |
| 21:29.10 | buttons840 | so 5060 would be one, (my softphone i login with is sip) |
| 21:29.14 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 21:29.15 | ariel_ | ~nat |
| 21:29.16 | infobot | from memory, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:30.06 | spackle | D-fender, not that you care but I got the Adtran phones to register. Adtran support was no help. I will write it up when I figure out exactly what is required. |
| 21:31.30 | carrar | perhaps UDP 5060,16348-32768 |
| 21:32.46 | ariel_ | those ports appear to be ones from Cisco, you should set them up with your rtp.conf settings as those are what asterisk normally uses. |
| 21:32.55 | [TK]D-Fender | ~sipnat |
| 21:32.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:32.58 | [TK]D-Fender | buttons840: ^^^^^^^^^^ |
| 21:33.27 | [TK]D-Fender | spackle: Was it actually complex enough to warrant writing up? |
| 21:33.41 | [TK]D-Fender | spackle: And I do care though somewhat passively. |
| 21:38.18 | Orbixx | What's the best way to play a menu and wait for input? |
| 21:39.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:39.50 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:40.42 | Orbixx | nevermind |
| 21:49.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:54.08 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 21:55.21 | *** join/#asterisk Whitor (n=Whitor@64.128.237.124) |
| 21:59.46 | *** join/#asterisk w3wt (n=sean@24-247-31-58.dhcp.bycy.mi.charter.com) |
| 22:04.06 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:05.43 | *** join/#asterisk citywok (n=chatzill@67-148-102-4.dia.static.qwest.net) |
| 22:06.05 | citywok | What is the recommended cordless phone to use as a receptionist phone? |
| 22:07.29 | bmoraca | on an SPA3102, how do I direct it to automatically dial an extension on the SIP proxy when a call comes in on the FXO port? |
| 22:16.08 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 22:16.47 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:20.16 | *** join/#asterisk Rob3Rt (i=R0b3Rt@181.45.96.58.static.exetel.com.au) |
| 22:23.26 | *** join/#asterisk sjobeck (n=Adium@ip67-95-217-18.z217-95-67.customer.algx.net) |
| 22:24.05 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:24.54 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 22:26.35 | *** part/#asterisk sjobeck (n=Adium@ip67-95-217-18.z217-95-67.customer.algx.net) |
| 22:27.44 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:27.52 | _ShrikE | Has anyone ever seen an odbc function append a \ to the query result? |
| 22:31.36 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 22:31.44 | *** join/#asterisk luca`gervasi (n=ashura@host167-170-dynamic.36-79-r.retail.telecomitalia.it) |
| 22:31.48 | luca`gervasi | Hallo! |
| 22:32.05 | Orbixx | Anybody know the correct format to include asterisk variables in exten commands? |
| 22:32.06 | _ShrikE | Actually it only appends the \ to the CLI output. It is not in the database record. |
| 22:32.12 | luca`gervasi | is there any working / free skype solution? |
| 22:32.36 | _ShrikE | Orbixx: ${VARIABLE} |
| 22:32.47 | Orbixx | exten => 4,n,Record(/temp/${CALLERID(num).wav) |
| 22:32.59 | Orbixx | Works, but .wav doesn't get included for some peculiar reason. |
| 22:33.10 | Orbixx | Then Asterisk throws |
| 22:33.11 | Orbixx | NEVERMIND |
| 22:33.15 | Orbixx | I'm an idiot. |
| 22:33.25 | _ShrikE | } |
| 22:33.28 | jong2 | ? |
| 22:36.54 | *** join/#asterisk dug (n=chatzill@ppp-71-139-42-138.dsl.snfc21.pacbell.net) |
| 22:39.59 | luca`gervasi | is there any working / free skype solution? |
| 22:40.59 | dug | I have my dahdi-channels.conf set correctly and added the include for dahdi-channels.conf in my chan_dahdi.conf but I still dont see it ring? I have also check dahdi_monitor and see it as ringing |
| 22:41.14 | timeshell_atwork | luca`gervasi http://www.chanskype.com |
| 22:41.29 | *** part/#asterisk CryWolf (n=freedomb@mn01.freedombi.com) |
| 22:41.38 | timeshell_atwork | Only works with 1.4 thogh |
| 22:41.42 | timeshell_atwork | s/thogh/though |
| 22:42.16 | FreakGuard | luca`gervasi: isn't 100% free (as in speech) |
| 22:42.19 | luca`gervasi | i'm using 1.6 trunk |
| 22:42.30 | luca`gervasi | i already get there |
| 22:42.39 | luca`gervasi | 19$ is not too much |
| 22:42.54 | luca`gervasi | but there is no documentation at all |
| 22:43.01 | luca`gervasi | it's like a blind jump |
| 22:45.24 | luca`gervasi | it needs vnc... |
| 22:45.29 | luca`gervasi | not quite a solution... |
| 22:45.51 | luca`gervasi | seems like a workaround... |
| 23:02.11 | *** join/#asterisk rue_mohr (n=Dennis@24.207.122.10) |
| 23:02.30 | rue_mohr | hey, anyone know whats up with digium support? it just rings and rings |
| 23:03.58 | rue_mohr | hey anuyone know whats up with #asterisk? its like the channel is dead |
| 23:05.03 | telnettech | im here |
| 23:05.10 | rue_mohr | ah |
| 23:05.19 | rue_mohr | you know why digium sorrort isn't asnwering? |
| 23:05.25 | telnettech | no clue |
| 23:05.34 | telnettech | it is 6pm there |
| 23:05.47 | telnettech | do they do 24/7 support> |
| 23:05.49 | telnettech | ? |
| 23:06.00 | rue_mohr | I thought so, international and all |
| 23:06.05 | telnettech | huh |
| 23:06.44 | rue_mohr | well, how about if I say this, anyone know why nomatter what I do I cant get enough volume out of the phones? |
| 23:09.14 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 23:09.22 | drmessano | Maybe your number drops directly in the janitorial queue and they're out unplugging a shitter |
| 23:09.54 | rue_mohr | sounds like I good reason to insist on a refund |
| 23:10.29 | telnettech | Rue: check the TX and RX of the phone if you can |
| 23:11.02 | rue_mohr | its an ip set, how am I supposed to do that |
| 23:11.09 | rue_mohr | I have the recieve set properly as per 1mw from the CO |
| 23:11.12 | drmessano | Your sense of entitlement amuses me |
| 23:11.32 | rue_mohr | I'm REALLY frustrated |
| 23:12.13 | rue_mohr | I'd be less frustrated if I'd not followed advise to go with polycom sets instead of using aastra's all thru |
| 23:14.07 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:14.41 | drmessano | Maybe you should turn this whole project over to someone with some asterisk knowledge.. With the time you put into this X standard hourly wages, you could have flown in most of Digium to fix it onsite. |
| 23:15.20 | rue_mohr | you in north america? |
| 23:15.35 | drmessano | I am |
| 23:16.00 | rue_mohr | you think you could make it work? |
| 23:16.53 | *** join/#asterisk tfrew (n=tfrew@c-68-57-86-168.hsd1.va.comcast.net) |
| 23:17.59 | drmessano | Without a doubt.. All your problems as you have described them, endlessly, should have been resolved long ago when the time spent on each individual issue became inversely proportional to a resolution involving replace or repair of key components |
| 23:19.22 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 23:20.10 | drmessano | A $400 card isn't worth 6 months of anyones time.. if you think the card is bad, buy a new damn card.. If you think theres a problem with the lines, MAKE the telco come out and prove to YOU the line is ok. |
| 23:20.46 | rue_mohr | how am I gonna tell the boss the brand new $1400 card is no good? |
| 23:21.00 | drmessano | If its brand new, RMA it.. |
| 23:21.08 | rue_mohr | I checked the lines, their 0db within .3db |
| 23:21.22 | rue_mohr | ok, so your saying I shoudl RMA the card |
| 23:21.48 | bmoraca | man, working with that fxo port on the SPA3102 is freaking obtuse as heck |
| 23:21.54 | rue_mohr | is that what you would do if you were here? |
| 23:22.11 | *** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30) |
| 23:22.14 | drmessano | You're also putting way too much time in overly trivial line measurements when the other 99.9999% set a card up, screw with the levels a little til "it works" and move on.. If theres that much of a problem, microanalyzing it wont help |
| 23:22.29 | drmessano | If you suspect the card is bad, RMA IT |
| 23:22.45 | rue_mohr | every 3 days the secretary comes to me and says she cant hear the people on the other end |
| 23:23.02 | rue_mohr | I'v dialed the polycoms gain up to something like 12db |
| 23:23.28 | rue_mohr | I cant dial the digium card past the 2db that made it recieve the 1mw properly or I get echo city |
| 23:23.45 | *** join/#asterisk howie (n=howie@71-95-220-206.dhcp.mtpk.ca.charter.com) |
| 23:23.49 | drmessano | So every 3 days the POLYCOMS... ALL decide to drop by x db to the point you need to jack them up? Wouldnt you suspect something a little easier to believe? |
| 23:24.15 | rue_mohr | I suspect the card is doing exactly what its designed to do, which is not working |
| 23:24.41 | rue_mohr | no, the gain on the poly comes stays the same |
| 23:25.00 | rue_mohr | it takes 3 days for the secretary to get frustrated enough to phone me |
| 23:25.02 | Sargun_Screen | What's a good softphone for Linux? What's a low-bandwidth, crappy quality codec? |
| 23:25.11 | drmessano | Have the telco test the line and if the line checks fine, call support.. Let them have their way with it.. and if you cant prove the card is a problem and cant prove its not, RMA |
| 23:25.57 | rue_mohr | i already said twice that the 1mw on the lines is within 0.3db |
| 23:26.42 | mercutioviz | Sargun_Screen: g729 is low bandwidth and crappy. :) |
| 23:26.55 | rue_mohr | you also have to pay for it |
| 23:26.59 | mercutioviz | true |
| 23:27.10 | drmessano | You've been complaining about this line for 6 months.. or is it longer? |
| 23:27.12 | mercutioviz | if you're not unscrupulous, of course. |
| 23:27.15 | rue_mohr | longer |
| 23:27.33 | Sargun_Screen | mercutioviz: free. |
| 23:27.38 | mercutioviz | which card is it? i was afk and I don't want to scroll back |
| 23:27.47 | rue_mohr | TDM800 with an echo can |
| 23:27.50 | drmessano | Apparently something is broken here.. The line, the card, or you.. Check/replace all 3.. |
| 23:28.01 | mercutioviz | ugh, analog |
| 23:28.07 | rue_mohr | i already said 3 times that the 1mw on the lines is within 0.3db |
| 23:28.35 | rue_mohr | its not possable for me to check the levels on the sip connection |
| 23:28.42 | mercutioviz | do all ports on the card exhibit the same behavior |
| 23:28.45 | mercutioviz | ? |
| 23:28.54 | drmessano | I dont give a shit about your measurements.. Ive been hearing about your measurements for months.. youve not shown me any real reference except what is gained from questionable cards and questionable software along with a questionable line |
| 23:29.27 | Sargun_Screen | o.O |
| 23:29.37 | rue_mohr | "yes" but when I use an analog line with that the cordless is on and I have local bridging turned on that phone is fine |
| 23:29.52 | drmessano | So RMA the card.. what are you waiting for? |
| 23:30.11 | rue_mohr | would you like a picture of the meter showing you 0.25db from the COs milliwatt? |
| 23:30.27 | Orbixx | Faulty cards can play havoc with things, ESPECIALLY if they're analogous. |
| 23:30.30 | drmessano | Which meter would this be? |
| 23:30.46 | Orbixx | (the "things" being analogues, not the card - lol) |
| 23:31.10 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-166-138.dsl.stlsmo.sbcglobal.net) |
| 23:31.17 | rue_mohr | http://eds.dyndns.org/~ircjunk/images/p1010034.jpg |
| 23:31.41 | rue_mohr | same meter I used to tell that the milliwatt() is out by -11db compared to the milliwatt(o) |
| 23:31.54 | rue_mohr | did you know the new milliwatt source is out by -11db!? |
| 23:32.18 | rue_mohr | I have to leave work now, talk to you all later |
| 23:32.37 | SaiSoma | speaking of weird things with digium cards . . . .. |
| 23:32.43 | rue_mohr | (at the bottom of the meter screen is the signal strength in dbm) |
| 23:34.31 | drmessano | .25db sure looks a lot like 250 millivolt to me |
| 23:35.25 | SaiSoma | I have a TDM800P and a TE122 in the same box. Randomly, they stop functioning. Asterisk shows the TE122 is working and the TDM800P having all channels red, but the Nortel CS1000M shows the connection down on the PRI and all analog extensions hooked to the TDM800P idle. I suspected the CS1000 for the last few weeks, but I'm starting to suspect the bus on the PC. Sound like I'm on the... |
| 23:35.26 | SaiSoma | ...right track? |
| 23:35.47 | SaiSoma | The only thing that fixes the issue is a reboot of the PC. |
| 23:36.03 | LemensTS | does it take anymore resources to play a wav file than it does a gsm? |
| 23:36.05 | bmoraca | 1.21 gigawats? |
| 23:36.13 | drmessano | Sounds to me like rue_mohr needs to learn how to read a fucking DMM |
| 23:36.23 | LemensTS | *gonna be playing 500 concurrently is why im askiung |
| 23:36.43 | citywok | LemensTS: depends what codec the channel is in |
| 23:37.14 | bmoraca | LemensTS: if you're playing a sound file that differs in codec from your channel, it's all bad news. |
| 23:37.26 | citywok | LemensTS: if asterisk has to do any transcoding, it's going to take CPU. therefore if the channel is a SIP GSM stream, wav will be more. if it's a g729 stream, it has to transcode either way, and at that point the wav may be faster than the gsm version |
| 23:37.54 | citywok | the goal is to encode your audio files in the codec that gets used most commonly |
| 23:39.44 | LemensTS | citywok: ok im passing it off to a sip provider so i guess gsm would be what i want |
| 23:39.58 | bmoraca | LemensTS: again, that depends. |
| 23:39.59 | citywok | LemensTS: if GSM is the codec your SIP provider uses, then yes |
| 23:40.16 | citywok | LemensTS: if you are using G711, G729, ilbc, or something else, you will have a headache |
| 23:40.49 | bmoraca | and why are you going to be playing 500 concurrently to an ITSP? |
| 23:41.07 | citywok | yea, that's a lot of concurrent calls |
| 23:41.32 | citywok | you better make sure the provider doesnt shut you down for slamming their systems with a huge burst of traffic |
| 23:41.38 | bmoraca | that's a lot of bandwidth |
| 23:42.10 | citywok | 6K/s * 500... 30mbit give or take? |
| 23:42.18 | LemensTS | citywok: g711 and g729 is what i can use. just looked it up |
| 23:42.35 | bmoraca | i'd expect g711 to be about double that |
| 23:42.39 | citywok | then you need licensing for g729 @ $10/channel, or use g711 which is about 10 calls per 1.5mbit of bandwidth |
| 23:42.47 | bmoraca | or $5k capital to do g729 :P |
| 23:42.59 | citywok | 50 * 1.5 is 75mbit of bandwidth for g711 |
| 23:43.10 | LemensTS | so with those 2 codecs, it doesnt matter wav or gsm? |
| 23:43.18 | Qwell | LemensTS: it matters a great deal |
| 23:43.19 | bmoraca | with those two, use wav |
| 23:43.21 | citywok | i've found 10-12 calls is the max a t1 can do |
| 23:43.34 | Qwell | use *neither*. if you doing anywhere near that many calls, you want zero transcoding. |
| 23:43.48 | citywok | yea, you need it encoded in the codec the channel is using |
| 23:43.55 | citywok | that much transcoding would crush a major system |
| 23:44.05 | LemensTS | so i can record in g711 codec? |
| 23:44.23 | LemensTS | using the Record cmd |
| 23:44.24 | bmoraca | yes |
| 23:44.33 | bmoraca | or at least you can transcode to it after you record |
| 23:45.21 | drmessano | [TK]D-Fender: You around? |
| 23:45.26 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:46.18 | LemensTS | ahh ok thats why i was confused, i didnt know I could do that. transcoding it after i record, you mean if i recorded it in wav but wanted it in g729 i could do it after i recorded it? |
| 23:46.32 | bmoraca | yes |
| 23:47.00 | drmessano | Qwell: You see this pic from Rue_mohr? |
| 23:47.14 | LemensTS | bmoraca: is there a dial plan cmd to do that? |
| 23:47.20 | LemensTS | or do i have to use sox |
| 23:47.30 | bmoraca | LemensTS: no, but you can issue system commands from the dialplan |
| 23:47.38 | bmoraca | actually |
| 23:47.40 | bmoraca | i lied |
| 23:47.47 | bmoraca | there is an asterisk command for recoding files |
| 23:47.51 | bmoraca | i don't remember it though |
| 23:47.54 | LemensTS | Record |
| 23:48.00 | LemensTS | oh recoding |
| 23:48.02 | LemensTS | ok cool ill search |
| 23:48.20 | bmoraca | drmessano: definitely measuring volts there |
| 23:48.32 | mercutioviz | sox is your best buddy for resampling/recoding sound files |
| 23:49.26 | LemensTS | Alright im gonna record in g711 and recode to wav file (they need to hear it in a web browser so i need it in both formats) |
| 23:49.38 | drmessano | bmoraca: Hes a dumbass.. The TOP scale he claims proves that he has 0.25db of signal and Dahdi's reference is off by 11db is disputed by the ACTUAL DBM scale on HIS METER showing -10.17 which is EXACTLY what Dahdi is telling him!!! |
| 23:49.42 | Qwell | bmoraca: more importantly - volts *AC* |
| 23:50.17 | Qwell | I tried pointing that flaw out before :p |
| 23:50.32 | bmoraca | lol |
| 23:50.39 | drmessano | He says DAHDI is off by -11dbm because he knows he has .025 "db" on the line.. and his OWN DMM is showing him -10.17 |
| 23:50.44 | drmessano | WTFFFFF |
| 23:51.15 | bmoraca | well, he's in the fucking back country...he wonders why there's going to be issues on the line |
| 23:51.42 | drmessano | He needs to replace either the line, the card, or himself.. I vote for himself |
| 23:51.53 | bmoraca | i just installed a system in Fairbanks, Alaska with analog lines...every time the wind blows, they get static or echo...and they call me to bitch about it...bugs me |
| 23:51.54 | citywok | LemensTS: how much bandwidth do you have to your provider? |
| 23:52.14 | LemensTS | If i encoded in g729, and passed it off to an itsp in g729 via sip, would i need any licenses besides when initially doing the recordings? |
| 23:52.34 | drmessano | bmoraca: Ironically.. the one who had a bug showing an improper reading skewed by 11db was rue_mohr himself |
| 23:52.41 | bmoraca | LemensTS: no, Asterisk doesn't require licenses for passthrough of g729 |
| 23:54.47 | LemensTS | citywok: im told 100mb's |
| 23:54.52 | LemensTS | 100mb/s |
| 23:55.05 | citywok | and what is the hardware of the box doign the 500 calls? |
| 23:55.09 | bmoraca | you have a 100mb/s internet connection? do want |
| 23:55.22 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 23:56.12 | LemensTS | bmoraca: lol yea thats what i thought |
| 23:56.34 | LemensTS | so im not sure what it actually is. its server beach |
